+++ /dev/null
-[general]
-upstream_branch = tizen
-upstream_tag = ${upstreamversion}
+++ /dev/null
-Wim Taymans <wim.taymans@collabora.co.uk>
-Hyunjun Ko <zzoon.ko@samsung.com>
+++ /dev/null
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+++ /dev/null
- GNU LIBRARY GENERAL PUBLIC LICENSE
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- Ty Coon, President of Vice
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-That's all there is to it!
+++ /dev/null
-=== release 1.4.5 ===
-
-2014-12-18 Sebastian Dröge <slomo@coaxion.net>
-
- * configure.ac:
- releasing 1.4.5
-
-2014-10-31 23:01:53 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
-
- * 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 <sebastian@centricular.com>
-
- * ChangeLog:
- * NEWS:
- * RELEASE:
- * configure.ac:
- * gst-rtsp-server.doap:
- Release 1.4.4
-
-2014-10-29 21:01:39 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
-
- * 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 <sebastian@centricular.com>
-
- * 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 <sebastian@centricular.com>
-
- * ChangeLog:
- * NEWS:
- * RELEASE:
- * configure.ac:
- * gst-rtsp-server.doap:
- Release 1.4.2
-
-2014-09-08 09:26:23 +0200 Srimanta Panda <srimanta@axis.com>
-
- * 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 <goranjn@axis.com>
-
- * 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 <sebastian@centricular.com>
-
- * ChangeLog:
- * NEWS:
- * RELEASE:
- * configure.ac:
- * gst-rtsp-server.doap:
- Release 1.4.1
-
-2014-08-05 16:12:19 +0200 Sebastian Dröge <sebastian@centricular.com>
-
- * 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 <arun@accosted.net>
-
- * 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 <sebastian@centricular.com>
-
- * ChangeLog:
- * NEWS:
- * RELEASE:
- * configure.ac:
- * gst-rtsp-server.doap:
- Release 1.4.0
-
-2014-07-16 20:39:42 +0900 Hyunjun Ko <zzoonis@gmail.com>
-
- * 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 <sebastian@centricular.com>
-
- * ChangeLog:
- * NEWS:
- * RELEASE:
- * configure.ac:
- * gst-rtsp-server.doap:
- Release 1.3.91
-
-2014-07-10 17:37:45 +0200 Wim Taymans <wtaymans@redhat.com>
-
- * docs/libs/gst-rtsp-server-sections.txt:
- docs: update docs
-
-2014-07-10 17:10:06 +0200 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-server.c:
- server: implement client REMOVE filter
-
-2014-07-10 17:05:13 +0200 Wim Taymans <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-session-pool.c:
- * gst/rtsp-server/rtsp-session.c:
- * gst/rtsp-server/rtsp-stream.c:
- 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 <ognyan@axis.com>
-
- * 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 <ognyan@axis.com>
-
- * tests/check/gst/client.c:
- client tests: send teardown to cleanup session
-
-2014-07-09 14:17:46 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * tests/check/gst/rtspserver.c:
- server tests: send teardown to cleanup session
-
-2014-07-09 15:01:31 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <tim@centricular.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-client.c:
- client: fix build
-
-2014-07-01 14:41:14 +0200 Wim Taymans <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <ognyan.tonchev at axis.com>
- See https://bugzilla.gnome.org/show_bug.cgi?id=732226
-
-2014-06-30 15:14:34 +0200 Wim Taymans <wtaymans@redhat.com>
-
- * 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 <evan@nemerson.com>
-
- * 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 <sebastian@centricular.com>
-
- * ChangeLog:
- * NEWS:
- * RELEASE:
- * configure.ac:
- * gst-rtsp-server.doap:
- Release 1.3.90
-
-2014-06-27 16:54:22 +0200 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: crypto can be NULL
-
-2014-06-11 16:42:08 -0700 Evan Nemerson <evan@nemerson.com>
-
- * 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 <evan@nemerson.com>
-
- * 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 <evan@nemerson.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-context.h:
- context: add type macro
-
-2014-06-24 09:34:50 +0200 Wim Taymans <wtaymans@redhat.com>
-
- * 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 <sebastian@centricular.com>
-
- * configure.ac:
- Back to development
-
-=== release 1.3.3 ===
-
-2014-06-22 19:36:14 +0200 Sebastian Dröge <sebastian@centricular.com>
-
- * ChangeLog:
- * NEWS:
- * RELEASE:
- * configure.ac:
- * gst-rtsp-server.doap:
- Release 1.3.3
-
-2014-05-20 14:48:37 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
-
- * 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 <otonchev@gmail.com>
-
- * 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 <ognyan@axis.com>
-
- * 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 <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- client: Destroy the rtsp watch after connection close
-
-2014-06-13 16:46:06 +0200 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-media.c:
- media: fix confusing comment
-
-2014-05-27 12:36:52 +0200 Göran Jönsson <goranjn@axis.com>
-
- * 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 <sebastian@centricular.com>
-
- * configure.ac:
- Back to development
-
-=== release 1.3.2 ===
-
-2014-05-21 13:06:36 +0200 Sebastian Dröge <sebastian@centricular.com>
-
- * ChangeLog:
- * NEWS:
- * RELEASE:
- * common:
- * configure.ac:
- * gst-rtsp-server.doap:
- Release 1.3.2
-
-2014-05-21 10:54:05 +0200 Sebastian Dröge <sebastian@centricular.com>
-
- * common:
- Automatic update of common submodule
- From 211fa5f to 1f5d3c3
-
-2014-05-20 15:57:30 +0200 Wim Taymans <wtaymans@redhat.com>
-
- * 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 <aleix@oblong.com>
-
- * 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 <ognyan@axis.com>
-
- * 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 <aleix@oblong.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <tim@centricular.com>
-
- * 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 <tim@centricular.com>
-
- * 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 <sebastian@centricular.com>
-
- * configure.ac:
- Back to development
-
-=== release 1.3.1 ===
-
-2014-05-03 18:40:24 +0200 Sebastian Dröge <sebastian@centricular.com>
-
- * ChangeLog:
- * NEWS:
- * RELEASE:
- * configure.ac:
- * gst-rtsp-server.doap:
- Release 1.3.1
-
-2014-05-03 10:18:00 +0200 Sebastian Dröge <sebastian@centricular.com>
-
- * common:
- Automatic update of common submodule
- From bcb1518 to 211fa5f
-
-2014-05-02 19:58:15 +0100 Tim-Philipp Müller <tim@centricular.com>
-
- * .gitignore:
- Update .gitignore
-
-2014-05-02 19:57:23 +0100 Tim-Philipp Müller <tim@centricular.com>
-
- * tests/check/gst/sessionmedia.c:
- tests: fix memory leak in sessionmedia unit test
-
-2014-05-01 06:17:06 +0200 Wim Taymans <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-client.c:
- client: fix typo in comment
-
-2014-04-14 15:17:14 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <linusp.svensson@gmail.com>
-
- * 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 <ognyan@axis.com>
-
- * 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 <ognyan@axis.com>
-
- * 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 <ognyan@axis.com>
-
- * 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 <vincent.penquerch@collabora.co.uk>
-
- * 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 <goranjn@axis.com>
-
- * 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 <ognyan@axis.com>
-
- * 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 <ognyan@axis.com>
-
- * 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 <ognyan@axis.com>
-
- * 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 <linussn@axis.com>
-
- * tests/check/gst/media.c:
- media test: cleanup the thread pool in tests
-
-2014-04-01 13:16:26 +0200 Linus Svensson <linussn@axis.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-client.c:
- client: cleanup error paths
-
-2014-04-02 12:27:24 +0200 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-media.c:
- media: fix docs
-
-2014-03-25 12:42:39 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * examples/test-video.c:
- test: print an error on failure
-
-2014-03-13 17:35:21 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * 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 <sebras@hotmail.com>
-
- * 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 <sebras@hotmail.com>
-
- * 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 <sebras@hotmail.com>
-
- * 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 <sebras@hotmail.com>
-
- 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 <sebras@hotmail.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <davidsf@axis.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <sebras@hotmail.com>
-
- * .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 <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: release some locks in error cases
-
-2014-03-02 05:12:10 +0100 Sebastian Rasmussen <sebras@hotmail.com>
-
- 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 <sebastian@centricular.com>
-
- * common:
- Automatic update of common submodule
- From fe1672e to bcb1518
-
-2014-02-26 22:15:51 +0100 Stefan Sauer <ensonic@users.sf.net>
-
- * common:
- Automatic update of common submodule
- From 1a07da9 to fe1672e
-
-2014-02-25 15:13:40 +0000 Tim-Philipp Müller <tim@centricular.com>
-
- * examples/Makefile.am:
- examples: use LDADD for libs instead of LDFLAGS
-
-2014-02-25 14:42:09 +0000 Tim-Philipp Müller <tim@centricular.com>
-
- * configure.ac:
- configure: make sure releases are in .doap file
-
-2014-02-25 14:11:00 +0000 Tim-Philipp Müller <tim@centricular.com>
-
- * examples/test-cgroups.c:
- examples: test-cgroups: don't put code with side effects into g_assert()
- The g_assert() might get compiled out with the right
- compiler/preprocessor flags.
-
-2014-02-25 14:07:50 +0000 Tim-Philipp Müller <tim@centricular.com>
-
- * examples/.gitignore:
- examples: add cgroup test binary to .gitignore
-
-2014-02-25 14:06:47 +0000 Tim-Philipp Müller <tim@centricular.com>
-
- * examples/test-cgroups.c:
- examples: fix cgroup test build
- Fixes build failure caused by compiler warning:
- test-cgroups.c:82:35: error: no previous prototype for ‘gst_rtsp_cgroup_pool_get_type’ [-Werror=missing-prototypes]
-
-2014-02-21 16:46:45 +0000 Tim-Philipp Müller <tim@centricular.com>
-
- * .gitignore:
- .gitignore: ignore temp files created in the course of 'make check'
-
-2014-02-18 09:44:34 +0100 Branko Subasic <branko@axis.com>
-
- * gst/rtsp-server/rtsp-media.c:
- rtsp-media: don't loose frames handling new PLAY request
- If client supplied a range check if the range specifies the start point.
- If not, then do an accurate seek to the current position. If a start
- point was specified do do a key unit seek to make sure the streaming
- starts with decodeable frames.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724611
-
-2014-02-18 16:58:45 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * 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 <wtaymans@redhat.com>
-
- * 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 <goranjn@axis.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: set ttl-mc before adding the socket
- Set ttl-mc before adding the socket. Otherwise the value ttl-mc will
- never be set on socket.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724531
-
-2014-02-11 14:20:39 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
-
- * gst/rtsp-server/rtsp-media.c:
- media: stop thread if media is already prepared
- in gst_rtsp_media_prepare() the thread is not used if media is already
- prepared (e.g. media shared) so we want to stop the thread. otherwise, a
- leak occurs.
- https://bugzilla.gnome.org/show_bug.cgi?id=724182
-
-2014-02-09 10:52:29 +0100 Sebastian Dröge <sebastian@centricular.com>
-
- * Makefile.am:
- build: Ship gst-rtsp-server.doap file
-
-2014-02-09 10:47:09 +0100 Sebastian Dröge <sebastian@centricular.com>
-
- * tests/check/gst/rtspserver.c:
- tests: Fix another compiler warning with gcc
-
-2014-02-09 10:45:28 +0100 Sebastian Dröge <sebastian@centricular.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-mount-points.c:
- * gst/rtsp-server/rtsp-stream.c:
- * tests/check/gst/client.c:
- rtsp-server: Fix lots of compiler warnings with clang
-
-2014-02-09 10:41:14 +0100 Sebastian Dröge <sebastian@centricular.com>
-
- * configure.ac:
- * gst-rtsp-server.doap:
- * tests/Makefile.am:
- configure: Synchronise with the configure scripts of the other modules
-
-2014-02-09 10:25:44 +0100 Sebastian Dröge <sebastian@centricular.com>
-
- * configure.ac:
- configure: Update version to 1.3.0.1 and require GStreamer 1.3.0
-
-2014-02-09 10:19:50 +0100 Sebastian Dröge <sebastian@centricular.com>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-stream.c:
- Revert "rtsp-server: support build against last stable release"
- This reverts commit 099a10f61f11413ad0ada8ee0b7b7ad1210b1b2f.
- Let us require 1.2.3 now, which is going to be released in a few
- minutes.
-
-2014-02-07 16:39:49 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-stream-transport.c:
- session: improve RTP-Info
- Ignore streams that can't generate RTP-Info instead of failing.
- Don't return the empty string when all streams are unconfigured but
- return NULL so that we don't generate and empty RTP-Info header.
- Improve docs a little.
-
-2014-02-03 22:41:48 +0200 Andrey Utkin <andrey.krieger.utkin@gmail.com>
-
- * gst/rtsp-server/rtsp-session-media.c:
- Don't free rtpinfo GString when it is NULL
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
-
-2014-02-06 09:48:05 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-media.c:
- media: only set keyframe flag when modifying start
- Only set the keyframe flag when we modify the start position. The
- keyframe flag should probably be ignored when no change is requested but
- until we can claim this is all documented properly and all demuxer
- implement this, avoid setting the flag.
- See also https://bugzilla.gnome.org/show_bug.cgi?id=723075
-
-2014-02-06 09:03:50 +0100 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-thread-pool.c:
- thread-pool: Unref source after mainloop has quit to avoid races in GLib
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723741
-
-2014-02-04 16:27:12 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: handle NULL seqnum and rtptime arguments
-
-2014-01-31 15:02:22 +0100 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-thread-pool.c:
- * tests/check/gst/threadpool.c:
- thread-pool: Unref reused threads in gst_rtsp_thread_stop()
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723519
-
-2014-02-04 10:14:45 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: add fallback for missing stats property
- Use a fallback when the payloader does not have a stats property
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=723554
-
-2014-01-30 10:45:56 +0100 Edward Hervey <bilboed@bilboed.com>
-
- * common:
- Automatic update of common submodule
- From f7bc1c3 to 1a07da9
-
-2014-01-28 14:51:26 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: don't leak stats structure
- Don't leak the stats structure and deal with NULL stats.
-
-2014-01-22 22:03:14 +0100 Sebastian Rasmussen <sebrn@axis.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: Get rtpinfo properties atomically from payloader
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=722844
-
-2014-01-21 14:46:47 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-media.c:
- media: refactor state change functions and signals
- Make functions to set the target state and the pipeline state and emit
- the signals from those functions.
-
-2014-01-21 12:01:25 +0100 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- media: add signal to notify of pending state changes
-
-2014-01-12 16:55:21 +0000 Tim-Philipp Müller <tim@centricular.com>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-stream.c:
- rtsp-server: support build against last stable release
- Until 1.2.3 is out with the new get_type function and we
- can require that.
-
-2014-01-07 15:28:05 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: fix compilation
-
-2014-01-07 12:21:09 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- stream: add property to configure profiles
-
-2014-01-07 12:28:47 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-client.c:
- client: let stream check supported transport
- Delegate the check if a transport is allowed to the stream.
- See https://bugzilla.gnome.org/show_bug.cgi?id=720696
-
-2014-01-07 12:14:15 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- stream: add method to check supported transport
- Add a method to check if a transport is supported
-
-2013-12-27 13:11:45 +0100 Sebastian Dröge <sebastian@centricular.com>
-
- * configure.ac:
- configure.ac: Only check for gstreamer-check, not check
- We include check in gstreamer-check since quite some time now.
-
-2013-12-26 17:02:50 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-stream-transport.c:
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- stream: return clock-rate from get_rtpinfo
- And use it to correct the rtptime to the requested start-time.
- See https://bugzilla.gnome.org/show_bug.cgi?id=712198
-
-2013-12-26 16:28:59 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-stream-transport.c:
- * gst/rtsp-server/rtsp-stream-transport.h:
- session-media: calculate start-time
-
-2013-12-26 14:43:35 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-stream-transport.c:
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- stream: also return the running-time
- Return the running-time in the rtpinfo as well.
-
-2013-12-26 15:41:14 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session-media.h:
- * gst/rtsp-server/rtsp-stream-transport.c:
- * gst/rtsp-server/rtsp-stream-transport.h:
- session-media: let the session-media make the RTPInfo
- Add method to create the RTPInfo for a stream-transport.
- Add method to create the RTPInfo for all stream-transports in a
- session-media.
- Use the session-media RTPInfo code in client. This allows us to refactor
- another method to link the TCP callbacks.
-
-2013-12-20 16:39:07 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
-
- mount-points: sort sequence before g_sequence_lookup
- * gst/rtsp-server/rtsp-mount-points.c (gst_rtsp_mount_points_remove_factory):
- sort sequence if dirty, otherwise lookup will fail.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720855
-
-2013-12-22 23:16:56 +0000 Tim-Philipp Müller <tim@centricular.com>
-
- * configure.ac:
- configure: rename package from gst-rtsp to gst-rtsp-server
- To match git module name and avoid confusion with the
- rtsp lib in gst-plugins-base and rtsp plugin in -good.
-
-2013-12-22 23:15:02 +0000 Tim-Philipp Müller <tim@centricular.com>
-
- * configure.ac:
- configure: bump core/base/good requirement to 1.2.0
- Bump to released stable version and make implicit
- requirements explicit.
-
-2013-12-22 23:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
-
- * autogen.sh:
- * common:
- * configure.ac:
- Fix broken gettext setup which is not used anyway
-
-2013-12-22 22:36:06 +0000 Tim-Philipp Müller <tim@centricular.com>
-
- * common:
- Automatic update of common submodule
- From dbedaa0 to d48bed3
-
-2013-12-18 16:37:27 +0100 Aleix Conchillo Flaqué <aleix@oblong.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- media: add setup_sdp vmethod
- gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public
- gst_rtsp_media_setup_sdp.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
-
-2013-12-19 14:26:34 +0100 Edward Hervey <bilboed@bilboed.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- rtsp-stream: Check return value of sscanf
- streamid is only valid if sscanf matched something.
-
-2013-12-19 14:24:54 +0100 Edward Hervey <bilboed@bilboed.com>
-
- * gst/rtsp-server/rtsp-client.c:
- rtsp-client: Fix iteration
- Wouldn't even enter the code block otherwise (i++ was used as the check
- and not the postfix).
-
-2013-12-18 15:57:03 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: add vmethod to configure media and streams
- Implement a vmethod that can be used to configure the media and the
- streams based on the current context. Handle the blocksize handling in
- the default handler.
- See https://bugzilla.gnome.org/show_bug.cgi?id=720667
-
-2013-12-12 00:38:07 +0000 Tim-Philipp Müller <tim@centricular.com>
-
- * .gitignore:
- Make git ignore more unit test binaries
-
-2013-12-12 00:36:07 +0000 Tim-Philipp Müller <tim@centricular.com>
-
- * gst/rtsp-server/rtsp-address-pool.h:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-context.h:
- * gst/rtsp-server/rtsp-media-factory-uri.h:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media.h:
- * gst/rtsp-server/rtsp-mount-points.h:
- * gst/rtsp-server/rtsp-server.h:
- * gst/rtsp-server/rtsp-session-media.h:
- * gst/rtsp-server/rtsp-session-pool.h:
- * gst/rtsp-server/rtsp-session.h:
- * gst/rtsp-server/rtsp-stream-transport.h:
- * gst/rtsp-server/rtsp-stream.h:
- * gst/rtsp-server/rtsp-thread-pool.h:
- * gst/rtsp-server/rtsp-token.h:
- rtsp-server: add padding to many public structures
- Not mini objects though, since they are not subclassable
- anyway, nor kept on the stack or inlined in a structure.
-
-2013-12-03 11:54:42 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
-
- media: add new create_rtpbin vmethod
- * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod.
- https://bugzilla.gnome.org/show_bug.cgi?id=719734
-
-2013-12-03 00:34:52 +0100 Sebastian Rasmussen <sebras@gmail.com>
-
- * tests/check/gst/media.c:
- tests: fix memory leak, free test's thread pool
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=719733
-
-2013-11-29 15:50:52 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-stream-transport.c:
- stream-transport: free url in finalize
-
-2013-11-29 15:50:23 +0100 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-media.c:
- media: also do state change in suspended state
-
-2013-11-29 10:53:08 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media.c:
- media: also handle prepare and range in suspended state
- When we are suspended, we are already prepared.
- We can get the range in the suspended state.
-
-2013-11-27 15:04:04 +0100 Branko Subasic <branko@axis.com>
-
- * tests/check/Makefile.am:
- * tests/check/gst/sessionmedia.c:
- check: add test for uri in setup
- Added unit tests for the new functionality in GstRTSPStreamTransport.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
-
-2013-11-28 17:47:18 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-client.c:
- client: store setup uri and use in PLAY response
- Store the uri used when doing the setup and use that in the PLAY
- response.
- fixes https://bugzilla.gnome.org/show_bug.cgi?id=715168
-
-2013-11-28 17:35:45 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-stream-transport.c:
- * gst/rtsp-server/rtsp-stream-transport.h:
- stream-transport: add method to get/set url
-
-2013-11-28 14:14:35 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-client.c:
- client: suspend after SDP and unsuspend before PLAYING
- Based on patches by Ognyan Tonchev <ognyan@axis.com>
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711257
-
-2013-11-28 14:10:19 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session.c:
- * tests/check/gst/media.c:
- * tests/check/gst/mediafactory.c:
- media: add suspend modes
- Add support for different suspend modes. The stream is suspended right after
- producing the SDP and after PAUSE. Different suspend modes are available that
- affect the state of the pipeline. NONE leaves the pipeline state unchanged and
- is the current and old behaviour, PAUSE will set the pipeline to the PAUSED
- state and RESET will bring the pipeline to the NULL state.
- A stream is also unsuspended when it goes back to PLAYING, for RESET streams,
- this means that the pipeline needs to be prerolled again.
- Base on patches by Ognyan Tonchev <ognyan@axis.com>
- See https://bugzilla.gnome.org/show_bug.cgi?id=711257
-
-2013-11-28 14:06:53 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-media.c:
- media: start live streams in blocked state
- Start live streams in the blocked state and make them preroll using the
- messages. This ensure that no data is played by the sink until we explicitly
- unblock the stream right before going to PLAYING.
- See https://bugzilla.gnome.org/show_bug.cgi?id=711257
-
-2013-11-28 13:58:05 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-media.c:
- media: refactor starting and waiting for preroll
- Based on patches from Ognyan Tonchev <ognyan@axis.com>
- See https://bugzilla.gnome.org/show_bug.cgi?id=711257
-
-2013-11-28 13:42:21 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- stream: add API to block streams
- Add an API to block on the streams and make it post a message.
- Based on patch by Ognyan Tonchev <ognyan@axis.com>
- See https://bugzilla.gnome.org/show_bug.cgi?id=711257
-
-2013-11-27 15:42:45 +0100 Edward Hervey <edward@collabora.com>
-
- * docs/libs/Makefile.am:
- docs: Specify the override file
- Even if it's empty (for now) it avoids make distcheck complaining
-
-2013-11-26 17:23:04 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-media.c:
- media: move default implementations to where they are used
-
-2013-11-26 16:25:37 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-media.c:
- media: take the right lock in gst_rtsp_media_set_pipeline_state()
- We need to take the state_lock when calling this method.
-
-2013-11-26 16:24:35 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-media.c:
- media: handle add-added on non-bins too
- Handle dynamic payloaders that are not bins, as used in the unit-test.
-
-2013-11-22 01:30:53 +0100 Sebastian Rasmussen <sebras@hotmail.com>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media.c:
- rtsp-media/-factory: Fix request pad name comments
- These must be escaped for gtk-doc to parse the comments without warnings.
-
-2013-11-20 15:51:54 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
-
- rtsp-media: remove transports if media is in error status
- * gst/rtsp-server/rtsp-media.c (gst_rtsp_media_set_state): if we are
- trying to change to GST_STATE_NULL and media is in error status, we
- remove all transports.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712776
-
-2013-11-22 11:16:20 +0100 Wim Taymans <wtaymans@redhat.com>
-
- * gst/rtsp-server/rtsp-media.c:
- rtsp-media: use element metadata to find payloader
- Use the element metadata to find the payloader instead of checking
- for the base class.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=712396
-
-2013-11-15 12:14:32 -0800 Aleix Conchillo Flaque <aleix@oblong.com>
-
- rtsp-stream: add getter for payload type
- * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt.
- * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader
- element and create the stream with this one instead of the dynpay%d
- element.
- https://bugzilla.gnome.org/show_bug.cgi?id=712396
-
-2013-11-22 02:28:28 +0100 Sebastian Rasmussen <sebras@hotmail.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-context.h:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-mount-points.c:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-token.c:
- rtsp-*: Refer to NULL as a constant in comments
- Plus one typo fix.
- https://bugzilla.gnome.org/show_bug.cgi?id=714988
-
-2013-11-22 03:10:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
-
- rtsp-*: Fix type name typos in comments
- * rtsp-auth: Refer to GstRTSPToken, not GstRTSPtoken
- * rtsp-auth: Refer to part of constant name as text
- * rtsp-auth/-permissions/-token: Refer to Permissions not Permission
- * rtsp-session-media: Fix GstRTSPSessionMedia typo
- * rtsp-stream: Fix typo when refering to GstBin
- https://bugzilla.gnome.org/show_bug.cgi?id=714988
-
-2013-11-22 00:45:17 +0100 Sebastian Rasmussen <sebras@hotmail.com>
-
- * docs/README:
- * docs/libs/gst-rtsp-server-docs.sgml:
- * docs/libs/gst-rtsp-server-sections.txt:
- docs: Improve documentation
- * Include annotation-glossary to quiet gtk-doc
- * Rename remaining ClientState -> Context
- * Rename object hierarchy file
- * Remove stale chapter references
- * Add missing function and object references
- * Include missing GstRTSPAddressPoolResult
- https://bugzilla.gnome.org/show_bug.cgi?id=714988
-
-2013-11-18 10:47:04 +0000 Tim-Philipp Müller <tim@centricular.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-session-pool.c:
- * gst/rtsp-server/rtsp-session.c:
- * gst/rtsp-server/rtsp-stream.c:
- rtsp-server: sprinkle some allow-none annotations for g-i
-
-2013-11-18 11:18:15 +0100 Wim Taymans <wim.taymans@gmail.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- stream: add method to filter transports
- Add a method to safely iterate and collect the stream transports
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
-
-2013-11-15 16:35:05 +0100 Wim Taymans <wim.taymans@gmail.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-session-pool.c:
- * gst/rtsp-server/rtsp-session.c:
- rtsp: allow NULL func in filters
- Passing a null function make the filters return a list of
- refcounted objects.
-
-2013-11-12 16:52:35 +0100 Wim Taymans <wim.taymans@gmail.com>
-
- * gst/rtsp-server/rtsp-address-pool.c:
- * tests/check/gst/addresspool.c:
- address-pool: fix address increment
- Use a guint instead of guint8 to increment the address. It's still not
- completely correct because a guint might not be able to hold the complete
- address range, but that's an enhacement for later.
- Add unit test to test improved behaviour.
- https://bugzilla.gnome.org/show_bug.cgi?id=708237
-
-2013-11-12 10:55:14 +0100 Patricia Muscalu <patricia@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * tests/check/gst/client.c:
- client: allow absolute path in requests
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689
-
-2013-11-07 13:22:09 +0100 Patricia Muscalu <patricia@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: make make_path_from_uri a vmethod
-
-2013-11-12 12:04:55 +0100 Wim Taymans <wim.taymans@gmail.com>
-
- * docs/libs/gst-rtsp-server-sections.txt:
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- * tests/check/Makefile.am:
- * tests/check/gst/stream.c:
- stream: Add functions to get rtp and rtcp sockets
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
-
-2013-11-12 11:21:55 +0100 Wim Taymans <wim.taymans@gmail.com>
-
- * gst/rtsp-server/rtsp-context.c:
- * gst/rtsp-server/rtsp-context.h:
- context: defing a GType for the context
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018
-
-2013-10-12 23:56:00 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
-
- * gst/rtsp-server/Makefile.am:
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-context.c:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-mount-points.c:
- * gst/rtsp-server/rtsp-server.h:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session.c:
- * gst/rtsp-server/rtsp-stream.c:
- Fixed several GIR warnings
-
-2013-11-12 11:15:46 +0100 Wim Taymans <wim.taymans@gmail.com>
-
- * gst/rtsp-server/rtsp-auth.c:
- auth: small typos
-
-2013-10-19 19:25:27 +0200 Sebastian Rasmussen <sebras@hotmail.com>
-
- * tests/check/Makefile.am:
- * tests/check/gst/token.c:
- tests: Add unit tests for token
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
-
-2013-10-19 19:24:34 +0200 Sebastian Rasmussen <sebras@hotmail.com>
-
- * gst/rtsp-server/rtsp-token.c:
- token: Validate args for gst_rtsp_token_is_allowed
- See https://bugzilla.gnome.org/show_bug.cgi?id=710520
-
-2013-10-19 19:21:53 +0200 Sebastian Rasmussen <sebras@hotmail.com>
-
- * gst/rtsp-server/rtsp-token.c:
- token: Fix bug when creating empty token
- We always want to have a valid GstStructure in the token.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710520
-
-2013-11-12 10:28:55 +0100 Wim Taymans <wim.taymans@gmail.com>
-
- * gst/rtsp-server/rtsp-thread-pool.c:
- thread-pool: avoid race in shutdown
- If we call g_main_loop_quit before the thread has entered g_main_loop_run, we
- don't actually stop the mainloop ever. Solve this race by adding an idle source
- to the mainloop that calls the _quit. This way we immediately exit the mainloop
- if quit was called before we started it.
-
-2013-10-19 17:36:05 +0200 Sebastian Rasmussen <sebras@hotmail.com>
-
- * tests/check/Makefile.am:
- * tests/check/gst/permissions.c:
- tests: Add unit tests for permissions
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710202
-
-2013-10-15 18:50:47 +0200 Sebastian Rasmussen <sebras@hotmail.com>
-
- * tests/check/gst/mediafactory.c:
- tests: Test mediafactory permissions
- See https://bugzilla.gnome.org/show_bug.cgi?id=710202
-
-2013-10-19 17:39:35 +0200 Sebastian Rasmussen <sebras@hotmail.com>
-
- * gst/rtsp-server/rtsp-permissions.c:
- permissions: Fix refcounting when adding/removing roles
- Previously a role that was removed was unreffed twice, and when
- replacing an existing role the replaced role was freed while still being
- referenced. Both bugs are now fixed.
- See https://bugzilla.gnome.org/show_bug.cgi?id=710202
-
-2013-10-15 18:01:38 +0200 Sebastian Rasmussen <sebras@hotmail.com>
-
- * tests/check/gst/media.c:
- * tests/check/gst/mediafactory.c:
- * tests/check/gst/rtspserver.c:
- tests: Check gst_rtsp_url_parse return value
- See https://bugzilla.gnome.org/show_bug.cgi?id=710202
-
-2013-11-05 11:22:51 +0000 Tim-Philipp Müller <tim@centricular.com>
-
- * common:
- Automatic update of common submodule
- From 865aa20 to dbedaa0
-
-2013-10-14 12:03:07 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-server.c:
- rtsp-server: Fix socket leak
- https://bugzilla.gnome.org/show_bug.cgi?id=710088
-
-2013-10-30 22:16:54 +0100 Sebastian Dröge <sebastian@centricular.com>
-
- * gst/rtsp-server/rtsp-session-pool.c:
- rtsp-session-pool: Make sure session IDs are properly URI-escaped
- https://bugzilla.gnome.org/show_bug.cgi?id=643812
-
-2013-10-15 16:37:34 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
-
- * examples/.gitignore:
- * examples/test-video.c:
- examples: fix compilation when WITH_AUTH is defined
- https://bugzilla.gnome.org/show_bug.cgi?id=710228
-
-2013-10-30 19:10:59 +0100 Sebastian Dröge <sebastian@centricular.com>
-
- * .gitignore:
- gitignore: Add new test binary
-
-2013-10-09 15:19:12 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * tests/check/Makefile.am:
- * tests/check/gst/threadpool.c:
- thread-pool: Add unit test for the thread pools
- https://bugzilla.gnome.org/show_bug.cgi?id=710228
-
-2013-10-09 15:25:10 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-thread-pool.c:
- thread-pool: Fix thread leak when reusing threads
- https://bugzilla.gnome.org/show_bug.cgi?id=709730
-
-2013-10-14 08:30:33 +0200 Patricia Muscalu <patricia@axis.com>
-
- * gst/rtsp-server/rtsp-server.c:
- * tests/check/gst/rtspserver.c:
- tests: fixed racy behavior in rtspserver tests
- https://bugzilla.gnome.org/show_bug.cgi?id=710078
-
-2013-10-14 19:36:24 +0200 Sebastian Rasmussen <sebras@hotmail.com>
-
- * tests/check/gst/addresspool.c:
- tests: Improve address pool unit tests
- Add a range with mixed IPV4 and IPV6 addresses to pool.
- Get an IPV4 address from an IPV6-only pool.
- Get an IPV6 address from an IPV4-only pool.
- Reserve a IPV6 address from an IPV4-only pool.
- Check for unicast addresses in multicast-only pool.
- Check for unicast addresses in uni-/multicast-mixed pool.
- https://bugzilla.gnome.org/show_bug.cgi?id=710128
-
-2013-10-04 06:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: append query string in PAUSE/PLAY/TEARDOWN as well
-
-2013-10-01 14:04:17 +0200 Jonas Holmberg <jonashg@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- client: Add query to control path
- If the SETUP url contains a query it must be appended to the control
- path so that it matches any already created stream in the media. The
- query will also be appended to the session media path.
-
-2013-10-04 05:48:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- rtsp-media: remove old line
-
-2013-10-01 13:15:19 +0200 Jonas Holmberg <jonashg@axis.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: Correct control comparison
- https://bugzilla.gnome.org/show_bug.cgi?id=709176
-
-2013-09-09 21:51:44 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: Check dynamically if the pipeline supports seeking
- We should not depend on whether or not the pipeline state change
- returned NO_PREROLL or not. A media could dynamically change its
- element and switch from seekable to non seekable so it's best to test
- the seekable nature of the pipeline dynamically when we try to do a seek.
-
-2013-09-09 21:51:23 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: Return FALSE if seeking is not supported
-
-2013-10-01 17:16:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- rtsp-media: don't seek accurate by default
- Accurate seeking is perhaps a little overkill in the most common situation and
- causes some formats (mp3) over slow media to seek extremely slowly.
-
-2013-09-26 14:36:58 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * tests/check/gst/rtspserver.c:
- tests: fix unit test
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708742
-
-2013-09-26 11:20:05 +0200 Jonas Holmberg <jonashg@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- client: Reply 400 if media cannot be constructed
- Reply 400 Bad Request instead of 503 Service Unavailable if media
- cannot be constructed in SETUP.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708821
-
-2013-09-26 09:41:10 +0200 Jonas Holmberg <jonashg@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- client: Send setup reply once only
- If find_media() failed in handle_setup_request() two replies was sent.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708819
-
-2013-09-24 18:35:36 +0100 Tim-Philipp Müller <tim@centricular.net>
-
- * common:
- Automatic update of common submodule
- From 6b03ba7 to 865aa20
-
-2013-09-23 14:28:04 +0200 Jonas Holmberg <jonashg@axis.com>
-
- * gst/rtsp-server/rtsp-server.c:
- server: Emit client-connected signal earlier
- Emit client-connected before the client ref is given to a GSource,
- otherwise client-connected can be emitted after the client object has
- been freed.
-
-2013-09-24 17:30:18 +0200 Patrick Radizi <patrick.radizi at axis.com>
-
- * gst/rtsp-server/rtsp-address-pool.c:
- * gst/rtsp-server/rtsp-address-pool.h:
- * gst/rtsp-server/rtsp-stream.c:
- * tests/check/gst/addresspool.c:
- addresspool: return reason of failure
- Let gst_rtsp_address_pool_reserve_address() return the reason why
- the address could not be reserved.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=708229
-
-2013-09-20 16:47:56 +0200 Edward Hervey <edward@collabora.com>
-
- * autogen.sh:
- autogen.sh: Sync behaviour with other GStreamer modules
- Allows building from outside of tree amongst other things
-
-2013-09-20 16:18:54 +0200 Edward Hervey <edward@collabora.com>
-
- * common:
- Automatic update of common submodule
- From b613661 to 6b03ba7
-
-2013-09-19 18:46:14 +0100 Tim-Philipp Müller <tim@centricular.net>
-
- * common:
- Automatic update of common submodule
- From 74a6857 to b613661
-
-2013-09-19 17:39:24 +0100 Tim-Philipp Müller <tim@centricular.net>
-
- * common:
- Automatic update of common submodule
- From 01a7a46 to 74a6857
-
-2013-09-19 15:44:26 +0200 Jonas Holmberg <jonashg@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- client: Do not read beyond end of path string
- If the setup was done without a control url, make sure we don't try to read the
- non-existing control string and crash.
-
-2013-09-17 14:39:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: Fix RTPInfo header
- Refactor the method to make the content_base.
- Use the content-base and the control url to construct the RTPInfo
- url.
-
-2013-09-17 12:21:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: map url to path only in describe
- Only map the request url to a path in the DESCRIBE method. The SDP then
- contains the base and control urls that should be used to SETUP/PAUSE/
- PLAY/TEARDOWN the media.
-
-2013-09-17 11:41:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- Revert "client: map URL to path in requests"
- This reverts commit e3fded2cec897a2ec003450607b916cc1601fd2d.
- This is not correct, we only remap the URL to a path in DESCRIBE, the SDP then
- contains the base and control urls which are used in the SETUP, PLAY,
- PAUSE and TEARDOWN requests.
-
-2013-09-16 17:16:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: map URL to path in requests
-
-2013-09-16 16:47:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-mount-points.c:
- * gst/rtsp-server/rtsp-mount-points.h:
- mount-points: make vmethod to make path from uri
- Make a vmethod to transform an url into a path. The path is then used to lookup
- the factory. This makes it possible to also use other bits of the url, such as
- the query parameters, to locate the factory.
-
-2013-09-09 11:05:26 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-thread-pool.c:
- * gst/rtsp-server/rtsp-thread-pool.h:
- thread-pool: Add cleanup to wait for the threadpool to finish
- Also fix race condition if two threads are asking for the first
- thread from the thread pool at once. This would case two internal
- GThreadPools to be created.
- https://bugzilla.gnome.org/show_bug.cgi?id=707753
-
-2013-09-05 08:56:02 +0200 Jonas Holmberg <jonashg@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * tests/check/gst/client.c:
- client: free threadpool
- https://bugzilla.gnome.org/show_bug.cgi?id=707638
-
-2013-09-06 17:23:20 +0200 Jonas Holmberg <jonashg@axis.com>
-
- * tests/check/gst/mountpoints.c:
- mountpoints tests: unref matched factories
- https://bugzilla.gnome.org/show_bug.cgi?id=707638
-
-2013-09-05 18:01:18 +0200 Jonas Holmberg <jonashg@axis.com>
-
- * tests/check/gst/media.c:
- media tests: unref thread pool and caps
- https://bugzilla.gnome.org/show_bug.cgi?id=707638
-
-2013-09-05 08:53:55 +0200 Jonas Holmberg <jonashg@axis.com>
-
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media.c:
- auth, media, media-factory: unref permissions
- https://bugzilla.gnome.org/show_bug.cgi?id=707638
-
-2013-08-23 15:15:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/Makefile.am:
- Makefile: add rule for appsrc example
-
-2013-08-23 15:14:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-appsrc.c:
- tests: add appsrc example
- Add an example on how to use appsrc to feed the server pipeline with data.
-
-2013-08-22 12:10:39 +0200 Patricia Muscalu <patricia@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- rtsp-client: remove query part from content-base string
- Make sure that after the control url has been resolved, it's
- not a part of the query-string.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706568
-
-2013-08-23 10:38:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: don't check url in response
- There is no url or method in the response to check
-
-2013-08-08 10:57:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- Add handle-response signal for when we receive a GET_PARAMETER response
-
-2013-08-16 12:42:22 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- Fix gst_rtsp_server_client_filter, using wrong variable type
-
-2013-08-22 18:39:59 +0100 Tim-Philipp Müller <tim@centricular.net>
-
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- rtsp-media-factory-uri: check AAC properly for whether it's parsed or not
- For AAC we need to check for framed=true instead of parsed=true.
- https://bugzilla.gnome.org/show_bug.cgi?id=701384
-
-2013-08-16 17:05:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: optimize pipeline for protocols
- When TCP is not an allowed protocol for the stream, avoid creating the
- appsrc/appsink/queue and tee elements.
-
-2013-08-16 16:34:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: set protocols on streams
-
-2013-08-16 16:16:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: use protocols supported by stream
-
-2013-08-16 16:16:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-stream.c:
- media-factory: allow all protocols
-
-2013-08-16 16:10:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: configure protocols in new streams
-
-2013-08-16 16:08:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- stream: add protocols property
-
-2013-08-05 10:46:33 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- rtsp-media: send state in "new-state" signal
- https://bugzilla.gnome.org/show_bug.cgi?id=705110
-
-2013-08-02 14:11:01 +0200 Lubosz Sarnecki <lubosz@gmail.com>
-
- * configure.ac:
- build: add subdir-objects to AM_INIT_AUTOMAKE
- Fixes warnings with automake 1.14
- https://bugzilla.gnome.org/show_bug.cgi?id=705350
-
-2013-08-02 17:15:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * docs/libs/gst-rtsp-server-sections.txt:
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- server: add method to iterate clients of server
-
-2013-06-11 19:10:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- Add vmethod for rtsp-media subclass to access rtpbin
-
-2013-07-11 16:12:04 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.h:
- small documentation fix
-
-2013-07-11 16:11:55 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- Do not take range header if range is invalid
-
-2013-08-02 16:57:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * docs/libs/gst-rtsp-server-sections.txt:
- * gst/rtsp-server/rtsp-media.c:
- media: add docs for new method
-
-2013-07-02 18:55:28 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- Add API to rtsp-media set the pipeline's state
-
-2013-06-11 19:09:42 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- Update current position/duration when gst_rtsp_media_get_range_string is called
-
-2013-07-22 17:27:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-cgroups.c:
- tests: add some more docs
-
-2013-07-22 14:25:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-cgroups.c:
- * gst/rtsp-server/Makefile.am:
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-context.c:
- * gst/rtsp-server/rtsp-context.h:
- * gst/rtsp-server/rtsp-params.c:
- * gst/rtsp-server/rtsp-params.h:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-thread-pool.c:
- * gst/rtsp-server/rtsp-thread-pool.h:
- * tests/check/gst/client.c:
- ClientState -> Context
- Rename the clientstate to context and put the code in a separate file.
-
-2013-07-18 12:19:25 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-auth.c:
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-auth.h:
- auth: add support for default token
- The default token is used when the user is not authenticated and can be used to
- give minimal permissions.
-
-2013-07-18 11:44:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-auth.c:
- * gst/rtsp-server/rtsp-auth.c:
- auth: use defines when possible
-
-2013-07-18 11:44:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-address-pool.c:
- address-pool: improve docs
-
-2013-07-18 12:26:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-permissions.c:
- permissions: add the role to the copy
-
-2013-07-17 19:35:33 -0400 Olivier Crête <olivier.crete@collabora.com>
-
- * gst/rtsp-server/rtsp-permissions.c:
- permissions: Also copy the roles
-
-2013-07-17 19:32:09 -0400 Olivier Crête <olivier.crete@collabora.com>
-
- * gst/rtsp-server/rtsp-permissions.c:
- permissions: Make it build
-
-2013-07-16 12:36:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-address-pool.h:
- docs: small fixes
-
-2013-07-16 12:32:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * docs/libs/gst-rtsp-server-sections.txt:
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-stream-transport.c:
- * gst/rtsp-server/rtsp-stream-transport.h:
- * gst/rtsp-server/rtsp-stream.c:
- * tests/check/gst/client.c:
- docs: improve docs
-
-2013-07-16 12:32:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * docs/libs/gst-rtsp-server-sections.txt:
- * gst/rtsp-server/rtsp-address-pool.c:
- * gst/rtsp-server/rtsp-address-pool.h:
- * tests/check/gst/addresspool.c:
- * tests/check/gst/rtspserver.c:
- address-pool: cleanups
- Remove redundant method, improve docs.
-
-2013-07-15 17:31:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * docs/libs/gst-rtsp-server-sections.txt:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-permissions.c:
- * gst/rtsp-server/rtsp-permissions.h:
- * gst/rtsp-server/rtsp-token.c:
- docs: improve docs
-
-2013-07-15 17:12:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-permissions.c:
- permissions: implement _remove_role
-
-2013-07-15 17:12:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-permissions.c:
- permissions: update docs
-
-2013-07-15 16:48:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/gst/client.c:
- tests: simplify tests
- Client settings are now disabled by default so we don't need an auth
- module to disable them.
-
-2013-07-15 16:47:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-auth.c:
- auth: add default authorizations
- When no auth module is specified, use our table of defaults to look up the
- default value of the check instead of always allowing everything. This was
- we can disallow client settings by default.
-
-2013-07-15 16:05:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * docs/README:
- README: update readme
-
-2013-07-15 15:25:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-thread-pool.c:
- * gst/rtsp-server/rtsp-thread-pool.h:
- thread-pool: add more docs
-
-2013-07-15 14:50:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-thread-pool.c:
- * gst/rtsp-server/rtsp-thread-pool.h:
- thread-pool: fix race in thread reuse
- If we try to reuse a thread right after we made it stop, we end up using a
- stopped thread. Catch this case and only reuse threads that are not stopping.
-
-2013-07-15 14:50:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- server: add small debug
-
-2013-07-15 11:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/gst/client.c:
- client: fix test
- Add some permissions to media so we can use the auth and enable
- client settings.
-
-2013-07-15 11:57:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: support pushed context in handle_request
- If we already have a pushed state, reuse it and add our own things. This makes
- it easier to write tests.
-
-2013-07-15 11:56:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-auth.c:
- auth: don't auth on methods
- Don't authorize on methods anymore but on the resources that we
- try to access, this is more flexible.
- Move the authorization checks to where they are needed and let the
- check return the response on error.
-
-2013-07-15 11:51:34 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-mount-points.c:
- mount-points: add some debug
-
-2013-07-12 17:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/gst/client.c:
- tests: almost fix test
-
-2013-07-12 17:07:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- auth: let the auth module check client_settings
- Let the auth module decide if client settings are allowed for the
- current client.
-
-2013-07-12 17:06:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-token.c:
- * gst/rtsp-server/rtsp-token.h:
- token: add method to check boolean permission
-
-2013-07-12 16:36:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-auth.c:
- * examples/test-cgroups.c:
- * gst/rtsp-server/rtsp-token.c:
- * gst/rtsp-server/rtsp-token.h:
- token: simplify token constructor
- Use variable arguments to make easier API.
-
-2013-07-12 16:17:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-auth.c:
- * examples/test-cgroups.c:
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media-factory.h:
- media-factory: add convenience API for factory
-
-2013-07-12 16:03:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-auth.c:
- * examples/test-cgroups.c:
- * gst/rtsp-server/rtsp-permissions.c:
- * gst/rtsp-server/rtsp-permissions.h:
- permissions: simplify API a little
- Avoid passing GstStructure in the add_role method, use varargs instead
- to construct the structure behind the scenes. We can then also use the
- structure name as the role and simplify some more logic.
-
-2013-07-12 16:01:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-auth.c:
- auth: fix typo
-
-2013-07-12 15:19:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-client.c:
- auth: handle unauthorized response
- Move handling of the unauthorized response to the auth module, it can add
- the appropriate headers to request authorization for the required method
- much better than the client.
-
-2013-07-12 15:13:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: allow for sending any message, not only requests
- Change the _send_request() method to _send_message() so that we
- can both send requests and replies.
-
-2013-07-12 14:10:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * docs/libs/gst-rtsp-server-sections.txt:
- * gst/rtsp-server/rtsp-server.h:
- docs: fix docs
-
-2013-07-12 12:41:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-video.c:
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- auth: move TLS handling to auth module
- Remove the TLS settings on the server and move it to the auth module because
- that is where security related bits go.
-
-2013-07-12 12:38:54 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: add state push/pop
-
-2013-07-12 12:36:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: add connection to state
-
-2013-07-11 20:45:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-mount-points.c:
- mount-points: fix debug
-
-2013-07-11 17:28:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/gst/media.c:
- tests: fix media test
-
-2013-07-11 17:28:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-thread-pool.c:
- thread-pool: we don't require a state
-
-2013-07-11 17:18:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- server: let context ref the server
- So that we don't risk losing the server object early anc crash.
-
-2013-07-11 17:05:00 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/gst/client.c:
- tests: fix client test
-
-2013-07-11 16:57:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * docs/README:
- * docs/libs/gst-rtsp-server-docs.sgml:
- * docs/libs/gst-rtsp-server-sections.txt:
- * gst/rtsp-server/rtsp-address-pool.c:
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-mount-points.c:
- * gst/rtsp-server/rtsp-params.c:
- * gst/rtsp-server/rtsp-permissions.c:
- * gst/rtsp-server/rtsp-sdp.c:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session-pool.c:
- * gst/rtsp-server/rtsp-session.c:
- * gst/rtsp-server/rtsp-stream-transport.c:
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-thread-pool.c:
- * gst/rtsp-server/rtsp-token.c:
- docs: improve docs
-
-2013-07-11 16:28:09 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-session-pool.c:
- * gst/rtsp-server/rtsp-session-pool.h:
- session-pool: make vmethod to create a session
- Make a vmethod to create a sessions so that subclasses can create
- custom session objects
-
-2013-07-11 12:24:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media.h:
- * gst/rtsp-server/rtsp-mount-points.h:
- * gst/rtsp-server/rtsp-session-pool.h:
- * gst/rtsp-server/rtsp-stream.h:
- docs: more updates
-
-2013-07-11 12:18:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * docs/libs/gst-rtsp-server-docs.sgml:
- * docs/libs/gst-rtsp-server-sections.txt:
- * gst/rtsp-server/rtsp-address-pool.c:
- * gst/rtsp-server/rtsp-address-pool.h:
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- * gst/rtsp-server/rtsp-permissions.c:
- * gst/rtsp-server/rtsp-permissions.h:
- * gst/rtsp-server/rtsp-server.h:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session-media.h:
- * gst/rtsp-server/rtsp-session-pool.h:
- * gst/rtsp-server/rtsp-session.h:
- * gst/rtsp-server/rtsp-stream-transport.h:
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-thread-pool.h:
- docs: update docs
-
-2013-07-11 10:28:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- * examples/Makefile.am:
- configure: compile cgroup example conditionally
- Only compile the cgroup example when we have libcgroup
-
-2013-07-10 20:57:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- * examples/Makefile.am:
- * examples/test-cgroups.c:
- examples: add cgroups example
-
-2013-07-10 20:55:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/gst/rtspserver.c:
- tests: fix compilation
-
-2013-07-10 20:48:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-thread-pool.c:
- thread-pool: fix vmethod invocation
-
-2013-07-10 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-thread-pool.c:
- * gst/rtsp-server/rtsp-thread-pool.h:
- thread-pool: store thread type in thread
-
-2013-07-10 17:09:27 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: pass thread from pool to media _prepare
- Get a thread from the configured threadpool and pass it to the prepare method of
- the media.
-
-2013-07-10 17:08:14 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- media: Accept a thread in _prepare
- Remove out own threadpool handling and use the provided thread and
- maincontext for the bus messages and the state changes.
-
-2013-07-10 17:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- server: configure client thread pool
-
-2013-07-10 17:06:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: add method to configure thread pool
-
-2013-07-10 16:49:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- server: use thread pool
- Use the thread pool instead of doing our own thing.
-
-2013-07-10 16:47:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/Makefile.am:
- * gst/rtsp-server/rtsp-thread-pool.c:
- * gst/rtsp-server/rtsp-thread-pool.h:
- thread-pool: add object to manage threads
- Add an object to manage the client and media threads.
-
-2013-07-10 15:28:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-auth.c:
- auth: debug authorization check
-
-2013-07-09 20:44:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: start media pipeline in context
- Start the media pipeline in the provided context (or our default one
- when NULL). This makes sure that we run the bus thread in this context and that
- all media threads are children of this context.
-
-2013-07-09 16:38:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- factory: pass permissions to media by default
-
-2013-07-09 16:09:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-auth.c:
- test: add permissions to auth test
- Ass some permissions to the media factory in the test.
-
-2013-07-09 16:04:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-client.c:
- auth: simplify auth checks
- Remove client from methods, it's now in the state
- Perform the check specified by the string, use the information from the
- thread local context.
-
-2013-07-09 16:01:29 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: add state to current thread
- Add the client to the ClientState object.
- Place the ClientState on the current thread.
-
-2013-07-09 14:33:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- media: make it possible to set permissions
- Make it possible to set permissions on media and media factory objects
-
-2013-07-09 14:31:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/Makefile.am:
- * gst/rtsp-server/rtsp-permissions.c:
- * gst/rtsp-server/rtsp-permissions.h:
- permissions: add permissions object
- Add a mini object to store permissions based on a role.
-
-2013-07-08 16:29:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-auth.c:
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-client.c:
- auth: add auth checks
- Add an enum with auth checks and implement the checks in the auth object.
- Perform the checks from the client.
-
-2013-07-05 20:48:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-auth.c:
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-client.h:
- auth: use the token after authentication
- After we authenticated a user, keep the Token around in the state.
-
-2013-07-05 20:43:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- * tests/check/gst/media.c:
- media: add optional context for bus messages
- Add an optional mainloop to _prepare that will handle the bus messages instead
- of always using the shared mainloop.
-
-2013-07-05 20:34:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/Makefile.am:
- * gst/rtsp-server/rtsp-token.c:
- * gst/rtsp-server/rtsp-token.h:
- token: add authorization token
- Add a simply miniobject that contains the authorizations. The object contains a
- GstStructure that hold all authorization fields. When a user is authenticated,
- the auth module will create a Token for the user. The token is then used to
- check what operations the user is allowed to do and various other configuration
- values.
-
-2013-07-05 12:08:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-auth.c:
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- auth: remove auth from media and factory
- Remove the auth object from media and factory. We want to have the RTSPClient
- authenticate and authorize resources, there is no need to place another auth
- manager on the media/factory.
-
-2013-07-04 14:33:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-auth.c:
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-client.h:
- auth: add support for multiple basic auth tokens
- Make it possible to add multiple basic authorisation tokens to one authorization
- object. Associate with each token an authorization group that will define what
- capabilities are allowed.
-
-2013-07-03 16:15:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: error out on non-aggregate control
- We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.
-
-2013-07-03 15:55:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: rework setup request a little
- Cache the media in DESCRIBE based on the longest matching path with the uri
- that we can find in the mount points.
- Rework the setup request a little to get the media from the session or from
- the longest matching path, this way we can derive the control string as
- everything after the path instead of hardcoding it.
- Find the stream based on the control string and only open a session when all
- this can be done.
-
-2013-07-03 15:14:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- media: add method to find a stream by control url
-
-2013-07-03 15:13:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- stream: add method to check control url of stream
-
-2013-07-03 12:37:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session-media.h:
- * gst/rtsp-server/rtsp-session.c:
- * gst/rtsp-server/rtsp-session.h:
- session: use path matching for session media
- Use a path string instead of a uri to lookup session media in the sessions. Also
- use path matching to find the largest possible path that matches.
-
-2013-07-03 11:04:53 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-mount-points.c:
- * gst/rtsp-server/rtsp-mount-points.h:
- * tests/check/gst/mountpoints.c:
- mount-points: remove useless vmethod
- Making lookups in the mount points should not be done with a URL, if there is a
- mapping to be done from URL to mount points, we'll need to do it somewhere
- else.
-
-2013-07-03 10:25:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-mount-points.c:
- * gst/rtsp-server/rtsp-mount-points.h:
- * tests/check/gst/mountpoints.c:
- mount-points: improve mount point searching
- Use a GSequence to keep track of the mount points.
- Match a URL to the longest matching registered mount point. This should be the
- URL to perform aggreagate control and the remainder is the stream specific
- control part.
- Add some unit tests for this.
-
-2013-07-03 10:40:33 +0200 Sebastian Dröge <slomo@circular-chaos.org>
-
- * gst/rtsp-server/Makefile.am:
- rtsp-server: Allow building of static library
-
-2013-07-02 15:59:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/gst/mediafactory.c:
- tests: fix compilation
-
-2013-07-02 15:54:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-sdp.c:
- sdp: get control string from stream
- Use the control string as configured in the stream.
-
-2013-07-02 14:44:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- stream: add methods and property to set control string
-
-2013-07-02 11:58:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: cleanups
- Rename variables for clarity
- Keep media in state when we can
-
-2013-07-01 16:46:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- stream: add more support for IPv6
- Rename _get_address to _get_multicast_address in GstRTSPStream to
- make it clear that this function only deals with multicast.
- Make it possible to have both an IPv4 and IPv6 multicast address on
- a stream. Give the client an IPv4 or IPv6 address depending on the
- address it used to connect to the server.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
-
-2013-07-01 15:18:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: fix comment
-
-2013-07-01 14:45:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: handle failed port allocation
- Allow for ipv4 or ipv6 socket allocations to fail. Only report failure if we
- can't allocate any family at all. Also keep track of what port families we
- allocated.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703175
-
-2013-07-01 12:20:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: improve docs
-
-2013-07-01 12:04:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream-transport.c:
- stream-transport: remove old if 0 block
-
-2013-06-27 11:21:42 +0200 Patricia Muscalu <patricia@axis.com>
-
- * tests/check/gst/client.c:
- tests: fix tests
- gst_rtsp_client_get_uri() has been removed
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703173
-
-2013-06-26 17:18:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: add method to filter managed sessions
- Add a method to filter the sessions managed by this client connection.
- See https://bugzilla.gnome.org/show_bug.cgi?id=703016
-
-2013-06-26 16:32:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: remove _get_uri() method
- Remove the get_uri() method on the client. A client has no uri, the uri
- property is an internal property to manage the last cached media for
- the client.
-
-2013-06-26 16:31:39 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory.h:
- media-factory: fix typo
-
-2013-06-26 14:42:15 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-media.c:
- rtsp-media: Do not leak the query in default_query_stop
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120
-
-2013-06-25 15:46:41 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: don't unlock when conversion fails
- Don't unlock the state lock when conversion fails because it was not locked.
-
-2013-06-10 17:32:40 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- Add query_position and query_stop vmethods to rtsp-media
-
-2013-06-10 17:33:01 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- Fix typo in property install for rtsp-media's time-provider
-
-2013-06-25 15:09:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: clean some variables
- Clean some variables and add some guards to _send_request()
-
-2013-06-10 17:32:12 -0400 Youness Alaoui <youness.alaoui@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- Add gst_rtsp_client_send_request API
- This makes it possible to send arbitrary messages to a client, such as
- SET_PARAMETER or GET_PARAMETER
-
-2013-06-24 23:56:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- media: add _get_element() method
- Add method to get the element used when creating the media.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
-
-2013-06-24 23:51:38 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: fix docs
-
-2013-06-24 11:41:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- stream: allow access to the rtp session
- https://bugzilla.gnome.org/show_bug.cgi?id=703004
-
-2013-06-24 10:43:59 +0200 Alexander Schrab <alexas@axis.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- dscp qos support in gst-rtsp-stream
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
-
-2013-06-20 17:30:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/gst/rtspserver.c:
- tests: fix test
- Actually do what the comment says. Also keep the old code around, not sure what
- should happen when you get a 454 from a TEARDOWN, does it close the connection?
- it currently doesn't.
-
-2013-06-20 12:20:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: also watch newly created session
- When we newly created a session, start watching it immediately instead of
- on the next request.
-
-2013-06-20 12:18:23 +0200 Patricia Muscalu <patricia@axis.com>
-
- * tests/check/gst/client.c:
- tests: add unit test for new-session
- See https://bugzilla.gnome.org/show_bug.cgi?id=701587
-
-2013-06-20 12:16:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: emit new-session when new session is created
- Only emit new-session when we created a new session for a client, not when a
- client picked up a previous session.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587
-
-2013-06-20 11:17:29 +0200 Alexander Schrab <alexas@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- client: handle asterisk as path in requests
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266
-
-2013-06-20 11:14:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: handle segment query format mismatch
- It's possible that the segment query returns with a different format than what
- we asked for, handle this case also.
