From bb04132f230f5a3f8305c51676da377d9ade903f Mon Sep 17 00:00:00 2001 From: =?utf8?q?Sebastian=20Dr=C3=B6ge?= Date: Fri, 30 Oct 2015 17:04:16 +0200 Subject: [PATCH 01/16] Release 1.6.1 --- ChangeLog | 42 +++++++++++++++++++++++++-- NEWS | 81 +++++++++++++++------------------------------------- RELEASE | 27 ++++++++++-------- configure.ac | 12 ++++---- gst-rtsp-server.doap | 10 +++++++ 5 files changed, 94 insertions(+), 78 deletions(-) diff --git a/ChangeLog b/ChangeLog index 62ce38f..eba4d72 100644 --- a/ChangeLog +++ b/ChangeLog @@ -1,9 +1,47 @@ +=== release 1.6.1 === + +2015-10-30 Sebastian Dröge + + * configure.ac: + releasing 1.6.1 + +2015-10-22 09:15:21 +0200 David Svensson Fors + + * gst/rtsp-server/rtsp-stream.c: + rtsp-stream: Always unref return value of gst_object_get_parent() + Fixes a leak of a GstBin in the udp-mcast case. + https://bugzilla.gnome.org/show_bug.cgi?id=756968 + +2015-09-29 13:04:53 +0100 Tim-Philipp Müller + + * common: + common: update for new suppression + Makes check-valgrind pass with glib 2.46 + +2015-09-28 17:40:59 +0200 Sebastian Rasmussen + + * gst/rtsp-server/rtsp-media.c: + rtsp-media: Take reference to media that will be prepared + default_prepare() takes a transfer-none reference GstRTSPMedia object. + Later on a g_idle_source_new() is created and a pointer to the media + object is passed as user data. If the media is freed before the idle + source is dispatched the media object pointer is invalid, but the idle + source callback expects it to still be valid. To fix this a reference to + the media object is taken when registering the source callback function + and a corresponding release of the reference is done when the souce is + destroyed. + Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755748 + === release 1.6.0 === -2015-09-25 Sebastian Dröge +2015-09-25 23:32:52 +0200 Sebastian Dröge + * ChangeLog: + * NEWS: + * RELEASE: * configure.ac: - releasing 1.6.0 + * gst-rtsp-server.doap: + Release 1.6.0 === release 1.5.91 === diff --git a/NEWS b/NEWS index e04f318..ed483fd 100644 --- a/NEWS +++ b/NEWS @@ -1,64 +1,29 @@ -This is GStreamer 1.6.0 +This is GStreamer 1.6.1 -The GStreamer team is proud to announce a new major feature release in the -stable 1.x API series of your favourite cross-platform multimedia framework! +The GStreamer team is proud to announce the first bugfix release in the stable +1.6 release series of your favourite cross-platform multimedia framework! -This release has been in the works for more than a year and is packed with new -features, bug fixes and other improvements. +This release only contains bugfixes and it is safe to update from 1.6.0. For a +full list of bugfixes see Bugzilla: + https://bugzilla.gnome.org/buglist.cgi?bug_status=RESOLVED&bug_status=VERIFIED&limit=0&list_id=73005&order=bug_id&product=GStreamer&resolution=FIXED&target_milestone=1.6.1 -See http://gstreamer.freedesktop.org/releases/1.6/ for the full list of -changes. +See http://gstreamer.freedesktop.org/releases/1.6/ for the latest version of this document. -Highlights +Major bugfixes -- Stereoscopic 3D and multiview video support -- Trick mode API for key-frame only fast-forward/fast-reverse playback etc. -- Improved DTS (decoding timestamp) vs. PTS (presentation timestamp) handling - to account for negative DTS -- New GstVideoConverter API for more optimised and more correct conversion of - raw video frames between all supported formats, with rescaling -- v4l2src now supports renegotiation -- v4l2transform can now do scaling -- V4L2 Element now report Colorimetry properly -- Easier chunked recording of MP4, Matroska, Ogg, MPEG-TS: new splitmuxsink - and multifilesink improvements -- Content Protection signalling API and Common Encryption (CENC) support for - DASH/MP4 -- Many adaptive streaming (DASH, HLS and MSS) improvements -- New PTP and NTP network client clocks and better remote clock tracking - stability -- High-quality text subtitle overlay at display resolutions with glimagesink - or gtkglsink -- RECORD support for the GStreamer RTSP Server -- Retransmissions (RTX) support in RTSP server and client -- RTSP seeking support in client and server has been fixed -- RTCP scheduling improvements and reduced size RTCP support -- MP4/MOV muxer acquired a new "robust" mode of operation which attempts to - keep the output file in a valid state at all times -- Live mixing support in aggregator, audiomixer and compositor was improved a - lot -- compositor now supports rescaling and converting inputs streams on the fly -- New audiointerleave element with proper input synchronisation and live input - support -- Blackmagic Design DeckLink capture and playback card support was rewritten - from scratch; 2k/4k support; mode sensing -- KLV metadata support in RTP and MPEG-TS -- H.265 video encoder (x265), decoders (libav, libde265) and RTP payloader and - depayloaders -- New DTLS plugin and SRTP/DTLS support -- OpenGL3 support, multiple contexts and context propagation, 3D video, - transfer/conversion separation, subtitle blending -- New OpenGL-based QML video sink, Gtk GL video sink, CoreAnimation - CAOpenGLLayerSink video sink -- gst-libav switched to ffmpeg as libav-provider, gains support for - 3D/multiview video, trick modes, and the CAVS codec -- GstHarness API for unit tests -- gst-editing-services got a completely new ges-launch-1.0 interface, improved - mixing support and integration into gst-validate -- gnonlin has been deprecated in favor of nle (Non Linear Engine) in - gst-editing-services -- gst-validate has a new plugin system, an extensive default testsuite, - support for concurrent test runs and valgrind support -- cerbero build tool for SDK binary packages gains new 'bundle-source' command -- Various improvements to the Android, iOS, OS X and Windows platform support +- Crashes in the gst-libav encoders were fixed +- More DASH-IF test streams are working now +- Live DASH, HLS and MS SmoothStreaming streams work more reliable and other + fixes for the adaptive streaming protocols +- Reverse playback works with scaletempo to keep the audio pitch +- Correct stream-time is reported for negative applied_rate +- SRTP packet validation during decoding does not reject valid packets anymore +- Fixes for audioaggregator and aggregator to start producing output at the + right time, and e.g. not outputting lots of silence in the beginning +- gst-libav's internal ffmpeg snapshot was updated to 2.8.1 +- cerbero has support for Mac OS X 10.11 (El Capitan) +- Various memory leaks were fixed, including major leaks in playbin, playsink + and decodebin +- Various GObject-Introspection annotation fixes for bindings +- and many, many more: https://bugzilla.gnome.org/buglist.cgi?bug_status=RESOLVED&bug_status=VERIFIED&limit=0&list_id=73005&order=bug_id&product=GStreamer&resolution=FIXED&target_milestone=1.6.1 diff --git a/RELEASE b/RELEASE index 0230fd5..599d3e1 100644 --- a/RELEASE +++ b/RELEASE @@ -1,24 +1,27 @@ -Release notes for GStreamer RTSP Server Library 1.6.0 +Release notes for GStreamer RTSP Server Library 1.6.1 -The GStreamer team is proud to announce a new major feature release in the -stable 1.x API series of your favourite cross-platform multimedia framework! +The GStreamer team is proud to announce the first bugfix release in the stable +1.6 release series of your favourite cross-platform multimedia framework! -This release has been in the works for more than a year and is packed with new -features, bug fixes and other improvements. +This release only contains bugfixes and it is safe to update from 1.6.0. For a +full list of bugfixes see Bugzilla. -See -http://gstreamer.freedesktop.org/releases/1.6/ -for the full list of changes. + + +See http://gstreamer.freedesktop.org/releases/1.6/ +for the full release notes. -There were no bugs fixed in this release - +Bugs fixed in this release + + * 755748 : gst-rtsp-server: Retain reference to rtsp-media when preparing + * 756968 : rtsp-stream: GstBin leak in udp-mcast case ==== Download ==== @@ -55,7 +58,7 @@ subscribe to the gstreamer-devel list. Contributors to this release - * Jan Schmidt - * Sebastian Dröge + * David Svensson Fors + * Sebastian Rasmussen * Tim-Philipp Müller   \ No newline at end of file diff --git a/configure.ac b/configure.ac index 50b6128..8e0f930 100644 --- a/configure.ac +++ b/configure.ac @@ -2,7 +2,7 @@ AC_PREREQ(2.69) 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], [1.6.0], +AC_INIT([GStreamer RTSP Server Library], [1.6.1], [http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer], [gst-rtsp-server]) AG_GST_INIT @@ -53,13 +53,13 @@ dnl 1.2.5 => 205 dnl 1.10.9 (who knows) => 1009 dnl dnl sets GST_LT_LDFLAGS -AS_LIBTOOL(GST, 600, 0, 600) +AS_LIBTOOL(GST, 601, 0, 601) dnl *** required versions of GStreamer stuff *** -GST_REQ=1.6.0 -GSTPB_REQ=1.6.0 -GSTPG_REQ=1.6.0 -GSTPD_REQ=1.6.0 +GST_REQ=1.6.1 +GSTPB_REQ=1.6.1 +GSTPG_REQ=1.6.1 +GSTPD_REQ=1.6.1 dnl *** autotools stuff **** diff --git a/gst-rtsp-server.doap b/gst-rtsp-server.doap index 88dae1f..7f874cb 100644 --- a/gst-rtsp-server.doap +++ b/gst-rtsp-server.doap @@ -32,6 +32,16 @@ RTSP server library based on GStreamer + 1.6.1 + 1.6 + + 2015-10-30 + + + + + + 1.6.0 1.6 -- 2.7.4 From 1b7c6507105e9ce387eaf5d0978e19950c002613 Mon Sep 17 00:00:00 2001 From: Hyunjun Ko Date: Wed, 4 Nov 2015 15:45:00 +0900 Subject: [PATCH 02/16] Add temporary build config Change-Id: I59102e9f67356872d2ce9f2b2d9a0add615a12b5 --- .gbs.conf | 3 +++ 1 file changed, 3 insertions(+) create mode 100644 .gbs.conf diff --git a/.gbs.conf b/.gbs.conf new file mode 100644 index 0000000..1b88822 --- /dev/null +++ b/.gbs.conf @@ -0,0 +1,3 @@ +[general] +upstream_branch = tizen +upstream_tag = ${upstreamversion} -- 2.7.4 From 40fbe91cd9a6d88efa5c5eed7d7f2cb27c0adef1 Mon Sep 17 00:00:00 2001 From: Hyunjun Ko Date: Wed, 25 Nov 2015 16:36:41 +0900 Subject: [PATCH 03/16] Initialize to upgrade upstream/1.6 Change-Id: 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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 deleted file mode 100644 index c8c6360..0000000 --- a/ChangeLog +++ /dev/null @@ -1,7726 +0,0 @@ -=== 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 deleted file mode 100644 index 2099840..0000000 --- a/INSTALL +++ /dev/null @@ -1,370 +0,0 @@ -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 -`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 deleted file mode 100644 index 100b512..0000000 --- a/Makefile.am +++ /dev/null @@ -1,79 +0,0 @@ -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 deleted file mode 100644 index cc265ee..0000000 --- a/NEWS +++ /dev/null @@ -1,2 +0,0 @@ -This is GStreamer RTSP Server 1.4.5 - diff --git a/README b/README deleted file mode 100644 index 2373730..0000000 --- a/README +++ /dev/null @@ -1,4 +0,0 @@ -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 deleted file mode 100644 index f5876ce..0000000 --- a/RELEASE +++ /dev/null @@ -1,69 +0,0 @@ - -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 deleted file mode 100644 index 14e4172..0000000 --- a/REQUIREMENTS +++ /dev/null @@ -1,3 +0,0 @@ -You need to have GStreamer. You can use an installed version of -GStreamer or from its build dir. - diff --git a/TODO b/TODO deleted file mode 100644 index 53d7d1c..0000000 --- a/TODO +++ /dev/null @@ -1,3 +0,0 @@ - - - use a config file to configure the server - - error recovery diff --git a/autogen.sh b/autogen.sh deleted file mode 100755 index 267b167..0000000 --- a/autogen.sh +++ /dev/null @@ -1,113 +0,0 @@ -#!/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 deleted file mode 100644 index 92d0d25..0000000 --- a/common/ChangeLog +++ /dev/null @@ -1,1712 +0,0 @@ -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; 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a/common/README +++ /dev/null @@ -1,255 +0,0 @@ -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 deleted file mode 100644 index 7a7a35b..0000000 --- a/common/c-to-xml.py +++ /dev/null @@ -1,36 +0,0 @@ -# -*- 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 deleted file mode 100755 index ee01ff7..0000000 --- a/common/check-exports +++ /dev/null @@ -1,54 +0,0 @@ -#!/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 deleted file mode 100644 index 611546a..0000000 --- a/common/check.mak +++ /dev/null @@ -1,252 +0,0 @@ -# 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 deleted file mode 100644 index d0036b3..0000000 --- a/common/coverage/coverage-report-entry.pl +++ /dev/null @@ -1,69 +0,0 @@ -#!/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
deleted file mode 100644
index 18bd6f1..0000000
--- a/common/coverage/coverage-report.pl
+++ /dev/null
@@ -1,125 +0,0 @@
-#!/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
deleted file mode 100644
index 3fe124b..0000000
--- a/common/coverage/coverage-report.xsl
+++ /dev/null
@@ -1,235 +0,0 @@
-
-
-
-
-  
-
-  
-    
-      
-        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
deleted file mode 100644
index b45728a..0000000
--- a/common/coverage/lcov.mak
+++ /dev/null
@@ -1,47 +0,0 @@
-## .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
deleted file mode 100644
index 53eec24..0000000
--- a/common/cruft.mak
+++ /dev/null
@@ -1,56 +0,0 @@
-# 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
deleted file mode 100755
index aef1d31..0000000
--- a/common/download-translations
+++ /dev/null
@@ -1,152 +0,0 @@
-#!/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
deleted file mode 100755
index f57e307..0000000
--- a/common/extract-release-date-from-doap-file
+++ /dev/null
@@ -1,32 +0,0 @@
-#!/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
deleted file mode 100644
index 682b905..0000000
--- a/common/gettext.patch
+++ /dev/null
@@ -1,12 +0,0 @@
---- 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
deleted file mode 100644
index ef93a5f..0000000
--- a/common/glib-gen.mak
+++ /dev/null
@@ -1,44 +0,0 @@
-# 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
deleted file mode 100644
index 78cab66..0000000
--- a/common/gst-autogen.sh
+++ /dev/null
@@ -1,345 +0,0 @@
-# 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 $hook hook **********";
-          fi
-        }
-      fi
-    done
-  fi
-}
diff --git a/common/gst-indent b/common/gst-indent
deleted file mode 100755
index 732b2ba..0000000
--- a/common/gst-indent
+++ /dev/null
@@ -1,49 +0,0 @@
-#!/bin/sh
-#
-# Check that the code follows a consistant code style
-#
-
-# Check for existence of indent, and error out if not present.
-# On some *bsd systems the binary seems to be called gnunindent,
-# so check for that first.
-
-version=`gnuindent --version 2>/dev/null`
-if test "x$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
deleted file mode 100644
index 2c60392..0000000
--- a/common/gst.supp
+++ /dev/null
@@ -1,3967 +0,0 @@
-### 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
deleted file mode 100755
index 4ad9786..0000000
--- a/common/gstdoc-scangobj
+++ /dev/null
@@ -1,1617 +0,0 @@
-#!/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
deleted file mode 100644
index f19e7d7..0000000
--- a/common/gtk-doc-plugins.mak
+++ /dev/null
@@ -1,382 +0,0 @@
-# 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
deleted file mode 100644
index 9d76889..0000000
--- a/common/gtk-doc.mak
+++ /dev/null
@@ -1,227 +0,0 @@
-###########################################################################
-# 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
deleted file mode 100644
index 3d387ee..0000000
--- a/common/m4/Makefile.am
+++ /dev/null
@@ -1,40 +0,0 @@
-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 \
-	gst-plugin-docs.m4 \
-	gst-valgrind.m4 \
-	gst-x11.m4 \
-	gst.m4 \
-	gtk-doc.m4 \
-	introspection.m4 \
-	pkg.m4 \
-	check.m4 \
-	orc.m4
diff --git a/common/m4/Makefile.in b/common/m4/Makefile.in
deleted file mode 100644
index 80550ab..0000000
--- a/common/m4/Makefile.in
+++ /dev/null
@@ -1,594 +0,0 @@
-# Makefile.in generated by automake 1.14.1 from Makefile.am.
-# @configure_input@
-
-# Copyright (C) 1994-2013 Free Software Foundation, Inc.
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-
-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
deleted file mode 100644
index 867a344..0000000
--- a/common/m4/README
+++ /dev/null
@@ -1,3 +0,0 @@
-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
deleted file mode 100644
index d6c9e33..0000000
--- a/common/m4/as-ac-expand.m4
+++ /dev/null
@@ -1,43 +0,0 @@
-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
deleted file mode 100644
index 3f7920d..0000000
--- a/common/m4/as-auto-alt.m4
+++ /dev/null
@@ -1,50 +0,0 @@
-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
deleted file mode 100644
index 8bb853a..0000000
--- a/common/m4/as-compiler-flag.m4
+++ /dev/null
@@ -1,96 +0,0 @@
-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
deleted file mode 100644
index 309a060..0000000
--- a/common/m4/as-compiler.m4
+++ /dev/null
@@ -1,44 +0,0 @@
-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
deleted file mode 100644
index 2e27050..0000000
--- a/common/m4/as-docbook.m4
+++ /dev/null
@@ -1,78 +0,0 @@
-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
deleted file mode 100644
index af32104..0000000
--- a/common/m4/as-gcc-inline-assembly.m4
+++ /dev/null
@@ -1,52 +0,0 @@
-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
deleted file mode 100644
index 06f0ae4..0000000
--- a/common/m4/as-libtool-tags.m4
+++ /dev/null
@@ -1,83 +0,0 @@
-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
deleted file mode 100644
index 3b16095..0000000
--- a/common/m4/as-libtool.m4
+++ /dev/null
@@ -1,46 +0,0 @@
-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
deleted file mode 100644
index eb9b175..0000000
--- a/common/m4/as-python.m4
+++ /dev/null
@@ -1,152 +0,0 @@
-## ------------------------
-## 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
deleted file mode 100644
index 22ff774..0000000
--- a/common/m4/as-version.m4
+++ /dev/null
@@ -1,75 +0,0 @@
