From: jk7744.park Date: Sat, 24 Oct 2015 07:50:02 +0000 (+0900) Subject: tizen 2.4 release X-Git-Tag: accepted/tizen/2.4/mobile/20151029.032644^0 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=4cdb77e7e201326ff3f4f947ac991da3096953de;p=framework%2Fmultimedia%2Fgst-rtsp-server.git tizen 2.4 release --- diff --git a/AUTHORS b/AUTHORS new file mode 100644 index 0000000..6b672b9 --- /dev/null +++ b/AUTHORS @@ -0,0 +1,2 @@ +Wim Taymans +Hyunjun Ko diff --git a/COPYING b/COPYING new file mode 100644 index 0000000..c87cfe8 --- /dev/null +++ b/COPYING @@ -0,0 +1,481 @@ + GNU LIBRARY GENERAL PUBLIC LICENSE + Version 2, June 1991 + + Copyright (C) 1991 Free Software Foundation, Inc. + 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA + Everyone is permitted to copy and distribute verbatim copies + of this license document, but changing it is not allowed. + +[This is the first released version of the library GPL. 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Here is a sample; alter the names: + + Yoyodyne, Inc., hereby disclaims all copyright interest in the + library `Frob' (a library for tweaking knobs) written by James Random Hacker. + + , 1 April 1990 + Ty Coon, President of Vice + +That's all there is to it! diff --git a/ChangeLog b/ChangeLog new file mode 100644 index 0000000..c8c6360 --- /dev/null +++ b/ChangeLog @@ -0,0 +1,7726 @@ +=== release 1.4.5 === + +2014-12-18 Sebastian Dröge + + * configure.ac: + releasing 1.4.5 + +2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: unref srtp decoder when leaving bin + https://bugzilla.gnome.org/show_bug.cgi?id=739481 + +=== release 1.4.4 === + +2014-11-06 13:18:04 +0100 Sebastian Dröge + + * ChangeLog: + * NEWS: + * RELEASE: + * configure.ac: + * gst-rtsp-server.doap: + Release 1.4.4 + +2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: mikey memory leaks + https://bugzilla.gnome.org/show_bug.cgi?id=739383 + +=== release 1.4.3 === + +2014-09-24 12:51:21 +0300 Sebastian Dröge + + * ChangeLog: + * NEWS: + * RELEASE: + * configure.ac: + * gst-rtsp-server.doap: + Release 1.4.3 + +=== release 1.4.2 === + +2014-09-19 15:13:37 +0300 Sebastian Dröge + + * ChangeLog: + * NEWS: + * RELEASE: + * configure.ac: + * gst-rtsp-server.doap: + Release 1.4.2 + +2014-09-08 09:26:23 +0200 Srimanta Panda + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + rtsp-media: Make sure that sequence numbers are monotonic after pause + The sequence number is not monotonic for RTP packets after pause. The + reason is basepayloader generates a randon sequence number when the + pipeline goes from ready to pause. With this fix generation of sequence + number will be monotonic when going from pause to play request. + https://bugzilla.gnome.org/show_bug.cgi?id=736017 + +2014-08-28 13:35:15 +0200 Göran Jönsson + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: Protect saved clients watch with a mutex + Fixes a crash when close() is called while merging clients + in handle_tunnel(). In that case close() would destroy the + watch while it is still being used in handle_tunnel(). + https://bugzilla.gnome.org/show_bug.cgi?id=735570 + +=== release 1.4.1 === + +2014-08-27 15:05:07 +0300 Sebastian Dröge + + * ChangeLog: + * NEWS: + * RELEASE: + * configure.ac: + * gst-rtsp-server.doap: + Release 1.4.1 + +2014-08-05 16:12:19 +0200 Sebastian Dröge + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + rtsp-media: Query position and stop time only on the RTP parts of the pipeline + The RTCP parts, in specific the RTCP udpsinks, are not flushed when + seeking and will always continue counting the time. This leads to + the NPT after a backwards seek to be something completely different + to the actual seek position. + https://bugzilla.gnome.org/show_bug.cgi?id=732644 + +2014-07-02 22:45:07 +0530 Arun Raghavan + + * gst/rtsp-server/rtsp-media.c: + signals: Fix copy-pasto in target-state signal offset + +=== release 1.4.0 === + +2014-07-19 17:56:31 +0200 Sebastian Dröge + + * ChangeLog: + * NEWS: + * RELEASE: + * configure.ac: + * gst-rtsp-server.doap: + Release 1.4.0 + +2014-07-16 20:39:42 +0900 Hyunjun Ko + + * gst/rtsp-server/rtsp-media.h: + media: correct misspelled words in description + https://bugzilla.gnome.org/show_bug.cgi?id=733244 + +=== release 1.3.91 === + +2014-07-11 12:19:08 +0200 Sebastian Dröge + + * ChangeLog: + * NEWS: + * RELEASE: + * configure.ac: + * gst-rtsp-server.doap: + Release 1.3.91 + +2014-07-10 17:37:45 +0200 Wim Taymans + + * docs/libs/gst-rtsp-server-sections.txt: + docs: update docs + +2014-07-10 17:10:06 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + server: implement client REMOVE filter + +2014-07-10 17:05:13 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: expose _close() method + Expose a previously internal close method to close the client + connection. + +2014-07-10 12:20:15 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-session-pool.c: + session-pool: signal session-removed outside of the lock + Release the lock before emiting the session-removed signal. + +2014-07-10 11:32:20 +0200 Wim Taymans + + * 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: + filter: Release lock in filter functions + Release the object lock before calling the filter functions. We need to + keep a cookie to detect when the list changed during the filter + callback. We also keep a hashtable to make sure we only call the filter + function once for each object in case of concurrent modification. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732950 + +2014-07-09 15:16:08 +0200 Ognyan Tonchev + + * gst/rtsp-server/rtsp-client.c: + client: check if watch is set in handle_teardown() + The unit tests run without a watch + +2014-07-09 14:19:10 +0200 Ognyan Tonchev + + * tests/check/gst/client.c: + client tests: send teardown to cleanup session + +2014-07-09 14:17:46 +0200 Ognyan Tonchev + + * tests/check/gst/rtspserver.c: + server tests: send teardown to cleanup session + +2014-07-09 15:01:31 +0200 Ognyan Tonchev + + * gst/rtsp-server/rtsp-client.c: + client: keep ref to client for the session removed handler + This extra ref will be dropped when all client sessions have been + removed. A session is removed when a client sends teardown, closes its + endpoint of the TCP connection or the sessions expires. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226 + +2014-07-08 12:36:12 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-session.c: + * tests/check/gst/client.c: + client: manage media in session as a last step + Once we manage a media in a session, we can't unmanage it anymore + without destroying it. Therefore, first check everything before we + manage the media, otherwise if something is wrong we have no way to + unmanage the media. + If we created a new session and something went wrong, remove the session + again. Fixes a leak in the unit test. + +2014-07-03 19:52:42 +0100 Tim-Philipp Müller + + * examples/test-mp4.c: + * examples/test-ogg.c: + examples: print 'stream ready at url' for mp4 and ogg example + +2014-07-02 16:04:53 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-sdp.c: + rtsp: fix for MIKEY api change + +2014-07-01 16:12:13 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: free watch context only once + The watch context is freed when the source is destroyed. Avoids + a CRITICAL when we try to unref the context twice. + +2014-07-01 15:02:15 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: fix build + +2014-07-01 14:41:14 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: protect sessions with lock + Protect the list of sessions with the lock. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=732226 + +2014-07-01 12:13:47 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + Client: keep a ref to the session + Don't just keep a weak ref to the session objects but use a hard ref. We + will be notified when a session is removed from the pool (expired) with + the new session-removed signal. + Don't automatically close the RTSP connection when all the sessions of + a client are removed, a client can continue to operate and it can create + a new session if it wants. If you want to remove the client from the + server, you have to use gst_rtsp_server_client_filter() now. + Based on patch from Ognyan Tonchev + See https://bugzilla.gnome.org/show_bug.cgi?id=732226 + +2014-06-30 15:14:34 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session-pool.h: + session-pool: add session-removed signal + Add a signal to be notified when a session is removed from the pool. + +2014-06-30 00:37:59 -0700 Evan Nemerson + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/rtsp-server.h: + Make rtsp-server.h a single-include header, use it for G-I + https://bugzilla.gnome.org/show_bug.cgi?id=732411 + +=== release 1.3.90 === + +2014-06-28 11:48:29 +0200 Sebastian Dröge + + * ChangeLog: + * NEWS: + * RELEASE: + * configure.ac: + * gst-rtsp-server.doap: + Release 1.3.90 + +2014-06-27 16:54:22 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-stream.c: + stream: crypto can be NULL + +2014-06-11 16:42:08 -0700 Evan Nemerson + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-mount-points.c: + introspection: add missing allow-none annotations + https://bugzilla.gnome.org/show_bug.cgi?id=730952 + +2014-06-11 16:38:36 -0700 Evan Nemerson + + * gst/rtsp-server/rtsp-address-pool.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-session-media.c: + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-stream-transport.c: + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-token.c: + introspection: add (nullable) annotations to return values + https://bugzilla.gnome.org/show_bug.cgi?id=730952 + +2014-06-24 09:48:45 +0200 Evan Nemerson + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-stream.c: + gi: improve annotations + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730953 + +2014-06-24 09:43:44 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-server.c: + signals: use generic marshal function + Use the generic C marshal function. + Use more explicit type instead of G_TYPE_POINTER + +2014-06-24 09:42:47 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-context.h: + context: add type macro + +2014-06-24 09:34:50 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-sdp.h: + sdp: hide key length defines + They don't have a namespace. + +2014-06-22 19:37:31 +0200 Sebastian Dröge + + * configure.ac: + Back to development + +=== release 1.3.3 === + +2014-06-22 19:36:14 +0200 Sebastian Dröge + + * ChangeLog: + * NEWS: + * RELEASE: + * configure.ac: + * gst-rtsp-server.doap: + Release 1.3.3 + +2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-sdp.h: + mikey: add different key length parameters + Add encryption and authentication key length parameters to MIKEY. For + the encoders, the key lengths are obtained from the cipher and auth + algorithms set in the caps. For the decoders, they are obtained while + parsing the key management from the client. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730472 + +2014-03-16 17:29:48 +0100 Ognyan Tonchev + + * tests/check/gst/stream.c: + stream tests: Make sure we get right multicast address from stream + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731577 + +2014-06-12 13:49:17 +0200 Ognyan Tonchev + + * gst/rtsp-server/rtsp-client.c: + client: ref the context until rtsp watch is alive + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=731569 + +2014-06-12 13:48:44 +0200 Ognyan Tonchev + + * gst/rtsp-server/rtsp-client.c: + client: Destroy the rtsp watch after connection close + +2014-06-13 16:46:06 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: fix confusing comment + +2014-05-27 12:36:52 +0200 Göran Jönsson + + * gst/rtsp-server/rtsp-session.c: + rtsp-session: Timeout in header. + Adding the possbilty to always have timout in header. + This is configurabe with setting "timeout-always-visible". + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728264 + +2014-05-21 13:23:40 +0200 Sebastian Dröge + + * configure.ac: + Back to development + +=== release 1.3.2 === + +2014-05-21 13:06:36 +0200 Sebastian Dröge + + * ChangeLog: + * NEWS: + * RELEASE: + * common: + * configure.ac: + * gst-rtsp-server.doap: + Release 1.3.2 + +2014-05-21 10:54:05 +0200 Sebastian Dröge + + * common: + Automatic update of common submodule + From 211fa5f to 1f5d3c3 + +2014-05-20 15:57:30 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: store TCP ports in transport + Store the TCP ports in the transport when we are doing RTSP over TCP. + This way, we can easily get to the ports from the transport. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=729776 + +2014-05-15 18:15:04 -0700 Aleix Conchillo Flaqué + + * gst/rtsp-server/rtsp-stream.c: + stream: add signals for new RTP/RTCP encoders + New signals to allow the user to configure the dynamically created + encoders. + https://bugzilla.gnome.org/show_bug.cgi?id=730228 + +2014-05-14 09:31:31 +0200 Ognyan Tonchev + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: Make suspend()/unsuspend() virtual + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109 + +2014-05-09 17:25:07 -0700 Aleix Conchillo Flaqué + + * gst/rtsp-server/rtsp-client.c: + client: fix send-message signal marshaller + Use generic marshalling for the send-message signal. It has + two POINTER arguments, not just one. + https://bugzilla.gnome.org/show_bug.cgi?id=729900 + +2014-05-09 15:08:48 +0200 Wim Taymans + + * tests/check/gst/media.c: + tests: add and remove pads only once + In this test we simulate a dynamic pad by watching the caps event. + Because of renegotiation in the base payloader now, this caps is sent + multiple times but we can only deal with 1 invocation, use a variable to + only 'add and remove' the pad once. + +2014-05-02 20:06:29 +0100 Tim-Philipp Müller + + * tests/check/gst/rtspserver.c: + tests: add unit test for correct handling of Require headers + https://bugzilla.gnome.org/show_bug.cgi?id=729426 + +2014-05-02 19:59:23 +0100 Tim-Philipp Müller + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: handle Require headers and respond with OPTION_NOT_SUPPORTED + Servers must handle Require headers and must report a failure + if they don't handle any of the Required options, see RFC 2326, + section 12.32: https://tools.ietf.org/html/rfc2326#page-54 + https://bugzilla.gnome.org/show_bug.cgi?id=729426 + +2014-05-03 20:48:43 +0200 Sebastian Dröge + + * configure.ac: + Back to development + +=== release 1.3.1 === + +2014-05-03 18:40:24 +0200 Sebastian Dröge + + * ChangeLog: + * NEWS: + * RELEASE: + * configure.ac: + * gst-rtsp-server.doap: + Release 1.3.1 + +2014-05-03 10:18:00 +0200 Sebastian Dröge + + * common: + Automatic update of common submodule + From bcb1518 to 211fa5f + +2014-05-02 19:58:15 +0100 Tim-Philipp Müller + + * .gitignore: + Update .gitignore + +2014-05-02 19:57:23 +0100 Tim-Philipp Müller + + * tests/check/gst/sessionmedia.c: + tests: fix memory leak in sessionmedia unit test + +2014-05-01 06:17:06 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: emit a signal before sending a message + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728970 + +2014-05-01 06:07:08 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: pass context to send_message + Pass the current context to send_message, we will need it later. + +2014-05-01 05:29:54 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: fix typo in comment + +2014-04-14 15:17:14 +0200 Ognyan Tonchev + + * gst/rtsp-server/rtsp-media.c: + media: Do not stop thread twice if default_prepare() fails + +2014-04-15 16:51:17 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: set the watch to flushing before going to NULL + First set the watch to flushing so that we unblock any current and + future attempt to send data on the watch, Then set the pipeline to + NULL. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728153 + +2014-04-11 23:52:49 +0200 Linus Svensson + + * gst/rtsp-server/rtsp-session-pool.c: + * tests/check/gst/sessionpool.c: + rtsp-session-pool: Fixes annotation + Fixes annotation for gst_rtsp_session_pool_create() and memory leaks + in the sessionpool test. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728060 + +2014-04-09 16:44:21 +0200 Ognyan Tonchev + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: make media_prepare virtual + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029 + +2014-04-12 05:57:00 +0200 Ognyan Tonchev + + * gst/rtsp-server/rtsp-media.c: + * tests/check/gst/media.c: + media: stop the thread in more error cases + +2014-04-12 05:53:15 +0200 Ognyan Tonchev + + * gst/rtsp-server/rtsp-media.c: + * tests/check/gst/media.c: + media: allow NULL as the thread + Use the default context whan passing a NULL thread. + +2014-04-10 16:39:11 +0100 Vincent Penquerc'h + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: indent cleanup + Coverity was moaning about unreachable code, and I think it was just + confused by { being before the label. We'll see if it pops up again. + Coverity 1197705 + +2014-04-01 13:04:21 +0200 Göran Jönsson + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + client: Add drop-backlog property + When we have too many messages queued for a client (currently hardcoded + to 100) we overflow and drop the messages. Add a drop-backlog property + to control this behaviour. Setting this property to FALSE will retry + to send the messages to the client by waiting for more room in the + backlog. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898 + +2014-04-03 12:19:51 +0200 Ognyan Tonchev + + * gst/rtsp-server/rtsp-client.c: + client: support for POST before GET when setting up a tunnel + +2014-04-02 12:03:32 +0200 Ognyan Tonchev + + * gst/rtsp-server/rtsp-client.c: + client: remove watch of the second client after http tunnel setup + The second client will be freed after the HTTP tunnel has been set up. + Make sure it's RTSP watch is never dispatched again. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727488 + +2014-03-31 11:00:11 +0200 Ognyan Tonchev + + * gst/rtsp-server/rtsp-media.c: + * tests/check/gst/media.c: + media: Make media_prepare() fail if port allocation fails + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727376 + +2014-04-01 16:55:13 +0200 Linus Svensson + + * tests/check/gst/media.c: + media test: cleanup the thread pool in tests + +2014-04-01 13:16:26 +0200 Linus Svensson + + * gst/rtsp-server/rtsp-media.c: + * tests/check/gst/media.c: + rtsp-media: Unblock blocked streams in unprepare + The streams will be blocked when a live media is prepared. + The streams should be unblocked in gst_rtsp_media_unprepare. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727231 + +2014-04-08 14:49:41 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: release the state lock when going to NULL + Set our state to UNPREPARING and release the state-lock before + setting the pipeline to the NULL state. This way, any pad-added + callback will be able to take the state-lock and check that we are now + unpreparing instead of deadlocking. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=727102 + +2014-04-08 12:08:17 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: protect status with lock + Make sure we only update the status with the lock. + +2014-04-04 17:39:36 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-sdp.c: + rtsp: update for MIKEY API changes + +2014-04-03 12:52:51 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: parse the mikey response from the client + Parse the mikey response from the client and update the policy for + each SSRC. + +2014-04-02 12:36:16 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: add method to set crypto info + Make a method to configure the crypto information of a stream. + Set udpsrc in READY instead of PAUSED so that we can configure caps + later. + +2014-04-03 12:57:13 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: cleanup error paths + +2014-04-02 12:27:24 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: fix docs + +2014-03-25 12:42:39 +0100 Wim Taymans + + * examples/test-video.c: + test: enable SRTP only on RTSPS + We only want to enable SRTP when doing rtsp over TLS so that we can + exchange the keys in a secure way. + +2014-03-25 12:41:33 +0100 Wim Taymans + + * examples/test-video.c: + test: print an error on failure + +2014-03-13 17:35:21 +0100 Wim Taymans + + * configure.ac: + * examples/test-video.c: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-stream.c: + * tests/check/Makefile.am: + stream: add SRTP support + Install srtp encoder and decoder elements in rtpbin + Add MIKEY in SDP + +2014-03-16 19:45:26 +0100 Sebastian Rasmussen + + * tests/check/Makefile.am: + * tests/check/gst/sessionpool.c: + tests: Add unit tests for sessionpool + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726470 + +2014-03-22 13:24:27 +0100 Sebastian Rasmussen + + * tests/check/gst/threadpool.c: + tests: Improve code coverage of rtsp-threadpool tests + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726873 + +2014-03-23 21:26:00 +0100 Sebastian Rasmussen + + * tests/check/gst/sessionmedia.c: + tests: Improve code coverage for rtsp-session-media + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726940 + +2014-03-23 21:24:48 +0100 Sebastian Rasmussen + + gobject-introspection: Add annotations to support language bindings + In addition a few cosmetic changes: + * Adjust the order of arguments + * Fix typo: occured -> occurred + * Fix indentation after Return:-clauses + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941 + +2014-03-14 19:03:24 +0100 Sebastian Rasmussen + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: Don't mix IPv4 and IPv6 addresses + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726362 + +2014-03-13 14:27:15 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-stream.c: + stream: take caps after the session manager + Take the caps for the SDP after they leave the rtpbin so that we can + also get the properties added by rtpbin elements. + +2014-03-13 14:20:17 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-stream.c: + stream: release lock while pushing out packets + Keep a cache of the transports and use this to iterate the transport + while pushing packets. This allows us to release the lock early. + See https://bugzilla.gnome.org/show_bug.cgi?id=725898 + +2014-03-06 13:52:02 +0100 David Svensson Fors + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + rtsp-client: vmethod for modifying tunnel GET response + Add a vmethod tunnel_http_response where the response to the HTTP GET + for tunneled connections can be modified. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725879 + +2014-03-03 16:56:53 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-sdp.c: + sdp: make 1 media line per profile + If we have multiple profiles (AVP or AVPF) for a stream, make one m= + line in the SDP for each profile. The client is then supposed to pick + one of the profiles in the SETUP request. Because the m= lines have the + same pt, the client also knows that only 1 option is possible. + +2014-03-03 16:55:48 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + factory: add profile property and pass to media and streams + +2014-03-03 15:12:55 +0100 Wim Taymans + + * examples/test-multicast.c: + * gst/rtsp-server/rtsp-sdp.c: + sdp: pass multicast connection for multicast-only stream + Pass the multicast address of the stream in the connection info in the + SDP so that clients try a multicast connection first. + Only allow multicast connections in the test-multicast example. Also + increase the TTL a little. + +2014-03-02 05:12:01 +0100 Sebastian Rasmussen + + * .gitignore: + .gitignore: Ignore gcov intermediate files + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725484 + +2014-03-03 12:17:48 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-stream.c: + stream: release some locks in error cases + +2014-03-02 05:12:10 +0100 Sebastian Rasmussen + + docs: Enable and fix gtk-doc warnings + * Makefile: Enable gtk-doc warnings, like the rest of GStreamer + * addresspool/mediafactory: Add missing annotation colon + * stream: Annotate return value + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725528 + +2014-02-28 09:36:49 +0100 Sebastian Dröge + + * common: + Automatic update of common submodule + From fe1672e to bcb1518 + +2014-02-26 22:15:51 +0100 Stefan Sauer + + * common: + Automatic update of common submodule + From 1a07da9 to fe1672e + +2014-02-25 15:13:40 +0000 Tim-Philipp Müller + + * examples/Makefile.am: + examples: use LDADD for libs instead of LDFLAGS + +2014-02-25 14:42:09 +0000 Tim-Philipp Müller + + * configure.ac: + configure: make sure releases are in .doap file + +2014-02-25 14:11:00 +0000 Tim-Philipp Müller + + * 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 + + * examples/.gitignore: + examples: add cgroup test binary to .gitignore + +2014-02-25 14:06:47 +0000 Tim-Philipp Müller + + * 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 + + * .gitignore: + .gitignore: ignore temp files created in the course of 'make check' + +2014-02-18 09:44:34 +0100 Branko Subasic + + * 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 + +2014-02-18 16:58:45 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + Revert "media: only flush when setting a new start position" + This reverts commit f67fc23aab59f28796bebf130504ff46ccb97b0a. + We need to do the flush in all cases, demuxer block currently for + non-flushing seeks. + +2014-02-18 16:38:39 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: only flush when setting a new start position + Only flush the pipeline when we change the start position with + a seek. + See https://bugzilla.gnome.org/show_bug.cgi?id=724611 + +2014-02-17 10:43:05 +0100 Göran Jönsson + + * 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 + + * 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 + +2014-02-09 10:52:29 +0100 Sebastian Dröge + + * Makefile.am: + build: Ship gst-rtsp-server.doap file + +2014-02-09 10:47:09 +0100 Sebastian Dröge + + * tests/check/gst/rtspserver.c: + tests: Fix another compiler warning with gcc + +2014-02-09 10:45:28 +0100 Sebastian Dröge + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-stream.c: + stream: handle NULL seqnum and rtptime arguments + +2014-01-31 15:02:22 +0100 Ognyan Tonchev + + * 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 + + * 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 + + * common: + Automatic update of common submodule + From f7bc1c3 to 1a07da9 + +2014-01-28 14:51:26 +0100 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-stream.c: + stream: fix compilation + +2014-01-07 12:21:09 +0100 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + 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 + + * 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 + + * 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 + + * autogen.sh: + * common: + * configure.ac: + Fix broken gettext setup which is not used anyway + +2013-12-22 22:36:06 +0000 Tim-Philipp Müller + + * common: + Automatic update of common submodule + From dbedaa0 to d48bed3 + +2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué + + * 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 + + * 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 + + * 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 + + * 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 + + * .gitignore: + Make git ignore more unit test binaries + +2013-12-12 00:36:07 +0000 Tim-Philipp Müller + + * 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 + + 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 + + * 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 + + * gst/rtsp-server/rtsp-stream-transport.c: + stream-transport: free url in finalize + +2013-11-29 15:50:23 +0100 Ognyan Tonchev + + * gst/rtsp-server/rtsp-media.c: + media: also do state change in suspended state + +2013-11-29 10:53:08 +0100 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-client.c: + client: suspend after SDP and unsuspend before PLAYING + Based on patches by Ognyan Tonchev + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257 + +2013-11-28 14:10:19 +0100 Wim Taymans + + * 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 + See https://bugzilla.gnome.org/show_bug.cgi?id=711257 + +2013-11-28 14:06:53 +0100 Wim Taymans + + * 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 + + * gst/rtsp-server/rtsp-media.c: + media: refactor starting and waiting for preroll + Based on patches from Ognyan Tonchev + See https://bugzilla.gnome.org/show_bug.cgi?id=711257 + +2013-11-28 13:42:21 +0100 Wim Taymans + + * 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 + See https://bugzilla.gnome.org/show_bug.cgi?id=711257 + +2013-11-27 15:42:45 +0100 Edward Hervey + + * 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 + + * gst/rtsp-server/rtsp-media.c: + media: move default implementations to where they are used + +2013-11-26 16:25:37 +0100 Wim Taymans + + * 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 + + * 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 + + * 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 + + 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 + + * 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 + + 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 + + * 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 + + 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-auth.c: + auth: small typos + +2013-10-19 19:25:27 +0200 Sebastian Rasmussen + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * common: + Automatic update of common submodule + From 865aa20 to dbedaa0 + +2013-10-14 12:03:07 +0200 Ognyan Tonchev + + * 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 + + * 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 + + * 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 + + * .gitignore: + gitignore: Add new test binary + +2013-10-09 15:19:12 +0200 Ognyan Tonchev + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: remove old line + +2013-10-01 13:15:19 +0200 Jonas Holmberg + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-media.c: + media: Return FALSE if seeking is not supported + +2013-10-01 17:16:11 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * common: + Automatic update of common submodule + From 6b03ba7 to 865aa20 + +2013-09-23 14:28:04 +0200 Jonas Holmberg + + * 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 + + * 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 + + * 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 + + * common: + Automatic update of common submodule + From b613661 to 6b03ba7 + +2013-09-19 18:46:14 +0100 Tim-Philipp Müller + + * common: + Automatic update of common submodule + From 74a6857 to b613661 + +2013-09-19 17:39:24 +0100 Tim-Philipp Müller + + * common: + Automatic update of common submodule + From 01a7a46 to 74a6857 + +2013-09-19 15:44:26 +0200 Jonas Holmberg + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-client.c: + client: map URL to path in requests + +2013-09-16 16:47:40 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * examples/Makefile.am: + Makefile: add rule for appsrc example + +2013-08-23 15:14:29 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-media.c: + media: set protocols on streams + +2013-08-16 16:16:31 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: use protocols supported by stream + +2013-08-16 16:16:00 +0200 Wim Taymans + + * 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 + + * gst/rtsp-server/rtsp-media.c: + media: configure protocols in new streams + +2013-08-16 16:08:43 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-client.h: + small documentation fix + +2013-07-11 16:11:55 -0400 Youness Alaoui + + * gst/rtsp-server/rtsp-client.c: + Do not take range header if range is invalid + +2013-08-02 16:57:26 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * examples/test-cgroups.c: + tests: add some more docs + +2013-07-22 14:25:04 +0200 Wim Taymans + + * 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 + + * 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 + + * examples/test-auth.c: + * gst/rtsp-server/rtsp-auth.c: + auth: use defines when possible + +2013-07-18 11:44:21 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-address-pool.c: + address-pool: improve docs + +2013-07-18 12:26:45 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-permissions.c: + permissions: add the role to the copy + +2013-07-17 19:35:33 -0400 Olivier Crête + + * gst/rtsp-server/rtsp-permissions.c: + permissions: Also copy the roles + +2013-07-17 19:32:09 -0400 Olivier Crête + + * gst/rtsp-server/rtsp-permissions.c: + permissions: Make it build + +2013-07-16 12:36:56 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-address-pool.h: + docs: small fixes + +2013-07-16 12:32:51 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-permissions.c: + permissions: implement _remove_role + +2013-07-15 17:12:43 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-permissions.c: + permissions: update docs + +2013-07-15 16:48:37 +0200 Wim Taymans + + * 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 + + * 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 + + * docs/README: + README: update readme + +2013-07-15 15:25:00 +0200 Wim Taymans + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-server.c: + server: add small debug + +2013-07-15 11:58:58 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-mount-points.c: + mount-points: add some debug + +2013-07-12 17:26:55 +0200 Wim Taymans + + * tests/check/gst/client.c: + tests: almost fix test + +2013-07-12 17:07:53 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-auth.c: + auth: fix typo + +2013-07-12 15:19:29 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-mount-points.c: + mount-points: fix debug + +2013-07-11 17:28:17 +0200 Wim Taymans + + * tests/check/gst/media.c: + tests: fix media test + +2013-07-11 17:28:04 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-thread-pool.c: + thread-pool: we don't require a state + +2013-07-11 17:18:58 +0200 Wim Taymans + + * 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 + + * tests/check/gst/client.c: + tests: fix client test + +2013-07-11 16:57:14 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * configure.ac: + * examples/Makefile.am: + * examples/test-cgroups.c: + examples: add cgroups example + +2013-07-10 20:55:03 +0200 Wim Taymans + + * tests/check/gst/rtspserver.c: + tests: fix compilation + +2013-07-10 20:48:47 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-thread-pool.c: + thread-pool: fix vmethod invocation + +2013-07-10 20:48:18 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-server.c: + server: configure client thread pool + +2013-07-10 17:06:36 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-auth.c: + auth: debug authorization check + +2013-07-09 20:44:51 +0200 Wim Taymans + + * 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 + + * gst/rtsp-server/rtsp-media-factory.c: + factory: pass permissions to media by default + +2013-07-09 16:09:07 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/Makefile.am: + rtsp-server: Allow building of static library + +2013-07-02 15:59:16 +0200 Wim Taymans + + * tests/check/gst/mediafactory.c: + tests: fix compilation + +2013-07-02 15:54:43 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-client.c: + client: fix comment + +2013-07-01 14:45:49 +0200 Wim Taymans + + * 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 + + * gst/rtsp-server/rtsp-stream.c: + stream: improve docs + +2013-07-01 12:04:45 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-stream-transport.c: + stream-transport: remove old if 0 block + +2013-06-27 11:21:42 +0200 Patricia Muscalu + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-media-factory.h: + media-factory: fix typo + +2013-06-26 14:42:15 +0200 Ognyan Tonchev + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-media.c: + media: fix docs + +2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * common: + Automatic update of common submodule + From 098c0d7 to 01a7a46 + +2013-05-29 13:45:00 +0200 David Svensson Fors + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * examples/Makefile.am: + * examples/test-video.c: + example: add TLS example + +2013-05-31 11:42:36 +0200 Wim Taymans + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-server.c: + server: don't crash on NULL GError + +2013-05-30 10:46:33 +0200 Patricia Muscalu + + * 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 + + * gst/rtsp-server/rtsp-client.c: + client: improve debug + +2013-05-30 07:18:22 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * common: + Automatic update of common submodule + From 5edcd85 to 098c0d7 + +2013-04-23 11:28:39 +0200 Ognyan Tonchev + + * tests/check/gst/media.c: + tests: add dynamic payloader prepare/unprepare check + +2013-04-23 10:27:35 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: release lock when removing fakesink + +2013-04-23 10:16:17 +0200 Wim Taymans + + * 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 + + * common: + Automatic update of common submodule + From 3cb3d3c to 5edcd85 + +2013-04-22 17:34:37 +0200 Wim Taymans + + * 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 + + * examples/test-sdp.c: + tests: add example of reusable pipelines + +2013-04-22 17:32:31 +0200 Ognyan Tonchev + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * common: + Automatic update of common submodule + From aed87ae to 3cb3d3c + +2013-04-12 11:34:38 +0100 Wim Taymans + + * 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 + + * 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 + + * common: + Automatic update of common submodule + From 04c7a1e to aed87ae + +2013-04-09 20:39:58 +0200 Wim Taymans + + * 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 + + * gst/rtsp-server/rtsp-media.c: + media: small cleanup + +2013-03-20 12:33:54 +0100 David Svensson Fors + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * configure.ac: + configure: remove out-of-date comment + +2013-03-12 00:05:49 +0000 Tim-Philipp Müller + + * .gitignore: + .gitignore: ignore more build files + +2013-03-12 00:03:36 +0000 Tim-Philipp Müller + + * tests/check/Makefile.am: + tests: use right _LIBS variable for gst-plugins-base libs + +2013-03-11 11:35:14 +0100 Wim Taymans + + * tests/check/Makefile.am: + check: add librtp to libs + +2013-02-20 19:37:51 -0500 Olivier Crête + + * 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 + + * tests/check/gst/rtspserver.c: + tests: Make sure packets are actually received + +2013-02-19 18:27:20 -0500 Olivier Crête + + * gst/rtsp-server/rtsp-stream.c: + stream: Select unicast address from pool if appropriate + +2013-02-19 16:43:08 -0500 Olivier Crête + + * gst/rtsp-server/rtsp-stream.c: + stream: Properties are always there in Gst 1.0 + +2013-02-19 16:36:20 -0500 Olivier Crête + + * tests/check/gst/addresspool.c: + tests: Add tests for unicast addresses in pool + +2013-02-20 14:26:03 -0500 Olivier Crête + + * 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 + + * 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 + + * 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 + + * 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 + + * tests/check/gst/rtspserver.c: + tests: Add test for client disconnection + +2013-02-18 20:15:41 -0500 Olivier Crête + + * tests/check/gst/rtspserver.c: + tests: Test client and session timeouts with multiple threads + +2013-02-18 14:59:58 -0500 Olivier Crête + + * 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 + + * tests/check/gst/rtspserver.c: + tests: Test that slow DESCRIBE don't block other clients + +2013-02-14 19:52:09 -0500 Olivier Crête + + * tests/check/gst/client.c: + tests: Add tests for client-requested multicast address + +2013-02-14 13:44:54 -0500 Olivier Crête + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * tests/check/gst/addresspool.c: + tests: Add more tests for the address pool + +2013-02-12 16:29:25 -0500 Olivier Crête + + * 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 + + * common: + Automatic update of common submodule + From 2de221c to 04c7a1e + +2013-02-07 16:18:08 -0600 George McCollister + + * 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 + + * common: + Automatic update of common submodule + From a942293 to 2de221c + +2013-01-28 10:31:50 +0100 Wim Taymans + + * 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 + + * 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 + + * common: + Automatic update of common submodule + From acb04d9 to a942293 + +2012-12-14 11:58:29 +0100 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-media.c: + media: improve debug + +2012-11-30 12:53:02 +0100 Wim Taymans + + * 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 + + * gst/rtsp-server/rtsp-media.c: + media: use g_object_ref/unref for GObjects + +2012-11-30 07:05:25 +0100 Alessandro Decina + + * 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 + + * gst/rtsp-server/rtsp-media.c: + Fix compiler warning + +2012-11-30 06:14:49 +0100 Alessandro Decina + + * 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 + + * gst/rtsp-server/rtsp-session-media.h: + small cleanup + +2012-11-29 17:20:56 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * tests/check/gst/media.c: + media: avoid element leak + +2012-11-29 17:20:26 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: require an element in media constructor + +2012-11-29 17:07:30 +0100 Wim Taymans + + * 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 + + * gst/rtsp-server/rtsp-client.c: + client: TEARDOWN brings that state to Init again + +2012-11-29 11:11:05 +0100 Wim Taymans + + * 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 + + * tests/check/Makefile.am: + * tests/check/gst/media.c: + tests: add media tests + +2012-11-28 14:45:30 +0100 Wim Taymans + + * 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 + + * tests/check/Makefile.am: + * tests/check/gst/mediafactory.c: + tests: add mediafactory test + +2012-11-28 12:40:18 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-stream.c: + stream: improve debug + +2012-11-28 12:39:37 +0100 Wim Taymans + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-media-factory.c: + media-factory: require an url + +2012-11-28 11:40:33 +0100 Wim Taymans + + * examples/test-uri.c: + examples: fix include + +2012-11-28 11:17:27 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.h: + server: remove unused include + +2012-11-28 11:07:57 +0100 Wim Taymans + + * tests/check/Makefile.am: + * tests/check/gst/mountpoints.c: + tests: add test for mountpoints + +2012-11-28 11:05:08 +0100 Wim Taymans + + * 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 + + * gst/rtsp-server/rtsp-mount-points.c: + mounts: add g_return_if guards + +2012-11-27 12:51:55 +0100 Wim Taymans + + * tests/check/gst/client.c: + tests: add more tests + +2012-11-27 12:33:02 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: improve debug + +2012-11-27 12:24:21 +0100 Wim Taymans + + * 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 + + * common: + update common + +2012-11-27 12:13:59 +0100 Wim Taymans + + * tests/check/gst/client.c: + test: add test for session in options request + +2012-11-27 12:11:41 +0100 Wim Taymans + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-client.c: + client: small cleanup + +2012-11-26 17:34:35 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.h: + client: remove unused include + +2012-11-26 17:34:24 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: fix compilation + +2012-11-26 17:28:29 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: call destroy without the lock + +2012-11-26 17:20:39 +0100 Wim Taymans + + * 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 + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: small cleanup + +2012-11-26 16:39:26 +0100 Wim Taymans + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-client.c: + client: log more errors + +2012-11-26 13:35:48 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: fix compilation + +2012-11-26 13:16:59 +0100 Wim Taymans + + * 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 + + * 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 + + * configure.ac: + configure: bump required version of -base + +2012-11-21 17:21:28 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: fix seeking + +2012-11-21 16:41:56 +0100 Wim Taymans + + * 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 + + * examples/test-launch.c: + examples: fix whitespace + +2012-11-20 13:34:46 +0100 Wim Taymans + + * 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 + + * examples/test-uri.c: + test-uri: remove mapping example + +2012-11-20 12:47:20 +0100 Wim Taymans + + * examples/test-uri.c: + test-uri: fix callback signature + +2012-11-20 12:29:55 +0100 Wim Taymans + + * 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 + + * gst/rtsp-server/rtsp-media-factory-uri.c: + factory-uri: add some debug + +2012-11-20 12:24:13 +0100 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-server.c: + server: fix small leak + +2012-11-20 09:42:51 +0100 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * common: + Automatic update of common submodule + From 6bb6951 to a72faea + +2012-11-17 00:11:27 +0000 Tim-Philipp Müller + + * 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 + + * 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 + + * 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 + + * 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 + + * examples/test-multicast.c: + test: set shared + +2012-11-15 16:18:29 +0100 Wim Taymans + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-media.c: + media: configure address pool in new streams + +2012-11-15 15:36:21 +0100 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-address-pool.c: + address-pool: small cleanups + +2012-11-14 15:50:42 +0100 Wim Taymans + + * tests/check/Makefile.am: + * tests/check/gst/addresspool.c: + tests: add addresspool unit test + +2012-11-14 15:49:06 +0100 Wim Taymans + + * 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 + + * gst/rtsp-server/rtsp-server.c: + server: set default max-threads property + +2012-11-13 11:54:17 +0100 Wim Taymans + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-stream.c: + * gst/rtsp-server/rtsp-stream.h: + stream: add locking + +2012-11-12 17:11:18 +0100 Wim Taymans + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + session: add locking + +2012-11-12 16:30:16 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + server: free old socket + +2012-11-12 16:18:57 +0100 Wim Taymans + + * 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 + + * gst/rtsp-server/rtsp-media-factory.c: + media-factory: add locking + +2012-11-12 16:03:21 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + auth: add locking + +2012-11-12 15:53:28 +0100 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-client.c: + client: small cleanup + +2012-10-27 23:49:24 +0200 Wim Taymans + + * 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 + + * gst/rtsp-server/rtsp-client.c: + client: refactor transport parsing + +2012-10-27 21:05:03 +0200 Wim Taymans + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-stream.c: + stream: transports must already have been removed + +2012-10-26 17:28:10 +0200 Wim Taymans + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-media.c: + media: signal unprepared when we actually finish + +2012-10-26 15:19:23 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * docs/README: + docs: update docs + +2012-10-26 11:24:55 +0100 Tim-Philipp Müller + + * configure.ac: + configure: bump version number after refactoring + +2012-10-25 21:29:58 +0200 Wim Taymans + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/Makefile.am: + Explicitly link against gio. Fix link error on mac. + +2012-10-10 11:13:10 +0200 Ognyan Tonchev + + * 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 + + * 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 + + * common: + Automatic update of common submodule + From 6c0b52c to 6bb6951 + +2012-10-01 16:13:50 +0200 Patricia Muscalu + + * 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 + + * common: + Automatic update of common submodule + From 4f962f7 to 6c0b52c + +2012-09-10 16:25:57 +0200 Ognyan Tonchev + + * gst/rtsp-server/rtsp-media.c: + media: fix check for seekability + +2012-09-07 17:14:30 +0200 Wim Taymans + + * 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 + + * gst/rtsp-server/rtsp-server.c: + server: remove obsolete includes + +2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque + + 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 + + * 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 + + * 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 + + * common: + Automatic update of common submodule + From 668acee to 4f962f7 + +2012-08-15 15:54:32 +0200 Patricia Muscalu + + * 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 + + * common: + Automatic update of common submodule + From 94ccf4c to 668acee + +2012-07-18 15:54:49 +0200 Patricia Muscalu + + * 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 + + * common: + Automatic update of common submodule + From 98e386f to 94ccf4c + +2012-07-10 11:39:58 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: fix docs + +2012-07-03 18:06:00 +0200 Ognyan Tonchev + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * gst/rtsp-server/rtsp-client.c: + client: don't leak transports + +2012-06-12 14:45:39 +0200 David Svensson Fors + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * 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 + + * common: + Automatic update of common submodule + From 03a0e57 to 98e386f + +2012-06-06 18:20:49 +0200 Edward Hervey + + * common: + Automatic update of common submodule + From 1fab359 to 03a0e57 + +2012-06-06 14:49:02 +0200 David Svensson Fors + + * 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 + + * common: + Automatic update of common submodule + From f1b5a96 to 1fab359 + +2012-05-31 13:11:43 +0200 Sebastian Dröge + + * common: + Automatic update of common submodule + From 92b7266 to f1b5a96 + +2012-05-30 12:48:51 +0200 Sebastian Dröge + + * common: + Automatic update of common submodule + From ec1c4a8 to 92b7266 + +2012-05-30 11:27:31 +0200 Sebastian Dröge + + * common: + Automatic update of common submodule + From 3429ba6 to ec1c4a8 + +2012-05-22 15:37:25 +0200 David Svensson Fors + + * 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 + + * common: + Automatic update of common submodule + From dc70203 to 3429ba6 + +2012-05-11 09:42:47 +0200 Wim Taymans + + * 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 + + * common: + Automatic update of common submodule + From 6db25be to dc70203 + +2012-04-13 15:27:22 +0200 Sebastian Dröge + + * 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 + + * 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 + + * common: + Automatic update of common submodule + From 464fe15 to 6db25be + +2012-04-05 18:45:43 +0200 Sebastian Dröge + + * common: + Automatic update of common submodule + From 7fda524 to 464fe15 + +2012-04-04 14:45:55 +0200 Sebastian Dröge + + * configure.ac: + * docs/libs/Makefile.am: + * docs/version.entities.in: + * gst-rtsp.spec.in: + * gst/rtsp-server/Makefile.am: + * pkgconfig/Makefile.am: + * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in: + * pkgconfig/gstreamer-rtsp-server.pc.in: + * tests/Makefile.am: + rtsp-server: Update versioning + +2012-03-29 15:12:21 +0200 Sebastian Dröge + + Merge remote-tracking branch 'origin/0.10' + Conflicts: + gst/rtsp-server/rtsp-session-pool.c + +2012-03-27 10:13:20 +0200 Sebastian Dröge + + * gst/rtsp-server/rtsp-session-pool.c: + rtsp-server: Don't use deprecated GLib API + +2012-03-26 12:23:36 +0200 Wim Taymans + + Replace master with 0.11 + +2012-03-26 12:22:05 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + +2012-03-26 12:20:51 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + +2012-03-19 10:48:09 +0000 Vincent Penquerc'h + + * docs/README: + A couple minor typo fixes + +2012-03-13 18:10:53 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: fix state of the appqueue + +2012-03-13 16:06:50 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory-uri.c: + factory: use videoconvert + +2012-03-13 16:02:47 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory-uri.c: + factory: change to new style caps + +2012-03-07 15:03:55 +0100 Wim Taymans + + * 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.c: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + * gst/rtsp-server/rtsp-session-pool.c: + rtsp-server: port to GIO + Port to GIO + +2012-03-07 15:03:24 +0100 Wim Taymans + + * configure.ac: + configure: fix build + +2012-02-29 15:56:06 +0000 Tim-Philipp Müller + + * docs/README: + docs: fix for gst_rtsp_server_set_port() -> _set_service() + https://bugzilla.gnome.org/show_bug.cgi?id=666548 + +2012-02-13 11:42:51 +0000 Tim-Philipp Müller + + * configure.ac: + * examples/Makefile.am: + First rule of gst-rtsp-server club: don't talk about gst-phonon + +2012-02-13 11:40:44 +0000 Tim-Philipp Müller + + * configure.ac: + * pkgconfig/Makefile.am: + * pkgconfig/gst-rtsp-server-uninstalled.pc.in: + * pkgconfig/gst-rtsp-server.pc.in: + * pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in: + * pkgconfig/gstreamer-rtsp-server.pc.in: + pkg-config: rename gst-rtsp-server-0.11.pc to gstreamer-rtsp-server-0.11.pc + For consistency with all other modules. + +2012-02-13 11:06:33 +0000 Tim-Philipp Müller + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: update for new map API + +2012-02-13 10:37:37 +0000 Tim-Philipp Müller + + * .gitignore: + * bindings/Makefile.am: + * bindings/python/Makefile.am: + * bindings/python/arg-types.py: + * bindings/python/codegen/Makefile.am: + * bindings/python/codegen/__init__.py: + * bindings/python/codegen/argtypes.py: + * bindings/python/codegen/code-coverage.py: + * bindings/python/codegen/codegen.py: + * bindings/python/codegen/definitions.py: + * bindings/python/codegen/defsparser.py: + * bindings/python/codegen/docextract.py: + * bindings/python/codegen/docgen.py: + * bindings/python/codegen/fileprefix.override: + * bindings/python/codegen/fileprefixmodule.c: + * bindings/python/codegen/h2def.py: + * bindings/python/codegen/mergedefs.py: + * bindings/python/codegen/mkskel.py: + * bindings/python/codegen/override.py: + * bindings/python/codegen/reversewrapper.py: + * bindings/python/codegen/scmexpr.py: + * bindings/python/rtspserver-types.defs: + * bindings/python/rtspserver.defs: + * bindings/python/rtspserver.override: + * bindings/python/rtspservermodule.c: + * bindings/python/test.py: + * configure.ac: + python: remove pygst-based python bindings + pygi is the future, apparently. + +2012-01-25 14:12:41 +0100 Thomas Vander Stichele + + * common: + Automatic update of common submodule + From c463bc0 to 7fda524 + +2012-01-25 11:40:59 +0100 Sebastian Dröge + + * common: + Automatic update of common submodule + From 2a59016 to c463bc0 + +2012-01-18 16:48:41 +0100 Sebastian Dröge + + * common: + Automatic update of common submodule + From 0807187 to 2a59016 + +2012-01-04 19:56:02 +0000 Tim-Philipp Müller + + * common: + Automatic update of common submodule + From 11f0cd5 to 0807187 + +2011-12-09 11:00:46 +0100 Wim Taymans + + * examples/test-auth.c: + example: update for new caps + +2011-12-09 10:53:30 +0100 Wim Taymans + + * examples/test-video.c: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + rtsp-server: port some more to 0.11 + Fix caps. + Remove bufferlist stuff + Update for new API. + Add queue before appsink now that preroll-queue-len is gone. + Update for request pad changes. + +2011-11-03 16:14:03 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-11-03 16:06:23 +0100 Fabian Deutsch + + * bindings/vala/packages/gst-rtsp-server-0.10.metadata: + bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership. + Signed-off-by: Fabian Deutsch + +2011-11-03 16:06:23 +0100 Fabian Deutsch + + * bindings/vala/packages/gst-rtsp-server-0.10.metadata: + bindings: Fix vala binding of gst_rtsp_media_mapping_add_factory to transfer ownership. + Signed-off-by: Fabian Deutsch + +2011-11-03 12:58:42 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-11-03 12:55:24 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: add a seekable boolean + Maintain the seekable state with a new variable instead of reusing the + is_live variable. + +2011-09-16 11:31:17 -0400 Victor Gottardi + + * gst/rtsp-server/rtsp-media.c: + Disallow seek in live media + +2011-11-03 11:58:42 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-11-03 10:48:40 +0100 mat + + * gst/rtsp-server/rtsp-server.c: + #ifdef statements for windows socket creation were missing + +2011-09-06 21:53:46 +0200 Stefan Sauer + + * common: + Automatic update of common submodule + From a39eb83 to 11f0cd5 + +2011-09-06 16:07:18 +0200 Stefan Sauer + + * common: + Automatic update of common submodule + From 605cd9a to a39eb83 + +2011-08-16 16:39:26 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-08-16 16:07:04 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: use method to access property + +2011-08-16 15:15:19 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + media-factory: add protocols property + Add a property to configure the allowed protocols in the media created from the + factory. + +2011-08-16 15:03:06 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + media-factory: add media-configure signal + Add signal to allow the application to configure the media after it was created + from the factory. + +2011-08-16 16:07:04 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: use method to access property + +2011-08-16 15:15:19 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + media-factory: add protocols property + Add a property to configure the allowed protocols in the media created from the + factory. + +2011-08-16 15:03:06 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + media-factory: add media-configure signal + Add signal to allow the application to configure the media after it was created + from the factory. + +2011-08-16 14:50:50 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-08-16 13:43:44 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: use media multicast group + +2011-08-16 13:37:50 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-server.h: + * gst/rtsp-server/rtsp-session-pool.h: + * gst/rtsp-server/rtsp-session.h: + retab some .h + +2011-08-16 13:31:52 +0200 Robert Krakora + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-sdp.h: + sdp: copy and free the server ip address + Copy and free the server ip address to make memory management easier later. + +2011-08-16 13:27:39 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + media-factory: configure multicast in media + +2011-08-16 13:25:16 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: add property for multicast group + Add a property to configure the multicast group in the media. + Based on patches from Marc Leeman and Robert Krakora. + +2011-08-16 13:13:36 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + media-factory: add property for multicast group + Add a property to configure the multicast group in the media factory. + Based on patches from Marc Leeman and Robert Krakora. + +2011-08-16 12:51:44 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: do configuration of transport in one place + Move the configuration of the transport destination address to where we also + configure the other bits. + +2011-08-16 13:43:44 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: use media multicast group + +2011-08-16 13:37:50 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-server.h: + * gst/rtsp-server/rtsp-session-pool.h: + * gst/rtsp-server/rtsp-session.h: + retab some .h + +2011-08-16 13:31:52 +0200 Robert Krakora + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-sdp.h: + sdp: copy and free the server ip address + Copy and free the server ip address to make memory management easier later. + +2011-08-16 13:27:39 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + media-factory: configure multicast in media + +2011-08-16 13:25:16 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: add property for multicast group + Add a property to configure the multicast group in the media. + Based on patches from Marc Leeman and Robert Krakora. + +2011-08-16 13:13:36 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + media-factory: add property for multicast group + Add a property to configure the multicast group in the media factory. + Based on patches from Marc Leeman and Robert Krakora. + +2011-08-16 12:51:44 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: do configuration of transport in one place + Move the configuration of the transport destination address to where we also + configure the other bits. + +2011-08-16 12:11:59 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-08-16 12:09:48 +0200 Robert Krakora + + * gst/rtsp-server/rtsp-client.c: + client: destroy pipeline on client disconnect with no prior TEARDOWN. + The problem occurs when the client abruptly closes the connection without + issuing a TEARDOWN. The TEARDOWN handler in the rtsp-client.c file of the RTSP + server is where the pipeline gets torn down. Since this handler is not called, + the pipeline remains and is up and running. Subsequent clients get their own + pipelines and if the do not issue TEARDOWNs then those pipelines will also + remain up and running. This is a resource leak. + +2011-08-16 11:53:37 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-06-30 10:13:59 +0200 Emmanuel Pacaud + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + media-factory: add a "media-constructed" signal to GstRTSPMediaFactory + For example, it can be used to retrieve source elements like appsrc, in a more + convenient way than subclassing get_element. + +2011-08-16 11:12:33 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-08-11 18:07:08 -0700 David Schleef + + * gst/rtsp-server/rtsp-server.c: + rtsp-server: hold on to reference while using object + +2011-08-04 08:59:17 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: use new api + +2011-08-04 08:58:58 +0200 Wim Taymans + + * configure.ac: + configure: use unstable api + +2011-06-27 11:26:26 -0700 David Schleef + + * gst/rtsp-server/rtsp-client.c: + client: fix reference counting + +2011-07-20 17:16:42 +0200 Thijs Vermeir + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + fix compiler warnings about unused variables + +2011-07-19 16:10:39 +0200 Stefan Sauer + + * examples/test-launch.c: + * examples/test-readme.c: + * examples/test-uri.c: + * examples/test-video.c: + examples: tell rtsp uri when ready + +2011-06-23 11:30:14 -0700 David Schleef + + * common: + Automatic update of common submodule + From 69b981f to 605cd9a + +2011-06-13 19:05:57 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: update for buffer API change + +2011-06-07 10:54:26 +0200 Edward Hervey + + * gst/rtsp-server/Makefile.am: + Makefile.am: 0.10 => @GST_MAJORMINOR@ + +2011-06-07 10:59:16 +0200 Edward Hervey + + * gst/rtsp-server/rtsp-media-factory-uri.c: + rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer + +2011-06-07 10:59:03 +0200 Edward Hervey + + * gst/rtsp-server/.gitignore: + .gitignore: 0.10 => 0.11 + +2011-06-07 10:54:26 +0200 Edward Hervey + + * gst/rtsp-server/Makefile.am: + Makefile.am: 0.10 => @GST_MAJORMINOR@ + +2011-05-24 18:26:06 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-05-19 23:00:52 +0300 Stefan Kost + + * common: + Automatic update of common submodule + From 9e5bbd5 to 69b981f + +2011-05-18 16:14:10 +0300 Stefan Kost + + * common: + Automatic update of common submodule + From fd35073 to 9e5bbd5 + +2011-05-18 12:27:35 +0300 Stefan Kost + + * common: + Automatic update of common submodule + From 46dfcea to fd35073 + +2011-05-17 09:48:13 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-media.c: + media: port to new caps API + +2011-05-17 09:45:04 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + +2011-05-03 21:13:15 +0200 Fabian Deutsch + + * bindings/vala/gst-rtsp-server-0.10.vapi: + Updated Vala bindings. + Signed-off-by: Fabian Deutsch + +2011-05-03 16:24:28 +0200 Fabian Deutsch + + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + Add a signal for newly connected clients. + Signed-off-by: Fabian Deutsch + +2011-05-08 13:15:19 +0200 Alessandro Decina + + * bindings/python/rtspserver.override: + python: override gst_rtsp_media_mapping_add_factory to fix refcounting + +2011-04-26 19:22:50 +0200 Wim Taymans + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-funnel.c: + * gst/rtsp-server/rtsp-funnel.h: + * gst/rtsp-server/rtsp-media.c: + rtsp-server: port to 0.11 + +2011-04-26 19:14:18 +0200 Wim Taymans + + * common: + add common + +2011-04-26 19:07:13 +0200 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + common + configure.ac + +2011-04-24 14:07:11 +0100 Tim-Philipp Müller + + * common: + Automatic update of common submodule + From c3cafe1 to 46dfcea + +2011-04-20 11:19:38 +0200 Alessandro Decina + + * bindings/python/Makefile.am: + * bindings/python/rtspserver.defs: + python bindings: wrap GstRTSPMediaFactoryClass vfuncs + +2011-04-20 11:13:56 +0200 Alessandro Decina + + * bindings/python/arg-types.py: + python bindings: add GstRTSPUrlParam + Needed to implement MediaFactory virtual proxies + +2011-04-20 10:19:46 +0200 Alessandro Decina + + * bindings/python/arg-types.py: + python bindings: fix returning GstRTSPUrl types + +2011-04-20 10:17:07 +0200 Alessandro Decina + + * bindings/python/arg-types.py: + python bindings: add arg type for GstRTSPUrl + +2011-04-20 10:16:08 +0200 Alessandro Decina + + * bindings/python/rtspserver.defs: + python bindings: fix the definition of MediaFactory.collect_stream + +2011-04-04 15:59:50 +0300 Stefan Kost + + * common: + Automatic update of common submodule + From 1ccbe09 to c3cafe1 + +2011-03-25 22:38:06 +0100 Sebastian Dröge + + * common: + Automatic update of common submodule + From 193b717 to 1ccbe09 + +2011-03-25 14:58:34 +0200 Stefan Kost + + * common: + Automatic update of common submodule + From b77e2bf to 193b717 + +2011-03-25 10:04:57 +0100 Sebastian Dröge + + * Makefile.am: + build: Include lcov.mak to allow test coverage report generation + +2011-03-25 09:35:15 +0100 Sebastian Dröge + + * common: + Automatic update of common submodule + From d8814b6 to b77e2bf + +2011-03-25 09:11:40 +0100 Sebastian Dröge + + * common: + Automatic update of common submodule + From 6aaa286 to d8814b6 + +2011-03-24 18:51:37 +0200 Stefan Kost + + * common: + Automatic update of common submodule + From 6aec6b9 to 6aaa286 + +2011-03-18 19:34:57 +0100 Luis de Bethencourt + + * autogen.sh: + autogen: wingo signed comment + +2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya + + * gst/rtsp-server/rtsp-session-pool.c: + session: use full charset for RTSP session ID + As specified in RFC 2326 section 3.4 use full valid charset to make guessing + session ID more difficult. + https://bugzilla.gnome.org/show_bug.cgi?id=643812 + +2011-03-07 10:23:06 +0100 Sebastian Dröge + + * gst/rtsp-server/Makefile.am: + rtsp-server: Don't install the funnel header + +2011-02-28 18:35:03 +0100 Mark Nauwelaerts + + * common: + Automatic update of common submodule + From 1de7f6a to 6aec6b9 + +2011-02-26 19:58:02 +0000 Tim-Philipp Müller + + * configure.ac: + configure: require core/base 0.10.31 + Needed at least for gst_plugin_feature_rank_compare_func(). + +2011-02-14 12:56:29 +0200 Stefan Kost + + * common: + Automatic update of common submodule + From f94d739 to 1de7f6a + +2011-02-02 15:37:03 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: remove more unused code + +2011-02-02 15:30:45 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: remove duplicate filtering + Remove the duplicate filtering code now that we have a released -good version. + Give a warning instead. + +2011-01-31 17:38:47 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media.c: + media: fix default buffer size + +2011-01-31 17:37:02 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + media-factory: add property to configure the buffer-size + Add a property to configure the kernel UDP buffer size. + +2011-01-31 17:28:22 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: add property to configure kernel buffer sizes + Add a property to configure the kernel UDP buffer size. + +2011-01-26 15:52:54 +0000 Tim-Philipp Müller + + * configure.ac: + configure: set PYGOBJECT_REQ before using it + https://bugzilla.gnome.org/show_bug.cgi?id=640641 + +2011-01-24 11:59:22 +0000 Tim-Philipp Müller + + * docs/Makefile.am: + docs: recursive into sub-directories on 'make upload' + +2011-01-24 11:53:17 +0000 Tim-Philipp Müller + + * docs/libs/gst-rtsp-server-docs.sgml: + * docs/version.entities.in: + docs: mention full version these docs are for, not just major-minor + +2011-01-24 12:07:17 +0100 Wim Taymans + + * configure.ac: + back to development + +=== release 0.10.8 === + +2011-01-24 11:57:12 +0100 Wim Taymans + + * configure.ac: + release 0.10.8 + +2011-01-19 15:29:55 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + rtsp-server: clarify docs a little + +2011-01-13 18:57:15 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: init debug category before starting thread + +2011-01-13 18:40:48 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-auth.c: + auth: add realm to make it more spec compliant + +2011-01-12 18:57:41 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + server: add locking + +2011-01-12 18:33:49 +0100 Wim Taymans + + * examples/test-video.c: + example: improve example docs a little + +2011-01-12 18:26:57 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + server: ensure the watch has a ref to the server + +2011-01-12 18:24:44 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + server: simpify channel function + +2011-01-12 18:18:13 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + server: simplify management of channel and source + We don't need to keep around the channel and source objects. Let the mainloop + and the source manage the source and channel respectively. + +2011-01-12 18:17:26 +0100 Wim Taymans + + * Makefile.am: + * configure.ac: + build tests + +2011-01-12 18:16:46 +0100 Wim Taymans + + * tests/.gitignore: + * tests/Makefile.am: + * tests/test-cleanup.c: + tests: add tests directory and cleanup test + +2011-01-12 18:14:48 +0100 Wim Taymans + + * 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-session-pool.c: + * gst/rtsp-server/rtsp-session.c: + server: improve debugging in various objects + +2011-01-12 16:38:34 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + server: chain up to the parent finalize + +2010-09-21 17:04:02 -0300 André Dieb Martins + + * bindings/python/rtspserver-types.defs: + * bindings/python/rtspserver.defs: + * bindings/python/rtspserver.override: + * bindings/python/test.py: + gst-rtsp-server: update python bindings + +2011-01-12 15:37:39 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: use the response from the clientstate + Create the response object only once and store in the client state. + Make all methods use the state response, + +2011-01-12 15:36:22 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + server: use signal to keep track of clients + Keep track of all the clients that the server creates and remove them when they + fire the 'closed' signal. + +2011-01-12 15:35:51 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: emit signal when closing + +2011-01-12 13:57:09 +0100 Wim Taymans + + * examples/.gitignore: + * examples/Makefile.am: + * examples/test-auth.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-media-factory.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-session-pool.h: + * gst/rtsp-server/rtsp-session.h: + media: enable per factory authorisations + Allow for adding a GstRTSPAuth on the factory and media level and check + permissions when accessing the factory. + Add hints to the auth methods for future more fine grained authorisation. + Add example application for per factory authentication. + +2011-01-12 13:16:08 +0100 Wim Taymans + + * 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-params.c: + * gst/rtsp-server/rtsp-params.h: + rtsp-server: Pass ClientState structure arround + Pass the collected information for the ongoing request in a GstRTSPClientState + structure that we can then pass around to simplify the method arguments. This + will also be handy when we implement logging functionality. + +2011-01-12 12:07:40 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + media-factory: add methods to configure authorisation + +2011-01-12 12:07:20 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: unref auth in finalize + +2011-01-12 12:07:04 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + server: unref auth in finalize + +2011-01-12 11:07:26 +0100 Wim Taymans + + * docs/libs/gst-rtsp-server-docs.sgml: + * docs/libs/gst-rtsp-server-sections.txt: + * docs/libs/gst-rtsp-server.types: + docs: add more docs + +2011-01-12 10:57:08 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + server: separate create and accept + Create separate create and accept methods so that subclasses can create custom + client object. + Configure the server in the client object and prepare for keeping track of + connected clients. + +2011-01-12 10:42:52 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + client: add support for setting the server. + Add support for keeping a ref to the server that started this client + connection. + +2011-01-12 10:41:42 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-auth.c: + auth: fix memleak and add some docs + Fix a memleak of the basic auth token. + Add docs for the helper function + +2011-01-12 00:35:28 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-auth.c: + * gst/rtsp-server/rtsp-auth.h: + * gst/rtsp-server/rtsp-client.c: + client: delegate setup of auth to the manager + Delegate the configuration of the authentication tokens to the manager object + when configured. + +2011-01-12 00:17:54 +0100 Wim Taymans + + * examples/test-video.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-server.c: + * gst/rtsp-server/rtsp-server.h: + auth: add authentication object + Add an object that can check the authorization of requests. + Implement basic authentication. + Add example authentication to test-video + +2011-01-12 00:20:36 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + server: move includes back + the includes are needed for sockaddr_in. + +2011-01-11 22:41:12 +0100 Wim Taymans + + * 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: move network includes where they are needed + +2011-01-07 23:45:32 +0200 Sreerenj Balachandran + + * gst/rtsp-server/rtsp-media.h: + rtsp-media.h: Minor corrections in comments. + Fixes #638944 + +2011-01-11 15:52:44 +0200 Stefan Kost + + * common: + Automatic update of common submodule + From e572c87 to f94d739 + +2011-01-11 13:01:44 +0100 Edward Hervey + + * .gitignore: + * docs/.gitignore: + * docs/libs/.gitignore: + * examples/.gitignore: + * gst/rtsp-server/.gitignore: + gitignore: updates + +2011-01-11 12:58:39 +0100 Edward Hervey + + * docs/libs/Makefile.am: + docs: We don't build ps/pdf for API reference docs + +2011-01-10 16:39:36 +0000 Tim-Philipp Müller + + * common: + Automatic update of common submodule + From ccbaa85 to e572c87 + +2011-01-10 14:56:39 +0000 Tim-Philipp Müller + + * common: + Automatic update of common submodule + From 46445ad to ccbaa85 + +2011-01-10 15:10:53 +0100 Wim Taymans + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/fs-funnel.c: + * gst/rtsp-server/fs-funnel.h: + * gst/rtsp-server/rtsp-funnel.c: + * gst/rtsp-server/rtsp-funnel.h: + * gst/rtsp-server/rtsp-media.c: + funnel: rename fsfunnel to rtspfunnel + Rename the funnel to avoid conflicts with the farsight one. + +2011-01-10 13:41:43 +0100 Wim Taymans + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/fs-funnel.c: + * gst/rtsp-server/fs-funnel.h: + * gst/rtsp-server/rtsp-media.c: + rtsp-media: add and use fsfunnel + Add a copy of fsfunnel to the build because input-selector removed the (broken) + select-all property that we need. + +2011-01-08 01:58:44 +0000 Tim-Philipp Müller + + * gst/rtsp-server/Makefile.am: + gobject-introspection: use PKG_CONFIG_PATH specified at configure time + Use PKG_CONFIG_PATH specified at configure time (if any) as well + for the g-ir-compiler, rather than just assuming the env var has + been set. + +2011-01-08 01:55:06 +0000 Tim-Philipp Müller + + * .gitignore: + * Makefile.am: + * configure.ac: + * m4/Makefile.am: + * m4/codeset.m4: + build: make autotools put all .m4 cruft into m4/ rather than polluting common/m4 + +2011-01-08 01:15:35 +0000 Tim-Philipp Müller + + * configure.ac: + * gst/rtsp-server/Makefile.am: + gobject-introspection: fix g-i build for uninstalled setup + Requires gst-plugins-base git (> 0.10.31.2). + +2011-01-07 11:27:57 +0100 Wim Taymans + + * examples/test-uri.c: + examples: add some more options and comments + +2011-01-07 11:24:39 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory-uri.c: + factory-uri: use right property type + +2011-01-05 12:07:42 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory-uri.c: + factory-uri: attempt to configure buffer-lists + Attempt to configure buffer lists in the payloader for improved performance. + +2011-01-05 12:06:23 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: attempt to configure bigger UDP buffers + Attempt to configure bigger udp kernel send buffers to avoid overflowing the + send buffers with high bitrate streams. + +2011-01-05 11:26:30 +0100 Jonas Larsson + + * gst/rtsp-server/rtsp-client.c: + client: use the socket length from getsockname + Use the length returned by getsockname to perform the getnameinfo call because + the size can depend on the socket type and platform. + Fixes #638723 + +2010-12-30 12:41:53 +0100 Wim Taymans + + * docs/libs/gst-rtsp-server-docs.sgml: + * docs/libs/gst-rtsp-server-sections.txt: + docs: add uri factory to the docs + +2010-12-30 12:41:31 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.h: + docs: improve docs + +2010-12-29 16:26:41 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + rtsp-server: add support for buffer lists + Add support for sending bufferlists received from appsink. + Fixes #635832 + +2010-12-28 18:35:01 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-sdp.c: + media: make method to retrieve the play range + Make a method to retrieve the playback range so that we can conditionally create + a different range for the SDP and the PLAY requests. + +2010-12-28 18:34:10 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: add signal to notify of state changes + +2010-12-28 18:31:26 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.h: + client: cleanup headers + +2010-12-28 12:18:41 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: fix typo + +2010-12-23 18:53:01 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-media-factory-uri.h: + factory-uri: add support for gstpay + Add an option to prefer gstpay over decoder + raw payloader. + +2010-12-23 15:58:14 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-media-factory-uri.h: + factory-uri: rework the autoplugger. + Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers + before payloaders. + +2010-12-21 17:37:26 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory-uri.c: + factory-uri: use better factory filter + Make better payloader filter based on autoplug rank and RTP use case. + +2010-12-20 17:48:41 +0100 Edward Hervey + + * common: + Automatic update of common submodule + From 169462a to 46445ad + +2010-12-18 11:24:48 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + server: set SO_REUSEADDR before bind + Set the SO_REUSEADDR _before_ bind() to make it actually work. + +2010-12-13 16:58:36 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: emit prepared signal when prepared + Make a 'prepared' signal and emit it when we successfully prepared the element. + This signal can be used to configure the media object after it has been prepared + for streaming. + +2010-12-15 14:58:00 +0200 Stefan Kost + + * common: + Automatic update of common submodule + From 011bcc8 to 169462a + +2010-12-13 16:38:09 +0100 Andy Wingo + + python an optional dependency + * configure.ac: Move up valgrind and g-i checks. Make the python + dependency optional, as it was before. + +2010-12-13 11:43:13 +0100 Wim Taymans + + Merge branch 'master' into 0.11 + Conflicts: + common + configure.ac + +2010-12-12 15:48:47 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: update range when active clients changed + When we changed the number of active clients, update the current range + information because we want the second client connecting to a shared resource + continue from where the stream currently. + +2010-12-12 04:06:41 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-media-factory-uri.h: + factory-uri: add colorspace and fix pt + Rework the way we pass data to the autoplugger. + When we have raw caps, plug a converter element to make pluggin to raw + payloaders more successful. + Make sure all dynamically plugged payloaders have a unique payload types. + +2010-12-11 18:06:26 +0100 Wim Taymans + + * examples/Makefile.am: + * examples/test-uri.c: + example: add example of the uri factory + +2010-12-11 18:01:53 +0100 Wim Taymans + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/rtsp-media-factory-uri.c: + * gst/rtsp-server/rtsp-media-factory-uri.h: + * gst/rtsp-server/rtsp-server.h: + factory-uri: add a factory to stream any URI + Make a factory that uses uridecodebin to decode any uri and autoplug a payloader + when we have one. + +2010-12-11 17:31:44 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: ignore spurious ASYNC_DONE messages + When we are dynamically adding pads, the addition of the udpsrc elements will + trigger an ASYNC_DONE. We have to ignore this because we only want to react to + the real ASYNC_DONE when everything is prerolled. + +2010-12-11 13:41:24 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + media-factory: make lock macro + +2010-12-11 10:53:28 +0100 Edward Hervey + + * gst/rtsp-server/rtsp-client.c: + rtsp-server: Remove unused variable and dead assignment + +2010-12-11 10:49:30 +0100 Edward Hervey + + * examples/test-launch.c: + * examples/test-mp4.c: + * examples/test-ogg.c: + * examples/test-readme.c: + * examples/test-sdp.c: + * examples/test-video.c: + examples: Run gst-indent + +2010-12-11 10:48:42 +0100 Edward Hervey + + * gst/rtsp-server/rtsp-client.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-params.c: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session.c: + rtsp-server: Run gst-indent + Since it wasn't using the upstream common previously, there was no + indentation check before commiting. + +2010-12-11 10:48:25 +0100 Edward Hervey + + * gst/rtsp-server/rtsp-media-mapping.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-session-pool.h: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + rtsp-server: Some more doc fixups + +2010-12-07 18:56:03 +0100 Edward Hervey + + * Makefile.am: + Makefile: Add cruft-cleaning support + +2010-12-07 18:52:15 +0100 Edward Hervey + + * Makefile.am: + * configure.ac: + * docs/Makefile.am: + * docs/libs/Makefile.am: + * docs/libs/gst-rtsp-server-docs.sgml: + * docs/libs/gst-rtsp-server-sections.txt: + * docs/libs/gst-rtsp-server.types: + * docs/version.entities.in: + docs: Add gtk-doc build system + +2010-12-07 18:14:39 +0100 Edward Hervey + + * gst/rtsp-server/Makefile.am: + Makefile.am: Use standard GIR make behaviour + +2010-12-07 18:14:22 +0100 Edward Hervey + + * autogen.sh: + * configure.ac: + autogen/configure: Bring more in sync to standard gst module behaviour + +2010-12-06 19:29:53 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: warn and fail when gstrtpbin is not found + +2010-12-06 12:40:30 +0100 Wim Taymans + + * configure.ac: + configure: open 0.11 branch + +2010-12-01 20:00:22 +0100 Edward Hervey + + * .gitmodules: + * common: + Add common submodule + +2010-12-01 19:58:49 +0100 Edward Hervey + + * common/ChangeLog: + * common/Makefile.am: + * common/c-to-xml.py: + * common/check.mak: + * common/coverage/coverage-report-entry.pl: + * common/coverage/coverage-report.pl: + * common/coverage/coverage-report.xsl: + * common/coverage/lcov.mak: + * common/gettext.patch: + * common/glib-gen.mak: + * common/gst-autogen.sh: + * common/gst-xmlinspect.py: + * common/gst.supp: + * common/gstdoc-scangobj: + * common/gtk-doc-plugins.mak: + * common/gtk-doc.mak: + * common/m4/.gitignore: + * common/m4/Makefile.am: + * common/m4/README: + * common/m4/as-ac-expand.m4: + * common/m4/as-auto-alt.m4: + * common/m4/as-compiler-flag.m4: + * common/m4/as-compiler.m4: + * common/m4/as-docbook.m4: + * common/m4/as-libtool-tags.m4: + * common/m4/as-libtool.m4: + * common/m4/as-python.m4: + * common/m4/as-scrub-include.m4: + * common/m4/as-version.m4: + * common/m4/ax_create_stdint_h.m4: + * common/m4/check.m4: + * common/m4/glib-gettext.m4: + * common/m4/gst-arch.m4: + * common/m4/gst-args.m4: + * common/m4/gst-check.m4: + * common/m4/gst-debuginfo.m4: + * common/m4/gst-default.m4: + * common/m4/gst-doc.m4: + * common/m4/gst-error.m4: + * common/m4/gst-feature.m4: + * common/m4/gst-function.m4: + * common/m4/gst-gettext.m4: + * common/m4/gst-glib2.m4: + * common/m4/gst-libxml2.m4: + * common/m4/gst-plugindir.m4: + * common/m4/gst-valgrind.m4: + * common/m4/gtk-doc.m4: + * common/m4/introspection.m4: + * common/m4/pkg.m4: + * common/mangle-tmpl.py: + * common/plugins.xsl: + * common/po.mak: + * common/release.mak: + * common/scangobj-merge.py: + * common/upload.mak: + common: Remove static version + +2010-11-08 17:04:00 +0000 Bastien Nocera + + * common/m4/introspection.m4: + Update introspection.m4 to match usage + +2010-10-30 13:26:12 +0200 Wim Taymans + + * README: + README: update + Remove old stuff from the README + +2010-10-11 11:12:11 +0200 Wim Taymans + + * configure.ac: + back to development + +=== release 0.10.7 === + +2010-10-11 11:05:40 +0200 Wim Taymans + + * configure.ac: + release 0.10.7 + +2010-10-04 17:16:40 +0200 Wim Taymans + + * examples/test-ogg.c: + test-ogg: remove parsers + Remove the parsers, they are not needed anymore as oggdemux now outputs normal + buffers with timestamps. Using the parsers also seems to break things. + +2010-09-23 12:44:18 +0200 Sebastian Pölsterl + + * bindings/vala/gst-rtsp-server-0.10.vapi: + * bindings/vala/packages/gst-rtsp-server-0.10.metadata: + Updated Vala bindings + +2010-09-22 23:13:37 +0200 Sebastian Pölsterl + + * common/m4/introspection.m4: + * configure.ac: + * gst/rtsp-server/Makefile.am: + Added initial gobject-introspection support + +2010-09-23 11:32:58 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + media-factory: don't use host for shared hash key + When we generate the key to share made between connections, don't include the + host used to connect so that we can share media even if between clients that + connected with localhost and ones with the ip address. + +2010-09-22 21:16:03 +0100 Tim-Philipp Müller + + * bindings/vala/Makefile.am: + build: fix distcheck + +2010-09-22 18:24:12 +0200 Sebastian Dröge + + * bindings/vala/gst-rtsp-server-0.10.vapi: + * bindings/vala/packages/gst-rtsp-server-0.10.gi: + * bindings/vala/packages/gst-rtsp-server-0.10.metadata: + Update Vala bindings + +2010-09-22 18:12:50 +0200 Sebastian Dröge + + * bindings/vala/Makefile.am: + * configure.ac: + Fix configure checks and installation location for Vala bindings + Fixes bug #628676. + +2010-09-22 16:32:30 +0200 Wim Taymans + + * configure.ac: + back to development + +=== release 0.10.6 === + +2010-09-22 16:22:49 +0200 Wim Taymans + + * configure.ac: + configure: release 0.10.6 + +2010-09-22 16:15:56 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: help the compiler a little + +2010-08-24 16:47:30 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-session.c: + media: cleanup media transport before freeing + Cleanup the media transport data before freeing. In particular, remove the qdata + from the rtpsource object. + +2010-08-20 18:17:08 +0200 Wim Taymans + + * 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-factory: add eos-shutdown property + Add an eos-shutdown property that will send an EOS to the pipeline before + shutting it down. This allows for nice cleanup in case of a muxer. + Fixes #625597 + +2010-08-20 15:58:39 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: use multiudpsink send-duplicates when we can + If we have a new enough multiudpsink with the send-duplicates property, use this + instead of doing our own filtering. Our custom filtering code should eventually + be removed when we can depend on a released -good. + +2010-08-20 13:19:56 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: don't leak destinations + Refactor and cleanup the destinations array when the stream is destroyed. + +2010-08-20 13:09:12 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: don't add udp addresses multiple times + Keep track of the udp addresses we added to udpsink and never add the same udp + destination twice. This avoids duplicate packets when using multicast. + +2010-08-20 10:18:34 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + server: disable use of SO_LINGER + SO_LINGER cause the client to fail to receive a TEARDOWN message because the + server close()s the connection. + +2010-08-19 18:52:47 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + server: use 5 second linger period in SO_LINGER + Wait 5 seconds before clearing the send buffers and reseting the connection with + the client when we do a close. This should be enough time to get the message to + the client. + See #622757 + +2010-08-16 12:32:28 +0200 Robert Krakora + + * gst/rtsp-server/rtsp-server.c: + server: use SO_LINGER + SO_LINGER on the socket will make sure that any pending data on the socket is + flushed ASAP and that the socket connection is reset. This makes sure that the + socket can be reused immediately. + Fixes 622757 + +2010-08-16 12:24:50 +0200 Wim Taymans + + * docs/README: + README: add blurb about shared media factories + +2010-08-09 12:56:23 -0700 David Schleef + + * gst/rtsp-server/rtsp-media.c: + Add stdlib.h for atoi() + +2010-05-20 14:33:24 +0100 Tim-Philipp Müller + + * bindings/python/Makefile.am: + * bindings/vala/Makefile.am: + build: distcheck fixes + Fix 'make distcheck', somewhat (it still fails because it tries to + install files into /usr/share/vala/vapi/ irrespective of the + configured prefix). + +2010-05-20 14:09:18 +0100 Tim-Philipp Müller + + * configure.ac: + configure: bump core/base requirements to released version + Makes things less confusing for people. + +2010-04-25 16:35:30 +0100 Tim-Philipp Müller + + * configure.ac: + configure: fail if GStreamer core/base requirements are not met + +2010-04-06 17:08:40 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: improve client cleanups + Make sure the session does not timeout when using TCP. We need to do this + because quicktime player does not send RTCP for some reason in tunneled + mode. + Refactor some cleanup code. + Fixes #612915 + +2010-04-06 17:07:27 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + session: add support for prevent session timeouts + Add an atomix counter to prevent session timeouts when we are, for example, + streaming over TCP. + +2010-04-06 15:45:56 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: fix unlink on session timeouts + When our session times out, make sure we unlink all streams in this + session. + Remove the tunnelid when closing the connection. + +2010-04-06 15:44:45 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-session.c: + session: small cleanups + +2010-04-06 11:13:51 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: handle lost_tunnel callbacks + Handle lost_tunnel callbacks and use it to store the tunnelid back into the + hashtable so that we can reuse it for when the client reopens the POST + socket. + Close the connection after a TEARDOWN. + Make sure or watchid is cleared when the watch is removed. + Fixes #612915 + +2010-03-19 18:03:40 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-sdp.c: + rtsp-server: add more support for multicast + +2010-03-19 15:15:29 +0100 Wim Taymans + + * configure.ac: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: allow configuration of allowed lower transport + +2010-03-16 18:37:18 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-sdp.h: + * gst/rtsp-server/rtsp-server.c: + rtsp: keep track of server ip and ipv6 + Keep track of how the client connected to the server and setup the udp ports + with the same protocol. + Copy the server ip address in the SDP so that clients can send RTCP back to + us. + +2010-03-16 18:34:43 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-session.c: + session: indent + +2010-03-16 18:33:23 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: use right size for malloc + +2010-03-10 11:45:30 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + server: comment ipv6 server listening address + +2010-03-10 11:45:06 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: allow for ipv6 sockets + +2010-03-09 13:49:00 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + server: rework server part + Allow setting a bind address, make sure we can deal with ipv6. + Remove the port property and change with the service property. + +2010-03-09 13:44:20 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.h: + media: update comments a little + +2010-03-09 13:43:29 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: make content-base better + Use the URI formatting functions to make a content-base. Also make sure that + there is a trailing / at the end. + +2010-03-09 13:42:50 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: guard against invalid paths + +2010-03-09 13:41:33 +0100 Wim Taymans + + * examples/test-video.c: + test: catch server bind errors + +2010-03-09 10:27:38 +0100 Alessandro Decina + + * gst/rtsp-server/rtsp-media.c: + rtspmedia: emit "unprepared" if _prepare fails. + Emit the unprepared signal if gst_rtsp_media_prepare fails so that the + media object is removed from its factory's cache. + +2010-03-05 19:08:08 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: collect media position when seek completes + +2010-03-05 18:37:17 +0100 Luca Ognibene + + * gst/rtsp-server/rtsp-client.c: + client: call unlink_streams in client finalize + Fixes #599027 + +2010-03-05 18:23:18 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: limit the time to wait to something huge + Avoid waiting forever but limit the timeout to 20 seconds. + +2010-03-05 17:57:08 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-sdp.c: + sdp: reindent and check for prepared status + +2010-03-05 17:51:26 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-session.c: + media: avoid doing _get_state() for state changes + When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait + until the media is prerolled or in error. This avoids doing a blocking call of + gst_element_get_state() that can cause lockups when there is an error. + Fixes #611899 + +2010-03-05 16:20:08 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: reindent + +2010-03-05 13:34:15 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + media-factory: better error handling + Improve the error handling a bit. + +2010-03-05 13:31:37 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: rework transport parsing + Rework the transport parsing code so that we can ignore transports we don't + support instead of just picking the first one we can parse. + Configure a (for now hardcoded) destination for multicast transports. + +2010-03-05 13:28:58 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: set multicast sink parameters + Disable loop and automatic multicast join on the udpsink elements. + Add some more debug info. + Reset some state variables in the right place. + Use the right port numbers for multicast. + +2010-03-05 13:27:18 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-session.c: + session: handle transport setup correctly + Handle UDP, MCAST and TCP transport negotiation more correctly. + Store the server session SSRC in the transport. + +2010-01-27 18:38:27 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + rtsp-client: implement error_full + Implement error_full to avoid some segfaults when the rtspconnection calls it. + See #608245 + +2009-12-25 18:24:10 +0100 Wim Taymans + + * docs/README: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-server.c: + docs: update docs and comments + +2009-12-25 15:22:23 +0100 Nikolay Ivanov + + * gst/rtsp-server/rtsp-sdp.c: + sdp: make server work better when behind a proxy + +2009-11-21 01:17:25 +0100 Sebastian Pölsterl + + * gst/rtsp-server/rtsp-client.c: + client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG + +2009-11-21 19:20:23 +0100 Sebastian Pölsterl + + * gst/rtsp-server/rtsp-client.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: + Use GStreamer's debugging subsystem + +2009-11-21 01:00:39 +0100 Sebastian Pölsterl + + * gst/rtsp-server/rtsp-media-factory.c: + server: Set ghost pad active in gst_rtsp_media_factory_collect_streams + +2009-11-05 11:22:44 +0100 Wim Taymans + + * configure.ac: + back to development + +=== release 0.10.5 === + +2009-11-05 11:20:45 +0100 Wim Taymans + + * configure.ac: + release 0.10.5 + +2009-10-14 12:11:31 +0200 Wim Taymans + + * configure.ac: + configure: bump required versions + +2009-10-11 13:57:54 +0200 Luca Ognibene + + * gst/rtsp-server/rtsp-client.c: + client: call weak-unref on client->sessions from finalize + Fixes bug #596305 + +2009-10-09 23:08:18 +0200 Sebastian Pölsterl + + * gst/rtsp-server/rtsp-media.c: + media: Fixed crasher where caps got unref'ed too often + +2009-10-09 16:26:30 +0200 Sebastian Pölsterl + + * configure.ac: + * pkgconfig/.gitignore: + * pkgconfig/Makefile.am: + * pkgconfig/gst-rtsp-server-uninstalled.pc.in: + Added pkg-config file to use gst-rtsp-server uninstalled + +2009-09-11 13:52:27 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: add some docs + +2009-08-24 13:27:00 +0200 Peter Kjellerstedt + + * gst/rtsp-server/rtsp-client.c: + rtsp: Use gst_rtsp_watch_send_message(). + Use gst_rtsp_watch_send_message() since the old API which used + gst_rtsp_watch_queue_message() has been deprecated. + +2009-08-05 11:53:56 +0200 Wim Taymans + + * configure.ac: + back to development + +=== release 0.10.4 === + +2009-08-05 11:44:49 +0200 Wim Taymans + + * configure.ac: + Release 0.10.4 + +2009-07-27 19:42:44 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + rtsp: allocate channels in TCP mode + When the client does not provide us with channels in TCP mode, allocate channels + ourselves. + +2009-07-24 12:49:41 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: don't crash when tunnelid is missing + When a clients tries to open an HTTP tunnel but fails to provide a tunnelid, + don't crash but return an error response to the client. + Fixes #589489 + +2009-07-13 11:31:23 +0200 Sebastian Pölsterl + + * bindings/vala/gst-rtsp-server-0.10.vapi: + * bindings/vala/packages/gst-rtsp-server-0.10.gi: + * bindings/vala/packages/gst-rtsp-server-0.10.metadata: + bindings: update vala bindings with new method + +2009-06-30 21:27:53 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session-pool.h: + sessionpool: add function to filter sessions + Add generic function to retrieve/remove sessions. + +2009-06-22 18:57:25 +0100 Tim-Philipp Müller + + * configure.ac: + configure: bump core/base requirements to release + +2009-06-18 16:05:18 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: fix indentation + +2009-06-14 23:12:13 +0200 Sebastian Pölsterl + + * gst/rtsp-server/rtsp-media.c: + Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often. + +2009-06-13 16:05:02 +0200 Sebastian Pölsterl + + * gst/rtsp-server/rtsp-media.c: + set state and remove elements of media in for loop + +2009-06-13 14:38:39 +0200 Sebastian + + * bindings/vala/gst-rtsp-server-0.10.vapi: + * bindings/vala/packages/gst-rtsp-server-0.10.gi: + Added gst_rtsp_media_remove_elements function to Vala bindings + +2009-06-13 14:38:20 +0200 Sebastian + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + Added gst_rtsp_media_remove_elements function + +2009-06-12 22:22:40 +0200 Sebastian + + * gst/rtsp-server/rtsp-media.c: + Don't use name for gstrtpbin so we can add multiple instances to the pipeline + +2009-06-12 19:28:04 +0200 Sebastian Pölsterl + + * bindings/vala/gst-rtsp-server-0.10.vapi: + * bindings/vala/packages/gst-rtsp-server-0.10.gi: + * bindings/vala/packages/gst-rtsp-server-0.10.metadata: + Updated Vala bindings + +2009-06-12 18:05:30 +0200 Sebastian Pölsterl + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + Added vmethod unprepare to GstRTSPMedia + The default implementation sets the state of the pipeline to GST_STATE_NULL + +2009-06-12 17:51:44 +0200 Sebastian Pölsterl + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + Made collect_streams function public + +2009-06-12 17:45:29 +0200 Sebastian Pölsterl + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + Added vmethod create_pipeline to GstRTSPMediaFactory + The pipeline is created in this method and the GstRTSPMedia's element is added to it + +2009-06-11 11:27:47 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: use g_source_destroy() + We need to use g_source_destroy() because we might have added the source to a + different main context than the default one. + +2009-06-10 00:01:07 +0200 Wim Taymans + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-params.c: + * gst/rtsp-server/rtsp-params.h: + rtsp: prepare for handling GET/SET_PARAMETER + Add helper functions to handle GET/SET_PARAMETER. Reply with an error when there + is a body now. + Fix return codes of handlers. + +2009-06-04 19:20:26 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: don't leak session pads + +2009-06-04 18:32:15 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: clean up the messages a bit + +2009-06-03 12:13:21 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-sdp.c: + sdp: warn and skip streams without media + +2009-05-30 14:38:34 +0200 Sebastian Pölsterl + + * bindings/vala/gst-rtsp-server-0.10.vapi: + * bindings/vala/packages/gst-rtsp-server-0.10.metadata: + vala: Fixed typo in header file of RTSPMediaStream + +2009-05-27 11:15:22 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: fix message + Fix a debug message + Make dumping RTCP stats configurable + +2009-05-26 19:20:07 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: be less verbose and leak less + +2009-05-26 19:05:07 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: don't leak the destination address + +2009-05-26 19:01:10 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + rtsp: use RTCP to keep the session alive + Use the RTCP rtcp-from stats field to find the associated session and use this + to keep the session alive. + +2009-05-26 17:27:07 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-session.c: + session: add 5sec to the real session timeout + Allow the session to live 5sec longer before really timing out. This should give + clients some extra time to keep the session active. + +2009-05-26 17:25:59 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: replay OK to GET/SET_PARAMETER + Some clients (vlc) use GET/SET_PARAMETER to keep the TCP session open. Make it + so that we return OK for those requests. + +2009-05-26 11:42:41 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: keep track of active transports + Keep track of which transport is active to avoid closing the connection too + soon. + Remove the destination transport also when going to NULL. + Print some stats about the SDES and other RTCP messages we receive from the + clients. + +2009-05-24 20:00:19 +0200 Wim Taymans + + * examples/.gitignore: + * examples/Makefile.am: + * examples/test-sdp.c: + example: add SDP relay example + +2009-05-24 19:56:45 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: also count active TCP connections + +2009-05-24 19:34:52 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + rtsp: add support for dynamic elements + Add support for dynamic elements. + Don't set live pipelines back to paused. + +2009-05-24 19:33:22 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-sdp.c: + sdp: don't add encoding name when absent in caps + +2009-05-23 16:30:55 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: warn when we can't do RTP-Info + +2009-05-23 16:18:04 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + factory: factor out the stream construction + +2009-05-23 16:17:02 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: only add RTP-Info when we have the info + Only add RTP-Info for a stream when we can get the seqnum and timestamp from the + depayloader. + +2009-05-17 14:04:31 +0200 Wim Taymans + + * configure.ac: + back to development + +=== release 0.10.3 === + +2009-05-17 13:59:10 +0200 Wim Taymans + + * configure.ac: + release: 0.10.3 + - Fixes a bug where it put the wrong verion in pkgconfig + - Link RTP and RTCP sources + +2009-05-15 17:58:44 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: link the RTP udpsrc to the session manager + Link the RTP udpsrc and the appsrc to the session manager so that they don't + shut down when the client sends a packet to open firewalls. + +2009-05-15 17:10:44 +0200 Sebastian Pölsterl + + * pkgconfig/gst-rtsp-server.pc.in: + Don't use hard-coded version number in pkg-config file + +2009-05-11 10:51:47 +0200 Wim Taymans + + * configure.ac: + back to development + +=== release 0.10.2 === + +2009-05-11 10:50:31 +0200 Wim Taymans + + * configure.ac: + release 0.10.2 + +2009-05-11 10:38:44 +0200 Wim Taymans + + * .gitignore: + * common/m4/.gitignore: + * examples/.gitignore: + * pkgconfig/.gitignore: + add some .gitignore files + +2009-04-29 17:24:46 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: seek to key frames + +2009-04-21 22:44:05 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + media: emit the unprepared signal by id + Emit the unprepared signal by id instead of name and set the media as + reused. + +2009-04-21 22:23:54 +0200 Sebastian Pölsterl + + * gst/rtsp-server/rtsp-media.c: + Set pipeline's state to NULL no matter if the media is reusable and emit unprepared signal in gst_rtsp_media_unprepare + +2009-04-18 16:10:59 +0200 Sebastian Pölsterl + + * gst/rtsp-server/rtsp-server.c: + Added finalize function to GstRTPSPServer to unref session pool and media mapping + +2009-04-17 21:13:07 +0200 Sebastian Pölsterl + + * bindings/vala/gst-rtsp-server-0.10.vapi: + * bindings/vala/packages/gst-rtsp-server-0.10.gi: + * bindings/vala/packages/gst-rtsp-server-0.10.metadata: + Updated vala bindings + +2009-04-14 23:38:58 +0200 Wim Taymans + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + server: use appsink and appsrc with the API + Use the appsink/appsrc API instead of the signals for higher + performance. + +2009-04-14 23:38:15 +0200 Wim Taymans + + * examples/test-ogg.c: + tests: set the payload type correctly + +2009-04-03 22:46:22 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + factory: connect to the unprepare signal + Connect to the unprepare signal for non-reusable media so that we can remove + them from the cache. + +2009-04-03 22:45:57 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: add signal to notify of unprepare + +2009-04-03 22:22:30 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + media: more work on making the media shared + Add a reusable flag to medias, indicating that they can be reused after a state + change to NULL. + Small cleanups. + +2009-04-03 19:47:38 +0200 Wim Taymans + + * examples/test-readme.c: + examples: mark the example as shared for testing + +2009-04-03 19:44:37 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + client: support shared media + Always perform the state actions even if the target state of the pipeline is + already correct, we still want to add/remove the transports when we are dealing + with shared media. + Keep a counter of the number of active transports for a media so that we can use + this to perform a state change when needed. + Perform a state change of the pipeline only when the first transport was added + or when there are no active transports. + +2009-04-03 09:03:59 +0200 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + client: fix refcounting crasher + Don't need to remove the weak refs in the finalize methods, they are already + removed in the dispose. + Don't register the callback with a DestroyNofity. + +2009-04-01 01:01:46 +0100 Tim-Philipp Müller + + * gst/rtsp-server/rtsp-client.c: + Fix rtsp client refcount management in TCP mode. + Don't unref a client ref we never had. Fixes an unref + of an already-free client object after a client + teardown request for me. + +2009-04-01 00:45:17 +0100 Tim-Philipp Müller + + * gst/rtsp-server/rtsp-session.c: + docs: fix typo in API docs + +2009-03-13 15:57:42 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + More seeking fixes. + Keep the udp sources in playing even if we go to paused. unlock the sources when + we shut down. + Add some more debug info. + Only seek when we need to. + Keep track of the position when we go to paused. + +2009-03-12 20:32:14 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + Add beginnings of seeking. + Parse the Range header and perform a seek on the pipeline for the requested + position. It's disabled currently until I figure out what's going wrong. + +2009-03-12 20:31:22 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + allow pause requests for now. + -- + +2009-03-11 20:03:06 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + Remove weak ref on the session in teardown + We need to remove our weakref from the session when we do a teardown because + else we close the TCP connection prematurely. + +2009-03-11 19:38:06 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-session-pool.c: + Do some more session cleanup + Make session timeout kill the TCP connection that currently watches the + session. + Remove the client timeout property. + +2009-03-11 16:45:12 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + Add TCP transports + Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP + connection. + +2009-03-11 16:39:20 +0100 Wim Taymans + + * examples/Makefile.am: + * examples/test-launch.c: + Add example server that takes launch lines + Add an example server that streams any -launch line. + +2009-03-06 19:34:14 +0100 Wim Taymans + + * examples/test-readme.c: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + Add support for live streams + Add support for live streams and ranges + Start on handling TCP data transfer. + +2009-03-04 16:33:59 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + Free the pipeline before other things + --- + +2009-03-04 16:33:21 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + Only free the pending tunnel if there is one + -- + +2009-03-04 12:44:01 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + * gst/rtsp-server/rtsp-media.c: + rtsp-server: Add support for tunneling + Add support for tunneling over HTTP. + Use new connection methods to retrieve the url. + Dispatch messages based on the message type instead of blindly + assuming it's always a request. + Keep track of the watch id so that we can remove it later. + Set the media pipeline to NULL before unreffing the pipeline. + +2009-02-19 15:53:50 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + Fix for channel -> watch rename in gstreamer + Rename the RTSPChannel to RTSPWatch and remove an unused variable. + +2009-02-18 18:57:31 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + Use ASYNC RTSP io + Use the async RTSP channels instead of spawning a new thread for each client. + If a sessionid is specified in a request, fail if we don't have the session. + +2009-02-18 17:49:03 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + Add better debug info + Add some better debug info. + +2009-02-13 20:00:34 +0100 Wim Taymans + + * examples/test-video.c: + Time out sessions + Add support for session timeouts in the example. + +2009-02-13 19:58:17 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session-pool.h: + Pass GTimeVal around for performance reasons + Get the current time only once and pass it around so that sessions don't have to + get the current time anymore. + Add experimental support for a GSource that dispatches when the session needs to + be cleaned up. + +2009-02-13 19:56:01 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + Add better support for session timeouts + Add a method to request the number of milliseconds when a session will timeout. + +2009-02-13 19:54:18 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + Add suport for RTP manager monitoring + Add the first stage in monitoring the rtp manager. + Make sure we don't update the state to something we don't want. + +2009-02-13 19:52:05 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + Add support for session keepalive + Get and update the session timeout for all requests. get the session as early as + possible. + +2009-02-13 16:39:36 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.h: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + Handle media bus messages + Handle media bus messages in a custom mainloop and dispatch them to the + RTSPMedia objects. Let the default implementation handle some common messages. + +2009-02-13 12:57:45 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session.c: + Some more session timeout handling + Move the session header setting code to a central place so that we always add + the timeout parameter too. + Handle timeouts by running the session cleanup code. + Stop media before cleaning up. + +2009-02-10 16:24:13 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + Add timeout property + Add a timeout property ot the client and make the other properties into GObject + properties. + +2009-02-10 16:21:17 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-session-pool.c: + Use getters and setters in property code + Use the getters and setters for the timeout property instead of locking + ourselves. + +2009-02-04 20:13:32 +0100 Wim Taymans + + Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server + +2009-02-04 20:10:39 +0100 Wim Taymans + + * 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: + Add more timeout stuff + Add method to check if a session is expired. + Add method to perform cleanup on a session pool. + +2009-02-04 19:52:50 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * 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: + Add beginnings of session timeouts and limits + Add the timeout value to the Session header for unusual timeout values. + Allow us to configure a limit to the amount of active sessions in a pool. Set a + limit on the amount of retry we do after a sessionid collision. + Add properties to the sessionid and the timeout of a session. Keep track of + creation time and last access time for sessions. + +2009-02-04 17:00:42 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-sdp.c: + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + Cleanup of sessions and more + Fix the refcounting of media and sessions in the client. Properly clean up the + session data when the client performs a teardown. + Add Server header to responses. + Allow for multiple uri setups in one session. + Add Range header to the PLAY response and add the range attribute to the SDP + message. + Fix the session pool remove method, it used the wrong key in the hashtable. Also + give the ownership of the sessionid to the session object. + +2009-02-04 09:57:55 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + Rename a variable + Rename the 'server_port' variable to simply 'port'. + +2009-02-03 19:32:38 +0100 Wim Taymans + + * configure.ac: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + Rework the way we handle transports for streams + Make the media accept an array of transports for the streams that we have + configured for the play/pause requests. + Implement server states for a client and its media. + Require 0.10.22.1 (git HEAD) of gstreamer. + +2009-01-31 19:50:33 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media-factory.c: + Drop const from functions dealing with urls + Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't + have the right const in them. + +2009-01-30 17:06:26 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-sdp.c: + Fix various leaks + Fix some leaks. + +2009-01-30 16:24:10 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + More cleanups + Don't keep a reference to the GstRTSPMedia in the stream. + Free more things when freeing the GstRTSPMedia. + +2009-01-30 14:53:28 +0100 Wim Taymans + + * docs/README: + * 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: + More docs and small cleanups + Add some more docs and update the README + Cleanup some method names. + Remove an unneeded idx field in the GstRTSPMediaStream + +2009-01-30 13:24:04 +0100 Wim Taymans + + * docs/README: + * examples/Makefile.am: + * examples/test-readme.c: + Add a README and more example code + Add a README file that contains a small introduction on how to use the server + along with the example code explained in the readme. + +2009-01-30 11:06:31 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-server.c: + Fix some leaks and change default port + Fix some memory leaks by setting the udpsrc elements to the unlocked state after + we finished the initial preroll. If we keep them locked, setting the pipeline to + NULL will not stop and clean up the sources correctly. + Change the default RTSP port to 8554 aka the official alternative RTSP port. + +2009-01-29 18:55:22 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + Cleanups to the session object + Remove some unneeded variables in the session state of a stream such as the + owner media and the server transport. + Get the configuration of a media stream in a session based on the media_stream + in the original object instead of our cached index. + Free more data in the finalize method. + +2009-01-29 18:51:02 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + Cleanups and reuse media from DESCRIBE + Handle thread create errors. + Rename some internal methods to better match what they actually do. + Handle misconfiguration of session_pool and media_mapping gracefully. + Cache the DESCRIBE media and uri in the client connection and reuse them when + we receive a SETUP request in the same connection for the same uri. + Cleanup the client connection object. + +2009-01-29 17:20:27 +0100 Wim Taymans + + * 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: + Add shared properties to media and factory + Add the shared property to media. + Implement some simple caching in the factory depending on if the media is shared + or not. + +2009-01-29 17:19:21 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + Add a little comment + Add some comment about the content-base header. + +2009-01-29 13:31:27 +0100 Wim Taymans + + * examples/Makefile.am: + * examples/main.c: + * examples/test-mp4.c: + * examples/test-ogg.c: + * examples/test-video.c: + * gst/rtsp-server/Makefile.am: + * 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: + * 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.c: + * gst/rtsp-server/rtsp-session.h: + Reorganize things, prepare for media sharing + Added various other test server examples + Move the SDP message generation to a separate helper. + Refactor common code for finding the session. + Add content-base for realplayer compatibility + Clean up request uris before processing for better vlc compatibility. + Move prerolling and pipeline construction to the RTSPMedia object. + Use multiudpsink for future pipeline reuse. + +2009-01-30 11:23:57 +0100 Wim Taymans + + * configure.ac: + Back to development + Back to 0.10.1.1 + +=== release 0.10.1 === + +2009-01-30 11:20:18 +0100 Wim Taymans + + * configure.ac: + Make 0.10.1 release + Release 0.10.1 + +2009-01-29 15:19:01 +0100 Wim Taymans + + * bindings/vala/Makefile.am: + Fix make dist + Add more directories and files to the dist. + +2009-01-24 14:34:35 +0100 Sebastian Pölsterl + + * bindings/python/Makefile.am: + * bindings/python/rtspserver.override: + Fixed compile error of python bindings + +2009-01-23 21:03:53 +0100 Sebastian Pölsterl + + * bindings/vala/gst-rtsp-server-0.10.vapi: + * bindings/vala/packages/gst-rtsp-server-0.10.metadata: + Marked values as nullable accordingly + +2009-01-23 20:31:11 +0100 Sebastian Pölsterl + + * bindings/vala/gst-rtsp-server-0.10.vapi: + * bindings/vala/packages/gst-rtsp-server-0.10.excludes: + * bindings/vala/packages/gst-rtsp-server-0.10.gi: + * bindings/vala/packages/gst-rtsp-server-0.10.metadata: + Updated Vala bindings + +2009-01-22 18:35:17 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-media-mapping.c: + * gst/rtsp-server/rtsp-media-mapping.h: + * gst/rtsp-server/rtsp-media.h: + * gst/rtsp-server/rtsp-session-pool.h: + Cleanups and doc updates + Add some more documentation and do some minor cleanups here and there. + +2009-01-22 17:58:19 +0100 Wim Taymans + + * 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-session.c: + * gst/rtsp-server/rtsp-session.h: + More improvements + Rename GstRTSPMediaBin to GstRTSPMedia + Parse the request url into a GstRTSPUri object and pass this object to the + various handlers and methods that require the uri. + +2009-01-22 16:54:07 +0100 Wim Taymans + + * examples/main.c: + Update example + Add some more docs and remove some old code from the example. + +2009-01-22 16:53:16 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + Handle state change failures better + Handle state change failures better when changing the state of the pipeline to + determine the SDP. + +2009-01-22 16:51:08 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + Make element creation more extendible + Add get_element vmethod to the default MediaFactory so that subclasses can just + override that method and still use the default logic for making a MediaBin from + that. + +2009-01-22 15:33:29 +0100 Wim Taymans + + * examples/main.c: + * gst/rtsp-server/Makefile.am: + * 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-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-server.c: + * gst/rtsp-server/rtsp-server.h: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + Make the server handle arbitrary pipelines + Make GstMediaFactory an object that can instantiate GstMediaBin objects. + The GstMediaBin object has a handle to a bin with elements and to a list of + GstMediaStream objects that this bin produces. + Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along + with methods to register and remove those mappings. + Add methods and a property to GstRTSPServer to manage the GstMediaMapper object + used by the server instance. + Modify the example application so that it shows how to create custom pipelines + attached to a specific mount point. + Various misc cleanps. + +2009-01-20 19:47:07 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + Allow setting a custom media factory for a server + +2009-01-20 19:46:21 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + Allow setting a custom media factory for a client. + +2009-01-20 19:45:28 +0100 Wim Taymans + + * gst/rtsp-server/Makefile.am: + Add Makefile entry for the media factory + +2009-01-20 19:44:45 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media-factory.c: + * gst/rtsp-server/rtsp-media-factory.h: + Add media factory to map urls to media pipeline objects. + +2009-01-20 19:43:47 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-media.c: + * gst/rtsp-server/rtsp-media.h: + Add comments. Remove unused field + +2009-01-20 19:41:53 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-session-pool.c: + * gst/rtsp-server/rtsp-session-pool.h: + Allow custom session pools to override the session id allocation algorithms Add some comments. + +2009-01-20 19:40:42 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-session.h: + Add some comments. + +2009-01-20 13:57:47 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + Move the connection code in one place Add some comments + +2009-01-20 13:19:36 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + Make vmethod to create and accept new clients. Add some docs. + +2009-01-19 19:36:23 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + * gst/rtsp-server/rtsp-server.h: + Make more properties configurable in the server. Expose the GIOChannel and GSource better to allow for more customisations. + +2009-01-19 19:34:29 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.h: + Name the parameters more appropriately. + +2009-01-19 19:32:28 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-session-pool.c: + Do some more cleanup of the session pool. + +2009-01-08 16:28:24 +0100 Wim Taymans + + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/rtsp-client.c: + Check if return value of gst_rtsp_session_get_media is not NULL + +2009-01-08 15:02:42 +0100 Wim Taymans + + * gst/rtsp-server/Makefile.am: + Install rtsp-session and rtsp-session-pool headers + +2009-01-08 14:57:55 +0100 Wim Taymans + + * .gitignore: + * Makefile.am: + * acinclude.m4: + * bindings/python/Makefile.am: + * bindings/python/arg-types.py: + * bindings/python/codegen/Makefile.am: + * bindings/python/codegen/__init__.py: + * bindings/python/codegen/argtypes.py: + * bindings/python/codegen/code-coverage.py: + * bindings/python/codegen/codegen.py: + * bindings/python/codegen/definitions.py: + * bindings/python/codegen/defsparser.py: + * bindings/python/codegen/docextract.py: + * bindings/python/codegen/docgen.py: + * bindings/python/codegen/fileprefix.override: + * bindings/python/codegen/fileprefixmodule.c: + * bindings/python/codegen/h2def.py: + * bindings/python/codegen/mergedefs.py: + * bindings/python/codegen/mkskel.py: + * bindings/python/codegen/override.py: + * bindings/python/codegen/reversewrapper.py: + * bindings/python/codegen/scmexpr.py: + * bindings/python/rtspserver-types.defs: + * bindings/python/rtspserver.defs: + * bindings/python/rtspserver.override: + * bindings/python/rtspservermodule.c: + * configure.ac: + Add python bindings. + +2009-01-08 14:53:47 +0100 Wim Taymans + + * bindings/Makefile.am: + * configure.ac: + Don't go into python dir when requirements for python bindings are missing + +2009-01-08 14:49:57 +0100 Wim Taymans + + * bindings/Makefile.am: + * bindings/vala/Makefile.am: + * configure.ac: + Install Vala bindings if vala is available + +2008-12-12 16:22:02 +0100 Sebastian Pölsterl + + * bindings/vala/gst-rtsp-server-0.10.deps: + * bindings/vala/gst-rtsp-server-0.10.vapi: + * bindings/vala/gst-rtsp-server.vapi: + * bindings/vala/packages/gst-rtsp-server-0.10.deps: + * bindings/vala/packages/gst-rtsp-server-0.10.excludes: + * 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: + * bindings/vala/packages/gst-rtsp-server.deps: + * bindings/vala/packages/gst-rtsp-server.excludes: + * bindings/vala/packages/gst-rtsp-server.files: + * bindings/vala/packages/gst-rtsp-server.gi: + * bindings/vala/packages/gst-rtsp-server.metadata: + * bindings/vala/packages/gst-rtsp-server.namespace: + Regenerated Vala bindings + +2008-12-08 13:19:40 +0100 Sebastian Pölsterl + + * bindings/vala/gst-rtsp-server.vapi: + * bindings/vala/packages/gst-rtsp-server.metadata: + Fixed typo in included headers for vala bindings + +2009-01-08 14:42:10 +0100 Wim Taymans + + * Makefile.am: + * configure.ac: + * pkgconfig/Makefile.am: + * pkgconfig/gst-rtsp-server.pc.in: + Added pkgconfig file + +2008-11-30 23:57:26 +0100 Sebastian Pölsterl + + * bindings/vala/gst-rtsp-server.vapi: + * bindings/vala/packages/gst-rtsp-server.excludes: + * bindings/vala/packages/gst-rtsp-server.gi: + * bindings/vala/packages/gst-rtsp-server.metadata: + Adjusted included headersfor Vala bindings. Ignore rtsp-url-compat.h + +2008-11-30 23:41:20 +0100 Sebastian Pölsterl + + * bindings/vala/gst-rtsp-server.vapi: + * bindings/vala/packages/gst-rtsp-server.deps: + * bindings/vala/packages/gst-rtsp-server.files: + * bindings/vala/packages/gst-rtsp-server.gi: + * bindings/vala/packages/gst-rtsp-server.metadata: + * bindings/vala/packages/gst-rtsp-server.namespace: + Added Vala bindings + +2008-10-25 23:36:16 +0200 Alessandro Decina + + * gst/rtsp-server/rtsp-session.c: + Change an obviously wrong return FALSE to return NULL; (cherry picked from commit 56d4fb48030db3ae45f3f0e60b29b36f3134322b) + +2008-11-13 19:43:10 +0100 Sebastian Pölsterl + + * examples/Makefile.am: + * gst/rtsp-server/Makefile.am: + Put GStreamer version in library name + +2009-01-08 13:51:26 +0100 Wim Taymans + + * examples/Makefile.am: + * gst/rtsp-server/Makefile.am: + Fix some issues to pass distcheck + +2009-01-08 13:41:33 +0100 Wim Taymans + + * gst/rtsp-server/rtsp-server.c: + Added port property to GstRTSPServer class. + +2009-01-08 13:18:55 +0100 Wim Taymans + + * Makefile.am: + * autogen.sh: + * configure.ac: + * examples/Makefile.am: + * examples/main.c: + * gst/Makefile.am: + * gst/rtsp-server/Makefile.am: + * gst/rtsp-server/rtsp-client.c: + * gst/rtsp-server/rtsp-client.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: + * gst/rtsp-server/rtsp-session.c: + * gst/rtsp-server/rtsp-session.h: + * src/Makefile.am: + * src/main.c: + * src/rtsp-client.c: + * src/rtsp-client.h: + * src/rtsp-media.c: + * src/rtsp-media.h: + * src/rtsp-server.c: + * src/rtsp-server.h: + * src/rtsp-session-pool.c: + * src/rtsp-session-pool.h: + * src/rtsp-session.c: + * src/rtsp-session.h: + Split in library and example program + +2008-11-10 20:59:35 +0100 Sebastian Pölsterl + + * src/rtsp-client.h: + Removed obsolete variable + +2008-11-10 21:03:15 +0100 Sebastian Pölsterl + + * src/rtsp-client.c: + * src/rtsp-client.h: + Removed pipeline variable GstRTSPClient, because it's only used in one function + +2009-01-08 11:22:58 +0100 Wim Taymans + + * src/rtsp-media.c: + Set the payload types for the different payloaders. Maybe this shoulde be done automatically instead. + +2008-10-23 12:23:27 +0200 Wim Taymans + + * src/rtsp-session.c: + Initialize some more vars. + +2008-10-23 12:14:55 +0200 Wim Taymans + + * src/rtsp-session.c: + Initialize variable to avoid compiler warning. + +2008-10-09 13:30:47 +0100 Simon McVittie + + * .gitignore: + Add a reasonable generic .gitignore + diff --git a/INSTALL b/INSTALL new file mode 100644 index 0000000..2099840 --- /dev/null +++ b/INSTALL @@ -0,0 +1,370 @@ +Installation Instructions +************************* + +Copyright (C) 1994-1996, 1999-2002, 2004-2013 Free Software Foundation, +Inc. + + Copying and distribution of this file, with or without modification, +are permitted in any medium without royalty provided the copyright +notice and this notice are preserved. 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If GNU +CC is not installed, it is recommended to use the following options in +order to use an ANSI C compiler: + + ./configure CC="cc -Ae -D_XOPEN_SOURCE=500" + +and if that doesn't work, install pre-built binaries of GCC for HP-UX. + + HP-UX `make' updates targets which have the same time stamps as +their prerequisites, which makes it generally unusable when shipped +generated files such as `configure' are involved. Use GNU `make' +instead. + + On OSF/1 a.k.a. Tru64, some versions of the default C compiler cannot +parse its `' header file. The option `-nodtk' can be used as +a workaround. If GNU CC is not installed, it is therefore recommended +to try + + ./configure CC="cc" + +and if that doesn't work, try + + ./configure CC="cc -nodtk" + + On Solaris, don't put `/usr/ucb' early in your `PATH'. This +directory contains several dysfunctional programs; working variants of +these programs are available in `/usr/bin'. So, if you need `/usr/ucb' +in your `PATH', put it _after_ `/usr/bin'. + + On Haiku, software installed for all users goes in `/boot/common', +not `/usr/local'. It is recommended to use the following options: + + ./configure --prefix=/boot/common + +Specifying the System Type +========================== + + There may be some features `configure' cannot figure out +automatically, but needs to determine by the type of machine the package +will run on. Usually, assuming the package is built to be run on the +_same_ architectures, `configure' can figure that out, but if it prints +a message saying it cannot guess the machine type, give it the +`--build=TYPE' option. TYPE can either be a short name for the system +type, such as `sun4', or a canonical name which has the form: + + CPU-COMPANY-SYSTEM + +where SYSTEM can have one of these forms: + + OS + KERNEL-OS + + See the file `config.sub' for the possible values of each field. If +`config.sub' isn't included in this package, then this package doesn't +need to know the machine type. + + If you are _building_ compiler tools for cross-compiling, you should +use the option `--target=TYPE' to select the type of system they will +produce code for. + + If you want to _use_ a cross compiler, that generates code for a +platform different from the build platform, you should specify the +"host" platform (i.e., that on which the generated programs will +eventually be run) with `--host=TYPE'. + +Sharing Defaults +================ + + If you want to set default values for `configure' scripts to share, +you can create a site shell script called `config.site' that gives +default values for variables like `CC', `cache_file', and `prefix'. +`configure' looks for `PREFIX/share/config.site' if it exists, then +`PREFIX/etc/config.site' if it exists. Or, you can set the +`CONFIG_SITE' environment variable to the location of the site script. +A warning: not all `configure' scripts look for a site script. + +Defining Variables +================== + + Variables not defined in a site shell script can be set in the +environment passed to `configure'. However, some packages may run +configure again during the build, and the customized values of these +variables may be lost. In order to avoid this problem, you should set +them in the `configure' command line, using `VAR=value'. For example: + + ./configure CC=/usr/local2/bin/gcc + +causes the specified `gcc' to be used as the C compiler (unless it is +overridden in the site shell script). + +Unfortunately, this technique does not work for `CONFIG_SHELL' due to +an Autoconf limitation. Until the limitation is lifted, you can use +this workaround: + + CONFIG_SHELL=/bin/bash ./configure CONFIG_SHELL=/bin/bash + +`configure' Invocation +====================== + + `configure' recognizes the following options to control how it +operates. + +`--help' +`-h' + Print a summary of all of the options to `configure', and exit. + +`--help=short' +`--help=recursive' + Print a summary of the options unique to this package's + `configure', and exit. The `short' variant lists options used + only in the top level, while the `recursive' variant lists options + also present in any nested packages. + +`--version' +`-V' + Print the version of Autoconf used to generate the `configure' + script, and exit. + +`--cache-file=FILE' + Enable the cache: use and save the results of the tests in FILE, + traditionally `config.cache'. FILE defaults to `/dev/null' to + disable caching. + +`--config-cache' +`-C' + Alias for `--cache-file=config.cache'. + +`--quiet' +`--silent' +`-q' + Do not print messages saying which checks are being made. To + suppress all normal output, redirect it to `/dev/null' (any error + messages will still be shown). + +`--srcdir=DIR' + Look for the package's source code in directory DIR. Usually + `configure' can determine that directory automatically. + +`--prefix=DIR' + Use DIR as the installation prefix. *note Installation Names:: + for more details, including other options available for fine-tuning + the installation locations. + +`--no-create' +`-n' + Run the configure checks, but stop before creating any output + files. + +`configure' also accepts some other, not widely useful, options. Run +`configure --help' for more details. diff --git a/Makefile.am b/Makefile.am new file mode 100644 index 0000000..100b512 --- /dev/null +++ b/Makefile.am @@ -0,0 +1,79 @@ +DISTCHECK_CONFIGURE_FLAGS=--enable-gtk-doc + +SUBDIRS = \ + gst \ + common \ + examples \ + pkgconfig \ + tests + +DIST_SUBDIRS = $(SUBDIRS) + +EXTRA_DIST = \ + ChangeLog autogen.sh depcomp \ + AUTHORS COPYING NEWS README RELEASE REQUIREMENTS \ + gst-rtsp-server.spec docs/design/gst-rtp-server-design \ + gst-rtsp-server.doap + +ACLOCAL_AMFLAGS = -I m4 -I common/m4 + +DISTCLEANFILES = _stdint.h gst-rtsp-server.spec + +include $(top_srcdir)/common/release.mak +include $(top_srcdir)/common/po.mak + +include $(top_srcdir)/common/coverage/lcov.mak + +check-valgrind: + cd tests/check && make check-valgrind + +if HAVE_CHECK +check-torture: + cd tests/check && make torture +else +check-torture: + true +endif + +# cruft: plugins that have been merged or moved or renamed +CRUFT_FILES = \ + $(top_builddir)/common/shave \ + $(top_builddir)/common/shave-libtool \ + $(top_builddir)/common/m4/codeset.m4 \ + $(top_builddir)/common/m4/gettext.m4 \ + $(top_builddir)/common/m4/glibc2.m4 \ + $(top_builddir)/common/m4/glibc21.m4 \ + $(top_builddir)/common/m4/iconv.m4 \ + $(top_builddir)/common/m4/intdiv0.m4 \ + $(top_builddir)/common/m4/intl.m4 \ + $(top_builddir)/common/m4/intldir.m4 \ + $(top_builddir)/common/m4/intlmacosx.m4 \ + $(top_builddir)/common/m4/intmax.m4 \ + $(top_builddir)/common/m4/inttypes-pri.m4 \ + $(top_builddir)/common/m4/inttypes_h.m4 \ + $(top_builddir)/common/m4/lcmessage.m4 \ + $(top_builddir)/common/m4/lib-ld.m4 \ + $(top_builddir)/common/m4/lib-link.m4 \ + $(top_builddir)/common/m4/lib-prefix.m4 \ + $(top_builddir)/common/m4/libtool.m4 \ + $(top_builddir)/common/m4/lock.m4 \ + $(top_builddir)/common/m4/longlong.m4 \ + $(top_builddir)/common/m4/ltoptions.m4 \ + $(top_builddir)/common/m4/ltsugar.m4 \ + $(top_builddir)/common/m4/ltversion.m4 \ + $(top_builddir)/common/m4/lt~obsolete.m4 \ + $(top_builddir)/common/m4/nls.m4 \ + $(top_builddir)/common/m4/po.m4 \ + $(top_builddir)/common/m4/printf-posix.m4 \ + $(top_builddir)/common/m4/progtest.m4 \ + $(top_builddir)/common/m4/size_max.m4 \ + $(top_builddir)/common/m4/stdint_h.m4 \ + $(top_builddir)/common/m4/uintmax_t.m4 \ + $(top_builddir)/common/m4/visibility.m4 \ + $(top_builddir)/common/m4/wchar_t.m4 \ + $(top_builddir)/common/m4/wint_t.m4 \ + $(top_builddir)/common/m4/xsize.m4 + +include $(top_srcdir)/common/cruft.mak + +all-local: check-cruft diff --git a/NEWS b/NEWS new file mode 100644 index 0000000..cc265ee --- /dev/null +++ b/NEWS @@ -0,0 +1,2 @@ +This is GStreamer RTSP Server 1.4.5 + diff --git a/README b/README new file mode 100644 index 0000000..2373730 --- /dev/null +++ b/README @@ -0,0 +1,4 @@ +gst-rtsp-server is a library ion top of GStreamer for building an RTSP server + +There are some examples in the examples/ directory and more comprehensive +documentation in docs/README. diff --git a/RELEASE b/RELEASE new file mode 100644 index 0000000..f5876ce --- /dev/null +++ b/RELEASE @@ -0,0 +1,69 @@ + +Release notes for GStreamer RTSP Server Library 1.4.5 + +The GStreamer team is pleased to announce a bugfix release of the stable +1.4 release series. The 1.4 release series is adding new features on top +of the 1.2 series and is part of the API and ABI-stable 1.x release +series of the GStreamer multimedia framework that contains new features. +The 1.4.x bugfix releases only contain important bugfixes compared to 1.4.0. + + +Binaries for Android, iOS, Mac OS X and Windows are provided by the +GStreamer project for this release. + + +The 1.x series is a stable series targeted at end users. It is not API +or ABI compatible with the 0.10.x series. It can, however, be installed +in parallel with the 0.10.x series and will not affect an existing +0.10.x installation. + + +The stable 1.4.x release series is API and ABI compatible with 1.0.x and +any other 1.x release series in the future. Compared to 1.0.x it contains +some new features and more intrusive changes that were considered too +risky as a bugfix. + + + + +Bugs fixed in this release + + * 739481 : rtsp-stream: leaks srtp decoder when leaving rtpbin + +==== 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 +  \ No newline at end of file diff --git a/REQUIREMENTS b/REQUIREMENTS new file mode 100644 index 0000000..14e4172 --- /dev/null +++ b/REQUIREMENTS @@ -0,0 +1,3 @@ +You need to have GStreamer. You can use an installed version of +GStreamer or from its build dir. + diff --git a/TODO b/TODO new file mode 100644 index 0000000..53d7d1c --- /dev/null +++ b/TODO @@ -0,0 +1,3 @@ + + - use a config file to configure the server + - error recovery diff --git a/autogen.sh b/autogen.sh new file mode 100755 index 0000000..267b167 --- /dev/null +++ b/autogen.sh @@ -0,0 +1,113 @@ +#!/bin/sh +# Run this to generate all the initial makefiles, etc. + + +test -n "$srcdir" || srcdir=`dirname "$0"` +test -n "$srcdir" || srcdir=. + +olddir=`pwd` +cd "$srcdir" + +DIE=0 +package=gst-rtsp +srcfile=gst/rtsp-server/rtsp-server.c + +# Make sure we have common +if test ! -f common/gst-autogen.sh; +then + echo "+ Setting up common submodule" + git submodule init +fi +git submodule update + +# source helper functions +if test ! -f common/gst-autogen.sh; +then + echo There is something wrong with your source tree. + echo You are missing common/gst-autogen.sh + exit 1 +fi +. common/gst-autogen.sh + +# install pre-commit hook for doing clean commits +if test ! \( -x .git/hooks/pre-commit -a -L .git/hooks/pre-commit \); +then + rm -f .git/hooks/pre-commit + ln -s ../../common/hooks/pre-commit.hook .git/hooks/pre-commit +fi + +NOCONFIGURE=1 +CONFIGURE_DEF_OPT='--enable-maintainer-mode --enable-gtk-doc' + +autogen_options $@ + +printf "+ check for build tools" +if test ! -z "$NOCHECK"; then echo ": skipped version checks"; else echo; fi +version_check "autoconf" "$AUTOCONF autoconf autoconf270 autoconf269 autoconf268 " \ + "ftp://ftp.gnu.org/pub/gnu/autoconf/" 2 68 || DIE=1 +version_check "automake" "$AUTOMAKE automake automake-1.11" \ + "ftp://ftp.gnu.org/pub/gnu/automake/" 1 11 || DIE=1 +#version_check "autopoint" "autopoint" \ +# "ftp://ftp.gnu.org/pub/gnu/gettext/" 0 17 || DIE=1 +version_check "libtoolize" "$LIBTOOLIZE libtoolize glibtoolize" \ + "ftp://ftp.gnu.org/pub/gnu/libtool/" 2 2 6 || DIE=1 +version_check "pkg-config" "" \ + "http://www.freedesktop.org/software/pkgconfig" 0 8 0 || DIE=1 + +die_check $DIE + +aclocal_check || DIE=1 +autoheader_check || DIE=1 + +die_check $DIE + +# if no arguments specified then this will be printed +if test -z "$*" && test -z "$NOCONFIGURE"; then + echo "+ checking for autogen.sh options" + echo " This autogen script will automatically run ./configure as:" + echo " ./configure $CONFIGURE_DEF_OPT" + echo " To pass any additional options, please specify them on the $0" + echo " command line." +fi + +toplevel_check $srcfile + +# autopoint +#if test -d po ; then +# tool_run "$autopoint" "--force" +#fi + +# aclocal +# if test -f acinclude.m4; then rm acinclude.m4; fi + +tool_run "$libtoolize" "--copy --force" +tool_run "$aclocal" "-I m4 -I common/m4 $ACLOCAL_FLAGS" +tool_run "$autoheader" + +# touch the stamp-h.in build stamp so we don't re-run autoheader in maintainer mode +echo timestamp > stamp-h.in 2> /dev/null + +tool_run "$autoconf" +debug "automake: $automake" +tool_run "$automake" "--add-missing --copy" + +test -n "$NOCONFIGURE" && { + echo "+ skipping configure stage for package $package, as requested." + echo "+ autogen.sh done." + exit 0 +} + +cd "$olddir" + +echo "+ running configure ... " +test ! -z "$CONFIGURE_DEF_OPT" && echo " default flags: $CONFIGURE_DEF_OPT" +test ! -z "$CONFIGURE_EXT_OPT" && echo " external flags: $CONFIGURE_EXT_OPT" +echo + +echo "$srcdir/configure" $CONFIGURE_DEF_OPT $CONFIGURE_EXT_OPT +"$srcdir/configure" $CONFIGURE_DEF_OPT $CONFIGURE_EXT_OPT || { + echo " configure failed" + exit 1 +} + +echo "Now type 'make' to compile $package." diff --git a/common/ChangeLog b/common/ChangeLog new file mode 100644 index 0000000..92d0d25 --- /dev/null +++ b/common/ChangeLog @@ -0,0 +1,1712 @@ +2008-12-17 Edward Hervey + + * gst.supp: + And yet another variation of the GstAudioFilter leak. + +2008-12-15 Sebastian Dröge + + Patch by: Roland Illig + + * m4/gst-parser.m4: + Fix AG_GST_BISON_CHECK to handle version numbers with more than + two components (i.e. 2.4.1). Fixes bug #564507. + +2008-12-14 Edward Hervey + + * gst.supp: + And yet another variant of the GstAudioFilter leak. + +2008-12-13 Edward Hervey + + * gst.supp: + Added variants of leaks of dynamic pad templates created in + GstAudioFilter. + Add conditional jump triggered by getaddrinfo (maybe glibc-2.9). + +2008-12-12 Edward Hervey + + * gst.supp: + Fix leak in GIO called by gnomevfs. Nothing we can do about this. + +2008-12-12 Edward Hervey + + * gst.supp: + Added another suppression for dynamic pad templates, in this case + GstAudioFilter. + Added suppression for PangoLanguage which can never be freed + according to the Pango API. + +2008-12-12 Edward Hervey + + * gst.supp: + A whole bunch of suppressions detected on latest gentoo ~amd64. + Make some existing suppressions more generic (for subtle dependecy + code changes). + Added suppressions for glibc-2.9. + Added suppressions for new variants of ALSA leaks. + Added suppressions for a series of leaks in plugins registrations due + to some pad templates' caps calculated at runtime. + Added suppressions for variants of some leaks in pango/fontconfig. + Added suppressions for leak in gstffmpegcsp.c (nothing we can do + about it, but will only exist once). + +2008-12-04 Sebastian Dröge + + * m4/gst-plugin-docs.m4: + Remove the check if $have_gtk_doc equals yes as it's not defined + and $enable_gtk_doc should be good enough. + Also this restores the build of the plugin documentation. + +2008-12-01 Mark Nauwelaerts + + * gst.supp: + Add suppression variant for Ubuntu Hardy x86/64bit. + +2008-12-01 Stefan Kost + + * gtk-doc-plugins.mak: + * gtk-doc.mak: + Simplily uninstall rule. Its closer to upstream and fixes #150331. + +2008-11-29 Sebastian Dröge + + * m4/glib-gettext.m4: + Update glib-gettext.m4 from latest stable GLib release. + +2008-11-29 Sebastian Dröge + + Patch by: Cygwin Ports maintainer + + + * gettext.patch: + Update the gettext patch for use with gettext 0.17 which is + required to build with libtool 2.2 because of conflicts. + First part of bug #556091. + +2008-11-29 Sebastian Dröge + + * m4/gtk-doc.m4: + * m4/pkg.m4: + Update gtk-doc and pkg-config m4 macros from their latest releases. + +2008-11-20 Michael Smith + + * m4/as-objc.m4: + Fix objective C test macro when none of the compilers are found at all. + +2008-10-30 Stefan Kost + + * gtk-doc.mak: + Also cp the entities here to all xinlcude based docs (workaround for + not being able to set up a search path). + +2008-10-17 Jan Schmidt + + * gtk-doc.mak: + Don't clobber the real registry cache file when + building docs. + +2008-10-07 Jan Schmidt - Sun Microsystems + + * m4/gst-error.m4: + Also disable the bogus "loop not entered at top" warnings appearing on Sparc Forte builds. + +2008-10-06 Stefan Kost + + * gtk-doc.mak: + Apply the same fix as below to gtk-doc.mak. Somehow did not end up in + CVS. + +2008-09-05 David Schleef + + * gtk-doc-plugins.mak: Fix the check for gtkdoc-rebase: don't + pass the 'which' error back to make. This fix is more specific + than what is in upstream. + +2008-09-05 David Schleef + + * gtk-doc.mak: Fix the check for gtkdoc-rebase: don't pass the + 'which' error back to make. This fix is more specific than + what is in upstream. + +2008-09-04 Stefan Kost + + * gtk-doc-plugins.mak: + * gtk-doc.mak: + Get closer to upstream makefiles. Don't install index.sgml twice. Call + gtkdoc-rebase (if exists). + +2008-08-21 Stefan Kost + + * gtk-doc-plugins.mak: + Revert $(top_builddir) -> $(builddir) change of rev. 1.39 as there is + no variable called builddir. + +2008-07-31 Mark Nauwelaerts + + * gst.supp: + Add suppressions for Ubunty Hardy x86/64bit, similar to earlier + versions and 32bit variant. + +2008-07-31 Sebastian Dröge + + * m4/gst-feature.m4: + Remove GST_DISABLE_(ENUMTYPES|INDEX|URI). + +2008-07-21 Tim-Philipp Müller + + * m4/gst-error.m4:: + When checking for GST_ERROR_CXXFLAGS, check each compiler flag + individually, not all together. + +2008-07-20 Tim-Philipp Müller + + * m4/gst-parser.m4:: + Fix bison version number detection for older --version + output format (as bison 1.28 on OSX 10.4 outputs). + Fixes #543853. + +2008-07-12 Stefan Kost + + * plugins.xsl: + Split refsect2 also here to make "Element Pads" subtitle visible. + +2008-07-08 Sebastian Dröge + + * m4/gst-error.m4: + Add compiler flags to warn if declarations after statements or + variable length arrays are used. These are C99/GCC extensions and + are not supported by some compilers we want to support. + +2008-07-02 Mark Nauwelaerts + + * gtk-doc-plugins.mak: + Only clean doc maintainer stamps in maintainer-clean. Fixes #539977. + +2008-06-20 Sebastian Dröge + + * gstdoc-scangobj: + Always use format strings for printf-like functions, even if they just + print a string. Fixes bug #536981. + +2008-06-20 Sebastian Dröge + + * gtk-doc-plugins.mak: + * gtk-doc.mak: + Include CFLAGS and LDFLAGS in GTKDOC_CFLAGS and GTKDOC_LDFLAGS, + otherwise the values passed to configure are ignored. + Fixes bug #536978. + +2008-06-05 Tim-Philipp Müller + + * m4/gst-error.m4: + Add -fno-strict-aliasing when compiling with -Werror, to work around + warnings caused by G_LOCK with recent GLib versions (2.16.x) (#316221). + +2008-06-05 Jan Schmidt + + * gtk-doc.mak: + Don't copy html/*.png files unless they don't already exist + in the destdir. Fixes distcheck failure caused by permissions + problems trying to copy a file into the destdir when it already + exists. + +2008-05-28 Stefan Kost + + * plugins.xsl: + The class was not shown in plugin docs. Fix typo in changelog below. + +2008-05-22 Jan Schmidt + + * gstdoc-scangobj: + Emit warnings if one of the GTypes we're expecting is 0 + when scanning. + +2008-05-21 Felipe Contreras + + * gtk-doc-plugins.mak: + * gtk-doc.mak: + Fix installing png images when gtk-doc is disabled. + +2008-05-21 Felipe Contreras + + * gtk-doc-plugins.mak: + * gtk-doc.mak: + Fix make clean when gtk-doc is disabled and other cleanups. + +2008-05-17 Jan Schmidt + + * gtk-doc-plugins.mak: + Be more quiet when the files don't yet exist. + +2008-05-16 Jan Schmidt + + * gstdoc-scangobj: + Add a mechanism for adding 'implicitly created' GTypes into the + scan, allowing for documenting plugin-private base classes that + provide signals or properties for public elements. + + * gtk-doc-plugins.mak: + Use $(builddir) instead of $(top_builddir) in a few places - there's + no need to hard code 'docs/plugins' as the only useable path. + +2008-05-14 Peter Kjellerstedt + + * m4/gst-feature.m4: + Report plug-ins without external dependencies that will not be built + even when the name of the plug-in is a substring of another plug-in, + e.g., goom vs. goom2k1. + +2008-05-14 Tim-Philipp Müller + + * gst.supp: + Add suppression for glibc bug on gutsy/x86-64 + +2008-05-12 Stefan Kost + + * plugins.xsl: + Improve the layout of the caps, but splitting them on ";". + +2008-05-09 Sebastian Dröge + + Patch by: Brian Cameron + + * m4/gst-default.m4: + Don't set the default audio sink to the default visualizer. + Fixes bug #532295. + +2008-05-07 Tim-Philipp Müller + + * check.mak: (help): + Document GST_CHECKS environment variable in checks 'make help'. + +2008-05-06 Sebastian Dröge + + Patch by: Marc-Andre Lureau + + * scangobj-merge.py: + Don't depend on Twisted just for the OrderedDict but implement our + own ordered dictionary class. Fixes bug #531577. + +2008-04-23 Edward Hervey + + * gst.supp: + Re-arrange latest suppressions. + Add all known suppressions for ubuntu hardy. Same as for older + ubuntus, but with different codepaths. + +2008-04-22 Edward Hervey + + * gst.supp: Make tls leak suppression a bit more generic. + +2008-04-22 Edward Hervey + + * gst.supp: Fix ommission in latest commit. + Make tls leak suppression more generic in order to cover more + distributions (and hopefully also future distributions). + +2008-04-22 Edward Hervey + + * gst.supp: Add suppressions for Hardy. + They're just the newer versions of similar suppressions we had + for the previous versions of ubuntu. + +2008-04-15 Sebastian Dröge + + * Makefile.am: + * m4/Makefile.am: + Dist all files in common. Fixes bug #527984. + +2008-04-14 Tim-Philipp Müller + + * m4/gst-function.m4: + Rename AC_CACHE_VAL cache-ids to contain '_cv_' in order to make + autoconf-2.62 complain less. + +2008-04-13 Tim-Philipp Müller + + * m4/gst-args.m4: + * m4/gst-valgrind.m4: + Bump valgrind requirement to 3.0 (which was released in August 2005). + Fixes #489269. Also, check for version >=REQ and not >REQ. + +2008-04-09 Tim-Philipp Müller + + * m4/gst-default.m4: + Add --with-default-{audiosink|audiosrc|videosink|videosrc|visualizer} + configure switches (#519417). + +2008-04-03 Tim-Philipp Müller + + * m4/gst-args.m4: + Add --disable-foo switch for dependency-less plugins (#525586). + +2008-04-01 Sebastian Dröge + + * m4/gst-parser.m4: + Unconditionally require flex 2.5.31 and bison 1.875. + +2008-03-23 Sebastian Dröge + + * m4/gst-arch.m4: + amd64/x86_64 allows unaligned memory access too. + +2008-03-21 Sebastian Dröge + + * m4/gst-dowhile.m4: + Add macro that checks if the compiler supports do {} while (0) + macros and define HAVE_DOWHILE_MACROS if it does. This is + needed by glib/gmacros.h to use something else than + if (1) else for G_STMT_START/END when compling C++, which + causes compiler warnings because of ambigious else with g++ 4.3. + +2008-03-21 Sebastian Dröge + + * m4/gst-plugin-docs.m4: + * mangle-tmpl.py: + Don't depend on PyXML and use only XML modules that are shipped + with python. Fixes bug #519635. + +2008-03-07 Edward Hervey + + * m4/gtk-doc.m4: (GTK_DOC_CHECK): + The previous commit to this file by Stefan Kost mentionned checking for + SED, but NOT checking for gtkdoc-check (wth is that doing there ??). + Therefore, removing the check for gtkdoc-check + +2008-03-03 David Schleef + + * m4/ax_create_stdint_h.m4: Oops, checked in the wrong copy of + this file. (Update from upstream) + +2008-03-03 David Schleef + + * m4/ax_create_stdint_h.m4: Update from upstream. Fixes a bug + compiling with MSVC. + +2008-03-03 Edward Hervey + + * m4/pkg.m4: + Allow override of pkg-config results, as proposed by configure --help. + This is in fact just a backport from upstream pkg.m4. + Fixes #518892 + +2008-03-03 Peter Kjellerstedt + + * ChangeLog: + Changelog surgery of my previous commit to add bugzilla reference. + * m4/gst-args.m4: + Add AG_GST_CHECK_PLUGIN and AG_GST_DISABLE_PLUGIN to make it easier + to include and exclude plug-ins without external references, i.e., + plug-ins listed in GST_PLUGINS_SELECTED. (#498222) + +2008-03-03 Sebastian Dröge + + * gst.supp: + Add another glibc suppression. + +2008-02-29 Peter Kjellerstedt + + * m4/gst-feature.m4: + Make the comment before defines generated via AG_GST_CHECK_FEATURE + look nicer. (#498222) + +2008-02-26 Jan Schmidt + + * m4/Makefile.am: + * m4/as-gcc-inline-assembly.m4: + Add Dave Schleef's GCC inline assembly detection macro + for using in gst-plugins-good in the goom 2k4 plugin. + +2008-02-25 Andy Wingo + + * gst-autogen.sh: Instead of only passing certain arguments to + configure, pass anything that we didn't handle. Much friendlier. + Fixes #34412. + +2008-02-23 Jan Schmidt + + * m4/gst-error.m4: + Store the detected compiler flags into ERROR_CFLAGS rather than + ERROR_CXXFLAGS, and use the macro that checks the C compiler, not + the C++ one. + +2008-02-23 Tim-Philipp Müller + + * m4/gst-error.m4: + Reflow checks for additional warning flags so they're not + nested, which fixes the result reporting in the configure + output. + +2008-02-22 Tim-Philipp Müller + + * m4/as-compiler-flag.m4: + Add AS_CXX_COMPILER_FLAG + + * m4/gst-error.m4: + Add AG_GST_SET_ERROR_CXXFLAGS (Forte bits need testing) + +2008-02-22 Tim-Philipp Müller + + * gtk-doc-plugins.mak: + Add 'check-inspected-versions' target; this helps identify + files that should have been removed or where the version + number should (ideally) be updated before a release + (which doesn't happen automatically if the releaser doesn't + build that plugin locally). Not adding at a distcheck hook + yet though, because it's not really that important and would + probably also be a problem on buildbots. + +2008-02-22 Sebastian Dröge + + * gst.supp: + Add even more glibc 2.7 suppressions. + +2008-02-22 Sebastian Dröge + + * gst.supp: + Add another suppression for GLib caching some values after + the first call. + +2008-02-12 Sebastian Dröge + + Patch by: + Tim Mooney + + * m4/gst-error.m4: + Use no%E_MACRO_REDEFINED on Solaris to prevent compiler warnings. + Fixes bug #515905. + +2008-02-11 Sebastian Dröge + + * gst.supp: + Add a few more glibc 2.7 suppressions to make the avisubtitle unit + test valgrind clean. Fixes bug #515703. + +2008-02-08 Stefan Kost + + * ChangeLog: + Changelog surgery for last commit. + +2008-02-08 Stefan Kost + + * m4/gtk-doc.m4: + Conditionally check for SED. Also sync a bit with upstream macro. + +2008-02-08 Stefan Kost + + * gtk-doc-plugins.mak: + * gtk-doc.mak: + Use '$(SED)' instead of 'sed'. Don't use -i for in-place as its gnu + only, move to a temp file instead. + +2008-02-06 Stefan Kost + + * gtk-doc-plugins.mak: + * gtk-doc.mak: + As our docs are versioned, we need to patch the index.sgml file to have + correct paths there, unless we also want to fork gtk-doc's xsl (which + we don't). This hopefully fixes xrefs between modules. + +2008-02-02 Sebastian Dröge + + * m4/gst-feature.m4: + Use printf instead of echo as "echo -e" isn't POSIX and doesn't work + with strict POSIX shells like tcsh or dash and also not every platform + has a /bin/echo that supports it. + +2008-01-24 Stefan Kost + + * ChangeLog: + ChangeLog surgery. + + * gstdoc-scangobj: + Sync the object scanner with gtk-doc fixes. Update args and hierarchy + files. + +2008-01-20 Sebastian Dröge + + * check.mak: + * coverage/lcov.mak: + * gtk-doc-plugins.mak: + * release.mak: + Use $(MAKE) instead of make to fix the build if GNU make is called + something else on the system. + + * m4/as-docbook.m4: + Fix path for docbook.xsl if we have no /etc/xml/catalog and add a + docbook-xsl search path for FreeBSD. + +2008-01-18 Sebastian Dröge + + * gst.supp: + Add a suppression for a glibc bug: + http://valgrind.org/docs/manual/faq.html#faq.exit_errors> + +2008-01-18 Sebastian Dröge + + * gst.supp: + Add some more glibc 2.7 suppressions and make the GLib suppressions + for the home/tmp/etc directory caching a bit more generic. + +2008-01-18 Sebastian Dröge + + * gst.supp: + Add some glibc 2.7 supressions as found on Debian/unstable. + +2008-01-14 Jan Schmidt + + * download-translations: + Apparently I have problems with leaving things commented out when + I edit shell scripts. + +2008-01-12 Jan Schmidt + + * download-translations: + Remove bash-isms + +2008-01-12 Jan Schmidt + + * check-exports: + Restore the cleanup rm of our tmp file which I didn't mean to leave + commented out. + +2008-01-12 Jan Schmidt + + * check-exports: + Fixes to make check-export work on both Solaris and Linux + + * m4/gst-error.m4: + Disable extra warning category (argument mismatch) as an error + on Forte, as it prevents the libcheck fail_if macros from compiling. + + * win32.mak: + Substitute the GStreamer version so things will keep working in 0.11 + +2008-01-11 Tim-Philipp Müller + + Patch by: Peter Kjellerstedt + + * m4/gst-glib2.m4: + * m4/gst-libxml2.m4: + Improve/fix output from configure if either glib-2.0 or + libxml2 are not installed (#498222). + +2008-01-09 Stefan Kost + + * coverage/lcov.mak: + Update coverage make-rules: use them conditionaly, use libtool mode + and use lcov to cleanup. + +2007-12-18 Sebastian Dröge + + * glib-gen.mak: + Also use #include "header" instead of #include
for the + headers that were used to generate the source files for the same + reason as below. + + Remove whitespace before #include. + +2007-12-18 Sebastian Dröge + + * glib-gen.mak: + Use #include "header" instead of #include
for the generated + enum C files as the file will always be in the same directory and + some compilers seem to be a bit strict about that unless . is added + to the include path. + + Include all headers that were used to generate the source files in + the C file as they're used there. + +2007-12-17 Tim-Philipp Müller + + * win32.mak: (win32), (win32defs), (win32crlf): + Make check for CR LF in Visual C++ 6.0 project files + work, based on patch by David Schleef (#496722, #393626). + +2007-12-17 Tim-Philipp Müller + + * Makefile.am: + Don't forget to dist the new win32.mak. + +2007-12-17 Tim-Philipp Müller + + * win32.mak: (win32), (win32defs): + Move common win32 Makefile foo into this new file. + +2007-12-15 Stefan Kost + + * gtk-doc-plugins.mak: + * gtk-doc.mak: + We should have never forked this that much :/. + +2007-12-13 Tim-Philipp Müller + + * check-exports: + Fix build on the ppc64 build bot. + +2007-12-13 Tim-Philipp Müller + + * check-exports: + Suppress more unintentional exports (too much hassle to rename them, + since the win32 project files would need changing too). + +2007-12-12 Tim-Philipp Müller + + * Makefile.am: + check-exports should be disted. + +2007-12-12 Tim-Philipp Müller + + * check-exports: + Add quick'n'dirty script to check the exported symbols of a library + against the symbols in the corresponding .def file (#493983). Based + on script by Ole André Vadla Ravnås. + +2007-11-06 Jan Schmidt + + * gtk-doc-plugins.mak: + Fix distcheck by making sure the types files are treated like the + other gtkdoc-scangobj generated files. + +2007-09-21 Sebastian Dröge + + * m4/gst-args.m4: + Let the AG_GST_ARG_ENABLE_EXPERIMENTAL macro default to disable + building of experimental plugins. Nobody uses it yet and the + --enable--experimental stuff from gst-plugins-good defaults to + disable too. + +2007-09-06 Tim-Philipp Müller + + * gtk-doc-plugins.mak: + Just use the normal 'check' target and avoid a circular + dependency. + +2007-09-06 Tim-Philipp Müller + + * gtk-doc-plugins.mak: + Add rule to error out if .hierarchy file contains tabs. + +2007-08-20 Tim-Philipp Müller + + * download-translations: + * po.mak: + If there are new languages, they need to be added to po/LINGUAS. + +2007-08-20 Tim-Philipp Müller + + * download-translations: + * po.mak: + Fix up 'download-po' a bit, so that we find new translations + for languages that aren't in our po/LINGUAS file yet too. + +2007-07-16 Jan Schmidt + + * gst.supp: + Add a suppression for GLib caching the tmp dir seen on an + Ubuntu Feisty system. + +2007-07-13 Jan Schmidt + + * m4/gst-feature.m4: + If we want to use 'echo -e', call /bin/echo instead of the shell's + since -e is a bash extension, and our /bin/sh might not be being + provided by bash. + +2007-07-01 Thomas Vander Stichele + + * po.mak: + Translation project has moved. Also, no idea how this used to + work given that we weren't downloading a .po file. + +2007-06-25 Stefan Kost + + * gst-xmlinspect.py: + * plugins.xsl: + Also extract element caps for plugin-docs. Fixes parts of #117692. + +2007-06-21 Tim-Philipp Müller + + Patch by: Andreas Schwab + + * m4/gst-feature.m4: + Fix quoting (#449493). + +2007-06-10 Sebastian Dröge + + * m4/gst-parser.m4: + Only generate the parser if bison >= 1.875 _and_ flex >= 2.5.31 is + installed and use pre-generated sources otherwise. Fixes bug #444820. + +2007-05-11 Michael Smith + + * gst.supp: + Suppression variant for our good friend the TLS leak, this time for + Ubuntu Feisty/x86. + +2007-05-09 Tim-Philipp Müller + + * gtk-doc-plugins.mak: + Fix make distcheck again; change some spaces to tabs in makefile. + +2007-04-29 Thomas Vander Stichele + + * gtk-doc-plugins.mak (-module): + Error out when the html build step gives warnings, so they get + fixed properly. + +2007-04-23 Stefan Kost + + * m4/gst-feature.m4: + Add macro AG_GST_PARSE_SUBSYSTEM_DISABLES that checks the defines in + the configuration header and AC_DEFINES the setings. + +2007-04-19 Sebastian Dröge + + Patch by: Vincent Torri + + * m4/gst-parser.m4: + Put the AC_MSG_RESULT output in brackets to get it properly written to + the terminal. + +2007-04-18 Sebastian Dröge + + * m4/gst-parser.m4: + Check for flex >= 2.5.31 and set GENERATE_PARSER if we have at least + that version. Otherwise use pre-generated parser sources as we can't + raise the required flex version. HAVE_MT_SAVE_FLEX is obsolete now + as we use a new enough flex version anyway. First part of #349180 + +2007-04-10 Thomas Vander Stichele + + * m4/gst-check.m4: + Allow pre-setting the GST(PB)_TOOLS/PLUGINS_DIR variables to help + builds against older GStreamer. + +2007-03-25 Sebastian Dröge + + * m4/gst-parser.m4: + Fix the flex version check. It ignored the micro version before. + +2007-03-09 Jan Schmidt + + * check.mak: + Use the same timeout when generating valgrind suppressions as + running the valgrind test. + + * gst.supp: + Add some more suppressions and stuff. + +2007-03-08 Jan Schmidt + + * check.mak: + Make sure GSlice is disabled when building suppressions too. + + * gst.supp: + Add around *850* lines of suppressions for one-time initialisations + inside libasound and gconf/bonobo/ORBit. I feel so dirty. + +2007-03-07 Jan Schmidt + + * gst.supp: + add a suppression for this GConf flup on the FC5 buildbot. + +2007-03-06 Jan Schmidt + + * gst.supp: + Make the suppression a little more generic, to catch the FC5 + backtrace too. + +2007-03-06 Jan Schmidt + + * gst.supp: + Add a suppression for libcdio 0.76. It leaks an internal struct + when the CD-ROM device is not accessible. + +2007-02-28 Thomas Vander Stichele + + * m4/gst-arch.m4: + Move a line that was in the wrong macro + +2007-02-28 Thomas Vander Stichele + + * m4/gst.m4: + Add + * m4/gst-arch.m4: + * m4/gst-args.m4: + * m4/gst-check.m4: + * m4/gst-debuginfo.m4: + * m4/gst-default.m4: + * m4/gst-doc.m4: + * m4/gst-error.m4: + * m4/gst-feature.m4: + * m4/gst-function.m4: + * m4/gst-gettext.m4: + * m4/gst-glib2.m4: + * m4/gst-libxml2.m4: + * m4/gst-parser.m4: + * m4/gst-plugin-docs.m4: + * m4/gst-plugindir.m4: + * m4/gst-valgrind.m4: + * m4/gst-x11.m4: + Convert all macros to use AG_GST style so we can properly warn + when they're missing if configure.ac calls AG_GST_INIT + Will require update in all GStreamer modules. + +2007-02-11 Stefan Kost + + * m4/gst-args.m4: + Remove 'enable' from configure switch description as this leads to + confusing lines like "disable enable builing ...". + * m4/gst-feature.m4: + Fix comment to sound less horrible. + +2007-02-07 Tim-Philipp Müller + + Patch by: Will Newton + + * m4/gst-check.m4: + Use $PKG_CONFIG rather than pkg-config directly, the one in our path + might not be the one we want, like when cross-compiling. Also, other + macros such as PKG_CHECK_MODULES use $PKG_CONFIG, so we should + probably too just for consistency. Fixes #405288. + +2007-01-08 Tim-Philipp Müller + + * m4/gst-parser.m4: + Need to use double square brackets again so m4 doesn't remove them + (fixes #378931). + + * m4/gst-args.m4: + Use double square brackets here as well, for the same reason. + +2007-01-05 Tim-Philipp Müller + + * m4/gst-parser.m4: + Use 'sed' rather than 'tr' to strip trailing letters from version + numbers, since 'tr' might not be available and we know sed is + (#378931). + +2006-10-21 Tim-Philipp Müller + + * check.mak: + Increase default timeout under valgrind, 60 is just too short and + some tests take a bit longer these days and not everyone has a + beefy machine. + +2006-09-29 Michael Smith + + * gst.supp: + More suppressions for edgy. + +2006-09-28 Jan Schmidt + + * m4/gst-glib2.m4: + Use gmodule-no-export-2.0.pc instead of gmodule-2.0.pc - we neither + want nor need --export-dynamic (which ends up making us export a bunch + of unneeded symbols) + +2006-09-14 Tim-Philipp Müller + + * gst.supp: + Some suppressions for the more recent ld.so in ubuntu edgy. + +2006-08-23 Tim-Philipp Müller + + * gst.supp: + Shorten function trail so the suppression works on + my ubuntu dapper system with core cvs as well. + +2006-07-28 Jan Schmidt + + * gst.supp: + Extra suppressions from my Ubuntu x86_64 machine + +2006-07-24 Tim-Philipp Müller + + Patch by: Frederic Peters + + * m4/gst-parser.m4: + Need to double square brackets in .m4 files. Should fix bison + version detection with version numbers like 1.23a (#348354). + +2006-07-24 Jan Schmidt + + * check.mak: + Valgrind fails to find tests written in tests/check/ directly (rather + than a subdir) - because valgrind gets run with a filename that + doesn't contain a relative path, it goes searching /usr/bin instead. + Run with ./.... to make things work either way. + + * gtk-doc-plugins.mak: + Add $(top_builddir)/src as a place to look for plugins + when building too, since that's where gst-template keeps things + +2006-07-23 Stefan Kost + + Patch by: Frederic Peters + + * m4/gst-parser.m4: + Fix bison detection (#348354) + +2006-07-21 Stefan Kost + + * m4/gst-parser.m4: + check for bison and flex + +2006-07-13 Thomas Vander Stichele + + * m4/gst-plugin-docs.m4: + remove the configure argument for enabling plugin doc build; + having gtk-doc enabled and pyxml present is enough of a trigger + +2006-07-03 Thomas Vander Stichele + + * coverage/lcov.mak: + fix up rules to work with gst-python as well + run "make lcov" to test and generate the reports + run "make lcov-reset" to redo it after that + +2006-07-02 Thomas Vander Stichele + + * Makefile.am: + * check.mak: + add an inspect target that inspects every element feature, + so we can have that added for coverage + * coverage/lcov.mak: + add support for lcov + +2006-07-02 Thomas Vander Stichele + + * m4/gst-args.m4: + when building with gcov, reset CFLAGS and friends to O0 + +2006-07-02 Thomas Vander Stichele + + * m4/gst-args.m4: + Find the gcov that matches the gcc version + Only allow gcov if we use gcc + +2006-07-02 Thomas Vander Stichele + + * Makefile.am: + * coverage/coverage-report-entry.pl: + * coverage/coverage-report.pl: + * coverage/coverage-report.xsl: + copy coverage reporting files from dbus + +2006-07-01 Thomas Vander Stichele + + * m4/gst-args.m4: + libtool strips gcov's -f flags, so libgcov does not get + linked in. Setting GCOV_LIBS with -lgcov fixes libtool's + stripping + also show what pkg-config-path we set + +2006-06-22 Tim-Philipp Müller + + Patch by: Peter Kjellerstedt + + * m4/gst-feature.m4: + Show list of plugins without external dependencies that + will not be built as well (#344136). + +2006-06-15 Tim-Philipp Müller + + * m4/gst-plugin-docs.m4: + add GST_PLUGIN_DOCS, which checks for everything needed + to build the plugin docs (namely gtk-doc and pyxml); also + adds a new --enable-plugin-docs configure switch; will + set ENABLE_PLUGIN_DOCS conditional for use in Makefile.am + files (see #344039). + +2006-06-11 Thomas Vander Stichele + + * m4/gst-check.m4: + add GST_PKG_CHECK_MODULES, which in the normal case of checking + for a dependency lib for a plug-in only needs two arguments + to do the right thing. + * m4/gst-feature.m4: + clean up output a little of feature checking; also deal with + non-plug-in feature checks + * m4/Makefile.am: + * m4/gst-gstreamer.m4: + remove this file; it's a useless check + +2006-06-06 Thomas Vander Stichele + + * m4/gst-arch.m4: + add PPC64 so we can have separate structure sizes for it + +2006-06-05 Edward Hervey + + * gtk-doc.mak: + Check for the proper .devhelp2 file to remove. + +2006-05-31 Thomas Vander Stichele + + * gtk-doc.mak: + allow a magic variable to suppress errors from docbuilding + +2006-05-30 Thomas Vander Stichele + + * gtk-doc.mak: + error out if gtkdoc-mktmpl finds unused declarations + +2006-05-28 Edward Hervey + + * gst.supp: + Reverting previous commit. That's good to know, Edward, but why ? + +2006-05-28 Edward Hervey + + * gst.supp: + Added suppresion for memleak in g_option_context_parse on fc5-64 + +2006-05-19 Thomas Vander Stichele + + * m4/gst-check.m4: + set GSTPB_PLUGINS_DIR just like GST_PLUGINS_DIR + +2006-05-18 Tim-Philipp Müller + + * check.mak: + Fix 'make help' in check directories, it should be + 'valgrind.gen-suppressions' not 'valgrind-gen-suppressions' + (not changing target to match help string on purpose to keep + scripts etc. functional). + +2006-05-18 Thomas Vander Stichele + + Patch by: Peter Kjellerstedt + + * m4/gst-arch.m4: + add support for CRIS and CRISv32. + +2006-05-17 Jan Schmidt + + * m4/gst-args.m4: + Fix the macros for command-line supplied package and origin names + so they don't end up being configure as "" (Fixes #341479) + +2006-05-14 Jan Schmidt + + * gtk-doc.mak: + Add uninstall rule to remove .devhelp2 files. + +2006-05-09 Edward Hervey + + * gst.supp: + Add suppression for GSlice version of + g_type_init calloc leak + +2006-04-05 Michael Smith + + * gst.supp: + Delete a bogus suppression for the registry code. + Generalise a suppression for a glib bug (see #337404) + +2006-04-04 Michael Smith + + * gst.supp: + Add a leak suppression: the existing glibc-doesn't-free-TLS one + wasn't triggering here. + +2006-04-04 Michael Smith + + * gst.supp: + Add some minimally-neccesary suppressions for my x86/dapper system. + +2006-04-01 Thomas Vander Stichele + + * plugins.xsl: + Do not display an origin link if origin does not start with http + See #323798 + +2006-04-01 Thomas Vander Stichele + + * m4/gst-args.m4: + * m4/gst-feature.m4: + add more macros + * m4/gst-x11.m4: + X11-related checks + +2006-04-01 Thomas Vander Stichele + + * m4/as-version.m4: + newer version + * m4/gst-args.m4: + * m4/gst-doc.m4: + update and add other macros to be shared across projects + +2006-03-24 Thomas Vander Stichele + + * gst.supp: + add a suppression for g_parse_debug_string + +2006-03-23 Stefan Kost + + * gstdoc-scangobj: + sync fully with gtkdoc-0.15 + +2006-03-23 Stefan Kost + + * gstdoc-scangobj: + * gtk-doc.mak: + sync a little with gtk-doc mainline + +2006-03-17 Wim Taymans + + * gst.supp: + add another clone suppression + change all glibc suppressions to match 2.3.* + +2006-03-09 Thomas Vander Stichele + + * m4/check.m4: + fix test so it actually works when the normal check is used + over debian's/ubuntu's + +2006-03-08 Jan Schmidt + + * check.mak: + Set G_SLICE=always-malloc when valgrinding tests + (closes #333272) + +2006-02-21 Jan Schmidt + + * m4/gst-glib2.m4: + Fix debug output when the GLib version prerequisite is not found + +2006-02-13 Andy Wingo + + * m4/check.m4: Hack around Debian/Ubuntu's broken installation of + the PIC version of check as libcheck_pic.a. Should work with + cross-compilation too. Grr. + +2006-02-06 Thomas Vander Stichele + + * m4/gst-default.m4: + switch to auto* sinks for defaults + +2006-02-02 Wim Taymans + + * check.mak: + add a .valgrind.gen-suppressions target to aid in generating + suppressions + * gst.supp: + add more repressions from my debian glibc as of today + +2006-02-02 Thomas Vander Stichele + + * gtk-doc-plugins.mak: + only add srcdir/gst if it exists + +2006-01-30 Thomas Vander Stichele + + * release.mak: + don't complain about disted enums in win32 + +2006-01-20 Thomas Vander Stichele + + * m4/gst-check.m4: + AC_SUBST CFLAGS and LIBS + do a non-command because something is stripping out our AC_SUBST + +2006-01-20 Thomas Vander Stichele + + * m4/gst-args.m4: + * m4/gst-valgrind.m4: + properly give a "no" result manually when providing a + not-found action to fix configure output + +2006-01-20 Thomas Vander Stichele + + * m4/pkg.m4: + update with a more recent version + +2006-01-07 Thomas Vander Stichele + + * gettext.patch: + make Makefile depend on LINGUAS, so rebuilds work when adding + a language + +2006-01-03 Michael Smith + + * check.mak: + Clarify error message from valgrind test runs. + +2005-12-16 Thomas Vander Stichele + + * m4/gst-arch.m4: + define HOST_CPU + +2005-11-29 Thomas Vander Stichele + + * check.mak: + add a valgrind-forever target for tests + +2005-11-28 Thomas Vander Stichele + + * check.mak: + when a "make test.check" run fails, make it rerun the test with + at least debug level 2 + +2005-11-14 Thomas Vander Stichele + + * m4/Makefile.am: + * m4/gst-check.m4: + fix check for base plugins + * m4/gst-default.m4: + add m4 to set default elements + +2005-10-18 Thomas Vander Stichele + + * m4/gst-check.m4: + check for tools correctly + +2005-10-18 Thomas Vander Stichele + + * gtk-doc.mak: + only enable breaking on new API when make distcheck passes, + not before + +2005-10-18 Thomas Vander Stichele + + * m4/gst-check.m4: + Resurrect Julien's dead body and wipe his mind clean + +2005-10-18 Thomas Vander Stichele + + * m4/gst-check.m4: + Kill Julien + +2005-10-17 Julien MOUTTE + + * m4/gst-check.m4: I know Thomas will kill me but this + ifelse statement seems incorrect as it is always setting + required to "yes". With this one it seems to work. Fixes + build of gst-plugins-base on my setup where gstreamer-check + is definitely not present/required. + +2005-10-18 Stefan Kost + + * gtk-doc.mak: + make build break on new api that has not been added to the + sections file + +2005-10-17 Thomas Vander Stichele + + * m4/gst-glib2.m4: + * m4/Makefile.am: + * m4/gst-check.m4: + add macro for easy checks for GStreamer libs + +2005-10-16 Thomas Vander Stichele + + * m4/gst-glib2.m4: + update, warn in error cases + +2005-10-16 Thomas Vander Stichele + + * m4/gst-error.m4: + add GST_SET_DEFAULT_LEVEL + +2005-10-16 Thomas Vander Stichele + + * m4/Makefile.am: + * m4/gst-gettext.m4: + remove the AM_GNU_GETTEXT* calls, they need to be in configure.ac + * m4/gst-glib2.m4: + clean up and re-use in core soon + * m4/gst-plugindir.m4: + macro to set up PLUGINDIR and plugindir define/var + +2005-10-15 Thomas Vander Stichele + + * m4/Makefile.am: + * m4/gst-gettext.m4: + add macro for setting up gettext + +2005-10-15 Thomas Vander Stichele + + * m4/gst-args.m4: + add some .m4's for argument checking that can be shared among modules + +2005-10-15 Thomas Vander Stichele + + * m4/as-libtool.m4: + set _LT_LDFLAGS + * m4/gst-libxml2.m4: + document + +2005-10-15 Thomas Vander Stichele + + * m4/gst-arch.m4: + indent a little + add AC_REQUIRE + * m4/gst-error.m4: + clean up + +2005-10-12 Thomas Vander Stichele + + * gst-autogen.sh: + update version detection expression to catch stuff like + Libtool (libtool15) 1.5.0 + +2005-10-11 Thomas Vander Stichele + + * gst.supp: + commit 6 new suppressions related to g_module_open; can these + really not be folded into one ? + +2005-10-11 Edward Hervey + + * gst.supp: + made the suppression more generic + Added pthread memleak suppresions + Added nss_parse_* memleak suppresion (used by g_option_context_parse) + +2005-10-11 Thomas Vander Stichele + + * check.mak: + be more strict, more leak resolution + * gst.supp: + clean up the g_type_init suppressions + +2005-10-07 Thomas Vander Stichele + + * m4/Makefile.am: + * m4/gst-valgrind.m4: + put the valgrind detection in an .m4 + +2005-09-29 Thomas Vander Stichele + + * check.mak: + add some more targets, like "help", but also more intensive tests + +2005-09-23 Thomas Vander Stichele + + * gtk-doc.mak: + make certain doc warnings fatal so people maintain docs again + +2005-09-23 Thomas Vander Stichele + + * Makefile.am: + * gtk-doc-plugins.mak: + * scangobj-merge.py: + merge additions from the .signals.new and .args.new file in + the original ones, only updating if necessary + +2005-09-23 Thomas Vander Stichele + + * gst-xmlinspect.py: + * gstdoc-scangobj: + * gtk-doc-plugins.mak: + fix properly for new API; make update in plugins dir now works + +2005-09-20 Thomas Vander Stichele + + * gst-xmlinspect.py: + * gstdoc-scangobj: + some fixes for new API + * gtk-doc-plugins.mak: + set environment properly + +2005-09-17 David Schleef + + * gtk-doc-plugins.mak: Use new environment variables. + +2005-09-16 Michael Smith + + * gstdoc-scangobj: + Make the scanobj code reflect registry/plugin API changes + +2005-09-15 Thomas Vander Stichele + + * gtk-doc-plugins.mak: + split out scanobj step (which will be run by doc maintainer) + from scan step (which will be run on every build) + clean up some of the commands for make distcheck + +2005-09-15 Thomas Vander Stichele + + * gtk-doc-plugins.mak: + * mangle-tmpl.py: + first stab at reorganizing the plugins build so we can maintain + element docs + +2005-09-14 David Schleef + + * as-libtool.mak: Remove + * m4/as-libtool.m4: The libtool bug that this worked around has + been fixed. + * m4/as-version.m4: Don't define GST_RELEASE, since it causes + config.h to be regenerated needlessly, and we don't use it. + +2005-09-14 Thomas Vander Stichele + + * gtk-doc-plugins.mak: + error out on inspect failure + +2005-09-14 Michael Smith + + * glib-gen.mak: + Don't call glib-mkenums with arguments that confuse/break MinGW, + fixes 316155. + +2005-09-03 Thomas Vander Stichele + + * gtk-doc-plugins.mak: + * gtk-doc.mak: + * m4/gst-doc.m4: + separate out gtk-doc and docbook stuff + have two separate --enable configure flags + +2005-08-26 Thomas Vander Stichele + + * check.mak: + add a .gdb target; rebuild registry for each target, otherwise + a code rebuild always triggers a reg rebuild, and it's just too + annoying + * gstdoc-scangobj: + +2005-08-21 Thomas Vander Stichele + + * check.mak: + separate out REGISTRY_ENVIRONMENT; we want to use that from + our valgrind runs, but we also want TESTS_ENVIRONMENT to contain + everything that the first test, gst-register, needs + +2005-08-21 Thomas Vander Stichele + + * check.mak: + parse output of valgrind and check for definitely lost, and error + out; somehow I was led to believe valgrind returns non-zero for + leaks, but I can't make it do that, so let's parse + +2005-08-20 Thomas Vander Stichele + + * check.mak: + for some weird reason valgrind does not report actual memleaks + if GST_PLUGIN_PATH is set to anything but the core gstreamer dir + while valgrind is running. Since the registry is going to go + anyway, I don't want to waste any more time on this; I just run + valgrind without GST_PLUGIN_PATH set. Since the registry loading + doesn't check if GST_PLUGIN_PATH got changed as a reason to rebuild + the registry, that's actually fine. + +2005-08-15 Thomas Vander Stichele + + * mangle-tmpl.py: + keep original Long_Description; only insert an include if it's + not already the first line in there + * plugins.xsl: + output more information for plugins, including an origin hyperlink + +2005-08-15 Thomas Vander Stichele + + * gst-xmlinspect.py: + a first stab at inspecting plugins and outputting an xml description + * gtk-doc-plugins.mak: + a gtk-doc using snippet for plugins documentation + * plugins.xsl: + a stylesheet to convert gst-xmlinspect.py output to docbook output + for inclusion in the gtk-doc stuff + +2005-07-20 Ronald S. Bultje + + * m4/gst-doc.m4: + s/pdf/eps/ in test for whether we output EPS images (#309379). + +2005-07-18 Andy Wingo + + * m4/as-libtool-tags.m4: Ooh, backported from libtool 1.6. Much + better. Thanks, Paolo Bonzini! + + * m4/Makefile.am (EXTRA_DIST): + * m4/as-libtool-tags.m4: New file, tries to disable some CXX and + fortran checks. + +2005-07-08 Thomas Vander Stichele + + * m4/gst-error.m4: + add macro to set ERROR_CFLAGS + +2005-06-30 Jan Schmidt + + * gst-autogen.sh: + Remove the old autoregen.sh if it exists before recreating it, + to prevent confusing any shell process that might be reading it + currently. + +2005-06-29 Thomas Vander Stichele + + * m4/gtk-doc.m4: + added + +2005-06-03 Stefan Kost + + * gst-autogen.sh: create autoregen.sh *before* shifting the options + +2005-05-17 Thomas Vander Stichele + + * gst-autogen.sh: only update autoregen.sh on actual runs + +2005-03-11 Thomas Vander Stichele + + * m4/check.m4: m4 from the check unit test suite + +2004-12-14 David Schleef + + * m4/gst-arch.m4: remove MMX stuff, since it doesn't work and + isn't needed anywhere + +2004-12-08 Thomas Vander Stichele + + * gst-autogen.sh: + allow failure command to be run so we can clean upfrom autopoint + +2004-09-03 Zeeshan Ali Khattak + * m4/gst-feature.m4: Trying to correct the GST_CHECK_CONFIGPROG macro + +2004-07-21 Benjamin Otte + + * m4/.cvsignore: exciting updates for libtool m4 files + +2004-07-12 David Schleef + + * m4/as-objc.m4: Add a macro to test for objective C + +2004-06-12 Thomas Vander Stichele + + * m4/gst-feature.m4: + not all of them support --plugin-libs, so redirect stderr + +2004-06-12 Thomas Vander Stichele + + * m4/as-scrub-include.m4: + sync with upstream to 0.1.4. Fixes #132440 + +2004-06-07 Benjamin Otte + + * m4/gst-feature.m4: + write a big marker into configure output when checking next plugin + to allow easier parsing of why plugins are(n't) built. + +2004-06-01 Thomas Vander Stichele + + * m4/as-compiler-flag.m4: + * m4/as-compiler.m4: + * m4/as-libtool.m4: + * m4/as-version.m4: + sync with upstream, change sticky options to -ko + +2004-05-24 Thomas Vander Stichele + + * m4/as-scrub-include.m4: synced with upstream + +2004-05-03 Thomas Vander Stichele + + * po.mak: + snippet for updating .po files + +2004-03-18 Thomas Vander Stichele + + * Makefile.am: + * m4/Makefile.am: + integrate these with the dist + +2004-03-17 Thomas Vander Stichele + + * release.mak: add a release target + +2004-03-09 Thomas Vander Stichele + + patch by: Stephane Loeuillet + + * m4/ax_create_stdint_h.m4: + use head -n instead of head - (#136500) + +2004-03-05 Thomas Vander Stichele + + * m4/gst-doc.m4: don't build PS without dvips binary + +2004-02-22 Julio M. Merino Vidal + + reviewed by: Benjamin Otte + + * m4/as-docbook.m4: + don't use == operator with test(1) (fixes #135115) + +2004-02-16 Thomas Vander Stichele + + * common/m4/gst-arch.m4: x86_64 is x86 too (clue from Fedora 2 test) + +2004-02-13 Thomas Vander Stichele + + * m4/gst-feature.m4: + remove AM_CONDITIONAL for the subsystem since automake 1.6.x + requires that call be in configure.ac + +2004-02-13 Thomas Vander Stichele + + * m4/gst-libxml2.m4: + take required version as argument, and default to 2.4.9 if not + specified + +2004-02-12 Thomas Vander Stichele + + * m4/gst-feature.m4: + rename and fix up GST_CHECK_DISABLE_SUBSYSTEM + +2004-02-11 Thomas Vander Stichele + + * common/m4/as-ac-expand.m4: + * common/m4/as-auto-alt.m4: + * common/m4/as-compiler-flag.m4: + * common/m4/as-compiler.m4: + * common/m4/as-docbook.m4: + * common/m4/as-libtool.m4: + * common/m4/as-scrub-include.m4: + * common/m4/as-version.m4: + * common/m4/glib-gettext.m4: + * common/m4/gst-arch.m4: + * common/m4/gst-debuginfo.m4: + * common/m4/gst-doc.m4: + * common/m4/gst-feature.m4: + * common/m4/gst-function.m4: + * common/m4/gst-glib2.m4: + * common/m4/gst-gstreamer.m4: + * common/m4/gst-libxml2.m4: + * common/m4/gst-makecontext.m4: + * common/m4/gst-mcsc.m4: + * common/m4/pkg.m4: + fix underquoted macros as reported by automake 1.8.x (#133800) + +2004-02-11 Johan Dahlin + + * gst-autogen.sh: Use A-Z instead of A-z in sed expression to + avoid a warning + +2004-02-05 Thomas Vander Stichele + + * m4/gst-doc.m4: + we use --output-format=xml and --ingnore-files options to + gtkdoc-mkdb, which got added between 0.9 and 1.0 + +2004-02-04 Thomas Vander Stichele + + * m4/as-libtool.m4: remove AM_PROG_LIBTOOL so it can move back + to configure.ac to shut up libtoolize + +2004-02-03 Thomas Vander Stichele + + * glib-gen.mak: added; used to generate enums and marshal code + +2004-01-13 Thomas Vander Stichele + + * gettext.patch: added; used by autogen.sh to make sure + GETTEXT_PACKAGE is understood from po/Makefile.in.in -> po/Makefile.in + diff --git a/common/Makefile.am b/common/Makefile.am new file mode 100644 index 0000000..25966fc --- /dev/null +++ b/common/Makefile.am @@ -0,0 +1,22 @@ +SUBDIRS = m4 + +EXTRA_DIST = \ + ChangeLog \ + gettext.patch \ + glib-gen.mak gtk-doc.mak upload-doc.mak \ + cruft.mak release.mak win32.mak po.mak \ + parallel-subdirs.mak \ + gst-autogen.sh \ + check-exports \ + c-to-xml.py mangle-tmpl.py scangobj-merge.py \ + gtk-doc-plugins.mak \ + plugins.xsl gstdoc-scangobj \ + gst.supp check.mak \ + coverage/lcov.mak \ + coverage/coverage-report.pl \ + coverage/coverage-report.xsl \ + coverage/coverage-report-entry.pl \ + download-translations \ + extract-release-date-from-doap-file \ + gst-indent \ + orc.mak diff --git a/common/Makefile.in b/common/Makefile.in new file mode 100644 index 0000000..b4465d3 --- /dev/null +++ b/common/Makefile.in @@ -0,0 +1,756 @@ +# Makefile.in generated by automake 1.14.1 from Makefile.am. +# @configure_input@ + +# Copyright (C) 1994-2013 Free Software Foundation, Inc. + +# This Makefile.in is free software; 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/dev/null +++ b/common/README @@ -0,0 +1,255 @@ +GStreamer @SERIES_VERSION@ + +WHAT IT IS +---------- + +This is GStreamer, a framework for streaming media. + +WHERE TO START +-------------- + +We have a website at +http://gstreamer.freedesktop.org/ + +You should start by going through our FAQ at +http://gstreamer.freedesktop.org/data/doc/gstreamer/head/faq/html/ + +There is more documentation; go to +http://gstreamer.freedesktop.org/documentation + +You can subscribe to our mailing lists; see the website for details. + +We track bugs in GNOME's bugzilla; see the website for details. + +You can join us on IRC - #gstreamer on irc.freenode.org + +GStreamer 1.0 series +-------------------- + +Starring + + GSTREAMER + +The core around which all other modules revolve. Base functionality and +libraries, some essential elements, documentation, and testing. + + BASE + +A well-groomed and well-maintained collection of GStreamer plug-ins and +elements, spanning the range of possible types of elements one would want +to write for GStreamer. + +And introducing, for the first time ever, on the development screen ... + + THE GOOD + + --- "Such ingratitude. After all the times I've saved your life." + +A collection of plug-ins you'd want to have right next to you on the +battlefield. Shooting sharp and making no mistakes, these plug-ins have it +all: good looks, good code, and good licensing. Documented and dressed up +in tests. If you're looking for a role model to base your own plug-in on, +here it is. + +If you find a plot hole or a badly lip-synced line of code in them, +let us know - it is a matter of honour for us to ensure Blondie doesn't look +like he's been walking 100 miles through the desert without water. + + THE UGLY + + --- "When you have to shoot, shoot. Don't talk." + +There are times when the world needs a color between black and white. +Quality code to match the good's, but two-timing, backstabbing and ready to +sell your freedom down the river. These plug-ins might have a patent noose +around their neck, or a lock-up license, or any other problem that makes you +think twice about shipping them. + +We don't call them ugly because we like them less. Does a mother love her +son less because he's not as pretty as the other ones ? No - she commends +him on his great personality. These plug-ins are the life of the party. +And we'll still step in and set them straight if you report any unacceptable +behaviour - because there are two kinds of people in the world, my friend: +those with a rope around their neck and the people who do the cutting. + + THE BAD + + --- "That an accusation?" + +No perfectly groomed moustache or any amount of fine clothing is going to +cover up the truth - these plug-ins are Bad with a capital B. +They look fine on the outside, and might even appear to get the job done, but +at the end of the day they're a black sheep. Without a golden-haired angel +to watch over them, they'll probably land in an unmarked grave at the final +showdown. + +Don't bug us about their quality - exercise your Free Software rights, +patch up the offender and send us the patch on the fastest steed you can +steal from the Confederates. Because you see, in this world, there's two +kinds of people, my friend: those with loaded guns and those who dig. +You dig. + +The Lowdown +----------- + + --- "I've never seen so many plug-ins wasted so badly." + +GStreamer Plug-ins has grown so big that it's hard to separate the wheat from +the chaff. Also, distributors have brought up issues about the legal status +of some of the plug-ins we ship. To remedy this, we've divided the previous +set of available plug-ins into four modules: + +- gst-plugins-base: a small and fixed set of plug-ins, covering a wide range + of possible types of elements; these are continuously kept up-to-date + with any core changes during the development series. + + - We believe distributors can safely ship these plug-ins. + - People writing elements should base their code on these elements. + - These elements come with examples, documentation, and regression tests. + +- gst-plugins-good: a set of plug-ins that we consider to have good quality + code, correct functionality, our preferred license (LGPL for the plug-in + code, LGPL or LGPL-compatible for the supporting library). + + - We believe distributors can safely ship these plug-ins. + - People writing elements should base their code on these elements. + +- gst-plugins-ugly: a set of plug-ins that have good quality and correct + functionality, but distributing them might pose problems. The license + on either the plug-ins or the supporting libraries might not be how we'd + like. The code might be widely known to present patent problems. + + - Distributors should check if they want/can ship these plug-ins. + - People writing elements should base their code on these elements. + +- gst-plugins-bad: a set of plug-ins that aren't up to par compared to the + rest. They might be close to being good quality, but they're missing + something - be it a good code review, some documentation, a set of tests, + a real live maintainer, or some actual wide use. + If the blanks are filled in they might be upgraded to become part of + either gst-plugins-good or gst-plugins-ugly, depending on the other factors. + + - If the plug-ins break, you can't complain - instead, you can fix the + problem and send us a patch, or bribe someone into fixing them for you. + - New contributors can start here for things to work on. + +PLATFORMS +--------- + +- Linux is of course fully supported +- FreeBSD is reported to work; other BSDs should work too +- Solaris is reported to work; a specific sunaudiosink plugin has been written +- MacOSX works, binary 1.x packages can be built using the cerbero build tool +- Windows works; binary 1.x packages can be built using the cerbero build tool + - MSys/MinGW builds + - Microsoft Visual Studio builds are not yet available or supported +- Android works, binary 1.x packages can be built using the cerbero build tool +- iOS works + +INSTALLING FROM PACKAGES +------------------------ + +You should always prefer installing from packages first. GStreamer is +well-maintained for a number of distributions, including Fedora, Debian, +Ubuntu, Mandrake, Gentoo, ... + +Only in cases where you: +- want to hack on GStreamer +- want to verify that a bug has been fixed +- do not have a sane distribution +should you choose to build from source tarballs or git. + +Find more information about the various packages at +http://gstreamer.freedesktop.org/download/ + +COMPILING FROM SOURCE TARBALLS +------------------------------ + +- again, make sure that you really need to install from source ! + If GStreamer is one of your first projects ever that you build from source, + consider taking on an easier project. + +- check output of ./configure --help to see if any options apply to you +- run + ./configure + make + + to build GStreamer. +- if you want to install it (not required, but what you usually want to do), run + make install + +- try out a simple test: + gst-launch -v fakesrc num_buffers=5 ! fakesink + (If you didn't install GStreamer, prefix gst-launch with tools/) + + If it outputs a bunch of messages from fakesrc and fakesink, everything is + ok. + + If it did not work, keep in mind that you might need to adjust the + PATH and/or LD_LIBRARY_PATH environment variables to make the system + find GStreamer in the prefix where you installed (by default that is /usr/local). + +- After this, you're ready to install gst-plugins, which will provide the + functionality you're probably looking for by now, so go on and read + that README. + +COMPILING FROM GIT +------------------ + +When building from git sources, you will need to run autogen.sh to generate +the build system files. + +You will need a set of additional tools typical for building from git, +including: +- autoconf +- automake +- libtool + +autogen.sh will check for recent enough versions and complain if you don't have +them. You can also specify specific versions of automake and autoconf with +--with-automake and --with-autoconf + +Check autogen.sh options by running autogen.sh --help + +autogen.sh can pass on arguments to configure + +When you have done this once, you can use autoregen.sh to re-autogen with +the last passed options as a handy shortcut. Use it. + +After the autogen.sh stage, you can follow the directions listed in +"COMPILING FROM SOURCE" + +You can also run your whole git stack uninstalled in your home directory, +so that you can quickly test changes without affecting your system setup or +interfering with GStreamer installed from packages. Many GStreamer developers +use an uninstalled setup for their work. + +There is a 'create-uninstalled-setup.sh' script in + + http://cgit.freedesktop.org/gstreamer/gstreamer/tree/scripts/ + +to easily create an uninstalled setup from scratch. + + +PLUG-IN DEPENDENCIES AND LICENSES +--------------------------------- + +GStreamer is developed under the terms of the LGPL (see LICENSE file for +details). Some of our plug-ins however rely on libraries which are available +under other licenses. This means that if you are distributing an application +which has a non-GPL compatible license (for instance a closed-source +application) with GStreamer, you have to make sure not to distribute GPL-linked +plug-ins. + +When using GPL-linked plug-ins, GStreamer is for all practical reasons +under the GPL itself. + +HISTORY +------- + +The fundamental design comes from the video pipeline at Oregon Graduate +Institute, as well as some ideas from DirectMedia. It's based on plug-ins that +will provide the various codec and other functionality. The interface +hopefully is generic enough for various companies (ahem, Apple) to release +binary codecs for Linux, until such time as they get a clue and release the +source. diff --git a/common/c-to-xml.py b/common/c-to-xml.py new file mode 100644 index 0000000..7a7a35b --- /dev/null +++ b/common/c-to-xml.py @@ -0,0 +1,36 @@ +# -*- Mode: Python -*- +# vi:si:et:sw=4:sts=4:ts=4 + +""" +Convert a C program to valid XML to be included in docbook +""" + +from __future__ import print_function, unicode_literals + +import sys +import os +from xml.sax import saxutils + +def main(): + if len(sys.argv) == 1: + sys.stderr.write("Please specify a source file to convert") + sys.exit(1) + source = sys.argv[1] + + if not os.path.exists(source): + sys.stderr.write("%s does not exist.\n" % source) + sys.exit(1) + + content = open(source, "r").read() + + # print header + print ('') + print ('') + print () + print ('') + + # print content + print (saxutils.escape(content)) + print ('') + +main() diff --git a/common/check-exports b/common/check-exports new file mode 100755 index 0000000..ee01ff7 --- /dev/null +++ b/common/check-exports @@ -0,0 +1,54 @@ +#!/bin/sh +# check-exports +# +# quick'n'dirty script that retrieves the list of exported symbols of a given +# library using 'nm', and compares that against the list of symbols-to-export +# of our win32/common/libfoo.def files. + +if [ $# -ne 2 ]; then + echo "Usage: $0 library.def library.so" + exit 1 +fi + +def_path="$1" +def_name="$(basename $def_path)" +lib_path="$2" + +lib_result="`mktemp /tmp/defname.XXXXXX`" + +LC_ALL=C +export LC_ALL + +# On Solaris, add -p to get the correct output format +NMARGS= +if nm -V 2>&1 |grep Solaris > /dev/null; then + NMARGS=-p +fi + +# _end is special cased because for some reason it is reported as an exported +# BSS symbol, unlike on linux where it's a local absolute symbol. +nm $NMARGS $lib_path | awk \ + '{ + if ($3 ~ /^[_]?(gst_|Gst|GST_).*/) + { + if ($2 ~ /^[BSDG]$/) + print "\t" $3 " DATA" + else if ($2 == "T") + print "\t" $3 + } + }' | sort | awk '{ if (NR == 1) print "EXPORTS"; print $0; }' \ + > $lib_result + +diffoutput=`diff -u $def_path $lib_result` +diffresult=$? + +rm $lib_result + +if test "$diffresult" -eq 0; then + exit 0; +else + echo -n "$diffoutput" >&2 + echo >&2 + exit 1; +fi + diff --git a/common/check.mak b/common/check.mak new file mode 100644 index 0000000..611546a --- /dev/null +++ b/common/check.mak @@ -0,0 +1,252 @@ +# keep target around, since it's referenced in the modules' Makefiles +clean-local-check: + @echo + +if HAVE_VALGRIND +# hangs spectacularly on some machines, so let's not do this by default yet +check-valgrind: + $(MAKE) valgrind +else +check-valgrind: + @true +endif + +LOOPS ?= 10 + +# run any given test by running make test.check +# if the test fails, run it again at at least debug level 2 +%.check: % + @$(TESTS_ENVIRONMENT) \ + CK_DEFAULT_TIMEOUT=20 \ + $* || \ + $(TESTS_ENVIRONMENT) \ + GST_DEBUG=$$GST_DEBUG,*:2 \ + CK_DEFAULT_TIMEOUT=20 \ + $* + +# just like 'check', but don't run it again if it fails (useful for debugging) +%.check-norepeat: % + @$(TESTS_ENVIRONMENT) \ + CK_DEFAULT_TIMEOUT=20 \ + $* + +# run any given test in a loop +%.torture: % + @for i in `seq 1 $(LOOPS)`; do \ + $(TESTS_ENVIRONMENT) \ + CK_DEFAULT_TIMEOUT=20 \ + $*; done + +# run any given test in an infinite loop +%.forever: % + @while true; do \ + $(TESTS_ENVIRONMENT) \ + CK_DEFAULT_TIMEOUT=20 \ + $* || break; done + +# valgrind any given test by running make test.valgrind +%.valgrind: % + @$(TESTS_ENVIRONMENT) \ + CK_DEFAULT_TIMEOUT=360 \ + G_SLICE=always-malloc \ + $(LIBTOOL) --mode=execute \ + $(VALGRIND_PATH) -q \ + $(foreach s,$(SUPPRESSIONS),--suppressions=$(s)) \ + --tool=memcheck --leak-check=full --trace-children=yes \ + --show-possibly-lost=no \ + --leak-resolution=high --num-callers=20 \ + ./$* 2>&1 | tee valgrind.log + @if grep "==" valgrind.log > /dev/null 2>&1; then \ + rm valgrind.log; \ + exit 1; \ + fi + @rm valgrind.log + +# valgrind any given test and generate suppressions for it +%.valgrind.gen-suppressions: % + @$(TESTS_ENVIRONMENT) \ + CK_DEFAULT_TIMEOUT=360 \ + G_SLICE=always-malloc \ + $(LIBTOOL) --mode=execute \ + $(VALGRIND_PATH) -q \ + $(foreach s,$(SUPPRESSIONS),--suppressions=$(s)) \ + --tool=memcheck --leak-check=full --trace-children=yes \ + --show-possibly-lost=no \ + --leak-resolution=high --num-callers=20 \ + --gen-suppressions=all \ + ./$* 2>&1 | tee suppressions.log + +# valgrind torture any given test +%.valgrind-torture: % + @for i in `seq 1 $(LOOPS)`; do \ + $(MAKE) $*.valgrind || \ + (echo "Failure after $$i runs"; exit 1) || \ + exit 1; \ + done + @banner="All $(LOOPS) loops passed"; \ + dashes=`echo "$$banner" | sed s/./=/g`; \ + echo $$dashes; echo $$banner; echo $$dashes + +# valgrind any given test until failure by running make test.valgrind-forever +%.valgrind-forever: % + @while $(MAKE) $*.valgrind; do \ + true; done + +# gdb any given test by running make test.gdb +%.gdb: % + @$(TESTS_ENVIRONMENT) \ + CK_FORK=no \ + $(LIBTOOL) --mode=execute \ + gdb $* + +%.lcov-reset: + $(MAKE) $*.lcov-run + $(MAKE) $*.lcov-report + +%.lcov: % + $(MAKE) $*.lcov-reset + +if GST_GCOV_ENABLED +%.lcov-clean: + $(MAKE) -C $(top_builddir) lcov-clean + +%.lcov-run: + $(MAKE) $*.lcov-clean + $(MAKE) $*.check + +%.lcov-report: + $(MAKE) -C $(top_builddir) lcov-report +else +%.lcov-run: + echo "Need to reconfigure with --enable-gcov" + +%.lcov-report: + echo "Need to reconfigure with --enable-gcov" +endif + +# torture tests +torture: $(TESTS) + -rm test-registry.* + @echo "Torturing tests ..." + @for i in `seq 1 $(LOOPS)`; do \ + $(MAKE) check || \ + (echo "Failure after $$i runs"; exit 1) || \ + exit 1; \ + done + @banner="All $(LOOPS) loops passed"; \ + dashes=`echo "$$banner" | sed s/./=/g`; \ + echo $$dashes; echo $$banner; echo $$dashes + +# forever tests +forever: $(TESTS) + -rm test-registry.* + @echo "Forever tests ..." + @while true; do \ + $(MAKE) check || \ + (echo "Failure"; exit 1) || \ + exit 1; \ + done + +# valgrind all tests +valgrind: $(TESTS) + @echo "Valgrinding tests ..." + @failed=0; \ + for t in $(filter-out $(VALGRIND_TESTS_DISABLE),$(TESTS)); do \ + $(MAKE) $$t.valgrind; \ + if test "$$?" -ne 0; then \ + echo "Valgrind error for test $$t"; \ + failed=`expr $$failed + 1`; \ + whicht="$$whicht $$t"; \ + fi; \ + done; \ + if test "$$failed" -ne 0; then \ + echo "$$failed tests had leaks or errors under valgrind:"; \ + echo "$$whicht"; \ + false; \ + fi + +# valgrind all tests until failure +valgrind-forever: $(TESTS) + -rm test-registry.* + @echo "Forever valgrinding tests ..." + @while true; do \ + $(MAKE) valgrind || \ + (echo "Failure"; exit 1) || \ + exit 1; \ + done + +# valgrind torture all tests +valgrind-torture: $(TESTS) + -rm test-registry.* + @echo "Torturing and valgrinding tests ..." + @for i in `seq 1 $(LOOPS)`; do \ + $(MAKE) valgrind || \ + (echo "Failure after $$i runs"; exit 1) || \ + exit 1; \ + done + @banner="All $(LOOPS) loops passed"; \ + dashes=`echo "$$banner" | sed s/./=/g`; \ + echo $$dashes; echo $$banner; echo $$dashes + +# valgrind all tests and generate suppressions +valgrind.gen-suppressions: $(TESTS) + @echo "Valgrinding tests ..." + @failed=0; \ + for t in $(filter-out $(VALGRIND_TESTS_DISABLE),$(TESTS)); do \ + $(MAKE) $$t.valgrind.gen-suppressions; \ + if test "$$?" -ne 0; then \ + echo "Valgrind error for test $$t"; \ + failed=`expr $$failed + 1`; \ + whicht="$$whicht $$t"; \ + fi; \ + done; \ + if test "$$failed" -ne 0; then \ + echo "$$failed tests had leaks or errors under valgrind:"; \ + echo "$$whicht"; \ + false; \ + fi + +# inspect every plugin feature +GST_INSPECT = $(GST_TOOLS_DIR)/gst-inspect-$(GST_API_VERSION) +inspect: + @echo "Inspecting features ..." + @for e in `$(TESTS_ENVIRONMENT) $(GST_INSPECT) | head -n -2 \ + | cut -d: -f2`; \ + do echo Inspecting $$e; \ + $(GST_INSPECT) $$e > /dev/null 2>&1; done + +help: + @echo + @echo "make check -- run all checks" + @echo "make torture -- run all checks $(LOOPS) times" + @echo "make (dir)/(test).check -- run the given check once, repeat with GST_DEBUG=*:2 if it fails" + @echo "make (dir)/(test).check-norepeat -- run the given check once, but don't run it again if it fails" + @echo "make (dir)/(test).forever -- run the given check forever" + @echo "make (dir)/(test).torture -- run the given check $(LOOPS) times" + @echo + @echo "make (dir)/(test).gdb -- start up gdb for the given test" + @echo + @echo "make valgrind -- valgrind all tests" + @echo "make valgrind-forever -- valgrind all tests forever" + @echo "make valgrind-torture -- valgrind all tests $(LOOPS) times" + @echo "make valgrind.gen-suppressions -- generate suppressions for all tests" + @echo " and save to suppressions.log" + @echo "make (dir)/(test).valgrind -- valgrind the given test" + @echo "make (dir)/(test).valgrind-forever -- valgrind the given test forever" + @echo "make (dir)/(test).valgrind-torture -- valgrind the given test $(LOOPS) times" + @echo "make (dir)/(test).valgrind.gen-suppressions -- generate suppressions" + @echo " and save to suppressions.log" + @echo "make inspect -- inspect all plugin features" + @echo + @echo + @echo "Additionally, you can use the GST_CHECKS environment variable to" + @echo "specify which test(s) should be run. This is useful if you are" + @echo "debugging a failure in one particular test, or want to reproduce" + @echo "a race condition in a single test." + @echo + @echo "Examples:" + @echo + @echo " GST_CHECKS=test_this,test_that make element/foobar.check" + @echo " GST_CHECKS=test_many_threads make element/foobar.forever" + @echo + diff --git a/common/coverage/coverage-report-entry.pl b/common/coverage/coverage-report-entry.pl new file mode 100644 index 0000000..d0036b3 --- /dev/null +++ b/common/coverage/coverage-report-entry.pl @@ -0,0 +1,69 @@ +#!/usr/bin/perl +# +# Copyright (C) 2006 Daniel Berrange +# +# This program is free software; you can redistribute it and/or modify +# it under the terms of the GNU General Public License as published by +# the Free Software Foundation; either version 2 of the License, or +# (at your option) any later version. +# +# This program is distributed in the hope that it will be useful, +# but WITHOUT ANY WARRANTY; without even the implied warranty of +# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +# GNU General Public License for more details. +# +# You should have received a copy of the GNU General Public License +# along with this program; if not, write to the Free Software +# Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA + + +print < + +Coverage report for $ARGV[0] + + + +

Coverage report for $ARGV[0]

+ +
+EOF
+
+
+while (<>) {
+    s/&/&/g;
+    s//>/g;
+
+    if (/^\s*function (\S+) called (\d+) returned \d+% blocks executed \d+%/) {
+	my $class = $2 > 0 ? "perfect" : "terrible";
+	$_ = "$_";
+    } elsif (/^\s*branch\s+\d+\s+taken\s+(\d+)%\s+.*$/) {
+	my $class = $1 > 0 ? "perfect" : "terrible";
+	$_ = "$_";
+    } elsif (/^\s*branch\s+\d+\s+never executed.*$/) {
+	my $class = "terrible";
+	$_ = "$_";
+    } elsif (/^\s*call\s+\d+\s+never executed.*$/) {
+	my $class = "terrible";
+	$_ = "$_";
+    } elsif (/^\s*call\s+\d+\s+returned\s+(\d+)%.*$/) {
+	my $class = $1 > 0 ? "perfect" : "terrible";
+	$_ = "$_";
+    }
+
+    print;
+}
+
+print <
+
+
+EOF
diff --git a/common/coverage/coverage-report.pl b/common/coverage/coverage-report.pl
new file mode 100644
index 0000000..18bd6f1
--- /dev/null
+++ b/common/coverage/coverage-report.pl
@@ -0,0 +1,125 @@
+#!/usr/bin/perl
+#
+# Copyright (C) 2006 Daniel Berrange
+#
+# This program is free software; you can redistribute it and/or modify
+# it under the terms of the GNU General Public License as published by
+# the Free Software Foundation; either version 2 of the License, or
+# (at your option) any later version.
+#
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+# GNU General Public License for more details.
+#
+# You should have received a copy of the GNU General Public License
+# along with this program; if not, write to the Free Software
+# Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA  02110-1301  USA
+
+use warnings;
+use strict;
+
+my %coverage = ( functions => {}, files => {} );
+
+my %filemap;
+
+my $type;
+my $name;
+
+my @functions;
+
+while (<>) {
+    if (/^Function '(.*)'\s*$/) {
+	$type = "function";
+	$name = $1;
+	$coverage{$type}->{$name} = {};
+	push @functions, $name;
+    } elsif (/^File '(.*?)'\s*$/) {
+	$type = "file";
+	$name = $1;
+	$coverage{$type}->{$name} = {};
+
+	foreach my $func (@functions) {
+	    $coverage{"function"}->{$func}->{file} = $name;
+	}
+	@functions = ();
+    } elsif (/^Lines executed:(.*)%\s*of\s*(\d+)\s*$/) {
+	$coverage{$type}->{$name}->{lines} = $2;
+	$coverage{$type}->{$name}->{linesCoverage} = $1;
+    } elsif (/^Branches executed:(.*)%\s*of\s*(\d+)\s*$/) {
+	$coverage{$type}->{$name}->{branches} = $2;
+	$coverage{$type}->{$name}->{branchesCoverage} = $1;
+    } elsif (/^Taken at least once:(.*)%\s*of\s*(\d+)\s*$/) {
+	$coverage{$type}->{$name}->{conds} = $2;
+	$coverage{$type}->{$name}->{condsCoverage} = $1;
+    } elsif (/^Calls executed:(.*)%\s*of\s*(\d+)\s*$/) {
+	$coverage{$type}->{$name}->{calls} = $2;
+	$coverage{$type}->{$name}->{callsCoverage} = $1;
+    } elsif (/^No branches$/) {
+	$coverage{$type}->{$name}->{branches} = 0;
+	$coverage{$type}->{$name}->{branchesCoverage} = "100.00";
+	$coverage{$type}->{$name}->{conds} = 0;
+	$coverage{$type}->{$name}->{condsCoverage} = "100.00";
+    } elsif (/^No calls$/) {
+	$coverage{$type}->{$name}->{calls} = 0;
+	$coverage{$type}->{$name}->{callsCoverage} = "100.00";
+    } elsif (/^\s*(.*):creating '(.*)'\s*$/) {
+	$filemap{$1} = $2;
+    } elsif (/^\s*$/) {
+	# nada
+    } else {
+	warn "Shit [$_]\n";
+    }
+}
+
+my %summary;
+foreach my $type ("function", "file") {
+    $summary{$type} = {};
+    foreach my $m ("lines", "branches", "conds", "calls") {
+	my $totalGot = 0;
+	my $totalMiss = 0;
+	my $count = 0;
+	foreach my $func (keys %{$coverage{function}}) {
+	    $count++;
+	    my $got = $coverage{function}->{$func}->{$m};
+	    $totalGot += $got;
+	    my $miss = $got * $coverage{function}->{$func}->{$m ."Coverage"} / 100;
+	    $totalMiss += $miss;
+	}
+	$summary{$type}->{$m} = sprintf("%d", $totalGot);
+	$summary{$type}->{$m . "Coverage"} = sprintf("%.2f", $totalMiss / $totalGot * 100);
+    }
+}
+
+
+
+print "\n";
+
+foreach my $type ("function", "file") {
+    printf "<%ss>\n", $type;
+    foreach my $name (sort { $a cmp $b } keys %{$coverage{$type}}) {
+	my $rec = $coverage{$type}->{$name};
+	printf "  \n", $name, ($type eq "file" ? $filemap{$name} : $filemap{$rec->{file}});
+	printf "    \n", $rec->{lines}, $rec->{linesCoverage};
+	if (exists $rec->{branches}) {
+	    printf "    \n", $rec->{branches}, $rec->{branchesCoverage};
+	}
+	if (exists $rec->{conds}) {
+	    printf "    \n", $rec->{conds}, $rec->{condsCoverage};
+	}
+	if (exists $rec->{calls}) {
+	    printf "    \n", $rec->{calls}, $rec->{callsCoverage};
+	}
+	print  "  \n";
+    }
+
+    printf "  \n";
+    printf "    \n", $summary{$type}->{lines}, $summary{$type}->{linesCoverage};
+    printf "    \n", $summary{$type}->{branches}, $summary{$type}->{branchesCoverage};
+    printf "    \n", $summary{$type}->{conds}, $summary{$type}->{condsCoverage};
+    printf "    \n", $summary{$type}->{calls}, $summary{$type}->{callsCoverage};
+    printf  "  \n";
+    printf "\n", $type;
+}
+
+print "\n";
diff --git a/common/coverage/coverage-report.xsl b/common/coverage/coverage-report.xsl
new file mode 100644
index 0000000..3fe124b
--- /dev/null
+++ b/common/coverage/coverage-report.xsl
@@ -0,0 +1,235 @@
+
+
+
+
+  
+
+  
+    
+      
+        Coverage report
+        
+      
+      
+        
Coverage report

+        
+      
+    
+  
+
+  
+    
Function coverage

+    
+      
+    
+  
+  
+
+  
+    
File coverage

+    
+      
+    
+  
+
+  
+    
+    
+      
+        
+          
+          
+          
+          
+          
+        
+      
+      
+        
+          
+            
+            
+              
+                
+                  odd
+                
+                
+                  even
+                
+              
+            
+          
+        
+      
+      
+        
+          
+            
+            
+              
+                
+                  odd
+                
+                
+                  even
+                
+              
+            
+          
+        
+      
+    
NameLinesBranchesConditionsCalls