-
-2013-06-11 15:28:32 +0200 David Svensson Fors <davidsf@axis.com>
-
- * gst/rtsp-server/rtsp-media.c:
- media: use segment stop in collect_media_stats
- Use segment stop instead of duration as range end point.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185
-
-2013-06-17 16:47:56 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-media.c:
- * tests/check/gst/media.c:
- rtsp-media: Do not leak the element in take_pipeline
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470
-
-2013-06-17 16:18:37 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- rtsp-client: Make configure_client_transport virtual
- This patch makes configure_client_transport virtual. The functionality is
- needed to handle some weird clients sending multicast transport settings as url
- options.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173
-
-2013-06-12 12:23:56 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- rtsp-client: Make param_set and param_get virtual
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072
-
-2013-06-05 15:49:45 +0200 David Svensson Fors <davidsf@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- media: convert_range replaces get_range_times
- get_range_times worked for handling UTC ranges for seeks, but we also
- need to convert back from NPT to the requested unit in
- get_range_string. convert_range is now used for both.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
-
-2013-06-14 16:05:59 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-sdp.c:
- * gst/rtsp-server/rtsp-sdp.h:
- sdp: cleanup sdp info
- We don't need to pass the proto, we can more easily check a boolean.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
-
-2013-06-12 15:22:57 +0200 Alexander Schrab <alexas@axis.com>
-
- * gst/rtsp-server/rtsp-sdp.c:
- use 0.0.0.0 or :: for c= line instead of server address
-
-2013-06-12 10:56:16 +0200 Alexander Schrab <alexas@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- use local address, not remote, in SDP
- See https://bugzilla.gnome.org/show_bug.cgi?id=702063
-
-2013-06-05 15:18:26 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 098c0d7 to 01a7a46
-
-2013-05-29 13:45:00 +0200 David Svensson Fors <davidsf@axis.com>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- media: possibility to override range time conversion
- Make it possible to override the conversion from GstRTSPTimeRange to
- GstClockTimes, that is done before seeking on the media
- pipeline. Overriding can be useful for UTC ranges, where the default
- conversion gives nanoseconds since 1900.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
-
-2013-06-03 12:04:44 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- rtsp-server: Expose the use_client_settings API
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
-
-2013-05-30 08:07:48 +0200 Alexander Schrab <alexas@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- rtspstream: handle both ipv4 and ipv6 clients
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
-
-2013-05-31 15:28:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-sdp.c:
- Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
- This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.
- We already have a way to place extra attributes in the SDP by using a string
- property with prefix x- or a- in the caps.
-
-2013-05-31 15:27:48 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-sdp.c:
- Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
- This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.
- We already have a way to place extra attributes in the SDP, just make a string
- property in the payloader with a- or x- prefix.
-
-2013-05-31 15:41:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-sdp.c:
- rtsp: place a- and x- properties as attributes
- application/x-rtp has properties with a- and x- prefixes that should be
- placed as attributes in the SDP for the media instead of being added to the
- fmtp.
-
-2013-05-31 12:10:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/Makefile.am:
- * examples/test-video.c:
- example: add TLS example
-
-2013-05-31 11:42:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- server: add support for TLS
- Add methods to set and get a TLS certificate.
- Add vmethod to configure a new connection. By default, configure the TLS
- certificate in a new connection if needed.
-
-2013-05-31 11:14:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- server: remove accept_client vmethod
- This vmethod is not very useful so remove it.
-
-2013-05-30 17:23:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- server: don't crash on NULL GError
-
-2013-05-30 10:46:33 +0200 Patricia Muscalu <patricia@axis.com>
-
- * gst/rtsp-server/rtsp-session-pool.c:
- rtsp-session-pool: corrected session timeout detection
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
-
-2013-05-30 10:52:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: improve debug
-
-2013-05-30 07:18:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-server.c:
- server: refactor connection setup
- Let the server accept the socket connection and construct a GstRTSPConnection
- from it. Remove the code from the client and let the client only deal with
- a fully configure GstRTSPConnection object.
- We will need this later when the server will configure the connection for
- TLS.
-
-2013-05-30 06:49:20 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: keep the transport object alive
- Keep the transport object alive while we have it as qdata on the
- source.
-
-2013-05-27 12:58:07 +0200 Alexander Schrab <alexas@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-server.c:
- rtsp-server: Do not crash on nmapping of server
- * generate error when gst_rtsp_connection_accept fails
- * do not stop accepting incoming connections because
- accepting a client fails
- https://bugzilla.gnome.org/show_bug.cgi?id=701072
-
-2013-05-24 13:39:50 +0200 Alexander Schrab <alexas@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
- https://bugzilla.gnome.org/show_bug.cgi?id=700953
-
-2013-05-22 03:29:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
-
- * gst/rtsp-server/rtsp-sdp.c:
- rtsp-sdp: Parse framerate caps field and set SDP attribute
- The SDP attribute and its format is described in RFC4566.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
-
-2013-05-22 03:29:30 +0200 Sebastian Rasmussen <sebrn@axis.com>
-
- * gst/rtsp-server/rtsp-sdp.c:
- rtsp-sdp: Parse width/height from caps and set SDP attribute
- The SDP attribute and its format is described in RFC6064.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
-
-2013-04-29 14:46:30 +0200 Patricia Muscalu <patricia@axis.com>
-
- * gst/rtsp-server/rtsp-sdp.c:
- * tests/check/gst/client.c:
- rtsp-sdp: add bandwidth line
- https://bugzilla.gnome.org/show_bug.cgi?id=699220
-
-2013-05-15 10:55:09 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 5edcd85 to 098c0d7
-
-2013-04-23 11:28:39 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * tests/check/gst/media.c:
- tests: add dynamic payloader prepare/unprepare check
-
-2013-04-23 10:27:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: release lock when removing fakesink
-
-2013-04-23 10:16:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: set elements to NULL before removing
- When removing a stream, set the elements to NULL first. This avoids
- element-is-not-in-NULL-state errors when we dispose the elements.
-
-2013-04-22 23:55:48 +0100 Tim-Philipp Müller <tim@centricular.net>
-
- * common:
- Automatic update of common submodule
- From 3cb3d3c to 5edcd85
-
-2013-04-22 17:34:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- media: listen to pad-removed signals
- Listen to the pad-removed signal and remove the stream associated with the
- removed pad.
- Add signal to be notified of the removed pad.
- Remove the fakesink in unprepare()
- Fix signatures of the signal methods
-
-2013-04-22 17:33:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-sdp.c:
- tests: add example of reusable pipelines
-
-2013-04-22 17:32:31 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- stream: add method to get the srcpad
-
-2013-04-22 16:49:39 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * tests/check/gst/media.c:
- check: add media prepare/unprepare test
- See https://bugzilla.gnome.org/show_bug.cgi?id=698376
-
-2013-04-22 16:40:48 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-media.c:
- media: disconnect from signal handlers in unprepare()
- We connected to the pad-added and no-more-pads signals in prepare() so
- we need to disconnect from them in unprepare().
- See https://bugzilla.gnome.org/show_bug.cgi?id=698376
-
-2013-04-22 16:25:17 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-media.c:
- media: don't free streams array
- Don't free the streams array in the unprepare() method, they were not
- added in prepare().
- See https://bugzilla.gnome.org/show_bug.cgi?id=698376
-
-2013-04-22 16:19:35 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-media.c:
- media: don't unref the pipeline in unprepare
- Unprepare() should undo what prepare() does. Because the pipeline is
- not created in prepare(), we should not unref it in unprepare()
-
-2013-04-22 16:09:22 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: clear session and caps for reuse
- Set the session and caps to NULL after unref otherwise we might unref
- them again later.
- See https://bugzilla.gnome.org/show_bug.cgi?id=698376
-
-2013-04-15 12:21:54 +0200 David Svensson Fors <davidsf@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- client: send out teardown signal before tearing down
- The advantage is that in the signal handler you get direct access to
- information about what streams are about to get torn down (in the
- GstRTSPClientState).
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697686
-
-2013-04-15 12:17:34 +0200 David Svensson Fors <davidsf@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: expose connection
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546
-
-2013-04-14 17:58:22 +0100 Tim-Philipp Müller <tim@centricular.net>
-
- * common:
- Automatic update of common submodule
- From aed87ae to 3cb3d3c
-
-2013-04-12 11:34:38 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session-media.h:
- media: add method to get the base_time of the pipeline
- Together with a shared clock, this base-time could eventually be sent to
- the client so that it can reconstruct the exact running-time of the clock
- on the server.
-
-2013-04-09 22:35:28 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/Makefile.am:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- * gst/rtsp-server/rtsp-sdp.c:
- media: add GstNetTimeProvider support
- Add a property to let the media provide a GstNetTimeProvider for its clock.
- Make methods to get the clock and nettimeprovider
- Add a x-gst-clock property to the SDP with the IP and port number of the nettime
- provider and also the current time of the clock. This should make it possible
- for (GStreamer) clients to slave their clock to the server clock.
-
-2013-04-09 21:02:47 +0200 Stefan Sauer <ensonic@users.sf.net>
-
- * common:
- Automatic update of common submodule
- From 04c7a1e to aed87ae
-
-2013-04-09 20:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: wait for buffering to complete
- Wait for buffering to complete before changing the state to the target state.
-
-2013-04-09 20:11:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: small cleanup
-
-2013-03-20 12:33:54 +0100 David Svensson Fors <davidsf@axis.com>
-
- * tests/check/gst/rtspserver.c:
- tests: remove extra unref in test_setup_non_existing_stream
- The unref is not needed anymore, teardown runs without it.
- https://bugzilla.gnome.org/show_bug.cgi?id=696542
-
-2013-03-20 11:28:11 +0100 David Svensson Fors <davidsf@axis.com>
-
- * tests/check/gst/rtspserver.c:
- tests: GSocketService cleanup in test_bind_already_in_use
- Use g_socket_service_stop so the rtspserver test stops listening for
- incoming connections in test_bind_already_in_use.
- https://bugzilla.gnome.org/show_bug.cgi?id=696541
-
-2013-03-22 18:25:07 -0400 Olivier Crête <olivier.crete@collabora.com>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
- Instead use a GWeakRef which is safe to use
- This is a known GLib bug, see:
- https://bugzilla.gnome.org/show_bug.cgi?id=667145
-
-2013-02-22 14:17:29 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- * gst/rtsp-server/rtsp-sdp.c:
- * tests/check/gst/media.c:
- * tests/check/gst/rtspserver.c:
- rtsp-media/client: Reply to PLAY request with same type of Range
- Remember the type of Range from the PLAY request and use the same type for
- the reply.
-
-2013-03-18 09:25:54 +0100 Patricia Muscalu <patricia@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- * tests/check/gst/client.c:
- rtsp-client: expose uri
-
-2013-03-13 17:46:58 -0400 Olivier Crête <olivier.crete@collabora.com>
-
- * tests/check/gst/mediafactory.c:
- tests: Hold ref while creating second media
- To test if the media aren't shared, make sure we keep the first one while creating a second
- otherwise the same memory address may be reused.
-
-2013-03-12 00:10:18 +0000 Tim-Philipp Müller <tim@centricular.net>
-
- * configure.ac:
- configure: remove out-of-date comment
-
-2013-03-12 00:05:49 +0000 Tim-Philipp Müller <tim@centricular.net>
-
- * .gitignore:
- .gitignore: ignore more build files
-
-2013-03-12 00:03:36 +0000 Tim-Philipp Müller <tim@centricular.net>
-
- * tests/check/Makefile.am:
- tests: use right _LIBS variable for gst-plugins-base libs
-
-2013-03-11 11:35:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/Makefile.am:
- check: add librtp to libs
-
-2013-02-20 19:37:51 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * tests/check/gst/rtspserver.c:
- tests: Add test to check selecting a port the server will send from
-
-2013-02-20 18:30:01 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * tests/check/gst/rtspserver.c:
- tests: Make sure packets are actually received
-
-2013-02-19 18:27:20 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: Select unicast address from pool if appropriate
-
-2013-02-19 16:43:08 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: Properties are always there in Gst 1.0
-
-2013-02-19 16:36:20 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * tests/check/gst/addresspool.c:
- tests: Add tests for unicast addresses in pool
-
-2013-02-20 14:26:03 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * gst/rtsp-server/rtsp-address-pool.c:
- * tests/check/gst/addresspool.c:
- address-pool: Verify that multicast addresses are used for multicast and vice-versa
-
-2013-02-19 16:34:16 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * docs/libs/gst-rtsp-server-sections.txt:
- * gst/rtsp-server/rtsp-address-pool.c:
- * gst/rtsp-server/rtsp-address-pool.h:
- * gst/rtsp-server/rtsp-stream.c:
- * tests/check/gst/addresspool.c:
- address-pool: Add unicast addresses
-
-2013-02-19 13:19:41 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * configure.ac:
- * gst/rtsp-server/rtsp-server.c:
- * tests/check/gst/rtspserver.c:
- rtsp-server: Limit the number of threads per server instance
- If we exceed the maximum, just round robin the clients over the existing
- threads.
-
-2013-02-19 12:31:23 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * gst/rtsp-server/rtsp-server.c:
- rtsp-server: No need to store the GMainContext in the client context
-
-2013-02-18 20:22:18 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * tests/check/gst/rtspserver.c:
- tests: Add test for client disconnection
-
-2013-02-18 20:15:41 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * tests/check/gst/rtspserver.c:
- tests: Test client and session timeouts with multiple threads
-
-2013-02-18 14:59:58 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * gst/rtsp-server/rtsp-address-pool.c:
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-mount-points.c:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session-pool.c:
- * gst/rtsp-server/rtsp-session.c:
- Document locking and its order
-
-2013-02-15 20:02:31 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * tests/check/gst/rtspserver.c:
- tests: Test that slow DESCRIBE don't block other clients
-
-2013-02-14 19:52:09 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * tests/check/gst/client.c:
- tests: Add tests for client-requested multicast address
-
-2013-02-14 13:44:54 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * docs/libs/gst-rtsp-server-sections.txt:
- docs: Put the various functions in the right sections
-
-2013-02-14 13:38:07 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * docs/libs/gst-rtsp-server-docs.sgml:
- * docs/libs/gst-rtsp-server-sections.txt:
- * gst/rtsp-server/rtsp-address-pool.c:
- * gst/rtsp-server/rtsp-address-pool.h:
- docs: Generate docs for GstRTSPAddressPool
-
-2013-02-13 18:32:20 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- client: Check client provided addresses against the address pool
-
-2013-02-13 18:01:43 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * gst/rtsp-server/rtsp-address-pool.c:
- * gst/rtsp-server/rtsp-address-pool.h:
- * tests/check/gst/addresspool.c:
- address-pool: Add API to request a specific address from the pool
- Also add relevant unit tests.
-
-2013-02-12 19:34:24 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * tests/check/gst/mediafactory.c:
- tests: Check the passing around of a RTSPAddressPool
- Make sure the RTSPAddressPool is propagated from the MediaFactory all the
- way down to the stream.
-
-2013-02-12 16:34:37 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * tests/check/gst/addresspool.c:
- tests: Add more tests for the address pool
-
-2013-02-12 16:29:25 -0500 Olivier Crête <olivier.crete@collabora.com>
-
- * gst/rtsp-server/rtsp-address-pool.c:
- address-pool: Fix off by one error
- When splitting a port range, the port after a skip is not part of range.
-
-2013-03-07 00:04:19 +0000 Tim-Philipp Müller <tim@centricular.net>
-
- * common:
- Automatic update of common submodule
- From 2de221c to 04c7a1e
-
-2013-02-07 16:18:08 -0600 George McCollister <george.mccollister@gmail.com>
-
- * configure.ac:
- configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
- AM_CONFIG_HEADER was removed in automake 1.13
- https://bugzilla.gnome.org/show_bug.cgi?id=693368
-
-2013-01-28 20:45:44 +0100 Stefan Sauer <ensonic@users.sf.net>
-
- * common:
- Automatic update of common submodule
- From a942293 to 2de221c
-
-2013-01-28 10:31:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: make sure the watch exists while sending data
- Protect the send_func with a lock. This allows us to wait for sending
- to complete before changing the send_func and user_data. We add an
- extra ref to the watch to make sure that it remains valid during
- sending.
- When closing the connection, set the send_func to NULL
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=692433
-
-2013-01-16 12:16:32 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-
- * tests/check/Makefile.am:
- tests: use GST_*_1_0 environment variables everywhere
- The _1_0 suffixed environment variables override the
- non-suffixed ones, so if we're in an environment that
- sets the _1_0 suffixed ones, such as jhbuild, we need
- to set those to make sure ours actually always get
- used.
-
-2013-01-15 15:09:24 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From acb04d9 to a942293
-
-2012-12-14 11:58:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- rtsp-client: set the client backlog
- Set the client backlog to a reasonable default
-
-2012-12-04 09:47:35 +0100 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-media.c:
- rtsp-media: Make the element a constructor parameter
- https://bugzilla.gnome.org/show_bug.cgi?id=689594
-
-2012-12-04 01:05:31 +0100 Sebastian Rasmussen <sebras@hotmail.com>
-
- * docs/libs/Makefile.am:
- docs: Link with gcov library when gcov is enabled
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=689583
-
-2012-11-30 15:03:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: match prepare with unprepare
- Really unprepare when there were an equal amount of prepare calls.
-
-2012-11-30 14:58:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: media has to be unprepared in finalize
- Because unprepare takes away the last ref on the media.
-
-2012-11-30 14:36:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it"
- This reverts commit ba5b78ff2ff223049188eb456e228c709ccd3e05.
- We can't use the refcount to trigger unprepare because it is the unprepare call
- that removes the last refcount after all messages are consumed. What we should
- probably do is make a prepared refcount and only unprepare when the refcount
- reaches 0.
-
-2012-11-30 13:35:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: let the source unref the last media ref
- the last ref to the media is held by the source so we don't need to add more ref
- and unrefs, we simply destroy the media when the source is gone.
-
-2012-11-30 12:54:10 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: improve debug
-
-2012-11-30 12:53:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: check state
- Make sure we are in the right state when collecting the position and duration.
- Only make ourselves PREPARED when we were previously PREPARING.
-
-2012-11-30 10:05:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: use g_object_ref/unref for GObjects
-
-2012-11-30 07:05:25 +0100 Alessandro Decina <alessandro.d@gmail.com>
-
- * gst/rtsp-server/rtsp-client.c:
- client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it
- Calling gst_rtsp_media_unprepare breaks shared medias. Just unref
- GstRTSPMedia instances and let gst_rtsp_media_finalize unprepare when a media
- isn't being used anymore.
-
-2012-11-30 06:17:46 +0100 Alessandro Decina <alessandro.d@gmail.com>
-
- * gst/rtsp-server/rtsp-media.c:
- Fix compiler warning
-
-2012-11-30 06:14:49 +0100 Alessandro Decina <alessandro.d@gmail.com>
-
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- Add missing g_type_class_add_private in GstRTSPMediaFactoryURI
-
-2012-11-29 17:21:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-session-media.h:
- small cleanup
-
-2012-11-29 17:20:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- * tests/check/gst/media.c:
- media: avoid element leak
-
-2012-11-29 17:20:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: require an element in media constructor
-
-2012-11-29 17:07:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- Revert "client: TEARDOWN brings that state to Init again"
- This reverts commit 4b61fdad85a3ca84752bf074fdb2fa203954b32e.
- The object is already disposed, there is no point in setting the state.
-
-2012-11-29 12:30:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: TEARDOWN brings that state to Init again
-
-2012-11-29 11:11:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * docs/libs/gst-rtsp-server-sections.txt:
- * examples/test-auth.c:
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- * gst/rtsp-server/rtsp-media-factory-uri.h:
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- * gst/rtsp-server/rtsp-mount-points.c:
- * gst/rtsp-server/rtsp-mount-points.h:
- * gst/rtsp-server/rtsp-sdp.c:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session-media.h:
- * gst/rtsp-server/rtsp-session-pool.c:
- * gst/rtsp-server/rtsp-session-pool.h:
- * gst/rtsp-server/rtsp-session.c:
- * gst/rtsp-server/rtsp-session.h:
- * gst/rtsp-server/rtsp-stream-transport.c:
- * gst/rtsp-server/rtsp-stream-transport.h:
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- * tests/check/gst/media.c:
- rtsp: make object details private
- Make all object details private
- Add methods to access private bits
-
-2012-11-28 14:50:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/Makefile.am:
- * tests/check/gst/media.c:
- tests: add media tests
-
-2012-11-28 14:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: check if prepared for some methods
- Check that the media object is prepared before doing seek and getting the
- current position etc.
- Add some g_return checks.
-
-2012-11-28 12:40:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/Makefile.am:
- * tests/check/gst/mediafactory.c:
- tests: add mediafactory test
-
-2012-11-28 12:40:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: improve debug
-
-2012-11-28 12:39:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- media: unref pipeline in finalize to avoid leaking it
-
-2012-11-28 12:10:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- * gst/rtsp-server/rtsp-media.c:
- rtsp: use gst_object_unref on GstObjects
-
-2012-11-28 12:10:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- media-factory: require an url
-
-2012-11-28 11:40:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-uri.c:
- examples: fix include
-
-2012-11-28 11:17:27 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.h:
- server: remove unused include
-
-2012-11-28 11:07:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/Makefile.am:
- * tests/check/gst/mountpoints.c:
- tests: add test for mountpoints
-
-2012-11-28 11:05:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: fix factory leak
- Keep the factory in the state object only for authorization checks and make
- sure we unref it on failure. Also don't keep invalid objects in the state
- object.
-
-2012-11-28 10:40:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-mount-points.c:
- mounts: add g_return_if guards
-
-2012-11-27 12:51:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/gst/client.c:
- tests: add more tests
-
-2012-11-27 12:33:02 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: improve debug
-
-2012-11-27 12:24:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: improve debug and fix leaks
- Cleanup the uri and session when there is a bad request.
-
-2012-11-27 12:17:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * common:
- update common
-
-2012-11-27 12:13:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/gst/client.c:
- test: add test for session in options request
-
-2012-11-27 12:11:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: use 454 when session can't be found
- We should use 454 when a session can't be found because there was no session
- pool configured in the server. This is not a server configuration problem
- because the server on which the request is done might not be the same one that
- will keep the sessions for us and so it does not need to support sessions.
-
-2012-11-27 11:17:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: only free connection when there is one
- It's possible that the client doesn't have a connection when we try to free it.
-
-2012-11-27 11:17:31 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/Makefile.am:
- * tests/check/gst/client.c:
- tests: add unit test for the client object
-
-2012-11-26 17:35:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: small cleanup
-
-2012-11-26 17:34:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.h:
- client: remove unused include
-
-2012-11-26 17:34:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: fix compilation
-
-2012-11-26 17:28:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: call destroy without the lock
-
-2012-11-26 17:20:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: make the client usable without a socket
- Make a method to let the client handle a message and a callback when the client
- wants us to send a response message back. This makes it possible to also use the
- client object without the sockets, which should make it easier to test.
-
-2012-11-26 16:45:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: small cleanup
-
-2012-11-26 16:39:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * docs/libs/gst-rtsp-server-sections.txt:
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-server.c:
- client: remove reference to server
- We don't need to keep a ref to the server
-
-2012-11-26 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: add locking
- Also add some g_return_if()
-
-2012-11-26 13:37:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: log more errors
-
-2012-11-26 13:35:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: fix compilation
-
-2012-11-26 13:16:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: add generic close-after-send support
- Add a property to send_response() to close the connection after the response has
- been sent to the client.
-
-2012-11-26 12:34:05 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * docs/README:
- * docs/libs/gst-rtsp-server-docs.sgml:
- * docs/libs/gst-rtsp-server-sections.txt:
- * docs/libs/gst-rtsp-server.types:
- * examples/test-auth.c:
- * examples/test-launch.c:
- * examples/test-mp4.c:
- * examples/test-multicast.c:
- * examples/test-multicast2.c:
- * examples/test-ogg.c:
- * examples/test-readme.c:
- * examples/test-sdp.c:
- * examples/test-uri.c:
- * examples/test-video.c:
- * gst/rtsp-server/Makefile.am:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-media-mapping.c:
- * gst/rtsp-server/rtsp-media-mapping.h:
- * gst/rtsp-server/rtsp-mount-points.c:
- * gst/rtsp-server/rtsp-mount-points.h:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session-pool.c:
- * gst/rtsp-server/rtsp-session-pool.h:
- * tests/check/gst/rtspserver.c:
- MediaMapping -> MountPoints
- Describes better what the object manages.
-
-2012-11-26 09:36:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- configure: bump required version of -base
-
-2012-11-21 17:21:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: fix seeking
-
-2012-11-21 16:41:56 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- media: support more Range formats
- Use the new -base methods to convert the Range string into a seek start and stop
- value.
-
-2012-11-21 16:41:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-launch.c:
- examples: fix whitespace
-
-2012-11-20 13:34:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-auth.c:
- test-auth: add example of how to remove sessions
- Add an example of the session filter api.
-
-2012-11-20 12:47:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-uri.c:
- test-uri: remove mapping example
-
-2012-11-20 12:47:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-uri.c:
- test-uri: fix callback signature
-
-2012-11-20 12:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- factory: keep ref to factory while media active
- While the media from a factory is alive, keep a ref to the factory.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
-
-2012-11-20 12:29:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- factory-uri: add some debug
-
-2012-11-20 12:24:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: set udp sources to PLAYING
- Set the UDP sources to PLAYING and locked state before we add it to the pipeline
- so that it doesn't cause our pipeline to produce ASYNC-DONE.
-
-2012-11-20 12:10:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- factory-uri: take ref to factory
- Take a ref to the factory that we place in our list.
-
-2012-11-20 11:30:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/Makefile.am:
- * tests/test-reuse.c:
- test: add test for server reuse
- See https://bugzilla.gnome.org/show_bug.cgi?id=688395
-
-2012-11-15 14:02:37 +0100 David Svensson Fors <davidsf@axis.com>
-
- * gst/rtsp-server/rtsp-server.c:
- server: start and stop multiple times
- Stop listening on the RTSP port when the GSource is removed, so clients
- can't connect and the server can be started again.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
-
-2012-11-20 11:24:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- server: fix small leak
-
-2012-11-20 09:42:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: unref source in finish_unprepare
- The source is created in prepare, unref it in finish_unprepare.
- See https://bugzilla.gnome.org/show_bug.cgi?id=688707
-
-2012-11-19 15:47:08 +0100 David Svensson Fors <davidsf@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media.c:
- rtsp-media: remove bus watch before finalizing
- * A GDestroyNotify function is set for the bus watch in gst_rtsp_media_prepare.
- * An extra media ref is added for the bus watch. This extra ref is unreffed by
- the GDestroyNotify function.
- * gst_rtsp_media_unprepare destroys the source so the bus watch is removed.
- * GstRTSPClient, which calls gst_rtsp_media_prepare, also calls
- gst_rtsp_media_unprepare before unreffing the media.
- This way, the bus watch will be removed before the media is finalized.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688707
-
-2012-11-17 14:51:52 +0100 Alessandro Decina <alessandro.d@gmail.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: wait until the TEARDOWN response is sent to close the connection
- Responses can be sent async so we need to wait until the TEARDOWN response has
- been written before we close the connection to the client. This avoids the risk
- of writing/polling closed sockets.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688535
-
-2012-11-19 15:44:27 +0100 David Svensson Fors <davidsf@axis.com>
-
- * gst/rtsp-server/rtsp-stream.c:
- rtsp-stream: plug socket leak
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
-
-2012-11-19 11:31:12 +0000 Tim-Philipp Müller <tim@centricular.net>
-
- * common:
- Automatic update of common submodule
- From 6bb6951 to a72faea
-
-2012-11-17 00:11:27 +0000 Tim-Philipp Müller <tim@centricular.net>
-
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- rtsp-server: don't use deprecated API
-
-2012-11-17 00:03:42 +0000 Tim-Philipp Müller <tim@centricular.net>
-
- * gst/rtsp-server/rtsp-client.c:
- rtsp-client: fix unused-but-set-variable compiler warning
- rtsp-client.c:1260:21: error: variable 'protocols' set but not used
-
-2012-11-15 17:11:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * TODO:
- * docs/libs/gst-rtsp-server-sections.txt:
- * gst/rtsp-server/rtsp-client.c:
- rtsp: cleanups
-
-2012-11-15 16:52:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/Makefile.am:
- * examples/test-multicast2.c:
- examples: add another multicast example
- Add an example for how to configure separate multicast ranges for each media
- stream.
-
-2012-11-15 16:21:51 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-multicast.c:
- test: set shared
-
-2012-11-15 16:18:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session-media.h:
- * gst/rtsp-server/rtsp-stream-transport.c:
- * gst/rtsp-server/rtsp-stream-transport.h:
- stream: use the address managed by the stream
- Use the address managed by the stream for multicast. This allows us to have 1
- multicast address for each stream.
- Because the address is now managed by the stream we don't have to pass it around
- anymore.
- Set the address pool on the streams.
-
-2012-11-15 16:15:20 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-stream.c:
- rtsp: improve debug
-
-2012-11-15 15:41:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- media: add signal for new streams
- This allows applications to listen for new streams and configure properties on
- them, like the address pool.
-
-2012-11-15 15:41:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: configure address pool in new streams
-
-2012-11-15 15:36:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- stream: add methods to deal with address pool
- Add methods to get and set the address pool for the stream
- Add method to allocate and get the multicast addresses for this stream.
-
-2012-11-15 15:32:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * docs/libs/gst-rtsp-server-sections.txt:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- media: remove MTU property
- It is a stream property
-
-2012-11-15 15:29:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: set blocksize only on stream
- Set the blocksize only on the current stream.
-
-2012-11-15 13:52:07 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: share src and sink sockets
- the allocated socket is in the used-socket property, not socket.
-
-2012-11-15 13:25:14 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-address-pool.c:
- * gst/rtsp-server/rtsp-address-pool.h:
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session-media.h:
- * gst/rtsp-server/rtsp-stream-transport.c:
- * gst/rtsp-server/rtsp-stream-transport.h:
- * tests/check/gst/addresspool.c:
- rtsp: make address-pool return an address object
- Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to
- store more info in the structure and allows us to more easily return the address
- to the right pool when no longer needed.
- Pass the address to the StreamTransport so that we can return it to the pool
- when the stream transport is freed or changed.
-
-2012-11-15 13:22:54 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/Makefile.am:
- * examples/test-multicast.c:
- examples: add multicast example
- Show how to set up the multicast address pool so that media can be
- server with multicast.
-
-2012-11-14 17:23:59 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- rtsp: use AddressPool
- Remove the multicast_group property.
- Use the configured addresspool to allocate multicast addresses.
-
-2012-11-14 16:17:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-address-pool.c:
- * gst/rtsp-server/rtsp-address-pool.h:
- address-pool: add clear method
-
-2012-11-14 16:10:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-address-pool.c:
- address-pool: small cleanups
-
-2012-11-14 15:50:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * tests/check/Makefile.am:
- * tests/check/gst/addresspool.c:
- tests: add addresspool unit test
-
-2012-11-14 15:49:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/Makefile.am:
- * gst/rtsp-server/rtsp-address-pool.c:
- * gst/rtsp-server/rtsp-address-pool.h:
- address-pool: add object to manage multicast addresses
- Make an object that can manage a rage of multicast addresses and ports.
-
-2012-11-13 12:05:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- server: set default max-threads property
-
-2012-11-13 11:54:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: wait for concurrent _prepare
- If a prepare is busy, wait for the result.
-
-2012-11-13 11:49:08 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: add lock around message handler
- We don't want to dispatch messages while we are still processing the result of
- the state change.
-
-2012-11-13 11:15:35 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- media: add lock to protect state changes
-
-2012-11-13 11:14:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- stream: add locking
-
-2012-11-12 17:11:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream-transport.c:
- * gst/rtsp-server/rtsp-stream-transport.h:
- * gst/rtsp-server/rtsp-stream.c:
- stream-transport: add keep-alive method
-
-2012-11-12 17:06:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream-transport.c:
- * gst/rtsp-server/rtsp-stream-transport.h:
- * gst/rtsp-server/rtsp-stream.c:
- stream-transport: add method to handle RTP/RTCP
- Call new methods instead of poking into the structures directly.
-
-2012-11-12 16:51:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session-media.h:
- session-media: add locking
-
-2012-11-12 16:42:37 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-session.c:
- * gst/rtsp-server/rtsp-session.h:
- session: add locking
-
-2012-11-12 16:30:16 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- server: free old socket
-
-2012-11-12 16:18:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-mapping.c:
- * gst/rtsp-server/rtsp-media-mapping.h:
- mapping: add locking
-
-2012-11-12 16:14:19 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- media-factory: add locking
-
-2012-11-12 16:03:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-auth.h:
- auth: add locking
-
-2012-11-12 15:53:28 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- server: add max-thread property
-
-2012-11-12 15:29:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- server: use a threadpool for the mainloops
-
-2012-11-12 14:30:43 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- client: rename method
- gst_rtsp_client_create_from_socket -> gst_rtsp_client_use_socket: we
- don't really create the client from the socket, we use the socket for the
- client.
-
-2012-11-12 14:09:09 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-server.c:
- server: rework maincontext handling in clients
- Make a separate method to attach a client to a MainContext.
- Let the server decide in what GMainContext the client will operate and give this
- context to the client in attach. Then the server can later decide to use a
- separate thread for each client or just use the mainthread.
-
-2012-11-12 12:40:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-session.c:
- * gst/rtsp-server/rtsp-session.h:
- session: move session header code in session object
-
-2012-11-04 00:14:25 +0000 Tim-Philipp Müller <tim@centricular.net>
-
- * COPYING:
- * COPYING.LIB:
- * examples/test-auth.c:
- * examples/test-launch.c:
- * examples/test-mp4.c:
- * examples/test-ogg.c:
- * examples/test-readme.c:
- * examples/test-sdp.c:
- * examples/test-uri.c:
- * examples/test-video.c:
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- * gst/rtsp-server/rtsp-media-factory-uri.h:
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media-mapping.c:
- * gst/rtsp-server/rtsp-media-mapping.h:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- * gst/rtsp-server/rtsp-params.c:
- * gst/rtsp-server/rtsp-params.h:
- * gst/rtsp-server/rtsp-sdp.c:
- * gst/rtsp-server/rtsp-sdp.h:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session-media.h:
- * gst/rtsp-server/rtsp-session-pool.c:
- * gst/rtsp-server/rtsp-session-pool.h:
- * gst/rtsp-server/rtsp-session.c:
- * gst/rtsp-server/rtsp-session.h:
- * gst/rtsp-server/rtsp-stream-transport.c:
- * gst/rtsp-server/rtsp-stream-transport.h:
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- * tests/check/gst/rtspserver.c:
- * tests/test-cleanup.c:
- Fix FSF address
-
-2012-10-28 13:48:44 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session.c:
- rtsp-server: added annotations to indicate type of ownership transfer of return values
- https://bugzilla.gnome.org/show_bug.cgi?id=680777
-
-2012-10-28 15:37:51 +0000 Tim-Philipp Müller <tim@centricular.net>
-
- * configure.ac:
- No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
-
-2012-10-28 15:09:04 +0000 Tim-Philipp Müller <tim@centricular.net>
-
- * Makefile.am:
- * bindings/Makefile.am:
- * bindings/vala/Makefile.am:
- * bindings/vala/gst-rtsp-server-0.10.deps:
- * bindings/vala/gst-rtsp-server-0.10.vapi:
- * bindings/vala/packages/gst-rtsp-server-0.10.deps:
- * bindings/vala/packages/gst-rtsp-server-0.10.files:
- * bindings/vala/packages/gst-rtsp-server-0.10.gi:
- * bindings/vala/packages/gst-rtsp-server-0.10.metadata:
- * bindings/vala/packages/gst-rtsp-server-0.10.namespace:
- * configure.ac:
- bindings: remove vala bindings
- They'll be reunited with the other GStreamer bindings
- https://bugzilla.gnome.org/show_bug.cgi?id=680777
-
-2012-10-28 00:23:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session-media.h:
- * gst/rtsp-server/rtsp-stream-transport.c:
- * gst/rtsp-server/rtsp-stream-transport.h:
- rtsp: only create transport when needed
- Only create the StreamTransport when configured.
-
-2012-10-27 23:53:35 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: small cleanup
-
-2012-10-27 23:49:24 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-stream-transport.c:
- * gst/rtsp-server/rtsp-stream-transport.h:
- rtsp: refactor configuration of transport
- Move the configuration of the transport to a place where it makes
- more sense.
-
-2012-10-27 21:26:55 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: refactor transport parsing
-
-2012-10-27 21:05:03 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: refuse to change the MTU on shared media
- If we change the MTU of chared media, it changes for all clients.
- We don't want to set the MTU to something large for clients that
- stream over UDP.
-
-2012-10-27 11:53:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-mp4.c:
- * gst/rtsp-server/rtsp-media.c:
- small fixes to docs and debug
-
-2012-10-26 17:29:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-stream.c:
- stream: transports must already have been removed
-
-2012-10-26 17:28:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- stream: improve join and leave of the pipeline
- simplify code
- Do the cleanup properly
- Add some docs
-
-2012-10-26 15:23:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: move unprepare below default implementation
- Makes it easier to find the default implementation
-
-2012-10-26 15:21:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: signal unprepared when we actually finish
-
-2012-10-26 15:19:23 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: no need to unlock, unprepare does that when needed
-
-2012-10-26 12:33:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * docs/libs/gst-rtsp-server-sections.txt:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media-mapping.c:
- * gst/rtsp-server/rtsp-media.h:
- * gst/rtsp-server/rtsp-params.c:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-session-pool.h:
- * gst/rtsp-server/rtsp-session.c:
- * gst/rtsp-server/rtsp-session.h:
- * gst/rtsp-server/rtsp-stream-transport.h:
- * gst/rtsp-server/rtsp-stream.h:
- docs: update docs
-
-2012-10-26 12:04:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media-mapping.h:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- * gst/rtsp-server/rtsp-server.h:
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- rtsp: fix MTU setting
- Fix setting of the MTU. There is no need for a vmethod.
-
-2012-10-26 11:02:43 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * docs/README:
- docs: update docs
-
-2012-10-26 11:24:55 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-
- * configure.ac:
- configure: bump version number after refactoring
-
-2012-10-25 21:29:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/Makefile.am:
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- * gst/rtsp-server/rtsp-sdp.c:
- * gst/rtsp-server/rtsp-session-media.c:
- * gst/rtsp-server/rtsp-session-media.h:
- * gst/rtsp-server/rtsp-session.c:
- * gst/rtsp-server/rtsp-session.h:
- * gst/rtsp-server/rtsp-stream-transport.c:
- * gst/rtsp-server/rtsp-stream-transport.h:
- * gst/rtsp-server/rtsp-stream.c:
- * gst/rtsp-server/rtsp-stream.h:
- rtsp: massive refactoring
- Make GObjects from the remaining simple structures.
- Remove GstRTSPSessionStream, it's not needed.
- Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter
- Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how
- a GstRTSPStream should be transported to a client.
- Rename GstRTSPMediaFactory::get_element -> create_element because that
- more accurately describes what it does.
- Make nice methods instead of poking in the structures.
- Move some methods inside the relevant object source code.
- Use GPtrArray to store objects instead of plain arrays, it is more
- natural and allows us to more easily clean up.
- Move the allocation of udp ports to the Stream object. The Stream object
- contains the elements needed to stream the media to a client.
- Improve the prepare and unprepare methods. Unprepare should now undo
- everything prepare did. Improve also async unprepare when doing EOS on
- shutdown. Make sure we always unprepare correctly.
-
-2012-10-23 22:11:17 +0200 Sebastian Rasmussen <sebrn@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- rtsp-client: Unref server address clients connected to
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
-
-2012-10-22 16:09:24 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-server.c:
- rtsp-server: don't ref server socket if it is NULL
- Fixes test_bind_already_in_use unit test again after commit 6a497440.
- https://bugzilla.gnome.org/show_bug.cgi?id=686644
-
-2012-10-22 16:29:09 +0200 Sebastian Rasmussen <sebrn@axis.com>
-
- * tests/check/Makefile.am:
- tests: Add libgio link dependency
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
-
-2012-10-01 20:03:43 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
-
- * gst/rtsp-server/rtsp-media-mapping.c:
- * gst/rtsp-server/rtsp-media-mapping.h:
- rtsp-media-mapping: rename find_media vfunc to find_factory
- The virtual method and class method should have the same name
- so it is correctly represented in GIR file
- https://bugzilla.gnome.org/show_bug.cgi?id=680777
-
-2012-10-01 19:46:15 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
-
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media-mapping.c:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-session-pool.c:
- * gst/rtsp-server/rtsp-session.c:
- rtsp-server: fixed comments and GIR annotations
- https://bugzilla.gnome.org/show_bug.cgi?id=680777
-
-2012-10-12 07:18:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
-
- * gst/rtsp-server/rtsp-media-mapping.c:
- media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
-
-2012-10-12 07:08:57 +0200 Alessandro Decina <alessandro.d@gmail.com>
-
- * gst/rtsp-server/rtsp-server.c:
- rtsp-server: allow binding on port 0 (binds on a random port)
-
-2012-10-12 06:21:24 +0200 Alessandro Decina <alessandro.d@gmail.com>
-
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- rtsp-server: add bound-port property
- bound-port can be used to retrieve the port number when the server is bound on
- port 0, which binds on a random port.
-
-2012-10-12 06:11:36 +0200 Alessandro Decina <alessandro.d@gmail.com>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media-factory.h:
- rtsp-media-factory: make ::get_element overridable by GI bindings
- The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
- for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
- as the invoker for ::get_element(), making it overridable by GI generated
- bindings.
-
-2012-10-12 06:07:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
-
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- rtsp-media-factory-uri: don't autoplug parsers in a loop
- Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
- h264parse forever.
-
-2012-10-06 15:49:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
-
- * gst/rtsp-server/Makefile.am:
- Explicitly link against gio. Fix link error on mac.
-
-2012-10-10 11:13:10 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
-
- * gst/rtsp-server/rtsp-session.c:
- session: add ttl to the transport header in SETUP
- See https://bugzilla.gnome.org/show_bug.cgi?id=685561
-
-2012-10-10 11:06:02 +0200 Ognyan Tonchev <ognyan.tonchev at axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-media.c:
- client: Use client transport settings for multicast if allowed.
- This patch makes it possible for the client to send transport settings for
- multicast (destination && ttl). Client settings must be explicitly allowed or
- the server will use its own settings.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685561
-
-2012-10-06 15:02:27 +0100 Tim-Philipp Müller <tim@centricular.net>
-
- * common:
- Automatic update of common submodule
- From 6c0b52c to 6bb6951
-
-2012-10-01 16:13:50 +0200 Patricia Muscalu <patricia@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- rtsp-client: do not destroy the rtsp watch
- Don't destroy the client watch while dispatching. The rtsp watch is
- automatically destroyed after the rtsp watch function closed() has
- been called.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=685220
-
-2012-09-22 16:11:48 +0100 Tim-Philipp Müller <tim@centricular.net>
-
- * common:
- Automatic update of common submodule
- From 4f962f7 to 6c0b52c
-
-2012-09-10 16:25:57 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-media.c:
- media: fix check for seekability
-
-2012-09-07 17:14:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: use more GIO
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
-
-2012-09-07 17:14:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- server: remove obsolete includes
-
-2012-09-03 17:33:17 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
-
- rtsp-media: also initialize transports in on_ssrc_active (bug #683304)
- * gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
- be available in "on_new_ssrc". The transports are added in
- gst_rtsp_media_set_state when going to PLAYING state. However,
- "on_new_ssrc" might be called before this happens.
- https://bugzilla.gnome.org/show_bug.cgi?id=683304
-
-2012-09-03 10:48:14 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- rtsp-client: add signals for rtsp requests (fixes #683287)
-
-2012-08-30 12:03:27 -0700 Aleix Conchillo Flaque <aleix@oblong.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- add new-session signal to rtsp-client (fixes #683058)
-
-2012-08-22 13:34:55 +0200 Stefan Sauer <ensonic@users.sf.net>
-
- * common:
- Automatic update of common submodule
- From 668acee to 4f962f7
-
-2012-08-15 15:54:32 +0200 Patricia Muscalu <patricia@axis.com>
-
- * gst/rtsp-server/rtsp-server.c:
- * tests/check/gst/rtspserver.c:
- rtsp-server: fixed segfault in gst_rtsp_server_create_socket
- Do not assume that *error is set in g_socket_address_enumerator_next.
- Added test_bind_already_in_use unit-test.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
-
-2012-08-05 16:43:53 +0100 Tim-Philipp Müller <tim@centricular.net>
-
- * common:
- Automatic update of common submodule
- From 94ccf4c to 668acee
-
-2012-07-18 15:54:49 +0200 Patricia Muscalu <patricia@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- rtsp-client: make create_sdp virtual method
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
-
-2012-07-23 08:48:25 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 98e386f to 94ccf4c
-
-2012-07-10 11:39:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: fix docs
-
-2012-07-03 18:06:00 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- rtsp-server: use an existing socket to establish HTTP tunnel
- Make it possible to transfer a socket from an HTTP server to be used as
- an RTSP over HTTP tunnel.
-
-2012-07-03 13:26:30 +0200 Ognyan Tonchev <ognyan@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- rtsp: Handle the blocksize parameter
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
-
-2012-06-25 14:28:10 +0200 Sebastian Rasmussen <sebrn@axis.com>
-
- * tests/check/Makefile.am:
- * tests/check/gst/rtspserver.c:
- Have unit test get header from source dir, not installed dir
- This makes compilation of unit tests work in a build directory other
- than the source directory.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
-
-2012-06-23 15:06:11 +0100 Tim-Philipp Müller <tim@centricular.net>
-
- * gst/rtsp-server/rtsp-media.c:
- rtsp-media: update for gst_element_make_from_uri() changes
-
-2012-06-19 15:25:36 +0200 David Svensson Fors <davidsf@axis.com>
-
- * configure.ac:
- * tests/Makefile.am:
- * tests/check/Makefile.am:
- * tests/check/gst/rtspserver.c:
- rtsp: add unit test
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
-
-2012-06-13 11:43:17 +0200 David Svensson Fors <davidsf@axis.com>
-
- * gst/rtsp-server/rtsp-media.c:
- rtsp-media: don't collect media stats when going to NULL
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
-
-2012-06-14 09:59:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: don't leak transports
-
-2012-06-12 14:45:39 +0200 David Svensson Fors <davidsf@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- rtsp-client: free transport on no_stream in SETUP handler
-
-2012-06-12 14:33:35 +0200 David Svensson Fors <davidsf@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- rtsp-client: changed session media iteration
- In client_unlink_session: now don't iterate in session->medias
- list where items are removed by gst_rtsp_session_release_media.
- Instead, repeatedly remove the first item.
-
-2012-06-12 13:39:35 +0200 David Svensson Fors <davidsf@axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- rtsp-client: don't use g_object_unref on GstRTSPSessionMedia
- GstRTSPSessionMedia is not a GObject type. When the
- GstRTSPSession is freed, it will free the media.
-
-2012-06-12 13:36:57 +0200 David Svensson Fors <davidsf@axis.com>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- factory: plug pad leak in collect_streams
- In gst_rtsp_media_factory_collect_streams: unref the srcpad that
- was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
- will take one reference, and the other reference will otherwise
- give a memory leak.
-
-2012-05-25 16:43:38 +0200 Sebastian Rasmussen <sebrn@axis.com>
-
- * configure.ac:
- configure: suppress some warnings when debug is disabled
- Warnings about unused variables should be suppressed if core has the
- debug system disabled.
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
-
-2012-06-09 17:41:05 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-
- * docs/libs/Makefile.am:
- docs: fix build in uninstalled setup
- Include gst-plugins-base libs properly.
-
-2012-05-25 16:38:15 +0200 Sebastian Rasmussen <sebrn@axis.com>
-
- * docs/libs/gst-rtsp-server.types:
- docs: include headers defining rtsp-server object types
- Fixes compiler warnings during docs build.
- https://bugzilla.gnome.org/show_bug.cgi?id=676824
-
-2012-05-25 17:11:53 +0200 Sebastian Rasmussen <sebrn@axis.com>
-
- * configure.ac:
- configure: Add warning flags for compiler when configuring
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676824
-
-2012-06-08 15:07:06 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 03a0e57 to 98e386f
-
-2012-06-06 18:20:49 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 1fab359 to 03a0e57
-
-2012-06-06 14:49:02 +0200 David Svensson Fors <davidsf at axis.com>
-
- * gst/rtsp-server/rtsp-client.c:
- client: fix GSocketAddress leak in gst_rtsp_client_accept
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463
-
-2012-06-01 10:30:58 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From f1b5a96 to 1fab359
-
-2012-05-31 13:11:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 92b7266 to f1b5a96
-
-2012-05-30 12:48:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From ec1c4a8 to 92b7266
-
-2012-05-30 11:27:31 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 3429ba6 to ec1c4a8
-
-2012-05-22 15:37:25 +0200 David Svensson Fors <davidsf at axis.com>
-
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- * gst/rtsp-server/rtsp-server.c:
- rtsp: fix compiler warnings
- Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500
-
-2012-05-13 15:59:10 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From dc70203 to 3429ba6
-
-2012-05-11 09:42:47 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- * gst/rtsp-server/rtsp-session-pool.c:
- * gst/rtsp-server/rtsp-session-pool.h:
- rtsp-server: port to new thread API
-
-2012-04-16 09:11:54 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 6db25be to dc70203
-
-2012-04-13 15:27:22 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-client.c:
- rtsp-server: Fix compilation and compiler warnings
-
-2012-04-13 13:49:08 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * autogen.sh:
- * configure.ac:
- * gst/rtsp-server/Makefile.am:
- configure: Modernize autotools setup a bit
- Also we now only create tar.bz2 and tar.xz tarballs.
-
-2012-04-13 13:39:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 464fe15 to 6db25be
-
-2012-04-05 18:45:43 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 7fda524 to 464fe15
-
-2012-04-04 14:45:55 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * 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 <sebastian.droege@collabora.co.uk>
-
- Merge remote-tracking branch 'origin/0.10'
- Conflicts:
- gst/rtsp-server/rtsp-session-pool.c
-
-2012-03-27 10:13:20 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-session-pool.c:
- rtsp-server: Don't use deprecated GLib API
-
-2012-03-26 12:23:36 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- Replace master with 0.11
-
-2012-03-26 12:22:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- Merge branch 'master' into 0.11
-
-2012-03-26 12:20:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- Merge branch 'master' into 0.11
-
-2012-03-19 10:48:09 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
-
- * docs/README:
- A couple minor typo fixes
-
-2012-03-13 18:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: fix state of the appqueue
-
-2012-03-13 16:06:50 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- factory: use videoconvert
-
-2012-03-13 16:02:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- factory: change to new style caps
-
-2012-03-07 15:03:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- configure: fix build
-
-2012-02-29 15:56:06 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-
- * 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 <tim.muller@collabora.co.uk>
-
- * 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 <tim.muller@collabora.co.uk>
-
- * 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 <tim.muller@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- rtsp-client: update for new map API
-
-2012-02-13 10:37:37 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-
- * .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 <thomas (at) apestaart (dot) org>
-
- * common:
- Automatic update of common submodule
- From c463bc0 to 7fda524
-
-2012-01-25 11:40:59 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 2a59016 to c463bc0
-
-2012-01-18 16:48:41 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 0807187 to 2a59016
-
-2012-01-04 19:56:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 11f0cd5 to 0807187
-
-2011-12-09 11:00:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-auth.c:
- example: update for new caps
-
-2011-12-09 10:53:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- Merge branch 'master' into 0.11
-
-2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
-
- * 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 <fabian.deutsch@gmx.de>
-
-2011-11-03 16:06:23 +0100 Fabian Deutsch <fabian.deutsch@gmx.de>
-
- * 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 <fabian.deutsch@gmx.de>
-
-2011-11-03 12:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- Merge branch 'master' into 0.11
-
-2011-11-03 12:55:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <vgottardi@hotmail.com>
-
- * gst/rtsp-server/rtsp-media.c:
- Disallow seek in live media
-
-2011-11-03 11:58:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- Merge branch 'master' into 0.11
-
-2011-11-03 10:48:40 +0100 mat <matzepopatze@gmx.de>
-
- * gst/rtsp-server/rtsp-server.c:
- #ifdef statements for windows socket creation were missing
-
-2011-09-06 21:53:46 +0200 Stefan Sauer <ensonic@users.sf.net>
-
- * common:
- Automatic update of common submodule
- From a39eb83 to 11f0cd5
-
-2011-09-06 16:07:18 +0200 Stefan Sauer <ensonic@users.sf.net>
-
- * common:
- Automatic update of common submodule
- From 605cd9a to a39eb83
-
-2011-08-16 16:39:26 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- Merge branch 'master' into 0.11
-
-2011-08-16 16:07:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: use method to access property
-
-2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: use method to access property
-
-2011-08-16 15:15:19 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- Merge branch 'master' into 0.11
-
-2011-08-16 13:43:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: use media multicast group
-
-2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <rob.krakora at messagenetsystems.com>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- media-factory: configure multicast in media
-
-2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: use media multicast group
-
-2011-08-16 13:37:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <rob.krakora at messagenetsystems.com>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- media-factory: configure multicast in media
-
-2011-08-16 13:25:16 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- Merge branch 'master' into 0.11
-
-2011-08-16 12:09:48 +0200 Robert Krakora <rob.krakora at messagenetsystems.com>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- Merge branch 'master' into 0.11
-
-2011-06-30 10:13:59 +0200 Emmanuel Pacaud <emmanuel@gnome.org>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- Merge branch 'master' into 0.11
-
-2011-08-11 18:07:08 -0700 David Schleef <ds@schleef.org>
-
- * gst/rtsp-server/rtsp-server.c:
- rtsp-server: hold on to reference while using object
-
-2011-08-04 08:59:17 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: use new api
-
-2011-08-04 08:58:58 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- configure: use unstable api
-
-2011-06-27 11:26:26 -0700 David Schleef <ds@schleef.org>
-
- * gst/rtsp-server/rtsp-client.c:
- client: fix reference counting
-
-2011-07-20 17:16:42 +0200 Thijs Vermeir <thijsvermeir@gmail.com>
-
- * 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 <ensonic@google.com>
-
- * 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 <ds@schleef.org>
-
- * common:
- Automatic update of common submodule
- From 69b981f to 605cd9a
-
-2011-06-13 19:05:57 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: update for buffer API change
-
-2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
-
- * gst/rtsp-server/Makefile.am:
- Makefile.am: 0.10 => @GST_MAJORMINOR@
-
-2011-06-07 10:59:16 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
-
- * 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 <edward.hervey@collabora.co.uk>
-
- * gst/rtsp-server/.gitignore:
- .gitignore: 0.10 => 0.11
-
-2011-06-07 10:54:26 +0200 Edward Hervey <edward.hervey@collabora.co.uk>
-
- * gst/rtsp-server/Makefile.am:
- Makefile.am: 0.10 => @GST_MAJORMINOR@
-
-2011-05-24 18:26:06 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- Merge branch 'master' into 0.11
-
-2011-05-19 23:00:52 +0300 Stefan Kost <ensonic@users.sf.net>
-
- * common:
- Automatic update of common submodule
- From 9e5bbd5 to 69b981f
-
-2011-05-18 16:14:10 +0300 Stefan Kost <ensonic@users.sf.net>
-
- * common:
- Automatic update of common submodule
- From fd35073 to 9e5bbd5
-
-2011-05-18 12:27:35 +0300 Stefan Kost <ensonic@users.sf.net>
-
- * common:
- Automatic update of common submodule
- From 46dfcea to fd35073
-
-2011-05-17 09:48:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- Merge branch 'master' into 0.11
-
-2011-05-03 21:13:15 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
-
- * bindings/vala/gst-rtsp-server-0.10.vapi:
- Updated Vala bindings.
- Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
-
-2011-05-03 16:24:28 +0200 Fabian Deutsch <fabian.deutsch@gmx.de>
-
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- Add a signal for newly connected clients.
- Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
-
-2011-05-08 13:15:19 +0200 Alessandro Decina <alessandro.d@gmail.com>
-
- * bindings/python/rtspserver.override:
- python: override gst_rtsp_media_mapping_add_factory to fix refcounting
-
-2011-04-26 19:22:50 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * common:
- add common
-
-2011-04-26 19:07:13 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- Merge branch 'master' into 0.11
- Conflicts:
- common
- configure.ac
-
-2011-04-24 14:07:11 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From c3cafe1 to 46dfcea
-
-2011-04-20 11:19:38 +0200 Alessandro Decina <alessandro.d@gmail.com>
-
- * bindings/python/Makefile.am:
- * bindings/python/rtspserver.defs:
- python bindings: wrap GstRTSPMediaFactoryClass vfuncs
-
-2011-04-20 11:13:56 +0200 Alessandro Decina <alessandro.d@gmail.com>
-
- * bindings/python/arg-types.py:
- python bindings: add GstRTSPUrlParam
- Needed to implement MediaFactory virtual proxies
-
-2011-04-20 10:19:46 +0200 Alessandro Decina <alessandro.d@gmail.com>
-
- * bindings/python/arg-types.py:
- python bindings: fix returning GstRTSPUrl types
-
-2011-04-20 10:17:07 +0200 Alessandro Decina <alessandro.d@gmail.com>
-
- * bindings/python/arg-types.py:
- python bindings: add arg type for GstRTSPUrl
-
-2011-04-20 10:16:08 +0200 Alessandro Decina <alessandro.d@gmail.com>
-
- * bindings/python/rtspserver.defs:
- python bindings: fix the definition of MediaFactory.collect_stream
-
-2011-04-04 15:59:50 +0300 Stefan Kost <ensonic@users.sf.net>
-
- * common:
- Automatic update of common submodule
- From 1ccbe09 to c3cafe1
-
-2011-03-25 22:38:06 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 193b717 to 1ccbe09
-
-2011-03-25 14:58:34 +0200 Stefan Kost <ensonic@users.sf.net>
-
- * common:
- Automatic update of common submodule
- From b77e2bf to 193b717
-
-2011-03-25 10:04:57 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * Makefile.am:
- build: Include lcov.mak to allow test coverage report generation
-
-2011-03-25 09:35:15 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From d8814b6 to b77e2bf
-
-2011-03-25 09:11:40 +0100 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 6aaa286 to d8814b6
-
-2011-03-24 18:51:37 +0200 Stefan Kost <ensonic@users.sf.net>
-
- * common:
- Automatic update of common submodule
- From 6aec6b9 to 6aaa286
-
-2011-03-18 19:34:57 +0100 Luis de Bethencourt <luis@debethencourt.com>
-
- * autogen.sh:
- autogen: wingo signed comment
-
-2011-03-03 20:38:03 +0100 Miguel Angel Cabrera Moya <madmac2501@gmail.com>
-
- * 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 <sebastian.droege@collabora.co.uk>
-
- * gst/rtsp-server/Makefile.am:
- rtsp-server: Don't install the funnel header
-
-2011-02-28 18:35:03 +0100 Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 1de7f6a to 6aec6b9
-
-2011-02-26 19:58:02 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-
- * 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 <ensonic@users.sf.net>
-
- * common:
- Automatic update of common submodule
- From f94d739 to 1de7f6a
-
-2011-02-02 15:37:03 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: remove more unused code
-
-2011-02-02 15:30:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <tim.muller@collabora.co.uk>
-
- * 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 <tim.muller@collabora.co.uk>
-
- * docs/Makefile.am:
- docs: recursive into sub-directories on 'make upload'
-
-2011-01-24 11:53:17 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- back to development
-
-=== release 0.10.8 ===
-
-2011-01-24 11:57:12 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- release 0.10.8
-
-2011-01-19 15:29:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- rtsp-server: clarify docs a little
-
-2011-01-13 18:57:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: init debug category before starting thread
-
-2011-01-13 18:40:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-auth.c:
- auth: add realm to make it more spec compliant
-
-2011-01-12 18:57:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- * gst/rtsp-server/rtsp-server.h:
- server: add locking
-
-2011-01-12 18:33:49 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-video.c:
- example: improve example docs a little
-
-2011-01-12 18:26:57 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- server: simpify channel function
-
-2011-01-12 18:18:13 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * Makefile.am:
- * configure.ac:
- build tests
-
-2011-01-12 18:16:46 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- server: chain up to the parent finalize
-
-2010-09-21 17:04:02 -0300 André Dieb Martins <andre.dieb@gmail.com>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-auth.c:
- * gst/rtsp-server/rtsp-auth.h:
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-client.h:
- * gst/rtsp-server/rtsp-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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: unref auth in finalize
-
-2011-01-12 12:07:04 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- server: unref auth in finalize
-
-2011-01-12 11:07:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <sreerenj.balachandran@nokia.com>
-
- * gst/rtsp-server/rtsp-media.h:
- rtsp-media.h: Minor corrections in comments.