-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
deleted file mode 100644
index 13bf699..0000000
--- a/common/m4/ax_create_stdint_h.m4
+++ /dev/null
@@ -1,734 +0,0 @@
-##### 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
deleted file mode 100644
index afd26eb..0000000
--- a/common/m4/check.m4
+++ /dev/null
@@ -1,181 +0,0 @@
-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
deleted file mode 100644
index f8d442f..0000000
--- a/common/m4/glib-gettext.m4
+++ /dev/null
@@ -1,432 +0,0 @@
-# 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
deleted file mode 100644
index 077a20b..0000000
--- a/common/m4/gst-arch.m4
+++ /dev/null
@@ -1,140 +0,0 @@
-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
deleted file mode 100644
index b478c82..0000000
--- a/common/m4/gst-args.m4
+++ /dev/null
@@ -1,360 +0,0 @@
-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 $PACKAGE_VERSION_MICRO '<' 90`" = "x1"; then
-        FATAL_WARNINGS=yes
-      else
-        FATAL_WARNINGS=no
-      fi
-    ])
-])
diff --git a/common/m4/gst-check.m4 b/common/m4/gst-check.m4
deleted file mode 100644
index 4277053..0000000
--- a/common/m4/gst-check.m4
+++ /dev/null
@@ -1,294 +0,0 @@
-dnl pkg-config-based checks for GStreamer modules and dependency modules
-
-dnl generic:
-dnl AG_GST_PKG_CHECK_MODULES([PREFIX], [WHICH], [REQUIRED])
-dnl sets HAVE_[$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 $GST_PLUGINS_LIBAV_DIR])
-    AC_SUBST(GST_PLUGINS_LIBAV_DIR)
-  fi
-])
diff --git a/common/m4/gst-debuginfo.m4 b/common/m4/gst-debuginfo.m4
deleted file mode 100644
index b48854d..0000000
--- a/common/m4/gst-debuginfo.m4
+++ /dev/null
@@ -1,46 +0,0 @@
-AC_DEFUN([AG_GST_DEBUGINFO], [
-AC_ARG_ENABLE(debug,
-AC_HELP_STRING([--disable-debug],[disable addition of -g debugging info]),
-[case "${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
deleted file mode 100644
index 8de9756..0000000
--- a/common/m4/gst-default.m4
+++ /dev/null
@@ -1,120 +0,0 @@
-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
deleted file mode 100644
index 5d3f0fd..0000000
--- a/common/m4/gst-doc.m4
+++ /dev/null
@@ -1,92 +0,0 @@
-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$DOC_PS = xyes)
-])
diff --git a/common/m4/gst-dowhile.m4 b/common/m4/gst-dowhile.m4
deleted file mode 100644
index 069808d..0000000
--- a/common/m4/gst-dowhile.m4
+++ /dev/null
@@ -1,24 +0,0 @@
-dnl
-dnl Check for working do while(0) macros. This is used by G_STMT_START
-dnl and G_STMT_END in glib/gmacros.h. Without having this defined we
-dnl get "ambigious if-else" compiler warnings when compling C++ code.
-dnl
-dnl Copied from GLib's configure.in
-dnl
-AC_DEFUN([AG_GST_CHECK_DOWHILE_MACROS],[
-
-dnl *** check for working do while(0) macros ***
-AC_CACHE_CHECK([for working do while(0) macros], _cv_g_support_dowhile_macros, [
-	AC_TRY_COMPILE([],[
-	#define STMT_START do
-	#define STMT_END while(0)
-	#define STMT_TEST STMT_START { i = 0; } STMT_END
-	int main(void) { int i = 1; STMT_TEST; return i; }],
-	[_cv_g_support_dowhile_macros=yes],
-	[_cv_g_support_dowhile_macros=no],
-	[_cv_g_support_dowhile_macros=yes])
-])
-if test x$_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
deleted file mode 100644
index bce80d2..0000000
--- a/common/m4/gst-error.m4
+++ /dev/null
@@ -1,298 +0,0 @@
-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
deleted file mode 100644
index 876215e..0000000
--- a/common/m4/gst-feature.m4
+++ /dev/null
@@ -1,297 +0,0 @@
-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
deleted file mode 100644
index 61adfd3..0000000
--- a/common/m4/gst-function.m4
+++ /dev/null
@@ -1,63 +0,0 @@
-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, $function, [macro to use to show function name])
-])
diff --git a/common/m4/gst-gettext.m4 b/common/m4/gst-gettext.m4
deleted file mode 100644
index df817eb..0000000
--- a/common/m4/gst-gettext.m4
+++ /dev/null
@@ -1,28 +0,0 @@
-dnl gettext setup
-
-dnl AG_GST_GETTEXT([gettext-package])
-dnl defines GETTEXT_PACKAGE and LOCALEDIR
-
-AC_DEFUN([AG_GST_GETTEXT],
-[
-  if test "$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
deleted file mode 100644
index 63f0f46..0000000
--- a/common/m4/gst-glib2.m4
+++ /dev/null
@@ -1,126 +0,0 @@
-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
deleted file mode 100644
index 4a843f0..0000000
--- a/common/m4/gst-libxml2.m4
+++ /dev/null
@@ -1,52 +0,0 @@
-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
deleted file mode 100644
index bc885e3..0000000
--- a/common/m4/gst-package-release-datetime.m4
+++ /dev/null
@@ -1,89 +0,0 @@
-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 ], [ $3 ])
-  else
-    AG_GST_SET_PACKAGE_RELEASE_DATETIME([no])
-  fi
-])
diff --git a/common/m4/gst-parser.m4 b/common/m4/gst-parser.m4
deleted file mode 100644
index b002047..0000000
--- a/common/m4/gst-parser.m4
+++ /dev/null
@@ -1,55 +0,0 @@
-AC_DEFUN([AG_GST_BISON_CHECK],
-[
-  dnl FIXME: check if AC_PROG_YACC is suitable here
-  dnl FIXME: make precious
-  AC_PATH_PROG(BISON_PATH, bison, no)
-  if test x$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
deleted file mode 100644
index 40d6faf..0000000
--- a/common/m4/gst-platform.m4
+++ /dev/null
@@ -1,67 +0,0 @@
-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
deleted file mode 100644
index 0e2ab6e..0000000
--- a/common/m4/gst-plugin-docs.m4
+++ /dev/null
@@ -1,25 +0,0 @@
-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
deleted file mode 100644
index c9e1301..0000000
--- a/common/m4/gst-plugindir.m4
+++ /dev/null
@@ -1,17 +0,0 @@
-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
deleted file mode 100644
index 5c0d608..0000000
--- a/common/m4/gst-valgrind.m4
+++ /dev/null
@@ -1,35 +0,0 @@
-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$VALGRIND_PATH" = "xno")
-])
diff --git a/common/m4/gst-x11.m4 b/common/m4/gst-x11.m4
deleted file mode 100644
index d3baf2d..0000000
--- a/common/m4/gst-x11.m4
+++ /dev/null
@@ -1,70 +0,0 @@
-dnl macros for X-related detections
-dnl AC_SUBST's HAVE_X, X_CFLAGS, X_LIBS
-AC_DEFUN([AG_GST_CHECK_X],
-[
-  AC_PATH_XTRA
-  ac_cflags_save="$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
deleted file mode 100644
index d4c53cb..0000000
--- a/common/m4/gst.m4
+++ /dev/null
@@ -1,36 +0,0 @@
-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
deleted file mode 100644
index b243f1c..0000000
--- a/common/m4/gtk-doc.m4
+++ /dev/null
@@ -1,70 +0,0 @@
-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
deleted file mode 100644
index 589721c..0000000
--- a/common/m4/introspection.m4
+++ /dev/null
@@ -1,94 +0,0 @@
-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
deleted file mode 100644
index 26b2459..0000000
--- a/common/m4/orc.m4
+++ /dev/null
@@ -1,70 +0,0 @@
-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 >= $ORC_REQ, which was\n"
-      printf "               not found.  Slower code paths will be used.\n"
-    fi
-  fi
-  printf "\n"
-])
-
diff --git a/common/m4/pkg.m4 b/common/m4/pkg.m4
deleted file mode 100644
index 996e294..0000000
--- a/common/m4/pkg.m4
+++ /dev/null
@@ -1,157 +0,0 @@
-# pkg.m4 - Macros to locate and utilise pkg-config.            -*- Autoconf -*-
-#
-# Copyright © 2004 Scott James Remnant .
-#
-# This program is free software; you can redistribute it and/or modify
-# it under the terms of the GNU General Public License as published by
-# the Free Software Foundation; either version 2 of the License, or
-# (at your option) any later version.
-#
-# This program is distributed in the hope that it will be useful, but
-# WITHOUT ANY WARRANTY; without even the implied warranty of
-# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
-# General Public License for more details.
-#
-# You should have received a copy of the GNU General Public License
-# along with this program; if not, write to the Free Software
-# Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
-#
-# As a special exception to the GNU General Public License, if you
-# distribute this file as part of a program that contains a
-# configuration script generated by Autoconf, you may include it under
-# the same distribution terms that you use for the rest of that program.
-
-# PKG_PROG_PKG_CONFIG([MIN-VERSION])
-# ----------------------------------
-AC_DEFUN([PKG_PROG_PKG_CONFIG],
-[m4_pattern_forbid([^_?PKG_[A-Z_]+$])
-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
deleted file mode 100644
index 51ea8c2..0000000
--- a/common/mangle-tmpl.py
+++ /dev/null
@@ -1,165 +0,0 @@
-# -*- 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
deleted file mode 100644
index a232b5d..0000000
--- a/common/orc.mak
+++ /dev/null
@@ -1,76 +0,0 @@
-#
-# 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
deleted file mode 100644
index 36885df..0000000
--- a/common/parallel-subdirs.mak
+++ /dev/null
@@ -1,13 +0,0 @@
-# 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
deleted file mode 100644
index 60515b6..0000000
--- a/common/plugins.xsl
+++ /dev/null
@@ -1,209 +0,0 @@
- 
-
-
-
-
-
-
-  
-    
-      
-        
-          -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
deleted file mode 100644
index e019fac..0000000
--- a/common/po.mak
+++ /dev/null
@@ -1,4 +0,0 @@
-# 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
deleted file mode 100644
index 715657b..0000000
--- a/common/release.mak
+++ /dev/null
@@ -1,34 +0,0 @@
-# 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
deleted file mode 100755
index 4a9f1fc..0000000
--- a/common/scangobj-merge.py
+++ /dev/null
@@ -1,311 +0,0 @@
-#!/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
deleted file mode 100644
index c1c4945..0000000
--- a/common/upload-doc.mak
+++ /dev/null
@@ -1,64 +0,0 @@
-# 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 $$DIR; \
-	\
-	if echo $(FORMATS) | grep html > /dev/null; then \
-	  echo "Un-preparing docs for upload (rebasing cross-references) ..." ; \
-	  gtkdoc-rebase --html-dir=$(builddir)/html ; \
-	fi; \
-	echo Done
diff --git a/common/win32.mak b/common/win32.mak
deleted file mode 100644
index 30e347e..0000000
--- a/common/win32.mak
+++ /dev/null
@@ -1,78 +0,0 @@
-# various tests to make sure we dist the win32 stuff (for MSVC builds) right
-
-# the MANIFEST contains all win32 related files that should be disted
-win32 = $(shell cat $(top_srcdir)/win32/MANIFEST)
-
-# wildcard is apparently not portable to other makes, hence the use of find
-# these are library .def files with the symbols to export
-win32defs = $(shell find $(top_srcdir)/win32/common -name '*.def')
-
-# wildcard is apparently not portable to other makes, hence the use of find
-# these are files that need to be disted with CRLF line endings:
-win32crlf = $(shell find $(top_srcdir)/win32 -name '*.dsw' -o -name '*.dsp')
-
-win32-debug:
-	@echo; \
-	echo win32     = $(win32); \
-	echo; \
-	echo win32defs = $(win32defs); \
-	echo; \
-	echo win32crlf = $(win32crlf); \
-	echo
-
-win32-check-crlf:
-	@echo Checking win32 files for CR LF line endings ...; \
-	fail=0 ; \
-	for each in $(win32crlf) ; do \
-	  result=`perl -e 'print grep(/\r\n/,<>)' "$$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
deleted file mode 100644
index 7a65c6f..0000000
--- a/configure.ac
+++ /dev/null
@@ -1,344 +0,0 @@
-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
deleted file mode 100644
index 9f9f150..0000000
--- a/docs/Makefile.am
+++ /dev/null
@@ -1,20 +0,0 @@
-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 $$a; \
-	    make upload; \
-	    cd ..; \
-	  done; \
-	fi
diff --git a/docs/Makefile.in b/docs/Makefile.in
deleted file mode 100644
index 3aa1228..0000000
--- a/docs/Makefile.in
+++ /dev/null
@@ -1,751 +0,0 @@
-# Makefile.in generated by automake 1.14.1 from Makefile.am.
-# @configure_input@
-
-# Copyright (C) 1994-2013 Free Software Foundation, Inc.
-
-# This Makefile.in is free software; the Free Software Foundation
-# gives unlimited permission to copy and/or distribute it,
-# with or without modifications, as long as this notice is preserved.
-
-# This program is distributed in the hope that it will be useful,
-# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
-# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
-# PARTICULAR PURPOSE.
-
-@SET_MAKE@
-VPATH = @srcdir@
-am__is_gnu_make = test -n '$(MAKEFILE_LIST)' && test -n '$(MAKELEVEL)'
-am__make_running_with_option = \
-  case $${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 \
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-          | sed "s/$$bs$$bs[$$bs $$bs	]*//g"`;; \
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-  fi; \
-  skip_next=no; \
-  strip_trailopt () \
-  { \
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-      *=*|--*) continue;; \
-        -*I) strip_trailopt 'I'; skip_next=yes;; \
-      -*I?*) strip_trailopt 'I';; \
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-      -*O?*) strip_trailopt 'O';; \
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-      -*l?*) strip_trailopt 'l';; \
-      -[dEDm]) skip_next=yes;; \
-      -[JT]) skip_next=yes;; \
-    esac; \
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-      *$$target_option*) has_opt=yes; break;; \
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-  done; \
-  test $$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))
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-am__cd = CDPATH="$${ZSH_VERSION+.}$(PATH_SEPARATOR)" && cd
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-INSTALL_HEADER = $(INSTALL_DATA)
-transform = $(program_transform_name)
-NORMAL_INSTALL = :
-PRE_INSTALL = :
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-POST_UNINSTALL = :
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-subdir = docs
-DIST_COMMON = $(srcdir)/Makefile.in $(srcdir)/Makefile.am \
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-ACLOCAL_M4 = $(top_srcdir)/aclocal.m4
-am__aclocal_m4_deps = $(top_srcdir)/common/m4/as-ac-expand.m4 \
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-ps-am:
-
-uninstall-am:
-
-.MAKE: $(am__recursive_targets) install-am install-strip
-
-.PHONY: $(am__recursive_targets) CTAGS GTAGS TAGS all all-am check \
-	check-am clean clean-generic clean-libtool cscopelist-am ctags \
-	ctags-am distclean distclean-generic distclean-libtool \
-	distclean-tags distdir dvi dvi-am html html-am info info-am \
-	install install-am install-data install-data-am install-dvi \
-	install-dvi-am install-exec install-exec-am install-html \
-	install-html-am install-info install-info-am install-man \
-	install-pdf install-pdf-am install-ps install-ps-am \
-	install-strip installcheck installcheck-am installdirs \
-	installdirs-am maintainer-clean maintainer-clean-generic \
-	mostlyclean mostlyclean-generic mostlyclean-libtool pdf pdf-am \
-	ps ps-am tags tags-am uninstall uninstall-am
-
-
-upload:
-	@if test "x$(SUBDIRS)" != x; then \
-	  for a in $(SUBDIRS); do \
-	    cd $$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
deleted file mode 100644
index e1cdc9c..0000000
--- a/docs/README
+++ /dev/null
@@ -1,498 +0,0 @@
-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
deleted file mode 100644
index 7da8283..0000000
--- a/docs/design/gst-rtp-server-design
+++ /dev/null
@@ -1,35 +0,0 @@
-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
deleted file mode 100644
index 64b7aa5..0000000
--- a/docs/libs/Makefile.am
+++ /dev/null
@@ -1,106 +0,0 @@
-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
deleted file mode 100644
index 95811a2..0000000
--- a/docs/libs/Makefile.in
+++ /dev/null
@@ -1,946 +0,0 @@
-# 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.
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-GST_PKG_CONFIG_PATH = @GST_PKG_CONFIG_PATH@
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-GST_REVISION = @GST_REVISION@
-GST_TOOLS_DIR = @GST_TOOLS_DIR@
-GTKDOC_CHECK = @GTKDOC_CHECK@
-GTKDOC_DEPS_CFLAGS = @GTKDOC_DEPS_CFLAGS@
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-GTKDOC_MKPDF = @GTKDOC_MKPDF@
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-INTROSPECTION_MAKEFILE = @INTROSPECTION_MAKEFILE@
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-LDFLAGS = @LDFLAGS@
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-PACKAGE_NAME = @PACKAGE_NAME@
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-MODULE = gst-rtsp-server
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-
-# 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
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-	    *$$dep*) \
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-.PRECIOUS: Makefile
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-	@case '$?' in \
-	  *config.status*) \
-	    cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh;; \
-	  *) \
-	    echo ' cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@ $(am__depfiles_maybe)'; \
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-	  sed -e "s|^$$srcdirstrip/||;t" \
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-	  */*) $(MKDIR_P) `echo "$$dist_files" | \
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-	    cp -fpR $$d/$$file "$(distdir)$$dir" || exit 1; \
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-	done
-	$(MAKE) $(AM_MAKEFLAGS) \
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-	fi
-mostlyclean-generic:
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-
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-	install-dvi install-dvi-am install-exec install-exec-am \
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-	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
deleted file mode 100644
index 06141f8..0000000
--- a/docs/libs/gst-rtsp-server-docs.sgml
+++ /dev/null
@@ -1,49 +0,0 @@
-
-
-%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
deleted file mode 100644
index e69de29..0000000
diff --git a/docs/libs/gst-rtsp-server-sections.txt b/docs/libs/gst-rtsp-server-sections.txt
deleted file mode 100644
index e832c4e..0000000
--- a/docs/libs/gst-rtsp-server-sections.txt
+++ /dev/null
@@ -1,697 +0,0 @@
-