+  
+  
+  
+    
+    
+    
+      
+        
+          
+        
+        
+          
+        
+        
+          Summary
+        
+      
+
+      
+        
+      
+      
+        
+      
+
+      
+        
+      
+      
+        
+      
+
+      
+        
+      
+      
+        
+      
+
+      
+        
+      
+      
+        
+      
+
+    
+  
+  
+  
+    
+  
+
+  
+    
+  
+
+  
+    
+  
+
+  
+    
+  
+
+  
+    
+  
+
+  
+    
+      
+        
+          perfect
+        
+        
+          excellant
+        
+        
+          good
+        
+        
+          poor
+        
+        
+          bad
+        
+        
+          terrible
+        
+      
+    
+    
+    % of 
+  
+
+
diff --git a/common/coverage/lcov.mak b/common/coverage/lcov.mak
new file mode 100644
index 0000000..b45728a
--- /dev/null
+++ b/common/coverage/lcov.mak
@@ -0,0 +1,47 @@
+## .PHONY so it always rebuilds it
+.PHONY: lcov-reset lcov lcov-run lcov-report lcov-upload lcov-clean
+
+# run lcov from scratch, always
+lcov-reset:
+	$(MAKE) lcov-run
+	$(MAKE) lcov-report
+
+# run lcov from scratch if the dir is not there
+lcov:
+	$(MAKE) lcov-reset
+
+if GST_GCOV_ENABLED
+# reset lcov stats
+lcov-clean:
+	@-rm -rf lcov
+	lcov --directory . --zerocounters
+
+# reset run coverage tests
+lcov-run:
+	-$(MAKE) lcov-clean
+	-if test -d tests/check; then $(MAKE) -C tests/check inspect; fi
+	-$(MAKE) check
+
+# generate report based on current coverage data
+lcov-report:
+	mkdir lcov
+	lcov --compat-libtool --directory . --capture --output-file lcov/lcov.info
+	lcov --list-full-path -l lcov/lcov.info | grep -v "`cd $(top_srcdir) && pwd`" | cut -d\| -f1 > lcov/remove
+	lcov --list-full-path -l lcov/lcov.info | grep "tests/check/" | cut -d\| -f1 >> lcov/remove
+	lcov --list-full-path -l lcov/lcov.info | grep "docs/plugins/" | cut -d\| -f1 >> lcov/remove
+	lcov -r lcov/lcov.info `cat lcov/remove` > lcov/lcov.cleaned.info
+	rm lcov/remove
+	mv lcov/lcov.cleaned.info lcov/lcov.info
+	genhtml -t "$(PACKAGE_STRING)" -o lcov --num-spaces 2 lcov/lcov.info
+
+lcov-upload: lcov
+	rsync -rvz -e ssh --delete lcov/* gstreamer.freedesktop.org:/srv/gstreamer.freedesktop.org/www/data/coverage/lcov/$(PACKAGE)
+
+else
+lcov-run:
+	echo "Need to reconfigure with --enable-gcov"
+
+lcov-report:
+	echo "Need to reconfigure with --enable-gcov"
+endif
+
diff --git a/common/cruft.mak b/common/cruft.mak
new file mode 100644
index 0000000..53eec24
--- /dev/null
+++ b/common/cruft.mak
@@ -0,0 +1,56 @@
+# checks for left-over files in the (usually uninstalled) tree, ie. for
+# stuff that best be deleted to avoid problems like having old plugin binaries
+# lying around.
+#
+# set CRUFT_FILES and/or CRUFT_DIRS in your Makefile.am when you include this
+
+check-cruft:
+	@cruft_files=""; cruft_dirs=""; \
+	for f in $(CRUFT_FILES); do \
+	  if test -e $$f; then \
+	    cruft_files="$$cruft_files $$f"; \
+	  fi \
+	done; \
+	for d in $(CRUFT_DIRS); do \
+	  if test -e $$d; then \
+	    cruft_dirs="$$cruft_dirs $$d"; \
+	  fi \
+	done; \
+	if test "x$$cruft_files$$cruft_dirs" != x; then \
+	  echo; \
+	  echo "**** CRUFT ALERT *****"; \
+	  echo; \
+	  echo "The following files and directories may not be needed any "; \
+	  echo "longer (usually because a plugin has been merged into     "; \
+	  echo "another plugin, moved to a different module, or been      "; \
+	  echo "renamed), and you probably want to clean them up if you   "; \
+	  echo "don't have local changes:                                 "; \
+	  echo; \
+	  for f in $$cruft_files; do echo "file $$f"; done; \
+	  echo; \
+	  for d in $$cruft_dirs; do echo "directory $$d"; done; \
+	  echo; \
+	  echo "'make clean-cruft' will remove these for you."; \
+	  echo; \
+	fi
+
+clean-cruft-dirs:
+	@for d in $(CRUFT_DIRS); do \
+	  if test -e $$d; then \
+	    rm -r "$$d" && echo "Removed directory $$d"; \
+	  fi \
+	done
+
+clean-cruft-files:
+	@for f in $(CRUFT_FILES); do \
+	  if test -e $$f; then \
+	    rm "$$f" && echo "Removed file $$f"; \
+	  fi \
+	done
+
+clean-cruft: clean-cruft-dirs clean-cruft-files
+
+# also might want to add this to your Makefile.am:
+#
+# all-local: check-cruft
+
diff --git a/common/download-translations b/common/download-translations
new file mode 100755
index 0000000..aef1d31
--- /dev/null
+++ b/common/download-translations
@@ -0,0 +1,152 @@
+#!/bin/sh
+# Shell script to download the latest translations for a given GStreamer
+# package from translationproject.org
+
+
+# DOMAINS based on http://translationproject.org/extra/matrix.html
+# We need to check all domains, not only po/LINGUAS, since there might be
+# new translations
+DOMAINS=\
+"af am ar az be bg pt_BR bs ca zh_CN cs cy da de el eo es et eu fa fi fr "\
+"ga en_GB gl gu he hi zh_HK hr hu id is it ja ko ku ky lg lt lv mk mn ms "\
+"mt nb ne nl nn or pa pl pt rm ro ru rw sk sl sq sr sv ta tq th tk "\
+"tr zh_TW uk ven vi wa xh zu"
+
+# for testing/debugging:
+#DOMAINS="es fr hu sv pl xx"
+
+# check for 'diff' program
+diff --version 2>/dev/null >/dev/null
+if [ ! $? ]; then
+  echo "==== You must have the 'diff' program installed for this script ===="
+  exit 1
+fi
+
+# check for 'wget' program
+wget --version 2>/dev/null >/dev/null
+if [ ! $? ]; then
+  echo "==== You must have the 'wget' program installed for this script ===="
+  exit 1
+fi
+
+# make sure we're in the top-level directory
+if [ ! -d ./po ]; then
+  echo "==== No ./po directory in the current working directory ===="
+  exit 1
+fi
+
+# make sure a package argument was passed to us
+if [ -z "$1" ]; then
+  echo "Usage: $0 PACKAGE, e.g. $0 gst-plugins-good"
+  exit 1
+fi
+
+if test "$1" != "gstreamer" -a \
+        "$1" != "gst-plugins-base" -a \
+        "$1" != "gst-plugins-good" -a \
+        "$1" != "gst-plugins-ugly" -a \
+        "$1" != "gst-plugins-bad"; then
+  echo "Unexpected package '$1' ?!"
+  exit 1
+fi
+
+PACKAGE="$1"
+
+DOMAINS_TO_ADD=""
+DOMAINS_UPDATED=""
+DOMAINS_NOT_IN_LINGUAS=""
+
+echo "Downloading latest translation files for package $PACKAGE ..."
+echo
+
+for d in $DOMAINS
+do
+  PACKAGE_PO_URL_BASE="http://translationproject.org/latest/$PACKAGE"
+  PO_URL="$PACKAGE_PO_URL_BASE/$d.po"
+  PO_FILENAME="$PACKAGE.$d.po"
+  if wget -q -nc -O $PO_FILENAME $PO_URL; then
+    # we want all .po files in UTF-8 format really, so convert if needed..
+    CHARSET=`grep Content-Type $PO_FILENAME | sed -e 's/.*charset=\(.*\)\\\\n.*/\1/'`
+    if test "x$CHARSET" != "xUTF-8" -a "x$CHARSET" != "xutf-8"; then
+      # note: things like the bugs address will be added back by make update-po
+      if msguniq $PO_FILENAME --no-location \
+                              --output-file=$PO_FILENAME.utf8 \
+                              --to-code=UTF-8; then
+        mv $PO_FILENAME.utf8 $PO_FILENAME
+      else
+        echo "**** $d: conversion from $CHARSET to UTF-8 failed ****"
+      fi
+    fi
+    if [ -f "po/$d.po" ]; then
+      # ./po/foo.po exists, so let's check if ours matches the latest from the
+      # translation project website
+      REVDATE_NEW=`grep PO-Revision-Date $PO_FILENAME`;
+      REVDATE_OLD=`grep PO-Revision-Date po/$d.po`;
+      CHARSET_OLD=`grep Content-Type po/$d.po | sed -e 's/.*charset=\(.*\)\\\\n.*/\1/'`
+      if test "x$REVDATE_NEW" = "x$REVDATE_OLD" -a "x$CHARSET_OLD" = "xUTF-8"; then
+        # note: source code line markers will be removed later by make upload-po
+        echo "$d.po: up-to-date"
+        rm -f $PO_FILENAME
+      else
+        mv $PO_FILENAME "po/$d.po"
+        if test "x$CHARSET_OLD" != "xUTF-8" -a "x$CHARSET_OLD" != "xutf-8"; then
+          echo "$d.po: update (and charset converted from $CHARSET_OLD to UTF-8)"
+        else
+          echo "$d.po: updated"
+        fi
+        DOMAINS_UPDATED="$DOMAINS_UPDATED $d"
+      fi
+      # make sure domain is listed in LINGUAS
+      if ! grep $d "po/LINGUAS" >/dev/null 2>/dev/null; then
+        DOMAINS_NOT_IN_LINGUAS="$DOMAINS_NOT_IN_LINGUAS $d"
+      fi
+    else
+      # ./po/foo.po doesn't exist, but foo.po exists on the translation project
+      # website, so it's probably a new translation
+      echo "$d.po: new language"
+      mv $PO_FILENAME "po/$d.po"
+      DOMAINS_UPDATED="$DOMAINS_UPDATED $d"
+      DOMAINS_TO_ADD="$DOMAINS_TO_ADD $d"
+    fi
+  else
+    rm -f $PO_FILENAME
+    echo "$d.po: failure (does probably not exist)"
+  fi
+done
+
+if [ -n "$DOMAINS_UPDATED" ]; then
+  echo "===================================================================="
+  echo
+  echo "Language domains updated    :$DOMAINS_UPDATED"
+  echo "Language domains to git add :$DOMAINS_TO_ADD"
+  echo
+  echo "Source: http://translationproject.org/latest/$PACKAGE/"
+  echo
+  if [ -n "$DOMAINS_TO_ADD" ]; then
+    CMD_STRING="git add"
+    for d in $DOMAINS_TO_ADD; do
+      CMD_STRING="$CMD_STRING po/$d.po"
+    done
+    echo "Please run"
+    echo
+    echo "  $CMD_STRING"
+    echo
+    echo "now and add the following domains to the po/LINGUAS file:"
+    echo
+    echo "  $DOMAINS_TO_ADD"
+    echo
+    echo
+  fi
+  echo "===================================================================="
+fi
+
+if [ -n "$DOMAINS_NOT_IN_LINGUAS" ]; then
+  echo
+  echo "Existing domains missing from the po/LINGUAS file:"
+  echo
+  echo "  $DOMAINS_NOT_IN_LINGUAS"
+  echo
+  echo
+fi
+
+
diff --git a/common/extract-release-date-from-doap-file b/common/extract-release-date-from-doap-file
new file mode 100755
index 0000000..f57e307
--- /dev/null
+++ b/common/extract-release-date-from-doap-file
@@ -0,0 +1,32 @@
+#!/bin/sh
+# Shell script to extract the date given a release version and a .doap file
+
+if test "x$1" = "x" -o "x$2" = "x" -o ! -s "$2"; then
+  echo "Usage: $0 RELEASE-VERSION-NUMBER DOAP-FILE" >&2;
+  exit 1
+fi
+
+if ! grep '/dev/null ; then
+  echo "$2 does not look lika a .doap file" >&2;
+  exit 1
+fi
+
+if ! grep "$1" "$2" >/dev/null ; then
+  echo "$2 contains no reference to a version $1" >&2;
+  exit 1
+fi
+
+awk 'BEGIN {x=0}
+{
+if ( $0 ~ // ) {x=1; chunk=""}
+if (x==1) {
+  if ($0 ~ //) { chunk = chunk $0 }
+  if ($0 ~ //) { chunk = chunk $0 }
+}
+if ($0 ~ /<\/release>/) {x=0; print chunk}
+}' < "$2" | \
+\
+grep ''"$1"'' | \
+\
+sed -e 's/^.*//' -e 's/<\/created>.*$//'
+
diff --git a/common/gettext.patch b/common/gettext.patch
new file mode 100644
index 0000000..682b905
--- /dev/null
+++ b/common/gettext.patch
@@ -0,0 +1,12 @@
+--- po/Makefile.in.in.orig	2006-01-07 12:03:45.000000000 +0100
++++ po/Makefile.in.in	2006-01-07 12:04:23.000000000 +0100
+@@ -11,6 +11,9 @@
+ PACKAGE = @PACKAGE@
+ VERSION = @VERSION@
+ 
++# thomas: add GETTEXT_PACKAGE substitution as used in Makevars
++GETTEXT_PACKAGE = @GETTEXT_PACKAGE@
++
+ SHELL = /bin/sh
+ @SET_MAKE@
+ 
diff --git a/common/glib-gen.mak b/common/glib-gen.mak
new file mode 100644
index 0000000..ef93a5f
--- /dev/null
+++ b/common/glib-gen.mak
@@ -0,0 +1,44 @@
+# these are the variables your Makefile.am should set
+# the example is based on the colorbalance interface
+
+#glib_enum_headers=$(colorbalance_headers)
+#glib_enum_define=GST_COLOR_BALANCE
+#glib_enum_prefix=gst_color_balance
+
+enum_headers=$(foreach h,$(glib_enum_headers),\n\#include \"$(h)\")
+
+# these are all the rules generating the relevant files
+%-marshal.h: %-marshal.list
+	$(AM_V_GEN)glib-genmarshal --header --prefix=$(glib_enum_prefix)_marshal $^ > $*-marshal.h.tmp && \
+	mv $*-marshal.h.tmp $*-marshal.h
+
+%-marshal.c: %-marshal.list
+	$(AM_V_GEN)echo "#include \"$*-marshal.h\"" >> $*-marshal.c.tmp && \
+	glib-genmarshal --body --prefix=$(glib_enum_prefix)_marshal $^ >> $*-marshal.c.tmp && \
+	mv $*-marshal.c.tmp $*-marshal.c
+
+%-enumtypes.h: $(glib_enum_headers)
+	$(AM_V_GEN)glib-mkenums \
+	--fhead "#ifndef __$(glib_enum_define)_ENUM_TYPES_H__\n#define __$(glib_enum_define)_ENUM_TYPES_H__\n\n#include \n\nG_BEGIN_DECLS\n" \
+	--fprod "\n/* enumerations from \"@filename@\" */\n" \
+	--vhead "GType @enum_name@_get_type (void);\n#define GST_TYPE_@ENUMSHORT@ (@enum_name@_get_type())\n"         \
+	--ftail "G_END_DECLS\n\n#endif /* __$(glib_enum_define)_ENUM_TYPES_H__ */" \
+	$^ > $@
+
+%-enumtypes.c: $(glib_enum_headers)
+	@if test "x$(glib_enum_headers)" = "x"; then echo "ERROR: glib_enum_headers is empty, please fix Makefile"; exit 1; fi
+	$(AM_V_GEN)glib-mkenums \
+	--fhead "#include \"$*-enumtypes.h\"\n$(enum_headers)" \
+	--fprod "\n/* enumerations from \"@filename@\" */" \
+	--vhead "GType\n@enum_name@_get_type (void)\n{\n  static volatile gsize g_define_type_id__volatile = 0;\n  if (g_once_init_enter (&g_define_type_id__volatile)) {\n    static const G@Type@Value values[] = {"     \
+	--vprod "      { @VALUENAME@, \"@VALUENAME@\", \"@valuenick@\" }," \
+	--vtail "      { 0, NULL, NULL }\n    };\n    GType g_define_type_id = g_@type@_register_static (\"@EnumName@\", values);\n    g_once_init_leave (&g_define_type_id__volatile, g_define_type_id);\n  }\n  return g_define_type_id__volatile;\n}\n" \
+	$^ > $@
+
+# a hack rule to make sure .Plo files exist because they get include'd
+# from Makefile's
+.deps/%-marshal.Plo:
+	@touch $@
+
+.deps/%-enumtypes.Plo:
+	@touch $@
diff --git a/common/gst-autogen.sh b/common/gst-autogen.sh
new file mode 100644
index 0000000..78cab66
--- /dev/null
+++ b/common/gst-autogen.sh
@@ -0,0 +1,345 @@
+# a silly hack that generates autoregen.sh but it's handy
+# Remove the old autoregen.sh first to create a new file,
+# as the current one may be being read by the shell executing
+# this script.
+if [ -f "autoregen.sh" ]; then
+  rm autoregen.sh
+fi
+echo "#!/bin/sh" > autoregen.sh
+echo "./autogen.sh $@ \$@" >> autoregen.sh
+chmod +x autoregen.sh
+
+# helper functions for autogen.sh
+
+debug ()
+# print out a debug message if DEBUG is a defined variable
+{
+  if test ! -z "$DEBUG"
+  then
+    echo "DEBUG: $1"
+  fi
+}
+
+version_get ()
+# based on the command's version output, set variables
+# _MAJOR, _MINOR, _MICRO, _VERSION, using the given prefix as variable prefix
+#
+# arg 1: command binary name
+# arg 2: (uppercased) variable name prefix
+{
+  COMMAND=$1
+  VARPREFIX=`echo $2 | tr .,- _`
+  local ${VARPREFIX}_VERSION
+
+  # strip everything that's not a digit, then use cut to get the first field
+  pkg_version=`$COMMAND --version|head -n 1|sed 's/^.*)[^0-9]*//'|cut -d' ' -f1`
+  debug "pkg_version $pkg_version"
+  # remove any non-digit characters from the version numbers to permit numeric
+  # comparison
+  pkg_major=`echo $pkg_version | cut -d. -f1 | sed s/[a-zA-Z\-].*//g`
+  pkg_minor=`echo $pkg_version | cut -d. -f2 | sed s/[a-zA-Z\-].*//g`
+  pkg_micro=`echo $pkg_version | cut -d. -f3 | sed s/[a-zA-Z\-].*//g`
+  test -z "$pkg_major" && pkg_major=0
+  test -z "$pkg_minor" && pkg_minor=0
+  test -z "$pkg_micro" && pkg_micro=0
+  debug "found major $pkg_major minor $pkg_minor micro $pkg_micro"
+  eval ${VARPREFIX}_MAJOR=$pkg_major
+  eval ${VARPREFIX}_MINOR=$pkg_minor
+  eval ${VARPREFIX}_MICRO=$pkg_micro
+  eval ${VARPREFIX}_VERSION=$pkg_version
+}
+
+version_compare ()
+# Checks whether the version of VARPREFIX is equal to or
+# newer than the requested version
+# arg1: VARPREFIX
+# arg2: MAJOR
+# arg3: MINOR
+# arg4: MICRO
+{
+  VARPREFIX=`echo $1 | tr .,- _`
+  MAJOR=$2
+  MINOR=$3
+  MICRO=$4
+
+  eval pkg_major=\$${VARPREFIX}_MAJOR;
+  eval pkg_minor=\$${VARPREFIX}_MINOR;
+  eval pkg_micro=\$${VARPREFIX}_MICRO;
+
+  #start checking the version
+  debug "version_compare: $VARPREFIX against $MAJOR.$MINOR.$MICRO"
+
+    # reset check
+    WRONG=
+
+    if [ ! "$pkg_major" -gt "$MAJOR" ]; then
+      debug "major: $pkg_major <= $MAJOR"
+      if [ "$pkg_major" -lt "$MAJOR" ]; then
+        debug "major: $pkg_major < $MAJOR"
+        WRONG=1
+      elif [ ! "$pkg_minor" -gt "$MINOR" ]; then
+        debug "minor: $pkg_minor <= $MINOR"
+        if [ "$pkg_minor" -lt "$MINOR" ]; then
+          debug "minor: $pkg_minor < $MINOR"
+          WRONG=1
+        elif [ "$pkg_micro" -lt "$MICRO" ]; then
+          debug "micro: $pkg_micro < $MICRO"
+	  WRONG=1
+        fi
+      fi
+    fi
+    if test ! -z "$WRONG"; then
+      debug "version_compare: $VARPREFIX older than $MAJOR.$MINOR.$MICRO"
+      return 1
+    fi
+    debug "version_compare: $VARPREFIX equal to/newer than $MAJOR.$MINOR.$MICRO"
+    return 0
+}
+
+
+version_check ()
+# check the version of a package
+# first argument : package name (executable)
+# second argument : optional path where to look for it instead
+# third argument : source download url
+# rest of arguments : major, minor, micro version
+# all consecutive ones : suggestions for binaries to use
+# (if not specified in second argument)
+{
+  PACKAGE=$1
+  PKG_PATH=$2
+  URL=$3
+  MAJOR=$4
+  MINOR=$5
+  MICRO=$6
+
+  # for backwards compatibility, we let PKG_PATH=PACKAGE when PKG_PATH null
+  if test -z "$PKG_PATH"; then PKG_PATH=$PACKAGE; fi
+  debug "major $MAJOR minor $MINOR micro $MICRO"
+  VERSION=$MAJOR
+  if test ! -z "$MINOR"; then VERSION=$VERSION.$MINOR; else MINOR=0; fi
+  if test ! -z "$MICRO"; then VERSION=$VERSION.$MICRO; else MICRO=0; fi
+
+  debug "major $MAJOR minor $MINOR micro $MICRO"
+
+  for SUGGESTION in $PKG_PATH; do
+    COMMAND="$SUGGESTION"
+
+    # don't check if asked not to
+    test -z "$NOCHECK" && {
+      printf "  checking for $COMMAND >= $VERSION ... "
+    } || {
+      # we set a var with the same name as the package, but stripped of
+      # unwanted chars
+      VAR=`echo $PACKAGE | sed 's/-//g'`
+      debug "setting $VAR"
+      eval $VAR="$COMMAND"
+      return 0
+    }
+
+    which $COMMAND > /dev/null 2>&1
+    if test $? -eq 1;
+    then 
+      debug "$COMMAND not found"
+      continue
+    fi
+
+    VARPREFIX=`echo $COMMAND | sed 's/-//g' | tr [:lower:] [:upper:]`
+    version_get $COMMAND $VARPREFIX
+
+    version_compare $VARPREFIX $MAJOR $MINOR $MICRO
+    if test $? -ne 0; then
+      echo "found $pkg_version, not ok !"
+      continue
+    else
+      echo "found $pkg_version, ok."
+      # we set a var with the same name as the package, but stripped of
+      # unwanted chars
+      VAR=`echo $PACKAGE | sed 's/-//g'`
+      debug "setting $VAR"
+      eval $VAR="$COMMAND"
+      return 0
+    fi
+  done
+
+  echo "$PACKAGE not found !"
+  echo "You must have $PACKAGE installed to compile $package."
+  echo "Download the appropriate package for your distribution,"
+  echo "or get the source tarball at $URL"
+  return 1;
+}
+
+aclocal_check ()
+{
+  # normally aclocal is part of automake
+  # so we expect it to be in the same place as automake
+  # so if a different automake is supplied, we need to adapt as well
+  # so how's about replacing automake with aclocal in the set var,
+  # and saving that in $aclocal ?
+  # note, this will fail if the actual automake isn't called automake*
+  # or if part of the path before it contains it
+  if [ -z "$automake" ]; then
+    echo "Error: no automake variable set !"
+    return 1
+  else
+    aclocal=`echo $automake | sed s/automake/aclocal/`
+    debug "aclocal: $aclocal"
+    if [ "$aclocal" != "aclocal" ];
+    then
+      CONFIGURE_DEF_OPT="$CONFIGURE_DEF_OPT --with-aclocal=$aclocal"
+    fi
+    if [ ! -x `which $aclocal` ]; then
+      echo "Error: cannot execute $aclocal !"
+      return 1
+    fi
+  fi
+}
+
+autoheader_check ()
+{
+  # same here - autoheader is part of autoconf
+  # use the same voodoo
+  if [ -z "$autoconf" ]; then
+    echo "Error: no autoconf variable set !"
+    return 1
+  else
+    autoheader=`echo $autoconf | sed s/autoconf/autoheader/`
+    debug "autoheader: $autoheader"
+    if [ "$autoheader" != "autoheader" ];
+    then
+      CONFIGURE_DEF_OPT="$CONFIGURE_DEF_OPT --with-autoheader=$autoheader"
+    fi
+    if [ ! -x `which $autoheader` ]; then
+      echo "Error: cannot execute $autoheader !"
+      return 1
+    fi
+  fi
+
+}
+
+die_check ()
+{
+  # call with $DIE
+  # if set to 1, we need to print something helpful then die
+  DIE=$1
+  if test "x$DIE" = "x1";
+  then
+    echo
+    echo "- Please get the right tools before proceeding."
+    echo "- Alternatively, if you're sure we're wrong, run with --nocheck."
+    exit 1
+  fi
+}
+
+autogen_options ()
+{
+  if test "x$1" = "x"; then
+    return 0
+  fi
+
+  while test "x$1" != "x" ; do
+    optarg=`expr "x$1" : 'x[^=]*=\(.*\)'`
+    case "$1" in
+      --noconfigure)
+          NOCONFIGURE=defined
+	  AUTOGEN_EXT_OPT="$AUTOGEN_EXT_OPT --noconfigure"
+          echo "+ configure run disabled"
+          shift
+          ;;
+      --nocheck)
+	  AUTOGEN_EXT_OPT="$AUTOGEN_EXT_OPT --nocheck"
+          NOCHECK=defined
+          echo "+ autotools version check disabled"
+          shift
+          ;;
+      -d|--debug)
+          DEBUG=defined
+	  AUTOGEN_EXT_OPT="$AUTOGEN_EXT_OPT --debug"
+          echo "+ debug output enabled"
+          shift
+          ;;
+      -h|--help)
+          echo "autogen.sh (autogen options) -- (configure options)"
+          echo "autogen.sh help options: "
+          echo " --noconfigure            don't run the configure script"
+          echo " --nocheck                don't do version checks"
+          echo " --debug                  debug the autogen process"
+          echo
+          echo " --with-autoconf PATH     use autoconf in PATH"
+          echo " --with-automake PATH     use automake in PATH"
+          echo
+          echo "Any argument either not in the above list or after a '--' will be "
+          echo "passed to ./configure."
+	  exit 1
+          ;;
+      --with-automake=*)
+          AUTOMAKE=$optarg
+          echo "+ using alternate automake in $optarg"
+	  CONFIGURE_DEF_OPT="$CONFIGURE_DEF_OPT --with-automake=$AUTOMAKE"
+          shift
+          ;;
+      --with-autoconf=*)
+          AUTOCONF=$optarg
+          echo "+ using alternate autoconf in $optarg"
+	  CONFIGURE_DEF_OPT="$CONFIGURE_DEF_OPT --with-autoconf=$AUTOCONF"
+          shift
+          ;;
+      --) shift ; break ;;
+      *)
+          echo "+ passing argument $1 to configure"
+	  CONFIGURE_EXT_OPT="$CONFIGURE_EXT_OPT $1"
+          shift
+          ;;
+    esac
+  done
+
+  for arg do CONFIGURE_EXT_OPT="$CONFIGURE_EXT_OPT $arg"; done
+  if test ! -z "$CONFIGURE_EXT_OPT"
+  then
+    echo "+ options passed to configure: $CONFIGURE_EXT_OPT"
+  fi
+}
+
+toplevel_check ()
+{
+  srcfile=$1
+  test -f $srcfile || {
+        echo "You must run this script in the top-level $package directory"
+        exit 1
+  }
+}
+
+tool_run ()
+{
+  tool=$1
+  options=$2
+  run_if_fail=$3
+  echo "+ running $tool $options..."
+  $tool $options || {
+    echo
+    echo $tool failed
+    eval $run_if_fail
+    exit 1
+  }
+}
+
+install_git_hooks ()
+{
+  if test -d .git; then
+    # install pre-commit hook for doing clean commits
+    for hook in pre-commit; do
+      if test ! \( -x .git/hooks/$hook -a -L .git/hooks/$hook \); then
+        echo "+ Installing git $hook hook"
+        rm -f .git/hooks/$hook
+        ln -s ../../common/hooks/$hook.hook .git/hooks/$hook || {
+          # if we couldn't create a symbolic link, try doing a plain cp
+          if cp common/hooks/pre-commit.hook .git/hooks/pre-commit; then
+            chmod +x .git/hooks/pre-commit;
+          else
+            echo "********** Couldn't install git $hook hook **********";
+          fi
+        }
+      fi
+    done
+  fi
+}
diff --git a/common/gst-indent b/common/gst-indent
new file mode 100755
index 0000000..732b2ba
--- /dev/null
+++ b/common/gst-indent
@@ -0,0 +1,49 @@
+#!/bin/sh
+#
+# Check that the code follows a consistant code style
+#
+
+# Check for existence of indent, and error out if not present.
+# On some *bsd systems the binary seems to be called gnunindent,
+# so check for that first.
+
+version=`gnuindent --version 2>/dev/null`
+if test "x$version" = "x"; then
+  version=`indent --version 2>/dev/null`
+  if test "x$version" = "x"; then
+    echo "GStreamer git pre-commit hook:"
+    echo "Did not find GNU indent, please install it before continuing."
+    exit 1
+  fi
+  INDENT=indent
+else
+  INDENT=gnuindent
+fi
+
+case `$INDENT --version` in
+  GNU*)
+      ;;
+  default)
+      echo "GStreamer git pre-commit hook:"
+      echo "Did not find GNU indent, please install it before continuing."
+      echo "(Found $INDENT, but it doesn't seem to be GNU indent)"
+      exit 1
+      ;;
+esac
+
+INDENT_PARAMETERS="--braces-on-if-line \
+	--case-brace-indentation0 \
+	--case-indentation2 \
+	--braces-after-struct-decl-line \
+	--line-length80 \
+	--no-tabs \
+	--cuddle-else \
+	--dont-line-up-parentheses \
+	--continuation-indentation4 \
+	--honour-newlines \
+	--tab-size8 \
+	--indent-level2 \
+	--leave-preprocessor-space"
+
+$INDENT ${INDENT_PARAMETERS} $@
+
diff --git a/common/gst.supp b/common/gst.supp
new file mode 100644
index 0000000..2c60392
--- /dev/null
+++ b/common/gst.supp
@@ -0,0 +1,3967 @@
+### this file contains suppressions for valgrind when running
+### the gstreamer unit tests
+### it might be useful for wider use as well
+
+### syscall suppressions
+
+{
+   
+   Memcheck:Param
+   clone(parent_tidptr)
+   fun:clone
+   fun:clone
+}
+
+{
+   
+   Memcheck:Param
+   clone(child_tidptr)
+   fun:clone
+   fun:clone
+}
+
+{
+   
+   Memcheck:Param
+   clone(tlsinfo)
+   fun:clone
+   fun:clone
+}
+
+### glibc suppressions
+
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.*.so
+   fun:dl_open_worker
+   obj:/lib/ld-2.*.so
+   fun:_dl_open
+   fun:dlopen_doit
+   obj:/lib/ld-2.*.so
+   fun:_dlerror_run
+   fun:dlopen
+   fun:g_module_open
+   fun:gst_plugin_load_file
+}
+
+{
+   
+   Memcheck:Cond
+   fun:strlen
+   fun:fillin_rpath
+   fun:_dl_init_paths
+   fun:dl_main
+   fun:_dl_sysdep_start
+   fun:_dl_start
+   obj:/lib64/ld-2.*.so
+   obj:*
+   obj:*
+}
+
+{
+   
+   Memcheck:Cond
+   fun:_dl_relocate_object
+   fun:dl_main
+   fun:_dl_sysdep_start
+   fun:_dl_start
+}
+
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.2.5
+}
+
+# glibc does not deallocate thread-local storage
+
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:_dl_allocate_tls
+   fun:pthread_create@@*
+}
+
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:allocate_dtv
+   fun:_dl_allocate_tls
+}
+
+# I get an extra stack entry on x86/dapper
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   obj:/lib/ld-2.3.*.so
+   fun:_dl_allocate_tls
+   fun:pthread_create@@*
+}
+
+
+{
+   
+   Memcheck:Cond
+   fun:strstr
+   fun:__pthread_initialize_minimal
+   obj:/lib/libpthread-*.so
+   obj:/lib/libpthread-*.so
+   fun:call_init
+   fun:_dl_init
+   obj:/lib/ld-*.so
+}
+
+# a thread-related free problem in glibc from Edgard
+{
+   __libc_freeres_rw_acess
+   Memcheck:Addr4
+   obj:*
+   obj:*
+   obj:*
+   obj:*
+   obj:*
+   fun:__libc_freeres
+}
+
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+}
+
+# g_module_open-related problems
+{
+   
+   Memcheck:Addr2
+   fun:memcpy
+   fun:_dl_map_object_deps
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.1
+   fun:g_module_open
+   fun:gst_plugin_load_file
+   fun:gst_registry_scan_path_level
+   fun:gst_registry_scan_path_level
+   fun:gst_registry_scan_path_level
+   fun:init_post
+   fun:g_option_context_parse
+   fun:gst_init_check
+   fun:gst_init
+   fun:gst_check_init
+   fun:main
+}
+
+{
+   
+   Memcheck:Addr4
+   fun:memcpy
+   fun:_dl_map_object_deps
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.1
+   fun:g_module_open
+   fun:gst_plugin_load_file
+   fun:gst_registry_scan_path_level
+   fun:gst_registry_scan_path_level
+   fun:gst_registry_scan_path_level
+   fun:init_post
+   fun:g_option_context_parse
+   fun:gst_init_check
+   fun:gst_init
+   fun:gst_check_init
+   fun:main
+}
+
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   fun:do_sym
+   fun:_dl_sym
+   fun:dlsym_doit
+   obj:/lib/ld-2.3.*.so
+   fun:_dlerror_run
+   fun:dlsym
+   fun:g_module_symbol
+   fun:g_module_open
+   fun:gst_plugin_load_file
+}
+
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   fun:dl_open_worker
+   obj:/lib/ld-2.3.*.so
+   fun:_dl_open
+   fun:dlopen_doit
+   obj:/lib/ld-2.3.*.so
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.1
+   fun:g_module_open
+   fun:gst_plugin_load_file
+}
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.3.*.so
+   fun:dl_open_worker
+   obj:/lib/ld-2.3.*.so
+   fun:_dl_open
+   fun:dlopen_doit
+   obj:/lib/ld-2.3.*.so
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.1
+   fun:g_module_open
+   fun:gst_plugin_load_file
+   fun:gst_plugin_load_by_name
+   fun:gst_plugin_feature_load
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   obj:/lib/ld-2.3.*.so
+   fun:dl_open_worker
+   obj:/lib/ld-2.3.*.so
+   fun:_dl_open
+   fun:dlopen_doit
+   obj:/lib/ld-2.3.*.so
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.1
+   fun:g_module_open
+   fun:gst_plugin_load_file
+   fun:gst_plugin_load_by_name
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   fun:dl_open_worker
+   obj:/lib/ld-2.3.*.so
+   fun:_dl_open
+   fun:dlopen_doit
+   obj:/lib/ld-2.3.*.so
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   fun:dl_open_worker
+   obj:/lib/ld-2.3.*.so
+   fun:_dl_open
+   fun:dlopen_doit
+   obj:/lib/ld-2.3.*.so
+   fun:_dlerror_run
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   fun:dl_open_worker
+   obj:/lib/ld-2.3.*.so
+   fun:_dl_open
+   fun:dlopen_doit
+   obj:/lib/ld-2.3.*.so
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.1
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   fun:dl_open_worker
+   obj:/lib/ld-2.3.*.so
+   fun:_dl_open
+   fun:dlopen_doit
+   obj:/lib/ld-2.3.*.so
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.1
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   fun:do_sym
+   fun:_dl_sym
+   fun:dlsym_doit
+   obj:/lib/ld-2.3.*.so
+   fun:_dlerror_run
+   fun:dlsym
+   fun:g_module_symbol
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Param
+   futex(uaddr2)
+   fun:pthread_once
+   obj:/lib/libc-2.3.*.so
+   obj:/lib/libc-2.3.*.so
+   fun:mbsnrtowcs
+   fun:vfprintf
+   fun:vsprintf
+   fun:sprintf
+   obj:/lib/libc-2.3.*.so
+   fun:tmpfile
+   fun:setup_pipe
+   fun:setup_messaging_with_key
+   fun:setup_messaging
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libc-2.7.so
+   fun:_dl_sym
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlsym
+   fun:g_module_symbol
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libc-2.7.so
+   obj:/lib/ld-2.7.so
+   fun:__libc_dlopen_mode
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libc-2.7.so
+   obj:/lib/ld-2.7.so
+   fun:__libc_dlopen_mode
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libc-2.7.so
+   obj:/lib/ld-2.7.so
+   fun:__libc_dlopen_mode
+   obj:/lib/i686/cmov/libc-2.7.so
+   obj:/lib/i686/cmov/libc-2.7.so
+   obj:/lib/i686/cmov/libc-2.7.so
+   obj:/lib/i686/cmov/libc-2.7.so
+   obj:/lib/i686/cmov/libc-2.7.so
+   fun:iconv_open
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libc-2.7.so
+   obj:/lib/ld-2.7.so
+   fun:__libc_dlopen_mode
+   obj:/lib/i686/cmov/libc-2.7.so
+   obj:/lib/i686/cmov/libc-2.7.so
+   obj:/lib/i686/cmov/libc-2.7.so
+   obj:/lib/i686/cmov/libc-2.7.so
+   obj:/lib/i686/cmov/libc-2.7.so
+   fun:iconv_open
+}
+
+{
+   
+   Memcheck:Addr8
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libc-2.7.so
+   obj:/lib/ld-2.7.so
+   fun:__libc_dlopen_mode
+   obj:/lib/libc-2.7.so
+   obj:/lib/libc-2.7.so
+   obj:/lib/libc-2.7.so
+   obj:/lib/libc-2.7.so
+   obj:/lib/libc-2.7.so
+   fun:iconv_open
+}
+
+{
+   
+   Memcheck:Addr8
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libc-2.7.so
+   obj:/lib/ld-2.7.so
+   fun:__libc_dlopen_mode
+   obj:/lib/libc-2.7.so
+   obj:/lib/libc-2.7.so
+   obj:/lib/libc-2.7.so
+   obj:/lib/libc-2.7.so
+   obj:/lib/libc-2.7.so
+   fun:iconv_open
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+}
+
+{
+   
+   Memcheck:Addr8
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libc-2.7.so
+   obj:/lib/ld-2.7.so
+   fun:__libc_dlopen_mode
+}
+
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+}
+
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+}
+
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+}
+
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+}
+
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+}
+
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/i686/cmov/libdl-2.7.so
+   fun:dlopen
+}
+
+# suppression for a glibc bug:
+# http://valgrind.org/docs/manual/faq.html#faq.exit_errors>
+{
+   
+   Memcheck:Free
+   fun:free
+   obj:*libc-*.so
+   fun:__libc_freeres
+   fun:*
+   fun:_Exit
+}
+
+# same as above, just so it works for tpm on gutsy/x86-64
+{
+   
+   Memcheck:Free
+   fun:free
+   fun:free_mem
+   fun:__libc_freeres
+}
+
+# valgrind doesn't allow me to specify a suppression for Addr1, Addr2, Addr4
+# as Addr*, so 3 copies for that; and then 2 of each for that pesky memcpy
+{
+   
+   Memcheck:Addr1
+   fun:_dl_signal_error
+   fun:_dl_map_object_deps
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.1
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr2
+   fun:_dl_signal_error
+   fun:_dl_map_object_deps
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.1
+   fun:g_module_open
+}
+{
+   
+   Memcheck:Addr4
+   fun:_dl_signal_error
+   fun:_dl_map_object_deps
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.1
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr1
+   fun:memcpy
+   fun:_dl_signal_error
+   fun:_dl_map_object_deps
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.1
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr2
+   fun:memcpy
+   fun:_dl_signal_error
+   fun:_dl_map_object_deps
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.1
+   fun:g_module_open
+}
+{
+   
+   Memcheck:Addr4
+   fun:memcpy
+   fun:_dl_signal_error
+   fun:_dl_map_object_deps
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.1
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr8
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/libc-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   fun:_dl_open
+   obj:/lib/libdl-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+}
+
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/libc-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   fun:_dl_open
+   obj:/lib/libdl-2.3.*.so
+   obj:/lib/ld-2.3.*.so
+   obj:/lib/libdl-2.3.*.so
+   fun:dlopen
+   fun:g_module_open
+   fun:gst_plugin_load_file
+   fun:gst_plugin_load_by_name
+   fun:gst_plugin_feature_load
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.3.6.so
+   obj:/lib/ld-2.3.6.so
+   obj:/lib/tls/i686/cmov/libc-2.3.6.so
+   obj:/lib/ld-2.3.6.so
+   fun:_dl_open
+   obj:/lib/tls/i686/cmov/libdl-2.3.6.so
+   obj:/lib/ld-2.3.6.so
+   obj:/lib/tls/i686/cmov/libdl-2.3.6.so
+   fun:dlopen
+}
+
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.3.6.so
+   obj:/lib/tls/i686/cmov/libc-2.3.6.so
+   obj:/lib/ld-2.3.6.so
+   fun:_dl_open
+   obj:/lib/tls/i686/cmov/libdl-2.3.6.so
+   obj:/lib/ld-2.3.6.so
+   obj:/lib/tls/i686/cmov/libdl-2.3.6.so
+   fun:dlopen
+}
+
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.3.6.so
+   obj:/lib/ld-2.3.6.so
+   obj:/lib/ld-2.3.6.so
+   obj:/lib/tls/i686/cmov/libc-2.3.6.so
+   obj:/lib/ld-2.3.6.so
+   fun:_dl_open
+   obj:/lib/tls/i686/cmov/libdl-2.3.6.so
+   obj:/lib/ld-2.3.6.so
+   obj:/lib/tls/i686/cmov/libdl-2.3.6.so
+   fun:dlopen
+}
+
+### glib suppressions
+{
+   
+   Memcheck:Cond
+   fun:g_parse_debug_string
+   obj:/usr/lib*/libglib-2.0.so.*
+   fun:g_slice_alloc
+   fun:g_slice_alloc0
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   ...
+   fun:g_type_init*
+   fun:init_pre*
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   ...
+   fun:g_type_register_fundamental
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:realloc
+   fun:g_realloc
+   fun:type_node_any_new_W
+}
+
+{
+   
+   Memcheck:Leak
+   fun:realloc
+   fun:g_realloc
+   fun:type_node_any_new_W
+}
+
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:g_type_class_ref
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:realloc
+   fun:g_realloc
+   fun:type_add_flags_W
+}
+
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:type_add_flags_W
+}
+
+#pthread memleaks
+
+{
+   Thread creation leak
+   Memcheck:Leak
+   fun:calloc
+   fun:allocate_dtv
+   fun:_dl_allocate*
+   fun:_dl_allocate*
+   fun:__pthread_initialize_minimal
+}
+
+{
+   Thread management leak
+   Memcheck:Leak
+   fun:calloc
+   fun:allocate_dtv
+   fun:_dl_allocate*
+   fun:_dl_allocate*
+   fun:__pthread_*
+}
+
+{
+   Thread management leak 2
+   Memcheck:Leak
+   fun:memalign
+   fun:_dl_allocate*
+   fun:_dl_allocate*
+   fun:__pthread_*
+}
+
+{
+   pthread_create Syscall param write(buf) points to uninitialised byte(s)
+   Memcheck:Param
+   write(buf)
+   fun:pthread_create@@GLIBC_2.2.5
+   fun:g_thread_create*
+
+}
+
+# nss_parse_* memleak (used by g_option_context_parse)
+{
+   nss_parse_* memleak
+   Memcheck:Leak
+   fun:malloc
+   fun:nss_parse_service_list
+   fun:__nss_database_lookup
+}
+
+# liboil suppressions
+{
+   
+   Memcheck:Value8
+   obj:/usr/lib/liboil-0.3.so.0.1.0
+   obj:/usr/lib/liboil-0.3.so.0.1.0
+   obj:/usr/lib/liboil-0.3.so.0.1.0
+   fun:oil_cpu_fault_check_try
+   fun:oil_test_check_impl
+   fun:oil_class_optimize
+   fun:oil_optimize_all
+   fun:oil_init
+}
+
+{
+   
+   Memcheck:Addr8
+   obj:/lib/ld-2.3.6.so
+}
+
+{
+   
+   Memcheck:Param
+   futex(uaddr2)
+   fun:pthread_once
+   obj:/lib/libc-2.3.6.so
+   obj:/lib/libc-2.3.6.so
+   fun:setlocale
+   fun:init_pre
+   fun:g_option_context_parse
+   fun:gst_init_check
+   fun:gst_init
+   fun:gst_check_init
+   fun:main
+}
+
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.3.6.so
+   obj:/lib/ld-2.3.6.so
+   fun:_dl_open
+   obj:/lib/libdl-2.3.6.so
+   obj:/lib/ld-2.3.6.so
+   obj:/lib/libdl-2.3.6.so
+   fun:dlopen
+   fun:g_module_open
+   fun:gst_plugin_load_file
+}
+# this exists in a bunch of different variations, hence the short tail/trace
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.4.so
+   obj:/lib/ld-2.4.so
+}
+{
+   
+   Memcheck:Addr8
+   obj:/lib/ld-2.4.so
+   obj:/lib/ld-2.4.so
+}
+
+# More edgy suppressions (Mike)
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.4.so
+   obj:/lib/ld-2.4.so
+   obj:/lib/ld-2.4.so
+   obj:/lib/ld-2.4.so
+   fun:dlopen_doit
+   obj:/lib/ld-2.4.so
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.1
+}
+
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.4.so
+   obj:/lib/ld-2.4.so
+   obj:/lib/ld-2.4.so
+   obj:/lib/ld-2.4.so
+   obj:/lib/ld-2.4.so
+   obj:/lib/ld-2.4.so
+   fun:dlopen_doit
+   obj:/lib/ld-2.4.so
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.1
+}
+
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.4.so
+   obj:/lib/ld-2.4.so
+   obj:/lib/ld-2.4.so
+   fun:do_sym
+   fun:_dl_sym
+}
+
+# This one's overly general, but there's zero other information in the stack
+# trace - just these five lines!
+{
+   
+   Memcheck:Cond
+   obj:/lib/ld-2.4.so
+   obj:/lib/ld-2.4.so
+   obj:/lib/ld-2.4.so
+   obj:/lib/ld-2.4.so
+   obj:/lib/ld-2.4.so
+}
+
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   obj:/lib/ld-2.4.so
+   fun:_dl_allocate_tls
+   fun:pthread_create@@GLIBC_2.1
+}
+
+# TLS leaks for feisty/x86
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:allocate_dtv
+   fun:_dl_allocate_tls
+   fun:pthread_create@@GLIBC_2.1
+}
+
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   obj:/usr/lib/libcdio.so.6.0.1
+   fun:cdio_open_am_linux
+   obj:/usr/lib/libcdio.so.6.0.1
+   fun:cdio_open_am
+}
+
+{
+   
+   Memcheck:Addr8
+   obj:/lib/ld-2.5.so
+}
+
+{
+   
+   Memcheck:Cond
+   fun:snd_pcm_direct_shm_create_or_connect
+   fun:snd_pcm_dsnoop_open
+   fun:_snd_pcm_dsnoop_open
+   obj:/*lib/libasound.so.2.0.0
+   obj:/*lib/libasound.so.2.0.0
+   fun:snd_pcm_open_slave
+   fun:_snd_pcm_plug_open
+   obj:/*lib/libasound.so.2.0.0
+   fun:snd_pcm_open_slave
+   fun:_snd_pcm_asym_open
+   obj:/*lib/libasound.so.2.0.0
+   obj:/*lib/libasound.so.2.0.0
+}
+
+{
+   
+   Memcheck:Cond
+   fun:snd*_pcm_hw_param_set_near
+}
+
+{
+   
+   Memcheck:Cond
+   ...
+   fun:snd*_pcm_hw_param_set_near
+}
+
+{
+   
+   Memcheck:Cond
+   obj:/*lib/libasound.so.2.0.0
+   obj:/*lib/libasound.so.2.0.0
+   fun:snd_pcm_close
+   obj:/*lib/libasound.so.2.0.0
+}
+{
+   
+   Memcheck:Cond
+   fun:snd_pcm_direct_shm_create_or_connect
+   fun:snd_pcm_dmix_open
+   fun:_snd_pcm_dmix_open
+   obj:/*lib/libasound.so.2.0.0
+   obj:/*lib/libasound.so.2.0.0
+   fun:snd_pcm_open_slave
+   fun:_snd_pcm_softvol_open
+   obj:/*lib/libasound.so.2.0.0
+   fun:snd_pcm_open_slave
+   fun:_snd_pcm_plug_open
+   obj:/*lib/libasound.so.2.0.0
+   fun:snd_pcm_open_slave
+   fun:_snd_pcm_asym_open
+   obj:/*lib/libasound.so.2.0.0
+   obj:/*lib/libasound.so.2.0.0
+}
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:strdup
+   fun:snd_dlobj_cache_add
+   obj:/*lib/libasound.so.2.0.0
+   fun:snd_pcm_open_slave
+   fun:snd_pcm_dsnoop_open
+   fun:_snd_pcm_dsnoop_open
+   obj:/*lib/libasound.so.2.0.0
+   obj:/*lib/libasound.so.2.0.0
+   fun:snd_pcm_open_slave
+   fun:_snd_pcm_plug_open
+   obj:/*lib/libasound.so.2.0.0
+   fun:snd_pcm_open_slave
+   fun:_snd_pcm_asym_open
+   obj:/*lib/libasound.so.2.0.0
+   obj:/*lib/libasound.so.2.0.0
+}
+# Catch about 15 variations on inserting info into an ALSA
+# internal cache
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   ...
+   fun:snd*_dlobj_cache_add
+   obj:/*lib*/libasound.so.2.0.0
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   ...
+   fun:snd_pcm_open_conf
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   obj:/*lib*/libasound.so.2.0.0
+   ...
+   fun:snd_config_hook_load
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   obj:/*lib*/libasound.so.2.0.0
+   ...
+   fun:snd_config_update_r
+   fun:snd_config_update
+}
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:strdup
+   ...
+   fun:snd_config_update_r
+   fun:snd_config_update
+}
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:_dl_close_worker
+   ...
+   fun:snd_config_searcha_hooks
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   obj:/lib/libc*.so
+   fun:__nss_database_lookup
+   obj:*
+   obj:*
+   fun:getgrnam_r
+   fun:getgrnam
+   fun:snd_pcm_direct_parse_open_conf
+}
+
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:_XCBInitDisplayLock
+   fun:XOpenDisplay
+}
+
+# GConf internal initialisations related to getting the default client.
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc_tcval
+   obj:/usr/lib/libORBit-2.so.*
+   fun:ORBit_demarshal_IOR
+   fun:ORBit_demarshal_object
+   fun:CORBA_ORB_string_to_object
+   obj:/usr/lib/libgconf-2.so.*
+   fun:gconf_get_current_lock_holder
+   fun:gconf_activate_server
+   obj:/usr/lib/libgconf-2.so.*
+   obj:/usr/lib/libgconf-2.so.*
+   fun:gconf_engine_get_default
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc_tcval
+   obj:*
+   fun:PortableServer_POA_servant_to_reference
+   fun:*
+   fun:*
+   fun:*
+   fun:gconf_engine_get_default
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc_tcval
+   obj:/usr/lib/libORBit-2.so.*
+   fun:ORBit_demarshal_IOR
+   fun:ORBit_demarshal_object
+   fun:CORBA_ORB_string_to_object
+   obj:/usr/lib/libgconf-2.so.*
+   fun:gconf_get_current_lock_holder
+   fun:gconf_activate_server
+   obj:/usr/lib/libgconf-2.so.*
+   obj:/usr/lib/libgconf-2.so.*
+   fun:gconf_engine_get_default
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc*
+   fun:*
+   fun:ORBit_demarshal_IOR
+   fun:ORBit_demarshal_object
+   fun:ORBit_demarshal_value
+   fun:*
+   fun:ORBit_small_invoke_stub
+   fun:ConfigServer_get_default_database
+   fun:*
+   fun:gconf_engine_get_default
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc*
+   fun:*
+   fun:IOP_generate_profiles
+   fun:ORBit_marshal_object
+   fun:ORBit_marshal_value
+   fun:*
+   fun:ORBit_small_invoke_stub
+   fun:ConfigServer_add_client
+   fun:*
+   fun:*
+   fun:gconf_engine_get_default
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc_by_tc
+   fun:*
+   fun:PortableServer_POA_servant_to_reference
+   fun:*
+   fun:*
+   fun:*
+   fun:gconf_engine_get_default
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc_by_tc
+   obj:/usr/lib/libORBit-2.so.*
+   fun:ORBit_demarshal_IOR
+   fun:ORBit_demarshal_object
+   fun:CORBA_ORB_string_to_object
+   obj:/usr/lib/libgconf-2.so.*
+   fun:gconf_get_current_lock_holder
+   fun:gconf_activate_server
+   obj:/usr/lib/libgconf-2.so.*
+   obj:/usr/lib/libgconf-2.so.*
+   fun:gconf_engine_get_default
+}
+
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc*
+   fun:*
+   fun:ORBit_demarshal_IOR
+   fun:ORBit_demarshal_object
+   fun:*
+   fun:*
+   fun:gconf_activate_server
+}
+
+# Some libORBit/bonobo initialisation stuff
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:g_malloc
+   fun:ORBit_alloc_string
+   fun:CORBA_string_dup
+   fun:Bonobo_ActivationEnvValue_set
+   fun:bonobo_activation_init_activation_env
+   fun:bonobo_activation_orb_init
+   fun:bonobo_activation_init
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc*
+   fun:ORBit_small_alloc*
+   obj:/usr/lib/libORBit-2.so*
+   fun:PortableServer_POA_servant_to_reference
+   obj:/usr/lib/libbonobo-2.so*
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc_tcval
+   fun:ORBit_small_allocbuf
+   fun:ORBit_adaptor_setup
+   obj:/usr/lib/libORBit-2.so*
+   fun:ORBit_POA_setup_root
+   fun:ORBit_init_internals
+   fun:CORBA_ORB_init
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc_tcval
+   fun:ORBit_adaptor_setup
+   fun:*
+   fun:ORBit_POA_setup_root
+   fun:ORBit_init_internals
+   fun:CORBA_ORB_init
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc*
+   fun:ORBit_small_allocbuf
+   fun:bonobo_activation_init_activation_env
+   fun:bonobo_activation_orb_init
+   fun:bonobo_activation_init
+}
+
+# More GConf stuff from the FC5 buildbot, mostly variations on the
+# above stack traces
+{
+   
+   Memcheck:Param
+   writev(vector[...])
+   fun:writev
+   obj:/usr/lib/libORBit-2.so*
+   fun:link_connection_writev
+   fun:giop_send_buffer_write
+   obj:/usr/lib/libORBit-2.so*
+   fun:ORBit_small_invoke_stub
+   fun:ORBit_small_invoke_stub_n
+   fun:ORBit_c_stub_invoke
+   fun:ConfigServer_ping
+   fun:gconf_activate_server
+   obj:/usr/lib/libgconf-2.so*
+   obj:/usr/lib/libgconf-2.so*
+   fun:gconf_engine_get_default
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc*
+   fun:ORBit_small_alloc*
+   obj:/usr/lib/libORBit-2.so*
+   fun:PortableServer_POA_servant_to_reference
+   obj:/usr/lib/libgconf-2.so*
+   obj:/usr/lib/libgconf-2.so*
+   obj:/usr/lib/libgconf-2.so*
+   fun:gconf_engine_get_default
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc*
+   fun:ORBit_small_alloc
+   obj:/usr/lib/libORBit-2.so*
+   fun:ORBit_demarshal_IOR
+   fun:ORBit_demarshal_object
+   fun:CORBA_ORB_string_to_object
+   obj:/usr/lib/libgconf-2.so*
+   fun:gconf_get_current_lock_holder
+   fun:gconf_activate_server
+   obj:/usr/lib/libgconf-2.so*
+   obj:/usr/lib/libgconf-2.so*
+   fun:gconf_engine_get_default
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc*
+   fun:ORBit_small_alloc*
+   obj:/usr/lib/libORBit-2.so*
+   fun:ORBit_demarshal_IOR
+   fun:ORBit_demarshal_object
+   fun:CORBA_ORB_string_to_object
+   obj:/usr/lib/libgconf-2.so*
+   fun:gconf_get_current_lock_holder
+   fun:gconf_activate_server
+   obj:/usr/lib/libgconf-2.so*
+   obj:/usr/lib/libgconf-2.so*
+   fun:gconf_engine_get_default
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc*
+   fun:ORBit_small_alloc*
+   obj:/usr/lib/libORBit-2.so*
+   fun:ORBit_demarshal_IOR
+   fun:ORBit_demarshal_object
+   fun:ORBit_demarshal_value
+   obj:/usr/lib/libORBit-2.so*
+   fun:ORBit_small_invoke_stub
+   fun:ORBit_small_invoke_stub_n
+   fun:ORBit_c_stub_invoke
+   fun:ConfigServer_get_default_database
+   obj:/usr/lib/libgconf-2.so*
+   fun:gconf_engine_get_default
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:ORBit_alloc*
+   fun:ORBit_small_alloc*
+   obj:/usr/lib/libORBit-2.so*
+   fun:ORBit_OAObject_object_to_objkey
+   fun:IOP_generate_profiles
+   fun:ORBit_marshal_object
+   fun:ORBit_marshal_value
+   obj:/usr/lib/libORBit-2.so*
+   fun:ORBit_small_invoke_stub
+   fun:ORBit_small_invoke_stub_n
+   fun:ORBit_c_stub_invoke
+   fun:ConfigServer_add_client
+   obj:/usr/lib/libgconf-2.so*
+   obj:/usr/lib/libgconf-2.so*
+   fun:gconf_engine_get_default
+}
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   obj:*libc-*.so
+   fun:__nss_database_lookup
+   obj:*
+   obj:*
+   fun:getpwnam_r
+   obj:/usr/lib*/libglib-2.0.so.*
+   fun:g_get_home_dir
+}
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   obj:*libc-*.so
+   fun:__nss_database_lookup
+   obj:*
+   obj:*
+   fun:getpwnam_r
+   obj:/usr/lib*/libglib-2.0.so.*
+   fun:g_get_user_name
+}
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   obj:*libc-*.so
+   fun:__nss_database_lookup
+   obj:*
+   obj:*
+   fun:getpwnam_r
+   obj:/usr/lib*/libglib-2.0.so.*
+   fun:g_get_tmp_dir
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   obj:*libc-*.so
+   fun:__nss_database_lookup
+   obj:*
+   obj:*
+   fun:getpwnam_r
+   obj:/usr/lib*/libglib-2.0.so.0.*
+   fun:g_get_host_name
+}
+
+
+## Some Fontconfig errors.
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:FcPatternObjectInsertElt
+   fun:FcPatternObjectAddWithBinding
+   fun:FcPatternAppend
+   fun:FcEndElement
+   obj:/usr/lib/libexpat.so.*
+   obj:/usr/lib/libexpat.so.*
+   obj:/usr/lib/libexpat.so.*
+   obj:/usr/lib/libexpat.so.*
+   fun:XML_ParseBuffer
+   fun:FcConfigParseAndLoad
+   fun:FcConfigParseAndLoad
+   fun:FcParseInclude
+   fun:FcEndElement
+   obj:/usr/lib/libexpat.so.*
+   obj:/usr/lib/libexpat.so.*
+   obj:/usr/lib/libexpat.so.*
+   obj:/usr/lib/libexpat.so.*
+   fun:XML_ParseBuffer
+   fun:FcConfigParseAndLoad
+}
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   ...
+   fun:FcInitLoadConfig
+}
+
+# Issues with ubuntu Hardy, same crack as for previous ubuntus
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   obj:*
+   fun:_dl_allocate_tls
+   fun:pthread_create@@*
+   obj:/usr/lib/libgthread*
+   fun:g_thread_*
+}
+
+# I've made this version generic, so that it covers future modifications
+# of library names
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   obj:*
+   fun:_dl_allocate_tls
+   fun:pthread_create@@*
+   fun:g_thread_*
+}
+
+# series of invalid read of size 4 in g_module_open for ubuntu
+# hardy x86/32bit
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/tls/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/tls/i686/cmov/libdl-2.7.so
+   fun:dlopen
+   fun:g_module_open
+   fun:gst_plugin_load_*
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/tls/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/tls/i686/cmov/libdl-2.7.so
+   fun:dlopen
+   fun:g_module_open
+   fun:gst_plugin_load_*
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/tls/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/tls/i686/cmov/libdl-2.7.so
+   fun:dlopen
+   fun:g_module_open
+   fun:gst_plugin_load_*
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/tls/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/tls/i686/cmov/libdl-2.7.so
+   fun:dlopen
+   fun:g_module_open
+   fun:gst_plugin_load_*
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/tls/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/tls/i686/cmov/libdl-2.7.so
+   fun:dlopen
+   fun:g_module_open
+   fun:gst_plugin_load*
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/tls/i686/cmov/libc-2.7.so
+   fun:_dl_sym
+   obj:/lib/tls/i686/cmov/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/tls/i686/cmov/libdl-2.7.so
+   fun:dlsym
+   fun:g_module_symbol
+   fun:g_module_open
+   fun:gst_plugin_load_*
+}
+
+# series of invalid read of size 8 in g_module_open for ubuntu
+# hardy x86/64bit
+{
+   
+   Memcheck:Addr8
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libdl-2.7.so
+   fun:dlopen
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr8
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libdl-2.7.so
+   fun:dlopen
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr8
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libdl-2.7.so
+   fun:dlopen
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr8
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libdl-2.7.so
+   fun:dlopen
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr8
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libdl-2.7.so
+   fun:dlopen
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr8
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libdl-2.7.so
+   fun:dlopen
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr8
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libc-2.7.so
+   obj:/lib/libdl-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libdl-2.7.so
+   fun:dlsym
+   fun:g_module_symbol
+   fun:g_module_open
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/tls/i686/cmov/libc-2.7.so
+   obj:/lib/ld-2.7.so
+   fun:__libc_dlopen_mode
+   fun:__nss_lookup_function
+   obj:/lib/tls/i686/cmov/libc-2.7.so
+   fun:__nss_passwd_lookup
+   fun:getpwnam_r
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/tls/i686/cmov/libc-2.7.so
+   obj:/lib/ld-2.7.so
+   fun:__libc_dlopen_mode
+   fun:__nss_lookup_function
+   obj:/lib/tls/i686/cmov/libc-2.7.so
+   fun:__nss_passwd_lookup
+   fun:getpwnam_r
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/tls/i686/cmov/libc-2.7.so
+   obj:/lib/ld-2.7.so
+   fun:__libc_dlopen_mode
+   fun:__nss_lookup_function
+   obj:/lib/tls/i686/cmov/libnss_compat-2.7.so
+   fun:_nss_compat_getpwnam_r
+   fun:getpwnam_r
+}
+
+{
+   
+   Memcheck:Addr4
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/tls/i686/cmov/libc-2.7.so
+   obj:/lib/ld-2.7.so
+   fun:__libc_dlopen_mode
+   fun:__nss_lookup_function
+   obj:/lib/tls/i686/cmov/libnss_compat-2.7.so
+   fun:_nss_compat_getpwnam_r
+   fun:getpwnam_r
+}
+
+{
+   
+   Memcheck:Addr8
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/ld-2.7.so
+   obj:/lib/libc-2.7.so
+   obj:/lib/ld-2.7.so
+   fun:__libc_dlopen_mode
+   fun:__nss_lookup_function
+   obj:/lib/libc-2.7.so
+   fun:getpwnam_r
+}
+
+## Leaks in ALSA (variations of leak from snd_config_load1)
+
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:_snd_config_make
+   fun:_snd_config_make_add
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:snd_config_load1
+}
+
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:_snd_config_make
+   fun:_snd_config_make_add
+   fun:*
+   fun:*
+   fun:snd_config_load1
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:_snd_config_make
+   fun:_snd_config_make_add
+   fun:*
+   fun:*
+   fun:*
+   fun:snd_config_load1
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:_snd_config_make
+   fun:_snd_config_make_add
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:snd_config_load1
+}
+
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:_snd_config_make
+   fun:_snd_config_make_add
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:snd_config_load1
+}
+
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:_snd_config_make
+   fun:_snd_config_make_add
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:snd_config_load1
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:_snd_config_make
+   fun:_snd_config_make_add
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:snd_config_load1
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:snd1_dlobj_cache_add
+   fun:snd_ctl_open_noupdate
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:*
+   fun:snd1_dlobj_cache_add
+   fun:snd_ctl_open_noupdate
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:*
+   fun:*
+   fun:*
+   fun:snd_config_load1
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:snd_config_load1
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:snd_config_load1
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:snd_config_load1
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:snd_config_load1
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:snd_config_load1
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:snd_config_load1
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:snd_config_load1
+}
+
+
+# The following are leaks of caps that need to be created dynamically
+# in the type registration of the plugin (used for pad templates).
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:g_malloc
+   fun:g_slice_alloc
+   fun:gst_caps_new_empty
+   fun:gst_caps_new_simple
+   fun:*
+   fun:g_type_class_ref
+   fun:gst_element_register
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:g_malloc
+   fun:g_slice_alloc
+   fun:gst_caps_new_empty
+   fun:*
+   fun:*
+   fun:g_type_class_ref
+   fun:gst_element_register
+   fun:gst_ogm_parse_plugin_init
+   fun:plugin_init
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:g_malloc
+   fun:g_slice_alloc
+   fun:gst_caps_new_empty
+   fun:gst_caps_copy
+   fun:gst_video_test_src_base_init
+   fun:g_type_class_ref
+   fun:gst_element_register
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:g_malloc
+   fun:g_slice_alloc
+   fun:gst_caps_new_empty
+   fun:gst_caps_copy
+   fun:gst_video_test_src_getcaps
+   fun:gst_video_test_src_base_init
+   fun:g_type_class_ref
+   fun:gst_element_register
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:g_malloc
+   fun:g_slice_alloc
+   fun:gst_caps_new_empty
+   fun:gst_ffmpegcsp_codectype_to_caps
+   fun:gst_ffmpegcolorspace_register
+   fun:plugin_init
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:g_malloc
+   fun:g_slice_alloc
+   fun:gst_caps_new_empty
+   fun:gst_caps_copy
+   fun:gst_ffmpegcolorspace_register
+   fun:plugin_init
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:g_malloc
+   fun:g_slice_alloc
+   fun:gst_caps_new_empty
+   fun:gst_caps_new_any
+   fun:gst_ffmpegdemux_register
+   fun:plugin_init
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:g_malloc
+   fun:g_slice_alloc
+   fun:gst_caps_new_empty
+   fun:gst_caps_copy
+   fun:gst_audio_filter_class_add_pad_templates
+}
+
+{
+   
+   Memcheck:Leak
+   fun:realloc
+   fun:g_realloc
+   fun:g_ptr_array_maybe_expand
+   fun:g_ptr_array_add
+   fun:gst_caps_append
+   fun:gst_audio_filter_class_add_pad_templates
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:realloc
+   fun:g_realloc
+   fun:g_ptr_array_maybe_expand
+   fun:g_ptr_array_add
+   fun:gst_caps_append
+   fun:gst_audio_filter_class_add_pad_templates
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:realloc
+   fun:g_realloc
+   fun:g_ptr_array_maybe_expand
+   fun:g_ptr_array_add
+   fun:gst_caps_copy
+   fun:gst_audio_filter_class_add_pad_templates
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:g_malloc
+   fun:g_slice_alloc
+   fun:g_ptr_array_sized_new
+   fun:gst_caps_new_empty
+   fun:gst_caps_copy
+   fun:gst_audio_filter_class_add_pad_templates
+}
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:realloc
+   fun:g_realloc
+   fun:g_array_maybe_expand
+   fun:g_array_sized_new
+   fun:*
+   fun:*
+   fun:*
+   fun:gst_value_init_and_copy
+   fun:gst_structure_copy
+   fun:gst_caps_copy
+   fun:gst_audio_filter_class_add_pad_templates
+}
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:realloc
+   fun:g_realloc
+   fun:g_array_maybe_expand
+   fun:g_array_sized_new
+   fun:*
+   fun:gst_structure_copy
+   fun:gst_caps_copy
+   fun:gst_audio_filter_class_add_pad_templates
+}
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:g_malloc
+   fun:g_slice_alloc
+   fun:g_array_sized_new
+   fun:*
+   fun:gst_structure_copy
+   fun:gst_caps_copy
+   fun:gst_audio_filter_class_add_pad_templates
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:g_malloc
+   fun:g_slice_alloc
+   fun:gst_caps_new_empty
+   fun:gst_riff_create_*_template_caps
+}
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:realloc
+   fun:g_realloc
+   fun:*
+   fun:*
+   fun:*
+   fun:gst_structure_copy
+   fun:gst_caps_copy
+   fun:gst_caps_append
+   fun:gst_riff_create_*_template_caps
+}
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:g_malloc
+   fun:g_slice_alloc
+   fun:g_array_sized_new
+   fun:*
+   fun:gst_structure_copy
+   fun:gst_caps_copy
+   fun:gst_caps_append
+   fun:gst_riff_create_*_template_caps
+}
+
+## Leaks in pango (bilboed: gentoo unstable amd64)
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   ...
+   fun:pango_layout_get_pixel_extents
+}
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:pango_language_from_string
+   fun:pango_language_get_default
+   fun:pango_context_init
+   fun:g_type_create_instance
+   fun:g_object_constructor
+   fun:g_object_newv
+   fun:g_object_new_valist
+   fun:g_object_new
+   fun:pango_font_map_create_context
+}
+
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   fun:g_malloc0
+   fun:pango_language_from_string
+}
+
+
+## Leak of everything allocated by gst-libav plugin init
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   ...
+   fun:gst_ffmpeg_cfg_init
+}
+
+## Leak of GIO module through gnomevfs
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:g_malloc
+   fun:*
+   fun:*
+   fun:g_type_create_instance
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:g_io_module_new
+   fun:g_io_modules_load_all_in_directory
+   fun:*
+   fun:get_default_vfs
+}
+
+## Conditional jump in getaddrinfo (bilboed, gentoo ~amd64, Dec 13 2008)
+{
+   
+   Memcheck:Cond
+   fun:gaih_inet
+   fun:getaddrinfo
+}
+
+## Dynamic pad templates in mxfmux
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:g_malloc
+   fun:g_slice_alloc
+   fun:gst_caps_new_empty
+   fun:gst_caps_from_string
+   fun:mxf_*_init
+   fun:plugin_init
+}
+
+## We don't know if ffmpeg frees this or not and better pass a copy for safety
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   fun:g_malloc
+   fun:g_strdup
+   fun:gst_ffmpeg_cfg_fill_context
+   fun:gst_ffmpegenc_setcaps
+   fun:gst_pad_set_caps
+}
+
+## Leak/overreads with glibc-2.10
+
+{
+   
+   Memcheck:Value8
+   fun:do_sym
+   fun:dlsym_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlsym
+}
+{
+   
+   Memcheck:Cond
+   fun:do_sym
+   fun:dlsym_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlsym
+}
+
+{
+   
+   Memcheck:Value8
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen*
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_relocate_object
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen*
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_check_map_versions
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen*
+}
+
+{
+   
+   Memcheck:Cond
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen*
+}
+
+{
+   
+   Memcheck:Cond
+   fun:_dl_relocate_object
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen*
+}
+
+{
+   
+   Memcheck:Cond
+   fun:_dl_check_map_versions
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen*
+}
+
+{
+   
+   Memcheck:Cond
+   fun:_dl_map_object*
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen*
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_map_object*
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen*
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_check_caller
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen*
+}
+
+{
+   
+   Memcheck:Cond
+   fun:_dl_check_caller
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen*
+}
+
+{
+   
+   Memcheck:Value8
+   obj:/lib*/libc-2.10.*.so
+   obj:/lib*/libc-2.10.*.so
+   fun:_vgnU_freeres
+}
+{
+   
+   Memcheck:Cond
+   obj:/lib*/libc-2.10.*.so
+   obj:/lib*/libc-2.10.*.so
+   fun:_vgnU_freeres
+}
+{
+   
+   Memcheck:Free
+   fun:free
+   obj:/lib*/libc-2.10.*.so
+   obj:/lib*/libc-2.10.*.so
+   fun:_vgnU_freeres
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_fini
+   fun:__run_exit_handlers
+   fun:exit
+}
+
+{
+   
+   Memcheck:Cond
+   fun:_dl_fini
+   fun:__run_exit_handlers
+   fun:exit
+}
+{
+   
+   Memcheck:Value8
+   fun:_dl_sort_fini
+   fun:_dl_fini
+   fun:__run_exit_handlers
+   fun:exit
+}
+
+{
+   
+   Memcheck:Cond
+   fun:_dl_sort_fini
+   fun:_dl_fini
+   fun:__run_exit_handlers
+   fun:exit
+}
+
+# glibc-2.10 dl overreads
+{
+   
+   Memcheck:Value8
+   fun:_dl_fixup
+   fun:_dl_runtime_resolve
+}
+{
+   
+   Memcheck:Cond
+   fun:_dl_fixup
+   fun:_dl_runtime_resolve
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_lookup_symbol_x
+   fun:_dl_fixup
+   fun:_dl_runtime_resolve
+}
+{
+   
+   Memcheck:Cond
+   fun:_dl_lookup_symbol_x
+   fun:_dl_fixup
+   fun:_dl_runtime_resolve
+}
+{
+   
+   Memcheck:Value8
+   fun:call_init
+   fun:_dl_init
+}
+{
+   
+   Memcheck:Value8
+   fun:_dl_init
+}
+{
+   
+   Memcheck:Value8
+   fun:do_lookup_x
+   fun:_dl_lookup_symbol_x
+   fun:_dl_relocate_object
+   fun:dl_main
+}
+{
+   
+   Memcheck:Cond
+   fun:do_lookup_x
+   fun:_dl_lookup_symbol_x
+   fun:_dl_relocate_object
+   fun:dl_main
+}
+{
+   
+   Memcheck:Value8
+   fun:_dl_lookup_symbol_x
+   fun:_dl_relocate_object
+   fun:dl_main
+}
+{
+   
+   Memcheck:Value8
+   fun:_dl_relocate_object
+   fun:dl_main
+}
+{
+   
+   Memcheck:Value8
+   fun:dl_main
+   fun:_dl_sysdep_start
+   fun:_dl_start
+}
+{
+   
+   Memcheck:Cond
+   fun:dl_main
+   fun:_dl_sysdep_start
+   fun:_dl_start
+}
+
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:do_lookup_x
+   fun:_dl_lookup_symbol_x
+   fun:_dl_relocate_object
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:do_lookup_x
+   fun:_dl_lookup_symbol_x
+   fun:_dl_relocate_object
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_check_map_versions
+   fun:_dl_check_all_versions
+   fun:version_check_doit
+   fun:_dl_receive_error
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Cond
+   fun:_dl_check_map_versions
+   fun:_dl_check_all_versions
+   fun:version_check_doit
+   fun:_dl_receive_error
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_check_all_versions
+   fun:version_check_doit
+   fun:_dl_receive_error
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Cond
+   fun:_dl_check_all_versions
+   fun:version_check_doit
+   fun:_dl_receive_error
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:_dl_check_map_versions
+   fun:_dl_check_all_versions
+   fun:version_check_doit
+   fun:_dl_receive_error
+   fun:dl_main
+}
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:_dl_check_map_versions
+   fun:_dl_check_all_versions
+   fun:version_check_doit
+   fun:_dl_receive_error
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Value8
+   fun:init_tls
+   fun:dl_main
+}
+{
+   
+   Memcheck:Cond
+   fun:init_tls
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Cond
+   fun:_dl_map_object_deps
+   fun:dl_main
+}
+{
+   
+   Memcheck:Value8
+   fun:_dl_map_object_deps
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_protect_relro
+   fun:_dl_relocate_object
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:do_lookup_x
+   fun:_dl_lookup_symbol_x
+   fun:_dl_relocate_object
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_setup_hash
+   fun:_dl_map_object_from_fd
+   fun:_dl_map_object
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:_dl_new_object
+   fun:_dl_map_object_from_fd
+   fun:_dl_map_object
+}
+
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:_dl_new_object
+   fun:_dl_map_object_from_fd
+   fun:_dl_map_object
+}
+
+{
+   
+   Memcheck:Value8
+   fun:openaux
+   fun:_dl_catch_error
+   fun:_dl_map_object_deps
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:_dl_map_object
+}
+
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:_dl_map_object
+}
+
+{
+   
+   Memcheck:Cond
+   fun:_dl_map_object
+   fun:openaux
+   fun:_dl_catch_error
+   fun:_dl_map_object_deps
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_map_object
+   fun:openaux
+   fun:_dl_catch_error
+   fun:_dl_map_object_deps
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:_dl_map_object
+   fun:openaux
+   fun:_dl_catch_error
+   fun:_dl_map_object_deps
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:open_path
+   fun:_dl_map_object
+   fun:openaux
+   fun:_dl_catch_error
+   fun:_dl_map_object_deps
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:open_path
+   fun:_dl_map_object
+   fun:openaux
+   fun:_dl_catch_error
+   fun:_dl_map_object_deps
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_map_object_from_fd
+   fun:_dl_map_object
+}
+
+{
+   
+   Memcheck:Cond
+   fun:_dl_map_object_from_fd
+   fun:_dl_map_object
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:_dl_new_object
+   fun:_dl_map_object_from_fd
+   fun:_dl_map_object
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_new_object
+   fun:_dl_map_object_from_fd
+   fun:_dl_map_object
+}
+
+{
+   
+   Memcheck:Cond
+   fun:_dl_new_object
+   fun:_dl_map_object_from_fd
+   fun:_dl_map_object
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:_dl_name_match_p
+   fun:_dl_map_object
+}
+
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:*
+   fun:_dl_map_object
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:_dl_name_match_p
+   fun:_dl_check_map_versions
+   fun:_dl_check_all_versions
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:*
+   fun:do_lookup_x
+   fun:_dl_lookup_symbol_x
+}
+
+{
+   
+   Memcheck:Cond
+   fun:do_lookup_x
+   fun:_dl_lookup_symbol_x
+}
+
+{
+   
+   Memcheck:Value8
+   fun:do_lookup_x
+   fun:_dl_lookup_symbol_x
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:do_lookup_x
+   fun:_dl_lookup_symbol_x
+}
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:do_lookup_x
+   fun:_dl_lookup_symbol_x
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_name_match_p
+   fun:_dl_map_object
+   fun:dl_open_worker
+}
+{
+   
+   Memcheck:Cond
+   fun:_dl_name_match_p
+   fun:_dl_map_object
+   fun:dl_open_worker
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:_dl_name_match_p
+   fun:_dl_map_object
+   fun:dl_open_worker
+}
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:_dl_name_match_p
+   fun:_dl_map_object
+   fun:dl_open_worker
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_lookup_symbol_x
+   fun:_dl_relocate_object
+}
+{
+   
+   Memcheck:Cond
+   fun:_dl_lookup_symbol_x
+   fun:_dl_relocate_object
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:*
+   fun:_dl_check_map_versions
+}
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:_dl_check_map_versions
+}
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:*
+   fun:_dl_check_map_versions
+}
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:_dl_check_map_versions
+}
+{
+   
+   Memcheck:Value8
+   fun:openaux
+}
+{
+   
+   Memcheck:Value8
+   fun:_dl_name_match_p
+   fun:_dl_map_object
+}
+
+{
+   
+   Memcheck:Cond
+   fun:_dl_close_worker
+   fun:_dl_close
+   fun:_dl_catch_error
+   fun:dlerror_run
+}
+{
+   
+   Memcheck:Value8
+   fun:_dl_close_worker
+   fun:_dl_close
+   fun:_dl_catch_error
+   fun:dlerror_run
+}
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:_dl_close_worker
+   fun:_dl_close
+   fun:_dl_catch_error
+   fun:dlerror_run
+}
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:_dl_close_worker
+   fun:_dl_close
+   fun:_dl_catch_error
+   fun:dlerror_run
+}
+
+{
+   
+   Memcheck:Cond
+   fun:fillin_rpath
+   fun:_dl_init_paths
+   fun:dl_main
+}
+{
+   
+   Memcheck:Value8
+   fun:fillin_rpath
+   fun:_dl_init_paths
+   fun:dl_main
+}
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:fillin_rpath
+   fun:_dl_init_paths
+   fun:dl_main
+}
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:fillin_rpath
+   fun:_dl_init_paths
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Cond
+   fun:_dl_map_object
+   fun:map_doit
+   fun:_dl_catch_error
+   fun:do_preload
+   fun:dl_main
+}
+{
+   
+   Memcheck:Value8
+   fun:_dl_map_object
+   fun:map_doit
+   fun:_dl_catch_error
+   fun:do_preload
+   fun:dl_main
+}
+{
+   
+   Memcheck:Param
+   open(filename)
+   fun:open
+   fun:open_verify
+   fun:_dl_map_object
+   fun:map_doit
+   fun:_dl_catch_error
+   fun:do_preload
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Param
+   stat(file_name)
+   fun:_xstat
+   fun:open_path
+   fun:_dl_map_object
+   fun:openaux
+   fun:_dl_catch_error
+   fun:_dl_map_object_deps
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_catch_error
+   fun:_dl_map_object_deps
+   fun:dl_open_worker
+}
+
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:_dl_map_object_deps
+   fun:dl_main
+}
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:_dl_map_object_deps
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:*
+   fun:_dl_map_object_deps
+   fun:dl_main
+}
+
+# glibc-2.10 tls issues
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:init_tls
+   fun:dl_main
+}
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:init_tls
+   fun:dl_main
+}
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:*
+   fun:init_tls
+   fun:dl_main
+}
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:*
+   fun:init_tls
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Cond
+   fun:_dl_allocate_tls_init
+   fun:dl_main
+}
+{
+   
+   Memcheck:Value8
+   fun:_dl_allocate_tls_init
+   fun:dl_main
+}
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:_dl_allocate_tls_init
+   fun:dl_main
+}
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:_dl_allocate_tls_init
+   fun:dl_main
+}
+
+{
+   
+   Memcheck:Cond
+   fun:__tls*
+   obj:*
+   obj:*
+   fun:_vgnU_freeres
+}
+
+{
+   
+   Memcheck:Param
+   arch_prctl(arg2)
+   fun:init_tls
+}
+# GLib caching tmp/home directories (glibc-2.10 variants)
+{
+   
+   Memcheck:Cond
+   fun:*
+   fun:dl_open_worker
+   fun:*
+   fun:*
+   fun:*
+   fun:_dl_catch_error
+   fun:dlerror_run
+   fun:*
+   fun:__nss_lookup_function
+   fun:__nss_lookup
+   fun:getpwnam*
+}
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:dl_open_worker
+   fun:*
+   fun:*
+   fun:*
+   fun:_dl_catch_error
+   fun:dlerror_run
+   fun:*
+   fun:__nss_lookup_function
+   fun:__nss_lookup
+   fun:getpwnam*
+}
+{
+   
+   Memcheck:Cond
+   fun:dl_open_worker
+   fun:*
+   fun:*
+   fun:do_dlopen
+   fun:*
+   fun:dlerror_run
+   fun:*
+   fun:__nss_lookup_function
+   fun:__nss_lookup
+   fun:getpwnam*
+}
+{
+   
+   Memcheck:Value8
+   fun:dl_open_worker
+   fun:*
+   fun:*
+   fun:do_dlopen
+   fun:*
+   fun:dlerror_run
+   fun:*
+   fun:__nss_lookup_function
+   fun:__nss_lookup
+   fun:getpwnam*
+}
+
+{
+   
+   Memcheck:Value8
+   fun:_dl_add_to_slotinfo
+   fun:dl_main
+}
+{
+   
+   Memcheck:Param
+   open(filename)
+   fun:open
+   fun:open_verify
+   fun:open_path
+   fun:_dl_map_object
+}
+
+
+
+# GModule issues with glibc-2.10
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:*
+   fun:dlsym
+   fun:g_module_symbol
+}
+{
+   
+   Memcheck:Value8
+   fun:g_module_*
+   fun:gst_plugin*
+}
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:g_module_*
+   fun:gst_plugin*
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:*
+   fun:dlopen*
+   fun:g_module_open
+}
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:dlsym
+   fun:g_module_symbol
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:*
+   fun:dlopen*
+   fun:g_module_open
+}
+
+# Leak in GSlice
+{
+   
+   Memcheck:Value8
+   fun:g_parse_debug_string
+   fun:slice_config_init
+   fun:g_slice_init_nomessage
+   fun:_g_slice_thread_init_nomessage
+   fun:g_thread_init_glib
+}
+
+# 2.10 pthread issues
+{
+   
+   Memcheck:Value8
+   fun:__pthread_initialize_minimal
+}
+
+# glibc 2.11 conditional
+{
+   
+   Memcheck:Cond
+   fun:_dl_relocate_object
+   fun:dl_main
+   fun:_dl_sysdep_start
+   fun:_dl_start
+   obj:/lib64/ld-2.11.so
+}
+
+# glibc 2.11 Leak
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.2.5
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:_dl_*
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.2.5
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:_dl_*
+   fun:_dl_*
+   fun:_dl_*
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.2.5
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:*
+   fun:_dl_*
+   fun:openaux
+   fun:_dl_catch_error
+   fun:_dl_map_object_deps
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.2.5
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:*
+   fun:_dl_map_object
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.2.5
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:_dl_new_object
+   fun:_dl_map_object_from_fd
+   fun:_dl_map_object
+   fun:openaux
+   fun:_dl_catch_error
+   fun:_dl_map_object_deps
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.2.5
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:*
+   fun:_dl_*
+   fun:_dl_*
+   fun:_dl_*
+   fun:dl_open_worker
+   fun:_dl_catch_error
+   fun:_dl_open
+   fun:dlopen_doit
+   fun:_dl_catch_error
+   fun:_dlerror_run
+   fun:dlopen@@GLIBC_2.2.5
+}
+
+# glib type leaks
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   ...
+   fun:g_type_register_static
+}
+
+# new registry system
+# all of this will only be created once when loading registry.
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   ...
+   fun:_priv_gst_registry_chunks_load_plugin
+}
+
+# system-wide tags
+# these tags are registered once
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:*
+   fun:*
+   fun:gst_tag_register
+   fun:_gst_tag_initialize
+}
+
+# system-wide type classes that we keep referenced
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   ...
+   fun:g_type_class_ref
+}
+
+# leaking cached queries which are only initialized once
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   ...
+   fun:_gst_query_initialize
+   fun:init_post
+}
+
+# macosx (leopard) library loader leak
+{
+   
+   Memcheck:Leak
+   fun:_Znwm
+   fun:_ZNSs4_Rep9_S_createEmmRKSaIcE
+   fun:_ZNSs12_S_constructIPKcEEPcT_S3_RKSaIcESt20forward_iterator_tag
+   fun:_ZNSsC2EPKcRKSaIcE
+   fun:_Z41__static_initialization_and_destruction_0ii
+   fun:_ZN16ImageLoaderMachO18doModInitFunctionsERKN11ImageLoader11LinkContextE
+}
+
+# GObject type registration
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   ...
+   fun:_g_atomic_array_copy
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   fun:getdelim
+   obj:*libselinux*
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   ...
+   obj:*/sed
+}
+
+{
+   
+   Memcheck:Addr8
+   ...
+   obj:*/sed
+}
+
+# GLib 2.23 interface vtable
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   ...
+   fun:g_type_add_interface_static
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   obj:*/dash
+}
+
+# libtool/gentoo fake leak
+# it actually runs bash and valgrind complains
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   obj:/bin/bash
+}
+
+{
+   
+   Memcheck:Leak
+   fun:*alloc
+   ...
+   fun:_gst_plugin_loader_client_run
+   fun:main
+}
+
+{
+   
+   Memcheck:Cond
+   fun:*strcasecmp*
+   ...
+   fun:__dcigettext
+}
+
+{
+   
+   Memcheck:Value8
+   fun:*strcasecmp*
+   ...
+   fun:__dcigettext
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   ...
+   fun:gst_poll_new
+   fun:gst_poll_new_timer
+   fun:gst_system_clock_init
+}
+
+{
+   
+   Memcheck:Leak
+   fun:calloc
+   ...
+   fun:gobject_init_ctor
+}
+
+{
+   
+   Memcheck:Leak
+   fun:malloc
+   ...
+   fun:g_quark_from*_string
+}
diff --git a/common/gstdoc-scangobj b/common/gstdoc-scangobj
new file mode 100755
index 0000000..4ad9786
--- /dev/null
+++ b/common/gstdoc-scangobj
@@ -0,0 +1,1617 @@
+#!/usr/bin/env perl
+# -*- cperl -*-
+#
+# gtk-doc - GTK DocBook documentation generator.
+# Copyright (C) 1998  Damon Chaplin
+#
+# This program is free software; you can redistribute it and/or modify
+# it under the terms of the GNU General Public License as published by
+# the Free Software Foundation; either version 2 of the License, or
+# (at your option) any later version.
+#
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+# GNU General Public License for more details.
+#
+# You should have received a copy of the GNU General Public License
+# along with this program; if not, write to the Free Software
+# Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA.
+#
+
+#
+# This gets information about object hierarchies and signals
+# by compiling a small C program. CFLAGS and LDFLAGS must be
+# set appropriately before running this script.
+#
+
+use Getopt::Long;
+
+my $GTK_DOC_PREFIX=`pkg-config --variable prefix gtk-doc`;
+if ($GTK_DOC_PREFIX) {
+  chomp $GTK_DOC_PREFIX;
+  #print "Adding $GTK_DOC_PREFIX/share/gtk-doc/data to \@INC\n";
+  unshift @INC, "$GTK_DOC_PREFIX/share/gtk-doc/data";
+} else {
+  unshift @INC, '/usr/share/gtk-doc/data';
+}
+require "gtkdoc-common.pl";
+
+# Options
+
+# name of documentation module
+my $MODULE;
+my $OUTPUT_DIR;
+my $INSPECT_DIR;
+my $VERBOSE;
+my $PRINT_VERSION;
+my $PRINT_HELP;
+my $TYPE_INIT_FUNC="g_type_init ()";
+
+# --nogtkinit is deprecated, as it is the default now anyway.
+%optctl = (module => \$MODULE,
+           source => \$SOURCE,
+	   types => \$TYPES_FILE,
+	   nogtkinit => \$NO_GTK_INIT,
+	   'type-init-func' => \$TYPE_INIT_FUNC,
+	   'output-dir' => \$OUTPUT_DIR,
+	   'inspect-dir' => \$INSPECT_DIR,
+	   'verbose' => \$VERBOSE,
+	   'version' => \$PRINT_VERSION,
+	   'help' => \$PRINT_HELP);
+
+GetOptions(\%optctl, "module=s", "source=s", "types:s", "output-dir:s", "inspect-dir:s", "nogtkinit", "type-init-func:s", "verbose", "version", "help");
+
+if ($NO_GTK_INIT) {
+  # Do nothing. This just avoids a warning.
+  # the option is not used anymore
+}
+
+if ($PRINT_VERSION) {
+    print "1.5\n";
+    exit 0;
+}
+
+if (!$MODULE) {
+    $PRINT_HELP = 1;
+}
+
+if ($PRINT_HELP) {
+    print <$MODULE-scan.c") || die "Cannot open $MODULE-scan.c: $!\n";
+
+my $old_signals_filename = "$OUTPUT_DIR/$MODULE.signals";
+my $new_signals_filename = "$OUTPUT_DIR/$MODULE.signals.new";
+my $old_hierarchy_filename = "$OUTPUT_DIR/$MODULE.hierarchy";
+my $new_hierarchy_filename = "$OUTPUT_DIR/$MODULE.hierarchy.new";
+my $old_interfaces_filename = "$OUTPUT_DIR/$MODULE.interfaces";
+my $new_interfaces_filename = "$OUTPUT_DIR/$MODULE.interfaces.new";
+my $old_prerequisites_filename = "$OUTPUT_DIR/$MODULE.prerequisites";
+my $new_prerequisites_filename = "$OUTPUT_DIR/$MODULE.prerequisites.new";
+my $old_args_filename = "$OUTPUT_DIR/$MODULE.args";
+my $new_args_filename = "$OUTPUT_DIR/$MODULE.args.new";
+
+my $debug_log="g_message";
+if (!defined($VERBOSE) or $VERBOSE eq "0") {
+    $debug_log="//$debug_log";
+}
+
+# write a C program to scan the types
+
+$includes = "";
+@types = ();
+@impl_types = ();
+
+for () {
+    if (/^#include/) {
+	$includes .= $_;
+    } elsif (/^%/) {
+	next;
+    } elsif (/^\s*$/) {
+	next;
+    } elsif (/^type:(.*)$/) {
+	$t = $1;
+        chomp $t;
+	push @impl_types, $t;
+    } else {
+	chomp;
+	push @types, $_;
+    }
+}
+
+$ntypes = @types + @impl_types + 1;
+
+print OUTPUT <
+#include 
+#include 
+#include 
+
+#include 
+EOT
+
+if ($includes) {
+    print OUTPUT $includes;
+} else {
+    for (@types) {
+        print OUTPUT "extern GType $_ (void);\n";
+    }
+}
+
+print OUTPUT <
+#endif
+
+static GType *object_types = NULL;
+
+static GString *xmlstr = NULL;
+
+static const gchar*
+xmlprint (gint indent, const gchar *tag, const gchar *data)
+{
+  const gchar indent_str[] = "                                               ";
+
+  /* reset */
+  g_string_truncate (xmlstr, 0);
+  g_string_append_len (xmlstr, indent_str, MIN (indent, strlen (indent_str)));
+  g_string_append_printf (xmlstr, "<%s>", tag);
+
+  if (data) {
+    gchar *s;
+
+    s = g_markup_escape_text (data, -1);
+    g_string_append (xmlstr, s);
+    g_free (s);
+  }
+
+  g_string_append_printf (xmlstr, "\\n", tag);
+  return xmlstr->str;
+}
+
+static gint
+gst_feature_sort_compare (gconstpointer a, gconstpointer b)
+{
+  const gchar *name_a = gst_plugin_feature_get_name ((GstPluginFeature *) a);
+  const gchar *name_b = gst_plugin_feature_get_name ((GstPluginFeature *) b);
+  return strcmp (name_a, name_b);
+}
+
+static gint
+static_pad_template_compare (gconstpointer a, gconstpointer b)
+{
+  GstStaticPadTemplate *spt_a = (GstStaticPadTemplate *) a;
+  GstStaticPadTemplate *spt_b = (GstStaticPadTemplate *) b;
+
+  /* we want SINK before SRC (enum is UNKNOWN, SRC, SINK) */
+  if (spt_a->direction != spt_b->direction)
+    return spt_b->direction - spt_a->direction;
+
+  /* we want ALWAYS first, SOMETIMES second, REQUEST last
+   * (enum is ALWAYS, SOMETIMES, REQUEST) */
+  if (spt_a->presence != spt_b->presence)
+    return spt_a->presence - spt_b->presence;
+
+  return strcmp (spt_a->name_template, spt_b->name_template);
+}
+
+static GType *
+get_object_types (void)
+{
+    gpointer g_object_class;
+    GList *plugins = NULL;
+    GList *factories = NULL;
+    GList *l;
+    GstElementFactory *factory = NULL;
+    GType type;
+    gint i = 0;
+    gboolean reinspect;
+
+    /* get a list of features from plugins in our source module */
+    plugins = gst_registry_get_plugin_list (gst_registry_get ());
+
+    xmlstr = g_string_new ("");
+
+    reinspect = !g_file_test ("scanobj-build.stamp", G_FILE_TEST_EXISTS);
+
+    while (plugins) {
+      GList *features;
+      GstPlugin *plugin;
+      const gchar *source;
+      FILE *inspect = NULL;
+      gchar *inspect_name;
+
+      plugin = (GstPlugin *) (plugins->data);
+      plugins = g_list_next (plugins);
+      source = gst_plugin_get_source (plugin);
+      if (!source || strcmp (source, "$SOURCE") != 0) {
+        continue;
+      }
+
+      /* skip static coreelements plugin with pipeline and bin element factory */
+      if (gst_plugin_get_filename (plugin) == NULL)
+        continue;
+
+      $debug_log ("plugin: %s source: %s", gst_plugin_get_name (plugin), source);
+
+      if (reinspect) {
+        gchar *basename;
+
+        inspect_name = g_strdup_printf ("$INSPECT_DIR" G_DIR_SEPARATOR_S "plugin-%s.xml",
+            gst_plugin_get_name (plugin));
+        inspect = fopen (inspect_name, "w");
+        if (inspect == NULL) {
+          g_error ("Could not open %s for writing: %s\\n", inspect_name,
+              g_strerror (errno));
+        }
+        g_free (inspect_name);
+
+		  basename = g_path_get_basename (gst_plugin_get_filename (plugin));
+
+        /* output plugin data */
+        fputs ("\\n",inspect);
+        fputs (xmlprint(2, "name", gst_plugin_get_name (plugin)),inspect);
+        fputs (xmlprint(2, "description", gst_plugin_get_description (plugin)),inspect);
+        fputs (xmlprint(2, "filename", gst_plugin_get_filename (plugin)),inspect);
+        fputs (xmlprint(2, "basename", basename),inspect);
+        fputs (xmlprint(2, "version", gst_plugin_get_version (plugin)),inspect);
+        fputs (xmlprint(2, "license", gst_plugin_get_license (plugin)),inspect);
+        fputs (xmlprint(2, "source", gst_plugin_get_source (plugin)),inspect);
+        fputs (xmlprint(2, "package", gst_plugin_get_package (plugin)),inspect);
+        fputs (xmlprint(2, "origin", gst_plugin_get_origin (plugin)),inspect);
+        fputs ("  \\n", inspect);
+
+		  g_free (basename);
+      }
+
+      features =
+          gst_registry_get_feature_list_by_plugin (gst_registry_get (),
+          gst_plugin_get_name (plugin));
+
+      /* sort factories by feature->name */
+      features = g_list_sort (features, gst_feature_sort_compare);
+
+      while (features) {
+        GstPluginFeature *feature;
+        feature = GST_PLUGIN_FEATURE (features->data);
+        feature = gst_plugin_feature_load (feature);
+        if (!feature) {
+          g_warning ("Could not load plugin feature %s",
+                     gst_plugin_feature_get_name (feature));
+        }
+
+        if (GST_IS_ELEMENT_FACTORY (feature)) {
+          const gchar *pad_dir[] = { "unknown","source","sink" };
+          const gchar *pad_pres[] = { "always","sometimes","request" };
+          GList *pads, *pad;
+
+          $debug_log ("  feature: %s", gst_plugin_feature_get_name (feature));
+
+          factory = GST_ELEMENT_FACTORY (feature);
+          factories = g_list_prepend (factories, factory);
+
+          if (reinspect) {
+            /* output element data */
+            fputs ("    \\n", inspect);
+            fputs (xmlprint(6, "name", gst_plugin_feature_get_name (feature)),inspect);
+            fputs (xmlprint(6, "longname", gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_LONGNAME)),inspect);
+            fputs (xmlprint(6, "class", gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_KLASS)),inspect);
+            fputs (xmlprint(6, "description", gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_DESCRIPTION)),inspect);
+            fputs (xmlprint(6, "author", gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_AUTHOR)),inspect);
+            fputs ("      \\n", inspect);
+
+            /* output pad-template data */
+            pads = g_list_copy ((GList *) gst_element_factory_get_static_pad_templates (factory));
+            pads = g_list_sort (pads, static_pad_template_compare);
+            for (pad = pads; pad != NULL; pad = pad->next) {
+              GstStaticPadTemplate *pt = pad->data;
+
+              fputs ("        \\n", inspect);
+              fputs (xmlprint(10, "name", pt->name_template),inspect);
+              fputs (xmlprint(10, "direction", pad_dir[pt->direction]),inspect);
+              fputs (xmlprint(10, "presence", pad_pres[pt->presence]),inspect);
+              fputs (xmlprint(10, "details", pt->static_caps.string),inspect);
+              fputs ("        \\n", inspect);
+            }
+            g_list_free (pads);
+            fputs ("      \\n    \\n", inspect);
+          }
+        }
+        features = g_list_next (features);
+      }
+
+      if (reinspect) {
+        fputs ("  \\n", inspect);
+        fclose (inspect);
+      }
+    }
+
+    g_string_free (xmlstr, TRUE);
+
+    $debug_log ("number of element factories: %d", g_list_length (factories));
+
+    /* allocate the object_types array to hold them */
+    object_types = g_new0 (GType, g_list_length (factories)+$ntypes+1);
+
+    l = factories;
+    i = 0;
+
+    /* fill it */
+    while (l) {
+      factory = GST_ELEMENT_FACTORY (l->data);
+      type = gst_element_factory_get_element_type (factory);
+      if (type != 0) {
+        $debug_log ("adding type for factory %s", gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_LONGNAME));
+        object_types[i++] = type;
+      } else {
+        g_message ("type info for factory %s not found",
+            gst_element_factory_get_metadata (factory, GST_ELEMENT_METADATA_LONGNAME));
+      }
+      l = g_list_next (l);
+    }
+
+EOT
+
+# get_type functions:
+for (@types) {
+print OUTPUT < uppercase with '_'
+   * GFileMonitor -> file_monitor
+   * GIOExtensionPoint -> extension_point
+   * GtkTreeView -> tree_view
+   * if 2nd char is upper case too
+   *   search for first lower case and go back one char
+   * else
+   *   search for next upper case
+   */
+  if (!strncmp (object_name, "Gtk", 3))
+      object_arg = object_name + 3;
+  else if (!strncmp (object_name, "Gnome", 5))
+      object_arg = object_name + 5;
+  else
+      object_arg = object_name;
+
+  object_arg_lower = g_ascii_strdown (object_arg, -1);
+  sprintf (pos, "*%s\\n", object_arg_lower);
+  pos += strlen (pos);
+  if (!strncmp (object_arg_lower, "widget", 6))
+    widget_num = 2;
+  g_free(object_arg_lower);
+
+  /* Convert signal name to use underscores rather than dashes '-'. */
+  strncpy (signal_name, query_info.signal_name, 127);
+  signal_name[127] = '\\0';
+  for (i = 0; signal_name[i]; i++)
+    {
+      if (signal_name[i] == '-')
+	signal_name[i] = '_';
+    }
+
+  /* Output the signal parameters. */
+  for (param = 0; param < query_info.n_params; param++)
+    {
+      type_name = get_type_name (query_info.param_types[param] & ~G_SIGNAL_TYPE_STATIC_SCOPE, &is_pointer);
+
+      /* Most arguments to the callback are called "arg1", "arg2", etc.
+         GtkWidgets are called "widget", "widget2", ...
+         GtkCallbacks are called "callback", "callback2", ... */
+      if (!strcmp (type_name, "GtkWidget"))
+        {
+          arg_name = "widget";
+          arg_num = &widget_num;
+        }
+      else if (!strcmp (type_name, "GtkCallback")
+               || !strcmp (type_name, "GtkCCallback"))
+        {
+          arg_name = "callback";
+          arg_num = &callback_num;
+        }
+      else
+        {
+          arg_name = "arg";
+          arg_num = ¶m_num;
+        }
+      sprintf (pos, "%s ", type_name);
+      pos += strlen (pos);
+
+      if (!arg_num || *arg_num == 0)
+        sprintf (pos, "%s%s\\n", is_pointer ? "*" : " ", arg_name);
+      else
+        sprintf (pos, "%s%s%i\\n", is_pointer ? "*" : " ", arg_name,
+                 *arg_num);
+      pos += strlen (pos);
+
+      if (arg_num)
+        {
+          if (*arg_num == 0)
+            *arg_num = 2;
+          else
+            *arg_num += 1;
+        }
+    }
+
+  pos = flags;
+  /* We use one-character flags for simplicity. */
+  if (query_info.signal_flags & G_SIGNAL_RUN_FIRST)
+    *pos++ = 'f';
+  if (query_info.signal_flags & G_SIGNAL_RUN_LAST)
+    *pos++ = 'l';
+  if (query_info.signal_flags & G_SIGNAL_RUN_CLEANUP)
+    *pos++ = 'c';
+  if (query_info.signal_flags & G_SIGNAL_NO_RECURSE)
+    *pos++ = 'r';
+  if (query_info.signal_flags & G_SIGNAL_DETAILED)
+    *pos++ = 'd';
+  if (query_info.signal_flags & G_SIGNAL_ACTION)
+    *pos++ = 'a';
+  if (query_info.signal_flags & G_SIGNAL_NO_HOOKS)
+    *pos++ = 'h';
+  *pos = 0;
+
+  /* Output the return type and function name. */
+  ret_type = get_type_name (query_info.return_type & ~G_SIGNAL_TYPE_STATIC_SCOPE, &is_pointer);
+
+  fprintf (fp,
+	   "\\n%s::%s\\n%s%s\\n%s\\n%s\\n\\n",
+	   object_name, query_info.signal_name, ret_type, is_pointer ? "*" : "", flags, buffer);
+}
+
+
+/* Returns the type name to use for a signal argument or return value, given
+   the GtkType from the signal info. It also sets is_pointer to TRUE if the
+   argument needs a '*' since it is a pointer. */
+static const gchar *
+get_type_name (GType type, gboolean * is_pointer)
+{
+  const gchar *type_name;
+
+  *is_pointer = FALSE;
+  type_name = g_type_name (type);
+
+  switch (type) {
+  case G_TYPE_NONE:
+  case G_TYPE_CHAR:
+  case G_TYPE_UCHAR:
+  case G_TYPE_BOOLEAN:
+  case G_TYPE_INT:
+  case G_TYPE_UINT:
+  case G_TYPE_LONG:
+  case G_TYPE_ULONG:
+  case G_TYPE_FLOAT:
+  case G_TYPE_DOUBLE:
+  case G_TYPE_POINTER:
+    /* These all have normal C type names so they are OK. */
+    return type_name;
+
+  case G_TYPE_STRING:
+    /* A GtkString is really a gchar*. */
+    *is_pointer = TRUE;
+    return "gchar";
+
+  case G_TYPE_ENUM:
+  case G_TYPE_FLAGS:
+    /* We use a gint for both of these. Hopefully a subtype with a decent
+       name will be registered and used instead, as GTK+ does itself. */
+    return "gint";
+
+  case G_TYPE_BOXED:
+    /* The boxed type shouldn't be used itself, only subtypes. Though we
+       return 'gpointer' just in case. */
+    return "gpointer";
+
+  case G_TYPE_PARAM:
+    /* A GParam is really a GParamSpec*. */
+    *is_pointer = TRUE;
+    return "GParamSpec";
+
+#if GLIB_CHECK_VERSION (2, 25, 9)
+  case G_TYPE_VARIANT:
+    *is_pointer = TRUE;
+    return "GVariant";
+#endif
+
+default:
+    break;
+  }
+
+  /* For all GObject subclasses we can use the class name with a "*",
+     e.g. 'GtkWidget *'. */
+  if (g_type_is_a (type, G_TYPE_OBJECT))
+    *is_pointer = TRUE;
+
+  /* Also catch non GObject root types */
+  if (G_TYPE_IS_CLASSED (type))
+    *is_pointer = TRUE;
+
+  /* All boxed subtypes will be pointers as well. */
+  /* Exception: GStrv */
+  if (g_type_is_a (type, G_TYPE_BOXED) &&
+      !g_type_is_a (type, G_TYPE_STRV))
+    *is_pointer = TRUE;
+
+  /* All pointer subtypes will be pointers as well. */
+  if (g_type_is_a (type, G_TYPE_POINTER))
+    *is_pointer = TRUE;
+
+  /* But enums are not */
+  if (g_type_is_a (type, G_TYPE_ENUM) ||
+      g_type_is_a (type, G_TYPE_FLAGS))
+    *is_pointer = FALSE;
+
+  return type_name;
+}
+
+
+/* This outputs the hierarchy of all objects which have been initialized,
+   i.e. by calling their XXX_get_type() initialization function. */
+static void
+output_object_hierarchy (void)
+{
+  FILE *fp;
+  gint i,j;
+  GType root, type;
+  GType root_types[$ntypes] = { G_TYPE_INVALID, };
+
+  fp = fopen (hierarchy_filename, "w");
+  if (fp == NULL)
+    {
+      g_warning ("Couldn't open output file: %s : %s", hierarchy_filename, g_strerror(errno));
+      return;
+    }
+  output_hierarchy (fp, G_TYPE_OBJECT, 0);
+  output_hierarchy (fp, G_TYPE_INTERFACE, 0);
+
+  for (i=0; object_types[i]; i++) {
+    root = object_types[i];
+    while ((type = g_type_parent (root))) {
+      root = type;
+    }
+    if ((root != G_TYPE_OBJECT) && (root != G_TYPE_INTERFACE)) {
+      for (j=0; root_types[j]; j++) {
+        if (root == root_types[j]) {
+          root = G_TYPE_INVALID; break;
+        }
+      }
+      if(root) {
+        root_types[j] = root;
+        output_hierarchy (fp, root, 0);
+      }
+    }
+  }
+
+  fclose (fp);
+}
+
+static int
+compare_types (const void *a, const void *b)
+{
+  const char *na = g_type_name (*((GType *)a));
+  const char *nb = g_type_name (*((GType *)b));
+
+  return g_strcmp0 (na, nb);
+}
+
+
+/* This is called recursively to output the hierarchy of a object. */
+static void
+output_hierarchy (FILE  *fp,
+		  GType  type,
+		  guint   level)
+{
+  guint i;
+  GType *children;
+  guint n_children;
+
+  if (!type)
+    return;
+
+  for (i = 0; i < level; i++)
+    fprintf (fp, "  ");
+  fprintf (fp, "%s\\n", g_type_name (type));
+
+  children = g_type_children (type, &n_children);
+  qsort (children, n_children, sizeof (GType), compare_types);
+
+
+  for (i=0; i < n_children; i++)
+    output_hierarchy (fp, children[i], level + 1);
+
+  g_free (children);
+}
+
+static void output_object_interfaces (void)
+{
+  guint i;
+  FILE *fp;
+
+  fp = fopen (interfaces_filename, "w");
+  if (fp == NULL)
+    {
+      g_warning ("Couldn't open output file: %s : %s", interfaces_filename, g_strerror(errno));
+      return;
+    }
+  output_interfaces (fp, G_TYPE_OBJECT);
+
+  for (i = 0; object_types[i]; i++)
+    {
+      if (!g_type_parent (object_types[i]) &&
+          (object_types[i] != G_TYPE_OBJECT) &&
+          G_TYPE_IS_INSTANTIATABLE (object_types[i]))
+        {
+          output_interfaces (fp, object_types[i]);
+        }
+    }
+  fclose (fp);
+}
+
+static void
+output_interfaces (FILE  *fp,
+		   GType  type)
+{
+  guint i;
+  GType *children, *interfaces;
+  guint n_children, n_interfaces;
+
+  if (!type)
+    return;
+
+  interfaces = g_type_interfaces (type, &n_interfaces);
+
+  if (n_interfaces > 0)
+    {
+      fprintf (fp, "%s", g_type_name (type));
+      for (i=0; i < n_interfaces; i++)
+          fprintf (fp, " %s", g_type_name (interfaces[i]));
+      fprintf (fp, "\\n");
+     }
+  g_free (interfaces);
+
+  children = g_type_children (type, &n_children);
+
+  for (i=0; i < n_children; i++)
+    output_interfaces (fp, children[i]);
+
+  g_free (children);
+}
+
+static void output_interface_prerequisites (void)
+{
+  FILE *fp;
+
+  fp = fopen (prerequisites_filename, "w");
+  if (fp == NULL)
+    {
+      g_warning ("Couldn't open output file: %s : %s", prerequisites_filename, g_strerror(errno));
+      return;
+    }
+  output_prerequisites (fp, G_TYPE_INTERFACE);
+  fclose (fp);
+}
+
+static void
+output_prerequisites (FILE  *fp,
+		      GType  type)
+{
+#if GLIB_CHECK_VERSION(2,1,0)
+  guint i;
+  GType *children, *prerequisites;
+  guint n_children, n_prerequisites;
+
+  if (!type)
+    return;
+
+  prerequisites = g_type_interface_prerequisites (type, &n_prerequisites);
+
+  if (n_prerequisites > 0)
+    {
+      fprintf (fp, "%s", g_type_name (type));
+      for (i=0; i < n_prerequisites; i++)
+          fprintf (fp, " %s", g_type_name (prerequisites[i]));
+      fprintf (fp, "\\n");
+     }
+  g_free (prerequisites);
+
+  children = g_type_children (type, &n_children);
+
+  for (i=0; i < n_children; i++)
+    output_prerequisites (fp, children[i]);
+
+  g_free (children);
+#endif
+}
+
+static void
+output_args (void)
+{
+  FILE *fp;
+  gint i;
+
+  fp = fopen (args_filename, "w");
+  if (fp == NULL)
+    {
+      g_warning ("Couldn't open output file: %s : %s", args_filename, g_strerror(errno));
+      return;
+    }
+
+  for (i = 0; object_types[i]; i++) {
+    output_object_args (fp, object_types[i]);
+  }
+
+  fclose (fp);
+}
+
+static gint
+compare_param_specs (const void *a, const void *b)
+{
+  GParamSpec *spec_a = *(GParamSpec **)a;
+  GParamSpec *spec_b = *(GParamSpec **)b;
+
+  return strcmp (g_param_spec_get_name (spec_a), g_param_spec_get_name (spec_b));
+}
+
+/* Its common to have unsigned properties restricted
+ * to the signed range. Therefore we make this look
+ * a bit nicer by spelling out the max constants.
+ */
+
+/* Don't use "==" with floats, it might trigger a gcc warning.  */
+#define GTKDOC_COMPARE_FLOAT(x, y) (x <= y && x >= y)
+
+static gchar*
+describe_double_constant (gdouble value)
+{
+  gchar *desc;
+
+  if (GTKDOC_COMPARE_FLOAT (value, G_MAXDOUBLE))
+    desc = g_strdup ("G_MAXDOUBLE");
+  else if (GTKDOC_COMPARE_FLOAT (value, G_MINDOUBLE))
+    desc = g_strdup ("G_MINDOUBLE");
+  else if (GTKDOC_COMPARE_FLOAT (value, -G_MAXDOUBLE))
+    desc = g_strdup ("-G_MAXDOUBLE");
+  else if (GTKDOC_COMPARE_FLOAT (value, G_MAXFLOAT))
+    desc = g_strdup ("G_MAXFLOAT");
+  else if (GTKDOC_COMPARE_FLOAT (value, G_MINFLOAT))
+    desc = g_strdup ("G_MINFLOAT");
+  else if (GTKDOC_COMPARE_FLOAT (value, -G_MAXFLOAT))
+    desc = g_strdup ("-G_MAXFLOAT");
+  else{
+    /* make sure floats are output with a decimal dot irrespective of
+    * current locale. Use formatd since we want human-readable numbers
+    * and do not need the exact same bit representation when deserialising */
+    desc = g_malloc0 (G_ASCII_DTOSTR_BUF_SIZE);
+    g_ascii_formatd (desc, G_ASCII_DTOSTR_BUF_SIZE, "%g", value);
+  }
+
+  return desc;
+}
+
+static gchar*
+describe_signed_constant (gsize size, gint64 value)
+{
+  gchar *desc = NULL;
+
+  switch (size) {
+    case 8:
+      if (value == G_MAXINT64)
+        desc = g_strdup ("G_MAXINT64");
+      else if (value == G_MININT64)
+        desc = g_strdup ("G_MININT64");
+      /* fall through */
+    case 4:
+      if (sizeof (int) == 4) {
+        if (value == G_MAXINT)
+          desc = g_strdup ("G_MAXINT");
+        else if (value == G_MININT)
+          desc = g_strdup ("G_MININT");
+        else if (value == (gint64)G_MAXUINT)
+          desc = g_strdup ("G_MAXUINT");
+      }
+      if (value == G_MAXLONG)
+        desc = g_strdup ("G_MAXLONG");
+      else if (value == G_MINLONG)
+        desc = g_strdup ("G_MINLONG");
+      else if (value == (gint64)G_MAXULONG)
+        desc = g_strdup ("G_MAXULONG");
+      /* fall through */
+    case 2:
+      if (sizeof (int) == 2) {
+        if (value == G_MAXINT)
+          desc = g_strdup ("G_MAXINT");
+        else if (value == G_MININT)
+          desc = g_strdup ("G_MININT");
+        else if (value == (gint64)G_MAXUINT)
+          desc = g_strdup ("G_MAXUINT");
+      }
+      break;
+    default:
+      break;
+  }
+  if (!desc)
+    desc = g_strdup_printf ("%" G_GINT64_FORMAT, value);
+
+  return desc;
+}
+
+static gchar*
+describe_unsigned_constant (gsize size, guint64 value)
+{
+  gchar *desc = NULL;
+
+  switch (size) {
+    case 8:
+      if (value == G_MAXINT64)
+        desc = g_strdup ("G_MAXINT64");
+      else if (value == G_MAXUINT64)
+        desc = g_strdup ("G_MAXUINT64");
+      /* fall through */
+    case 4:
+      if (sizeof (int) == 4) {
+        if (value == (guint64)G_MAXINT)
+          desc = g_strdup ("G_MAXINT");
+        else if (value == G_MAXUINT)
+          desc = g_strdup ("G_MAXUINT");
+      }
+      if (value == (guint64)G_MAXLONG)
+        desc = g_strdup ("G_MAXLONG");
+      else if (value == G_MAXULONG)
+        desc = g_strdup ("G_MAXULONG");
+      /* fall through */
+    case 2:
+      if (sizeof (int) == 2) {
+        if (value == (guint64)G_MAXINT)
+          desc = g_strdup ("G_MAXINT");
+        else if (value == G_MAXUINT)
+          desc = g_strdup ("G_MAXUINT");
+      }
+      break;
+    default:
+      break;
+  }
+  if (!desc)
+    desc = g_strdup_printf ("%" G_GUINT64_FORMAT, value);
+
+  return desc;
+}
+
+static gchar*
+describe_type (GParamSpec *spec)
+{
+  gchar *desc;
+  gchar *lower;
+  gchar *upper;
+
+  if (G_IS_PARAM_SPEC_CHAR (spec))
+    {
+      GParamSpecChar *pspec = G_PARAM_SPEC_CHAR (spec);
+
+      lower = describe_signed_constant (sizeof(gchar), pspec->minimum);
+      upper = describe_signed_constant (sizeof(gchar), pspec->maximum);
+      if (pspec->minimum == G_MININT8 && pspec->maximum == G_MAXINT8)
+	desc = g_strdup ("");
+      else if (pspec->minimum == G_MININT8)
+	desc = g_strdup_printf ("<= %s", upper);
+      else if (pspec->maximum == G_MAXINT8)
+	desc = g_strdup_printf (">= %s", lower);
+      else
+	desc = g_strdup_printf ("[%s,%s]", lower, upper);
+      g_free (lower);
+      g_free (upper);
+    }
+  else if (G_IS_PARAM_SPEC_UCHAR (spec))
+    {
+      GParamSpecUChar *pspec = G_PARAM_SPEC_UCHAR (spec);
+
+      lower = describe_unsigned_constant (sizeof(guchar), pspec->minimum);
+      upper = describe_unsigned_constant (sizeof(guchar), pspec->maximum);
+      if (pspec->minimum == 0 && pspec->maximum == G_MAXUINT8)
+	desc = g_strdup ("");
+      else if (pspec->minimum == 0)
+	desc = g_strdup_printf ("<= %s", upper);
+      else if (pspec->maximum == G_MAXUINT8)
+	desc = g_strdup_printf (">= %s", lower);
+      else
+	desc = g_strdup_printf ("[%s,%s]", lower, upper);
+      g_free (lower);
+      g_free (upper);
+    }
+  else if (G_IS_PARAM_SPEC_INT (spec))
+    {
+      GParamSpecInt *pspec = G_PARAM_SPEC_INT (spec);
+
+      lower = describe_signed_constant (sizeof(gint), pspec->minimum);
+      upper = describe_signed_constant (sizeof(gint), pspec->maximum);
+      if (pspec->minimum == G_MININT && pspec->maximum == G_MAXINT)
+	desc = g_strdup ("");
+      else if (pspec->minimum == G_MININT)
+	desc = g_strdup_printf ("<= %s", upper);
+      else if (pspec->maximum == G_MAXINT)
+	desc = g_strdup_printf (">= %s", lower);
+      else
+	desc = g_strdup_printf ("[%s,%s]", lower, upper);
+      g_free (lower);
+      g_free (upper);
+    }
+  else if (G_IS_PARAM_SPEC_UINT (spec))
+    {
+      GParamSpecUInt *pspec = G_PARAM_SPEC_UINT (spec);
+
+      lower = describe_unsigned_constant (sizeof(guint), pspec->minimum);
+      upper = describe_unsigned_constant (sizeof(guint), pspec->maximum);
+      if (pspec->minimum == 0 && pspec->maximum == G_MAXUINT)
+	desc = g_strdup ("");
+      else if (pspec->minimum == 0)
+	desc = g_strdup_printf ("<= %s", upper);
+      else if (pspec->maximum == G_MAXUINT)
+	desc = g_strdup_printf (">= %s", lower);
+      else
+	desc = g_strdup_printf ("[%s,%s]", lower, upper);
+      g_free (lower);
+      g_free (upper);
+    }
+  else if (G_IS_PARAM_SPEC_LONG (spec))
+    {
+      GParamSpecLong *pspec = G_PARAM_SPEC_LONG (spec);
+
+      lower = describe_signed_constant (sizeof(glong), pspec->minimum);
+      upper = describe_signed_constant (sizeof(glong), pspec->maximum);
+      if (pspec->minimum == G_MINLONG && pspec->maximum == G_MAXLONG)
+	desc = g_strdup ("");
+      else if (pspec->minimum == G_MINLONG)
+	desc = g_strdup_printf ("<= %s", upper);
+      else if (pspec->maximum == G_MAXLONG)
+	desc = g_strdup_printf (">= %s", lower);
+      else
+	desc = g_strdup_printf ("[%s,%s]", lower, upper);
+      g_free (lower);
+      g_free (upper);
+    }
+  else if (G_IS_PARAM_SPEC_ULONG (spec))
+    {
+      GParamSpecULong *pspec = G_PARAM_SPEC_ULONG (spec);
+
+      lower = describe_unsigned_constant (sizeof(gulong), pspec->minimum);
+      upper = describe_unsigned_constant (sizeof(gulong), pspec->maximum);
+      if (pspec->minimum == 0 && pspec->maximum == G_MAXULONG)
+	desc = g_strdup ("");
+      else if (pspec->minimum == 0)
+	desc = g_strdup_printf ("<= %s", upper);
+      else if (pspec->maximum == G_MAXULONG)
+	desc = g_strdup_printf (">= %s", lower);
+      else
+	desc = g_strdup_printf ("[%s,%s]", lower, upper);
+      g_free (lower);
+      g_free (upper);
+    }
+  else if (G_IS_PARAM_SPEC_INT64 (spec))
+    {
+      GParamSpecInt64 *pspec = G_PARAM_SPEC_INT64 (spec);
+
+      lower = describe_signed_constant (sizeof(gint64), pspec->minimum);
+      upper = describe_signed_constant (sizeof(gint64), pspec->maximum);
+      if (pspec->minimum == G_MININT64 && pspec->maximum == G_MAXINT64)
+	desc = g_strdup ("");
+      else if (pspec->minimum == G_MININT64)
+	desc = g_strdup_printf ("<= %s", upper);
+      else if (pspec->maximum == G_MAXINT64)
+	desc = g_strdup_printf (">= %s", lower);
+      else
+	desc = g_strdup_printf ("[%s,%s]", lower, upper);
+      g_free (lower);
+      g_free (upper);
+    }
+  else if (G_IS_PARAM_SPEC_UINT64 (spec))
+    {
+      GParamSpecUInt64 *pspec = G_PARAM_SPEC_UINT64 (spec);
+
+      lower = describe_unsigned_constant (sizeof(guint64), pspec->minimum);
+      upper = describe_unsigned_constant (sizeof(guint64), pspec->maximum);
+      if (pspec->minimum == 0 && pspec->maximum == G_MAXUINT64)
+	desc = g_strdup ("");
+      else if (pspec->minimum == 0)
+	desc = g_strdup_printf ("<= %s", upper);
+      else if (pspec->maximum == G_MAXUINT64)
+	desc = g_strdup_printf (">= %s", lower);
+      else
+	desc = g_strdup_printf ("[%s,%s]", lower, upper);
+      g_free (lower);
+      g_free (upper);
+    }
+  else if (G_IS_PARAM_SPEC_FLOAT (spec))
+    {
+      GParamSpecFloat *pspec = G_PARAM_SPEC_FLOAT (spec);
+
+      lower = describe_double_constant (pspec->minimum);
+      upper = describe_double_constant (pspec->maximum);
+      if (GTKDOC_COMPARE_FLOAT (pspec->minimum, -G_MAXFLOAT))
+	{
+	  if (GTKDOC_COMPARE_FLOAT (pspec->maximum, G_MAXFLOAT))
+	    desc = g_strdup ("");
+	  else
+	    desc = g_strdup_printf ("<= %s", upper);
+	}
+      else if (GTKDOC_COMPARE_FLOAT (pspec->maximum, G_MAXFLOAT))
+	desc = g_strdup_printf (">= %s", lower);
+      else
+	desc = g_strdup_printf ("[%s,%s]", lower, upper);
+      g_free (lower);
+      g_free (upper);
+    }
+  else if (G_IS_PARAM_SPEC_DOUBLE (spec))
+    {
+      GParamSpecDouble *pspec = G_PARAM_SPEC_DOUBLE (spec);
+
+      lower = describe_double_constant (pspec->minimum);
+      upper = describe_double_constant (pspec->maximum);
+      if (GTKDOC_COMPARE_FLOAT (pspec->minimum, -G_MAXDOUBLE))
+	{
+	  if (GTKDOC_COMPARE_FLOAT (pspec->maximum, G_MAXDOUBLE))
+	    desc = g_strdup ("");
+	  else
+	    desc = g_strdup_printf ("<= %s", upper);
+	}
+      else if (GTKDOC_COMPARE_FLOAT (pspec->maximum, G_MAXDOUBLE))
+	desc = g_strdup_printf (">= %s", lower);
+      else
+	desc = g_strdup_printf ("[%s,%s]", lower, upper);
+      g_free (lower);
+      g_free (upper);
+    }
+#if GLIB_CHECK_VERSION (2, 12, 0)
+  else if (G_IS_PARAM_SPEC_GTYPE (spec))
+    {
+      GParamSpecGType *pspec = G_PARAM_SPEC_GTYPE (spec);
+      gboolean is_pointer;
+
+      desc = g_strdup (get_type_name (pspec->is_a_type, &is_pointer));
+    }
+#endif
+#if GLIB_CHECK_VERSION (2, 25, 9)
+  else if (G_IS_PARAM_SPEC_VARIANT (spec))
+    {
+      GParamSpecVariant *pspec = G_PARAM_SPEC_VARIANT (spec);
+      gchar *variant_type;
+
+      variant_type = g_variant_type_dup_string (pspec->type);
+      desc = g_strdup_printf ("GVariant<%s>", variant_type);
+      g_free (variant_type);
+    }
+#endif
+  else
+    {
+      desc = g_strdup ("");
+    }
+
+  return desc;
+}
+
+static gchar*
+describe_default (GParamSpec *spec)
+{
+  gchar *desc;
+
+  if (G_IS_PARAM_SPEC_CHAR (spec))
+    {
+      GParamSpecChar *pspec = G_PARAM_SPEC_CHAR (spec);
+
+      desc = g_strdup_printf ("%d", pspec->default_value);
+    }
+  else if (G_IS_PARAM_SPEC_UCHAR (spec))
+    {
+      GParamSpecUChar *pspec = G_PARAM_SPEC_UCHAR (spec);
+
+      desc = g_strdup_printf ("%u", pspec->default_value);
+    }
+  else if (G_IS_PARAM_SPEC_BOOLEAN (spec))
+    {
+      GParamSpecBoolean *pspec = G_PARAM_SPEC_BOOLEAN (spec);
+
+      desc = g_strdup_printf ("%s", pspec->default_value ? "TRUE" : "FALSE");
+    }
+  else if (G_IS_PARAM_SPEC_INT (spec))
+    {
+      GParamSpecInt *pspec = G_PARAM_SPEC_INT (spec);
+
+      desc = g_strdup_printf ("%d", pspec->default_value);
+    }
+  else if (G_IS_PARAM_SPEC_UINT (spec))
+    {
+      GParamSpecUInt *pspec = G_PARAM_SPEC_UINT (spec);
+
+      desc = g_strdup_printf ("%u", pspec->default_value);
+    }
+  else if (G_IS_PARAM_SPEC_LONG (spec))
+    {
+      GParamSpecLong *pspec = G_PARAM_SPEC_LONG (spec);
+
+      desc = g_strdup_printf ("%ld", pspec->default_value);
+    }
+  else if (G_IS_PARAM_SPEC_LONG (spec))
+    {
+      GParamSpecULong *pspec = G_PARAM_SPEC_ULONG (spec);
+
+      desc = g_strdup_printf ("%lu", pspec->default_value);
+    }
+  else if (G_IS_PARAM_SPEC_INT64 (spec))
+    {
+      GParamSpecInt64 *pspec = G_PARAM_SPEC_INT64 (spec);
+
+      desc = g_strdup_printf ("%" G_GINT64_FORMAT, pspec->default_value);
+    }
+  else if (G_IS_PARAM_SPEC_UINT64 (spec))
+    {
+      GParamSpecUInt64 *pspec = G_PARAM_SPEC_UINT64 (spec);
+
+      desc = g_strdup_printf ("%" G_GUINT64_FORMAT, pspec->default_value);
+    }
+  else if (G_IS_PARAM_SPEC_UNICHAR (spec))
+    {
+      GParamSpecUnichar *pspec = G_PARAM_SPEC_UNICHAR (spec);
+
+      if (g_unichar_isprint (pspec->default_value))
+	desc = g_strdup_printf ("'%c'", pspec->default_value);
+      else
+	desc = g_strdup_printf ("%u", pspec->default_value);
+    }
+  else if (G_IS_PARAM_SPEC_ENUM (spec))
+    {
+      GParamSpecEnum *pspec = G_PARAM_SPEC_ENUM (spec);
+
+      GEnumValue *value = g_enum_get_value (pspec->enum_class, pspec->default_value);
+      if (value)
+	desc = g_strdup_printf ("%s", value->value_name);
+      else
+	desc = g_strdup_printf ("%d", pspec->default_value);
+    }
+  else if (G_IS_PARAM_SPEC_FLAGS (spec))
+    {
+      GParamSpecFlags *pspec = G_PARAM_SPEC_FLAGS (spec);
+      guint default_value;
+      GString *acc;
+
+      default_value = pspec->default_value;
+      acc = g_string_new ("");
+
+      while (default_value)
+	{
+	  GFlagsValue *value = g_flags_get_first_value (pspec->flags_class, default_value);
+
+	  if (!value)
+	    break;
+
+	  if (acc->len > 0)
+	    g_string_append (acc, "|");
+	  g_string_append (acc, value->value_name);
+
+	  default_value &= ~value->value;
+	}
+
+      if (default_value == 0)
+	desc = g_string_free (acc, FALSE);
+      else
+	{
+	  desc = g_strdup_printf ("%d", pspec->default_value);
+	  g_string_free (acc, TRUE);
+	}
+    }
+  else if (G_IS_PARAM_SPEC_FLOAT (spec))
+    {
+      GParamSpecFloat *pspec = G_PARAM_SPEC_FLOAT (spec);
+
+      /* make sure floats are output with a decimal dot irrespective of
+       * current locale. Use formatd since we want human-readable numbers
+       * and do not need the exact same bit representation when deserialising */
+      desc = g_malloc0 (G_ASCII_DTOSTR_BUF_SIZE);
+      g_ascii_formatd (desc, G_ASCII_DTOSTR_BUF_SIZE, "%g",
+          pspec->default_value);
+    }
+  else if (G_IS_PARAM_SPEC_DOUBLE (spec))
+    {
+      GParamSpecDouble *pspec = G_PARAM_SPEC_DOUBLE (spec);
+
+      /* make sure floats are output with a decimal dot irrespective of
+       * current locale. Use formatd since we want human-readable numbers
+       * and do not need the exact same bit representation when deserialising */
+      desc = g_malloc0 (G_ASCII_DTOSTR_BUF_SIZE);
+      g_ascii_formatd (desc, G_ASCII_DTOSTR_BUF_SIZE, "%g",
+          pspec->default_value);
+    }
+  else if (G_IS_PARAM_SPEC_STRING (spec))
+    {
+      GParamSpecString *pspec = G_PARAM_SPEC_STRING (spec);
+
+      if (pspec->default_value)
+	{
+	  gchar *esc = g_strescape (pspec->default_value, NULL);
+
+	  desc = g_strdup_printf ("\\"%s\\"", esc);
+
+	  g_free (esc);
+	}
+      else
+	desc = g_strdup_printf ("NULL");
+    }
+  else
+    {
+      desc = g_strdup ("");
+    }
+
+  return desc;
+}
+
+
+static void
+output_object_args (FILE *fp, GType object_type)
+{
+  gpointer class;
+  const gchar *object_class_name;
+  guint arg;
+  gchar flags[16], *pos;
+  GParamSpec **properties;
+  guint n_properties;
+  gboolean child_prop;
+  gboolean style_prop;
+  gboolean is_pointer;
+  const gchar *type_name;
+  gchar *type_desc;
+  gchar *default_value;
+
+  if (G_TYPE_IS_OBJECT (object_type))
+    {
+      class = g_type_class_peek (object_type);
+      if (!class)
+	return;
+
+      properties = g_object_class_list_properties (class, &n_properties);
+    }
+#if GLIB_MAJOR_VERSION > 2 || (GLIB_MAJOR_VERSION == 2 && GLIB_MINOR_VERSION >= 3)
+  else if (G_TYPE_IS_INTERFACE (object_type))
+    {
+      class = g_type_default_interface_ref (object_type);
+
+      if (!class)
+	return;
+
+      properties = g_object_interface_list_properties (class, &n_properties);
+    }
+#endif
+  else
+    return;
+
+  object_class_name = g_type_name (object_type);
+
+  child_prop = FALSE;
+  style_prop = FALSE;
+
+  while (TRUE) {
+    qsort (properties, n_properties, sizeof (GParamSpec *), compare_param_specs);
+    for (arg = 0; arg < n_properties; arg++)
+      {
+        GParamSpec *spec = properties[arg];
+        const gchar *nick, *blurb, *dot;
+
+        if (spec->owner_type != object_type)
+          continue;
+
+        pos = flags;
+        /* We use one-character flags for simplicity. */
+        if (child_prop && !style_prop)
+   	  *pos++ = 'c';
+        if (style_prop)
+   	  *pos++ = 's';
+        if (spec->flags & G_PARAM_READABLE)
+ 	  *pos++ = 'r';
+        if (spec->flags & G_PARAM_WRITABLE)
+	  *pos++ = 'w';
+        if (spec->flags & G_PARAM_CONSTRUCT)
+	  *pos++ = 'x';
+        if (spec->flags & G_PARAM_CONSTRUCT_ONLY)
+	  *pos++ = 'X';
+        *pos = 0;
+
+        nick = g_param_spec_get_nick (spec);
+        blurb = g_param_spec_get_blurb (spec);
+
+        dot = "";
+        if (blurb) {
+          int str_len = strlen (blurb);
+          if (str_len > 0  && blurb[str_len - 1] != '.')
+            dot = ".";
+        }
+
+	type_desc = describe_type (spec);
+	default_value = describe_default (spec);
+	type_name = get_type_name (spec->value_type, &is_pointer);
+        fprintf (fp, "\\n%s::%s\\n%s%s\\n%s\\n%s\\n%s\\n%s%s\\n%s\\n\\n\\n",
+	         object_class_name, g_param_spec_get_name (spec), type_name, is_pointer ? "*" : "", type_desc, flags, nick ? nick : "(null)", blurb ? blurb : "(null)", dot, default_value);
+	g_free (type_desc);
+	g_free (default_value);
+      }
+
+    g_free (properties);
+
+#ifdef GTK_IS_CONTAINER_CLASS
+    if (!child_prop && GTK_IS_CONTAINER_CLASS (class)) {
+      properties = gtk_container_class_list_child_properties (class, &n_properties);
+      child_prop = TRUE;
+      continue;
+    }
+#endif
+
+#ifdef GTK_IS_CELL_AREA_CLASS
+    if (!child_prop && GTK_IS_CELL_AREA_CLASS (class)) {
+      properties = gtk_cell_area_class_list_cell_properties (class, &n_properties);
+      child_prop = TRUE;
+      continue;
+    }
+#endif
+
+#ifdef GTK_IS_WIDGET_CLASS
+#if GTK_CHECK_VERSION(2,1,0)
+    if (!style_prop && GTK_IS_WIDGET_CLASS (class)) {
+      properties = gtk_widget_class_list_style_properties (GTK_WIDGET_CLASS (class), &n_properties);
+      style_prop = TRUE;
+      continue;
+    }
+#endif
+#endif
+
+    break;
+  }
+}
+EOT
+
+close OUTPUT;
+
+# Compile and run our file
+
+$CC = $ENV{CC} ? $ENV{CC} : "gcc";
+$LD = $ENV{LD} ? $ENV{LD} : $CC;
+$CFLAGS = $ENV{CFLAGS} ? "$ENV{CFLAGS}" : "";
+$LDFLAGS = $ENV{LDFLAGS} ? $ENV{LDFLAGS} : "";
+
+my $o_file;
+if ($CC =~ /libtool/) {
+  $o_file  = "$MODULE-scan.lo"
+} else {
+  $o_file = "$MODULE-scan.o"
+}
+
+my $stdout="";
+if (!defined($VERBOSE) or $VERBOSE eq "0") {
+    $stdout=">/dev/null";
+}
+
+# Compiling scanner
+$command = "$CC $stdout $CFLAGS -c -o $o_file $MODULE-scan.c";
+system("($command)") == 0 or die "Compilation of scanner failed: $!\n";
+
+# Linking scanner
+$command = "$LD $stdout -o $MODULE-scan $o_file $LDFLAGS";
+system($command) == 0 or die "Linking of scanner failed: $!\n";
+
+# Running scanner $MODULE-scan ";
+system("sh -c ./$MODULE-scan") == 0 or die "Scan failed: $!\n";
+
+if (!defined($ENV{"GTK_DOC_KEEP_INTERMEDIATE"})) {
+  unlink "./$MODULE-scan.c", "./$MODULE-scan.o", "./$MODULE-scan.lo", "./$MODULE-scan";
+}
+
+&UpdateFileIfChanged ($old_hierarchy_filename, $new_hierarchy_filename, 0);
+# we will merge these in scangobj-merge.py
+#&UpdateFileIfChanged ($old_interfaces_filename, $new_interfaces_filename, 0);
+#&UpdateFileIfChanged ($old_prerequisites_filename, $new_prerequisites_filename, 0);
+#&UpdateFileIfChanged ($old_signals_filename, $new_signals_filename, 0);
+#&UpdateFileIfChanged ($old_args_filename, $new_args_filename, 0);
+
diff --git a/common/gtk-doc-plugins.mak b/common/gtk-doc-plugins.mak
new file mode 100644
index 0000000..f19e7d7
--- /dev/null
+++ b/common/gtk-doc-plugins.mak
@@ -0,0 +1,382 @@
+# This is an include file specifically tuned for building documentation
+# for GStreamer plug-ins
+
+help:
+	@echo
+	@echo "If you are a doc maintainer, run 'make update' to update"
+	@echo "the documentation files maintained in git"
+	@echo
+	@echo Other useful make targets:
+	@echo
+	@echo  check-inspected-versions: make sure the inspected plugin info
+	@echo                            is up to date before a release
+	@echo
+
+# update the stuff maintained by doc maintainers
+update: scanobj-update
+	$(MAKE) check-outdated-docs
+
+# We set GPATH here; this gives us semantics for GNU make
+# which are more like other make's VPATH, when it comes to
+# whether a source that is a target of one rule is then
+# searched for in VPATH/GPATH.
+#
+GPATH = $(srcdir)
+
+# thomas: make docs parallel installable
+TARGET_DIR=$(HTML_DIR)/$(DOC_MODULE)-@GST_API_VERSION@
+
+MAINTAINER_DOC_STAMPS =			\
+	scanobj-build.stamp
+
+EXTRA_DIST = 				\
+	$(MAINTAINER_DOC_STAMPS)		\
+	$(srcdir)/inspect/*.xml		\
+	$(SCANOBJ_FILES)		\
+	$(content_files)		\
+	$(extra_files)			\
+	$(HTML_IMAGES)			\
+	$(DOC_MAIN_SGML_FILE)	\
+	$(DOC_OVERRIDES)		\
+	$(DOC_MODULE)-sections.txt
+
+# we don't add scanobj-build.stamp here since they are built manually by docs
+# maintainers and result is commited to git
+DOC_STAMPS =				\
+	scan-build.stamp		\
+	tmpl-build.stamp		\
+	sgml-build.stamp		\
+	html-build.stamp		\
+	scan.stamp			\
+	tmpl.stamp			\
+	sgml.stamp			\
+	html.stamp
+
+# files generated/updated by gtkdoc-scangobj
+SCANOBJ_FILES =				\
+	$(DOC_MODULE).args              \
+	$(DOC_MODULE).hierarchy         \
+	$(DOC_MODULE).interfaces        \
+	$(DOC_MODULE).prerequisites     \
+	$(DOC_MODULE).signals           \
+	$(DOC_MODULE).types
+
+SCANOBJ_FILES_O =			\
+	.libs/$(DOC_MODULE)-scan.o
+
+# files generated/updated by gtkdoc-scan
+SCAN_FILES =				\
+	$(DOC_MODULE)-sections.txt	\
+	$(DOC_MODULE)-overrides.txt	\
+	$(DOC_MODULE)-decl.txt		\
+	$(DOC_MODULE)-decl-list.txt
+
+
+REPORT_FILES = \
+	$(DOC_MODULE)-undocumented.txt \
+	$(DOC_MODULE)-undeclared.txt \
+	$(DOC_MODULE)-unused.txt
+
+CLEANFILES = \
+	$(SCANOBJ_FILES_O) \
+	$(REPORT_FILES) \
+	$(DOC_STAMPS) \
+	inspect-registry.xml
+
+INSPECT_DIR = inspect
+
+if ENABLE_GTK_DOC
+all-local: html-build.stamp
+
+### inspect GStreamer plug-ins; done by documentation maintainer ###
+
+# only look at the plugins in this module when building inspect .xml stuff
+INSPECT_REGISTRY=$(top_builddir)/docs/plugins/inspect-registry.xml
+INSPECT_ENVIRONMENT=\
+	LC_ALL=C \
+	GST_PLUGIN_SYSTEM_PATH_1_0= \
+	GST_PLUGIN_PATH_1_0=$(top_builddir)/gst:$(top_builddir)/sys:$(top_builddir)/ext:$(top_builddir)/plugins:$(top_builddir)/src:$(top_builddir)/gnl \
+	GST_REGISTRY_1_0=$(INSPECT_REGISTRY) \
+	PKG_CONFIG_PATH="$(GST_PKG_CONFIG_PATH)" \
+	$(INSPECT_EXTRA_ENVIRONMENT)
+
+#### scan gobjects; done by documentation maintainer ####
+scanobj-update:
+	-rm scanobj-build.stamp
+	$(MAKE) scanobj-build.stamp
+
+# gstdoc-scanobj produces 5 output files (.new)
+# scangobj-merge.py merges them into the file which we commit later
+# TODO: also merge the hierarchy
+scanobj-build.stamp: $(SCANOBJ_DEPS) $(basefiles)
+	@echo "  DOC   Introspecting gobjects"
+	@if test x"$(srcdir)" != x. ; then				\
+	    for f in $(SCANOBJ_FILES) $(SCAN_FILES);			\
+	    do								\
+	        if test -e $(srcdir)/$$f; then cp -u $(srcdir)/$$f . ; fi;	\
+	    done;							\
+	fi;								\
+	mkdir -p $(INSPECT_DIR); \
+	scanobj_options=""; \
+	if test "x$(V)" = "x1"; then \
+	    scanobj_options="--verbose"; \
+	fi; \
+	$(INSPECT_ENVIRONMENT) 					\
+	CC="$(GTKDOC_CC)" LD="$(GTKDOC_LD)"				\
+	CFLAGS="$(GTKDOC_CFLAGS) $(CFLAGS) $(WARNING_CFLAGS)"	\
+	LDFLAGS="$(GTKDOC_LIBS) $(LDFLAGS)"				\
+	$(GST_DOC_SCANOBJ) $$scanobj_options --type-init-func="gst_init(NULL,NULL)"	\
+	    --module=$(DOC_MODULE) --source=$(PACKAGE) --inspect-dir=$(INSPECT_DIR) &&		\
+	    echo "  DOC   Merging introspection data" && \
+	    $(PYTHON)						\
+	    $(top_srcdir)/common/scangobj-merge.py $(DOC_MODULE) || exit 1;	\
+	if test x"$(srcdir)" != x. ; then				\
+	    for f in $(SCANOBJ_FILES);					\
+	    do								\
+	        cmp -s ./$$f $(srcdir)/$$f || cp ./$$f $(srcdir)/ ;		\
+	    done;							\
+	fi;								\
+	touch scanobj-build.stamp
+
+$(DOC_MODULE)-decl.txt $(SCANOBJ_FILES) $(SCANOBJ_FILES_O): scan-build.stamp
+	@true
+
+### scan headers; done on every build ###
+scan-build.stamp: $(HFILE_GLOB) $(EXTRA_HFILES) $(basefiles) scanobj-build.stamp
+	@echo '  DOC   Scanning header files'
+	@if test x"$(srcdir)" != x. ; then				\
+	    for f in $(SCANOBJ_FILES) $(SCAN_FILES);			\
+	    do								\
+	        if test -e $(srcdir)/$$f; then cp -u $(srcdir)/$$f . ; fi;	\
+	    done;							\
+	fi
+	@_source_dir='' ;						\
+	for i in $(DOC_SOURCE_DIR) ; do					\
+	    _source_dir="$${_source_dir} --source-dir=$$i" ;	        \
+	done ;							        \
+	gtkdoc-scan							\
+	    $(SCAN_OPTIONS) $(EXTRA_HFILES)				\
+	    --module=$(DOC_MODULE)					\
+	    $${_source_dir}             				\
+	    --ignore-headers="$(IGNORE_HFILES)";			\
+	touch scan-build.stamp
+
+#### update templates; done on every build ####
+
+# in a non-srcdir build, we need to copy files from the previous step
+# and the files from previous runs of this step
+tmpl-build.stamp: $(DOC_MODULE)-decl.txt $(SCANOBJ_FILES) $(DOC_MODULE)-sections.txt $(DOC_OVERRIDES)
+	@echo '  DOC   Rebuilding template files'
+	@if test x"$(srcdir)" != x. ; then				\
+	    for f in $(SCANOBJ_FILES) $(SCAN_FILES);			\
+	    do								\
+	        if test -e $(srcdir)/$$f; then cp -u $(srcdir)/$$f . ; fi;	\
+	    done;							\
+	fi
+	@gtkdoc-mktmpl --module=$(DOC_MODULE)
+	@$(PYTHON) \
+		$(top_srcdir)/common/mangle-tmpl.py $(srcdir)/$(INSPECT_DIR) tmpl
+	@touch tmpl-build.stamp
+
+tmpl.stamp: tmpl-build.stamp
+	@true
+
+#### xml ####
+
+sgml-build.stamp: tmpl.stamp scan-build.stamp $(CFILE_GLOB) $(top_srcdir)/common/plugins.xsl $(expand_content_files)
+	@echo '  DOC   Building XML'
+	@-mkdir -p xml
+	@for a in $(srcdir)/$(INSPECT_DIR)/*.xml; do \
+	    xsltproc --stringparam module $(MODULE) \
+		$(top_srcdir)/common/plugins.xsl $$a > xml/`basename $$a`; done
+	@for f in $(EXAMPLE_CFILES); do \
+		$(PYTHON) $(top_srcdir)/common/c-to-xml.py $$f > xml/element-`basename $$f .c`.xml; done
+	@gtkdoc-mkdb \
+		--module=$(DOC_MODULE) \
+		--source-dir=$(DOC_SOURCE_DIR) \
+		 --expand-content-files="$(expand_content_files)" \
+		--main-sgml-file=$(srcdir)/$(DOC_MAIN_SGML_FILE) \
+		--output-format=xml \
+		--ignore-files="$(IGNORE_HFILES) $(IGNORE_CFILES)" \
+		$(MKDB_OPTIONS)
+	@cp ../version.entities xml
+	@touch sgml-build.stamp
+
+sgml.stamp: sgml-build.stamp
+	@true
+
+#### html ####
+
+html-build.stamp: sgml.stamp $(DOC_MAIN_SGML_FILE) $(content_files)
+	@echo '  DOC   Building HTML'
+	@rm -rf html
+	@mkdir html
+	@cp $(srcdir)/$(DOC_MAIN_SGML_FILE) html
+	@for f in $(content_files); do cp $(srcdir)/$$f html; done
+	@cp -pr xml html
+	@cp ../version.entities html
+	@mkhtml_options=""; \
+	gtkdoc-mkhtml 2>&1 --help | grep  >/dev/null "\-\-verbose"; \
+	if test "$(?)" = "0"; then \
+	  if test "x$(V)" = "x1"; then \
+	    mkhtml_options="$$mkhtml_options --verbose"; \
+	  fi; \
+	fi; \
+	cd html && gtkdoc-mkhtml $$mkhtml_options $(DOC_MODULE)-@GST_API_VERSION@ $(DOC_MAIN_SGML_FILE)
+	@rm -f html/$(DOC_MAIN_SGML_FILE)
+	@rm -rf html/xml
+	@rm -f html/version.entities
+	@test "x$(HTML_IMAGES)" = "x" || for i in "" $(HTML_IMAGES) ; do \
+	    if test "$$i" != ""; then cp $(srcdir)/$$i html ; fi; done
+	@echo '  DOC   Fixing cross-references'
+	@gtkdoc-fixxref --module=$(DOC_MODULE) --module-dir=html --html-dir=$(HTML_DIR) $(FIXXREF_OPTIONS)
+	@touch html-build.stamp
+
+clean-local-gtkdoc:
+	@rm -rf xml tmpl html
+# clean files copied for nonsrcdir templates build
+	@if test x"$(srcdir)" != x. ; then \
+	    rm -rf $(SCANOBJ_FILES) $(SCAN_FILES) $(REPORT_FILES) \
+	        $(MAINTAINER_DOC_STAMPS); \
+	fi
+else
+all-local:
+clean-local-gtkdoc:
+endif
+
+clean-local: clean-local-gtkdoc
+	@rm -f *~ *.bak
+	@rm -rf .libs
+
+distclean-local:
+	@rm -f $(REPORT_FILES) \
+	        $(DOC_MODULE)-decl-list.txt $(DOC_MODULE)-decl.txt
+	@rm -rf tmpl/*.sgml.bak
+	@rm -f $(DOC_MODULE).hierarchy
+	@rm -f *.stamp || true
+	@if test "$(abs_srcdir)" != "$(abs_builddir)" ; then \
+	    rm -f $(DOC_MODULE)-docs.sgml ; \
+	    rm -f $(DOC_MODULE).types ; \
+	    rm -f $(DOC_MODULE).interfaces ; \
+	    rm -f $(DOC_MODULE)-overrides.txt ; \
+	    rm -f $(DOC_MODULE).prerequisites ; \
+	    rm -f $(DOC_MODULE)-sections.txt ; \
+	    rm -rf tmpl/*.sgml ; \
+	    rm -rf $(INSPECT_DIR); \
+	fi
+	@rm -rf *.o
+
+MAINTAINERCLEANFILES = $(MAINTAINER_DOC_STAMPS)
+
+# thomas: make docs parallel installable; devhelp requires majorminor too
+install-data-local:
+	(installfiles=`echo $(builddir)/html/*.sgml $(builddir)/html/*.html $(builddir)/html/*.png $(builddir)/html/*.css`; \
+	if test "$$installfiles" = '$(builddir)/html/*.sgml $(builddir)/html/*.html $(builddir)/html/*.png $(builddir)/html/*.css'; \
+	then echo '-- Nothing to install' ; \
+	else \
+	  $(mkinstalldirs) $(DESTDIR)$(TARGET_DIR); \
+	  for i in $$installfiles; do \
+	    echo '-- Installing '$$i ; \
+	    $(INSTALL_DATA) $$i $(DESTDIR)$(TARGET_DIR); \
+	  done; \
+	  pngfiles=`echo ./html/*.png`; \
+	  if test "$$pngfiles" != './html/*.png'; then \
+	    for i in $$pngfiles; do \
+	      echo '-- Installing '$$i ; \
+	      $(INSTALL_DATA) $$i $(DESTDIR)$(TARGET_DIR); \
+	    done; \
+	  fi; \
+	  echo '-- Installing $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2' ; \
+	  if test -e $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2; then \
+	            $(INSTALL_DATA) $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2 \
+	            $(DESTDIR)$(TARGET_DIR)/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2; \
+	  fi; \
+	  $(GTKDOC_REBASE) --relative --dest-dir=$(DESTDIR) --html-dir=$(DESTDIR)$(TARGET_DIR) || true ; \
+	fi)
+uninstall-local:
+	if test -d $(DESTDIR)$(TARGET_DIR); then \
+	  rm -rf $(DESTDIR)$(TARGET_DIR)/*; \
+	  rmdir -p $(DESTDIR)$(TARGET_DIR) 2>/dev/null || true; \
+	else \
+	  echo '-- Nothing to uninstall' ; \
+	fi;
+
+#
+# Checks
+#
+if ENABLE_GTK_DOC
+check-hierarchy: $(DOC_MODULE).hierarchy
+	@if grep '	' $(DOC_MODULE).hierarchy; then \
+	    echo "$(DOC_MODULE).hierarchy contains tabs, please fix"; \
+	    /bin/false; \
+	fi
+
+check: check-hierarchy
+endif
+
+# wildcard is apparently not portable to other makes, hence the use of find
+inspect_files = $(shell find $(srcdir)/$(INSPECT_DIR) -name '*.xml')
+
+check-inspected-versions:
+	@echo Checking plugin versions of inspected plugin data ...; \
+	fail=0 ; \
+	for each in $(inspect_files) ; do \
+	  if (grep -H '' $$each | grep -v '$(VERSION)'); then \
+	    echo $$each should be fixed to say version $(VERSION) or be removed ; \
+	    echo "sed -i -e 's//$(VERSION)<\/version>/'" $$each; \
+	    echo ; \
+	    fail=1; \
+	  fi ; \
+	done ; \
+	exit $$fail
+
+check-outdated-docs:
+	$(AM_V_GEN)echo Checking for outdated plugin inspect data ...; \
+	fail=0 ; \
+	if [ -d $(top_srcdir)/.git/ ]; then \
+	  files=`find $(srcdir)/inspect/ -name '*xml'`; \
+	  for f in $$files; do \
+	    ver=`grep '$(PACKAGE_VERSION)' $$f`; \
+	    if test "x$$ver" = "x"; then \
+	      plugin=`echo $$f | sed -e 's/^.*plugin-//' -e 's/.xml//'`; \
+	      # echo "Checking $$plugin $$f"; \
+	      pushd "$(top_srcdir)" >/dev/null; \
+	      pinit=`git grep -A3 GST_PLUGIN_DEFINE -- ext/ gst/ sys/ | grep "\"$$plugin\""`; \
+	      popd >/dev/null; \
+	      # echo "[$$pinit]"; \
+	      if test "x$$pinit" = "x"; then \
+	        printf " **** outdated docs for plugin %-15s: %s\n" $$plugin $$f; \
+	        fail=1; \
+	      fi; \
+	    fi; \
+	  done; \
+	fi ; \
+	exit $$fail
+
+#
+# Require gtk-doc when making dist
+#
+if ENABLE_GTK_DOC
+dist-check-gtkdoc:
+else
+dist-check-gtkdoc:
+	@echo "*** gtk-doc must be installed and enabled in order to make dist"
+	@false
+endif
+
+# FIXME: decide whether we want to dist generated html or not
+# also this only works, if the project has been build before
+# we could dist html only if its there, but that might lead to missing html in
+# tarballs
+dist-hook: dist-check-gtkdoc dist-hook-local
+	mkdir $(distdir)/html
+	cp html/* $(distdir)/html
+	-cp $(srcdir)/$(DOC_MODULE).types $(distdir)/
+	-cp $(srcdir)/$(DOC_MODULE)-sections.txt $(distdir)/
+	cd $(distdir) && rm -f $(DISTCLEANFILES)
+	-gtkdoc-rebase --online --relative --html-dir=$(distdir)/html
+
+.PHONY : dist-hook-local docs check-outdated-docs inspect
+
+# avoid spurious build errors when distchecking with -jN
+.NOTPARALLEL:
diff --git a/common/gtk-doc.mak b/common/gtk-doc.mak
new file mode 100644
index 0000000..9d76889
--- /dev/null
+++ b/common/gtk-doc.mak
@@ -0,0 +1,227 @@
+###########################################################################
+# Everything below here is generic and you shouldn't need to change it.
+###########################################################################
+# thomas: except of course that we did
+
+# thomas: copied from glib-2
+# We set GPATH here; this gives us semantics for GNU make
+# which are more like other make's VPATH, when it comes to
+# whether a source that is a target of one rule is then
+# searched for in VPATH/GPATH.
+#
+GPATH = $(srcdir)
+
+# thomas: make docs parallel installable
+TARGET_DIR=$(HTML_DIR)/$(DOC_MODULE)-@GST_API_VERSION@
+
+EXTRA_DIST = 				\
+	$(content_files)		\
+	$(extra_files)			\
+	$(HTML_IMAGES)			\
+	$(DOC_MAIN_SGML_FILE)		\
+	$(DOC_MODULE).types		\
+	$(DOC_OVERRIDES)		\
+	$(DOC_MODULE)-sections.txt
+
+DOC_STAMPS =				\
+	setup-build.stamp		\
+	scan-build.stamp		\
+	sgml-build.stamp		\
+	html-build.stamp		\
+	sgml.stamp		\
+	html.stamp
+
+SCANOBJ_FILES =				\
+	$(DOC_MODULE).args		\
+	$(DOC_MODULE).hierarchy		\
+	$(DOC_MODULE).interfaces		\
+	$(DOC_MODULE).prerequisites	\
+	$(DOC_MODULE).signals		\
+	.libs/$(DOC_MODULE)-scan.o
+
+REPORT_FILES = \
+	$(DOC_MODULE)-undocumented.txt \
+	$(DOC_MODULE)-undeclared.txt \
+	$(DOC_MODULE)-unused.txt
+
+CLEANFILES = $(SCANOBJ_FILES) $(REPORT_FILES) $(DOC_STAMPS) doc-registry.xml
+
+if ENABLE_GTK_DOC
+all-local: html-build.stamp
+
+#### setup ####
+
+setup-build.stamp: $(content_files)
+	-@if test "$(abs_srcdir)" != "$(abs_builddir)" ; then \
+	    echo '  DOC   Preparing build'; \
+	    files=`echo $(DOC_MAIN_SGML_FILE) $(DOC_OVERRIDES) $(DOC_MODULE)-sections.txt $(DOC_MODULE).types $(content_files)`; \
+	    if test "x$$files" != "x" ; then \
+	        for file in $$files ; do \
+	            test -f $(abs_srcdir)/$$file && \
+	                cp -pu $(abs_srcdir)/$$file $(abs_builddir)/ || true; \
+	        done; \
+	    fi; \
+	fi
+	@touch setup-build.stamp
+
+#### scan ####
+
+# in the case of non-srcdir builds, the built gst directory gets added
+# to gtk-doc scanning; but only then, to avoid duplicates
+scan-build.stamp: $(HFILE_GLOB) $(CFILE_GLOB)
+	@echo '  DOC   Scanning header files'
+	@_source_dir='' ;						\
+	for i in $(DOC_SOURCE_DIR) ; do					\
+	    _source_dir="$${_source_dir} --source-dir=$$i" ;	        \
+	done ;							        \
+	gtkdoc-scan							\
+		$(SCAN_OPTIONS) $(EXTRA_HFILES)				\
+		--module=$(DOC_MODULE)					\
+		$${_source_dir}         				\
+		--ignore-headers="$(IGNORE_HFILES)"
+	@if grep -l '^..*$$' $(DOC_MODULE).types > /dev/null; then	\
+	    echo "  DOC   Introspecting gobjects"; \
+	    GST_PLUGIN_SYSTEM_PATH_1_0=`cd $(top_builddir) && pwd`		\
+	    GST_PLUGIN_PATH_1_0=						\
+	    GST_REGISTRY_1_0=doc-registry.xml				\
+	    $(GTKDOC_EXTRA_ENVIRONMENT)					\
+	    CC="$(GTKDOC_CC)" LD="$(GTKDOC_LD)"				\
+	    CFLAGS="$(GTKDOC_CFLAGS) $(CFLAGS)"				\
+	    LDFLAGS="$(GTKDOC_LIBS) $(LDFLAGS)"				\
+	    gtkdoc-scangobj --type-init-func="gst_init(NULL,NULL)"	\
+	        --module=$(DOC_MODULE) ;				\
+	else								\
+	    for i in $(SCANOBJ_FILES) ; do				\
+	       test -f $$i || touch $$i ;				\
+	    done							\
+	fi
+	@touch scan-build.stamp
+
+$(DOC_MODULE)-decl.txt $(SCANOBJ_FILES) $(DOC_MODULE)-sections.txt $(DOC_MODULE)-overrides.txt: scan-build.stamp
+	@true
+
+#### xml ####
+
+sgml-build.stamp: setup-build.stamp $(DOC_MODULE)-decl.txt $(SCANOBJ_FILES) $(DOC_MODULE)-sections.txt $(expand_content_files)
+	@echo '  DOC   Building XML'
+	@gtkdoc-mkdb --module=$(DOC_MODULE) --source-dir=$(DOC_SOURCE_DIR)  --expand-content-files="$(expand_content_files)" --main-sgml-file=$(DOC_MAIN_SGML_FILE) --output-format=xml $(MKDB_OPTIONS)
+	@cp ../version.entities xml
+	@touch sgml-build.stamp
+
+sgml.stamp: sgml-build.stamp
+	@true
+
+#### html ####
+
+html-build.stamp: sgml.stamp $(DOC_MAIN_SGML_FILE) $(content_files)
+	@echo '  DOC   Building HTML'
+	@rm -rf html
+	@mkdir html
+	@cp -pr xml html
+	@cp ../version.entities ./
+	@mkhtml_options=""; \
+	gtkdoc-mkhtml 2>&1 --help | grep  >/dev/null "\-\-verbose"; \
+	if test "$(?)" = "0"; then \
+	  if test "x$(V)" = "x1"; then \
+	    mkhtml_options="$$mkhtml_options --verbose"; \
+	  fi; \
+	fi; \
+	@gtkdoc-mkhtml 2>&1 --help | grep  >/dev/null "\-\-path"; \
+	if test "$(?)" = "0"; then \
+	  mkhtml_options=--path="$(abs_srcdir)"; \
+	fi; \
+	cd html && gtkdoc-mkhtml $$mkhtml_options $(MKHTML_OPTIONS) $(DOC_MODULE)-@GST_API_VERSION@ ../$(DOC_MAIN_SGML_FILE)
+	@rm -rf html/xml
+	@rm -f version.entities
+	@test "x$(HTML_IMAGES)" = "x" ||  ( cd $(srcdir) && cp $(HTML_IMAGES) $(abs_builddir)/html )
+	@echo '  DOC   Fixing cross-references'
+	@gtkdoc-fixxref --module=$(DOC_MODULE) --module-dir=html --html-dir=$(HTML_DIR) $(FIXXREF_OPTIONS)
+	@touch html-build.stamp
+
+clean-local-gtkdoc:
+	@rm -rf xml tmpl html
+# clean files copied for nonsrcdir templates build
+	@if test x"$(srcdir)" != x. ; then \
+	        rm -rf $(DOC_MODULE).types; \
+	fi
+else
+all-local:
+clean-local-gtkdoc:
+endif
+
+clean-local: clean-local-gtkdoc
+	@rm -f *~ *.bak
+	@rm -rf .libs
+
+distclean-local:
+	@rm -f $(REPORT_FILES) \
+	        $(DOC_MODULE)-decl-list.txt $(DOC_MODULE)-decl.txt
+	@rm -rf tmpl/*.sgml.bak
+	@rm -f $(DOC_MODULE).hierarchy
+	@rm -f *.stamp || true
+	@if test "$(abs_srcdir)" != "$(abs_builddir)" ; then \
+	    rm -f $(DOC_MAIN_SGML_FILE) ; \
+	    rm -f $(DOC_OVERRIDES) ; \
+	    rm -f $(DOC_MODULE).types ; \
+	    rm -f $(DOC_MODULE).interfaces ; \
+	    rm -f $(DOC_MODULE).prerequisites ; \
+	    rm -f $(DOC_MODULE)-sections.txt ; \
+	    rm -f $(content_files) ; \
+	    rm -rf tmpl/*.sgml ; \
+	fi
+	@rm -rf *.o
+
+maintainer-clean-local: clean
+	@cd $(srcdir) && rm -rf html \
+		xml $(DOC_MODULE)-decl-list.txt $(DOC_MODULE)-decl.txt
+
+# thomas: make docs parallel installable; devhelp requires majorminor too
+install-data-local:
+	(installfiles=`echo $(builddir)/html/*.sgml $(builddir)/html/*.html $(builddir)/html/*.png $(builddir)/html/*.css`; \
+	if test "$$installfiles" = '$(builddir)/html/*.sgml $(builddir)/html/*.html $(builddir)/html/*.png $(builddir)/html/*.css'; \
+	then echo '-- Nothing to install' ; \
+	else \
+	  $(mkinstalldirs) $(DESTDIR)$(TARGET_DIR); \
+	  for i in $$installfiles; do \
+	    echo '-- Installing '$$i ; \
+	    $(INSTALL_DATA) $$i $(DESTDIR)$(TARGET_DIR); \
+	  done; \
+	  echo '-- Installing $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2' ; \
+	  if test -e $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2; then \
+	            $(INSTALL_DATA) $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2 \
+	            $(DESTDIR)$(TARGET_DIR)/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2; \
+	  fi; \
+	  $(GTKDOC_REBASE) --relative --dest-dir=$(DESTDIR) --html-dir=$(DESTDIR)$(TARGET_DIR) || true ; \
+	fi)
+uninstall-local:
+	if test -d $(DESTDIR)$(TARGET_DIR); then \
+	  rm -rf $(DESTDIR)$(TARGET_DIR)/*; \
+	  rmdir -p $(DESTDIR)$(TARGET_DIR) 2>/dev/null || true; \
+	else \
+	  echo '-- Nothing to uninstall' ; \
+	fi;
+
+
+#
+# Require gtk-doc when making dist
+#
+if ENABLE_GTK_DOC
+dist-check-gtkdoc:
+else
+dist-check-gtkdoc:
+	@echo "*** gtk-doc must be installed and enabled in order to make dist"
+	@false
+endif
+
+dist-hook: dist-check-gtkdoc dist-hook-local
+	mkdir $(distdir)/html
+	cp html/* $(distdir)/html
+	-cp $(srcdir)/$(DOC_MODULE).types $(distdir)/
+	-cp $(srcdir)/$(DOC_MODULE)-sections.txt $(distdir)/
+	cd $(distdir) && rm -f $(DISTCLEANFILES)
+	-gtkdoc-rebase --online --relative --html-dir=$(distdir)/html
+
+.PHONY : dist-hook-local docs
+
+# avoid spurious build errors when distchecking with -jN
+.NOTPARALLEL:
diff --git a/common/m4/Makefile.am b/common/m4/Makefile.am
new file mode 100644
index 0000000..3d387ee
--- /dev/null
+++ b/common/m4/Makefile.am
@@ -0,0 +1,40 @@
+EXTRA_DIST = \
+	README \
+	as-ac-expand.m4 \
+	as-auto-alt.m4 \
+	as-compiler-flag.m4 \
+	as-compiler.m4 \
+	as-docbook.m4 \
+	as-gcc-inline-assembly.m4 \
+	as-libtool.m4 \
+	as-libtool-tags.m4 \
+	as-python.m4 \
+	as-version.m4 \
+	ax_create_stdint_h.m4 \
+	glib-gettext.m4 \
+	gst-arch.m4 \
+	gst-args.m4 \
+	gst-check.m4 \
+	gst-debuginfo.m4 \
+	gst-default.m4 \
+	gst-doc.m4 \
+	gst-dowhile.m4 \
+	gst-error.m4 \
+	gst-feature.m4 \
+	gst-function.m4 \
+	gst-gettext.m4 \
+	gst-glib2.m4 \
+	gst-libxml2.m4 \
+	gst-parser.m4 \
+	gst-package-release-datetime.m4 \
+	gst-platform.m4 \
+	gst-plugindir.m4 \
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+distclean: distclean-am
+	-rm -f Makefile
+distclean-am: clean-am distclean-generic
+
+dvi: dvi-am
+
+dvi-am:
+
+html: html-am
+
+html-am:
+
+info: info-am
+
+info-am:
+
+install-data-am:
+
+install-dvi: install-dvi-am
+
+install-dvi-am:
+
+install-exec-am:
+
+install-html: install-html-am
+
+install-html-am:
+
+install-info: install-info-am
+
+install-info-am:
+
+install-man:
+
+install-pdf: install-pdf-am
+
+install-pdf-am:
+
+install-ps: install-ps-am
+
+install-ps-am:
+
+installcheck-am:
+
+maintainer-clean: maintainer-clean-am
+	-rm -f Makefile
+maintainer-clean-am: distclean-am maintainer-clean-generic
+
+mostlyclean: mostlyclean-am
+
+mostlyclean-am: mostlyclean-generic mostlyclean-libtool
+
+pdf: pdf-am
+
+pdf-am:
+
+ps: ps-am
+
+ps-am:
+
+uninstall-am:
+
+.MAKE: install-am install-strip
+
+.PHONY: all all-am check check-am clean clean-generic clean-libtool \
+	cscopelist-am ctags-am distclean distclean-generic \
+	distclean-libtool distdir dvi dvi-am html html-am info info-am \
+	install install-am install-data install-data-am install-dvi \
+	install-dvi-am install-exec install-exec-am install-html \
+	install-html-am install-info install-info-am install-man \
+	install-pdf install-pdf-am install-ps install-ps-am \
+	install-strip installcheck installcheck-am installdirs \
+	maintainer-clean maintainer-clean-generic mostlyclean \
+	mostlyclean-generic mostlyclean-libtool pdf pdf-am ps ps-am \
+	tags-am uninstall uninstall-am
+
+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
diff --git a/common/m4/README b/common/m4/README
new file mode 100644
index 0000000..867a344
--- /dev/null
+++ b/common/m4/README
@@ -0,0 +1,3 @@
+All aclocal .m4 files we need are put here and cat'd to acinclude.m4 in
+the source root.  Official ones (taken from the relevant devel packages)
+are named as-is, unofficial ones (or changed ones) get a gst-prefix.
diff --git a/common/m4/as-ac-expand.m4 b/common/m4/as-ac-expand.m4
new file mode 100644
index 0000000..d6c9e33
--- /dev/null
+++ b/common/m4/as-ac-expand.m4
@@ -0,0 +1,43 @@
+dnl as-ac-expand.m4 0.2.0
+dnl autostars m4 macro for expanding directories using configure's prefix
+dnl thomas@apestaart.org
+
+dnl AS_AC_EXPAND(VAR, CONFIGURE_VAR)
+dnl example
+dnl AS_AC_EXPAND(SYSCONFDIR, $sysconfdir)
+dnl will set SYSCONFDIR to /usr/local/etc if prefix=/usr/local
+
+AC_DEFUN([AS_AC_EXPAND],
+[
+  EXP_VAR=[$1]
+  FROM_VAR=[$2]
+
+  dnl first expand prefix and exec_prefix if necessary
+  prefix_save=$prefix
+  exec_prefix_save=$exec_prefix
+
+  dnl if no prefix given, then use /usr/local, the default prefix
+  if test "x$prefix" = "xNONE"; then
+    prefix="$ac_default_prefix"
+  fi
+  dnl if no exec_prefix given, then use prefix
+  if test "x$exec_prefix" = "xNONE"; then
+    exec_prefix=$prefix
+  fi
+
+  full_var="$FROM_VAR"
+  dnl loop until it doesn't change anymore
+  while true; do
+    new_full_var="`eval echo $full_var`"
+    if test "x$new_full_var" = "x$full_var"; then break; fi
+    full_var=$new_full_var
+  done
+
+  dnl clean up
+  full_var=$new_full_var
+  AC_SUBST([$1], "$full_var")
+
+  dnl restore prefix and exec_prefix
+  prefix=$prefix_save
+  exec_prefix=$exec_prefix_save
+])
diff --git a/common/m4/as-auto-alt.m4 b/common/m4/as-auto-alt.m4
new file mode 100644
index 0000000..3f7920d
--- /dev/null
+++ b/common/m4/as-auto-alt.m4
@@ -0,0 +1,50 @@
+dnl as-auto-alt.m4 0.0.2
+dnl autostars m4 macro for supplying alternate autotools versions to configure
+dnl thomas@apestaart.org
+dnl
+dnl AS_AUTOTOOLS_ALTERNATE()
+dnl
+dnl supplies --with arguments for autoconf, autoheader, automake, aclocal
+
+AC_DEFUN([AS_AUTOTOOLS_ALTERNATE],
+[
+  dnl allow for different autoconf version
+  AC_ARG_WITH(autoconf,
+    AC_HELP_STRING([--with-autoconf],
+                   [use a different autoconf for regeneration of Makefiles]),
+    [
+      unset AUTOCONF
+      AM_MISSING_PROG(AUTOCONF, ${withval})
+      AC_MSG_NOTICE([Using $AUTOCONF as autoconf])
+    ])
+
+  dnl allow for different autoheader version
+  AC_ARG_WITH(autoheader,
+    AC_HELP_STRING([--with-autoheader],
+                   [use a different autoheader for regeneration of Makefiles]),
+    [
+      unset AUTOHEADER
+      AM_MISSING_PROG(AUTOHEADER, ${withval})
+      AC_MSG_NOTICE([Using $AUTOHEADER as autoheader])
+    ])
+
+  dnl allow for different automake version
+  AC_ARG_WITH(automake,
+    AC_HELP_STRING([--with-automake],
+                   [use a different automake for regeneration of Makefiles]),
+    [
+      unset AUTOMAKE
+      AM_MISSING_PROG(AUTOMAKE, ${withval})
+      AC_MSG_NOTICE([Using $AUTOMAKE as automake])
+    ])
+
+  dnl allow for different aclocal version
+  AC_ARG_WITH(aclocal,
+    AC_HELP_STRING([--with-aclocal],
+                   [use a different aclocal for regeneration of Makefiles]),
+    [
+      unset ACLOCAL
+      AM_MISSING_PROG(ACLOCAL, ${withval})
+      AC_MSG_NOTICE([Using $ACLOCAL as aclocal])
+    ])
+])
diff --git a/common/m4/as-compiler-flag.m4 b/common/m4/as-compiler-flag.m4
new file mode 100644
index 0000000..8bb853a
--- /dev/null
+++ b/common/m4/as-compiler-flag.m4
@@ -0,0 +1,96 @@
+dnl as-compiler-flag.m4 0.1.0
+
+dnl autostars m4 macro for detection of compiler flags
+
+dnl David Schleef 
+dnl Tim-Philipp Müller 
+
+dnl AS_COMPILER_FLAG(CFLAGS, ACTION-IF-ACCEPTED, [ACTION-IF-NOT-ACCEPTED])
+dnl Tries to compile with the given CFLAGS.
+dnl Runs ACTION-IF-ACCEPTED if the compiler can compile with the flags,
+dnl and ACTION-IF-NOT-ACCEPTED otherwise.
+
+AC_DEFUN([AS_COMPILER_FLAG],
+[
+  AC_MSG_CHECKING([to see if compiler understands $1])
+
+  save_CFLAGS="$CFLAGS"
+  CFLAGS="$CFLAGS $1"
+
+  AC_TRY_COMPILE([ ], [], [flag_ok=yes], [flag_ok=no])
+  CFLAGS="$save_CFLAGS"
+
+  if test "X$flag_ok" = Xyes ; then
+    $2
+    true
+  else
+    $3
+    true
+  fi
+  AC_MSG_RESULT([$flag_ok])
+])
+
+dnl AS_CXX_COMPILER_FLAG(CPPFLAGS, ACTION-IF-ACCEPTED, [ACTION-IF-NOT-ACCEPTED])
+dnl Tries to compile with the given CPPFLAGS.
+dnl Runs ACTION-IF-ACCEPTED if the compiler can compile with the flags,
+dnl and ACTION-IF-NOT-ACCEPTED otherwise.
+
+AC_DEFUN([AS_CXX_COMPILER_FLAG],
+[
+  AC_REQUIRE([AC_PROG_CXX])
+
+  AC_MSG_CHECKING([to see if c++ compiler understands $1])
+
+  save_CPPFLAGS="$CPPFLAGS"
+  CPPFLAGS="$CPPFLAGS $1"
+
+  AC_LANG_PUSH(C++)
+
+  AC_TRY_COMPILE([ ], [], [flag_ok=yes], [flag_ok=no])
+  CPPFLAGS="$save_CPPFLAGS"
+
+  if test "X$flag_ok" = Xyes ; then
+    $2
+    true
+  else
+    $3
+    true
+  fi
+
+  AC_LANG_POP(C++)
+
+  AC_MSG_RESULT([$flag_ok])
+])
+
+dnl AS_OBJC_COMPILER_FLAG(CPPFLAGS, ACTION-IF-ACCEPTED, [ACTION-IF-NOT-ACCEPTED])
+dnl Tries to compile with the given CPPFLAGS.
+dnl Runs ACTION-IF-ACCEPTED if the compiler can compile with the flags,
+dnl and ACTION-IF-NOT-ACCEPTED otherwise.
+
+AC_DEFUN([AS_OBJC_COMPILER_FLAG],
+[
+  AC_REQUIRE([AC_PROG_OBJC])
+
+  AC_MSG_CHECKING([to see if Objective C compiler understands $1])
+
+  save_CPPFLAGS="$CPPFLAGS"
+  CPPFLAGS="$CPPFLAGS $1"
+
+  AC_LANG_PUSH([Objective C])
+
+  AC_TRY_COMPILE([ ], [], [flag_ok=yes], [flag_ok=no])
+  CPPFLAGS="$save_CPPFLAGS"
+
+  if test "X$flag_ok" = Xyes ; then
+    $2
+    true
+  else
+    $3
+    true
+  fi
+
+  AC_LANG_POP([Objective C])
+
+  AC_MSG_RESULT([$flag_ok])
+])
+
diff --git a/common/m4/as-compiler.m4 b/common/m4/as-compiler.m4
new file mode 100644
index 0000000..309a060
--- /dev/null
+++ b/common/m4/as-compiler.m4
@@ -0,0 +1,44 @@
+dnl as-compiler.m4 0.1.0
+
+dnl autostars m4 macro for detection of compiler flavor
+
+dnl Thomas Vander Stichele 
+
+dnl $Id: as-compiler.m4,v 1.4 2004/06/01 09:44:19 thomasvs Exp $
+
+dnl AS_COMPILER(COMPILER)
+dnl will set variable COMPILER to
+dnl - gcc
+dnl - forte
+dnl - (empty) if no guess could be made
+
+AC_DEFUN([AS_COMPILER],
+[
+  as_compiler=
+  AC_MSG_CHECKING(for compiler flavour)
+
+  dnl is it gcc ?
+  if test "x$GCC" = "xyes"; then
+    as_compiler="gcc"
+  fi
+
+  dnl is it forte ?
+  AC_TRY_RUN([
+int main
+(int argc, char *argv[])
+{
+#ifdef __sun
+  return 0;
+#else
+  return 1;
+#endif
+}
+  ], as_compiler="forte", ,)
+
+  if test "x$as_compiler" = "x"; then
+    AC_MSG_RESULT([unknown !])
+  else
+    AC_MSG_RESULT($as_compiler)
+  fi
+  [$1]=$as_compiler
+])
diff --git a/common/m4/as-docbook.m4 b/common/m4/as-docbook.m4
new file mode 100644
index 0000000..2e27050
--- /dev/null
+++ b/common/m4/as-docbook.m4
@@ -0,0 +1,78 @@
+dnl AS_DOCBOOK([, ACTION-IF-FOUND [, ACTION-IF-NOT-FOUND]])
+dnl checks if xsltproc can build docbook documentation
+dnl (which is possible if the catalog is set up properly
+dnl I also tried checking for a specific version and type of docbook
+dnl but xsltproc seemed to happily run anyway, so we can't check for that
+dnl and version
+dnl this macro takes inspiration from
+dnl http://www.movement.uklinux.net/docs/docbook-autotools/configure.html
+AC_DEFUN([AS_DOCBOOK],
+[
+  XSLTPROC_FLAGS=--nonet
+  DOCBOOK_ROOT=
+  TYPE_LC=xml
+  TYPE_UC=XML
+  DOCBOOK_VERSION=4.1.2
+
+  if test -n "$XML_CATALOG_FILES"; then
+    oldIFS=$IFS
+    IFS=' '
+    for xml_catalog_file in $XML_CATALOG_FILES; do
+      if test -f $xml_catalog_file; then
+        XML_CATALOG=$xml_catalog_file
+        CAT_ENTRY_START=''
+        break
+      fi
+    done
+    IFS=$oldIFS
+  elif test ! -f /etc/xml/catalog; then
+    for i in /usr/share/sgml/docbook/stylesheet/xsl/nwalsh /usr/share/sgml/docbook/xsl-stylesheets/ /usr/local/share/xsl/docbook ;
+    do
+      if test -d "$i"; then
+        DOCBOOK_ROOT=$i
+      fi
+    done
+  else
+    XML_CATALOG=/etc/xml/catalog
+    CAT_ENTRY_START=''
+  fi
+
+  dnl We need xsltproc to process the test
+  AC_CHECK_PROG(XSLTPROC,xsltproc,xsltproc,)
+  XSLTPROC_WORKS=no
+  if test -n "$XSLTPROC"; then
+    AC_MSG_CHECKING([whether xsltproc docbook processing works])
+
+    if test -n "$XML_CATALOG"; then
+      DB_FILE="http://docbook.sourceforge.net/release/xsl/current/xhtml/docbook.xsl"
+    else
+      DB_FILE="$DOCBOOK_ROOT/xhtml/docbook.xsl"
+    fi
+    $XSLTPROC $XSLTPROC_FLAGS $DB_FILE >/dev/null 2>&1 << END
+
+
+
+
+END
+    if test "$?" = 0; then
+      XSLTPROC_WORKS=yes
+    fi
+    AC_MSG_RESULT($XSLTPROC_WORKS)
+  fi
+
+  if test "x$XSLTPROC_WORKS" = "xyes"; then
+    dnl execute ACTION-IF-FOUND
+    ifelse([$1], , :, [$1])
+  else
+    dnl execute ACTION-IF-NOT-FOUND
+    ifelse([$2], , :, [$2])
+  fi
+
+  AC_SUBST(XML_CATALOG)
+  AC_SUBST(XSLTPROC_FLAGS)
+  AC_SUBST(DOCBOOK_ROOT)
+  AC_SUBST(CAT_ENTRY_START)
+  AC_SUBST(CAT_ENTRY_END)
+])
diff --git a/common/m4/as-gcc-inline-assembly.m4 b/common/m4/as-gcc-inline-assembly.m4
new file mode 100644
index 0000000..af32104
--- /dev/null
+++ b/common/m4/as-gcc-inline-assembly.m4
@@ -0,0 +1,52 @@
+dnl as-gcc-inline-assembly.m4 0.1.0
+
+dnl autostars m4 macro for detection of gcc inline assembly
+
+dnl David Schleef 
+
+dnl $Id$
+
+dnl AS_COMPILER_FLAG(ACTION-IF-ACCEPTED, [ACTION-IF-NOT-ACCEPTED])
+dnl Tries to compile with the given CFLAGS.
+dnl Runs ACTION-IF-ACCEPTED if the compiler can compile with the flags,
+dnl and ACTION-IF-NOT-ACCEPTED otherwise.
+
+AC_DEFUN([AS_GCC_INLINE_ASSEMBLY],
+[
+  AC_MSG_CHECKING([if compiler supports gcc-style inline assembly])
+
+  AC_TRY_COMPILE([], [
+#ifdef __GNUC_MINOR__
+#if (__GNUC__ * 1000 + __GNUC_MINOR__) < 3004
+#error GCC before 3.4 has critical bugs compiling inline assembly
+#endif
+#endif
+__asm__ (""::) ], [flag_ok=yes], [flag_ok=no])
+
+  if test "X$flag_ok" = Xyes ; then
+    $1
+    true
+  else
+    $2
+    true
+  fi
+  AC_MSG_RESULT([$flag_ok])
+])
+
+
+AC_DEFUN([AS_GCC_ASM_POWERPC_FPU],
+[
+  AC_MSG_CHECKING([if compiler supports FPU instructions on PowerPC])
+
+  AC_TRY_COMPILE([], [__asm__ ("fadd 0,0,0"::) ], [flag_ok=yes], [flag_ok=no])
+
+  if test "X$flag_ok" = Xyes ; then
+    $1
+    true
+  else
+    $2
+    true
+  fi
+  AC_MSG_RESULT([$flag_ok])
+])
+
diff --git a/common/m4/as-libtool-tags.m4 b/common/m4/as-libtool-tags.m4
new file mode 100644
index 0000000..06f0ae4
--- /dev/null
+++ b/common/m4/as-libtool-tags.m4
@@ -0,0 +1,83 @@
+dnl as-libtool-tags.m4 0.1.4
+
+dnl autostars m4 macro for selecting libtool "tags" (languages)
+
+dnl Andy Wingo does not claim credit for this macro
+dnl backported from libtool 1.6 by Paolo Bonzini
+dnl see http://lists.gnu.org/archive/html/libtool/2003-12/msg00007.html
+
+dnl $Id$
+
+dnl AS_LIBTOOL_TAGS([tags...])
+
+dnl example
+dnl AS_LIBTOOL_TAGS([]) for only C (no fortran, etc)
+
+dnl When AC_LIBTOOL_TAGS is used, I redefine _LT_AC_TAGCONFIG
+dnl to be more similar to the libtool 1.6 implementation, which
+dnl uses an m4 loop and m4 case instead of a shell loop.  This
+dnl way the CXX/GCJ/F77/RC tests are not always expanded.
+
+dnl AS_LIBTOOL_TAGS
+dnl ---------------
+dnl tags to enable
+AC_DEFUN([AS_LIBTOOL_TAGS],
+[m4_define([_LT_TAGS],[$1])
+m4_define([_LT_AC_TAGCONFIG], [
+  # redefined LT AC TAGCONFIG
+  if test -f "$ltmain"; then
+    if test ! -f "${ofile}"; then
+      AC_MSG_WARN([output file `$ofile' does not exist])
+    fi
+
+    if test -z "$LTCC"; then
+      eval "`$SHELL ${ofile} --config | grep '^LTCC='`"
+      if test -z "$LTCC"; then
+        AC_MSG_WARN([output file `$ofile' does not look like a libtool script])
+      else
+        AC_MSG_WARN([using `LTCC=$LTCC', extracted from `$ofile'])
+      fi
+    fi
+
+    AC_FOREACH([_LT_TAG], _LT_TAGS,
+      echo THOMAS: tag _LT_TAG
+      [m4_case(_LT_TAG,
+      [CXX], [
+    if test -n "$CXX" && test "X$CXX" != "Xno"; then
+      echo "THOMAS: YAY CXX"
+      AC_LIBTOOL_LANG_CXX_CONFIG
+      available_tags="$available_tags _LT_TAG"
+    fi],
+      [F77], [
+    if test -n "$F77" && test "X$F77" != "Xno"; then
+      AC_LIBTOOL_LANG_F77_CONFIG
+      available_tags="$available_tags _LT_TAG"
+    fi],
+      [GCJ], [
+    if test -n "$GCJ" && test "X$GCJ" != "Xno"; then
+      AC_LIBTOOL_LANG_GCJ_CONFIG
+      available_tags="$available_tags _LT_TAG"
+    fi],
+      [RC], [
+    if test -n "$RC" && test "X$RC" != "Xno"; then
+      AC_LIBTOOL_LANG_RC_CONFIG
+      available_tags="$available_tags _LT_TAG"
+    fi],
+      [m4_errprintn(m4_location[: error: invalid tag name: ]"_LT_TAG")
+      m4_exit(1)])
+    ])
+    echo THOMAS: available tags: $available_tags
+  fi
+  # Now substitute the updated list of available tags.
+  if eval "sed -e 's/^available_tags=.*\$/available_tags=\"$available_tags\"/' \"$ofile\" > \"${ofile}T\""; then
+    mv "${ofile}T" "$ofile"
+    chmod +x "$ofile"
+    AC_MSG_NOTICE([updated available libtool tags with $available_tags.])
+  else
+    rm -f "${ofile}T"
+    AC_MSG_ERROR([unable to update list of available tagged configurations.])
+
+  fi
+
+])dnl _LT_AC_TAG_CONFIG
+])
diff --git a/common/m4/as-libtool.m4 b/common/m4/as-libtool.m4
new file mode 100644
index 0000000..3b16095
--- /dev/null
+++ b/common/m4/as-libtool.m4
@@ -0,0 +1,46 @@
+dnl as-libtool.m4 0.1.4
+
+dnl autostars m4 macro for libtool versioning
+
+dnl Thomas Vander Stichele 
+
+dnl $Id: as-libtool.m4,v 1.10 2005/10/15 13:44:23 thomasvs Exp $
+
+dnl AS_LIBTOOL(PREFIX, CURRENT, REVISION, AGE, [RELEASE])
+
+dnl example
+dnl AS_LIBTOOL(GST, 2, 0, 0)
+
+dnl this macro
+dnl - defines [$PREFIX]_CURRENT, REVISION and AGE
+dnl - defines [$PREFIX]_LIBVERSION
+dnl - defines [$PREFIX]_LT_LDFLAGS to set versioning
+dnl - AC_SUBST's them all
+
+dnl if RELEASE is given, then add a -release option to the LDFLAGS
+dnl with the given release version
+dnl then use [$PREFIX]_LT_LDFLAGS in the relevant Makefile.am's
+
+dnl call AM_PROG_LIBTOOL after this call
+
+AC_DEFUN([AS_LIBTOOL],
+[
+  [$1]_CURRENT=[$2]
+  [$1]_REVISION=[$3]
+  [$1]_AGE=[$4]
+  [$1]_LIBVERSION=[$2]:[$3]:[$4]
+  AC_SUBST([$1]_CURRENT)
+  AC_SUBST([$1]_REVISION)
+  AC_SUBST([$1]_AGE)
+  AC_SUBST([$1]_LIBVERSION)
+
+  [$1]_LT_LDFLAGS="$[$1]_LT_LDFLAGS -version-info $[$1]_LIBVERSION"
+  if test ! -z "[$5]"
+  then
+    [$1]_LT_LDFLAGS="$[$1]_LT_LDFLAGS -release [$5]"
+  fi
+  AC_SUBST([$1]_LT_LDFLAGS)
+
+  LT_PREREQ([2.2.6])
+  LT_INIT([dlopen win32-dll disable-static])
+])
diff --git a/common/m4/as-python.m4 b/common/m4/as-python.m4
new file mode 100644
index 0000000..eb9b175
--- /dev/null
+++ b/common/m4/as-python.m4
@@ -0,0 +1,152 @@
+## ------------------------
+## Python file handling
+## From Andrew Dalke
+## Updated by James Henstridge
+## Updated by Andy Wingo to loop through possible pythons
+## ------------------------
+
+# AS_PATH_PYTHON([MINIMUM-VERSION])
+
+# Adds support for distributing Python modules and packages.  To
+# install modules, copy them to $(pythondir), using the python_PYTHON
+# automake variable.  To install a package with the same name as the
+# automake package, install to $(pkgpythondir), or use the
+# pkgpython_PYTHON automake variable.
+
+# The variables $(pyexecdir) and $(pkgpyexecdir) are provided as
+# locations to install python extension modules (shared libraries).
+# Another macro is required to find the appropriate flags to compile
+# extension modules.
+
+# If your package is configured with a different prefix to python,
+# users will have to add the install directory to the PYTHONPATH
+# environment variable, or create a .pth file (see the python
+# documentation for details).
+
+# If the MINIMUM-VERSION argument is passed, AS_PATH_PYTHON will
+# cause an error if the version of python installed on the system
+# doesn't meet the requirement.  MINIMUM-VERSION should consist of
+# numbers and dots only.
+
+# Updated to loop over all possible python binaries by Andy Wingo
+# 
+# Updated to only warn and unset PYTHON if no good one is found
+
+AC_DEFUN([AS_PATH_PYTHON],
+ [
+  dnl Find a version of Python.  I could check for python versions 1.4
+  dnl or earlier, but the default installation locations changed from
+  dnl $prefix/lib/site-python in 1.4 to $prefix/lib/python1.5/site-packages
+  dnl in 1.5, and I don't want to maintain that logic.
+
+  dnl should we do the version check?
+  PYTHON_CANDIDATES="python python2.2 python2.1 python2.0 python2 \
+                     python1.6 python1.5"
+  ifelse([$1],[],
+         [AC_PATH_PROG(PYTHON, $PYTHON_CANDIDATES)],
+         [
+     AC_MSG_NOTICE(Looking for Python version >= $1)
+    changequote(<<, >>)dnl
+    prog="
+import sys, string
+minver = '$1'
+# split string by '.' and convert to numeric
+minver_info = map(string.atoi, string.split(minver, '.'))
+# we can now do comparisons on the two lists:
+if sys.version_info >= tuple(minver_info):
+	sys.exit(0)
+else:
+	sys.exit(1)"
+    changequote([, ])dnl
+
+    python_good=false
+    for python_candidate in $PYTHON_CANDIDATES; do
+      unset PYTHON
+      AC_PATH_PROG(PYTHON, $python_candidate) 1> /dev/null 2> /dev/null
+
+      if test "x$PYTHON" = "x"; then continue; fi
+
+      if $PYTHON -c "$prog" 1>&AC_FD_CC 2>&AC_FD_CC; then
+        AC_MSG_CHECKING(["$PYTHON":])
+	AC_MSG_RESULT([okay])
+        python_good=true
+        break;
+      else
+        dnl clear the cache val
+        unset ac_cv_path_PYTHON
+      fi
+    done
+  ])
+
+  if test "$python_good" != "true"; then
+    AC_MSG_WARN([No suitable version of python found])
+    PYTHON=
+  else
+
+  AC_MSG_CHECKING([local Python configuration])
+
+  dnl Query Python for its version number.  Getting [:3] seems to be
+  dnl the best way to do this; it's what "site.py" does in the standard
+  dnl library.  Need to change quote character because of [:3]
+
+  AC_SUBST(PYTHON_VERSION)
+  changequote(<<, >>)dnl
+  PYTHON_VERSION=`$PYTHON -c "import sys; print sys.version[:3]"`
+  changequote([, ])dnl
+
+
+  dnl Use the values of $prefix and $exec_prefix for the corresponding
+  dnl values of PYTHON_PREFIX and PYTHON_EXEC_PREFIX.  These are made
+  dnl distinct variables so they can be overridden if need be.  However,
+  dnl general consensus is that you shouldn't need this ability.
+
+  AC_SUBST(PYTHON_PREFIX)
+  PYTHON_PREFIX='${prefix}'
+
+  AC_SUBST(PYTHON_EXEC_PREFIX)
+  PYTHON_EXEC_PREFIX='${exec_prefix}'
+
+  dnl At times (like when building shared libraries) you may want
+  dnl to know which OS platform Python thinks this is.
+
+  AC_SUBST(PYTHON_PLATFORM)
+  PYTHON_PLATFORM=`$PYTHON -c "import sys; print sys.platform"`
+
+
+  dnl Set up 4 directories:
+
+  dnl pythondir -- where to install python scripts.  This is the
+  dnl   site-packages directory, not the python standard library
+  dnl   directory like in previous automake betas.  This behaviour
+  dnl   is more consistent with lispdir.m4 for example.
+  dnl
+  dnl Also, if the package prefix isn't the same as python's prefix,
+  dnl then the old $(pythondir) was pretty useless.
+
+  AC_SUBST(pythondir)
+  pythondir=$PYTHON_PREFIX"/lib/python"$PYTHON_VERSION/site-packages
+
+  dnl pkgpythondir -- $PACKAGE directory under pythondir.  Was
+  dnl   PYTHON_SITE_PACKAGE in previous betas, but this naming is
+  dnl   more consistent with the rest of automake.
+  dnl   Maybe this should be put in python.am?
+
+  AC_SUBST(pkgpythondir)
+  pkgpythondir=\${pythondir}/$PACKAGE
+
+  dnl pyexecdir -- directory for installing python extension modules
+  dnl   (shared libraries)  Was PYTHON_SITE_EXEC in previous betas.
+
+  AC_SUBST(pyexecdir)
+  pyexecdir=$PYTHON_EXEC_PREFIX"/lib/python"$PYTHON_VERSION/site-packages
+
+  dnl pkgpyexecdir -- $(pyexecdir)/$(PACKAGE)
+  dnl   Maybe this should be put in python.am?
+
+  AC_SUBST(pkgpyexecdir)
+  pkgpyexecdir=\${pyexecdir}/$PACKAGE
+
+  AC_MSG_RESULT([looks good])
+
+  fi
+])
diff --git a/common/m4/as-version.m4 b/common/m4/as-version.m4
new file mode 100644
index 0000000..22ff774
--- /dev/null
+++ b/common/m4/as-version.m4
@@ -0,0 +1,75 @@
+dnl as-version.m4 0.2.0
+
+dnl autostars m4 macro for versioning
+
+dnl Thomas Vander Stichele 
+
+dnl $Id: as-version.m4,v 1.15 2006/04/01 09:40:24 thomasvs Exp $
+
+dnl AS_VERSION
+
+dnl example
+dnl AS_VERSION
+
+dnl this macro
+dnl - AC_SUBST's PACKAGE_VERSION_MAJOR, _MINOR, _MICRO
+dnl - AC_SUBST's PACKAGE_VERSION_RELEASE,
+dnl    which can be used for rpm release fields
+dnl - doesn't call AM_INIT_AUTOMAKE anymore because it prevents
+dnl   maintainer mode from running correctly
+dnl
+dnl don't forget to put #undef PACKAGE_VERSION_RELEASE in acconfig.h
+dnl if you use acconfig.h
+
+AC_DEFUN([AS_VERSION],
+[
+  PACKAGE_VERSION_MAJOR=$(echo AC_PACKAGE_VERSION | cut -d'.' -f1)
+  PACKAGE_VERSION_MINOR=$(echo AC_PACKAGE_VERSION | cut -d'.' -f2)
+  PACKAGE_VERSION_MICRO=$(echo AC_PACKAGE_VERSION | cut -d'.' -f3)
+
+  AC_SUBST(PACKAGE_VERSION_MAJOR)
+  AC_SUBST(PACKAGE_VERSION_MINOR)
+  AC_SUBST(PACKAGE_VERSION_MICRO)
+])
+
+dnl AS_NANO(ACTION-IF-NANO-NON-NULL, [ACTION-IF-NANO-NULL])
+
+dnl requires AC_INIT to be called before
+dnl For projects using a fourth or nano number in your versioning to indicate
+dnl development or prerelease snapshots, this macro allows the build to be
+dnl set up differently accordingly.
+
+dnl this macro:
+dnl - parses AC_PACKAGE_VERSION, set by AC_INIT, and extracts the nano number
+dnl - sets the variable PACKAGE_VERSION_NANO
+dnl - sets the variable PACKAGE_VERSION_RELEASE, which can be used
+dnl   for rpm release fields
+dnl - executes ACTION-IF-NANO-NON-NULL or ACTION-IF-NANO-NULL
+
+dnl example:
+dnl AS_NANO(RELEASE="yes", RELEASE="no")
+
+AC_DEFUN([AS_NANO],
+[
+  AC_MSG_CHECKING(nano version)
+
+  NANO=$(echo AC_PACKAGE_VERSION | cut -d'.' -f4)
+
+  if test x"$NANO" = x || test "x$NANO" = "x0" ; then
+    AC_MSG_RESULT([0 (release)])
+    NANO=0
+    PACKAGE_VERSION_RELEASE=1
+    ifelse([$1], , :, [$1])
+  else
+    AC_MSG_RESULT($NANO)
+    PACKAGE_VERSION_RELEASE=0.`date +%Y%m%d.%H%M%S`
+    if test "x$NANO" != "x1" ; then
+       ifelse([$1], , :, [$1])
+    else
+       ifelse([$2], , :, [$2])
+    fi
+  fi
+  PACKAGE_VERSION_NANO=$NANO
+  AC_SUBST(PACKAGE_VERSION_NANO)
+  AC_SUBST(PACKAGE_VERSION_RELEASE)
+])
diff --git a/common/m4/ax_create_stdint_h.m4 b/common/m4/ax_create_stdint_h.m4
new file mode 100644
index 0000000..13bf699
--- /dev/null
+++ b/common/m4/ax_create_stdint_h.m4
@@ -0,0 +1,734 @@
+##### http://autoconf-archive.cryp.to/ax_create_stdint_h.html
+#
+# SYNOPSIS
+#
+#   AX_CREATE_STDINT_H [( HEADER-TO-GENERATE [, HEDERS-TO-CHECK])]
+#
+# DESCRIPTION
+#
+#   the "ISO C9X: 7.18 Integer types " section requires the
+#   existence of an include file  that defines a set of
+#   typedefs, especially uint8_t,int32_t,uintptr_t. Many older
+#   installations will not provide this file, but some will have the
+#   very same definitions in . In other enviroments we can
+#   use the inet-types in  which would define the typedefs
+#   int8_t and u_int8_t respectivly.
+#
+#   This macros will create a local "_stdint.h" or the headerfile given
+#   as an argument. In many cases that file will just "#include
+#   " or "#include ", while in other environments
+#   it will provide the set of basic 'stdint's definitions/typedefs:
+#
+#     int8_t,uint8_t,int16_t,uint16_t,int32_t,uint32_t,intptr_t,uintptr_t
+#     int_least32_t.. int_fast32_t.. intmax_t
+#
+#   which may or may not rely on the definitions of other files, or
+#   using the AC_CHECK_SIZEOF macro to determine the actual sizeof each
+#   type.
+#
+#   if your header files require the stdint-types you will want to
+#   create an installable file mylib-int.h that all your other
+#   installable header may include. So if you have a library package
+#   named "mylib", just use
+#
+#        AX_CREATE_STDINT_H(mylib-int.h)
+#
+#   in configure.ac and go to install that very header file in
+#   Makefile.am along with the other headers (mylib.h) - and the
+#   mylib-specific headers can simply use "#include " to
+#   obtain the stdint-types.
+#
+#   Remember, if the system already had a valid , the
+#   generated file will include it directly. No need for fuzzy
+#   HAVE_STDINT_H things... (oops, GCC 4.2.x has deliberatly disabled
+#   its stdint.h for non-c99 compilation and the c99-mode is not the
+#   default. Therefore this macro will not use the compiler's stdint.h
+#   - please complain to the GCC developers).
+#
+# LAST MODIFICATION
+#
+#   2007-06-27
+#
+# COPYLEFT
+#
+#   Copyright (c) 2007 Guido U. Draheim 
+#
+#   This program is free software; you can redistribute it and/or
+#   modify it under the terms of the GNU General Public License as
+#   published by the Free Software Foundation; either version 2 of the
+#   License, or (at your option) any later version.
+#
+#   This program is distributed in the hope that it will be useful, but
+#   WITHOUT ANY WARRANTY; without even the implied warranty of
+#   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+#   General Public License for more details.
+#
+#   You should have received a copy of the GNU General Public License
+#   along with this program; if not, write to the Free Software
+#   Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA
+#   02111-1307, USA.
+#
+#   As a special exception, the respective Autoconf Macro's copyright
+#   owner gives unlimited permission to copy, distribute and modify the
+#   configure scripts that are the output of Autoconf when processing
+#   the Macro. You need not follow the terms of the GNU General Public
+#   License when using or distributing such scripts, even though
+#   portions of the text of the Macro appear in them. The GNU General
+#   Public License (GPL) does govern all other use of the material that
+#   constitutes the Autoconf Macro.
+#
+#   This special exception to the GPL applies to versions of the
+#   Autoconf Macro released by the Autoconf Macro Archive. When you
+#   make and distribute a modified version of the Autoconf Macro, you
+#   may extend this special exception to the GPL to apply to your
+#   modified version as well.
+
+AC_DEFUN([AX_CHECK_DATA_MODEL],[
+   AC_CHECK_SIZEOF(char)
+   AC_CHECK_SIZEOF(short)
+   AC_CHECK_SIZEOF(int)
+   AC_CHECK_SIZEOF(long)
+   AC_CHECK_SIZEOF(void*)
+   ac_cv_char_data_model=""
+   ac_cv_char_data_model="$ac_cv_char_data_model$ac_cv_sizeof_char"
+   ac_cv_char_data_model="$ac_cv_char_data_model$ac_cv_sizeof_short"
+   ac_cv_char_data_model="$ac_cv_char_data_model$ac_cv_sizeof_int"
+   ac_cv_long_data_model=""
+   ac_cv_long_data_model="$ac_cv_long_data_model$ac_cv_sizeof_int"
+   ac_cv_long_data_model="$ac_cv_long_data_model$ac_cv_sizeof_long"
+   ac_cv_long_data_model="$ac_cv_long_data_model$ac_cv_sizeof_voidp"
+   AC_MSG_CHECKING([data model])
+   case "$ac_cv_char_data_model/$ac_cv_long_data_model" in
+    122/242)     ac_cv_data_model="IP16"  ; n="standard 16bit machine" ;;
+    122/244)     ac_cv_data_model="LP32"  ; n="standard 32bit machine" ;;
+    122/*)       ac_cv_data_model="i16"   ; n="unusual int16 model" ;;
+    124/444)     ac_cv_data_model="ILP32" ; n="standard 32bit unixish" ;;
+    124/488)     ac_cv_data_model="LP64"  ; n="standard 64bit unixish" ;;
+    124/448)     ac_cv_data_model="LLP64" ; n="unusual 64bit unixish" ;;
+    124/*)       ac_cv_data_model="i32"   ; n="unusual int32 model" ;;
+    128/888)     ac_cv_data_model="ILP64" ; n="unusual 64bit numeric" ;;
+    128/*)       ac_cv_data_model="i64"   ; n="unusual int64 model" ;;
+    222/*2)      ac_cv_data_model="DSP16" ; n="strict 16bit dsptype" ;;
+    333/*3)      ac_cv_data_model="DSP24" ; n="strict 24bit dsptype" ;;
+    444/*4)      ac_cv_data_model="DSP32" ; n="strict 32bit dsptype" ;;
+    666/*6)      ac_cv_data_model="DSP48" ; n="strict 48bit dsptype" ;;
+    888/*8)      ac_cv_data_model="DSP64" ; n="strict 64bit dsptype" ;;
+    222/*|333/*|444/*|666/*|888/*) :
+                 ac_cv_data_model="iDSP"  ; n="unusual dsptype" ;;
+     *)          ac_cv_data_model="none"  ; n="very unusual model" ;;
+   esac
+   AC_MSG_RESULT([$ac_cv_data_model ($ac_cv_long_data_model, $n)])
+])
+
+dnl AX_CHECK_HEADER_STDINT_X([HEADERLIST][,ACTION-IF])
+AC_DEFUN([AX_CHECK_HEADER_STDINT_X],[
+AC_CACHE_CHECK([for stdint uintptr_t], [ac_cv_header_stdint_x],[
+ ac_cv_header_stdint_x="" # the 1997 typedefs (inttypes.h)
+  AC_MSG_RESULT([(..)])
+  for i in m4_ifval([$1],[$1],[stdint.h inttypes.h sys/inttypes.h sys/types.h])
+  do
+   unset ac_cv_type_uintptr_t
+   unset ac_cv_type_uint64_t
+   AC_CHECK_TYPE(uintptr_t,[ac_cv_header_stdint_x=$i],continue,[#include <$i>])
+   AC_CHECK_TYPE(uint64_t,[and64="/uint64_t"],[and64=""],[#include<$i>])
+   m4_ifvaln([$2],[$2]) break
+  done
+  AC_MSG_CHECKING([for stdint uintptr_t])
+ ])
+])
+
+AC_DEFUN([AX_CHECK_HEADER_STDINT_O],[
+AC_CACHE_CHECK([for stdint uint32_t], [ac_cv_header_stdint_o],[
+ ac_cv_header_stdint_o="" # the 1995 typedefs (sys/inttypes.h)
+  AC_MSG_RESULT([(..)])
+  for i in m4_ifval([$1],[$1],[inttypes.h sys/inttypes.h sys/types.h stdint.h])
+  do
+   unset ac_cv_type_uint32_t
+   unset ac_cv_type_uint64_t
+   AC_CHECK_TYPE(uint32_t,[ac_cv_header_stdint_o=$i],continue,[#include <$i>])
+   AC_CHECK_TYPE(uint64_t,[and64="/uint64_t"],[and64=""],[#include<$i>])
+   m4_ifvaln([$2],[$2]) break
+   break;
+  done
+  AC_MSG_CHECKING([for stdint uint32_t])
+ ])
+])
+
+AC_DEFUN([AX_CHECK_HEADER_STDINT_U],[
+AC_CACHE_CHECK([for stdint u_int32_t], [ac_cv_header_stdint_u],[
+ ac_cv_header_stdint_u="" # the BSD typedefs (sys/types.h)
+  AC_MSG_RESULT([(..)])
+  for i in m4_ifval([$1],[$1],[sys/types.h inttypes.h sys/inttypes.h]) ; do
+   unset ac_cv_type_u_int32_t
+   unset ac_cv_type_u_int64_t
+   AC_CHECK_TYPE(u_int32_t,[ac_cv_header_stdint_u=$i],continue,[#include <$i>])
+   AC_CHECK_TYPE(u_int64_t,[and64="/u_int64_t"],[and64=""],[#include<$i>])
+   m4_ifvaln([$2],[$2]) break
+   break;
+  done
+  AC_MSG_CHECKING([for stdint u_int32_t])
+ ])
+])
+
+AC_DEFUN([AX_CREATE_STDINT_H],
+[# ------ AX CREATE STDINT H -------------------------------------
+AC_MSG_CHECKING([for stdint types])
+ac_stdint_h=`echo ifelse($1, , _stdint.h, $1)`
+# try to shortcircuit - if the default include path of the compiler
+# can find a "stdint.h" header then we assume that all compilers can.
+AC_CACHE_VAL([ac_cv_header_stdint_t],[
+old_CXXFLAGS="$CXXFLAGS" ; CXXFLAGS=""
+old_CPPFLAGS="$CPPFLAGS" ; CPPFLAGS=""
+old_CFLAGS="$CFLAGS"     ; CFLAGS=""
+AC_TRY_COMPILE([#include ],[int_least32_t v = 0;],
+[ac_cv_stdint_result="(assuming C99 compatible system)"
+ ac_cv_header_stdint_t="stdint.h"; ],
+[ac_cv_header_stdint_t=""])
+if test "$GCC" = "yes" && test ".$ac_cv_header_stdint_t" = "."; then
+CFLAGS="-std=c99"
+AC_TRY_COMPILE([#include ],[int_least32_t v = 0;],
+[AC_MSG_WARN(your GCC compiler has a defunct stdint.h for its default-mode)])
+fi
+CXXFLAGS="$old_CXXFLAGS"
+CPPFLAGS="$old_CPPFLAGS"
+CFLAGS="$old_CFLAGS" ])
+
+v="... $ac_cv_header_stdint_h"
+if test "$ac_stdint_h" = "stdint.h" ; then
+ AC_MSG_RESULT([(are you sure you want them in ./stdint.h?)])
+elif test "$ac_stdint_h" = "inttypes.h" ; then
+ AC_MSG_RESULT([(are you sure you want them in ./inttypes.h?)])
+elif test "_$ac_cv_header_stdint_t" = "_" ; then
+ AC_MSG_RESULT([(putting them into $ac_stdint_h)$v])
+else
+ ac_cv_header_stdint="$ac_cv_header_stdint_t"
+ AC_MSG_RESULT([$ac_cv_header_stdint (shortcircuit)])
+fi
+
+if test "_$ac_cv_header_stdint_t" = "_" ; then # can not shortcircuit..
+
+dnl .....intro message done, now do a few system checks.....
+dnl btw, all old CHECK_TYPE macros do automatically "DEFINE" a type,
+dnl therefore we use the autoconf implementation detail CHECK_TYPE_NEW
+dnl instead that is triggered with 3 or more arguments (see types.m4)
+
+inttype_headers=`echo $2 | sed -e 's/,/ /g'`
+
+ac_cv_stdint_result="(no helpful system typedefs seen)"
+AX_CHECK_HEADER_STDINT_X(dnl
+   stdint.h inttypes.h sys/inttypes.h $inttype_headers,
+   ac_cv_stdint_result="(seen uintptr_t$and64 in $i)")
+
+if test "_$ac_cv_header_stdint_x" = "_" ; then
+AX_CHECK_HEADER_STDINT_O(dnl,
+   inttypes.h sys/inttypes.h stdint.h $inttype_headers,
+   ac_cv_stdint_result="(seen uint32_t$and64 in $i)")
+fi
+
+if test "_$ac_cv_header_stdint_x" = "_" ; then
+if test "_$ac_cv_header_stdint_o" = "_" ; then
+AX_CHECK_HEADER_STDINT_U(dnl,
+   sys/types.h inttypes.h sys/inttypes.h $inttype_headers,
+   ac_cv_stdint_result="(seen u_int32_t$and64 in $i)")
+fi fi
+
+dnl if there was no good C99 header file, do some typedef checks...
+if test "_$ac_cv_header_stdint_x" = "_" ; then
+   AC_MSG_CHECKING([for stdint datatype model])
+   AC_MSG_RESULT([(..)])
+   AX_CHECK_DATA_MODEL
+fi
+
+if test "_$ac_cv_header_stdint_x" != "_" ; then
+   ac_cv_header_stdint="$ac_cv_header_stdint_x"
+elif  test "_$ac_cv_header_stdint_o" != "_" ; then
+   ac_cv_header_stdint="$ac_cv_header_stdint_o"
+elif  test "_$ac_cv_header_stdint_u" != "_" ; then
+   ac_cv_header_stdint="$ac_cv_header_stdint_u"
+else
+   ac_cv_header_stdint="stddef.h"
+fi
+
+AC_MSG_CHECKING([for extra inttypes in chosen header])
+AC_MSG_RESULT([($ac_cv_header_stdint)])
+dnl see if int_least and int_fast types are present in _this_ header.
+unset ac_cv_type_int_least32_t
+unset ac_cv_type_int_fast32_t
+AC_CHECK_TYPE(int_least32_t,,,[#include <$ac_cv_header_stdint>])
+AC_CHECK_TYPE(int_fast32_t,,,[#include<$ac_cv_header_stdint>])
+AC_CHECK_TYPE(intmax_t,,,[#include <$ac_cv_header_stdint>])
+
+fi # shortcircut to system "stdint.h"
+# ------------------ PREPARE VARIABLES ------------------------------
+if test "$GCC" = "yes" ; then
+ac_cv_stdint_message="using gnu compiler "`$CC --version | head -1`
+else
+ac_cv_stdint_message="using $CC"
+fi
+
+AC_MSG_RESULT([make use of $ac_cv_header_stdint in $ac_stdint_h dnl
+$ac_cv_stdint_result])
+
+dnl -----------------------------------------------------------------
+# ----------------- DONE inttypes.h checks START header -------------
+AC_CONFIG_COMMANDS([$ac_stdint_h],[
+AC_MSG_NOTICE(creating $ac_stdint_h : $_ac_stdint_h)
+ac_stdint=$tmp/_stdint.h
+
+echo "#ifndef" $_ac_stdint_h >$ac_stdint
+echo "#define" $_ac_stdint_h "1" >>$ac_stdint
+echo "#ifndef" _GENERATED_STDINT_H >>$ac_stdint
+echo "#define" _GENERATED_STDINT_H '"'$PACKAGE $VERSION'"' >>$ac_stdint
+echo "/* generated $ac_cv_stdint_message */" >>$ac_stdint
+if test "_$ac_cv_header_stdint_t" != "_" ; then
+echo "#define _STDINT_HAVE_STDINT_H" "1" >>$ac_stdint
+echo "#include " >>$ac_stdint
+echo "#endif" >>$ac_stdint
+echo "#endif" >>$ac_stdint
+else
+
+cat >>$ac_stdint <
+#else
+#include 
+
+/* .................... configured part ............................ */
+
+STDINT_EOF
+
+echo "/* whether we have a C99 compatible stdint header file */" >>$ac_stdint
+if test "_$ac_cv_header_stdint_x" != "_" ; then
+  ac_header="$ac_cv_header_stdint_x"
+  echo "#define _STDINT_HEADER_INTPTR" '"'"$ac_header"'"' >>$ac_stdint
+else
+  echo "/* #undef _STDINT_HEADER_INTPTR */" >>$ac_stdint
+fi
+
+echo "/* whether we have a C96 compatible inttypes header file */" >>$ac_stdint
+if  test "_$ac_cv_header_stdint_o" != "_" ; then
+  ac_header="$ac_cv_header_stdint_o"
+  echo "#define _STDINT_HEADER_UINT32" '"'"$ac_header"'"' >>$ac_stdint
+else
+  echo "/* #undef _STDINT_HEADER_UINT32 */" >>$ac_stdint
+fi
+
+echo "/* whether we have a BSD compatible inet types header */" >>$ac_stdint
+if  test "_$ac_cv_header_stdint_u" != "_" ; then
+  ac_header="$ac_cv_header_stdint_u"
+  echo "#define _STDINT_HEADER_U_INT32" '"'"$ac_header"'"' >>$ac_stdint
+else
+  echo "/* #undef _STDINT_HEADER_U_INT32 */" >>$ac_stdint
+fi
+
+echo "" >>$ac_stdint
+
+if test "_$ac_header" != "_" ; then if test "$ac_header" != "stddef.h" ; then
+  echo "#include <$ac_header>" >>$ac_stdint
+  echo "" >>$ac_stdint
+fi fi
+
+echo "/* which 64bit typedef has been found */" >>$ac_stdint
+if test "$ac_cv_type_uint64_t" = "yes" ; then
+echo "#define   _STDINT_HAVE_UINT64_T" "1"  >>$ac_stdint
+else
+echo "/* #undef _STDINT_HAVE_UINT64_T */" >>$ac_stdint
+fi
+if test "$ac_cv_type_u_int64_t" = "yes" ; then
+echo "#define   _STDINT_HAVE_U_INT64_T" "1"  >>$ac_stdint
+else
+echo "/* #undef _STDINT_HAVE_U_INT64_T */" >>$ac_stdint
+fi
+echo "" >>$ac_stdint
+
+echo "/* which type model has been detected */" >>$ac_stdint
+if test "_$ac_cv_char_data_model" != "_" ; then
+echo "#define   _STDINT_CHAR_MODEL" "$ac_cv_char_data_model" >>$ac_stdint
+echo "#define   _STDINT_LONG_MODEL" "$ac_cv_long_data_model" >>$ac_stdint
+else
+echo "/* #undef _STDINT_CHAR_MODEL // skipped */" >>$ac_stdint
+echo "/* #undef _STDINT_LONG_MODEL // skipped */" >>$ac_stdint
+fi
+echo "" >>$ac_stdint
+
+echo "/* whether int_least types were detected */" >>$ac_stdint
+if test "$ac_cv_type_int_least32_t" = "yes"; then
+echo "#define   _STDINT_HAVE_INT_LEAST32_T" "1"  >>$ac_stdint
+else
+echo "/* #undef _STDINT_HAVE_INT_LEAST32_T */" >>$ac_stdint
+fi
+echo "/* whether int_fast types were detected */" >>$ac_stdint
+if test "$ac_cv_type_int_fast32_t" = "yes"; then
+echo "#define   _STDINT_HAVE_INT_FAST32_T" "1" >>$ac_stdint
+else
+echo "/* #undef _STDINT_HAVE_INT_FAST32_T */" >>$ac_stdint
+fi
+echo "/* whether intmax_t type was detected */" >>$ac_stdint
+if test "$ac_cv_type_intmax_t" = "yes"; then
+echo "#define   _STDINT_HAVE_INTMAX_T" "1" >>$ac_stdint
+else
+echo "/* #undef _STDINT_HAVE_INTMAX_T */" >>$ac_stdint
+fi
+echo "" >>$ac_stdint
+
+  cat >>$ac_stdint <= 199901L
+#define _HAVE_UINT64_T
+#define _HAVE_LONGLONG_UINT64_T
+typedef long long int64_t;
+typedef unsigned long long uint64_t;
+
+#elif !defined __STRICT_ANSI__
+#if defined _MSC_VER || defined __WATCOMC__ || defined __BORLANDC__
+#define _HAVE_UINT64_T
+typedef __int64 int64_t;
+typedef unsigned __int64 uint64_t;
+
+#elif defined __GNUC__ || defined __MWERKS__ || defined __ELF__
+/* note: all ELF-systems seem to have loff-support which needs 64-bit */
+#if !defined _NO_LONGLONG
+#define _HAVE_UINT64_T
+#define _HAVE_LONGLONG_UINT64_T
+typedef long long int64_t;
+typedef unsigned long long uint64_t;
+#endif
+
+#elif defined __alpha || (defined __mips && defined _ABIN32)
+#if !defined _NO_LONGLONG
+typedef long int64_t;
+typedef unsigned long uint64_t;
+#endif
+  /* compiler/cpu type to define int64_t */
+#endif
+#endif
+#endif
+
+#if defined _STDINT_HAVE_U_INT_TYPES
+/* int8_t int16_t int32_t defined by inet code, redeclare the u_intXX types */
+typedef u_int8_t uint8_t;
+typedef u_int16_t uint16_t;
+typedef u_int32_t uint32_t;
+
+/* glibc compatibility */
+#ifndef __int8_t_defined
+#define __int8_t_defined
+#endif
+#endif
+
+#ifdef _STDINT_NEED_INT_MODEL_T
+/* we must guess all the basic types. Apart from byte-adressable system, */
+/* there a few 32-bit-only dsp-systems that we guard with BYTE_MODEL 8-} */
+/* (btw, those nibble-addressable systems are way off, or so we assume) */
+
+dnl   /* have a look at "64bit and data size neutrality" at */
+dnl   /* http://unix.org/version2/whatsnew/login_64bit.html */
+dnl   /* (the shorthand "ILP" types always have a "P" part) */
+
+#if defined _STDINT_BYTE_MODEL
+#if _STDINT_LONG_MODEL+0 == 242
+/* 2:4:2 =  IP16 = a normal 16-bit system                */
+typedef unsigned char   uint8_t;
+typedef unsigned short  uint16_t;
+typedef unsigned long   uint32_t;
+#ifndef __int8_t_defined
+#define __int8_t_defined
+typedef          char    int8_t;
+typedef          short   int16_t;
+typedef          long    int32_t;
+#endif
+#elif _STDINT_LONG_MODEL+0 == 244 || _STDINT_LONG_MODEL == 444
+/* 2:4:4 =  LP32 = a 32-bit system derived from a 16-bit */
+/* 4:4:4 = ILP32 = a normal 32-bit system                */
+typedef unsigned char   uint8_t;
+typedef unsigned short  uint16_t;
+typedef unsigned int    uint32_t;
+#ifndef __int8_t_defined
+#define __int8_t_defined
+typedef          char    int8_t;
+typedef          short   int16_t;
+typedef          int     int32_t;
+#endif
+#elif _STDINT_LONG_MODEL+0 == 484 || _STDINT_LONG_MODEL+0 == 488
+/* 4:8:4 =  IP32 = a 32-bit system prepared for 64-bit    */
+/* 4:8:8 =  LP64 = a normal 64-bit system                 */
+typedef unsigned char   uint8_t;
+typedef unsigned short  uint16_t;
+typedef unsigned int    uint32_t;
+#ifndef __int8_t_defined
+#define __int8_t_defined
+typedef          char    int8_t;
+typedef          short   int16_t;
+typedef          int     int32_t;
+#endif
+/* this system has a "long" of 64bit */
+#ifndef _HAVE_UINT64_T
+#define _HAVE_UINT64_T
+typedef unsigned long   uint64_t;
+typedef          long    int64_t;
+#endif
+#elif _STDINT_LONG_MODEL+0 == 448
+/*      LLP64   a 64-bit system derived from a 32-bit system */
+typedef unsigned char   uint8_t;
+typedef unsigned short  uint16_t;
+typedef unsigned int    uint32_t;
+#ifndef __int8_t_defined
+#define __int8_t_defined
+typedef          char    int8_t;
+typedef          short   int16_t;
+typedef          int     int32_t;
+#endif
+/* assuming the system has a "long long" */
+#ifndef _HAVE_UINT64_T
+#define _HAVE_UINT64_T
+#define _HAVE_LONGLONG_UINT64_T
+typedef unsigned long long uint64_t;
+typedef          long long  int64_t;
+#endif
+#else
+#define _STDINT_NO_INT32_T
+#endif
+#else
+#define _STDINT_NO_INT8_T
+#define _STDINT_NO_INT32_T
+#endif
+#endif
+
+/*
+ * quote from SunOS-5.8 sys/inttypes.h:
+ * Use at your own risk.  As of February 1996, the committee is squarely
+ * behind the fixed sized types; the "least" and "fast" types are still being
+ * discussed.  The probability that the "fast" types may be removed before
+ * the standard is finalized is high enough that they are not currently
+ * implemented.
+ */
+
+#if defined _STDINT_NEED_INT_LEAST_T
+typedef  int8_t    int_least8_t;
+typedef  int16_t   int_least16_t;
+typedef  int32_t   int_least32_t;
+#ifdef _HAVE_UINT64_T
+typedef  int64_t   int_least64_t;
+#endif
+
+typedef uint8_t   uint_least8_t;
+typedef uint16_t  uint_least16_t;
+typedef uint32_t  uint_least32_t;
+#ifdef _HAVE_UINT64_T
+typedef uint64_t  uint_least64_t;
+#endif
+  /* least types */
+#endif
+
+#if defined _STDINT_NEED_INT_FAST_T
+typedef  int8_t    int_fast8_t;
+typedef  int       int_fast16_t;
+typedef  int32_t   int_fast32_t;
+#ifdef _HAVE_UINT64_T
+typedef  int64_t   int_fast64_t;
+#endif
+
+typedef uint8_t   uint_fast8_t;
+typedef unsigned  uint_fast16_t;
+typedef uint32_t  uint_fast32_t;
+#ifdef _HAVE_UINT64_T
+typedef uint64_t  uint_fast64_t;
+#endif
+  /* fast types */
+#endif
+
+#ifdef _STDINT_NEED_INTMAX_T
+#ifdef _HAVE_UINT64_T
+typedef  int64_t       intmax_t;
+typedef uint64_t      uintmax_t;
+#else
+typedef          long  intmax_t;
+typedef unsigned long uintmax_t;
+#endif
+#endif
+
+#ifdef _STDINT_NEED_INTPTR_T
+#ifndef __intptr_t_defined
+#define __intptr_t_defined
+/* we encourage using "long" to store pointer values, never use "int" ! */
+#if   _STDINT_LONG_MODEL+0 == 242 || _STDINT_LONG_MODEL+0 == 484
+typedef  unsigned int   uintptr_t;
+typedef           int    intptr_t;
+#elif _STDINT_LONG_MODEL+0 == 244 || _STDINT_LONG_MODEL+0 == 444
+typedef  unsigned long  uintptr_t;
+typedef           long   intptr_t;
+#elif _STDINT_LONG_MODEL+0 == 448 && defined _HAVE_UINT64_T
+typedef        uint64_t uintptr_t;
+typedef         int64_t  intptr_t;
+#else /* matches typical system types ILP32 and LP64 - but not IP16 or LLP64 */
+typedef  unsigned long  uintptr_t;
+typedef           long   intptr_t;
+#endif
+#endif
+#endif
+
+/* The ISO C99 standard specifies that in C++ implementations these
+   should only be defined if explicitly requested.  */
+#if !defined __cplusplus || defined __STDC_CONSTANT_MACROS
+#ifndef UINT32_C
+
+/* Signed.  */
+# define INT8_C(c)      c
+# define INT16_C(c)     c
+# define INT32_C(c)     c
+# ifdef _HAVE_LONGLONG_UINT64_T
+#  define INT64_C(c)    c ## L
+# else
+#  define INT64_C(c)    c ## LL
+# endif
+
+/* Unsigned.  */
+# define UINT8_C(c)     c ## U
+# define UINT16_C(c)    c ## U
+# define UINT32_C(c)    c ## U
+# ifdef _HAVE_LONGLONG_UINT64_T
+#  define UINT64_C(c)   c ## UL
+# else
+#  define UINT64_C(c)   c ## ULL
+# endif
+
+/* Maximal type.  */
+# ifdef _HAVE_LONGLONG_UINT64_T
+#  define INTMAX_C(c)   c ## L
+#  define UINTMAX_C(c)  c ## UL
+# else
+#  define INTMAX_C(c)   c ## LL
+#  define UINTMAX_C(c)  c ## ULL
+# endif
+
+  /* literalnumbers */
+#endif
+#endif
+
+/* These limits are merily those of a two complement byte-oriented system */
+
+/* Minimum of signed integral types.  */
+# define INT8_MIN               (-128)
+# define INT16_MIN              (-32767-1)
+# define INT32_MIN              (-2147483647-1)
+# define INT64_MIN              (-__INT64_C(9223372036854775807)-1)
+/* Maximum of signed integral types.  */
+# define INT8_MAX               (127)
+# define INT16_MAX              (32767)
+# define INT32_MAX              (2147483647)
+# define INT64_MAX              (__INT64_C(9223372036854775807))
+
+/* Maximum of unsigned integral types.  */
+# define UINT8_MAX              (255)
+# define UINT16_MAX             (65535)
+# define UINT32_MAX             (4294967295U)
+# define UINT64_MAX             (__UINT64_C(18446744073709551615))
+
+/* Minimum of signed integral types having a minimum size.  */
+# define INT_LEAST8_MIN         INT8_MIN
+# define INT_LEAST16_MIN        INT16_MIN
+# define INT_LEAST32_MIN        INT32_MIN
+# define INT_LEAST64_MIN        INT64_MIN
+/* Maximum of signed integral types having a minimum size.  */
+# define INT_LEAST8_MAX         INT8_MAX
+# define INT_LEAST16_MAX        INT16_MAX
+# define INT_LEAST32_MAX        INT32_MAX
+# define INT_LEAST64_MAX        INT64_MAX
+
+/* Maximum of unsigned integral types having a minimum size.  */
+# define UINT_LEAST8_MAX        UINT8_MAX
+# define UINT_LEAST16_MAX       UINT16_MAX
+# define UINT_LEAST32_MAX       UINT32_MAX
+# define UINT_LEAST64_MAX       UINT64_MAX
+
+  /* shortcircuit*/
+#endif
+  /* once */
+#endif
+#endif
+STDINT_EOF
+fi
+    if cmp -s $ac_stdint_h $ac_stdint 2>/dev/null; then
+      AC_MSG_NOTICE([$ac_stdint_h is unchanged])
+    else
+      ac_dir=`AS_DIRNAME(["$ac_stdint_h"])`
+      AS_MKDIR_P(["$ac_dir"])
+      rm -f $ac_stdint_h
+      mv $ac_stdint $ac_stdint_h
+    fi
+],[# variables for create stdint.h replacement
+PACKAGE="$PACKAGE"
+VERSION="$VERSION"
+ac_stdint_h="$ac_stdint_h"
+_ac_stdint_h=AS_TR_CPP(_$PACKAGE-$ac_stdint_h)
+ac_cv_stdint_message="$ac_cv_stdint_message"
+ac_cv_header_stdint_t="$ac_cv_header_stdint_t"
+ac_cv_header_stdint_x="$ac_cv_header_stdint_x"
+ac_cv_header_stdint_o="$ac_cv_header_stdint_o"
+ac_cv_header_stdint_u="$ac_cv_header_stdint_u"
+ac_cv_type_uint64_t="$ac_cv_type_uint64_t"
+ac_cv_type_u_int64_t="$ac_cv_type_u_int64_t"
+ac_cv_char_data_model="$ac_cv_char_data_model"
+ac_cv_long_data_model="$ac_cv_long_data_model"
+ac_cv_type_int_least32_t="$ac_cv_type_int_least32_t"
+ac_cv_type_int_fast32_t="$ac_cv_type_int_fast32_t"
+ac_cv_type_intmax_t="$ac_cv_type_intmax_t"
+])
+])
diff --git a/common/m4/check.m4 b/common/m4/check.m4
new file mode 100644
index 0000000..afd26eb
--- /dev/null
+++ b/common/m4/check.m4
@@ -0,0 +1,181 @@
+dnl _AM_TRY_CHECK(MINIMUM-VERSION, EXTRA-CFLAGS, EXTRA-LIBS, CHECK-LIB-NAME
+dnl               [, ACTION-IF-FOUND [, ACTION-IF-NOT-FOUND]])
+dnl Test for check, and define CHECK_CFLAGS and CHECK_LIBS
+dnl Done this way because of the brokenness that is
+dnl https://launchpad.net/distros/ubuntu/+source/check/+bug/5840
+dnl
+
+AC_DEFUN([_AM_TRY_CHECK],
+[
+  min_check_version=$1
+  extra_cflags=$2
+  extra_libs=$3
+  check_lib_name=$4
+
+  CHECK_CFLAGS="$extra_cflags"
+  CHECK_LIBS="$extra_libs -l$check_lib_name"
+
+  ac_save_CFLAGS="$CFLAGS"
+  ac_save_LIBS="$LIBS"
+
+  CFLAGS="$CFLAGS $CHECK_CFLAGS"
+  LIBS="$CHECK_LIBS $LIBS"
+
+  AC_MSG_CHECKING(for check named $check_lib_name  - version >= $min_check_version)
+
+  rm -f conf.check-test
+  dnl unset no_check, since in our second run it would have been set to yes
+  dnl before
+  no_check=
+  AC_TRY_RUN([
+#include 
+#include 
+
+#include 
+
+int main ()
+{
+  int major, minor, micro;
+  char *tmp_version;
+
+  system ("touch conf.check-test");
+
+  /* HP/UX 9 (%@#!) writes to sscanf strings */
+  tmp_version = strdup("$min_check_version");
+  if (sscanf(tmp_version, "%d.%d.%d", &major, &minor, µ) != 3) {
+     printf("%s, bad version string\n", "$min_check_version");
+     return 1;
+   }
+
+  if ((CHECK_MAJOR_VERSION != check_major_version) ||
+      (CHECK_MINOR_VERSION != check_minor_version) ||
+      (CHECK_MICRO_VERSION != check_micro_version))
+    {
+      printf("\n*** The check header file (version %d.%d.%d) does not match\n",
+	     CHECK_MAJOR_VERSION, CHECK_MINOR_VERSION, CHECK_MICRO_VERSION);
+      printf("*** the check library (version %d.%d.%d).\n",
+	     check_major_version, check_minor_version, check_micro_version);
+      return 1;
+    }
+
+  if ((check_major_version > major) ||
+      ((check_major_version == major) && (check_minor_version > minor)) ||
+      ((check_major_version == major) && (check_minor_version == minor) && (check_micro_version >= micro)))
+    {
+      return 0;
+    }
+  else
+    {
+      printf("\n*** An old version of check (%d.%d.%d) was found.\n",
+             check_major_version, check_minor_version, check_micro_version);
+      printf("*** You need a version of check being at least %d.%d.%d.\n", major, minor, micro);
+      printf("***\n");
+      printf("*** If you have already installed a sufficiently new version, this error\n");
+      printf("*** probably means that the wrong copy of the check library and header\n");
+      printf("*** file is being found. Rerun configure with the --with-check=PATH option\n");
+      printf("*** to specify the prefix where the correct version was installed.\n");
+    }
+
+  return 1;
+}
+],, no_check=yes, [echo $ac_n "cross compiling; assumed OK... $ac_c"])
+
+  CFLAGS="$ac_save_CFLAGS"
+  LIBS="$ac_save_LIBS"
+
+  if test "x$no_check" = x ; then
+    AC_MSG_RESULT(yes)
+    ifelse([$5], , :, [$5])
+  else
+    AC_MSG_RESULT(no)
+    if test -f conf.check-test ; then
+      :
+    else
+      echo "*** Could not run check test program, checking why..."
+      CFLAGS="$CFLAGS $CHECK_CFLAGS"
+      LIBS="$CHECK_LIBS $LIBS"
+      AC_TRY_LINK([
+#include 
+#include 
+
+#include 
+], ,  [ echo "*** The test program compiled, but did not run. This usually means"
+        echo "*** that the run-time linker is not finding check. You'll need to set your"
+        echo "*** LD_LIBRARY_PATH environment variable, or edit /etc/ld.so.conf to point"
+        echo "*** to the installed location  Also, make sure you have run ldconfig if that"
+        echo "*** is required on your system"
+	echo "***"
+        echo "*** If you have an old version installed, it is best to remove it, although"
+        echo "*** you may also be able to get things to work by modifying LD_LIBRARY_PATH"],
+      [ echo "*** The test program failed to compile or link. See the file config.log for"
+        echo "*** the exact error that occured." ])
+
+      CFLAGS="$ac_save_CFLAGS"
+      LIBS="$ac_save_LIBS"
+    fi
+
+    CHECK_CFLAGS=""
+    CHECK_LIBS=""
+
+    rm -f conf.check-test
+    ifelse([$6], , AC_MSG_ERROR([check not found]), [$6])
+  fi
+])
+
+
+dnl AM_PATH_CHECK([MINIMUM-VERSION, [ACTION-IF-FOUND [, ACTION-IF-NOT-FOUND]]])
+dnl Test for check, and define CHECK_CFLAGS and CHECK_LIBS
+dnl
+
+AC_DEFUN([AM_PATH_CHECK],
+[
+  AC_ARG_WITH(check,
+  [  --with-check=PATH       prefix where check is installed [default=auto]])
+
+  AC_ARG_WITH(checklibname,
+    AC_HELP_STRING([--with-check-lib-name=NAME],
+      [name of the PIC check library (default=check)]))
+
+  min_check_version=ifelse([$1], ,0.8.2,$1)
+
+  if test x$with_check = xno; then
+    AC_MSG_RESULT(disabled)
+    ifelse([$3], , AC_MSG_ERROR([disabling check is not supported]), [$3])
+  else
+    if test "x$with_check" != x; then
+      CHECK_EXTRA_CFLAGS="-I$with_check/include"
+      CHECK_EXTRA_LIBS="-L$with_check/lib"
+    else
+      CHECK_EXTRA_CFLAGS=""
+      CHECK_EXTRA_LIBS=""
+    fi
+
+    if test x$with_checklibname = x; then
+      _AM_TRY_CHECK($min_check_version, $CHECK_EXTRA_CFLAGS, $CHECK_EXTRA_LIBS,
+        check_pic, [have_check=true], [have_check=false])
+      if test x$have_check = xtrue; then
+        ifelse([$2], , :, [$2])
+      else
+        _AM_TRY_CHECK($min_check_version, $CHECK_EXTRA_CFLAGS, $CHECK_EXTRA_LIBS,
+          check, [have_check=true], [have_check=false])
+        if test x$have_check = xtrue; then
+          ifelse([$2], , :, [$2])
+        else
+          ifelse([$3], , AC_MSG_ERROR([check not found]), [$3])
+        fi
+      fi
+    else
+      _AM_TRY_CHECK($min_check_version, $CHECK_EXTRA_CFLAGS, $CHECK_EXTRA_LIBS,
+        $with_checklibname, [have_check=true], [have_check=false])
+      if test x$have_check = xtrue; then
+        ifelse([$2], , :, [$2])
+      else
+        ifelse([$3], , AC_MSG_ERROR([check not found]), [$3])
+      fi
+    fi
+
+    AC_SUBST(CHECK_CFLAGS)
+    AC_SUBST(CHECK_LIBS)
+    rm -f conf.check-test
+  fi
+])
diff --git a/common/m4/glib-gettext.m4 b/common/m4/glib-gettext.m4
new file mode 100644
index 0000000..f8d442f
--- /dev/null
+++ b/common/m4/glib-gettext.m4
@@ -0,0 +1,432 @@
+# Copyright (C) 1995-2002 Free Software Foundation, Inc.
+# Copyright (C) 2001-2003,2004 Red Hat, Inc.
+#
+# This file is free software, distributed under the terms of the GNU
+# General Public License.  As a special exception to the GNU General
+# Public License, this file may be distributed as part of a program
+# that contains a configuration script generated by Autoconf, under
+# the same distribution terms as the rest of that program.
+#
+# This file can be copied and used freely without restrictions.  It can
+# be used in projects which are not available under the GNU Public License
+# but which still want to provide support for the GNU gettext functionality.
+#
+# Macro to add for using GNU gettext.
+# Ulrich Drepper , 1995, 1996
+#
+# Modified to never use included libintl.
+# Owen Taylor , 12/15/1998
+#
+# Major rework to remove unused code
+# Owen Taylor , 12/11/2002
+#
+# Added better handling of ALL_LINGUAS from GNU gettext version
+# written by Bruno Haible, Owen Taylor  5/30/3002
+#
+# Modified to require ngettext
+# Matthias Clasen  08/06/2004
+#
+# We need this here as well, since someone might use autoconf-2.5x
+# to configure GLib then an older version to configure a package
+# using AM_GLIB_GNU_GETTEXT
+AC_PREREQ(2.53)
+
+dnl
+dnl We go to great lengths to make sure that aclocal won't
+dnl try to pull in the installed version of these macros
+dnl when running aclocal in the glib directory.
+dnl
+m4_copy([AC_DEFUN],[glib_DEFUN])
+m4_copy([AC_REQUIRE],[glib_REQUIRE])
+dnl
+dnl At the end, if we're not within glib, we'll define the public
+dnl definitions in terms of our private definitions.
+dnl
+
+# GLIB_LC_MESSAGES
+#--------------------
+glib_DEFUN([GLIB_LC_MESSAGES],
+  [AC_CHECK_HEADERS([locale.h])
+    if test $ac_cv_header_locale_h = yes; then
+    AC_CACHE_CHECK([for LC_MESSAGES], am_cv_val_LC_MESSAGES,
+      [AC_TRY_LINK([#include ], [return LC_MESSAGES],
+       am_cv_val_LC_MESSAGES=yes, am_cv_val_LC_MESSAGES=no)])
+    if test $am_cv_val_LC_MESSAGES = yes; then
+      AC_DEFINE(HAVE_LC_MESSAGES, 1,
+        [Define if your  file defines LC_MESSAGES.])
+    fi
+  fi])
+
+# GLIB_PATH_PROG_WITH_TEST
+#----------------------------
+dnl GLIB_PATH_PROG_WITH_TEST(VARIABLE, PROG-TO-CHECK-FOR,
+dnl   TEST-PERFORMED-ON-FOUND_PROGRAM [, VALUE-IF-NOT-FOUND [, PATH]])
+glib_DEFUN([GLIB_PATH_PROG_WITH_TEST],
+[# Extract the first word of "$2", so it can be a program name with args.
+set dummy $2; ac_word=[$]2
+AC_MSG_CHECKING([for $ac_word])
+AC_CACHE_VAL(ac_cv_path_$1,
+[case "[$]$1" in
+  /*)
+  ac_cv_path_$1="[$]$1" # Let the user override the test with a path.
+  ;;
+  *)
+  IFS="${IFS= 	}"; ac_save_ifs="$IFS"; IFS="${IFS}:"
+  for ac_dir in ifelse([$5], , $PATH, [$5]); do
+    test -z "$ac_dir" && ac_dir=.
+    if test -f $ac_dir/$ac_word; then
+      if [$3]; then
+	ac_cv_path_$1="$ac_dir/$ac_word"
+	break
+      fi
+    fi
+  done
+  IFS="$ac_save_ifs"
+dnl If no 4th arg is given, leave the cache variable unset,
+dnl so AC_PATH_PROGS will keep looking.
+ifelse([$4], , , [  test -z "[$]ac_cv_path_$1" && ac_cv_path_$1="$4"
+])dnl
+  ;;
+esac])dnl
+$1="$ac_cv_path_$1"
+if test ifelse([$4], , [-n "[$]$1"], ["[$]$1" != "$4"]); then
+  AC_MSG_RESULT([$]$1)
+else
+  AC_MSG_RESULT(no)
+fi
+AC_SUBST($1)dnl
+])
+
+# GLIB_WITH_NLS
+#-----------------
+glib_DEFUN([GLIB_WITH_NLS],
+  dnl NLS is obligatory
+  [USE_NLS=yes
+    AC_SUBST(USE_NLS)
+
+    gt_cv_have_gettext=no
+
+    CATOBJEXT=NONE
+    XGETTEXT=:
+    INTLLIBS=
+
+    AC_CHECK_HEADER(libintl.h,
+     [gt_cv_func_dgettext_libintl="no"
+      libintl_extra_libs=""
+
+      #
+      # First check in libc
+      #
+      AC_CACHE_CHECK([for ngettext in libc], gt_cv_func_ngettext_libc,
+        [AC_TRY_LINK([
+#include 
+],
+         [return !ngettext ("","", 1)],
+	  gt_cv_func_ngettext_libc=yes,
+          gt_cv_func_ngettext_libc=no)
+        ])
+
+      if test "$gt_cv_func_ngettext_libc" = "yes" ; then
+	      AC_CACHE_CHECK([for dgettext in libc], gt_cv_func_dgettext_libc,
+        	[AC_TRY_LINK([
+#include 
+],
+	          [return !dgettext ("","")],
+		  gt_cv_func_dgettext_libc=yes,
+	          gt_cv_func_dgettext_libc=no)
+        	])
+      fi
+
+      if test "$gt_cv_func_ngettext_libc" = "yes" ; then
+        AC_CHECK_FUNCS(bind_textdomain_codeset)
+      fi
+
+      #
+      # If we don't have everything we want, check in libintl
+      #
+      if test "$gt_cv_func_dgettext_libc" != "yes" \
+	 || test "$gt_cv_func_ngettext_libc" != "yes" \
+         || test "$ac_cv_func_bind_textdomain_codeset" != "yes" ; then
+
+        AC_CHECK_LIB(intl, bindtextdomain,
+	    [AC_CHECK_LIB(intl, ngettext,
+		    [AC_CHECK_LIB(intl, dgettext,
+			          gt_cv_func_dgettext_libintl=yes)])])
+
+	if test "$gt_cv_func_dgettext_libintl" != "yes" ; then
+	  AC_MSG_CHECKING([if -liconv is needed to use gettext])
+	  AC_MSG_RESULT([])
+  	  AC_CHECK_LIB(intl, ngettext,
+          	[AC_CHECK_LIB(intl, dcgettext,
+		       [gt_cv_func_dgettext_libintl=yes
+			libintl_extra_libs=-liconv],
+			:,-liconv)],
+		:,-liconv)
+        fi
+
+        #
+        # If we found libintl, then check in it for bind_textdomain_codeset();
+        # we'll prefer libc if neither have bind_textdomain_codeset(),
+        # and both have dgettext and ngettext
+        #
+        if test "$gt_cv_func_dgettext_libintl" = "yes" ; then
+          glib_save_LIBS="$LIBS"
+          LIBS="$LIBS -lintl $libintl_extra_libs"
+          unset ac_cv_func_bind_textdomain_codeset
+          AC_CHECK_FUNCS(bind_textdomain_codeset)
+          LIBS="$glib_save_LIBS"
+
+          if test "$ac_cv_func_bind_textdomain_codeset" = "yes" ; then
+            gt_cv_func_dgettext_libc=no
+          else
+            if test "$gt_cv_func_dgettext_libc" = "yes" \
+		&& test "$gt_cv_func_ngettext_libc" = "yes"; then
+              gt_cv_func_dgettext_libintl=no
+            fi
+          fi
+        fi
+      fi
+
+      if test "$gt_cv_func_dgettext_libc" = "yes" \
+	|| test "$gt_cv_func_dgettext_libintl" = "yes"; then
+        gt_cv_have_gettext=yes
+      fi
+
+      if test "$gt_cv_func_dgettext_libintl" = "yes"; then
+        INTLLIBS="-lintl $libintl_extra_libs"
+      fi
+
+      if test "$gt_cv_have_gettext" = "yes"; then
+	AC_DEFINE(HAVE_GETTEXT,1,
+	  [Define if the GNU gettext() function is already present or preinstalled.])
+	GLIB_PATH_PROG_WITH_TEST(MSGFMT, msgfmt,
+	  [test -z "`$ac_dir/$ac_word -h 2>&1 | grep 'dv '`"], no)dnl
+	if test "$MSGFMT" != "no"; then
+          glib_save_LIBS="$LIBS"
+          LIBS="$LIBS $INTLLIBS"
+	  AC_CHECK_FUNCS(dcgettext)
+	  MSGFMT_OPTS=
+	  AC_MSG_CHECKING([if msgfmt accepts -c])
+	  GLIB_RUN_PROG([$MSGFMT -c -o /dev/null],[
+msgid ""
+msgstr ""
+"Content-Type: text/plain; charset=UTF-8\n"
+"Project-Id-Version: test 1.0\n"
+"PO-Revision-Date: 2007-02-15 12:01+0100\n"
+"Last-Translator: test \n"
+"Language-Team: C \n"
+"MIME-Version: 1.0\n"
+"Content-Transfer-Encoding: 8bit\n"
+], [MSGFMT_OPTS=-c; AC_MSG_RESULT([yes])], [AC_MSG_RESULT([no])])
+	  AC_SUBST(MSGFMT_OPTS)
+	  AC_PATH_PROG(GMSGFMT, gmsgfmt, $MSGFMT)
+	  GLIB_PATH_PROG_WITH_TEST(XGETTEXT, xgettext,
+	    [test -z "`$ac_dir/$ac_word -h 2>&1 | grep '(HELP)'`"], :)
+	  AC_TRY_LINK(, [extern int _nl_msg_cat_cntr;
+			 return _nl_msg_cat_cntr],
+	    [CATOBJEXT=.gmo
+             DATADIRNAME=share],
+	    [case $host in
+	    *-*-solaris*)
+	    dnl On Solaris, if bind_textdomain_codeset is in libc,
+	    dnl GNU format message catalog is always supported,
+            dnl since both are added to the libc all together.
+	    dnl Hence, we'd like to go with DATADIRNAME=share and
+	    dnl and CATOBJEXT=.gmo in this case.
+            AC_CHECK_FUNC(bind_textdomain_codeset,
+	      [CATOBJEXT=.gmo
+               DATADIRNAME=share],
+	      [CATOBJEXT=.mo
+               DATADIRNAME=lib])
+	    ;;
+	    *)
+	    CATOBJEXT=.mo
+            DATADIRNAME=lib
+	    ;;
+	    esac])
+          LIBS="$glib_save_LIBS"
+	  INSTOBJEXT=.mo
+	else
+	  gt_cv_have_gettext=no
+	fi
+      fi
+    ])
+
+    if test "$gt_cv_have_gettext" = "yes" ; then
+      AC_DEFINE(ENABLE_NLS, 1,
+        [always defined to indicate that i18n is enabled])
+    fi
+
+    dnl Test whether we really found GNU xgettext.
+    if test "$XGETTEXT" != ":"; then
+      dnl If it is not GNU xgettext we define it as : so that the
+      dnl Makefiles still can work.
+      if $XGETTEXT --omit-header /dev/null 2> /dev/null; then
+        : ;
+      else
+        AC_MSG_RESULT(
+	  [found xgettext program is not GNU xgettext; ignore it])
+        XGETTEXT=":"
+      fi
+    fi
+
+    # We need to process the po/ directory.
+    POSUB=po
+
+    AC_OUTPUT_COMMANDS(
+      [case "$CONFIG_FILES" in *po/Makefile.in*)
+        sed -e "/POTFILES =/r po/POTFILES" po/Makefile.in > po/Makefile
+      esac])
+
+    dnl These rules are solely for the distribution goal.  While doing this
+    dnl we only have to keep exactly one list of the available catalogs
+    dnl in configure.in.
+    for lang in $ALL_LINGUAS; do
+      GMOFILES="$GMOFILES $lang.gmo"
+      POFILES="$POFILES $lang.po"
+    done
+
+    dnl Make all variables we use known to autoconf.
+    AC_SUBST(CATALOGS)
+    AC_SUBST(CATOBJEXT)
+    AC_SUBST(DATADIRNAME)
+    AC_SUBST(GMOFILES)
+    AC_SUBST(INSTOBJEXT)
+    AC_SUBST(INTLLIBS)
+    AC_SUBST(PO_IN_DATADIR_TRUE)
+    AC_SUBST(PO_IN_DATADIR_FALSE)
+    AC_SUBST(POFILES)
+    AC_SUBST(POSUB)
+  ])
+
+# AM_GLIB_GNU_GETTEXT
+# -------------------
+# Do checks necessary for use of gettext. If a suitable implementation
+# of gettext is found in either in libintl or in the C library,
+# it will set INTLLIBS to the libraries needed for use of gettext
+# and AC_DEFINE() HAVE_GETTEXT and ENABLE_NLS. (The shell variable
+# gt_cv_have_gettext will be set to "yes".) It will also call AC_SUBST()
+# on various variables needed by the Makefile.in.in installed by
+# glib-gettextize.
+dnl
+glib_DEFUN([GLIB_GNU_GETTEXT],
+  [AC_REQUIRE([AC_PROG_CC])dnl
+   AC_REQUIRE([AC_HEADER_STDC])dnl
+
+   GLIB_LC_MESSAGES
+   GLIB_WITH_NLS
+
+   if test "$gt_cv_have_gettext" = "yes"; then
+     if test "x$ALL_LINGUAS" = "x"; then
+       LINGUAS=
+     else
+       AC_MSG_CHECKING(for catalogs to be installed)
+       NEW_LINGUAS=
+       for presentlang in $ALL_LINGUAS; do
+         useit=no
+         if test "%UNSET%" != "${LINGUAS-%UNSET%}"; then
+           desiredlanguages="$LINGUAS"
+         else
+           desiredlanguages="$ALL_LINGUAS"
+         fi
+         for desiredlang in $desiredlanguages; do
+ 	   # Use the presentlang catalog if desiredlang is
+           #   a. equal to presentlang, or
+           #   b. a variant of presentlang (because in this case,
+           #      presentlang can be used as a fallback for messages
+           #      which are not translated in the desiredlang catalog).
+           case "$desiredlang" in
+             "$presentlang"*) useit=yes;;
+           esac
+         done
+         if test $useit = yes; then
+           NEW_LINGUAS="$NEW_LINGUAS $presentlang"
+         fi
+       done
+       LINGUAS=$NEW_LINGUAS
+       AC_MSG_RESULT($LINGUAS)
+     fi
+
+     dnl Construct list of names of catalog files to be constructed.
+     if test -n "$LINGUAS"; then
+       for lang in $LINGUAS; do CATALOGS="$CATALOGS $lang$CATOBJEXT"; done
+     fi
+   fi
+
+   dnl If the AC_CONFIG_AUX_DIR macro for autoconf is used we possibly
+   dnl find the mkinstalldirs script in another subdir but ($top_srcdir).
+   dnl Try to locate is.
+   MKINSTALLDIRS=
+   if test -n "$ac_aux_dir"; then
+     MKINSTALLDIRS="$ac_aux_dir/mkinstalldirs"
+   fi
+   if test -z "$MKINSTALLDIRS"; then
+     MKINSTALLDIRS="\$(top_srcdir)/mkinstalldirs"
+   fi
+   AC_SUBST(MKINSTALLDIRS)
+
+   dnl Generate list of files to be processed by xgettext which will
+   dnl be included in po/Makefile.
+   test -d po || mkdir po
+   if test "x$srcdir" != "x."; then
+     if test "x`echo $srcdir | sed 's@/.*@@'`" = "x"; then
+       posrcprefix="$srcdir/"
+     else
+       posrcprefix="../$srcdir/"
+     fi
+   else
+     posrcprefix="../"
+   fi
+   rm -f po/POTFILES
+   sed -e "/^#/d" -e "/^\$/d" -e "s,.*,	$posrcprefix& \\\\," -e "\$s/\(.*\) \\\\/\1/" \
+	< $srcdir/po/POTFILES.in > po/POTFILES
+  ])
+
+# AM_GLIB_DEFINE_LOCALEDIR(VARIABLE)
+# -------------------------------
+# Define VARIABLE to the location where catalog files will
+# be installed by po/Makefile.
+glib_DEFUN([GLIB_DEFINE_LOCALEDIR],
+[glib_REQUIRE([GLIB_GNU_GETTEXT])dnl
+glib_save_prefix="$prefix"
+glib_save_exec_prefix="$exec_prefix"
+glib_save_datarootdir="$datarootdir"
+test "x$prefix" = xNONE && prefix=$ac_default_prefix
+test "x$exec_prefix" = xNONE && exec_prefix=$prefix
+datarootdir=`eval echo "${datarootdir}"`
+if test "x$CATOBJEXT" = "x.mo" ; then
+  localedir=`eval echo "${libdir}/locale"`
+else
+  localedir=`eval echo "${datadir}/locale"`
+fi
+prefix="$glib_save_prefix"
+exec_prefix="$glib_save_exec_prefix"
+datarootdir="$glib_save_datarootdir"
+AC_DEFINE_UNQUOTED($1, "$localedir",
+  [Define the location where the catalogs will be installed])
+])
+
+dnl
+dnl Now the definitions that aclocal will find
+dnl
+ifdef(glib_configure_in,[],[
+AC_DEFUN([AM_GLIB_GNU_GETTEXT],[GLIB_GNU_GETTEXT($@)])
+AC_DEFUN([AM_GLIB_DEFINE_LOCALEDIR],[GLIB_DEFINE_LOCALEDIR($@)])
+])dnl
+
+# GLIB_RUN_PROG(PROGRAM, TEST-FILE, [ACTION-IF-PASS], [ACTION-IF-FAIL])
+#
+# Create a temporary file with TEST-FILE as its contents and pass the
+# file name to PROGRAM.  Perform ACTION-IF-PASS if PROGRAM exits with
+# 0 and perform ACTION-IF-FAIL for any other exit status.
+AC_DEFUN([GLIB_RUN_PROG],
+[cat >conftest.foo <<_ACEOF
+$2
+_ACEOF
+if AC_RUN_LOG([$1 conftest.foo]); then
+  m4_ifval([$3], [$3], [:])
+m4_ifvaln([$4], [else $4])dnl
+echo "$as_me: failed input was:" >&AS_MESSAGE_LOG_FD
+sed 's/^/| /' conftest.foo >&AS_MESSAGE_LOG_FD
+fi])
+
diff --git a/common/m4/gst-arch.m4 b/common/m4/gst-arch.m4
new file mode 100644
index 0000000..077a20b
--- /dev/null
+++ b/common/m4/gst-arch.m4
@@ -0,0 +1,140 @@
+dnl AG_GST_ARCH
+dnl sets up defines and automake conditionals for host architecture
+dnl checks endianness
+dnl defines HOST_CPU
+
+AC_DEFUN([AG_GST_ARCH],
+[
+  dnl Determine CPU
+  case "x${target_cpu}" in
+    xi?86 | xk? | xi?86_64)
+      case $target_os in
+         solaris*)
+            AC_CHECK_DECL([__i386],	[I386_ABI="yes"], [I386_ABI="no"])
+            AC_CHECK_DECL([__amd64], [AMD64_ABI="yes"], [AMD64_ABI="no"])
+
+            if test "x$I386_ABI" = "xyes" ; then
+               HAVE_CPU_I386=yes
+               AC_DEFINE(HAVE_CPU_I386, 1, [Define if the target CPU is an x86])
+            fi
+            if test "x$AMD64_ABI" = "xyes" ; then
+                HAVE_CPU_X86_64=yes
+                AC_DEFINE(HAVE_CPU_X86_64, 1, [Define if the target CPU is a x86_64])
+            fi
+            ;;
+         *)
+            HAVE_CPU_I386=yes
+            AC_DEFINE(HAVE_CPU_I386, 1, [Define if the target CPU is an x86])
+
+            dnl FIXME could use some better detection
+            dnl (ie CPUID)
+            case "x${target_cpu}" in
+              xi386 | xi486) ;;
+            *)
+              AC_DEFINE(HAVE_RDTSC, 1, [Define if RDTSC is available]) ;;
+            esac
+            ;;
+      esac
+      ;;
+    xpowerpc)
+      HAVE_CPU_PPC=yes
+      AC_DEFINE(HAVE_CPU_PPC, 1, [Define if the target CPU is a PowerPC]) ;;
+    xpowerpc64)
+      HAVE_CPU_PPC64=yes
+      AC_DEFINE(HAVE_CPU_PPC64, 1, [Define if the target CPU is a 64 bit PowerPC]) ;;
+    xalpha*)
+      HAVE_CPU_ALPHA=yes
+      AC_DEFINE(HAVE_CPU_ALPHA, 1, [Define if the target CPU is an Alpha]) ;;
+    xarm*)
+      HAVE_CPU_ARM=yes
+      AC_DEFINE(HAVE_CPU_ARM, 1, [Define if the target CPU is an ARM]) ;;
+    xsparc*)
+      HAVE_CPU_SPARC=yes
+      AC_DEFINE(HAVE_CPU_SPARC, 1, [Define if the target CPU is a SPARC]) ;;
+    xmips*)
+      HAVE_CPU_MIPS=yes
+      AC_DEFINE(HAVE_CPU_MIPS, 1, [Define if the target CPU is a MIPS]) ;;
+    xhppa*)
+      HAVE_CPU_HPPA=yes
+      AC_DEFINE(HAVE_CPU_HPPA, 1, [Define if the target CPU is a HPPA]) ;;
+    xs390*)
+      HAVE_CPU_S390=yes
+      AC_DEFINE(HAVE_CPU_S390, 1, [Define if the target CPU is a S390]) ;;
+    xia64*)
+      HAVE_CPU_IA64=yes
+      AC_DEFINE(HAVE_CPU_IA64, 1, [Define if the target CPU is a IA64]) ;;
+    xm68k*)
+      HAVE_CPU_M68K=yes
+      AC_DEFINE(HAVE_CPU_M68K, 1, [Define if the target CPU is a M68K]) ;;
+    xx86_64)
+      HAVE_CPU_X86_64=yes
+      AC_DEFINE(HAVE_CPU_X86_64, 1, [Define if the target CPU is a x86_64]) ;;
+    xcris)
+      HAVE_CPU_CRIS=yes
+      AC_DEFINE(HAVE_CPU_CRIS, 1, [Define if the target CPU is a CRIS]) ;;
+    xcrisv32)
+      HAVE_CPU_CRISV32=yes
+      AC_DEFINE(HAVE_CPU_CRISV32, 1, [Define if the target CPU is a CRISv32]) ;;
+  esac
+
+  dnl Determine endianness
+  AC_C_BIGENDIAN
+
+  AM_CONDITIONAL(HAVE_CPU_I386,       test "x$HAVE_CPU_I386" = "xyes")
+  AM_CONDITIONAL(HAVE_CPU_PPC,        test "x$HAVE_CPU_PPC" = "xyes")
+  AM_CONDITIONAL(HAVE_CPU_PPC64,      test "x$HAVE_CPU_PPC64" = "xyes")
+  AM_CONDITIONAL(HAVE_CPU_ALPHA,      test "x$HAVE_CPU_ALPHA" = "xyes")
+  AM_CONDITIONAL(HAVE_CPU_ARM,        test "x$HAVE_CPU_ARM" = "xyes")
+  AM_CONDITIONAL(HAVE_CPU_SPARC,      test "x$HAVE_CPU_SPARC" = "xyes")
+  AM_CONDITIONAL(HAVE_CPU_HPPA,       test "x$HAVE_CPU_HPPA" = "xyes")
+  AM_CONDITIONAL(HAVE_CPU_MIPS,       test "x$HAVE_CPU_MIPS" = "xyes")
+  AM_CONDITIONAL(HAVE_CPU_S390,       test "x$HAVE_CPU_S390" = "xyes")
+  AM_CONDITIONAL(HAVE_CPU_IA64,       test "x$HAVE_CPU_IA64" = "xyes")
+  AM_CONDITIONAL(HAVE_CPU_M68K,       test "x$HAVE_CPU_M68K" = "xyes")
+  AM_CONDITIONAL(HAVE_CPU_X86_64,     test "x$HAVE_CPU_X86_64" = "xyes")
+  AM_CONDITIONAL(HAVE_CPU_CRIS,       test "x$HAVE_CPU_CRIS" = "xyes")
+  AM_CONDITIONAL(HAVE_CPU_CRISV32,    test "x$HAVE_CPU_CRISV32" = "xyes")
+
+  AC_DEFINE_UNQUOTED(HOST_CPU, "$host_cpu", [the host CPU])
+  AC_DEFINE_UNQUOTED(TARGET_CPU, "$target_cpu", [the target CPU])
+])
+
+dnl check if unaligned memory access works correctly
+AC_DEFUN([AG_GST_UNALIGNED_ACCESS], [
+  AC_MSG_CHECKING([if unaligned memory access works correctly])
+  if test x"$as_cv_unaligned_access" = x ; then
+    case $host in
+      alpha*|arm*|hp*|mips*|sh*|sparc*|ia64*)
+        _AS_ECHO_N([(blacklisted) ])
+        as_cv_unaligned_access=no
+	;;
+      i?86*|x86_64*|amd64*|powerpc*|m68k*|cris*)
+        _AS_ECHO_N([(whitelisted) ])
+        as_cv_unaligned_access=yes
+	;;
+    esac
+  else
+    _AS_ECHO_N([(cached) ])
+  fi
+  if test x"$as_cv_unaligned_access" = x ; then
+    AC_TRY_RUN([
+int main(int argc, char **argv)
+{
+  char array[] = "ABCDEFGH";
+  unsigned int iarray[2];
+  memcpy(iarray,array,8);
+#define GET(x) (*(unsigned int *)((char *)iarray + (x)))
+  if(GET(0) != 0x41424344 && GET(0) != 0x44434241) return 1;
+  if(GET(1) != 0x42434445 && GET(1) != 0x45444342) return 1;
+  if(GET(2) != 0x43444546 && GET(2) != 0x46454443) return 1;
+  if(GET(3) != 0x44454647 && GET(3) != 0x47464544) return 1;
+  return 0;
+}
+    ], as_cv_unaligned_access="yes", as_cv_unaligned_access="no")
+  fi
+  AC_MSG_RESULT($as_cv_unaligned_access)
+  if test "$as_cv_unaligned_access" = "yes"; then
+    AC_DEFINE_UNQUOTED(HAVE_UNALIGNED_ACCESS, 1,
+      [defined if unaligned memory access works correctly])
+  fi
+])
diff --git a/common/m4/gst-args.m4 b/common/m4/gst-args.m4
new file mode 100644
index 0000000..b478c82
--- /dev/null
+++ b/common/m4/gst-args.m4
@@ -0,0 +1,360 @@
+dnl configure-time options shared among gstreamer modules
+
+dnl AG_GST_ARG_DEBUG
+dnl AG_GST_ARG_PROFILING
+dnl AG_GST_ARG_VALGRIND
+dnl AG_GST_ARG_GCOV
+
+dnl AG_GST_ARG_EXAMPLES
+
+dnl AG_GST_ARG_WITH_PKG_CONFIG_PATH
+dnl AG_GST_ARG_WITH_PACKAGE_NAME
+dnl AG_GST_ARG_WITH_PACKAGE_ORIGIN
+
+dnl AG_GST_ARG_WITH_PLUGINS
+dnl AG_GST_CHECK_PLUGIN
+dnl AG_GST_DISABLE_PLUGIN
+
+dnl AG_GST_ARG_ENABLE_EXTERNAL
+dnl AG_GST_ARG_ENABLE_EXPERIMENTAL
+dnl AG_GST_ARG_ENABLE_BROKEN
+
+dnl AG_GST_ARG_DISABLE_FATAL_WARNINGS
+AC_DEFUN([AG_GST_ARG_DEBUG],
+[
+  dnl debugging stuff
+  AC_ARG_ENABLE(debug,
+    AC_HELP_STRING([--disable-debug],[disable addition of -g debugging info]),
+    [
+      case "${enableval}" in
+        yes) USE_DEBUG=yes ;;
+        no)  USE_DEBUG=no ;;
+        *)   AC_MSG_ERROR(bad value ${enableval} for --enable-debug) ;;
+      esac
+    ],
+    [USE_DEBUG=yes]) dnl Default value
+])
+
+AC_DEFUN([AG_GST_ARG_PROFILING],
+[
+  AC_ARG_ENABLE(profiling,
+    AC_HELP_STRING([--enable-profiling],
+      [adds -pg to compiler commandline, for profiling]),
+    [
+      case "${enableval}" in
+        yes) USE_PROFILING=yes ;;
+        no)  USE_PROFILING=no ;;
+        *)   AC_MSG_ERROR(bad value ${enableval} for --enable-profiling) ;;
+      esac
+    ],
+    [USE_PROFILING=no]) dnl Default value
+])
+
+AC_DEFUN([AG_GST_ARG_VALGRIND],
+[
+  dnl valgrind inclusion
+  AC_ARG_ENABLE(valgrind,
+    AC_HELP_STRING([--disable-valgrind],[disable run-time valgrind detection]),
+    [
+      case "${enableval}" in
+        yes) USE_VALGRIND="$USE_DEBUG" ;;
+        no)  USE_VALGRIND=no ;;
+        *)   AC_MSG_ERROR(bad value ${enableval} for --enable-valgrind) ;;
+      esac
+    ],
+    [USE_VALGRIND="$USE_DEBUG"]) dnl Default value
+  VALGRIND_REQ="3.0"
+  if test "x$USE_VALGRIND" = xyes; then
+    PKG_CHECK_MODULES(VALGRIND, valgrind >= $VALGRIND_REQ,
+      USE_VALGRIND="yes",
+      USE_VALGRIND="no")
+  fi
+  if test "x$USE_VALGRIND" = xyes; then
+    AC_DEFINE(HAVE_VALGRIND, 1, [Define if valgrind should be used])
+    AC_MSG_NOTICE(Using extra code paths for valgrind)
+  fi
+])
+
+AC_DEFUN([AG_GST_ARG_GCOV],
+[
+  AC_ARG_ENABLE(gcov,
+    AC_HELP_STRING([--enable-gcov],
+      [compile with coverage profiling instrumentation (gcc only)]),
+    enable_gcov=$enableval,
+    enable_gcov=no)
+  if test x$enable_gcov = xyes ; then
+    if test "x$GCC" != "xyes"
+    then
+      AC_MSG_ERROR([gcov only works if gcc is used])
+    fi
+
+    AS_COMPILER_FLAG(["-fprofile-arcs"],
+      [GCOV_CFLAGS="$GCOV_CFLAGS -fprofile-arcs"],
+      true)
+    AS_COMPILER_FLAG(["-ftest-coverage"],
+      [GCOV_CFLAGS="$GCOV_CFLAGS -ftest-coverage"],
+      true)
+    dnl remove any -O flags - FIXME: is this needed ?
+    GCOV_CFLAGS=`echo "$GCOV_CFLAGS" | sed -e 's/-O[[0-9]]*//g'`
+    dnl libtool 1.5.22 and lower strip -fprofile-arcs from the flags
+    dnl passed to the linker, which is a bug; -fprofile-arcs implicitly
+    dnl links in -lgcov, so we do it explicitly here for the same effect
+    GCOV_LIBS=-lgcov
+    AC_SUBST(GCOV_CFLAGS)
+    AC_SUBST(GCOV_LIBS)
+    GCOV=`echo $CC | sed s/gcc/gcov/g`
+    AC_SUBST(GCOV)
+
+    GST_GCOV_ENABLED=yes
+    AC_DEFINE_UNQUOTED(GST_GCOV_ENABLED, 1,
+      [Defined if gcov is enabled to force a rebuild due to config.h changing])
+    dnl if gcov is used, we do not want default -O2 CFLAGS
+    if test "x$GST_GCOV_ENABLED" = "xyes"
+    then
+      CFLAGS="$CFLAGS -O0"
+      AC_SUBST(CFLAGS)
+      CXXFLAGS="$CXXFLAGS -O0"
+      AC_SUBST(CXXFLAGS)
+      FFLAGS="$FFLAGS -O0"
+      AC_SUBST(FFLAGS)
+      CCASFLAGS="$CCASFLAGS -O0"
+      AC_SUBST(CCASFLAGS)
+      AC_MSG_NOTICE([gcov enabled, setting CFLAGS and friends to $CFLAGS])
+    fi
+  fi
+  AM_CONDITIONAL(GST_GCOV_ENABLED, test x$enable_gcov = xyes)
+])
+
+AC_DEFUN([AG_GST_ARG_EXAMPLES],
+[
+  AC_ARG_ENABLE(examples,
+    AC_HELP_STRING([--disable-examples], [disable building examples]),
+      [
+        case "${enableval}" in
+          yes) BUILD_EXAMPLES=yes ;;
+          no)  BUILD_EXAMPLES=no ;;
+          *)   AC_MSG_ERROR(bad value ${enableval} for --disable-examples) ;;
+        esac
+      ],
+      [BUILD_EXAMPLES=yes]) dnl Default value
+  AM_CONDITIONAL(BUILD_EXAMPLES,      test "x$BUILD_EXAMPLES" = "xyes")
+])
+
+AC_DEFUN([AG_GST_ARG_WITH_PKG_CONFIG_PATH],
+[
+  dnl possibly modify pkg-config path
+  AC_ARG_WITH(pkg-config-path,
+     AC_HELP_STRING([--with-pkg-config-path],
+                    [colon-separated list of pkg-config(1) dirs]),
+     [
+       export PKG_CONFIG_PATH=${withval}
+       AC_MSG_NOTICE(Set PKG_CONFIG_PATH to $PKG_CONFIG_PATH)
+     ])
+])
+
+
+dnl This macro requires that GST_GIT or GST_CVS is set to yes or no (release)
+AC_DEFUN([AG_GST_ARG_WITH_PACKAGE_NAME],
+[
+  dnl package name in plugins
+  AC_ARG_WITH(package-name,
+    AC_HELP_STRING([--with-package-name],
+      [specify package name to use in plugins]),
+    [
+      case "${withval}" in
+        yes) AC_MSG_ERROR(bad value ${withval} for --with-package-name) ;;
+        no)  AC_MSG_ERROR(bad value ${withval} for --with-package-name) ;;
+        *)   GST_PACKAGE_NAME="${withval}" ;;
+      esac
+    ],
+    [
+      P=$1
+      if test "x$P" = "x"
+      then
+        P=$PACKAGE_NAME
+      fi
+
+      if test "x$PACKAGE_VERSION_NANO" = "x0"
+      then
+        GST_PACKAGE_NAME="$P source release"
+      else
+        if test "x$PACKAGE_VERSION_NANO" = "x1"
+        then
+          GST_PACKAGE_NAME="$P git"
+        else
+          GST_PACKAGE_NAME="$P prerelease"
+        fi
+      fi
+    ]
+  )
+  AC_MSG_NOTICE(Using $GST_PACKAGE_NAME as package name)
+  AC_DEFINE_UNQUOTED(GST_PACKAGE_NAME, "$GST_PACKAGE_NAME",
+      [package name in plugins])
+  AC_SUBST(GST_PACKAGE_NAME)
+])
+
+AC_DEFUN([AG_GST_ARG_WITH_PACKAGE_ORIGIN],
+[
+  dnl package origin URL
+  AC_ARG_WITH(package-origin,
+    AC_HELP_STRING([--with-package-origin],
+      [specify package origin URL to use in plugins]),
+    [
+      case "${withval}" in
+        yes) AC_MSG_ERROR(bad value ${withval} for --with-package-origin) ;;
+        no)  AC_MSG_ERROR(bad value ${withval} for --with-package-origin) ;;
+        *)   GST_PACKAGE_ORIGIN="${withval}" ;;
+      esac
+    ],
+    [GST_PACKAGE_ORIGIN="[Unknown package origin]"] dnl Default value
+  )
+  AC_MSG_NOTICE(Using $GST_PACKAGE_ORIGIN as package origin)
+  AC_DEFINE_UNQUOTED(GST_PACKAGE_ORIGIN, "$GST_PACKAGE_ORIGIN",
+      [package origin])
+  AC_SUBST(GST_PACKAGE_ORIGIN)
+])
+
+dnl sets WITH_PLUGINS to the list of plug-ins given as an argument
+dnl also clears GST_PLUGINS_ALL and GST_PLUGINS_SELECTED
+AC_DEFUN([AG_GST_ARG_WITH_PLUGINS],
+[
+  AC_ARG_WITH(plugins,
+    AC_HELP_STRING([--with-plugins],
+      [comma-separated list of dependencyless plug-ins to compile]),
+    [WITH_PLUGINS=$withval],
+    [WITH_PLUGINS=])
+
+  GST_PLUGINS_ALL=""
+  GST_PLUGINS_SELECTED=""
+  GST_PLUGINS_NONPORTED=""
+
+  AC_SUBST(GST_PLUGINS_ALL)
+  AC_SUBST(GST_PLUGINS_SELECTED)
+  AC_SUBST(GST_PLUGINS_NONPORTED)
+])
+
+dnl AG_GST_CHECK_PLUGIN(PLUGIN-NAME)
+dnl
+dnl This macro adds the plug-in  to GST_PLUGINS_ALL. Then it
+dnl checks if WITH_PLUGINS is empty or the plugin is present in WITH_PLUGINS,
+dnl and if so adds it to GST_PLUGINS_SELECTED. Then it checks if the plugin
+dnl is present in WITHOUT_PLUGINS (ie. was disabled specifically) and if so
+dnl removes it from GST_PLUGINS_SELECTED.
+dnl
+dnl The macro will call AM_CONDITIONAL(USE_PLUGIN_, ...) to allow
+dnl control of what is built in Makefile.ams.
+AC_DEFUN([AG_GST_CHECK_PLUGIN],
+[
+  GST_PLUGINS_ALL="$GST_PLUGINS_ALL [$1]"
+
+  define([pname_def],translit([$1], -a-z, _a-z))
+
+  AC_ARG_ENABLE([$1],
+    AC_HELP_STRING([--disable-[$1]], [disable dependency-less $1 plugin]),
+    [
+      case "${enableval}" in
+        yes) [gst_use_]pname_def=yes ;;
+        no) [gst_use_]pname_def=no ;;
+        *) AC_MSG_ERROR([bad value ${enableval} for --enable-$1]) ;;
+       esac
+    ],
+    [[gst_use_]pname_def=yes]) dnl Default value
+
+  if test x$[gst_use_]pname_def = xno; then
+    AC_MSG_NOTICE(disabling dependency-less plugin $1)
+    WITHOUT_PLUGINS="$WITHOUT_PLUGINS [$1]"
+  fi
+  undefine([pname_def])
+
+  dnl First check inclusion
+  if [[ -z "$WITH_PLUGINS" ]] || echo " [$WITH_PLUGINS] " | tr , ' ' | grep -i " [$1] " > /dev/null; then
+    GST_PLUGINS_SELECTED="$GST_PLUGINS_SELECTED [$1]"
+  fi
+  dnl Then check exclusion
+  if echo " [$WITHOUT_PLUGINS] " | tr , ' ' | grep -i " [$1] " > /dev/null; then
+    GST_PLUGINS_SELECTED=`echo " $GST_PLUGINS_SELECTED " | $SED -e 's/ [$1] / /'`
+  fi
+  dnl Finally check if the plugin is ported or not
+  if echo " [$GST_PLUGINS_NONPORTED] " | tr , ' ' | grep -i " [$1] " > /dev/null; then
+    GST_PLUGINS_SELECTED=`echo " $GST_PLUGINS_SELECTED " | $SED -e 's/ [$1] / /'`
+  fi
+  AM_CONDITIONAL([USE_PLUGIN_]translit([$1], a-z, A-Z), echo " $GST_PLUGINS_SELECTED " | grep -i " [$1] " > /dev/null)
+])
+
+dnl AG_GST_DISABLE_PLUGIN(PLUGIN-NAME)
+dnl
+dnl This macro disables the plug-in  by removing it from
+dnl GST_PLUGINS_SELECTED.
+AC_DEFUN([AG_GST_DISABLE_PLUGIN],
+[
+  GST_PLUGINS_SELECTED=`echo " $GST_PLUGINS_SELECTED " | $SED -e 's/ [$1] / /'`
+  AM_CONDITIONAL([USE_PLUGIN_]translit([$1], a-z, A-Z), false)
+])
+
+AC_DEFUN([AG_GST_ARG_ENABLE_EXTERNAL],
+[
+  AG_GST_CHECK_FEATURE(EXTERNAL, [building of plug-ins with external deps],,
+    HAVE_EXTERNAL=yes, enabled,
+    [
+      AC_MSG_NOTICE(building external plug-ins)
+      BUILD_EXTERNAL="yes"
+    ],[
+      AC_MSG_WARN(all plug-ins with external dependencies will not be built)
+      BUILD_EXTERNAL="no"
+    ])
+  # make BUILD_EXTERNAL available to Makefile.am
+  AM_CONDITIONAL(BUILD_EXTERNAL, test "x$BUILD_EXTERNAL" = "xyes")
+])
+
+dnl experimental plug-ins; stuff that hasn't had the dust settle yet
+dnl read 'builds, but might not work'
+AC_DEFUN([AG_GST_ARG_ENABLE_EXPERIMENTAL],
+[
+  AG_GST_CHECK_FEATURE(EXPERIMENTAL, [building of experimental plug-ins],,
+    HAVE_EXPERIMENTAL=yes, disabled,
+    [
+      AC_MSG_WARN(building experimental plug-ins)
+      BUILD_EXPERIMENTAL="yes"
+    ],[
+      AC_MSG_NOTICE(not building experimental plug-ins)
+      BUILD_EXPERIMENTAL="no"
+    ])
+  # make BUILD_EXPERIMENTAL available to Makefile.am
+  AM_CONDITIONAL(BUILD_EXPERIMENTAL, test "x$BUILD_EXPERIMENTAL" = "xyes")
+])
+
+dnl broken plug-ins; stuff that doesn't seem to build at the moment
+AC_DEFUN([AG_GST_ARG_ENABLE_BROKEN],
+[
+  AG_GST_CHECK_FEATURE(BROKEN, [building of broken plug-ins],,
+    HAVE_BROKEN=yes, disabled,
+    [
+      AC_MSG_WARN([building broken plug-ins -- no bug reports on these, only patches ...])
+    ],[
+      AC_MSG_NOTICE([not building broken plug-ins])
+    ])
+])
+
+dnl allow people (or build tools) to override default behaviour
+dnl for fatal compiler warnings
+dnl Enable fatal warnings by default only for development versions
+AC_DEFUN([AG_GST_ARG_DISABLE_FATAL_WARNINGS],
+[
+  AC_ARG_ENABLE(fatal-warnings,
+    AC_HELP_STRING([--disable-fatal-warnings],
+                   [Don't turn compiler warnings into fatal errors]),
+    [
+      case "${enableval}" in
+        yes) FATAL_WARNINGS=yes ;;
+        no)  FATAL_WARNINGS=no ;;
+        *)   AC_MSG_ERROR(bad value ${enableval} for --disable-fatal-warnings) ;;
+      esac
+    ],
+    [
+      if test "x`expr $PACKAGE_VERSION_MINOR % 2`" = "x1" -a "x`expr $PACKAGE_VERSION_MICRO '<' 90`" = "x1"; then
+        FATAL_WARNINGS=yes
+      else
+        FATAL_WARNINGS=no
+      fi
+    ])
+])
diff --git a/common/m4/gst-check.m4 b/common/m4/gst-check.m4
new file mode 100644
index 0000000..4277053
--- /dev/null
+++ b/common/m4/gst-check.m4
@@ -0,0 +1,294 @@
+dnl pkg-config-based checks for GStreamer modules and dependency modules
+
+dnl generic:
+dnl AG_GST_PKG_CHECK_MODULES([PREFIX], [WHICH], [REQUIRED])
+dnl sets HAVE_[$PREFIX], [$PREFIX]_*
+dnl AG_GST_CHECK_MODULES([PREFIX], [MODULE], [MINVER], [NAME], [REQUIRED])
+dnl sets HAVE_[$PREFIX], [$PREFIX]_*
+
+dnl specific:
+dnl AG_GST_CHECK_GST([MAJMIN], [MINVER], [REQUIRED])
+dnl   also sets/ACSUBSTs GST_TOOLS_DIR and GST_PLUGINS_DIR
+dnl AG_GST_CHECK_GST_BASE([MAJMIN], [MINVER], [REQUIRED])
+dnl AG_GST_CHECK_GST_CONTROLLER([MAJMIN], [MINVER], [REQUIRED])
+dnl AG_GST_CHECK_GST_NET([MAJMIN], [MINVER], [REQUIRED])
+dnl AG_GST_CHECK_GST_CHECK([MAJMIN], [MINVER], [REQUIRED])
+dnl AG_GST_CHECK_GST_PLUGINS_BASE([MAJMIN], [MINVER], [REQUIRED])
+dnl   also sets/ACSUBSTs GSTPB_PLUGINS_DIR
+
+AC_DEFUN([AG_GST_PKG_CHECK_MODULES],
+[
+  which="[$2]"
+  dnl not required by default, since we use this mostly for plugin deps
+  required=ifelse([$3], , "no", [$3])
+
+  PKG_CHECK_MODULES([$1], $which,
+    [
+      HAVE_[$1]="yes"
+    ],
+    [
+      HAVE_[$1]="no"
+      if test "x$required" = "xyes"; then
+        AC_MSG_ERROR($[$1]_PKG_ERRORS)
+      else
+        AC_MSG_NOTICE($[$1]_PKG_ERRORS)
+      fi
+    ])
+
+  dnl AC_SUBST of CFLAGS and LIBS was not done before automake 1.7
+  dnl It gets done automatically in automake >= 1.7, which we now require
+]))
+
+AC_DEFUN([AG_GST_CHECK_MODULES],
+[
+  module=[$2]
+  minver=[$3]
+  name="[$4]"
+  required=ifelse([$5], , "yes", [$5]) dnl required by default
+
+  PKG_CHECK_MODULES([$1], $module >= $minver,
+    [
+      HAVE_[$1]="yes"
+    ],
+    [
+      HAVE_[$1]="no"
+      AC_MSG_NOTICE($[$1]_PKG_ERRORS)
+      if test "x$required" = "xyes"; then
+        AC_MSG_ERROR([no $module >= $minver ($name) found])
+      else
+        AC_MSG_NOTICE([no $module >= $minver ($name) found])
+      fi
+    ])
+
+  dnl AC_SUBST of CFLAGS and LIBS was not done before automake 1.7
+  dnl It gets done automatically in automake >= 1.7, which we now require
+]))
+
+AC_DEFUN([AG_GST_CHECK_GST],
+[
+  AG_GST_CHECK_MODULES(GST, gstreamer-[$1], [$2], [GStreamer], [$3])
+  dnl allow setting before calling this macro to override
+  if test -z $GST_TOOLS_DIR; then
+    GST_TOOLS_DIR=`$PKG_CONFIG --variable=toolsdir gstreamer-[$1]`
+    if test -z $GST_TOOLS_DIR; then
+      AC_MSG_ERROR(
+        [no tools dir set in GStreamer pkg-config file, core upgrade needed.])
+    fi
+  fi
+  AC_MSG_NOTICE([using GStreamer tools in $GST_TOOLS_DIR])
+  AC_SUBST(GST_TOOLS_DIR)
+
+  dnl check for where core plug-ins got installed
+  dnl this is used for unit tests
+  dnl allow setting before calling this macro to override
+  if test -z $GST_PLUGINS_DIR; then
+    GST_PLUGINS_DIR=`$PKG_CONFIG --variable=pluginsdir gstreamer-[$1]`
+    if test -z $GST_PLUGINS_DIR; then
+      AC_MSG_ERROR(
+        [no pluginsdir set in GStreamer pkg-config file, core upgrade needed.])
+    fi
+  fi
+  AC_MSG_NOTICE([using GStreamer plug-ins in $GST_PLUGINS_DIR])
+  AC_SUBST(GST_PLUGINS_DIR)
+])
+
+AC_DEFUN([AG_GST_CHECK_GST_BASE],
+[
+  AG_GST_CHECK_MODULES(GST_BASE, gstreamer-base-[$1], [$2],
+    [GStreamer Base Libraries], [$3])
+])
+
+AC_DEFUN([AG_GST_CHECK_GST_CONTROLLER],
+[
+  AG_GST_CHECK_MODULES(GST_CONTROLLER, gstreamer-controller-[$1], [$2],
+    [GStreamer Controller Library], [$3])
+])
+
+AC_DEFUN([AG_GST_CHECK_GST_NET],
+[
+  AG_GST_CHECK_MODULES(GST_NET, gstreamer-net-[$1], [$2],
+    [GStreamer Network Library], [$3])
+])
+
+AC_DEFUN([AG_GST_CHECK_GST_CHECK],
+[
+  AG_GST_CHECK_MODULES(GST_CHECK, gstreamer-check-[$1], [$2],
+    [GStreamer Check unittest Library], [$3])
+])
+
+dnl ===========================================================================
+dnl AG_GST_CHECK_UNINSTALLED_SETUP([ACTION-IF-UNINSTALLED], [ACTION-IF-NOT])
+dnl
+dnl ACTION-IF-UNINSTALLED  (optional) extra actions to perform if the setup
+dnl                        is an uninstalled setup
+dnl ACTION-IF-NOT          (optional) extra actions to perform if the setup
+dnl                        is not an uninstalled setup
+dnl ===========================================================================
+AC_DEFUN([AG_GST_CHECK_UNINSTALLED_SETUP],
+[
+  AC_MSG_CHECKING([whether this is an uninstalled GStreamer setup])
+  AC_CACHE_VAL(gst_cv_is_uninstalled_setup,[
+    gst_cv_is_uninstalled_setup=no
+    if (set -u; : $GST_PLUGIN_SYSTEM_PATH) 2>/dev/null ; then
+      if test -z "$GST_PLUGIN_SYSTEM_PATH" \
+           -a -n "$GST_PLUGIN_SCANNER"     \
+           -a -n "$GST_PLUGIN_PATH"        \
+           -a -n "$GST_REGISTRY"           \
+           -a -n "$DYLD_LIBRARY_PATH"      \
+           -a -n "$LD_LIBRARY_PATH"; then
+        gst_cv_is_uninstalled_setup=yes;
+      fi
+    fi
+  ])
+  AC_MSG_RESULT($gst_cv_is_uninstalled_setup)
+  if test "x$gst_cv_is_uninstalled_setup" = "xyes"; then
+    ifelse([$1], , :, [$1])
+  else
+    ifelse([$2], , :, [$2])
+  fi
+])
+
+dnl ===========================================================================
+dnl AG_GST_CHECK_GST_PLUGINS_BASE([GST-API_VERSION], [MIN-VERSION], [REQUIRED])
+dnl
+dnl Sets GST_PLUGINS_BASE_CFLAGS and GST_PLUGINS_BASE_LIBS.
+dnl
+dnl Also sets GSTPB_PLUGINS_DIR (and for consistency also GST_PLUGINS_BASE_DIR)
+dnl for use in Makefile.am. This is only really needed/useful in uninstalled
+dnl setups, since in an installed setup all plugins will be found in
+dnl GST_PLUGINS_DIR anyway.
+dnl ===========================================================================
+AC_DEFUN([AG_GST_CHECK_GST_PLUGINS_BASE],
+[
+  AG_GST_CHECK_MODULES(GST_PLUGINS_BASE, gstreamer-plugins-base-[$1], [$2],
+    [GStreamer Base Plugins], [$3])
+
+  if test "x$HAVE_GST_PLUGINS_BASE" = "xyes"; then
+    dnl check for where base plugins got installed
+    dnl this is used for unit tests
+    dnl allow setting before calling this macro to override
+    if test -z $GSTPB_PLUGINS_DIR; then
+      GSTPB_PLUGINS_DIR=`$PKG_CONFIG --variable=pluginsdir gstreamer-plugins-base-[$1]`
+      if test -z $GSTPB_PLUGINS_DIR; then
+        AC_MSG_ERROR(
+          [no pluginsdir set in GStreamer Base Plugins pkg-config file])
+      fi
+    fi
+    AC_MSG_NOTICE([using GStreamer Base Plugins in $GSTPB_PLUGINS_DIR])
+    GST_PLUGINS_BASE_DIR="$GSTPB_PLUGINS_DIR/gst:$GSTPB_PLUGINS_DIR/sys:$GSTPB_PLUGINS_DIR/ext"
+    AC_SUBST(GST_PLUGINS_BASE_DIR)
+    AC_SUBST(GSTPB_PLUGINS_DIR)
+  fi
+])
+
+dnl ===========================================================================
+dnl AG_GST_CHECK_GST_PLUGINS_GOOD([GST-API_VERSION], [MIN-VERSION])
+dnl
+dnl Will set GST_PLUGINS_GOOD_DIR for use in Makefile.am. Note that this will
+dnl only be set in an uninstalled setup, since -good ships no .pc file and in
+dnl an installed setup all plugins will be found in GST_PLUGINS_DIR anyway.
+dnl ===========================================================================
+AC_DEFUN([AG_GST_CHECK_GST_PLUGINS_GOOD],
+[
+  AG_GST_CHECK_MODULES(GST_PLUGINS_GOOD, gstreamer-plugins-good-[$1], [$2],
+    [GStreamer Good Plugins], [no])
+
+  if test "x$HAVE_GST_PLUGINS_GOOD" = "xyes"; then
+    dnl check for where good plugins got installed
+    dnl this is used for unit tests
+    dnl allow setting before calling this macro to override
+    if test -z $GST_PLUGINS_GOOD_DIR; then
+      GST_PLUGINS_GOOD_DIR=`$PKG_CONFIG --variable=pluginsdir gstreamer-plugins-good-[$1]`
+      if test -z $GST_PLUGINS_GOOD_DIR; then
+        AC_MSG_ERROR([no pluginsdir set in GStreamer Good Plugins pkg-config file])
+      fi
+    fi
+    AC_MSG_NOTICE([using GStreamer Good Plugins in $GST_PLUGINS_GOOD_DIR])
+    GST_PLUGINS_GOOD_DIR="$GST_PLUGINS_GOOD_DIR/gst:$GST_PLUGINS_GOOD_DIR/sys:$GST_PLUGINS_GOOD_DIR/ext"
+    AC_SUBST(GST_PLUGINS_GOOD_DIR)
+  fi
+])
+
+dnl ===========================================================================
+dnl AG_GST_CHECK_GST_PLUGINS_UGLY([GST-API_VERSION], [MIN-VERSION])
+dnl
+dnl Will set GST_PLUGINS_UGLY_DIR for use in Makefile.am. Note that this will
+dnl only be set in an uninstalled setup, since -bad ships no .pc file and in
+dnl an installed setup all plugins will be found in GST_PLUGINS_DIR anyway.
+dnl ===========================================================================
+AC_DEFUN([AG_GST_CHECK_GST_PLUGINS_UGLY],
+[
+  AG_GST_CHECK_MODULES(GST_PLUGINS_UGLY, gstreamer-plugins-ugly-[$1], [$2],
+    [GStreamer Ugly Plugins], [no])
+
+  if test "x$HAVE_GST_PLUGINS_UGLY" = "xyes"; then
+    dnl check for where ugly plugins got installed
+    dnl this is used for unit tests
+    dnl allow setting before calling this macro to override
+    if test -z $GST_PLUGINS_UGLY_DIR; then
+      GST_PLUGINS_UGLY_DIR=`$PKG_CONFIG --variable=pluginsdir gstreamer-plugins-ugly-[$1]`
+      if test -z $GST_PLUGINS_UGLY_DIR; then
+        AC_MSG_ERROR([no pluginsdir set in GStreamer Ugly Plugins pkg-config file])
+      fi
+    fi
+    AC_MSG_NOTICE([using GStreamer Ugly Plugins in $GST_PLUGINS_UGLY_DIR])
+    GST_PLUGINS_UGLY_DIR="$GST_PLUGINS_UGLY_DIR/gst:$GST_PLUGINS_UGLY_DIR/sys:$GST_PLUGINS_UGLY_DIR/ext"
+    AC_SUBST(GST_PLUGINS_UGLY_DIR)
+  fi
+])
+
+dnl ===========================================================================
+dnl AG_GST_CHECK_GST_PLUGINS_BAD([GST-API_VERSION], [MIN-VERSION])
+dnl
+dnl Will set GST_PLUGINS_BAD_DIR for use in Makefile.am. Note that this will
+dnl only be set in an uninstalled setup, since -ugly ships no .pc file and in
+dnl an installed setup all plugins will be found in GST_PLUGINS_DIR anyway.
+dnl ===========================================================================
+AC_DEFUN([AG_GST_CHECK_GST_PLUGINS_BAD],
+[
+  AG_GST_CHECK_MODULES(GST_PLUGINS_BAD, gstreamer-plugins-bad-[$1], [$2],
+    [GStreamer Bad Plugins], [no])
+
+  if test "x$HAVE_GST_PLUGINS_BAD" = "xyes"; then
+    dnl check for where bad plugins got installed
+    dnl this is used for unit tests
+    dnl allow setting before calling this macro to override
+    if test -z $GST_PLUGINS_BAD_DIR; then
+      GST_PLUGINS_BAD_DIR=`$PKG_CONFIG --variable=pluginsdir gstreamer-plugins-bad-[$1]`
+      if test -z $GST_PLUGINS_BAD_DIR; then
+        AC_MSG_ERROR([no pluginsdir set in GStreamer Bad Plugins pkg-config file])
+      fi
+    fi
+    AC_MSG_NOTICE([using GStreamer Bad Plugins in $GST_PLUGINS_BAD_DIR])
+    GST_PLUGINS_BAD_DIR="$GST_PLUGINS_BAD_DIR/gst:$GST_PLUGINS_BAD_DIR/sys:$GST_PLUGINS_BAD_DIR/ext"
+    AC_SUBST(GST_PLUGINS_BAD_DIR)
+  fi
+])
+
+dnl ===========================================================================
+dnl AG_GST_CHECK_GST_PLUGINS_LIBAV([GST-API_VERSION], [MIN-VERSION])
+dnl
+dnl Will set GST_PLUGINS_LIBAV_DIR for use in Makefile.am. Note that this will
+dnl only be set in an uninstalled setup, since -libav ships no .pc file and in
+dnl an installed setup all plugins will be found in GST_PLUGINS_DIR anyway.
+dnl ===========================================================================
+AC_DEFUN([AG_GST_CHECK_GST_PLUGINS_LIBAV],
+[
+  AG_GST_CHECK_MODULES(GST_PLUGINS_LIBAV, gstreamer-plugins-libav-[$1], [$2],
+    [GStreamer Libav Plugins], [no])
+
+  if test "x$HAVE_GST_PLUGINS_LIBAV" = "xyes"; then
+    dnl check for where libav plugins got installed
+    dnl this is used for unit tests
+    dnl allow setting before calling this macro to override
+    if test -z $GST_PLUGINS_LIBAV_DIR; then
+      GST_PLUGINS_LIBAV_DIR=`$PKG_CONFIG --variable=pluginsdir gstreamer-plugins-libav-[$1]`
+      if test -z $GST_PLUGINS_LIBAV_DIR; then
+        AC_MSG_ERROR([no pluginsdir set in GStreamer Libav Plugins pkg-config file])
+      fi
+    fi
+    GST_PLUGINS_LIBAV_DIR="$GST_PLUGINS_LIBAV_DIR/ext/libav"
+    AC_MSG_NOTICE([using GStreamer Libav Plugins in $GST_PLUGINS_LIBAV_DIR])
+    AC_SUBST(GST_PLUGINS_LIBAV_DIR)
+  fi
+])
diff --git a/common/m4/gst-debuginfo.m4 b/common/m4/gst-debuginfo.m4
new file mode 100644
index 0000000..b48854d
--- /dev/null
+++ b/common/m4/gst-debuginfo.m4
@@ -0,0 +1,46 @@
+AC_DEFUN([AG_GST_DEBUGINFO], [
+AC_ARG_ENABLE(debug,
+AC_HELP_STRING([--disable-debug],[disable addition of -g debugging info]),
+[case "${enableval}" in
+  yes) USE_DEBUG=yes ;;
+  no)  USE_DEBUG=no ;;
+  *) AC_MSG_ERROR(bad value ${enableval} for --enable-debug) ;;
+esac],
+[USE_DEBUG=yes]) dnl Default value
+
+AC_ARG_ENABLE(DEBUG,
+AC_HELP_STRING([--disable-DEBUG],[disables compilation of debugging messages]),
+[case "${enableval}" in
+  yes) ENABLE_DEBUG=yes ;;
+  no)  ENABLE_DEBUG=no ;;
+  *) AC_MSG_ERROR(bad value ${enableval} for --enable-DEBUG) ;;
+esac],
+[ENABLE_DEBUG=yes]) dnl Default value
+if test x$ENABLE_DEBUG = xyes; then
+  AC_DEFINE(GST_DEBUG_ENABLED, 1, [Define if DEBUG statements should be compiled in])
+fi
+
+AC_ARG_ENABLE(INFO,
+AC_HELP_STRING([--disable-INFO],[disables compilation of informational messages]),
+[case "${enableval}" in
+  yes) ENABLE_INFO=yes ;;
+  no)  ENABLE_INFO=no ;;
+  *) AC_MSG_ERROR(bad value ${enableval} for --enable-INFO) ;;
+esac],
+[ENABLE_INFO=yes]) dnl Default value
+if test x$ENABLE_INFO = xyes; then
+  AC_DEFINE(GST_INFO_ENABLED, 1, [Define if INFO statements should be compiled in])
+fi
+
+AC_ARG_ENABLE(debug-color,
+AC_HELP_STRING([--disable-debug-color],[disables color output of DEBUG and INFO output]),
+[case "${enableval}" in
+  yes) ENABLE_DEBUG_COLOR=yes ;;
+  no)  ENABLE_DEBUG_COLOR=no ;;
+  *) AC_MSG_ERROR(bad value ${enableval} for --enable-debug-color) ;;
+esac],
+[ENABLE_DEBUG_COLOR=yes]) dnl Default value
+if test "x$ENABLE_DEBUG_COLOR" = xyes; then
+  AC_DEFINE(GST_DEBUG_COLOR, 1, [Define if debugging messages should be colorized])
+fi
+])
diff --git a/common/m4/gst-default.m4 b/common/m4/gst-default.m4
new file mode 100644
index 0000000..8de9756
--- /dev/null
+++ b/common/m4/gst-default.m4
@@ -0,0 +1,120 @@
+dnl default elements used for tests and such
+
+dnl AG_GST_DEFAULT_ELEMENTS
+
+AC_DEFUN([AG_GST_DEFAULT_ELEMENTS],
+[
+  dnl decide on default elements
+  dnl FIXME: describe where exactly this gets used
+  dnl FIXME: decide if it's a problem that this could point to sinks from
+  dnl        depending plugin modules
+  dnl FIXME: when can we just use autoaudiosrc and autovideosrc?
+  DEFAULT_AUDIOSINK="autoaudiosink"
+  DEFAULT_VIDEOSINK="autovideosink"
+  DEFAULT_AUDIOSRC="alsasrc"
+  DEFAULT_VIDEOSRC="v4l2src"
+  DEFAULT_VISUALIZER="goom"
+  case "$host" in
+    *-sun-* | *pc-solaris* )
+      DEFAULT_AUDIOSRC="sunaudiosrc"
+      ;;
+    *-darwin* )
+      DEFAULT_AUDIOSRC="osxaudiosrc"
+      ;;
+  esac
+
+  dnl Default audio sink
+  AC_ARG_WITH(default-audiosink,
+    AC_HELP_STRING([--with-default-audiosink], [specify default audio sink]),
+    [
+      case "${withval}" in
+        yes) AC_MSG_ERROR(bad value ${withval} for --with-default-audiosink) ;;
+        no)  AC_MSG_ERROR(bad value ${withval} for --with-default-audiosink) ;;
+        *)   DEFAULT_AUDIOSINK="${withval}" ;;
+      esac
+    ],
+    [
+      DEFAULT_AUDIOSINK="$DEFAULT_AUDIOSINK"
+    ] dnl Default value as determined above
+  )
+  AC_MSG_NOTICE(Using $DEFAULT_AUDIOSINK as default audio sink)
+  AC_SUBST(DEFAULT_AUDIOSINK)
+  AC_DEFINE_UNQUOTED(DEFAULT_AUDIOSINK, "$DEFAULT_AUDIOSINK",
+    [Default audio sink])
+
+  dnl Default audio source
+  AC_ARG_WITH(default-audiosrc,
+    AC_HELP_STRING([--with-default-audiosrc], [specify default audio source]),
+    [
+      case "${withval}" in
+        yes) AC_MSG_ERROR(bad value ${withval} for --with-default-audiosrc) ;;
+        no)  AC_MSG_ERROR(bad value ${withval} for --with-default-audiosrc) ;;
+        *)   DEFAULT_AUDIOSRC="${withval}" ;;
+      esac
+    ],
+    [
+      DEFAULT_AUDIOSRC="$DEFAULT_AUDIOSRC"
+    ] dnl Default value as determined above
+  )
+  AC_MSG_NOTICE(Using $DEFAULT_AUDIOSRC as default audio source)
+  AC_SUBST(DEFAULT_AUDIOSRC)
+  AC_DEFINE_UNQUOTED(DEFAULT_AUDIOSRC, "$DEFAULT_AUDIOSRC",
+    [Default audio source])
+
+  dnl Default video sink
+  AC_ARG_WITH(default-videosink,
+    AC_HELP_STRING([--with-default-videosink], [specify default video sink]),
+    [
+      case "${withval}" in
+        yes) AC_MSG_ERROR(bad value ${withval} for --with-default-videosink) ;;
+        no)  AC_MSG_ERROR(bad value ${withval} for --with-default-videosink) ;;
+        *)   DEFAULT_VIDEOSINK="${withval}" ;;
+      esac
+    ],
+    [
+      DEFAULT_VIDEOSINK="$DEFAULT_VIDEOSINK"
+    ] dnl Default value as determined above
+  )
+  AC_MSG_NOTICE(Using $DEFAULT_VIDEOSINK as default video sink)
+  AC_SUBST(DEFAULT_VIDEOSINK)
+  AC_DEFINE_UNQUOTED(DEFAULT_VIDEOSINK, "$DEFAULT_VIDEOSINK",
+    [Default video sink])
+
+  dnl Default video source
+  AC_ARG_WITH(default-videosrc,
+    AC_HELP_STRING([--with-default-videosrc], [specify default video source]),
+    [
+      case "${withval}" in
+        yes) AC_MSG_ERROR(bad value ${withval} for --with-default-videosrc) ;;
+        no)  AC_MSG_ERROR(bad value ${withval} for --with-default-videosrc) ;;
+        *)   DEFAULT_VIDEOSRC="${withval}" ;;
+      esac
+    ],
+    [
+      DEFAULT_VIDEOSRC="$DEFAULT_VIDEOSRC"
+    ] dnl Default value as determined above
+  )
+  AC_MSG_NOTICE(Using $DEFAULT_VIDEOSRC as default video source)
+  AC_SUBST(DEFAULT_VIDEOSRC)
+  AC_DEFINE_UNQUOTED(DEFAULT_VIDEOSRC, "$DEFAULT_VIDEOSRC",
+    [Default video source])
+
+    dnl Default visualizer
+  AC_ARG_WITH(default-visualizer,
+    AC_HELP_STRING([--with-default-visualizer], [specify default visualizer]),
+    [
+      case "${withval}" in
+        yes) AC_MSG_ERROR(bad value ${withval} for --with-default-visualizer) ;;
+        no)  AC_MSG_ERROR(bad value ${withval} for --with-default-visualizer) ;;
+        *)   DEFAULT_VISUALIZER="${withval}" ;;
+      esac
+    ],
+    [
+      DEFAULT_VISUALIZER="$DEFAULT_VISUALIZER"
+    ] dnl Default value as determined above
+  )
+  AC_MSG_NOTICE(Using $DEFAULT_VISUALIZER as default visualizer)
+  AC_SUBST(DEFAULT_VISUALIZER)
+  AC_DEFINE_UNQUOTED(DEFAULT_VISUALIZER, "$DEFAULT_VISUALIZER",
+    [Default visualizer])
+])
diff --git a/common/m4/gst-doc.m4 b/common/m4/gst-doc.m4
new file mode 100644
index 0000000..5d3f0fd
--- /dev/null
+++ b/common/m4/gst-doc.m4
@@ -0,0 +1,92 @@
+AC_DEFUN([AG_GST_DOCBOOK_CHECK],
+[
+  dnl choose a location to install docbook docs in
+  if test "x$PACKAGE_TARNAME" = "x"
+  then
+    AC_MSG_ERROR([Internal error - PACKAGE_TARNAME not set])
+  fi
+  docdir="\$(datadir)/doc/$PACKAGE_TARNAME-$GST_API_VERSION"
+
+  dnl enable/disable docbook documentation building
+  AC_ARG_ENABLE(docbook,
+  AC_HELP_STRING([--enable-docbook],
+                 [use docbook to build documentation [default=no]]),,
+                 enable_docbook=no)
+
+  have_docbook=no
+
+  if test x$enable_docbook = xyes; then
+    dnl check if we actually have everything we need
+
+    dnl check for docbook tools
+    AC_CHECK_PROG(HAVE_DOCBOOK2PS, docbook2ps, yes, no)
+    AC_CHECK_PROG(HAVE_XSLTPROC, xsltproc, yes, no)
+    AC_CHECK_PROG(HAVE_JADETEX, jadetex, yes, no)
+    AC_CHECK_PROG(HAVE_PS2PDF, ps2pdf, yes, no)
+
+    dnl check if we can process docbook stuff
+    AS_DOCBOOK(have_docbook=yes, have_docbook=no)
+
+    dnl check for extra tools
+    AC_CHECK_PROG(HAVE_DVIPS, dvips, yes, no)
+    AC_CHECK_PROG(HAVE_XMLLINT, xmllint, yes, no)
+
+    AC_CHECK_PROG(HAVE_PNGTOPNM, pngtopnm, yes, no)
+    AC_CHECK_PROG(HAVE_PNMTOPS,  pnmtops,  yes, no)
+    AC_CHECK_PROG(HAVE_EPSTOPDF, epstopdf, yes, no)
+
+    dnl check if we can generate HTML
+    if test "x$HAVE_XSLTPROC" = "xyes" && \
+       test "x$enable_docbook" = "xyes" && \
+       test "x$HAVE_XMLLINT" = "xyes"; then
+      DOC_HTML=yes
+      AC_MSG_NOTICE(Will output HTML documentation)
+     else
+      DOC_HTML=no
+      AC_MSG_NOTICE(Will not output HTML documentation)
+    fi
+
+    dnl check if we can generate PS
+    if test "x$HAVE_DOCBOOK2PS" = "xyes" && \
+       test "x$enable_docbook" = "xyes" && \
+       test "x$HAVE_XMLLINT" = "xyes" && \
+       test "x$HAVE_JADETEX" = "xyes" && \
+       test "x$HAVE_DVIPS" = "xyes" && \
+       test "x$HAVE_PNGTOPNM" = "xyes" && \
+       test "x$HAVE_PNMTOPS" = "xyes"; then
+      DOC_PS=yes
+      AC_MSG_NOTICE(Will output PS documentation)
+    else
+      DOC_PS=no
+      AC_MSG_NOTICE(Will not output PS documentation)
+    fi
+
+    dnl check if we can generate PDF - using only ps2pdf
+    if test "x$DOC_PS" = "xyes" && \
+       test "x$enable_docbook" = "xyes" && \
+       test "x$HAVE_XMLLINT" = "xyes" && \
+       test "x$HAVE_PS2PDF" = "xyes"; then
+      DOC_PDF=yes
+      AC_MSG_NOTICE(Will output PDF documentation)
+    else
+      DOC_PDF=no
+      AC_MSG_NOTICE(Will not output PDF documentation)
+    fi
+
+    dnl if we don't have everything, we should disable
+    if test "x$have_docbook" != "xyes"; then
+      enable_docbook=no
+    fi
+  fi
+
+  dnl if we're going to install documentation, tell us where
+  if test "x$have_docbook" = "xyes"; then
+    AC_MSG_NOTICE(Installing documentation in $docdir)
+    AC_SUBST(docdir)
+  fi
+
+  AM_CONDITIONAL(ENABLE_DOCBOOK,      test x$enable_docbook = xyes)
+  AM_CONDITIONAL(DOC_HTML,            test x$DOC_HTML = xyes)
+  AM_CONDITIONAL(DOC_PDF,             test x$DOC_PDF = xyes)
+  AM_CONDITIONAL(DOC_PS,              test x$DOC_PS = xyes)
+])
diff --git a/common/m4/gst-dowhile.m4 b/common/m4/gst-dowhile.m4
new file mode 100644
index 0000000..069808d
--- /dev/null
+++ b/common/m4/gst-dowhile.m4
@@ -0,0 +1,24 @@
+dnl
+dnl Check for working do while(0) macros. This is used by G_STMT_START
+dnl and G_STMT_END in glib/gmacros.h. Without having this defined we
+dnl get "ambigious if-else" compiler warnings when compling C++ code.
+dnl
+dnl Copied from GLib's configure.in
+dnl
+AC_DEFUN([AG_GST_CHECK_DOWHILE_MACROS],[
+
+dnl *** check for working do while(0) macros ***
+AC_CACHE_CHECK([for working do while(0) macros], _cv_g_support_dowhile_macros, [
+	AC_TRY_COMPILE([],[
+	#define STMT_START do
+	#define STMT_END while(0)
+	#define STMT_TEST STMT_START { i = 0; } STMT_END
+	int main(void) { int i = 1; STMT_TEST; return i; }],
+	[_cv_g_support_dowhile_macros=yes],
+	[_cv_g_support_dowhile_macros=no],
+	[_cv_g_support_dowhile_macros=yes])
+])
+if test x$_cv_g_support_dowhile_macros = xyes; then
+  AC_DEFINE(HAVE_DOWHILE_MACROS, 1, [define for working do while(0) macros])
+fi
+])
diff --git a/common/m4/gst-error.m4 b/common/m4/gst-error.m4
new file mode 100644
index 0000000..bce80d2
--- /dev/null
+++ b/common/m4/gst-error.m4
@@ -0,0 +1,298 @@
+dnl handle various error-related things
+
+dnl Thomas Vander Stichele 
+dnl Tim-Philipp Müller 
+
+dnl Last modification: 2008-02-18
+
+dnl AG_GST_SET_ERROR_CFLAGS([ADD-WERROR], [MORE_FLAGS])
+dnl AG_GST_SET_ERROR_CXXFLAGS([ADD-WERROR], [MORE_FLAGS])
+dnl AG_GST_SET_LEVEL_DEFAULT([IS-GIT-VERSION])
+
+
+dnl Sets WARNING_CFLAGS and ERROR_CFLAGS to something the compiler 
+dnl will accept and AC_SUBST them so they are available in Makefile
+dnl
+dnl WARNING_CFLAGS will contain flags to make the compiler emit more
+dnl   warnings.
+dnl ERROR_CFLAGS will contain flags to make those warnings fatal,
+dnl   unless ADD-WERROR is set to "no"
+dnl 
+dnl If MORE_FLAGS is set, tries to add each of the given flags
+dnl to WARNING_CFLAGS if the compiler supports them. Each flag is 
+dnl tested separately.
+dnl
+dnl These flags can be overridden at make time:
+dnl make ERROR_CFLAGS=
+AC_DEFUN([AG_GST_SET_ERROR_CFLAGS],
+[
+  AC_REQUIRE([AC_PROG_CC])
+  AC_REQUIRE([AS_COMPILER_FLAG])
+
+  WARNING_CFLAGS=""
+  ERROR_CFLAGS=""
+
+  dnl if we support -Wall, set it unconditionally
+  AS_COMPILER_FLAG(-Wall,
+                   WARNING_CFLAGS="$WARNING_CFLAGS -Wall")
+
+  dnl Warn if declarations after statements are used (C99 extension)
+  AS_COMPILER_FLAG(-Wdeclaration-after-statement,
+        WARNING_CFLAGS="$WARNING_CFLAGS -Wdeclaration-after-statement")
+
+  dnl Warn if variable length arrays are used (C99 extension)
+  AS_COMPILER_FLAG(-Wvla,
+        WARNING_CFLAGS="$WARNING_CFLAGS -Wvla")
+
+  dnl Warn for invalid pointer arithmetic
+  AS_COMPILER_FLAG(-Wpointer-arith,
+        WARNING_CFLAGS="$WARNING_CFLAGS -Wpointer-arith")
+
+  dnl if asked for, add -Werror if supported
+  if test "x$1" != "xno"
+  then
+    AS_COMPILER_FLAG(-Werror, ERROR_CFLAGS="$ERROR_CFLAGS -Werror")
+
+    dnl if -Werror isn't suported, try -errwarn=%all (Sun Forte case)
+    if test "x$ERROR_CFLAGS" = "x"
+    then
+      AS_COMPILER_FLAG([-errwarn=%all], [
+          ERROR_CFLAGS="-errwarn=%all"
+          dnl try -errwarn=%all,no%E_EMPTY_DECLARATION,
+          dnl no%E_STATEMENT_NOT_REACHED,no%E_ARGUEMENT_MISMATCH,
+          dnl no%E_MACRO_REDEFINED (Sun Forte case)
+          dnl For Forte we need disable "empty declaration" warning produced by un-needed semicolon
+          dnl "statement not reached" disabled because there is g_assert_not_reached () in some places
+          dnl "macro redefined" because of gst/gettext.h
+          dnl FIXME: is it really supposed to be 'ARGUEMENT' and not 'ARGUMENT'?
+          for f in 'no%E_EMPTY_DECLARATION' \
+                   'no%E_STATEMENT_NOT_REACHED' \
+                   'no%E_ARGUEMENT_MISMATCH' \
+                   'no%E_MACRO_REDEFINED' \
+                   'no%E_LOOP_NOT_ENTERED_AT_TOP'
+          do
+            AS_COMPILER_FLAG([-errwarn=%all,$f], [
+              ERROR_CFLAGS="$ERROR_CFLAGS,$f"
+            ])
+          done
+      ])
+    else
+      dnl Add -fno-strict-aliasing for GLib versions before 2.19.8
+      dnl as before G_LOCK and friends caused strict aliasing compiler
+      dnl warnings.
+      PKG_CHECK_EXISTS([glib-2.0 < 2.19.8], [
+        AS_COMPILER_FLAG(-fno-strict-aliasing,
+            ERROR_CFLAGS="$ERROR_CFLAGS -fno-strict-aliasing")
+	])
+    fi
+  fi
+
+  if test "x$2" != "x"
+  then
+    UNSUPPORTED=""
+    list="$2"
+    for each in $list
+    do
+      AS_COMPILER_FLAG($each,
+          WARNING_CFLAGS="$WARNING_CFLAGS $each",
+          UNSUPPORTED="$UNSUPPORTED $each")
+    done
+    if test "X$UNSUPPORTED" != X ; then
+      AC_MSG_NOTICE([unsupported compiler flags: $UNSUPPORTED])
+    fi
+  fi
+
+  AC_SUBST(WARNING_CFLAGS)
+  AC_SUBST(ERROR_CFLAGS)
+  AC_MSG_NOTICE([set WARNING_CFLAGS to $WARNING_CFLAGS])
+  AC_MSG_NOTICE([set ERROR_CFLAGS to $ERROR_CFLAGS])
+])
+
+dnl Sets WARNING_CXXFLAGS and ERROR_CXXFLAGS to something the compiler 
+dnl will accept and AC_SUBST them so they are available in Makefile
+dnl
+dnl WARNING_CXXFLAGS will contain flags to make the compiler emit more
+dnl   warnings.
+dnl ERROR_CXXFLAGS will contain flags to make those warnings fatal,
+dnl   unless ADD-WERROR is set to "no"
+dnl 
+dnl If MORE_FLAGS is set, tries to add each of the given flags
+dnl to WARNING_CFLAGS if the compiler supports them. Each flag is 
+dnl tested separately.
+dnl
+dnl These flags can be overridden at make time:
+dnl make ERROR_CXXFLAGS=
+AC_DEFUN([AG_GST_SET_ERROR_CXXFLAGS],
+[
+  AC_REQUIRE([AC_PROG_CXX])
+  AC_REQUIRE([AS_CXX_COMPILER_FLAG])
+
+  ERROR_CXXFLAGS=""
+  WARNING_CXXFLAGS=""
+
+  dnl if we support -Wall, set it unconditionally
+  AS_CXX_COMPILER_FLAG(-Wall, WARNING_CXXFLAGS="$WARNING_CXXFLAGS -Wall")
+
+  dnl if asked for, add -Werror if supported
+  if test "x$1" != "xno"
+  then
+    AS_CXX_COMPILER_FLAG(-Werror, ERROR_CXXFLAGS="$ERROR_CXXFLAGS -Werror")
+
+    if test "x$ERROR_CXXFLAGS" != "x"
+    then
+        dnl add exceptions
+        AS_CXX_COMPILER_FLAG([-Wno-non-virtual-dtor], ERROR_CXXFLAGS="$ERROR_CXXFLAGS -Wno-non-virtual-dtor")
+
+	dnl Add -fno-strict-aliasing for GLib versions before 2.19.8
+	dnl as before G_LOCK and friends caused strict aliasing compiler
+	dnl warnings.
+	PKG_CHECK_EXISTS([glib-2.0 < 2.19.8], [
+	  AS_CXX_COMPILER_FLAG([-fno-strict-aliasing],
+	    ERROR_CXXFLAGS="$ERROR_CXXFLAGS -fno-strict-aliasing")
+	  ])
+    else
+      dnl if -Werror isn't suported, try -errwarn=%all
+      AS_CXX_COMPILER_FLAG([-errwarn=%all], ERROR_CXXFLAGS="$ERROR_CXXFLAGS -errwarn=%all")
+      if test "x$ERROR_CXXFLAGS" != "x"; then
+        dnl try -errwarn=%all,no%E_EMPTY_DECLARATION,
+        dnl no%E_STATEMENT_NOT_REACHED,no%E_ARGUEMENT_MISMATCH,
+        dnl no%E_MACRO_REDEFINED (Sun Forte case)
+        dnl For Forte we need disable "empty declaration" warning produced by un-needed semicolon
+        dnl "statement not reached" disabled because there is g_assert_not_reached () in some places
+        dnl "macro redefined" because of gst/gettext.h
+        dnl FIXME: is it really supposed to be 'ARGUEMENT' and not 'ARGUMENT'?
+        dnl FIXME: do any of these work with the c++ compiler? if not, why
+        dnl do we check at all?
+        for f in 'no%E_EMPTY_DECLARATION' \
+                 'no%E_STATEMENT_NOT_REACHED' \
+                 'no%E_ARGUEMENT_MISMATCH' \
+                 'no%E_MACRO_REDEFINED' \
+                 'no%E_LOOP_NOT_ENTERED_AT_TOP'
+        do
+          AS_CXX_COMPILER_FLAG([-errwarn=%all,$f], [ERROR_CXXFLAGS="$ERROR_CXXFLAGS,$f"])
+        done
+      fi
+    fi
+  fi
+
+  if test "x$2" != "x"
+  then
+    UNSUPPORTED=""
+    list="$2"
+    for each in $list
+    do
+      AS_CXX_COMPILER_FLAG($each,
+          WARNING_CXXFLAGS="$WARNING_CXXFLAGS $each",
+          UNSUPPORTED="$UNSUPPORTED $each")
+    done
+    if test "X$UNSUPPORTED" != X ; then
+      AC_MSG_NOTICE([unsupported compiler flags: $UNSUPPORTED])
+    fi
+  fi
+
+  AC_SUBST(WARNING_CXXFLAGS)
+  AC_SUBST(ERROR_CXXFLAGS)
+  AC_MSG_NOTICE([set WARNING_CXXFLAGS to $WARNING_CXXFLAGS])
+  AC_MSG_NOTICE([set ERROR_CXXFLAGS to $ERROR_CXXFLAGS])
+])
+
+dnl Sets WARNING_OBJCFLAGS and ERROR_OBJCFLAGS to something the compiler 
+dnl will accept and AC_SUBST them so they are available in Makefile
+dnl
+dnl WARNING_OBJCFLAGS will contain flags to make the compiler emit more
+dnl   warnings.
+dnl ERROR_OBJCFLAGS will contain flags to make those warnings fatal,
+dnl   unless ADD-WERROR is set to "no"
+dnl 
+dnl If MORE_FLAGS is set, tries to add each of the given flags
+dnl to WARNING_CFLAGS if the compiler supports them. Each flag is 
+dnl tested separately.
+dnl
+dnl These flags can be overridden at make time:
+dnl make ERROR_OBJCFLAGS=
+AC_DEFUN([AG_GST_SET_ERROR_OBJCFLAGS],
+[
+  AC_REQUIRE([AC_PROG_OBJC])
+  AC_REQUIRE([AS_OBJC_COMPILER_FLAG])
+
+  ERROR_OBJCFLAGS=""
+  WARNING_OBJCFLAGS=""
+
+  dnl if we support -Wall, set it unconditionally
+  AS_OBJC_COMPILER_FLAG(-Wall, WARNING_OBJCFLAGS="$WARNING_OBJCFLAGS -Wall")
+
+  dnl if asked for, add -Werror if supported
+  if test "x$1" != "xno"
+  then
+    AS_OBJC_COMPILER_FLAG(-Werror, ERROR_OBJCFLAGS="$ERROR_OBJCFLAGS -Werror")
+
+    if test "x$ERROR_OBJCFLAGS" != "x"
+    then
+	dnl Add -fno-strict-aliasing for GLib versions before 2.19.8
+	dnl as before G_LOCK and friends caused strict aliasing compiler
+	dnl warnings.
+	PKG_CHECK_EXISTS([glib-2.0 < 2.19.8], [
+	  AS_OBJC_COMPILER_FLAG([-fno-strict-aliasing],
+	    ERROR_OBJCFLAGS="$ERROR_OBJCFLAGS -fno-strict-aliasing")
+	  ])
+    else
+      dnl if -Werror isn't suported, try -errwarn=%all
+      AS_OBJC_COMPILER_FLAG([-errwarn=%all], ERROR_OBJCFLAGS="$ERROR_OBJCFLAGS -errwarn=%all")
+      if test "x$ERROR_OBJCFLAGS" != "x"; then
+        dnl try -errwarn=%all,no%E_EMPTY_DECLARATION,
+        dnl no%E_STATEMENT_NOT_REACHED,no%E_ARGUEMENT_MISMATCH,
+        dnl no%E_MACRO_REDEFINED (Sun Forte case)
+        dnl For Forte we need disable "empty declaration" warning produced by un-needed semicolon
+        dnl "statement not reached" disabled because there is g_assert_not_reached () in some places
+        dnl "macro redefined" because of gst/gettext.h
+        dnl FIXME: is it really supposed to be 'ARGUEMENT' and not 'ARGUMENT'?
+        dnl FIXME: do any of these work with the c++ compiler? if not, why
+        dnl do we check at all?
+        for f in 'no%E_EMPTY_DECLARATION' \
+                 'no%E_STATEMENT_NOT_REACHED' \
+                 'no%E_ARGUEMENT_MISMATCH' \
+                 'no%E_MACRO_REDEFINED' \
+                 'no%E_LOOP_NOT_ENTERED_AT_TOP'
+        do
+          AS_OBJC_COMPILER_FLAG([-errwarn=%all,$f], [ERROR_OBJCFLAGS="$ERROR_OBJCFLAGS,$f"])
+        done
+      fi
+    fi
+  fi
+
+  if test "x$2" != "x"
+  then
+    UNSUPPORTED=""
+    list="$2"
+    for each in $list
+    do
+      AS_OBJC_COMPILER_FLAG($each,
+          WARNING_OBJCFLAGS="$WARNING_OBJCFLAGS $each",
+          UNSUPPORTED="$UNSUPPORTED $each")
+    done
+    if test "X$UNSUPPORTED" != X ; then
+      AC_MSG_NOTICE([unsupported compiler flags: $UNSUPPORTED])
+    fi
+  fi
+
+  AC_SUBST(WARNING_OBJCFLAGS)
+  AC_SUBST(ERROR_OBJCFLAGS)
+  AC_MSG_NOTICE([set WARNING_OBJCFLAGS to $WARNING_OBJCFLAGS])
+  AC_MSG_NOTICE([set ERROR_OBJCFLAGS to $ERROR_OBJCFLAGS])
+])
+
+dnl Sets the default error level for debugging messages
+AC_DEFUN([AG_GST_SET_LEVEL_DEFAULT],
+[
+  dnl define correct errorlevel for debugging messages. We want to have
+  dnl GST_ERROR messages printed when running cvs builds
+  if test "x[$1]" = "xyes"; then
+    GST_LEVEL_DEFAULT=GST_LEVEL_ERROR
+  else
+    GST_LEVEL_DEFAULT=GST_LEVEL_NONE
+  fi
+  AC_DEFINE_UNQUOTED(GST_LEVEL_DEFAULT, $GST_LEVEL_DEFAULT,
+    [Default errorlevel to use])
+  dnl AC_SUBST so we can use it for win32/common/config.h
+  AC_SUBST(GST_LEVEL_DEFAULT)
+])
diff --git a/common/m4/gst-feature.m4 b/common/m4/gst-feature.m4
new file mode 100644
index 0000000..876215e
--- /dev/null
+++ b/common/m4/gst-feature.m4
@@ -0,0 +1,297 @@
+dnl Perform a check for a feature for GStreamer
+dnl Richard Boulton 
+dnl Thomas Vander Stichele  added useful stuff
+dnl Last modification: 25/06/2001
+dnl
+dnl AG_GST_CHECK_FEATURE(FEATURE-NAME, FEATURE-DESCRIPTION,
+dnl                   DEPENDENT-PLUGINS, TEST-FOR-FEATURE,
+dnl                   DISABLE-BY-DEFAULT, ACTION-IF-USE, ACTION-IF-NOTUSE)
+dnl
+dnl This macro adds a command line argument to allow the user to enable
+dnl or disable a feature, and if the feature is enabled, performs a supplied
+dnl test to check if the feature is available.
+dnl
+dnl The test should define HAVE_ to "yes" or "no" depending
+dnl on whether the feature is available.
+dnl
+dnl The macro will set USE_ to "yes" or "no" depending on
+dnl whether the feature is to be used.
+dnl Thomas changed this, so that when USE_ was already set
+dnl to no, then it stays that way.
+dnl
+dnl The macro will call AM_CONDITIONAL(USE_, ...) to allow
+dnl the feature to control what is built in Makefile.ams.  If you want
+dnl additional actions resulting from the test, you can add them with the
+dnl ACTION-IF-USE and ACTION-IF-NOTUSE parameters.
+dnl
+dnl FEATURE-NAME        is the name of the feature, and should be in
+dnl                     purely upper case characters.
+dnl FEATURE-DESCRIPTION is used to describe the feature in help text for
+dnl                     the command line argument.
+dnl DEPENDENT-PLUGINS   lists any plug-ins which depend on this feature.
+dnl TEST-FOR-FEATURE    is a test which sets HAVE_ to "yes"
+dnl                     or "no" depending on whether the feature is
+dnl                     available.
+dnl DISABLE-BY-DEFAULT  if "disabled", the feature is disabled by default,
+dnl                     if any other value, the feature is enabled by default.
+dnl ACTION-IF-USE       any extra actions to perform if the feature is to be
+dnl                     used.
+dnl ACTION-IF-NOTUSE    any extra actions to perform if the feature is not to
+dnl                     be used.
+dnl
+dnl
+dnl thomas :
+dnl we also added a history.
+dnl GST_PLUGINS_YES will contain all plugins to be built
+dnl                 that were checked through AG_GST_CHECK_FEATURE
+dnl GST_PLUGINS_NO will contain those that won't be built
+
+AC_DEFUN([AG_GST_CHECK_FEATURE],
+[echo
+AC_MSG_NOTICE(*** checking feature: [$2] ***)
+if test "x[$3]" != "x"
+then
+  AC_MSG_NOTICE(*** for plug-ins: [$3] ***)
+fi
+dnl
+builtin(define, [gst_endisable], ifelse($5, [disabled], [enable], [disable]))dnl
+dnl if it is set to NO, then don't even consider it for building
+NOUSE=
+if test "x$USE_[$1]" = "xno"; then
+  NOUSE="yes"
+fi
+AC_ARG_ENABLE(translit([$1], A-Z, a-z),
+  [  ]builtin(format, --%-26s gst_endisable %s, gst_endisable-translit([$1], A-Z, a-z), [$2]ifelse([$3],,,: [$3])),
+  [ case "${enableval}" in
+      yes) USE_[$1]=yes;;
+      no) USE_[$1]=no;;
+      *) AC_MSG_ERROR(bad value ${enableval} for --enable-translit([$1], A-Z, a-z)) ;;
+    esac],
+  [ USE_$1=]ifelse($5, [disabled], [no], [yes]))           dnl DEFAULT
+
+dnl *** set it back to no if it was preset to no
+if test "x$NOUSE" = "xyes"; then
+  USE_[$1]="no"
+  AC_MSG_WARN(*** $3 pre-configured not to be built)
+fi
+NOUSE=
+
+dnl *** Check if it is ported or not
+if echo " [$GST_PLUGINS_NONPORTED] " | tr , ' ' | grep -i " [$1] " > /dev/null; then
+  USE_[$1]="no"
+  AC_MSG_WARN(*** $3 not ported)
+fi
+
+dnl *** If it's enabled
+
+if test x$USE_[$1] = xyes; then
+  dnl save compile variables before the test
+
+  gst_check_save_LIBS=$LIBS
+  gst_check_save_LDFLAGS=$LDFLAGS
+  gst_check_save_CFLAGS=$CFLAGS
+  gst_check_save_CPPFLAGS=$CPPFLAGS
+  gst_check_save_CXXFLAGS=$CXXFLAGS
+
+  HAVE_[$1]=no
+  dnl TEST_FOR_FEATURE
+  $4
+
+  LIBS=$gst_check_save_LIBS
+  LDFLAGS=$gst_check_save_LDFLAGS
+  CFLAGS=$gst_check_save_CFLAGS
+  CPPFLAGS=$gst_check_save_CPPFLAGS
+  CXXFLAGS=$gst_check_save_CXXFLAGS
+
+  dnl If it isn't found, unset USE_[$1]
+  if test x$HAVE_[$1] = xno; then
+    USE_[$1]=no
+  else
+    ifelse([$3], , :, [AC_MSG_NOTICE(*** These plugins will be built: [$3])])
+  fi
+fi
+dnl *** Warn if it's disabled or not found
+if test x$USE_[$1] = xyes; then
+  ifelse([$6], , :, [$6])
+  if test "x$3" != "x"; then
+    GST_PLUGINS_YES="\t[$3]\n$GST_PLUGINS_YES"
+  fi
+  AC_DEFINE(HAVE_[$1], , [Define to enable $2]ifelse($3,,, [ (used by $3)]).)
+else
+  ifelse([$3], , :, [AC_MSG_NOTICE(*** These plugins will not be built: [$3])])
+  if test "x$3" != "x"; then
+    GST_PLUGINS_NO="\t[$3]\n$GST_PLUGINS_NO"
+  fi
+  ifelse([$7], , :, [$7])
+fi
+dnl *** Define the conditional as appropriate
+AM_CONDITIONAL(USE_[$1], test x$USE_[$1] = xyes)
+])
+
+dnl Use AC_CHECK_LIB and AC_CHECK_HEADER to do both tests at once
+dnl sets HAVE_module if we have it
+dnl Richard Boulton 
+dnl Last modification: 26/06/2001
+dnl AG_GST_CHECK_LIBHEADER(FEATURE-NAME, LIB NAME, LIB FUNCTION, EXTRA LD FLAGS,
+dnl                     HEADER NAME, ACTION-IF-FOUND, ACTION-IF-NOT-FOUND)
+dnl
+dnl This check was written for GStreamer: it should be renamed and checked
+dnl for portability if you decide to use it elsewhere.
+dnl
+AC_DEFUN([AG_GST_CHECK_LIBHEADER],
+[
+  AC_CHECK_LIB([$2], [$3], HAVE_[$1]=yes, HAVE_[$1]=no,[$4])
+  if test "x$HAVE_[$1]" = "xyes"; then
+    AC_CHECK_HEADER([$5], :, HAVE_[$1]=no)
+    if test "x$HAVE_[$1]" = "xyes"; then
+      dnl execute what needs to be
+      ifelse([$6], , :, [$6])
+    else
+      ifelse([$7], , :, [$7])
+    fi
+  else
+    ifelse([$7], , :, [$7])
+  fi
+  AC_SUBST(HAVE_[$1])
+]
+)
+
+dnl 2004-02-14 Thomas - changed to get set properly and use proper output
+dnl 2003-06-27 Benjamin Otte - changed to make this work with gstconfig.h
+dnl
+dnl Add a subsystem --disable flag and all the necessary symbols and substitions
+dnl
+dnl AG_GST_CHECK_SUBSYSTEM_DISABLE(SYSNAME, [subsystem name])
+dnl
+AC_DEFUN([AG_GST_CHECK_SUBSYSTEM_DISABLE],
+[
+  dnl this define will replace each literal subsys_def occurrence with
+  dnl the lowercase hyphen-separated subsystem
+  dnl e.g. if $1 is GST_DEBUG then subsys_def will be a macro with gst-debug
+  define([subsys_def],translit([$1], _A-Z, -a-z))
+
+  AC_ARG_ENABLE(subsys_def,
+    AC_HELP_STRING(--disable-subsys_def, [disable $2]),
+    [
+      case "${enableval}" in
+        yes) GST_DISABLE_[$1]=no ;;
+        no) GST_DISABLE_[$1]=yes ;;
+        *) AC_MSG_ERROR([bad value ${enableval} for --enable-subsys_def]) ;;
+       esac
+    ],
+    [GST_DISABLE_[$1]=no]) dnl Default value
+
+  if test x$GST_DISABLE_[$1] = xyes; then
+    AC_MSG_NOTICE([disabled subsystem [$2]])
+    GST_DISABLE_[$1]_DEFINE="#define GST_DISABLE_$1 1"
+  else
+    GST_DISABLE_[$1]_DEFINE="/* #undef GST_DISABLE_$1 */"
+  fi
+  AC_SUBST(GST_DISABLE_[$1]_DEFINE)
+  undefine([subsys_def])
+])
+
+
+dnl Parse gstconfig.h for feature and defines add the symbols and substitions
+dnl
+dnl AG_GST_PARSE_SUBSYSTEM_DISABLE(GST_CONFIGPATH, FEATURE)
+dnl
+AC_DEFUN([AG_GST_PARSE_SUBSYSTEM_DISABLE],
+[
+  grep >/dev/null "#undef GST_DISABLE_$2" $1
+  if test $? = 0; then
+    GST_DISABLE_[$2]=0
+  else
+    GST_DISABLE_[$2]=1
+  fi
+  AC_SUBST(GST_DISABLE_[$2])
+])
+
+dnl Parse gstconfig.h and defines add the symbols and substitions
+dnl
+dnl GST_CONFIGPATH=`$PKG_CONFIG --variable=includedir gstreamer-1.0`"/gst/gstconfig.h"
+dnl AG_GST_PARSE_SUBSYSTEM_DISABLES(GST_CONFIGPATH)
+dnl
+AC_DEFUN([AG_GST_PARSE_SUBSYSTEM_DISABLES],
+[
+  AG_GST_PARSE_SUBSYSTEM_DISABLE($1,GST_DEBUG)
+  AG_GST_PARSE_SUBSYSTEM_DISABLE($1,LOADSAVE)
+  AG_GST_PARSE_SUBSYSTEM_DISABLE($1,PARSE)
+  AG_GST_PARSE_SUBSYSTEM_DISABLE($1,TRACE)
+  AG_GST_PARSE_SUBSYSTEM_DISABLE($1,ALLOC_TRACE)
+  AG_GST_PARSE_SUBSYSTEM_DISABLE($1,REGISTRY)
+  AG_GST_PARSE_SUBSYSTEM_DISABLE($1,PLUGIN)
+  AG_GST_PARSE_SUBSYSTEM_DISABLE($1,XML)
+])
+
+dnl AG_GST_CHECK_GST_DEBUG_DISABLED(ACTION-IF-DISABLED, ACTION-IF-NOT-DISABLED)
+dnl
+dnl Checks if the GStreamer debugging system is disabled in the core version
+dnl we are compiling against (by checking gstconfig.h)
+dnl
+AC_DEFUN([AG_GST_CHECK_GST_DEBUG_DISABLED],
+[
+  AC_REQUIRE([AG_GST_CHECK_GST])
+
+  AC_MSG_CHECKING([whether the GStreamer debugging system is enabled])
+  AC_LANG_PUSH([C])
+  save_CFLAGS="$CFLAGS"
+  CFLAGS="$GST_CFLAGS $CFLAGS"
+  AC_COMPILE_IFELSE([
+    AC_LANG_SOURCE([[
+      #include 
+      #ifdef GST_DISABLE_GST_DEBUG
+      #error "debugging disabled, make compiler fail"
+      #endif]])], [ debug_system_enabled=yes], [debug_system_enabled=no])
+  CFLAGS="$save_CFLAGS"
+  AC_LANG_POP([C])
+
+  AC_MSG_RESULT([$debug_system_enabled])
+
+  if test "x$debug_system_enabled" = "xyes" ; then
+    $2
+    true
+  else
+    $1
+    true
+  fi
+])
+
+dnl relies on GST_PLUGINS_ALL, GST_PLUGINS_SELECTED, GST_PLUGINS_YES,
+dnl GST_PLUGINS_NO, and BUILD_EXTERNAL
+AC_DEFUN([AG_GST_OUTPUT_PLUGINS], [
+
+printf "configure: *** Plug-ins without external dependencies that will be built:\n"
+( for i in $GST_PLUGINS_SELECTED; do printf '\t'$i'\n'; done ) | sort
+printf "\n"
+
+printf "configure: *** Plug-ins without external dependencies that will NOT be built:\n"
+( for i in $GST_PLUGINS_ALL; do
+    case " $GST_PLUGINS_SELECTED " in
+      *\ $i\ *)
+	;;
+      *)
+	printf '\t'$i'\n'
+	;;
+    esac
+  done ) | sort
+printf "\n"
+
+printf "configure: *** Plug-ins that have NOT been ported:\n"
+( for i in $GST_PLUGINS_NONPORTED; do
+	printf '\t'$i'\n'
+  done ) | sort
+printf "\n"
+
+if test "x$BUILD_EXTERNAL" = "xno"; then
+  printf "configure: *** No plug-ins with external dependencies will be built\n"
+else
+  printf "configure: *** Plug-ins with dependencies that will be built:"
+  printf "$GST_PLUGINS_YES\n" | sort
+  printf "\n"
+  printf "configure: *** Plug-ins with dependencies that will NOT be built:"
+  printf "$GST_PLUGINS_NO\n" | sort
+  printf "\n"
+fi
+])
+
diff --git a/common/m4/gst-function.m4 b/common/m4/gst-function.m4
new file mode 100644
index 0000000..61adfd3
--- /dev/null
+++ b/common/m4/gst-function.m4
@@ -0,0 +1,63 @@
+dnl
+dnl Check for compiler mechanism to show functions in debugging
+dnl copied from an Ali patch floating on the internet
+dnl
+AC_DEFUN([AG_GST_CHECK_FUNCTION],[
+  dnl #1: __PRETTY_FUNCTION__
+  AC_MSG_CHECKING(whether $CC implements __PRETTY_FUNCTION__)
+  AC_CACHE_VAL(gst_cv_have_pretty_function,[
+    AC_TRY_LINK([#include ],
+                [printf("%s", __PRETTY_FUNCTION__);],
+                gst_cv_have_pretty_function=yes,
+                gst_cv_have_pretty_function=no)
+  ])
+  AC_MSG_RESULT($gst_cv_have_pretty_function)
+  if test "$gst_cv_have_pretty_function" = yes; then
+    AC_DEFINE(HAVE_PRETTY_FUNCTION, 1,
+              [defined if the compiler implements __PRETTY_FUNCTION__])
+  fi
+
+dnl #2: __FUNCTION__
+  AC_MSG_CHECKING(whether $CC implements __FUNCTION__)
+  AC_CACHE_VAL(gst_cv_have_function,[
+    AC_TRY_LINK([#include ],
+                [printf("%s", __FUNCTION__);],
+                gst_cv_have_function=yes,
+                gst_cv_have_function=no)
+  ])
+  AC_MSG_RESULT($gst_cv_have_function)
+  if test "$gst_cv_have_function" = yes; then
+    AC_DEFINE(HAVE_FUNCTION, 1,
+              [defined if the compiler implements __FUNCTION__])
+  fi
+
+dnl #3: __func__
+  AC_MSG_CHECKING(whether $CC implements __func__)
+  AC_CACHE_VAL(gst_cv_have_func,[
+    AC_TRY_LINK([#include ],
+                [printf("%s", __func__);],
+                gst_cv_have_func=yes,
+                gst_cv_have_func=no)
+  ])
+  AC_MSG_RESULT($gst_cv_have_func)
+  if test "$gst_cv_have_func" = yes; then
+    AC_DEFINE(HAVE_FUNC, 1,
+              [defined if the compiler implements __func__])
+  fi
+
+dnl now define FUNCTION to whatever works, and fallback to ""
+  if test "$gst_cv_have_pretty_function" = yes; then
+    function=__PRETTY_FUNCTION__
+  else
+    if test "$gst_cv_have_function" = yes; then
+      function=__FUNCTION__
+    else
+      if test "$gst_cv_have_func" = yes; then
+        function=__func__
+      else
+        function=\"\"
+      fi
+    fi
+  fi
+  AC_DEFINE_UNQUOTED(GST_FUNCTION, $function, [macro to use to show function name])
+])
diff --git a/common/m4/gst-gettext.m4 b/common/m4/gst-gettext.m4
new file mode 100644
index 0000000..df817eb
--- /dev/null
+++ b/common/m4/gst-gettext.m4
@@ -0,0 +1,28 @@
+dnl gettext setup
+
+dnl AG_GST_GETTEXT([gettext-package])
+dnl defines GETTEXT_PACKAGE and LOCALEDIR
+
+AC_DEFUN([AG_GST_GETTEXT],
+[
+  if test "$USE_NLS" = "yes"; then
+    GETTEXT_PACKAGE=[$1]
+  else
+    GETTEXT_PACKAGE=[NULL]
+  fi
+  AC_SUBST(GETTEXT_PACKAGE)
+  AC_DEFINE_UNQUOTED([GETTEXT_PACKAGE], "$GETTEXT_PACKAGE",
+                     [gettext package name])
+
+  dnl make sure po/Makevars is kept in sync with GETTEXT_PACKAGE
+  if test -e "${srcdir}/po/Makevars"; then
+    if ! grep -e "$1" "${srcdir}/po/Makevars"; then
+      AC_MSG_ERROR([DOMAIN in po/Makevars does not match GETTEXT_PACKAGE $1])
+    fi
+  fi
+
+  dnl define LOCALEDIR in config.h
+  AS_AC_EXPAND(LOCALEDIR, $datadir/locale)
+  AC_DEFINE_UNQUOTED([LOCALEDIR], "$LOCALEDIR",
+                     [gettext locale dir])
+])
diff --git a/common/m4/gst-glib2.m4 b/common/m4/gst-glib2.m4
new file mode 100644
index 0000000..63f0f46
--- /dev/null
+++ b/common/m4/gst-glib2.m4
@@ -0,0 +1,126 @@
+dnl check for a minimum version of GLib
+
+dnl AG_GST_GLIB_CHECK([minimum-version-required])
+
+AC_DEFUN([AG_GST_GLIB_CHECK],
+[
+  AC_REQUIRE([AS_NANO])
+
+  dnl Minimum required version of GLib
+  GLIB_REQ=[$1]
+  if test "x$GLIB_REQ" = "x"
+  then
+    AC_MSG_ERROR([Please specify a required version for GLib 2.0])
+  fi
+  AC_SUBST(GLIB_REQ)
+
+  dnl Check for glib with everything
+  AG_GST_PKG_CHECK_MODULES(GLIB,
+    glib-2.0 >= $GLIB_REQ gobject-2.0 gmodule-no-export-2.0)
+
+  if test "x$HAVE_GLIB" = "xno"; then
+    AC_MSG_ERROR([This package requires GLib >= $GLIB_REQ to compile.])
+  fi
+
+  dnl Add define to tell GLib that threading is always enabled within GStreamer
+  dnl code (optimisation, bypasses checks if the threading system is enabled
+  dnl when using threading primitives)
+  GLIB_EXTRA_CFLAGS="$GLIB_EXTRA_CFLAGS -DG_THREADS_MANDATORY"
+
+  dnl Define G_DISABLE_DEPRECATED for development versions
+  if test "x`expr $PACKAGE_VERSION_MINOR % 2`" = "x1" -a "x`expr $PACKAGE_VERSION_MICRO '<' 90`" = "x1"; then
+    GLIB_EXTRA_CFLAGS="$GLIB_EXTRA_CFLAGS -DG_DISABLE_DEPRECATED"
+  fi
+
+  AC_ARG_ENABLE(gobject-cast-checks,
+    AS_HELP_STRING([--enable-gobject-cast-checks[=@<:@no/auto/yes@:>@]],
+      [Enable GObject cast checks]),[enable_gobject_cast_checks=$enableval],
+    [enable_gobject_cast_checks=auto])
+
+  if test "x$enable_gobject_cast_checks" = "xauto"; then
+    dnl Turn on cast checks only for development versions
+    if test "x`expr $PACKAGE_VERSION_MINOR % 2`" = "x1" -a "x`expr $PACKAGE_VERSION_MICRO '<' 90`" = "x1"; then
+      enable_gobject_cast_checks=yes
+    else
+      enable_gobject_cast_checks=no
+    fi
+  fi
+
+  if test "x$enable_gobject_cast_checks" = "xno"; then
+    GLIB_EXTRA_CFLAGS="$GLIB_EXTRA_CFLAGS -DG_DISABLE_CAST_CHECKS"
+  fi
+
+  AC_ARG_ENABLE(glib-asserts,
+    AS_HELP_STRING([--enable-glib-asserts[=@<:@no/auto/yes@:>@]],
+      [Enable GLib assertion]),[enable_glib_assertions=$enableval],
+    [enable_glib_assertions=auto])
+
+  if test "x$enable_glib_assertions" = "xauto"; then
+    dnl Enable assertions only for development versions
+    if test "x`expr $PACKAGE_VERSION_MINOR % 2`" = "x1" -a "x`expr $PACKAGE_VERSION_MICRO '<' 90`" = "x1"; then
+      enable_glib_assertions=yes
+    else
+      enable_glib_assertions=no
+    fi
+  fi
+
+  if test "x$enable_glib_assertions" = "xno"; then
+    GLIB_EXTRA_CFLAGS="$GLIB_EXTRA_CFLAGS -DG_DISABLE_ASSERT"
+  fi
+
+  dnl Find location of glib utils. People may want to or have to override these,
+  dnl e.g. in a cross-compile situation where PATH is a bit messed up. We need
+  dnl for these tools to work on the host, so can't just use the one from the
+  dnl GLib installation that pkg-config picks up, as that might be for a
+  dnl different target architecture.
+  dnl
+  dnl glib-genmarshal:
+  AC_MSG_CHECKING(for glib-genmarshal)
+  if test "x$GLIB_GENMARSHAL" != "x"; then
+    AC_MSG_RESULT([$GLIB_GENMARSHAL (from environment)])
+  else
+    GLIB_GENMARSHAL=`$PKG_CONFIG --variable=glib_genmarshal glib-2.0`
+    if $GLIB_GENMARSHAL --version 2>/dev/null >/dev/null; then
+      AC_MSG_RESULT([$GLIB_GENMARSHAL (from pkg-config path)])
+    else
+      AC_PATH_PROG(GLIB_GENMARSHAL, [glib-genmarshal], [glib-genmarshal])
+      AC_MSG_RESULT([$GLIB_GENMARSHAL])
+    fi
+  fi
+  if ! $GLIB_GENMARSHAL --version 2>/dev/null >/dev/null; then
+    AC_MSG_WARN([$GLIB_GENMARSHAL does not seem to work!])
+  fi
+  AC_SUBST(GLIB_GENMARSHAL)
+
+  dnl glib-mkenums:
+  AC_MSG_CHECKING(for glib-mkenums)
+  if test "x$GLIB_MKENUMS" != "x"; then
+    AC_MSG_RESULT([$GLIB_MKENUMS (from environment)])
+  else
+    dnl glib-mkenums is written in perl so should always work really
+    GLIB_MKENUMS=`$PKG_CONFIG --variable=glib_mkenums glib-2.0`
+    AC_MSG_RESULT([$GLIB_MKENUMS])
+  fi
+  if ! $GLIB_MKENUMS --version 2>/dev/null >/dev/null; then
+    AC_MSG_WARN([$GLIB_MKENUMS does not seem to work!])
+  fi
+  AC_SUBST(GLIB_MKENUMS)
+
+  AC_SUBST(GLIB_EXTRA_CFLAGS)
+
+  dnl Now check for GIO
+  PKG_CHECK_MODULES(GIO, gio-2.0 >= $GLIB_REQ)
+  if test "x$HAVE_GIO" = "xno"; then
+    AC_MSG_ERROR([This package requires GIO >= $GLIB_REQ to compile.])
+  fi
+  
+  GIO_MODULE_DIR="`$PKG_CONFIG --variable=giomoduledir gio-2.0`"
+  AC_DEFINE_UNQUOTED(GIO_MODULE_DIR, "$GIO_MODULE_DIR",
+      [The GIO modules directory.])
+  GIO_LIBDIR="`$PKG_CONFIG --variable=libdir gio-2.0`"
+  AC_DEFINE_UNQUOTED(GIO_LIBDIR, "$GIO_LIBDIR",
+      [The GIO library directory.])
+  AC_SUBST(GIO_CFLAGS)
+  AC_SUBST(GIO_LIBS)
+  AC_SUBST(GIO_LDFLAGS)
+])
diff --git a/common/m4/gst-libxml2.m4 b/common/m4/gst-libxml2.m4
new file mode 100644
index 0000000..4a843f0
--- /dev/null
+++ b/common/m4/gst-libxml2.m4
@@ -0,0 +1,52 @@
+dnl call this macro with the minimum required version as an argument
+dnl this macro sets and AC_SUBSTs XML_CFLAGS and XML_LIBS
+dnl it also sets LIBXML_PKG, used for the pkg-config file
+
+AC_DEFUN([AG_GST_LIBXML2_CHECK],
+[
+  dnl Minimum required version of libxml2
+  dnl default to 2.4.9 if not specified
+  LIBXML2_REQ=ifelse([$1],,2.4.9,[$1])
+  AC_SUBST(LIBXML2_REQ)
+
+  dnl check for libxml2
+  PKG_CHECK_MODULES(XML, libxml-2.0 >= $LIBXML2_REQ,
+                    HAVE_LIBXML2=yes, [
+                      AC_MSG_RESULT(no)
+                      HAVE_LIBXML2=no
+                    ])
+  if test "x$HAVE_LIBXML2" = "xyes"; then
+    AC_DEFINE(HAVE_LIBXML2, 1, [Define if libxml2 is available])
+  else
+    AC_MSG_ERROR([
+        Need libxml2 and development headers/files to build GStreamer.
+
+        You can do without libxml2 if you pass --disable-loadsave to
+        configure, but that breaks ABI, so don't do that unless you
+        are building for an embedded setup and know what you are doing.
+    ])
+  fi
+  dnl this is for the .pc file
+  LIBXML_PKG=', libxml-2.0'
+  AC_SUBST(LIBXML_PKG)
+  AC_SUBST(XML_LIBS)
+  AC_SUBST(XML_CFLAGS)
+
+  dnl XML_LIBS might pull in -lz without zlib actually being on the system, so
+  dnl try linking with these LIBS and CFLAGS
+  ac_save_CFLAGS=$CFLAGS
+  ac_save_LIBS=$LIBS
+  CFLAGS="$CFLAGS $XML_CFLAGS"
+  LIBS="$LIBS $XML_LIBS"
+  AC_TRY_LINK([
+#include 
+#include 
+],[
+/* function body */
+],
+    AC_MSG_NOTICE([Test xml2 program linked]),
+    AC_MSG_ERROR([Could not link libxml2 test program.  Check if you have the necessary dependencies.])
+  )
+  CFLAGS="$ac_save_CFLAGS"
+  LIBS="$ac_save_LIBS"
+])
diff --git a/common/m4/gst-package-release-datetime.m4 b/common/m4/gst-package-release-datetime.m4
new file mode 100644
index 0000000..bc885e3
--- /dev/null
+++ b/common/m4/gst-package-release-datetime.m4
@@ -0,0 +1,89 @@
+dnl macros to set GST_PACKAGE_RELEASE_DATETIME
+
+dnl ===========================================================================
+dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME
+dnl
+dnl Usage:
+dnl
+dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME()
+dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([no]...)
+dnl sets the release datetime to the current date
+dnl (no = this is not a release, but git or prerelease)
+dnl
+dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([YYYY-MM-DD])
+dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([yes], [YYYY-MM-DD])
+dnl sets the release datetime to the specified date (and time, if given)
+dnl (yes = this is a release, not git or prerelease)
+dnl
+dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([yes], [DOAP-FILE], [RELEASE-VERSION])
+dnl sets the release date to the release date associated with version
+dnl RELEASE-VERSION in the .doap file DOAP-FILE
+dnl (yes = this is a release, not git or prerelease)
+dnl
+dnl We need to treat pre-releases like git because there won't be an entry
+dnl in the .doap file for pre-releases yet, and we don't want to use the
+dnl date of the last release either.
+dnl ===========================================================================
+AC_DEFUN([AG_GST_SET_PACKAGE_RELEASE_DATETIME],
+[
+  dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME()
+  dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([yes]...)
+  if test "x$1" = "xno" -o "x$1" = "x"; then
+    GST_PACKAGE_RELEASE_DATETIME=`date -u "+%Y-%m-%dT%H:%MZ"`
+  elif test "x$1" = "xyes"; then
+    dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([no], ["YYYY-MM-DD"])
+    dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([no], [DOAP-FILE], [RELEASE-VERSION])
+    if ( echo $1 | grep '^20[1-9][0-9]-[0-1][0-9]-[0-3][0-9]' >/dev/null ) ; then
+      GST_PACKAGE_RELEASE_DATETIME=$1
+    else
+      dnl we assume the .doap file contains the date as YYYY-MM-DD
+      YYYY_MM_DD=`sh "${srcdir}/common/extract-release-date-from-doap-file" $3 $2`;
+      if test "x$YYYY_MM_DD" != "x"; then
+        GST_PACKAGE_RELEASE_DATETIME=$YYYY_MM_DD
+      else
+        AC_MSG_ERROR([SET_PACKAGE_RELEASE_DATETIME: could not extract
+            release date for release version $3 from $2])
+        GST_PACKAGE_RELEASE_DATETIME=""
+      fi
+    fi
+  dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([YYYY-MM-DD])
+  elif ( echo $1 | grep '^20[1-9][0-9]-[0-1][0-9]-[0-3][0-9]' >/dev/null ) ; then
+    GST_PACKAGE_RELEASE_DATETIME=$1
+  else
+    AC_MSG_WARN([SET_PACKAGE_RELEASE_DATETIME: invalid first argument])
+    GST_PACKAGE_RELEASE_DATETIME=""
+  fi
+
+  if test "x$GST_PACKAGE_RELEASE_DATETIME" = "x"; then
+    AC_MSG_WARN([Invalid package release date time: $GST_PACKAGE_RELEASE_DATETIME])
+  else
+    AC_MSG_NOTICE([Setting GST_PACKAGE_RELEASE_DATETIME to $GST_PACKAGE_RELEASE_DATETIME])
+
+    AC_DEFINE_UNQUOTED([GST_PACKAGE_RELEASE_DATETIME],
+        ["$GST_PACKAGE_RELEASE_DATETIME"],
+        [GStreamer package release date/time for plugins as YYYY-MM-DD])
+  fi
+])
+
+dnl ===========================================================================
+dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME_WITH_NANO
+dnl
+dnl Usage:
+dnl
+dnl AG_GST_SET_PACKAGE_RELEASE_DATETIME([NANO-VERSION], [DOAP-FILE], [RELEASE-VERSION])
+dnl if NANO-VERSION is 0, sets the release date to the release date associated
+dnl with version RELEASE-VERSION in the .doap file DOAP-FILE, otherwise sets
+dnl the release date and time to the current date/time.
+dnl
+dnl We need to treat pre-releases like git because there won't be an entry
+dnl in the .doap file for pre-releases yet, and we don't want to use the
+dnl date of the last release either.
+dnl ===========================================================================
+AC_DEFUN([AG_GST_SET_PACKAGE_RELEASE_DATETIME_WITH_NANO],
+[
+  if test "x$1" = "x0"; then
+    AG_GST_SET_PACKAGE_RELEASE_DATETIME([yes], [ $2 ], [ $3 ])
+  else
+    AG_GST_SET_PACKAGE_RELEASE_DATETIME([no])
+  fi
+])
diff --git a/common/m4/gst-parser.m4 b/common/m4/gst-parser.m4
new file mode 100644
index 0000000..b002047
--- /dev/null
+++ b/common/m4/gst-parser.m4
@@ -0,0 +1,55 @@
+AC_DEFUN([AG_GST_BISON_CHECK],
+[
+  dnl FIXME: check if AC_PROG_YACC is suitable here
+  dnl FIXME: make precious
+  AC_PATH_PROG(BISON_PATH, bison, no)
+  if test x$BISON_PATH = xno; then
+    AC_MSG_ERROR(Could not find bison)
+  fi
+
+  dnl check bison version
+  dnl we need version >= 2.4 for the '<>' support
+  dnl in the parser.
+  dnl First lines observed: 'bison (GNU Bison) 2.3' or 'GNU Bison version 1.28'
+  bison_min_version=2.4
+  bison_version=`$BISON_PATH --version | head -n 1 |  sed 's/^[[^0-9]]*//' | sed 's/[[^0-9]]*$//' | cut -d' ' -f1`
+  AC_MSG_CHECKING([bison version $bison_version >= $bison_min_version])
+
+  if perl -we "exit ((v$bison_version ge v$bison_min_version) ? 0 : 1)"; then
+    AC_MSG_RESULT([yes])
+  else
+    AC_MSG_ERROR([no])
+  fi
+])
+
+AC_DEFUN([AG_GST_FLEX_CHECK],
+[
+  dnl we require flex for building the parser
+  AC_PATH_PROG(FLEX_PATH, flex, no)
+  if test x$FLEX_PATH = xno; then
+    AC_MSG_ERROR(Could not find flex)
+  fi
+
+  dnl check flex version
+  dnl we need version >= 2.5.31 for the reentrancy support
+  dnl in the parser.
+  flex_min_version=2.5.31
+  flex_version=`$FLEX_PATH --version | head -n 1 | awk '{print $2}'`
+  AC_MSG_CHECKING([flex version $flex_version >= $flex_min_version])
+  if perl -w < \$min_version_major) ||
+     ((\$flex_version_major == \$min_version_major) &&
+      (\$flex_version_minor > \$min_version_minor)) ||
+     ((\$flex_version_major == \$min_version_major) &&
+      (\$flex_version_minor == \$min_version_minor) &&
+      (\$flex_version_micro >= \$min_version_micro)))
+     ? 0 : 1);
+EOF
+  then
+    AC_MSG_RESULT(yes)
+  else
+    AC_MSG_ERROR([no])
+  fi
+])
diff --git a/common/m4/gst-platform.m4 b/common/m4/gst-platform.m4
new file mode 100644
index 0000000..40d6faf
--- /dev/null
+++ b/common/m4/gst-platform.m4
@@ -0,0 +1,67 @@
+dnl AG_GST_PLATFORM
+dnl Check for platform specific features and define some variables
+dnl
+dnl GST_EXTRA_MODULE_SUFFIX: contains a platform specific
+dnl   extra module suffix additional to G_MODULE_SUFFIX
+dnl
+dnl HAVE_OSX: Defined if compiling for OS X
+dnl
+dnl GST_HAVE_UNSAFE_FORK: Defined if fork is unsafe (Windows)
+dnl
+dnl HAVE_WIN32: Defined if compiling on Win32
+dnl
+
+AC_DEFUN([AG_GST_PLATFORM],
+[
+  AC_REQUIRE([AC_CANONICAL_HOST])
+
+  case $host_os in
+    rhapsody*)
+      AC_DEFINE_UNQUOTED(GST_EXTRA_MODULE_SUFFIX, [".dylib"], [Extra platform specific plugin suffix])
+      ;;
+    darwin*)
+      AC_DEFINE_UNQUOTED(GST_EXTRA_MODULE_SUFFIX, [".dylib"], [Extra platform specific plugin suffix])
+      AC_DEFINE_UNQUOTED(HAVE_OSX, 1, [Defined if compiling for OSX])
+      ;;
+    cygwin*)
+      AC_DEFINE_UNQUOTED(GST_HAVE_UNSAFE_FORK, 1, [Defined when registry scanning through fork is unsafe])
+      ;;
+    mingw* | msvc* | mks*)
+      dnl HAVE_WIN32 currently means "disable POSIXisms".
+      AC_DEFINE_UNQUOTED(HAVE_WIN32, 1, [Defined if compiling for Windows])
+
+      dnl define __MSVCRT_VERSION__ version if not set already by the
+      dnl compiler (ie. mostly for mingw). This is needed for things like
+      dnl __stat64 to be available. If set by the compiler, ensure it's
+      dnl new enough - we need at least WinXP SP2.
+      AC_TRY_COMPILE([ ], [ return __MSVCRT_VERSION__; ], [
+          AC_TRY_COMPILE([ ], [
+            #if __MSVCRT_VERSION__ < 0x0601
+            #error "MSVCRT too old"
+            #endif
+          ], [
+            AC_MSG_NOTICE([MSVCRT version looks ok])
+          ], [
+            AC_MSG_ERROR([MSVCRT version too old, need at least WinXP SP2])
+          ])
+      ], [
+        AC_MSG_NOTICE([Setting MSVCRT version to 0x0601])
+        AC_DEFINE_UNQUOTED(__MSVCRT_VERSION__, 0x0601, [We need at least WinXP SP2 for __stat64])
+      ])
+      ;;
+     *)
+      ;;
+  esac
+])
+
+AC_DEFUN([AG_GST_LIBTOOL_PREPARE],
+[
+  dnl Persuade libtool to also link (-l) a 'pure' (DirectX) static lib,
+  dnl i.e. as opposed to only import lib with dll counterpart.
+  dnl Needs to be tweaked before libtool's checks.
+  case $host_os in
+  cygwin* | mingw*)
+    lt_cv_deplibs_check_method=pass_all
+    ;;
+  esac
+])
\ No newline at end of file
diff --git a/common/m4/gst-plugin-docs.m4 b/common/m4/gst-plugin-docs.m4
new file mode 100644
index 0000000..0e2ab6e
--- /dev/null
+++ b/common/m4/gst-plugin-docs.m4
@@ -0,0 +1,25 @@
+dnl AG_GST_PLUGIN_DOCS([MINIMUM-GTK-DOC-VERSION])
+dnl
+dnl checks for prerequisites for the common/mangle-tmpl.py script
+dnl used when building the plugin documentation
+
+AC_DEFUN([AG_GST_PLUGIN_DOCS],
+[
+  AC_BEFORE([GTK_DOC_CHECK],[$0])dnl check for gtk-doc first
+  AC_REQUIRE([AM_PATH_PYTHON])dnl find python first
+
+  build_plugin_docs=no
+  AC_MSG_CHECKING([whether to build plugin documentation])
+  if test x$enable_gtk_doc = xyes; then
+    if test x$PYTHON != x; then
+      build_plugin_docs=yes
+      AC_MSG_RESULT([yes])
+    else
+      AC_MSG_RESULT([no (python not found)])
+    fi
+  else
+    AC_MSG_RESULT([no (gtk-doc disabled or not available)])
+  fi
+
+  AM_CONDITIONAL(ENABLE_PLUGIN_DOCS, test x$build_plugin_docs = xyes)
+])
diff --git a/common/m4/gst-plugindir.m4 b/common/m4/gst-plugindir.m4
new file mode 100644
index 0000000..c9e1301
--- /dev/null
+++ b/common/m4/gst-plugindir.m4
@@ -0,0 +1,17 @@
+dnl AG_GST_SET_PLUGINDIR
+
+dnl AC_DEFINE PLUGINDIR to the full location where plug-ins will be installed
+dnl AC_SUBST plugindir, to be used in Makefile.am's
+
+AC_DEFUN([AG_GST_SET_PLUGINDIR],
+[
+  dnl define location of plugin directory
+  AS_AC_EXPAND(PLUGINDIR, ${libdir}/gstreamer-$GST_API_VERSION)
+  AC_DEFINE_UNQUOTED(PLUGINDIR, "$PLUGINDIR",
+    [directory where plugins are located])
+  AC_MSG_NOTICE([Using $PLUGINDIR as the plugin install location])
+
+  dnl plugin directory configure-time variable for use in Makefile.am
+  plugindir="\$(libdir)/gstreamer-$GST_API_VERSION"
+  AC_SUBST(plugindir)
+])
diff --git a/common/m4/gst-valgrind.m4 b/common/m4/gst-valgrind.m4
new file mode 100644
index 0000000..5c0d608
--- /dev/null
+++ b/common/m4/gst-valgrind.m4
@@ -0,0 +1,35 @@
+AC_DEFUN([AG_GST_VALGRIND_CHECK],
+[
+  dnl valgrind inclusion
+  AC_ARG_ENABLE(valgrind,
+    AC_HELP_STRING([--disable-valgrind], [disable run-time valgrind detection]),
+    [
+      case "${enableval}" in
+        yes) USE_VALGRIND="$USE_DEBUG" ;;
+        no)  USE_VALGRIND=no ;;
+        *) AC_MSG_ERROR(bad value ${enableval} for --enable-valgrind) ;;
+      esac],
+    [
+      USE_VALGRIND="$USE_DEBUG"
+    ]) dnl Default value
+
+  VALGRIND_REQ="3.0"
+  if test "x$USE_VALGRIND" = xyes; then
+    PKG_CHECK_MODULES(VALGRIND, valgrind >= $VALGRIND_REQ,
+      USE_VALGRIND="yes",
+      [
+        USE_VALGRIND="no"
+        AC_MSG_RESULT([no])
+      ])
+  fi
+
+  if test "x$USE_VALGRIND" = xyes; then
+    AC_DEFINE(HAVE_VALGRIND, 1, [Define if valgrind should be used])
+    AC_MSG_NOTICE(Using extra code paths for valgrind)
+  fi
+  AC_SUBST(VALGRIND_CFLAGS)
+  AC_SUBST(VALGRIND_LIBS)
+
+  AC_PATH_PROG(VALGRIND_PATH, valgrind, no)
+  AM_CONDITIONAL(HAVE_VALGRIND, test ! "x$VALGRIND_PATH" = "xno")
+])
diff --git a/common/m4/gst-x11.m4 b/common/m4/gst-x11.m4
new file mode 100644
index 0000000..d3baf2d
--- /dev/null
+++ b/common/m4/gst-x11.m4
@@ -0,0 +1,70 @@
+dnl macros for X-related detections
+dnl AC_SUBST's HAVE_X, X_CFLAGS, X_LIBS
+AC_DEFUN([AG_GST_CHECK_X],
+[
+  AC_PATH_XTRA
+  ac_cflags_save="$CFLAGS"
+  ac_cppflags_save="$CPPFLAGS"
+  CFLAGS="$CFLAGS $X_CFLAGS"
+  CPPFLAGS="$CPPFLAGS $X_CFLAGS"
+
+  dnl now try to find the HEADER
+  AC_CHECK_HEADER(X11/Xlib.h, HAVE_X="yes", HAVE_X="no")
+
+  if test "x$HAVE_X" = "xno"
+  then
+    AC_MSG_NOTICE([cannot find X11 development files])
+  else
+    dnl this is much more than we want
+    X_LIBS="$X_LIBS $X_PRE_LIBS $X_EXTRA_LIBS"
+    dnl AC_PATH_XTRA only defines the path needed to find the X libs,
+    dnl it does not add the libs; therefore we add them here
+    X_LIBS="$X_LIBS -lX11"
+    AC_SUBST(X_CFLAGS)
+    AC_SUBST(X_LIBS)
+  fi
+  AC_SUBST(HAVE_X)
+
+  CFLAGS="$ac_cflags_save"
+  CPPFLAGS="$ac_cppflags_save"
+])
+
+dnl *** XVideo ***
+dnl Look for the PIC library first, Debian requires it.
+dnl Check debian-devel archives for gory details.
+dnl 20020110:
+dnl At the moment XFree86 doesn't distribute shared libXv due
+dnl to unstable API.  On many platforms you CAN NOT link a shared
+dnl lib to a static non-PIC lib.  This is what the xvideo GStreamer
+dnl plug-in wants to do.  So Debian distributes a PIC compiled
+dnl version of the static lib for plug-ins to link to when it is
+dnl inappropriate to link the main application to libXv directly.
+dnl FIXME: add check if this platform can support linking to a
+dnl        non-PIC libXv, if not then don not use Xv.
+dnl FIXME: perhaps warn user if they have a shared libXv since
+dnl        this is an error until XFree86 starts shipping one
+AC_DEFUN([AG_GST_CHECK_XV],
+[
+  if test x$HAVE_X = xyes; then
+    AC_CHECK_LIB(Xv_pic, XvQueryExtension,
+                 HAVE_XVIDEO="yes", HAVE_XVIDEO="no",
+                 $X_LIBS -lXext)
+
+    if test x$HAVE_XVIDEO = xyes; then
+      XVIDEO_LIBS="-lXv_pic -lXext"
+      AC_SUBST(XVIDEO_LIBS)
+    else
+      dnl try again using something else if we didn't find it first
+      if test x$HAVE_XVIDEO = xno; then
+        AC_CHECK_LIB(Xv, XvQueryExtension,
+                   HAVE_XVIDEO="yes", HAVE_XVIDEO="no",
+                   $X_LIBS -lXext)
+
+        if test x$HAVE_XVIDEO = xyes; then
+          XVIDEO_LIBS="-lXv -lXext"
+          AC_SUBST(XVIDEO_LIBS)
+        fi
+      fi
+    fi
+  fi
+])
diff --git a/common/m4/gst.m4 b/common/m4/gst.m4
new file mode 100644
index 0000000..d4c53cb
--- /dev/null
+++ b/common/m4/gst.m4
@@ -0,0 +1,36 @@
+dnl AG_GST_INIT
+dnl sets up use of GStreamer configure.ac macros
+dnl all GStreamer autoconf macros are prefixed
+dnl with AG_GST_ for public macros
+dnl with _AG_GST_ for private macros
+dnl
+dnl We call AC_CANONICAL_TARGET and AC_CANONICAL_HOST so that
+dnl it is valid before AC_ARG_PROGRAM is called
+
+AC_DEFUN([AG_GST_INIT],
+[
+  m4_pattern_forbid(^_?AG_GST_)
+  AC_REQUIRE([AC_CANONICAL_HOST]) dnl we use host_ variables
+  AC_REQUIRE([AC_CANONICAL_TARGET]) dnl we use target_ variables
+])
+
+dnl AG_GST_PKG_CONFIG_PATH
+dnl
+dnl sets up a GST_PKG_CONFIG_PATH variable for use in Makefile.am
+dnl which contains the path of the in-tree pkgconfig directory first
+dnl and then any paths specified in PKG_CONFIG_PATH.
+dnl
+dnl We do this mostly so we don't have to use unportable shell constructs
+dnl such as ${PKG_CONFIG_PATH:+:$PKG_CONFIG_PATH} in Makefile.am to handle
+dnl the case where the environment variable is not set, but also in order
+dnl to avoid a trailing ':' in the PKG_CONFIG_PATH which apparently causes
+dnl problems with pkg-config on windows with msys/mingw.
+AC_DEFUN([AG_GST_PKG_CONFIG_PATH],
+[
+  GST_PKG_CONFIG_PATH="\$(top_builddir)/pkgconfig"
+  if test "x$PKG_CONFIG_PATH" != "x"; then
+    GST_PKG_CONFIG_PATH="$GST_PKG_CONFIG_PATH:$PKG_CONFIG_PATH"
+  fi
+  AC_SUBST([GST_PKG_CONFIG_PATH])
+  AC_MSG_NOTICE([Using GST_PKG_CONFIG_PATH = $GST_PKG_CONFIG_PATH])
+])
diff --git a/common/m4/gtk-doc.m4 b/common/m4/gtk-doc.m4
new file mode 100644
index 0000000..b243f1c
--- /dev/null
+++ b/common/m4/gtk-doc.m4
@@ -0,0 +1,70 @@
+dnl -*- mode: autoconf -*-
+
+# serial 1
+
+dnl Usage:
+dnl   GTK_DOC_CHECK([minimum-gtk-doc-version])
+AC_DEFUN([GTK_DOC_CHECK],
+[
+  AC_REQUIRE([PKG_PROG_PKG_CONFIG])
+  AC_BEFORE([AC_PROG_LIBTOOL],[$0])dnl setup libtool first
+  AC_BEFORE([AM_PROG_LIBTOOL],[$0])dnl setup libtool first
+
+  dnl check for tools we added during development
+  AC_PATH_PROG([GTKDOC_CHECK],[gtkdoc-check])
+  AC_PATH_PROGS([GTKDOC_REBASE],[gtkdoc-rebase],[true])
+  AC_PATH_PROG([GTKDOC_MKPDF],[gtkdoc-mkpdf])
+
+  dnl for overriding the documentation installation directory
+  AC_ARG_WITH([html-dir],
+    AS_HELP_STRING([--with-html-dir=PATH], [path to installed docs]),,
+    [with_html_dir='${datadir}/gtk-doc/html'])
+  HTML_DIR="$with_html_dir"
+  AC_SUBST([HTML_DIR])
+
+  dnl enable/disable documentation building
+  AC_ARG_ENABLE([gtk-doc],
+    AS_HELP_STRING([--enable-gtk-doc],
+                   [use gtk-doc to build documentation [[default=no]]]),,
+    [enable_gtk_doc=no])
+
+  if test x$enable_gtk_doc = xyes; then
+    ifelse([$1],[],
+      [PKG_CHECK_EXISTS([gtk-doc],,
+                        AC_MSG_ERROR([gtk-doc not installed and --enable-gtk-doc requested]))],
+      [PKG_CHECK_EXISTS([gtk-doc >= $1],,
+                        AC_MSG_ERROR([You need to have gtk-doc >= $1 installed to build $PACKAGE_NAME]))])
+    dnl don't check for glib if we build glib
+    if test "x$PACKAGE_NAME" != "xglib"; then
+      dnl don't fail if someone does not have glib
+      PKG_CHECK_MODULES(GTKDOC_DEPS, glib-2.0 >= 2.10.0 gobject-2.0  >= 2.10.0,,)
+    fi
+    dnl don't rely on sed being pulled in implicitly. Fixes Solaris build.
+    if test -z "$SED"; then
+      AC_PROG_SED
+    fi
+  fi
+
+  AC_MSG_CHECKING([whether to build gtk-doc documentation])
+  AC_MSG_RESULT($enable_gtk_doc)
+
+  dnl enable/disable output formats
+  AC_ARG_ENABLE([gtk-doc-html],
+    AS_HELP_STRING([--enable-gtk-doc-html],
+                   [build documentation in html format [[default=yes]]]),,
+    [enable_gtk_doc_html=yes])
+    AC_ARG_ENABLE([gtk-doc-pdf],
+      AS_HELP_STRING([--enable-gtk-doc-pdf],
+                     [build documentation in pdf format [[default=no]]]),,
+      [enable_gtk_doc_pdf=no])
+
+  if test -z "$GTKDOC_MKPDF"; then
+    enable_gtk_doc_pdf=no
+  fi
+
+
+  AM_CONDITIONAL([ENABLE_GTK_DOC], [test x$enable_gtk_doc = xyes])
+  AM_CONDITIONAL([GTK_DOC_BUILD_HTML], [test x$enable_gtk_doc_html = xyes])
+  AM_CONDITIONAL([GTK_DOC_BUILD_PDF], [test x$enable_gtk_doc_pdf = xyes])
+  AM_CONDITIONAL([GTK_DOC_USE_LIBTOOL], [test -n "$LIBTOOL"])
+])
diff --git a/common/m4/introspection.m4 b/common/m4/introspection.m4
new file mode 100644
index 0000000..589721c
--- /dev/null
+++ b/common/m4/introspection.m4
@@ -0,0 +1,94 @@
+dnl -*- mode: autoconf -*-
+dnl Copyright 2009 Johan Dahlin
+dnl
+dnl This file is free software; the author(s) gives unlimited
+dnl permission to copy and/or distribute it, with or without
+dnl modifications, as long as this notice is preserved.
+dnl
+
+# serial 1
+
+m4_define([_GOBJECT_INTROSPECTION_CHECK_INTERNAL],
+[
+    AC_BEFORE([AC_PROG_LIBTOOL],[$0])dnl setup libtool first
+    AC_BEFORE([AM_PROG_LIBTOOL],[$0])dnl setup libtool first
+    AC_BEFORE([LT_INIT],[$0])dnl setup libtool first
+
+    dnl enable/disable introspection
+    m4_if([$2], [require],
+    [dnl
+        enable_introspection=yes
+    ],[dnl
+        AC_ARG_ENABLE(introspection,
+                  AS_HELP_STRING([--enable-introspection[=@<:@no/auto/yes@:>@]],
+                                 [Enable introspection for this build]),, 
+                                 [enable_introspection=auto])
+    ])dnl
+
+    AC_MSG_CHECKING([for gobject-introspection])
+
+    dnl presence/version checking
+    AS_CASE([$enable_introspection],
+    [no], [dnl
+        found_introspection="no (disabled, use --enable-introspection to enable)"
+    ],dnl
+    [yes],[dnl
+        PKG_CHECK_EXISTS([gobject-introspection-1.0],,
+                         AC_MSG_ERROR([gobject-introspection-1.0 is not installed]))
+        PKG_CHECK_EXISTS([gobject-introspection-1.0 >= $1],
+                         found_introspection=yes,
+                         AC_MSG_ERROR([You need to have gobject-introspection >= $1 installed to build AC_PACKAGE_NAME]))
+    ],dnl
+    [auto],[dnl
+        PKG_CHECK_EXISTS([gobject-introspection-1.0 >= $1], found_introspection=yes, found_introspection=no)
+    ],dnl
+    [dnl	
+        AC_MSG_ERROR([invalid argument passed to --enable-introspection, should be one of @<:@no/auto/yes@:>@])
+    ])dnl
+
+    AC_MSG_RESULT([$found_introspection])
+
+    INTROSPECTION_SCANNER=
+    INTROSPECTION_COMPILER=
+    INTROSPECTION_GENERATE=
+    INTROSPECTION_GIRDIR=
+    INTROSPECTION_TYPELIBDIR=
+    if test "x$found_introspection" = "xyes"; then
+       INTROSPECTION_SCANNER=`$PKG_CONFIG --variable=g_ir_scanner gobject-introspection-1.0`
+       INTROSPECTION_COMPILER=`$PKG_CONFIG --variable=g_ir_compiler gobject-introspection-1.0`
+       INTROSPECTION_GENERATE=`$PKG_CONFIG --variable=g_ir_generate gobject-introspection-1.0`
+       INTROSPECTION_GIRDIR=`$PKG_CONFIG --variable=girdir gobject-introspection-1.0`
+       INTROSPECTION_TYPELIBDIR="$($PKG_CONFIG --variable=typelibdir gobject-introspection-1.0)"
+       INTROSPECTION_CFLAGS=`$PKG_CONFIG --cflags gobject-introspection-1.0`
+       INTROSPECTION_LIBS=`$PKG_CONFIG --libs gobject-introspection-1.0`
+       INTROSPECTION_MAKEFILE=`$PKG_CONFIG --variable=datadir gobject-introspection-1.0`/gobject-introspection-1.0/Makefile.introspection
+    fi
+    AC_SUBST(INTROSPECTION_SCANNER)
+    AC_SUBST(INTROSPECTION_COMPILER)
+    AC_SUBST(INTROSPECTION_GENERATE)
+    AC_SUBST(INTROSPECTION_GIRDIR)
+    AC_SUBST(INTROSPECTION_TYPELIBDIR)
+    AC_SUBST(INTROSPECTION_CFLAGS)
+    AC_SUBST(INTROSPECTION_LIBS)
+    AC_SUBST(INTROSPECTION_MAKEFILE)
+
+    AM_CONDITIONAL(HAVE_INTROSPECTION, test "x$found_introspection" = "xyes")
+])
+
+
+dnl Usage:
+dnl   GOBJECT_INTROSPECTION_CHECK([minimum-g-i-version])
+
+AC_DEFUN([GOBJECT_INTROSPECTION_CHECK],
+[
+  _GOBJECT_INTROSPECTION_CHECK_INTERNAL([$1])
+])
+
+dnl Usage:
+dnl   GOBJECT_INTROSPECTION_REQUIRE([minimum-g-i-version])
+
+
+AC_DEFUN([GOBJECT_INTROSPECTION_REQUIRE],
+[
+  _GOBJECT_INTROSPECTION_CHECK_INTERNAL([$1], [require])
+])
diff --git a/common/m4/orc.m4 b/common/m4/orc.m4
new file mode 100644
index 0000000..26b2459
--- /dev/null
+++ b/common/m4/orc.m4
@@ -0,0 +1,70 @@
+dnl pkg-config-based checks for Orc
+
+dnl specific:
+dnl ORC_CHECK([REQUIRED_VERSION])
+
+AC_DEFUN([ORC_CHECK],
+[
+  ORC_REQ=ifelse([$1], , "0.4.6", [$1])
+
+  AC_ARG_ENABLE(orc,
+  AC_HELP_STRING([--enable-orc],[use Orc if installed]),
+  [case "${enableval}" in
+    auto) enable_orc=auto ;;
+    yes) enable_orc=yes ;;
+    no)  enable_orc=no ;;
+    *) AC_MSG_ERROR(bad value ${enableval} for --enable-orc) ;;
+  esac
+  ],
+  [enable_orc=auto]) dnl Default value
+
+  if test "x$enable_orc" != "xno" ; then
+    PKG_CHECK_MODULES(ORC, orc-0.4 >= $ORC_REQ, [
+      AC_DEFINE(HAVE_ORC, 1, [Use Orc])
+      HAVE_ORC=yes
+      if test "x$ORCC" = "x" ; then
+        AC_MSG_CHECKING(for usable orcc)
+        ORCC=`$PKG_CONFIG --variable=orcc orc-0.4`
+        dnl check whether the orcc found by pkg-config can be run from the build environment
+        dnl if this is not the case (e.g. when cross-compiling) fall back to orcc from PATH
+        AS_IF([$ORCC --version 1> /dev/null 2> /dev/null], [], [ORCC=`which orcc`])
+        AC_MSG_RESULT($ORCC)
+      fi
+      AC_SUBST(ORCC)
+      ORCC_FLAGS="--compat $ORC_REQ"
+      AC_SUBST(ORCC_FLAGS)
+      AS_IF([test "x$ORCC" = "x"], [HAVE_ORCC=no], [HAVE_ORCC=yes])
+    ], [
+      if test "x$enable_orc" = "xyes" ; then
+        AC_MSG_ERROR([--enable-orc specified, but Orc >= $ORC_REQ not found])
+      fi
+      AC_DEFINE(DISABLE_ORC, 1, [Disable Orc])
+      HAVE_ORC=no
+      HAVE_ORCC=no
+    ])
+  else
+    AC_DEFINE(DISABLE_ORC, 1, [Disable Orc])
+    HAVE_ORC=no
+    HAVE_ORCC=no
+  fi
+  AM_CONDITIONAL(HAVE_ORC, [test "x$HAVE_ORC" = "xyes"])
+  AM_CONDITIONAL(HAVE_ORCC, [test "x$HAVE_ORCC" = "xyes"])
+
+]))
+
+AC_DEFUN([ORC_OUTPUT],
+[
+  if test "$HAVE_ORC" = yes ; then
+    printf "configure: *** Orc acceleration enabled.\n"
+  else
+    if test "x$enable_orc" = "xno" ; then
+      printf "configure: *** Orc acceleration disabled by --disable-orc.  Slower code paths\n"
+      printf "               will be used.\n"
+    else
+      printf "configure: *** Orc acceleration disabled.  Requires Orc >= $ORC_REQ, which was\n"
+      printf "               not found.  Slower code paths will be used.\n"
+    fi
+  fi
+  printf "\n"
+])
+
diff --git a/common/m4/pkg.m4 b/common/m4/pkg.m4
new file mode 100644
index 0000000..996e294
--- /dev/null
+++ b/common/m4/pkg.m4
@@ -0,0 +1,157 @@
+# pkg.m4 - Macros to locate and utilise pkg-config.            -*- Autoconf -*-
+#
+# Copyright © 2004 Scott James Remnant .
+#
+# This program is free software; you can redistribute it and/or modify
+# it under the terms of the GNU General Public License as published by
+# the Free Software Foundation; either version 2 of the License, or
+# (at your option) any later version.
+#
+# This program is distributed in the hope that it will be useful, but
+# WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+# General Public License for more details.
+#
+# You should have received a copy of the GNU General Public License
+# along with this program; if not, write to the Free Software
+# Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
+#
+# As a special exception to the GNU General Public License, if you
+# distribute this file as part of a program that contains a
+# configuration script generated by Autoconf, you may include it under
+# the same distribution terms that you use for the rest of that program.
+
+# PKG_PROG_PKG_CONFIG([MIN-VERSION])
+# ----------------------------------
+AC_DEFUN([PKG_PROG_PKG_CONFIG],
+[m4_pattern_forbid([^_?PKG_[A-Z_]+$])
+m4_pattern_allow([^PKG_CONFIG(_PATH)?$])
+AC_ARG_VAR([PKG_CONFIG], [path to pkg-config utility])dnl
+if test "x$ac_cv_env_PKG_CONFIG_set" != "xset"; then
+	AC_PATH_TOOL([PKG_CONFIG], [pkg-config])
+fi
+if test -n "$PKG_CONFIG"; then
+	_pkg_min_version=m4_default([$1], [0.9.0])
+	AC_MSG_CHECKING([pkg-config is at least version $_pkg_min_version])
+	if $PKG_CONFIG --atleast-pkgconfig-version $_pkg_min_version; then
+		AC_MSG_RESULT([yes])
+	else
+		AC_MSG_RESULT([no])
+		PKG_CONFIG=""
+	fi
+
+fi[]dnl
+])# PKG_PROG_PKG_CONFIG
+
+# PKG_CHECK_EXISTS(MODULES, [ACTION-IF-FOUND], [ACTION-IF-NOT-FOUND])
+#
+# Check to see whether a particular set of modules exists.  Similar
+# to PKG_CHECK_MODULES(), but does not set variables or print errors.
+#
+#
+# Similar to PKG_CHECK_MODULES, make sure that the first instance of
+# this or PKG_CHECK_MODULES is called, or make sure to call
+# PKG_CHECK_EXISTS manually
+# --------------------------------------------------------------
+AC_DEFUN([PKG_CHECK_EXISTS],
+[AC_REQUIRE([PKG_PROG_PKG_CONFIG])dnl
+if test -n "$PKG_CONFIG" && \
+    AC_RUN_LOG([$PKG_CONFIG --exists --print-errors "$1"]); then
+  m4_ifval([$2], [$2], [:])
+m4_ifvaln([$3], [else
+  $3])dnl
+fi])
+
+
+# _PKG_CONFIG([VARIABLE], [COMMAND], [MODULES])
+# ---------------------------------------------
+m4_define([_PKG_CONFIG],
+[if test -n "$PKG_CONFIG"; then
+    if test -n "$$1"; then
+        pkg_cv_[]$1="$$1"
+    else
+        PKG_CHECK_EXISTS([$3],
+                         [pkg_cv_[]$1=`$PKG_CONFIG --[]$2 "$3" 2>/dev/null`],
+			 [pkg_failed=yes])
+    fi
+else
+	pkg_failed=untried
+fi[]dnl
+])# _PKG_CONFIG
+
+# _PKG_SHORT_ERRORS_SUPPORTED
+# -----------------------------
+AC_DEFUN([_PKG_SHORT_ERRORS_SUPPORTED],
+[AC_REQUIRE([PKG_PROG_PKG_CONFIG])
+if $PKG_CONFIG --atleast-pkgconfig-version 0.20; then
+        _pkg_short_errors_supported=yes
+else
+        _pkg_short_errors_supported=no
+fi[]dnl
+])# _PKG_SHORT_ERRORS_SUPPORTED
+
+
+# PKG_CHECK_MODULES(VARIABLE-PREFIX, MODULES, [ACTION-IF-FOUND],
+# [ACTION-IF-NOT-FOUND])
+#
+#
+# Note that if there is a possibility the first call to
+# PKG_CHECK_MODULES might not happen, you should be sure to include an
+# explicit call to PKG_PROG_PKG_CONFIG in your configure.ac
+#
+#
+# --------------------------------------------------------------
+AC_DEFUN([PKG_CHECK_MODULES],
+[AC_REQUIRE([PKG_PROG_PKG_CONFIG])dnl
+AC_ARG_VAR([$1][_CFLAGS], [C compiler flags for $1, overriding pkg-config])dnl
+AC_ARG_VAR([$1][_LIBS], [linker flags for $1, overriding pkg-config])dnl
+
+pkg_failed=no
+AC_MSG_CHECKING([for $1])
+
+_PKG_CONFIG([$1][_CFLAGS], [cflags], [$2])
+_PKG_CONFIG([$1][_LIBS], [libs], [$2])
+
+m4_define([_PKG_TEXT], [Alternatively, you may set the environment variables $1[]_CFLAGS
+and $1[]_LIBS to avoid the need to call pkg-config.
+See the pkg-config man page for more details.])
+
+if test $pkg_failed = yes; then
+        _PKG_SHORT_ERRORS_SUPPORTED
+        if test $_pkg_short_errors_supported = yes; then
+	        $1[]_PKG_ERRORS=`$PKG_CONFIG --short-errors --errors-to-stdout --print-errors "$2"`
+        else
+	        $1[]_PKG_ERRORS=`$PKG_CONFIG --errors-to-stdout --print-errors "$2"`
+        fi
+	# Put the nasty error message in config.log where it belongs
+	echo "$$1[]_PKG_ERRORS" >&AS_MESSAGE_LOG_FD
+
+	ifelse([$4], , [AC_MSG_ERROR(dnl
+[Package requirements ($2) were not met:
+
+$$1_PKG_ERRORS
+
+Consider adjusting the PKG_CONFIG_PATH environment variable if you
+installed software in a non-standard prefix.
+
+_PKG_TEXT
+])],
+		[AC_MSG_RESULT([no])
+                $4])
+elif test $pkg_failed = untried; then
+	ifelse([$4], , [AC_MSG_FAILURE(dnl
+[The pkg-config script could not be found or is too old.  Make sure it
+is in your PATH or set the PKG_CONFIG environment variable to the full
+path to pkg-config.
+
+_PKG_TEXT
+
+To get pkg-config, see .])],
+		[$4])
+else
+	$1[]_CFLAGS=$pkg_cv_[]$1[]_CFLAGS
+	$1[]_LIBS=$pkg_cv_[]$1[]_LIBS
+        AC_MSG_RESULT([yes])
+	ifelse([$3], , :, [$3])
+fi[]dnl
+])# PKG_CHECK_MODULES
diff --git a/common/mangle-tmpl.py b/common/mangle-tmpl.py
new file mode 100644
index 0000000..51ea8c2
--- /dev/null
+++ b/common/mangle-tmpl.py
@@ -0,0 +1,165 @@
+# -*- Mode: Python -*-
+# vi:si:et:sw=4:sts=4:ts=4
+
+"""
+use the output from gst-xmlinspect.py to mangle tmpl/*.sgml and
+insert/overwrite Short Description and Long Description
+"""
+
+# FIXME: right now it uses pygst and scans on its own;
+# we really should use inspect/*.xml instead since the result of
+# gst-xmlinspect.py is committed by the docs maintainer, who can be
+# expected to have pygst, but this step should be done for every docs build,
+# so no pygst allowed
+
+# read in inspect/*.xml
+# for every tmpl/element-(name).xml: mangle with details from element
+
+from __future__ import print_function, unicode_literals
+
+import glob
+import re
+import sys
+import os
+
+class Tmpl:
+    def __init__(self, filename):
+        self.filename = filename
+        self._sectionids = []
+        self._sections = {}
+
+    def read(self):
+        """
+        Read and parse the sections from the given file.
+        """
+        lines = open(self.filename).readlines()
+        matcher = re.compile("\n")
+        id = None
+
+        for line in lines:
+            match = matcher.search(line)
+            if match:
+                id = match.expand("\\1")
+                self._sectionids.append(id)
+                self._sections[id] = []
+            else:
+                if not id:
+                    sys.stderr.write(
+                        "WARNING: line before a SECTION header: %s" % line)
+                else:
+                    self._sections[id].append(line)
+
+    def get_section(self, id):
+        """
+        Get the content from the given section.
+        """
+        return self._sections[id]
+
+    def set_section(self, id, content):
+        """
+        Replace the given section id with the given content.
+        """
+        self._sections[id] = content
+
+    def output(self):
+        """
+        Return the output of the current template in the tmpl/*.sgml format.
+        """
+        lines = []
+        for id in self._sectionids:
+            lines.append("\n" % id)
+            for line in self._sections[id]:
+                lines.append(line)
+
+        return "".join(lines)
+
+    def write(self, backup=False):
+        """
+        Write out the template file again, backing up the previous one.
+        """
+        if backup:
+            target = self.filename + ".mangle.bak"
+            os.rename(self.filename, target)
+
+        handle = open(self.filename, "w")
+        handle.write(self.output())
+        handle.close()
+
+import xml.dom.minidom
+
+def get_elements(file):
+    elements = {}
+    doc = xml.dom.minidom.parse(file)
+
+    elem = None
+    for e in doc.childNodes:
+        if e.nodeType == e.ELEMENT_NODE and e.localName == 'plugin':
+            elem = e
+            break
+    if elem == None:
+        return None
+
+    elem2 = None
+    for e in elem.childNodes:
+        if e.nodeType == e.ELEMENT_NODE and e.localName == 'elements':
+            elem2 = e
+            break
+    if elem2 == None:
+        return None
+
+    elem = elem2
+
+    for e in elem.childNodes:
+        if e.nodeType == e.ELEMENT_NODE and e.localName == 'element':
+            name = None
+            description = None
+
+            for e2 in e.childNodes:
+                if e2.nodeType == e2.ELEMENT_NODE and e2.localName == 'name':
+                    name = e2.childNodes[0].nodeValue.encode("UTF-8")
+                elif e2.nodeType == e2.ELEMENT_NODE and e2.localName == 'description':
+                    if e2.childNodes:
+                      description = e2.childNodes[0].nodeValue.encode("UTF-8")
+                    else:
+                      description = 'No description'
+
+            if name != None and description != None:
+                elements[name] = {'description': description}
+
+    return elements
+
+def main():
+    if not len(sys.argv) == 3:
+        sys.stderr.write('Please specify the inspect/ dir and the tmpl/ dir')
+        sys.exit(1)
+
+    inspectdir = sys.argv[1]
+    tmpldir = sys.argv[2]
+
+    # parse all .xml files; build map of element name -> short desc
+    #for file in glob.glob("inspect/plugin-*.xml"):
+    elements = {}
+    for file in glob.glob("%s/plugin-*.xml" % inspectdir):
+        elements.update(get_elements(file))
+
+    for file in glob.glob("%s/element-*.sgml" % tmpldir):
+        base = os.path.basename(file)
+        element = base[len("element-"):-len(".sgml")]
+        tmpl = Tmpl(file)
+        tmpl.read()
+        if element in elements.keys():
+            description = elements[element]['description']
+            tmpl.set_section("Short_Description", "%s\n\n" % description)
+
+        # put in an include if not yet there
+        line = '' + \
+            '' + \
+            '\n'
+        section = tmpl.get_section("Long_Description")
+        if not section[0]  == line:
+            section.insert(0, line)
+        tmpl.set_section("Long_Description", section)
+        tmpl.write()
+
+main()
diff --git a/common/orc.mak b/common/orc.mak
new file mode 100644
index 0000000..a232b5d
--- /dev/null
+++ b/common/orc.mak
@@ -0,0 +1,76 @@
+#
+# This is a makefile.am fragment to build Orc code.
+#
+# Define ORC_SOURCE and then include this file, such as:
+#
+#  ORC_SOURCE=gstadderorc
+#  include $(top_srcdir)/common/orc.mak
+#
+# This fragment will create tmp-orc.c and gstadderorc.h from
+# gstadderorc.orc.
+#
+# When 'make dist' is run at the top level, or 'make orc-update'
+# in a directory including this fragment, the generated source 
+# files will be copied to $(ORC_SOURCE)-dist.[ch].  These files
+# should be checked in to git, since they are used if Orc is
+# disabled.
+# 
+# Note that this file defines BUILT_SOURCES, so any later usage
+# of BUILT_SOURCES in the Makefile.am that includes this file
+# must use '+='.
+#
+
+
+EXTRA_DIST = $(ORC_SOURCE).orc
+
+ORC_NODIST_SOURCES = tmp-orc.c $(ORC_SOURCE).h
+BUILT_SOURCES = tmp-orc.c $(ORC_SOURCE).h
+
+
+orc-update: tmp-orc.c $(ORC_SOURCE).h
+	$(top_srcdir)/common/gst-indent tmp-orc.c
+	cp tmp-orc.c $(srcdir)/$(ORC_SOURCE)-dist.c
+	cp $(ORC_SOURCE).h $(srcdir)/$(ORC_SOURCE)-dist.h
+
+orcc_v_gen = $(orcc_v_gen_$(V))
+orcc_v_gen_ = $(orcc_v_gen_$(AM_DEFAULT_VERBOSITY))
+orcc_v_gen_0 = @echo "  ORCC   $@";
+
+cp_v_gen = $(cp_v_gen_$(V))
+cp_v_gen_ = $(cp_v_gen_$(AM_DEFAULT_VERBOSITY))
+cp_v_gen_0 = @echo "  CP     $@";
+
+if HAVE_ORCC
+tmp-orc.c: $(srcdir)/$(ORC_SOURCE).orc
+	$(orcc_v_gen)$(ORCC) $(ORCC_FLAGS) --implementation --include glib.h -o tmp-orc.c $(srcdir)/$(ORC_SOURCE).orc
+
+$(ORC_SOURCE).h: $(srcdir)/$(ORC_SOURCE).orc
+	$(orcc_v_gen)$(ORCC) $(ORCC_FLAGS) --header --include glib.h -o $(ORC_SOURCE).h $(srcdir)/$(ORC_SOURCE).orc
+else
+tmp-orc.c: $(srcdir)/$(ORC_SOURCE).orc $(srcdir)/$(ORC_SOURCE)-dist.c
+	$(cp_v_gen)cp $(srcdir)/$(ORC_SOURCE)-dist.c tmp-orc.c
+
+$(ORC_SOURCE).h: $(srcdir)/$(ORC_SOURCE).orc $(srcdir)/$(ORC_SOURCE)-dist.c
+	$(cp_v_gen)cp $(srcdir)/$(ORC_SOURCE)-dist.h $(ORC_SOURCE).h
+endif
+
+clean-local: clean-orc
+.PHONY: clean-orc
+clean-orc:
+	rm -f tmp-orc.c $(ORC_SOURCE).h
+
+dist-hook: dist-hook-orc
+.PHONY: dist-hook-orc
+
+# we try and copy updated orc -dist files below, but don't fail if it
+# doesn't work as the srcdir might not be writable
+dist-hook-orc: tmp-orc.c $(ORC_SOURCE).h
+	$(top_srcdir)/common/gst-indent tmp-orc.c
+	rm -f tmp-orc.c~
+	cmp -s tmp-orc.c $(srcdir)/$(ORC_SOURCE)-dist.c || \
+	  cp tmp-orc.c $(srcdir)/$(ORC_SOURCE)-dist.c || true
+	cmp -s $(ORC_SOURCE).h $(srcdir)/$(ORC_SOURCE)-dist.h || \
+	  cp $(ORC_SOURCE).h $(srcdir)/$(ORC_SOURCE)-dist.h || true
+	cp -p tmp-orc.c $(distdir)/$(ORC_SOURCE)-dist.c
+	cp -p $(ORC_SOURCE).h $(distdir)/$(ORC_SOURCE)-dist.h
+
diff --git a/common/parallel-subdirs.mak b/common/parallel-subdirs.mak
new file mode 100644
index 0000000..36885df
--- /dev/null
+++ b/common/parallel-subdirs.mak
@@ -0,0 +1,13 @@
+# include this at the end of $MODULE/ext/Makefile.am to force make to
+# build subdirectories in parallel when make -jN is used. We will end up
+# descending into all subdirectories a second time, but only after the first
+# (parallel) run has finished, so it should go right through the second time.
+
+.PHONY: independent-subdirs $(SUBDIRS)
+
+independent-subdirs: $(SUBDIRS)
+
+$(SUBDIRS):
+	$(MAKE) -C $@
+
+all-recursive: independent-subdirs
diff --git a/common/plugins.xsl b/common/plugins.xsl
new file mode 100644
index 0000000..60515b6
--- /dev/null
+++ b/common/plugins.xsl
@@ -0,0 +1,209 @@
+ 
+
+
+
+
+
+
+  
+    
+      
+        
+          -plugins-
+          
+        
+      
+      
+        
+      
+    
+    
+    
+    
+
+      
+        
+          Element Information
+          
+
+            
+            
+              plugin
+              
+                
+                  
+                    plugin-
+                    
+                  
+                
+              
+            
+
+            
+              author
+              
+                
+              
+            
+
+            
+              class
+              
+                
+              
+            
+
+           
+         
+
+        
+          Element Pads
+          
+          
+            
+              
+                name
+                
+                  
+                
+              
+
+              
+                direction
+                
+                  
+                
+              
+
+              
+                presence
+                
+                  
+                
+              
+
+              
+                
+                  
+                    details
+                  
+                  
+                    
+                  
+                
+              
+
+             
+
+            
+
+          
+         
+       
+
+    
+  
+
+  
+    
+      -plugins-plugin-
+
+      
+        
+          
+        
+        3
+        FIXME Library
+       
+
+      
+        
+          
+        
+
+        
+          
+            plugin-
+          
+          
+        
+      
+
+      
+        Plugin Information
+        
+
+          
+            filename
+            
+              
+            
+          
+
+          
+            version
+            
+              
+            
+          
+
+          
+            run-time license
+            
+              
+            
+          
+
+          
+            package
+            
+              
+            
+          
+
+          
+            origin
+            
+              
+                
+                
+                  
+                    
+                      
+                      
+                    
+                  
+                  
+                    
+                  
+                
+              
+            
+          
+
+        
+      
+
+      
+        Elements
+        
+        
+          
+        
+      
+
+    
+
+  
+
+  
+  
+
+
diff --git a/common/po.mak b/common/po.mak
new file mode 100644
index 0000000..e019fac
--- /dev/null
+++ b/common/po.mak
@@ -0,0 +1,4 @@
+# rule to download the latest .po files
+download-po: $(top_srcdir)/common/download-translations
+	$(top_srcdir)/common/download-translations $(PACKAGE)
+
diff --git a/common/release.mak b/common/release.mak
new file mode 100644
index 0000000..715657b
--- /dev/null
+++ b/common/release.mak
@@ -0,0 +1,34 @@
+# include this snippet to add a common release: target by using
+# include $(top_srcdir)/common/release.mak
+
+release: dist
+	@$(MAKE) $(PACKAGE)-$(VERSION).tar.xz.sha256sum
+	@echo
+	@echo "================================================================================================="
+	@echo "http://gstreamer.freedesktop.org/src/$(PACKAGE)/$(PACKAGE)-$(VERSION).tar.xz"
+	@cat $(PACKAGE)-$(VERSION).tar.xz.sha256sum
+	@echo "================================================================================================="
+	@if [ -d ~/releases/ ]; then \
+	  cp -v $(PACKAGE)-$(VERSION).tar.xz ~/releases/; \
+	fi
+	@if [ -d ../www/data/src ]; then \
+	  mv -v $(PACKAGE)-$(VERSION).tar.xz ../www/data/src/$(PACKAGE)/ ; \
+	  mv -v $(PACKAGE)-$(VERSION).tar.xz.sha256sum ../www/data/src/$(PACKAGE)/ ; \
+	fi
+	@echo "================================================================================================="
+
+# generate sha256 sum files
+%.sha256sum: %
+	@sha256sum $< > $@
+
+# check that no marshal or enumtypes files are included
+# this in turn ensures that distcheck fails for missing .list files which is currently
+# shadowed when the corresponding .c and .h files are included.
+distcheck-hook:
+	@test "x" = "x`find $(distdir) -name \*-enumtypes.[ch] | grep -v win32`" && \
+	test "x" = "x`find $(distdir) -name \*-marshal.[ch]`" || \
+	( echo "*** Leftover enumtypes or marshal files in the tarball." && \
+	  echo "*** Make sure the following files are not disted:" && \
+	  find $(distdir) -name \*-enumtypes.[ch] | grep -v win32 && \
+	  find $(distdir) -name \*-marshal.[ch] && \
+	  false )
diff --git a/common/scangobj-merge.py b/common/scangobj-merge.py
new file mode 100755
index 0000000..4a9f1fc
--- /dev/null
+++ b/common/scangobj-merge.py
@@ -0,0 +1,311 @@
+#!/usr/bin/python
+# -*- Mode: Python -*-
+# vi:si:et:sw=4:sts=4:ts=4
+
+"""
+parse, merge and write gstdoc-scanobj files
+"""
+
+from __future__ import print_function, unicode_literals
+
+import sys
+import os
+
+def debug(*args):
+    pass
+
+# OrderedDict class based on
+# http://aspn.activestate.com/ASPN/Cookbook/Python/Recipe/107747
+# Licensed under the Python License
+class OrderedDict(dict):
+    def __init__(self):
+        self._keys = []
+        dict.__init__(self)
+
+    def __delitem__(self, key):
+        dict.__delitem__(self, key)
+        self._keys.remove(key)
+
+    def __setitem__(self, key, item):
+        dict.__setitem__(self, key, item)
+        if key not in self._keys: self._keys.append(key)
+
+    def clear(self):
+        dict.clear(self)
+        self._keys = []
+
+    def copy(self):
+        dict = dict.copy(self)
+        dict._keys = self._keys[:]
+        return dict
+
+    def items(self):
+        return zip(self._keys, self.values())
+
+    def keys(self):
+        return self._keys
+
+    def popitem(self):
+        try:
+            key = self._keys[-1]
+        except IndexError:
+            raise KeyError('dictionary is empty')
+
+        val = self[key]
+        del self[key]
+
+        return (key, val)
+
+    def setdefault(self, key, failobj = None):
+        dict.setdefault(self, key, failobj)
+        if key not in self._keys: self._keys.append(key)
+
+    def update(self, dict):
+        dict.update(self, dict)
+        for key in dict.keys():
+            if key not in self._keys: self._keys.append(key)
+
+    def values(self):
+        return map(self.get, self._keys)
+
+class Object:
+    def __init__(self, name):
+        self._signals = OrderedDict()
+        self._args = OrderedDict()
+        self.name = name
+
+    def __repr__(self):
+        return "" % self.name
+
+    def add_signal(self, signal, overwrite=True):
+        if not overwrite and signal.name in self._signals:
+            raise IndexError("signal %s already in %r" % (signal.name, self))
+        self._signals[signal.name] = signal
+
+    def add_arg(self, arg, overwrite=True):
+        if not overwrite and arg.name in self._args:
+            raise IndexError("arg %s already in %r" % (arg.name, self))
+        self._args[arg.name] = arg
+
+class Docable:
+    def __init__(self, **kwargs):
+        for key in self.attrs:
+            setattr(self, key, kwargs[key])
+        self.dict = kwargs
+
+    def __repr__(self):
+        return "<%r %s>" % (str(self.__class__), self.name)
+
+class Signal(Docable):
+    attrs = ['name', 'returns', 'args']
+
+class Arg(Docable):
+    attrs = ['name', 'type', 'range', 'flags', 'nick', 'blurb', 'default']
+
+class GDoc:
+    def load_file(self, filename):
+        try:
+            lines = open(filename).readlines()
+            self.load_data("".join(lines))
+        except IOError:
+            print ("WARNING - could not read from %s" % filename)
+
+    def save_file(self, filename, backup=False):
+        """
+        Save the information to the given file if the file content changed.
+        """
+        olddata = None
+        try:
+            lines = open(filename).readlines()
+            olddata = "".join(lines)
+        except IOError:
+            print ("WARNING - could not read from %s" % filename)
+        newdata = self.get_data()
+        if olddata and olddata == newdata:
+            return
+
+        if olddata:
+            if backup:
+                os.rename(filename, filename + '.bak')
+
+        handle = open(filename, "w")
+        handle.write(newdata)
+        handle.close()
+
+class Signals(GDoc):
+    def __init__(self):
+        self._objects = OrderedDict()
+
+    def load_data(self, data):
+        """
+        Load the .signals lines, creating our list of objects and signals.
+        """
+        import re
+        smatcher = re.compile(
+            '(?s)'                                      # make . match \n
+            '\n(.*?)\n'
+            )
+        nmatcher = re.compile(
+            ''
+            '(?P\S*)'                           # store object
+            '::'
+            '(?P\S*)'                           # store signal
+            ''
+        )
+        rmatcher = re.compile(
+            '(?s)'                                      # make . match \n
+            '(?P\S*)\n'     # store returns
+            '(?P.*)'                              # store args
+        )
+        for block in smatcher.findall(data):
+            nmatch = nmatcher.search(block)
+            if nmatch:
+                o = nmatch.group('object')
+                debug("Found object", o)
+                debug("Found signal", nmatch.group('signal'))
+                if o not in self._objects:
+                    object = Object(o)
+                    self._objects[o] = object
+
+                rmatch = rmatcher.search(block)
+                if rmatch:
+                    dict = rmatch.groupdict().copy()
+                    dict['name'] = nmatch.group('signal')
+                    signal = Signal(**dict)
+                    self._objects[o].add_signal(signal)
+
+    def get_data(self):
+        lines = []
+        for o in self._objects.values():
+            for s in o._signals.values():
+                block = """
+%(object)s::%(name)s
+%(returns)s
+%(args)s
+"""
+                d = s.dict.copy()
+                d['object'] = o.name
+                lines.append(block % d)
+
+        return "\n".join(lines) + '\n'
+
+class Args(GDoc):
+    def __init__(self):
+        self._objects = OrderedDict()
+
+    def load_data(self, data):
+        """
+        Load the .args lines, creating our list of objects and args.
+        """
+        import re
+        amatcher = re.compile(
+            '(?s)'                                      # make . match \n
+            '\n(.*?)\n'
+            )
+        nmatcher = re.compile(
+            ''
+            '(?P\S*)'                           # store object
+            '::'
+            '(?P\S*)'                              # store arg
+            ''
+        )
+        rmatcher = re.compile(
+            '(?s)'                                      # make . match \n
+            '(?P\S*)\n'              # store type
+            '(?P.*?)\n'           # store range
+            '(?P\S*)\n'           # store flags
+            '(?P.*?)\n'              # store nick
+            '(?P.*?)\n'           # store blurb
+            '(?P.*?)\n'     # store default
+        )
+        for block in amatcher.findall(data):
+            nmatch = nmatcher.search(block)
+            if nmatch:
+                o = nmatch.group('object')
+                debug("Found object", o)
+                debug("Found arg", nmatch.group('arg'))
+                if o not in self._objects:
+                    object = Object(o)
+                    self._objects[o] = object
+
+                rmatch = rmatcher.search(block)
+                if rmatch:
+                    dict = rmatch.groupdict().copy()
+                    dict['name'] = nmatch.group('arg')
+                    arg = Arg(**dict)
+                    self._objects[o].add_arg(arg)
+                else:
+                    print ("ERROR: could not match arg from block %s" % block)
+
+    def get_data(self):
+        lines = []
+        for o in self._objects.values():
+            for a in o._args.values():
+                block = """
+%(object)s::%(name)s
+%(type)s
+%(range)s
+%(flags)s
+%(nick)s
+%(blurb)s
+%(default)s
+
+"""
+                d = a.dict.copy()
+                d['object'] = o.name
+                lines.append(block % d)
+
+        return "\n".join(lines) + '\n'
+
+class SingleLine(GDoc):
+    def __init__(self):
+        self._objects = []
+
+    def load_data(self, data):
+        """
+        Load the .interfaces/.prerequisites lines, merge duplicates
+        """
+        # split data on '\n'
+        lines = data.splitlines();
+        # merge them into self._objects
+        for line in lines:
+            if line not in self._objects:
+                self._objects.append(line)
+
+    def get_data(self):
+        lines = sorted(self._objects)
+        return "\n".join(lines) + '\n'
+
+def main(argv):
+    modulename = None
+    try:
+        modulename = argv[1]
+    except IndexError:
+        sys.stderr.write('Please provide a documentation module name\n')
+        sys.exit(1)
+
+    signals = Signals()
+    signals.load_file(modulename + '.signals')
+    signals.load_file(modulename + '.signals.new')
+    signals.save_file(modulename + '.signals', backup=True)
+    os.unlink(modulename + '.signals.new')
+
+    args = Args()
+    args.load_file(modulename + '.args')
+    args.load_file(modulename + '.args.new')
+    args.save_file(modulename + '.args', backup=True)
+    os.unlink(modulename + '.args.new')
+
+    ifaces = SingleLine()
+    ifaces.load_file(modulename + '.interfaces')
+    ifaces.load_file(modulename + '.interfaces.new')
+    ifaces.save_file(modulename + '.interfaces', backup=True)
+    os.unlink(modulename + '.interfaces.new')
+
+    prereq = SingleLine()
+    prereq.load_file(modulename + '.prerequisites')
+    prereq.load_file(modulename + '.prerequisites.new')
+    prereq.save_file(modulename + '.prerequisites', backup=True)
+    os.unlink(modulename + '.prerequisites.new')
+
+main(sys.argv)
diff --git a/common/upload-doc.mak b/common/upload-doc.mak
new file mode 100644
index 0000000..c1c4945
--- /dev/null
+++ b/common/upload-doc.mak
@@ -0,0 +1,64 @@
+# this snippet is to be included by both our docbook manuals
+# and gtk-doc API references
+
+# it adds an upload target to each of these dir's Makefiles
+
+# each Makefile.am should define the following variables:
+# - DOC: the base name of the documentation
+#        (faq, manual, pwg, gstreamer, gstreamer-libs)
+# - FORMATS: the formats in which DOC is output
+#            (html ps pdf)
+
+# if you want to use it, make sure your $HOME/.ssh/config file contains the
+# correct User entry for the Host entry for the DOC_SERVER
+
+# these variables define the location of the online docs
+DOC_SERVER = gstreamer.freedesktop.org
+DOC_BASE = /srv/gstreamer.freedesktop.org/www/data/doc
+DOC_URL = $(DOC_SERVER):$(DOC_BASE)
+
+upload: $(FORMATS)
+	@if echo $(FORMATS) | grep html > /dev/null; then \
+	  echo "Preparing docs for upload (rebasing cross-references) ..." ; \
+	  if test x$(builddir) != x$(srcdir); then \
+	    echo "make upload can only be used if srcdir == builddir"; \
+	    exit 1; \
+	  fi; \
+	  # gtkdoc-rebase sometimes gets confused, so reset everything to \
+	  # local links before rebasing to online links                   \
+	  gtkdoc-rebase --html-dir=$(builddir)/html 2>/dev/null 2>/dev/null ; \
+	  rebase=`gtkdoc-rebase --verbose --online --html-dir=$(builddir)/html` ; \
+	  echo "$$rebase" | grep -e "On-*line"; \
+	  for req in glib gobject gstreamer gstreamer-libs gst-plugins-base-libs; do \
+	    if ! ( echo "$$rebase" | grep -i -e "On-*line.*/$$req/" ); then \
+	      echo "===============================================================================" ; \
+	      echo " Could not determine online location for $$req docs. Cross-referencing will be " ; \
+	      echo " broken, so not uploading. Make sure the library's gtk-doc documentation is    " ; \
+	      echo " installed somewhere in /usr/share/gtk-doc.                                    " ; \
+	      echo "===============================================================================" ; \
+	      exit 1; \
+	    fi; \
+	  done; \
+	  export SRC="$$SRC html"; \
+	fi; \
+	if echo $(FORMATS) | grep ps > /dev/null; then export SRC="$$SRC $(DOC).ps"; fi; \
+	if echo $(FORMATS) | grep pdf > /dev/null; then export SRC="$$SRC $(DOC).pdf"; fi; \
+	\
+	# upload releases to both X.Y/ and head/ subdirectories \
+	export DIR=$(DOC_BASE)/gstreamer/$(PACKAGE_VERSION_MAJOR).$(PACKAGE_VERSION_MINOR)/$(DOC); \
+	echo Uploading $$SRC to $(DOC_SERVER):$$DIR; \
+	ssh $(DOC_SERVER) mkdir -p $$DIR; \
+	rsync -rv -e ssh --delete $$SRC $(DOC_SERVER):$$DIR; \
+	ssh $(DOC_SERVER) chmod -R g+w $$DIR; \
+	\
+	export DIR=$(DOC_BASE)/gstreamer/head/$(DOC); \
+	echo Uploading $$SRC to $(DOC_SERVER):$$DIR; \
+	ssh $(DOC_SERVER) mkdir -p $$DIR; \
+	rsync -rv -e ssh --delete $$SRC $(DOC_SERVER):$$DIR; \
+	ssh $(DOC_SERVER) chmod -R g+w $$DIR; \
+	\
+	if echo $(FORMATS) | grep html > /dev/null; then \
+	  echo "Un-preparing docs for upload (rebasing cross-references) ..." ; \
+	  gtkdoc-rebase --html-dir=$(builddir)/html ; \
+	fi; \
+	echo Done
diff --git a/common/win32.mak b/common/win32.mak
new file mode 100644
index 0000000..30e347e
--- /dev/null
+++ b/common/win32.mak
@@ -0,0 +1,78 @@
+# various tests to make sure we dist the win32 stuff (for MSVC builds) right
+
+# the MANIFEST contains all win32 related files that should be disted
+win32 = $(shell cat $(top_srcdir)/win32/MANIFEST)
+
+# wildcard is apparently not portable to other makes, hence the use of find
+# these are library .def files with the symbols to export
+win32defs = $(shell find $(top_srcdir)/win32/common -name '*.def')
+
+# wildcard is apparently not portable to other makes, hence the use of find
+# these are files that need to be disted with CRLF line endings:
+win32crlf = $(shell find $(top_srcdir)/win32 -name '*.dsw' -o -name '*.dsp')
+
+win32-debug:
+	@echo; \
+	echo win32     = $(win32); \
+	echo; \
+	echo win32defs = $(win32defs); \
+	echo; \
+	echo win32crlf = $(win32crlf); \
+	echo
+
+win32-check-crlf:
+	@echo Checking win32 files for CR LF line endings ...; \
+	fail=0 ; \
+	for each in $(win32crlf) ; do \
+	  result=`perl -e 'print grep(/\r\n/,<>)' "$$each" | wc -l`; \
+	  if test "$$result" = 0 ; then \
+	    echo $$each must be fixed to have CRLF line endings ; \
+	    fail=1; \
+	  fi ; \
+	done ; \
+	exit $$fail
+
+# make sure all symbols we export on linux are defined in the win32 .def too
+# (don't care about other unixes for now, it's enough if it works on one of
+# the linux build bots; we assume .so )
+check-exports:
+	@fail=0 ; \
+	for l in $(win32defs); do \
+	  libbase=`basename "$$l" ".def"`; \
+	  libso=`find "$(top_builddir)" -name "$$libbase-@GST_API_VERSION@.so" | grep -v /_build/ | head -n1`; \
+	  libdef="$(top_srcdir)/win32/common/$$libbase.def"; \
+	  if test "x$$libso" != "x"; then \
+	    echo Checking symbols in $$libso; \
+	    if ! ($(top_srcdir)/common/check-exports $$libdef $$libso) ; then \
+	      fail=1; \
+	    fi; \
+	  fi; \
+	done ; \
+	if test $$fail != 0; then \
+	  echo '-----------------------------------------------------------'; \
+	  echo 'Run this to update the .def files:'; \
+	  echo 'make update-exports'; \
+	  echo '-----------------------------------------------------------'; \
+	fi; \
+	exit $$fail
+
+update-exports:
+	make check-exports 2>&1 | patch -p1
+	git add win32/common/libgst*.def
+	git diff --cached -- win32/common/
+	echo '^^^--- updated and staged changes above'
+
+# complain about nonportable printf format strings (%lld, %llu, %zu etc.)
+check-nonportable-print-format:
+	@fail=0 ; \
+	loc=`find "$(top_srcdir)" -name '*.c' | xargs grep -n -e '%[0-9]*ll[udx]' -e '%[0-9]*z[udx]'`; \
+	if test "x$$loc" != "x"; then \
+	  echo "Please fix the following print format strings:" ; \
+	  find "$(top_srcdir)" -name '*.c' | xargs grep -n -e '%[0-9]*ll[udx]' -e '%[0-9]*z[udx]'; \
+	  fail=1; \
+	fi; \
+	exit $$fail
+
+dist-hook: check-exports win32-check-crlf
+
+
diff --git a/configure.ac b/configure.ac
new file mode 100644
index 0000000..7a65c6f
--- /dev/null
+++ b/configure.ac
@@ -0,0 +1,344 @@
+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], [1.4.5],
+    [http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],
+    [gst-rtsp-server])
+AG_GST_INIT
+
+dnl initialize automake
+AM_INIT_AUTOMAKE([-Wno-portability 1.11 no-dist-gzip dist-xz tar-ustar subdir-objects])
+
+dnl define PACKAGE_VERSION_* variables
+AS_VERSION
+
+dnl check if this is a release version
+AS_NANO(GST_GIT="no", GST_GIT="yes")
+
+dnl can autoconf find the source ?
+AC_CONFIG_SRCDIR([gst/rtsp-server/rtsp-server.c])
+
+dnl define the output header for config
+AC_CONFIG_HEADERS([config.h])
+
+dnl AM_MAINTAINER_MODE only provides the option to configure to enable it
+AM_MAINTAINER_MODE([enable])
+
+dnl sets host_* variables
+AC_CANONICAL_HOST
+
+dnl use pretty build output with automake >= 1.11
+m4_ifdef([AM_SILENT_RULES],[AM_SILENT_RULES([yes])],
+  [AM_DEFAULT_VERBOSITY=1
+   AC_SUBST(AM_DEFAULT_VERBOSITY)])
+
+dnl our libraries and install dirs use major.minor as a version
+dnl GST_API_VERSION=$PACKAGE_VERSION_MAJOR.$PACKAGE_VERSION_MINOR
+dnl we override it here if we need to for the release candidate of new series
+GST_API_VERSION=1.0
+AC_SUBST(GST_API_VERSION)
+
+dnl CURRENT, REVISION, AGE
+dnl - library source changed -> increment REVISION
+dnl - interfaces added/removed/changed -> increment CURRENT, REVISION = 0
+dnl - interfaces added -> increment AGE
+dnl - interfaces removed -> AGE = 0
+dnl
+dnl Keep CURRENT as MINOR * 100 + MICRO
+dnl Ex : 1.0.0 => 0
+dnl      1.0.3 => 3
+dnl      1.1.0 => 100
+dnl      1.2.5 => 205
+dnl      1.10.9 (who knows) => 1009
+dnl
+dnl sets GST_LT_LDFLAGS
+AS_LIBTOOL(GST, 405, 0, 405)
+
+dnl *** required versions of GStreamer stuff ***
+GST_REQ=1.4.0
+GSTPB_REQ=1.4.0
+GSTPG_REQ=1.4.0
+GSTPD_REQ=1.4.0
+
+dnl *** autotools stuff ****
+
+dnl allow for different autotools
+AS_AUTOTOOLS_ALTERNATE
+
+dnl Add parameters for aclocal
+AC_SUBST(ACLOCAL_AMFLAGS, "-I m4 -I common/m4")
+AC_CONFIG_MACRO_DIR([m4])
+
+dnl set up gettext
+dnl the version check needs to stay here because autopoint greps for it
+#AM_GNU_GETTEXT_VERSION([0.17])
+#AM_GNU_GETTEXT([external])
+#AG_GST_GETTEXT([gst-rtsp-server-$GST_API_VERSION])
+
+dnl *** check for arguments to configure ***
+
+AG_GST_ARG_DISABLE_FATAL_WARNINGS
+
+AG_GST_ARG_DEBUG
+AG_GST_ARG_VALGRIND
+AG_GST_ARG_GCOV
+AG_GST_ARG_WITH_PKG_CONFIG_PATH
+AG_GST_ARG_WITH_PACKAGE_NAME
+AG_GST_ARG_WITH_PACKAGE_ORIGIN
+
+AG_GST_PKG_CONFIG_PATH
+
+AG_GST_SET_PACKAGE_RELEASE_DATETIME_WITH_NANO([$PACKAGE_VERSION_NANO],
+  ["${srcdir}/gst-rtsp-server.doap"],
+  [$PACKAGE_VERSION_MAJOR.$PACKAGE_VERSION_MINOR.$PACKAGE_VERSION_MICRO])
+
+dnl building of tests
+AC_ARG_ENABLE(tests,
+  AS_HELP_STRING([--disable-tests],[disable building test apps]),
+  [
+    case "${enableval}" in
+      yes) BUILD_TESTS=yes ;;
+      no)  BUILD_TESTS=no ;;
+      *)   AC_MSG_ERROR(bad value ${enableval} for --disable-tests) ;;
+    esac
+  ],
+[BUILD_TESTS=yes]) dnl Default value
+AM_CONDITIONAL(BUILD_TESTS,         test "x$BUILD_TESTS" = "xyes")
+
+dnl *** checks for platform ***
+
+dnl * hardware/architecture *
+
+dnl *** checks for programs ***
+
+dnl find a compiler
+AC_PROG_CC
+AC_PROG_CC_STDC
+
+dnl check if the compiler supports '-c' and '-o' options
+AM_PROG_CC_C_O
+
+dnl find an assembler
+AM_PROG_AS
+
+AC_PATH_PROG(VALGRIND_PATH, valgrind, no)
+AM_CONDITIONAL(HAVE_VALGRIND, test ! "x$VALGRIND_PATH" = "xno")
+
+dnl check for gobject-introspection
+GOBJECT_INTROSPECTION_CHECK([1.31.1])
+
+dnl check for documentation tools
+AG_GST_DOCBOOK_CHECK
+GTK_DOC_CHECK([1.12])
+
+dnl *** checks for libraries ***
+
+dnl *** checks for header files ***
+
+dnl *** checks for types/defines ***
+
+dnl *** checks for structures ***
+
+dnl *** checks for compiler characteristics ***
+
+dnl *** checks for library functions ***
+
+dnl *** checks for dependancy libraries ***
+
+dnl GLib is required
+GLIB_REQ=2.32.0
+AC_SUBST([GLIB_REQ])
+AG_GST_GLIB_CHECK([$GLIB_REQ])
+
+dnl checks for gstreamer
+dnl uninstalled is selected preferentially -- see pkg-config(1)
+AG_GST_CHECK_GST($GST_API_VERSION, [$GST_REQ], [yes])
+
+GST_TOOLS_DIR=`$PKG_CONFIG --variable=toolsdir gstreamer-$GST_API_VERSION`
+if test -z $GST_TOOLS_DIR; then
+  AC_MSG_ERROR([no tools dir defined in GStreamer pkg-config file; core upgrade needed.])
+fi
+AC_SUBST(GST_TOOLS_DIR)
+
+GST_PLUGINS_DIR=`$PKG_CONFIG gstreamer-$GST_API_VERSION --variable pluginsdir`
+AC_SUBST(GST_PLUGINS_DIR)
+AC_MSG_NOTICE(Using GStreamer Core Plugins in $GST_PLUGINS_DIR)
+
+AG_GST_CHECK_GST_BASE($GST_API_VERSION, [$GST_REQ], [yes])
+
+AG_GST_CHECK_GST_PLUGINS_BASE($GST_API_VERSION, [$GSTPB_REQ], [yes])
+GSTPB_PLUGINS_DIR=`$PKG_CONFIG gstreamer-plugins-base-$GST_API_VERSION --variable pluginsdir`
+AC_SUBST(GSTPB_PLUGINS_DIR)
+AC_MSG_NOTICE(Using GStreamer Base Plugins in $GSTPB_PLUGINS_DIR)
+
+AG_GST_CHECK_GST_PLUGINS_GOOD($GST_API_VERSION, [$GSTPG_REQ], [yes])
+GSTPG_PLUGINS_DIR=`$PKG_CONFIG gstreamer-plugins-good-$GST_API_VERSION --variable pluginsdir`
+AC_SUBST(GSTPG_PLUGINS_DIR)
+AC_MSG_NOTICE(Using GStreamer Good Plugins in $GSTPG_PLUGINS_DIR)
+
+AG_GST_CHECK_GST_PLUGINS_BAD($GST_API_VERSION, [$GSTPD_REQ], [yes])
+GSTPD_PLUGINS_DIR=`$PKG_CONFIG gstreamer-plugins-bad-$GST_API_VERSION --variable pluginsdir`
+AC_SUBST(GSTPD_PLUGINS_DIR)
+AC_MSG_NOTICE(Using GStreamer Bad Plugins in $GSTPD_PLUGINS_DIR)
+
+AG_GST_CHECK_GST_CHECK($GST_API_VERSION, [$GST_REQ], no)
+AM_CONDITIONAL(HAVE_CHECK, test "x$HAVE_GST_CHECK" = "xyes")
+
+dnl Check for -Bsymbolic-functions linker flag used to avoid
+dnl intra-library PLT jumps, if available.
+AC_ARG_ENABLE(Bsymbolic,
+              [AS_HELP_STRING([--disable-Bsymbolic],[avoid linking with -Bsymbolic])],,
+              [SAVED_LDFLAGS="${LDFLAGS}"
+               AC_MSG_CHECKING([for -Bsymbolic-functions linker flag])
+               LDFLAGS=-Wl,-Bsymbolic-functions
+               AC_LINK_IFELSE([AC_LANG_PROGRAM([[]], [[int main (void) { return 0; }]])],[
+                           AC_MSG_RESULT(yes)
+                           enable_Bsymbolic=yes],[
+                           AC_MSG_RESULT(no)
+                           enable_Bsymbolic=no])
+               LDFLAGS="${SAVED_LDFLAGS}"])
+
+dnl *** set variables based on configure arguments ***
+
+dnl set license and copyright notice
+GST_LICENSE="LGPL"
+AC_DEFINE_UNQUOTED(GST_LICENSE, "$GST_LICENSE", [GStreamer license])
+AC_SUBST(GST_LICENSE)
+
+dnl set location of plugin directory
+AG_GST_SET_PLUGINDIR
+
+dnl set release date/time (and check that release version is in doap file)
+AG_GST_SET_PACKAGE_RELEASE_DATETIME_WITH_NANO([$PACKAGE_VERSION_NANO],
+  ["${srcdir}/gst-rtsp-server.doap"],
+  [$PACKAGE_VERSION_MAJOR.$PACKAGE_VERSION_MINOR.$PACKAGE_VERSION_MICRO])
+
+# set by AG_GST_PARSE_SUBSYSTEM_DISABLES above
+dnl make sure it doesn't complain about unused variables if debugging is disabled
+NO_WARNINGS=""
+AG_GST_CHECK_GST_DEBUG_DISABLED([NO_WARNINGS="-Wno-unused"], [NO_WARNINGS=""])
+
+dnl define an ERROR_CFLAGS Makefile variable
+AG_GST_SET_ERROR_CFLAGS($FATAL_WARNINGS, [-Wmissing-declarations -Wmissing-prototypes -Wredundant-decls -Wundef -Wwrite-strings -Wformat-nonliteral -Wformat-security -Wold-style-definition -Winit-self -Wmissing-include-dirs -Waddress -Waggregate-return -Wno-multichar -Wnested-externs $NO_WARNINGS])
+
+dnl define correct level for debugging messages
+AG_GST_SET_LEVEL_DEFAULT($GST_GIT)
+
+dnl used in examples
+AG_GST_DEFAULT_ELEMENTS
+
+dnl *** finalize CFLAGS, LDFLAGS, LIBS
+
+dnl Overview:
+dnl GST_OPTION_CFLAGS:  common flags for profiling, debugging, errors, ...
+dnl GST_*:              flags shared by built objects to link against GStreamer
+dnl GST_ALL_LDFLAGS:    linker flags shared by all
+dnl GST_LIB_LDFLAGS:    additional linker flags for all libaries
+dnl GST_LT_LDFLAGS:     library versioning of our libraries
+dnl GST_PLUGIN_LDFLAGS: flags to be used for all plugins
+
+dnl GST_OPTION_CFLAGS
+if test "x$USE_DEBUG" = xyes; then
+   PROFILE_CFLAGS="-g"
+fi
+AC_SUBST(PROFILE_CFLAGS)
+
+# GST_DISABLE_DEPRECATED: hide the visibility of deprecated
+# functionality from the API that gstreamer uses
+# GST_REMOVE_DEPRECATED: don't compile deprecated functionality (breaks ABI)
+if test "x$PACKAGE_VERSION_NANO" = "x1"; then
+  dnl Define _only_ when compiling from git (not for pre-releases or releases)
+  DEPRECATED_CFLAGS="-DGST_DISABLE_DEPRECATED"
+else
+  DEPRECATED_CFLAGS=""
+fi
+AC_SUBST(DEPRECATED_CFLAGS)
+
+dnl every flag in GST_OPTION_CFLAGS can be overridden at make time
+GST_OPTION_CFLAGS="\$(WARNING_CFLAGS) \$(ERROR_CFLAGS) \$(DEBUG_CFLAGS) \$(PROFILE_CFLAGS) \$(GCOV_CFLAGS) \$(OPT_CFLAGS) \$(DEPRECATED_CFLAGS)"
+AC_SUBST(GST_OPTION_CFLAGS)
+
+dnl FIXME: do we want to rename to GST_ALL_* ?
+dnl prefer internal headers to already installed ones
+dnl add GST_OPTION_CFLAGS, but overridable
+GST_CFLAGS="$GST_CFLAGS \$(GST_OPTION_CFLAGS)"
+AC_SUBST(GST_CFLAGS)
+AC_SUBST(GST_LIBS)
+
+dnl GST_ALL_*
+dnl vars common to for all internal objects (core libs, elements, applications)
+dnl CFLAGS:
+dnl - src and build dirs need to be added because every piece that gets built
+dnl   will need the GStreamer source and generated headers
+GST_ALL_CFLAGS="-I\$(top_srcdir) -I\$(top_builddir) $GST_PLUGINS_BASE_CFLAGS $GST_CFLAGS \$(GST_OPTION_CFLAGS)"
+AC_SUBST([GST_ALL_CFLAGS])
+
+dnl FIXME: check if LTLIBINTL is needed everywhere
+dnl I presume it is given that it contains the symbols that _() stuff maps to
+GST_ALL_LIBS="$GST_LIBS $LTLIBINTL \$(GCOV_LIBS)"
+AC_SUBST([GST_ALL_LIBS])
+
+dnl LDFLAGS really should only contain flags, not libs - they get added before
+dnl whatevertarget_LIBS and -L flags here affect the rest of the linking
+GST_ALL_LDFLAGS="-no-undefined"
+if test "x${enable_Bsymbolic}" = "xyes"; then
+  GST_ALL_LDFLAGS="$GST_ALL_LDFLAGS -Wl,-Bsymbolic-functions"
+fi
+
+AC_SUBST(GST_ALL_LDFLAGS)
+
+dnl GST_LIB_LDFLAGS
+dnl linker flags shared by all libraries
+dnl LDFLAGS modifier defining exported symbols from built libraries
+GST_LIB_LDFLAGS="-export-symbols-regex \^[_]?\(gst_\|Gst\|GST_\).*"
+AC_SUBST(GST_LIB_LDFLAGS)
+
+dnl GST_OBJ_*
+dnl default vars for all internal objects built on libgstrtspserver
+dnl includes GST_ALL_*
+GST_OBJ_CFLAGS="\$(GST_ALL_CFLAGS)"
+AC_SUBST([GST_OBJ_CFLAGS])
+GST_OBJ_LIBS="\$(top_builddir)/gst/rtsp-server/libgstrtspserver-$GST_API_VERSION.la \$(GST_ALL_LIBS)"
+AC_SUBST([GST_OBJ_LIBS])
+
+PKG_CHECK_MODULES(LIBCGROUP, libcgroup >= 0.26, HAVE_LIBCGROUP="yes", HAVE_LIBCGROUP="no")
+AC_SUBST(LIBCGROUP_CFLAGS)
+AC_SUBST(LIBCGROUP_LIBS)
+AM_CONDITIONAL(HAVE_LIBCGROUP, test "x$HAVE_LIBCGROUP" = "xyes")
+
+dnl this really should only contain flags, not libs - they get added before
+dnl whatevertarget_LIBS and -L flags here affect the rest of the linking
+
+dnl *** output files ***
+
+dnl keep this alphabetic per directory, please
+AC_CONFIG_FILES([
+Makefile
+common/Makefile
+common/m4/Makefile
+gst/Makefile
+gst/rtsp-server/Makefile
+examples/Makefile
+tests/Makefile
+tests/check/Makefile
+pkgconfig/Makefile
+pkgconfig/gstreamer-rtsp-server.pc
+pkgconfig/gstreamer-rtsp-server-uninstalled.pc
+docs/Makefile
+docs/version.entities
+docs/libs/Makefile
+])
+AC_OUTPUT
+
+echo "
+
+Configuration
+	Version                    : ${VERSION}
+	Source code location       : ${srcdir}
+	Prefix                     : ${prefix}
+	Compiler                   : ${CC}
+	CGroups example            : ${HAVE_LIBCGROUP}
+
+gst-rtsp-server configured. Type 'make' to build.
+"
diff --git a/docs/Makefile.am b/docs/Makefile.am
new file mode 100644
index 0000000..9f9f150
--- /dev/null
+++ b/docs/Makefile.am
@@ -0,0 +1,20 @@
+if ENABLE_GTK_DOC
+DOCS_SUBDIRS = libs
+else
+DOCS_SUBDIRS = 
+endif
+
+SUBDIRS = $(DOCS_SUBDIRS)
+DIST_SUBDIRS = libs
+
+EXTRA_DIST = \
+	version.entities.in
+
+upload:
+	@if test "x$(SUBDIRS)" != x; then \
+	  for a in $(SUBDIRS); do \
+	    cd $$a; \
+	    make upload; \
+	    cd ..; \
+	  done; \
+	fi
diff --git a/docs/Makefile.in b/docs/Makefile.in
new file mode 100644
index 0000000..3aa1228
--- /dev/null
+++ b/docs/Makefile.in
@@ -0,0 +1,751 @@
+# Makefile.in generated by automake 1.14.1 from Makefile.am.
+# @configure_input@
+
+# Copyright (C) 1994-2013 Free Software Foundation, Inc.
+
+# This Makefile.in is free software; the Free Software Foundation
+# gives unlimited permission to copy and/or distribute it,
+# with or without modifications, as long as this notice is preserved.
+
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
+# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
+# PARTICULAR PURPOSE.
+
+@SET_MAKE@
+VPATH = @srcdir@
+am__is_gnu_make = test -n '$(MAKEFILE_LIST)' && test -n '$(MAKELEVEL)'
+am__make_running_with_option = \
+  case $${target_option-} in \
+      ?) ;; \
+      *) echo "am__make_running_with_option: internal error: invalid" \
+              "target option '$${target_option-}' specified" >&2; \
+         exit 1;; \
+  esac; \
+  has_opt=no; \
+  sane_makeflags=$$MAKEFLAGS; \
+  if $(am__is_gnu_make); then \
+    sane_makeflags=$$MFLAGS; \
+  else \
+    case $$MAKEFLAGS in \
+      *\\[\ \	]*) \
+        bs=\\; \
+        sane_makeflags=`printf '%s\n' "$$MAKEFLAGS" \
+          | sed "s/$$bs$$bs[$$bs $$bs	]*//g"`;; \
+    esac; \
+  fi; \
+  skip_next=no; \
+  strip_trailopt () \
+  { \
+    flg=`printf '%s\n' "$$flg" | sed "s/$$1.*$$//"`; \
+  }; \
+  for flg in $$sane_makeflags; do \
+    test $$skip_next = yes && { skip_next=no; continue; }; \
+    case $$flg in \
+      *=*|--*) continue;; \
+        -*I) strip_trailopt 'I'; skip_next=yes;; \
+      -*I?*) strip_trailopt 'I';; \
+        -*O) strip_trailopt 'O'; skip_next=yes;; \
+      -*O?*) strip_trailopt 'O';; \
+        -*l) strip_trailopt 'l'; skip_next=yes;; \
+      -*l?*) strip_trailopt 'l';; \
+      -[dEDm]) skip_next=yes;; \
+      -[JT]) skip_next=yes;; \
+    esac; \
+    case $$flg in \
+      *$$target_option*) has_opt=yes; break;; \
+    esac; \
+  done; \
+  test $$has_opt = yes
+am__make_dryrun = (target_option=n; $(am__make_running_with_option))
+am__make_keepgoing = (target_option=k; $(am__make_running_with_option))
+pkgdatadir = $(datadir)/@PACKAGE@
+pkgincludedir = $(includedir)/@PACKAGE@
+pkglibdir = $(libdir)/@PACKAGE@
+pkglibexecdir = $(libexecdir)/@PACKAGE@
+am__cd = CDPATH="$${ZSH_VERSION+.}$(PATH_SEPARATOR)" && cd
+install_sh_DATA = $(install_sh) -c -m 644
+install_sh_PROGRAM = $(install_sh) -c
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+INSTALL_HEADER = $(INSTALL_DATA)
+transform = $(program_transform_name)
+NORMAL_INSTALL = :
+PRE_INSTALL = :
+POST_INSTALL = :
+NORMAL_UNINSTALL = :
+PRE_UNINSTALL = :
+POST_UNINSTALL = :
+build_triplet = @build@
+host_triplet = @host@
+target_triplet = @target@
+subdir = docs
+DIST_COMMON = $(srcdir)/Makefile.in $(srcdir)/Makefile.am \
+	$(srcdir)/version.entities.in README
+ACLOCAL_M4 = $(top_srcdir)/aclocal.m4
+am__aclocal_m4_deps = $(top_srcdir)/common/m4/as-ac-expand.m4 \
+	$(top_srcdir)/common/m4/as-auto-alt.m4 \
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+	$(top_srcdir)/common/m4/as-docbook.m4 \
+	$(top_srcdir)/common/m4/as-libtool.m4 \
+	$(top_srcdir)/common/m4/as-version.m4 \
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+	$(top_srcdir)/common/m4/gst-check.m4 \
+	$(top_srcdir)/common/m4/gst-default.m4 \
+	$(top_srcdir)/common/m4/gst-doc.m4 \
+	$(top_srcdir)/common/m4/gst-error.m4 \
+	$(top_srcdir)/common/m4/gst-feature.m4 \
+	$(top_srcdir)/common/m4/gst-glib2.m4 \
+	$(top_srcdir)/common/m4/gst-package-release-datetime.m4 \
+	$(top_srcdir)/common/m4/gst-plugindir.m4 \
+	$(top_srcdir)/common/m4/gst.m4 \
+	$(top_srcdir)/common/m4/gtk-doc.m4 \
+	$(top_srcdir)/common/m4/introspection.m4 \
+	$(top_srcdir)/common/m4/pkg.m4 $(top_srcdir)/m4/libtool.m4 \
+	$(top_srcdir)/m4/ltoptions.m4 $(top_srcdir)/m4/ltsugar.m4 \
+	$(top_srcdir)/m4/ltversion.m4 $(top_srcdir)/m4/lt~obsolete.m4 \
+	$(top_srcdir)/acinclude.m4 $(top_srcdir)/configure.ac
+am__configure_deps = $(am__aclocal_m4_deps) $(CONFIGURE_DEPENDENCIES) \
+	$(ACLOCAL_M4)
+mkinstalldirs = $(install_sh) -d
+CONFIG_HEADER = $(top_builddir)/config.h
+CONFIG_CLEAN_FILES = version.entities
+CONFIG_CLEAN_VPATH_FILES =
+AM_V_P = $(am__v_P_@AM_V@)
+am__v_P_ = $(am__v_P_@AM_DEFAULT_V@)
+am__v_P_0 = false
+am__v_P_1 = :
+AM_V_GEN = $(am__v_GEN_@AM_V@)
+am__v_GEN_ = $(am__v_GEN_@AM_DEFAULT_V@)
+am__v_GEN_0 = @echo "  GEN     " $@;
+am__v_GEN_1 = 
+AM_V_at = $(am__v_at_@AM_V@)
+am__v_at_ = $(am__v_at_@AM_DEFAULT_V@)
+am__v_at_0 = @
+am__v_at_1 = 
+SOURCES =
+DIST_SOURCES =
+RECURSIVE_TARGETS = all-recursive check-recursive cscopelist-recursive \
+	ctags-recursive dvi-recursive html-recursive info-recursive \
+	install-data-recursive install-dvi-recursive \
+	install-exec-recursive install-html-recursive \
+	install-info-recursive install-pdf-recursive \
+	install-ps-recursive install-recursive installcheck-recursive \
+	installdirs-recursive pdf-recursive ps-recursive \
+	tags-recursive uninstall-recursive
+am__can_run_installinfo = \
+  case $$AM_UPDATE_INFO_DIR in \
+    n|no|NO) false;; \
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+
+html: html-recursive
+
+html-am:
+
+info: info-recursive
+
+info-am:
+
+install-data-am:
+
+install-dvi: install-dvi-recursive
+
+install-dvi-am:
+
+install-exec-am:
+
+install-html: install-html-recursive
+
+install-html-am:
+
+install-info: install-info-recursive
+
+install-info-am:
+
+install-man:
+
+install-pdf: install-pdf-recursive
+
+install-pdf-am:
+
+install-ps: install-ps-recursive
+
+install-ps-am:
+
+installcheck-am:
+
+maintainer-clean: maintainer-clean-recursive
+	-rm -f Makefile
+maintainer-clean-am: distclean-am maintainer-clean-generic
+
+mostlyclean: mostlyclean-recursive
+
+mostlyclean-am: mostlyclean-generic mostlyclean-libtool
+
+pdf: pdf-recursive
+
+pdf-am:
+
+ps: ps-recursive
+
+ps-am:
+
+uninstall-am:
+
+.MAKE: $(am__recursive_targets) install-am install-strip
+
+.PHONY: $(am__recursive_targets) CTAGS GTAGS TAGS all all-am check \
+	check-am clean clean-generic clean-libtool cscopelist-am ctags \
+	ctags-am distclean distclean-generic distclean-libtool \
+	distclean-tags distdir dvi dvi-am html html-am info info-am \
+	install install-am install-data install-data-am install-dvi \
+	install-dvi-am install-exec install-exec-am install-html \
+	install-html-am install-info install-info-am install-man \
+	install-pdf install-pdf-am install-ps install-ps-am \
+	install-strip installcheck installcheck-am installdirs \
+	installdirs-am maintainer-clean maintainer-clean-generic \
+	mostlyclean mostlyclean-generic mostlyclean-libtool pdf pdf-am \
+	ps ps-am tags tags-am uninstall uninstall-am
+
+
+upload:
+	@if test "x$(SUBDIRS)" != x; then \
+	  for a in $(SUBDIRS); do \
+	    cd $$a; \
+	    make upload; \
+	    cd ..; \
+	  done; \
+	fi
+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
diff --git a/docs/README b/docs/README
new file mode 100644
index 0000000..e1cdc9c
--- /dev/null
+++ b/docs/README
@@ -0,0 +1,498 @@
+README 
+------
+
+(Last updated on Mon 15 jul 2013, version 0.11.90.1)
+
+This HOWTO describes the basic usage of the GStreamer RTSP libraries and how you
+can build simple server applications with it.
+
+* General
+
+ The server relies heavily on the RTSP infrastructure of GStreamer. This includes
+ all of the media acquisition, decoding, encoding, payloading and UDP/TCP
+ streaming. We use the rtpbin element for all the session management. Most of
+ the RTSP message parsing and construction in the server is done using the RTSP
+ library that comes with gst-plugins-base. 
+
+ The result is that the server is rather small (a few 11000 lines of code) and easy
+ to understand and extend. In its current state of development, things change
+ fast, API and ABI are unstable. We encourage people to use it for their various
+ use cases and participate by suggesting changes/features.
+
+ Most of the server is built as a library containing a bunch of GObject objects
+ that provide reasonable default functionality but has a fair amount of hooks
+ to override the default behaviour.
+
+ The server currently integrates with the glib mainloop nicely.  It's currently
+ not meant to be used in high-load scenarios and because no security audit has
+ been done, you should probably not put it on a public IP address.
+
+* Initialisation
+
+ You need to initialize GStreamer before using any of the RTSP server functions.
+
+   #include 
+
+   int
+   main (int argc, char *argv[])
+   {
+     gst_init (&argc, &argv);
+
+     ...
+   }
+ 
+ The server itself currently does not have any specific initialisation function
+ but that might change in the future.
+
+
+* Creating the server
+
+ The first thing you want to do is create a new GstRTSPServer object. This object
+ will handle all the new client connections to your server once it is added to a
+ GMainLoop. You can create a new server object like this:
+
+   #include 
+
+   GstRTSPServer *server;
+
+   server = gst_rtsp_server_new ();
+
+ The server will by default listen on port 8554 for new connections. This can be
+ changed by calling gst_rtsp_server_set_service() or with the 'service' GObject
+ property. This makes it possible to run multiple server instances listening on
+ multiple ports on one machine.
+
+ We can make the server start listening on its default port by attaching it to a
+ mainloop. The following example shows how this is done and will start a server
+ on the default 8554 port. For any request we make, we will get a NOT_FOUND
+ error code because we need to configure more things before the server becomes
+ useful.
+
+   #include 
+   #include 
+
+   int
+   main (int argc, char *argv[])
+   {
+     GstRTSPServer *server;
+     GMainLoop *loop;
+
+     gst_init (&argc, &argv);
+
+     server = gst_rtsp_server_new ();
+
+     /* make a mainloop for the default context */
+     loop = g_main_loop_new (NULL, FALSE);
+
+     /* attach the server to the default maincontext */
+     gst_rtsp_server_attach (server, NULL);
+
+     /* start serving */
+     g_main_loop_run (loop);
+   }
+
+ The server manages four other objects: GstRTSPSessionPool,
+ GstRTSPMountPoints, GstRTSPAuth and GstRTSPThreadPool.
+
+ The GstRTSPSessionPool is an object that keeps track of all the active sessions
+ in the server. A session will usually be kept for each client that performed a
+ SETUP request for a certain media stream. It contains the configuration that
+ the client negotiated with the server to receive the particular stream, ie. the
+ transport used and port pairs for UDP along with the state of the streaming.
+ The default implementation of the session pool is usually sufficient but
+ alternative implementation can be used by the server.
+
+ The GstRTSPMountPoints object is more interesting and needs more configuration
+ before the server object is useful. This object manages the mapping from a
+ request URL to a specific stream and its configuration. We explain in the next
+ topic how to configure this object.
+
+ GstRTSPAuth is an object that authenticates users and authorizes actions
+ performed by users. By default, a server does not have a GstRTSPAuth object and
+ thus does not try to perform any authentication or authorization.
+
+ GstRTSPThreadPool manages the threads used for client connections and media
+ pipelines. The server has a default implementation of a threadpool that should
+ be sufficient in most cases.
+
+
+* Making url mount points
+
+ Next we need to define what media is attached to a particular URL. What we want
+ to achieve is that when the user asks our server for a specific URL, say /test,
+ that we create (or reuse) a GStreamer pipeline that produces one or more RTP
+ streams. 
+ 
+ The object that can create such pipeline is called a GstRTSPMediaFactory object. 
+ The default implementation of GstRTSPMediaFactory allows you to easily create
+ GStreamer pipelines using the gst-launch syntax. It is possible to create a
+ GstRTSPMediaFactory subclass that uses different methods for constructing
+ pipelines.
+
+ The default GstRTSPMediaFactory can be configured with a gst-launch line that
+ produces a toplevel bin (use '(' and ')' around the pipeline description to
+ force a toplevel GstBin instead of the default GstPipeline toplevel element).
+ The pipeline description should contain elements named payN, one for each
+ stream (ex. pay0, pay1, ...). Also, for increased compatibility each stream
+ should have a different payload type which can be configured on the payloader.
+
+ The following code snippet illustrates how to create a media factory that
+ creates an RTP feed of an H264 encoded test video signal.
+
+   GstRTSPMediaFactory *factory;
+
+   factory = gst_rtsp_media_factory_new ();
+
+   gst_rtsp_media_factory_set_launch (factory, 
+                "( videotestsrc ! x264enc ! rtph264pay pt=96 name=pay0 )");
+
+ Now that we have the media factory, we can attach it to a specific url. To do
+ this we get the default GstRTSPMountPoints from our server and add the url to
+ factory mount points to it like this:
+
+   GstRTSPMountPoints *mounts;
+
+   ...create server..create factory..
+
+   /* get the default mount points from the server */
+   mounts = gst_rtsp_server_get_mount_points (server);
+
+   /* attach the video test signal to the "/test" URL */
+   gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+   g_object_unref (mounts);
+
+ When starting the server now and directing an RTP client to the URL (like with 
+ vlc, mplayer or gstreamer):
+
+   rtsp://localhost:8554/test 
+
+ a test signal will be streamed to the client. The full example code can be
+ found in the examples/test-readme.c file.
+
+ Note that by default the factory will create a new pipeline for each client. If
+ you want to share a pipeline between clients, use
+ gst_rtsp_media_factory_set_shared().
+
+
+* more on GstRTSPMediaFactory
+
+ The GstRTSPMediaFactory is responsible for creating and caching GstRTSPMedia
+ objects. 
+
+ A freshly created GstRTSPMedia object from the factory initially only contains a
+ GstElement containing the elements to produce the RTP streams for the media and
+ a GPtrArray of GstRTSPStream objects describing the payloader and its source
+ pad. The media is unprepared in this state.
+
+ Usually the url will determine what kind of pipeline should be created. You can
+ for example use query parameters to configure certain parts of the pipeline or
+ select encoders and payloaders based on some url pattern.
+
+ When dealing with a live stream from, for example, a webcam, it can be
+ interesting to share the pipeline with multiple clients. This must be done when
+ only one instance of the video capture element can be used at a time. In this
+ case, the shared property of GstRTSPMedia must be used to instruct the default
+ GstRTSPMediaFactory implementation to cache the media.
+
+ When all objects created from a factory can be shared, you can set the shared
+ property directly on the factory.
+
+* more on GstRTSPMedia
+
+ After creating the GstRTSPMedia object from the factory, it can be prepared
+ with gst_rtsp_media_prepare(). This method will put those objects in a
+ GstPipeline and will construct and link the streaming elements and the
+ rtpbin session manager object.
+
+ The _prepare() method will then preroll the pipeline in order to figure out the
+ caps on the payloaders. After the GstRTSPMedia prerolled it will be in the
+ prepared state and can be used for creating SDP files or for streaming to
+ clients.
+
+ The prepare method will also create 2 UDP ports for each stream that can be
+ used for sending and receiving RTP/RTCP from clients. These port numbers will
+ have to be negotiated with the client in the SETUP requests.
+
+ When preparing a GstRTSPMedia, an appsink and asppsrc is also constructed
+ for streaming the stream over TCP when requested.
+
+ Media is prepared by the server when DESCRIBE or SETUP requests are received
+ from the client.
+
+
+* the GstRTSPClient object
+
+ When a server detects a new client connection on its port, it will accept the
+ connection, check if the connection is allowed and then call the vmethod
+ create_client. The default implementation of this function will create
+ a new GstRTCPClient object, will configure the session pool, mount points,
+ auth and thread pool objects in it.
+
+ The server will then attach the new client to a server mainloop to let it
+ handle further communication with the client. In RTSP it is usual to keep
+ the connection open between multiple RTSP requests. The client watch will
+ be dispatched by the server mainloop when a new GstRTSPMessage is received,
+ which will then be handled and a response will be sent.
+
+ The GstRTSPClient object remains alive for as long as a client has a TCP
+ connection open with the server. Since is possible for a client to open and close
+ the TCP connection between requests, we cannot store the state related
+ to the configured RTSP session in the GstRTSPClient object. This server state
+ is instead stored in the GstRTSPSession object, identified with the session
+ id.
+
+
+* GstRTSPSession
+
+ This object contains state about a specific RTSP session identified with a
+ session id. This state contains the configured streams and their associated
+ transports.
+ 
+ When a GstRTSPClient performs a SETUP request, the server will allocate a new
+ GstRTSPSession with a unique session id from the GstRTSPSessionPool. The pool
+ maintains a list of all existing sessions and makes sure that no session id is
+ used multiple times. The session id is sent to the client so that the client
+ can refer to its previously configured state by sending the session id in
+ further requests.
+
+ A client will then use the session id to configure one or more
+ GstRTSPSessionMedia objects, identified by their url. This SessionMedia object
+ contains the configuration of a GstRTSPMedia and its configured
+ GstRTSPStreamTransport.
+
+
+* GstRTSPSessionMedia and GstRTSPStreamTransport
+
+ A GstRTSPSessionMedia is identified by a URL and is referenced by a
+ GstRTSPSession. It is created as soon as a client performs a SETUP operation on
+ a particular URL. It will contain a link to the GstRTSPMedia object associated
+ with the URL along with the state of the media and the configured transports
+ for each of the streams in the media.
+
+ Each SETUP request performed by the client will configure a
+ GstRTSPStreamTransport object linked to by the GstRTSPSessionMedia structure.
+ It will contain the transport information needed to send this stream to the
+ client. The GstRTSPStreamTransport also contains a link to the GstRTSPStream
+ object that generates the actual data to be streamed to the client.
+
+ Note how GstRTSPMedia and GstRTSPStream (the providers of the data to
+ stream) are decoupled from GstRTSPSessionMedia and GstRTSPStreamTransport (the
+ configuration of how to send this stream to a client) in order to be able to
+ send the data of one GstRTSPMedia to multiple clients.
+
+
+* media control
+
+ After a client has configured the transports for a GstRTSPMedia and its
+ GstRTSPStreams, the client can play/pause/stop the stream.
+
+ The GstRTSPMedia object was prepared in the DESCRIBE call (or during SETUP when
+ the client skipped the DESCRIBE request). As seen earlier, this configures a
+ couple of udpsink and udpsrc elements to respectively send and receive the
+ media to clients.
+
+ When a client performs a PLAY request, its configured destination UDP ports are
+ added to the GstRTSPStream target destinations, at which point data will
+ be sent to the client. The corresponding GstRTSPMedia object will be set to the
+ PLAYING state if it was not allready in order to send the data to the
+ destination.
+
+ The server needs to prepare an RTP-Info header field in the PLAY response,
+ which consists of the sequence number and the RTP timestamp of the next RTP
+ packet. In order to achive this, the server queries the payloaders for this
+ information when it prerolled the pipeline.
+
+ When a client performs a PAUSE request, the destination UDP ports are removed
+ from the GstRTSPStream object and the GstRTSPMedia object is set to PAUSED
+ if no other destinations are configured anymore.
+
+
+* seeking
+
+ A seek is performed when a client sends a Range header in the PLAY request.
+ This only works when not dealing with shared (live) streams.
+
+ The server performs a GStreamer flushing seek on the media, waits for the
+ pipeline to preroll again and then responds to the client after collecting the
+ new RTP sequence number and timestamp from the payloaders.
+
+
+* session management
+
+ The server has to react to clients that suddenly disappear because of network
+ problems or otherwise. It needs to make sure that it can reasonable free the
+ resources that are used by the various objects in use for streaming when the
+ client appears to be gone.
+
+ Each of the GstRTSPSession objects managed by a GstRTSPSessionPool has
+ therefore a last_access field that contains the timestamp of when activity from
+ a client was last recorded.
+ 
+ Various ways exist to detect activity from a client:
+
+  - RTSP keepalive requests. When a client is receiving RTP data, the RTSP TCP
+    connection is largely unused. It is the client's responsability to
+    periodically send keep-alive requests over the TCP channel.
+
+    Whenever a keep-alive request is received by the server (any request that
+    contains a session id, usually an OPTION or GET_PARAMETER request) the
+    last_access of the session is updated.
+
+  - Since it is not required for a client to keep the RTSP TCP connection open
+    while streaming, gst-rtsp-server also detects activity from clients by
+    looking at the RTCP messages it receives.
+
+    When an RTCP message is received from a client, the server looks in its list
+    of active ports if this message originates from a known host/port pair that
+    is currently active in a GstRTSPSession. If this is the case, the session is
+    kept alive.
+
+    Since the server does not know anything about the port number that will be
+    used by the client to send RTCP, this method does not always work. Later
+    RTSP RFCs will include support for negotiating this port number with the
+    server. Most clients however use the same port number for sending and
+    receiving RTCP exactly for this reason.
+
+ If there was no activity in a particular session for a long time (by default 60
+ seconds), the application should remove the session from the pool. For this,
+ the application should periodically (say every 2 seconds) check if no sessions
+ expired and call gst_rtsp_session_pool_cleanup() to remove them.
+ 
+ When a session is removed from the sessionpool and its last reference is
+ unreffef, all related objects and media are destroyed as if a TEARDOWN happened
+ from the client.
+
+
+* TEARDOWN
+
+ A TEARDOWN request will first locate the GstRTSPSessionMedia of the URL. It
+ will then remove all transports from the streams, making sure that streaming
+ stops to the clients. It will then remove the GstRTSPSessionMedia and
+ GstRTSPStreamTransport objects. Finally the GstRTSPSession is released back
+ into the pool.
+
+ When there are no more references to the GstRTSPMedia, the media pipeline is
+ shut down (with _unprepare) and destroyed. This will then also destroy the
+ GstRTSPStream objects.
+
+
+* Security
+
+ The security of the server and the policy is implemented in a GstRTSPAuth
+ object. The object is reponsible for:
+
+  - authenticate the user of the server.
+
+  - check if the current user is authorized to perform an operation.
+
+ For critical operations, the server will call gst_rtsp_auth_check() with
+ a string describing the operation which should be validated. The installed
+ GstRTSPAuth object is then responsible for checking if the operation is
+ allowed.
+
+ Implementations of GstRTSPAuth objects can use the following infrastructure
+ bits of the rtsp server to implement these checks:
+
+  - GstRTSPToken: a generic structure describing roles and permissions granted
+    to a user.
+
+  - GstRTSPPermissions: a generic list of roles and matching permissions. These
+    can be attached to media and facties currently.
+
+ An Auth implementation will usually authenticate a user, using method such as
+ Basic authentication or client certificates or perhaps simply use the IP address.
+ The result of the authentication of the user will be a GstRTSPToken that is made
+ current in the context of the ongoing request.
+
+ The auth module can then implement the various checks in the server by looking
+ at the current token and, if needed, compare it to the required GstRTSPPermissions
+ of the current object.
+
+ The security is deliberately kept generic with a default implementation of the
+ GstRTSPAuth object providing a usable and simple implementaion. To make more
+ complicated security modules, the auth object should be subclassed and new
+ implementations for the checks needs to be made.
+
+
+Objects
+-------
+
+GstRTSPServer
+ - Toplevel object listening for connections and creating new
+   GstRTSPClient objects
+
+GstRTSPClient
+ - Handle RTSP Requests from connected clients. All other objects
+   are called by this object.
+
+GstRTSPContext
+ - Helper structure contaning the current state of the request
+   handled by the client.
+
+
+GstRTSPMountPoints
+ - Maps a url to a GstRTSPMediaFactory implementation. The default
+   implementation uses a simple hashtable to map a url to a factory.
+
+GstRTSPMediaFactory
+ - Creates and caches GstRTSPMedia objects. The default implementation
+   can create GstRTSPMedia objects based on gst-launch syntax.
+
+GstRTSPMediaFactoryURI
+ - Specialized GstRTSPMediaFactory that can stream the content of any
+   URI.
+
+GstRTSPMedia
+ - The object that contains the media pipeline and various GstRTSPStream
+   objects that produce RTP packets
+
+GstRTSPStream
+ - Manages the elements to stream a stream of a GstRTSPMedia to one or
+   more GstRTSPStreamTransports.
+
+
+GstRTSPSessionPool
+ - Creates and manages GstRTSPSession objects identified by an id.
+
+GstRTSPSession
+ - An object containing the various GstRTSPSessionMedia objects managed
+   by this session.
+
+GstRTSPSessionMedia
+ - The state of a GstRTSPMedia and the configuration of a GstRTSPStream
+   objects. The configuration for the GstRTSPStream is stored in
+   GstRTSPStreamTransport objects.
+
+GstRTSPStreamTransport
+ - Configuration of how a GstRTSPStream is send to a particular client. It
+   contains the transport that was negotiated with the client in the SETUP
+   request.
+
+
+GstRTSPSDP
+ - helper functions for creating SDP messages from gstRTSPMedia
+
+GstRTSPAddressPool
+ - a pool of multicast and unicast addresses used in streaming
+
+GstRTSPThreadPool
+ - a pool of threads used for various server tasks such as handling clients and
+   managin media pipelines.
+
+
+GstRTSPAuth
+ - Hooks for checking authorizations, all client activity will call this
+   object with the GstRTSPContext structure. By default it supports
+   basic authentication.
+
+GstRTSPToken
+ - Credentials of a user. This contrains the roles that the user is allowed
+   to assume and other permissions or capabilities of the user.
+
+GstRTSPPermissions
+ - A list of permissions for each role. The permissions are usually attached
+   to objects to describe what roles have what permissions.
+
+GstRTSPParams
+ - object to handle get and set parameter requests.
+
diff --git a/docs/design/gst-rtp-server-design b/docs/design/gst-rtp-server-design
new file mode 100644
index 0000000..7da8283
--- /dev/null
+++ b/docs/design/gst-rtp-server-design
@@ -0,0 +1,35 @@
+RTSP server
+-----------
+
+This directory contains an example RTSP server built with various GStreamer
+components and libraries. It also uses GStreamer for all of the multimedia
+procesing and RTP bits. The following features are implemented:
+
+ - 
+
+Server Design
+-------------
+
+The toplevel component of the server is a GstRTSPServer object. This object
+creates and binds on the server socket and attaches into the mainloop.
+
+For each request a new GstRTSPClient object is created that will accept the
+request and a thread is started to handle further communication with the
+client until the connection is closed.
+
+When a client issues a SETUP request we create a GstRTSPSession object,
+identified with a sessionid, that will keep track of the state of a client.
+The object is destroyed when a TEARDOWN request is made for that sessionid.
+
+We also maintain a pool of URL to media pipeline mappings. Each url is mapped to
+an object that is able to provide a pipeline for that media. We provide
+pipelines to stream live captured data, on-demand file streaming or on-demand
+transcoding of a file or stream.
+
+A pool of currently active pipelines is also maintained. Usually the active
+pipelines are in use by one or more GstRTSPSession objects. An active pipeline
+becomes inactive when no more sessions refer to it.
+
+A client can choose to start a new pipeline or join a currently active pipeline.
+Some active pipeline cannot be joined (such as on-demand streams) but a new
+instance of that pipeline can be created.
diff --git a/docs/libs/Makefile.am b/docs/libs/Makefile.am
new file mode 100644
index 0000000..64b7aa5
--- /dev/null
+++ b/docs/libs/Makefile.am
@@ -0,0 +1,106 @@
+GST_DOC_SCANOBJ = $(top_srcdir)/common/gstdoc-scangobj
+
+## Process this file with automake to produce Makefile.in
+
+# The name of the module, e.g. 'glib'.
+MODULE=gst-rtsp-server
+DOC_MODULE=$(MODULE)
+
+# for upload-doc.mak
+DOC=$(MODULE)
+FORMATS=html
+html: html-build.stamp
+include $(top_srcdir)/common/upload-doc.mak
+
+# generated basefiles
+#basefiles = \
+##		$(DOC_MODULE).types \
+#		$(DOC_MODULE)-sections.txt \
+#		$(DOC_MODULE)-docs.sgml
+
+# ugly hack to make -unused.sgml work
+#unused-build.stamp:
+#	BUILDDIR=`pwd` && \
+#	cd $(srcdir)/tmpl && \
+#	ln -sf gstreamer-libs-unused.sgml \
+#		$$BUILDDIR/tmpl/gstreamer-libs-@GST_API_VERSION@-unused.sgml
+#	touch unused-build.stamp
+
+# these rules are added to create parallel docs using GST_API_VERSION
+#$(basefiles): gstreamer-libs-@GST_API_VERSION@%: gstreamer-libs%
+#	cp $< $@
+
+#CLEANFILES = $(basefiles)
+
+# The top-level SGML file. Change it if you want.
+DOC_MAIN_SGML_FILE=$(DOC_MODULE)-docs.sgml
+
+# The directory containing the source code. Relative to $(top_srcdir).
+# gtk-doc will search all .c & .h files beneath here for inline comments
+# documenting functions and macros.
+DOC_SOURCE_DIR = $(top_srcdir)/gst/rtsp-server/
+DOC_BUILD_DIR=$(top_builddir)/gst/rtsp-server/
+
+SCAN_OPTIONS= 
+
+# FIXME :
+# there's something wrong with gstreamer-sections.txt not being in the dist
+# maybe it doesn't resolve; we're adding it below for now
+#EXTRA_DIST = gstreamer.types.in gstreamer.hierarchy $(DOC_MODULE)-sections.txt gstreamer-sections.txt $(DOC_MAIN_SGML_FILE)
+
+# Extra options to supply to gtkdoc-mkdb.
+MKDB_OPTIONS=--sgml-mode --source-suffixes=c,h,cc,m
+
+# Extra options to supply to gtkdoc-fixref.
+FIXXREF_OPTIONS=--extra-dir=$(GLIB_PREFIX)/share/gtk-doc/html \
+	--extra-dir=$(GST_PREFIX)/share/gtk-doc/html \
+	--extra-dir=$(GSTPB_PREFIX)/share/gtk-doc/html
+
+# Used for dependencies.
+HFILE_GLOB=$(DOC_SOURCE_DIR)/*.h
+CFILE_GLOB=$(DOC_SOURCE_DIR)/*.c
+
+SCANOBJ_DEPS = \
+	$(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la
+
+# Extra options to supply to gtkdoc-scan.
+SCANOBJ_OPTIONS=--type-init-func="g_type_init();gst_init(&argc,&argv)"
+
+# Header files to ignore when scanning.
+IGNORE_HFILES = 
+IGNORE_CFILES =
+
+# we add all .h files of elements that have signals/args we want
+# sadly this also pulls in the private methods - maybe we should
+# move those around in the source ?
+# also, we should add some stuff here conditionally based on whether
+# or not the plugin will actually build
+# but I'm not sure about that - it might be this Just Works given that
+# the registry won't have the element
+
+EXTRA_HFILES = 
+
+# Images to copy into HTML directory.
+HTML_IMAGES =
+
+# Extra SGML files that are included by $(DOC_MAIN_SGML_FILE).
+content_files = 
+
+# Other files to distribute.
+extra_files =
+
+# CFLAGS and LDFLAGS for compiling scan program. Only needed if your app/lib
+# contains GtkObjects/GObjects and you want to document signals and properties.
+GTKDOC_CFLAGS = -I$(top_srcdir) $(GST_PLUGINS_BASE_CFLAGS) \
+	$(GST_BASE_CFLAGS) $(GST_CFLAGS)
+GTKDOC_LIBS = $(SCANOBJ_DEPS) $(GST_PLUGINS_BASE_LIBS) \
+	$(GST_BASE_LIBS) $(GST_LIBS) $(GCOV_LIBS)
+
+GTKDOC_CC=$(LIBTOOL) --tag=CC --mode=compile $(CC)
+GTKDOC_LD=$(LIBTOOL) --tag=CC --mode=link $(CC)
+
+# If you need to override some of the declarations, place them in this file
+# and uncomment this line.
+DOC_OVERRIDES = $(DOC_MODULE)-overrides.txt
+
+include $(top_srcdir)/common/gtk-doc.mak
diff --git a/docs/libs/Makefile.in b/docs/libs/Makefile.in
new file mode 100644
index 0000000..95811a2
--- /dev/null
+++ b/docs/libs/Makefile.in
@@ -0,0 +1,946 @@
+# Makefile.in generated by automake 1.14.1 from Makefile.am.
+# @configure_input@
+
+# Copyright (C) 1994-2013 Free Software Foundation, Inc.
+
+# This Makefile.in is free software; the Free Software Foundation
+# gives unlimited permission to copy and/or distribute it,
+# with or without modifications, as long as this notice is preserved.
+
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
+# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
+# PARTICULAR PURPOSE.
+
+@SET_MAKE@
+
+# this snippet is to be included by both our docbook manuals
+# and gtk-doc API references
+
+# it adds an upload target to each of these dir's Makefiles
+
+# each Makefile.am should define the following variables:
+# - DOC: the base name of the documentation
+#        (faq, manual, pwg, gstreamer, gstreamer-libs)
+# - FORMATS: the formats in which DOC is output
+#            (html ps pdf)
+
+# if you want to use it, make sure your $HOME/.ssh/config file contains the
+# correct User entry for the Host entry for the DOC_SERVER
+
+###########################################################################
+# Everything below here is generic and you shouldn't need to change it.
+###########################################################################
+# thomas: except of course that we did
+VPATH = @srcdir@
+am__is_gnu_make = test -n '$(MAKEFILE_LIST)' && test -n '$(MAKELEVEL)'
+am__make_running_with_option = \
+  case $${target_option-} in \
+      ?) ;; \
+      *) echo "am__make_running_with_option: internal error: invalid" \
+              "target option '$${target_option-}' specified" >&2; \
+         exit 1;; \
+  esac; \
+  has_opt=no; \
+  sane_makeflags=$$MAKEFLAGS; \
+  if $(am__is_gnu_make); then \
+    sane_makeflags=$$MFLAGS; \
+  else \
+    case $$MAKEFLAGS in \
+      *\\[\ \	]*) \
+        bs=\\; \
+        sane_makeflags=`printf '%s\n' "$$MAKEFLAGS" \
+          | sed "s/$$bs$$bs[$$bs $$bs	]*//g"`;; \
+    esac; \
+  fi; \
+  skip_next=no; \
+  strip_trailopt () \
+  { \
+    flg=`printf '%s\n' "$$flg" | sed "s/$$1.*$$//"`; \
+  }; \
+  for flg in $$sane_makeflags; do \
+    test $$skip_next = yes && { skip_next=no; continue; }; \
+    case $$flg in \
+      *=*|--*) continue;; \
+        -*I) strip_trailopt 'I'; skip_next=yes;; \
+      -*I?*) strip_trailopt 'I';; \
+        -*O) strip_trailopt 'O'; skip_next=yes;; \
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+      -[JT]) skip_next=yes;; \
+    esac; \
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+  test $$has_opt = yes
+am__make_dryrun = (target_option=n; $(am__make_running_with_option))
+am__make_keepgoing = (target_option=k; $(am__make_running_with_option))
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+NORMAL_INSTALL = :
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+GST_AGE = @GST_AGE@
+GST_ALL_CFLAGS = @GST_ALL_CFLAGS@
+GST_ALL_LDFLAGS = @GST_ALL_LDFLAGS@
+GST_ALL_LIBS = @GST_ALL_LIBS@
+GST_API_VERSION = @GST_API_VERSION@
+GST_BASE_CFLAGS = @GST_BASE_CFLAGS@
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+GST_PKG_CONFIG_PATH = @GST_PKG_CONFIG_PATH@
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+HAVE_PS2PDF = @HAVE_PS2PDF@
+HAVE_XMLLINT = @HAVE_XMLLINT@
+HAVE_XSLTPROC = @HAVE_XSLTPROC@
+HTML_DIR = @HTML_DIR@
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+install_sh = @install_sh@
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+localstatedir = @localstatedir@
+mandir = @mandir@
+mkdir_p = @mkdir_p@
+oldincludedir = @oldincludedir@
+pdfdir = @pdfdir@
+plugindir = @plugindir@
+prefix = @prefix@
+program_transform_name = @program_transform_name@
+psdir = @psdir@
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+target_vendor = @target_vendor@
+top_build_prefix = @top_build_prefix@
+top_builddir = @top_builddir@
+top_srcdir = @top_srcdir@
+GST_DOC_SCANOBJ = $(top_srcdir)/common/gstdoc-scangobj
+
+# The name of the module, e.g. 'glib'.
+MODULE = gst-rtsp-server
+DOC_MODULE = $(MODULE)
+
+# for upload-doc.mak
+DOC = $(MODULE)
+FORMATS = html
+
+# these variables define the location of the online docs
+DOC_SERVER = gstreamer.freedesktop.org
+DOC_BASE = /srv/gstreamer.freedesktop.org/www/data/doc
+DOC_URL = $(DOC_SERVER):$(DOC_BASE)
+
+# generated basefiles
+#basefiles = \
+#		$(DOC_MODULE)-sections.txt \
+#		$(DOC_MODULE)-docs.sgml
+
+# ugly hack to make -unused.sgml work
+#unused-build.stamp:
+#	BUILDDIR=`pwd` && \
+#	cd $(srcdir)/tmpl && \
+#	ln -sf gstreamer-libs-unused.sgml \
+#		$$BUILDDIR/tmpl/gstreamer-libs-@GST_API_VERSION@-unused.sgml
+#	touch unused-build.stamp
+
+# these rules are added to create parallel docs using GST_API_VERSION
+#$(basefiles): gstreamer-libs-@GST_API_VERSION@%: gstreamer-libs%
+#	cp $< $@
+
+#CLEANFILES = $(basefiles)
+
+# The top-level SGML file. Change it if you want.
+DOC_MAIN_SGML_FILE = $(DOC_MODULE)-docs.sgml
+
+# The directory containing the source code. Relative to $(top_srcdir).
+# gtk-doc will search all .c & .h files beneath here for inline comments
+# documenting functions and macros.
+DOC_SOURCE_DIR = $(top_srcdir)/gst/rtsp-server/
+DOC_BUILD_DIR = $(top_builddir)/gst/rtsp-server/
+SCAN_OPTIONS = 
+
+# FIXME :
+# there's something wrong with gstreamer-sections.txt not being in the dist
+# maybe it doesn't resolve; we're adding it below for now
+#EXTRA_DIST = gstreamer.types.in gstreamer.hierarchy $(DOC_MODULE)-sections.txt gstreamer-sections.txt $(DOC_MAIN_SGML_FILE)
+
+# Extra options to supply to gtkdoc-mkdb.
+MKDB_OPTIONS = --sgml-mode --source-suffixes=c,h,cc,m
+
+# Extra options to supply to gtkdoc-fixref.
+FIXXREF_OPTIONS = --extra-dir=$(GLIB_PREFIX)/share/gtk-doc/html \
+	--extra-dir=$(GST_PREFIX)/share/gtk-doc/html \
+	--extra-dir=$(GSTPB_PREFIX)/share/gtk-doc/html
+
+
+# Used for dependencies.
+HFILE_GLOB = $(DOC_SOURCE_DIR)/*.h
+CFILE_GLOB = $(DOC_SOURCE_DIR)/*.c
+SCANOBJ_DEPS = \
+	$(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la
+
+
+# Extra options to supply to gtkdoc-scan.
+SCANOBJ_OPTIONS = --type-init-func="g_type_init();gst_init(&argc,&argv)"
+
+# Header files to ignore when scanning.
+IGNORE_HFILES = 
+IGNORE_CFILES = 
+
+# we add all .h files of elements that have signals/args we want
+# sadly this also pulls in the private methods - maybe we should
+# move those around in the source ?
+# also, we should add some stuff here conditionally based on whether
+# or not the plugin will actually build
+# but I'm not sure about that - it might be this Just Works given that
+# the registry won't have the element
+EXTRA_HFILES = 
+
+# Images to copy into HTML directory.
+HTML_IMAGES = 
+
+# Extra SGML files that are included by $(DOC_MAIN_SGML_FILE).
+content_files = 
+
+# Other files to distribute.
+extra_files = 
+
+# CFLAGS and LDFLAGS for compiling scan program. Only needed if your app/lib
+# contains GtkObjects/GObjects and you want to document signals and properties.
+GTKDOC_CFLAGS = -I$(top_srcdir) $(GST_PLUGINS_BASE_CFLAGS) \
+	$(GST_BASE_CFLAGS) $(GST_CFLAGS)
+
+GTKDOC_LIBS = $(SCANOBJ_DEPS) $(GST_PLUGINS_BASE_LIBS) \
+	$(GST_BASE_LIBS) $(GST_LIBS) $(GCOV_LIBS)
+
+GTKDOC_CC = $(LIBTOOL) --tag=CC --mode=compile $(CC)
+GTKDOC_LD = $(LIBTOOL) --tag=CC --mode=link $(CC)
+
+# If you need to override some of the declarations, place them in this file
+# and uncomment this line.
+DOC_OVERRIDES = $(DOC_MODULE)-overrides.txt
+
+# thomas: copied from glib-2
+# We set GPATH here; this gives us semantics for GNU make
+# which are more like other make's VPATH, when it comes to
+# whether a source that is a target of one rule is then
+# searched for in VPATH/GPATH.
+#
+GPATH = $(srcdir)
+
+# thomas: make docs parallel installable
+TARGET_DIR = $(HTML_DIR)/$(DOC_MODULE)-@GST_API_VERSION@
+EXTRA_DIST = \
+	$(content_files)		\
+	$(extra_files)			\
+	$(HTML_IMAGES)			\
+	$(DOC_MAIN_SGML_FILE)		\
+	$(DOC_MODULE).types		\
+	$(DOC_OVERRIDES)		\
+	$(DOC_MODULE)-sections.txt
+
+DOC_STAMPS = \
+	setup-build.stamp		\
+	scan-build.stamp		\
+	sgml-build.stamp		\
+	html-build.stamp		\
+	sgml.stamp		\
+	html.stamp
+
+SCANOBJ_FILES = \
+	$(DOC_MODULE).args		\
+	$(DOC_MODULE).hierarchy		\
+	$(DOC_MODULE).interfaces		\
+	$(DOC_MODULE).prerequisites	\
+	$(DOC_MODULE).signals		\
+	.libs/$(DOC_MODULE)-scan.o
+
+REPORT_FILES = \
+	$(DOC_MODULE)-undocumented.txt \
+	$(DOC_MODULE)-undeclared.txt \
+	$(DOC_MODULE)-unused.txt
+
+CLEANFILES = $(SCANOBJ_FILES) $(REPORT_FILES) $(DOC_STAMPS) doc-registry.xml
+all: all-am
+
+.SUFFIXES:
+$(srcdir)/Makefile.in: @MAINTAINER_MODE_TRUE@ $(srcdir)/Makefile.am $(top_srcdir)/common/upload-doc.mak $(top_srcdir)/common/gtk-doc.mak $(am__configure_deps)
+	@for dep in $?; do \
+	  case '$(am__configure_deps)' in \
+	    *$$dep*) \
+	      ( cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh ) \
+	        && { if test -f $@; then exit 0; else break; fi; }; \
+	      exit 1;; \
+	  esac; \
+	done; \
+	echo ' cd $(top_srcdir) && $(AUTOMAKE) --gnu docs/libs/Makefile'; \
+	$(am__cd) $(top_srcdir) && \
+	  $(AUTOMAKE) --gnu docs/libs/Makefile
+.PRECIOUS: Makefile
+Makefile: $(srcdir)/Makefile.in $(top_builddir)/config.status
+	@case '$?' in \
+	  *config.status*) \
+	    cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh;; \
+	  *) \
+	    echo ' cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@ $(am__depfiles_maybe)'; \
+	    cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@ $(am__depfiles_maybe);; \
+	esac;
+$(top_srcdir)/common/upload-doc.mak $(top_srcdir)/common/gtk-doc.mak:
+
+$(top_builddir)/config.status: $(top_srcdir)/configure $(CONFIG_STATUS_DEPENDENCIES)
+	cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+
+$(top_srcdir)/configure: @MAINTAINER_MODE_TRUE@ $(am__configure_deps)
+	cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+$(ACLOCAL_M4): @MAINTAINER_MODE_TRUE@ $(am__aclocal_m4_deps)
+	cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+$(am__aclocal_m4_deps):
+
+mostlyclean-libtool:
+	-rm -f *.lo
+
+clean-libtool:
+	-rm -rf .libs _libs
+tags TAGS:
+
+ctags CTAGS:
+
+cscope cscopelist:
+
+
+distdir: $(DISTFILES)
+	@srcdirstrip=`echo "$(srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+	topsrcdirstrip=`echo "$(top_srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+	list='$(DISTFILES)'; \
+	  dist_files=`for file in $$list; do echo $$file; done | \
+	  sed -e "s|^$$srcdirstrip/||;t" \
+	      -e "s|^$$topsrcdirstrip/|$(top_builddir)/|;t"`; \
+	case $$dist_files in \
+	  */*) $(MKDIR_P) `echo "$$dist_files" | \
+			   sed '/\//!d;s|^|$(distdir)/|;s,/[^/]*$$,,' | \
+			   sort -u` ;; \
+	esac; \
+	for file in $$dist_files; do \
+	  if test -f $$file || test -d $$file; then d=.; else d=$(srcdir); fi; \
+	  if test -d $$d/$$file; then \
+	    dir=`echo "/$$file" | sed -e 's,/[^/]*$$,,'`; \
+	    if test -d "$(distdir)/$$file"; then \
+	      find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+	    fi; \
+	    if test -d $(srcdir)/$$file && test $$d != $(srcdir); then \
+	      cp -fpR $(srcdir)/$$file "$(distdir)$$dir" || exit 1; \
+	      find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+	    fi; \
+	    cp -fpR $$d/$$file "$(distdir)$$dir" || exit 1; \
+	  else \
+	    test -f "$(distdir)/$$file" \
+	    || cp -p $$d/$$file "$(distdir)/$$file" \
+	    || exit 1; \
+	  fi; \
+	done
+	$(MAKE) $(AM_MAKEFLAGS) \
+	  top_distdir="$(top_distdir)" distdir="$(distdir)" \
+	  dist-hook
+check-am: all-am
+check: check-am
+all-am: Makefile all-local
+installdirs:
+install: install-am
+install-exec: install-exec-am
+install-data: install-data-am
+uninstall: uninstall-am
+
+install-am: all-am
+	@$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
+
+installcheck: installcheck-am
+install-strip:
+	if test -z '$(STRIP)'; then \
+	  $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	    install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	      install; \
+	else \
+	  $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	    install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	    "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'" install; \
+	fi
+mostlyclean-generic:
+
+clean-generic:
+	-test -z "$(CLEANFILES)" || rm -f $(CLEANFILES)
+
+distclean-generic:
+	-test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
+	-test . = "$(srcdir)" || test -z "$(CONFIG_CLEAN_VPATH_FILES)" || rm -f $(CONFIG_CLEAN_VPATH_FILES)
+
+maintainer-clean-generic:
+	@echo "This command is intended for maintainers to use"
+	@echo "it deletes files that may require special tools to rebuild."
+clean: clean-am
+
+clean-am: clean-generic clean-libtool clean-local mostlyclean-am
+
+distclean: distclean-am
+	-rm -f Makefile
+distclean-am: clean-am distclean-generic distclean-local
+
+dvi: dvi-am
+
+dvi-am:
+
+html-am:
+
+info: info-am
+
+info-am:
+
+install-data-am: install-data-local
+
+install-dvi: install-dvi-am
+
+install-dvi-am:
+
+install-exec-am:
+
+install-html: install-html-am
+
+install-html-am:
+
+install-info: install-info-am
+
+install-info-am:
+
+install-man:
+
+install-pdf: install-pdf-am
+
+install-pdf-am:
+
+install-ps: install-ps-am
+
+install-ps-am:
+
+installcheck-am:
+
+maintainer-clean: maintainer-clean-am
+	-rm -f Makefile
+maintainer-clean-am: distclean-am maintainer-clean-generic \
+	maintainer-clean-local
+
+mostlyclean: mostlyclean-am
+
+mostlyclean-am: mostlyclean-generic mostlyclean-libtool
+
+pdf: pdf-am
+
+pdf-am:
+
+ps: ps-am
+
+ps-am:
+
+uninstall-am: uninstall-local
+
+.MAKE: install-am install-strip
+
+.PHONY: all all-am all-local check check-am clean clean-generic \
+	clean-libtool clean-local cscopelist-am ctags-am dist-hook \
+	distclean distclean-generic distclean-libtool distclean-local \
+	distdir dvi dvi-am html html-am info info-am install \
+	install-am install-data install-data-am install-data-local \
+	install-dvi install-dvi-am install-exec install-exec-am \
+	install-html install-html-am install-info install-info-am \
+	install-man install-pdf install-pdf-am install-ps \
+	install-ps-am install-strip installcheck installcheck-am \
+	installdirs maintainer-clean maintainer-clean-generic \
+	maintainer-clean-local mostlyclean mostlyclean-generic \
+	mostlyclean-libtool pdf pdf-am ps ps-am tags-am uninstall \
+	uninstall-am uninstall-local
+
+html: html-build.stamp
+
+upload: $(FORMATS)
+	@if echo $(FORMATS) | grep html > /dev/null; then \
+	  echo "Preparing docs for upload (rebasing cross-references) ..." ; \
+	  if test x$(builddir) != x$(srcdir); then \
+	    echo "make upload can only be used if srcdir == builddir"; \
+	    exit 1; \
+	  fi; \
+	  # gtkdoc-rebase sometimes gets confused, so reset everything to \
+	  # local links before rebasing to online links                   \
+	  gtkdoc-rebase --html-dir=$(builddir)/html 2>/dev/null 2>/dev/null ; \
+	  rebase=`gtkdoc-rebase --verbose --online --html-dir=$(builddir)/html` ; \
+	  echo "$$rebase" | grep -e "On-*line"; \
+	  for req in glib gobject gstreamer gstreamer-libs gst-plugins-base-libs; do \
+	    if ! ( echo "$$rebase" | grep -i -e "On-*line.*/$$req/" ); then \
+	      echo "===============================================================================" ; \
+	      echo " Could not determine online location for $$req docs. Cross-referencing will be " ; \
+	      echo " broken, so not uploading. Make sure the library's gtk-doc documentation is    " ; \
+	      echo " installed somewhere in /usr/share/gtk-doc.                                    " ; \
+	      echo "===============================================================================" ; \
+	      exit 1; \
+	    fi; \
+	  done; \
+	  export SRC="$$SRC html"; \
+	fi; \
+	if echo $(FORMATS) | grep ps > /dev/null; then export SRC="$$SRC $(DOC).ps"; fi; \
+	if echo $(FORMATS) | grep pdf > /dev/null; then export SRC="$$SRC $(DOC).pdf"; fi; \
+	\
+	# upload releases to both X.Y/ and head/ subdirectories \
+	export DIR=$(DOC_BASE)/gstreamer/$(PACKAGE_VERSION_MAJOR).$(PACKAGE_VERSION_MINOR)/$(DOC); \
+	echo Uploading $$SRC to $(DOC_SERVER):$$DIR; \
+	ssh $(DOC_SERVER) mkdir -p $$DIR; \
+	rsync -rv -e ssh --delete $$SRC $(DOC_SERVER):$$DIR; \
+	ssh $(DOC_SERVER) chmod -R g+w $$DIR; \
+	\
+	export DIR=$(DOC_BASE)/gstreamer/head/$(DOC); \
+	echo Uploading $$SRC to $(DOC_SERVER):$$DIR; \
+	ssh $(DOC_SERVER) mkdir -p $$DIR; \
+	rsync -rv -e ssh --delete $$SRC $(DOC_SERVER):$$DIR; \
+	ssh $(DOC_SERVER) chmod -R g+w $$DIR; \
+	\
+	if echo $(FORMATS) | grep html > /dev/null; then \
+	  echo "Un-preparing docs for upload (rebasing cross-references) ..." ; \
+	  gtkdoc-rebase --html-dir=$(builddir)/html ; \
+	fi; \
+	echo Done
+
+@ENABLE_GTK_DOC_TRUE@all-local: html-build.stamp
+
+#### setup ####
+
+@ENABLE_GTK_DOC_TRUE@setup-build.stamp: $(content_files)
+@ENABLE_GTK_DOC_TRUE@	-@if test "$(abs_srcdir)" != "$(abs_builddir)" ; then \
+@ENABLE_GTK_DOC_TRUE@	    echo '  DOC   Preparing build'; \
+@ENABLE_GTK_DOC_TRUE@	    files=`echo $(DOC_MAIN_SGML_FILE) $(DOC_OVERRIDES) $(DOC_MODULE)-sections.txt $(DOC_MODULE).types $(content_files)`; \
+@ENABLE_GTK_DOC_TRUE@	    if test "x$$files" != "x" ; then \
+@ENABLE_GTK_DOC_TRUE@	        for file in $$files ; do \
+@ENABLE_GTK_DOC_TRUE@	            test -f $(abs_srcdir)/$$file && \
+@ENABLE_GTK_DOC_TRUE@	                cp -pu $(abs_srcdir)/$$file $(abs_builddir)/ || true; \
+@ENABLE_GTK_DOC_TRUE@	        done; \
+@ENABLE_GTK_DOC_TRUE@	    fi; \
+@ENABLE_GTK_DOC_TRUE@	fi
+@ENABLE_GTK_DOC_TRUE@	@touch setup-build.stamp
+
+#### scan ####
+
+# in the case of non-srcdir builds, the built gst directory gets added
+# to gtk-doc scanning; but only then, to avoid duplicates
+@ENABLE_GTK_DOC_TRUE@scan-build.stamp: $(HFILE_GLOB) $(CFILE_GLOB)
+@ENABLE_GTK_DOC_TRUE@	@echo '  DOC   Scanning header files'
+@ENABLE_GTK_DOC_TRUE@	@_source_dir='' ;						\
+@ENABLE_GTK_DOC_TRUE@	for i in $(DOC_SOURCE_DIR) ; do					\
+@ENABLE_GTK_DOC_TRUE@	    _source_dir="$${_source_dir} --source-dir=$$i" ;	        \
+@ENABLE_GTK_DOC_TRUE@	done ;							        \
+@ENABLE_GTK_DOC_TRUE@	gtkdoc-scan							\
+@ENABLE_GTK_DOC_TRUE@		$(SCAN_OPTIONS) $(EXTRA_HFILES)				\
+@ENABLE_GTK_DOC_TRUE@		--module=$(DOC_MODULE)					\
+@ENABLE_GTK_DOC_TRUE@		$${_source_dir}         				\
+@ENABLE_GTK_DOC_TRUE@		--ignore-headers="$(IGNORE_HFILES)"
+@ENABLE_GTK_DOC_TRUE@	@if grep -l '^..*$$' $(DOC_MODULE).types > /dev/null; then	\
+@ENABLE_GTK_DOC_TRUE@	    echo "  DOC   Introspecting gobjects"; \
+@ENABLE_GTK_DOC_TRUE@	    GST_PLUGIN_SYSTEM_PATH_1_0=`cd $(top_builddir) && pwd`		\
+@ENABLE_GTK_DOC_TRUE@	    GST_PLUGIN_PATH_1_0=						\
+@ENABLE_GTK_DOC_TRUE@	    GST_REGISTRY_1_0=doc-registry.xml				\
+@ENABLE_GTK_DOC_TRUE@	    $(GTKDOC_EXTRA_ENVIRONMENT)					\
+@ENABLE_GTK_DOC_TRUE@	    CC="$(GTKDOC_CC)" LD="$(GTKDOC_LD)"				\
+@ENABLE_GTK_DOC_TRUE@	    CFLAGS="$(GTKDOC_CFLAGS) $(CFLAGS)"				\
+@ENABLE_GTK_DOC_TRUE@	    LDFLAGS="$(GTKDOC_LIBS) $(LDFLAGS)"				\
+@ENABLE_GTK_DOC_TRUE@	    gtkdoc-scangobj --type-init-func="gst_init(NULL,NULL)"	\
+@ENABLE_GTK_DOC_TRUE@	        --module=$(DOC_MODULE) ;				\
+@ENABLE_GTK_DOC_TRUE@	else								\
+@ENABLE_GTK_DOC_TRUE@	    for i in $(SCANOBJ_FILES) ; do				\
+@ENABLE_GTK_DOC_TRUE@	       test -f $$i || touch $$i ;				\
+@ENABLE_GTK_DOC_TRUE@	    done							\
+@ENABLE_GTK_DOC_TRUE@	fi
+@ENABLE_GTK_DOC_TRUE@	@touch scan-build.stamp
+
+@ENABLE_GTK_DOC_TRUE@$(DOC_MODULE)-decl.txt $(SCANOBJ_FILES) $(DOC_MODULE)-sections.txt $(DOC_MODULE)-overrides.txt: scan-build.stamp
+@ENABLE_GTK_DOC_TRUE@	@true
+
+#### xml ####
+
+@ENABLE_GTK_DOC_TRUE@sgml-build.stamp: setup-build.stamp $(DOC_MODULE)-decl.txt $(SCANOBJ_FILES) $(DOC_MODULE)-sections.txt $(expand_content_files)
+@ENABLE_GTK_DOC_TRUE@	@echo '  DOC   Building XML'
+@ENABLE_GTK_DOC_TRUE@	@gtkdoc-mkdb --module=$(DOC_MODULE) --source-dir=$(DOC_SOURCE_DIR)  --expand-content-files="$(expand_content_files)" --main-sgml-file=$(DOC_MAIN_SGML_FILE) --output-format=xml $(MKDB_OPTIONS)
+@ENABLE_GTK_DOC_TRUE@	@cp ../version.entities xml
+@ENABLE_GTK_DOC_TRUE@	@touch sgml-build.stamp
+
+@ENABLE_GTK_DOC_TRUE@sgml.stamp: sgml-build.stamp
+@ENABLE_GTK_DOC_TRUE@	@true
+
+#### html ####
+
+@ENABLE_GTK_DOC_TRUE@html-build.stamp: sgml.stamp $(DOC_MAIN_SGML_FILE) $(content_files)
+@ENABLE_GTK_DOC_TRUE@	@echo '  DOC   Building HTML'
+@ENABLE_GTK_DOC_TRUE@	@rm -rf html
+@ENABLE_GTK_DOC_TRUE@	@mkdir html
+@ENABLE_GTK_DOC_TRUE@	@cp -pr xml html
+@ENABLE_GTK_DOC_TRUE@	@cp ../version.entities ./
+@ENABLE_GTK_DOC_TRUE@	@mkhtml_options=""; \
+@ENABLE_GTK_DOC_TRUE@	gtkdoc-mkhtml 2>&1 --help | grep  >/dev/null "\-\-verbose"; \
+@ENABLE_GTK_DOC_TRUE@	if test "$(?)" = "0"; then \
+@ENABLE_GTK_DOC_TRUE@	  if test "x$(V)" = "x1"; then \
+@ENABLE_GTK_DOC_TRUE@	    mkhtml_options="$$mkhtml_options --verbose"; \
+@ENABLE_GTK_DOC_TRUE@	  fi; \
+@ENABLE_GTK_DOC_TRUE@	fi; \
+@ENABLE_GTK_DOC_TRUE@	@gtkdoc-mkhtml 2>&1 --help | grep  >/dev/null "\-\-path"; \
+@ENABLE_GTK_DOC_TRUE@	if test "$(?)" = "0"; then \
+@ENABLE_GTK_DOC_TRUE@	  mkhtml_options=--path="$(abs_srcdir)"; \
+@ENABLE_GTK_DOC_TRUE@	fi; \
+@ENABLE_GTK_DOC_TRUE@	cd html && gtkdoc-mkhtml $$mkhtml_options $(MKHTML_OPTIONS) $(DOC_MODULE)-@GST_API_VERSION@ ../$(DOC_MAIN_SGML_FILE)
+@ENABLE_GTK_DOC_TRUE@	@rm -rf html/xml
+@ENABLE_GTK_DOC_TRUE@	@rm -f version.entities
+@ENABLE_GTK_DOC_TRUE@	@test "x$(HTML_IMAGES)" = "x" ||  ( cd $(srcdir) && cp $(HTML_IMAGES) $(abs_builddir)/html )
+@ENABLE_GTK_DOC_TRUE@	@echo '  DOC   Fixing cross-references'
+@ENABLE_GTK_DOC_TRUE@	@gtkdoc-fixxref --module=$(DOC_MODULE) --module-dir=html --html-dir=$(HTML_DIR) $(FIXXREF_OPTIONS)
+@ENABLE_GTK_DOC_TRUE@	@touch html-build.stamp
+
+@ENABLE_GTK_DOC_TRUE@clean-local-gtkdoc:
+@ENABLE_GTK_DOC_TRUE@	@rm -rf xml tmpl html
+# clean files copied for nonsrcdir templates build
+@ENABLE_GTK_DOC_TRUE@	@if test x"$(srcdir)" != x. ; then \
+@ENABLE_GTK_DOC_TRUE@	        rm -rf $(DOC_MODULE).types; \
+@ENABLE_GTK_DOC_TRUE@	fi
+@ENABLE_GTK_DOC_FALSE@all-local:
+@ENABLE_GTK_DOC_FALSE@clean-local-gtkdoc:
+
+clean-local: clean-local-gtkdoc
+	@rm -f *~ *.bak
+	@rm -rf .libs
+
+distclean-local:
+	@rm -f $(REPORT_FILES) \
+	        $(DOC_MODULE)-decl-list.txt $(DOC_MODULE)-decl.txt
+	@rm -rf tmpl/*.sgml.bak
+	@rm -f $(DOC_MODULE).hierarchy
+	@rm -f *.stamp || true
+	@if test "$(abs_srcdir)" != "$(abs_builddir)" ; then \
+	    rm -f $(DOC_MAIN_SGML_FILE) ; \
+	    rm -f $(DOC_OVERRIDES) ; \
+	    rm -f $(DOC_MODULE).types ; \
+	    rm -f $(DOC_MODULE).interfaces ; \
+	    rm -f $(DOC_MODULE).prerequisites ; \
+	    rm -f $(DOC_MODULE)-sections.txt ; \
+	    rm -f $(content_files) ; \
+	    rm -rf tmpl/*.sgml ; \
+	fi
+	@rm -rf *.o
+
+maintainer-clean-local: clean
+	@cd $(srcdir) && rm -rf html \
+		xml $(DOC_MODULE)-decl-list.txt $(DOC_MODULE)-decl.txt
+
+# thomas: make docs parallel installable; devhelp requires majorminor too
+install-data-local:
+	(installfiles=`echo $(builddir)/html/*.sgml $(builddir)/html/*.html $(builddir)/html/*.png $(builddir)/html/*.css`; \
+	if test "$$installfiles" = '$(builddir)/html/*.sgml $(builddir)/html/*.html $(builddir)/html/*.png $(builddir)/html/*.css'; \
+	then echo '-- Nothing to install' ; \
+	else \
+	  $(mkinstalldirs) $(DESTDIR)$(TARGET_DIR); \
+	  for i in $$installfiles; do \
+	    echo '-- Installing '$$i ; \
+	    $(INSTALL_DATA) $$i $(DESTDIR)$(TARGET_DIR); \
+	  done; \
+	  echo '-- Installing $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2' ; \
+	  if test -e $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2; then \
+	            $(INSTALL_DATA) $(builddir)/html/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2 \
+	            $(DESTDIR)$(TARGET_DIR)/$(DOC_MODULE)-@GST_API_VERSION@.devhelp2; \
+	  fi; \
+	  $(GTKDOC_REBASE) --relative --dest-dir=$(DESTDIR) --html-dir=$(DESTDIR)$(TARGET_DIR) || true ; \
+	fi)
+uninstall-local:
+	if test -d $(DESTDIR)$(TARGET_DIR); then \
+	  rm -rf $(DESTDIR)$(TARGET_DIR)/*; \
+	  rmdir -p $(DESTDIR)$(TARGET_DIR) 2>/dev/null || true; \
+	else \
+	  echo '-- Nothing to uninstall' ; \
+	fi;
+
+#
+# Require gtk-doc when making dist
+#
+@ENABLE_GTK_DOC_TRUE@dist-check-gtkdoc:
+@ENABLE_GTK_DOC_FALSE@dist-check-gtkdoc:
+@ENABLE_GTK_DOC_FALSE@	@echo "*** gtk-doc must be installed and enabled in order to make dist"
+@ENABLE_GTK_DOC_FALSE@	@false
+
+dist-hook: dist-check-gtkdoc dist-hook-local
+	mkdir $(distdir)/html
+	cp html/* $(distdir)/html
+	-cp $(srcdir)/$(DOC_MODULE).types $(distdir)/
+	-cp $(srcdir)/$(DOC_MODULE)-sections.txt $(distdir)/
+	cd $(distdir) && rm -f $(DISTCLEANFILES)
+	-gtkdoc-rebase --online --relative --html-dir=$(distdir)/html
+
+.PHONY : dist-hook-local docs
+
+# avoid spurious build errors when distchecking with -jN
+.NOTPARALLEL:
+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
diff --git a/docs/libs/gst-rtsp-server-docs.sgml b/docs/libs/gst-rtsp-server-docs.sgml
new file mode 100644
index 0000000..06141f8
--- /dev/null
+++ b/docs/libs/gst-rtsp-server-docs.sgml
@@ -0,0 +1,49 @@
+
+
+%version-entities;
+]>
+
+
+  
+    GStreamer RTSP Server Reference Manual
+    
+      for GStreamer RTSP Server &GST_VERSION;
+    
+  
+
+  
+    
+    
+    
+    
+    
+    
+    
+    
+    
+    
+    
+    
+    
+    
+    
+    
+    
+    
+    
+  
+
+  
+    Object Hierarchy
+    
+  
+
+  
+    API Index
+    
+  
+
+  
+
diff --git a/docs/libs/gst-rtsp-server-overrides.txt b/docs/libs/gst-rtsp-server-overrides.txt
new file mode 100644
index 0000000..e69de29
diff --git a/docs/libs/gst-rtsp-server-sections.txt b/docs/libs/gst-rtsp-server-sections.txt
new file mode 100644
index 0000000..e832c4e
--- /dev/null
+++ b/docs/libs/gst-rtsp-server-sections.txt
@@ -0,0 +1,697 @@
+