- Fixes #638944
-
-2011-01-11 15:52:44 +0200 Stefan Kost <ensonic@users.sf.net>
-
- * common:
- Automatic update of common submodule
- From e572c87 to f94d739
-
-2011-01-11 13:01:44 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
-
- * .gitignore:
- * docs/.gitignore:
- * docs/libs/.gitignore:
- * examples/.gitignore:
- * gst/rtsp-server/.gitignore:
- gitignore: updates
-
-2011-01-11 12:58:39 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
-
- * 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 <tim.muller@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From ccbaa85 to e572c87
-
-2011-01-10 14:56:39 +0000 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 46445ad to ccbaa85
-
-2011-01-10 15:10:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <tim.muller@collabora.co.uk>
-
- * 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 <tim.muller@collabora.co.uk>
-
- * .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 <tim.muller@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * examples/test-uri.c:
- examples: add some more options and comments
-
-2011-01-07 11:24:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory-uri.c:
- factory-uri: use right property type
-
-2011-01-05 12:07:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <jonas at hallerud dot se>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media.h:
- docs: improve docs
-
-2010-12-29 16:26:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.h:
- client: cleanup headers
-
-2010-12-28 12:18:41 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: fix typo
-
-2010-12-23 18:53:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <edward.hervey@collabora.co.uk>
-
- * common:
- Automatic update of common submodule
- From 169462a to 46445ad
-
-2010-12-18 11:24:48 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <ensonic@users.sf.net>
-
- * common:
- Automatic update of common submodule
- From 011bcc8 to 169462a
-
-2010-12-13 16:38:09 +0100 Andy Wingo <wingo@oblong.com>
-
- 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 <wim.taymans@collabora.co.uk>
-
- Merge branch 'master' into 0.11
- Conflicts:
- common
- configure.ac
-
-2010-12-12 15:48:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * examples/Makefile.am:
- * examples/test-uri.c:
- example: add example of the uri factory
-
-2010-12-11 18:01:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <bilboed@bilboed.com>
-
- * gst/rtsp-server/rtsp-client.c:
- rtsp-server: Remove unused variable and dead assignment
-
-2010-12-11 10:49:30 +0100 Edward Hervey <bilboed@bilboed.com>
-
- * 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 <bilboed@bilboed.com>
-
- * 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 <bilboed@bilboed.com>
-
- * 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 <edward.hervey@collabora.co.uk>
-
- * Makefile.am:
- Makefile: Add cruft-cleaning support
-
-2010-12-07 18:52:15 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
-
- * 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 <edward.hervey@collabora.co.uk>
-
- * gst/rtsp-server/Makefile.am:
- Makefile.am: Use standard GIR make behaviour
-
-2010-12-07 18:14:22 +0100 Edward Hervey <edward.hervey@collabora.co.uk>
-
- * autogen.sh:
- * configure.ac:
- autogen/configure: Bring more in sync to standard gst module behaviour
-
-2010-12-06 19:29:53 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: warn and fail when gstrtpbin is not found
-
-2010-12-06 12:40:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- configure: open 0.11 branch
-
-2010-12-01 20:00:22 +0100 Edward Hervey <bilboed@bilboed.com>
-
- * .gitmodules:
- * common:
- Add common submodule
-
-2010-12-01 19:58:49 +0100 Edward Hervey <bilboed@bilboed.com>
-
- * 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 <hadess@hadess.net>
-
- * common/m4/introspection.m4:
- Update introspection.m4 to match usage
-
-2010-10-30 13:26:12 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * README:
- README: update
- Remove old stuff from the README
-
-2010-10-11 11:12:11 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- back to development
-
-=== release 0.10.7 ===
-
-2010-10-11 11:05:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- release 0.10.7
-
-2010-10-04 17:16:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <tim.muller@collabora.co.uk>
-
- * bindings/vala/Makefile.am:
- build: fix distcheck
-
-2010-09-22 18:24:12 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * 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 <sebastian.droege@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- back to development
-
-=== release 0.10.6 ===
-
-2010-09-22 16:22:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- configure: release 0.10.6
-
-2010-09-22 16:15:56 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: help the compiler a little
-
-2010-08-24 16:47:30 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- media-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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <rob.krakora at messagenetsystems.com>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * docs/README:
- README: add blurb about shared media factories
-
-2010-08-09 12:56:23 -0700 David Schleef <ds@schleef.org>
-
- * gst/rtsp-server/rtsp-media.c:
- Add stdlib.h for atoi()
-
-2010-05-20 14:33:24 +0100 Tim-Philipp Müller <tim.muller@collabora.co.uk>
-
- * 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 <tim.muller@collabora.co.uk>
-
- * 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 <tim.muller@collabora.co.uk>
-
- * configure.ac:
- configure: fail if GStreamer core/base requirements are not met
-
-2010-04-06 17:08:40 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-session.c:
- session: small cleanups
-
-2010-04-06 11:13:51 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-session.c:
- session: indent
-
-2010-03-16 18:33:23 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: use right size for malloc
-
-2010-03-10 11:45:30 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- server: comment ipv6 server listening address
-
-2010-03-10 11:45:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: allow for ipv6 sockets
-
-2010-03-09 13:49:00 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.h:
- media: update comments a little
-
-2010-03-09 13:43:29 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: guard against invalid paths
-
-2010-03-09 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-video.c:
- test: catch server bind errors
-
-2010-03-09 10:27:38 +0100 Alessandro Decina <alessandro.d@gmail.com>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: collect media position when seek completes
-
-2010-03-05 18:37:17 +0100 Luca Ognibene <luca.ognibene at gmail.com>
-
- * gst/rtsp-server/rtsp-client.c:
- client: call unlink_streams in client finalize
- Fixes #599027
-
-2010-03-05 18:23:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-sdp.c:
- sdp: reindent and check for prepared status
-
-2010-03-05 17:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- * gst/rtsp-server/rtsp-session.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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: reindent
-
-2010-03-05 13:34:15 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <ivnik@mail.ru>
-
- * gst/rtsp-server/rtsp-sdp.c:
- sdp: make server work better when behind a proxy
-
-2009-11-21 01:17:25 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- back to development
-
-=== release 0.10.5 ===
-
-2009-11-05 11:20:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- release 0.10.5
-
-2009-10-14 12:11:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- configure: bump required versions
-
-2009-10-11 13:57:54 +0200 Luca Ognibene <luca.ognibene@gmail.com>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: add some docs
-
-2009-08-24 13:27:00 +0200 Peter Kjellerstedt <pkj@axis.com>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- back to development
-
-=== release 0.10.4 ===
-
-2009-08-05 11:44:49 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- Release 0.10.4
-
-2009-07-27 19:42:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-session.c:
- * gst/rtsp-server/rtsp-session.h:
- 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 <wim.taymans@collabora.co.uk>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <tim.muller@collabora.co.uk>
-
- * configure.ac:
- configure: bump core/base requirements to release
-
-2009-06-18 16:05:18 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: fix indentation
-
-2009-06-14 23:12:13 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
-
- * 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 <sebp@k-d-w.org>
-
- * gst/rtsp-server/rtsp-media.c:
- set state and remove elements of media in for loop
-
-2009-06-13 14:38:39 +0200 Sebastian <sebastian@ubuntu.(none)>
-
- * 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 <sebastian@ubuntu.(none)>
-
- * 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 <sebastian@ubuntu.(none)>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: don't leak session pads
-
-2009-06-04 18:32:15 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: clean up the messages a bit
-
-2009-06-03 12:13:21 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-sdp.c:
- sdp: warn and skip streams without media
-
-2009-05-30 14:38:34 +0200 Sebastian Pölsterl <sebp@k-d-w.org>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: be less verbose and leak less
-
-2009-05-26 19:05:07 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: don't leak the destination address
-
-2009-05-26 19:01:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * examples/.gitignore:
- * examples/Makefile.am:
- * examples/test-sdp.c:
- example: add SDP relay example
-
-2009-05-24 19:56:45 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: also count active TCP connections
-
-2009-05-24 19:34:52 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- client: warn when we can't do RTP-Info
-
-2009-05-23 16:18:04 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- factory: factor out the stream construction
-
-2009-05-23 16:17:02 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- back to development
-
-=== release 0.10.3 ===
-
-2009-05-17 13:59:10 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- back to development
-
-=== release 0.10.2 ===
-
-2009-05-11 10:50:31 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- release 0.10.2
-
-2009-05-11 10:38:44 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * .gitignore:
- * common/m4/.gitignore:
- * examples/.gitignore:
- * pkgconfig/.gitignore:
- add some .gitignore files
-
-2009-04-29 17:24:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- media: seek to key frames
-
-2009-04-21 22:44:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * examples/test-ogg.c:
- tests: set the payload type correctly
-
-2009-04-03 22:46:22 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * examples/test-readme.c:
- examples: mark the example as shared for testing
-
-2009-04-03 19:44:37 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <tim.muller@collabora.co.uk>
-
- * 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 <tim.muller@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-session.c:
- docs: fix typo in API docs
-
-2009-03-13 15:57:42 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- allow pause requests for now.
- --
-
-2009-03-11 20:03:06 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- Free the pipeline before other things
- ---
-
-2009-03-04 16:33:21 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- Only free the pending tunnel if there is one
- --
-
-2009-03-04 12:44:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media.c:
- Add better debug info
- Add some better debug info.
-
-2009-02-13 20:00:34 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/test-video.c:
- Time out sessions
- Add support for session timeouts in the example.
-
-2009-02-13 19:58:17 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- Merge branch 'master' of git+ssh://git.collabora.co.uk/git/gst-rtsp-server
-
-2009-02-04 20:10:39 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- Back to development
- Back to 0.10.1.1
-
-=== release 0.10.1 ===
-
-2009-01-30 11:20:18 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * configure.ac:
- Make 0.10.1 release
- Release 0.10.1
-
-2009-01-29 15:19:01 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * bindings/vala/Makefile.am:
- Fix make dist
- Add more directories and files to the dist.
-
-2009-01-24 14:34:35 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
-
- * bindings/python/Makefile.am:
- * bindings/python/rtspserver.override:
- Fixed compile error of python bindings
-
-2009-01-23 21:03:53 +0100 Sebastian Pölsterl <sebp@k-d-w.org>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-client.c:
- * gst/rtsp-server/rtsp-media-factory.c:
- * gst/rtsp-server/rtsp-media-factory.h:
- * gst/rtsp-server/rtsp-media.c:
- * gst/rtsp-server/rtsp-media.h:
- * gst/rtsp-server/rtsp-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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/Makefile.am:
- Add Makefile entry for the media factory
-
-2009-01-20 19:44:45 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-session.h:
- Add some comments.
-
-2009-01-20 13:57:47 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-session-pool.c:
- Do some more cleanup of the session pool.
-
-2009-01-08 16:28:24 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/Makefile.am:
- Install rtsp-session and rtsp-session-pool headers
-
-2009-01-08 14:57:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * .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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <sebp@k-d-w.org>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <marduk@k-d-w.org>
-
- * 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 <marduk@k-d-w.org>
-
- * 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 <alessandro.d@gmail.com>
-
- * 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 <sebp@ubuntu.(none)>
-
- * examples/Makefile.am:
- * gst/rtsp-server/Makefile.am:
- Put GStreamer version in library name
-
-2009-01-08 13:51:26 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * examples/Makefile.am:
- * gst/rtsp-server/Makefile.am:
- Fix some issues to pass distcheck
-
-2009-01-08 13:41:33 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * gst/rtsp-server/rtsp-server.c:
- Added port property to GstRTSPServer class.
-
-2009-01-08 13:18:55 +0100 Wim Taymans <wim.taymans@collabora.co.uk>
-
- * 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 <sebp@ubuntu.(none)>
-
- * src/rtsp-client.h:
- Removed obsolete variable
-
-2008-11-10 21:03:15 +0100 Sebastian Pölsterl <sebp@ubuntu.(none)>
-
- * 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 <wim.taymans@collabora.co.uk>
-
- * 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 <wim@metal.(none)>
-
- * src/rtsp-session.c:
- Initialize some more vars.
-
-2008-10-23 12:14:55 +0200 Wim Taymans <wim@metal.(none)>
-
- * src/rtsp-session.c:
- Initialize variable to avoid compiler warning.
-
-2008-10-09 13:30:47 +0100 Simon McVittie <simon.mcvittie@collabora.co.uk>
-
- * .gitignore:
- Add a reasonable generic .gitignore
-
+++ /dev/null
-Installation Instructions
-*************************
-
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-Inc.
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-Basic Installation
-==================
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-====================================
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-is known as a "VPATH" build.
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-
-`--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.
+++ /dev/null
-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
+++ /dev/null
-This is GStreamer RTSP Server 1.4.5
-
+++ /dev/null
-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.
+++ /dev/null
-
-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
+++ /dev/null
-You need to have GStreamer. You can use an installed version of
-GStreamer or from its build dir.
-
+++ /dev/null
-
- - use a config file to configure the server
- - error recovery
+++ /dev/null
-#!/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."
+++ /dev/null
-2008-12-17 Edward Hervey <bilboed@gmail.com>
-
- * gst.supp:
- And yet another variation of the GstAudioFilter leak.
-
-2008-12-15 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- Patch by: Roland Illig <roland dot illig at gmx dot de>
-
- * 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 <bilboed@gmail.com>
-
- * gst.supp:
- And yet another variant of the GstAudioFilter leak.
-
-2008-12-13 Edward Hervey <edward.hervey@collabora.co.uk>
-
- * 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 <edward.hervey@collabora.co.uk>
-
- * gst.supp:
- Fix leak in GIO called by gnomevfs. Nothing we can do about this.
-
-2008-12-12 Edward Hervey <edward.hervey@collabora.co.uk>
-
- * 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 <edward.hervey@collabora.co.uk>
-
- * 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 <sebastian.droege@collabora.co.uk>
-
- * 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 <mark.nauwelaerts@collabora.co.uk>
-
- * gst.supp:
- Add suppression variant for Ubuntu Hardy x86/64bit.
-
-2008-12-01 Stefan Kost <ensonic@users.sf.net>
-
- * gtk-doc-plugins.mak:
- * gtk-doc.mak:
- Simplily uninstall rule. Its closer to upstream and fixes #150331.
-
-2008-11-29 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * m4/glib-gettext.m4:
- Update glib-gettext.m4 from latest stable GLib release.
-
-2008-11-29 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- Patch by: Cygwin Ports maintainer
- <yselkowitz at users dot sourceforge dot net>
-
- * 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 <sebastian.droege@collabora.co.uk>
-
- * m4/gtk-doc.m4:
- * m4/pkg.m4:
- Update gtk-doc and pkg-config m4 macros from their latest releases.
-
-2008-11-20 Michael Smith <msmith@songbirdnest.com>
-
- * m4/as-objc.m4:
- Fix objective C test macro when none of the compilers are found at all.
-
-2008-10-30 Stefan Kost <ensonic@users.sf.net>
-
- * 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 <jan.schmidt@sun.com>
-
- * gtk-doc.mak:
- Don't clobber the real registry cache file when
- building docs.
-
-2008-10-07 Jan Schmidt - Sun Microsystems <jan.schmidt@sun.com>
-
- * m4/gst-error.m4:
- Also disable the bogus "loop not entered at top" warnings appearing on Sparc Forte builds.
-
-2008-10-06 Stefan Kost <ensonic@users.sf.net>
-
- * 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 <ds@schleef.org>
-
- * 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 <ds@schleef.org>
-
- * 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 <ensonic@users.sf.net>
-
- * 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 <ensonic@users.sf.net>
-
- * 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 <mark.nauwelaerts@collabora.co.uk>
-
- * gst.supp:
- Add suppressions for Ubunty Hardy x86/64bit, similar to earlier
- versions and 32bit variant.
-
-2008-07-31 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * m4/gst-feature.m4:
- Remove GST_DISABLE_(ENUMTYPES|INDEX|URI).
-
-2008-07-21 Tim-Philipp Müller <tim.muller at collabora co uk>
-
- * m4/gst-error.m4::
- When checking for GST_ERROR_CXXFLAGS, check each compiler flag
- individually, not all together.
-
-2008-07-20 Tim-Philipp Müller <tim.muller at collabora co uk>
-
- * 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 <ensonic@users.sf.net>
-
- * plugins.xsl:
- Split refsect2 also here to make "Element Pads" subtitle visible.
-
-2008-07-08 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-
- * 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 <mark.nauwelaerts@collabora.co.uk>
-
- * gtk-doc-plugins.mak:
- Only clean doc maintainer stamps in maintainer-clean. Fixes #539977.
-
-2008-06-20 Sebastian Dröge <slomo@circular-chaos.org>
-
- * 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 <slomo@circular-chaos.org>
-
- * 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 <tim.muller at collabora co uk>
-
- * 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 <jan.schmidt@sun.com>
-
- * 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 <ensonic@users.sf.net>
-
- * plugins.xsl:
- The class was not shown in plugin docs. Fix typo in changelog below.
-
-2008-05-22 Jan Schmidt <jan.schmidt@sun.com>
-
- * gstdoc-scangobj:
- Emit warnings if one of the GTypes we're expecting is 0
- when scanning.
-
-2008-05-21 Felipe Contreras <felipe.contreras@gmail.com>
-
- * gtk-doc-plugins.mak:
- * gtk-doc.mak:
- Fix installing png images when gtk-doc is disabled.
-
-2008-05-21 Felipe Contreras <felipe.contreras@gmail.com>
-
- * gtk-doc-plugins.mak:
- * gtk-doc.mak:
- Fix make clean when gtk-doc is disabled and other cleanups.
-
-2008-05-17 Jan Schmidt <jan.schmidt@sun.com>
-
- * gtk-doc-plugins.mak:
- Be more quiet when the files don't yet exist.
-
-2008-05-16 Jan Schmidt <jan.schmidt@sun.com>
-
- * 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 <pkj@axis.com>
-
- * 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 <tim.muller at collabora co uk>
-
- * gst.supp:
- Add suppression for glibc bug on gutsy/x86-64
-
-2008-05-12 Stefan Kost <ensonic@users.sf.net>
-
- * plugins.xsl:
- Improve the layout of the caps, but splitting them on ";".
-
-2008-05-09 Sebastian Dröge <slomo@circular-chaos.org>
-
- Patch by: Brian Cameron <brian dot cameron at sun dot com>
-
- * m4/gst-default.m4:
- Don't set the default audio sink to the default visualizer.
- Fixes bug #532295.
-
-2008-05-07 Tim-Philipp Müller <tim.muller at collabora co uk>
-
- * check.mak: (help):
- Document GST_CHECKS environment variable in checks 'make help'.
-
-2008-05-06 Sebastian Dröge <slomo@circular-chaos.org>
-
- Patch by: Marc-Andre Lureau <marcandre dot lureau at gmail dot com>
-
- * 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 <edward.hervey@collabora.co.uk>
-
- * 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 <bilboed@gmail.com>
-
- * gst.supp: Make tls leak suppression a bit more generic.
-
-2008-04-22 Edward Hervey <bilboed@gmail.com>
-
- * 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 <bilboed@gmail.com>
-
- * 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 <slomo@circular-chaos.org>
-
- * Makefile.am:
- * m4/Makefile.am:
- Dist all files in common. Fixes bug #527984.
-
-2008-04-14 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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 <tim at centricular dot net>
-
- * 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 <tim at centricular dot net>
-
- * m4/gst-default.m4:
- Add --with-default-{audiosink|audiosrc|videosink|videosrc|visualizer}
- configure switches (#519417).
-
-2008-04-03 Tim-Philipp Müller <tim at centricular dot net>
-
- * m4/gst-args.m4:
- Add --disable-foo switch for dependency-less plugins (#525586).
-
-2008-04-01 Sebastian Dröge <slomo@circular-chaos.org>
-
- * m4/gst-parser.m4:
- Unconditionally require flex 2.5.31 and bison 1.875.
-
-2008-03-23 Sebastian Dröge <slomo@circular-chaos.org>
-
- * m4/gst-arch.m4:
- amd64/x86_64 allows unaligned memory access too.
-
-2008-03-21 Sebastian Dröge <slomo@circular-chaos.org>
-
- * 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 <slomo@circular-chaos.org>
-
- * 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 <edward.hervey@collabora.co.uk>
-
- * 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 <ds@schleef.org>
-
- * m4/ax_create_stdint_h.m4: Oops, checked in the wrong copy of
- this file. (Update from upstream)
-
-2008-03-03 David Schleef <ds@schleef.org>
-
- * m4/ax_create_stdint_h.m4: Update from upstream. Fixes a bug
- compiling with MSVC.
-
-2008-03-03 Edward Hervey <edward.hervey@collabora.co.uk>
-
- * 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 <pkj@axis.com>
-
- * 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 <slomo@circular-chaos.org>
-
- * gst.supp:
- Add another glibc suppression.
-
-2008-02-29 Peter Kjellerstedt <pkj@axis.com>
-
- * m4/gst-feature.m4:
- Make the comment before defines generated via AG_GST_CHECK_FEATURE
- look nicer. (#498222)
-
-2008-02-26 Jan Schmidt <jan.schmidt@sun.com>
-
- * 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 <wingo@pobox.com>
-
- * 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 <Jan.Schmidt@sun.com>
-
- * 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 <tim at centricular dot net>
-
- * 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 <tim at centricular dot net>
-
- * 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 <tim at centricular dot net>
-
- * 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 <slomo@circular-chaos.org>
-
- * gst.supp:
- Add even more glibc 2.7 suppressions.
-
-2008-02-22 Sebastian Dröge <slomo@circular-chaos.org>
-
- * gst.supp:
- Add another suppression for GLib caching some values after
- the first call.
-
-2008-02-12 Sebastian Dröge <slomo@circular-chaos.org>
-
- Patch by:
- Tim Mooney <mooney at dogbert dot cc dot ndsu dot nodak dot edu>
-
- * m4/gst-error.m4:
- Use no%E_MACRO_REDEFINED on Solaris to prevent compiler warnings.
- Fixes bug #515905.
-
-2008-02-11 Sebastian Dröge <slomo@circular-chaos.org>
-
- * 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 <ensonic@users.sf.net>
-
- * ChangeLog:
- Changelog surgery for last commit.
-
-2008-02-08 Stefan Kost <ensonic@users.sf.net>
-
- * m4/gtk-doc.m4:
- Conditionally check for SED. Also sync a bit with upstream macro.
-
-2008-02-08 Stefan Kost <ensonic@users.sf.net>
-
- * 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 <ensonic@users.sf.net>
-
- * 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 <slomo@circular-chaos.org>
-
- * 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 <ensonic@users.sf.net>
-
- * ChangeLog:
- ChangeLog surgery.
-
- * gstdoc-scangobj:
- Sync the object scanner with gtk-doc fixes. Update args and hierarchy
- files.
-
-2008-01-20 Sebastian Dröge <slomo@circular-chaos.org>
-
- * 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 <slomo@circular-chaos.org>
-
- * gst.supp:
- Add a suppression for a glibc bug:
- http://valgrind.org/docs/manual/faq.html#faq.exit_errors>
-
-2008-01-18 Sebastian Dröge <slomo@circular-chaos.org>
-
- * 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 <slomo@circular-chaos.org>
-
- * gst.supp:
- Add some glibc 2.7 supressions as found on Debian/unstable.
-
-2008-01-14 Jan Schmidt <jan.schmidt@sun.com>
-
- * download-translations:
- Apparently I have problems with leaving things commented out when
- I edit shell scripts.
-
-2008-01-12 Jan Schmidt <Jan.Schmidt@sun.com>
-
- * download-translations:
- Remove bash-isms
-
-2008-01-12 Jan Schmidt <Jan.Schmidt@sun.com>
-
- * check-exports:
- Restore the cleanup rm of our tmp file which I didn't mean to leave
- commented out.
-
-2008-01-12 Jan Schmidt <Jan.Schmidt@sun.com>
-
- * 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 <tim at centricular dot net>
-
- Patch by: Peter Kjellerstedt <pkj axis com>
-
- * 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 <ensonic@users.sf.net>
-
- * coverage/lcov.mak:
- Update coverage make-rules: use them conditionaly, use libtool mode
- and use lcov to cleanup.
-
-2007-12-18 Sebastian Dröge <slomo@circular-chaos.org>
-
- * glib-gen.mak:
- Also use #include "header" instead of #include <header> 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 <slomo@circular-chaos.org>
-
- * glib-gen.mak:
- Use #include "header" instead of #include <header> 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 <tim at centricular dot net>
-
- * 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 <tim at centricular dot net>
-
- * Makefile.am:
- Don't forget to dist the new win32.mak.
-
-2007-12-17 Tim-Philipp Müller <tim at centricular dot net>
-
- * win32.mak: (win32), (win32defs):
- Move common win32 Makefile foo into this new file.
-
-2007-12-15 Stefan Kost <ensonic@users.sf.net>
-
- * gtk-doc-plugins.mak:
- * gtk-doc.mak:
- We should have never forked this that much :/.
-
-2007-12-13 Tim-Philipp Müller <tim at centricular dot net>
-
- * check-exports:
- Fix build on the ppc64 build bot.
-
-2007-12-13 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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 <tim at centricular dot net>
-
- * Makefile.am:
- check-exports should be disted.
-
-2007-12-12 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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 <jan.schmidt@sun.com>
-
- * 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 <slomo@circular-chaos.org>
-
- * 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 <tim at centricular dot net>
-
- * gtk-doc-plugins.mak:
- Just use the normal 'check' target and avoid a circular
- dependency.
-
-2007-09-06 Tim-Philipp Müller <tim at centricular dot net>
-
- * gtk-doc-plugins.mak:
- Add rule to error out if .hierarchy file contains tabs.
-
-2007-08-20 Tim-Philipp Müller <tim at centricular dot net>
-
- * download-translations:
- * po.mak:
- If there are new languages, they need to be added to po/LINGUAS.
-
-2007-08-20 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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 <thaytan@mad.scientist.com>
-
- * gst.supp:
- Add a suppression for GLib caching the tmp dir seen on an
- Ubuntu Feisty system.
-
-2007-07-13 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <ensonic@users.sf.net>
-
- * gst-xmlinspect.py:
- * plugins.xsl:
- Also extract element caps for plugin-docs. Fixes parts of #117692.
-
-2007-06-21 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Andreas Schwab
-
- * m4/gst-feature.m4:
- Fix quoting (#449493).
-
-2007-06-10 Sebastian Dröge <slomo@circular-chaos.org>
-
- * 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 <msmith@fluendo.com>
-
- * gst.supp:
- Suppression variant for our good friend the TLS leak, this time for
- Ubuntu Feisty/x86.
-
-2007-05-09 Tim-Philipp Müller <tim at centricular dot net>
-
- * gtk-doc-plugins.mak:
- Fix make distcheck again; change some spaces to tabs in makefile.
-
-2007-04-29 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gtk-doc-plugins.mak (-module):
- Error out when the html build step gives warnings, so they get
- fixed properly.
-
-2007-04-23 Stefan Kost <ensonic@users.sf.net>
-
- * 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 <slomo@circular-chaos.org>
-
- Patch by: Vincent Torri <vtorri at univ-evry dot fr>
-
- * 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 <slomo@circular-chaos.org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <slomo@circular-chaos.org>
-
- * m4/gst-parser.m4:
- Fix the flex version check. It ignored the micro version before.
-
-2007-03-09 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 <thaytan@mad.scientist.com>
-
- * 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 <thaytan@mad.scientist.com>
-
- * gst.supp:
- add a suppression for this GConf flup on the FC5 buildbot.
-
-2007-03-06 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst.supp:
- Make the suppression a little more generic, to catch the FC5
- backtrace too.
-
-2007-03-06 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 <thomas at apestaart dot org>
-
- * m4/gst-arch.m4:
- Move a line that was in the wrong macro
-
-2007-02-28 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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 <ensonic@users.sf.net>
-
- * 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 <tim at centricular dot net>
-
- Patch by: Will Newton <will.newton gmail com>
-
- * 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 <tim at centricular dot net>
-
- * 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 <tim at centricular dot net>
-
- * 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 <tim at centricular dot net>
-
- * 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 <msmith@fluendo.com>
-
- * gst.supp:
- More suppressions for edgy.
-
-2006-09-28 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 <tim at centricular dot net>
-
- * gst.supp:
- Some suppressions for the more recent ld.so in ubuntu edgy.
-
-2006-08-23 Tim-Philipp Müller <tim at centricular dot net>
-
- * gst.supp:
- Shorten function trail so the suppression works on
- my ubuntu dapper system with core cvs as well.
-
-2006-07-28 Jan Schmidt <thaytan@mad.scientist.com>
-
- * gst.supp:
- Extra suppressions from my Ubuntu x86_64 machine
-
-2006-07-24 Tim-Philipp Müller <tim at centricular dot net>
-
- Patch by: Frederic Peters <fpeters at entrouvert com>
-
- * 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 <thaytan@mad.scientist.com>
-
- * 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 <ensonic@users.sf.net>
-
- Patch by: Frederic Peters <fpeters@entrouvert.com>
-
- * m4/gst-parser.m4:
- Fix bison detection (#348354)
-
-2006-07-21 Stefan Kost <ensonic@users.sf.net>
-
- * m4/gst-parser.m4:
- check for bison and flex
-
-2006-07-13 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * m4/gst-args.m4:
- when building with gcov, reset CFLAGS and friends to O0
-
-2006-07-02 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <tim at centricular dot net>
-
- Patch by: Peter Kjellerstedt <pkj at axis com>
-
- * 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 <tim at centricular dot net>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * m4/gst-arch.m4:
- add PPC64 so we can have separate structure sizes for it
-
-2006-06-05 Edward Hervey <edward@fluendo.com>
-
- * gtk-doc.mak:
- Check for the proper .devhelp2 file to remove.
-
-2006-05-31 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gtk-doc.mak:
- allow a magic variable to suppress errors from docbuilding
-
-2006-05-30 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * gtk-doc.mak:
- error out if gtkdoc-mktmpl finds unused declarations
-
-2006-05-28 Edward Hervey <edward@fluendo.com>
-
- * gst.supp:
- Reverting previous commit. That's good to know, Edward, but why ?
-
-2006-05-28 Edward Hervey <edward@fluendo.com>
-
- * gst.supp:
- Added suppresion for memleak in g_option_context_parse on fc5-64
-
-2006-05-19 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * m4/gst-check.m4:
- set GSTPB_PLUGINS_DIR just like GST_PLUGINS_DIR
-
-2006-05-18 Tim-Philipp Müller <tim at centricular dot net>
-
- * 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 <thomas at apestaart dot org>
-
- Patch by: Peter Kjellerstedt
-
- * m4/gst-arch.m4:
- add support for CRIS and CRISv32.
-
-2006-05-17 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 <thaytan@mad.scientist.com>
-
- * gtk-doc.mak:
- Add uninstall rule to remove .devhelp2 files.
-
-2006-05-09 Edward Hervey <edward@fluendo.com>
-
- * gst.supp:
- Add suppression for GSlice version of
- g_type_init calloc leak
-
-2006-04-05 Michael Smith <msmith@fluendo.com>
-
- * gst.supp:
- Delete a bogus suppression for the registry code.
- Generalise a suppression for a glib bug (see #337404)
-
-2006-04-04 Michael Smith <msmith@fluendo.com>
-
- * gst.supp:
- Add a leak suppression: the existing glibc-doesn't-free-TLS one
- wasn't triggering here.
-
-2006-04-04 Michael Smith <msmith@fluendo.com>
-
- * gst.supp:
- Add some minimally-neccesary suppressions for my x86/dapper system.
-
-2006-04-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * plugins.xsl:
- Do not display an origin link if origin does not start with http
- See #323798
-
-2006-04-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * m4/gst-args.m4:
- * m4/gst-feature.m4:
- add more macros
- * m4/gst-x11.m4:
- X11-related checks
-
-2006-04-01 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * gst.supp:
- add a suppression for g_parse_debug_string
-
-2006-03-23 Stefan Kost <ensonic@users.sf.net>
-
- * gstdoc-scangobj:
- sync fully with gtkdoc-0.15
-
-2006-03-23 Stefan Kost <ensonic@users.sf.net>
-
- * gstdoc-scangobj:
- * gtk-doc.mak:
- sync a little with gtk-doc mainline
-
-2006-03-17 Wim Taymans <wim@fluendo.com>
-
- * gst.supp:
- add another clone suppression
- change all glibc suppressions to match 2.3.*
-
-2006-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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 <thaytan@mad.scientist.com>
-
- * check.mak:
- Set G_SLICE=always-malloc when valgrinding tests
- (closes #333272)
-
-2006-02-21 Jan Schmidt <thaytan@mad.scientist.com>
-
- * m4/gst-glib2.m4:
- Fix debug output when the GLib version prerequisite is not found
-
-2006-02-13 Andy Wingo <wingo@pobox.com>
-
- * 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 <thomas at apestaart dot org>
-
- * m4/gst-default.m4:
- switch to auto* sinks for defaults
-
-2006-02-02 Wim Taymans <wim@fluendo.com>
-
- * 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 <thomas at apestaart dot org>
-
- * gtk-doc-plugins.mak:
- only add srcdir/gst if it exists
-
-2006-01-30 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * release.mak:
- don't complain about disted enums in win32
-
-2006-01-20 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * m4/pkg.m4:
- update with a more recent version
-
-2006-01-07 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gettext.patch:
- make Makefile depend on LINGUAS, so rebuilds work when adding
- a language
-
-2006-01-03 Michael Smith <msmith@fluendo.com>
-
- * check.mak:
- Clarify error message from valgrind test runs.
-
-2005-12-16 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * m4/gst-arch.m4:
- define HOST_CPU
-
-2005-11-29 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * check.mak:
- add a valgrind-forever target for tests
-
-2005-11-28 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * m4/gst-check.m4:
- check for tools correctly
-
-2005-10-18 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gtk-doc.mak:
- only enable breaking on new API when make distcheck passes,
- not before
-
-2005-10-18 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * m4/gst-check.m4:
- Resurrect Julien's dead body and wipe his mind clean
-
-2005-10-18 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * m4/gst-check.m4:
- Kill Julien
-
-2005-10-17 Julien MOUTTE <julien@moutte.net>
-
- * 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 <ensonic@users.sf.net>
-
- * gtk-doc.mak:
- make build break on new api that has not been added to the
- sections file
-
-2005-10-17 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * m4/gst-glib2.m4:
- update, warn in error cases
-
-2005-10-16 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * m4/gst-error.m4:
- add GST_SET_DEFAULT_LEVEL
-
-2005-10-16 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * m4/Makefile.am:
- * m4/gst-gettext.m4:
- add macro for setting up gettext
-
-2005-10-15 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * m4/gst-args.m4:
- add some .m4's for argument checking that can be shared among modules
-
-2005-10-15 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * m4/as-libtool.m4:
- set _LT_LDFLAGS
- * m4/gst-libxml2.m4:
- document
-
-2005-10-15 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * m4/gst-arch.m4:
- indent a little
- add AC_REQUIRE
- * m4/gst-error.m4:
- clean up
-
-2005-10-12 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-autogen.sh:
- update version detection expression to catch stuff like
- Libtool (libtool15) 1.5.0
-
-2005-10-11 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst.supp:
- commit 6 new suppressions related to g_module_open; can these
- really not be folded into one ?
-
-2005-10-11 Edward Hervey <edward@fluendo.com>
-
- * gst.supp:
- made the <g_type_init calloc 2> suppression more generic
- Added pthread memleak suppresions
- Added nss_parse_* memleak suppresion (used by g_option_context_parse)
-
-2005-10-11 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * check.mak:
- be more strict, more leak resolution
- * gst.supp:
- clean up the g_type_init suppressions
-
-2005-10-07 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * m4/Makefile.am:
- * m4/gst-valgrind.m4:
- put the valgrind detection in an .m4
-
-2005-09-29 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * check.mak:
- add some more targets, like "help", but also more intensive tests
-
-2005-09-23 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gtk-doc.mak:
- make certain doc warnings fatal so people maintain docs again
-
-2005-09-23 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * gst-xmlinspect.py:
- * gstdoc-scangobj:
- some fixes for new API
- * gtk-doc-plugins.mak:
- set environment properly
-
-2005-09-17 David Schleef <ds@schleef.org>
-
- * gtk-doc-plugins.mak: Use new environment variables.
-
-2005-09-16 Michael Smith <msmith@fluendo.com>
-
- * gstdoc-scangobj:
- Make the scanobj code reflect registry/plugin API changes
-
-2005-09-15 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <ds@schleef.org>
-
- * 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 <thomas at apestaart dot org>
-
- * gtk-doc-plugins.mak:
- error out on inspect failure
-
-2005-09-14 Michael Smith <msmith@fluendo.com>
-
- * glib-gen.mak:
- Don't call glib-mkenums with arguments that confuse/break MinGW,
- fixes 316155.
-
-2005-09-03 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <rbultje@ronald.bitfreak.net>
-
- * m4/gst-doc.m4:
- s/pdf/eps/ in test for whether we output EPS images (#309379).
-
-2005-07-18 Andy Wingo <wingo@pobox.com>
-
- * 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 <thomas at apestaart dot org>
-
- * m4/gst-error.m4:
- add macro to set ERROR_CFLAGS
-
-2005-06-30 Jan Schmidt <thaytan@mad.scientist.com>
-
- * 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 <thomas at apestaart dot org>
-
- * m4/gtk-doc.m4:
- added
-
-2005-06-03 Stefan Kost <ensonic@users.sf.net>
-
- * gst-autogen.sh: create autoregen.sh *before* shifting the options
-
-2005-05-17 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-autogen.sh: only update autoregen.sh on actual runs
-
-2005-03-11 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * m4/check.m4: m4 from the check unit test suite
-
-2004-12-14 David Schleef <ds@schleef.org>
-
- * m4/gst-arch.m4: remove MMX stuff, since it doesn't work and
- isn't needed anywhere
-
-2004-12-08 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gst-autogen.sh:
- allow failure command to be run so we can clean upfrom autopoint
-
-2004-09-03 Zeeshan Ali Khattak <zeenix@gmail.com>
- * m4/gst-feature.m4: Trying to correct the GST_CHECK_CONFIGPROG macro
-
-2004-07-21 Benjamin Otte <otte@gnome.org>
-
- * m4/.cvsignore: exciting updates for libtool m4 files
-
-2004-07-12 David Schleef <ds@schleef.org>
-
- * m4/as-objc.m4: Add a macro to test for objective C
-
-2004-06-12 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * m4/gst-feature.m4:
- not all of them support --plugin-libs, so redirect stderr
-
-2004-06-12 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * m4/as-scrub-include.m4:
- sync with upstream to 0.1.4. Fixes #132440
-
-2004-06-07 Benjamin Otte <otte@gnome.org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * m4/as-scrub-include.m4: synced with upstream
-
-2004-05-03 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * po.mak:
- snippet for updating .po files
-
-2004-03-18 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * Makefile.am:
- * m4/Makefile.am:
- integrate these with the dist
-
-2004-03-17 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * release.mak: add a release target
-
-2004-03-09 Thomas Vander Stichele <thomas at apestaart dot org>
-
- patch by: Stephane Loeuillet
-
- * m4/ax_create_stdint_h.m4:
- use head -n instead of head - (#136500)
-
-2004-03-05 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * m4/gst-doc.m4: don't build PS without dvips binary
-
-2004-02-22 Julio M. Merino Vidal <jmmv@menta.net>
-
- reviewed by: Benjamin Otte <otte@gnome.org>
-
- * m4/as-docbook.m4:
- don't use == operator with test(1) (fixes #135115)
-
-2004-02-16 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * common/m4/gst-arch.m4: x86_64 is x86 too (clue from Fedora 2 test)
-
-2004-02-13 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * m4/gst-libxml2.m4:
- take required version as argument, and default to 2.4.9 if not
- specified
-
-2004-02-12 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * m4/gst-feature.m4:
- rename and fix up GST_CHECK_DISABLE_SUBSYSTEM
-
-2004-02-11 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * 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 <johan@gnome.org>
-
- * gst-autogen.sh: Use A-Z instead of A-z in sed expression to
- avoid a warning
-
-2004-02-05 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
-
- * 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 <thomas at apestaart dot org>
-
- * 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 <thomas at apestaart dot org>
-
- * glib-gen.mak: added; used to generate enums and marshal code
-
-2004-01-13 Thomas Vander Stichele <thomas at apestaart dot org>
-
- * gettext.patch: added; used by autogen.sh to make sure
- GETTEXT_PACKAGE is understood from po/Makefile.in.in -> po/Makefile.in
-
+++ /dev/null
-SUBDIRS = m4
-
-EXTRA_DIST = \
- ChangeLog \
- gettext.patch \
- glib-gen.mak gtk-doc.mak upload-doc.mak \
- cruft.mak release.mak win32.mak po.mak \
- parallel-subdirs.mak \
- gst-autogen.sh \
- check-exports \
- c-to-xml.py mangle-tmpl.py scangobj-merge.py \
- gtk-doc-plugins.mak \
- plugins.xsl gstdoc-scangobj \
- gst.supp check.mak \
- coverage/lcov.mak \
- coverage/coverage-report.pl \
- coverage/coverage-report.xsl \
- coverage/coverage-report-entry.pl \
- download-translations \
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+++ /dev/null
-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.
+++ /dev/null
-# -*- 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 ('<?xml version="1.0"?>')
- print ('<!DOCTYPE refentry PUBLIC "-//OASIS//DTD DocBook XML V4.1.2//EN" "http://www.oasis-open.org/docbook/xml/4.1.2/docbookx.dtd">')
- print ()
- print ('<programlisting>')
-
- # print content
- print (saxutils.escape(content))
- print ('</programlisting>')
-
-main()
+++ /dev/null
-#!/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
-
+++ /dev/null
-# 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
-
+++ /dev/null
-#!/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 <<EOF;
-<html>
-<head>
-<title>Coverage report for $ARGV[0]</title>
-<style type="text/css">
- span.perfect {
- background: rgb(0,255,0);
- }
- span.terrible {
- background: rgb(255,0,0);
- }
-</style>
-</head>
-<body>
-<h1>Coverage report for $ARGV[0]</h1>
-
-<pre>
-EOF
-
-
-while (<>) {
- s/&/&/g;
- s/</</g;
- s/>/>/g;
-
- if (/^\s*function (\S+) called (\d+) returned \d+% blocks executed \d+%/) {
- my $class = $2 > 0 ? "perfect" : "terrible";
- $_ = "<span class=\"$class\" id=\"" . $1 . "\">$_</span>";
- } elsif (/^\s*branch\s+\d+\s+taken\s+(\d+)%\s+.*$/) {
- my $class = $1 > 0 ? "perfect" : "terrible";
- $_ = "<span class=\"$class\">$_</span>";
- } elsif (/^\s*branch\s+\d+\s+never executed.*$/) {
- my $class = "terrible";
- $_ = "<span class=\"$class\">$_</span>";
- } elsif (/^\s*call\s+\d+\s+never executed.*$/) {
- my $class = "terrible";
- $_ = "<span class=\"$class\">$_</span>";
- } elsif (/^\s*call\s+\d+\s+returned\s+(\d+)%.*$/) {
- my $class = $1 > 0 ? "perfect" : "terrible";
- $_ = "<span class=\"$class\">$_</span>";
- }
-
- print;
-}
-
-print <<EOF;
-</pre>
-</body>
-</html>
-EOF
+++ /dev/null
-#!/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 "<coverage>\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 " <entry name=\"%s\" details=\"%s\">\n", $name, ($type eq "file" ? $filemap{$name} : $filemap{$rec->{file}});
- printf " <lines count=\"%s\" coverage=\"%s\"/>\n", $rec->{lines}, $rec->{linesCoverage};
- if (exists $rec->{branches}) {
- printf " <branches count=\"%s\" coverage=\"%s\"/>\n", $rec->{branches}, $rec->{branchesCoverage};
- }
- if (exists $rec->{conds}) {
- printf " <conditions count=\"%s\" coverage=\"%s\"/>\n", $rec->{conds}, $rec->{condsCoverage};
- }
- if (exists $rec->{calls}) {
- printf " <calls count=\"%s\" coverage=\"%s\"/>\n", $rec->{calls}, $rec->{callsCoverage};
- }
- print " </entry>\n";
- }
-
- printf " <summary>\n";
- printf " <lines count=\"%s\" coverage=\"%s\"/>\n", $summary{$type}->{lines}, $summary{$type}->{linesCoverage};
- printf " <branches count=\"%s\" coverage=\"%s\"/>\n", $summary{$type}->{branches}, $summary{$type}->{branchesCoverage};
- printf " <conditions count=\"%s\" coverage=\"%s\"/>\n", $summary{$type}->{conds}, $summary{$type}->{condsCoverage};
- printf " <calls count=\"%s\" coverage=\"%s\"/>\n", $summary{$type}->{calls}, $summary{$type}->{callsCoverage};
- printf " </summary>\n";
- printf "</%ss>\n", $type;
-}
-
-print "</coverage>\n";
+++ /dev/null
-<?xml version="1.0" encoding="utf-8"?>\r
-<!--\r
-#\r
-# Copyright (C) 2006 Daniel Berrange\r
-#\r
-# This program is free software; you can redistribute it and/or modify\r
-# it under the terms of the GNU General Public License as published by\r
-# the Free Software Foundation; either version 2 of the License, or\r
-# (at your option) any later version.\r
-#\r
-# This program is distributed in the hope that it will be useful,\r
-# but WITHOUT ANY WARRANTY; without even the implied warranty of\r
-# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the\r
-# GNU General Public License for more details.\r
-#\r
-# You should have received a copy of the GNU General Public License\r
-# along with this program; if not, write to the Free Software\r
-# Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA\r
--->\r
-<xsl:stylesheet xmlns:xsl="http://www.w3.org/1999/XSL/Transform"\r
- version="1.0">\r
-\r
- <xsl:output method="html"/>\r
-\r
- <xsl:template match="coverage">\r
- <html>\r
- <head>\r
- <title>Coverage report</title>\r
- <style type="text/css">\r
- tbody tr.odd td.label {\r
- border-top: 1px solid rgb(128,128,128);\r
- border-bottom: 1px solid rgb(128,128,128);\r
- }\r
- tbody tr.odd td.label {\r
- background: rgb(200,200,200);\r
- }\r
- \r
- thead, tfoot {\r
- background: rgb(60,60,60);\r
- color: white;\r
- font-weight: bold;\r
- }\r
-\r
- tr td.perfect {\r
- background: rgb(0,255,0);\r
- color: black;\r
- }\r
- tr td.excellant {\r
- background: rgb(140,255,140);\r
- color: black;\r
- }\r
- tr td.good {\r
- background: rgb(160,255,0);\r
- color: black;\r
- }\r
- tr td.poor {\r
- background: rgb(255,160,0);\r
- color: black;\r
- }\r
- tr td.bad {\r
- background: rgb(255,140,140);\r
- color: black;\r
- }\r
- tr td.terrible {\r
- background: rgb(255,0,0);\r
- color: black;\r
- }\r
- </style>\r
- </head>\r
- <body>\r
- <h1>Coverage report</h1>\r
- <xsl:apply-templates/>\r
- </body>\r
- </html>\r
- </xsl:template>\r
-\r
- <xsl:template match="functions">\r
- <h2>Function coverage</h2>\r
- <xsl:call-template name="content">\r
- <xsl:with-param name="type" select="'function'"/>\r
- </xsl:call-template>\r
- </xsl:template>\r
- \r
-\r
- <xsl:template match="files">\r
- <h2>File coverage</h2>\r
- <xsl:call-template name="content">\r
- <xsl:with-param name="type" select="'file'"/>\r
- </xsl:call-template>\r
- </xsl:template>\r
-\r
- <xsl:template name="content">\r
- <xsl:param name="type"/>\r
- <table>\r
- <thead>\r
- <tr>\r
- <th>Name</th>\r
- <th>Lines</th>\r
- <th>Branches</th>\r
- <th>Conditions</th>\r
- <th>Calls</th>\r
- </tr>\r
- </thead>\r
- <tbody>\r
- <xsl:for-each select="entry">\r
- <xsl:call-template name="entry">\r
- <xsl:with-param name="type" select="$type"/>\r
- <xsl:with-param name="class">\r
- <xsl:choose>\r
- <xsl:when test="position() mod 2">\r
- <xsl:text>odd</xsl:text>\r
- </xsl:when>\r
- <xsl:otherwise>\r
- <xsl:text>even</xsl:text>\r
- </xsl:otherwise>\r
- </xsl:choose>\r
- </xsl:with-param>\r
- </xsl:call-template>\r
- </xsl:for-each>\r
- </tbody>\r
- <tfoot>\r
- <xsl:for-each select="summary">\r
- <xsl:call-template name="entry">\r
- <xsl:with-param name="type" select="'summary'"/>\r
- <xsl:with-param name="class">\r
- <xsl:choose>\r
- <xsl:when test="position() mod 2">\r
- <xsl:text>odd</xsl:text>\r
- </xsl:when>\r
- <xsl:otherwise>\r
- <xsl:text>even</xsl:text>\r
- </xsl:otherwise>\r
- </xsl:choose>\r
- </xsl:with-param>\r
- </xsl:call-template>\r
- </xsl:for-each>\r
- </tfoot>\r
- </table>\r
- </xsl:template>\r
- \r
- <xsl:template name="entry">\r
- <xsl:param name="type"/>\r
- <xsl:param name="class"/>\r
- <tr class="{$class}">\r
- <xsl:choose>\r
- <xsl:when test="$type = 'function'">\r
- <td class="label"><a href="{@details}.html#{@name}"><xsl:value-of select="@name"/></a></td>\r
- </xsl:when>\r
- <xsl:when test="$type = 'file'">\r
- <td class="label"><a href="{@details}.html"><xsl:value-of select="@name"/></a></td>\r
- </xsl:when>\r
- <xsl:otherwise>\r
- <td class="label">Summary</td>\r
- </xsl:otherwise>\r
- </xsl:choose>\r
-\r
- <xsl:if test="count(lines)">\r
- <xsl:apply-templates select="lines"/>\r
- </xsl:if>\r
- <xsl:if test="not(count(lines))">\r
- <xsl:call-template name="missing"/>\r
- </xsl:if>\r
-\r
- <xsl:if test="count(branches)">\r
- <xsl:apply-templates select="branches"/>\r
- </xsl:if>\r
- <xsl:if test="not(count(branches))">\r
- <xsl:call-template name="missing"/>\r
- </xsl:if>\r
-\r
- <xsl:if test="count(conditions)">\r
- <xsl:apply-templates select="conditions"/>\r
- </xsl:if>\r
- <xsl:if test="not(count(conditions))">\r
- <xsl:call-template name="missing"/>\r
- </xsl:if>\r
-\r
- <xsl:if test="count(calls)">\r
- <xsl:apply-templates select="calls"/>\r
- </xsl:if>\r
- <xsl:if test="not(count(calls))">\r
- <xsl:call-template name="missing"/>\r
- </xsl:if>\r
-\r
- </tr>\r
- </xsl:template>\r
- \r
- <xsl:template match="lines">\r
- <xsl:call-template name="row"/>\r
- </xsl:template>\r
-\r
- <xsl:template match="branches">\r
- <xsl:call-template name="row"/>\r
- </xsl:template>\r
-\r
- <xsl:template match="conditions">\r
- <xsl:call-template name="row"/>\r
- </xsl:template>\r
-\r
- <xsl:template match="calls">\r
- <xsl:call-template name="row"/>\r
- </xsl:template>\r
-\r
- <xsl:template name="missing">\r
- <td></td>\r
- </xsl:template>\r
-\r
- <xsl:template name="row">\r
- <xsl:variable name="quality">\r
- <xsl:choose>\r
- <xsl:when test="@coverage = 100">\r
- <xsl:text>perfect</xsl:text>\r
- </xsl:when>\r
- <xsl:when test="@coverage >= 80.0">\r
- <xsl:text>excellant</xsl:text>\r
- </xsl:when>\r
- <xsl:when test="@coverage >= 60.0">\r
- <xsl:text>good</xsl:text>\r
- </xsl:when>\r
- <xsl:when test="@coverage >= 40.0">\r
- <xsl:text>poor</xsl:text>\r
- </xsl:when>\r
- <xsl:when test="@coverage >= 20.0">\r
- <xsl:text>bad</xsl:text>\r
- </xsl:when>\r
- <xsl:otherwise>\r
- <xsl:text>terrible</xsl:text>\r
- </xsl:otherwise>\r
- </xsl:choose>\r
- </xsl:variable>\r
- \r
- <td class="{$quality}"><xsl:value-of select="@coverage"/>% of <xsl:value-of select="@count"/></td>\r
- </xsl:template>\r
-\r
-</xsl:stylesheet>\r
+++ /dev/null
-## .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
-
+++ /dev/null
-# 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
-
+++ /dev/null
-#!/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
-
-
+++ /dev/null
-#!/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 '<Project' "$2" >/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 ~ /<release>/ ) {x=1; chunk=""}
-if (x==1) {
- if ($0 ~ /<revision>/) { chunk = chunk $0 }
- if ($0 ~ /<created>/) { chunk = chunk $0 }
-}
-if ($0 ~ /<\/release>/) {x=0; print chunk}
-}' < "$2" | \
-\
-grep '<revision>'"$1"'</revision>' | \
-\
-sed -e 's/^.*<created>//' -e 's/<\/created>.*$//'
-
+++ /dev/null
---- 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@
-
+++ /dev/null
-# 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 __$(glib_enum_define)_ENUM_TYPES_H__\n\n#include <glib-object.h>\n\nG_BEGIN_DECLS\n" \
- --fprod "\n/* enumerations from \"@filename@\" */\n" \
- --vhead "GType @enum_name@_get_type (void);\n#define GST_TYPE_@ENUMSHORT@ (@enum_name@_get_type())\n" \
- --ftail "G_END_DECLS\n\n#endif /* __$(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 $@
+++ /dev/null
-# 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 $hook hook **********";
- fi
- }
- fi
- done
- fi
-}
+++ /dev/null
-#!/bin/sh
-#
-# Check that the code follows a consistant code style
-#
-
-# Check for existence of indent, and error out if not present.
-# On some *bsd systems the binary seems to be called gnunindent,
-# so check for that first.
-
-version=`gnuindent --version 2>/dev/null`
-if test "x$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} $@
-
+++ /dev/null
-### this file contains suppressions for valgrind when running
-### the gstreamer unit tests
-### it might be useful for wider use as well
-
-### syscall suppressions
-
-{
- <clone on Wim's Debian>
- Memcheck:Param
- clone(parent_tidptr)
- fun:clone
- fun:clone
-}
-
-{
- <clone on Wim's Debian>
- Memcheck:Param
- clone(child_tidptr)
- fun:clone
- fun:clone
-}
-
-{
- <clone on Wim's Debian>
- Memcheck:Param
- clone(tlsinfo)
- fun:clone
- fun:clone
-}
-
-### glibc suppressions
-
-{
- <conditional jump on wim's debian 2/2/06>
- 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
-}
-
-{
- <Conditional jump>
- 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:*
-}
-
-{
- <Conditional jump>
- Memcheck:Cond
- fun:_dl_relocate_object
- fun:dl_main
- fun:_dl_sysdep_start
- fun:_dl_start
-}
-
-{
- <insert a suppression name here>
- 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
-
-{
- <tls>
- Memcheck:Leak
- fun:calloc
- fun:_dl_allocate_tls
- fun:pthread_create@@*
-}
-
-{
- <tls>
- Memcheck:Leak
- fun:calloc
- fun:allocate_dtv
- fun:_dl_allocate_tls
-}
-
-# I get an extra stack entry on x86/dapper
-{
- <tls>
- Memcheck:Leak
- fun:calloc
- obj:/lib/ld-2.3.*.so
- fun:_dl_allocate_tls
- fun:pthread_create@@*
-}
-
-
-{
- <pthread strstr>
- 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
-}
-
-{
- <a conditional jump on wim's debian>
- 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
-{
- <started showing up on fc4-quick>
- 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
-}
-
-{
- <started showing up on fc4-quick>
- 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
-}
-
-{
- <g_module_open on wim's debian>
- 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
-}
-
-{
- <g_module_open on wim's debian>
- 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
-}
-{
- <g_module_open on wim's debian>
- 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
-}
-
-{
- <leak on wim's debian in g_module_open>
- 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
-}
-
-{
- <invalid read on wim's debian>
- 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
-}
-
-{
- <invalid read on wim's debian>
- 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
-}
-
-{
- <invalid read on wim's debian - 2006-02-02>
- 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
-}
-
-{
- <invalid read on wim's debian - 2006-02-02>
- 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
-}
-
-{
- <invalid read on wim's debian - 2006-02-02>
- 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
-}
-
-{
- <futex on Andy's 64-bit ubuntu>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on Ubunty Hardy 64-bit>
- 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
-}
-
-{
- <suppression for glibc 2.7 on Ubunty Hardy 64-bit>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on Ubunty Hardy 64-bit>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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
-}
-
-{
- <suppression for glibc 2.7 on debian>
- 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>
-{
- <Workaround for a glibc bug>
- 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
-{
- <workaround glibc bug 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
-{
- <Invalid read of size 1, 2, 4 on thomas's FC4>
- 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
-}
-
-{
- <Invalid read of size 1, 2, 4 on thomas's FC4>
- 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
-}
-{
- <Invalid read of size 1, 2, 4 on thomas's FC4>
- 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
-}
-
-{
- <Invalid read of size 1, 2, 4 on thomas's FC4>
- 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
-}
-
-{
- <Invalid read of size 1, 2, 4 on thomas's FC4>
- 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
-}
-{
- <Invalid read of size 1, 2, 4 on thomas's FC4>
- 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
-}
-
-{
- <Addr8 on Andy's AMD64 ubuntu in dl_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
-}
-
-{
- <Conditional jump on Andy's AMD64 ubuntu>
- 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
-}
-
-{
- <Mike's x86 dapper>
- 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
-}
-
-{
- <Mike's x86 dapper>
- 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
-}
-
-{
- <Another dapper one>
- 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
-{
- <g_parse_debug_string>
- Memcheck:Cond
- fun:g_parse_debug_string
- obj:/usr/lib*/libglib-2.0.so.*
- fun:g_slice_alloc
- fun:g_slice_alloc0
-}
-
-{
- <g_type_init leaks>
- Memcheck:Leak
- fun:*alloc
- ...
- fun:g_type_init*
- fun:init_pre*
-}
-
-{
- <g_type_register_fundamental leaks>
- Memcheck:Leak
- fun:*alloc
- ...
- fun:g_type_register_fundamental
-}
-
-{
- <glib 2.21 static type data>
- Memcheck:Leak
- fun:malloc
- fun:realloc
- fun:g_realloc
- fun:type_node_any_new_W
-}
-
-{
- <glib 2.21 static type data>
- Memcheck:Leak
- fun:realloc
- fun:g_realloc
- fun:type_node_any_new_W
-}
-
-{
- <glib 2.21 static type data>
- Memcheck:Leak
- fun:calloc
- fun:g_malloc0
- fun:g_type_class_ref
-}
-
-{
- <glib 2.21 static type data>
- Memcheck:Leak
- fun:malloc
- fun:realloc
- fun:g_realloc
- fun:type_add_flags_W
-}
-
-{
- <glib 2.21 static type data>
- 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
-{
- <liboil cpu_fault_check_try>
- 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
-}
-
-{
- <annoying read error inside dlopen stuff on Ubuntu Dapper x86_64>
- Memcheck:Addr8
- obj:/lib/ld-2.3.6.so
-}
-
-{
- <Ubuntu Dapper x86_64>
- 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
-}
-
-{
- <Ubuntu Dapper x86_64 dlopen stuff again>
- 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
-{
- <dlopen invalid read of size 4 suppression on tpm's Ubuntu edgy/x86>
- Memcheck:Addr4
- obj:/lib/ld-2.4.so
- obj:/lib/ld-2.4.so
-}
-{
- <and the same for 64bit systems>
- Memcheck:Addr8
- obj:/lib/ld-2.4.so
- obj:/lib/ld-2.4.so
-}
-
-# More edgy suppressions (Mike)
-{
- <dlopen Condition jump suppressions for Ubuntu Edgy/x86>
- 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
-}
-
-{
- <dlopen Condition jump suppressions for Ubuntu Edgy/x86>
- 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
-}
-
-{
- <dlopen Condition jump suppressions for Ubuntu Edgy/x86>
- 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!