-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
deleted file mode 100644
index f4476f2..0000000
--- a/docs/libs/gst-rtsp-server.types
+++ /dev/null
@@ -1,25 +0,0 @@
-#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
deleted file mode 100644
index b51425a..0000000
--- a/docs/libs/html/GstRTSPAuth.html
+++ /dev/null
@@ -1,613 +0,0 @@
-
-
-
-
-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

-

----
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
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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

-

-----
-
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-
-
-
-
-
-
-
-
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-
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-
-
-





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
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-GStreamer RTSP Server Reference Manual: GstRTSPClient
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GstRTSPClient

-
GstRTSPClient — A client connection state

-

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Functions

-

----
-
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-





-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

-

-----
-
-
-
-
-
-
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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

-

-----
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
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-
-
-
-
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-





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

-

-----
-
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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

-

-----
-
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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

-

-----
-
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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

-

-----
-
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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

-

-----
-
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-





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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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

-

-----
-
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-





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
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--- a/docs/libs/html/GstRTSPMedia.html
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-GStreamer RTSP Server Reference Manual: GstRTSPMedia
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GstRTSPMedia

-
GstRTSPMedia — The media pipeline

-

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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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-
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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

-

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

-

-

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Types and Values

-

----
-
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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);
-};
-

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The RTSP media class

-

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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 ()
  

-

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enum GstRTSPMediaStatus

-
The state of the media pipeline.

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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

-		 

-

-

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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.

-

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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

-

-

-

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The “eos-shutdown” property

-
  “eos-shutdown”             gboolean

-
Send an EOS event to the pipeline before unpreparing.

-
Flags: Read / Write

-
Default value: FALSE

-

-

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-
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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-GStreamer RTSP Server Reference Manual: GstRTSPMediaFactory
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GstRTSPMediaFactory

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GstRTSPMediaFactory — A factory for media pipelines

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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 ()
-

-

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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

-

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-

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voidmedia-configureRun Last
voidmedia-constructedRun Last

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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

-

-----
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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

-

-----
-
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-
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-
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-





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

-

-----
-
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-





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

-

-----
-
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-
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-
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-





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

-

-----
-
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-
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-
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-
-





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

-

-----
-
-
-
-
-
-
-
-
-
-
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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

-

-----
-
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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.

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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

-
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GstRTSPMountPoints

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GstRTSPMountPoints — Map a path to media

-

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-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 ()
-

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-

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-
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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

-

-----
-
-
-
-
-
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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

-

-----
-
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-
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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

-

-----
-
-
-
-
-
-
-
-
-
-
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-
-





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

-

-----
-
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-
-





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

-

-

-

-

-
See Also

-
GstRTSPMediaFactory, GstRTSPClient

-

-

-

-

-          Generated by GTK-Doc V1.21

-
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-GStreamer RTSP Server Reference Manual: GstRTSPServer
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GstRTSPServer

-
GstRTSPServer — The main server object

-

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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 ()
-

-

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-

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

-GstRTSPMountPoints *mount-pointsRead / Write

-gchar *serviceRead / Write

-GstRTSPSessionPool *session-poolRead / Write

-

-

-
Signals

-

-----
-
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-





voidclient-connectedRun Last

-

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-
Types and Values

-

----
-
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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
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--- a/docs/libs/html/GstRTSPSession.html
+++ /dev/null
@@ -1,881 +0,0 @@
-
-
-
-
-GStreamer RTSP Server Reference Manual: GstRTSPSession
-
-
-
-
-
-
-
-
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-
-
-
-
-
-

-

-

-
-
-

-
GstRTSPSession

-
GstRTSPSession — An object to manage media

-

-

-
Functions

-

----
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
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-
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-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

-

-----
-
-
-
-
-
-
-
-
-
-
-
-
-
-
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-
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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

-

-----
-
-
-
-
-
-
-
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-
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-
-
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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

-

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-
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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

-
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GstRTSPSessionPool — An object for managing sessions

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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 ()
-

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-

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guintmax-sessionsRead / Write

-

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-

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voidsession-removedRun Last

-

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structGstRTSPSessionPool
structGstRTSPSessionPoolClass

-

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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

-

-----
-
-
-
-
-
-
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-
-
-
-





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

-

-----
-
-
-
-
-
-
-
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-
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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

-

-----
-
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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);
-};
-

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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

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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
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+++ /dev/null
@@ -1,70 +0,0 @@
-
-
-
-
-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
deleted file mode 100644
index 671e5d5..0000000
--- a/docs/libs/html/api-index-full.html
+++ /dev/null
@@ -1,1470 +0,0 @@
-
-
-
-
-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

-
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\ No newline at end of file
diff --git a/docs/libs/html/ch01.html b/docs/libs/html/ch01.html
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-GStreamer RTSP Server Reference Manual: 
-
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-
-
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-
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-

-

-

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-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
-

-

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-          Generated by GTK-Doc V1.21

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\ 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
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diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPAddressPool.html b/docs/libs/html/gst-rtsp-server-GstRTSPAddressPool.html
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-GStreamer RTSP Server Reference Manual: GstRTSPAddressPool
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GstRTSPAddressPool

-
GstRTSPAddressPool — A pool of network addresses

-

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-
Functions

-

----
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-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

-

----
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#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

-

-----
-
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-





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.

-

-
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-

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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

-

-----
-
-
-
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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

-

-----
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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

-

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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

-

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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

-

-----
-
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-
-





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

-

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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

-
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\ 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
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-GStreamer RTSP Server Reference Manual: GstRTSPContext
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GstRTSPContext

-
GstRTSPContext — A client request context

-

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-

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-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

-

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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

-
-
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diff --git a/docs/libs/html/gst-rtsp-server-GstRTSPMediaFactoryURI.html b/docs/libs/html/gst-rtsp-server-GstRTSPMediaFactoryURI.html
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GstRTSPMediaFactoryURI

-
GstRTSPMediaFactoryURI — A factory for URI sources

-

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-GstRTSPMediaFactoryURI *
-
-gst_rtsp_media_factory_uri_new ()
-

-void
-
-gst_rtsp_media_factory_uri_set_uri ()
-

-gchar *
-
-gst_rtsp_media_factory_uri_get_uri ()
-

-

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Types and Values

-

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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

-
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\ 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
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GstRTSPParams

-
GstRTSPParams — Param get and set implementation

-

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-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

-

-----
-
-
-
-
-
-
-
-
-
-
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-
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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

-

-

-

-

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\ 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
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GstRTSPPermissions

-
GstRTSPPermissions — Roles and associated permissions

-

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Functions

-

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-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

-

-----
-
-
-
-
-
-
-
-
-
-
-
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-
-
-
-
-
-
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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

-

-----
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
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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

-

-----
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
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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
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GstRTSPSdp

-
GstRTSPSdp — Make SDP messages

-

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-

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-gboolean
-
-gst_rtsp_sdp_from_media ()
-

-

-

-
Types and Values

-

----
-
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-
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-




 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

-

-----
-
-
-
-
-
-
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-
-
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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

-
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\ 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
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GstRTSPSessionMedia

-
GstRTSPSessionMedia — Media managed in a session

-

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-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
deleted file mode 100644
index 6ae49fe..0000000
--- a/docs/libs/html/gst-rtsp-server-GstRTSPStreamTransport.html
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-GStreamer RTSP Server Reference Manual: GstRTSPStreamTransport
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-

-

-
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-
GstRTSPStreamTransport

-
GstRTSPStreamTransport — A media stream transport configuration

-

-

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Functions

-

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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

-

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-
-
-
-
-
-
-
-
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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
deleted file mode 100644
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--- a/docs/libs/html/gst-rtsp-server-GstRTSPThreadPool.html
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-
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-
-GStreamer RTSP Server Reference Manual: GstRTSPThreadPool
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-

-

-

-
-
-

-
GstRTSPThreadPool

-
GstRTSPThreadPool — A pool of threads

-

-

-
Functions

-

----
-
-
-
-
-
-
-
-
-
-
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-
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-
-





-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
deleted file mode 100644
index 9bd5be3..0000000
--- a/docs/libs/html/gst-rtsp-server-GstRTSPToken.html
+++ /dev/null
@@ -1,443 +0,0 @@
-
-
-
-
-GStreamer RTSP Server Reference Manual: GstRTSPToken
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-

-

-

-
-
-

-
GstRTSPToken

-
GstRTSPToken — Roles and permissions for a client

-

-

-
Functions

-

----
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
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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
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-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
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-
MT safe. 