+rtsp-address-pool
+GstRTSPAddressPool
+
+
+GST_RTSP_ADDRESS_POOL_ANY_IPV4
+GST_RTSP_ADDRESS_POOL_ANY_IPV6
+GstRTSPAddress
+GstRTSPAddressFlags
+gst_rtsp_address_copy
+gst_rtsp_address_free
+
+
+GstRTSPAddressPool
+GstRTSPAddressPoolClass
+GstRTSPAddressPoolResult
+gst_rtsp_address_pool_new
+gst_rtsp_address_pool_clear
+gst_rtsp_address_pool_dump
+gst_rtsp_address_pool_add_range
+gst_rtsp_address_pool_has_unicast_addresses
+gst_rtsp_address_pool_acquire_address
+gst_rtsp_address_pool_reserve_address
+
+GST_RTSP_ADDRESS_POOL_CAST
+GST_RTSP_ADDRESS_POOL_CLASS_CAST
+GST_IS_RTSP_ADDRESS_POOL
+GST_IS_RTSP_ADDRESS_POOL_CLASS
+GST_RTSP_ADDRESS_POOL
+GST_RTSP_ADDRESS_POOL_CLASS
+GST_RTSP_ADDRESS_POOL_GET_CLASS
+GST_TYPE_RTSP_ADDRESS_POOL
+GstRTSPAddressPoolPrivate
+gst_rtsp_address_get_type
+gst_rtsp_address_pool_get_type
+

+
+

+rtsp-auth
+GstRTSPAuth
+GstRTSPAuth
+GstRTSPAuthClass
+gst_rtsp_auth_new
+
+gst_rtsp_auth_get_tls_certificate
+gst_rtsp_auth_set_tls_certificate
+gst_rtsp_auth_make_basic
+gst_rtsp_auth_add_basic
+gst_rtsp_auth_remove_basic
+gst_rtsp_auth_check
+gst_rtsp_auth_get_default_token
+gst_rtsp_auth_set_default_token
+
+
+GST_RTSP_AUTH_CHECK_CONNECT
+GST_RTSP_AUTH_CHECK_URL
+GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS
+GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT
+GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS
+
+
+GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE
+GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS
+
+
+GST_RTSP_PERM_MEDIA_FACTORY_ACCESS
+GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT
+
+GST_RTSP_AUTH_CAST
+GST_RTSP_AUTH_CLASS_CAST
+GST_IS_RTSP_AUTH
+GST_IS_RTSP_AUTH_CLASS
+GST_RTSP_AUTH
+GST_RTSP_AUTH_CLASS
+GST_RTSP_AUTH_GET_CLASS
+GST_TYPE_RTSP_AUTH
+GstRTSPAuthPrivate
+gst_rtsp_auth_get_type
+

+
+

+rtsp-client
+GstRTSPClient
+
+GstRTSPClient
+GstRTSPClientClass
+
+gst_rtsp_client_new
+gst_rtsp_client_close
+
+gst_rtsp_client_get_session_pool
+gst_rtsp_client_set_session_pool
+
+gst_rtsp_client_get_mount_points
+gst_rtsp_client_set_mount_points
+
+gst_rtsp_client_get_auth
+gst_rtsp_client_set_auth
+
+gst_rtsp_client_get_thread_pool
+gst_rtsp_client_set_thread_pool
+
+gst_rtsp_client_get_connection
+gst_rtsp_client_set_connection
+
+gst_rtsp_client_attach
+
+GstRTSPClientSendFunc
+gst_rtsp_client_set_send_func
+
+gst_rtsp_client_handle_message
+gst_rtsp_client_send_message
+
+GstRTSPClientSessionFilterFunc
+gst_rtsp_client_session_filter
+
+GST_RTSP_CLIENT_CAST
+GST_RTSP_CLIENT_CLASS_CAST
+GST_IS_RTSP_CLIENT
+GST_IS_RTSP_CLIENT_CLASS
+GST_RTSP_CLIENT
+GST_RTSP_CLIENT_CLASS
+GST_RTSP_CLIENT_GET_CLASS
+GST_TYPE_RTSP_CLIENT
+GstRTSPClientPrivate
+gst_rtsp_client_get_type
+

+
+

+rtsp-context
+GstRTSPContext
+GstRTSPContext
+gst_rtsp_context_get_current
+gst_rtsp_context_push_current
+gst_rtsp_context_pop_current
+
+GST_TYPE_RTSP_CONTEXT
+gst_rtsp_context_get_type
+

+
+

+rtsp-media
+GstRTSPMedia
+GstRTSPMedia
+GstRTSPMediaClass
+gst_rtsp_media_new
+gst_rtsp_media_get_element
+gst_rtsp_media_take_pipeline
+
+gst_rtsp_media_set_permissions
+gst_rtsp_media_get_permissions
+
+gst_rtsp_media_set_shared
+gst_rtsp_media_is_shared
+
+gst_rtsp_media_set_reusable
+gst_rtsp_media_is_reusable
+
+gst_rtsp_media_set_profiles
+gst_rtsp_media_get_profiles
+
+gst_rtsp_media_set_protocols
+gst_rtsp_media_get_protocols
+
+gst_rtsp_media_set_eos_shutdown
+gst_rtsp_media_is_eos_shutdown
+
+gst_rtsp_media_set_address_pool
+gst_rtsp_media_get_address_pool
+
+gst_rtsp_media_set_buffer_size
+gst_rtsp_media_get_buffer_size
+
+gst_rtsp_media_setup_sdp
+
+
+gst_rtsp_media_prepare
+gst_rtsp_media_unprepare
+GstRTSPMediaStatus
+gst_rtsp_media_get_status
+
+
+gst_rtsp_media_set_suspend_mode
+gst_rtsp_media_get_suspend_mode
+
+GstRTSPSuspendMode
+gst_rtsp_media_suspend
+gst_rtsp_media_unsuspend
+
+
+gst_rtsp_media_collect_streams
+gst_rtsp_media_create_stream
+gst_rtsp_media_n_streams
+gst_rtsp_media_get_stream
+gst_rtsp_media_find_stream
+
+
+gst_rtsp_media_seek
+gst_rtsp_media_get_range_string
+
+gst_rtsp_media_set_state
+gst_rtsp_media_set_pipeline_state
+
+
+gst_rtsp_media_get_clock
+gst_rtsp_media_get_base_time
+gst_rtsp_media_use_time_provider
+gst_rtsp_media_is_time_provider
+gst_rtsp_media_get_time_provider
+
+GST_RTSP_MEDIA_CAST
+GST_RTSP_MEDIA_CLASS_CAST
+GST_IS_RTSP_MEDIA
+GST_IS_RTSP_MEDIA_CLASS
+GST_RTSP_MEDIA
+GST_RTSP_MEDIA_CLASS
+GST_RTSP_MEDIA_GET_CLASS
+GST_TYPE_RTSP_MEDIA
+GstRTSPMediaPrivate
+gst_rtsp_media_get_type
+GST_TYPE_RTSP_SUSPEND_MODE
+gst_rtsp_suspend_mode_get_type
+

+
+

+rtsp-media-factory
+GstRTSPMediaFactory
+GstRTSPMediaFactory
+GstRTSPMediaFactoryClass
+gst_rtsp_media_factory_new
+
+gst_rtsp_media_factory_get_launch
+gst_rtsp_media_factory_set_launch
+
+gst_rtsp_media_factory_get_permissions
+gst_rtsp_media_factory_set_permissions
+gst_rtsp_media_factory_add_role
+
+gst_rtsp_media_factory_set_shared
+gst_rtsp_media_factory_is_shared
+
+gst_rtsp_media_factory_is_eos_shutdown
+gst_rtsp_media_factory_set_eos_shutdown
+
+gst_rtsp_media_factory_get_protocols
+gst_rtsp_media_factory_set_protocols
+
+gst_rtsp_media_factory_set_profiles
+gst_rtsp_media_factory_get_profiles
+
+gst_rtsp_media_factory_get_address_pool
+gst_rtsp_media_factory_set_address_pool
+
+gst_rtsp_media_factory_get_buffer_size
+gst_rtsp_media_factory_set_buffer_size
+
+gst_rtsp_media_factory_get_suspend_mode
+gst_rtsp_media_factory_set_suspend_mode
+
+gst_rtsp_media_factory_construct
+gst_rtsp_media_factory_create_element
+
+
+GST_RTSP_MEDIA_FACTORY_CAST
+GST_RTSP_MEDIA_FACTORY_CLASS_CAST
+GST_IS_RTSP_MEDIA_FACTORY
+GST_IS_RTSP_MEDIA_FACTORY_CLASS
+GST_RTSP_MEDIA_FACTORY
+GST_RTSP_MEDIA_FACTORY_CLASS
+GST_RTSP_MEDIA_FACTORY_GET_CLASS
+GST_TYPE_RTSP_MEDIA_FACTORY
+GstRTSPMediaFactoryPrivate
+gst_rtsp_media_factory_get_type
+

+
+

+rtsp-media-factory-uri
+GstRTSPMediaFactoryURI
+GstRTSPMediaFactoryURI
+GstRTSPMediaFactoryURIClass
+gst_rtsp_media_factory_uri_new
+gst_rtsp_media_factory_uri_set_uri
+gst_rtsp_media_factory_uri_get_uri
+
+GST_RTSP_MEDIA_FACTORY_URI_CAST
+GST_RTSP_MEDIA_FACTORY_URI_CLASS_CAST
+GST_IS_RTSP_MEDIA_FACTORY_URI
+GST_IS_RTSP_MEDIA_FACTORY_URI_CLASS
+GST_RTSP_MEDIA_FACTORY_URI
+GST_RTSP_MEDIA_FACTORY_URI_CLASS
+GST_RTSP_MEDIA_FACTORY_URI_GET_CLASS
+GST_TYPE_RTSP_MEDIA_FACTORY_URI
+GstRTSPMediaFactoryURIPrivate
+gst_rtsp_media_factory_uri_get_type
+

+
+

+rtsp-mount-points
+GstRTSPMountPoints
+GstRTSPMountPoints
+GstRTSPMountPointsClass
+gst_rtsp_mount_points_new
+gst_rtsp_mount_points_add_factory
+gst_rtsp_mount_points_remove_factory
+gst_rtsp_mount_points_match
+gst_rtsp_mount_points_make_path
+
+GST_RTSP_MOUNT_POINTS_CAST
+GST_RTSP_MOUNT_POINTS_CLASS_CAST
+GST_IS_RTSP_MOUNT_POINTS
+GST_IS_RTSP_MOUNT_POINTS_CLASS
+GST_RTSP_MOUNT_POINTS
+GST_RTSP_MOUNT_POINTS_CLASS
+GST_RTSP_MOUNT_POINTS_GET_CLASS
+GST_TYPE_RTSP_MOUNT_POINTS
+GstRTSPMountPointsPrivate
+gst_rtsp_mount_points_get_type
+

+
+

+rtsp-params
+GstRTSPParams
+gst_rtsp_params_get
+gst_rtsp_params_set
+

+
+

+rtsp-permissions
+GstRTSPPermissions
+GstRTSPPermissions
+gst_rtsp_permissions_new
+gst_rtsp_permissions_ref
+gst_rtsp_permissions_unref
+gst_rtsp_permissions_add_role
+gst_rtsp_permissions_add_role_valist
+gst_rtsp_permissions_remove_role
+gst_rtsp_permissions_get_role
+gst_rtsp_permissions_is_allowed
+
+GST_RTSP_PERMISSIONS_CAST
+GST_IS_RTSP_PERMISSIONS
+GST_RTSP_PERMISSIONS
+GST_TYPE_RTSP_PERMISSIONS
+gst_rtsp_permissions_get_type
+

+
+

+rtsp-sdp
+GstRTSPSdp
+GstSDPInfo
+gst_rtsp_sdp_from_media
+

+
+

+rtsp-server
+GstRTSPServer
+GstRTSPServer
+GstRTSPServerClass
+
+gst_rtsp_server_new
+
+gst_rtsp_server_get_address
+gst_rtsp_server_set_address
+
+gst_rtsp_server_get_service
+gst_rtsp_server_set_service
+
+gst_rtsp_server_get_backlog
+gst_rtsp_server_set_backlog
+
+gst_rtsp_server_get_bound_port
+
+gst_rtsp_server_get_mount_points
+gst_rtsp_server_set_mount_points
+
+gst_rtsp_server_get_session_pool
+gst_rtsp_server_set_session_pool
+
+gst_rtsp_server_get_thread_pool
+gst_rtsp_server_set_thread_pool
+
+gst_rtsp_server_get_auth
+gst_rtsp_server_set_auth
+
+gst_rtsp_server_transfer_connection
+gst_rtsp_server_io_func
+gst_rtsp_server_create_socket
+gst_rtsp_server_create_source
+gst_rtsp_server_attach
+
+GstRTSPServerClientFilterFunc
+gst_rtsp_server_client_filter
+
+
+GST_IS_RTSP_SERVER
+GST_RTSP_SERVER_CAST
+GST_RTSP_SERVER_CLASS_CAST
+GST_IS_RTSP_SERVER_CLASS
+GST_RTSP_SERVER
+GST_RTSP_SERVER_CLASS
+GST_RTSP_SERVER_GET_CLASS
+GST_TYPE_RTSP_SERVER
+GstRTSPServerPrivate
+gst_rtsp_server_get_type
+

+
+

+rtsp-session
+GstRTSPSession
+GstRTSPSession
+GstRTSPSessionClass
+
+gst_rtsp_session_new
+gst_rtsp_session_get_sessionid
+
+gst_rtsp_session_get_header
+
+gst_rtsp_session_set_timeout
+gst_rtsp_session_get_timeout
+
+gst_rtsp_session_touch
+gst_rtsp_session_prevent_expire
+gst_rtsp_session_allow_expire
+gst_rtsp_session_next_timeout
+gst_rtsp_session_is_expired
+
+gst_rtsp_session_manage_media
+gst_rtsp_session_release_media
+
+gst_rtsp_session_get_media
+
+GstRTSPFilterResult
+GstRTSPSessionFilterFunc
+gst_rtsp_session_filter
+
+GST_RTSP_SESSION_CAST
+GST_RTSP_SESSION_CLASS_CAST
+GST_IS_RTSP_SESSION
+GST_IS_RTSP_SESSION_CLASS
+GST_RTSP_SESSION
+GST_RTSP_SESSION_CLASS
+GST_RTSP_SESSION_GET_CLASS
+GST_TYPE_RTSP_SESSION
+GstRTSPSessionPrivate
+gst_rtsp_session_get_type
+

+
+

+rtsp-session-media
+GstRTSPSessionMedia
+GstRTSPSessionMedia
+GstRTSPSessionMediaClass
+gst_rtsp_session_media_new
+gst_rtsp_session_media_matches
+
+gst_rtsp_session_media_get_media
+gst_rtsp_session_media_get_base_time
+gst_rtsp_session_media_get_rtpinfo
+
+gst_rtsp_session_media_set_state
+
+gst_rtsp_session_media_get_rtsp_state
+gst_rtsp_session_media_set_rtsp_state
+
+gst_rtsp_session_media_get_transport
+gst_rtsp_session_media_set_transport
+
+gst_rtsp_session_media_alloc_channels
+
+GST_RTSP_SESSION_MEDIA_CAST
+GST_RTSP_SESSION_MEDIA_CLASS_CAST
+GST_IS_RTSP_SESSION_MEDIA
+GST_IS_RTSP_SESSION_MEDIA_CLASS
+GST_RTSP_SESSION_MEDIA
+GST_RTSP_SESSION_MEDIA_CLASS
+GST_RTSP_SESSION_MEDIA_GET_CLASS
+GST_TYPE_RTSP_SESSION_MEDIA
+GstRTSPSessionMediaPrivate
+gst_rtsp_session_media_get_type
+

+
+

+rtsp-session-pool
+GstRTSPSessionPool
+GstRTSPSessionPool
+GstRTSPSessionPoolClass
+gst_rtsp_session_pool_new
+
+gst_rtsp_session_pool_get_max_sessions
+gst_rtsp_session_pool_set_max_sessions
+
+gst_rtsp_session_pool_get_n_sessions
+
+gst_rtsp_session_pool_create
+gst_rtsp_session_pool_find
+gst_rtsp_session_pool_remove
+
+gst_rtsp_session_pool_cleanup
+
+GstRTSPSessionPoolFunc
+gst_rtsp_session_pool_create_watch
+
+GstRTSPSessionPoolFilterFunc
+gst_rtsp_session_pool_filter
+
+GST_RTSP_SESSION_POOL_CAST
+GST_RTSP_SESSION_POOL_CLASS_CAST
+GST_IS_RTSP_SESSION_POOL
+GST_IS_RTSP_SESSION_POOL_CLASS
+GST_RTSP_SESSION_POOL
+GST_RTSP_SESSION_POOL_CLASS
+GST_RTSP_SESSION_POOL_GET_CLASS
+GST_TYPE_RTSP_SESSION_POOL
+GstRTSPSessionPoolPrivate
+gst_rtsp_session_pool_get_type
+

+
+

+rtsp-stream
+GstRTSPStream
+GstRTSPStream
+GstRTSPStreamClass
+gst_rtsp_stream_new
+
+gst_rtsp_stream_get_index
+gst_rtsp_stream_get_srcpad
+
+gst_rtsp_stream_get_control
+gst_rtsp_stream_set_control
+gst_rtsp_stream_has_control
+
+gst_rtsp_stream_get_mtu
+gst_rtsp_stream_set_mtu
+
+gst_rtsp_stream_get_dscp_qos
+gst_rtsp_stream_set_dscp_qos
+
+gst_rtsp_stream_set_profiles
+gst_rtsp_stream_get_profiles
+
+gst_rtsp_stream_get_protocols
+gst_rtsp_stream_set_protocols
+
+gst_rtsp_stream_is_transport_supported
+
+gst_rtsp_stream_get_address_pool
+gst_rtsp_stream_set_address_pool
+gst_rtsp_stream_reserve_address
+
+gst_rtsp_stream_join_bin
+gst_rtsp_stream_leave_bin
+
+gst_rtsp_stream_get_server_port
+gst_rtsp_stream_get_multicast_address
+gst_rtsp_stream_get_rtpsession
+gst_rtsp_stream_get_ssrc
+gst_rtsp_stream_get_rtpinfo
+gst_rtsp_stream_get_caps
+gst_rtsp_stream_get_pt
+
+gst_rtsp_stream_recv_rtcp
+gst_rtsp_stream_recv_rtp
+
+gst_rtsp_stream_add_transport
+gst_rtsp_stream_remove_transport
+
+gst_rtsp_stream_get_rtp_socket
+gst_rtsp_stream_get_rtcp_socket
+
+gst_rtsp_stream_set_blocked
+gst_rtsp_stream_is_blocking
+
+gst_rtsp_stream_update_crypto
+
+GstRTSPStreamTransportFilterFunc
+gst_rtsp_stream_transport_filter
+
+
+GST_RTSP_STREAM_CAST
+GST_RTSP_STREAM_CLASS_CAST
+GST_IS_RTSP_STREAM
+GST_IS_RTSP_STREAM_CLASS
+GST_RTSP_STREAM
+GST_RTSP_STREAM_CLASS
+GST_RTSP_STREAM_GET_CLASS
+GST_TYPE_RTSP_STREAM
+GstRTSPStreamPrivate
+gst_rtsp_stream_get_type
+

+
+

+rtsp-stream-transport
+GstRTSPStreamTransport
+GstRTSPStreamTransport
+GstRTSPStreamTransportClass
+gst_rtsp_stream_transport_new
+
+gst_rtsp_stream_transport_get_stream
+
+gst_rtsp_stream_transport_get_transport
+gst_rtsp_stream_transport_set_transport
+
+gst_rtsp_stream_transport_get_url
+gst_rtsp_stream_transport_set_url
+
+gst_rtsp_stream_transport_get_rtpinfo
+
+GstRTSPSendFunc
+gst_rtsp_stream_transport_set_callbacks
+
+GstRTSPKeepAliveFunc
+gst_rtsp_stream_transport_set_keepalive
+gst_rtsp_stream_transport_keep_alive
+
+gst_rtsp_stream_transport_set_active
+
+gst_rtsp_stream_transport_set_timed_out
+gst_rtsp_stream_transport_is_timed_out
+
+gst_rtsp_stream_transport_send_rtcp
+gst_rtsp_stream_transport_send_rtp
+
+
+GST_RTSP_STREAM_TRANSPORT_CAST
+GST_RTSP_STREAM_TRANSPORT_CLASS_CAST
+GST_IS_RTSP_STREAM_TRANSPORT
+GST_IS_RTSP_STREAM_TRANSPORT_CLASS
+GST_RTSP_STREAM_TRANSPORT
+GST_RTSP_STREAM_TRANSPORT_CLASS
+GST_RTSP_STREAM_TRANSPORT_GET_CLASS
+GST_TYPE_RTSP_STREAM_TRANSPORT
+GstRTSPStreamTransportPrivate
+gst_rtsp_stream_transport_get_type
+

+
+

+rtsp-thread-pool
+GstRTSPThreadPool
+
+GstRTSPThreadType
+GstRTSPThread
+
+gst_rtsp_thread_new
+gst_rtsp_thread_ref
+gst_rtsp_thread_unref
+gst_rtsp_thread_reuse
+gst_rtsp_thread_stop
+
+
+GstRTSPThreadPool
+GstRTSPThreadPoolClass
+gst_rtsp_thread_pool_new
+
+gst_rtsp_thread_pool_get_max_threads
+gst_rtsp_thread_pool_set_max_threads
+
+gst_rtsp_thread_pool_get_thread
+gst_rtsp_thread_pool_cleanup
+
+GST_RTSP_THREAD_CAST
+GST_RTSP_THREAD_POOL_CAST
+GST_RTSP_THREAD_POOL_CLASS_CAST
+GST_IS_RTSP_THREAD
+GST_IS_RTSP_THREAD_POOL
+GST_IS_RTSP_THREAD_POOL_CLASS
+GST_RTSP_THREAD
+GST_RTSP_THREAD_POOL
+GST_RTSP_THREAD_POOL_CLASS
+GST_RTSP_THREAD_POOL_GET_CLASS
+GST_TYPE_RTSP_THREAD
+GST_TYPE_RTSP_THREAD_POOL
+GstRTSPThreadPoolPrivate
+gst_rtsp_thread_get_type
+gst_rtsp_thread_pool_get_type
+

+
+

+rtsp-token
+GstRTSPToken
+GstRTSPToken
+gst_rtsp_token_new_empty
+gst_rtsp_token_new
+gst_rtsp_token_new_valist
+gst_rtsp_token_ref
+gst_rtsp_token_unref
+gst_rtsp_token_get_structure
+gst_rtsp_token_writable_structure
+gst_rtsp_token_get_string
+gst_rtsp_token_is_allowed
+
+GST_RTSP_TOKEN_CAST
+GST_IS_RTSP_TOKEN
+GST_RTSP_TOKEN
+GST_TYPE_RTSP_TOKEN
+gst_rtsp_token_get_type
+

+
diff --git a/docs/libs/gst-rtsp-server.types b/docs/libs/gst-rtsp-server.types
new file mode 100644
index 0000000..f4476f2
--- /dev/null
+++ b/docs/libs/gst-rtsp-server.types
@@ -0,0 +1,25 @@
+#include 
+
+#include 
+gst_rtsp_auth_get_type
+
+#include 
+gst_rtsp_mount_points_get_type
+
+#include 
+gst_rtsp_media_factory_get_type
+
+#include 
+gst_rtsp_media_get_type
+
+#include 
+gst_rtsp_server_get_type
+
+#include 
+gst_rtsp_session_pool_get_type
+
+#include 
+gst_rtsp_session_get_type
+
+#include 
+gst_rtsp_client_get_type
diff --git a/docs/libs/html/GstRTSPAuth.html b/docs/libs/html/GstRTSPAuth.html
new file mode 100644
index 0000000..b51425a
--- /dev/null
+++ b/docs/libs/html/GstRTSPAuth.html
@@ -0,0 +1,613 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPAuth
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPAuth

+
GstRTSPAuth — Authentication and authorization

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





+GstRTSPAuth *
+
+gst_rtsp_auth_new ()
+

+GTlsCertificate *
+
+gst_rtsp_auth_get_tls_certificate ()
+

+void
+
+gst_rtsp_auth_set_tls_certificate ()
+

+gchar *
+
+gst_rtsp_auth_make_basic ()
+

+void
+
+gst_rtsp_auth_add_basic ()
+

+void
+
+gst_rtsp_auth_remove_basic ()
+

+gboolean
+
+gst_rtsp_auth_check ()
+

+GstRTSPToken *
+
+gst_rtsp_auth_get_default_token ()
+

+void
+
+gst_rtsp_auth_set_default_token ()
+

+

+

+
Types and Values

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+




structGstRTSPAuth
structGstRTSPAuthClass
#defineGST_RTSP_AUTH_CHECK_CONNECT
#defineGST_RTSP_AUTH_CHECK_URL
#defineGST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS
#defineGST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT
#defineGST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS
#defineGST_RTSP_TOKEN_MEDIA_FACTORY_ROLE
#defineGST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS
#defineGST_RTSP_PERM_MEDIA_FACTORY_ACCESS
#defineGST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT

+

+

+
Object Hierarchy

+
    GObject
+    ╰── GstRTSPAuth
+

+

+

+
Description

+
The GstRTSPAuth object is responsible for checking if the current user is
+allowed to perform requested actions. The default implementation has some
+reasonable checks but subclasses can implement custom security policies.

+
A new auth object is made with gst_rtsp_auth_new(). It is usually configured
+on the GstRTSPServer object.

+
The RTSP server will call gst_rtsp_auth_check() with a string describing the
+check to perform. The possible checks are prefixed with
+GST_RTSP_AUTH_CHECK_*. Depending on the check, the default implementation
+will use the current GstRTSPToken, GstRTSPContext and
+GstRTSPPermissions on the object to check if an operation is allowed.

+
The default GstRTSPAuth object has support for basic authentication. With
+gst_rtsp_auth_add_basic() you can add a basic authentication string together
+with the GstRTSPToken that will become active when successfully
+authenticated.

+
When a TLS certificate has been set with gst_rtsp_auth_set_tls_certificate(),
+the default auth object will require the client to connect with a TLS
+connection.

+
Last reviewed on 2013-07-16 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_auth_new ()

+
GstRTSPAuth *
+gst_rtsp_auth_new (void);

+
Create a new GstRTSPAuth instance.

+

+
Returns

+
 a new GstRTSPAuth. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_auth_get_tls_certificate ()

+
GTlsCertificate *
+gst_rtsp_auth_get_tls_certificate (GstRTSPAuth *auth);

+
Get the GTlsCertificate used for negotiating TLS auth
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





auth
a GstRTSPAuth
 

+

+

+
Returns

+
 the GTlsCertificate of auth
+. g_object_unref() after
+usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_auth_set_tls_certificate ()

+
void
+gst_rtsp_auth_set_tls_certificate (GstRTSPAuth *auth,
+                                   GTlsCertificate *cert);

+
Set the TLS certificate for the auth. Client connections will only
+be accepted when TLS is negotiated.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





auth
a GstRTSPAuth
 
cert
 a GTlsCertificate. 
[transfer none][allow-none]

+

+

+

+

+
gst_rtsp_auth_make_basic ()

+
gchar *
+gst_rtsp_auth_make_basic (const gchar *user,
+                          const gchar *pass);

+
Construct a Basic authorisation token from user
+ and pass
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





user
a userid
 
pass
a password
 

+

+

+
Returns

+
 the base64 encoding of the string user
+:pass
+.
+g_free() after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_auth_add_basic ()

+
void
+gst_rtsp_auth_add_basic (GstRTSPAuth *auth,
+                         const gchar *basic,
+                         GstRTSPToken *token);

+
Add a basic token for the default authentication algorithm that
+enables the client with privileges listed in token
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





auth
a GstRTSPAuth
 
basic
the basic token
 
token
 authorisation token. 
[transfer none]

+

+

+

+

+
gst_rtsp_auth_remove_basic ()

+
void
+gst_rtsp_auth_remove_basic (GstRTSPAuth *auth,
+                            const gchar *basic);

+
Add a basic token for the default authentication algorithm that
+enables the client with privileges from authgroup
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





auth
a GstRTSPAuth
 
basic
 the basic token. 
[transfer none]

+

+

+

+

+
gst_rtsp_auth_check ()

+
gboolean
+gst_rtsp_auth_check (const gchar *check);

+
Check if check
+ is allowed in the current context.

+

+
Parameters

+

+++++
+
+
+
+
+
+





check
the item to check
 

+

+

+
Returns

+
 FALSE if check failed.

+

+

+

+

+

+
gst_rtsp_auth_get_default_token ()

+
GstRTSPToken *
+gst_rtsp_auth_get_default_token (GstRTSPAuth *auth);

+
Get the default token for auth
+. This token will be used for unauthenticated
+users.

+

+
Parameters

+

+++++
+
+
+
+
+
+





auth
a GstRTSPAuth
 

+

+

+
Returns

+
 the GstRTSPToken of auth
+. gst_rtsp_token_unref() after
+usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_auth_set_default_token ()

+
void
+gst_rtsp_auth_set_default_token (GstRTSPAuth *auth,
+                                 GstRTSPToken *token);

+
Set the default GstRTSPToken to token
+ in auth
+. The default token will
+be used for unauthenticated users.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





auth
a GstRTSPAuth
 
token
 a GstRTSPToken. 
[transfer none][allow-none]

+

+

+

+

+
Types and Values

+

+
struct GstRTSPAuth

+
struct GstRTSPAuth;

+
The authentication structure.

+

+

+

+
struct GstRTSPAuthClass

+
struct GstRTSPAuthClass {
+  GObjectClass  parent_class;
+
+  gboolean           (*authenticate) (GstRTSPAuth *auth, GstRTSPContext *ctx);
+  gboolean           (*check)        (GstRTSPAuth *auth, GstRTSPContext *ctx,
+                                      const gchar *check);
+};
+

+
The authentication class.

+

+
Members

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





GObjectClass parent_class;
  
authenticate ()
check the authentication of a client. The default implementation
+checks if the authentication in the header matches one of the basic
+authentication tokens. This function should set the authgroup field
+in the context.
 
check ()
check if a resource can be accessed. this function should
+call authenticate to authenticate the client when needed. The method
+should also construct and send an appropriate response message on
+error.
 

+

+

+

+

+
GST_RTSP_AUTH_CHECK_CONNECT

+
#define GST_RTSP_AUTH_CHECK_CONNECT                  "auth.check.connect"
+

+
Check a new connection

+

+

+

+
GST_RTSP_AUTH_CHECK_URL

+
#define GST_RTSP_AUTH_CHECK_URL                      "auth.check.url"
+

+
Check the URL and methods

+

+

+

+
GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS

+
#define GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS     "auth.check.media.factory.access"
+

+
Check if access is allowed to a factory.
+When access is not allowed an 404 Not Found is sent in the response.

+

+

+

+
GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT

+
#define GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT  "auth.check.media.factory.construct"
+

+
Check if media can be constructed from a media factory
+A response should be sent on error.

+

+

+

+
GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS

+
#define GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS  "auth.check.transport.client-settings"
+

+
Check if the client can specify TTL, destination and
+port pair in multicast. No response is sent when the check returns
+FALSE.

+

+

+

+
GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE

+
#define GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE      "media.factory.role"
+

+
G_TYPE_STRING, the role to use when dealing with media factories

+
The default GstRTSPAuth object uses this string in the token to find the
+role of the media factory. It will then retrieve the GstRTSPPermissions of
+the media factory and retrieve the role with the same name.

+

+

+

+
GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS

+
#define GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS   "transport.client-settings"
+

+
G_TYPE_BOOLEAN, TRUE if the client can specify TTL, destination and
+    port pair in multicast.

+

+

+

+
GST_RTSP_PERM_MEDIA_FACTORY_ACCESS

+
#define GST_RTSP_PERM_MEDIA_FACTORY_ACCESS      "media.factory.access"
+

+
G_TYPE_BOOLEAN, TRUE if the media can be accessed, FALSE will
+return a 404 Not Found error when trying to access the media.

+

+

+

+
GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT

+
#define GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT   "media.factory.construct"
+

+
G_TYPE_BOOLEAN, TRUE if the media can be constructed, FALSE will
+return a 404 Not Found error when trying to access the media.

+

+

+

+
See Also

+
GstRTSPPermissions, GstRTSPToken

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/GstRTSPClient.html b/docs/libs/html/GstRTSPClient.html
new file mode 100644
index 0000000..d806b44
--- /dev/null
+++ b/docs/libs/html/GstRTSPClient.html
@@ -0,0 +1,1371 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPClient
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPClient

+
GstRTSPClient — A client connection state

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





+GstRTSPClient *
+
+gst_rtsp_client_new ()
+

+void
+
+gst_rtsp_client_close ()
+

+GstRTSPSessionPool *
+
+gst_rtsp_client_get_session_pool ()
+

+void
+
+gst_rtsp_client_set_session_pool ()
+

+GstRTSPMountPoints *
+
+gst_rtsp_client_get_mount_points ()
+

+void
+
+gst_rtsp_client_set_mount_points ()
+

+GstRTSPAuth *
+
+gst_rtsp_client_get_auth ()
+

+void
+
+gst_rtsp_client_set_auth ()
+

+GstRTSPThreadPool *
+
+gst_rtsp_client_get_thread_pool ()
+

+void
+
+gst_rtsp_client_set_thread_pool ()
+

+GstRTSPConnection *
+
+gst_rtsp_client_get_connection ()
+

+gboolean
+
+gst_rtsp_client_set_connection ()
+

+guint
+
+gst_rtsp_client_attach ()
+

+gboolean
+
+(*GstRTSPClientSendFunc) ()
+

+void
+
+gst_rtsp_client_set_send_func ()
+

+GstRTSPResult
+
+gst_rtsp_client_handle_message ()
+

+GstRTSPResult
+
+gst_rtsp_client_send_message ()
+

+GstRTSPFilterResult
+
+(*GstRTSPClientSessionFilterFunc) ()
+

+GList *
+
+gst_rtsp_client_session_filter ()
+

+

+

+
Properties

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





gbooleandrop-backlogRead / Write

+GstRTSPMountPoints *mount-pointsRead / Write

+GstRTSPSessionPool *session-poolRead / Write

+

+

+
Signals

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





voidclosedRun Last
voiddescribe-requestRun Last
voidget-parameter-requestRun Last
voidhandle-responseRun Last
voidnew-sessionRun Last
voidoptions-requestRun Last
voidpause-requestRun Last
voidplay-requestRun Last
voidsend-messageRun Last
voidset-parameter-requestRun Last
voidsetup-requestRun Last
voidteardown-requestRun Last

+

+

+
Types and Values

+

++++
+
+
+
+
+
+
+
+
+
+
+




structGstRTSPClient
structGstRTSPClientClass

+

+

+
Object Hierarchy

+
    GObject
+    ╰── GstRTSPClient
+

+

+

+
Description

+
The client object handles the connection with a client for as long as a TCP
+connection is open.

+
A GstRTSPClient is created by GstRTSPServer when a new connection is
+accepted and it inherits the GstRTSPMountPoints, GstRTSPSessionPool,
+GstRTSPAuth and GstRTSPThreadPool from the server.

+
The client connection should be configured with the GstRTSPConnection using
+gst_rtsp_client_set_connection() before it can be attached to a GMainContext
+using gst_rtsp_client_attach(). From then on the client will handle requests
+on the connection.

+
Use gst_rtsp_client_session_filter() to iterate or modify all the
+GstRTSPSession objects managed by the client object.

+
Last reviewed on 2013-07-11 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_client_new ()

+
GstRTSPClient *
+gst_rtsp_client_new (void);

+
Create a new GstRTSPClient instance.

+

+
Returns

+
 a new GstRTSPClient. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_client_close ()

+
void
+gst_rtsp_client_close (GstRTSPClient *client);

+
Close the connection of client
+ and remove all media it was managing.

+

+
Parameters

+

+++++
+
+
+
+
+
+





client
a GstRTSPClient
 

+

+
Since 1.4

+

+

+

+
gst_rtsp_client_get_session_pool ()

+
GstRTSPSessionPool *
+gst_rtsp_client_get_session_pool (GstRTSPClient *client);

+
Get the GstRTSPSessionPool object that client
+ uses to manage its sessions.

+

+
Parameters

+

+++++
+
+
+
+
+
+





client
a GstRTSPClient
 

+

+

+
Returns

+
 a GstRTSPSessionPool, unref after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_client_set_session_pool ()

+
void
+gst_rtsp_client_set_session_pool (GstRTSPClient *client,
+                                  GstRTSPSessionPool *pool);

+
Set pool
+ as the sessionpool for client
+ which it will use to find
+or allocate sessions. the sessionpool is usually inherited from the server
+that created the client but can be overridden later.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





client
a GstRTSPClient
 
pool
 a GstRTSPSessionPool. 
[transfer none]

+

+

+

+

+
gst_rtsp_client_get_mount_points ()

+
GstRTSPMountPoints *
+gst_rtsp_client_get_mount_points (GstRTSPClient *client);

+
Get the GstRTSPMountPoints object that client
+ uses to manage its sessions.

+

+
Parameters

+

+++++
+
+
+
+
+
+





client
a GstRTSPClient
 

+

+

+
Returns

+
 a GstRTSPMountPoints, unref after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_client_set_mount_points ()

+
void
+gst_rtsp_client_set_mount_points (GstRTSPClient *client,
+                                  GstRTSPMountPoints *mounts);

+
Set mounts
+ as the mount points for client
+ which it will use to map urls
+to media streams. These mount points are usually inherited from the server that
+created the client but can be overriden later.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





client
a GstRTSPClient
 
mounts
 a GstRTSPMountPoints. 
[transfer none]

+

+

+

+

+
gst_rtsp_client_get_auth ()

+
GstRTSPAuth *
+gst_rtsp_client_get_auth (GstRTSPClient *client);

+
Get the GstRTSPAuth used as the authentication manager of client
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





client
a GstRTSPClient
 

+

+

+
Returns

+
 the GstRTSPAuth of client
+. g_object_unref() after
+usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_client_set_auth ()

+
void
+gst_rtsp_client_set_auth (GstRTSPClient *client,
+                          GstRTSPAuth *auth);

+
configure auth
+ to be used as the authentication manager of client
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





client
a GstRTSPClient
 
auth
 a GstRTSPAuth. 
[transfer none]

+

+

+

+

+
gst_rtsp_client_get_thread_pool ()

+
GstRTSPThreadPool *
+gst_rtsp_client_get_thread_pool (GstRTSPClient *client);

+
Get the GstRTSPThreadPool used as the thread pool of client
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





client
a GstRTSPClient
 

+

+

+
Returns

+
 the GstRTSPThreadPool of client
+. g_object_unref() after
+usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_client_set_thread_pool ()

+
void
+gst_rtsp_client_set_thread_pool (GstRTSPClient *client,
+                                 GstRTSPThreadPool *pool);

+
configure pool
+ to be used as the thread pool of client
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





client
a GstRTSPClient
 
pool
 a GstRTSPThreadPool. 
[transfer none]

+

+

+

+

+
gst_rtsp_client_get_connection ()

+
GstRTSPConnection *
+gst_rtsp_client_get_connection (GstRTSPClient *client);

+
Get the GstRTSPConnection of client
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





client
a GstRTSPClient
 

+

+

+
Returns

+
 the GstRTSPConnection of client
+.
+The connection object returned remains valid until the client is freed. 

+
[transfer none]

+

+

+

+

+
gst_rtsp_client_set_connection ()

+
gboolean
+gst_rtsp_client_set_connection (GstRTSPClient *client,
+                                GstRTSPConnection *conn);

+
Set the GstRTSPConnection of client
+. This function takes ownership of
+conn
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





client
a GstRTSPClient
 
conn
 a GstRTSPConnection. 
[transfer full]

+

+

+
Returns

+
 TRUE on success.

+

+

+

+

+

+
gst_rtsp_client_attach ()

+
guint
+gst_rtsp_client_attach (GstRTSPClient *client,
+                        GMainContext *context);

+
Attaches client
+ to context
+. When the mainloop for context
+ is run, the
+client will be dispatched. When context
+ is NULL, the default context will be
+used).

+
This function should be called when the client properties and urls are fully
+configured and the client is ready to start.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





client
a GstRTSPClient
 
context
 a GMainContext. 
[allow-none]

+

+

+
Returns

+
 the ID (greater than 0) for the source within the GMainContext.

+

+

+

+

+

+
GstRTSPClientSendFunc ()

+
gboolean
+(*GstRTSPClientSendFunc) (GstRTSPClient *client,
+                          GstRTSPMessage *message,
+                          gboolean close,
+                          gpointer user_data);

+
This callback is called when client
+ wants to send message
+. When close
+ is
+TRUE, the connection should be closed when the message has been sent.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





client
a GstRTSPClient
 
message
a GstRTSPMessage
 
close
close the connection
 
user_data
user data when registering the callback
 

+

+

+
Returns

+
 TRUE on success.

+

+

+

+

+

+
gst_rtsp_client_set_send_func ()

+
void
+gst_rtsp_client_set_send_func (GstRTSPClient *client,
+                               GstRTSPClientSendFunc func,
+                               gpointer user_data,
+                               GDestroyNotify notify);

+
Set func
+ as the callback that will be called when a new message needs to be
+sent to the client. user_data
+ is passed to func
+ and notify
+ is called when
+user_data
+ is no longer in use.

+
By default, the client will send the messages on the GstRTSPConnection that
+was configured with gst_rtsp_client_attach() was called.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





client
a GstRTSPClient
 
func
 a GstRTSPClientSendFunc. 
[scope notified]
user_data
 user data passed to func
+. 
[closure]
notify
 called when user_data
+is no longer in use. 
[allow-none]

+

+

+

+

+
gst_rtsp_client_handle_message ()

+
GstRTSPResult
+gst_rtsp_client_handle_message (GstRTSPClient *client,
+                                GstRTSPMessage *message);

+
Let the client handle message
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





client
a GstRTSPClient
 
message
 an GstRTSPMessage. 
[transfer none]

+

+

+
Returns

+
 a GstRTSPResult.

+

+

+

+

+

+
gst_rtsp_client_send_message ()

+
GstRTSPResult
+gst_rtsp_client_send_message (GstRTSPClient *client,
+                              GstRTSPSession *session,
+                              GstRTSPMessage *message);

+
Send a message message to the remote end. message
+ must be a
+GST_RTSP_MESSAGE_REQUEST or a GST_RTSP_MESSAGE_RESPONSE.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





client
a GstRTSPClient
 
session
 a GstRTSPSession to send
+the message to or NULL. 
[allow-none][transfer none]
message
 The GstRTSPMessage to send. 
[transfer none]

+

+

+

+

+
GstRTSPClientSessionFilterFunc ()

+
GstRTSPFilterResult
+(*GstRTSPClientSessionFilterFunc) (GstRTSPClient *client,
+                                   GstRTSPSession *sess,
+                                   gpointer user_data);

+
This function will be called by the gst_rtsp_client_session_filter(). An
+implementation should return a value of GstRTSPFilterResult.

+
When this function returns GST_RTSP_FILTER_REMOVE, sess
+ will be removed
+from client
+.

+
A return value of GST_RTSP_FILTER_KEEP will leave sess
+ untouched in
+client
+.

+
A value of GST_RTSP_FILTER_REF will add sess
+ to the result GList of
+gst_rtsp_client_session_filter().

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





client
a GstRTSPClient object
 
sess
a GstRTSPSession in client
+
 
user_data
user data that has been given to gst_rtsp_client_session_filter()
 

+

+

+
Returns

+
 a GstRTSPFilterResult.

+

+

+

+

+

+
gst_rtsp_client_session_filter ()

+
GList *
+gst_rtsp_client_session_filter (GstRTSPClient *client,
+                                GstRTSPClientSessionFilterFunc func,
+                                gpointer user_data);

+
Call func
+ for each session managed by client
+. The result value of func
+
+determines what happens to the session. func
+ will be called with client
+
+locked so no further actions on client
+ can be performed from func
+.

+
If func
+ returns GST_RTSP_FILTER_REMOVE, the session will be removed from
+client
+.

+
If func
+ returns GST_RTSP_FILTER_KEEP, the session will remain in client
+.

+
If func
+ returns GST_RTSP_FILTER_REF, the session will remain in client
+ but
+will also be added with an additional ref to the result GList of this
+function..

+
When func
+ is NULL, GST_RTSP_FILTER_REF will be assumed for each session.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





client
a GstRTSPClient
 
func
 a callback. 
[scope call][allow-none]
user_data
user data passed to func
+
 

+

+

+
Returns

+
 a GList with all
+sessions for which func
+returned GST_RTSP_FILTER_REF. After usage, each
+element in the GList should be unreffed before the list is freed. 

+
[element-type GstRTSPSession][transfer full]

+

+

+

+

+
Types and Values

+

+
struct GstRTSPClient

+
struct GstRTSPClient;

+
The client object represents the connection and its state with a client.

+

+

+

+
struct GstRTSPClientClass

+
struct GstRTSPClientClass {
+  GObjectClass  parent_class;
+
+  GstSDPMessage * (*create_sdp) (GstRTSPClient *client, GstRTSPMedia *media);
+  gboolean        (*configure_client_media)     (GstRTSPClient * client,
+                                                 GstRTSPMedia * media, GstRTSPStream * stream,
+                                                 GstRTSPContext * ctx);
+  gboolean        (*configure_client_transport) (GstRTSPClient * client,
+                                                 GstRTSPContext * ctx,
+                                                 GstRTSPTransport * ct);
+  GstRTSPResult   (*params_set) (GstRTSPClient *client, GstRTSPContext *ctx);
+  GstRTSPResult   (*params_get) (GstRTSPClient *client, GstRTSPContext *ctx);
+  gchar *         (*make_path_from_uri) (GstRTSPClient *client, const GstRTSPUrl *uri);
+
+  /* signals */
+  void     (*closed)                  (GstRTSPClient *client);
+  void     (*new_session)             (GstRTSPClient *client, GstRTSPSession *session);
+  void     (*options_request)         (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*describe_request)        (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*setup_request)           (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*play_request)            (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*pause_request)           (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*teardown_request)        (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*set_parameter_request)   (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*get_parameter_request)   (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*handle_response)         (GstRTSPClient *client, GstRTSPContext *ctx);
+
+  void     (*tunnel_http_response)    (GstRTSPClient * client, GstRTSPMessage * request,
+                                       GstRTSPMessage * response);
+};
+

+
The client class structure.

+

+
Members

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
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+
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+
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+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





GObjectClass parent_class;
  
create_sdp ()
called when the SDP needs to be created for media.
 
configure_client_media ()
called when the stream in media needs to be configured.
+The default implementation will configure the blocksize on the payloader when
+spcified in the request headers.
 
configure_client_transport ()
called when the client transport needs to be
+configured.
 
params_set ()
set parameters. This function should also initialize the
+RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
 
params_get ()
get parameters. This function should also initialize the
+RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
 
make_path_from_uri ()
  
closed ()
  
new_session ()
  
options_request ()
  
describe_request ()
  
setup_request ()
  
play_request ()
  
pause_request ()
  
teardown_request ()
  
set_parameter_request ()
  
get_parameter_request ()
  
handle_response ()
  
tunnel_http_response ()
called when a response to the GET request is about to
+be sent for a tunneled connection. The response can be modified. Since 1.4
 

+

+

+

+

+
Property Details

+

+
The “drop-backlog” property

+
  “drop-backlog”             gboolean

+
Drop data when the backlog queue is full.

+
Flags: Read / Write

+
Default value: TRUE

+

+

+

+
The “mount-points” property

+
  “mount-points”             GstRTSPMountPoints *

+
The mount points to use for client session.

+
Flags: Read / Write

+

+

+

+
The “session-pool” property

+
  “session-pool”             GstRTSPSessionPool *

+
The session pool to use for client session.

+
Flags: Read / Write

+

+

+

+
Signal Details

+

+
The “closed” signal

+
void
+user_function (GstRTSPClient *gstrtspclient,
+               gpointer       user_data)

+
Flags: Run Last

+

+

+

+
The “describe-request” signal

+
void
+user_function (GstRTSPClient  *gstrtspclient,
+               GstRTSPContext *arg1,
+               gpointer        user_data)

+
Flags: Run Last

+

+

+

+
The “get-parameter-request” signal

+
void
+user_function (GstRTSPClient  *gstrtspclient,
+               GstRTSPContext *arg1,
+               gpointer        user_data)

+
Flags: Run Last

+

+

+

+
The “handle-response” signal

+
void
+user_function (GstRTSPClient  *gstrtspclient,
+               GstRTSPContext *arg1,
+               gpointer        user_data)

+
Flags: Run Last

+

+

+

+
The “new-session” signal

+
void
+user_function (GstRTSPClient  *gstrtspclient,
+               GstRTSPSession *arg1,
+               gpointer        user_data)

+
Flags: Run Last

+

+

+

+
The “options-request” signal

+
void
+user_function (GstRTSPClient  *gstrtspclient,
+               GstRTSPContext *arg1,
+               gpointer        user_data)

+
Flags: Run Last

+

+

+

+
The “pause-request” signal

+
void
+user_function (GstRTSPClient  *gstrtspclient,
+               GstRTSPContext *arg1,
+               gpointer        user_data)

+
Flags: Run Last

+

+

+

+
The “play-request” signal

+
void
+user_function (GstRTSPClient  *gstrtspclient,
+               GstRTSPContext *arg1,
+               gpointer        user_data)

+
Flags: Run Last

+

+

+

+
The “send-message” signal

+
void
+user_function (GstRTSPClient  *client,
+               GstRTSPContext *session,
+               gpointer        message,
+               gpointer        user_data)

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





client
The RTSP client
 
session
 The session. 
[type GstRtspServer.RTSPSession]
message
 The message. 
[type GstRtsp.RTSPMessage]
user_data
user data set when the signal handler was connected.
 

+

+
Flags: Run Last

+

+

+

+
The “set-parameter-request” signal

+
void
+user_function (GstRTSPClient  *gstrtspclient,
+               GstRTSPContext *arg1,
+               gpointer        user_data)

+
Flags: Run Last

+

+

+

+
The “setup-request” signal

+
void
+user_function (GstRTSPClient  *gstrtspclient,
+               GstRTSPContext *arg1,
+               gpointer        user_data)

+
Flags: Run Last

+

+

+

+
The “teardown-request” signal

+
void
+user_function (GstRTSPClient  *gstrtspclient,
+               GstRTSPContext *arg1,
+               gpointer        user_data)

+
Flags: Run Last

+

+

+

+
See Also

+
GstRTSPServer, GstRTSPThreadPool

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/GstRTSPMedia.html b/docs/libs/html/GstRTSPMedia.html
new file mode 100644
index 0000000..7005873
--- /dev/null
+++ b/docs/libs/html/GstRTSPMedia.html
@@ -0,0 +1,2286 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPMedia
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPMedia

+
GstRTSPMedia — The media pipeline

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
+
+
+
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+
+
+
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+
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+
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+
+
+
+
+
+
+
+
+





+GstRTSPMedia *
+
+gst_rtsp_media_new ()
+

+GstElement *
+
+gst_rtsp_media_get_element ()
+

+void
+
+gst_rtsp_media_take_pipeline ()
+

+void
+
+gst_rtsp_media_set_permissions ()
+

+GstRTSPPermissions *
+
+gst_rtsp_media_get_permissions ()
+

+void
+
+gst_rtsp_media_set_shared ()
+

+gboolean
+
+gst_rtsp_media_is_shared ()
+

+void
+
+gst_rtsp_media_set_reusable ()
+

+gboolean
+
+gst_rtsp_media_is_reusable ()
+

+void
+
+gst_rtsp_media_set_profiles ()
+

+GstRTSPProfile
+
+gst_rtsp_media_get_profiles ()
+

+void
+
+gst_rtsp_media_set_protocols ()
+

+GstRTSPLowerTrans
+
+gst_rtsp_media_get_protocols ()
+

+void
+
+gst_rtsp_media_set_eos_shutdown ()
+

+gboolean
+
+gst_rtsp_media_is_eos_shutdown ()
+

+void
+
+gst_rtsp_media_set_address_pool ()
+

+GstRTSPAddressPool *
+
+gst_rtsp_media_get_address_pool ()
+

+void
+
+gst_rtsp_media_set_buffer_size ()
+

+guint
+
+gst_rtsp_media_get_buffer_size ()
+

+gboolean
+
+gst_rtsp_media_setup_sdp ()
+

+gboolean
+
+gst_rtsp_media_prepare ()
+

+gboolean
+
+gst_rtsp_media_unprepare ()
+

+GstRTSPMediaStatus
+
+gst_rtsp_media_get_status ()
+

+void
+
+gst_rtsp_media_set_suspend_mode ()
+

+GstRTSPSuspendMode
+
+gst_rtsp_media_get_suspend_mode ()
+

+gboolean
+
+gst_rtsp_media_suspend ()
+

+gboolean
+
+gst_rtsp_media_unsuspend ()
+

+void
+
+gst_rtsp_media_collect_streams ()
+

+GstRTSPStream *
+
+gst_rtsp_media_create_stream ()
+

+guint
+
+gst_rtsp_media_n_streams ()
+

+GstRTSPStream *
+
+gst_rtsp_media_get_stream ()
+

+GstRTSPStream *
+
+gst_rtsp_media_find_stream ()
+

+gboolean
+
+gst_rtsp_media_seek ()
+

+gchar *
+
+gst_rtsp_media_get_range_string ()
+

+gboolean
+
+gst_rtsp_media_set_state ()
+

+void
+
+gst_rtsp_media_set_pipeline_state ()
+

+GstClock *
+
+gst_rtsp_media_get_clock ()
+

+GstClockTime
+
+gst_rtsp_media_get_base_time ()
+

+void
+
+gst_rtsp_media_use_time_provider ()
+

+gboolean
+
+gst_rtsp_media_is_time_provider ()
+

+GstNetTimeProvider *
+
+gst_rtsp_media_get_time_provider ()
+

+

+

+
Properties

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
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+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





guintbuffer-sizeRead / Write

+GstElement *elementRead / Write / Construct Only
gbooleaneos-shutdownRead / Write
GstRTSPProfileprofilesRead / Write
GstRTSPLowerTransprotocolsRead / Write
gbooleanreusableRead / Write
gbooleansharedRead / Write
GstRTSPSuspendModesuspend-modeRead / Write
gbooleantime-providerRead / Write

+

+

+
Signals

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





voidnew-stateRun Last
voidnew-streamRun Last
voidpreparedRun Last
voidremoved-streamRun Last
voidtarget-stateRun Last
voidunpreparedRun Last

+

+

+
Types and Values

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+




structGstRTSPMedia
structGstRTSPMediaClass
enumGstRTSPMediaStatus
enumGstRTSPSuspendMode

+

+

+
Object Hierarchy

+
    GObject
+    ╰── GstRTSPMedia
+

+

+

+
Description

+
a GstRTSPMedia contains the complete GStreamer pipeline to manage the
+streaming to the clients. The actual data transfer is done by the
+GstRTSPStream objects that are created and exposed by the GstRTSPMedia.

+
The GstRTSPMedia is usually created from a GstRTSPMediaFactory when the
+client does a DESCRIBE or SETUP of a resource.

+
A media is created with gst_rtsp_media_new() that takes the element that will
+provide the streaming elements. For each of the streams, a new GstRTSPStream
+object needs to be made with the gst_rtsp_media_create_stream() which takes
+the payloader element and the source pad that produces the RTP stream.

+
The pipeline of the media is set to PAUSED with gst_rtsp_media_prepare(). The
+prepare method will add rtpbin and sinks and sources to send and receive RTP
+and RTCP packets from the clients. Each stream srcpad is connected to an
+input into the internal rtpbin.

+
It is also possible to dynamically create GstRTSPStream objects during the
+prepare phase. With gst_rtsp_media_get_status() you can check the status of
+the prepare phase.

+
After the media is prepared, it is ready for streaming. It will usually be
+managed in a session with gst_rtsp_session_manage_media(). See
+GstRTSPSession and GstRTSPSessionMedia.

+
The state of the media can be controlled with gst_rtsp_media_set_state().
+Seeking can be done with gst_rtsp_media_seek().

+
With gst_rtsp_media_unprepare() the pipeline is stopped and shut down. When
+gst_rtsp_media_set_eos_shutdown() an EOS will be sent to the pipeline to
+cleanly shut down.

+
With gst_rtsp_media_set_shared(), the media can be shared between multiple
+clients. With gst_rtsp_media_set_reusable() you can control if the pipeline
+can be prepared again after an unprepare.

+
Last reviewed on 2013-07-11 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_media_new ()

+
GstRTSPMedia *
+gst_rtsp_media_new (GstElement *element);

+
Create a new GstRTSPMedia instance. element
+ is the bin element that
+provides the different streams. The GstRTSPMedia object contains the
+element to produce RTP data for one or more related (audio/video/..)
+streams.

+
Ownership is taken of element
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





element
 a GstElement. 
[transfer full]

+

+

+
Returns

+
 a new GstRTSPMedia object. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_media_get_element ()

+
GstElement *
+gst_rtsp_media_get_element (GstRTSPMedia *media);

+
Get the element that was used when constructing media
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 a GstElement. Unref after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_media_take_pipeline ()

+
void
+gst_rtsp_media_take_pipeline (GstRTSPMedia *media,
+                              GstPipeline *pipeline);

+
Set pipeline
+ as the GstPipeline for media
+. Ownership is
+taken of pipeline
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
pipeline
 a GstPipeline. 
[transfer full]

+

+

+

+

+
gst_rtsp_media_set_permissions ()

+
void
+gst_rtsp_media_set_permissions (GstRTSPMedia *media,
+                                GstRTSPPermissions *permissions);

+
Set permissions
+ on media
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
permissions
 a GstRTSPPermissions. 
[transfer none]

+

+

+

+

+
gst_rtsp_media_get_permissions ()

+
GstRTSPPermissions *
+gst_rtsp_media_get_permissions (GstRTSPMedia *media);

+
Get the permissions object from media
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 a GstRTSPPermissions object, unref after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_media_set_shared ()

+
void
+gst_rtsp_media_set_shared (GstRTSPMedia *media,
+                           gboolean shared);

+
Set or unset if the pipeline for media
+ can be shared will multiple clients.
+When shared
+ is TRUE, client requests for this media will share the media
+pipeline.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
shared
the new value
 

+

+

+

+

+
gst_rtsp_media_is_shared ()

+
gboolean
+gst_rtsp_media_is_shared (GstRTSPMedia *media);

+
Check if the pipeline for media
+ can be shared between multiple clients.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 TRUE if the media can be shared between clients.

+

+

+

+

+

+
gst_rtsp_media_set_reusable ()

+
void
+gst_rtsp_media_set_reusable (GstRTSPMedia *media,
+                             gboolean reusable);

+
Set or unset if the pipeline for media
+ can be reused after the pipeline has
+been unprepared.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
reusable
the new value
 

+

+

+

+

+
gst_rtsp_media_is_reusable ()

+
gboolean
+gst_rtsp_media_is_reusable (GstRTSPMedia *media);

+
Check if the pipeline for media
+ can be reused after an unprepare.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 TRUE if the media can be reused

+

+

+

+

+

+
gst_rtsp_media_set_profiles ()

+
void
+gst_rtsp_media_set_profiles (GstRTSPMedia *media,
+                             GstRTSPProfile profiles);

+
Configure the allowed lower transport for media
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
profiles
the new flags
 

+

+

+

+

+
gst_rtsp_media_get_profiles ()

+
GstRTSPProfile
+gst_rtsp_media_get_profiles (GstRTSPMedia *media);

+
Get the allowed profiles of media
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 a GstRTSPProfile

+

+

+

+

+

+
gst_rtsp_media_set_protocols ()

+
void
+gst_rtsp_media_set_protocols (GstRTSPMedia *media,
+                              GstRTSPLowerTrans protocols);

+
Configure the allowed lower transport for media
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
protocols
the new flags
 

+

+

+

+

+
gst_rtsp_media_get_protocols ()

+
GstRTSPLowerTrans
+gst_rtsp_media_get_protocols (GstRTSPMedia *media);

+
Get the allowed protocols of media
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 a GstRTSPLowerTrans

+

+

+

+

+

+
gst_rtsp_media_set_eos_shutdown ()

+
void
+gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media,
+                                 gboolean eos_shutdown);

+
Set or unset if an EOS event will be sent to the pipeline for media
+ before
+it is unprepared.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
eos_shutdown
the new value
 

+

+

+

+

+
gst_rtsp_media_is_eos_shutdown ()

+
gboolean
+gst_rtsp_media_is_eos_shutdown (GstRTSPMedia *media);

+
Check if the pipeline for media
+ will send an EOS down the pipeline before
+unpreparing.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 TRUE if the media will send EOS before unpreparing.

+

+

+

+

+

+
gst_rtsp_media_set_address_pool ()

+
void
+gst_rtsp_media_set_address_pool (GstRTSPMedia *media,
+                                 GstRTSPAddressPool *pool);

+
configure pool
+ to be used as the address pool of media
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
pool
 a GstRTSPAddressPool. 
[transfer none]

+

+

+

+

+
gst_rtsp_media_get_address_pool ()

+
GstRTSPAddressPool *
+gst_rtsp_media_get_address_pool (GstRTSPMedia *media);

+
Get the GstRTSPAddressPool used as the address pool of media
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 the GstRTSPAddressPool of media
+. g_object_unref() after
+usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_media_set_buffer_size ()

+
void
+gst_rtsp_media_set_buffer_size (GstRTSPMedia *media,
+                                guint size);

+
Set the kernel UDP buffer size.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
size
the new value
 

+

+

+

+

+
gst_rtsp_media_get_buffer_size ()

+
guint
+gst_rtsp_media_get_buffer_size (GstRTSPMedia *media);

+
Get the kernel UDP buffer size.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 the kernel UDP buffer size.

+

+

+

+

+

+
gst_rtsp_media_setup_sdp ()

+
gboolean
+gst_rtsp_media_setup_sdp (GstRTSPMedia *media,
+                          GstSDPMessage *sdp,
+                          GstSDPInfo *info);

+
Add media
+ specific info to sdp
+. info
+ is used to configure the connection
+information in the SDP.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
sdp
 a GstSDPMessage. 
[transfer none]
info
 a GstSDPInfo. 
[transfer none]

+

+

+
Returns

+
 TRUE on success.

+

+

+

+

+

+
gst_rtsp_media_prepare ()

+
gboolean
+gst_rtsp_media_prepare (GstRTSPMedia *media,
+                        GstRTSPThread *thread);

+
Prepare media
+ for streaming. This function will create the objects
+to manage the streaming. A pipeline must have been set on media
+ with
+gst_rtsp_media_take_pipeline().

+
It will preroll the pipeline and collect vital information about the streams
+such as the duration.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
thread
 a GstRTSPThread to run the
+bus handler or NULL. 
[transfer full][allow-none]

+

+

+
Returns

+
 TRUE on success.

+

+

+

+

+

+
gst_rtsp_media_unprepare ()

+
gboolean
+gst_rtsp_media_unprepare (GstRTSPMedia *media);

+
Unprepare media
+. After this call, the media should be prepared again before
+it can be used again. If the media is set to be non-reusable, a new instance
+must be created.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 TRUE on success.

+

+

+

+

+

+
gst_rtsp_media_get_status ()

+
GstRTSPMediaStatus
+gst_rtsp_media_get_status (GstRTSPMedia *media);

+
Get the status of media
+. When media
+ is busy preparing, this function waits
+until media
+ is prepared or in error.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 the status of media
+.

+

+

+

+

+

+
gst_rtsp_media_set_suspend_mode ()

+
void
+gst_rtsp_media_set_suspend_mode (GstRTSPMedia *media,
+                                 GstRTSPSuspendMode mode);

+
Control how @ media will be suspended after the SDP has been generated and
+after a PAUSE request has been performed.

+
Media must be unprepared when setting the suspend mode.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
mode
the new GstRTSPSuspendMode
 

+

+

+

+

+
gst_rtsp_media_get_suspend_mode ()

+
GstRTSPSuspendMode
+gst_rtsp_media_get_suspend_mode (GstRTSPMedia *media);

+
Get how media
+ will be suspended.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 GstRTSPSuspendMode.

+

+

+

+

+

+
gst_rtsp_media_suspend ()

+
gboolean
+gst_rtsp_media_suspend (GstRTSPMedia *media);

+
Suspend media
+. The state of the pipeline managed by media
+ is set to
+GST_STATE_NULL but all streams are kept. media
+ can be prepared again
+with gst_rtsp_media_unsuspend()

+
media
+ must be prepared with gst_rtsp_media_prepare();

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 TRUE on success.

+

+

+

+

+

+
gst_rtsp_media_unsuspend ()

+
gboolean
+gst_rtsp_media_unsuspend (GstRTSPMedia *media);

+
Unsuspend media
+ if it was in a suspended state. This method does nothing
+when the media was not in the suspended state.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 TRUE on success.

+

+

+

+

+

+
gst_rtsp_media_collect_streams ()

+
void
+gst_rtsp_media_collect_streams (GstRTSPMedia *media);

+
Find all payloader elements, they should be named pay%d in the
+element of media
+, and create GstRTSPStreams for them.

+
Collect all dynamic elements, named dynpay%d, and add them to
+the list of dynamic elements.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+

+

+
gst_rtsp_media_create_stream ()

+
GstRTSPStream *
+gst_rtsp_media_create_stream (GstRTSPMedia *media,
+                              GstElement *payloader,
+                              GstPad *srcpad);

+
Create a new stream in media
+ that provides RTP data on srcpad
+.
+srcpad
+ should be a pad of an element inside media->element
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
payloader
a GstElement
 
srcpad
a source GstPad
 

+

+

+
Returns

+
 a new GstRTSPStream that remains valid for as long
+as media
+exists. 

+
[transfer none]

+

+

+

+

+
gst_rtsp_media_n_streams ()

+
guint
+gst_rtsp_media_n_streams (GstRTSPMedia *media);

+
Get the number of streams in this media.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 The number of streams.

+

+

+

+

+

+
gst_rtsp_media_get_stream ()

+
GstRTSPStream *
+gst_rtsp_media_get_stream (GstRTSPMedia *media,
+                           guint idx);

+
Retrieve the stream with index idx
+ from media
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
idx
the stream index
 

+

+

+
Returns

+
 the GstRTSPStream at index
+idx
+or NULL when a stream with that index did not exist. 

+
[nullable][transfer none]

+

+

+

+

+
gst_rtsp_media_find_stream ()

+
GstRTSPStream *
+gst_rtsp_media_find_stream (GstRTSPMedia *media,
+                            const gchar *control);

+
Find a stream in media
+ with control
+ as the control uri.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
control
the control of the stream
 

+

+

+
Returns

+
 the GstRTSPStream with
+control uri control
+or NULL when a stream with that control did
+not exist. 

+
[nullable][transfer none]

+

+

+

+

+
gst_rtsp_media_seek ()

+
gboolean
+gst_rtsp_media_seek (GstRTSPMedia *media,
+                     GstRTSPTimeRange *range);

+
Seek the pipeline of media
+ to range
+. media
+ must be prepared with
+gst_rtsp_media_prepare().

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
range
 a GstRTSPTimeRange. 
[transfer none]

+

+

+
Returns

+
 TRUE on success.

+

+

+

+

+

+
gst_rtsp_media_get_range_string ()

+
gchar *
+gst_rtsp_media_get_range_string (GstRTSPMedia *media,
+                                 gboolean play,
+                                 GstRTSPRangeUnit unit);

+
Get the current range as a string. media
+ must be prepared with
+gst_rtsp_media_prepare().

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
play
for the PLAY request
 
unit
the unit to use for the string
 

+

+

+
Returns

+
 The range as a string, g_free() after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_media_set_state ()

+
gboolean
+gst_rtsp_media_set_state (GstRTSPMedia *media,
+                          GstState state,
+                          GPtrArray *transports);

+
Set the state of media
+ to state
+ and for the transports in transports
+.

+
media
+ must be prepared with gst_rtsp_media_prepare();

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
state
the target state of the media
 
transports
a GPtrArray of GstRTSPStreamTransport pointers. 
[transfer none][element-type GstRtspServer.RTSPStreamTransport]

+

+

+
Returns

+
 TRUE on success.

+

+

+

+

+

+
gst_rtsp_media_set_pipeline_state ()

+
void
+gst_rtsp_media_set_pipeline_state (GstRTSPMedia *media,
+                                   GstState state);

+
Set the state of the pipeline managed by media
+ to state
+

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
state
the target state of the pipeline
 

+

+

+

+

+
gst_rtsp_media_get_clock ()

+
GstClock *
+gst_rtsp_media_get_clock (GstRTSPMedia *media);

+
Get the clock that is used by the pipeline in media
+.

+
media
+ must be prepared before this method returns a valid clock object.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 the GstClock used by media
+. unref after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_media_get_base_time ()

+
GstClockTime
+gst_rtsp_media_get_base_time (GstRTSPMedia *media);

+
Get the base_time that is used by the pipeline in media
+.

+
media
+ must be prepared before this method returns a valid base_time.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 the base_time used by media
+.

+

+

+

+

+

+
gst_rtsp_media_use_time_provider ()

+
void
+gst_rtsp_media_use_time_provider (GstRTSPMedia *media,
+                                  gboolean time_provider);

+
Set media
+ to provide a GstNetTimeProvider.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
time_provider
if a GstNetTimeProvider should be used
 

+

+

+

+

+
gst_rtsp_media_is_time_provider ()

+
gboolean
+gst_rtsp_media_is_time_provider (GstRTSPMedia *media);

+
Check if media
+ can provide a GstNetTimeProvider for its pipeline clock.

+
Use gst_rtsp_media_get_time_provider() to get the network clock.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPMedia
 

+

+

+
Returns

+
 TRUE if media
+can provide a GstNetTimeProvider.

+

+

+

+

+

+
gst_rtsp_media_get_time_provider ()

+
GstNetTimeProvider *
+gst_rtsp_media_get_time_provider (GstRTSPMedia *media,
+                                  const gchar *address,
+                                  guint16 port);

+
Get the GstNetTimeProvider for the clock used by media
+. The time provider
+will listen on address
+ and port
+ for client time requests.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPMedia
 
address
 an address or NULL. 
[allow-none]
port
a port or 0
 

+

+

+
Returns

+
 the GstNetTimeProvider of media
+. 

+
[transfer full]

+

+

+

+

+
Types and Values

+

+
struct GstRTSPMedia

+
struct GstRTSPMedia;

+
A class that contains the GStreamer element along with a list of
+GstRTSPStream objects that can produce data.

+
This object is usually created from a GstRTSPMediaFactory.

+

+

+

+
struct GstRTSPMediaClass

+
struct GstRTSPMediaClass {
+  GObjectClass  parent_class;
+
+  /* vmethods */
+  gboolean        (*handle_message)  (GstRTSPMedia *media, GstMessage *message);
+  gboolean        (*prepare)         (GstRTSPMedia *media, GstRTSPThread *thread);
+  gboolean        (*unprepare)       (GstRTSPMedia *media);
+  gboolean        (*suspend)         (GstRTSPMedia *media);
+  gboolean        (*unsuspend)       (GstRTSPMedia *media);
+  gboolean        (*convert_range)   (GstRTSPMedia *media, GstRTSPTimeRange *range,
+                                      GstRTSPRangeUnit unit);
+  gboolean        (*query_position)  (GstRTSPMedia *media, gint64 *position);
+  gboolean        (*query_stop)      (GstRTSPMedia *media, gint64 *stop);
+  GstElement *    (*create_rtpbin)   (GstRTSPMedia *media);
+  gboolean        (*setup_rtpbin)    (GstRTSPMedia *media, GstElement *rtpbin);
+  gboolean        (*setup_sdp)       (GstRTSPMedia *media, GstSDPMessage *sdp, GstSDPInfo *info);
+
+  /* signals */
+  void            (*new_stream)      (GstRTSPMedia *media, GstRTSPStream * stream);
+  void            (*removed_stream)  (GstRTSPMedia *media, GstRTSPStream * stream);
+
+  void            (*prepared)        (GstRTSPMedia *media);
+  void            (*unprepared)      (GstRTSPMedia *media);
+
+  void            (*target_state)    (GstRTSPMedia *media, GstState state);
+  void            (*new_state)       (GstRTSPMedia *media, GstState state);
+};
+

+
The RTSP media class

+

+
Members

+

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GObjectClass parent_class;
  
handle_message ()
handle a message
 
prepare ()
the default implementation adds all elements and sets the
+pipeline's state to GST_STATE_PAUSED (or GST_STATE_PLAYING
+in case of NO_PREROLL elements).
 
unprepare ()
the default implementation sets the pipeline's state
+to GST_STATE_NULL and removes all elements.
 
suspend ()
the default implementation sets the pipeline's state to
+GST_STATE_NULL GST_STATE_PAUSED depending on the selected
+suspend mode.
 
unsuspend ()
the default implementation reverts the suspend operation.
+The pipeline will be prerolled again if it's state was
+set to GST_STATE_NULL in suspend.
 
convert_range ()
convert a range to the given unit
 
query_position ()
query the current position in the pipeline
 
query_stop ()
query when playback will stop
 
create_rtpbin ()
  
setup_rtpbin ()
  
setup_sdp ()
  
new_stream ()
  
removed_stream ()
  
prepared ()
  
unprepared ()
  
target_state ()
  
new_state ()
  

+

+

+

+

+
enum GstRTSPMediaStatus

+
The state of the media pipeline.

+

+
Members

+

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GST_RTSP_MEDIA_STATUS_UNPREPARED

+
media pipeline not prerolled

+		 
GST_RTSP_MEDIA_STATUS_UNPREPARING

+
media pipeline is busy doing a clean
+                                    shutdown.

+		 
GST_RTSP_MEDIA_STATUS_PREPARING

+
media pipeline is prerolling

+		 
GST_RTSP_MEDIA_STATUS_PREPARED

+
media pipeline is prerolled

+		 
GST_RTSP_MEDIA_STATUS_SUSPENDED

+
media is suspended

+		 
GST_RTSP_MEDIA_STATUS_ERROR

+
media pipeline is in error

+		 

+

+

+

+

+
enum GstRTSPSuspendMode

+
The suspend mode of the media pipeline. A media pipeline is suspended right
+after creating the SDP and when the client performs a PAUSED request.

+

+
Members

+

+++++
+
+
+
+
+
+
+
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+
+
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+
+
+
+





GST_RTSP_SUSPEND_MODE_NONE

+
Media is not suspended

+		 
GST_RTSP_SUSPEND_MODE_PAUSE

+
Media is PAUSED in suspend

+		 
GST_RTSP_SUSPEND_MODE_RESET

+
The media is set to NULL when suspended

+		 

+

+

+

+

+
Property Details

+

+
The “buffer-size” property

+
  “buffer-size”              guint

+
The kernel UDP buffer size to use.

+
Flags: Read / Write

+
Default value: 524288

+

+

+

+
The “element” property

+
  “element”                  GstElement *

+
The GstBin to use for streaming the media.

+
Flags: Read / Write / Construct Only

+

+

+

+
The “eos-shutdown” property

+
  “eos-shutdown”             gboolean

+
Send an EOS event to the pipeline before unpreparing.

+
Flags: Read / Write

+
Default value: FALSE

+

+

+

+
The “profiles” property

+
  “profiles”                 GstRTSPProfile

+
Allowed transfer profiles.

+
Flags: Read / Write

+
Default value: GST_RTSP_PROFILE_AVP

+

+

+

+
The “protocols” property

+
  “protocols”                GstRTSPLowerTrans

+
Allowed lower transport protocols.

+
Flags: Read / Write

+
Default value: GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP

+

+

+

+
The “reusable” property

+
  “reusable”                 gboolean

+
If this media pipeline can be reused after an unprepare.

+
Flags: Read / Write

+
Default value: FALSE

+

+

+

+
The “shared” property

+
  “shared”                   gboolean

+
If this media pipeline can be shared.

+
Flags: Read / Write

+
Default value: FALSE

+

+

+

+
The “suspend-mode” property

+
  “suspend-mode”             GstRTSPSuspendMode

+
How to suspend the media in PAUSED.

+
Flags: Read / Write

+
Default value: GST_RTSP_SUSPEND_MODE_NONE

+

+

+

+
The “time-provider” property

+
  “time-provider”            gboolean

+
Use a NetTimeProvider for clients.

+
Flags: Read / Write

+
Default value: FALSE

+

+

+

+
Signal Details

+

+
The “new-state” signal

+
void
+user_function (GstRTSPMedia *gstrtspmedia,
+               gint          arg1,
+               gpointer      user_data)

+
Flags: Run Last

+

+

+

+
The “new-stream” signal

+
void
+user_function (GstRTSPMedia  *gstrtspmedia,
+               GstRTSPStream *arg1,
+               gpointer       user_data)

+
Flags: Run Last

+

+

+

+
The “prepared” signal

+
void
+user_function (GstRTSPMedia *gstrtspmedia,
+               gpointer      user_data)

+
Flags: Run Last

+

+

+

+
The “removed-stream” signal

+
void
+user_function (GstRTSPMedia  *gstrtspmedia,
+               GstRTSPStream *arg1,
+               gpointer       user_data)

+
Flags: Run Last

+

+

+

+
The “target-state” signal

+
void
+user_function (GstRTSPMedia *gstrtspmedia,
+               gint          arg1,
+               gpointer      user_data)

+
Flags: Run Last

+

+

+

+
The “unprepared” signal

+
void
+user_function (GstRTSPMedia *gstrtspmedia,
+               gpointer      user_data)

+
Flags: Run Last

+

+

+

+
See Also

+
GstRTSPMediaFactory, GstRTSPStream, GstRTSPSession,
+    GstRTSPSessionMedia

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
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diff --git a/docs/libs/html/GstRTSPMediaFactory.html b/docs/libs/html/GstRTSPMediaFactory.html
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+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPMediaFactory
+
+
+
+
+
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+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPMediaFactory

+
GstRTSPMediaFactory — A factory for media pipelines

+

+

+
Functions

+

++++
+
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+





+GstRTSPMediaFactory *
+
+gst_rtsp_media_factory_new ()
+

+gchar *
+
+gst_rtsp_media_factory_get_launch ()
+

+void
+
+gst_rtsp_media_factory_set_launch ()
+

+GstRTSPPermissions *
+
+gst_rtsp_media_factory_get_permissions ()
+

+void
+
+gst_rtsp_media_factory_set_permissions ()
+

+void
+
+gst_rtsp_media_factory_add_role ()
+

+void
+
+gst_rtsp_media_factory_set_shared ()
+

+gboolean
+
+gst_rtsp_media_factory_is_shared ()
+

+gboolean
+
+gst_rtsp_media_factory_is_eos_shutdown ()
+

+void
+
+gst_rtsp_media_factory_set_eos_shutdown ()
+

+GstRTSPLowerTrans
+
+gst_rtsp_media_factory_get_protocols ()
+

+void
+
+gst_rtsp_media_factory_set_protocols ()
+

+void
+
+gst_rtsp_media_factory_set_profiles ()
+

+GstRTSPProfile
+
+gst_rtsp_media_factory_get_profiles ()
+

+GstRTSPAddressPool *
+
+gst_rtsp_media_factory_get_address_pool ()
+

+void
+
+gst_rtsp_media_factory_set_address_pool ()
+

+guint
+
+gst_rtsp_media_factory_get_buffer_size ()
+

+void
+
+gst_rtsp_media_factory_set_buffer_size ()
+

+GstRTSPSuspendMode
+
+gst_rtsp_media_factory_get_suspend_mode ()
+

+void
+
+gst_rtsp_media_factory_set_suspend_mode ()
+

+GstRTSPMedia *
+
+gst_rtsp_media_factory_construct ()
+

+GstElement *
+
+gst_rtsp_media_factory_create_element ()
+

+

+

+
Properties

+

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+
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+





guintbuffer-sizeRead / Write
gbooleaneos-shutdownRead / Write

+gchar *launchRead / Write
GstRTSPProfileprofilesRead / Write
GstRTSPLowerTransprotocolsRead / Write
gbooleansharedRead / Write
GstRTSPSuspendModesuspend-modeRead / Write

+

+

+
Signals

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





voidmedia-configureRun Last
voidmedia-constructedRun Last

+

+

+
Types and Values

+

++++
+
+
+
+
+
+
+
+
+
+
+




structGstRTSPMediaFactory
structGstRTSPMediaFactoryClass

+

+

+
Object Hierarchy

+
    GObject
+    ╰── GstRTSPMediaFactory
+

+

+

+
Description

+
The GstRTSPMediaFactory is responsible for creating or recycling
+GstRTSPMedia objects based on the passed URL.

+
The default implementation of the object can create GstRTSPMedia objects
+containing a pipeline created from a launch description set with
+gst_rtsp_media_factory_set_launch().

+
Media from a factory can be shared by setting the shared flag with
+gst_rtsp_media_factory_set_shared(). When a factory is shared,
+gst_rtsp_media_factory_construct() will return the same GstRTSPMedia when
+the url matches.

+
Last reviewed on 2013-07-11 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_media_factory_new ()

+
GstRTSPMediaFactory *
+gst_rtsp_media_factory_new (void);

+
Create a new GstRTSPMediaFactory instance.

+

+
Returns

+
 a new GstRTSPMediaFactory object. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_media_factory_get_launch ()

+
gchar *
+gst_rtsp_media_factory_get_launch (GstRTSPMediaFactory *factory);

+
Get the gst_parse_launch() pipeline description that will be used in the
+default prepare vmethod.

+

+
Parameters

+

+++++
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 

+

+

+
Returns

+
 the configured launch description. g_free() after
+usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_media_factory_set_launch ()

+
void
+gst_rtsp_media_factory_set_launch (GstRTSPMediaFactory *factory,
+                                   const gchar *launch);

+
The gst_parse_launch() line to use for constructing the pipeline in the
+default prepare vmethod.

+
The pipeline description should return a GstBin as the toplevel element
+which can be accomplished by enclosing the dscription with brackets '('
+')'.

+
The description should return a pipeline with payloaders named pay0, pay1,
+etc.. Each of the payloaders will result in a stream.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 
launch
the launch description
 

+

+

+

+

+
gst_rtsp_media_factory_get_permissions ()

+
GstRTSPPermissions *
+gst_rtsp_media_factory_get_permissions
+                               (GstRTSPMediaFactory *factory);

+
Get the permissions object from factory
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 

+

+

+
Returns

+
 a GstRTSPPermissions object, unref after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_media_factory_set_permissions ()

+
void
+gst_rtsp_media_factory_set_permissions
+                               (GstRTSPMediaFactory *factory,
+                                GstRTSPPermissions *permissions);

+
Set permissions
+ on factory
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 
permissions
 a GstRTSPPermissions. 
[transfer none]

+

+

+

+

+
gst_rtsp_media_factory_add_role ()

+
void
+gst_rtsp_media_factory_add_role (GstRTSPMediaFactory *factory,
+                                 const gchar *role,
+                                 const gchar *fieldname,
+                                 ...);

+
A convenience method to add role
+ with fieldname
+ and additional arguments to
+the permissions of factory
+. If factory
+ had no permissions, new permissions
+will be created and the role will be added to it.

+

+
Parameters

+

+++++
+
+
+
+
+
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+





factory
a GstRTSPMediaFactory
 
role
a role
 
fieldname
the first field name
 
...
additional arguments
 

+

+

+

+

+
gst_rtsp_media_factory_set_shared ()

+
void
+gst_rtsp_media_factory_set_shared (GstRTSPMediaFactory *factory,
+                                   gboolean shared);

+
Configure if media created from this factory can be shared between clients.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 
shared
the new value
 

+

+

+

+

+
gst_rtsp_media_factory_is_shared ()

+
gboolean
+gst_rtsp_media_factory_is_shared (GstRTSPMediaFactory *factory);

+
Get if media created from this factory can be shared between clients.

+

+
Parameters

+

+++++
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 

+

+

+
Returns

+
 TRUE if the media will be shared between clients.

+

+

+

+

+

+
gst_rtsp_media_factory_is_eos_shutdown ()

+
gboolean
+gst_rtsp_media_factory_is_eos_shutdown
+                               (GstRTSPMediaFactory *factory);

+
Get if media created from this factory will have an EOS event sent to the
+pipeline before shutdown.

+

+
Parameters

+

+++++
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 

+

+

+
Returns

+
 TRUE if the media will receive EOS before shutdown.

+

+

+

+

+

+
gst_rtsp_media_factory_set_eos_shutdown ()

+
void
+gst_rtsp_media_factory_set_eos_shutdown
+                               (GstRTSPMediaFactory *factory,
+                                gboolean eos_shutdown);

+
Configure if media created from this factory will have an EOS sent to the
+pipeline before shutdown.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 
eos_shutdown
the new value
 

+

+

+

+

+
gst_rtsp_media_factory_get_protocols ()

+
GstRTSPLowerTrans
+gst_rtsp_media_factory_get_protocols (GstRTSPMediaFactory *factory);

+
Get the allowed protocols of factory
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 

+

+

+
Returns

+
 a GstRTSPLowerTrans

+

+

+

+

+

+
gst_rtsp_media_factory_set_protocols ()

+
void
+gst_rtsp_media_factory_set_protocols (GstRTSPMediaFactory *factory,
+                                      GstRTSPLowerTrans protocols);

+
Configure the allowed lower transport for factory
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 
protocols
the new flags
 

+

+

+

+

+
gst_rtsp_media_factory_set_profiles ()

+
void
+gst_rtsp_media_factory_set_profiles (GstRTSPMediaFactory *factory,
+                                     GstRTSPProfile profiles);

+
Configure the allowed profiles for factory
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 
profiles
the new flags
 

+

+

+

+

+
gst_rtsp_media_factory_get_profiles ()

+
GstRTSPProfile
+gst_rtsp_media_factory_get_profiles (GstRTSPMediaFactory *factory);

+
Get the allowed profiles of factory
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 

+

+

+
Returns

+
 a GstRTSPProfile

+

+

+

+

+

+
gst_rtsp_media_factory_get_address_pool ()

+
GstRTSPAddressPool *
+gst_rtsp_media_factory_get_address_pool
+                               (GstRTSPMediaFactory *factory);

+
Get the GstRTSPAddressPool used as the address pool of factory
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 

+

+

+
Returns

+
 the GstRTSPAddressPool of factory
+. g_object_unref() after
+usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_media_factory_set_address_pool ()

+
void
+gst_rtsp_media_factory_set_address_pool
+                               (GstRTSPMediaFactory *factory,
+                                GstRTSPAddressPool *pool);

+
configure pool
+ to be used as the address pool of factory
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 
pool
 a GstRTSPAddressPool. 
[transfer none]

+

+

+

+

+
gst_rtsp_media_factory_get_buffer_size ()

+
guint
+gst_rtsp_media_factory_get_buffer_size
+                               (GstRTSPMediaFactory *factory);

+
Get the kernel UDP buffer size.

+

+
Parameters

+

+++++
+
+
+
+
+
+





factory
a GstRTSPMedia
 

+

+

+
Returns

+
 the kernel UDP buffer size.

+

+

+

+

+

+
gst_rtsp_media_factory_set_buffer_size ()

+
void
+gst_rtsp_media_factory_set_buffer_size
+                               (GstRTSPMediaFactory *factory,
+                                guint size);

+
Set the kernel UDP buffer size.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





factory
a GstRTSPMedia
 
size
the new value
 

+

+

+

+

+
gst_rtsp_media_factory_get_suspend_mode ()

+
GstRTSPSuspendMode
+gst_rtsp_media_factory_get_suspend_mode
+                               (GstRTSPMediaFactory *factory);

+
Get how media created from this factory will be suspended.

+

+
Parameters

+

+++++
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 

+

+

+
Returns

+
 a GstRTSPSuspendMode.

+

+

+

+

+

+
gst_rtsp_media_factory_set_suspend_mode ()

+
void
+gst_rtsp_media_factory_set_suspend_mode
+                               (GstRTSPMediaFactory *factory,
+                                GstRTSPSuspendMode mode);

+
Configure how media created from this factory will be suspended.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 
mode
the new GstRTSPSuspendMode
 

+

+

+

+

+
gst_rtsp_media_factory_construct ()

+
GstRTSPMedia *
+gst_rtsp_media_factory_construct (GstRTSPMediaFactory *factory,
+                                  const GstRTSPUrl *url);

+
Construct the media object and create its streams. Implementations
+should create the needed gstreamer elements and add them to the result
+object. No state changes should be performed on them yet.

+
One or more GstRTSPStream objects should be created from the result
+with gst_rtsp_media_create_stream().

+
After the media is constructed, it can be configured and then prepared
+with gst_rtsp_media_prepare().

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 
url
the url used
 

+

+

+
Returns

+
 a new GstRTSPMedia if the media could be prepared. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_media_factory_create_element ()

+
GstElement *
+gst_rtsp_media_factory_create_element (GstRTSPMediaFactory *factory,
+                                       const GstRTSPUrl *url);

+
Construct and return a GstElement that is a GstBin containing
+the elements to use for streaming the media.

+
The bin should contain payloaders pay%d for each stream. The default
+implementation of this function returns the bin created from the
+launch parameter.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 
url
the url used
 

+

+

+
Returns

+
 a new GstElement. 

+
[transfer floating]

+

+

+

+

+
Types and Values

+

+
struct GstRTSPMediaFactory

+
struct GstRTSPMediaFactory;

+
The definition and logic for constructing the pipeline for a media. The media
+can contain multiple streams like audio and video.

+

+

+

+
struct GstRTSPMediaFactoryClass

+
struct GstRTSPMediaFactoryClass {
+  GObjectClass  parent_class;
+
+  gchar *         (*gen_key)            (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
+
+  GstElement *    (*create_element)     (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
+  GstRTSPMedia *  (*construct)          (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
+  GstElement *    (*create_pipeline)    (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+  void            (*configure)          (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+
+  /* signals */
+  void            (*media_constructed)  (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+  void            (*media_configure)    (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+};
+

+
The GstRTSPMediaFactory class structure.

+

+
Members

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
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+
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+
+
+
+
+
+
+
+
+
+
+
+
+
+





GObjectClass parent_class;
  
gen_key ()
convert url
+to a key for caching shared GstRTSPMedia objects.
+The default implementation of this function will use the complete URL
+including the query parameters to return a key.
 
create_element ()
Construct and return a GstElement that is a GstBin containing
+the elements to use for streaming the media. The bin should contain
+payloaders pay%d for each stream. The default implementation of this
+function returns the bin created from the launch parameter.
 
construct ()
the vmethod that will be called when the factory has to create the
+GstRTSPMedia for url
+. The default implementation of this
+function calls create_element to retrieve an element and then looks for
+pay%d to create the streams.
 
create_pipeline ()
create a new pipeline or re-use an existing one and
+add the GstRTSPMedia's element created by construct
+to the pipeline.
 
configure ()
configure the media created with construct
+. The default
+implementation will configure the 'shared' property of the media.
 
media_constructed ()
signal emited when a media was constructed
 
media_configure ()
signal emited when a media should be configured
 

+

+

+

+

+
Property Details

+

+
The “buffer-size” property

+
  “buffer-size”              guint

+
The kernel UDP buffer size to use.

+
Flags: Read / Write

+
Default value: 524288

+

+

+

+
The “eos-shutdown” property

+
  “eos-shutdown”             gboolean

+
Send EOS down the pipeline before shutting down.

+
Flags: Read / Write

+
Default value: FALSE

+

+

+

+
The “launch” property

+
  “launch”                   gchar *

+
A launch description of the pipeline.

+
Flags: Read / Write

+
Default value: NULL

+

+

+

+
The “profiles” property

+
  “profiles”                 GstRTSPProfile

+
Allowed transfer profiles.

+
Flags: Read / Write

+
Default value: GST_RTSP_PROFILE_AVP

+

+

+

+
The “protocols” property

+
  “protocols”                GstRTSPLowerTrans

+
Allowed lower transport protocols.

+
Flags: Read / Write

+
Default value: GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP

+

+

+

+
The “shared” property

+
  “shared”                   gboolean

+
If media from this factory is shared.

+
Flags: Read / Write

+
Default value: FALSE

+

+

+

+
The “suspend-mode” property

+
  “suspend-mode”             GstRTSPSuspendMode

+
Control how media will be suspended.