-{
- <dlopen Condition jump suppressions for Ubuntu Edgy/x86>
- 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
-}
-
-{
- <tls leaks on Edgy/x86>
- 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
-{
- <tls leaks on Feisty/x86>
- Memcheck:Leak
- fun:calloc
- fun:allocate_dtv
- fun:_dl_allocate_tls
- fun:pthread_create@@GLIBC_2.1
-}
-
-{
- <libcdio 0.76 leak>
- 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
-}
-
-{
- <Addr8 on Jan's AMD64 ubuntu Feisty in dl_open>
- Memcheck:Addr8
- obj:/lib/ld-2.5.so
-}
-
-{
- <First of many Alsa errors>
- 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
-}
-
-{
- <alsa error>
- Memcheck:Cond
- fun:snd*_pcm_hw_param_set_near
-}
-
-{
- <alsa error>
- Memcheck:Cond
- ...
- fun:snd*_pcm_hw_param_set_near
-}
-
-{
- <alsa error>
- 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
-}
-{
- <alsa error>
- 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
-}
-{
- <alsa error>
- 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
-{
- <alsa error>
- Memcheck:Leak
- fun:malloc
- ...
- fun:snd*_dlobj_cache_add
- obj:/*lib*/libasound.so.2.0.0
-}
-
-{
- <alsa leak in loading configuration>
- Memcheck:Leak
- fun:*alloc
- ...
- fun:snd_pcm_open_conf
-}
-
-{
- <alsa leak snd_config_hook_load>
- Memcheck:Leak
- fun:*alloc
- obj:/*lib*/libasound.so.2.0.0
- ...
- fun:snd_config_hook_load
-}
-
-{
- <alsa leak snd_config_update_r>
- Memcheck:Leak
- fun:*alloc
- obj:/*lib*/libasound.so.2.0.0
- ...
- fun:snd_config_update_r
- fun:snd_config_update
-}
-{
- <alsa leak snd_config_update_r>
- Memcheck:Leak
- fun:*alloc
- fun:strdup
- ...
- fun:snd_config_update_r
- fun:snd_config_update
-}
-{
- <alsa leak snd_config_searcha_hooks>
- Memcheck:Leak
- fun:*alloc
- fun:_dl_close_worker
- ...
- fun:snd_config_searcha_hooks
-}
-
-{
- <nss lookup within ALSA>
- 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
-}
-
-{
- <libxcb leak on Ubuntu Feisty>
- Memcheck:Leak
- fun:calloc
- fun:_XCBInitDisplayLock
- fun:XOpenDisplay
-}
-
-# GConf internal initialisations related to getting the default client.
-{
- <Orbit something or other>
- 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
-}
-{
- <gconf internal leak>
- 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
-}
-{
- <gconf internal leak>
- 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
-}
-{
- <gconf internal initialisation>
- 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
-}
-{
- <gconf internal init>
- 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
-}
-{
- <gconf internal init>
- 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
-}
-{
- <gconf internal init>
- 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
-}
-
-{
- <insert a suppression name here>
- 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
-{
- <bonobo init>
- 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
-}
-{
- <bonobo 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*
-}
-{
- <bonobo init>
- 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
-}
-{
- <bonobo init - more recent variant of above>
- 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
-}
-{
- <bonobo 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
-{
- <incompletely initialised ORBit buffer>
- 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
-}
-{
- <gconf 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/libgconf-2.so*
- obj:/usr/lib/libgconf-2.so*
- obj:/usr/lib/libgconf-2.so*
- fun:gconf_engine_get_default
-}
-{
- <gconf init>
- 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
-}
-{
- <gconf init>
- 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
-}
-{
- <bonobo init>
- 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
-}
-{
- <gconf init>
- 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
-}
-{
- <GLib caching the 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_home_dir
-}
-{
- <GLib caching the 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_user_name
-}
-{
- <GLib caching the 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.*
- fun:g_get_tmp_dir
-}
-
-{
- <GLib caching the host name>
- 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.
-{
- <First time load of a font - feisty x86_64>
- 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
-}
-{
- <First time load of a font - generic>
- Memcheck:Leak
- fun:*alloc
- ...
- fun:FcInitLoadConfig
-}
-
-# Issues with ubuntu Hardy, same crack as for previous ubuntus
-{
- <tls leak generic ubuntu hardy x86>
- 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
-{
- <tls leak generic>
- 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
-{
- <invalid read of size 4 within <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/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_*
-}
-
-{
- <invalid read of size 4 within <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/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_*
-}
-
-{
- <invalid read of size 4 within <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/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_*
-}
-
-{
- <invalid read of size 4 within <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/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_*
-}
-
-{
- <invalid read of size 4 within <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/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*
-}
-
-{
- <invalid read of size 4 within <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/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
-{
- <invalid read of size 8 within <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/libdl-2.7.so
- obj:/lib/ld-2.7.so
- obj:/lib/libdl-2.7.so
- fun:dlopen
- fun:g_module_open
-}
-
-{
- <invalid read of size 8 within <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
-}
-
-{
- <invalid read of size 8 within <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
-}
-
-{
- <invalid read of size 8 within <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
-}
-
-{
- <invalid read of size 8 within <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
-}
-
-{
- <invalid read of size 8 within <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
-}
-
-{
- <invalid read of size 8 within <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
-}
-
-{
- <GLib caching>
- 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
-}
-
-{
- <GLib caching>
- 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
-}
-
-{
- <GLib caching>
- 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
-}
-
-{
- <GLib caching>
- 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
-}
-
-{
- <GLib caching>
- 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)
-
-{
- <Alsa leak>
- Memcheck:Leak
- fun:calloc
- fun:_snd_config_make
- fun:_snd_config_make_add
- fun:*
- fun:*
- fun:*
- fun:*
- fun:snd_config_load1
-}
-
-{
- <Alsa leak>
- Memcheck:Leak
- fun:calloc
- fun:_snd_config_make
- fun:_snd_config_make_add
- fun:*
- fun:*
- fun:snd_config_load1
-}
-{
- <Alsa leak>
- Memcheck:Leak
- fun:calloc
- fun:_snd_config_make
- fun:_snd_config_make_add
- fun:*
- fun:*
- fun:*
- fun:snd_config_load1
-}
-{
- <Alsa leak>
- Memcheck:Leak
- fun:calloc
- fun:_snd_config_make
- fun:_snd_config_make_add
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:snd_config_load1
-}
-
-{
- <Alsa leak>
- Memcheck:Leak
- fun:calloc
- fun:_snd_config_make
- fun:_snd_config_make_add
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:snd_config_load1
-}
-
-{
- <Alsa leak>
- Memcheck:Leak
- fun:calloc
- fun:_snd_config_make
- fun:_snd_config_make_add
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:snd_config_load1
-}
-{
- <Alsa leak>
- 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
-}
-
-{
- <Alsa leak>
- Memcheck:Leak
- fun:malloc
- fun:snd1_dlobj_cache_add
- fun:snd_ctl_open_noupdate
-}
-
-{
- <Alsa leak>
- Memcheck:Leak
- fun:malloc
- fun:*
- fun:snd1_dlobj_cache_add
- fun:snd_ctl_open_noupdate
-}
-
-{
- <Alsa leak>
- Memcheck:Leak
- fun:*alloc
- fun:*
- fun:*
- fun:*
- fun:snd_config_load1
-}
-
-{
- <Alsa leak>
- Memcheck:Leak
- fun:*alloc
- fun:*
- fun:*
- fun:*
- fun:*
- fun:snd_config_load1
-}
-
-{
- <Alsa leak>
- Memcheck:Leak
- fun:*alloc
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:snd_config_load1
-}
-
-{
- <Alsa leak>
- Memcheck:Leak
- fun:*alloc
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:snd_config_load1
-}
-
-{
- <Alsa leak>
- Memcheck:Leak
- fun:*alloc
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:snd_config_load1
-}
-
-{
- <Alsa leak>
- Memcheck:Leak
- fun:*alloc
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:snd_config_load1
-}
-
-{
- <Alsa leak>
- Memcheck:Leak
- fun:*alloc
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:snd_config_load1
-}
-
-{
- <Alsa leak>
- 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).
-
-{
- <Leak in ogmparsers>
- 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
-}
-
-{
- <Leak in ogmparsers>
- 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
-}
-
-{
- <Leak in videotestsrc>
- 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
-}
-
-{
- <Leak in videotestsrc>
- 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
-}
-
-{
- <Leak in ffmpegcolorspace>
- 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
-}
-
-{
- <Leak in ffmpegocolorspace>
- 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
-}
-
-{
- <Leak in gstffmpegdemux>
- 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
-}
-
-{
- <Leak in GstAudioFilter subclasses>
- 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
-}
-
-{
- <Leak in GstAudioFilter subclasses, variant>
- 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
-}
-
-{
- <Leak in GstAudioFilter subclasses, variant>
- 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
-}
-
-{
- <Leak in GstAudioFilter subclasses, variant>
- 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
-}
-
-{
- <Leak in GstAudioFilter subclasses, variant2>
- 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
-}
-{
- <Leak in GstAudioFilter subclasses, variant3>
- 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
-}
-{
- <Leak in GstAudioFilter subclasses, variant4>
- 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
-}
-{
- <Leak in GstAudioFilter subclasses, variant5>
- 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
-}
-
-{
- <Leak in riff-media>
- Memcheck:Leak
- fun:malloc
- fun:g_malloc
- fun:g_slice_alloc
- fun:gst_caps_new_empty
- fun:gst_riff_create_*_template_caps
-}
-{
- <Leak in riff-media>
- 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
-}
-{
- <Leak in riff-media>
- 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)
-
-{
- <Pango leak - generic>
- Memcheck:Leak
- fun:*alloc
- ...
- fun:pango_layout_get_pixel_extents
-}
-{
- <insert a suppression name here>
- 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
-}
-
-{
- <PangoLanguage can never be freed>
- Memcheck:Leak
- fun:calloc
- fun:g_malloc0
- fun:pango_language_from_string
-}
-
-
-## Leak of everything allocated by gst-libav plugin init
-{
- <insert_a_suppression_name_here>
- Memcheck:Leak
- fun:*alloc
- ...
- fun:gst_ffmpeg_cfg_init
-}
-
-## Leak of GIO module through gnomevfs
-
-{
- <gio leak>
- 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)
-{
- <Leak of addrinfo in esd>
- Memcheck:Cond
- fun:gaih_inet
- fun:getaddrinfo
-}
-
-## Dynamic pad templates in mxfmux
-{
- <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
-{
- <insert a suppression name here>
- 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
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:do_sym
- fun:dlsym_doit
- fun:_dl_catch_error
- fun:_dlerror_run
- fun:dlsym
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:do_sym
- fun:dlsym_doit
- fun:_dl_catch_error
- fun:_dlerror_run
- fun:dlsym
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- 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*
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- 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*
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- 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*
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- 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*
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- 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*
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- 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*
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- 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*
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- 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*
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- 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*
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- 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*
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- obj:/lib*/libc-2.10.*.so
- obj:/lib*/libc-2.10.*.so
- fun:_vgnU_freeres
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- obj:/lib*/libc-2.10.*.so
- obj:/lib*/libc-2.10.*.so
- fun:_vgnU_freeres
-}
-{
- <glibc-2.10 mysterious invalid free on exit>
- Memcheck:Free
- fun:free
- obj:/lib*/libc-2.10.*.so
- obj:/lib*/libc-2.10.*.so
- fun:_vgnU_freeres
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_fini
- fun:__run_exit_handlers
- fun:exit
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:_dl_fini
- fun:__run_exit_handlers
- fun:exit
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_sort_fini
- fun:_dl_fini
- fun:__run_exit_handlers
- fun:exit
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:_dl_sort_fini
- fun:_dl_fini
- fun:__run_exit_handlers
- fun:exit
-}
-
-# glibc-2.10 dl overreads
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_fixup
- fun:_dl_runtime_resolve
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:_dl_fixup
- fun:_dl_runtime_resolve
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_lookup_symbol_x
- fun:_dl_fixup
- fun:_dl_runtime_resolve
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:_dl_lookup_symbol_x
- fun:_dl_fixup
- fun:_dl_runtime_resolve
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:call_init
- fun:_dl_init
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_init
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:do_lookup_x
- fun:_dl_lookup_symbol_x
- fun:_dl_relocate_object
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:do_lookup_x
- fun:_dl_lookup_symbol_x
- fun:_dl_relocate_object
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_lookup_symbol_x
- fun:_dl_relocate_object
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_relocate_object
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:dl_main
- fun:_dl_sysdep_start
- fun:_dl_start
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:dl_main
- fun:_dl_sysdep_start
- fun:_dl_start
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:do_lookup_x
- fun:_dl_lookup_symbol_x
- fun:_dl_relocate_object
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:do_lookup_x
- fun:_dl_lookup_symbol_x
- fun:_dl_relocate_object
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_check_map_versions
- fun:_dl_check_all_versions
- fun:version_check_doit
- fun:_dl_receive_error
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:_dl_check_map_versions
- fun:_dl_check_all_versions
- fun:version_check_doit
- fun:_dl_receive_error
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_check_all_versions
- fun:version_check_doit
- fun:_dl_receive_error
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:_dl_check_all_versions
- fun:version_check_doit
- fun:_dl_receive_error
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:_dl_check_map_versions
- fun:_dl_check_all_versions
- fun:version_check_doit
- fun:_dl_receive_error
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:_dl_check_map_versions
- fun:_dl_check_all_versions
- fun:version_check_doit
- fun:_dl_receive_error
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:init_tls
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:init_tls
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:_dl_map_object_deps
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_map_object_deps
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_protect_relro
- fun:_dl_relocate_object
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:do_lookup_x
- fun:_dl_lookup_symbol_x
- fun:_dl_relocate_object
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_setup_hash
- fun:_dl_map_object_from_fd
- fun:_dl_map_object
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:_dl_new_object
- fun:_dl_map_object_from_fd
- fun:_dl_map_object
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:_dl_new_object
- fun:_dl_map_object_from_fd
- fun:_dl_map_object
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:openaux
- fun:_dl_catch_error
- fun:_dl_map_object_deps
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:_dl_map_object
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:_dl_map_object
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:_dl_map_object
- fun:openaux
- fun:_dl_catch_error
- fun:_dl_map_object_deps
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_map_object
- fun:openaux
- fun:_dl_catch_error
- fun:_dl_map_object_deps
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:_dl_map_object
- fun:openaux
- fun:_dl_catch_error
- fun:_dl_map_object_deps
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:open_path
- fun:_dl_map_object
- fun:openaux
- fun:_dl_catch_error
- fun:_dl_map_object_deps
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:open_path
- fun:_dl_map_object
- fun:openaux
- fun:_dl_catch_error
- fun:_dl_map_object_deps
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_map_object_from_fd
- fun:_dl_map_object
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:_dl_map_object_from_fd
- fun:_dl_map_object
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:_dl_new_object
- fun:_dl_map_object_from_fd
- fun:_dl_map_object
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_new_object
- fun:_dl_map_object_from_fd
- fun:_dl_map_object
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:_dl_new_object
- fun:_dl_map_object_from_fd
- fun:_dl_map_object
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:_dl_name_match_p
- fun:_dl_map_object
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:*
- fun:_dl_map_object
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:_dl_name_match_p
- fun:_dl_check_map_versions
- fun:_dl_check_all_versions
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:*
- fun:do_lookup_x
- fun:_dl_lookup_symbol_x
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:do_lookup_x
- fun:_dl_lookup_symbol_x
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:do_lookup_x
- fun:_dl_lookup_symbol_x
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:do_lookup_x
- fun:_dl_lookup_symbol_x
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:do_lookup_x
- fun:_dl_lookup_symbol_x
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_name_match_p
- fun:_dl_map_object
- fun:dl_open_worker
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:_dl_name_match_p
- fun:_dl_map_object
- fun:dl_open_worker
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:_dl_name_match_p
- fun:_dl_map_object
- fun:dl_open_worker
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:_dl_name_match_p
- fun:_dl_map_object
- fun:dl_open_worker
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_lookup_symbol_x
- fun:_dl_relocate_object
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:_dl_lookup_symbol_x
- fun:_dl_relocate_object
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:*
- fun:_dl_check_map_versions
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:_dl_check_map_versions
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:*
- fun:_dl_check_map_versions
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:_dl_check_map_versions
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:openaux
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_name_match_p
- fun:_dl_map_object
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:_dl_close_worker
- fun:_dl_close
- fun:_dl_catch_error
- fun:dlerror_run
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_close_worker
- fun:_dl_close
- fun:_dl_catch_error
- fun:dlerror_run
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:_dl_close_worker
- fun:_dl_close
- fun:_dl_catch_error
- fun:dlerror_run
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:_dl_close_worker
- fun:_dl_close
- fun:_dl_catch_error
- fun:dlerror_run
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:fillin_rpath
- fun:_dl_init_paths
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:fillin_rpath
- fun:_dl_init_paths
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:fillin_rpath
- fun:_dl_init_paths
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:fillin_rpath
- fun:_dl_init_paths
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:_dl_map_object
- fun:map_doit
- fun:_dl_catch_error
- fun:do_preload
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_map_object
- fun:map_doit
- fun:_dl_catch_error
- fun:do_preload
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- 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
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- 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
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_catch_error
- fun:_dl_map_object_deps
- fun:dl_open_worker
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:_dl_map_object_deps
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:_dl_map_object_deps
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:*
- fun:_dl_map_object_deps
- fun:dl_main
-}
-
-# glibc-2.10 tls issues
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:init_tls
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:init_tls
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:*
- fun:init_tls
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:*
- fun:init_tls
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:_dl_allocate_tls_init
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:_dl_allocate_tls_init
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:*
- fun:_dl_allocate_tls_init
- fun:dl_main
-}
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Value8
- fun:*
- fun:_dl_allocate_tls_init
- fun:dl_main
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Cond
- fun:__tls*
- obj:*
- obj:*
- fun:_vgnU_freeres
-}
-
-{
- <glibc-2.10 overreads/conditionals>
- Memcheck:Param
- arch_prctl(arg2)
- fun:init_tls
-}
-# GLib caching tmp/home directories (glibc-2.10 variants)
-{
- <glibc-2.10 GLIB leaks>
- 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*
-}
-{
- <glibc-2.10 GLIB leaks>
- 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*
-}
-{
- <glibc-2.10 GLIB leaks>
- 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*
-}
-{
- <glibc-2.10 GLIB leaks>
- 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*
-}
-
-{
- <glibc-2.10 GLIB leaks>
- Memcheck:Value8
- fun:_dl_add_to_slotinfo
- fun:dl_main
-}
-{
- <glibc-2.10 GLIB leaks>
- Memcheck:Param
- open(filename)
- fun:open
- fun:open_verify
- fun:open_path
- fun:_dl_map_object
-}
-
-
-
-# GModule issues with glibc-2.10
-{
- <glibc-2.10 GLIB leaks>
- Memcheck:Value8
- fun:*
- fun:*
- fun:dlsym
- fun:g_module_symbol
-}
-{
- <glibc-2.10 GLIB leaks>
- Memcheck:Value8
- fun:g_module_*
- fun:gst_plugin*
-}
-{
- <glibc-2.10 GLIB leaks>
- Memcheck:Value8
- fun:*
- fun:g_module_*
- fun:gst_plugin*
-}
-
-{
- <glibc-2.10 GLIB leaks>
- Memcheck:Value8
- fun:*
- fun:*
- fun:dlopen*
- fun:g_module_open
-}
-{
- <glibc-2.10 GLIB leaks>
- Memcheck:Value8
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:dlsym
- fun:g_module_symbol
-}
-
-{
- <glibc-2.10 GLIB leaks>
- Memcheck:Value8
- fun:*
- fun:*
- fun:*
- fun:*
- fun:*
- fun:dlopen*
- fun:g_module_open
-}
-
-# Leak in GSlice
-{
- <insert a suppression name here>
- 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
-{
- <insert a suppression name here>
- Memcheck:Value8
- fun:__pthread_initialize_minimal
-}
-
-# glibc 2.11 conditional
-{
- <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
-
-{
- <insert_a_suppression_name_here>
- 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
-}
-
-{
- <insert_a_suppression_name_here>
- 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
-}
-
-{
- <insert_a_suppression_name_here>
- 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
-}
-
-{
- <insert_a_suppression_name_here>
- 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
-}
-
-{
- <insert_a_suppression_name_here>
- 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
-}
-
-{
- <insert_a_suppression_name_here>
- 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
-}
-
-{
- <insert_a_suppression_name_here>
- 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
-{
- <insert_a_suppression_name_here>
- Memcheck:Leak
- fun:*alloc
- ...
- fun:g_type_register_static
-}
-
-# new registry system
-# all of this will only be created once when loading registry.
-
-{
- <insert_a_suppression_name_here>
- Memcheck:Leak
- fun:*alloc
- ...
- fun:_priv_gst_registry_chunks_load_plugin
-}
-
-# system-wide tags
-# these tags are registered once
-
-{
- <insert_a_suppression_name_here>
- Memcheck:Leak
- fun:*alloc
- fun:*
- fun:*
- fun:gst_tag_register
- fun:_gst_tag_initialize
-}
-
-# system-wide type classes that we keep referenced
-
-{
- <g_type_class_ref leaks>
- Memcheck:Leak
- fun:*alloc
- ...
- fun:g_type_class_ref
-}
-
-# leaking cached queries which are only initialized once
-{
- <insert_a_suppression_name_here>
- Memcheck:Leak
- fun:*alloc
- ...
- fun:_gst_query_initialize
- fun:init_post
-}
-
-# macosx (leopard) library loader leak
-{
- <insert_a_suppression_name_here>
- 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
-{
- <insert_a_suppression_name_here>
- Memcheck:Leak
- fun:*alloc
- ...
- fun:_g_atomic_array_copy
-}
-
-{
- <getdelim one-time inits called from libselinux>
- Memcheck:Leak
- fun:*alloc
- fun:getdelim
- obj:*libselinux*
-}
-
-{
- <weird one when re-reading registry>
- Memcheck:Leak
- fun:*alloc
- ...
- obj:*/sed
-}
-
-{
- <weird one when re-reading registry>
- Memcheck:Addr8
- ...
- obj:*/sed
-}
-
-# GLib 2.23 interface vtable
-{
- <insert_a_suppression_name_here>
- Memcheck:Leak
- fun:*alloc
- ...
- fun:g_type_add_interface_static
-}
-
-{
- <leak in dash on debian sid>
- Memcheck:Leak
- fun:*alloc
- obj:*/dash
-}
-
-# libtool/gentoo fake leak
-# it actually runs bash and valgrind complains
-{
- <insert_a_suppression_name_here>
- Memcheck:Leak
- fun:*alloc
- obj:/bin/bash
-}
-
-{
- <ignore possbly-lost leaks in the plugin scanner which doesn't clean up properly>
- Memcheck:Leak
- fun:*alloc
- ...
- fun:_gst_plugin_loader_client_run
- fun:main
-}
-
-{
- <warning with libc 2.13-2 as in Debian/unstable on amd64>
- Memcheck:Cond
- fun:*strcasecmp*
- ...
- fun:__dcigettext
-}
-
-{
- <warning with libc 2.13-2 as in Debian/unstable on amd64>
- Memcheck:Value8
- fun:*strcasecmp*
- ...
- fun:__dcigettext
-}
-
-{
- <GstSystemClock is a singleton and does not leak>
- Memcheck:Leak
- fun:malloc
- ...
- fun:gst_poll_new
- fun:gst_poll_new_timer
- fun:gst_system_clock_init
-}
-
-{
- <glib types are singletons>
- Memcheck:Leak
- fun:calloc
- ...
- fun:gobject_init_ctor
-}
-
-{
- <quark table is leaked on purpose if it grows too big>
- Memcheck:Leak
- fun:malloc
- ...
- fun:g_quark_from*_string
-}
+++ /dev/null
-#!/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 <<EOF;
-gstdoc-scangobj version 1.5 - introspect gstreamer-plugins
-
---module=MODULE_NAME Name of the doc module being parsed
---source=SOURCE_NAME Name of the source module for plugins
---types=FILE The name of the file to store the types in
---type-init-func=FUNC The init function to call instead of g_type_init()
---output-dir=DIRNAME The directory where the results are stored
---inspect-dir=DIRNAME The directory where the plugin inspect data is stored
---verbose Print extra output while processing
---version Print the version of this program
---help Print this help
-EOF
- exit 0;
-}
-
-$OUTPUT_DIR = $OUTPUT_DIR ? $OUTPUT_DIR : ".";
-
-$TYPES_FILE = $TYPES_FILE ? $TYPES_FILE : "$OUTPUT_DIR/$MODULE.types";
-
-open (TYPES, $TYPES_FILE) || die "Cannot open $TYPES_FILE: $!\n";
-open (OUTPUT, ">$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 (<TYPES>) {
- 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 <<EOT;
-
-/* file generated by common/gstdoc-scangobj */
-
-#include <string.h>
-#include <stdlib.h>
-#include <stdio.h>
-#include <errno.h>
-
-#include <gst/gst.h>
-EOT
-
-if ($includes) {
- print OUTPUT $includes;
-} else {
- for (@types) {
- print OUTPUT "extern GType $_ (void);\n";
- }
-}
-
-print OUTPUT <<EOT;
-
-#ifdef GTK_IS_WIDGET_CLASS
-#include <gtk/gtkversion.h>
-#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, "</%s>\\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 ("<plugin>\\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 (" <elements>\\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 (" <element>\\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 (" <pads>\\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 (" <caps>\\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 (" </caps>\\n", inspect);
- }
- g_list_free (pads);
- fputs (" </pads>\\n </element>\\n", inspect);
- }
- }
- features = g_list_next (features);
- }
-
- if (reinspect) {
- fputs (" </elements>\\n</plugin>", 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 <<EOT;
- type = $_ ();
- if (type == 0) {
- g_message ("$_ () didn't return a valid type");
- }
- else {
- object_types[i++] = type;
- }
-EOT
-}
-
-# Implicit types retrieved from GLib:
-for (@impl_types) {
-print OUTPUT <<EOT;
- type = g_type_from_name ("$_");
- if (type == 0) {
- g_message ("Implicit type $_ not found");
- }
- else {
- object_types[i++] = type;
- }
-EOT
-}
-
-print OUTPUT <<EOT;
-
- object_types[i] = 0;
-
- /* reference the GObjectClass to initialize the param spec pool
- * potentially needed by interfaces. See http://bugs.gnome.org/571820 */
- g_object_class = g_type_class_ref (G_TYPE_OBJECT);
-
- /* Need to make sure all the types are loaded in and initialize
- * their signals and properties.
- */
- for (i=0; object_types[i]; i++)
- {
- if (G_TYPE_IS_CLASSED (object_types[i]))
- g_type_class_ref (object_types[i]);
- if (G_TYPE_IS_INTERFACE (object_types[i]))
- g_type_default_interface_ref (object_types[i]);
- }
-
- g_type_class_unref (g_object_class);
-
- return object_types;
-}
-
-/*
- * This uses GObject type functions to output signal prototypes and the object
- * hierarchy.
- */
-
-/* The output files */
-const gchar *signals_filename = "$new_signals_filename";
-const gchar *hierarchy_filename = "$new_hierarchy_filename";
-const gchar *interfaces_filename = "$new_interfaces_filename";
-const gchar *prerequisites_filename = "$new_prerequisites_filename";
-const gchar *args_filename = "$new_args_filename";
-
-
-static void output_signals (void);
-static void output_object_signals (FILE *fp,
- GType object_type);
-static void output_object_signal (FILE *fp,
- const gchar *object_class_name,
- guint signal_id);
-static const gchar * get_type_name (GType type,
- gboolean * is_pointer);
-static void output_object_hierarchy (void);
-static void output_hierarchy (FILE *fp,
- GType type,
- guint level);
-
-static void output_object_interfaces (void);
-static void output_interfaces (FILE *fp,
- GType type);
-
-static void output_interface_prerequisites (void);
-static void output_prerequisites (FILE *fp,
- GType type);
-
-static void output_args (void);
-static void output_object_args (FILE *fp, GType object_type);
-
-int
-main (G_GNUC_UNUSED int argc, G_GNUC_UNUSED char *argv[])
-{
- $TYPE_INIT_FUNC;
-
- get_object_types ();
-
- output_signals ();
- output_object_hierarchy ();
- output_object_interfaces ();
- output_interface_prerequisites ();
- output_args ();
-
- return 0;
-}
-
-
-static void
-output_signals (void)
-{
- FILE *fp;
- gint i;
-
- fp = fopen (signals_filename, "w");
- if (fp == NULL)
- {
- g_warning ("Couldn't open output file: %s : %s", signals_filename, g_strerror(errno));
- return;
- }
-
- for (i = 0; object_types[i]; i++)
- output_object_signals (fp, object_types[i]);
-
- fclose (fp);
-}
-
-static gint
-compare_signals (const void *a, const void *b)
-{
- const guint *signal_a = a;
- const guint *signal_b = b;
-
- return strcmp (g_signal_name (*signal_a), g_signal_name (*signal_b));
-}
-
-/* This outputs all the signals of one object. */
-static void
-output_object_signals (FILE *fp, GType object_type)
-{
- const gchar *object_class_name;
- guint *signals, n_signals;
- guint sig;
-
- if (G_TYPE_IS_INSTANTIATABLE (object_type) ||
- G_TYPE_IS_INTERFACE (object_type))
- {
-
- object_class_name = g_type_name (object_type);
-
- signals = g_signal_list_ids (object_type, &n_signals);
- qsort (signals, n_signals, sizeof (guint), compare_signals);
-
- for (sig = 0; sig < n_signals; sig++)
- {
- output_object_signal (fp, object_class_name, signals[sig]);
- }
- g_free (signals);
- }
-}
-
-
-/* This outputs one signal. */
-static void
-output_object_signal (FILE *fp,
- const gchar *object_name,
- guint signal_id)
-{
- GSignalQuery query_info;
- const gchar *type_name, *ret_type, *object_arg, *arg_name;
- gchar *pos, *object_arg_lower;
- gboolean is_pointer;
- gchar buffer[1024];
- guint i, param;
- gint param_num, widget_num, event_num, callback_num;
- gint *arg_num;
- gchar signal_name[128];
- gchar flags[16];
-
- $debug_log ("Object: %s Signal: %u", object_name, signal_id);
-
- param_num = 1;
- widget_num = event_num = callback_num = 0;
-
- g_signal_query (signal_id, &query_info);
-
- /* Output the signal object type and the argument name. We assume the
- type is a pointer - I think that is OK. We remove "Gtk" or "Gnome" and
- convert to lower case for the argument name. */
- pos = buffer;
- sprintf (pos, "%s ", object_name);
- pos += strlen (pos);
-
- /* Try to come up with a sensible variable name for the first arg
- * It chops off 2 know prefixes :/ and makes the name lowercase
- * It should replace lowercase -> 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,
- "<SIGNAL>\\n<NAME>%s::%s</NAME>\\n<RETURNS>%s%s</RETURNS>\\n<FLAGS>%s</FLAGS>\\n%s</SIGNAL>\\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, "<ARG>\\n<NAME>%s::%s</NAME>\\n<TYPE>%s%s</TYPE>\\n<RANGE>%s</RANGE>\\n<FLAGS>%s</FLAGS>\\n<NICK>%s</NICK>\\n<BLURB>%s%s</BLURB>\\n<DEFAULT>%s</DEFAULT>\\n</ARG>\\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);
-
+++ /dev/null
-# 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 '<version>' $$each | grep -v '<version>$(VERSION)'); then \
- echo $$each should be fixed to say version $(VERSION) or be removed ; \
- echo "sed -i -e 's/<version.*version>/<version>$(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 $$files; do \
- ver=`grep '<version>$(PACKAGE_VERSION)</version>' $$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:
+++ /dev/null
-###########################################################################
-# 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:
+++ /dev/null
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+++ /dev/null
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-dnl David Schleef <ds@schleef.org>
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-])
-
-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])
-])
-
+++ /dev/null
-dnl as-compiler.m4 0.1.0
-
-dnl autostars m4 macro for detection of compiler flavor
-
-dnl Thomas Vander Stichele <thomas at apestaart dot org>
-
-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
-])
+++ /dev/null
-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='<!--'
- CAT_ENTRY_END='-->'
- 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=$i
- fi
- done
- else
- XML_CATALOG=/etc/xml/catalog
- CAT_ENTRY_START='<!--'
- CAT_ENTRY_END='-->'
- fi
-
- dnl We need xsltproc to process the test
- AC_CHECK_PROG(XSLTPROC,xsltproc,xsltproc,)
- XSLTPROC_WORKS=no
- if test -n "$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
-<?xml version="1.0" encoding='ISO-8859-1'?>
-<!DOCTYPE book PUBLIC "-//OASIS//DTD DocBook $TYPE_UC V$DOCBOOK_VERSION//EN" "http://www.oasis-open.org/docbook/$TYPE_LC/$DOCBOOK_VERSION/docbookx.dtd">
-<book id="test">
-</book>
-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)
-])
+++ /dev/null
-dnl as-gcc-inline-assembly.m4 0.1.0
-
-dnl autostars m4 macro for detection of gcc inline assembly
-
-dnl David Schleef <ds@schleef.org>
-
-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])
-])
-
+++ /dev/null
-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
-])
+++ /dev/null
-dnl as-libtool.m4 0.1.4
-
-dnl autostars m4 macro for libtool versioning
-
-dnl Thomas Vander Stichele <thomas at apestaart dot org>
-
-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])
-])
+++ /dev/null
-## ------------------------
-## 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
-# <wingo@pobox.com>
-# 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
-])
+++ /dev/null
-dnl as-version.m4 0.2.0
-
-dnl autostars m4 macro for versioning
-
-dnl Thomas Vander Stichele <thomas at apestaart dot org>
-
-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)
-])
+++ /dev/null
-##### 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 <stdint.h>" section requires the
-# existence of an include file <stdint.h> 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 <inttypes.h>. In other enviroments we can
-# use the inet-types in <sys/types.h> 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
-# <stdint.h>" or "#include <inttypes.h>", 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 <mylib-int.h>" to
-# obtain the stdint-types.
-#
-# Remember, if the system already had a valid <stdint.h>, 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 <guidod@gmx.de>
-#
-# 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 <stdint.h>],[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 <stdint.h>],[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 <stdint.h>" >>$ac_stdint
-echo "#endif" >>$ac_stdint
-echo "#endif" >>$ac_stdint
-else
-
-cat >>$ac_stdint <<STDINT_EOF
-
-/* ................... shortcircuit part ........................... */
-
-#if defined HAVE_STDINT_H || defined _STDINT_HAVE_STDINT_H
-#include <stdint.h>
-#else
-#include <stddef.h>
-
-/* .................... 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 <<STDINT_EOF
-/* .................... detections part ............................ */
-
-/* whether we need to define bitspecific types from compiler base types */
-#ifndef _STDINT_HEADER_INTPTR
-#ifndef _STDINT_HEADER_UINT32
-#ifndef _STDINT_HEADER_U_INT32
-#define _STDINT_NEED_INT_MODEL_T
-#else
-#define _STDINT_HAVE_U_INT_TYPES
-#endif
-#endif
-#endif
-
-#ifdef _STDINT_HAVE_U_INT_TYPES
-#undef _STDINT_NEED_INT_MODEL_T
-#endif
-
-#ifdef _STDINT_CHAR_MODEL
-#if _STDINT_CHAR_MODEL+0 == 122 || _STDINT_CHAR_MODEL+0 == 124
-#ifndef _STDINT_BYTE_MODEL
-#define _STDINT_BYTE_MODEL 12
-#endif
-#endif
-#endif
-
-#ifndef _STDINT_HAVE_INT_LEAST32_T
-#define _STDINT_NEED_INT_LEAST_T
-#endif
-
-#ifndef _STDINT_HAVE_INT_FAST32_T
-#define _STDINT_NEED_INT_FAST_T
-#endif
-
-#ifndef _STDINT_HEADER_INTPTR
-#define _STDINT_NEED_INTPTR_T
-#ifndef _STDINT_HAVE_INTMAX_T
-#define _STDINT_NEED_INTMAX_T
-#endif
-#endif
-
-
-/* .................... definition part ............................ */
-
-/* some system headers have good uint64_t */
-#ifndef _HAVE_UINT64_T
-#if defined _STDINT_HAVE_UINT64_T || defined HAVE_UINT64_T
-#define _HAVE_UINT64_T
-#elif defined _STDINT_HAVE_U_INT64_T || defined HAVE_U_INT64_T
-#define _HAVE_UINT64_T
-typedef u_int64_t uint64_t;
-#endif
-#endif
-
-#ifndef _HAVE_UINT64_T
-/* .. here are some common heuristics using compiler runtime specifics */
-#if defined __STDC_VERSION__ && defined __STDC_VERSION__ >= 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"
-])
-])
+++ /dev/null
-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 >= $min_check_version)
-
- rm -f conf.check-test
- dnl unset no_check, since in our second run it would have been set to yes
- dnl before
- no_check=
- AC_TRY_RUN([
-#include <stdio.h>
-#include <stdlib.h>
-
-#include <check.h>
-
-int main ()
-{
- int major, minor, micro;
- char *tmp_version;
-
- system ("touch conf.check-test");
-
- /* HP/UX 9 (%@#!) writes to sscanf strings */
- tmp_version = strdup("$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 $LIBS"
- AC_TRY_LINK([
-#include <stdio.h>
-#include <stdlib.h>
-
-#include <check.h>
-], , [ echo "*** The test program compiled, but did not run. This usually means"
- echo "*** that the run-time linker is not finding check. You'll need to set your"
- echo "*** LD_LIBRARY_PATH environment variable, or edit /etc/ld.so.conf to point"
- echo "*** to the installed location Also, make sure you have run ldconfig if that"
- echo "*** is required on your system"
- echo "***"
- echo "*** If you have an old version installed, it is best to remove it, although"
- echo "*** you may also be able to get things to work by modifying LD_LIBRARY_PATH"],
- [ echo "*** The test program failed to compile or link. See the file config.log for"
- echo "*** the exact error that occured." ])
-
- CFLAGS="$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
-])
+++ /dev/null
-# 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 <drepper@cygnus.com>, 1995, 1996
-#
-# Modified to never use included libintl.
-# Owen Taylor <otaylor@redhat.com>, 12/15/1998
-#
-# Major rework to remove unused code
-# Owen Taylor <otaylor@redhat.com>, 12/11/2002
-#
-# Added better handling of ALL_LINGUAS from GNU gettext version
-# written by Bruno Haible, Owen Taylor <otaylor.redhat.com> 5/30/3002
-#
-# Modified to require ngettext
-# Matthias Clasen <mclasen@redhat.com> 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 <locale.h>], [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 <locale.h> 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($1)dnl
-])
-
-# GLIB_WITH_NLS
-#-----------------
-glib_DEFUN([GLIB_WITH_NLS],
- dnl NLS is obligatory
- [USE_NLS=yes
- AC_SUBST(USE_NLS)
-
- gt_cv_have_gettext=no
-
- CATOBJEXT=NONE
- XGETTEXT=:
- INTLLIBS=
-
- AC_CHECK_HEADER(libintl.h,
- [gt_cv_func_dgettext_libintl="no"
- libintl_extra_libs=""
-
- #
- # First check in libc
- #
- AC_CACHE_CHECK([for ngettext in libc], gt_cv_func_ngettext_libc,
- [AC_TRY_LINK([
-#include <libintl.h>
-],
- [return !ngettext ("","", 1)],
- gt_cv_func_ngettext_libc=yes,
- gt_cv_func_ngettext_libc=no)
- ])
-
- if test "$gt_cv_func_ngettext_libc" = "yes" ; then
- AC_CACHE_CHECK([for dgettext in libc], gt_cv_func_dgettext_libc,
- [AC_TRY_LINK([
-#include <libintl.h>
-],
- [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 <foo@bar.xx>\n"
-"Language-Team: C <LL@li.org>\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])
-
+++ /dev/null
-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
-])
+++ /dev/null
-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=$withval],
- [WITH_PLUGINS=])
-
- GST_PLUGINS_ALL=""
- GST_PLUGINS_SELECTED=""
- GST_PLUGINS_NONPORTED=""
-
- AC_SUBST(GST_PLUGINS_ALL)
- AC_SUBST(GST_PLUGINS_SELECTED)
- AC_SUBST(GST_PLUGINS_NONPORTED)
-])
-
-dnl AG_GST_CHECK_PLUGIN(PLUGIN-NAME)
-dnl
-dnl This macro adds the plug-in <PLUGIN-NAME> to GST_PLUGINS_ALL. Then it
-dnl checks if WITH_PLUGINS is empty or the plugin is present in WITH_PLUGINS,
-dnl and if so adds it to GST_PLUGINS_SELECTED. Then it checks if the plugin
-dnl is present in WITHOUT_PLUGINS (ie. was disabled specifically) and if so
-dnl removes it from GST_PLUGINS_SELECTED.
-dnl
-dnl The macro will call AM_CONDITIONAL(USE_PLUGIN_<PLUGIN-NAME>, ...) to allow
-dnl control of what is built in Makefile.ams.
-AC_DEFUN([AG_GST_CHECK_PLUGIN],
-[
- GST_PLUGINS_ALL="$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 <PLUGIN-NAME> 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 $PACKAGE_VERSION_MICRO '<' 90`" = "x1"; then
- FATAL_WARNINGS=yes
- else
- FATAL_WARNINGS=no
- fi
- ])
-])
+++ /dev/null
-dnl pkg-config-based checks for GStreamer modules and dependency modules
-
-dnl generic:
-dnl AG_GST_PKG_CHECK_MODULES([PREFIX], [WHICH], [REQUIRED])
-dnl sets HAVE_[$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 $GST_PLUGINS_LIBAV_DIR])
- AC_SUBST(GST_PLUGINS_LIBAV_DIR)
- fi
-])
+++ /dev/null
-AC_DEFUN([AG_GST_DEBUGINFO], [
-AC_ARG_ENABLE(debug,
-AC_HELP_STRING([--disable-debug],[disable addition of -g debugging info]),
-[case "${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
-])
+++ /dev/null
-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])
-])
+++ /dev/null
-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$DOC_PS = xyes)
-])
+++ /dev/null
-dnl
-dnl Check for working do while(0) macros. This is used by G_STMT_START
-dnl and G_STMT_END in glib/gmacros.h. Without having this defined we
-dnl get "ambigious if-else" compiler warnings when compling C++ code.
-dnl
-dnl Copied from GLib's configure.in
-dnl
-AC_DEFUN([AG_GST_CHECK_DOWHILE_MACROS],[
-
-dnl *** check for working do while(0) macros ***
-AC_CACHE_CHECK([for working do while(0) macros], _cv_g_support_dowhile_macros, [
- AC_TRY_COMPILE([],[
- #define STMT_START do
- #define STMT_END while(0)
- #define STMT_TEST STMT_START { i = 0; } STMT_END
- int main(void) { int i = 1; STMT_TEST; return i; }],
- [_cv_g_support_dowhile_macros=yes],
- [_cv_g_support_dowhile_macros=no],
- [_cv_g_support_dowhile_macros=yes])
-])
-if test x$_cv_g_support_dowhile_macros = xyes; then
- AC_DEFINE(HAVE_DOWHILE_MACROS, 1, [define for working do while(0) macros])
-fi
-])
+++ /dev/null
-dnl handle various error-related things
-
-dnl Thomas Vander Stichele <thomas@apestaart.org>
-dnl Tim-Philipp Müller <tim centricular net>
-
-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)
-])
+++ /dev/null
-dnl Perform a check for a feature for GStreamer
-dnl Richard Boulton <richard-alsa@tartarus.org>
-dnl Thomas Vander Stichele <thomas@apestaart.org> 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_<FEATURE-NAME> to "yes" or "no" depending
-dnl on whether the feature is available.
-dnl
-dnl The macro will set USE_<FEATURE-NAME> to "yes" or "no" depending on
-dnl whether the feature is to be used.
-dnl Thomas changed this, so that when USE_<FEATURE-NAME> was already set
-dnl to no, then it stays that way.
-dnl
-dnl The macro will call AM_CONDITIONAL(USE_<FEATURE-NAME>, ...) 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_<FEATURE-NAME> 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 <richard-alsa@tartarus.org>
-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 $CFLAGS"
- AC_COMPILE_IFELSE([
- AC_LANG_SOURCE([[
- #include <gst/gstconfig.h>
- #ifdef GST_DISABLE_GST_DEBUG
- #error "debugging disabled, make compiler fail"
- #endif]])], [ debug_system_enabled=yes], [debug_system_enabled=no])
- CFLAGS="$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
-])
-
+++ /dev/null
-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 $CC implements __PRETTY_FUNCTION__)
- AC_CACHE_VAL(gst_cv_have_pretty_function,[
- AC_TRY_LINK([#include <stdio.h>],
- [printf("%s", __PRETTY_FUNCTION__);],
- gst_cv_have_pretty_function=yes,
- gst_cv_have_pretty_function=no)
- ])
- AC_MSG_RESULT($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 $CC implements __FUNCTION__)
- AC_CACHE_VAL(gst_cv_have_function,[
- AC_TRY_LINK([#include <stdio.h>],
- [printf("%s", __FUNCTION__);],
- gst_cv_have_function=yes,
- gst_cv_have_function=no)
- ])
- AC_MSG_RESULT($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 $CC implements __func__)
- AC_CACHE_VAL(gst_cv_have_func,[
- AC_TRY_LINK([#include <stdio.h>],
- [printf("%s", __func__);],
- gst_cv_have_func=yes,
- gst_cv_have_func=no)
- ])
- AC_MSG_RESULT($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, $function, [macro to use to show function name])
-])
+++ /dev/null
-dnl gettext setup
-
-dnl AG_GST_GETTEXT([gettext-package])
-dnl defines GETTEXT_PACKAGE and LOCALEDIR
-
-AC_DEFUN([AG_GST_GETTEXT],
-[
- if test "$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])
-])
+++ /dev/null
-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)
-])
+++ /dev/null
-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 $XML_LIBS"
- AC_TRY_LINK([
-#include <libxml/tree.h>
-#include <stdio.h>
-],[
-/* function body */
-],
- AC_MSG_NOTICE([Test xml2 program linked]),
- AC_MSG_ERROR([Could not link libxml2 test program. Check if you have the necessary dependencies.])
- )
- CFLAGS="$ac_save_CFLAGS"
- LIBS="$ac_save_LIBS"
-])
+++ /dev/null
-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 ], [ $3 ])
- else
- AG_GST_SET_PACKAGE_RELEASE_DATETIME([no])
- fi
-])
+++ /dev/null
-AC_DEFUN([AG_GST_BISON_CHECK],
-[
- dnl FIXME: check if AC_PROG_YACC is suitable here
- dnl FIXME: make precious
- AC_PATH_PROG(BISON_PATH, bison, no)
- if test x$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 >= $flex_min_version])
- if perl -w <<EOF
- (\$min_version_major, \$min_version_minor, \$min_version_micro ) = "$flex_min_version" =~ /(\d+)\.(\d+)\.(\d+)/;
- (\$flex_version_major, \$flex_version_minor, \$flex_version_micro ) = "$flex_version" =~ /(\d+)\.(\d+)\.(\d+)/;
- exit (((\$flex_version_major > \$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
-])
+++ /dev/null
-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
+++ /dev/null
-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)
-])
+++ /dev/null
-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)
-])
+++ /dev/null
-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$VALGRIND_PATH" = "xno")
-])
+++ /dev/null
-dnl macros for X-related detections
-dnl AC_SUBST's HAVE_X, X_CFLAGS, X_LIBS
-AC_DEFUN([AG_GST_CHECK_X],
-[
- AC_PATH_XTRA
- ac_cflags_save="$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
-])
+++ /dev/null
-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])
-])
+++ /dev/null
-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"])
-])
+++ /dev/null
-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])
-])
+++ /dev/null
-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 >= $ORC_REQ, which was\n"
- printf " not found. Slower code paths will be used.\n"
- fi
- fi
- printf "\n"
-])
-
+++ /dev/null
-# pkg.m4 - Macros to locate and utilise pkg-config. -*- Autoconf -*-
-#
-# Copyright © 2004 Scott James Remnant <scott@netsplit.com>.
-#
-# This program is free software; you can redistribute it and/or modify
-# it under the terms of the GNU General Public License as published by
-# the Free Software Foundation; either version 2 of the License, or
-# (at your option) any later version.
-#
-# This program is distributed in the hope that it will be useful, but
-# WITHOUT ANY WARRANTY; without even the implied warranty of
-# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
-# General Public License for more details.
-#
-# You should have received a copy of the GNU General Public License
-# along with this program; if not, write to the Free Software
-# Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
-#
-# As a special exception to the GNU General Public License, if you
-# distribute this file as part of a program that contains a
-# configuration script generated by Autoconf, you may include it under
-# the same distribution terms that you use for the rest of that program.
-
-# PKG_PROG_PKG_CONFIG([MIN-VERSION])
-# ----------------------------------
-AC_DEFUN([PKG_PROG_PKG_CONFIG],
-[m4_pattern_forbid([^_?PKG_[A-Z_]+$])
-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 <http://pkg-config.freedesktop.org/>.])],
- [$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
+++ /dev/null
-# -*- 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("<!-- ##### SECTION (\S+) ##### -->\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("<!-- ##### SECTION %s ##### -->\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 = '<include xmlns="http://www.w3.org/2003/XInclude" href="' + \
- 'element-' + element + '-details.xml">' + \
- '<fallback xmlns="http://www.w3.org/2003/XInclude" />' + \
- '</include>\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()
+++ /dev/null
-#
-# 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
-
+++ /dev/null
-# 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 $@
-
-all-recursive: independent-subdirs
+++ /dev/null
-<?xml version='1.0'?> <!--*- mode: xml -*-->
-
-<xsl:stylesheet
- xmlns:xsl="http://www.w3.org/1999/XSL/Transform"
- xmlns:exsl="http://exslt.org/common"
- xmlns:str="http://exslt.org/strings"
- extension-element-prefixes="exsl str"
- version="1.0">
-<xsl:output method="xml" indent="yes"
- doctype-public ="-//OASIS//DTD DocBook XML V4.1.2//EN"
- doctype-system = "http://www.oasis-open.org/docbook/xml/4.1.2/docbookx.dtd"/>
-
-<xsl:param name="module" />
-
- <xsl:template match="element">
- <xsl:element name="varlistentry">
- <xsl:element name="term">
- <xsl:element name="link">
- <xsl:attribute name="linkend"><xsl:value-of select="$module" />-plugins-<xsl:value-of select="name"/></xsl:attribute>
- <xsl:value-of select="name" />
- </xsl:element>
- </xsl:element>
- <xsl:element name="listitem">
- <xsl:element name="simpara"><xsl:value-of select="description" /></xsl:element>
- </xsl:element>
- </xsl:element>
- <xsl:variable name="name"><xsl:copy-of select="name"/></xsl:variable>
- <!-- here we write an element-(name)-details.xml file for the element -->
- <exsl:document href="{concat ('xml/element-', $name, '-details.xml')}" method="xml" indent="yes">
-
- <xsl:element name="refsynopsisdiv">
- <xsl:element name="refsect2">
- <xsl:element name="title">Element Information</xsl:element>
- <xsl:element name="variablelist">
-
- <!-- plugin name and link -->
- <xsl:element name="varlistentry">
- <xsl:element name="term">plugin</xsl:element>
- <xsl:element name="listitem">
- <xsl:element name="simpara">
- <xsl:element name="link">
- <xsl:attribute name="linkend">plugin-<xsl:value-of select="../../name"/></xsl:attribute>
- <xsl:value-of select="../../name" />
- </xsl:element>
- </xsl:element>
- </xsl:element>
- </xsl:element>
-
- <xsl:element name="varlistentry">
- <xsl:element name="term">author</xsl:element>
- <xsl:element name="listitem">
- <xsl:element name="simpara"><xsl:value-of select="author" /></xsl:element>
- </xsl:element>
- </xsl:element>
-
- <xsl:element name="varlistentry">
- <xsl:element name="term">class</xsl:element>
- <xsl:element name="listitem">
- <xsl:element name="simpara"><xsl:value-of select="class" /></xsl:element>
- </xsl:element>
- </xsl:element>
-
- </xsl:element> <!-- variablelist -->
- </xsl:element> <!-- refsect2 -->
-
- <xsl:element name="refsect2">
- <xsl:element name="title">Element Pads</xsl:element>
- <!-- process all caps -->
- <xsl:for-each select="pads/caps">
- <xsl:element name="variablelist">
- <xsl:element name="varlistentry">
- <xsl:element name="term">name</xsl:element>
- <xsl:element name="listitem">
- <xsl:element name="simpara"><xsl:value-of select="name" /></xsl:element>
- </xsl:element>
- </xsl:element>
-
- <xsl:element name="varlistentry">
- <xsl:element name="term">direction</xsl:element>
- <xsl:element name="listitem">
- <xsl:element name="simpara"><xsl:value-of select="direction" /></xsl:element>
- </xsl:element>
- </xsl:element>
-
- <xsl:element name="varlistentry">
- <xsl:element name="term">presence</xsl:element>
- <xsl:element name="listitem">
- <xsl:element name="simpara"><xsl:value-of select="presence" /></xsl:element>
- </xsl:element>
- </xsl:element>
-
- <xsl:for-each select='str:tokenize(details, ";")'>
- <xsl:element name="varlistentry">
- <xsl:element name="term">
- <xsl:if test="position()=1">details</xsl:if>
- </xsl:element>
- <xsl:element name="listitem">
- <xsl:element name="simpara"><xsl:value-of select='.'/></xsl:element>
- </xsl:element>
- </xsl:element>
- </xsl:for-each>
-
- </xsl:element> <!-- variablelist -->
-
- <!--xsl:element name="programlisting"><xsl:value-of select="details" /></xsl:element-->
-
- </xsl:for-each>
- </xsl:element> <!-- refsect2 -->
- </xsl:element> <!-- refsynopsisdiv -->
-
- </exsl:document>
- </xsl:template>
-
- <xsl:template match="plugin">
- <xsl:element name="refentry">
- <xsl:attribute name="id"><xsl:value-of select="$module" />-plugins-plugin-<xsl:value-of select="name"/></xsl:attribute>
-
- <xsl:element name="refmeta">
- <xsl:element name="refentrytitle">
- <xsl:value-of select="name"/>
- </xsl:element>
- <xsl:element name="manvolnum">3</xsl:element>
- <xsl:element name="refmiscinfo">FIXME Library</xsl:element>
- </xsl:element> <!-- refmeta -->
-
- <xsl:element name="refnamediv">
- <xsl:element name="refname">
- <xsl:value-of select="name"/>
- </xsl:element>
-
- <xsl:element name="refpurpose">
- <xsl:element name="anchor">
- <xsl:attribute name="id">plugin-<xsl:value-of select="name"/></xsl:attribute>
- </xsl:element>
- <xsl:value-of select="description"/>
- </xsl:element>
- </xsl:element>
-
- <xsl:element name="refsect1">
- <xsl:element name="title">Plugin Information</xsl:element>
- <xsl:element name="variablelist">
-
- <xsl:element name="varlistentry">
- <xsl:element name="term">filename</xsl:element>
- <xsl:element name="listitem">
- <xsl:element name="simpara"><xsl:value-of select="basename" /></xsl:element>
- </xsl:element>
- </xsl:element>
-
- <xsl:element name="varlistentry">
- <xsl:element name="term">version</xsl:element>
- <xsl:element name="listitem">
- <xsl:element name="simpara"><xsl:value-of select="version" /></xsl:element>
- </xsl:element>
- </xsl:element>
-
- <xsl:element name="varlistentry">
- <xsl:element name="term">run-time license</xsl:element>
- <xsl:element name="listitem">
- <xsl:element name="simpara"><xsl:value-of select="license" /></xsl:element>
- </xsl:element>
- </xsl:element>
-
- <xsl:element name="varlistentry">
- <xsl:element name="term">package</xsl:element>
- <xsl:element name="listitem">
- <xsl:element name="simpara"><xsl:value-of select="package" /></xsl:element>
- </xsl:element>
- </xsl:element>
-
- <xsl:element name="varlistentry">
- <xsl:element name="term">origin</xsl:element>
- <xsl:element name="listitem">
- <xsl:element name="simpara">
- <!-- only show origin as link if it starts with http -->
- <xsl:choose>
- <xsl:when test="substring(@href, 1, 4) = 'http'">
- <xsl:element name="ulink">
- <xsl:attribute name="url"><xsl:value-of select="origin" /></xsl:attribute>
- <xsl:value-of select="origin" />
- </xsl:element>
- </xsl:when>
- <xsl:otherwise>
- <xsl:value-of select="origin" />
- </xsl:otherwise>
- </xsl:choose>
- </xsl:element>
- </xsl:element>
- </xsl:element>
-
- </xsl:element>
- </xsl:element>
-
- <xsl:element name="refsect1">
- <xsl:element name="title">Elements</xsl:element>
- <!-- process all elements -->
- <xsl:element name="variablelist">
- <xsl:apply-templates select="elements"/>
- </xsl:element>
- </xsl:element>
-
- </xsl:element>
-
- </xsl:template>
-
- <!-- ignore -->
- <xsl:template match="gst-plugin-paths" />
-
-</xsl:stylesheet>
+++ /dev/null
-# rule to download the latest .po files
-download-po: $(top_srcdir)/common/download-translations
- $(top_srcdir)/common/download-translations $(PACKAGE)
-
+++ /dev/null
-# 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 )
+++ /dev/null
-#!/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 "<Object %s>" % 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
- '<SIGNAL>\n(.*?)</SIGNAL>\n'
- )
- nmatcher = re.compile(
- '<NAME>'
- '(?P<object>\S*)' # store object
- '::'
- '(?P<signal>\S*)' # store signal
- '</NAME>'
- )
- rmatcher = re.compile(
- '(?s)' # make . match \n
- '<RETURNS>(?P<returns>\S*)</RETURNS>\n' # store returns
- '(?P<args>.*)' # 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 = """<SIGNAL>
-<NAME>%(object)s::%(name)s</NAME>
-<RETURNS>%(returns)s</RETURNS>
-%(args)s</SIGNAL>
-"""
- 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
- '<ARG>\n(.*?)</ARG>\n'
- )
- nmatcher = re.compile(
- '<NAME>'
- '(?P<object>\S*)' # store object
- '::'
- '(?P<arg>\S*)' # store arg
- '</NAME>'
- )
- rmatcher = re.compile(
- '(?s)' # make . match \n
- '<TYPE>(?P<type>\S*)</TYPE>\n' # store type
- '<RANGE>(?P<range>.*?)</RANGE>\n' # store range
- '<FLAGS>(?P<flags>\S*)</FLAGS>\n' # store flags
- '<NICK>(?P<nick>.*?)</NICK>\n' # store nick
- '<BLURB>(?P<blurb>.*?)</BLURB>\n' # store blurb
- '<DEFAULT>(?P<default>.*?)</DEFAULT>\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 = """<ARG>
-<NAME>%(object)s::%(name)s</NAME>
-<TYPE>%(type)s</TYPE>
-<RANGE>%(range)s</RANGE>
-<FLAGS>%(flags)s</FLAGS>
-<NICK>%(nick)s</NICK>
-<BLURB>%(blurb)s</BLURB>
-<DEFAULT>%(default)s</DEFAULT>
-</ARG>
-"""
- 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)
+++ /dev/null
-# 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 $$DIR; \
- \
- if echo $(FORMATS) | grep html > /dev/null; then \
- echo "Un-preparing docs for upload (rebasing cross-references) ..." ; \
- gtkdoc-rebase --html-dir=$(builddir)/html ; \
- fi; \
- echo Done
+++ /dev/null
-# various tests to make sure we dist the win32 stuff (for MSVC builds) right
-
-# the MANIFEST contains all win32 related files that should be disted
-win32 = $(shell cat $(top_srcdir)/win32/MANIFEST)
-
-# wildcard is apparently not portable to other makes, hence the use of find
-# these are library .def files with the symbols to export
-win32defs = $(shell find $(top_srcdir)/win32/common -name '*.def')
-
-# wildcard is apparently not portable to other makes, hence the use of find
-# these are files that need to be disted with CRLF line endings:
-win32crlf = $(shell find $(top_srcdir)/win32 -name '*.dsw' -o -name '*.dsp')
-
-win32-debug:
- @echo; \
- echo win32 = $(win32); \
- echo; \
- echo win32defs = $(win32defs); \
- echo; \
- echo win32crlf = $(win32crlf); \
- echo
-
-win32-check-crlf:
- @echo Checking win32 files for CR LF line endings ...; \
- fail=0 ; \
- for each in $(win32crlf) ; do \
- result=`perl -e 'print grep(/\r\n/,<>)' "$$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
-
-
+++ /dev/null
-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.