-
[transfer none]

-

-

-

-

-
gst_rtsp_token_get_string ()

-
const gchar *
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-
Get the string value of field
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-.

-

-
Parameters

-

-----
-
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-
-
-
-
-
-
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token
a GstRTSPToken
 
field
a field name
 

-

-

-
Returns

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 the string value of field
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-. The string
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-. 

-
[transfer none][nullable]

-

-

-

-

-
gst_rtsp_token_is_allowed ()

-
gboolean
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-
Check if token
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- and if it is set to TRUE.

-

-
Parameters

-

-----
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-
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-
-
-
-
-
-
-
-
-
-





token
a GstRTSPToken
 
field
a field name
 

-

-

-
Returns

-
 TRUE if token
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-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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-	-test . = "$(srcdir)" || test -z "$(CONFIG_CLEAN_VPATH_FILES)" || rm -f $(CONFIG_CLEAN_VPATH_FILES)
-
-maintainer-clean-generic:
-	@echo "This command is intended for maintainers to use"
-	@echo "it deletes files that may require special tools to rebuild."
-clean: clean-am
-
-clean-am: clean-generic clean-libtool clean-noinstPROGRAMS \
-	mostlyclean-am
-
-distclean: distclean-am
-	-rm -rf ./$(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 ./$(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: install-am install-strip
-
-.PHONY: 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 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/examples/test-appsrc.c b/examples/test-appsrc.c
deleted file mode 100644
index 9d271ea..0000000
--- a/examples/test-appsrc.c
+++ /dev/null
@@ -1,137 +0,0 @@
-/* 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
deleted file mode 100644
index eff3c74..0000000
--- a/examples/test-auth.c
+++ /dev/null
@@ -1,186 +0,0 @@
-/* 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
deleted file mode 100644
index 600fa3c..0000000
--- a/examples/test-cgroups.c
+++ /dev/null
@@ -1,276 +0,0 @@
-/* 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
deleted file mode 100644
index 0466224..0000000
--- a/examples/test-launch.c
+++ /dev/null
@@ -1,71 +0,0 @@
-/* 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
deleted file mode 100644
index 52e082d..0000000
--- a/examples/test-mp4.c
+++ /dev/null
@@ -1,76 +0,0 @@
-/* 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
deleted file mode 100644
index c7d55fc..0000000
--- a/examples/test-multicast.c
+++ /dev/null
@@ -1,104 +0,0 @@
-/* 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
deleted file mode 100644
index 1d8e6f2..0000000
--- a/examples/test-multicast2.c
+++ /dev/null
@@ -1,125 +0,0 @@
-/* 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
deleted file mode 100644
index f4d104c..0000000
--- a/examples/test-ogg.c
+++ /dev/null
@@ -1,76 +0,0 @@
-/* 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
deleted file mode 100644
index 2e2caa6..0000000
--- a/examples/test-readme.c
+++ /dev/null
@@ -1,67 +0,0 @@
-/* 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
deleted file mode 100644
index 98efe9e..0000000
--- a/examples/test-sdp.c
+++ /dev/null
@@ -1,98 +0,0 @@
-/* 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
deleted file mode 100644
index b0daebc..0000000
--- a/examples/test-uri.c
+++ /dev/null
@@ -1,114 +0,0 @@
-/* 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
deleted file mode 100644
index 0d4a3a5..0000000
--- a/examples/test-video.c
+++ /dev/null
@@ -1,177 +0,0 @@
-/* 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
deleted file mode 100644
index 26af281..0000000
--- a/examples/test-wfd.c
+++ /dev/null
@@ -1,133 +0,0 @@
-/* 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
deleted file mode 100644
index 8dfa661..0000000
--- a/gst-rtsp-server.doap
+++ /dev/null
@@ -1,160 +0,0 @@
-
-
- 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
deleted file mode 100755
index a76fdba..0000000
--- a/gst-rtsp-server.manifest
+++ /dev/null
@@ -1,5 +0,0 @@
-
-	
-		
-	
-
diff --git a/gst/Makefile.am b/gst/Makefile.am
deleted file mode 100644
index e37bbc6..0000000
--- a/gst/Makefile.am
+++ /dev/null
@@ -1 +0,0 @@
-SUBDIRS = rtsp-server
diff --git a/gst/Makefile.in b/gst/Makefile.in
deleted file mode 100644
index 35a2115..0000000
--- a/gst/Makefile.in
+++ /dev/null
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diff --git a/gst/rtsp-server/Makefile.am b/gst/rtsp-server/Makefile.am
deleted file mode 100644
index 60eaf3e..0000000
--- a/gst/rtsp-server/Makefile.am
+++ /dev/null
@@ -1,128 +0,0 @@
-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
-
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-    $(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)
-
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-
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-	$(AM_V_GEN)PKG_CONFIG_PATH="$(GST_PKG_CONFIG_PATH)" \
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-		--strip-prefix=Gst \
-		--warn-all \
-		-I$(top_srcdir) \
-		-I$(top_builddir) \
-		-DIN_GOBJECT_INTROSPECTION=1 \
-		--c-include='gst/rtsp-server/rtsp-server.h' \
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-		--add-include-path=`$(PKG_CONFIG) --variable=girdir gstreamer-rtsp-@GST_API_VERSION@` \
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-		--add-include-path=`$(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" \
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-		--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 = $(libdir)/girepository-1.0/
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-	--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-rtsp-@GST_API_VERSION@` \
-	--includedir=`$(PKG_CONFIG) --variable=girdir gstreamer-sdp-@GST_API_VERSION@` \
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-
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diff --git a/gst/rtsp-server/Makefile.in b/gst/rtsp-server/Makefile.in
deleted file mode 100644
index c7c73ae..0000000
--- a/gst/rtsp-server/Makefile.in
+++ /dev/null
@@ -1,1065 +0,0 @@
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-install: install-am
-install-exec: install-exec-am
-install-data: install-data-am
-uninstall: uninstall-am
-
-install-am: all-am
-	@$(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 $(CONFIG_CLEAN_VPATH_FILES)
-
-maintainer-clean-generic:
-	@echo "This command is intended for maintainers to use"
-	@echo "it deletes files that may require special tools to rebuild."
-clean: clean-am
-
-clean-am: clean-generic clean-libLTLIBRARIES clean-libtool \
-	mostlyclean-am
-
-distclean: distclean-am
-	-rm -rf ./$(DEPDIR)
-	-rm -f Makefile
-distclean-am: clean-am distclean-compile distclean-generic
-
-dvi: dvi-am
-
-dvi-am:
-
-html: html-am
-
-html-am:
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-info: info-am
-
-info-am:
-
-install-data-am: install-girDATA \
-	install-libgstrtspserver_@GST_API_VERSION@includeHEADERS \
-	install-typelibsDATA
-
-install-dvi: install-dvi-am
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-install-dvi-am:
-
-install-exec-am: install-libLTLIBRARIES
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-install-html: install-html-am
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-install-html-am:
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-install-info: install-info-am
-
-install-info-am:
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-install-man:
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-install-pdf-am:
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-install-ps-am:
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-installcheck-am:
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-maintainer-clean: maintainer-clean-am
-	-rm -rf ./$(DEPDIR)
-	-rm -f Makefile
-maintainer-clean-am: distclean-am maintainer-clean-generic
-
-mostlyclean: mostlyclean-am
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-mostlyclean-am: mostlyclean-compile mostlyclean-generic \
-	mostlyclean-libtool
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-pdf: pdf-am
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-ps: ps-am
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-ps-am:
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-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 \
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-	maintainer-clean-generic mostlyclean mostlyclean-compile \
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-	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
deleted file mode 100755
index 92883d3..0000000
--- a/gst/rtsp-server/gstwfdmessage.c
+++ /dev/null
@@ -1,1968 +0,0 @@
-/* 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 EDID_BLOCK_COUNT_MAX_SIZE 256
-#define MAX_PORT_SIZE 65535
-
-#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) {
-      if (msg->display_edid->edid_block_count > 0 &&
-          msg->display_edid->edid_block_count <= EDID_BLOCK_COUNT_MAX_SIZE) {
-        g_string_append_printf (lines," %04x", msg->display_edid->edid_block_count);
-        g_string_append_printf(lines, " %s", msg->display_edid->edid_payload);
-      } 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;
-  if (edid_blockcount > 0 && edid_blockcount <= EDID_BLOCK_COUNT_MAX_SIZE) {
-    msg->display_edid->edid_block_count = edid_blockcount;
-    msg->display_edid->edid_payload = g_malloc(EDID_BLOCK_SIZE * edid_blockcount);
-    if (msg->display_edid->edid_payload)
-      memcpy(msg->display_edid->edid_payload, edid_playload, EDID_BLOCK_SIZE * edid_blockcount);
-    else
-      msg->display_edid->edid_supported = FALSE;
-  } else
-    msg->display_edid->edid_supported = FALSE;
-
-  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);
-  g_return_val_if_fail (edid_supported != NULL, GST_WFD_EINVAL);
-  g_return_val_if_fail (edid_blockcount != NULL, GST_WFD_EINVAL);
-  g_return_val_if_fail (edid_playload != NULL, GST_WFD_EINVAL);
-
-  *edid_supported = FALSE;
-  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 > 0 && msg->display_edid->edid_block_count <= EDID_BLOCK_COUNT_MAX_SIZE) {
-        char *temp;
-        temp = g_malloc0(EDID_BLOCK_SIZE * msg->display_edid->edid_block_count);
-        if (temp) {
-          memcpy(temp, msg->display_edid->edid_payload, EDID_BLOCK_SIZE * msg->display_edid->edid_block_count);
-          *edid_playload = temp;
-          *edid_supported = TRUE;
-        }
-      }
-    }
-  }
-  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);
-  g_return_val_if_fail (TCPPort <= MAX_PORT_SIZE, 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
deleted file mode 100755
index 3261289..0000000
--- a/gst/rtsp-server/gstwfdmessage.h
+++ /dev/null
@@ -1,618 +0,0 @@
-/* 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
deleted file mode 100644
index 1343839..0000000
--- a/gst/rtsp-server/rtsp-address-pool.c
+++ /dev/null
@@ -1,751 +0,0 @@
-/* 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
deleted file mode 100644
index a473bdf..0000000
--- a/gst/rtsp-server/rtsp-address-pool.h
+++ /dev/null
@@ -1,182 +0,0 @@
-/* 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
deleted file mode 100644
index b0e924e..0000000
--- a/gst/rtsp-server/rtsp-auth.c
+++ /dev/null
@@ -1,636 +0,0 @@
-/* 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
deleted file mode 100644
index cb281f0..0000000
--- a/gst/rtsp-server/rtsp-auth.h
+++ /dev/null
@@ -1,173 +0,0 @@
-/* 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
deleted file mode 100644
index 8c68fdb..0000000
--- a/gst/rtsp-server/rtsp-client-wfd.c
+++ /dev/null
@@ -1,2959 +0,0 @@
-/* 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_KEEP_ALIVE_FAIL,
-  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);
-
-  gst_rtsp_client_wfd_signals[SIGNAL_WFD_KEEP_ALIVE_FAIL] =
-      g_signal_new ("wfd-keep-alive-fail", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
-      G_STRUCT_OFFSET (GstRTSPWFDClientClass, wfd_keep_alive_fail), NULL, NULL,
-      g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
-
-  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);
-
-    g_signal_emit (client,
-        gst_rtsp_client_wfd_signals[SIGNAL_WFD_KEEP_ALIVE_FAIL], 0,
-        NULL);
-    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;
-}
-
-guint64
-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;
-}
-
-guint64
-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;
-}
-
-guint64
-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, guint64 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, guint64 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, guint64 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
deleted file mode 100644
index a7486a4..0000000
--- a/gst/rtsp-server/rtsp-client-wfd.h
+++ /dev/null
@@ -1,221 +0,0 @@
-/* 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);
-  void     (*wfd_keep_alive_fail)         (GstRTSPWFDClient *client);
-
-  /*< 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);
-guint64   gst_rtsp_wfd_client_get_video_cea_resolution(GstRTSPWFDClient *client);
-guint64   gst_rtsp_wfd_client_get_video_vesa_resolution(GstRTSPWFDClient *client);
-guint64   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, guint64 res);
-void gst_rtsp_wfd_client_set_video_vesa_resolution(GstRTSPWFDClient *client, guint64 res);
-void gst_rtsp_wfd_client_set_video_hh_resolution(GstRTSPWFDClient *client, guint64 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
deleted file mode 100644
index ac1ce96..0000000
--- a/gst/rtsp-server/rtsp-client.c
+++ /dev/null
@@ -1,3564 +0,0 @@
-/* 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;
-  }
-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;
-  }
-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;
-  }
-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;
-  }
-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
deleted file mode 100644
index 2ddb82e..0000000
--- a/gst/rtsp-server/rtsp-client.h
+++ /dev/null
@@ -1,196 +0,0 @@
-/* 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
deleted file mode 100644
index 919ca2d..0000000
--- a/gst/rtsp-server/rtsp-context.c
+++ /dev/null
@@ -1,92 +0,0 @@
-/* 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
deleted file mode 100644
index c2be87e..0000000
--- a/gst/rtsp-server/rtsp-context.h
+++ /dev/null
@@ -1,88 +0,0 @@
-/* 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
deleted file mode 100644
index 01f3f99..0000000
--- a/gst/rtsp-server/rtsp-media-factory-uri.c
+++ /dev/null
@@ -1,645 +0,0 @@
-/* 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
deleted file mode 100644
index 6dc3a40..0000000
--- a/gst/rtsp-server/rtsp-media-factory-uri.h
+++ /dev/null
@@ -1,80 +0,0 @@
-/* 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
deleted file mode 100644
index 3c1b34d..0000000
--- a/gst/rtsp-server/rtsp-media-factory-wfd.c
+++ /dev/null
@@ -1,1200 +0,0 @@
-/* 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_waylandsrc_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 waylandsrc as video source");
-
-  videosrc = gst_element_factory_make ("waylandsrc", "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, "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, 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_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;
-    case GST_WFD_VSRC_WAYLANDSRC:
-      if (!_rtsp_media_factory_wfd_create_waylandsrc_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
deleted file mode 100644
index 9b8fc75..0000000
--- a/gst/rtsp-server/rtsp-media-factory-wfd.h
+++ /dev/null
@@ -1,131 +0,0 @@
-/* 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,
-  GST_WFD_VSRC_WAYLANDSRC
-};
-
-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
deleted file mode 100644
index 7d85dcc..0000000
--- a/gst/rtsp-server/rtsp-media-factory.c
+++ /dev/null
@@ -1,1147 +0,0 @@
-/* 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
deleted file mode 100644
index bb6aec3..0000000
--- a/gst/rtsp-server/rtsp-media-factory.h
+++ /dev/null
@@ -1,154 +0,0 @@
-/* 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
deleted file mode 100644
index 6b81e25..0000000
--- a/gst/rtsp-server/rtsp-media.c
+++ /dev/null
@@ -1,3092 +0,0 @@
-/* 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
deleted file mode 100644
index a78650d..0000000
--- a/gst/rtsp-server/rtsp-media.h
+++ /dev/null
@@ -1,235 +0,0 @@
-/* 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
deleted file mode 100644
index c6a8c06..0000000
--- a/gst/rtsp-server/rtsp-mount-points.c
+++ /dev/null
@@ -1,370 +0,0 @@
-/* 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
deleted file mode 100644
index ed13d59..0000000
--- a/gst/rtsp-server/rtsp-mount-points.h
+++ /dev/null
@@ -1,91 +0,0 @@
-/* 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
deleted file mode 100644
index 6025722..0000000
--- a/gst/rtsp-server/rtsp-params.c
+++ /dev/null
@@ -1,77 +0,0 @@
-/* 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
deleted file mode 100644
index 563757d..0000000
--- a/gst/rtsp-server/rtsp-params.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/* 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
deleted file mode 100644
index 66e9e44..0000000
--- a/gst/rtsp-server/rtsp-permissions.c
+++ /dev/null
@@ -1,276 +0,0 @@
-/* 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
deleted file mode 100644
index 9e739c7..0000000
--- a/gst/rtsp-server/rtsp-permissions.h
+++ /dev/null
@@ -1,101 +0,0 @@
-/* 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
deleted file mode 100644
index 07aea90..0000000
--- a/gst/rtsp-server/rtsp-sdp.c
+++ /dev/null
@@ -1,482 +0,0 @@
-/* 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
deleted file mode 100644
index 7732f36..0000000
--- a/gst/rtsp-server/rtsp-sdp.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/* 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
deleted file mode 100644
index 15dac76..0000000
--- a/gst/rtsp-server/rtsp-server-wfd.c
+++ /dev/null
@@ -1,332 +0,0 @@
-/* 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
deleted file mode 100644
index 13a5f3a..0000000
--- a/gst/rtsp-server/rtsp-server-wfd.h
+++ /dev/null
@@ -1,155 +0,0 @@
-/* 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
deleted file mode 100644
index 686ba03..0000000
--- a/gst/rtsp-server/rtsp-server.c
+++ /dev/null
@@ -1,1430 +0,0 @@
-/* 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
deleted file mode 100644
index 9a6f9d4..0000000
--- a/gst/rtsp-server/rtsp-server.h
+++ /dev/null
@@ -1,169 +0,0 @@
-/* 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
deleted file mode 100644
index 8eb1f28..0000000
--- a/gst/rtsp-server/rtsp-session-media.c
+++ /dev/null
@@ -1,505 +0,0 @@
-/* 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
deleted file mode 100644
index e2af5ad..0000000
--- a/gst/rtsp-server/rtsp-session-media.h
+++ /dev/null
@@ -1,97 +0,0 @@
-/* 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
deleted file mode 100644
index fad5288..0000000
--- a/gst/rtsp-server/rtsp-session-pool.c
+++ /dev/null
@@ -1,765 +0,0 @@
-/* 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
deleted file mode 100644
index e2ba951..0000000
--- a/gst/rtsp-server/rtsp-session-pool.h
+++ /dev/null
@@ -1,143 +0,0 @@
-/* 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
deleted file mode 100644
index 372746a..0000000
--- a/gst/rtsp-server/rtsp-session.c
+++ /dev/null
@@ -1,649 +0,0 @@
-/* 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
deleted file mode 100644
index d9c0f18..0000000
--- a/gst/rtsp-server/rtsp-session.h
+++ /dev/null
@@ -1,143 +0,0 @@
-/* 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
deleted file mode 100644
index a99295b..0000000
--- a/gst/rtsp-server/rtsp-stream-transport.c
+++ /dev/null
@@ -1,503 +0,0 @@
-/* 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
deleted file mode 100644
index 7735ee7..0000000
--- a/gst/rtsp-server/rtsp-stream-transport.h
+++ /dev/null
@@ -1,133 +0,0 @@
-/* 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
deleted file mode 100644
index 8e1b216..0000000
--- a/gst/rtsp-server/rtsp-stream.c
+++ /dev/null
@@ -1,2799 +0,0 @@
-/* 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
deleted file mode 100644
index dddbbf4..0000000
--- a/gst/rtsp-server/rtsp-stream.h
+++ /dev/null
@@ -1,190 +0,0 @@
-/* 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
deleted file mode 100644
index a8cce78..0000000
--- a/gst/rtsp-server/rtsp-thread-pool.c
+++ /dev/null
@@ -1,561 +0,0 @@
-/* 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
deleted file mode 100644
index e360034..0000000
--- a/gst/rtsp-server/rtsp-thread-pool.h
+++ /dev/null
@@ -1,179 +0,0 @@
-/* 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
deleted file mode 100644
index 69250f9..0000000
--- a/gst/rtsp-server/rtsp-token.c
+++ /dev/null
@@ -1,251 +0,0 @@
-/* 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
deleted file mode 100644
index 265d8d3..0000000
--- a/gst/rtsp-server/rtsp-token.h
+++ /dev/null
@@ -1,97 +0,0 @@
-/* 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
deleted file mode 100644
index 7d47de6..0000000
--- a/packaging/gst-rtsp-server.spec
+++ /dev/null
@@ -1,73 +0,0 @@
-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
deleted file mode 100644
index c45755e..0000000
--- a/pkgconfig/Makefile.am
+++ /dev/null
@@ -1,21 +0,0 @@
-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) $(pcfiles_uninstalled)
diff --git a/pkgconfig/Makefile.in b/pkgconfig/Makefile.in
deleted file mode 100644
index bafc540..0000000
--- a/pkgconfig/Makefile.in
+++ /dev/null
@@ -1,637 +0,0 @@
-# Makefile.in generated by automake 1.14.1 from Makefile.am.
-# @configure_input@
-
-# Copyright (C) 1994-2013 Free Software Foundation, Inc.
-
-# This Makefile.in is free software; the Free Software Foundation
-# gives unlimited permission to copy and/or distribute it,
-# with or without modifications, as long as this notice is preserved.
-
-# This program is distributed in the hope that it will be useful,
-# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
-# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
-# PARTICULAR PURPOSE.
-
-@SET_MAKE@
-
-VPATH = @srcdir@
-am__is_gnu_make = test -n '$(MAKEFILE_LIST)' && test -n '$(MAKELEVEL)'
-am__make_running_with_option = \
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-      ?) ;; \
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-  if $(am__is_gnu_make); then \
-    sane_makeflags=$$MFLAGS; \
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-    case $$MAKEFLAGS in \
-      *\\[\ \	]*) \
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-          | sed "s/$$bs$$bs[$$bs $$bs	]*//g"`;; \
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-  test $$has_opt = yes
-am__make_dryrun = (target_option=n; $(am__make_running_with_option))
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-pkgdatadir = $(datadir)/@PACKAGE@
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-pkglibdir = $(libdir)/@PACKAGE@
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-am__cd = CDPATH="$${ZSH_VERSION+.}$(PATH_SEPARATOR)" && cd
-install_sh_DATA = $(install_sh) -c -m 644
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-am__configure_deps = $(am__aclocal_m4_deps) $(CONFIGURE_DEPENDENCIES) \
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diff --git a/pkgconfig/gstreamer-rtsp-server.pc.in b/pkgconfig/gstreamer-rtsp-server.pc.in
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-	@rm valgrind.log
-
-# valgrind any given test and generate suppressions for it
-%.valgrind.gen-suppressions: %
-	@$(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
deleted file mode 100644
index 9a0ff54..0000000
--- a/tests/check/gst/addresspool.c
+++ /dev/null
@@ -1,286 +0,0 @@
-/* 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
deleted file mode 100644
index 1bbd67d..0000000
--- a/tests/check/gst/client.c
+++ /dev/null
@@ -1,929 +0,0 @@
-/* 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
deleted file mode 100644
index c7d680e..0000000
--- a/tests/check/gst/media.c
+++ /dev/null
@@ -1,434 +0,0 @@
-/* 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
deleted file mode 100644
index b6b250b..0000000
--- a/tests/check/gst/mediafactory.c
+++ /dev/null
@@ -1,336 +0,0 @@
-/* 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
deleted file mode 100644
index 31a223e..0000000
--- a/tests/check/gst/mountpoints.c
+++ /dev/null
@@ -1,147 +0,0 @@
-/* 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
deleted file mode 100644
index 0dcf770..0000000
--- a/tests/check/gst/permissions.c
+++ /dev/null
@@ -1,107 +0,0 @@
-/* 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
deleted file mode 100644
index 28910f7..0000000
--- a/tests/check/gst/rtspserver.c
+++ /dev/null
@@ -1,1454 +0,0 @@
-/* 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
deleted file mode 100644
index 13445b4..0000000
--- a/tests/check/gst/sessionmedia.c
+++ /dev/null
@@ -1,399 +0,0 @@
-/* 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
deleted file mode 100644
index c1cc149..0000000
--- a/tests/check/gst/sessionpool.c
+++ /dev/null
@@ -1,197 +0,0 @@
-/* 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
deleted file mode 100644
index 582e674..0000000
--- a/tests/check/gst/stream.c
+++ /dev/null
@@ -1,172 +0,0 @@
-/* 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
deleted file mode 100644
index c92e64f..0000000
--- a/tests/check/gst/threadpool.c
+++ /dev/null
@@ -1,236 +0,0 @@
-/* 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
deleted file mode 100644
index df5bbd5..0000000
--- a/tests/check/gst/token.c
+++ /dev/null
@@ -1,99 +0,0 @@
-/* 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
deleted file mode 100644
index 3b6f7f6..0000000
--- a/tests/test-cleanup.c
+++ /dev/null
@@ -1,68 +0,0 @@
-/* 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
deleted file mode 100644
index 67063a8..0000000
--- a/tests/test-reuse.c
+++ /dev/null
@@ -1,90 +0,0 @@
-/* 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;
-  }
-}
-- 
2.7.4