+
Flags: Read / Write

+
Default value: GST_RTSP_SUSPEND_MODE_NONE

+

+

+

+
Signal Details

+

+
The “media-configure” signal

+
void
+user_function (GstRTSPMediaFactory *gstrtspmediafactory,
+               GstRTSPMedia        *arg1,
+               gpointer             user_data)

+
Flags: Run Last

+

+

+

+
The “media-constructed” signal

+
void
+user_function (GstRTSPMediaFactory *gstrtspmediafactory,
+               GstRTSPMedia        *arg1,
+               gpointer             user_data)

+
Flags: Run Last

+

+

+

+
See Also

+
GstRTSPMountPoints, GstRTSPMedia

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/GstRTSPMountPoints.html b/docs/libs/html/GstRTSPMountPoints.html
new file mode 100644
index 0000000..a685b7b
--- /dev/null
+++ b/docs/libs/html/GstRTSPMountPoints.html
@@ -0,0 +1,353 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPMountPoints
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPMountPoints

+
GstRTSPMountPoints — Map a path to media

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





+GstRTSPMountPoints *
+
+gst_rtsp_mount_points_new ()
+

+void
+
+gst_rtsp_mount_points_add_factory ()
+

+void
+
+gst_rtsp_mount_points_remove_factory ()
+

+GstRTSPMediaFactory *
+
+gst_rtsp_mount_points_match ()
+

+gchar *
+
+gst_rtsp_mount_points_make_path ()
+

+

+

+
Types and Values

+

++++
+
+
+
+
+
+
+
+
+
+
+




structGstRTSPMountPoints
structGstRTSPMountPointsClass

+

+

+
Object Hierarchy

+
    GObject
+    ╰── GstRTSPMountPoints
+

+

+

+
Description

+
A GstRTSPMountPoints object maintains a relation between paths
+and GstRTSPMediaFactory objects. This object is usually given to
+GstRTSPClient and used to find the media attached to a path.

+
With gst_rtsp_mount_points_add_factory() and
+gst_rtsp_mount_points_remove_factory(), factories can be added and
+removed.

+
With gst_rtsp_mount_points_match() you can find the GstRTSPMediaFactory
+object that completely matches the given path.

+
Last reviewed on 2013-07-11 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_mount_points_new ()

+
GstRTSPMountPoints *
+gst_rtsp_mount_points_new (void);

+
Make a new mount points object.

+

+
Returns

+
 a new GstRTSPMountPoints. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_mount_points_add_factory ()

+
void
+gst_rtsp_mount_points_add_factory (GstRTSPMountPoints *mounts,
+                                   const gchar *path,
+                                   GstRTSPMediaFactory *factory);

+
Attach factory
+ to the mount point path
+ in mounts
+.

+
path
+ is of the form (/node)+. Any previous mount point will be freed.

+
Ownership is taken of the reference on factory
+ so that factory
+ should not be
+used after calling this function.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





mounts
a GstRTSPMountPoints
 
path
a mount point
 
factory
 a GstRTSPMediaFactory. 
[transfer full]

+

+

+

+

+
gst_rtsp_mount_points_remove_factory ()

+
void
+gst_rtsp_mount_points_remove_factory (GstRTSPMountPoints *mounts,
+                                      const gchar *path);

+
Remove the GstRTSPMediaFactory associated with path
+ in mounts
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





mounts
a GstRTSPMountPoints
 
path
a mount point
 

+

+

+

+

+
gst_rtsp_mount_points_match ()

+
GstRTSPMediaFactory *
+gst_rtsp_mount_points_match (GstRTSPMountPoints *mounts,
+                             const gchar *path,
+                             gint *matched);

+
Find the factory in mounts
+ that has the longest match with path
+.

+
If matched
+ is NULL, path
+ will match the factory exactly otherwise
+the amount of characters that matched is returned in matched
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





mounts
a GstRTSPMountPoints
 
path
a mount point
 
matched
 the amount of path
+matched. 
[out][allow-none]

+

+

+
Returns

+
 the GstRTSPMediaFactory for path
+.
+g_object_unref() after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_mount_points_make_path ()

+
gchar *
+gst_rtsp_mount_points_make_path (GstRTSPMountPoints *mounts,
+                                 const GstRTSPUrl *url);

+
Make a path string from url
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





mounts
a GstRTSPMountPoints
 
url
a GstRTSPUrl
 

+

+

+
Returns

+
 a path string for url
+, g_free() after usage. 

+
[transfer full]

+

+

+

+

+
Types and Values

+

+
struct GstRTSPMountPoints

+
struct GstRTSPMountPoints;

+
Creates a GstRTSPMediaFactory object for a given url.

+

+

+

+
struct GstRTSPMountPointsClass

+
struct GstRTSPMountPointsClass {
+  GObjectClass  parent_class;
+
+  gchar * (*make_path) (GstRTSPMountPoints *mounts,
+                        const GstRTSPUrl *url);
+};
+

+
The class for the media mounts object.

+

+
Members

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





GObjectClass parent_class;
  
make_path ()
make a path from the given url.
 

+

+

+

+

+
See Also

+
GstRTSPMediaFactory, GstRTSPClient

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/GstRTSPServer.html b/docs/libs/html/GstRTSPServer.html
new file mode 100644
index 0000000..e9547e9
--- /dev/null
+++ b/docs/libs/html/GstRTSPServer.html
@@ -0,0 +1,1302 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPServer
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPServer

+
GstRTSPServer — The main server object

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
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+
+
+
+
+
+
+
+
+
+
+
+
+





+GstRTSPServer *
+
+gst_rtsp_server_new ()
+

+gchar *
+
+gst_rtsp_server_get_address ()
+

+void
+
+gst_rtsp_server_set_address ()
+

+gchar *
+
+gst_rtsp_server_get_service ()
+

+void
+
+gst_rtsp_server_set_service ()
+

+gint
+
+gst_rtsp_server_get_backlog ()
+

+void
+
+gst_rtsp_server_set_backlog ()
+

+int
+
+gst_rtsp_server_get_bound_port ()
+

+GstRTSPMountPoints *
+
+gst_rtsp_server_get_mount_points ()
+

+void
+
+gst_rtsp_server_set_mount_points ()
+

+GstRTSPSessionPool *
+
+gst_rtsp_server_get_session_pool ()
+

+void
+
+gst_rtsp_server_set_session_pool ()
+

+GstRTSPThreadPool *
+
+gst_rtsp_server_get_thread_pool ()
+

+void
+
+gst_rtsp_server_set_thread_pool ()
+

+GstRTSPAuth *
+
+gst_rtsp_server_get_auth ()
+

+void
+
+gst_rtsp_server_set_auth ()
+

+gboolean
+
+gst_rtsp_server_transfer_connection ()
+

+gboolean
+
+gst_rtsp_server_io_func ()
+

+GSocket *
+
+gst_rtsp_server_create_socket ()
+

+GSource *
+
+gst_rtsp_server_create_source ()
+

+guint
+
+gst_rtsp_server_attach ()
+

+GstRTSPFilterResult
+
+(*GstRTSPServerClientFilterFunc) ()
+

+GList *
+
+gst_rtsp_server_client_filter ()
+

+

+

+
Properties

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+






+gchar *addressRead / Write
gintbacklogRead / Write
gintbound-portRead

+GstRTSPMountPoints *mount-pointsRead / Write

+gchar *serviceRead / Write

+GstRTSPSessionPool *session-poolRead / Write

+

+

+
Signals

+

+++++
+
+
+
+
+
+





voidclient-connectedRun Last

+

+

+
Types and Values

+

++++
+
+
+
+
+
+
+
+
+
+
+




structGstRTSPServer
structGstRTSPServerClass

+

+

+
Object Hierarchy

+
    GObject
+    ╰── GstRTSPServer
+

+

+

+
Description

+
The server object is the object listening for connections on a port and
+creating GstRTSPClient objects to handle those connections.

+
The server will listen on the address set with gst_rtsp_server_set_address()
+and the port or service configured with gst_rtsp_server_set_service().
+Use gst_rtsp_server_set_backlog() to configure the amount of pending requests
+that the server will keep. By default the server listens on the current
+network (0.0.0.0) and port 8554.

+
The server will require an SSL connection when a TLS certificate has been
+set in the auth object with gst_rtsp_auth_set_tls_certificate().

+
To start the server, use gst_rtsp_server_attach() to attach it to a
+GMainContext. For more control, gst_rtsp_server_create_source() and
+gst_rtsp_server_create_socket() can be used to get a GSource and GSocket
+respectively.

+
gst_rtsp_server_transfer_connection() can be used to transfer an existing
+socket to the RTSP server, for example from an HTTP server.

+
Once the server socket is attached to a mainloop, it will start accepting
+connections. When a new connection is received, a new GstRTSPClient object
+is created to handle the connection. The new client will be configured with
+the server GstRTSPAuth, GstRTSPMountPoints, GstRTSPSessionPool and
+GstRTSPThreadPool.

+
The server uses the configured GstRTSPThreadPool object to handle the
+remainder of the communication with this client.

+
Last reviewed on 2013-07-11 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_server_new ()

+
GstRTSPServer *
+gst_rtsp_server_new (void);

+
Create a new GstRTSPServer instance.

+

+
Returns

+
 a new GstRTSPServer. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_server_get_address ()

+
gchar *
+gst_rtsp_server_get_address (GstRTSPServer *server);

+
Get the address on which the server will accept connections.

+

+
Parameters

+

+++++
+
+
+
+
+
+





server
a GstRTSPServer
 

+

+

+
Returns

+
 the server address. g_free() after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_server_set_address ()

+
void
+gst_rtsp_server_set_address (GstRTSPServer *server,
+                             const gchar *address);

+
Configure server
+ to accept connections on the given address.

+
This function must be called before the server is bound.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





server
a GstRTSPServer
 
address
the address
 

+

+

+

+

+
gst_rtsp_server_get_service ()

+
gchar *
+gst_rtsp_server_get_service (GstRTSPServer *server);

+
Get the service on which the server will accept connections.

+

+
Parameters

+

+++++
+
+
+
+
+
+





server
a GstRTSPServer
 

+

+

+
Returns

+
 the service. use g_free() after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_server_set_service ()

+
void
+gst_rtsp_server_set_service (GstRTSPServer *server,
+                             const gchar *service);

+
Configure server
+ to accept connections on the given service.
+service
+ should be a string containing the service name (see services(5)) or
+a string containing a port number between 1 and 65535.

+
When service
+ is set to "0", the server will listen on a random free
+port. The actual used port can be retrieved with
+gst_rtsp_server_get_bound_port().

+
This function must be called before the server is bound.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





server
a GstRTSPServer
 
service
the service
 

+

+

+

+

+
gst_rtsp_server_get_backlog ()

+
gint
+gst_rtsp_server_get_backlog (GstRTSPServer *server);

+
The maximum amount of queued requests for the server.

+

+
Parameters

+

+++++
+
+
+
+
+
+





server
a GstRTSPServer
 

+

+

+
Returns

+
 the server backlog.

+

+

+

+

+

+
gst_rtsp_server_set_backlog ()

+
void
+gst_rtsp_server_set_backlog (GstRTSPServer *server,
+                             gint backlog);

+
configure the maximum amount of requests that may be queued for the
+server.

+
This function must be called before the server is bound.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





server
a GstRTSPServer
 
backlog
the backlog
 

+

+

+

+

+
gst_rtsp_server_get_bound_port ()

+
int
+gst_rtsp_server_get_bound_port (GstRTSPServer *server);

+
Get the port number where the server was bound to.

+

+
Parameters

+

+++++
+
+
+
+
+
+





server
a GstRTSPServer
 

+

+

+
Returns

+
 the port number

+

+

+

+

+

+
gst_rtsp_server_get_mount_points ()

+
GstRTSPMountPoints *
+gst_rtsp_server_get_mount_points (GstRTSPServer *server);

+
Get the GstRTSPMountPoints used as the mount points of server
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





server
a GstRTSPServer
 

+

+

+
Returns

+
 the GstRTSPMountPoints of server
+. g_object_unref() after
+usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_server_set_mount_points ()

+
void
+gst_rtsp_server_set_mount_points (GstRTSPServer *server,
+                                  GstRTSPMountPoints *mounts);

+
configure mounts
+ to be used as the mount points of server
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





server
a GstRTSPServer
 
mounts
 a GstRTSPMountPoints. 
[transfer none]

+

+

+

+

+
gst_rtsp_server_get_session_pool ()

+
GstRTSPSessionPool *
+gst_rtsp_server_get_session_pool (GstRTSPServer *server);

+
Get the GstRTSPSessionPool used as the session pool of server
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





server
a GstRTSPServer
 

+

+

+
Returns

+
 the GstRTSPSessionPool used for sessions. g_object_unref() after
+usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_server_set_session_pool ()

+
void
+gst_rtsp_server_set_session_pool (GstRTSPServer *server,
+                                  GstRTSPSessionPool *pool);

+
configure pool
+ to be used as the session pool of server
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





server
a GstRTSPServer
 
pool
 a GstRTSPSessionPool. 
[transfer none]

+

+

+

+

+
gst_rtsp_server_get_thread_pool ()

+
GstRTSPThreadPool *
+gst_rtsp_server_get_thread_pool (GstRTSPServer *server);

+
Get the GstRTSPThreadPool used as the thread pool of server
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





server
a GstRTSPServer
 

+

+

+
Returns

+
 the GstRTSPThreadPool of server
+. g_object_unref() after
+usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_server_set_thread_pool ()

+
void
+gst_rtsp_server_set_thread_pool (GstRTSPServer *server,
+                                 GstRTSPThreadPool *pool);

+
configure pool
+ to be used as the thread pool of server
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





server
a GstRTSPServer
 
pool
 a GstRTSPThreadPool. 
[transfer none]

+

+

+

+

+
gst_rtsp_server_get_auth ()

+
GstRTSPAuth *
+gst_rtsp_server_get_auth (GstRTSPServer *server);

+
Get the GstRTSPAuth used as the authentication manager of server
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





server
a GstRTSPServer
 

+

+

+
Returns

+
 the GstRTSPAuth of server
+. g_object_unref() after
+usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_server_set_auth ()

+
void
+gst_rtsp_server_set_auth (GstRTSPServer *server,
+                          GstRTSPAuth *auth);

+
configure auth
+ to be used as the authentication manager of server
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





server
a GstRTSPServer
 
auth
 a GstRTSPAuth. 
[transfer none]

+

+

+

+

+
gst_rtsp_server_transfer_connection ()

+
gboolean
+gst_rtsp_server_transfer_connection (GstRTSPServer *server,
+                                     GSocket *socket,
+                                     const gchar *ip,
+                                     gint port,
+                                     const gchar *initial_buffer);

+
Take an existing network socket and use it for an RTSP connection. This
+is used when transferring a socket from an HTTP server which should be used
+as an RTSP over HTTP tunnel. The initial_buffer
+ contains any remaining data
+that the HTTP server read from the socket while parsing the HTTP header.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





server
a GstRTSPServer
 
socket
 a network socket. 
[transfer full]
ip
the IP address of the remote client
 
port
the port used by the other end
 
initial_buffer
any initial data that was already read from the socket
 

+

+

+
Returns

+
 TRUE if all was ok, FALSE if an error occurred.

+

+

+

+

+

+
gst_rtsp_server_io_func ()

+
gboolean
+gst_rtsp_server_io_func (GSocket *socket,
+                         GIOCondition condition,
+                         GstRTSPServer *server);

+
A default GSocketSourceFunc that creates a new GstRTSPClient to accept and handle a
+new connection on socket
+ or server
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





socket
a GSocket
 
condition
the condition on source
+
 
server
 a GstRTSPServer. 
[transfer none]

+

+

+
Returns

+
 TRUE if the source could be connected, FALSE if an error occurred.

+

+

+

+

+

+
gst_rtsp_server_create_socket ()

+
GSocket *
+gst_rtsp_server_create_socket (GstRTSPServer *server,
+                               GCancellable *cancellable,
+                               GError **error);

+
Create a GSocket for server
+. The socket will listen on the
+configured service.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





server
a GstRTSPServer
 
cancellable
 a GCancellable. 
[allow-none]
error
 a GError. 
[out]

+

+

+
Returns

+
 the GSocket for server
+or NULL when an error
+occurred. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_server_create_source ()

+
GSource *
+gst_rtsp_server_create_source (GstRTSPServer *server,
+                               GCancellable *cancellable,
+                               GError **error);

+
Create a GSource for server
+. The new source will have a default
+GSocketSourceFunc of gst_rtsp_server_io_func().

+
cancellable
+ if not NULL can be used to cancel the source, which will cause
+the source to trigger, reporting the current condition (which is likely 0
+unless cancellation happened at the same time as a condition change). You can
+check for this in the callback using g_cancellable_is_cancelled().

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





server
a GstRTSPServer
 
cancellable
 a GCancellable or NULL. 
[allow-none]
error
 a GError. 
[out]

+

+

+
Returns

+
 the GSource for server
+or NULL when an error
+occurred. Free with g_source_unref(). 

+
[transfer full]

+

+

+

+

+
gst_rtsp_server_attach ()

+
guint
+gst_rtsp_server_attach (GstRTSPServer *server,
+                        GMainContext *context);

+
Attaches server
+ to context
+. When the mainloop for context
+ is run, the
+server will be dispatched. When context
+ is NULL, the default context will be
+used).

+
This function should be called when the server properties and urls are fully
+configured and the server is ready to start.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





server
a GstRTSPServer
 
context
 a GMainContext. 
[allow-none]

+

+

+
Returns

+
 the ID (greater than 0) for the source within the GMainContext.

+

+

+

+

+

+
GstRTSPServerClientFilterFunc ()

+
GstRTSPFilterResult
+(*GstRTSPServerClientFilterFunc) (GstRTSPServer *server,
+                                  GstRTSPClient *client,
+                                  gpointer user_data);

+
This function will be called by the gst_rtsp_server_client_filter(). An
+implementation should return a value of GstRTSPFilterResult.

+
When this function returns GST_RTSP_FILTER_REMOVE, client
+ will be removed
+from server
+.

+
A return value of GST_RTSP_FILTER_KEEP will leave client
+ untouched in
+server
+.

+
A value of GST_RTSP_FILTER_REF will add client
+ to the result GList of
+gst_rtsp_server_client_filter().

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





server
a GstRTSPServer object
 
client
a GstRTSPClient in server
+
 
user_data
user data that has been given to gst_rtsp_server_client_filter()
 

+

+

+
Returns

+
 a GstRTSPFilterResult.

+

+

+

+

+

+
gst_rtsp_server_client_filter ()

+
GList *
+gst_rtsp_server_client_filter (GstRTSPServer *server,
+                               GstRTSPServerClientFilterFunc func,
+                               gpointer user_data);

+
Call func
+ for each client managed by server
+. The result value of func
+
+determines what happens to the client. func
+ will be called with server
+
+locked so no further actions on server
+ can be performed from func
+.

+
If func
+ returns GST_RTSP_FILTER_REMOVE, the client will be removed from
+server
+.

+
If func
+ returns GST_RTSP_FILTER_KEEP, the client will remain in server
+.

+
If func
+ returns GST_RTSP_FILTER_REF, the client will remain in server
+ but
+will also be added with an additional ref to the result GList of this
+function..

+
When func
+ is NULL, GST_RTSP_FILTER_REF will be assumed for each client.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





server
a GstRTSPServer
 
func
 a callback. 
[scope call][allow-none]
user_data
user data passed to func
+
 

+

+

+
Returns

+
 a GList with all
+clients for which func
+returned GST_RTSP_FILTER_REF. After usage, each
+element in the GList should be unreffed before the list is freed. 

+
[element-type GstRTSPClient][transfer full]

+

+

+

+

+
Types and Values

+

+
struct GstRTSPServer

+
struct GstRTSPServer;

+
This object listens on a port, creates and manages the clients connected to
+it.

+

+

+

+
struct GstRTSPServerClass

+
struct GstRTSPServerClass {
+  GObjectClass  parent_class;
+
+  GstRTSPClient * (*create_client)      (GstRTSPServer *server);
+
+  /* signals */
+  void            (*client_connected)   (GstRTSPServer *server, GstRTSPClient *client);
+};
+

+
The RTSP server class structure

+

+
Members

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





GObjectClass parent_class;
  
create_client ()
Create, configure a new GstRTSPClient
+object that handles the new connection on socket
+. The default
+implementation will create a GstRTSPClient and will configure the
+mount-points, auth, session-pool and thread-pool on the client.
 
client_connected ()
emited when a new client connected.
 

+

+

+

+

+
Property Details

+

+
The “address” property

+
  “address”                  gchar *

+
The address the server uses to listen on.

+
Flags: Read / Write

+
Default value: "0.0.0.0"

+

+

+

+
The “backlog” property

+
  “backlog”                  gint

+
The maximum length to which the queue of pending connections may grow.

+
Flags: Read / Write

+
Allowed values: >= 0

+
Default value: 5

+

+

+

+
The “bound-port” property

+
  “bound-port”               gint

+
The port number the server is listening on.

+
Flags: Read

+
Allowed values: [-1,65535]

+
Default value: -1

+

+

+

+
The “mount-points” property

+
  “mount-points”             GstRTSPMountPoints *

+
The mount points to use for client session.

+
Flags: Read / Write

+

+

+

+
The “service” property

+
  “service”                  gchar *

+
The service or port number the server uses to listen on.

+
Flags: Read / Write

+
Default value: "8554"

+

+

+

+
The “session-pool” property

+
  “session-pool”             GstRTSPSessionPool *

+
The session pool to use for client session.

+
Flags: Read / Write

+

+

+

+
Signal Details

+

+
The “client-connected” signal

+
void
+user_function (GstRTSPServer *gstrtspserver,
+               GstRTSPClient *arg1,
+               gpointer       user_data)

+
Flags: Run Last

+

+

+

+
See Also

+
GstRTSPClient, GstRTSPThreadPool

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/GstRTSPSession.html b/docs/libs/html/GstRTSPSession.html
new file mode 100644
index 0000000..38cc654
--- /dev/null
+++ b/docs/libs/html/GstRTSPSession.html
@@ -0,0 +1,881 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPSession
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPSession

+
GstRTSPSession — An object to manage media

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





+GstRTSPSession *
+
+gst_rtsp_session_new ()
+
const gchar *
+
+gst_rtsp_session_get_sessionid ()
+

+gchar *
+
+gst_rtsp_session_get_header ()
+

+void
+
+gst_rtsp_session_set_timeout ()
+

+guint
+
+gst_rtsp_session_get_timeout ()
+

+void
+
+gst_rtsp_session_touch ()
+

+void
+
+gst_rtsp_session_prevent_expire ()
+

+void
+
+gst_rtsp_session_allow_expire ()
+

+gint
+
+gst_rtsp_session_next_timeout ()
+

+gboolean
+
+gst_rtsp_session_is_expired ()
+

+GstRTSPSessionMedia *
+
+gst_rtsp_session_manage_media ()
+

+gboolean
+
+gst_rtsp_session_release_media ()
+

+GstRTSPSessionMedia *
+
+gst_rtsp_session_get_media ()
+

+GstRTSPFilterResult
+
+(*GstRTSPSessionFilterFunc) ()
+

+GList *
+
+gst_rtsp_session_filter ()
+

+

+

+
Properties

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+






+gchar *sessionidRead / Write / Construct Only
guinttimeoutRead / Write
gbooleantimeout-always-visibleRead / Write

+

+

+
Types and Values

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+




structGstRTSPSession
structGstRTSPSessionClass
enumGstRTSPFilterResult

+

+

+
Object Hierarchy

+
    GObject
+    ╰── GstRTSPSession
+

+

+

+
Description

+
The GstRTSPSession is identified by an id, unique in the
+GstRTSPSessionPool that created the session and manages media and its
+configuration.

+
A GstRTSPSession has a timeout that can be retrieved with
+gst_rtsp_session_get_timeout(). You can check if the sessions is expired with
+gst_rtsp_session_is_expired(). gst_rtsp_session_touch() will reset the
+expiration counter of the session.

+
When a client configures a media with SETUP, a session will be created to
+keep track of the configuration of that media. With
+gst_rtsp_session_manage_media(), the media is added to the managed media
+in the session. With gst_rtsp_session_release_media() the media can be
+released again from the session. Managed media is identified in the sessions
+with a url. Use gst_rtsp_session_get_media() to get the media that matches
+(part of) the given url.

+
The media in a session can be iterated with gst_rtsp_session_filter().

+
Last reviewed on 2013-07-11 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_session_new ()

+
GstRTSPSession *
+gst_rtsp_session_new (const gchar *sessionid);

+
Create a new GstRTSPSession instance with sessionid
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





sessionid
a session id
 

+

+

+
Returns

+
 a new GstRTSPSession. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_session_get_sessionid ()

+
const gchar *
+gst_rtsp_session_get_sessionid (GstRTSPSession *session);

+
Get the sessionid of session
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





session
a GstRTSPSession
 

+

+

+
Returns

+
 the sessionid of session
+. The value remains valid
+as long as session
+is alive. 

+
[transfer none]

+

+

+

+

+
gst_rtsp_session_get_header ()

+
gchar *
+gst_rtsp_session_get_header (GstRTSPSession *session);

+
Get the string that can be placed in the Session header field.

+

+
Parameters

+

+++++
+
+
+
+
+
+





session
a GstRTSPSession
 

+

+

+
Returns

+
 the Session header of session
+. g_free() after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_session_set_timeout ()

+
void
+gst_rtsp_session_set_timeout (GstRTSPSession *session,
+                              guint timeout);

+
Configure session
+ for a timeout of timeout
+ seconds. The session will be
+cleaned up when there is no activity for timeout
+ seconds.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





session
a GstRTSPSession
 
timeout
the new timeout
 

+

+

+

+

+
gst_rtsp_session_get_timeout ()

+
guint
+gst_rtsp_session_get_timeout (GstRTSPSession *session);

+
Get the timeout value of session
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





session
a GstRTSPSession
 

+

+

+
Returns

+
 the timeout of session
+in seconds.

+

+

+

+

+

+
gst_rtsp_session_touch ()

+
void
+gst_rtsp_session_touch (GstRTSPSession *session);

+
Update the last_access time of the session to the current time.

+

+
Parameters

+

+++++
+
+
+
+
+
+





session
a GstRTSPSession
 

+

+

+

+

+
gst_rtsp_session_prevent_expire ()

+
void
+gst_rtsp_session_prevent_expire (GstRTSPSession *session);

+
Prevent session
+ from expiring.

+

+
Parameters

+

+++++
+
+
+
+
+
+





session
a GstRTSPSession
 

+

+

+

+

+
gst_rtsp_session_allow_expire ()

+
void
+gst_rtsp_session_allow_expire (GstRTSPSession *session);

+
Allow session
+ to expire. This method must be called an equal
+amount of time as gst_rtsp_session_prevent_expire().

+

+
Parameters

+

+++++
+
+
+
+
+
+





session
a GstRTSPSession
 

+

+

+

+

+
gst_rtsp_session_next_timeout ()

+
gint
+gst_rtsp_session_next_timeout (GstRTSPSession *session,
+                               GTimeVal *now);

+
Get the amount of milliseconds till the session will expire.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





session
a GstRTSPSession
 
now
 the current system time. 
[transfer none]

+

+

+
Returns

+
 the amount of milliseconds since the session will time out.

+

+

+

+

+

+
gst_rtsp_session_is_expired ()

+
gboolean
+gst_rtsp_session_is_expired (GstRTSPSession *session,
+                             GTimeVal *now);

+
Check if session
+ timeout out.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





session
a GstRTSPSession
 
now
 the current system time. 
[transfer none]

+

+

+
Returns

+
 TRUE if session
+timed out

+

+

+

+

+

+
gst_rtsp_session_manage_media ()

+
GstRTSPSessionMedia *
+gst_rtsp_session_manage_media (GstRTSPSession *sess,
+                               const gchar *path,
+                               GstRTSPMedia *media);

+
Manage the media object obj
+ in sess
+. path
+ will be used to retrieve this
+media from the session with gst_rtsp_session_get_media().

+
Ownership is taken from media
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





sess
a GstRTSPSession
 
path
the path for the media
 
media
 a GstRTSPMedia. 
[transfer full]

+

+

+
Returns

+
 a new GstRTSPSessionMedia
+object. 

+
[transfer none]

+

+

+

+

+
gst_rtsp_session_release_media ()

+
gboolean
+gst_rtsp_session_release_media (GstRTSPSession *sess,
+                                GstRTSPSessionMedia *media);

+
Release the managed media
+ in sess
+, freeing the memory allocated by it.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





sess
a GstRTSPSession
 
media
 a GstRTSPMedia. 
[transfer none]

+

+

+
Returns

+
 TRUE if there are more media session left in sess
+.

+

+

+

+

+

+
gst_rtsp_session_get_media ()

+
GstRTSPSessionMedia *
+gst_rtsp_session_get_media (GstRTSPSession *sess,
+                            const gchar *path,
+                            gint *matched);

+
Get the session media for path
+. matched
+ will contain the number of matched
+characters of path
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





sess
a GstRTSPSession
 
path
the path for the media
 
matched
 the amount of matched characters. 
[out]

+

+

+
Returns

+
 the configuration for path
+in sess
+. 

+
[transfer none]

+

+

+

+

+
GstRTSPSessionFilterFunc ()

+
GstRTSPFilterResult
+(*GstRTSPSessionFilterFunc) (GstRTSPSession *sess,
+                             GstRTSPSessionMedia *media,
+                             gpointer user_data);

+
This function will be called by the gst_rtsp_session_filter(). An
+implementation should return a value of GstRTSPFilterResult.

+
When this function returns GST_RTSP_FILTER_REMOVE, media
+ will be removed
+from sess
+.

+
A return value of GST_RTSP_FILTER_KEEP will leave media
+ untouched in
+sess
+.

+
A value of GST_RTSP_FILTER_REF will add media
+ to the result GList of
+gst_rtsp_session_filter().

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





sess
a GstRTSPSession object
 
media
a GstRTSPSessionMedia in sess
+
 
user_data
user data that has been given to gst_rtsp_session_filter()
 

+

+

+
Returns

+
 a GstRTSPFilterResult.

+

+

+

+

+

+
gst_rtsp_session_filter ()

+
GList *
+gst_rtsp_session_filter (GstRTSPSession *sess,
+                         GstRTSPSessionFilterFunc func,
+                         gpointer user_data);

+
Call func
+ for each media in sess
+. The result value of func
+ determines
+what happens to the media. func
+ will be called with sess
+
+locked so no further actions on sess
+ can be performed from func
+.

+
If func
+ returns GST_RTSP_FILTER_REMOVE, the media will be removed from
+sess
+.

+
If func
+ returns GST_RTSP_FILTER_KEEP, the media will remain in sess
+.

+
If func
+ returns GST_RTSP_FILTER_REF, the media will remain in sess
+ but
+will also be added with an additional ref to the result GList of this
+function..

+
When func
+ is NULL, GST_RTSP_FILTER_REF will be assumed for all media.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





sess
a GstRTSPSession
 
func
 a callback. 
[scope call][allow-none]
user_data
 user data passed to func
+. 
[closure]

+

+

+
Returns

+
 a GList with all
+media for which func
+returned GST_RTSP_FILTER_REF. After usage, each
+element in the GList should be unreffed before the list is freed. 

+
[element-type GstRTSPSessionMedia][transfer full]

+

+

+

+

+
Types and Values

+

+
struct GstRTSPSession

+
struct GstRTSPSession;

+
Session information kept by the server for a specific client.
+One client session, identified with a session id, can handle multiple medias
+identified with the url of a media.

+

+

+

+
struct GstRTSPSessionClass

+
struct GstRTSPSessionClass {
+  GObjectClass  parent_class;
+};
+

+

+

+

+
enum GstRTSPFilterResult

+
Possible return values for gst_rtsp_session_pool_filter().

+

+
Members

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





GST_RTSP_FILTER_REMOVE

+
Remove session

+		 
GST_RTSP_FILTER_KEEP

+
Keep session in the pool

+		 
GST_RTSP_FILTER_REF

+
Ref session in the result list

+		 

+

+

+

+

+
Property Details

+

+
The “sessionid” property

+
  “sessionid”                gchar *

+
the session id.

+
Flags: Read / Write / Construct Only

+
Default value: NULL

+

+

+

+
The “timeout” property

+
  “timeout”                  guint

+
the timeout of the session (0 = never).

+
Flags: Read / Write

+
Default value: 60

+

+

+

+
The “timeout-always-visible” property

+
  “timeout-always-visible”   gboolean

+
timeout always visible in header.

+
Flags: Read / Write

+
Default value: FALSE

+

+

+

+
See Also

+
GstRTSPSessionPool, GstRTSPSessionMedia, GstRTSPMedia

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/GstRTSPSessionPool.html b/docs/libs/html/GstRTSPSessionPool.html
new file mode 100644
index 0000000..718aef7
--- /dev/null
+++ b/docs/libs/html/GstRTSPSessionPool.html
@@ -0,0 +1,730 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPSessionPool
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPSessionPool

+
GstRTSPSessionPool — An object for managing sessions

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
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+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





+GstRTSPSessionPool *
+
+gst_rtsp_session_pool_new ()
+

+guint
+
+gst_rtsp_session_pool_get_max_sessions ()
+

+void
+
+gst_rtsp_session_pool_set_max_sessions ()
+

+guint
+
+gst_rtsp_session_pool_get_n_sessions ()
+

+GstRTSPSession *
+
+gst_rtsp_session_pool_create ()
+

+GstRTSPSession *
+
+gst_rtsp_session_pool_find ()
+

+gboolean
+
+gst_rtsp_session_pool_remove ()
+

+guint
+
+gst_rtsp_session_pool_cleanup ()
+

+gboolean
+
+(*GstRTSPSessionPoolFunc) ()
+

+GSource *
+
+gst_rtsp_session_pool_create_watch ()
+

+GstRTSPFilterResult
+
+(*GstRTSPSessionPoolFilterFunc) ()
+

+GList *
+
+gst_rtsp_session_pool_filter ()
+

+

+

+
Properties

+

+++++
+
+
+
+
+
+





guintmax-sessionsRead / Write

+

+

+
Signals

+

+++++
+
+
+
+
+
+





voidsession-removedRun Last

+

+

+
Types and Values

+

++++
+
+
+
+
+
+
+
+
+
+
+




structGstRTSPSessionPool
structGstRTSPSessionPoolClass

+

+

+
Object Hierarchy

+
    GObject
+    ╰── GstRTSPSessionPool
+

+

+

+
Description

+
The GstRTSPSessionPool object manages a list of GstRTSPSession objects.

+
The maximum number of sessions can be configured with
+gst_rtsp_session_pool_set_max_sessions(). The current number of sessions can
+be retrieved with gst_rtsp_session_pool_get_n_sessions().

+
Use gst_rtsp_session_pool_create() to create a new GstRTSPSession object.
+The session object can be found again with its id and
+gst_rtsp_session_pool_find().

+
All sessions can be iterated with gst_rtsp_session_pool_filter().

+
Run gst_rtsp_session_pool_cleanup() periodically to remove timed out sessions
+or use gst_rtsp_session_pool_create_watch() to be notified when session
+cleanup should be performed.

+
Last reviewed on 2013-07-11 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_session_pool_new ()

+
GstRTSPSessionPool *
+gst_rtsp_session_pool_new (void);

+
Create a new GstRTSPSessionPool instance.

+

+
Returns

+
 A new GstRTSPSessionPool. g_object_unref() after
+usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_session_pool_get_max_sessions ()

+
guint
+gst_rtsp_session_pool_get_max_sessions
+                               (GstRTSPSessionPool *pool);

+
Get the maximum allowed number of sessions in pool
+. 0 means an unlimited
+amount of sessions.

+

+
Parameters

+

+++++
+
+
+
+
+
+





pool
a GstRTSPSessionPool
 

+

+

+
Returns

+
 the maximum allowed number of sessions.

+

+

+

+

+

+
gst_rtsp_session_pool_set_max_sessions ()

+
void
+gst_rtsp_session_pool_set_max_sessions
+                               (GstRTSPSessionPool *pool,
+                                guint max);

+
Configure the maximum allowed number of sessions in pool
+ to max
+.
+A value of 0 means an unlimited amount of sessions.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





pool
a GstRTSPSessionPool
 
max
the maximum number of sessions
 

+

+

+

+

+
gst_rtsp_session_pool_get_n_sessions ()

+
guint
+gst_rtsp_session_pool_get_n_sessions (GstRTSPSessionPool *pool);

+
Get the amount of active sessions in pool
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





pool
a GstRTSPSessionPool
 

+

+

+
Returns

+
 the amount of active sessions in pool
+.

+

+

+

+

+

+
gst_rtsp_session_pool_create ()

+
GstRTSPSession *
+gst_rtsp_session_pool_create (GstRTSPSessionPool *pool);

+
Create a new GstRTSPSession object in pool
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





pool
a GstRTSPSessionPool
 

+

+

+
Returns

+
 a new GstRTSPSession. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_session_pool_find ()

+
GstRTSPSession *
+gst_rtsp_session_pool_find (GstRTSPSessionPool *pool,
+                            const gchar *sessionid);

+
Find the session with sessionid
+ in pool
+. The access time of the session
+will be updated with gst_rtsp_session_touch().

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





pool
the pool to search
 
sessionid
the session id
 

+

+

+
Returns

+
 the GstRTSPSession with sessionid
+or NULL when the session did not exist. g_object_unref() after usage. 

+
[transfer full][nullable]

+

+

+

+

+
gst_rtsp_session_pool_remove ()

+
gboolean
+gst_rtsp_session_pool_remove (GstRTSPSessionPool *pool,
+                              GstRTSPSession *sess);

+
Remove sess
+ from pool
+, releasing the ref that the pool has on sess
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





pool
a GstRTSPSessionPool
 
sess
 a GstRTSPSession. 
[transfer none]

+

+

+
Returns

+
 TRUE if the session was found and removed.

+

+

+

+

+

+
gst_rtsp_session_pool_cleanup ()

+
guint
+gst_rtsp_session_pool_cleanup (GstRTSPSessionPool *pool);

+
Inspect all the sessions in pool
+ and remove the sessions that are inactive
+for more than their timeout.

+

+
Parameters

+

+++++
+
+
+
+
+
+





pool
a GstRTSPSessionPool
 

+

+

+
Returns

+
 the amount of sessions that got removed.

+

+

+

+

+

+
GstRTSPSessionPoolFunc ()

+
gboolean
+(*GstRTSPSessionPoolFunc) (GstRTSPSessionPool *pool,
+                           gpointer user_data);

+
The function that will be called from the GSource watch on the session pool.

+
The function will be called when the pool must be cleaned up because one or
+more sessions timed out.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





pool
a GstRTSPSessionPool object
 
user_data
user data that has been given when registering the handler
 

+

+

+
Returns

+
 FALSE if the source should be removed.

+

+

+

+

+

+
gst_rtsp_session_pool_create_watch ()

+
GSource *
+gst_rtsp_session_pool_create_watch (GstRTSPSessionPool *pool);

+
Create a GSource that will be dispatched when the session should be cleaned
+up.

+

+
Parameters

+

+++++
+
+
+
+
+
+





pool
a GstRTSPSessionPool
 

+

+

+
Returns

+
 a GSource. 

+
[transfer full]

+

+

+

+

+
GstRTSPSessionPoolFilterFunc ()

+
GstRTSPFilterResult
+(*GstRTSPSessionPoolFilterFunc) (GstRTSPSessionPool *pool,
+                                 GstRTSPSession *session,
+                                 gpointer user_data);

+
This function will be called by the gst_rtsp_session_pool_filter(). An
+implementation should return a value of GstRTSPFilterResult.

+
When this function returns GST_RTSP_FILTER_REMOVE, session
+ will be removed
+from pool
+.

+
A return value of GST_RTSP_FILTER_KEEP will leave session
+ untouched in
+pool
+.

+
A value of GST_RTSP_FILTER_REF will add session
+ to the result GList of
+gst_rtsp_session_pool_filter().

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





pool
a GstRTSPSessionPool object
 
session
a GstRTSPSession in pool
+
 
user_data
user data that has been given to gst_rtsp_session_pool_filter()
 

+

+

+
Returns

+
 a GstRTSPFilterResult.

+

+

+

+

+

+
gst_rtsp_session_pool_filter ()

+
GList *
+gst_rtsp_session_pool_filter (GstRTSPSessionPool *pool,
+                              GstRTSPSessionPoolFilterFunc func,
+                              gpointer user_data);

+
Call func
+ for each session in pool
+. The result value of func
+ determines
+what happens to the session. func
+ will be called with the session pool
+locked so no further actions on pool
+ can be performed from func
+.

+
If func
+ returns GST_RTSP_FILTER_REMOVE, the session will be set to the
+expired state with gst_rtsp_session_set_expired() and removed from
+pool
+.

+
If func
+ returns GST_RTSP_FILTER_KEEP, the session will remain in pool
+.

+
If func
+ returns GST_RTSP_FILTER_REF, the session will remain in pool
+ but
+will also be added with an additional ref to the result GList of this
+function..

+
When func
+ is NULL, GST_RTSP_FILTER_REF will be assumed for all sessions.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





pool
a GstRTSPSessionPool
 
func
 a callback. 
[scope call][allow-none]
user_data
 user data passed to func
+. 
[closure]

+

+

+
Returns

+
 a GList with all
+sessions for which func
+returned GST_RTSP_FILTER_REF. After usage, each
+element in the GList should be unreffed before the list is freed. 

+
[element-type GstRTSPSession][transfer full]

+

+

+

+

+
Types and Values

+

+
struct GstRTSPSessionPool

+
struct GstRTSPSessionPool;

+
An object that keeps track of the active sessions. This object is usually
+attached to a GstRTSPServer object to manage the sessions in that server.

+

+

+

+
struct GstRTSPSessionPoolClass

+
struct GstRTSPSessionPoolClass {
+  GObjectClass  parent_class;
+
+  gchar *          (*create_session_id)   (GstRTSPSessionPool *pool);
+  GstRTSPSession * (*create_session)      (GstRTSPSessionPool *pool, const gchar *id);
+
+  /* signals */
+  void             (*session_removed)     (GstRTSPSessionPool *pool,
+                                           GstRTSPSession *session);
+};
+

+

+
Members

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





GObjectClass parent_class;
  
create_session_id ()
create a new random session id. Subclasses can create
+custom session ids and should not check if the session exists.
 
create_session ()
make a new session object.
 
session_removed ()
a session was removed from the pool
 

+

+

+

+

+
Property Details

+

+
The “max-sessions” property

+
  “max-sessions”             guint

+
the maximum amount of sessions (0 = unlimited).

+
Flags: Read / Write

+
Default value: 0

+

+

+

+
Signal Details

+

+
The “session-removed” signal

+
void
+user_function (GstRTSPSessionPool *gstrtspsessionpool,
+               GstRTSPSession     *arg1,
+               gpointer            user_data)

+
Flags: Run Last

+

+

+

+
See Also

+
GstRTSPSession

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/GstRTSPStream.html b/docs/libs/html/GstRTSPStream.html
new file mode 100644
index 0000000..fee8a7d
--- /dev/null
+++ b/docs/libs/html/GstRTSPStream.html
@@ -0,0 +1,1832 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPStream
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPStream

+
GstRTSPStream — A media stream

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
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+
+
+





+GstRTSPStream *
+
+gst_rtsp_stream_new ()
+

+guint
+
+gst_rtsp_stream_get_index ()
+

+GstPad *
+
+gst_rtsp_stream_get_srcpad ()
+

+gchar *
+
+gst_rtsp_stream_get_control ()
+

+void
+
+gst_rtsp_stream_set_control ()
+

+gboolean
+
+gst_rtsp_stream_has_control ()
+

+guint
+
+gst_rtsp_stream_get_mtu ()
+

+void
+
+gst_rtsp_stream_set_mtu ()
+

+gint
+
+gst_rtsp_stream_get_dscp_qos ()
+

+void
+
+gst_rtsp_stream_set_dscp_qos ()
+

+void
+
+gst_rtsp_stream_set_profiles ()
+

+GstRTSPProfile
+
+gst_rtsp_stream_get_profiles ()
+

+GstRTSPLowerTrans
+
+gst_rtsp_stream_get_protocols ()
+

+void
+
+gst_rtsp_stream_set_protocols ()
+

+gboolean
+
+gst_rtsp_stream_is_transport_supported ()
+

+GstRTSPAddressPool *
+
+gst_rtsp_stream_get_address_pool ()
+

+void
+
+gst_rtsp_stream_set_address_pool ()
+

+GstRTSPAddress *
+
+gst_rtsp_stream_reserve_address ()
+

+gboolean
+
+gst_rtsp_stream_join_bin ()
+

+gboolean
+
+gst_rtsp_stream_leave_bin ()
+

+void
+
+gst_rtsp_stream_get_server_port ()
+

+GstRTSPAddress *
+
+gst_rtsp_stream_get_multicast_address ()
+

+GObject *
+
+gst_rtsp_stream_get_rtpsession ()
+

+void
+
+gst_rtsp_stream_get_ssrc ()
+

+gboolean
+
+gst_rtsp_stream_get_rtpinfo ()
+

+GstCaps *
+
+gst_rtsp_stream_get_caps ()
+

+guint
+
+gst_rtsp_stream_get_pt ()
+

+GstFlowReturn
+
+gst_rtsp_stream_recv_rtcp ()
+

+GstFlowReturn
+
+gst_rtsp_stream_recv_rtp ()
+

+gboolean
+
+gst_rtsp_stream_add_transport ()
+

+gboolean
+
+gst_rtsp_stream_remove_transport ()
+

+GSocket *
+
+gst_rtsp_stream_get_rtp_socket ()
+

+GSocket *
+
+gst_rtsp_stream_get_rtcp_socket ()
+

+gboolean
+
+gst_rtsp_stream_set_blocked ()
+

+gboolean
+
+gst_rtsp_stream_is_blocking ()
+

+gboolean
+
+gst_rtsp_stream_update_crypto ()
+

+GstRTSPFilterResult
+
+(*GstRTSPStreamTransportFilterFunc) ()
+

+GList *
+
+gst_rtsp_stream_transport_filter ()
+

+

+

+
Types and Values

+

++++
+
+
+
+
+
+
+
+
+
+
+




structGstRTSPStream
 GstRTSPStreamClass

+

+

+
Object Hierarchy

+
    GObject
+    ╰── GstRTSPStream
+

+

+

+
Description

+
The GstRTSPStream object manages the data transport for one stream. It
+is created from a payloader element and a source pad that produce the RTP
+packets for the stream.

+
With gst_rtsp_stream_join_bin() the streaming elements are added to the bin
+and rtpbin. gst_rtsp_stream_leave_bin() removes the elements again.

+
The GstRTSPStream will use the configured addresspool, as set with
+gst_rtsp_stream_set_address_pool(), to allocate multicast addresses for the
+stream. With gst_rtsp_stream_get_multicast_address() you can get the
+configured address.

+
With gst_rtsp_stream_get_server_port() you can get the port that the server
+will use to receive RTCP. This is the part that the clients will use to send
+RTCP to.

+
With gst_rtsp_stream_add_transport() destinations can be added where the
+stream should be sent to. Use gst_rtsp_stream_remove_transport() to remove
+the destination again.

+
Last reviewed on 2013-07-16 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_stream_new ()

+
GstRTSPStream *
+gst_rtsp_stream_new (guint idx,
+                     GstElement *payloader,
+                     GstPad *srcpad);

+
Create a new media stream with index idx
+ that handles RTP data on
+srcpad
+ and has a payloader element payloader
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





idx
an index
 
srcpad
a GstPad
 
payloader
a GstElement
 

+

+

+
Returns

+
 a new GstRTSPStream. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_stream_get_index ()

+
guint
+gst_rtsp_stream_get_index (GstRTSPStream *stream);

+
Get the stream index.

+
Return: the stream index.

+

+
Parameters

+

+++++
+
+
+
+
+
+





stream
a GstRTSPStream
 

+

+

+

+

+
gst_rtsp_stream_get_srcpad ()

+
GstPad *
+gst_rtsp_stream_get_srcpad (GstRTSPStream *stream);

+
Get the srcpad associated with stream
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





stream
a GstRTSPStream
 

+

+

+
Returns

+
 the srcpad. Unref after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_stream_get_control ()

+
gchar *
+gst_rtsp_stream_get_control (GstRTSPStream *stream);

+
Get the control string to identify this stream.

+

+
Parameters

+

+++++
+
+
+
+
+
+





stream
a GstRTSPStream
 

+

+

+
Returns

+
 the control string. g_free() after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_stream_set_control ()

+
void
+gst_rtsp_stream_set_control (GstRTSPStream *stream,
+                             const gchar *control);

+
Set the control string in stream
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
control
a control string
 

+

+

+

+

+
gst_rtsp_stream_has_control ()

+
gboolean
+gst_rtsp_stream_has_control (GstRTSPStream *stream,
+                             const gchar *control);

+
Check if stream
+ has the control string control
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
control
a control string
 

+

+

+
Returns

+
 TRUE is stream
+has control
+as the control string

+

+

+

+

+

+
gst_rtsp_stream_get_mtu ()

+
guint
+gst_rtsp_stream_get_mtu (GstRTSPStream *stream);

+
Get the configured MTU in the payloader of stream
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





stream
a GstRTSPStream
 

+

+

+
Returns

+
 the MTU of the payloader.

+

+

+

+

+

+
gst_rtsp_stream_set_mtu ()

+
void
+gst_rtsp_stream_set_mtu (GstRTSPStream *stream,
+                         guint mtu);

+
Configure the mtu in the payloader of stream
+ to mtu
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
mtu
a new MTU
 

+

+

+

+

+
gst_rtsp_stream_get_dscp_qos ()

+
gint
+gst_rtsp_stream_get_dscp_qos (GstRTSPStream *stream);

+
Get the configured DSCP QoS in of the outgoing sockets.

+

+
Parameters

+

+++++
+
+
+
+
+
+





stream
a GstRTSPStream
 

+

+

+
Returns

+
 the DSCP QoS value of the outgoing sockets, or -1 if disbled.

+

+

+

+

+

+
gst_rtsp_stream_set_dscp_qos ()

+
void
+gst_rtsp_stream_set_dscp_qos (GstRTSPStream *stream,
+                              gint dscp_qos);

+
Configure the dscp qos of the outgoing sockets to dscp_qos
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
dscp_qos
a new dscp qos value (0-63, or -1 to disable)
 

+

+

+

+

+
gst_rtsp_stream_set_profiles ()

+
void
+gst_rtsp_stream_set_profiles (GstRTSPStream *stream,
+                              GstRTSPProfile profiles);

+
Configure the allowed profiles for stream
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
profiles
the new profiles
 

+

+

+

+

+
gst_rtsp_stream_get_profiles ()

+
GstRTSPProfile
+gst_rtsp_stream_get_profiles (GstRTSPStream *stream);

+
Get the allowed profiles of stream
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





stream
a GstRTSPStream
 

+

+

+
Returns

+
 a GstRTSPProfile

+

+

+

+

+

+
gst_rtsp_stream_get_protocols ()

+
GstRTSPLowerTrans
+gst_rtsp_stream_get_protocols (GstRTSPStream *stream);

+
Get the allowed protocols of stream
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





stream
a GstRTSPStream
 

+

+

+
Returns

+
 a GstRTSPLowerTrans

+

+

+

+

+

+
gst_rtsp_stream_set_protocols ()

+
void
+gst_rtsp_stream_set_protocols (GstRTSPStream *stream,
+                               GstRTSPLowerTrans protocols);

+
Configure the allowed lower transport for stream
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
protocols
the new flags
 

+

+

+

+

+
gst_rtsp_stream_is_transport_supported ()

+
gboolean
+gst_rtsp_stream_is_transport_supported
+                               (GstRTSPStream *stream,
+                                GstRTSPTransport *transport);

+
Check if transport
+ can be handled by stream

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
transport
 a GstRTSPTransport. 
[transfer none]

+

+

+
Returns

+
 TRUE if transport
+can be handled by stream
+.

+

+

+

+

+

+
gst_rtsp_stream_get_address_pool ()

+
GstRTSPAddressPool *
+gst_rtsp_stream_get_address_pool (GstRTSPStream *stream);

+
Get the GstRTSPAddressPool used as the address pool of stream
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





stream
a GstRTSPStream
 

+

+

+
Returns

+
 the GstRTSPAddressPool of stream
+. g_object_unref() after
+usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_stream_set_address_pool ()

+
void
+gst_rtsp_stream_set_address_pool (GstRTSPStream *stream,
+                                  GstRTSPAddressPool *pool);

+
configure pool
+ to be used as the address pool of stream
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
pool
 a GstRTSPAddressPool. 
[transfer none]

+

+

+

+

+
gst_rtsp_stream_reserve_address ()

+
GstRTSPAddress *
+gst_rtsp_stream_reserve_address (GstRTSPStream *stream,
+                                 const gchar *address,
+                                 guint port,
+                                 guint n_ports,
+                                 guint ttl);

+
Reserve address
+ and port
+ as the address and port of stream
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
address
an address
 
port
a port
 
n_ports
n_ports
 
ttl
a TTL
 

+

+

+
Returns

+
 the GstRTSPAddress of stream
+or NULL when
+the address could be reserved. gst_rtsp_address_free() after usage. 

+
[nullable]

+

+

+

+

+
gst_rtsp_stream_join_bin ()

+
gboolean
+gst_rtsp_stream_join_bin (GstRTSPStream *stream,
+                          GstBin *bin,
+                          GstElement *rtpbin,
+                          GstState state);

+
Join the GstBin bin
+ that contains the element rtpbin
+.

+
stream
+ will link to rtpbin
+, which must be inside bin
+. The elements
+added to bin
+ will be set to the state given in state
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
bin
 a GstBin to join. 
[transfer none]
rtpbin
 a rtpbin element in bin
+. 
[transfer none]
state
the target state of the new elements
 

+

+

+
Returns

+
 TRUE on success.

+

+

+

+

+

+
gst_rtsp_stream_leave_bin ()

+
gboolean
+gst_rtsp_stream_leave_bin (GstRTSPStream *stream,
+                           GstBin *bin,
+                           GstElement *rtpbin);

+
Remove the elements of stream
+ from bin
+.

+
Return: TRUE on success.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
bin
 a GstBin. 
[transfer none]
rtpbin
 a rtpbin GstElement. 
[transfer none]

+

+

+

+

+
gst_rtsp_stream_get_server_port ()

+
void
+gst_rtsp_stream_get_server_port (GstRTSPStream *stream,
+                                 GstRTSPRange *server_port,
+                                 GSocketFamily family);

+
Fill server_port
+ with the port pair used by the server. This function can
+only be called when stream
+ has been joined.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
server_port
 result server port. 
[out]
family
the port family to get
 

+

+

+

+

+
gst_rtsp_stream_get_multicast_address ()

+
GstRTSPAddress *
+gst_rtsp_stream_get_multicast_address (GstRTSPStream *stream,
+                                       GSocketFamily family);

+
Get the multicast address of stream
+ for family
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
family
the GSocketFamily
 

+

+

+
Returns

+
 the GstRTSPAddress of stream
+or NULL when no address could be allocated. gst_rtsp_address_free()
+after usage. 

+
[transfer full][nullable]

+

+

+

+

+
gst_rtsp_stream_get_rtpsession ()

+
GObject *
+gst_rtsp_stream_get_rtpsession (GstRTSPStream *stream);

+
Get the RTP session of this stream.

+

+
Parameters

+

+++++
+
+
+
+
+
+





stream
a GstRTSPStream
 

+

+

+
Returns

+
 The RTP session of this stream. Unref after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_stream_get_ssrc ()

+
void
+gst_rtsp_stream_get_ssrc (GstRTSPStream *stream,
+                          guint *ssrc);

+
Get the SSRC used by the RTP session of this stream. This function can only
+be called when stream
+ has been joined.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
ssrc
 result ssrc. 
[out]

+

+

+

+

+
gst_rtsp_stream_get_rtpinfo ()

+
gboolean
+gst_rtsp_stream_get_rtpinfo (GstRTSPStream *stream,
+                             guint *rtptime,
+                             guint *seq,
+                             guint *clock_rate,
+                             GstClockTime *running_time);

+
Retrieve the current rtptime, seq and running-time. This is used to
+construct a RTPInfo reply header.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
rtptime
 result RTP timestamp. 
[allow-none]
seq
 result RTP seqnum. 
[allow-none]
clock_rate
 the clock rate. 
[allow-none]
running_time
 result running-time. 
[allow-none]

+

+

+
Returns

+
 TRUE when rtptime, seq and running-time could be determined.

+

+

+

+

+

+
gst_rtsp_stream_get_caps ()

+
GstCaps *
+gst_rtsp_stream_get_caps (GstRTSPStream *stream);

+
Retrieve the current caps of stream
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





stream
a GstRTSPStream
 

+

+

+
Returns

+
 the GstCaps of stream
+. use gst_caps_unref()
+after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_stream_get_pt ()

+
guint
+gst_rtsp_stream_get_pt (GstRTSPStream *stream);

+
Get the stream payload type.

+
Return: the stream payload type.

+

+
Parameters

+

+++++
+
+
+
+
+
+





stream
a GstRTSPStream
 

+

+

+

+

+
gst_rtsp_stream_recv_rtcp ()

+
GstFlowReturn
+gst_rtsp_stream_recv_rtcp (GstRTSPStream *stream,
+                           GstBuffer *buffer);

+
Handle an RTCP buffer for the stream. This method is usually called when a
+message has been received from a client using the TCP transport.

+
This function takes ownership of buffer
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
buffer
 a GstBuffer. 
[transfer full]

+

+

+
Returns

+
 a GstFlowReturn.

+

+

+

+

+

+
gst_rtsp_stream_recv_rtp ()

+
GstFlowReturn
+gst_rtsp_stream_recv_rtp (GstRTSPStream *stream,
+                          GstBuffer *buffer);

+
Handle an RTP buffer for the stream. This method is usually called when a
+message has been received from a client using the TCP transport.

+
This function takes ownership of buffer
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
buffer
 a GstBuffer. 
[transfer full]

+

+

+
Returns

+
 a GstFlowReturn.

+

+

+

+

+

+
gst_rtsp_stream_add_transport ()

+
gboolean
+gst_rtsp_stream_add_transport (GstRTSPStream *stream,
+                               GstRTSPStreamTransport *trans);

+
Add the transport in trans
+ to stream
+. The media of stream
+ will
+then also be send to the values configured in trans
+.

+
stream
+ must be joined to a bin.

+
trans
+ must contain a valid GstRTSPTransport.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
trans
 a GstRTSPStreamTransport. 
[transfer none]

+

+

+
Returns

+
 TRUE if trans
+was added

+

+

+

+

+

+
gst_rtsp_stream_remove_transport ()

+
gboolean
+gst_rtsp_stream_remove_transport (GstRTSPStream *stream,
+                                  GstRTSPStreamTransport *trans);

+
Remove the transport in trans
+ from stream
+. The media of stream
+ will
+not be sent to the values configured in trans
+.

+
stream
+ must be joined to a bin.

+
trans
+ must contain a valid GstRTSPTransport.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
trans
 a GstRTSPStreamTransport. 
[transfer none]

+

+

+
Returns

+
 TRUE if trans
+was removed

+

+

+

+

+

+
gst_rtsp_stream_get_rtp_socket ()

+
GSocket *
+gst_rtsp_stream_get_rtp_socket (GstRTSPStream *stream,
+                                GSocketFamily family);

+
Get the RTP socket from stream
+ for a family
+.

+
stream
+ must be joined to a bin.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
family
the socket family
 

+

+

+
Returns

+
 the RTP socket or NULL if no
+socket could be allocated for family
+. Unref after usage. 

+
[transfer full][nullable]

+

+

+

+

+
gst_rtsp_stream_get_rtcp_socket ()

+
GSocket *
+gst_rtsp_stream_get_rtcp_socket (GstRTSPStream *stream,
+                                 GSocketFamily family);

+
Get the RTCP socket from stream
+ for a family
+.

+
stream
+ must be joined to a bin.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
family
the socket family
 

+

+

+
Returns

+
 the RTCP socket or NULL if no
+socket could be allocated for family
+. Unref after usage. 

+
[transfer full][nullable]

+

+

+

+

+
gst_rtsp_stream_set_blocked ()

+
gboolean
+gst_rtsp_stream_set_blocked (GstRTSPStream *stream,
+                             gboolean blocked);

+
Blocks or unblocks the dataflow on stream
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
blocked
boolean indicating we should block or unblock
 

+

+

+
Returns

+
 TRUE on success

+

+

+

+

+

+
gst_rtsp_stream_is_blocking ()

+
gboolean
+gst_rtsp_stream_is_blocking (GstRTSPStream *stream);

+
Check if stream
+ is blocking on a GstBuffer.

+

+
Parameters

+

+++++
+
+
+
+
+
+





stream
a GstRTSPStream
 

+

+

+
Returns

+
 TRUE if stream
+is blocking

+

+

+

+

+

+
gst_rtsp_stream_update_crypto ()

+
gboolean
+gst_rtsp_stream_update_crypto (GstRTSPStream *stream,
+                               guint ssrc,
+                               GstCaps *crypto);

+
Update the new crypto information for ssrc
+ in stream
+. If information
+for ssrc
+ did not exist, it will be added. If information
+for ssrc
+ existed, it will be replaced. If crypto
+ is NULL, it will
+be removed from stream
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
ssrc
the SSRC
 
crypto
 a GstCaps with crypto info. 
[transfer none][allow-none]

+

+

+
Returns

+
 TRUE if crypto
+could be updated

+

+

+

+

+

+
GstRTSPStreamTransportFilterFunc ()

+
GstRTSPFilterResult
+(*GstRTSPStreamTransportFilterFunc) (GstRTSPStream *stream,
+                                     GstRTSPStreamTransport *trans,
+                                     gpointer user_data);

+
This function will be called by the gst_rtsp_stream_transport_filter(). An
+implementation should return a value of GstRTSPFilterResult.

+
When this function returns GST_RTSP_FILTER_REMOVE, trans
+ will be removed
+from stream
+.

+
A return value of GST_RTSP_FILTER_KEEP will leave trans
+ untouched in
+stream
+.

+
A value of GST_RTSP_FILTER_REF will add trans
+ to the result GList of
+gst_rtsp_stream_transport_filter().

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream object
 
trans
a GstRTSPStreamTransport in stream
+
 
user_data
user data that has been given to gst_rtsp_stream_transport_filter()
 

+

+

+
Returns

+
 a GstRTSPFilterResult.

+

+

+

+

+

+
gst_rtsp_stream_transport_filter ()

+
GList *
+gst_rtsp_stream_transport_filter (GstRTSPStream *stream,
+                                  GstRTSPStreamTransportFilterFunc func,
+                                  gpointer user_data);

+
Call func
+ for each transport managed by stream
+. The result value of func
+
+determines what happens to the transport. func
+ will be called with stream
+
+locked so no further actions on stream
+ can be performed from func
+.

+
If func
+ returns GST_RTSP_FILTER_REMOVE, the transport will be removed from
+stream
+.

+
If func
+ returns GST_RTSP_FILTER_KEEP, the transport will remain in stream
+.

+
If func
+ returns GST_RTSP_FILTER_REF, the transport will remain in stream
+ but
+will also be added with an additional ref to the result GList of this
+function..

+
When func
+ is NULL, GST_RTSP_FILTER_REF will be assumed for each transport.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
func
 a callback. 
[scope call][allow-none]
user_data
 user data passed to func
+. 
[closure]

+

+

+
Returns

+
 a GList with all
+transports for which func
+returned GST_RTSP_FILTER_REF. After usage, each
+element in the GList should be unreffed before the list is freed. 

+
[element-type GstRTSPStreamTransport][transfer full]

+

+

+

+

+
Types and Values

+

+
struct GstRTSPStream

+
struct GstRTSPStream;

+
The definition of a media stream.

+

+

+

+
GstRTSPStreamClass

+
typedef struct _GstRTSPStreamClass GstRTSPStreamClass;

+

+

+

+
See Also

+
GstRTSPMedia

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/annotation-glossary.html b/docs/libs/html/annotation-glossary.html
new file mode 100644
index 0000000..4326d28
--- /dev/null
+++ b/docs/libs/html/annotation-glossary.html
@@ -0,0 +1,70 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: Annotation Glossary
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+Annotation Glossary

+
A

+
allow-none

+
NULL is OK, both for passing and for returning.

+
C

+
closure

+
This parameter is a 'user_data', for callbacks; many bindings can pass NULL here.

+
E

+
element-type

+
Generics and defining elements of containers and arrays.

+
N

+
nullable

+
NULL may be passed as the value in, out, in-out; or as a return value.

+
O

+
out

+
Parameter for returning results. Default is transfer full.

+
S

+
scope call

+
The callback is valid only during the call to the method.

+
scope notified

+
The callback is valid until the GDestroyNotify argument is called.

+
T

+
transfer floating

+
Alias for transfer none, used for objects with floating refs.

+
transfer full

+
Free data after the code is done.

+
transfer none

+
Don't free data after the code is done.

+
type

+
Override the parsed C type with given type.

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/api-index-full.html b/docs/libs/html/api-index-full.html
new file mode 100644
index 0000000..671e5d5
--- /dev/null
+++ b/docs/libs/html/api-index-full.html
@@ -0,0 +1,1470 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: API Index
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+API Index

+
R

+

+GstRTSPAddress, struct in GstRTSPAddressPool
+

+

+

+GstRTSPAddressFlags, enum in GstRTSPAddressPool
+

+

+

+GstRTSPAddressPool, struct in GstRTSPAddressPool
+

+

+

+GstRTSPAddressPoolClass, struct in GstRTSPAddressPool
+

+

+

+GstRTSPAddressPoolResult, enum in GstRTSPAddressPool
+

+

+

+GstRTSPAuth, struct in GstRTSPAuth
+

+

+

+GstRTSPAuthClass, struct in GstRTSPAuth
+

+

+

+GstRTSPClient, struct in GstRTSPClient
+

+

+

+GstRTSPClient::closed, object signal in GstRTSPClient
+

+

+

+GstRTSPClient::describe-request, object signal in GstRTSPClient
+

+

+

+GstRTSPClient::get-parameter-request, object signal in GstRTSPClient
+

+

+

+GstRTSPClient::handle-response, object signal in GstRTSPClient
+

+

+

+GstRTSPClient::new-session, object signal in GstRTSPClient
+

+

+

+GstRTSPClient::options-request, object signal in GstRTSPClient
+

+

+

+GstRTSPClient::pause-request, object signal in GstRTSPClient
+

+

+

+GstRTSPClient::play-request, object signal in GstRTSPClient
+

+

+

+GstRTSPClient::send-message, object signal in GstRTSPClient
+

+

+

+GstRTSPClient::set-parameter-request, object signal in GstRTSPClient
+

+

+

+GstRTSPClient::setup-request, object signal in GstRTSPClient
+

+

+

+GstRTSPClient::teardown-request, object signal in GstRTSPClient
+

+

+

+GstRTSPClient:drop-backlog, object property in GstRTSPClient
+

+

+

+GstRTSPClient:mount-points, object property in GstRTSPClient
+

+

+

+GstRTSPClient:session-pool, object property in GstRTSPClient
+

+

+

+GstRTSPClientClass, struct in GstRTSPClient
+

+

+

+GstRTSPClientSendFunc, user_function in GstRTSPClient
+

+

+

+GstRTSPClientSessionFilterFunc, user_function in GstRTSPClient
+

+

+

+GstRTSPContext, struct in GstRTSPContext
+

+

+

+GstRTSPFilterResult, enum in GstRTSPSession
+

+

+

+GstRTSPKeepAliveFunc, user_function in GstRTSPStreamTransport
+

+

+

+GstRTSPMedia, struct in GstRTSPMedia
+

+

+

+GstRTSPMedia::new-state, object signal in GstRTSPMedia
+

+

+

+GstRTSPMedia::new-stream, object signal in GstRTSPMedia
+

+

+

+GstRTSPMedia::prepared, object signal in GstRTSPMedia
+

+

+

+GstRTSPMedia::removed-stream, object signal in GstRTSPMedia
+

+

+

+GstRTSPMedia::target-state, object signal in GstRTSPMedia
+

+

+

+GstRTSPMedia::unprepared, object signal in GstRTSPMedia
+

+

+

+GstRTSPMedia:buffer-size, object property in GstRTSPMedia
+

+

+

+GstRTSPMedia:element, object property in GstRTSPMedia
+

+

+

+GstRTSPMedia:eos-shutdown, object property in GstRTSPMedia
+

+

+

+GstRTSPMedia:profiles, object property in GstRTSPMedia
+

+

+

+GstRTSPMedia:protocols, object property in GstRTSPMedia
+

+

+

+GstRTSPMedia:reusable, object property in GstRTSPMedia
+

+

+

+GstRTSPMedia:shared, object property in GstRTSPMedia
+

+

+

+GstRTSPMedia:suspend-mode, object property in GstRTSPMedia
+

+

+

+GstRTSPMedia:time-provider, object property in GstRTSPMedia
+

+

+

+GstRTSPMediaClass, struct in GstRTSPMedia
+

+

+

+GstRTSPMediaFactory, struct in GstRTSPMediaFactory
+

+

+

+GstRTSPMediaFactory::media-configure, object signal in GstRTSPMediaFactory
+

+

+

+GstRTSPMediaFactory::media-constructed, object signal in GstRTSPMediaFactory
+

+

+

+GstRTSPMediaFactory:buffer-size, object property in GstRTSPMediaFactory
+

+

+

+GstRTSPMediaFactory:eos-shutdown, object property in GstRTSPMediaFactory
+

+

+

+GstRTSPMediaFactory:launch, object property in GstRTSPMediaFactory
+

+

+

+GstRTSPMediaFactory:profiles, object property in GstRTSPMediaFactory
+

+

+

+GstRTSPMediaFactory:protocols, object property in GstRTSPMediaFactory
+

+

+

+GstRTSPMediaFactory:shared, object property in GstRTSPMediaFactory
+

+

+

+GstRTSPMediaFactory:suspend-mode, object property in GstRTSPMediaFactory
+

+

+

+GstRTSPMediaFactoryClass, struct in GstRTSPMediaFactory
+

+

+

+GstRTSPMediaFactoryURI, struct in GstRTSPMediaFactoryURI
+

+

+

+GstRTSPMediaFactoryURIClass, struct in GstRTSPMediaFactoryURI
+

+

+

+GstRTSPMediaStatus, enum in GstRTSPMedia
+

+

+

+GstRTSPMountPoints, struct in GstRTSPMountPoints
+

+

+

+GstRTSPMountPointsClass, struct in GstRTSPMountPoints
+

+

+

+GstRTSPPermissions, struct in GstRTSPPermissions
+

+

+

+GstRTSPSendFunc, user_function in GstRTSPStreamTransport
+

+

+

+GstRTSPServer, struct in GstRTSPServer
+

+

+

+GstRTSPServer::client-connected, object signal in GstRTSPServer
+

+

+

+GstRTSPServer:address, object property in GstRTSPServer
+

+

+

+GstRTSPServer:backlog, object property in GstRTSPServer
+

+

+

+GstRTSPServer:bound-port, object property in GstRTSPServer
+

+

+

+GstRTSPServer:mount-points, object property in GstRTSPServer
+

+

+

+GstRTSPServer:service, object property in GstRTSPServer
+

+

+

+GstRTSPServer:session-pool, object property in GstRTSPServer
+

+

+

+GstRTSPServerClass, struct in GstRTSPServer
+

+

+

+GstRTSPServerClientFilterFunc, user_function in GstRTSPServer
+

+

+

+GstRTSPSession, struct in GstRTSPSession
+

+

+

+GstRTSPSession:sessionid, object property in GstRTSPSession
+

+

+

+GstRTSPSession:timeout, object property in GstRTSPSession
+

+

+

+GstRTSPSession:timeout-always-visible, object property in GstRTSPSession
+

+

+

+GstRTSPSessionClass, struct in GstRTSPSession
+

+

+

+GstRTSPSessionFilterFunc, user_function in GstRTSPSession
+

+

+

+GstRTSPSessionMedia, struct in GstRTSPSessionMedia
+

+

+

+GstRTSPSessionMediaClass, struct in GstRTSPSessionMedia
+

+

+

+GstRTSPSessionPool, struct in GstRTSPSessionPool
+

+

+

+GstRTSPSessionPool::session-removed, object signal in GstRTSPSessionPool
+

+

+

+GstRTSPSessionPool:max-sessions, object property in GstRTSPSessionPool
+

+

+

+GstRTSPSessionPoolClass, struct in GstRTSPSessionPool
+

+

+

+GstRTSPSessionPoolFilterFunc, user_function in GstRTSPSessionPool
+

+

+

+GstRTSPSessionPoolFunc, user_function in GstRTSPSessionPool
+

+

+

+GstRTSPStream, struct in GstRTSPStream
+

+

+

+GstRTSPStreamClass, struct in GstRTSPStream
+

+

+

+GstRTSPStreamTransport, struct in GstRTSPStreamTransport
+

+

+

+GstRTSPStreamTransportClass, struct in GstRTSPStreamTransport
+

+

+

+GstRTSPStreamTransportFilterFunc, user_function in GstRTSPStream
+

+

+

+GstRTSPSuspendMode, enum in GstRTSPMedia
+

+

+

+GstRTSPThread, struct in GstRTSPThreadPool
+

+

+

+GstRTSPThreadPool, struct in GstRTSPThreadPool
+

+

+

+GstRTSPThreadPoolClass, struct in GstRTSPThreadPool
+

+

+

+GstRTSPThreadType, enum in GstRTSPThreadPool
+

+

+

+GstRTSPToken, struct in GstRTSPToken
+

+

+

+gst_rtsp_address_copy, function in GstRTSPAddressPool
+

+

+

+gst_rtsp_address_free, function in GstRTSPAddressPool
+

+

+

+gst_rtsp_address_pool_acquire_address, function in GstRTSPAddressPool
+

+

+

+gst_rtsp_address_pool_add_range, function in GstRTSPAddressPool
+

+

+

+GST_RTSP_ADDRESS_POOL_ANY_IPV4, macro in GstRTSPAddressPool
+

+

+

+GST_RTSP_ADDRESS_POOL_ANY_IPV6, macro in GstRTSPAddressPool
+

+

+

+gst_rtsp_address_pool_clear, function in GstRTSPAddressPool
+

+

+

+gst_rtsp_address_pool_dump, function in GstRTSPAddressPool
+

+

+

+gst_rtsp_address_pool_has_unicast_addresses, function in GstRTSPAddressPool
+

+

+

+gst_rtsp_address_pool_new, function in GstRTSPAddressPool
+

+

+

+gst_rtsp_address_pool_reserve_address, function in GstRTSPAddressPool
+

+

+

+gst_rtsp_auth_add_basic, function in GstRTSPAuth
+

+

+

+gst_rtsp_auth_check, function in GstRTSPAuth
+

+

+

+GST_RTSP_AUTH_CHECK_CONNECT, macro in GstRTSPAuth
+

+

+

+GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS, macro in GstRTSPAuth
+

+

+

+GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT, macro in GstRTSPAuth
+

+

+

+GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS, macro in GstRTSPAuth
+

+

+

+GST_RTSP_AUTH_CHECK_URL, macro in GstRTSPAuth
+

+

+

+gst_rtsp_auth_get_default_token, function in GstRTSPAuth
+

+

+

+gst_rtsp_auth_get_tls_certificate, function in GstRTSPAuth
+

+

+

+gst_rtsp_auth_make_basic, function in GstRTSPAuth
+

+

+

+gst_rtsp_auth_new, function in GstRTSPAuth
+

+

+

+gst_rtsp_auth_remove_basic, function in GstRTSPAuth
+

+

+

+gst_rtsp_auth_set_default_token, function in GstRTSPAuth
+

+

+

+gst_rtsp_auth_set_tls_certificate, function in GstRTSPAuth
+

+

+

+gst_rtsp_client_attach, function in GstRTSPClient
+

+

+

+gst_rtsp_client_close, function in GstRTSPClient
+

+

+

+gst_rtsp_client_get_auth, function in GstRTSPClient
+

+

+

+gst_rtsp_client_get_connection, function in GstRTSPClient
+

+

+

+gst_rtsp_client_get_mount_points, function in GstRTSPClient
+

+

+

+gst_rtsp_client_get_session_pool, function in GstRTSPClient
+

+

+

+gst_rtsp_client_get_thread_pool, function in GstRTSPClient
+

+

+

+gst_rtsp_client_handle_message, function in GstRTSPClient
+

+

+

+gst_rtsp_client_new, function in GstRTSPClient
+

+

+

+gst_rtsp_client_send_message, function in GstRTSPClient
+

+

+

+gst_rtsp_client_session_filter, function in GstRTSPClient
+

+

+

+gst_rtsp_client_set_auth, function in GstRTSPClient
+

+

+

+gst_rtsp_client_set_connection, function in GstRTSPClient
+

+

+

+gst_rtsp_client_set_mount_points, function in GstRTSPClient
+

+

+

+gst_rtsp_client_set_send_func, function in GstRTSPClient
+

+

+

+gst_rtsp_client_set_session_pool, function in GstRTSPClient
+

+

+

+gst_rtsp_client_set_thread_pool, function in GstRTSPClient
+

+

+

+gst_rtsp_context_get_current, function in GstRTSPContext
+

+

+

+gst_rtsp_context_pop_current, function in GstRTSPContext
+

+

+

+gst_rtsp_context_push_current, function in GstRTSPContext
+

+

+

+gst_rtsp_media_collect_streams, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_create_stream, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_factory_add_role, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_construct, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_create_element, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_get_address_pool, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_get_buffer_size, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_get_launch, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_get_permissions, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_get_profiles, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_get_protocols, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_get_suspend_mode, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_is_eos_shutdown, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_is_shared, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_new, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_set_address_pool, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_set_buffer_size, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_set_eos_shutdown, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_set_launch, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_set_permissions, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_set_profiles, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_set_protocols, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_set_shared, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_set_suspend_mode, function in GstRTSPMediaFactory
+

+

+

+gst_rtsp_media_factory_uri_get_uri, function in GstRTSPMediaFactoryURI
+

+

+

+gst_rtsp_media_factory_uri_new, function in GstRTSPMediaFactoryURI
+

+

+

+gst_rtsp_media_factory_uri_set_uri, function in GstRTSPMediaFactoryURI
+

+

+

+gst_rtsp_media_find_stream, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_get_address_pool, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_get_base_time, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_get_buffer_size, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_get_clock, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_get_element, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_get_permissions, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_get_profiles, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_get_protocols, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_get_range_string, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_get_status, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_get_stream, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_get_suspend_mode, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_get_time_provider, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_is_eos_shutdown, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_is_reusable, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_is_shared, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_is_time_provider, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_new, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_n_streams, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_prepare, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_seek, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_setup_sdp, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_set_address_pool, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_set_buffer_size, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_set_eos_shutdown, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_set_permissions, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_set_pipeline_state, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_set_profiles, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_set_protocols, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_set_reusable, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_set_shared, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_set_state, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_set_suspend_mode, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_suspend, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_take_pipeline, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_unprepare, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_unsuspend, function in GstRTSPMedia
+

+

+

+gst_rtsp_media_use_time_provider, function in GstRTSPMedia
+

+

+

+gst_rtsp_mount_points_add_factory, function in GstRTSPMountPoints
+

+

+

+gst_rtsp_mount_points_make_path, function in GstRTSPMountPoints
+

+

+

+gst_rtsp_mount_points_match, function in GstRTSPMountPoints
+

+

+

+gst_rtsp_mount_points_new, function in GstRTSPMountPoints
+

+

+

+gst_rtsp_mount_points_remove_factory, function in GstRTSPMountPoints
+

+

+

+gst_rtsp_params_get, function in GstRTSPParams
+

+

+

+gst_rtsp_params_set, function in GstRTSPParams
+

+

+

+gst_rtsp_permissions_add_role, function in GstRTSPPermissions
+

+

+

+gst_rtsp_permissions_add_role_valist, function in GstRTSPPermissions
+

+

+

+gst_rtsp_permissions_get_role, function in GstRTSPPermissions
+

+

+

+gst_rtsp_permissions_is_allowed, function in GstRTSPPermissions
+

+

+

+gst_rtsp_permissions_new, function in GstRTSPPermissions
+

+

+

+gst_rtsp_permissions_ref, function in GstRTSPPermissions
+

+

+

+gst_rtsp_permissions_remove_role, function in GstRTSPPermissions
+

+

+

+gst_rtsp_permissions_unref, function in GstRTSPPermissions
+

+

+

+GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, macro in GstRTSPAuth
+

+

+

+GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, macro in GstRTSPAuth
+

+

+

+gst_rtsp_sdp_from_media, function in GstRTSPSdp
+

+

+

+gst_rtsp_server_attach, function in GstRTSPServer
+

+

+

+gst_rtsp_server_client_filter, function in GstRTSPServer
+

+

+

+gst_rtsp_server_create_socket, function in GstRTSPServer
+

+

+

+gst_rtsp_server_create_source, function in GstRTSPServer
+

+

+

+gst_rtsp_server_get_address, function in GstRTSPServer
+

+

+

+gst_rtsp_server_get_auth, function in GstRTSPServer
+

+

+

+gst_rtsp_server_get_backlog, function in GstRTSPServer
+

+

+

+gst_rtsp_server_get_bound_port, function in GstRTSPServer
+

+

+

+gst_rtsp_server_get_mount_points, function in GstRTSPServer
+

+

+

+gst_rtsp_server_get_service, function in GstRTSPServer
+

+

+

+gst_rtsp_server_get_session_pool, function in GstRTSPServer
+

+

+

+gst_rtsp_server_get_thread_pool, function in GstRTSPServer
+

+

+

+gst_rtsp_server_io_func, function in GstRTSPServer
+

+

+

+gst_rtsp_server_new, function in GstRTSPServer
+

+

+

+gst_rtsp_server_set_address, function in GstRTSPServer
+

+

+

+gst_rtsp_server_set_auth, function in GstRTSPServer
+

+

+

+gst_rtsp_server_set_backlog, function in GstRTSPServer
+

+

+

+gst_rtsp_server_set_mount_points, function in GstRTSPServer
+

+

+

+gst_rtsp_server_set_service, function in GstRTSPServer
+

+

+

+gst_rtsp_server_set_session_pool, function in GstRTSPServer
+

+

+

+gst_rtsp_server_set_thread_pool, function in GstRTSPServer
+

+

+

+gst_rtsp_server_transfer_connection, function in GstRTSPServer
+

+

+

+gst_rtsp_session_allow_expire, function in GstRTSPSession
+

+

+

+gst_rtsp_session_filter, function in GstRTSPSession
+

+

+

+gst_rtsp_session_get_header, function in GstRTSPSession
+

+

+

+gst_rtsp_session_get_media, function in GstRTSPSession
+

+

+

+gst_rtsp_session_get_sessionid, function in GstRTSPSession
+

+

+

+gst_rtsp_session_get_timeout, function in GstRTSPSession
+

+

+

+gst_rtsp_session_is_expired, function in GstRTSPSession
+

+

+

+gst_rtsp_session_manage_media, function in GstRTSPSession
+

+

+

+gst_rtsp_session_media_alloc_channels, function in GstRTSPSessionMedia
+

+

+

+gst_rtsp_session_media_get_base_time, function in GstRTSPSessionMedia
+

+

+

+gst_rtsp_session_media_get_media, function in GstRTSPSessionMedia
+

+

+

+gst_rtsp_session_media_get_rtpinfo, function in GstRTSPSessionMedia
+

+

+

+gst_rtsp_session_media_get_rtsp_state, function in GstRTSPSessionMedia
+

+

+

+gst_rtsp_session_media_get_transport, function in GstRTSPSessionMedia
+

+

+

+gst_rtsp_session_media_matches, function in GstRTSPSessionMedia
+

+

+

+gst_rtsp_session_media_new, function in GstRTSPSessionMedia
+

+

+

+gst_rtsp_session_media_set_rtsp_state, function in GstRTSPSessionMedia
+

+

+

+gst_rtsp_session_media_set_state, function in GstRTSPSessionMedia
+

+

+

+gst_rtsp_session_media_set_transport, function in GstRTSPSessionMedia
+

+

+

+gst_rtsp_session_new, function in GstRTSPSession
+

+

+

+gst_rtsp_session_next_timeout, function in GstRTSPSession
+

+

+

+gst_rtsp_session_pool_cleanup, function in GstRTSPSessionPool
+

+

+

+gst_rtsp_session_pool_create, function in GstRTSPSessionPool
+

+

+

+gst_rtsp_session_pool_create_watch, function in GstRTSPSessionPool
+

+

+

+gst_rtsp_session_pool_filter, function in GstRTSPSessionPool
+

+

+

+gst_rtsp_session_pool_find, function in GstRTSPSessionPool
+

+

+

+gst_rtsp_session_pool_get_max_sessions, function in GstRTSPSessionPool
+

+

+

+gst_rtsp_session_pool_get_n_sessions, function in GstRTSPSessionPool
+

+

+

+gst_rtsp_session_pool_new, function in GstRTSPSessionPool
+

+

+

+gst_rtsp_session_pool_remove, function in GstRTSPSessionPool
+

+

+

+gst_rtsp_session_pool_set_max_sessions, function in GstRTSPSessionPool
+

+

+

+gst_rtsp_session_prevent_expire, function in GstRTSPSession
+

+

+

+gst_rtsp_session_release_media, function in GstRTSPSession
+

+

+

+gst_rtsp_session_set_timeout, function in GstRTSPSession
+

+

+

+gst_rtsp_session_touch, function in GstRTSPSession
+

+

+

+gst_rtsp_stream_add_transport, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_address_pool, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_caps, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_control, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_dscp_qos, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_index, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_mtu, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_multicast_address, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_profiles, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_protocols, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_pt, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_rtcp_socket, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_rtpinfo, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_rtpsession, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_rtp_socket, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_server_port, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_srcpad, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_get_ssrc, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_has_control, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_is_blocking, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_is_transport_supported, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_join_bin, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_leave_bin, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_new, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_recv_rtcp, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_recv_rtp, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_remove_transport, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_reserve_address, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_set_address_pool, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_set_blocked, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_set_control, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_set_dscp_qos, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_set_mtu, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_set_profiles, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_set_protocols, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_transport_filter, function in GstRTSPStream
+

+

+

+gst_rtsp_stream_transport_get_rtpinfo, function in GstRTSPStreamTransport
+

+

+

+gst_rtsp_stream_transport_get_stream, function in GstRTSPStreamTransport
+

+

+

+gst_rtsp_stream_transport_get_transport, function in GstRTSPStreamTransport
+

+

+

+gst_rtsp_stream_transport_get_url, function in GstRTSPStreamTransport
+

+

+

+gst_rtsp_stream_transport_is_timed_out, function in GstRTSPStreamTransport
+

+

+

+gst_rtsp_stream_transport_keep_alive, function in GstRTSPStreamTransport
+

+

+

+gst_rtsp_stream_transport_new, function in GstRTSPStreamTransport
+

+

+

+gst_rtsp_stream_transport_send_rtcp, function in GstRTSPStreamTransport
+

+

+

+gst_rtsp_stream_transport_send_rtp, function in GstRTSPStreamTransport
+

+

+

+gst_rtsp_stream_transport_set_active, function in GstRTSPStreamTransport
+

+

+

+gst_rtsp_stream_transport_set_callbacks, function in GstRTSPStreamTransport
+

+

+

+gst_rtsp_stream_transport_set_keepalive, function in GstRTSPStreamTransport
+

+

+

+gst_rtsp_stream_transport_set_timed_out, function in GstRTSPStreamTransport
+

+

+

+gst_rtsp_stream_transport_set_transport, function in GstRTSPStreamTransport
+

+

+

+gst_rtsp_stream_transport_set_url, function in GstRTSPStreamTransport
+

+

+

+gst_rtsp_stream_update_crypto, function in GstRTSPStream
+

+

+

+gst_rtsp_thread_new, function in GstRTSPThreadPool
+

+

+

+gst_rtsp_thread_pool_cleanup, function in GstRTSPThreadPool
+

+

+

+gst_rtsp_thread_pool_get_max_threads, function in GstRTSPThreadPool
+

+

+

+gst_rtsp_thread_pool_get_thread, function in GstRTSPThreadPool
+

+

+

+gst_rtsp_thread_pool_new, function in GstRTSPThreadPool
+

+

+

+gst_rtsp_thread_pool_set_max_threads, function in GstRTSPThreadPool
+

+

+

+gst_rtsp_thread_ref, function in GstRTSPThreadPool
+

+

+

+gst_rtsp_thread_reuse, function in GstRTSPThreadPool
+

+

+

+gst_rtsp_thread_stop, function in GstRTSPThreadPool
+

+

+

+gst_rtsp_thread_unref, function in GstRTSPThreadPool
+

+

+

+gst_rtsp_token_get_string, function in GstRTSPToken
+

+

+

+gst_rtsp_token_get_structure, function in GstRTSPToken
+

+

+

+gst_rtsp_token_is_allowed, function in GstRTSPToken
+

+

+

+GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, macro in GstRTSPAuth
+

+

+

+gst_rtsp_token_new, function in GstRTSPToken
+

+

+

+gst_rtsp_token_new_empty, function in GstRTSPToken
+

+

+

+gst_rtsp_token_new_valist, function in GstRTSPToken
+

+

+

+gst_rtsp_token_ref, function in GstRTSPToken
+

+

+

+GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS, macro in GstRTSPAuth
+

+

+

+gst_rtsp_token_unref, function in GstRTSPToken
+

+

+

+gst_rtsp_token_writable_structure, function in GstRTSPToken
+

+

+
S

+

+GstSDPInfo, struct in GstRTSPSdp
+

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/ch01.html b/docs/libs/html/ch01.html
new file mode 100644
index 0000000..0173744
--- /dev/null
+++ b/docs/libs/html/ch01.html
@@ -0,0 +1,88 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: 
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+

+GstRTSPServer — The main server object
+

+

+GstRTSPClient — A client connection state
+

+

+GstRTSPContext — A client request context
+

+

+GstRTSPMountPoints — Map a path to media
+

+

+GstRTSPMediaFactory — A factory for media pipelines
+

+

+GstRTSPMediaFactoryURI — A factory for URI sources
+

+

+GstRTSPMedia — The media pipeline
+

+

+GstRTSPStream — A media stream
+

+

+GstRTSPSessionPool — An object for managing sessions
+

+

+GstRTSPSession — An object to manage media
+

+

+GstRTSPSessionMedia — Media managed in a session
+

+

+GstRTSPStreamTransport — A media stream transport configuration
+

+

+GstRTSPSdp — Make SDP messages
+

+

+GstRTSPAddressPool — A pool of network addresses
+

+

+GstRTSPThreadPool — A pool of threads
+

+

+GstRTSPAuth — Authentication and authorization
+

+

+GstRTSPToken — Roles and permissions for a client
+

+

+GstRTSPPermissions — Roles and associated permissions
+

+

+GstRTSPParams — Param get and set implementation
+

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/gst-rtsp-server-1.0.devhelp2 b/docs/libs/html/gst-rtsp-server-1.0.devhelp2
new file mode 100644
index 0000000..3569b7c
--- /dev/null
+++ b/docs/libs/html/gst-rtsp-server-1.0.devhelp2
@@ -0,0 +1,391 @@
+
+
+
+  
+    
+      
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+
diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPAddressPool.html b/docs/libs/html/gst-rtsp-server-GstRTSPAddressPool.html
new file mode 100644
index 0000000..aec2ed1
--- /dev/null
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPAddressPool.html
@@ -0,0 +1,734 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPAddressPool
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPAddressPool

+
GstRTSPAddressPool — A pool of network addresses

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





+GstRTSPAddress *
+
+gst_rtsp_address_copy ()
+

+void
+
+gst_rtsp_address_free ()
+

+GstRTSPAddressPool *
+
+gst_rtsp_address_pool_new ()
+

+void
+
+gst_rtsp_address_pool_clear ()
+

+void
+
+gst_rtsp_address_pool_dump ()
+

+gboolean
+
+gst_rtsp_address_pool_add_range ()
+

+gboolean
+
+gst_rtsp_address_pool_has_unicast_addresses ()
+

+GstRTSPAddress *
+
+gst_rtsp_address_pool_acquire_address ()
+

+GstRTSPAddressPoolResult
+
+gst_rtsp_address_pool_reserve_address ()
+

+

+

+
Types and Values

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+




#defineGST_RTSP_ADDRESS_POOL_ANY_IPV4
#defineGST_RTSP_ADDRESS_POOL_ANY_IPV6
structGstRTSPAddress
enumGstRTSPAddressFlags
structGstRTSPAddressPool
structGstRTSPAddressPoolClass
enumGstRTSPAddressPoolResult

+

+

+
Description

+
The GstRTSPAddressPool is an object that maintains a collection of network
+addresses. It is used to allocate server ports and server multicast addresses
+but also to reserve client provided destination addresses.

+
A range of addresses can be added with gst_rtsp_address_pool_add_range().
+Both multicast and unicast addresses can be added.

+
With gst_rtsp_address_pool_acquire_address() an unused address and port range
+can be acquired from the pool. With gst_rtsp_address_pool_reserve_address() a
+specific address can be retrieved. Both methods return a boxed
+GstRTSPAddress that should be freed with gst_rtsp_address_free() after
+usage, which brings the address back into the pool.

+
Last reviewed on 2013-07-16 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_address_copy ()

+
GstRTSPAddress *
+gst_rtsp_address_copy (GstRTSPAddress *addr);

+
Make a copy of addr
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





addr
a GstRTSPAddress
 

+

+

+
Returns

+
 a copy of addr
+.

+

+

+

+

+

+
gst_rtsp_address_free ()

+
void
+gst_rtsp_address_free (GstRTSPAddress *addr);

+
Free addr
+ and releasing it back into the pool when owned by a
+pool.

+

+
Parameters

+

+++++
+
+
+
+
+
+





addr
a GstRTSPAddress
 

+

+

+

+

+
gst_rtsp_address_pool_new ()

+
GstRTSPAddressPool *
+gst_rtsp_address_pool_new (void);

+
Make a new GstRTSPAddressPool.

+

+
Returns

+
 a new GstRTSPAddressPool. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_address_pool_clear ()

+
void
+gst_rtsp_address_pool_clear (GstRTSPAddressPool *pool);

+
Clear all addresses in pool
+. There should be no outstanding
+allocations.

+

+
Parameters

+

+++++
+
+
+
+
+
+





pool
a GstRTSPAddressPool
 

+

+

+

+

+
gst_rtsp_address_pool_dump ()

+
void
+gst_rtsp_address_pool_dump (GstRTSPAddressPool *pool);

+
Dump the free and allocated addresses to stdout.

+

+
Parameters

+

+++++
+
+
+
+
+
+





pool
a GstRTSPAddressPool
 

+

+

+

+

+
gst_rtsp_address_pool_add_range ()

+
gboolean
+gst_rtsp_address_pool_add_range (GstRTSPAddressPool *pool,
+                                 const gchar *min_address,
+                                 const gchar *max_address,
+                                 guint16 min_port,
+                                 guint16 max_port,
+                                 guint8 ttl);

+
Adds the addresses from min_addess
+ to max_address
+ (inclusive)
+to pool
+. The valid port range for the addresses will be from min_port
+ to
+max_port
+ inclusive.

+
When ttl
+ is 0, min_address
+ and max_address
+ should be unicast addresses.
+min_address
+ and max_address
+ can be set to
+GST_RTSP_ADDRESS_POOL_ANY_IPV4 or GST_RTSP_ADDRESS_POOL_ANY_IPV6 to bind
+to all available IPv4 or IPv6 addresses.

+
When ttl
+ > 0, min_address
+ and max_address
+ should be multicast addresses.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





pool
a GstRTSPAddressPool
 
min_address
a minimum address to add
 
max_address
a maximum address to add
 
min_port
the minimum port
 
max_port
the maximum port
 
ttl
a TTL or 0 for unicast addresses
 

+

+

+
Returns

+
 TRUE if the addresses could be added.

+

+

+

+

+

+
gst_rtsp_address_pool_has_unicast_addresses ()

+
gboolean
+gst_rtsp_address_pool_has_unicast_addresses
+                               (GstRTSPAddressPool *pool);

+
Used to know if the pool includes any unicast addresses.

+

+
Parameters

+

+++++
+
+
+
+
+
+





pool
a GstRTSPAddressPool
 

+

+

+
Returns

+
 TRUE if the pool includes any unicast addresses, FALSE otherwise

+

+

+

+

+

+
gst_rtsp_address_pool_acquire_address ()

+
GstRTSPAddress *
+gst_rtsp_address_pool_acquire_address (GstRTSPAddressPool *pool,
+                                       GstRTSPAddressFlags flags,
+                                       gint n_ports);

+
Take an address and ports from pool
+. flags
+ can be used to control the
+allocation. n_ports
+ consecutive ports will be allocated of which the first
+one can be found in port
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





pool
a GstRTSPAddressPool
 
flags
flags
 
n_ports
the amount of ports
 

+

+

+
Returns

+
 a GstRTSPAddress that should be freed with
+gst_rtsp_address_free after use or NULL when no address could be
+acquired. 

+
[nullable]

+

+

+

+

+
gst_rtsp_address_pool_reserve_address ()

+
GstRTSPAddressPoolResult
+gst_rtsp_address_pool_reserve_address (GstRTSPAddressPool *pool,
+                                       const gchar *ip_address,
+                                       guint port,
+                                       guint n_ports,
+                                       guint ttl,
+                                       GstRTSPAddress **address);

+
Take a specific address and ports from pool
+. n_ports
+ consecutive
+ports will be allocated of which the first one can be found in
+port
+.

+
If ttl
+ is 0, address
+ should be a unicast address. If ttl
+ > 0, address
+
+should be a valid multicast address.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





pool
a GstRTSPAddressPool
 
ip_address
The IP address to reserve
 
port
The first port to reserve
 
n_ports
The number of ports
 
ttl
The requested ttl
 
address
 storage for a GstRTSPAddress. 
[out]

+

+

+
Returns

+
 GST_RTSP_ADDRESS_POOL_OK if an address was reserved. The address
+is returned in address
+and should be freed with gst_rtsp_address_free
+after use.

+

+

+

+

+

+
Types and Values

+

+
GST_RTSP_ADDRESS_POOL_ANY_IPV4

+
#define GST_RTSP_ADDRESS_POOL_ANY_IPV4  "0.0.0.0"
+

+
Used with gst_rtsp_address_pool_add_range() to bind to all
+IPv4 addresses

+

+

+

+
GST_RTSP_ADDRESS_POOL_ANY_IPV6

+
#define GST_RTSP_ADDRESS_POOL_ANY_IPV6  "::"
+

+
Used with gst_rtsp_address_pool_add_range() to bind to all
+IPv6 addresses

+

+

+

+
struct GstRTSPAddress

+
struct GstRTSPAddress {
+  GstRTSPAddressPool *pool;
+
+  gchar *address;
+  guint16 port;
+  gint n_ports;
+  guint8 ttl;
+};
+

+
An address

+

+
Members

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





GstRTSPAddressPool *pool;
the GstRTSPAddressPool owner of this address
 
gchar *address;
the address
 
guint16 port;
the port number
 
gint n_ports;
number of ports
 
guint8 ttl;
TTL or 0 for unicast addresses
 

+

+

+

+

+
enum GstRTSPAddressFlags

+
Flags used to control allocation of addresses

+

+
Members

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





GST_RTSP_ADDRESS_FLAG_NONE

+
no flags

+		 
GST_RTSP_ADDRESS_FLAG_IPV4

+
an IPv4 address

+		 
GST_RTSP_ADDRESS_FLAG_IPV6

+
and IPv6 address

+		 
GST_RTSP_ADDRESS_FLAG_EVEN_PORT

+
address with an even port

+		 
GST_RTSP_ADDRESS_FLAG_MULTICAST

+
a multicast address

+		 
GST_RTSP_ADDRESS_FLAG_UNICAST

+
a unicast address

+		 

+

+

+

+

+
struct GstRTSPAddressPool

+
struct GstRTSPAddressPool {
+  GObject       parent;
+};
+

+
An address pool, all member are private

+

+
Members

+

+++++
+
+
+
+
+
+





GObject parent;
the parent GObject
 

+

+

+

+

+
struct GstRTSPAddressPoolClass

+
struct GstRTSPAddressPoolClass {
+  GObjectClass  parent_class;
+};
+

+
Opaque Address pool class.

+

+

+

+
enum GstRTSPAddressPoolResult

+
Result codes from RTSP address pool functions.

+

+
Members

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





GST_RTSP_ADDRESS_POOL_OK

+
no error

+		 
GST_RTSP_ADDRESS_POOL_EINVAL

+
invalid arguments were provided to a function

+		 
GST_RTSP_ADDRESS_POOL_ERESERVED

+
the addres has already been reserved

+		 
GST_RTSP_ADDRESS_POOL_ERANGE

+
the address is not in the pool

+		 
GST_RTSP_ADDRESS_POOL_ELAST

+
last error

+		 

+

+

+

+

+
See Also

+
GstRTSPStream, GstRTSPStreamTransport

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPContext.html b/docs/libs/html/gst-rtsp-server-GstRTSPContext.html
new file mode 100644
index 0000000..00c9702
--- /dev/null
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPContext.html
@@ -0,0 +1,267 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPContext
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPContext

+
GstRTSPContext — A client request context

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





+GstRTSPContext *
+
+gst_rtsp_context_get_current ()
+

+void
+
+gst_rtsp_context_push_current ()
+

+void
+
+gst_rtsp_context_pop_current ()
+

+

+

+
Types and Values

+

++++
+
+
+
+
+




structGstRTSPContext

+

+

+
Description

+
Last reviewed on 2013-07-11 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_context_get_current ()

+
GstRTSPContext *
+gst_rtsp_context_get_current (void);

+
Get the current GstRTSPContext. This object is retrieved from the
+current thread that is handling the request for a client.

+

+
Returns

+
 a GstRTSPContext

+

+

+

+

+

+
gst_rtsp_context_push_current ()

+
void
+gst_rtsp_context_push_current (GstRTSPContext *ctx);

+
Pushes ctx
+ onto the context stack. The current
+context can then be received using gst_rtsp_context_get_current().

+

+
Parameters

+

+++++
+
+
+
+
+
+





ctx
a #GstRTSPContext
 

+

+

+

+

+
gst_rtsp_context_pop_current ()

+
void
+gst_rtsp_context_pop_current (GstRTSPContext *ctx);

+
Pops ctx
+ off the context stack (verifying that ctx
+
+is on the top of the stack).

+

+
Parameters

+

+++++
+
+
+
+
+
+





ctx
a GstRTSPContext
 

+

+

+

+

+
Types and Values

+

+
struct GstRTSPContext

+
struct GstRTSPContext {
+  GstRTSPServer       *server;
+  GstRTSPConnection   *conn;
+  GstRTSPClient       *client;
+  GstRTSPMessage      *request;
+  GstRTSPUrl          *uri;
+  GstRTSPMethod        method;
+  GstRTSPAuth         *auth;
+  GstRTSPToken        *token;
+  GstRTSPSession      *session;
+  GstRTSPSessionMedia *sessmedia;
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia        *media;
+  GstRTSPStream       *stream;
+  GstRTSPMessage      *response;
+};
+

+
Information passed around containing the context of a request.

+

+
Members

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





GstRTSPServer *server;
the server
 
GstRTSPConnection *conn;
the connection
 
GstRTSPClient *client;
the client
 
GstRTSPMessage *request;
the complete request
 
GstRTSPUrl *uri;
the complete url parsed from request
+
 
GstRTSPMethod method;
the parsed method of uri
+
 
GstRTSPAuth *auth;
the current auth object or NULL
 
GstRTSPToken *token;
authorisation token
 
GstRTSPSession *session;
the session, can be NULL
 
GstRTSPSessionMedia *sessmedia;
the session media for the url can be NULL
 
GstRTSPMediaFactory *factory;
the media factory for the url, can be NULL
 
GstRTSPMedia *media;
the media for the url can be NULL
 
GstRTSPStream *stream;
the stream for the url can be NULL
 
GstRTSPMessage *response;
the response
 

+

+

+

+

+
See Also

+
GstRTSPServer, GstRTSPClient

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPMediaFactoryURI.html b/docs/libs/html/gst-rtsp-server-GstRTSPMediaFactoryURI.html
new file mode 100644
index 0000000..ba6ee82
--- /dev/null
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPMediaFactoryURI.html
@@ -0,0 +1,196 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPMediaFactoryURI
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPMediaFactoryURI

+
GstRTSPMediaFactoryURI — A factory for URI sources

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





+GstRTSPMediaFactoryURI *
+
+gst_rtsp_media_factory_uri_new ()
+

+void
+
+gst_rtsp_media_factory_uri_set_uri ()
+

+gchar *
+
+gst_rtsp_media_factory_uri_get_uri ()
+

+

+

+
Types and Values

+

++++
+
+
+
+
+
+
+
+
+
+
+




structGstRTSPMediaFactoryURI
structGstRTSPMediaFactoryURIClass

+

+

+
Description

+
This specialized GstRTSPMediaFactory constructs media pipelines from a URI,
+given with gst_rtsp_media_factory_uri_set_uri().

+
It will automatically demux and payload the different streams found in the
+media at URL.

+
Last reviewed on 2013-07-11 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_media_factory_uri_new ()

+
GstRTSPMediaFactoryURI *
+gst_rtsp_media_factory_uri_new (void);

+
Create a new GstRTSPMediaFactoryURI instance.

+

+
Returns

+
 a new GstRTSPMediaFactoryURI object. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_media_factory_uri_set_uri ()

+
void
+gst_rtsp_media_factory_uri_set_uri (GstRTSPMediaFactoryURI *factory,
+                                    const gchar *uri);

+
Set the URI of the resource that will be streamed by this factory.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 
uri
the uri the stream
 

+

+

+

+

+
gst_rtsp_media_factory_uri_get_uri ()

+
gchar *
+gst_rtsp_media_factory_uri_get_uri (GstRTSPMediaFactoryURI *factory);

+
Get the URI that will provide media for this factory.

+

+
Parameters

+

+++++
+
+
+
+
+
+





factory
a GstRTSPMediaFactory
 

+

+

+
Returns

+
 the configured URI. g_free() after usage. 

+
[transfer full]

+

+

+

+

+
Types and Values

+

+
struct GstRTSPMediaFactoryURI

+
struct GstRTSPMediaFactoryURI {
+  GstRTSPMediaFactory   parent;
+};
+

+
A media factory that creates a pipeline to play and uri.

+

+

+

+
struct GstRTSPMediaFactoryURIClass

+
struct GstRTSPMediaFactoryURIClass {
+  GstRTSPMediaFactoryClass  parent_class;
+};
+

+
The GstRTSPMediaFactoryURI class structure.

+

+

+

+
See Also

+
GstRTSPMediaFactory, GstRTSPMedia

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPParams.html b/docs/libs/html/gst-rtsp-server-GstRTSPParams.html
new file mode 100644
index 0000000..2831087
--- /dev/null
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPParams.html
@@ -0,0 +1,149 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPParams
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPParams

+
GstRTSPParams — Param get and set implementation

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
+
+
+





+GstRTSPResult
+
+gst_rtsp_params_get ()
+

+GstRTSPResult
+
+gst_rtsp_params_set ()
+

+

+

+
Description

+
Last reviewed on 2013-07-11 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_params_get ()

+
GstRTSPResult
+gst_rtsp_params_get (GstRTSPClient *client,
+                     GstRTSPContext *ctx);

+
Get parameters (not implemented yet)

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





client
a GstRTSPClient
 
ctx
 a GstRTSPContext. 
[transfer none]

+

+

+
Returns

+
 a GstRTSPResult

+

+

+

+

+

+
gst_rtsp_params_set ()

+
GstRTSPResult
+gst_rtsp_params_set (GstRTSPClient *client,
+                     GstRTSPContext *ctx);

+
Set parameters (not implemented yet)

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





client
a GstRTSPClient
 
ctx
 a GstRTSPContext. 
[transfer none]

+

+

+
Returns

+
 a GstRTSPResult

+

+

+

+

+

+
Types and Values

+

+

+
See Also

+
GstRTSPClient

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPPermissions.html b/docs/libs/html/gst-rtsp-server-GstRTSPPermissions.html
new file mode 100644
index 0000000..d6cfc0f
--- /dev/null
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPPermissions.html
@@ -0,0 +1,428 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPPermissions
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPPermissions

+
GstRTSPPermissions — Roles and associated permissions

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





+GstRTSPPermissions *
+
+gst_rtsp_permissions_new ()
+

+GstRTSPPermissions *
+
+gst_rtsp_permissions_ref ()
+

+void
+
+gst_rtsp_permissions_unref ()
+

+void
+
+gst_rtsp_permissions_add_role ()
+

+void
+
+gst_rtsp_permissions_add_role_valist ()
+

+void
+
+gst_rtsp_permissions_remove_role ()
+
const GstStructure *
+
+gst_rtsp_permissions_get_role ()
+

+gboolean
+
+gst_rtsp_permissions_is_allowed ()
+

+

+

+
Types and Values

+

++++
+
+
+
+
+




structGstRTSPPermissions

+

+

+
Description

+
The GstRTSPPermissions object contains an array of roles and associated
+permissions. The roles are represented with a string and the permissions with
+a generic GstStructure.

+
The permissions are deliberately kept generic. The possible values of the
+roles and GstStructure keys and values are only determined by the GstRTSPAuth
+object that performs the checks on the permissions and the current
+GstRTSPToken.

+
As a convenience function, gst_rtsp_permissions_is_allowed() can be used to
+check if the permissions contains a role that contains the boolean value
+TRUE for the the given key.

+
Last reviewed on 2013-07-15 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_permissions_new ()

+
GstRTSPPermissions *
+gst_rtsp_permissions_new (void);

+
Create a new empty Authorization permissions.

+

+
Returns

+
 a new empty authorization permissions. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_permissions_ref ()

+
GstRTSPPermissions *
+gst_rtsp_permissions_ref (GstRTSPPermissions *permissions);

+
Increase the refcount of this permissions.

+

+
Parameters

+

+++++
+
+
+
+
+
+





permissions
The permissions to refcount
 

+

+

+
Returns

+
 permissions
+(for convenience when doing assignments). 

+
[transfer full]

+

+

+

+

+
gst_rtsp_permissions_unref ()

+
void
+gst_rtsp_permissions_unref (GstRTSPPermissions *permissions);

+
Decrease the refcount of an permissions, freeing it if the refcount reaches 0.

+

+
Parameters

+

+++++
+
+
+
+
+
+





permissions
 the permissions to refcount. 
[transfer full]

+

+

+

+

+
gst_rtsp_permissions_add_role ()

+
void
+gst_rtsp_permissions_add_role (GstRTSPPermissions *permissions,
+                               const gchar *role,
+                               const gchar *fieldname,
+                               ...);

+
Add a new role
+ to permissions
+ with the given variables. The fields
+are the same layout as gst_structure_new().

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





permissions
a GstRTSPPermissions
 
role
a role
 
fieldname
the first field name
 
...
additional arguments
 

+

+

+

+

+
gst_rtsp_permissions_add_role_valist ()

+
void
+gst_rtsp_permissions_add_role_valist (GstRTSPPermissions *permissions,
+                                      const gchar *role,
+                                      const gchar *fieldname,
+                                      va_list var_args);

+
Add a new role
+ to permissions
+ with the given variables. Structure fields
+are set according to the varargs in a manner similar to gst_structure_new().

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





permissions
a GstRTSPPermissions
 
role
a role
 
fieldname
the first field name
 
var_args
additional fields to add
 

+

+

+

+

+
gst_rtsp_permissions_remove_role ()

+
void
+gst_rtsp_permissions_remove_role (GstRTSPPermissions *permissions,
+                                  const gchar *role);

+
Remove all permissions for role
+ in permissions
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





permissions
a GstRTSPPermissions
 
role
a role
 

+

+

+

+

+
gst_rtsp_permissions_get_role ()

+
const GstStructure *
+gst_rtsp_permissions_get_role (GstRTSPPermissions *permissions,
+                               const gchar *role);

+
Get all permissions for role
+ in permissions
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





permissions
a GstRTSPPermissions
 
role
a role
 

+

+

+
Returns

+
 the structure with permissions for role
+. It
+remains valid for as long as permissions
+is valid. 

+
[transfer none]

+

+

+

+

+
gst_rtsp_permissions_is_allowed ()

+
gboolean
+gst_rtsp_permissions_is_allowed (GstRTSPPermissions *permissions,
+                                 const gchar *role,
+                                 const gchar *permission);

+
Check if role
+ in permissions
+ is given permission for permission
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





permissions
a GstRTSPPermissions
 
role
a role
 
permission
a permission
 

+

+

+
Returns

+
 TRUE if role
+is allowed permission
+.

+

+

+

+

+

+
Types and Values

+

+
struct GstRTSPPermissions

+
struct GstRTSPPermissions {
+  GstMiniObject mini_object;
+};
+

+
The opaque permissions structure. It is used to define the permissions
+of objects in different roles.

+

+

+

+
See Also

+
GstRTSPToken, GstRTSPAuth

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPSdp.html b/docs/libs/html/gst-rtsp-server-GstRTSPSdp.html
new file mode 100644
index 0000000..88f0d70
--- /dev/null
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPSdp.html
@@ -0,0 +1,135 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPSdp
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPSdp

+
GstRTSPSdp — Make SDP messages

+

+

+
Functions

+

++++
+
+
+
+
+





+gboolean
+
+gst_rtsp_sdp_from_media ()
+

+

+

+
Types and Values

+

++++
+
+
+
+
+




 GstSDPInfo

+

+

+
Description

+
Last reviewed on 2013-07-11 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_sdp_from_media ()

+
gboolean
+gst_rtsp_sdp_from_media (GstSDPMessage *sdp,
+                         GstSDPInfo *info,
+                         GstRTSPMedia *media);

+
Add media
+ specific info to sdp
+. info
+ is used to configure the connection
+information in the SDP.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





sdp
a GstSDPMessage
 
info
 a GstSDPInfo. 
[transfer none]
media
 a GstRTSPMedia. 
[transfer none]

+

+

+
Returns

+
 TRUE on success.

+

+

+

+

+

+
Types and Values

+

+
GstSDPInfo

+
typedef struct {
+  gboolean is_ipv6;
+  const gchar *server_ip;
+} GstSDPInfo;
+

+

+

+

+
See Also

+
GstRTSPMedia

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPSessionMedia.html b/docs/libs/html/gst-rtsp-server-GstRTSPSessionMedia.html
new file mode 100644
index 0000000..f36b686
--- /dev/null
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPSessionMedia.html
@@ -0,0 +1,589 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPSessionMedia
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPSessionMedia

+
GstRTSPSessionMedia — Media managed in a session

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





+GstRTSPSessionMedia *
+
+gst_rtsp_session_media_new ()
+

+gboolean
+
+gst_rtsp_session_media_matches ()
+

+GstRTSPMedia *
+
+gst_rtsp_session_media_get_media ()
+

+GstClockTime
+
+gst_rtsp_session_media_get_base_time ()
+

+gchar *
+
+gst_rtsp_session_media_get_rtpinfo ()
+

+gboolean
+
+gst_rtsp_session_media_set_state ()
+

+GstRTSPState
+
+gst_rtsp_session_media_get_rtsp_state ()
+

+void
+
+gst_rtsp_session_media_set_rtsp_state ()
+

+GstRTSPStreamTransport *
+
+gst_rtsp_session_media_get_transport ()
+

+GstRTSPStreamTransport *
+
+gst_rtsp_session_media_set_transport ()
+

+gboolean
+
+gst_rtsp_session_media_alloc_channels ()
+

+

+

+
Types and Values

+

++++
+
+
+
+
+
+
+
+
+
+
+




structGstRTSPSessionMedia
structGstRTSPSessionMediaClass

+

+

+
Description

+
The GstRTSPSessionMedia object manages a GstRTSPMedia with a given path.

+
With gst_rtsp_session_media_get_transport() and
+gst_rtsp_session_media_set_transport() the transports of a GstRTSPStream of
+the managed GstRTSPMedia can be retrieved and configured.

+
Use gst_rtsp_session_media_set_state() to control the media state and
+transports.

+
Last reviewed on 2013-07-16 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_session_media_new ()

+
GstRTSPSessionMedia *
+gst_rtsp_session_media_new (const gchar *path,
+                            GstRTSPMedia *media);

+
Create a new GstRTSPSessionMedia that manages the streams
+in media
+ for path
+. media
+ should be prepared.

+
Ownership is taken of media
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





path
the path
 
media
 the GstRTSPMedia. 
[transfer full]

+

+

+
Returns

+
 a new GstRTSPSessionMedia. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_session_media_matches ()

+
gboolean
+gst_rtsp_session_media_matches (GstRTSPSessionMedia *media,
+                                const gchar *path,
+                                gint *matched);

+
Check if the path of media
+ matches path
+. It path
+ matches, the amount of
+matched characters is returned in matched
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPSessionMedia
 
path
a path
 
matched
 the amount of matched characters of path
+. 
[out]

+

+

+
Returns

+
 TRUE when path
+matches the path of media
+.

+

+

+

+

+

+
gst_rtsp_session_media_get_media ()

+
GstRTSPMedia *
+gst_rtsp_session_media_get_media (GstRTSPSessionMedia *media);

+
Get the GstRTSPMedia that was used when constructing media
+

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPSessionMedia
 

+

+

+
Returns

+
 the GstRTSPMedia of media
+. Remains valid as long
+as media
+is valid. 

+
[transfer none]

+

+

+

+

+
gst_rtsp_session_media_get_base_time ()

+
GstClockTime
+gst_rtsp_session_media_get_base_time (GstRTSPSessionMedia *media);

+
Get the base_time of the GstRTSPMedia in media
+

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPSessionMedia
 

+

+

+
Returns

+
 the base_time of the media.

+

+

+

+

+

+
gst_rtsp_session_media_get_rtpinfo ()

+
gchar *
+gst_rtsp_session_media_get_rtpinfo (GstRTSPSessionMedia *media);

+
Retrieve the RTP-Info header string for all streams in media
+
+with configured transports.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPSessionMedia
 

+

+

+
Returns

+
 The RTP-Info as a string or
+NULL when no RTP-Info could be generated, g_free() after usage. 

+
[transfer full][nullable]

+

+

+

+

+
gst_rtsp_session_media_set_state ()

+
gboolean
+gst_rtsp_session_media_set_state (GstRTSPSessionMedia *media,
+                                  GstState state);

+
Tell the media object media
+ to change to state
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPSessionMedia
 
state
the new state
 

+

+

+
Returns

+
 TRUE on success.

+

+

+

+

+

+
gst_rtsp_session_media_get_rtsp_state ()

+
GstRTSPState
+gst_rtsp_session_media_get_rtsp_state (GstRTSPSessionMedia *media);

+
Get the current RTSP state of media
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





media
a GstRTSPSessionMedia
 

+

+

+
Returns

+
 the current RTSP state of media
+.

+

+

+

+

+

+
gst_rtsp_session_media_set_rtsp_state ()

+
void
+gst_rtsp_session_media_set_rtsp_state (GstRTSPSessionMedia *media,
+                                       GstRTSPState state);

+
Set the RTSP state of media
+ to state
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPSessionMedia
 
state
a GstRTSPState
 

+

+

+

+

+
gst_rtsp_session_media_get_transport ()

+
GstRTSPStreamTransport *
+gst_rtsp_session_media_get_transport (GstRTSPSessionMedia *media,
+                                      guint idx);

+
Get a previously created GstRTSPStreamTransport for the stream at idx
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPSessionMedia
 
idx
the stream index
 

+

+

+
Returns

+
 a GstRTSPStreamTransport that is valid until the
+session of media
+is unreffed. 

+
[transfer none]

+

+

+

+

+
gst_rtsp_session_media_set_transport ()

+
GstRTSPStreamTransport *
+gst_rtsp_session_media_set_transport (GstRTSPSessionMedia *media,
+                                      GstRTSPStream *stream,
+                                      GstRTSPTransport *tr);

+
Configure the transport for stream
+ to tr
+ in media
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPSessionMedia
 
stream
a GstRTSPStream
 
tr
 a GstRTSPTransport. 
[transfer full]

+

+

+
Returns

+
 the new or updated GstRTSPStreamTransport for stream
+. 

+
[transfer none]

+

+

+

+

+
gst_rtsp_session_media_alloc_channels ()

+
gboolean
+gst_rtsp_session_media_alloc_channels (GstRTSPSessionMedia *media,
+                                       GstRTSPRange *range);

+
Fill range
+ with the next available min and max channels for
+interleaved transport.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





media
a GstRTSPSessionMedia
 
range
 a GstRTSPRange. 
[out]

+

+

+
Returns

+
 TRUE on success.

+

+

+

+

+

+
Types and Values

+

+
struct GstRTSPSessionMedia

+
struct GstRTSPSessionMedia {
+  GObject  parent;
+};
+

+
State of a client session regarding a specific media identified by path.

+

+

+

+
struct GstRTSPSessionMediaClass

+
struct GstRTSPSessionMediaClass {
+  GObjectClass  parent_class;
+};
+

+

+

+

+
See Also

+
GstRTSPMedia, GstRTSPSession

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPStreamTransport.html b/docs/libs/html/gst-rtsp-server-GstRTSPStreamTransport.html
new file mode 100644
index 0000000..6ae49fe
--- /dev/null
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPStreamTransport.html
@@ -0,0 +1,851 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPStreamTransport
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPStreamTransport

+
GstRTSPStreamTransport — A media stream transport configuration

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
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+
+
+
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+
+
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+
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+
+
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+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





+GstRTSPStreamTransport *
+
+gst_rtsp_stream_transport_new ()
+

+GstRTSPStream *
+
+gst_rtsp_stream_transport_get_stream ()
+
const GstRTSPTransport *
+
+gst_rtsp_stream_transport_get_transport ()
+

+void
+
+gst_rtsp_stream_transport_set_transport ()
+
const GstRTSPUrl *
+
+gst_rtsp_stream_transport_get_url ()
+

+void
+
+gst_rtsp_stream_transport_set_url ()
+

+gchar *
+
+gst_rtsp_stream_transport_get_rtpinfo ()
+

+gboolean
+
+(*GstRTSPSendFunc) ()
+

+void
+
+gst_rtsp_stream_transport_set_callbacks ()
+

+void
+
+(*GstRTSPKeepAliveFunc) ()
+

+void
+
+gst_rtsp_stream_transport_set_keepalive ()
+

+void
+
+gst_rtsp_stream_transport_keep_alive ()
+

+gboolean
+
+gst_rtsp_stream_transport_set_active ()
+

+void
+
+gst_rtsp_stream_transport_set_timed_out ()
+

+gboolean
+
+gst_rtsp_stream_transport_is_timed_out ()
+

+gboolean
+
+gst_rtsp_stream_transport_send_rtcp ()
+

+gboolean
+
+gst_rtsp_stream_transport_send_rtp ()
+

+

+

+
Types and Values

+

++++
+
+
+
+
+
+
+
+
+
+
+




structGstRTSPStreamTransport
structGstRTSPStreamTransportClass

+

+

+
Description

+
The GstRTSPStreamTransport configures the transport used by a
+GstRTSPStream. It is usually manages by a GstRTSPSessionMedia object.

+
With gst_rtsp_stream_transport_set_callbacks(), callbacks can be configured
+to handle the RTP and RTCP packets from the stream, for example when they
+need to be sent over TCP.

+
With  gst_rtsp_stream_transport_set_active() the transports are added and
+removed from the stream.

+
A GstRTSPStream will call gst_rtsp_stream_transport_keep_alive() when RTCP
+is received from the client. It will also call
+gst_rtsp_stream_transport_set_timed_out() when a receiver has timed out.

+
Last reviewed on 2013-07-16 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_stream_transport_new ()

+
GstRTSPStreamTransport *
+gst_rtsp_stream_transport_new (GstRTSPStream *stream,
+                               GstRTSPTransport *tr);

+
Create a new GstRTSPStreamTransport that can be used to manage
+stream
+ with transport tr
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





stream
a GstRTSPStream
 
tr
 a GstRTSPTransport. 
[transfer full]

+

+

+
Returns

+
 a new GstRTSPStreamTransport. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_stream_transport_get_stream ()

+
GstRTSPStream *
+gst_rtsp_stream_transport_get_stream (GstRTSPStreamTransport *trans);

+
Get the GstRTSPStream used when constructing trans
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





trans
a GstRTSPStreamTransport
 

+

+

+
Returns

+
 the stream used when constructing trans
+. 

+
[transfer none]

+

+

+

+

+
gst_rtsp_stream_transport_get_transport ()

+
const GstRTSPTransport *
+gst_rtsp_stream_transport_get_transport
+                               (GstRTSPStreamTransport *trans);

+
Get the transport configured in trans
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





trans
a GstRTSPStreamTransport
 

+

+

+
Returns

+
 the transport configured in trans
+. It remains
+valid for as long as trans
+is valid. 

+
[transfer none]

+

+

+

+

+
gst_rtsp_stream_transport_set_transport ()

+
void
+gst_rtsp_stream_transport_set_transport
+                               (GstRTSPStreamTransport *trans,
+                                GstRTSPTransport *tr);

+
Set tr
+ as the client transport. This function takes ownership of the
+passed tr
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





trans
a GstRTSPStreamTransport
 
tr
 a client GstRTSPTransport. 
[transfer full]

+

+

+

+

+
gst_rtsp_stream_transport_get_url ()

+
const GstRTSPUrl *
+gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport *trans);

+
Get the url configured in trans
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





trans
a GstRTSPStreamTransport
 

+

+

+
Returns

+
 the url configured in trans
+. It remains
+valid for as long as trans
+is valid. 

+
[transfer none]

+

+

+

+

+
gst_rtsp_stream_transport_set_url ()

+
void
+gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport *trans,
+                                   const GstRTSPUrl *url);

+
Set url
+ as the client url.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





trans
a GstRTSPStreamTransport
 
url
 a client GstRTSPUrl. 
[transfer none]

+

+

+

+

+
gst_rtsp_stream_transport_get_rtpinfo ()

+
gchar *
+gst_rtsp_stream_transport_get_rtpinfo (GstRTSPStreamTransport *trans,
+                                       GstClockTime start_time);

+
Get the RTP-Info string for trans
+ and start_time
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





trans
a GstRTSPStreamTransport
 
start_time
a star time
 

+

+

+
Returns

+
 the RTPInfo string for trans
+and start_time
+or NULL when the RTP-Info could not be
+determined. g_free() after usage. 

+
[transfer full][nullable]

+

+

+

+

+
GstRTSPSendFunc ()

+
gboolean
+(*GstRTSPSendFunc) (GstBuffer *buffer,
+                    guint8 channel,
+                    gpointer user_data);

+
Function registered with gst_rtsp_stream_transport_set_callbacks() and
+called when buffer
+ must be sent on channel
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





buffer
a GstBuffer
 
channel
a channel
 
user_data
user data
 

+

+

+
Returns

+
 TRUE on success

+

+

+

+

+

+
gst_rtsp_stream_transport_set_callbacks ()

+
void
+gst_rtsp_stream_transport_set_callbacks
+                               (GstRTSPStreamTransport *trans,
+                                GstRTSPSendFunc send_rtp,
+                                GstRTSPSendFunc send_rtcp,
+                                gpointer user_data,
+                                GDestroyNotify notify);

+
Install callbacks that will be called when data for a stream should be sent
+to a client. This is usually used when sending RTP/RTCP over TCP.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





trans
a GstRTSPStreamTransport
 
send_rtp
 a callback called when RTP should be sent. 
[scope notified]
send_rtcp
 a callback called when RTCP should be sent. 
[scope notified]
user_data
 user data passed to callbacks. 
[closure]
notify
 called with the user_data when no longer needed. 
[allow-none]

+

+

+

+

+
GstRTSPKeepAliveFunc ()

+
void
+(*GstRTSPKeepAliveFunc) (gpointer user_data);

+
Function registered with gst_rtsp_stream_transport_set_keepalive() and called
+when the stream is active.

+

+
Parameters

+

+++++
+
+
+
+
+
+





user_data
user data
 

+

+

+

+

+
gst_rtsp_stream_transport_set_keepalive ()

+
void
+gst_rtsp_stream_transport_set_keepalive
+                               (GstRTSPStreamTransport *trans,
+                                GstRTSPKeepAliveFunc keep_alive,
+                                gpointer user_data,
+                                GDestroyNotify notify);

+
Install callbacks that will be called when RTCP packets are received from the
+receiver of trans
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





trans
a GstRTSPStreamTransport
 
keep_alive
 a callback called when the receiver is active. 
[scope notified]
user_data
 user data passed to callback. 
[closure]
notify
 called with the user_data when no longer needed. 
[allow-none]

+

+

+

+

+
gst_rtsp_stream_transport_keep_alive ()

+
void
+gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport *trans);

+
Signal the installed keep_alive callback for trans
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+





trans
a GstRTSPStreamTransport
 

+

+

+

+

+
gst_rtsp_stream_transport_set_active ()

+
gboolean
+gst_rtsp_stream_transport_set_active (GstRTSPStreamTransport *trans,
+                                      gboolean active);

+
Activate or deactivate datatransfer configured in trans
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





trans
a GstRTSPStreamTransport
 
active
new state of trans
+
 

+

+

+
Returns

+
 TRUE when the state was changed.

+

+

+

+

+

+
gst_rtsp_stream_transport_set_timed_out ()

+
void
+gst_rtsp_stream_transport_set_timed_out
+                               (GstRTSPStreamTransport *trans,
+                                gboolean timedout);

+
Set the timed out state of trans
+ to timedout
+

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





trans
a GstRTSPStreamTransport
 
timedout
timed out value
 

+

+

+

+

+
gst_rtsp_stream_transport_is_timed_out ()

+
gboolean
+gst_rtsp_stream_transport_is_timed_out
+                               (GstRTSPStreamTransport *trans);

+
Check if trans
+ is timed out.

+

+
Parameters

+

+++++
+
+
+
+
+
+





trans
a GstRTSPStreamTransport
 

+

+

+
Returns

+
 TRUE if trans
+timed out.

+

+

+

+

+

+
gst_rtsp_stream_transport_send_rtcp ()

+
gboolean
+gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport *trans,
+                                     GstBuffer *buffer);

+
Send buffer
+ to the installed RTCP callback for trans
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





trans
a GstRTSPStreamTransport
 
buffer
 a GstBuffer. 
[transfer none]

+

+

+
Returns

+
 TRUE on success

+

+

+

+

+

+
gst_rtsp_stream_transport_send_rtp ()

+
gboolean
+gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport *trans,
+                                    GstBuffer *buffer);

+
Send buffer
+ to the installed RTP callback for trans
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





trans
a GstRTSPStreamTransport
 
buffer
 a GstBuffer. 
[transfer none]

+

+

+
Returns

+
 TRUE on success

+

+

+

+

+

+
Types and Values

+

+
struct GstRTSPStreamTransport

+
struct GstRTSPStreamTransport {
+  GObject              parent;
+};
+

+
A Transport description for a stream

+

+
Members

+

+++++
+
+
+
+
+
+





GObject parent;
parent instance
 

+

+

+

+

+
struct GstRTSPStreamTransportClass

+
struct GstRTSPStreamTransportClass {
+  GObjectClass parent_class;
+};
+

+

+

+

+
See Also

+
GstRTSPStream, GstRTSPSessionMedia

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPThreadPool.html b/docs/libs/html/gst-rtsp-server-GstRTSPThreadPool.html
new file mode 100644
index 0000000..f793fe9
--- /dev/null
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPThreadPool.html
@@ -0,0 +1,592 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPThreadPool
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPThreadPool

+
GstRTSPThreadPool — A pool of threads

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





+GstRTSPThread *
+
+gst_rtsp_thread_new ()
+

+GstRTSPThread *
+
+gst_rtsp_thread_ref ()
+

+void
+
+gst_rtsp_thread_unref ()
+

+gboolean
+
+gst_rtsp_thread_reuse ()
+

+void
+
+gst_rtsp_thread_stop ()
+

+GstRTSPThreadPool *
+
+gst_rtsp_thread_pool_new ()
+

+gint
+
+gst_rtsp_thread_pool_get_max_threads ()
+

+void
+
+gst_rtsp_thread_pool_set_max_threads ()
+

+GstRTSPThread *
+
+gst_rtsp_thread_pool_get_thread ()
+

+void
+
+gst_rtsp_thread_pool_cleanup ()
+

+

+

+
Types and Values

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+




enumGstRTSPThreadType
structGstRTSPThread
structGstRTSPThreadPool
structGstRTSPThreadPoolClass

+

+

+
Description

+
A GstRTSPThreadPool manages reusable threads for various server tasks.
+Currently the defined thread types can be found in GstRTSPThreadType.

+
Threads of type GST_RTSP_THREAD_TYPE_CLIENT are used to handle requests from
+a connected client. With gst_rtsp_thread_pool_get_max_threads() a maximum
+number of threads can be set after which the pool will start to reuse the
+same thread for multiple clients.

+
Threads of type GST_RTSP_THREAD_TYPE_MEDIA will be used to perform the state
+changes of the media pipelines and handle its bus messages.

+
gst_rtsp_thread_pool_get_thread() can be used to create a GstRTSPThread
+object of the right type. The thread object contains a mainloop and context
+that run in a seperate thread and can be used to attached sources to.

+
gst_rtsp_thread_reuse() can be used to reuse a thread for multiple purposes.
+If all gst_rtsp_thread_reuse() calls are matched with a
+gst_rtsp_thread_stop() call, the mainloop will be quit and the thread will
+stop.

+
To configure the threads, a subclass of this object should be made and the
+virtual methods should be overriden to implement the desired functionality.

+
Last reviewed on 2013-07-11 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_thread_new ()

+
GstRTSPThread *
+gst_rtsp_thread_new (GstRTSPThreadType type);

+
Create a new thread object that can run a mainloop.

+

+
Parameters

+

+++++
+
+
+
+
+
+





type
the thread type
 

+

+

+
Returns

+
 a GstRTSPThread. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_thread_ref ()

+
GstRTSPThread *
+gst_rtsp_thread_ref (GstRTSPThread *thread);

+
Increase the refcount of this thread.

+

+
Parameters

+

+++++
+
+
+
+
+
+





thread
The thread to refcount
 

+

+

+
Returns

+
 thread
+(for convenience when doing assignments). 

+
[transfer full]

+

+

+

+

+
gst_rtsp_thread_unref ()

+
void
+gst_rtsp_thread_unref (GstRTSPPermissions *thread);

+
Decrease the refcount of an thread, freeing it if the refcount reaches 0.

+

+
Parameters

+

+++++
+
+
+
+
+
+





thread
 the thread to refcount. 
[transfer full]

+

+

+

+

+
gst_rtsp_thread_reuse ()

+
gboolean
+gst_rtsp_thread_reuse (GstRTSPThread *thread);

+
Reuse the mainloop of thread
+

+

+
Parameters

+

+++++
+
+
+
+
+
+





thread
 a GstRTSPThread. 
[transfer none]

+

+

+
Returns

+
 TRUE if the mainloop could be reused

+

+

+

+

+

+
gst_rtsp_thread_stop ()

+
void
+gst_rtsp_thread_stop (GstRTSPThread *thread);

+
Stop and unref thread
+. When no threads are using the mainloop, the thread
+will be stopped and the final ref to thread
+ will be released.

+

+
Parameters

+

+++++
+
+
+
+
+
+





thread
 a GstRTSPThread. 
[transfer full]

+

+

+

+

+
gst_rtsp_thread_pool_new ()

+
GstRTSPThreadPool *
+gst_rtsp_thread_pool_new (void);

+
Create a new GstRTSPThreadPool instance.

+

+
Returns

+
 a new GstRTSPThreadPool. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_thread_pool_get_max_threads ()

+
gint
+gst_rtsp_thread_pool_get_max_threads (GstRTSPThreadPool *pool);

+
Get the maximum number of threads used for client connections.
+See gst_rtsp_thread_pool_set_max_threads().

+

+
Parameters

+

+++++
+
+
+
+
+
+





pool
a GstRTSPThreadPool
 

+

+

+
Returns

+
 the maximum number of threads.

+

+

+

+

+

+
gst_rtsp_thread_pool_set_max_threads ()

+
void
+gst_rtsp_thread_pool_set_max_threads (GstRTSPThreadPool *pool,
+                                      gint max_threads);

+
Set the maximum threads used by the pool to handle client requests.
+A value of 0 will use the pool mainloop, a value of -1 will use an
+unlimited number of threads.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





pool
a GstRTSPThreadPool
 
max_threads
maximum threads
 

+

+

+

+

+
gst_rtsp_thread_pool_get_thread ()

+
GstRTSPThread *
+gst_rtsp_thread_pool_get_thread (GstRTSPThreadPool *pool,
+                                 GstRTSPThreadType type,
+                                 GstRTSPContext *ctx);

+
Get a new GstRTSPThread for type
+ and ctx
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





pool
a GstRTSPThreadPool
 
type
the GstRTSPThreadType
 
ctx
 a GstRTSPContext. 
[transfer none]

+

+

+
Returns

+
 a new GstRTSPThread, gst_rtsp_thread_stop() after usage. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_thread_pool_cleanup ()

+
void
+gst_rtsp_thread_pool_cleanup (void);

+
Wait for all tasks to be stopped and free all allocated resources. This is
+mainly used in test suites to ensure proper cleanup of internal data
+structures.

+

+

+

+
Types and Values

+

+
enum GstRTSPThreadType

+
Different thread types

+

+
Members

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





GST_RTSP_THREAD_TYPE_CLIENT

+
a thread to handle the client communication

+		 
GST_RTSP_THREAD_TYPE_MEDIA

+
a thread to handle media 

+		 

+

+

+

+

+
struct GstRTSPThread

+
struct GstRTSPThread {
+  GstMiniObject mini_object;
+
+  GstRTSPThreadType type;
+  GMainContext *context;
+  GMainLoop *loop;
+};
+

+
Structure holding info about a mainloop running in a thread

+

+
Members

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





GstMiniObject mini_object;
parent GstMiniObject
 
GstRTSPThreadType type;
the thread type
 
GMainContext *context;
a GMainContext
 
GMainLoop *loop;
a GMainLoop
 

+

+

+

+

+
struct GstRTSPThreadPool

+
struct GstRTSPThreadPool {
+  GObject       parent;
+};
+

+
The thread pool structure.

+

+

+

+
struct GstRTSPThreadPoolClass

+
struct GstRTSPThreadPoolClass {
+  GObjectClass  parent_class;
+
+  GThreadPool *pool;
+
+  GstRTSPThread * (*get_thread)        (GstRTSPThreadPool *pool,
+                                        GstRTSPThreadType type,
+                                        GstRTSPContext *ctx);
+  void            (*configure_thread)  (GstRTSPThreadPool *pool,
+                                        GstRTSPThread * thread,
+                                        GstRTSPContext *ctx);
+
+  void            (*thread_enter)      (GstRTSPThreadPool *pool,
+                                        GstRTSPThread *thread);
+  void            (*thread_leave)      (GstRTSPThreadPool *pool,
+                                        GstRTSPThread *thread);
+};
+

+
Class for managing threads.

+

+
Members

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+





GObjectClass parent_class;
  
GThreadPool *pool;
a GThreadPool used internally
 
get_thread ()
this function should make or reuse an existing thread that runs
+a mainloop.
 
configure_thread ()
configure a thread object. this vmethod is called when
+a new thread has been created and should be configured.
 
thread_enter ()
called from the thread when it is entered
 
thread_leave ()
called from the thread when it is left
 

+

+

+

+

+
See Also

+
GstRTSPMedia, GstRTSPClient

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPToken.html b/docs/libs/html/gst-rtsp-server-GstRTSPToken.html
new file mode 100644
index 0000000..9bd5be3
--- /dev/null
+++ b/docs/libs/html/gst-rtsp-server-GstRTSPToken.html
@@ -0,0 +1,443 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GstRTSPToken
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+

+
+
+

+
GstRTSPToken

+
GstRTSPToken — Roles and permissions for a client

+

+

+
Functions

+

++++
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
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+
+
+
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+
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+
+
+
+
+
+
+
+





+GstRTSPToken *
+
+gst_rtsp_token_new_empty ()
+

+GstRTSPToken *
+
+gst_rtsp_token_new ()
+

+GstRTSPToken *
+
+gst_rtsp_token_new_valist ()
+

+GstRTSPToken *
+
+gst_rtsp_token_ref ()
+

+void
+
+gst_rtsp_token_unref ()
+
const GstStructure *
+
+gst_rtsp_token_get_structure ()
+

+GstStructure *
+
+gst_rtsp_token_writable_structure ()
+
const gchar *
+
+gst_rtsp_token_get_string ()
+

+gboolean
+
+gst_rtsp_token_is_allowed ()
+

+

+

+
Types and Values

+

++++
+
+
+
+
+




structGstRTSPToken

+

+

+
Description

+
A GstRTSPToken contains the permissions and roles of the user
+performing the current request. A token is usually created when a user is
+authenticated by the GstRTSPAuth object and is then placed as the current
+token for the current request.

+
GstRTSPAuth can use the token and its contents to check authorization for
+various operations by comparing the token to the GstRTSPPermissions of the
+object.

+
The accepted values of the token are entirely defined by the GstRTSPAuth
+object that implements the security policy.

+
Last reviewed on 2013-07-15 (1.0.0)

+

+

+
Functions

+

+
gst_rtsp_token_new_empty ()

+
GstRTSPToken *
+gst_rtsp_token_new_empty (void);

+
Create a new empty Authorization token.

+

+
Returns

+
 a new empty authorization token. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_token_new ()

+
GstRTSPToken *
+gst_rtsp_token_new (const gchar *firstfield,
+                    ...);

+
Create a new Authorization token with the given fieldnames and values.
+Arguments are given similar to gst_structure_new().

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





firstfield
the first fieldname
 
...
additional arguments
 

+

+

+
Returns

+
 a new authorization token. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_token_new_valist ()

+
GstRTSPToken *
+gst_rtsp_token_new_valist (const gchar *firstfield,
+                           va_list var_args);

+
Create a new Authorization token with the given fieldnames and values.
+Arguments are given similar to gst_structure_new_valist().

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





firstfield
the first fieldname
 
var_args
additional arguments
 

+

+

+
Returns

+
 a new authorization token. 

+
[transfer full]

+

+

+

+

+
gst_rtsp_token_ref ()

+
GstRTSPToken *
+gst_rtsp_token_ref (GstRTSPToken *token);

+
Increase the refcount of this token.

+

+
Parameters

+

+++++
+
+
+
+
+
+





token
The token to refcount
 

+

+

+
Returns

+
 token
+(for convenience when doing assignments). 

+
[transfer full]

+

+

+

+

+
gst_rtsp_token_unref ()

+
void
+gst_rtsp_token_unref (GstRTSPToken *token);

+
Decrease the refcount of an token, freeing it if the refcount reaches 0.

+

+
Parameters

+

+++++
+
+
+
+
+
+





token
 the token to refcount. 
[transfer full]

+

+

+

+

+
gst_rtsp_token_get_structure ()

+
const GstStructure *
+gst_rtsp_token_get_structure (GstRTSPToken *token);

+
Access the structure of the token.

+

+
Parameters

+

+++++
+
+
+
+
+
+





token
The GstRTSPToken.
 

+

+

+
Returns

+
 The structure of the token. The structure is still
+owned by the token, which means that you should not free it and that the
+pointer becomes invalid when you free the token.

+
MT safe. 

+
[transfer none]

+

+

+

+

+
gst_rtsp_token_writable_structure ()

+
GstStructure *
+gst_rtsp_token_writable_structure (GstRTSPToken *token);

+
Get a writable version of the structure.

+

+
Parameters

+

+++++
+
+
+
+
+
+





token
The GstRTSPToken.
 

+

+

+
Returns

+
 The structure of the token. The structure is still
+owned by the token, which means that you should not free it and that the
+pointer becomes invalid when you free the token. This function checks if
+token
+is writable and will never return NULL.

+
MT safe. 

+
[transfer none]

+

+

+

+

+
gst_rtsp_token_get_string ()

+
const gchar *
+gst_rtsp_token_get_string (GstRTSPToken *token,
+                           const gchar *field);

+
Get the string value of field
+ in token
+.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





token
a GstRTSPToken
 
field
a field name
 

+

+

+
Returns

+
 the string value of field
+in
+token
+or NULL when field
+is not defined in token
+. The string
+becomes invalid when you free token
+. 

+
[transfer none][nullable]

+

+

+

+

+
gst_rtsp_token_is_allowed ()

+
gboolean
+gst_rtsp_token_is_allowed (GstRTSPToken *token,
+                           const gchar *field);

+
Check if token
+ has a boolean field
+ and if it is set to TRUE.

+

+
Parameters

+

+++++
+
+
+
+
+
+
+
+
+
+
+
+
+





token
a GstRTSPToken
 
field
a field name
 

+

+

+
Returns

+
 TRUE if token
+has a boolean field named field
+set to TRUE.

+

+

+

+

+

+
Types and Values

+

+
struct GstRTSPToken

+
struct GstRTSPToken {
+  GstMiniObject mini_object;
+};
+

+
An opaque object used for checking authorisations.
+It is generated after successful authentication.

+

+

+

+
See Also

+
GstRTSPClient, GstRTSPPermissions, GstRTSPAuth

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
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diff --git a/docs/libs/html/index.html b/docs/libs/html/index.html
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+++ b/docs/libs/html/index.html
@@ -0,0 +1,93 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: GStreamer RTSP Server Reference Manual
+
+
+
+
+
+
+
+

+

+

+

+

+      for GStreamer RTSP Server 1.4.5
+    

+

+

+

+

+

+

+

+GstRTSPServer — The main server object
+

+

+GstRTSPClient — A client connection state
+

+

+GstRTSPContext — A client request context
+

+

+GstRTSPMountPoints — Map a path to media
+

+

+GstRTSPMediaFactory — A factory for media pipelines
+

+

+GstRTSPMediaFactoryURI — A factory for URI sources
+

+

+GstRTSPMedia — The media pipeline
+

+

+GstRTSPStream — A media stream
+

+

+GstRTSPSessionPool — An object for managing sessions
+

+

+GstRTSPSession — An object to manage media
+

+

+GstRTSPSessionMedia — Media managed in a session
+

+

+GstRTSPStreamTransport — A media stream transport configuration
+

+

+GstRTSPSdp — Make SDP messages
+

+

+GstRTSPAddressPool — A pool of network addresses
+

+

+GstRTSPThreadPool — A pool of threads
+

+

+GstRTSPAuth — Authentication and authorization
+

+

+GstRTSPToken — Roles and permissions for a client
+

+

+GstRTSPPermissions — Roles and associated permissions
+

+

+GstRTSPParams — Param get and set implementation
+

+

+
Object Hierarchy

+
API Index

+
Annotation Glossary

+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/index.sgml b/docs/libs/html/index.sgml
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diff --git a/docs/libs/html/left-insensitive.png b/docs/libs/html/left-insensitive.png
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diff --git a/docs/libs/html/rtsp-server-hierarchy.html b/docs/libs/html/rtsp-server-hierarchy.html
new file mode 100644
index 0000000..4d752f2
--- /dev/null
+++ b/docs/libs/html/rtsp-server-hierarchy.html
@@ -0,0 +1,42 @@
+
+
+
+
+GStreamer RTSP Server Reference Manual: Object Hierarchy
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+

+

+Object Hierarchy

+
+    GObject
+    ├── GstRTSPAuth
+    ├── GstRTSPMountPoints
+    ├── GstRTSPMediaFactory
+    ├── GstRTSPMedia
+    ├── GstRTSPServer
+    ├── GstRTSPSessionPool
+    ├── GstRTSPSession
+    ├── GstRTSPClient
+    ╰── GstRTSPStream
+