-"
+++ /dev/null
-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
+++ /dev/null
-# 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.
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-@SET_MAKE@
-VPATH = @srcdir@
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-# Otherwise a system limit (for SysV at least) may be exceeded.
-.NOEXPORT:
+++ /dev/null
-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 <gst/gst.h>
-
- 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 <gst/rtsp-server/rtsp-server.h>
-
- 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 <gst/gst.h>
- #include <gst/rtsp-server/rtsp-server.h>
-
- 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.
-
+++ /dev/null
-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.
+++ /dev/null
-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:
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-# 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
+++ /dev/null
-# 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
-
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-# (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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-# The name of the module, e.g. 'glib'.
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-# for upload-doc.mak
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-# these variables define the location of the online docs
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-
-# generated basefiles
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-# ugly hack to make -unused.sgml work
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-# The directory containing the source code. Relative to $(top_srcdir).
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-# FIXME :
-# there's something wrong with gstreamer-sections.txt not being in the dist
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-
-# Extra options to supply to gtkdoc-fixref.
-FIXXREF_OPTIONS = --extra-dir=$(GLIB_PREFIX)/share/gtk-doc/html \
- --extra-dir=$(GST_PREFIX)/share/gtk-doc/html \
- --extra-dir=$(GSTPB_PREFIX)/share/gtk-doc/html
-
-
-# Used for dependencies.
-HFILE_GLOB = $(DOC_SOURCE_DIR)/*.h
-CFILE_GLOB = $(DOC_SOURCE_DIR)/*.c
-SCANOBJ_DEPS = \
- $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la
-
-
-# Extra options to supply to gtkdoc-scan.
-SCANOBJ_OPTIONS = --type-init-func="g_type_init();gst_init(&argc,&argv)"
-
-# Header files to ignore when scanning.
-IGNORE_HFILES =
-IGNORE_CFILES =
-
-# we add all .h files of elements that have signals/args we want
-# sadly this also pulls in the private methods - maybe we should
-# move those around in the source ?
-# also, we should add some stuff here conditionally based on whether
-# or not the plugin will actually build
-# but I'm not sure about that - it might be this Just Works given that
-# the registry won't have the element
-EXTRA_HFILES =
-
-# Images to copy into HTML directory.
-HTML_IMAGES =
-
-# Extra SGML files that are included by $(DOC_MAIN_SGML_FILE).
-content_files =
-
-# Other files to distribute.
-extra_files =
-
-# CFLAGS and LDFLAGS for compiling scan program. Only needed if your app/lib
-# contains GtkObjects/GObjects and you want to document signals and properties.
-GTKDOC_CFLAGS = -I$(top_srcdir) $(GST_PLUGINS_BASE_CFLAGS) \
- $(GST_BASE_CFLAGS) $(GST_CFLAGS)
-
-GTKDOC_LIBS = $(SCANOBJ_DEPS) $(GST_PLUGINS_BASE_LIBS) \
- $(GST_BASE_LIBS) $(GST_LIBS) $(GCOV_LIBS)
-
-GTKDOC_CC = $(LIBTOOL) --tag=CC --mode=compile $(CC)
-GTKDOC_LD = $(LIBTOOL) --tag=CC --mode=link $(CC)
-
-# If you need to override some of the declarations, place them in this file
-# and uncomment this line.
-DOC_OVERRIDES = $(DOC_MODULE)-overrides.txt
-
-# thomas: copied from glib-2
-# We set GPATH here; this gives us semantics for GNU make
-# which are more like other make's VPATH, when it comes to
-# whether a source that is a target of one rule is then
-# searched for in VPATH/GPATH.
-#
-GPATH = $(srcdir)
-
-# thomas: make docs parallel installable
-TARGET_DIR = $(HTML_DIR)/$(DOC_MODULE)-@GST_API_VERSION@
-EXTRA_DIST = \
- $(content_files) \
- $(extra_files) \
- $(HTML_IMAGES) \
- $(DOC_MAIN_SGML_FILE) \
- $(DOC_MODULE).types \
- $(DOC_OVERRIDES) \
- $(DOC_MODULE)-sections.txt
-
-DOC_STAMPS = \
- setup-build.stamp \
- scan-build.stamp \
- sgml-build.stamp \
- html-build.stamp \
- sgml.stamp \
- html.stamp
-
-SCANOBJ_FILES = \
- $(DOC_MODULE).args \
- $(DOC_MODULE).hierarchy \
- $(DOC_MODULE).interfaces \
- $(DOC_MODULE).prerequisites \
- $(DOC_MODULE).signals \
- .libs/$(DOC_MODULE)-scan.o
-
-REPORT_FILES = \
- $(DOC_MODULE)-undocumented.txt \
- $(DOC_MODULE)-undeclared.txt \
- $(DOC_MODULE)-unused.txt
-
-CLEANFILES = $(SCANOBJ_FILES) $(REPORT_FILES) $(DOC_STAMPS) doc-registry.xml
-all: all-am
-
-.SUFFIXES:
-$(srcdir)/Makefile.in: @MAINTAINER_MODE_TRUE@ $(srcdir)/Makefile.am $(top_srcdir)/common/upload-doc.mak $(top_srcdir)/common/gtk-doc.mak $(am__configure_deps)
- @for dep in $?; do \
- case '$(am__configure_deps)' in \
- *$$dep*) \
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- && { if test -f $@; then exit 0; else break; fi; }; \
- exit 1;; \
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- done; \
- echo ' cd $(top_srcdir) && $(AUTOMAKE) --gnu docs/libs/Makefile'; \
- $(am__cd) $(top_srcdir) && \
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- list='$(DISTFILES)'; \
- dist_files=`for file in $$list; do echo $$file; done | \
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-
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-
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-
-html: html-build.stamp
-
-upload: $(FORMATS)
- @if echo $(FORMATS) | grep html > /dev/null; then \
- echo "Preparing docs for upload (rebasing cross-references) ..." ; \
- if test x$(builddir) != x$(srcdir); then \
- echo "make upload can only be used if srcdir == builddir"; \
- exit 1; \
- fi; \
- # gtkdoc-rebase sometimes gets confused, so reset everything to \
- # local links before rebasing to online links \
- gtkdoc-rebase --html-dir=$(builddir)/html 2>/dev/null 2>/dev/null ; \
- rebase=`gtkdoc-rebase --verbose --online --html-dir=$(builddir)/html` ; \
- echo "$$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:
+++ /dev/null
-<?xml version="1.0"?>
-<!DOCTYPE book PUBLIC "-//OASIS//DTD DocBook XML V4.1.2//EN"
- "http://www.oasis-open.org/docbook/xml/4.1.2/docbookx.dtd" [
-<!ENTITY % version-entities SYSTEM "version.entities">
-%version-entities;
-]>
-
-<book id="index" xmlns:xi="http://www.w3.org/2003/XInclude">
- <bookinfo>
- <title>GStreamer RTSP Server Reference Manual</title>
- <releaseinfo>
- for GStreamer RTSP Server &GST_VERSION;
- </releaseinfo>
- </bookinfo>
-
- <chapter>
- <xi:include href="xml/rtsp-server.xml"/>
- <xi:include href="xml/rtsp-client.xml"/>
- <xi:include href="xml/rtsp-context.xml"/>
- <xi:include href="xml/rtsp-mount-points.xml"/>
- <xi:include href="xml/rtsp-media-factory.xml"/>
- <xi:include href="xml/rtsp-media-factory-uri.xml"/>
- <xi:include href="xml/rtsp-media.xml"/>
- <xi:include href="xml/rtsp-stream.xml"/>
- <xi:include href="xml/rtsp-session-pool.xml"/>
- <xi:include href="xml/rtsp-session.xml"/>
- <xi:include href="xml/rtsp-session-media.xml"/>
- <xi:include href="xml/rtsp-stream-transport.xml"/>
- <xi:include href="xml/rtsp-sdp.xml"/>
- <xi:include href="xml/rtsp-address-pool.xml"/>
- <xi:include href="xml/rtsp-thread-pool.xml"/>
- <xi:include href="xml/rtsp-auth.xml"/>
- <xi:include href="xml/rtsp-token.xml"/>
- <xi:include href="xml/rtsp-permissions.xml"/>
- <xi:include href="xml/rtsp-params.xml"/>
- </chapter>
-
- <chapter id="rtsp-server-hierarchy">
- <title>Object Hierarchy</title>
- <xi:include href="xml/tree_index.sgml"/>
- </chapter>
-
- <index id="api-index-full">
- <title>API Index</title>
- <xi:include href="xml/api-index-full.xml"><xi:fallback /></xi:include>
- </index>
-
- <xi:include href="xml/annotation-glossary.xml"><xi:fallback /></xi:include>
-</book>
+++ /dev/null
-<SECTION>
-<FILE>rtsp-address-pool</FILE>
-<TITLE>GstRTSPAddressPool</TITLE>
-
-<SUBSECTION Address>
-GST_RTSP_ADDRESS_POOL_ANY_IPV4
-GST_RTSP_ADDRESS_POOL_ANY_IPV6
-GstRTSPAddress
-GstRTSPAddressFlags
-gst_rtsp_address_copy
-gst_rtsp_address_free
-
-<SUBSECTION AddressPool>
-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
-<SUBSECTION Standard>
-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
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-auth</FILE>
-<TITLE>GstRTSPAuth</TITLE>
-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
-
-<SUBSECTION AuthChecks>
-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
-
-<SUBSECTION AuthTokens>
-GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE
-GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS
-
-<SUBSECTION AuthPermissions>
-GST_RTSP_PERM_MEDIA_FACTORY_ACCESS
-GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT
-<SUBSECTION Standard>
-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
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-client</FILE>
-<TITLE>GstRTSPClient</TITLE>
-<SUBSECTION Client>
-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
-<SUBSECTION Standard>
-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
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-context</FILE>
-<TITLE>GstRTSPContext</TITLE>
-GstRTSPContext
-gst_rtsp_context_get_current
-gst_rtsp_context_push_current
-gst_rtsp_context_pop_current
-<SUBSECTION Standard>
-GST_TYPE_RTSP_CONTEXT
-gst_rtsp_context_get_type
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-media</FILE>
-<TITLE>GstRTSPMedia</TITLE>
-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
-
-<SUBSECTION MediaPrepare>
-gst_rtsp_media_prepare
-gst_rtsp_media_unprepare
-GstRTSPMediaStatus
-gst_rtsp_media_get_status
-
-<SUBSECTION MediaSuspend>
-gst_rtsp_media_set_suspend_mode
-gst_rtsp_media_get_suspend_mode
-
-GstRTSPSuspendMode
-gst_rtsp_media_suspend
-gst_rtsp_media_unsuspend
-
-<SUBSECTION MediaStreams>
-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
-
-<SUBSECTION MediaState>
-gst_rtsp_media_seek
-gst_rtsp_media_get_range_string
-
-gst_rtsp_media_set_state
-gst_rtsp_media_set_pipeline_state
-
-<SUBSECTION MediaClocks>
-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
-<SUBSECTION Standard>
-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
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-media-factory</FILE>
-<TITLE>GstRTSPMediaFactory</TITLE>
-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
-
-<SUBSECTION Standard>
-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
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-media-factory-uri</FILE>
-<TITLE>GstRTSPMediaFactoryURI</TITLE>
-GstRTSPMediaFactoryURI
-GstRTSPMediaFactoryURIClass
-gst_rtsp_media_factory_uri_new
-gst_rtsp_media_factory_uri_set_uri
-gst_rtsp_media_factory_uri_get_uri
-<SUBSECTION Standard>
-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
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-mount-points</FILE>
-<TITLE>GstRTSPMountPoints</TITLE>
-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
-<SUBSECTION Standard>
-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
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-params</FILE>
-<TITLE>GstRTSPParams</TITLE>
-gst_rtsp_params_get
-gst_rtsp_params_set
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-permissions</FILE>
-<TITLE>GstRTSPPermissions</TITLE>
-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
-<SUBSECTION Standard>
-GST_RTSP_PERMISSIONS_CAST
-GST_IS_RTSP_PERMISSIONS
-GST_RTSP_PERMISSIONS
-GST_TYPE_RTSP_PERMISSIONS
-gst_rtsp_permissions_get_type
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-sdp</FILE>
-<TITLE>GstRTSPSdp</TITLE>
-GstSDPInfo
-gst_rtsp_sdp_from_media
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-server</FILE>
-<TITLE>GstRTSPServer</TITLE>
-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
-
-<SUBSECTION Standard>
-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
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-session</FILE>
-<TITLE>GstRTSPSession</TITLE>
-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
-<SUBSECTION Standard>
-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
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-session-media</FILE>
-<TITLE>GstRTSPSessionMedia</TITLE>
-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
-<SUBSECTION Standard>
-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
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-session-pool</FILE>
-<TITLE>GstRTSPSessionPool</TITLE>
-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
-<SUBSECTION Standard>
-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
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-stream</FILE>
-<TITLE>GstRTSPStream</TITLE>
-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
-
-<SUBSECTION Standard>
-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
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-stream-transport</FILE>
-<TITLE>GstRTSPStreamTransport</TITLE>
-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
-
-<SUBSECTION Standard>
-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
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-thread-pool</FILE>
-<TITLE>GstRTSPThreadPool</TITLE>
-<SUBSECTION Thread>
-GstRTSPThreadType
-GstRTSPThread
-
-gst_rtsp_thread_new
-gst_rtsp_thread_ref
-gst_rtsp_thread_unref
-gst_rtsp_thread_reuse
-gst_rtsp_thread_stop
-
-<SUBSECTION ThreadPool>
-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
-<SUBSECTION Standard>
-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
-</SECTION>
-
-<SECTION>
-<FILE>rtsp-token</FILE>
-<TITLE>GstRTSPToken</TITLE>
-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
-<SUBSECTION Standard>
-GST_RTSP_TOKEN_CAST
-GST_IS_RTSP_TOKEN
-GST_RTSP_TOKEN
-GST_TYPE_RTSP_TOKEN
-gst_rtsp_token_get_type
-</SECTION>
-
+++ /dev/null
-#include <gst/gst.h>
-
-#include <gst/rtsp-server/rtsp-auth.h>
-gst_rtsp_auth_get_type
-
-#include <gst/rtsp-server/rtsp-mount-points.h>
-gst_rtsp_mount_points_get_type
-
-#include <gst/rtsp-server/rtsp-media-factory.h>
-gst_rtsp_media_factory_get_type
-
-#include <gst/rtsp-server/rtsp-media.h>
-gst_rtsp_media_get_type
-
-#include <gst/rtsp-server/rtsp-server.h>
-gst_rtsp_server_get_type
-
-#include <gst/rtsp-server/rtsp-session-pool.h>
-gst_rtsp_session_pool_get_type
-
-#include <gst/rtsp-server/rtsp-session.h>
-gst_rtsp_session_get_type
-
-#include <gst/rtsp-server/rtsp-client.h>
-gst_rtsp_client_get_type
+++ /dev/null
-<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
-<html>
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-<title>GStreamer RTSP Server Reference Manual: GstRTSPAuth</title>
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-<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
- <a href="#GstRTSPAuth.description" class="shortcut">Description</a></span><span id="nav_hierarchy"> <span class="dim">|</span>
- <a href="#GstRTSPAuth.object-hierarchy" class="shortcut">Object Hierarchy</a></span>
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-<div class="refentry">
-<a name="GstRTSPAuth"></a><div class="titlepage"></div>
-<div class="refnamediv"><table width="100%"><tr>
-<td valign="top">
-<h2><span class="refentrytitle"><a name="GstRTSPAuth.top_of_page"></a>GstRTSPAuth</span></h2>
-<p>GstRTSPAuth — Authentication and authorization</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="GstRTSPAuth.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="returnvalue">GstRTSPAuth</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-new" title="gst_rtsp_auth_new ()">gst_rtsp_auth_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/gio/unstable/GTlsCertificate.html"><span class="returnvalue">GTlsCertificate</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-get-tls-certificate" title="gst_rtsp_auth_get_tls_certificate ()">gst_rtsp_auth_get_tls_certificate</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-set-tls-certificate" title="gst_rtsp_auth_set_tls_certificate ()">gst_rtsp_auth_set_tls_certificate</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-make-basic" title="gst_rtsp_auth_make_basic ()">gst_rtsp_auth_make_basic</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-add-basic" title="gst_rtsp_auth_add_basic ()">gst_rtsp_auth_add_basic</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-remove-basic" title="gst_rtsp_auth_remove_basic ()">gst_rtsp_auth_remove_basic</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-check" title="gst_rtsp_auth_check ()">gst_rtsp_auth_check</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-get-default-token" title="gst_rtsp_auth_get_default_token ()">gst_rtsp_auth_get_default_token</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-set-default-token" title="gst_rtsp_auth_set_default_token ()">gst_rtsp_auth_set_default_token</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPAuth.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPAuth.html#GstRTSPAuth-struct" title="struct GstRTSPAuth">GstRTSPAuth</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPAuth.html#GstRTSPAuthClass" title="struct GstRTSPAuthClass">GstRTSPAuthClass</a></td>
-</tr>
-<tr>
-<td class="define_keyword">#define</td>
-<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-CONNECT:CAPS" title="GST_RTSP_AUTH_CHECK_CONNECT">GST_RTSP_AUTH_CHECK_CONNECT</a></td>
-</tr>
-<tr>
-<td class="define_keyword">#define</td>
-<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-URL:CAPS" title="GST_RTSP_AUTH_CHECK_URL">GST_RTSP_AUTH_CHECK_URL</a></td>
-</tr>
-<tr>
-<td class="define_keyword">#define</td>
-<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-MEDIA-FACTORY-ACCESS:CAPS" title="GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS">GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS</a></td>
-</tr>
-<tr>
-<td class="define_keyword">#define</td>
-<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-MEDIA-FACTORY-CONSTRUCT:CAPS" title="GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT">GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT</a></td>
-</tr>
-<tr>
-<td class="define_keyword">#define</td>
-<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-TRANSPORT-CLIENT-SETTINGS:CAPS" title="GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS">GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS</a></td>
-</tr>
-<tr>
-<td class="define_keyword">#define</td>
-<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-TOKEN-MEDIA-FACTORY-ROLE:CAPS" title="GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE">GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE</a></td>
-</tr>
-<tr>
-<td class="define_keyword">#define</td>
-<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-TOKEN-TRANSPORT-CLIENT-SETTINGS:CAPS" title="GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS">GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS</a></td>
-</tr>
-<tr>
-<td class="define_keyword">#define</td>
-<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-PERM-MEDIA-FACTORY-ACCESS:CAPS" title="GST_RTSP_PERM_MEDIA_FACTORY_ACCESS">GST_RTSP_PERM_MEDIA_FACTORY_ACCESS</a></td>
-</tr>
-<tr>
-<td class="define_keyword">#define</td>
-<td class="function_name"><a class="link" href="GstRTSPAuth.html#GST-RTSP-PERM-MEDIA-FACTORY-CONSTRUCT:CAPS" title="GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT">GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT</a></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPAuth.object-hierarchy"></a><h2>Object Hierarchy</h2>
-<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
- <span class="lineart">╰──</span> GstRTSPAuth
-</pre>
-</div>
-<div class="refsect1">
-<a name="GstRTSPAuth.description"></a><h2>Description</h2>
-<p>The <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> 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.</p>
-<p>A new auth object is made with <a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-new" title="gst_rtsp_auth_new ()"><code class="function">gst_rtsp_auth_new()</code></a>. It is usually configured
-on the <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> object.</p>
-<p>The RTSP server will call <a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-check" title="gst_rtsp_auth_check ()"><code class="function">gst_rtsp_auth_check()</code></a> 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 <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> and
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> on the object to check if an operation is allowed.</p>
-<p>The default <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> object has support for basic authentication. With
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-add-basic" title="gst_rtsp_auth_add_basic ()"><code class="function">gst_rtsp_auth_add_basic()</code></a> you can add a basic authentication string together
-with the <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> that will become active when successfully
-authenticated.</p>
-<p>When a TLS certificate has been set with <a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-set-tls-certificate" title="gst_rtsp_auth_set_tls_certificate ()"><code class="function">gst_rtsp_auth_set_tls_certificate()</code></a>,
-the default auth object will require the client to connect with a TLS
-connection.</p>
-<p>Last reviewed on 2013-07-16 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="GstRTSPAuth.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-auth-new"></a><h3>gst_rtsp_auth_new ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="returnvalue">GstRTSPAuth</span></a> *
-gst_rtsp_auth_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
-<p>Create a new <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> instance.</p>
-<div class="refsect3">
-<a name="id-1.2.16.7.2.5"></a><h4>Returns</h4>
-<p> a new <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a>. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-auth-get-tls-certificate"></a><h3>gst_rtsp_auth_get_tls_certificate ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/gio/unstable/GTlsCertificate.html"><span class="returnvalue">GTlsCertificate</span></a> *
-gst_rtsp_auth_get_tls_certificate (<em class="parameter"><code><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *auth</code></em>);</pre>
-<p>Get the <a href="https://developer.gnome.org/gio/unstable/GTlsCertificate.html"><span class="type">GTlsCertificate</span></a> used for negotiating TLS <em class="parameter"><code>auth</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.16.7.3.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>auth</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.16.7.3.6"></a><h4>Returns</h4>
-<p> the <a href="https://developer.gnome.org/gio/unstable/GTlsCertificate.html"><span class="type">GTlsCertificate</span></a> of <em class="parameter"><code>auth</code></em>
-. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
-usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-auth-set-tls-certificate"></a><h3>gst_rtsp_auth_set_tls_certificate ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_auth_set_tls_certificate (<em class="parameter"><code><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *auth</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GTlsCertificate.html"><span class="type">GTlsCertificate</span></a> *cert</code></em>);</pre>
-<p>Set the TLS certificate for the auth. Client connections will only
-be accepted when TLS is negotiated.</p>
-<div class="refsect3">
-<a name="id-1.2.16.7.4.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>auth</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>cert</p></td>
-<td class="parameter_description"><p> a <a href="https://developer.gnome.org/gio/unstable/GTlsCertificate.html"><span class="type">GTlsCertificate</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-auth-make-basic"></a><h3>gst_rtsp_auth_make_basic ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-gst_rtsp_auth_make_basic (<em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *user</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *pass</code></em>);</pre>
-<p>Construct a Basic authorisation token from <em class="parameter"><code>user</code></em>
- and <em class="parameter"><code>pass</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.16.7.5.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>user</p></td>
-<td class="parameter_description"><p>a userid</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>pass</p></td>
-<td class="parameter_description"><p>a password</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.16.7.5.6"></a><h4>Returns</h4>
-<p> the base64 encoding of the string <em class="parameter"><code>user</code></em>
-:<em class="parameter"><code>pass</code></em>
-.
-<a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-auth-add-basic"></a><h3>gst_rtsp_auth_add_basic ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_auth_add_basic (<em class="parameter"><code><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *auth</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *basic</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *token</code></em>);</pre>
-<p>Add a basic token for the default authentication algorithm that
-enables the client with privileges listed in <em class="parameter"><code>token</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.16.7.6.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>auth</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>basic</p></td>
-<td class="parameter_description"><p>the basic token</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>token</p></td>
-<td class="parameter_description"><p> authorisation token. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-auth-remove-basic"></a><h3>gst_rtsp_auth_remove_basic ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_auth_remove_basic (<em class="parameter"><code><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *auth</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *basic</code></em>);</pre>
-<p>Add a basic token for the default authentication algorithm that
-enables the client with privileges from <em class="parameter"><code>authgroup</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.16.7.7.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>auth</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>basic</p></td>
-<td class="parameter_description"><p> the basic token. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-auth-check"></a><h3>gst_rtsp_auth_check ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_auth_check (<em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *check</code></em>);</pre>
-<p>Check if <em class="parameter"><code>check</code></em>
- is allowed in the current context.</p>
-<div class="refsect3">
-<a name="id-1.2.16.7.8.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>check</p></td>
-<td class="parameter_description"><p>the item to check</p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.16.7.8.6"></a><h4>Returns</h4>
-<p> FALSE if check failed.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-auth-get-default-token"></a><h3>gst_rtsp_auth_get_default_token ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
-gst_rtsp_auth_get_default_token (<em class="parameter"><code><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *auth</code></em>);</pre>
-<p>Get the default token for <em class="parameter"><code>auth</code></em>
-. This token will be used for unauthenticated
-users.</p>
-<div class="refsect3">
-<a name="id-1.2.16.7.9.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>auth</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.16.7.9.6"></a><h4>Returns</h4>
-<p> the <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> of <em class="parameter"><code>auth</code></em>
-. <a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-unref" title="gst_rtsp_token_unref ()"><code class="function">gst_rtsp_token_unref()</code></a> after
-usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-auth-set-default-token"></a><h3>gst_rtsp_auth_set_default_token ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_auth_set_default_token (<em class="parameter"><code><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *auth</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *token</code></em>);</pre>
-<p>Set the default <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> to <em class="parameter"><code>token</code></em>
- in <em class="parameter"><code>auth</code></em>
-. The default token will
-be used for unauthenticated users.</p>
-<div class="refsect3">
-<a name="id-1.2.16.7.10.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>auth</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>token</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPAuth.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstRTSPAuth-struct"></a><h3>struct GstRTSPAuth</h3>
-<pre class="programlisting">struct GstRTSPAuth;</pre>
-<p>The authentication structure.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPAuthClass"></a><h3>struct GstRTSPAuthClass</h3>
-<pre class="programlisting">struct GstRTSPAuthClass {
- GObjectClass parent_class;
-
- gboolean (*authenticate) (GstRTSPAuth *auth, GstRTSPContext *ctx);
- gboolean (*check) (GstRTSPAuth *auth, GstRTSPContext *ctx,
- const gchar *check);
-};
-</pre>
-<p>The authentication class.</p>
-<div class="refsect3">
-<a name="id-1.2.16.8.3.5"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="struct_members_name">
-<col class="struct_members_description">
-<col width="200px" class="struct_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObjectClass"><span class="type">GObjectClass</span></a> <em class="structfield"><code><a name="GstRTSPAuthClass.parent-class"></a>parent_class</code></em>;</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPAuthClass.authenticate"></a>authenticate</code></em> ()</p></td>
-<td class="struct_member_description"><p>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.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPAuthClass.check"></a>check</code></em> ()</p></td>
-<td class="struct_member_description"><p>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.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GST-RTSP-AUTH-CHECK-CONNECT:CAPS"></a><h3>GST_RTSP_AUTH_CHECK_CONNECT</h3>
-<pre class="programlisting">#define GST_RTSP_AUTH_CHECK_CONNECT "auth.check.connect"
-</pre>
-<p>Check a new connection</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GST-RTSP-AUTH-CHECK-URL:CAPS"></a><h3>GST_RTSP_AUTH_CHECK_URL</h3>
-<pre class="programlisting">#define GST_RTSP_AUTH_CHECK_URL "auth.check.url"
-</pre>
-<p>Check the URL and methods</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GST-RTSP-AUTH-CHECK-MEDIA-FACTORY-ACCESS:CAPS"></a><h3>GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS</h3>
-<pre class="programlisting">#define GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS "auth.check.media.factory.access"
-</pre>
-<p>Check if access is allowed to a factory.
-When access is not allowed an 404 Not Found is sent in the response.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GST-RTSP-AUTH-CHECK-MEDIA-FACTORY-CONSTRUCT:CAPS"></a><h3>GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT</h3>
-<pre class="programlisting">#define GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT "auth.check.media.factory.construct"
-</pre>
-<p>Check if media can be constructed from a media factory
-A response should be sent on error.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GST-RTSP-AUTH-CHECK-TRANSPORT-CLIENT-SETTINGS:CAPS"></a><h3>GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS</h3>
-<pre class="programlisting">#define GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS "auth.check.transport.client-settings"
-</pre>
-<p>Check if the client can specify TTL, destination and
-port pair in multicast. No response is sent when the check returns
-<a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#FALSE:CAPS"><code class="literal">FALSE</code></a>.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GST-RTSP-TOKEN-MEDIA-FACTORY-ROLE:CAPS"></a><h3>GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE</h3>
-<pre class="programlisting">#define GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE "media.factory.role"
-</pre>
-<p>G_TYPE_STRING, the role to use when dealing with media factories</p>
-<p>The default <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> object uses this string in the token to find the
-role of the media factory. It will then retrieve the <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> of
-the media factory and retrieve the role with the same name.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GST-RTSP-TOKEN-TRANSPORT-CLIENT-SETTINGS:CAPS"></a><h3>GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS</h3>
-<pre class="programlisting">#define GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS "transport.client-settings"
-</pre>
-<p>G_TYPE_BOOLEAN, <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the client can specify TTL, destination and
- port pair in multicast.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GST-RTSP-PERM-MEDIA-FACTORY-ACCESS:CAPS"></a><h3>GST_RTSP_PERM_MEDIA_FACTORY_ACCESS</h3>
-<pre class="programlisting">#define GST_RTSP_PERM_MEDIA_FACTORY_ACCESS "media.factory.access"
-</pre>
-<p>G_TYPE_BOOLEAN, <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the media can be accessed, <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#FALSE:CAPS"><code class="literal">FALSE</code></a> will
-return a 404 Not Found error when trying to access the media.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GST-RTSP-PERM-MEDIA-FACTORY-CONSTRUCT:CAPS"></a><h3>GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT</h3>
-<pre class="programlisting">#define GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT "media.factory.construct"
-</pre>
-<p>G_TYPE_BOOLEAN, <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the media can be constructed, <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#FALSE:CAPS"><code class="literal">FALSE</code></a> will
-return a 404 Not Found error when trying to access the media.</p>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPAuth.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a></p>
-</div>
-</div>
-<div class="footer">
-<hr>
- Generated by GTK-Doc V1.21</div>
-</body>
-</html>
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-<div class="refentry">
-<a name="GstRTSPClient"></a><div class="titlepage"></div>
-<div class="refnamediv"><table width="100%"><tr>
-<td valign="top">
-<h2><span class="refentrytitle"><a name="GstRTSPClient.top_of_page"></a>GstRTSPClient</span></h2>
-<p>GstRTSPClient — A client connection state</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="GstRTSPClient.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="returnvalue">GstRTSPClient</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-new" title="gst_rtsp_client_new ()">gst_rtsp_client_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-close" title="gst_rtsp_client_close ()">gst_rtsp_client_close</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="returnvalue">GstRTSPSessionPool</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-session-pool" title="gst_rtsp_client_get_session_pool ()">gst_rtsp_client_get_session_pool</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-session-pool" title="gst_rtsp_client_set_session_pool ()">gst_rtsp_client_set_session_pool</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="returnvalue">GstRTSPMountPoints</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-mount-points" title="gst_rtsp_client_get_mount_points ()">gst_rtsp_client_get_mount_points</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-mount-points" title="gst_rtsp_client_set_mount_points ()">gst_rtsp_client_set_mount_points</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="returnvalue">GstRTSPAuth</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-auth" title="gst_rtsp_client_get_auth ()">gst_rtsp_client_get_auth</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-auth" title="gst_rtsp_client_set_auth ()">gst_rtsp_client_set_auth</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="returnvalue">GstRTSPThreadPool</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-thread-pool" title="gst_rtsp_client_get_thread_pool ()">gst_rtsp_client_get_thread_pool</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-thread-pool" title="gst_rtsp_client_set_thread_pool ()">gst_rtsp_client_set_thread_pool</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="returnvalue">GstRTSPConnection</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-connection" title="gst_rtsp_client_get_connection ()">gst_rtsp_client_get_connection</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-connection" title="gst_rtsp_client_set_connection ()">gst_rtsp_client_set_connection</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-attach" title="gst_rtsp_client_attach ()">gst_rtsp_client_attach</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<span class="c_punctuation">(</span><a class="link" href="GstRTSPClient.html#GstRTSPClientSendFunc" title="GstRTSPClientSendFunc ()">*GstRTSPClientSendFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-send-func" title="gst_rtsp_client_set_send_func ()">gst_rtsp_client_set_send_func</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="returnvalue">GstRTSPResult</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-handle-message" title="gst_rtsp_client_handle_message ()">gst_rtsp_client_handle_message</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="returnvalue">GstRTSPResult</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-send-message" title="gst_rtsp_client_send_message ()">gst_rtsp_client_send_message</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
-</td>
-<td class="function_name">
-<span class="c_punctuation">(</span><a class="link" href="GstRTSPClient.html#GstRTSPClientSessionFilterFunc" title="GstRTSPClientSessionFilterFunc ()">*GstRTSPClientSessionFilterFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-session-filter" title="gst_rtsp_client_session_filter ()">gst_rtsp_client_session_filter</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPClient.properties"></a><h2>Properties</h2>
-<div class="informaltable"><table border="0">
-<colgroup>
-<col width="150px" class="properties_type">
-<col width="300px" class="properties_name">
-<col width="200px" class="properties_flags">
-</colgroup>
-<tbody>
-<tr>
-<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient--drop-backlog" title="The “drop-backlog” property">drop-backlog</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type">
-<a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *</td>
-<td class="property_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient--mount-points" title="The “mount-points” property">mount-points</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type">
-<a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *</td>
-<td class="property_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient--session-pool" title="The “session-pool” property">session-pool</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPClient.signals"></a><h2>Signals</h2>
-<div class="informaltable"><table border="0">
-<colgroup>
-<col width="150px" class="signals_return">
-<col width="300px" class="signals_name">
-<col width="200px" class="signals_flags">
-</colgroup>
-<tbody>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-closed" title="The “closed” signal">closed</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-describe-request" title="The “describe-request” signal">describe-request</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-get-parameter-request" title="The “get-parameter-request” signal">get-parameter-request</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-handle-response" title="The “handle-response” signal">handle-response</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-new-session" title="The “new-session” signal">new-session</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-options-request" title="The “options-request” signal">options-request</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-pause-request" title="The “pause-request” signal">pause-request</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-play-request" title="The “play-request” signal">play-request</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-send-message" title="The “send-message” signal">send-message</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-set-parameter-request" title="The “set-parameter-request” signal">set-parameter-request</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-setup-request" title="The “setup-request” signal">setup-request</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-teardown-request" title="The “teardown-request” signal">teardown-request</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPClient.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPClient.html#GstRTSPClient-struct" title="struct GstRTSPClient">GstRTSPClient</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPClient.html#GstRTSPClientClass" title="struct GstRTSPClientClass">GstRTSPClientClass</a></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPClient.object-hierarchy"></a><h2>Object Hierarchy</h2>
-<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
- <span class="lineart">╰──</span> GstRTSPClient
-</pre>
-</div>
-<div class="refsect1">
-<a name="GstRTSPClient.description"></a><h2>Description</h2>
-<p>The client object handles the connection with a client for as long as a TCP
-connection is open.</p>
-<p>A <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> is created by <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> when a new connection is
-accepted and it inherits the <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a>, <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a>,
-<a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> and <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> from the server.</p>
-<p>The client connection should be configured with the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="type">GstRTSPConnection</span></a> using
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-connection" title="gst_rtsp_client_set_connection ()"><code class="function">gst_rtsp_client_set_connection()</code></a> before it can be attached to a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainContext"><span class="type">GMainContext</span></a>
-using <a class="link" href="GstRTSPClient.html#gst-rtsp-client-attach" title="gst_rtsp_client_attach ()"><code class="function">gst_rtsp_client_attach()</code></a>. From then on the client will handle requests
-on the connection.</p>
-<p>Use <a class="link" href="GstRTSPClient.html#gst-rtsp-client-session-filter" title="gst_rtsp_client_session_filter ()"><code class="function">gst_rtsp_client_session_filter()</code></a> to iterate or modify all the
-<a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> objects managed by the client object.</p>
-<p>Last reviewed on 2013-07-11 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="GstRTSPClient.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-client-new"></a><h3>gst_rtsp_client_new ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="returnvalue">GstRTSPClient</span></a> *
-gst_rtsp_client_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
-<p>Create a new <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> instance.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.2.5"></a><h4>Returns</h4>
-<p> a new <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a>. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-client-close"></a><h3>gst_rtsp_client_close ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_client_close (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>);</pre>
-<p>Close the connection of <em class="parameter"><code>client</code></em>
- and remove all media it was managing.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.3.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<p class="since">Since 1.4</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-client-get-session-pool"></a><h3>gst_rtsp_client_get_session_pool ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="returnvalue">GstRTSPSessionPool</span></a> *
-gst_rtsp_client_get_session_pool (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>);</pre>
-<p>Get the <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> object that <em class="parameter"><code>client</code></em>
- uses to manage its sessions.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.4.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.2.9.4.6"></a><h4>Returns</h4>
-<p> a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a>, unref after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-client-set-session-pool"></a><h3>gst_rtsp_client_set_session_pool ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_client_set_session_pool (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>);</pre>
-<p>Set <em class="parameter"><code>pool</code></em>
- as the sessionpool for <em class="parameter"><code>client</code></em>
- 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.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.5.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p> a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-client-get-mount-points"></a><h3>gst_rtsp_client_get_mount_points ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="returnvalue">GstRTSPMountPoints</span></a> *
-gst_rtsp_client_get_mount_points (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>);</pre>
-<p>Get the <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> object that <em class="parameter"><code>client</code></em>
- uses to manage its sessions.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.6.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.2.9.6.6"></a><h4>Returns</h4>
-<p> a <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a>, unref after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-client-set-mount-points"></a><h3>gst_rtsp_client_set_mount_points ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_client_set_mount_points (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *mounts</code></em>);</pre>
-<p>Set <em class="parameter"><code>mounts</code></em>
- as the mount points for <em class="parameter"><code>client</code></em>
- 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.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.7.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>mounts</p></td>
-<td class="parameter_description"><p> a <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-client-get-auth"></a><h3>gst_rtsp_client_get_auth ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="returnvalue">GstRTSPAuth</span></a> *
-gst_rtsp_client_get_auth (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>);</pre>
-<p>Get the <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> used as the authentication manager of <em class="parameter"><code>client</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.8.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.2.9.8.6"></a><h4>Returns</h4>
-<p> the <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> of <em class="parameter"><code>client</code></em>
-. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
-usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-client-set-auth"></a><h3>gst_rtsp_client_set_auth ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_client_set_auth (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *auth</code></em>);</pre>
-<p>configure <em class="parameter"><code>auth</code></em>
- to be used as the authentication manager of <em class="parameter"><code>client</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.9.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>auth</p></td>
-<td class="parameter_description"><p> a <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-client-get-thread-pool"></a><h3>gst_rtsp_client_get_thread_pool ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="returnvalue">GstRTSPThreadPool</span></a> *
-gst_rtsp_client_get_thread_pool (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>);</pre>
-<p>Get the <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> used as the thread pool of <em class="parameter"><code>client</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.10.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.2.9.10.6"></a><h4>Returns</h4>
-<p> the <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> of <em class="parameter"><code>client</code></em>
-. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
-usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-client-set-thread-pool"></a><h3>gst_rtsp_client_set_thread_pool ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_client_set_thread_pool (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> *pool</code></em>);</pre>
-<p>configure <em class="parameter"><code>pool</code></em>
- to be used as the thread pool of <em class="parameter"><code>client</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.11.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-client-get-connection"></a><h3>gst_rtsp_client_get_connection ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="returnvalue">GstRTSPConnection</span></a> *
-gst_rtsp_client_get_connection (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>);</pre>
-<p>Get the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="type">GstRTSPConnection</span></a> of <em class="parameter"><code>client</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.12.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.2.9.12.6"></a><h4>Returns</h4>
-<p> the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="type">GstRTSPConnection</span></a> of <em class="parameter"><code>client</code></em>
-.
-The connection object returned remains valid until the client is freed. </p>
-<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-client-set-connection"></a><h3>gst_rtsp_client_set_connection ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_client_set_connection (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="type">GstRTSPConnection</span></a> *conn</code></em>);</pre>
-<p>Set the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="type">GstRTSPConnection</span></a> of <em class="parameter"><code>client</code></em>
-. This function takes ownership of
-<em class="parameter"><code>conn</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.13.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>conn</p></td>
-<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="type">GstRTSPConnection</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.2.9.13.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-client-attach"></a><h3>gst_rtsp_client_attach ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-gst_rtsp_client_attach (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainContext"><span class="type">GMainContext</span></a> *context</code></em>);</pre>
-<p>Attaches <em class="parameter"><code>client</code></em>
- to <em class="parameter"><code>context</code></em>
-. When the mainloop for <em class="parameter"><code>context</code></em>
- is run, the
-client will be dispatched. When <em class="parameter"><code>context</code></em>
- is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, the default context will be
-used).</p>
-<p>This function should be called when the client properties and urls are fully
-configured and the client is ready to start.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.14.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>context</p></td>
-<td class="parameter_description"><p> a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainContext"><span class="type">GMainContext</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.2.9.14.7"></a><h4>Returns</h4>
-<p> the ID (greater than 0) for the source within the GMainContext.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPClientSendFunc"></a><h3>GstRTSPClientSendFunc ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-<span class="c_punctuation">(</span>*GstRTSPClientSendFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#GstRTSPMessage"><span class="type">GstRTSPMessage</span></a> *message</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> close</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
-<p>This callback is called when <em class="parameter"><code>client</code></em>
- wants to send <em class="parameter"><code>message</code></em>
-. When <em class="parameter"><code>close</code></em>
- is
-<a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a>, the connection should be closed when the message has been sent.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.15.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>message</p></td>
-<td class="parameter_description"><p>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#GstRTSPMessage"><span class="type">GstRTSPMessage</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>close</p></td>
-<td class="parameter_description"><p>close the connection</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p>user data when registering the callback</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.2.9.15.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-client-set-send-func"></a><h3>gst_rtsp_client_set_send_func ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_client_set_send_func (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPClient.html#GstRTSPClientSendFunc" title="GstRTSPClientSendFunc ()"><span class="type">GstRTSPClientSendFunc</span></a> func</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Datasets.html#GDestroyNotify"><span class="type">GDestroyNotify</span></a> notify</code></em>);</pre>
-<p>Set <em class="parameter"><code>func</code></em>
- as the callback that will be called when a new message needs to be
-sent to the client. <em class="parameter"><code>user_data</code></em>
- is passed to <em class="parameter"><code>func</code></em>
- and <em class="parameter"><code>notify</code></em>
- is called when
-<em class="parameter"><code>user_data</code></em>
- is no longer in use.</p>
-<p>By default, the client will send the messages on the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="type">GstRTSPConnection</span></a> that
-was configured with <a class="link" href="GstRTSPClient.html#gst-rtsp-client-attach" title="gst_rtsp_client_attach ()"><code class="function">gst_rtsp_client_attach()</code></a> was called.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.16.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>func</p></td>
-<td class="parameter_description"><p> a <a class="link" href="GstRTSPClient.html#GstRTSPClientSendFunc" title="GstRTSPClientSendFunc ()"><span class="type">GstRTSPClientSendFunc</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid until the GDestroyNotify argument is called."><span class="acronym">scope notified</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p> user data passed to <em class="parameter"><code>func</code></em>
-. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="This parameter is a 'user_data', for callbacks; many bindings can pass NULL here."><span class="acronym">closure</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>notify</p></td>
-<td class="parameter_description"><p> called when <em class="parameter"><code>user_data</code></em>
-is no longer in use. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-client-handle-message"></a><h3>gst_rtsp_client_handle_message ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="returnvalue">GstRTSPResult</span></a>
-gst_rtsp_client_handle_message (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#GstRTSPMessage"><span class="type">GstRTSPMessage</span></a> *message</code></em>);</pre>
-<p>Let the client handle <em class="parameter"><code>message</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.17.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>message</p></td>
-<td class="parameter_description"><p> an <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#GstRTSPMessage"><span class="type">GstRTSPMessage</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.2.9.17.6"></a><h4>Returns</h4>
-<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="type">GstRTSPResult</span></a>.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-client-send-message"></a><h3>gst_rtsp_client_send_message ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="returnvalue">GstRTSPResult</span></a>
-gst_rtsp_client_send_message (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#GstRTSPMessage"><span class="type">GstRTSPMessage</span></a> *message</code></em>);</pre>
-<p>Send a message message to the remote end. <em class="parameter"><code>message</code></em>
- must be a
-<span class="type">GST_RTSP_MESSAGE_REQUEST</span> or a <span class="type">GST_RTSP_MESSAGE_RESPONSE</span>.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.18.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>session</p></td>
-<td class="parameter_description"><p> a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> to send
-the message to or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>][<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>message</p></td>
-<td class="parameter_description"><p> The <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#GstRTSPMessage"><span class="type">GstRTSPMessage</span></a> to send. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPClientSessionFilterFunc"></a><h3>GstRTSPClientSessionFilterFunc ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
-<span class="c_punctuation">(</span>*GstRTSPClientSessionFilterFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *sess</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
-<p>This function will be called by the <a class="link" href="GstRTSPClient.html#gst-rtsp-client-session-filter" title="gst_rtsp_client_session_filter ()"><code class="function">gst_rtsp_client_session_filter()</code></a>. An
-implementation should return a value of <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
-<p>When this function returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, <em class="parameter"><code>sess</code></em>
- will be removed
-from <em class="parameter"><code>client</code></em>
-.</p>
-<p>A return value of <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a> will leave <em class="parameter"><code>sess</code></em>
- untouched in
-<em class="parameter"><code>client</code></em>
-.</p>
-<p>A value of <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a> will add <em class="parameter"><code>sess</code></em>
- to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-session-filter" title="gst_rtsp_client_session_filter ()"><code class="function">gst_rtsp_client_session_filter()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.19.8"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> object</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>sess</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> in <em class="parameter"><code>client</code></em>
-</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p>user data that has been given to <a class="link" href="GstRTSPClient.html#gst-rtsp-client-session-filter" title="gst_rtsp_client_session_filter ()"><code class="function">gst_rtsp_client_session_filter()</code></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.2.9.19.9"></a><h4>Returns</h4>
-<p> a <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-client-session-filter"></a><h3>gst_rtsp_client_session_filter ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
-gst_rtsp_client_session_filter (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPClient.html#GstRTSPClientSessionFilterFunc" title="GstRTSPClientSessionFilterFunc ()"><span class="type">GstRTSPClientSessionFilterFunc</span></a> func</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
-<p>Call <em class="parameter"><code>func</code></em>
- for each session managed by <em class="parameter"><code>client</code></em>
-. The result value of <em class="parameter"><code>func</code></em>
-
-determines what happens to the session. <em class="parameter"><code>func</code></em>
- will be called with <em class="parameter"><code>client</code></em>
-
-locked so no further actions on <em class="parameter"><code>client</code></em>
- can be performed from <em class="parameter"><code>func</code></em>
-.</p>
-<p>If <em class="parameter"><code>func</code></em>
- returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, the session will be removed from
-<em class="parameter"><code>client</code></em>
-.</p>
-<p>If <em class="parameter"><code>func</code></em>
- returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a>, the session will remain in <em class="parameter"><code>client</code></em>
-.</p>
-<p>If <em class="parameter"><code>func</code></em>
- returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>, the session will remain in <em class="parameter"><code>client</code></em>
- but
-will also be added with an additional ref to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of this
-function..</p>
-<p>When <em class="parameter"><code>func</code></em>
- is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a> will be assumed for each session.</p>
-<div class="refsect3">
-<a name="id-1.2.2.9.20.9"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>func</p></td>
-<td class="parameter_description"><p> a callback. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid only during the call to the method."><span class="acronym">scope call</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p>user data passed to <em class="parameter"><code>func</code></em>
-</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.2.9.20.10"></a><h4>Returns</h4>
-<p> a <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> with all
-sessions for which <em class="parameter"><code>func</code></em>
-returned <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>. After usage, each
-element in the <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> should be unreffed before the list is freed. </p>
-<p><span class="annotation">[<acronym title="Generics and defining elements of containers and arrays."><span class="acronym">element-type</span></acronym> GstRTSPSession][<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPClient.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstRTSPClient-struct"></a><h3>struct GstRTSPClient</h3>
-<pre class="programlisting">struct GstRTSPClient;</pre>
-<p>The client object represents the connection and its state with a client.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPClientClass"></a><h3>struct GstRTSPClientClass</h3>
-<pre class="programlisting">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);
-};
-</pre>
-<p>The client class structure.</p>
-<div class="refsect3">
-<a name="id-1.2.2.10.3.5"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="struct_members_name">
-<col class="struct_members_description">
-<col width="200px" class="struct_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObjectClass"><span class="type">GObjectClass</span></a> <em class="structfield"><code><a name="GstRTSPClientClass.parent-class"></a>parent_class</code></em>;</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.create-sdp"></a>create_sdp</code></em> ()</p></td>
-<td class="struct_member_description"><p>called when the SDP needs to be created for media.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.configure-client-media"></a>configure_client_media</code></em> ()</p></td>
-<td class="struct_member_description"><p>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.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.configure-client-transport"></a>configure_client_transport</code></em> ()</p></td>
-<td class="struct_member_description"><p>called when the client transport needs to be
-configured.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.params-set"></a>params_set</code></em> ()</p></td>
-<td class="struct_member_description"><p>set parameters. This function should also initialize the
-RTSP response(ctx->response) via a call to <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#gst-rtsp-message-init-response"><code class="function">gst_rtsp_message_init_response()</code></a></p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.params-get"></a>params_get</code></em> ()</p></td>
-<td class="struct_member_description"><p>get parameters. This function should also initialize the
-RTSP response(ctx->response) via a call to <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#gst-rtsp-message-init-response"><code class="function">gst_rtsp_message_init_response()</code></a></p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.make-path-from-uri"></a>make_path_from_uri</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.closed"></a>closed</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.new-session"></a>new_session</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.options-request"></a>options_request</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.describe-request"></a>describe_request</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.setup-request"></a>setup_request</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.play-request"></a>play_request</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.pause-request"></a>pause_request</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.teardown-request"></a>teardown_request</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.set-parameter-request"></a>set_parameter_request</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.get-parameter-request"></a>get_parameter_request</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.handle-response"></a>handle_response</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPClientClass.tunnel-http-response"></a>tunnel_http_response</code></em> ()</p></td>
-<td class="struct_member_description"><p>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</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPClient.property-details"></a><h2>Property Details</h2>
-<div class="refsect2">
-<a name="GstRTSPClient--drop-backlog"></a><h3>The <code class="literal">“drop-backlog”</code> property</h3>
-<pre class="programlisting"> “drop-backlog” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></pre>
-<p>Drop data when the backlog queue is full.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: TRUE</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPClient--mount-points"></a><h3>The <code class="literal">“mount-points”</code> property</h3>
-<pre class="programlisting"> “mount-points” <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *</pre>
-<p>The mount points to use for client session.</p>
-<p>Flags: Read / Write</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPClient--session-pool"></a><h3>The <code class="literal">“session-pool”</code> property</h3>
-<pre class="programlisting"> “session-pool” <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *</pre>
-<p>The session pool to use for client session.</p>
-<p>Flags: Read / Write</p>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPClient.signal-details"></a><h2>Signal Details</h2>
-<div class="refsect2">
-<a name="GstRTSPClient-closed"></a><h3>The <code class="literal">“closed”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPClient-describe-request"></a><h3>The <code class="literal">“describe-request”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
- <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPClient-get-parameter-request"></a><h3>The <code class="literal">“get-parameter-request”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
- <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPClient-handle-response"></a><h3>The <code class="literal">“handle-response”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
- <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPClient-new-session"></a><h3>The <code class="literal">“new-session”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
- <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPClient-options-request"></a><h3>The <code class="literal">“options-request”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
- <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPClient-pause-request"></a><h3>The <code class="literal">“pause-request”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
- <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPClient-play-request"></a><h3>The <code class="literal">“play-request”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
- <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPClient-send-message"></a><h3>The <code class="literal">“send-message”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client,
- <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *session,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> message,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<div class="refsect3">
-<a name="id-1.2.2.12.10.4"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>The RTSP client</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>session</p></td>
-<td class="parameter_description"><p> The session. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Override the parsed C type with given type."><span class="acronym">type</span></acronym> GstRtspServer.RTSPSession]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>message</p></td>
-<td class="parameter_description"><p> The message. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Override the parsed C type with given type."><span class="acronym">type</span></acronym> GstRtsp.RTSPMessage]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p>user data set when the signal handler was connected.</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPClient-set-parameter-request"></a><h3>The <code class="literal">“set-parameter-request”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
- <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPClient-setup-request"></a><h3>The <code class="literal">“setup-request”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
- <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPClient-teardown-request"></a><h3>The <code class="literal">“teardown-request”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *gstrtspclient,
- <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPClient.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a></p>
-</div>
-</div>
-<div class="footer">
-<hr>
- Generated by GTK-Doc V1.21</div>
-</body>
-</html>
\ No newline at end of file
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-<html>
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-<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
-<title>GStreamer RTSP Server Reference Manual: GstRTSPMedia</title>
-<meta name="generator" content="DocBook XSL Stylesheets V1.78.1">
-<link rel="home" href="index.html" title="GStreamer RTSP Server Reference Manual">
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- <a href="#GstRTSPMedia.object-hierarchy" class="shortcut">Object Hierarchy</a></span><span id="nav_properties"> <span class="dim">|</span>
- <a href="#GstRTSPMedia.properties" class="shortcut">Properties</a></span><span id="nav_signals"> <span class="dim">|</span>
- <a href="#GstRTSPMedia.signals" class="shortcut">Signals</a></span>
-</td>
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-</tr></table>
-<div class="refentry">
-<a name="GstRTSPMedia"></a><div class="titlepage"></div>
-<div class="refnamediv"><table width="100%"><tr>
-<td valign="top">
-<h2><span class="refentrytitle"><a name="GstRTSPMedia.top_of_page"></a>GstRTSPMedia</span></h2>
-<p>GstRTSPMedia — The media pipeline</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="GstRTSPMedia.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="returnvalue">GstRTSPMedia</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-new" title="gst_rtsp_media_new ()">gst_rtsp_media_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="returnvalue">GstElement</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-element" title="gst_rtsp_media_get_element ()">gst_rtsp_media_get_element</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-take-pipeline" title="gst_rtsp_media_take_pipeline ()">gst_rtsp_media_take_pipeline</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-permissions" title="gst_rtsp_media_set_permissions ()">gst_rtsp_media_set_permissions</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="returnvalue">GstRTSPPermissions</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-permissions" title="gst_rtsp_media_get_permissions ()">gst_rtsp_media_get_permissions</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-shared" title="gst_rtsp_media_set_shared ()">gst_rtsp_media_set_shared</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-is-shared" title="gst_rtsp_media_is_shared ()">gst_rtsp_media_is_shared</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-reusable" title="gst_rtsp_media_set_reusable ()">gst_rtsp_media_set_reusable</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-is-reusable" title="gst_rtsp_media_is_reusable ()">gst_rtsp_media_is_reusable</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-profiles" title="gst_rtsp_media_set_profiles ()">gst_rtsp_media_set_profiles</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="returnvalue">GstRTSPProfile</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-profiles" title="gst_rtsp_media_get_profiles ()">gst_rtsp_media_get_profiles</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-protocols" title="gst_rtsp_media_set_protocols ()">gst_rtsp_media_set_protocols</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="returnvalue">GstRTSPLowerTrans</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-protocols" title="gst_rtsp_media_get_protocols ()">gst_rtsp_media_get_protocols</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-eos-shutdown" title="gst_rtsp_media_set_eos_shutdown ()">gst_rtsp_media_set_eos_shutdown</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-is-eos-shutdown" title="gst_rtsp_media_is_eos_shutdown ()">gst_rtsp_media_is_eos_shutdown</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-address-pool" title="gst_rtsp_media_set_address_pool ()">gst_rtsp_media_set_address_pool</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="returnvalue">GstRTSPAddressPool</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-address-pool" title="gst_rtsp_media_get_address_pool ()">gst_rtsp_media_get_address_pool</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-buffer-size" title="gst_rtsp_media_set_buffer_size ()">gst_rtsp_media_set_buffer_size</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-buffer-size" title="gst_rtsp_media_get_buffer_size ()">gst_rtsp_media_get_buffer_size</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-setup-sdp" title="gst_rtsp_media_setup_sdp ()">gst_rtsp_media_setup_sdp</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-prepare" title="gst_rtsp_media_prepare ()">gst_rtsp_media_prepare</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-unprepare" title="gst_rtsp_media_unprepare ()">gst_rtsp_media_unprepare</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPMedia.html#GstRTSPMediaStatus" title="enum GstRTSPMediaStatus"><span class="returnvalue">GstRTSPMediaStatus</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-status" title="gst_rtsp_media_get_status ()">gst_rtsp_media_get_status</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-suspend-mode" title="gst_rtsp_media_set_suspend_mode ()">gst_rtsp_media_set_suspend_mode</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="returnvalue">GstRTSPSuspendMode</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-suspend-mode" title="gst_rtsp_media_get_suspend_mode ()">gst_rtsp_media_get_suspend_mode</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-suspend" title="gst_rtsp_media_suspend ()">gst_rtsp_media_suspend</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-unsuspend" title="gst_rtsp_media_unsuspend ()">gst_rtsp_media_unsuspend</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-collect-streams" title="gst_rtsp_media_collect_streams ()">gst_rtsp_media_collect_streams</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-create-stream" title="gst_rtsp_media_create_stream ()">gst_rtsp_media_create_stream</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-n-streams" title="gst_rtsp_media_n_streams ()">gst_rtsp_media_n_streams</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-stream" title="gst_rtsp_media_get_stream ()">gst_rtsp_media_get_stream</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-find-stream" title="gst_rtsp_media_find_stream ()">gst_rtsp_media_find_stream</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-seek" title="gst_rtsp_media_seek ()">gst_rtsp_media_seek</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-range-string" title="gst_rtsp_media_get_range_string ()">gst_rtsp_media_get_range_string</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-state" title="gst_rtsp_media_set_state ()">gst_rtsp_media_set_state</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-pipeline-state" title="gst_rtsp_media_set_pipeline_state ()">gst_rtsp_media_set_pipeline_state</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html"><span class="returnvalue">GstClock</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-clock" title="gst_rtsp_media_get_clock ()">gst_rtsp_media_get_clock</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html#GstClockTime"><span class="returnvalue">GstClockTime</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-base-time" title="gst_rtsp_media_get_base_time ()">gst_rtsp_media_get_base_time</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-use-time-provider" title="gst_rtsp_media_use_time_provider ()">gst_rtsp_media_use_time_provider</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-is-time-provider" title="gst_rtsp_media_is_time_provider ()">gst_rtsp_media_is_time_provider</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstNetTimeProvider.html"><span class="returnvalue">GstNetTimeProvider</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-time-provider" title="gst_rtsp_media_get_time_provider ()">gst_rtsp_media_get_time_provider</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMedia.properties"></a><h2>Properties</h2>
-<div class="informaltable"><table border="0">
-<colgroup>
-<col width="150px" class="properties_type">
-<col width="300px" class="properties_name">
-<col width="200px" class="properties_flags">
-</colgroup>
-<tbody>
-<tr>
-<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--buffer-size" title="The “buffer-size” property">buffer-size</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> *</td>
-<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--element" title="The “element” property">element</a></td>
-<td class="property_flags">Read / Write / Construct Only</td>
-</tr>
-<tr>
-<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--eos-shutdown" title="The “eos-shutdown” property">eos-shutdown</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--profiles" title="The “profiles” property">profiles</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--protocols" title="The “protocols” property">protocols</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--reusable" title="The “reusable” property">reusable</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--shared" title="The “shared” property">shared</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type"><a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--suspend-mode" title="The “suspend-mode” property">suspend-mode</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia--time-provider" title="The “time-provider” property">time-provider</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMedia.signals"></a><h2>Signals</h2>
-<div class="informaltable"><table border="0">
-<colgroup>
-<col width="150px" class="signals_return">
-<col width="300px" class="signals_name">
-<col width="200px" class="signals_flags">
-</colgroup>
-<tbody>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia-new-state" title="The “new-state” signal">new-state</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia-new-stream" title="The “new-stream” signal">new-stream</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia-prepared" title="The “prepared” signal">prepared</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia-removed-stream" title="The “removed-stream” signal">removed-stream</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia-target-state" title="The “target-state” signal">target-state</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia-unprepared" title="The “unprepared” signal">unprepared</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMedia.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMedia-struct" title="struct GstRTSPMedia">GstRTSPMedia</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMediaClass" title="struct GstRTSPMediaClass">GstRTSPMediaClass</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">enum</td>
-<td class="function_name"><a class="link" href="GstRTSPMedia.html#GstRTSPMediaStatus" title="enum GstRTSPMediaStatus">GstRTSPMediaStatus</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">enum</td>
-<td class="function_name"><a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode">GstRTSPSuspendMode</a></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMedia.object-hierarchy"></a><h2>Object Hierarchy</h2>
-<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
- <span class="lineart">╰──</span> GstRTSPMedia
-</pre>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMedia.description"></a><h2>Description</h2>
-<p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> contains the complete GStreamer pipeline to manage the
-streaming to the clients. The actual data transfer is done by the
-<a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> objects that are created and exposed by the <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a>.</p>
-<p>The <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> is usually created from a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> when the
-client does a DESCRIBE or SETUP of a resource.</p>
-<p>A media is created with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-new" title="gst_rtsp_media_new ()"><code class="function">gst_rtsp_media_new()</code></a> that takes the element that will
-provide the streaming elements. For each of the streams, a new <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a>
-object needs to be made with the <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-create-stream" title="gst_rtsp_media_create_stream ()"><code class="function">gst_rtsp_media_create_stream()</code></a> which takes
-the payloader element and the source pad that produces the RTP stream.</p>
-<p>The pipeline of the media is set to PAUSED with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-prepare" title="gst_rtsp_media_prepare ()"><code class="function">gst_rtsp_media_prepare()</code></a>. 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.</p>
-<p>It is also possible to dynamically create <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> objects during the
-prepare phase. With <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-status" title="gst_rtsp_media_get_status ()"><code class="function">gst_rtsp_media_get_status()</code></a> you can check the status of
-the prepare phase.</p>
-<p>After the media is prepared, it is ready for streaming. It will usually be
-managed in a session with <a class="link" href="GstRTSPSession.html#gst-rtsp-session-manage-media" title="gst_rtsp_session_manage_media ()"><code class="function">gst_rtsp_session_manage_media()</code></a>. See
-<a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> and <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a>.</p>
-<p>The state of the media can be controlled with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-state" title="gst_rtsp_media_set_state ()"><code class="function">gst_rtsp_media_set_state()</code></a>.