From 4aad84e8446f34dec2dd206ce9586acb7d031874 Mon Sep 17 00:00:00 2001
From: Hyunjun Ko 
Date: Wed, 25 Nov 2015 17:24:27 +0900
Subject: [PATCH 04/16] Apply Wifi Display feature

Change-Id: I5c928adb9525f9c69bafcd263d0d538490ce42c1
---
 .gitignore                               |   68 -
 .gitmodules                              |    3 -
 gst-rtsp-server.manifest                 |    5 +
 gst/rtsp-server/Makefile.am              |   10 +-
 gst/rtsp-server/gstwfdmessage.c          | 1968 ++++++++++++++++++++
 gst/rtsp-server/gstwfdmessage.h          |  618 +++++++
 gst/rtsp-server/rtsp-client-wfd.c        | 2959 ++++++++++++++++++++++++++++++
 gst/rtsp-server/rtsp-client-wfd.h        |  221 +++
 gst/rtsp-server/rtsp-client.c            |   36 +-
 gst/rtsp-server/rtsp-client.h            |    5 +-
 gst/rtsp-server/rtsp-media-factory-wfd.c | 1200 ++++++++++++
 gst/rtsp-server/rtsp-media-factory-wfd.h |  131 ++
 gst/rtsp-server/rtsp-server-wfd.c        |  332 ++++
 gst/rtsp-server/rtsp-server-wfd.h        |  155 ++
 gst/rtsp-server/rtsp-stream.c            |   23 +
 gst/rtsp-server/rtsp-stream.h            |    1 +
 packaging/common.tar.bz2                 |  Bin 0 -> 102577 bytes
 packaging/gst-rtsp-server.spec           |   76 +
 18 files changed, 7732 insertions(+), 79 deletions(-)
 delete mode 100644 .gitignore
 delete mode 100644 .gitmodules
 create mode 100644 gst-rtsp-server.manifest
 create mode 100755 gst/rtsp-server/gstwfdmessage.c
 create mode 100755 gst/rtsp-server/gstwfdmessage.h
 create mode 100644 gst/rtsp-server/rtsp-client-wfd.c
 create mode 100644 gst/rtsp-server/rtsp-client-wfd.h
 create mode 100644 gst/rtsp-server/rtsp-media-factory-wfd.c
 create mode 100644 gst/rtsp-server/rtsp-media-factory-wfd.h
 create mode 100644 gst/rtsp-server/rtsp-server-wfd.c
 create mode 100644 gst/rtsp-server/rtsp-server-wfd.h
 create mode 100644 packaging/common.tar.bz2
 create mode 100644 packaging/gst-rtsp-server.spec