+

+

+

+          Generated by GTK-Doc V1.21

+
+
\ No newline at end of file
diff --git a/docs/libs/html/style.css b/docs/libs/html/style.css
new file mode 100644
index 0000000..c141ddd
--- /dev/null
+++ b/docs/libs/html/style.css
@@ -0,0 +1,476 @@
+body
+{
+  font-family: cantarell, sans-serif;
+}
+.synopsis, .classsynopsis
+{
+  /* tango:aluminium 1/2 */
+  background: #eeeeec;
+  background: rgba(238, 238, 236, 0.5);
+  border: solid 1px rgb(238, 238, 236);
+  padding: 0.5em;
+}
+.programlisting
+{
+  /* tango:sky blue 0/1 */
+  /* fallback for no rgba support */
+  background: #e6f3ff;
+  border: solid 1px #729fcf;
+  background: rgba(114, 159, 207, 0.1);
+  border: solid 1px rgba(114, 159, 207, 0.2);
+  padding: 0.5em;
+}
+.variablelist
+{
+  padding: 4px;
+  margin-left: 3em;
+}
+.variablelist td:first-child
+{
+  vertical-align: top;
+}
+
+div.gallery-float
+{
+  float: left;
+  padding: 10px;
+}
+div.gallery-float img
+{
+  border-style: none;
+}
+div.gallery-spacer
+{
+  clear: both;
+}
+
+a, a:visited
+{
+  text-decoration: none;
+  /* tango:sky blue 2 */
+  color: #3465a4;
+}
+a:hover
+{
+  text-decoration: underline;
+  /* tango:sky blue 1 */
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+
+div.informaltable table
+{
+  border-collapse: separate;
+  border-spacing: 1em 0.5em;
+  border: none;
+}
+
+div.informaltable table td, div.informaltable table th
+{
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+}
+
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+{
+  text-align: right;
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+
+/* dim non-primary columns */
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+{
+  color: #888a85;
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+
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+.signal_flags a:hover
+{
+ color: #729fcf;
+}
+
+td p
+{
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+
+div.table table
+{
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+  border: solid 1px #babdb6;
+}
+
+div.table table td, div.table table th
+{
+  /* tango:aluminium 3 */
+  border: solid 1px #babdb6;
+  padding: 3px;
+  vertical-align: top;
+}
+
+div.table table th
+{
+  /* tango:aluminium 2 */
+  background-color: #d3d7cf;
+}
+
+h4
+{
+  color: #555753;
+}
+
+hr
+{
+  /* tango:aluminium 1 */
+  color: #d3d7cf;
+  background: #d3d7cf;
+  border: none 0px;
+  height: 1px;
+  clear: both;
+  margin: 2.0em 0em 2.0em 0em;
+}
+
+dl.toc dt
+{
+  padding-bottom: 0.25em;
+}
+
+dl.toc > dd > dl > dt
+{
+  padding-top: 0.25em;
+  padding-bottom: 0.25em;
+}
+
+dl.toc > dt
+{
+  padding-top: 1em;
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+  font-weight: bold;
+}
+
+.parameter
+{
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+
+.footer
+{
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+  /* tango:aluminium 3 */
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+  text-align: center;
+  font-size: 80%;
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+
+.informalfigure,
+.figure
+{
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+}
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+.informalexample,
+.example
+{
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+  margin-bottom: 1em;
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+
+.warning
+{
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+}
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+{
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+  /* tango:sky blue 0.5 */
+  background: #a6c5e3;
+  background: rgba(114, 159, 207, 0.2);
+  /* tango:aluminium 6 */
+  color: #2e3436;
+}
+.listing_code {
+  /* tango:sky blue 0 */
+  background: #e6f3ff;
+  background: rgba(114, 159, 207, 0.1);
+}
+.listing_code .programlisting {
+  /* override from previous */
+  border: none 0px;
+  padding: 0px;
+  background: none;
+}
+.listing_lines pre, .listing_code pre {
+  margin: 0px;
+}
+
+@media screen {
+  sup a.footnote
+  {
+    position: relative;
+    top: 0em ! important;
+  }
+  /* this is needed so that the local anchors are displayed below the naviagtion */
+  div.footnote a[name], div.refnamediv a[name], div.refsect1 a[name], div.refsect2 a[name], div.index a[name], div.glossary a[name], div.sect1 a[name]
+  {
+    display: inline-block;
+    position: relative;
+    top:-5em;
+  }
+  /* this seems to be a bug in the xsl style sheets when generating indexes */
+  div.index div.index
+  {
+    top: 0em;
+  }
+  /* make space for the fixed navigation bar and add space at the bottom so that
+   * link targets appear somewhat close to top
+   */
+  body
+  {
+    padding-top: 2.5em;
+    padding-bottom: 500px;
+    max-width: 60em;
+  }
+  p
+  {
+    max-width: 60em;
+  }
+  /* style and size the navigation bar */
+  table.navigation#top
+  {
+    position: fixed;
+    background: #e2e2e2;
+    border-bottom: solid 1px #babdb6;
+    border-spacing: 5px;
+    margin-top: 0;
+    margin-bottom: 0;
+    top: 0;
+    left: 0;
+    z-index: 10;
+  }
+  table.navigation#top td
+  {
+    padding-left: 6px;
+    padding-right: 6px;
+  }
+  .navigation a, .navigation a:visited
+  {
+    /* tango:sky blue 3 */
+    color: #204a87;
+  }
+  .navigation a:hover
+  {
+    /* tango:sky blue 2 */
+    color: #3465a4;
+  }
+  td.shortcuts
+  {
+    /* tango:sky blue 2 */
+    color: #3465a4;
+    font-size: 80%;
+    white-space: nowrap;
+  }
+  td.shortcuts .dim
+  {
+    color: #babdb6;
+  }
+  .navigation .title
+  {
+    font-size: 80%;
+    max-width: none;
+    margin: 0px;
+    font-weight: normal;
+  }
+}
+@media screen and (min-width: 60em) {
+  /* screen larger than 60em */
+  body { margin: auto; }
+}
+@media screen and (max-width: 60em) {
+  /* screen less than 60em */
+  #nav_hierarchy { display: none; }
+  #nav_interfaces { display: none; }
+  #nav_prerequisites { display: none; }
+  #nav_derived_interfaces { display: none; }
+  #nav_implementations { display: none; }
+  #nav_child_properties { display: none; }
+  #nav_style_properties { display: none; }
+  #nav_index { display: none; }
+  #nav_glossary { display: none; }
+  .gallery_image { display: none; }
+  .property_flags { display: none; }
+  .signal_flags { display: none; }
+  .parameter_annotations { display: none; }
+  .enum_member_annotations { display: none; }
+  .struct_member_annotations { display: none; }
+  .union_member_annotations { display: none; }
+  /* now that a column is hidden, optimize space */
+  col.parameters_name { width: auto; }
+  col.parameters_description { width: auto; }
+  col.struct_members_name { width: auto; }
+  col.struct_members_description { width: auto; }
+  col.enum_members_name { width: auto; }
+  col.enum_members_description { width: auto; }
+  col.union_members_name { width: auto; }
+  col.union_members_description { width: auto; }
+  .listing_lines { display: none; }
+}
+@media print {
+  table.navigation {
+    visibility: collapse;
+    display: none;
+  }
+  div.titlepage table.navigation {
+    visibility: visible;
+    display: table;
+    background: #e2e2e2;
+    border: solid 1px #babdb6;
+    margin-top: 0;
+    margin-bottom: 0;
+    top: 0;
+    left: 0;
+    height: 3em;
+  }
+}
+
diff --git a/docs/libs/html/up-insensitive.png b/docs/libs/html/up-insensitive.png
new file mode 100644
index 0000000..f404986
Binary files /dev/null and b/docs/libs/html/up-insensitive.png differ
diff --git a/docs/libs/html/up.png b/docs/libs/html/up.png
new file mode 100644
index 0000000..80b4b37
Binary files /dev/null and b/docs/libs/html/up.png differ
diff --git a/docs/version.entities.in b/docs/version.entities.in
new file mode 100644
index 0000000..286989f
--- /dev/null
+++ b/docs/version.entities.in
@@ -0,0 +1,2 @@
+
+
diff --git a/examples/Makefile.am b/examples/Makefile.am
new file mode 100644
index 0000000..eb5ddb8
--- /dev/null
+++ b/examples/Makefile.am
@@ -0,0 +1,14 @@
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new file mode 100644
index 0000000..82559a7
--- /dev/null
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+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
diff --git a/examples/test-appsrc.c b/examples/test-appsrc.c
new file mode 100644
index 0000000..9d271ea
--- /dev/null
+++ b/examples/test-appsrc.c
@@ -0,0 +1,137 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+typedef struct
+{
+  gboolean white;
+  GstClockTime timestamp;
+} MyContext;
+
+/* called when we need to give data to appsrc */
+static void
+need_data (GstElement * appsrc, guint unused, MyContext * ctx)
+{
+  GstBuffer *buffer;
+  guint size;
+  GstFlowReturn ret;
+
+  size = 385 * 288 * 2;
+
+  buffer = gst_buffer_new_allocate (NULL, size, NULL);
+
+  /* this makes the image black/white */
+  gst_buffer_memset (buffer, 0, ctx->white ? 0xff : 0x0, size);
+
+  ctx->white = !ctx->white;
+
+  /* increment the timestamp every 1/2 second */
+  GST_BUFFER_PTS (buffer) = ctx->timestamp;
+  GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale_int (1, GST_SECOND, 2);
+  ctx->timestamp += GST_BUFFER_DURATION (buffer);
+
+  g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
+}
+
+/* called when a new media pipeline is constructed. We can query the
+ * pipeline and configure our appsrc */
+static void
+media_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media,
+    gpointer user_data)
+{
+  GstElement *element, *appsrc;
+  MyContext *ctx;
+
+  /* get the element used for providing the streams of the media */
+  element = gst_rtsp_media_get_element (media);
+
+  /* get our appsrc, we named it 'mysrc' with the name property */
+  appsrc = gst_bin_get_by_name_recurse_up (GST_BIN (element), "mysrc");
+
+  /* this instructs appsrc that we will be dealing with timed buffer */
+  gst_util_set_object_arg (G_OBJECT (appsrc), "format", "time");
+  /* configure the caps of the video */
+  g_object_set (G_OBJECT (appsrc), "caps",
+      gst_caps_new_simple ("video/x-raw",
+          "format", G_TYPE_STRING, "RGB16",
+          "width", G_TYPE_INT, 384,
+          "height", G_TYPE_INT, 288,
+          "framerate", GST_TYPE_FRACTION, 0, 1, NULL), NULL);
+
+  ctx = g_new0 (MyContext, 1);
+  ctx->white = FALSE;
+  ctx->timestamp = 0;
+  /* make sure ther datais freed when the media is gone */
+  g_object_set_data_full (G_OBJECT (media), "my-extra-data", ctx,
+      (GDestroyNotify) g_free);
+
+  /* install the callback that will be called when a buffer is needed */
+  g_signal_connect (appsrc, "need-data", (GCallback) need_data, ctx);
+}
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactory *factory;
+
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* get the mount points for this server, every server has a default object
+   * that be used to map uri mount points to media factories */
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  /* make a media factory for a test stream. The default media factory can use
+   * gst-launch syntax to create pipelines.
+   * any launch line works as long as it contains elements named pay%d. Each
+   * element with pay%d names will be a stream */
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory,
+      "( appsrc name=mysrc ! videoconvert ! x264enc ! rtph264pay name=pay0 pt=96 )");
+
+  /* notify when our media is ready, This is called whenever someone asks for
+   * the media and a new pipeline with our appsrc is created */
+  g_signal_connect (factory, "media-configure", (GCallback) media_configure,
+      NULL);
+
+  /* attach the test factory to the /test url */
+  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+  /* don't need the ref to the mounts anymore */
+  g_object_unref (mounts);
+
+  /* attach the server to the default maincontext */
+  gst_rtsp_server_attach (server, NULL);
+
+  /* start serving */
+  g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+  g_main_loop_run (loop);
+
+  return 0;
+}
diff --git a/examples/test-auth.c b/examples/test-auth.c
new file mode 100644
index 0000000..eff3c74
--- /dev/null
+++ b/examples/test-auth.c
@@ -0,0 +1,186 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+static gboolean
+remove_func (GstRTSPSessionPool * pool, GstRTSPSession * session,
+    GstRTSPServer * server)
+{
+  return GST_RTSP_FILTER_REMOVE;
+}
+
+static gboolean
+remove_sessions (GstRTSPServer * server)
+{
+  GstRTSPSessionPool *pool;
+
+  g_print ("removing all sessions\n");
+  pool = gst_rtsp_server_get_session_pool (server);
+  gst_rtsp_session_pool_filter (pool,
+      (GstRTSPSessionPoolFilterFunc) remove_func, server);
+  g_object_unref (pool);
+
+  return FALSE;
+}
+
+static gboolean
+timeout (GstRTSPServer * server)
+{
+  GstRTSPSessionPool *pool;
+
+  pool = gst_rtsp_server_get_session_pool (server);
+  gst_rtsp_session_pool_cleanup (pool);
+  g_object_unref (pool);
+
+  return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactory *factory;
+  GstRTSPAuth *auth;
+  GstRTSPToken *token;
+  gchar *basic;
+
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* get the mounts for this server, every server has a default mapper object
+   * that be used to map uri mount points to media factories */
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+
+  /* make a media factory for a test stream. The default media factory can use
+   * gst-launch syntax to create pipelines. 
+   * any launch line works as long as it contains elements named pay%d. Each
+   * element with pay%d names will be a stream */
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory, "( "
+      "videotestsrc ! video/x-raw,width=352,height=288,framerate=15/1 ! "
+      "x264enc ! rtph264pay name=pay0 pt=96 "
+      "audiotestsrc ! audio/x-raw,rate=8000 ! "
+      "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+  /* attach the test factory to the /test url */
+  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+  /* allow user and admin to access this resource */
+  gst_rtsp_media_factory_add_role (factory, "user",
+      GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+      GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+  gst_rtsp_media_factory_add_role (factory, "admin",
+      GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+      GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+  /* admin2 can look at the media but not construct so he gets a
+   * 401 Unauthorized */
+  gst_rtsp_media_factory_add_role (factory, "admin2",
+      GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+      GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, FALSE, NULL);
+
+  /* make another factory */
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory, "( "
+      "videotestsrc ! video/x-raw,width=352,height=288,framerate=30/1 ! "
+      "x264enc ! rtph264pay name=pay0 pt=96 )");
+  /* attach the test factory to the /test url */
+  gst_rtsp_mount_points_add_factory (mounts, "/test2", factory);
+
+  /* allow admin2 to access this resource */
+  /* user and admin have no permissions so they can't even see the
+   * media and get a 404 Not Found */
+  gst_rtsp_media_factory_add_role (factory, "admin2",
+      GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+      GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+
+  /* don't need the ref to the mapper anymore */
+  g_object_unref (mounts);
+
+  /* make a new authentication manager */
+  auth = gst_rtsp_auth_new ();
+
+  /* make default token, it has the same permissions as admin2 */
+  token =
+      gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+      "admin2", NULL);
+  gst_rtsp_auth_set_default_token (auth, token);
+  gst_rtsp_token_unref (token);
+
+  /* make user token */
+  token =
+      gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+      "user", NULL);
+  basic = gst_rtsp_auth_make_basic ("user", "password");
+  gst_rtsp_auth_add_basic (auth, basic, token);
+  g_free (basic);
+  gst_rtsp_token_unref (token);
+
+  /* make admin token */
+  token =
+      gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+      "admin", NULL);
+  basic = gst_rtsp_auth_make_basic ("admin", "power");
+  gst_rtsp_auth_add_basic (auth, basic, token);
+  g_free (basic);
+  gst_rtsp_token_unref (token);
+
+  /* make admin2 token */
+  token =
+      gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+      "admin2", NULL);
+  basic = gst_rtsp_auth_make_basic ("admin2", "power2");
+  gst_rtsp_auth_add_basic (auth, basic, token);
+  g_free (basic);
+  gst_rtsp_token_unref (token);
+
+  /* set as the server authentication manager */
+  gst_rtsp_server_set_auth (server, auth);
+  g_object_unref (auth);
+
+  /* attach the server to the default maincontext */
+  if (gst_rtsp_server_attach (server, NULL) == 0)
+    goto failed;
+
+  g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+  g_timeout_add_seconds (10, (GSourceFunc) remove_sessions, server);
+
+  /* start serving */
+  g_print ("stream with user:password ready at rtsp://127.0.0.1:8554/test\n");
+  g_print ("stream with admin:power ready at rtsp://127.0.0.1:8554/test\n");
+  g_print ("stream with admin2:power2 ready at rtsp://127.0.0.1:8554/test2\n");
+  g_main_loop_run (loop);
+
+  return 0;
+
+  /* ERRORS */
+failed:
+  {
+    g_print ("failed to attach the server\n");
+    return -1;
+  }
+}
diff --git a/examples/test-cgroups.c b/examples/test-cgroups.c
new file mode 100644
index 0000000..600fa3c
--- /dev/null
+++ b/examples/test-cgroups.c
@@ -0,0 +1,276 @@
+/* GStreamer
+ * Copyright (C) 2013 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/* Runs a pipeline and clasifies the media pipelines based on the
+ * authenticated user.
+ *
+ * This test requires 2 cpu cgroups to exist named 'user' and 'admin'.
+ * The rtsp server should have permission to add its threads to the
+ * cgroups.
+ *
+ *   sudo cgcreate -t uid:gid -g cpu:/user
+ *   sudo cgcreate -t uid:gid -g cpu:/admin
+ *
+ * With -t you can give the user and group access to the task file to
+ * write the thread ids. The user running the server can be used.
+ *
+ * Then you would want to change the cpu shares assigned to each group:
+ *
+ *   sudo cgset -r cpu.shares=100 user
+ *   sudo cgset -r cpu.shares=1024 admin
+ *
+ * Then start clients for 'user' until the stream is degraded because of
+ * lack of CPU. Then start a client for 'admin' and check that the stream
+ * is not degraded.
+ */
+
+#include 
+
+#include 
+#include 
+
+typedef struct _GstRTSPCGroupPool GstRTSPCGroupPool;
+typedef struct _GstRTSPCGroupPoolClass GstRTSPCGroupPoolClass;
+
+#define GST_TYPE_RTSP_CGROUP_POOL              (gst_rtsp_cgroup_pool_get_type ())
+#define GST_IS_RTSP_CGROUP_POOL(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_CGROUP_POOL))
+#define GST_IS_RTSP_CGROUP_POOL_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_CGROUP_POOL))
+#define GST_RTSP_CGROUP_POOL_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_CGROUP_POOL, GstRTSPCGroupPoolClass))
+#define GST_RTSP_CGROUP_POOL(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_CGROUP_POOL, GstRTSPCGroupPool))
+#define GST_RTSP_CGROUP_POOL_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_CGROUP_POOL, GstRTSPCGroupPoolClass))
+#define GST_RTSP_CGROUP_POOL_CAST(obj)         ((GstRTSPCGroupPool*)(obj))
+#define GST_RTSP_CGROUP_POOL_CLASS_CAST(klass) ((GstRTSPCGroupPoolClass*)(klass))
+
+struct _GstRTSPCGroupPool
+{
+  GstRTSPThreadPool parent;
+
+  struct cgroup *user;
+  struct cgroup *admin;
+};
+
+struct _GstRTSPCGroupPoolClass
+{
+  GstRTSPThreadPoolClass parent_class;
+};
+
+static GQuark thread_cgroup;
+
+static void gst_rtsp_cgroup_pool_finalize (GObject * obj);
+
+static void default_thread_enter (GstRTSPThreadPool * pool,
+    GstRTSPThread * thread);
+static void default_configure_thread (GstRTSPThreadPool * pool,
+    GstRTSPThread * thread, GstRTSPContext * ctx);
+
+static GType gst_rtsp_cgroup_pool_get_type (void);
+
+G_DEFINE_TYPE (GstRTSPCGroupPool, gst_rtsp_cgroup_pool,
+    GST_TYPE_RTSP_THREAD_POOL);
+
+static void
+gst_rtsp_cgroup_pool_class_init (GstRTSPCGroupPoolClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstRTSPThreadPoolClass *tpool_class;
+
+  gobject_class = G_OBJECT_CLASS (klass);
+  tpool_class = GST_RTSP_THREAD_POOL_CLASS (klass);
+
+  gobject_class->finalize = gst_rtsp_cgroup_pool_finalize;
+
+  tpool_class->configure_thread = default_configure_thread;
+  tpool_class->thread_enter = default_thread_enter;
+
+  thread_cgroup = g_quark_from_string ("cgroup.pool.thread.cgroup");
+
+  cgroup_init ();
+}
+
+static void
+gst_rtsp_cgroup_pool_init (GstRTSPCGroupPool * pool)
+{
+  pool->user = cgroup_new_cgroup ("user");
+  if (cgroup_add_controller (pool->user, "cpu") == NULL)
+    g_error ("Failed to add cpu controller to user cgroup");
+  pool->admin = cgroup_new_cgroup ("admin");
+  if (cgroup_add_controller (pool->admin, "cpu") == NULL)
+    g_error ("Failed to add cpu controller to admin cgroup");
+}
+
+static void
+gst_rtsp_cgroup_pool_finalize (GObject * obj)
+{
+  GstRTSPCGroupPool *pool = GST_RTSP_CGROUP_POOL (obj);
+
+  GST_INFO ("finalize pool %p", pool);
+
+  cgroup_free (&pool->user);
+  cgroup_free (&pool->admin);
+
+  G_OBJECT_CLASS (gst_rtsp_cgroup_pool_parent_class)->finalize (obj);
+}
+
+static void
+default_thread_enter (GstRTSPThreadPool * pool, GstRTSPThread * thread)
+{
+  struct cgroup *cgroup;
+
+  cgroup = gst_mini_object_get_qdata (GST_MINI_OBJECT (thread), thread_cgroup);
+  if (cgroup) {
+    gint res = 0;
+
+    res = cgroup_attach_task (cgroup);
+
+    if (res != 0)
+      GST_ERROR ("error: %d (%s)", res, cgroup_strerror (res));
+  }
+}
+
+static void
+default_configure_thread (GstRTSPThreadPool * pool,
+    GstRTSPThread * thread, GstRTSPContext * ctx)
+{
+  GstRTSPCGroupPool *cpool = GST_RTSP_CGROUP_POOL (pool);
+  const gchar *cls;
+  struct cgroup *cgroup;
+
+  if (ctx->token)
+    cls = gst_rtsp_token_get_string (ctx->token, "cgroup.pool.media.class");
+  else
+    cls = NULL;
+
+  GST_DEBUG ("manage cgroup %s", cls);
+
+  if (!g_strcmp0 (cls, "admin"))
+    cgroup = cpool->admin;
+  else
+    cgroup = cpool->user;
+
+  /* attach the cgroup to the thread */
+  gst_mini_object_set_qdata (GST_MINI_OBJECT (thread), thread_cgroup,
+      cgroup, NULL);
+}
+
+static gboolean
+timeout (GstRTSPServer * server)
+{
+  GstRTSPSessionPool *pool;
+
+  pool = gst_rtsp_server_get_session_pool (server);
+  gst_rtsp_session_pool_cleanup (pool);
+  g_object_unref (pool);
+
+  return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactory *factory;
+  GstRTSPAuth *auth;
+  GstRTSPToken *token;
+  gchar *basic;
+  GstRTSPThreadPool *thread_pool;
+
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* get the mounts for this server, every server has a default mapper object
+   * that be used to map uri mount points to media factories */
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  /* make a media factory for a test stream. The default media factory can use
+   * gst-launch syntax to create pipelines. 
+   * any launch line works as long as it contains elements named pay%d. Each
+   * element with pay%d names will be a stream */
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory, "( "
+      "videotestsrc ! video/x-raw,width=640,height=480,framerate=50/1 ! "
+      "x264enc ! rtph264pay name=pay0 pt=96 "
+      "audiotestsrc ! audio/x-raw,rate=8000 ! "
+      "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+  /* attach the test factory to the /test url */
+  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+  /* allow user and admin to access this resource */
+  gst_rtsp_media_factory_add_role (factory, "user",
+      "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+      "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
+  gst_rtsp_media_factory_add_role (factory, "admin",
+      "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+      "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
+
+  /* don't need the ref to the mapper anymore */
+  g_object_unref (mounts);
+
+  /* make a new authentication manager */
+  auth = gst_rtsp_auth_new ();
+
+  /* make user token */
+  token = gst_rtsp_token_new ("cgroup.pool.media.class", G_TYPE_STRING, "user",
+      "media.factory.role", G_TYPE_STRING, "user", NULL);
+  basic = gst_rtsp_auth_make_basic ("user", "password");
+  gst_rtsp_auth_add_basic (auth, basic, token);
+  g_free (basic);
+  gst_rtsp_token_unref (token);
+
+  /* make admin token */
+  token = gst_rtsp_token_new ("cgroup.pool.media.class", G_TYPE_STRING, "admin",
+      "media.factory.role", G_TYPE_STRING, "admin", NULL);
+  basic = gst_rtsp_auth_make_basic ("admin", "power");
+  gst_rtsp_auth_add_basic (auth, basic, token);
+  g_free (basic);
+  gst_rtsp_token_unref (token);
+
+  /* set as the server authentication manager */
+  gst_rtsp_server_set_auth (server, auth);
+  g_object_unref (auth);
+
+  thread_pool = g_object_new (GST_TYPE_RTSP_CGROUP_POOL, NULL);
+  gst_rtsp_server_set_thread_pool (server, thread_pool);
+  g_object_unref (thread_pool);
+
+  /* attach the server to the default maincontext */
+  if (gst_rtsp_server_attach (server, NULL) == 0)
+    goto failed;
+
+  g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+  /* start serving */
+  g_print ("stream with user:password ready at rtsp://127.0.0.1:8554/test\n");
+  g_print ("stream with admin:power ready at rtsp://127.0.0.1:8554/test\n");
+  g_main_loop_run (loop);
+
+  return 0;
+
+  /* ERRORS */
+failed:
+  {
+    g_print ("failed to attach the server\n");
+    return -1;
+  }
+}
diff --git a/examples/test-launch.c b/examples/test-launch.c
new file mode 100644
index 0000000..0466224
--- /dev/null
+++ b/examples/test-launch.c
@@ -0,0 +1,71 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactory *factory;
+
+  gst_init (&argc, &argv);
+
+  if (argc < 2) {
+    g_print ("usage: %s  \n"
+        "example: %s \"( videotestsrc ! x264enc ! rtph264pay name=pay0 pt=96 )\"\n",
+        argv[0], argv[0]);
+    return -1;
+  }
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* get the mount points for this server, every server has a default object
+   * that be used to map uri mount points to media factories */
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  /* make a media factory for a test stream. The default media factory can use
+   * gst-launch syntax to create pipelines.
+   * any launch line works as long as it contains elements named pay%d. Each
+   * element with pay%d names will be a stream */
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory, argv[1]);
+
+  /* attach the test factory to the /test url */
+  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+  /* don't need the ref to the mapper anymore */
+  g_object_unref (mounts);
+
+  /* attach the server to the default maincontext */
+  gst_rtsp_server_attach (server, NULL);
+
+  /* start serving */
+  g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+  g_main_loop_run (loop);
+
+  return 0;
+}
diff --git a/examples/test-mp4.c b/examples/test-mp4.c
new file mode 100644
index 0000000..52e082d
--- /dev/null
+++ b/examples/test-mp4.c
@@ -0,0 +1,76 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactory *factory;
+  gchar *str;
+
+  gst_init (&argc, &argv);
+
+  if (argc < 2) {
+    g_message ("usage: %s ", argv[0]);
+    return -1;
+  }
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* get the mount points for this server, every server has a default object
+   * that be used to map uri mount points to media factories */
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  str = g_strdup_printf ("( "
+      "filesrc location=%s ! qtdemux name=d "
+      "d. ! queue ! rtph264pay pt=96 name=pay0 "
+      "d. ! queue ! rtpmp4apay pt=97 name=pay1 " ")", argv[1]);
+
+  /* make a media factory for a test stream. The default media factory can use
+   * gst-launch syntax to create pipelines. 
+   * any launch line works as long as it contains elements named pay%d. Each
+   * element with pay%d names will be a stream */
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory, str);
+  g_free (str);
+
+  /* attach the test factory to the /test url */
+  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+  /* don't need the ref to the mapper anymore */
+  g_object_unref (mounts);
+
+  /* attach the server to the default maincontext */
+  gst_rtsp_server_attach (server, NULL);
+
+  /* start serving */
+  g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+  g_main_loop_run (loop);
+
+  return 0;
+}
diff --git a/examples/test-multicast.c b/examples/test-multicast.c
new file mode 100644
index 0000000..c7d55fc
--- /dev/null
+++ b/examples/test-multicast.c
@@ -0,0 +1,104 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+
+static gboolean
+timeout (GstRTSPServer * server, gboolean ignored)
+{
+  GstRTSPSessionPool *pool;
+
+  pool = gst_rtsp_server_get_session_pool (server);
+  gst_rtsp_session_pool_cleanup (pool);
+  g_object_unref (pool);
+
+  return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactory *factory;
+  GstRTSPAddressPool *pool;
+
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* get the mount points for this server, every server has a default object
+   * that be used to map uri mount points to media factories */
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  /* make a media factory for a test stream. The default media factory can use
+   * gst-launch syntax to create pipelines. 
+   * any launch line works as long as it contains elements named pay%d. Each
+   * element with pay%d names will be a stream */
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory, "( "
+      "videotestsrc ! video/x-raw,width=352,height=288,framerate=15/1 ! "
+      "x264enc ! rtph264pay name=pay0 pt=96 "
+      "audiotestsrc ! audio/x-raw,rate=8000 ! "
+      "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+
+  gst_rtsp_media_factory_set_shared (factory, TRUE);
+
+  /* make a new address pool */
+  pool = gst_rtsp_address_pool_new ();
+  gst_rtsp_address_pool_add_range (pool,
+      "224.3.0.0", "224.3.0.10", 5000, 5010, 16);
+  gst_rtsp_media_factory_set_address_pool (factory, pool);
+  /* only allow multicast */
+  gst_rtsp_media_factory_set_protocols (factory,
+      GST_RTSP_LOWER_TRANS_UDP_MCAST);
+  g_object_unref (pool);
+
+  /* attach the test factory to the /test url */
+  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+  /* don't need the ref to the mapper anymore */
+  g_object_unref (mounts);
+
+  /* attach the server to the default maincontext */
+  if (gst_rtsp_server_attach (server, NULL) == 0)
+    goto failed;
+
+  g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+  /* start serving */
+  g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+  g_main_loop_run (loop);
+
+  return 0;
+
+  /* ERRORS */
+failed:
+  {
+    g_print ("failed to attach the server\n");
+    return -1;
+  }
+}
diff --git a/examples/test-multicast2.c b/examples/test-multicast2.c
new file mode 100644
index 0000000..1d8e6f2
--- /dev/null
+++ b/examples/test-multicast2.c
@@ -0,0 +1,125 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+
+static gboolean
+timeout (GstRTSPServer * server, gboolean ignored)
+{
+  GstRTSPSessionPool *pool;
+
+  pool = gst_rtsp_server_get_session_pool (server);
+  gst_rtsp_session_pool_cleanup (pool);
+  g_object_unref (pool);
+
+  return TRUE;
+}
+
+static void
+media_constructed (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
+{
+  guint i, n_streams;
+
+  n_streams = gst_rtsp_media_n_streams (media);
+
+  for (i = 0; i < n_streams; i++) {
+    GstRTSPAddressPool *pool;
+    GstRTSPStream *stream;
+    gchar *min, *max;
+
+    stream = gst_rtsp_media_get_stream (media, i);
+
+    /* make a new address pool */
+    pool = gst_rtsp_address_pool_new ();
+
+    min = g_strdup_printf ("224.3.0.%d", (2 * i) + 1);
+    max = g_strdup_printf ("224.3.0.%d", (2 * i) + 2);
+    gst_rtsp_address_pool_add_range (pool, min, max,
+        5000 + (10 * i), 5010 + (10 * i), 1);
+    g_free (min);
+    g_free (max);
+
+    gst_rtsp_stream_set_address_pool (stream, pool);
+    g_object_unref (pool);
+  }
+}
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactory *factory;
+
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* get the mount points for this server, every server has a default object
+   * that be used to map uri mount points to media factories */
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  /* make a media factory for a test stream. The default media factory can use
+   * gst-launch syntax to create pipelines. 
+   * any launch line works as long as it contains elements named pay%d. Each
+   * element with pay%d names will be a stream */
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory, "( "
+      "videotestsrc ! video/x-raw,width=352,height=288,framerate=15/1 ! "
+      "x264enc ! rtph264pay name=pay0 pt=96 "
+      "audiotestsrc ! audio/x-raw,rate=8000 ! "
+      "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+
+  gst_rtsp_media_factory_set_shared (factory, TRUE);
+
+  g_signal_connect (factory, "media-constructed", (GCallback)
+      media_constructed, NULL);
+
+  /* attach the test factory to the /test url */
+  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+  /* don't need the ref to the mapper anymore */
+  g_object_unref (mounts);
+
+  /* attach the server to the default maincontext */
+  if (gst_rtsp_server_attach (server, NULL) == 0)
+    goto failed;
+
+  g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+  /* start serving */
+  g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+  g_main_loop_run (loop);
+
+  return 0;
+
+  /* ERRORS */
+failed:
+  {
+    g_print ("failed to attach the server\n");
+    return -1;
+  }
+}
diff --git a/examples/test-ogg.c b/examples/test-ogg.c
new file mode 100644
index 0000000..f4d104c
--- /dev/null
+++ b/examples/test-ogg.c
@@ -0,0 +1,76 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactory *factory;
+  gchar *str;
+
+  gst_init (&argc, &argv);
+
+  if (argc < 2) {
+    g_message ("usage: %s ", argv[0]);
+    return -1;
+  }
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* get the mount points for this server, every server has a default object
+   * that be used to map uri mount points to media factories */
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  str = g_strdup_printf ("( "
+      "filesrc location=%s ! oggdemux name=d "
+      "d. ! queue ! rtptheorapay name=pay0 pt=96 "
+      "d. ! queue ! rtpvorbispay name=pay1 pt=97 " ")", argv[1]);
+
+  /* make a media factory for a test stream. The default media factory can use
+   * gst-launch syntax to create pipelines. 
+   * any launch line works as long as it contains elements named pay%d. Each
+   * element with pay%d names will be a stream */
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory, str);
+  g_free (str);
+
+  /* attach the test factory to the /test url */
+  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+  /* don't need the ref to the mapper anymore */
+  g_object_unref (mounts);
+
+  /* attach the server to the default maincontext */
+  gst_rtsp_server_attach (server, NULL);
+
+  /* start serving */
+  g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+  g_main_loop_run (loop);
+
+  return 0;
+}
diff --git a/examples/test-readme.c b/examples/test-readme.c
new file mode 100644
index 0000000..2e2caa6
--- /dev/null
+++ b/examples/test-readme.c
@@ -0,0 +1,67 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactory *factory;
+
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* get the mount points for this server, every server has a default object
+   * that be used to map uri mount points to media factories */
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  /* make a media factory for a test stream. The default media factory can use
+   * gst-launch syntax to create pipelines. 
+   * any launch line works as long as it contains elements named pay%d. Each
+   * element with pay%d names will be a stream */
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory,
+      "( videotestsrc is-live=1 ! x264enc ! rtph264pay name=pay0 pt=96 )");
+
+  gst_rtsp_media_factory_set_shared (factory, TRUE);
+
+  /* attach the test factory to the /test url */
+  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+  /* don't need the ref to the mapper anymore */
+  g_object_unref (mounts);
+
+  /* attach the server to the default maincontext */
+  gst_rtsp_server_attach (server, NULL);
+
+  /* start serving */
+  g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+  g_main_loop_run (loop);
+
+  return 0;
+}
diff --git a/examples/test-sdp.c b/examples/test-sdp.c
new file mode 100644
index 0000000..98efe9e
--- /dev/null
+++ b/examples/test-sdp.c
@@ -0,0 +1,98 @@
+/* GStreamer
+ * Copyright (C) 2009 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+
+static gboolean
+timeout (GstRTSPServer * server, gboolean ignored)
+{
+  GstRTSPSessionPool *pool;
+
+  pool = gst_rtsp_server_get_session_pool (server);
+  gst_rtsp_session_pool_cleanup (pool);
+  g_object_unref (pool);
+
+  return TRUE;
+}
+
+static void
+media_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
+{
+  gst_rtsp_media_set_reusable (media, TRUE);
+}
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactory *factory;
+  gchar *str;
+
+  gst_init (&argc, &argv);
+
+  if (argc < 2) {
+    g_message ("usage: %s ", argv[0]);
+    return -1;
+  }
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* get the mount points for this server, every server has a default object
+   * that be used to map uri mount points to media factories */
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  /* make a media factory for a test stream. The default media factory can use
+   * gst-launch syntax to create pipelines. 
+   * any launch line works as long as it contains elements named pay%d. Each
+   * element with pay%d names will be a stream */
+  factory = gst_rtsp_media_factory_new ();
+
+  str =
+      g_strdup_printf ("( filesrc location=%s ! sdpdemux name=dynpay0 )",
+      argv[1]);
+  gst_rtsp_media_factory_set_launch (factory, str);
+  gst_rtsp_media_factory_set_shared (factory, TRUE);
+  g_signal_connect (factory, "media-configure", (GCallback) media_configure,
+      NULL);
+  g_free (str);
+
+  /* attach the test factory to the /test url */
+  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+  /* don't need the ref to the mapper anymore */
+  g_object_unref (mounts);
+
+  /* attach the server to the default maincontext */
+  gst_rtsp_server_attach (server, NULL);
+
+  g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+  /* start serving */
+  g_main_loop_run (loop);
+
+  return 0;
+}
diff --git a/examples/test-uri.c b/examples/test-uri.c
new file mode 100644
index 0000000..b0daebc
--- /dev/null
+++ b/examples/test-uri.c
@@ -0,0 +1,114 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+#include 
+
+
+static gboolean
+timeout (GstRTSPServer * server)
+{
+  GstRTSPSessionPool *pool;
+
+  pool = gst_rtsp_server_get_session_pool (server);
+  gst_rtsp_session_pool_cleanup (pool);
+  g_object_unref (pool);
+
+  return TRUE;
+}
+
+static gboolean
+remove_map (GstRTSPServer * server)
+{
+  GstRTSPMountPoints *mounts;
+
+  g_print ("removing /test mount point\n");
+  mounts = gst_rtsp_server_get_mount_points (server);
+  gst_rtsp_mount_points_remove_factory (mounts, "/test");
+  g_object_unref (mounts);
+
+  return FALSE;
+}
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactoryURI *factory;
+
+  gst_init (&argc, &argv);
+
+  if (argc < 2) {
+    g_message ("usage: %s ", argv[0]);
+    return -1;
+  }
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* get the mount points for this server, every server has a default object
+   * that be used to map uri mount points to media factories */
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  /* make a URI media factory for a test stream. */
+  factory = gst_rtsp_media_factory_uri_new ();
+  /* when using GStreamer as a client, one can use the gst payloader, which is
+   * more efficient when there is no payloader for the compressed format */
+  /* g_object_set (factory, "use-gstpay", TRUE, NULL); */
+  gst_rtsp_media_factory_uri_set_uri (factory, argv[1]);
+  /* if you want multiple clients to see the same video, set the shared property
+   * to TRUE */
+  /* gst_rtsp_media_factory_set_shared ( GST_RTSP_MEDIA_FACTORY (factory), TRUE); */
+
+  /* attach the test factory to the /test url */
+  gst_rtsp_mount_points_add_factory (mounts, "/test",
+      GST_RTSP_MEDIA_FACTORY (factory));
+
+  /* don't need the ref to the mapper anymore */
+  g_object_unref (mounts);
+
+  /* attach the server to the default maincontext */
+  if (gst_rtsp_server_attach (server, NULL) == 0)
+    goto failed;
+
+  /* do session cleanup every 2 seconds */
+  g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+  /* remove the mount point after 10 seconds, new clients won't be able to use the
+   * /test url anymore */
+  g_timeout_add_seconds (10, (GSourceFunc) remove_map, server);
+
+  /* start serving */
+  g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+  g_main_loop_run (loop);
+
+  return 0;
+
+  /* ERRORS */
+failed:
+  {
+    g_print ("failed to attach the server\n");
+    return -1;
+  }
+}
diff --git a/examples/test-video.c b/examples/test-video.c
new file mode 100644
index 0000000..0d4a3a5
--- /dev/null
+++ b/examples/test-video.c
@@ -0,0 +1,177 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+/* define this if you want the resource to only be available when using
+ * user/password as the password */
+#undef WITH_AUTH
+
+/* define this if you want the server to use TLS (it will also need WITH_AUTH
+ * to be defined) */
+#undef WITH_TLS
+
+/* this timeout is periodically run to clean up the expired sessions from the
+ * pool. This needs to be run explicitly currently but might be done
+ * automatically as part of the mainloop. */
+static gboolean
+timeout (GstRTSPServer * server, gboolean ignored)
+{
+  GstRTSPSessionPool *pool;
+
+  pool = gst_rtsp_server_get_session_pool (server);
+  gst_rtsp_session_pool_cleanup (pool);
+  g_object_unref (pool);
+
+  return TRUE;
+}
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactory *factory;
+#ifdef WITH_AUTH
+  GstRTSPAuth *auth;
+  GstRTSPToken *token;
+  gchar *basic;
+  GstRTSPPermissions *permissions;
+#endif
+#ifdef WITH_TLS
+  GTlsCertificate *cert;
+  GError *error = NULL;
+#endif
+
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+#ifdef WITH_AUTH
+  /* make a new authentication manager. it can be added to control access to all
+   * the factories on the server or on individual factories. */
+  auth = gst_rtsp_auth_new ();
+#ifdef WITH_TLS
+  cert = g_tls_certificate_new_from_pem ("-----BEGIN CERTIFICATE-----"
+      "MIICJjCCAY+gAwIBAgIBBzANBgkqhkiG9w0BAQUFADCBhjETMBEGCgmSJomT8ixk"
+      "ARkWA0NPTTEXMBUGCgmSJomT8ixkARkWB0VYQU1QTEUxHjAcBgNVBAsTFUNlcnRp"
+      "ZmljYXRlIEF1dGhvcml0eTEXMBUGA1UEAxMOY2EuZXhhbXBsZS5jb20xHTAbBgkq"
+      "hkiG9w0BCQEWDmNhQGV4YW1wbGUuY29tMB4XDTExMDExNzE5NDcxN1oXDTIxMDEx"
+      "NDE5NDcxN1owSzETMBEGCgmSJomT8ixkARkWA0NPTTEXMBUGCgmSJomT8ixkARkW"
+      "B0VYQU1QTEUxGzAZBgNVBAMTEnNlcnZlci5leGFtcGxlLmNvbTBcMA0GCSqGSIb3"
+      "DQEBAQUAA0sAMEgCQQDYScTxk55XBmbDM9zzwO+grVySE4rudWuzH2PpObIonqbf"
+      "hRoAalKVluG9jvbHI81eXxCdSObv1KBP1sbN5RzpAgMBAAGjIjAgMAkGA1UdEwQC"
+      "MAAwEwYDVR0lBAwwCgYIKwYBBQUHAwEwDQYJKoZIhvcNAQEFBQADgYEAYx6fMqT1"
+      "Gvo0jq88E8mc+bmp4LfXD4wJ7KxYeadQxt75HFRpj4FhFO3DOpVRFgzHlOEo3Fwk"
+      "PZOKjvkT0cbcoEq5whLH25dHoQxGoVQgFyAP5s+7Vp5AlHh8Y/vAoXeEVyy/RCIH"
+      "QkhUlAflfDMcrrYjsmwoOPSjhx6Mm/AopX4="
+      "-----END CERTIFICATE-----"
+      "-----BEGIN PRIVATE KEY-----"
+      "MIIBVAIBADANBgkqhkiG9w0BAQEFAASCAT4wggE6AgEAAkEA2EnE8ZOeVwZmwzPc"
+      "88DvoK1ckhOK7nVrsx9j6TmyKJ6m34UaAGpSlZbhvY72xyPNXl8QnUjm79SgT9bG"
+      "zeUc6QIDAQABAkBRFJZ32VbqWMP9OVwDJLiwC01AlYLnka0mIQZbT/2xq9dUc9GW"
+      "U3kiVw4lL8v/+sPjtTPCYYdzHHOyDen6znVhAiEA9qJT7BtQvRxCvGrAhr9MS022"
+      "tTdPbW829BoUtIeH64cCIQDggG5i48v7HPacPBIH1RaSVhXl8qHCpQD3qrIw3FMw"
+      "DwIga8PqH5Sf5sHedy2+CiK0V4MRfoU4c3zQ6kArI+bEgSkCIQCLA1vXBiE31B5s"
+      "bdHoYa1BXebfZVd+1Hd95IfEM5mbRwIgSkDuQwV55BBlvWph3U8wVIMIb4GStaH8"
+      "W535W8UBbEg=" "-----END PRIVATE KEY-----", -1, &error);
+  if (cert == NULL) {
+    g_printerr ("failed to parse PEM: %s\n", error->message);
+    return -1;
+  }
+  gst_rtsp_auth_set_tls_certificate (auth, cert);
+  g_object_unref (cert);
+#endif
+
+  /* make user token */
+  token =
+      gst_rtsp_token_new (GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+      "user", NULL);
+  basic = gst_rtsp_auth_make_basic ("user", "password");
+  gst_rtsp_auth_add_basic (auth, basic, token);
+  g_free (basic);
+  gst_rtsp_token_unref (token);
+
+  /* configure in the server */
+  gst_rtsp_server_set_auth (server, auth);
+#endif
+
+  /* get the mount points for this server, every server has a default object
+   * that be used to map uri mount points to media factories */
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  /* make a media factory for a test stream. The default media factory can use
+   * gst-launch syntax to create pipelines.
+   * any launch line works as long as it contains elements named pay%d. Each
+   * element with pay%d names will be a stream */
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory, "( "
+      "videotestsrc ! video/x-raw,width=352,height=288,framerate=15/1 ! "
+      "x264enc ! rtph264pay name=pay0 pt=96 "
+      "audiotestsrc ! audio/x-raw,rate=8000 ! "
+      "alawenc ! rtppcmapay name=pay1 pt=97 " ")");
+#ifdef WITH_AUTH
+  /* add permissions for the user media role */
+  permissions = gst_rtsp_permissions_new ();
+  gst_rtsp_permissions_add_role (permissions, "user",
+      GST_RTSP_PERM_MEDIA_FACTORY_ACCESS, G_TYPE_BOOLEAN, TRUE,
+      GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT, G_TYPE_BOOLEAN, TRUE, NULL);
+  gst_rtsp_media_factory_set_permissions (factory, permissions);
+  gst_rtsp_permissions_unref (permissions);
+#ifdef WITH_TLS
+  gst_rtsp_media_factory_set_profiles (factory, GST_RTSP_PROFILE_SAVP);
+#endif
+#endif
+
+  /* attach the test factory to the /test url */
+  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+  /* don't need the ref to the mapper anymore */
+  g_object_unref (mounts);
+
+  /* attach the server to the default maincontext */
+  if (gst_rtsp_server_attach (server, NULL) == 0)
+    goto failed;
+
+  /* add a timeout for the session cleanup */
+  g_timeout_add_seconds (2, (GSourceFunc) timeout, server);
+
+  /* start serving, this never stops */
+#ifdef WITH_TLS
+  g_print ("stream ready at rtsps://127.0.0.1:8554/test\n");
+#else
+  g_print ("stream ready at rtsp://127.0.0.1:8554/test\n");
+#endif
+  g_main_loop_run (loop);
+
+  return 0;
+
+  /* ERRORS */
+failed:
+  {
+    g_print ("failed to attach the server\n");
+    return -1;
+  }
+}
diff --git a/examples/test-wfd.c b/examples/test-wfd.c
new file mode 100644
index 0000000..26af281
--- /dev/null
+++ b/examples/test-wfd.c
@@ -0,0 +1,133 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+#include 
+
+#include 
+#include 
+
+#if 1
+#define VIDEO_PIPELINE "ximagesrc ! videorate ! videoscale ! videoconvert ! " \
+  "video/x-raw,width=640,height=480,framerate=30/1 ! " \
+  "x264enc aud=false byte-stream=true bitrate=512 ! video/x-h264,profile=baseline ! mpegtsmux wfd-mode=TRUE ! " \
+  "rtpmp2tpay name=pay0 pt=33"
+#define AUDIO_PIPELINE "pulsesrc device=alsa_output.pci-0000_00_1b.0.analog-stereo.monitor ! audioconvert ! " \
+  "faac ! mpegtsmux wfd-mode=TRUE ! " \
+  "rtpmp2tpay name=pay0 pt=33"
+#else
+#define VIDEO_PIPELINE "ximagesrc ! videoscale ! videoconvert ! " \
+  "video/x-raw,width=640,height=480,framerate=60/1 ! " \
+  "x264enc aud=false byte-stream=true bitrate=512 ! video/x-h264,profile=baseline ! mpegtsmux name=mux " \
+  "pulsesrc device=alsa_output.pci-0000_00_1b.0.analog-stereo.monitor ! audioconvert ! " \
+  "faac ! mux. mux. ! " \
+  "rtpmp2tpay name=pay0 pt=33"
+#if 0
+#define VIDEO_PIPELINE "ximagesrc startx=0 starty=0 endx=1919 endy=1079 ! videorate ! videoscale ! video/x-raw,width=1280,height=720,framerate=30/1 ! videoconvert ! " \
+  "x264enc aud=false byte-stream=true bitrate=512 ! video/x-h264,profile=baseline ! mpegtsmux name=mux wfd-mode=TRUE " \
+  "pulsesrc device=alsa_output.pci-0000_00_1b.0.analog-stereo.monitor ! audioconvert ! " \
+  "faac ! mux. mux. ! " \
+  "rtpmp2tpay name=pay0 pt=33"
+#endif
+/*
+#define VIDEO_PIPELINE "ximagesrc do-timestamp=true ! videoscale ! video/x-raw,width=1280,height=720,framerate=30/1 ! videoconvert ! " \
+  "x264enc aud=false byte-stream=true bitrate=512 ! video/x-h264,profile=baseline ! mpegtsmux name=mux wfd-mode=TRUE " \
+  "pulsesrc device=alsa_output.pci-0000_00_1b.0.analog-stereo.monitor provide-clock=false ! audioconvert ! " \
+  "faac ! mux. mux. ! " \
+  "rtpmp2tpay name=pay0 pt=33"
+*/
+#endif
+
+#define WFD_RTSP_PORT "2022"
+#define TEST_MOUNT_POINT  "/wfd1.0/streamid=0"
+
+GMainLoop *loop;
+
+static gboolean
+teardown(gpointer data)
+{
+  GstRTSPWFDServer *server = NULL;
+  server = (GstRTSPWFDServer *) data;
+
+  g_print("teardown\n");
+
+  if (server == NULL) return FALSE;
+
+  gst_rtsp_wfd_server_trigger_request (GST_RTSP_SERVER(server), WFD_TRIGGER_TEARDOWN);
+
+  return FALSE;
+}
+
+int main (int argc, char *argv[])
+{
+  GstRTSPWFDServer *server;
+  guint id;
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactoryWFD *factory;
+  GMainContext *context = NULL;
+  GSource * signal_handler_src = NULL;
+
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+  context = g_main_loop_get_context(loop);
+
+  /* create a server instance */
+  server = gst_rtsp_wfd_server_new ();
+
+  gst_rtsp_server_set_address(GST_RTSP_SERVER(server), "192.168.3.100"); 
+  //gst_rtsp_server_set_address(GST_RTSP_SERVER(server), "192.168.49.194"); 
+  //gst_rtsp_server_set_address(GST_RTSP_SERVER(server), "192.168.49.20"); 
+  gst_rtsp_server_set_service(GST_RTSP_SERVER(server), WFD_RTSP_PORT);
+  mounts = gst_rtsp_server_get_mount_points (GST_RTSP_SERVER(server));
+
+  factory = gst_rtsp_media_factory_wfd_new ();
+
+  gst_rtsp_media_factory_set_launch (GST_RTSP_MEDIA_FACTORY(factory),
+      "( " VIDEO_PIPELINE " )");
+  g_object_ref (factory);
+  gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, GST_RTSP_MEDIA_FACTORY(factory));
+  if (mounts) g_object_unref (mounts);
+
+  /* attach the server to the default maincontext */
+  if ((id = gst_rtsp_server_attach (GST_RTSP_SERVER_CAST(server), NULL)) == 0)
+    goto failed;
+
+  signal_handler_src = (GSource *)g_unix_signal_source_new (SIGINT);
+  /* Set callback to be called when socket is readable */
+  g_source_set_callback(signal_handler_src, teardown, server, NULL);
+  g_source_attach(signal_handler_src, context);
+
+  /* start serving */
+  g_main_loop_run (loop);
+
+  /* cleanup */
+  g_source_remove (id);
+  g_object_unref (server);
+  g_main_loop_unref (loop);
+
+  return 0;
+
+  /* ERRORS */
+failed:
+  {
+    g_print ("failed to attach the server\n");
+    return -1;
+  }
+}
diff --git a/gst-rtsp-server.doap b/gst-rtsp-server.doap
new file mode 100644
index 0000000..8dfa661
--- /dev/null
+++ b/gst-rtsp-server.doap
@@ -0,0 +1,160 @@
+
+
+ GStreamer RTSP Server
+ gst-rtsp-server
+ 
+ 1999-10-31
+ 
+RTSP server library based on GStreamer
+
+ 
+RTSP server library based on GStreamer
+ 
+ 
+ 
+ 
+ 
+ C
+ 
+ 
+
+ 
+   
+     
+     
+   
+ 
+
+ 
+  
+   1.4.5
+   1.4
+   
+   2014-12-18
+   
+  
+ 
+
+ 
+  
+   1.4.4
+   1.4
+   
+   2014-11-06
+   
+  
+ 
+
+ 
+  
+   1.4.3
+   1.4
+   
+   2014-09-24
+   
+  
+ 
+
+ 
+  
+   1.4.2
+   1.4
+   
+   2014-09-19
+   
+  
+ 
+
+ 
+  
+   1.4.1
+   1.4
+   
+   2014-08-27
+   
+  
+ 
+
+ 
+  
+   1.4.0
+   1.4
+   
+   2014-07-19
+   
+  
+ 
+
+ 
+  
+   1.3.91
+   1.3
+   
+   2014-07-11
+   
+  
+ 
+
+ 
+  
+   1.3.90
+   1.3
+   
+   2014-06-28
+   
+  
+ 
+
+ 
+  
+   1.3.3
+   1.3
+   
+   2014-06-22
+   
+  
+ 
+
+ 
+  
+   1.3.2
+   1.3
+   
+   2014-05-21
+   
+  
+ 
+
+ 
+  
+   1.3.1
+   1.3
+   
+   2014-05-03
+   
+  
+ 
+
+ 
+  
+   1.1.90
+   1.1
+   
+   2014-02-09
+   
+  
+ 
+
+ 
+  
+     Wim Taymans
+     0d93fde052812d51a05fd86de9bdbf674423daa2
+  
+ 
+
+
diff --git a/gst-rtsp-server.manifest b/gst-rtsp-server.manifest
new file mode 100644
index 0000000..a76fdba
--- /dev/null
+++ b/gst-rtsp-server.manifest
@@ -0,0 +1,5 @@
+
+	
+		
+	
+
diff --git a/gst/Makefile.am b/gst/Makefile.am
new file mode 100644
index 0000000..e37bbc6
--- /dev/null
+++ b/gst/Makefile.am
@@ -0,0 +1 @@
+SUBDIRS = rtsp-server
diff --git a/gst/Makefile.in b/gst/Makefile.in
new file mode 100644
index 0000000..35a2115
--- /dev/null
+++ b/gst/Makefile.in
@@ -0,0 +1,734 @@
+# Makefile.in generated by automake 1.14.1 from Makefile.am.
+# @configure_input@
+
+# Copyright (C) 1994-2013 Free Software Foundation, Inc.
+
+# This Makefile.in is free software; the Free Software Foundation
+# gives unlimited permission to copy and/or distribute it,
+# with or without modifications, as long as this notice is preserved.
+
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
+# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
+# PARTICULAR PURPOSE.
+
+@SET_MAKE@
+VPATH = @srcdir@
+am__is_gnu_make = test -n '$(MAKEFILE_LIST)' && test -n '$(MAKELEVEL)'
+am__make_running_with_option = \
+  case $${target_option-} in \
+      ?) ;; \
+      *) echo "am__make_running_with_option: internal error: invalid" \
+              "target option '$${target_option-}' specified" >&2; \
+         exit 1;; \
+  esac; \
+  has_opt=no; \
+  sane_makeflags=$$MAKEFLAGS; \
+  if $(am__is_gnu_make); then \
+    sane_makeflags=$$MFLAGS; \
+  else \
+    case $$MAKEFLAGS in \
+      *\\[\ \	]*) \
+        bs=\\; \
+        sane_makeflags=`printf '%s\n' "$$MAKEFLAGS" \
+          | sed "s/$$bs$$bs[$$bs $$bs	]*//g"`;; \
+    esac; \
+  fi; \
+  skip_next=no; \
+  strip_trailopt () \
+  { \
+    flg=`printf '%s\n' "$$flg" | sed "s/$$1.*$$//"`; \
+  }; \
+  for flg in $$sane_makeflags; do \
+    test $$skip_next = yes && { skip_next=no; continue; }; \
+    case $$flg in \
+      *=*|--*) continue;; \
+        -*I) strip_trailopt 'I'; skip_next=yes;; \
+      -*I?*) strip_trailopt 'I';; \
+        -*O) strip_trailopt 'O'; skip_next=yes;; \
+      -*O?*) strip_trailopt 'O';; \
+        -*l) strip_trailopt 'l'; skip_next=yes;; \
+      -*l?*) strip_trailopt 'l';; \
+      -[dEDm]) skip_next=yes;; \
+      -[JT]) skip_next=yes;; \
+    esac; \
+    case $$flg in \
+      *$$target_option*) has_opt=yes; break;; \
+    esac; \
+  done; \
+  test $$has_opt = yes
+am__make_dryrun = (target_option=n; $(am__make_running_with_option))
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+	  if test -f $$file || test -d $$file; then d=.; else d=$(srcdir); fi; \
+	  if test -d $$d/$$file; then \
+	    dir=`echo "/$$file" | sed -e 's,/[^/]*$$,,'`; \
+	    if test -d "$(distdir)/$$file"; then \
+	      find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+	    fi; \
+	    if test -d $(srcdir)/$$file && test $$d != $(srcdir); then \
+	      cp -fpR $(srcdir)/$$file "$(distdir)$$dir" || exit 1; \
+	      find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+	    fi; \
+	    cp -fpR $$d/$$file "$(distdir)$$dir" || exit 1; \
+	  else \
+	    test -f "$(distdir)/$$file" \
+	    || cp -p $$d/$$file "$(distdir)/$$file" \
+	    || exit 1; \
+	  fi; \
+	done
+	@list='$(DIST_SUBDIRS)'; for subdir in $$list; do \
+	  if test "$$subdir" = .; then :; else \
+	    $(am__make_dryrun) \
+	      || test -d "$(distdir)/$$subdir" \
+	      || $(MKDIR_P) "$(distdir)/$$subdir" \
+	      || exit 1; \
+	    dir1=$$subdir; dir2="$(distdir)/$$subdir"; \
+	    $(am__relativize); \
+	    new_distdir=$$reldir; \
+	    dir1=$$subdir; dir2="$(top_distdir)"; \
+	    $(am__relativize); \
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+	    echo " (cd $$subdir && $(MAKE) $(AM_MAKEFLAGS) top_distdir="$$new_top_distdir" distdir="$$new_distdir" \\"; \
+	    echo "     am__remove_distdir=: am__skip_length_check=: am__skip_mode_fix=: distdir)"; \
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+		am__skip_length_check=: \
+		am__skip_mode_fix=: \
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+	  $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	    install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	      install; \
+	else \
+	  $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	    install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	    "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'" install; \
+	fi
+mostlyclean-generic:
+
+clean-generic:
+
+distclean-generic:
+	-test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
+	-test . = "$(srcdir)" || test -z "$(CONFIG_CLEAN_VPATH_FILES)" || rm -f $(CONFIG_CLEAN_VPATH_FILES)
+
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+	@echo "This command is intended for maintainers to use"
+	@echo "it deletes files that may require special tools to rebuild."
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+	-rm -f Makefile
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diff --git a/gst/rtsp-server/Makefile.am b/gst/rtsp-server/Makefile.am
new file mode 100644
index 0000000..60eaf3e
--- /dev/null
+++ b/gst/rtsp-server/Makefile.am
@@ -0,0 +1,128 @@
+public_headers = \
+		rtsp-auth.h \
+		rtsp-address-pool.h \
+		rtsp-context.h \
+		rtsp-params.h \
+		rtsp-sdp.h \
+		rtsp-thread-pool.h \
+		rtsp-media.h \
+		rtsp-media-factory.h \
+		rtsp-media-factory-wfd.h \
+		rtsp-media-factory-uri.h \
+		rtsp-mount-points.h \
+		rtsp-permissions.h \
+		rtsp-stream.h \
+		rtsp-stream-transport.h \
+		rtsp-session.h \
+		rtsp-session-media.h \
+		rtsp-session-pool.h \
+		rtsp-token.h \
+		rtsp-client-wfd.h \
+		rtsp-client.h \
+		rtsp-server-wfd.h \
+		rtsp-server.h \
+		gstwfdmessage.h
+
+c_sources = \
+	rtsp-auth.c \
+	rtsp-address-pool.c \
+	rtsp-context.c \
+	rtsp-params.c \
+	rtsp-sdp.c \
+	rtsp-thread-pool.c \
+	rtsp-media.c \
+	rtsp-media-factory.c \
+	rtsp-media-factory-wfd.c \
+	rtsp-media-factory-uri.c \
+	rtsp-mount-points.c \
+	rtsp-permissions.c \
+	rtsp-stream.c \
+	rtsp-stream-transport.c \
+	rtsp-session.c \
+	rtsp-session-media.c \
+	rtsp-session-pool.c \
+	rtsp-token.c \
+	gstwfdmessage.c \
+	rtsp-client-wfd.c \
+	rtsp-client.c \
+	rtsp-server-wfd.c \
+	rtsp-server.c
+
+noinst_HEADERS = 
+
+lib_LTLIBRARIES = \
+	libgstrtspserver-@GST_API_VERSION@.la
+
+libgstrtspserver_@GST_API_VERSION@_la_SOURCES = \
+    $(c_sources)
+
+libgstrtspserver_@GST_API_VERSION@_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
+libgstrtspserver_@GST_API_VERSION@_la_LDFLAGS = $(GST_LIB_LDFLAGS) $(GST_ALL_LDFLAGS) $(GST_LT_LDFLAGS)
+libgstrtspserver_@GST_API_VERSION@_la_LIBADD = \
+	$(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) \
+    -lgstrtp-@GST_API_VERSION@ -lgstrtsp-@GST_API_VERSION@ \
+            -lgstnet-@GST_API_VERSION@ \
+            -lgstsdp-@GST_API_VERSION@ \
+            -lgstapp-@GST_API_VERSION@ \
+	    $(GST_LIBS) $(GIO_LIBS) $(LIBM)
+
+libgstrtspserver_@GST_API_VERSION@includedir = $(includedir)/gstreamer-@GST_API_VERSION@/gst/rtsp-server
+libgstrtspserver_@GST_API_VERSION@include_HEADERS = $(public_headers)
+
+CLEANFILES =
+
+if HAVE_INTROSPECTION
+BUILT_GIRSOURCES = GstRtspServer-@GST_API_VERSION@.gir
+
+gir_headers=$(patsubst %,$(srcdir)/%, $(libgstrtspserver_@GST_API_VERSION@include_HEADERS))
+gir_sources=$(patsubst %,$(srcdir)/%, $(libgstrtspserver_@GST_API_VERSION@_la_SOURCES))
+
+GstRtspServer-@GST_API_VERSION@.gir: $(INTROSPECTION_SCANNER) libgstrtspserver-@GST_API_VERSION@.la
+	$(AM_V_GEN)PKG_CONFIG_PATH="$(GST_PKG_CONFIG_PATH)" \
+		$(INTROSPECTION_SCANNER) -v --namespace GstRtspServer \
+		--nsversion=@GST_API_VERSION@ \
+		--strip-prefix=Gst \
+		--warn-all \
+		-I$(top_srcdir) \
+		-I$(top_builddir) \
+		-DIN_GOBJECT_INTROSPECTION=1 \
+		--c-include='gst/rtsp-server/rtsp-server.h' \
+		--add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-@GST_API_VERSION@` \
+		--add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-rtsp-@GST_API_VERSION@` \
+		--add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-sdp-@GST_API_VERSION@` \
+		--add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-net-@GST_API_VERSION@` \
+		--library=libgstrtspserver-@GST_API_VERSION@.la \
+		--include=Gst-@GST_API_VERSION@ \
+		--include=GstRtsp-@GST_API_VERSION@ \
+		--include=GstNet-@GST_API_VERSION@ \
+		--libtool="$(top_builddir)/libtool" \
+		--pkg gstreamer-@GST_API_VERSION@ \
+		--pkg gstreamer-rtsp-@GST_API_VERSION@ \
+		--pkg gstreamer-net-@GST_API_VERSION@ \
+		--pkg-export gstreamer-rtsp-server-@GST_API_VERSION@ \
+		--output $@ \
+		$(gir_headers) \
+		$(gir_sources)
+
+# INTROSPECTION_GIRDIR/INTROSPECTION_TYPELIBDIR aren't the right place to
+# install anything - we need to install inside our prefix.
+girdir = $(datadir)/gir-1.0
+gir_DATA = $(BUILT_GIRSOURCES)
+
+typelibsdir = $(libdir)/girepository-1.0/
+
+typelibs_DATA = $(BUILT_GIRSOURCES:.gir=.typelib)
+
+%.typelib: %.gir $(INTROSPECTION_COMPILER)
+	$(AM_V_GEN)PKG_CONFIG_PATH="$(GST_PKG_CONFIG_PATH)" \
+	$(INTROSPECTION_COMPILER) \
+	--includedir=$(srcdir) \
+	--includedir=$(builddir) \
+	--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-@GST_API_VERSION@` \
+	--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-rtsp-@GST_API_VERSION@` \
+	--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-sdp-@GST_API_VERSION@` \
+	--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-net-@GST_API_VERSION@` \
+	$(INTROSPECTION_COMPILER_OPTS) $< -o $(@F)
+
+CLEANFILES += $(BUILT_GIRSOURCES) $(typelibs_DATA)
+endif
diff --git a/gst/rtsp-server/Makefile.in b/gst/rtsp-server/Makefile.in
new file mode 100644
index 0000000..c7c73ae
--- /dev/null
+++ b/gst/rtsp-server/Makefile.in
@@ -0,0 +1,1065 @@
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+WARNING_CFLAGS = @WARNING_CFLAGS@
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+top_build_prefix = @top_build_prefix@
+top_builddir = @top_builddir@
+top_srcdir = @top_srcdir@
+public_headers = \
+		rtsp-auth.h \
+		rtsp-address-pool.h \
+		rtsp-context.h \
+		rtsp-params.h \
+		rtsp-sdp.h \
+		rtsp-thread-pool.h \
+		rtsp-media.h \
+		rtsp-media-factory.h \
+		rtsp-media-factory-uri.h \
+		rtsp-mount-points.h \
+		rtsp-permissions.h \
+		rtsp-stream.h \
+		rtsp-stream-transport.h \
+		rtsp-session.h \
+		rtsp-session-media.h \
+		rtsp-session-pool.h \
+		rtsp-token.h \
+		rtsp-client.h \
+		rtsp-server.h
+
+c_sources = \
+	rtsp-auth.c \
+	rtsp-address-pool.c \
+	rtsp-context.c \
+	rtsp-params.c \
+	rtsp-sdp.c \
+	rtsp-thread-pool.c \
+	rtsp-media.c \
+	rtsp-media-factory.c \
+	rtsp-media-factory-uri.c \
+	rtsp-mount-points.c \
+	rtsp-permissions.c \
+	rtsp-stream.c \
+	rtsp-stream-transport.c \
+	rtsp-session.c \
+	rtsp-session-media.c \
+	rtsp-session-pool.c \
+	rtsp-token.c \
+	rtsp-client.c \
+	rtsp-server.c
+
+noinst_HEADERS = 
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+	libgstrtspserver-@GST_API_VERSION@.la
+
+libgstrtspserver_@GST_API_VERSION@_la_SOURCES = \
+    $(c_sources)
+
+libgstrtspserver_@GST_API_VERSION@_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
+libgstrtspserver_@GST_API_VERSION@_la_LDFLAGS = $(GST_LIB_LDFLAGS) $(GST_ALL_LDFLAGS) $(GST_LT_LDFLAGS)
+libgstrtspserver_@GST_API_VERSION@_la_LIBADD = \
+	$(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) \
+    -lgstrtp-@GST_API_VERSION@ -lgstrtsp-@GST_API_VERSION@ \
+            -lgstnet-@GST_API_VERSION@ \
+            -lgstsdp-@GST_API_VERSION@ \
+            -lgstapp-@GST_API_VERSION@ \
+	    $(GST_LIBS) $(GIO_LIBS) $(LIBM)
+
+libgstrtspserver_@GST_API_VERSION@includedir = $(includedir)/gstreamer-@GST_API_VERSION@/gst/rtsp-server
+libgstrtspserver_@GST_API_VERSION@include_HEADERS = $(public_headers)
+CLEANFILES = $(am__append_1)
+@HAVE_INTROSPECTION_TRUE@BUILT_GIRSOURCES = GstRtspServer-@GST_API_VERSION@.gir
+@HAVE_INTROSPECTION_TRUE@gir_headers = $(patsubst %,$(srcdir)/%, $(libgstrtspserver_@GST_API_VERSION@include_HEADERS))
+@HAVE_INTROSPECTION_TRUE@gir_sources = $(patsubst %,$(srcdir)/%, $(libgstrtspserver_@GST_API_VERSION@_la_SOURCES))
+
+# INTROSPECTION_GIRDIR/INTROSPECTION_TYPELIBDIR aren't the right place to
+# install anything - we need to install inside our prefix.
+@HAVE_INTROSPECTION_TRUE@girdir = $(datadir)/gir-1.0
+@HAVE_INTROSPECTION_TRUE@gir_DATA = $(BUILT_GIRSOURCES)
+@HAVE_INTROSPECTION_TRUE@typelibsdir = $(libdir)/girepository-1.0/
+@HAVE_INTROSPECTION_TRUE@typelibs_DATA = $(BUILT_GIRSOURCES:.gir=.typelib)
+all: all-am
+
+.SUFFIXES:
+.SUFFIXES: .c .lo .o .obj
+$(srcdir)/Makefile.in: @MAINTAINER_MODE_TRUE@ $(srcdir)/Makefile.am  $(am__configure_deps)
+	@for dep in $?; do \
+	  case '$(am__configure_deps)' in \
+	    *$$dep*) \
+	      ( cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh ) \
+	        && { if test -f $@; then exit 0; else break; fi; }; \
+	      exit 1;; \
+	  esac; \
+	done; \
+	echo ' cd $(top_srcdir) && $(AUTOMAKE) --gnu gst/rtsp-server/Makefile'; \
+	$(am__cd) $(top_srcdir) && \
+	  $(AUTOMAKE) --gnu gst/rtsp-server/Makefile
+.PRECIOUS: Makefile
+Makefile: $(srcdir)/Makefile.in $(top_builddir)/config.status
+	@case '$?' in \
+	  *config.status*) \
+	    cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh;; \
+	  *) \
+	    echo ' cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@ $(am__depfiles_maybe)'; \
+	    cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@ $(am__depfiles_maybe);; \
+	esac;
+
+$(top_builddir)/config.status: $(top_srcdir)/configure $(CONFIG_STATUS_DEPENDENCIES)
+	cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
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+$(top_srcdir)/configure: @MAINTAINER_MODE_TRUE@ $(am__configure_deps)
+	cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
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+	cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
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+install-libLTLIBRARIES: $(lib_LTLIBRARIES)
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+	@list='$(lib_LTLIBRARIES)'; test -n "$(libdir)" || list=; \
+	list2=; for p in $$list; do \
+	  if test -f $$p; then \
+	    list2="$$list2 $$p"; \
+	  else :; fi; \
+	done; \
+	test -z "$$list2" || { \
+	  echo " $(MKDIR_P) '$(DESTDIR)$(libdir)'"; \
+	  $(MKDIR_P) "$(DESTDIR)$(libdir)" || exit 1; \
+	  echo " $(LIBTOOL) $(AM_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=install $(INSTALL) $(INSTALL_STRIP_FLAG) $$list2 '$(DESTDIR)$(libdir)'"; \
+	  $(LIBTOOL) $(AM_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=install $(INSTALL) $(INSTALL_STRIP_FLAG) $$list2 "$(DESTDIR)$(libdir)"; \
+	}
+
+uninstall-libLTLIBRARIES:
+	@$(NORMAL_UNINSTALL)
+	@list='$(lib_LTLIBRARIES)'; test -n "$(libdir)" || list=; \
+	for p in $$list; do \
+	  $(am__strip_dir) \
+	  echo " $(LIBTOOL) $(AM_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=uninstall rm -f '$(DESTDIR)$(libdir)/$$f'"; \
+	  $(LIBTOOL) $(AM_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=uninstall rm -f "$(DESTDIR)$(libdir)/$$f"; \
+	done
+
+clean-libLTLIBRARIES:
+	-test -z "$(lib_LTLIBRARIES)" || rm -f $(lib_LTLIBRARIES)
+	@list='$(lib_LTLIBRARIES)'; \
+	locs=`for p in $$list; do echo $$p; done | \
+	      sed 's|^[^/]*$$|.|; s|/[^/]*$$||; s|$$|/so_locations|' | \
+	      sort -u`; \
+	test -z "$$locs" || { \
+	  echo rm -f $${locs}; \
+	  rm -f $${locs}; \
+	}
+
+libgstrtspserver-@GST_API_VERSION@.la: $(libgstrtspserver_@GST_API_VERSION@_la_OBJECTS) $(libgstrtspserver_@GST_API_VERSION@_la_DEPENDENCIES) $(EXTRA_libgstrtspserver_@GST_API_VERSION@_la_DEPENDENCIES) 
+	$(AM_V_CCLD)$(libgstrtspserver_@GST_API_VERSION@_la_LINK) -rpath $(libdir) $(libgstrtspserver_@GST_API_VERSION@_la_OBJECTS) $(libgstrtspserver_@GST_API_VERSION@_la_LIBADD) $(LIBS)
+
+mostlyclean-compile:
+	-rm -f *.$(OBJEXT)
+
+distclean-compile:
+	-rm -f *.tab.c
+
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-address-pool.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-auth.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-client.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-context.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-media-factory-uri.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-media-factory.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-media.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-mount-points.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-params.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-permissions.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-sdp.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-server.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-session-media.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-session-pool.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-session.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-stream-transport.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-stream.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-thread-pool.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstrtspserver_@GST_API_VERSION@_la-rtsp-token.Plo@am__quote@
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+mostlyclean: mostlyclean-am
+
+mostlyclean-am: mostlyclean-compile mostlyclean-generic \
+	mostlyclean-libtool
+
+pdf: pdf-am
+
+pdf-am:
+
+ps: ps-am
+
+ps-am:
+
+uninstall-am: uninstall-girDATA uninstall-libLTLIBRARIES \
+	uninstall-libgstrtspserver_@GST_API_VERSION@includeHEADERS \
+	uninstall-typelibsDATA
+
+.MAKE: install-am install-strip
+
+.PHONY: all all-am check check-am clean clean-generic \
+	clean-libLTLIBRARIES clean-libtool cscopelist-am ctags-am \
+	distclean distclean-compile distclean-generic \
+	distclean-libtool distdir dvi dvi-am html html-am info info-am \
+	install install-am install-data install-data-am install-dvi \
+	install-dvi-am install-exec install-exec-am install-girDATA \
+	install-html install-html-am install-info install-info-am \
+	install-libLTLIBRARIES \
+	install-libgstrtspserver_@GST_API_VERSION@includeHEADERS \
+	install-man install-pdf install-pdf-am install-ps \
+	install-ps-am install-strip install-typelibsDATA installcheck \
+	installcheck-am installdirs maintainer-clean \
+	maintainer-clean-generic mostlyclean mostlyclean-compile \
+	mostlyclean-generic mostlyclean-libtool pdf pdf-am ps ps-am \
+	tags-am uninstall uninstall-am uninstall-girDATA \
+	uninstall-libLTLIBRARIES \
+	uninstall-libgstrtspserver_@GST_API_VERSION@includeHEADERS \
+	uninstall-typelibsDATA
+
+
+@HAVE_INTROSPECTION_TRUE@GstRtspServer-@GST_API_VERSION@.gir: $(INTROSPECTION_SCANNER) libgstrtspserver-@GST_API_VERSION@.la
+@HAVE_INTROSPECTION_TRUE@	$(AM_V_GEN)PKG_CONFIG_PATH="$(GST_PKG_CONFIG_PATH)" \
+@HAVE_INTROSPECTION_TRUE@		$(INTROSPECTION_SCANNER) -v --namespace GstRtspServer \
+@HAVE_INTROSPECTION_TRUE@		--nsversion=@GST_API_VERSION@ \
+@HAVE_INTROSPECTION_TRUE@		--strip-prefix=Gst \
+@HAVE_INTROSPECTION_TRUE@		--warn-all \
+@HAVE_INTROSPECTION_TRUE@		-I$(top_srcdir) \
+@HAVE_INTROSPECTION_TRUE@		-I$(top_builddir) \
+@HAVE_INTROSPECTION_TRUE@		-DIN_GOBJECT_INTROSPECTION=1 \
+@HAVE_INTROSPECTION_TRUE@		--c-include='gst/rtsp-server/rtsp-server.h' \
+@HAVE_INTROSPECTION_TRUE@		--add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-@GST_API_VERSION@` \
+@HAVE_INTROSPECTION_TRUE@		--add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-rtsp-@GST_API_VERSION@` \
+@HAVE_INTROSPECTION_TRUE@		--add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-sdp-@GST_API_VERSION@` \
+@HAVE_INTROSPECTION_TRUE@		--add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-net-@GST_API_VERSION@` \
+@HAVE_INTROSPECTION_TRUE@		--library=libgstrtspserver-@GST_API_VERSION@.la \
+@HAVE_INTROSPECTION_TRUE@		--include=Gst-@GST_API_VERSION@ \
+@HAVE_INTROSPECTION_TRUE@		--include=GstRtsp-@GST_API_VERSION@ \
+@HAVE_INTROSPECTION_TRUE@		--include=GstNet-@GST_API_VERSION@ \
+@HAVE_INTROSPECTION_TRUE@		--libtool="$(top_builddir)/libtool" \
+@HAVE_INTROSPECTION_TRUE@		--pkg gstreamer-@GST_API_VERSION@ \
+@HAVE_INTROSPECTION_TRUE@		--pkg gstreamer-rtsp-@GST_API_VERSION@ \
+@HAVE_INTROSPECTION_TRUE@		--pkg gstreamer-net-@GST_API_VERSION@ \
+@HAVE_INTROSPECTION_TRUE@		--pkg-export gstreamer-rtsp-server-@GST_API_VERSION@ \
+@HAVE_INTROSPECTION_TRUE@		--output $@ \
+@HAVE_INTROSPECTION_TRUE@		$(gir_headers) \
+@HAVE_INTROSPECTION_TRUE@		$(gir_sources)
+
+@HAVE_INTROSPECTION_TRUE@%.typelib: %.gir $(INTROSPECTION_COMPILER)
+@HAVE_INTROSPECTION_TRUE@	$(AM_V_GEN)PKG_CONFIG_PATH="$(GST_PKG_CONFIG_PATH)" \
+@HAVE_INTROSPECTION_TRUE@	$(INTROSPECTION_COMPILER) \
+@HAVE_INTROSPECTION_TRUE@	--includedir=$(srcdir) \
+@HAVE_INTROSPECTION_TRUE@	--includedir=$(builddir) \
+@HAVE_INTROSPECTION_TRUE@	--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-@GST_API_VERSION@` \
+@HAVE_INTROSPECTION_TRUE@	--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-rtsp-@GST_API_VERSION@` \
+@HAVE_INTROSPECTION_TRUE@	--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-sdp-@GST_API_VERSION@` \
+@HAVE_INTROSPECTION_TRUE@	--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-net-@GST_API_VERSION@` \
+@HAVE_INTROSPECTION_TRUE@	$(INTROSPECTION_COMPILER_OPTS) $< -o $(@F)
+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
diff --git a/gst/rtsp-server/gstwfdmessage.c b/gst/rtsp-server/gstwfdmessage.c
new file mode 100755
index 0000000..46301bd
--- /dev/null
+++ b/gst/rtsp-server/gstwfdmessage.c
@@ -0,0 +1,1963 @@
+/* GStreamer
+ * Copyright (C) <2005,2006> Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+/**
+ * SECTION:gstwfdmessage
+ * @short_description: Helper methods for dealing with WFD messages
+ *
+ * 
+ * 
+ * The GstWFDMessage helper functions makes it easy to parse and create WFD
+ * messages.
+ * 
+ * 
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include 
+#include 
+#include 
+
+#include 
+
+#include "gstwfdmessage.h"
+
+#define EDID_BLOCK_SIZE 128
+
+#define FREE_STRING(field)              g_free (field); (field) = NULL
+#define REPLACE_STRING(field, val)      FREE_STRING(field); (field) = g_strdup (val)
+
+#define INIT_ARRAY(field, type, init_func)              \
+G_STMT_START {                                          \
+  if (field) {                                          \
+    guint i;                                            \
+    for(i = 0; i < (field)->len; i++)                   \
+      init_func (&g_array_index ((field), type, i));    \
+    g_array_set_size ((field), 0);                      \
+  }                                                     \
+  else                                                  \
+    (field) = g_array_new (FALSE, TRUE, sizeof (type)); \
+} G_STMT_END
+
+#define FREE_ARRAY(field)         \
+G_STMT_START {                    \
+  if (field)                      \
+    g_array_free ((field), TRUE); \
+  (field) = NULL;                 \
+} G_STMT_END
+
+#define DEFINE_STRING_SETTER(field)                                     \
+GstWFDResult gst_wfd_message_set_##field (GstWFDMessage *msg, const gchar *val) { \
+  g_free (msg->field);                                                  \
+  msg->field = g_strdup (val);                                          \
+  return GST_WFD_OK;                                                    \
+}
+#define DEFINE_STRING_GETTER(field)                                     \
+const gchar* gst_wfd_message_get_##field (const GstWFDMessage *msg) {   \
+  return msg->field;                                                    \
+}
+
+#define DEFINE_ARRAY_LEN(field)                                         \
+guint gst_wfd_message_##field##_len (const GstWFDMessage *msg) {        \
+  return msg->field->len;                                               \
+}
+#define DEFINE_ARRAY_GETTER(method, field, type)                        \
+const type * gst_wfd_message_get_##method (const GstWFDMessage *msg, guint idx) {  \
+  return &g_array_index (msg->field, type, idx);                        \
+}
+#define DEFINE_PTR_ARRAY_GETTER(method, field, type)                    \
+const type gst_wfd_message_get_##method (const GstWFDMessage *msg, guint idx) {    \
+  return g_array_index (msg->field, type, idx);                         \
+}
+#define DEFINE_ARRAY_INSERT(method, field, intype, dup_method, type)         \
+GstWFDResult gst_wfd_message_insert_##method (GstWFDMessage *msg, gint idx, intype val) {   \
+  type vt;                                                              \
+  type* v = &vt;                                                         \
+  dup_method (v, val);                                                  \
+  if (idx == -1)                                                        \
+    g_array_append_val (msg->field, vt);                                \
+  else                                                                  \
+    g_array_insert_val (msg->field, idx, vt);                           \
+  return GST_WFD_OK;                                                    \
+}
+
+#define DEFINE_ARRAY_REPLACE(method, field, intype, free_method, dup_method, type)         \
+GstWFDResult gst_wfd_message_replace_##method (GstWFDMessage *msg, guint idx, intype val) {   \
+  type *v = &g_array_index (msg->field, type, idx);                   \
+  free_method (v);                                                    \
+  dup_method (v, val);                                                  \
+  return GST_WFD_OK;                                                    \
+}
+#define DEFINE_ARRAY_REMOVE(method, field, type, free_method)                        \
+GstWFDResult gst_wfd_message_remove_##method (GstWFDMessage *msg, guint idx) {  \
+  type *v = &g_array_index (msg->field, type, idx);                     \
+  free_method (v);                                                      \
+  g_array_remove_index (msg->field, idx);                               \
+  return GST_WFD_OK;                                                    \
+}
+#define DEFINE_ARRAY_ADDER(method, type)                                \
+GstWFDResult gst_wfd_message_add_##method (GstWFDMessage *msg, const type val) {   \
+  return gst_wfd_message_insert_##method (msg, -1, val);                \
+}
+
+#define dup_string(v,val) ((*v) = g_strdup (val))
+#define INIT_STR_ARRAY(field) \
+    INIT_ARRAY (field, gchar *, free_string)
+#define DEFINE_STR_ARRAY_GETTER(method, field) \
+    DEFINE_PTR_ARRAY_GETTER(method, field, gchar *)
+#define DEFINE_STR_ARRAY_INSERT(method, field) \
+    DEFINE_ARRAY_INSERT (method, field, const gchar *, dup_string, gchar *)
+#define DEFINE_STR_ARRAY_ADDER(method, field) \
+    DEFINE_ARRAY_ADDER (method, gchar *)
+#define DEFINE_STR_ARRAY_REPLACE(method, field) \
+    DEFINE_ARRAY_REPLACE (method, field, const gchar *, free_string, dup_string, gchar *)
+#define DEFINE_STR_ARRAY_REMOVE(method, field) \
+    DEFINE_ARRAY_REMOVE (method, field, gchar *, free_string)
+
+static GstWFDMessage *gst_wfd_message_boxed_copy (GstWFDMessage * orig);
+static void gst_wfd_message_boxed_free (GstWFDMessage * msg);
+
+G_DEFINE_BOXED_TYPE (GstWFDMessage, gst_wfd_message, gst_wfd_message_boxed_copy,
+    gst_wfd_message_boxed_free);
+
+static GstWFDMessage *
+gst_wfd_message_boxed_copy (GstWFDMessage * orig)
+{
+  GstWFDMessage *copy;
+
+  if (gst_wfd_message_copy (orig, ©) == GST_WFD_OK)
+    return copy;
+
+  return NULL;
+}
+
+static void
+gst_wfd_message_boxed_free (GstWFDMessage * msg)
+{
+  gst_wfd_message_free (msg);
+}
+
+/**
+ * gst_wfd_message_new:
+ * @msg: (out) (transfer full): pointer to new #GstWFDMessage
+ *
+ * Allocate a new GstWFDMessage and store the result in @msg.
+ *
+ * Returns: a #GstWFDResult.
+ */
+GstWFDResult
+gst_wfd_message_new (GstWFDMessage ** msg)
+{
+  GstWFDMessage *newmsg;
+
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+  newmsg = g_new0 (GstWFDMessage, 1);
+
+  *msg = newmsg;
+
+  return gst_wfd_message_init (newmsg);
+}
+
+/**
+ * gst_wfd_message_init:
+ * @msg: a #GstWFDMessage
+ *
+ * Initialize @msg so that its contents are as if it was freshly allocated
+ * with gst_wfd_message_new(). This function is mostly used to initialize a message
+ * allocated on the stack. gst_wfd_message_uninit() undoes this operation.
+ *
+ * When this function is invoked on newly allocated data (with malloc or on the
+ * stack), its contents should be set to 0 before calling this function.
+ *
+ * Returns: a #GstWFDResult.
+ */
+GstWFDResult
+gst_wfd_message_init (GstWFDMessage * msg)
+{
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+  return GST_WFD_OK;
+}
+
+/**
+ * gst_wfd_message_uninit:
+ * @msg: a #GstWFDMessage
+ *
+ * Free all resources allocated in @msg. @msg should not be used anymore after
+ * this function. This function should be used when @msg was allocated on the
+ * stack and initialized with gst_wfd_message_init().
+ *
+ * Returns: a #GstWFDResult.
+ */
+GstWFDResult
+gst_wfd_message_uninit (GstWFDMessage * msg)
+{
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+  if (msg->audio_codecs) {
+    guint i = 0;
+    if (msg->audio_codecs->list) {
+      for (; i < msg->audio_codecs->count; i++) {
+        FREE_STRING(msg->audio_codecs->list[i].audio_format);
+        msg->audio_codecs->list[i].modes = 0;
+        msg->audio_codecs->list[i].latency = 0;
+      }
+      FREE_STRING(msg->audio_codecs->list);
+    }
+    FREE_STRING(msg->audio_codecs);
+  }
+
+  if (msg->video_formats) {
+    FREE_STRING(msg->video_formats->list);
+    FREE_STRING(msg->video_formats);
+  }
+
+  if (msg->video_3d_formats) {
+    FREE_STRING(msg->video_3d_formats->list);
+    FREE_STRING(msg->video_3d_formats);
+  }
+
+  if (msg->content_protection) {
+    if (msg->content_protection->hdcp2_spec) {
+      FREE_STRING(msg->content_protection->hdcp2_spec->hdcpversion);
+      FREE_STRING(msg->content_protection->hdcp2_spec->TCPPort);
+      FREE_STRING(msg->content_protection->hdcp2_spec);
+    }
+    FREE_STRING(msg->content_protection);
+  }
+
+  if (msg->display_edid) {
+    if (msg->display_edid->edid_payload)
+      FREE_STRING(msg->display_edid->edid_payload);
+    FREE_STRING(msg->display_edid);
+  }
+
+  if (msg->coupled_sink) {
+    if (msg->coupled_sink->coupled_sink_cap) {
+      FREE_STRING(msg->coupled_sink->coupled_sink_cap->sink_address);
+      FREE_STRING(msg->coupled_sink->coupled_sink_cap);
+    }
+    FREE_STRING(msg->coupled_sink);
+  }
+
+  if (msg->trigger_method) {
+    FREE_STRING(msg->trigger_method->wfd_trigger_method);
+    FREE_STRING(msg->trigger_method);
+  }
+
+  if (msg->presentation_url) {
+    FREE_STRING(msg->presentation_url->wfd_url0);
+    FREE_STRING(msg->presentation_url->wfd_url1);
+    FREE_STRING(msg->presentation_url);
+  }
+
+  if (msg->client_rtp_ports) {
+    FREE_STRING(msg->client_rtp_ports->profile);
+    FREE_STRING(msg->client_rtp_ports->mode);
+    FREE_STRING(msg->client_rtp_ports);
+  }
+
+  if (msg->route) {
+    FREE_STRING(msg->route->destination);
+    FREE_STRING(msg->route);
+  }
+
+  if (msg->I2C) {
+    FREE_STRING(msg->I2C);
+  }
+
+  if (msg->av_format_change_timing) {
+    FREE_STRING(msg->av_format_change_timing);
+  }
+
+  if (msg->preferred_display_mode) {
+    FREE_STRING(msg->preferred_display_mode);
+  }
+
+  if (msg->standby_resume_capability) {
+    FREE_STRING(msg->standby_resume_capability);
+  }
+
+  if (msg->standby) {
+    FREE_STRING(msg->standby);
+  }
+
+  if (msg->connector_type) {
+    FREE_STRING(msg->connector_type);
+  }
+
+  if (msg->idr_request) {
+    FREE_STRING(msg->idr_request);
+  }
+
+  return GST_WFD_OK;
+}
+
+/**
+ * gst_wfd_message_copy:
+ * @msg: a #GstWFDMessage
+ * @copy: (out) (transfer full): pointer to new #GstWFDMessage
+ *
+ * Allocate a new copy of @msg and store the result in @copy. The value in
+ * @copy should be release with gst_wfd_message_free function.
+ *
+ * Returns: a #GstWFDResult
+ *
+ * Since: 1.6
+ */
+GstWFDResult
+gst_wfd_message_copy (const GstWFDMessage * msg, GstWFDMessage ** copy)
+{
+  GstWFDResult ret;
+  GstWFDMessage *cp;
+
+  if (msg == NULL)
+    return GST_WFD_EINVAL;
+
+  ret = gst_wfd_message_new (copy);
+  if (ret != GST_WFD_OK)
+    return ret;
+
+  cp = *copy;
+
+  /* TODO-WFD */
+  if (msg->client_rtp_ports) {
+    cp->client_rtp_ports = g_malloc (sizeof (GstWFDClientRtpPorts));
+    if (cp->client_rtp_ports) {
+      cp->client_rtp_ports->profile = g_strdup (msg->client_rtp_ports->profile);
+      cp->client_rtp_ports->rtp_port0 = msg->client_rtp_ports->rtp_port0;
+      cp->client_rtp_ports->rtp_port1 = msg->client_rtp_ports->rtp_port1;
+      cp->client_rtp_ports->mode = g_strdup (msg->client_rtp_ports->mode);
+    }
+  }
+
+  return GST_WFD_OK;
+}
+
+
+static void
+_read_string_space_ended (gchar * dest, guint size, gchar * src)
+{
+  guint idx = 0;
+
+  while (!g_ascii_isspace (*src) && *src != '\0') {
+    if (idx < size - 1)
+      dest[idx++] = *src;
+    src++;
+  }
+
+  if (size > 0)
+    dest[idx] = '\0';
+
+  return;
+}
+
+static void
+_read_string_attr_and_value (gchar * attr, gchar * value, guint tsize,
+    guint vsize, gchar del, gchar * src)
+{
+  guint idx;
+
+  idx = 0;
+
+  while (*src != del && *src != '\0') {
+    if (idx < tsize - 1)
+      attr[idx++] = *src;
+    src++;
+  }
+
+  if (tsize > 0)
+    attr[idx] = '\0';
+
+  src++;
+  idx = 0;
+
+  while (*src != '\0') {
+    if (idx < vsize - 1)
+      value[idx++] = *src;
+    src++;
+  }
+
+  if (vsize > 0)
+    value[idx] = '\0';
+
+  return;
+}
+
+static void
+gst_wfd_parse_attribute (gchar * buffer, GstWFDMessage * msg)
+{
+  gchar attr[8192] = { 0 };
+  gchar value[8192] = { 0 };
+  gchar temp[8192] = { 0 };
+  gchar *p = buffer;
+  gchar *v = value;
+
+#define WFD_SKIP_SPACE(q) if (*q && g_ascii_isspace (*q)) q++
+#define WFD_SKIP_EQUAL(q) if (*q && *q == '=') q++
+#define WFD_SKIP_COMMA(q) if (*q && g_ascii_ispunct (*q)) q++
+#define WFD_READ_STRING(field) _read_string_space_ended (temp, sizeof (temp), v); v+=strlen(temp); REPLACE_STRING (field, temp)
+#define WFD_READ_UINT32(field) _read_string_space_ended (temp, sizeof (temp), v); v+=strlen(temp); field = strtoul (temp, NULL, 16)
+#define WFD_READ_UINT32_DIGIT(field) _read_string_space_ended (temp, sizeof (temp), v); v+=strlen(temp); field = strtoul (temp, NULL, 10)
+
+  _read_string_attr_and_value (attr, value, sizeof (attr), sizeof (value), ':',
+      p);
+
+  if (!g_strcmp0 (attr, GST_STRING_WFD_AUDIO_CODECS)) {
+    msg->audio_codecs = g_new0 (GstWFDAudioCodeclist, 1);
+    if (strlen (v)) {
+      guint i = 0;
+      msg->audio_codecs->count = strlen (v) / 16;
+      msg->audio_codecs->list =
+          g_new0 (GstWFDAudioCodec, msg->audio_codecs->count);
+      for (; i < msg->audio_codecs->count; i++) {
+        WFD_SKIP_SPACE (v);
+        WFD_READ_STRING (msg->audio_codecs->list[i].audio_format);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->audio_codecs->list[i].modes);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->audio_codecs->list[i].latency);
+        WFD_SKIP_COMMA (v);
+      }
+    }
+  } else if (!g_strcmp0 (attr, GST_STRING_WFD_VIDEO_FORMATS)) {
+    msg->video_formats = g_new0 (GstWFDVideoCodeclist, 1);
+    if (strlen (v)) {
+      msg->video_formats->count = 1;
+      msg->video_formats->list = g_new0 (GstWFDVideoCodec, 1);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_formats->list->native);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_formats->
+          list->preferred_display_mode_supported);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_formats->list->H264_codec.profile);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_formats->list->H264_codec.level);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_formats->list->H264_codec.
+          misc_params.CEA_Support);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_formats->list->H264_codec.
+          misc_params.VESA_Support);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_formats->list->H264_codec.
+          misc_params.HH_Support);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_formats->list->H264_codec.
+          misc_params.latency);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_formats->list->H264_codec.
+          misc_params.min_slice_size);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_formats->list->H264_codec.
+          misc_params.slice_enc_params);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_formats->list->H264_codec.
+          misc_params.frame_rate_control_support);
+      WFD_SKIP_SPACE (v);
+      if (msg->video_formats->list->preferred_display_mode_supported == 1) {
+        WFD_READ_UINT32 (msg->video_formats->list->H264_codec.max_hres);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->video_formats->list->H264_codec.max_vres);
+        WFD_SKIP_SPACE (v);
+      }
+    }
+  } else if (!g_strcmp0 (attr, GST_STRING_WFD_3D_VIDEO_FORMATS)) {
+    msg->video_3d_formats = g_new0 (GstWFD3DFormats, 1);
+    if (strlen (v)) {
+      msg->video_3d_formats->count = 1;
+      msg->video_3d_formats->list = g_new0 (GstWFD3dCapList, 1);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_3d_formats->list->native);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_3d_formats->
+          list->preferred_display_mode_supported);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_3d_formats->list->H264_codec.profile);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_3d_formats->list->H264_codec.level);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_3d_formats->list->H264_codec.
+          misc_params.video_3d_capability);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_3d_formats->list->H264_codec.
+          misc_params.latency);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_3d_formats->list->H264_codec.
+          misc_params.min_slice_size);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_3d_formats->list->H264_codec.
+          misc_params.slice_enc_params);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->video_3d_formats->list->H264_codec.
+          misc_params.frame_rate_control_support);
+      WFD_SKIP_SPACE (v);
+      if (msg->video_formats->list->preferred_display_mode_supported == 1) {
+        WFD_READ_UINT32 (msg->video_formats->list->H264_codec.max_hres);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->video_formats->list->H264_codec.max_vres);
+        WFD_SKIP_SPACE (v);
+      }
+    }
+  } else if (!g_strcmp0 (attr, GST_STRING_WFD_CONTENT_PROTECTION)) {
+    msg->content_protection = g_new0 (GstWFDContentProtection, 1);
+    if (strlen (v)) {
+      WFD_SKIP_SPACE (v);
+      msg->content_protection->hdcp2_spec = g_new0 (GstWFDHdcp2Spec, 1);
+      if (strstr (v, "none")) {
+        msg->content_protection->hdcp2_spec->hdcpversion = g_strdup ("none");
+      } else {
+        WFD_READ_STRING (msg->content_protection->hdcp2_spec->hdcpversion);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_STRING (msg->content_protection->hdcp2_spec->TCPPort);
+      }
+    }
+  } else if (!g_strcmp0 (attr, GST_STRING_WFD_DISPLAY_EDID)) {
+    msg->display_edid = g_new0 (GstWFDDisplayEdid, 1);
+    if (strlen (v)) {
+      WFD_SKIP_SPACE (v);
+      if (strstr (v, "none")) {
+        msg->display_edid->edid_supported = 0;
+      } else {
+        msg->display_edid->edid_supported = 1;
+        WFD_READ_UINT32 (msg->display_edid->edid_block_count);
+        WFD_SKIP_SPACE (v);
+        if (msg->display_edid->edid_block_count) {
+          gchar *edid_string = v;
+          int i = 0, j = 0;
+          guint32 payload_size =
+              EDID_BLOCK_SIZE * msg->display_edid->edid_block_count;
+          msg->display_edid->edid_payload = g_malloc (payload_size);
+          for (;
+              i < (EDID_BLOCK_SIZE * msg->display_edid->edid_block_count * 2);
+              j++) {
+            int k = 0, kk = 0;
+            if (edid_string[i] > 0x29 && edid_string[i] < 0x40)
+              k = edid_string[i] - 48;
+            else if (edid_string[i] > 0x60 && edid_string[i] < 0x67)
+              k = edid_string[i] - 87;
+            else if (edid_string[i] > 0x40 && edid_string[i] < 0x47)
+              k = edid_string[i] - 55;
+
+            if (edid_string[i + 1] > 0x29 && edid_string[i + 1] < 0x40)
+              kk = edid_string[i + 1] - 48;
+            else if (edid_string[i + 1] > 0x60 && edid_string[i + 1] < 0x67)
+              kk = edid_string[i + 1] - 87;
+            else if (edid_string[i + 1] > 0x40 && edid_string[i + 1] < 0x47)
+              kk = edid_string[i + 1] - 55;
+
+            msg->display_edid->edid_payload[j] = (k << 4) | kk;
+            i += 2;
+          }
+          //memcpy(msg->display_edid->edid_payload, v, payload_size);
+          v += (payload_size * 2);
+        } else
+          v += strlen (v);
+      }
+    }
+  } else if (!g_strcmp0 (attr, GST_STRING_WFD_COUPLED_SINK)) {
+    msg->coupled_sink = g_new0 (GstWFDCoupledSink, 1);
+    if (strlen (v)) {
+      msg->coupled_sink->coupled_sink_cap = g_new0 (GstWFDCoupled_sink_cap, 1);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->coupled_sink->coupled_sink_cap->status);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_STRING (msg->coupled_sink->coupled_sink_cap->sink_address);
+    }
+  } else if (!g_strcmp0 (attr, GST_STRING_WFD_TRIGGER_METHOD)) {
+    msg->trigger_method = g_new0 (GstWFDTriggerMethod, 1);
+    if (strlen (v)) {
+      WFD_SKIP_SPACE (v);
+      WFD_READ_STRING (msg->trigger_method->wfd_trigger_method);
+    }
+  } else if (!g_strcmp0 (attr, GST_STRING_WFD_PRESENTATION_URL)) {
+    msg->presentation_url = g_new0 (GstWFDPresentationUrl, 1);
+    if (strlen (v)) {
+      WFD_SKIP_SPACE (v);
+      WFD_READ_STRING (msg->presentation_url->wfd_url0);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_STRING (msg->presentation_url->wfd_url1);
+    }
+  } else if (!g_strcmp0 (attr, GST_STRING_WFD_CLIENT_RTP_PORTS)) {
+    msg->client_rtp_ports = g_new0 (GstWFDClientRtpPorts, 1);
+    if (strlen (v)) {
+      WFD_SKIP_SPACE (v);
+      WFD_READ_STRING (msg->client_rtp_ports->profile);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32_DIGIT (msg->client_rtp_ports->rtp_port0);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32_DIGIT (msg->client_rtp_ports->rtp_port1);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_STRING (msg->client_rtp_ports->mode);
+    }
+  } else if (!g_strcmp0 (attr, GST_STRING_WFD_ROUTE)) {
+    msg->route = g_new0 (GstWFDRoute, 1);
+    if (strlen (v)) {
+      WFD_SKIP_SPACE (v);
+      WFD_READ_STRING (msg->route->destination);
+    }
+  } else if (!g_strcmp0 (attr, GST_STRING_WFD_I2C)) {
+    msg->I2C = g_new0 (GstWFDI2C, 1);
+    if (strlen (v)) {
+      msg->I2C->I2CPresent = TRUE;
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32_DIGIT (msg->I2C->I2C_port);
+      if (msg->I2C->I2C_port)
+        msg->I2C->I2CPresent = TRUE;
+    }
+  } else if (!g_strcmp0 (attr, GST_STRING_WFD_AV_FORMAT_CHANGE_TIMING)) {
+    msg->av_format_change_timing = g_new0 (GstWFDAVFormatChangeTiming, 1);
+    if (strlen (v)) {
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->av_format_change_timing->PTS);
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->av_format_change_timing->DTS);
+    }
+  } else if (!g_strcmp0 (attr, GST_STRING_WFD_PREFERRED_DISPLAY_MODE)) {
+    msg->preferred_display_mode = g_new0 (GstWFDPreferredDisplayMode, 1);
+    if (strlen (v)) {
+      WFD_SKIP_SPACE (v);
+      if (!strstr (v, "none")) {
+        msg->preferred_display_mode->displaymodesupported = FALSE;
+      } else {
+        WFD_READ_UINT32 (msg->preferred_display_mode->p_clock);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->H);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->HB);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->HSPOL_HSOFF);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->HSW);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->V);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->VB);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->VSPOL_VSOFF);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->VSW);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->VBS3D);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->V2d_s3d_modes);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->P_depth);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.profile);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.level);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.
+            misc_params.CEA_Support);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.
+            misc_params.VESA_Support);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.
+            misc_params.HH_Support);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.
+            misc_params.latency);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.
+            misc_params.min_slice_size);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.
+            misc_params.slice_enc_params);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.
+            misc_params.frame_rate_control_support);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.max_hres);
+        WFD_SKIP_SPACE (v);
+        WFD_READ_UINT32 (msg->preferred_display_mode->H264_codec.max_vres);
+        WFD_SKIP_SPACE (v);
+      }
+    }
+  } else if (!g_strcmp0 (attr, GST_STRING_WFD_STANDBY_RESUME_CAPABILITY)) {
+    msg->standby_resume_capability = g_new0 (GstWFDStandbyResumeCapability, 1);
+    if (strlen (v)) {
+      WFD_SKIP_SPACE (v);
+      if (!g_strcmp0 (v, "supported"))
+        msg->standby_resume_capability->standby_resume_cap = TRUE;
+      else
+        msg->standby_resume_capability->standby_resume_cap = FALSE;
+    }
+  } else if (!g_strcmp0 (attr, GST_STRING_WFD_STANDBY)) {
+    msg->standby = g_new0 (GstWFDStandby, 1);
+    msg->standby->wfd_standby = TRUE;
+  } else if (!g_strcmp0 (attr, GST_STRING_WFD_CONNECTOR_TYPE)) {
+    msg->connector_type = g_new0 (GstWFDConnectorType, 1);
+    if (strlen (v)) {
+      msg->connector_type->supported = TRUE;
+      WFD_SKIP_SPACE (v);
+      WFD_READ_UINT32 (msg->connector_type->connector_type);
+    }
+  } else if (!g_strcmp0 (attr, GST_STRING_WFD_IDR_REQUEST)) {
+    msg->idr_request = g_new0 (GstWFDIdrRequest, 1);
+    msg->idr_request->idr_request = TRUE;
+  }
+  return;
+}
+
+/**
+ * gst_wfd_message_parse_buffer:
+ * @data: the start of the buffer
+ * @size: the size of the buffer
+ * @msg: the result #GstSDPMessage
+ *
+ * Parse the contents of @size bytes pointed to by @data and store the result in
+ * @msg.
+ *
+ * Returns: #GST_SDP_OK on success.
+ */
+GstWFDResult
+gst_wfd_message_parse_buffer (const guint8 * data, guint size,
+    GstWFDMessage * msg)
+{
+  gchar *p;
+  gchar buffer[255] = { 0 };
+  guint idx = 0;
+
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+  g_return_val_if_fail (data != NULL, GST_WFD_EINVAL);
+  g_return_val_if_fail (size != 0, GST_WFD_EINVAL);
+
+  p = (gchar *) data;
+  while (TRUE) {
+
+    if (*p == '\0')
+      break;
+
+    idx = 0;
+    while (*p != '\n' && *p != '\r' && *p != '\0') {
+      if (idx < sizeof (buffer) - 1)
+        buffer[idx++] = *p;
+      p++;
+    }
+    buffer[idx] = '\0';
+    gst_wfd_parse_attribute (buffer, msg);
+
+    if (*p == '\0')
+      break;
+    p += 2;
+  }
+  return GST_WFD_OK;
+}
+
+/**
+ * gst_wfd_message_free:
+ * @msg: a #GstWFDMessage
+ *
+ * Free all resources allocated by @msg. @msg should not be used anymore after
+ * this function. This function should be used when @msg was dynamically
+ * allocated with gst_wfd_message_new().
+ *
+ * Returns: a #GstWFDResult.
+ */
+GstWFDResult
+gst_wfd_message_free (GstWFDMessage * msg)
+{
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+  gst_wfd_message_uninit (msg);
+  g_free (msg);
+
+  return GST_WFD_OK;
+}
+
+/**
+ * gst_wfd_message_as_text:
+ * @msg: a #GstWFDMessage
+ *
+ * Convert the contents of @msg to a text string.
+ *
+ * Returns: A dynamically allocated string representing the WFD description.
+ */
+gchar *
+gst_wfd_message_as_text (const GstWFDMessage * msg)
+{
+  /* change all vars so they match rfc? */
+  GString *lines;
+  guint i;
+
+  g_return_val_if_fail (msg != NULL, NULL);
+
+  lines = g_string_new ("");
+
+  /* list of audio codecs */
+  if (msg->audio_codecs) {
+    g_string_append_printf (lines, GST_STRING_WFD_AUDIO_CODECS);
+    if (msg->audio_codecs->list) {
+      g_string_append_printf (lines, ":");
+      for (i = 0; i < msg->audio_codecs->count; i++) {
+        g_string_append_printf (lines, " %s",
+            msg->audio_codecs->list[i].audio_format);
+        g_string_append_printf (lines, " %08x",
+            msg->audio_codecs->list[i].modes);
+        g_string_append_printf (lines, " %02x",
+            msg->audio_codecs->list[i].latency);
+        if ((i + 1) < msg->audio_codecs->count)
+          g_string_append_printf (lines, ",");
+      }
+    }
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  /* list of video codecs */
+  if (msg->video_formats) {
+    g_string_append_printf (lines, GST_STRING_WFD_VIDEO_FORMATS);
+    if (msg->video_formats->list) {
+      g_string_append_printf (lines, ":");
+      g_string_append_printf (lines, " %02x", msg->video_formats->list->native);
+      g_string_append_printf (lines, " %02x",
+          msg->video_formats->list->preferred_display_mode_supported);
+      g_string_append_printf (lines, " %02x",
+          msg->video_formats->list->H264_codec.profile);
+      g_string_append_printf (lines, " %02x",
+          msg->video_formats->list->H264_codec.level);
+      g_string_append_printf (lines, " %08x",
+          msg->video_formats->list->H264_codec.misc_params.CEA_Support);
+      g_string_append_printf (lines, " %08x",
+          msg->video_formats->list->H264_codec.misc_params.VESA_Support);
+      g_string_append_printf (lines, " %08x",
+          msg->video_formats->list->H264_codec.misc_params.HH_Support);
+      g_string_append_printf (lines, " %02x",
+          msg->video_formats->list->H264_codec.misc_params.latency);
+      g_string_append_printf (lines, " %04x",
+          msg->video_formats->list->H264_codec.misc_params.min_slice_size);
+      g_string_append_printf (lines, " %04x",
+          msg->video_formats->list->H264_codec.misc_params.slice_enc_params);
+      g_string_append_printf (lines, " %02x",
+          msg->video_formats->list->H264_codec.
+          misc_params.frame_rate_control_support);
+
+      if (msg->video_formats->list->H264_codec.max_hres)
+        g_string_append_printf (lines, " %04x",
+            msg->video_formats->list->H264_codec.max_hres);
+      else
+        g_string_append_printf (lines, " none");
+
+      if (msg->video_formats->list->H264_codec.max_vres)
+        g_string_append_printf (lines, " %04x",
+            msg->video_formats->list->H264_codec.max_vres);
+      else
+        g_string_append_printf (lines, " none");
+    }
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  /* list of video 3D codecs */
+  if (msg->video_3d_formats) {
+    g_string_append_printf (lines, GST_STRING_WFD_3D_VIDEO_FORMATS);
+    g_string_append_printf (lines, ":");
+    if (msg->video_3d_formats->list) {
+      g_string_append_printf (lines, " %02x",
+          msg->video_3d_formats->list->native);
+      g_string_append_printf (lines, " %02x",
+          msg->video_3d_formats->list->preferred_display_mode_supported);
+      g_string_append_printf (lines, " %02x",
+          msg->video_3d_formats->list->H264_codec.profile);
+      g_string_append_printf (lines, " %02x",
+          msg->video_3d_formats->list->H264_codec.level);
+      g_string_append_printf (lines, " %16x",
+          msg->video_3d_formats->list->H264_codec.
+          misc_params.video_3d_capability);
+      g_string_append_printf (lines, " %02x",
+          msg->video_3d_formats->list->H264_codec.misc_params.latency);
+      g_string_append_printf (lines, " %04x",
+          msg->video_3d_formats->list->H264_codec.misc_params.min_slice_size);
+      g_string_append_printf (lines, " %04x",
+          msg->video_3d_formats->list->H264_codec.misc_params.slice_enc_params);
+      g_string_append_printf (lines, " %02x",
+          msg->video_3d_formats->list->H264_codec.
+          misc_params.frame_rate_control_support);
+      if (msg->video_3d_formats->list->H264_codec.max_hres)
+        g_string_append_printf (lines, " %04x",
+            msg->video_3d_formats->list->H264_codec.max_hres);
+      else
+        g_string_append_printf (lines, " none");
+      if (msg->video_3d_formats->list->H264_codec.max_vres)
+        g_string_append_printf (lines, " %04x",
+            msg->video_3d_formats->list->H264_codec.max_vres);
+      else
+        g_string_append_printf (lines, " none");
+    } else {
+      g_string_append_printf (lines, " none");
+    }
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  if (msg->content_protection) {
+    g_string_append_printf (lines, GST_STRING_WFD_CONTENT_PROTECTION);
+    g_string_append_printf (lines, ":");
+    if (msg->content_protection->hdcp2_spec) {
+      if (msg->content_protection->hdcp2_spec->hdcpversion) {
+        g_string_append_printf (lines, " %s",
+            msg->content_protection->hdcp2_spec->hdcpversion);
+        g_string_append_printf (lines, " %s",
+            msg->content_protection->hdcp2_spec->TCPPort);
+      } else {
+        g_string_append_printf (lines, " none");
+      }
+    } else {
+      g_string_append_printf (lines, " none");
+    }
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  if (msg->display_edid) {
+    g_string_append_printf (lines, GST_STRING_WFD_DISPLAY_EDID);
+    g_string_append_printf (lines, ":");
+    if (msg->display_edid->edid_supported) {
+      g_string_append_printf (lines, " %d", msg->display_edid->edid_supported);
+      if (msg->display_edid->edid_block_count)
+        g_string_append_printf (lines, " %d",
+            msg->display_edid->edid_block_count);
+      else
+        g_string_append_printf (lines, " none");
+    } else {
+      g_string_append_printf (lines, " none");
+    }
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  if (msg->coupled_sink) {
+    g_string_append_printf (lines, GST_STRING_WFD_COUPLED_SINK);
+    g_string_append_printf (lines, ":");
+    if (msg->coupled_sink->coupled_sink_cap) {
+      g_string_append_printf (lines, " %02x",
+          msg->coupled_sink->coupled_sink_cap->status);
+      if (msg->coupled_sink->coupled_sink_cap->sink_address)
+        g_string_append_printf (lines, " %s",
+            msg->coupled_sink->coupled_sink_cap->sink_address);
+      else
+        g_string_append_printf (lines, " none");
+    } else {
+      g_string_append_printf (lines, " none");
+    }
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  if (msg->trigger_method) {
+    g_string_append_printf (lines, GST_STRING_WFD_TRIGGER_METHOD);
+    g_string_append_printf (lines, ":");
+    g_string_append_printf (lines, " %s",
+        msg->trigger_method->wfd_trigger_method);
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  if (msg->presentation_url) {
+    g_string_append_printf (lines, GST_STRING_WFD_PRESENTATION_URL);
+    g_string_append_printf (lines, ":");
+    if (msg->presentation_url->wfd_url0)
+      g_string_append_printf (lines, " %s", msg->presentation_url->wfd_url0);
+    else
+      g_string_append_printf (lines, " none");
+    if (msg->presentation_url->wfd_url1)
+      g_string_append_printf (lines, " %s", msg->presentation_url->wfd_url1);
+    else
+      g_string_append_printf (lines, " none");
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  if (msg->client_rtp_ports) {
+    g_string_append_printf (lines, GST_STRING_WFD_CLIENT_RTP_PORTS);
+    if (msg->client_rtp_ports->profile) {
+      g_string_append_printf (lines, ":");
+      g_string_append_printf (lines, " %s", msg->client_rtp_ports->profile);
+      g_string_append_printf (lines, " %d", msg->client_rtp_ports->rtp_port0);
+      g_string_append_printf (lines, " %d", msg->client_rtp_ports->rtp_port1);
+      g_string_append_printf (lines, " %s", msg->client_rtp_ports->mode);
+    }
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  if (msg->route) {
+    g_string_append_printf (lines, GST_STRING_WFD_ROUTE);
+    g_string_append_printf (lines, ":");
+    g_string_append_printf (lines, " %s", msg->route->destination);
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  if (msg->I2C) {
+    g_string_append_printf (lines, GST_STRING_WFD_I2C);
+    g_string_append_printf (lines, ":");
+    if (msg->I2C->I2CPresent)
+      g_string_append_printf (lines, " %x", msg->I2C->I2C_port);
+    else
+      g_string_append_printf (lines, " none");
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  if (msg->av_format_change_timing) {
+    g_string_append_printf (lines, GST_STRING_WFD_AV_FORMAT_CHANGE_TIMING);
+    g_string_append_printf (lines, ":");
+    g_string_append_printf (lines, " %010llx",
+        msg->av_format_change_timing->PTS);
+    g_string_append_printf (lines, " %010llx",
+        msg->av_format_change_timing->DTS);
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  if (msg->preferred_display_mode) {
+    g_string_append_printf (lines, GST_STRING_WFD_PREFERRED_DISPLAY_MODE);
+    g_string_append_printf (lines, ":");
+    if (msg->preferred_display_mode->displaymodesupported) {
+      g_string_append_printf (lines, " %06llx",
+          msg->preferred_display_mode->p_clock);
+      g_string_append_printf (lines, " %04x", msg->preferred_display_mode->H);
+      g_string_append_printf (lines, " %04x", msg->preferred_display_mode->HB);
+      g_string_append_printf (lines, " %04x",
+          msg->preferred_display_mode->HSPOL_HSOFF);
+      g_string_append_printf (lines, " %04x", msg->preferred_display_mode->HSW);
+      g_string_append_printf (lines, " %04x", msg->preferred_display_mode->V);
+      g_string_append_printf (lines, " %04x", msg->preferred_display_mode->VB);
+      g_string_append_printf (lines, " %04x",
+          msg->preferred_display_mode->VSPOL_VSOFF);
+      g_string_append_printf (lines, " %04x", msg->preferred_display_mode->VSW);
+      g_string_append_printf (lines, " %02x",
+          msg->preferred_display_mode->VBS3D);
+      g_string_append_printf (lines, " %02x",
+          msg->preferred_display_mode->V2d_s3d_modes);
+      g_string_append_printf (lines, " %02x",
+          msg->preferred_display_mode->P_depth);
+    } else
+      g_string_append_printf (lines, " none");
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  if (msg->standby_resume_capability) {
+    g_string_append_printf (lines, GST_STRING_WFD_STANDBY_RESUME_CAPABILITY);
+    g_string_append_printf (lines, ":");
+    if (msg->standby_resume_capability->standby_resume_cap)
+      g_string_append_printf (lines, " supported");
+    else
+      g_string_append_printf (lines, " none");
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  if (msg->standby) {
+    g_string_append_printf (lines, GST_STRING_WFD_STANDBY);
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  if (msg->connector_type) {
+    g_string_append_printf (lines, GST_STRING_WFD_CONNECTOR_TYPE);
+    g_string_append_printf (lines, ":");
+    if (msg->connector_type->connector_type)
+      g_string_append_printf (lines, " %02x",
+          msg->connector_type->connector_type);
+    else
+      g_string_append_printf (lines, " none");
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  if (msg->idr_request) {
+    g_string_append_printf (lines, GST_STRING_WFD_IDR_REQUEST);
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  return g_string_free (lines, FALSE);
+}
+
+gchar *
+gst_wfd_message_param_names_as_text (const GstWFDMessage * msg)
+{
+  /* change all vars so they match rfc? */
+  GString *lines;
+  g_return_val_if_fail (msg != NULL, NULL);
+
+  lines = g_string_new ("");
+
+  /* list of audio codecs */
+  if (msg->audio_codecs) {
+    g_string_append_printf (lines, GST_STRING_WFD_AUDIO_CODECS);
+    g_string_append_printf (lines, "\r\n");
+  }
+  /* list of video codecs */
+  if (msg->video_formats) {
+    g_string_append_printf (lines, GST_STRING_WFD_VIDEO_FORMATS);
+    g_string_append_printf (lines, "\r\n");
+  }
+  /* list of video 3D codecs */
+  if (msg->video_3d_formats) {
+    g_string_append_printf (lines, GST_STRING_WFD_3D_VIDEO_FORMATS);
+    g_string_append_printf (lines, "\r\n");
+  }
+  if (msg->content_protection) {
+    g_string_append_printf (lines, GST_STRING_WFD_CONTENT_PROTECTION);
+    g_string_append_printf (lines, "\r\n");
+  }
+  if (msg->display_edid) {
+    g_string_append_printf (lines, GST_STRING_WFD_DISPLAY_EDID);
+    g_string_append_printf (lines, "\r\n");
+  }
+  if (msg->coupled_sink) {
+    g_string_append_printf (lines, GST_STRING_WFD_COUPLED_SINK);
+    g_string_append_printf (lines, "\r\n");
+  }
+  if (msg->trigger_method) {
+    g_string_append_printf (lines, GST_STRING_WFD_TRIGGER_METHOD);
+    g_string_append_printf (lines, "\r\n");
+  }
+  if (msg->presentation_url) {
+    g_string_append_printf (lines, GST_STRING_WFD_PRESENTATION_URL);
+    g_string_append_printf (lines, "\r\n");
+  }
+  if (msg->client_rtp_ports) {
+    g_string_append_printf (lines, GST_STRING_WFD_CLIENT_RTP_PORTS);
+    g_string_append_printf (lines, "\r\n");
+  }
+  if (msg->route) {
+    g_string_append_printf (lines, GST_STRING_WFD_ROUTE);
+    g_string_append_printf (lines, "\r\n");
+  }
+  if (msg->I2C) {
+    g_string_append_printf (lines, GST_STRING_WFD_I2C);
+    g_string_append_printf (lines, "\r\n");
+  }
+  if (msg->av_format_change_timing) {
+    g_string_append_printf (lines, GST_STRING_WFD_AV_FORMAT_CHANGE_TIMING);
+    g_string_append_printf (lines, "\r\n");
+  }
+  if (msg->preferred_display_mode) {
+    g_string_append_printf (lines, GST_STRING_WFD_PREFERRED_DISPLAY_MODE);
+    g_string_append_printf (lines, "\r\n");
+  }
+  if (msg->standby_resume_capability) {
+    g_string_append_printf (lines, GST_STRING_WFD_STANDBY_RESUME_CAPABILITY);
+    g_string_append_printf (lines, "\r\n");
+  }
+  if (msg->standby) {
+    g_string_append_printf (lines, GST_STRING_WFD_STANDBY);
+    g_string_append_printf (lines, "\r\n");
+  }
+  if (msg->connector_type) {
+    g_string_append_printf (lines, GST_STRING_WFD_CONNECTOR_TYPE);
+    g_string_append_printf (lines, "\r\n");
+  }
+  if (msg->idr_request) {
+    g_string_append_printf (lines, GST_STRING_WFD_IDR_REQUEST);
+    g_string_append_printf (lines, "\r\n");
+  }
+
+  return g_string_free (lines, FALSE);
+}
+
+/**
+ * gst_wfd_message_dump:
+ * @msg: a #GstWFDMessage
+ *
+ * Dump the parsed contents of @msg to stdout.
+ *
+ * Returns: a #GstWFDResult.
+ */
+GstWFDResult
+gst_wfd_message_dump (const GstWFDMessage * msg)
+{
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+  if (msg->audio_codecs) {
+    guint i = 0;
+    g_print ("Audio supported formats : \n");
+    for (; i < msg->audio_codecs->count; i++) {
+      g_print ("Codec: %s\n", msg->audio_codecs->list[i].audio_format);
+      if (!strcmp (msg->audio_codecs->list[i].audio_format, "LPCM")) {
+        if (msg->audio_codecs->list[i].modes & GST_WFD_FREQ_44100)
+          g_print ("	Freq: %d\n", 44100);
+        if (msg->audio_codecs->list[i].modes & GST_WFD_FREQ_48000)
+          g_print ("	Freq: %d\n", 48000);
+        g_print ("	Channels: %d\n", 2);
+      }
+      if (!strcmp (msg->audio_codecs->list[i].audio_format, "AAC")) {
+        g_print ("	Freq: %d\n", 48000);
+        if (msg->audio_codecs->list[i].modes & GST_WFD_CHANNEL_2)
+          g_print ("	Channels: %d\n", 2);
+        if (msg->audio_codecs->list[i].modes & GST_WFD_CHANNEL_4)
+          g_print ("	Channels: %d\n", 4);
+        if (msg->audio_codecs->list[i].modes & GST_WFD_CHANNEL_6)
+          g_print ("	Channels: %d\n", 6);
+        if (msg->audio_codecs->list[i].modes & GST_WFD_CHANNEL_8)
+          g_print ("	Channels: %d\n", 8);
+      }
+      if (!strcmp (msg->audio_codecs->list[i].audio_format, "AC3")) {
+        g_print ("	Freq: %d\n", 48000);
+        if (msg->audio_codecs->list[i].modes & GST_WFD_CHANNEL_2)
+          g_print ("	Channels: %d\n", 2);
+        if (msg->audio_codecs->list[i].modes & GST_WFD_CHANNEL_4)
+          g_print ("	Channels: %d\n", 4);
+        if (msg->audio_codecs->list[i].modes & GST_WFD_CHANNEL_6)
+          g_print ("	Channels: %d\n", 6);
+      }
+      g_print ("	Bitwidth: %d\n", 16);
+      g_print ("	Latency: %d\n", msg->audio_codecs->list[i].latency);
+    }
+  }
+
+
+  if (msg->video_formats) {
+    g_print ("Video supported formats : \n");
+    if (msg->video_formats->list) {
+      guint nativeindex = 0;
+      g_print ("Codec: H264\n");
+      if ((msg->video_formats->list->native & 0x7) ==
+          GST_WFD_VIDEO_CEA_RESOLUTION) {
+        g_print ("	Native type: CEA\n");
+      } else if ((msg->video_formats->list->native & 0x7) ==
+          GST_WFD_VIDEO_VESA_RESOLUTION) {
+        g_print ("	Native type: VESA\n");
+      } else if ((msg->video_formats->list->native & 0x7) ==
+          GST_WFD_VIDEO_HH_RESOLUTION) {
+        g_print ("	Native type: HH\n");
+      }
+      nativeindex = msg->video_formats->list->native >> 3;
+      g_print ("	Resolution: %d\n", (1 << nativeindex));
+
+      if (msg->video_formats->list->
+          H264_codec.profile & GST_WFD_H264_BASE_PROFILE) {
+        g_print ("	Profile: BASE\n");
+      } else if (msg->video_formats->list->
+          H264_codec.profile & GST_WFD_H264_HIGH_PROFILE) {
+        g_print ("	Profile: HIGH\n");
+      }
+      if (msg->video_formats->list->H264_codec.level & GST_WFD_H264_LEVEL_3_1) {
+        g_print ("	Level: 3.1\n");
+      } else if (msg->video_formats->list->
+          H264_codec.level & GST_WFD_H264_LEVEL_3_2) {
+        g_print ("	Level: 3.2\n");
+      } else if (msg->video_formats->list->
+          H264_codec.level & GST_WFD_H264_LEVEL_4) {
+        g_print ("	Level: 4\n");
+      } else if (msg->video_formats->list->
+          H264_codec.level & GST_WFD_H264_LEVEL_4_1) {
+        g_print ("	Level: 4.1\n");
+      } else if (msg->video_formats->list->
+          H264_codec.level & GST_WFD_H264_LEVEL_4_2) {
+        g_print ("	Level: 4.2\n");
+      }
+      g_print ("	Latency: %d\n",
+          msg->video_formats->list->H264_codec.misc_params.latency);
+      g_print ("	min_slice_size: %x\n",
+          msg->video_formats->list->H264_codec.misc_params.min_slice_size);
+      g_print ("	slice_enc_params: %x\n",
+          msg->video_formats->list->H264_codec.misc_params.slice_enc_params);
+      g_print ("	frame_rate_control_support: %x\n",
+          msg->video_formats->list->H264_codec.
+          misc_params.frame_rate_control_support);
+      if (msg->video_formats->list->H264_codec.max_hres) {
+        g_print ("	Max Height: %04d\n",
+            msg->video_formats->list->H264_codec.max_hres);
+      }
+      if (msg->video_formats->list->H264_codec.max_vres) {
+        g_print ("	Max Width: %04d\n",
+            msg->video_formats->list->H264_codec.max_vres);
+      }
+    }
+  }
+
+  if (msg->video_3d_formats) {
+    g_print ("wfd_3d_formats");
+    g_print ("\r\n");
+  }
+
+  if (msg->content_protection) {
+    g_print (GST_STRING_WFD_CONTENT_PROTECTION);
+    g_print ("\r\n");
+  }
+
+  if (msg->display_edid) {
+    g_print (GST_STRING_WFD_DISPLAY_EDID);
+    g_print ("\r\n");
+  }
+
+  if (msg->coupled_sink) {
+    g_print (GST_STRING_WFD_COUPLED_SINK);
+    g_print ("\r\n");
+  }
+
+  if (msg->trigger_method) {
+    g_print ("	Trigger type: %s\n", msg->trigger_method->wfd_trigger_method);
+  }
+
+  if (msg->presentation_url) {
+    g_print (GST_STRING_WFD_PRESENTATION_URL);
+    g_print ("\r\n");
+  }
+
+  if (msg->client_rtp_ports) {
+    g_print (" Client RTP Ports : \n");
+    if (msg->client_rtp_ports->profile) {
+      g_print ("%s\n", msg->client_rtp_ports->profile);
+      g_print ("	%d\n", msg->client_rtp_ports->rtp_port0);
+      g_print ("	%d\n", msg->client_rtp_ports->rtp_port1);
+      g_print ("	%s\n", msg->client_rtp_ports->mode);
+    }
+    g_print ("\r\n");
+  }
+
+  if (msg->route) {
+    g_print (GST_STRING_WFD_ROUTE);
+    g_print ("\r\n");
+  }
+
+  if (msg->I2C) {
+    g_print (GST_STRING_WFD_I2C);
+    g_print ("\r\n");
+  }
+
+  if (msg->av_format_change_timing) {
+    g_print (GST_STRING_WFD_AV_FORMAT_CHANGE_TIMING);
+    g_print ("\r\n");
+  }
+
+  if (msg->preferred_display_mode) {
+    g_print (GST_STRING_WFD_PREFERRED_DISPLAY_MODE);
+    g_print ("\r\n");
+  }
+
+  if (msg->standby_resume_capability) {
+    g_print (GST_STRING_WFD_STANDBY_RESUME_CAPABILITY);
+    g_print ("\r\n");
+  }
+
+  if (msg->standby) {
+    g_print (GST_STRING_WFD_STANDBY);
+    g_print ("\r\n");
+  }
+
+  if (msg->connector_type) {
+    g_print (GST_STRING_WFD_CONNECTOR_TYPE);
+    g_print ("\r\n");
+  }
+
+  if (msg->idr_request) {
+    g_print (GST_STRING_WFD_IDR_REQUEST);
+    g_print ("\r\n");
+  }
+
+  return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_set_supported_audio_format (GstWFDMessage * msg,
+    GstWFDAudioFormats a_codec,
+    guint a_freq, guint a_channels, guint a_bitwidth, guint32 a_latency)
+{
+  guint temp = a_codec;
+  guint i = 0;
+  guint pcm = 0, aac = 0, ac3 = 0;
+
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+  if (!msg->audio_codecs)
+    msg->audio_codecs = g_new0 (GstWFDAudioCodeclist, 1);
+
+  if (a_codec != GST_WFD_AUDIO_UNKNOWN) {
+    while (temp) {
+      msg->audio_codecs->count++;
+      temp >>= 1;
+    }
+    msg->audio_codecs->list =
+        g_new0 (GstWFDAudioCodec, msg->audio_codecs->count);
+    for (; i < msg->audio_codecs->count; i++) {
+      if ((a_codec & GST_WFD_AUDIO_LPCM) && (!pcm)) {
+        msg->audio_codecs->list[i].audio_format = g_strdup ("LPCM");
+        msg->audio_codecs->list[i].modes = a_freq;
+        msg->audio_codecs->list[i].latency = a_latency;
+        pcm = 1;
+      } else if ((a_codec & GST_WFD_AUDIO_AAC) && (!aac)) {
+        msg->audio_codecs->list[i].audio_format = g_strdup ("AAC");
+        msg->audio_codecs->list[i].modes = a_channels;
+        msg->audio_codecs->list[i].latency = a_latency;
+        aac = 1;
+      } else if ((a_codec & GST_WFD_AUDIO_AC3) && (!ac3)) {
+        msg->audio_codecs->list[i].audio_format = g_strdup ("AC3");
+        msg->audio_codecs->list[i].modes = a_channels;
+        msg->audio_codecs->list[i].latency = a_latency;
+        ac3 = 1;
+      }
+    }
+  }
+  return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_set_prefered_audio_format (GstWFDMessage * msg,
+    GstWFDAudioFormats a_codec,
+    GstWFDAudioFreq a_freq,
+    GstWFDAudioChannels a_channels, guint a_bitwidth, guint32 a_latency)
+{
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+  if (!msg->audio_codecs)
+    msg->audio_codecs = g_new0 (GstWFDAudioCodeclist, 1);
+
+  msg->audio_codecs->list = g_new0 (GstWFDAudioCodec, 1);
+  msg->audio_codecs->count = 1;
+  if (a_codec == GST_WFD_AUDIO_LPCM) {
+    msg->audio_codecs->list->audio_format = g_strdup ("LPCM");
+    msg->audio_codecs->list->modes = a_freq;
+    msg->audio_codecs->list->latency = a_latency;
+  } else if (a_codec == GST_WFD_AUDIO_AAC) {
+    msg->audio_codecs->list->audio_format = g_strdup ("AAC");
+    msg->audio_codecs->list->modes = a_channels;
+    msg->audio_codecs->list->latency = a_latency;
+  } else if (a_codec == GST_WFD_AUDIO_AC3) {
+    msg->audio_codecs->list->audio_format = g_strdup ("AC3");
+    msg->audio_codecs->list->modes = a_channels;
+    msg->audio_codecs->list->latency = a_latency;
+  }
+  return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_supported_audio_format (GstWFDMessage * msg,
+    guint * a_codec,
+    guint * a_freq, guint * a_channels, guint * a_bitwidth, guint32 * a_latency)
+{
+  guint i = 0;
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+  g_return_val_if_fail (msg->audio_codecs != NULL, GST_WFD_EINVAL);
+
+  for (; i < msg->audio_codecs->count; i++) {
+    if (!g_strcmp0 (msg->audio_codecs->list[i].audio_format, "LPCM")) {
+      *a_codec |= GST_WFD_AUDIO_LPCM;
+      *a_freq |= msg->audio_codecs->list[i].modes;
+      *a_channels |= GST_WFD_CHANNEL_2;
+      *a_bitwidth = 16;
+      *a_latency = msg->audio_codecs->list[i].latency;
+    } else if (!g_strcmp0 (msg->audio_codecs->list[i].audio_format, "AAC")) {
+      *a_codec |= GST_WFD_AUDIO_AAC;
+      *a_freq |= GST_WFD_FREQ_48000;
+      *a_channels |= msg->audio_codecs->list[i].modes;
+      *a_bitwidth = 16;
+      *a_latency = msg->audio_codecs->list[i].latency;
+    } else if (!g_strcmp0 (msg->audio_codecs->list[i].audio_format, "AC3")) {
+      *a_codec |= GST_WFD_AUDIO_AC3;
+      *a_freq |= GST_WFD_FREQ_48000;
+      *a_channels |= msg->audio_codecs->list[i].modes;
+      *a_bitwidth = 16;
+      *a_latency = msg->audio_codecs->list[i].latency;
+    }
+  }
+  return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_prefered_audio_format (GstWFDMessage * msg,
+    GstWFDAudioFormats * a_codec,
+    GstWFDAudioFreq * a_freq,
+    GstWFDAudioChannels * a_channels, guint * a_bitwidth, guint32 * a_latency)
+{
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+  if (!g_strcmp0 (msg->audio_codecs->list->audio_format, "LPCM")) {
+    *a_codec = GST_WFD_AUDIO_LPCM;
+    *a_freq = msg->audio_codecs->list->modes;
+    *a_channels = GST_WFD_CHANNEL_2;
+    *a_bitwidth = 16;
+    *a_latency = msg->audio_codecs->list->latency;
+  } else if (!g_strcmp0 (msg->audio_codecs->list->audio_format, "AAC")) {
+    *a_codec = GST_WFD_AUDIO_AAC;
+    *a_freq = GST_WFD_FREQ_48000;
+    *a_channels = msg->audio_codecs->list->modes;
+    *a_bitwidth = 16;
+    *a_latency = msg->audio_codecs->list->latency;
+  } else if (!g_strcmp0 (msg->audio_codecs->list->audio_format, "AC3")) {
+    *a_codec = GST_WFD_AUDIO_AC3;
+    *a_freq = GST_WFD_FREQ_48000;
+    *a_channels = msg->audio_codecs->list->modes;
+    *a_bitwidth = 16;
+    *a_latency = msg->audio_codecs->list->latency;
+  }
+  return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_set_supported_video_format (GstWFDMessage * msg,
+    GstWFDVideoCodecs v_codec,
+    GstWFDVideoNativeResolution v_native,
+    guint64 v_native_resolution,
+    guint64 v_cea_resolution,
+    guint64 v_vesa_resolution,
+    guint64 v_hh_resolution,
+    guint v_profile,
+    guint v_level,
+    guint32 v_latency,
+    guint32 v_max_height,
+    guint32 v_max_width,
+    guint32 min_slice_size, guint32 slice_enc_params, guint frame_rate_control)
+{
+  guint nativeindex = 0;
+  guint64 temp = v_native_resolution;
+
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+  if (!msg->video_formats)
+    msg->video_formats = g_new0 (GstWFDVideoCodeclist, 1);
+
+  if (v_codec != GST_WFD_VIDEO_UNKNOWN) {
+    msg->video_formats->list = g_new0 (GstWFDVideoCodec, 1);
+    while (temp) {
+      nativeindex++;
+      temp >>= 1;
+    }
+
+    msg->video_formats->list->native = nativeindex - 1;
+    msg->video_formats->list->native <<= 3;
+
+    if (v_native == GST_WFD_VIDEO_VESA_RESOLUTION)
+      msg->video_formats->list->native |= 1;
+    else if (v_native == GST_WFD_VIDEO_HH_RESOLUTION)
+      msg->video_formats->list->native |= 2;
+
+    msg->video_formats->list->preferred_display_mode_supported = 1;
+    msg->video_formats->list->H264_codec.profile = v_profile;
+    msg->video_formats->list->H264_codec.level = v_level;
+    msg->video_formats->list->H264_codec.max_hres = v_max_height;
+    msg->video_formats->list->H264_codec.max_vres = v_max_width;
+    msg->video_formats->list->H264_codec.misc_params.CEA_Support =
+        v_cea_resolution;
+    msg->video_formats->list->H264_codec.misc_params.VESA_Support =
+        v_vesa_resolution;
+    msg->video_formats->list->H264_codec.misc_params.HH_Support =
+        v_hh_resolution;
+    msg->video_formats->list->H264_codec.misc_params.latency = v_latency;
+    msg->video_formats->list->H264_codec.misc_params.min_slice_size =
+        min_slice_size;
+    msg->video_formats->list->H264_codec.misc_params.slice_enc_params =
+        slice_enc_params;
+    msg->video_formats->list->H264_codec.
+        misc_params.frame_rate_control_support = frame_rate_control;
+  }
+  return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_set_prefered_video_format (GstWFDMessage * msg,
+    GstWFDVideoCodecs v_codec,
+    GstWFDVideoNativeResolution v_native,
+    guint64 v_native_resolution,
+    GstWFDVideoCEAResolution v_cea_resolution,
+    GstWFDVideoVESAResolution v_vesa_resolution,
+    GstWFDVideoHHResolution v_hh_resolution,
+    GstWFDVideoH264Profile v_profile,
+    GstWFDVideoH264Level v_level,
+    guint32 v_latency,
+    guint32 v_max_height,
+    guint32 v_max_width,
+    guint32 min_slice_size, guint32 slice_enc_params, guint frame_rate_control)
+{
+  guint nativeindex = 0;
+  guint64 temp = v_native_resolution;
+
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+  if (!msg->video_formats)
+    msg->video_formats = g_new0 (GstWFDVideoCodeclist, 1);
+  msg->video_formats->list = g_new0 (GstWFDVideoCodec, 1);
+
+  while (temp) {
+    nativeindex++;
+    temp >>= 1;
+  }
+
+  if (nativeindex)
+    msg->video_formats->list->native = nativeindex - 1;
+  msg->video_formats->list->native <<= 3;
+
+  if (v_native == GST_WFD_VIDEO_VESA_RESOLUTION)
+    msg->video_formats->list->native |= 1;
+  else if (v_native == GST_WFD_VIDEO_HH_RESOLUTION)
+    msg->video_formats->list->native |= 2;
+
+  msg->video_formats->list->preferred_display_mode_supported = 0;
+  msg->video_formats->list->H264_codec.profile = v_profile;
+  msg->video_formats->list->H264_codec.level = v_level;
+  msg->video_formats->list->H264_codec.max_hres = v_max_height;
+  msg->video_formats->list->H264_codec.max_vres = v_max_width;
+  msg->video_formats->list->H264_codec.misc_params.CEA_Support =
+      v_cea_resolution;
+  msg->video_formats->list->H264_codec.misc_params.VESA_Support =
+      v_vesa_resolution;
+  msg->video_formats->list->H264_codec.misc_params.HH_Support = v_hh_resolution;
+  msg->video_formats->list->H264_codec.misc_params.latency = v_latency;
+  msg->video_formats->list->H264_codec.misc_params.min_slice_size =
+      min_slice_size;
+  msg->video_formats->list->H264_codec.misc_params.slice_enc_params =
+      slice_enc_params;
+  msg->video_formats->list->H264_codec.misc_params.frame_rate_control_support =
+      frame_rate_control;
+  return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_supported_video_format (GstWFDMessage * msg,
+    GstWFDVideoCodecs * v_codec,
+    GstWFDVideoNativeResolution * v_native,
+    guint64 * v_native_resolution,
+    guint64 * v_cea_resolution,
+    guint64 * v_vesa_resolution,
+    guint64 * v_hh_resolution,
+    guint * v_profile,
+    guint * v_level,
+    guint32 * v_latency,
+    guint32 * v_max_height,
+    guint32 * v_max_width,
+    guint32 * min_slice_size,
+    guint32 * slice_enc_params, guint * frame_rate_control)
+{
+  guint nativeindex = 0;
+
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+  *v_codec = GST_WFD_VIDEO_H264;
+  *v_native = msg->video_formats->list->native & 0x7;
+  nativeindex = msg->video_formats->list->native >> 3;
+  *v_native_resolution = ((guint64) 1) << nativeindex;
+  *v_profile = msg->video_formats->list->H264_codec.profile;
+  *v_level = msg->video_formats->list->H264_codec.level;
+  *v_max_height = msg->video_formats->list->H264_codec.max_hres;
+  *v_max_width = msg->video_formats->list->H264_codec.max_vres;
+  *v_cea_resolution =
+      msg->video_formats->list->H264_codec.misc_params.CEA_Support;
+  *v_vesa_resolution =
+      msg->video_formats->list->H264_codec.misc_params.VESA_Support;
+  *v_hh_resolution =
+      msg->video_formats->list->H264_codec.misc_params.HH_Support;
+  *v_latency = msg->video_formats->list->H264_codec.misc_params.latency;
+  *min_slice_size =
+      msg->video_formats->list->H264_codec.misc_params.min_slice_size;
+  *slice_enc_params =
+      msg->video_formats->list->H264_codec.misc_params.slice_enc_params;
+  *frame_rate_control =
+      msg->video_formats->list->H264_codec.
+      misc_params.frame_rate_control_support;
+  return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_prefered_video_format (GstWFDMessage * msg,
+    GstWFDVideoCodecs * v_codec,
+    GstWFDVideoNativeResolution * v_native,
+    guint64 * v_native_resolution,
+    GstWFDVideoCEAResolution * v_cea_resolution,
+    GstWFDVideoVESAResolution * v_vesa_resolution,
+    GstWFDVideoHHResolution * v_hh_resolution,
+    GstWFDVideoH264Profile * v_profile,
+    GstWFDVideoH264Level * v_level,
+    guint32 * v_latency,
+    guint32 * v_max_height,
+    guint32 * v_max_width,
+    guint32 * min_slice_size,
+    guint32 * slice_enc_params, guint * frame_rate_control)
+{
+  guint nativeindex = 0;
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+  *v_codec = GST_WFD_VIDEO_H264;
+  *v_native = msg->video_formats->list->native & 0x7;
+  nativeindex = msg->video_formats->list->native >> 3;
+  *v_native_resolution = ((guint64) 1) << nativeindex;
+  *v_profile = msg->video_formats->list->H264_codec.profile;
+  *v_level = msg->video_formats->list->H264_codec.level;
+  *v_max_height = msg->video_formats->list->H264_codec.max_hres;
+  *v_max_width = msg->video_formats->list->H264_codec.max_vres;
+  *v_cea_resolution =
+      msg->video_formats->list->H264_codec.misc_params.CEA_Support;
+  *v_vesa_resolution =
+      msg->video_formats->list->H264_codec.misc_params.VESA_Support;
+  *v_hh_resolution =
+      msg->video_formats->list->H264_codec.misc_params.HH_Support;
+  *v_latency = msg->video_formats->list->H264_codec.misc_params.latency;
+  *min_slice_size =
+      msg->video_formats->list->H264_codec.misc_params.min_slice_size;
+  *slice_enc_params =
+      msg->video_formats->list->H264_codec.misc_params.slice_enc_params;
+  *frame_rate_control =
+      msg->video_formats->list->H264_codec.
+      misc_params.frame_rate_control_support;
+  return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_set_display_edid (GstWFDMessage * msg,
+    gboolean edid_supported, guint32 edid_blockcount, gchar * edid_playload)
+{
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+  if (!msg->display_edid)
+    msg->display_edid = g_new0 (GstWFDDisplayEdid, 1);
+  msg->display_edid->edid_supported = edid_supported;
+  if (!edid_supported)
+    return GST_WFD_OK;
+  msg->display_edid->edid_block_count = edid_blockcount;
+  if (edid_blockcount) {
+    msg->display_edid->edid_payload = g_malloc (128 * edid_blockcount);
+    if (!msg->display_edid->edid_payload)
+      memcpy (msg->display_edid->edid_payload, edid_playload,
+          128 * edid_blockcount);
+  } else
+    msg->display_edid->edid_payload = g_strdup ("none");
+  return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_display_edid (GstWFDMessage * msg,
+    gboolean * edid_supported,
+    guint32 * edid_blockcount, gchar ** edid_playload)
+{
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+  if (msg->display_edid) {
+    if (msg->display_edid->edid_supported) {
+      *edid_blockcount = msg->display_edid->edid_block_count;
+      if (msg->display_edid->edid_block_count) {
+        char *temp;
+        temp = g_malloc (EDID_BLOCK_SIZE * msg->display_edid->edid_block_count);
+        if (temp) {
+          memset (temp, 0,
+              EDID_BLOCK_SIZE * msg->display_edid->edid_block_count);
+          memcpy (temp, msg->display_edid->edid_payload,
+              EDID_BLOCK_SIZE * msg->display_edid->edid_block_count);
+          *edid_playload = temp;
+          *edid_supported = TRUE;
+        }
+      } else
+        *edid_playload = g_strdup ("none");
+    }
+  } else
+    *edid_supported = FALSE;
+  return GST_WFD_OK;
+}
+
+
+GstWFDResult
+gst_wfd_message_set_contentprotection_type (GstWFDMessage * msg,
+    GstWFDHDCPProtection hdcpversion, guint32 TCPPort)
+{
+  char str[11] = { 0, };
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+  if (!msg->content_protection)
+    msg->content_protection = g_new0 (GstWFDContentProtection, 1);
+  if (hdcpversion == GST_WFD_HDCP_NONE)
+    return GST_WFD_OK;
+  msg->content_protection->hdcp2_spec = g_new0 (GstWFDHdcp2Spec, 1);
+  if (hdcpversion == GST_WFD_HDCP_2_0)
+    msg->content_protection->hdcp2_spec->hdcpversion = g_strdup ("HDCP2.0");
+  else if (hdcpversion == GST_WFD_HDCP_2_1)
+    msg->content_protection->hdcp2_spec->hdcpversion = g_strdup ("HDCP2.1");
+  snprintf (str, sizeof (str), "port=%d", TCPPort);
+  msg->content_protection->hdcp2_spec->TCPPort = g_strdup (str);
+  return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_contentprotection_type (GstWFDMessage * msg,
+    GstWFDHDCPProtection * hdcpversion, guint32 * TCPPort)
+{
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+  if (msg->content_protection && msg->content_protection->hdcp2_spec) {
+    char *result = NULL;
+    char *ptr = NULL;
+    if (!g_strcmp0 (msg->content_protection->hdcp2_spec->hdcpversion, "none")) {
+      *hdcpversion = GST_WFD_HDCP_NONE;
+      *TCPPort = 0;
+      return GST_WFD_OK;
+    }
+    if (!g_strcmp0 (msg->content_protection->hdcp2_spec->hdcpversion,
+            "HDCP2.0"))
+      *hdcpversion = GST_WFD_HDCP_2_0;
+    else if (!g_strcmp0 (msg->content_protection->hdcp2_spec->hdcpversion,
+            "HDCP2.1"))
+      *hdcpversion = GST_WFD_HDCP_2_1;
+    else {
+      *hdcpversion = GST_WFD_HDCP_NONE;
+      *TCPPort = 0;
+      return GST_WFD_OK;
+    }
+
+    result = strtok_r (msg->content_protection->hdcp2_spec->TCPPort, "=", &ptr);
+    while (result != NULL) {
+      result = strtok_r (NULL, "=", &ptr);
+      *TCPPort = atoi (result);
+      break;
+    }
+  } else
+    *hdcpversion = GST_WFD_HDCP_NONE;
+  return GST_WFD_OK;
+}
+
+
+GstWFDResult
+gst_wfd_messge_set_prefered_rtp_ports (GstWFDMessage * msg,
+    GstWFDRTSPTransMode trans,
+    GstWFDRTSPProfile profile,
+    GstWFDRTSPLowerTrans lowertrans, guint32 rtp_port0, guint32 rtp_port1)
+{
+  GString *lines;
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+  if (!msg->client_rtp_ports)
+    msg->client_rtp_ports = g_new0 (GstWFDClientRtpPorts, 1);
+
+  if (trans != GST_WFD_RTSP_TRANS_UNKNOWN) {
+    lines = g_string_new ("");
+    if (trans == GST_WFD_RTSP_TRANS_RTP)
+      g_string_append_printf (lines, "RTP");
+    else if (trans == GST_WFD_RTSP_TRANS_RDT)
+      g_string_append_printf (lines, "RDT");
+
+    if (profile == GST_WFD_RTSP_PROFILE_AVP)
+      g_string_append_printf (lines, "/AVP");
+    else if (profile == GST_WFD_RTSP_PROFILE_SAVP)
+      g_string_append_printf (lines, "/SAVP");
+
+    if (lowertrans == GST_WFD_RTSP_LOWER_TRANS_UDP)
+      g_string_append_printf (lines, "/UDP;unicast");
+    else if (lowertrans == GST_WFD_RTSP_LOWER_TRANS_UDP_MCAST)
+      g_string_append_printf (lines, "/UDP;multicast");
+    else if (lowertrans == GST_WFD_RTSP_LOWER_TRANS_TCP)
+      g_string_append_printf (lines, "/TCP;unicast");
+    else if (lowertrans == GST_WFD_RTSP_LOWER_TRANS_HTTP)
+      g_string_append_printf (lines, "/HTTP");
+
+    msg->client_rtp_ports->profile = g_string_free (lines, FALSE);
+    msg->client_rtp_ports->rtp_port0 = rtp_port0;
+    msg->client_rtp_ports->rtp_port1 = rtp_port1;
+    msg->client_rtp_ports->mode = g_strdup ("mode=play");
+  }
+  return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_prefered_rtp_ports (GstWFDMessage * msg,
+    GstWFDRTSPTransMode * trans,
+    GstWFDRTSPProfile * profile,
+    GstWFDRTSPLowerTrans * lowertrans, guint32 * rtp_port0, guint32 * rtp_port1)
+{
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+  g_return_val_if_fail (msg->client_rtp_ports != NULL, GST_WFD_EINVAL);
+
+  if (g_strrstr (msg->client_rtp_ports->profile, "RTP"))
+    *trans = GST_WFD_RTSP_TRANS_RTP;
+  if (g_strrstr (msg->client_rtp_ports->profile, "RDT"))
+    *trans = GST_WFD_RTSP_TRANS_RDT;
+  if (g_strrstr (msg->client_rtp_ports->profile, "AVP"))
+    *profile = GST_WFD_RTSP_PROFILE_AVP;
+  if (g_strrstr (msg->client_rtp_ports->profile, "SAVP"))
+    *profile = GST_WFD_RTSP_PROFILE_SAVP;
+  if (g_strrstr (msg->client_rtp_ports->profile, "UDP;unicast"))
+    *lowertrans = GST_WFD_RTSP_LOWER_TRANS_UDP;
+  if (g_strrstr (msg->client_rtp_ports->profile, "UDP;multicast"))
+    *lowertrans = GST_WFD_RTSP_LOWER_TRANS_UDP_MCAST;
+  if (g_strrstr (msg->client_rtp_ports->profile, "TCP;unicast"))
+    *lowertrans = GST_WFD_RTSP_LOWER_TRANS_TCP;
+  if (g_strrstr (msg->client_rtp_ports->profile, "HTTP"))
+    *lowertrans = GST_WFD_RTSP_LOWER_TRANS_HTTP;
+
+  *rtp_port0 = msg->client_rtp_ports->rtp_port0;
+  *rtp_port1 = msg->client_rtp_ports->rtp_port1;
+
+  return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_set_presentation_url (GstWFDMessage * msg, gchar * wfd_url0,
+    gchar * wfd_url1)
+{
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+  if (!msg->presentation_url)
+    msg->presentation_url = g_new0 (GstWFDPresentationUrl, 1);
+  if (wfd_url0)
+    msg->presentation_url->wfd_url0 = g_strdup (wfd_url0);
+  if (wfd_url1)
+    msg->presentation_url->wfd_url1 = g_strdup (wfd_url1);
+  return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_presentation_url (GstWFDMessage * msg, gchar ** wfd_url0,
+    gchar ** wfd_url1)
+{
+  g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
+
+  if (msg->presentation_url) {
+    *wfd_url0 = g_strdup (msg->presentation_url->wfd_url0);
+    *wfd_url1 = g_strdup (msg->presentation_url->wfd_url1);
+  }
+  return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_set_av_format_change_timing(GstWFDMessage *msg, guint64 PTS, guint64 DTS)
+{
+  g_return_val_if_fail(msg != NULL, GST_WFD_EINVAL);
+
+  if (!msg->av_format_change_timing)
+    msg->av_format_change_timing = g_new0(GstWFDAVFormatChangeTiming, 1);
+
+  msg->av_format_change_timing->PTS = PTS;
+  msg->av_format_change_timing->DTS = DTS;
+  return GST_WFD_OK;
+}
+
+GstWFDResult
+gst_wfd_message_get_av_format_change_timing(GstWFDMessage *msg, guint64 *PTS, guint64 *DTS)
+{
+  g_return_val_if_fail(msg != NULL, GST_WFD_EINVAL);
+  g_return_val_if_fail(PTS != NULL, GST_WFD_EINVAL);
+  g_return_val_if_fail(DTS != NULL, GST_WFD_EINVAL);
+
+  if (msg->av_format_change_timing) {
+    *PTS = msg->av_format_change_timing->PTS;
+    *DTS = msg->av_format_change_timing->DTS;
+  }
+
+  return GST_WFD_OK;
+}
diff --git a/gst/rtsp-server/gstwfdmessage.h b/gst/rtsp-server/gstwfdmessage.h
new file mode 100755
index 0000000..3261289
--- /dev/null
+++ b/gst/rtsp-server/gstwfdmessage.h
@@ -0,0 +1,618 @@
+/* GStreamer
+ * Copyright (C) <2005,2006> Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifndef __GST_WFD_MESSAGE_H__
+#define __GST_WFD_MESSAGE_H__
+
+#include 
+#include 
+
+G_BEGIN_DECLS
+
+#define GST_STRING_WFD_AUDIO_CODECS               "wfd_audio_codecs"
+#define GST_STRING_WFD_VIDEO_FORMATS              "wfd_video_formats"
+#define GST_STRING_WFD_3D_VIDEO_FORMATS           "wfd_3d_video_formats"
+#define GST_STRING_WFD_CONTENT_PROTECTION         "wfd_content_protection"
+#define GST_STRING_WFD_DISPLAY_EDID               "wfd_display_edid"
+#define GST_STRING_WFD_COUPLED_SINK               "wfd_coupled_sink"
+#define GST_STRING_WFD_TRIGGER_METHOD             "wfd_trigger_method"
+#define GST_STRING_WFD_PRESENTATION_URL           "wfd_presentation_URL"
+#define GST_STRING_WFD_CLIENT_RTP_PORTS           "wfd_client_rtp_ports"
+#define GST_STRING_WFD_ROUTE                      "wfd_route"
+#define GST_STRING_WFD_I2C                        "wfd_I2C"
+#define GST_STRING_WFD_AV_FORMAT_CHANGE_TIMING    "wfd_av_format_change_timing"
+#define GST_STRING_WFD_PREFERRED_DISPLAY_MODE     "wfd_preferred_display_mode"
+#define GST_STRING_WFD_STANDBY_RESUME_CAPABILITY  "wfd_standby_resume_capability"
+#define GST_STRING_WFD_STANDBY                    "wfd_standby"
+#define GST_STRING_WFD_CONNECTOR_TYPE             "wfd_connector_type"
+#define GST_STRING_WFD_IDR_REQUEST                "wfd_idr_request"
+
+/**
+ * GstWFDResult:
+ * @GST_WFD_OK: A successful return value
+ * @GST_WFD_EINVAL: a function was given invalid parameters
+ *
+ * Return values for the WFD functions.
+ */
+typedef enum {
+  GST_WFD_OK     = 0,
+  GST_WFD_EINVAL = -1
+} GstWFDResult;
+
+
+typedef enum {
+  GST_WFD_AUDIO_UNKNOWN 	= 0,
+  GST_WFD_AUDIO_LPCM		= (1 << 0),
+  GST_WFD_AUDIO_AAC		= (1 << 1),
+  GST_WFD_AUDIO_AC3		= (1 << 2)
+} GstWFDAudioFormats;
+
+typedef enum {
+  GST_WFD_FREQ_UNKNOWN = 0,
+  GST_WFD_FREQ_44100 	 = (1 << 0),
+  GST_WFD_FREQ_48000	 = (1 << 1)
+} GstWFDAudioFreq;
+
+typedef enum {
+  GST_WFD_CHANNEL_UNKNOWN = 0,
+  GST_WFD_CHANNEL_2 	 	= (1 << 0),
+  GST_WFD_CHANNEL_4		= (1 << 1),
+  GST_WFD_CHANNEL_6		= (1 << 2),
+  GST_WFD_CHANNEL_8		= (1 << 3)
+} GstWFDAudioChannels;
+
+
+typedef enum {
+  GST_WFD_VIDEO_UNKNOWN = 0,
+  GST_WFD_VIDEO_H264	  = (1 << 0)
+} GstWFDVideoCodecs;
+
+typedef enum {
+  GST_WFD_VIDEO_CEA_RESOLUTION = 0,
+  GST_WFD_VIDEO_VESA_RESOLUTION,
+  GST_WFD_VIDEO_HH_RESOLUTION
+} GstWFDVideoNativeResolution;
+
+typedef enum {
+  GST_WFD_CEA_UNKNOWN		= 0,
+  GST_WFD_CEA_640x480P60 	= (1 << 0),
+  GST_WFD_CEA_720x480P60	= (1 << 1),
+  GST_WFD_CEA_720x480I60	= (1 << 2),
+  GST_WFD_CEA_720x576P50	= (1 << 3),
+  GST_WFD_CEA_720x576I50	= (1 << 4),
+  GST_WFD_CEA_1280x720P30	= (1 << 5),
+  GST_WFD_CEA_1280x720P60	= (1 << 6),
+  GST_WFD_CEA_1920x1080P30= (1 << 7),
+  GST_WFD_CEA_1920x1080P60= (1 << 8),
+  GST_WFD_CEA_1920x1080I60= (1 << 9),
+  GST_WFD_CEA_1280x720P25	= (1 << 10),
+  GST_WFD_CEA_1280x720P50	= (1 << 11),
+  GST_WFD_CEA_1920x1080P25= (1 << 12),
+  GST_WFD_CEA_1920x1080P50= (1 << 13),
+  GST_WFD_CEA_1920x1080I50= (1 << 14),
+  GST_WFD_CEA_1280x720P24	= (1 << 15),
+  GST_WFD_CEA_1920x1080P24= (1 << 16)
+} GstWFDVideoCEAResolution;
+
+typedef enum {
+  GST_WFD_VESA_UNKNOWN		= 0,
+  GST_WFD_VESA_800x600P30 	= (1 << 0),
+  GST_WFD_VESA_800x600P60		= (1 << 1),
+  GST_WFD_VESA_1024x768P30	= (1 << 2),
+  GST_WFD_VESA_1024x768P60	= (1 << 3),
+  GST_WFD_VESA_1152x864P30	= (1 << 4),
+  GST_WFD_VESA_1152x864P60	= (1 << 5),
+  GST_WFD_VESA_1280x768P30	= (1 << 6),
+  GST_WFD_VESA_1280x768P60	= (1 << 7),
+  GST_WFD_VESA_1280x800P30	= (1 << 8),
+  GST_WFD_VESA_1280x800P60	= (1 << 9),
+  GST_WFD_VESA_1360x768P30	= (1 << 10),
+  GST_WFD_VESA_1360x768P60	= (1 << 11),
+  GST_WFD_VESA_1366x768P30	= (1 << 12),
+  GST_WFD_VESA_1366x768P60	= (1 << 13),
+  GST_WFD_VESA_1280x1024P30	= (1 << 14),
+  GST_WFD_VESA_1280x1024P60	= (1 << 15),
+  GST_WFD_VESA_1400x1050P30	= (1 << 16),
+  GST_WFD_VESA_1400x1050P60	= (1 << 17),
+  GST_WFD_VESA_1440x900P30	= (1 << 18),
+  GST_WFD_VESA_1440x900P60	= (1 << 19),
+  GST_WFD_VESA_1600x900P30	= (1 << 20),
+  GST_WFD_VESA_1600x900P60	= (1 << 21),
+  GST_WFD_VESA_1600x1200P30	= (1 << 22),
+  GST_WFD_VESA_1600x1200P60	= (1 << 23),
+  GST_WFD_VESA_1680x1024P30	= (1 << 24),
+  GST_WFD_VESA_1680x1024P60	= (1 << 25),
+  GST_WFD_VESA_1680x1050P30	= (1 << 26),
+  GST_WFD_VESA_1680x1050P60	= (1 << 27),
+  GST_WFD_VESA_1920x1200P30	= (1 << 28),
+  GST_WFD_VESA_1920x1200P60	= (1 << 29)
+} GstWFDVideoVESAResolution;
+
+typedef enum {
+  GST_WFD_HH_UNKNOWN		= 0,
+  GST_WFD_HH_800x480P30 	= (1 << 0),
+  GST_WFD_HH_800x480P60	= (1 << 1),
+  GST_WFD_HH_854x480P30	= (1 << 2),
+  GST_WFD_HH_854x480P60	= (1 << 3),
+  GST_WFD_HH_864x480P30	= (1 << 4),
+  GST_WFD_HH_864x480P60	= (1 << 5),
+  GST_WFD_HH_640x360P30	= (1 << 6),
+  GST_WFD_HH_640x360P60	= (1 << 7),
+  GST_WFD_HH_960x540P30	= (1 << 8),
+  GST_WFD_HH_960x540P60	= (1 << 9),
+  GST_WFD_HH_848x480P30	= (1 << 10),
+  GST_WFD_HH_848x480P60	= (1 << 11)
+} GstWFDVideoHHResolution;
+
+typedef enum {
+  GST_WFD_H264_UNKNOWN_PROFILE= 0,
+  GST_WFD_H264_BASE_PROFILE	= (1 << 0),
+  GST_WFD_H264_HIGH_PROFILE	= (1 << 1)
+} GstWFDVideoH264Profile;
+
+typedef enum {
+  GST_WFD_H264_LEVEL_UNKNOWN = 0,
+  GST_WFD_H264_LEVEL_3_1   = (1 << 0),
+  GST_WFD_H264_LEVEL_3_2   = (1 << 1),
+  GST_WFD_H264_LEVEL_4       = (1 << 2),
+  GST_WFD_H264_LEVEL_4_1   = (1 << 3),
+  GST_WFD_H264_LEVEL_4_2   = (1 << 4)
+} GstWFDVideoH264Level;
+
+typedef enum {
+  GST_WFD_HDCP_NONE	= 0,
+  GST_WFD_HDCP_2_0	= (1 << 0),
+  GST_WFD_HDCP_2_1	= (1 << 1)
+} GstWFDHDCPProtection;
+
+typedef enum {
+  GST_WFD_SINK_UNKNOWN = -1,
+  GST_WFD_SINK_NOT_COUPLED	= 0,
+  GST_WFD_SINK_COUPLED,
+  GST_WFD_SINK_TEARDOWN_COUPLING,
+  GST_WFD_SINK_RESERVED
+} GstWFDCoupledSinkStatus;
+
+typedef enum {
+  GST_WFD_TRIGGER_UNKNOWN = 0,
+  GST_WFD_TRIGGER_SETUP,
+  GST_WFD_TRIGGER_PAUSE,
+  GST_WFD_TRIGGER_TEARDOWN,
+  GST_WFD_TRIGGER_PLAY
+} GstWFDTrigger;
+
+typedef enum {
+  GST_WFD_RTSP_TRANS_UNKNOWN =  0,
+  GST_WFD_RTSP_TRANS_RTP     = (1 << 0),
+  GST_WFD_RTSP_TRANS_RDT     = (1 << 1)
+} GstWFDRTSPTransMode;
+
+typedef enum {
+  GST_WFD_RTSP_PROFILE_UNKNOWN =  0,
+  GST_WFD_RTSP_PROFILE_AVP     = (1 << 0),
+  GST_WFD_RTSP_PROFILE_SAVP    = (1 << 1)
+} GstWFDRTSPProfile;
+
+typedef enum {
+  GST_WFD_RTSP_LOWER_TRANS_UNKNOWN   = 0,
+  GST_WFD_RTSP_LOWER_TRANS_UDP       = (1 << 0),
+  GST_WFD_RTSP_LOWER_TRANS_UDP_MCAST = (1 << 1),
+  GST_WFD_RTSP_LOWER_TRANS_TCP       = (1 << 2),
+  GST_WFD_RTSP_LOWER_TRANS_HTTP      = (1 << 3)
+} GstWFDRTSPLowerTrans;
+
+typedef enum {
+  GST_WFD_PRIMARY_SINK   = 0,
+  GST_WFD_SECONDARY_SINK
+} GstWFDSinkType;
+
+typedef enum {
+  GST_WFD_CONNECTOR_VGA           = 0,
+  GST_WFD_CONNECTOR_S,
+  GST_WFD_CONNECTOR_COMPOSITE,
+  GST_WFD_CONNECTOR_COMPONENT,
+  GST_WFD_CONNECTOR_DVI,
+  GST_WFD_CONNECTOR_HDMI,
+  GST_WFD_CONNECTOR_LVDS,
+  GST_WFD_CONNECTOR_RESERVED_7,
+  GST_WFD_CONNECTOR_JAPANESE_D,
+  GST_WFD_CONNECTOR_SDI,
+  GST_WFD_CONNECTOR_DP,
+  GST_WFD_CONNECTOR_RESERVED_11,
+  GST_WFD_CONNECTOR_UDI,
+  GST_WFD_CONNECTOR_NO           = 254,
+  GST_WFD_CONNECTOR_PHYSICAL     = 255
+} GstWFDConnector;
+
+
+typedef struct {
+  gchar	*audio_format;
+  guint32 modes;
+  guint latency;
+} GstWFDAudioCodec;
+
+typedef struct {
+  guint	count;
+  GstWFDAudioCodec *list;
+} GstWFDAudioCodeclist;
+
+
+typedef struct {
+  guint CEA_Support;
+  guint VESA_Support;
+  guint HH_Support;
+  guint latency;
+  guint min_slice_size;
+  guint slice_enc_params;
+  guint frame_rate_control_support;
+} GstWFDVideoH264MiscParams;
+
+typedef struct {
+  guint profile;
+  guint level;
+  guint max_hres;
+  guint max_vres;
+  GstWFDVideoH264MiscParams misc_params;
+} GstWFDVideoH264Codec;
+
+typedef struct {
+  guint	native;
+  guint preferred_display_mode_supported;
+  GstWFDVideoH264Codec H264_codec;
+} GstWFDVideoCodec;
+
+typedef struct {
+  guint			count;
+  GstWFDVideoCodec *list;
+} GstWFDVideoCodeclist;
+
+typedef struct {
+  guint video_3d_capability;
+  guint latency;
+  guint min_slice_size;
+  guint slice_enc_params;
+  guint frame_rate_control_support;
+} GstWFD3DVideoH264MiscParams;
+
+typedef struct {
+  guint profile;
+  guint level;
+  GstWFD3DVideoH264MiscParams misc_params;
+  guint max_hres;
+  guint max_vres;
+} GstWFD3DVideoH264Codec;
+
+typedef struct {
+  guint native;
+  guint preferred_display_mode_supported;
+  GstWFD3DVideoH264Codec H264_codec;
+} GstWFD3dCapList;
+
+typedef struct {
+  guint			count;
+  GstWFD3dCapList *list;
+} GstWFD3DFormats;
+
+typedef struct {
+  gchar *hdcpversion;
+  gchar *TCPPort;
+} GstWFDHdcp2Spec;
+
+typedef struct {
+  GstWFDHdcp2Spec *hdcp2_spec;
+} GstWFDContentProtection;
+
+typedef struct {
+  guint edid_supported;
+  guint edid_block_count;
+  gchar *edid_payload;
+} GstWFDDisplayEdid;
+
+
+typedef struct {
+  guint status;
+  gchar *sink_address;
+} GstWFDCoupled_sink_cap;
+
+typedef struct {
+  GstWFDCoupled_sink_cap *coupled_sink_cap;
+} GstWFDCoupledSink;
+
+typedef struct {
+  gchar *wfd_trigger_method;
+} GstWFDTriggerMethod;
+
+typedef struct {
+  gchar *wfd_url0;
+  gchar *wfd_url1;
+} GstWFDPresentationUrl;
+
+typedef struct {
+  gchar *profile;
+  guint32 rtp_port0;
+  guint32 rtp_port1;
+  gchar *mode;
+} GstWFDClientRtpPorts;
+
+typedef struct {
+ gchar *destination;
+} GstWFDRoute;
+
+typedef struct {
+  gboolean I2CPresent;
+  guint32 I2C_port;
+} GstWFDI2C;
+
+typedef struct {
+  guint64 PTS;
+  guint64 DTS;
+} GstWFDAVFormatChangeTiming;
+
+typedef struct {
+  gboolean displaymodesupported;
+  guint64 p_clock;
+  guint32 H;
+  guint32 HB;
+  guint32 HSPOL_HSOFF;
+  guint32 HSW;
+  guint32 V;
+  guint32 VB;
+  guint32 VSPOL_VSOFF;
+  guint32 VSW;
+  guint VBS3D;
+  guint R;
+  guint V2d_s3d_modes;
+  guint P_depth;
+  GstWFDVideoH264Codec H264_codec;
+} GstWFDPreferredDisplayMode;
+
+typedef struct {
+  gboolean standby_resume_cap;
+} GstWFDStandbyResumeCapability;
+
+typedef struct {
+  gboolean wfd_standby;
+} GstWFDStandby;
+
+typedef struct {
+  gboolean supported;
+  gint32 connector_type;
+} GstWFDConnectorType;
+
+typedef struct {
+  gboolean idr_request;
+} GstWFDIdrRequest;
+
+/**
+ * GstWFDMessage:
+ * @version: the protocol version
+ * @origin: owner/creator and session identifier
+ * @session_name: session name
+ * @information: session information
+ * @uri: URI of description
+ * @emails: array of #gchar with email addresses
+ * @phones: array of #gchar with phone numbers
+ * @connection: connection information for the session
+ * @bandwidths: array of #GstWFDBandwidth with bandwidth information
+ * @times: array of #GstWFDTime with time descriptions
+ * @zones: array of #GstWFDZone with time zone adjustments
+ * @key: encryption key
+ * @attributes: array of #GstWFDAttribute with session attributes
+ * @medias: array of #GstWFDMedia with media descriptions
+ *
+ * The contents of the WFD message.
+ */
+typedef struct {
+  GstWFDAudioCodeclist *audio_codecs;
+  GstWFDVideoCodeclist *video_formats;
+  GstWFD3DFormats *video_3d_formats;
+  GstWFDContentProtection *content_protection;
+  GstWFDDisplayEdid *display_edid;
+  GstWFDCoupledSink *coupled_sink;
+  GstWFDTriggerMethod *trigger_method;
+  GstWFDPresentationUrl *presentation_url;
+  GstWFDClientRtpPorts *client_rtp_ports;
+  GstWFDRoute *route;
+  GstWFDI2C *I2C;
+  GstWFDAVFormatChangeTiming *av_format_change_timing;
+  GstWFDPreferredDisplayMode *preferred_display_mode;
+  GstWFDStandbyResumeCapability *standby_resume_capability;
+  GstWFDStandby *standby;
+  GstWFDConnectorType *connector_type;
+  GstWFDIdrRequest *idr_request;
+} GstWFDMessage;
+
+GType                   gst_wfd_message_get_type            (void);
+
+#define GST_TYPE_WFD_MESSAGE           (gst_wfd_message_get_type())
+#define GST_WFD_MESSAGE_CAST(object)   ((GstWFDMessage *)(object))
+#define GST_WFD_MESSAGE(object)        (GST_WFD_MESSAGE_CAST(object))
+
+/* Session descriptions */
+GstWFDResult            gst_wfd_message_new                 (GstWFDMessage **msg);
+GstWFDResult            gst_wfd_message_init                (GstWFDMessage *msg);
+GstWFDResult            gst_wfd_message_uninit              (GstWFDMessage *msg);
+GstWFDResult            gst_wfd_message_free                (GstWFDMessage *msg);
+GstWFDResult            gst_wfd_message_copy                (const GstWFDMessage *msg, GstWFDMessage **copy);
+
+GstWFDResult            gst_wfd_message_parse_buffer        (const guint8 *data, guint size, GstWFDMessage *msg);
+gchar*                  gst_wfd_message_as_text             (const GstWFDMessage *msg);
+gchar*                  gst_wfd_message_param_names_as_text (const GstWFDMessage *msg);
+GstWFDResult            gst_wfd_message_dump                (const GstWFDMessage *msg);
+
+
+GstWFDResult gst_wfd_message_set_supported_audio_format(GstWFDMessage *msg,
+                                        GstWFDAudioFormats a_codec,
+                                        guint a_freq, guint a_channels,
+                                        guint a_bitwidth, guint32 a_latency);
+
+GstWFDResult gst_wfd_message_set_prefered_audio_format(GstWFDMessage *msg,
+                                        GstWFDAudioFormats a_codec,
+                                        GstWFDAudioFreq a_freq,
+                                        GstWFDAudioChannels a_channels,
+                                        guint a_bitwidth, guint32 a_latency);
+
+GstWFDResult gst_wfd_message_get_supported_audio_format (GstWFDMessage *msg,
+                                        guint *a_codec,
+                                        guint *a_freq,
+                                        guint *a_channels,
+                                        guint *a_bitwidth,
+                                        guint32 *a_latency);
+
+GstWFDResult gst_wfd_message_get_prefered_audio_format (GstWFDMessage *msg,
+                                        GstWFDAudioFormats *a_codec,
+                                        GstWFDAudioFreq *a_freq,
+                                        GstWFDAudioChannels *a_channels,
+                                        guint *a_bitwidth, guint32 *a_latency);
+
+GstWFDResult gst_wfd_message_set_supported_video_format (GstWFDMessage *msg,
+                                        GstWFDVideoCodecs v_codec,
+                                        GstWFDVideoNativeResolution v_native,
+                                        guint64 v_native_resolution,
+                                        guint64 v_cea_resolution,
+                                        guint64 v_vesa_resolution,
+                                        guint64 v_hh_resolution,
+                                        guint v_profile,
+                                        guint v_level,
+                                        guint32 v_latency,
+                                        guint32 v_max_height,
+                                        guint32 v_max_width,
+                                        guint32 min_slice_size,
+                                        guint32 slice_enc_params,
+                                        guint frame_rate_control);
+
+GstWFDResult gst_wfd_message_set_prefered_video_format(GstWFDMessage *msg,
+                                        GstWFDVideoCodecs v_codec,
+                                        GstWFDVideoNativeResolution v_native,
+                                        guint64 v_native_resolution,
+                                        GstWFDVideoCEAResolution v_cea_resolution,
+                                        GstWFDVideoVESAResolution v_vesa_resolution,
+                                        GstWFDVideoHHResolution v_hh_resolution,
+                                        GstWFDVideoH264Profile v_profile,
+                                        GstWFDVideoH264Level v_level,
+                                        guint32 v_latency,
+                                        guint32 v_max_height,
+                                        guint32 v_max_width,
+                                        guint32 min_slice_size,
+                                        guint32 slice_enc_params,
+                                        guint frame_rate_control);
+
+GstWFDResult gst_wfd_message_get_supported_video_format(GstWFDMessage *msg,
+                                        GstWFDVideoCodecs *v_codec,
+                                        GstWFDVideoNativeResolution *v_native,
+                                        guint64 *v_native_resolution,
+                                        guint64 *v_cea_resolution,
+                                        guint64 *v_vesa_resolution,
+                                        guint64 *v_hh_resolution,
+                                        guint *v_profile,
+                                        guint *v_level,
+                                        guint32 *v_latency,
+                                        guint32 *v_max_height,
+                                        guint32 *v_max_width,
+                                        guint32 *min_slice_size,
+                                        guint32 *slice_enc_params,
+                                        guint *frame_rate_control);
+
+GstWFDResult gst_wfd_message_get_prefered_video_format(GstWFDMessage *msg,
+                                        GstWFDVideoCodecs *v_codec,
+                                        GstWFDVideoNativeResolution *v_native,
+                                        guint64 *v_native_resolution,
+                                        GstWFDVideoCEAResolution *v_cea_resolution,
+                                        GstWFDVideoVESAResolution *v_vesa_resolution,
+                                        GstWFDVideoHHResolution *v_hh_resolution,
+                                        GstWFDVideoH264Profile *v_profile,
+                                        GstWFDVideoH264Level *v_level,
+                                        guint32 *v_latency,
+                                        guint32 *v_max_height,
+                                        guint32 *v_max_width,
+                                        guint32 *min_slice_size,
+                                        guint32 *slice_enc_params,
+                                        guint *frame_rate_control);
+
+GstWFDResult gst_wfd_message_set_display_edid (GstWFDMessage *msg,
+                                        gboolean edid_supported,
+                                        guint32 edid_blockcount,
+                                        gchar *edid_playload);
+
+GstWFDResult gst_wfd_message_get_display_edid (GstWFDMessage *msg,
+                                        gboolean *edid_supported,
+                                        guint32 *edid_blockcount,
+                                        gchar **edid_playload);
+
+GstWFDResult gst_wfd_message_set_contentprotection_type (GstWFDMessage *msg,
+                                        GstWFDHDCPProtection hdcpversion,
+                                        guint32 TCPPort);
+
+GstWFDResult gst_wfd_message_get_contentprotection_type (GstWFDMessage *msg,
+                                        GstWFDHDCPProtection *hdcpversion,
+                                        guint32 *TCPPort);
+
+GstWFDResult gst_wfd_messge_set_prefered_rtp_ports (GstWFDMessage *msg,
+                                        GstWFDRTSPTransMode trans,
+                                        GstWFDRTSPProfile profile,
+                                        GstWFDRTSPLowerTrans lowertrans,
+                                        guint32 rtp_port0,
+                                        guint32 rtp_port1);
+
+GstWFDResult gst_wfd_message_get_prefered_rtp_ports (GstWFDMessage *msg,
+                                        GstWFDRTSPTransMode *trans,
+                                        GstWFDRTSPProfile *profile,
+                                        GstWFDRTSPLowerTrans *lowertrans,
+                                        guint32 *rtp_port0,
+                                        guint32 *rtp_port1);
+
+GstWFDResult gst_wfd_message_set_presentation_url(GstWFDMessage *msg,
+                                        gchar *wfd_url0, gchar *wfd_url1);
+
+GstWFDResult gst_wfd_message_get_presentation_url(GstWFDMessage *msg, gchar **wfd_url0,
+                                        gchar **wfd_url1);
+
+GstWFDResult gst_wfd_message_set_av_format_change_timing(GstWFDMessage *msg,
+                                        guint64 PTS,
+                                        guint64 DTS);
+
+GstWFDResult gst_wfd_message_get_av_format_change_timing(GstWFDMessage *msg,
+                                        guint64 *PTS,
+                                        guint64 *DTS);
+G_END_DECLS
+
+#endif /* __GST_WFD_MESSAGE_H__ */
diff --git a/gst/rtsp-server/rtsp-address-pool.c b/gst/rtsp-server/rtsp-address-pool.c
new file mode 100644
index 0000000..1343839
--- /dev/null
+++ b/gst/rtsp-server/rtsp-address-pool.c
@@ -0,0 +1,751 @@
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-address-pool
+ * @short_description: A pool of network addresses
+ * @see_also: #GstRTSPStream, #GstRTSPStreamTransport
+ *
+ * The #GstRTSPAddressPool is an object that maintains a collection of network
+ * addresses. It is used to allocate server ports and server multicast addresses
+ * but also to reserve client provided destination addresses.
+ *
+ * A range of addresses can be added with gst_rtsp_address_pool_add_range().
+ * Both multicast and unicast addresses can be added.
+ *
+ * With gst_rtsp_address_pool_acquire_address() an unused address and port range
+ * can be acquired from the pool. With gst_rtsp_address_pool_reserve_address() a
+ * specific address can be retrieved. Both methods return a boxed
+ * #GstRTSPAddress that should be freed with gst_rtsp_address_free() after
+ * usage, which brings the address back into the pool.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
+
+#include 
+#include 
+
+#include "rtsp-address-pool.h"
+
+/**
+ * gst_rtsp_address_copy:
+ * @addr: a #GstRTSPAddress
+ *
+ * Make a copy of @addr.
+ *
+ * Returns: a copy of @addr.
+ */
+GstRTSPAddress *
+gst_rtsp_address_copy (GstRTSPAddress * addr)
+{
+  GstRTSPAddress *copy;
+
+  g_return_val_if_fail (addr != NULL, NULL);
+
+  copy = g_slice_dup (GstRTSPAddress, addr);
+  /* only release to the pool when the original is freed. It's a bit
+   * weird but this will do for now as it avoid us to use refcounting. */
+  copy->pool = NULL;
+  copy->address = g_strdup (copy->address);
+
+  return copy;
+}
+
+static void gst_rtsp_address_pool_release_address (GstRTSPAddressPool * pool,
+    GstRTSPAddress * addr);
+
+/**
+ * gst_rtsp_address_free:
+ * @addr: a #GstRTSPAddress
+ *
+ * Free @addr and releasing it back into the pool when owned by a
+ * pool.
+ */
+void
+gst_rtsp_address_free (GstRTSPAddress * addr)
+{
+  g_return_if_fail (addr != NULL);
+
+  if (addr->pool) {
+    /* unrefs the pool and sets it to NULL */
+    gst_rtsp_address_pool_release_address (addr->pool, addr);
+  }
+  g_free (addr->address);
+  g_slice_free (GstRTSPAddress, addr);
+}
+
+G_DEFINE_BOXED_TYPE (GstRTSPAddress, gst_rtsp_address,
+    (GBoxedCopyFunc) gst_rtsp_address_copy,
+    (GBoxedFreeFunc) gst_rtsp_address_free);
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_address_pool_debug);
+#define GST_CAT_DEFAULT rtsp_address_pool_debug
+
+#define GST_RTSP_ADDRESS_POOL_GET_PRIVATE(obj)  \
+     (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_ADDRESS_POOL, GstRTSPAddressPoolPrivate))
+
+struct _GstRTSPAddressPoolPrivate
+{
+  GMutex lock;                  /* protects everything in this struct */
+  GList *addresses;
+  GList *allocated;
+
+  gboolean has_unicast_addresses;
+};
+
+#define ADDR_IS_IPV4(a)      ((a)->size == 4)
+#define ADDR_IS_IPV6(a)      ((a)->size == 16)
+#define ADDR_IS_EVEN_PORT(a) (((a)->port & 1) == 0)
+
+typedef struct
+{
+  guint8 bytes[16];
+  gsize size;
+  guint16 port;
+} Addr;
+
+typedef struct
+{
+  Addr min;
+  Addr max;
+  guint8 ttl;
+} AddrRange;
+
+#define RANGE_IS_SINGLE(r) (memcmp ((r)->min.bytes, (r)->max.bytes, (r)->min.size) == 0)
+
+#define gst_rtsp_address_pool_parent_class parent_class
+G_DEFINE_TYPE (GstRTSPAddressPool, gst_rtsp_address_pool, G_TYPE_OBJECT);
+
+static void gst_rtsp_address_pool_finalize (GObject * obj);
+
+static void
+gst_rtsp_address_pool_class_init (GstRTSPAddressPoolClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  gobject_class = G_OBJECT_CLASS (klass);
+
+  gobject_class->finalize = gst_rtsp_address_pool_finalize;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPAddressPoolPrivate));
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_address_pool_debug, "rtspaddresspool", 0,
+      "GstRTSPAddressPool");
+}
+
+static void
+gst_rtsp_address_pool_init (GstRTSPAddressPool * pool)
+{
+  pool->priv = GST_RTSP_ADDRESS_POOL_GET_PRIVATE (pool);
+
+  g_mutex_init (&pool->priv->lock);
+}
+
+static void
+free_range (AddrRange * range)
+{
+  g_slice_free (AddrRange, range);
+}
+
+static void
+gst_rtsp_address_pool_finalize (GObject * obj)
+{
+  GstRTSPAddressPool *pool;
+
+  pool = GST_RTSP_ADDRESS_POOL (obj);
+
+  g_list_free_full (pool->priv->addresses, (GDestroyNotify) free_range);
+  g_list_free_full (pool->priv->allocated, (GDestroyNotify) free_range);
+  g_mutex_clear (&pool->priv->lock);
+
+  G_OBJECT_CLASS (parent_class)->finalize (obj);
+}
+
+/**
+ * gst_rtsp_address_pool_new:
+ *
+ * Make a new #GstRTSPAddressPool.
+ *
+ * Returns: (transfer full): a new #GstRTSPAddressPool
+ */
+GstRTSPAddressPool *
+gst_rtsp_address_pool_new (void)
+{
+  GstRTSPAddressPool *pool;
+
+  pool = g_object_new (GST_TYPE_RTSP_ADDRESS_POOL, NULL);
+
+  return pool;
+}
+
+/**
+ * gst_rtsp_address_pool_clear:
+ * @pool: a #GstRTSPAddressPool
+ *
+ * Clear all addresses in @pool. There should be no outstanding
+ * allocations.
+ */
+void
+gst_rtsp_address_pool_clear (GstRTSPAddressPool * pool)
+{
+  GstRTSPAddressPoolPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool));
+  g_return_if_fail (pool->priv->allocated == NULL);
+
+  priv = pool->priv;
+
+  g_mutex_lock (&priv->lock);
+  g_list_free_full (priv->addresses, (GDestroyNotify) free_range);
+  priv->addresses = NULL;
+  g_mutex_unlock (&priv->lock);
+}
+
+static gboolean
+fill_address (const gchar * address, guint16 port, Addr * addr,
+    gboolean is_multicast)
+{
+  GInetAddress *inet;
+
+  inet = g_inet_address_new_from_string (address);
+  if (inet == NULL)
+    return FALSE;
+
+  if (is_multicast != g_inet_address_get_is_multicast (inet)) {
+    g_object_unref (inet);
+    return FALSE;
+  }
+
+  addr->size = g_inet_address_get_native_size (inet);
+  memcpy (addr->bytes, g_inet_address_to_bytes (inet), addr->size);
+  g_object_unref (inet);
+  addr->port = port;
+
+  return TRUE;
+}
+
+static gchar *
+get_address_string (Addr * addr)
+{
+  gchar *res;
+  GInetAddress *inet;
+
+  inet = g_inet_address_new_from_bytes (addr->bytes,
+      addr->size == 4 ? G_SOCKET_FAMILY_IPV4 : G_SOCKET_FAMILY_IPV6);
+  res = g_inet_address_to_string (inet);
+  g_object_unref (inet);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_address_pool_add_range:
+ * @pool: a #GstRTSPAddressPool
+ * @min_address: a minimum address to add
+ * @max_address: a maximum address to add
+ * @min_port: the minimum port
+ * @max_port: the maximum port
+ * @ttl: a TTL or 0 for unicast addresses
+ *
+ * Adds the addresses from @min_addess to @max_address (inclusive)
+ * to @pool. The valid port range for the addresses will be from @min_port to
+ * @max_port inclusive.
+ *
+ * When @ttl is 0, @min_address and @max_address should be unicast addresses.
+ * @min_address and @max_address can be set to
+ * #GST_RTSP_ADDRESS_POOL_ANY_IPV4 or #GST_RTSP_ADDRESS_POOL_ANY_IPV6 to bind
+ * to all available IPv4 or IPv6 addresses.
+ *
+ * When @ttl > 0, @min_address and @max_address should be multicast addresses.
+ *
+ * Returns: %TRUE if the addresses could be added.
+ */
+gboolean
+gst_rtsp_address_pool_add_range (GstRTSPAddressPool * pool,
+    const gchar * min_address, const gchar * max_address,
+    guint16 min_port, guint16 max_port, guint8 ttl)
+{
+  AddrRange *range;
+  GstRTSPAddressPoolPrivate *priv;
+  gboolean is_multicast;
+
+  g_return_val_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool), FALSE);
+  g_return_val_if_fail (min_port <= max_port, FALSE);
+
+  priv = pool->priv;
+
+  is_multicast = ttl != 0;
+
+  range = g_slice_new0 (AddrRange);
+
+  if (!fill_address (min_address, min_port, &range->min, is_multicast))
+    goto invalid;
+  if (!fill_address (max_address, max_port, &range->max, is_multicast))
+    goto invalid;
+
+  if (range->min.size != range->max.size)
+    goto invalid;
+  if (memcmp (range->min.bytes, range->max.bytes, range->min.size) > 0)
+    goto invalid;
+
+  range->ttl = ttl;
+
+  GST_DEBUG_OBJECT (pool, "adding %s-%s:%u-%u ttl %u", min_address, max_address,
+      min_port, max_port, ttl);
+
+  g_mutex_lock (&priv->lock);
+  priv->addresses = g_list_prepend (priv->addresses, range);
+
+  if (!is_multicast)
+    priv->has_unicast_addresses = TRUE;
+  g_mutex_unlock (&priv->lock);
+
+  return TRUE;
+
+  /* ERRORS */
+invalid:
+  {
+    GST_ERROR_OBJECT (pool, "invalid address range %s-%s", min_address,
+        max_address);
+    g_slice_free (AddrRange, range);
+    return FALSE;
+  }
+}
+
+static void
+inc_address (Addr * addr, guint count)
+{
+  gint i;
+  guint carry;
+
+  carry = count;
+  for (i = addr->size - 1; i >= 0 && carry > 0; i--) {
+    carry += addr->bytes[i];
+    addr->bytes[i] = carry & 0xff;
+    carry >>= 8;
+  }
+}
+
+/* tells us the number of addresses between min_addr and max_addr */
+static guint
+diff_address (Addr * max_addr, Addr * min_addr)
+{
+  gint i;
+  guint result = 0;
+
+  g_return_val_if_fail (min_addr->size == max_addr->size, 0);
+
+  for (i = 0; i < min_addr->size; i++) {
+    g_return_val_if_fail (result < (1 << 24), result);
+
+    result <<= 8;
+    result += max_addr->bytes[i] - min_addr->bytes[i];
+  }
+
+  return result;
+}
+
+static AddrRange *
+split_range (GstRTSPAddressPool * pool, AddrRange * range, guint skip_addr,
+    guint skip_port, gint n_ports)
+{
+  GstRTSPAddressPoolPrivate *priv = pool->priv;
+  AddrRange *temp;
+
+  if (skip_addr) {
+    temp = g_slice_dup (AddrRange, range);
+    memcpy (temp->max.bytes, temp->min.bytes, temp->min.size);
+    inc_address (&temp->max, skip_addr - 1);
+    priv->addresses = g_list_prepend (priv->addresses, temp);
+
+    inc_address (&range->min, skip_addr);
+  }
+
+  if (!RANGE_IS_SINGLE (range)) {
+    /* min and max are not the same, we have more than one address. */
+    temp = g_slice_dup (AddrRange, range);
+    /* increment the range min address */
+    inc_address (&temp->min, 1);
+    /* and store back in pool */
+    priv->addresses = g_list_prepend (priv->addresses, temp);
+
+    /* adjust range with only the first address */
+    memcpy (range->max.bytes, range->min.bytes, range->min.size);
+  }
+
+  /* range now contains only one single address */
+  if (skip_port > 0) {
+    /* make a range with the skipped ports */
+    temp = g_slice_dup (AddrRange, range);
+    temp->max.port = temp->min.port + skip_port - 1;
+    /* and store back in pool */
+    priv->addresses = g_list_prepend (priv->addresses, temp);
+
+    /* increment range port */
+    range->min.port += skip_port;
+  }
+  /* range now contains single address with desired port number */
+  if (range->max.port - range->min.port + 1 > n_ports) {
+    /* make a range with the remaining ports */
+    temp = g_slice_dup (AddrRange, range);
+    temp->min.port += n_ports;
+    /* and store back in pool */
+    priv->addresses = g_list_prepend (priv->addresses, temp);
+
+    /* and truncate port */
+    range->max.port = range->min.port + n_ports - 1;
+  }
+  return range;
+}
+
+/**
+ * gst_rtsp_address_pool_acquire_address:
+ * @pool: a #GstRTSPAddressPool
+ * @flags: flags
+ * @n_ports: the amount of ports
+ *
+ * Take an address and ports from @pool. @flags can be used to control the
+ * allocation. @n_ports consecutive ports will be allocated of which the first
+ * one can be found in @port.
+ *
+ * Returns: (nullable): a #GstRTSPAddress that should be freed with
+ * gst_rtsp_address_free after use or %NULL when no address could be
+ * acquired.
+ */
+GstRTSPAddress *
+gst_rtsp_address_pool_acquire_address (GstRTSPAddressPool * pool,
+    GstRTSPAddressFlags flags, gint n_ports)
+{
+  GstRTSPAddressPoolPrivate *priv;
+  GList *walk, *next;
+  AddrRange *result;
+  GstRTSPAddress *addr;
+
+  g_return_val_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool), NULL);
+  g_return_val_if_fail (n_ports > 0, NULL);
+
+  priv = pool->priv;
+  result = NULL;
+  addr = NULL;
+
+  g_mutex_lock (&priv->lock);
+  /* go over available ranges */
+  for (walk = priv->addresses; walk; walk = next) {
+    AddrRange *range;
+    gint ports, skip;
+
+    range = walk->data;
+    next = walk->next;
+
+    /* check address type when given */
+    if (flags & GST_RTSP_ADDRESS_FLAG_IPV4 && !ADDR_IS_IPV4 (&range->min))
+      continue;
+    if (flags & GST_RTSP_ADDRESS_FLAG_IPV6 && !ADDR_IS_IPV6 (&range->min))
+      continue;
+    if (flags & GST_RTSP_ADDRESS_FLAG_MULTICAST && range->ttl == 0)
+      continue;
+    if (flags & GST_RTSP_ADDRESS_FLAG_UNICAST && range->ttl != 0)
+      continue;
+
+    /* check for enough ports */
+    ports = range->max.port - range->min.port + 1;
+    if (flags & GST_RTSP_ADDRESS_FLAG_EVEN_PORT
+        && !ADDR_IS_EVEN_PORT (&range->min))
+      skip = 1;
+    else
+      skip = 0;
+    if (ports - skip < n_ports)
+      continue;
+
+    /* we found a range, remove from the list */
+    priv->addresses = g_list_delete_link (priv->addresses, walk);
+    /* now split and exit our loop */
+    result = split_range (pool, range, 0, skip, n_ports);
+    priv->allocated = g_list_prepend (priv->allocated, result);
+    break;
+  }
+  g_mutex_unlock (&priv->lock);
+
+  if (result) {
+    addr = g_slice_new0 (GstRTSPAddress);
+    addr->pool = g_object_ref (pool);
+    addr->address = get_address_string (&result->min);
+    addr->n_ports = n_ports;
+    addr->port = result->min.port;
+    addr->ttl = result->ttl;
+    addr->priv = result;
+
+    GST_DEBUG_OBJECT (pool, "got address %s:%u ttl %u", addr->address,
+        addr->port, addr->ttl);
+  }
+
+  return addr;
+}
+
+/**
+ * gst_rtsp_address_pool_release_address:
+ * @pool: a #GstRTSPAddressPool
+ * @id: an address id
+ *
+ * Release a previously acquired address (with
+ * gst_rtsp_address_pool_acquire_address()) back into @pool.
+ */
+static void
+gst_rtsp_address_pool_release_address (GstRTSPAddressPool * pool,
+    GstRTSPAddress * addr)
+{
+  GstRTSPAddressPoolPrivate *priv;
+  GList *find;
+  AddrRange *range;
+
+  g_return_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool));
+  g_return_if_fail (addr != NULL);
+  g_return_if_fail (addr->pool == pool);
+
+  priv = pool->priv;
+  range = addr->priv;
+
+  /* we don't want to free twice */
+  addr->priv = NULL;
+  addr->pool = NULL;
+
+  g_mutex_lock (&priv->lock);
+  find = g_list_find (priv->allocated, range);
+  if (find == NULL)
+    goto not_found;
+
+  priv->allocated = g_list_delete_link (priv->allocated, find);
+
+  /* FIXME, merge and do something clever */
+  priv->addresses = g_list_prepend (priv->addresses, range);
+  g_mutex_unlock (&priv->lock);
+
+  g_object_unref (pool);
+
+  return;
+
+  /* ERRORS */
+not_found:
+  {
+    g_warning ("Released unknown address %p", addr);
+    g_mutex_unlock (&priv->lock);
+    return;
+  }
+}
+
+static void
+dump_range (AddrRange * range, GstRTSPAddressPool * pool)
+{
+  gchar *addr1, *addr2;
+
+  addr1 = get_address_string (&range->min);
+  addr2 = get_address_string (&range->max);
+  g_print ("  address %s-%s, port %u-%u, ttl %u\n", addr1, addr2,
+      range->min.port, range->max.port, range->ttl);
+  g_free (addr1);
+  g_free (addr2);
+}
+
+/**
+ * gst_rtsp_address_pool_dump:
+ * @pool: a #GstRTSPAddressPool
+ *
+ * Dump the free and allocated addresses to stdout.
+ */
+void
+gst_rtsp_address_pool_dump (GstRTSPAddressPool * pool)
+{
+  GstRTSPAddressPoolPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool));
+
+  priv = pool->priv;
+
+  g_mutex_lock (&priv->lock);
+  g_print ("free:\n");
+  g_list_foreach (priv->addresses, (GFunc) dump_range, pool);
+  g_print ("allocated:\n");
+  g_list_foreach (priv->allocated, (GFunc) dump_range, pool);
+  g_mutex_unlock (&priv->lock);
+}
+
+static GList *
+find_address_in_ranges (GList * addresses, Addr * addr, guint port,
+    guint n_ports, guint ttl)
+{
+  GList *walk, *next;
+
+  /* go over available ranges */
+  for (walk = addresses; walk; walk = next) {
+    AddrRange *range;
+
+    range = walk->data;
+    next = walk->next;
+
+    /* Not the right type of address */
+    if (range->min.size != addr->size)
+      continue;
+
+    /* Check that the address is in the interval */
+    if (memcmp (range->min.bytes, addr->bytes, addr->size) > 0 ||
+        memcmp (range->max.bytes, addr->bytes, addr->size) < 0)
+      continue;
+
+    /* Make sure the requested ports are inside the range */
+    if (port < range->min.port || port + n_ports - 1 > range->max.port)
+      continue;
+
+    if (ttl != range->ttl)
+      continue;
+
+    break;
+  }
+
+  return walk;
+}
+
+/**
+ * gst_rtsp_address_pool_reserve_address:
+ * @pool: a #GstRTSPAddressPool
+ * @ip_address: The IP address to reserve
+ * @port: The first port to reserve
+ * @n_ports: The number of ports
+ * @ttl: The requested ttl
+ * @address: (out): storage for a #GstRTSPAddress
+ *
+ * Take a specific address and ports from @pool. @n_ports consecutive
+ * ports will be allocated of which the first one can be found in
+ * @port.
+ *
+ * If @ttl is 0, @address should be a unicast address. If @ttl > 0, @address
+ * should be a valid multicast address.
+ *
+ * Returns: #GST_RTSP_ADDRESS_POOL_OK if an address was reserved. The address
+ * is returned in @address and should be freed with gst_rtsp_address_free
+ * after use.
+ */
+GstRTSPAddressPoolResult
+gst_rtsp_address_pool_reserve_address (GstRTSPAddressPool * pool,
+    const gchar * ip_address, guint port, guint n_ports, guint ttl,
+    GstRTSPAddress ** address)
+{
+  GstRTSPAddressPoolPrivate *priv;
+  Addr input_addr;
+  GList *list;
+  AddrRange *addr_range;
+  GstRTSPAddress *addr;
+  gboolean is_multicast;
+  GstRTSPAddressPoolResult result;
+
+  g_return_val_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool),
+      GST_RTSP_ADDRESS_POOL_EINVAL);
+  g_return_val_if_fail (ip_address != NULL, GST_RTSP_ADDRESS_POOL_EINVAL);
+  g_return_val_if_fail (port > 0, GST_RTSP_ADDRESS_POOL_EINVAL);
+  g_return_val_if_fail (n_ports > 0, GST_RTSP_ADDRESS_POOL_EINVAL);
+  g_return_val_if_fail (address != NULL, GST_RTSP_ADDRESS_POOL_EINVAL);
+
+  priv = pool->priv;
+  addr_range = NULL;
+  addr = NULL;
+  is_multicast = ttl != 0;
+
+  if (!fill_address (ip_address, port, &input_addr, is_multicast))
+    goto invalid;
+
+  g_mutex_lock (&priv->lock);
+  list = find_address_in_ranges (priv->addresses, &input_addr, port, n_ports,
+      ttl);
+  if (list != NULL) {
+    AddrRange *range = list->data;
+    guint skip_port, skip_addr;
+
+    skip_addr = diff_address (&input_addr, &range->min);
+    skip_port = port - range->min.port;
+
+    GST_DEBUG_OBJECT (pool, "diff 0x%08x/%u", skip_addr, skip_port);
+
+    /* we found a range, remove from the list */
+    priv->addresses = g_list_delete_link (priv->addresses, list);
+    /* now split and exit our loop */
+    addr_range = split_range (pool, range, skip_addr, skip_port, n_ports);
+    priv->allocated = g_list_prepend (priv->allocated, addr_range);
+  }
+
+  if (addr_range) {
+    addr = g_slice_new0 (GstRTSPAddress);
+    addr->pool = g_object_ref (pool);
+    addr->address = get_address_string (&addr_range->min);
+    addr->n_ports = n_ports;
+    addr->port = addr_range->min.port;
+    addr->ttl = addr_range->ttl;
+    addr->priv = addr_range;
+
+    result = GST_RTSP_ADDRESS_POOL_OK;
+    GST_DEBUG_OBJECT (pool, "reserved address %s:%u ttl %u", addr->address,
+        addr->port, addr->ttl);
+  } else {
+    /* We failed to reserve the address. Check if it was because the address
+     * was already in use or if it wasn't in the pool to begin with */
+    list = find_address_in_ranges (priv->allocated, &input_addr, port, n_ports,
+        ttl);
+    if (list != NULL) {
+      result = GST_RTSP_ADDRESS_POOL_ERESERVED;
+    } else {
+      result = GST_RTSP_ADDRESS_POOL_ERANGE;
+    }
+  }
+  g_mutex_unlock (&priv->lock);
+
+  *address = addr;
+  return result;
+
+  /* ERRORS */
+invalid:
+  {
+    GST_ERROR_OBJECT (pool, "invalid address %s:%u/%u/%u", ip_address,
+        port, n_ports, ttl);
+    *address = NULL;
+    return GST_RTSP_ADDRESS_POOL_EINVAL;
+  }
+}
+
+/**
+ * gst_rtsp_address_pool_has_unicast_addresses:
+ * @pool: a #GstRTSPAddressPool
+ *
+ * Used to know if the pool includes any unicast addresses.
+ *
+ * Returns: %TRUE if the pool includes any unicast addresses, %FALSE otherwise
+ */
+
+gboolean
+gst_rtsp_address_pool_has_unicast_addresses (GstRTSPAddressPool * pool)
+{
+  GstRTSPAddressPoolPrivate *priv;
+  gboolean has_unicast_addresses;
+
+  g_return_val_if_fail (GST_IS_RTSP_ADDRESS_POOL (pool), FALSE);
+
+  priv = pool->priv;
+
+  g_mutex_lock (&priv->lock);
+  has_unicast_addresses = priv->has_unicast_addresses;
+  g_mutex_unlock (&priv->lock);
+
+  return has_unicast_addresses;
+}
diff --git a/gst/rtsp-server/rtsp-address-pool.h b/gst/rtsp-server/rtsp-address-pool.h
new file mode 100644
index 0000000..a473bdf
--- /dev/null
+++ b/gst/rtsp-server/rtsp-address-pool.h
@@ -0,0 +1,182 @@
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#ifndef __GST_RTSP_ADDRESS_POOL_H__
+#define __GST_RTSP_ADDRESS_POOL_H__
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_ADDRESS_POOL              (gst_rtsp_address_pool_get_type ())
+#define GST_IS_RTSP_ADDRESS_POOL(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_ADDRESS_POOL))
+#define GST_IS_RTSP_ADDRESS_POOL_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_ADDRESS_POOL))
+#define GST_RTSP_ADDRESS_POOL_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_ADDRESS_POOL, GstRTSPAddressPoolClass))
+#define GST_RTSP_ADDRESS_POOL(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_ADDRESS_POOL, GstRTSPAddressPool))
+#define GST_RTSP_ADDRESS_POOL_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_ADDRESS_POOL, GstRTSPAddressPoolClass))
+#define GST_RTSP_ADDRESS_POOL_CAST(obj)         ((GstRTSPAddressPool*)(obj))
+#define GST_RTSP_ADDRESS_POOL_CLASS_CAST(klass) ((GstRTSPAddressPoolClass*)(klass))
+
+/**
+ * GstRTSPAddressPoolResult:
+ * @GST_RTSP_ADDRESS_POOL_OK: no error
+ * @GST_RTSP_ADDRESS_POOL_EINVAL:invalid arguments were provided to a function
+ * @GST_RTSP_ADDRESS_POOL_ERESERVED: the addres has already been reserved
+ * @GST_RTSP_ADDRESS_POOL_ERANGE: the address is not in the pool
+ * @GST_RTSP_ADDRESS_POOL_ELAST: last error
+ *
+ * Result codes from RTSP address pool functions.
+ */
+typedef enum {
+  GST_RTSP_ADDRESS_POOL_OK         =  0,
+  /* errors */
+  GST_RTSP_ADDRESS_POOL_EINVAL     = -1,
+  GST_RTSP_ADDRESS_POOL_ERESERVED  = -2,
+  GST_RTSP_ADDRESS_POOL_ERANGE     = -3,
+
+  GST_RTSP_ADDRESS_POOL_ELAST      = -4,
+} GstRTSPAddressPoolResult;
+
+
+typedef struct _GstRTSPAddress GstRTSPAddress;
+
+typedef struct _GstRTSPAddressPool GstRTSPAddressPool;
+typedef struct _GstRTSPAddressPoolClass GstRTSPAddressPoolClass;
+typedef struct _GstRTSPAddressPoolPrivate GstRTSPAddressPoolPrivate;
+
+/**
+ * GstRTSPAddress:
+ * @pool: the #GstRTSPAddressPool owner of this address
+ * @address: the address
+ * @port: the port number
+ * @n_ports: number of ports
+ * @ttl: TTL or 0 for unicast addresses
+ *
+ * An address
+ */
+struct _GstRTSPAddress {
+  GstRTSPAddressPool *pool;
+
+  gchar *address;
+  guint16 port;
+  gint n_ports;
+  guint8 ttl;
+
+  /**/
+  gpointer priv;
+};
+
+GType gst_rtsp_address_get_type        (void);
+
+GstRTSPAddress * gst_rtsp_address_copy (GstRTSPAddress *addr);
+void             gst_rtsp_address_free (GstRTSPAddress *addr);
+
+/**
+ * GstRTSPAddressFlags:
+ * @GST_RTSP_ADDRESS_FLAG_NONE: no flags
+ * @GST_RTSP_ADDRESS_FLAG_IPV4: an IPv4 address
+ * @GST_RTSP_ADDRESS_FLAG_IPV6: and IPv6 address
+ * @GST_RTSP_ADDRESS_FLAG_EVEN_PORT: address with an even port
+ * @GST_RTSP_ADDRESS_FLAG_MULTICAST: a multicast address
+ * @GST_RTSP_ADDRESS_FLAG_UNICAST: a unicast address
+ *
+ * Flags used to control allocation of addresses
+ */
+typedef enum {
+  GST_RTSP_ADDRESS_FLAG_NONE      = 0,
+  GST_RTSP_ADDRESS_FLAG_IPV4      = (1 << 0),
+  GST_RTSP_ADDRESS_FLAG_IPV6      = (1 << 1),
+  GST_RTSP_ADDRESS_FLAG_EVEN_PORT = (1 << 2),
+  GST_RTSP_ADDRESS_FLAG_MULTICAST = (1 << 3),
+  GST_RTSP_ADDRESS_FLAG_UNICAST   = (1 << 4),
+} GstRTSPAddressFlags;
+
+/**
+ * GST_RTSP_ADDRESS_POOL_ANY_IPV4:
+ *
+ * Used with gst_rtsp_address_pool_add_range() to bind to all
+ * IPv4 addresses
+ */
+#define GST_RTSP_ADDRESS_POOL_ANY_IPV4  "0.0.0.0"
+
+/**
+ * GST_RTSP_ADDRESS_POOL_ANY_IPV6:
+ *
+ * Used with gst_rtsp_address_pool_add_range() to bind to all
+ * IPv6 addresses
+ */
+#define GST_RTSP_ADDRESS_POOL_ANY_IPV6  "::"
+
+/**
+ * GstRTSPAddressPool:
+ * @parent: the parent GObject
+ *
+ * An address pool, all member are private
+ */
+struct _GstRTSPAddressPool {
+  GObject       parent;
+
+  /*< private >*/
+  GstRTSPAddressPoolPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPAddressPoolClass:
+ *
+ * Opaque Address pool class.
+ */
+struct _GstRTSPAddressPoolClass {
+  GObjectClass  parent_class;
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+GType                  gst_rtsp_address_pool_get_type        (void);
+
+/* create a new address pool */
+GstRTSPAddressPool *   gst_rtsp_address_pool_new             (void);
+
+void                   gst_rtsp_address_pool_clear           (GstRTSPAddressPool * pool);
+void                   gst_rtsp_address_pool_dump            (GstRTSPAddressPool * pool);
+
+gboolean               gst_rtsp_address_pool_add_range       (GstRTSPAddressPool * pool,
+                                                              const gchar *min_address,
+                                                              const gchar *max_address,
+                                                              guint16 min_port,
+                                                              guint16 max_port,
+                                                              guint8 ttl);
+
+GstRTSPAddress *       gst_rtsp_address_pool_acquire_address (GstRTSPAddressPool * pool,
+                                                              GstRTSPAddressFlags flags,
+                                                              gint n_ports);
+
+GstRTSPAddressPoolResult  gst_rtsp_address_pool_reserve_address (GstRTSPAddressPool * pool,
+                                                              const gchar *ip_address,
+                                                              guint port,
+                                                              guint n_ports,
+                                                              guint ttl,
+                                                              GstRTSPAddress ** address);
+
+gboolean               gst_rtsp_address_pool_has_unicast_addresses (GstRTSPAddressPool * pool);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_ADDRESS_POOL_H__ */
diff --git a/gst/rtsp-server/rtsp-auth.c b/gst/rtsp-server/rtsp-auth.c
new file mode 100644
index 0000000..b0e924e
--- /dev/null
+++ b/gst/rtsp-server/rtsp-auth.c
@@ -0,0 +1,636 @@
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-auth
+ * @short_description: Authentication and authorization
+ * @see_also: #GstRTSPPermissions, #GstRTSPToken
+ *
+ * The #GstRTSPAuth object is responsible for checking if the current user is
+ * allowed to perform requested actions. The default implementation has some
+ * reasonable checks but subclasses can implement custom security policies.
+ *
+ * A new auth object is made with gst_rtsp_auth_new(). It is usually configured
+ * on the #GstRTSPServer object.
+ *
+ * The RTSP server will call gst_rtsp_auth_check() with a string describing the
+ * check to perform. The possible checks are prefixed with
+ * GST_RTSP_AUTH_CHECK_*. Depending on the check, the default implementation
+ * will use the current #GstRTSPToken, #GstRTSPContext and
+ * #GstRTSPPermissions on the object to check if an operation is allowed.
+ *
+ * The default #GstRTSPAuth object has support for basic authentication. With
+ * gst_rtsp_auth_add_basic() you can add a basic authentication string together
+ * with the #GstRTSPToken that will become active when successfully
+ * authenticated.
+ *
+ * When a TLS certificate has been set with gst_rtsp_auth_set_tls_certificate(),
+ * the default auth object will require the client to connect with a TLS
+ * connection.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
+
+#include 
+
+#include "rtsp-auth.h"
+
+#define GST_RTSP_AUTH_GET_PRIVATE(obj)  \
+   (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_AUTH, GstRTSPAuthPrivate))
+
+struct _GstRTSPAuthPrivate
+{
+  GMutex lock;
+
+  /* the TLS certificate */
+  GTlsCertificate *certificate;
+  GHashTable *basic;            /* protected by lock */
+  GstRTSPToken *default_token;
+  GstRTSPMethod methods;
+};
+
+enum
+{
+  PROP_0,
+  PROP_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_auth_debug);
+#define GST_CAT_DEFAULT rtsp_auth_debug
+
+static void gst_rtsp_auth_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec);
+static void gst_rtsp_auth_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_auth_finalize (GObject * obj);
+
+static gboolean default_authenticate (GstRTSPAuth * auth, GstRTSPContext * ctx);
+static gboolean default_check (GstRTSPAuth * auth, GstRTSPContext * ctx,
+    const gchar * check);
+
+G_DEFINE_TYPE (GstRTSPAuth, gst_rtsp_auth, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_auth_class_init (GstRTSPAuthClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPAuthPrivate));
+
+  gobject_class = G_OBJECT_CLASS (klass);
+
+  gobject_class->get_property = gst_rtsp_auth_get_property;
+  gobject_class->set_property = gst_rtsp_auth_set_property;
+  gobject_class->finalize = gst_rtsp_auth_finalize;
+
+  klass->authenticate = default_authenticate;
+  klass->check = default_check;
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_auth_debug, "rtspauth", 0, "GstRTSPAuth");
+}
+
+static void
+gst_rtsp_auth_init (GstRTSPAuth * auth)
+{
+  GstRTSPAuthPrivate *priv;
+
+  auth->priv = priv = GST_RTSP_AUTH_GET_PRIVATE (auth);
+
+  g_mutex_init (&priv->lock);
+
+  priv->basic = g_hash_table_new_full (g_str_hash, g_str_equal, g_free,
+      (GDestroyNotify) gst_rtsp_token_unref);
+
+  /* bitwise or of all methods that need authentication */
+  priv->methods = 0;
+}
+
+static void
+gst_rtsp_auth_finalize (GObject * obj)
+{
+  GstRTSPAuth *auth = GST_RTSP_AUTH (obj);
+  GstRTSPAuthPrivate *priv = auth->priv;
+
+  GST_INFO ("finalize auth %p", auth);
+
+  if (priv->certificate)
+    g_object_unref (priv->certificate);
+  g_hash_table_unref (priv->basic);
+  g_mutex_clear (&priv->lock);
+
+  G_OBJECT_CLASS (gst_rtsp_auth_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_auth_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec)
+{
+  switch (propid) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+static void
+gst_rtsp_auth_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec)
+{
+  switch (propid) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+/**
+ * gst_rtsp_auth_new:
+ *
+ * Create a new #GstRTSPAuth instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPAuth
+ */
+GstRTSPAuth *
+gst_rtsp_auth_new (void)
+{
+  GstRTSPAuth *result;
+
+  result = g_object_new (GST_TYPE_RTSP_AUTH, NULL);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_auth_set_tls_certificate:
+ * @auth: a #GstRTSPAuth
+ * @cert: (transfer none) (allow-none): a #GTlsCertificate
+ *
+ * Set the TLS certificate for the auth. Client connections will only
+ * be accepted when TLS is negotiated.
+ */
+void
+gst_rtsp_auth_set_tls_certificate (GstRTSPAuth * auth, GTlsCertificate * cert)
+{
+  GstRTSPAuthPrivate *priv;
+  GTlsCertificate *old;
+
+  g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+
+  priv = auth->priv;
+
+  if (cert)
+    g_object_ref (cert);
+
+  g_mutex_lock (&priv->lock);
+  old = priv->certificate;
+  priv->certificate = cert;
+  g_mutex_unlock (&priv->lock);
+
+  if (old)
+    g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_auth_get_tls_certificate:
+ * @auth: a #GstRTSPAuth
+ *
+ * Get the #GTlsCertificate used for negotiating TLS @auth.
+ *
+ * Returns: (transfer full): the #GTlsCertificate of @auth. g_object_unref() after
+ * usage.
+ */
+GTlsCertificate *
+gst_rtsp_auth_get_tls_certificate (GstRTSPAuth * auth)
+{
+  GstRTSPAuthPrivate *priv;
+  GTlsCertificate *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_AUTH (auth), NULL);
+
+  priv = auth->priv;
+
+  g_mutex_lock (&priv->lock);
+  if ((result = priv->certificate))
+    g_object_ref (result);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_auth_set_default_token:
+ * @auth: a #GstRTSPAuth
+ * @token: (transfer none) (allow-none): a #GstRTSPToken
+ *
+ * Set the default #GstRTSPToken to @token in @auth. The default token will
+ * be used for unauthenticated users.
+ */
+void
+gst_rtsp_auth_set_default_token (GstRTSPAuth * auth, GstRTSPToken * token)
+{
+  GstRTSPAuthPrivate *priv;
+  GstRTSPToken *old;
+
+  g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+
+  priv = auth->priv;
+
+  if (token)
+    gst_rtsp_token_ref (token);
+
+  g_mutex_lock (&priv->lock);
+  old = priv->default_token;
+  priv->default_token = token;
+  g_mutex_unlock (&priv->lock);
+
+  if (old)
+    gst_rtsp_token_unref (old);
+}
+
+/**
+ * gst_rtsp_auth_get_default_token:
+ * @auth: a #GstRTSPAuth
+ *
+ * Get the default token for @auth. This token will be used for unauthenticated
+ * users.
+ *
+ * Returns: (transfer full): the #GstRTSPToken of @auth. gst_rtsp_token_unref() after
+ * usage.
+ */
+GstRTSPToken *
+gst_rtsp_auth_get_default_token (GstRTSPAuth * auth)
+{
+  GstRTSPAuthPrivate *priv;
+  GstRTSPToken *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_AUTH (auth), NULL);
+
+  priv = auth->priv;
+
+  g_mutex_lock (&priv->lock);
+  if ((result = priv->default_token))
+    gst_rtsp_token_ref (result);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_auth_add_basic:
+ * @auth: a #GstRTSPAuth
+ * @basic: the basic token
+ * @token: (transfer none): authorisation token
+ *
+ * Add a basic token for the default authentication algorithm that
+ * enables the client with privileges listed in @token.
+ */
+void
+gst_rtsp_auth_add_basic (GstRTSPAuth * auth, const gchar * basic,
+    GstRTSPToken * token)
+{
+  GstRTSPAuthPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+  g_return_if_fail (basic != NULL);
+  g_return_if_fail (GST_IS_RTSP_TOKEN (token));
+
+  priv = auth->priv;
+
+  g_mutex_lock (&priv->lock);
+  g_hash_table_replace (priv->basic, g_strdup (basic),
+      gst_rtsp_token_ref (token));
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_auth_remove_basic:
+ * @auth: a #GstRTSPAuth
+ * @basic: (transfer none): the basic token
+ *
+ * Add a basic token for the default authentication algorithm that
+ * enables the client with privileges from @authgroup.
+ */
+void
+gst_rtsp_auth_remove_basic (GstRTSPAuth * auth, const gchar * basic)
+{
+  GstRTSPAuthPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_AUTH (auth));
+  g_return_if_fail (basic != NULL);
+
+  priv = auth->priv;
+
+  g_mutex_lock (&priv->lock);
+  g_hash_table_remove (priv->basic, basic);
+  g_mutex_unlock (&priv->lock);
+}
+
+static gboolean
+default_authenticate (GstRTSPAuth * auth, GstRTSPContext * ctx)
+{
+  GstRTSPAuthPrivate *priv = auth->priv;
+  GstRTSPResult res;
+  gchar *authorization;
+
+  GST_DEBUG_OBJECT (auth, "authenticate");
+
+  g_mutex_lock (&priv->lock);
+  /* FIXME, need to ref but we have no way to unref when the ctx is
+   * popped */
+  ctx->token = priv->default_token;
+  g_mutex_unlock (&priv->lock);
+
+  res =
+      gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_AUTHORIZATION,
+      &authorization, 0);
+  if (res < 0)
+    goto no_auth;
+
+  /* parse type */
+  if (g_ascii_strncasecmp (authorization, "basic ", 6) == 0) {
+    GstRTSPToken *token;
+
+    GST_DEBUG_OBJECT (auth, "check Basic auth");
+    g_mutex_lock (&priv->lock);
+    if ((token = g_hash_table_lookup (priv->basic, &authorization[6]))) {
+      GST_DEBUG_OBJECT (auth, "setting token %p", token);
+      ctx->token = token;
+    }
+    g_mutex_unlock (&priv->lock);
+  } else if (g_ascii_strncasecmp (authorization, "digest ", 7) == 0) {
+    GST_DEBUG_OBJECT (auth, "check Digest auth");
+    /* not implemented yet */
+  }
+  return TRUE;
+
+no_auth:
+  {
+    GST_DEBUG_OBJECT (auth, "no authorization header found");
+    return TRUE;
+  }
+}
+
+static void
+send_response (GstRTSPAuth * auth, GstRTSPStatusCode code, GstRTSPContext * ctx)
+{
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
+
+  if (code == GST_RTSP_STS_UNAUTHORIZED) {
+    /* we only have Basic for now */
+    gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_WWW_AUTHENTICATE,
+        "Basic realm=\"GStreamer RTSP Server\"");
+  }
+  gst_rtsp_client_send_message (ctx->client, ctx->session, ctx->response);
+}
+
+static gboolean
+ensure_authenticated (GstRTSPAuth * auth, GstRTSPContext * ctx)
+{
+  GstRTSPAuthClass *klass;
+
+  klass = GST_RTSP_AUTH_GET_CLASS (auth);
+
+  /* we need a token to check */
+  if (ctx->token == NULL) {
+    if (klass->authenticate) {
+      if (!klass->authenticate (auth, ctx))
+        goto authenticate_failed;
+    }
+  }
+  if (ctx->token == NULL)
+    goto no_auth;
+
+  return TRUE;
+
+/* ERRORS */
+authenticate_failed:
+  {
+    GST_DEBUG_OBJECT (auth, "authenticate failed");
+    send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
+    return FALSE;
+  }
+no_auth:
+  {
+    GST_DEBUG_OBJECT (auth, "no authorization token found");
+    send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
+    return FALSE;
+  }
+}
+
+/* new connection */
+static gboolean
+check_connect (GstRTSPAuth * auth, GstRTSPContext * ctx, const gchar * check)
+{
+  GstRTSPAuthPrivate *priv = auth->priv;
+
+  if (priv->certificate) {
+    GTlsConnection *tls;
+
+    /* configure the connection */
+    tls = gst_rtsp_connection_get_tls (ctx->conn, NULL);
+    g_tls_connection_set_certificate (tls, priv->certificate);
+  }
+  return TRUE;
+}
+
+/* check url and methods */
+static gboolean
+check_url (GstRTSPAuth * auth, GstRTSPContext * ctx, const gchar * check)
+{
+  GstRTSPAuthPrivate *priv = auth->priv;
+
+  if ((ctx->method & priv->methods) != 0)
+    if (!ensure_authenticated (auth, ctx))
+      goto not_authenticated;
+
+  return TRUE;
+
+  /* ERRORS */
+not_authenticated:
+  {
+    return FALSE;
+  }
+}
+
+/* check access to media factory */
+static gboolean
+check_factory (GstRTSPAuth * auth, GstRTSPContext * ctx, const gchar * check)
+{
+  const gchar *role;
+  GstRTSPPermissions *perms;
+
+  if (!ensure_authenticated (auth, ctx))
+    return FALSE;
+
+  if (!(role = gst_rtsp_token_get_string (ctx->token,
+              GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE)))
+    goto no_media_role;
+  if (!(perms = gst_rtsp_media_factory_get_permissions (ctx->factory)))
+    goto no_permissions;
+
+  if (g_str_equal (check, GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS)) {
+    if (!gst_rtsp_permissions_is_allowed (perms, role,
+            GST_RTSP_PERM_MEDIA_FACTORY_ACCESS))
+      goto no_access;
+  } else if (g_str_equal (check, GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT)) {
+    if (!gst_rtsp_permissions_is_allowed (perms, role,
+            GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT))
+      goto no_construct;
+  }
+
+  gst_rtsp_permissions_unref (perms);
+
+  return TRUE;
+
+  /* ERRORS */
+no_media_role:
+  {
+    GST_DEBUG_OBJECT (auth, "no media factory role found");
+    send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
+    return FALSE;
+  }
+no_permissions:
+  {
+    GST_DEBUG_OBJECT (auth, "no permissions on media factory found");
+    send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
+    return FALSE;
+  }
+no_access:
+  {
+    GST_DEBUG_OBJECT (auth, "no permissions to access media factory");
+    gst_rtsp_permissions_unref (perms);
+    send_response (auth, GST_RTSP_STS_NOT_FOUND, ctx);
+    return FALSE;
+  }
+no_construct:
+  {
+    GST_DEBUG_OBJECT (auth, "no permissions to construct media factory");
+    gst_rtsp_permissions_unref (perms);
+    send_response (auth, GST_RTSP_STS_UNAUTHORIZED, ctx);
+    return FALSE;
+  }
+}
+
+static gboolean
+check_client_settings (GstRTSPAuth * auth, GstRTSPContext * ctx,
+    const gchar * check)
+{
+  if (!ensure_authenticated (auth, ctx))
+    return FALSE;
+
+  return gst_rtsp_token_is_allowed (ctx->token,
+      GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS);
+}
+
+static gboolean
+default_check (GstRTSPAuth * auth, GstRTSPContext * ctx, const gchar * check)
+{
+  gboolean res = FALSE;
+
+  /* FIXME, use hastable or so */
+  if (g_str_equal (check, GST_RTSP_AUTH_CHECK_CONNECT)) {
+    res = check_connect (auth, ctx, check);
+  } else if (g_str_equal (check, GST_RTSP_AUTH_CHECK_URL)) {
+    res = check_url (auth, ctx, check);
+  } else if (g_str_has_prefix (check, "auth.check.media.factory.")) {
+    res = check_factory (auth, ctx, check);
+  } else if (g_str_equal (check, GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS)) {
+    res = check_client_settings (auth, ctx, check);
+  }
+  return res;
+}
+
+static gboolean
+no_auth_check (const gchar * check)
+{
+  gboolean res;
+
+  if (g_str_equal (check, GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS))
+    res = FALSE;
+  else
+    res = TRUE;
+
+  return res;
+}
+
+/**
+ * gst_rtsp_auth_check:
+ * @check: the item to check
+ *
+ * Check if @check is allowed in the current context.
+ *
+ * Returns: FALSE if check failed.
+ */
+gboolean
+gst_rtsp_auth_check (const gchar * check)
+{
+  gboolean result = FALSE;
+  GstRTSPAuthClass *klass;
+  GstRTSPContext *ctx;
+  GstRTSPAuth *auth;
+
+  g_return_val_if_fail (check != NULL, FALSE);
+
+  if (!(ctx = gst_rtsp_context_get_current ()))
+    goto no_context;
+
+  /* no auth, we don't need to check */
+  if (!(auth = ctx->auth))
+    return no_auth_check (check);
+
+  klass = GST_RTSP_AUTH_GET_CLASS (auth);
+
+  GST_DEBUG_OBJECT (auth, "check authorization '%s'", check);
+
+  if (klass->check)
+    result = klass->check (auth, ctx, check);
+
+  return result;
+
+  /* ERRORS */
+no_context:
+  {
+    GST_ERROR ("no context found");
+    return FALSE;
+  }
+}
+
+/**
+ * gst_rtsp_auth_make_basic:
+ * @user: a userid
+ * @pass: a password
+ *
+ * Construct a Basic authorisation token from @user and @pass.
+ *
+ * Returns: (transfer full): the base64 encoding of the string @user:@pass.
+ * g_free() after usage.
+ */
+gchar *
+gst_rtsp_auth_make_basic (const gchar * user, const gchar * pass)
+{
+  gchar *user_pass;
+  gchar *result;
+
+  g_return_val_if_fail (user != NULL, NULL);
+  g_return_val_if_fail (pass != NULL, NULL);
+
+  user_pass = g_strjoin (":", user, pass, NULL);
+  result = g_base64_encode ((guchar *) user_pass, strlen (user_pass));
+  g_free (user_pass);
+
+  return result;
+}
diff --git a/gst/rtsp-server/rtsp-auth.h b/gst/rtsp-server/rtsp-auth.h
new file mode 100644
index 0000000..cb281f0
--- /dev/null
+++ b/gst/rtsp-server/rtsp-auth.h
@@ -0,0 +1,173 @@
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#ifndef __GST_RTSP_AUTH_H__
+#define __GST_RTSP_AUTH_H__
+
+typedef struct _GstRTSPAuth GstRTSPAuth;
+typedef struct _GstRTSPAuthClass GstRTSPAuthClass;
+typedef struct _GstRTSPAuthPrivate GstRTSPAuthPrivate;
+
+#include "rtsp-client.h"
+#include "rtsp-token.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_AUTH              (gst_rtsp_auth_get_type ())
+#define GST_IS_RTSP_AUTH(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_AUTH))
+#define GST_IS_RTSP_AUTH_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_AUTH))
+#define GST_RTSP_AUTH_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_AUTH, GstRTSPAuthClass))
+#define GST_RTSP_AUTH(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_AUTH, GstRTSPAuth))
+#define GST_RTSP_AUTH_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_AUTH, GstRTSPAuthClass))
+#define GST_RTSP_AUTH_CAST(obj)         ((GstRTSPAuth*)(obj))
+#define GST_RTSP_AUTH_CLASS_CAST(klass) ((GstRTSPAuthClass*)(klass))
+
+/**
+ * GstRTSPAuth:
+ *
+ * The authentication structure.
+ */
+struct _GstRTSPAuth {
+  GObject       parent;
+
+  /*< private >*/
+  GstRTSPAuthPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPAuthClass:
+ * @authenticate: check the authentication of a client. The default implementation
+ *         checks if the authentication in the header matches one of the basic
+ *         authentication tokens. This function should set the authgroup field
+ *         in the context.
+ * @check: check if a resource can be accessed. this function should
+ *         call authenticate to authenticate the client when needed. The method
+ *         should also construct and send an appropriate response message on
+ *         error.
+ *
+ * The authentication class.
+ */
+struct _GstRTSPAuthClass {
+  GObjectClass  parent_class;
+
+  gboolean           (*authenticate) (GstRTSPAuth *auth, GstRTSPContext *ctx);
+  gboolean           (*check)        (GstRTSPAuth *auth, GstRTSPContext *ctx,
+                                      const gchar *check);
+  /*< private >*/
+  gpointer            _gst_reserved[GST_PADDING];
+};
+
+GType               gst_rtsp_auth_get_type          (void);
+
+GstRTSPAuth *       gst_rtsp_auth_new               (void);
+
+void                gst_rtsp_auth_set_tls_certificate (GstRTSPAuth *auth, GTlsCertificate *cert);
+GTlsCertificate *   gst_rtsp_auth_get_tls_certificate (GstRTSPAuth *auth);
+
+void                gst_rtsp_auth_set_default_token (GstRTSPAuth *auth, GstRTSPToken *token);
+GstRTSPToken *      gst_rtsp_auth_get_default_token (GstRTSPAuth *auth);
+
+void                gst_rtsp_auth_add_basic         (GstRTSPAuth *auth, const gchar * basic,
+                                                     GstRTSPToken *token);
+void                gst_rtsp_auth_remove_basic      (GstRTSPAuth *auth, const gchar * basic);
+
+gboolean            gst_rtsp_auth_check             (const gchar *check);
+
+
+/* helpers */
+gchar *             gst_rtsp_auth_make_basic        (const gchar * user, const gchar * pass);
+
+/* checks */
+/**
+ * GST_RTSP_AUTH_CHECK_CONNECT:
+ *
+ * Check a new connection
+ */
+#define GST_RTSP_AUTH_CHECK_CONNECT                  "auth.check.connect"
+/**
+ * GST_RTSP_AUTH_CHECK_URL:
+ *
+ * Check the URL and methods
+ */
+#define GST_RTSP_AUTH_CHECK_URL                      "auth.check.url"
+/**
+ * GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS:
+ *
+ * Check if access is allowed to a factory.
+ * When access is not allowed an 404 Not Found is sent in the response.
+ */
+#define GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS     "auth.check.media.factory.access"
+/**
+ * GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT:
+ *
+ * Check if media can be constructed from a media factory
+ * A response should be sent on error.
+ */
+#define GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT  "auth.check.media.factory.construct"
+/**
+ * GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS:
+ *
+ * Check if the client can specify TTL, destination and
+ * port pair in multicast. No response is sent when the check returns
+ * %FALSE.
+ */
+#define GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS  "auth.check.transport.client-settings"
+
+
+/* tokens */
+/**
+ * GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE:
+ *
+ * G_TYPE_STRING, the role to use when dealing with media factories
+ *
+ * The default #GstRTSPAuth object uses this string in the token to find the
+ * role of the media factory. It will then retrieve the #GstRTSPPermissions of
+ * the media factory and retrieve the role with the same name.
+ */
+#define GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE      "media.factory.role"
+/**
+ * GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS:
+ *
+ * G_TYPE_BOOLEAN, %TRUE if the client can specify TTL, destination and
+ *     port pair in multicast.
+ */
+#define GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS   "transport.client-settings"
+
+/* permissions */
+/**
+ * GST_RTSP_PERM_MEDIA_FACTORY_ACCESS:
+ *
+ * G_TYPE_BOOLEAN, %TRUE if the media can be accessed, %FALSE will
+ * return a 404 Not Found error when trying to access the media.
+ */
+#define GST_RTSP_PERM_MEDIA_FACTORY_ACCESS      "media.factory.access"
+/**
+ * GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT:
+ *
+ * G_TYPE_BOOLEAN, %TRUE if the media can be constructed, %FALSE will
+ * return a 404 Not Found error when trying to access the media.
+ */
+#define GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT   "media.factory.construct"
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_AUTH_H__ */
diff --git a/gst/rtsp-server/rtsp-client-wfd.c b/gst/rtsp-server/rtsp-client-wfd.c
new file mode 100644
index 0000000..e594160
--- /dev/null
+++ b/gst/rtsp-server/rtsp-client-wfd.c
@@ -0,0 +1,2950 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-client
+ * @short_description: A client connection state
+ * @see_also: #GstRTSPServer, #GstRTSPThreadPool
+ *
+ * The client object handles the connection with a client for as long as a TCP
+ * connection is open.
+ *
+ * A #GstRTSPWFDClient is created by #GstRTSPServer when a new connection is
+ * accepted and it inherits the #GstRTSPMountPoints, #GstRTSPSessionPool,
+ * #GstRTSPAuth and #GstRTSPThreadPool from the server.
+ *
+ * The client connection should be configured with the #GstRTSPConnection using
+ * gst_rtsp_wfd_client_set_connection() before it can be attached to a #GMainContext
+ * using gst_rtsp_wfd_client_attach(). From then on the client will handle requests
+ * on the connection.
+ *
+ * Use gst_rtsp_wfd_client_session_filter() to iterate or modify all the
+ * #GstRTSPSession objects managed by the client object.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include 
+#include 
+
+#include "rtsp-client-wfd.h"
+#include "rtsp-media-factory-wfd.h"
+#include "rtsp-sdp.h"
+#include "rtsp-params.h"
+
+#define GST_RTSP_WFD_CLIENT_GET_PRIVATE(obj)  \
+   (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_WFD_CLIENT, GstRTSPWFDClientPrivate))
+
+typedef struct _GstRTSPClientRTPStats GstRTSPClientRTPStats;
+
+struct _GstRTSPClientRTPStats {
+  GstRTSPStream *stream;
+  guint64 last_sent_bytes;
+  guint64 sent_bytes;
+  guint last_seqnum;
+  guint seqnum;
+
+  /* Info in RR (Receiver Report) */
+  guint8 fraction_lost;
+  guint32 cumulative_lost_num;
+  guint16 max_seqnum;
+  guint32 arrival_jitter;
+  guint32 lsr;
+  guint32 dlsr;
+  guint32 rtt;
+  guint resent_packets;
+};
+
+struct _GstRTSPWFDClientPrivate
+{
+  GstRTSPWFDClientSendFunc send_func;   /* protected by send_lock */
+  gpointer send_data;           /* protected by send_lock */
+  GDestroyNotify send_notify;   /* protected by send_lock */
+
+  /* used to cache the media in the last requested DESCRIBE so that
+   * we can pick it up in the next SETUP immediately */
+  gchar *path;
+  GstRTSPMedia *media;
+
+  GList *transports;
+  GList *sessions;
+
+  guint8 m1_done;
+  guint8 m3_done;
+  guint8 m4_done;
+
+  /* Host's URL info */
+  gchar *host_address;
+
+  /* Parameters for WIFI-DISPLAY */
+  guint caCodec;
+  guint8 audio_codec;
+  guint cFreq;
+  guint cChanels;
+  guint cBitwidth;
+  guint caLatency;
+  guint cvCodec;
+  guint cNative;
+  guint64 cNativeResolution;
+  guint64 video_resolution_supported;
+  gint video_native_resolution;
+  guint64 cCEAResolution;
+  guint64 cVESAResolution;
+  guint64 cHHResolution;
+  guint cProfile;
+  guint cLevel;
+  guint32 cMaxHeight;
+  guint32 cMaxWidth;
+  guint32 cFramerate;
+  guint32 cInterleaved;
+  guint32 cmin_slice_size;
+  guint32 cslice_enc_params;
+  guint cframe_rate_control;
+  guint cvLatency;
+  guint ctrans;
+  guint cprofile;
+  guint clowertrans;
+  guint32 crtp_port0;
+  guint32 crtp_port1;
+
+  gboolean protection_enabled;
+  GstWFDHDCPProtection hdcp_version;
+  guint32 hdcp_tcpport;
+
+  gboolean edid_supported;
+  guint32 edid_hres;
+  guint32 edid_vres;
+
+  gboolean keep_alive_flag;
+  GMutex keep_alive_lock;
+
+  /* RTP statistics */
+  GstRTSPClientRTPStats stats;
+  GMutex stats_lock;
+  guint stats_timer_id;
+  gboolean rtcp_stats_enabled;
+};
+
+#define DEFAULT_WFD_TIMEOUT 60
+#define WFD_MOUNT_POINT "/wfd1.0/streamid=0"
+
+enum
+{
+  SIGNAL_WFD_OPTIONS_REQUEST,
+  SIGNAL_WFD_GET_PARAMETER_REQUEST,
+  SIGNAL_WFD_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_wfd_client_debug);
+#define GST_CAT_DEFAULT rtsp_wfd_client_debug
+
+static guint gst_rtsp_client_wfd_signals[SIGNAL_WFD_LAST] = { 0 };
+
+static void gst_rtsp_wfd_client_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec);
+static void gst_rtsp_wfd_client_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_wfd_client_finalize (GObject * obj);
+
+static gboolean handle_wfd_options_request (GstRTSPClient * client,
+    GstRTSPContext * ctx);
+static gboolean handle_wfd_set_param_request (GstRTSPClient * client,
+    GstRTSPContext * ctx);
+static gboolean handle_wfd_get_param_request (GstRTSPClient * client,
+    GstRTSPContext * ctx);
+
+static void send_generic_wfd_response (GstRTSPWFDClient * client,
+    GstRTSPStatusCode code, GstRTSPContext * ctx);
+static gchar *wfd_make_path_from_uri (GstRTSPClient * client,
+    const GstRTSPUrl * uri);
+static void wfd_options_request_done (GstRTSPWFDClient * client, GstRTSPContext *ctx);
+static void wfd_get_param_request_done (GstRTSPWFDClient * client, GstRTSPContext *ctx);
+static void handle_wfd_response (GstRTSPClient * client, GstRTSPContext * ctx);
+static void handle_wfd_play (GstRTSPClient * client, GstRTSPContext * ctx);
+static void wfd_set_keep_alive_condition(GstRTSPWFDClient * client);
+static gboolean wfd_ckeck_keep_alive_response (gpointer userdata);
+static gboolean keep_alive_condition(gpointer userdata);
+static gboolean wfd_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
+    GstRTSPStream * stream, GstRTSPContext * ctx);
+
+GstRTSPResult prepare_trigger_request (GstRTSPWFDClient * client,
+    GstRTSPMessage * request, GstWFDTriggerType trigger_type, gchar * url);
+
+GstRTSPResult prepare_request (GstRTSPWFDClient * client,
+    GstRTSPMessage * request, GstRTSPMethod method, gchar * url);
+
+void
+send_request (GstRTSPWFDClient * client, GstRTSPSession * session,
+    GstRTSPMessage * request);
+
+GstRTSPResult
+prepare_response (GstRTSPWFDClient * client, GstRTSPMessage * request,
+    GstRTSPMessage * response, GstRTSPMethod method);
+
+static GstRTSPResult handle_M1_message (GstRTSPWFDClient * client);
+static GstRTSPResult handle_M3_message (GstRTSPWFDClient * client);
+static GstRTSPResult handle_M4_message (GstRTSPWFDClient * client);
+static GstRTSPResult handle_M16_message (GstRTSPWFDClient * client);
+
+G_DEFINE_TYPE (GstRTSPWFDClient, gst_rtsp_wfd_client, GST_TYPE_RTSP_CLIENT);
+
+static void
+gst_rtsp_wfd_client_class_init (GstRTSPWFDClientClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstRTSPClientClass *rtsp_client_class;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPWFDClientPrivate));
+
+  gobject_class = G_OBJECT_CLASS (klass);
+  rtsp_client_class = GST_RTSP_CLIENT_CLASS (klass);
+
+  gobject_class->get_property = gst_rtsp_wfd_client_get_property;
+  gobject_class->set_property = gst_rtsp_wfd_client_set_property;
+  gobject_class->finalize = gst_rtsp_wfd_client_finalize;
+
+  //klass->create_sdp = create_sdp;
+  //klass->configure_client_transport = default_configure_client_transport;
+  //klass->params_set = default_params_set;
+  //klass->params_get = default_params_get;
+
+  rtsp_client_class->handle_options_request = handle_wfd_options_request;
+  rtsp_client_class->handle_set_param_request = handle_wfd_set_param_request;
+  rtsp_client_class->handle_get_param_request = handle_wfd_get_param_request;
+  rtsp_client_class->make_path_from_uri = wfd_make_path_from_uri;
+  rtsp_client_class->configure_client_media = wfd_configure_client_media;
+
+  rtsp_client_class->handle_response = handle_wfd_response;
+  rtsp_client_class->play_request = handle_wfd_play;
+
+  gst_rtsp_client_wfd_signals[SIGNAL_WFD_OPTIONS_REQUEST] =
+      g_signal_new ("wfd-options-request", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPWFDClientClass,
+          wfd_options_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
+      G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+  gst_rtsp_client_wfd_signals[SIGNAL_WFD_GET_PARAMETER_REQUEST] =
+      g_signal_new ("wfd-get-parameter-request", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPWFDClientClass,
+          wfd_get_param_request), NULL, NULL, g_cclosure_marshal_VOID__POINTER,
+      G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+  klass->wfd_options_request = wfd_options_request_done;
+  klass->wfd_get_param_request = wfd_get_param_request_done;
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_wfd_client_debug, "rtspwfdclient", 0,
+      "GstRTSPWFDClient");
+}
+
+static void
+gst_rtsp_wfd_client_init (GstRTSPWFDClient * client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+  g_return_if_fail (priv != NULL);
+
+  client->priv = priv;
+  priv->protection_enabled = FALSE;
+  priv->video_native_resolution = GST_WFD_VIDEO_CEA_RESOLUTION;
+  priv->video_resolution_supported = GST_WFD_CEA_640x480P60;
+  priv->audio_codec = GST_WFD_AUDIO_AAC;
+  priv->keep_alive_flag = FALSE;
+
+  g_mutex_init (&priv->keep_alive_lock);
+  g_mutex_init (&priv->stats_lock);
+
+  priv->host_address = NULL;
+
+  priv->stats_timer_id = -1;
+  priv->rtcp_stats_enabled = FALSE;
+  memset (&priv->stats, 0x00, sizeof (GstRTSPClientRTPStats));
+
+  GST_INFO_OBJECT (client, "Client is initialized");
+}
+
+/* A client is finalized when the connection is broken */
+static void
+gst_rtsp_wfd_client_finalize (GObject * obj)
+{
+  GstRTSPWFDClient *client = GST_RTSP_WFD_CLIENT (obj);
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+  g_return_if_fail (GST_IS_RTSP_WFD_CLIENT (obj));
+  g_return_if_fail (priv != NULL);
+
+  GST_INFO ("finalize client %p", client);
+
+  if (priv->host_address)
+    g_free (priv->host_address);
+
+  if (priv->stats_timer_id > 0)
+    g_source_remove(priv->stats_timer_id);
+
+  g_mutex_clear (&priv->keep_alive_lock);
+  g_mutex_clear (&priv->stats_lock);
+  G_OBJECT_CLASS (gst_rtsp_wfd_client_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_wfd_client_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec)
+{
+  //GstRTSPWFDClient *client = GST_RTSP_WFD_CLIENT (object);
+
+  switch (propid) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+static void
+gst_rtsp_wfd_client_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec)
+{
+  //GstRTSPWFDClient *client = GST_RTSP_WFD_CLIENT (object);
+
+  switch (propid) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+/**
+ * gst_rtsp_wfd_client_new:
+ *
+ * Create a new #GstRTSPWFDClient instance.
+ *
+ * Returns: a new #GstRTSPWFDClient
+ */
+GstRTSPWFDClient *
+gst_rtsp_wfd_client_new (void)
+{
+  GstRTSPWFDClient *result;
+
+  result = g_object_new (GST_TYPE_RTSP_WFD_CLIENT, NULL);
+
+  return result;
+}
+
+void
+gst_rtsp_wfd_client_start_wfd (GstRTSPWFDClient * client)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GST_INFO_OBJECT (client, "gst_rtsp_wfd_client_start_wfd");
+
+  res = handle_M1_message (client);
+  if (res < GST_RTSP_OK) {
+    GST_ERROR_OBJECT (client, "handle_M1_message failed : %d", res);
+  }
+
+  return;
+}
+
+static gboolean
+wfd_display_rtp_stats (gpointer userdata)
+{
+  guint16 seqnum = 0;
+  guint64 bytes = 0;
+
+  GstRTSPWFDClient *client = NULL;
+  GstRTSPWFDClientPrivate *priv = NULL;
+
+  client = (GstRTSPWFDClient *) userdata;
+  priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+  if (!priv) {
+    GST_ERROR("No priv");
+    return FALSE;
+  }
+
+  g_mutex_lock(&priv->stats_lock);
+
+  seqnum = gst_rtsp_stream_get_current_seqnum (priv->stats.stream);
+  bytes = gst_rtsp_stream_get_udp_sent_bytes (priv->stats.stream);
+
+  GST_INFO ("----------------------------------------------------\n");
+  GST_INFO ("Sent RTP packets : %d", seqnum - priv->stats.last_seqnum);
+  GST_INFO ("Sent Bytes of RTP packets : %lld bytes", bytes - priv->stats.last_sent_bytes);
+
+  priv->stats.last_seqnum = seqnum;
+  priv->stats.last_sent_bytes = bytes;
+
+  if (priv->rtcp_stats_enabled) {
+    GST_INFO ("Fraction Lost: %d", priv->stats.fraction_lost);
+    GST_INFO ("Cumulative number of packets lost: %d", priv->stats.cumulative_lost_num);
+    GST_INFO ("Extended highest sequence number received: %d", priv->stats.max_seqnum);
+    GST_INFO ("Interarrival Jitter: %d", priv->stats.arrival_jitter);
+    GST_INFO ("Round trip time : %d", priv->stats.rtt);
+  }
+
+  GST_INFO ("----------------------------------------------------\n");
+
+  g_mutex_unlock(&priv->stats_lock);
+
+  return TRUE;
+}
+
+static void
+on_rtcp_stats (GstRTSPStream *stream, GstStructure *stats, GstRTSPClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+  guint fraction_lost, exthighestseq, jitter, lsr, dlsr, rtt;
+  gint packetslost;
+
+  if (!priv) return;
+
+  g_mutex_lock(&priv->stats_lock);
+
+  gst_structure_get_uint (stats, "rb-fractionlost", &fraction_lost);
+  gst_structure_get_int (stats, "rb-packetslost", &packetslost);
+  gst_structure_get_uint (stats, "rb-exthighestseq", &exthighestseq);
+  gst_structure_get_uint (stats, "rb-jitter", &jitter);
+  gst_structure_get_uint (stats, "rb-lsr", &lsr);
+  gst_structure_get_uint (stats, "rb-dlsr", &dlsr);
+  gst_structure_get_uint (stats, "rb-round-trip", &rtt);
+
+  if (!priv->rtcp_stats_enabled)
+    priv->rtcp_stats_enabled = TRUE;
+
+  priv->stats.stream = stream;
+  priv->stats.fraction_lost = (guint8)fraction_lost;
+  priv->stats.cumulative_lost_num += (guint32)fraction_lost;
+  priv->stats.max_seqnum = (guint16)exthighestseq;
+  priv->stats.arrival_jitter = (guint32)jitter;
+  priv->stats.lsr = (guint32)lsr;
+  priv->stats.dlsr = (guint32)dlsr;
+  priv->stats.rtt = (guint32)rtt;
+
+  g_mutex_unlock(&priv->stats_lock);
+}
+
+static gboolean
+wfd_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
+    GstRTSPStream * stream, GstRTSPContext * ctx)
+{
+  if (stream) {
+    GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+    if (priv)
+      priv->stats.stream = stream;
+    g_signal_connect (stream, "rtcp-statistics", (GCallback) on_rtcp_stats, client);
+  }
+
+  return GST_RTSP_CLIENT_CLASS (gst_rtsp_wfd_client_parent_class)->configure_client_media (client, media, stream, ctx);
+}
+static void
+wfd_options_request_done (GstRTSPWFDClient * client, GstRTSPContext *ctx)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPWFDClientClass *klass = GST_RTSP_WFD_CLIENT_GET_CLASS (client);
+
+  g_return_if_fail (klass != NULL);
+
+  GST_INFO_OBJECT (client, "M2 done..");
+
+  res = handle_M3_message (client);
+  if (res < GST_RTSP_OK) {
+    GST_ERROR_OBJECT (client, "handle_M3_message failed : %d", res);
+  }
+
+  if (klass->prepare_resource) {
+    klass->prepare_resource (client, ctx);
+  }
+
+  return;
+}
+
+static void
+wfd_get_param_request_done (GstRTSPWFDClient * client, GstRTSPContext *ctx)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  GstRTSPWFDClientClass *klass = GST_RTSP_WFD_CLIENT_GET_CLASS (client);
+
+  g_return_if_fail (priv != NULL && klass != NULL);
+
+  priv->m3_done = TRUE;
+  GST_INFO_OBJECT (client, "M3 done..");
+
+  res = handle_M4_message (client);
+  if (res < GST_RTSP_OK) {
+    GST_ERROR_OBJECT (client, "handle_M4_message failed : %d", res);
+  }
+
+  if (klass->confirm_resource) {
+    klass->confirm_resource (client, ctx);
+  }
+
+  return;
+}
+
+static guint
+wfd_get_prefered_audio_codec (guint8 srcAudioCodec,
+    guint sinkAudioCodec)
+{
+  int i = 0;
+  guint codec = 0;
+  for (i = 0; i < 8; i++) {
+    if (((sinkAudioCodec << i) & 0x80)
+        && ((srcAudioCodec << i) & 0x80)) {
+      codec = (0x01 << (7 - i));
+      break;
+    }
+  }
+  return codec;
+}
+
+static guint64
+wfd_get_prefered_resolution (guint64 srcResolution,
+    guint64 sinkResolution,
+    GstWFDVideoNativeResolution native,
+    guint32 * cMaxWidth,
+    guint32 * cMaxHeight, guint32 * cFramerate, guint32 * interleaved)
+{
+  int i = 0;
+  guint64 resolution = 0;
+  for (i = 0; i < 32; i++) {
+    if (((sinkResolution << i) & 0x80000000)
+        && ((srcResolution << i) & 0x80000000)) {
+      resolution = ((guint64) 0x00000001 << (31 - i));
+      break;
+    }
+  }
+  switch (native) {
+    case GST_WFD_VIDEO_CEA_RESOLUTION:
+    {
+      switch (resolution) {
+        case GST_WFD_CEA_640x480P60:
+          *cMaxWidth = 640;
+          *cMaxHeight = 480;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_CEA_720x480P60:
+          *cMaxWidth = 720;
+          *cMaxHeight = 480;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_CEA_720x480I60:
+          *cMaxWidth = 720;
+          *cMaxHeight = 480;
+          *cFramerate = 60;
+          *interleaved = 1;
+          break;
+        case GST_WFD_CEA_720x576P50:
+          *cMaxWidth = 720;
+          *cMaxHeight = 576;
+          *cFramerate = 50;
+          *interleaved = 0;
+          break;
+        case GST_WFD_CEA_720x576I50:
+          *cMaxWidth = 720;
+          *cMaxHeight = 576;
+          *cFramerate = 50;
+          *interleaved = 1;
+          break;
+        case GST_WFD_CEA_1280x720P30:
+          *cMaxWidth = 1280;
+          *cMaxHeight = 720;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_CEA_1280x720P60:
+          *cMaxWidth = 1280;
+          *cMaxHeight = 720;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_CEA_1920x1080P30:
+          *cMaxWidth = 1920;
+          *cMaxHeight = 1080;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_CEA_1920x1080P60:
+          *cMaxWidth = 1920;
+          *cMaxHeight = 1080;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_CEA_1920x1080I60:
+          *cMaxWidth = 1920;
+          *cMaxHeight = 1080;
+          *cFramerate = 60;
+          *interleaved = 1;
+          break;
+        case GST_WFD_CEA_1280x720P25:
+          *cMaxWidth = 1280;
+          *cMaxHeight = 720;
+          *cFramerate = 25;
+          *interleaved = 0;
+          break;
+        case GST_WFD_CEA_1280x720P50:
+          *cMaxWidth = 1280;
+          *cMaxHeight = 720;
+          *cFramerate = 50;
+          *interleaved = 0;
+          break;
+        case GST_WFD_CEA_1920x1080P25:
+          *cMaxWidth = 1920;
+          *cMaxHeight = 1080;
+          *cFramerate = 25;
+          *interleaved = 0;
+          break;
+        case GST_WFD_CEA_1920x1080P50:
+          *cMaxWidth = 1920;
+          *cMaxHeight = 1080;
+          *cFramerate = 50;
+          *interleaved = 0;
+          break;
+        case GST_WFD_CEA_1920x1080I50:
+          *cMaxWidth = 1920;
+          *cMaxHeight = 1080;
+          *cFramerate = 50;
+          *interleaved = 1;
+          break;
+        case GST_WFD_CEA_1280x720P24:
+          *cMaxWidth = 1280;
+          *cMaxHeight = 720;
+          *cFramerate = 24;
+          *interleaved = 0;
+          break;
+        case GST_WFD_CEA_1920x1080P24:
+          *cMaxWidth = 1920;
+          *cMaxHeight = 1080;
+          *cFramerate = 24;
+          *interleaved = 0;
+          break;
+        default:
+          *cMaxWidth = 0;
+          *cMaxHeight = 0;
+          *cFramerate = 0;
+          *interleaved = 0;
+          break;
+      }
+    }
+      break;
+    case GST_WFD_VIDEO_VESA_RESOLUTION:
+    {
+      switch (resolution) {
+        case GST_WFD_VESA_800x600P30:
+          *cMaxWidth = 800;
+          *cMaxHeight = 600;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_800x600P60:
+          *cMaxWidth = 800;
+          *cMaxHeight = 600;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1024x768P30:
+          *cMaxWidth = 1024;
+          *cMaxHeight = 768;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1024x768P60:
+          *cMaxWidth = 1024;
+          *cMaxHeight = 768;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1152x864P30:
+          *cMaxWidth = 1152;
+          *cMaxHeight = 864;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1152x864P60:
+          *cMaxWidth = 1152;
+          *cMaxHeight = 864;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1280x768P30:
+          *cMaxWidth = 1280;
+          *cMaxHeight = 768;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1280x768P60:
+          *cMaxWidth = 1280;
+          *cMaxHeight = 768;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1280x800P30:
+          *cMaxWidth = 1280;
+          *cMaxHeight = 800;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1280x800P60:
+          *cMaxWidth = 1280;
+          *cMaxHeight = 800;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1360x768P30:
+          *cMaxWidth = 1360;
+          *cMaxHeight = 768;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1360x768P60:
+          *cMaxWidth = 1360;
+          *cMaxHeight = 768;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1366x768P30:
+          *cMaxWidth = 1366;
+          *cMaxHeight = 768;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1366x768P60:
+          *cMaxWidth = 1366;
+          *cMaxHeight = 768;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1280x1024P30:
+          *cMaxWidth = 1280;
+          *cMaxHeight = 1024;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1280x1024P60:
+          *cMaxWidth = 1280;
+          *cMaxHeight = 1024;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1400x1050P30:
+          *cMaxWidth = 1400;
+          *cMaxHeight = 1050;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1400x1050P60:
+          *cMaxWidth = 1400;
+          *cMaxHeight = 1050;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1440x900P30:
+          *cMaxWidth = 1440;
+          *cMaxHeight = 900;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1440x900P60:
+          *cMaxWidth = 1440;
+          *cMaxHeight = 900;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1600x900P30:
+          *cMaxWidth = 1600;
+          *cMaxHeight = 900;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1600x900P60:
+          *cMaxWidth = 1600;
+          *cMaxHeight = 900;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1600x1200P30:
+          *cMaxWidth = 1600;
+          *cMaxHeight = 1200;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1600x1200P60:
+          *cMaxWidth = 1600;
+          *cMaxHeight = 1200;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1680x1024P30:
+          *cMaxWidth = 1680;
+          *cMaxHeight = 1024;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1680x1024P60:
+          *cMaxWidth = 1680;
+          *cMaxHeight = 1024;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1680x1050P30:
+          *cMaxWidth = 1680;
+          *cMaxHeight = 1050;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1680x1050P60:
+          *cMaxWidth = 1680;
+          *cMaxHeight = 1050;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1920x1200P30:
+          *cMaxWidth = 1920;
+          *cMaxHeight = 1200;
+          *cFramerate = 30;
+          *interleaved = 0;
+          break;
+        case GST_WFD_VESA_1920x1200P60:
+          *cMaxWidth = 1920;
+          *cMaxHeight = 1200;
+          *cFramerate = 60;
+          *interleaved = 0;
+          break;
+        default:
+          *cMaxWidth = 0;
+          *cMaxHeight = 0;
+          *cFramerate = 0;
+          *interleaved = 0;
+          break;
+      }
+    }
+      break;
+    case GST_WFD_VIDEO_HH_RESOLUTION:
+    {
+      *interleaved = 0;
+      switch (resolution) {
+        case GST_WFD_HH_800x480P30:
+          *cMaxWidth = 800;
+          *cMaxHeight = 480;
+          *cFramerate = 30;
+          break;
+        case GST_WFD_HH_800x480P60:
+          *cMaxWidth = 800;
+          *cMaxHeight = 480;
+          *cFramerate = 60;
+          break;
+        case GST_WFD_HH_854x480P30:
+          *cMaxWidth = 854;
+          *cMaxHeight = 480;
+          *cFramerate = 30;
+          break;
+        case GST_WFD_HH_854x480P60:
+          *cMaxWidth = 854;
+          *cMaxHeight = 480;
+          *cFramerate = 60;
+          break;
+        case GST_WFD_HH_864x480P30:
+          *cMaxWidth = 864;
+          *cMaxHeight = 480;
+          *cFramerate = 30;
+          break;
+        case GST_WFD_HH_864x480P60:
+          *cMaxWidth = 864;
+          *cMaxHeight = 480;
+          *cFramerate = 60;
+          break;
+        case GST_WFD_HH_640x360P30:
+          *cMaxWidth = 640;
+          *cMaxHeight = 360;
+          *cFramerate = 30;
+          break;
+        case GST_WFD_HH_640x360P60:
+          *cMaxWidth = 640;
+          *cMaxHeight = 360;
+          *cFramerate = 60;
+          break;
+        case GST_WFD_HH_960x540P30:
+          *cMaxWidth = 960;
+          *cMaxHeight = 540;
+          *cFramerate = 30;
+          break;
+        case GST_WFD_HH_960x540P60:
+          *cMaxWidth = 960;
+          *cMaxHeight = 540;
+          *cFramerate = 60;
+          break;
+        case GST_WFD_HH_848x480P30:
+          *cMaxWidth = 848;
+          *cMaxHeight = 480;
+          *cFramerate = 30;
+          break;
+        case GST_WFD_HH_848x480P60:
+          *cMaxWidth = 848;
+          *cMaxHeight = 480;
+          *cFramerate = 60;
+          break;
+        default:
+          *cMaxWidth = 0;
+          *cMaxHeight = 0;
+          *cFramerate = 0;
+          *interleaved = 0;
+          break;
+      }
+    }
+    break;
+
+    default:
+      *cMaxWidth = 0;
+      *cMaxHeight = 0;
+      *cFramerate = 0;
+      *interleaved = 0;
+      break;
+  }
+  return resolution;
+}
+
+static gchar *
+wfd_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
+{
+  gchar *path;
+
+  GST_DEBUG_OBJECT (client, "Got URI host : %s", uri->host);
+  GST_DEBUG_OBJECT (client, "Got URI abspath : %s", uri->abspath);
+
+  path = g_strdup ("/wfd1.0/streamid=0");
+
+  return path;
+}
+
+static void
+handle_wfd_play (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPWFDClient *_client = GST_RTSP_WFD_CLIENT (client);
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+  g_return_if_fail (priv != NULL);
+
+  wfd_set_keep_alive_condition(_client);
+  
+  priv->stats_timer_id = g_timeout_add (2000, wfd_display_rtp_stats, _client);
+}
+
+static void
+handle_wfd_response (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  guint8 *data = NULL;
+  guint size = 0;
+
+  GstRTSPWFDClient *_client = GST_RTSP_WFD_CLIENT (client);
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+  g_return_if_fail (priv != NULL);
+
+  GST_INFO_OBJECT (_client, "Handling response..");
+
+  if (!ctx) {
+    GST_ERROR_OBJECT (_client, "Context is NULL");
+    goto error;
+  }
+
+  if (!ctx->response) {
+    GST_ERROR_OBJECT (_client, "Response is NULL");
+    goto error;
+  }
+
+  /* parsing the GET_PARAMTER response */
+  res = gst_rtsp_message_get_body (ctx->response, (guint8 **) & data, &size);
+  if (res != GST_RTSP_OK) {
+    GST_ERROR_OBJECT (_client, "Failed to get body of response...");
+    return;
+  }
+
+  GST_INFO_OBJECT (_client, "Response body is %d", size);
+  if (size > 0) {
+    if (!priv->m3_done) {
+      GstWFDResult wfd_res;
+      GstWFDMessage *msg = NULL;
+      /* Parse M3 response from sink */
+      wfd_res = gst_wfd_message_new (&msg);
+      if (wfd_res != GST_WFD_OK) {
+        GST_ERROR_OBJECT (client, "Failed to create wfd message...");
+        goto error;
+      }
+
+      wfd_res = gst_wfd_message_init (msg);
+      if (wfd_res != GST_WFD_OK) {
+        GST_ERROR_OBJECT (client, "Failed to init wfd message...");
+        goto error;
+      }
+
+      wfd_res = gst_wfd_message_parse_buffer (data, size, msg);
+
+      GST_DEBUG_OBJECT (client, "M3 response server side message body: %s",
+          gst_wfd_message_as_text (msg));
+
+      /* Get the audio formats supported by WFDSink */
+      if (msg->audio_codecs) {
+        wfd_res =
+            gst_wfd_message_get_supported_audio_format (msg, &priv->caCodec,
+            &priv->cFreq, &priv->cChanels, &priv->cBitwidth, &priv->caLatency);
+        if (wfd_res != GST_WFD_OK) {
+          GST_WARNING_OBJECT (client,
+              "Failed to get wfd support audio formats...");
+          goto error;
+        }
+      }
+
+      /* Get the Video formats supported by WFDSink */
+      wfd_res =
+          gst_wfd_message_get_supported_video_format (msg, &priv->cvCodec,
+          &priv->cNative, &priv->cNativeResolution,
+          (guint64 *) & priv->cCEAResolution,
+          (guint64 *) & priv->cVESAResolution,
+          (guint64 *) & priv->cHHResolution, &priv->cProfile, &priv->cLevel,
+          &priv->cvLatency, &priv->cMaxHeight, &priv->cMaxWidth,
+          &priv->cmin_slice_size, &priv->cslice_enc_params,
+          &priv->cframe_rate_control);
+      if (wfd_res != GST_WFD_OK) {
+        GST_WARNING_OBJECT (client,
+            "Failed to get wfd supported video formats...");
+        goto error;
+      }
+
+      if (msg->client_rtp_ports) {
+        /* Get the RTP ports preferred by WFDSink */
+        wfd_res =
+            gst_wfd_message_get_prefered_rtp_ports (msg, &priv->ctrans,
+            &priv->cprofile, &priv->clowertrans, &priv->crtp_port0,
+            &priv->crtp_port1);
+        if (wfd_res != GST_WFD_OK) {
+          GST_WARNING_OBJECT (client,
+              "Failed to get wfd prefered RTP ports...");
+          goto error;
+        }
+      }
+
+      if (msg->display_edid) {
+        guint32 edid_block_count = 0;
+        gchar *edid_payload = NULL;
+        priv->edid_supported = FALSE;
+        /* Get the display edid preferred by WFDSink */
+        GST_DEBUG_OBJECT (client, "Going to gst_wfd_message_get_display_edid");
+        wfd_res =
+            gst_wfd_message_get_display_edid (msg, &priv->edid_supported,
+            &edid_block_count, &edid_payload);
+        if (wfd_res != GST_WFD_OK) {
+          GST_ERROR_OBJECT (client, "Failed to get wfd display edid...");
+          goto error;
+        }
+        GST_DEBUG_OBJECT (client, " edid supported: %d edid_block_count: %d",
+            priv->edid_supported, edid_block_count);
+        if (priv->edid_supported) {
+          priv->edid_hres = 0;
+          priv->edid_vres = 0;
+          priv->edid_hres =
+              (guint32) (((edid_payload[54 + 4] >> 4) << 8) | edid_payload[54 +
+                  2]);
+          priv->edid_vres =
+              (guint32) (((edid_payload[54 + 7] >> 4) << 8) | edid_payload[54 +
+                  5]);
+          GST_DEBUG_OBJECT (client, " edid supported Hres: %d Wres: %d",
+              priv->edid_hres, priv->edid_vres);
+          if ((priv->edid_hres < 640) || (priv->edid_vres < 480)
+              || (priv->edid_hres > 1920) || (priv->edid_vres > 1080)) {
+            priv->edid_hres = 0;
+            priv->edid_vres = 0;
+            priv->edid_supported = FALSE;
+            GST_WARNING_OBJECT (client, " edid invalid resolutions");
+          }
+        }
+      }
+
+      if (msg->content_protection) {
+#if 0
+        /*Get the hdcp version and tcp port by WFDSink */
+        wfd_res =
+            gst_wfd_message_get_contentprotection_type (msg,
+            &priv->hdcp_version, &priv->hdcp_tcpport);
+        GST_DEBUG ("hdcp version =%d, tcp port = %d", priv->hdcp_version,
+            priv->hdcp_tcpport);
+        if (priv->hdcp_version > 0 && priv->hdcp_tcpport > 0)
+          priv->protection_enabled = TRUE;
+
+        if (wfd_res != GST_WFD_OK) {
+          GST_WARNING_OBJECT (client,
+              "Failed to get wfd content protection...");
+          goto error;
+        }
+#else
+        GST_WARNING_OBJECT (client, "Don't use content protection");
+#endif
+      }
+
+      g_signal_emit (_client,
+          gst_rtsp_client_wfd_signals[SIGNAL_WFD_GET_PARAMETER_REQUEST], 0,
+          ctx);
+    } else {
+      /* TODO-WFD: Handle another GET_PARAMETER response with body */
+    }
+  } else if (size == 0) {
+    if (!priv->m1_done) {
+      GST_INFO_OBJECT (_client, "M1 response is done");
+      priv->m1_done = TRUE;
+    } else if (!priv->m4_done) {
+      GST_INFO_OBJECT (_client, "M4 response is done");
+      priv->m4_done = TRUE;
+
+      gst_rtsp_wfd_client_trigger_request (_client, WFD_TRIGGER_SETUP);
+    } else {
+      g_mutex_lock(&priv->keep_alive_lock);
+      if (priv->keep_alive_flag == FALSE) {
+        GST_INFO_OBJECT (_client, "M16 response is done");
+        priv->keep_alive_flag = TRUE;
+      }
+      g_mutex_unlock(&priv->keep_alive_lock);
+    }
+  }
+
+  return;
+
+error:
+  return;
+}
+
+static gboolean
+handle_wfd_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPMethod options;
+  gchar *tmp = NULL;
+  gchar *str = NULL;
+  gchar *user_agent = NULL;
+
+  GstRTSPWFDClient *_client = GST_RTSP_WFD_CLIENT (client);
+
+  options = GST_RTSP_OPTIONS |
+      GST_RTSP_PAUSE |
+      GST_RTSP_PLAY |
+      GST_RTSP_SETUP |
+      GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
+
+  str = gst_rtsp_options_as_text (options);
+
+  /*append WFD specific method */
+  tmp = g_strdup (", org.wfa.wfd1.0");
+  g_strlcat (str, tmp, strlen (tmp) + strlen (str) + 1);
+
+  gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+      gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
+
+  gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
+  g_free (str);
+  str = NULL;
+
+  res =
+      gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_USER_AGENT,
+      &user_agent, 0);
+  if (res == GST_RTSP_OK) {
+    gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_USER_AGENT,
+        user_agent);
+  } else {
+    return FALSE;
+  }
+
+  res = gst_rtsp_client_send_message (client, NULL, ctx->response);
+  if (res != GST_RTSP_OK) {
+    GST_ERROR_OBJECT (client, "gst_rtsp_client_send_message failed : %d", res);
+    return FALSE;
+  }
+
+  GST_DEBUG_OBJECT (client, "Sent M2 response...");
+
+  g_signal_emit (_client,
+      gst_rtsp_client_wfd_signals[SIGNAL_WFD_OPTIONS_REQUEST], 0, ctx);
+
+  return TRUE;
+}
+
+static gboolean
+handle_wfd_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  guint8 *data = NULL;
+  guint size = 0;
+
+  GstRTSPWFDClient *_client = GST_RTSP_WFD_CLIENT (client);
+
+  /* parsing the GET_PARAMTER request */
+  res = gst_rtsp_message_get_body (ctx->request, (guint8 **) & data, &size);
+  if (res != GST_RTSP_OK) {
+    GST_ERROR_OBJECT (_client, "Failed to get body of request...");
+    return FALSE;
+  }
+
+  if (size == 0) {
+    send_generic_wfd_response (_client, GST_RTSP_STS_OK, ctx);
+  } else {
+    /* TODO-WFD: Handle other GET_PARAMETER request from sink */
+  }
+
+  return TRUE;
+}
+
+static gboolean
+handle_wfd_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  guint8 *data = NULL;
+  guint size = 0;
+
+  GstRTSPWFDClient *_client = GST_RTSP_WFD_CLIENT (client);
+
+  res = gst_rtsp_message_get_body (ctx->request, &data, &size);
+  if (res != GST_RTSP_OK)
+    goto bad_request;
+
+  if (size == 0) {
+    /* no body, keep-alive request */
+    send_generic_wfd_response (_client, GST_RTSP_STS_OK, ctx);
+  } else {
+    if (data != NULL) {
+      GST_INFO_OBJECT (_client, "SET_PARAMETER Request : %s(%d)", data, size);
+      if (g_strcmp0 ((const gchar *) data, "wfd_idr_request"))
+        send_generic_wfd_response (_client, GST_RTSP_STS_OK, ctx);
+#if 0
+      else
+        /* TODO-WFD : Handle other set param request */
+        send_generic_wfd_response (_client, GST_RTSP_STS_OK, ctx);
+#endif
+    } else {
+      goto bad_request;
+    }
+  }
+
+  return TRUE;
+
+  /* ERRORS */
+bad_request:
+  {
+    GST_ERROR ("_client %p: bad request", _client);
+    send_generic_wfd_response (_client, GST_RTSP_STS_BAD_REQUEST, ctx);
+    return FALSE;
+  }
+}
+
+#if 0
+static gboolean
+gst_rtsp_wfd_client_parse_methods (GstRTSPWFDClient * client,
+    GstRTSPMessage * response)
+{
+  GstRTSPHeaderField field;
+  gchar *respoptions;
+  gchar **options;
+  gint indx = 0;
+  gint i;
+  gboolean found_wfd_method = FALSE;
+
+  /* reset supported methods */
+  client->supported_methods = 0;
+
+  /* Try Allow Header first */
+  field = GST_RTSP_HDR_ALLOW;
+  while (TRUE) {
+    respoptions = NULL;
+    gst_rtsp_message_get_header (response, field, &respoptions, indx);
+    if (indx == 0 && !respoptions) {
+      /* if no Allow header was found then try the Public header... */
+      field = GST_RTSP_HDR_PUBLIC;
+      gst_rtsp_message_get_header (response, field, &respoptions, indx);
+    }
+    if (!respoptions)
+      break;
+
+    /* If we get here, the server gave a list of supported methods, parse
+     * them here. The string is like:
+     *
+     * OPTIONS,  PLAY, SETUP, ...
+     */
+    options = g_strsplit (respoptions, ",", 0);
+
+    for (i = 0; options[i]; i++) {
+      gchar *stripped;
+      gint method;
+
+      stripped = g_strstrip (options[i]);
+      method = gst_rtsp_find_method (stripped);
+
+      if (!g_ascii_strcasecmp ("org.wfa.wfd1.0", stripped))
+        found_wfd_method = TRUE;
+
+      /* keep bitfield of supported methods */
+      if (method != GST_RTSP_INVALID)
+        client->supported_methods |= method;
+    }
+    g_strfreev (options);
+
+    indx++;
+  }
+
+  if (!found_wfd_method) {
+    GST_ERROR_OBJECT (client,
+        "WFD client is not supporting WFD mandatory message : org.wfa.wfd1.0...");
+    goto no_required_methods;
+  }
+
+  /* Checking mandatory method */
+  if (!(client->supported_methods & GST_RTSP_SET_PARAMETER)) {
+    GST_ERROR_OBJECT (client,
+        "WFD client is not supporting WFD mandatory message : SET_PARAMETER...");
+    goto no_required_methods;
+  }
+
+  /* Checking mandatory method */
+  if (!(client->supported_methods & GST_RTSP_GET_PARAMETER)) {
+    GST_ERROR_OBJECT (client,
+        "WFD client is not supporting WFD mandatory message : GET_PARAMETER...");
+    goto no_required_methods;
+  }
+
+  if (!(client->supported_methods & GST_RTSP_OPTIONS)) {
+    GST_INFO_OBJECT (client, "assuming OPTIONS is supported by client...");
+    client->supported_methods |= GST_RTSP_OPTIONS;
+  }
+
+  return TRUE;
+
+/* ERRORS */
+no_required_methods:
+  {
+    GST_ELEMENT_ERROR (client, RESOURCE, OPEN_READ, (NULL),
+        ("WFD Client does not support mandatory methods."));
+    return FALSE;
+  }
+}
+#endif
+
+typedef enum
+{
+  M1_REQ_MSG,
+  M1_RES_MSG,
+  M2_REQ_MSG,
+  M2_RES_MSG,
+  M3_REQ_MSG,
+  M3_RES_MSG,
+  M4_REQ_MSG,
+  M4_RES_MSG,
+  M5_REQ_MSG,
+  TEARDOWN_TRIGGER,
+  PLAY_TRIGGER,
+  PAUSE_TRIGGER,
+} GstWFDMessageType;
+
+static gboolean
+_set_negotiated_audio_codec (GstRTSPWFDClient *client,
+    guint audio_codec)
+{
+  GstRTSPClient *parent_client = GST_RTSP_CLIENT_CAST (client);
+
+  GstRTSPMediaFactory *factory = NULL;
+  GstRTSPMountPoints *mount_points = NULL;
+  gchar *path = NULL;
+  gint matched = 0;
+  gboolean ret = TRUE;
+
+  if (!(mount_points = gst_rtsp_client_get_mount_points (parent_client))) {
+    ret = FALSE;
+    GST_ERROR_OBJECT (client, "Failed to set negotiated audio codec: no mount points...");
+    goto no_mount_points;
+  }
+
+  path = g_strdup(WFD_MOUNT_POINT);
+  if (!path) {
+    ret = FALSE;
+    GST_ERROR_OBJECT (client, "Failed to set negotiated audio codec: no path...");
+    goto no_path;
+  }
+
+  if (!(factory = gst_rtsp_mount_points_match (mount_points,
+          path, &matched))) {
+    GST_ERROR_OBJECT (client, "Failed to set negotiated audio codec: no factory...");
+    ret = FALSE;
+    goto no_factory;
+  }
+
+  gst_rtsp_media_factory_wfd_set_audio_codec (factory,
+      audio_codec);
+  ret = TRUE;
+
+  g_object_unref(factory);
+
+no_factory:
+  g_free(path);
+no_path:
+  g_object_unref(mount_points);
+no_mount_points:
+  return ret;
+}
+
+static gboolean
+_set_negotiated_resolution(GstRTSPWFDClient *client,
+    guint32 width, guint32 height)
+{
+  GstRTSPClient *parent_client = GST_RTSP_CLIENT_CAST (client);
+
+  GstRTSPMediaFactory *factory = NULL;
+  GstRTSPMountPoints *mount_points = NULL;
+  gchar *path = NULL;
+  gint matched = 0;
+  gboolean ret = TRUE;
+
+  if (!(mount_points = gst_rtsp_client_get_mount_points (parent_client))) {
+    ret = FALSE;
+    GST_ERROR_OBJECT (client, "Failed to set negotiated resolution: no mount points...");
+    goto no_mount_points;
+  }
+
+  path = g_strdup(WFD_MOUNT_POINT);
+  if (!path) {
+    ret = FALSE;
+    GST_ERROR_OBJECT (client, "Failed to set negotiated resolution: no path...");
+    goto no_path;
+  }
+
+  if (!(factory = gst_rtsp_mount_points_match (mount_points,
+          path, &matched))) {
+    GST_ERROR_OBJECT (client, "Failed to set negotiated resolution: no factory...");
+    ret = FALSE;
+    goto no_factory;
+  }
+
+  gst_rtsp_media_factory_wfd_set_negotiated_resolution(factory,
+      width, height);
+  ret = TRUE;
+
+  g_object_unref(factory);
+
+no_factory:
+  g_free(path);
+no_path:
+  g_object_unref(mount_points);
+no_mount_points:
+  return ret;
+}
+
+static void
+_set_wfd_message_body (GstRTSPWFDClient * client, GstWFDMessageType msg_type,
+    gchar ** data, guint * len)
+{
+  GString *buf = NULL;
+  GstWFDMessage *msg = NULL;
+  GstWFDResult wfd_res = GST_WFD_EINVAL;
+  GstRTSPWFDClientPrivate *priv = NULL;
+  priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+  g_return_if_fail (priv != NULL);
+
+  buf = g_string_new ("");
+  g_return_if_fail (buf != NULL);
+
+  if (msg_type == M3_REQ_MSG) {
+    /* create M3 request to be sent */
+    wfd_res = gst_wfd_message_new (&msg);
+    if (wfd_res != GST_WFD_OK) {
+      GST_ERROR_OBJECT (client, "Failed to create wfd message...");
+      goto error;
+    }
+
+    wfd_res = gst_wfd_message_init (msg);
+    if (wfd_res != GST_WFD_OK) {
+      GST_ERROR_OBJECT (client, "Failed to init wfd message...");
+      goto error;
+    }
+
+    /* set the supported audio formats by the WFD server */
+    wfd_res =
+        gst_wfd_message_set_supported_audio_format (msg, GST_WFD_AUDIO_UNKNOWN,
+        GST_WFD_FREQ_UNKNOWN, GST_WFD_CHANNEL_UNKNOWN, 0, 0);
+    if (wfd_res != GST_WFD_OK) {
+      GST_ERROR_OBJECT (client,
+          "Failed to set supported audio formats on wfd message...");
+      goto error;
+    }
+
+    /* set the supported Video formats by the WFD server */
+    wfd_res =
+        gst_wfd_message_set_supported_video_format (msg, GST_WFD_VIDEO_UNKNOWN,
+        GST_WFD_VIDEO_CEA_RESOLUTION, GST_WFD_CEA_UNKNOWN, GST_WFD_CEA_UNKNOWN,
+        GST_WFD_VESA_UNKNOWN, GST_WFD_HH_UNKNOWN, GST_WFD_H264_UNKNOWN_PROFILE,
+        GST_WFD_H264_LEVEL_UNKNOWN, 0, 0, 0, 0, 0, 0);
+    if (wfd_res != GST_WFD_OK) {
+      GST_ERROR_OBJECT (client,
+          "Failed to set supported video formats on wfd message...");
+      goto error;
+    }
+
+    wfd_res = gst_wfd_message_set_display_edid (msg, 0, 0, NULL);
+    if (wfd_res != GST_WFD_OK) {
+      GST_ERROR_OBJECT (client,
+          "Failed to set display edid type on wfd message...");
+      goto error;
+    }
+
+    if (priv->protection_enabled) {
+      wfd_res =
+          gst_wfd_message_set_contentprotection_type (msg, GST_WFD_HDCP_NONE,
+          0);
+      if (wfd_res != GST_WFD_OK) {
+        GST_ERROR_OBJECT (client,
+            "Failed to set supported content protection type on wfd message...");
+        goto error;
+      }
+    }
+
+    /* set the preffered RTP ports for the WFD server */
+    wfd_res =
+        gst_wfd_messge_set_prefered_rtp_ports (msg, GST_WFD_RTSP_TRANS_UNKNOWN,
+        GST_WFD_RTSP_PROFILE_UNKNOWN, GST_WFD_RTSP_LOWER_TRANS_UNKNOWN, 0, 0);
+    if (wfd_res != GST_WFD_OK) {
+      GST_ERROR_OBJECT (client,
+          "Failed to set supported video formats on wfd message...");
+      goto error;
+    }
+
+    *data = gst_wfd_message_param_names_as_text (msg);
+    if (*data == NULL) {
+      GST_ERROR_OBJECT (client, "Failed to get wfd message as text...");
+      goto error;
+    } else {
+      *len = strlen (*data);
+    }
+  } else if (msg_type == M4_REQ_MSG) {
+    GstRTSPUrl *url = NULL;
+
+    GstRTSPClient *parent_client = GST_RTSP_CLIENT_CAST (client);
+    GstRTSPConnection *connection =
+        gst_rtsp_client_get_connection (parent_client);
+
+    /* Parameters for the preffered audio formats */
+    GstWFDAudioFormats taudiocodec = GST_WFD_AUDIO_UNKNOWN;
+    GstWFDAudioFreq taudiofreq = GST_WFD_FREQ_UNKNOWN;
+    GstWFDAudioChannels taudiochannels = GST_WFD_CHANNEL_UNKNOWN;
+
+    /* Parameters for the preffered video formats */
+    GstWFDVideoCEAResolution tcCEAResolution = GST_WFD_CEA_UNKNOWN;
+    GstWFDVideoVESAResolution tcVESAResolution = GST_WFD_VESA_UNKNOWN;
+    GstWFDVideoHHResolution tcHHResolution = GST_WFD_HH_UNKNOWN;
+    GstWFDVideoH264Profile tcProfile;
+    GstWFDVideoH264Level tcLevel;
+    guint64 resolution_supported = 0;
+
+    url = gst_rtsp_connection_get_url (connection);
+    if (url == NULL) {
+      GST_ERROR_OBJECT (client, "Failed to get connection URL");
+      return;
+    }
+
+    /* Logic to negotiate with information of M3 response */
+    /* create M4 request to be sent */
+    wfd_res = gst_wfd_message_new (&msg);
+    if (wfd_res != GST_WFD_OK) {
+      GST_ERROR_OBJECT (client, "Failed to create wfd message...");
+      goto error;
+    }
+
+    wfd_res = gst_wfd_message_init (msg);
+    if (wfd_res != GST_WFD_OK) {
+      GST_ERROR_OBJECT (client, "Failed to init wfd message...");
+      goto error;
+    }
+
+    g_string_append_printf (buf, "rtsp://");
+
+    if (priv->host_address) {
+      g_string_append (buf, priv->host_address);
+    } else {
+      GST_ERROR_OBJECT (client, "Failed to get host address");
+      if (buf) g_string_free (buf, TRUE);
+      goto error;
+    }
+
+    g_string_append_printf (buf, "/wfd1.0/streamid=0");
+    wfd_res =
+        gst_wfd_message_set_presentation_url (msg, g_string_free (buf, FALSE),
+        NULL);
+
+    if (wfd_res != GST_WFD_OK) {
+      GST_ERROR_OBJECT (client, "Failed to set presentation url");
+      goto error;
+    }
+
+    taudiocodec = wfd_get_prefered_audio_codec (priv->audio_codec, priv->caCodec);
+    priv->caCodec = taudiocodec;
+    if (!_set_negotiated_audio_codec(client, priv->caCodec)) {
+      GST_ERROR_OBJECT (client, "Failed to set negotiated "
+          "audio codec to media factory...");
+    }
+
+    if (priv->cFreq & GST_WFD_FREQ_48000)
+      taudiofreq = GST_WFD_FREQ_48000;
+    else if (priv->cFreq & GST_WFD_FREQ_44100)
+      taudiofreq = GST_WFD_FREQ_44100;
+    priv->cFreq = taudiofreq;
+
+    /* TODO-WFD: Currently only 2 channels is present */
+    if (priv->cChanels & GST_WFD_CHANNEL_8)
+      taudiochannels = GST_WFD_CHANNEL_2;
+    else if (priv->cChanels & GST_WFD_CHANNEL_6)
+      taudiochannels = GST_WFD_CHANNEL_2;
+    else if (priv->cChanels & GST_WFD_CHANNEL_4)
+      taudiochannels = GST_WFD_CHANNEL_2;
+    else if (priv->cChanels & GST_WFD_CHANNEL_2)
+      taudiochannels = GST_WFD_CHANNEL_2;
+    priv->cChanels = taudiochannels;
+
+    wfd_res =
+        gst_wfd_message_set_prefered_audio_format (msg, taudiocodec, taudiofreq,
+        taudiochannels, priv->cBitwidth, priv->caLatency);
+    if (wfd_res != GST_WFD_OK) {
+      GST_ERROR_OBJECT (priv, "Failed to set preffered audio formats...");
+      goto error;
+    }
+
+    /* Set the preffered video formats */
+    priv->cvCodec = GST_WFD_VIDEO_H264;
+    priv->cProfile = tcProfile = GST_WFD_H264_BASE_PROFILE;
+    priv->cLevel = tcLevel = GST_WFD_H264_LEVEL_3_1;
+
+    resolution_supported = priv->video_resolution_supported;
+
+    /* TODO-WFD: Need to verify this logic
+       if(priv->edid_supported) {
+       if (priv->edid_hres < 1920) resolution_supported = resolution_supported & 0x8C7F;
+       if (priv->edid_hres < 1280) resolution_supported = resolution_supported & 0x1F;
+       if (priv->edid_hres < 720) resolution_supported = resolution_supported & 0x01;
+       }
+     */
+
+    if (priv->video_native_resolution == GST_WFD_VIDEO_CEA_RESOLUTION) {
+      tcCEAResolution =
+          wfd_get_prefered_resolution (resolution_supported,
+          priv->cCEAResolution, priv->video_native_resolution, &priv->cMaxWidth,
+          &priv->cMaxHeight, &priv->cFramerate, &priv->cInterleaved);
+      GST_DEBUG
+          ("wfd negotiated resolution: %08x, width: %d, height: %d, framerate: %d, interleaved: %d",
+          tcCEAResolution, priv->cMaxWidth, priv->cMaxHeight, priv->cFramerate,
+          priv->cInterleaved);
+    } else if (priv->video_native_resolution == GST_WFD_VIDEO_VESA_RESOLUTION) {
+      tcVESAResolution =
+          wfd_get_prefered_resolution (resolution_supported,
+          priv->cVESAResolution, priv->video_native_resolution,
+          &priv->cMaxWidth, &priv->cMaxHeight, &priv->cFramerate,
+          &priv->cInterleaved);
+      GST_DEBUG
+          ("wfd negotiated resolution: %08x, width: %d, height: %d, framerate: %d, interleaved: %d",
+          tcVESAResolution, priv->cMaxWidth, priv->cMaxHeight, priv->cFramerate,
+          priv->cInterleaved);
+    } else if (priv->video_native_resolution == GST_WFD_VIDEO_HH_RESOLUTION) {
+      tcHHResolution =
+          wfd_get_prefered_resolution (resolution_supported,
+          priv->cHHResolution, priv->video_native_resolution, &priv->cMaxWidth,
+          &priv->cMaxHeight, &priv->cFramerate, &priv->cInterleaved);
+      GST_DEBUG
+          ("wfd negotiated resolution: %08x, width: %d, height: %d, framerate: %d, interleaved: %d",
+          tcHHResolution, priv->cMaxWidth, priv->cMaxHeight, priv->cFramerate,
+          priv->cInterleaved);
+    }
+
+    if (!_set_negotiated_resolution(client, priv->cMaxWidth,
+          priv->cMaxHeight)) {
+      GST_ERROR_OBJECT (client, "Failed to set negotiated "
+          "resolution to media factory...");
+    }
+
+    wfd_res =
+        gst_wfd_message_set_prefered_video_format (msg, priv->cvCodec,
+        priv->video_native_resolution, GST_WFD_CEA_UNKNOWN, tcCEAResolution,
+        tcVESAResolution, tcHHResolution, tcProfile, tcLevel, priv->cvLatency,
+        priv->cMaxWidth, priv->cMaxHeight, priv->cmin_slice_size,
+        priv->cslice_enc_params, priv->cframe_rate_control);
+
+    if (wfd_res != GST_WFD_OK) {
+      GST_ERROR_OBJECT (client, "Failed to set preffered video formats...");
+      goto error;
+    }
+
+    /* set the preffered RTP ports for the WFD server */
+    wfd_res =
+        gst_wfd_messge_set_prefered_rtp_ports (msg, GST_WFD_RTSP_TRANS_RTP,
+        GST_WFD_RTSP_PROFILE_AVP, GST_WFD_RTSP_LOWER_TRANS_UDP, priv->crtp_port0, priv->crtp_port1);
+    if (wfd_res != GST_WFD_OK) {
+      GST_ERROR_OBJECT (client,
+          "Failed to set supported video formats on wfd message...");
+      goto error;
+    }
+
+    *data = gst_wfd_message_as_text (msg);
+    if (*data == NULL) {
+      GST_ERROR_OBJECT (client, "Failed to get wfd message as text...");
+      goto error;
+    } else {
+      *len = strlen (*data);
+    }
+  } else if (msg_type == M5_REQ_MSG) {
+    g_string_append (buf, "wfd_trigger_method: SETUP");
+    g_string_append (buf, "\r\n");
+    *len = buf->len;
+    *data = g_string_free (buf, FALSE);
+  } else if (msg_type == TEARDOWN_TRIGGER) {
+    g_string_append (buf, "wfd_trigger_method: TEARDOWN");
+    g_string_append (buf, "\r\n");
+    *len = buf->len;
+    *data = g_string_free (buf, FALSE);
+  } else if (msg_type == PLAY_TRIGGER) {
+    g_string_append (buf, "wfd_trigger_method: PLAY");
+    g_string_append (buf, "\r\n");
+    *len = buf->len;
+    *data = g_string_free (buf, FALSE);
+  } else if (msg_type == PAUSE_TRIGGER) {
+    g_string_append (buf, "wfd_trigger_method: PAUSE");
+    g_string_append (buf, "\r\n");
+    *len = buf->len;
+    *data = g_string_free (buf, FALSE);
+  } else {
+    return;
+  }
+
+  return;
+
+error:
+  *data = NULL;
+  *len = 0;
+
+  return;
+}
+
+/**
+* prepare_request:
+* @client: client object
+* @request : requst message to be prepared
+* @method : RTSP method of the request
+* @url : url need to be in the request
+* @message_type : WFD message type
+* @trigger_type : trigger method to be used for M5 mainly
+*
+* Prepares request based on @method & @message_type
+*
+* Returns: a #GstRTSPResult.
+*/
+GstRTSPResult
+prepare_request (GstRTSPWFDClient * client, GstRTSPMessage * request,
+    GstRTSPMethod method, gchar * url)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  gchar *str = NULL;
+
+  if (method == GST_RTSP_GET_PARAMETER || method == GST_RTSP_SET_PARAMETER) {
+    g_free (url);
+    url = g_strdup ("rtsp://localhost/wfd1.0");
+  }
+
+  GST_DEBUG_OBJECT (client, "Preparing request: %d", method);
+
+  /* initialize the request */
+  res = gst_rtsp_message_init_request (request, method, url);
+  if (res < 0) {
+    GST_ERROR ("init request failed");
+    return res;
+  }
+
+  switch (method) {
+      /* Prepare OPTIONS request to send */
+    case GST_RTSP_OPTIONS:{
+      /* add wfd specific require filed "org.wfa.wfd1.0" */
+      str = g_strdup ("org.wfa.wfd1.0");
+      res = gst_rtsp_message_add_header (request, GST_RTSP_HDR_REQUIRE, str);
+      if (res < 0) {
+        GST_ERROR ("Failed to add header");
+        g_free (str);
+        return res;
+      }
+
+      g_free (str);
+      break;
+    }
+
+      /* Prepare GET_PARAMETER request */
+    case GST_RTSP_GET_PARAMETER:{
+      gchar *msg = NULL;
+      guint msglen = 0;
+      GString *msglength;
+
+      /* add content type */
+      res =
+          gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+          "text/parameters");
+      if (res < 0) {
+        GST_ERROR ("Failed to add header");
+        return res;
+      }
+
+      _set_wfd_message_body (client, M3_REQ_MSG, &msg, &msglen);
+      msglength = g_string_new ("");
+      g_string_append_printf (msglength, "%d", msglen);
+      GST_DEBUG ("M3 server side message body: %s", msg);
+
+      /* add content-length type */
+      res =
+          gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_LENGTH,
+          g_string_free (msglength, FALSE));
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+        goto error;
+      }
+
+      res = gst_rtsp_message_set_body (request, (guint8 *) msg, msglen);
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+        goto error;
+      }
+
+      g_free (msg);
+      break;
+    }
+
+      /* Prepare SET_PARAMETER request */
+    case GST_RTSP_SET_PARAMETER:{
+      gchar *msg = NULL;
+      guint msglen = 0;
+      GString *msglength;
+
+      /* add content type */
+      res =
+          gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+          "text/parameters");
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp request...");
+        goto error;
+      }
+
+      _set_wfd_message_body (client, M4_REQ_MSG, &msg, &msglen);
+      msglength = g_string_new ("");
+      g_string_append_printf (msglength, "%d", msglen);
+      GST_DEBUG ("M4 server side message body: %s", msg);
+
+      /* add content-length type */
+      res =
+          gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_LENGTH,
+          g_string_free (msglength, FALSE));
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+        goto error;
+      }
+
+      res = gst_rtsp_message_set_body (request, (guint8 *) msg, msglen);
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+        goto error;
+      }
+
+      g_free (msg);
+      break;
+    }
+
+    default:{
+    }
+  }
+
+  return res;
+
+error:
+  return GST_RTSP_ERROR;
+}
+
+GstRTSPResult
+prepare_trigger_request (GstRTSPWFDClient * client, GstRTSPMessage * request,
+    GstWFDTriggerType trigger_type, gchar * url)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+
+  /* initialize the request */
+  res = gst_rtsp_message_init_request (request, GST_RTSP_SET_PARAMETER, url);
+  if (res < 0) {
+    GST_ERROR ("init request failed");
+    return res;
+  }
+
+  switch (trigger_type) {
+    case WFD_TRIGGER_SETUP:{
+      gchar *msg;
+      guint msglen = 0;
+      GString *msglength;
+
+      /* add content type */
+      res =
+          gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+          "text/parameters");
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp request...");
+        goto error;
+      }
+
+      _set_wfd_message_body (client, M5_REQ_MSG, &msg, &msglen);
+      msglength = g_string_new ("");
+      g_string_append_printf (msglength, "%d", msglen);
+      GST_DEBUG ("M5 server side message body: %s", msg);
+
+      /* add content-length type */
+      res =
+          gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_LENGTH,
+          g_string_free (msglength, FALSE));
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+        goto error;
+      }
+
+      res = gst_rtsp_message_set_body (request, (guint8 *) msg, msglen);
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+        goto error;
+      }
+
+      g_free (msg);
+      break;
+    }
+    case WFD_TRIGGER_TEARDOWN:{
+      gchar *msg;
+      guint msglen = 0;
+      GString *msglength;
+
+      /* add content type */
+      res =
+          gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+          "text/parameters");
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp request...");
+        goto error;
+      }
+
+      _set_wfd_message_body (client, TEARDOWN_TRIGGER, &msg, &msglen);
+      msglength = g_string_new ("");
+      g_string_append_printf (msglength, "%d", msglen);
+      GST_DEBUG ("Trigger TEARDOWN server side message body: %s", msg);
+
+      /* add content-length type */
+      res =
+          gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_LENGTH,
+          g_string_free (msglength, FALSE));
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+        goto error;
+      }
+
+      res = gst_rtsp_message_set_body (request, (guint8 *) msg, msglen);
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+        goto error;
+      }
+
+      g_free (msg);
+      break;
+    }
+    case WFD_TRIGGER_PLAY:{
+      gchar *msg;
+      guint msglen = 0;
+      GString *msglength;
+
+      /* add content type */
+      res =
+          gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+          "text/parameters");
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp request...");
+        goto error;
+      }
+
+      _set_wfd_message_body (client, PLAY_TRIGGER, &msg, &msglen);
+      msglength = g_string_new ("");
+      g_string_append_printf (msglength, "%d", msglen);
+      GST_DEBUG ("Trigger PLAY server side message body: %s", msg);
+
+      /* add content-length type */
+      res =
+          gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_LENGTH,
+          g_string_free (msglength, FALSE));
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+        goto error;
+      }
+
+      res = gst_rtsp_message_set_body (request, (guint8 *) msg, msglen);
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+        goto error;
+      }
+
+      g_free (msg);
+      break;
+    }
+    case WFD_TRIGGER_PAUSE:{
+      gchar *msg;
+      guint msglen = 0;
+      GString *msglength;
+
+      /* add content type */
+      res =
+          gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_TYPE,
+          "text/parameters");
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp request...");
+        goto error;
+      }
+
+      _set_wfd_message_body (client, PAUSE_TRIGGER, &msg, &msglen);
+      msglength = g_string_new ("");
+      g_string_append_printf (msglength, "%d", msglen);
+      GST_DEBUG ("Trigger PAUSE server side message body: %s", msg);
+
+      /* add content-length type */
+      res =
+          gst_rtsp_message_add_header (request, GST_RTSP_HDR_CONTENT_LENGTH,
+          g_string_free (msglength, FALSE));
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+        goto error;
+      }
+
+      res = gst_rtsp_message_set_body (request, (guint8 *) msg, msglen);
+      if (res != GST_RTSP_OK) {
+        GST_ERROR_OBJECT (client, "Failed to add header to rtsp message...");
+        goto error;
+      }
+
+      g_free (msg);
+      break;
+    }
+      /* TODO-WFD: implement to handle other trigger type */
+    default:{
+    }
+  }
+
+  return res;
+
+error:
+  return res;
+}
+
+
+void
+send_request (GstRTSPWFDClient * client, GstRTSPSession * session,
+    GstRTSPMessage * request)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPClient *parent_client = GST_RTSP_CLIENT_CAST (client);
+
+  /* remove any previous header */
+  gst_rtsp_message_remove_header (request, GST_RTSP_HDR_SESSION, -1);
+
+  /* add the new session header for new session ids */
+  if (session) {
+    guint timeout;
+    const gchar *sessionid = NULL;
+    gchar *str;
+
+    sessionid = gst_rtsp_session_get_sessionid (session);
+    GST_INFO_OBJECT (client, "Session id : %s", sessionid);
+
+    timeout = gst_rtsp_session_get_timeout (session);
+    if (timeout != DEFAULT_WFD_TIMEOUT)
+      str = g_strdup_printf ("%s; timeout=%d", sessionid, timeout);
+    else
+      str = g_strdup (sessionid);
+
+    gst_rtsp_message_take_header (request, GST_RTSP_HDR_SESSION, str);
+  }
+#if 0
+  if (gst_debug_category_get_threshold (rtsp_wfd_client_debug) >= GST_LEVEL_LOG) {
+    gst_rtsp_message_dump (request);
+  }
+#endif
+  res = gst_rtsp_client_send_message (parent_client, session, request);
+  if (res != GST_RTSP_OK) {
+    GST_ERROR_OBJECT (client, "gst_rtsp_client_send_message failed : %d", res);
+  }
+
+  gst_rtsp_message_unset (request);
+}
+
+/**
+* prepare_response:
+* @client: client object
+* @request : requst message received
+* @response : response to be prepare based on request
+* @method : RTSP method
+*
+* prepare response to the request based on @method & @message_type
+*
+* Returns: a #GstRTSPResult.
+*/
+GstRTSPResult
+prepare_response (GstRTSPWFDClient * client, GstRTSPMessage * request,
+    GstRTSPMessage * response, GstRTSPMethod method)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+
+  switch (method) {
+      /* prepare OPTIONS response */
+    case GST_RTSP_OPTIONS:{
+      GstRTSPMethod options;
+      gchar *tmp = NULL;
+      gchar *str = NULL;
+      gchar *user_agent = NULL;
+
+      options = GST_RTSP_OPTIONS |
+          GST_RTSP_PAUSE |
+          GST_RTSP_PLAY |
+          GST_RTSP_SETUP |
+          GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
+
+      str = gst_rtsp_options_as_text (options);
+
+      /*append WFD specific method */
+      tmp = g_strdup (", org.wfa.wfd1.0");
+      g_strlcat (str, tmp, strlen (tmp) + strlen (str) + 1);
+
+      gst_rtsp_message_init_response (response, GST_RTSP_STS_OK,
+          gst_rtsp_status_as_text (GST_RTSP_STS_OK), request);
+
+      gst_rtsp_message_add_header (response, GST_RTSP_HDR_PUBLIC, str);
+      g_free (str);
+      str = NULL;
+      res =
+          gst_rtsp_message_get_header (request, GST_RTSP_HDR_USER_AGENT,
+          &user_agent, 0);
+      if (res == GST_RTSP_OK) {
+        gst_rtsp_message_add_header (response, GST_RTSP_HDR_USER_AGENT,
+            user_agent);
+      } else
+        res = GST_RTSP_OK;
+      break;
+    }
+    default:
+      GST_ERROR_OBJECT (client, "Unhandled method...");
+      return GST_RTSP_EINVAL;
+      break;
+  }
+
+  return res;
+}
+
+static void
+send_generic_wfd_response (GstRTSPWFDClient * client, GstRTSPStatusCode code,
+    GstRTSPContext * ctx)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPClient *parent_client = GST_RTSP_CLIENT_CAST (client);
+
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
+
+  res = gst_rtsp_client_send_message (parent_client, NULL, ctx->response);
+  if (res != GST_RTSP_OK) {
+    GST_ERROR_OBJECT (client, "gst_rtsp_client_send_message failed : %d", res);
+  }
+}
+
+
+static GstRTSPResult
+handle_M1_message (GstRTSPWFDClient * client)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPMessage request = { 0 };
+
+  res = prepare_request (client, &request, GST_RTSP_OPTIONS, (gchar *) "*");
+  if (GST_RTSP_OK != res) {
+    GST_ERROR_OBJECT (client, "Failed to prepare M1 request....\n");
+    return res;
+  }
+
+  GST_DEBUG_OBJECT (client, "Sending M1 request.. (OPTIONS request)");
+
+  send_request (client, NULL, &request);
+
+  return res;
+}
+
+/**
+* handle_M3_message:
+* @client: client object
+*
+* Handles M3 WFD message.
+* This API will send M3 message (GET_PARAMETER) to WFDSink to query supported formats by the WFDSink.
+* After getting supported formats info, this API will set those values on WFDConfigMessage obj
+*
+* Returns: a #GstRTSPResult.
+*/
+static GstRTSPResult
+handle_M3_message (GstRTSPWFDClient * client)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPMessage request = { 0 };
+  GstRTSPUrl *url = NULL;
+  gchar *url_str = NULL;
+
+  GstRTSPClient *parent_client = GST_RTSP_CLIENT_CAST (client);
+  GstRTSPConnection *connection =
+      gst_rtsp_client_get_connection (parent_client);
+
+  url = gst_rtsp_connection_get_url (connection);
+  if (url == NULL) {
+    GST_ERROR_OBJECT (client, "Failed to get connection URL");
+    res = GST_RTSP_ERROR;
+    goto error;
+  }
+
+  url_str = gst_rtsp_url_get_request_uri (url);
+  if (url_str == NULL) {
+    GST_ERROR_OBJECT (client, "Failed to get connection URL");
+    res = GST_RTSP_ERROR;
+    goto error;
+  }
+
+  res = prepare_request (client, &request, GST_RTSP_GET_PARAMETER, url_str);
+  if (GST_RTSP_OK != res) {
+    GST_ERROR_OBJECT (client, "Failed to prepare M3 request....\n");
+    goto error;
+  }
+
+  GST_DEBUG_OBJECT (client, "Sending GET_PARAMETER request message (M3)...");
+
+  send_request (client, NULL, &request);
+
+  return res;
+
+error:
+  return res;
+}
+
+static GstRTSPResult
+handle_M4_message (GstRTSPWFDClient * client)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPMessage request = { 0 };
+  GstRTSPUrl *url = NULL;
+  gchar *url_str = NULL;
+
+  GstRTSPClient *parent_client = GST_RTSP_CLIENT_CAST (client);
+  GstRTSPConnection *connection =
+      gst_rtsp_client_get_connection (parent_client);
+
+  url = gst_rtsp_connection_get_url (connection);
+  if (url == NULL) {
+    GST_ERROR_OBJECT (client, "Failed to get connection URL");
+    res = GST_RTSP_ERROR;
+    goto error;
+  }
+
+  url_str = gst_rtsp_url_get_request_uri (url);
+  if (url_str == NULL) {
+    GST_ERROR_OBJECT (client, "Failed to get connection URL");
+    res = GST_RTSP_ERROR;
+    goto error;
+  }
+
+  res = prepare_request (client, &request, GST_RTSP_SET_PARAMETER, url_str);
+  if (GST_RTSP_OK != res) {
+    GST_ERROR_OBJECT (client, "Failed to prepare M4 request....\n");