-Seeking can be done with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-seek" title="gst_rtsp_media_seek ()"><code class="function">gst_rtsp_media_seek()</code></a>.</p>
-<p>With <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-unprepare" title="gst_rtsp_media_unprepare ()"><code class="function">gst_rtsp_media_unprepare()</code></a> the pipeline is stopped and shut down. When
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-eos-shutdown" title="gst_rtsp_media_set_eos_shutdown ()"><code class="function">gst_rtsp_media_set_eos_shutdown()</code></a> an EOS will be sent to the pipeline to
-cleanly shut down.</p>
-<p>With <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-shared" title="gst_rtsp_media_set_shared ()"><code class="function">gst_rtsp_media_set_shared()</code></a>, the media can be shared between multiple
-clients. With <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-reusable" title="gst_rtsp_media_set_reusable ()"><code class="function">gst_rtsp_media_set_reusable()</code></a> you can control if the pipeline
-can be prepared again after an unprepare.</p>
-<p>Last reviewed on 2013-07-11 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMedia.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-media-new"></a><h3>gst_rtsp_media_new ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="returnvalue">GstRTSPMedia</span></a> *
-gst_rtsp_media_new (<em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> *element</code></em>);</pre>
-<p>Create a new <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> instance. <em class="parameter"><code>element</code></em>
- is the bin element that
-provides the different streams. The <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> object contains the
-element to produce RTP data for one or more related (audio/video/..)
-streams.</p>
-<p>Ownership is taken of <em class="parameter"><code>element</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.2.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>element</p></td>
-<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.2.7"></a><h4>Returns</h4>
-<p> a new <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> object. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-get-element"></a><h3>gst_rtsp_media_get_element ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="returnvalue">GstElement</span></a> *
-gst_rtsp_media_get_element (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Get the element that was used when constructing <em class="parameter"><code>media</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.3.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.3.6"></a><h4>Returns</h4>
-<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a>. Unref after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-take-pipeline"></a><h3>gst_rtsp_media_take_pipeline ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_take_pipeline (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPipeline.html"><span class="type">GstPipeline</span></a> *pipeline</code></em>);</pre>
-<p>Set <em class="parameter"><code>pipeline</code></em>
- as the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPipeline.html"><span class="type">GstPipeline</span></a> for <em class="parameter"><code>media</code></em>
-. Ownership is
-taken of <em class="parameter"><code>pipeline</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.4.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>pipeline</p></td>
-<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPipeline.html"><span class="type">GstPipeline</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-set-permissions"></a><h3>gst_rtsp_media_set_permissions ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_set_permissions (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>);</pre>
-<p>Set <em class="parameter"><code>permissions</code></em>
- on <em class="parameter"><code>media</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.5.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>permissions</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-get-permissions"></a><h3>gst_rtsp_media_get_permissions ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="returnvalue">GstRTSPPermissions</span></a> *
-gst_rtsp_media_get_permissions (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Get the permissions object from <em class="parameter"><code>media</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.6.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.6.6"></a><h4>Returns</h4>
-<p> a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> object, unref after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-set-shared"></a><h3>gst_rtsp_media_set_shared ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_set_shared (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> shared</code></em>);</pre>
-<p>Set or unset if the pipeline for <em class="parameter"><code>media</code></em>
- can be shared will multiple clients.
-When <em class="parameter"><code>shared</code></em>
- is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a>, client requests for this media will share the media
-pipeline.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.7.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>shared</p></td>
-<td class="parameter_description"><p>the new value</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-is-shared"></a><h3>gst_rtsp_media_is_shared ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_media_is_shared (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Check if the pipeline for <em class="parameter"><code>media</code></em>
- can be shared between multiple clients.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.8.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.8.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the media can be shared between clients.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-set-reusable"></a><h3>gst_rtsp_media_set_reusable ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_set_reusable (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> reusable</code></em>);</pre>
-<p>Set or unset if the pipeline for <em class="parameter"><code>media</code></em>
- can be reused after the pipeline has
-been unprepared.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.9.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>reusable</p></td>
-<td class="parameter_description"><p>the new value</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-is-reusable"></a><h3>gst_rtsp_media_is_reusable ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_media_is_reusable (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Check if the pipeline for <em class="parameter"><code>media</code></em>
- can be reused after an unprepare.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.10.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.10.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the media can be reused</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-set-profiles"></a><h3>gst_rtsp_media_set_profiles ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_set_profiles (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a> profiles</code></em>);</pre>
-<p>Configure the allowed lower transport for <em class="parameter"><code>media</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.11.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>profiles</p></td>
-<td class="parameter_description"><p>the new flags</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-get-profiles"></a><h3>gst_rtsp_media_get_profiles ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="returnvalue">GstRTSPProfile</span></a>
-gst_rtsp_media_get_profiles (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Get the allowed profiles of <em class="parameter"><code>media</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.12.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.12.6"></a><h4>Returns</h4>
-<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a></p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-set-protocols"></a><h3>gst_rtsp_media_set_protocols ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_set_protocols (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a> protocols</code></em>);</pre>
-<p>Configure the allowed lower transport for <em class="parameter"><code>media</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.13.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>protocols</p></td>
-<td class="parameter_description"><p>the new flags</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-get-protocols"></a><h3>gst_rtsp_media_get_protocols ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="returnvalue">GstRTSPLowerTrans</span></a>
-gst_rtsp_media_get_protocols (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Get the allowed protocols of <em class="parameter"><code>media</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.14.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.14.6"></a><h4>Returns</h4>
-<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a></p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-set-eos-shutdown"></a><h3>gst_rtsp_media_set_eos_shutdown ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_set_eos_shutdown (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> eos_shutdown</code></em>);</pre>
-<p>Set or unset if an EOS event will be sent to the pipeline for <em class="parameter"><code>media</code></em>
- before
-it is unprepared.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.15.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>eos_shutdown</p></td>
-<td class="parameter_description"><p>the new value</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-is-eos-shutdown"></a><h3>gst_rtsp_media_is_eos_shutdown ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_media_is_eos_shutdown (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Check if the pipeline for <em class="parameter"><code>media</code></em>
- will send an EOS down the pipeline before
-unpreparing.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.16.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.16.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the media will send EOS before unpreparing.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-set-address-pool"></a><h3>gst_rtsp_media_set_address_pool ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_set_address_pool (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>);</pre>
-<p>configure <em class="parameter"><code>pool</code></em>
- to be used as the address pool of <em class="parameter"><code>media</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.17.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-get-address-pool"></a><h3>gst_rtsp_media_get_address_pool ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="returnvalue">GstRTSPAddressPool</span></a> *
-gst_rtsp_media_get_address_pool (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Get the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> used as the address pool of <em class="parameter"><code>media</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.18.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.18.6"></a><h4>Returns</h4>
-<p> the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> of <em class="parameter"><code>media</code></em>
-. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
-usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-set-buffer-size"></a><h3>gst_rtsp_media_set_buffer_size ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_set_buffer_size (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> size</code></em>);</pre>
-<p>Set the kernel UDP buffer size.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.19.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>size</p></td>
-<td class="parameter_description"><p>the new value</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-get-buffer-size"></a><h3>gst_rtsp_media_get_buffer_size ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-gst_rtsp_media_get_buffer_size (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Get the kernel UDP buffer size.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.20.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.20.6"></a><h4>Returns</h4>
-<p> the kernel UDP buffer size.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-setup-sdp"></a><h3>gst_rtsp_media_setup_sdp ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_media_setup_sdp (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstsdpmessage.html#GstSDPMessage"><span class="type">GstSDPMessage</span></a> *sdp</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSdp.html#GstSDPInfo" title="GstSDPInfo"><span class="type">GstSDPInfo</span></a> *info</code></em>);</pre>
-<p>Add <em class="parameter"><code>media</code></em>
- specific info to <em class="parameter"><code>sdp</code></em>
-. <em class="parameter"><code>info</code></em>
- is used to configure the connection
-information in the SDP.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.21.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>sdp</p></td>
-<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstsdpmessage.html#GstSDPMessage"><span class="type">GstSDPMessage</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>info</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPSdp.html#GstSDPInfo" title="GstSDPInfo"><span class="type">GstSDPInfo</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.21.6"></a><h4>Returns</h4>
-<p> TRUE on success.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-prepare"></a><h3>gst_rtsp_media_prepare ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_media_prepare (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a> *thread</code></em>);</pre>
-<p>Prepare <em class="parameter"><code>media</code></em>
- for streaming. This function will create the objects
-to manage the streaming. A pipeline must have been set on <em class="parameter"><code>media</code></em>
- with
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-take-pipeline" title="gst_rtsp_media_take_pipeline ()"><code class="function">gst_rtsp_media_take_pipeline()</code></a>.</p>
-<p>It will preroll the pipeline and collect vital information about the streams
-such as the duration.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.22.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>thread</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a> to run the
-bus handler or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.22.7"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-unprepare"></a><h3>gst_rtsp_media_unprepare ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_media_unprepare (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Unprepare <em class="parameter"><code>media</code></em>
-. 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.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.23.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.23.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-get-status"></a><h3>gst_rtsp_media_get_status ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPMedia.html#GstRTSPMediaStatus" title="enum GstRTSPMediaStatus"><span class="returnvalue">GstRTSPMediaStatus</span></a>
-gst_rtsp_media_get_status (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Get the status of <em class="parameter"><code>media</code></em>
-. When <em class="parameter"><code>media</code></em>
- is busy preparing, this function waits
-until <em class="parameter"><code>media</code></em>
- is prepared or in error.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.24.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.24.6"></a><h4>Returns</h4>
-<p> the status of <em class="parameter"><code>media</code></em>
-.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-set-suspend-mode"></a><h3>gst_rtsp_media_set_suspend_mode ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_set_suspend_mode (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a> mode</code></em>);</pre>
-<p>Control how @ media will be suspended after the SDP has been generated and
-after a PAUSE request has been performed.</p>
-<p>Media must be unprepared when setting the suspend mode.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.25.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>mode</p></td>
-<td class="parameter_description"><p>the new <a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-get-suspend-mode"></a><h3>gst_rtsp_media_get_suspend_mode ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="returnvalue">GstRTSPSuspendMode</span></a>
-gst_rtsp_media_get_suspend_mode (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Get how <em class="parameter"><code>media</code></em>
- will be suspended.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.26.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.26.6"></a><h4>Returns</h4>
-<p> <a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a>.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-suspend"></a><h3>gst_rtsp_media_suspend ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_media_suspend (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Suspend <em class="parameter"><code>media</code></em>
-. The state of the pipeline managed by <em class="parameter"><code>media</code></em>
- is set to
-GST_STATE_NULL but all streams are kept. <em class="parameter"><code>media</code></em>
- can be prepared again
-with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-unsuspend" title="gst_rtsp_media_unsuspend ()"><code class="function">gst_rtsp_media_unsuspend()</code></a></p>
-<p><em class="parameter"><code>media</code></em>
- must be prepared with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-prepare" title="gst_rtsp_media_prepare ()"><code class="function">gst_rtsp_media_prepare()</code></a>;</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.27.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.27.7"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-unsuspend"></a><h3>gst_rtsp_media_unsuspend ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_media_unsuspend (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Unsuspend <em class="parameter"><code>media</code></em>
- if it was in a suspended state. This method does nothing
-when the media was not in the suspended state.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.28.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.28.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-collect-streams"></a><h3>gst_rtsp_media_collect_streams ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_collect_streams (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Find all payloader elements, they should be named pay%d in the
-element of <em class="parameter"><code>media</code></em>
-, and create <a href="../gst-rtsp-server-1.0/GstRTSPStream.html"><span class="type">GstRTSPStreams</span></a> for them.</p>
-<p>Collect all dynamic elements, named dynpay%d, and add them to
-the list of dynamic elements.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.29.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-create-stream"></a><h3>gst_rtsp_media_create_stream ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
-gst_rtsp_media_create_stream (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> *payloader</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html"><span class="type">GstPad</span></a> *srcpad</code></em>);</pre>
-<p>Create a new stream in <em class="parameter"><code>media</code></em>
- that provides RTP data on <em class="parameter"><code>srcpad</code></em>
-.
-<em class="parameter"><code>srcpad</code></em>
- should be a pad of an element inside <em class="parameter"><code>media->element</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.30.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>payloader</p></td>
-<td class="parameter_description"><p>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>srcpad</p></td>
-<td class="parameter_description"><p>a source <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html"><span class="type">GstPad</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.30.6"></a><h4>Returns</h4>
-<p> a new <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> that remains valid for as long
-as <em class="parameter"><code>media</code></em>
-exists. </p>
-<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-n-streams"></a><h3>gst_rtsp_media_n_streams ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-gst_rtsp_media_n_streams (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Get the number of streams in this media.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.31.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.31.6"></a><h4>Returns</h4>
-<p> The number of streams.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-get-stream"></a><h3>gst_rtsp_media_get_stream ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
-gst_rtsp_media_get_stream (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> idx</code></em>);</pre>
-<p>Retrieve the stream with index <em class="parameter"><code>idx</code></em>
- from <em class="parameter"><code>media</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.32.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>idx</p></td>
-<td class="parameter_description"><p>the stream index</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.32.6"></a><h4>Returns</h4>
-<p> the <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> at index
-<em class="parameter"><code>idx</code></em>
-or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when a stream with that index did not exist. </p>
-<p><span class="annotation">[<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>][<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-find-stream"></a><h3>gst_rtsp_media_find_stream ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
-gst_rtsp_media_find_stream (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *control</code></em>);</pre>
-<p>Find a stream in <em class="parameter"><code>media</code></em>
- with <em class="parameter"><code>control</code></em>
- as the control uri.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.33.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>control</p></td>
-<td class="parameter_description"><p>the control of the stream</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.33.6"></a><h4>Returns</h4>
-<p> the <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> with
-control uri <em class="parameter"><code>control</code></em>
-or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when a stream with that control did
-not exist. </p>
-<p><span class="annotation">[<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>][<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-seek"></a><h3>gst_rtsp_media_seek ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_media_seek (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsprange.html#GstRTSPTimeRange"><span class="type">GstRTSPTimeRange</span></a> *range</code></em>);</pre>
-<p>Seek the pipeline of <em class="parameter"><code>media</code></em>
- to <em class="parameter"><code>range</code></em>
-. <em class="parameter"><code>media</code></em>
- must be prepared with
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-prepare" title="gst_rtsp_media_prepare ()"><code class="function">gst_rtsp_media_prepare()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.34.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>range</p></td>
-<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsprange.html#GstRTSPTimeRange"><span class="type">GstRTSPTimeRange</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.34.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-get-range-string"></a><h3>gst_rtsp_media_get_range_string ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-gst_rtsp_media_get_range_string (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> play</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsprange.html#GstRTSPRangeUnit"><span class="type">GstRTSPRangeUnit</span></a> unit</code></em>);</pre>
-<p>Get the current range as a string. <em class="parameter"><code>media</code></em>
- must be prepared with
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-prepare" title="gst_rtsp_media_prepare ()"><code class="function">gst_rtsp_media_prepare()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.35.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>play</p></td>
-<td class="parameter_description"><p>for the PLAY request</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>unit</p></td>
-<td class="parameter_description"><p>the unit to use for the string</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.35.6"></a><h4>Returns</h4>
-<p> The range as a string, <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-set-state"></a><h3>gst_rtsp_media_set_state ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_media_set_state (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#GstState"><span class="type">GstState</span></a> state</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Pointer-Arrays.html#GPtrArray"><span class="type">GPtrArray</span></a> *transports</code></em>);</pre>
-<p>Set the state of <em class="parameter"><code>media</code></em>
- to <em class="parameter"><code>state</code></em>
- and for the transports in <em class="parameter"><code>transports</code></em>
-.</p>
-<p><em class="parameter"><code>media</code></em>
- must be prepared with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-prepare" title="gst_rtsp_media_prepare ()"><code class="function">gst_rtsp_media_prepare()</code></a>;</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.36.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>state</p></td>
-<td class="parameter_description"><p>the target state of the media</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>transports</p></td>
-<td class="parameter_description"><p>a <a href="https://developer.gnome.org/glib/unstable/glib-Pointer-Arrays.html#GPtrArray"><span class="type">GPtrArray</span></a> of <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> pointers. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>][<acronym title="Generics and defining elements of containers and arrays."><span class="acronym">element-type</span></acronym> GstRtspServer.RTSPStreamTransport]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.36.7"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-set-pipeline-state"></a><h3>gst_rtsp_media_set_pipeline_state ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_set_pipeline_state (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#GstState"><span class="type">GstState</span></a> state</code></em>);</pre>
-<p>Set the state of the pipeline managed by <em class="parameter"><code>media</code></em>
- to <em class="parameter"><code>state</code></em>
-</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.37.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>state</p></td>
-<td class="parameter_description"><p>the target state of the pipeline</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-get-clock"></a><h3>gst_rtsp_media_get_clock ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html"><span class="returnvalue">GstClock</span></a> *
-gst_rtsp_media_get_clock (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Get the clock that is used by the pipeline in <em class="parameter"><code>media</code></em>
-.</p>
-<p><em class="parameter"><code>media</code></em>
- must be prepared before this method returns a valid clock object.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.38.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.38.7"></a><h4>Returns</h4>
-<p> the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html"><span class="type">GstClock</span></a> used by <em class="parameter"><code>media</code></em>
-. unref after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-get-base-time"></a><h3>gst_rtsp_media_get_base_time ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html#GstClockTime"><span class="returnvalue">GstClockTime</span></a>
-gst_rtsp_media_get_base_time (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Get the base_time that is used by the pipeline in <em class="parameter"><code>media</code></em>
-.</p>
-<p><em class="parameter"><code>media</code></em>
- must be prepared before this method returns a valid base_time.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.39.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.39.7"></a><h4>Returns</h4>
-<p> the base_time used by <em class="parameter"><code>media</code></em>
-.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-use-time-provider"></a><h3>gst_rtsp_media_use_time_provider ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_use_time_provider (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> time_provider</code></em>);</pre>
-<p>Set <em class="parameter"><code>media</code></em>
- to provide a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstNetTimeProvider.html"><span class="type">GstNetTimeProvider</span></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.40.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>time_provider</p></td>
-<td class="parameter_description"><p>if a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstNetTimeProvider.html"><span class="type">GstNetTimeProvider</span></a> should be used</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-is-time-provider"></a><h3>gst_rtsp_media_is_time_provider ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_media_is_time_provider (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Check if <em class="parameter"><code>media</code></em>
- can provide a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstNetTimeProvider.html"><span class="type">GstNetTimeProvider</span></a> for its pipeline clock.</p>
-<p>Use <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-time-provider" title="gst_rtsp_media_get_time_provider ()"><code class="function">gst_rtsp_media_get_time_provider()</code></a> to get the network clock.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.41.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.41.7"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>media</code></em>
-can provide a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstNetTimeProvider.html"><span class="type">GstNetTimeProvider</span></a>.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-get-time-provider"></a><h3>gst_rtsp_media_get_time_provider ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstNetTimeProvider.html"><span class="returnvalue">GstNetTimeProvider</span></a> *
-gst_rtsp_media_get_time_provider (<em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *address</code></em>,
- <em class="parameter"><code><span class="type">guint16</span> port</code></em>);</pre>
-<p>Get the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstNetTimeProvider.html"><span class="type">GstNetTimeProvider</span></a> for the clock used by <em class="parameter"><code>media</code></em>
-. The time provider
-will listen on <em class="parameter"><code>address</code></em>
- and <em class="parameter"><code>port</code></em>
- for client time requests.</p>
-<div class="refsect3">
-<a name="id-1.2.7.9.42.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>address</p></td>
-<td class="parameter_description"><p> an address or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>port</p></td>
-<td class="parameter_description"><p>a port or 0</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.7.9.42.6"></a><h4>Returns</h4>
-<p> the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstNetTimeProvider.html"><span class="type">GstNetTimeProvider</span></a> of <em class="parameter"><code>media</code></em>
-. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMedia.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstRTSPMedia-struct"></a><h3>struct GstRTSPMedia</h3>
-<pre class="programlisting">struct GstRTSPMedia;</pre>
-<p>A class that contains the GStreamer element along with a list of
-<a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> objects that can produce data.</p>
-<p>This object is usually created from a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a>.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMediaClass"></a><h3>struct GstRTSPMediaClass</h3>
-<pre class="programlisting">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);
-};
-</pre>
-<p>The RTSP media class</p>
-<div class="refsect3">
-<a name="id-1.2.7.10.3.5"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="struct_members_name">
-<col class="struct_members_description">
-<col width="200px" class="struct_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObjectClass"><span class="type">GObjectClass</span></a> <em class="structfield"><code><a name="GstRTSPMediaClass.parent-class"></a>parent_class</code></em>;</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.handle-message"></a>handle_message</code></em> ()</p></td>
-<td class="struct_member_description"><p>handle a message</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.prepare"></a>prepare</code></em> ()</p></td>
-<td class="struct_member_description"><p>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).</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.unprepare"></a>unprepare</code></em> ()</p></td>
-<td class="struct_member_description"><p>the default implementation sets the pipeline's state
-to GST_STATE_NULL and removes all elements.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.suspend"></a>suspend</code></em> ()</p></td>
-<td class="struct_member_description"><p>the default implementation sets the pipeline's state to
-GST_STATE_NULL GST_STATE_PAUSED depending on the selected
-suspend mode.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.unsuspend"></a>unsuspend</code></em> ()</p></td>
-<td class="struct_member_description"><p>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.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.convert-range"></a>convert_range</code></em> ()</p></td>
-<td class="struct_member_description"><p>convert a range to the given unit</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.query-position"></a>query_position</code></em> ()</p></td>
-<td class="struct_member_description"><p>query the current position in the pipeline</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.query-stop"></a>query_stop</code></em> ()</p></td>
-<td class="struct_member_description"><p>query when playback will stop</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.create-rtpbin"></a>create_rtpbin</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.setup-rtpbin"></a>setup_rtpbin</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.setup-sdp"></a>setup_sdp</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.new-stream"></a>new_stream</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.removed-stream"></a>removed_stream</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.prepared"></a>prepared</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.unprepared"></a>unprepared</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.target-state"></a>target_state</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaClass.new-state"></a>new_state</code></em> ()</p></td>
-<td> </td>
-<td> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMediaStatus"></a><h3>enum GstRTSPMediaStatus</h3>
-<p>The state of the media pipeline.</p>
-<div class="refsect3">
-<a name="id-1.2.7.10.4.4"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="enum_members_name">
-<col class="enum_members_description">
-<col width="200px" class="enum_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-MEDIA-STATUS-UNPREPARED:CAPS"></a>GST_RTSP_MEDIA_STATUS_UNPREPARED</p></td>
-<td class="enum_member_description">
-<p>media pipeline not prerolled</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-MEDIA-STATUS-UNPREPARING:CAPS"></a>GST_RTSP_MEDIA_STATUS_UNPREPARING</p></td>
-<td class="enum_member_description">
-<p>media pipeline is busy doing a clean
- shutdown.</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-MEDIA-STATUS-PREPARING:CAPS"></a>GST_RTSP_MEDIA_STATUS_PREPARING</p></td>
-<td class="enum_member_description">
-<p>media pipeline is prerolling</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-MEDIA-STATUS-PREPARED:CAPS"></a>GST_RTSP_MEDIA_STATUS_PREPARED</p></td>
-<td class="enum_member_description">
-<p>media pipeline is prerolled</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-MEDIA-STATUS-SUSPENDED:CAPS"></a>GST_RTSP_MEDIA_STATUS_SUSPENDED</p></td>
-<td class="enum_member_description">
-<p>media is suspended</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-MEDIA-STATUS-ERROR:CAPS"></a>GST_RTSP_MEDIA_STATUS_ERROR</p></td>
-<td class="enum_member_description">
-<p>media pipeline is in error</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPSuspendMode"></a><h3>enum GstRTSPSuspendMode</h3>
-<p>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.</p>
-<div class="refsect3">
-<a name="id-1.2.7.10.5.4"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="enum_members_name">
-<col class="enum_members_description">
-<col width="200px" class="enum_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-SUSPEND-MODE-NONE:CAPS"></a>GST_RTSP_SUSPEND_MODE_NONE</p></td>
-<td class="enum_member_description">
-<p>Media is not suspended</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-SUSPEND-MODE-PAUSE:CAPS"></a>GST_RTSP_SUSPEND_MODE_PAUSE</p></td>
-<td class="enum_member_description">
-<p>Media is PAUSED in suspend</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-SUSPEND-MODE-RESET:CAPS"></a>GST_RTSP_SUSPEND_MODE_RESET</p></td>
-<td class="enum_member_description">
-<p>The media is set to NULL when suspended</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMedia.property-details"></a><h2>Property Details</h2>
-<div class="refsect2">
-<a name="GstRTSPMedia--buffer-size"></a><h3>The <code class="literal">“buffer-size”</code> property</h3>
-<pre class="programlisting"> “buffer-size” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a></pre>
-<p>The kernel UDP buffer size to use.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: 524288</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMedia--element"></a><h3>The <code class="literal">“element”</code> property</h3>
-<pre class="programlisting"> “element” <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> *</pre>
-<p>The GstBin to use for streaming the media.</p>
-<p>Flags: Read / Write / Construct Only</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMedia--eos-shutdown"></a><h3>The <code class="literal">“eos-shutdown”</code> property</h3>
-<pre class="programlisting"> “eos-shutdown” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></pre>
-<p>Send an EOS event to the pipeline before unpreparing.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: FALSE</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMedia--profiles"></a><h3>The <code class="literal">“profiles”</code> property</h3>
-<pre class="programlisting"> “profiles” <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a></pre>
-<p>Allowed transfer profiles.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: GST_RTSP_PROFILE_AVP</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMedia--protocols"></a><h3>The <code class="literal">“protocols”</code> property</h3>
-<pre class="programlisting"> “protocols” <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a></pre>
-<p>Allowed lower transport protocols.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMedia--reusable"></a><h3>The <code class="literal">“reusable”</code> property</h3>
-<pre class="programlisting"> “reusable” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></pre>
-<p>If this media pipeline can be reused after an unprepare.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: FALSE</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMedia--shared"></a><h3>The <code class="literal">“shared”</code> property</h3>
-<pre class="programlisting"> “shared” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></pre>
-<p>If this media pipeline can be shared.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: FALSE</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMedia--suspend-mode"></a><h3>The <code class="literal">“suspend-mode”</code> property</h3>
-<pre class="programlisting"> “suspend-mode” <a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a></pre>
-<p>How to suspend the media in PAUSED.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: GST_RTSP_SUSPEND_MODE_NONE</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMedia--time-provider"></a><h3>The <code class="literal">“time-provider”</code> property</h3>
-<pre class="programlisting"> “time-provider” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></pre>
-<p>Use a NetTimeProvider for clients.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: FALSE</p>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMedia.signal-details"></a><h2>Signal Details</h2>
-<div class="refsect2">
-<a name="GstRTSPMedia-new-state"></a><h3>The <code class="literal">“new-state”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *gstrtspmedia,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMedia-new-stream"></a><h3>The <code class="literal">“new-stream”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *gstrtspmedia,
- <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMedia-prepared"></a><h3>The <code class="literal">“prepared”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *gstrtspmedia,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMedia-removed-stream"></a><h3>The <code class="literal">“removed-stream”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *gstrtspmedia,
- <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMedia-target-state"></a><h3>The <code class="literal">“target-state”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *gstrtspmedia,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMedia-unprepared"></a><h3>The <code class="literal">“unprepared”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *gstrtspmedia,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMedia.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a>, <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a>, <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a>,
- <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p>
-</div>
-</div>
-<div class="footer">
-<hr>
- Generated by GTK-Doc V1.21</div>
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-<a name="GstRTSPMediaFactory"></a><div class="titlepage"></div>
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-<td valign="top">
-<h2><span class="refentrytitle"><a name="GstRTSPMediaFactory.top_of_page"></a>GstRTSPMediaFactory</span></h2>
-<p>GstRTSPMediaFactory — A factory for media pipelines</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="GstRTSPMediaFactory.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="returnvalue">GstRTSPMediaFactory</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-new" title="gst_rtsp_media_factory_new ()">gst_rtsp_media_factory_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-launch" title="gst_rtsp_media_factory_get_launch ()">gst_rtsp_media_factory_get_launch</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-launch" title="gst_rtsp_media_factory_set_launch ()">gst_rtsp_media_factory_set_launch</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="returnvalue">GstRTSPPermissions</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-permissions" title="gst_rtsp_media_factory_get_permissions ()">gst_rtsp_media_factory_get_permissions</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-permissions" title="gst_rtsp_media_factory_set_permissions ()">gst_rtsp_media_factory_set_permissions</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-add-role" title="gst_rtsp_media_factory_add_role ()">gst_rtsp_media_factory_add_role</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-shared" title="gst_rtsp_media_factory_set_shared ()">gst_rtsp_media_factory_set_shared</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-is-shared" title="gst_rtsp_media_factory_is_shared ()">gst_rtsp_media_factory_is_shared</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-is-eos-shutdown" title="gst_rtsp_media_factory_is_eos_shutdown ()">gst_rtsp_media_factory_is_eos_shutdown</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-eos-shutdown" title="gst_rtsp_media_factory_set_eos_shutdown ()">gst_rtsp_media_factory_set_eos_shutdown</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="returnvalue">GstRTSPLowerTrans</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-protocols" title="gst_rtsp_media_factory_get_protocols ()">gst_rtsp_media_factory_get_protocols</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-protocols" title="gst_rtsp_media_factory_set_protocols ()">gst_rtsp_media_factory_set_protocols</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-profiles" title="gst_rtsp_media_factory_set_profiles ()">gst_rtsp_media_factory_set_profiles</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="returnvalue">GstRTSPProfile</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-profiles" title="gst_rtsp_media_factory_get_profiles ()">gst_rtsp_media_factory_get_profiles</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="returnvalue">GstRTSPAddressPool</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-address-pool" title="gst_rtsp_media_factory_get_address_pool ()">gst_rtsp_media_factory_get_address_pool</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-address-pool" title="gst_rtsp_media_factory_set_address_pool ()">gst_rtsp_media_factory_set_address_pool</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-buffer-size" title="gst_rtsp_media_factory_get_buffer_size ()">gst_rtsp_media_factory_get_buffer_size</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-buffer-size" title="gst_rtsp_media_factory_set_buffer_size ()">gst_rtsp_media_factory_set_buffer_size</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="returnvalue">GstRTSPSuspendMode</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-suspend-mode" title="gst_rtsp_media_factory_get_suspend_mode ()">gst_rtsp_media_factory_get_suspend_mode</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-suspend-mode" title="gst_rtsp_media_factory_set_suspend_mode ()">gst_rtsp_media_factory_set_suspend_mode</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="returnvalue">GstRTSPMedia</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-construct" title="gst_rtsp_media_factory_construct ()">gst_rtsp_media_factory_construct</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="returnvalue">GstElement</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-create-element" title="gst_rtsp_media_factory_create_element ()">gst_rtsp_media_factory_create_element</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMediaFactory.properties"></a><h2>Properties</h2>
-<div class="informaltable"><table border="0">
-<colgroup>
-<col width="150px" class="properties_type">
-<col width="300px" class="properties_name">
-<col width="200px" class="properties_flags">
-</colgroup>
-<tbody>
-<tr>
-<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--buffer-size" title="The “buffer-size” property">buffer-size</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--eos-shutdown" title="The “eos-shutdown” property">eos-shutdown</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *</td>
-<td class="property_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--launch" title="The “launch” property">launch</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--profiles" title="The “profiles” property">profiles</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--protocols" title="The “protocols” property">protocols</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--shared" title="The “shared” property">shared</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type"><a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--suspend-mode" title="The “suspend-mode” property">suspend-mode</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMediaFactory.signals"></a><h2>Signals</h2>
-<div class="informaltable"><table border="0">
-<colgroup>
-<col width="150px" class="signals_return">
-<col width="300px" class="signals_name">
-<col width="200px" class="signals_flags">
-</colgroup>
-<tbody>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory-media-configure" title="The “media-configure” signal">media-configure</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-<tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory-media-constructed" title="The “media-constructed” signal">media-constructed</a></td>
-<td class="signal_flags">Run Last</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMediaFactory.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory-struct" title="struct GstRTSPMediaFactory">GstRTSPMediaFactory</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactoryClass" title="struct GstRTSPMediaFactoryClass">GstRTSPMediaFactoryClass</a></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMediaFactory.object-hierarchy"></a><h2>Object Hierarchy</h2>
-<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
- <span class="lineart">╰──</span> GstRTSPMediaFactory
-</pre>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMediaFactory.description"></a><h2>Description</h2>
-<p>The <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> is responsible for creating or recycling
-<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> objects based on the passed URL.</p>
-<p>The default implementation of the object can create <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> objects
-containing a pipeline created from a launch description set with
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-launch" title="gst_rtsp_media_factory_set_launch ()"><code class="function">gst_rtsp_media_factory_set_launch()</code></a>.</p>
-<p>Media from a factory can be shared by setting the shared flag with
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-shared" title="gst_rtsp_media_factory_set_shared ()"><code class="function">gst_rtsp_media_factory_set_shared()</code></a>. When a factory is shared,
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-construct" title="gst_rtsp_media_factory_construct ()"><code class="function">gst_rtsp_media_factory_construct()</code></a> will return the same <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> when
-the url matches.</p>
-<p>Last reviewed on 2013-07-11 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMediaFactory.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-new"></a><h3>gst_rtsp_media_factory_new ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="returnvalue">GstRTSPMediaFactory</span></a> *
-gst_rtsp_media_factory_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
-<p>Create a new <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> instance.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.2.5"></a><h4>Returns</h4>
-<p> a new <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> object. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-get-launch"></a><h3>gst_rtsp_media_factory_get_launch ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-gst_rtsp_media_factory_get_launch (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
-<p>Get the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstParse.html#gst-parse-launch"><code class="function">gst_parse_launch()</code></a> pipeline description that will be used in the
-default prepare vmethod.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.3.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.5.9.3.6"></a><h4>Returns</h4>
-<p> the configured launch description. <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after
-usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-set-launch"></a><h3>gst_rtsp_media_factory_set_launch ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_factory_set_launch (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *launch</code></em>);</pre>
-<p>The <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstParse.html#gst-parse-launch"><code class="function">gst_parse_launch()</code></a> line to use for constructing the pipeline in the
-default prepare vmethod.</p>
-<p>The pipeline description should return a GstBin as the toplevel element
-which can be accomplished by enclosing the dscription with brackets '('
-')'.</p>
-<p>The description should return a pipeline with payloaders named pay0, pay1,
-etc.. Each of the payloaders will result in a stream.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.4.7"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>launch</p></td>
-<td class="parameter_description"><p>the launch description</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-get-permissions"></a><h3>gst_rtsp_media_factory_get_permissions ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="returnvalue">GstRTSPPermissions</span></a> *
-gst_rtsp_media_factory_get_permissions
- (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
-<p>Get the permissions object from <em class="parameter"><code>factory</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.5.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.5.9.5.6"></a><h4>Returns</h4>
-<p> a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> object, unref after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-set-permissions"></a><h3>gst_rtsp_media_factory_set_permissions ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_factory_set_permissions
- (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>);</pre>
-<p>Set <em class="parameter"><code>permissions</code></em>
- on <em class="parameter"><code>factory</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.6.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>permissions</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-add-role"></a><h3>gst_rtsp_media_factory_add_role ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_factory_add_role (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *role</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *fieldname</code></em>,
- <em class="parameter"><code>...</code></em>);</pre>
-<p>A convenience method to add <em class="parameter"><code>role</code></em>
- with <em class="parameter"><code>fieldname</code></em>
- and additional arguments to
-the permissions of <em class="parameter"><code>factory</code></em>
-. If <em class="parameter"><code>factory</code></em>
- had no permissions, new permissions
-will be created and the role will be added to it.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.7.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>role</p></td>
-<td class="parameter_description"><p>a role</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>fieldname</p></td>
-<td class="parameter_description"><p>the first field name</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>...</p></td>
-<td class="parameter_description"><p>additional arguments</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-set-shared"></a><h3>gst_rtsp_media_factory_set_shared ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_factory_set_shared (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> shared</code></em>);</pre>
-<p>Configure if media created from this factory can be shared between clients.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.8.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>shared</p></td>
-<td class="parameter_description"><p>the new value</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-is-shared"></a><h3>gst_rtsp_media_factory_is_shared ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_media_factory_is_shared (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
-<p>Get if media created from this factory can be shared between clients.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.9.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.5.9.9.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the media will be shared between clients.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-is-eos-shutdown"></a><h3>gst_rtsp_media_factory_is_eos_shutdown ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_media_factory_is_eos_shutdown
- (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
-<p>Get if media created from this factory will have an EOS event sent to the
-pipeline before shutdown.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.10.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.5.9.10.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the media will receive EOS before shutdown.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-set-eos-shutdown"></a><h3>gst_rtsp_media_factory_set_eos_shutdown ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_factory_set_eos_shutdown
- (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> eos_shutdown</code></em>);</pre>
-<p>Configure if media created from this factory will have an EOS sent to the
-pipeline before shutdown.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.11.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>eos_shutdown</p></td>
-<td class="parameter_description"><p>the new value</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-get-protocols"></a><h3>gst_rtsp_media_factory_get_protocols ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="returnvalue">GstRTSPLowerTrans</span></a>
-gst_rtsp_media_factory_get_protocols (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
-<p>Get the allowed protocols of <em class="parameter"><code>factory</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.12.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.5.9.12.6"></a><h4>Returns</h4>
-<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a></p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-set-protocols"></a><h3>gst_rtsp_media_factory_set_protocols ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_factory_set_protocols (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a> protocols</code></em>);</pre>
-<p>Configure the allowed lower transport for <em class="parameter"><code>factory</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.13.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>protocols</p></td>
-<td class="parameter_description"><p>the new flags</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-set-profiles"></a><h3>gst_rtsp_media_factory_set_profiles ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_factory_set_profiles (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a> profiles</code></em>);</pre>
-<p>Configure the allowed profiles for <em class="parameter"><code>factory</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.14.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>profiles</p></td>
-<td class="parameter_description"><p>the new flags</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-get-profiles"></a><h3>gst_rtsp_media_factory_get_profiles ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="returnvalue">GstRTSPProfile</span></a>
-gst_rtsp_media_factory_get_profiles (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
-<p>Get the allowed profiles of <em class="parameter"><code>factory</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.15.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.5.9.15.6"></a><h4>Returns</h4>
-<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a></p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-get-address-pool"></a><h3>gst_rtsp_media_factory_get_address_pool ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="returnvalue">GstRTSPAddressPool</span></a> *
-gst_rtsp_media_factory_get_address_pool
- (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
-<p>Get the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> used as the address pool of <em class="parameter"><code>factory</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.16.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.5.9.16.6"></a><h4>Returns</h4>
-<p> the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> of <em class="parameter"><code>factory</code></em>
-. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
-usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-set-address-pool"></a><h3>gst_rtsp_media_factory_set_address_pool ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_factory_set_address_pool
- (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>);</pre>
-<p>configure <em class="parameter"><code>pool</code></em>
- to be used as the address pool of <em class="parameter"><code>factory</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.17.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-get-buffer-size"></a><h3>gst_rtsp_media_factory_get_buffer_size ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-gst_rtsp_media_factory_get_buffer_size
- (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
-<p>Get the kernel UDP buffer size.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.18.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.5.9.18.6"></a><h4>Returns</h4>
-<p> the kernel UDP buffer size.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-set-buffer-size"></a><h3>gst_rtsp_media_factory_set_buffer_size ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_factory_set_buffer_size
- (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> size</code></em>);</pre>
-<p>Set the kernel UDP buffer size.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.19.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>size</p></td>
-<td class="parameter_description"><p>the new value</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-get-suspend-mode"></a><h3>gst_rtsp_media_factory_get_suspend_mode ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="returnvalue">GstRTSPSuspendMode</span></a>
-gst_rtsp_media_factory_get_suspend_mode
- (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
-<p>Get how media created from this factory will be suspended.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.20.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.5.9.20.6"></a><h4>Returns</h4>
-<p> a <a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a>.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-set-suspend-mode"></a><h3>gst_rtsp_media_factory_set_suspend_mode ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_factory_set_suspend_mode
- (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a> mode</code></em>);</pre>
-<p>Configure how media created from this factory will be suspended.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.21.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>mode</p></td>
-<td class="parameter_description"><p>the new <a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-construct"></a><h3>gst_rtsp_media_factory_construct ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="returnvalue">GstRTSPMedia</span></a> *
-gst_rtsp_media_factory_construct (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
- <em class="parameter"><code>const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="type">GstRTSPUrl</span></a> *url</code></em>);</pre>
-<p>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.</p>
-<p>One or more GstRTSPStream objects should be created from the result
-with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-create-stream" title="gst_rtsp_media_create_stream ()"><code class="function">gst_rtsp_media_create_stream()</code></a>.</p>
-<p>After the media is constructed, it can be configured and then prepared
-with <a class="link" href="GstRTSPMedia.html#gst-rtsp-media-prepare" title="gst_rtsp_media_prepare ()"><code class="function">gst_rtsp_media_prepare()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.22.7"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>url</p></td>
-<td class="parameter_description"><p>the url used</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.5.9.22.8"></a><h4>Returns</h4>
-<p> a new <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> if the media could be prepared. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-create-element"></a><h3>gst_rtsp_media_factory_create_element ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="returnvalue">GstElement</span></a> *
-gst_rtsp_media_factory_create_element (<em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>,
- <em class="parameter"><code>const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="type">GstRTSPUrl</span></a> *url</code></em>);</pre>
-<p>Construct and return a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> that is a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBin.html"><span class="type">GstBin</span></a> containing
-the elements to use for streaming the media.</p>
-<p>The bin should contain payloaders pay%d for each stream. The default
-implementation of this function returns the bin created from the
-launch parameter.</p>
-<div class="refsect3">
-<a name="id-1.2.5.9.23.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>url</p></td>
-<td class="parameter_description"><p>the url used</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.5.9.23.7"></a><h4>Returns</h4>
-<p> a new <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a>. </p>
-<p><span class="annotation">[<acronym title="Alias for transfer none, used for objects with floating refs."><span class="acronym">transfer floating</span></acronym>]</span></p>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMediaFactory.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstRTSPMediaFactory-struct"></a><h3>struct GstRTSPMediaFactory</h3>
-<pre class="programlisting">struct GstRTSPMediaFactory;</pre>
-<p>The definition and logic for constructing the pipeline for a media. The media
-can contain multiple streams like audio and video.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMediaFactoryClass"></a><h3>struct GstRTSPMediaFactoryClass</h3>
-<pre class="programlisting">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);
-};
-</pre>
-<p>The <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> class structure.</p>
-<div class="refsect3">
-<a name="id-1.2.5.10.3.5"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="struct_members_name">
-<col class="struct_members_description">
-<col width="200px" class="struct_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObjectClass"><span class="type">GObjectClass</span></a> <em class="structfield"><code><a name="GstRTSPMediaFactoryClass.parent-class"></a>parent_class</code></em>;</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaFactoryClass.gen-key"></a>gen_key</code></em> ()</p></td>
-<td class="struct_member_description"><p>convert <em class="parameter"><code>url</code></em>
-to a key for caching shared <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> objects.
-The default implementation of this function will use the complete URL
-including the query parameters to return a key.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaFactoryClass.create-element"></a>create_element</code></em> ()</p></td>
-<td class="struct_member_description"><p>Construct and return a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> that is a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBin.html"><span class="type">GstBin</span></a> 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.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaFactoryClass.construct"></a>construct</code></em> ()</p></td>
-<td class="struct_member_description"><p>the vmethod that will be called when the factory has to create the
-<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> for <em class="parameter"><code>url</code></em>
-. The default implementation of this
-function calls create_element to retrieve an element and then looks for
-pay%d to create the streams.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaFactoryClass.create-pipeline"></a>create_pipeline</code></em> ()</p></td>
-<td class="struct_member_description"><p>create a new pipeline or re-use an existing one and
-add the <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a>'s element created by <em class="parameter"><code>construct</code></em>
-to the pipeline.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaFactoryClass.configure"></a>configure</code></em> ()</p></td>
-<td class="struct_member_description"><p>configure the media created with <em class="parameter"><code>construct</code></em>
-. The default
-implementation will configure the 'shared' property of the media.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaFactoryClass.media-constructed"></a>media_constructed</code></em> ()</p></td>
-<td class="struct_member_description"><p>signal emited when a media was constructed</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMediaFactoryClass.media-configure"></a>media_configure</code></em> ()</p></td>
-<td class="struct_member_description"><p>signal emited when a media should be configured</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMediaFactory.property-details"></a><h2>Property Details</h2>
-<div class="refsect2">
-<a name="GstRTSPMediaFactory--buffer-size"></a><h3>The <code class="literal">“buffer-size”</code> property</h3>
-<pre class="programlisting"> “buffer-size” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a></pre>
-<p>The kernel UDP buffer size to use.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: 524288</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMediaFactory--eos-shutdown"></a><h3>The <code class="literal">“eos-shutdown”</code> property</h3>
-<pre class="programlisting"> “eos-shutdown” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></pre>
-<p>Send EOS down the pipeline before shutting down.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: FALSE</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMediaFactory--launch"></a><h3>The <code class="literal">“launch”</code> property</h3>
-<pre class="programlisting"> “launch” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *</pre>
-<p>A launch description of the pipeline.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: NULL</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMediaFactory--profiles"></a><h3>The <code class="literal">“profiles”</code> property</h3>
-<pre class="programlisting"> “profiles” <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a></pre>
-<p>Allowed transfer profiles.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: GST_RTSP_PROFILE_AVP</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMediaFactory--protocols"></a><h3>The <code class="literal">“protocols”</code> property</h3>
-<pre class="programlisting"> “protocols” <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a></pre>
-<p>Allowed lower transport protocols.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMediaFactory--shared"></a><h3>The <code class="literal">“shared”</code> property</h3>
-<pre class="programlisting"> “shared” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></pre>
-<p>If media from this factory is shared.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: FALSE</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMediaFactory--suspend-mode"></a><h3>The <code class="literal">“suspend-mode”</code> property</h3>
-<pre class="programlisting"> “suspend-mode” <a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode"><span class="type">GstRTSPSuspendMode</span></a></pre>
-<p>Control how media will be suspended.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: GST_RTSP_SUSPEND_MODE_NONE</p>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMediaFactory.signal-details"></a><h2>Signal Details</h2>
-<div class="refsect2">
-<a name="GstRTSPMediaFactory-media-configure"></a><h3>The <code class="literal">“media-configure”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *gstrtspmediafactory,
- <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMediaFactory-media-constructed"></a><h3>The <code class="literal">“media-constructed”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *gstrtspmediafactory,
- <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMediaFactory.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a>, <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p>
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-<h2><span class="refentrytitle"><a name="GstRTSPMountPoints.top_of_page"></a>GstRTSPMountPoints</span></h2>
-<p>GstRTSPMountPoints — Map a path to media</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="GstRTSPMountPoints.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="returnvalue">GstRTSPMountPoints</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-new" title="gst_rtsp_mount_points_new ()">gst_rtsp_mount_points_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-add-factory" title="gst_rtsp_mount_points_add_factory ()">gst_rtsp_mount_points_add_factory</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-remove-factory" title="gst_rtsp_mount_points_remove_factory ()">gst_rtsp_mount_points_remove_factory</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="returnvalue">GstRTSPMediaFactory</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-match" title="gst_rtsp_mount_points_match ()">gst_rtsp_mount_points_match</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-make-path" title="gst_rtsp_mount_points_make_path ()">gst_rtsp_mount_points_make_path</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMountPoints.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPMountPoints.html#GstRTSPMountPoints-struct" title="struct GstRTSPMountPoints">GstRTSPMountPoints</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPMountPoints.html#GstRTSPMountPointsClass" title="struct GstRTSPMountPointsClass">GstRTSPMountPointsClass</a></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMountPoints.object-hierarchy"></a><h2>Object Hierarchy</h2>
-<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
- <span class="lineart">╰──</span> GstRTSPMountPoints
-</pre>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMountPoints.description"></a><h2>Description</h2>
-<p>A <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> object maintains a relation between paths
-and <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> objects. This object is usually given to
-<a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> and used to find the media attached to a path.</p>
-<p>With <a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-add-factory" title="gst_rtsp_mount_points_add_factory ()"><code class="function">gst_rtsp_mount_points_add_factory()</code></a> and
-<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-remove-factory" title="gst_rtsp_mount_points_remove_factory ()"><code class="function">gst_rtsp_mount_points_remove_factory()</code></a>, factories can be added and
-removed.</p>
-<p>With <a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-match" title="gst_rtsp_mount_points_match ()"><code class="function">gst_rtsp_mount_points_match()</code></a> you can find the <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a>
-object that completely matches the given path.</p>
-<p>Last reviewed on 2013-07-11 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMountPoints.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-mount-points-new"></a><h3>gst_rtsp_mount_points_new ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="returnvalue">GstRTSPMountPoints</span></a> *
-gst_rtsp_mount_points_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
-<p>Make a new mount points object.</p>
-<div class="refsect3">
-<a name="id-1.2.4.7.2.5"></a><h4>Returns</h4>
-<p> a new <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a>. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-mount-points-add-factory"></a><h3>gst_rtsp_mount_points_add_factory ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_mount_points_add_factory (<em class="parameter"><code><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *mounts</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *path</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *factory</code></em>);</pre>
-<p>Attach <em class="parameter"><code>factory</code></em>
- to the mount point <em class="parameter"><code>path</code></em>
- in <em class="parameter"><code>mounts</code></em>
-.</p>
-<p><em class="parameter"><code>path</code></em>
- is of the form (/node)+. Any previous mount point will be freed.</p>
-<p>Ownership is taken of the reference on <em class="parameter"><code>factory</code></em>
- so that <em class="parameter"><code>factory</code></em>
- should not be
-used after calling this function.</p>
-<div class="refsect3">
-<a name="id-1.2.4.7.3.7"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>mounts</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>path</p></td>
-<td class="parameter_description"><p>a mount point</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p> a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-mount-points-remove-factory"></a><h3>gst_rtsp_mount_points_remove_factory ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_mount_points_remove_factory (<em class="parameter"><code><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *mounts</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *path</code></em>);</pre>
-<p>Remove the <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> associated with <em class="parameter"><code>path</code></em>
- in <em class="parameter"><code>mounts</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.4.7.4.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>mounts</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>path</p></td>
-<td class="parameter_description"><p>a mount point</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-mount-points-match"></a><h3>gst_rtsp_mount_points_match ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="returnvalue">GstRTSPMediaFactory</span></a> *
-gst_rtsp_mount_points_match (<em class="parameter"><code><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *mounts</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *path</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> *matched</code></em>);</pre>
-<p>Find the factory in <em class="parameter"><code>mounts</code></em>
- that has the longest match with <em class="parameter"><code>path</code></em>
-.</p>
-<p>If <em class="parameter"><code>matched</code></em>
- is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, <em class="parameter"><code>path</code></em>
- will match the factory exactly otherwise
-the amount of characters that matched is returned in <em class="parameter"><code>matched</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.4.7.5.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>mounts</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>path</p></td>
-<td class="parameter_description"><p>a mount point</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>matched</p></td>
-<td class="parameter_description"><p> the amount of <em class="parameter"><code>path</code></em>
-matched. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.4.7.5.7"></a><h4>Returns</h4>
-<p> the <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> for <em class="parameter"><code>path</code></em>
-.
-<a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-mount-points-make-path"></a><h3>gst_rtsp_mount_points_make_path ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-gst_rtsp_mount_points_make_path (<em class="parameter"><code><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *mounts</code></em>,
- <em class="parameter"><code>const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="type">GstRTSPUrl</span></a> *url</code></em>);</pre>
-<p>Make a path string from <em class="parameter"><code>url</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.4.7.6.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>mounts</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>url</p></td>
-<td class="parameter_description"><p>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="type">GstRTSPUrl</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.4.7.6.6"></a><h4>Returns</h4>
-<p> a path string for <em class="parameter"><code>url</code></em>
-, <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMountPoints.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstRTSPMountPoints-struct"></a><h3>struct GstRTSPMountPoints</h3>
-<pre class="programlisting">struct GstRTSPMountPoints;</pre>
-<p>Creates a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> object for a given url.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMountPointsClass"></a><h3>struct GstRTSPMountPointsClass</h3>
-<pre class="programlisting">struct GstRTSPMountPointsClass {
- GObjectClass parent_class;
-
- gchar * (*make_path) (GstRTSPMountPoints *mounts,
- const GstRTSPUrl *url);
-};
-</pre>
-<p>The class for the media mounts object.</p>
-<div class="refsect3">
-<a name="id-1.2.4.8.3.5"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="struct_members_name">
-<col class="struct_members_description">
-<col width="200px" class="struct_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObjectClass"><span class="type">GObjectClass</span></a> <em class="structfield"><code><a name="GstRTSPMountPointsClass.parent-class"></a>parent_class</code></em>;</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPMountPointsClass.make-path"></a>make_path</code></em> ()</p></td>
-<td class="struct_member_description"><p>make a path from the given url.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPMountPoints.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a>, <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p>
-</div>
-</div>
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-<td valign="top">
-<h2><span class="refentrytitle"><a name="GstRTSPServer.top_of_page"></a>GstRTSPServer</span></h2>
-<p>GstRTSPServer — The main server object</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="GstRTSPServer.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="returnvalue">GstRTSPServer</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-new" title="gst_rtsp_server_new ()">gst_rtsp_server_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-address" title="gst_rtsp_server_get_address ()">gst_rtsp_server_get_address</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-address" title="gst_rtsp_server_set_address ()">gst_rtsp_server_set_address</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-service" title="gst_rtsp_server_get_service ()">gst_rtsp_server_get_service</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-service" title="gst_rtsp_server_set_service ()">gst_rtsp_server_set_service</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="returnvalue">gint</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-backlog" title="gst_rtsp_server_get_backlog ()">gst_rtsp_server_get_backlog</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-backlog" title="gst_rtsp_server_set_backlog ()">gst_rtsp_server_set_backlog</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">int</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-bound-port" title="gst_rtsp_server_get_bound_port ()">gst_rtsp_server_get_bound_port</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="returnvalue">GstRTSPMountPoints</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-mount-points" title="gst_rtsp_server_get_mount_points ()">gst_rtsp_server_get_mount_points</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-mount-points" title="gst_rtsp_server_set_mount_points ()">gst_rtsp_server_set_mount_points</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="returnvalue">GstRTSPSessionPool</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-session-pool" title="gst_rtsp_server_get_session_pool ()">gst_rtsp_server_get_session_pool</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-session-pool" title="gst_rtsp_server_set_session_pool ()">gst_rtsp_server_set_session_pool</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="returnvalue">GstRTSPThreadPool</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-thread-pool" title="gst_rtsp_server_get_thread_pool ()">gst_rtsp_server_get_thread_pool</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-thread-pool" title="gst_rtsp_server_set_thread_pool ()">gst_rtsp_server_set_thread_pool</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="returnvalue">GstRTSPAuth</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-auth" title="gst_rtsp_server_get_auth ()">gst_rtsp_server_get_auth</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-auth" title="gst_rtsp_server_set_auth ()">gst_rtsp_server_set_auth</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-transfer-connection" title="gst_rtsp_server_transfer_connection ()">gst_rtsp_server_transfer_connection</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-io-func" title="gst_rtsp_server_io_func ()">gst_rtsp_server_io_func</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="returnvalue">GSocket</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-create-socket" title="gst_rtsp_server_create_socket ()">gst_rtsp_server_create_socket</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="returnvalue">GSource</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-create-source" title="gst_rtsp_server_create_source ()">gst_rtsp_server_create_source</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-attach" title="gst_rtsp_server_attach ()">gst_rtsp_server_attach</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
-</td>
-<td class="function_name">
-<span class="c_punctuation">(</span><a class="link" href="GstRTSPServer.html#GstRTSPServerClientFilterFunc" title="GstRTSPServerClientFilterFunc ()">*GstRTSPServerClientFilterFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-client-filter" title="gst_rtsp_server_client_filter ()">gst_rtsp_server_client_filter</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPServer.properties"></a><h2>Properties</h2>
-<div class="informaltable"><table border="0">
-<colgroup>
-<col width="150px" class="properties_type">
-<col width="300px" class="properties_name">
-<col width="200px" class="properties_flags">
-</colgroup>
-<tbody>
-<tr>
-<td class="property_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *</td>
-<td class="property_name"><a class="link" href="GstRTSPServer.html#GstRTSPServer--address" title="The “address” property">address</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPServer.html#GstRTSPServer--backlog" title="The “backlog” property">backlog</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPServer.html#GstRTSPServer--bound-port" title="The “bound-port” property">bound-port</a></td>
-<td class="property_flags">Read</td>
-</tr>
-<tr>
-<td class="property_type">
-<a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *</td>
-<td class="property_name"><a class="link" href="GstRTSPServer.html#GstRTSPServer--mount-points" title="The “mount-points” property">mount-points</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *</td>
-<td class="property_name"><a class="link" href="GstRTSPServer.html#GstRTSPServer--service" title="The “service” property">service</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type">
-<a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *</td>
-<td class="property_name"><a class="link" href="GstRTSPServer.html#GstRTSPServer--session-pool" title="The “session-pool” property">session-pool</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPServer.signals"></a><h2>Signals</h2>
-<div class="informaltable"><table border="0">
-<colgroup>
-<col width="150px" class="signals_return">
-<col width="300px" class="signals_name">
-<col width="200px" class="signals_flags">
-</colgroup>
-<tbody><tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPServer.html#GstRTSPServer-client-connected" title="The “client-connected” signal">client-connected</a></td>
-<td class="signal_flags">Run Last</td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPServer.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPServer.html#GstRTSPServer-struct" title="struct GstRTSPServer">GstRTSPServer</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPServer.html#GstRTSPServerClass" title="struct GstRTSPServerClass">GstRTSPServerClass</a></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPServer.object-hierarchy"></a><h2>Object Hierarchy</h2>
-<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
- <span class="lineart">╰──</span> GstRTSPServer
-</pre>
-</div>
-<div class="refsect1">
-<a name="GstRTSPServer.description"></a><h2>Description</h2>
-<p>The server object is the object listening for connections on a port and
-creating <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> objects to handle those connections.</p>
-<p>The server will listen on the address set with <a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-address" title="gst_rtsp_server_set_address ()"><code class="function">gst_rtsp_server_set_address()</code></a>
-and the port or service configured with <a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-service" title="gst_rtsp_server_set_service ()"><code class="function">gst_rtsp_server_set_service()</code></a>.