diff --git a/.gitignore b/.gitignore
deleted file mode 100644
index 574f665..0000000
--- a/.gitignore
+++ /dev/null
@@ -1,68 +0,0 @@
-*.[oa]
-*.pyc
-*.gcda
-*.gcno
-*.la
-*.lo
-*.loT
-*.sw[po]
-*.tar.*
-*~
-*.gc??
-.deps
-.libs
-ABOUT-NLS
-INSTALL
-Makefile
-Makefile.in
-aclocal.m4
-autom4te.cache
-autoregen.sh
-compile
-config.guess
-config.h
-config.h.in
-config.log
-config.rpath
-config.status
-config.sub
-configure
-depcomp
-install-sh
-libtool
-ltmain.sh
-missing
-stamp-h1
-tags
-gst-rtsp.spec
-stamp-h.in
-.dirstamp
-
-/m4/*m4
-
-/gst/rtsp-server/GstRtspServer-1.0.gir
-/gst/rtsp-server/GstRtspServer-1.0.typelib
-
-/examples/test-multicast
-/examples/test-multicast2
-
-/test-driver
-/tests/check/gst/*.log
-/tests/check/gst/*.trs
-/tests/check/test-suite.log
-/tests/check/gst/addresspool
-/tests/check/gst/client
-/tests/check/gst/media
-/tests/check/gst/mediafactory
-/tests/check/gst/mountpoints
-/tests/check/gst/permissions
-/tests/check/gst/rtspserver
-/tests/check/gst/sessionmedia
-/tests/check/gst/sessionpool
-/tests/check/gst/stream
-/tests/check/gst/threadpool
-/tests/check/gst/token
-/tests/check/test-registry.reg
-/tests/test-reuse
-
-/po
diff --git a/.gitmodules b/.gitmodules
deleted file mode 100644
index a6b1eda..0000000
--- a/.gitmodules
+++ /dev/null
@@ -1,3 +0,0 @@
-[submodule "common"]
-	path = common
-	url = git://anongit.freedesktop.org/gstreamer/common
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/rtsp-server/Makefile.am b/gst/rtsp-server/Makefile.am
index 4fcd366..60eaf3e 100644
--- a/gst/rtsp-server/Makefile.am
+++ b/gst/rtsp-server/Makefile.am
@@ -7,6 +7,7 @@ public_headers = \
 		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 \
@@ -16,8 +17,11 @@ public_headers = \
 		rtsp-session-media.h \
 		rtsp-session-pool.h \
 		rtsp-token.h \
+		rtsp-client-wfd.h \
 		rtsp-client.h \
-		rtsp-server.h
+		rtsp-server-wfd.h \
+		rtsp-server.h \
+		gstwfdmessage.h
 
 c_sources = \
 	rtsp-auth.c \
@@ -28,6 +32,7 @@ c_sources = \
 	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 \
@@ -37,7 +42,10 @@ c_sources = \
 	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 = 
diff --git a/gst/rtsp-server/gstwfdmessage.c b/gst/rtsp-server/gstwfdmessage.c
new file mode 100755
index 0000000..b70e57d
--- /dev/null
+++ b/gst/rtsp-server/gstwfdmessage.c
@@ -0,0 +1,1968 @@
+/* GStreamer
+ * Copyright (C) 2015 Samsung Electronics Hyunjun Ko 
+ *
+ * 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 EDID_BLOCK_COUNT_MAX_SIZE 256
+#define MAX_PORT_SIZE 65535
+
+#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) {
+      if (msg->display_edid->edid_block_count > 0 &&
+          msg->display_edid->edid_block_count <= EDID_BLOCK_COUNT_MAX_SIZE) {
+        g_string_append_printf (lines," %04x", msg->display_edid->edid_block_count);
+        g_string_append_printf(lines, " %s", msg->display_edid->edid_payload);
+      } 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;
+  if (edid_blockcount > 0 && edid_blockcount <= EDID_BLOCK_COUNT_MAX_SIZE) {
+    msg->display_edid->edid_block_count = edid_blockcount;
+    msg->display_edid->edid_payload = g_malloc(EDID_BLOCK_SIZE * edid_blockcount);
+    if (msg->display_edid->edid_payload)
+      memcpy(msg->display_edid->edid_payload, edid_playload, EDID_BLOCK_SIZE * edid_blockcount);
+    else
+      msg->display_edid->edid_supported = FALSE;
+  } else
+    msg->display_edid->edid_supported = FALSE;
+
+  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);
+  g_return_val_if_fail (edid_supported != NULL, GST_WFD_EINVAL);
+  g_return_val_if_fail (edid_blockcount != NULL, GST_WFD_EINVAL);
+  g_return_val_if_fail (edid_playload != NULL, GST_WFD_EINVAL);
+
+  *edid_supported = FALSE;
+  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 > 0 && msg->display_edid->edid_block_count <= EDID_BLOCK_COUNT_MAX_SIZE) {
+        char *temp;
+        temp = g_malloc0(EDID_BLOCK_SIZE * msg->display_edid->edid_block_count);
+        if (temp) {
+          memcpy(temp, msg->display_edid->edid_payload, EDID_BLOCK_SIZE * msg->display_edid->edid_block_count);
+          *edid_playload = temp;
+          *edid_supported = TRUE;
+        }
+      }
+    }
+  }
+  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);
+  g_return_val_if_fail (TCPPort <= MAX_PORT_SIZE, 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..b390ef3
--- /dev/null
+++ b/gst/rtsp-server/gstwfdmessage.h
@@ -0,0 +1,618 @@
+/* GStreamer
+ * Copyright (C) 2015 Samsung Electronics Hyunjun Ko 
+ *
+ * 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-client-wfd.c b/gst/rtsp-server/rtsp-client-wfd.c
new file mode 100644
index 0000000..561a722
--- /dev/null
+++ b/gst/rtsp-server/rtsp-client-wfd.c
@@ -0,0 +1,2959 @@
+/* GStreamer
+ * Copyright (C) 2015 Samsung Electronics Hyunjun Ko 
+ *
+ * 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_KEEP_ALIVE_FAIL,
+  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);
+
+  gst_rtsp_client_wfd_signals[SIGNAL_WFD_KEEP_ALIVE_FAIL] =
+      g_signal_new ("wfd-keep-alive-fail", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+      G_STRUCT_OFFSET (GstRTSPWFDClientClass, wfd_keep_alive_fail), NULL, NULL,
+      g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
+
+  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);
+
+    g_signal_emit (client,
+        gst_rtsp_client_wfd_signals[SIGNAL_WFD_KEEP_ALIVE_FAIL], 0,
+        NULL);
+    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;
+}
+
+guint64
+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;
+}
+
+guint64
+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;
+}
+
+guint64
+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, guint64 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, guint64 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, guint64 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..baf54b9
--- /dev/null
+++ b/gst/rtsp-server/rtsp-client-wfd.h
@@ -0,0 +1,221 @@
+/* GStreamer
+ * Copyright (C) 2015 Samsung Electronics Hyunjun Ko 
+ *
+ * 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);
+  void     (*wfd_keep_alive_fail)         (GstRTSPWFDClient *client);
+
+  /*< 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);
+guint64   gst_rtsp_wfd_client_get_video_cea_resolution(GstRTSPWFDClient *client);
+guint64   gst_rtsp_wfd_client_get_video_vesa_resolution(GstRTSPWFDClient *client);
+guint64   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, guint64 res);
+void gst_rtsp_wfd_client_set_video_vesa_resolution(GstRTSPWFDClient *client, guint64 res);
+void gst_rtsp_wfd_client_set_video_hh_resolution(GstRTSPWFDClient *client, guint64 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
index 55f6259..13ccf36 100644
--- a/gst/rtsp-server/rtsp-client.c
+++ b/gst/rtsp-server/rtsp-client.c
@@ -155,6 +155,12 @@ 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);
 
@@ -180,6 +186,9 @@ gst_rtsp_client_class_init (GstRTSPClientClass * klass)
   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",
@@ -929,7 +938,7 @@ default_params_get (GstRTSPClient * client, GstRTSPContext * ctx)
 }
 
 static gboolean
-handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
+default_handle_get_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
 {
   GstRTSPResult res;
   guint8 *data;
@@ -966,7 +975,7 @@ bad_request:
 }
 
 static gboolean
-handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
+default_handle_set_param_request (GstRTSPClient * client, GstRTSPContext * ctx)
 {
   GstRTSPResult res;
   guint8 *data;
@@ -1833,6 +1842,8 @@ handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
   if (media == NULL)
     goto media_not_found_no_reply;
 
+  /* FIXME-WFD : wfd url problem */
+#if 0
   if (path[matched] == '\0') {
     if (gst_rtsp_media_n_streams (media) == 1) {
       stream = gst_rtsp_media_get_stream (media, 0);
@@ -1848,6 +1859,12 @@ handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
     /* find the stream now using the control part */
     stream = gst_rtsp_media_find_stream (media, control);
   }
+#else
+  control = g_strdup ("stream=0");
+
+  /* find the stream now using the control part */
+  stream = gst_rtsp_media_find_stream (media, control);
+#endif
 
   if (stream == NULL)
     goto stream_not_found;
@@ -1944,6 +1961,10 @@ handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
   /* create and serialize the server transport */
   st = make_server_transport (client, media, 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 */
@@ -2548,7 +2569,7 @@ unsuspend_failed:
 }
 
 static gboolean
-handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
+default_handle_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
 {
   GstRTSPMethod options;
   gchar *str;
@@ -2704,6 +2725,9 @@ handle_request (GstRTSPClient * client, GstRTSPMessage * request)
   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;
@@ -2828,7 +2852,7 @@ handle_request (GstRTSPClient * client, GstRTSPMessage * request)
   /* now see what is asked and dispatch to a dedicated handler */
   switch (method) {
     case GST_RTSP_OPTIONS:
-      handle_options_request (client, ctx);
+      klass->handle_options_request (client, ctx);
       break;
     case GST_RTSP_DESCRIBE:
       handle_describe_request (client, ctx);
@@ -2846,10 +2870,10 @@ handle_request (GstRTSPClient * client, GstRTSPMessage * request)
       handle_teardown_request (client, ctx);
       break;
     case GST_RTSP_SET_PARAMETER:
-      handle_set_param_request (client, ctx);
+      klass->handle_set_param_request (client, ctx);
       break;
     case GST_RTSP_GET_PARAMETER:
-      handle_get_param_request (client, ctx);
+      klass->handle_get_param_request (client, ctx);
       break;
     case GST_RTSP_ANNOUNCE:
       handle_announce_request (client, ctx);
diff --git a/gst/rtsp-server/rtsp-client.h b/gst/rtsp-server/rtsp-client.h
index 6b05448..9b6d879 100644
--- a/gst/rtsp-server/rtsp-client.h
+++ b/gst/rtsp-server/rtsp-client.h
@@ -103,6 +103,9 @@ struct _GstRTSPClientClass {
   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);
@@ -129,7 +132,7 @@ struct _GstRTSPClientClass {
   gchar*   (*check_requirements)      (GstRTSPClient *client, GstRTSPContext *ctx, gchar ** arr);
 
   /*< private >*/
-  gpointer _gst_reserved[GST_PADDING_LARGE-6];
+  gpointer _gst_reserved[GST_PADDING_LARGE-9];
 };
 
 GType                 gst_rtsp_client_get_type          (void);
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..f08a5c2
--- /dev/null
+++ b/gst/rtsp-server/rtsp-media-factory-wfd.c
@@ -0,0 +1,1200 @@
+/* GStreamer
+ * Copyright (C) 2015 Samsung Electronics Hyunjun Ko 
+ *
+ * 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_waylandsrc_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 waylandsrc as video source");
+
+  videosrc = gst_element_factory_make ("waylandsrc", "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, "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, 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_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;
+    case GST_WFD_VSRC_WAYLANDSRC:
+      if (!_rtsp_media_factory_wfd_create_waylandsrc_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..1751a22
--- /dev/null
+++ b/gst/rtsp-server/rtsp-media-factory-wfd.h
@@ -0,0 +1,131 @@
+/* GStreamer
+ * Copyright (C) 2015 Samsung Electronics Hyunjun Ko 
+ *
+ * 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,
+  GST_WFD_VSRC_WAYLANDSRC
+};
+
+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-server-wfd.c b/gst/rtsp-server/rtsp-server-wfd.c
new file mode 100644
index 0000000..4e72093
--- /dev/null
+++ b/gst/rtsp-server/rtsp-server-wfd.c
@@ -0,0 +1,332 @@
+/* GStreamer
+ * Copyright (C) 2015 Samsung Electronics Hyunjun Ko 
+ *
+ * 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..e718e62
--- /dev/null
+++ b/gst/rtsp-server/rtsp-server-wfd.h
@@ -0,0 +1,155 @@
+/* GStreamer
+ * Copyright (C) 2015 Samsung Electronics Hyunjun Ko 
+ *
+ * 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-stream.c b/gst/rtsp-server/rtsp-stream.c
index 1c4366e..4c1f3b7 100644
--- a/gst/rtsp-server/rtsp-stream.c
+++ b/gst/rtsp-server/rtsp-stream.c
@@ -1103,6 +1103,9 @@ again:
       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);
@@ -1280,10 +1283,15 @@ alloc_ports (GstRTSPStream * stream)
       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 (stream, 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;
 }
@@ -3154,6 +3162,21 @@ gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
   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
diff --git a/gst/rtsp-server/rtsp-stream.h b/gst/rtsp-server/rtsp-stream.h
index 0042f67..eca190c 100644
--- a/gst/rtsp-server/rtsp-stream.h
+++ b/gst/rtsp-server/rtsp-stream.h
@@ -155,6 +155,7 @@ gboolean          gst_rtsp_stream_query_stop       (GstRTSPStream * stream,
 
 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);
 void              gst_rtsp_stream_set_retransmission_time     (GstRTSPStream *stream, GstClockTime time);
 GstClockTime      gst_rtsp_stream_get_retransmission_time     (GstRTSPStream *stream);
 guint             gst_rtsp_stream_get_retransmission_pt       (GstRTSPStream * stream);
diff --git a/packaging/common.tar.bz2 b/packaging/common.tar.bz2
new file mode 100644
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diff --git a/packaging/gst-rtsp-server.spec b/packaging/gst-rtsp-server.spec
new file mode 100644
index 0000000..0474352
--- /dev/null
+++ b/packaging/gst-rtsp-server.spec
@@ -0,0 +1,76 @@
+Name:       gst-rtsp-server
+Summary:    Multimedia Framework Library
+Version:    1.6.1
+Release:    0
+Url:        http://gstreamer.freedesktop.org/
+Group:      System/Libraries
+License:    LGPL-2.0+
+Source:     http://gstreamer.freedesktop.org/src/gst-rtsp-server/gst-rtsp-server-%{version}.tar.xz
+Source100:  common.tar.bz2
+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 -n gst-rtsp-server-%{version}
+%setup -q -T -D -a 100 
+
+%build
+
+NOCONFIGURE=1 ./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/*
-- 
2.7.4


From b109694c1d78f9c7883aa934ce53221c96739a17 Mon Sep 17 00:00:00 2001
From: Hyunjun Ko 
Date: Thu, 3 Dec 2015 17:51:02 +0900
Subject: [PATCH 05/16] client: makes handle_play_request inheritable