+    goto error;
+  }
+
+  GST_DEBUG_OBJECT (client, "Sending SET_PARAMETER request message (M4)...");
+
+  send_request (client, NULL, &request);
+
+  return res;
+
+error:
+  return res;
+}
+
+GstRTSPResult
+gst_rtsp_wfd_client_trigger_request (GstRTSPWFDClient * client,
+    GstWFDTriggerType type)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPMessage request = { 0 };
+  GstRTSPUrl *url = NULL;
+  gchar *url_str = NULL;
+
+  GstRTSPClient *parent_client = GST_RTSP_CLIENT_CAST (client);
+  GstRTSPConnection *connection =
+      gst_rtsp_client_get_connection (parent_client);
+
+  url = gst_rtsp_connection_get_url (connection);
+  if (url == NULL) {
+    GST_ERROR_OBJECT (client, "Failed to get connection URL");
+    res = GST_RTSP_ERROR;
+    goto error;
+  }
+
+  url_str = gst_rtsp_url_get_request_uri (url);
+  if (url_str == NULL) {
+    GST_ERROR_OBJECT (client, "Failed to get connection URL");
+    res = GST_RTSP_ERROR;
+    goto error;
+  }
+
+  res = prepare_trigger_request (client, &request, type, url_str);
+  if (GST_RTSP_OK != res) {
+    GST_ERROR_OBJECT (client, "Failed to prepare M5 request....\n");
+    goto error;
+  }
+
+  GST_DEBUG_OBJECT (client, "Sending trigger request message...: %d", type);
+
+  send_request (client, NULL, &request);
+
+  return res;
+
+error:
+  return res;
+}
+
+GstRTSPResult
+gst_rtsp_wfd_client_set_video_supported_resolution (GstRTSPWFDClient * client,
+    guint64 supported_reso)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+  g_return_val_if_fail (priv != NULL, GST_RTSP_EINVAL);
+
+  priv->video_resolution_supported = supported_reso;
+  GST_DEBUG ("Resolution : %"G_GUINT64_FORMAT, supported_reso);
+
+  return res;
+}
+
+GstRTSPResult
+gst_rtsp_wfd_client_set_video_native_resolution (GstRTSPWFDClient * client,
+    guint64 native_reso)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+  g_return_val_if_fail (priv != NULL, GST_RTSP_EINVAL);
+
+  priv->video_native_resolution = native_reso;
+  GST_DEBUG ("Native Resolution : %"G_GUINT64_FORMAT, native_reso);
+
+  return res;
+}
+
+GstRTSPResult
+gst_rtsp_wfd_client_set_audio_codec (GstRTSPWFDClient * client,
+    guint8 audio_codec)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+  g_return_val_if_fail (priv != NULL, GST_RTSP_EINVAL);
+
+  priv->audio_codec = audio_codec;
+  GST_DEBUG ("Audio codec : %d", audio_codec);
+
+  return res;
+}
+
+static gboolean
+wfd_ckeck_keep_alive_response (gpointer userdata)
+{
+  GstRTSPWFDClient *client = (GstRTSPWFDClient *)userdata;
+  GstRTSPWFDClientPrivate *priv = NULL;
+  if (!client) {
+    return FALSE;
+  }
+
+  priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, GST_RTSP_EINVAL);
+
+  if (priv->keep_alive_flag) {
+    return FALSE;
+  }
+  else {
+    GST_INFO ("%p: source error notification", client);
+    // FIXME Do something here. Maybe emit some signal?
+    return FALSE;
+  }
+}
+
+/*Sending keep_alive (M16) message.
+  Without calling prepare_request function.*/
+static GstRTSPResult
+handle_M16_message (GstRTSPWFDClient * client)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPMessage request = { 0 };
+  gchar *url_str = NULL;
+
+  url_str = g_strdup("rtsp://localhost/wfd1.0");
+
+  res = gst_rtsp_message_init_request (&request, GST_RTSP_GET_PARAMETER, url_str);
+  if (res < 0) {
+    GST_ERROR ("init request failed");
+    return FALSE;
+  }
+
+  send_request (client, NULL, &request);
+  return GST_RTSP_OK;
+}
+
+/*CHecking whether source has got response of any request.
+ * If yes, keep alive message is sent otherwise error message
+ * will be displayed.*/
+static gboolean
+keep_alive_condition(gpointer userdata)
+{
+  GstRTSPWFDClient *client;
+  GstRTSPWFDClientPrivate *priv;
+  GstRTSPResult res;
+  client = (GstRTSPWFDClient *)userdata;
+  if (!client) {
+    return FALSE;
+  }
+  priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+  g_return_val_if_fail (priv != NULL, FALSE);
+
+  g_mutex_lock(&priv->keep_alive_lock);
+  if(!priv->keep_alive_flag) {
+    g_timeout_add(5000, wfd_ckeck_keep_alive_response, client);
+  }
+  else {
+    GST_DEBUG_OBJECT (client, "have received last keep alive message response");
+  }
+
+  GST_DEBUG("sending keep alive message");
+  res = handle_M16_message(client);
+  if(res == GST_RTSP_OK) {
+    priv->keep_alive_flag = FALSE;
+  } else {
+    GST_ERROR_OBJECT (client, "Failed to send Keep Alive Message");
+    g_mutex_unlock(&priv->keep_alive_lock);
+    return FALSE;
+  }
+
+  g_mutex_unlock(&priv->keep_alive_lock);
+  return TRUE;
+}
+
+static
+void wfd_set_keep_alive_condition(GstRTSPWFDClient * client)
+{
+  g_timeout_add((DEFAULT_WFD_TIMEOUT-5)*1000, keep_alive_condition, client);
+}
+
+void
+gst_rtsp_wfd_client_set_host_address (GstRTSPWFDClient *client, const gchar * address)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+
+  g_return_if_fail (priv != NULL);
+
+  if (priv->host_address) {
+    g_free (priv->host_address);
+  }
+
+  priv->host_address = g_strdup (address);
+}
+
+guint
+gst_rtsp_wfd_client_get_audio_codec(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->caCodec;
+}
+
+guint
+gst_rtsp_wfd_client_get_audio_freq(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cFreq;
+}
+
+guint
+gst_rtsp_wfd_client_get_audio_channels(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cChanels;
+}
+
+guint
+gst_rtsp_wfd_client_get_audio_bit_width(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cBitwidth;
+}
+
+guint
+gst_rtsp_wfd_client_get_audio_latency(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->caLatency;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_codec(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cvCodec;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_native(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cNative;
+}
+
+guint64
+gst_rtsp_wfd_client_get_video_native_resolution(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cNativeResolution;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_cea_resolution(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cCEAResolution;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_vesa_resolution(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cVESAResolution;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_hh_resolution(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cHHResolution;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_profile(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cProfile;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_level(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cLevel;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_latency(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cvLatency;
+}
+
+guint32
+gst_rtsp_wfd_client_get_video_max_height(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cMaxHeight;
+}
+
+guint32
+gst_rtsp_wfd_client_get_video_max_width(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cMaxWidth;
+}
+
+guint32
+gst_rtsp_wfd_client_get_video_framerate(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cFramerate;
+}
+
+guint32
+gst_rtsp_wfd_client_get_video_min_slice_size(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cmin_slice_size;
+}
+
+guint32
+gst_rtsp_wfd_client_get_video_slice_enc_params(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cslice_enc_params;
+}
+
+guint
+gst_rtsp_wfd_client_get_video_framerate_control(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->cframe_rate_control;
+}
+
+guint32
+gst_rtsp_wfd_client_get_rtp_port0(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->crtp_port0;
+}
+
+guint32
+gst_rtsp_wfd_client_get_rtp_port1(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->crtp_port1;
+}
+
+gboolean
+gst_rtsp_wfd_client_get_edid_supported(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->edid_supported;
+}
+
+guint32
+gst_rtsp_wfd_client_get_edid_hresolution(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->edid_hres;
+}
+
+guint32
+gst_rtsp_wfd_client_get_edid_vresolution(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->edid_vres;
+}
+
+gboolean
+gst_rtsp_wfd_client_get_protection_enabled(GstRTSPWFDClient *client)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  return priv->protection_enabled;
+}
+
+void
+gst_rtsp_wfd_client_set_audio_freq(GstRTSPWFDClient *client, guint freq)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  priv->cFreq = freq;
+}
+
+void
+gst_rtsp_wfd_client_set_edid_supported(GstRTSPWFDClient *client, gboolean supported)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  priv->edid_supported = supported;
+}
+
+void
+gst_rtsp_wfd_client_set_edid_hresolution(GstRTSPWFDClient *client, guint32 reso)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  priv->edid_hres = reso;
+}
+
+void
+gst_rtsp_wfd_client_set_edid_vresolution(GstRTSPWFDClient *client, guint32 reso)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  priv->edid_vres = reso;
+}
+
+void
+gst_rtsp_wfd_client_set_protection_enabled(GstRTSPWFDClient *client, gboolean enable)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  priv->protection_enabled = enable;
+}
+
+void gst_rtsp_wfd_client_set_keep_alive_flag(GstRTSPWFDClient *client, gboolean flag)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_val_if_fail (priv != NULL, 0);
+
+  g_mutex_lock(&priv->keep_alive_lock);
+  if (priv->keep_alive_flag == !(flag))
+    priv->keep_alive_flag = flag;
+  g_mutex_unlock(&priv->keep_alive_lock);
+}
+
+void
+gst_rtsp_wfd_client_set_aud_codec (GstRTSPWFDClient *client, guint acodec)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->caCodec = acodec;
+}
+
+void
+gst_rtsp_wfd_client_set_audio_channels(GstRTSPWFDClient *client, guint channels)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cChanels = channels;
+}
+
+void
+gst_rtsp_wfd_client_set_audio_bit_width(GstRTSPWFDClient *client, guint bwidth)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cBitwidth = bwidth;
+}
+
+void
+gst_rtsp_wfd_client_set_audio_latency(GstRTSPWFDClient *client, guint latency)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->caLatency = latency;
+}
+
+void
+gst_rtsp_wfd_client_set_video_codec(GstRTSPWFDClient *client, guint vcodec)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cvCodec = vcodec;
+}
+
+void
+gst_rtsp_wfd_client_set_video_native(GstRTSPWFDClient *client, guint native)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cNative = native;
+}
+
+void
+gst_rtsp_wfd_client_set_vid_native_resolution(GstRTSPWFDClient *client, guint64 res)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cNativeResolution = res;
+}
+
+void
+gst_rtsp_wfd_client_set_video_cea_resolution(GstRTSPWFDClient *client, guint res)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cCEAResolution = res;
+}
+
+void
+gst_rtsp_wfd_client_set_video_vesa_resolution(GstRTSPWFDClient *client, guint res)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cVESAResolution = res;
+}
+
+void
+gst_rtsp_wfd_client_set_video_hh_resolution(GstRTSPWFDClient *client, guint res)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cHHResolution = res;
+}
+
+void
+gst_rtsp_wfd_client_set_video_profile(GstRTSPWFDClient *client, guint profile)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cProfile = profile;
+}
+
+void
+gst_rtsp_wfd_client_set_video_level(GstRTSPWFDClient *client, guint level)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cLevel = level;
+}
+
+void
+gst_rtsp_wfd_client_set_video_latency(GstRTSPWFDClient *client, guint latency)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cvLatency = latency;
+}
+
+void
+gst_rtsp_wfd_client_set_video_max_height(GstRTSPWFDClient *client, guint32 height)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cMaxHeight = height;
+}
+
+void
+gst_rtsp_wfd_client_set_video_max_width(GstRTSPWFDClient *client, guint32 width)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cMaxWidth = width;
+}
+
+void
+gst_rtsp_wfd_client_set_video_framerate(GstRTSPWFDClient *client, guint32 framerate)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cFramerate = framerate;
+}
+
+void
+gst_rtsp_wfd_client_set_video_min_slice_size(GstRTSPWFDClient *client, guint32 slice_size)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cmin_slice_size = slice_size;
+}
+
+void
+gst_rtsp_wfd_client_set_video_slice_enc_params(GstRTSPWFDClient *client, guint32 enc_params)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cslice_enc_params = enc_params;
+}
+
+void
+gst_rtsp_wfd_client_set_video_framerate_control(GstRTSPWFDClient *client, guint framerate)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->cframe_rate_control = framerate;
+}
+
+void
+gst_rtsp_wfd_client_set_rtp_port0(GstRTSPWFDClient *client, guint32 port)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->crtp_port0 = port;
+}
+
+void
+gst_rtsp_wfd_client_set_rtp_port1(GstRTSPWFDClient *client, guint32 port)
+{
+  GstRTSPWFDClientPrivate *priv = GST_RTSP_WFD_CLIENT_GET_PRIVATE (client);
+  g_return_if_fail (priv != NULL);
+
+  priv->crtp_port1 = port;
+}
diff --git a/gst/rtsp-server/rtsp-client-wfd.h b/gst/rtsp-server/rtsp-client-wfd.h
new file mode 100644
index 0000000..25d96de
--- /dev/null
+++ b/gst/rtsp-server/rtsp-client-wfd.h
@@ -0,0 +1,220 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+#include 
+
+#ifndef __GST_RTSP_WFD_CLIENT_H__
+#define __GST_RTSP_WFD_CLIENT_H__
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPWFDClient GstRTSPWFDClient;
+typedef struct _GstRTSPWFDClientClass GstRTSPWFDClientClass;
+typedef struct _GstRTSPWFDClientPrivate GstRTSPWFDClientPrivate;
+
+#include "rtsp-context.h"
+#include "rtsp-mount-points.h"
+#include "rtsp-sdp.h"
+#include "rtsp-auth.h"
+#include "rtsp-client.h"
+#include "gstwfdmessage.h"
+
+#define GST_TYPE_RTSP_WFD_CLIENT              (gst_rtsp_wfd_client_get_type ())
+#define GST_IS_RTSP_WFD_CLIENT(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_WFD_CLIENT))
+#define GST_IS_RTSP_WFD_CLIENT_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_WFD_CLIENT))
+#define GST_RTSP_WFD_CLIENT_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_WFD_CLIENT, GstRTSPWFDClientClass))
+#define GST_RTSP_WFD_CLIENT(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_WFD_CLIENT, GstRTSPWFDClient))
+#define GST_RTSP_WFD_CLIENT_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_WFD_CLIENT, GstRTSPWFDClientClass))
+#define GST_RTSP_WFD_CLIENT_CAST(obj)         ((GstRTSPWFDClient*)(obj))
+#define GST_RTSP_WFD_CLIENT_CLASS_CAST(klass) ((GstRTSPWFDClientClass*)(klass))
+
+
+/**
+ *
+ */
+typedef enum {
+  WFD_TRIGGER_SETUP,
+  WFD_TRIGGER_PAUSE,
+  WFD_TRIGGER_TEARDOWN,
+  WFD_TRIGGER_PLAY
+} GstWFDTriggerType;
+
+/**
+ * GstRTSPWFDClientSendFunc:
+ * @client: a #GstRTSPWFDClient
+ * @message: a #GstRTSPMessage
+ * @close: close the connection
+ * @user_data: user data when registering the callback
+ *
+ * This callback is called when @client wants to send @message. When @close is
+ * %TRUE, the connection should be closed when the message has been sent.
+ *
+ * Returns: %TRUE on success.
+ */
+typedef gboolean (*GstRTSPWFDClientSendFunc)      (GstRTSPWFDClient *client,
+                                                GstRTSPMessage *message,
+                                                gboolean close,
+                                                gpointer user_data);
+
+/**
+ * GstRTSPWFDClient:
+ *
+ * The client object represents the connection and its state with a client.
+ */
+struct _GstRTSPWFDClient {
+  GstRTSPClient  parent;
+
+  gint           supported_methods;
+  /*< private >*/
+  GstRTSPWFDClientPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPWFDClientClass:
+ * @create_sdp: called when the SDP needs to be created for media.
+ * @configure_client_media: called when the stream in media needs to be configured.
+ *    The default implementation will configure the blocksize on the payloader when
+ *    spcified in the request headers.
+ * @configure_client_transport: called when the client transport needs to be
+ *    configured.
+ * @params_set: set parameters. This function should also initialize the
+ *    RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
+ * @params_get: get parameters. This function should also initialize the
+ *    RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
+ *
+ * The client class structure.
+ */
+struct _GstRTSPWFDClientClass {
+  GstRTSPClientClass  parent_class;
+
+  GstRTSPResult       (*prepare_resource) (GstRTSPWFDClient *client, GstRTSPContext *ctx);
+  GstRTSPResult       (*confirm_resource) (GstRTSPWFDClient *client, GstRTSPContext *ctx);
+
+  /* signals */
+  void     (*wfd_options_request)         (GstRTSPWFDClient *client, GstRTSPContext *ctx);
+  void     (*wfd_get_param_request)       (GstRTSPWFDClient *client, GstRTSPContext *ctx);
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+GType                 gst_rtsp_wfd_client_get_type          (void);
+
+GstRTSPWFDClient *    gst_rtsp_wfd_client_new               (void);
+
+void                  gst_rtsp_wfd_client_set_host_address  (
+                          GstRTSPWFDClient *client, const gchar * address);
+
+void                  gst_rtsp_wfd_client_start_wfd(GstRTSPWFDClient *client);
+GstRTSPResult         gst_rtsp_wfd_client_trigger_request (
+                          GstRTSPWFDClient * client, GstWFDTriggerType type);
+
+GstRTSPResult         gst_rtsp_wfd_client_set_video_supported_resolution (
+                          GstRTSPWFDClient * client, guint64 supported_reso);
+GstRTSPResult         gst_rtsp_wfd_client_set_video_native_resolution (
+		                      GstRTSPWFDClient * client, guint64 native_reso);
+GstRTSPResult         gst_rtsp_wfd_client_set_audio_codec (
+		                      GstRTSPWFDClient * client, guint8 audio_codec);
+
+guint     gst_rtsp_wfd_client_get_audio_codec(GstRTSPWFDClient *client);
+guint     gst_rtsp_wfd_client_get_audio_freq(GstRTSPWFDClient *client);
+guint     gst_rtsp_wfd_client_get_audio_channels(GstRTSPWFDClient *client);
+guint     gst_rtsp_wfd_client_get_audio_bit_width(GstRTSPWFDClient *client);
+guint     gst_rtsp_wfd_client_get_audio_latency(GstRTSPWFDClient *client);
+guint     gst_rtsp_wfd_client_get_video_codec(GstRTSPWFDClient *client);
+guint     gst_rtsp_wfd_client_get_video_native(GstRTSPWFDClient *client);
+guint64   gst_rtsp_wfd_client_get_video_native_resolution(GstRTSPWFDClient *client);
+guint     gst_rtsp_wfd_client_get_video_cea_resolution(GstRTSPWFDClient *client);
+guint     gst_rtsp_wfd_client_get_video_vesa_resolution(GstRTSPWFDClient *client);
+guint     gst_rtsp_wfd_client_get_video_hh_resolution(GstRTSPWFDClient *client);
+guint     gst_rtsp_wfd_client_get_video_profile(GstRTSPWFDClient *client);
+guint     gst_rtsp_wfd_client_get_video_level(GstRTSPWFDClient *client);
+guint     gst_rtsp_wfd_client_get_video_latency(GstRTSPWFDClient *client);
+guint32   gst_rtsp_wfd_client_get_video_max_height(GstRTSPWFDClient *client);
+guint32   gst_rtsp_wfd_client_get_video_max_width(GstRTSPWFDClient *client);
+guint32   gst_rtsp_wfd_client_get_video_framerate(GstRTSPWFDClient *client);
+guint32   gst_rtsp_wfd_client_get_video_min_slice_size(GstRTSPWFDClient *client);
+guint32   gst_rtsp_wfd_client_get_video_slice_enc_params(GstRTSPWFDClient *client);
+guint     gst_rtsp_wfd_client_get_video_framerate_control(GstRTSPWFDClient *client);
+guint32   gst_rtsp_wfd_client_get_rtp_port0(GstRTSPWFDClient *client);
+guint32   gst_rtsp_wfd_client_get_rtp_port1(GstRTSPWFDClient *client);
+gboolean  gst_rtsp_wfd_client_get_edid_supported(GstRTSPWFDClient *client);
+guint32   gst_rtsp_wfd_client_get_edid_hresolution(GstRTSPWFDClient *client);
+guint32   gst_rtsp_wfd_client_get_edid_vresolution(GstRTSPWFDClient *client);
+gboolean  gst_rtsp_wfd_client_get_protection_enabled(GstRTSPWFDClient *client);
+
+void gst_rtsp_wfd_client_set_audio_freq(GstRTSPWFDClient *client, guint freq);
+void gst_rtsp_wfd_client_set_edid_supported(GstRTSPWFDClient *client, gboolean supported);
+void gst_rtsp_wfd_client_set_edid_hresolution(GstRTSPWFDClient *client, guint32 reso);
+void gst_rtsp_wfd_client_set_edid_vresolution(GstRTSPWFDClient *client, guint32 reso);
+void gst_rtsp_wfd_client_set_protection_enabled(GstRTSPWFDClient *client, gboolean enable);
+void gst_rtsp_wfd_client_set_keep_alive_flag(GstRTSPWFDClient *client, gboolean flag);
+void gst_rtsp_wfd_client_set_aud_codec(GstRTSPWFDClient *client, guint acodec);
+void gst_rtsp_wfd_client_set_audio_channels(GstRTSPWFDClient *client, guint channels);
+void gst_rtsp_wfd_client_set_audio_bit_width(GstRTSPWFDClient *client, guint bwidth);
+void gst_rtsp_wfd_client_set_audio_latency(GstRTSPWFDClient *client, guint latency);
+void gst_rtsp_wfd_client_set_video_codec(GstRTSPWFDClient *client, guint vcodec);
+void gst_rtsp_wfd_client_set_video_native(GstRTSPWFDClient *client, guint native);
+void gst_rtsp_wfd_client_set_vid_native_resolution(GstRTSPWFDClient *client, guint64 res);
+void gst_rtsp_wfd_client_set_video_cea_resolution(GstRTSPWFDClient *client, guint res);
+void gst_rtsp_wfd_client_set_video_vesa_resolution(GstRTSPWFDClient *client, guint res);
+void gst_rtsp_wfd_client_set_video_hh_resolution(GstRTSPWFDClient *client, guint res);
+void gst_rtsp_wfd_client_set_video_profile(GstRTSPWFDClient *client, guint profile);
+void gst_rtsp_wfd_client_set_video_level(GstRTSPWFDClient *client, guint level);
+void gst_rtsp_wfd_client_set_video_latency(GstRTSPWFDClient *client, guint latency);
+void gst_rtsp_wfd_client_set_video_max_height(GstRTSPWFDClient *client, guint32 height);
+void gst_rtsp_wfd_client_set_video_max_width(GstRTSPWFDClient *client, guint32 width);
+void gst_rtsp_wfd_client_set_video_framerate(GstRTSPWFDClient *client, guint32 framerate);
+void gst_rtsp_wfd_client_set_video_min_slice_size(GstRTSPWFDClient *client, guint32 slice_size);
+void gst_rtsp_wfd_client_set_video_slice_enc_params(GstRTSPWFDClient *client, guint32 enc_params);
+void gst_rtsp_wfd_client_set_video_framerate_control(GstRTSPWFDClient *client, guint framerate);
+void gst_rtsp_wfd_client_set_rtp_port0(GstRTSPWFDClient *client, guint32 port);
+void gst_rtsp_wfd_client_set_rtp_port1(GstRTSPWFDClient *client, guint32 port);
+
+/**
+ * GstRTSPWFDClientSessionFilterFunc:
+ * @client: a #GstRTSPWFDClient object
+ * @sess: a #GstRTSPSession in @client
+ * @user_data: user data that has been given to gst_rtsp_wfd_client_session_filter()
+ *
+ * This function will be called by the gst_rtsp_wfd_client_session_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @sess will be removed
+ * from @client.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @sess untouched in
+ * @client.
+ *
+ * A value of #GST_RTSP_FILTER_REF will add @sess to the result #GList of
+ * gst_rtsp_wfd_client_session_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPWFDClientSessionFilterFunc)  (GstRTSPWFDClient *client,
+                                                                GstRTSPSession *sess,
+                                                                gpointer user_data);
+
+
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_WFD_CLIENT_H__ */
diff --git a/gst/rtsp-server/rtsp-client.c b/gst/rtsp-server/rtsp-client.c
new file mode 100644
index 0000000..b169d4b
--- /dev/null
+++ b/gst/rtsp-server/rtsp-client.c
@@ -0,0 +1,3568 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-client
+ * @short_description: A client connection state
+ * @see_also: #GstRTSPServer, #GstRTSPThreadPool
+ *
+ * The client object handles the connection with a client for as long as a TCP
+ * connection is open.
+ *
+ * A #GstRTSPClient is created by #GstRTSPServer when a new connection is
+ * accepted and it inherits the #GstRTSPMountPoints, #GstRTSPSessionPool,
+ * #GstRTSPAuth and #GstRTSPThreadPool from the server.
+ *
+ * The client connection should be configured with the #GstRTSPConnection using
+ * gst_rtsp_client_set_connection() before it can be attached to a #GMainContext
+ * using gst_rtsp_client_attach(). From then on the client will handle requests
+ * on the connection.
+ *
+ * Use gst_rtsp_client_session_filter() to iterate or modify all the
+ * #GstRTSPSession objects managed by the client object.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include 
+#include 
+
+#include 
+
+#include "rtsp-client.h"
+#include "rtsp-sdp.h"
+#include "rtsp-params.h"
+
+#define GST_RTSP_CLIENT_GET_PRIVATE(obj)  \
+   (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientPrivate))
+
+/* locking order:
+ * send_lock, lock, tunnels_lock
+ */
+
+struct _GstRTSPClientPrivate
+{
+  GMutex lock;                  /* protects everything else */
+  GMutex send_lock;
+  GMutex watch_lock;
+  GstRTSPConnection *connection;
+  GstRTSPWatch *watch;
+  GMainContext *watch_context;
+  guint close_seq;
+  gchar *server_ip;
+  gboolean is_ipv6;
+
+  GstRTSPClientSendFunc send_func;      /* protected by send_lock */
+  gpointer send_data;           /* protected by send_lock */
+  GDestroyNotify send_notify;   /* protected by send_lock */
+
+  GstRTSPSessionPool *session_pool;
+  gulong session_removed_id;
+  GstRTSPMountPoints *mount_points;
+  GstRTSPAuth *auth;
+  GstRTSPThreadPool *thread_pool;
+
+  /* used to cache the media in the last requested DESCRIBE so that
+   * we can pick it up in the next SETUP immediately */
+  gchar *path;
+  GstRTSPMedia *media;
+
+  GList *transports;
+  GList *sessions;
+  guint sessions_cookie;
+
+  gboolean drop_backlog;
+};
+
+static GMutex tunnels_lock;
+static GHashTable *tunnels;     /* protected by tunnels_lock */
+
+#define DEFAULT_SESSION_POOL            NULL
+#define DEFAULT_MOUNT_POINTS            NULL
+#define DEFAULT_DROP_BACKLOG            TRUE
+
+enum
+{
+  PROP_0,
+  PROP_SESSION_POOL,
+  PROP_MOUNT_POINTS,
+  PROP_DROP_BACKLOG,
+  PROP_LAST
+};
+
+enum
+{
+  SIGNAL_CLOSED,
+  SIGNAL_NEW_SESSION,
+  SIGNAL_OPTIONS_REQUEST,
+  SIGNAL_DESCRIBE_REQUEST,
+  SIGNAL_SETUP_REQUEST,
+  SIGNAL_PLAY_REQUEST,
+  SIGNAL_PAUSE_REQUEST,
+  SIGNAL_TEARDOWN_REQUEST,
+  SIGNAL_SET_PARAMETER_REQUEST,
+  SIGNAL_GET_PARAMETER_REQUEST,
+  SIGNAL_HANDLE_RESPONSE,
+  SIGNAL_SEND_MESSAGE,
+  SIGNAL_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_client_debug);
+#define GST_CAT_DEFAULT rtsp_client_debug
+
+static guint gst_rtsp_client_signals[SIGNAL_LAST] = { 0 };
+
+static void gst_rtsp_client_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec);
+static void gst_rtsp_client_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_client_finalize (GObject * obj);
+
+static GstSDPMessage *create_sdp (GstRTSPClient * client, GstRTSPMedia * media);
+static void unlink_session_transports (GstRTSPClient * client,
+    GstRTSPSession * session, GstRTSPSessionMedia * sessmedia);
+static gboolean default_configure_client_media (GstRTSPClient * client,
+    GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
+static gboolean default_configure_client_transport (GstRTSPClient * client,
+    GstRTSPContext * ctx, GstRTSPTransport * ct);
+static GstRTSPResult default_params_set (GstRTSPClient * client,
+    GstRTSPContext * ctx);
+static GstRTSPResult default_params_get (GstRTSPClient * client,
+    GstRTSPContext * ctx);
+static gchar *default_make_path_from_uri (GstRTSPClient * client,
+    const GstRTSPUrl * uri);
+static gboolean default_handle_options_request (GstRTSPClient * client,
+    GstRTSPContext * ctx);
+static gboolean default_handle_set_param_request (GstRTSPClient * client,
+    GstRTSPContext * ctx);
+static gboolean default_handle_get_param_request (GstRTSPClient * client,
+    GstRTSPContext * ctx);
+static void client_session_removed (GstRTSPSessionPool * pool,
+    GstRTSPSession * session, GstRTSPClient * client);
+
+G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_client_class_init (GstRTSPClientClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPClientPrivate));
+
+  gobject_class = G_OBJECT_CLASS (klass);
+
+  gobject_class->get_property = gst_rtsp_client_get_property;
+  gobject_class->set_property = gst_rtsp_client_set_property;
+  gobject_class->finalize = gst_rtsp_client_finalize;
+
+  klass->create_sdp = create_sdp;
+  klass->configure_client_media = default_configure_client_media;
+  klass->configure_client_transport = default_configure_client_transport;
+  klass->params_set = default_params_set;
+  klass->params_get = default_params_get;
+  klass->make_path_from_uri = default_make_path_from_uri;
+  klass->handle_options_request = default_handle_options_request;
+  klass->handle_set_param_request = default_handle_set_param_request;
+  klass->handle_get_param_request = default_handle_get_param_request;
+
+  g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
+      g_param_spec_object ("session-pool", "Session Pool",
+          "The session pool to use for client session",
+          GST_TYPE_RTSP_SESSION_POOL,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
+      g_param_spec_object ("mount-points", "Mount Points",
+          "The mount points to use for client session",
+          GST_TYPE_RTSP_MOUNT_POINTS,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_DROP_BACKLOG,
+      g_param_spec_boolean ("drop-backlog", "Drop Backlog",
+          "Drop data when the backlog queue is full",
+          DEFAULT_DROP_BACKLOG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  gst_rtsp_client_signals[SIGNAL_CLOSED] =
+      g_signal_new ("closed", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+      G_STRUCT_OFFSET (GstRTSPClientClass, closed), NULL, NULL,
+      g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
+
+  gst_rtsp_client_signals[SIGNAL_NEW_SESSION] =
+      g_signal_new ("new-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+      G_STRUCT_OFFSET (GstRTSPClientClass, new_session), NULL, NULL,
+      g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_SESSION);
+
+  gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST] =
+      g_signal_new ("options-request", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, options_request),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+      GST_TYPE_RTSP_CONTEXT);
+
+  gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST] =
+      g_signal_new ("describe-request", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, describe_request),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+      GST_TYPE_RTSP_CONTEXT);
+
+  gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST] =
+      g_signal_new ("setup-request", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, setup_request),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+      GST_TYPE_RTSP_CONTEXT);
+
+  gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST] =
+      g_signal_new ("play-request", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, play_request),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+      GST_TYPE_RTSP_CONTEXT);
+
+  gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST] =
+      g_signal_new ("pause-request", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, pause_request),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+      GST_TYPE_RTSP_CONTEXT);
+
+  gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST] =
+      g_signal_new ("teardown-request", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass, teardown_request),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+      GST_TYPE_RTSP_CONTEXT);
+
+  gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST] =
+      g_signal_new ("set-parameter-request", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+          set_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
+      G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+  gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST] =
+      g_signal_new ("get-parameter-request", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+          get_parameter_request), NULL, NULL, g_cclosure_marshal_generic,
+      G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+  gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE] =
+      g_signal_new ("handle-response", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPClientClass,
+          handle_response), NULL, NULL, g_cclosure_marshal_generic,
+      G_TYPE_NONE, 1, GST_TYPE_RTSP_CONTEXT);
+
+  /**
+   * GstRTSPClient::send-message:
+   * @client: The RTSP client
+   * @session: (type GstRtspServer.RTSPSession): The session
+   * @message: (type GstRtsp.RTSPMessage): The message
+   */
+  gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE] =
+      g_signal_new ("send-message", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
+      G_TYPE_NONE, 2, GST_TYPE_RTSP_CONTEXT, G_TYPE_POINTER);
+
+  tunnels =
+      g_hash_table_new_full (g_str_hash, g_str_equal, g_free, g_object_unref);
+  g_mutex_init (&tunnels_lock);
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_client_debug, "rtspclient", 0, "GstRTSPClient");
+}
+
+static void
+gst_rtsp_client_init (GstRTSPClient * client)
+{
+  GstRTSPClientPrivate *priv = GST_RTSP_CLIENT_GET_PRIVATE (client);
+
+  client->priv = priv;
+
+  g_mutex_init (&priv->lock);
+  g_mutex_init (&priv->send_lock);
+  g_mutex_init (&priv->watch_lock);
+  priv->close_seq = 0;
+  priv->drop_backlog = DEFAULT_DROP_BACKLOG;
+}
+
+static GstRTSPFilterResult
+filter_session_media (GstRTSPSession * sess, GstRTSPSessionMedia * sessmedia,
+    gpointer user_data)
+{
+  GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+
+  gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
+  unlink_session_transports (client, sess, sessmedia);
+
+  /* unmanage the media in the session */
+  return GST_RTSP_FILTER_REMOVE;
+}
+
+static void
+client_watch_session (GstRTSPClient * client, GstRTSPSession * session)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+
+  g_mutex_lock (&priv->lock);
+  /* check if we already know about this session */
+  if (g_list_find (priv->sessions, session) == NULL) {
+    GST_INFO ("watching session %p", session);
+
+    priv->sessions = g_list_prepend (priv->sessions, g_object_ref (session));
+    priv->sessions_cookie++;
+
+    /* connect removed session handler, it will be disconnected when the last
+     * session gets removed  */
+    if (priv->session_removed_id == 0)
+      priv->session_removed_id = g_signal_connect_data (priv->session_pool,
+          "session-removed", G_CALLBACK (client_session_removed),
+          g_object_ref (client), (GClosureNotify) g_object_unref, 0);
+  }
+  g_mutex_unlock (&priv->lock);
+
+  return;
+}
+
+/* should be called with lock */
+static void
+client_unwatch_session (GstRTSPClient * client, GstRTSPSession * session,
+    GList * link)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+
+  GST_INFO ("client %p: unwatch session %p", client, session);
+
+  if (link == NULL) {
+    link = g_list_find (priv->sessions, session);
+    if (link == NULL)
+      return;
+  }
+
+  priv->sessions = g_list_delete_link (priv->sessions, link);
+  priv->sessions_cookie++;
+
+  /* if this was the last session, disconnect the handler.
+   * This will also drop the extra client ref */
+  if (!priv->sessions) {
+    g_signal_handler_disconnect (priv->session_pool, priv->session_removed_id);
+    priv->session_removed_id = 0;
+  }
+
+  /* unlink all media managed in this session */
+  gst_rtsp_session_filter (session, filter_session_media, client);
+
+  /* remove the session */
+  g_object_unref (session);
+}
+
+static GstRTSPFilterResult
+cleanup_session (GstRTSPClient * client, GstRTSPSession * sess,
+    gpointer user_data)
+{
+  return GST_RTSP_FILTER_REMOVE;
+}
+
+/* A client is finalized when the connection is broken */
+static void
+gst_rtsp_client_finalize (GObject * obj)
+{
+  GstRTSPClient *client = GST_RTSP_CLIENT (obj);
+  GstRTSPClientPrivate *priv = client->priv;
+
+  GST_INFO ("finalize client %p", client);
+
+  if (priv->watch)
+    gst_rtsp_watch_set_flushing (priv->watch, TRUE);
+  gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+
+  if (priv->watch)
+    g_source_destroy ((GSource *) priv->watch);
+
+  if (priv->watch_context)
+    g_main_context_unref (priv->watch_context);
+
+  /* all sessions should have been removed by now. We keep a ref to
+   * the client object for the session removed handler. The ref is
+   * dropped when the last session is removed from the list. */
+  g_assert (priv->sessions == NULL);
+  g_assert (priv->session_removed_id == 0);
+
+  if (priv->connection)
+    gst_rtsp_connection_free (priv->connection);
+  if (priv->session_pool) {
+    g_object_unref (priv->session_pool);
+  }
+  if (priv->mount_points)
+    g_object_unref (priv->mount_points);
+  if (priv->auth)
+    g_object_unref (priv->auth);
+  if (priv->thread_pool)
+    g_object_unref (priv->thread_pool);
+
+  if (priv->path)
+    g_free (priv->path);
+  if (priv->media) {
+    gst_rtsp_media_unprepare (priv->media);
+    g_object_unref (priv->media);
+  }
+
+  g_free (priv->server_ip);
+  g_mutex_clear (&priv->lock);
+  g_mutex_clear (&priv->send_lock);
+  g_mutex_clear (&priv->watch_lock);
+
+  G_OBJECT_CLASS (gst_rtsp_client_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_client_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRTSPClient *client = GST_RTSP_CLIENT (object);
+  GstRTSPClientPrivate *priv = client->priv;
+
+  switch (propid) {
+    case PROP_SESSION_POOL:
+      g_value_take_object (value, gst_rtsp_client_get_session_pool (client));
+      break;
+    case PROP_MOUNT_POINTS:
+      g_value_take_object (value, gst_rtsp_client_get_mount_points (client));
+      break;
+    case PROP_DROP_BACKLOG:
+      g_value_set_boolean (value, priv->drop_backlog);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+static void
+gst_rtsp_client_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRTSPClient *client = GST_RTSP_CLIENT (object);
+  GstRTSPClientPrivate *priv = client->priv;
+
+  switch (propid) {
+    case PROP_SESSION_POOL:
+      gst_rtsp_client_set_session_pool (client, g_value_get_object (value));
+      break;
+    case PROP_MOUNT_POINTS:
+      gst_rtsp_client_set_mount_points (client, g_value_get_object (value));
+      break;
+    case PROP_DROP_BACKLOG:
+      g_mutex_lock (&priv->lock);
+      priv->drop_backlog = g_value_get_boolean (value);
+      g_mutex_unlock (&priv->lock);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+/**
+ * gst_rtsp_client_new:
+ *
+ * Create a new #GstRTSPClient instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPClient
+ */
+GstRTSPClient *
+gst_rtsp_client_new (void)
+{
+  GstRTSPClient *result;
+
+  result = g_object_new (GST_TYPE_RTSP_CLIENT, NULL);
+
+  return result;
+}
+
+static void
+send_message (GstRTSPClient * client, GstRTSPContext * ctx,
+    GstRTSPMessage * message, gboolean close)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+
+  gst_rtsp_message_add_header (message, GST_RTSP_HDR_SERVER,
+      "GStreamer RTSP server");
+
+  /* remove any previous header */
+  gst_rtsp_message_remove_header (message, GST_RTSP_HDR_SESSION, -1);
+
+  /* add the new session header for new session ids */
+  if (ctx->session) {
+    gst_rtsp_message_take_header (message, GST_RTSP_HDR_SESSION,
+        gst_rtsp_session_get_header (ctx->session));
+  }
+
+  if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
+    gst_rtsp_message_dump (message);
+  }
+
+  if (close)
+    gst_rtsp_message_add_header (message, GST_RTSP_HDR_CONNECTION, "close");
+
+  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SEND_MESSAGE],
+      0, ctx, message);
+
+  g_mutex_lock (&priv->send_lock);
+  if (priv->send_func)
+    priv->send_func (client, message, close, priv->send_data);
+  g_mutex_unlock (&priv->send_lock);
+
+  gst_rtsp_message_unset (message);
+}
+
+static void
+send_generic_response (GstRTSPClient * client, GstRTSPStatusCode code,
+    GstRTSPContext * ctx)
+{
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
+
+  ctx->session = NULL;
+
+  send_message (client, ctx, ctx->response, FALSE);
+}
+
+static void
+send_option_not_supported_response (GstRTSPClient * client,
+    GstRTSPContext * ctx, const gchar * unsupported_options)
+{
+  GstRTSPStatusCode code = GST_RTSP_STS_OPTION_NOT_SUPPORTED;
+
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
+
+  if (unsupported_options != NULL) {
+    gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_UNSUPPORTED,
+        unsupported_options);
+  }
+
+  ctx->session = NULL;
+
+  send_message (client, ctx, ctx->response, FALSE);
+}
+
+static gboolean
+paths_are_equal (const gchar * path1, const gchar * path2, gint len2)
+{
+  if (path1 == NULL || path2 == NULL)
+    return FALSE;
+
+  if (strlen (path1) != len2)
+    return FALSE;
+
+  if (strncmp (path1, path2, len2))
+    return FALSE;
+
+  return TRUE;
+}
+
+/* this function is called to initially find the media for the DESCRIBE request
+ * but is cached for when the same client (without breaking the connection) is
+ * doing a setup for the exact same url. */
+static GstRTSPMedia *
+find_media (GstRTSPClient * client, GstRTSPContext * ctx, gchar * path,
+    gint * matched)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia *media;
+  gint path_len;
+
+  /* find the longest matching factory for the uri first */
+  if (!(factory = gst_rtsp_mount_points_match (priv->mount_points,
+              path, matched)))
+    goto no_factory;
+
+  ctx->factory = factory;
+
+  if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS))
+    goto no_factory_access;
+
+  if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT))
+    goto not_authorized;
+
+  if (matched)
+    path_len = *matched;
+  else
+    path_len = strlen (path);
+
+  if (!paths_are_equal (priv->path, path, path_len)) {
+    GstRTSPThread *thread;
+
+    /* remove any previously cached values before we try to construct a new
+     * media for uri */
+    if (priv->path)
+      g_free (priv->path);
+    priv->path = NULL;
+    if (priv->media) {
+      gst_rtsp_media_unprepare (priv->media);
+      g_object_unref (priv->media);
+    }
+    priv->media = NULL;
+
+    /* prepare the media and add it to the pipeline */
+    if (!(media = gst_rtsp_media_factory_construct (factory, ctx->uri)))
+      goto no_media;
+
+    ctx->media = media;
+
+    thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
+        GST_RTSP_THREAD_TYPE_MEDIA, ctx);
+    if (thread == NULL)
+      goto no_thread;
+
+    /* prepare the media */
+    if (!(gst_rtsp_media_prepare (media, thread)))
+      goto no_prepare;
+
+    /* now keep track of the uri and the media */
+    priv->path = g_strndup (path, path_len);
+    priv->media = media;
+  } else {
+    /* we have seen this path before, used cached media */
+    media = priv->media;
+    ctx->media = media;
+    GST_INFO ("reusing cached media %p for path %s", media, priv->path);
+  }
+
+  g_object_unref (factory);
+  ctx->factory = NULL;
+
+  if (media)
+    g_object_ref (media);
+
+  return media;
+
+  /* ERRORS */
+no_factory:
+  {
+    GST_ERROR ("client %p: no factory for path %s", client, path);
+    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+    return NULL;
+  }
+no_factory_access:
+  {
+    GST_ERROR ("client %p: not authorized to see factory path %s", client,
+        path);
+    /* error reply is already sent */
+    return NULL;
+  }
+not_authorized:
+  {
+    GST_ERROR ("client %p: not authorized for factory path %s", client, path);
+    /* error reply is already sent */
+    return NULL;
+  }
+no_media:
+  {
+    GST_ERROR ("client %p: can't create media", client);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+    g_object_unref (factory);
+    ctx->factory = NULL;
+    return NULL;
+  }
+no_thread:
+  {
+    GST_ERROR ("client %p: can't create thread", client);
+    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+    g_object_unref (media);
+    ctx->media = NULL;
+    g_object_unref (factory);
+    ctx->factory = NULL;
+    return NULL;
+  }
+no_prepare:
+  {
+    GST_ERROR ("client %p: can't prepare media", client);
+    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+    g_object_unref (media);
+    ctx->media = NULL;
+    g_object_unref (factory);
+    ctx->factory = NULL;
+    return NULL;
+  }
+}
+
+static gboolean
+do_send_data (GstBuffer * buffer, guint8 channel, GstRTSPClient * client)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+  GstRTSPMessage message = { 0 };
+  GstMapInfo map_info;
+  guint8 *data;
+  guint usize;
+
+  gst_rtsp_message_init_data (&message, channel);
+
+  /* FIXME, need some sort of iovec RTSPMessage here */
+  if (!gst_buffer_map (buffer, &map_info, GST_MAP_READ))
+    return FALSE;
+
+  gst_rtsp_message_take_body (&message, map_info.data, map_info.size);
+
+  g_mutex_lock (&priv->send_lock);
+  if (priv->send_func)
+    priv->send_func (client, &message, FALSE, priv->send_data);
+  g_mutex_unlock (&priv->send_lock);
+
+  gst_rtsp_message_steal_body (&message, &data, &usize);
+  gst_buffer_unmap (buffer, &map_info);
+
+  gst_rtsp_message_unset (&message);
+
+  return TRUE;
+}
+
+static void
+link_transport (GstRTSPClient * client, GstRTSPSession * session,
+    GstRTSPStreamTransport * trans)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+
+  GST_DEBUG ("client %p: linking transport %p", client, trans);
+
+  gst_rtsp_stream_transport_set_callbacks (trans,
+      (GstRTSPSendFunc) do_send_data,
+      (GstRTSPSendFunc) do_send_data, client, NULL);
+
+  priv->transports = g_list_prepend (priv->transports, trans);
+
+  /* make sure our session can't expire */
+  gst_rtsp_session_prevent_expire (session);
+}
+
+static void
+link_session_transports (GstRTSPClient * client, GstRTSPSession * session,
+    GstRTSPSessionMedia * sessmedia)
+{
+  guint n_streams, i;
+
+  n_streams =
+      gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia));
+  for (i = 0; i < n_streams; i++) {
+    GstRTSPStreamTransport *trans;
+    const GstRTSPTransport *tr;
+
+    /* get the transport, if there is no transport configured, skip this stream */
+    trans = gst_rtsp_session_media_get_transport (sessmedia, i);
+    if (trans == NULL)
+      continue;
+
+    tr = gst_rtsp_stream_transport_get_transport (trans);
+
+    if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
+      /* for TCP, link the stream to the TCP connection of the client */
+      link_transport (client, session, trans);
+    }
+  }
+}
+
+static void
+unlink_transport (GstRTSPClient * client, GstRTSPSession * session,
+    GstRTSPStreamTransport * trans)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+
+  GST_DEBUG ("client %p: unlinking transport %p", client, trans);
+
+  gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
+
+  priv->transports = g_list_remove (priv->transports, trans);
+
+  /* our session can now expire */
+  gst_rtsp_session_allow_expire (session);
+}
+
+static void
+unlink_session_transports (GstRTSPClient * client, GstRTSPSession * session,
+    GstRTSPSessionMedia * sessmedia)
+{
+  guint n_streams, i;
+
+  n_streams =
+      gst_rtsp_media_n_streams (gst_rtsp_session_media_get_media (sessmedia));
+  for (i = 0; i < n_streams; i++) {
+    GstRTSPStreamTransport *trans;
+    const GstRTSPTransport *tr;
+
+    /* get the transport, if there is no transport configured, skip this stream */
+    trans = gst_rtsp_session_media_get_transport (sessmedia, i);
+    if (trans == NULL)
+      continue;
+
+    tr = gst_rtsp_stream_transport_get_transport (trans);
+
+    if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
+      /* for TCP, unlink the stream from the TCP connection of the client */
+      unlink_transport (client, session, trans);
+    }
+  }
+}
+
+/**
+ * gst_rtsp_client_close:
+ * @client: a #GstRTSPClient
+ *
+ * Close the connection of @client and remove all media it was managing.
+ *
+ * Since: 1.4
+ */
+void
+gst_rtsp_client_close (GstRTSPClient * client)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+  const gchar *tunnelid;
+
+  GST_DEBUG ("client %p: closing connection", client);
+
+  if (priv->connection) {
+    if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
+      g_mutex_lock (&tunnels_lock);
+      /* remove from tunnelids */
+      g_hash_table_remove (tunnels, tunnelid);
+      g_mutex_unlock (&tunnels_lock);
+    }
+    gst_rtsp_connection_close (priv->connection);
+  }
+
+  /* connection is now closed, destroy the watch which will also cause the
+   * closed signal to be emitted */
+  if (priv->watch) {
+    GST_DEBUG ("client %p: destroying watch", client);
+    g_source_destroy ((GSource *) priv->watch);
+    priv->watch = NULL;
+    gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+  }
+}
+
+static gchar *
+default_make_path_from_uri (GstRTSPClient * client, const GstRTSPUrl * uri)
+{
+  gchar *path;
+
+  if (uri->query)
+    path = g_strconcat (uri->abspath, "?", uri->query, NULL);
+  else
+    path = g_strdup (uri->abspath);
+
+  return path;
+}
+
+static gboolean
+handle_teardown_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+  GstRTSPClientClass *klass;
+  GstRTSPSession *session;
+  GstRTSPSessionMedia *sessmedia;
+  GstRTSPStatusCode code;
+  gchar *path;
+  gint matched;
+  gboolean keep_session;
+
+  if (!ctx->session)
+    goto no_session;
+
+  session = ctx->session;
+
+  if (!ctx->uri)
+    goto no_uri;
+
+  klass = GST_RTSP_CLIENT_GET_CLASS (client);
+  path = klass->make_path_from_uri (client, ctx->uri);
+
+  /* get a handle to the configuration of the media in the session */
+  sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+  if (!sessmedia)
+    goto not_found;
+
+  /* only aggregate control for now.. */
+  if (path[matched] != '\0')
+    goto no_aggregate;
+
+  g_free (path);
+
+  ctx->sessmedia = sessmedia;
+
+  /* we emit the signal before closing the connection */
+  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_TEARDOWN_REQUEST],
+      0, ctx);
+
+  /* make sure we unblock the backlog and don't accept new messages
+   * on the watch */
+  if (priv->watch != NULL)
+    gst_rtsp_watch_set_flushing (priv->watch, TRUE);
+
+  /* unlink the all TCP callbacks */
+  unlink_session_transports (client, session, sessmedia);
+
+  gst_rtsp_session_media_set_state (sessmedia, GST_STATE_NULL);
+
+  /* allow messages again so that we can send the reply */
+  if (priv->watch != NULL)
+    gst_rtsp_watch_set_flushing (priv->watch, FALSE);
+
+  /* unmanage the media in the session, returns false if all media session
+   * are torn down. */
+  keep_session = gst_rtsp_session_release_media (session, sessmedia);
+
+  /* construct the response now */
+  code = GST_RTSP_STS_OK;
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
+
+  send_message (client, ctx, ctx->response, TRUE);
+
+  if (!keep_session) {
+    /* remove the session */
+    gst_rtsp_session_pool_remove (priv->session_pool, session);
+  }
+
+  return TRUE;
+
+  /* ERRORS */
+no_session:
+  {
+    GST_ERROR ("client %p: no session", client);
+    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+    return FALSE;
+  }
+no_uri:
+  {
+    GST_ERROR ("client %p: no uri supplied", client);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+    return FALSE;
+  }
+not_found:
+  {
+    GST_ERROR ("client %p: no media for uri", client);
+    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+    g_free (path);
+    return FALSE;
+  }
+no_aggregate:
+  {
+    GST_ERROR ("client %p: no aggregate path %s", client, path);
+    send_generic_response (client,
+        GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+    g_free (path);
+    return FALSE;
+  }
+}
+
+static GstRTSPResult
+default_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPResult res;
+
+  res = gst_rtsp_params_set (client, ctx);
+
+  return res;
+}
+
+static GstRTSPResult
+default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPResult res;
+
+  res = gst_rtsp_params_get (client, ctx);
+
+  return res;
+}
+
+static gboolean
+default_handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPResult res;
+  guint8 *data;
+  guint size;
+
+  res = gst_rtsp_message_get_body (ctx->request, &data, &size);
+  if (res != GST_RTSP_OK)
+    goto bad_request;
+
+  if (size == 0) {
+    /* no body, keep-alive request */
+    send_generic_response (client, GST_RTSP_STS_OK, ctx);
+  } else {
+    /* there is a body, handle the params */
+    res = GST_RTSP_CLIENT_GET_CLASS (client)->params_get (client, ctx);
+    if (res != GST_RTSP_OK)
+      goto bad_request;
+
+    send_message (client, ctx, ctx->response, FALSE);
+  }
+
+  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_GET_PARAMETER_REQUEST],
+      0, ctx);
+
+  return TRUE;
+
+  /* ERRORS */
+bad_request:
+  {
+    GST_ERROR ("client %p: bad request", client);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+    return FALSE;
+  }
+}
+
+static gboolean
+default_handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPResult res;
+  guint8 *data;
+  guint size;
+
+  res = gst_rtsp_message_get_body (ctx->request, &data, &size);
+  if (res != GST_RTSP_OK)
+    goto bad_request;
+
+  if (size == 0) {
+    /* no body, keep-alive request */
+    send_generic_response (client, GST_RTSP_STS_OK, ctx);
+  } else {
+    /* there is a body, handle the params */
+    res = GST_RTSP_CLIENT_GET_CLASS (client)->params_set (client, ctx);
+    if (res != GST_RTSP_OK)
+      goto bad_request;
+
+    send_message (client, ctx, ctx->response, FALSE);
+  }
+
+  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SET_PARAMETER_REQUEST],
+      0, ctx);
+
+  return TRUE;
+
+  /* ERRORS */
+bad_request:
+  {
+    GST_ERROR ("client %p: bad request", client);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+    return FALSE;
+  }
+}
+
+static gboolean
+handle_pause_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPSession *session;
+  GstRTSPClientClass *klass;
+  GstRTSPSessionMedia *sessmedia;
+  GstRTSPStatusCode code;
+  GstRTSPState rtspstate;
+  gchar *path;
+  gint matched;
+
+  if (!(session = ctx->session))
+    goto no_session;
+
+  if (!ctx->uri)
+    goto no_uri;
+
+  klass = GST_RTSP_CLIENT_GET_CLASS (client);
+  path = klass->make_path_from_uri (client, ctx->uri);
+
+  /* get a handle to the configuration of the media in the session */
+  sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+  if (!sessmedia)
+    goto not_found;
+
+  if (path[matched] != '\0')
+    goto no_aggregate;
+
+  g_free (path);
+
+  ctx->sessmedia = sessmedia;
+
+  rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
+  /* the session state must be playing or recording */
+  if (rtspstate != GST_RTSP_STATE_PLAYING &&
+      rtspstate != GST_RTSP_STATE_RECORDING)
+    goto invalid_state;
+
+  /* unlink the all TCP callbacks */
+  unlink_session_transports (client, session, sessmedia);
+
+  /* then pause sending */
+  gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PAUSED);
+
+  /* construct the response now */
+  code = GST_RTSP_STS_OK;
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
+
+  send_message (client, ctx, ctx->response, FALSE);
+
+  /* the state is now READY */
+  gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
+
+  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PAUSE_REQUEST], 0, ctx);
+
+  return TRUE;
+
+  /* ERRORS */
+no_session:
+  {
+    GST_ERROR ("client %p: no seesion", client);
+    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+    return FALSE;
+  }
+no_uri:
+  {
+    GST_ERROR ("client %p: no uri supplied", client);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+    return FALSE;
+  }
+not_found:
+  {
+    GST_ERROR ("client %p: no media for uri", client);
+    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+    g_free (path);
+    return FALSE;
+  }
+no_aggregate:
+  {
+    GST_ERROR ("client %p: no aggregate path %s", client, path);
+    send_generic_response (client,
+        GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+    g_free (path);
+    return FALSE;
+  }
+invalid_state:
+  {
+    GST_ERROR ("client %p: not PLAYING or RECORDING", client);
+    send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
+        ctx);
+    return FALSE;
+  }
+}
+
+/* convert @url and @path to a URL used as a content base for the factory
+ * located at @path */
+static gchar *
+make_base_url (GstRTSPClient * client, GstRTSPUrl * url, const gchar * path)
+{
+  GstRTSPUrl tmp;
+  gchar *result;
+  const gchar *trail;
+
+  /* check for trailing '/' and append one */
+  trail = (path[strlen (path) - 1] != '/' ? "/" : "");
+
+  tmp = *url;
+  tmp.user = NULL;
+  tmp.passwd = NULL;
+  tmp.abspath = g_strdup_printf ("%s%s", path, trail);
+  tmp.query = NULL;
+  result = gst_rtsp_url_get_request_uri (&tmp);
+  g_free (tmp.abspath);
+
+  return result;
+}
+
+static gboolean
+handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPSession *session;
+  GstRTSPClientClass *klass;
+  GstRTSPSessionMedia *sessmedia;
+  GstRTSPMedia *media;
+  GstRTSPStatusCode code;
+  GstRTSPUrl *uri;
+  gchar *str;
+  GstRTSPTimeRange *range;
+  GstRTSPResult res;
+  GstRTSPState rtspstate;
+  GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
+  gchar *path, *rtpinfo;
+  gint matched;
+
+  if (!(session = ctx->session))
+    goto no_session;
+
+  if (!(uri = ctx->uri))
+    goto no_uri;
+
+  klass = GST_RTSP_CLIENT_GET_CLASS (client);
+  path = klass->make_path_from_uri (client, uri);
+
+  /* get a handle to the configuration of the media in the session */
+  sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+  if (!sessmedia)
+    goto not_found;
+
+  if (path[matched] != '\0')
+    goto no_aggregate;
+
+  g_free (path);
+
+  ctx->sessmedia = sessmedia;
+  ctx->media = media = gst_rtsp_session_media_get_media (sessmedia);
+
+  /* the session state must be playing or ready */
+  rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
+  if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
+    goto invalid_state;
+
+  /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
+  if (!gst_rtsp_media_unsuspend (media))
+    goto unsuspend_failed;
+
+  /* parse the range header if we have one */
+  res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
+  if (res == GST_RTSP_OK) {
+    if (gst_rtsp_range_parse (str, &range) == GST_RTSP_OK) {
+      /* we have a range, seek to the position */
+      unit = range->unit;
+      gst_rtsp_media_seek (media, range);
+      gst_rtsp_range_free (range);
+    }
+  }
+
+  /* link the all TCP callbacks */
+  link_session_transports (client, session, sessmedia);
+
+  /* grab RTPInfo from the media now */
+  rtpinfo = gst_rtsp_session_media_get_rtpinfo (sessmedia);
+
+  /* construct the response now */
+  code = GST_RTSP_STS_OK;
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
+
+  /* add the RTP-Info header */
+  if (rtpinfo)
+    gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RTP_INFO,
+        rtpinfo);
+
+  /* add the range */
+  str = gst_rtsp_media_get_range_string (media, TRUE, unit);
+  if (str)
+    gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_RANGE, str);
+
+  send_message (client, ctx, ctx->response, FALSE);
+
+  /* start playing after sending the response */
+  gst_rtsp_session_media_set_state (sessmedia, GST_STATE_PLAYING);
+
+  gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_PLAYING);
+
+  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_PLAY_REQUEST], 0, ctx);
+
+  return TRUE;
+
+  /* ERRORS */
+no_session:
+  {
+    GST_ERROR ("client %p: no session", client);
+    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+    return FALSE;
+  }
+no_uri:
+  {
+    GST_ERROR ("client %p: no uri supplied", client);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+    return FALSE;
+  }
+not_found:
+  {
+    GST_ERROR ("client %p: media not found", client);
+    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+    return FALSE;
+  }
+no_aggregate:
+  {
+    GST_ERROR ("client %p: no aggregate path %s", client, path);
+    send_generic_response (client,
+        GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED, ctx);
+    g_free (path);
+    return FALSE;
+  }
+invalid_state:
+  {
+    GST_ERROR ("client %p: not PLAYING or READY", client);
+    send_generic_response (client, GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE,
+        ctx);
+    return FALSE;
+  }
+unsuspend_failed:
+  {
+    GST_ERROR ("client %p: unsuspend failed", client);
+    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+    return FALSE;
+  }
+}
+
+static void
+do_keepalive (GstRTSPSession * session)
+{
+  GST_INFO ("keep session %p alive", session);
+  gst_rtsp_session_touch (session);
+}
+
+/* parse @transport and return a valid transport in @tr. only transports
+ * supported by @stream are returned. Returns FALSE if no valid transport
+ * was found. */
+static gboolean
+parse_transport (const char *transport, GstRTSPStream * stream,
+    GstRTSPTransport * tr)
+{
+  gint i;
+  gboolean res;
+  gchar **transports;
+
+  res = FALSE;
+  gst_rtsp_transport_init (tr);
+
+  GST_DEBUG ("parsing transports %s", transport);
+
+  transports = g_strsplit (transport, ",", 0);
+
+  /* loop through the transports, try to parse */
+  for (i = 0; transports[i]; i++) {
+    res = gst_rtsp_transport_parse (transports[i], tr);
+    if (res != GST_RTSP_OK) {
+      /* no valid transport, search some more */
+      GST_WARNING ("could not parse transport %s", transports[i]);
+      goto next;
+    }
+
+    /* we have a transport, see if it's supported */
+    if (!gst_rtsp_stream_is_transport_supported (stream, tr)) {
+      GST_WARNING ("unsupported transport %s", transports[i]);
+      goto next;
+    }
+
+    /* we have a valid transport */
+    GST_INFO ("found valid transport %s", transports[i]);
+    res = TRUE;
+    break;
+
+  next:
+    gst_rtsp_transport_init (tr);
+  }
+  g_strfreev (transports);
+
+  return res;
+}
+
+static gboolean
+default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
+    GstRTSPStream * stream, GstRTSPContext * ctx)
+{
+  GstRTSPMessage *request = ctx->request;
+  gchar *blocksize_str;
+
+  if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
+          &blocksize_str, 0) == GST_RTSP_OK) {
+    guint64 blocksize;
+    gchar *end;
+
+    blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
+    if (end == blocksize_str)
+      goto parse_failed;
+
+    /* we don't want to change the mtu when this media
+     * can be shared because it impacts other clients */
+    if (gst_rtsp_media_is_shared (media))
+      goto done;
+
+    if (blocksize > G_MAXUINT)
+      blocksize = G_MAXUINT;
+
+    gst_rtsp_stream_set_mtu (stream, blocksize);
+  }
+done:
+  return TRUE;
+
+  /* ERRORS */
+parse_failed:
+  {
+    GST_ERROR_OBJECT (client, "failed to parse blocksize");
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+    return FALSE;
+  }
+}
+
+static gboolean
+default_configure_client_transport (GstRTSPClient * client,
+    GstRTSPContext * ctx, GstRTSPTransport * ct)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+
+  /* we have a valid transport now, set the destination of the client. */
+  if (ct->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
+    gboolean use_client_settings;
+
+    use_client_settings =
+        gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS);
+
+    if (ct->destination && use_client_settings) {
+      GstRTSPAddress *addr;
+
+      addr = gst_rtsp_stream_reserve_address (ctx->stream, ct->destination,
+          ct->port.min, ct->port.max - ct->port.min + 1, ct->ttl);
+
+      if (addr == NULL)
+        goto no_address;
+
+      gst_rtsp_address_free (addr);
+    } else {
+      GstRTSPAddress *addr;
+      GSocketFamily family;
+
+      family = priv->is_ipv6 ? G_SOCKET_FAMILY_IPV6 : G_SOCKET_FAMILY_IPV4;
+
+      addr = gst_rtsp_stream_get_multicast_address (ctx->stream, family);
+      if (addr == NULL)
+        goto no_address;
+
+      g_free (ct->destination);
+      ct->destination = g_strdup (addr->address);
+      ct->port.min = addr->port;
+      ct->port.max = addr->port + addr->n_ports - 1;
+      ct->ttl = addr->ttl;
+
+      gst_rtsp_address_free (addr);
+    }
+  } else {
+    GstRTSPUrl *url;
+
+    url = gst_rtsp_connection_get_url (priv->connection);
+    g_free (ct->destination);
+    ct->destination = g_strdup (url->host);
+
+    if (ct->lower_transport & GST_RTSP_LOWER_TRANS_TCP) {
+      GSocket *sock;
+      GSocketAddress *addr;
+
+      sock = gst_rtsp_connection_get_read_socket (priv->connection);
+      if ((addr = g_socket_get_remote_address (sock, NULL))) {
+        /* our read port is the sender port of client */
+        ct->client_port.min =
+            g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
+        g_object_unref (addr);
+      }
+      if ((addr = g_socket_get_local_address (sock, NULL))) {
+        ct->server_port.max =
+            g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
+        g_object_unref (addr);
+      }
+      sock = gst_rtsp_connection_get_write_socket (priv->connection);
+      if ((addr = g_socket_get_remote_address (sock, NULL))) {
+        /* our write port is the receiver port of client */
+        ct->client_port.max =
+            g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
+        g_object_unref (addr);
+      }
+      if ((addr = g_socket_get_local_address (sock, NULL))) {
+        ct->server_port.min =
+            g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (addr));
+        g_object_unref (addr);
+      }
+      /* check if the client selected channels for TCP */
+      if (ct->interleaved.min == -1 || ct->interleaved.max == -1) {
+        gst_rtsp_session_media_alloc_channels (ctx->sessmedia,
+            &ct->interleaved);
+      }
+    }
+  }
+  return TRUE;
+
+  /* ERRORS */
+no_address:
+  {
+    GST_ERROR_OBJECT (client, "failed to acquire address for stream");
+    return FALSE;
+  }
+}
+
+static GstRTSPTransport *
+make_server_transport (GstRTSPClient * client, GstRTSPContext * ctx,
+    GstRTSPTransport * ct)
+{
+  GstRTSPTransport *st;
+  GInetAddress *addr;
+  GSocketFamily family;
+
+  /* prepare the server transport */
+  gst_rtsp_transport_new (&st);
+
+  st->trans = ct->trans;
+  st->profile = ct->profile;
+  st->lower_transport = ct->lower_transport;
+
+  addr = g_inet_address_new_from_string (ct->destination);
+
+  if (!addr) {
+    GST_ERROR ("failed to get inet addr from client destination");
+    family = G_SOCKET_FAMILY_IPV4;
+  } else {
+    family = g_inet_address_get_family (addr);
+    g_object_unref (addr);
+    addr = NULL;
+  }
+
+  switch (st->lower_transport) {
+    case GST_RTSP_LOWER_TRANS_UDP:
+      st->client_port = ct->client_port;
+      gst_rtsp_stream_get_server_port (ctx->stream, &st->server_port, family);
+      break;
+    case GST_RTSP_LOWER_TRANS_UDP_MCAST:
+      st->port = ct->port;
+      st->destination = g_strdup (ct->destination);
+      st->ttl = ct->ttl;
+      break;
+    case GST_RTSP_LOWER_TRANS_TCP:
+      st->interleaved = ct->interleaved;
+      st->client_port = ct->client_port;
+      st->server_port = ct->server_port;
+    default:
+      break;
+  }
+
+  gst_rtsp_stream_get_ssrc (ctx->stream, &st->ssrc);
+
+  return st;
+}
+
+#define AES_128_KEY_LEN 16
+#define AES_256_KEY_LEN 32
+
+#define HMAC_32_KEY_LEN 4
+#define HMAC_80_KEY_LEN 10
+
+static gboolean
+mikey_apply_policy (GstCaps * caps, GstMIKEYMessage * msg, guint8 policy)
+{
+  const gchar *srtp_cipher;
+  const gchar *srtp_auth;
+  const GstMIKEYPayload *sp;
+  guint i;
+
+  /* loop over Security policy until we find one containing policy */
+  for (i = 0;; i++) {
+    if ((sp = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, i)) == NULL)
+      break;
+
+    if (((GstMIKEYPayloadSP *) sp)->policy == policy)
+      break;
+  }
+
+  /* the default ciphers */
+  srtp_cipher = "aes-128-icm";
+  srtp_auth = "hmac-sha1-80";
+
+  /* now override the defaults with what is in the Security Policy */
+  if (sp != NULL) {
+    guint len;
+
+    /* collect all the params and go over them */
+    len = gst_mikey_payload_sp_get_n_params (sp);
+    for (i = 0; i < len; i++) {
+      const GstMIKEYPayloadSPParam *param =
+          gst_mikey_payload_sp_get_param (sp, i);
+
+      switch (param->type) {
+        case GST_MIKEY_SP_SRTP_ENC_ALG:
+          switch (param->val[0]) {
+            case 0:
+              srtp_cipher = "null";
+              break;
+            case 2:
+            case 1:
+              srtp_cipher = "aes-128-icm";
+              break;
+            default:
+              break;
+          }
+          break;
+        case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
+          switch (param->val[0]) {
+            case AES_128_KEY_LEN:
+              srtp_cipher = "aes-128-icm";
+              break;
+            case AES_256_KEY_LEN:
+              srtp_cipher = "aes-256-icm";
+              break;
+            default:
+              break;
+          }
+          break;
+        case GST_MIKEY_SP_SRTP_AUTH_ALG:
+          switch (param->val[0]) {
+            case 0:
+              srtp_auth = "null";
+              break;
+            case 2:
+            case 1:
+              srtp_auth = "hmac-sha1-80";
+              break;
+            default:
+              break;
+          }
+          break;
+        case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
+          switch (param->val[0]) {
+            case HMAC_32_KEY_LEN:
+              srtp_auth = "hmac-sha1-32";
+              break;
+            case HMAC_80_KEY_LEN:
+              srtp_auth = "hmac-sha1-80";
+              break;
+            default:
+              break;
+          }
+          break;
+        case GST_MIKEY_SP_SRTP_SRTP_ENC:
+          break;
+        case GST_MIKEY_SP_SRTP_SRTCP_ENC:
+          break;
+        default:
+          break;
+      }
+    }
+  }
+  /* now configure the SRTP parameters */
+  gst_caps_set_simple (caps,
+      "srtp-cipher", G_TYPE_STRING, srtp_cipher,
+      "srtp-auth", G_TYPE_STRING, srtp_auth,
+      "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
+      "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
+
+  return TRUE;
+}
+
+static gboolean
+handle_mikey_data (GstRTSPClient * client, GstRTSPContext * ctx,
+    guint8 * data, gsize size)
+{
+  GstMIKEYMessage *msg;
+  guint i, n_cs;
+  GstCaps *caps = NULL;
+  GstMIKEYPayloadKEMAC *kemac;
+  const GstMIKEYPayloadKeyData *pkd;
+  GstBuffer *key;
+
+  /* the MIKEY message contains a CSB or crypto session bundle. It is a
+   * set of Crypto Sessions protected with the same master key.
+   * In the context of SRTP, an RTP and its RTCP stream is part of a
+   * crypto session */
+  if ((msg = gst_mikey_message_new_from_data (data, size, NULL, NULL)) == NULL)
+    goto parse_failed;
+
+  /* we can only handle SRTP crypto sessions for now */
+  if (msg->map_type != GST_MIKEY_MAP_TYPE_SRTP)
+    goto invalid_map_type;
+
+  /* get the number of crypto sessions. This maps SSRC to its
+   * security parameters */
+  n_cs = gst_mikey_message_get_n_cs (msg);
+  if (n_cs == 0)
+    goto no_crypto_sessions;
+
+  /* we also need keys */
+  if (!(kemac = (GstMIKEYPayloadKEMAC *) gst_mikey_message_find_payload
+          (msg, GST_MIKEY_PT_KEMAC, 0)))
+    goto no_keys;
+
+  /* we don't support encrypted keys */
+  if (kemac->enc_alg != GST_MIKEY_ENC_NULL
+      || kemac->mac_alg != GST_MIKEY_MAC_NULL)
+    goto unsupported_encryption;
+
+  /* get Key data sub-payload */
+  pkd = (const GstMIKEYPayloadKeyData *)
+      gst_mikey_payload_kemac_get_sub (&kemac->pt, 0);
+
+  key =
+      gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
+      pkd->key_len);
+
+  /* go over all crypto sessions and create the security policy for each
+   * SSRC */
+  for (i = 0; i < n_cs; i++) {
+    const GstMIKEYMapSRTP *map = gst_mikey_message_get_cs_srtp (msg, i);
+
+    caps = gst_caps_new_simple ("application/x-srtp",
+        "ssrc", G_TYPE_UINT, map->ssrc,
+        "roc", G_TYPE_UINT, map->roc, "srtp-key", GST_TYPE_BUFFER, key, NULL);
+    mikey_apply_policy (caps, msg, map->policy);
+
+    gst_rtsp_stream_update_crypto (ctx->stream, map->ssrc, caps);
+    gst_caps_unref (caps);
+  }
+  gst_mikey_message_unref (msg);
+  gst_buffer_unref (key);
+
+  return TRUE;
+
+  /* ERRORS */
+parse_failed:
+  {
+    GST_DEBUG_OBJECT (client, "failed to parse MIKEY message");
+    return FALSE;
+  }
+invalid_map_type:
+  {
+    GST_DEBUG_OBJECT (client, "invalid map type %d", msg->map_type);
+    goto cleanup_message;
+  }
+no_crypto_sessions:
+  {
+    GST_DEBUG_OBJECT (client, "no crypto sessions");
+    goto cleanup_message;
+  }
+no_keys:
+  {
+    GST_DEBUG_OBJECT (client, "no keys found");
+    goto cleanup_message;
+  }
+unsupported_encryption:
+  {
+    GST_DEBUG_OBJECT (client, "unsupported key encryption");
+    goto cleanup_message;
+  }
+cleanup_message:
+  {
+    gst_mikey_message_unref (msg);
+    return FALSE;
+  }
+}
+
+#define IS_STRIP_CHAR(c) (g_ascii_isspace ((guchar)(c)) || ((c) == '\"'))
+
+static void
+strip_chars (gchar * str)
+{
+  gchar *s;
+  gsize len;
+
+  len = strlen (str);
+  while (len--) {
+    if (!IS_STRIP_CHAR (str[len]))
+      break;
+    str[len] = '\0';
+  }
+  for (s = str; *s && IS_STRIP_CHAR (*s); s++);
+  memmove (str, s, len + 1);
+}
+
+/* KeyMgmt = "KeyMgmt" ":" key-mgmt-spec 0*("," key-mgmt-spec)
+ * key-mgmt-spec = "prot" "=" KMPID ";" ["uri" "=" %x22 URI %x22 ";"]
+ */
+static gboolean
+handle_keymgmt (GstRTSPClient * client, GstRTSPContext * ctx, gchar * keymgmt)
+{
+  gchar **specs;
+  gint i, j;
+
+  specs = g_strsplit (keymgmt, ",", 0);
+  for (i = 0; specs[i]; i++) {
+    gchar **split;
+
+    split = g_strsplit (specs[i], ";", 0);
+    for (j = 0; split[j]; j++) {
+      g_strstrip (split[j]);
+      if (g_str_has_prefix (split[j], "prot=")) {
+        g_strstrip (split[j] + 5);
+        if (!g_str_equal (split[j] + 5, "mikey"))
+          break;
+        GST_DEBUG ("found mikey");
+      } else if (g_str_has_prefix (split[j], "uri=")) {
+        strip_chars (split[j] + 4);
+        GST_DEBUG ("found uri '%s'", split[j] + 4);
+      } else if (g_str_has_prefix (split[j], "data=")) {
+        guchar *data;
+        gsize size;
+        strip_chars (split[j] + 5);
+        GST_DEBUG ("found data '%s'", split[j] + 5);
+        data = g_base64_decode_inplace (split[j] + 5, &size);
+        handle_mikey_data (client, ctx, data, size);
+      }
+    }
+    g_strfreev (split);
+  }
+  g_strfreev (specs);
+  return TRUE;
+}
+
+static gboolean
+handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+  GstRTSPResult res;
+  GstRTSPUrl *uri;
+  gchar *transport, *keymgmt;
+  GstRTSPTransport *ct, *st;
+  GstRTSPStatusCode code;
+  GstRTSPSession *session;
+  GstRTSPStreamTransport *trans;
+  gchar *trans_str;
+  GstRTSPSessionMedia *sessmedia;
+  GstRTSPMedia *media;
+  GstRTSPStream *stream;
+  GstRTSPState rtspstate;
+  GstRTSPClientClass *klass;
+  gchar *path, *control;
+  gint matched;
+  gboolean new_session = FALSE;
+
+  if (!ctx->uri)
+    goto no_uri;
+
+  uri = ctx->uri;
+  klass = GST_RTSP_CLIENT_GET_CLASS (client);
+  path = klass->make_path_from_uri (client, uri);
+
+  /* parse the transport */
+  res =
+      gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_TRANSPORT,
+      &transport, 0);
+  if (res != GST_RTSP_OK)
+    goto no_transport;
+
+  /* we create the session after parsing stuff so that we don't make
+   * a session for malformed requests */
+  if (priv->session_pool == NULL)
+    goto no_pool;
+
+  session = ctx->session;
+
+  if (session) {
+    g_object_ref (session);
+    /* get a handle to the configuration of the media in the session, this can
+     * return NULL if this is a new url to manage in this session. */
+    sessmedia = gst_rtsp_session_get_media (session, path, &matched);
+  } else {
+    /* we need a new media configuration in this session */
+    sessmedia = NULL;
+  }
+
+  /* we have no session media, find one and manage it */
+  if (sessmedia == NULL) {
+    /* get a handle to the configuration of the media in the session */
+    media = find_media (client, ctx, path, &matched);
+  } else {
+    if ((media = gst_rtsp_session_media_get_media (sessmedia)))
+      g_object_ref (media);
+    else
+      goto media_not_found;
+  }
+  /* no media, not found then */
+  if (media == NULL)
+    goto media_not_found_no_reply;
+
+  /* FIXME-WFD : wfd url problem */
+#if 0
+  if (path[matched] == '\0')
+    goto control_not_found;
+
+  /* path is what matched. */
+  path[matched] = '\0';
+  /* control is remainder */
+  control = &path[matched + 1];
+#else
+  control = g_strdup ("stream=0");
+#endif
+
+  /* find the stream now using the control part */
+  stream = gst_rtsp_media_find_stream (media, control);
+  if (stream == NULL)
+    goto stream_not_found;
+
+  /* now we have a uri identifying a valid media and stream */
+  ctx->stream = stream;
+  ctx->media = media;
+
+  if (session == NULL) {
+    /* create a session if this fails we probably reached our session limit or
+     * something. */
+    if (!(session = gst_rtsp_session_pool_create (priv->session_pool)))
+      goto service_unavailable;
+
+    /* make sure this client is closed when the session is closed */
+    client_watch_session (client, session);
+
+    new_session = TRUE;
+    /* signal new session */
+    g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_NEW_SESSION], 0,
+        session);
+
+    ctx->session = session;
+  }
+
+  if (!klass->configure_client_media (client, media, stream, ctx))
+    goto configure_media_failed_no_reply;
+
+  gst_rtsp_transport_new (&ct);
+
+  /* parse and find a usable supported transport */
+  if (!parse_transport (transport, stream, ct))
+    goto unsupported_transports;
+
+  /* update the client transport */
+  if (!klass->configure_client_transport (client, ctx, ct))
+    goto unsupported_client_transport;
+
+  /* parse the keymgmt */
+  if (gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_KEYMGMT,
+          &keymgmt, 0) == GST_RTSP_OK) {
+    if (!handle_keymgmt (client, ctx, keymgmt))
+      goto keymgmt_error;
+  }
+
+  if (sessmedia == NULL) {
+    /* manage the media in our session now, if not done already  */
+    sessmedia = gst_rtsp_session_manage_media (session, path, media);
+    /* if we stil have no media, error */
+    if (sessmedia == NULL)
+      goto sessmedia_unavailable;
+  } else {
+    g_object_unref (media);
+  }
+
+  ctx->sessmedia = sessmedia;
+
+  /* set in the session media transport */
+  trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
+
+  /* configure the url used to set this transport, this we will use when
+   * generating the response for the PLAY request */
+  gst_rtsp_stream_transport_set_url (trans, uri);
+
+  /* configure keepalive for this transport */
+  gst_rtsp_stream_transport_set_keepalive (trans,
+      (GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
+
+  /* create and serialize the server transport */
+  st = make_server_transport (client, ctx, ct);
+  trans_str = gst_rtsp_transport_as_text (st);
+
+  /* FIXME-WFD : Temporarily force to set profile string */
+  trans_str = g_strjoinv ("RTP/AVP/UDP", g_strsplit (trans_str, "RTP/AVP", -1));
+
+  gst_rtsp_transport_free (st);
+
+  /* construct the response now */
+  code = GST_RTSP_STS_OK;
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
+
+  gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_TRANSPORT,
+      trans_str);
+  g_free (trans_str);
+
+  send_message (client, ctx, ctx->response, FALSE);
+
+  /* update the state */
+  rtspstate = gst_rtsp_session_media_get_rtsp_state (sessmedia);
+  switch (rtspstate) {
+    case GST_RTSP_STATE_PLAYING:
+    case GST_RTSP_STATE_RECORDING:
+    case GST_RTSP_STATE_READY:
+      /* no state change */
+      break;
+    default:
+      gst_rtsp_session_media_set_rtsp_state (sessmedia, GST_RTSP_STATE_READY);
+      break;
+  }
+  g_object_unref (session);
+  g_free (path);
+
+  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_SETUP_REQUEST], 0, ctx);
+
+  return TRUE;
+
+  /* ERRORS */
+no_uri:
+  {
+    GST_ERROR ("client %p: no uri", client);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+    return FALSE;
+  }
+no_transport:
+  {
+    GST_ERROR ("client %p: no transport", client);
+    send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
+    goto cleanup_path;
+  }
+no_pool:
+  {
+    GST_ERROR ("client %p: no session pool configured", client);
+    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+    goto cleanup_path;
+  }
+media_not_found_no_reply:
+  {
+    GST_ERROR ("client %p: media '%s' not found", client, path);
+    /* error reply is already sent */
+    goto cleanup_path;
+  }
+media_not_found:
+  {
+    GST_ERROR ("client %p: media '%s' not found", client, path);
+    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+    goto cleanup_path;
+  }
+#if 0
+control_not_found:
+  {
+    GST_ERROR ("client %p: no control in path '%s'", client, path);
+    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+    g_object_unref (media);
+    goto cleanup_path;
+  }
+#endif
+stream_not_found:
+  {
+    GST_ERROR ("client %p: stream '%s' not found", client, control);
+    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+    g_object_unref (media);
+    goto cleanup_path;
+  }
+service_unavailable:
+  {
+    GST_ERROR ("client %p: can't create session", client);
+    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+    g_object_unref (media);
+    goto cleanup_path;
+  }
+sessmedia_unavailable:
+  {
+    GST_ERROR ("client %p: can't create session media", client);
+    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+    g_object_unref (media);
+    goto cleanup_session;
+  }
+configure_media_failed_no_reply:
+  {
+    GST_ERROR ("client %p: configure_media failed", client);
+    /* error reply is already sent */
+    goto cleanup_session;
+  }
+unsupported_transports:
+  {
+    GST_ERROR ("client %p: unsupported transports", client);
+    send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
+    goto cleanup_transport;
+  }
+unsupported_client_transport:
+  {
+    GST_ERROR ("client %p: unsupported client transport", client);
+    send_generic_response (client, GST_RTSP_STS_UNSUPPORTED_TRANSPORT, ctx);
+    goto cleanup_transport;
+  }
+keymgmt_error:
+  {
+    GST_ERROR ("client %p: keymgmt error", client);
+    send_generic_response (client, GST_RTSP_STS_KEY_MANAGEMENT_FAILURE, ctx);
+    goto cleanup_transport;
+  }
+  {
+  cleanup_transport:
+    gst_rtsp_transport_free (ct);
+  cleanup_session:
+    if (new_session)
+      gst_rtsp_session_pool_remove (priv->session_pool, session);
+    g_object_unref (session);
+  cleanup_path:
+    g_free (path);
+    return FALSE;
+  }
+}
+
+static GstSDPMessage *
+create_sdp (GstRTSPClient * client, GstRTSPMedia * media)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+  GstSDPMessage *sdp;
+  GstSDPInfo info;
+  const gchar *proto;
+
+  gst_sdp_message_new (&sdp);
+
+  /* some standard things first */
+  gst_sdp_message_set_version (sdp, "0");
+
+  if (priv->is_ipv6)
+    proto = "IP6";
+  else
+    proto = "IP4";
+
+  gst_sdp_message_set_origin (sdp, "-", "1188340656180883", "1", "IN", proto,
+      priv->server_ip);
+
+  gst_sdp_message_set_session_name (sdp, "Session streamed with GStreamer");
+  gst_sdp_message_set_information (sdp, "rtsp-server");
+  gst_sdp_message_add_time (sdp, "0", "0", NULL);
+  gst_sdp_message_add_attribute (sdp, "tool", "GStreamer");
+  gst_sdp_message_add_attribute (sdp, "type", "broadcast");
+  gst_sdp_message_add_attribute (sdp, "control", "*");
+
+  info.is_ipv6 = priv->is_ipv6;
+  info.server_ip = priv->server_ip;
+
+  /* create an SDP for the media object */
+  if (!gst_rtsp_media_setup_sdp (media, sdp, &info))
+    goto no_sdp;
+
+  return sdp;
+
+  /* ERRORS */
+no_sdp:
+  {
+    GST_ERROR ("client %p: could not create SDP", client);
+    gst_sdp_message_free (sdp);
+    return NULL;
+  }
+}
+
+/* for the describe we must generate an SDP */
+static gboolean
+handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+  GstRTSPResult res;
+  GstSDPMessage *sdp;
+  guint i;
+  gchar *path, *str;
+  GstRTSPMedia *media;
+  GstRTSPClientClass *klass;
+
+  klass = GST_RTSP_CLIENT_GET_CLASS (client);
+
+  if (!ctx->uri)
+    goto no_uri;
+
+  /* check what kind of format is accepted, we don't really do anything with it
+   * and always return SDP for now. */
+  for (i = 0;; i++) {
+    gchar *accept;
+
+    res =
+        gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_ACCEPT,
+        &accept, i);
+    if (res == GST_RTSP_ENOTIMPL)
+      break;
+
+    if (g_ascii_strcasecmp (accept, "application/sdp") == 0)
+      break;
+  }
+
+  if (!priv->mount_points)
+    goto no_mount_points;
+
+  if (!(path = gst_rtsp_mount_points_make_path (priv->mount_points, ctx->uri)))
+    goto no_path;
+
+  /* find the media object for the uri */
+  if (!(media = find_media (client, ctx, path, NULL)))
+    goto no_media;
+
+  /* create an SDP for the media object on this client */
+  if (!(sdp = klass->create_sdp (client, media)))
+    goto no_sdp;
+
+  /* we suspend after the describe */
+  gst_rtsp_media_suspend (media);
+  g_object_unref (media);
+
+  gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+      gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
+
+  gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_TYPE,
+      "application/sdp");
+
+  /* content base for some clients that might screw up creating the setup uri */
+  str = make_base_url (client, ctx->uri, path);
+  g_free (path);
+
+  GST_INFO ("adding content-base: %s", str);
+  gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
+
+  /* add SDP to the response body */
+  str = gst_sdp_message_as_text (sdp);
+  gst_rtsp_message_take_body (ctx->response, (guint8 *) str, strlen (str));
+  gst_sdp_message_free (sdp);
+
+  send_message (client, ctx, ctx->response, FALSE);
+
+  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_DESCRIBE_REQUEST],
+      0, ctx);
+
+  return TRUE;
+
+  /* ERRORS */
+no_uri:
+  {
+    GST_ERROR ("client %p: no uri", client);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+    return FALSE;
+  }
+no_mount_points:
+  {
+    GST_ERROR ("client %p: no mount points configured", client);
+    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+    return FALSE;
+  }
+no_path:
+  {
+    GST_ERROR ("client %p: can't find path for url", client);
+    send_generic_response (client, GST_RTSP_STS_NOT_FOUND, ctx);
+    return FALSE;
+  }
+no_media:
+  {
+    GST_ERROR ("client %p: no media", client);
+    g_free (path);
+    /* error reply is already sent */
+    return FALSE;
+  }
+no_sdp:
+  {
+    GST_ERROR ("client %p: can't create SDP", client);
+    send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+    g_free (path);
+    g_object_unref (media);
+    return FALSE;
+  }
+}
+
+static gboolean
+default_handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPMethod options;
+  gchar *str;
+
+  options = GST_RTSP_DESCRIBE |
+      GST_RTSP_OPTIONS |
+      GST_RTSP_PAUSE |
+      GST_RTSP_PLAY |
+      GST_RTSP_SETUP |
+      GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN;
+
+  str = gst_rtsp_options_as_text (options);
+
+  gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
+      gst_rtsp_status_as_text (GST_RTSP_STS_OK), ctx->request);
+
+  gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
+  g_free (str);
+
+  send_message (client, ctx, ctx->response, FALSE);
+
+  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_OPTIONS_REQUEST],
+      0, ctx);
+
+  return TRUE;
+}
+
+/* remove duplicate and trailing '/' */
+static void
+sanitize_uri (GstRTSPUrl * uri)
+{
+  gint i, len;
+  gchar *s, *d;
+  gboolean have_slash, prev_slash;
+
+  s = d = uri->abspath;
+  len = strlen (uri->abspath);
+
+  prev_slash = FALSE;
+
+  for (i = 0; i < len; i++) {
+    have_slash = s[i] == '/';
+    *d = s[i];
+    if (!have_slash || !prev_slash)
+      d++;
+    prev_slash = have_slash;
+  }
+  len = d - uri->abspath;
+  /* don't remove the first slash if that's the only thing left */
+  if (len > 1 && *(d - 1) == '/')
+    d--;
+  *d = '\0';
+}
+
+/* is called when the session is removed from its session pool. */
+static void
+client_session_removed (GstRTSPSessionPool * pool, GstRTSPSession * session,
+    GstRTSPClient * client)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+
+  GST_INFO ("client %p: session %p removed", client, session);
+
+  g_mutex_lock (&priv->lock);
+  client_unwatch_session (client, session, NULL);
+  g_mutex_unlock (&priv->lock);
+}
+
+/* Returns TRUE if there are no Require headers, otherwise returns FALSE
+ * and also returns a newly-allocated string of (comma-separated) unsupported
+ * options in the unsupported_reqs variable .
+ *
+ * There may be multiple Require headers, but we must send one single
+ * Unsupported header with all the unsupported options as response. If
+ * an incoming Require header contained a comma-separated list of options
+ * GstRtspConnection will already have split that list up into multiple
+ * headers.
+ *
+ * TODO: allow the application to decide what features are supported
+ */
+static gboolean
+check_request_requirements (GstRTSPMessage * msg, gchar ** unsupported_reqs)
+{
+  GstRTSPResult res;
+  GPtrArray *arr = NULL;
+  gchar *reqs = NULL;
+  gint i;
+
+  i = 0;
+  do {
+    res = gst_rtsp_message_get_header (msg, GST_RTSP_HDR_REQUIRE, &reqs, i++);
+
+    if (res == GST_RTSP_ENOTIMPL)
+      break;
+
+    if (arr == NULL)
+      arr = g_ptr_array_new_with_free_func ((GDestroyNotify) g_free);
+
+    g_ptr_array_add (arr, g_strdup (reqs));
+  }
+  while (TRUE);
+
+  /* if we don't have any Require headers at all, all is fine */
+  if (i == 1)
+    return TRUE;
+
+  /* otherwise we've now processed at all the Require headers */
+  g_ptr_array_add (arr, NULL);
+
+  /* for now we don't commit to supporting anything, so will just report
+   * all of the required options as unsupported */
+  *unsupported_reqs = g_strjoinv (", ", (gchar **) arr->pdata);
+
+  g_ptr_array_unref (arr);
+  return FALSE;
+}
+
+static void
+handle_request (GstRTSPClient * client, GstRTSPMessage * request)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+  GstRTSPMethod method;
+  const gchar *uristr;
+  GstRTSPUrl *uri = NULL;
+  GstRTSPVersion version;
+  GstRTSPResult res;
+  GstRTSPSession *session = NULL;
+  GstRTSPContext sctx = { NULL }, *ctx;
+  GstRTSPMessage response = { 0 };
+  gchar *unsupported_reqs = NULL;
+  gchar *sessid;
+  GstRTSPClientClass *klass;
+
+  klass = GST_RTSP_CLIENT_GET_CLASS (client);
+
+  if (!(ctx = gst_rtsp_context_get_current ())) {
+    ctx = &sctx;
+    ctx->auth = priv->auth;
+    gst_rtsp_context_push_current (ctx);
+  }
+
+  ctx->conn = priv->connection;
+  ctx->client = client;
+  ctx->request = request;
+  ctx->response = &response;
+
+  if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
+    gst_rtsp_message_dump (request);
+  }
+
+  gst_rtsp_message_parse_request (request, &method, &uristr, &version);
+
+  GST_INFO ("client %p: received a request %s %s %s", client,
+      gst_rtsp_method_as_text (method), uristr,
+      gst_rtsp_version_as_text (version));
+
+  /* we can only handle 1.0 requests */
+  if (version != GST_RTSP_VERSION_1_0)
+    goto not_supported;
+
+  ctx->method = method;
+
+  /* we always try to parse the url first */
+  if (strcmp (uristr, "*") == 0) {
+    /* special case where we have * as uri, keep uri = NULL */
+  } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
+    /* check if the uristr is an absolute path <=> scheme and host information
+     * is missing */
+    gchar *scheme;
+
+    scheme = g_uri_parse_scheme (uristr);
+    if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
+      gchar *absolute_uristr = NULL;
+
+      GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
+      if (priv->server_ip == NULL) {
+        GST_WARNING_OBJECT (client, "host information missing");
+        goto bad_request;
+      }
+
+      absolute_uristr =
+          g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
+
+      GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
+      if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
+        g_free (absolute_uristr);
+        goto bad_request;
+      }
+      g_free (absolute_uristr);
+    } else {
+      g_free (scheme);
+      goto bad_request;
+    }
+  }
+
+  /* get the session if there is any */
+  res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
+  if (res == GST_RTSP_OK) {
+    if (priv->session_pool == NULL)
+      goto no_pool;
+
+    /* we had a session in the request, find it again */
+    if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
+      goto session_not_found;
+
+    /* we add the session to the client list of watched sessions. When a session
+     * disappears because it times out, we will be notified. If all sessions are
+     * gone, we will close the connection */
+    client_watch_session (client, session);
+  }
+
+  /* sanitize the uri */
+  if (uri)
+    sanitize_uri (uri);
+  ctx->uri = uri;
+  ctx->session = session;
+
+  if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_URL))
+    goto not_authorized;
+
+#if 0
+  /* FIXME-WFD : How does it handle this */
+  /* handle any 'Require' headers */
+  if (!check_request_requirements (ctx->request, &unsupported_reqs))
+    goto unsupported_requirement;
+#endif
+
+  /* now see what is asked and dispatch to a dedicated handler */
+  switch (method) {
+    case GST_RTSP_OPTIONS:
+      klass->handle_options_request (client, ctx);
+      break;
+    case GST_RTSP_DESCRIBE:
+      handle_describe_request (client, ctx);
+      break;
+    case GST_RTSP_SETUP:
+      handle_setup_request (client, ctx);
+      break;
+    case GST_RTSP_PLAY:
+      handle_play_request (client, ctx);
+      break;
+    case GST_RTSP_PAUSE:
+      handle_pause_request (client, ctx);
+      break;
+    case GST_RTSP_TEARDOWN:
+      handle_teardown_request (client, ctx);
+      break;
+    case GST_RTSP_SET_PARAMETER:
+      klass->handle_set_param_request (client, ctx);
+      break;
+    case GST_RTSP_GET_PARAMETER:
+      klass->handle_get_param_request (client, ctx);
+      break;
+    case GST_RTSP_ANNOUNCE:
+    case GST_RTSP_RECORD:
+    case GST_RTSP_REDIRECT:
+      goto not_implemented;
+    case GST_RTSP_INVALID:
+    default:
+      goto bad_request;
+  }
+
+done:
+  if (ctx == &sctx)
+    gst_rtsp_context_pop_current (ctx);
+  if (session)
+    g_object_unref (session);
+  if (uri)
+    gst_rtsp_url_free (uri);
+  return;
+
+  /* ERRORS */
+not_supported:
+  {
+    GST_ERROR ("client %p: version %d not supported", client, version);
+    send_generic_response (client, GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED,
+        ctx);
+    goto done;
+  }
+bad_request:
+  {
+    GST_ERROR ("client %p: bad request", client);
+    send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+    goto done;
+  }
+no_pool:
+  {
+    GST_ERROR ("client %p: no pool configured", client);
+    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+    goto done;
+  }
+session_not_found:
+  {
+    GST_ERROR ("client %p: session not found", client);
+    send_generic_response (client, GST_RTSP_STS_SESSION_NOT_FOUND, ctx);
+    goto done;
+  }
+not_authorized:
+  {
+    GST_ERROR ("client %p: not allowed", client);
+    /* error reply is already sent */
+    goto done;
+  }
+#if 0
+unsupported_requirement:
+  {
+    GST_ERROR ("client %p: Required option is not supported (%s)", client,
+        unsupported_reqs);
+    send_option_not_supported_response (client, ctx, unsupported_reqs);
+    g_free (unsupported_reqs);
+    goto done;
+  }
+#endif
+not_implemented:
+  {
+    GST_ERROR ("client %p: method %d not implemented", client, method);
+    send_generic_response (client, GST_RTSP_STS_NOT_IMPLEMENTED, ctx);
+    goto done;
+  }
+}
+
+
+static void
+handle_response (GstRTSPClient * client, GstRTSPMessage * response)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+  GstRTSPResult res;
+  GstRTSPSession *session = NULL;
+  GstRTSPContext sctx = { NULL }, *ctx;
+  gchar *sessid;
+
+  if (!(ctx = gst_rtsp_context_get_current ())) {
+    ctx = &sctx;
+    ctx->auth = priv->auth;
+    gst_rtsp_context_push_current (ctx);
+  }
+
+  ctx->conn = priv->connection;
+  ctx->client = client;
+  ctx->request = NULL;
+  ctx->uri = NULL;
+  ctx->method = GST_RTSP_INVALID;
+  ctx->response = response;
+
+  if (gst_debug_category_get_threshold (rtsp_client_debug) >= GST_LEVEL_LOG) {
+    gst_rtsp_message_dump (response);
+  }
+
+  GST_INFO ("client %p: received a response", client);
+
+  /* get the session if there is any */
+  res =
+      gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &sessid, 0);
+  if (res == GST_RTSP_OK) {
+    if (priv->session_pool == NULL)
+      goto no_pool;
+
+    /* we had a session in the request, find it again */
+    if (!(session = gst_rtsp_session_pool_find (priv->session_pool, sessid)))
+      goto session_not_found;
+
+    /* we add the session to the client list of watched sessions. When a session
+     * disappears because it times out, we will be notified. If all sessions are
+     * gone, we will close the connection */
+    client_watch_session (client, session);
+  }
+
+  ctx->session = session;
+
+  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_HANDLE_RESPONSE],
+      0, ctx);
+
+done:
+  if (ctx == &sctx)
+    gst_rtsp_context_pop_current (ctx);
+  if (session)
+    g_object_unref (session);
+  return;
+
+no_pool:
+  {
+    GST_ERROR ("client %p: no pool configured", client);
+    goto done;
+  }
+session_not_found:
+  {
+    GST_ERROR ("client %p: session not found", client);
+    goto done;
+  }
+}
+
+static void
+handle_data (GstRTSPClient * client, GstRTSPMessage * message)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+  GstRTSPResult res;
+  guint8 channel;
+  GList *walk;
+  guint8 *data;
+  guint size;
+  GstBuffer *buffer;
+  gboolean handled;
+
+  /* find the stream for this message */
+  res = gst_rtsp_message_parse_data (message, &channel);
+  if (res != GST_RTSP_OK)
+    return;
+
+  gst_rtsp_message_steal_body (message, &data, &size);
+
+  buffer = gst_buffer_new_wrapped (data, size);
+
+  handled = FALSE;
+  for (walk = priv->transports; walk; walk = g_list_next (walk)) {
+    GstRTSPStreamTransport *trans;
+    GstRTSPStream *stream;
+    const GstRTSPTransport *tr;
+
+    trans = walk->data;
+
+    tr = gst_rtsp_stream_transport_get_transport (trans);
+    stream = gst_rtsp_stream_transport_get_stream (trans);
+
+    /* check for TCP transport */
+    if (tr->lower_transport == GST_RTSP_LOWER_TRANS_TCP) {
+      /* dispatch to the stream based on the channel number */
+      if (tr->interleaved.min == channel) {
+        gst_rtsp_stream_recv_rtp (stream, buffer);
+        handled = TRUE;
+        break;
+      } else if (tr->interleaved.max == channel) {
+        gst_rtsp_stream_recv_rtcp (stream, buffer);
+        handled = TRUE;
+        break;
+      }
+    }
+  }
+  if (!handled)
+    gst_buffer_unref (buffer);
+}
+
+/**
+ * gst_rtsp_client_set_session_pool:
+ * @client: a #GstRTSPClient
+ * @pool: (transfer none): a #GstRTSPSessionPool
+ *
+ * Set @pool as the sessionpool for @client which it will use to find
+ * or allocate sessions. the sessionpool is usually inherited from the server
+ * that created the client but can be overridden later.
+ */
+void
+gst_rtsp_client_set_session_pool (GstRTSPClient * client,
+    GstRTSPSessionPool * pool)
+{
+  GstRTSPSessionPool *old;
+  GstRTSPClientPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+  priv = client->priv;
+
+  if (pool)
+    g_object_ref (pool);
+
+  g_mutex_lock (&priv->lock);
+  old = priv->session_pool;
+  priv->session_pool = pool;
+
+  if (priv->session_removed_id) {
+    g_signal_handler_disconnect (old, priv->session_removed_id);
+    priv->session_removed_id = 0;
+  }
+  g_mutex_unlock (&priv->lock);
+
+  /* FIXME, should remove all sessions from the old pool for this client */
+  if (old)
+    g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_client_get_session_pool:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPSessionPool object that @client uses to manage its sessions.
+ *
+ * Returns: (transfer full): a #GstRTSPSessionPool, unref after usage.
+ */
+GstRTSPSessionPool *
+gst_rtsp_client_get_session_pool (GstRTSPClient * client)
+{
+  GstRTSPClientPrivate *priv;
+  GstRTSPSessionPool *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+  priv = client->priv;
+
+  g_mutex_lock (&priv->lock);
+  if ((result = priv->session_pool))
+    g_object_ref (result);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_client_set_mount_points:
+ * @client: a #GstRTSPClient
+ * @mounts: (transfer none): a #GstRTSPMountPoints
+ *
+ * Set @mounts as the mount points for @client which it will use to map urls
+ * to media streams. These mount points are usually inherited from the server that
+ * created the client but can be overriden later.
+ */
+void
+gst_rtsp_client_set_mount_points (GstRTSPClient * client,
+    GstRTSPMountPoints * mounts)
+{
+  GstRTSPClientPrivate *priv;
+  GstRTSPMountPoints *old;
+
+  g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+  priv = client->priv;
+
+  if (mounts)
+    g_object_ref (mounts);
+
+  g_mutex_lock (&priv->lock);
+  old = priv->mount_points;
+  priv->mount_points = mounts;
+  g_mutex_unlock (&priv->lock);
+
+  if (old)
+    g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_client_get_mount_points:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPMountPoints object that @client uses to manage its sessions.
+ *
+ * Returns: (transfer full): a #GstRTSPMountPoints, unref after usage.
+ */
+GstRTSPMountPoints *
+gst_rtsp_client_get_mount_points (GstRTSPClient * client)
+{
+  GstRTSPClientPrivate *priv;
+  GstRTSPMountPoints *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+  priv = client->priv;
+
+  g_mutex_lock (&priv->lock);
+  if ((result = priv->mount_points))
+    g_object_ref (result);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_client_set_auth:
+ * @client: a #GstRTSPClient
+ * @auth: (transfer none): a #GstRTSPAuth
+ *
+ * configure @auth to be used as the authentication manager of @client.
+ */
+void
+gst_rtsp_client_set_auth (GstRTSPClient * client, GstRTSPAuth * auth)
+{
+  GstRTSPClientPrivate *priv;
+  GstRTSPAuth *old;
+
+  g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+  priv = client->priv;
+
+  if (auth)
+    g_object_ref (auth);
+
+  g_mutex_lock (&priv->lock);
+  old = priv->auth;
+  priv->auth = auth;
+  g_mutex_unlock (&priv->lock);
+
+  if (old)
+    g_object_unref (old);
+}
+
+
+/**
+ * gst_rtsp_client_get_auth:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPAuth used as the authentication manager of @client.
+ *
+ * Returns: (transfer full): the #GstRTSPAuth of @client. g_object_unref() after
+ * usage.
+ */
+GstRTSPAuth *
+gst_rtsp_client_get_auth (GstRTSPClient * client)
+{
+  GstRTSPClientPrivate *priv;
+  GstRTSPAuth *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+  priv = client->priv;
+
+  g_mutex_lock (&priv->lock);
+  if ((result = priv->auth))
+    g_object_ref (result);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_client_set_thread_pool:
+ * @client: a #GstRTSPClient
+ * @pool: (transfer none): a #GstRTSPThreadPool
+ *
+ * configure @pool to be used as the thread pool of @client.
+ */
+void
+gst_rtsp_client_set_thread_pool (GstRTSPClient * client,
+    GstRTSPThreadPool * pool)
+{
+  GstRTSPClientPrivate *priv;
+  GstRTSPThreadPool *old;
+
+  g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+  priv = client->priv;
+
+  if (pool)
+    g_object_ref (pool);
+
+  g_mutex_lock (&priv->lock);
+  old = priv->thread_pool;
+  priv->thread_pool = pool;
+  g_mutex_unlock (&priv->lock);
+
+  if (old)
+    g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_client_get_thread_pool:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPThreadPool used as the thread pool of @client.
+ *
+ * Returns: (transfer full): the #GstRTSPThreadPool of @client. g_object_unref() after
+ * usage.
+ */
+GstRTSPThreadPool *
+gst_rtsp_client_get_thread_pool (GstRTSPClient * client)
+{
+  GstRTSPClientPrivate *priv;
+  GstRTSPThreadPool *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+  priv = client->priv;
+
+  g_mutex_lock (&priv->lock);
+  if ((result = priv->thread_pool))
+    g_object_ref (result);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_client_set_connection:
+ * @client: a #GstRTSPClient
+ * @conn: (transfer full): a #GstRTSPConnection
+ *
+ * Set the #GstRTSPConnection of @client. This function takes ownership of
+ * @conn.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_client_set_connection (GstRTSPClient * client,
+    GstRTSPConnection * conn)
+{
+  GstRTSPClientPrivate *priv;
+  GSocket *read_socket;
+  GSocketAddress *address;
+  GstRTSPUrl *url;
+  GError *error = NULL;
+
+  g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), FALSE);
+  g_return_val_if_fail (conn != NULL, FALSE);
+
+  priv = client->priv;
+
+  read_socket = gst_rtsp_connection_get_read_socket (conn);
+
+  if (!(address = g_socket_get_local_address (read_socket, &error)))
+    goto no_address;
+
+  g_free (priv->server_ip);
+  /* keep the original ip that the client connected to */
+  if (G_IS_INET_SOCKET_ADDRESS (address)) {
+    GInetAddress *iaddr;
+
+    iaddr = g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (address));
+
+    /* socket might be ipv6 but adress still ipv4 */
+    priv->is_ipv6 = g_inet_address_get_family (iaddr) == G_SOCKET_FAMILY_IPV6;
+    priv->server_ip = g_inet_address_to_string (iaddr);
+    g_object_unref (address);
+  } else {
+    priv->is_ipv6 = g_socket_get_family (read_socket) == G_SOCKET_FAMILY_IPV6;
+    priv->server_ip = g_strdup ("unknown");
+  }
+
+  GST_INFO ("client %p connected to server ip %s, ipv6 = %d", client,
+      priv->server_ip, priv->is_ipv6);
+
+  url = gst_rtsp_connection_get_url (conn);
+  GST_INFO ("added new client %p ip %s:%d", client, url->host, url->port);
+
+  priv->connection = conn;
+
+  return TRUE;
+
+  /* ERRORS */
+no_address:
+  {
+    GST_ERROR ("could not get local address %s", error->message);
+    g_error_free (error);
+    return FALSE;
+  }
+}
+
+/**
+ * gst_rtsp_client_get_connection:
+ * @client: a #GstRTSPClient
+ *
+ * Get the #GstRTSPConnection of @client.
+ *
+ * Returns: (transfer none): the #GstRTSPConnection of @client.
+ * The connection object returned remains valid until the client is freed.
+ */
+GstRTSPConnection *
+gst_rtsp_client_get_connection (GstRTSPClient * client)
+{
+  g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+  return client->priv->connection;
+}
+
+/**
+ * gst_rtsp_client_set_send_func:
+ * @client: a #GstRTSPClient
+ * @func: (scope notified): a #GstRTSPClientSendFunc
+ * @user_data: (closure): user data passed to @func
+ * @notify: (allow-none): called when @user_data is no longer in use
+ *
+ * Set @func as the callback that will be called when a new message needs to be
+ * sent to the client. @user_data is passed to @func and @notify is called when
+ * @user_data is no longer in use.
+ *
+ * By default, the client will send the messages on the #GstRTSPConnection that
+ * was configured with gst_rtsp_client_attach() was called.
+ */
+void
+gst_rtsp_client_set_send_func (GstRTSPClient * client,
+    GstRTSPClientSendFunc func, gpointer user_data, GDestroyNotify notify)
+{
+  GstRTSPClientPrivate *priv;
+  GDestroyNotify old_notify;
+  gpointer old_data;
+
+  g_return_if_fail (GST_IS_RTSP_CLIENT (client));
+
+  priv = client->priv;
+
+  g_mutex_lock (&priv->send_lock);
+  priv->send_func = func;
+  old_notify = priv->send_notify;
+  old_data = priv->send_data;
+  priv->send_notify = notify;
+  priv->send_data = user_data;
+  g_mutex_unlock (&priv->send_lock);
+
+  if (old_notify)
+    old_notify (old_data);
+}
+
+/**
+ * gst_rtsp_client_handle_message:
+ * @client: a #GstRTSPClient
+ * @message: (transfer none): an #GstRTSPMessage
+ *
+ * Let the client handle @message.
+ *
+ * Returns: a #GstRTSPResult.
+ */
+GstRTSPResult
+gst_rtsp_client_handle_message (GstRTSPClient * client,
+    GstRTSPMessage * message)
+{
+  g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
+  g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
+
+  switch (message->type) {
+    case GST_RTSP_MESSAGE_REQUEST:
+      handle_request (client, message);
+      break;
+    case GST_RTSP_MESSAGE_RESPONSE:
+      handle_response (client, message);
+      break;
+    case GST_RTSP_MESSAGE_DATA:
+      handle_data (client, message);
+      break;
+    default:
+      break;
+  }
+  return GST_RTSP_OK;
+}
+
+/**
+ * gst_rtsp_client_send_message:
+ * @client: a #GstRTSPClient
+ * @session: (allow-none) (transfer none): a #GstRTSPSession to send
+ *   the message to or %NULL
+ * @message: (transfer none): The #GstRTSPMessage to send
+ *
+ * Send a message message to the remote end. @message must be a
+ * #GST_RTSP_MESSAGE_REQUEST or a #GST_RTSP_MESSAGE_RESPONSE.
+ */
+GstRTSPResult
+gst_rtsp_client_send_message (GstRTSPClient * client, GstRTSPSession * session,
+    GstRTSPMessage * message)
+{
+  GstRTSPContext sctx = { NULL }
+  , *ctx;
+  GstRTSPClientPrivate *priv;
+
+  g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), GST_RTSP_EINVAL);
+  g_return_val_if_fail (message != NULL, GST_RTSP_EINVAL);
+  g_return_val_if_fail (message->type == GST_RTSP_MESSAGE_REQUEST ||
+      message->type == GST_RTSP_MESSAGE_RESPONSE, GST_RTSP_EINVAL);
+
+  priv = client->priv;
+
+  if (!(ctx = gst_rtsp_context_get_current ())) {
+    ctx = &sctx;
+    ctx->auth = priv->auth;
+    gst_rtsp_context_push_current (ctx);
+  }
+
+  ctx->conn = priv->connection;
+  ctx->client = client;
+  ctx->session = session;
+
+  send_message (client, ctx, message, FALSE);
+
+  if (ctx == &sctx)
+    gst_rtsp_context_pop_current (ctx);
+
+  return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+do_send_message (GstRTSPClient * client, GstRTSPMessage * message,
+    gboolean close, gpointer user_data)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+  GstRTSPResult ret;
+  GTimeVal time;
+
+  time.tv_sec = 1;
+  time.tv_usec = 0;
+
+  do {
+    /* send the response and store the seq number so we can wait until it's
+     * written to the client to close the connection */
+    ret =
+        gst_rtsp_watch_send_message (priv->watch, message,
+        close ? &priv->close_seq : NULL);
+    if (ret == GST_RTSP_OK)
+      break;
+
+    if (ret != GST_RTSP_ENOMEM)
+      goto error;
+
+    /* drop backlog */
+    if (priv->drop_backlog)
+      break;
+
+    /* queue was full, wait for more space */
+    GST_DEBUG_OBJECT (client, "waiting for backlog");
+    ret = gst_rtsp_watch_wait_backlog (priv->watch, &time);
+    GST_DEBUG_OBJECT (client, "Resend due to backlog full");
+  } while (ret != GST_RTSP_EINTR);
+
+  return ret;
+
+  /* ERRORS */
+error:
+  {
+    GST_DEBUG_OBJECT (client, "got error %d", ret);
+    return ret;
+  }
+}
+
+static GstRTSPResult
+message_received (GstRTSPWatch * watch, GstRTSPMessage * message,
+    gpointer user_data)
+{
+  return gst_rtsp_client_handle_message (GST_RTSP_CLIENT (user_data), message);
+}
+
+static GstRTSPResult
+message_sent (GstRTSPWatch * watch, guint cseq, gpointer user_data)
+{
+  GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+  GstRTSPClientPrivate *priv = client->priv;
+
+  if (priv->close_seq && priv->close_seq == cseq) {
+    GST_INFO ("client %p: send close message", client);
+    priv->close_seq = 0;
+    gst_rtsp_client_close (client);
+  }
+
+  return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+closed (GstRTSPWatch * watch, gpointer user_data)
+{
+  GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+  GstRTSPClientPrivate *priv = client->priv;
+  const gchar *tunnelid;
+
+  GST_INFO ("client %p: connection closed", client);
+
+  if ((tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection))) {
+    g_mutex_lock (&tunnels_lock);
+    /* remove from tunnelids */
+    g_hash_table_remove (tunnels, tunnelid);
+    g_mutex_unlock (&tunnels_lock);
+  }
+
+  gst_rtsp_watch_set_flushing (watch, TRUE);
+  g_mutex_lock (&priv->watch_lock);
+  gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+  g_mutex_unlock (&priv->watch_lock);
+
+  return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+error (GstRTSPWatch * watch, GstRTSPResult result, gpointer user_data)
+{
+  GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+  gchar *str;
+
+  str = gst_rtsp_strresult (result);
+  GST_INFO ("client %p: received an error %s", client, str);
+  g_free (str);
+
+  return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+error_full (GstRTSPWatch * watch, GstRTSPResult result,
+    GstRTSPMessage * message, guint id, gpointer user_data)
+{
+  GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+  gchar *str;
+
+  str = gst_rtsp_strresult (result);
+  GST_INFO
+      ("client %p: error when handling message %p with id %d: %s",
+      client, message, id, str);
+  g_free (str);
+
+  return GST_RTSP_OK;
+}
+
+static gboolean
+remember_tunnel (GstRTSPClient * client)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+  const gchar *tunnelid;
+
+  /* store client in the pending tunnels */
+  tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
+  if (tunnelid == NULL)
+    goto no_tunnelid;
+
+  GST_INFO ("client %p: inserting tunnel session %s", client, tunnelid);
+
+  /* we can't have two clients connecting with the same tunnelid */
+  g_mutex_lock (&tunnels_lock);
+  if (g_hash_table_lookup (tunnels, tunnelid))
+    goto tunnel_existed;
+
+  g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
+  g_mutex_unlock (&tunnels_lock);
+
+  return TRUE;
+
+  /* ERRORS */
+no_tunnelid:
+  {
+    GST_ERROR ("client %p: no tunnelid provided", client);
+    return FALSE;
+  }
+tunnel_existed:
+  {
+    g_mutex_unlock (&tunnels_lock);
+    GST_ERROR ("client %p: tunnel session %s already existed", client,
+        tunnelid);
+    return FALSE;
+  }
+}
+
+static GstRTSPResult
+tunnel_lost (GstRTSPWatch * watch, gpointer user_data)
+{
+  GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+  GstRTSPClientPrivate *priv = client->priv;
+
+  GST_WARNING ("client %p: tunnel lost (connection %p)", client,
+      priv->connection);
+
+  /* ignore error, it'll only be a problem when the client does a POST again */
+  remember_tunnel (client);
+
+  return GST_RTSP_OK;
+}
+
+static gboolean
+handle_tunnel (GstRTSPClient * client)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+  GstRTSPClient *oclient;
+  GstRTSPClientPrivate *opriv;
+  const gchar *tunnelid;
+
+  tunnelid = gst_rtsp_connection_get_tunnelid (priv->connection);
+  if (tunnelid == NULL)
+    goto no_tunnelid;
+
+  /* check for previous tunnel */
+  g_mutex_lock (&tunnels_lock);
+  oclient = g_hash_table_lookup (tunnels, tunnelid);
+
+  if (oclient == NULL) {
+    /* no previous tunnel, remember tunnel */
+    g_hash_table_insert (tunnels, g_strdup (tunnelid), g_object_ref (client));
+    g_mutex_unlock (&tunnels_lock);
+
+    GST_INFO ("client %p: no previous tunnel found, remembering tunnel (%p)",
+        client, priv->connection);
+  } else {
+    /* merge both tunnels into the first client */
+    /* remove the old client from the table. ref before because removing it will
+     * remove the ref to it. */
+    g_object_ref (oclient);
+    g_hash_table_remove (tunnels, tunnelid);
+    g_mutex_unlock (&tunnels_lock);
+
+    opriv = oclient->priv;
+
+    g_mutex_lock (&opriv->watch_lock);
+    if (opriv->watch == NULL)
+      goto tunnel_closed;
+
+    GST_INFO ("client %p: found previous tunnel %p (old %p, new %p)", client,
+        oclient, opriv->connection, priv->connection);
+
+    gst_rtsp_connection_do_tunnel (opriv->connection, priv->connection);
+    gst_rtsp_watch_reset (priv->watch);
+    gst_rtsp_watch_reset (opriv->watch);
+    g_mutex_unlock (&opriv->watch_lock);
+    g_object_unref (oclient);
+
+    /* the old client owns the tunnel now, the new one will be freed */
+    g_source_destroy ((GSource *) priv->watch);
+    priv->watch = NULL;
+    gst_rtsp_client_set_send_func (client, NULL, NULL, NULL);
+  }
+
+  return TRUE;
+
+  /* ERRORS */
+no_tunnelid:
+  {
+    GST_ERROR ("client %p: no tunnelid provided", client);
+    return FALSE;
+  }
+tunnel_closed:
+  {
+    GST_ERROR ("client %p: tunnel session %s was closed", client, tunnelid);
+    g_mutex_unlock (&opriv->watch_lock);
+    g_object_unref (oclient);
+    return FALSE;
+  }
+}
+
+static GstRTSPStatusCode
+tunnel_get (GstRTSPWatch * watch, gpointer user_data)
+{
+  GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+
+  GST_INFO ("client %p: tunnel get (connection %p)", client,
+      client->priv->connection);
+
+  if (!handle_tunnel (client)) {
+    return GST_RTSP_STS_SERVICE_UNAVAILABLE;
+  }
+
+  return GST_RTSP_STS_OK;
+}
+
+static GstRTSPResult
+tunnel_post (GstRTSPWatch * watch, gpointer user_data)
+{
+  GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+
+  GST_INFO ("client %p: tunnel post (connection %p)", client,
+      client->priv->connection);
+
+  if (!handle_tunnel (client)) {
+    return GST_RTSP_ERROR;
+  }
+
+  return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+tunnel_http_response (GstRTSPWatch * watch, GstRTSPMessage * request,
+    GstRTSPMessage * response, gpointer user_data)
+{
+  GstRTSPClientClass *klass;
+
+  GstRTSPClient *client = GST_RTSP_CLIENT (user_data);
+  klass = GST_RTSP_CLIENT_GET_CLASS (client);
+
+  if (klass->tunnel_http_response) {
+    klass->tunnel_http_response (client, request, response);
+  }
+
+  return GST_RTSP_OK;
+}
+
+static GstRTSPWatchFuncs watch_funcs = {
+  message_received,
+  message_sent,
+  closed,
+  error,
+  tunnel_get,
+  tunnel_post,
+  error_full,
+  tunnel_lost,
+  tunnel_http_response
+};
+
+static void
+client_watch_notify (GstRTSPClient * client)
+{
+  GstRTSPClientPrivate *priv = client->priv;
+
+  GST_INFO ("client %p: watch destroyed", client);
+  priv->watch = NULL;
+  g_main_context_unref (priv->watch_context);
+  priv->watch_context = NULL;
+  /* remove all sessions and so drop the extra client ref */
+  gst_rtsp_client_session_filter (client, cleanup_session, NULL);
+  g_signal_emit (client, gst_rtsp_client_signals[SIGNAL_CLOSED], 0, NULL);
+  g_object_unref (client);
+}
+
+/**
+ * gst_rtsp_client_attach:
+ * @client: a #GstRTSPClient
+ * @context: (allow-none): a #GMainContext
+ *
+ * Attaches @client to @context. When the mainloop for @context is run, the
+ * client will be dispatched. When @context is %NULL, the default context will be
+ * used).
+ *
+ * This function should be called when the client properties and urls are fully
+ * configured and the client is ready to start.
+ *
+ * Returns: the ID (greater than 0) for the source within the GMainContext.
+ */
+guint
+gst_rtsp_client_attach (GstRTSPClient * client, GMainContext * context)
+{
+  GstRTSPClientPrivate *priv;
+  guint res;
+
+  g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), 0);
+  priv = client->priv;
+  g_return_val_if_fail (priv->connection != NULL, 0);
+  g_return_val_if_fail (priv->watch == NULL, 0);
+
+  /* make sure noone will free the context before the watch is destroyed */
+  priv->watch_context = g_main_context_ref (context);
+
+  /* create watch for the connection and attach */
+  priv->watch = gst_rtsp_watch_new (priv->connection, &watch_funcs,
+      g_object_ref (client), (GDestroyNotify) client_watch_notify);
+  gst_rtsp_client_set_send_func (client, do_send_message, priv->watch,
+      (GDestroyNotify) gst_rtsp_watch_unref);
+
+  /* FIXME make this configurable. We don't want to do this yet because it will
+   * be superceeded by a cache object later */
+  gst_rtsp_watch_set_send_backlog (priv->watch, 0, 100);
+
+  GST_INFO ("client %p: attaching to context %p", client, context);
+  res = gst_rtsp_watch_attach (priv->watch, context);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_client_session_filter:
+ * @client: a #GstRTSPClient
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: user data passed to @func
+ *
+ * Call @func for each session managed by @client. The result value of @func
+ * determines what happens to the session. @func will be called with @client
+ * locked so no further actions on @client can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be removed from
+ * @client.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @client.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @client but
+ * will also be added with an additional ref to the result #GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
+ *
+ * Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
+ * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the #GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_client_session_filter (GstRTSPClient * client,
+    GstRTSPClientSessionFilterFunc func, gpointer user_data)
+{
+  GstRTSPClientPrivate *priv;
+  GList *result, *walk, *next;
+  GHashTable *visited;
+  guint cookie;
+
+  g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
+
+  priv = client->priv;
+
+  result = NULL;
+  if (func)
+    visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+  g_mutex_lock (&priv->lock);
+restart:
+  cookie = priv->sessions_cookie;
+  for (walk = priv->sessions; walk; walk = next) {
+    GstRTSPSession *sess = walk->data;
+    GstRTSPFilterResult res;
+    gboolean changed;
+
+    next = g_list_next (walk);
+
+    if (func) {
+      /* only visit each session once */
+      if (g_hash_table_contains (visited, sess))
+        continue;
+
+      g_hash_table_add (visited, g_object_ref (sess));
+      g_mutex_unlock (&priv->lock);
+
+      res = func (client, sess, user_data);
+
+      g_mutex_lock (&priv->lock);
+    } else
+      res = GST_RTSP_FILTER_REF;
+
+    changed = (cookie != priv->sessions_cookie);
+
+    switch (res) {
+      case GST_RTSP_FILTER_REMOVE:
+        /* stop watching the session and pretend it went away, if the list was
+         * changed, we can't use the current list position, try to see if we
+         * still have the session */
+        client_unwatch_session (client, sess, changed ? NULL : walk);
+        cookie = priv->sessions_cookie;
+        break;
+      case GST_RTSP_FILTER_REF:
+        result = g_list_prepend (result, g_object_ref (sess));
+        break;
+      case GST_RTSP_FILTER_KEEP:
+      default:
+        break;
+    }
+    if (changed)
+      goto restart;
+  }
+  g_mutex_unlock (&priv->lock);
+
+  if (func)
+    g_hash_table_unref (visited);
+
+  return result;
+}
diff --git a/gst/rtsp-server/rtsp-client.h b/gst/rtsp-server/rtsp-client.h
new file mode 100644
index 0000000..2ddb82e
--- /dev/null
+++ b/gst/rtsp-server/rtsp-client.h
@@ -0,0 +1,196 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+#include 
+
+#ifndef __GST_RTSP_CLIENT_H__
+#define __GST_RTSP_CLIENT_H__
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPClient GstRTSPClient;
+typedef struct _GstRTSPClientClass GstRTSPClientClass;
+typedef struct _GstRTSPClientPrivate GstRTSPClientPrivate;
+
+#include "rtsp-context.h"
+#include "rtsp-mount-points.h"
+#include "rtsp-sdp.h"
+#include "rtsp-auth.h"
+
+#define GST_TYPE_RTSP_CLIENT              (gst_rtsp_client_get_type ())
+#define GST_IS_RTSP_CLIENT(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_CLIENT))
+#define GST_IS_RTSP_CLIENT_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_CLIENT))
+#define GST_RTSP_CLIENT_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClientClass))
+#define GST_RTSP_CLIENT(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_CLIENT, GstRTSPClient))
+#define GST_RTSP_CLIENT_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_CLIENT, GstRTSPClientClass))
+#define GST_RTSP_CLIENT_CAST(obj)         ((GstRTSPClient*)(obj))
+#define GST_RTSP_CLIENT_CLASS_CAST(klass) ((GstRTSPClientClass*)(klass))
+
+/**
+ * GstRTSPClientSendFunc:
+ * @client: a #GstRTSPClient
+ * @message: a #GstRTSPMessage
+ * @close: close the connection
+ * @user_data: user data when registering the callback
+ *
+ * This callback is called when @client wants to send @message. When @close is
+ * %TRUE, the connection should be closed when the message has been sent.
+ *
+ * Returns: %TRUE on success.
+ */
+typedef gboolean (*GstRTSPClientSendFunc)      (GstRTSPClient *client,
+                                                GstRTSPMessage *message,
+                                                gboolean close,
+                                                gpointer user_data);
+
+/**
+ * GstRTSPClient:
+ *
+ * The client object represents the connection and its state with a client.
+ */
+struct _GstRTSPClient {
+  GObject       parent;
+
+  /*< private >*/
+  GstRTSPClientPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPClientClass:
+ * @create_sdp: called when the SDP needs to be created for media.
+ * @configure_client_media: called when the stream in media needs to be configured.
+ *    The default implementation will configure the blocksize on the payloader when
+ *    spcified in the request headers.
+ * @configure_client_transport: called when the client transport needs to be
+ *    configured.
+ * @params_set: set parameters. This function should also initialize the
+ *    RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
+ * @params_get: get parameters. This function should also initialize the
+ *    RTSP response(ctx->response) via a call to gst_rtsp_message_init_response()
+ * @tunnel_http_response: called when a response to the GET request is about to
+ *   be sent for a tunneled connection. The response can be modified. Since 1.4
+ *
+ * The client class structure.
+ */
+struct _GstRTSPClientClass {
+  GObjectClass  parent_class;
+
+  GstSDPMessage * (*create_sdp) (GstRTSPClient *client, GstRTSPMedia *media);
+  gboolean        (*configure_client_media)     (GstRTSPClient * client,
+                                                 GstRTSPMedia * media, GstRTSPStream * stream,
+                                                 GstRTSPContext * ctx);
+  gboolean        (*configure_client_transport) (GstRTSPClient * client,
+                                                 GstRTSPContext * ctx,
+                                                 GstRTSPTransport * ct);
+  GstRTSPResult   (*params_set) (GstRTSPClient *client, GstRTSPContext *ctx);
+  GstRTSPResult   (*params_get) (GstRTSPClient *client, GstRTSPContext *ctx);
+  gchar *         (*make_path_from_uri) (GstRTSPClient *client, const GstRTSPUrl *uri);
+  gboolean        (*handle_options_request) (GstRTSPClient * client, GstRTSPContext * ctx);
+  gboolean        (*handle_set_param_request) (GstRTSPClient * client, GstRTSPContext * ctx);
+  gboolean        (*handle_get_param_request) (GstRTSPClient * client, GstRTSPContext * ctx);
+
+  /* signals */
+  void     (*closed)                  (GstRTSPClient *client);
+  void     (*new_session)             (GstRTSPClient *client, GstRTSPSession *session);
+  void     (*options_request)         (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*describe_request)        (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*setup_request)           (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*play_request)            (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*pause_request)           (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*teardown_request)        (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*set_parameter_request)   (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*get_parameter_request)   (GstRTSPClient *client, GstRTSPContext *ctx);
+  void     (*handle_response)         (GstRTSPClient *client, GstRTSPContext *ctx);
+
+  void     (*tunnel_http_response)    (GstRTSPClient * client, GstRTSPMessage * request,
+                                       GstRTSPMessage * response);
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING_LARGE-1];
+};
+
+GType                 gst_rtsp_client_get_type          (void);
+
+GstRTSPClient *       gst_rtsp_client_new               (void);
+
+void                  gst_rtsp_client_set_session_pool  (GstRTSPClient *client,
+                                                         GstRTSPSessionPool *pool);
+GstRTSPSessionPool *  gst_rtsp_client_get_session_pool  (GstRTSPClient *client);
+
+void                  gst_rtsp_client_set_mount_points  (GstRTSPClient *client,
+                                                         GstRTSPMountPoints *mounts);
+GstRTSPMountPoints *  gst_rtsp_client_get_mount_points  (GstRTSPClient *client);
+
+void                  gst_rtsp_client_set_auth          (GstRTSPClient *client, GstRTSPAuth *auth);
+GstRTSPAuth *         gst_rtsp_client_get_auth          (GstRTSPClient *client);
+
+void                  gst_rtsp_client_set_thread_pool   (GstRTSPClient *client, GstRTSPThreadPool *pool);
+GstRTSPThreadPool *   gst_rtsp_client_get_thread_pool   (GstRTSPClient *client);
+
+gboolean              gst_rtsp_client_set_connection    (GstRTSPClient *client, GstRTSPConnection *conn);
+GstRTSPConnection *   gst_rtsp_client_get_connection    (GstRTSPClient *client);
+
+guint                 gst_rtsp_client_attach            (GstRTSPClient *client,
+                                                         GMainContext *context);
+void                  gst_rtsp_client_close             (GstRTSPClient * client);
+
+void                  gst_rtsp_client_set_send_func     (GstRTSPClient *client,
+                                                         GstRTSPClientSendFunc func,
+                                                         gpointer user_data,
+                                                         GDestroyNotify notify);
+
+GstRTSPResult         gst_rtsp_client_handle_message    (GstRTSPClient *client,
+                                                         GstRTSPMessage *message);
+GstRTSPResult         gst_rtsp_client_send_message      (GstRTSPClient * client,
+                                                         GstRTSPSession *session,
+                                                         GstRTSPMessage *message);
+/**
+ * GstRTSPClientSessionFilterFunc:
+ * @client: a #GstRTSPClient object
+ * @sess: a #GstRTSPSession in @client
+ * @user_data: user data that has been given to gst_rtsp_client_session_filter()
+ *
+ * This function will be called by the gst_rtsp_client_session_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @sess will be removed
+ * from @client.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @sess untouched in
+ * @client.
+ *
+ * A value of #GST_RTSP_FILTER_REF will add @sess to the result #GList of
+ * gst_rtsp_client_session_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPClientSessionFilterFunc)  (GstRTSPClient *client,
+                                                                GstRTSPSession *sess,
+                                                                gpointer user_data);
+
+GList *                gst_rtsp_client_session_filter    (GstRTSPClient *client,
+                                                          GstRTSPClientSessionFilterFunc func,
+                                                          gpointer user_data);
+
+
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_CLIENT_H__ */
diff --git a/gst/rtsp-server/rtsp-context.c b/gst/rtsp-server/rtsp-context.c
new file mode 100644
index 0000000..919ca2d
--- /dev/null
+++ b/gst/rtsp-server/rtsp-context.c
@@ -0,0 +1,92 @@
+/* GStreamer
+ * Copyright (C) 2013 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-context
+ * @short_description: A client request context
+ * @see_also: #GstRTSPServer, #GstRTSPClient
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include "rtsp-context.h"
+
+G_DEFINE_POINTER_TYPE (GstRTSPContext, gst_rtsp_context);
+
+static GPrivate current_context;
+
+/**
+ * gst_rtsp_context_get_current:
+ *
+ * Get the current #GstRTSPContext. This object is retrieved from the
+ * current thread that is handling the request for a client.
+ *
+ * Returns: a #GstRTSPContext
+ */
+GstRTSPContext *
+gst_rtsp_context_get_current (void)
+{
+  GSList *l;
+
+  l = g_private_get (¤t_context);
+  if (l == NULL)
+    return NULL;
+
+  return (GstRTSPContext *) (l->data);
+
+}
+
+/**
+ * gst_rtsp_context_push_current:
+ * @ctx: a ##GstRTSPContext
+ *
+ * Pushes @ctx onto the context stack. The current
+ * context can then be received using gst_rtsp_context_get_current().
+ **/
+void
+gst_rtsp_context_push_current (GstRTSPContext * ctx)
+{
+  GSList *l;
+
+  g_return_if_fail (ctx != NULL);
+
+  l = g_private_get (¤t_context);
+  l = g_slist_prepend (l, ctx);
+  g_private_set (¤t_context, l);
+}
+
+/**
+ * gst_rtsp_context_pop_current:
+ * @ctx: a #GstRTSPContext
+ *
+ * Pops @ctx off the context stack (verifying that @ctx
+ * is on the top of the stack).
+ **/
+void
+gst_rtsp_context_pop_current (GstRTSPContext * ctx)
+{
+  GSList *l;
+
+  l = g_private_get (¤t_context);
+
+  g_return_if_fail (l != NULL);
+  g_return_if_fail (l->data == ctx);
+
+  l = g_slist_delete_link (l, l);
+  g_private_set (¤t_context, l);
+}
diff --git a/gst/rtsp-server/rtsp-context.h b/gst/rtsp-server/rtsp-context.h
new file mode 100644
index 0000000..c2be87e
--- /dev/null
+++ b/gst/rtsp-server/rtsp-context.h
@@ -0,0 +1,88 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+#include 
+
+#ifndef __GST_RTSP_CONTEXT_H__
+#define __GST_RTSP_CONTEXT_H__
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_CONTEXT              (gst_rtsp_context_get_type ())
+
+typedef struct _GstRTSPContext GstRTSPContext;
+
+#include "rtsp-server.h"
+#include "rtsp-media.h"
+#include "rtsp-media-factory.h"
+#include "rtsp-session-media.h"
+#include "rtsp-auth.h"
+#include "rtsp-thread-pool.h"
+#include "rtsp-token.h"
+
+/**
+ * GstRTSPContext:
+ * @server: the server
+ * @conn: the connection
+ * @client: the client
+ * @request: the complete request
+ * @uri: the complete url parsed from @request
+ * @method: the parsed method of @uri
+ * @auth: the current auth object or %NULL
+ * @token: authorisation token
+ * @session: the session, can be %NULL
+ * @sessmedia: the session media for the url can be %NULL
+ * @factory: the media factory for the url, can be %NULL
+ * @media: the media for the url can be %NULL
+ * @stream: the stream for the url can be %NULL
+ * @response: the response
+ *
+ * Information passed around containing the context of a request.
+ */
+struct _GstRTSPContext {
+  GstRTSPServer       *server;
+  GstRTSPConnection   *conn;
+  GstRTSPClient       *client;
+  GstRTSPMessage      *request;
+  GstRTSPUrl          *uri;
+  GstRTSPMethod        method;
+  GstRTSPAuth         *auth;
+  GstRTSPToken        *token;
+  GstRTSPSession      *session;
+  GstRTSPSessionMedia *sessmedia;
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia        *media;
+  GstRTSPStream       *stream;
+  GstRTSPMessage      *response;
+
+  /*< private >*/
+  gpointer            _gst_reserved[GST_PADDING];
+};
+
+GType gst_rtsp_context_get_type (void);
+
+GstRTSPContext *     gst_rtsp_context_get_current   (void);
+void                 gst_rtsp_context_push_current  (GstRTSPContext * ctx);
+void                 gst_rtsp_context_pop_current   (GstRTSPContext * ctx);
+
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_CONTEXT_H__ */
diff --git a/gst/rtsp-server/rtsp-media-factory-uri.c b/gst/rtsp-server/rtsp-media-factory-uri.c
new file mode 100644
index 0000000..01f3f99
--- /dev/null
+++ b/gst/rtsp-server/rtsp-media-factory-uri.c
@@ -0,0 +1,645 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-media-factory-uri
+ * @short_description: A factory for URI sources
+ * @see_also: #GstRTSPMediaFactory, #GstRTSPMedia
+ *
+ * This specialized #GstRTSPMediaFactory constructs media pipelines from a URI,
+ * given with gst_rtsp_media_factory_uri_set_uri().
+ *
+ * It will automatically demux and payload the different streams found in the
+ * media at URL.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include 
+
+#include "rtsp-media-factory-uri.h"
+
+#define GST_RTSP_MEDIA_FACTORY_URI_GET_PRIVATE(obj)  \
+    (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_URI, GstRTSPMediaFactoryURIPrivate))
+
+struct _GstRTSPMediaFactoryURIPrivate
+{
+  GMutex lock;
+  gchar *uri;                   /* protected by lock */
+  gboolean use_gstpay;
+
+  GstCaps *raw_vcaps;
+  GstCaps *raw_acaps;
+  GList *demuxers;
+  GList *payloaders;
+  GList *decoders;
+};
+
+#define DEFAULT_URI         NULL
+#define DEFAULT_USE_GSTPAY  FALSE
+
+enum
+{
+  PROP_0,
+  PROP_URI,
+  PROP_USE_GSTPAY,
+  PROP_LAST
+};
+
+
+#define RAW_VIDEO_CAPS \
+    "video/x-raw"
+
+#define RAW_AUDIO_CAPS \
+    "audio/x-raw"
+
+static GstStaticCaps raw_video_caps = GST_STATIC_CAPS (RAW_VIDEO_CAPS);
+static GstStaticCaps raw_audio_caps = GST_STATIC_CAPS (RAW_AUDIO_CAPS);
+
+typedef struct
+{
+  GstRTSPMediaFactoryURI *factory;
+  guint pt;
+} FactoryData;
+
+static void
+free_data (FactoryData * data)
+{
+  g_object_unref (data->factory);
+  g_free (data);
+}
+
+static const gchar *factory_key = "GstRTSPMediaFactoryURI";
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_factory_uri_debug);
+#define GST_CAT_DEFAULT rtsp_media_factory_uri_debug
+
+static void gst_rtsp_media_factory_uri_get_property (GObject * object,
+    guint propid, GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_factory_uri_set_property (GObject * object,
+    guint propid, const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_factory_uri_finalize (GObject * obj);
+
+static GstElement *rtsp_media_factory_uri_create_element (GstRTSPMediaFactory *
+    factory, const GstRTSPUrl * url);
+
+G_DEFINE_TYPE (GstRTSPMediaFactoryURI, gst_rtsp_media_factory_uri,
+    GST_TYPE_RTSP_MEDIA_FACTORY);
+
+static void
+gst_rtsp_media_factory_uri_class_init (GstRTSPMediaFactoryURIClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstRTSPMediaFactoryClass *mediafactory_class;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPMediaFactoryURIPrivate));
+
+  gobject_class = G_OBJECT_CLASS (klass);
+  mediafactory_class = GST_RTSP_MEDIA_FACTORY_CLASS (klass);
+
+  gobject_class->get_property = gst_rtsp_media_factory_uri_get_property;
+  gobject_class->set_property = gst_rtsp_media_factory_uri_set_property;
+  gobject_class->finalize = gst_rtsp_media_factory_uri_finalize;
+
+  /**
+   * GstRTSPMediaFactoryURI::uri:
+   *
+   * The uri of the resource that will be served by this factory.
+   */
+  g_object_class_install_property (gobject_class, PROP_URI,
+      g_param_spec_string ("uri", "URI",
+          "The URI of the resource to stream", DEFAULT_URI,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRTSPMediaFactoryURI::use-gstpay:
+   *
+   * Allow the usage of gstpay in order to avoid decoding of compressed formats
+   * without a payloader.
+   */
+  g_object_class_install_property (gobject_class, PROP_USE_GSTPAY,
+      g_param_spec_boolean ("use-gstpay", "Use gstpay",
+          "Use the gstpay payloader to avoid decoding", DEFAULT_USE_GSTPAY,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  mediafactory_class->create_element = rtsp_media_factory_uri_create_element;
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_media_factory_uri_debug, "rtspmediafactoryuri",
+      0, "GstRTSPMediaFactoryUri");
+}
+
+typedef struct
+{
+  GList *demux;
+  GList *payload;
+  GList *decode;
+} FilterData;
+
+static gboolean
+payloader_filter (GstPluginFeature * feature, FilterData * data)
+{
+  const gchar *klass;
+  GstElementFactory *fact;
+  GList **list = NULL;
+
+  /* we only care about element factories */
+  if (G_UNLIKELY (!GST_IS_ELEMENT_FACTORY (feature)))
+    return FALSE;
+
+  if (gst_plugin_feature_get_rank (feature) < GST_RANK_MARGINAL)
+    return FALSE;
+
+  fact = GST_ELEMENT_FACTORY_CAST (feature);
+
+  klass = gst_element_factory_get_metadata (fact, GST_ELEMENT_METADATA_KLASS);
+
+  if (strstr (klass, "Decoder"))
+    list = &data->decode;
+  else if (strstr (klass, "Demux"))
+    list = &data->demux;
+  else if (strstr (klass, "Parser") && strstr (klass, "Codec"))
+    list = &data->demux;
+  else if (strstr (klass, "Payloader") && strstr (klass, "RTP"))
+    list = &data->payload;
+
+  if (list) {
+    GST_DEBUG ("adding %s", GST_OBJECT_NAME (fact));
+    *list = g_list_prepend (*list, gst_object_ref (fact));
+  }
+
+  return FALSE;
+}
+
+static void
+gst_rtsp_media_factory_uri_init (GstRTSPMediaFactoryURI * factory)
+{
+  GstRTSPMediaFactoryURIPrivate *priv =
+      GST_RTSP_MEDIA_FACTORY_URI_GET_PRIVATE (factory);
+  FilterData data = { NULL, NULL, NULL };
+
+  GST_DEBUG_OBJECT (factory, "new");
+
+  factory->priv = priv;
+
+  priv->uri = g_strdup (DEFAULT_URI);
+  priv->use_gstpay = DEFAULT_USE_GSTPAY;
+  g_mutex_init (&priv->lock);
+
+  /* get the feature list using the filter */
+  gst_registry_feature_filter (gst_registry_get (), (GstPluginFeatureFilter)
+      payloader_filter, FALSE, &data);
+  /* sort */
+  priv->demuxers =
+      g_list_sort (data.demux, gst_plugin_feature_rank_compare_func);
+  priv->payloaders =
+      g_list_sort (data.payload, gst_plugin_feature_rank_compare_func);
+  priv->decoders =
+      g_list_sort (data.decode, gst_plugin_feature_rank_compare_func);
+
+  priv->raw_vcaps = gst_static_caps_get (&raw_video_caps);
+  priv->raw_acaps = gst_static_caps_get (&raw_audio_caps);
+}
+
+static void
+gst_rtsp_media_factory_uri_finalize (GObject * obj)
+{
+  GstRTSPMediaFactoryURI *factory = GST_RTSP_MEDIA_FACTORY_URI (obj);
+  GstRTSPMediaFactoryURIPrivate *priv = factory->priv;
+
+  GST_DEBUG_OBJECT (factory, "finalize");
+
+  g_free (priv->uri);
+  gst_plugin_feature_list_free (priv->demuxers);
+  gst_plugin_feature_list_free (priv->payloaders);
+  gst_plugin_feature_list_free (priv->decoders);
+  gst_caps_unref (priv->raw_vcaps);
+  gst_caps_unref (priv->raw_acaps);
+  g_mutex_clear (&priv->lock);
+
+  G_OBJECT_CLASS (gst_rtsp_media_factory_uri_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_media_factory_uri_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRTSPMediaFactoryURI *factory = GST_RTSP_MEDIA_FACTORY_URI (object);
+  GstRTSPMediaFactoryURIPrivate *priv = factory->priv;
+
+  switch (propid) {
+    case PROP_URI:
+      g_value_take_string (value, gst_rtsp_media_factory_uri_get_uri (factory));
+      break;
+    case PROP_USE_GSTPAY:
+      g_value_set_boolean (value, priv->use_gstpay);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+static void
+gst_rtsp_media_factory_uri_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRTSPMediaFactoryURI *factory = GST_RTSP_MEDIA_FACTORY_URI (object);
+  GstRTSPMediaFactoryURIPrivate *priv = factory->priv;
+
+  switch (propid) {
+    case PROP_URI:
+      gst_rtsp_media_factory_uri_set_uri (factory, g_value_get_string (value));
+      break;
+    case PROP_USE_GSTPAY:
+      priv->use_gstpay = g_value_get_boolean (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+/**
+ * gst_rtsp_media_factory_uri_new:
+ *
+ * Create a new #GstRTSPMediaFactoryURI instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPMediaFactoryURI object.
+ */
+GstRTSPMediaFactoryURI *
+gst_rtsp_media_factory_uri_new (void)
+{
+  GstRTSPMediaFactoryURI *result;
+
+  result = g_object_new (GST_TYPE_RTSP_MEDIA_FACTORY_URI, NULL);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_media_factory_uri_set_uri:
+ * @factory: a #GstRTSPMediaFactory
+ * @uri: the uri the stream
+ *
+ * Set the URI of the resource that will be streamed by this factory.
+ */
+void
+gst_rtsp_media_factory_uri_set_uri (GstRTSPMediaFactoryURI * factory,
+    const gchar * uri)
+{
+  GstRTSPMediaFactoryURIPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY_URI (factory));
+  g_return_if_fail (uri != NULL);
+
+  priv = factory->priv;
+
+  g_mutex_lock (&priv->lock);
+  g_free (priv->uri);
+  priv->uri = g_strdup (uri);
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_factory_uri_get_uri:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the URI that will provide media for this factory.
+ *
+ * Returns: (transfer full): the configured URI. g_free() after usage.
+ */
+gchar *
+gst_rtsp_media_factory_uri_get_uri (GstRTSPMediaFactoryURI * factory)
+{
+  GstRTSPMediaFactoryURIPrivate *priv;
+  gchar *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY_URI (factory), NULL);
+
+  priv = factory->priv;
+
+  g_mutex_lock (&priv->lock);
+  result = g_strdup (priv->uri);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+static GstElementFactory *
+find_payloader (GstRTSPMediaFactoryURI * urifact, GstCaps * caps)
+{
+  GstRTSPMediaFactoryURIPrivate *priv = urifact->priv;
+  GList *list;
+  GstElementFactory *factory = NULL;
+  gboolean autoplug_more = FALSE;
+
+  /* first find a demuxer that can link */
+  list = gst_element_factory_list_filter (priv->demuxers, caps,
+      GST_PAD_SINK, FALSE);
+
+  if (list) {
+    GstStructure *structure = gst_caps_get_structure (caps, 0);
+    gboolean parsed = FALSE;
+    gint mpegversion = 0;
+
+    if (!gst_structure_get_boolean (structure, "parsed", &parsed) &&
+        gst_structure_has_name (structure, "audio/mpeg") &&
+        gst_structure_get_int (structure, "mpegversion", &mpegversion) &&
+        (mpegversion == 2 || mpegversion == 4)) {
+      /* for AAC it's framed=true instead of parsed=true */
+      gst_structure_get_boolean (structure, "framed", &parsed);
+    }
+
+    /* Avoid plugging parsers in a loop. This is not 100% correct, as some
+     * parsers don't set parsed=true in caps. We should do something like
+     * decodebin does and track decode chains and elements plugged in those
+     * chains...
+     */
+    if (parsed) {
+      GList *walk;
+      const gchar *klass;
+
+      for (walk = list; walk; walk = walk->next) {
+        factory = GST_ELEMENT_FACTORY (walk->data);
+        klass = gst_element_factory_get_metadata (factory,
+            GST_ELEMENT_METADATA_KLASS);
+        if (strstr (klass, "Parser"))
+          /* caps have parsed=true, so skip this parser to avoid loops */
+          continue;
+
+        autoplug_more = TRUE;
+        break;
+      }
+    } else {
+      /* caps don't have parsed=true set and we have a demuxer/parser */
+      autoplug_more = TRUE;
+    }
+
+    gst_plugin_feature_list_free (list);
+  }
+
+  if (autoplug_more)
+    /* we have a demuxer, try that one first */
+    return NULL;
+
+  /* no demuxer try a depayloader */
+  list = gst_element_factory_list_filter (priv->payloaders, caps,
+      GST_PAD_SINK, FALSE);
+
+  if (list == NULL) {
+    if (priv->use_gstpay) {
+      /* no depayloader or parser/demuxer, use gstpay when allowed */
+      factory = gst_element_factory_find ("rtpgstpay");
+    } else {
+      /* no depayloader, try a decoder, we'll get to a payloader for a decoded
+       * video or audio format, worst case. */
+      list = gst_element_factory_list_filter (priv->decoders, caps,
+          GST_PAD_SINK, FALSE);
+
+      if (list != NULL) {
+        /* we have a decoder, try that one first */
+        gst_plugin_feature_list_free (list);
+        return NULL;
+      }
+    }
+  }
+
+  if (list != NULL) {
+    factory = GST_ELEMENT_FACTORY_CAST (list->data);
+    g_object_ref (factory);
+    gst_plugin_feature_list_free (list);
+  }
+  return factory;
+}
+
+static gboolean
+autoplug_continue_cb (GstElement * uribin, GstPad * pad, GstCaps * caps,
+    GstElement * element)
+{
+  FactoryData *data;
+  GstElementFactory *factory;
+
+  GST_DEBUG ("found pad %s:%s of caps %" GST_PTR_FORMAT,
+      GST_DEBUG_PAD_NAME (pad), caps);
+
+  data = g_object_get_data (G_OBJECT (element), factory_key);
+
+  if (!(factory = find_payloader (data->factory, caps)))
+    goto no_factory;
+
+  /* we found a payloader, stop autoplugging so we can plug the
+   * payloader. */
+  GST_DEBUG ("found factory %s",
+      gst_plugin_feature_get_name (GST_PLUGIN_FEATURE (factory)));
+  gst_object_unref (factory);
+
+  return FALSE;
+
+  /* ERRORS */
+no_factory:
+  {
+    /* no payloader, continue autoplugging */
+    GST_DEBUG ("no payloader found");
+    return TRUE;
+  }
+}
+
+static void
+pad_added_cb (GstElement * uribin, GstPad * pad, GstElement * element)
+{
+  GstRTSPMediaFactoryURI *urifact;
+  GstRTSPMediaFactoryURIPrivate *priv;
+  FactoryData *data;
+  GstElementFactory *factory;
+  GstElement *payloader;
+  GstCaps *caps;
+  GstPad *sinkpad, *srcpad, *ghostpad;
+  GstElement *convert;
+  gchar *padname;
+
+  GST_DEBUG ("added pad %s:%s", GST_DEBUG_PAD_NAME (pad));
+
+  /* link the element now and expose the pad */
+  data = g_object_get_data (G_OBJECT (element), factory_key);
+  urifact = data->factory;
+  priv = urifact->priv;
+
+  /* ref to make refcounting easier later */
+  gst_object_ref (pad);
+  padname = gst_pad_get_name (pad);
+
+  /* get pad caps first, then call get_caps, then fail */
+  if ((caps = gst_pad_get_current_caps (pad)) == NULL)
+    if ((caps = gst_pad_query_caps (pad, NULL)) == NULL)
+      goto no_caps;
+
+  /* check for raw caps */
+  if (gst_caps_can_intersect (caps, priv->raw_vcaps)) {
+    /* we have raw video caps, insert converter */
+    convert = gst_element_factory_make ("videoconvert", NULL);
+  } else if (gst_caps_can_intersect (caps, priv->raw_acaps)) {
+    /* we have raw audio caps, insert converter */
+    convert = gst_element_factory_make ("audioconvert", NULL);
+  } else {
+    convert = NULL;
+  }
+
+  if (convert) {
+    gst_bin_add (GST_BIN_CAST (element), convert);
+    gst_element_set_state (convert, GST_STATE_PLAYING);
+
+    sinkpad = gst_element_get_static_pad (convert, "sink");
+    gst_pad_link (pad, sinkpad);
+    gst_object_unref (sinkpad);
+
+    /* unref old pad, we reffed before */
+    gst_object_unref (pad);
+
+    /* continue with new pad and caps */
+    pad = gst_element_get_static_pad (convert, "src");
+    if ((caps = gst_pad_get_current_caps (pad)) == NULL)
+      if ((caps = gst_pad_query_caps (pad, NULL)) == NULL)
+        goto no_caps;
+  }
+
+  if (!(factory = find_payloader (urifact, caps)))
+    goto no_factory;
+
+  gst_caps_unref (caps);
+
+  /* we have a payloader now */
+  GST_DEBUG ("found payloader factory %s",
+      gst_plugin_feature_get_name (GST_PLUGIN_FEATURE (factory)));
+
+  payloader = gst_element_factory_create (factory, NULL);
+  if (payloader == NULL)
+    goto no_payloader;
+
+  g_object_set (payloader, "pt", data->pt, NULL);
+  data->pt++;
+
+  if (g_object_class_find_property (G_OBJECT_GET_CLASS (payloader),
+          "buffer-list"))
+    g_object_set (payloader, "buffer-list", TRUE, NULL);
+
+  /* add the payloader to the pipeline */
+  gst_bin_add (GST_BIN_CAST (element), payloader);
+  gst_element_set_state (payloader, GST_STATE_PLAYING);
+
+  /* link the pad to the sinkpad of the payloader */
+  sinkpad = gst_element_get_static_pad (payloader, "sink");
+  gst_pad_link (pad, sinkpad);
+  gst_object_unref (sinkpad);
+  gst_object_unref (pad);
+
+  /* now expose the srcpad of the payloader as a ghostpad with the same name
+   * as the uridecodebin pad name. */
+  srcpad = gst_element_get_static_pad (payloader, "src");
+  ghostpad = gst_ghost_pad_new (padname, srcpad);
+  gst_object_unref (srcpad);
+  g_free (padname);
+
+  gst_pad_set_active (ghostpad, TRUE);
+  gst_element_add_pad (element, ghostpad);
+
+  return;
+
+  /* ERRORS */
+no_caps:
+  {
+    GST_WARNING ("could not get caps from pad");
+    g_free (padname);
+    gst_object_unref (pad);
+    return;
+  }
+no_factory:
+  {
+    GST_DEBUG ("no payloader found");
+    g_free (padname);
+    gst_caps_unref (caps);
+    gst_object_unref (pad);
+    return;
+  }
+no_payloader:
+  {
+    GST_ERROR ("could not create payloader from factory");
+    g_free (padname);
+    gst_caps_unref (caps);
+    gst_object_unref (pad);
+    return;
+  }
+}
+
+static void
+no_more_pads_cb (GstElement * uribin, GstElement * element)
+{
+  GST_DEBUG ("no-more-pads");
+  gst_element_no_more_pads (element);
+}
+
+static GstElement *
+rtsp_media_factory_uri_create_element (GstRTSPMediaFactory * factory,
+    const GstRTSPUrl * url)
+{
+  GstRTSPMediaFactoryURIPrivate *priv;
+  GstElement *topbin, *element, *uribin;
+  GstRTSPMediaFactoryURI *urifact;
+  FactoryData *data;
+
+  urifact = GST_RTSP_MEDIA_FACTORY_URI_CAST (factory);
+  priv = urifact->priv;
+
+  GST_LOG ("creating element");
+
+  topbin = gst_bin_new ("GstRTSPMediaFactoryURI");
+  g_assert (topbin != NULL);
+
+  /* our bin will dynamically expose payloaded pads */
+  element = gst_bin_new ("dynpay0");
+  g_assert (element != NULL);
+
+  uribin = gst_element_factory_make ("uridecodebin", "uribin");
+  if (uribin == NULL)
+    goto no_uridecodebin;
+
+  g_object_set (uribin, "uri", priv->uri, NULL);
+
+  /* keep factory data around */
+  data = g_new0 (FactoryData, 1);
+  data->factory = g_object_ref (factory);
+  data->pt = 96;
+
+  g_object_set_data_full (G_OBJECT (element), factory_key,
+      data, (GDestroyNotify) free_data);
+
+  /* connect to the signals */
+  g_signal_connect (uribin, "autoplug-continue",
+      (GCallback) autoplug_continue_cb, element);
+  g_signal_connect (uribin, "pad-added", (GCallback) pad_added_cb, element);
+  g_signal_connect (uribin, "no-more-pads", (GCallback) no_more_pads_cb,
+      element);
+
+  gst_bin_add (GST_BIN_CAST (element), uribin);
+  gst_bin_add (GST_BIN_CAST (topbin), element);
+
+  return topbin;
+
+no_uridecodebin:
+  {
+    g_critical ("can't create uridecodebin element");
+    gst_object_unref (element);
+    return NULL;
+  }
+}
diff --git a/gst/rtsp-server/rtsp-media-factory-uri.h b/gst/rtsp-server/rtsp-media-factory-uri.h
new file mode 100644
index 0000000..6dc3a40
--- /dev/null
+++ b/gst/rtsp-server/rtsp-media-factory-uri.h
@@ -0,0 +1,80 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include "rtsp-media-factory.h"
+
+#ifndef __GST_RTSP_MEDIA_FACTORY_URI_H__
+#define __GST_RTSP_MEDIA_FACTORY_URI_H__
+
+G_BEGIN_DECLS
+
+/* types for the media factory */
+#define GST_TYPE_RTSP_MEDIA_FACTORY_URI              (gst_rtsp_media_factory_uri_get_type ())
+#define GST_IS_RTSP_MEDIA_FACTORY_URI(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_URI))
+#define GST_IS_RTSP_MEDIA_FACTORY_URI_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA_FACTORY_URI))
+#define GST_RTSP_MEDIA_FACTORY_URI_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_URI, GstRTSPMediaFactoryURIClass))
+#define GST_RTSP_MEDIA_FACTORY_URI(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_URI, GstRTSPMediaFactoryURI))
+#define GST_RTSP_MEDIA_FACTORY_URI_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA_FACTORY_URI, GstRTSPMediaFactoryURIClass))
+#define GST_RTSP_MEDIA_FACTORY_URI_CAST(obj)         ((GstRTSPMediaFactoryURI*)(obj))
+#define GST_RTSP_MEDIA_FACTORY_URI_CLASS_CAST(klass) ((GstRTSPMediaFactoryURIClass*)(klass))
+
+typedef struct _GstRTSPMediaFactoryURI GstRTSPMediaFactoryURI;
+typedef struct _GstRTSPMediaFactoryURIClass GstRTSPMediaFactoryURIClass;
+typedef struct _GstRTSPMediaFactoryURIPrivate GstRTSPMediaFactoryURIPrivate;
+
+/**
+ * GstRTSPMediaFactoryURI:
+ *
+ * A media factory that creates a pipeline to play and uri.
+ */
+struct _GstRTSPMediaFactoryURI {
+  GstRTSPMediaFactory   parent;
+
+  /*< private >*/
+  GstRTSPMediaFactoryURIPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPMediaFactoryURIClass:
+ *
+ * The #GstRTSPMediaFactoryURI class structure.
+ */
+struct _GstRTSPMediaFactoryURIClass {
+  GstRTSPMediaFactoryClass  parent_class;
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+GType                 gst_rtsp_media_factory_uri_get_type   (void);
+
+/* creating the factory */
+GstRTSPMediaFactoryURI * gst_rtsp_media_factory_uri_new     (void);
+
+/* configuring the factory */
+void                  gst_rtsp_media_factory_uri_set_uri  (GstRTSPMediaFactoryURI *factory,
+                                                           const gchar *uri);
+gchar *               gst_rtsp_media_factory_uri_get_uri  (GstRTSPMediaFactoryURI *factory);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_MEDIA_FACTORY_URI_H__ */
diff --git a/gst/rtsp-server/rtsp-media-factory-wfd.c b/gst/rtsp-server/rtsp-media-factory-wfd.c
new file mode 100644
index 0000000..e083fa6
--- /dev/null
+++ b/gst/rtsp-server/rtsp-media-factory-wfd.c
@@ -0,0 +1,1111 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-media-factory
+ * @short_description: A factory for media pipelines
+ * @see_also: #GstRTSPMountPoints, #GstRTSPMedia
+ *
+ * The #GstRTSPMediaFactoryWFD is responsible for creating or recycling
+ * #GstRTSPMedia objects based on the passed URL.
+ *
+ * The default implementation of the object can create #GstRTSPMedia objects
+ * containing a pipeline created from a launch description set with
+ * gst_rtsp_media_factory_wfd_set_launch().
+ *
+ * Media from a factory can be shared by setting the shared flag with
+ * gst_rtsp_media_factory_wfd_set_shared(). When a factory is shared,
+ * gst_rtsp_media_factory_wfd_construct() will return the same #GstRTSPMedia when
+ * the url matches.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include 
+#include "rtsp-media-factory-wfd.h"
+#include "gstwfdmessage.h"
+
+#define GST_RTSP_MEDIA_FACTORY_WFD_GET_PRIVATE(obj)  \
+       (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_WFD, GstRTSPMediaFactoryWFDPrivate))
+
+#define GST_RTSP_MEDIA_FACTORY_WFD_GET_LOCK(f)       (&(GST_RTSP_MEDIA_FACTORY_WFD_CAST(f)->priv->lock))
+#define GST_RTSP_MEDIA_FACTORY_WFD_LOCK(f)           (g_mutex_lock(GST_RTSP_MEDIA_FACTORY_WFD_GET_LOCK(f)))
+#define GST_RTSP_MEDIA_FACTORY_WFD_UNLOCK(f)         (g_mutex_unlock(GST_RTSP_MEDIA_FACTORY_WFD_GET_LOCK(f)))
+
+struct _GstRTSPMediaFactoryWFDPrivate
+{
+  GMutex lock;
+  GstRTSPPermissions *permissions;
+  gchar *launch;
+  gboolean shared;
+  GstRTSPLowerTrans protocols;
+  guint buffer_size;
+  guint mtu_size;
+
+  guint8 videosrc_type;
+  guint8 video_codec;
+  gchar *video_encoder;
+  guint video_bitrate;
+  guint video_width;
+  guint video_height;
+  guint video_framerate;
+  guint video_enc_skip_inbuf_value;
+  GstElement *video_queue;
+
+  gchar *audio_device;
+  gchar *audio_encoder_aac;
+  gchar *audio_encoder_ac3;
+  guint8 audio_codec;
+  guint64 audio_latency_time;
+  guint64 audio_buffer_time;
+  gboolean audio_do_timestamp;
+  guint8 audio_channels;
+  guint8 audio_freq;
+  guint8 audio_bitrate;
+  GstElement *audio_queue;
+
+  guint64 video_resolution_supported;
+
+  gboolean dump_ts;
+};
+
+#define DEFAULT_LAUNCH          NULL
+#define DEFAULT_SHARED          FALSE
+#define DEFAULT_PROTOCOLS       GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
+                                        GST_RTSP_LOWER_TRANS_TCP
+#define DEFAULT_BUFFER_SIZE     0x80000
+
+enum
+{
+  PROP_0,
+  PROP_LAUNCH,
+  PROP_SHARED,
+  PROP_SUSPEND_MODE,
+  PROP_EOS_SHUTDOWN,
+  PROP_PROTOCOLS,
+  PROP_BUFFER_SIZE,
+  PROP_LAST
+};
+
+enum
+{
+  SIGNAL_MEDIA_CONSTRUCTED,
+  SIGNAL_MEDIA_CONFIGURE,
+  SIGNAL_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_wfd_debug);
+#define GST_CAT_DEFAULT rtsp_media_wfd_debug
+
+static void gst_rtsp_media_factory_wfd_get_property (GObject * object,
+    guint propid, GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_factory_wfd_set_property (GObject * object,
+    guint propid, const GValue * value, GParamSpec * pspec);
+
+static void gst_rtsp_media_factory_wfd_finalize (GObject * obj);
+
+
+static GstElement *rtsp_media_factory_wfd_create_element (GstRTSPMediaFactory *
+    factory, const GstRTSPUrl * url);
+static GstRTSPMedia *rtsp_media_factory_wfd_construct (GstRTSPMediaFactory *
+    factory, const GstRTSPUrl * url);
+
+G_DEFINE_TYPE (GstRTSPMediaFactoryWFD, gst_rtsp_media_factory_wfd,
+    GST_TYPE_RTSP_MEDIA_FACTORY);
+
+static void
+gst_rtsp_media_factory_wfd_class_init (GstRTSPMediaFactoryWFDClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstRTSPMediaFactoryClass *factory_class;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPMediaFactoryWFDPrivate));
+
+  gobject_class = G_OBJECT_CLASS (klass);
+  factory_class = GST_RTSP_MEDIA_FACTORY_CLASS (klass);
+
+  gobject_class->get_property = gst_rtsp_media_factory_wfd_get_property;
+  gobject_class->set_property = gst_rtsp_media_factory_wfd_set_property;
+  gobject_class->finalize = gst_rtsp_media_factory_wfd_finalize;
+
+  factory_class->construct = rtsp_media_factory_wfd_construct;
+  factory_class->create_element = rtsp_media_factory_wfd_create_element;
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_media_wfd_debug, "rtspmediafactorywfd", 0,
+      "GstRTSPMediaFactoryWFD");
+}
+
+void  gst_rtsp_media_factory_wfd_set (GstRTSPMediaFactoryWFD * factory,
+    guint8 videosrc_type, gchar *audio_device, guint64 audio_latency_time,
+    guint64 audio_buffer_time, gboolean audio_do_timestamp, guint mtu_size)
+{
+  GstRTSPMediaFactoryWFDPrivate *priv =
+      GST_RTSP_MEDIA_FACTORY_WFD_GET_PRIVATE (factory);
+  factory->priv = priv;
+
+  priv->videosrc_type = videosrc_type;
+  priv->audio_device = audio_device;
+  priv->audio_latency_time = audio_latency_time;
+  priv->audio_buffer_time = audio_buffer_time;
+  priv->audio_do_timestamp = audio_do_timestamp;
+  priv->mtu_size = mtu_size;
+}
+
+void  gst_rtsp_media_factory_wfd_set_encoders (GstRTSPMediaFactoryWFD * factory,
+    gchar *video_encoder, gchar *audio_encoder_aac, gchar *audio_encoder_ac3)
+{
+  GstRTSPMediaFactoryWFDPrivate *priv =
+      GST_RTSP_MEDIA_FACTORY_WFD_GET_PRIVATE (factory);
+  factory->priv = priv;
+
+  priv->video_encoder = video_encoder;
+  priv->audio_encoder_aac = audio_encoder_aac;
+  priv->audio_encoder_ac3 = audio_encoder_ac3;
+}
+
+void  gst_rtsp_media_factory_wfd_set_dump_ts (GstRTSPMediaFactoryWFD * factory,
+    gboolean dump_ts)
+{
+  GstRTSPMediaFactoryWFDPrivate *priv =
+      GST_RTSP_MEDIA_FACTORY_WFD_GET_PRIVATE (factory);
+  factory->priv = priv;
+
+  priv->dump_ts = dump_ts;
+}
+void gst_rtsp_media_factory_wfd_set_negotiated_resolution (GstRTSPMediaFactory *factory,
+   guint32 width, guint32 height)
+{
+  GstRTSPMediaFactoryWFD *factory_wfd = GST_RTSP_MEDIA_FACTORY_WFD (factory);
+  GstRTSPMediaFactoryWFDPrivate *priv = factory_wfd->priv;
+
+  priv->video_width = width;
+  priv->video_height = height;
+}
+void gst_rtsp_media_factory_wfd_set_audio_codec (GstRTSPMediaFactory *factory,
+   guint audio_codec)
+{
+  GstRTSPMediaFactoryWFD *factory_wfd = GST_RTSP_MEDIA_FACTORY_WFD (factory);
+  GstRTSPMediaFactoryWFDPrivate *priv = factory_wfd->priv;
+
+  priv->audio_codec = audio_codec;
+}
+
+static void
+gst_rtsp_media_factory_wfd_init (GstRTSPMediaFactoryWFD * factory)
+{
+  GstRTSPMediaFactoryWFDPrivate *priv =
+      GST_RTSP_MEDIA_FACTORY_WFD_GET_PRIVATE (factory);
+  factory->priv = priv;
+
+  priv->launch = g_strdup (DEFAULT_LAUNCH);
+  priv->shared = DEFAULT_SHARED;
+  priv->protocols = DEFAULT_PROTOCOLS;
+  priv->buffer_size = DEFAULT_BUFFER_SIZE;
+
+  //priv->videosrc_type = GST_WFD_VSRC_XIMAGESRC;
+  //priv->videosrc_type = GST_WFD_VSRC_XVIMAGESRC;
+  //priv->videosrc_type = GST_WFD_VSRC_CAMERASRC;
+  priv->videosrc_type = GST_WFD_VSRC_VIDEOTESTSRC;
+  priv->video_codec = GST_WFD_VIDEO_H264;
+  priv->video_encoder = g_strdup ("omxh264enc");
+  priv->video_bitrate = 200000;
+  priv->video_width = 640;
+  priv->video_height = 480;
+  priv->video_framerate = 30;
+  priv->video_enc_skip_inbuf_value = 5;
+
+  priv->audio_device = g_strdup ("alsa_output.1.analog-stereo.monitor");
+  priv->audio_codec = GST_WFD_AUDIO_AAC;
+  priv->audio_encoder_aac = g_strdup ("avenc_aac");
+  priv->audio_encoder_ac3 = g_strdup ("avenc_ac3");
+  priv->audio_latency_time = 10000;
+  priv->audio_buffer_time = 200000;
+  priv->audio_do_timestamp = FALSE;
+  priv->audio_channels = GST_WFD_CHANNEL_2;
+  priv->audio_freq = GST_WFD_FREQ_48000;
+
+  g_mutex_init (&priv->lock);
+}
+
+static void
+gst_rtsp_media_factory_wfd_finalize (GObject * obj)
+{
+  GstRTSPMediaFactoryWFD *factory = GST_RTSP_MEDIA_FACTORY_WFD (obj);
+  GstRTSPMediaFactoryWFDPrivate *priv = factory->priv;
+
+  if (priv->permissions)
+    gst_rtsp_permissions_unref (priv->permissions);
+  g_free (priv->launch);
+  g_mutex_clear (&priv->lock);
+
+  if (priv->audio_device)
+    g_free (priv->audio_device);
+  if (priv->audio_encoder_aac)
+    g_free (priv->audio_encoder_aac);
+  if (priv->audio_encoder_ac3)
+    g_free (priv->audio_encoder_ac3);
+
+  if (priv->video_encoder)
+    g_free (priv->video_encoder);
+
+  G_OBJECT_CLASS (gst_rtsp_media_factory_wfd_parent_class)->finalize (obj);
+}
+
+GstRTSPMediaFactoryWFD *
+gst_rtsp_media_factory_wfd_new (void)
+{
+  GstRTSPMediaFactoryWFD *result;
+
+  result = g_object_new (GST_TYPE_RTSP_MEDIA_FACTORY_WFD, NULL);
+
+  return result;
+}
+
+static void
+gst_rtsp_media_factory_wfd_get_property (GObject * object,
+    guint propid, GValue * value, GParamSpec * pspec)
+{
+  //GstRTSPMediaFactoryWFD *factory = GST_RTSP_MEDIA_FACTORY_WFD (object);
+
+  switch (propid) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+static void
+gst_rtsp_media_factory_wfd_set_property (GObject * object,
+    guint propid, const GValue * value, GParamSpec * pspec)
+{
+  //GstRTSPMediaFactoryWFD *factory = GST_RTSP_MEDIA_FACTORY_WFD (object);
+
+  switch (propid) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+static GstPadProbeReturn
+rtsp_media_wfd_dump_data (GstPad * pad, GstPadProbeInfo *info, gpointer u_data)
+{
+  guint8 *data;
+  gsize size;
+  FILE *f;
+  GstMapInfo mapinfo;
+
+  if (info->type == (GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_PUSH)) {
+    GstBuffer *buffer = gst_pad_probe_info_get_buffer (info);
+
+    gst_buffer_map (buffer, &mapinfo, GST_MAP_READ);
+    data = mapinfo.data;
+    size = gst_buffer_get_size (buffer);
+
+    f = fopen ("/root/probe.ts", "a");
+    if (f != NULL) {
+      fwrite (data, size, 1, f);
+      fclose (f);
+    }
+    gst_buffer_unmap (buffer, &mapinfo);
+  }
+
+  return GST_PAD_PROBE_OK;
+}
+
+static gboolean
+_rtsp_media_factory_wfd_create_audio_capture_bin (GstRTSPMediaFactoryWFD *
+    factory, GstBin * srcbin)
+{
+  GstElement *audiosrc = NULL;
+  GstElement *acaps = NULL;
+  GstElement *acaps2 = NULL;
+  GstElement *aenc = NULL;
+  GstElement *audio_convert = NULL;
+  GstElement *aqueue = NULL;
+  GstRTSPMediaFactoryWFDPrivate *priv = NULL;
+
+  guint channels = 0;
+  gboolean is_enc_req = TRUE;
+  guint freq = 0;
+  gchar *acodec = NULL;
+
+  priv = factory->priv;
+
+  /* create audio src element */
+  audiosrc = gst_element_factory_make ("pulsesrc", "audiosrc");
+  if (!audiosrc) {
+    GST_ERROR_OBJECT (factory, "failed to create audiosrc element");
+    goto create_error;
+  }
+
+  GST_INFO_OBJECT (factory, "audio device : %s", priv->audio_device);
+  GST_INFO_OBJECT (factory, "audio latency time  : %"G_GUINT64_FORMAT,
+      priv->audio_latency_time);
+  GST_INFO_OBJECT (factory, "audio_buffer_time  : %"G_GUINT64_FORMAT,
+      priv->audio_buffer_time);
+  GST_INFO_OBJECT (factory, "audio_do_timestamp  : %d",
+      priv->audio_do_timestamp);
+
+  g_object_set (audiosrc, "device", priv->audio_device, NULL);
+  g_object_set (audiosrc, "buffer-time", (gint64) priv->audio_buffer_time,
+      NULL);
+  g_object_set (audiosrc, "latency-time", (gint64) priv->audio_latency_time,
+      NULL);
+  g_object_set (audiosrc, "do-timestamp", (gboolean) priv->audio_do_timestamp,
+      NULL);
+  g_object_set (audiosrc, "provide-clock", (gboolean) FALSE, NULL);
+  g_object_set (audiosrc, "is-live", (gboolean) TRUE, NULL);
+
+  if (priv->audio_codec == GST_WFD_AUDIO_LPCM) {
+    /* To meet miracast certification */
+    gint64 block_size = 1920;
+    g_object_set (audiosrc, "blocksize", (gint64) block_size, NULL);
+
+    audio_convert = gst_element_factory_make ("capssetter", "audio_convert");
+    if (NULL == audio_convert) {
+      GST_ERROR_OBJECT (factory, "failed to create audio convert element");
+      goto create_error;
+    }
+    g_object_set (audio_convert, "caps", gst_caps_new_simple("audio/x-lpcm",
+              "width", G_TYPE_INT, 16,
+              "rate", G_TYPE_INT, 48000,
+              "channels", G_TYPE_INT, 2,
+              "dynamic_range", G_TYPE_INT, 0,
+              "emphasis", G_TYPE_BOOLEAN, FALSE,
+              "mute", G_TYPE_BOOLEAN, FALSE, NULL), NULL);
+    g_object_set (audio_convert, "join", (gboolean)FALSE, NULL);
+    g_object_set (audio_convert, "replace", (gboolean)TRUE, NULL);
+
+    acaps2 = gst_element_factory_make ("capsfilter", "audiocaps2");
+    if (NULL == acaps2) {
+      GST_ERROR_OBJECT (factory, "failed to create audio capsilfter element");
+      goto create_error;
+    }
+    /* In case of LPCM, uses big endian */
+        g_object_set (G_OBJECT (acaps2), "caps",
+            gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, "S16BE",
+                /* In case of LPCM, uses big endian */
+                "rate", G_TYPE_INT, 48000,
+                "channels", G_TYPE_INT, 2, NULL), NULL);
+  }
+
+  /* create audio caps element */
+  acaps = gst_element_factory_make ("capsfilter", "audiocaps");
+  if (NULL == acaps) {
+    GST_ERROR_OBJECT (factory, "failed to create audio capsilfter element");
+    goto create_error;
+  }
+
+  if (priv->audio_channels == GST_WFD_CHANNEL_2)
+    channels = 2;
+  else if (priv->audio_channels == GST_WFD_CHANNEL_4)
+    channels = 4;
+  else if (priv->audio_channels == GST_WFD_CHANNEL_6)
+    channels = 6;
+  else if (priv->audio_channels == GST_WFD_CHANNEL_8)
+    channels = 8;
+  else
+    channels = 2;
+
+  if (priv->audio_freq == GST_WFD_FREQ_44100)
+    freq = 44100;
+  else if (priv->audio_freq == GST_WFD_FREQ_48000)
+    freq = 48000;
+  else
+    freq = 44100;
+
+  if (priv->audio_codec == GST_WFD_AUDIO_LPCM) {
+    g_object_set (G_OBJECT (acaps), "caps",
+        gst_caps_new_simple ("audio/x-lpcm", "width", G_TYPE_INT, 16,
+            "rate", G_TYPE_INT, 48000,
+            "channels", G_TYPE_INT, 2,
+            "dynamic_range", G_TYPE_INT, 0,
+            "emphasis", G_TYPE_BOOLEAN, FALSE,
+            "mute", G_TYPE_BOOLEAN, FALSE, NULL), NULL);
+  } else if ((priv->audio_codec == GST_WFD_AUDIO_AAC)
+      || (priv->audio_codec == GST_WFD_AUDIO_AC3)) {
+    g_object_set (G_OBJECT (acaps), "caps", gst_caps_new_simple ("audio/x-raw",
+            "endianness", G_TYPE_INT, 1234, "signed", G_TYPE_BOOLEAN, TRUE,
+            "depth", G_TYPE_INT, 16, "rate", G_TYPE_INT, freq, "channels",
+            G_TYPE_INT, channels, NULL), NULL);
+  }
+
+  if (priv->audio_codec == GST_WFD_AUDIO_AAC) {
+    acodec = g_strdup (priv->audio_encoder_aac);
+    is_enc_req = TRUE;
+  } else if (priv->audio_codec == GST_WFD_AUDIO_AC3) {
+    acodec = g_strdup (priv->audio_encoder_ac3);
+    is_enc_req = TRUE;
+  } else if (priv->audio_codec == GST_WFD_AUDIO_LPCM) {
+    GST_DEBUG_OBJECT (factory, "No codec required, raw data will be sent");
+    is_enc_req = FALSE;
+  } else {
+    GST_ERROR_OBJECT (factory, "Yet to support other than H264 format");
+    goto create_error;
+  }
+
+  if (is_enc_req) {
+    aenc = gst_element_factory_make (acodec, "audioenc");
+    if (NULL == aenc) {
+      GST_ERROR_OBJECT (factory, "failed to create audio encoder element");
+      goto create_error;
+    }
+
+    g_object_set (aenc, "compliance", -2, NULL);
+    g_object_set (aenc, "tolerance", 400000000, NULL);
+    g_object_set (aenc, "bitrate", (guint) 128000, NULL);
+    g_object_set (aenc, "rate-control", 2, NULL);
+
+    aqueue = gst_element_factory_make ("queue", "audio-queue");
+    if (!aqueue) {
+      GST_ERROR_OBJECT (factory, "failed to create audio queue element");
+      goto create_error;
+    }
+
+    gst_bin_add_many (srcbin, audiosrc, acaps, aenc, aqueue, NULL);
+
+    if (!gst_element_link_many (audiosrc, acaps, aenc, aqueue, NULL)) {
+      GST_ERROR_OBJECT (factory, "Failed to link audio src elements...");
+      goto create_error;
+    }
+  } else {
+    aqueue = gst_element_factory_make ("queue", "audio-queue");
+    if (!aqueue) {
+      GST_ERROR_OBJECT (factory, "failed to create audio queue element");
+      goto create_error;
+    }
+
+    gst_bin_add_many (srcbin, audiosrc, acaps2, audio_convert, acaps, aqueue, NULL);
+
+    if (!gst_element_link_many (audiosrc, acaps2, audio_convert, acaps, aqueue, NULL)) {
+      GST_ERROR_OBJECT (factory, "Failed to link audio src elements...");
+      goto create_error;
+    }
+  }
+
+  priv->audio_queue = aqueue;
+  if (acodec) g_free (acodec);
+
+  return TRUE;
+
+create_error:
+  if (acodec) g_free (acodec);
+  return FALSE;
+}
+
+static gboolean
+_rtsp_media_factory_wfd_create_videotest_bin (GstRTSPMediaFactoryWFD * factory,
+    GstBin * srcbin)
+{
+  GstElement *videosrc = NULL;
+  GstElement *vcaps = NULL;
+  GstElement *videoconvert = NULL;
+  GstElement *venc_caps = NULL;
+  gchar *vcodec = NULL;
+  GstElement *venc = NULL;
+  GstElement *vparse = NULL;
+  GstElement *vqueue = NULL;
+  GstRTSPMediaFactoryWFDPrivate *priv = NULL;
+
+  priv = factory->priv;
+
+  GST_INFO_OBJECT (factory, "picked videotestsrc as video source");
+
+  videosrc = gst_element_factory_make ("videotestsrc", "videosrc");
+  if (NULL == videosrc) {
+    GST_ERROR_OBJECT (factory, "failed to create ximagesrc element");
+    goto create_error;
+  }
+
+  /* create video caps element */
+  vcaps = gst_element_factory_make ("capsfilter", "videocaps");
+  if (NULL == vcaps) {
+    GST_ERROR_OBJECT (factory, "failed to create video capsilfter element");
+    goto create_error;
+  }
+
+  g_object_set (G_OBJECT (vcaps), "caps",
+      gst_caps_new_simple ("video/x-raw",
+          "format", G_TYPE_STRING, "I420",
+          "width", G_TYPE_INT, priv->video_width,
+          "height", G_TYPE_INT, priv->video_height,
+          "framerate", GST_TYPE_FRACTION, priv->video_framerate, 1, NULL),
+      NULL);
+
+  /* create video convert element */
+  videoconvert = gst_element_factory_make ("videoconvert", "videoconvert");
+  if (NULL == videoconvert) {
+    GST_ERROR_OBJECT (factory, "failed to create video videoconvert element");
+    goto create_error;
+  }
+
+  venc_caps = gst_element_factory_make ("capsfilter", "venc_caps");
+  if (NULL == venc_caps) {
+    GST_ERROR_OBJECT (factory, "failed to create video capsilfter element");
+    goto create_error;
+  }
+
+  g_object_set (G_OBJECT (venc_caps), "caps",
+      gst_caps_new_simple ("video/x-raw",
+          "format", G_TYPE_STRING, "SN12",
+          "width", G_TYPE_INT, priv->video_width,
+          "height", G_TYPE_INT, priv->video_height,
+          "framerate", GST_TYPE_FRACTION, priv->video_framerate, 1, NULL),
+      NULL);
+
+  if (priv->video_codec == GST_WFD_VIDEO_H264)
+    vcodec = g_strdup (priv->video_encoder);
+  else {
+    GST_ERROR_OBJECT (factory, "Yet to support other than H264 format");
+    goto create_error;
+  }
+
+  venc = gst_element_factory_make (vcodec, "videoenc");
+  if (vcodec) g_free (vcodec);
+
+  if (!venc) {
+    GST_ERROR_OBJECT (factory, "failed to create video encoder element");
+    goto create_error;
+  }
+
+  g_object_set (venc, "aud", 0, NULL);
+  g_object_set (venc, "byte-stream", 1, NULL);
+  g_object_set (venc, "bitrate", 512, NULL);
+
+  vparse = gst_element_factory_make ("h264parse", "videoparse");
+  if (NULL == vparse) {
+    GST_ERROR_OBJECT (factory, "failed to create h264 parse element");
+    goto create_error;
+  }
+  g_object_set (vparse, "config-interval", 1, NULL);
+
+  vqueue = gst_element_factory_make ("queue", "video-queue");
+  if (!vqueue) {
+    GST_ERROR_OBJECT (factory, "failed to create video queue element");
+    goto create_error;
+  }
+
+  gst_bin_add_many (srcbin, videosrc, vcaps, videoconvert, venc_caps, venc, vparse, vqueue, NULL);
+  if (!gst_element_link_many (videosrc, vcaps, videoconvert, venc_caps, venc, vparse, vqueue, NULL)) {
+    GST_ERROR_OBJECT (factory, "Failed to link video src elements...");
+    goto create_error;
+  }
+
+  priv->video_queue = vqueue;
+
+  return TRUE;
+
+create_error:
+  return FALSE;
+}
+
+static gboolean
+_rtsp_media_factory_wfd_create_camera_capture_bin (GstRTSPMediaFactoryWFD *
+    factory, GstBin * srcbin)
+{
+  GstElement *videosrc = NULL;
+  GstElement *vcaps = NULL;
+  GstElement *venc = NULL;
+  GstElement *vparse = NULL;
+  GstElement *vqueue = NULL;
+  gchar *vcodec = NULL;
+  GstRTSPMediaFactoryWFDPrivate *priv = NULL;
+
+  priv = factory->priv;
+
+  videosrc = gst_element_factory_make ("camerasrc", "videosrc");
+  if (NULL == videosrc) {
+    GST_ERROR_OBJECT (factory, "failed to create camerasrc element");
+    goto create_error;
+  }
+
+  /* create video caps element */
+  vcaps = gst_element_factory_make ("capsfilter", "videocaps");
+  if (NULL == vcaps) {
+    GST_ERROR_OBJECT (factory, "failed to create video capsilfter element");
+    goto create_error;
+  }
+
+  GST_INFO_OBJECT (factory, "picked camerasrc as video source");
+  g_object_set (G_OBJECT (vcaps), "caps",
+      gst_caps_new_simple ("video/x-raw",
+          "width", G_TYPE_INT, priv->video_width,
+          "height", G_TYPE_INT, priv->video_height,
+          "format", G_TYPE_STRING, "SN12",
+          "framerate", GST_TYPE_FRACTION, priv->video_framerate, 1, NULL),
+      NULL);
+
+  if (priv->video_codec == GST_WFD_VIDEO_H264)
+    vcodec = g_strdup (priv->video_encoder);
+  else {
+    GST_ERROR_OBJECT (factory, "Yet to support other than H264 format");
+    goto create_error;
+  }
+
+  venc = gst_element_factory_make (vcodec, "videoenc");
+  if (!venc) {
+    GST_ERROR_OBJECT (factory, "failed to create video encoder element");
+    goto create_error;
+  }
+  if (vcodec) g_free (vcodec);
+
+  g_object_set (venc, "bitrate", priv->video_bitrate, NULL);
+  g_object_set (venc, "byte-stream", 1, NULL);
+  g_object_set (venc, "append-dci", 1, NULL);
+
+  vparse = gst_element_factory_make ("h264parse", "videoparse");
+  if (NULL == vparse) {
+    GST_ERROR_OBJECT (factory, "failed to create h264 parse element");
+    goto create_error;
+  }
+  g_object_set (vparse, "config-interval", 1, NULL);
+
+  vqueue = gst_element_factory_make ("queue", "video-queue");
+  if (!vqueue) {
+    GST_ERROR_OBJECT (factory, "failed to create video queue element");
+    goto create_error;
+  }
+
+  gst_bin_add_many (srcbin, videosrc, vcaps, venc, vparse, vqueue, NULL);
+
+  if (!gst_element_link_many (videosrc, vcaps, venc, vparse, vqueue, NULL)) {
+    GST_ERROR_OBJECT (factory, "Failed to link video src elements...");
+    goto create_error;
+  }
+
+  priv->video_queue = vqueue;
+
+  return TRUE;
+
+create_error:
+  return FALSE;
+}
+
+static gboolean
+_rtsp_media_factory_wfd_create_xcapture_bin (GstRTSPMediaFactoryWFD * factory,
+    GstBin * srcbin)
+{
+  GstElement *videosrc = NULL;
+  GstElement *vcaps = NULL;
+  GstElement *venc_caps = NULL;
+  GstElement *videoconvert = NULL, *videoscale = NULL;
+  gchar *vcodec = NULL;
+  GstElement *venc = NULL;
+  GstElement *vparse = NULL;
+  GstElement *vqueue = NULL;
+  GstRTSPMediaFactoryWFDPrivate *priv = NULL;
+
+  priv = factory->priv;
+
+  GST_INFO_OBJECT (factory, "picked ximagesrc as video source");
+
+  videosrc = gst_element_factory_make ("ximagesrc", "videosrc");
+  if (NULL == videosrc) {
+    GST_ERROR_OBJECT (factory, "failed to create ximagesrc element");
+    goto create_error;
+  }
+
+  videoscale = gst_element_factory_make ("videoscale", "videoscale");
+  if (NULL == videoscale) {
+    GST_ERROR_OBJECT (factory, "failed to create videoscale element");
+    goto create_error;
+  }
+
+  videoconvert = gst_element_factory_make ("videoconvert", "videoconvert");
+  if (NULL == videoconvert) {
+    GST_ERROR_OBJECT (factory, "failed to create videoconvert element");
+    goto create_error;
+  }
+
+  /* create video caps element */
+  vcaps = gst_element_factory_make ("capsfilter", "videocaps");
+  if (NULL == vcaps) {
+    GST_ERROR_OBJECT (factory, "failed to create video capsilfter element");
+    goto create_error;
+  }
+
+  g_object_set (G_OBJECT (vcaps), "caps",
+      gst_caps_new_simple ("video/x-raw",
+          "width", G_TYPE_INT, priv->video_width,
+          "height", G_TYPE_INT, priv->video_height,
+          "framerate", GST_TYPE_FRACTION, priv->video_framerate, 1, NULL),
+      NULL);
+
+  if (priv->video_codec == GST_WFD_VIDEO_H264)
+    vcodec = g_strdup (priv->video_encoder);
+  else {
+    GST_ERROR_OBJECT (factory, "Yet to support other than H264 format");
+    goto create_error;
+  }
+
+  venc = gst_element_factory_make (vcodec, "videoenc");
+  if (vcodec) g_free (vcodec);
+
+  if (!venc) {
+    GST_ERROR_OBJECT (factory, "failed to create video encoder element");
+    goto create_error;
+  }
+
+  g_object_set (venc, "aud", 0, NULL);
+  g_object_set (venc, "byte-stream", 1, NULL);
+  g_object_set (venc, "bitrate", 512, NULL);
+
+  venc_caps = gst_element_factory_make ("capsfilter", "venc_caps");
+  if (NULL == venc_caps) {
+    GST_ERROR_OBJECT (factory, "failed to create video capsilfter element");
+    goto create_error;
+  }
+
+  g_object_set (G_OBJECT (venc_caps), "caps",
+      gst_caps_new_simple ("video/x-h264",
+          "profile", G_TYPE_STRING, "baseline", NULL), NULL);
+
+  vparse = gst_element_factory_make ("h264parse", "videoparse");
+  if (NULL == vparse) {
+    GST_ERROR_OBJECT (factory, "failed to create h264 parse element");
+    goto create_error;
+  }
+  g_object_set (vparse, "config-interval", 1, NULL);
+
+  vqueue = gst_element_factory_make ("queue", "video-queue");
+  if (!vqueue) {
+    GST_ERROR_OBJECT (factory, "failed to create video queue element");
+    goto create_error;
+  }
+
+  gst_bin_add_many (srcbin, videosrc, videoscale, videoconvert, vcaps, venc,
+      venc_caps, vparse, vqueue, NULL);
+  if (!gst_element_link_many (videosrc, videoscale, videoconvert, vcaps, venc,
+          venc_caps, vparse, vqueue, NULL)) {
+    GST_ERROR_OBJECT (factory, "Failed to link video src elements...");
+    goto create_error;
+  }
+
+  priv->video_queue = vqueue;
+
+  return TRUE;
+
+create_error:
+  return FALSE;
+}
+
+static gboolean
+_rtsp_media_factory_wfd_create_xvcapture_bin (GstRTSPMediaFactoryWFD * factory,
+    GstBin * srcbin)
+{
+  GstElement *videosrc = NULL;
+  GstElement *vcaps = NULL;
+  gchar *vcodec = NULL;
+  GstElement *venc = NULL;
+  GstElement *vparse = NULL;
+  GstElement *vqueue = NULL;
+  GstRTSPMediaFactoryWFDPrivate *priv = NULL;
+
+  priv = factory->priv;
+
+  GST_INFO_OBJECT (factory, "picked xvimagesrc as video source");
+
+  videosrc = gst_element_factory_make ("xvimagesrc", "videosrc");
+  if (NULL == videosrc) {
+    GST_ERROR_OBJECT (factory, "failed to create xvimagesrc element");
+    goto create_error;
+  }
+
+  /* create video caps element */
+  vcaps = gst_element_factory_make ("capsfilter", "videocaps");
+  if (NULL == vcaps) {
+    GST_ERROR_OBJECT (factory, "failed to create video capsilfter element");
+    goto create_error;
+  }
+
+  g_object_set (G_OBJECT (vcaps), "caps",
+      gst_caps_new_simple ("video/x-raw",
+          "width", G_TYPE_INT, priv->video_width,
+          "height", G_TYPE_INT, priv->video_height,
+          "format", G_TYPE_STRING, "SN12",
+          "framerate", GST_TYPE_FRACTION, priv->video_framerate, 1, NULL),
+      NULL);
+
+  if (priv->video_codec == GST_WFD_VIDEO_H264) {
+    vcodec = g_strdup (priv->video_encoder);
+  } else {
+    GST_ERROR_OBJECT (factory, "Yet to support other than H264 format");
+    goto create_error;
+  }
+
+  venc = gst_element_factory_make (vcodec, "videoenc");
+  if (!venc) {
+    GST_ERROR_OBJECT (factory, "failed to create video encoder element");
+    goto create_error;
+  }
+  g_object_set (venc, "bitrate", priv->video_bitrate, NULL);
+  g_object_set (venc, "byte-stream", 1, NULL);
+  g_object_set (venc, "append-dci", 1, NULL);
+  g_object_set (venc, "idr-period", 120, NULL);
+  g_object_set (venc, "skip-inbuf", priv->video_enc_skip_inbuf_value, NULL);
+
+  vparse = gst_element_factory_make ("h264parse", "videoparse");
+  if (NULL == vparse) {
+    GST_ERROR_OBJECT (factory, "failed to create h264 parse element");
+    goto create_error;
+  }
+  g_object_set (vparse, "config-interval", 1, NULL);
+
+  vqueue = gst_element_factory_make ("queue", "video-queue");
+  if (!vqueue) {
+    GST_ERROR_OBJECT (factory, "failed to create video queue element");
+    goto create_error;
+  }
+
+  gst_bin_add_many (srcbin, videosrc, vcaps, venc, vparse, vqueue, NULL);
+  if (!gst_element_link_many (videosrc, vcaps, venc, vparse, vqueue, NULL)) {
+    GST_ERROR_OBJECT (factory, "Failed to link video src elements...");
+    goto create_error;
+  }
+
+  priv->video_queue = vqueue;
+  if (vcodec) g_free (vcodec);
+
+  return TRUE;
+
+create_error:
+  if (vcodec) g_free (vcodec);
+  return FALSE;
+}
+
+static GstElement *
+_rtsp_media_factory_wfd_create_srcbin (GstRTSPMediaFactoryWFD * factory)
+{
+  GstRTSPMediaFactoryWFDPrivate *priv = NULL;
+
+  GstBin *srcbin = NULL;
+  GstElement *mux = NULL;
+  GstElement *mux_queue = NULL;
+  GstElement *payload = NULL;
+  GstPad *srcpad = NULL;
+  GstPad *mux_vsinkpad = NULL;
+  GstPad *mux_asinkpad = NULL;
+
+  priv = factory->priv;
+
+  /* create source bin */
+  srcbin = GST_BIN (gst_bin_new ("srcbin"));
+  if (!srcbin) {
+    GST_ERROR_OBJECT (factory, "failed to create source bin...");
+    goto create_error;
+  }
+
+  /* create video src element */
+  switch (priv->videosrc_type) {
+    case GST_WFD_VSRC_XIMAGESRC:
+      if (!_rtsp_media_factory_wfd_create_xcapture_bin (factory, srcbin)) {
+        GST_ERROR_OBJECT (factory, "failed to create xcapture bin...");
+        goto create_error;
+      }
+      break;
+    case GST_WFD_VSRC_XVIMAGESRC:
+      if (!_rtsp_media_factory_wfd_create_xvcapture_bin (factory, srcbin)) {
+        GST_ERROR_OBJECT (factory, "failed to create xvcapture bin...");
+        goto create_error;
+      }
+      break;
+    case GST_WFD_VSRC_CAMERASRC:
+      if (!_rtsp_media_factory_wfd_create_camera_capture_bin (factory, srcbin)) {
+        GST_ERROR_OBJECT (factory, "failed to create camera capture bin...");
+        goto create_error;
+      }
+      break;
+    case GST_WFD_VSRC_VIDEOTESTSRC:
+      if (!_rtsp_media_factory_wfd_create_videotest_bin (factory, srcbin)) {
+        GST_ERROR_OBJECT (factory, "failed to create videotestsrc bin...");
+        goto create_error;
+      }
+      break;
+    default:
+      GST_ERROR_OBJECT (factory, "unknow mode selected...");
+      goto create_error;
+  }
+
+  mux = gst_element_factory_make ("mpegtsmux", "tsmux");
+  if (!mux) {
+    GST_ERROR_OBJECT (factory, "failed to create muxer element");
+    goto create_error;
+  }
+
+  g_object_set (mux, "wfd-mode", TRUE, NULL);
+
+  mux_queue = gst_element_factory_make ("queue", "muxer-queue");
+  if (!mux_queue) {
+    GST_ERROR_OBJECT (factory, "failed to create muxer-queue element");
+    goto create_error;
+  }
+
+  g_object_set (mux_queue, "max-size-buffers", 20000, NULL);
+
+  payload = gst_element_factory_make ("rtpmp2tpay", "pay0");
+  if (!payload) {
+    GST_ERROR_OBJECT (factory, "failed to create payload element");
+    goto create_error;
+  }
+
+  g_object_set (payload, "pt", 33, NULL);
+  g_object_set (payload, "mtu", priv->mtu_size, NULL);
+  g_object_set (payload, "rtp-flush", (gboolean) TRUE, NULL);
+
+  gst_bin_add_many (srcbin, mux, mux_queue, payload, NULL);
+
+  if (!gst_element_link_many (mux, mux_queue, payload, NULL)) {
+    GST_ERROR_OBJECT (factory, "Failed to link muxer & payload...");
+    goto create_error;
+  }
+
+  /* request video sink pad from muxer, which has elementary pid 0x1011 */
+  mux_vsinkpad = gst_element_get_request_pad (mux, "sink_4113");
+  if (!mux_vsinkpad) {
+    GST_ERROR_OBJECT (factory, "Failed to get sink pad from muxer...");
+    goto create_error;
+  }
+
+  /* request srcpad from video queue */
+  srcpad = gst_element_get_static_pad (priv->video_queue, "src");
+  if (!srcpad) {
+    GST_ERROR_OBJECT (factory, "Failed to get srcpad from video queue...");
+    goto create_error;
+  }
+
+  if (gst_pad_link (srcpad, mux_vsinkpad) != GST_PAD_LINK_OK) {
+    GST_ERROR_OBJECT (factory,
+        "Failed to link video queue src pad & muxer video sink pad...");
+    goto create_error;
+  }
+
+  gst_object_unref (mux_vsinkpad);
+  gst_object_unref (srcpad);
+  srcpad = NULL;
+
+  /* create audio source elements & add to pipeline */
+  if (!_rtsp_media_factory_wfd_create_audio_capture_bin (factory, srcbin))
+    goto create_error;
+
+  /* request audio sink pad from muxer, which has elementary pid 0x1100 */
+  mux_asinkpad = gst_element_get_request_pad (mux, "sink_4352");
+  if (!mux_asinkpad) {
+    GST_ERROR_OBJECT (factory, "Failed to get sinkpad from muxer...");
+    goto create_error;
+  }
+
+  /* request srcpad from audio queue */
+  srcpad = gst_element_get_static_pad (priv->audio_queue, "src");
+  if (!srcpad) {
+    GST_ERROR_OBJECT (factory, "Failed to get srcpad from audio queue...");
+    goto create_error;
+  }
+
+  /* link audio queue's srcpad & muxer sink pad */
+  if (gst_pad_link (srcpad, mux_asinkpad) != GST_PAD_LINK_OK) {
+    GST_ERROR_OBJECT (factory,
+        "Failed to link audio queue src pad & muxer audio sink pad...");
+    goto create_error;
+  }
+  gst_object_unref (mux_asinkpad);
+  gst_object_unref (srcpad);
+
+  if (priv->dump_ts)
+  {
+    GstPad *pad_probe = NULL;
+    pad_probe = gst_element_get_static_pad (mux, "src");
+
+    if (NULL == pad_probe) {
+      GST_INFO_OBJECT (factory, "pad for probe not created");
+    } else {
+      GST_INFO_OBJECT (factory, "pad for probe SUCCESSFUL");
+    }
+    gst_pad_add_probe (pad_probe, GST_PAD_PROBE_TYPE_BUFFER,
+        rtsp_media_wfd_dump_data, factory, NULL);
+  }
+
+  GST_DEBUG_OBJECT (factory, "successfully created source bin...");
+
+  return GST_ELEMENT_CAST (srcbin);
+
+create_error:
+  GST_ERROR_OBJECT (factory, "Failed to create pipeline");
+  return NULL;
+}
+
+static GstElement *
+rtsp_media_factory_wfd_create_element (GstRTSPMediaFactory * factory,
+    const GstRTSPUrl * url)
+{
+  GstRTSPMediaFactoryWFD *_factory = GST_RTSP_MEDIA_FACTORY_WFD_CAST (factory);
+  GstElement *element = NULL;
+
+  GST_RTSP_MEDIA_FACTORY_WFD_LOCK (factory);
+
+  element = _rtsp_media_factory_wfd_create_srcbin (_factory);
+
+  GST_RTSP_MEDIA_FACTORY_WFD_UNLOCK (factory);
+
+  return element;
+}
+
+static GstRTSPMedia *
+rtsp_media_factory_wfd_construct (GstRTSPMediaFactory * factory,
+    const GstRTSPUrl * url)
+{
+  GstRTSPMedia *media;
+  GstElement *element, *pipeline;
+  GstRTSPMediaFactoryClass *klass;
+
+  klass = GST_RTSP_MEDIA_FACTORY_GET_CLASS (factory);
+
+  if (!klass->create_pipeline)
+    goto no_create;
+
+  element = gst_rtsp_media_factory_create_element (factory, url);
+  if (element == NULL)
+    goto no_element;
+
+  /* create a new empty media */
+  media = gst_rtsp_media_new (element);
+
+  gst_rtsp_media_collect_streams (media);
+
+  pipeline = klass->create_pipeline (factory, media);
+  if (pipeline == NULL)
+    goto no_pipeline;
+
+  return media;
+
+  /* ERRORS */
+no_create:
+  {
+    g_critical ("no create_pipeline function");
+    return NULL;
+  }
+no_element:
+  {
+    g_critical ("could not create element");
+    return NULL;
+  }
+no_pipeline:
+  {
+    g_critical ("can't create pipeline");
+    g_object_unref (media);
+    return NULL;
+  }
+}
diff --git a/gst/rtsp-server/rtsp-media-factory-wfd.h b/gst/rtsp-server/rtsp-media-factory-wfd.h
new file mode 100644
index 0000000..5320d63
--- /dev/null
+++ b/gst/rtsp-server/rtsp-media-factory-wfd.h
@@ -0,0 +1,130 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include "rtsp-media-factory.h"
+
+#ifndef __GST_RTSP_MEDIA_FACTORY_WFD_H__
+#define __GST_RTSP_MEDIA_FACTORY_WFD_H__
+
+G_BEGIN_DECLS
+/* types for the media factory */
+#define GST_TYPE_RTSP_MEDIA_FACTORY_WFD              (gst_rtsp_media_factory_wfd_get_type ())
+#define GST_IS_RTSP_MEDIA_FACTORY_WFD(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_WFD))
+#define GST_IS_RTSP_MEDIA_FACTORY_WFD_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA_FACTORY_WFD))
+#define GST_RTSP_MEDIA_FACTORY_WFD_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_WFD, GstRTSPMediaFactoryWFDClass))
+#define GST_RTSP_MEDIA_FACTORY_WFD(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA_FACTORY_WFD, GstRTSPMediaFactoryWFD))
+#define GST_RTSP_MEDIA_FACTORY_WFD_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA_FACTORY_WFD, GstRTSPMediaFactoryWFDClass))
+#define GST_RTSP_MEDIA_FACTORY_WFD_CAST(obj)         ((GstRTSPMediaFactoryWFD*)(obj))
+#define GST_RTSP_MEDIA_FACTORY_WFD_CLASS_CAST(klass) ((GstRTSPMediaFactoryWFDClass*)(klass))
+    enum
+{
+  GST_WFD_VSRC_XIMAGESRC,
+  GST_WFD_VSRC_XVIMAGESRC,
+  GST_WFD_VSRC_CAMERASRC,
+  GST_WFD_VSRC_VIDEOTESTSRC
+};
+
+typedef struct _GstRTSPMediaFactoryWFD GstRTSPMediaFactoryWFD;
+typedef struct _GstRTSPMediaFactoryWFDClass GstRTSPMediaFactoryWFDClass;
+typedef struct _GstRTSPMediaFactoryWFDPrivate GstRTSPMediaFactoryWFDPrivate;
+
+/**
+ * GstRTSPMediaFactoryWFD:
+ *
+ * The definition and logic for constructing the pipeline for a media. The media
+ * can contain multiple streams like audio and video.
+ */
+struct _GstRTSPMediaFactoryWFD
+{
+  GstRTSPMediaFactory parent;
+
+  /*< private > */
+  GstRTSPMediaFactoryWFDPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPMediaFactoryWFDClass:
+ * @gen_key: convert @url to a key for caching shared #GstRTSPMedia objects.
+ *       The default implementation of this function will use the complete URL
+ *       including the query parameters to return a key.
+ * @create_element: Construct and return a #GstElement that is a #GstBin containing
+ *       the elements to use for streaming the media. The bin should contain
+ *       payloaders pay\%d for each stream. The default implementation of this
+ *       function returns the bin created from the launch parameter.
+ * @construct: the vmethod that will be called when the factory has to create the
+ *       #GstRTSPMedia for @url. The default implementation of this
+ *       function calls create_element to retrieve an element and then looks for
+ *       pay\%d to create the streams.
+ * @create_pipeline: create a new pipeline or re-use an existing one and
+ *       add the #GstRTSPMedia's element created by @construct to the pipeline.
+ * @configure: configure the media created with @construct. The default
+ *       implementation will configure the 'shared' property of the media.
+ * @media_constructed: signal emited when a media was constructed
+ * @media_configure: signal emited when a media should be configured
+ *
+ * The #GstRTSPMediaFactoryWFD class structure.
+ */
+struct _GstRTSPMediaFactoryWFDClass
+{
+  GstRTSPMediaFactoryClass parent_class;
+
+  gchar *(*gen_key) (GstRTSPMediaFactoryWFD * factory, const GstRTSPUrl * url);
+
+  GstElement *(*create_element) (GstRTSPMediaFactoryWFD * factory,
+      const GstRTSPUrl * url);
+  GstRTSPMedia *(*construct) (GstRTSPMediaFactoryWFD * factory,
+      const GstRTSPUrl * url);
+  GstElement *(*create_pipeline) (GstRTSPMediaFactoryWFD * factory,
+      GstRTSPMedia * media);
+  void (*configure) (GstRTSPMediaFactoryWFD * factory, GstRTSPMedia * media);
+
+  /* signals */
+  void (*media_constructed) (GstRTSPMediaFactoryWFD * factory,
+      GstRTSPMedia * media);
+  void (*media_configure) (GstRTSPMediaFactoryWFD * factory,
+      GstRTSPMedia * media);
+
+  /*< private > */
+  gpointer _gst_reserved[GST_PADDING_LARGE];
+};
+
+GType gst_rtsp_media_factory_wfd_get_type (void);
+
+/* creating the factory */
+GstRTSPMediaFactoryWFD *gst_rtsp_media_factory_wfd_new (void);
+GstElement *gst_rtsp_media_factory_wfd_create_element (GstRTSPMediaFactoryWFD *
+    factory, const GstRTSPUrl * url);
+
+void  gst_rtsp_media_factory_wfd_set (GstRTSPMediaFactoryWFD * factory,
+    guint8 videosrc_type, gchar *audio_device, guint64 audio_latency_time,
+    guint64 audio_buffer_time, gboolean audio_do_timestamp, guint mtu_size);
+void  gst_rtsp_media_factory_wfd_set_encoders (GstRTSPMediaFactoryWFD * factory,
+    gchar *video_encoder, gchar *audio_encoder_aac, gchar *audio_encoder_ac3);
+void  gst_rtsp_media_factory_wfd_set_dump_ts (GstRTSPMediaFactoryWFD * factory,
+    gboolean dump_ts);
+void gst_rtsp_media_factory_wfd_set_negotiated_resolution (GstRTSPMediaFactory *factory,
+   guint32 width, guint32 height);
+void gst_rtsp_media_factory_wfd_set_audio_codec (GstRTSPMediaFactory *factory,
+    guint audio_codec);
+
+G_END_DECLS
+#endif /* __GST_RTSP_MEDIA_FACTORY_WFD_H__ */
diff --git a/gst/rtsp-server/rtsp-media-factory.c b/gst/rtsp-server/rtsp-media-factory.c
new file mode 100644
index 0000000..7d85dcc
--- /dev/null
+++ b/gst/rtsp-server/rtsp-media-factory.c
@@ -0,0 +1,1147 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-media-factory
+ * @short_description: A factory for media pipelines
+ * @see_also: #GstRTSPMountPoints, #GstRTSPMedia
+ *
+ * The #GstRTSPMediaFactory is responsible for creating or recycling
+ * #GstRTSPMedia objects based on the passed URL.
+ *
+ * The default implementation of the object can create #GstRTSPMedia objects
+ * containing a pipeline created from a launch description set with
+ * gst_rtsp_media_factory_set_launch().
+ *
+ * Media from a factory can be shared by setting the shared flag with
+ * gst_rtsp_media_factory_set_shared(). When a factory is shared,
+ * gst_rtsp_media_factory_construct() will return the same #GstRTSPMedia when
+ * the url matches.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include "rtsp-media-factory.h"
+
+#define GST_RTSP_MEDIA_FACTORY_GET_PRIVATE(obj)  \
+       (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MEDIA_FACTORY, GstRTSPMediaFactoryPrivate))
+
+#define GST_RTSP_MEDIA_FACTORY_GET_LOCK(f)       (&(GST_RTSP_MEDIA_FACTORY_CAST(f)->priv->lock))
+#define GST_RTSP_MEDIA_FACTORY_LOCK(f)           (g_mutex_lock(GST_RTSP_MEDIA_FACTORY_GET_LOCK(f)))
+#define GST_RTSP_MEDIA_FACTORY_UNLOCK(f)         (g_mutex_unlock(GST_RTSP_MEDIA_FACTORY_GET_LOCK(f)))
+
+struct _GstRTSPMediaFactoryPrivate
+{
+  GMutex lock;                  /* protects everything but medias */
+  GstRTSPPermissions *permissions;
+  gchar *launch;
+  gboolean shared;
+  GstRTSPSuspendMode suspend_mode;
+  gboolean eos_shutdown;
+  GstRTSPProfile profiles;
+  GstRTSPLowerTrans protocols;
+  guint buffer_size;
+  GstRTSPAddressPool *pool;
+
+  GMutex medias_lock;
+  GHashTable *medias;           /* protected by medias_lock */
+};
+
+#define DEFAULT_LAUNCH          NULL
+#define DEFAULT_SHARED          FALSE
+#define DEFAULT_SUSPEND_MODE    GST_RTSP_SUSPEND_MODE_NONE
+#define DEFAULT_EOS_SHUTDOWN    FALSE
+#define DEFAULT_PROFILES        GST_RTSP_PROFILE_AVP
+#define DEFAULT_PROTOCOLS       GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
+                                        GST_RTSP_LOWER_TRANS_TCP
+#define DEFAULT_BUFFER_SIZE     0x80000
+
+enum
+{
+  PROP_0,
+  PROP_LAUNCH,
+  PROP_SHARED,
+  PROP_SUSPEND_MODE,
+  PROP_EOS_SHUTDOWN,
+  PROP_PROFILES,
+  PROP_PROTOCOLS,
+  PROP_BUFFER_SIZE,
+  PROP_LAST
+};
+
+enum
+{
+  SIGNAL_MEDIA_CONSTRUCTED,
+  SIGNAL_MEDIA_CONFIGURE,
+  SIGNAL_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
+#define GST_CAT_DEFAULT rtsp_media_debug
+
+static guint gst_rtsp_media_factory_signals[SIGNAL_LAST] = { 0 };
+
+static void gst_rtsp_media_factory_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_factory_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_factory_finalize (GObject * obj);
+
+static gchar *default_gen_key (GstRTSPMediaFactory * factory,
+    const GstRTSPUrl * url);
+static GstElement *default_create_element (GstRTSPMediaFactory * factory,
+    const GstRTSPUrl * url);
+static GstRTSPMedia *default_construct (GstRTSPMediaFactory * factory,
+    const GstRTSPUrl * url);
+static void default_configure (GstRTSPMediaFactory * factory,
+    GstRTSPMedia * media);
+static GstElement *default_create_pipeline (GstRTSPMediaFactory * factory,
+    GstRTSPMedia * media);
+
+G_DEFINE_TYPE (GstRTSPMediaFactory, gst_rtsp_media_factory, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_media_factory_class_init (GstRTSPMediaFactoryClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPMediaFactoryPrivate));
+
+  gobject_class = G_OBJECT_CLASS (klass);
+
+  gobject_class->get_property = gst_rtsp_media_factory_get_property;
+  gobject_class->set_property = gst_rtsp_media_factory_set_property;
+  gobject_class->finalize = gst_rtsp_media_factory_finalize;
+
+  /**
+   * GstRTSPMediaFactory::launch:
+   *
+   * The gst_parse_launch() line to use for constructing the pipeline in the
+   * default prepare vmethod.
+   *
+   * The pipeline description should return a GstBin as the toplevel element
+   * which can be accomplished by enclosing the dscription with brackets '('
+   * ')'.
+   *
+   * The description should return a pipeline with payloaders named pay0, pay1,
+   * etc.. Each of the payloaders will result in a stream.
+   *
+   * Support for dynamic payloaders can be accomplished by adding payloaders
+   * named dynpay0, dynpay1, etc..
+   */
+  g_object_class_install_property (gobject_class, PROP_LAUNCH,
+      g_param_spec_string ("launch", "Launch",
+          "A launch description of the pipeline", DEFAULT_LAUNCH,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_SHARED,
+      g_param_spec_boolean ("shared", "Shared",
+          "If media from this factory is shared", DEFAULT_SHARED,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
+      g_param_spec_enum ("suspend-mode", "Suspend Mode",
+          "Control how media will be suspended", GST_TYPE_RTSP_SUSPEND_MODE,
+          DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
+      g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
+          "Send EOS down the pipeline before shutting down",
+          DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_PROFILES,
+      g_param_spec_flags ("profiles", "Profiles",
+          "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
+          DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
+      g_param_spec_flags ("protocols", "Protocols",
+          "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
+          DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
+      g_param_spec_uint ("buffer-size", "Buffer Size",
+          "The kernel UDP buffer size to use", 0, G_MAXUINT,
+          DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  gst_rtsp_media_factory_signals[SIGNAL_MEDIA_CONSTRUCTED] =
+      g_signal_new ("media-constructed", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaFactoryClass,
+          media_constructed), NULL, NULL, g_cclosure_marshal_generic,
+      G_TYPE_NONE, 1, GST_TYPE_RTSP_MEDIA);
+
+  gst_rtsp_media_factory_signals[SIGNAL_MEDIA_CONFIGURE] =
+      g_signal_new ("media-configure", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaFactoryClass,
+          media_configure), NULL, NULL, g_cclosure_marshal_generic,
+      G_TYPE_NONE, 1, GST_TYPE_RTSP_MEDIA);
+
+  klass->gen_key = default_gen_key;
+  klass->create_element = default_create_element;
+  klass->construct = default_construct;
+  klass->configure = default_configure;
+  klass->create_pipeline = default_create_pipeline;
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmediafactory", 0,
+      "GstRTSPMediaFactory");
+}
+
+static void
+gst_rtsp_media_factory_init (GstRTSPMediaFactory * factory)
+{
+  GstRTSPMediaFactoryPrivate *priv =
+      GST_RTSP_MEDIA_FACTORY_GET_PRIVATE (factory);
+  factory->priv = priv;
+
+  priv->launch = g_strdup (DEFAULT_LAUNCH);
+  priv->shared = DEFAULT_SHARED;
+  priv->suspend_mode = DEFAULT_SUSPEND_MODE;
+  priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
+  priv->profiles = DEFAULT_PROFILES;
+  priv->protocols = DEFAULT_PROTOCOLS;
+  priv->buffer_size = DEFAULT_BUFFER_SIZE;
+
+  g_mutex_init (&priv->lock);
+  g_mutex_init (&priv->medias_lock);
+  priv->medias = g_hash_table_new_full (g_str_hash, g_str_equal,
+      g_free, g_object_unref);
+}
+
+static void
+gst_rtsp_media_factory_finalize (GObject * obj)
+{
+  GstRTSPMediaFactory *factory = GST_RTSP_MEDIA_FACTORY (obj);
+  GstRTSPMediaFactoryPrivate *priv = factory->priv;
+
+  if (priv->permissions)
+    gst_rtsp_permissions_unref (priv->permissions);
+  g_hash_table_unref (priv->medias);
+  g_mutex_clear (&priv->medias_lock);
+  g_free (priv->launch);
+  g_mutex_clear (&priv->lock);
+  if (priv->pool)
+    g_object_unref (priv->pool);
+
+  G_OBJECT_CLASS (gst_rtsp_media_factory_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_media_factory_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRTSPMediaFactory *factory = GST_RTSP_MEDIA_FACTORY (object);
+
+  switch (propid) {
+    case PROP_LAUNCH:
+      g_value_take_string (value, gst_rtsp_media_factory_get_launch (factory));
+      break;
+    case PROP_SHARED:
+      g_value_set_boolean (value, gst_rtsp_media_factory_is_shared (factory));
+      break;
+    case PROP_SUSPEND_MODE:
+      g_value_set_enum (value,
+          gst_rtsp_media_factory_get_suspend_mode (factory));
+      break;
+    case PROP_EOS_SHUTDOWN:
+      g_value_set_boolean (value,
+          gst_rtsp_media_factory_is_eos_shutdown (factory));
+      break;
+    case PROP_PROFILES:
+      g_value_set_flags (value, gst_rtsp_media_factory_get_profiles (factory));
+      break;
+    case PROP_PROTOCOLS:
+      g_value_set_flags (value, gst_rtsp_media_factory_get_protocols (factory));
+      break;
+    case PROP_BUFFER_SIZE:
+      g_value_set_uint (value,
+          gst_rtsp_media_factory_get_buffer_size (factory));
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+static void
+gst_rtsp_media_factory_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRTSPMediaFactory *factory = GST_RTSP_MEDIA_FACTORY (object);
+
+  switch (propid) {
+    case PROP_LAUNCH:
+      gst_rtsp_media_factory_set_launch (factory, g_value_get_string (value));
+      break;
+    case PROP_SHARED:
+      gst_rtsp_media_factory_set_shared (factory, g_value_get_boolean (value));
+      break;
+    case PROP_SUSPEND_MODE:
+      gst_rtsp_media_factory_set_suspend_mode (factory,
+          g_value_get_enum (value));
+      break;
+    case PROP_EOS_SHUTDOWN:
+      gst_rtsp_media_factory_set_eos_shutdown (factory,
+          g_value_get_boolean (value));
+      break;
+    case PROP_PROFILES:
+      gst_rtsp_media_factory_set_profiles (factory, g_value_get_flags (value));
+      break;
+    case PROP_PROTOCOLS:
+      gst_rtsp_media_factory_set_protocols (factory, g_value_get_flags (value));
+      break;
+    case PROP_BUFFER_SIZE:
+      gst_rtsp_media_factory_set_buffer_size (factory,
+          g_value_get_uint (value));
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+/**
+ * gst_rtsp_media_factory_new:
+ *
+ * Create a new #GstRTSPMediaFactory instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPMediaFactory object.
+ */
+GstRTSPMediaFactory *
+gst_rtsp_media_factory_new (void)
+{
+  GstRTSPMediaFactory *result;
+
+  result = g_object_new (GST_TYPE_RTSP_MEDIA_FACTORY, NULL);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_permissions:
+ * @factory: a #GstRTSPMediaFactory
+ * @permissions: (transfer none): a #GstRTSPPermissions
+ *
+ * Set @permissions on @factory.
+ */
+void
+gst_rtsp_media_factory_set_permissions (GstRTSPMediaFactory * factory,
+    GstRTSPPermissions * permissions)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  if (priv->permissions)
+    gst_rtsp_permissions_unref (priv->permissions);
+  if ((priv->permissions = permissions))
+    gst_rtsp_permissions_ref (permissions);
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_permissions:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the permissions object from @factory.
+ *
+ * Returns: (transfer full): a #GstRTSPPermissions object, unref after usage.
+ */
+GstRTSPPermissions *
+gst_rtsp_media_factory_get_permissions (GstRTSPMediaFactory * factory)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+  GstRTSPPermissions *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  if ((result = priv->permissions))
+    gst_rtsp_permissions_ref (result);
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_media_factory_add_role:
+ * @factory: a #GstRTSPMediaFactory
+ * @role: a role
+ * @fieldname: the first field name
+ * @...: additional arguments
+ *
+ * A convenience method to add @role with @fieldname and additional arguments to
+ * the permissions of @factory. If @factory had no permissions, new permissions
+ * will be created and the role will be added to it.
+ */
+void
+gst_rtsp_media_factory_add_role (GstRTSPMediaFactory * factory,
+    const gchar * role, const gchar * fieldname, ...)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+  va_list var_args;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+  g_return_if_fail (role != NULL);
+  g_return_if_fail (fieldname != NULL);
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  if (priv->permissions == NULL)
+    priv->permissions = gst_rtsp_permissions_new ();
+
+  va_start (var_args, fieldname);
+  gst_rtsp_permissions_add_role_valist (priv->permissions, role, fieldname,
+      var_args);
+  va_end (var_args);
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_set_launch:
+ * @factory: a #GstRTSPMediaFactory
+ * @launch: the launch description
+ *
+ *
+ * The gst_parse_launch() line to use for constructing the pipeline in the
+ * default prepare vmethod.
+ *
+ * The pipeline description should return a GstBin as the toplevel element
+ * which can be accomplished by enclosing the dscription with brackets '('
+ * ')'.
+ *
+ * The description should return a pipeline with payloaders named pay0, pay1,
+ * etc.. Each of the payloaders will result in a stream.
+ */
+void
+gst_rtsp_media_factory_set_launch (GstRTSPMediaFactory * factory,
+    const gchar * launch)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+  g_return_if_fail (launch != NULL);
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  g_free (priv->launch);
+  priv->launch = g_strdup (launch);
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_launch:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the gst_parse_launch() pipeline description that will be used in the
+ * default prepare vmethod.
+ *
+ * Returns: (transfer full): the configured launch description. g_free() after
+ * usage.
+ */
+gchar *
+gst_rtsp_media_factory_get_launch (GstRTSPMediaFactory * factory)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+  gchar *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  result = g_strdup (priv->launch);
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_suspend_mode:
+ * @factory: a #GstRTSPMediaFactory
+ * @mode: the new #GstRTSPSuspendMode
+ *
+ * Configure how media created from this factory will be suspended.
+ */
+void
+gst_rtsp_media_factory_set_suspend_mode (GstRTSPMediaFactory * factory,
+    GstRTSPSuspendMode mode)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  priv->suspend_mode = mode;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_suspend_mode:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get how media created from this factory will be suspended.
+ *
+ * Returns: a #GstRTSPSuspendMode.
+ */
+GstRTSPSuspendMode
+gst_rtsp_media_factory_get_suspend_mode (GstRTSPMediaFactory * factory)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+  GstRTSPSuspendMode result;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory),
+      GST_RTSP_SUSPEND_MODE_NONE);
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  result = priv->suspend_mode;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_shared:
+ * @factory: a #GstRTSPMediaFactory
+ * @shared: the new value
+ *
+ * Configure if media created from this factory can be shared between clients.
+ */
+void
+gst_rtsp_media_factory_set_shared (GstRTSPMediaFactory * factory,
+    gboolean shared)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  priv->shared = shared;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_is_shared:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get if media created from this factory can be shared between clients.
+ *
+ * Returns: %TRUE if the media will be shared between clients.
+ */
+gboolean
+gst_rtsp_media_factory_is_shared (GstRTSPMediaFactory * factory)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+  gboolean result;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), FALSE);
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  result = priv->shared;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_eos_shutdown:
+ * @factory: a #GstRTSPMediaFactory
+ * @eos_shutdown: the new value
+ *
+ * Configure if media created from this factory will have an EOS sent to the
+ * pipeline before shutdown.
+ */
+void
+gst_rtsp_media_factory_set_eos_shutdown (GstRTSPMediaFactory * factory,
+    gboolean eos_shutdown)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  priv->eos_shutdown = eos_shutdown;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_is_eos_shutdown:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get if media created from this factory will have an EOS event sent to the
+ * pipeline before shutdown.
+ *
+ * Returns: %TRUE if the media will receive EOS before shutdown.
+ */
+gboolean
+gst_rtsp_media_factory_is_eos_shutdown (GstRTSPMediaFactory * factory)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+  gboolean result;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), FALSE);
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  result = priv->eos_shutdown;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_buffer_size:
+ * @factory: a #GstRTSPMedia
+ * @size: the new value
+ *
+ * Set the kernel UDP buffer size.
+ */
+void
+gst_rtsp_media_factory_set_buffer_size (GstRTSPMediaFactory * factory,
+    guint size)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  priv->buffer_size = size;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_buffer_size:
+ * @factory: a #GstRTSPMedia
+ *
+ * Get the kernel UDP buffer size.
+ *
+ * Returns: the kernel UDP buffer size.
+ */
+guint
+gst_rtsp_media_factory_get_buffer_size (GstRTSPMediaFactory * factory)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+  guint result;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), 0);
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  result = priv->buffer_size;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_address_pool:
+ * @factory: a #GstRTSPMediaFactory
+ * @pool: (transfer none): a #GstRTSPAddressPool
+ *
+ * configure @pool to be used as the address pool of @factory.
+ */
+void
+gst_rtsp_media_factory_set_address_pool (GstRTSPMediaFactory * factory,
+    GstRTSPAddressPool * pool)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+  GstRTSPAddressPool *old;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  if ((old = priv->pool) != pool)
+    priv->pool = pool ? g_object_ref (pool) : NULL;
+  else
+    old = NULL;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+  if (old)
+    g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_media_factory_get_address_pool:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the #GstRTSPAddressPool used as the address pool of @factory.
+ *
+ * Returns: (transfer full): the #GstRTSPAddressPool of @factory. g_object_unref() after
+ * usage.
+ */
+GstRTSPAddressPool *
+gst_rtsp_media_factory_get_address_pool (GstRTSPMediaFactory * factory)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+  GstRTSPAddressPool *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  if ((result = priv->pool))
+    g_object_ref (result);
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_media_factory_set_profiles:
+ * @factory: a #GstRTSPMediaFactory
+ * @profiles: the new flags
+ *
+ * Configure the allowed profiles for @factory.
+ */
+void
+gst_rtsp_media_factory_set_profiles (GstRTSPMediaFactory * factory,
+    GstRTSPProfile profiles)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+  priv = factory->priv;
+
+  GST_DEBUG_OBJECT (factory, "profiles %d", profiles);
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  priv->profiles = profiles;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_profiles:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the allowed profiles of @factory.
+ *
+ * Returns: a #GstRTSPProfile
+ */
+GstRTSPProfile
+gst_rtsp_media_factory_get_profiles (GstRTSPMediaFactory * factory)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+  GstRTSPProfile res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory),
+      GST_RTSP_PROFILE_UNKNOWN);
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  res = priv->profiles;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_media_factory_set_protocols:
+ * @factory: a #GstRTSPMediaFactory
+ * @protocols: the new flags
+ *
+ * Configure the allowed lower transport for @factory.
+ */
+void
+gst_rtsp_media_factory_set_protocols (GstRTSPMediaFactory * factory,
+    GstRTSPLowerTrans protocols)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+
+  priv = factory->priv;
+
+  GST_DEBUG_OBJECT (factory, "protocols %d", protocols);
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  priv->protocols = protocols;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+}
+
+/**
+ * gst_rtsp_media_factory_get_protocols:
+ * @factory: a #GstRTSPMediaFactory
+ *
+ * Get the allowed protocols of @factory.
+ *
+ * Returns: a #GstRTSPLowerTrans
+ */
+GstRTSPLowerTrans
+gst_rtsp_media_factory_get_protocols (GstRTSPMediaFactory * factory)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+  GstRTSPLowerTrans res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory),
+      GST_RTSP_LOWER_TRANS_UNKNOWN);
+
+  priv = factory->priv;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  res = priv->protocols;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+  return res;
+}
+
+static gboolean
+compare_media (gpointer key, GstRTSPMedia * media1, GstRTSPMedia * media2)
+{
+  return (media1 == media2);
+}
+
+static void
+media_unprepared (GstRTSPMedia * media, GWeakRef * ref)
+{
+  GstRTSPMediaFactory *factory = g_weak_ref_get (ref);
+  GstRTSPMediaFactoryPrivate *priv;
+
+  if (!factory)
+    return;
+
+  priv = factory->priv;;
+
+  g_mutex_lock (&priv->medias_lock);
+  g_hash_table_foreach_remove (priv->medias, (GHRFunc) compare_media, media);
+  g_mutex_unlock (&priv->medias_lock);
+
+  g_object_unref (factory);
+}
+
+static GWeakRef *
+weak_ref_new (gpointer obj)
+{
+  GWeakRef *ref = g_slice_new (GWeakRef);
+
+  g_weak_ref_init (ref, obj);
+  return ref;
+}
+
+static void
+weak_ref_free (GWeakRef * ref)
+{
+  g_weak_ref_clear (ref);
+  g_slice_free (GWeakRef, ref);
+}
+
+/**
+ * gst_rtsp_media_factory_construct:
+ * @factory: a #GstRTSPMediaFactory
+ * @url: the url used
+ *
+ * Construct the media object and create its streams. Implementations
+ * should create the needed gstreamer elements and add them to the result
+ * object. No state changes should be performed on them yet.
+ *
+ * One or more GstRTSPStream objects should be created from the result
+ * with gst_rtsp_media_create_stream ().
+ *
+ * After the media is constructed, it can be configured and then prepared
+ * with gst_rtsp_media_prepare ().
+ *
+ * Returns: (transfer full): a new #GstRTSPMedia if the media could be prepared.
+ */
+GstRTSPMedia *
+gst_rtsp_media_factory_construct (GstRTSPMediaFactory * factory,
+    const GstRTSPUrl * url)
+{
+  GstRTSPMediaFactoryPrivate *priv;
+  gchar *key;
+  GstRTSPMedia *media;
+  GstRTSPMediaFactoryClass *klass;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+  g_return_val_if_fail (url != NULL, NULL);
+
+  priv = factory->priv;;
+  klass = GST_RTSP_MEDIA_FACTORY_GET_CLASS (factory);
+
+  /* convert the url to a key for the hashtable. NULL return or a NULL function
+   * will not cache anything for this factory. */
+  if (klass->gen_key)
+    key = klass->gen_key (factory, url);
+  else
+    key = NULL;
+
+  g_mutex_lock (&priv->medias_lock);
+  if (key) {
+    /* we have a key, see if we find a cached media */
+    media = g_hash_table_lookup (priv->medias, key);
+    if (media)
+      g_object_ref (media);
+  } else
+    media = NULL;
+
+  if (media == NULL) {
+    /* nothing cached found, try to create one */
+    if (klass->construct) {
+      media = klass->construct (factory, url);
+      if (media)
+        g_signal_emit (factory,
+            gst_rtsp_media_factory_signals[SIGNAL_MEDIA_CONSTRUCTED], 0, media,
+            NULL);
+    } else
+      media = NULL;
+
+    if (media) {
+      /* configure the media */
+      if (klass->configure)
+        klass->configure (factory, media);
+
+      g_signal_emit (factory,
+          gst_rtsp_media_factory_signals[SIGNAL_MEDIA_CONFIGURE], 0, media,
+          NULL);
+
+      /* check if we can cache this media */
+      if (gst_rtsp_media_is_shared (media)) {
+        /* insert in the hashtable, takes ownership of the key */
+        g_object_ref (media);
+        g_hash_table_insert (priv->medias, key, media);
+        key = NULL;
+      }
+      if (!gst_rtsp_media_is_reusable (media)) {
+        /* when not reusable, connect to the unprepare signal to remove the item
+         * from our cache when it gets unprepared */
+        g_signal_connect_data (media, "unprepared",
+            (GCallback) media_unprepared, weak_ref_new (factory),
+            (GClosureNotify) weak_ref_free, 0);
+      }
+    }
+  }
+  g_mutex_unlock (&priv->medias_lock);
+
+  if (key)
+    g_free (key);
+
+  GST_INFO ("constructed media %p for url %s", media, url->abspath);
+
+  return media;
+}
+
+static gchar *
+default_gen_key (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
+{
+  gchar *result;
+  const gchar *pre_query;
+  const gchar *query;
+
+  pre_query = url->query ? "?" : "";
+  query = url->query ? url->query : "";
+
+  result =
+      g_strdup_printf ("%u%s%s%s", url->port, url->abspath, pre_query, query);
+
+  return result;
+}
+
+static GstElement *
+default_create_element (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
+{
+  GstRTSPMediaFactoryPrivate *priv = factory->priv;
+  GstElement *element;
+  GError *error = NULL;
+
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  /* we need a parse syntax */
+  if (priv->launch == NULL)
+    goto no_launch;
+
+  /* parse the user provided launch line */
+  element = gst_parse_launch (priv->launch, &error);
+  if (element == NULL)
+    goto parse_error;
+
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+  if (error != NULL) {
+    /* a recoverable error was encountered */
+    GST_WARNING ("recoverable parsing error: %s", error->message);
+    g_error_free (error);
+  }
+  return element;
+
+  /* ERRORS */
+no_launch:
+  {
+    GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+    g_critical ("no launch line specified");
+    return NULL;
+  }
+parse_error:
+  {
+    g_critical ("could not parse launch syntax (%s): %s", priv->launch,
+        (error ? error->message : "unknown reason"));
+    GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+    if (error)
+      g_error_free (error);
+    return NULL;
+  }
+}
+
+static GstRTSPMedia *
+default_construct (GstRTSPMediaFactory * factory, const GstRTSPUrl * url)
+{
+  GstRTSPMedia *media;
+  GstElement *element, *pipeline;
+  GstRTSPMediaFactoryClass *klass;
+
+  klass = GST_RTSP_MEDIA_FACTORY_GET_CLASS (factory);
+
+  if (!klass->create_pipeline)
+    goto no_create;
+
+  element = gst_rtsp_media_factory_create_element (factory, url);
+  if (element == NULL)
+    goto no_element;
+
+  /* create a new empty media */
+  media = gst_rtsp_media_new (element);
+
+  gst_rtsp_media_collect_streams (media);
+
+  pipeline = klass->create_pipeline (factory, media);
+  if (pipeline == NULL)
+    goto no_pipeline;
+
+  return media;
+
+  /* ERRORS */
+no_create:
+  {
+    g_critical ("no create_pipeline function");
+    return NULL;
+  }
+no_element:
+  {
+    g_critical ("could not create element");
+    return NULL;
+  }
+no_pipeline:
+  {
+    g_critical ("can't create pipeline");
+    g_object_unref (media);
+    return NULL;
+  }
+}
+
+static GstElement *
+default_create_pipeline (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
+{
+  GstElement *pipeline;
+
+  pipeline = gst_pipeline_new ("media-pipeline");
+  gst_rtsp_media_take_pipeline (media, GST_PIPELINE_CAST (pipeline));
+
+  return pipeline;
+}
+
+static void
+default_configure (GstRTSPMediaFactory * factory, GstRTSPMedia * media)
+{
+  GstRTSPMediaFactoryPrivate *priv = factory->priv;
+  gboolean shared, eos_shutdown;
+  guint size;
+  GstRTSPSuspendMode suspend_mode;
+  GstRTSPProfile profiles;
+  GstRTSPLowerTrans protocols;
+  GstRTSPAddressPool *pool;
+  GstRTSPPermissions *perms;
+
+  /* configure the sharedness */
+  GST_RTSP_MEDIA_FACTORY_LOCK (factory);
+  suspend_mode = priv->suspend_mode;
+  shared = priv->shared;
+  eos_shutdown = priv->eos_shutdown;
+  size = priv->buffer_size;
+  profiles = priv->profiles;
+  protocols = priv->protocols;
+  GST_RTSP_MEDIA_FACTORY_UNLOCK (factory);
+
+  gst_rtsp_media_set_suspend_mode (media, suspend_mode);
+  gst_rtsp_media_set_shared (media, shared);
+  gst_rtsp_media_set_eos_shutdown (media, eos_shutdown);
+  gst_rtsp_media_set_buffer_size (media, size);
+  gst_rtsp_media_set_profiles (media, profiles);
+  gst_rtsp_media_set_protocols (media, protocols);
+
+  if ((pool = gst_rtsp_media_factory_get_address_pool (factory))) {
+    gst_rtsp_media_set_address_pool (media, pool);
+    g_object_unref (pool);
+  }
+  if ((perms = gst_rtsp_media_factory_get_permissions (factory))) {
+    gst_rtsp_media_set_permissions (media, perms);
+    gst_rtsp_permissions_unref (perms);
+  }
+}
+
+/**
+ * gst_rtsp_media_factory_create_element:
+ * @factory: a #GstRTSPMediaFactory
+ * @url: the url used
+ *
+ * Construct and return a #GstElement that is a #GstBin containing
+ * the elements to use for streaming the media.
+ *
+ * The bin should contain payloaders pay\%d for each stream. The default
+ * implementation of this function returns the bin created from the
+ * launch parameter.
+ *
+ * Returns: (transfer floating): a new #GstElement.
+ */
+GstElement *
+gst_rtsp_media_factory_create_element (GstRTSPMediaFactory * factory,
+    const GstRTSPUrl * url)
+{
+  GstRTSPMediaFactoryClass *klass;
+  GstElement *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory), NULL);
+  g_return_val_if_fail (url != NULL, NULL);
+
+  klass = GST_RTSP_MEDIA_FACTORY_GET_CLASS (factory);
+
+  if (klass->create_element)
+    result = klass->create_element (factory, url);
+  else
+    result = NULL;
+
+  return result;
+}
diff --git a/gst/rtsp-server/rtsp-media-factory.h b/gst/rtsp-server/rtsp-media-factory.h
new file mode 100644
index 0000000..bb6aec3
--- /dev/null
+++ b/gst/rtsp-server/rtsp-media-factory.h
@@ -0,0 +1,154 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+#include 
+
+#include "rtsp-media.h"
+#include "rtsp-permissions.h"
+#include "rtsp-address-pool.h"
+
+#ifndef __GST_RTSP_MEDIA_FACTORY_H__
+#define __GST_RTSP_MEDIA_FACTORY_H__
+
+G_BEGIN_DECLS
+
+/* types for the media factory */
+#define GST_TYPE_RTSP_MEDIA_FACTORY              (gst_rtsp_media_factory_get_type ())
+#define GST_IS_RTSP_MEDIA_FACTORY(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA_FACTORY))
+#define GST_IS_RTSP_MEDIA_FACTORY_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA_FACTORY))
+#define GST_RTSP_MEDIA_FACTORY_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA_FACTORY, GstRTSPMediaFactoryClass))
+#define GST_RTSP_MEDIA_FACTORY(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA_FACTORY, GstRTSPMediaFactory))
+#define GST_RTSP_MEDIA_FACTORY_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA_FACTORY, GstRTSPMediaFactoryClass))
+#define GST_RTSP_MEDIA_FACTORY_CAST(obj)         ((GstRTSPMediaFactory*)(obj))
+#define GST_RTSP_MEDIA_FACTORY_CLASS_CAST(klass) ((GstRTSPMediaFactoryClass*)(klass))
+
+typedef struct _GstRTSPMediaFactory GstRTSPMediaFactory;
+typedef struct _GstRTSPMediaFactoryClass GstRTSPMediaFactoryClass;
+typedef struct _GstRTSPMediaFactoryPrivate GstRTSPMediaFactoryPrivate;
+
+/**
+ * GstRTSPMediaFactory:
+ *
+ * The definition and logic for constructing the pipeline for a media. The media
+ * can contain multiple streams like audio and video.
+ */
+struct _GstRTSPMediaFactory {
+  GObject            parent;
+
+  /*< private >*/
+  GstRTSPMediaFactoryPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPMediaFactoryClass:
+ * @gen_key: convert @url to a key for caching shared #GstRTSPMedia objects.
+ *       The default implementation of this function will use the complete URL
+ *       including the query parameters to return a key.
+ * @create_element: Construct and return a #GstElement that is a #GstBin containing
+ *       the elements to use for streaming the media. The bin should contain
+ *       payloaders pay\%d for each stream. The default implementation of this
+ *       function returns the bin created from the launch parameter.
+ * @construct: the vmethod that will be called when the factory has to create the
+ *       #GstRTSPMedia for @url. The default implementation of this
+ *       function calls create_element to retrieve an element and then looks for
+ *       pay\%d to create the streams.
+ * @create_pipeline: create a new pipeline or re-use an existing one and
+ *       add the #GstRTSPMedia's element created by @construct to the pipeline.
+ * @configure: configure the media created with @construct. The default
+ *       implementation will configure the 'shared' property of the media.
+ * @media_constructed: signal emited when a media was constructed
+ * @media_configure: signal emited when a media should be configured
+ *
+ * The #GstRTSPMediaFactory class structure.
+ */
+struct _GstRTSPMediaFactoryClass {
+  GObjectClass  parent_class;
+
+  gchar *         (*gen_key)            (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
+
+  GstElement *    (*create_element)     (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
+  GstRTSPMedia *  (*construct)          (GstRTSPMediaFactory *factory, const GstRTSPUrl *url);
+  GstElement *    (*create_pipeline)    (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+  void            (*configure)          (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+
+  /* signals */
+  void            (*media_constructed)  (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+  void            (*media_configure)    (GstRTSPMediaFactory *factory, GstRTSPMedia *media);
+
+  /*< private >*/
+  gpointer         _gst_reserved[GST_PADDING_LARGE];
+};
+
+GType                 gst_rtsp_media_factory_get_type     (void);
+
+/* creating the factory */
+GstRTSPMediaFactory * gst_rtsp_media_factory_new          (void);
+
+/* configuring the factory */
+void                  gst_rtsp_media_factory_set_launch       (GstRTSPMediaFactory *factory,
+                                                               const gchar *launch);
+gchar *               gst_rtsp_media_factory_get_launch       (GstRTSPMediaFactory *factory);
+
+void                  gst_rtsp_media_factory_set_permissions  (GstRTSPMediaFactory *factory,
+                                                               GstRTSPPermissions *permissions);
+GstRTSPPermissions *  gst_rtsp_media_factory_get_permissions  (GstRTSPMediaFactory *factory);
+void                  gst_rtsp_media_factory_add_role         (GstRTSPMediaFactory *factory,
+                                                               const gchar *role,
+                                                               const gchar *fieldname, ...);
+
+void                  gst_rtsp_media_factory_set_shared       (GstRTSPMediaFactory *factory,
+                                                               gboolean shared);
+gboolean              gst_rtsp_media_factory_is_shared        (GstRTSPMediaFactory *factory);
+
+void                  gst_rtsp_media_factory_set_suspend_mode (GstRTSPMediaFactory *factory,
+                                                               GstRTSPSuspendMode mode);
+GstRTSPSuspendMode    gst_rtsp_media_factory_get_suspend_mode (GstRTSPMediaFactory *factory);
+
+void                  gst_rtsp_media_factory_set_eos_shutdown (GstRTSPMediaFactory *factory,
+                                                               gboolean eos_shutdown);
+gboolean              gst_rtsp_media_factory_is_eos_shutdown  (GstRTSPMediaFactory *factory);
+
+void                  gst_rtsp_media_factory_set_profiles     (GstRTSPMediaFactory *factory,
+                                                               GstRTSPProfile profiles);
+GstRTSPProfile        gst_rtsp_media_factory_get_profiles     (GstRTSPMediaFactory *factory);
+
+void                  gst_rtsp_media_factory_set_protocols    (GstRTSPMediaFactory *factory,
+                                                               GstRTSPLowerTrans protocols);
+GstRTSPLowerTrans     gst_rtsp_media_factory_get_protocols    (GstRTSPMediaFactory *factory);
+
+void                  gst_rtsp_media_factory_set_address_pool (GstRTSPMediaFactory * factory,
+                                                               GstRTSPAddressPool * pool);
+GstRTSPAddressPool *  gst_rtsp_media_factory_get_address_pool (GstRTSPMediaFactory * factory);
+
+void                  gst_rtsp_media_factory_set_buffer_size  (GstRTSPMediaFactory * factory,
+                                                               guint size);
+guint                 gst_rtsp_media_factory_get_buffer_size  (GstRTSPMediaFactory * factory);
+
+/* creating the media from the factory and a url */
+GstRTSPMedia *        gst_rtsp_media_factory_construct        (GstRTSPMediaFactory *factory,
+                                                               const GstRTSPUrl *url);
+
+GstElement *          gst_rtsp_media_factory_create_element   (GstRTSPMediaFactory *factory,
+                                                               const GstRTSPUrl *url);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_MEDIA_FACTORY_H__ */
diff --git a/gst/rtsp-server/rtsp-media.c b/gst/rtsp-server/rtsp-media.c
new file mode 100644
index 0000000..6b81e25
--- /dev/null
+++ b/gst/rtsp-server/rtsp-media.c
@@ -0,0 +1,3092 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-media
+ * @short_description: The media pipeline
+ * @see_also: #GstRTSPMediaFactory, #GstRTSPStream, #GstRTSPSession,
+ *     #GstRTSPSessionMedia
+ *
+ * a #GstRTSPMedia contains the complete GStreamer pipeline to manage the
+ * streaming to the clients. The actual data transfer is done by the
+ * #GstRTSPStream objects that are created and exposed by the #GstRTSPMedia.
+ *
+ * The #GstRTSPMedia is usually created from a #GstRTSPMediaFactory when the
+ * client does a DESCRIBE or SETUP of a resource.
+ *
+ * A media is created with gst_rtsp_media_new() that takes the element that will
+ * provide the streaming elements. For each of the streams, a new #GstRTSPStream
+ * object needs to be made with the gst_rtsp_media_create_stream() which takes
+ * the payloader element and the source pad that produces the RTP stream.
+ *
+ * The pipeline of the media is set to PAUSED with gst_rtsp_media_prepare(). The
+ * prepare method will add rtpbin and sinks and sources to send and receive RTP
+ * and RTCP packets from the clients. Each stream srcpad is connected to an
+ * input into the internal rtpbin.
+ *
+ * It is also possible to dynamically create #GstRTSPStream objects during the
+ * prepare phase. With gst_rtsp_media_get_status() you can check the status of
+ * the prepare phase.
+ *
+ * After the media is prepared, it is ready for streaming. It will usually be
+ * managed in a session with gst_rtsp_session_manage_media(). See
+ * #GstRTSPSession and #GstRTSPSessionMedia.
+ *
+ * The state of the media can be controlled with gst_rtsp_media_set_state ().
+ * Seeking can be done with gst_rtsp_media_seek().
+ *
+ * With gst_rtsp_media_unprepare() the pipeline is stopped and shut down. When
+ * gst_rtsp_media_set_eos_shutdown() an EOS will be sent to the pipeline to
+ * cleanly shut down.
+ *
+ * With gst_rtsp_media_set_shared(), the media can be shared between multiple
+ * clients. With gst_rtsp_media_set_reusable() you can control if the pipeline
+ * can be prepared again after an unprepare.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include 
+#include 
+
+#include 
+#include 
+
+#include "rtsp-media.h"
+
+#define GST_RTSP_MEDIA_GET_PRIVATE(obj)  \
+     (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaPrivate))
+
+struct _GstRTSPMediaPrivate
+{
+  GMutex lock;
+  GCond cond;
+
+  /* protected by lock */
+  GstRTSPPermissions *permissions;
+  gboolean shared;
+  gboolean suspend_mode;
+  gboolean reusable;
+  GstRTSPProfile profiles;
+  GstRTSPLowerTrans protocols;
+  gboolean reused;
+  gboolean eos_shutdown;
+  guint buffer_size;
+  GstRTSPAddressPool *pool;
+  gboolean blocked;
+
+  GstElement *element;
+  GRecMutex state_lock;         /* locking order: state lock, lock */
+  GPtrArray *streams;           /* protected by lock */
+  GList *dynamic;               /* protected by lock */
+  GstRTSPMediaStatus status;    /* protected by lock */
+  gint prepare_count;
+  gint n_active;
+  gboolean adding;
+
+  /* the pipeline for the media */
+  GstElement *pipeline;
+  GstElement *fakesink;         /* protected by lock */
+  GSource *source;
+  guint id;
+  GstRTSPThread *thread;
+
+  gboolean time_provider;
+  GstNetTimeProvider *nettime;
+
+  gboolean is_live;
+  gboolean seekable;
+  gboolean buffering;
+  GstState target_state;
+
+  /* RTP session manager */
+  GstElement *rtpbin;
+
+  /* the range of media */
+  GstRTSPTimeRange range;       /* protected by lock */
+  GstClockTime range_start;
+  GstClockTime range_stop;
+};
+
+#define DEFAULT_SHARED          FALSE
+#define DEFAULT_SUSPEND_MODE    GST_RTSP_SUSPEND_MODE_NONE
+#define DEFAULT_REUSABLE        FALSE
+#define DEFAULT_PROFILES        GST_RTSP_PROFILE_AVP
+#define DEFAULT_PROTOCOLS       GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
+                                        GST_RTSP_LOWER_TRANS_TCP
+#define DEFAULT_EOS_SHUTDOWN    FALSE
+#define DEFAULT_BUFFER_SIZE     0x80000
+#define DEFAULT_TIME_PROVIDER   FALSE
+
+/* define to dump received RTCP packets */
+#undef DUMP_STATS
+
+enum
+{
+  PROP_0,
+  PROP_SHARED,
+  PROP_SUSPEND_MODE,
+  PROP_REUSABLE,
+  PROP_PROFILES,
+  PROP_PROTOCOLS,
+  PROP_EOS_SHUTDOWN,
+  PROP_BUFFER_SIZE,
+  PROP_ELEMENT,
+  PROP_TIME_PROVIDER,
+  PROP_LAST
+};
+
+enum
+{
+  SIGNAL_NEW_STREAM,
+  SIGNAL_REMOVED_STREAM,
+  SIGNAL_PREPARED,
+  SIGNAL_UNPREPARED,
+  SIGNAL_TARGET_STATE,
+  SIGNAL_NEW_STATE,
+  SIGNAL_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
+#define GST_CAT_DEFAULT rtsp_media_debug
+
+static void gst_rtsp_media_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_media_finalize (GObject * obj);
+
+static gboolean default_handle_message (GstRTSPMedia * media,
+    GstMessage * message);
+static void finish_unprepare (GstRTSPMedia * media);
+static gboolean default_prepare (GstRTSPMedia * media, GstRTSPThread * thread);
+static gboolean default_unprepare (GstRTSPMedia * media);
+static gboolean default_suspend (GstRTSPMedia * media);
+static gboolean default_unsuspend (GstRTSPMedia * media);
+static gboolean default_convert_range (GstRTSPMedia * media,
+    GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
+static gboolean default_query_position (GstRTSPMedia * media,
+    gint64 * position);
+static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
+static GstElement *default_create_rtpbin (GstRTSPMedia * media);
+static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+    GstSDPInfo * info);
+
+static gboolean wait_preroll (GstRTSPMedia * media);
+
+static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
+
+#define C_ENUM(v) ((gint) v)
+
+#define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type())
+GType
+gst_rtsp_suspend_mode_get_type (void)
+{
+  static gsize id = 0;
+  static const GEnumValue values[] = {
+    {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
+    {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
+        "pause"},
+    {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
+        "reset"},
+    {0, NULL, NULL}
+  };
+
+  if (g_once_init_enter (&id)) {
+    GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
+    g_once_init_leave (&id, tmp);
+  }
+  return (GType) id;
+}
+
+G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate));
+
+  gobject_class = G_OBJECT_CLASS (klass);
+
+  gobject_class->get_property = gst_rtsp_media_get_property;
+  gobject_class->set_property = gst_rtsp_media_set_property;
+  gobject_class->finalize = gst_rtsp_media_finalize;
+
+  g_object_class_install_property (gobject_class, PROP_SHARED,
+      g_param_spec_boolean ("shared", "Shared",
+          "If this media pipeline can be shared", DEFAULT_SHARED,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
+      g_param_spec_enum ("suspend-mode", "Suspend Mode",
+          "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
+          DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_REUSABLE,
+      g_param_spec_boolean ("reusable", "Reusable",
+          "If this media pipeline can be reused after an unprepare",
+          DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_PROFILES,
+      g_param_spec_flags ("profiles", "Profiles",
+          "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
+          DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
+      g_param_spec_flags ("protocols", "Protocols",
+          "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
+          DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
+      g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
+          "Send an EOS event to the pipeline before unpreparing",
+          DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
+      g_param_spec_uint ("buffer-size", "Buffer Size",
+          "The kernel UDP buffer size to use", 0, G_MAXUINT,
+          DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_ELEMENT,
+      g_param_spec_object ("element", "The Element",
+          "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
+          G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
+
+  g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
+      g_param_spec_boolean ("time-provider", "Time Provider",
+          "Use a NetTimeProvider for clients",
+          DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
+      g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+      G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
+      g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
+
+  gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
+      g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+      GST_TYPE_RTSP_STREAM);
+
+  gst_rtsp_media_signals[SIGNAL_PREPARED] =
+      g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+      G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
+      g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
+
+  gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
+      g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+      G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
+      g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
+
+  gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
+      g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, target_state),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
+
+  gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
+      g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+      G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
+      g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
+
+  klass->handle_message = default_handle_message;
+  klass->prepare = default_prepare;
+  klass->unprepare = default_unprepare;
+  klass->suspend = default_suspend;
+  klass->unsuspend = default_unsuspend;
+  klass->convert_range = default_convert_range;
+  klass->query_position = default_query_position;
+  klass->query_stop = default_query_stop;
+  klass->create_rtpbin = default_create_rtpbin;
+  klass->setup_sdp = default_setup_sdp;
+}
+
+static void
+gst_rtsp_media_init (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = GST_RTSP_MEDIA_GET_PRIVATE (media);
+
+  media->priv = priv;
+
+  priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
+  g_mutex_init (&priv->lock);
+  g_cond_init (&priv->cond);
+  g_rec_mutex_init (&priv->state_lock);
+
+  priv->shared = DEFAULT_SHARED;
+  priv->suspend_mode = DEFAULT_SUSPEND_MODE;
+  priv->reusable = DEFAULT_REUSABLE;
+  priv->profiles = DEFAULT_PROFILES;
+  priv->protocols = DEFAULT_PROTOCOLS;
+  priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
+  priv->buffer_size = DEFAULT_BUFFER_SIZE;
+  priv->time_provider = DEFAULT_TIME_PROVIDER;
+}
+
+static void
+gst_rtsp_media_finalize (GObject * obj)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPMedia *media;
+
+  media = GST_RTSP_MEDIA (obj);
+  priv = media->priv;
+
+  GST_INFO ("finalize media %p", media);
+
+  if (priv->permissions)
+    gst_rtsp_permissions_unref (priv->permissions);
+
+  g_ptr_array_unref (priv->streams);
+
+  g_list_free_full (priv->dynamic, gst_object_unref);
+
+  if (priv->pipeline)
+    gst_object_unref (priv->pipeline);
+  if (priv->nettime)
+    gst_object_unref (priv->nettime);
+  gst_object_unref (priv->element);
+  if (priv->pool)
+    g_object_unref (priv->pool);
+  g_mutex_clear (&priv->lock);
+  g_cond_clear (&priv->cond);
+  g_rec_mutex_clear (&priv->state_lock);
+
+  G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_media_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRTSPMedia *media = GST_RTSP_MEDIA (object);
+
+  switch (propid) {
+    case PROP_ELEMENT:
+      g_value_set_object (value, media->priv->element);
+      break;
+    case PROP_SHARED:
+      g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
+      break;
+    case PROP_SUSPEND_MODE:
+      g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
+      break;
+    case PROP_REUSABLE:
+      g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
+      break;
+    case PROP_PROFILES:
+      g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
+      break;
+    case PROP_PROTOCOLS:
+      g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
+      break;
+    case PROP_EOS_SHUTDOWN:
+      g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
+      break;
+    case PROP_BUFFER_SIZE:
+      g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
+      break;
+    case PROP_TIME_PROVIDER:
+      g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+static void
+gst_rtsp_media_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRTSPMedia *media = GST_RTSP_MEDIA (object);
+
+  switch (propid) {
+    case PROP_ELEMENT:
+      media->priv->element = g_value_get_object (value);
+      gst_object_ref_sink (media->priv->element);
+      break;
+    case PROP_SHARED:
+      gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
+      break;
+    case PROP_SUSPEND_MODE:
+      gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
+      break;
+    case PROP_REUSABLE:
+      gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
+      break;
+    case PROP_PROFILES:
+      gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
+      break;
+    case PROP_PROTOCOLS:
+      gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
+      break;
+    case PROP_EOS_SHUTDOWN:
+      gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
+      break;
+    case PROP_BUFFER_SIZE:
+      gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
+      break;
+    case PROP_TIME_PROVIDER:
+      gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+typedef struct
+{
+  gint64 position;
+  gboolean ret;
+} DoQueryPositionData;
+
+static void
+do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
+{
+  gint64 tmp;
+
+  if (gst_rtsp_stream_query_position (stream, &tmp)) {
+    data->position = MAX (data->position, tmp);
+    data->ret = TRUE;
+  }
+}
+
+static gboolean
+default_query_position (GstRTSPMedia * media, gint64 * position)
+{
+  GstRTSPMediaPrivate *priv;
+  DoQueryPositionData data;
+
+  priv = media->priv;
+
+  data.position = -1;
+  data.ret = FALSE;
+
+  g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
+
+  *position = data.position;
+
+  return data.ret;
+}
+
+typedef struct
+{
+  gint64 stop;
+  gboolean ret;
+} DoQueryStopData;
+
+static void
+do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
+{
+  gint64 tmp;
+
+  if (gst_rtsp_stream_query_stop (stream, &tmp)) {
+    data->stop = MAX (data->stop, tmp);
+    data->ret = TRUE;
+  }
+}
+
+static gboolean
+default_query_stop (GstRTSPMedia * media, gint64 * stop)
+{
+  GstRTSPMediaPrivate *priv;
+  DoQueryStopData data;
+
+  priv = media->priv;
+
+  data.stop = -1;
+  data.ret = FALSE;
+
+  g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
+
+  *stop = data.stop;
+
+  return data.ret;
+}
+
+static GstElement *
+default_create_rtpbin (GstRTSPMedia * media)
+{
+  GstElement *rtpbin;
+
+  rtpbin = gst_element_factory_make ("rtpbin", NULL);
+
+  return rtpbin;
+}
+
+/* must be called with state lock */
+static void
+collect_media_stats (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  gint64 position = 0, stop = -1;
+
+  if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
+      priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
+    return;
+
+  priv->range.unit = GST_RTSP_RANGE_NPT;
+
+  GST_INFO ("collect media stats");
+
+  if (priv->is_live) {
+    priv->range.min.type = GST_RTSP_TIME_NOW;
+    priv->range.min.seconds = -1;
+    priv->range_start = -1;
+    priv->range.max.type = GST_RTSP_TIME_END;
+    priv->range.max.seconds = -1;
+    priv->range_stop = -1;
+  } else {
+    GstRTSPMediaClass *klass;
+    gboolean ret;
+
+    klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+    /* get the position */
+    ret = FALSE;
+    if (klass->query_position)
+      ret = klass->query_position (media, &position);
+
+    if (!ret) {
+      GST_INFO ("position query failed");
+      position = 0;
+    }
+
+    /* get the current segment stop */
+    ret = FALSE;
+    if (klass->query_stop)
+      ret = klass->query_stop (media, &stop);
+
+    if (!ret) {
+      GST_INFO ("stop query failed");
+      stop = -1;
+    }
+
+    GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
+        GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
+
+    if (position == -1) {
+      priv->range.min.type = GST_RTSP_TIME_NOW;
+      priv->range.min.seconds = -1;
+      priv->range_start = -1;
+    } else {
+      priv->range.min.type = GST_RTSP_TIME_SECONDS;
+      priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
+      priv->range_start = position;
+    }
+    if (stop == -1) {
+      priv->range.max.type = GST_RTSP_TIME_END;
+      priv->range.max.seconds = -1;
+      priv->range_stop = -1;
+    } else {
+      priv->range.max.type = GST_RTSP_TIME_SECONDS;
+      priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
+      priv->range_stop = stop;
+    }
+  }
+}
+
+/**
+ * gst_rtsp_media_new:
+ * @element: (transfer full): a #GstElement
+ *
+ * Create a new #GstRTSPMedia instance. @element is the bin element that
+ * provides the different streams. The #GstRTSPMedia object contains the
+ * element to produce RTP data for one or more related (audio/video/..)
+ * streams.
+ *
+ * Ownership is taken of @element.
+ *
+ * Returns: (transfer full): a new #GstRTSPMedia object.
+ */
+GstRTSPMedia *
+gst_rtsp_media_new (GstElement * element)
+{
+  GstRTSPMedia *result;
+
+  g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
+
+  result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_media_get_element:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the element that was used when constructing @media.
+ *
+ * Returns: (transfer full): a #GstElement. Unref after usage.
+ */
+GstElement *
+gst_rtsp_media_get_element (GstRTSPMedia * media)
+{
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+  return gst_object_ref (media->priv->element);
+}
+
+/**
+ * gst_rtsp_media_take_pipeline:
+ * @media: a #GstRTSPMedia
+ * @pipeline: (transfer full): a #GstPipeline
+ *
+ * Set @pipeline as the #GstPipeline for @media. Ownership is
+ * taken of @pipeline.
+ */
+void
+gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
+{
+  GstRTSPMediaPrivate *priv;
+  GstElement *old;
+  GstNetTimeProvider *nettime;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+  g_return_if_fail (GST_IS_PIPELINE (pipeline));
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  old = priv->pipeline;
+  priv->pipeline = GST_ELEMENT_CAST (pipeline);
+  nettime = priv->nettime;
+  priv->nettime = NULL;
+  g_mutex_unlock (&priv->lock);
+
+  if (old)
+    gst_object_unref (old);
+
+  if (nettime)
+    gst_object_unref (nettime);
+
+  gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
+}
+
+/**
+ * gst_rtsp_media_set_permissions:
+ * @media: a #GstRTSPMedia
+ * @permissions: (transfer none): a #GstRTSPPermissions
+ *
+ * Set @permissions on @media.
+ */
+void
+gst_rtsp_media_set_permissions (GstRTSPMedia * media,
+    GstRTSPPermissions * permissions)
+{
+  GstRTSPMediaPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  if (priv->permissions)
+    gst_rtsp_permissions_unref (priv->permissions);
+  if ((priv->permissions = permissions))
+    gst_rtsp_permissions_ref (permissions);
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_permissions:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the permissions object from @media.
+ *
+ * Returns: (transfer full): a #GstRTSPPermissions object, unref after usage.
+ */
+GstRTSPPermissions *
+gst_rtsp_media_get_permissions (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPPermissions *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  if ((result = priv->permissions))
+    gst_rtsp_permissions_ref (result);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_media_set_suspend_mode:
+ * @media: a #GstRTSPMedia
+ * @mode: the new #GstRTSPSuspendMode
+ *
+ * Control how @ media will be suspended after the SDP has been generated and
+ * after a PAUSE request has been performed.
+ *
+ * Media must be unprepared when setting the suspend mode.
+ */
+void
+gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
+{
+  GstRTSPMediaPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  priv = media->priv;
+
+  g_rec_mutex_lock (&priv->state_lock);
+  if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
+    goto was_prepared;
+  priv->suspend_mode = mode;
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return;
+
+  /* ERRORS */
+was_prepared:
+  {
+    GST_WARNING ("media %p was prepared", media);
+    g_rec_mutex_unlock (&priv->state_lock);
+  }
+}
+
+/**
+ * gst_rtsp_media_get_suspend_mode:
+ * @media: a #GstRTSPMedia
+ *
+ * Get how @media will be suspended.
+ *
+ * Returns: #GstRTSPSuspendMode.
+ */
+GstRTSPSuspendMode
+gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPSuspendMode res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
+
+  priv = media->priv;
+
+  g_rec_mutex_lock (&priv->state_lock);
+  res = priv->suspend_mode;
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_media_set_shared:
+ * @media: a #GstRTSPMedia
+ * @shared: the new value
+ *
+ * Set or unset if the pipeline for @media can be shared will multiple clients.
+ * When @shared is %TRUE, client requests for this media will share the media
+ * pipeline.
+ */
+void
+gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
+{
+  GstRTSPMediaPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->shared = shared;
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_is_shared:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if the pipeline for @media can be shared between multiple clients.
+ *
+ * Returns: %TRUE if the media can be shared between clients.
+ */
+gboolean
+gst_rtsp_media_is_shared (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+  gboolean res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  res = priv->shared;
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_media_set_reusable:
+ * @media: a #GstRTSPMedia
+ * @reusable: the new value
+ *
+ * Set or unset if the pipeline for @media can be reused after the pipeline has
+ * been unprepared.
+ */
+void
+gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
+{
+  GstRTSPMediaPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->reusable = reusable;
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_is_reusable:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if the pipeline for @media can be reused after an unprepare.
+ *
+ * Returns: %TRUE if the media can be reused
+ */
+gboolean
+gst_rtsp_media_is_reusable (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+  gboolean res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  res = priv->reusable;
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+static void
+do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
+{
+  gst_rtsp_stream_set_profiles (stream, *profiles);
+}
+
+/**
+ * gst_rtsp_media_set_profiles:
+ * @media: a #GstRTSPMedia
+ * @profiles: the new flags
+ *
+ * Configure the allowed lower transport for @media.
+ */
+void
+gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
+{
+  GstRTSPMediaPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->profiles = profiles;
+  g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_profiles:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the allowed profiles of @media.
+ *
+ * Returns: a #GstRTSPProfile
+ */
+GstRTSPProfile
+gst_rtsp_media_get_profiles (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPProfile res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  res = priv->profiles;
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+static void
+do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
+{
+  gst_rtsp_stream_set_protocols (stream, *protocols);
+}
+
+/**
+ * gst_rtsp_media_set_protocols:
+ * @media: a #GstRTSPMedia
+ * @protocols: the new flags
+ *
+ * Configure the allowed lower transport for @media.
+ */
+void
+gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
+{
+  GstRTSPMediaPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->protocols = protocols;
+  g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_protocols:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the allowed protocols of @media.
+ *
+ * Returns: a #GstRTSPLowerTrans
+ */
+GstRTSPLowerTrans
+gst_rtsp_media_get_protocols (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPLowerTrans res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
+      GST_RTSP_LOWER_TRANS_UNKNOWN);
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  res = priv->protocols;
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_media_set_eos_shutdown:
+ * @media: a #GstRTSPMedia
+ * @eos_shutdown: the new value
+ *
+ * Set or unset if an EOS event will be sent to the pipeline for @media before
+ * it is unprepared.
+ */
+void
+gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
+{
+  GstRTSPMediaPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->eos_shutdown = eos_shutdown;
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_is_eos_shutdown:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if the pipeline for @media will send an EOS down the pipeline before
+ * unpreparing.
+ *
+ * Returns: %TRUE if the media will send EOS before unpreparing.
+ */
+gboolean
+gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+  gboolean res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  res = priv->eos_shutdown;
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_media_set_buffer_size:
+ * @media: a #GstRTSPMedia
+ * @size: the new value
+ *
+ * Set the kernel UDP buffer size.
+ */
+void
+gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
+{
+  GstRTSPMediaPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  GST_LOG_OBJECT (media, "set buffer size %u", size);
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->buffer_size = size;
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_buffer_size:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the kernel UDP buffer size.
+ *
+ * Returns: the kernel UDP buffer size.
+ */
+guint
+gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+  guint res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+  priv = media->priv;
+
+  g_mutex_unlock (&priv->lock);
+  res = priv->buffer_size;
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_media_use_time_provider:
+ * @media: a #GstRTSPMedia
+ * @time_provider: if a #GstNetTimeProvider should be used
+ *
+ * Set @media to provide a #GstNetTimeProvider.
+ */
+void
+gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
+{
+  GstRTSPMediaPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->time_provider = time_provider;
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_is_time_provider:
+ * @media: a #GstRTSPMedia
+ *
+ * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
+ *
+ * Use gst_rtsp_media_get_time_provider() to get the network clock.
+ *
+ * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
+ */
+gboolean
+gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+  gboolean res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+  priv = media->priv;
+
+  g_mutex_unlock (&priv->lock);
+  res = priv->time_provider;
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_media_set_address_pool:
+ * @media: a #GstRTSPMedia
+ * @pool: (transfer none): a #GstRTSPAddressPool
+ *
+ * configure @pool to be used as the address pool of @media.
+ */
+void
+gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
+    GstRTSPAddressPool * pool)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPAddressPool *old;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  priv = media->priv;
+
+  GST_LOG_OBJECT (media, "set address pool %p", pool);
+
+  g_mutex_lock (&priv->lock);
+  if ((old = priv->pool) != pool)
+    priv->pool = pool ? g_object_ref (pool) : NULL;
+  else
+    old = NULL;
+  g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
+      pool);
+  g_mutex_unlock (&priv->lock);
+
+  if (old)
+    g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_media_get_address_pool:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the #GstRTSPAddressPool used as the address pool of @media.
+ *
+ * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
+ * usage.
+ */
+GstRTSPAddressPool *
+gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPAddressPool *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  if ((result = priv->pool))
+    g_object_ref (result);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_media_collect_streams:
+ * @media: a #GstRTSPMedia
+ *
+ * Find all payloader elements, they should be named pay\%d in the
+ * element of @media, and create #GstRTSPStreams for them.
+ *
+ * Collect all dynamic elements, named dynpay\%d, and add them to
+ * the list of dynamic elements.
+ */
+void
+gst_rtsp_media_collect_streams (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+  GstElement *element, *elem;
+  GstPad *pad;
+  gint i;
+  gboolean have_elem;
+
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  priv = media->priv;
+  element = priv->element;
+
+  have_elem = TRUE;
+  for (i = 0; have_elem; i++) {
+    gchar *name;
+
+    have_elem = FALSE;
+
+    name = g_strdup_printf ("pay%d", i);
+    if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
+      GST_INFO ("found stream %d with payloader %p", i, elem);
+
+      /* take the pad of the payloader */
+      pad = gst_element_get_static_pad (elem, "src");
+      /* create the stream */
+      gst_rtsp_media_create_stream (media, elem, pad);
+      gst_object_unref (pad);
+      gst_object_unref (elem);
+
+      have_elem = TRUE;
+    }
+    g_free (name);
+
+    name = g_strdup_printf ("dynpay%d", i);
+    if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
+      /* a stream that will dynamically create pads to provide RTP packets */
+      GST_INFO ("found dynamic element %d, %p", i, elem);
+
+      g_mutex_lock (&priv->lock);
+      priv->dynamic = g_list_prepend (priv->dynamic, elem);
+      g_mutex_unlock (&priv->lock);
+
+      have_elem = TRUE;
+    }
+    g_free (name);
+  }
+}
+
+/**
+ * gst_rtsp_media_create_stream:
+ * @media: a #GstRTSPMedia
+ * @payloader: a #GstElement
+ * @srcpad: a source #GstPad
+ *
+ * Create a new stream in @media that provides RTP data on @srcpad.
+ * @srcpad should be a pad of an element inside @media->element.
+ *
+ * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
+ * as @media exists.
+ */
+GstRTSPStream *
+gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
+    GstPad * pad)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPStream *stream;
+  GstPad *srcpad;
+  gchar *name;
+  gint idx;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+  g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
+  g_return_val_if_fail (GST_IS_PAD (pad), NULL);
+  g_return_val_if_fail (GST_PAD_IS_SRC (pad), NULL);
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  idx = priv->streams->len;
+
+  GST_DEBUG ("media %p: creating stream with index %d", media, idx);
+
+  name = g_strdup_printf ("src_%u", idx);
+  srcpad = gst_ghost_pad_new (name, pad);
+  gst_pad_set_active (srcpad, TRUE);
+  gst_element_add_pad (priv->element, srcpad);
+  g_free (name);
+
+  stream = gst_rtsp_stream_new (idx, payloader, srcpad);
+  if (priv->pool)
+    gst_rtsp_stream_set_address_pool (stream, priv->pool);
+  gst_rtsp_stream_set_profiles (stream, priv->profiles);
+  gst_rtsp_stream_set_protocols (stream, priv->protocols);
+
+  g_ptr_array_add (priv->streams, stream);
+  g_mutex_unlock (&priv->lock);
+
+  g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
+      NULL);
+
+  return stream;
+}
+
+static void
+gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
+{
+  GstRTSPMediaPrivate *priv;
+  GstPad *srcpad;
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  /* remove the ghostpad */
+  srcpad = gst_rtsp_stream_get_srcpad (stream);
+  gst_element_remove_pad (priv->element, srcpad);
+  gst_object_unref (srcpad);
+  /* now remove the stream */
+  g_object_ref (stream);
+  g_ptr_array_remove (priv->streams, stream);
+  g_mutex_unlock (&priv->lock);
+
+  g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
+      stream, NULL);
+
+  g_object_unref (stream);
+}
+
+/**
+ * gst_rtsp_media_n_streams:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the number of streams in this media.
+ *
+ * Returns: The number of streams.
+ */
+guint
+gst_rtsp_media_n_streams (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+  guint res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  res = priv->streams->len;
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_media_get_stream:
+ * @media: a #GstRTSPMedia
+ * @idx: the stream index
+ *
+ * Retrieve the stream with index @idx from @media.
+ *
+ * Returns: (nullable) (transfer none): the #GstRTSPStream at index
+ * @idx or %NULL when a stream with that index did not exist.
+ */
+GstRTSPStream *
+gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPStream *res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  if (idx < priv->streams->len)
+    res = g_ptr_array_index (priv->streams, idx);
+  else
+    res = NULL;
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_media_find_stream:
+ * @media: a #GstRTSPMedia
+ * @control: the control of the stream
+ *
+ * Find a stream in @media with @control as the control uri.
+ *
+ * Returns: (nullable) (transfer none): the #GstRTSPStream with
+ * control uri @control or %NULL when a stream with that control did
+ * not exist.
+ */
+GstRTSPStream *
+gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPStream *res;
+  gint i;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+  g_return_val_if_fail (control != NULL, NULL);
+
+  priv = media->priv;
+
+  res = NULL;
+
+  g_mutex_lock (&priv->lock);
+  for (i = 0; i < priv->streams->len; i++) {
+    GstRTSPStream *test;
+
+    test = g_ptr_array_index (priv->streams, i);
+    if (gst_rtsp_stream_has_control (test, control)) {
+      res = test;
+      break;
+    }
+  }
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+/* called with state-lock */
+static gboolean
+default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
+    GstRTSPRangeUnit unit)
+{
+  return gst_rtsp_range_convert_units (range, unit);
+}
+
+/**
+ * gst_rtsp_media_get_range_string:
+ * @media: a #GstRTSPMedia
+ * @play: for the PLAY request
+ * @unit: the unit to use for the string
+ *
+ * Get the current range as a string. @media must be prepared with
+ * gst_rtsp_media_prepare ().
+ *
+ * Returns: (transfer full): The range as a string, g_free() after usage.
+ */
+gchar *
+gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
+    GstRTSPRangeUnit unit)
+{
+  GstRTSPMediaClass *klass;
+  GstRTSPMediaPrivate *priv;
+  gchar *result;
+  GstRTSPTimeRange range;
+
+  klass = GST_RTSP_MEDIA_GET_CLASS (media);
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+  g_return_val_if_fail (klass->convert_range != NULL, FALSE);
+
+  priv = media->priv;
+
+  g_rec_mutex_lock (&priv->state_lock);
+  if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
+      priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
+    goto not_prepared;
+
+  g_mutex_lock (&priv->lock);
+
+  /* Update the range value with current position/duration */
+  collect_media_stats (media);
+
+  /* make copy */
+  range = priv->range;
+
+  if (!play && priv->n_active > 0) {
+    range.min.type = GST_RTSP_TIME_NOW;
+    range.min.seconds = -1;
+  }
+  g_mutex_unlock (&priv->lock);
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  if (!klass->convert_range (media, &range, unit))
+    goto conversion_failed;
+
+  result = gst_rtsp_range_to_string (&range);
+
+  return result;
+
+  /* ERRORS */
+not_prepared:
+  {
+    GST_WARNING ("media %p was not prepared", media);
+    g_rec_mutex_unlock (&priv->state_lock);
+    return NULL;
+  }
+conversion_failed:
+  {
+    GST_WARNING ("range conversion to unit %d failed", unit);
+    return NULL;
+  }
+}
+
+static void
+stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
+{
+  gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
+}
+
+static void
+media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+
+  GST_DEBUG ("media %p set blocked %d", media, blocked);
+  priv->blocked = blocked;
+  g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
+}
+
+static void
+gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->status = status;
+  GST_DEBUG ("setting new status to %d", status);
+  g_cond_broadcast (&priv->cond);
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_media_get_status:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the status of @media. When @media is busy preparing, this function waits
+ * until @media is prepared or in error.
+ *
+ * Returns: the status of @media.
+ */
+GstRTSPMediaStatus
+gst_rtsp_media_get_status (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstRTSPMediaStatus result;
+  gint64 end_time;
+
+  g_mutex_lock (&priv->lock);
+  end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
+  /* while we are preparing, wait */
+  while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
+    GST_DEBUG ("waiting for status change");
+    if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
+      GST_DEBUG ("timeout, assuming error status");
+      priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
+    }
+  }
+  /* could be success or error */
+  result = priv->status;
+  GST_DEBUG ("got status %d", result);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_media_seek:
+ * @media: a #GstRTSPMedia
+ * @range: (transfer none): a #GstRTSPTimeRange
+ *
+ * Seek the pipeline of @media to @range. @media must be prepared with
+ * gst_rtsp_media_prepare().
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
+{
+  GstRTSPMediaClass *klass;
+  GstRTSPMediaPrivate *priv;
+  gboolean res;
+  GstClockTime start, stop;
+  GstSeekType start_type, stop_type;
+  GstQuery *query;
+
+  klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+  g_return_val_if_fail (range != NULL, FALSE);
+  g_return_val_if_fail (klass->convert_range != NULL, FALSE);
+
+  priv = media->priv;
+
+  g_rec_mutex_lock (&priv->state_lock);
+  if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
+    goto not_prepared;
+
+  /* Update the seekable state of the pipeline in case it changed */
+  query = gst_query_new_seeking (GST_FORMAT_TIME);
+  if (gst_element_query (priv->pipeline, query)) {
+    GstFormat format;
+    gboolean seekable;
+    gint64 start, end;
+
+    gst_query_parse_seeking (query, &format, &seekable, &start, &end);
+    priv->seekable = seekable;
+  }
+  gst_query_unref (query);
+
+  if (!priv->seekable)
+    goto not_seekable;
+
+  start_type = stop_type = GST_SEEK_TYPE_NONE;
+
+  if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
+    goto not_supported;
+  gst_rtsp_range_get_times (range, &start, &stop);
+
+  GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
+  GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
+
+  if (start != GST_CLOCK_TIME_NONE)
+    start_type = GST_SEEK_TYPE_SET;
+
+  if (priv->range_stop == stop)
+    stop = GST_CLOCK_TIME_NONE;
+  else if (stop != GST_CLOCK_TIME_NONE)
+    stop_type = GST_SEEK_TYPE_SET;
+
+  if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
+    GstSeekFlags flags;
+
+    GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
+        GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
+
+    gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+    if (priv->blocked)
+      media_streams_set_blocked (media, TRUE);
+
+    /* depends on the current playing state of the pipeline. We might need to
+     * queue this until we get EOS. */
+    flags = GST_SEEK_FLAG_FLUSH;
+
+    /* if range start was not supplied we must continue from current position.
+     * but since we're doing a flushing seek, let us query the current position
+     * so we end up at exactly the same position after the seek. */
+    if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
+      gint64 position;
+      gboolean ret = FALSE;
+
+      if (klass->query_position)
+        ret = klass->query_position (media, &position);
+
+      if (!ret) {
+        GST_WARNING ("position query failed");
+      } else {
+        GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
+            GST_TIME_ARGS (position));
+        start = position;
+        start_type = GST_SEEK_TYPE_SET;
+        flags |= GST_SEEK_FLAG_ACCURATE;
+      }
+    } else {
+      /* only set keyframe flag when modifying start */
+      if (start_type != GST_SEEK_TYPE_NONE)
+        flags |= GST_SEEK_FLAG_KEY_UNIT;
+    }
+
+    /* FIXME, we only do forwards playback, no trick modes yet */
+    res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
+        flags, start_type, start, stop_type, stop);
+
+    /* and block for the seek to complete */
+    GST_INFO ("done seeking %d", res);
+    g_rec_mutex_unlock (&priv->state_lock);
+
+    /* wait until pipeline is prerolled again, this will also collect stats */
+    if (!wait_preroll (media))
+      goto preroll_failed;
+
+    g_rec_mutex_lock (&priv->state_lock);
+    GST_INFO ("prerolled again");
+  } else {
+    GST_INFO ("no seek needed");
+    res = TRUE;
+  }
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return res;
+
+  /* ERRORS */
+not_prepared:
+  {
+    g_rec_mutex_unlock (&priv->state_lock);
+    GST_INFO ("media %p is not prepared", media);
+    return FALSE;
+  }
+not_seekable:
+  {
+    g_rec_mutex_unlock (&priv->state_lock);
+    GST_INFO ("pipeline is not seekable");
+    return FALSE;
+  }
+not_supported:
+  {
+    g_rec_mutex_unlock (&priv->state_lock);
+    GST_WARNING ("conversion to npt not supported");
+    return FALSE;
+  }
+preroll_failed:
+  {
+    GST_WARNING ("failed to preroll after seek");
+    return FALSE;
+  }
+}
+
+static void
+stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
+{
+  *blocked &= gst_rtsp_stream_is_blocking (stream);
+}
+
+static gboolean
+media_streams_blocking (GstRTSPMedia * media)
+{
+  gboolean blocking = TRUE;
+
+  g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
+      &blocking);
+
+  return blocking;
+}
+
+static GstStateChangeReturn
+set_state (GstRTSPMedia * media, GstState state)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstStateChangeReturn ret;
+
+  GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
+      media);
+  ret = gst_element_set_state (priv->pipeline, state);
+
+  return ret;
+}
+
+static GstStateChangeReturn
+set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstStateChangeReturn ret;
+
+  GST_INFO ("set target state to %s for media %p",
+      gst_element_state_get_name (state), media);
+  priv->target_state = state;
+
+  g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
+      priv->target_state, NULL);
+
+  if (do_state)
+    ret = set_state (media, state);
+  else
+    ret = GST_STATE_CHANGE_SUCCESS;
+
+  return ret;
+}
+
+/* called with state-lock */
+static gboolean
+default_handle_message (GstRTSPMedia * media, GstMessage * message)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstMessageType type;