-Use <a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-backlog" title="gst_rtsp_server_set_backlog ()"><code class="function">gst_rtsp_server_set_backlog()</code></a> 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.</p>
-<p>The server will require an SSL connection when a TLS certificate has been
-set in the auth object with <a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-set-tls-certificate" title="gst_rtsp_auth_set_tls_certificate ()"><code class="function">gst_rtsp_auth_set_tls_certificate()</code></a>.</p>
-<p>To start the server, use <a class="link" href="GstRTSPServer.html#gst-rtsp-server-attach" title="gst_rtsp_server_attach ()"><code class="function">gst_rtsp_server_attach()</code></a> to attach it to a
-<a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainContext"><span class="type">GMainContext</span></a>. For more control, <a class="link" href="GstRTSPServer.html#gst-rtsp-server-create-source" title="gst_rtsp_server_create_source ()"><code class="function">gst_rtsp_server_create_source()</code></a> and
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-create-socket" title="gst_rtsp_server_create_socket ()"><code class="function">gst_rtsp_server_create_socket()</code></a> can be used to get a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="type">GSource</span></a> and <a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="type">GSocket</span></a>
-respectively.</p>
-<p>gst_rtsp_server_transfer_connection() can be used to transfer an existing
-socket to the RTSP server, for example from an HTTP server.</p>
-<p>Once the server socket is attached to a mainloop, it will start accepting
-connections. When a new connection is received, a new <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> object
-is created to handle the connection. The new client will be configured with
-the server <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a>, <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a>, <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> and
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a>.</p>
-<p>The server uses the configured <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> object to handle the
-remainder of the communication with this client.</p>
-<p>Last reviewed on 2013-07-11 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="GstRTSPServer.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-server-new"></a><h3>gst_rtsp_server_new ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="returnvalue">GstRTSPServer</span></a> *
-gst_rtsp_server_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
-<p>Create a new <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> instance.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.2.5"></a><h4>Returns</h4>
-<p> a new <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a>. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-get-address"></a><h3>gst_rtsp_server_get_address ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-gst_rtsp_server_get_address (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
-<p>Get the address on which the server will accept connections.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.3.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.1.9.3.6"></a><h4>Returns</h4>
-<p> the server address. <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-set-address"></a><h3>gst_rtsp_server_set_address ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_server_set_address (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *address</code></em>);</pre>
-<p>Configure <em class="parameter"><code>server</code></em>
- to accept connections on the given address.</p>
-<p>This function must be called before the server is bound.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.4.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>address</p></td>
-<td class="parameter_description"><p>the address</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-get-service"></a><h3>gst_rtsp_server_get_service ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-gst_rtsp_server_get_service (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
-<p>Get the service on which the server will accept connections.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.5.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.1.9.5.6"></a><h4>Returns</h4>
-<p> the service. use <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-set-service"></a><h3>gst_rtsp_server_set_service ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_server_set_service (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *service</code></em>);</pre>
-<p>Configure <em class="parameter"><code>server</code></em>
- to accept connections on the given service.
-<em class="parameter"><code>service</code></em>
- should be a string containing the service name (see services(5)) or
-a string containing a port number between 1 and 65535.</p>
-<p>When <em class="parameter"><code>service</code></em>
- is set to "0", the server will listen on a random free
-port. The actual used port can be retrieved with
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-bound-port" title="gst_rtsp_server_get_bound_port ()"><code class="function">gst_rtsp_server_get_bound_port()</code></a>.</p>
-<p>This function must be called before the server is bound.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.6.7"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>service</p></td>
-<td class="parameter_description"><p>the service</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-get-backlog"></a><h3>gst_rtsp_server_get_backlog ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="returnvalue">gint</span></a>
-gst_rtsp_server_get_backlog (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
-<p>The maximum amount of queued requests for the server.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.7.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.1.9.7.6"></a><h4>Returns</h4>
-<p> the server backlog.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-set-backlog"></a><h3>gst_rtsp_server_set_backlog ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_server_set_backlog (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> backlog</code></em>);</pre>
-<p>configure the maximum amount of requests that may be queued for the
-server.</p>
-<p>This function must be called before the server is bound.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.8.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>backlog</p></td>
-<td class="parameter_description"><p>the backlog</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-get-bound-port"></a><h3>gst_rtsp_server_get_bound_port ()</h3>
-<pre class="programlisting"><span class="returnvalue">int</span>
-gst_rtsp_server_get_bound_port (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
-<p>Get the port number where the server was bound to.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.9.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.1.9.9.6"></a><h4>Returns</h4>
-<p> the port number</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-get-mount-points"></a><h3>gst_rtsp_server_get_mount_points ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="returnvalue">GstRTSPMountPoints</span></a> *
-gst_rtsp_server_get_mount_points (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
-<p>Get the <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> used as the mount points of <em class="parameter"><code>server</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.10.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.1.9.10.6"></a><h4>Returns</h4>
-<p> the <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> of <em class="parameter"><code>server</code></em>
-. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
-usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-set-mount-points"></a><h3>gst_rtsp_server_set_mount_points ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_server_set_mount_points (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *mounts</code></em>);</pre>
-<p>configure <em class="parameter"><code>mounts</code></em>
- to be used as the mount points of <em class="parameter"><code>server</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.11.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>mounts</p></td>
-<td class="parameter_description"><p> a <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-get-session-pool"></a><h3>gst_rtsp_server_get_session_pool ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="returnvalue">GstRTSPSessionPool</span></a> *
-gst_rtsp_server_get_session_pool (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
-<p>Get the <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> used as the session pool of <em class="parameter"><code>server</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.12.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.1.9.12.6"></a><h4>Returns</h4>
-<p> the <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> used for sessions. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
-usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-set-session-pool"></a><h3>gst_rtsp_server_set_session_pool ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_server_set_session_pool (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>);</pre>
-<p>configure <em class="parameter"><code>pool</code></em>
- to be used as the session pool of <em class="parameter"><code>server</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.13.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p> a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-get-thread-pool"></a><h3>gst_rtsp_server_get_thread_pool ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="returnvalue">GstRTSPThreadPool</span></a> *
-gst_rtsp_server_get_thread_pool (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
-<p>Get the <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> used as the thread pool of <em class="parameter"><code>server</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.14.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.1.9.14.6"></a><h4>Returns</h4>
-<p> the <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> of <em class="parameter"><code>server</code></em>
-. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
-usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-set-thread-pool"></a><h3>gst_rtsp_server_set_thread_pool ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_server_set_thread_pool (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> *pool</code></em>);</pre>
-<p>configure <em class="parameter"><code>pool</code></em>
- to be used as the thread pool of <em class="parameter"><code>server</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.15.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-get-auth"></a><h3>gst_rtsp_server_get_auth ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="returnvalue">GstRTSPAuth</span></a> *
-gst_rtsp_server_get_auth (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
-<p>Get the <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> used as the authentication manager of <em class="parameter"><code>server</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.16.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.1.9.16.6"></a><h4>Returns</h4>
-<p> the <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> of <em class="parameter"><code>server</code></em>
-. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
-usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-set-auth"></a><h3>gst_rtsp_server_set_auth ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_server_set_auth (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *auth</code></em>);</pre>
-<p>configure <em class="parameter"><code>auth</code></em>
- to be used as the authentication manager of <em class="parameter"><code>server</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.17.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>auth</p></td>
-<td class="parameter_description"><p> a <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-transfer-connection"></a><h3>gst_rtsp_server_transfer_connection ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_server_transfer_connection (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="type">GSocket</span></a> *socket</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *ip</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> port</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *initial_buffer</code></em>);</pre>
-<p>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 <em class="parameter"><code>initial_buffer</code></em>
- contains any remaining data
-that the HTTP server read from the socket while parsing the HTTP header.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.18.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>socket</p></td>
-<td class="parameter_description"><p> a network socket. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>ip</p></td>
-<td class="parameter_description"><p>the IP address of the remote client</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>port</p></td>
-<td class="parameter_description"><p>the port used by the other end</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>initial_buffer</p></td>
-<td class="parameter_description"><p>any initial data that was already read from the socket</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.1.9.18.6"></a><h4>Returns</h4>
-<p> TRUE if all was ok, FALSE if an error occurred.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-io-func"></a><h3>gst_rtsp_server_io_func ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_server_io_func (<em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="type">GSocket</span></a> *socket</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-IO-Channels.html#GIOCondition"><span class="type">GIOCondition</span></a> condition</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>);</pre>
-<p>A default <a href="https://developer.gnome.org/gio/unstable/GSocket.html#GSocketSourceFunc"><span class="type">GSocketSourceFunc</span></a> that creates a new <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> to accept and handle a
-new connection on <em class="parameter"><code>socket</code></em>
- or <em class="parameter"><code>server</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.19.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>socket</p></td>
-<td class="parameter_description"><p>a <a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="type">GSocket</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>condition</p></td>
-<td class="parameter_description"><p>the condition on <em class="parameter"><code>source</code></em>
-</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p> a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.1.9.19.6"></a><h4>Returns</h4>
-<p> TRUE if the source could be connected, FALSE if an error occurred.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-create-socket"></a><h3>gst_rtsp_server_create_socket ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="returnvalue">GSocket</span></a> *
-gst_rtsp_server_create_socket (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GCancellable.html"><span class="type">GCancellable</span></a> *cancellable</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Error-Reporting.html#GError"><span class="type">GError</span></a> **error</code></em>);</pre>
-<p>Create a <a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="type">GSocket</span></a> for <em class="parameter"><code>server</code></em>
-. The socket will listen on the
-configured service.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.20.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>cancellable</p></td>
-<td class="parameter_description"><p> a <a href="https://developer.gnome.org/gio/unstable/GCancellable.html"><span class="type">GCancellable</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>error</p></td>
-<td class="parameter_description"><p> a <a href="https://developer.gnome.org/glib/unstable/glib-Error-Reporting.html#GError"><span class="type">GError</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.1.9.20.6"></a><h4>Returns</h4>
-<p> the <a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="type">GSocket</span></a> for <em class="parameter"><code>server</code></em>
-or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when an error
-occurred. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-create-source"></a><h3>gst_rtsp_server_create_source ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="returnvalue">GSource</span></a> *
-gst_rtsp_server_create_source (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GCancellable.html"><span class="type">GCancellable</span></a> *cancellable</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Error-Reporting.html#GError"><span class="type">GError</span></a> **error</code></em>);</pre>
-<p>Create a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="type">GSource</span></a> for <em class="parameter"><code>server</code></em>
-. The new source will have a default
-<a href="https://developer.gnome.org/gio/unstable/GSocket.html#GSocketSourceFunc"><span class="type">GSocketSourceFunc</span></a> of <a class="link" href="GstRTSPServer.html#gst-rtsp-server-io-func" title="gst_rtsp_server_io_func ()"><code class="function">gst_rtsp_server_io_func()</code></a>.</p>
-<p><em class="parameter"><code>cancellable</code></em>
- if not <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> 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 <a href="https://developer.gnome.org/gio/unstable/GCancellable.html#g-cancellable-is-cancelled"><code class="function">g_cancellable_is_cancelled()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.21.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>cancellable</p></td>
-<td class="parameter_description"><p> a <a href="https://developer.gnome.org/gio/unstable/GCancellable.html"><span class="type">GCancellable</span></a> or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>error</p></td>
-<td class="parameter_description"><p> a <a href="https://developer.gnome.org/glib/unstable/glib-Error-Reporting.html#GError"><span class="type">GError</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.1.9.21.7"></a><h4>Returns</h4>
-<p> the <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="type">GSource</span></a> for <em class="parameter"><code>server</code></em>
-or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when an error
-occurred. Free with <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#g-source-unref"><code class="function">g_source_unref()</code></a>. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-attach"></a><h3>gst_rtsp_server_attach ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-gst_rtsp_server_attach (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainContext"><span class="type">GMainContext</span></a> *context</code></em>);</pre>
-<p>Attaches <em class="parameter"><code>server</code></em>
- to <em class="parameter"><code>context</code></em>
-. When the mainloop for <em class="parameter"><code>context</code></em>
- is run, the
-server will be dispatched. When <em class="parameter"><code>context</code></em>
- is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, the default context will be
-used).</p>
-<p>This function should be called when the server properties and urls are fully
-configured and the server is ready to start.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.22.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>context</p></td>
-<td class="parameter_description"><p> a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainContext"><span class="type">GMainContext</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.1.9.22.7"></a><h4>Returns</h4>
-<p> the ID (greater than 0) for the source within the GMainContext.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPServerClientFilterFunc"></a><h3>GstRTSPServerClientFilterFunc ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
-<span class="c_punctuation">(</span>*GstRTSPServerClientFilterFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
-<p>This function will be called by the <a class="link" href="GstRTSPServer.html#gst-rtsp-server-client-filter" title="gst_rtsp_server_client_filter ()"><code class="function">gst_rtsp_server_client_filter()</code></a>. An
-implementation should return a value of <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
-<p>When this function returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, <em class="parameter"><code>client</code></em>
- will be removed
-from <em class="parameter"><code>server</code></em>
-.</p>
-<p>A return value of <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a> will leave <em class="parameter"><code>client</code></em>
- untouched in
-<em class="parameter"><code>server</code></em>
-.</p>
-<p>A value of <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a> will add <em class="parameter"><code>client</code></em>
- to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-client-filter" title="gst_rtsp_server_client_filter ()"><code class="function">gst_rtsp_server_client_filter()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.23.8"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> object</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> in <em class="parameter"><code>server</code></em>
-</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p>user data that has been given to <a class="link" href="GstRTSPServer.html#gst-rtsp-server-client-filter" title="gst_rtsp_server_client_filter ()"><code class="function">gst_rtsp_server_client_filter()</code></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.1.9.23.9"></a><h4>Returns</h4>
-<p> a <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-server-client-filter"></a><h3>gst_rtsp_server_client_filter ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
-gst_rtsp_server_client_filter (<em class="parameter"><code><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *server</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPServer.html#GstRTSPServerClientFilterFunc" title="GstRTSPServerClientFilterFunc ()"><span class="type">GstRTSPServerClientFilterFunc</span></a> func</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
-<p>Call <em class="parameter"><code>func</code></em>
- for each client managed by <em class="parameter"><code>server</code></em>
-. The result value of <em class="parameter"><code>func</code></em>
-
-determines what happens to the client. <em class="parameter"><code>func</code></em>
- will be called with <em class="parameter"><code>server</code></em>
-
-locked so no further actions on <em class="parameter"><code>server</code></em>
- can be performed from <em class="parameter"><code>func</code></em>
-.</p>
-<p>If <em class="parameter"><code>func</code></em>
- returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, the client will be removed from
-<em class="parameter"><code>server</code></em>
-.</p>
-<p>If <em class="parameter"><code>func</code></em>
- returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a>, the client will remain in <em class="parameter"><code>server</code></em>
-.</p>
-<p>If <em class="parameter"><code>func</code></em>
- returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>, the client will remain in <em class="parameter"><code>server</code></em>
- but
-will also be added with an additional ref to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of this
-function..</p>
-<p>When <em class="parameter"><code>func</code></em>
- is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a> will be assumed for each client.</p>
-<div class="refsect3">
-<a name="id-1.2.1.9.24.9"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>server</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>func</p></td>
-<td class="parameter_description"><p> a callback. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid only during the call to the method."><span class="acronym">scope call</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p>user data passed to <em class="parameter"><code>func</code></em>
-</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.1.9.24.10"></a><h4>Returns</h4>
-<p> a <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> with all
-clients for which <em class="parameter"><code>func</code></em>
-returned <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>. After usage, each
-element in the <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> should be unreffed before the list is freed. </p>
-<p><span class="annotation">[<acronym title="Generics and defining elements of containers and arrays."><span class="acronym">element-type</span></acronym> GstRTSPClient][<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPServer.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstRTSPServer-struct"></a><h3>struct GstRTSPServer</h3>
-<pre class="programlisting">struct GstRTSPServer;</pre>
-<p>This object listens on a port, creates and manages the clients connected to
-it.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPServerClass"></a><h3>struct GstRTSPServerClass</h3>
-<pre class="programlisting">struct GstRTSPServerClass {
- GObjectClass parent_class;
-
- GstRTSPClient * (*create_client) (GstRTSPServer *server);
-
- /* signals */
- void (*client_connected) (GstRTSPServer *server, GstRTSPClient *client);
-};
-</pre>
-<p>The RTSP server class structure</p>
-<div class="refsect3">
-<a name="id-1.2.1.10.3.5"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="struct_members_name">
-<col class="struct_members_description">
-<col width="200px" class="struct_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObjectClass"><span class="type">GObjectClass</span></a> <em class="structfield"><code><a name="GstRTSPServerClass.parent-class"></a>parent_class</code></em>;</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPServerClass.create-client"></a>create_client</code></em> ()</p></td>
-<td class="struct_member_description"><p>Create, configure a new GstRTSPClient
-object that handles the new connection on <em class="parameter"><code>socket</code></em>
-. The default
-implementation will create a GstRTSPClient and will configure the
-mount-points, auth, session-pool and thread-pool on the client.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPServerClass.client-connected"></a>client_connected</code></em> ()</p></td>
-<td class="struct_member_description"><p>emited when a new client connected.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPServer.property-details"></a><h2>Property Details</h2>
-<div class="refsect2">
-<a name="GstRTSPServer--address"></a><h3>The <code class="literal">“address”</code> property</h3>
-<pre class="programlisting"> “address” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *</pre>
-<p>The address the server uses to listen on.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: "0.0.0.0"</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPServer--backlog"></a><h3>The <code class="literal">“backlog”</code> property</h3>
-<pre class="programlisting"> “backlog” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a></pre>
-<p>The maximum length to which the queue of pending connections may grow.</p>
-<p>Flags: Read / Write</p>
-<p>Allowed values: >= 0</p>
-<p>Default value: 5</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPServer--bound-port"></a><h3>The <code class="literal">“bound-port”</code> property</h3>
-<pre class="programlisting"> “bound-port” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a></pre>
-<p>The port number the server is listening on.</p>
-<p>Flags: Read</p>
-<p>Allowed values: [-1,65535]</p>
-<p>Default value: -1</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPServer--mount-points"></a><h3>The <code class="literal">“mount-points”</code> property</h3>
-<pre class="programlisting"> “mount-points” <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints"><span class="type">GstRTSPMountPoints</span></a> *</pre>
-<p>The mount points to use for client session.</p>
-<p>Flags: Read / Write</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPServer--service"></a><h3>The <code class="literal">“service”</code> property</h3>
-<pre class="programlisting"> “service” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *</pre>
-<p>The service or port number the server uses to listen on.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: "8554"</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPServer--session-pool"></a><h3>The <code class="literal">“session-pool”</code> property</h3>
-<pre class="programlisting"> “session-pool” <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *</pre>
-<p>The session pool to use for client session.</p>
-<p>Flags: Read / Write</p>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPServer.signal-details"></a><h2>Signal Details</h2>
-<div class="refsect2">
-<a name="GstRTSPServer-client-connected"></a><h3>The <code class="literal">“client-connected”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *gstrtspserver,
- <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPServer.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a></p>
-</div>
-</div>
-<div class="footer">
-<hr>
- Generated by GTK-Doc V1.21</div>
-</body>
-</html>
\ No newline at end of file
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-<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
-<html>
-<head>
-<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
-<title>GStreamer RTSP Server Reference Manual: GstRTSPSession</title>
-<meta name="generator" content="DocBook XSL Stylesheets V1.78.1">
-<link rel="home" href="index.html" title="GStreamer RTSP Server Reference Manual">
-<link rel="up" href="ch01.html" title="">
-<link rel="prev" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">
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-<td width="100%" align="left" class="shortcuts">
-<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
- <a href="#GstRTSPSession.description" class="shortcut">Description</a></span><span id="nav_hierarchy"> <span class="dim">|</span>
- <a href="#GstRTSPSession.object-hierarchy" class="shortcut">Object Hierarchy</a></span><span id="nav_properties"> <span class="dim">|</span>
- <a href="#GstRTSPSession.properties" class="shortcut">Properties</a></span>
-</td>
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-</tr></table>
-<div class="refentry">
-<a name="GstRTSPSession"></a><div class="titlepage"></div>
-<div class="refnamediv"><table width="100%"><tr>
-<td valign="top">
-<h2><span class="refentrytitle"><a name="GstRTSPSession.top_of_page"></a>GstRTSPSession</span></h2>
-<p>GstRTSPSession — An object to manage media</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="GstRTSPSession.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="returnvalue">GstRTSPSession</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-new" title="gst_rtsp_session_new ()">gst_rtsp_session_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-sessionid" title="gst_rtsp_session_get_sessionid ()">gst_rtsp_session_get_sessionid</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-header" title="gst_rtsp_session_get_header ()">gst_rtsp_session_get_header</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-set-timeout" title="gst_rtsp_session_set_timeout ()">gst_rtsp_session_set_timeout</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-timeout" title="gst_rtsp_session_get_timeout ()">gst_rtsp_session_get_timeout</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-touch" title="gst_rtsp_session_touch ()">gst_rtsp_session_touch</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-prevent-expire" title="gst_rtsp_session_prevent_expire ()">gst_rtsp_session_prevent_expire</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-allow-expire" title="gst_rtsp_session_allow_expire ()">gst_rtsp_session_allow_expire</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="returnvalue">gint</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-next-timeout" title="gst_rtsp_session_next_timeout ()">gst_rtsp_session_next_timeout</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-is-expired" title="gst_rtsp_session_is_expired ()">gst_rtsp_session_is_expired</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="returnvalue">GstRTSPSessionMedia</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-manage-media" title="gst_rtsp_session_manage_media ()">gst_rtsp_session_manage_media</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-release-media" title="gst_rtsp_session_release_media ()">gst_rtsp_session_release_media</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="returnvalue">GstRTSPSessionMedia</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-media" title="gst_rtsp_session_get_media ()">gst_rtsp_session_get_media</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
-</td>
-<td class="function_name">
-<span class="c_punctuation">(</span><a class="link" href="GstRTSPSession.html#GstRTSPSessionFilterFunc" title="GstRTSPSessionFilterFunc ()">*GstRTSPSessionFilterFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-filter" title="gst_rtsp_session_filter ()">gst_rtsp_session_filter</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSession.properties"></a><h2>Properties</h2>
-<div class="informaltable"><table border="0">
-<colgroup>
-<col width="150px" class="properties_type">
-<col width="300px" class="properties_name">
-<col width="200px" class="properties_flags">
-</colgroup>
-<tbody>
-<tr>
-<td class="property_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *</td>
-<td class="property_name"><a class="link" href="GstRTSPSession.html#GstRTSPSession--sessionid" title="The “sessionid” property">sessionid</a></td>
-<td class="property_flags">Read / Write / Construct Only</td>
-</tr>
-<tr>
-<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPSession.html#GstRTSPSession--timeout" title="The “timeout” property">timeout</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-<tr>
-<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPSession.html#GstRTSPSession--timeout-always-visible" title="The “timeout-always-visible” property">timeout-always-visible</a></td>
-<td class="property_flags">Read / Write</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSession.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPSession.html#GstRTSPSession-struct" title="struct GstRTSPSession">GstRTSPSession</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPSession.html#GstRTSPSessionClass" title="struct GstRTSPSessionClass">GstRTSPSessionClass</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">enum</td>
-<td class="function_name"><a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult">GstRTSPFilterResult</a></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSession.object-hierarchy"></a><h2>Object Hierarchy</h2>
-<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
- <span class="lineart">╰──</span> GstRTSPSession
-</pre>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSession.description"></a><h2>Description</h2>
-<p>The <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> is identified by an id, unique in the
-<a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> that created the session and manages media and its
-configuration.</p>
-<p>A <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> has a timeout that can be retrieved with
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-timeout" title="gst_rtsp_session_get_timeout ()"><code class="function">gst_rtsp_session_get_timeout()</code></a>. You can check if the sessions is expired with
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-is-expired" title="gst_rtsp_session_is_expired ()"><code class="function">gst_rtsp_session_is_expired()</code></a>. <a class="link" href="GstRTSPSession.html#gst-rtsp-session-touch" title="gst_rtsp_session_touch ()"><code class="function">gst_rtsp_session_touch()</code></a> will reset the
-expiration counter of the session.</p>
-<p>When a client configures a media with SETUP, a session will be created to
-keep track of the configuration of that media. With
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-manage-media" title="gst_rtsp_session_manage_media ()"><code class="function">gst_rtsp_session_manage_media()</code></a>, the media is added to the managed media
-in the session. With <a class="link" href="GstRTSPSession.html#gst-rtsp-session-release-media" title="gst_rtsp_session_release_media ()"><code class="function">gst_rtsp_session_release_media()</code></a> the media can be
-released again from the session. Managed media is identified in the sessions
-with a url. Use <a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-media" title="gst_rtsp_session_get_media ()"><code class="function">gst_rtsp_session_get_media()</code></a> to get the media that matches
-(part of) the given url.</p>
-<p>The media in a session can be iterated with <a class="link" href="GstRTSPSession.html#gst-rtsp-session-filter" title="gst_rtsp_session_filter ()"><code class="function">gst_rtsp_session_filter()</code></a>.</p>
-<p>Last reviewed on 2013-07-11 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSession.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-session-new"></a><h3>gst_rtsp_session_new ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="returnvalue">GstRTSPSession</span></a> *
-gst_rtsp_session_new (<em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *sessionid</code></em>);</pre>
-<p>Create a new <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> instance with <em class="parameter"><code>sessionid</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.10.8.2.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>sessionid</p></td>
-<td class="parameter_description"><p>a session id</p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.10.8.2.6"></a><h4>Returns</h4>
-<p> a new <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a>. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-get-sessionid"></a><h3>gst_rtsp_session_get_sessionid ()</h3>
-<pre class="programlisting">const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-gst_rtsp_session_get_sessionid (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>);</pre>
-<p>Get the sessionid of <em class="parameter"><code>session</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.10.8.3.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>session</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.10.8.3.6"></a><h4>Returns</h4>
-<p> the sessionid of <em class="parameter"><code>session</code></em>
-. The value remains valid
-as long as <em class="parameter"><code>session</code></em>
-is alive. </p>
-<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-get-header"></a><h3>gst_rtsp_session_get_header ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-gst_rtsp_session_get_header (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>);</pre>
-<p>Get the string that can be placed in the Session header field.</p>
-<div class="refsect3">
-<a name="id-1.2.10.8.4.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>session</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.10.8.4.6"></a><h4>Returns</h4>
-<p> the Session header of <em class="parameter"><code>session</code></em>
-. <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-set-timeout"></a><h3>gst_rtsp_session_set_timeout ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_session_set_timeout (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> timeout</code></em>);</pre>
-<p>Configure <em class="parameter"><code>session</code></em>
- for a timeout of <em class="parameter"><code>timeout</code></em>
- seconds. The session will be
-cleaned up when there is no activity for <em class="parameter"><code>timeout</code></em>
- seconds.</p>
-<div class="refsect3">
-<a name="id-1.2.10.8.5.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>session</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>timeout</p></td>
-<td class="parameter_description"><p>the new timeout</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-get-timeout"></a><h3>gst_rtsp_session_get_timeout ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-gst_rtsp_session_get_timeout (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>);</pre>
-<p>Get the timeout value of <em class="parameter"><code>session</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.10.8.6.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>session</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.10.8.6.6"></a><h4>Returns</h4>
-<p> the timeout of <em class="parameter"><code>session</code></em>
-in seconds.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-touch"></a><h3>gst_rtsp_session_touch ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_session_touch (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>);</pre>
-<p>Update the last_access time of the session to the current time.</p>
-<div class="refsect3">
-<a name="id-1.2.10.8.7.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>session</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-prevent-expire"></a><h3>gst_rtsp_session_prevent_expire ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_session_prevent_expire (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>);</pre>
-<p>Prevent <em class="parameter"><code>session</code></em>
- from expiring.</p>
-<div class="refsect3">
-<a name="id-1.2.10.8.8.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>session</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-allow-expire"></a><h3>gst_rtsp_session_allow_expire ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_session_allow_expire (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>);</pre>
-<p>Allow <em class="parameter"><code>session</code></em>
- to expire. This method must be called an equal
-amount of time as <a class="link" href="GstRTSPSession.html#gst-rtsp-session-prevent-expire" title="gst_rtsp_session_prevent_expire ()"><code class="function">gst_rtsp_session_prevent_expire()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.10.8.9.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>session</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-next-timeout"></a><h3>gst_rtsp_session_next_timeout ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="returnvalue">gint</span></a>
-gst_rtsp_session_next_timeout (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Date-and-Time-Functions.html#GTimeVal"><span class="type">GTimeVal</span></a> *now</code></em>);</pre>
-<p>Get the amount of milliseconds till the session will expire.</p>
-<div class="refsect3">
-<a name="id-1.2.10.8.10.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>session</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>now</p></td>
-<td class="parameter_description"><p> the current system time. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.10.8.10.6"></a><h4>Returns</h4>
-<p> the amount of milliseconds since the session will time out.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-is-expired"></a><h3>gst_rtsp_session_is_expired ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_session_is_expired (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Date-and-Time-Functions.html#GTimeVal"><span class="type">GTimeVal</span></a> *now</code></em>);</pre>
-<p>Check if <em class="parameter"><code>session</code></em>
- timeout out.</p>
-<div class="refsect3">
-<a name="id-1.2.10.8.11.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>session</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>now</p></td>
-<td class="parameter_description"><p> the current system time. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.10.8.11.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>session</code></em>
-timed out</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-manage-media"></a><h3>gst_rtsp_session_manage_media ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="returnvalue">GstRTSPSessionMedia</span></a> *
-gst_rtsp_session_manage_media (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *sess</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *path</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Manage the media object <em class="parameter"><code>obj</code></em>
- in <em class="parameter"><code>sess</code></em>
-. <em class="parameter"><code>path</code></em>
- will be used to retrieve this
-media from the session with <a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-media" title="gst_rtsp_session_get_media ()"><code class="function">gst_rtsp_session_get_media()</code></a>.</p>
-<p>Ownership is taken from <em class="parameter"><code>media</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.10.8.12.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>sess</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>path</p></td>
-<td class="parameter_description"><p>the path for the media</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p> a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.10.8.12.7"></a><h4>Returns</h4>
-<p> a new <em class="parameter"><code>GstRTSPSessionMedia</code></em>
-object. </p>
-<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-release-media"></a><h3>gst_rtsp_session_release_media ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_session_release_media (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *sess</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>);</pre>
-<p>Release the managed <em class="parameter"><code>media</code></em>
- in <em class="parameter"><code>sess</code></em>
-, freeing the memory allocated by it.</p>
-<div class="refsect3">
-<a name="id-1.2.10.8.13.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>sess</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p> a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.10.8.13.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if there are more media session left in <em class="parameter"><code>sess</code></em>
-.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-get-media"></a><h3>gst_rtsp_session_get_media ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="returnvalue">GstRTSPSessionMedia</span></a> *
-gst_rtsp_session_get_media (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *sess</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *path</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> *matched</code></em>);</pre>
-<p>Get the session media for <em class="parameter"><code>path</code></em>
-. <em class="parameter"><code>matched</code></em>
- will contain the number of matched
-characters of <em class="parameter"><code>path</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.10.8.14.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>sess</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>path</p></td>
-<td class="parameter_description"><p>the path for the media</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>matched</p></td>
-<td class="parameter_description"><p> the amount of matched characters. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.10.8.14.6"></a><h4>Returns</h4>
-<p> the configuration for <em class="parameter"><code>path</code></em>
-in <em class="parameter"><code>sess</code></em>
-. </p>
-<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPSessionFilterFunc"></a><h3>GstRTSPSessionFilterFunc ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
-<span class="c_punctuation">(</span>*GstRTSPSessionFilterFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *sess</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
-<p>This function will be called by the <a class="link" href="GstRTSPSession.html#gst-rtsp-session-filter" title="gst_rtsp_session_filter ()"><code class="function">gst_rtsp_session_filter()</code></a>. An
-implementation should return a value of <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
-<p>When this function returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, <em class="parameter"><code>media</code></em>
- will be removed
-from <em class="parameter"><code>sess</code></em>
-.</p>
-<p>A return value of <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a> will leave <em class="parameter"><code>media</code></em>
- untouched in
-<em class="parameter"><code>sess</code></em>
-.</p>
-<p>A value of GST_RTSP_FILTER_REF will add <em class="parameter"><code>media</code></em>
- to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-filter" title="gst_rtsp_session_filter ()"><code class="function">gst_rtsp_session_filter()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.10.8.15.8"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>sess</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> object</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> in <em class="parameter"><code>sess</code></em>
-</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p>user data that has been given to <a class="link" href="GstRTSPSession.html#gst-rtsp-session-filter" title="gst_rtsp_session_filter ()"><code class="function">gst_rtsp_session_filter()</code></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.10.8.15.9"></a><h4>Returns</h4>
-<p> a <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-filter"></a><h3>gst_rtsp_session_filter ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
-gst_rtsp_session_filter (<em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *sess</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPSession.html#GstRTSPSessionFilterFunc" title="GstRTSPSessionFilterFunc ()"><span class="type">GstRTSPSessionFilterFunc</span></a> func</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
-<p>Call <em class="parameter"><code>func</code></em>
- for each media in <em class="parameter"><code>sess</code></em>
-. The result value of <em class="parameter"><code>func</code></em>
- determines
-what happens to the media. <em class="parameter"><code>func</code></em>
- will be called with <em class="parameter"><code>sess</code></em>
-
-locked so no further actions on <em class="parameter"><code>sess</code></em>
- can be performed from <em class="parameter"><code>func</code></em>
-.</p>
-<p>If <em class="parameter"><code>func</code></em>
- returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, the media will be removed from
-<em class="parameter"><code>sess</code></em>
-.</p>
-<p>If <em class="parameter"><code>func</code></em>
- returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a>, the media will remain in <em class="parameter"><code>sess</code></em>
-.</p>
-<p>If <em class="parameter"><code>func</code></em>
- returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>, the media will remain in <em class="parameter"><code>sess</code></em>
- but
-will also be added with an additional ref to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of this
-function..</p>
-<p>When <em class="parameter"><code>func</code></em>
- is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a> will be assumed for all media.</p>
-<div class="refsect3">
-<a name="id-1.2.10.8.16.9"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>sess</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>func</p></td>
-<td class="parameter_description"><p> a callback. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid only during the call to the method."><span class="acronym">scope call</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p> user data passed to <em class="parameter"><code>func</code></em>
-. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="This parameter is a 'user_data', for callbacks; many bindings can pass NULL here."><span class="acronym">closure</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.10.8.16.10"></a><h4>Returns</h4>
-<p> a GList with all
-media for which <em class="parameter"><code>func</code></em>
-returned <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>. After usage, each
-element in the <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> should be unreffed before the list is freed. </p>
-<p><span class="annotation">[<acronym title="Generics and defining elements of containers and arrays."><span class="acronym">element-type</span></acronym> GstRTSPSessionMedia][<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSession.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstRTSPSession-struct"></a><h3>struct GstRTSPSession</h3>
-<pre class="programlisting">struct GstRTSPSession;</pre>
-<p>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.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPSessionClass"></a><h3>struct GstRTSPSessionClass</h3>
-<pre class="programlisting">struct GstRTSPSessionClass {
- GObjectClass parent_class;
-};
-</pre>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPFilterResult"></a><h3>enum GstRTSPFilterResult</h3>
-<p>Possible return values for <a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-filter" title="gst_rtsp_session_pool_filter ()"><code class="function">gst_rtsp_session_pool_filter()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.10.9.4.4"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="enum_members_name">
-<col class="enum_members_description">
-<col width="200px" class="enum_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-FILTER-REMOVE:CAPS"></a>GST_RTSP_FILTER_REMOVE</p></td>
-<td class="enum_member_description">
-<p>Remove session</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-FILTER-KEEP:CAPS"></a>GST_RTSP_FILTER_KEEP</p></td>
-<td class="enum_member_description">
-<p>Keep session in the pool</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-FILTER-REF:CAPS"></a>GST_RTSP_FILTER_REF</p></td>
-<td class="enum_member_description">
-<p>Ref session in the result list</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSession.property-details"></a><h2>Property Details</h2>
-<div class="refsect2">
-<a name="GstRTSPSession--sessionid"></a><h3>The <code class="literal">“sessionid”</code> property</h3>
-<pre class="programlisting"> “sessionid” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *</pre>
-<p>the session id.</p>
-<p>Flags: Read / Write / Construct Only</p>
-<p>Default value: NULL</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPSession--timeout"></a><h3>The <code class="literal">“timeout”</code> property</h3>
-<pre class="programlisting"> “timeout” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a></pre>
-<p>the timeout of the session (0 = never).</p>
-<p>Flags: Read / Write</p>
-<p>Default value: 60</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPSession--timeout-always-visible"></a><h3>The <code class="literal">“timeout-always-visible”</code> property</h3>
-<pre class="programlisting"> “timeout-always-visible” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a></pre>
-<p>timeout always visible in header.</p>
-<p>Flags: Read / Write</p>
-<p>Default value: FALSE</p>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSession.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a>, <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p>
-</div>
-</div>
-<div class="footer">
-<hr>
- Generated by GTK-Doc V1.21</div>
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-<div class="refentry">
-<a name="GstRTSPSessionPool"></a><div class="titlepage"></div>
-<div class="refnamediv"><table width="100%"><tr>
-<td valign="top">
-<h2><span class="refentrytitle"><a name="GstRTSPSessionPool.top_of_page"></a>GstRTSPSessionPool</span></h2>
-<p>GstRTSPSessionPool — An object for managing sessions</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="GstRTSPSessionPool.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="returnvalue">GstRTSPSessionPool</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-new" title="gst_rtsp_session_pool_new ()">gst_rtsp_session_pool_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-get-max-sessions" title="gst_rtsp_session_pool_get_max_sessions ()">gst_rtsp_session_pool_get_max_sessions</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-set-max-sessions" title="gst_rtsp_session_pool_set_max_sessions ()">gst_rtsp_session_pool_set_max_sessions</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-get-n-sessions" title="gst_rtsp_session_pool_get_n_sessions ()">gst_rtsp_session_pool_get_n_sessions</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="returnvalue">GstRTSPSession</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-create" title="gst_rtsp_session_pool_create ()">gst_rtsp_session_pool_create</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="returnvalue">GstRTSPSession</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-find" title="gst_rtsp_session_pool_find ()">gst_rtsp_session_pool_find</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-remove" title="gst_rtsp_session_pool_remove ()">gst_rtsp_session_pool_remove</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-cleanup" title="gst_rtsp_session_pool_cleanup ()">gst_rtsp_session_pool_cleanup</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<span class="c_punctuation">(</span><a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPoolFunc" title="GstRTSPSessionPoolFunc ()">*GstRTSPSessionPoolFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="returnvalue">GSource</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-create-watch" title="gst_rtsp_session_pool_create_watch ()">gst_rtsp_session_pool_create_watch</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
-</td>
-<td class="function_name">
-<span class="c_punctuation">(</span><a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPoolFilterFunc" title="GstRTSPSessionPoolFilterFunc ()">*GstRTSPSessionPoolFilterFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-filter" title="gst_rtsp_session_pool_filter ()">gst_rtsp_session_pool_filter</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSessionPool.properties"></a><h2>Properties</h2>
-<div class="informaltable"><table border="0">
-<colgroup>
-<col width="150px" class="properties_type">
-<col width="300px" class="properties_name">
-<col width="200px" class="properties_flags">
-</colgroup>
-<tbody><tr>
-<td class="property_type"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a></td>
-<td class="property_name"><a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPool--max-sessions" title="The “max-sessions” property">max-sessions</a></td>
-<td class="property_flags">Read / Write</td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSessionPool.signals"></a><h2>Signals</h2>
-<div class="informaltable"><table border="0">
-<colgroup>
-<col width="150px" class="signals_return">
-<col width="300px" class="signals_name">
-<col width="200px" class="signals_flags">
-</colgroup>
-<tbody><tr>
-<td class="signal_type"><span class="returnvalue">void</span></td>
-<td class="signal_name"><a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPool-session-removed" title="The “session-removed” signal">session-removed</a></td>
-<td class="signal_flags">Run Last</td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSessionPool.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPool-struct" title="struct GstRTSPSessionPool">GstRTSPSessionPool</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPoolClass" title="struct GstRTSPSessionPoolClass">GstRTSPSessionPoolClass</a></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSessionPool.object-hierarchy"></a><h2>Object Hierarchy</h2>
-<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
- <span class="lineart">╰──</span> GstRTSPSessionPool
-</pre>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSessionPool.description"></a><h2>Description</h2>
-<p>The <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> object manages a list of <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> objects.</p>
-<p>The maximum number of sessions can be configured with
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-set-max-sessions" title="gst_rtsp_session_pool_set_max_sessions ()"><code class="function">gst_rtsp_session_pool_set_max_sessions()</code></a>. The current number of sessions can
-be retrieved with <a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-get-n-sessions" title="gst_rtsp_session_pool_get_n_sessions ()"><code class="function">gst_rtsp_session_pool_get_n_sessions()</code></a>.</p>
-<p>Use <a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-create" title="gst_rtsp_session_pool_create ()"><code class="function">gst_rtsp_session_pool_create()</code></a> to create a new <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> object.
-The session object can be found again with its id and
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-find" title="gst_rtsp_session_pool_find ()"><code class="function">gst_rtsp_session_pool_find()</code></a>.</p>
-<p>All sessions can be iterated with <a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-filter" title="gst_rtsp_session_pool_filter ()"><code class="function">gst_rtsp_session_pool_filter()</code></a>.</p>
-<p>Run <a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-cleanup" title="gst_rtsp_session_pool_cleanup ()"><code class="function">gst_rtsp_session_pool_cleanup()</code></a> periodically to remove timed out sessions
-or use <a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-create-watch" title="gst_rtsp_session_pool_create_watch ()"><code class="function">gst_rtsp_session_pool_create_watch()</code></a> to be notified when session
-cleanup should be performed.</p>
-<p>Last reviewed on 2013-07-11 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSessionPool.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-session-pool-new"></a><h3>gst_rtsp_session_pool_new ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="returnvalue">GstRTSPSessionPool</span></a> *
-gst_rtsp_session_pool_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
-<p>Create a new <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> instance.</p>
-<div class="refsect3">
-<a name="id-1.2.9.9.2.5"></a><h4>Returns</h4>
-<p> A new <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a>. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
-usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-pool-get-max-sessions"></a><h3>gst_rtsp_session_pool_get_max_sessions ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-gst_rtsp_session_pool_get_max_sessions
- (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>);</pre>
-<p>Get the maximum allowed number of sessions in <em class="parameter"><code>pool</code></em>
-. 0 means an unlimited
-amount of sessions.</p>
-<div class="refsect3">
-<a name="id-1.2.9.9.3.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.9.9.3.6"></a><h4>Returns</h4>
-<p> the maximum allowed number of sessions.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-pool-set-max-sessions"></a><h3>gst_rtsp_session_pool_set_max_sessions ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_session_pool_set_max_sessions
- (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> max</code></em>);</pre>
-<p>Configure the maximum allowed number of sessions in <em class="parameter"><code>pool</code></em>
- to <em class="parameter"><code>max</code></em>
-.