Change-Id: I1a0449d7610d1d1019eee880ed18873c148f0d0c
---
 gst/rtsp-server/rtsp-client.c | 8 ++++++--
 gst/rtsp-server/rtsp-client.h | 3 ++-
 2 files changed, 8 insertions(+), 3 deletions(-)

diff --git a/gst/rtsp-server/rtsp-client.c b/gst/rtsp-server/rtsp-client.c
index 13ccf36..05ff274 100644
--- a/gst/rtsp-server/rtsp-client.c
+++ b/gst/rtsp-server/rtsp-client.c
@@ -161,6 +161,9 @@ static gboolean default_handle_set_param_request (GstRTSPClient * client,
     GstRTSPContext * ctx);
 static gboolean default_handle_get_param_request (GstRTSPClient * client,
     GstRTSPContext * ctx);
+static gboolean default_handle_play_request (GstRTSPClient * client,
+    GstRTSPContext * ctx);
+
 static void client_session_removed (GstRTSPSessionPool * pool,
     GstRTSPSession * session, GstRTSPClient * client);
 
@@ -189,6 +192,7 @@ gst_rtsp_client_class_init (GstRTSPClientClass * klass)
   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;
+  klass->handle_play_request = default_handle_play_request;
 
   g_object_class_install_property (gobject_class, PROP_SESSION_POOL,
       g_param_spec_object ("session-pool", "Session Pool",
@@ -1127,7 +1131,7 @@ make_base_url (GstRTSPClient * client, GstRTSPUrl * url, const gchar * path)
 }
 
 static gboolean
-handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
+default_handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
 {
   GstRTSPSession *session;
   GstRTSPClientClass *klass;
@@ -2861,7 +2865,7 @@ handle_request (GstRTSPClient * client, GstRTSPMessage * request)
       handle_setup_request (client, ctx);
       break;
     case GST_RTSP_PLAY:
-      handle_play_request (client, ctx);
+      klass->handle_play_request (client, ctx);
       break;
     case GST_RTSP_PAUSE:
       handle_pause_request (client, ctx);
diff --git a/gst/rtsp-server/rtsp-client.h b/gst/rtsp-server/rtsp-client.h
index 9b6d879..13fa2d1 100644
--- a/gst/rtsp-server/rtsp-client.h
+++ b/gst/rtsp-server/rtsp-client.h
@@ -106,6 +106,7 @@ struct _GstRTSPClientClass {
   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);
+  gboolean        (*handle_play_request) (GstRTSPClient * client, GstRTSPContext * ctx);
 
   /* signals */
   void     (*closed)                  (GstRTSPClient *client);
@@ -132,7 +133,7 @@ struct _GstRTSPClientClass {
   gchar*   (*check_requirements)      (GstRTSPClient *client, GstRTSPContext *ctx, gchar ** arr);
 
   /*< private >*/
-  gpointer _gst_reserved[GST_PADDING_LARGE-9];
+  gpointer _gst_reserved[GST_PADDING_LARGE-10];
 };
 
 GType                 gst_rtsp_client_get_type          (void);
-- 
2.7.4


From 2e221d5eb45afa3d4aea5bc3b18c9da3f7f72e78 Mon Sep 17 00:00:00 2001
From: Hyunjun Ko 
Date: Thu, 3 Dec 2015 17:51:38 +0900
Subject: [PATCH 06/16] stream: Remove one FIXME-WFD, which could be resolved
 for user to set address pool

Change-Id: Ibd48e6f585bf5be9be72ddda4b6a4fc75abc9103
---
 gst/rtsp-server/rtsp-stream.c | 3 ---
 1 file changed, 3 deletions(-)

diff --git a/gst/rtsp-server/rtsp-stream.c b/gst/rtsp-server/rtsp-stream.c
index 4c1f3b7..9805c1e 100644
--- a/gst/rtsp-server/rtsp-stream.c
+++ b/gst/rtsp-server/rtsp-stream.c
@@ -1103,9 +1103,6 @@ again:
       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);
-- 
2.7.4


From 267446a6e04437f16e56731e99d54625063a11d6 Mon Sep 17 00:00:00 2001
From: Hyunjun Ko 
Date: Thu, 3 Dec 2015 17:52:03 +0900
Subject: [PATCH 07/16] Fix svace defects

Change-Id: Ibc6aec1fd85e3d4d539f8e5fa207b39c73740b3b
---
 gst/rtsp-server/rtsp-client-wfd.c        | 12 ++++++------
 gst/rtsp-server/rtsp-media-factory-wfd.c |  3 ++-
 2 files changed, 8 insertions(+), 7 deletions(-)

diff --git a/gst/rtsp-server/rtsp-client-wfd.c b/gst/rtsp-server/rtsp-client-wfd.c
index 561a722..38cbae1 100644
--- a/gst/rtsp-server/rtsp-client-wfd.c
+++ b/gst/rtsp-server/rtsp-client-wfd.c
@@ -2717,7 +2717,7 @@ 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);
+  g_return_if_fail (priv != NULL);
 
   priv->cFreq = freq;
 }
@@ -2726,7 +2726,7 @@ 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);
+  g_return_if_fail (priv != NULL);
 
   priv->edid_supported = supported;
 }
@@ -2735,7 +2735,7 @@ 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);
+  g_return_if_fail (priv != NULL);
 
   priv->edid_hres = reso;
 }
@@ -2744,7 +2744,7 @@ 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);
+  g_return_if_fail (priv != NULL);
 
   priv->edid_vres = reso;
 }
@@ -2753,7 +2753,7 @@ 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);
+  g_return_if_fail (priv != NULL);
 
   priv->protection_enabled = enable;
 }
@@ -2761,7 +2761,7 @@ gst_rtsp_wfd_client_set_protection_enabled(GstRTSPWFDClient *client, gboolean en
 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_return_if_fail (priv != NULL);
 
   g_mutex_lock(&priv->keep_alive_lock);
   if (priv->keep_alive_flag == !(flag))
diff --git a/gst/rtsp-server/rtsp-media-factory-wfd.c b/gst/rtsp-server/rtsp-media-factory-wfd.c
index f08a5c2..0b7e76d 100644
--- a/gst/rtsp-server/rtsp-media-factory-wfd.c
+++ b/gst/rtsp-server/rtsp-media-factory-wfd.c
@@ -740,11 +740,12 @@ _rtsp_media_factory_wfd_create_camera_capture_bin (GstRTSPMediaFactoryWFD *
   }
 
   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;
   }
-  if (vcodec) g_free (vcodec);
 
   g_object_set (venc, "bitrate", priv->video_bitrate, NULL);
   g_object_set (venc, "byte-stream", 1, NULL);
-- 
2.7.4


From 735ba122b8a0ee5bc2a00b6ac7084ebe58ce8ed1 Mon Sep 17 00:00:00 2001
From: Hyunjun Ko 
Date: Thu, 10 Dec 2015 16:31:25 +0900
Subject: [PATCH 08/16] Remove FIXME-WFD temporary code

Change-Id: Ib59e88b3a089538239896a2714abc2c8c65aa596
---
 gst/rtsp-server/rtsp-client.c | 8 --------
 gst/rtsp-server/rtsp-stream.c | 5 -----
 2 files changed, 13 deletions(-)

diff --git a/gst/rtsp-server/rtsp-client.c b/gst/rtsp-server/rtsp-client.c
index 05ff274..4f3212d 100644
--- a/gst/rtsp-server/rtsp-client.c
+++ b/gst/rtsp-server/rtsp-client.c
@@ -1846,8 +1846,6 @@ handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
   if (media == NULL)
     goto media_not_found_no_reply;
 
-  /* FIXME-WFD : wfd url problem */
-#if 0
   if (path[matched] == '\0') {
     if (gst_rtsp_media_n_streams (media) == 1) {
       stream = gst_rtsp_media_get_stream (media, 0);
@@ -1863,12 +1861,6 @@ handle_setup_request (GstRTSPClient * client, GstRTSPContext * ctx)
     /* find the stream now using the control part */
     stream = gst_rtsp_media_find_stream (media, control);
   }
-#else
-  control = g_strdup ("stream=0");
-
-  /* find the stream now using the control part */
-  stream = gst_rtsp_media_find_stream (media, control);
-#endif
 
   if (stream == NULL)
     goto stream_not_found;
diff --git a/gst/rtsp-server/rtsp-stream.c b/gst/rtsp-server/rtsp-stream.c
index 9805c1e..a015346 100644
--- a/gst/rtsp-server/rtsp-stream.c
+++ b/gst/rtsp-server/rtsp-stream.c
@@ -1280,15 +1280,10 @@ alloc_ports (GstRTSPStream * stream)
       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 (stream, 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;
 }
-- 
2.7.4


From 8d671acd29a3420e775d9a255f6142c4b1337d64 Mon Sep 17 00:00:00 2001
From: Hyunjun Ko 
Date: Wed, 16 Dec 2015 10:09:11 +0900
Subject: [PATCH 09/16] Fix dereferencing null pointer problem

Change-Id: I2524622ca707ed8b554e6feef3171b29e90732c1
---
 gst/rtsp-server/gstwfdmessage.c | 16 ++++++++++------
 1 file changed, 10 insertions(+), 6 deletions(-)

diff --git a/gst/rtsp-server/gstwfdmessage.c b/gst/rtsp-server/gstwfdmessage.c
index b70e57d..3dbd6db 100755
--- a/gst/rtsp-server/gstwfdmessage.c
+++ b/gst/rtsp-server/gstwfdmessage.c
@@ -1808,7 +1808,7 @@ gst_wfd_message_get_contentprotection_type (GstWFDMessage * msg,
   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;
+    char *ptr[2] = {0, };
     if (!g_strcmp0 (msg->content_protection->hdcp2_spec->hdcpversion, "none")) {
       *hdcpversion = GST_WFD_HDCP_NONE;
       *TCPPort = 0;
@@ -1826,11 +1826,15 @@ gst_wfd_message_get_contentprotection_type (GstWFDMessage * msg,
       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;
+    if (msg->content_protection->hdcp2_spec->TCPPort) {
+      result = strtok_r (msg->content_protection->hdcp2_spec->TCPPort, "=", &ptr[0]);
+      while (result != NULL) {
+        result = strtok_r (NULL, "=", &ptr[1]);
+        *TCPPort = atoi (result);
+        break;
+      }
+    } else {
+      *TCPPort = 0;
     }
   } else
     *hdcpversion = GST_WFD_HDCP_NONE;
-- 
2.7.4


From 814ff75c36a19e6e5a54095309c3591343e3be5b Mon Sep 17 00:00:00 2001
From: Hyunjun Ko 
Date: Fri, 18 Dec 2015 15:56:02 +0900
Subject: [PATCH 10/16] Fix crash by misusing strtok_r

Change-Id: I7f4a4e6728efb4651abe076621d139329b4bbb8d
---
 gst/rtsp-server/gstwfdmessage.c | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

diff --git a/gst/rtsp-server/gstwfdmessage.c b/gst/rtsp-server/gstwfdmessage.c
index 3dbd6db..9aa873e 100755
--- a/gst/rtsp-server/gstwfdmessage.c
+++ b/gst/rtsp-server/gstwfdmessage.c
@@ -1808,7 +1808,7 @@ gst_wfd_message_get_contentprotection_type (GstWFDMessage * msg,
   g_return_val_if_fail (msg != NULL, GST_WFD_EINVAL);
   if (msg->content_protection && msg->content_protection->hdcp2_spec) {
     char *result = NULL;
-    char *ptr[2] = {0, };
+    char *ptr = NULL;
     if (!g_strcmp0 (msg->content_protection->hdcp2_spec->hdcpversion, "none")) {
       *hdcpversion = GST_WFD_HDCP_NONE;
       *TCPPort = 0;
@@ -1827,9 +1827,9 @@ gst_wfd_message_get_contentprotection_type (GstWFDMessage * msg,
     }
 
     if (msg->content_protection->hdcp2_spec->TCPPort) {
-      result = strtok_r (msg->content_protection->hdcp2_spec->TCPPort, "=", &ptr[0]);
-      while (result != NULL) {
-        result = strtok_r (NULL, "=", &ptr[1]);
+      result = strtok_r (msg->content_protection->hdcp2_spec->TCPPort, "=", &ptr);
+      while (result != NULL && ptr != NULL) {
+        result = strtok_r (NULL, "=", &ptr);
         *TCPPort = atoi (result);
         break;
       }
-- 
2.7.4


From bbf1ce4729d6429add27d772cbc7a3c894e6c0c2 Mon Sep 17 00:00:00 2001
From: Hyunjun Ko 
Date: Mon, 18 Jan 2016 16:56:39 +0900
Subject: [PATCH 11/16] Add signal to notify when M7 done

Change-Id: I56802c4cb7c6fee8407a689d51af4cc443887ef0
---
 gst/rtsp-server/rtsp-client-wfd.c | 11 ++++++++++-
 gst/rtsp-server/rtsp-client-wfd.h |  1 +
 2 files changed, 11 insertions(+), 1 deletion(-)

diff --git a/gst/rtsp-server/rtsp-client-wfd.c b/gst/rtsp-server/rtsp-client-wfd.c
index 38cbae1..8566638 100644
--- a/gst/rtsp-server/rtsp-client-wfd.c
+++ b/gst/rtsp-server/rtsp-client-wfd.c
@@ -148,6 +148,7 @@ enum
   SIGNAL_WFD_OPTIONS_REQUEST,
   SIGNAL_WFD_GET_PARAMETER_REQUEST,
   SIGNAL_WFD_KEEP_ALIVE_FAIL,
+  SIGNAL_WFD_PLAYING_DONE,
   SIGNAL_WFD_LAST
 };
 
@@ -250,6 +251,11 @@ gst_rtsp_wfd_client_class_init (GstRTSPWFDClientClass * klass)
       G_STRUCT_OFFSET (GstRTSPWFDClientClass, wfd_keep_alive_fail), NULL, NULL,
       g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
 
+  gst_rtsp_client_wfd_signals[SIGNAL_WFD_PLAYING_DONE] =
+      g_signal_new ("wfd-playing-done", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
+      G_STRUCT_OFFSET (GstRTSPWFDClientClass, wfd_playing_done), NULL, NULL,
+      g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
+
   klass->wfd_options_request = wfd_options_request_done;
   klass->wfd_get_param_request = wfd_get_param_request_done;
 
@@ -944,8 +950,11 @@ handle_wfd_play (GstRTSPClient * client, GstRTSPContext * ctx)
   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);
+
+  g_signal_emit (client,
+      gst_rtsp_client_wfd_signals[SIGNAL_WFD_PLAYING_DONE], 0, NULL);
 }
 
 static void
diff --git a/gst/rtsp-server/rtsp-client-wfd.h b/gst/rtsp-server/rtsp-client-wfd.h
index baf54b9..67faab7 100644
--- a/gst/rtsp-server/rtsp-client-wfd.h
+++ b/gst/rtsp-server/rtsp-client-wfd.h
@@ -112,6 +112,7 @@ struct _GstRTSPWFDClientClass {
   void     (*wfd_options_request)         (GstRTSPWFDClient *client, GstRTSPContext *ctx);
   void     (*wfd_get_param_request)       (GstRTSPWFDClient *client, GstRTSPContext *ctx);
   void     (*wfd_keep_alive_fail)         (GstRTSPWFDClient *client);
+  void     (*wfd_playing_done)            (GstRTSPWFDClient *client);
 