+
+  type = GST_MESSAGE_TYPE (message);
+
+  switch (type) {
+    case GST_MESSAGE_STATE_CHANGED:
+      break;
+    case GST_MESSAGE_BUFFERING:
+    {
+      gint percent;
+
+      gst_message_parse_buffering (message, &percent);
+
+      /* no state management needed for live pipelines */
+      if (priv->is_live)
+        break;
+
+      if (percent == 100) {
+        /* a 100% message means buffering is done */
+        priv->buffering = FALSE;
+        /* if the desired state is playing, go back */
+        if (priv->target_state == GST_STATE_PLAYING) {
+          GST_INFO ("Buffering done, setting pipeline to PLAYING");
+          set_state (media, GST_STATE_PLAYING);
+        } else {
+          GST_INFO ("Buffering done");
+        }
+      } else {
+        /* buffering busy */
+        if (priv->buffering == FALSE) {
+          if (priv->target_state == GST_STATE_PLAYING) {
+            /* we were not buffering but PLAYING, PAUSE  the pipeline. */
+            GST_INFO ("Buffering, setting pipeline to PAUSED ...");
+            set_state (media, GST_STATE_PAUSED);
+          } else {
+            GST_INFO ("Buffering ...");
+          }
+        }
+        priv->buffering = TRUE;
+      }
+      break;
+    }
+    case GST_MESSAGE_LATENCY:
+    {
+      gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
+      break;
+    }
+    case GST_MESSAGE_ERROR:
+    {
+      GError *gerror;
+      gchar *debug;
+
+      gst_message_parse_error (message, &gerror, &debug);
+      GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
+      g_error_free (gerror);
+      g_free (debug);
+
+      gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+      break;
+    }
+    case GST_MESSAGE_WARNING:
+    {
+      GError *gerror;
+      gchar *debug;
+
+      gst_message_parse_warning (message, &gerror, &debug);
+      GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
+      g_error_free (gerror);
+      g_free (debug);
+      break;
+    }
+    case GST_MESSAGE_ELEMENT:
+    {
+      const GstStructure *s;
+
+      s = gst_message_get_structure (message);
+      if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
+        GST_DEBUG ("media received blocking message");
+        if (priv->blocked && media_streams_blocking (media)) {
+          GST_DEBUG ("media is blocking");
+          collect_media_stats (media);
+
+          if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+            gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+        }
+      }
+      break;
+    }
+    case GST_MESSAGE_STREAM_STATUS:
+      break;
+    case GST_MESSAGE_ASYNC_DONE:
+      if (priv->adding) {
+        /* when we are dynamically adding pads, the addition of the udpsrc will
+         * temporarily produce ASYNC_DONE messages. We have to ignore them and
+         * wait for the final ASYNC_DONE after everything prerolled */
+        GST_INFO ("%p: ignoring ASYNC_DONE", media);
+      } else {
+        GST_INFO ("%p: got ASYNC_DONE", media);
+        collect_media_stats (media);
+
+        if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+          gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+      }
+      break;
+    case GST_MESSAGE_EOS:
+      GST_INFO ("%p: got EOS", media);
+
+      if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
+        GST_DEBUG ("shutting down after EOS");
+        finish_unprepare (media);
+      }
+      break;
+    default:
+      GST_INFO ("%p: got message type %d (%s)", media, type,
+          gst_message_type_get_name (type));
+      break;
+  }
+  return TRUE;
+}
+
+static gboolean
+bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstRTSPMediaClass *klass;
+  gboolean ret;
+
+  klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+  g_rec_mutex_lock (&priv->state_lock);
+  if (klass->handle_message)
+    ret = klass->handle_message (media, message);
+  else
+    ret = FALSE;
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return ret;
+}
+
+static void
+watch_destroyed (GstRTSPMedia * media)
+{
+  GST_DEBUG_OBJECT (media, "source destroyed");
+  g_object_unref (media);
+}
+
+static GstElement *
+find_payload_element (GstElement * payloader)
+{
+  GstElement *pay = NULL;
+
+  if (GST_IS_BIN (payloader)) {
+    GstIterator *iter;
+    GValue item = { 0 };
+
+    iter = gst_bin_iterate_recurse (GST_BIN (payloader));
+    while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
+      GstElement *element = (GstElement *) g_value_get_object (&item);
+      GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
+      const gchar *klass;
+
+      klass =
+          gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
+      if (klass == NULL)
+        continue;
+
+      if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
+        pay = gst_object_ref (element);
+        g_value_unset (&item);
+        break;
+      }
+      g_value_unset (&item);
+    }
+    gst_iterator_free (iter);
+  } else {
+    pay = g_object_ref (payloader);
+  }
+
+  return pay;
+}
+
+/* called from streaming threads */
+static void
+pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstRTSPStream *stream;
+  GstElement *pay;
+
+  /* find the real payload element */
+  pay = find_payload_element (element);
+  stream = gst_rtsp_media_create_stream (media, pay, pad);
+  gst_object_unref (pay);
+
+  GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
+
+  g_rec_mutex_lock (&priv->state_lock);
+  if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
+    goto not_preparing;
+
+  g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
+
+  /* we will be adding elements below that will cause ASYNC_DONE to be
+   * posted in the bus. We want to ignore those messages until the
+   * pipeline really prerolled. */
+  priv->adding = TRUE;
+
+  /* join the element in the PAUSED state because this callback is
+   * called from the streaming thread and it is PAUSED */
+  if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
+          priv->rtpbin, GST_STATE_PAUSED)) {
+    GST_WARNING ("failed to join bin element");
+  }
+
+  priv->adding = FALSE;
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return;
+
+  /* ERRORS */
+not_preparing:
+  {
+    gst_rtsp_media_remove_stream (media, stream);
+    g_rec_mutex_unlock (&priv->state_lock);
+    GST_INFO ("ignore pad because we are not preparing");
+    return;
+  }
+}
+
+static void
+pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstRTSPStream *stream;
+
+  stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
+  if (stream == NULL)
+    return;
+
+  GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
+
+  g_rec_mutex_lock (&priv->state_lock);
+  gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  gst_rtsp_media_remove_stream (media, stream);
+}
+
+static void
+remove_fakesink (GstRTSPMediaPrivate * priv)
+{
+  GstElement *fakesink;
+
+  g_mutex_lock (&priv->lock);
+  if ((fakesink = priv->fakesink))
+    gst_object_ref (fakesink);
+  priv->fakesink = NULL;
+  g_mutex_unlock (&priv->lock);
+
+  if (fakesink) {
+    gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
+    gst_element_set_state (fakesink, GST_STATE_NULL);
+    gst_object_unref (fakesink);
+    GST_INFO ("removed fakesink");
+  }
+}
+
+static void
+no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+
+  GST_INFO ("no more pads");
+  remove_fakesink (priv);
+}
+
+typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
+
+struct _DynPaySignalHandlers
+{
+  gulong pad_added_handler;
+  gulong pad_removed_handler;
+  gulong no_more_pads_handler;
+};
+
+static gboolean
+start_preroll (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstStateChangeReturn ret;
+
+  GST_INFO ("setting pipeline to PAUSED for media %p", media);
+  /* first go to PAUSED */
+  ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
+
+  switch (ret) {
+    case GST_STATE_CHANGE_SUCCESS:
+      GST_INFO ("SUCCESS state change for media %p", media);
+      priv->seekable = TRUE;
+      break;
+    case GST_STATE_CHANGE_ASYNC:
+      GST_INFO ("ASYNC state change for media %p", media);
+      priv->seekable = TRUE;
+      break;
+    case GST_STATE_CHANGE_NO_PREROLL:
+      /* we need to go to PLAYING */
+      GST_INFO ("NO_PREROLL state change: live media %p", media);
+      /* FIXME we disable seeking for live streams for now. We should perform a
+       * seeking query in preroll instead */
+      priv->seekable = FALSE;
+      priv->is_live = TRUE;
+      /* start blocked  to make sure nothing goes to the sink */
+      media_streams_set_blocked (media, TRUE);
+      ret = set_state (media, GST_STATE_PLAYING);
+      if (ret == GST_STATE_CHANGE_FAILURE)
+        goto state_failed;
+      break;
+    case GST_STATE_CHANGE_FAILURE:
+      goto state_failed;
+  }
+
+  return TRUE;
+
+state_failed:
+  {
+    GST_WARNING ("failed to preroll pipeline");
+    return FALSE;
+  }
+}
+
+static gboolean
+wait_preroll (GstRTSPMedia * media)
+{
+  GstRTSPMediaStatus status;
+
+  GST_DEBUG ("wait to preroll pipeline");
+
+  /* wait until pipeline is prerolled */
+  status = gst_rtsp_media_get_status (media);
+  if (status == GST_RTSP_MEDIA_STATUS_ERROR)
+    goto preroll_failed;
+
+  return TRUE;
+
+preroll_failed:
+  {
+    GST_WARNING ("failed to preroll pipeline");
+    return FALSE;
+  }
+}
+
+static gboolean
+start_prepare (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  guint i;
+  GList *walk;
+
+  /* link streams we already have, other streams might appear when we have
+   * dynamic elements */
+  for (i = 0; i < priv->streams->len; i++) {
+    GstRTSPStream *stream;
+
+    stream = g_ptr_array_index (priv->streams, i);
+
+    if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
+            priv->rtpbin, GST_STATE_NULL)) {
+      goto join_bin_failed;
+    }
+  }
+
+  for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
+    GstElement *elem = walk->data;
+    DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
+
+    GST_INFO ("adding callbacks for dynamic element %p", elem);
+
+    handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
+        (GCallback) pad_added_cb, media);
+    handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
+        (GCallback) pad_removed_cb, media);
+    handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
+        (GCallback) no_more_pads_cb, media);
+
+    g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
+
+    /* we add a fakesink here in order to make the state change async. We remove
+     * the fakesink again in the no-more-pads callback. */
+    priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
+    gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
+  }
+
+  if (!start_preroll (media))
+    goto preroll_failed;
+
+  return FALSE;
+
+join_bin_failed:
+  {
+    GST_WARNING ("failed to join bin element");
+    gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+    return FALSE;
+  }
+preroll_failed:
+  {
+    GST_WARNING ("failed to preroll pipeline");
+    gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+    return FALSE;
+  }
+}
+
+static gboolean
+default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPMediaClass *klass;
+  GstBus *bus;
+  GMainContext *context;
+  GSource *source;
+
+  priv = media->priv;
+
+  klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+  if (!klass->create_rtpbin)
+    goto no_create_rtpbin;
+
+  priv->rtpbin = klass->create_rtpbin (media);
+  if (priv->rtpbin != NULL) {
+    gboolean success = TRUE;
+
+    if (klass->setup_rtpbin)
+      success = klass->setup_rtpbin (media, priv->rtpbin);
+
+    if (success == FALSE) {
+      gst_object_unref (priv->rtpbin);
+      priv->rtpbin = NULL;
+    }
+  }
+  if (priv->rtpbin == NULL)
+    goto no_rtpbin;
+
+  priv->thread = thread;
+  context = (thread != NULL) ? (thread->context) : NULL;
+
+  bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
+
+  /* add the pipeline bus to our custom mainloop */
+  priv->source = gst_bus_create_watch (bus);
+  gst_object_unref (bus);
+
+  g_source_set_callback (priv->source, (GSourceFunc) bus_message,
+      g_object_ref (media), (GDestroyNotify) watch_destroyed);
+
+  priv->id = g_source_attach (priv->source, context);
+
+  /* add stuff to the bin */
+  gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
+
+  /* do remainder in context */
+  source = g_idle_source_new ();
+  g_source_set_callback (source, (GSourceFunc) start_prepare, media, NULL);
+  g_source_attach (source, context);
+  g_source_unref (source);
+
+  return TRUE;
+
+  /* ERRORS */
+no_create_rtpbin:
+  {
+    GST_ERROR ("no create_rtpbin function");
+    g_critical ("no create_rtpbin vmethod function set");
+    return FALSE;
+  }
+no_rtpbin:
+  {
+    GST_WARNING ("no rtpbin element");
+    g_warning ("failed to create element 'rtpbin', check your installation");
+    return FALSE;
+  }
+}
+
+/**
+ * gst_rtsp_media_prepare:
+ * @media: a #GstRTSPMedia
+ * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
+ *   bus handler or %NULL
+ *
+ * Prepare @media for streaming. This function will create the objects
+ * to manage the streaming. A pipeline must have been set on @media with
+ * gst_rtsp_media_take_pipeline().
+ *
+ * It will preroll the pipeline and collect vital information about the streams
+ * such as the duration.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPMediaClass *klass;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+  priv = media->priv;
+
+  g_rec_mutex_lock (&priv->state_lock);
+  priv->prepare_count++;
+
+  if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
+      priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
+    goto was_prepared;
+
+  if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
+    goto is_preparing;
+
+  if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
+    goto not_unprepared;
+
+  if (!priv->reusable && priv->reused)
+    goto is_reused;
+
+  GST_INFO ("preparing media %p", media);
+
+  /* reset some variables */
+  priv->is_live = FALSE;
+  priv->seekable = FALSE;
+  priv->buffering = FALSE;
+
+  /* we're preparing now */
+  gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+
+  klass = GST_RTSP_MEDIA_GET_CLASS (media);
+  if (klass->prepare) {
+    if (!klass->prepare (media, thread))
+      goto prepare_failed;
+  }
+
+wait_status:
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  /* now wait for all pads to be prerolled, FIXME, we should somehow be
+   * able to do this async so that we don't block the server thread. */
+  if (!wait_preroll (media))
+    goto preroll_failed;
+
+  g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
+
+  GST_INFO ("object %p is prerolled", media);
+
+  return TRUE;
+
+  /* OK */
+is_preparing:
+  {
+    /* we are not going to use the giving thread, so stop it. */
+    if (thread)
+      gst_rtsp_thread_stop (thread);
+    goto wait_status;
+  }
+was_prepared:
+  {
+    GST_LOG ("media %p was prepared", media);
+    /* we are not going to use the giving thread, so stop it. */
+    if (thread)
+      gst_rtsp_thread_stop (thread);
+    g_rec_mutex_unlock (&priv->state_lock);
+    return TRUE;
+  }
+  /* ERRORS */
+not_unprepared:
+  {
+    /* we are not going to use the giving thread, so stop it. */
+    if (thread)
+      gst_rtsp_thread_stop (thread);
+    GST_WARNING ("media %p was not unprepared", media);
+    priv->prepare_count--;
+    g_rec_mutex_unlock (&priv->state_lock);
+    return FALSE;
+  }
+is_reused:
+  {
+    /* we are not going to use the giving thread, so stop it. */
+    if (thread)
+      gst_rtsp_thread_stop (thread);
+    priv->prepare_count--;
+    g_rec_mutex_unlock (&priv->state_lock);
+    GST_WARNING ("can not reuse media %p", media);
+    return FALSE;
+  }
+prepare_failed:
+  {
+    /* we are not going to use the giving thread, so stop it. */
+    if (thread)
+      gst_rtsp_thread_stop (thread);
+    priv->prepare_count--;
+    g_rec_mutex_unlock (&priv->state_lock);
+    GST_ERROR ("failed to prepare media");
+    return FALSE;
+  }
+preroll_failed:
+  {
+    GST_WARNING ("failed to preroll pipeline");
+    gst_rtsp_media_unprepare (media);
+    return FALSE;
+  }
+}
+
+/* must be called with state-lock */
+static void
+finish_unprepare (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  gint i;
+  GList *walk;
+
+  GST_DEBUG ("shutting down");
+
+  /* release the lock on shutdown, otherwise pad_added_cb might try to
+   * acquire the lock and then we deadlock */
+  g_rec_mutex_unlock (&priv->state_lock);
+  set_state (media, GST_STATE_NULL);
+  g_rec_mutex_lock (&priv->state_lock);
+  remove_fakesink (priv);
+
+  for (i = 0; i < priv->streams->len; i++) {
+    GstRTSPStream *stream;
+
+    GST_INFO ("Removing elements of stream %d from pipeline", i);
+
+    stream = g_ptr_array_index (priv->streams, i);
+
+    gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
+  }
+
+  /* remove the pad signal handlers */
+  for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
+    GstElement *elem = walk->data;
+    DynPaySignalHandlers *handlers;
+
+    handlers =
+        g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
+    g_assert (handlers != NULL);
+
+    g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
+    g_signal_handler_disconnect (G_OBJECT (elem),
+        handlers->pad_removed_handler);
+    g_signal_handler_disconnect (G_OBJECT (elem),
+        handlers->no_more_pads_handler);
+
+    g_slice_free (DynPaySignalHandlers, handlers);
+  }
+
+  gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
+  priv->rtpbin = NULL;
+
+  if (priv->nettime)
+    gst_object_unref (priv->nettime);
+  priv->nettime = NULL;
+
+  priv->reused = TRUE;
+  gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
+
+  /* when the media is not reusable, this will effectively unref the media and
+   * recreate it */
+  g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
+
+  /* the source has the last ref to the media */
+  if (priv->source) {
+    GST_DEBUG ("destroy source");
+    g_source_destroy (priv->source);
+    g_source_unref (priv->source);
+  }
+  if (priv->thread) {
+    GST_DEBUG ("stop thread");
+    gst_rtsp_thread_stop (priv->thread);
+  }
+}
+
+/* called with state-lock */
+static gboolean
+default_unprepare (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+
+  if (priv->eos_shutdown) {
+    GST_DEBUG ("sending EOS for shutdown");
+    /* ref so that we don't disappear */
+    gst_element_send_event (priv->pipeline, gst_event_new_eos ());
+    /* we need to go to playing again for the EOS to propagate, normally in this
+     * state, nothing is receiving data from us anymore so this is ok. */
+    set_state (media, GST_STATE_PLAYING);
+  } else {
+    finish_unprepare (media);
+  }
+  return TRUE;
+}
+
+/**
+ * gst_rtsp_media_unprepare:
+ * @media: a #GstRTSPMedia
+ *
+ * Unprepare @media. After this call, the media should be prepared again before
+ * it can be used again. If the media is set to be non-reusable, a new instance
+ * must be created.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_unprepare (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv;
+  gboolean success;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+  priv = media->priv;
+
+  g_rec_mutex_lock (&priv->state_lock);
+  if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
+    goto was_unprepared;
+
+  priv->prepare_count--;
+  if (priv->prepare_count > 0)
+    goto is_busy;
+
+  GST_INFO ("unprepare media %p", media);
+  if (priv->blocked)
+    media_streams_set_blocked (media, FALSE);
+  set_target_state (media, GST_STATE_NULL, FALSE);
+  success = TRUE;
+
+  gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
+
+  if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
+    GstRTSPMediaClass *klass;
+
+    klass = GST_RTSP_MEDIA_GET_CLASS (media);
+    if (klass->unprepare)
+      success = klass->unprepare (media);
+  } else {
+    finish_unprepare (media);
+  }
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return success;
+
+was_unprepared:
+  {
+    g_rec_mutex_unlock (&priv->state_lock);
+    GST_INFO ("media %p was already unprepared", media);
+    return TRUE;
+  }
+is_busy:
+  {
+    GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
+    g_rec_mutex_unlock (&priv->state_lock);
+    return TRUE;
+  }
+}
+
+/* should be called with state-lock */
+static GstClock *
+get_clock_unlocked (GstRTSPMedia * media)
+{
+  if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
+    GST_DEBUG_OBJECT (media, "media was not prepared");
+    return NULL;
+  }
+  return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
+}
+
+/**
+ * gst_rtsp_media_get_clock:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the clock that is used by the pipeline in @media.
+ *
+ * @media must be prepared before this method returns a valid clock object.
+ *
+ * Returns: (transfer full): the #GstClock used by @media. unref after usage.
+ */
+GstClock *
+gst_rtsp_media_get_clock (GstRTSPMedia * media)
+{
+  GstClock *clock;
+  GstRTSPMediaPrivate *priv;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+  priv = media->priv;
+
+  g_rec_mutex_lock (&priv->state_lock);
+  clock = get_clock_unlocked (media);
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return clock;
+}
+
+/**
+ * gst_rtsp_media_get_base_time:
+ * @media: a #GstRTSPMedia
+ *
+ * Get the base_time that is used by the pipeline in @media.
+ *
+ * @media must be prepared before this method returns a valid base_time.
+ *
+ * Returns: the base_time used by @media.
+ */
+GstClockTime
+gst_rtsp_media_get_base_time (GstRTSPMedia * media)
+{
+  GstClockTime result;
+  GstRTSPMediaPrivate *priv;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
+
+  priv = media->priv;
+
+  g_rec_mutex_lock (&priv->state_lock);
+  if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
+    goto not_prepared;
+
+  result = gst_element_get_base_time (media->priv->pipeline);
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return result;
+
+  /* ERRORS */
+not_prepared:
+  {
+    g_rec_mutex_unlock (&priv->state_lock);
+    GST_DEBUG_OBJECT (media, "media was not prepared");
+    return GST_CLOCK_TIME_NONE;
+  }
+}
+
+/**
+ * gst_rtsp_media_get_time_provider:
+ * @media: a #GstRTSPMedia
+ * @address: (allow-none): an address or %NULL
+ * @port: a port or 0
+ *
+ * Get the #GstNetTimeProvider for the clock used by @media. The time provider
+ * will listen on @address and @port for client time requests.
+ *
+ * Returns: (transfer full): the #GstNetTimeProvider of @media.
+ */
+GstNetTimeProvider *
+gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
+    guint16 port)
+{
+  GstRTSPMediaPrivate *priv;
+  GstNetTimeProvider *provider = NULL;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+
+  priv = media->priv;
+
+  g_rec_mutex_lock (&priv->state_lock);
+  if (priv->time_provider) {
+    if ((provider = priv->nettime) == NULL) {
+      GstClock *clock;
+
+      if (priv->time_provider && (clock = get_clock_unlocked (media))) {
+        provider = gst_net_time_provider_new (clock, address, port);
+        gst_object_unref (clock);
+
+        priv->nettime = provider;
+      }
+    }
+  }
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  if (provider)
+    gst_object_ref (provider);
+
+  return provider;
+}
+
+static gboolean
+default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
+{
+  return gst_rtsp_sdp_from_media (sdp, info, media);
+}
+
+/**
+ * gst_rtsp_media_setup_sdp:
+ * @media: a #GstRTSPMedia
+ * @sdp: (transfer none): a #GstSDPMessage
+ * @info: (transfer none): a #GstSDPInfo
+ *
+ * Add @media specific info to @sdp. @info is used to configure the connection
+ * information in the SDP.
+ *
+ * Returns: TRUE on success.
+ */
+gboolean
+gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+    GstSDPInfo * info)
+{
+  GstRTSPMediaPrivate *priv;
+  GstRTSPMediaClass *klass;
+  gboolean res;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+  g_return_val_if_fail (sdp != NULL, FALSE);
+  g_return_val_if_fail (info != NULL, FALSE);
+
+  priv = media->priv;
+
+  g_rec_mutex_lock (&priv->state_lock);
+
+  klass = GST_RTSP_MEDIA_GET_CLASS (media);
+
+  if (!klass->setup_sdp)
+    goto no_setup_sdp;
+
+  res = klass->setup_sdp (media, sdp, info);
+
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return res;
+
+  /* ERRORS */
+no_setup_sdp:
+  {
+    g_rec_mutex_unlock (&priv->state_lock);
+    GST_ERROR ("no setup_sdp function");
+    g_critical ("no setup_sdp vmethod function set");
+    return FALSE;
+  }
+}
+
+static void
+do_set_seqnum (GstRTSPStream * stream)
+{
+  guint16 seq_num;
+  seq_num = gst_rtsp_stream_get_current_seqnum (stream);
+  gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
+}
+
+/* call with state_lock */
+gboolean
+default_suspend (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstStateChangeReturn ret;
+
+  switch (priv->suspend_mode) {
+    case GST_RTSP_SUSPEND_MODE_NONE:
+      GST_DEBUG ("media %p no suspend", media);
+      break;
+    case GST_RTSP_SUSPEND_MODE_PAUSE:
+      GST_DEBUG ("media %p suspend to PAUSED", media);
+      ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
+      if (ret == GST_STATE_CHANGE_FAILURE)
+        goto state_failed;
+      break;
+    case GST_RTSP_SUSPEND_MODE_RESET:
+      GST_DEBUG ("media %p suspend to NULL", media);
+      ret = set_target_state (media, GST_STATE_NULL, TRUE);
+      if (ret == GST_STATE_CHANGE_FAILURE)
+        goto state_failed;
+      /* Because payloader needs to set the sequence number as
+       * monotonic, we need to preserve the sequence number
+       * after pause. (otherwise going from pause to play,  which
+       * is actually from NULL to PLAY will create a new sequence
+       * number. */
+      g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
+      break;
+    default:
+      break;
+  }
+
+  /* let the streams do the state changes freely, if any */
+  media_streams_set_blocked (media, FALSE);
+
+  return TRUE;
+
+  /* ERRORS */
+state_failed:
+  {
+    GST_WARNING ("failed changing pipeline's state for media %p", media);
+    return FALSE;
+  }
+}
+
+/**
+ * gst_rtsp_media_suspend:
+ * @media: a #GstRTSPMedia
+ *
+ * Suspend @media. The state of the pipeline managed by @media is set to
+ * GST_STATE_NULL but all streams are kept. @media can be prepared again
+ * with gst_rtsp_media_unsuspend()
+ *
+ * @media must be prepared with gst_rtsp_media_prepare();
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_suspend (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstRTSPMediaClass *klass;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+  GST_FIXME ("suspend for dynamic pipelines needs fixing");
+
+  g_rec_mutex_lock (&priv->state_lock);
+  if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
+    goto not_prepared;
+
+  /* don't attempt to suspend when something is busy */
+  if (priv->n_active > 0)
+    goto done;
+
+  klass = GST_RTSP_MEDIA_GET_CLASS (media);
+  if (klass->suspend) {
+    if (!klass->suspend (media))
+      goto suspend_failed;
+  }
+
+  gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
+done:
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return TRUE;
+
+  /* ERRORS */
+not_prepared:
+  {
+    g_rec_mutex_unlock (&priv->state_lock);
+    GST_WARNING ("media %p was not prepared", media);
+    return FALSE;
+  }
+suspend_failed:
+  {
+    g_rec_mutex_unlock (&priv->state_lock);
+    gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+    GST_WARNING ("failed to suspend media %p", media);
+    return FALSE;
+  }
+}
+
+/* call with state_lock */
+gboolean
+default_unsuspend (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+
+  switch (priv->suspend_mode) {
+    case GST_RTSP_SUSPEND_MODE_NONE:
+      gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+      break;
+    case GST_RTSP_SUSPEND_MODE_PAUSE:
+      gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
+      break;
+    case GST_RTSP_SUSPEND_MODE_RESET:
+    {
+      gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
+      if (!start_preroll (media))
+        goto start_failed;
+      g_rec_mutex_unlock (&priv->state_lock);
+
+      if (!wait_preroll (media))
+        goto preroll_failed;
+
+      g_rec_mutex_lock (&priv->state_lock);
+    }
+    default:
+      break;
+  }
+
+  return TRUE;
+
+  /* ERRORS */
+start_failed:
+  {
+    GST_WARNING ("failed to preroll pipeline");
+    return FALSE;
+  }
+preroll_failed:
+  {
+    GST_WARNING ("failed to preroll pipeline");
+    return FALSE;
+  }
+}
+
+/**
+ * gst_rtsp_media_unsuspend:
+ * @media: a #GstRTSPMedia
+ *
+ * Unsuspend @media if it was in a suspended state. This method does nothing
+ * when the media was not in the suspended state.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_unsuspend (GstRTSPMedia * media)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+  GstRTSPMediaClass *klass;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+
+  g_rec_mutex_lock (&priv->state_lock);
+  if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
+    goto done;
+
+  klass = GST_RTSP_MEDIA_GET_CLASS (media);
+  if (klass->unsuspend) {
+    if (!klass->unsuspend (media))
+      goto unsuspend_failed;
+  }
+
+done:
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return TRUE;
+
+  /* ERRORS */
+unsuspend_failed:
+  {
+    g_rec_mutex_unlock (&priv->state_lock);
+    GST_WARNING ("failed to unsuspend media %p", media);
+    gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
+    return FALSE;
+  }
+}
+
+/* must be called with state-lock */
+static void
+media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
+{
+  GstRTSPMediaPrivate *priv = media->priv;
+
+  if (state == GST_STATE_NULL) {
+    gst_rtsp_media_unprepare (media);
+  } else {
+    GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
+    set_target_state (media, state, FALSE);
+    /* when we are buffering, don't update the state yet, this will be done
+     * when buffering finishes */
+    if (priv->buffering) {
+      GST_INFO ("Buffering busy, delay state change");
+    } else {
+      if (state == GST_STATE_PLAYING)
+        /* make sure pads are not blocking anymore when going to PLAYING */
+        media_streams_set_blocked (media, FALSE);
+
+      set_state (media, state);
+
+      /* and suspend after pause */
+      if (state == GST_STATE_PAUSED)
+        gst_rtsp_media_suspend (media);
+    }
+  }
+}
+
+/**
+ * gst_rtsp_media_set_pipeline_state:
+ * @media: a #GstRTSPMedia
+ * @state: the target state of the pipeline
+ *
+ * Set the state of the pipeline managed by @media to @state
+ */
+void
+gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
+{
+  g_return_if_fail (GST_IS_RTSP_MEDIA (media));
+
+  g_rec_mutex_lock (&media->priv->state_lock);
+  media_set_pipeline_state_locked (media, state);
+  g_rec_mutex_unlock (&media->priv->state_lock);
+}
+
+/**
+ * gst_rtsp_media_set_state:
+ * @media: a #GstRTSPMedia
+ * @state: the target state of the media
+ * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
+ * a #GPtrArray of #GstRTSPStreamTransport pointers
+ *
+ * Set the state of @media to @state and for the transports in @transports.
+ *
+ * @media must be prepared with gst_rtsp_media_prepare();
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
+    GPtrArray * transports)
+{
+  GstRTSPMediaPrivate *priv;
+  gint i;
+  gboolean activate, deactivate, do_state;
+  gint old_active;
+
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
+  g_return_val_if_fail (transports != NULL, FALSE);
+
+  priv = media->priv;
+
+  g_rec_mutex_lock (&priv->state_lock);
+  if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
+    goto error_status;
+  if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
+      priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
+    goto not_prepared;
+
+  /* NULL and READY are the same */
+  if (state == GST_STATE_READY)
+    state = GST_STATE_NULL;
+
+  activate = deactivate = FALSE;
+
+  GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
+      media);
+
+  switch (state) {
+    case GST_STATE_NULL:
+    case GST_STATE_PAUSED:
+      /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
+      if (priv->target_state == GST_STATE_PLAYING)
+        deactivate = TRUE;
+      break;
+    case GST_STATE_PLAYING:
+      /* we're going to PLAYING, activate */
+      activate = TRUE;
+      break;
+    default:
+      break;
+  }
+  old_active = priv->n_active;
+
+  for (i = 0; i < transports->len; i++) {
+    GstRTSPStreamTransport *trans;
+
+    /* we need a non-NULL entry in the array */
+    trans = g_ptr_array_index (transports, i);
+    if (trans == NULL)
+      continue;
+
+    if (activate) {
+      if (gst_rtsp_stream_transport_set_active (trans, TRUE))
+        priv->n_active++;
+    } else if (deactivate) {
+      if (gst_rtsp_stream_transport_set_active (trans, FALSE))
+        priv->n_active--;
+    }
+  }
+
+  /* we just activated the first media, do the playing state change */
+  if (old_active == 0 && activate)
+    do_state = TRUE;
+  /* if we have no more active media, do the downward state changes */
+  else if (priv->n_active == 0)
+    do_state = TRUE;
+  else
+    do_state = FALSE;
+
+  GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
+      media, do_state);
+
+  if (priv->target_state != state) {
+    if (do_state)
+      media_set_pipeline_state_locked (media, state);
+
+    g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
+        NULL);
+  }
+
+  /* remember where we are */
+  if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
+          old_active != priv->n_active))
+    collect_media_stats (media);
+
+  g_rec_mutex_unlock (&priv->state_lock);
+
+  return TRUE;
+
+  /* ERRORS */
+not_prepared:
+  {
+    GST_WARNING ("media %p was not prepared", media);
+    g_rec_mutex_unlock (&priv->state_lock);
+    return FALSE;
+  }
+error_status:
+  {
+    GST_WARNING ("media %p in error status while changing to state %d",
+        media, state);
+    if (state == GST_STATE_NULL) {
+      for (i = 0; i < transports->len; i++) {
+        GstRTSPStreamTransport *trans;
+
+        /* we need a non-NULL entry in the array */
+        trans = g_ptr_array_index (transports, i);
+        if (trans == NULL)
+          continue;
+
+        gst_rtsp_stream_transport_set_active (trans, FALSE);
+      }
+      priv->n_active = 0;
+    }
+    g_rec_mutex_unlock (&priv->state_lock);
+    return FALSE;
+  }
+}
diff --git a/gst/rtsp-server/rtsp-media.h b/gst/rtsp-server/rtsp-media.h
new file mode 100644
index 0000000..a78650d
--- /dev/null
+++ b/gst/rtsp-server/rtsp-media.h
@@ -0,0 +1,235 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+#include 
+#include 
+#include 
+
+#ifndef __GST_RTSP_MEDIA_H__
+#define __GST_RTSP_MEDIA_H__
+
+G_BEGIN_DECLS
+
+/* types for the media */
+#define GST_TYPE_RTSP_MEDIA              (gst_rtsp_media_get_type ())
+#define GST_IS_RTSP_MEDIA(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA))
+#define GST_IS_RTSP_MEDIA_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA))
+#define GST_RTSP_MEDIA_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
+#define GST_RTSP_MEDIA(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMedia))
+#define GST_RTSP_MEDIA_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
+#define GST_RTSP_MEDIA_CAST(obj)         ((GstRTSPMedia*)(obj))
+#define GST_RTSP_MEDIA_CLASS_CAST(klass) ((GstRTSPMediaClass*)(klass))
+
+typedef struct _GstRTSPMedia GstRTSPMedia;
+typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
+typedef struct _GstRTSPMediaPrivate GstRTSPMediaPrivate;
+
+/**
+ * GstRTSPMediaStatus:
+ * @GST_RTSP_MEDIA_STATUS_UNPREPARED: media pipeline not prerolled
+ * @GST_RTSP_MEDIA_STATUS_UNPREPARING: media pipeline is busy doing a clean
+ *                                     shutdown.
+ * @GST_RTSP_MEDIA_STATUS_PREPARING: media pipeline is prerolling
+ * @GST_RTSP_MEDIA_STATUS_PREPARED: media pipeline is prerolled
+ * @GST_RTSP_MEDIA_STATUS_SUSPENDED: media is suspended
+ * @GST_RTSP_MEDIA_STATUS_ERROR: media pipeline is in error
+ *
+ * The state of the media pipeline.
+ */
+typedef enum {
+  GST_RTSP_MEDIA_STATUS_UNPREPARED  = 0,
+  GST_RTSP_MEDIA_STATUS_UNPREPARING = 1,
+  GST_RTSP_MEDIA_STATUS_PREPARING   = 2,
+  GST_RTSP_MEDIA_STATUS_PREPARED    = 3,
+  GST_RTSP_MEDIA_STATUS_SUSPENDED   = 4,
+  GST_RTSP_MEDIA_STATUS_ERROR       = 5
+} GstRTSPMediaStatus;
+
+/**
+ * GstRTSPSuspendMode:
+ * @GST_RTSP_SUSPEND_MODE_NONE: Media is not suspended
+ * @GST_RTSP_SUSPEND_MODE_PAUSE: Media is PAUSED in suspend
+ * @GST_RTSP_SUSPEND_MODE_RESET: The media is set to NULL when suspended
+ *
+ * The suspend mode of the media pipeline. A media pipeline is suspended right
+ * after creating the SDP and when the client performs a PAUSED request.
+ */
+typedef enum {
+  GST_RTSP_SUSPEND_MODE_NONE   = 0,
+  GST_RTSP_SUSPEND_MODE_PAUSE  = 1,
+  GST_RTSP_SUSPEND_MODE_RESET  = 2
+} GstRTSPSuspendMode;
+
+#define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type())
+GType gst_rtsp_suspend_mode_get_type (void);
+
+#include "rtsp-stream.h"
+#include "rtsp-thread-pool.h"
+#include "rtsp-permissions.h"
+#include "rtsp-address-pool.h"
+#include "rtsp-sdp.h"
+
+/**
+ * GstRTSPMedia:
+ *
+ * A class that contains the GStreamer element along with a list of
+ * #GstRTSPStream objects that can produce data.
+ *
+ * This object is usually created from a #GstRTSPMediaFactory.
+ */
+struct _GstRTSPMedia {
+  GObject            parent;
+
+  /*< private >*/
+  GstRTSPMediaPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPMediaClass:
+ * @handle_message: handle a message
+ * @prepare: the default implementation adds all elements and sets the
+ *           pipeline's state to GST_STATE_PAUSED (or GST_STATE_PLAYING
+ *           in case of NO_PREROLL elements).
+ * @unprepare: the default implementation sets the pipeline's state
+ *             to GST_STATE_NULL and removes all elements.
+ * @suspend: the default implementation sets the pipeline's state to
+ *           GST_STATE_NULL GST_STATE_PAUSED depending on the selected
+ *           suspend mode.
+ * @unsuspend: the default implementation reverts the suspend operation.
+ *             The pipeline will be prerolled again if it's state was
+ *             set to GST_STATE_NULL in suspend.
+ * @convert_range: convert a range to the given unit
+ * @query_position: query the current position in the pipeline
+ * @query_stop: query when playback will stop
+ *
+ * The RTSP media class
+ */
+struct _GstRTSPMediaClass {
+  GObjectClass  parent_class;
+
+  /* vmethods */
+  gboolean        (*handle_message)  (GstRTSPMedia *media, GstMessage *message);
+  gboolean        (*prepare)         (GstRTSPMedia *media, GstRTSPThread *thread);
+  gboolean        (*unprepare)       (GstRTSPMedia *media);
+  gboolean        (*suspend)         (GstRTSPMedia *media);
+  gboolean        (*unsuspend)       (GstRTSPMedia *media);
+  gboolean        (*convert_range)   (GstRTSPMedia *media, GstRTSPTimeRange *range,
+                                      GstRTSPRangeUnit unit);
+  gboolean        (*query_position)  (GstRTSPMedia *media, gint64 *position);
+  gboolean        (*query_stop)      (GstRTSPMedia *media, gint64 *stop);
+  GstElement *    (*create_rtpbin)   (GstRTSPMedia *media);
+  gboolean        (*setup_rtpbin)    (GstRTSPMedia *media, GstElement *rtpbin);
+  gboolean        (*setup_sdp)       (GstRTSPMedia *media, GstSDPMessage *sdp, GstSDPInfo *info);
+
+  /* signals */
+  void            (*new_stream)      (GstRTSPMedia *media, GstRTSPStream * stream);
+  void            (*removed_stream)  (GstRTSPMedia *media, GstRTSPStream * stream);
+
+  void            (*prepared)        (GstRTSPMedia *media);
+  void            (*unprepared)      (GstRTSPMedia *media);
+
+  void            (*target_state)    (GstRTSPMedia *media, GstState state);
+  void            (*new_state)       (GstRTSPMedia *media, GstState state);
+
+  /*< private >*/
+  gpointer         _gst_reserved[GST_PADDING_LARGE];
+};
+
+GType                 gst_rtsp_media_get_type         (void);
+
+/* creating the media */
+GstRTSPMedia *        gst_rtsp_media_new              (GstElement *element);
+GstElement *          gst_rtsp_media_get_element      (GstRTSPMedia *media);
+
+void                  gst_rtsp_media_take_pipeline    (GstRTSPMedia *media, GstPipeline *pipeline);
+
+GstRTSPMediaStatus    gst_rtsp_media_get_status       (GstRTSPMedia *media);
+
+void                  gst_rtsp_media_set_permissions  (GstRTSPMedia *media,
+                                                       GstRTSPPermissions *permissions);
+GstRTSPPermissions *  gst_rtsp_media_get_permissions  (GstRTSPMedia *media);
+
+void                  gst_rtsp_media_set_shared       (GstRTSPMedia *media, gboolean shared);
+gboolean              gst_rtsp_media_is_shared        (GstRTSPMedia *media);
+
+void                  gst_rtsp_media_set_reusable     (GstRTSPMedia *media, gboolean reusable);
+gboolean              gst_rtsp_media_is_reusable      (GstRTSPMedia *media);
+
+void                  gst_rtsp_media_set_profiles     (GstRTSPMedia *media, GstRTSPProfile profiles);
+GstRTSPProfile        gst_rtsp_media_get_profiles     (GstRTSPMedia *media);
+
+void                  gst_rtsp_media_set_protocols    (GstRTSPMedia *media, GstRTSPLowerTrans protocols);
+GstRTSPLowerTrans     gst_rtsp_media_get_protocols    (GstRTSPMedia *media);
+
+void                  gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media, gboolean eos_shutdown);
+gboolean              gst_rtsp_media_is_eos_shutdown  (GstRTSPMedia *media);
+
+void                  gst_rtsp_media_set_address_pool (GstRTSPMedia *media, GstRTSPAddressPool *pool);
+GstRTSPAddressPool *  gst_rtsp_media_get_address_pool (GstRTSPMedia *media);
+
+void                  gst_rtsp_media_set_buffer_size  (GstRTSPMedia *media, guint size);
+guint                 gst_rtsp_media_get_buffer_size  (GstRTSPMedia *media);
+
+void                  gst_rtsp_media_use_time_provider (GstRTSPMedia *media, gboolean time_provider);
+gboolean              gst_rtsp_media_is_time_provider  (GstRTSPMedia *media);
+GstNetTimeProvider *  gst_rtsp_media_get_time_provider (GstRTSPMedia *media,
+                                                        const gchar *address, guint16 port);
+
+/* prepare the media for playback */
+gboolean              gst_rtsp_media_prepare          (GstRTSPMedia *media, GstRTSPThread *thread);
+gboolean              gst_rtsp_media_unprepare        (GstRTSPMedia *media);
+
+void                  gst_rtsp_media_set_suspend_mode (GstRTSPMedia *media, GstRTSPSuspendMode mode);
+GstRTSPSuspendMode    gst_rtsp_media_get_suspend_mode (GstRTSPMedia *media);
+
+gboolean              gst_rtsp_media_suspend          (GstRTSPMedia *media);
+gboolean              gst_rtsp_media_unsuspend        (GstRTSPMedia *media);
+
+gboolean              gst_rtsp_media_setup_sdp        (GstRTSPMedia * media, GstSDPMessage * sdp,
+                                                       GstSDPInfo * info);
+
+/* creating streams */
+void                  gst_rtsp_media_collect_streams  (GstRTSPMedia *media);
+GstRTSPStream *       gst_rtsp_media_create_stream    (GstRTSPMedia *media,
+                                                       GstElement *payloader,
+                                                       GstPad *srcpad);
+
+/* dealing with the media */
+GstClock *            gst_rtsp_media_get_clock        (GstRTSPMedia *media);
+GstClockTime          gst_rtsp_media_get_base_time    (GstRTSPMedia *media);
+
+guint                 gst_rtsp_media_n_streams        (GstRTSPMedia *media);
+GstRTSPStream *       gst_rtsp_media_get_stream       (GstRTSPMedia *media, guint idx);
+GstRTSPStream *       gst_rtsp_media_find_stream      (GstRTSPMedia *media, const gchar * control);
+
+gboolean              gst_rtsp_media_seek             (GstRTSPMedia *media, GstRTSPTimeRange *range);
+gchar *               gst_rtsp_media_get_range_string (GstRTSPMedia *media,
+                                                       gboolean play,
+                                                       GstRTSPRangeUnit unit);
+
+gboolean              gst_rtsp_media_set_state        (GstRTSPMedia *media, GstState state,
+                                                       GPtrArray *transports);
+void                  gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media,
+                                                         GstState state);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_MEDIA_H__ */
diff --git a/gst/rtsp-server/rtsp-mount-points.c b/gst/rtsp-server/rtsp-mount-points.c
new file mode 100644
index 0000000..c6a8c06
--- /dev/null
+++ b/gst/rtsp-server/rtsp-mount-points.c
@@ -0,0 +1,370 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-mount-points
+ * @short_description: Map a path to media
+ * @see_also: #GstRTSPMediaFactory, #GstRTSPClient
+ *
+ * A #GstRTSPMountPoints object maintains a relation between paths
+ * and #GstRTSPMediaFactory objects. This object is usually given to
+ * #GstRTSPClient and used to find the media attached to a path.
+ *
+ * With gst_rtsp_mount_points_add_factory () and
+ * gst_rtsp_mount_points_remove_factory(), factories can be added and
+ * removed.
+ *
+ * With gst_rtsp_mount_points_match() you can find the #GstRTSPMediaFactory
+ * object that completely matches the given path.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#include 
+
+#include "rtsp-mount-points.h"
+
+#define GST_RTSP_MOUNT_POINTS_GET_PRIVATE(obj)  \
+       (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MOUNT_POINTS, GstRTSPMountPointsPrivate))
+
+typedef struct
+{
+  gchar *path;
+  gint len;
+  GstRTSPMediaFactory *factory;
+} DataItem;
+
+static DataItem *
+data_item_new (gchar * path, gint len, GstRTSPMediaFactory * factory)
+{
+  DataItem *item;
+
+  item = g_slice_alloc (sizeof (DataItem));
+  item->path = path;
+  item->len = len;
+  item->factory = factory;
+
+  return item;
+}
+
+static void
+data_item_free (gpointer data)
+{
+  DataItem *item = data;
+
+  g_free (item->path);
+  g_object_unref (item->factory);
+  g_slice_free1 (sizeof (DataItem), item);
+}
+
+static void
+data_item_dump (gconstpointer a, gconstpointer prefix)
+{
+  const DataItem *item = a;
+
+  GST_DEBUG ("%s%s %p", (gchar *) prefix, item->path, item->factory);
+}
+
+static gint
+data_item_compare (gconstpointer a, gconstpointer b, gpointer user_data)
+{
+  const DataItem *item1 = a, *item2 = b;
+  gint res;
+
+  res = g_strcmp0 (item1->path, item2->path);
+
+  return res;
+}
+
+struct _GstRTSPMountPointsPrivate
+{
+  GMutex lock;
+  GSequence *mounts;            /* protected by lock */
+  gboolean dirty;
+};
+
+G_DEFINE_TYPE (GstRTSPMountPoints, gst_rtsp_mount_points, G_TYPE_OBJECT);
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
+#define GST_CAT_DEFAULT rtsp_media_debug
+
+static gchar *default_make_path (GstRTSPMountPoints * mounts,
+    const GstRTSPUrl * url);
+static void gst_rtsp_mount_points_finalize (GObject * obj);
+
+static void
+gst_rtsp_mount_points_class_init (GstRTSPMountPointsClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPMountPointsPrivate));
+
+  gobject_class = G_OBJECT_CLASS (klass);
+
+  gobject_class->finalize = gst_rtsp_mount_points_finalize;
+
+  klass->make_path = default_make_path;
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmountpoints", 0,
+      "GstRTSPMountPoints");
+}
+
+static void
+gst_rtsp_mount_points_init (GstRTSPMountPoints * mounts)
+{
+  GstRTSPMountPointsPrivate *priv = GST_RTSP_MOUNT_POINTS_GET_PRIVATE (mounts);
+
+  GST_DEBUG_OBJECT (mounts, "created");
+
+  mounts->priv = priv;
+
+  g_mutex_init (&priv->lock);
+  priv->mounts = g_sequence_new (data_item_free);
+  priv->dirty = FALSE;
+}
+
+static void
+gst_rtsp_mount_points_finalize (GObject * obj)
+{
+  GstRTSPMountPoints *mounts = GST_RTSP_MOUNT_POINTS (obj);
+  GstRTSPMountPointsPrivate *priv = mounts->priv;
+
+  GST_DEBUG_OBJECT (mounts, "finalized");
+
+  g_sequence_free (priv->mounts);
+  g_mutex_clear (&priv->lock);
+
+  G_OBJECT_CLASS (gst_rtsp_mount_points_parent_class)->finalize (obj);
+}
+
+/**
+ * gst_rtsp_mount_points_new:
+ *
+ * Make a new mount points object.
+ *
+ * Returns: (transfer full): a new #GstRTSPMountPoints
+ */
+GstRTSPMountPoints *
+gst_rtsp_mount_points_new (void)
+{
+  GstRTSPMountPoints *result;
+
+  result = g_object_new (GST_TYPE_RTSP_MOUNT_POINTS, NULL);
+
+  return result;
+}
+
+static gchar *
+default_make_path (GstRTSPMountPoints * mounts, const GstRTSPUrl * url)
+{
+  return g_strdup (url->abspath);
+}
+
+/**
+ * gst_rtsp_mount_points_make_path:
+ * @mounts: a #GstRTSPMountPoints
+ * @url: a #GstRTSPUrl
+ *
+ * Make a path string from @url.
+ *
+ * Returns: (transfer full): a path string for @url, g_free() after usage.
+ */
+gchar *
+gst_rtsp_mount_points_make_path (GstRTSPMountPoints * mounts,
+    const GstRTSPUrl * url)
+{
+  GstRTSPMountPointsClass *klass;
+  gchar *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_MOUNT_POINTS (mounts), NULL);
+  g_return_val_if_fail (url != NULL, NULL);
+
+  klass = GST_RTSP_MOUNT_POINTS_GET_CLASS (mounts);
+
+  if (klass->make_path)
+    result = klass->make_path (mounts, url);
+  else
+    result = NULL;
+
+  return result;
+}
+
+static gboolean
+has_prefix (DataItem * str, DataItem * prefix)
+{
+  /* prefix needs to be smaller than str */
+  if (str->len < prefix->len)
+    return FALSE;
+
+  /* if str is larger, it there should be a / following the prefix */
+  if (str->len > prefix->len && str->path[prefix->len] != '/')
+    return FALSE;
+
+  return strncmp (str->path, prefix->path, prefix->len) == 0;
+}
+
+/**
+ * gst_rtsp_mount_points_match:
+ * @mounts: a #GstRTSPMountPoints
+ * @path: a mount point
+ * @matched: (out) (allow-none): the amount of @path matched
+ *
+ * Find the factory in @mounts that has the longest match with @path.
+ *
+ * If @matched is %NULL, @path will match the factory exactly otherwise
+ * the amount of characters that matched is returned in @matched.
+ *
+ * Returns: (transfer full): the #GstRTSPMediaFactory for @path.
+ * g_object_unref() after usage.
+ */
+GstRTSPMediaFactory *
+gst_rtsp_mount_points_match (GstRTSPMountPoints * mounts,
+    const gchar * path, gint * matched)
+{
+  GstRTSPMountPointsPrivate *priv;
+  GstRTSPMediaFactory *result = NULL;
+  GSequenceIter *iter, *best;
+  DataItem item, *ritem;
+
+  g_return_val_if_fail (GST_IS_RTSP_MOUNT_POINTS (mounts), NULL);
+  g_return_val_if_fail (path != NULL, NULL);
+
+  priv = mounts->priv;
+
+  item.path = (gchar *) path;
+  item.len = strlen (path);
+
+  g_mutex_lock (&priv->lock);
+  if (priv->dirty) {
+    g_sequence_sort (priv->mounts, data_item_compare, mounts);
+    g_sequence_foreach (priv->mounts, (GFunc) data_item_dump,
+        (gpointer) "sort :");
+    priv->dirty = FALSE;
+  }
+
+  /* find the location of the media in the hashtable we only use the absolute
+   * path of the uri to find a media factory. If the factory depends on other
+   * properties found in the url, this method should be overridden. */
+  iter = g_sequence_get_begin_iter (priv->mounts);
+  best = NULL;
+  while (!g_sequence_iter_is_end (iter)) {
+    ritem = g_sequence_get (iter);
+
+    data_item_dump (ritem, "inspect: ");
+
+    if (best == NULL) {
+      if (has_prefix (&item, ritem)) {
+        data_item_dump (ritem, "prefix: ");
+        best = iter;
+      }
+    } else {
+      if (!has_prefix (&item, ritem))
+        break;
+
+      best = iter;
+      data_item_dump (ritem, "new best: ");
+    }
+    iter = g_sequence_iter_next (iter);
+  }
+  if (best) {
+    ritem = g_sequence_get (best);
+    data_item_dump (ritem, "result: ");
+    if (matched || ritem->len == item.len) {
+      result = g_object_ref (ritem->factory);
+      if (matched)
+        *matched = ritem->len;
+    }
+  }
+  g_mutex_unlock (&priv->lock);
+
+  GST_INFO ("found media factory %p for path %s", result, path);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_mount_points_add_factory:
+ * @mounts: a #GstRTSPMountPoints
+ * @path: a mount point
+ * @factory: (transfer full): a #GstRTSPMediaFactory
+ *
+ * Attach @factory to the mount point @path in @mounts.
+ *
+ * @path is of the form (/node)+. Any previous mount point will be freed.
+ *
+ * Ownership is taken of the reference on @factory so that @factory should not be
+ * used after calling this function.
+ */
+void
+gst_rtsp_mount_points_add_factory (GstRTSPMountPoints * mounts,
+    const gchar * path, GstRTSPMediaFactory * factory)
+{
+  GstRTSPMountPointsPrivate *priv;
+  DataItem *item;
+
+  g_return_if_fail (GST_IS_RTSP_MOUNT_POINTS (mounts));
+  g_return_if_fail (GST_IS_RTSP_MEDIA_FACTORY (factory));
+  g_return_if_fail (path != NULL);
+
+  priv = mounts->priv;
+
+  item = data_item_new (g_strdup (path), strlen (path), factory);
+
+  GST_INFO ("adding media factory %p for path %s", factory, path);
+
+  g_mutex_lock (&priv->lock);
+  g_sequence_append (priv->mounts, item);
+  priv->dirty = TRUE;
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_mount_points_remove_factory:
+ * @mounts: a #GstRTSPMountPoints
+ * @path: a mount point
+ *
+ * Remove the #GstRTSPMediaFactory associated with @path in @mounts.
+ */
+void
+gst_rtsp_mount_points_remove_factory (GstRTSPMountPoints * mounts,
+    const gchar * path)
+{
+  GstRTSPMountPointsPrivate *priv;
+  DataItem item;
+  GSequenceIter *iter;
+
+  g_return_if_fail (GST_IS_RTSP_MOUNT_POINTS (mounts));
+  g_return_if_fail (path != NULL);
+
+  priv = mounts->priv;
+
+  item.path = (gchar *) path;
+
+  GST_INFO ("removing media factory for path %s", path);
+
+  g_mutex_lock (&priv->lock);
+  if (priv->dirty) {
+    g_sequence_sort (priv->mounts, data_item_compare, mounts);
+    priv->dirty = FALSE;
+  }
+  iter = g_sequence_lookup (priv->mounts, &item, data_item_compare, mounts);
+  if (iter) {
+    g_sequence_remove (iter);
+    priv->dirty = TRUE;
+  }
+  g_mutex_unlock (&priv->lock);
+}
diff --git a/gst/rtsp-server/rtsp-mount-points.h b/gst/rtsp-server/rtsp-mount-points.h
new file mode 100644
index 0000000..ed13d59
--- /dev/null
+++ b/gst/rtsp-server/rtsp-mount-points.h
@@ -0,0 +1,91 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include "rtsp-media-factory.h"
+
+#ifndef __GST_RTSP_MOUNT_POINTS_H__
+#define __GST_RTSP_MOUNT_POINTS_H__
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_MOUNT_POINTS              (gst_rtsp_mount_points_get_type ())
+#define GST_IS_RTSP_MOUNT_POINTS(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MOUNT_POINTS))
+#define GST_IS_RTSP_MOUNT_POINTS_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MOUNT_POINTS))
+#define GST_RTSP_MOUNT_POINTS_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MOUNT_POINTS, GstRTSPMountPointsClass))
+#define GST_RTSP_MOUNT_POINTS(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MOUNT_POINTS, GstRTSPMountPoints))
+#define GST_RTSP_MOUNT_POINTS_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MOUNT_POINTS, GstRTSPMountPointsClass))
+#define GST_RTSP_MOUNT_POINTS_CAST(obj)         ((GstRTSPMountPoints*)(obj))
+#define GST_RTSP_MOUNT_POINTS_CLASS_CAST(klass) ((GstRTSPMountPointsClass*)(klass))
+
+typedef struct _GstRTSPMountPoints GstRTSPMountPoints;
+typedef struct _GstRTSPMountPointsClass GstRTSPMountPointsClass;
+typedef struct _GstRTSPMountPointsPrivate GstRTSPMountPointsPrivate;
+
+/**
+ * GstRTSPMountPoints:
+ *
+ * Creates a #GstRTSPMediaFactory object for a given url.
+ */
+struct _GstRTSPMountPoints {
+  GObject       parent;
+
+  /*< private >*/
+  GstRTSPMountPointsPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPMountPointsClass:
+ * @make_path: make a path from the given url.
+ *
+ * The class for the media mounts object.
+ */
+struct _GstRTSPMountPointsClass {
+  GObjectClass  parent_class;
+
+  gchar * (*make_path) (GstRTSPMountPoints *mounts,
+                        const GstRTSPUrl *url);
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+GType                 gst_rtsp_mount_points_get_type       (void);
+
+/* creating a mount points */
+GstRTSPMountPoints *  gst_rtsp_mount_points_new            (void);
+
+gchar *               gst_rtsp_mount_points_make_path      (GstRTSPMountPoints *mounts,
+                                                            const GstRTSPUrl * url);
+/* finding a media factory */
+GstRTSPMediaFactory * gst_rtsp_mount_points_match          (GstRTSPMountPoints *mounts,
+                                                            const gchar *path,
+                                                            gint * matched);
+/* managing media to a mount point */
+void                  gst_rtsp_mount_points_add_factory    (GstRTSPMountPoints *mounts,
+                                                            const gchar *path,
+                                                            GstRTSPMediaFactory *factory);
+void                  gst_rtsp_mount_points_remove_factory (GstRTSPMountPoints *mounts,
+                                                            const gchar *path);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_MOUNT_POINTS_H__ */
diff --git a/gst/rtsp-server/rtsp-params.c b/gst/rtsp-server/rtsp-params.c
new file mode 100644
index 0000000..6025722
--- /dev/null
+++ b/gst/rtsp-server/rtsp-params.c
@@ -0,0 +1,77 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-params
+ * @short_description: Param get and set implementation
+ * @see_also: #GstRTSPClient
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include 
+
+#include "rtsp-params.h"
+
+/**
+ * gst_rtsp_params_set:
+ * @client: a #GstRTSPClient
+ * @ctx: (transfer none): a #GstRTSPContext
+ *
+ * Set parameters (not implemented yet)
+ *
+ * Returns: a #GstRTSPResult
+ */
+GstRTSPResult
+gst_rtsp_params_set (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPStatusCode code;
+
+  /* FIXME, actually parse the request based on the mime type and try to repond
+   * with a list of the parameters */
+  code = GST_RTSP_STS_PARAMETER_NOT_UNDERSTOOD;
+
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
+
+  return GST_RTSP_OK;
+}
+
+/**
+ * gst_rtsp_params_get:
+ * @client: a #GstRTSPClient
+ * @ctx: (transfer none): a #GstRTSPContext
+ *
+ * Get parameters (not implemented yet)
+ *
+ * Returns: a #GstRTSPResult
+ */
+GstRTSPResult
+gst_rtsp_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
+{
+  GstRTSPStatusCode code;
+
+  /* FIXME, actually parse the request based on the mime type and try to repond
+   * with a list of the parameters */
+  code = GST_RTSP_STS_PARAMETER_NOT_UNDERSTOOD;
+
+  gst_rtsp_message_init_response (ctx->response, code,
+      gst_rtsp_status_as_text (code), ctx->request);
+
+  return GST_RTSP_OK;
+}
diff --git a/gst/rtsp-server/rtsp-params.h b/gst/rtsp-server/rtsp-params.h
new file mode 100644
index 0000000..563757d
--- /dev/null
+++ b/gst/rtsp-server/rtsp-params.h
@@ -0,0 +1,38 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+#include 
+
+#ifndef __GST_RTSP_PARAMS_H__
+#define __GST_RTSP_PARAMS_H__
+
+#include "rtsp-client.h"
+#include "rtsp-session.h"
+
+G_BEGIN_DECLS
+
+GstRTSPResult    gst_rtsp_params_set      (GstRTSPClient * client, GstRTSPContext * ctx);
+GstRTSPResult    gst_rtsp_params_get      (GstRTSPClient * client, GstRTSPContext * ctx);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_PARAMS_H__ */
diff --git a/gst/rtsp-server/rtsp-permissions.c b/gst/rtsp-server/rtsp-permissions.c
new file mode 100644
index 0000000..66e9e44
--- /dev/null
+++ b/gst/rtsp-server/rtsp-permissions.c
@@ -0,0 +1,276 @@
+/* GStreamer
+ * Copyright (C) 2013 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-permissions
+ * @short_description: Roles and associated permissions
+ * @see_also: #GstRTSPToken, #GstRTSPAuth
+ *
+ * The #GstRTSPPermissions object contains an array of roles and associated
+ * permissions. The roles are represented with a string and the permissions with
+ * a generic #GstStructure.
+ *
+ * The permissions are deliberately kept generic. The possible values of the
+ * roles and #GstStructure keys and values are only determined by the #GstRTSPAuth
+ * object that performs the checks on the permissions and the current
+ * #GstRTSPToken.
+ *
+ * As a convenience function, gst_rtsp_permissions_is_allowed() can be used to
+ * check if the permissions contains a role that contains the boolean value
+ * %TRUE for the the given key.
+ *
+ * Last reviewed on 2013-07-15 (1.0.0)
+ */
+
+#include 
+
+#include "rtsp-permissions.h"
+
+typedef struct _GstRTSPPermissionsImpl
+{
+  GstRTSPPermissions permissions;
+
+  /* Roles, array of GstStructure */
+  GPtrArray *roles;
+} GstRTSPPermissionsImpl;
+
+static void
+free_structure (GstStructure * structure)
+{
+  gst_structure_set_parent_refcount (structure, NULL);
+  gst_structure_free (structure);
+}
+
+//GST_DEBUG_CATEGORY_STATIC (rtsp_permissions_debug);
+//#define GST_CAT_DEFAULT rtsp_permissions_debug
+
+GST_DEFINE_MINI_OBJECT_TYPE (GstRTSPPermissions, gst_rtsp_permissions);
+
+static void gst_rtsp_permissions_init (GstRTSPPermissionsImpl * permissions);
+
+static void
+_gst_rtsp_permissions_free (GstRTSPPermissions * permissions)
+{
+  GstRTSPPermissionsImpl *impl = (GstRTSPPermissionsImpl *) permissions;
+
+  g_ptr_array_free (impl->roles, TRUE);
+
+  g_slice_free1 (sizeof (GstRTSPPermissionsImpl), permissions);
+}
+
+static GstRTSPPermissions *
+_gst_rtsp_permissions_copy (GstRTSPPermissionsImpl * permissions)
+{
+  GstRTSPPermissionsImpl *copy;
+  guint i;
+
+  copy = (GstRTSPPermissionsImpl *) gst_rtsp_permissions_new ();
+
+  for (i = 0; i < permissions->roles->len; i++) {
+    GstStructure *entry = g_ptr_array_index (permissions->roles, i);
+    GstStructure *entry_copy = gst_structure_copy (entry);
+
+    gst_structure_set_parent_refcount (entry_copy,
+        ©->permissions.mini_object.refcount);
+    g_ptr_array_add (copy->roles, entry_copy);
+  }
+
+  return GST_RTSP_PERMISSIONS (copy);
+}
+
+static void
+gst_rtsp_permissions_init (GstRTSPPermissionsImpl * permissions)
+{
+  gst_mini_object_init (GST_MINI_OBJECT_CAST (permissions), 0,
+      GST_TYPE_RTSP_PERMISSIONS,
+      (GstMiniObjectCopyFunction) _gst_rtsp_permissions_copy, NULL,
+      (GstMiniObjectFreeFunction) _gst_rtsp_permissions_free);
+
+  permissions->roles =
+      g_ptr_array_new_with_free_func ((GDestroyNotify) free_structure);
+}
+
+/**
+ * gst_rtsp_permissions_new:
+ *
+ * Create a new empty Authorization permissions.
+ *
+ * Returns: (transfer full): a new empty authorization permissions.
+ */
+GstRTSPPermissions *
+gst_rtsp_permissions_new (void)
+{
+  GstRTSPPermissionsImpl *permissions;
+
+  permissions = g_slice_new0 (GstRTSPPermissionsImpl);
+  gst_rtsp_permissions_init (permissions);
+
+  return GST_RTSP_PERMISSIONS (permissions);
+}
+
+/**
+ * gst_rtsp_permissions_add_role:
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ * @fieldname: the first field name
+ * @...: additional arguments
+ *
+ * Add a new @role to @permissions with the given variables. The fields
+ * are the same layout as gst_structure_new().
+ */
+void
+gst_rtsp_permissions_add_role (GstRTSPPermissions * permissions,
+    const gchar * role, const gchar * fieldname, ...)
+{
+  va_list var_args;
+
+  va_start (var_args, fieldname);
+  gst_rtsp_permissions_add_role_valist (permissions, role, fieldname, var_args);
+  va_end (var_args);
+}
+
+/**
+ * gst_rtsp_permissions_add_role_valist:
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ * @fieldname: the first field name
+ * @var_args: additional fields to add
+ *
+ * Add a new @role to @permissions with the given variables. Structure fields
+ * are set according to the varargs in a manner similar to gst_structure_new().
+ */
+void
+gst_rtsp_permissions_add_role_valist (GstRTSPPermissions * permissions,
+    const gchar * role, const gchar * fieldname, va_list var_args)
+{
+  GstRTSPPermissionsImpl *impl = (GstRTSPPermissionsImpl *) permissions;
+  GstStructure *structure;
+  guint i, len;
+
+  g_return_if_fail (GST_IS_RTSP_PERMISSIONS (permissions));
+  g_return_if_fail (gst_mini_object_is_writable (&permissions->mini_object));
+  g_return_if_fail (role != NULL);
+  g_return_if_fail (fieldname != NULL);
+
+  structure = gst_structure_new_valist (role, fieldname, var_args);
+  g_return_if_fail (structure != NULL);
+
+  len = impl->roles->len;
+  for (i = 0; i < len; i++) {
+    GstStructure *entry = g_ptr_array_index (impl->roles, i);
+
+    if (gst_structure_has_name (entry, role)) {
+      g_ptr_array_remove_index_fast (impl->roles, i);
+      break;
+    }
+  }
+
+  gst_structure_set_parent_refcount (structure,
+      &impl->permissions.mini_object.refcount);
+  g_ptr_array_add (impl->roles, structure);
+}
+
+/**
+ * gst_rtsp_permissions_remove_role:
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ *
+ * Remove all permissions for @role in @permissions.
+ */
+void
+gst_rtsp_permissions_remove_role (GstRTSPPermissions * permissions,
+    const gchar * role)
+{
+  GstRTSPPermissionsImpl *impl = (GstRTSPPermissionsImpl *) permissions;
+  guint i, len;
+
+  g_return_if_fail (GST_IS_RTSP_PERMISSIONS (permissions));
+  g_return_if_fail (gst_mini_object_is_writable (&permissions->mini_object));
+  g_return_if_fail (role != NULL);
+
+  len = impl->roles->len;
+  for (i = 0; i < len; i++) {
+    GstStructure *entry = g_ptr_array_index (impl->roles, i);
+
+    if (gst_structure_has_name (entry, role)) {
+      g_ptr_array_remove_index_fast (impl->roles, i);
+      break;
+    }
+  }
+}
+
+/**
+ * gst_rtsp_permissions_get_role:
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ *
+ * Get all permissions for @role in @permissions.
+ *
+ * Returns: (transfer none): the structure with permissions for @role. It
+ * remains valid for as long as @permissions is valid.
+ */
+const GstStructure *
+gst_rtsp_permissions_get_role (GstRTSPPermissions * permissions,
+    const gchar * role)
+{
+  GstRTSPPermissionsImpl *impl = (GstRTSPPermissionsImpl *) permissions;
+  guint i, len;
+
+  g_return_val_if_fail (GST_IS_RTSP_PERMISSIONS (permissions), NULL);
+  g_return_val_if_fail (role != NULL, NULL);
+
+  len = impl->roles->len;
+  for (i = 0; i < len; i++) {
+    GstStructure *entry = g_ptr_array_index (impl->roles, i);
+
+    if (gst_structure_has_name (entry, role))
+      return entry;
+  }
+  return NULL;
+}
+
+/**
+ * gst_rtsp_permissions_is_allowed:
+ * @permissions: a #GstRTSPPermissions
+ * @role: a role
+ * @permission: a permission
+ *
+ * Check if @role in @permissions is given permission for @permission.
+ *
+ * Returns: %TRUE if @role is allowed @permission.
+ */
+gboolean
+gst_rtsp_permissions_is_allowed (GstRTSPPermissions * permissions,
+    const gchar * role, const gchar * permission)
+{
+  const GstStructure *str;
+  gboolean result;
+
+  g_return_val_if_fail (GST_IS_RTSP_PERMISSIONS (permissions), FALSE);
+  g_return_val_if_fail (role != NULL, FALSE);
+  g_return_val_if_fail (permission != NULL, FALSE);
+
+  str = gst_rtsp_permissions_get_role (permissions, role);
+  if (str == NULL)
+    return FALSE;
+
+  if (!gst_structure_get_boolean (str, permission, &result))
+    result = FALSE;
+
+  return result;
+}
diff --git a/gst/rtsp-server/rtsp-permissions.h b/gst/rtsp-server/rtsp-permissions.h
new file mode 100644
index 0000000..9e739c7
--- /dev/null
+++ b/gst/rtsp-server/rtsp-permissions.h
@@ -0,0 +1,101 @@
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#ifndef __GST_RTSP_PERMISSIONS_H__
+#define __GST_RTSP_PERMISSIONS_H__
+
+typedef struct _GstRTSPPermissions GstRTSPPermissions;
+
+G_BEGIN_DECLS
+
+GType gst_rtsp_permissions_get_type (void);
+
+#define GST_TYPE_RTSP_PERMISSIONS        (gst_rtsp_permissions_get_type ())
+#define GST_IS_RTSP_PERMISSIONS(obj)     (GST_IS_MINI_OBJECT_TYPE (obj, GST_TYPE_RTSP_PERMISSIONS))
+#define GST_RTSP_PERMISSIONS_CAST(obj)   ((GstRTSPPermissions*)(obj))
+#define GST_RTSP_PERMISSIONS(obj)        (GST_RTSP_PERMISSIONS_CAST(obj))
+
+/**
+ * GstRTSPPermissions:
+ *
+ * The opaque permissions structure. It is used to define the permissions
+ * of objects in different roles.
+ */
+struct _GstRTSPPermissions {
+  GstMiniObject mini_object;
+};
+
+/* refcounting */
+/**
+ * gst_rtsp_permissions_ref:
+ * @permissions: The permissions to refcount
+ *
+ * Increase the refcount of this permissions.
+ *
+ * Returns: (transfer full): @permissions (for convenience when doing assignments)
+ */
+#ifdef _FOOL_GTK_DOC_
+G_INLINE_FUNC GstRTSPPermissions * gst_rtsp_permissions_ref (GstRTSPPermissions * permissions);
+#endif
+
+static inline GstRTSPPermissions *
+gst_rtsp_permissions_ref (GstRTSPPermissions * permissions)
+{
+  return (GstRTSPPermissions *) gst_mini_object_ref (GST_MINI_OBJECT_CAST (permissions));
+}
+
+/**
+ * gst_rtsp_permissions_unref:
+ * @permissions: (transfer full): the permissions to refcount
+ *
+ * Decrease the refcount of an permissions, freeing it if the refcount reaches 0.
+ */
+#ifdef _FOOL_GTK_DOC_
+G_INLINE_FUNC void gst_rtsp_permissions_unref (GstRTSPPermissions * permissions);
+#endif
+
+static inline void
+gst_rtsp_permissions_unref (GstRTSPPermissions * permissions)
+{
+  gst_mini_object_unref (GST_MINI_OBJECT_CAST (permissions));
+}
+
+
+GstRTSPPermissions *  gst_rtsp_permissions_new             (void);
+
+void                  gst_rtsp_permissions_add_role        (GstRTSPPermissions *permissions,
+                                                            const gchar *role,
+                                                            const gchar *fieldname, ...);
+void                  gst_rtsp_permissions_add_role_valist (GstRTSPPermissions *permissions,
+                                                            const gchar *role,
+                                                            const gchar *fieldname,
+                                                            va_list var_args);
+void                  gst_rtsp_permissions_remove_role     (GstRTSPPermissions *permissions,
+                                                            const gchar *role);
+const GstStructure *  gst_rtsp_permissions_get_role        (GstRTSPPermissions *permissions,
+                                                            const gchar *role);
+
+gboolean              gst_rtsp_permissions_is_allowed      (GstRTSPPermissions *permissions,
+                                                            const gchar *role, const gchar *permission);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_PERMISSIONS_H__ */
diff --git a/gst/rtsp-server/rtsp-sdp.c b/gst/rtsp-server/rtsp-sdp.c
new file mode 100644
index 0000000..07aea90
--- /dev/null
+++ b/gst/rtsp-server/rtsp-sdp.c
@@ -0,0 +1,482 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-sdp
+ * @short_description: Make SDP messages
+ * @see_also: #GstRTSPMedia
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include 
+
+#include 
+
+#include "rtsp-sdp.h"
+
+#define AES_128_KEY_LEN 16
+#define AES_256_KEY_LEN 32
+
+#define HMAC_32_KEY_LEN 4
+#define HMAC_80_KEY_LEN 10
+
+static gboolean
+get_info_from_tags (GstPad * pad, GstEvent ** event, gpointer user_data)
+{
+  GstSDPMedia *media = (GstSDPMedia *) user_data;
+
+  if (GST_EVENT_TYPE (*event) == GST_EVENT_TAG) {
+    GstTagList *tags;
+    guint bitrate = 0;
+
+    gst_event_parse_tag (*event, &tags);
+
+    if (gst_tag_list_get_scope (tags) != GST_TAG_SCOPE_STREAM)
+      return TRUE;
+
+    if (!gst_tag_list_get_uint (tags, GST_TAG_MAXIMUM_BITRATE,
+            &bitrate) || bitrate == 0)
+      if (!gst_tag_list_get_uint (tags, GST_TAG_BITRATE, &bitrate) ||
+          bitrate == 0)
+        return TRUE;
+
+    /* set bandwidth (kbits/s) */
+    gst_sdp_media_add_bandwidth (media, GST_SDP_BWTYPE_AS, bitrate / 1000);
+
+    return FALSE;
+
+  }
+
+  return TRUE;
+}
+
+static void
+update_sdp_from_tags (GstRTSPStream * stream, GstSDPMedia * stream_media)
+{
+  GstPad *src_pad;
+
+  src_pad = gst_rtsp_stream_get_srcpad (stream);
+
+  gst_pad_sticky_events_foreach (src_pad, get_info_from_tags, stream_media);
+
+  gst_object_unref (src_pad);
+}
+
+static guint8
+enc_key_length_from_cipher_name (const gchar * cipher)
+{
+  if (g_strcmp0 (cipher, "aes-128-icm") == 0)
+    return AES_128_KEY_LEN;
+  else if (g_strcmp0 (cipher, "aes-256-icm") == 0)
+    return AES_256_KEY_LEN;
+  else {
+    GST_ERROR ("encryption algorithm '%s' not supported", cipher);
+    return 0;
+  }
+}
+
+static guint8
+auth_key_length_from_auth_name (const gchar * auth)
+{
+  if (g_strcmp0 (auth, "hmac-sha1-32") == 0)
+    return HMAC_32_KEY_LEN;
+  else if (g_strcmp0 (auth, "hmac-sha1-80") == 0)
+    return HMAC_80_KEY_LEN;
+  else {
+    GST_ERROR ("authentication algorithm '%s' not supported", auth);
+    return 0;
+  }
+}
+
+static void
+make_media (GstSDPMessage * sdp, GstSDPInfo * info, GstRTSPMedia * media,
+    GstRTSPStream * stream, GstStructure * s, GstRTSPProfile profile)
+{
+  GstSDPMedia *smedia;
+  const gchar *caps_str, *caps_enc, *caps_params;
+  gchar *tmp;
+  gint caps_pt, caps_rate;
+  guint n_fields, j;
+  gboolean first;
+  GString *fmtp;
+  GstRTSPLowerTrans ltrans;
+  GSocketFamily family;
+  const gchar *addrtype, *proto;
+  gchar *address;
+  guint ttl;
+
+  gst_sdp_media_new (&smedia);
+
+  /* get media type and payload for the m= line */
+  caps_str = gst_structure_get_string (s, "media");
+  gst_sdp_media_set_media (smedia, caps_str);
+
+  gst_structure_get_int (s, "payload", &caps_pt);
+  tmp = g_strdup_printf ("%d", caps_pt);
+  gst_sdp_media_add_format (smedia, tmp);
+  g_free (tmp);
+
+  gst_sdp_media_set_port_info (smedia, 0, 1);
+
+  switch (profile) {
+    case GST_RTSP_PROFILE_AVP:
+      proto = "RTP/AVP";
+      break;
+    case GST_RTSP_PROFILE_AVPF:
+      proto = "RTP/AVPF";
+      break;
+    case GST_RTSP_PROFILE_SAVP:
+      proto = "RTP/SAVP";
+      break;
+    case GST_RTSP_PROFILE_SAVPF:
+      proto = "RTP/SAVPF";
+      break;
+    default:
+      proto = "udp";
+      break;
+  }
+  gst_sdp_media_set_proto (smedia, proto);
+
+  if (info->is_ipv6) {
+    addrtype = "IP6";
+    family = G_SOCKET_FAMILY_IPV6;
+  } else {
+    addrtype = "IP4";
+    family = G_SOCKET_FAMILY_IPV4;
+  }
+
+  ltrans = gst_rtsp_stream_get_protocols (stream);
+  if (ltrans == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
+    GstRTSPAddress *addr;
+
+    addr = gst_rtsp_stream_get_multicast_address (stream, family);
+    if (addr == NULL)
+      goto no_multicast;
+
+    address = g_strdup (addr->address);
+    ttl = addr->ttl;
+    gst_rtsp_address_free (addr);
+  } else {
+    ttl = 16;
+    if (info->is_ipv6)
+      address = g_strdup ("::");
+    else
+      address = g_strdup ("0.0.0.0");
+  }
+
+  /* for the c= line */
+  gst_sdp_media_add_connection (smedia, "IN", addrtype, address, ttl, 1);
+  g_free (address);
+
+  /* get clock-rate, media type and params for the rtpmap attribute */
+  gst_structure_get_int (s, "clock-rate", &caps_rate);
+  caps_enc = gst_structure_get_string (s, "encoding-name");
+  caps_params = gst_structure_get_string (s, "encoding-params");
+
+  if (caps_enc) {
+    if (caps_params)
+      tmp = g_strdup_printf ("%d %s/%d/%s", caps_pt, caps_enc, caps_rate,
+          caps_params);
+    else
+      tmp = g_strdup_printf ("%d %s/%d", caps_pt, caps_enc, caps_rate);
+
+    gst_sdp_media_add_attribute (smedia, "rtpmap", tmp);
+    g_free (tmp);
+  }
+
+  /* the config uri */
+  tmp = gst_rtsp_stream_get_control (stream);
+  gst_sdp_media_add_attribute (smedia, "control", tmp);
+  g_free (tmp);
+
+
+  /* check for srtp */
+  do {
+    GstBuffer *srtpkey;
+    const GValue *val;
+    const gchar *srtpcipher, *srtpauth, *srtcpcipher, *srtcpauth;
+    GstMIKEYMessage *msg;
+    GstMIKEYPayload *payload, *pkd;
+    GBytes *bytes;
+    GstMapInfo info;
+    const guint8 *data;
+    gsize size;
+    gchar *base64;
+    guint8 byte;
+    guint32 ssrc;
+
+    val = gst_structure_get_value (s, "srtp-key");
+    if (val == NULL)
+      break;
+
+    srtpkey = gst_value_get_buffer (val);
+    if (srtpkey == NULL)
+      break;
+
+    srtpcipher = gst_structure_get_string (s, "srtp-cipher");
+    srtpauth = gst_structure_get_string (s, "srtp-auth");
+    srtcpcipher = gst_structure_get_string (s, "srtcp-cipher");
+    srtcpauth = gst_structure_get_string (s, "srtcp-auth");
+
+    if (srtpcipher == NULL || srtpauth == NULL || srtcpcipher == NULL ||
+        srtcpauth == NULL)
+      break;
+
+    msg = gst_mikey_message_new ();
+    /* unencrypted MIKEY message, we send this over TLS so this is allowed */
+    gst_mikey_message_set_info (msg, GST_MIKEY_VERSION, GST_MIKEY_TYPE_PSK_INIT,
+        FALSE, GST_MIKEY_PRF_MIKEY_1, 0, GST_MIKEY_MAP_TYPE_SRTP);
+    /* add policy '0' for our SSRC */
+    gst_rtsp_stream_get_ssrc (stream, &ssrc);
+    gst_mikey_message_add_cs_srtp (msg, 0, ssrc, 0);
+    /* timestamp is now */
+    gst_mikey_message_add_t_now_ntp_utc (msg);
+    /* add some random data */
+    gst_mikey_message_add_rand_len (msg, 16);
+
+    /* the policy '0' is SRTP with the above discovered algorithms */
+    payload = gst_mikey_payload_new (GST_MIKEY_PT_SP);
+    gst_mikey_payload_sp_set (payload, 0, GST_MIKEY_SEC_PROTO_SRTP);
+
+    /* only AES-CM is supported */
+    byte = 1;
+    gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_ALG, 1,
+        &byte);
+    /* Encryption key length */
+    byte = enc_key_length_from_cipher_name (srtpcipher);
+    gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_KEY_LEN, 1,
+        &byte);
+    /* only HMAC-SHA1 */
+    gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_ALG, 1,
+        &byte);
+    /* Authentication key length */
+    byte = auth_key_length_from_auth_name (srtpauth);
+    gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_KEY_LEN, 1,
+        &byte);
+    /* we enable encryption on RTP and RTCP */
+    gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_ENC, 1,
+        &byte);
+    gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTCP_ENC, 1,
+        &byte);
+    /* we enable authentication on RTP and RTCP */
+    gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_AUTH, 1,
+        &byte);
+    gst_mikey_message_add_payload (msg, payload);
+
+    /* make unencrypted KEMAC */
+    payload = gst_mikey_payload_new (GST_MIKEY_PT_KEMAC);
+    gst_mikey_payload_kemac_set (payload, GST_MIKEY_ENC_NULL,
+        GST_MIKEY_MAC_NULL);
+
+    /* add the key in key data */
+    pkd = gst_mikey_payload_new (GST_MIKEY_PT_KEY_DATA);
+    gst_buffer_map (srtpkey, &info, GST_MAP_READ);
+    gst_mikey_payload_key_data_set_key (pkd, GST_MIKEY_KD_TEK, info.size,
+        info.data);
+    gst_buffer_unmap (srtpkey, &info);
+    /* add key data to KEMAC */
+    gst_mikey_payload_kemac_add_sub (payload, pkd);
+    gst_mikey_message_add_payload (msg, payload);
+
+    /* now serialize this to bytes */
+    bytes = gst_mikey_message_to_bytes (msg, NULL, NULL);
+    gst_mikey_message_unref (msg);
+    /* and make it into base64 */
+    data = g_bytes_get_data (bytes, &size);
+    base64 = g_base64_encode (data, size);
+    g_bytes_unref (bytes);
+
+    tmp = g_strdup_printf ("mikey %s", base64);
+    g_free (base64);
+
+    gst_sdp_media_add_attribute (smedia, "key-mgmt", tmp);
+    g_free (tmp);
+  } while (FALSE);
+
+  /* collect all other properties and add them to fmtp or attributes */
+  fmtp = g_string_new ("");
+  g_string_append_printf (fmtp, "%d ", caps_pt);
+  first = TRUE;
+  n_fields = gst_structure_n_fields (s);
+  for (j = 0; j < n_fields; j++) {
+    const gchar *fname, *fval;
+
+    fname = gst_structure_nth_field_name (s, j);
+
+    /* filter out standard properties */
+    if (!strcmp (fname, "media"))
+      continue;
+    if (!strcmp (fname, "payload"))
+      continue;
+    if (!strcmp (fname, "clock-rate"))
+      continue;
+    if (!strcmp (fname, "encoding-name"))
+      continue;
+    if (!strcmp (fname, "encoding-params"))
+      continue;
+    if (!strcmp (fname, "ssrc"))
+      continue;
+    if (!strcmp (fname, "clock-base"))
+      continue;
+    if (!strcmp (fname, "seqnum-base"))
+      continue;
+    if (g_str_has_prefix (fname, "srtp-"))
+      continue;
+    if (g_str_has_prefix (fname, "srtcp-"))
+      continue;
+
+    if (g_str_has_prefix (fname, "a-")) {
+      /* attribute */
+      if ((fval = gst_structure_get_string (s, fname)))
+        gst_sdp_media_add_attribute (smedia, fname + 2, fval);
+      continue;
+    }
+    if (g_str_has_prefix (fname, "x-")) {
+      /* attribute */
+      if ((fval = gst_structure_get_string (s, fname)))
+        gst_sdp_media_add_attribute (smedia, fname, fval);
+      continue;
+    }
+
+    if ((fval = gst_structure_get_string (s, fname))) {
+      g_string_append_printf (fmtp, "%s%s=%s", first ? "" : ";", fname, fval);
+      first = FALSE;
+    }
+  }
+  if (!first) {
+    tmp = g_string_free (fmtp, FALSE);
+    gst_sdp_media_add_attribute (smedia, "fmtp", tmp);
+    g_free (tmp);
+  } else {
+    g_string_free (fmtp, TRUE);
+  }
+
+  update_sdp_from_tags (stream, smedia);
+
+  gst_sdp_message_add_media (sdp, smedia);
+  gst_sdp_media_free (smedia);
+
+  return;
+
+  /* ERRORS */
+no_multicast:
+  {
+    gst_sdp_media_free (smedia);
+    g_warning ("ignoring stream %d without multicast address",
+        gst_rtsp_stream_get_index (stream));
+    return;
+  }
+}
+
+/**
+ * gst_rtsp_sdp_from_media:
+ * @sdp: a #GstSDPMessage
+ * @info: (transfer none): a #GstSDPInfo
+ * @media: (transfer none): a #GstRTSPMedia
+ *
+ * Add @media specific info to @sdp. @info is used to configure the connection
+ * information in the SDP.
+ *
+ * Returns: TRUE on success.
+ */
+gboolean
+gst_rtsp_sdp_from_media (GstSDPMessage * sdp, GstSDPInfo * info,
+    GstRTSPMedia * media)
+{
+  guint i, n_streams;
+  gchar *rangestr;
+
+  n_streams = gst_rtsp_media_n_streams (media);
+
+  rangestr = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
+  if (rangestr == NULL)
+    goto not_prepared;
+
+  gst_sdp_message_add_attribute (sdp, "range", rangestr);
+  g_free (rangestr);
+
+  for (i = 0; i < n_streams; i++) {
+    GstRTSPStream *stream;
+    GstCaps *caps;
+    GstStructure *s;
+    GstRTSPProfile profiles;
+    guint mask;
+
+    stream = gst_rtsp_media_get_stream (media, i);
+    caps = gst_rtsp_stream_get_caps (stream);
+
+    if (caps == NULL) {
+      g_warning ("ignoring stream %d without media type", i);
+      continue;
+    }
+
+    s = gst_caps_get_structure (caps, 0);
+    if (s == NULL) {
+      gst_caps_unref (caps);
+      g_warning ("ignoring stream %d without media type", i);
+      continue;
+    }
+
+    /* make a new media for each profile */
+    profiles = gst_rtsp_stream_get_profiles (stream);
+    mask = 1;
+    while (profiles >= mask) {
+      GstRTSPProfile prof = profiles & mask;
+
+      if (prof)
+        make_media (sdp, info, media, stream, s, prof);
+
+      mask <<= 1;
+    }
+    gst_caps_unref (caps);
+  }
+
+  {
+    GstNetTimeProvider *provider;
+
+    if ((provider =
+            gst_rtsp_media_get_time_provider (media, info->server_ip, 0))) {
+      GstClock *clock;
+      gchar *address, *str;
+      gint port;
+
+      g_object_get (provider, "clock", &clock, "address", &address, "port",
+          &port, NULL);
+
+      str = g_strdup_printf ("GstNetTimeProvider %s %s:%d %" G_GUINT64_FORMAT,
+          g_type_name (G_TYPE_FROM_INSTANCE (clock)), address, port,
+          gst_clock_get_time (clock));
+
+      gst_sdp_message_add_attribute (sdp, "x-gst-clock", str);
+      g_free (str);
+      gst_object_unref (clock);
+      g_free (address);
+      gst_object_unref (provider);
+    }
+  }
+
+  return TRUE;
+
+  /* ERRORS */
+not_prepared:
+  {
+    GST_ERROR ("media %p is not prepared", media);
+    return FALSE;
+  }
+}
diff --git a/gst/rtsp-server/rtsp-sdp.h b/gst/rtsp-server/rtsp-sdp.h
new file mode 100644
index 0000000..7732f36
--- /dev/null
+++ b/gst/rtsp-server/rtsp-sdp.h
@@ -0,0 +1,40 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+#include 
+
+#include "rtsp-media.h"
+
+#ifndef __GST_RTSP_SDP_H__
+#define __GST_RTSP_SDP_H__
+
+G_BEGIN_DECLS
+
+typedef struct {
+  gboolean is_ipv6;
+  const gchar *server_ip;
+} GstSDPInfo;
+
+/* creating SDP */
+gboolean            gst_rtsp_sdp_from_media      (GstSDPMessage *sdp, GstSDPInfo *info, GstRTSPMedia * media);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SDP_H__ */
diff --git a/gst/rtsp-server/rtsp-server-wfd.c b/gst/rtsp-server/rtsp-server-wfd.c
new file mode 100644
index 0000000..15dac76
--- /dev/null
+++ b/gst/rtsp-server/rtsp-server-wfd.c
@@ -0,0 +1,332 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-server
+ * @short_description: The main server object
+ * @see_also: #GstRTSPClient, #GstRTSPThreadPool
+ *
+ * The server object is the object listening for connections on a port and
+ * creating #GstRTSPClient objects to handle those connections.
+ *
+ * The server will listen on the address set with gst_rtsp_server_set_address()
+ * and the port or service configured with gst_rtsp_server_set_service().
+ * Use gst_rtsp_server_set_backlog() to configure the amount of pending requests
+ * that the server will keep. By default the server listens on the current
+ * network (0.0.0.0) and port 8554.
+ *
+ * The server will require an SSL connection when a TLS certificate has been
+ * set in the auth object with gst_rtsp_auth_set_tls_certificate().
+ *
+ * To start the server, use gst_rtsp_server_attach() to attach it to a
+ * #GMainContext. For more control, gst_rtsp_server_create_source() and
+ * gst_rtsp_server_create_socket() can be used to get a #GSource and #GSocket
+ * respectively.
+ *
+ * gst_rtsp_server_transfer_connection() can be used to transfer an existing
+ * socket to the RTSP server, for example from an HTTP server.
+ *
+ * Once the server socket is attached to a mainloop, it will start accepting
+ * connections. When a new connection is received, a new #GstRTSPClient object
+ * is created to handle the connection. The new client will be configured with
+ * the server #GstRTSPAuth, #GstRTSPMountPoints, #GstRTSPSessionPool and
+ * #GstRTSPThreadPool.
+ *
+ * The server uses the configured #GstRTSPThreadPool object to handle the
+ * remainder of the communication with this client.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#include 
+#include 
+
+#include "rtsp-server-wfd.h"
+#include "rtsp-client-wfd.h"
+
+#define GST_RTSP_WFD_SERVER_GET_PRIVATE(obj)  \
+       (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_WFD_SERVER, GstRTSPWFDServerPrivate))
+
+#define GST_RTSP_WFD_SERVER_GET_LOCK(server)  (&(GST_RTSP_WFD_SERVER_CAST(server)->priv->lock))
+#define GST_RTSP_WFD_SERVER_LOCK(server)      (g_mutex_lock(GST_RTSP_WFD_SERVER_GET_LOCK(server)))
+#define GST_RTSP_WFD_SERVER_UNLOCK(server)    (g_mutex_unlock(GST_RTSP_WFD_SERVER_GET_LOCK(server)))
+
+struct _GstRTSPWFDServerPrivate
+{
+  GMutex lock;                  /* protects everything in this struct */
+
+  /* the clients that are connected */
+  GList *clients;
+  guint64 native_resolution;
+  guint64 supported_resolution;
+  guint8 audio_codec;
+};
+
+G_DEFINE_TYPE (GstRTSPWFDServer, gst_rtsp_wfd_server, GST_TYPE_RTSP_SERVER);
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_wfd_server_debug);
+#define GST_CAT_DEFAULT rtsp_wfd_server_debug
+
+static void gst_rtsp_wfd_server_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec);
+static void gst_rtsp_wfd_server_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_wfd_server_finalize (GObject * object);
+
+static GstRTSPClient *create_client_wfd (GstRTSPServer * server);
+static void client_connected_wfd (GstRTSPServer * server,
+    GstRTSPClient * client);
+
+static void
+gst_rtsp_wfd_server_class_init (GstRTSPWFDServerClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstRTSPServerClass *rtsp_server_class;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPWFDServerPrivate));
+
+  gobject_class = G_OBJECT_CLASS (klass);
+  rtsp_server_class = GST_RTSP_SERVER_CLASS (klass);
+
+  gobject_class->get_property = gst_rtsp_wfd_server_get_property;
+  gobject_class->set_property = gst_rtsp_wfd_server_set_property;
+  gobject_class->finalize = gst_rtsp_wfd_server_finalize;
+
+  rtsp_server_class->create_client = create_client_wfd;
+  rtsp_server_class->client_connected = client_connected_wfd;
+
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_wfd_server_debug, "rtspwfdserver", 0,
+      "GstRTSPWFDServer");
+}
+
+static void
+gst_rtsp_wfd_server_init (GstRTSPWFDServer * server)
+{
+  GstRTSPWFDServerPrivate *priv = GST_RTSP_WFD_SERVER_GET_PRIVATE (server);
+
+  g_return_if_fail (priv != NULL);
+
+  server->priv = priv;
+  server->priv->native_resolution = 0;
+  server->priv->supported_resolution = 1;
+  server->priv->audio_codec = 2;
+  GST_INFO_OBJECT (server, "New server is initialized");
+}
+
+static void
+gst_rtsp_wfd_server_finalize (GObject * object)
+{
+  GstRTSPWFDServer *server = GST_RTSP_WFD_SERVER (object);
+  //GstRTSPWFDServerPrivate *priv = server->priv;
+
+  GST_DEBUG_OBJECT (server, "finalize server");
+
+  G_OBJECT_CLASS (gst_rtsp_wfd_server_parent_class)->finalize (object);
+}
+
+/**
+ * gst_rtsp_server_new:
+ *
+ * Create a new #GstRTSPWFDServer instance.
+ */
+GstRTSPWFDServer *
+gst_rtsp_wfd_server_new (void)
+{
+  GstRTSPWFDServer *result;
+
+  result = g_object_new (GST_TYPE_RTSP_WFD_SERVER, NULL);
+
+  return result;
+}
+
+static void
+gst_rtsp_wfd_server_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec)
+{
+  //GstRTSPWFDServer *server = GST_RTSP_WFD_SERVER (object);
+
+  switch (propid) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+static void
+gst_rtsp_wfd_server_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec)
+{
+  //GstRTSPWFDServer *server = GST_RTSP_WFD_SERVER (object);
+
+  switch (propid) {
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+static gboolean
+_start_wfd (gpointer data)
+{
+  GstRTSPWFDClient *client = (GstRTSPWFDClient *) data;
+
+  GST_INFO_OBJECT (client, "WFD client is STARTing");
+
+  gst_rtsp_wfd_client_start_wfd (client);
+  return FALSE;
+}
+
+static void
+client_connected_wfd (GstRTSPServer * server, GstRTSPClient * client)
+{
+  GST_INFO_OBJECT (server, "Client is connected");
+
+  gst_rtsp_wfd_client_set_host_address (GST_RTSP_WFD_CLIENT_CAST (client),
+      gst_rtsp_server_get_address (server));
+  g_idle_add (_start_wfd, client);
+  return;
+}
+
+static GstRTSPClient *
+create_client_wfd (GstRTSPServer * server)
+{
+  GstRTSPWFDClient *client;
+  GstRTSPThreadPool *thread_pool = NULL;
+  GstRTSPSessionPool *session_pool = NULL;
+  GstRTSPMountPoints *mount_points = NULL;
+  GstRTSPAuth *auth = NULL;
+  GstRTSPWFDServerPrivate *priv = GST_RTSP_WFD_SERVER_GET_PRIVATE(server);
+
+  g_return_val_if_fail (priv != NULL, NULL);
+
+  GST_INFO_OBJECT (server, "New Client is being created");
+
+  /* a new client connected, create a session to handle the client. */
+  client = gst_rtsp_wfd_client_new ();
+
+  thread_pool = gst_rtsp_server_get_thread_pool (server);
+  session_pool = gst_rtsp_server_get_session_pool (server);
+  mount_points = gst_rtsp_server_get_mount_points (server);
+  auth = gst_rtsp_server_get_auth (server);
+
+  /* set the session pool that this client should use */
+  GST_RTSP_WFD_SERVER_LOCK (server);
+  gst_rtsp_client_set_session_pool (GST_RTSP_CLIENT_CAST (client),
+      session_pool);
+  /* set the mount points that this client should use */
+  gst_rtsp_client_set_mount_points (GST_RTSP_CLIENT_CAST (client),
+      mount_points);
+  /* set authentication manager */
+  gst_rtsp_client_set_auth (GST_RTSP_CLIENT_CAST (client), auth);
+  /* set threadpool */
+  gst_rtsp_client_set_thread_pool (GST_RTSP_CLIENT_CAST (client), thread_pool);
+
+  gst_rtsp_wfd_client_set_video_supported_resolution (client,
+        priv->supported_resolution);
+
+  gst_rtsp_wfd_client_set_video_native_resolution (client,
+        priv->native_resolution);
+
+  gst_rtsp_wfd_client_set_audio_codec (client,
+        priv->audio_codec);
+
+  GST_RTSP_WFD_SERVER_UNLOCK (server);
+
+  return GST_RTSP_CLIENT (client);
+}
+
+GstRTSPResult
+gst_rtsp_wfd_server_trigger_request (GstRTSPServer * server,
+    GstWFDTriggerType type)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GList *clients, *walk, *next;
+
+  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), GST_RTSP_ERROR);
+
+  clients = gst_rtsp_server_client_filter (server, NULL, NULL);
+  if (clients == NULL) {
+    GST_ERROR_OBJECT (server, "There is no client in this server");
+  }
+
+  for (walk = clients; walk; walk = next) {
+    GstRTSPClient *client = walk->data;
+
+    next = g_list_next (walk);
+
+    res =
+        gst_rtsp_wfd_client_trigger_request (GST_RTSP_WFD_CLIENT (client),
+        type);
+    if (res != GST_RTSP_OK) {
+      GST_ERROR_OBJECT (server, "Failed to send trigger request %d", type);
+    }
+    g_object_unref (client);
+  }
+
+  return res;
+
+}
+
+GstRTSPResult
+gst_rtsp_wfd_server_set_supported_reso(GstRTSPWFDServer *server, guint64 supported_reso)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPWFDServerPrivate *priv = GST_RTSP_WFD_SERVER_GET_PRIVATE(server);
+
+  g_return_val_if_fail (GST_IS_RTSP_WFD_SERVER (server), GST_RTSP_ERROR);
+  g_return_val_if_fail (priv != NULL, GST_RTSP_ERROR);
+
+  GST_RTSP_WFD_SERVER_LOCK (server);
+
+  priv->supported_resolution = supported_reso;
+
+  GST_RTSP_WFD_SERVER_UNLOCK (server);
+  return res;
+}
+
+GstRTSPResult
+gst_rtsp_wfd_server_set_video_native_reso (GstRTSPWFDServer *server, guint64 native_reso)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPWFDServerPrivate *priv = GST_RTSP_WFD_SERVER_GET_PRIVATE(server);
+
+  g_return_val_if_fail (GST_IS_RTSP_WFD_SERVER (server), GST_RTSP_ERROR);
+  g_return_val_if_fail (priv != NULL, GST_RTSP_ERROR);
+
+  GST_RTSP_WFD_SERVER_LOCK (server);
+
+  priv->native_resolution = native_reso;
+
+  GST_RTSP_WFD_SERVER_UNLOCK (server);
+  return res;
+}
+
+GstRTSPResult
+gst_rtsp_wfd_server_set_audio_codec (GstRTSPWFDServer *server, guint8 audio_codec)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPWFDServerPrivate *priv = GST_RTSP_WFD_SERVER_GET_PRIVATE(server);
+
+  g_return_val_if_fail (GST_IS_RTSP_WFD_SERVER (server), GST_RTSP_ERROR);
+  g_return_val_if_fail (priv != NULL, GST_RTSP_ERROR);
+
+  GST_RTSP_WFD_SERVER_LOCK (server);
+
+  priv->audio_codec = audio_codec;
+
+  GST_RTSP_WFD_SERVER_UNLOCK (server);
+  return res;
+}
diff --git a/gst/rtsp-server/rtsp-server-wfd.h b/gst/rtsp-server/rtsp-server-wfd.h
new file mode 100644
index 0000000..13a5f3a
--- /dev/null
+++ b/gst/rtsp-server/rtsp-server-wfd.h
@@ -0,0 +1,155 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTSP_SERVER_WFD_H__
+#define __GST_RTSP_SERVER_WFD_H__
+
+#include 
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPWFDServer GstRTSPWFDServer;
+typedef struct _GstRTSPWFDServerClass GstRTSPWFDServerClass;
+typedef struct _GstRTSPWFDServerPrivate GstRTSPWFDServerPrivate;
+
+#include "rtsp-session-pool.h"
+#include "rtsp-mount-points.h"
+#include "rtsp-server.h"
+#include "rtsp-client-wfd.h"
+#include "rtsp-auth.h"
+
+#define GST_TYPE_RTSP_WFD_SERVER              (gst_rtsp_wfd_server_get_type ())
+#define GST_IS_RTSP_WFD_SERVER(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_WFD_SERVER))
+#define GST_IS_RTSP_WFD_SERVER_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_WFD_SERVER))
+#define GST_RTSP_WFD_SERVER_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_WFD_SERVER, GstRTSPWFDServerClass))
+#define GST_RTSP_WFD_SERVER(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_WFD_SERVER, GstRTSPWFDServer))
+#define GST_RTSP_WFD_SERVER_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_WFD_SERVER, GstRTSPWFDServerClass))
+#define GST_RTSP_WFD_SERVER_CAST(obj)         ((GstRTSPWFDServer*)(obj))
+#define GST_RTSP_WFD_SERVER_CLASS_CAST(klass) ((GstRTSPWFDServerClass*)(klass))
+
+/**
+ * GstRTSPWFDServer:
+ *
+ * This object listens on a port, creates and manages the clients connected to
+ * it.
+ */
+struct _GstRTSPWFDServer {
+  GstRTSPServer parent;
+
+  /*< private >*/
+  GstRTSPWFDServerPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPServerClass:
+ * @create_client: Create, configure a new GstRTSPClient
+ *          object that handles the new connection on @socket. The default
+ *          implementation will create a GstRTSPClient and will configure the
+ *          mount-points, auth, session-pool and thread-pool on the client.
+ * @client_connected: emited when a new client connected.
+ *
+ * The RTSP server class structure
+ */
+struct _GstRTSPWFDServerClass {
+  GstRTSPServerClass  parent_class;
+
+  /*< private >*/
+  gpointer         _gst_reserved[GST_PADDING_LARGE];
+};
+
+GType                 gst_rtsp_wfd_server_get_type             (void);
+GstRTSPWFDServer *    gst_rtsp_wfd_server_new                  (void);
+GstRTSPResult         gst_rtsp_wfd_server_trigger_request      (GstRTSPServer *server, GstWFDTriggerType type);
+
+GstRTSPResult         gst_rtsp_wfd_server_set_supported_reso (GstRTSPWFDServer *server, guint64 supported_reso);
+GstRTSPResult         gst_rtsp_wfd_server_set_video_native_reso    (GstRTSPWFDServer *server, guint64 native_reso);
+GstRTSPResult         gst_rtsp_wfd_server_set_audio_codec    (GstRTSPWFDServer *server, guint8 audio_codec);
+
+#if 0
+void                  gst_rtsp_server_set_address          (GstRTSPServer *server, const gchar *address);
+gchar *               gst_rtsp_server_get_address          (GstRTSPServer *server);
+
+void                  gst_rtsp_server_set_service          (GstRTSPServer *server, const gchar *service);
+gchar *               gst_rtsp_server_get_service          (GstRTSPServer *server);
+
+void                  gst_rtsp_server_set_backlog          (GstRTSPServer *server, gint backlog);
+gint                  gst_rtsp_server_get_backlog          (GstRTSPServer *server);
+
+int                   gst_rtsp_server_get_bound_port       (GstRTSPServer *server);
+
+void                  gst_rtsp_server_set_session_pool     (GstRTSPServer *server, GstRTSPSessionPool *pool);
+GstRTSPSessionPool *  gst_rtsp_server_get_session_pool     (GstRTSPServer *server);
+
+void                  gst_rtsp_server_set_mount_points     (GstRTSPServer *server, GstRTSPMountPoints *mounts);
+GstRTSPMountPoints *  gst_rtsp_server_get_mount_points     (GstRTSPServer *server);
+
+void                  gst_rtsp_server_set_auth             (GstRTSPServer *server, GstRTSPAuth *auth);
+GstRTSPAuth *         gst_rtsp_server_get_auth             (GstRTSPServer *server);
+
+void                  gst_rtsp_server_set_thread_pool      (GstRTSPServer *server, GstRTSPThreadPool *pool);
+GstRTSPThreadPool *   gst_rtsp_server_get_thread_pool      (GstRTSPServer *server);
+
+gboolean              gst_rtsp_server_transfer_connection  (GstRTSPServer * server, GSocket *socket,
+                                                            const gchar * ip, gint port,
+                                                            const gchar *initial_buffer);
+
+gboolean              gst_rtsp_server_io_func              (GSocket *socket, GIOCondition condition,
+                                                            GstRTSPServer *server);
+
+GSocket *             gst_rtsp_server_create_socket        (GstRTSPServer *server,
+                                                            GCancellable  *cancellable,
+                                                            GError **error);
+GSource *             gst_rtsp_server_create_source        (GstRTSPServer *server,
+                                                            GCancellable * cancellable,
+                                                            GError **error);
+guint                 gst_rtsp_server_attach               (GstRTSPServer *server,
+                                                            GMainContext *context);
+/**
+ * GstRTSPServerClientFilterFunc:
+ * @server: a #GstRTSPServer object
+ * @client: a #GstRTSPClient in @server
+ * @user_data: user data that has been given to gst_rtsp_server_client_filter()
+ *
+ * This function will be called by the gst_rtsp_server_client_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @client will be removed
+ * from @server.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @client untouched in
+ * @server.
+ *
+ * A value of #GST_RTSP_FILTER_REF will add @client to the result #GList of
+ * gst_rtsp_server_client_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPServerClientFilterFunc)  (GstRTSPServer *server,
+                                                               GstRTSPClient *client,
+                                                               gpointer user_data);
+
+GList *                gst_rtsp_server_client_filter    (GstRTSPServer *server,
+                                                         GstRTSPServerClientFilterFunc func,
+                                                         gpointer user_data);
+#endif
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SERVER_WFD_H__ */
diff --git a/gst/rtsp-server/rtsp-server.c b/gst/rtsp-server/rtsp-server.c
new file mode 100644
index 0000000..686ba03
--- /dev/null
+++ b/gst/rtsp-server/rtsp-server.c
@@ -0,0 +1,1430 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-server
+ * @short_description: The main server object
+ * @see_also: #GstRTSPClient, #GstRTSPThreadPool
+ *
+ * The server object is the object listening for connections on a port and
+ * creating #GstRTSPClient objects to handle those connections.
+ *
+ * The server will listen on the address set with gst_rtsp_server_set_address()
+ * and the port or service configured with gst_rtsp_server_set_service().
+ * Use gst_rtsp_server_set_backlog() to configure the amount of pending requests
+ * that the server will keep. By default the server listens on the current
+ * network (0.0.0.0) and port 8554.
+ *
+ * The server will require an SSL connection when a TLS certificate has been
+ * set in the auth object with gst_rtsp_auth_set_tls_certificate().
+ *
+ * To start the server, use gst_rtsp_server_attach() to attach it to a
+ * #GMainContext. For more control, gst_rtsp_server_create_source() and
+ * gst_rtsp_server_create_socket() can be used to get a #GSource and #GSocket
+ * respectively.
+ *
+ * gst_rtsp_server_transfer_connection() can be used to transfer an existing
+ * socket to the RTSP server, for example from an HTTP server.
+ *
+ * Once the server socket is attached to a mainloop, it will start accepting
+ * connections. When a new connection is received, a new #GstRTSPClient object
+ * is created to handle the connection. The new client will be configured with
+ * the server #GstRTSPAuth, #GstRTSPMountPoints, #GstRTSPSessionPool and
+ * #GstRTSPThreadPool.
+ *
+ * The server uses the configured #GstRTSPThreadPool object to handle the
+ * remainder of the communication with this client.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+#include 
+#include 
+
+#include "rtsp-server.h"
+#include "rtsp-client.h"
+
+#define GST_RTSP_SERVER_GET_PRIVATE(obj)  \
+       (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_SERVER, GstRTSPServerPrivate))
+
+#define GST_RTSP_SERVER_GET_LOCK(server)  (&(GST_RTSP_SERVER_CAST(server)->priv->lock))
+#define GST_RTSP_SERVER_LOCK(server)      (g_mutex_lock(GST_RTSP_SERVER_GET_LOCK(server)))
+#define GST_RTSP_SERVER_UNLOCK(server)    (g_mutex_unlock(GST_RTSP_SERVER_GET_LOCK(server)))
+
+struct _GstRTSPServerPrivate
+{
+  GMutex lock;                  /* protects everything in this struct */
+
+  /* server information */
+  gchar *address;
+  gchar *service;
+  gint backlog;
+
+  GSocket *socket;
+
+  /* sessions on this server */
+  GstRTSPSessionPool *session_pool;
+
+  /* mount points for this server */
+  GstRTSPMountPoints *mount_points;
+
+  /* authentication manager */
+  GstRTSPAuth *auth;
+
+  /* resource manager */
+  GstRTSPThreadPool *thread_pool;
+
+  /* the clients that are connected */
+  GList *clients;
+  guint clients_cookie;
+};
+
+#define DEFAULT_ADDRESS         "0.0.0.0"
+#define DEFAULT_BOUND_PORT      -1
+/* #define DEFAULT_ADDRESS         "::0" */
+#define DEFAULT_SERVICE         "8554"
+#define DEFAULT_BACKLOG         5
+
+/* Define to use the SO_LINGER option so that the server sockets can be resused
+ * sooner. Disabled for now because it is not very well implemented by various
+ * OSes and it causes clients to fail to read the TEARDOWN response. */
+#undef USE_SOLINGER
+
+enum
+{
+  PROP_0,
+  PROP_ADDRESS,
+  PROP_SERVICE,
+  PROP_BOUND_PORT,
+  PROP_BACKLOG,
+
+  PROP_SESSION_POOL,
+  PROP_MOUNT_POINTS,
+  PROP_LAST
+};
+
+enum
+{
+  SIGNAL_CLIENT_CONNECTED,
+  SIGNAL_LAST
+};
+
+G_DEFINE_TYPE (GstRTSPServer, gst_rtsp_server, G_TYPE_OBJECT);
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_server_debug);
+#define GST_CAT_DEFAULT rtsp_server_debug
+
+typedef struct _ClientContext ClientContext;
+
+static guint gst_rtsp_server_signals[SIGNAL_LAST] = { 0 };
+
+static void gst_rtsp_server_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec);
+static void gst_rtsp_server_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_server_finalize (GObject * object);
+
+static GstRTSPClient *default_create_client (GstRTSPServer * server);
+
+static void
+gst_rtsp_server_class_init (GstRTSPServerClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPServerPrivate));
+
+  gobject_class = G_OBJECT_CLASS (klass);
+
+  gobject_class->get_property = gst_rtsp_server_get_property;
+  gobject_class->set_property = gst_rtsp_server_set_property;
+  gobject_class->finalize = gst_rtsp_server_finalize;
+
+  /**
+   * GstRTSPServer::address:
+   *
+   * The address of the server. This is the address where the server will
+   * listen on.
+   */
+  g_object_class_install_property (gobject_class, PROP_ADDRESS,
+      g_param_spec_string ("address", "Address",
+          "The address the server uses to listen on", DEFAULT_ADDRESS,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRTSPServer::service:
+   *
+   * The service of the server. This is either a string with the service name or
+   * a port number (as a string) the server will listen on.
+   */
+  g_object_class_install_property (gobject_class, PROP_SERVICE,
+      g_param_spec_string ("service", "Service",
+          "The service or port number the server uses to listen on",
+          DEFAULT_SERVICE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRTSPServer::bound-port:
+   *
+   * The actual port the server is listening on. Can be used to retrieve the
+   * port number when the server is started on port 0, which means bind to a
+   * random port. Set to -1 if the server has not been bound yet.
+   */
+  g_object_class_install_property (gobject_class, PROP_BOUND_PORT,
+      g_param_spec_int ("bound-port", "Bound port",
+          "The port number the server is listening on",
+          -1, G_MAXUINT16, DEFAULT_BOUND_PORT,
+          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRTSPServer::backlog:
+   *
+   * The backlog argument defines the maximum length to which the queue of
+   * pending connections for the server may grow. If a connection request arrives
+   * when the queue is full, the client may receive an error with an indication of
+   * ECONNREFUSED or, if the underlying protocol supports retransmission, the
+   * request may be ignored so that a later reattempt at connection succeeds.
+   */
+  g_object_class_install_property (gobject_class, PROP_BACKLOG,
+      g_param_spec_int ("backlog", "Backlog",
+          "The maximum length to which the queue "
+          "of pending connections may grow", 0, G_MAXINT, DEFAULT_BACKLOG,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRTSPServer::session-pool:
+   *
+   * The session pool of the server. By default each server has a separate
+   * session pool but sessions can be shared between servers by setting the same
+   * session pool on multiple servers.
+   */
+  g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
+      g_param_spec_object ("session-pool", "Session Pool",
+          "The session pool to use for client session",
+          GST_TYPE_RTSP_SESSION_POOL,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRTSPServer::mount-points:
+   *
+   * The mount points to use for this server. By default the server has no
+   * mount points and thus cannot map urls to media streams.
+   */
+  g_object_class_install_property (gobject_class, PROP_MOUNT_POINTS,
+      g_param_spec_object ("mount-points", "Mount Points",
+          "The mount points to use for client session",
+          GST_TYPE_RTSP_MOUNT_POINTS,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  gst_rtsp_server_signals[SIGNAL_CLIENT_CONNECTED] =
+      g_signal_new ("client-connected", G_TYPE_FROM_CLASS (gobject_class),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPServerClass, client_connected),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+      GST_TYPE_RTSP_CLIENT);
+
+  klass->create_client = default_create_client;
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_server_debug, "rtspserver", 0, "GstRTSPServer");
+}
+
+static void
+gst_rtsp_server_init (GstRTSPServer * server)
+{
+  GstRTSPServerPrivate *priv = GST_RTSP_SERVER_GET_PRIVATE (server);
+
+  server->priv = priv;
+
+  g_mutex_init (&priv->lock);
+  priv->address = g_strdup (DEFAULT_ADDRESS);
+  priv->service = g_strdup (DEFAULT_SERVICE);
+  priv->socket = NULL;
+  priv->backlog = DEFAULT_BACKLOG;
+  priv->session_pool = gst_rtsp_session_pool_new ();
+  priv->mount_points = gst_rtsp_mount_points_new ();
+  priv->thread_pool = gst_rtsp_thread_pool_new ();
+}
+
+static void
+gst_rtsp_server_finalize (GObject * object)
+{
+  GstRTSPServer *server = GST_RTSP_SERVER (object);
+  GstRTSPServerPrivate *priv = server->priv;
+
+  GST_DEBUG_OBJECT (server, "finalize server");
+
+  g_free (priv->address);
+  g_free (priv->service);
+
+  if (priv->socket)
+    g_object_unref (priv->socket);
+
+  if (priv->session_pool)
+    g_object_unref (priv->session_pool);
+  if (priv->mount_points)
+    g_object_unref (priv->mount_points);
+  if (priv->thread_pool)
+    g_object_unref (priv->thread_pool);
+
+  if (priv->auth)
+    g_object_unref (priv->auth);
+
+  g_mutex_clear (&priv->lock);
+
+  G_OBJECT_CLASS (gst_rtsp_server_parent_class)->finalize (object);
+}
+
+/**
+ * gst_rtsp_server_new:
+ *
+ * Create a new #GstRTSPServer instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPServer
+ */
+GstRTSPServer *
+gst_rtsp_server_new (void)
+{
+  GstRTSPServer *result;
+
+  result = g_object_new (GST_TYPE_RTSP_SERVER, NULL);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_server_set_address:
+ * @server: a #GstRTSPServer
+ * @address: the address
+ *
+ * Configure @server to accept connections on the given address.
+ *
+ * This function must be called before the server is bound.
+ */
+void
+gst_rtsp_server_set_address (GstRTSPServer * server, const gchar * address)
+{
+  GstRTSPServerPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_SERVER (server));
+  g_return_if_fail (address != NULL);
+
+  priv = server->priv;
+
+  GST_RTSP_SERVER_LOCK (server);
+  g_free (priv->address);
+  priv->address = g_strdup (address);
+  GST_RTSP_SERVER_UNLOCK (server);
+}
+
+/**
+ * gst_rtsp_server_get_address:
+ * @server: a #GstRTSPServer
+ *
+ * Get the address on which the server will accept connections.
+ *
+ * Returns: (transfer full): the server address. g_free() after usage.
+ */
+gchar *
+gst_rtsp_server_get_address (GstRTSPServer * server)
+{
+  GstRTSPServerPrivate *priv;
+  gchar *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+  priv = server->priv;
+
+  GST_RTSP_SERVER_LOCK (server);
+  result = g_strdup (priv->address);
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_server_get_bound_port:
+ * @server: a #GstRTSPServer
+ *
+ * Get the port number where the server was bound to.
+ *
+ * Returns: the port number
+ */
+int
+gst_rtsp_server_get_bound_port (GstRTSPServer * server)
+{
+  GstRTSPServerPrivate *priv;
+  GSocketAddress *address;
+  int result = -1;
+
+  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), result);
+
+  priv = server->priv;
+
+  GST_RTSP_SERVER_LOCK (server);
+  if (priv->socket == NULL)
+    goto out;
+
+  address = g_socket_get_local_address (priv->socket, NULL);
+  result = g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (address));
+  g_object_unref (address);
+
+out:
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_server_set_service:
+ * @server: a #GstRTSPServer
+ * @service: the service
+ *
+ * Configure @server to accept connections on the given service.
+ * @service should be a string containing the service name (see services(5)) or
+ * a string containing a port number between 1 and 65535.
+ *
+ * When @service is set to "0", the server will listen on a random free
+ * port. The actual used port can be retrieved with
+ * gst_rtsp_server_get_bound_port().
+ *
+ * This function must be called before the server is bound.
+ */
+void
+gst_rtsp_server_set_service (GstRTSPServer * server, const gchar * service)
+{
+  GstRTSPServerPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_SERVER (server));
+  g_return_if_fail (service != NULL);
+
+  priv = server->priv;
+
+  GST_RTSP_SERVER_LOCK (server);
+  g_free (priv->service);
+  priv->service = g_strdup (service);
+  GST_RTSP_SERVER_UNLOCK (server);
+}
+
+/**
+ * gst_rtsp_server_get_service:
+ * @server: a #GstRTSPServer
+ *
+ * Get the service on which the server will accept connections.
+ *
+ * Returns: (transfer full): the service. use g_free() after usage.
+ */
+gchar *
+gst_rtsp_server_get_service (GstRTSPServer * server)
+{
+  GstRTSPServerPrivate *priv;
+  gchar *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+  priv = server->priv;
+
+  GST_RTSP_SERVER_LOCK (server);
+  result = g_strdup (priv->service);
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_server_set_backlog:
+ * @server: a #GstRTSPServer
+ * @backlog: the backlog
+ *
+ * configure the maximum amount of requests that may be queued for the
+ * server.
+ *
+ * This function must be called before the server is bound.
+ */
+void
+gst_rtsp_server_set_backlog (GstRTSPServer * server, gint backlog)
+{
+  GstRTSPServerPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+  priv = server->priv;
+
+  GST_RTSP_SERVER_LOCK (server);
+  priv->backlog = backlog;
+  GST_RTSP_SERVER_UNLOCK (server);
+}
+
+/**
+ * gst_rtsp_server_get_backlog:
+ * @server: a #GstRTSPServer
+ *
+ * The maximum amount of queued requests for the server.
+ *
+ * Returns: the server backlog.
+ */
+gint
+gst_rtsp_server_get_backlog (GstRTSPServer * server)
+{
+  GstRTSPServerPrivate *priv;
+  gint result;
+
+  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), -1);
+
+  priv = server->priv;
+
+  GST_RTSP_SERVER_LOCK (server);
+  result = priv->backlog;
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_server_set_session_pool:
+ * @server: a #GstRTSPServer
+ * @pool: (transfer none): a #GstRTSPSessionPool
+ *
+ * configure @pool to be used as the session pool of @server.
+ */
+void
+gst_rtsp_server_set_session_pool (GstRTSPServer * server,
+    GstRTSPSessionPool * pool)
+{
+  GstRTSPServerPrivate *priv;
+  GstRTSPSessionPool *old;
+
+  g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+  priv = server->priv;
+
+  if (pool)
+    g_object_ref (pool);
+
+  GST_RTSP_SERVER_LOCK (server);
+  old = priv->session_pool;
+  priv->session_pool = pool;
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  if (old)
+    g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_server_get_session_pool:
+ * @server: a #GstRTSPServer
+ *
+ * Get the #GstRTSPSessionPool used as the session pool of @server.
+ *
+ * Returns: (transfer full): the #GstRTSPSessionPool used for sessions. g_object_unref() after
+ * usage.
+ */
+GstRTSPSessionPool *
+gst_rtsp_server_get_session_pool (GstRTSPServer * server)
+{
+  GstRTSPServerPrivate *priv;
+  GstRTSPSessionPool *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+  priv = server->priv;
+
+  GST_RTSP_SERVER_LOCK (server);
+  if ((result = priv->session_pool))
+    g_object_ref (result);
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_server_set_mount_points:
+ * @server: a #GstRTSPServer
+ * @mounts: (transfer none): a #GstRTSPMountPoints
+ *
+ * configure @mounts to be used as the mount points of @server.
+ */
+void
+gst_rtsp_server_set_mount_points (GstRTSPServer * server,
+    GstRTSPMountPoints * mounts)
+{
+  GstRTSPServerPrivate *priv;
+  GstRTSPMountPoints *old;
+
+  g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+  priv = server->priv;
+
+  if (mounts)
+    g_object_ref (mounts);
+
+  GST_RTSP_SERVER_LOCK (server);
+  old = priv->mount_points;
+  priv->mount_points = mounts;
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  if (old)
+    g_object_unref (old);
+}
+
+
+/**
+ * gst_rtsp_server_get_mount_points:
+ * @server: a #GstRTSPServer
+ *
+ * Get the #GstRTSPMountPoints used as the mount points of @server.
+ *
+ * Returns: (transfer full): the #GstRTSPMountPoints of @server. g_object_unref() after
+ * usage.
+ */
+GstRTSPMountPoints *
+gst_rtsp_server_get_mount_points (GstRTSPServer * server)
+{
+  GstRTSPServerPrivate *priv;
+  GstRTSPMountPoints *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+  priv = server->priv;
+
+  GST_RTSP_SERVER_LOCK (server);
+  if ((result = priv->mount_points))
+    g_object_ref (result);
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_server_set_auth:
+ * @server: a #GstRTSPServer
+ * @auth: (transfer none): a #GstRTSPAuth
+ *
+ * configure @auth to be used as the authentication manager of @server.
+ */
+void
+gst_rtsp_server_set_auth (GstRTSPServer * server, GstRTSPAuth * auth)
+{
+  GstRTSPServerPrivate *priv;
+  GstRTSPAuth *old;
+
+  g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+  priv = server->priv;
+
+  if (auth)
+    g_object_ref (auth);
+
+  GST_RTSP_SERVER_LOCK (server);
+  old = priv->auth;
+  priv->auth = auth;
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  if (old)
+    g_object_unref (old);
+}
+
+
+/**
+ * gst_rtsp_server_get_auth:
+ * @server: a #GstRTSPServer
+ *
+ * Get the #GstRTSPAuth used as the authentication manager of @server.
+ *
+ * Returns: (transfer full): the #GstRTSPAuth of @server. g_object_unref() after
+ * usage.
+ */
+GstRTSPAuth *
+gst_rtsp_server_get_auth (GstRTSPServer * server)
+{
+  GstRTSPServerPrivate *priv;
+  GstRTSPAuth *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+  priv = server->priv;
+
+  GST_RTSP_SERVER_LOCK (server);
+  if ((result = priv->auth))
+    g_object_ref (result);
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_server_set_thread_pool:
+ * @server: a #GstRTSPServer
+ * @pool: (transfer none): a #GstRTSPThreadPool
+ *
+ * configure @pool to be used as the thread pool of @server.
+ */
+void
+gst_rtsp_server_set_thread_pool (GstRTSPServer * server,
+    GstRTSPThreadPool * pool)
+{
+  GstRTSPServerPrivate *priv;
+  GstRTSPThreadPool *old;
+
+  g_return_if_fail (GST_IS_RTSP_SERVER (server));
+
+  priv = server->priv;
+
+  if (pool)
+    g_object_ref (pool);
+
+  GST_RTSP_SERVER_LOCK (server);
+  old = priv->thread_pool;
+  priv->thread_pool = pool;
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  if (old)
+    g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_server_get_thread_pool:
+ * @server: a #GstRTSPServer
+ *
+ * Get the #GstRTSPThreadPool used as the thread pool of @server.
+ *
+ * Returns: (transfer full): the #GstRTSPThreadPool of @server. g_object_unref() after
+ * usage.
+ */
+GstRTSPThreadPool *
+gst_rtsp_server_get_thread_pool (GstRTSPServer * server)
+{
+  GstRTSPServerPrivate *priv;
+  GstRTSPThreadPool *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+  priv = server->priv;
+
+  GST_RTSP_SERVER_LOCK (server);
+  if ((result = priv->thread_pool))
+    g_object_ref (result);
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  return result;
+}
+
+static void
+gst_rtsp_server_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRTSPServer *server = GST_RTSP_SERVER (object);
+
+  switch (propid) {
+    case PROP_ADDRESS:
+      g_value_take_string (value, gst_rtsp_server_get_address (server));
+      break;
+    case PROP_SERVICE:
+      g_value_take_string (value, gst_rtsp_server_get_service (server));
+      break;
+    case PROP_BOUND_PORT:
+      g_value_set_int (value, gst_rtsp_server_get_bound_port (server));
+      break;
+    case PROP_BACKLOG:
+      g_value_set_int (value, gst_rtsp_server_get_backlog (server));
+      break;
+    case PROP_SESSION_POOL:
+      g_value_take_object (value, gst_rtsp_server_get_session_pool (server));
+      break;
+    case PROP_MOUNT_POINTS:
+      g_value_take_object (value, gst_rtsp_server_get_mount_points (server));
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+static void
+gst_rtsp_server_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRTSPServer *server = GST_RTSP_SERVER (object);
+
+  switch (propid) {
+    case PROP_ADDRESS:
+      gst_rtsp_server_set_address (server, g_value_get_string (value));
+      break;
+    case PROP_SERVICE:
+      gst_rtsp_server_set_service (server, g_value_get_string (value));
+      break;
+    case PROP_BACKLOG:
+      gst_rtsp_server_set_backlog (server, g_value_get_int (value));
+      break;
+    case PROP_SESSION_POOL:
+      gst_rtsp_server_set_session_pool (server, g_value_get_object (value));
+      break;
+    case PROP_MOUNT_POINTS:
+      gst_rtsp_server_set_mount_points (server, g_value_get_object (value));
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+/**
+ * gst_rtsp_server_create_socket:
+ * @server: a #GstRTSPServer
+ * @cancellable: (allow-none): a #GCancellable
+ * @error: (out): a #GError
+ *
+ * Create a #GSocket for @server. The socket will listen on the
+ * configured service.
+ *
+ * Returns: (transfer full): the #GSocket for @server or %NULL when an error
+ * occurred.
+ */
+GSocket *
+gst_rtsp_server_create_socket (GstRTSPServer * server,
+    GCancellable * cancellable, GError ** error)
+{
+  GstRTSPServerPrivate *priv;
+  GSocketConnectable *conn;
+  GSocketAddressEnumerator *enumerator;
+  GSocket *socket = NULL;
+#ifdef USE_SOLINGER
+  struct linger linger;
+#endif
+  GError *sock_error = NULL;
+  GError *bind_error = NULL;
+  guint16 port;
+
+  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+  priv = server->priv;
+
+  GST_RTSP_SERVER_LOCK (server);
+  GST_DEBUG_OBJECT (server, "getting address info of %s/%s", priv->address,
+      priv->service);
+
+  /* resolve the server IP address */
+  port = atoi (priv->service);
+  if (port != 0 || !strcmp (priv->service, "0"))
+    conn = g_network_address_new (priv->address, port);
+  else
+    conn = g_network_service_new (priv->service, "tcp", priv->address);
+
+  enumerator = g_socket_connectable_enumerate (conn);
+  g_object_unref (conn);
+
+  /* create server socket, we loop through all the addresses until we manage to
+   * create a socket and bind. */
+  while (TRUE) {
+    GSocketAddress *sockaddr;
+
+    sockaddr =
+        g_socket_address_enumerator_next (enumerator, cancellable, error);
+    if (!sockaddr) {
+      if (!*error)
+        GST_DEBUG_OBJECT (server, "no more addresses %s",
+            *error ? (*error)->message : "");
+      else
+        GST_DEBUG_OBJECT (server, "failed to retrieve next address %s",
+            (*error)->message);
+      break;
+    }
+
+    /* only keep the first error */
+    socket = g_socket_new (g_socket_address_get_family (sockaddr),
+        G_SOCKET_TYPE_STREAM, G_SOCKET_PROTOCOL_TCP,
+        sock_error ? NULL : &sock_error);
+
+    if (socket == NULL) {
+      GST_DEBUG_OBJECT (server, "failed to make socket (%s), try next",
+          sock_error->message);
+      g_object_unref (sockaddr);
+      continue;
+    }
+
+    if (g_socket_bind (socket, sockaddr, TRUE, bind_error ? NULL : &bind_error)) {
+      /* ask what port the socket has been bound to */
+      if (port == 0 || !strcmp (priv->service, "0")) {
+        GError *addr_error = NULL;
+
+        g_object_unref (sockaddr);
+        sockaddr = g_socket_get_local_address (socket, &addr_error);
+
+        if (addr_error != NULL) {
+          GST_DEBUG_OBJECT (server,
+              "failed to get the local address of a bound socket %s",
+              addr_error->message);
+          g_clear_error (&addr_error);
+          break;
+        }
+        port =
+            g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr));
+
+        if (port != 0) {
+          g_free (priv->service);
+          priv->service = g_strdup_printf ("%d", port);
+        } else {
+          GST_DEBUG_OBJECT (server, "failed to get the port of a bound socket");
+        }
+      }
+      g_object_unref (sockaddr);
+      break;
+    }
+
+    GST_DEBUG_OBJECT (server, "failed to bind socket (%s), try next",
+        bind_error->message);
+    g_object_unref (sockaddr);
+    g_object_unref (socket);
+    socket = NULL;
+  }
+  g_object_unref (enumerator);
+
+  if (socket == NULL)
+    goto no_socket;
+
+  g_clear_error (&sock_error);
+  g_clear_error (&bind_error);
+
+  GST_DEBUG_OBJECT (server, "opened sending server socket");
+
+  /* keep connection alive; avoids SIGPIPE during write */
+  g_socket_set_keepalive (socket, TRUE);
+
+#if 0
+#ifdef USE_SOLINGER
+  /* make sure socket is reset 5 seconds after close. This ensure that we can
+   * reuse the socket quickly while still having a chance to send data to the
+   * client. */
+  linger.l_onoff = 1;
+  linger.l_linger = 5;
+  if (setsockopt (sockfd, SOL_SOCKET, SO_LINGER,
+          (void *) &linger, sizeof (linger)) < 0)
+    goto linger_failed;
+#endif
+#endif
+
+  /* set the server socket to nonblocking */
+  g_socket_set_blocking (socket, FALSE);
+
+  /* set listen backlog */
+  g_socket_set_listen_backlog (socket, priv->backlog);
+
+  if (!g_socket_listen (socket, error))
+    goto listen_failed;
+
+  GST_DEBUG_OBJECT (server, "listening on server socket %p with queue of %d",
+      socket, priv->backlog);
+
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  return socket;
+
+  /* ERRORS */
+no_socket:
+  {
+    GST_ERROR_OBJECT (server, "failed to create socket");
+    goto close_error;
+  }
+#if 0
+#ifdef USE_SOLINGER
+linger_failed:
+  {
+    GST_ERROR_OBJECT (server, "failed to no linger socket: %s",
+        g_strerror (errno));
+    goto close_error;
+  }
+#endif
+#endif
+listen_failed:
+  {
+    GST_ERROR_OBJECT (server, "failed to listen on socket: %s",
+        (*error)->message);
+    goto close_error;
+  }
+close_error:
+  {
+    if (socket)
+      g_object_unref (socket);
+
+    if (sock_error) {
+      if (error == NULL)
+        g_propagate_error (error, sock_error);
+      else
+        g_error_free (sock_error);
+    }
+    if (bind_error) {
+      if ((error == NULL) || (*error == NULL))
+        g_propagate_error (error, bind_error);
+      else
+        g_error_free (bind_error);
+    }
+    GST_RTSP_SERVER_UNLOCK (server);
+    return NULL;
+  }
+}
+
+struct _ClientContext
+{
+  GstRTSPServer *server;
+  GstRTSPThread *thread;
+  GstRTSPClient *client;
+};
+
+static gboolean
+free_client_context (ClientContext * ctx)
+{
+  GST_DEBUG ("free context %p", ctx);
+
+  GST_RTSP_SERVER_LOCK (ctx->server);
+  if (ctx->thread)
+    gst_rtsp_thread_stop (ctx->thread);
+  GST_RTSP_SERVER_UNLOCK (ctx->server);
+
+  g_object_unref (ctx->client);
+  g_object_unref (ctx->server);
+  g_slice_free (ClientContext, ctx);
+
+  return G_SOURCE_REMOVE;
+}
+
+static void
+unmanage_client (GstRTSPClient * client, ClientContext * ctx)
+{
+  GstRTSPServer *server = ctx->server;
+  GstRTSPServerPrivate *priv = server->priv;
+
+  GST_DEBUG_OBJECT (server, "unmanage client %p", client);
+
+  GST_RTSP_SERVER_LOCK (server);
+  priv->clients = g_list_remove (priv->clients, ctx);
+  priv->clients_cookie++;
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  if (ctx->thread) {
+    GSource *src;
+
+    src = g_idle_source_new ();
+    g_source_set_callback (src, (GSourceFunc) free_client_context, ctx, NULL);
+    g_source_attach (src, ctx->thread->context);
+    g_source_unref (src);
+  } else {
+    free_client_context (ctx);
+  }
+}
+
+/* add the client context to the active list of clients, takes ownership
+ * of client */
+static void
+manage_client (GstRTSPServer * server, GstRTSPClient * client)
+{
+  ClientContext *cctx;
+  GstRTSPServerPrivate *priv = server->priv;
+  GMainContext *mainctx = NULL;
+  GstRTSPContext ctx = { NULL };
+
+  GST_DEBUG_OBJECT (server, "manage client %p", client);
+
+  g_signal_emit (server, gst_rtsp_server_signals[SIGNAL_CLIENT_CONNECTED], 0,
+      client);
+
+  cctx = g_slice_new0 (ClientContext);
+  cctx->server = g_object_ref (server);
+  cctx->client = client;
+
+  GST_RTSP_SERVER_LOCK (server);
+
+  ctx.server = server;
+  ctx.client = client;
+
+  cctx->thread = gst_rtsp_thread_pool_get_thread (priv->thread_pool,
+      GST_RTSP_THREAD_TYPE_CLIENT, &ctx);
+  if (cctx->thread)
+    mainctx = cctx->thread->context;
+  else {
+    GSource *source;
+    /* find the context to add the watch */
+    if ((source = g_main_current_source ()))
+      mainctx = g_source_get_context (source);
+  }
+
+  g_signal_connect (client, "closed", (GCallback) unmanage_client, cctx);
+  priv->clients = g_list_prepend (priv->clients, cctx);
+  priv->clients_cookie++;
+
+  gst_rtsp_client_attach (client, mainctx);
+
+  GST_RTSP_SERVER_UNLOCK (server);
+}
+
+static GstRTSPClient *
+default_create_client (GstRTSPServer * server)
+{
+  GstRTSPClient *client;
+  GstRTSPServerPrivate *priv = server->priv;
+
+  /* a new client connected, create a session to handle the client. */
+  client = gst_rtsp_client_new ();
+
+  /* set the session pool that this client should use */
+  GST_RTSP_SERVER_LOCK (server);
+  gst_rtsp_client_set_session_pool (client, priv->session_pool);
+  /* set the mount points that this client should use */
+  gst_rtsp_client_set_mount_points (client, priv->mount_points);
+  /* set authentication manager */
+  gst_rtsp_client_set_auth (client, priv->auth);
+  /* set threadpool */
+  gst_rtsp_client_set_thread_pool (client, priv->thread_pool);
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  return client;
+}
+
+/**
+ * gst_rtsp_server_transfer_connection:
+ * @server: a #GstRTSPServer
+ * @socket: (transfer full): a network socket
+ * @ip: the IP address of the remote client
+ * @port: the port used by the other end
+ * @initial_buffer: any initial data that was already read from the socket
+ *
+ * Take an existing network socket and use it for an RTSP connection. This
+ * is used when transferring a socket from an HTTP server which should be used
+ * as an RTSP over HTTP tunnel. The @initial_buffer contains any remaining data
+ * that the HTTP server read from the socket while parsing the HTTP header.
+ *
+ * Returns: TRUE if all was ok, FALSE if an error occurred.
+ */
+gboolean
+gst_rtsp_server_transfer_connection (GstRTSPServer * server, GSocket * socket,
+    const gchar * ip, gint port, const gchar * initial_buffer)
+{
+  GstRTSPClient *client = NULL;
+  GstRTSPServerClass *klass;
+  GstRTSPConnection *conn;
+  GstRTSPResult res;
+
+  klass = GST_RTSP_SERVER_GET_CLASS (server);
+
+  if (klass->create_client)
+    client = klass->create_client (server);
+  if (client == NULL)
+    goto client_failed;
+
+  GST_RTSP_CHECK (gst_rtsp_connection_create_from_socket (socket, ip, port,
+          initial_buffer, &conn), no_connection);
+  g_object_unref (socket);
+
+  /* set connection on the client now */
+  gst_rtsp_client_set_connection (client, conn);
+
+  /* manage the client connection */
+  manage_client (server, client);
+
+  return TRUE;
+
+  /* ERRORS */
+client_failed:
+  {
+    GST_ERROR_OBJECT (server, "failed to create a client");
+    g_object_unref (socket);
+    return FALSE;
+  }
+no_connection:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+    GST_ERROR ("could not create connection from socket %p: %s", socket, str);
+    g_free (str);
+    g_object_unref (socket);
+    return FALSE;
+  }
+}
+
+/**
+ * gst_rtsp_server_io_func:
+ * @socket: a #GSocket
+ * @condition: the condition on @source
+ * @server: (transfer none): a #GstRTSPServer
+ *
+ * A default #GSocketSourceFunc that creates a new #GstRTSPClient to accept and handle a
+ * new connection on @socket or @server.
+ *
+ * Returns: TRUE if the source could be connected, FALSE if an error occurred.
+ */
+gboolean
+gst_rtsp_server_io_func (GSocket * socket, GIOCondition condition,
+    GstRTSPServer * server)
+{
+  GstRTSPServerPrivate *priv = server->priv;
+  GstRTSPClient *client = NULL;
+  GstRTSPServerClass *klass;
+  GstRTSPResult res;
+  GstRTSPConnection *conn = NULL;
+  GstRTSPContext ctx = { NULL };
+
+  if (condition & G_IO_IN) {
+    /* a new client connected. */
+    GST_RTSP_CHECK (gst_rtsp_connection_accept (socket, &conn, NULL),
+        accept_failed);
+
+    ctx.server = server;
+    ctx.conn = conn;
+    ctx.auth = priv->auth;
+    gst_rtsp_context_push_current (&ctx);
+
+    if (!gst_rtsp_auth_check (GST_RTSP_AUTH_CHECK_CONNECT))
+      goto connection_refused;
+
+    klass = GST_RTSP_SERVER_GET_CLASS (server);
+    /* a new client connected, create a client object to handle the client. */
+    if (klass->create_client)
+      client = klass->create_client (server);
+    if (client == NULL)
+      goto client_failed;
+
+    /* set connection on the client now */
+    gst_rtsp_client_set_connection (client, conn);
+
+    /* manage the client connection */
+    manage_client (server, client);
+  } else {
+    GST_WARNING_OBJECT (server, "received unknown event %08x", condition);
+  }
+exit:
+  gst_rtsp_context_pop_current (&ctx);
+
+  return G_SOURCE_CONTINUE;
+
+  /* ERRORS */
+accept_failed:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+    GST_ERROR_OBJECT (server, "Could not accept client on socket %p: %s",
+        socket, str);
+    g_free (str);
+    goto exit;
+  }
+connection_refused:
+  {
+    GST_ERROR_OBJECT (server, "connection refused");
+    gst_rtsp_connection_free (conn);
+    goto exit;
+  }
+client_failed:
+  {
+    GST_ERROR_OBJECT (server, "failed to create a client");
+    gst_rtsp_connection_free (conn);
+    goto exit;
+  }
+}
+
+static void
+watch_destroyed (GstRTSPServer * server)
+{
+  GstRTSPServerPrivate *priv = server->priv;
+
+  GST_DEBUG_OBJECT (server, "source destroyed");
+
+  g_object_unref (priv->socket);
+  priv->socket = NULL;
+  g_object_unref (server);
+}
+
+/**
+ * gst_rtsp_server_create_source:
+ * @server: a #GstRTSPServer
+ * @cancellable: (allow-none): a #GCancellable or %NULL.
+ * @error: (out): a #GError
+ *
+ * Create a #GSource for @server. The new source will have a default
+ * #GSocketSourceFunc of gst_rtsp_server_io_func().
+ *
+ * @cancellable if not %NULL can be used to cancel the source, which will cause
+ * the source to trigger, reporting the current condition (which is likely 0
+ * unless cancellation happened at the same time as a condition change). You can
+ * check for this in the callback using g_cancellable_is_cancelled().
+ *
+ * Returns: (transfer full): the #GSource for @server or %NULL when an error
+ * occurred. Free with g_source_unref ()
+ */
+GSource *
+gst_rtsp_server_create_source (GstRTSPServer * server,
+    GCancellable * cancellable, GError ** error)
+{
+  GstRTSPServerPrivate *priv;
+  GSocket *socket, *old;
+  GSource *source;
+
+  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+  priv = server->priv;
+
+  socket = gst_rtsp_server_create_socket (server, NULL, error);
+  if (socket == NULL)
+    goto no_socket;
+
+  GST_RTSP_SERVER_LOCK (server);
+  old = priv->socket;
+  priv->socket = g_object_ref (socket);
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  if (old)
+    g_object_unref (old);
+
+  /* create a watch for reads (new connections) and possible errors */
+  source = g_socket_create_source (socket, G_IO_IN |
+      G_IO_ERR | G_IO_HUP | G_IO_NVAL, cancellable);
+  g_object_unref (socket);
+
+  /* configure the callback */
+  g_source_set_callback (source,
+      (GSourceFunc) gst_rtsp_server_io_func, g_object_ref (server),
+      (GDestroyNotify) watch_destroyed);
+
+  return source;
+
+no_socket:
+  {
+    GST_ERROR_OBJECT (server, "failed to create socket");
+    return NULL;
+  }
+}
+
+/**
+ * gst_rtsp_server_attach:
+ * @server: a #GstRTSPServer
+ * @context: (allow-none): a #GMainContext
+ *
+ * Attaches @server to @context. When the mainloop for @context is run, the
+ * server will be dispatched. When @context is %NULL, the default context will be
+ * used).
+ *
+ * This function should be called when the server properties and urls are fully
+ * configured and the server is ready to start.
+ *
+ * Returns: the ID (greater than 0) for the source within the GMainContext.
+ */
+guint
+gst_rtsp_server_attach (GstRTSPServer * server, GMainContext * context)
+{
+  guint res;
+  GSource *source;
+  GError *error = NULL;
+
+  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), 0);
+
+  source = gst_rtsp_server_create_source (server, NULL, &error);
+  if (source == NULL)
+    goto no_source;
+
+  res = g_source_attach (source, context);
+  g_source_unref (source);
+
+  return res;
+
+  /* ERRORS */
+no_source:
+  {
+    GST_ERROR_OBJECT (server, "failed to create watch: %s", error->message);
+    g_error_free (error);
+    return 0;
+  }
+}
+
+/**
+ * gst_rtsp_server_client_filter:
+ * @server: a #GstRTSPServer
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: user data passed to @func
+ *
+ * Call @func for each client managed by @server. The result value of @func
+ * determines what happens to the client. @func will be called with @server
+ * locked so no further actions on @server can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the client will be removed from
+ * @server.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the client will remain in @server.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the client will remain in @server but
+ * will also be added with an additional ref to the result #GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each client.
+ *
+ * Returns: (element-type GstRTSPClient) (transfer full): a #GList with all
+ * clients for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the #GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_server_client_filter (GstRTSPServer * server,
+    GstRTSPServerClientFilterFunc func, gpointer user_data)
+{
+  GstRTSPServerPrivate *priv;
+  GList *result, *walk, *next;
+  GHashTable *visited;
+  guint cookie;
+
+  g_return_val_if_fail (GST_IS_RTSP_SERVER (server), NULL);
+
+  priv = server->priv;
+
+  result = NULL;
+  if (func)
+    visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+  GST_RTSP_SERVER_LOCK (server);
+restart:
+  cookie = priv->clients_cookie;
+  for (walk = priv->clients; walk; walk = next) {
+    ClientContext *cctx = walk->data;
+    GstRTSPClient *client = cctx->client;
+    GstRTSPFilterResult res;
+    gboolean changed;
+
+    next = g_list_next (walk);
+
+    if (func) {
+      /* only visit each media once */
+      if (g_hash_table_contains (visited, client))
+        continue;
+
+      g_hash_table_add (visited, g_object_ref (client));
+      GST_RTSP_SERVER_UNLOCK (server);
+
+      res = func (server, client, user_data);
+
+      GST_RTSP_SERVER_LOCK (server);
+    } else
+      res = GST_RTSP_FILTER_REF;
+
+    changed = (cookie != priv->clients_cookie);
+
+    switch (res) {
+      case GST_RTSP_FILTER_REMOVE:
+        GST_RTSP_SERVER_UNLOCK (server);
+
+        gst_rtsp_client_close (client);
+
+        GST_RTSP_SERVER_LOCK (server);
+        changed |= (cookie != priv->clients_cookie);
+        break;
+      case GST_RTSP_FILTER_REF:
+        result = g_list_prepend (result, g_object_ref (client));
+        break;
+      case GST_RTSP_FILTER_KEEP:
+      default:
+        break;
+    }
+    if (changed)
+      goto restart;
+  }
+  GST_RTSP_SERVER_UNLOCK (server);
+
+  if (func)
+    g_hash_table_unref (visited);
+
+  return result;
+}
diff --git a/gst/rtsp-server/rtsp-server.h b/gst/rtsp-server/rtsp-server.h
new file mode 100644
index 0000000..9a6f9d4
--- /dev/null
+++ b/gst/rtsp-server/rtsp-server.h
@@ -0,0 +1,169 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_RTSP_SERVER_H__
+#define __GST_RTSP_SERVER_H__
+
+#include 
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPServer GstRTSPServer;
+typedef struct _GstRTSPServerClass GstRTSPServerClass;
+typedef struct _GstRTSPServerPrivate GstRTSPServerPrivate;
+
+#include "rtsp-session-pool.h"
+#include "rtsp-session.h"
+#include "rtsp-media.h"
+#include "rtsp-stream.h"
+#include "rtsp-stream-transport.h"
+#include "rtsp-address-pool.h"
+#include "rtsp-thread-pool.h"
+#include "rtsp-client.h"
+#include "rtsp-context.h"
+#include "rtsp-server.h"
+#include "rtsp-mount-points.h"
+#include "rtsp-media-factory.h"
+#include "rtsp-permissions.h"
+#include "rtsp-auth.h"
+#include "rtsp-token.h"
+#include "rtsp-session-media.h"
+#include "rtsp-sdp.h"
+#include "rtsp-media-factory-uri.h"
+#include "rtsp-params.h"
+
+#define GST_TYPE_RTSP_SERVER              (gst_rtsp_server_get_type ())
+#define GST_IS_RTSP_SERVER(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_SERVER))
+#define GST_IS_RTSP_SERVER_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_SERVER))
+#define GST_RTSP_SERVER_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_SERVER, GstRTSPServerClass))
+#define GST_RTSP_SERVER(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_SERVER, GstRTSPServer))
+#define GST_RTSP_SERVER_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_SERVER, GstRTSPServerClass))
+#define GST_RTSP_SERVER_CAST(obj)         ((GstRTSPServer*)(obj))
+#define GST_RTSP_SERVER_CLASS_CAST(klass) ((GstRTSPServerClass*)(klass))
+
+/**
+ * GstRTSPServer:
+ *
+ * This object listens on a port, creates and manages the clients connected to
+ * it.
+ */
+struct _GstRTSPServer {
+  GObject      parent;
+
+  /*< private >*/
+  GstRTSPServerPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPServerClass:
+ * @create_client: Create, configure a new GstRTSPClient
+ *          object that handles the new connection on @socket. The default
+ *          implementation will create a GstRTSPClient and will configure the
+ *          mount-points, auth, session-pool and thread-pool on the client.
+ * @client_connected: emited when a new client connected.
+ *
+ * The RTSP server class structure
+ */
+struct _GstRTSPServerClass {
+  GObjectClass  parent_class;
+
+  GstRTSPClient * (*create_client)      (GstRTSPServer *server);
+
+  /* signals */
+  void            (*client_connected)   (GstRTSPServer *server, GstRTSPClient *client);
+
+  /*< private >*/
+  gpointer         _gst_reserved[GST_PADDING_LARGE];
+};
+
+GType                 gst_rtsp_server_get_type             (void);
+
+GstRTSPServer *       gst_rtsp_server_new                  (void);
+
+void                  gst_rtsp_server_set_address          (GstRTSPServer *server, const gchar *address);
+gchar *               gst_rtsp_server_get_address          (GstRTSPServer *server);
+
+void                  gst_rtsp_server_set_service          (GstRTSPServer *server, const gchar *service);
+gchar *               gst_rtsp_server_get_service          (GstRTSPServer *server);
+
+void                  gst_rtsp_server_set_backlog          (GstRTSPServer *server, gint backlog);
+gint                  gst_rtsp_server_get_backlog          (GstRTSPServer *server);
+
+int                   gst_rtsp_server_get_bound_port       (GstRTSPServer *server);
+
+void                  gst_rtsp_server_set_session_pool     (GstRTSPServer *server, GstRTSPSessionPool *pool);
+GstRTSPSessionPool *  gst_rtsp_server_get_session_pool     (GstRTSPServer *server);
+
+void                  gst_rtsp_server_set_mount_points     (GstRTSPServer *server, GstRTSPMountPoints *mounts);
+GstRTSPMountPoints *  gst_rtsp_server_get_mount_points     (GstRTSPServer *server);
+
+void                  gst_rtsp_server_set_auth             (GstRTSPServer *server, GstRTSPAuth *auth);
+GstRTSPAuth *         gst_rtsp_server_get_auth             (GstRTSPServer *server);
+
+void                  gst_rtsp_server_set_thread_pool      (GstRTSPServer *server, GstRTSPThreadPool *pool);
+GstRTSPThreadPool *   gst_rtsp_server_get_thread_pool      (GstRTSPServer *server);
+
+gboolean              gst_rtsp_server_transfer_connection  (GstRTSPServer * server, GSocket *socket,
+                                                            const gchar * ip, gint port,
+                                                            const gchar *initial_buffer);
+
+gboolean              gst_rtsp_server_io_func              (GSocket *socket, GIOCondition condition,
+                                                            GstRTSPServer *server);
+
+GSocket *             gst_rtsp_server_create_socket        (GstRTSPServer *server,
+                                                            GCancellable  *cancellable,
+                                                            GError **error);
+GSource *             gst_rtsp_server_create_source        (GstRTSPServer *server,
+                                                            GCancellable * cancellable,
+                                                            GError **error);
+guint                 gst_rtsp_server_attach               (GstRTSPServer *server,
+                                                            GMainContext *context);
+
+/**
+ * GstRTSPServerClientFilterFunc:
+ * @server: a #GstRTSPServer object
+ * @client: a #GstRTSPClient in @server
+ * @user_data: user data that has been given to gst_rtsp_server_client_filter()
+ *
+ * This function will be called by the gst_rtsp_server_client_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @client will be removed
+ * from @server.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @client untouched in
+ * @server.
+ *
+ * A value of #GST_RTSP_FILTER_REF will add @client to the result #GList of
+ * gst_rtsp_server_client_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPServerClientFilterFunc)  (GstRTSPServer *server,
+                                                               GstRTSPClient *client,
+                                                               gpointer user_data);
+
+GList *                gst_rtsp_server_client_filter    (GstRTSPServer *server,
+                                                         GstRTSPServerClientFilterFunc func,
+                                                         gpointer user_data);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SERVER_H__ */
diff --git a/gst/rtsp-server/rtsp-session-media.c b/gst/rtsp-server/rtsp-session-media.c
new file mode 100644
index 0000000..8eb1f28
--- /dev/null
+++ b/gst/rtsp-server/rtsp-session-media.c
@@ -0,0 +1,505 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-session-media
+ * @short_description: Media managed in a session
+ * @see_also: #GstRTSPMedia, #GstRTSPSession
+ *
+ * The #GstRTSPSessionMedia object manages a #GstRTSPMedia with a given path.
+ *
+ * With gst_rtsp_session_media_get_transport() and
+ * gst_rtsp_session_media_set_transport() the transports of a #GstRTSPStream of
+ * the managed #GstRTSPMedia can be retrieved and configured.
+ *
+ * Use gst_rtsp_session_media_set_state() to control the media state and
+ * transports.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
+
+#include 
+
+#include "rtsp-session.h"
+
+#define GST_RTSP_SESSION_MEDIA_GET_PRIVATE(obj)  \
+    (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMediaPrivate))
+
+struct _GstRTSPSessionMediaPrivate
+{
+  GMutex lock;
+  gchar *path;                  /* unmutable */
+  gint path_len;                /* unmutable */
+  GstRTSPMedia *media;          /* unmutable */
+  GstRTSPState state;           /* protected by lock */
+  guint counter;                /* protected by lock */
+
+  GPtrArray *transports;        /* protected by lock */
+};
+
+enum
+{
+  PROP_0,
+  PROP_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_session_media_debug);
+#define GST_CAT_DEFAULT rtsp_session_media_debug
+
+static void gst_rtsp_session_media_finalize (GObject * obj);
+
+G_DEFINE_TYPE (GstRTSPSessionMedia, gst_rtsp_session_media, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_session_media_class_init (GstRTSPSessionMediaClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPSessionMediaPrivate));
+
+  gobject_class = G_OBJECT_CLASS (klass);
+
+  gobject_class->finalize = gst_rtsp_session_media_finalize;
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_session_media_debug, "rtspsessionmedia", 0,
+      "GstRTSPSessionMedia");
+}
+
+static void
+gst_rtsp_session_media_init (GstRTSPSessionMedia * media)
+{
+  GstRTSPSessionMediaPrivate *priv = GST_RTSP_SESSION_MEDIA_GET_PRIVATE (media);
+
+  media->priv = priv;
+
+  g_mutex_init (&priv->lock);
+  priv->state = GST_RTSP_STATE_INIT;
+}
+
+static void
+gst_rtsp_session_media_finalize (GObject * obj)
+{
+  GstRTSPSessionMedia *media;
+  GstRTSPSessionMediaPrivate *priv;
+
+  media = GST_RTSP_SESSION_MEDIA (obj);
+  priv = media->priv;
+
+  GST_INFO ("free session media %p", media);
+
+  gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
+
+  g_ptr_array_unref (priv->transports);
+
+  g_free (priv->path);
+  g_object_unref (priv->media);
+  g_mutex_clear (&priv->lock);
+
+  G_OBJECT_CLASS (gst_rtsp_session_media_parent_class)->finalize (obj);
+}
+
+static void
+free_session_media (gpointer data)
+{
+  if (data)
+    g_object_unref (data);
+}
+
+/**
+ * gst_rtsp_session_media_new:
+ * @path: the path
+ * @media: (transfer full): the #GstRTSPMedia
+ *
+ * Create a new #GstRTSPSessionMedia that manages the streams
+ * in @media for @path. @media should be prepared.
+ *
+ * Ownership is taken of @media.
+ *
+ * Returns: (transfer full): a new #GstRTSPSessionMedia.
+ */
+GstRTSPSessionMedia *
+gst_rtsp_session_media_new (const gchar * path, GstRTSPMedia * media)
+{
+  GstRTSPSessionMediaPrivate *priv;
+  GstRTSPSessionMedia *result;
+  guint n_streams;
+  GstRTSPMediaStatus status;
+
+  g_return_val_if_fail (path != NULL, NULL);
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+  status = gst_rtsp_media_get_status (media);
+  g_return_val_if_fail (status == GST_RTSP_MEDIA_STATUS_PREPARED || status ==
+      GST_RTSP_MEDIA_STATUS_SUSPENDED, NULL);
+
+  result = g_object_new (GST_TYPE_RTSP_SESSION_MEDIA, NULL);
+  priv = result->priv;
+
+  priv->path = g_strdup (path);
+  priv->path_len = strlen (path);
+  priv->media = media;
+
+  /* prealloc the streams now, filled with NULL */
+  n_streams = gst_rtsp_media_n_streams (media);
+  priv->transports = g_ptr_array_new_full (n_streams, free_session_media);
+  g_ptr_array_set_size (priv->transports, n_streams);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_session_media_matches:
+ * @media: a #GstRTSPSessionMedia
+ * @path: a path
+ * @matched: (out): the amount of matched characters of @path
+ *
+ * Check if the path of @media matches @path. It @path matches, the amount of
+ * matched characters is returned in @matched.
+ *
+ * Returns: %TRUE when @path matches the path of @media.
+ */
+gboolean
+gst_rtsp_session_media_matches (GstRTSPSessionMedia * media,
+    const gchar * path, gint * matched)
+{
+  GstRTSPSessionMediaPrivate *priv;
+  gint len;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
+  g_return_val_if_fail (path != NULL, FALSE);
+  g_return_val_if_fail (matched != NULL, FALSE);
+
+  priv = media->priv;
+  len = strlen (path);
+
+  /* path needs to be smaller than the media path */
+  if (len < priv->path_len)
+    return FALSE;
+
+  /* if media path is larger, it there should be a / following the path */
+  if (len > priv->path_len && path[priv->path_len] != '/')
+    return FALSE;
+
+  *matched = priv->path_len;
+
+  return strncmp (path, priv->path, priv->path_len) == 0;
+}
+
+/**
+ * gst_rtsp_session_media_get_media:
+ * @media: a #GstRTSPSessionMedia
+ *
+ * Get the #GstRTSPMedia that was used when constructing @media
+ *
+ * Returns: (transfer none): the #GstRTSPMedia of @media. Remains valid as long
+ * as @media is valid.
+ */
+GstRTSPMedia *
+gst_rtsp_session_media_get_media (GstRTSPSessionMedia * media)
+{
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
+
+  return media->priv->media;
+}
+
+/**
+ * gst_rtsp_session_media_get_base_time:
+ * @media: a #GstRTSPSessionMedia
+ *
+ * Get the base_time of the #GstRTSPMedia in @media
+ *
+ * Returns: the base_time of the media.
+ */
+GstClockTime
+gst_rtsp_session_media_get_base_time (GstRTSPSessionMedia * media)
+{
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), GST_CLOCK_TIME_NONE);
+
+  return gst_rtsp_media_get_base_time (media->priv->media);
+}
+
+/**
+ * gst_rtsp_session_media_get_rtpinfo:
+ * @media: a #GstRTSPSessionMedia
+ *
+ * Retrieve the RTP-Info header string for all streams in @media
+ * with configured transports.
+ *
+ * Returns: (transfer full) (nullable): The RTP-Info as a string or
+ * %NULL when no RTP-Info could be generated, g_free() after usage.
+ */
+gchar *
+gst_rtsp_session_media_get_rtpinfo (GstRTSPSessionMedia * media)
+{
+  GstRTSPSessionMediaPrivate *priv;
+  GString *rtpinfo = NULL;
+  GstRTSPStreamTransport *transport;
+  GstRTSPStream *stream;
+  guint i, n_streams;
+  GstClockTime earliest = GST_CLOCK_TIME_NONE;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
+
+  priv = media->priv;
+  g_mutex_lock (&priv->lock);
+
+  if (gst_rtsp_media_get_status (priv->media) != GST_RTSP_MEDIA_STATUS_PREPARED)
+    goto not_prepared;
+
+  n_streams = priv->transports->len;
+
+  /* first step, take lowest running-time from all streams */
+  GST_LOG_OBJECT (media, "determining start time among %d transports",
+      n_streams);
+
+  for (i = 0; i < n_streams; i++) {
+    GstClockTime running_time;
+
+    transport = g_ptr_array_index (priv->transports, i);
+    if (transport == NULL) {
+      GST_DEBUG_OBJECT (media, "ignoring unconfigured transport %d", i);
+      continue;
+    }
+
+    stream = gst_rtsp_stream_transport_get_stream (transport);
+    if (!gst_rtsp_stream_get_rtpinfo (stream, NULL, NULL, NULL, &running_time))
+      continue;
+
+    GST_LOG_OBJECT (media, "running time of %d stream: %" GST_TIME_FORMAT, i,
+        GST_TIME_ARGS (running_time));
+
+    if (!GST_CLOCK_TIME_IS_VALID (earliest)) {
+      earliest = running_time;
+    } else {
+      earliest = MIN (earliest, running_time);
+    }
+  }
+
+  GST_LOG_OBJECT (media, "media start time: %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (earliest));
+
+  /* next step, scale all rtptime of all streams to lowest running-time */
+  GST_LOG_OBJECT (media, "collecting RTP info for %d transports", n_streams);
+
+  for (i = 0; i < n_streams; i++) {
+    gchar *stream_rtpinfo;
+
+    transport = g_ptr_array_index (priv->transports, i);
+    if (transport == NULL) {
+      GST_DEBUG_OBJECT (media, "ignoring unconfigured transport %d", i);
+      continue;
+    }
+
+    stream_rtpinfo =
+        gst_rtsp_stream_transport_get_rtpinfo (transport, earliest);
+    if (stream_rtpinfo == NULL) {
+      GST_DEBUG_OBJECT (media, "ignoring unknown RTPInfo %d", i);
+      continue;
+    }
+
+    if (rtpinfo == NULL)
+      rtpinfo = g_string_new ("");
+    else
+      g_string_append (rtpinfo, ", ");
+
+    g_string_append (rtpinfo, stream_rtpinfo);
+    g_free (stream_rtpinfo);
+  }
+  g_mutex_unlock (&priv->lock);
+
+  if (rtpinfo == NULL) {
+    GST_WARNING_OBJECT (media, "RTP info is empty");
+    return NULL;
+  }
+  return g_string_free (rtpinfo, FALSE);
+
+  /* ERRORS */
+not_prepared:
+  {
+    g_mutex_unlock (&priv->lock);
+    GST_ERROR_OBJECT (media, "media was not prepared");
+    return NULL;
+  }
+}
+
+/**
+ * gst_rtsp_session_media_set_transport:
+ * @media: a #GstRTSPSessionMedia
+ * @stream: a #GstRTSPStream
+ * @tr: (transfer full): a #GstRTSPTransport
+ *
+ * Configure the transport for @stream to @tr in @media.
+ *
+ * Returns: (transfer none): the new or updated #GstRTSPStreamTransport for @stream.
+ */
+GstRTSPStreamTransport *
+gst_rtsp_session_media_set_transport (GstRTSPSessionMedia * media,
+    GstRTSPStream * stream, GstRTSPTransport * tr)
+{
+  GstRTSPSessionMediaPrivate *priv;
+  GstRTSPStreamTransport *result;
+  guint idx;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+  g_return_val_if_fail (tr != NULL, NULL);
+  priv = media->priv;
+  idx = gst_rtsp_stream_get_index (stream);
+  g_return_val_if_fail (idx < priv->transports->len, NULL);
+
+  g_mutex_lock (&priv->lock);
+  result = g_ptr_array_index (priv->transports, idx);
+  if (result == NULL) {
+    result = gst_rtsp_stream_transport_new (stream, tr);
+    g_ptr_array_index (priv->transports, idx) = result;
+    g_mutex_unlock (&priv->lock);
+  } else {
+    gst_rtsp_stream_transport_set_transport (result, tr);
+    g_mutex_unlock (&priv->lock);
+  }
+
+  return result;
+}
+
+/**
+ * gst_rtsp_session_media_get_transport:
+ * @media: a #GstRTSPSessionMedia
+ * @idx: the stream index
+ *
+ * Get a previously created #GstRTSPStreamTransport for the stream at @idx.
+ *
+ * Returns: (transfer none): a #GstRTSPStreamTransport that is valid until the
+ * session of @media is unreffed.
+ */
+GstRTSPStreamTransport *
+gst_rtsp_session_media_get_transport (GstRTSPSessionMedia * media, guint idx)
+{
+  GstRTSPSessionMediaPrivate *priv;
+  GstRTSPStreamTransport *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
+  priv = media->priv;
+  g_return_val_if_fail (idx < priv->transports->len, NULL);
+
+  g_mutex_lock (&priv->lock);
+  result = g_ptr_array_index (priv->transports, idx);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_session_media_alloc_channels:
+ * @media: a #GstRTSPSessionMedia
+ * @range: (out): a #GstRTSPRange
+ *
+ * Fill @range with the next available min and max channels for
+ * interleaved transport.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_session_media_alloc_channels (GstRTSPSessionMedia * media,
+    GstRTSPRange * range)
+{
+  GstRTSPSessionMediaPrivate *priv;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  range->min = priv->counter++;
+  range->max = priv->counter++;
+  g_mutex_unlock (&priv->lock);
+
+  return TRUE;
+}
+
+/**
+ * gst_rtsp_session_media_set_state:
+ * @media: a #GstRTSPSessionMedia
+ * @state: the new state
+ *
+ * Tell the media object @media to change to @state.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_session_media_set_state (GstRTSPSessionMedia * media, GstState state)
+{
+  GstRTSPSessionMediaPrivate *priv;
+  gboolean ret;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  ret = gst_rtsp_media_set_state (priv->media, state, priv->transports);
+  g_mutex_unlock (&priv->lock);
+
+  return ret;
+}
+
+/**
+ * gst_rtsp_session_media_set_rtsp_state:
+ * @media: a #GstRTSPSessionMedia
+ * @state: a #GstRTSPState
+ *
+ * Set the RTSP state of @media to @state.
+ */
+void
+gst_rtsp_session_media_set_rtsp_state (GstRTSPSessionMedia * media,
+    GstRTSPState state)
+{
+  GstRTSPSessionMediaPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_SESSION_MEDIA (media));
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->state = state;
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_session_media_get_rtsp_state:
+ * @media: a #GstRTSPSessionMedia
+ *
+ * Get the current RTSP state of @media.
+ *
+ * Returns: the current RTSP state of @media.
+ */
+GstRTSPState
+gst_rtsp_session_media_get_rtsp_state (GstRTSPSessionMedia * media)
+{
+  GstRTSPSessionMediaPrivate *priv;
+  GstRTSPState ret;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media),
+      GST_RTSP_STATE_INVALID);
+
+  priv = media->priv;
+
+  g_mutex_lock (&priv->lock);
+  ret = priv->state;
+  g_mutex_unlock (&priv->lock);
+
+  return ret;
+}
diff --git a/gst/rtsp-server/rtsp-session-media.h b/gst/rtsp-server/rtsp-session-media.h
new file mode 100644
index 0000000..e2af5ad
--- /dev/null
+++ b/gst/rtsp-server/rtsp-session-media.h
@@ -0,0 +1,97 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+#ifndef __GST_RTSP_SESSION_MEDIA_H__
+#define __GST_RTSP_SESSION_MEDIA_H__
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_SESSION_MEDIA              (gst_rtsp_session_media_get_type ())
+#define GST_IS_RTSP_SESSION_MEDIA(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_SESSION_MEDIA))
+#define GST_IS_RTSP_SESSION_MEDIA_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_SESSION_MEDIA))
+#define GST_RTSP_SESSION_MEDIA_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMediaClass))
+#define GST_RTSP_SESSION_MEDIA(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMedia))
+#define GST_RTSP_SESSION_MEDIA_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_SESSION_MEDIA, GstRTSPSessionMediaClass))
+#define GST_RTSP_SESSION_MEDIA_CAST(obj)         ((GstRTSPSessionMedia*)(obj))
+#define GST_RTSP_SESSION_MEDIA_CLASS_CAST(klass) ((GstRTSPSessionMediaClass*)(klass))
+
+typedef struct _GstRTSPSessionMedia GstRTSPSessionMedia;
+typedef struct _GstRTSPSessionMediaClass GstRTSPSessionMediaClass;
+typedef struct _GstRTSPSessionMediaPrivate GstRTSPSessionMediaPrivate;
+
+/**
+ * GstRTSPSessionMedia:
+ *
+ * State of a client session regarding a specific media identified by path.
+ */
+struct _GstRTSPSessionMedia
+{
+  GObject  parent;
+
+  /*< private >*/
+  GstRTSPSessionMediaPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+struct _GstRTSPSessionMediaClass
+{
+  GObjectClass  parent_class;
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+GType                    gst_rtsp_session_media_get_type       (void);
+
+GstRTSPSessionMedia *    gst_rtsp_session_media_new            (const gchar *path,
+                                                                GstRTSPMedia *media);
+
+gboolean                 gst_rtsp_session_media_matches        (GstRTSPSessionMedia *media,
+                                                                const gchar *path,
+                                                                gint * matched);
+GstRTSPMedia *           gst_rtsp_session_media_get_media      (GstRTSPSessionMedia *media);
+
+GstClockTime             gst_rtsp_session_media_get_base_time  (GstRTSPSessionMedia *media);
+/* control media */
+gboolean                 gst_rtsp_session_media_set_state      (GstRTSPSessionMedia *media,
+                                                                GstState state);
+
+void                     gst_rtsp_session_media_set_rtsp_state (GstRTSPSessionMedia *media,
+                                                                GstRTSPState state);
+GstRTSPState             gst_rtsp_session_media_get_rtsp_state (GstRTSPSessionMedia *media);
+
+/* get stream transport config */
+GstRTSPStreamTransport * gst_rtsp_session_media_set_transport  (GstRTSPSessionMedia *media,
+                                                                GstRTSPStream *stream,
+                                                                GstRTSPTransport *tr);
+GstRTSPStreamTransport * gst_rtsp_session_media_get_transport  (GstRTSPSessionMedia *media,
+                                                                guint idx);
+
+gboolean                 gst_rtsp_session_media_alloc_channels (GstRTSPSessionMedia *media,
+                                                                GstRTSPRange *range);
+
+gchar *                  gst_rtsp_session_media_get_rtpinfo    (GstRTSPSessionMedia * media);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SESSION_MEDIA_H__ */
diff --git a/gst/rtsp-server/rtsp-session-pool.c b/gst/rtsp-server/rtsp-session-pool.c
new file mode 100644
index 0000000..fad5288
--- /dev/null
+++ b/gst/rtsp-server/rtsp-session-pool.c
@@ -0,0 +1,765 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-session-pool
+ * @short_description: An object for managing sessions
+ * @see_also: #GstRTSPSession
+ *
+ * The #GstRTSPSessionPool object manages a list of #GstRTSPSession objects.
+ *
+ * The maximum number of sessions can be configured with
+ * gst_rtsp_session_pool_set_max_sessions(). The current number of sessions can
+ * be retrieved with gst_rtsp_session_pool_get_n_sessions().
+ *
+ * Use gst_rtsp_session_pool_create() to create a new #GstRTSPSession object.
+ * The session object can be found again with its id and
+ * gst_rtsp_session_pool_find().
+ *
+ * All sessions can be iterated with gst_rtsp_session_pool_filter().
+ *
+ * Run gst_rtsp_session_pool_cleanup() periodically to remove timed out sessions
+ * or use gst_rtsp_session_pool_create_watch() to be notified when session
+ * cleanup should be performed.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include "rtsp-session-pool.h"
+
+#define GST_RTSP_SESSION_POOL_GET_PRIVATE(obj)  \
+         (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_SESSION_POOL, GstRTSPSessionPoolPrivate))
+
+struct _GstRTSPSessionPoolPrivate
+{
+  GMutex lock;                  /* protects everything in this struct */
+  guint max_sessions;
+  GHashTable *sessions;
+  guint sessions_cookie;
+};
+
+#define DEFAULT_MAX_SESSIONS 0
+
+enum
+{
+  PROP_0,
+  PROP_MAX_SESSIONS,
+  PROP_LAST
+};
+
+static const gchar session_id_charset[] =
+    { 'a', 'b', 'c', 'd', 'e', 'f', 'g', 'h', 'i', 'j', 'k', 'l', 'm', 'n', 'o',
+  'p', 'q', 'r', 's', 't', 'u', 'v', 'w', 'x', 'y', 'z', 'A', 'B', 'C', 'D',
+  'E', 'F', 'G', 'H', 'I', 'J', 'K', 'L', 'M', 'N', 'O', 'P', 'Q', 'R', 'S',
+  'T', 'U', 'V', 'W', 'X', 'Y', 'Z', '0', '1', '2', '3', '4', '5', '6', '7',
+  '8', '9', '$', '-', '_', '.', '+'
+};
+
+enum
+{
+  SIGNAL_SESSION_REMOVED,
+  SIGNAL_LAST
+};
+
+static guint gst_rtsp_session_pool_signals[SIGNAL_LAST] = { 0 };
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_session_debug);
+#define GST_CAT_DEFAULT rtsp_session_debug
+
+static void gst_rtsp_session_pool_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec);
+static void gst_rtsp_session_pool_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_session_pool_finalize (GObject * object);
+
+static gchar *create_session_id (GstRTSPSessionPool * pool);
+static GstRTSPSession *create_session (GstRTSPSessionPool * pool,
+    const gchar * id);
+
+G_DEFINE_TYPE (GstRTSPSessionPool, gst_rtsp_session_pool, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_session_pool_class_init (GstRTSPSessionPoolClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPSessionPoolPrivate));
+
+  gobject_class = G_OBJECT_CLASS (klass);
+
+  gobject_class->get_property = gst_rtsp_session_pool_get_property;
+  gobject_class->set_property = gst_rtsp_session_pool_set_property;
+  gobject_class->finalize = gst_rtsp_session_pool_finalize;
+
+  g_object_class_install_property (gobject_class, PROP_MAX_SESSIONS,
+      g_param_spec_uint ("max-sessions", "Max Sessions",
+          "the maximum amount of sessions (0 = unlimited)",
+          0, G_MAXUINT, DEFAULT_MAX_SESSIONS,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  gst_rtsp_session_pool_signals[SIGNAL_SESSION_REMOVED] =
+      g_signal_new ("session-removed", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPSessionPoolClass,
+          session_removed), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE,
+      1, GST_TYPE_RTSP_SESSION);
+
+  klass->create_session_id = create_session_id;
+  klass->create_session = create_session;
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_session_debug, "rtspsessionpool", 0,
+      "GstRTSPSessionPool");
+}
+
+static void
+gst_rtsp_session_pool_init (GstRTSPSessionPool * pool)
+{
+  GstRTSPSessionPoolPrivate *priv = GST_RTSP_SESSION_POOL_GET_PRIVATE (pool);
+
+  pool->priv = priv;
+
+  g_mutex_init (&priv->lock);
+  priv->sessions = g_hash_table_new_full (g_str_hash, g_str_equal,
+      NULL, g_object_unref);
+  priv->max_sessions = DEFAULT_MAX_SESSIONS;
+}
+
+static GstRTSPFilterResult
+remove_sessions_func (GstRTSPSessionPool * pool, GstRTSPSession * session,
+    gpointer user_data)
+{
+  return GST_RTSP_FILTER_REMOVE;
+}
+
+static void
+gst_rtsp_session_pool_finalize (GObject * object)
+{
+  GstRTSPSessionPool *pool = GST_RTSP_SESSION_POOL (object);
+  GstRTSPSessionPoolPrivate *priv = pool->priv;
+
+  gst_rtsp_session_pool_filter (pool, remove_sessions_func, NULL);
+  g_hash_table_unref (priv->sessions);
+  g_mutex_clear (&priv->lock);
+
+  G_OBJECT_CLASS (gst_rtsp_session_pool_parent_class)->finalize (object);
+}
+
+static void
+gst_rtsp_session_pool_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRTSPSessionPool *pool = GST_RTSP_SESSION_POOL (object);
+
+  switch (propid) {
+    case PROP_MAX_SESSIONS:
+      g_value_set_uint (value, gst_rtsp_session_pool_get_max_sessions (pool));
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+      break;
+  }
+}
+
+static void
+gst_rtsp_session_pool_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRTSPSessionPool *pool = GST_RTSP_SESSION_POOL (object);
+
+  switch (propid) {
+    case PROP_MAX_SESSIONS:
+      gst_rtsp_session_pool_set_max_sessions (pool, g_value_get_uint (value));
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+      break;
+  }
+}
+
+/**
+ * gst_rtsp_session_pool_new:
+ *
+ * Create a new #GstRTSPSessionPool instance.
+ *
+ * Returns: (transfer full): A new #GstRTSPSessionPool. g_object_unref() after
+ * usage.
+ */
+GstRTSPSessionPool *
+gst_rtsp_session_pool_new (void)
+{
+  GstRTSPSessionPool *result;
+
+  result = g_object_new (GST_TYPE_RTSP_SESSION_POOL, NULL);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_session_pool_set_max_sessions:
+ * @pool: a #GstRTSPSessionPool
+ * @max: the maximum number of sessions
+ *
+ * Configure the maximum allowed number of sessions in @pool to @max.
+ * A value of 0 means an unlimited amount of sessions.
+ */
+void
+gst_rtsp_session_pool_set_max_sessions (GstRTSPSessionPool * pool, guint max)
+{
+  GstRTSPSessionPoolPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_SESSION_POOL (pool));
+
+  priv = pool->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->max_sessions = max;
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_session_pool_get_max_sessions:
+ * @pool: a #GstRTSPSessionPool
+ *
+ * Get the maximum allowed number of sessions in @pool. 0 means an unlimited
+ * amount of sessions.
+ *
+ * Returns: the maximum allowed number of sessions.
+ */
+guint
+gst_rtsp_session_pool_get_max_sessions (GstRTSPSessionPool * pool)
+{
+  GstRTSPSessionPoolPrivate *priv;
+  guint result;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), 0);
+
+  priv = pool->priv;
+
+  g_mutex_lock (&priv->lock);
+  result = priv->max_sessions;
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_session_pool_get_n_sessions:
+ * @pool: a #GstRTSPSessionPool
+ *
+ * Get the amount of active sessions in @pool.
+ *
+ * Returns: the amount of active sessions in @pool.
+ */
+guint
+gst_rtsp_session_pool_get_n_sessions (GstRTSPSessionPool * pool)
+{
+  GstRTSPSessionPoolPrivate *priv;
+  guint result;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), 0);
+
+  priv = pool->priv;
+
+  g_mutex_lock (&priv->lock);
+  result = g_hash_table_size (priv->sessions);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_session_pool_find:
+ * @pool: the pool to search
+ * @sessionid: the session id
+ *
+ * Find the session with @sessionid in @pool. The access time of the session
+ * will be updated with gst_rtsp_session_touch().
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPSession with @sessionid
+ * or %NULL when the session did not exist. g_object_unref() after usage.
+ */
+GstRTSPSession *
+gst_rtsp_session_pool_find (GstRTSPSessionPool * pool, const gchar * sessionid)
+{
+  GstRTSPSessionPoolPrivate *priv;
+  GstRTSPSession *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), NULL);
+  g_return_val_if_fail (sessionid != NULL, NULL);
+
+  priv = pool->priv;
+
+  g_mutex_lock (&priv->lock);
+  result = g_hash_table_lookup (priv->sessions, sessionid);
+  if (result) {
+    g_object_ref (result);
+    gst_rtsp_session_touch (result);
+  }
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+static gchar *
+create_session_id (GstRTSPSessionPool * pool)
+{
+  gchar id[17];
+  gint i;
+
+  for (i = 0; i < 16; i++) {
+    id[i] =
+        session_id_charset[g_random_int_range (0,
+            G_N_ELEMENTS (session_id_charset))];
+  }
+  id[16] = 0;
+
+  return g_uri_escape_string (id, NULL, FALSE);
+}
+
+static GstRTSPSession *
+create_session (GstRTSPSessionPool * pool, const gchar * id)
+{
+  return gst_rtsp_session_new (id);
+}
+
+/**
+ * gst_rtsp_session_pool_create:
+ * @pool: a #GstRTSPSessionPool
+ *
+ * Create a new #GstRTSPSession object in @pool.
+ *
+ * Returns: (transfer full): a new #GstRTSPSession.
+ */
+GstRTSPSession *
+gst_rtsp_session_pool_create (GstRTSPSessionPool * pool)
+{
+  GstRTSPSessionPoolPrivate *priv;
+  GstRTSPSession *result = NULL;
+  GstRTSPSessionPoolClass *klass;
+  gchar *id = NULL;
+  guint retry;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), NULL);
+
+  priv = pool->priv;
+
+  klass = GST_RTSP_SESSION_POOL_GET_CLASS (pool);
+
+  retry = 0;
+  do {
+    /* start by creating a new random session id, we assume that this is random
+     * enough to not cause a collision, which we will check later  */
+    if (klass->create_session_id)
+      id = klass->create_session_id (pool);
+    else
+      goto no_function;
+
+    if (id == NULL)
+      goto no_session;
+
+    g_mutex_lock (&priv->lock);
+    /* check session limit */
+    if (priv->max_sessions > 0) {
+      if (g_hash_table_size (priv->sessions) >= priv->max_sessions)
+        goto too_many_sessions;
+    }
+    /* check if the sessionid existed */
+    result = g_hash_table_lookup (priv->sessions, id);
+    if (result) {
+      /* found, retry with a different session id */
+      result = NULL;
+      retry++;
+      if (retry > 100)
+        goto collision;
+    } else {
+      /* not found, create session and insert it in the pool */
+      if (klass->create_session)
+        result = create_session (pool, id);
+      if (result == NULL)
+        goto too_many_sessions;
+      /* take additional ref for the pool */
+      g_object_ref (result);
+      g_hash_table_insert (priv->sessions,
+          (gchar *) gst_rtsp_session_get_sessionid (result), result);
+      priv->sessions_cookie++;
+    }
+    g_mutex_unlock (&priv->lock);
+
+    g_free (id);
+  } while (result == NULL);
+
+  return result;
+
+  /* ERRORS */
+no_function:
+  {
+    GST_WARNING ("no create_session_id vmethod in GstRTSPSessionPool %p", pool);
+    return NULL;
+  }
+no_session:
+  {
+    GST_WARNING ("can't create session id with GstRTSPSessionPool %p", pool);
+    return NULL;
+  }
+collision:
+  {
+    GST_WARNING ("can't find unique sessionid for GstRTSPSessionPool %p", pool);
+    g_mutex_unlock (&priv->lock);
+    g_free (id);
+    return NULL;
+  }
+too_many_sessions:
+  {
+    GST_WARNING ("session pool reached max sessions of %d", priv->max_sessions);
+    g_mutex_unlock (&priv->lock);
+    g_free (id);
+    return NULL;
+  }
+}
+
+/**
+ * gst_rtsp_session_pool_remove:
+ * @pool: a #GstRTSPSessionPool
+ * @sess: (transfer none): a #GstRTSPSession
+ *
+ * Remove @sess from @pool, releasing the ref that the pool has on @sess.
+ *
+ * Returns: %TRUE if the session was found and removed.
+ */
+gboolean
+gst_rtsp_session_pool_remove (GstRTSPSessionPool * pool, GstRTSPSession * sess)
+{
+  GstRTSPSessionPoolPrivate *priv;
+  gboolean found;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), FALSE);
+  g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), FALSE);
+
+  priv = pool->priv;
+
+  g_mutex_lock (&priv->lock);
+  g_object_ref (sess);
+  found =
+      g_hash_table_remove (priv->sessions,
+      gst_rtsp_session_get_sessionid (sess));
+  if (found)
+    priv->sessions_cookie++;
+  g_mutex_unlock (&priv->lock);
+
+  if (found)
+    g_signal_emit (pool, gst_rtsp_session_pool_signals[SIGNAL_SESSION_REMOVED],
+        0, sess);
+
+  g_object_unref (sess);
+
+  return found;
+}
+
+typedef struct
+{
+  GTimeVal now;
+  GstRTSPSessionPool *pool;
+  GList *removed;
+} CleanupData;
+
+static gboolean
+cleanup_func (gchar * sessionid, GstRTSPSession * sess, CleanupData * data)
+{
+  gboolean expired;
+
+  expired = gst_rtsp_session_is_expired (sess, &data->now);
+  if (expired) {
+    GST_DEBUG ("session expired");
+    data->removed = g_list_prepend (data->removed, g_object_ref (sess));
+  }
+  return expired;
+}
+
+/**
+ * gst_rtsp_session_pool_cleanup:
+ * @pool: a #GstRTSPSessionPool
+ *
+ * Inspect all the sessions in @pool and remove the sessions that are inactive
+ * for more than their timeout.
+ *
+ * Returns: the amount of sessions that got removed.
+ */
+guint
+gst_rtsp_session_pool_cleanup (GstRTSPSessionPool * pool)
+{
+  GstRTSPSessionPoolPrivate *priv;
+  guint result;
+  CleanupData data;
+  GList *walk;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), 0);
+
+  priv = pool->priv;
+
+  g_get_current_time (&data.now);
+  data.pool = pool;
+  data.removed = NULL;
+
+  g_mutex_lock (&priv->lock);
+  result =
+      g_hash_table_foreach_remove (priv->sessions, (GHRFunc) cleanup_func,
+      &data);
+  if (result > 0)
+    priv->sessions_cookie++;
+  g_mutex_unlock (&priv->lock);
+
+  for (walk = data.removed; walk; walk = walk->next) {
+    GstRTSPSession *sess = walk->data;
+
+    g_signal_emit (pool,
+        gst_rtsp_session_pool_signals[SIGNAL_SESSION_REMOVED], 0, sess);
+
+    g_object_unref (sess);
+  }
+  g_list_free (data.removed);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_session_pool_filter:
+ * @pool: a #GstRTSPSessionPool
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: (closure): user data passed to @func
+ *
+ * Call @func for each session in @pool. The result value of @func determines
+ * what happens to the session. @func will be called with the session pool
+ * locked so no further actions on @pool can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the session will be set to the
+ * expired state with gst_rtsp_session_set_expired() and removed from
+ * @pool.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the session will remain in @pool.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the session will remain in @pool but
+ * will also be added with an additional ref to the result GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for all sessions.
+ *
+ * Returns: (element-type GstRTSPSession) (transfer full): a GList with all
+ * sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_session_pool_filter (GstRTSPSessionPool * pool,
+    GstRTSPSessionPoolFilterFunc func, gpointer user_data)
+{
+  GstRTSPSessionPoolPrivate *priv;
+  GHashTableIter iter;
+  gpointer key, value;
+  GList *result;
+  GHashTable *visited;
+  guint cookie;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), NULL);
+
+  priv = pool->priv;
+
+  result = NULL;
+  if (func)
+    visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+  g_mutex_lock (&priv->lock);
+restart:
+  g_hash_table_iter_init (&iter, priv->sessions);
+  cookie = priv->sessions_cookie;
+  while (g_hash_table_iter_next (&iter, &key, &value)) {
+    GstRTSPSession *session = value;
+    GstRTSPFilterResult res;
+    gboolean changed;
+
+    if (func) {
+      /* only visit each session once */
+      if (g_hash_table_contains (visited, session))
+        continue;
+
+      g_hash_table_add (visited, g_object_ref (session));
+      g_mutex_unlock (&priv->lock);
+
+      res = func (pool, session, user_data);
+
+      g_mutex_lock (&priv->lock);
+    } else
+      res = GST_RTSP_FILTER_REF;
+
+    changed = (cookie != priv->sessions_cookie);
+
+    switch (res) {
+      case GST_RTSP_FILTER_REMOVE:
+      {
+        gboolean removed = TRUE;
+
+        if (changed)
+          /* something changed, check if we still have the session */
+          removed = g_hash_table_remove (priv->sessions, key);
+        else
+          g_hash_table_iter_remove (&iter);
+
+        if (removed) {
+          /* if we managed to remove the session, update the cookie and
+           * signal */
+          cookie = ++priv->sessions_cookie;
+          g_mutex_unlock (&priv->lock);
+
+          g_signal_emit (pool,
+              gst_rtsp_session_pool_signals[SIGNAL_SESSION_REMOVED], 0,
+              session);
+
+          g_mutex_lock (&priv->lock);
+          /* cookie could have changed again, make sure we restart */
+          changed |= (cookie != priv->sessions_cookie);
+        }
+        break;
+      }
+      case GST_RTSP_FILTER_REF:
+        /* keep ref */
+        result = g_list_prepend (result, g_object_ref (session));
+        break;
+      case GST_RTSP_FILTER_KEEP:
+      default:
+        break;
+    }
+    if (changed)
+      goto restart;
+  }
+  g_mutex_unlock (&priv->lock);
+
+  if (func)
+    g_hash_table_unref (visited);
+
+  return result;
+}
+
+typedef struct
+{
+  GSource source;
+  GstRTSPSessionPool *pool;
+  gint timeout;
+} GstPoolSource;
+
+static void
+collect_timeout (gchar * sessionid, GstRTSPSession * sess, GstPoolSource * psrc)
+{
+  gint timeout;
+  GTimeVal now;
+
+  g_get_current_time (&now);
+
+  timeout = gst_rtsp_session_next_timeout (sess, &now);
+  GST_INFO ("%p: next timeout: %d", sess, timeout);
+  if (psrc->timeout == -1 || timeout < psrc->timeout)
+    psrc->timeout = timeout;
+}
+
+static gboolean
+gst_pool_source_prepare (GSource * source, gint * timeout)
+{
+  GstRTSPSessionPoolPrivate *priv;
+  GstPoolSource *psrc;
+  gboolean result;
+
+  psrc = (GstPoolSource *) source;
+  psrc->timeout = -1;
+  priv = psrc->pool->priv;
+
+  g_mutex_lock (&priv->lock);
+  g_hash_table_foreach (priv->sessions, (GHFunc) collect_timeout, psrc);
+  g_mutex_unlock (&priv->lock);
+
+  if (timeout)
+    *timeout = psrc->timeout;
+
+  result = psrc->timeout == 0;
+
+  GST_INFO ("prepare %d, %d", psrc->timeout, result);
+
+  return result;
+}
+
+static gboolean
+gst_pool_source_check (GSource * source)
+{
+  GST_INFO ("check");
+
+  return gst_pool_source_prepare (source, NULL);
+}
+
+static gboolean
+gst_pool_source_dispatch (GSource * source, GSourceFunc callback,
+    gpointer user_data)
+{
+  gboolean res;
+  GstPoolSource *psrc = (GstPoolSource *) source;
+  GstRTSPSessionPoolFunc func = (GstRTSPSessionPoolFunc) callback;
+
+  GST_INFO ("dispatch");
+
+  if (func)
+    res = func (psrc->pool, user_data);
+  else
+    res = FALSE;
+
+  return res;
+}
+
+static void
+gst_pool_source_finalize (GSource * source)
+{
+  GstPoolSource *psrc = (GstPoolSource *) source;
+
+  GST_INFO ("finalize %p", psrc);
+
+  g_object_unref (psrc->pool);
+  psrc->pool = NULL;
+}
+
+static GSourceFuncs gst_pool_source_funcs = {
+  gst_pool_source_prepare,
+  gst_pool_source_check,
+  gst_pool_source_dispatch,
+  gst_pool_source_finalize
+};
+
+/**
+ * gst_rtsp_session_pool_create_watch:
+ * @pool: a #GstRTSPSessionPool
+ *
+ * Create a #GSource that will be dispatched when the session should be cleaned
+ * up.
+ *
+ * Returns: (transfer full): a #GSource
+ */
+GSource *
+gst_rtsp_session_pool_create_watch (GstRTSPSessionPool * pool)
+{
+  GstPoolSource *source;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION_POOL (pool), NULL);
+
+  source = (GstPoolSource *) g_source_new (&gst_pool_source_funcs,
+      sizeof (GstPoolSource));
+  source->pool = g_object_ref (pool);
+
+  return (GSource *) source;
+}
diff --git a/gst/rtsp-server/rtsp-session-pool.h b/gst/rtsp-server/rtsp-session-pool.h
new file mode 100644
index 0000000..e2ba951
--- /dev/null
+++ b/gst/rtsp-server/rtsp-session-pool.h
@@ -0,0 +1,143 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#ifndef __GST_RTSP_SESSION_POOL_H__
+#define __GST_RTSP_SESSION_POOL_H__
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPSessionPool GstRTSPSessionPool;
+typedef struct _GstRTSPSessionPoolClass GstRTSPSessionPoolClass;
+typedef struct _GstRTSPSessionPoolPrivate GstRTSPSessionPoolPrivate;
+
+#include "rtsp-session.h"
+
+#define GST_TYPE_RTSP_SESSION_POOL              (gst_rtsp_session_pool_get_type ())
+#define GST_IS_RTSP_SESSION_POOL(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_SESSION_POOL))
+#define GST_IS_RTSP_SESSION_POOL_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_SESSION_POOL))
+#define GST_RTSP_SESSION_POOL_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_SESSION_POOL, GstRTSPSessionPoolClass))
+#define GST_RTSP_SESSION_POOL(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_SESSION_POOL, GstRTSPSessionPool))
+#define GST_RTSP_SESSION_POOL_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_SESSION_POOL, GstRTSPSessionPoolClass))
+#define GST_RTSP_SESSION_POOL_CAST(obj)         ((GstRTSPSessionPool*)(obj))
+#define GST_RTSP_SESSION_POOL_CLASS_CAST(klass) ((GstRTSPSessionPoolClass*)(klass))
+
+/**
+ * GstRTSPSessionPool:
+ *
+ * An object that keeps track of the active sessions. This object is usually
+ * attached to a #GstRTSPServer object to manage the sessions in that server.
+ */
+struct _GstRTSPSessionPool {
+  GObject       parent;
+
+  /*< private >*/
+  GstRTSPSessionPoolPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPSessionPoolClass:
+ * @create_session_id: create a new random session id. Subclasses can create
+ *    custom session ids and should not check if the session exists.
+ * @create_session: make a new session object.
+ * @session_removed: a session was removed from the pool
+ */
+struct _GstRTSPSessionPoolClass {
+  GObjectClass  parent_class;
+
+  gchar *          (*create_session_id)   (GstRTSPSessionPool *pool);
+  GstRTSPSession * (*create_session)      (GstRTSPSessionPool *pool, const gchar *id);
+
+  /* signals */
+  void             (*session_removed)     (GstRTSPSessionPool *pool,
+                                           GstRTSPSession *session);
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING_LARGE - 1];
+};
+
+/**
+ * GstRTSPSessionPoolFunc:
+ * @pool: a #GstRTSPSessionPool object
+ * @user_data: user data that has been given when registering the handler
+ *
+ * The function that will be called from the GSource watch on the session pool.
+ *
+ * The function will be called when the pool must be cleaned up because one or
+ * more sessions timed out.
+ *
+ * Returns: %FALSE if the source should be removed.
+ */
+typedef gboolean (*GstRTSPSessionPoolFunc)  (GstRTSPSessionPool *pool, gpointer user_data);
+
+/**
+ * GstRTSPSessionPoolFilterFunc:
+ * @pool: a #GstRTSPSessionPool object
+ * @session: a #GstRTSPSession in @pool
+ * @user_data: user data that has been given to gst_rtsp_session_pool_filter()
+ *
+ * This function will be called by the gst_rtsp_session_pool_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @session will be removed
+ * from @pool.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @session untouched in
+ * @pool.
+ *
+ * A value of GST_RTSP_FILTER_REF will add @session to the result #GList of
+ * gst_rtsp_session_pool_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPSessionPoolFilterFunc)  (GstRTSPSessionPool *pool,
+                                                              GstRTSPSession *session,
+                                                              gpointer user_data);
+
+
+GType                 gst_rtsp_session_pool_get_type          (void);
+
+/* creating a session pool */
+GstRTSPSessionPool *  gst_rtsp_session_pool_new               (void);
+
+/* counting sessions */
+void                  gst_rtsp_session_pool_set_max_sessions  (GstRTSPSessionPool *pool, guint max);
+guint                 gst_rtsp_session_pool_get_max_sessions  (GstRTSPSessionPool *pool);
+
+guint                 gst_rtsp_session_pool_get_n_sessions    (GstRTSPSessionPool *pool);
+
+/* managing sessions */
+GstRTSPSession *      gst_rtsp_session_pool_create            (GstRTSPSessionPool *pool);
+GstRTSPSession *      gst_rtsp_session_pool_find              (GstRTSPSessionPool *pool,
+                                                               const gchar *sessionid);
+gboolean              gst_rtsp_session_pool_remove            (GstRTSPSessionPool *pool,
+                                                               GstRTSPSession *sess);
+
+/* perform session maintenance */
+GList *               gst_rtsp_session_pool_filter            (GstRTSPSessionPool *pool,
+                                                               GstRTSPSessionPoolFilterFunc func,
+                                                               gpointer user_data);
+guint                 gst_rtsp_session_pool_cleanup           (GstRTSPSessionPool *pool);
+GSource *             gst_rtsp_session_pool_create_watch      (GstRTSPSessionPool *pool);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SESSION_POOL_H__ */
diff --git a/gst/rtsp-server/rtsp-session.c b/gst/rtsp-server/rtsp-session.c
new file mode 100644
index 0000000..372746a
--- /dev/null
+++ b/gst/rtsp-server/rtsp-session.c
@@ -0,0 +1,649 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-session
+ * @short_description: An object to manage media
+ * @see_also: #GstRTSPSessionPool, #GstRTSPSessionMedia, #GstRTSPMedia
+ *
+ * The #GstRTSPSession is identified by an id, unique in the
+ * #GstRTSPSessionPool that created the session and manages media and its
+ * configuration.
+ *
+ * A #GstRTSPSession has a timeout that can be retrieved with
+ * gst_rtsp_session_get_timeout(). You can check if the sessions is expired with
+ * gst_rtsp_session_is_expired(). gst_rtsp_session_touch() will reset the
+ * expiration counter of the session.
+ *
+ * When a client configures a media with SETUP, a session will be created to
+ * keep track of the configuration of that media. With
+ * gst_rtsp_session_manage_media(), the media is added to the managed media
+ * in the session. With gst_rtsp_session_release_media() the media can be
+ * released again from the session. Managed media is identified in the sessions
+ * with a url. Use gst_rtsp_session_get_media() to get the media that matches
+ * (part of) the given url.
+ *
+ * The media in a session can be iterated with gst_rtsp_session_filter().
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include 
+
+#include "rtsp-session.h"
+
+#define GST_RTSP_SESSION_GET_PRIVATE(obj)  \
+       (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_SESSION, GstRTSPSessionPrivate))
+
+struct _GstRTSPSessionPrivate
+{
+  GMutex lock;                  /* protects everything but sessionid and create_time */
+  gchar *sessionid;
+
+  guint timeout;
+  gboolean timeout_always_visible;
+  GTimeVal create_time;         /* immutable */
+  GTimeVal last_access;
+  gint expire_count;
+
+  GList *medias;
+  guint medias_cookie;
+};
+
+#undef DEBUG
+
+#define DEFAULT_TIMEOUT	       60
+#define DEFAULT_ALWAYS_VISIBLE  FALSE
+
+enum
+{
+  PROP_0,
+  PROP_SESSIONID,
+  PROP_TIMEOUT,
+  PROP_TIMEOUT_ALWAYS_VISIBLE,
+  PROP_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_session_debug);
+#define GST_CAT_DEFAULT rtsp_session_debug
+
+static void gst_rtsp_session_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec);
+static void gst_rtsp_session_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_session_finalize (GObject * obj);
+
+G_DEFINE_TYPE (GstRTSPSession, gst_rtsp_session, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_session_class_init (GstRTSPSessionClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPSessionPrivate));
+
+  gobject_class = G_OBJECT_CLASS (klass);
+
+  gobject_class->get_property = gst_rtsp_session_get_property;
+  gobject_class->set_property = gst_rtsp_session_set_property;
+  gobject_class->finalize = gst_rtsp_session_finalize;
+
+  g_object_class_install_property (gobject_class, PROP_SESSIONID,
+      g_param_spec_string ("sessionid", "Sessionid", "the session id",
+          NULL, G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY |
+          G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_TIMEOUT,
+      g_param_spec_uint ("timeout", "timeout",
+          "the timeout of the session (0 = never)", 0, G_MAXUINT,
+          DEFAULT_TIMEOUT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_TIMEOUT_ALWAYS_VISIBLE,
+      g_param_spec_boolean ("timeout-always-visible", "Timeout Always Visible ",
+          "timeout always visible in header",
+          DEFAULT_ALWAYS_VISIBLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_session_debug, "rtspsession", 0,
+      "GstRTSPSession");
+}
+
+static void
+gst_rtsp_session_init (GstRTSPSession * session)
+{
+  GstRTSPSessionPrivate *priv = GST_RTSP_SESSION_GET_PRIVATE (session);
+
+  session->priv = priv;
+
+  GST_INFO ("init session %p", session);
+
+  g_mutex_init (&priv->lock);
+  priv->timeout = DEFAULT_TIMEOUT;
+  g_get_current_time (&priv->create_time);
+  gst_rtsp_session_touch (session);
+}
+
+static void
+gst_rtsp_session_finalize (GObject * obj)
+{
+  GstRTSPSession *session;
+  GstRTSPSessionPrivate *priv;
+
+  session = GST_RTSP_SESSION (obj);
+  priv = session->priv;
+
+  GST_INFO ("finalize session %p", session);
+
+  /* free all media */
+  g_list_free_full (priv->medias, g_object_unref);
+
+  /* free session id */
+  g_free (priv->sessionid);
+  g_mutex_clear (&priv->lock);
+
+  G_OBJECT_CLASS (gst_rtsp_session_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_session_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRTSPSession *session = GST_RTSP_SESSION (object);
+  GstRTSPSessionPrivate *priv = session->priv;
+
+  switch (propid) {
+    case PROP_SESSIONID:
+      g_value_set_string (value, priv->sessionid);
+      break;
+    case PROP_TIMEOUT:
+      g_value_set_uint (value, gst_rtsp_session_get_timeout (session));
+      break;
+    case PROP_TIMEOUT_ALWAYS_VISIBLE:
+      g_value_set_boolean (value, priv->timeout_always_visible);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+static void
+gst_rtsp_session_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRTSPSession *session = GST_RTSP_SESSION (object);
+  GstRTSPSessionPrivate *priv = session->priv;
+
+  switch (propid) {
+    case PROP_SESSIONID:
+      g_free (priv->sessionid);
+      priv->sessionid = g_value_dup_string (value);
+      break;
+    case PROP_TIMEOUT:
+      gst_rtsp_session_set_timeout (session, g_value_get_uint (value));
+      break;
+    case PROP_TIMEOUT_ALWAYS_VISIBLE:
+      g_mutex_lock (&priv->lock);
+      priv->timeout_always_visible = g_value_get_boolean (value);
+      g_mutex_unlock (&priv->lock);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+/**
+ * gst_rtsp_session_manage_media:
+ * @sess: a #GstRTSPSession
+ * @path: the path for the media
+ * @media: (transfer full): a #GstRTSPMedia
+ *
+ * Manage the media object @obj in @sess. @path will be used to retrieve this
+ * media from the session with gst_rtsp_session_get_media().
+ *
+ * Ownership is taken from @media.
+ *
+ * Returns: (transfer none): a new @GstRTSPSessionMedia object.
+ */
+GstRTSPSessionMedia *
+gst_rtsp_session_manage_media (GstRTSPSession * sess, const gchar * path,
+    GstRTSPMedia * media)
+{
+  GstRTSPSessionPrivate *priv;
+  GstRTSPSessionMedia *result;
+  GstRTSPMediaStatus status;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), NULL);
+  g_return_val_if_fail (path != NULL, NULL);
+  g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
+  status = gst_rtsp_media_get_status (media);
+  g_return_val_if_fail (status == GST_RTSP_MEDIA_STATUS_PREPARED || status ==
+      GST_RTSP_MEDIA_STATUS_SUSPENDED, NULL);
+
+  priv = sess->priv;
+
+  result = gst_rtsp_session_media_new (path, media);
+
+  g_mutex_lock (&priv->lock);
+  priv->medias = g_list_prepend (priv->medias, result);
+  priv->medias_cookie++;
+  g_mutex_unlock (&priv->lock);
+
+  GST_INFO ("manage new media %p in session %p", media, result);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_session_release_media:
+ * @sess: a #GstRTSPSession
+ * @media: (transfer none): a #GstRTSPMedia
+ *
+ * Release the managed @media in @sess, freeing the memory allocated by it.
+ *
+ * Returns: %TRUE if there are more media session left in @sess.
+ */
+gboolean
+gst_rtsp_session_release_media (GstRTSPSession * sess,
+    GstRTSPSessionMedia * media)
+{
+  GstRTSPSessionPrivate *priv;
+  GList *find;
+  gboolean more;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), FALSE);
+  g_return_val_if_fail (media != NULL, FALSE);
+
+  priv = sess->priv;
+
+  g_mutex_lock (&priv->lock);
+  find = g_list_find (priv->medias, media);
+  if (find) {
+    priv->medias = g_list_delete_link (priv->medias, find);
+    priv->medias_cookie++;
+  }
+  more = (priv->medias != NULL);
+  g_mutex_unlock (&priv->lock);
+
+  if (find)
+    g_object_unref (media);
+
+  return more;
+}
+
+/**
+ * gst_rtsp_session_get_media:
+ * @sess: a #GstRTSPSession
+ * @path: the path for the media
+ * @matched: (out): the amount of matched characters
+ *
+ * Get the session media for @path. @matched will contain the number of matched
+ * characters of @path.
+ *
+ * Returns: (transfer none): the configuration for @path in @sess.
+ */
+GstRTSPSessionMedia *
+gst_rtsp_session_get_media (GstRTSPSession * sess, const gchar * path,
+    gint * matched)
+{
+  GstRTSPSessionPrivate *priv;
+  GstRTSPSessionMedia *result;
+  GList *walk;
+  gint best;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), NULL);
+  g_return_val_if_fail (path != NULL, NULL);
+
+  priv = sess->priv;
+  result = NULL;
+  best = 0;
+
+  g_mutex_lock (&priv->lock);
+  for (walk = priv->medias; walk; walk = g_list_next (walk)) {
+    GstRTSPSessionMedia *test;
+
+    test = (GstRTSPSessionMedia *) walk->data;
+
+    /* find largest match */
+    if (gst_rtsp_session_media_matches (test, path, matched)) {
+      if (best < *matched) {
+        result = test;
+        best = *matched;
+      }
+    }
+  }
+  g_mutex_unlock (&priv->lock);
+
+  *matched = best;
+
+  return result;
+}
+
+/**
+ * gst_rtsp_session_filter:
+ * @sess: a #GstRTSPSession
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: (closure): user data passed to @func
+ *
+ * Call @func for each media in @sess. The result value of @func determines
+ * what happens to the media. @func will be called with @sess
+ * locked so no further actions on @sess can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the media will be removed from
+ * @sess.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the media will remain in @sess.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the media will remain in @sess but
+ * will also be added with an additional ref to the result #GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for all media.
+ *
+ * Returns: (element-type GstRTSPSessionMedia) (transfer full): a GList with all
+ * media for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the #GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_session_filter (GstRTSPSession * sess,
+    GstRTSPSessionFilterFunc func, gpointer user_data)
+{
+  GstRTSPSessionPrivate *priv;
+  GList *result, *walk, *next;
+  GHashTable *visited;
+  guint cookie;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION (sess), NULL);
+
+  priv = sess->priv;
+
+  result = NULL;
+  if (func)
+    visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+  g_mutex_lock (&priv->lock);
+restart:
+  cookie = priv->medias_cookie;
+  for (walk = priv->medias; walk; walk = next) {
+    GstRTSPSessionMedia *media = walk->data;
+    GstRTSPFilterResult res;
+    gboolean changed;
+
+    next = g_list_next (walk);
+
+    if (func) {
+      /* only visit each media once */
+      if (g_hash_table_contains (visited, media))
+        continue;
+
+      g_hash_table_add (visited, g_object_ref (media));
+      g_mutex_unlock (&priv->lock);
+
+      res = func (sess, media, user_data);
+
+      g_mutex_lock (&priv->lock);
+    } else
+      res = GST_RTSP_FILTER_REF;
+
+    changed = (cookie != priv->medias_cookie);
+
+    switch (res) {
+      case GST_RTSP_FILTER_REMOVE:
+        if (changed)
+          priv->medias = g_list_remove (priv->medias, media);
+        else
+          priv->medias = g_list_delete_link (priv->medias, walk);
+        cookie = ++priv->medias_cookie;
+        g_object_unref (media);
+        break;
+      case GST_RTSP_FILTER_REF:
+        result = g_list_prepend (result, g_object_ref (media));
+        break;
+      case GST_RTSP_FILTER_KEEP:
+      default:
+        break;
+    }
+    if (changed)
+      goto restart;
+  }
+  g_mutex_unlock (&priv->lock);
+
+  if (func)
+    g_hash_table_unref (visited);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_session_new:
+ * @sessionid: a session id
+ *
+ * Create a new #GstRTSPSession instance with @sessionid.
+ *
+ * Returns: (transfer full): a new #GstRTSPSession
+ */
+GstRTSPSession *
+gst_rtsp_session_new (const gchar * sessionid)
+{
+  GstRTSPSession *result;
+
+  g_return_val_if_fail (sessionid != NULL, NULL);
+
+  result = g_object_new (GST_TYPE_RTSP_SESSION, "sessionid", sessionid, NULL);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_session_get_sessionid:
+ * @session: a #GstRTSPSession
+ *
+ * Get the sessionid of @session.
+ *
+ * Returns: (transfer none): the sessionid of @session. The value remains valid
+ * as long as @session is alive.
+ */
+const gchar *
+gst_rtsp_session_get_sessionid (GstRTSPSession * session)
+{
+  g_return_val_if_fail (GST_IS_RTSP_SESSION (session), NULL);
+
+  return session->priv->sessionid;
+}
+
+/**
+ * gst_rtsp_session_get_header:
+ * @session: a #GstRTSPSession
+ *
+ * Get the string that can be placed in the Session header field.
+ *
+ * Returns: (transfer full): the Session header of @session. g_free() after usage.
+ */
+gchar *
+gst_rtsp_session_get_header (GstRTSPSession * session)
+{
+  GstRTSPSessionPrivate *priv;
+  gchar *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION (session), NULL);
+
+  priv = session->priv;
+
+
+  g_mutex_lock (&priv->lock);
+  if (priv->timeout_always_visible || priv->timeout != 60)
+    result = g_strdup_printf ("%s; timeout=%d", priv->sessionid, priv->timeout);
+  else
+    result = g_strdup (priv->sessionid);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_session_set_timeout:
+ * @session: a #GstRTSPSession
+ * @timeout: the new timeout
+ *
+ * Configure @session for a timeout of @timeout seconds. The session will be
+ * cleaned up when there is no activity for @timeout seconds.
+ */
+void
+gst_rtsp_session_set_timeout (GstRTSPSession * session, guint timeout)
+{
+  GstRTSPSessionPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_SESSION (session));
+
+  priv = session->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->timeout = timeout;
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_session_get_timeout:
+ * @session: a #GstRTSPSession
+ *
+ * Get the timeout value of @session.
+ *
+ * Returns: the timeout of @session in seconds.
+ */
+guint
+gst_rtsp_session_get_timeout (GstRTSPSession * session)
+{
+  GstRTSPSessionPrivate *priv;
+  guint res;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION (session), 0);
+
+  priv = session->priv;
+
+  g_mutex_lock (&priv->lock);
+  res = priv->timeout;
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_session_touch:
+ * @session: a #GstRTSPSession
+ *
+ * Update the last_access time of the session to the current time.
+ */
+void
+gst_rtsp_session_touch (GstRTSPSession * session)
+{
+  GstRTSPSessionPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_SESSION (session));
+
+  priv = session->priv;
+
+  g_mutex_lock (&priv->lock);
+  g_get_current_time (&priv->last_access);
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_session_prevent_expire:
+ * @session: a #GstRTSPSession
+ *
+ * Prevent @session from expiring.
+ */
+void
+gst_rtsp_session_prevent_expire (GstRTSPSession * session)
+{
+  g_return_if_fail (GST_IS_RTSP_SESSION (session));
+
+  g_atomic_int_add (&session->priv->expire_count, 1);
+}
+
+/**
+ * gst_rtsp_session_allow_expire:
+ * @session: a #GstRTSPSession
+ *
+ * Allow @session to expire. This method must be called an equal
+ * amount of time as gst_rtsp_session_prevent_expire().
+ */
+void
+gst_rtsp_session_allow_expire (GstRTSPSession * session)
+{
+  g_atomic_int_add (&session->priv->expire_count, -1);
+}
+
+/**
+ * gst_rtsp_session_next_timeout:
+ * @session: a #GstRTSPSession
+ * @now: (transfer none): the current system time
+ *
+ * Get the amount of milliseconds till the session will expire.
+ *
+ * Returns: the amount of milliseconds since the session will time out.
+ */
+gint
+gst_rtsp_session_next_timeout (GstRTSPSession * session, GTimeVal * now)
+{
+  GstRTSPSessionPrivate *priv;
+  gint res;
+  GstClockTime last_access, now_ns;
+
+  g_return_val_if_fail (GST_IS_RTSP_SESSION (session), -1);
+  g_return_val_if_fail (now != NULL, -1);
+
+  priv = session->priv;
+
+  g_mutex_lock (&priv->lock);
+  if (g_atomic_int_get (&priv->expire_count) != 0) {
+    /* touch session when the expire count is not 0 */
+    g_get_current_time (&priv->last_access);
+  }
+
+  last_access = GST_TIMEVAL_TO_TIME (priv->last_access);
+  /* add timeout allow for 5 seconds of extra time */
+  last_access += priv->timeout * GST_SECOND + (5 * GST_SECOND);
+  g_mutex_unlock (&priv->lock);
+
+  now_ns = GST_TIMEVAL_TO_TIME (*now);
+
+  if (last_access > now_ns)
+    res = GST_TIME_AS_MSECONDS (last_access - now_ns);
+  else
+    res = 0;
+
+  return res;
+}
+
+/**
+ * gst_rtsp_session_is_expired:
+ * @session: a #GstRTSPSession
+ * @now: (transfer none): the current system time
+ *
+ * Check if @session timeout out.
+ *
+ * Returns: %TRUE if @session timed out
+ */
+gboolean
+gst_rtsp_session_is_expired (GstRTSPSession * session, GTimeVal * now)
+{
+  gboolean res;
+
+  res = (gst_rtsp_session_next_timeout (session, now) == 0);
+
+  return res;
+}
diff --git a/gst/rtsp-server/rtsp-session.h b/gst/rtsp-server/rtsp-session.h
new file mode 100644
index 0000000..d9c0f18
--- /dev/null
+++ b/gst/rtsp-server/rtsp-session.h
@@ -0,0 +1,143 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+#ifndef __GST_RTSP_SESSION_H__
+#define __GST_RTSP_SESSION_H__
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_SESSION              (gst_rtsp_session_get_type ())
+#define GST_IS_RTSP_SESSION(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_SESSION))
+#define GST_IS_RTSP_SESSION_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_SESSION))
+#define GST_RTSP_SESSION_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_SESSION, GstRTSPSessionClass))
+#define GST_RTSP_SESSION(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_SESSION, GstRTSPSession))
+#define GST_RTSP_SESSION_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_SESSION, GstRTSPSessionClass))
+#define GST_RTSP_SESSION_CAST(obj)         ((GstRTSPSession*)(obj))
+#define GST_RTSP_SESSION_CLASS_CAST(klass) ((GstRTSPSessionClass*)(klass))
+
+typedef struct _GstRTSPSession GstRTSPSession;
+typedef struct _GstRTSPSessionClass GstRTSPSessionClass;
+typedef struct _GstRTSPSessionPrivate GstRTSPSessionPrivate;
+
+/**
+ * GstRTSPFilterResult:
+ * @GST_RTSP_FILTER_REMOVE: Remove session
+ * @GST_RTSP_FILTER_KEEP: Keep session in the pool
+ * @GST_RTSP_FILTER_REF: Ref session in the result list
+ *
+ * Possible return values for gst_rtsp_session_pool_filter().
+ */
+typedef enum
+{
+  GST_RTSP_FILTER_REMOVE,
+  GST_RTSP_FILTER_KEEP,
+  GST_RTSP_FILTER_REF,
+} GstRTSPFilterResult;
+
+#include "rtsp-media.h"
+#include "rtsp-session-media.h"
+
+/**
+ * GstRTSPSession:
+ *
+ * Session information kept by the server for a specific client.
+ * One client session, identified with a session id, can handle multiple medias
+ * identified with the url of a media.
+ */
+struct _GstRTSPSession {
+  GObject       parent;
+
+  /*< private >*/
+  GstRTSPSessionPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+struct _GstRTSPSessionClass {
+  GObjectClass  parent_class;
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+GType                  gst_rtsp_session_get_type             (void);
+
+/* create a new session */
+GstRTSPSession *       gst_rtsp_session_new                  (const gchar *sessionid);
+
+const gchar *          gst_rtsp_session_get_sessionid        (GstRTSPSession *session);
+
+gchar *                gst_rtsp_session_get_header           (GstRTSPSession *session);
+
+void                   gst_rtsp_session_set_timeout          (GstRTSPSession *session, guint timeout);
+guint                  gst_rtsp_session_get_timeout          (GstRTSPSession *session);
+
+/* session timeout stuff */
+void                   gst_rtsp_session_touch                (GstRTSPSession *session);
+void                   gst_rtsp_session_prevent_expire       (GstRTSPSession *session);
+void                   gst_rtsp_session_allow_expire         (GstRTSPSession *session);
+gint                   gst_rtsp_session_next_timeout         (GstRTSPSession *session, GTimeVal *now);
+gboolean               gst_rtsp_session_is_expired           (GstRTSPSession *session, GTimeVal *now);
+
+/* handle media in a session */
+GstRTSPSessionMedia *  gst_rtsp_session_manage_media         (GstRTSPSession *sess,
+                                                              const gchar *path,
+                                                              GstRTSPMedia *media);
+gboolean               gst_rtsp_session_release_media        (GstRTSPSession *sess,
+                                                              GstRTSPSessionMedia *media);
+/* get media in a session */
+GstRTSPSessionMedia *  gst_rtsp_session_get_media            (GstRTSPSession *sess,
+                                                              const gchar *path,
+                                                              gint * matched);
+
+/**
+ * GstRTSPSessionFilterFunc:
+ * @sess: a #GstRTSPSession object
+ * @media: a #GstRTSPSessionMedia in @sess
+ * @user_data: user data that has been given to gst_rtsp_session_filter()
+ *
+ * This function will be called by the gst_rtsp_session_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @media will be removed
+ * from @sess.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @media untouched in
+ * @sess.
+ *
+ * A value of GST_RTSP_FILTER_REF will add @media to the result #GList of
+ * gst_rtsp_session_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPSessionFilterFunc)  (GstRTSPSession *sess,
+                                                          GstRTSPSessionMedia *media,
+                                                          gpointer user_data);
+
+GList *                gst_rtsp_session_filter           (GstRTSPSession *sess,
+                                                          GstRTSPSessionFilterFunc func,
+                                                          gpointer user_data);
+
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_SESSION_H__ */
diff --git a/gst/rtsp-server/rtsp-stream-transport.c b/gst/rtsp-server/rtsp-stream-transport.c
new file mode 100644
index 0000000..a99295b
--- /dev/null
+++ b/gst/rtsp-server/rtsp-stream-transport.c
@@ -0,0 +1,503 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-stream-transport
+ * @short_description: A media stream transport configuration
+ * @see_also: #GstRTSPStream, #GstRTSPSessionMedia
+ *
+ * The #GstRTSPStreamTransport configures the transport used by a
+ * #GstRTSPStream. It is usually manages by a #GstRTSPSessionMedia object.
+ *
+ * With gst_rtsp_stream_transport_set_callbacks(), callbacks can be configured
+ * to handle the RTP and RTCP packets from the stream, for example when they
+ * need to be sent over TCP.
+ *
+ * With  gst_rtsp_stream_transport_set_active() the transports are added and
+ * removed from the stream.
+ *
+ * A #GstRTSPStream will call gst_rtsp_stream_transport_keep_alive() when RTCP
+ * is received from the client. It will also call
+ * gst_rtsp_stream_transport_set_timed_out() when a receiver has timed out.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
+
+#include 
+#include 
+
+#include "rtsp-stream-transport.h"
+
+#define GST_RTSP_STREAM_TRANSPORT_GET_PRIVATE(obj)  \
+       (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportPrivate))
+
+struct _GstRTSPStreamTransportPrivate
+{
+  GstRTSPStream *stream;
+
+  GstRTSPSendFunc send_rtp;
+  GstRTSPSendFunc send_rtcp;
+  gpointer user_data;
+  GDestroyNotify notify;
+
+  GstRTSPKeepAliveFunc keep_alive;
+  gpointer ka_user_data;
+  GDestroyNotify ka_notify;
+  gboolean active;
+  gboolean timed_out;
+
+  GstRTSPTransport *transport;
+  GstRTSPUrl *url;
+
+  GObject *rtpsource;
+};
+
+enum
+{
+  PROP_0,
+  PROP_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_stream_transport_debug);
+#define GST_CAT_DEFAULT rtsp_stream_transport_debug
+
+static void gst_rtsp_stream_transport_finalize (GObject * obj);
+
+G_DEFINE_TYPE (GstRTSPStreamTransport, gst_rtsp_stream_transport,
+    G_TYPE_OBJECT);
+
+static void
+gst_rtsp_stream_transport_class_init (GstRTSPStreamTransportClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPStreamTransportPrivate));
+
+  gobject_class = G_OBJECT_CLASS (klass);
+
+  gobject_class->finalize = gst_rtsp_stream_transport_finalize;
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_stream_transport_debug, "rtspmediatransport",
+      0, "GstRTSPStreamTransport");
+}
+
+static void
+gst_rtsp_stream_transport_init (GstRTSPStreamTransport * trans)
+{
+  GstRTSPStreamTransportPrivate *priv =
+      GST_RTSP_STREAM_TRANSPORT_GET_PRIVATE (trans);
+
+  trans->priv = priv;
+}
+
+static void
+gst_rtsp_stream_transport_finalize (GObject * obj)
+{
+  GstRTSPStreamTransportPrivate *priv;
+  GstRTSPStreamTransport *trans;
+
+  trans = GST_RTSP_STREAM_TRANSPORT (obj);
+  priv = trans->priv;
+
+  /* remove callbacks now */
+  gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
+  gst_rtsp_stream_transport_set_keepalive (trans, NULL, NULL, NULL);
+
+  if (priv->transport)
+    gst_rtsp_transport_free (priv->transport);
+
+  if (priv->url)
+    gst_rtsp_url_free (priv->url);
+
+  G_OBJECT_CLASS (gst_rtsp_stream_transport_parent_class)->finalize (obj);
+}
+
+/**
+ * gst_rtsp_stream_transport_new:
+ * @stream: a #GstRTSPStream
+ * @tr: (transfer full): a GstRTSPTransport
+ *
+ * Create a new #GstRTSPStreamTransport that can be used to manage
+ * @stream with transport @tr.
+ *
+ * Returns: (transfer full): a new #GstRTSPStreamTransport
+ */
+GstRTSPStreamTransport *
+gst_rtsp_stream_transport_new (GstRTSPStream * stream, GstRTSPTransport * tr)
+{
+  GstRTSPStreamTransportPrivate *priv;
+  GstRTSPStreamTransport *trans;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+  g_return_val_if_fail (tr != NULL, NULL);
+
+  trans = g_object_new (GST_TYPE_RTSP_STREAM_TRANSPORT, NULL);
+  priv = trans->priv;
+  priv->stream = stream;
+  priv->transport = tr;
+
+  return trans;
+}
+
+/**
+ * gst_rtsp_stream_transport_get_stream:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Get the #GstRTSPStream used when constructing @trans.
+ *
+ * Returns: (transfer none): the stream used when constructing @trans.
+ */
+GstRTSPStream *
+gst_rtsp_stream_transport_get_stream (GstRTSPStreamTransport * trans)
+{
+  g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+  return trans->priv->stream;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_callbacks:
+ * @trans: a #GstRTSPStreamTransport
+ * @send_rtp: (scope notified): a callback called when RTP should be sent
+ * @send_rtcp: (scope notified): a callback called when RTCP should be sent
+ * @user_data: (closure): user data passed to callbacks
+ * @notify: (allow-none): called with the user_data when no longer needed.
+ *
+ * Install callbacks that will be called when data for a stream should be sent
+ * to a client. This is usually used when sending RTP/RTCP over TCP.
+ */
+void
+gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport * trans,
+    GstRTSPSendFunc send_rtp, GstRTSPSendFunc send_rtcp,
+    gpointer user_data, GDestroyNotify notify)
+{
+  GstRTSPStreamTransportPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+  priv = trans->priv;
+
+  priv->send_rtp = send_rtp;
+  priv->send_rtcp = send_rtcp;
+  if (priv->notify)
+    priv->notify (priv->user_data);
+  priv->user_data = user_data;
+  priv->notify = notify;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_keepalive:
+ * @trans: a #GstRTSPStreamTransport
+ * @keep_alive: (scope notified): a callback called when the receiver is active
+ * @user_data: (closure): user data passed to callback
+ * @notify: (allow-none): called with the user_data when no longer needed.
+ *
+ * Install callbacks that will be called when RTCP packets are received from the
+ * receiver of @trans.
+ */
+void
+gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport * trans,
+    GstRTSPKeepAliveFunc keep_alive, gpointer user_data, GDestroyNotify notify)
+{
+  GstRTSPStreamTransportPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+  priv = trans->priv;
+
+  priv->keep_alive = keep_alive;
+  if (priv->ka_notify)
+    priv->ka_notify (priv->ka_user_data);
+  priv->ka_user_data = user_data;
+  priv->ka_notify = notify;
+}
+
+
+/**
+ * gst_rtsp_stream_transport_set_transport:
+ * @trans: a #GstRTSPStreamTransport
+ * @tr: (transfer full): a client #GstRTSPTransport
+ *
+ * Set @tr as the client transport. This function takes ownership of the
+ * passed @tr.
+ */
+void
+gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans,
+    GstRTSPTransport * tr)
+{
+  GstRTSPStreamTransportPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+  g_return_if_fail (tr != NULL);
+
+  priv = trans->priv;
+
+  /* keep track of the transports in the stream. */
+  if (priv->transport)
+    gst_rtsp_transport_free (priv->transport);
+  priv->transport = tr;
+}
+
+/**
+ * gst_rtsp_stream_transport_get_transport:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Get the transport configured in @trans.
+ *
+ * Returns: (transfer none): the transport configured in @trans. It remains
+ * valid for as long as @trans is valid.
+ */
+const GstRTSPTransport *
+gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport * trans)
+{
+  g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+  return trans->priv->transport;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_url:
+ * @trans: a #GstRTSPStreamTransport
+ * @url: (transfer none): a client #GstRTSPUrl
+ *
+ * Set @url as the client url.
+ */
+void
+gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport * trans,
+    const GstRTSPUrl * url)
+{
+  GstRTSPStreamTransportPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+  priv = trans->priv;
+
+  /* keep track of the transports in the stream. */
+  if (priv->url)
+    gst_rtsp_url_free (priv->url);
+  priv->url = (url ? gst_rtsp_url_copy (url) : NULL);
+}
+
+/**
+ * gst_rtsp_stream_transport_get_url:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Get the url configured in @trans.
+ *
+ * Returns: (transfer none): the url configured in @trans. It remains
+ * valid for as long as @trans is valid.
+ */
+const GstRTSPUrl *
+gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport * trans)
+{
+  g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+  return trans->priv->url;
+}
+
+ /**
+ * gst_rtsp_stream_transport_get_rtpinfo:
+ * @trans: a #GstRTSPStreamTransport
+ * @start_time: a star time
+ *
+ * Get the RTP-Info string for @trans and @start_time.
+ *
+ * Returns: (transfer full) (nullable): the RTPInfo string for @trans
+ * and @start_time or %NULL when the RTP-Info could not be
+ * determined. g_free() after usage.
+ */
+gchar *
+gst_rtsp_stream_transport_get_rtpinfo (GstRTSPStreamTransport * trans,
+    GstClockTime start_time)
+{
+  GstRTSPStreamTransportPrivate *priv;
+  gchar *url_str;
+  GString *rtpinfo;
+  guint rtptime, seq, clock_rate;
+  GstClockTime running_time = GST_CLOCK_TIME_NONE;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
+
+  priv = trans->priv;
+
+  if (!gst_rtsp_stream_get_rtpinfo (priv->stream, &rtptime, &seq, &clock_rate,
+          &running_time))
+    return NULL;
+
+  GST_DEBUG ("RTP time %u, seq %u, rate %u, running-time %" GST_TIME_FORMAT,
+      rtptime, seq, clock_rate, GST_TIME_ARGS (running_time));
+
+  if (GST_CLOCK_TIME_IS_VALID (running_time)
+      && GST_CLOCK_TIME_IS_VALID (start_time)) {
+    if (running_time > start_time) {
+      rtptime -=
+          gst_util_uint64_scale_int (running_time - start_time, clock_rate,
+          GST_SECOND);
+    } else {
+      rtptime +=
+          gst_util_uint64_scale_int (start_time - running_time, clock_rate,
+          GST_SECOND);
+    }
+  }
+  GST_DEBUG ("RTP time %u, for start-time %" GST_TIME_FORMAT,
+      rtptime, GST_TIME_ARGS (start_time));
+
+  rtpinfo = g_string_new ("");
+
+  url_str = gst_rtsp_url_get_request_uri (trans->priv->url);
+  g_string_append_printf (rtpinfo, "url=%s;seq=%u;rtptime=%u",
+      url_str, seq, rtptime);
+  g_free (url_str);
+
+  return g_string_free (rtpinfo, FALSE);
+}
+
+/**
+ * gst_rtsp_stream_transport_set_active:
+ * @trans: a #GstRTSPStreamTransport
+ * @active: new state of @trans
+ *
+ * Activate or deactivate datatransfer configured in @trans.
+ *
+ * Returns: %TRUE when the state was changed.
+ */
+gboolean
+gst_rtsp_stream_transport_set_active (GstRTSPStreamTransport * trans,
+    gboolean active)
+{
+  GstRTSPStreamTransportPrivate *priv;
+  gboolean res;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
+
+  priv = trans->priv;
+
+  if (priv->active == active)
+    return FALSE;
+
+  if (active)
+    res = gst_rtsp_stream_add_transport (priv->stream, trans);
+  else
+    res = gst_rtsp_stream_remove_transport (priv->stream, trans);
+
+  if (res)
+    priv->active = active;
+
+  return res;
+}
+
+/**
+ * gst_rtsp_stream_transport_set_timed_out:
+ * @trans: a #GstRTSPStreamTransport
+ * @timedout: timed out value
+ *
+ * Set the timed out state of @trans to @timedout
+ */
+void
+gst_rtsp_stream_transport_set_timed_out (GstRTSPStreamTransport * trans,
+    gboolean timedout)
+{
+  g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
+
+  trans->priv->timed_out = timedout;
+}
+
+/**
+ * gst_rtsp_stream_transport_is_timed_out:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Check if @trans is timed out.
+ *
+ * Returns: %TRUE if @trans timed out.
+ */
+gboolean
+gst_rtsp_stream_transport_is_timed_out (GstRTSPStreamTransport * trans)
+{
+  g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
+
+  return trans->priv->timed_out;
+}
+
+/**
+ * gst_rtsp_stream_transport_send_rtp:
+ * @trans: a #GstRTSPStreamTransport
+ * @buffer: (transfer none): a #GstBuffer
+ *
+ * Send @buffer to the installed RTP callback for @trans.
+ *
+ * Returns: %TRUE on success
+ */
+gboolean
+gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport * trans,
+    GstBuffer * buffer)
+{
+  GstRTSPStreamTransportPrivate *priv;
+  gboolean res = FALSE;
+
+  priv = trans->priv;
+
+  if (priv->send_rtp)
+    res =
+        priv->send_rtp (buffer, priv->transport->interleaved.min,
+        priv->user_data);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_stream_transport_send_rtcp:
+ * @trans: a #GstRTSPStreamTransport
+ * @buffer: (transfer none): a #GstBuffer
+ *
+ * Send @buffer to the installed RTCP callback for @trans.
+ *
+ * Returns: %TRUE on success
+ */
+gboolean
+gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport * trans,
+    GstBuffer * buffer)
+{
+  GstRTSPStreamTransportPrivate *priv;
+  gboolean res = FALSE;
+
+  priv = trans->priv;
+
+  if (priv->send_rtcp)
+    res =
+        priv->send_rtcp (buffer, priv->transport->interleaved.max,
+        priv->user_data);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_stream_transport_keep_alive:
+ * @trans: a #GstRTSPStreamTransport
+ *
+ * Signal the installed keep_alive callback for @trans.
+ */
+void
+gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport * trans)
+{
+  GstRTSPStreamTransportPrivate *priv;
+
+  priv = trans->priv;
+
+  if (priv->keep_alive)
+    priv->keep_alive (priv->ka_user_data);
+}
diff --git a/gst/rtsp-server/rtsp-stream-transport.h b/gst/rtsp-server/rtsp-stream-transport.h
new file mode 100644
index 0000000..7735ee7
--- /dev/null
+++ b/gst/rtsp-server/rtsp-stream-transport.h
@@ -0,0 +1,133 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+#include 
+#include 
+
+#ifndef __GST_RTSP_STREAM_TRANSPORT_H__
+#define __GST_RTSP_STREAM_TRANSPORT_H__
+
+G_BEGIN_DECLS
+
+/* types for the media */
+#define GST_TYPE_RTSP_STREAM_TRANSPORT              (gst_rtsp_stream_transport_get_type ())
+#define GST_IS_RTSP_STREAM_TRANSPORT(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT))
+#define GST_IS_RTSP_STREAM_TRANSPORT_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_STREAM_TRANSPORT))
+#define GST_RTSP_STREAM_TRANSPORT_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportClass))
+#define GST_RTSP_STREAM_TRANSPORT(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransport))
+#define GST_RTSP_STREAM_TRANSPORT_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportClass))
+#define GST_RTSP_STREAM_TRANSPORT_CAST(obj)         ((GstRTSPStreamTransport*)(obj))
+#define GST_RTSP_STREAM_TRANSPORT_CLASS_CAST(klass) ((GstRTSPStreamTransportClass*)(klass))
+
+typedef struct _GstRTSPStreamTransport GstRTSPStreamTransport;
+typedef struct _GstRTSPStreamTransportClass GstRTSPStreamTransportClass;
+typedef struct _GstRTSPStreamTransportPrivate GstRTSPStreamTransportPrivate;
+
+#include "rtsp-stream.h"
+
+/**
+ * GstRTSPSendFunc:
+ * @buffer: a #GstBuffer
+ * @channel: a channel
+ * @user_data: user data
+ *
+ * Function registered with gst_rtsp_stream_transport_set_callbacks() and
+ * called when @buffer must be sent on @channel.
+ *
+ * Returns: %TRUE on success
+ */
+typedef gboolean (*GstRTSPSendFunc)      (GstBuffer *buffer, guint8 channel, gpointer user_data);
+/**
+ * GstRTSPKeepAliveFunc:
+ * @user_data: user data
+ *
+ * Function registered with gst_rtsp_stream_transport_set_keepalive() and called
+ * when the stream is active.
+ */
+typedef void     (*GstRTSPKeepAliveFunc) (gpointer user_data);
+
+/**
+ * GstRTSPStreamTransport:
+ * @parent: parent instance
+ *
+ * A Transport description for a stream
+ */
+struct _GstRTSPStreamTransport {
+  GObject              parent;
+
+  /*< private >*/
+  GstRTSPStreamTransportPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+struct _GstRTSPStreamTransportClass {
+  GObjectClass parent_class;
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+GType                    gst_rtsp_stream_transport_get_type (void);
+
+GstRTSPStreamTransport * gst_rtsp_stream_transport_new           (GstRTSPStream *stream,
+                                                                  GstRTSPTransport *tr);
+
+GstRTSPStream *          gst_rtsp_stream_transport_get_stream    (GstRTSPStreamTransport *trans);
+
+void                     gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport *trans,
+                                                                  GstRTSPTransport * tr);
+const GstRTSPTransport * gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport *trans);
+
+void                     gst_rtsp_stream_transport_set_url       (GstRTSPStreamTransport *trans,
+                                                                  const GstRTSPUrl * url);
+const GstRTSPUrl *       gst_rtsp_stream_transport_get_url       (GstRTSPStreamTransport *trans);
+
+
+gchar *                  gst_rtsp_stream_transport_get_rtpinfo   (GstRTSPStreamTransport *trans,
+                                                                  GstClockTime start_time);
+
+void                     gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport *trans,
+                                                                  GstRTSPSendFunc send_rtp,
+                                                                  GstRTSPSendFunc send_rtcp,
+                                                                  gpointer user_data,
+                                                                  GDestroyNotify  notify);
+void                     gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport *trans,
+                                                                  GstRTSPKeepAliveFunc keep_alive,
+                                                                  gpointer user_data,
+                                                                  GDestroyNotify  notify);
+void                     gst_rtsp_stream_transport_keep_alive    (GstRTSPStreamTransport *trans);
+
+gboolean                 gst_rtsp_stream_transport_set_active    (GstRTSPStreamTransport *trans,
+                                                                  gboolean active);
+
+void                     gst_rtsp_stream_transport_set_timed_out (GstRTSPStreamTransport *trans,
+                                                                  gboolean timedout);
+gboolean                 gst_rtsp_stream_transport_is_timed_out  (GstRTSPStreamTransport *trans);
+
+
+
+gboolean                 gst_rtsp_stream_transport_send_rtp      (GstRTSPStreamTransport *trans,
+                                                                  GstBuffer *buffer);
+gboolean                 gst_rtsp_stream_transport_send_rtcp     (GstRTSPStreamTransport *trans,
+                                                                  GstBuffer *buffer);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_STREAM_TRANSPORT_H__ */
diff --git a/gst/rtsp-server/rtsp-stream.c b/gst/rtsp-server/rtsp-stream.c
new file mode 100644
index 0000000..8e1b216
--- /dev/null
+++ b/gst/rtsp-server/rtsp-stream.c
@@ -0,0 +1,2799 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-stream
+ * @short_description: A media stream
+ * @see_also: #GstRTSPMedia
+ *
+ * The #GstRTSPStream object manages the data transport for one stream. It
+ * is created from a payloader element and a source pad that produce the RTP
+ * packets for the stream.
+ *
+ * With gst_rtsp_stream_join_bin() the streaming elements are added to the bin
+ * and rtpbin. gst_rtsp_stream_leave_bin() removes the elements again.
+ *
+ * The #GstRTSPStream will use the configured addresspool, as set with
+ * gst_rtsp_stream_set_address_pool(), to allocate multicast addresses for the
+ * stream. With gst_rtsp_stream_get_multicast_address() you can get the
+ * configured address.
+ *
+ * With gst_rtsp_stream_get_server_port () you can get the port that the server
+ * will use to receive RTCP. This is the part that the clients will use to send
+ * RTCP to.
+ *
+ * With gst_rtsp_stream_add_transport() destinations can be added where the
+ * stream should be sent to. Use gst_rtsp_stream_remove_transport() to remove
+ * the destination again.
+ *
+ * Last reviewed on 2013-07-16 (1.0.0)
+ */
+
+#include 
+#include 
+#include 
+
+#include 
+
+#include 
+#include 
+
+#include "rtsp-stream.h"
+
+#define GST_RTSP_STREAM_GET_PRIVATE(obj)  \
+     (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamPrivate))
+
+struct _GstRTSPStreamPrivate
+{
+  GMutex lock;
+  guint idx;
+  GstPad *srcpad;
+  GstElement *payloader;
+  guint buffer_size;
+  gboolean is_joined;
+  gchar *control;
+
+  GstRTSPProfile profiles;
+  GstRTSPLowerTrans protocols;
+
+  /* pads on the rtpbin */
+  GstPad *send_rtp_sink;
+  GstPad *recv_sink[2];
+  GstPad *send_src[2];
+
+  /* the RTPSession object */
+  GObject *session;
+
+  /* SRTP encoder/decoder */
+  GstElement *srtpenc;
+  GstElement *srtpdec;
+  GHashTable *keys;
+
+  /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
+   * sockets */
+  GstElement *udpsrc_v4[2];
+
+  /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
+   * sockets */
+  GstElement *udpsrc_v6[2];
+
+  GstElement *udpsink[2];
+
+  /* for TCP transport */
+  GstElement *appsrc[2];
+  GstElement *appqueue[2];
+  GstElement *appsink[2];
+
+  GstElement *tee[2];
+  GstElement *funnel[2];
+
+  /* server ports for sending/receiving over ipv4 */
+  GstRTSPRange server_port_v4;
+  GstRTSPAddress *server_addr_v4;
+  gboolean have_ipv4;
+
+  /* server ports for sending/receiving over ipv6 */
+  GstRTSPRange server_port_v6;
+  GstRTSPAddress *server_addr_v6;
+  gboolean have_ipv6;
+
+  /* multicast addresses */
+  GstRTSPAddressPool *pool;
+  GstRTSPAddress *addr_v4;
+  GstRTSPAddress *addr_v6;
+
+  /* the caps of the stream */
+  gulong caps_sig;
+  GstCaps *caps;
+
+  /* transports we stream to */
+  guint n_active;
+  GList *transports;
+  guint transports_cookie;
+  GList *tr_cache;
+  guint tr_cache_cookie;
+
+  gint dscp_qos;
+
+  /* stream blocking */
+  gulong blocked_id;
+  gboolean blocking;
+};
+
+#define DEFAULT_CONTROL         NULL
+#define DEFAULT_PROFILES        GST_RTSP_PROFILE_AVP
+#define DEFAULT_PROTOCOLS       GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
+                                        GST_RTSP_LOWER_TRANS_TCP
+
+enum
+{
+  PROP_0,
+  PROP_CONTROL,
+  PROP_PROFILES,
+  PROP_PROTOCOLS,
+  PROP_LAST
+};
+
+enum
+{
+  SIGNAL_NEW_RTP_ENCODER,
+  SIGNAL_NEW_RTCP_ENCODER,
+  SIGNAL_RTCP_STATS,
+  SIGNAL_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
+#define GST_CAT_DEFAULT rtsp_stream_debug
+
+static GQuark ssrc_stream_map_key;
+
+static void gst_rtsp_stream_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec);
+static void gst_rtsp_stream_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec);
+
+static void gst_rtsp_stream_finalize (GObject * obj);
+
+static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
+
+G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
+
+  gobject_class = G_OBJECT_CLASS (klass);
+
+  gobject_class->get_property = gst_rtsp_stream_get_property;
+  gobject_class->set_property = gst_rtsp_stream_set_property;
+  gobject_class->finalize = gst_rtsp_stream_finalize;
+
+  g_object_class_install_property (gobject_class, PROP_CONTROL,
+      g_param_spec_string ("control", "Control",
+          "The control string for this stream", DEFAULT_CONTROL,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_PROFILES,
+      g_param_spec_flags ("profiles", "Profiles",
+          "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
+          DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
+      g_param_spec_flags ("protocols", "Protocols",
+          "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
+          DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER] =
+      g_signal_new ("new-rtp-encoder", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
+      G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+
+  gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER] =
+      g_signal_new ("new-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
+      G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+
+  gst_rtsp_stream_signals[SIGNAL_RTCP_STATS] =
+      g_signal_new ("rtcp-statistics", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
+      G_TYPE_NONE, 1, GST_TYPE_STRUCTURE);
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
+
+  ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
+}
+
+static void
+gst_rtsp_stream_init (GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
+
+  GST_DEBUG ("new stream %p", stream);
+
+  stream->priv = priv;
+
+  priv->dscp_qos = -1;
+  priv->control = g_strdup (DEFAULT_CONTROL);
+  priv->profiles = DEFAULT_PROFILES;
+  priv->protocols = DEFAULT_PROTOCOLS;
+
+  g_mutex_init (&priv->lock);
+
+  priv->keys = g_hash_table_new_full (g_direct_hash, g_direct_equal,
+      NULL, (GDestroyNotify) gst_caps_unref);
+}
+
+static void
+gst_rtsp_stream_finalize (GObject * obj)
+{
+  GstRTSPStream *stream;
+  GstRTSPStreamPrivate *priv;
+
+  stream = GST_RTSP_STREAM (obj);
+  priv = stream->priv;
+
+  GST_DEBUG ("finalize stream %p", stream);
+
+  /* we really need to be unjoined now */
+  g_return_if_fail (!priv->is_joined);
+
+  if (priv->addr_v4)
+    gst_rtsp_address_free (priv->addr_v4);
+  if (priv->addr_v6)
+    gst_rtsp_address_free (priv->addr_v6);
+  if (priv->server_addr_v4)
+    gst_rtsp_address_free (priv->server_addr_v4);
+  if (priv->server_addr_v6)
+    gst_rtsp_address_free (priv->server_addr_v6);
+  if (priv->pool)
+    g_object_unref (priv->pool);
+  gst_object_unref (priv->payloader);
+  gst_object_unref (priv->srcpad);
+  g_free (priv->control);
+  g_mutex_clear (&priv->lock);
+
+  g_hash_table_unref (priv->keys);
+
+  G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_stream_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRTSPStream *stream = GST_RTSP_STREAM (object);
+
+  switch (propid) {
+    case PROP_CONTROL:
+      g_value_take_string (value, gst_rtsp_stream_get_control (stream));
+      break;
+    case PROP_PROFILES:
+      g_value_set_flags (value, gst_rtsp_stream_get_profiles (stream));
+      break;
+    case PROP_PROTOCOLS:
+      g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+static void
+gst_rtsp_stream_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRTSPStream *stream = GST_RTSP_STREAM (object);
+
+  switch (propid) {
+    case PROP_CONTROL:
+      gst_rtsp_stream_set_control (stream, g_value_get_string (value));
+      break;
+    case PROP_PROFILES:
+      gst_rtsp_stream_set_profiles (stream, g_value_get_flags (value));
+      break;
+    case PROP_PROTOCOLS:
+      gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+/**
+ * gst_rtsp_stream_new:
+ * @idx: an index
+ * @srcpad: a #GstPad
+ * @payloader: a #GstElement
+ *
+ * Create a new media stream with index @idx that handles RTP data on
+ * @srcpad and has a payloader element @payloader.
+ *
+ * Returns: (transfer full): a new #GstRTSPStream
+ */
+GstRTSPStream *
+gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
+{
+  GstRTSPStreamPrivate *priv;
+  GstRTSPStream *stream;
+
+  g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
+  g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
+  g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
+
+  stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
+  priv = stream->priv;
+  priv->idx = idx;
+  priv->payloader = gst_object_ref (payloader);
+  priv->srcpad = gst_object_ref (srcpad);
+
+  return stream;
+}
+
+/**
+ * gst_rtsp_stream_get_index:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the stream index.
+ *
+ * Return: the stream index.
+ */
+guint
+gst_rtsp_stream_get_index (GstRTSPStream * stream)
+{
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
+
+  return stream->priv->idx;
+}
+
+/**
+ * gst_rtsp_stream_get_pt:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the stream payload type.
+ *
+ * Return: the stream payload type.
+ */
+guint
+gst_rtsp_stream_get_pt (GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv;
+  guint pt;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
+
+  priv = stream->priv;
+
+  g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
+
+  return pt;
+}
+
+/**
+ * gst_rtsp_stream_get_srcpad:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the srcpad associated with @stream.
+ *
+ * Returns: (transfer full): the srcpad. Unref after usage.
+ */
+GstPad *
+gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
+{
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+  return gst_object_ref (stream->priv->srcpad);
+}
+
+/**
+ * gst_rtsp_stream_get_control:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the control string to identify this stream.
+ *
+ * Returns: (transfer full): the control string. g_free() after usage.
+ */
+gchar *
+gst_rtsp_stream_get_control (GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv;
+  gchar *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  if ((result = g_strdup (priv->control)) == NULL)
+    result = g_strdup_printf ("stream=%u", priv->idx);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_stream_set_control:
+ * @stream: a #GstRTSPStream
+ * @control: a control string
+ *
+ * Set the control string in @stream.
+ */
+void
+gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
+{
+  GstRTSPStreamPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  g_free (priv->control);
+  priv->control = g_strdup (control);
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_has_control:
+ * @stream: a #GstRTSPStream
+ * @control: a control string
+ *
+ * Check if @stream has the control string @control.
+ *
+ * Returns: %TRUE is @stream has @control as the control string
+ */
+gboolean
+gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
+{
+  GstRTSPStreamPrivate *priv;
+  gboolean res;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  if (priv->control)
+    res = (g_strcmp0 (priv->control, control) == 0);
+  else {
+    guint streamid;
+
+    if (sscanf (control, "stream=%u", &streamid) > 0)
+      res = (streamid == priv->idx);
+    else
+      res = FALSE;
+  }
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_stream_set_mtu:
+ * @stream: a #GstRTSPStream
+ * @mtu: a new MTU
+ *
+ * Configure the mtu in the payloader of @stream to @mtu.
+ */
+void
+gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
+{
+  GstRTSPStreamPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+  priv = stream->priv;
+
+  GST_LOG_OBJECT (stream, "set MTU %u", mtu);
+
+  g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
+}
+
+/**
+ * gst_rtsp_stream_get_mtu:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the configured MTU in the payloader of @stream.
+ *
+ * Returns: the MTU of the payloader.
+ */
+guint
+gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv;
+  guint mtu;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
+
+  priv = stream->priv;
+
+  g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
+
+  return mtu;
+}
+
+/* Update the dscp qos property on the udp sinks */
+static void
+update_dscp_qos (GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+  priv = stream->priv;
+
+  if (priv->udpsink[0]) {
+    g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
+        NULL);
+  }
+
+  if (priv->udpsink[1]) {
+    g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
+        NULL);
+  }
+}
+
+/**
+ * gst_rtsp_stream_set_dscp_qos:
+ * @stream: a #GstRTSPStream
+ * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
+ *
+ * Configure the dscp qos of the outgoing sockets to @dscp_qos.
+ */
+void
+gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
+{
+  GstRTSPStreamPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+  priv = stream->priv;
+
+  GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
+
+  if (dscp_qos < -1 || dscp_qos > 63) {
+    GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
+    return;
+  }
+
+  priv->dscp_qos = dscp_qos;
+
+  update_dscp_qos (stream);
+}
+
+/**
+ * gst_rtsp_stream_get_dscp_qos:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the configured DSCP QoS in of the outgoing sockets.
+ *
+ * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
+ */
+gint
+gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
+
+  priv = stream->priv;
+
+  return priv->dscp_qos;
+}
+
+/**
+ * gst_rtsp_stream_is_transport_supported:
+ * @stream: a #GstRTSPStream
+ * @transport: (transfer none): a #GstRTSPTransport
+ *
+ * Check if @transport can be handled by stream
+ *
+ * Returns: %TRUE if @transport can be handled by @stream.
+ */
+gboolean
+gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
+    GstRTSPTransport * transport)
+{
+  GstRTSPStreamPrivate *priv;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  if (transport->trans != GST_RTSP_TRANS_RTP)
+    goto unsupported_transmode;
+
+  if (!(transport->profile & priv->profiles))
+    goto unsupported_profile;
+
+  if (!(transport->lower_transport & priv->protocols))
+    goto unsupported_ltrans;
+
+  g_mutex_unlock (&priv->lock);
+
+  return TRUE;
+
+  /* ERRORS */
+unsupported_transmode:
+  {
+    GST_DEBUG ("unsupported transport mode %d", transport->trans);
+    g_mutex_unlock (&priv->lock);
+    return FALSE;
+  }
+unsupported_profile:
+  {
+    GST_DEBUG ("unsupported profile %d", transport->profile);
+    g_mutex_unlock (&priv->lock);
+    return FALSE;
+  }
+unsupported_ltrans:
+  {
+    GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
+    g_mutex_unlock (&priv->lock);
+    return FALSE;
+  }
+}
+
+/**
+ * gst_rtsp_stream_set_profiles:
+ * @stream: a #GstRTSPStream
+ * @profiles: the new profiles
+ *
+ * Configure the allowed profiles for @stream.
+ */
+void
+gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
+{
+  GstRTSPStreamPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->profiles = profiles;
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_profiles:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the allowed profiles of @stream.
+ *
+ * Returns: a #GstRTSPProfile
+ */
+GstRTSPProfile
+gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv;
+  GstRTSPProfile res;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  res = priv->profiles;
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_stream_set_protocols:
+ * @stream: a #GstRTSPStream
+ * @protocols: the new flags
+ *
+ * Configure the allowed lower transport for @stream.
+ */
+void
+gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
+    GstRTSPLowerTrans protocols)
+{
+  GstRTSPStreamPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->protocols = protocols;
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_protocols:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the allowed protocols of @stream.
+ *
+ * Returns: a #GstRTSPLowerTrans
+ */
+GstRTSPLowerTrans
+gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv;
+  GstRTSPLowerTrans res;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
+      GST_RTSP_LOWER_TRANS_UNKNOWN);
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  res = priv->protocols;
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_stream_set_address_pool:
+ * @stream: a #GstRTSPStream
+ * @pool: (transfer none): a #GstRTSPAddressPool
+ *
+ * configure @pool to be used as the address pool of @stream.
+ */
+void
+gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
+    GstRTSPAddressPool * pool)
+{
+  GstRTSPStreamPrivate *priv;
+  GstRTSPAddressPool *old;
+
+  g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+  priv = stream->priv;
+
+  GST_LOG_OBJECT (stream, "set address pool %p", pool);
+
+  g_mutex_lock (&priv->lock);
+  if ((old = priv->pool) != pool)
+    priv->pool = pool ? g_object_ref (pool) : NULL;
+  else
+    old = NULL;
+  g_mutex_unlock (&priv->lock);
+
+  if (old)
+    g_object_unref (old);
+}
+
+/**
+ * gst_rtsp_stream_get_address_pool:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the #GstRTSPAddressPool used as the address pool of @stream.
+ *
+ * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
+ * usage.
+ */
+GstRTSPAddressPool *
+gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv;
+  GstRTSPAddressPool *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  if ((result = priv->pool))
+    g_object_ref (result);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_stream_get_multicast_address:
+ * @stream: a #GstRTSPStream
+ * @family: the #GSocketFamily
+ *
+ * Get the multicast address of @stream for @family.
+ *
+ * Returns: (transfer full) (nullable): the #GstRTSPAddress of @stream
+ * or %NULL when no address could be allocated. gst_rtsp_address_free()
+ * after usage.
+ */
+GstRTSPAddress *
+gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
+    GSocketFamily family)
+{
+  GstRTSPStreamPrivate *priv;
+  GstRTSPAddress *result;
+  GstRTSPAddress **addrp;
+  GstRTSPAddressFlags flags;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+  priv = stream->priv;
+
+  if (family == G_SOCKET_FAMILY_IPV6) {
+    flags = GST_RTSP_ADDRESS_FLAG_IPV6;
+    addrp = &priv->addr_v6;
+  } else {
+    flags = GST_RTSP_ADDRESS_FLAG_IPV4;
+    addrp = &priv->addr_v4;
+  }
+
+  g_mutex_lock (&priv->lock);
+  if (*addrp == NULL) {
+    if (priv->pool == NULL)
+      goto no_pool;
+
+    flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
+
+    *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
+    if (*addrp == NULL)
+      goto no_address;
+  }
+  result = gst_rtsp_address_copy (*addrp);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+
+  /* ERRORS */
+no_pool:
+  {
+    GST_ERROR_OBJECT (stream, "no address pool specified");
+    g_mutex_unlock (&priv->lock);
+    return NULL;
+  }
+no_address:
+  {
+    GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
+    g_mutex_unlock (&priv->lock);
+    return NULL;
+  }
+}
+
+/**
+ * gst_rtsp_stream_reserve_address:
+ * @stream: a #GstRTSPStream
+ * @address: an address
+ * @port: a port
+ * @n_ports: n_ports
+ * @ttl: a TTL
+ *
+ * Reserve @address and @port as the address and port of @stream.
+ *
+ * Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
+ * the address could be reserved. gst_rtsp_address_free() after usage.
+ */
+GstRTSPAddress *
+gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
+    const gchar * address, guint port, guint n_ports, guint ttl)
+{
+  GstRTSPStreamPrivate *priv;
+  GstRTSPAddress *result;
+  GInetAddress *addr;
+  GSocketFamily family;
+  GstRTSPAddress **addrp;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+  g_return_val_if_fail (address != NULL, NULL);
+  g_return_val_if_fail (port > 0, NULL);
+  g_return_val_if_fail (n_ports > 0, NULL);
+  g_return_val_if_fail (ttl > 0, NULL);
+
+  priv = stream->priv;
+
+  addr = g_inet_address_new_from_string (address);
+  if (!addr) {
+    GST_ERROR ("failed to get inet addr from %s", address);
+    family = G_SOCKET_FAMILY_IPV4;
+  } else {
+    family = g_inet_address_get_family (addr);
+    g_object_unref (addr);
+  }
+
+  if (family == G_SOCKET_FAMILY_IPV6)
+    addrp = &priv->addr_v6;
+  else
+    addrp = &priv->addr_v4;
+
+  g_mutex_lock (&priv->lock);
+  if (*addrp == NULL) {
+    GstRTSPAddressPoolResult res;
+
+    if (priv->pool == NULL)
+      goto no_pool;
+
+    res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
+        port, n_ports, ttl, addrp);
+    if (res != GST_RTSP_ADDRESS_POOL_OK)
+      goto no_address;
+  } else {
+    if (strcmp ((*addrp)->address, address) ||
+        (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
+        (*addrp)->ttl != ttl)
+      goto different_address;
+  }
+  result = gst_rtsp_address_copy (*addrp);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+
+  /* ERRORS */
+no_pool:
+  {
+    GST_ERROR_OBJECT (stream, "no address pool specified");
+    g_mutex_unlock (&priv->lock);
+    return NULL;
+  }
+no_address:
+  {
+    GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
+        address);
+    g_mutex_unlock (&priv->lock);
+    return NULL;
+  }
+different_address:
+  {
+    GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
+        " reserved", address);
+    g_mutex_unlock (&priv->lock);
+    return NULL;
+  }
+}
+
+static gboolean
+alloc_ports_one_family (GstRTSPAddressPool * pool, gint buffer_size,
+    GSocketFamily family, GstElement * udpsrc_out[2],
+    GstElement * udpsink_out[2], GstRTSPRange * server_port_out,
+    GstRTSPAddress ** server_addr_out)
+{
+  GstStateChangeReturn ret;
+  GstElement *udpsrc0, *udpsrc1;
+  GstElement *udpsink0, *udpsink1;
+  GSocket *rtp_socket = NULL;
+  GSocket *rtcp_socket;
+  gint tmp_rtp, tmp_rtcp;
+  guint count;
+  gint rtpport, rtcpport;
+  GList *rejected_addresses = NULL;
+  GstRTSPAddress *addr = NULL;
+  GInetAddress *inetaddr = NULL;
+  GSocketAddress *rtp_sockaddr = NULL;
+  GSocketAddress *rtcp_sockaddr = NULL;
+  const gchar *multisink_socket;
+
+  if (family == G_SOCKET_FAMILY_IPV6)
+    multisink_socket = "socket-v6";
+  else
+    multisink_socket = "socket";
+
+  udpsrc0 = NULL;
+  udpsrc1 = NULL;
+  udpsink0 = NULL;
+  udpsink1 = NULL;
+  count = 0;
+
+  /* Start with random port */
+  tmp_rtp = 0;
+
+  rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
+      G_SOCKET_PROTOCOL_UDP, NULL);
+  if (!rtcp_socket)
+    goto no_udp_protocol;
+
+  if (*server_addr_out)
+    gst_rtsp_address_free (*server_addr_out);
+
+  /* try to allocate 2 UDP ports, the RTP port should be an even
+   * number and the RTCP port should be the next (uneven) port */
+again:
+
+  if (rtp_socket == NULL) {
+    rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
+        G_SOCKET_PROTOCOL_UDP, NULL);
+    if (!rtp_socket)
+      goto no_udp_protocol;
+  }
+
+  if (pool && gst_rtsp_address_pool_has_unicast_addresses (pool)) {
+    GstRTSPAddressFlags flags;
+
+    if (addr)
+      rejected_addresses = g_list_prepend (rejected_addresses, addr);
+
+    flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
+    if (family == G_SOCKET_FAMILY_IPV6)
+      flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
+    else
+      flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
+
+    addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
+
+    if (addr == NULL)
+      goto no_ports;
+
+    tmp_rtp = addr->port;
+
+    g_clear_object (&inetaddr);
+    inetaddr = g_inet_address_new_from_string (addr->address);
+  } else {
+    if (tmp_rtp != 0) {
+      tmp_rtp += 2;
+      if (++count > 20)
+        goto no_ports;
+    }
+
+    if (inetaddr == NULL)
+      inetaddr = g_inet_address_new_any (family);
+  }
+
+  /* FIXME-WFD : Force to set 19000 as port number */
+  tmp_rtp = 19000;
+
+  rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
+  if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
+    g_object_unref (rtp_sockaddr);
+    goto again;
+  }
+  g_object_unref (rtp_sockaddr);
+
+  rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
+  if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
+    g_clear_object (&rtp_sockaddr);
+    goto socket_error;
+  }
+
+  tmp_rtp =
+      g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
+  g_object_unref (rtp_sockaddr);
+
+  /* check if port is even */
+  if ((tmp_rtp & 1) != 0) {
+    /* port not even, close and allocate another */
+    tmp_rtp++;
+    g_clear_object (&rtp_socket);
+    goto again;
+  }
+
+  /* set port */
+  tmp_rtcp = tmp_rtp + 1;
+
+  rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
+  if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
+    g_object_unref (rtcp_sockaddr);
+    g_clear_object (&rtp_socket);
+    goto again;
+  }
+  g_object_unref (rtcp_sockaddr);
+
+  g_clear_object (&inetaddr);
+
+  udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
+  udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
+
+  if (udpsrc0 == NULL || udpsrc1 == NULL)
+    goto no_udp_protocol;
+
+  g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
+  g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
+
+  ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
+  if (ret == GST_STATE_CHANGE_FAILURE)
+    goto element_error;
+  ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
+  if (ret == GST_STATE_CHANGE_FAILURE)
+    goto element_error;
+
+  /* all fine, do port check */
+  g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
+  g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
+
+  /* this should not happen... */
+  if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
+    goto port_error;
+
+  if (udpsink_out[0])
+    udpsink0 = udpsink_out[0];
+  else
+    udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
+
+  if (!udpsink0)
+    goto no_udp_protocol;
+
+  g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
+  g_object_set (G_OBJECT (udpsink0), multisink_socket, rtp_socket, NULL);
+
+  if (udpsink_out[1])
+    udpsink1 = udpsink_out[1];
+  else
+    udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
+
+  if (!udpsink1)
+    goto no_udp_protocol;
+
+  g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
+  g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
+  g_object_set (G_OBJECT (udpsink0), "buffer-size", buffer_size, NULL);
+
+  g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
+  g_object_set (G_OBJECT (udpsink1), multisink_socket, rtcp_socket, NULL);
+  g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
+  g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
+  g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
+  g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
+  g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
+  g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
+
+  /* we keep these elements, we will further configure them when the
+   * client told us to really use the UDP ports. */
+  udpsrc_out[0] = udpsrc0;
+  udpsrc_out[1] = udpsrc1;
+  udpsink_out[0] = udpsink0;
+  udpsink_out[1] = udpsink1;
+  server_port_out->min = rtpport;
+  server_port_out->max = rtcpport;
+
+  *server_addr_out = addr;
+  g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
+
+  g_object_unref (rtp_socket);
+  g_object_unref (rtcp_socket);
+
+  return TRUE;
+
+  /* ERRORS */
+no_udp_protocol:
+  {
+    goto cleanup;
+  }
+no_ports:
+  {
+    goto cleanup;
+  }
+port_error:
+  {
+    goto cleanup;
+  }
+socket_error:
+  {
+    goto cleanup;
+  }
+element_error:
+  {
+    goto cleanup;
+  }
+cleanup:
+  {
+    if (udpsrc0) {
+      gst_element_set_state (udpsrc0, GST_STATE_NULL);
+      gst_object_unref (udpsrc0);
+    }
+    if (udpsrc1) {
+      gst_element_set_state (udpsrc1, GST_STATE_NULL);
+      gst_object_unref (udpsrc1);
+    }
+    if (udpsink0) {
+      gst_element_set_state (udpsink0, GST_STATE_NULL);
+      gst_object_unref (udpsink0);
+    }
+    if (inetaddr)
+      g_object_unref (inetaddr);
+    g_list_free_full (rejected_addresses,
+        (GDestroyNotify) gst_rtsp_address_free);
+    if (addr)
+      gst_rtsp_address_free (addr);
+    if (rtp_socket)
+      g_object_unref (rtp_socket);
+    if (rtcp_socket)
+      g_object_unref (rtcp_socket);
+    return FALSE;
+  }
+}
+
+/* must be called with lock */
+static gboolean
+alloc_ports (GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv = stream->priv;
+
+  priv->have_ipv4 = alloc_ports_one_family (priv->pool, priv->buffer_size,
+      G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4, priv->udpsink,
+      &priv->server_port_v4, &priv->server_addr_v4);
+
+  /* FIXME-WFD : force to disable ipv6 mode in WFD mode */
+#if 0
+  priv->have_ipv6 = alloc_ports_one_family (priv->pool, priv->buffer_size,
+      G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6, priv->udpsink,
+      &priv->server_port_v6, &priv->server_addr_v6);
+#else
+  priv->have_ipv6 = FALSE;
+#endif
+
+  return priv->have_ipv4 || priv->have_ipv6;
+}
+
+/**
+ * gst_rtsp_stream_get_server_port:
+ * @stream: a #GstRTSPStream
+ * @server_port: (out): result server port
+ * @family: the port family to get
+ *
+ * Fill @server_port with the port pair used by the server. This function can
+ * only be called when @stream has been joined.
+ */
+void
+gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
+    GstRTSPRange * server_port, GSocketFamily family)
+{
+  GstRTSPStreamPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+  priv = stream->priv;
+  g_return_if_fail (priv->is_joined);
+
+  g_mutex_lock (&priv->lock);
+  if (family == G_SOCKET_FAMILY_IPV4) {
+    if (server_port)
+      *server_port = priv->server_port_v4;
+  } else {
+    if (server_port)
+      *server_port = priv->server_port_v6;
+  }
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_stream_get_rtpsession:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the RTP session of this stream.
+ *
+ * Returns: (transfer full): The RTP session of this stream. Unref after usage.
+ */
+GObject *
+gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv;
+  GObject *session;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  if ((session = priv->session))
+    g_object_ref (session);
+  g_mutex_unlock (&priv->lock);
+
+  return session;
+}
+
+/**
+ * gst_rtsp_stream_get_ssrc:
+ * @stream: a #GstRTSPStream
+ * @ssrc: (out): result ssrc
+ *
+ * Get the SSRC used by the RTP session of this stream. This function can only
+ * be called when @stream has been joined.
+ */
+void
+gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
+{
+  GstRTSPStreamPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+  priv = stream->priv;
+  g_return_if_fail (priv->is_joined);
+
+  g_mutex_lock (&priv->lock);
+  if (ssrc && priv->session)
+    g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
+  g_mutex_unlock (&priv->lock);
+}
+
+/* executed from streaming thread */
+static void
+caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv = stream->priv;
+  GstCaps *newcaps, *oldcaps;
+
+  newcaps = gst_pad_get_current_caps (pad);
+
+  GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
+      newcaps);
+
+  g_mutex_lock (&priv->lock);
+  oldcaps = priv->caps;
+  priv->caps = newcaps;
+  g_mutex_unlock (&priv->lock);
+
+  if (oldcaps)
+    gst_caps_unref (oldcaps);
+}
+
+static void
+dump_structure (const GstStructure * s)
+{
+  gchar *sstr;
+
+  sstr = gst_structure_to_string (s);
+  GST_INFO ("structure: %s", sstr);
+  g_free (sstr);
+}
+
+static GstRTSPStreamTransport *
+find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
+{
+  GstRTSPStreamPrivate *priv = stream->priv;
+  GList *walk;
+  GstRTSPStreamTransport *result = NULL;
+  const gchar *tmp;
+  gchar *dest;
+  guint port;
+
+  if (rtcp_from == NULL)
+    return NULL;
+
+  tmp = g_strrstr (rtcp_from, ":");
+  if (tmp == NULL)
+    return NULL;
+
+  port = atoi (tmp + 1);
+  dest = g_strndup (rtcp_from, tmp - rtcp_from);
+
+  g_mutex_lock (&priv->lock);
+  GST_INFO ("finding %s:%d in %d transports", dest, port,
+      g_list_length (priv->transports));
+
+  for (walk = priv->transports; walk; walk = g_list_next (walk)) {
+    GstRTSPStreamTransport *trans = walk->data;
+    const GstRTSPTransport *tr;
+    gint min, max;
+
+    tr = gst_rtsp_stream_transport_get_transport (trans);
+
+    min = tr->client_port.min;
+    max = tr->client_port.max;
+
+    if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
+      result = trans;
+      break;
+    }
+  }
+  if (result)
+    g_object_ref (result);
+  g_mutex_unlock (&priv->lock);
+
+  g_free (dest);
+
+  return result;
+}
+
+static GstRTSPStreamTransport *
+check_transport (GObject * source, GstRTSPStream * stream)
+{
+  GstStructure *stats;
+  GstRTSPStreamTransport *trans;
+
+  /* see if we have a stream to match with the origin of the RTCP packet */
+  trans = g_object_get_qdata (source, ssrc_stream_map_key);
+  if (trans == NULL) {
+    g_object_get (source, "stats", &stats, NULL);
+    if (stats) {
+      const gchar *rtcp_from;
+
+      dump_structure (stats);
+
+      g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_RTCP_STATS], 0, stats);
+
+      rtcp_from = gst_structure_get_string (stats, "rtcp-from");
+      if ((trans = find_transport (stream, rtcp_from))) {
+        GST_INFO ("%p: found transport %p for source  %p", stream, trans,
+            source);
+        g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
+            g_object_unref);
+      }
+      gst_structure_free (stats);
+    }
+  }
+  return trans;
+}
+
+
+static void
+on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+  GstRTSPStreamTransport *trans;
+
+  GST_INFO ("%p: new source %p", stream, source);
+
+  trans = check_transport (source, stream);
+
+  if (trans)
+    GST_INFO ("%p: source %p for transport %p", stream, source, trans);
+}
+
+static void
+on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+  GST_INFO ("%p: new SDES %p", stream, source);
+}
+
+static void
+on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+  GstRTSPStreamTransport *trans;
+
+  trans = check_transport (source, stream);
+
+  if (trans) {
+    GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
+    gst_rtsp_stream_transport_keep_alive (trans);
+  }
+  {
+    GstStructure *stats;
+    g_object_get (source, "stats", &stats, NULL);
+    if (stats) {
+      g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_RTCP_STATS], 0, stats);
+
+#ifdef DUMP_STATS
+      dump_structure (stats);
+#endif
+      gst_structure_free (stats);
+    }
+  }
+}
+
+static void
+on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+  GST_INFO ("%p: source %p bye", stream, source);
+}
+
+static void
+on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+  GstRTSPStreamTransport *trans;
+
+  GST_INFO ("%p: source %p bye timeout", stream, source);
+
+  if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
+    gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
+    g_object_set_qdata (source, ssrc_stream_map_key, NULL);
+  }
+}
+
+static void
+on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+  GstRTSPStreamTransport *trans;
+
+  GST_INFO ("%p: source %p timeout", stream, source);
+
+  if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
+    gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
+    g_object_set_qdata (source, ssrc_stream_map_key, NULL);
+  }
+}
+
+static void
+clear_tr_cache (GstRTSPStreamPrivate * priv)
+{
+  g_list_foreach (priv->tr_cache, (GFunc) g_object_unref, NULL);
+  g_list_free (priv->tr_cache);
+  priv->tr_cache = NULL;
+}
+
+static GstFlowReturn
+handle_new_sample (GstAppSink * sink, gpointer user_data)
+{
+  GstRTSPStreamPrivate *priv;
+  GList *walk;
+  GstSample *sample;
+  GstBuffer *buffer;
+  GstRTSPStream *stream;
+  gboolean is_rtp;
+
+  sample = gst_app_sink_pull_sample (sink);
+  if (!sample)
+    return GST_FLOW_OK;
+
+  stream = (GstRTSPStream *) user_data;
+  priv = stream->priv;
+  buffer = gst_sample_get_buffer (sample);
+
+  is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
+
+  g_mutex_lock (&priv->lock);
+  if (priv->tr_cache_cookie != priv->transports_cookie) {
+    clear_tr_cache (priv);
+    for (walk = priv->transports; walk; walk = g_list_next (walk)) {
+      GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
+      priv->tr_cache = g_list_prepend (priv->tr_cache, g_object_ref (tr));
+    }
+    priv->tr_cache_cookie = priv->transports_cookie;
+  }
+  g_mutex_unlock (&priv->lock);
+
+  for (walk = priv->tr_cache; walk; walk = g_list_next (walk)) {
+    GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
+
+    if (is_rtp) {
+      gst_rtsp_stream_transport_send_rtp (tr, buffer);
+    } else {
+      gst_rtsp_stream_transport_send_rtcp (tr, buffer);
+    }
+  }
+  gst_sample_unref (sample);
+
+  return GST_FLOW_OK;
+}
+
+static GstAppSinkCallbacks sink_cb = {
+  NULL,                         /* not interested in EOS */
+  NULL,                         /* not interested in preroll samples */
+  handle_new_sample,
+};
+
+static GstElement *
+get_rtp_encoder (GstRTSPStream * stream, guint session)
+{
+  GstRTSPStreamPrivate *priv = stream->priv;
+
+  if (priv->srtpenc == NULL) {
+    gchar *name;
+
+    name = g_strdup_printf ("srtpenc_%u", session);
+    priv->srtpenc = gst_element_factory_make ("srtpenc", name);
+    g_free (name);
+
+    g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
+  }
+  return gst_object_ref (priv->srtpenc);
+}
+
+static GstElement *
+request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv = stream->priv;
+  GstElement *oldenc, *enc;
+  GstPad *pad;
+  gchar *name;
+
+  if (priv->idx != session)
+    return NULL;
+
+  GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
+
+  oldenc = priv->srtpenc;
+  enc = get_rtp_encoder (stream, session);
+  name = g_strdup_printf ("rtp_sink_%d", session);
+  pad = gst_element_get_request_pad (enc, name);
+  g_free (name);
+  gst_object_unref (pad);
+
+  if (oldenc == NULL)
+    g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
+        enc);
+
+  return enc;
+}
+
+static GstElement *
+request_rtcp_encoder (GstElement * rtpbin, guint session,
+    GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv = stream->priv;
+  GstElement *oldenc, *enc;
+  GstPad *pad;
+  gchar *name;
+
+  if (priv->idx != session)
+    return NULL;
+
+  GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
+
+  oldenc = priv->srtpenc;
+  enc = get_rtp_encoder (stream, session);
+  name = g_strdup_printf ("rtcp_sink_%d", session);
+  pad = gst_element_get_request_pad (enc, name);
+  g_free (name);
+  gst_object_unref (pad);
+
+  if (oldenc == NULL)
+    g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
+        enc);
+
+  return enc;
+}
+
+static GstCaps *
+request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv = stream->priv;
+  GstCaps *caps;
+
+  GST_DEBUG ("request key %08x", ssrc);
+
+  g_mutex_lock (&priv->lock);
+  if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
+    gst_caps_ref (caps);
+  g_mutex_unlock (&priv->lock);
+
+  return caps;
+}
+
+static GstElement *
+request_rtcp_decoder (GstElement * rtpbin, guint session,
+    GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv = stream->priv;
+
+  if (priv->idx != session)
+    return NULL;
+
+  if (priv->srtpdec == NULL) {
+    gchar *name;
+
+    name = g_strdup_printf ("srtpdec_%u", session);
+    priv->srtpdec = gst_element_factory_make ("srtpdec", name);
+    g_free (name);
+
+    g_signal_connect (priv->srtpdec, "request-key",
+        (GCallback) request_key, stream);
+  }
+  return gst_object_ref (priv->srtpdec);
+}
+
+/**
+ * gst_rtsp_stream_join_bin:
+ * @stream: a #GstRTSPStream
+ * @bin: (transfer none): a #GstBin to join
+ * @rtpbin: (transfer none): a rtpbin element in @bin
+ * @state: the target state of the new elements
+ *
+ * Join the #GstBin @bin that contains the element @rtpbin.
+ *
+ * @stream will link to @rtpbin, which must be inside @bin. The elements
+ * added to @bin will be set to the state given in @state.
+ *
+ * Returns: %TRUE on success.
+ */
+gboolean
+gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
+    GstElement * rtpbin, GstState state)
+{
+  GstRTSPStreamPrivate *priv;
+  gint i;
+  guint idx;
+  gchar *name;
+  GstPad *pad, *sinkpad, *selpad;
+  GstPadLinkReturn ret;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+  g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
+  g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  if (priv->is_joined)
+    goto was_joined;
+
+  /* create a session with the same index as the stream */
+  idx = priv->idx;
+
+  GST_INFO ("stream %p joining bin as session %u", stream, idx);
+
+  if (!alloc_ports (stream))
+    goto no_ports;
+
+  /* update the dscp qos field in the sinks */
+  update_dscp_qos (stream);
+
+  if (priv->profiles & GST_RTSP_PROFILE_SAVP
+      || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
+    /* For SRTP */
+    g_signal_connect (rtpbin, "request-rtp-encoder",
+        (GCallback) request_rtp_encoder, stream);
+    g_signal_connect (rtpbin, "request-rtcp-encoder",
+        (GCallback) request_rtcp_encoder, stream);
+    g_signal_connect (rtpbin, "request-rtcp-decoder",
+        (GCallback) request_rtcp_decoder, stream);
+  }
+
+  /* get a pad for sending RTP */
+  name = g_strdup_printf ("send_rtp_sink_%u", idx);
+  priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
+  g_free (name);
+  /* link the RTP pad to the session manager, it should not really fail unless
+   * this is not really an RTP pad */
+  ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
+  if (ret != GST_PAD_LINK_OK)
+    goto link_failed;
+
+  /* get pads from the RTP session element for sending and receiving
+   * RTP/RTCP*/
+  name = g_strdup_printf ("send_rtp_src_%u", idx);
+  priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
+  g_free (name);
+  name = g_strdup_printf ("send_rtcp_src_%u", idx);
+  priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
+  g_free (name);
+  name = g_strdup_printf ("recv_rtp_sink_%u", idx);
+  priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
+  g_free (name);
+  name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
+  priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
+  g_free (name);
+
+  /* get the session */
+  g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
+
+  g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
+      stream);
+  g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
+      stream);
+  g_signal_connect (priv->session, "on-ssrc-active",
+      (GCallback) on_ssrc_active, stream);
+  g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
+      stream);
+  g_signal_connect (priv->session, "on-bye-timeout",
+      (GCallback) on_bye_timeout, stream);
+  g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
+      stream);
+
+  for (i = 0; i < 2; i++) {
+    GstPad *teepad, *queuepad;
+    /* For the sender we create this bit of pipeline for both
+     * RTP and RTCP. Sync and preroll are enabled on udpsink so
+     * we need to add a queue before appsink to make the pipeline
+     * not block. For the TCP case, we want to pump data to the
+     * client as fast as possible anyway.
+     *
+     * .--------.      .-----.    .---------.
+     * | rtpbin |      | tee |    | udpsink |
+     * |       send->sink   src->sink       |
+     * '--------'      |     |    '---------'
+     *                 |     |    .---------.    .---------.
+     *                 |     |    |  queue  |    | appsink |
+     *                 |    src->sink      src->sink       |
+     *                 '-----'    '---------'    '---------'
+     *
+     * When only UDP is allowed, we skip the tee, queue and appsink and link the
+     * udpsink directly to the session.
+     */
+    /* add udpsink */
+    gst_bin_add (bin, priv->udpsink[i]);
+    sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
+
+    if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
+      /* make tee for RTP/RTCP */
+      priv->tee[i] = gst_element_factory_make ("tee", NULL);
+      gst_bin_add (bin, priv->tee[i]);
+
+      /* and link to rtpbin send pad */
+      pad = gst_element_get_static_pad (priv->tee[i], "sink");
+      gst_pad_link (priv->send_src[i], pad);
+      gst_object_unref (pad);
+
+      /* link tee to udpsink */
+      teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
+      gst_pad_link (teepad, sinkpad);
+      gst_object_unref (teepad);
+
+      /* make queue */
+      priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
+      gst_bin_add (bin, priv->appqueue[i]);
+      /* and link to tee */
+      teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
+      pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
+      gst_pad_link (teepad, pad);
+      gst_object_unref (pad);
+      gst_object_unref (teepad);
+
+      /* make appsink */
+      priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
+      g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
+      g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
+      gst_bin_add (bin, priv->appsink[i]);
+      gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
+          &sink_cb, stream, NULL);
+      /* and link to queue */
+      queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
+      pad = gst_element_get_static_pad (priv->appsink[i], "sink");
+      gst_pad_link (queuepad, pad);
+      gst_object_unref (pad);
+      gst_object_unref (queuepad);
+    } else {
+      /* else only udpsink needed, link it to the session */
+      gst_pad_link (priv->send_src[i], sinkpad);
+    }
+    gst_object_unref (sinkpad);
+
+    /* For the receiver we create this bit of pipeline for both
+     * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
+     * and it is all funneled into the rtpbin receive pad.
+     *
+     * .--------.     .--------.    .--------.
+     * | udpsrc |     | funnel |    | rtpbin |
+     * |       src->sink      src->sink      |
+     * '--------'     |        |    '--------'
+     * .--------.     |        |
+     * | appsrc |     |        |
+     * |       src->sink       |
+     * '--------'     '--------'
+     */
+    /* make funnel for the RTP/RTCP receivers */
+    priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
+    gst_bin_add (bin, priv->funnel[i]);
+
+    pad = gst_element_get_static_pad (priv->funnel[i], "src");
+    gst_pad_link (pad, priv->recv_sink[i]);
+    gst_object_unref (pad);
+
+    if (priv->udpsrc_v4[i]) {
+      /* we set and keep these to playing so that they don't cause NO_PREROLL return
+       * values */
+      gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
+      gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
+      /* add udpsrc */
+      gst_bin_add (bin, priv->udpsrc_v4[i]);
+
+      /* and link to the funnel v4 */
+      selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
+      pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
+      gst_pad_link (pad, selpad);
+      gst_object_unref (pad);
+      gst_object_unref (selpad);
+    }
+
+    if (priv->udpsrc_v6[i]) {
+      gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
+      gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
+      gst_bin_add (bin, priv->udpsrc_v6[i]);
+
+      /* and link to the funnel v6 */
+      selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
+      pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
+      gst_pad_link (pad, selpad);
+      gst_object_unref (pad);
+      gst_object_unref (selpad);
+    }
+
+    if (priv->protocols & GST_RTSP_LOWER_TRANS_TCP) {
+      /* make and add appsrc */
+      priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
+      gst_bin_add (bin, priv->appsrc[i]);
+      /* and link to the funnel */
+      selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
+      pad = gst_element_get_static_pad (priv->appsrc[i], "src");
+      gst_pad_link (pad, selpad);
+      gst_object_unref (pad);
+      gst_object_unref (selpad);
+    }
+
+    /* check if we need to set to a special state */
+    if (state != GST_STATE_NULL) {
+      if (priv->udpsink[i])
+        gst_element_set_state (priv->udpsink[i], state);
+      if (priv->appsink[i])