   /*< private >*/
   gpointer _gst_reserved[GST_PADDING_LARGE];
-- 
2.7.4


From 74fe786d124dfdc924d9a0dedc43540820d2dea6 Mon Sep 17 00:00:00 2001
From: "Hyunsoo, Park" 
Date: Tue, 2 Feb 2016 17:02:00 +0900
Subject: [PATCH 12/16] Add new property to pulsesrc instead of audio device

Change-Id: I59a28a8f3b1a39b84e5c55fb559cff3faf77838d
Signed-off-by: Hyunsoo, Park 
---
 gst/rtsp-server/rtsp-media-factory-wfd.c | 8 ++++++--
 1 file changed, 6 insertions(+), 2 deletions(-)

diff --git a/gst/rtsp-server/rtsp-media-factory-wfd.c b/gst/rtsp-server/rtsp-media-factory-wfd.c
index 0b7e76d..8cca5ec 100644
--- a/gst/rtsp-server/rtsp-media-factory-wfd.c
+++ b/gst/rtsp-server/rtsp-media-factory-wfd.c
@@ -337,6 +337,7 @@ _rtsp_media_factory_wfd_create_audio_capture_bin (GstRTSPMediaFactoryWFD *
   GstElement *audio_convert = NULL;
   GstElement *aqueue = NULL;
   GstRTSPMediaFactoryWFDPrivate *priv = NULL;
+  GstStructure *audio_properties_name = NULL;
 
   guint channels = 0;
   gboolean is_enc_req = TRUE;
@@ -360,7 +361,9 @@ _rtsp_media_factory_wfd_create_audio_capture_bin (GstRTSPMediaFactoryWFD *
   GST_INFO_OBJECT (factory, "audio_do_timestamp  : %d",
       priv->audio_do_timestamp);
 
-  g_object_set (audiosrc, "device", priv->audio_device, NULL);
+  audio_properties_name = gst_structure_new_from_string(priv->audio_device);
+
+  g_object_set (audiosrc, "stream-properties", audio_properties_name, NULL);
   g_object_set (audiosrc, "buffer-time", (gint64) priv->audio_buffer_time,
       NULL);
   g_object_set (audiosrc, "latency-time", (gint64) priv->audio_latency_time,
@@ -499,11 +502,12 @@ _rtsp_media_factory_wfd_create_audio_capture_bin (GstRTSPMediaFactoryWFD *
 
   priv->audio_queue = aqueue;
   if (acodec) g_free (acodec);
-
+  if (gst_structure_free) gst_structure_free(audio_properties_name);
   return TRUE;
 
 create_error:
   if (acodec) g_free (acodec);
+  if (gst_structure_free) gst_structure_free(audio_properties_name);
   return FALSE;
 }
 
-- 
2.7.4


From 0919d4bbaefd448c7cd988da6333639725e0763f Mon Sep 17 00:00:00 2001
From: "Hyunsoo, Park" 
Date: Thu, 3 Mar 2016 15:10:27 +0900
Subject: [PATCH 13/16] Fix warning of audio device property in free process

Change-Id: I41e4e7ab4b6d4d40ef90c65a377342e8efafa2e2
Signed-off-by: Hyunsoo, Park 
---
 gst/rtsp-server/rtsp-media-factory-wfd.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/gst/rtsp-server/rtsp-media-factory-wfd.c b/gst/rtsp-server/rtsp-media-factory-wfd.c
index 8cca5ec..60371b1 100644
--- a/gst/rtsp-server/rtsp-media-factory-wfd.c
+++ b/gst/rtsp-server/rtsp-media-factory-wfd.c
@@ -502,12 +502,12 @@ _rtsp_media_factory_wfd_create_audio_capture_bin (GstRTSPMediaFactoryWFD *
 
   priv->audio_queue = aqueue;
   if (acodec) g_free (acodec);
-  if (gst_structure_free) gst_structure_free(audio_properties_name);
+  if (audio_properties_name) gst_structure_free(audio_properties_name);
   return TRUE;
 
 create_error:
   if (acodec) g_free (acodec);
-  if (gst_structure_free) gst_structure_free(audio_properties_name);
+  if (audio_properties_name) gst_structure_free(audio_properties_name);
   return FALSE;
 }
 
-- 
2.7.4


From 9a09312feff357fc7a13e402765897a028df6063 Mon Sep 17 00:00:00 2001
From: SeokHoon Lee 
Date: Thu, 28 Apr 2016 11:31:55 +0900
Subject: [PATCH 14/16] modify audio counting algorithm from vd

Signed-off-by: SeokHoon Lee 
Change-Id: I90458b60a0508896d3f2289c9ef7dbfa4a0f9ee1
---
 gst/rtsp-server/gstwfdmessage.c | 11 +++++++----
 packaging/gst-rtsp-server.spec  |  2 +-
 2 files changed, 8 insertions(+), 5 deletions(-)
 mode change 100755 => 100644 gst/rtsp-server/gstwfdmessage.c

diff --git a/gst/rtsp-server/gstwfdmessage.c b/gst/rtsp-server/gstwfdmessage.c
old mode 100755
new mode 100644
index 9aa873e..fc81a9c
--- a/gst/rtsp-server/gstwfdmessage.c
+++ b/gst/rtsp-server/gstwfdmessage.c
@@ -1391,7 +1391,6 @@ 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;
 
@@ -1401,10 +1400,14 @@ gst_wfd_message_set_supported_audio_format (GstWFDMessage * msg,
     msg->audio_codecs = g_new0 (GstWFDAudioCodeclist, 1);
 
   if (a_codec != GST_WFD_AUDIO_UNKNOWN) {
-    while (temp) {
+
+    if (a_codec & GST_WFD_AUDIO_LPCM)
       msg->audio_codecs->count++;
-      temp >>= 1;
-    }
+    if (a_codec & GST_WFD_AUDIO_AAC)
+      msg->audio_codecs->count++;
+    if (a_codec & GST_WFD_AUDIO_AC3)
+      msg->audio_codecs->count++;
+
     msg->audio_codecs->list =
         g_new0 (GstWFDAudioCodec, msg->audio_codecs->count);
     for (; i < msg->audio_codecs->count; i++) {
diff --git a/packaging/gst-rtsp-server.spec b/packaging/gst-rtsp-server.spec
index 0474352..b0e3187 100644
--- a/packaging/gst-rtsp-server.spec
+++ b/packaging/gst-rtsp-server.spec
@@ -1,7 +1,7 @@
 Name:       gst-rtsp-server
 Summary:    Multimedia Framework Library
 Version:    1.6.1
-Release:    0
+Release:    1
 Url:        http://gstreamer.freedesktop.org/
 Group:      System/Libraries
 License:    LGPL-2.0+
-- 
2.7.4


From a214249b54e3424b9e0651c1acdb8c390e4e86c2 Mon Sep 17 00:00:00 2001
From: SeokHoon Lee 
Date: Tue, 3 May 2016 13:08:27 +0900
Subject: [PATCH 15/16] function name change.   prepare_request ->
 gst_prepare_request,   send_request -> gst_send_request.

Signed-off-by: SeokHoon Lee 
Change-Id: I5a149f92dc999ace6b14032656ac3760eb5a061b
---
 gst/rtsp-server/rtsp-client-wfd.c | 31 ++++++++++++-------------------
 gst/rtsp-server/rtsp-client-wfd.h |  4 ++++
 2 files changed, 16 insertions(+), 19 deletions(-)

diff --git a/gst/rtsp-server/rtsp-client-wfd.c b/gst/rtsp-server/rtsp-client-wfd.c
index 8566638..6833641 100644
--- a/gst/rtsp-server/rtsp-client-wfd.c
+++ b/gst/rtsp-server/rtsp-client-wfd.c
@@ -187,13 +187,6 @@ static gboolean wfd_configure_client_media (GstRTSPClient * client, GstRTSPMedia
 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);
@@ -1752,7 +1745,7 @@ error:
 }
 
 /**
-* prepare_request:
+* gst_prepare_request:
 * @client: client object
 * @request : requst message to be prepared
 * @method : RTSP method of the request
@@ -1765,7 +1758,7 @@ error:
 * Returns: a #GstRTSPResult.
 */
 GstRTSPResult
-prepare_request (GstRTSPWFDClient * client, GstRTSPMessage * request,
+gst_prepare_request (GstRTSPWFDClient * client, GstRTSPMessage * request,
     GstRTSPMethod method, gchar * url)
 {
   GstRTSPResult res = GST_RTSP_OK;
@@ -2064,7 +2057,7 @@ error:
 
 
 void
-send_request (GstRTSPWFDClient * client, GstRTSPSession * session,
+gst_send_request (GstRTSPWFDClient * client, GstRTSPSession * session,
     GstRTSPMessage * request)
 {
   GstRTSPResult res = GST_RTSP_OK;
@@ -2188,7 +2181,7 @@ handle_M1_message (GstRTSPWFDClient * client)
   GstRTSPResult res = GST_RTSP_OK;
   GstRTSPMessage request = { 0 };
 
-  res = prepare_request (client, &request, GST_RTSP_OPTIONS, (gchar *) "*");
+  res = gst_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;
@@ -2196,7 +2189,7 @@ handle_M1_message (GstRTSPWFDClient * client)
 
   GST_DEBUG_OBJECT (client, "Sending M1 request.. (OPTIONS request)");
 
-  send_request (client, NULL, &request);
+  gst_send_request (client, NULL, &request);
 
   return res;
 }
@@ -2237,7 +2230,7 @@ handle_M3_message (GstRTSPWFDClient * client)
     goto error;
   }
 
-  res = prepare_request (client, &request, GST_RTSP_GET_PARAMETER, url_str);
+  res = gst_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;
@@ -2245,7 +2238,7 @@ handle_M3_message (GstRTSPWFDClient * client)
 
   GST_DEBUG_OBJECT (client, "Sending GET_PARAMETER request message (M3)...");
 
-  send_request (client, NULL, &request);
+  gst_send_request (client, NULL, &request);
 
   return res;
 
@@ -2279,7 +2272,7 @@ handle_M4_message (GstRTSPWFDClient * client)
     goto error;
   }
 
-  res = prepare_request (client, &request, GST_RTSP_SET_PARAMETER, url_str);
+  res = gst_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;
@@ -2287,7 +2280,7 @@ handle_M4_message (GstRTSPWFDClient * client)
 
   GST_DEBUG_OBJECT (client, "Sending SET_PARAMETER request message (M4)...");
 
-  send_request (client, NULL, &request);
+  gst_send_request (client, NULL, &request);
 
   return res;
 
@@ -2330,7 +2323,7 @@ gst_rtsp_wfd_client_trigger_request (GstRTSPWFDClient * client,
 
   GST_DEBUG_OBJECT (client, "Sending trigger request message...: %d", type);
 
-  send_request (client, NULL, &request);
+  gst_send_request (client, NULL, &request);
 
   return res;
 
@@ -2409,7 +2402,7 @@ wfd_ckeck_keep_alive_response (gpointer userdata)
 }
 
 /*Sending keep_alive (M16) message.
-  Without calling prepare_request function.*/
+  Without calling gst_prepare_request function.*/
 static GstRTSPResult
 handle_M16_message (GstRTSPWFDClient * client)
 {
@@ -2425,7 +2418,7 @@ handle_M16_message (GstRTSPWFDClient * client)
     return FALSE;
   }
 
-  send_request (client, NULL, &request);
+  gst_send_request (client, NULL, &request);
   return GST_RTSP_OK;
 }
 
diff --git a/gst/rtsp-server/rtsp-client-wfd.h b/gst/rtsp-server/rtsp-client-wfd.h
index 67faab7..f9a836e 100644
--- a/gst/rtsp-server/rtsp-client-wfd.h
+++ b/gst/rtsp-server/rtsp-client-wfd.h
@@ -135,6 +135,10 @@ 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);
+GstRTSPResult         gst_prepare_request (GstRTSPWFDClient * client,
+                          GstRTSPMessage * request, GstRTSPMethod method, gchar * url);
+void                  gst_send_request (GstRTSPWFDClient * client,
+                          GstRTSPSession * session, GstRTSPMessage * request);
 
 guint     gst_rtsp_wfd_client_get_audio_codec(GstRTSPWFDClient *client);
 guint     gst_rtsp_wfd_client_get_audio_freq(GstRTSPWFDClient *client);
-- 
2.7.4


From f1b4c128effb154b75c3e3e96767c4da29871085 Mon Sep 17 00:00:00 2001
From: "Hyunsoo, Park" 
Date: Wed, 4 May 2016 11:08:09 +0900
Subject: [PATCH 16/16] Add preventing memory leak code for svace

Change-Id: I29a10d9976c8f6f644620a6cb1b486ae6e9a7073
Signed-off-by: Hyunsoo, Park 
---
 gst/rtsp-server/rtsp-client-wfd.c | 8 ++++++++
 1 file changed, 8 insertions(+)

diff --git a/gst/rtsp-server/rtsp-client-wfd.c b/gst/rtsp-server/rtsp-client-wfd.c
index 6833641..89e4704 100644
--- a/gst/rtsp-server/rtsp-client-wfd.c
+++ b/gst/rtsp-server/rtsp-client-wfd.c
@@ -1161,6 +1161,8 @@ handle_wfd_options_request (GstRTSPClient * client, GstRTSPContext * ctx)
 
   gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_PUBLIC, str);
   g_free (str);
+  g_free (tmp);
+
   str = NULL;
 
   res =
@@ -1773,6 +1775,9 @@ gst_prepare_request (GstRTSPWFDClient * client, GstRTSPMessage * request,
 
   /* initialize the request */
   res = gst_rtsp_message_init_request (request, method, url);
+  if (method == GST_RTSP_GET_PARAMETER || GST_RTSP_SET_PARAMETER) {
+    g_free(url);
+  }
   if (res < 0) {
     GST_ERROR ("init request failed");
     return res;
@@ -2138,6 +2143,7 @@ prepare_response (GstRTSPWFDClient * client, GstRTSPMessage * request,
 
       gst_rtsp_message_add_header (response, GST_RTSP_HDR_PUBLIC, str);
       g_free (str);
+      g_free (tmp);
       str = NULL;
       res =
           gst_rtsp_message_get_header (request, GST_RTSP_HDR_USER_AGENT,
@@ -2415,10 +2421,12 @@ handle_M16_message (GstRTSPWFDClient * client)
   res = gst_rtsp_message_init_request (&request, GST_RTSP_GET_PARAMETER, url_str);
   if (res < 0) {
     GST_ERROR ("init request failed");
+    g_free(url_str);
     return FALSE;
   }
 
   gst_send_request (client, NULL, &request);
+  g_free(url_str);
   return GST_RTSP_OK;
 }
 
-- 
2.7.4