+        gst_element_set_state (priv->appsink[i], state);
+      if (priv->appqueue[i])
+        gst_element_set_state (priv->appqueue[i], state);
+      if (priv->tee[i])
+        gst_element_set_state (priv->tee[i], state);
+      if (priv->funnel[i])
+        gst_element_set_state (priv->funnel[i], state);
+      if (priv->appsrc[i])
+        gst_element_set_state (priv->appsrc[i], state);
+    }
+  }
+
+  /* be notified of caps changes */
+  priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
+      (GCallback) caps_notify, stream);
+
+  priv->is_joined = TRUE;
+  g_mutex_unlock (&priv->lock);
+
+  return TRUE;
+
+  /* ERRORS */
+was_joined:
+  {
+    g_mutex_unlock (&priv->lock);
+    return TRUE;
+  }
+no_ports:
+  {
+    g_mutex_unlock (&priv->lock);
+    GST_WARNING ("failed to allocate ports %u", idx);
+    return FALSE;
+  }
+link_failed:
+  {
+    GST_WARNING ("failed to link stream %u", idx);
+    gst_object_unref (priv->send_rtp_sink);
+    priv->send_rtp_sink = NULL;
+    g_mutex_unlock (&priv->lock);
+    return FALSE;
+  }
+}
+
+/**
+ * gst_rtsp_stream_leave_bin:
+ * @stream: a #GstRTSPStream
+ * @bin: (transfer none): a #GstBin
+ * @rtpbin: (transfer none): a rtpbin #GstElement
+ *
+ * Remove the elements of @stream from @bin.
+ *
+ * Return: %TRUE on success.
+ */
+gboolean
+gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
+    GstElement * rtpbin)
+{
+  GstRTSPStreamPrivate *priv;
+  gint i;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+  g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
+  g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  if (!priv->is_joined)
+    goto was_not_joined;
+
+  /* all transports must be removed by now */
+  g_return_val_if_fail (priv->transports == NULL, FALSE);
+
+  clear_tr_cache (priv);
+
+  GST_INFO ("stream %p leaving bin", stream);
+
+  gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
+  g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
+  gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
+  gst_object_unref (priv->send_rtp_sink);
+  priv->send_rtp_sink = NULL;
+
+  for (i = 0; i < 2; i++) {
+    if (priv->udpsink[i])
+      gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
+    if (priv->appsink[i])
+      gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
+    if (priv->appqueue[i])
+      gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
+    if (priv->tee[i])
+      gst_element_set_state (priv->tee[i], GST_STATE_NULL);
+    if (priv->funnel[i])
+      gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
+    if (priv->appsrc[i])
+      gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
+    if (priv->udpsrc_v4[i]) {
+      /* and set udpsrc to NULL now before removing */
+      gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
+      gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
+      /* removing them should also nicely release the request
+       * pads when they finalize */
+      gst_bin_remove (bin, priv->udpsrc_v4[i]);
+    }
+    if (priv->udpsrc_v6[i]) {
+      gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
+      gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
+      gst_bin_remove (bin, priv->udpsrc_v6[i]);
+    }
+    if (priv->udpsink[i])
+      gst_bin_remove (bin, priv->udpsink[i]);
+    if (priv->appsrc[i])
+      gst_bin_remove (bin, priv->appsrc[i]);
+    if (priv->appsink[i])
+      gst_bin_remove (bin, priv->appsink[i]);
+    if (priv->appqueue[i])
+      gst_bin_remove (bin, priv->appqueue[i]);
+    if (priv->tee[i])
+      gst_bin_remove (bin, priv->tee[i]);
+    if (priv->funnel[i])
+      gst_bin_remove (bin, priv->funnel[i]);
+
+    gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
+    gst_object_unref (priv->recv_sink[i]);
+    priv->recv_sink[i] = NULL;
+
+    priv->udpsrc_v4[i] = NULL;
+    priv->udpsrc_v6[i] = NULL;
+    priv->udpsink[i] = NULL;
+    priv->appsrc[i] = NULL;
+    priv->appsink[i] = NULL;
+    priv->appqueue[i] = NULL;
+    priv->tee[i] = NULL;
+    priv->funnel[i] = NULL;
+  }
+  gst_object_unref (priv->send_src[0]);
+  priv->send_src[0] = NULL;
+
+  gst_element_release_request_pad (rtpbin, priv->send_src[1]);
+  gst_object_unref (priv->send_src[1]);
+  priv->send_src[1] = NULL;
+
+  g_object_unref (priv->session);
+  priv->session = NULL;
+  if (priv->caps)
+    gst_caps_unref (priv->caps);
+  priv->caps = NULL;
+
+  if (priv->srtpenc)
+    gst_object_unref (priv->srtpenc);
+  if (priv->srtpdec)
+    gst_object_unref (priv->srtpdec);
+
+  priv->is_joined = FALSE;
+  g_mutex_unlock (&priv->lock);
+
+  return TRUE;
+
+was_not_joined:
+  {
+    g_mutex_unlock (&priv->lock);
+    return TRUE;
+  }
+}
+
+/**
+ * gst_rtsp_stream_get_rtpinfo:
+ * @stream: a #GstRTSPStream
+ * @rtptime: (allow-none): result RTP timestamp
+ * @seq: (allow-none): result RTP seqnum
+ * @clock_rate: (allow-none): the clock rate
+ * @running_time: (allow-none): result running-time
+ *
+ * Retrieve the current rtptime, seq and running-time. This is used to
+ * construct a RTPInfo reply header.
+ *
+ * Returns: %TRUE when rtptime, seq and running-time could be determined.
+ */
+gboolean
+gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
+    guint * rtptime, guint * seq, guint * clock_rate,
+    GstClockTime * running_time)
+{
+  GstRTSPStreamPrivate *priv;
+  GstStructure *stats;
+  GObjectClass *payobjclass;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+  priv = stream->priv;
+
+  payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
+
+  g_mutex_lock (&priv->lock);
+
+  if (g_object_class_find_property (payobjclass, "stats")) {
+    g_object_get (priv->payloader, "stats", &stats, NULL);
+    if (stats == NULL)
+      goto no_stats;
+
+    if (seq)
+      gst_structure_get_uint (stats, "seqnum", seq);
+
+    if (rtptime)
+      gst_structure_get_uint (stats, "timestamp", rtptime);
+
+    if (running_time)
+      gst_structure_get_clock_time (stats, "running-time", running_time);
+
+    if (clock_rate) {
+      gst_structure_get_uint (stats, "clock-rate", clock_rate);
+      if (*clock_rate == 0 && running_time)
+        *running_time = GST_CLOCK_TIME_NONE;
+    }
+    gst_structure_free (stats);
+  } else {
+    if (!g_object_class_find_property (payobjclass, "seqnum") ||
+        !g_object_class_find_property (payobjclass, "timestamp"))
+      goto no_stats;
+
+    if (seq)
+      g_object_get (priv->payloader, "seqnum", seq, NULL);
+
+    if (rtptime)
+      g_object_get (priv->payloader, "timestamp", rtptime, NULL);
+
+    if (running_time)
+      *running_time = GST_CLOCK_TIME_NONE;
+  }
+  g_mutex_unlock (&priv->lock);
+
+  return TRUE;
+
+  /* ERRORS */
+no_stats:
+  {
+    GST_WARNING ("Could not get payloader stats");
+    g_mutex_unlock (&priv->lock);
+    return FALSE;
+  }
+}
+
+/**
+ * gst_rtsp_stream_get_caps:
+ * @stream: a #GstRTSPStream
+ *
+ * Retrieve the current caps of @stream.
+ *
+ * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
+ * after usage.
+ */
+GstCaps *
+gst_rtsp_stream_get_caps (GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv;
+  GstCaps *result;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  if ((result = priv->caps))
+    gst_caps_ref (result);
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_stream_recv_rtp:
+ * @stream: a #GstRTSPStream
+ * @buffer: (transfer full): a #GstBuffer
+ *
+ * Handle an RTP buffer for the stream. This method is usually called when a
+ * message has been received from a client using the TCP transport.
+ *
+ * This function takes ownership of @buffer.
+ *
+ * Returns: a GstFlowReturn.
+ */
+GstFlowReturn
+gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
+{
+  GstRTSPStreamPrivate *priv;
+  GstFlowReturn ret;
+  GstElement *element;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
+  priv = stream->priv;
+  g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
+  g_return_val_if_fail (priv->is_joined, FALSE);
+
+  g_mutex_lock (&priv->lock);
+  if (priv->appsrc[0])
+    element = gst_object_ref (priv->appsrc[0]);
+  else
+    element = NULL;
+  g_mutex_unlock (&priv->lock);
+
+  if (element) {
+    ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
+    gst_object_unref (element);
+  } else {
+    ret = GST_FLOW_OK;
+  }
+  return ret;
+}
+
+/**
+ * gst_rtsp_stream_recv_rtcp:
+ * @stream: a #GstRTSPStream
+ * @buffer: (transfer full): a #GstBuffer
+ *
+ * Handle an RTCP buffer for the stream. This method is usually called when a
+ * message has been received from a client using the TCP transport.
+ *
+ * This function takes ownership of @buffer.
+ *
+ * Returns: a GstFlowReturn.
+ */
+GstFlowReturn
+gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
+{
+  GstRTSPStreamPrivate *priv;
+  GstFlowReturn ret;
+  GstElement *element;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
+  priv = stream->priv;
+  g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
+  g_return_val_if_fail (priv->is_joined, FALSE);
+
+  g_mutex_lock (&priv->lock);
+  if (priv->appsrc[1])
+    element = gst_object_ref (priv->appsrc[1]);
+  else
+    element = NULL;
+  g_mutex_unlock (&priv->lock);
+
+  if (element) {
+    ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
+    gst_object_unref (element);
+  } else {
+    ret = GST_FLOW_OK;
+    gst_buffer_unref (buffer);
+  }
+  return ret;
+}
+
+/* must be called with lock */
+static gboolean
+update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
+    gboolean add)
+{
+  GstRTSPStreamPrivate *priv = stream->priv;
+  const GstRTSPTransport *tr;
+
+  tr = gst_rtsp_stream_transport_get_transport (trans);
+
+  switch (tr->lower_transport) {
+    case GST_RTSP_LOWER_TRANS_UDP:
+    case GST_RTSP_LOWER_TRANS_UDP_MCAST:
+    {
+      gchar *dest;
+      gint min, max;
+      guint ttl = 0;
+
+      dest = tr->destination;
+      if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
+        min = tr->port.min;
+        max = tr->port.max;
+        ttl = tr->ttl;
+      } else {
+        min = tr->client_port.min;
+        max = tr->client_port.max;
+      }
+
+      if (add) {
+        if (ttl > 0) {
+          GST_INFO ("setting ttl-mc %d", ttl);
+          g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
+          g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
+        }
+        GST_INFO ("adding %s:%d-%d", dest, min, max);
+        g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
+        g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
+        priv->transports = g_list_prepend (priv->transports, trans);
+      } else {
+        GST_INFO ("removing %s:%d-%d", dest, min, max);
+        g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
+        g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
+        priv->transports = g_list_remove (priv->transports, trans);
+      }
+      priv->transports_cookie++;
+      break;
+    }
+    case GST_RTSP_LOWER_TRANS_TCP:
+      if (add) {
+        GST_INFO ("adding TCP %s", tr->destination);
+        priv->transports = g_list_prepend (priv->transports, trans);
+      } else {
+        GST_INFO ("removing TCP %s", tr->destination);
+        priv->transports = g_list_remove (priv->transports, trans);
+      }
+      priv->transports_cookie++;
+      break;
+    default:
+      goto unknown_transport;
+  }
+  return TRUE;
+
+  /* ERRORS */
+unknown_transport:
+  {
+    GST_INFO ("Unknown transport %d", tr->lower_transport);
+    return FALSE;
+  }
+}
+
+
+/**
+ * gst_rtsp_stream_add_transport:
+ * @stream: a #GstRTSPStream
+ * @trans: (transfer none): a #GstRTSPStreamTransport
+ *
+ * Add the transport in @trans to @stream. The media of @stream will
+ * then also be send to the values configured in @trans.
+ *
+ * @stream must be joined to a bin.
+ *
+ * @trans must contain a valid #GstRTSPTransport.
+ *
+ * Returns: %TRUE if @trans was added
+ */
+gboolean
+gst_rtsp_stream_add_transport (GstRTSPStream * stream,
+    GstRTSPStreamTransport * trans)
+{
+  GstRTSPStreamPrivate *priv;
+  gboolean res;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+  priv = stream->priv;
+  g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
+  g_return_val_if_fail (priv->is_joined, FALSE);
+
+  g_mutex_lock (&priv->lock);
+  res = update_transport (stream, trans, TRUE);
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_stream_remove_transport:
+ * @stream: a #GstRTSPStream
+ * @trans: (transfer none): a #GstRTSPStreamTransport
+ *
+ * Remove the transport in @trans from @stream. The media of @stream will
+ * not be sent to the values configured in @trans.
+ *
+ * @stream must be joined to a bin.
+ *
+ * @trans must contain a valid #GstRTSPTransport.
+ *
+ * Returns: %TRUE if @trans was removed
+ */
+gboolean
+gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
+    GstRTSPStreamTransport * trans)
+{
+  GstRTSPStreamPrivate *priv;
+  gboolean res;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+  priv = stream->priv;
+  g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
+  g_return_val_if_fail (priv->is_joined, FALSE);
+
+  g_mutex_lock (&priv->lock);
+  res = update_transport (stream, trans, FALSE);
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+/**
+ * gst_rtsp_stream_update_crypto:
+ * @stream: a #GstRTSPStream
+ * @ssrc: the SSRC
+ * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
+ *
+ * Update the new crypto information for @ssrc in @stream. If information
+ * for @ssrc did not exist, it will be added. If information
+ * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
+ * be removed from @stream.
+ *
+ * Returns: %TRUE if @crypto could be updated
+ */
+gboolean
+gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
+    guint ssrc, GstCaps * crypto)
+{
+  GstRTSPStreamPrivate *priv;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+  g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
+
+  priv = stream->priv;
+
+  GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
+
+  g_mutex_lock (&priv->lock);
+  if (crypto)
+    g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
+        gst_caps_ref (crypto));
+  else
+    g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
+  g_mutex_unlock (&priv->lock);
+
+  return TRUE;
+}
+
+/**
+ * gst_rtsp_stream_get_rtp_socket:
+ * @stream: a #GstRTSPStream
+ * @family: the socket family
+ *
+ * Get the RTP socket from @stream for a @family.
+ *
+ * @stream must be joined to a bin.
+ *
+ * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
+ * socket could be allocated for @family. Unref after usage
+ */
+GSocket *
+gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
+{
+  GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
+  GSocket *socket;
+  const gchar *name;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+  g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
+      family == G_SOCKET_FAMILY_IPV6, NULL);
+  g_return_val_if_fail (priv->udpsink[0], NULL);
+
+  if (family == G_SOCKET_FAMILY_IPV6)
+    name = "socket-v6";
+  else
+    name = "socket";
+
+  g_object_get (priv->udpsink[0], name, &socket, NULL);
+
+  return socket;
+}
+
+/**
+ * gst_rtsp_stream_get_rtcp_socket:
+ * @stream: a #GstRTSPStream
+ * @family: the socket family
+ *
+ * Get the RTCP socket from @stream for a @family.
+ *
+ * @stream must be joined to a bin.
+ *
+ * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
+ * socket could be allocated for @family. Unref after usage
+ */
+GSocket *
+gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
+{
+  GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
+  GSocket *socket;
+  const gchar *name;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+  g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
+      family == G_SOCKET_FAMILY_IPV6, NULL);
+  g_return_val_if_fail (priv->udpsink[1], NULL);
+
+  if (family == G_SOCKET_FAMILY_IPV6)
+    name = "socket-v6";
+  else
+    name = "socket";
+
+  g_object_get (priv->udpsink[1], name, &socket, NULL);
+
+  return socket;
+}
+
+/**
+ * gst_rtsp_stream_set_seqnum:
+ * @stream: a #GstRTSPStream
+ * @seqnum: a new sequence number
+ *
+ * Configure the sequence number in the payloader of @stream to @seqnum.
+ */
+void
+gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
+{
+  GstRTSPStreamPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_STREAM (stream));
+
+  priv = stream->priv;
+
+  g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
+}
+
+/**
+ * gst_rtsp_stream_get_seqnum:
+ * @stream: a #GstRTSPStream
+ *
+ * Get the configured sequence number in the payloader of @stream.
+ *
+ * Returns: the sequence number of the payloader.
+ */
+guint16
+gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv;
+  guint seqnum;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
+
+  priv = stream->priv;
+
+  g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
+
+  return seqnum;
+}
+
+guint64
+gst_rtsp_stream_get_udp_sent_bytes (GstRTSPStream *stream)
+{
+  GstRTSPStreamPrivate *priv;
+  guint64 bytes = 0;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
+
+  priv = stream->priv;
+
+  g_object_get (G_OBJECT (priv->udpsink[0]), "bytes-to-serve", &bytes, NULL);
+
+  return bytes;
+}
+
+/**
+ * gst_rtsp_stream_transport_filter:
+ * @stream: a #GstRTSPStream
+ * @func: (scope call) (allow-none): a callback
+ * @user_data: (closure): user data passed to @func
+ *
+ * Call @func for each transport managed by @stream. The result value of @func
+ * determines what happens to the transport. @func will be called with @stream
+ * locked so no further actions on @stream can be performed from @func.
+ *
+ * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
+ * @stream.
+ *
+ * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
+ *
+ * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
+ * will also be added with an additional ref to the result #GList of this
+ * function..
+ *
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
+ *
+ * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
+ * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
+ * element in the #GList should be unreffed before the list is freed.
+ */
+GList *
+gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
+    GstRTSPStreamTransportFilterFunc func, gpointer user_data)
+{
+  GstRTSPStreamPrivate *priv;
+  GList *result, *walk, *next;
+  GHashTable *visited;
+  guint cookie;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
+
+  priv = stream->priv;
+
+  result = NULL;
+  if (func)
+    visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
+
+  g_mutex_lock (&priv->lock);
+restart:
+  cookie = priv->transports_cookie;
+  for (walk = priv->transports; walk; walk = next) {
+    GstRTSPStreamTransport *trans = walk->data;
+    GstRTSPFilterResult res;
+    gboolean changed;
+
+    next = g_list_next (walk);
+
+    if (func) {
+      /* only visit each transport once */
+      if (g_hash_table_contains (visited, trans))
+        continue;
+
+      g_hash_table_add (visited, g_object_ref (trans));
+      g_mutex_unlock (&priv->lock);
+
+      res = func (stream, trans, user_data);
+
+      g_mutex_lock (&priv->lock);
+    } else
+      res = GST_RTSP_FILTER_REF;
+
+    changed = (cookie != priv->transports_cookie);
+
+    switch (res) {
+      case GST_RTSP_FILTER_REMOVE:
+        update_transport (stream, trans, FALSE);
+        break;
+      case GST_RTSP_FILTER_REF:
+        result = g_list_prepend (result, g_object_ref (trans));
+        break;
+      case GST_RTSP_FILTER_KEEP:
+      default:
+        break;
+    }
+    if (changed)
+      goto restart;
+  }
+  g_mutex_unlock (&priv->lock);
+
+  if (func)
+    g_hash_table_unref (visited);
+
+  return result;
+}
+
+static GstPadProbeReturn
+pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
+{
+  GstRTSPStreamPrivate *priv;
+  GstRTSPStream *stream;
+
+  stream = user_data;
+  priv = stream->priv;
+
+  GST_DEBUG_OBJECT (pad, "now blocking");
+
+  g_mutex_lock (&priv->lock);
+  priv->blocking = TRUE;
+  g_mutex_unlock (&priv->lock);
+
+  gst_element_post_message (priv->payloader,
+      gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
+          gst_structure_new_empty ("GstRTSPStreamBlocking")));
+
+  return GST_PAD_PROBE_OK;
+}
+
+/**
+ * gst_rtsp_stream_set_blocked:
+ * @stream: a #GstRTSPStream
+ * @blocked: boolean indicating we should block or unblock
+ *
+ * Blocks or unblocks the dataflow on @stream.
+ *
+ * Returns: %TRUE on success
+ */
+gboolean
+gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
+{
+  GstRTSPStreamPrivate *priv;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  if (blocked) {
+    priv->blocking = FALSE;
+    if (priv->blocked_id == 0) {
+      priv->blocked_id = gst_pad_add_probe (priv->srcpad,
+          GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
+          GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
+          g_object_ref (stream), g_object_unref);
+    }
+  } else {
+    if (priv->blocked_id != 0) {
+      gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
+      priv->blocked_id = 0;
+      priv->blocking = FALSE;
+    }
+  }
+  g_mutex_unlock (&priv->lock);
+
+  return TRUE;
+}
+
+/**
+ * gst_rtsp_stream_is_blocking:
+ * @stream: a #GstRTSPStream
+ *
+ * Check if @stream is blocking on a #GstBuffer.
+ *
+ * Returns: %TRUE if @stream is blocking
+ */
+gboolean
+gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
+{
+  GstRTSPStreamPrivate *priv;
+  gboolean result;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  result = priv->blocking;
+  g_mutex_unlock (&priv->lock);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_stream_query_position:
+ * @stream: a #GstRTSPStream
+ *
+ * Query the position of the stream in %GST_FORMAT_TIME. This only considers
+ * the RTP parts of the pipeline and not the RTCP parts.
+ *
+ * Returns: %TRUE if the position could be queried
+ */
+gboolean
+gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
+{
+  GstRTSPStreamPrivate *priv;
+  GstElement *sink;
+  gboolean ret;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  if ((sink = priv->udpsink[0]))
+    gst_object_ref (sink);
+  g_mutex_unlock (&priv->lock);
+
+  if (!sink)
+    return FALSE;
+
+  ret = gst_element_query_position (sink, GST_FORMAT_TIME, position);
+  gst_object_unref (sink);
+
+  return ret;
+}
+
+/**
+ * gst_rtsp_stream_query_stop:
+ * @stream: a #GstRTSPStream
+ *
+ * Query the stop of the stream in %GST_FORMAT_TIME. This only considers
+ * the RTP parts of the pipeline and not the RTCP parts.
+ *
+ * Returns: %TRUE if the stop could be queried
+ */
+gboolean
+gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
+{
+  GstRTSPStreamPrivate *priv;
+  GstElement *sink;
+  GstQuery *query;
+  gboolean ret;
+
+  g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
+
+  priv = stream->priv;
+
+  g_mutex_lock (&priv->lock);
+  if ((sink = priv->udpsink[0]))
+    gst_object_ref (sink);
+  g_mutex_unlock (&priv->lock);
+
+  if (!sink)
+    return FALSE;
+
+  query = gst_query_new_segment (GST_FORMAT_TIME);
+  if ((ret = gst_element_query (sink, query))) {
+    GstFormat format;
+
+    gst_query_parse_segment (query, NULL, &format, NULL, stop);
+    if (format != GST_FORMAT_TIME)
+      *stop = -1;
+  }
+  gst_query_unref (query);
+  gst_object_unref (sink);
+
+  return ret;
+
+}
diff --git a/gst/rtsp-server/rtsp-stream.h b/gst/rtsp-server/rtsp-stream.h
new file mode 100644
index 0000000..dddbbf4
--- /dev/null
+++ b/gst/rtsp-server/rtsp-stream.h
@@ -0,0 +1,190 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+#include 
+#include 
+#include 
+
+#ifndef __GST_RTSP_STREAM_H__
+#define __GST_RTSP_STREAM_H__
+
+G_BEGIN_DECLS
+
+/* types for the media stream */
+#define GST_TYPE_RTSP_STREAM              (gst_rtsp_stream_get_type ())
+#define GST_IS_RTSP_STREAM(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_STREAM))
+#define GST_IS_RTSP_STREAM_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_STREAM))
+#define GST_RTSP_STREAM_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamClass))
+#define GST_RTSP_STREAM(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStream))
+#define GST_RTSP_STREAM_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_STREAM, GstRTSPStreamClass))
+#define GST_RTSP_STREAM_CAST(obj)         ((GstRTSPStream*)(obj))
+#define GST_RTSP_STREAM_CLASS_CAST(klass) ((GstRTSPStreamClass*)(klass))
+
+typedef struct _GstRTSPStream GstRTSPStream;
+typedef struct _GstRTSPStreamClass GstRTSPStreamClass;
+typedef struct _GstRTSPStreamPrivate GstRTSPStreamPrivate;
+
+#include "rtsp-stream-transport.h"
+#include "rtsp-address-pool.h"
+#include "rtsp-session.h"
+
+/**
+ * GstRTSPStream:
+ *
+ * The definition of a media stream.
+ */
+struct _GstRTSPStream {
+  GObject       parent;
+
+  /*< private >*/
+  GstRTSPStreamPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];};
+
+struct _GstRTSPStreamClass {
+  GObjectClass parent_class;
+
+  /*< private >*/
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+GType             gst_rtsp_stream_get_type         (void);
+
+GstRTSPStream *   gst_rtsp_stream_new              (guint idx, GstElement *payloader,
+                                                    GstPad *srcpad);
+guint             gst_rtsp_stream_get_index        (GstRTSPStream *stream);
+guint             gst_rtsp_stream_get_pt           (GstRTSPStream *stream);
+GstPad *          gst_rtsp_stream_get_srcpad       (GstRTSPStream *stream);
+
+void              gst_rtsp_stream_set_control      (GstRTSPStream *stream, const gchar *control);
+gchar *           gst_rtsp_stream_get_control      (GstRTSPStream *stream);
+gboolean          gst_rtsp_stream_has_control      (GstRTSPStream *stream, const gchar *control);
+
+void              gst_rtsp_stream_set_mtu          (GstRTSPStream *stream, guint mtu);
+guint             gst_rtsp_stream_get_mtu          (GstRTSPStream *stream);
+
+void              gst_rtsp_stream_set_dscp_qos     (GstRTSPStream *stream, gint dscp_qos);
+gint              gst_rtsp_stream_get_dscp_qos     (GstRTSPStream *stream);
+
+gboolean          gst_rtsp_stream_is_transport_supported  (GstRTSPStream *stream,
+                                                           GstRTSPTransport *transport);
+
+void              gst_rtsp_stream_set_profiles     (GstRTSPStream *stream, GstRTSPProfile profiles);
+GstRTSPProfile    gst_rtsp_stream_get_profiles     (GstRTSPStream *stream);
+
+void              gst_rtsp_stream_set_protocols    (GstRTSPStream *stream, GstRTSPLowerTrans protocols);
+GstRTSPLowerTrans gst_rtsp_stream_get_protocols    (GstRTSPStream *stream);
+
+void              gst_rtsp_stream_set_address_pool (GstRTSPStream *stream, GstRTSPAddressPool *pool);
+GstRTSPAddressPool *
+                  gst_rtsp_stream_get_address_pool (GstRTSPStream *stream);
+
+GstRTSPAddress *  gst_rtsp_stream_reserve_address  (GstRTSPStream *stream,
+                                                    const gchar * address,
+                                                    guint port,
+                                                    guint n_ports,
+                                                    guint ttl);
+
+gboolean          gst_rtsp_stream_join_bin         (GstRTSPStream *stream,
+                                                    GstBin *bin, GstElement *rtpbin,
+                                                    GstState state);
+gboolean          gst_rtsp_stream_leave_bin        (GstRTSPStream *stream,
+                                                    GstBin *bin, GstElement *rtpbin);
+
+gboolean          gst_rtsp_stream_set_blocked      (GstRTSPStream * stream,
+                                                    gboolean blocked);
+gboolean          gst_rtsp_stream_is_blocking      (GstRTSPStream * stream);
+
+void              gst_rtsp_stream_get_server_port  (GstRTSPStream *stream,
+                                                    GstRTSPRange *server_port,
+                                                    GSocketFamily family);
+GstRTSPAddress *  gst_rtsp_stream_get_multicast_address (GstRTSPStream *stream,
+                                                         GSocketFamily family);
+
+
+GObject *         gst_rtsp_stream_get_rtpsession   (GstRTSPStream *stream);
+
+void              gst_rtsp_stream_get_ssrc         (GstRTSPStream *stream,
+                                                    guint *ssrc);
+
+gboolean          gst_rtsp_stream_get_rtpinfo      (GstRTSPStream *stream,
+                                                    guint *rtptime, guint *seq,
+                                                    guint *clock_rate,
+                                                    GstClockTime *running_time);
+GstCaps *         gst_rtsp_stream_get_caps         (GstRTSPStream *stream);
+
+GstFlowReturn     gst_rtsp_stream_recv_rtp         (GstRTSPStream *stream,
+                                                    GstBuffer *buffer);
+GstFlowReturn     gst_rtsp_stream_recv_rtcp        (GstRTSPStream *stream,
+                                                    GstBuffer *buffer);
+
+gboolean          gst_rtsp_stream_add_transport    (GstRTSPStream *stream,
+                                                    GstRTSPStreamTransport *trans);
+gboolean          gst_rtsp_stream_remove_transport (GstRTSPStream *stream,
+                                                    GstRTSPStreamTransport *trans);
+
+GSocket *         gst_rtsp_stream_get_rtp_socket   (GstRTSPStream *stream,
+                                                    GSocketFamily family);
+GSocket *         gst_rtsp_stream_get_rtcp_socket  (GstRTSPStream *stream,
+                                                    GSocketFamily family);
+
+gboolean          gst_rtsp_stream_update_crypto    (GstRTSPStream * stream,
+                                                    guint ssrc, GstCaps * crypto);
+
+
+gboolean          gst_rtsp_stream_query_position   (GstRTSPStream * stream,
+                                                    gint64 * position);
+gboolean          gst_rtsp_stream_query_stop       (GstRTSPStream * stream,
+                                                    gint64 * stop);
+
+void              gst_rtsp_stream_set_seqnum_offset   (GstRTSPStream *stream, guint16 seqnum);
+guint16           gst_rtsp_stream_get_current_seqnum  (GstRTSPStream *stream);
+guint64           gst_rtsp_stream_get_udp_sent_bytes  (GstRTSPStream *stream);
+
+/**
+ * GstRTSPStreamTransportFilterFunc:
+ * @stream: a #GstRTSPStream object
+ * @trans: a #GstRTSPStreamTransport in @stream
+ * @user_data: user data that has been given to gst_rtsp_stream_transport_filter()
+ *
+ * This function will be called by the gst_rtsp_stream_transport_filter(). An
+ * implementation should return a value of #GstRTSPFilterResult.
+ *
+ * When this function returns #GST_RTSP_FILTER_REMOVE, @trans will be removed
+ * from @stream.
+ *
+ * A return value of #GST_RTSP_FILTER_KEEP will leave @trans untouched in
+ * @stream.
+ *
+ * A value of #GST_RTSP_FILTER_REF will add @trans to the result #GList of
+ * gst_rtsp_stream_transport_filter().
+ *
+ * Returns: a #GstRTSPFilterResult.
+ */
+typedef GstRTSPFilterResult (*GstRTSPStreamTransportFilterFunc) (GstRTSPStream *stream,
+                                                                 GstRTSPStreamTransport *trans,
+                                                                 gpointer user_data);
+
+GList *                gst_rtsp_stream_transport_filter  (GstRTSPStream *stream,
+                                                          GstRTSPStreamTransportFilterFunc func,
+                                                          gpointer user_data);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_STREAM_H__ */
diff --git a/gst/rtsp-server/rtsp-thread-pool.c b/gst/rtsp-server/rtsp-thread-pool.c
new file mode 100644
index 0000000..a8cce78
--- /dev/null
+++ b/gst/rtsp-server/rtsp-thread-pool.c
@@ -0,0 +1,561 @@
+/* GStreamer
+ * Copyright (C) 2013 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-thread-pool
+ * @short_description: A pool of threads
+ * @see_also: #GstRTSPMedia, #GstRTSPClient
+ *
+ * A #GstRTSPThreadPool manages reusable threads for various server tasks.
+ * Currently the defined thread types can be found in #GstRTSPThreadType.
+ *
+ * Threads of type #GST_RTSP_THREAD_TYPE_CLIENT are used to handle requests from
+ * a connected client. With gst_rtsp_thread_pool_get_max_threads() a maximum
+ * number of threads can be set after which the pool will start to reuse the
+ * same thread for multiple clients.
+ *
+ * Threads of type #GST_RTSP_THREAD_TYPE_MEDIA will be used to perform the state
+ * changes of the media pipelines and handle its bus messages.
+ *
+ * gst_rtsp_thread_pool_get_thread() can be used to create a #GstRTSPThread
+ * object of the right type. The thread object contains a mainloop and context
+ * that run in a seperate thread and can be used to attached sources to.
+ *
+ * gst_rtsp_thread_reuse() can be used to reuse a thread for multiple purposes.
+ * If all gst_rtsp_thread_reuse() calls are matched with a
+ * gst_rtsp_thread_stop() call, the mainloop will be quit and the thread will
+ * stop.
+ *
+ * To configure the threads, a subclass of this object should be made and the
+ * virtual methods should be overriden to implement the desired functionality.
+ *
+ * Last reviewed on 2013-07-11 (1.0.0)
+ */
+
+#include 
+
+#include "rtsp-thread-pool.h"
+
+typedef struct _GstRTSPThreadImpl
+{
+  GstRTSPThread thread;
+
+  gint reused;
+  GSource *source;
+} GstRTSPThreadImpl;
+
+GST_DEFINE_MINI_OBJECT_TYPE (GstRTSPThread, gst_rtsp_thread);
+
+static void gst_rtsp_thread_init (GstRTSPThreadImpl * impl);
+
+static void
+_gst_rtsp_thread_free (GstRTSPThreadImpl * impl)
+{
+  GST_DEBUG ("free thread %p", impl);
+
+  g_source_unref (impl->source);
+  g_main_loop_unref (impl->thread.loop);
+  g_main_context_unref (impl->thread.context);
+  g_slice_free1 (sizeof (GstRTSPThreadImpl), impl);
+}
+
+static GstRTSPThread *
+_gst_rtsp_thread_copy (GstRTSPThreadImpl * impl)
+{
+  GstRTSPThreadImpl *copy;
+
+  GST_DEBUG ("copy thread %p", impl);
+
+  copy = g_slice_new0 (GstRTSPThreadImpl);
+  gst_rtsp_thread_init (copy);
+  copy->thread.context = g_main_context_ref (impl->thread.context);
+  copy->thread.loop = g_main_loop_ref (impl->thread.loop);
+
+  return GST_RTSP_THREAD (copy);
+}
+
+static void
+gst_rtsp_thread_init (GstRTSPThreadImpl * impl)
+{
+  gst_mini_object_init (GST_MINI_OBJECT_CAST (impl), 0,
+      GST_TYPE_RTSP_THREAD,
+      (GstMiniObjectCopyFunction) _gst_rtsp_thread_copy, NULL,
+      (GstMiniObjectFreeFunction) _gst_rtsp_thread_free);
+
+  g_atomic_int_set (&impl->reused, 1);
+}
+
+/**
+ * gst_rtsp_thread_new:
+ * @type: the thread type
+ *
+ * Create a new thread object that can run a mainloop.
+ *
+ * Returns: (transfer full): a #GstRTSPThread.
+ */
+GstRTSPThread *
+gst_rtsp_thread_new (GstRTSPThreadType type)
+{
+  GstRTSPThreadImpl *impl;
+
+  impl = g_slice_new0 (GstRTSPThreadImpl);
+
+  gst_rtsp_thread_init (impl);
+  impl->thread.type = type;
+  impl->thread.context = g_main_context_new ();
+  impl->thread.loop = g_main_loop_new (impl->thread.context, TRUE);
+
+  return GST_RTSP_THREAD (impl);
+}
+
+/**
+ * gst_rtsp_thread_reuse:
+ * @thread: (transfer none): a #GstRTSPThread
+ *
+ * Reuse the mainloop of @thread
+ *
+ * Returns: %TRUE if the mainloop could be reused
+ */
+gboolean
+gst_rtsp_thread_reuse (GstRTSPThread * thread)
+{
+  GstRTSPThreadImpl *impl = (GstRTSPThreadImpl *) thread;
+  gboolean res;
+
+  g_return_val_if_fail (GST_IS_RTSP_THREAD (thread), FALSE);
+
+  GST_DEBUG ("reuse thread %p", thread);
+
+  res = g_atomic_int_add (&impl->reused, 1) > 0;
+  if (res)
+    gst_rtsp_thread_ref (thread);
+
+  return res;
+}
+
+static gboolean
+do_quit (GstRTSPThread * thread)
+{
+  GST_DEBUG ("stop mainloop of thread %p", thread);
+  g_main_loop_quit (thread->loop);
+  return FALSE;
+}
+
+/**
+ * gst_rtsp_thread_stop:
+ * @thread: (transfer full): a #GstRTSPThread
+ *
+ * Stop and unref @thread. When no threads are using the mainloop, the thread
+ * will be stopped and the final ref to @thread will be released.
+ */
+void
+gst_rtsp_thread_stop (GstRTSPThread * thread)
+{
+  GstRTSPThreadImpl *impl = (GstRTSPThreadImpl *) thread;
+
+  g_return_if_fail (GST_IS_RTSP_THREAD (thread));
+
+  GST_DEBUG ("stop thread %p", thread);
+
+  if (g_atomic_int_dec_and_test (&impl->reused)) {
+    GST_DEBUG ("add idle source to quit mainloop of thread %p", thread);
+    impl->source = g_idle_source_new ();
+    g_source_set_callback (impl->source, (GSourceFunc) do_quit,
+        thread, (GDestroyNotify) gst_rtsp_thread_unref);
+    g_source_attach (impl->source, thread->context);
+  } else
+    gst_rtsp_thread_unref (thread);
+}
+
+#define GST_RTSP_THREAD_POOL_GET_PRIVATE(obj)  \
+   (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_THREAD_POOL, GstRTSPThreadPoolPrivate))
+
+struct _GstRTSPThreadPoolPrivate
+{
+  GMutex lock;
+
+  gint max_threads;
+  /* currently used mainloops */
+  GQueue threads;
+};
+
+#define DEFAULT_MAX_THREADS 1
+
+enum
+{
+  PROP_0,
+  PROP_MAX_THREADS,
+  PROP_LAST
+};
+
+GST_DEBUG_CATEGORY_STATIC (rtsp_thread_pool_debug);
+#define GST_CAT_DEFAULT rtsp_thread_pool_debug
+
+static GQuark thread_pool;
+
+static void gst_rtsp_thread_pool_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec);
+static void gst_rtsp_thread_pool_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rtsp_thread_pool_finalize (GObject * obj);
+
+static gpointer do_loop (GstRTSPThread * thread);
+static GstRTSPThread *default_get_thread (GstRTSPThreadPool * pool,
+    GstRTSPThreadType type, GstRTSPContext * ctx);
+
+G_DEFINE_TYPE (GstRTSPThreadPool, gst_rtsp_thread_pool, G_TYPE_OBJECT);
+
+static void
+gst_rtsp_thread_pool_class_init (GstRTSPThreadPoolClass * klass)
+{
+  GObjectClass *gobject_class;
+
+  g_type_class_add_private (klass, sizeof (GstRTSPThreadPoolPrivate));
+
+  gobject_class = G_OBJECT_CLASS (klass);
+
+  gobject_class->get_property = gst_rtsp_thread_pool_get_property;
+  gobject_class->set_property = gst_rtsp_thread_pool_set_property;
+  gobject_class->finalize = gst_rtsp_thread_pool_finalize;
+
+  /**
+   * GstRTSPThreadPool::max-threads:
+   *
+   * The maximum amount of threads to use for client connections. A value of
+   * 0 means to use only the mainloop, -1 means an unlimited amount of
+   * threads.
+   */
+  g_object_class_install_property (gobject_class, PROP_MAX_THREADS,
+      g_param_spec_int ("max-threads", "Max Threads",
+          "The maximum amount of threads to use for client connections "
+          "(0 = only mainloop, -1 = unlimited)", -1, G_MAXINT,
+          DEFAULT_MAX_THREADS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  klass->get_thread = default_get_thread;
+
+  GST_DEBUG_CATEGORY_INIT (rtsp_thread_pool_debug, "rtspthreadpool", 0,
+      "GstRTSPThreadPool");
+
+  thread_pool = g_quark_from_string ("gst.rtsp.thread.pool");
+}
+
+static void
+gst_rtsp_thread_pool_init (GstRTSPThreadPool * pool)
+{
+  GstRTSPThreadPoolPrivate *priv;
+
+  pool->priv = priv = GST_RTSP_THREAD_POOL_GET_PRIVATE (pool);
+
+  g_mutex_init (&priv->lock);
+  priv->max_threads = DEFAULT_MAX_THREADS;
+  g_queue_init (&priv->threads);
+}
+
+static void
+gst_rtsp_thread_pool_finalize (GObject * obj)
+{
+  GstRTSPThreadPool *pool = GST_RTSP_THREAD_POOL (obj);
+  GstRTSPThreadPoolPrivate *priv = pool->priv;
+
+  GST_INFO ("finalize pool %p", pool);
+
+  g_queue_clear (&priv->threads);
+  g_mutex_clear (&priv->lock);
+
+  G_OBJECT_CLASS (gst_rtsp_thread_pool_parent_class)->finalize (obj);
+}
+
+static void
+gst_rtsp_thread_pool_get_property (GObject * object, guint propid,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRTSPThreadPool *pool = GST_RTSP_THREAD_POOL (object);
+
+  switch (propid) {
+    case PROP_MAX_THREADS:
+      g_value_set_int (value, gst_rtsp_thread_pool_get_max_threads (pool));
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+static void
+gst_rtsp_thread_pool_set_property (GObject * object, guint propid,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRTSPThreadPool *pool = GST_RTSP_THREAD_POOL (object);
+
+  switch (propid) {
+    case PROP_MAX_THREADS:
+      gst_rtsp_thread_pool_set_max_threads (pool, g_value_get_int (value));
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
+  }
+}
+
+static gpointer
+do_loop (GstRTSPThread * thread)
+{
+  GstRTSPThreadPoolPrivate *priv;
+  GstRTSPThreadPoolClass *klass;
+  GstRTSPThreadPool *pool;
+
+  pool = gst_mini_object_get_qdata (GST_MINI_OBJECT (thread), thread_pool);
+  priv = pool->priv;
+
+  klass = GST_RTSP_THREAD_POOL_GET_CLASS (pool);
+
+  if (klass->thread_enter)
+    klass->thread_enter (pool, thread);
+
+  GST_INFO ("enter mainloop of thread %p", thread);
+  g_main_loop_run (thread->loop);
+  GST_INFO ("exit mainloop of thread %p", thread);
+
+  if (klass->thread_leave)
+    klass->thread_leave (pool, thread);
+
+  g_mutex_lock (&priv->lock);
+  g_queue_remove (&priv->threads, thread);
+  g_mutex_unlock (&priv->lock);
+
+  gst_rtsp_thread_unref (thread);
+
+  return NULL;
+}
+
+/**
+ * gst_rtsp_thread_pool_new:
+ *
+ * Create a new #GstRTSPThreadPool instance.
+ *
+ * Returns: (transfer full): a new #GstRTSPThreadPool
+ */
+GstRTSPThreadPool *
+gst_rtsp_thread_pool_new (void)
+{
+  GstRTSPThreadPool *result;
+
+  result = g_object_new (GST_TYPE_RTSP_THREAD_POOL, NULL);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_thread_pool_set_max_threads:
+ * @pool: a #GstRTSPThreadPool
+ * @max_threads: maximum threads
+ *
+ * Set the maximum threads used by the pool to handle client requests.
+ * A value of 0 will use the pool mainloop, a value of -1 will use an
+ * unlimited number of threads.
+ */
+void
+gst_rtsp_thread_pool_set_max_threads (GstRTSPThreadPool * pool,
+    gint max_threads)
+{
+  GstRTSPThreadPoolPrivate *priv;
+
+  g_return_if_fail (GST_IS_RTSP_THREAD_POOL (pool));
+
+  priv = pool->priv;
+
+  g_mutex_lock (&priv->lock);
+  priv->max_threads = max_threads;
+  g_mutex_unlock (&priv->lock);
+}
+
+/**
+ * gst_rtsp_thread_pool_get_max_threads:
+ * @pool: a #GstRTSPThreadPool
+ *
+ * Get the maximum number of threads used for client connections.
+ * See gst_rtsp_thread_pool_set_max_threads().
+ *
+ * Returns: the maximum number of threads.
+ */
+gint
+gst_rtsp_thread_pool_get_max_threads (GstRTSPThreadPool * pool)
+{
+  GstRTSPThreadPoolPrivate *priv;
+  gint res;
+
+  g_return_val_if_fail (GST_IS_RTSP_THREAD_POOL (pool), -1);
+
+  priv = pool->priv;
+
+  g_mutex_lock (&priv->lock);
+  res = priv->max_threads;
+  g_mutex_unlock (&priv->lock);
+
+  return res;
+}
+
+static GstRTSPThread *
+make_thread (GstRTSPThreadPool * pool, GstRTSPThreadType type,
+    GstRTSPContext * ctx)
+{
+  GstRTSPThreadPoolClass *klass;
+  GstRTSPThread *thread;
+
+  klass = GST_RTSP_THREAD_POOL_GET_CLASS (pool);
+
+  thread = gst_rtsp_thread_new (type);
+  gst_mini_object_set_qdata (GST_MINI_OBJECT (thread), thread_pool,
+      g_object_ref (pool), g_object_unref);
+
+  GST_DEBUG_OBJECT (pool, "new thread %p", thread);
+
+  if (klass->configure_thread)
+    klass->configure_thread (pool, thread, ctx);
+
+  return thread;
+}
+
+static GstRTSPThread *
+default_get_thread (GstRTSPThreadPool * pool,
+    GstRTSPThreadType type, GstRTSPContext * ctx)
+{
+  GstRTSPThreadPoolPrivate *priv = pool->priv;
+  GstRTSPThreadPoolClass *klass;
+  GstRTSPThread *thread;
+  GError *error = NULL;
+
+  klass = GST_RTSP_THREAD_POOL_GET_CLASS (pool);
+
+  switch (type) {
+    case GST_RTSP_THREAD_TYPE_CLIENT:
+      if (priv->max_threads == 0) {
+        /* no threads allowed */
+        GST_DEBUG_OBJECT (pool, "no client threads allowed");
+        thread = NULL;
+      } else {
+        g_mutex_lock (&priv->lock);
+      retry:
+        if (priv->max_threads > 0 &&
+            g_queue_get_length (&priv->threads) >= priv->max_threads) {
+          /* max threads reached, recycle from queue */
+          thread = g_queue_pop_head (&priv->threads);
+          GST_DEBUG_OBJECT (pool, "recycle client thread %p", thread);
+          if (!gst_rtsp_thread_reuse (thread)) {
+            GST_DEBUG_OBJECT (pool, "thread %p stopping, retry", thread);
+            /* this can happen if we just decremented the reuse counter of the
+             * thread and signaled the mainloop that it should stop. We leave
+             * the thread out of the queue now, there is no point to add it
+             * again, it will be removed from the mainloop otherwise after it
+             * stops. */
+            goto retry;
+          }
+        } else {
+          /* make more threads */
+          GST_DEBUG_OBJECT (pool, "make new client thread");
+          thread = make_thread (pool, type, ctx);
+
+          if (!g_thread_pool_push (klass->pool, gst_rtsp_thread_ref (thread),
+                  &error))
+            goto thread_error;
+        }
+        g_queue_push_tail (&priv->threads, thread);
+        g_mutex_unlock (&priv->lock);
+      }
+      break;
+    case GST_RTSP_THREAD_TYPE_MEDIA:
+      GST_DEBUG_OBJECT (pool, "make new media thread");
+      thread = make_thread (pool, type, ctx);
+
+      if (!g_thread_pool_push (klass->pool, gst_rtsp_thread_ref (thread),
+              &error))
+        goto thread_error;
+      break;
+    default:
+      thread = NULL;
+      break;
+  }
+  return thread;
+
+  /* ERRORS */
+thread_error:
+  {
+    GST_ERROR_OBJECT (pool, "failed to push thread %s", error->message);
+    gst_rtsp_thread_unref (thread);
+    /* drop also the ref dedicated for the pool */
+    gst_rtsp_thread_unref (thread);
+    g_clear_error (&error);
+    return NULL;
+  }
+}
+
+/**
+ * gst_rtsp_thread_pool_get_thread:
+ * @pool: a #GstRTSPThreadPool
+ * @type: the #GstRTSPThreadType
+ * @ctx: (transfer none): a #GstRTSPContext
+ *
+ * Get a new #GstRTSPThread for @type and @ctx.
+ *
+ * Returns: (transfer full): a new #GstRTSPThread, gst_rtsp_thread_stop() after usage
+ */
+GstRTSPThread *
+gst_rtsp_thread_pool_get_thread (GstRTSPThreadPool * pool,
+    GstRTSPThreadType type, GstRTSPContext * ctx)
+{
+  GstRTSPThreadPoolClass *klass;
+  GstRTSPThread *result = NULL;
+
+  g_return_val_if_fail (GST_IS_RTSP_THREAD_POOL (pool), NULL);
+
+  klass = GST_RTSP_THREAD_POOL_GET_CLASS (pool);
+
+  /* We want to be thread safe as there might be 2 threads wanting to get new
+   * #GstRTSPThread at the same time
+   */
+  if (G_UNLIKELY (!g_atomic_pointer_get (&klass->pool))) {
+    GThreadPool *t_pool;
+    t_pool = g_thread_pool_new ((GFunc) do_loop, klass, -1, FALSE, NULL);
+    if (!g_atomic_pointer_compare_and_exchange (&klass->pool, NULL, t_pool))
+      g_thread_pool_free (t_pool, FALSE, TRUE);
+  }
+
+  if (klass->get_thread)
+    result = klass->get_thread (pool, type, ctx);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_thread_pool_cleanup:
+ *
+ * Wait for all tasks to be stopped and free all allocated resources. This is
+ * mainly used in test suites to ensure proper cleanup of internal data
+ * structures.
+ */
+void
+gst_rtsp_thread_pool_cleanup (void)
+{
+  GstRTSPThreadPoolClass *klass;
+
+  klass =
+      GST_RTSP_THREAD_POOL_CLASS (g_type_class_peek
+      (gst_rtsp_thread_pool_get_type ()));
+  if (klass->pool != NULL) {
+    g_thread_pool_free (klass->pool, FALSE, TRUE);
+    klass->pool = NULL;
+  }
+}
diff --git a/gst/rtsp-server/rtsp-thread-pool.h b/gst/rtsp-server/rtsp-thread-pool.h
new file mode 100644
index 0000000..e360034
--- /dev/null
+++ b/gst/rtsp-server/rtsp-thread-pool.h
@@ -0,0 +1,179 @@
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#ifndef __GST_RTSP_THREAD_POOL_H__
+#define __GST_RTSP_THREAD_POOL_H__
+
+typedef struct _GstRTSPThread GstRTSPThread;
+typedef struct _GstRTSPThreadPool GstRTSPThreadPool;
+typedef struct _GstRTSPThreadPoolClass GstRTSPThreadPoolClass;
+typedef struct _GstRTSPThreadPoolPrivate GstRTSPThreadPoolPrivate;
+
+#include "rtsp-client.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTSP_THREAD_POOL              (gst_rtsp_thread_pool_get_type ())
+#define GST_IS_RTSP_THREAD_POOL(obj)           (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_THREAD_POOL))
+#define GST_IS_RTSP_THREAD_POOL_CLASS(klass)   (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_THREAD_POOL))
+#define GST_RTSP_THREAD_POOL_GET_CLASS(obj)    (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_THREAD_POOL, GstRTSPThreadPoolClass))
+#define GST_RTSP_THREAD_POOL(obj)              (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_THREAD_POOL, GstRTSPThreadPool))
+#define GST_RTSP_THREAD_POOL_CLASS(klass)      (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_THREAD_POOL, GstRTSPThreadPoolClass))
+#define GST_RTSP_THREAD_POOL_CAST(obj)         ((GstRTSPThreadPool*)(obj))
+#define GST_RTSP_THREAD_POOL_CLASS_CAST(klass) ((GstRTSPThreadPoolClass*)(klass))
+
+GType gst_rtsp_thread_get_type (void);
+
+#define GST_TYPE_RTSP_THREAD        (gst_rtsp_thread_get_type ())
+#define GST_IS_RTSP_THREAD(obj)     (GST_IS_MINI_OBJECT_TYPE (obj, GST_TYPE_RTSP_THREAD))
+#define GST_RTSP_THREAD_CAST(obj)   ((GstRTSPThread*)(obj))
+#define GST_RTSP_THREAD(obj)        (GST_RTSP_THREAD_CAST(obj))
+
+/**
+ * GstRTSPThreadType:
+ * @GST_RTSP_THREAD_TYPE_CLIENT: a thread to handle the client communication
+ * @GST_RTSP_THREAD_TYPE_MEDIA: a thread to handle media 
+ *
+ * Different thread types
+ */
+typedef enum
+{
+  GST_RTSP_THREAD_TYPE_CLIENT,
+  GST_RTSP_THREAD_TYPE_MEDIA
+} GstRTSPThreadType;
+
+/**
+ * GstRTSPThread:
+ * @mini_object: parent #GstMiniObject
+ * @type: the thread type
+ * @context: a #GMainContext
+ * @loop: a #GMainLoop
+ *
+ * Structure holding info about a mainloop running in a thread
+ */
+struct _GstRTSPThread {
+  GstMiniObject mini_object;
+
+  GstRTSPThreadType type;
+  GMainContext *context;
+  GMainLoop *loop;
+};
+
+GstRTSPThread *   gst_rtsp_thread_new      (GstRTSPThreadType type);
+
+gboolean          gst_rtsp_thread_reuse    (GstRTSPThread * thread);
+void              gst_rtsp_thread_stop     (GstRTSPThread * thread);
+
+/**
+ * gst_rtsp_thread_ref:
+ * @thread: The thread to refcount
+ *
+ * Increase the refcount of this thread.
+ *
+ * Returns: (transfer full): @thread (for convenience when doing assignments)
+ */
+#ifdef _FOOL_GTK_DOC_
+G_INLINE_FUNC GstRTSPThread * gst_rtsp_thread_ref (GstRTSPThread * thread);
+#endif
+
+static inline GstRTSPThread *
+gst_rtsp_thread_ref (GstRTSPThread * thread)
+{
+  return (GstRTSPThread *) gst_mini_object_ref (GST_MINI_OBJECT_CAST (thread));
+}
+
+/**
+ * gst_rtsp_thread_unref:
+ * @thread: (transfer full): the thread to refcount
+ *
+ * Decrease the refcount of an thread, freeing it if the refcount reaches 0.
+ */
+#ifdef _FOOL_GTK_DOC_
+G_INLINE_FUNC void gst_rtsp_thread_unref (GstRTSPPermissions * thread);
+#endif
+
+
+static inline void
+gst_rtsp_thread_unref (GstRTSPThread * thread)
+{
+  gst_mini_object_unref (GST_MINI_OBJECT_CAST (thread));
+}
+
+/**
+ * GstRTSPThreadPool:
+ *
+ * The thread pool structure.
+ */
+struct _GstRTSPThreadPool {
+  GObject       parent;
+
+  /*< private >*/
+  GstRTSPThreadPoolPrivate *priv;
+  gpointer _gst_reserved[GST_PADDING];
+};
+
+/**
+ * GstRTSPThreadPoolClass:
+ * @pool: a #GThreadPool used internally
+ * @get_thread: this function should make or reuse an existing thread that runs
+ *        a mainloop.
+ * @configure_thread: configure a thread object. this vmethod is called when
+ *       a new thread has been created and should be configured.
+ * @thread_enter: called from the thread when it is entered
+ * @thread_leave: called from the thread when it is left
+ *
+ * Class for managing threads.
+ */
+struct _GstRTSPThreadPoolClass {
+  GObjectClass  parent_class;
+
+  GThreadPool *pool;
+
+  GstRTSPThread * (*get_thread)        (GstRTSPThreadPool *pool,
+                                        GstRTSPThreadType type,
+                                        GstRTSPContext *ctx);
+  void            (*configure_thread)  (GstRTSPThreadPool *pool,
+                                        GstRTSPThread * thread,
+                                        GstRTSPContext *ctx);
+
+  void            (*thread_enter)      (GstRTSPThreadPool *pool,
+                                        GstRTSPThread *thread);
+  void            (*thread_leave)      (GstRTSPThreadPool *pool,
+                                        GstRTSPThread *thread);
+
+  /*< private >*/
+  gpointer         _gst_reserved[GST_PADDING];
+};
+
+GType               gst_rtsp_thread_pool_get_type        (void);
+
+GstRTSPThreadPool * gst_rtsp_thread_pool_new             (void);
+
+void                gst_rtsp_thread_pool_set_max_threads (GstRTSPThreadPool * pool, gint max_threads);
+gint                gst_rtsp_thread_pool_get_max_threads (GstRTSPThreadPool * pool);
+
+GstRTSPThread *     gst_rtsp_thread_pool_get_thread      (GstRTSPThreadPool *pool,
+                                                          GstRTSPThreadType type,
+                                                          GstRTSPContext *ctx);
+void                gst_rtsp_thread_pool_cleanup         (void);
+G_END_DECLS
+
+#endif /* __GST_RTSP_THREAD_POOL_H__ */
diff --git a/gst/rtsp-server/rtsp-token.c b/gst/rtsp-server/rtsp-token.c
new file mode 100644
index 0000000..69250f9
--- /dev/null
+++ b/gst/rtsp-server/rtsp-token.c
@@ -0,0 +1,251 @@
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/**
+ * SECTION:rtsp-token
+ * @short_description: Roles and permissions for a client
+ * @see_also: #GstRTSPClient, #GstRTSPPermissions, #GstRTSPAuth
+ *
+ * A #GstRTSPToken contains the permissions and roles of the user
+ * performing the current request. A token is usually created when a user is
+ * authenticated by the #GstRTSPAuth object and is then placed as the current
+ * token for the current request.
+ *
+ * #GstRTSPAuth can use the token and its contents to check authorization for
+ * various operations by comparing the token to the #GstRTSPPermissions of the
+ * object.
+ *
+ * The accepted values of the token are entirely defined by the #GstRTSPAuth
+ * object that implements the security policy.
+ *
+ * Last reviewed on 2013-07-15 (1.0.0)
+ */
+
+#include 
+
+#include "rtsp-token.h"
+
+typedef struct _GstRTSPTokenImpl
+{
+  GstRTSPToken token;
+
+  GstStructure *structure;
+} GstRTSPTokenImpl;
+
+#define GST_RTSP_TOKEN_STRUCTURE(t)  (((GstRTSPTokenImpl *)(t))->structure)
+
+//GST_DEBUG_CATEGORY_STATIC (rtsp_token_debug);
+//#define GST_CAT_DEFAULT rtsp_token_debug
+
+GST_DEFINE_MINI_OBJECT_TYPE (GstRTSPToken, gst_rtsp_token);
+
+static void gst_rtsp_token_init (GstRTSPTokenImpl * token,
+    GstStructure * structure);
+
+static void
+_gst_rtsp_token_free (GstRTSPToken * token)
+{
+  GstRTSPTokenImpl *impl = (GstRTSPTokenImpl *) token;
+
+  gst_structure_set_parent_refcount (impl->structure, NULL);
+  gst_structure_free (impl->structure);
+
+  g_slice_free1 (sizeof (GstRTSPTokenImpl), token);
+}
+
+static GstRTSPToken *
+_gst_rtsp_token_copy (GstRTSPTokenImpl * token)
+{
+  GstRTSPTokenImpl *copy;
+  GstStructure *structure;
+
+  structure = gst_structure_copy (token->structure);
+
+  copy = g_slice_new0 (GstRTSPTokenImpl);
+  gst_rtsp_token_init (copy, structure);
+
+  return (GstRTSPToken *) copy;
+}
+
+static void
+gst_rtsp_token_init (GstRTSPTokenImpl * token, GstStructure * structure)
+{
+  gst_mini_object_init (GST_MINI_OBJECT_CAST (token), 0,
+      GST_TYPE_RTSP_TOKEN,
+      (GstMiniObjectCopyFunction) _gst_rtsp_token_copy, NULL,
+      (GstMiniObjectFreeFunction) _gst_rtsp_token_free);
+
+  token->structure = structure;
+  gst_structure_set_parent_refcount (token->structure,
+      &token->token.mini_object.refcount);
+}
+
+/**
+ * gst_rtsp_token_new_empty:
+ *
+ * Create a new empty Authorization token.
+ *
+ * Returns: (transfer full): a new empty authorization token.
+ */
+GstRTSPToken *
+gst_rtsp_token_new_empty (void)
+{
+  GstRTSPTokenImpl *token;
+  GstStructure *s;
+
+  s = gst_structure_new_empty ("GstRTSPToken");
+  g_return_val_if_fail (s != NULL, NULL);
+
+  token = g_slice_new0 (GstRTSPTokenImpl);
+  gst_rtsp_token_init (token, s);
+
+  return (GstRTSPToken *) token;
+}
+
+/**
+ * gst_rtsp_token_new:
+ * @firstfield: the first fieldname
+ * @...: additional arguments
+ *
+ * Create a new Authorization token with the given fieldnames and values.
+ * Arguments are given similar to gst_structure_new().
+ *
+ * Returns: (transfer full): a new authorization token.
+ */
+GstRTSPToken *
+gst_rtsp_token_new (const gchar * firstfield, ...)
+{
+  GstRTSPToken *result;
+  va_list var_args;
+
+  va_start (var_args, firstfield);
+  result = gst_rtsp_token_new_valist (firstfield, var_args);
+  va_end (var_args);
+
+  return result;
+}
+
+/**
+ * gst_rtsp_token_new_valist:
+ * @firstfield: the first fieldname
+ * @var_args: additional arguments
+ *
+ * Create a new Authorization token with the given fieldnames and values.
+ * Arguments are given similar to gst_structure_new_valist().
+ *
+ * Returns: (transfer full): a new authorization token.
+ */
+GstRTSPToken *
+gst_rtsp_token_new_valist (const gchar * firstfield, va_list var_args)
+{
+  GstRTSPToken *token;
+  GstStructure *s;
+
+  g_return_val_if_fail (firstfield != NULL, NULL);
+
+  token = gst_rtsp_token_new_empty ();
+  s = GST_RTSP_TOKEN_STRUCTURE (token);
+  gst_structure_set_valist (s, firstfield, var_args);
+
+  return token;
+}
+
+
+/**
+ * gst_rtsp_token_get_structure:
+ * @token: The #GstRTSPToken.
+ *
+ * Access the structure of the token.
+ *
+ * Returns: (transfer none): The structure of the token. The structure is still
+ * owned by the token, which means that you should not free it and that the
+ * pointer becomes invalid when you free the token.
+ *
+ * MT safe.
+ */
+const GstStructure *
+gst_rtsp_token_get_structure (GstRTSPToken * token)
+{
+  g_return_val_if_fail (GST_IS_RTSP_TOKEN (token), NULL);
+
+  return GST_RTSP_TOKEN_STRUCTURE (token);
+}
+
+/**
+ * gst_rtsp_token_writable_structure:
+ * @token: The #GstRTSPToken.
+ *
+ * Get a writable version of the structure.
+ *
+ * Returns: (transfer none): The structure of the token. The structure is still
+ * owned by the token, which means that you should not free it and that the
+ * pointer becomes invalid when you free the token. This function checks if
+ * @token is writable and will never return %NULL.
+ *
+ * MT safe.
+ */
+GstStructure *
+gst_rtsp_token_writable_structure (GstRTSPToken * token)
+{
+  g_return_val_if_fail (GST_IS_RTSP_TOKEN (token), NULL);
+  g_return_val_if_fail (gst_mini_object_is_writable (GST_MINI_OBJECT_CAST
+          (token)), NULL);
+
+  return GST_RTSP_TOKEN_STRUCTURE (token);
+}
+
+/**
+ * gst_rtsp_token_get_string:
+ * @token: a #GstRTSPToken
+ * @field: a field name
+ *
+ * Get the string value of @field in @token.
+ *
+ * Returns: (transfer none) (nullable): the string value of @field in
+ * @token or %NULL when @field is not defined in @token. The string
+ * becomes invalid when you free @token.
+ */
+const gchar *
+gst_rtsp_token_get_string (GstRTSPToken * token, const gchar * field)
+{
+  return gst_structure_get_string (GST_RTSP_TOKEN_STRUCTURE (token), field);
+}
+
+/**
+ * gst_rtsp_token_is_allowed:
+ * @token: a #GstRTSPToken
+ * @field: a field name
+ *
+ * Check if @token has a boolean @field and if it is set to %TRUE.
+ *
+ * Returns: %TRUE if @token has a boolean field named @field set to %TRUE.
+ */
+gboolean
+gst_rtsp_token_is_allowed (GstRTSPToken * token, const gchar * field)
+{
+  gboolean result;
+
+  g_return_val_if_fail (GST_IS_RTSP_TOKEN (token), FALSE);
+  g_return_val_if_fail (field != NULL, FALSE);
+
+  if (!gst_structure_get_boolean (GST_RTSP_TOKEN_STRUCTURE (token), field,
+          &result))
+    result = FALSE;
+
+  return result;
+}
diff --git a/gst/rtsp-server/rtsp-token.h b/gst/rtsp-server/rtsp-token.h
new file mode 100644
index 0000000..265d8d3
--- /dev/null
+++ b/gst/rtsp-server/rtsp-token.h
@@ -0,0 +1,97 @@
+/* GStreamer
+ * Copyright (C) 2010 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#ifndef __GST_RTSP_TOKEN_H__
+#define __GST_RTSP_TOKEN_H__
+
+typedef struct _GstRTSPToken GstRTSPToken;
+
+#include "rtsp-auth.h"
+
+G_BEGIN_DECLS
+
+GType gst_rtsp_token_get_type(void);
+
+#define GST_TYPE_RTSP_TOKEN        (gst_rtsp_token_get_type())
+#define GST_IS_RTSP_TOKEN(obj)     (GST_IS_MINI_OBJECT_TYPE (obj, GST_TYPE_RTSP_TOKEN))
+#define GST_RTSP_TOKEN_CAST(obj)   ((GstRTSPToken*)(obj))
+#define GST_RTSP_TOKEN(obj)        (GST_RTSP_TOKEN_CAST(obj))
+
+/**
+ * GstRTSPToken:
+ *
+ * An opaque object used for checking authorisations.
+ * It is generated after successful authentication.
+ */
+struct _GstRTSPToken {
+  GstMiniObject mini_object;
+};
+
+/* refcounting */
+/**
+ * gst_rtsp_token_ref:
+ * @token: The token to refcount
+ *
+ * Increase the refcount of this token.
+ *
+ * Returns: (transfer full): @token (for convenience when doing assignments)
+ */
+#ifdef _FOOL_GTK_DOC_
+G_INLINE_FUNC GstRTSPToken * gst_rtsp_token_ref (GstRTSPToken * token);
+#endif
+
+static inline GstRTSPToken *
+gst_rtsp_token_ref (GstRTSPToken * token)
+{
+  return (GstRTSPToken *) gst_mini_object_ref (GST_MINI_OBJECT_CAST (token));
+}
+
+/**
+ * gst_rtsp_token_unref:
+ * @token: (transfer full): the token to refcount
+ *
+ * Decrease the refcount of an token, freeing it if the refcount reaches 0.
+ */
+#ifdef _FOOL_GTK_DOC_
+G_INLINE_FUNC void gst_rtsp_token_unref (GstRTSPToken * token);
+#endif
+
+static inline void
+gst_rtsp_token_unref (GstRTSPToken * token)
+{
+  gst_mini_object_unref (GST_MINI_OBJECT_CAST (token));
+}
+
+
+GstRTSPToken *       gst_rtsp_token_new_empty          (void);
+GstRTSPToken *       gst_rtsp_token_new                (const gchar * firstfield, ...);
+GstRTSPToken *       gst_rtsp_token_new_valist         (const gchar * firstfield, va_list var_args);
+
+const GstStructure * gst_rtsp_token_get_structure      (GstRTSPToken *token);
+GstStructure *       gst_rtsp_token_writable_structure (GstRTSPToken *token);
+
+const gchar *        gst_rtsp_token_get_string         (GstRTSPToken *token,
+                                                        const gchar *field);
+gboolean             gst_rtsp_token_is_allowed         (GstRTSPToken *token,
+                                                        const gchar *field);
+G_END_DECLS
+
+#endif /* __GST_RTSP_TOKEN_H__ */
diff --git a/packaging/gst-rtsp-server.spec b/packaging/gst-rtsp-server.spec
new file mode 100644
index 0000000..7d47de6
--- /dev/null
+++ b/packaging/gst-rtsp-server.spec
@@ -0,0 +1,73 @@
+Name:       gst-rtsp-server
+Summary:    Multimedia Framework Library
+Version:    1.4.5
+Release:    3
+Group:      System/Libraries
+License:    LGPL-2.0+
+Source0:    %{name}-%{version}.tar.gz
+Requires(post):  /sbin/ldconfig
+Requires(postun):  /sbin/ldconfig
+BuildRequires:  pkgconfig(gstreamer-1.0)
+BuildRequires:  pkgconfig(gstreamer-plugins-base-1.0)
+
+BuildRoot:  %{_tmppath}/%{name}-%{version}-build
+
+%description
+
+%package devel
+Summary:    Multimedia Framework RTSP server library (DEV)
+Group:      Development/Libraries
+Requires:   %{name} = %{version}-%{release}
+
+%description devel
+
+%package factory
+Summary:    Multimedia Framework RTSP server Library (Factory)
+Group:      Development/Libraries
+Requires:   %{name} = %{version}-%{release}
+
+%description factory
+
+%prep
+%setup -q
+
+%build
+
+./autogen.sh
+
+CFLAGS+=" -DEXPORT_API=\"__attribute__((visibility(\\\"default\\\")))\" "; export CFLAGS
+LDFLAGS+="-Wl,--rpath=%{_prefix}/lib -Wl,--hash-style=both -Wl,--as-needed"; export LDFLAGS
+
+# always enable sdk build. This option should go away
+%configure --disable-static
+
+# Call make instruction with smp support
+make %{?jobs:-j%jobs}
+
+%install
+rm -rf %{buildroot}
+%make_install
+mkdir -p %{buildroot}/%{_datadir}/license
+cp -rf %{_builddir}/%{name}-%{version}/COPYING %{buildroot}%{_datadir}/license/%{name}
+
+%clean
+rm -rf %{buildroot}
+
+%post
+/sbin/ldconfig
+
+%postun
+/sbin/ldconfig
+
+%files
+%manifest gst-rtsp-server.manifest
+%defattr(-,root,root,-)
+%{_datadir}/license/%{name}
+%{_libdir}/*.so.*
+
+%files devel
+%defattr(-,root,root,-)
+%{_libdir}/*.so
+%{_includedir}/gstreamer-1.0/gst/rtsp-server/rtsp-*.h
+%{_includedir}/gstreamer-1.0/gst/rtsp-server/gstwfd*.h
+%{_libdir}/pkgconfig/*
diff --git a/pkgconfig/Makefile.am b/pkgconfig/Makefile.am
new file mode 100644
index 0000000..c45755e
--- /dev/null
+++ b/pkgconfig/Makefile.am
@@ -0,0 +1,21 @@
+pcfiles = \
+	gstreamer-rtsp-server-@GST_API_VERSION@.pc
+	
+pcfiles_uninstalled = \
+	gstreamer-rtsp-server-@GST_API_VERSION@-uninstalled.pc
+
+all-local: $(pcfiles) $(pcfiles_uninstalled)
+
+### how to generate pc files
+%-@GST_API_VERSION@.pc: %.pc
+	cp $< $@
+%-@GST_API_VERSION@-uninstalled.pc: %-uninstalled.pc
+	cp $< $@
+
+pkgconfigdir = $(libdir)/pkgconfig
+pkgconfig_DATA = $(pcfiles)
+
+EXTRA_DIST = \
+	gstreamer-rtsp-server.pc.in \
+	gstreamer-rtsp-server-uninstalled.pc.in
+CLEANFILES = $(pcfiles) $(pcfiles_uninstalled)
diff --git a/pkgconfig/Makefile.in b/pkgconfig/Makefile.in
new file mode 100644
index 0000000..bafc540
--- /dev/null
+++ b/pkgconfig/Makefile.in
@@ -0,0 +1,637 @@
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+# @configure_input@
+
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+
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+# with or without modifications, as long as this notice is preserved.
+
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+# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
+# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
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+
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+HAVE_JADETEX = @HAVE_JADETEX@
+HAVE_PNGTOPNM = @HAVE_PNGTOPNM@
+HAVE_PNMTOPS = @HAVE_PNMTOPS@
+HAVE_PS2PDF = @HAVE_PS2PDF@
+HAVE_XMLLINT = @HAVE_XMLLINT@
+HAVE_XSLTPROC = @HAVE_XSLTPROC@
+HTML_DIR = @HTML_DIR@
+INSTALL = @INSTALL@
+INSTALL_DATA = @INSTALL_DATA@
+INSTALL_PROGRAM = @INSTALL_PROGRAM@
+INSTALL_SCRIPT = @INSTALL_SCRIPT@
+INSTALL_STRIP_PROGRAM = @INSTALL_STRIP_PROGRAM@
+INTROSPECTION_CFLAGS = @INTROSPECTION_CFLAGS@
+INTROSPECTION_COMPILER = @INTROSPECTION_COMPILER@
+INTROSPECTION_GENERATE = @INTROSPECTION_GENERATE@
+INTROSPECTION_GIRDIR = @INTROSPECTION_GIRDIR@
+INTROSPECTION_LIBS = @INTROSPECTION_LIBS@
+INTROSPECTION_MAKEFILE = @INTROSPECTION_MAKEFILE@
+INTROSPECTION_SCANNER = @INTROSPECTION_SCANNER@
+INTROSPECTION_TYPELIBDIR = @INTROSPECTION_TYPELIBDIR@
+LD = @LD@
+LDFLAGS = @LDFLAGS@
+LIBCGROUP_CFLAGS = @LIBCGROUP_CFLAGS@
+LIBCGROUP_LIBS = @LIBCGROUP_LIBS@
+LIBOBJS = @LIBOBJS@
+LIBS = @LIBS@
+LIBTOOL = @LIBTOOL@
+LIPO = @LIPO@
+LN_S = @LN_S@
+LTLIBOBJS = @LTLIBOBJS@
+MAINT = @MAINT@
+MAKEINFO = @MAKEINFO@
+MANIFEST_TOOL = @MANIFEST_TOOL@
+MKDIR_P = @MKDIR_P@
+NM = @NM@
+NMEDIT = @NMEDIT@
+OBJDUMP = @OBJDUMP@
+OBJEXT = @OBJEXT@
+OTOOL = @OTOOL@
+OTOOL64 = @OTOOL64@
+PACKAGE = @PACKAGE@
+PACKAGE_BUGREPORT = @PACKAGE_BUGREPORT@
+PACKAGE_NAME = @PACKAGE_NAME@
+PACKAGE_STRING = @PACKAGE_STRING@
+PACKAGE_TARNAME = @PACKAGE_TARNAME@
+PACKAGE_URL = @PACKAGE_URL@
+PACKAGE_VERSION = @PACKAGE_VERSION@
+PACKAGE_VERSION_MAJOR = @PACKAGE_VERSION_MAJOR@
+PACKAGE_VERSION_MICRO = @PACKAGE_VERSION_MICRO@
+PACKAGE_VERSION_MINOR = @PACKAGE_VERSION_MINOR@
+PACKAGE_VERSION_NANO = @PACKAGE_VERSION_NANO@
+PACKAGE_VERSION_RELEASE = @PACKAGE_VERSION_RELEASE@
+PATH_SEPARATOR = @PATH_SEPARATOR@
+PKG_CONFIG = @PKG_CONFIG@
+PLUGINDIR = @PLUGINDIR@
+PROFILE_CFLAGS = @PROFILE_CFLAGS@
+RANLIB = @RANLIB@
+SED = @SED@
+SET_MAKE = @SET_MAKE@
+SHELL = @SHELL@
+STRIP = @STRIP@
+VALGRIND_CFLAGS = @VALGRIND_CFLAGS@
+VALGRIND_LIBS = @VALGRIND_LIBS@
+VALGRIND_PATH = @VALGRIND_PATH@
+VERSION = @VERSION@
+WARNING_CFLAGS = @WARNING_CFLAGS@
+XML_CATALOG = @XML_CATALOG@
+XSLTPROC = @XSLTPROC@
+XSLTPROC_FLAGS = @XSLTPROC_FLAGS@
+abs_builddir = @abs_builddir@
+abs_srcdir = @abs_srcdir@
+abs_top_builddir = @abs_top_builddir@
+abs_top_srcdir = @abs_top_srcdir@
+ac_ct_AR = @ac_ct_AR@
+ac_ct_CC = @ac_ct_CC@
+ac_ct_DUMPBIN = @ac_ct_DUMPBIN@
+am__include = @am__include@
+am__leading_dot = @am__leading_dot@
+am__quote = @am__quote@
+am__tar = @am__tar@
+am__untar = @am__untar@
+bindir = @bindir@
+build = @build@
+build_alias = @build_alias@
+build_cpu = @build_cpu@
+build_os = @build_os@
+build_vendor = @build_vendor@
+builddir = @builddir@
+datadir = @datadir@
+datarootdir = @datarootdir@
+docdir = @docdir@
+dvidir = @dvidir@
+exec_prefix = @exec_prefix@
+host = @host@
+host_alias = @host_alias@
+host_cpu = @host_cpu@
+host_os = @host_os@
+host_vendor = @host_vendor@
+htmldir = @htmldir@
+includedir = @includedir@
+infodir = @infodir@
+install_sh = @install_sh@
+libdir = @libdir@
+libexecdir = @libexecdir@
+localedir = @localedir@
+localstatedir = @localstatedir@
+mandir = @mandir@
+mkdir_p = @mkdir_p@
+oldincludedir = @oldincludedir@
+pdfdir = @pdfdir@
+plugindir = @plugindir@
+prefix = @prefix@
+program_transform_name = @program_transform_name@
+psdir = @psdir@
+sbindir = @sbindir@
+sharedstatedir = @sharedstatedir@
+srcdir = @srcdir@
+sysconfdir = @sysconfdir@
+target = @target@
+target_alias = @target_alias@
+target_cpu = @target_cpu@
+target_os = @target_os@
+target_vendor = @target_vendor@
+top_build_prefix = @top_build_prefix@
+top_builddir = @top_builddir@
+top_srcdir = @top_srcdir@
+pcfiles = \
+	gstreamer-rtsp-server-@GST_API_VERSION@.pc
+
+pcfiles_uninstalled = \
+	gstreamer-rtsp-server-@GST_API_VERSION@-uninstalled.pc
+
+pkgconfigdir = $(libdir)/pkgconfig
+pkgconfig_DATA = $(pcfiles)
+EXTRA_DIST = \
+	gstreamer-rtsp-server.pc.in \
+	gstreamer-rtsp-server-uninstalled.pc.in
+
+CLEANFILES = $(pcfiles) $(pcfiles_uninstalled)
+all: all-am
+
+.SUFFIXES:
+$(srcdir)/Makefile.in: @MAINTAINER_MODE_TRUE@ $(srcdir)/Makefile.am  $(am__configure_deps)
+	@for dep in $?; do \
+	  case '$(am__configure_deps)' in \
+	    *$$dep*) \
+	      ( cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh ) \
+	        && { if test -f $@; then exit 0; else break; fi; }; \
+	      exit 1;; \
+	  esac; \
+	done; \
+	echo ' cd $(top_srcdir) && $(AUTOMAKE) --gnu pkgconfig/Makefile'; \
+	$(am__cd) $(top_srcdir) && \
+	  $(AUTOMAKE) --gnu pkgconfig/Makefile
+.PRECIOUS: Makefile
+Makefile: $(srcdir)/Makefile.in $(top_builddir)/config.status
+	@case '$?' in \
+	  *config.status*) \
+	    cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh;; \
+	  *) \
+	    echo ' cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@ $(am__depfiles_maybe)'; \
+	    cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@ $(am__depfiles_maybe);; \
+	esac;
+
+$(top_builddir)/config.status: $(top_srcdir)/configure $(CONFIG_STATUS_DEPENDENCIES)
+	cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+
+$(top_srcdir)/configure: @MAINTAINER_MODE_TRUE@ $(am__configure_deps)
+	cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+$(ACLOCAL_M4): @MAINTAINER_MODE_TRUE@ $(am__aclocal_m4_deps)
+	cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+$(am__aclocal_m4_deps):
+gstreamer-rtsp-server.pc: $(top_builddir)/config.status $(srcdir)/gstreamer-rtsp-server.pc.in
+	cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@
+gstreamer-rtsp-server-uninstalled.pc: $(top_builddir)/config.status $(srcdir)/gstreamer-rtsp-server-uninstalled.pc.in
+	cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@
+
+mostlyclean-libtool:
+	-rm -f *.lo
+
+clean-libtool:
+	-rm -rf .libs _libs
+install-pkgconfigDATA: $(pkgconfig_DATA)
+	@$(NORMAL_INSTALL)
+	@list='$(pkgconfig_DATA)'; test -n "$(pkgconfigdir)" || list=; \
+	if test -n "$$list"; then \
+	  echo " $(MKDIR_P) '$(DESTDIR)$(pkgconfigdir)'"; \
+	  $(MKDIR_P) "$(DESTDIR)$(pkgconfigdir)" || exit 1; \
+	fi; \
+	for p in $$list; do \
+	  if test -f "$$p"; then d=; else d="$(srcdir)/"; fi; \
+	  echo "$$d$$p"; \
+	done | $(am__base_list) | \
+	while read files; do \
+	  echo " $(INSTALL_DATA) $$files '$(DESTDIR)$(pkgconfigdir)'"; \
+	  $(INSTALL_DATA) $$files "$(DESTDIR)$(pkgconfigdir)" || exit $$?; \
+	done
+
+uninstall-pkgconfigDATA:
+	@$(NORMAL_UNINSTALL)
+	@list='$(pkgconfig_DATA)'; test -n "$(pkgconfigdir)" || list=; \
+	files=`for p in $$list; do echo $$p; done | sed -e 's|^.*/||'`; \
+	dir='$(DESTDIR)$(pkgconfigdir)'; $(am__uninstall_files_from_dir)
+tags TAGS:
+
+ctags CTAGS:
+
+cscope cscopelist:
+
+
+distdir: $(DISTFILES)
+	@srcdirstrip=`echo "$(srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+	topsrcdirstrip=`echo "$(top_srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+	list='$(DISTFILES)'; \
+	  dist_files=`for file in $$list; do echo $$file; done | \
+	  sed -e "s|^$$srcdirstrip/||;t" \
+	      -e "s|^$$topsrcdirstrip/|$(top_builddir)/|;t"`; \
+	case $$dist_files in \
+	  */*) $(MKDIR_P) `echo "$$dist_files" | \
+			   sed '/\//!d;s|^|$(distdir)/|;s,/[^/]*$$,,' | \
+			   sort -u` ;; \
+	esac; \
+	for file in $$dist_files; do \
+	  if test -f $$file || test -d $$file; then d=.; else d=$(srcdir); fi; \
+	  if test -d $$d/$$file; then \
+	    dir=`echo "/$$file" | sed -e 's,/[^/]*$$,,'`; \
+	    if test -d "$(distdir)/$$file"; then \
+	      find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+	    fi; \
+	    if test -d $(srcdir)/$$file && test $$d != $(srcdir); then \
+	      cp -fpR $(srcdir)/$$file "$(distdir)$$dir" || exit 1; \
+	      find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+	    fi; \
+	    cp -fpR $$d/$$file "$(distdir)$$dir" || exit 1; \
+	  else \
+	    test -f "$(distdir)/$$file" \
+	    || cp -p $$d/$$file "$(distdir)/$$file" \
+	    || exit 1; \
+	  fi; \
+	done
+check-am: all-am
+check: check-am
+all-am: Makefile $(DATA) all-local
+installdirs:
+	for dir in "$(DESTDIR)$(pkgconfigdir)"; do \
+	  test -z "$$dir" || $(MKDIR_P) "$$dir"; \
+	done
+install: install-am
+install-exec: install-exec-am
+install-data: install-data-am
+uninstall: uninstall-am
+
+install-am: all-am
+	@$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
+
+installcheck: installcheck-am
+install-strip:
+	if test -z '$(STRIP)'; then \
+	  $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	    install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	      install; \
+	else \
+	  $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	    install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	    "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'" install; \
+	fi
+mostlyclean-generic:
+
+clean-generic:
+	-test -z "$(CLEANFILES)" || rm -f $(CLEANFILES)
+
+distclean-generic:
+	-test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
+	-test . = "$(srcdir)" || test -z "$(CONFIG_CLEAN_VPATH_FILES)" || rm -f $(CONFIG_CLEAN_VPATH_FILES)
+
+maintainer-clean-generic:
+	@echo "This command is intended for maintainers to use"
+	@echo "it deletes files that may require special tools to rebuild."
+clean: clean-am
+
+clean-am: clean-generic clean-libtool mostlyclean-am
+
+distclean: distclean-am
+	-rm -f Makefile
+distclean-am: clean-am distclean-generic
+
+dvi: dvi-am
+
+dvi-am:
+
+html: html-am
+
+html-am:
+
+info: info-am
+
+info-am:
+
+install-data-am: install-pkgconfigDATA
+
+install-dvi: install-dvi-am
+
+install-dvi-am:
+
+install-exec-am:
+
+install-html: install-html-am
+
+install-html-am:
+
+install-info: install-info-am
+
+install-info-am:
+
+install-man:
+
+install-pdf: install-pdf-am
+
+install-pdf-am:
+
+install-ps: install-ps-am
+
+install-ps-am:
+
+installcheck-am:
+
+maintainer-clean: maintainer-clean-am
+	-rm -f Makefile
+maintainer-clean-am: distclean-am maintainer-clean-generic
+
+mostlyclean: mostlyclean-am
+
+mostlyclean-am: mostlyclean-generic mostlyclean-libtool
+
+pdf: pdf-am
+
+pdf-am:
+
+ps: ps-am
+
+ps-am:
+
+uninstall-am: uninstall-pkgconfigDATA
+
+.MAKE: install-am install-strip
+
+.PHONY: all all-am all-local check check-am clean clean-generic \
+	clean-libtool cscopelist-am ctags-am distclean \
+	distclean-generic distclean-libtool distdir dvi dvi-am html \
+	html-am info info-am install install-am install-data \
+	install-data-am install-dvi install-dvi-am install-exec \
+	install-exec-am install-html install-html-am install-info \
+	install-info-am install-man install-pdf install-pdf-am \
+	install-pkgconfigDATA install-ps install-ps-am install-strip \
+	installcheck installcheck-am installdirs maintainer-clean \
+	maintainer-clean-generic mostlyclean mostlyclean-generic \
+	mostlyclean-libtool pdf pdf-am ps ps-am tags-am uninstall \
+	uninstall-am uninstall-pkgconfigDATA
+
+
+all-local: $(pcfiles) $(pcfiles_uninstalled)
+
+### how to generate pc files
+%-@GST_API_VERSION@.pc: %.pc
+	cp $< $@
+%-@GST_API_VERSION@-uninstalled.pc: %-uninstalled.pc
+	cp $< $@
+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
diff --git a/pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in b/pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in
new file mode 100644
index 0000000..e453805
--- /dev/null
+++ b/pkgconfig/gstreamer-rtsp-server-uninstalled.pc.in
@@ -0,0 +1,12 @@
+# the standard variables don't make sense for an uninstalled copy
+prefix=
+exec_prefix=
+libdir=${pcfiledir}/../gst/rtsp-server
+includedir=${pcfiledir}/..
+
+Name: gst-rtsp-server
+Description: GStreamer based RTSP server
+Version: @VERSION@
+Requires: gstreamer-@GST_API_VERSION@ gstreamer-plugins-base-@GST_API_VERSION@
+Libs: ${libdir}/libgstrtspserver-@GST_API_VERSION@.la
+Cflags: -I${includedir} -I@srcdir@/..
diff --git a/pkgconfig/gstreamer-rtsp-server.pc.in b/pkgconfig/gstreamer-rtsp-server.pc.in
new file mode 100644
index 0000000..3a1d820
--- /dev/null
+++ b/pkgconfig/gstreamer-rtsp-server.pc.in
@@ -0,0 +1,11 @@
+prefix=@prefix@
+exec_prefix=@exec_prefix@
+libdir=@libdir@
+includedir=@includedir@/gstreamer-@GST_API_VERSION@
+
+Name: gst-rtsp-server
+Description: GStreamer based RTSP server
+Version: @VERSION@
+Requires: gstreamer-@GST_API_VERSION@ gstreamer-base-@GST_API_VERSION@
+Libs: -L${libdir} -lgstrtspserver-@GST_API_VERSION@
+Cflags: -I${includedir}
diff --git a/tests/Makefile.am b/tests/Makefile.am
new file mode 100644
index 0000000..ca2ba66
--- /dev/null
+++ b/tests/Makefile.am
@@ -0,0 +1,19 @@
+noinst_PROGRAMS = test-cleanup test-reuse
+
+AM_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS)
+AM_CPPFLAGS = -I$(top_srcdir) -I$(srcdir)
+AM_LDFLAGS = \
+	$(GST_LIBS) \
+	$(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la
+
+if BUILD_TESTS
+SUBDIRS_CHECK = check
+else
+SUBDIRS_CHECK =
+endif
+
+SUBDIRS = 			\
+	$(SUBDIRS_CHECK)
+
+DIST_SUBDIRS = 			\
+	check
diff --git a/tests/Makefile.in b/tests/Makefile.in
new file mode 100644
index 0000000..63678e5
--- /dev/null
+++ b/tests/Makefile.in
@@ -0,0 +1,840 @@
+# Makefile.in generated by automake 1.14.1 from Makefile.am.
+# @configure_input@
+
+# Copyright (C) 1994-2013 Free Software Foundation, Inc.
+
+# This Makefile.in is free software; the Free Software Foundation
+# gives unlimited permission to copy and/or distribute it,
+# with or without modifications, as long as this notice is preserved.
+
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
+# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
+# PARTICULAR PURPOSE.
+
+@SET_MAKE@
+
+VPATH = @srcdir@
+am__is_gnu_make = test -n '$(MAKEFILE_LIST)' && test -n '$(MAKELEVEL)'
+am__make_running_with_option = \
+  case $${target_option-} in \
+      ?) ;; \
+      *) echo "am__make_running_with_option: internal error: invalid" \
+              "target option '$${target_option-}' specified" >&2; \
+         exit 1;; \
+  esac; \
+  has_opt=no; \
+  sane_makeflags=$$MAKEFLAGS; \
+  if $(am__is_gnu_make); then \
+    sane_makeflags=$$MFLAGS; \
+  else \
+    case $$MAKEFLAGS in \
+      *\\[\ \	]*) \
+        bs=\\; \
+        sane_makeflags=`printf '%s\n' "$$MAKEFLAGS" \
+          | sed "s/$$bs$$bs[$$bs $$bs	]*//g"`;; \
+    esac; \
+  fi; \
+  skip_next=no; \
+  strip_trailopt () \
+  { \
+    flg=`printf '%s\n' "$$flg" | sed "s/$$1.*$$//"`; \
+  }; \
+  for flg in $$sane_makeflags; do \
+    test $$skip_next = yes && { skip_next=no; continue; }; \
+    case $$flg in \
+      *=*|--*) continue;; \
+        -*I) strip_trailopt 'I'; skip_next=yes;; \
+      -*I?*) strip_trailopt 'I';; \
+        -*O) strip_trailopt 'O'; skip_next=yes;; \
+      -*O?*) strip_trailopt 'O';; \
+        -*l) strip_trailopt 'l'; skip_next=yes;; \
+      -*l?*) strip_trailopt 'l';; \
+      -[dEDm]) skip_next=yes;; \
+      -[JT]) skip_next=yes;; \
+    esac; \
+    case $$flg in \
+      *$$target_option*) has_opt=yes; break;; \
+    esac; \
+  done; \
+  test $$has_opt = yes
+am__make_dryrun = (target_option=n; $(am__make_running_with_option))
+am__make_keepgoing = (target_option=k; $(am__make_running_with_option))
+pkgdatadir = $(datadir)/@PACKAGE@
+pkgincludedir = $(includedir)/@PACKAGE@
+pkglibdir = $(libdir)/@PACKAGE@
+pkglibexecdir = $(libexecdir)/@PACKAGE@
+am__cd = CDPATH="$${ZSH_VERSION+.}$(PATH_SEPARATOR)" && cd
+install_sh_DATA = $(install_sh) -c -m 644
+install_sh_PROGRAM = $(install_sh) -c
+install_sh_SCRIPT = $(install_sh) -c
+INSTALL_HEADER = $(INSTALL_DATA)
+transform = $(program_transform_name)
+NORMAL_INSTALL = :
+PRE_INSTALL = :
+POST_INSTALL = :
+NORMAL_UNINSTALL = :
+PRE_UNINSTALL = :
+POST_UNINSTALL = :
+build_triplet = @build@
+host_triplet = @host@
+target_triplet = @target@
+noinst_PROGRAMS = test-cleanup$(EXEEXT) test-reuse$(EXEEXT)
+subdir = tests
+DIST_COMMON = $(srcdir)/Makefile.in $(srcdir)/Makefile.am \
+	$(top_srcdir)/depcomp
+ACLOCAL_M4 = $(top_srcdir)/aclocal.m4
+am__aclocal_m4_deps = $(top_srcdir)/common/m4/as-ac-expand.m4 \
+	$(top_srcdir)/common/m4/as-auto-alt.m4 \
+	$(top_srcdir)/common/m4/as-compiler-flag.m4 \
+	$(top_srcdir)/common/m4/as-docbook.m4 \
+	$(top_srcdir)/common/m4/as-libtool.m4 \
+	$(top_srcdir)/common/m4/as-version.m4 \
+	$(top_srcdir)/common/m4/gst-args.m4 \
+	$(top_srcdir)/common/m4/gst-check.m4 \
+	$(top_srcdir)/common/m4/gst-default.m4 \
+	$(top_srcdir)/common/m4/gst-doc.m4 \
+	$(top_srcdir)/common/m4/gst-error.m4 \
+	$(top_srcdir)/common/m4/gst-feature.m4 \
+	$(top_srcdir)/common/m4/gst-glib2.m4 \
+	$(top_srcdir)/common/m4/gst-package-release-datetime.m4 \
+	$(top_srcdir)/common/m4/gst-plugindir.m4 \
+	$(top_srcdir)/common/m4/gst.m4 \
+	$(top_srcdir)/common/m4/gtk-doc.m4 \
+	$(top_srcdir)/common/m4/introspection.m4 \
+	$(top_srcdir)/common/m4/pkg.m4 $(top_srcdir)/m4/libtool.m4 \
+	$(top_srcdir)/m4/ltoptions.m4 $(top_srcdir)/m4/ltsugar.m4 \
+	$(top_srcdir)/m4/ltversion.m4 $(top_srcdir)/m4/lt~obsolete.m4 \
+	$(top_srcdir)/acinclude.m4 $(top_srcdir)/configure.ac
+am__configure_deps = $(am__aclocal_m4_deps) $(CONFIGURE_DEPENDENCIES) \
+	$(ACLOCAL_M4)
+mkinstalldirs = $(install_sh) -d
+CONFIG_HEADER = $(top_builddir)/config.h
+CONFIG_CLEAN_FILES =
+CONFIG_CLEAN_VPATH_FILES =
+PROGRAMS = $(noinst_PROGRAMS)
+test_cleanup_SOURCES = test-cleanup.c
+test_cleanup_OBJECTS = test-cleanup.$(OBJEXT)
+test_cleanup_LDADD = $(LDADD)
+AM_V_lt = $(am__v_lt_@AM_V@)
+am__v_lt_ = $(am__v_lt_@AM_DEFAULT_V@)
+am__v_lt_0 = --silent
+am__v_lt_1 = 
+test_reuse_SOURCES = test-reuse.c
+test_reuse_OBJECTS = test-reuse.$(OBJEXT)
+test_reuse_LDADD = $(LDADD)
+AM_V_P = $(am__v_P_@AM_V@)
+am__v_P_ = $(am__v_P_@AM_DEFAULT_V@)
+am__v_P_0 = false
+am__v_P_1 = :
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+	  if test $$# -gt 0; then \
+	    $(ETAGS) $(ETAGSFLAGS) $(AM_ETAGSFLAGS) $(ETAGS_ARGS) \
+	      "$$@" $$unique; \
+	  else \
+	    $(ETAGS) $(ETAGSFLAGS) $(AM_ETAGSFLAGS) $(ETAGS_ARGS) \
+	      $$unique; \
+	  fi; \
+	fi
+ctags: ctags-recursive
+
+CTAGS: ctags
+ctags-am: $(TAGS_DEPENDENCIES) $(am__tagged_files)
+	$(am__define_uniq_tagged_files); \
+	test -z "$(CTAGS_ARGS)$$unique" \
+	  || $(CTAGS) $(CTAGSFLAGS) $(AM_CTAGSFLAGS) $(CTAGS_ARGS) \
+	     $$unique
+
+GTAGS:
+	here=`$(am__cd) $(top_builddir) && pwd` \
+	  && $(am__cd) $(top_srcdir) \
+	  && gtags -i $(GTAGS_ARGS) "$$here"
+cscopelist: cscopelist-recursive
+
+cscopelist-am: $(am__tagged_files)
+	list='$(am__tagged_files)'; \
+	case "$(srcdir)" in \
+	  [\\/]* | ?:[\\/]*) sdir="$(srcdir)" ;; \
+	  *) sdir=$(subdir)/$(srcdir) ;; \
+	esac; \
+	for i in $$list; do \
+	  if test -f "$$i"; then \
+	    echo "$(subdir)/$$i"; \
+	  else \
+	    echo "$$sdir/$$i"; \
+	  fi; \
+	done >> $(top_builddir)/cscope.files
+
+distclean-tags:
+	-rm -f TAGS ID GTAGS GRTAGS GSYMS GPATH tags
+
+distdir: $(DISTFILES)
+	@srcdirstrip=`echo "$(srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+	topsrcdirstrip=`echo "$(top_srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+	list='$(DISTFILES)'; \
+	  dist_files=`for file in $$list; do echo $$file; done | \
+	  sed -e "s|^$$srcdirstrip/||;t" \
+	      -e "s|^$$topsrcdirstrip/|$(top_builddir)/|;t"`; \
+	case $$dist_files in \
+	  */*) $(MKDIR_P) `echo "$$dist_files" | \
+			   sed '/\//!d;s|^|$(distdir)/|;s,/[^/]*$$,,' | \
+			   sort -u` ;; \
+	esac; \
+	for file in $$dist_files; do \
+	  if test -f $$file || test -d $$file; then d=.; else d=$(srcdir); fi; \
+	  if test -d $$d/$$file; then \
+	    dir=`echo "/$$file" | sed -e 's,/[^/]*$$,,'`; \
+	    if test -d "$(distdir)/$$file"; then \
+	      find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+	    fi; \
+	    if test -d $(srcdir)/$$file && test $$d != $(srcdir); then \
+	      cp -fpR $(srcdir)/$$file "$(distdir)$$dir" || exit 1; \
+	      find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+	    fi; \
+	    cp -fpR $$d/$$file "$(distdir)$$dir" || exit 1; \
+	  else \
+	    test -f "$(distdir)/$$file" \
+	    || cp -p $$d/$$file "$(distdir)/$$file" \
+	    || exit 1; \
+	  fi; \
+	done
+	@list='$(DIST_SUBDIRS)'; for subdir in $$list; do \
+	  if test "$$subdir" = .; then :; else \
+	    $(am__make_dryrun) \
+	      || test -d "$(distdir)/$$subdir" \
+	      || $(MKDIR_P) "$(distdir)/$$subdir" \
+	      || exit 1; \
+	    dir1=$$subdir; dir2="$(distdir)/$$subdir"; \
+	    $(am__relativize); \
+	    new_distdir=$$reldir; \
+	    dir1=$$subdir; dir2="$(top_distdir)"; \
+	    $(am__relativize); \
+	    new_top_distdir=$$reldir; \
+	    echo " (cd $$subdir && $(MAKE) $(AM_MAKEFLAGS) top_distdir="$$new_top_distdir" distdir="$$new_distdir" \\"; \
+	    echo "     am__remove_distdir=: am__skip_length_check=: am__skip_mode_fix=: distdir)"; \
+	    ($(am__cd) $$subdir && \
+	      $(MAKE) $(AM_MAKEFLAGS) \
+	        top_distdir="$$new_top_distdir" \
+	        distdir="$$new_distdir" \
+		am__remove_distdir=: \
+		am__skip_length_check=: \
+		am__skip_mode_fix=: \
+	        distdir) \
+	      || exit 1; \
+	  fi; \
+	done
+check-am: all-am
+check: check-recursive
+all-am: Makefile $(PROGRAMS)
+installdirs: installdirs-recursive
+installdirs-am:
+install: install-recursive
+install-exec: install-exec-recursive
+install-data: install-data-recursive
+uninstall: uninstall-recursive
+
+install-am: all-am
+	@$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
+
+installcheck: installcheck-recursive
+install-strip:
+	if test -z '$(STRIP)'; then \
+	  $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	    install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	      install; \
+	else \
+	  $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	    install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	    "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'" install; \
+	fi
+mostlyclean-generic:
+
+clean-generic:
+
+distclean-generic:
+	-test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
+	-test . = "$(srcdir)" || test -z "$(CONFIG_CLEAN_VPATH_FILES)" || rm -f $(CONFIG_CLEAN_VPATH_FILES)
+
+maintainer-clean-generic:
+	@echo "This command is intended for maintainers to use"
+	@echo "it deletes files that may require special tools to rebuild."
+clean: clean-recursive
+
+clean-am: clean-generic clean-libtool clean-noinstPROGRAMS \
+	mostlyclean-am
+
+distclean: distclean-recursive
+	-rm -rf ./$(DEPDIR)
+	-rm -f Makefile
+distclean-am: clean-am distclean-compile distclean-generic \
+	distclean-tags
+
+dvi: dvi-recursive
+
+dvi-am:
+
+html: html-recursive
+
+html-am:
+
+info: info-recursive
+
+info-am:
+
+install-data-am:
+
+install-dvi: install-dvi-recursive
+
+install-dvi-am:
+
+install-exec-am:
+
+install-html: install-html-recursive
+
+install-html-am:
+
+install-info: install-info-recursive
+
+install-info-am:
+
+install-man:
+
+install-pdf: install-pdf-recursive
+
+install-pdf-am:
+
+install-ps: install-ps-recursive
+
+install-ps-am:
+
+installcheck-am:
+
+maintainer-clean: maintainer-clean-recursive
+	-rm -rf ./$(DEPDIR)
+	-rm -f Makefile
+maintainer-clean-am: distclean-am maintainer-clean-generic
+
+mostlyclean: mostlyclean-recursive
+
+mostlyclean-am: mostlyclean-compile mostlyclean-generic \
+	mostlyclean-libtool
+
+pdf: pdf-recursive
+
+pdf-am:
+
+ps: ps-recursive
+
+ps-am:
+
+uninstall-am:
+
+.MAKE: $(am__recursive_targets) install-am install-strip
+
+.PHONY: $(am__recursive_targets) CTAGS GTAGS TAGS all all-am check \
+	check-am clean clean-generic clean-libtool \
+	clean-noinstPROGRAMS cscopelist-am ctags ctags-am distclean \
+	distclean-compile distclean-generic distclean-libtool \
+	distclean-tags distdir dvi dvi-am html html-am info info-am \
+	install install-am install-data install-data-am install-dvi \
+	install-dvi-am install-exec install-exec-am install-html \
+	install-html-am install-info install-info-am install-man \
+	install-pdf install-pdf-am install-ps install-ps-am \
+	install-strip installcheck installcheck-am installdirs \
+	installdirs-am maintainer-clean maintainer-clean-generic \
+	mostlyclean mostlyclean-compile mostlyclean-generic \
+	mostlyclean-libtool pdf pdf-am ps ps-am tags tags-am uninstall \
+	uninstall-am
+
+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am
new file mode 100644
index 0000000..82e6d42
--- /dev/null
+++ b/tests/check/Makefile.am
@@ -0,0 +1,62 @@
+include $(top_srcdir)/common/check.mak
+
+CHECK_REGISTRY = $(top_builddir)/tests/check/test-registry.reg
+TEST_FILES_DIRECTORY = $(top_srcdir)/tests/files
+
+REGISTRY_ENVIRONMENT = \
+	GST_REGISTRY_1_0=$(CHECK_REGISTRY)
+
+TESTS_ENVIRONMENT = \
+        CK_DEFAULT_TIMEOUT=120                                  \
+        GST_STATE_IGNORE_ELEMENTS="$(STATE_IGNORE_ELEMENTS)"	\
+	$(REGISTRY_ENVIRONMENT)					\
+	GST_PLUGIN_SYSTEM_PATH_1_0=					\
+	GST_PLUGIN_PATH_1_0=$(GST_PLUGINS_DIR):$(GSTPB_PLUGINS_DIR):$(GSTPG_PLUGINS_DIR):$(GSTPD_PLUGINS_DIR)	\
+	GST_PLUGIN_LOADING_WHITELIST="gstreamer:gst-plugins-base:gst-plugins-good:gst-plugins-bad"
+
+
+# ths core dumps of some machines have PIDs appended
+CLEANFILES = core.* test-registry.*
+
+clean-local: clean-local-check
+
+$(CHECK_REGISTRY):
+	$(TESTS_ENVIRONMENT)
+
+TESTS = $(check_PROGRAMS)
+
+check_PROGRAMS = \
+	gst/rtspserver \
+	gst/client \
+	gst/mountpoints \
+	gst/mediafactory \
+	gst/media \
+	gst/stream \
+	gst/addresspool \
+	gst/threadpool \
+	gst/permissions \
+	gst/token \
+	gst/sessionmedia \
+	gst/sessionpool
+
+# these tests don't even pass
+noinst_PROGRAMS =
+
+AM_CFLAGS = -I$(top_srcdir)/gst/rtsp-server \
+	$(GST_PLUGINS_BASE_CFLAGS) \
+        $(GST_BASE_CFLAGS) \
+        $(GIO_CFLAGS) \
+	$(GST_CFLAGS) \
+	$(GST_CHECK_CFLAGS) \
+	-DGST_TEST_FILES_PATH="\"$(TEST_FILES_DIRECTORY)\"" \
+	-UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS
+AM_CXXFLAGS = $(GST_CXXFLAGS) $(GST_CHECK_CFLAGS) \
+	-DGST_TEST_FILES_PATH="\"$(TEST_FILES_DIRECTORY)\"" \
+	-UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS
+LDADD = $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la \
+	$(GST_PLUGINS_BASE_LIBS) -lgstrtp-@GST_API_VERSION@ \
+	-lgstrtsp-@GST_API_VERSION@ -lgstsdp-@GST_API_VERSION@ \
+	$(GST_BASE_LIBS) $(GIO_LIBS) \
+	$(GST_LIBS) $(GST_CHECK_LIBS) $(GST_RTSP_SERVER_LIBS)
+
+SUPPRESSIONS = $(top_srcdir)/common/gst.supp
diff --git a/tests/check/Makefile.in b/tests/check/Makefile.in
new file mode 100644
index 0000000..a68c6ca
--- /dev/null
+++ b/tests/check/Makefile.in
@@ -0,0 +1,1635 @@
+# Makefile.in generated by automake 1.14.1 from Makefile.am.
+# @configure_input@
+
+# Copyright (C) 1994-2013 Free Software Foundation, Inc.
+
+# This Makefile.in is free software; the Free Software Foundation
+# gives unlimited permission to copy and/or distribute it,
+# with or without modifications, as long as this notice is preserved.
+
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
+# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
+# PARTICULAR PURPOSE.
+
+@SET_MAKE@
+
+VPATH = @srcdir@
+am__is_gnu_make = test -n '$(MAKEFILE_LIST)' && test -n '$(MAKELEVEL)'
+am__make_running_with_option = \
+  case $${target_option-} in \
+      ?) ;; \
+      *) echo "am__make_running_with_option: internal error: invalid" \
+              "target option '$${target_option-}' specified" >&2; \
+         exit 1;; \
+  esac; \
+  has_opt=no; \
+  sane_makeflags=$$MAKEFLAGS; \
+  if $(am__is_gnu_make); then \
+    sane_makeflags=$$MFLAGS; \
+  else \
+    case $$MAKEFLAGS in \
+      *\\[\ \	]*) \
+        bs=\\; \
+        sane_makeflags=`printf '%s\n' "$$MAKEFLAGS" \
+          | sed "s/$$bs$$bs[$$bs $$bs	]*//g"`;; \
+    esac; \
+  fi; \
+  skip_next=no; \
+  strip_trailopt () \
+  { \
+    flg=`printf '%s\n' "$$flg" | sed "s/$$1.*$$//"`; \
+  }; \
+  for flg in $$sane_makeflags; do \
+    test $$skip_next = yes && { skip_next=no; continue; }; \
+    case $$flg in \
+      *=*|--*) continue;; \
+        -*I) strip_trailopt 'I'; skip_next=yes;; \
+      -*I?*) strip_trailopt 'I';; \
+        -*O) strip_trailopt 'O'; skip_next=yes;; \
+      -*O?*) strip_trailopt 'O';; \
+        -*l) strip_trailopt 'l'; skip_next=yes;; \
+      -*l?*) strip_trailopt 'l';; \
+      -[dEDm]) skip_next=yes;; \
+      -[JT]) skip_next=yes;; \
+    esac; \
+    case $$flg in \
+      *$$target_option*) has_opt=yes; break;; \
+    esac; \
+  done; \
+  test $$has_opt = yes
+am__make_dryrun = (target_option=n; $(am__make_running_with_option))
+am__make_keepgoing = (target_option=k; $(am__make_running_with_option))
+pkgdatadir = $(datadir)/@PACKAGE@
+pkgincludedir = $(includedir)/@PACKAGE@
+pkglibdir = $(libdir)/@PACKAGE@
+pkglibexecdir = $(libexecdir)/@PACKAGE@
+am__cd = CDPATH="$${ZSH_VERSION+.}$(PATH_SEPARATOR)" && cd
+install_sh_DATA = $(install_sh) -c -m 644
+install_sh_PROGRAM = $(install_sh) -c
+install_sh_SCRIPT = $(install_sh) -c
+INSTALL_HEADER = $(INSTALL_DATA)
+transform = $(program_transform_name)
+NORMAL_INSTALL = :
+PRE_INSTALL = :
+POST_INSTALL = :
+NORMAL_UNINSTALL = :
+PRE_UNINSTALL = :
+POST_UNINSTALL = :
+build_triplet = @build@
+host_triplet = @host@
+target_triplet = @target@
+DIST_COMMON = $(top_srcdir)/common/check.mak $(srcdir)/Makefile.in \
+	$(srcdir)/Makefile.am $(top_srcdir)/depcomp \
+	$(top_srcdir)/test-driver
+check_PROGRAMS = gst/rtspserver$(EXEEXT) gst/client$(EXEEXT) \
+	gst/mountpoints$(EXEEXT) gst/mediafactory$(EXEEXT) \
+	gst/media$(EXEEXT) gst/stream$(EXEEXT) \
+	gst/addresspool$(EXEEXT) gst/threadpool$(EXEEXT) \
+	gst/permissions$(EXEEXT) gst/token$(EXEEXT) \
+	gst/sessionmedia$(EXEEXT) gst/sessionpool$(EXEEXT)
+noinst_PROGRAMS =
+subdir = tests/check
+ACLOCAL_M4 = $(top_srcdir)/aclocal.m4
+am__aclocal_m4_deps = $(top_srcdir)/common/m4/as-ac-expand.m4 \
+	$(top_srcdir)/common/m4/as-auto-alt.m4 \
+	$(top_srcdir)/common/m4/as-compiler-flag.m4 \
+	$(top_srcdir)/common/m4/as-docbook.m4 \
+	$(top_srcdir)/common/m4/as-libtool.m4 \
+	$(top_srcdir)/common/m4/as-version.m4 \
+	$(top_srcdir)/common/m4/gst-args.m4 \
+	$(top_srcdir)/common/m4/gst-check.m4 \
+	$(top_srcdir)/common/m4/gst-default.m4 \
+	$(top_srcdir)/common/m4/gst-doc.m4 \
+	$(top_srcdir)/common/m4/gst-error.m4 \
+	$(top_srcdir)/common/m4/gst-feature.m4 \
+	$(top_srcdir)/common/m4/gst-glib2.m4 \
+	$(top_srcdir)/common/m4/gst-package-release-datetime.m4 \
+	$(top_srcdir)/common/m4/gst-plugindir.m4 \
+	$(top_srcdir)/common/m4/gst.m4 \
+	$(top_srcdir)/common/m4/gtk-doc.m4 \
+	$(top_srcdir)/common/m4/introspection.m4 \
+	$(top_srcdir)/common/m4/pkg.m4 $(top_srcdir)/m4/libtool.m4 \
+	$(top_srcdir)/m4/ltoptions.m4 $(top_srcdir)/m4/ltsugar.m4 \
+	$(top_srcdir)/m4/ltversion.m4 $(top_srcdir)/m4/lt~obsolete.m4 \
+	$(top_srcdir)/acinclude.m4 $(top_srcdir)/configure.ac
+am__configure_deps = $(am__aclocal_m4_deps) $(CONFIGURE_DEPENDENCIES) \
+	$(ACLOCAL_M4)
+mkinstalldirs = $(install_sh) -d
+CONFIG_HEADER = $(top_builddir)/config.h
+CONFIG_CLEAN_FILES =
+CONFIG_CLEAN_VPATH_FILES =
+PROGRAMS = $(noinst_PROGRAMS)
+gst_addresspool_SOURCES = gst/addresspool.c
+am__dirstamp = $(am__leading_dot)dirstamp
+gst_addresspool_OBJECTS = gst/addresspool.$(OBJEXT)
+gst_addresspool_LDADD = $(LDADD)
+am__DEPENDENCIES_1 =
+gst_addresspool_DEPENDENCIES = $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1)
+AM_V_lt = $(am__v_lt_@AM_V@)
+am__v_lt_ = $(am__v_lt_@AM_DEFAULT_V@)
+am__v_lt_0 = --silent
+am__v_lt_1 = 
+gst_client_SOURCES = gst/client.c
+gst_client_OBJECTS = gst/client.$(OBJEXT)
+gst_client_LDADD = $(LDADD)
+gst_client_DEPENDENCIES = $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1)
+gst_media_SOURCES = gst/media.c
+gst_media_OBJECTS = gst/media.$(OBJEXT)
+gst_media_LDADD = $(LDADD)
+gst_media_DEPENDENCIES = $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1)
+gst_mediafactory_SOURCES = gst/mediafactory.c
+gst_mediafactory_OBJECTS = gst/mediafactory.$(OBJEXT)
+gst_mediafactory_LDADD = $(LDADD)
+gst_mediafactory_DEPENDENCIES = $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1)
+gst_mountpoints_SOURCES = gst/mountpoints.c
+gst_mountpoints_OBJECTS = gst/mountpoints.$(OBJEXT)
+gst_mountpoints_LDADD = $(LDADD)
+gst_mountpoints_DEPENDENCIES = $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1)
+gst_permissions_SOURCES = gst/permissions.c
+gst_permissions_OBJECTS = gst/permissions.$(OBJEXT)
+gst_permissions_LDADD = $(LDADD)
+gst_permissions_DEPENDENCIES = $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1)
+gst_rtspserver_SOURCES = gst/rtspserver.c
+gst_rtspserver_OBJECTS = gst/rtspserver.$(OBJEXT)
+gst_rtspserver_LDADD = $(LDADD)
+gst_rtspserver_DEPENDENCIES = $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1)
+gst_sessionmedia_SOURCES = gst/sessionmedia.c
+gst_sessionmedia_OBJECTS = gst/sessionmedia.$(OBJEXT)
+gst_sessionmedia_LDADD = $(LDADD)
+gst_sessionmedia_DEPENDENCIES = $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1)
+gst_sessionpool_SOURCES = gst/sessionpool.c
+gst_sessionpool_OBJECTS = gst/sessionpool.$(OBJEXT)
+gst_sessionpool_LDADD = $(LDADD)
+gst_sessionpool_DEPENDENCIES = $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1)
+gst_stream_SOURCES = gst/stream.c
+gst_stream_OBJECTS = gst/stream.$(OBJEXT)
+gst_stream_LDADD = $(LDADD)
+gst_stream_DEPENDENCIES = $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1)
+gst_threadpool_SOURCES = gst/threadpool.c
+gst_threadpool_OBJECTS = gst/threadpool.$(OBJEXT)
+gst_threadpool_LDADD = $(LDADD)
+gst_threadpool_DEPENDENCIES = $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1) $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1)
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+	gst/stream.c gst/threadpool.c gst/token.c
+DIST_SOURCES = gst/addresspool.c gst/client.c gst/media.c \
+	gst/mediafactory.c gst/mountpoints.c gst/permissions.c \
+	gst/rtspserver.c gst/sessionmedia.c gst/sessionpool.c \
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+target = @target@
+target_alias = @target_alias@
+target_cpu = @target_cpu@
+target_os = @target_os@
+target_vendor = @target_vendor@
+top_build_prefix = @top_build_prefix@
+top_builddir = @top_builddir@
+top_srcdir = @top_srcdir@
+
+# inspect every plugin feature
+GST_INSPECT = $(GST_TOOLS_DIR)/gst-inspect-$(GST_API_VERSION)
+CHECK_REGISTRY = $(top_builddir)/tests/check/test-registry.reg
+TEST_FILES_DIRECTORY = $(top_srcdir)/tests/files
+REGISTRY_ENVIRONMENT = \
+	GST_REGISTRY_1_0=$(CHECK_REGISTRY)
+
+TESTS_ENVIRONMENT = \
+        CK_DEFAULT_TIMEOUT=120                                  \
+        GST_STATE_IGNORE_ELEMENTS="$(STATE_IGNORE_ELEMENTS)"	\
+	$(REGISTRY_ENVIRONMENT)					\
+	GST_PLUGIN_SYSTEM_PATH_1_0=					\
+	GST_PLUGIN_PATH_1_0=$(GST_PLUGINS_DIR):$(GSTPB_PLUGINS_DIR):$(GSTPG_PLUGINS_DIR):$(GSTPD_PLUGINS_DIR)	\
+	GST_PLUGIN_LOADING_WHITELIST="gstreamer:gst-plugins-base:gst-plugins-good:gst-plugins-bad"
+
+
+# ths core dumps of some machines have PIDs appended
+CLEANFILES = core.* test-registry.*
+TESTS = $(check_PROGRAMS)
+AM_CFLAGS = -I$(top_srcdir)/gst/rtsp-server \
+	$(GST_PLUGINS_BASE_CFLAGS) \
+        $(GST_BASE_CFLAGS) \
+        $(GIO_CFLAGS) \
+	$(GST_CFLAGS) \
+	$(GST_CHECK_CFLAGS) \
+	-DGST_TEST_FILES_PATH="\"$(TEST_FILES_DIRECTORY)\"" \
+	-UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS
+
+AM_CXXFLAGS = $(GST_CXXFLAGS) $(GST_CHECK_CFLAGS) \
+	-DGST_TEST_FILES_PATH="\"$(TEST_FILES_DIRECTORY)\"" \
+	-UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS
+
+LDADD = $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la \
+	$(GST_PLUGINS_BASE_LIBS) -lgstrtp-@GST_API_VERSION@ \
+	-lgstrtsp-@GST_API_VERSION@ -lgstsdp-@GST_API_VERSION@ \
+	$(GST_BASE_LIBS) $(GIO_LIBS) \
+	$(GST_LIBS) $(GST_CHECK_LIBS) $(GST_RTSP_SERVER_LIBS)
+
+SUPPRESSIONS = $(top_srcdir)/common/gst.supp
+all: all-am
+
+.SUFFIXES:
+.SUFFIXES: .c .lo .log .o .obj .test .test$(EXEEXT) .trs
+$(srcdir)/Makefile.in: @MAINTAINER_MODE_TRUE@ $(srcdir)/Makefile.am $(top_srcdir)/common/check.mak $(am__configure_deps)
+	@for dep in $?; do \
+	  case '$(am__configure_deps)' in \
+	    *$$dep*) \
+	      ( cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh ) \
+	        && { if test -f $@; then exit 0; else break; fi; }; \
+	      exit 1;; \
+	  esac; \
+	done; \
+	echo ' cd $(top_srcdir) && $(AUTOMAKE) --gnu tests/check/Makefile'; \
+	$(am__cd) $(top_srcdir) && \
+	  $(AUTOMAKE) --gnu tests/check/Makefile
+.PRECIOUS: Makefile
+Makefile: $(srcdir)/Makefile.in $(top_builddir)/config.status
+	@case '$?' in \
+	  *config.status*) \
+	    cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh;; \
+	  *) \
+	    echo ' cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@ $(am__depfiles_maybe)'; \
+	    cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@ $(am__depfiles_maybe);; \
+	esac;
+$(top_srcdir)/common/check.mak:
+
+$(top_builddir)/config.status: $(top_srcdir)/configure $(CONFIG_STATUS_DEPENDENCIES)
+	cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+
+$(top_srcdir)/configure: @MAINTAINER_MODE_TRUE@ $(am__configure_deps)
+	cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+$(ACLOCAL_M4): @MAINTAINER_MODE_TRUE@ $(am__aclocal_m4_deps)
+	cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+$(am__aclocal_m4_deps):
+
+clean-checkPROGRAMS:
+	@list='$(check_PROGRAMS)'; test -n "$$list" || exit 0; \
+	echo " rm -f" $$list; \
+	rm -f $$list || exit $$?; \
+	test -n "$(EXEEXT)" || exit 0; \
+	list=`for p in $$list; do echo "$$p"; done | sed 's/$(EXEEXT)$$//'`; \
+	echo " rm -f" $$list; \
+	rm -f $$list
+
+clean-noinstPROGRAMS:
+	@list='$(noinst_PROGRAMS)'; test -n "$$list" || exit 0; \
+	echo " rm -f" $$list; \
+	rm -f $$list || exit $$?; \
+	test -n "$(EXEEXT)" || exit 0; \
+	list=`for p in $$list; do echo "$$p"; done | sed 's/$(EXEEXT)$$//'`; \
+	echo " rm -f" $$list; \
+	rm -f $$list
+gst/$(am__dirstamp):
+	@$(MKDIR_P) gst
+	@: > gst/$(am__dirstamp)
+gst/$(DEPDIR)/$(am__dirstamp):
+	@$(MKDIR_P) gst/$(DEPDIR)
+	@: > gst/$(DEPDIR)/$(am__dirstamp)
+gst/addresspool.$(OBJEXT): gst/$(am__dirstamp) \
+	gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/addresspool$(EXEEXT): $(gst_addresspool_OBJECTS) $(gst_addresspool_DEPENDENCIES) $(EXTRA_gst_addresspool_DEPENDENCIES) gst/$(am__dirstamp)
+	@rm -f gst/addresspool$(EXEEXT)
+	$(AM_V_CCLD)$(LINK) $(gst_addresspool_OBJECTS) $(gst_addresspool_LDADD) $(LIBS)
+gst/client.$(OBJEXT): gst/$(am__dirstamp) \
+	gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/client$(EXEEXT): $(gst_client_OBJECTS) $(gst_client_DEPENDENCIES) $(EXTRA_gst_client_DEPENDENCIES) gst/$(am__dirstamp)
+	@rm -f gst/client$(EXEEXT)
+	$(AM_V_CCLD)$(LINK) $(gst_client_OBJECTS) $(gst_client_LDADD) $(LIBS)
+gst/media.$(OBJEXT): gst/$(am__dirstamp) gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/media$(EXEEXT): $(gst_media_OBJECTS) $(gst_media_DEPENDENCIES) $(EXTRA_gst_media_DEPENDENCIES) gst/$(am__dirstamp)
+	@rm -f gst/media$(EXEEXT)
+	$(AM_V_CCLD)$(LINK) $(gst_media_OBJECTS) $(gst_media_LDADD) $(LIBS)
+gst/mediafactory.$(OBJEXT): gst/$(am__dirstamp) \
+	gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/mediafactory$(EXEEXT): $(gst_mediafactory_OBJECTS) $(gst_mediafactory_DEPENDENCIES) $(EXTRA_gst_mediafactory_DEPENDENCIES) gst/$(am__dirstamp)
+	@rm -f gst/mediafactory$(EXEEXT)
+	$(AM_V_CCLD)$(LINK) $(gst_mediafactory_OBJECTS) $(gst_mediafactory_LDADD) $(LIBS)
+gst/mountpoints.$(OBJEXT): gst/$(am__dirstamp) \
+	gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/mountpoints$(EXEEXT): $(gst_mountpoints_OBJECTS) $(gst_mountpoints_DEPENDENCIES) $(EXTRA_gst_mountpoints_DEPENDENCIES) gst/$(am__dirstamp)
+	@rm -f gst/mountpoints$(EXEEXT)
+	$(AM_V_CCLD)$(LINK) $(gst_mountpoints_OBJECTS) $(gst_mountpoints_LDADD) $(LIBS)
+gst/permissions.$(OBJEXT): gst/$(am__dirstamp) \
+	gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/permissions$(EXEEXT): $(gst_permissions_OBJECTS) $(gst_permissions_DEPENDENCIES) $(EXTRA_gst_permissions_DEPENDENCIES) gst/$(am__dirstamp)
+	@rm -f gst/permissions$(EXEEXT)
+	$(AM_V_CCLD)$(LINK) $(gst_permissions_OBJECTS) $(gst_permissions_LDADD) $(LIBS)
+gst/rtspserver.$(OBJEXT): gst/$(am__dirstamp) \
+	gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/rtspserver$(EXEEXT): $(gst_rtspserver_OBJECTS) $(gst_rtspserver_DEPENDENCIES) $(EXTRA_gst_rtspserver_DEPENDENCIES) gst/$(am__dirstamp)
+	@rm -f gst/rtspserver$(EXEEXT)
+	$(AM_V_CCLD)$(LINK) $(gst_rtspserver_OBJECTS) $(gst_rtspserver_LDADD) $(LIBS)
+gst/sessionmedia.$(OBJEXT): gst/$(am__dirstamp) \
+	gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/sessionmedia$(EXEEXT): $(gst_sessionmedia_OBJECTS) $(gst_sessionmedia_DEPENDENCIES) $(EXTRA_gst_sessionmedia_DEPENDENCIES) gst/$(am__dirstamp)
+	@rm -f gst/sessionmedia$(EXEEXT)
+	$(AM_V_CCLD)$(LINK) $(gst_sessionmedia_OBJECTS) $(gst_sessionmedia_LDADD) $(LIBS)
+gst/sessionpool.$(OBJEXT): gst/$(am__dirstamp) \
+	gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/sessionpool$(EXEEXT): $(gst_sessionpool_OBJECTS) $(gst_sessionpool_DEPENDENCIES) $(EXTRA_gst_sessionpool_DEPENDENCIES) gst/$(am__dirstamp)
+	@rm -f gst/sessionpool$(EXEEXT)
+	$(AM_V_CCLD)$(LINK) $(gst_sessionpool_OBJECTS) $(gst_sessionpool_LDADD) $(LIBS)
+gst/stream.$(OBJEXT): gst/$(am__dirstamp) \
+	gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/stream$(EXEEXT): $(gst_stream_OBJECTS) $(gst_stream_DEPENDENCIES) $(EXTRA_gst_stream_DEPENDENCIES) gst/$(am__dirstamp)
+	@rm -f gst/stream$(EXEEXT)
+	$(AM_V_CCLD)$(LINK) $(gst_stream_OBJECTS) $(gst_stream_LDADD) $(LIBS)
+gst/threadpool.$(OBJEXT): gst/$(am__dirstamp) \
+	gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/threadpool$(EXEEXT): $(gst_threadpool_OBJECTS) $(gst_threadpool_DEPENDENCIES) $(EXTRA_gst_threadpool_DEPENDENCIES) gst/$(am__dirstamp)
+	@rm -f gst/threadpool$(EXEEXT)
+	$(AM_V_CCLD)$(LINK) $(gst_threadpool_OBJECTS) $(gst_threadpool_LDADD) $(LIBS)
+gst/token.$(OBJEXT): gst/$(am__dirstamp) gst/$(DEPDIR)/$(am__dirstamp)
+
+gst/token$(EXEEXT): $(gst_token_OBJECTS) $(gst_token_DEPENDENCIES) $(EXTRA_gst_token_DEPENDENCIES) gst/$(am__dirstamp)
+	@rm -f gst/token$(EXEEXT)
+	$(AM_V_CCLD)$(LINK) $(gst_token_OBJECTS) $(gst_token_LDADD) $(LIBS)
+
+mostlyclean-compile:
+	-rm -f *.$(OBJEXT)
+	-rm -f gst/*.$(OBJEXT)
+
+distclean-compile:
+	-rm -f *.tab.c
+
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/addresspool.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/client.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/media.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/mediafactory.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/mountpoints.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/permissions.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/rtspserver.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/sessionmedia.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/sessionpool.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/stream.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/threadpool.Po@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@gst/$(DEPDIR)/token.Po@am__quote@
+
+.c.o:
+@am__fastdepCC_TRUE@	$(AM_V_CC)depbase=`echo $@ | sed 's|[^/]*$$|$(DEPDIR)/&|;s|\.o$$||'`;\
+@am__fastdepCC_TRUE@	$(COMPILE) -MT $@ -MD -MP -MF $$depbase.Tpo -c -o $@ $< &&\
+@am__fastdepCC_TRUE@	$(am__mv) $$depbase.Tpo $$depbase.Po
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	$(AM_V_CC)source='$<' object='$@' libtool=no @AMDEPBACKSLASH@
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	DEPDIR=$(DEPDIR) $(CCDEPMODE) $(depcomp) @AMDEPBACKSLASH@
+@am__fastdepCC_FALSE@	$(AM_V_CC@am__nodep@)$(COMPILE) -c -o $@ $<
+
+.c.obj:
+@am__fastdepCC_TRUE@	$(AM_V_CC)depbase=`echo $@ | sed 's|[^/]*$$|$(DEPDIR)/&|;s|\.obj$$||'`;\
+@am__fastdepCC_TRUE@	$(COMPILE) -MT $@ -MD -MP -MF $$depbase.Tpo -c -o $@ `$(CYGPATH_W) '$<'` &&\
+@am__fastdepCC_TRUE@	$(am__mv) $$depbase.Tpo $$depbase.Po
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	$(AM_V_CC)source='$<' object='$@' libtool=no @AMDEPBACKSLASH@
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	DEPDIR=$(DEPDIR) $(CCDEPMODE) $(depcomp) @AMDEPBACKSLASH@
+@am__fastdepCC_FALSE@	$(AM_V_CC@am__nodep@)$(COMPILE) -c -o $@ `$(CYGPATH_W) '$<'`
+
+.c.lo:
+@am__fastdepCC_TRUE@	$(AM_V_CC)depbase=`echo $@ | sed 's|[^/]*$$|$(DEPDIR)/&|;s|\.lo$$||'`;\
+@am__fastdepCC_TRUE@	$(LTCOMPILE) -MT $@ -MD -MP -MF $$depbase.Tpo -c -o $@ $< &&\
+@am__fastdepCC_TRUE@	$(am__mv) $$depbase.Tpo $$depbase.Plo
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	$(AM_V_CC)source='$<' object='$@' libtool=yes @AMDEPBACKSLASH@
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	DEPDIR=$(DEPDIR) $(CCDEPMODE) $(depcomp) @AMDEPBACKSLASH@
+@am__fastdepCC_FALSE@	$(AM_V_CC@am__nodep@)$(LTCOMPILE) -c -o $@ $<
+
+mostlyclean-libtool:
+	-rm -f *.lo
+
+clean-libtool:
+	-rm -rf .libs _libs
+	-rm -rf gst/.libs gst/_libs
+
+ID: $(am__tagged_files)
+	$(am__define_uniq_tagged_files); mkid -fID $$unique
+tags: tags-am
+TAGS: tags
+
+tags-am: $(TAGS_DEPENDENCIES) $(am__tagged_files)
+	set x; \
+	here=`pwd`; \
+	$(am__define_uniq_tagged_files); \
+	shift; \
+	if test -z "$(ETAGS_ARGS)$$*$$unique"; then :; else \
+	  test -n "$$unique" || unique=$$empty_fix; \
+	  if test $$# -gt 0; then \
+	    $(ETAGS) $(ETAGSFLAGS) $(AM_ETAGSFLAGS) $(ETAGS_ARGS) \
+	      "$$@" $$unique; \
+	  else \
+	    $(ETAGS) $(ETAGSFLAGS) $(AM_ETAGSFLAGS) $(ETAGS_ARGS) \
+	      $$unique; \
+	  fi; \
+	fi
+ctags: ctags-am
+
+CTAGS: ctags
+ctags-am: $(TAGS_DEPENDENCIES) $(am__tagged_files)
+	$(am__define_uniq_tagged_files); \
+	test -z "$(CTAGS_ARGS)$$unique" \
+	  || $(CTAGS) $(CTAGSFLAGS) $(AM_CTAGSFLAGS) $(CTAGS_ARGS) \
+	     $$unique
+
+GTAGS:
+	here=`$(am__cd) $(top_builddir) && pwd` \
+	  && $(am__cd) $(top_srcdir) \
+	  && gtags -i $(GTAGS_ARGS) "$$here"
+cscopelist: cscopelist-am
+
+cscopelist-am: $(am__tagged_files)
+	list='$(am__tagged_files)'; \
+	case "$(srcdir)" in \
+	  [\\/]* | ?:[\\/]*) sdir="$(srcdir)" ;; \
+	  *) sdir=$(subdir)/$(srcdir) ;; \
+	esac; \
+	for i in $$list; do \
+	  if test -f "$$i"; then \
+	    echo "$(subdir)/$$i"; \
+	  else \
+	    echo "$$sdir/$$i"; \
+	  fi; \
+	done >> $(top_builddir)/cscope.files
+
+distclean-tags:
+	-rm -f TAGS ID GTAGS GRTAGS GSYMS GPATH tags
+
+# Recover from deleted '.trs' file; this should ensure that
+# "rm -f foo.log; make foo.trs" re-run 'foo.test', and re-create
+# both 'foo.log' and 'foo.trs'.  Break the recipe in two subshells
+# to avoid problems with "make -n".
+.log.trs:
+	rm -f $< $@
+	$(MAKE) $(AM_MAKEFLAGS) $<
+
+# Leading 'am--fnord' is there to ensure the list of targets does not
+# expand to empty, as could happen e.g. with make check TESTS=''.
+am--fnord $(TEST_LOGS) $(TEST_LOGS:.log=.trs): $(am__force_recheck)
+am--force-recheck:
+	@:
+
+$(TEST_SUITE_LOG): $(TEST_LOGS)
+	@$(am__set_TESTS_bases); \
+	am__f_ok () { test -f "$$1" && test -r "$$1"; }; \
+	redo_bases=`for i in $$bases; do \
+	              am__f_ok $$i.trs && am__f_ok $$i.log || echo $$i; \
+	            done`; \
+	if test -n "$$redo_bases"; then \
+	  redo_logs=`for i in $$redo_bases; do echo $$i.log; done`; \
+	  redo_results=`for i in $$redo_bases; do echo $$i.trs; done`; \
+	  if $(am__make_dryrun); then :; else \
+	    rm -f $$redo_logs && rm -f $$redo_results || exit 1; \
+	  fi; \
+	fi; \
+	if test -n "$$am__remaking_logs"; then \
+	  echo "fatal: making $(TEST_SUITE_LOG): possible infinite" \
+	       "recursion detected" >&2; \
+	else \
+	  am__remaking_logs=yes $(MAKE) $(AM_MAKEFLAGS) $$redo_logs; \
+	fi; \
+	if $(am__make_dryrun); then :; else \
+	  st=0;  \
+	  errmsg="fatal: making $(TEST_SUITE_LOG): failed to create"; \
+	  for i in $$redo_bases; do \
+	    test -f $$i.trs && test -r $$i.trs \
+	      || { echo "$$errmsg $$i.trs" >&2; st=1; }; \
+	    test -f $$i.log && test -r $$i.log \
+	      || { echo "$$errmsg $$i.log" >&2; st=1; }; \
+	  done; \
+	  test $$st -eq 0 || exit 1; \
+	fi
+	@$(am__sh_e_setup); $(am__tty_colors); $(am__set_TESTS_bases); \
+	ws='[ 	]'; \
+	results=`for b in $$bases; do echo $$b.trs; done`; \
+	test -n "$$results" || results=/dev/null; \
+	all=`  grep "^$$ws*:test-result:"           $$results | wc -l`; \
+	pass=` grep "^$$ws*:test-result:$$ws*PASS"  $$results | wc -l`; \
+	fail=` grep "^$$ws*:test-result:$$ws*FAIL"  $$results | wc -l`; \
+	skip=` grep "^$$ws*:test-result:$$ws*SKIP"  $$results | wc -l`; \
+	xfail=`grep "^$$ws*:test-result:$$ws*XFAIL" $$results | wc -l`; \
+	xpass=`grep "^$$ws*:test-result:$$ws*XPASS" $$results | wc -l`; \
+	error=`grep "^$$ws*:test-result:$$ws*ERROR" $$results | wc -l`; \
+	if test `expr $$fail + $$xpass + $$error` -eq 0; then \
+	  success=true; \
+	else \
+	  success=false; \
+	fi; \
+	br='==================='; br=$$br$$br$$br$$br; \
+	result_count () \
+	{ \
+	    if test x"$$1" = x"--maybe-color"; then \
+	      maybe_colorize=yes; \
+	    elif test x"$$1" = x"--no-color"; then \
+	      maybe_colorize=no; \
+	    else \
+	      echo "$@: invalid 'result_count' usage" >&2; exit 4; \
+	    fi; \
+	    shift; \
+	    desc=$$1 count=$$2; \
+	    if test $$maybe_colorize = yes && test $$count -gt 0; then \
+	      color_start=$$3 color_end=$$std; \
+	    else \
+	      color_start= color_end=; \
+	    fi; \
+	    echo "$${color_start}# $$desc $$count$${color_end}"; \
+	}; \
+	create_testsuite_report () \
+	{ \
+	  result_count $$1 "TOTAL:" $$all   "$$brg"; \
+	  result_count $$1 "PASS: " $$pass  "$$grn"; \
+	  result_count $$1 "SKIP: " $$skip  "$$blu"; \
+	  result_count $$1 "XFAIL:" $$xfail "$$lgn"; \
+	  result_count $$1 "FAIL: " $$fail  "$$red"; \
+	  result_count $$1 "XPASS:" $$xpass "$$red"; \
+	  result_count $$1 "ERROR:" $$error "$$mgn"; \
+	}; \
+	{								\
+	  echo "$(PACKAGE_STRING): $(subdir)/$(TEST_SUITE_LOG)" |	\
+	    $(am__rst_title);						\
+	  create_testsuite_report --no-color;				\
+	  echo;								\
+	  echo ".. contents:: :depth: 2";				\
+	  echo;								\
+	  for b in $$bases; do echo $$b; done				\
+	    | $(am__create_global_log);					\
+	} >$(TEST_SUITE_LOG).tmp || exit 1;				\
+	mv $(TEST_SUITE_LOG).tmp $(TEST_SUITE_LOG);			\
+	if $$success; then						\
+	  col="$$grn";							\
+	 else								\
+	  col="$$red";							\
+	  test x"$$VERBOSE" = x || cat $(TEST_SUITE_LOG);		\
+	fi;								\
+	echo "$${col}$$br$${std}"; 					\
+	echo "$${col}Testsuite summary for $(PACKAGE_STRING)$${std}";	\
+	echo "$${col}$$br$${std}"; 					\
+	create_testsuite_report --maybe-color;				\
+	echo "$$col$$br$$std";						\
+	if $$success; then :; else					\
+	  echo "$${col}See $(subdir)/$(TEST_SUITE_LOG)$${std}";		\
+	  if test -n "$(PACKAGE_BUGREPORT)"; then			\
+	    echo "$${col}Please report to $(PACKAGE_BUGREPORT)$${std}";	\
+	  fi;								\
+	  echo "$$col$$br$$std";					\
+	fi;								\
+	$$success || exit 1
+
+check-TESTS:
+	@list='$(RECHECK_LOGS)';           test -z "$$list" || rm -f $$list
+	@list='$(RECHECK_LOGS:.log=.trs)'; test -z "$$list" || rm -f $$list
+	@test -z "$(TEST_SUITE_LOG)" || rm -f $(TEST_SUITE_LOG)
+	@set +e; $(am__set_TESTS_bases); \
+	log_list=`for i in $$bases; do echo $$i.log; done`; \
+	trs_list=`for i in $$bases; do echo $$i.trs; done`; \
+	log_list=`echo $$log_list`; trs_list=`echo $$trs_list`; \
+	$(MAKE) $(AM_MAKEFLAGS) $(TEST_SUITE_LOG) TEST_LOGS="$$log_list"; \
+	exit $$?;
+recheck: all $(check_PROGRAMS)
+	@test -z "$(TEST_SUITE_LOG)" || rm -f $(TEST_SUITE_LOG)
+	@set +e; $(am__set_TESTS_bases); \
+	bases=`for i in $$bases; do echo $$i; done \
+	         | $(am__list_recheck_tests)` || exit 1; \
+	log_list=`for i in $$bases; do echo $$i.log; done`; \
+	log_list=`echo $$log_list`; \
+	$(MAKE) $(AM_MAKEFLAGS) $(TEST_SUITE_LOG) \
+	        am__force_recheck=am--force-recheck \
+	        TEST_LOGS="$$log_list"; \
+	exit $$?
+gst/rtspserver.log: gst/rtspserver$(EXEEXT)
+	@p='gst/rtspserver$(EXEEXT)'; \
+	b='gst/rtspserver'; \
+	$(am__check_pre) $(LOG_DRIVER) --test-name "$$f" \
+	--log-file $$b.log --trs-file $$b.trs \
+	$(am__common_driver_flags) $(AM_LOG_DRIVER_FLAGS) $(LOG_DRIVER_FLAGS) -- $(LOG_COMPILE) \
+	"$$tst" $(AM_TESTS_FD_REDIRECT)
+gst/client.log: gst/client$(EXEEXT)
+	@p='gst/client$(EXEEXT)'; \
+	b='gst/client'; \
+	$(am__check_pre) $(LOG_DRIVER) --test-name "$$f" \
+	--log-file $$b.log --trs-file $$b.trs \
+	$(am__common_driver_flags) $(AM_LOG_DRIVER_FLAGS) $(LOG_DRIVER_FLAGS) -- $(LOG_COMPILE) \
+	"$$tst" $(AM_TESTS_FD_REDIRECT)
+gst/mountpoints.log: gst/mountpoints$(EXEEXT)
+	@p='gst/mountpoints$(EXEEXT)'; \
+	b='gst/mountpoints'; \
+	$(am__check_pre) $(LOG_DRIVER) --test-name "$$f" \
+	--log-file $$b.log --trs-file $$b.trs \
+	$(am__common_driver_flags) $(AM_LOG_DRIVER_FLAGS) $(LOG_DRIVER_FLAGS) -- $(LOG_COMPILE) \
+	"$$tst" $(AM_TESTS_FD_REDIRECT)
+gst/mediafactory.log: gst/mediafactory$(EXEEXT)
+	@p='gst/mediafactory$(EXEEXT)'; \
+	b='gst/mediafactory'; \
+	$(am__check_pre) $(LOG_DRIVER) --test-name "$$f" \
+	--log-file $$b.log --trs-file $$b.trs \
+	$(am__common_driver_flags) $(AM_LOG_DRIVER_FLAGS) $(LOG_DRIVER_FLAGS) -- $(LOG_COMPILE) \
+	"$$tst" $(AM_TESTS_FD_REDIRECT)
+gst/media.log: gst/media$(EXEEXT)
+	@p='gst/media$(EXEEXT)'; \
+	b='gst/media'; \
+	$(am__check_pre) $(LOG_DRIVER) --test-name "$$f" \
+	--log-file $$b.log --trs-file $$b.trs \
+	$(am__common_driver_flags) $(AM_LOG_DRIVER_FLAGS) $(LOG_DRIVER_FLAGS) -- $(LOG_COMPILE) \
+	"$$tst" $(AM_TESTS_FD_REDIRECT)
+gst/stream.log: gst/stream$(EXEEXT)
+	@p='gst/stream$(EXEEXT)'; \
+	b='gst/stream'; \
+	$(am__check_pre) $(LOG_DRIVER) --test-name "$$f" \
+	--log-file $$b.log --trs-file $$b.trs \
+	$(am__common_driver_flags) $(AM_LOG_DRIVER_FLAGS) $(LOG_DRIVER_FLAGS) -- $(LOG_COMPILE) \
+	"$$tst" $(AM_TESTS_FD_REDIRECT)
+gst/addresspool.log: gst/addresspool$(EXEEXT)
+	@p='gst/addresspool$(EXEEXT)'; \
+	b='gst/addresspool'; \
+	$(am__check_pre) $(LOG_DRIVER) --test-name "$$f" \
+	--log-file $$b.log --trs-file $$b.trs \
+	$(am__common_driver_flags) $(AM_LOG_DRIVER_FLAGS) $(LOG_DRIVER_FLAGS) -- $(LOG_COMPILE) \
+	"$$tst" $(AM_TESTS_FD_REDIRECT)
+gst/threadpool.log: gst/threadpool$(EXEEXT)
+	@p='gst/threadpool$(EXEEXT)'; \
+	b='gst/threadpool'; \
+	$(am__check_pre) $(LOG_DRIVER) --test-name "$$f" \
+	--log-file $$b.log --trs-file $$b.trs \
+	$(am__common_driver_flags) $(AM_LOG_DRIVER_FLAGS) $(LOG_DRIVER_FLAGS) -- $(LOG_COMPILE) \
+	"$$tst" $(AM_TESTS_FD_REDIRECT)
+gst/permissions.log: gst/permissions$(EXEEXT)
+	@p='gst/permissions$(EXEEXT)'; \
+	b='gst/permissions'; \
+	$(am__check_pre) $(LOG_DRIVER) --test-name "$$f" \
+	--log-file $$b.log --trs-file $$b.trs \
+	$(am__common_driver_flags) $(AM_LOG_DRIVER_FLAGS) $(LOG_DRIVER_FLAGS) -- $(LOG_COMPILE) \
+	"$$tst" $(AM_TESTS_FD_REDIRECT)
+gst/token.log: gst/token$(EXEEXT)
+	@p='gst/token$(EXEEXT)'; \
+	b='gst/token'; \
+	$(am__check_pre) $(LOG_DRIVER) --test-name "$$f" \
+	--log-file $$b.log --trs-file $$b.trs \
+	$(am__common_driver_flags) $(AM_LOG_DRIVER_FLAGS) $(LOG_DRIVER_FLAGS) -- $(LOG_COMPILE) \
+	"$$tst" $(AM_TESTS_FD_REDIRECT)
+gst/sessionmedia.log: gst/sessionmedia$(EXEEXT)
+	@p='gst/sessionmedia$(EXEEXT)'; \
+	b='gst/sessionmedia'; \
+	$(am__check_pre) $(LOG_DRIVER) --test-name "$$f" \
+	--log-file $$b.log --trs-file $$b.trs \
+	$(am__common_driver_flags) $(AM_LOG_DRIVER_FLAGS) $(LOG_DRIVER_FLAGS) -- $(LOG_COMPILE) \
+	"$$tst" $(AM_TESTS_FD_REDIRECT)
+gst/sessionpool.log: gst/sessionpool$(EXEEXT)
+	@p='gst/sessionpool$(EXEEXT)'; \
+	b='gst/sessionpool'; \
+	$(am__check_pre) $(LOG_DRIVER) --test-name "$$f" \
+	--log-file $$b.log --trs-file $$b.trs \
+	$(am__common_driver_flags) $(AM_LOG_DRIVER_FLAGS) $(LOG_DRIVER_FLAGS) -- $(LOG_COMPILE) \
+	"$$tst" $(AM_TESTS_FD_REDIRECT)
+.test.log:
+	@p='$<'; \
+	$(am__set_b); \
+	$(am__check_pre) $(TEST_LOG_DRIVER) --test-name "$$f" \
+	--log-file $$b.log --trs-file $$b.trs \
+	$(am__common_driver_flags) $(AM_TEST_LOG_DRIVER_FLAGS) $(TEST_LOG_DRIVER_FLAGS) -- $(TEST_LOG_COMPILE) \
+	"$$tst" $(AM_TESTS_FD_REDIRECT)
+@am__EXEEXT_TRUE@.test$(EXEEXT).log:
+@am__EXEEXT_TRUE@	@p='$<'; \
+@am__EXEEXT_TRUE@	$(am__set_b); \
+@am__EXEEXT_TRUE@	$(am__check_pre) $(TEST_LOG_DRIVER) --test-name "$$f" \
+@am__EXEEXT_TRUE@	--log-file $$b.log --trs-file $$b.trs \
+@am__EXEEXT_TRUE@	$(am__common_driver_flags) $(AM_TEST_LOG_DRIVER_FLAGS) $(TEST_LOG_DRIVER_FLAGS) -- $(TEST_LOG_COMPILE) \
+@am__EXEEXT_TRUE@	"$$tst" $(AM_TESTS_FD_REDIRECT)
+
+distdir: $(DISTFILES)
+	@srcdirstrip=`echo "$(srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+	topsrcdirstrip=`echo "$(top_srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+	list='$(DISTFILES)'; \
+	  dist_files=`for file in $$list; do echo $$file; done | \
+	  sed -e "s|^$$srcdirstrip/||;t" \
+	      -e "s|^$$topsrcdirstrip/|$(top_builddir)/|;t"`; \
+	case $$dist_files in \
+	  */*) $(MKDIR_P) `echo "$$dist_files" | \
+			   sed '/\//!d;s|^|$(distdir)/|;s,/[^/]*$$,,' | \
+			   sort -u` ;; \
+	esac; \
+	for file in $$dist_files; do \
+	  if test -f $$file || test -d $$file; then d=.; else d=$(srcdir); fi; \
+	  if test -d $$d/$$file; then \
+	    dir=`echo "/$$file" | sed -e 's,/[^/]*$$,,'`; \
+	    if test -d "$(distdir)/$$file"; then \
+	      find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+	    fi; \
+	    if test -d $(srcdir)/$$file && test $$d != $(srcdir); then \
+	      cp -fpR $(srcdir)/$$file "$(distdir)$$dir" || exit 1; \
+	      find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+	    fi; \
+	    cp -fpR $$d/$$file "$(distdir)$$dir" || exit 1; \
+	  else \
+	    test -f "$(distdir)/$$file" \
+	    || cp -p $$d/$$file "$(distdir)/$$file" \
+	    || exit 1; \
+	  fi; \
+	done
+check-am: all-am
+	$(MAKE) $(AM_MAKEFLAGS) $(check_PROGRAMS)
+	$(MAKE) $(AM_MAKEFLAGS) check-TESTS
+check: check-am
+all-am: Makefile $(PROGRAMS)
+installdirs:
+install: install-am
+install-exec: install-exec-am
+install-data: install-data-am
+uninstall: uninstall-am
+
+install-am: all-am
+	@$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
+
+installcheck: installcheck-am
+install-strip:
+	if test -z '$(STRIP)'; then \
+	  $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	    install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	      install; \
+	else \
+	  $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	    install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	    "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'" install; \
+	fi
+mostlyclean-generic:
+	-test -z "$(TEST_LOGS)" || rm -f $(TEST_LOGS)
+	-test -z "$(TEST_LOGS:.log=.trs)" || rm -f $(TEST_LOGS:.log=.trs)
+	-test -z "$(TEST_SUITE_LOG)" || rm -f $(TEST_SUITE_LOG)
+
+clean-generic:
+	-test -z "$(CLEANFILES)" || rm -f $(CLEANFILES)
+
+distclean-generic:
+	-test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
+	-test . = "$(srcdir)" || test -z "$(CONFIG_CLEAN_VPATH_FILES)" || rm -f $(CONFIG_CLEAN_VPATH_FILES)
+	-rm -f gst/$(DEPDIR)/$(am__dirstamp)
+	-rm -f gst/$(am__dirstamp)
+
+maintainer-clean-generic:
+	@echo "This command is intended for maintainers to use"
+	@echo "it deletes files that may require special tools to rebuild."
+clean: clean-am
+
+clean-am: clean-checkPROGRAMS clean-generic clean-libtool clean-local \
+	clean-noinstPROGRAMS mostlyclean-am
+
+distclean: distclean-am
+	-rm -rf gst/$(DEPDIR)
+	-rm -f Makefile
+distclean-am: clean-am distclean-compile distclean-generic \
+	distclean-tags
+
+dvi: dvi-am
+
+dvi-am:
+
+html: html-am
+
+html-am:
+
+info: info-am
+
+info-am:
+
+install-data-am:
+
+install-dvi: install-dvi-am
+
+install-dvi-am:
+
+install-exec-am:
+
+install-html: install-html-am
+
+install-html-am:
+
+install-info: install-info-am
+
+install-info-am:
+
+install-man:
+
+install-pdf: install-pdf-am
+
+install-pdf-am:
+
+install-ps: install-ps-am
+
+install-ps-am:
+
+installcheck-am:
+
+maintainer-clean: maintainer-clean-am
+	-rm -rf gst/$(DEPDIR)
+	-rm -f Makefile
+maintainer-clean-am: distclean-am maintainer-clean-generic
+
+mostlyclean: mostlyclean-am
+
+mostlyclean-am: mostlyclean-compile mostlyclean-generic \
+	mostlyclean-libtool
+
+pdf: pdf-am
+
+pdf-am:
+
+ps: ps-am
+
+ps-am:
+
+uninstall-am:
+
+.MAKE: check-am install-am install-strip
+
+.PHONY: CTAGS GTAGS TAGS all all-am check check-TESTS check-am clean \
+	clean-checkPROGRAMS clean-generic clean-libtool clean-local \
+	clean-noinstPROGRAMS cscopelist-am ctags ctags-am distclean \
+	distclean-compile distclean-generic distclean-libtool \
+	distclean-tags distdir dvi dvi-am html html-am info info-am \
+	install install-am install-data install-data-am install-dvi \
+	install-dvi-am install-exec install-exec-am install-html \
+	install-html-am install-info install-info-am install-man \
+	install-pdf install-pdf-am install-ps install-ps-am \
+	install-strip installcheck installcheck-am installdirs \
+	maintainer-clean maintainer-clean-generic mostlyclean \
+	mostlyclean-compile mostlyclean-generic mostlyclean-libtool \
+	pdf pdf-am ps ps-am recheck tags tags-am uninstall \
+	uninstall-am
+
+
+# keep target around, since it's referenced in the modules' Makefiles
+clean-local-check:
+	@echo
+
+# hangs spectacularly on some machines, so let's not do this by default yet
+@HAVE_VALGRIND_TRUE@check-valgrind:
+@HAVE_VALGRIND_TRUE@	$(MAKE) valgrind
+@HAVE_VALGRIND_FALSE@check-valgrind:
+@HAVE_VALGRIND_FALSE@	@true
+
+LOOPS ?= 10
+
+# run any given test by running make test.check
+# if the test fails, run it again at at least debug level 2
+%.check: %
+	@$(TESTS_ENVIRONMENT)					\
+	CK_DEFAULT_TIMEOUT=20					\
+	$* ||							\
+	$(TESTS_ENVIRONMENT)					\
+	GST_DEBUG=$$GST_DEBUG,*:2				\
+	CK_DEFAULT_TIMEOUT=20					\
+	$*
+
+# just like 'check', but don't run it again if it fails (useful for debugging)
+%.check-norepeat: %
+	@$(TESTS_ENVIRONMENT)					\
+	CK_DEFAULT_TIMEOUT=20					\
+	$*
+
+# run any given test in a loop
+%.torture: %
+	@for i in `seq 1 $(LOOPS)`; do				\
+	$(TESTS_ENVIRONMENT)					\
+	CK_DEFAULT_TIMEOUT=20					\
+	$*; done
+
+# run any given test in an infinite loop
+%.forever: %
+	@while true; do						\
+	$(TESTS_ENVIRONMENT)					\
+	CK_DEFAULT_TIMEOUT=20					\
+	$* || break; done
+
+# valgrind any given test by running make test.valgrind
+%.valgrind: %
+	@$(TESTS_ENVIRONMENT)					\
+	CK_DEFAULT_TIMEOUT=360					\
+	G_SLICE=always-malloc					\
+	$(LIBTOOL) --mode=execute				\
+	$(VALGRIND_PATH) -q					\
+	$(foreach s,$(SUPPRESSIONS),--suppressions=$(s))	\
+	--tool=memcheck --leak-check=full --trace-children=yes	\
+	--show-possibly-lost=no                                 \
+	--leak-resolution=high --num-callers=20			\
+	./$* 2>&1 | tee valgrind.log
+	@if grep "==" valgrind.log > /dev/null 2>&1; then	\
+	    rm valgrind.log;					\
+	    exit 1;						\
+	fi
+	@rm valgrind.log
+
+# valgrind any given test and generate suppressions for it
+%.valgrind.gen-suppressions: %
+	@$(TESTS_ENVIRONMENT)					\
+	CK_DEFAULT_TIMEOUT=360					\
+	G_SLICE=always-malloc					\
+	$(LIBTOOL) --mode=execute				\
+	$(VALGRIND_PATH) -q 					\
+	$(foreach s,$(SUPPRESSIONS),--suppressions=$(s))	\
+	--tool=memcheck --leak-check=full --trace-children=yes	\
+	--show-possibly-lost=no                                 \
+	--leak-resolution=high --num-callers=20			\
+	--gen-suppressions=all					\
+	./$* 2>&1 | tee suppressions.log
+
+# valgrind torture any given test
+%.valgrind-torture: %
+	@for i in `seq 1 $(LOOPS)`; do				\
+		$(MAKE) $*.valgrind ||				\
+		(echo "Failure after $$i runs"; exit 1) ||	\
+		exit 1;						\
+	done
+	@banner="All $(LOOPS) loops passed";			\
+	dashes=`echo "$$banner" | sed s/./=/g`;			\
+	echo $$dashes; echo $$banner; echo $$dashes
+
+# valgrind any given test until failure by running make test.valgrind-forever
+%.valgrind-forever: %
+	@while $(MAKE) $*.valgrind; do				\
+	  true; done
+
+# gdb any given test by running make test.gdb
+%.gdb: %
+	@$(TESTS_ENVIRONMENT)					\
+	CK_FORK=no						\
+	$(LIBTOOL) --mode=execute				\
+	gdb $*
+
+%.lcov-reset:
+	$(MAKE) $*.lcov-run
+	$(MAKE) $*.lcov-report
+
+%.lcov: %
+	$(MAKE) $*.lcov-reset
+
+@GST_GCOV_ENABLED_TRUE@%.lcov-clean:
+@GST_GCOV_ENABLED_TRUE@	$(MAKE) -C $(top_builddir) lcov-clean
+
+@GST_GCOV_ENABLED_TRUE@%.lcov-run:
+@GST_GCOV_ENABLED_TRUE@	$(MAKE) $*.lcov-clean
+@GST_GCOV_ENABLED_TRUE@	$(MAKE) $*.check
+
+@GST_GCOV_ENABLED_TRUE@%.lcov-report:
+@GST_GCOV_ENABLED_TRUE@	$(MAKE) -C $(top_builddir) lcov-report
+@GST_GCOV_ENABLED_FALSE@%.lcov-run:
+@GST_GCOV_ENABLED_FALSE@	echo "Need to reconfigure with --enable-gcov"
+
+@GST_GCOV_ENABLED_FALSE@%.lcov-report:
+@GST_GCOV_ENABLED_FALSE@	echo "Need to reconfigure with --enable-gcov"
+
+# torture tests
+torture: $(TESTS)
+	-rm test-registry.*
+	@echo "Torturing tests ..."
+	@for i in `seq 1 $(LOOPS)`; do				\
+		$(MAKE) check ||				\
+		(echo "Failure after $$i runs"; exit 1) ||	\
+		exit 1;						\
+	done
+	@banner="All $(LOOPS) loops passed";			\
+	dashes=`echo "$$banner" | sed s/./=/g`;			\
+	echo $$dashes; echo $$banner; echo $$dashes
+
+# forever tests
+forever: $(TESTS)
+	-rm test-registry.*
+	@echo "Forever tests ..."
+	@while true; do						\
+		$(MAKE) check ||				\
+		(echo "Failure"; exit 1) ||			\
+		exit 1;						\
+	done
+
+# valgrind all tests
+valgrind: $(TESTS)
+	@echo "Valgrinding tests ..."
+	@failed=0;							\
+	for t in $(filter-out $(VALGRIND_TESTS_DISABLE),$(TESTS)); do	\
+		$(MAKE) $$t.valgrind;					\
+		if test "$$?" -ne 0; then                               \
+			echo "Valgrind error for test $$t";		\
+			failed=`expr $$failed + 1`;			\
+			whicht="$$whicht $$t";				\
+		fi;							\
+	done;								\
+	if test "$$failed" -ne 0; then					\
+		echo "$$failed tests had leaks or errors under valgrind:";	\
+		echo "$$whicht";					\
+		false;							\
+	fi
+
+# valgrind all tests until failure
+valgrind-forever: $(TESTS)
+	-rm test-registry.*
+	@echo "Forever valgrinding tests ..."
+	@while true; do						\
+		$(MAKE) valgrind ||				\
+		(echo "Failure"; exit 1) ||			\
+		exit 1;						\
+	done
+
+# valgrind torture all tests
+valgrind-torture: $(TESTS)
+	-rm test-registry.*
+	@echo "Torturing and valgrinding tests ..."
+	@for i in `seq 1 $(LOOPS)`; do				\
+		$(MAKE) valgrind ||				\
+		(echo "Failure after $$i runs"; exit 1) ||	\
+		exit 1;						\
+	done
+	@banner="All $(LOOPS) loops passed";			\
+	dashes=`echo "$$banner" | sed s/./=/g`;			\
+	echo $$dashes; echo $$banner; echo $$dashes
+
+# valgrind all tests and generate suppressions
+valgrind.gen-suppressions: $(TESTS)
+	@echo "Valgrinding tests ..."
+	@failed=0;							\
+	for t in $(filter-out $(VALGRIND_TESTS_DISABLE),$(TESTS)); do	\
+		$(MAKE) $$t.valgrind.gen-suppressions;			\
+		if test "$$?" -ne 0; then                               \
+			echo "Valgrind error for test $$t";		\
+			failed=`expr $$failed + 1`;			\
+			whicht="$$whicht $$t";				\
+		fi;							\
+	done;								\
+	if test "$$failed" -ne 0; then					\
+		echo "$$failed tests had leaks or errors under valgrind:";	\
+		echo "$$whicht";					\
+		false;							\
+	fi
+inspect:
+	@echo "Inspecting features ..."
+	@for e in `$(TESTS_ENVIRONMENT) $(GST_INSPECT) | head -n -2 	\
+	  | cut -d: -f2`;						\
+	  do echo Inspecting $$e;					\
+	     $(GST_INSPECT) $$e > /dev/null 2>&1; done
+
+help:
+	@echo
+	@echo "make check                         -- run all checks"
+	@echo "make torture                       -- run all checks $(LOOPS) times"
+	@echo "make (dir)/(test).check            -- run the given check once, repeat with GST_DEBUG=*:2 if it fails"
+	@echo "make (dir)/(test).check-norepeat   -- run the given check once, but don't run it again if it fails"
+	@echo "make (dir)/(test).forever          -- run the given check forever"
+	@echo "make (dir)/(test).torture          -- run the given check $(LOOPS) times"
+	@echo
+	@echo "make (dir)/(test).gdb              -- start up gdb for the given test"
+	@echo
+	@echo "make valgrind                      -- valgrind all tests"
+	@echo "make valgrind-forever              -- valgrind all tests forever"
+	@echo "make valgrind-torture              -- valgrind all tests $(LOOPS) times"
+	@echo "make valgrind.gen-suppressions     -- generate suppressions for all tests"
+	@echo "                                      and save to suppressions.log"
+	@echo "make (dir)/(test).valgrind         -- valgrind the given test"
+	@echo "make (dir)/(test).valgrind-forever -- valgrind the given test forever"
+	@echo "make (dir)/(test).valgrind-torture -- valgrind the given test $(LOOPS) times"
+	@echo "make (dir)/(test).valgrind.gen-suppressions -- generate suppressions"
+	@echo "                                               and save to suppressions.log"
+	@echo "make inspect                       -- inspect all plugin features"
+	@echo
+	@echo
+	@echo "Additionally, you can use the GST_CHECKS environment variable to"
+	@echo "specify which test(s) should be run. This is useful if you are"
+	@echo "debugging a failure in one particular test, or want to reproduce"
+	@echo "a race condition in a single test."
+	@echo
+	@echo "Examples:"
+	@echo
+	@echo "  GST_CHECKS=test_this,test_that  make element/foobar.check"
+	@echo "  GST_CHECKS=test_many_threads    make element/foobar.forever"
+	@echo
+
+clean-local: clean-local-check
+
+$(CHECK_REGISTRY):
+	$(TESTS_ENVIRONMENT)
+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
diff --git a/tests/check/gst/addresspool.c b/tests/check/gst/addresspool.c
new file mode 100644
index 0000000..9a0ff54
--- /dev/null
+++ b/tests/check/gst/addresspool.c
@@ -0,0 +1,286 @@
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+GST_START_TEST (test_pool)
+{
+  GstRTSPAddressPool *pool;
+  GstRTSPAddress *addr, *addr2, *addr3;
+  GstRTSPAddressPoolResult res;
+
+  pool = gst_rtsp_address_pool_new ();
+
+  fail_if (gst_rtsp_address_pool_add_range (pool,
+          "233.252.0.1", "233.252.0.0", 5000, 5010, 1));
+  fail_if (gst_rtsp_address_pool_add_range (pool,
+          "233.252.0.1", "::1", 5000, 5010, 1));
+  fail_if (gst_rtsp_address_pool_add_range (pool,
+          "233.252.0.1", "ff02::1", 5000, 5010, 1));
+  fail_if (gst_rtsp_address_pool_add_range (pool,
+          "233.252.0.1.1", "233.252.0.1", 5000, 5010, 1));
+  fail_if (gst_rtsp_address_pool_add_range (pool,
+          "233.252.0.1", "233.252.0.1.1", 5000, 5010, 1));
+  ASSERT_CRITICAL (gst_rtsp_address_pool_add_range (pool,
+          "233.252.0.0", "233.252.0.1", 5010, 5000, 1));
+
+  fail_unless (gst_rtsp_address_pool_add_range (pool,
+          "233.252.0.0", "233.252.0.255", 5000, 5010, 1));
+  fail_unless (gst_rtsp_address_pool_add_range (pool,
+          "233.255.0.0", "233.255.0.0", 5000, 5010, 1));
+  fail_unless (gst_rtsp_address_pool_add_range (pool,
+          "233.255.0.0", "233.255.0.0", 5020, 5020, 1));
+
+  /* should fail, we can't allocate a block of 256 ports */
+  addr = gst_rtsp_address_pool_acquire_address (pool,
+      GST_RTSP_ADDRESS_FLAG_MULTICAST, 256);
+  fail_unless (addr == NULL);
+
+  addr = gst_rtsp_address_pool_acquire_address (pool,
+      GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+  fail_unless (addr != NULL);
+
+  addr2 = gst_rtsp_address_copy (addr);
+
+  gst_rtsp_address_free (addr2);
+  gst_rtsp_address_free (addr);
+
+  addr = gst_rtsp_address_pool_acquire_address (pool,
+      GST_RTSP_ADDRESS_FLAG_MULTICAST, 4);
+  fail_unless (addr != NULL);
+
+  /* Will fail because pool is NULL */
+  ASSERT_CRITICAL (gst_rtsp_address_pool_clear (NULL));
+
+  /* will fail because an address is allocated */
+  ASSERT_CRITICAL (gst_rtsp_address_pool_clear (pool));
+
+  gst_rtsp_address_free (addr);
+
+  gst_rtsp_address_pool_clear (pool);
+
+  /* start with odd port to make sure we are allocated address
+   * starting with even port
+   */
+  fail_unless (gst_rtsp_address_pool_add_range (pool,
+          "FF11:DB8::1", "FF11:DB8::1", 5001, 5003, 1));
+
+  addr = gst_rtsp_address_pool_acquire_address (pool,
+      GST_RTSP_ADDRESS_FLAG_IPV6 | GST_RTSP_ADDRESS_FLAG_EVEN_PORT |
+      GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+  fail_unless (addr != NULL);
+  fail_unless (addr->port == 5002);
+  fail_unless (!g_ascii_strcasecmp (addr->address, "FF11:DB8::1"));
+
+  /* Will fail becuse there is only one IPv6 address left */
+  addr2 = gst_rtsp_address_pool_acquire_address (pool,
+      GST_RTSP_ADDRESS_FLAG_IPV6 | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+  fail_unless (addr2 == NULL);
+
+  /* Will fail because the only IPv6 address left has an odd port */
+  addr2 = gst_rtsp_address_pool_acquire_address (pool,
+      GST_RTSP_ADDRESS_FLAG_IPV6 | GST_RTSP_ADDRESS_FLAG_EVEN_PORT |
+      GST_RTSP_ADDRESS_FLAG_MULTICAST, 1);
+  fail_unless (addr2 == NULL);
+
+  addr2 = gst_rtsp_address_pool_acquire_address (pool,
+      GST_RTSP_ADDRESS_FLAG_IPV4 | GST_RTSP_ADDRESS_FLAG_MULTICAST, 1);
+  fail_unless (addr2 == NULL);
+
+  gst_rtsp_address_free (addr);
+
+  gst_rtsp_address_pool_clear (pool);
+
+  fail_unless (gst_rtsp_address_pool_add_range (pool,
+          "233.252.0.0", "233.252.0.255", 5000, 5002, 1));
+
+  addr = gst_rtsp_address_pool_acquire_address (pool,
+      GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+  fail_unless (addr != NULL);
+  fail_unless (addr->port == 5000);
+  fail_unless (!strcmp (addr->address, "233.252.0.0"));
+
+  addr2 = gst_rtsp_address_pool_acquire_address (pool,
+      GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+  fail_unless (addr2 != NULL);
+  fail_unless (addr2->port == 5000);
+  fail_unless (!strcmp (addr2->address, "233.252.0.1"));
+
+  gst_rtsp_address_free (addr);
+  gst_rtsp_address_free (addr2);
+
+  addr = gst_rtsp_address_pool_acquire_address (pool,
+      GST_RTSP_ADDRESS_FLAG_IPV6 | GST_RTSP_ADDRESS_FLAG_MULTICAST, 1);
+  fail_unless (addr == NULL);
+
+  gst_rtsp_address_pool_clear (pool);
+
+  fail_unless (gst_rtsp_address_pool_add_range (pool,
+          "233.252.1.1", "233.252.1.1", 5000, 5001, 1));
+
+  res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 5000, 3,
+      1, &addr);
+  fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+  fail_unless (addr == NULL);
+
+  res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.2", 5000, 2,
+      1, &addr);
+  fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+  fail_unless (addr == NULL);
+
+  res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 500, 2, 1,
+      &addr);
+  fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+  fail_unless (addr == NULL);
+
+  res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 5000, 2,
+      2, &addr);
+  fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+  fail_unless (addr == NULL);
+
+  res = gst_rtsp_address_pool_reserve_address (pool, "2000::1", 5000, 2, 2,
+      &addr);
+  fail_unless (res == GST_RTSP_ADDRESS_POOL_EINVAL);
+  fail_unless (addr == NULL);
+
+  res = gst_rtsp_address_pool_reserve_address (pool, "ff02::1", 5000, 2, 2,
+      &addr);
+  fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+  fail_unless (addr == NULL);
+
+  res = gst_rtsp_address_pool_reserve_address (pool, "1.1", 5000, 2, 2, &addr);
+  fail_unless (res == GST_RTSP_ADDRESS_POOL_EINVAL);
+  fail_unless (addr == NULL);
+
+  res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 5000, 2,
+      1, &addr);
+  fail_unless (res == GST_RTSP_ADDRESS_POOL_OK);
+  fail_unless (addr != NULL);
+  fail_unless (addr->port == 5000);
+  fail_unless (!strcmp (addr->address, "233.252.1.1"));
+
+  res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 5000, 2,
+      1, &addr2);
+  fail_unless (res == GST_RTSP_ADDRESS_POOL_ERESERVED);
+  fail_unless (addr2 == NULL);
+
+  gst_rtsp_address_free (addr);
+  gst_rtsp_address_pool_clear (pool);
+
+  fail_unless (gst_rtsp_address_pool_add_range (pool,
+          "233.252.1.1", "233.252.1.3", 5000, 5001, 1));
+
+  res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.1", 5000, 2,
+      1, &addr);
+  fail_unless (addr != NULL);
+  fail_unless (addr->port == 5000);
+  fail_unless (!strcmp (addr->address, "233.252.1.1"));
+
+  res = gst_rtsp_address_pool_reserve_address (pool, "233.252.1.3", 5000, 2,
+      1, &addr2);
+  fail_unless (addr2 != NULL);
+  fail_unless (addr2->port == 5000);
+  fail_unless (!strcmp (addr2->address, "233.252.1.3"));
+
+  addr3 = gst_rtsp_address_pool_acquire_address (pool,
+      GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+  fail_unless (addr3 != NULL);
+  fail_unless (addr3->port == 5000);
+  fail_unless (!strcmp (addr3->address, "233.252.1.2"));
+
+  fail_unless (gst_rtsp_address_pool_acquire_address (pool,
+          GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2)
+      == NULL);
+
+  gst_rtsp_address_free (addr);
+  gst_rtsp_address_free (addr2);
+  gst_rtsp_address_free (addr3);
+  gst_rtsp_address_pool_clear (pool);
+
+  fail_unless (gst_rtsp_address_pool_add_range (pool,
+          "233.252.1.1", "233.252.1.1", 5000, 5001, 1));
+  fail_if (gst_rtsp_address_pool_has_unicast_addresses (pool));
+  fail_unless (gst_rtsp_address_pool_add_range (pool,
+          "192.168.1.1", "192.168.1.1", 6000, 6001, 0));
+  fail_unless (gst_rtsp_address_pool_has_unicast_addresses (pool));
+
+  addr = gst_rtsp_address_pool_acquire_address (pool,
+      GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
+  fail_unless (addr != NULL);
+  fail_unless (addr->port == 5000);
+  fail_unless (!strcmp (addr->address, "233.252.1.1"));
+  gst_rtsp_address_free (addr);
+
+  addr = gst_rtsp_address_pool_acquire_address (pool,
+      GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST, 2);
+  fail_unless (addr != NULL);
+  fail_unless (addr->port == 6000);
+  fail_unless (!strcmp (addr->address, "192.168.1.1"));
+  gst_rtsp_address_free (addr);
+
+  fail_unless (gst_rtsp_address_pool_add_range (pool,
+          GST_RTSP_ADDRESS_POOL_ANY_IPV4, GST_RTSP_ADDRESS_POOL_ANY_IPV4, 5000,
+          5001, 0));
+  res =
+      gst_rtsp_address_pool_reserve_address (pool, "192.168.0.1", 5000, 1, 0,
+      &addr);
+  fail_unless (res == GST_RTSP_ADDRESS_POOL_ERANGE);
+  res =
+      gst_rtsp_address_pool_reserve_address (pool, "0.0.0.0", 5000, 1, 0,
+      &addr);
+  fail_unless (res == GST_RTSP_ADDRESS_POOL_OK);
+  gst_rtsp_address_free (addr);
+  gst_rtsp_address_pool_clear (pool);
+
+  /* Error case 2. Using ANY as min address makes it possible to allocate the
+   * same address twice */
+  fail_unless (gst_rtsp_address_pool_add_range (pool,
+          GST_RTSP_ADDRESS_POOL_ANY_IPV4, "255.255.255.255", 5000, 5001, 0));
+  res =
+      gst_rtsp_address_pool_reserve_address (pool, "192.168.0.1", 5000, 1, 0,
+      &addr);
+  fail_unless (res == GST_RTSP_ADDRESS_POOL_OK);
+  res =
+      gst_rtsp_address_pool_reserve_address (pool, "192.168.0.1", 5000, 1, 0,
+      &addr2);
+  fail_unless (res == GST_RTSP_ADDRESS_POOL_ERESERVED);
+  gst_rtsp_address_free (addr);
+  gst_rtsp_address_pool_clear (pool);
+
+  g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspaddresspool_suite (void)
+{
+  Suite *s = suite_create ("rtspaddresspool");
+  TCase *tc = tcase_create ("general");
+
+  suite_add_tcase (s, tc);
+  tcase_set_timeout (tc, 20);
+  tcase_add_test (tc, test_pool);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtspaddresspool);
diff --git a/tests/check/gst/client.c b/tests/check/gst/client.c
new file mode 100644
index 0000000..1bbd67d
--- /dev/null
+++ b/tests/check/gst/client.c
@@ -0,0 +1,929 @@
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+static gchar * session_id;
+static gint cseq;
+static guint expected_session_timeout = 60;
+
+static gboolean
+test_response_200 (GstRTSPClient * client, GstRTSPMessage * response,
+    gboolean close, gpointer user_data)
+{
+  GstRTSPStatusCode code;
+  const gchar *reason;
+  GstRTSPVersion version;
+
+  fail_unless (gst_rtsp_message_get_type (response) ==
+      GST_RTSP_MESSAGE_RESPONSE);
+
+  fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+          &version)
+      == GST_RTSP_OK);
+  fail_unless (code == GST_RTSP_STS_OK);
+  fail_unless (g_str_equal (reason, "OK"));
+  fail_unless (version == GST_RTSP_VERSION_1_0);
+
+  return TRUE;
+}
+
+static gboolean
+test_response_400 (GstRTSPClient * client, GstRTSPMessage * response,
+    gboolean close, gpointer user_data)
+{
+  GstRTSPStatusCode code;
+  const gchar *reason;
+  GstRTSPVersion version;
+
+  fail_unless (gst_rtsp_message_get_type (response) ==
+      GST_RTSP_MESSAGE_RESPONSE);
+
+  fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+          &version)
+      == GST_RTSP_OK);
+  fail_unless (code == GST_RTSP_STS_BAD_REQUEST);
+  fail_unless (g_str_equal (reason, "Bad Request"));
+  fail_unless (version == GST_RTSP_VERSION_1_0);
+
+  return TRUE;
+}
+
+static gboolean
+test_response_404 (GstRTSPClient * client, GstRTSPMessage * response,
+    gboolean close, gpointer user_data)
+{
+  GstRTSPStatusCode code;
+  const gchar *reason;
+  GstRTSPVersion version;
+
+  fail_unless (gst_rtsp_message_get_type (response) ==
+      GST_RTSP_MESSAGE_RESPONSE);
+
+  fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+          &version)
+      == GST_RTSP_OK);
+  fail_unless (code == GST_RTSP_STS_NOT_FOUND);
+  fail_unless (g_str_equal (reason, "Not Found"));
+  fail_unless (version == GST_RTSP_VERSION_1_0);
+
+  return TRUE;
+}
+
+static gboolean
+test_response_454 (GstRTSPClient * client, GstRTSPMessage * response,
+    gboolean close, gpointer user_data)
+{
+  GstRTSPStatusCode code;
+  const gchar *reason;
+  GstRTSPVersion version;
+
+  fail_unless (gst_rtsp_message_get_type (response) ==
+      GST_RTSP_MESSAGE_RESPONSE);
+
+  fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+          &version)
+      == GST_RTSP_OK);
+  fail_unless (code == GST_RTSP_STS_SESSION_NOT_FOUND);
+  fail_unless (g_str_equal (reason, "Session Not Found"));
+  fail_unless (version == GST_RTSP_VERSION_1_0);
+
+  return TRUE;
+}
+
+static GstRTSPClient *
+setup_client (const gchar * launch_line)
+{
+  GstRTSPClient *client;
+  GstRTSPSessionPool *session_pool;
+  GstRTSPMountPoints *mount_points;
+  GstRTSPMediaFactory *factory;
+  GstRTSPThreadPool *thread_pool;
+
+  client = gst_rtsp_client_new ();
+
+  session_pool = gst_rtsp_session_pool_new ();
+  gst_rtsp_client_set_session_pool (client, session_pool);
+
+  mount_points = gst_rtsp_mount_points_new ();
+  factory = gst_rtsp_media_factory_new ();
+  if (launch_line == NULL)
+    gst_rtsp_media_factory_set_launch (factory,
+        "videotestsrc ! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96");
+  else
+    gst_rtsp_media_factory_set_launch (factory, launch_line);
+
+  gst_rtsp_mount_points_add_factory (mount_points, "/test", factory);
+  gst_rtsp_client_set_mount_points (client, mount_points);
+
+  thread_pool = gst_rtsp_thread_pool_new ();
+  gst_rtsp_client_set_thread_pool (client, thread_pool);
+
+  g_object_unref (mount_points);
+  g_object_unref (session_pool);
+  g_object_unref (thread_pool);
+
+  return client;
+}
+
+static void
+teardown_client (GstRTSPClient * client)
+{
+  gst_rtsp_client_set_thread_pool (client, NULL);
+  g_object_unref (client);
+}
+
+GST_START_TEST (test_request)
+{
+  GstRTSPClient *client;
+  GstRTSPMessage request = { 0, };
+  gchar *str;
+  GstRTSPConnection *conn;
+  GSocket *sock;
+  GError *error = NULL;
+
+  client = gst_rtsp_client_new ();
+
+  /* OPTIONS with invalid url */
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+          "foopy://padoop/") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+  g_free (str);
+
+  gst_rtsp_client_set_send_func (client, test_response_400, NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+
+  gst_rtsp_message_unset (&request);
+
+  /* OPTIONS with unknown session id */
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+          "rtsp://localhost/test") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+  g_free (str);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION, "foobar");
+
+  gst_rtsp_client_set_send_func (client, test_response_454, NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+
+  gst_rtsp_message_unset (&request);
+
+  /* OPTIONS with an absolute path instead of an absolute url */
+  /* set host information */
+  sock = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_STREAM,
+      G_SOCKET_PROTOCOL_TCP, &error);
+  g_assert_no_error (error);
+  gst_rtsp_connection_create_from_socket (sock, "localhost", 444, NULL, &conn);
+  fail_unless (gst_rtsp_client_set_connection (client, conn));
+  g_object_unref (sock);
+
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+          "/test") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+  g_free (str);
+
+  gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+
+  /* OPTIONS with an absolute path instead of an absolute url with invalid
+   * host information */
+  g_object_unref (client);
+  client = gst_rtsp_client_new ();
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+          "/test") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+  g_free (str);
+
+  gst_rtsp_client_set_send_func (client, test_response_400, NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+
+  g_object_unref (client);
+}
+
+GST_END_TEST;
+
+static gboolean
+test_option_response_200 (GstRTSPClient * client, GstRTSPMessage * response,
+    gboolean close, gpointer user_data)
+{
+  GstRTSPStatusCode code;
+  const gchar *reason;
+  GstRTSPVersion version;
+  gchar *str;
+  GstRTSPMethod methods;
+
+  fail_unless (gst_rtsp_message_get_type (response) ==
+      GST_RTSP_MESSAGE_RESPONSE);
+
+  fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+          &version)
+      == GST_RTSP_OK);
+  fail_unless (code == GST_RTSP_STS_OK);
+  fail_unless (g_str_equal (reason, "OK"));
+  fail_unless (version == GST_RTSP_VERSION_1_0);
+
+  fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
+          0) == GST_RTSP_OK);
+  fail_unless (atoi (str) == cseq++);
+
+  fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_PUBLIC, &str,
+          0) == GST_RTSP_OK);
+
+  methods = gst_rtsp_options_from_text (str);
+  fail_if (methods == 0);
+  fail_unless (methods == (GST_RTSP_DESCRIBE |
+          GST_RTSP_OPTIONS |
+          GST_RTSP_PAUSE |
+          GST_RTSP_PLAY |
+          GST_RTSP_SETUP |
+          GST_RTSP_GET_PARAMETER | GST_RTSP_SET_PARAMETER | GST_RTSP_TEARDOWN));
+
+  return TRUE;
+}
+
+GST_START_TEST (test_options)
+{
+  GstRTSPClient *client;
+  GstRTSPMessage request = { 0, };
+  gchar *str;
+
+  client = gst_rtsp_client_new ();
+
+  /* simple OPTIONS */
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+          "rtsp://localhost/test") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+  g_free (str);
+
+  gst_rtsp_client_set_send_func (client, test_option_response_200, NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+
+  g_object_unref (client);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_describe)
+{
+  GstRTSPClient *client;
+  GstRTSPMessage request = { 0, };
+  gchar *str;
+
+  client = gst_rtsp_client_new ();
+
+  /* simple DESCRIBE for non-existing url */
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
+          "rtsp://localhost/test") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+  g_free (str);
+
+  gst_rtsp_client_set_send_func (client, test_response_404, NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+
+  g_object_unref (client);
+
+  /* simple DESCRIBE for an existing url */
+  client = setup_client (NULL);
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
+          "rtsp://localhost/test") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+  g_free (str);
+
+  gst_rtsp_client_set_send_func (client, test_response_200, NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+
+  teardown_client (client);
+}
+
+GST_END_TEST;
+
+static const gchar *expected_transport = NULL;;
+
+static gboolean
+test_setup_response_200_multicast (GstRTSPClient * client,
+    GstRTSPMessage * response, gboolean close, gpointer user_data)
+{
+  GstRTSPStatusCode code;
+  const gchar *reason;
+  GstRTSPVersion version;
+  gchar *str;
+  GstRTSPSessionPool *session_pool;
+  GstRTSPSession *session;
+  gchar **session_hdr_params;
+
+  fail_unless (expected_transport != NULL);
+
+  fail_unless (gst_rtsp_message_get_type (response) ==
+      GST_RTSP_MESSAGE_RESPONSE);
+
+  fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+          &version)
+      == GST_RTSP_OK);
+  fail_unless (code == GST_RTSP_STS_OK);
+  fail_unless (g_str_equal (reason, "OK"));
+  fail_unless (version == GST_RTSP_VERSION_1_0);
+
+  fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
+          0) == GST_RTSP_OK);
+  fail_unless (atoi (str) == cseq++);
+
+  fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_TRANSPORT,
+          &str, 0) == GST_RTSP_OK);
+
+  fail_unless (!strcmp (str, expected_transport));
+
+  fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION,
+          &str, 0) == GST_RTSP_OK);
+  session_hdr_params = g_strsplit (str, ";", -1);
+
+  /* session-id value */
+  fail_unless (session_hdr_params[0] != NULL);
+
+  if (expected_session_timeout != 60) {
+    /* session timeout param */
+    gchar *timeout_str = g_strdup_printf ("timeout=%u",
+        expected_session_timeout);
+
+    fail_unless (session_hdr_params[1] != NULL);
+    g_strstrip (session_hdr_params[1]);
+    fail_unless (g_strcmp0 (session_hdr_params[1], timeout_str) == 0);
+    g_free (timeout_str);
+  }
+
+  session_pool = gst_rtsp_client_get_session_pool (client);
+  fail_unless (session_pool != NULL);
+
+  fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 1);
+  session = gst_rtsp_session_pool_find (session_pool, session_hdr_params[0]);
+  g_strfreev (session_hdr_params);
+
+  /* remember session id to be able to send teardown */
+  session_id = g_strdup (gst_rtsp_session_get_sessionid (session));
+  fail_unless (session_id != NULL);
+
+  fail_unless (session != NULL);
+  g_object_unref (session);
+
+  g_object_unref (session_pool);
+
+
+  return TRUE;
+}
+
+static gboolean
+test_teardown_response_200 (GstRTSPClient * client,
+    GstRTSPMessage * response, gboolean close, gpointer user_data)
+{
+  GstRTSPStatusCode code;
+  const gchar *reason;
+  GstRTSPVersion version;
+
+  fail_unless (gst_rtsp_message_get_type (response) ==
+      GST_RTSP_MESSAGE_RESPONSE);
+
+  fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+          &version)
+      == GST_RTSP_OK);
+  fail_unless (code == GST_RTSP_STS_OK);
+  fail_unless (g_str_equal (reason, "OK"));
+  fail_unless (version == GST_RTSP_VERSION_1_0);
+
+  return TRUE;
+}
+
+static void
+send_teardown (GstRTSPClient * client)
+{
+  GstRTSPMessage request = { 0, };
+  gchar *str;
+
+  fail_unless (session_id != NULL);
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN,
+          "rtsp://localhost/test") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SESSION,
+      session_id);
+  gst_rtsp_client_set_send_func (client, test_teardown_response_200,
+      NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+  g_free (session_id);
+  session_id = NULL;
+}
+
+static GstRTSPClient *
+setup_multicast_client (void)
+{
+  GstRTSPClient *client;
+  GstRTSPSessionPool *session_pool;
+  GstRTSPMountPoints *mount_points;
+  GstRTSPMediaFactory *factory;
+  GstRTSPAddressPool *address_pool;
+  GstRTSPThreadPool *thread_pool;
+
+  client = gst_rtsp_client_new ();
+
+  session_pool = gst_rtsp_session_pool_new ();
+  gst_rtsp_client_set_session_pool (client, session_pool);
+
+  mount_points = gst_rtsp_mount_points_new ();
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory,
+      "audiotestsrc ! audio/x-raw,rate=44100 ! audioconvert ! rtpL16pay name=pay0");
+  address_pool = gst_rtsp_address_pool_new ();
+  fail_unless (gst_rtsp_address_pool_add_range (address_pool,
+          "233.252.0.1", "233.252.0.1", 5000, 5010, 1));
+  gst_rtsp_media_factory_set_address_pool (factory, address_pool);
+  gst_rtsp_media_factory_add_role (factory, "user",
+      "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+      "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
+  gst_rtsp_mount_points_add_factory (mount_points, "/test", factory);
+  gst_rtsp_client_set_mount_points (client, mount_points);
+
+  thread_pool = gst_rtsp_thread_pool_new ();
+  gst_rtsp_client_set_thread_pool (client, thread_pool);
+
+  g_object_unref (mount_points);
+  g_object_unref (session_pool);
+  g_object_unref (address_pool);
+  g_object_unref (thread_pool);
+
+  return client;
+}
+
+GST_START_TEST (test_client_multicast_transport_404)
+{
+  GstRTSPClient *client;
+  GstRTSPMessage request = { 0, };
+  gchar *str;
+
+  client = setup_multicast_client ();
+
+  /* simple SETUP for non-existing url */
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+          "rtsp://localhost/test2/stream=0") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+      "RTP/AVP;multicast");
+
+  gst_rtsp_client_set_send_func (client, test_response_404, NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+
+  teardown_client (client);
+}
+
+GST_END_TEST;
+
+static void
+new_session_cb (GObject * client, GstRTSPSession * session, gpointer user_data)
+{
+  GST_DEBUG ("%p: new session %p", client, session);
+  gst_rtsp_session_set_timeout (session, expected_session_timeout);
+}
+
+GST_START_TEST (test_client_multicast_transport)
+{
+  GstRTSPClient *client;
+  GstRTSPMessage request = { 0, };
+  gchar *str;
+
+  client = setup_multicast_client ();
+
+  expected_session_timeout = 20;
+  g_signal_connect (G_OBJECT (client), "new-session",
+      G_CALLBACK (new_session_cb), NULL);
+
+  /* simple SETUP with a valid URI and multicast */
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+          "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+      "RTP/AVP;multicast");
+
+  expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
+      "ttl=1;port=5000-5001;mode=\"PLAY\"";
+  gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
+      NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+  expected_transport = NULL;
+  expected_session_timeout = 60;
+
+  send_teardown (client);
+
+  teardown_client (client);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_client_multicast_ignore_transport_specific)
+{
+  GstRTSPClient *client;
+  GstRTSPMessage request = { 0, };
+  gchar *str;
+
+  client = setup_multicast_client ();
+
+  /* simple SETUP with a valid URI and multicast and a specific dest,
+   * but ignore it  */
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+          "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+      "RTP/AVP;multicast;destination=233.252.0.2;ttl=2;port=5001-5006;");
+
+  expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
+      "ttl=1;port=5000-5001;mode=\"PLAY\"";
+  gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
+      NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+  expected_transport = NULL;
+
+  send_teardown (client);
+
+  teardown_client (client);
+}
+
+GST_END_TEST;
+
+static gboolean
+test_setup_response_461 (GstRTSPClient * client,
+    GstRTSPMessage * response, gboolean close, gpointer user_data)
+{
+  GstRTSPStatusCode code;
+  const gchar *reason;
+  GstRTSPVersion version;
+  gchar *str;
+
+  fail_unless (expected_transport == NULL);
+
+  fail_unless (gst_rtsp_message_get_type (response) ==
+      GST_RTSP_MESSAGE_RESPONSE);
+
+  fail_unless (gst_rtsp_message_parse_response (response, &code, &reason,
+          &version)
+      == GST_RTSP_OK);
+  fail_unless (code == GST_RTSP_STS_UNSUPPORTED_TRANSPORT);
+  fail_unless (g_str_equal (reason, "Unsupported transport"));
+  fail_unless (version == GST_RTSP_VERSION_1_0);
+
+  fail_unless (gst_rtsp_message_get_header (response, GST_RTSP_HDR_CSEQ, &str,
+          0) == GST_RTSP_OK);
+  fail_unless (atoi (str) == cseq++);
+
+
+  return TRUE;
+}
+
+GST_START_TEST (test_client_multicast_invalid_transport_specific)
+{
+  GstRTSPClient *client;
+  GstRTSPMessage request = { 0, };
+  gchar *str;
+  GstRTSPSessionPool *session_pool;
+  GstRTSPContext ctx = { NULL };
+
+  client = setup_multicast_client ();
+
+  ctx.client = client;
+  ctx.auth = gst_rtsp_auth_new ();
+  ctx.token =
+      gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
+      G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+      "user", NULL);
+  gst_rtsp_context_push_current (&ctx);
+
+  /* simple SETUP with a valid URI and multicast, but an invalid ip */
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+          "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+      "RTP/AVP;multicast;destination=233.252.0.2;ttl=1;port=5000-5001;");
+
+  gst_rtsp_client_set_send_func (client, test_setup_response_461, NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+
+  session_pool = gst_rtsp_client_get_session_pool (client);
+  fail_unless (session_pool != NULL);
+  /* FIXME: There seems to be a leak of a session here ! */
+  fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 0);
+  g_object_unref (session_pool);
+
+
+
+  /* simple SETUP with a valid URI and multicast, but an invalid prt */
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+          "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+      "RTP/AVP;multicast;destination=233.252.0.1;ttl=1;port=6000-6001;");
+
+  gst_rtsp_client_set_send_func (client, test_setup_response_461, NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+
+  session_pool = gst_rtsp_client_get_session_pool (client);
+  fail_unless (session_pool != NULL);
+  /* FIXME: There seems to be a leak of a session here ! */
+  fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 0);
+  g_object_unref (session_pool);
+
+
+
+  /* simple SETUP with a valid URI and multicast, but an invalid ttl */
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+          "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+      "RTP/AVP;multicast;destination=233.252.0.1;ttl=2;port=5000-5001;");
+
+  gst_rtsp_client_set_send_func (client, test_setup_response_461, NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+
+  session_pool = gst_rtsp_client_get_session_pool (client);
+  fail_unless (session_pool != NULL);
+  /* FIXME: There seems to be a leak of a session here ! */
+  fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 0);
+  g_object_unref (session_pool);
+
+
+  teardown_client (client);
+  g_object_unref (ctx.auth);
+  gst_rtsp_token_unref (ctx.token);
+  gst_rtsp_context_pop_current (&ctx);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_client_multicast_transport_specific)
+{
+  GstRTSPClient *client;
+  GstRTSPMessage request = { 0, };
+  gchar *str;
+  GstRTSPSessionPool *session_pool;
+  GstRTSPContext ctx = { NULL };
+
+  client = setup_multicast_client ();
+
+  ctx.client = client;
+  ctx.auth = gst_rtsp_auth_new ();
+  ctx.token =
+      gst_rtsp_token_new (GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS,
+      G_TYPE_BOOLEAN, TRUE, GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE, G_TYPE_STRING,
+      "user", NULL);
+  gst_rtsp_context_push_current (&ctx);
+
+  expected_transport = "RTP/AVP;multicast;destination=233.252.0.1;"
+      "ttl=1;port=5000-5001;mode=\"PLAY\"";
+
+  /* simple SETUP with a valid URI and multicast, but an invalid ip */
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+          "rtsp://localhost/test/stream=0") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_take_header (&request, GST_RTSP_HDR_CSEQ, str);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_TRANSPORT,
+      expected_transport);
+
+  gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
+      NULL, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+  expected_transport = NULL;
+
+  gst_rtsp_client_set_send_func (client, test_setup_response_200_multicast,
+      NULL, NULL);
+  session_pool = gst_rtsp_client_get_session_pool (client);
+  fail_unless (session_pool != NULL);
+  fail_unless (gst_rtsp_session_pool_get_n_sessions (session_pool) == 1);
+  g_object_unref (session_pool);
+
+  send_teardown (client);
+
+  teardown_client (client);
+  g_object_unref (ctx.auth);
+  gst_rtsp_token_unref (ctx.token);
+  gst_rtsp_context_pop_current (&ctx);
+}
+
+GST_END_TEST;
+
+static gboolean
+test_response_sdp (GstRTSPClient * client, GstRTSPMessage * response,
+    gboolean close, gpointer user_data)
+{
+  guint8 *data;
+  guint size;
+  GstSDPMessage *sdp_msg;
+  const GstSDPMedia *sdp_media;
+  const GstSDPBandwidth *bw;
+  gint bandwidth_val = GPOINTER_TO_INT (user_data);
+
+  fail_unless (gst_rtsp_message_get_body (response, &data, &size)
+      == GST_RTSP_OK);
+  gst_sdp_message_new (&sdp_msg);
+  fail_unless (gst_sdp_message_parse_buffer (data, size, sdp_msg)
+      == GST_SDP_OK);
+
+  /* session description */
+  /* v= */
+  fail_unless (gst_sdp_message_get_version (sdp_msg) != NULL);
+  /* o= */
+  fail_unless (gst_sdp_message_get_origin (sdp_msg) != NULL);
+  /* s= */
+  fail_unless (gst_sdp_message_get_session_name (sdp_msg) != NULL);
+  /* t=0 0 */
+  fail_unless (gst_sdp_message_times_len (sdp_msg) == 0);
+
+  /* verify number of medias */
+  fail_unless (gst_sdp_message_medias_len (sdp_msg) == 1);
+
+  /* media description */
+  sdp_media = gst_sdp_message_get_media (sdp_msg, 0);
+  fail_unless (sdp_media != NULL);
+
+  /* m= */
+  fail_unless (gst_sdp_media_get_media (sdp_media) != NULL);
+
+  /* media bandwidth */
+  if (bandwidth_val) {
+    fail_unless (gst_sdp_media_bandwidths_len (sdp_media) == 1);
+    bw = gst_sdp_media_get_bandwidth (sdp_media, 0);
+    fail_unless (bw != NULL);
+    fail_unless (g_strcmp0 (bw->bwtype, "AS") == 0);
+    fail_unless (bw->bandwidth == bandwidth_val);
+  } else {
+    fail_unless (gst_sdp_media_bandwidths_len (sdp_media) == 0);
+  }
+
+  gst_sdp_message_free (sdp_msg);
+
+  return TRUE;
+}
+
+static void
+test_client_sdp (const gchar * launch_line, guint * bandwidth_val)
+{
+  GstRTSPClient *client;
+  GstRTSPMessage request = { 0, };
+  gchar *str;
+
+  /* simple DESCRIBE for an existing url */
+  client = setup_client (launch_line);
+  fail_unless (gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
+          "rtsp://localhost/test") == GST_RTSP_OK);
+  str = g_strdup_printf ("%d", cseq);
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_CSEQ, str);
+  g_free (str);
+
+  gst_rtsp_client_set_send_func (client, test_response_sdp,
+      (gpointer) bandwidth_val, NULL);
+  fail_unless (gst_rtsp_client_handle_message (client,
+          &request) == GST_RTSP_OK);
+  gst_rtsp_message_unset (&request);
+
+  teardown_client (client);
+}
+
+GST_START_TEST (test_client_sdp_with_max_bitrate_tag)
+{
+  test_client_sdp ("videotestsrc "
+      "! taginject tags=\"maximum-bitrate=(uint)50000000\" "
+      "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+      GUINT_TO_POINTER (50000));
+
+
+  /* max-bitrate=0: no bandwidth line */
+  test_client_sdp ("videotestsrc "
+      "! taginject tags=\"maximum-bitrate=(uint)0\" "
+      "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+      GUINT_TO_POINTER (0));
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_client_sdp_with_bitrate_tag)
+{
+  test_client_sdp ("videotestsrc "
+      "! taginject tags=\"bitrate=(uint)7000000\" "
+      "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+      GUINT_TO_POINTER (7000));
+
+  /* bitrate=0: no bandwdith line */
+  test_client_sdp ("videotestsrc "
+      "! taginject tags=\"bitrate=(uint)0\" "
+      "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+      GUINT_TO_POINTER (0));
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_client_sdp_with_max_bitrate_and_bitrate_tags)
+{
+  test_client_sdp ("videotestsrc "
+      "! taginject tags=\"bitrate=(uint)7000000,maximum-bitrate=(uint)50000000\" "
+      "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+      GUINT_TO_POINTER (50000));
+
+  /* max-bitrate is zero: fallback to bitrate */
+  test_client_sdp ("videotestsrc "
+      "! taginject tags=\"bitrate=(uint)7000000,maximum-bitrate=(uint)0\" "
+      "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+      GUINT_TO_POINTER (7000));
+
+  /* max-bitrate=bitrate=0o: no bandwidth line */
+  test_client_sdp ("videotestsrc "
+      "! taginject tags=\"bitrate=(uint)0,maximum-bitrate=(uint)0\" "
+      "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96",
+      GUINT_TO_POINTER (0));
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_client_sdp_with_no_bitrate_tags)
+{
+  test_client_sdp ("videotestsrc "
+      "! video/x-raw,width=352,height=288 ! rtpgstpay name=pay0 pt=96", NULL);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspclient_suite (void)
+{
+  Suite *s = suite_create ("rtspclient");
+  TCase *tc = tcase_create ("general");
+
+  suite_add_tcase (s, tc);
+  tcase_set_timeout (tc, 20);
+  tcase_add_test (tc, test_request);
+  tcase_add_test (tc, test_options);
+  tcase_add_test (tc, test_describe);
+  tcase_add_test (tc, test_client_multicast_transport_404);
+  tcase_add_test (tc, test_client_multicast_transport);
+  tcase_add_test (tc, test_client_multicast_ignore_transport_specific);
+  tcase_add_test (tc, test_client_multicast_invalid_transport_specific);
+  tcase_add_test (tc, test_client_multicast_transport_specific);
+  tcase_add_test (tc, test_client_sdp_with_max_bitrate_tag);
+  tcase_add_test (tc, test_client_sdp_with_bitrate_tag);
+  tcase_add_test (tc, test_client_sdp_with_max_bitrate_and_bitrate_tags);
+  tcase_add_test (tc, test_client_sdp_with_no_bitrate_tags);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtspclient);
diff --git a/tests/check/gst/media.c b/tests/check/gst/media.c
new file mode 100644
index 0000000..c7d680e
--- /dev/null
+++ b/tests/check/gst/media.c
@@ -0,0 +1,434 @@
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+GST_START_TEST (test_launch)
+{
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia *media;
+  GstRTSPUrl *url;
+  GstRTSPStream *stream;
+  GstRTSPTimeRange *range;
+  gchar *str;
+  GstRTSPThreadPool *pool;
+  GstRTSPThread *thread;
+
+  factory = gst_rtsp_media_factory_new ();
+  fail_if (gst_rtsp_media_factory_is_shared (factory));
+  fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+          &url) == GST_RTSP_OK);
+
+  gst_rtsp_media_factory_set_launch (factory,
+      "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+
+  fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+  stream = gst_rtsp_media_get_stream (media, 0);
+  fail_unless (stream != NULL);
+
+  /* fails, need to be prepared */
+  str = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
+  fail_unless (str == NULL);
+
+  fail_unless (gst_rtsp_range_parse ("npt=5.0-", &range) == GST_RTSP_OK);
+  /* fails, need to be prepared */
+  fail_if (gst_rtsp_media_seek (media, range));
+
+  pool = gst_rtsp_thread_pool_new ();
+  thread = gst_rtsp_thread_pool_get_thread (pool,
+      GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+  fail_unless (gst_rtsp_media_prepare (media, thread));
+
+  str = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
+  fail_unless (g_str_equal (str, "npt=0-"));
+  g_free (str);
+
+  str = gst_rtsp_media_get_range_string (media, TRUE, GST_RTSP_RANGE_NPT);
+  fail_unless (g_str_equal (str, "npt=0-"));
+  g_free (str);
+
+  fail_unless (gst_rtsp_media_seek (media, range));
+
+  str = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
+  fail_unless (g_str_equal (str, "npt=5-"));
+  g_free (str);
+
+  str = gst_rtsp_media_get_range_string (media, TRUE, GST_RTSP_RANGE_NPT);
+  fail_unless (g_str_equal (str, "npt=5-"));
+  g_free (str);
+
+  fail_unless (gst_rtsp_media_unprepare (media));
+
+  /* should fail again */
+  str = gst_rtsp_media_get_range_string (media, FALSE, GST_RTSP_RANGE_NPT);
+  fail_unless (str == NULL);
+  fail_if (gst_rtsp_media_seek (media, range));
+
+  gst_rtsp_range_free (range);
+  g_object_unref (media);
+
+  gst_rtsp_url_free (url);
+  g_object_unref (factory);
+
+  g_object_unref (pool);
+
+  gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_media)
+{
+  GstRTSPMedia *media;
+  GstElement *bin, *e1, *e2;
+
+  bin = gst_bin_new ("bin");
+  fail_if (bin == NULL);
+
+  e1 = gst_element_factory_make ("videotestsrc", NULL);
+  fail_if (e1 == NULL);
+
+  e2 = gst_element_factory_make ("rtpvrawpay", "pay0");
+  fail_if (e2 == NULL);
+  g_object_set (e2, "pt", 96, NULL);
+
+  gst_bin_add_many (GST_BIN_CAST (bin), e1, e2, NULL);
+  gst_element_link_many (e1, e2, NULL);
+
+  media = gst_rtsp_media_new (bin);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+  g_object_unref (media);
+}
+
+GST_END_TEST;
+
+static void
+test_prepare_reusable (GstRTSPThreadPool * pool, const gchar * launch_line)
+{
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia *media;
+  GstRTSPUrl *url;
+  GstRTSPThread *thread;
+
+  factory = gst_rtsp_media_factory_new ();
+  fail_if (gst_rtsp_media_factory_is_shared (factory));
+  fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+          &url) == GST_RTSP_OK);
+
+  gst_rtsp_media_factory_set_launch (factory, launch_line);
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+  fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+  g_object_set (G_OBJECT (media), "reusable", TRUE, NULL);
+
+  thread = gst_rtsp_thread_pool_get_thread (pool,
+      GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+  fail_unless (gst_rtsp_media_prepare (media, thread));
+  fail_unless (gst_rtsp_media_unprepare (media));
+  fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+  thread = gst_rtsp_thread_pool_get_thread (pool,
+      GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+  fail_unless (gst_rtsp_media_prepare (media, thread));
+  fail_unless (gst_rtsp_media_unprepare (media));
+
+  g_object_unref (media);
+  gst_rtsp_url_free (url);
+  g_object_unref (factory);
+}
+
+GST_START_TEST (test_media_prepare)
+{
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia *media;
+  GstRTSPUrl *url;
+  GstRTSPThreadPool *pool;
+  GstRTSPThread *thread;
+
+  pool = gst_rtsp_thread_pool_new ();
+
+  /* test non-reusable media first */
+  factory = gst_rtsp_media_factory_new ();
+  fail_if (gst_rtsp_media_factory_is_shared (factory));
+  fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+          &url) == GST_RTSP_OK);
+
+  gst_rtsp_media_factory_set_launch (factory,
+      "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+  fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+  thread = gst_rtsp_thread_pool_get_thread (pool,
+      GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+  fail_unless (gst_rtsp_media_prepare (media, thread));
+  fail_unless (gst_rtsp_media_unprepare (media));
+  fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+  thread = gst_rtsp_thread_pool_get_thread (pool,
+      GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+  fail_if (gst_rtsp_media_prepare (media, thread));
+
+  g_object_unref (media);
+  gst_rtsp_url_free (url);
+  g_object_unref (factory);
+
+  /* test reusable media */
+  test_prepare_reusable (pool, "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+  test_prepare_reusable (pool,
+      "( videotestsrc is-live=true ! rtpvrawpay pt=96 name=pay0 )");
+
+  g_object_unref (pool);
+  gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+static void
+on_notify_caps (GstPad * pad, GParamSpec * pspec, GstElement * pay)
+{
+  GstCaps *caps;
+  static gboolean have_caps = FALSE;
+
+  g_object_get (pad, "caps", &caps, NULL);
+
+  GST_DEBUG ("notify %" GST_PTR_FORMAT, caps);
+
+  if (caps) {
+    if (!have_caps) {
+      g_signal_emit_by_name (pay, "pad-added", pad);
+      g_signal_emit_by_name (pay, "no-more-pads", NULL);
+      have_caps = TRUE;
+    }
+    gst_caps_unref (caps);
+  } else {
+    if (have_caps) {
+      g_signal_emit_by_name (pay, "pad-removed", pad);
+      have_caps = FALSE;
+    }
+  }
+}
+
+GST_START_TEST (test_media_dyn_prepare)
+{
+  GstRTSPMedia *media;
+  GstElement *bin, *src, *pay;
+  GstElement *pipeline;
+  GstPad *srcpad;
+  GstRTSPThreadPool *pool;
+  GstRTSPThread *thread;
+
+  bin = gst_bin_new ("bin");
+  fail_if (bin == NULL);
+
+  src = gst_element_factory_make ("videotestsrc", NULL);
+  fail_if (src == NULL);
+
+  pay = gst_element_factory_make ("rtpvrawpay", "dynpay0");
+  fail_if (pay == NULL);
+  g_object_set (pay, "pt", 96, NULL);
+
+  gst_bin_add_many (GST_BIN_CAST (bin), src, pay, NULL);
+  gst_element_link_many (src, pay, NULL);
+
+  media = gst_rtsp_media_new (bin);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+
+  g_object_set (G_OBJECT (media), "reusable", TRUE, NULL);
+
+  pipeline = gst_pipeline_new ("media-pipeline");
+  gst_rtsp_media_take_pipeline (media, GST_PIPELINE_CAST (pipeline));
+
+  gst_rtsp_media_collect_streams (media);
+
+  srcpad = gst_element_get_static_pad (pay, "src");
+
+  g_signal_connect (srcpad, "notify::caps", (GCallback) on_notify_caps, pay);
+
+  pool = gst_rtsp_thread_pool_new ();
+
+  fail_unless (gst_rtsp_media_n_streams (media) == 0);
+
+  thread = gst_rtsp_thread_pool_get_thread (pool,
+      GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+  fail_unless (gst_rtsp_media_prepare (media, thread));
+  fail_unless (gst_rtsp_media_n_streams (media) == 1);
+  fail_unless (gst_rtsp_media_unprepare (media));
+  fail_unless (gst_rtsp_media_n_streams (media) == 0);
+
+  fail_unless (gst_rtsp_media_n_streams (media) == 0);
+
+  thread = gst_rtsp_thread_pool_get_thread (pool,
+      GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+  fail_unless (gst_rtsp_media_prepare (media, thread));
+  fail_unless (gst_rtsp_media_n_streams (media) == 1);
+  fail_unless (gst_rtsp_media_unprepare (media));
+  fail_unless (gst_rtsp_media_n_streams (media) == 0);
+
+  gst_object_unref (srcpad);
+  g_object_unref (media);
+  g_object_unref (pool);
+
+  gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_media_prepare_port_alloc_fail)
+{
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia *media;
+  GstRTSPUrl *url;
+  GstRTSPThreadPool *pool;
+  GstRTSPThread *thread;
+  GstRTSPAddressPool *addrpool;
+
+  pool = gst_rtsp_thread_pool_new ();
+
+  factory = gst_rtsp_media_factory_new ();
+  fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+          &url) == GST_RTSP_OK);
+
+  gst_rtsp_media_factory_set_launch (factory,
+      "( fakesrc is-live=true ! text/plain ! rtpgstpay name=pay0 )");
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+
+  addrpool = gst_rtsp_address_pool_new ();
+  fail_unless (gst_rtsp_address_pool_add_range (addrpool, "192.168.1.1",
+          "192.168.1.1", 6000, 6001, 0));
+  gst_rtsp_media_set_address_pool (media, addrpool);
+
+  thread = gst_rtsp_thread_pool_get_thread (pool,
+      GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+  fail_if (gst_rtsp_media_prepare (media, thread));
+
+  g_object_unref (media);
+  g_object_unref (addrpool);
+  gst_rtsp_url_free (url);
+  g_object_unref (factory);
+  g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_media_take_pipeline)
+{
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia *media;
+  GstRTSPUrl *url;
+  GstElement *pipeline;
+
+  factory = gst_rtsp_media_factory_new ();
+  fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+          &url) == GST_RTSP_OK);
+  gst_rtsp_media_factory_set_launch (factory,
+      "( fakesrc ! text/plain ! rtpgstpay name=pay0 )");
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+
+  pipeline = gst_pipeline_new ("media-pipeline");
+  gst_rtsp_media_take_pipeline (media, GST_PIPELINE_CAST (pipeline));
+
+  g_object_unref (media);
+  gst_rtsp_url_free (url);
+  g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_media_reset)
+{
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia *media;
+  GstRTSPUrl *url;
+  GstRTSPThreadPool *pool;
+  GstRTSPThread *thread;
+
+  pool = gst_rtsp_thread_pool_new ();
+
+  factory = gst_rtsp_media_factory_new ();
+  fail_if (gst_rtsp_media_factory_is_shared (factory));
+  gst_rtsp_url_parse ("rtsp://localhost:8554/test", &url);
+
+  gst_rtsp_media_factory_set_launch (factory,
+      "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+
+  thread = gst_rtsp_thread_pool_get_thread (pool,
+      GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+  fail_unless (gst_rtsp_media_prepare (media, thread));
+  fail_unless (gst_rtsp_media_suspend (media));
+  fail_unless (gst_rtsp_media_unprepare (media));
+  g_object_unref (media);
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+
+  thread = gst_rtsp_thread_pool_get_thread (pool,
+      GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+  gst_rtsp_media_set_suspend_mode (media, GST_RTSP_SUSPEND_MODE_RESET);
+  fail_unless (gst_rtsp_media_prepare (media, thread));
+  fail_unless (gst_rtsp_media_suspend (media));
+  fail_unless (gst_rtsp_media_unprepare (media));
+  g_object_unref (media);
+
+  gst_rtsp_url_free (url);
+  g_object_unref (factory);
+  g_object_unref (pool);
+
+  gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspmedia_suite (void)
+{
+  Suite *s = suite_create ("rtspmedia");
+  TCase *tc = tcase_create ("general");
+
+  suite_add_tcase (s, tc);
+  tcase_set_timeout (tc, 20);
+  tcase_add_test (tc, test_launch);
+  tcase_add_test (tc, test_media);
+  tcase_add_test (tc, test_media_prepare);
+  tcase_add_test (tc, test_media_dyn_prepare);
+  tcase_add_test (tc, test_media_prepare_port_alloc_fail);
+  tcase_add_test (tc, test_media_take_pipeline);
+  tcase_add_test (tc, test_media_reset);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtspmedia);
diff --git a/tests/check/gst/mediafactory.c b/tests/check/gst/mediafactory.c
new file mode 100644
index 0000000..b6b250b
--- /dev/null
+++ b/tests/check/gst/mediafactory.c
@@ -0,0 +1,336 @@
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+GST_START_TEST (test_parse_error)
+{
+  GstRTSPMediaFactory *factory;
+  GstRTSPUrl *url;
+
+  factory = gst_rtsp_media_factory_new ();
+
+  gst_rtsp_media_factory_set_launch (factory, "foo");
+  fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+          &url) == GST_RTSP_OK);
+  ASSERT_CRITICAL (gst_rtsp_media_factory_create_element (factory, url));
+  ASSERT_CRITICAL (gst_rtsp_media_factory_construct (factory, url));
+
+  gst_rtsp_url_free (url);
+  g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_launch)
+{
+  GstRTSPMediaFactory *factory;
+  GstElement *element;
+  GstRTSPUrl *url;
+
+  factory = gst_rtsp_media_factory_new ();
+  fail_if (gst_rtsp_media_factory_is_shared (factory));
+  fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+          &url) == GST_RTSP_OK);
+
+  gst_rtsp_media_factory_set_launch (factory,
+      "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+  element = gst_rtsp_media_factory_create_element (factory, url);
+  fail_unless (GST_IS_BIN (element));
+  fail_if (GST_IS_PIPELINE (element));
+  gst_object_unref (element);
+
+  gst_rtsp_url_free (url);
+  g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_launch_construct)
+{
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia *media, *media2;
+  GstRTSPUrl *url;
+
+  factory = gst_rtsp_media_factory_new ();
+  fail_if (gst_rtsp_media_factory_is_shared (factory));
+  fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+          &url) == GST_RTSP_OK);
+
+  gst_rtsp_media_factory_set_launch (factory,
+      "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+
+  media2 = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media2));
+  fail_if (media == media2);
+
+  g_object_unref (media);
+  g_object_unref (media2);
+
+  gst_rtsp_url_free (url);
+  g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_shared)
+{
+  GstRTSPMediaFactory *factory;
+  GstElement *element;
+  GstRTSPMedia *media, *media2;
+  GstRTSPUrl *url;
+
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_shared (factory, TRUE);
+  fail_unless (gst_rtsp_media_factory_is_shared (factory));
+
+  fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+          &url) == GST_RTSP_OK);
+
+  gst_rtsp_media_factory_set_launch (factory,
+      "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+  element = gst_rtsp_media_factory_create_element (factory, url);
+  fail_unless (GST_IS_BIN (element));
+  fail_if (GST_IS_PIPELINE (element));
+  gst_object_unref (element);
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+
+  media2 = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media2));
+  fail_unless (media == media2);
+
+  g_object_unref (media);
+  g_object_unref (media2);
+
+  gst_rtsp_url_free (url);
+  g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_addresspool)
+{
+  GstRTSPMediaFactory *factory;
+  GstElement *element;
+  GstRTSPMedia *media;
+  GstRTSPUrl *url;
+  GstRTSPAddressPool *pool, *tmppool;
+  GstRTSPStream *stream;
+  GstRTSPAddress *addr;
+
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_shared (factory, TRUE);
+  fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+          &url) == GST_RTSP_OK);
+
+  gst_rtsp_media_factory_set_launch (factory,
+      "( videotestsrc ! rtpvrawpay pt=96 name=pay0 "
+      " audiotestsrc ! audioconvert ! rtpL16pay name=pay1 )");
+
+  pool = gst_rtsp_address_pool_new ();
+  fail_unless (gst_rtsp_address_pool_add_range (pool,
+          "233.252.0.1", "233.252.0.1", 5000, 5001, 3));
+
+  gst_rtsp_media_factory_set_address_pool (factory, pool);
+
+  tmppool = gst_rtsp_media_factory_get_address_pool (factory);
+  fail_unless (pool == tmppool);
+  g_object_unref (tmppool);
+
+  element = gst_rtsp_media_factory_create_element (factory, url);
+  fail_unless (GST_IS_BIN (element));
+  fail_if (GST_IS_PIPELINE (element));
+  gst_object_unref (element);
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+
+  tmppool = gst_rtsp_media_get_address_pool (media);
+  fail_unless (pool == tmppool);
+  g_object_unref (tmppool);
+
+  fail_unless (gst_rtsp_media_n_streams (media) == 2);
+
+  stream = gst_rtsp_media_get_stream (media, 0);
+  fail_unless (stream != NULL);
+
+  tmppool = gst_rtsp_stream_get_address_pool (stream);
+  fail_unless (pool == tmppool);
+  g_object_unref (tmppool);
+
+  addr = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
+  fail_unless (addr != NULL);
+  fail_unless (addr->port == 5000);
+  fail_unless (addr->n_ports == 2);
+  fail_unless (addr->ttl == 3);
+  gst_rtsp_address_free (addr);
+
+  stream = gst_rtsp_media_get_stream (media, 1);
+  fail_unless (stream != NULL);
+
+  tmppool = gst_rtsp_stream_get_address_pool (stream);
+  fail_unless (pool == tmppool);
+  g_object_unref (tmppool);
+
+  addr = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
+  fail_unless (addr == NULL);
+
+
+  g_object_unref (media);
+
+  g_object_unref (pool);
+  gst_rtsp_url_free (url);
+  g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_permissions)
+{
+  GstRTSPMediaFactory *factory;
+  GstRTSPPermissions *perms;
+  GstRTSPMedia *media;
+  GstRTSPUrl *url;
+
+  factory = gst_rtsp_media_factory_new ();
+  fail_if (gst_rtsp_media_factory_is_shared (factory));
+  fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+          &url) == GST_RTSP_OK);
+
+  gst_rtsp_media_factory_set_launch (factory,
+      "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+  gst_rtsp_media_factory_add_role (factory, "admin",
+      "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+      "media.factory.construct", G_TYPE_BOOLEAN, TRUE, NULL);
+
+  perms = gst_rtsp_media_factory_get_permissions (factory);
+  fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin",
+          "media.factory.access"));
+  fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin",
+          "media.factory.construct"));
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "missing",
+          "media.factory.access"));
+  gst_rtsp_permissions_unref (perms);
+
+  perms = gst_rtsp_permissions_new ();
+  gst_rtsp_permissions_add_role (perms, "user",
+      "media.factory.access", G_TYPE_BOOLEAN, TRUE,
+      "media.factory.construct", G_TYPE_BOOLEAN, FALSE, NULL);
+  gst_rtsp_media_factory_set_permissions (factory, perms);
+  gst_rtsp_permissions_unref (perms);
+
+  perms = gst_rtsp_media_factory_get_permissions (factory);
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "admin",
+          "media.factory.access"));
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "admin",
+          "media.factory.construct"));
+  fail_unless (gst_rtsp_permissions_is_allowed (perms, "user",
+          "media.factory.access"));
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "user",
+          "media.factory.construct"));
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "missing",
+          "media.factory.access"));
+  gst_rtsp_permissions_unref (perms);
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+  perms = gst_rtsp_media_get_permissions (media);
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "admin",
+          "media.factory.access"));
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "admin",
+          "media.factory.construct"));
+  fail_unless (gst_rtsp_permissions_is_allowed (perms, "user",
+          "media.factory.access"));
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "user",
+          "media.factory.construct"));
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "missing",
+          "media.factory.access"));
+  gst_rtsp_permissions_unref (perms);
+  g_object_unref (media);
+
+  gst_rtsp_url_free (url);
+  g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_reset)
+{
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia *media;
+  GstRTSPUrl *url;
+
+  factory = gst_rtsp_media_factory_new ();
+  fail_if (gst_rtsp_media_factory_is_shared (factory));
+  gst_rtsp_url_parse ("rtsp://localhost:8554/test", &url);
+
+  gst_rtsp_media_factory_set_launch (factory,
+      "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+  fail_if (gst_rtsp_media_get_suspend_mode (media) !=
+      GST_RTSP_SUSPEND_MODE_NONE);
+  g_object_unref (media);
+
+  gst_rtsp_media_factory_set_suspend_mode (factory,
+      GST_RTSP_SUSPEND_MODE_RESET);
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+  fail_if (gst_rtsp_media_get_suspend_mode (media) !=
+      GST_RTSP_SUSPEND_MODE_RESET);
+  g_object_unref (media);
+
+  gst_rtsp_url_free (url);
+  g_object_unref (factory);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspmediafactory_suite (void)
+{
+  Suite *s = suite_create ("rtspmediafactory");
+  TCase *tc = tcase_create ("general");
+
+  suite_add_tcase (s, tc);
+  tcase_set_timeout (tc, 20);
+  tcase_add_test (tc, test_parse_error);
+  tcase_add_test (tc, test_launch);
+  tcase_add_test (tc, test_launch_construct);
+  tcase_add_test (tc, test_shared);
+  tcase_add_test (tc, test_addresspool);
+  tcase_add_test (tc, test_permissions);
+  tcase_add_test (tc, test_reset);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtspmediafactory);
diff --git a/tests/check/gst/mountpoints.c b/tests/check/gst/mountpoints.c
new file mode 100644
index 0000000..31a223e
--- /dev/null
+++ b/tests/check/gst/mountpoints.c
@@ -0,0 +1,147 @@
+/* GStreamer
+ * Copyright (C) 2012 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+GST_START_TEST (test_create)
+{
+  GstRTSPMountPoints *mounts;
+  GstRTSPUrl *url, *url2;
+  GstRTSPMediaFactory *factory;
+
+  mounts = gst_rtsp_mount_points_new ();
+
+  fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test",
+          &url) == GST_RTSP_OK);
+  fail_unless (gst_rtsp_url_parse ("rtsp://localhost:8554/test2",
+          &url2) == GST_RTSP_OK);
+
+  fail_unless (gst_rtsp_mount_points_match (mounts, url->abspath,
+          NULL) == NULL);
+
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_mount_points_add_factory (mounts, "/test", factory);
+
+  fail_unless (gst_rtsp_mount_points_match (mounts, url->abspath,
+          NULL) == factory);
+  g_object_unref (factory);
+  fail_unless (gst_rtsp_mount_points_match (mounts, url2->abspath,
+          NULL) == NULL);
+
+  gst_rtsp_mount_points_remove_factory (mounts, "/test");
+
+  fail_unless (gst_rtsp_mount_points_match (mounts, url->abspath,
+          NULL) == NULL);
+  fail_unless (gst_rtsp_mount_points_match (mounts, url2->abspath,
+          NULL) == NULL);
+
+  gst_rtsp_url_free (url);
+  gst_rtsp_url_free (url2);
+
+  g_object_unref (mounts);
+}
+
+GST_END_TEST;
+
+static const gchar *paths[] = {
+  "/test",
+  "/booz/fooz",
+  "/booz/foo/zoop",
+  "/tark/bar",
+  "/tark/bar/baz",
+  "/tark/bar/baz/t",
+  "/boozop",
+};
+
+GST_START_TEST (test_match)
+{
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactory *f[G_N_ELEMENTS (paths)], *tmp;
+  gint i, matched;
+
+  mounts = gst_rtsp_mount_points_new ();
+
+  for (i = 0; i < G_N_ELEMENTS (paths); i++) {
+    f[i] = gst_rtsp_media_factory_new ();
+    gst_rtsp_mount_points_add_factory (mounts, paths[i], f[i]);
+  }
+
+  tmp = gst_rtsp_mount_points_match (mounts, "/test", &matched);
+  fail_unless (tmp == f[0]);
+  fail_unless (matched == 5);
+  g_object_unref (tmp);
+  tmp = gst_rtsp_mount_points_match (mounts, "/test/stream=1", &matched);
+  fail_unless (tmp == f[0]);
+  fail_unless (matched == 5);
+  g_object_unref (tmp);
+  tmp = gst_rtsp_mount_points_match (mounts, "/booz", &matched);
+  fail_unless (tmp == NULL);
+  tmp = gst_rtsp_mount_points_match (mounts, "/booz/foo", &matched);
+  fail_unless (tmp == NULL);
+  tmp = gst_rtsp_mount_points_match (mounts, "/booz/fooz", &matched);
+  fail_unless (tmp == f[1]);
+  fail_unless (matched == 10);
+  g_object_unref (tmp);
+  tmp = gst_rtsp_mount_points_match (mounts, "/booz/fooz/zoo", &matched);
+  fail_unless (tmp == f[1]);
+  fail_unless (matched == 10);
+  g_object_unref (tmp);
+  tmp = gst_rtsp_mount_points_match (mounts, "/booz/foo/zoop", &matched);
+  fail_unless (tmp == f[2]);
+  fail_unless (matched == 14);
+  g_object_unref (tmp);
+  tmp = gst_rtsp_mount_points_match (mounts, "/tark/bar", &matched);
+  fail_unless (tmp == f[3]);
+  fail_unless (matched == 9);
+  g_object_unref (tmp);
+  tmp = gst_rtsp_mount_points_match (mounts, "/tark/bar/boo", &matched);
+  fail_unless (tmp == f[3]);
+  fail_unless (matched == 9);
+  g_object_unref (tmp);
+  tmp = gst_rtsp_mount_points_match (mounts, "/tark/bar/ba", &matched);
+  fail_unless (tmp == f[3]);
+  fail_unless (matched == 9);
+  g_object_unref (tmp);
+  tmp = gst_rtsp_mount_points_match (mounts, "/tark/bar/baz", &matched);
+  fail_unless (tmp == f[4]);
+  fail_unless (matched == 13);
+  g_object_unref (tmp);
+
+  g_object_unref (mounts);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspmountpoints_suite (void)
+{
+  Suite *s = suite_create ("rtspmountpoints");
+  TCase *tc = tcase_create ("general");
+
+  suite_add_tcase (s, tc);
+  tcase_set_timeout (tc, 20);
+  tcase_add_test (tc, test_create);
+  tcase_add_test (tc, test_match);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtspmountpoints);
diff --git a/tests/check/gst/permissions.c b/tests/check/gst/permissions.c
new file mode 100644
index 0000000..0dcf770
--- /dev/null
+++ b/tests/check/gst/permissions.c
@@ -0,0 +1,107 @@
+/* GStreamer
+ * Copyright (C) 2013 Sebastian Rasmussen 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+GST_START_TEST (test_permissions)
+{
+  GstRTSPPermissions *perms;
+  GstRTSPPermissions *copy;
+
+  perms = gst_rtsp_permissions_new ();
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "missing", "permission1"));
+  gst_rtsp_permissions_unref (perms);
+
+  perms = gst_rtsp_permissions_new ();
+  gst_rtsp_permissions_add_role (perms, "user",
+      "permission1", G_TYPE_BOOLEAN, TRUE,
+      "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+  fail_unless (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "missing"));
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "missing", "permission1"));
+  copy = GST_RTSP_PERMISSIONS (gst_mini_object_copy (GST_MINI_OBJECT (perms)));
+  gst_rtsp_permissions_unref (perms);
+  fail_unless (gst_rtsp_permissions_is_allowed (copy, "user", "permission1"));
+  fail_if (gst_rtsp_permissions_is_allowed (copy, "user", "permission2"));
+  gst_rtsp_permissions_unref (copy);
+
+  perms = gst_rtsp_permissions_new ();
+  gst_rtsp_permissions_add_role (perms, "admin",
+      "permission1", G_TYPE_BOOLEAN, TRUE,
+      "permission2", G_TYPE_BOOLEAN, TRUE, NULL);
+  gst_rtsp_permissions_add_role (perms, "user",
+      "permission1", G_TYPE_BOOLEAN, TRUE,
+      "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+  fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission1"));
+  fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission2"));
+  fail_unless (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+  gst_rtsp_permissions_unref (perms);
+
+  perms = gst_rtsp_permissions_new ();
+  gst_rtsp_permissions_add_role (perms, "user",
+      "permission1", G_TYPE_BOOLEAN, TRUE,
+      "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+  fail_unless (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+  gst_rtsp_permissions_add_role (perms, "user",
+      "permission1", G_TYPE_BOOLEAN, FALSE,
+      "permission2", G_TYPE_BOOLEAN, TRUE, NULL);
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+  fail_unless (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+  gst_rtsp_permissions_unref (perms);
+
+  perms = gst_rtsp_permissions_new ();
+  gst_rtsp_permissions_add_role (perms, "admin",
+      "permission1", G_TYPE_BOOLEAN, TRUE,
+      "permission2", G_TYPE_BOOLEAN, TRUE, NULL);
+  gst_rtsp_permissions_add_role (perms, "user",
+      "permission1", G_TYPE_BOOLEAN, TRUE,
+      "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+  fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission1"));
+  fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission2"));
+  fail_unless (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+  gst_rtsp_permissions_remove_role (perms, "user");
+  fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission1"));
+  fail_unless (gst_rtsp_permissions_is_allowed (perms, "admin", "permission2"));
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission1"));
+  fail_if (gst_rtsp_permissions_is_allowed (perms, "user", "permission2"));
+  gst_rtsp_permissions_unref (perms);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtsppermissions_suite (void)
+{
+  Suite *s = suite_create ("rtsppermissions");
+  TCase *tc = tcase_create ("general");
+
+  suite_add_tcase (s, tc);
+  tcase_set_timeout (tc, 20);
+  tcase_add_test (tc, test_permissions);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtsppermissions);
diff --git a/tests/check/gst/rtspserver.c b/tests/check/gst/rtspserver.c
new file mode 100644
index 0000000..28910f7
--- /dev/null
+++ b/tests/check/gst/rtspserver.c
@@ -0,0 +1,1454 @@
+/* GStreamer
+ *
+ * unit test for GstRTSPServer
+ *
+ * Copyright (C) 2012 Axis Communications 
+ * @author David Svensson Fors 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+#include 
+#include 
+#include 
+
+#include 
+#include 
+
+#include "rtsp-server.h"
+
+#define VIDEO_PIPELINE "videotestsrc ! " \
+  "video/x-raw,width=352,height=288 ! " \
+  "rtpgstpay name=pay0 pt=96"
+#define AUDIO_PIPELINE "audiotestsrc ! " \
+  "audio/x-raw,rate=8000 ! " \
+  "rtpgstpay name=pay1 pt=97"
+
+#define TEST_MOUNT_POINT  "/test"
+#define TEST_PROTO        "RTP/AVP"
+#define TEST_ENCODING     "X-GST"
+#define TEST_CLOCK_RATE   "90000"
+
+/* tested rtsp server */
+static GstRTSPServer *server = NULL;
+
+/* tcp port that the test server listens for rtsp requests on */
+static gint test_port = 0;
+
+/* id of the server's source within the GMainContext */
+static guint source_id;
+
+/* iterate the default main loop until there are no events to dispatch */
+static void
+iterate (void)
+{
+  while (g_main_context_iteration (NULL, FALSE)) {
+    GST_DEBUG ("iteration");
+  }
+}
+
+static void
+get_client_ports_full (GstRTSPRange * range, GSocket ** rtp_socket,
+    GSocket ** rtcp_socket)
+{
+  GSocket *rtp = NULL;
+  GSocket *rtcp = NULL;
+  gint rtp_port = 0;
+  gint rtcp_port;
+  GInetAddress *anyaddr = g_inet_address_new_any (G_SOCKET_FAMILY_IPV4);
+  GSocketAddress *sockaddr;
+  gboolean bound;
+
+  for (;;) {
+    if (rtp_port != 0)
+      rtp_port += 2;
+
+    rtp = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_DATAGRAM,
+        G_SOCKET_PROTOCOL_UDP, NULL);
+    fail_unless (rtp != NULL);
+
+    sockaddr = g_inet_socket_address_new (anyaddr, rtp_port);
+    fail_unless (sockaddr != NULL);
+    bound = g_socket_bind (rtp, sockaddr, FALSE, NULL);
+    g_object_unref (sockaddr);
+    if (!bound) {
+      g_object_unref (rtp);
+      continue;
+    }
+
+    sockaddr = g_socket_get_local_address (rtp, NULL);
+    fail_unless (sockaddr != NULL && G_IS_INET_SOCKET_ADDRESS (sockaddr));
+    rtp_port =
+        g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr));
+    g_object_unref (sockaddr);
+
+    if (rtp_port % 2 != 0) {
+      rtp_port += 1;
+      g_object_unref (rtp);
+      continue;
+    }
+
+    rtcp_port = rtp_port + 1;
+
+    rtcp = g_socket_new (G_SOCKET_FAMILY_IPV4, G_SOCKET_TYPE_DATAGRAM,
+        G_SOCKET_PROTOCOL_UDP, NULL);
+    fail_unless (rtcp != NULL);
+
+    sockaddr = g_inet_socket_address_new (anyaddr, rtcp_port);
+    fail_unless (sockaddr != NULL);
+    bound = g_socket_bind (rtcp, sockaddr, FALSE, NULL);
+    g_object_unref (sockaddr);
+    if (!bound) {
+      g_object_unref (rtp);
+      g_object_unref (rtcp);
+      continue;
+    }
+
+    sockaddr = g_socket_get_local_address (rtcp, NULL);
+    fail_unless (sockaddr != NULL && G_IS_INET_SOCKET_ADDRESS (sockaddr));
+    fail_unless (rtcp_port ==
+        g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (sockaddr)));
+    g_object_unref (sockaddr);
+
+    break;
+  }
+
+  range->min = rtp_port;
+  range->max = rtcp_port;
+  if (rtp_socket)
+    *rtp_socket = rtp;
+  else
+    g_object_unref (rtp);
+  if (rtcp_socket)
+    *rtcp_socket = rtcp;
+  else
+    g_object_unref (rtcp);
+  GST_DEBUG ("client_port=%d-%d", range->min, range->max);
+  g_object_unref (anyaddr);
+}
+
+/* get a free rtp/rtcp client port pair */
+static void
+get_client_ports (GstRTSPRange * range)
+{
+  get_client_ports_full (range, NULL, NULL);
+}
+
+/* start the tested rtsp server */
+static void
+start_server (void)
+{
+  GstRTSPMountPoints *mounts;
+  gchar *service;
+  GstRTSPMediaFactory *factory;
+
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  factory = gst_rtsp_media_factory_new ();
+
+  gst_rtsp_media_factory_set_launch (factory,
+      "( " VIDEO_PIPELINE "  " AUDIO_PIPELINE " )");
+  gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
+  g_object_unref (mounts);
+
+  /* set port to any */
+  gst_rtsp_server_set_service (server, "0");
+
+  /* attach to default main context */
+  source_id = gst_rtsp_server_attach (server, NULL);
+  fail_if (source_id == 0);
+
+  /* get port */
+  service = gst_rtsp_server_get_service (server);
+  test_port = atoi (service);
+  fail_unless (test_port != 0);
+  g_free (service);
+
+  GST_DEBUG ("rtsp server listening on port %d", test_port);
+}
+
+/* stop the tested rtsp server */
+static void
+stop_server (void)
+{
+  g_source_remove (source_id);
+  source_id = 0;
+
+  GST_DEBUG ("rtsp server stopped");
+}
+
+/* create an rtsp connection to the server on test_port */
+static GstRTSPConnection *
+connect_to_server (gint port, const gchar * mount_point)
+{
+  GstRTSPConnection *conn = NULL;
+  gchar *address;
+  gchar *uri_string;
+  GstRTSPUrl *url = NULL;
+
+  address = gst_rtsp_server_get_address (server);
+  uri_string = g_strdup_printf ("rtsp://%s:%d%s", address, port, mount_point);
+  g_free (address);
+  fail_unless (gst_rtsp_url_parse (uri_string, &url) == GST_RTSP_OK);
+  g_free (uri_string);
+
+  fail_unless (gst_rtsp_connection_create (url, &conn) == GST_RTSP_OK);
+  gst_rtsp_url_free (url);
+
+  fail_unless (gst_rtsp_connection_connect (conn, NULL) == GST_RTSP_OK);
+
+  return conn;
+}
+
+/* create an rtsp request */
+static GstRTSPMessage *
+create_request (GstRTSPConnection * conn, GstRTSPMethod method,
+    const gchar * control)
+{
+  GstRTSPMessage *request = NULL;
+  gchar *base_uri;
+  gchar *full_uri;
+
+  base_uri = gst_rtsp_url_get_request_uri (gst_rtsp_connection_get_url (conn));
+  full_uri = g_strdup_printf ("%s/%s", base_uri, control ? control : "");
+  g_free (base_uri);
+  if (gst_rtsp_message_new_request (&request, method, full_uri) != GST_RTSP_OK) {
+    GST_DEBUG ("failed to create request object");
+    g_free (full_uri);
+    return NULL;
+  }
+  g_free (full_uri);
+  return request;
+}
+
+/* send an rtsp request */
+static gboolean
+send_request (GstRTSPConnection * conn, GstRTSPMessage * request)
+{
+  if (gst_rtsp_connection_send (conn, request, NULL) != GST_RTSP_OK) {
+    GST_DEBUG ("failed to send request");
+    return FALSE;
+  }
+  return TRUE;
+}
+
+/* read rtsp response. response must be freed by the caller */
+static GstRTSPMessage *
+read_response (GstRTSPConnection * conn)
+{
+  GstRTSPMessage *response = NULL;
+
+  if (gst_rtsp_message_new (&response) != GST_RTSP_OK) {
+    GST_DEBUG ("failed to create response object");
+    return NULL;
+  }
+  if (gst_rtsp_connection_receive (conn, response, NULL) != GST_RTSP_OK) {
+    GST_DEBUG ("failed to read response");
+    gst_rtsp_message_free (response);
+    return NULL;
+  }
+  fail_unless (gst_rtsp_message_get_type (response) ==
+      GST_RTSP_MESSAGE_RESPONSE);
+  return response;
+}
+
+/* send an rtsp request and receive response. gchar** parameters are out
+ * parameters that have to be freed by the caller */
+static GstRTSPStatusCode
+do_request_full (GstRTSPConnection * conn, GstRTSPMethod method,
+    const gchar * control, const gchar * session_in, const gchar * transport_in,
+    const gchar * range_in, const gchar * require_in,
+    gchar ** content_type, gchar ** content_base, gchar ** body,
+    gchar ** session_out, gchar ** transport_out, gchar ** range_out,
+    gchar ** unsupported_out)
+{
+  GstRTSPMessage *request;
+  GstRTSPMessage *response;
+  GstRTSPStatusCode code;
+  gchar *value;
+
+  /* create request */
+  request = create_request (conn, method, control);
+
+  /* add headers */
+  if (session_in) {
+    gst_rtsp_message_add_header (request, GST_RTSP_HDR_SESSION, session_in);
+  }
+  if (transport_in) {
+    gst_rtsp_message_add_header (request, GST_RTSP_HDR_TRANSPORT, transport_in);
+  }
+  if (range_in) {
+    gst_rtsp_message_add_header (request, GST_RTSP_HDR_RANGE, range_in);
+  }
+  if (require_in) {
+    gst_rtsp_message_add_header (request, GST_RTSP_HDR_REQUIRE, require_in);
+  }
+
+  /* send request */
+  fail_unless (send_request (conn, request));
+  gst_rtsp_message_free (request);
+
+  iterate ();
+
+  /* read response */
+  response = read_response (conn);
+
+  /* check status line */
+  gst_rtsp_message_parse_response (response, &code, NULL, NULL);
+  if (code != GST_RTSP_STS_OK) {
+    if (unsupported_out != NULL && code == GST_RTSP_STS_OPTION_NOT_SUPPORTED) {
+      gst_rtsp_message_get_header (response, GST_RTSP_HDR_UNSUPPORTED,
+          &value, 0);
+      *unsupported_out = g_strdup (value);
+    }
+    gst_rtsp_message_free (response);
+    return code;
+  }
+
+  /* get information from response */
+  if (content_type) {
+    gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_TYPE,
+        &value, 0);
+    *content_type = g_strdup (value);
+  }
+  if (content_base) {
+    gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
+        &value, 0);
+    *content_base = g_strdup (value);
+  }
+  if (body) {
+    *body = g_malloc (response->body_size + 1);
+    strncpy (*body, (gchar *) response->body, response->body_size);
+  }
+  if (session_out) {
+    gst_rtsp_message_get_header (response, GST_RTSP_HDR_SESSION, &value, 0);
+
+    value = g_strdup (value);
+
+    /* Remove the timeout */
+    if (value) {
+      char *pos = strchr (value, ';');
+      if (pos)
+        *pos = 0;
+    }
+    if (session_in) {
+      /* check that we got the same session back */
+      fail_unless (!g_strcmp0 (value, session_in));
+    }
+    *session_out = value;
+  }
+  if (transport_out) {
+    gst_rtsp_message_get_header (response, GST_RTSP_HDR_TRANSPORT, &value, 0);
+    *transport_out = g_strdup (value);
+  }
+  if (range_out) {
+    gst_rtsp_message_get_header (response, GST_RTSP_HDR_RANGE, &value, 0);
+    *range_out = g_strdup (value);
+  }
+
+  gst_rtsp_message_free (response);
+  return code;
+}
+
+/* send an rtsp request and receive response. gchar** parameters are out
+ * parameters that have to be freed by the caller */
+static GstRTSPStatusCode
+do_request (GstRTSPConnection * conn, GstRTSPMethod method,
+    const gchar * control, const gchar * session_in,
+    const gchar * transport_in, const gchar * range_in,
+    gchar ** content_type, gchar ** content_base, gchar ** body,
+    gchar ** session_out, gchar ** transport_out, gchar ** range_out)
+{
+  return do_request_full (conn, method, control, session_in, transport_in,
+      range_in, NULL, content_type, content_base, body, session_out,
+      transport_out, range_out, NULL);
+}
+
+/* send an rtsp request with a method and a session, and receive response */
+static GstRTSPStatusCode
+do_simple_request (GstRTSPConnection * conn, GstRTSPMethod method,
+    const gchar * session)
+{
+  return do_request (conn, method, NULL, session, NULL, NULL, NULL,
+      NULL, NULL, NULL, NULL, NULL);
+}
+
+/* send a DESCRIBE request and receive response. returns a received
+ * GstSDPMessage that must be freed by the caller */
+static GstSDPMessage *
+do_describe (GstRTSPConnection * conn, const gchar * mount_point)
+{
+  GstSDPMessage *sdp_message;
+  gchar *content_type;
+  gchar *content_base;
+  gchar *body;
+  gchar *address;
+  gchar *expected_content_base;
+
+  /* send DESCRIBE request */
+  fail_unless (do_request (conn, GST_RTSP_DESCRIBE, NULL, NULL, NULL, NULL,
+          &content_type, &content_base, &body, NULL, NULL, NULL) ==
+      GST_RTSP_STS_OK);
+
+  /* check response values */
+  fail_unless (!g_strcmp0 (content_type, "application/sdp"));
+  address = gst_rtsp_server_get_address (server);
+  expected_content_base =
+      g_strdup_printf ("rtsp://%s:%d%s/", address, test_port, mount_point);
+  fail_unless (!g_strcmp0 (content_base, expected_content_base));
+
+  /* create sdp message */
+  fail_unless (gst_sdp_message_new (&sdp_message) == GST_SDP_OK);
+  fail_unless (gst_sdp_message_parse_buffer ((guint8 *) body,
+          strlen (body), sdp_message) == GST_SDP_OK);
+
+  /* clean up */
+  g_free (content_type);
+  g_free (content_base);
+  g_free (body);
+  g_free (address);
+  g_free (expected_content_base);
+
+  return sdp_message;
+}
+
+/* send a SETUP request and receive response. if *session is not NULL,
+ * it is used in the request. otherwise, *session is set to a returned
+ * session string that must be freed by the caller. the returned
+ * transport must be freed by the caller. */
+static GstRTSPStatusCode
+do_setup_full (GstRTSPConnection * conn, const gchar * control,
+    const GstRTSPRange * client_ports, const gchar * require, gchar ** session,
+    GstRTSPTransport ** transport, gchar ** unsupported)
+{
+  GstRTSPStatusCode code;
+  gchar *session_in = NULL;
+  gchar *transport_string_in = NULL;
+  gchar **session_out = NULL;
+  gchar *transport_string_out = NULL;
+
+  /* prepare and send SETUP request */
+  if (session) {
+    if (*session) {
+      session_in = *session;
+    } else {
+      session_out = session;
+    }
+  }
+  transport_string_in =
+      g_strdup_printf (TEST_PROTO ";unicast;client_port=%d-%d",
+      client_ports->min, client_ports->max);
+  code =
+      do_request_full (conn, GST_RTSP_SETUP, control, session_in,
+      transport_string_in, NULL, require, NULL, NULL, NULL, session_out,
+      &transport_string_out, NULL, unsupported);
+  g_free (transport_string_in);
+
+  if (transport_string_out) {
+    /* create transport */
+    fail_unless (gst_rtsp_transport_new (transport) == GST_RTSP_OK);
+    fail_unless (gst_rtsp_transport_parse (transport_string_out,
+            *transport) == GST_RTSP_OK);
+    g_free (transport_string_out);
+  }
+  GST_INFO ("code=%d", code);
+  return code;
+}
+
+/* send a SETUP request and receive response. if *session is not NULL,
+ * it is used in the request. otherwise, *session is set to a returned
+ * session string that must be freed by the caller. the returned
+ * transport must be freed by the caller. */
+static GstRTSPStatusCode
+do_setup (GstRTSPConnection * conn, const gchar * control,
+    const GstRTSPRange * client_ports, gchar ** session,
+    GstRTSPTransport ** transport)
+{
+  return do_setup_full (conn, control, client_ports, NULL, session, transport,
+      NULL);
+}
+
+/* fixture setup function */
+static void
+setup (void)
+{
+  server = gst_rtsp_server_new ();
+}
+
+/* fixture clean-up function */
+static void
+teardown (void)
+{
+  if (server) {
+    g_object_unref (server);
+    server = NULL;
+  }
+  test_port = 0;
+}
+
+GST_START_TEST (test_connect)
+{
+  GstRTSPConnection *conn;
+
+  start_server ();
+
+  /* connect to server */
+  conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+  /* clean up */
+  gst_rtsp_connection_free (conn);
+  stop_server ();
+
+  /* iterate so the clean-up can finish */
+  iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_describe)
+{
+  GstRTSPConnection *conn;
+  GstSDPMessage *sdp_message = NULL;
+  const GstSDPMedia *sdp_media;
+  gint32 format;
+  gchar *expected_rtpmap;
+  const gchar *rtpmap;
+  const gchar *control_video;
+  const gchar *control_audio;
+
+  start_server ();
+
+  conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+  /* send DESCRIBE request */
+  sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+  fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+
+  /* check video sdp */
+  sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+  fail_unless (!g_strcmp0 (gst_sdp_media_get_proto (sdp_media), TEST_PROTO));
+  fail_unless (gst_sdp_media_formats_len (sdp_media) == 1);
+  sscanf (gst_sdp_media_get_format (sdp_media, 0), "%" G_GINT32_FORMAT,
+      &format);
+  expected_rtpmap =
+      g_strdup_printf ("%d " TEST_ENCODING "/" TEST_CLOCK_RATE, format);
+  rtpmap = gst_sdp_media_get_attribute_val (sdp_media, "rtpmap");
+  fail_unless (!g_strcmp0 (rtpmap, expected_rtpmap));
+  g_free (expected_rtpmap);
+  control_video = gst_sdp_media_get_attribute_val (sdp_media, "control");
+  fail_unless (!g_strcmp0 (control_video, "stream=0"));
+
+  /* check audio sdp */
+  sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+  fail_unless (!g_strcmp0 (gst_sdp_media_get_proto (sdp_media), TEST_PROTO));
+  fail_unless (gst_sdp_media_formats_len (sdp_media) == 1);
+  sscanf (gst_sdp_media_get_format (sdp_media, 0), "%" G_GINT32_FORMAT,
+      &format);
+  expected_rtpmap =
+      g_strdup_printf ("%d " TEST_ENCODING "/" TEST_CLOCK_RATE, format);
+  rtpmap = gst_sdp_media_get_attribute_val (sdp_media, "rtpmap");
+  fail_unless (!g_strcmp0 (rtpmap, expected_rtpmap));
+  g_free (expected_rtpmap);
+  control_audio = gst_sdp_media_get_attribute_val (sdp_media, "control");
+  fail_unless (!g_strcmp0 (control_audio, "stream=1"));
+
+  /* clean up and iterate so the clean-up can finish */
+  gst_sdp_message_free (sdp_message);
+  gst_rtsp_connection_free (conn);
+  stop_server ();
+  iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_describe_non_existing_mount_point)
+{
+  GstRTSPConnection *conn;
+
+  start_server ();
+
+  /* send DESCRIBE request for a non-existing mount point
+   * and check that we get a 404 Not Found */
+  conn = connect_to_server (test_port, "/non-existing");
+  fail_unless (do_simple_request (conn, GST_RTSP_DESCRIBE, NULL)
+      == GST_RTSP_STS_NOT_FOUND);
+
+  /* clean up and iterate so the clean-up can finish */
+  gst_rtsp_connection_free (conn);
+  stop_server ();
+  iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_setup)
+{
+  GstRTSPConnection *conn;
+  GstSDPMessage *sdp_message = NULL;
+  const GstSDPMedia *sdp_media;
+  const gchar *video_control;
+  const gchar *audio_control;
+  GstRTSPRange client_ports;
+  gchar *session = NULL;
+  GstRTSPTransport *video_transport = NULL;
+  GstRTSPTransport *audio_transport = NULL;
+
+  start_server ();
+
+  conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+  sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+  /* get control strings from DESCRIBE response */
+  fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+  sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+  video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+  sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+  audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+  get_client_ports (&client_ports);
+
+  /* send SETUP request for video */
+  fail_unless (do_setup (conn, video_control, &client_ports, &session,
+          &video_transport) == GST_RTSP_STS_OK);
+  GST_DEBUG ("set up video %s, got session '%s'", video_control, session);
+
+  /* check response from SETUP */
+  fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
+  fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
+  fail_unless (video_transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP);
+  fail_unless (video_transport->mode_play);
+  gst_rtsp_transport_free (video_transport);
+
+  /* send SETUP request for audio */
+  fail_unless (do_setup (conn, audio_control, &client_ports, &session,
+          &audio_transport) == GST_RTSP_STS_OK);
+  GST_DEBUG ("set up audio %s with session '%s'", audio_control, session);
+
+  /* check response from SETUP */
+  fail_unless (audio_transport->trans == GST_RTSP_TRANS_RTP);
+  fail_unless (audio_transport->profile == GST_RTSP_PROFILE_AVP);
+  fail_unless (audio_transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP);
+  fail_unless (audio_transport->mode_play);
+  gst_rtsp_transport_free (audio_transport);
+
+  /* send TEARDOWN request and check that we get 200 OK */
+  fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+          session) == GST_RTSP_STS_OK);
+
+  /* clean up and iterate so the clean-up can finish */
+  g_free (session);
+  gst_sdp_message_free (sdp_message);
+  gst_rtsp_connection_free (conn);
+  stop_server ();
+  iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_setup_with_require_header)
+{
+  GstRTSPConnection *conn;
+  GstSDPMessage *sdp_message = NULL;
+  const GstSDPMedia *sdp_media;
+  const gchar *video_control;
+  GstRTSPRange client_ports;
+  gchar *session = NULL;
+  gchar *unsupported = NULL;
+  GstRTSPTransport *video_transport = NULL;
+
+  start_server ();
+
+  conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+  sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+  /* get control strings from DESCRIBE response */
+  fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+  sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+  video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+  get_client_ports (&client_ports);
+
+  /* send SETUP request for video, with single Require header */
+  fail_unless_equals_int (do_setup_full (conn, video_control, &client_ports,
+          "funky-feature", &session, &video_transport, &unsupported),
+      GST_RTSP_STS_OPTION_NOT_SUPPORTED);
+  fail_unless_equals_string (unsupported, "funky-feature");
+  g_free (unsupported);
+  unsupported = NULL;
+
+  /* send SETUP request for video, with multiple Require headers */
+  fail_unless_equals_int (do_setup_full (conn, video_control, &client_ports,
+          "funky-feature, foo-bar, superburst", &session, &video_transport,
+          &unsupported), GST_RTSP_STS_OPTION_NOT_SUPPORTED);
+  fail_unless_equals_string (unsupported, "funky-feature, foo-bar, superburst");
+  g_free (unsupported);
+  unsupported = NULL;
+
+  /* ok, just do a normal setup then (make sure that still works) */
+  fail_unless_equals_int (do_setup (conn, video_control, &client_ports,
+          &session, &video_transport), GST_RTSP_STS_OK);
+
+  GST_DEBUG ("set up video %s, got session '%s'", video_control, session);
+
+  /* check response from SETUP */
+  fail_unless (video_transport->trans == GST_RTSP_TRANS_RTP);
+  fail_unless (video_transport->profile == GST_RTSP_PROFILE_AVP);
+  fail_unless (video_transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP);
+  fail_unless (video_transport->mode_play);
+  gst_rtsp_transport_free (video_transport);
+
+  /* send TEARDOWN request and check that we get 200 OK */
+  fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+          session) == GST_RTSP_STS_OK);
+
+  /* clean up and iterate so the clean-up can finish */
+  g_free (session);
+  gst_sdp_message_free (sdp_message);
+  gst_rtsp_connection_free (conn);
+  stop_server ();
+  iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_setup_non_existing_stream)
+{
+  GstRTSPConnection *conn;
+  GstRTSPRange client_ports;
+
+  start_server ();
+
+  conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+  get_client_ports (&client_ports);
+
+  /* send SETUP request with a non-existing stream and check that we get a
+   * 404 Not Found */
+  fail_unless (do_setup (conn, "stream=7", &client_ports, NULL,
+          NULL) == GST_RTSP_STS_NOT_FOUND);
+
+  /* clean up and iterate so the clean-up can finish */
+  gst_rtsp_connection_free (conn);
+  stop_server ();
+  iterate ();
+}
+
+GST_END_TEST;
+
+static void
+receive_rtp (GSocket * socket, GSocketAddress ** addr)
+{
+  GstBuffer *buffer = gst_buffer_new_allocate (NULL, 65536, NULL);
+
+  for (;;) {
+    gssize bytes;
+    GstMapInfo map = GST_MAP_INFO_INIT;
+    GstRTPBuffer rtpbuffer = GST_RTP_BUFFER_INIT;
+
+    gst_buffer_map (buffer, &map, GST_MAP_WRITE);
+    bytes = g_socket_receive_from (socket, addr, (gchar *) map.data,
+        map.maxsize, NULL, NULL);
+    fail_unless (bytes > 0);
+    gst_buffer_unmap (buffer, &map);
+    gst_buffer_set_size (buffer, bytes);
+
+    if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpbuffer)) {
+      gst_rtp_buffer_unmap (&rtpbuffer);
+      break;
+    }
+
+    if (addr)
+      g_clear_object (addr);
+  }
+
+  gst_buffer_unref (buffer);
+}
+
+static void
+receive_rtcp (GSocket * socket, GSocketAddress ** addr, GstRTCPType type)
+{
+  GstBuffer *buffer = gst_buffer_new_allocate (NULL, 65536, NULL);
+
+  for (;;) {
+    gssize bytes;
+    GstMapInfo map = GST_MAP_INFO_INIT;
+
+    gst_buffer_map (buffer, &map, GST_MAP_WRITE);
+    bytes = g_socket_receive_from (socket, addr, (gchar *) map.data,
+        map.maxsize, NULL, NULL);
+    fail_unless (bytes > 0);
+    gst_buffer_unmap (buffer, &map);
+    gst_buffer_set_size (buffer, bytes);
+
+    if (gst_rtcp_buffer_validate (buffer)) {
+      GstRTCPBuffer rtcpbuffer = GST_RTCP_BUFFER_INIT;
+      GstRTCPPacket packet;
+
+      if (type) {
+        fail_unless (gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcpbuffer));
+        fail_unless (gst_rtcp_buffer_get_first_packet (&rtcpbuffer, &packet));
+        do {
+          if (gst_rtcp_packet_get_type (&packet) == type) {
+            gst_rtcp_buffer_unmap (&rtcpbuffer);
+            goto done;
+          }
+        } while (gst_rtcp_packet_move_to_next (&packet));
+        gst_rtcp_buffer_unmap (&rtcpbuffer);
+      } else {
+        break;
+      }
+    }
+
+    if (addr)
+      g_clear_object (addr);
+  }
+
+done:
+
+  gst_buffer_unref (buffer);
+}
+
+static void
+do_test_play (const gchar * range)
+{
+  GstRTSPConnection *conn;
+  GstSDPMessage *sdp_message = NULL;
+  const GstSDPMedia *sdp_media;
+  const gchar *video_control;
+  const gchar *audio_control;
+  GstRTSPRange client_port;
+  gchar *session = NULL;
+  GstRTSPTransport *video_transport = NULL;
+  GstRTSPTransport *audio_transport = NULL;
+  GSocket *rtp_socket, *rtcp_socket;
+  gchar *range_out = NULL;
+
+  conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+  sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+  /* get control strings from DESCRIBE response */
+  fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+  sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+  video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+  sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+  audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+  get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
+
+  /* do SETUP for video and audio */
+  fail_unless (do_setup (conn, video_control, &client_port, &session,
+          &video_transport) == GST_RTSP_STS_OK);
+  fail_unless (do_setup (conn, audio_control, &client_port, &session,
+          &audio_transport) == GST_RTSP_STS_OK);
+
+  /* send PLAY request and check that we get 200 OK */
+  fail_unless (do_request (conn, GST_RTSP_PLAY, NULL, session, NULL, range,
+          NULL, NULL, NULL, NULL, NULL, &range_out) == GST_RTSP_STS_OK);
+  if (range)
+    fail_unless_equals_string (range, range_out);
+  g_free (range_out);
+
+  receive_rtp (rtp_socket, NULL);
+  receive_rtcp (rtcp_socket, NULL, 0);
+
+  /* send TEARDOWN request and check that we get 200 OK */
+  fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+          session) == GST_RTSP_STS_OK);
+
+  /* FIXME: The rtsp-server always disconnects the transport before
+   * sending the RTCP BYE
+   * receive_rtcp (rtcp_socket, NULL, GST_RTCP_TYPE_BYE);
+   */
+
+  /* clean up and iterate so the clean-up can finish */
+  g_object_unref (rtp_socket);
+  g_object_unref (rtcp_socket);
+  g_free (session);
+  gst_rtsp_transport_free (video_transport);
+  gst_rtsp_transport_free (audio_transport);
+  gst_sdp_message_free (sdp_message);
+  gst_rtsp_connection_free (conn);
+}
+
+
+GST_START_TEST (test_play)
+{
+  start_server ();
+
+  do_test_play (NULL);
+
+  stop_server ();
+  iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_play_without_session)
+{
+  GstRTSPConnection *conn;
+
+  start_server ();
+
+  conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+  /* send PLAY request without a session and check that we get a
+   * 454 Session Not Found */
+  fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+          NULL) == GST_RTSP_STS_SESSION_NOT_FOUND);
+
+  /* clean up and iterate so the clean-up can finish */
+  gst_rtsp_connection_free (conn);
+  stop_server ();
+  iterate ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_bind_already_in_use)
+{
+  GstRTSPServer *serv;
+  GSocketService *service;
+  GError *error = NULL;
+  guint16 port;
+  gchar *port_str;
+
+  serv = gst_rtsp_server_new ();
+  service = g_socket_service_new ();
+
+  /* bind service to port */
+  port =
+      g_socket_listener_add_any_inet_port (G_SOCKET_LISTENER (service), NULL,
+      &error);
+  g_assert_no_error (error);
+
+  port_str = g_strdup_printf ("%d\n", port);
+
+  /* try to bind server to the same port */
+  g_object_set (serv, "service", port_str, NULL);
+  g_free (port_str);
+
+  /* attach to default main context */
+  fail_unless (gst_rtsp_server_attach (serv, NULL) == 0);
+
+  /* cleanup */
+  g_object_unref (serv);
+  g_socket_service_stop (service);
+  g_object_unref (service);
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_play_multithreaded)
+{
+  GstRTSPThreadPool *pool;
+
+  pool = gst_rtsp_server_get_thread_pool (server);
+  gst_rtsp_thread_pool_set_max_threads (pool, 2);
+  g_object_unref (pool);
+
+  start_server ();
+
+  do_test_play (NULL);
+
+  stop_server ();
+  iterate ();
+}
+
+GST_END_TEST;
+
+enum
+{
+  BLOCK_ME,
+  BLOCKED,
+  UNBLOCK
+};
+
+
+static void
+media_constructed_block (GstRTSPMediaFactory * factory,
+    GstRTSPMedia * media, gpointer user_data)
+{
+  gint *block_state = user_data;
+
+  g_mutex_lock (&check_mutex);
+
+  *block_state = BLOCKED;
+  g_cond_broadcast (&check_cond);
+
+  while (*block_state != UNBLOCK)
+    g_cond_wait (&check_cond, &check_mutex);
+  g_mutex_unlock (&check_mutex);
+}
+
+
+GST_START_TEST (test_play_multithreaded_block_in_describe)
+{
+  GstRTSPConnection *conn;
+  GstRTSPMountPoints *mounts;
+  GstRTSPMediaFactory *factory;
+  gint block_state = BLOCK_ME;
+  GstRTSPMessage *request;
+  GstRTSPMessage *response;
+  GstRTSPStatusCode code;
+  GstRTSPThreadPool *pool;
+
+  pool = gst_rtsp_server_get_thread_pool (server);
+  gst_rtsp_thread_pool_set_max_threads (pool, 2);
+  g_object_unref (pool);
+
+  mounts = gst_rtsp_server_get_mount_points (server);
+  fail_unless (mounts != NULL);
+  factory = gst_rtsp_media_factory_new ();
+  gst_rtsp_media_factory_set_launch (factory,
+      "( " VIDEO_PIPELINE "  " AUDIO_PIPELINE " )");
+  g_signal_connect (factory, "media-constructed",
+      G_CALLBACK (media_constructed_block), &block_state);
+  gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT "2", factory);
+  g_object_unref (mounts);
+
+  start_server ();
+
+  conn = connect_to_server (test_port, TEST_MOUNT_POINT "2");
+  iterate ();
+
+  /* do describe, it will not return now as we've blocked it */
+  request = create_request (conn, GST_RTSP_DESCRIBE, NULL);
+  fail_unless (send_request (conn, request));
+  gst_rtsp_message_free (request);
+
+  g_mutex_lock (&check_mutex);
+  while (block_state != BLOCKED)
+    g_cond_wait (&check_cond, &check_mutex);
+  g_mutex_unlock (&check_mutex);
+
+  /* Do a second connection while the first one is blocked */
+  do_test_play (NULL);
+
+  /* Now unblock the describe */
+  g_mutex_lock (&check_mutex);
+  block_state = UNBLOCK;
+  g_cond_broadcast (&check_cond);
+  g_mutex_unlock (&check_mutex);
+
+  response = read_response (conn);
+  gst_rtsp_message_parse_response (response, &code, NULL, NULL);
+  fail_unless (code == GST_RTSP_STS_OK);
+  gst_rtsp_message_free (response);
+
+
+  gst_rtsp_connection_free (conn);
+  stop_server ();
+  iterate ();
+
+}
+
+GST_END_TEST;
+
+
+static void
+new_session_timeout_one (GstRTSPClient * client,
+    GstRTSPSession * session, gpointer user_data)
+{
+  gst_rtsp_session_set_timeout (session, 1);
+
+  g_signal_handlers_disconnect_by_func (client, new_session_timeout_one,
+      user_data);
+}
+
+static void
+session_connected_new_session_cb (GstRTSPServer * server,
+    GstRTSPClient * client, gpointer user_data)
+{
+
+  g_signal_connect (client, "new-session", user_data, NULL);
+}
+
+GST_START_TEST (test_play_multithreaded_timeout_client)
+{
+  GstRTSPConnection *conn;
+  GstSDPMessage *sdp_message = NULL;
+  const GstSDPMedia *sdp_media;
+  const gchar *video_control;
+  const gchar *audio_control;
+  GstRTSPRange client_port;
+  gchar *session = NULL;
+  GstRTSPTransport *video_transport = NULL;
+  GstRTSPTransport *audio_transport = NULL;
+  GstRTSPSessionPool *pool;
+  GstRTSPThreadPool *thread_pool;
+
+  thread_pool = gst_rtsp_server_get_thread_pool (server);
+  gst_rtsp_thread_pool_set_max_threads (thread_pool, 2);
+  g_object_unref (thread_pool);
+
+  pool = gst_rtsp_server_get_session_pool (server);
+  g_signal_connect (server, "client-connected",
+      G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
+
+  start_server ();
+
+
+  conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+  sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+  /* get control strings from DESCRIBE response */
+  fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+  sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+  video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+  sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+  audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+  get_client_ports (&client_port);
+
+  /* do SETUP for video and audio */
+  fail_unless (do_setup (conn, video_control, &client_port, &session,
+          &video_transport) == GST_RTSP_STS_OK);
+  fail_unless (do_setup (conn, audio_control, &client_port, &session,
+          &audio_transport) == GST_RTSP_STS_OK);
+
+  fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
+
+  /* send PLAY request and check that we get 200 OK */
+  fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+          session) == GST_RTSP_STS_OK);
+
+  sleep (7);
+
+  fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
+  fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 0);
+
+  /* clean up and iterate so the clean-up can finish */
+  g_object_unref (pool);
+  g_free (session);
+  gst_rtsp_transport_free (video_transport);
+  gst_rtsp_transport_free (audio_transport);
+  gst_sdp_message_free (sdp_message);
+  gst_rtsp_connection_free (conn);
+
+  stop_server ();
+  iterate ();
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_play_multithreaded_timeout_session)
+{
+  GstRTSPConnection *conn;
+  GstSDPMessage *sdp_message = NULL;
+  const GstSDPMedia *sdp_media;
+  const gchar *video_control;
+  const gchar *audio_control;
+  GstRTSPRange client_port;
+  gchar *session1 = NULL;
+  gchar *session2 = NULL;
+  GstRTSPTransport *video_transport = NULL;
+  GstRTSPTransport *audio_transport = NULL;
+  GstRTSPSessionPool *pool;
+  GstRTSPThreadPool *thread_pool;
+
+  thread_pool = gst_rtsp_server_get_thread_pool (server);
+  gst_rtsp_thread_pool_set_max_threads (thread_pool, 2);
+  g_object_unref (thread_pool);
+
+  pool = gst_rtsp_server_get_session_pool (server);
+  g_signal_connect (server, "client-connected",
+      G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
+
+  start_server ();
+
+
+  conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+  gst_rtsp_connection_set_remember_session_id (conn, FALSE);
+
+  sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+  /* get control strings from DESCRIBE response */
+  fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+  sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+  video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+  sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+  audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+  get_client_ports (&client_port);
+
+  /* do SETUP for video and audio */
+  fail_unless (do_setup (conn, video_control, &client_port, &session1,
+          &video_transport) == GST_RTSP_STS_OK);
+  fail_unless (do_setup (conn, audio_control, &client_port, &session2,
+          &audio_transport) == GST_RTSP_STS_OK);
+
+  fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 2);
+
+  /* send PLAY request and check that we get 200 OK */
+  fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+          session1) == GST_RTSP_STS_OK);
+  fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+          session2) == GST_RTSP_STS_OK);
+
+  sleep (7);
+
+  fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
+
+  /* send TEARDOWN request and check that we get 454 Session Not found */
+  fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+          session1) == GST_RTSP_STS_SESSION_NOT_FOUND);
+
+  fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+          session2) == GST_RTSP_STS_OK);
+
+  /* clean up and iterate so the clean-up can finish */
+  g_object_unref (pool);
+  g_free (session1);
+  g_free (session2);
+  gst_rtsp_transport_free (video_transport);
+  gst_rtsp_transport_free (audio_transport);
+  gst_sdp_message_free (sdp_message);
+  gst_rtsp_connection_free (conn);
+
+  stop_server ();
+  iterate ();
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_play_disconnect)
+{
+  GstRTSPConnection *conn;
+  GstSDPMessage *sdp_message = NULL;
+  const GstSDPMedia *sdp_media;
+  const gchar *video_control;
+  const gchar *audio_control;
+  GstRTSPRange client_port;
+  gchar *session = NULL;
+  GstRTSPTransport *video_transport = NULL;
+  GstRTSPTransport *audio_transport = NULL;
+  GstRTSPSessionPool *pool;
+
+  pool = gst_rtsp_server_get_session_pool (server);
+  g_signal_connect (server, "client-connected",
+      G_CALLBACK (session_connected_new_session_cb), new_session_timeout_one);
+
+  start_server ();
+
+  conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+  sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+  /* get control strings from DESCRIBE response */
+  fail_unless (gst_sdp_message_medias_len (sdp_message) == 2);
+  sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+  video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+  sdp_media = gst_sdp_message_get_media (sdp_message, 1);
+  audio_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+  get_client_ports (&client_port);
+
+  /* do SETUP for video and audio */
+  fail_unless (do_setup (conn, video_control, &client_port, &session,
+          &video_transport) == GST_RTSP_STS_OK);
+  fail_unless (do_setup (conn, audio_control, &client_port, &session,
+          &audio_transport) == GST_RTSP_STS_OK);
+
+  fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
+
+  /* send PLAY request and check that we get 200 OK */
+  fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+          session) == GST_RTSP_STS_OK);
+
+  gst_rtsp_connection_free (conn);
+
+  sleep (7);
+
+  fail_unless (gst_rtsp_session_pool_get_n_sessions (pool) == 1);
+  fail_unless (gst_rtsp_session_pool_cleanup (pool) == 1);
+
+
+  /* clean up and iterate so the clean-up can finish */
+  g_object_unref (pool);
+  g_free (session);
+  gst_rtsp_transport_free (video_transport);
+  gst_rtsp_transport_free (audio_transport);
+  gst_sdp_message_free (sdp_message);
+
+  stop_server ();
+  iterate ();
+}
+
+GST_END_TEST;
+
+/* Only different with test_play is the specific ports selected */
+
+GST_START_TEST (test_play_specific_server_port)
+{
+  GstRTSPMountPoints *mounts;
+  gchar *service;
+  GstRTSPMediaFactory *factory;
+  GstRTSPAddressPool *pool;
+  GstRTSPConnection *conn;
+  GstSDPMessage *sdp_message = NULL;
+  const GstSDPMedia *sdp_media;
+  const gchar *video_control;
+  GstRTSPRange client_port;
+  gchar *session = NULL;
+  GstRTSPTransport *video_transport = NULL;
+  GSocket *rtp_socket, *rtcp_socket;
+  GSocketAddress *rtp_address, *rtcp_address;
+  guint16 rtp_port, rtcp_port;
+
+  mounts = gst_rtsp_server_get_mount_points (server);
+
+  factory = gst_rtsp_media_factory_new ();
+  pool = gst_rtsp_address_pool_new ();
+  gst_rtsp_address_pool_add_range (pool, GST_RTSP_ADDRESS_POOL_ANY_IPV4,
+      GST_RTSP_ADDRESS_POOL_ANY_IPV4, 7770, 7780, 0);
+  gst_rtsp_media_factory_set_address_pool (factory, pool);
+  g_object_unref (pool);
+  gst_rtsp_media_factory_set_launch (factory, "( " VIDEO_PIPELINE " )");
+  gst_rtsp_mount_points_add_factory (mounts, TEST_MOUNT_POINT, factory);
+  g_object_unref (mounts);
+
+  /* set port to any */
+  gst_rtsp_server_set_service (server, "0");
+
+  /* attach to default main context */
+  source_id = gst_rtsp_server_attach (server, NULL);
+  fail_if (source_id == 0);
+
+  /* get port */
+  service = gst_rtsp_server_get_service (server);
+  test_port = atoi (service);
+  fail_unless (test_port != 0);
+  g_free (service);
+
+  GST_DEBUG ("rtsp server listening on port %d", test_port);
+
+
+  conn = connect_to_server (test_port, TEST_MOUNT_POINT);
+
+  sdp_message = do_describe (conn, TEST_MOUNT_POINT);
+
+  /* get control strings from DESCRIBE response */
+  fail_unless (gst_sdp_message_medias_len (sdp_message) == 1);
+  sdp_media = gst_sdp_message_get_media (sdp_message, 0);
+  video_control = gst_sdp_media_get_attribute_val (sdp_media, "control");
+
+  get_client_ports_full (&client_port, &rtp_socket, &rtcp_socket);
+
+  /* do SETUP for video */
+  fail_unless (do_setup (conn, video_control, &client_port, &session,
+          &video_transport) == GST_RTSP_STS_OK);
+
+  /* send PLAY request and check that we get 200 OK */
+  fail_unless (do_simple_request (conn, GST_RTSP_PLAY,
+          session) == GST_RTSP_STS_OK);
+
+  receive_rtp (rtp_socket, &rtp_address);
+  receive_rtcp (rtcp_socket, &rtcp_address, 0);
+
+  fail_unless (G_IS_INET_SOCKET_ADDRESS (rtp_address));
+  fail_unless (G_IS_INET_SOCKET_ADDRESS (rtcp_address));
+  rtp_port =
+      g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_address));
+  rtcp_port =
+      g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtcp_address));
+  fail_unless (rtp_port >= 7770 && rtp_port <= 7780 && rtp_port % 2 == 0);
+  fail_unless (rtcp_port >= 7770 && rtcp_port <= 7780 && rtcp_port % 2 == 1);
+  fail_unless (rtp_port + 1 == rtcp_port);
+
+  g_object_unref (rtp_address);
+  g_object_unref (rtcp_address);
+
+  /* send TEARDOWN request and check that we get 200 OK */
+  fail_unless (do_simple_request (conn, GST_RTSP_TEARDOWN,
+          session) == GST_RTSP_STS_OK);
+
+  /* FIXME: The rtsp-server always disconnects the transport before
+   * sending the RTCP BYE
+   * receive_rtcp (rtcp_socket, NULL, GST_RTCP_TYPE_BYE);
+   */
+
+  /* clean up and iterate so the clean-up can finish */
+  g_object_unref (rtp_socket);
+  g_object_unref (rtcp_socket);
+  g_free (session);
+  gst_rtsp_transport_free (video_transport);
+  gst_sdp_message_free (sdp_message);
+  gst_rtsp_connection_free (conn);
+
+
+  stop_server ();
+  iterate ();
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_play_smpte_range)
+{
+  start_server ();
+
+  do_test_play ("npt=5-");
+  do_test_play ("smpte=0:00:00-");
+  do_test_play ("smpte=1:00:00-");
+  do_test_play ("smpte=1:00:03-");
+  do_test_play ("clock=20120321T152256Z-");
+
+  stop_server ();
+  iterate ();
+}
+
+GST_END_TEST;
+
+
+static Suite *
+rtspserver_suite (void)
+{
+  Suite *s = suite_create ("rtspserver");
+  TCase *tc = tcase_create ("general");
+
+  suite_add_tcase (s, tc);
+  tcase_add_checked_fixture (tc, setup, teardown);
+  tcase_set_timeout (tc, 20);
+  tcase_add_test (tc, test_connect);
+  tcase_add_test (tc, test_describe);
+  tcase_add_test (tc, test_describe_non_existing_mount_point);
+  tcase_add_test (tc, test_setup);
+  tcase_add_test (tc, test_setup_with_require_header);
+  tcase_add_test (tc, test_setup_non_existing_stream);
+  tcase_add_test (tc, test_play);
+  tcase_add_test (tc, test_play_without_session);
+  tcase_add_test (tc, test_bind_already_in_use);
+  tcase_add_test (tc, test_play_multithreaded);
+  tcase_add_test (tc, test_play_multithreaded_block_in_describe);
+  tcase_add_test (tc, test_play_multithreaded_timeout_client);
+  tcase_add_test (tc, test_play_multithreaded_timeout_session);
+  tcase_add_test (tc, test_play_disconnect);
+  tcase_add_test (tc, test_play_specific_server_port);
+  tcase_add_test (tc, test_play_smpte_range);
+  return s;
+}
+
+GST_CHECK_MAIN (rtspserver);
diff --git a/tests/check/gst/sessionmedia.c b/tests/check/gst/sessionmedia.c
new file mode 100644
index 0000000..13445b4
--- /dev/null
+++ b/tests/check/gst/sessionmedia.c
@@ -0,0 +1,399 @@
+/* GStreamer
+ * Copyright (C) 2013 Branko Subasic 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+#include 
+
+#define TEST_PATH "rtsp://localhost:8554/test"
+#define SETUP_URL1 TEST_PATH "/stream=0"
+#define SETUP_URL2 TEST_PATH "/stream=1"
+
+GST_START_TEST (test_setup_url)
+{
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia *media;
+  GstRTSPUrl *url, *setup_url;
+  GstRTSPStream *stream;
+  GstRTSPThreadPool *pool;
+  GstRTSPThread *thread;
+  GstRTSPSessionMedia *sm;
+  GstRTSPStreamTransport *trans;
+  GstRTSPTransport *ct;
+  gint match_len;
+  gchar *url_str, *url_str2;
+
+  factory = gst_rtsp_media_factory_new ();
+  fail_if (gst_rtsp_media_factory_is_shared (factory));
+  fail_unless (gst_rtsp_url_parse (TEST_PATH, &url) == GST_RTSP_OK);
+
+  gst_rtsp_media_factory_set_launch (factory,
+      "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+
+  fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+  stream = gst_rtsp_media_get_stream (media, 0);
+  fail_unless (GST_IS_RTSP_STREAM (stream));
+
+  pool = gst_rtsp_thread_pool_new ();
+  thread = gst_rtsp_thread_pool_get_thread (pool,
+      GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+  fail_unless (gst_rtsp_media_prepare (media, thread));
+
+  /* create session media and make sure it matches test path
+   * note that gst_rtsp_session_media_new takes ownership of the media
+   * thus no need to unref it at the bottom of function */
+  sm = gst_rtsp_session_media_new (TEST_PATH, media);
+  fail_unless (GST_IS_RTSP_SESSION_MEDIA (sm));
+  fail_unless (gst_rtsp_session_media_matches (sm, TEST_PATH, &match_len));
+  fail_unless (match_len == strlen (TEST_PATH));
+  fail_unless (gst_rtsp_session_media_get_media (sm) == media);
+
+  /* make a transport for the stream */
+  gst_rtsp_transport_new (&ct);
+  trans = gst_rtsp_session_media_set_transport (sm, stream, ct);
+  fail_unless (gst_rtsp_session_media_get_transport (sm, 0) == trans);
+
+  /* make sure there's no setup url stored initially */
+  fail_unless (gst_rtsp_stream_transport_get_url (trans) == NULL);
+
+  /* now store a setup url and make sure it can be retrieved and that it's correct */
+  fail_unless (gst_rtsp_url_parse (SETUP_URL1, &setup_url) == GST_RTSP_OK);
+  gst_rtsp_stream_transport_set_url (trans, setup_url);
+
+  url_str = gst_rtsp_url_get_request_uri (setup_url);
+  url_str2 =
+      gst_rtsp_url_get_request_uri (gst_rtsp_stream_transport_get_url (trans));
+  fail_if (g_strcmp0 (url_str, url_str2) != 0);
+  g_free (url_str);
+  g_free (url_str2);
+
+  /* check that it's ok to try to store the same url again */
+  gst_rtsp_stream_transport_set_url (trans, setup_url);
+
+  fail_unless (gst_rtsp_media_unprepare (media));
+
+  gst_rtsp_url_free (setup_url);
+  gst_rtsp_url_free (url);
+
+  g_object_unref (sm);
+
+  g_object_unref (factory);
+  g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_rtsp_state)
+{
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia *media;
+  GstRTSPUrl *url;
+  GstRTSPStream *stream;
+  GstRTSPThreadPool *pool;
+  GstRTSPThread *thread;
+  GstRTSPSessionMedia *sm;
+
+  factory = gst_rtsp_media_factory_new ();
+  fail_if (gst_rtsp_media_factory_is_shared (factory));
+  fail_unless (gst_rtsp_url_parse (TEST_PATH, &url) == GST_RTSP_OK);
+
+  gst_rtsp_media_factory_set_launch (factory,
+      "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+
+  fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+  stream = gst_rtsp_media_get_stream (media, 0);
+  fail_unless (GST_IS_RTSP_STREAM (stream));
+
+  pool = gst_rtsp_thread_pool_new ();
+  thread = gst_rtsp_thread_pool_get_thread (pool,
+      GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+  fail_unless (gst_rtsp_media_prepare (media, thread));
+
+  sm = gst_rtsp_session_media_new (TEST_PATH, media);
+  fail_unless (GST_IS_RTSP_SESSION_MEDIA (sm));
+  fail_unless_equals_int (gst_rtsp_session_media_get_rtsp_state (sm),
+      GST_RTSP_STATE_INIT);
+
+  gst_rtsp_session_media_set_rtsp_state (sm, GST_RTSP_STATE_READY);
+  fail_unless_equals_int (gst_rtsp_session_media_get_rtsp_state (sm),
+      GST_RTSP_STATE_READY);
+
+  gst_rtsp_session_media_set_rtsp_state (sm, GST_RTSP_STATE_SEEKING);
+  fail_unless_equals_int (gst_rtsp_session_media_get_rtsp_state (sm),
+      GST_RTSP_STATE_SEEKING);
+
+  gst_rtsp_session_media_set_rtsp_state (sm, GST_RTSP_STATE_PLAYING);
+  fail_unless_equals_int (gst_rtsp_session_media_get_rtsp_state (sm),
+      GST_RTSP_STATE_PLAYING);
+
+  gst_rtsp_session_media_set_rtsp_state (sm, GST_RTSP_STATE_RECORDING);
+  fail_unless_equals_int (gst_rtsp_session_media_get_rtsp_state (sm),
+      GST_RTSP_STATE_RECORDING);
+
+  fail_unless (gst_rtsp_media_unprepare (media));
+
+  gst_rtsp_url_free (url);
+
+  g_object_unref (sm);
+
+  g_object_unref (factory);
+  g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_transports)
+{
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia *media;
+  GstRTSPUrl *url;
+  GstRTSPStream *stream1, *stream2;
+  GstRTSPThreadPool *pool;
+  GstRTSPThread *thread;
+  GstRTSPSessionMedia *sm;
+  GstRTSPStreamTransport *trans;
+  GstRTSPTransport *ct1, *ct2, *ct3, *ct4;
+  gint match_len;
+
+  factory = gst_rtsp_media_factory_new ();
+  fail_if (gst_rtsp_media_factory_is_shared (factory));
+  fail_unless (gst_rtsp_url_parse (TEST_PATH, &url) == GST_RTSP_OK);
+
+  gst_rtsp_media_factory_set_launch (factory,
+      "( videotestsrc ! rtpvrawpay pt=96 name=pay0 audiotestsrc ! rtpgstpay pt=97 name=pay1 )");
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+
+  fail_unless (gst_rtsp_media_n_streams (media) == 2);
+
+  stream1 = gst_rtsp_media_get_stream (media, 0);
+  fail_unless (GST_IS_RTSP_STREAM (stream1));
+
+  stream2 = gst_rtsp_media_get_stream (media, 1);
+  fail_unless (GST_IS_RTSP_STREAM (stream2));
+
+  pool = gst_rtsp_thread_pool_new ();
+  thread = gst_rtsp_thread_pool_get_thread (pool,
+      GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+  fail_unless (gst_rtsp_media_prepare (media, thread));
+
+  sm = gst_rtsp_session_media_new (TEST_PATH, media);
+  fail_unless (GST_IS_RTSP_SESSION_MEDIA (sm));
+  fail_unless (gst_rtsp_session_media_matches (sm, TEST_PATH, &match_len));
+  fail_unless (match_len == strlen (TEST_PATH));
+
+  gst_rtsp_transport_new (&ct1);
+  trans = gst_rtsp_session_media_set_transport (sm, stream1, ct1);
+  fail_unless (gst_rtsp_session_media_get_transport (sm, 0) == trans);
+
+  gst_rtsp_transport_new (&ct2);
+  trans = gst_rtsp_session_media_set_transport (sm, stream1, ct2);
+  fail_unless (gst_rtsp_session_media_get_transport (sm, 0) == trans);
+
+  gst_rtsp_transport_new (&ct3);
+  trans = gst_rtsp_session_media_set_transport (sm, stream2, ct3);
+  fail_unless (gst_rtsp_session_media_get_transport (sm, 1) == trans);
+
+  gst_rtsp_transport_new (&ct4);
+  trans = gst_rtsp_session_media_set_transport (sm, stream2, ct4);
+  fail_unless (gst_rtsp_session_media_get_transport (sm, 1) == trans);
+
+  fail_unless (gst_rtsp_media_unprepare (media));
+
+  gst_rtsp_url_free (url);
+
+  g_object_unref (sm);
+
+  g_object_unref (factory);
+  g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_time_and_rtpinfo)
+{
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia *media;
+  GstRTSPUrl *url;
+  GstRTSPStream *stream1, *stream2;
+  GstRTSPThreadPool *pool;
+  GstRTSPThread *thread;
+  GstRTSPSessionMedia *sm;
+  GstClockTime base_time;
+  gchar *rtpinfo;
+  GstRTSPTransport *ct1;
+  GstRTSPStreamTransport *trans;
+  GstRTSPUrl *setup_url;
+  gchar **streaminfo;
+
+  factory = gst_rtsp_media_factory_new ();
+  fail_if (gst_rtsp_media_factory_is_shared (factory));
+  fail_unless (gst_rtsp_url_parse (TEST_PATH, &url) == GST_RTSP_OK);
+
+  gst_rtsp_media_factory_set_launch (factory,
+      "( videotestsrc do-timestamp=true timestamp-offset=0 ! rtpvrawpay pt=96 name=pay0 "
+      "audiotestsrc do-timestamp=true timestamp-offset=1000000000 ! rtpgstpay pt=97 name=pay1 )");
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+
+  fail_unless (gst_rtsp_media_n_streams (media) == 2);
+
+  stream1 = gst_rtsp_media_get_stream (media, 0);
+  fail_unless (GST_IS_RTSP_STREAM (stream1));
+
+  stream2 = gst_rtsp_media_get_stream (media, 1);
+  fail_unless (GST_IS_RTSP_STREAM (stream2));
+
+  pool = gst_rtsp_thread_pool_new ();
+  thread = gst_rtsp_thread_pool_get_thread (pool,
+      GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+  fail_unless (gst_rtsp_media_prepare (media, thread));
+
+  sm = gst_rtsp_session_media_new (TEST_PATH, media);
+  fail_unless (GST_IS_RTSP_SESSION_MEDIA (sm));
+
+  base_time = gst_rtsp_session_media_get_base_time (sm);
+  fail_unless_equals_int64 (base_time, 0);
+
+  rtpinfo = gst_rtsp_session_media_get_rtpinfo (sm);
+  fail_unless (rtpinfo == NULL);
+
+  gst_rtsp_transport_new (&ct1);
+  trans = gst_rtsp_session_media_set_transport (sm, stream1, ct1);
+  fail_unless (gst_rtsp_session_media_get_transport (sm, 0) == trans);
+  fail_unless (gst_rtsp_url_parse (SETUP_URL1, &setup_url) == GST_RTSP_OK);
+  gst_rtsp_stream_transport_set_url (trans, setup_url);
+
+  base_time = gst_rtsp_session_media_get_base_time (sm);
+  fail_unless_equals_int64 (base_time, 0);
+
+  rtpinfo = gst_rtsp_session_media_get_rtpinfo (sm);
+  streaminfo = g_strsplit (rtpinfo, ",", 1);
+  g_free (rtpinfo);
+
+  fail_unless (g_strstr_len (streaminfo[0], -1, "url=") != NULL);
+  fail_unless (g_strstr_len (streaminfo[0], -1, "seq=") != NULL);
+  fail_unless (g_strstr_len (streaminfo[0], -1, "rtptime=") != NULL);
+  fail_unless (g_strstr_len (streaminfo[0], -1, SETUP_URL1) != NULL);
+
+  g_strfreev (streaminfo);
+
+  fail_unless (gst_rtsp_media_unprepare (media));
+
+  rtpinfo = gst_rtsp_session_media_get_rtpinfo (sm);
+  fail_unless (rtpinfo == NULL);
+
+  gst_rtsp_url_free (setup_url);
+  gst_rtsp_url_free (url);
+
+  g_object_unref (sm);
+
+  g_object_unref (factory);
+  g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_allocate_channels)
+{
+  GstRTSPMediaFactory *factory;
+  GstRTSPMedia *media;
+  GstRTSPUrl *url;
+  GstRTSPStream *stream;
+  GstRTSPThreadPool *pool;
+  GstRTSPThread *thread;
+  GstRTSPSessionMedia *sm;
+  GstRTSPRange range;
+
+  factory = gst_rtsp_media_factory_new ();
+  fail_if (gst_rtsp_media_factory_is_shared (factory));
+  fail_unless (gst_rtsp_url_parse (TEST_PATH, &url) == GST_RTSP_OK);
+
+  gst_rtsp_media_factory_set_launch (factory,
+      "( videotestsrc ! rtpvrawpay pt=96 name=pay0 )");
+
+  media = gst_rtsp_media_factory_construct (factory, url);
+  fail_unless (GST_IS_RTSP_MEDIA (media));
+
+  fail_unless (gst_rtsp_media_n_streams (media) == 1);
+
+  stream = gst_rtsp_media_get_stream (media, 0);
+  fail_unless (GST_IS_RTSP_STREAM (stream));
+
+  pool = gst_rtsp_thread_pool_new ();
+  thread = gst_rtsp_thread_pool_get_thread (pool,
+      GST_RTSP_THREAD_TYPE_MEDIA, NULL);
+
+  fail_unless (gst_rtsp_media_prepare (media, thread));
+
+  sm = gst_rtsp_session_media_new (TEST_PATH, media);
+  fail_unless (GST_IS_RTSP_SESSION_MEDIA (sm));
+
+  fail_unless (gst_rtsp_session_media_alloc_channels (sm, &range));
+  fail_unless_equals_int (range.min, 0);
+  fail_unless_equals_int (range.max, 1);
+
+  fail_unless (gst_rtsp_session_media_alloc_channels (sm, &range));
+  fail_unless_equals_int (range.min, 2);
+  fail_unless_equals_int (range.max, 3);
+
+  fail_unless (gst_rtsp_media_unprepare (media));
+
+  gst_rtsp_url_free (url);
+
+  g_object_unref (sm);
+
+  g_object_unref (factory);
+  g_object_unref (pool);
+}
+
+GST_END_TEST;
+static Suite *
+rtspsessionmedia_suite (void)
+{
+  Suite *s = suite_create ("rtspsessionmedia");
+  TCase *tc = tcase_create ("general");
+
+  suite_add_tcase (s, tc);
+  tcase_set_timeout (tc, 20);
+  tcase_add_test (tc, test_setup_url);
+  tcase_add_test (tc, test_rtsp_state);
+  tcase_add_test (tc, test_transports);
+  tcase_add_test (tc, test_time_and_rtpinfo);
+  tcase_add_test (tc, test_allocate_channels);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtspsessionmedia);
diff --git a/tests/check/gst/sessionpool.c b/tests/check/gst/sessionpool.c
new file mode 100644
index 0000000..c1cc149
--- /dev/null
+++ b/tests/check/gst/sessionpool.c
@@ -0,0 +1,197 @@
+/* GStreamer
+ * Copyright (C) 2014 Sebastian Rasmussen 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+#include 
+
+typedef struct
+{
+  GstRTSPSession *sessions[3];
+  GstRTSPFilterResult response[3];
+} Responses;
+
+static GstRTSPFilterResult
+filter_func (GstRTSPSessionPool * pool, GstRTSPSession * session,
+    gpointer user_data)
+{
+  Responses *responses = (Responses *) user_data;
+  gint i;
+
+  for (i = 0; i < 3; i++)
+    if (session == responses->sessions[i])
+      return responses->response[i];
+
+  return GST_RTSP_FILTER_KEEP;
+}
+
+GST_START_TEST (test_pool)
+{
+  GstRTSPSessionPool *pool;
+  GstRTSPSession *session1, *session2, *session3;
+  GstRTSPSession *compare;
+  gchar *session1id, *session2id, *session3id;
+  GList *list;
+  guint maxsessions;
+  GSource *source;
+  guint sourceid;
+
+  pool = gst_rtsp_session_pool_new ();
+  fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 0);
+  fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 0);
+
+  gst_rtsp_session_pool_set_max_sessions (pool, 3);
+  fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 3);
+
+  session1 = gst_rtsp_session_pool_create (pool);
+  fail_unless (GST_IS_RTSP_SESSION (session1));
+  fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 1);
+  fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 3);
+  session1id = g_strdup (gst_rtsp_session_get_sessionid (session1));
+
+  session2 = gst_rtsp_session_pool_create (pool);
+  fail_unless (GST_IS_RTSP_SESSION (session2));
+  fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 2);
+  fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 3);
+  session2id = g_strdup (gst_rtsp_session_get_sessionid (session2));
+
+  session3 = gst_rtsp_session_pool_create (pool);
+  fail_unless (GST_IS_RTSP_SESSION (session3));
+  fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 3);
+  fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 3);
+  session3id = g_strdup (gst_rtsp_session_get_sessionid (session3));
+
+  fail_if (GST_IS_RTSP_SESSION (gst_rtsp_session_pool_create (pool)));
+
+  compare = gst_rtsp_session_pool_find (pool, session1id);
+  fail_unless (compare == session1);
+  g_object_unref (compare);
+  compare = gst_rtsp_session_pool_find (pool, session2id);
+  fail_unless (compare == session2);
+  g_object_unref (compare);
+  compare = gst_rtsp_session_pool_find (pool, session3id);
+  fail_unless (compare == session3);
+  g_object_unref (compare);
+  fail_unless (gst_rtsp_session_pool_find (pool, "") == NULL);
+
+  fail_unless (gst_rtsp_session_pool_remove (pool, session2));
+  g_object_unref (session2);
+  fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 2);
+  fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 3);
+
+  gst_rtsp_session_pool_set_max_sessions (pool, 2);
+  fail_unless_equals_int (gst_rtsp_session_pool_get_n_sessions (pool), 2);
+  fail_unless_equals_int (gst_rtsp_session_pool_get_max_sessions (pool), 2);
+
+  session2 = gst_rtsp_session_pool_create (pool);
+  fail_if (GST_IS_RTSP_SESSION (session2));
+
+  {
+    list = gst_rtsp_session_pool_filter (pool, NULL, NULL);
+    fail_unless_equals_int (g_list_length (list), 2);
+    fail_unless (g_list_find (list, session1) != NULL);
+    fail_unless (g_list_find (list, session3) != NULL);
+    g_list_free_full (list, (GDestroyNotify) g_object_unref);
+  }
+
+  {
+    Responses responses = {
+      {session1, session2, session3},
+      {GST_RTSP_FILTER_KEEP, GST_RTSP_FILTER_KEEP, GST_RTSP_FILTER_KEEP},
+    };
+
+    list = gst_rtsp_session_pool_filter (pool, filter_func, &responses);
+    fail_unless (list == NULL);
+  }
+
+  {
+    Responses responses = {
+      {session1, session2, session3},
+      {GST_RTSP_FILTER_REF, GST_RTSP_FILTER_KEEP, GST_RTSP_FILTER_KEEP},
+    };
+
+    list = gst_rtsp_session_pool_filter (pool, filter_func, &responses);
+    fail_unless_equals_int (g_list_length (list), 1);
+    fail_unless (g_list_nth_data (list, 0) == session1);
+    g_list_free_full (list, (GDestroyNotify) g_object_unref);
+  }
+
+  {
+    Responses responses = {
+      {session1, session2, session3},
+      {GST_RTSP_FILTER_KEEP, GST_RTSP_FILTER_KEEP, GST_RTSP_FILTER_REMOVE},
+    };
+
+    list = gst_rtsp_session_pool_filter (pool, filter_func, &responses);
+    fail_unless_equals_int (g_list_length (list), 0);
+    g_list_free (list);
+  }
+
+  compare = gst_rtsp_session_pool_find (pool, session1id);
+  fail_unless (compare == session1);
+  g_object_unref (compare);
+  fail_unless (gst_rtsp_session_pool_find (pool, session2id) == NULL);
+  fail_unless (gst_rtsp_session_pool_find (pool, session3id) == NULL);
+
+  g_object_get (pool, "max-sessions", &maxsessions, NULL);
+  fail_unless_equals_int (maxsessions, 2);
+
+  g_object_set (pool, "max-sessions", 3, NULL);
+  g_object_get (pool, "max-sessions", &maxsessions, NULL);
+  fail_unless_equals_int (maxsessions, 3);
+
+  fail_unless_equals_int (gst_rtsp_session_pool_cleanup (pool), 0);
+
+  gst_rtsp_session_set_timeout (session1, 1);
+
+  source = gst_rtsp_session_pool_create_watch (pool);
+  fail_unless (source != NULL);
+
+  sourceid = g_source_attach (source, NULL);
+  fail_unless (sourceid != 0);
+
+  while (!g_main_context_iteration (NULL, TRUE));
+
+  g_source_unref (source);
+
+  g_object_unref (session1);
+  g_object_unref (session3);
+
+  g_free (session1id);
+  g_free (session2id);
+  g_free (session3id);
+
+  g_object_unref (pool);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspsessionpool_suite (void)
+{
+  Suite *s = suite_create ("rtspsessionpool");
+  TCase *tc = tcase_create ("general");
+
+  suite_add_tcase (s, tc);
+  tcase_set_timeout (tc, 15);
+  tcase_add_test (tc, test_pool);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtspsessionpool);
diff --git a/tests/check/gst/stream.c b/tests/check/gst/stream.c
new file mode 100644
index 0000000..582e674
--- /dev/null
+++ b/tests/check/gst/stream.c
@@ -0,0 +1,172 @@
+/* GStreamer
+ * Copyright (C) 2013 Axis Communications AB 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+#include 
+
+GST_START_TEST (test_get_sockets)
+{
+  GstPad *srcpad;
+  GstElement *pay;
+  GstRTSPStream *stream;
+  GstBin *bin;
+  GstElement *rtpbin;
+  GSocket *socket;
+  gboolean have_ipv4;
+  gboolean have_ipv6;
+
+  srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
+  fail_unless (srcpad != NULL);
+  gst_pad_set_active (srcpad, TRUE);
+  pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
+  fail_unless (pay != NULL);
+  stream = gst_rtsp_stream_new (0, pay, srcpad);
+  fail_unless (stream != NULL);
+  gst_object_unref (pay);
+  gst_object_unref (srcpad);
+  rtpbin = gst_element_factory_make ("rtpbin", "testrtpbin");
+  fail_unless (rtpbin != NULL);
+  bin = GST_BIN (gst_bin_new ("testbin"));
+  fail_unless (bin != NULL);
+  fail_unless (gst_bin_add (bin, rtpbin));
+
+  fail_unless (gst_rtsp_stream_join_bin (stream, bin, rtpbin, GST_STATE_NULL));
+
+  socket = gst_rtsp_stream_get_rtp_socket (stream, G_SOCKET_FAMILY_IPV4);
+  have_ipv4 = (socket != NULL);
+  if (have_ipv4) {
+    fail_unless (g_socket_get_fd (socket) >= 0);
+    g_object_unref (socket);
+  }
+
+  socket = gst_rtsp_stream_get_rtcp_socket (stream, G_SOCKET_FAMILY_IPV4);
+  if (have_ipv4) {
+    fail_unless (socket != NULL);
+    fail_unless (g_socket_get_fd (socket) >= 0);
+    g_object_unref (socket);
+  } else {
+    fail_unless (socket == NULL);
+  }
+
+  socket = gst_rtsp_stream_get_rtp_socket (stream, G_SOCKET_FAMILY_IPV6);
+  have_ipv6 = (socket != NULL);
+  if (have_ipv6) {
+    fail_unless (g_socket_get_fd (socket) >= 0);
+    g_object_unref (socket);
+  }
+
+  socket = gst_rtsp_stream_get_rtcp_socket (stream, G_SOCKET_FAMILY_IPV6);
+  if (have_ipv6) {
+    fail_unless (socket != NULL);
+    fail_unless (g_socket_get_fd (socket) >= 0);
+    g_object_unref (socket);
+  } else {
+    fail_unless (socket == NULL);
+  }
+
+  /* check that at least one family is available */
+  fail_unless (have_ipv4 || have_ipv6);
+
+  fail_unless (gst_rtsp_stream_leave_bin (stream, bin, rtpbin));
+
+  gst_object_unref (bin);
+  gst_object_unref (stream);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_get_multicast_address)
+{
+  GstPad *srcpad;
+  GstElement *pay;
+  GstRTSPStream *stream;
+  GstRTSPAddressPool *pool;
+  GstRTSPAddress *addr1;
+  GstRTSPAddress *addr2;
+
+  srcpad = gst_pad_new ("testsrcpad", GST_PAD_SRC);
+  fail_unless (srcpad != NULL);
+  gst_pad_set_active (srcpad, TRUE);
+  pay = gst_element_factory_make ("rtpgstpay", "testpayloader");
+  fail_unless (pay != NULL);
+  stream = gst_rtsp_stream_new (0, pay, srcpad);
+  fail_unless (stream != NULL);
+  gst_object_unref (pay);
+  gst_object_unref (srcpad);
+
+  pool = gst_rtsp_address_pool_new ();
+  fail_unless (gst_rtsp_address_pool_add_range (pool,
+          "233.252.0.0", "233.252.0.0", 5000, 5001, 1));
+  fail_unless (gst_rtsp_address_pool_add_range (pool,
+          "FF11:DB8::1", "FF11:DB8::1", 5002, 5003, 1));
+  gst_rtsp_stream_set_address_pool (stream, pool);
+
+  addr1 = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
+  fail_unless (addr1 != NULL);
+  fail_unless_equals_string (addr1->address, "233.252.0.0");
+  fail_unless_equals_int (addr1->port, 5000);
+  fail_unless_equals_int (addr1->n_ports, 2);
+
+  addr2 = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV4);
+  fail_unless (addr2 != NULL);
+  fail_unless_equals_string (addr2->address, "233.252.0.0");
+  fail_unless_equals_int (addr2->port, 5000);
+  fail_unless_equals_int (addr2->n_ports, 2);
+
+  gst_rtsp_address_free (addr1);
+  gst_rtsp_address_free (addr2);
+
+  addr1 = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV6);
+  fail_unless (addr1 != NULL);
+  fail_unless (!g_ascii_strcasecmp (addr1->address, "FF11:DB8::1"));
+  fail_unless_equals_int (addr1->port, 5002);
+  fail_unless_equals_int (addr1->n_ports, 2);
+
+  addr2 = gst_rtsp_stream_get_multicast_address (stream, G_SOCKET_FAMILY_IPV6);
+  fail_unless (addr2 != NULL);
+  fail_unless (!g_ascii_strcasecmp (addr2->address, "FF11:DB8::1"));
+  fail_unless_equals_int (addr2->port, 5002);
+  fail_unless_equals_int (addr2->n_ports, 2);
+
+  gst_rtsp_address_free (addr1);
+  gst_rtsp_address_free (addr2);
+
+  g_object_unref (pool);
+
+  gst_object_unref (stream);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspstream_suite (void)
+{
+  Suite *s = suite_create ("rtspstream");
+  TCase *tc = tcase_create ("general");
+
+  suite_add_tcase (s, tc);
+  tcase_add_test (tc, test_get_sockets);
+  tcase_add_test (tc, test_get_multicast_address);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtspstream);
diff --git a/tests/check/gst/threadpool.c b/tests/check/gst/threadpool.c
new file mode 100644
index 0000000..c92e64f
--- /dev/null
+++ b/tests/check/gst/threadpool.c
@@ -0,0 +1,236 @@
+/* GStreamer
+ * unit tests for GstRTSPThreadPool
+ * Copyright (C) 2013 Axis Communications 
+ * @author Ognyan Tonchev 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+GST_START_TEST (test_pool_get_thread)
+{
+  GstRTSPThreadPool *pool;
+  GstRTSPThread *thread;
+
+  pool = gst_rtsp_thread_pool_new ();
+  fail_unless (GST_IS_RTSP_THREAD_POOL (pool));
+
+  thread = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+      NULL);
+  fail_unless (GST_IS_RTSP_THREAD (thread));
+  /* one ref is hold by the pool */
+  fail_unless_equals_int (GST_MINI_OBJECT_REFCOUNT (thread), 2);
+
+  gst_rtsp_thread_stop (thread);
+  g_object_unref (pool);
+  gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_pool_get_media_thread)
+{
+  GstRTSPThreadPool *pool;
+  GstRTSPThread *thread;
+
+  pool = gst_rtsp_thread_pool_new ();
+  fail_unless (GST_IS_RTSP_THREAD_POOL (pool));
+
+  thread = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_MEDIA,
+      NULL);
+  fail_unless (GST_IS_RTSP_THREAD (thread));
+  /* one ref is hold by the pool */
+  fail_unless_equals_int (GST_MINI_OBJECT_REFCOUNT (thread), 2);
+
+  gst_rtsp_thread_stop (thread);
+  g_object_unref (pool);
+  gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_pool_get_thread_reuse)
+{
+  GstRTSPThreadPool *pool;
+  GstRTSPThread *thread1;
+  GstRTSPThread *thread2;
+
+  pool = gst_rtsp_thread_pool_new ();
+  fail_unless (GST_IS_RTSP_THREAD_POOL (pool));
+
+  gst_rtsp_thread_pool_set_max_threads (pool, 1);
+
+  thread1 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+      NULL);
+  fail_unless (GST_IS_RTSP_THREAD (thread1));
+
+  thread2 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+      NULL);
+  fail_unless (GST_IS_RTSP_THREAD (thread2));
+
+  fail_unless (thread2 == thread1);
+  /* one ref is hold by the pool */
+  fail_unless_equals_int (GST_MINI_OBJECT_REFCOUNT (thread1), 3);
+
+  gst_rtsp_thread_stop (thread1);
+  gst_rtsp_thread_stop (thread2);
+  g_object_unref (pool);
+
+  gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+static void
+do_test_pool_max_thread (gboolean use_property)
+{
+  GstRTSPThreadPool *pool;
+  GstRTSPThread *thread1;
+  GstRTSPThread *thread2;
+  GstRTSPThread *thread3;
+  gint max_threads;
+
+  pool = gst_rtsp_thread_pool_new ();
+  fail_unless (GST_IS_RTSP_THREAD_POOL (pool));
+
+  if (use_property) {
+    g_object_get (pool, "max-threads", &max_threads, NULL);
+    fail_unless_equals_int (max_threads, 1);
+  } else {
+    fail_unless_equals_int (gst_rtsp_thread_pool_get_max_threads (pool), 1);
+  }
+
+  thread1 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+      NULL);
+  fail_unless (GST_IS_RTSP_THREAD (thread1));
+
+  thread2 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+      NULL);
+  fail_unless (GST_IS_RTSP_THREAD (thread2));
+
+  fail_unless (thread1 == thread2);
+
+  gst_rtsp_thread_stop (thread1);
+  gst_rtsp_thread_stop (thread2);
+
+  if (use_property) {
+    g_object_set (pool, "max-threads", 2, NULL);
+    g_object_get (pool, "max-threads", &max_threads, NULL);
+    fail_unless_equals_int (max_threads, 2);
+  } else {
+    gst_rtsp_thread_pool_set_max_threads (pool, 2);
+    fail_unless_equals_int (gst_rtsp_thread_pool_get_max_threads (pool), 2);
+  }
+
+  thread1 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+      NULL);
+  fail_unless (GST_IS_RTSP_THREAD (thread1));
+
+  thread2 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+      NULL);
+  fail_unless (GST_IS_RTSP_THREAD (thread2));
+
+  thread3 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+      NULL);
+  fail_unless (GST_IS_RTSP_THREAD (thread3));
+
+  fail_unless (thread2 != thread1);
+  fail_unless (thread3 == thread2 || thread3 == thread1);
+
+  gst_rtsp_thread_stop (thread1);
+  gst_rtsp_thread_stop (thread2);
+  gst_rtsp_thread_stop (thread3);
+
+  if (use_property) {
+    g_object_set (pool, "max-threads", 0, NULL);
+    g_object_get (pool, "max-threads", &max_threads, NULL);
+    fail_unless_equals_int (max_threads, 0);
+  } else {
+    gst_rtsp_thread_pool_set_max_threads (pool, 0);
+    fail_unless_equals_int (gst_rtsp_thread_pool_get_max_threads (pool), 0);
+  }
+
+  thread1 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+      NULL);
+  fail_if (GST_IS_RTSP_THREAD (thread1));
+
+  g_object_unref (pool);
+
+  gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_START_TEST (test_pool_max_threads)
+{
+  do_test_pool_max_thread (FALSE);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_pool_max_threads_property)
+{
+  do_test_pool_max_thread (TRUE);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_pool_thread_copy)
+{
+  GstRTSPThreadPool *pool;
+  GstRTSPThread *thread1;
+  GstRTSPThread *thread2;
+
+  pool = gst_rtsp_thread_pool_new ();
+  fail_unless (GST_IS_RTSP_THREAD_POOL (pool));
+
+  thread1 = gst_rtsp_thread_pool_get_thread (pool, GST_RTSP_THREAD_TYPE_CLIENT,
+      NULL);
+  fail_unless (GST_IS_RTSP_THREAD (thread1));
+  fail_unless (GST_IS_MINI_OBJECT_TYPE (thread1, GST_TYPE_RTSP_THREAD));
+
+  thread2 = GST_RTSP_THREAD (gst_mini_object_copy (GST_MINI_OBJECT (thread1)));
+  fail_unless (GST_IS_RTSP_THREAD (thread2));
+  fail_unless (GST_IS_MINI_OBJECT_TYPE (thread2, GST_TYPE_RTSP_THREAD));
+
+  gst_rtsp_thread_stop (thread1);
+  gst_rtsp_thread_stop (thread2);
+  g_object_unref (pool);
+  gst_rtsp_thread_pool_cleanup ();
+}
+
+GST_END_TEST;
+
+static Suite *
+rtspthreadpool_suite (void)
+{
+  Suite *s = suite_create ("rtspthreadpool");
+  TCase *tc = tcase_create ("general");
+
+  suite_add_tcase (s, tc);
+  tcase_set_timeout (tc, 20);
+  tcase_add_test (tc, test_pool_get_thread);
+  tcase_add_test (tc, test_pool_get_media_thread);
+  tcase_add_test (tc, test_pool_get_thread_reuse);
+  tcase_add_test (tc, test_pool_max_threads);
+  tcase_add_test (tc, test_pool_max_threads_property);
+  tcase_add_test (tc, test_pool_thread_copy);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtspthreadpool);
diff --git a/tests/check/gst/token.c b/tests/check/gst/token.c
new file mode 100644
index 0000000..df5bbd5
--- /dev/null
+++ b/tests/check/gst/token.c
@@ -0,0 +1,99 @@
+/* GStreamer
+ * Copyright (C) 2013 Sebastian Rasmussen 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+GST_START_TEST (test_token)
+{
+  GstRTSPToken *token;
+  GstRTSPToken *token2;
+  GstRTSPToken *copy;
+  GstStructure *str;
+
+  token = gst_rtsp_token_new_empty ();
+  fail_if (gst_rtsp_token_is_allowed (token, "missing"));
+  gst_rtsp_token_unref (token);
+
+  token = gst_rtsp_token_new ("role", G_TYPE_STRING, "user",
+      "permission1", G_TYPE_BOOLEAN, TRUE,
+      "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+  fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"), "user");
+  fail_unless (gst_rtsp_token_is_allowed (token, "permission1"));
+  fail_if (gst_rtsp_token_is_allowed (token, "permission2"));
+  fail_if (gst_rtsp_token_is_allowed (token, "missing"));
+  copy = GST_RTSP_TOKEN (gst_mini_object_copy (GST_MINI_OBJECT (token)));
+  gst_rtsp_token_unref (token);
+  fail_unless_equals_string (gst_rtsp_token_get_string (copy, "role"), "user");
+  fail_unless (gst_rtsp_token_is_allowed (copy, "permission1"));
+  fail_if (gst_rtsp_token_is_allowed (copy, "permission2"));
+  fail_if (gst_rtsp_token_is_allowed (copy, "missing"));
+  gst_rtsp_token_unref (copy);
+
+  token = gst_rtsp_token_new ("role", G_TYPE_STRING, "user",
+      "permission1", G_TYPE_BOOLEAN, TRUE,
+      "permission2", G_TYPE_BOOLEAN, FALSE, NULL);
+  fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"), "user");
+  fail_unless (gst_rtsp_token_is_allowed (token, "permission1"));
+  fail_if (gst_rtsp_token_is_allowed (token, "permission2"));
+  fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"), "user");
+
+  fail_unless (gst_mini_object_is_writable (GST_MINI_OBJECT (token)));
+  fail_unless (gst_rtsp_token_writable_structure (token) != NULL);
+  fail_unless (gst_rtsp_token_get_structure (token) != NULL);
+
+  token2 = gst_rtsp_token_ref (token);
+
+  fail_if (gst_mini_object_is_writable (GST_MINI_OBJECT (token)));
+  ASSERT_CRITICAL (fail_unless (gst_rtsp_token_writable_structure (token) ==
+          NULL));
+  fail_unless (gst_rtsp_token_get_structure (token) != NULL);
+
+  gst_rtsp_token_unref (token2);
+
+  fail_unless (gst_mini_object_is_writable (GST_MINI_OBJECT (token)));
+  fail_unless (gst_rtsp_token_writable_structure (token) != NULL);
+  fail_unless (gst_rtsp_token_get_structure (token) != NULL);
+
+  str = gst_rtsp_token_writable_structure (token);
+  gst_structure_set (str, "permission2", G_TYPE_BOOLEAN, TRUE, NULL);
+  fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"), "user");
+  fail_unless (gst_rtsp_token_is_allowed (token, "permission1"));
+  fail_unless (gst_rtsp_token_is_allowed (token, "permission2"));
+  fail_unless_equals_string (gst_rtsp_token_get_string (token, "role"), "user");
+  gst_rtsp_token_unref (token);
+}
+
+GST_END_TEST;
+
+static Suite *
+rtsptoken_suite (void)
+{
+  Suite *s = suite_create ("rtsptoken");
+  TCase *tc = tcase_create ("general");
+
+  suite_add_tcase (s, tc);
+  tcase_set_timeout (tc, 20);
+  tcase_add_test (tc, test_token);
+
+  return s;
+}
+
+GST_CHECK_MAIN (rtsptoken);
diff --git a/tests/test-cleanup.c b/tests/test-cleanup.c
new file mode 100644
index 0000000..3b6f7f6
--- /dev/null
+++ b/tests/test-cleanup.c
@@ -0,0 +1,68 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+static gboolean
+timeout (GMainLoop * loop, gboolean ignored)
+{
+  g_main_loop_quit (loop);
+  return FALSE;
+}
+
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+  GstRTSPServer *server;
+  guint id;
+
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* attach the server to the default maincontext */
+  if ((id = gst_rtsp_server_attach (server, NULL)) == 0)
+    goto failed;
+
+  g_timeout_add_seconds (2, (GSourceFunc) timeout, loop);
+
+  /* start serving */
+  g_main_loop_run (loop);
+
+  /* cleanup */
+  g_source_remove (id);
+  g_object_unref (server);
+  g_main_loop_unref (loop);
+
+  return 0;
+
+  /* ERRORS */
+failed:
+  {
+    g_print ("failed to attach the server\n");
+    return -1;
+  }
+}
diff --git a/tests/test-reuse.c b/tests/test-reuse.c
new file mode 100644
index 0000000..67063a8
--- /dev/null
+++ b/tests/test-reuse.c
@@ -0,0 +1,90 @@
+/* GStreamer
+ * Copyright (C) 2008 Wim Taymans 
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include 
+
+#include 
+
+#define TIMEOUT 2
+
+static gboolean timeout_1 (GMainLoop * loop, gboolean ignored);
+
+static guint id;
+static gint rounds = 3;
+static GstRTSPServer *server;
+
+static gboolean
+timeout_2 (GMainLoop * loop, gboolean ignored)
+{
+  rounds--;
+  if (rounds > 0) {
+    id = gst_rtsp_server_attach (server, NULL);
+    g_print ("have attached\n");
+    g_timeout_add_seconds (TIMEOUT, (GSourceFunc) timeout_1, loop);
+  } else {
+    g_main_loop_quit (loop);
+  }
+  return FALSE;
+}
+
+static gboolean
+timeout_1 (GMainLoop * loop, gboolean ignored)
+{
+  g_source_remove (id);
+  g_print ("have removed\n");
+  g_timeout_add_seconds (TIMEOUT, (GSourceFunc) timeout_2, loop);
+  return FALSE;
+}
+
+int
+main (int argc, char *argv[])
+{
+  GMainLoop *loop;
+
+  gst_init (&argc, &argv);
+
+  loop = g_main_loop_new (NULL, FALSE);
+
+  /* create a server instance */
+  server = gst_rtsp_server_new ();
+
+  /* attach the server to the default maincontext */
+  if ((id = gst_rtsp_server_attach (server, NULL)) == 0)
+    goto failed;
+  g_print ("have attached\n");
+
+  g_timeout_add_seconds (TIMEOUT, (GSourceFunc) timeout_1, loop);
+
+  /* start serving */
+  g_main_loop_run (loop);
+
+  /* cleanup */
+  g_object_unref (server);
+  g_main_loop_unref (loop);
+
+  g_print ("quit\n");
+  return 0;
+
+  /* ERRORS */
+failed:
+  {
+    g_print ("failed to attach the server\n");
+    return -1;
+  }
+}