-A value of 0 means an unlimited amount of sessions.</p>
-<div class="refsect3">
-<a name="id-1.2.9.9.4.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>max</p></td>
-<td class="parameter_description"><p>the maximum number of sessions</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-pool-get-n-sessions"></a><h3>gst_rtsp_session_pool_get_n_sessions ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-gst_rtsp_session_pool_get_n_sessions (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>);</pre>
-<p>Get the amount of active sessions in <em class="parameter"><code>pool</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.9.9.5.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.9.9.5.6"></a><h4>Returns</h4>
-<p> the amount of active sessions in <em class="parameter"><code>pool</code></em>
-.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-pool-create"></a><h3>gst_rtsp_session_pool_create ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="returnvalue">GstRTSPSession</span></a> *
-gst_rtsp_session_pool_create (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>);</pre>
-<p>Create a new <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> object in <em class="parameter"><code>pool</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.9.9.6.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.9.9.6.6"></a><h4>Returns</h4>
-<p> a new <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a>. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-pool-find"></a><h3>gst_rtsp_session_pool_find ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="returnvalue">GstRTSPSession</span></a> *
-gst_rtsp_session_pool_find (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *sessionid</code></em>);</pre>
-<p>Find the session with <em class="parameter"><code>sessionid</code></em>
- in <em class="parameter"><code>pool</code></em>
-. The access time of the session
-will be updated with <a class="link" href="GstRTSPSession.html#gst-rtsp-session-touch" title="gst_rtsp_session_touch ()"><code class="function">gst_rtsp_session_touch()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.9.9.7.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>the pool to search</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>sessionid</p></td>
-<td class="parameter_description"><p>the session id</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.9.9.7.6"></a><h4>Returns</h4>
-<p> the <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> with <em class="parameter"><code>sessionid</code></em>
-or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when the session did not exist. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>][<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-pool-remove"></a><h3>gst_rtsp_session_pool_remove ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_session_pool_remove (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *sess</code></em>);</pre>
-<p>Remove <em class="parameter"><code>sess</code></em>
- from <em class="parameter"><code>pool</code></em>
-, releasing the ref that the pool has on <em class="parameter"><code>sess</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.9.9.8.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>sess</p></td>
-<td class="parameter_description"><p> a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.9.9.8.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the session was found and removed.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-pool-cleanup"></a><h3>gst_rtsp_session_pool_cleanup ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-gst_rtsp_session_pool_cleanup (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>);</pre>
-<p>Inspect all the sessions in <em class="parameter"><code>pool</code></em>
- and remove the sessions that are inactive
-for more than their timeout.</p>
-<div class="refsect3">
-<a name="id-1.2.9.9.9.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.9.9.9.6"></a><h4>Returns</h4>
-<p> the amount of sessions that got removed.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPSessionPoolFunc"></a><h3>GstRTSPSessionPoolFunc ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-<span class="c_punctuation">(</span>*GstRTSPSessionPoolFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
-<p>The function that will be called from the GSource watch on the session pool.</p>
-<p>The function will be called when the pool must be cleaned up because one or
-more sessions timed out.</p>
-<div class="refsect3">
-<a name="id-1.2.9.9.10.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> object</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p>user data that has been given when registering the handler</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.9.9.10.7"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#FALSE:CAPS"><code class="literal">FALSE</code></a> if the source should be removed.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-pool-create-watch"></a><h3>gst_rtsp_session_pool_create_watch ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="returnvalue">GSource</span></a> *
-gst_rtsp_session_pool_create_watch (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>);</pre>
-<p>Create a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="type">GSource</span></a> that will be dispatched when the session should be cleaned
-up.</p>
-<div class="refsect3">
-<a name="id-1.2.9.9.11.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.9.9.11.6"></a><h4>Returns</h4>
-<p> a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GSource"><span class="type">GSource</span></a>. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPSessionPoolFilterFunc"></a><h3>GstRTSPSessionPoolFilterFunc ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
-<span class="c_punctuation">(</span>*GstRTSPSessionPoolFilterFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *session</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
-<p>This function will be called by the <a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-filter" title="gst_rtsp_session_pool_filter ()"><code class="function">gst_rtsp_session_pool_filter()</code></a>. An
-implementation should return a value of <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
-<p>When this function returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, <em class="parameter"><code>session</code></em>
- will be removed
-from <em class="parameter"><code>pool</code></em>
-.</p>
-<p>A return value of <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a> will leave <em class="parameter"><code>session</code></em>
- untouched in
-<em class="parameter"><code>pool</code></em>
-.</p>
-<p>A value of GST_RTSP_FILTER_REF will add <em class="parameter"><code>session</code></em>
- to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-filter" title="gst_rtsp_session_pool_filter ()"><code class="function">gst_rtsp_session_pool_filter()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.9.9.12.8"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> object</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>session</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> in <em class="parameter"><code>pool</code></em>
-</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p>user data that has been given to <a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-filter" title="gst_rtsp_session_pool_filter ()"><code class="function">gst_rtsp_session_pool_filter()</code></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.9.9.12.9"></a><h4>Returns</h4>
-<p> a <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-pool-filter"></a><h3>gst_rtsp_session_pool_filter ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
-gst_rtsp_session_pool_filter (<em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *pool</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPoolFilterFunc" title="GstRTSPSessionPoolFilterFunc ()"><span class="type">GstRTSPSessionPoolFilterFunc</span></a> func</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
-<p>Call <em class="parameter"><code>func</code></em>
- for each session in <em class="parameter"><code>pool</code></em>
-. The result value of <em class="parameter"><code>func</code></em>
- determines
-what happens to the session. <em class="parameter"><code>func</code></em>
- will be called with the session pool
-locked so no further actions on <em class="parameter"><code>pool</code></em>
- can be performed from <em class="parameter"><code>func</code></em>
-.</p>
-<p>If <em class="parameter"><code>func</code></em>
- returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, the session will be set to the
-expired state with <code class="function">gst_rtsp_session_set_expired()</code> and removed from
-<em class="parameter"><code>pool</code></em>
-.</p>
-<p>If <em class="parameter"><code>func</code></em>
- returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a>, the session will remain in <em class="parameter"><code>pool</code></em>
-.</p>
-<p>If <em class="parameter"><code>func</code></em>
- returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>, the session will remain in <em class="parameter"><code>pool</code></em>
- but
-will also be added with an additional ref to the result GList of this
-function..</p>
-<p>When <em class="parameter"><code>func</code></em>
- is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a> will be assumed for all sessions.</p>
-<div class="refsect3">
-<a name="id-1.2.9.9.13.9"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>func</p></td>
-<td class="parameter_description"><p> a callback. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid only during the call to the method."><span class="acronym">scope call</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p> user data passed to <em class="parameter"><code>func</code></em>
-. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="This parameter is a 'user_data', for callbacks; many bindings can pass NULL here."><span class="acronym">closure</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.9.9.13.10"></a><h4>Returns</h4>
-<p> a GList with all
-sessions for which <em class="parameter"><code>func</code></em>
-returned <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>. After usage, each
-element in the GList should be unreffed before the list is freed. </p>
-<p><span class="annotation">[<acronym title="Generics and defining elements of containers and arrays."><span class="acronym">element-type</span></acronym> GstRTSPSession][<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSessionPool.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstRTSPSessionPool-struct"></a><h3>struct GstRTSPSessionPool</h3>
-<pre class="programlisting">struct GstRTSPSessionPool;</pre>
-<p>An object that keeps track of the active sessions. This object is usually
-attached to a <a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> object to manage the sessions in that server.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPSessionPoolClass"></a><h3>struct GstRTSPSessionPoolClass</h3>
-<pre class="programlisting">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);
-};
-</pre>
-<div class="refsect3">
-<a name="id-1.2.9.10.3.4"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="struct_members_name">
-<col class="struct_members_description">
-<col width="200px" class="struct_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObjectClass"><span class="type">GObjectClass</span></a> <em class="structfield"><code><a name="GstRTSPSessionPoolClass.parent-class"></a>parent_class</code></em>;</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPSessionPoolClass.create-session-id"></a>create_session_id</code></em> ()</p></td>
-<td class="struct_member_description"><p>create a new random session id. Subclasses can create
-custom session ids and should not check if the session exists.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPSessionPoolClass.create-session"></a>create_session</code></em> ()</p></td>
-<td class="struct_member_description"><p>make a new session object.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPSessionPoolClass.session-removed"></a>session_removed</code></em> ()</p></td>
-<td class="struct_member_description"><p>a session was removed from the pool</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSessionPool.property-details"></a><h2>Property Details</h2>
-<div class="refsect2">
-<a name="GstRTSPSessionPool--max-sessions"></a><h3>The <code class="literal">“max-sessions”</code> property</h3>
-<pre class="programlisting"> “max-sessions” <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a></pre>
-<p>the maximum amount of sessions (0 = unlimited).</p>
-<p>Flags: Read / Write</p>
-<p>Default value: 0</p>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSessionPool.signal-details"></a><h2>Signal Details</h2>
-<div class="refsect2">
-<a name="GstRTSPSessionPool-session-removed"></a><h3>The <code class="literal">“session-removed”</code> signal</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-user_function (<a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool"><span class="type">GstRTSPSessionPool</span></a> *gstrtspsessionpool,
- <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *arg1,
- <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data)</pre>
-<p>Flags: Run Last</p>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPSessionPool.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p>
-</div>
-</div>
-<div class="footer">
-<hr>
- Generated by GTK-Doc V1.21</div>
-</body>
-</html>
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-<div class="refentry">
-<a name="GstRTSPStream"></a><div class="titlepage"></div>
-<div class="refnamediv"><table width="100%"><tr>
-<td valign="top">
-<h2><span class="refentrytitle"><a name="GstRTSPStream.top_of_page"></a>GstRTSPStream</span></h2>
-<p>GstRTSPStream — A media stream</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="GstRTSPStream.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-new" title="gst_rtsp_stream_new ()">gst_rtsp_stream_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-index" title="gst_rtsp_stream_get_index ()">gst_rtsp_stream_get_index</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html"><span class="returnvalue">GstPad</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-srcpad" title="gst_rtsp_stream_get_srcpad ()">gst_rtsp_stream_get_srcpad</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-control" title="gst_rtsp_stream_get_control ()">gst_rtsp_stream_get_control</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-control" title="gst_rtsp_stream_set_control ()">gst_rtsp_stream_set_control</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-has-control" title="gst_rtsp_stream_has_control ()">gst_rtsp_stream_has_control</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-mtu" title="gst_rtsp_stream_get_mtu ()">gst_rtsp_stream_get_mtu</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-mtu" title="gst_rtsp_stream_set_mtu ()">gst_rtsp_stream_set_mtu</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="returnvalue">gint</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-dscp-qos" title="gst_rtsp_stream_get_dscp_qos ()">gst_rtsp_stream_get_dscp_qos</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-dscp-qos" title="gst_rtsp_stream_set_dscp_qos ()">gst_rtsp_stream_set_dscp_qos</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-profiles" title="gst_rtsp_stream_set_profiles ()">gst_rtsp_stream_set_profiles</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="returnvalue">GstRTSPProfile</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-profiles" title="gst_rtsp_stream_get_profiles ()">gst_rtsp_stream_get_profiles</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="returnvalue">GstRTSPLowerTrans</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-protocols" title="gst_rtsp_stream_get_protocols ()">gst_rtsp_stream_get_protocols</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-protocols" title="gst_rtsp_stream_set_protocols ()">gst_rtsp_stream_set_protocols</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-is-transport-supported" title="gst_rtsp_stream_is_transport_supported ()">gst_rtsp_stream_is_transport_supported</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="returnvalue">GstRTSPAddressPool</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-address-pool" title="gst_rtsp_stream_get_address_pool ()">gst_rtsp_stream_get_address_pool</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-address-pool" title="gst_rtsp_stream_set_address_pool ()">gst_rtsp_stream_set_address_pool</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="returnvalue">GstRTSPAddress</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-reserve-address" title="gst_rtsp_stream_reserve_address ()">gst_rtsp_stream_reserve_address</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-join-bin" title="gst_rtsp_stream_join_bin ()">gst_rtsp_stream_join_bin</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-leave-bin" title="gst_rtsp_stream_leave_bin ()">gst_rtsp_stream_leave_bin</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-server-port" title="gst_rtsp_stream_get_server_port ()">gst_rtsp_stream_get_server_port</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="returnvalue">GstRTSPAddress</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-multicast-address" title="gst_rtsp_stream_get_multicast_address ()">gst_rtsp_stream_get_multicast_address</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject"><span class="returnvalue">GObject</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-rtpsession" title="gst_rtsp_stream_get_rtpsession ()">gst_rtsp_stream_get_rtpsession</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-ssrc" title="gst_rtsp_stream_get_ssrc ()">gst_rtsp_stream_get_ssrc</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-rtpinfo" title="gst_rtsp_stream_get_rtpinfo ()">gst_rtsp_stream_get_rtpinfo</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstCaps.html"><span class="returnvalue">GstCaps</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-caps" title="gst_rtsp_stream_get_caps ()">gst_rtsp_stream_get_caps</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-pt" title="gst_rtsp_stream_get_pt ()">gst_rtsp_stream_get_pt</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#GstFlowReturn"><span class="returnvalue">GstFlowReturn</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-recv-rtcp" title="gst_rtsp_stream_recv_rtcp ()">gst_rtsp_stream_recv_rtcp</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#GstFlowReturn"><span class="returnvalue">GstFlowReturn</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-recv-rtp" title="gst_rtsp_stream_recv_rtp ()">gst_rtsp_stream_recv_rtp</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-add-transport" title="gst_rtsp_stream_add_transport ()">gst_rtsp_stream_add_transport</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-remove-transport" title="gst_rtsp_stream_remove_transport ()">gst_rtsp_stream_remove_transport</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="returnvalue">GSocket</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-rtp-socket" title="gst_rtsp_stream_get_rtp_socket ()">gst_rtsp_stream_get_rtp_socket</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="returnvalue">GSocket</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-rtcp-socket" title="gst_rtsp_stream_get_rtcp_socket ()">gst_rtsp_stream_get_rtcp_socket</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-blocked" title="gst_rtsp_stream_set_blocked ()">gst_rtsp_stream_set_blocked</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-is-blocking" title="gst_rtsp_stream_is_blocking ()">gst_rtsp_stream_is_blocking</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-update-crypto" title="gst_rtsp_stream_update_crypto ()">gst_rtsp_stream_update_crypto</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
-</td>
-<td class="function_name">
-<span class="c_punctuation">(</span><a class="link" href="GstRTSPStream.html#GstRTSPStreamTransportFilterFunc" title="GstRTSPStreamTransportFilterFunc ()">*GstRTSPStreamTransportFilterFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-transport-filter" title="gst_rtsp_stream_transport_filter ()">gst_rtsp_stream_transport_filter</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPStream.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="GstRTSPStream.html#GstRTSPStream-struct" title="struct GstRTSPStream">GstRTSPStream</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword"> </td>
-<td class="function_name"><a class="link" href="GstRTSPStream.html#GstRTSPStreamClass" title="GstRTSPStreamClass">GstRTSPStreamClass</a></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPStream.object-hierarchy"></a><h2>Object Hierarchy</h2>
-<pre class="screen"> <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject">GObject</a>
- <span class="lineart">╰──</span> GstRTSPStream
-</pre>
-</div>
-<div class="refsect1">
-<a name="GstRTSPStream.description"></a><h2>Description</h2>
-<p>The <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> 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.</p>
-<p>With <a class="link" href="GstRTSPStream.html#gst-rtsp-stream-join-bin" title="gst_rtsp_stream_join_bin ()"><code class="function">gst_rtsp_stream_join_bin()</code></a> the streaming elements are added to the bin
-and rtpbin. <a class="link" href="GstRTSPStream.html#gst-rtsp-stream-leave-bin" title="gst_rtsp_stream_leave_bin ()"><code class="function">gst_rtsp_stream_leave_bin()</code></a> removes the elements again.</p>
-<p>The <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> will use the configured addresspool, as set with
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-address-pool" title="gst_rtsp_stream_set_address_pool ()"><code class="function">gst_rtsp_stream_set_address_pool()</code></a>, to allocate multicast addresses for the
-stream. With <a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-multicast-address" title="gst_rtsp_stream_get_multicast_address ()"><code class="function">gst_rtsp_stream_get_multicast_address()</code></a> you can get the
-configured address.</p>
-<p>With <a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-server-port" title="gst_rtsp_stream_get_server_port ()"><code class="function">gst_rtsp_stream_get_server_port()</code></a> 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.</p>
-<p>With <a class="link" href="GstRTSPStream.html#gst-rtsp-stream-add-transport" title="gst_rtsp_stream_add_transport ()"><code class="function">gst_rtsp_stream_add_transport()</code></a> destinations can be added where the
-stream should be sent to. Use <a class="link" href="GstRTSPStream.html#gst-rtsp-stream-remove-transport" title="gst_rtsp_stream_remove_transport ()"><code class="function">gst_rtsp_stream_remove_transport()</code></a> to remove
-the destination again.</p>
-<p>Last reviewed on 2013-07-16 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="GstRTSPStream.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-stream-new"></a><h3>gst_rtsp_stream_new ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
-gst_rtsp_stream_new (<em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> idx</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> *payloader</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html"><span class="type">GstPad</span></a> *srcpad</code></em>);</pre>
-<p>Create a new media stream with index <em class="parameter"><code>idx</code></em>
- that handles RTP data on
-<em class="parameter"><code>srcpad</code></em>
- and has a payloader element <em class="parameter"><code>payloader</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.2.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>idx</p></td>
-<td class="parameter_description"><p>an index</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>srcpad</p></td>
-<td class="parameter_description"><p>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html"><span class="type">GstPad</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>payloader</p></td>
-<td class="parameter_description"><p>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.2.6"></a><h4>Returns</h4>
-<p> a new <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a>. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-index"></a><h3>gst_rtsp_stream_get_index ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-gst_rtsp_stream_get_index (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
-<p>Get the stream index.</p>
-<p>Return: the stream index.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.3.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-srcpad"></a><h3>gst_rtsp_stream_get_srcpad ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html"><span class="returnvalue">GstPad</span></a> *
-gst_rtsp_stream_get_srcpad (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
-<p>Get the srcpad associated with <em class="parameter"><code>stream</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.4.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.4.6"></a><h4>Returns</h4>
-<p> the srcpad. Unref after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-control"></a><h3>gst_rtsp_stream_get_control ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-gst_rtsp_stream_get_control (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
-<p>Get the control string to identify this stream.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.5.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.5.6"></a><h4>Returns</h4>
-<p> the control string. <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-set-control"></a><h3>gst_rtsp_stream_set_control ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_stream_set_control (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *control</code></em>);</pre>
-<p>Set the control string in <em class="parameter"><code>stream</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.6.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>control</p></td>
-<td class="parameter_description"><p>a control string</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-has-control"></a><h3>gst_rtsp_stream_has_control ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_stream_has_control (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *control</code></em>);</pre>
-<p>Check if <em class="parameter"><code>stream</code></em>
- has the control string <em class="parameter"><code>control</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.7.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>control</p></td>
-<td class="parameter_description"><p>a control string</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.7.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> is <em class="parameter"><code>stream</code></em>
-has <em class="parameter"><code>control</code></em>
-as the control string</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-mtu"></a><h3>gst_rtsp_stream_get_mtu ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-gst_rtsp_stream_get_mtu (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
-<p>Get the configured MTU in the payloader of <em class="parameter"><code>stream</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.8.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.8.6"></a><h4>Returns</h4>
-<p> the MTU of the payloader.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-set-mtu"></a><h3>gst_rtsp_stream_set_mtu ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_stream_set_mtu (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> mtu</code></em>);</pre>
-<p>Configure the mtu in the payloader of <em class="parameter"><code>stream</code></em>
- to <em class="parameter"><code>mtu</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.9.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>mtu</p></td>
-<td class="parameter_description"><p>a new MTU</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-dscp-qos"></a><h3>gst_rtsp_stream_get_dscp_qos ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="returnvalue">gint</span></a>
-gst_rtsp_stream_get_dscp_qos (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
-<p>Get the configured DSCP QoS in of the outgoing sockets.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.10.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.10.6"></a><h4>Returns</h4>
-<p> the DSCP QoS value of the outgoing sockets, or -1 if disbled.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-set-dscp-qos"></a><h3>gst_rtsp_stream_set_dscp_qos ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_stream_set_dscp_qos (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> dscp_qos</code></em>);</pre>
-<p>Configure the dscp qos of the outgoing sockets to <em class="parameter"><code>dscp_qos</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.11.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>dscp_qos</p></td>
-<td class="parameter_description"><p>a new dscp qos value (0-63, or -1 to disable)</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-set-profiles"></a><h3>gst_rtsp_stream_set_profiles ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_stream_set_profiles (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a> profiles</code></em>);</pre>
-<p>Configure the allowed profiles for <em class="parameter"><code>stream</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.12.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>profiles</p></td>
-<td class="parameter_description"><p>the new profiles</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-profiles"></a><h3>gst_rtsp_stream_get_profiles ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="returnvalue">GstRTSPProfile</span></a>
-gst_rtsp_stream_get_profiles (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
-<p>Get the allowed profiles of <em class="parameter"><code>stream</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.13.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.13.6"></a><h4>Returns</h4>
-<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPProfile"><span class="type">GstRTSPProfile</span></a></p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-protocols"></a><h3>gst_rtsp_stream_get_protocols ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="returnvalue">GstRTSPLowerTrans</span></a>
-gst_rtsp_stream_get_protocols (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
-<p>Get the allowed protocols of <em class="parameter"><code>stream</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.14.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.14.6"></a><h4>Returns</h4>
-<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a></p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-set-protocols"></a><h3>gst_rtsp_stream_set_protocols ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_stream_set_protocols (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPLowerTrans"><span class="type">GstRTSPLowerTrans</span></a> protocols</code></em>);</pre>
-<p>Configure the allowed lower transport for <em class="parameter"><code>stream</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.15.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>protocols</p></td>
-<td class="parameter_description"><p>the new flags</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-is-transport-supported"></a><h3>gst_rtsp_stream_is_transport_supported ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_stream_is_transport_supported
- (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a> *transport</code></em>);</pre>
-<p>Check if <em class="parameter"><code>transport</code></em>
- can be handled by stream</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.16.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>transport</p></td>
-<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.16.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>transport</code></em>
-can be handled by <em class="parameter"><code>stream</code></em>
-.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-address-pool"></a><h3>gst_rtsp_stream_get_address_pool ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="returnvalue">GstRTSPAddressPool</span></a> *
-gst_rtsp_stream_get_address_pool (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
-<p>Get the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> used as the address pool of <em class="parameter"><code>stream</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.17.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.17.6"></a><h4>Returns</h4>
-<p> the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> of <em class="parameter"><code>stream</code></em>
-. <a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#g-object-unref"><code class="function">g_object_unref()</code></a> after
-usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-set-address-pool"></a><h3>gst_rtsp_stream_set_address_pool ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_stream_set_address_pool (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>);</pre>
-<p>configure <em class="parameter"><code>pool</code></em>
- to be used as the address pool of <em class="parameter"><code>stream</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.18.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-reserve-address"></a><h3>gst_rtsp_stream_reserve_address ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="returnvalue">GstRTSPAddress</span></a> *
-gst_rtsp_stream_reserve_address (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *address</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> port</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> n_ports</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> ttl</code></em>);</pre>
-<p>Reserve <em class="parameter"><code>address</code></em>
- and <em class="parameter"><code>port</code></em>
- as the address and port of <em class="parameter"><code>stream</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.19.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>address</p></td>
-<td class="parameter_description"><p>an address</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>port</p></td>
-<td class="parameter_description"><p>a port</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>n_ports</p></td>
-<td class="parameter_description"><p>n_ports</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>ttl</p></td>
-<td class="parameter_description"><p>a TTL</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.19.6"></a><h4>Returns</h4>
-<p> the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a> of <em class="parameter"><code>stream</code></em>
-or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when
-the address could be reserved. <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-free" title="gst_rtsp_address_free ()"><code class="function">gst_rtsp_address_free()</code></a> after usage. </p>
-<p><span class="annotation">[<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-join-bin"></a><h3>gst_rtsp_stream_join_bin ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_stream_join_bin (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBin.html"><span class="type">GstBin</span></a> *bin</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> *rtpbin</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#GstState"><span class="type">GstState</span></a> state</code></em>);</pre>
-<p>Join the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBin.html"><span class="type">GstBin</span></a> <em class="parameter"><code>bin</code></em>
- that contains the element <em class="parameter"><code>rtpbin</code></em>
-.</p>
-<p><em class="parameter"><code>stream</code></em>
- will link to <em class="parameter"><code>rtpbin</code></em>
-, which must be inside <em class="parameter"><code>bin</code></em>
-. The elements
-added to <em class="parameter"><code>bin</code></em>
- will be set to the state given in <em class="parameter"><code>state</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.20.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>bin</p></td>
-<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBin.html"><span class="type">GstBin</span></a> to join. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>rtpbin</p></td>
-<td class="parameter_description"><p> a rtpbin element in <em class="parameter"><code>bin</code></em>
-. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>state</p></td>
-<td class="parameter_description"><p>the target state of the new elements</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.20.7"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-leave-bin"></a><h3>gst_rtsp_stream_leave_bin ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_stream_leave_bin (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBin.html"><span class="type">GstBin</span></a> *bin</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a> *rtpbin</code></em>);</pre>
-<p>Remove the elements of <em class="parameter"><code>stream</code></em>
- from <em class="parameter"><code>bin</code></em>
-.</p>
-<p>Return: <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.21.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>bin</p></td>
-<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBin.html"><span class="type">GstBin</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>rtpbin</p></td>
-<td class="parameter_description"><p> a rtpbin <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html"><span class="type">GstElement</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-server-port"></a><h3>gst_rtsp_stream_get_server_port ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_stream_get_server_port (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPRange"><span class="type">GstRTSPRange</span></a> *server_port</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GSocketAddress.html#GSocketFamily"><span class="type">GSocketFamily</span></a> family</code></em>);</pre>
-<p>Fill <em class="parameter"><code>server_port</code></em>
- with the port pair used by the server. This function can
-only be called when <em class="parameter"><code>stream</code></em>
- has been joined.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.22.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>server_port</p></td>
-<td class="parameter_description"><p> result server port. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>family</p></td>
-<td class="parameter_description"><p>the port family to get</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-multicast-address"></a><h3>gst_rtsp_stream_get_multicast_address ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="returnvalue">GstRTSPAddress</span></a> *
-gst_rtsp_stream_get_multicast_address (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GSocketAddress.html#GSocketFamily"><span class="type">GSocketFamily</span></a> family</code></em>);</pre>
-<p>Get the multicast address of <em class="parameter"><code>stream</code></em>
- for <em class="parameter"><code>family</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.23.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>family</p></td>
-<td class="parameter_description"><p>the <a href="https://developer.gnome.org/gio/unstable/GSocketAddress.html#GSocketFamily"><span class="type">GSocketFamily</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.23.6"></a><h4>Returns</h4>
-<p> the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a> of <em class="parameter"><code>stream</code></em>
-or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when no address could be allocated. <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-free" title="gst_rtsp_address_free ()"><code class="function">gst_rtsp_address_free()</code></a>
-after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>][<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-rtpsession"></a><h3>gst_rtsp_stream_get_rtpsession ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject"><span class="returnvalue">GObject</span></a> *
-gst_rtsp_stream_get_rtpsession (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
-<p>Get the RTP session of this stream.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.24.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.24.6"></a><h4>Returns</h4>
-<p> The RTP session of this stream. Unref after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-ssrc"></a><h3>gst_rtsp_stream_get_ssrc ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_stream_get_ssrc (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> *ssrc</code></em>);</pre>
-<p>Get the SSRC used by the RTP session of this stream. This function can only
-be called when <em class="parameter"><code>stream</code></em>
- has been joined.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.25.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>ssrc</p></td>
-<td class="parameter_description"><p> result ssrc. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-rtpinfo"></a><h3>gst_rtsp_stream_get_rtpinfo ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_stream_get_rtpinfo (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> *rtptime</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> *seq</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> *clock_rate</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html#GstClockTime"><span class="type">GstClockTime</span></a> *running_time</code></em>);</pre>
-<p>Retrieve the current rtptime, seq and running-time. This is used to
-construct a RTPInfo reply header.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.26.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>rtptime</p></td>
-<td class="parameter_description"><p> result RTP timestamp. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>seq</p></td>
-<td class="parameter_description"><p> result RTP seqnum. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>clock_rate</p></td>
-<td class="parameter_description"><p> the clock rate. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>running_time</p></td>
-<td class="parameter_description"><p> result running-time. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.26.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> when rtptime, seq and running-time could be determined.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-caps"></a><h3>gst_rtsp_stream_get_caps ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstCaps.html"><span class="returnvalue">GstCaps</span></a> *
-gst_rtsp_stream_get_caps (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
-<p>Retrieve the current caps of <em class="parameter"><code>stream</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.27.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.27.6"></a><h4>Returns</h4>
-<p> the <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstCaps.html"><span class="type">GstCaps</span></a> of <em class="parameter"><code>stream</code></em>
-. use <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstCaps.html#gst-caps-unref"><code class="function">gst_caps_unref()</code></a>
-after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-pt"></a><h3>gst_rtsp_stream_get_pt ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="returnvalue">guint</span></a>
-gst_rtsp_stream_get_pt (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
-<p>Get the stream payload type.</p>
-<p>Return: the stream payload type.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.28.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-recv-rtcp"></a><h3>gst_rtsp_stream_recv_rtcp ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#GstFlowReturn"><span class="returnvalue">GstFlowReturn</span></a>
-gst_rtsp_stream_recv_rtcp (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a> *buffer</code></em>);</pre>
-<p>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.</p>
-<p>This function takes ownership of <em class="parameter"><code>buffer</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.29.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>buffer</p></td>
-<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.29.7"></a><h4>Returns</h4>
-<p> a GstFlowReturn.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-recv-rtp"></a><h3>gst_rtsp_stream_recv_rtp ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#GstFlowReturn"><span class="returnvalue">GstFlowReturn</span></a>
-gst_rtsp_stream_recv_rtp (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a> *buffer</code></em>);</pre>
-<p>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.</p>
-<p>This function takes ownership of <em class="parameter"><code>buffer</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.30.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>buffer</p></td>
-<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.30.7"></a><h4>Returns</h4>
-<p> a GstFlowReturn.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-add-transport"></a><h3>gst_rtsp_stream_add_transport ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_stream_add_transport (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>);</pre>
-<p>Add the transport in <em class="parameter"><code>trans</code></em>
- to <em class="parameter"><code>stream</code></em>
-. The media of <em class="parameter"><code>stream</code></em>
- will
-then also be send to the values configured in <em class="parameter"><code>trans</code></em>
-.</p>
-<p><em class="parameter"><code>stream</code></em>
- must be joined to a bin.</p>
-<p><em class="parameter"><code>trans</code></em>
- must contain a valid <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.31.7"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.31.8"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>trans</code></em>
-was added</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-remove-transport"></a><h3>gst_rtsp_stream_remove_transport ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_stream_remove_transport (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>);</pre>
-<p>Remove the transport in <em class="parameter"><code>trans</code></em>
- from <em class="parameter"><code>stream</code></em>
-. The media of <em class="parameter"><code>stream</code></em>
- will
-not be sent to the values configured in <em class="parameter"><code>trans</code></em>
-.</p>
-<p><em class="parameter"><code>stream</code></em>
- must be joined to a bin.</p>
-<p><em class="parameter"><code>trans</code></em>
- must contain a valid <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.32.7"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.32.8"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>trans</code></em>
-was removed</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-rtp-socket"></a><h3>gst_rtsp_stream_get_rtp_socket ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="returnvalue">GSocket</span></a> *
-gst_rtsp_stream_get_rtp_socket (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GSocketAddress.html#GSocketFamily"><span class="type">GSocketFamily</span></a> family</code></em>);</pre>
-<p>Get the RTP socket from <em class="parameter"><code>stream</code></em>
- for a <em class="parameter"><code>family</code></em>
-.</p>
-<p><em class="parameter"><code>stream</code></em>
- must be joined to a bin.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.33.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>family</p></td>
-<td class="parameter_description"><p>the socket family</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.33.7"></a><h4>Returns</h4>
-<p> the RTP socket or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> if no
-socket could be allocated for <em class="parameter"><code>family</code></em>
-. Unref after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>][<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-get-rtcp-socket"></a><h3>gst_rtsp_stream_get_rtcp_socket ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/gio/unstable/GSocket.html"><span class="returnvalue">GSocket</span></a> *
-gst_rtsp_stream_get_rtcp_socket (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/gio/unstable/GSocketAddress.html#GSocketFamily"><span class="type">GSocketFamily</span></a> family</code></em>);</pre>
-<p>Get the RTCP socket from <em class="parameter"><code>stream</code></em>
- for a <em class="parameter"><code>family</code></em>
-.</p>
-<p><em class="parameter"><code>stream</code></em>
- must be joined to a bin.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.34.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>family</p></td>
-<td class="parameter_description"><p>the socket family</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.34.7"></a><h4>Returns</h4>
-<p> the RTCP socket or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> if no
-socket could be allocated for <em class="parameter"><code>family</code></em>
-. Unref after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>][<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-set-blocked"></a><h3>gst_rtsp_stream_set_blocked ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_stream_set_blocked (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> blocked</code></em>);</pre>
-<p>Blocks or unblocks the dataflow on <em class="parameter"><code>stream</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.35.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>blocked</p></td>
-<td class="parameter_description"><p>boolean indicating we should block or unblock</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.35.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-is-blocking"></a><h3>gst_rtsp_stream_is_blocking ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_stream_is_blocking (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>);</pre>
-<p>Check if <em class="parameter"><code>stream</code></em>
- is blocking on a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.36.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.36.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>stream</code></em>
-is blocking</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-update-crypto"></a><h3>gst_rtsp_stream_update_crypto ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_stream_update_crypto (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> ssrc</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstCaps.html"><span class="type">GstCaps</span></a> *crypto</code></em>);</pre>
-<p>Update the new crypto information for <em class="parameter"><code>ssrc</code></em>
- in <em class="parameter"><code>stream</code></em>
-. If information
-for <em class="parameter"><code>ssrc</code></em>
- did not exist, it will be added. If information
-for <em class="parameter"><code>ssrc</code></em>
- existed, it will be replaced. If <em class="parameter"><code>crypto</code></em>
- is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, it will
-be removed from <em class="parameter"><code>stream</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.37.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>ssrc</p></td>
-<td class="parameter_description"><p>the SSRC</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>crypto</p></td>
-<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstCaps.html"><span class="type">GstCaps</span></a> with crypto info. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.37.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>crypto</code></em>
-could be updated</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPStreamTransportFilterFunc"></a><h3>GstRTSPStreamTransportFilterFunc ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="returnvalue">GstRTSPFilterResult</span></a>
-<span class="c_punctuation">(</span>*GstRTSPStreamTransportFilterFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
-<p>This function will be called by the <a class="link" href="GstRTSPStream.html#gst-rtsp-stream-transport-filter" title="gst_rtsp_stream_transport_filter ()"><code class="function">gst_rtsp_stream_transport_filter()</code></a>. An
-implementation should return a value of <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
-<p>When this function returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, <em class="parameter"><code>trans</code></em>
- will be removed
-from <em class="parameter"><code>stream</code></em>
-.</p>
-<p>A return value of <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a> will leave <em class="parameter"><code>trans</code></em>
- untouched in
-<em class="parameter"><code>stream</code></em>
-.</p>
-<p>A value of <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a> will add <em class="parameter"><code>trans</code></em>
- to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-transport-filter" title="gst_rtsp_stream_transport_filter ()"><code class="function">gst_rtsp_stream_transport_filter()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.38.8"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> object</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> in <em class="parameter"><code>stream</code></em>
-</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p>user data that has been given to <a class="link" href="GstRTSPStream.html#gst-rtsp-stream-transport-filter" title="gst_rtsp_stream_transport_filter ()"><code class="function">gst_rtsp_stream_transport_filter()</code></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.38.9"></a><h4>Returns</h4>
-<p> a <a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult"><span class="type">GstRTSPFilterResult</span></a>.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-transport-filter"></a><h3>gst_rtsp_stream_transport_filter ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="returnvalue">GList</span></a> *
-gst_rtsp_stream_transport_filter (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPStream.html#GstRTSPStreamTransportFilterFunc" title="GstRTSPStreamTransportFilterFunc ()"><span class="type">GstRTSPStreamTransportFilterFunc</span></a> func</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
-<p>Call <em class="parameter"><code>func</code></em>
- for each transport managed by <em class="parameter"><code>stream</code></em>
-. The result value of <em class="parameter"><code>func</code></em>
-
-determines what happens to the transport. <em class="parameter"><code>func</code></em>
- will be called with <em class="parameter"><code>stream</code></em>
-
-locked so no further actions on <em class="parameter"><code>stream</code></em>
- can be performed from <em class="parameter"><code>func</code></em>
-.</p>
-<p>If <em class="parameter"><code>func</code></em>
- returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REMOVE:CAPS"><span class="type">GST_RTSP_FILTER_REMOVE</span></a>, the transport will be removed from
-<em class="parameter"><code>stream</code></em>
-.</p>
-<p>If <em class="parameter"><code>func</code></em>
- returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-KEEP:CAPS"><span class="type">GST_RTSP_FILTER_KEEP</span></a>, the transport will remain in <em class="parameter"><code>stream</code></em>
-.</p>
-<p>If <em class="parameter"><code>func</code></em>
- returns <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>, the transport will remain in <em class="parameter"><code>stream</code></em>
- but
-will also be added with an additional ref to the result <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> of this
-function..</p>
-<p>When <em class="parameter"><code>func</code></em>
- is <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>, <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a> will be assumed for each transport.</p>
-<div class="refsect3">
-<a name="id-1.2.8.7.39.9"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>func</p></td>
-<td class="parameter_description"><p> a callback. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid only during the call to the method."><span class="acronym">scope call</span></acronym>][<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p> user data passed to <em class="parameter"><code>func</code></em>
-. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="This parameter is a 'user_data', for callbacks; many bindings can pass NULL here."><span class="acronym">closure</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.8.7.39.10"></a><h4>Returns</h4>
-<p> a <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> with all
-transports for which <em class="parameter"><code>func</code></em>
-returned <a class="link" href="GstRTSPSession.html#GST-RTSP-FILTER-REF:CAPS"><span class="type">GST_RTSP_FILTER_REF</span></a>. After usage, each
-element in the <a href="https://developer.gnome.org/glib/unstable/glib-Doubly-Linked-Lists.html#GList"><span class="type">GList</span></a> should be unreffed before the list is freed. </p>
-<p><span class="annotation">[<acronym title="Generics and defining elements of containers and arrays."><span class="acronym">element-type</span></acronym> GstRTSPStreamTransport][<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPStream.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstRTSPStream-struct"></a><h3>struct GstRTSPStream</h3>
-<pre class="programlisting">struct GstRTSPStream;</pre>
-<p>The definition of a media stream.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPStreamClass"></a><h3>GstRTSPStreamClass</h3>
-<pre class="programlisting">typedef struct _GstRTSPStreamClass GstRTSPStreamClass;</pre>
-</div>
-</div>
-<div class="refsect1">
-<a name="GstRTSPStream.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p>
-</div>
-</div>
-<div class="footer">
-<hr>
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-<div class="glossary">
-<div class="titlepage"><div><div><h1 class="title">
-<a name="annotation-glossary"></a>Annotation Glossary</h1></div></div></div>
-<a name="glsA"></a><h3 class="title">A</h3>
-<dt><span class="glossterm"><a name="annotation-glossterm-allow-none"></a>allow-none</span></dt>
-<dd class="glossdef"><p>NULL is OK, both for passing and for returning.</p></dd>
-<a name="glsC"></a><h3 class="title">C</h3>
-<dt><span class="glossterm"><a name="annotation-glossterm-closure"></a>closure</span></dt>
-<dd class="glossdef"><p>This parameter is a 'user_data', for callbacks; many bindings can pass NULL here.</p></dd>
-<a name="glsE"></a><h3 class="title">E</h3>
-<dt><span class="glossterm"><a name="annotation-glossterm-element-type"></a>element-type</span></dt>
-<dd class="glossdef"><p>Generics and defining elements of containers and arrays.</p></dd>
-<a name="glsN"></a><h3 class="title">N</h3>
-<dt><span class="glossterm"><a name="annotation-glossterm-nullable"></a>nullable</span></dt>
-<dd class="glossdef"><p>NULL may be passed as the value in, out, in-out; or as a return value.</p></dd>
-<a name="glsO"></a><h3 class="title">O</h3>
-<dt><span class="glossterm"><a name="annotation-glossterm-out"></a>out</span></dt>
-<dd class="glossdef"><p>Parameter for returning results. Default is <acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>.</p></dd>
-<a name="glsS"></a><h3 class="title">S</h3>
-<dt><span class="glossterm"><a name="annotation-glossterm-scope%20call"></a>scope call</span></dt>
-<dd class="glossdef"><p>The callback is valid only during the call to the method.</p></dd>
-<dt><span class="glossterm"><a name="annotation-glossterm-scope%20notified"></a>scope notified</span></dt>
-<dd class="glossdef"><p>The callback is valid until the GDestroyNotify argument is called.</p></dd>
-<a name="glsT"></a><h3 class="title">T</h3>
-<dt><span class="glossterm"><a name="annotation-glossterm-transfer%20floating"></a>transfer floating</span></dt>
-<dd class="glossdef"><p>Alias for <acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>, used for objects with floating refs.</p></dd>
-<dt><span class="glossterm"><a name="annotation-glossterm-transfer%20full"></a>transfer full</span></dt>
-<dd class="glossdef"><p>Free data after the code is done.</p></dd>
-<dt><span class="glossterm"><a name="annotation-glossterm-transfer%20none"></a>transfer none</span></dt>
-<dd class="glossdef"><p>Don't free data after the code is done.</p></dd>
-<dt><span class="glossterm"><a name="annotation-glossterm-type"></a>type</span></dt>
-<dd class="glossdef"><p>Override the parsed C type with given type.</p></dd>
-</div>
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-<hr>
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-<a name="api-index-full"></a>API Index</h1></div></div></div>
-<a name="idx"></a><a name="idxR"></a><h3 class="title">R</h3>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress">GstRTSPAddress</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressFlags" title="enum GstRTSPAddressFlags">GstRTSPAddressFlags</a>, enum in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool">GstRTSPAddressPool</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPoolClass" title="struct GstRTSPAddressPoolClass">GstRTSPAddressPoolClass</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPoolResult" title="enum GstRTSPAddressPoolResult">GstRTSPAddressPoolResult</a>, enum in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#GstRTSPAuth-struct" title="struct GstRTSPAuth">GstRTSPAuth</a>, struct in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#GstRTSPAuthClass" title="struct GstRTSPAuthClass">GstRTSPAuthClass</a>, struct in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClient-struct" title="struct GstRTSPClient">GstRTSPClient</a>, struct in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClient-closed" title="The “closed” signal">GstRTSPClient::closed</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClient-describe-request" title="The “describe-request” signal">GstRTSPClient::describe-request</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClient-get-parameter-request" title="The “get-parameter-request” signal">GstRTSPClient::get-parameter-request</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClient-handle-response" title="The “handle-response” signal">GstRTSPClient::handle-response</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClient-new-session" title="The “new-session” signal">GstRTSPClient::new-session</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClient-options-request" title="The “options-request” signal">GstRTSPClient::options-request</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClient-pause-request" title="The “pause-request” signal">GstRTSPClient::pause-request</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClient-play-request" title="The “play-request” signal">GstRTSPClient::play-request</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClient-send-message" title="The “send-message” signal">GstRTSPClient::send-message</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClient-set-parameter-request" title="The “set-parameter-request” signal">GstRTSPClient::set-parameter-request</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClient-setup-request" title="The “setup-request” signal">GstRTSPClient::setup-request</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClient-teardown-request" title="The “teardown-request” signal">GstRTSPClient::teardown-request</a>, object signal in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClient--drop-backlog" title="The “drop-backlog” property">GstRTSPClient:drop-backlog</a>, object property in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClient--mount-points" title="The “mount-points” property">GstRTSPClient:mount-points</a>, object property in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClient--session-pool" title="The “session-pool” property">GstRTSPClient:session-pool</a>, object property in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClientClass" title="struct GstRTSPClientClass">GstRTSPClientClass</a>, struct in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClientSendFunc" title="GstRTSPClientSendFunc ()">GstRTSPClientSendFunc</a>, user_function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#GstRTSPClientSessionFilterFunc" title="GstRTSPClientSessionFilterFunc ()">GstRTSPClientSessionFilterFunc</a>, user_function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext">GstRTSPContext</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPContext.html" title="GstRTSPContext">GstRTSPContext</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#GstRTSPFilterResult" title="enum GstRTSPFilterResult">GstRTSPFilterResult</a>, enum in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPKeepAliveFunc" title="GstRTSPKeepAliveFunc ()">GstRTSPKeepAliveFunc</a>, user_function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMedia-struct" title="struct GstRTSPMedia">GstRTSPMedia</a>, struct in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMedia-new-state" title="The “new-state” signal">GstRTSPMedia::new-state</a>, object signal in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMedia-new-stream" title="The “new-stream” signal">GstRTSPMedia::new-stream</a>, object signal in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMedia-prepared" title="The “prepared” signal">GstRTSPMedia::prepared</a>, object signal in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMedia-removed-stream" title="The “removed-stream” signal">GstRTSPMedia::removed-stream</a>, object signal in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMedia-target-state" title="The “target-state” signal">GstRTSPMedia::target-state</a>, object signal in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMedia-unprepared" title="The “unprepared” signal">GstRTSPMedia::unprepared</a>, object signal in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--buffer-size" title="The “buffer-size” property">GstRTSPMedia:buffer-size</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--element" title="The “element” property">GstRTSPMedia:element</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--eos-shutdown" title="The “eos-shutdown” property">GstRTSPMedia:eos-shutdown</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--profiles" title="The “profiles” property">GstRTSPMedia:profiles</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--protocols" title="The “protocols” property">GstRTSPMedia:protocols</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--reusable" title="The “reusable” property">GstRTSPMedia:reusable</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--shared" title="The “shared” property">GstRTSPMedia:shared</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--suspend-mode" title="The “suspend-mode” property">GstRTSPMedia:suspend-mode</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMedia--time-provider" title="The “time-provider” property">GstRTSPMedia:time-provider</a>, object property in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMediaClass" title="struct GstRTSPMediaClass">GstRTSPMediaClass</a>, struct in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory-struct" title="struct GstRTSPMediaFactory">GstRTSPMediaFactory</a>, struct in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory-media-configure" title="The “media-configure” signal">GstRTSPMediaFactory::media-configure</a>, object signal in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory-media-constructed" title="The “media-constructed” signal">GstRTSPMediaFactory::media-constructed</a>, object signal in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--buffer-size" title="The “buffer-size” property">GstRTSPMediaFactory:buffer-size</a>, object property in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--eos-shutdown" title="The “eos-shutdown” property">GstRTSPMediaFactory:eos-shutdown</a>, object property in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--launch" title="The “launch” property">GstRTSPMediaFactory:launch</a>, object property in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--profiles" title="The “profiles” property">GstRTSPMediaFactory:profiles</a>, object property in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--protocols" title="The “protocols” property">GstRTSPMediaFactory:protocols</a>, object property in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--shared" title="The “shared” property">GstRTSPMediaFactory:shared</a>, object property in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactory--suspend-mode" title="The “suspend-mode” property">GstRTSPMediaFactory:suspend-mode</a>, object property in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#GstRTSPMediaFactoryClass" title="struct GstRTSPMediaFactoryClass">GstRTSPMediaFactoryClass</a>, struct in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI">GstRTSPMediaFactoryURI</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html" title="GstRTSPMediaFactoryURI">GstRTSPMediaFactoryURI</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURIClass" title="struct GstRTSPMediaFactoryURIClass">GstRTSPMediaFactoryURIClass</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html" title="GstRTSPMediaFactoryURI">GstRTSPMediaFactoryURI</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPMediaStatus" title="enum GstRTSPMediaStatus">GstRTSPMediaStatus</a>, enum in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMountPoints.html#GstRTSPMountPoints-struct" title="struct GstRTSPMountPoints">GstRTSPMountPoints</a>, struct in <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints">GstRTSPMountPoints</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMountPoints.html#GstRTSPMountPointsClass" title="struct GstRTSPMountPointsClass">GstRTSPMountPointsClass</a>, struct in <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints">GstRTSPMountPoints</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions">GstRTSPPermissions</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPSendFunc" title="GstRTSPSendFunc ()">GstRTSPSendFunc</a>, user_function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#GstRTSPServer-struct" title="struct GstRTSPServer">GstRTSPServer</a>, struct in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#GstRTSPServer-client-connected" title="The “client-connected” signal">GstRTSPServer::client-connected</a>, object signal in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#GstRTSPServer--address" title="The “address” property">GstRTSPServer:address</a>, object property in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#GstRTSPServer--backlog" title="The “backlog” property">GstRTSPServer:backlog</a>, object property in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#GstRTSPServer--bound-port" title="The “bound-port” property">GstRTSPServer:bound-port</a>, object property in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#GstRTSPServer--mount-points" title="The “mount-points” property">GstRTSPServer:mount-points</a>, object property in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#GstRTSPServer--service" title="The “service” property">GstRTSPServer:service</a>, object property in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#GstRTSPServer--session-pool" title="The “session-pool” property">GstRTSPServer:session-pool</a>, object property in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#GstRTSPServerClass" title="struct GstRTSPServerClass">GstRTSPServerClass</a>, struct in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#GstRTSPServerClientFilterFunc" title="GstRTSPServerClientFilterFunc ()">GstRTSPServerClientFilterFunc</a>, user_function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#GstRTSPSession-struct" title="struct GstRTSPSession">GstRTSPSession</a>, struct in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#GstRTSPSession--sessionid" title="The “sessionid” property">GstRTSPSession:sessionid</a>, object property in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#GstRTSPSession--timeout" title="The “timeout” property">GstRTSPSession:timeout</a>, object property in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#GstRTSPSession--timeout-always-visible" title="The “timeout-always-visible” property">GstRTSPSession:timeout-always-visible</a>, object property in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#GstRTSPSessionClass" title="struct GstRTSPSessionClass">GstRTSPSessionClass</a>, struct in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#GstRTSPSessionFilterFunc" title="GstRTSPSessionFilterFunc ()">GstRTSPSessionFilterFunc</a>, user_function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia">GstRTSPSessionMedia</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMediaClass" title="struct GstRTSPSessionMediaClass">GstRTSPSessionMediaClass</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPool-struct" title="struct GstRTSPSessionPool">GstRTSPSessionPool</a>, struct in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPool-session-removed" title="The “session-removed” signal">GstRTSPSessionPool::session-removed</a>, object signal in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPool--max-sessions" title="The “max-sessions” property">GstRTSPSessionPool:max-sessions</a>, object property in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPoolClass" title="struct GstRTSPSessionPoolClass">GstRTSPSessionPoolClass</a>, struct in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPoolFilterFunc" title="GstRTSPSessionPoolFilterFunc ()">GstRTSPSessionPoolFilterFunc</a>, user_function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSessionPool.html#GstRTSPSessionPoolFunc" title="GstRTSPSessionPoolFunc ()">GstRTSPSessionPoolFunc</a>, user_function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#GstRTSPStream-struct" title="struct GstRTSPStream">GstRTSPStream</a>, struct in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#GstRTSPStreamClass" title="GstRTSPStreamClass">GstRTSPStreamClass</a>, struct in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport">GstRTSPStreamTransport</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransportClass" title="struct GstRTSPStreamTransportClass">GstRTSPStreamTransportClass</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#GstRTSPStreamTransportFilterFunc" title="GstRTSPStreamTransportFilterFunc ()">GstRTSPStreamTransportFilterFunc</a>, user_function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#GstRTSPSuspendMode" title="enum GstRTSPSuspendMode">GstRTSPSuspendMode</a>, enum in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread">GstRTSPThread</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool">GstRTSPThreadPool</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPoolClass" title="struct GstRTSPThreadPoolClass">GstRTSPThreadPoolClass</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadType" title="enum GstRTSPThreadType">GstRTSPThreadType</a>, enum in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken">GstRTSPToken</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-copy" title="gst_rtsp_address_copy ()">gst_rtsp_address_copy</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-free" title="gst_rtsp_address_free ()">gst_rtsp_address_free</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-acquire-address" title="gst_rtsp_address_pool_acquire_address ()">gst_rtsp_address_pool_acquire_address</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-add-range" title="gst_rtsp_address_pool_add_range ()">gst_rtsp_address_pool_add_range</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-ANY-IPV4:CAPS" title="GST_RTSP_ADDRESS_POOL_ANY_IPV4">GST_RTSP_ADDRESS_POOL_ANY_IPV4</a>, macro in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-ANY-IPV6:CAPS" title="GST_RTSP_ADDRESS_POOL_ANY_IPV6">GST_RTSP_ADDRESS_POOL_ANY_IPV6</a>, macro in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-clear" title="gst_rtsp_address_pool_clear ()">gst_rtsp_address_pool_clear</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-dump" title="gst_rtsp_address_pool_dump ()">gst_rtsp_address_pool_dump</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-has-unicast-addresses" title="gst_rtsp_address_pool_has_unicast_addresses ()">gst_rtsp_address_pool_has_unicast_addresses</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-new" title="gst_rtsp_address_pool_new ()">gst_rtsp_address_pool_new</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-reserve-address" title="gst_rtsp_address_pool_reserve_address ()">gst_rtsp_address_pool_reserve_address</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html" title="GstRTSPAddressPool">GstRTSPAddressPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-add-basic" title="gst_rtsp_auth_add_basic ()">gst_rtsp_auth_add_basic</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-check" title="gst_rtsp_auth_check ()">gst_rtsp_auth_check</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-CONNECT:CAPS" title="GST_RTSP_AUTH_CHECK_CONNECT">GST_RTSP_AUTH_CHECK_CONNECT</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-MEDIA-FACTORY-ACCESS:CAPS" title="GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS">GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-MEDIA-FACTORY-CONSTRUCT:CAPS" title="GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT">GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-TRANSPORT-CLIENT-SETTINGS:CAPS" title="GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS">GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-URL:CAPS" title="GST_RTSP_AUTH_CHECK_URL">GST_RTSP_AUTH_CHECK_URL</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-get-default-token" title="gst_rtsp_auth_get_default_token ()">gst_rtsp_auth_get_default_token</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-get-tls-certificate" title="gst_rtsp_auth_get_tls_certificate ()">gst_rtsp_auth_get_tls_certificate</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-make-basic" title="gst_rtsp_auth_make_basic ()">gst_rtsp_auth_make_basic</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-new" title="gst_rtsp_auth_new ()">gst_rtsp_auth_new</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-remove-basic" title="gst_rtsp_auth_remove_basic ()">gst_rtsp_auth_remove_basic</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-set-default-token" title="gst_rtsp_auth_set_default_token ()">gst_rtsp_auth_set_default_token</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#gst-rtsp-auth-set-tls-certificate" title="gst_rtsp_auth_set_tls_certificate ()">gst_rtsp_auth_set_tls_certificate</a>, function in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-attach" title="gst_rtsp_client_attach ()">gst_rtsp_client_attach</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-close" title="gst_rtsp_client_close ()">gst_rtsp_client_close</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-auth" title="gst_rtsp_client_get_auth ()">gst_rtsp_client_get_auth</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-connection" title="gst_rtsp_client_get_connection ()">gst_rtsp_client_get_connection</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-mount-points" title="gst_rtsp_client_get_mount_points ()">gst_rtsp_client_get_mount_points</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-session-pool" title="gst_rtsp_client_get_session_pool ()">gst_rtsp_client_get_session_pool</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-get-thread-pool" title="gst_rtsp_client_get_thread_pool ()">gst_rtsp_client_get_thread_pool</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-handle-message" title="gst_rtsp_client_handle_message ()">gst_rtsp_client_handle_message</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-new" title="gst_rtsp_client_new ()">gst_rtsp_client_new</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-send-message" title="gst_rtsp_client_send_message ()">gst_rtsp_client_send_message</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-session-filter" title="gst_rtsp_client_session_filter ()">gst_rtsp_client_session_filter</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-auth" title="gst_rtsp_client_set_auth ()">gst_rtsp_client_set_auth</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-connection" title="gst_rtsp_client_set_connection ()">gst_rtsp_client_set_connection</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-mount-points" title="gst_rtsp_client_set_mount_points ()">gst_rtsp_client_set_mount_points</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-send-func" title="gst_rtsp_client_set_send_func ()">gst_rtsp_client_set_send_func</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-session-pool" title="gst_rtsp_client_set_session_pool ()">gst_rtsp_client_set_session_pool</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPClient.html#gst-rtsp-client-set-thread-pool" title="gst_rtsp_client_set_thread_pool ()">gst_rtsp_client_set_thread_pool</a>, function in <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-get-current" title="gst_rtsp_context_get_current ()">gst_rtsp_context_get_current</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPContext.html" title="GstRTSPContext">GstRTSPContext</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-pop-current" title="gst_rtsp_context_pop_current ()">gst_rtsp_context_pop_current</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPContext.html" title="GstRTSPContext">GstRTSPContext</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-push-current" title="gst_rtsp_context_push_current ()">gst_rtsp_context_push_current</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPContext.html" title="GstRTSPContext">GstRTSPContext</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-collect-streams" title="gst_rtsp_media_collect_streams ()">gst_rtsp_media_collect_streams</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-create-stream" title="gst_rtsp_media_create_stream ()">gst_rtsp_media_create_stream</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-add-role" title="gst_rtsp_media_factory_add_role ()">gst_rtsp_media_factory_add_role</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-construct" title="gst_rtsp_media_factory_construct ()">gst_rtsp_media_factory_construct</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-create-element" title="gst_rtsp_media_factory_create_element ()">gst_rtsp_media_factory_create_element</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-address-pool" title="gst_rtsp_media_factory_get_address_pool ()">gst_rtsp_media_factory_get_address_pool</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-buffer-size" title="gst_rtsp_media_factory_get_buffer_size ()">gst_rtsp_media_factory_get_buffer_size</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-launch" title="gst_rtsp_media_factory_get_launch ()">gst_rtsp_media_factory_get_launch</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-permissions" title="gst_rtsp_media_factory_get_permissions ()">gst_rtsp_media_factory_get_permissions</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-profiles" title="gst_rtsp_media_factory_get_profiles ()">gst_rtsp_media_factory_get_profiles</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-protocols" title="gst_rtsp_media_factory_get_protocols ()">gst_rtsp_media_factory_get_protocols</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-suspend-mode" title="gst_rtsp_media_factory_get_suspend_mode ()">gst_rtsp_media_factory_get_suspend_mode</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-is-eos-shutdown" title="gst_rtsp_media_factory_is_eos_shutdown ()">gst_rtsp_media_factory_is_eos_shutdown</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-is-shared" title="gst_rtsp_media_factory_is_shared ()">gst_rtsp_media_factory_is_shared</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-new" title="gst_rtsp_media_factory_new ()">gst_rtsp_media_factory_new</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-address-pool" title="gst_rtsp_media_factory_set_address_pool ()">gst_rtsp_media_factory_set_address_pool</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-buffer-size" title="gst_rtsp_media_factory_set_buffer_size ()">gst_rtsp_media_factory_set_buffer_size</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-eos-shutdown" title="gst_rtsp_media_factory_set_eos_shutdown ()">gst_rtsp_media_factory_set_eos_shutdown</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-launch" title="gst_rtsp_media_factory_set_launch ()">gst_rtsp_media_factory_set_launch</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-permissions" title="gst_rtsp_media_factory_set_permissions ()">gst_rtsp_media_factory_set_permissions</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-profiles" title="gst_rtsp_media_factory_set_profiles ()">gst_rtsp_media_factory_set_profiles</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-protocols" title="gst_rtsp_media_factory_set_protocols ()">gst_rtsp_media_factory_set_protocols</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-shared" title="gst_rtsp_media_factory_set_shared ()">gst_rtsp_media_factory_set_shared</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-suspend-mode" title="gst_rtsp_media_factory_set_suspend_mode ()">gst_rtsp_media_factory_set_suspend_mode</a>, function in <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-get-uri" title="gst_rtsp_media_factory_uri_get_uri ()">gst_rtsp_media_factory_uri_get_uri</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html" title="GstRTSPMediaFactoryURI">GstRTSPMediaFactoryURI</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-new" title="gst_rtsp_media_factory_uri_new ()">gst_rtsp_media_factory_uri_new</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html" title="GstRTSPMediaFactoryURI">GstRTSPMediaFactoryURI</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-set-uri" title="gst_rtsp_media_factory_uri_set_uri ()">gst_rtsp_media_factory_uri_set_uri</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html" title="GstRTSPMediaFactoryURI">GstRTSPMediaFactoryURI</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-find-stream" title="gst_rtsp_media_find_stream ()">gst_rtsp_media_find_stream</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-address-pool" title="gst_rtsp_media_get_address_pool ()">gst_rtsp_media_get_address_pool</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-base-time" title="gst_rtsp_media_get_base_time ()">gst_rtsp_media_get_base_time</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-buffer-size" title="gst_rtsp_media_get_buffer_size ()">gst_rtsp_media_get_buffer_size</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-clock" title="gst_rtsp_media_get_clock ()">gst_rtsp_media_get_clock</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-element" title="gst_rtsp_media_get_element ()">gst_rtsp_media_get_element</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-permissions" title="gst_rtsp_media_get_permissions ()">gst_rtsp_media_get_permissions</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-profiles" title="gst_rtsp_media_get_profiles ()">gst_rtsp_media_get_profiles</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-protocols" title="gst_rtsp_media_get_protocols ()">gst_rtsp_media_get_protocols</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-range-string" title="gst_rtsp_media_get_range_string ()">gst_rtsp_media_get_range_string</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-status" title="gst_rtsp_media_get_status ()">gst_rtsp_media_get_status</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-stream" title="gst_rtsp_media_get_stream ()">gst_rtsp_media_get_stream</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-suspend-mode" title="gst_rtsp_media_get_suspend_mode ()">gst_rtsp_media_get_suspend_mode</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-get-time-provider" title="gst_rtsp_media_get_time_provider ()">gst_rtsp_media_get_time_provider</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-is-eos-shutdown" title="gst_rtsp_media_is_eos_shutdown ()">gst_rtsp_media_is_eos_shutdown</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-is-reusable" title="gst_rtsp_media_is_reusable ()">gst_rtsp_media_is_reusable</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-is-shared" title="gst_rtsp_media_is_shared ()">gst_rtsp_media_is_shared</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-is-time-provider" title="gst_rtsp_media_is_time_provider ()">gst_rtsp_media_is_time_provider</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-new" title="gst_rtsp_media_new ()">gst_rtsp_media_new</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-n-streams" title="gst_rtsp_media_n_streams ()">gst_rtsp_media_n_streams</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-prepare" title="gst_rtsp_media_prepare ()">gst_rtsp_media_prepare</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-seek" title="gst_rtsp_media_seek ()">gst_rtsp_media_seek</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-setup-sdp" title="gst_rtsp_media_setup_sdp ()">gst_rtsp_media_setup_sdp</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-address-pool" title="gst_rtsp_media_set_address_pool ()">gst_rtsp_media_set_address_pool</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-buffer-size" title="gst_rtsp_media_set_buffer_size ()">gst_rtsp_media_set_buffer_size</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-eos-shutdown" title="gst_rtsp_media_set_eos_shutdown ()">gst_rtsp_media_set_eos_shutdown</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-permissions" title="gst_rtsp_media_set_permissions ()">gst_rtsp_media_set_permissions</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-pipeline-state" title="gst_rtsp_media_set_pipeline_state ()">gst_rtsp_media_set_pipeline_state</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-profiles" title="gst_rtsp_media_set_profiles ()">gst_rtsp_media_set_profiles</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-protocols" title="gst_rtsp_media_set_protocols ()">gst_rtsp_media_set_protocols</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-reusable" title="gst_rtsp_media_set_reusable ()">gst_rtsp_media_set_reusable</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-shared" title="gst_rtsp_media_set_shared ()">gst_rtsp_media_set_shared</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-state" title="gst_rtsp_media_set_state ()">gst_rtsp_media_set_state</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-set-suspend-mode" title="gst_rtsp_media_set_suspend_mode ()">gst_rtsp_media_set_suspend_mode</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-suspend" title="gst_rtsp_media_suspend ()">gst_rtsp_media_suspend</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-take-pipeline" title="gst_rtsp_media_take_pipeline ()">gst_rtsp_media_take_pipeline</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-unprepare" title="gst_rtsp_media_unprepare ()">gst_rtsp_media_unprepare</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-unsuspend" title="gst_rtsp_media_unsuspend ()">gst_rtsp_media_unsuspend</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMedia.html#gst-rtsp-media-use-time-provider" title="gst_rtsp_media_use_time_provider ()">gst_rtsp_media_use_time_provider</a>, function in <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia">GstRTSPMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-add-factory" title="gst_rtsp_mount_points_add_factory ()">gst_rtsp_mount_points_add_factory</a>, function in <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints">GstRTSPMountPoints</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-make-path" title="gst_rtsp_mount_points_make_path ()">gst_rtsp_mount_points_make_path</a>, function in <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints">GstRTSPMountPoints</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-match" title="gst_rtsp_mount_points_match ()">gst_rtsp_mount_points_match</a>, function in <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints">GstRTSPMountPoints</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-new" title="gst_rtsp_mount_points_new ()">gst_rtsp_mount_points_new</a>, function in <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints">GstRTSPMountPoints</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPMountPoints.html#gst-rtsp-mount-points-remove-factory" title="gst_rtsp_mount_points_remove_factory ()">gst_rtsp_mount_points_remove_factory</a>, function in <a class="link" href="GstRTSPMountPoints.html" title="GstRTSPMountPoints">GstRTSPMountPoints</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPParams.html#gst-rtsp-params-get" title="gst_rtsp_params_get ()">gst_rtsp_params_get</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPParams.html" title="GstRTSPParams">GstRTSPParams</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPParams.html#gst-rtsp-params-set" title="gst_rtsp_params_set ()">gst_rtsp_params_set</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPParams.html" title="GstRTSPParams">GstRTSPParams</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-add-role" title="gst_rtsp_permissions_add_role ()">gst_rtsp_permissions_add_role</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-add-role-valist" title="gst_rtsp_permissions_add_role_valist ()">gst_rtsp_permissions_add_role_valist</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-get-role" title="gst_rtsp_permissions_get_role ()">gst_rtsp_permissions_get_role</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-is-allowed" title="gst_rtsp_permissions_is_allowed ()">gst_rtsp_permissions_is_allowed</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-new" title="gst_rtsp_permissions_new ()">gst_rtsp_permissions_new</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-ref" title="gst_rtsp_permissions_ref ()">gst_rtsp_permissions_ref</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-remove-role" title="gst_rtsp_permissions_remove_role ()">gst_rtsp_permissions_remove_role</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-unref" title="gst_rtsp_permissions_unref ()">gst_rtsp_permissions_unref</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html" title="GstRTSPPermissions">GstRTSPPermissions</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#GST-RTSP-PERM-MEDIA-FACTORY-ACCESS:CAPS" title="GST_RTSP_PERM_MEDIA_FACTORY_ACCESS">GST_RTSP_PERM_MEDIA_FACTORY_ACCESS</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#GST-RTSP-PERM-MEDIA-FACTORY-CONSTRUCT:CAPS" title="GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT">GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPSdp.html#gst-rtsp-sdp-from-media" title="gst_rtsp_sdp_from_media ()">gst_rtsp_sdp_from_media</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSdp.html" title="GstRTSPSdp">GstRTSPSdp</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-attach" title="gst_rtsp_server_attach ()">gst_rtsp_server_attach</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-client-filter" title="gst_rtsp_server_client_filter ()">gst_rtsp_server_client_filter</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-create-socket" title="gst_rtsp_server_create_socket ()">gst_rtsp_server_create_socket</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-create-source" title="gst_rtsp_server_create_source ()">gst_rtsp_server_create_source</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-address" title="gst_rtsp_server_get_address ()">gst_rtsp_server_get_address</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-auth" title="gst_rtsp_server_get_auth ()">gst_rtsp_server_get_auth</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-backlog" title="gst_rtsp_server_get_backlog ()">gst_rtsp_server_get_backlog</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-bound-port" title="gst_rtsp_server_get_bound_port ()">gst_rtsp_server_get_bound_port</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-mount-points" title="gst_rtsp_server_get_mount_points ()">gst_rtsp_server_get_mount_points</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-service" title="gst_rtsp_server_get_service ()">gst_rtsp_server_get_service</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-session-pool" title="gst_rtsp_server_get_session_pool ()">gst_rtsp_server_get_session_pool</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-get-thread-pool" title="gst_rtsp_server_get_thread_pool ()">gst_rtsp_server_get_thread_pool</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-io-func" title="gst_rtsp_server_io_func ()">gst_rtsp_server_io_func</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-new" title="gst_rtsp_server_new ()">gst_rtsp_server_new</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-address" title="gst_rtsp_server_set_address ()">gst_rtsp_server_set_address</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-auth" title="gst_rtsp_server_set_auth ()">gst_rtsp_server_set_auth</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-backlog" title="gst_rtsp_server_set_backlog ()">gst_rtsp_server_set_backlog</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-mount-points" title="gst_rtsp_server_set_mount_points ()">gst_rtsp_server_set_mount_points</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-service" title="gst_rtsp_server_set_service ()">gst_rtsp_server_set_service</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-session-pool" title="gst_rtsp_server_set_session_pool ()">gst_rtsp_server_set_session_pool</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-set-thread-pool" title="gst_rtsp_server_set_thread_pool ()">gst_rtsp_server_set_thread_pool</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPServer.html#gst-rtsp-server-transfer-connection" title="gst_rtsp_server_transfer_connection ()">gst_rtsp_server_transfer_connection</a>, function in <a class="link" href="GstRTSPServer.html" title="GstRTSPServer">GstRTSPServer</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-allow-expire" title="gst_rtsp_session_allow_expire ()">gst_rtsp_session_allow_expire</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-filter" title="gst_rtsp_session_filter ()">gst_rtsp_session_filter</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-header" title="gst_rtsp_session_get_header ()">gst_rtsp_session_get_header</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-media" title="gst_rtsp_session_get_media ()">gst_rtsp_session_get_media</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-sessionid" title="gst_rtsp_session_get_sessionid ()">gst_rtsp_session_get_sessionid</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-get-timeout" title="gst_rtsp_session_get_timeout ()">gst_rtsp_session_get_timeout</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-is-expired" title="gst_rtsp_session_is_expired ()">gst_rtsp_session_is_expired</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-manage-media" title="gst_rtsp_session_manage_media ()">gst_rtsp_session_manage_media</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-alloc-channels" title="gst_rtsp_session_media_alloc_channels ()">gst_rtsp_session_media_alloc_channels</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-base-time" title="gst_rtsp_session_media_get_base_time ()">gst_rtsp_session_media_get_base_time</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-media" title="gst_rtsp_session_media_get_media ()">gst_rtsp_session_media_get_media</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-rtpinfo" title="gst_rtsp_session_media_get_rtpinfo ()">gst_rtsp_session_media_get_rtpinfo</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-rtsp-state" title="gst_rtsp_session_media_get_rtsp_state ()">gst_rtsp_session_media_get_rtsp_state</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-transport" title="gst_rtsp_session_media_get_transport ()">gst_rtsp_session_media_get_transport</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-matches" title="gst_rtsp_session_media_matches ()">gst_rtsp_session_media_matches</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-new" title="gst_rtsp_session_media_new ()">gst_rtsp_session_media_new</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-rtsp-state" title="gst_rtsp_session_media_set_rtsp_state ()">gst_rtsp_session_media_set_rtsp_state</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-state" title="gst_rtsp_session_media_set_state ()">gst_rtsp_session_media_set_state</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-transport" title="gst_rtsp_session_media_set_transport ()">gst_rtsp_session_media_set_transport</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html" title="GstRTSPSessionMedia">GstRTSPSessionMedia</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-new" title="gst_rtsp_session_new ()">gst_rtsp_session_new</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-next-timeout" title="gst_rtsp_session_next_timeout ()">gst_rtsp_session_next_timeout</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-cleanup" title="gst_rtsp_session_pool_cleanup ()">gst_rtsp_session_pool_cleanup</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-create" title="gst_rtsp_session_pool_create ()">gst_rtsp_session_pool_create</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-create-watch" title="gst_rtsp_session_pool_create_watch ()">gst_rtsp_session_pool_create_watch</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-filter" title="gst_rtsp_session_pool_filter ()">gst_rtsp_session_pool_filter</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-find" title="gst_rtsp_session_pool_find ()">gst_rtsp_session_pool_find</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-get-max-sessions" title="gst_rtsp_session_pool_get_max_sessions ()">gst_rtsp_session_pool_get_max_sessions</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-get-n-sessions" title="gst_rtsp_session_pool_get_n_sessions ()">gst_rtsp_session_pool_get_n_sessions</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-new" title="gst_rtsp_session_pool_new ()">gst_rtsp_session_pool_new</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-remove" title="gst_rtsp_session_pool_remove ()">gst_rtsp_session_pool_remove</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSessionPool.html#gst-rtsp-session-pool-set-max-sessions" title="gst_rtsp_session_pool_set_max_sessions ()">gst_rtsp_session_pool_set_max_sessions</a>, function in <a class="link" href="GstRTSPSessionPool.html" title="GstRTSPSessionPool">GstRTSPSessionPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-prevent-expire" title="gst_rtsp_session_prevent_expire ()">gst_rtsp_session_prevent_expire</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-release-media" title="gst_rtsp_session_release_media ()">gst_rtsp_session_release_media</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-set-timeout" title="gst_rtsp_session_set_timeout ()">gst_rtsp_session_set_timeout</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPSession.html#gst-rtsp-session-touch" title="gst_rtsp_session_touch ()">gst_rtsp_session_touch</a>, function in <a class="link" href="GstRTSPSession.html" title="GstRTSPSession">GstRTSPSession</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-add-transport" title="gst_rtsp_stream_add_transport ()">gst_rtsp_stream_add_transport</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-address-pool" title="gst_rtsp_stream_get_address_pool ()">gst_rtsp_stream_get_address_pool</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-caps" title="gst_rtsp_stream_get_caps ()">gst_rtsp_stream_get_caps</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-control" title="gst_rtsp_stream_get_control ()">gst_rtsp_stream_get_control</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-dscp-qos" title="gst_rtsp_stream_get_dscp_qos ()">gst_rtsp_stream_get_dscp_qos</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-index" title="gst_rtsp_stream_get_index ()">gst_rtsp_stream_get_index</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-mtu" title="gst_rtsp_stream_get_mtu ()">gst_rtsp_stream_get_mtu</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-multicast-address" title="gst_rtsp_stream_get_multicast_address ()">gst_rtsp_stream_get_multicast_address</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-profiles" title="gst_rtsp_stream_get_profiles ()">gst_rtsp_stream_get_profiles</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-protocols" title="gst_rtsp_stream_get_protocols ()">gst_rtsp_stream_get_protocols</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-pt" title="gst_rtsp_stream_get_pt ()">gst_rtsp_stream_get_pt</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-rtcp-socket" title="gst_rtsp_stream_get_rtcp_socket ()">gst_rtsp_stream_get_rtcp_socket</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-rtpinfo" title="gst_rtsp_stream_get_rtpinfo ()">gst_rtsp_stream_get_rtpinfo</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-rtpsession" title="gst_rtsp_stream_get_rtpsession ()">gst_rtsp_stream_get_rtpsession</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-rtp-socket" title="gst_rtsp_stream_get_rtp_socket ()">gst_rtsp_stream_get_rtp_socket</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-server-port" title="gst_rtsp_stream_get_server_port ()">gst_rtsp_stream_get_server_port</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-srcpad" title="gst_rtsp_stream_get_srcpad ()">gst_rtsp_stream_get_srcpad</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-get-ssrc" title="gst_rtsp_stream_get_ssrc ()">gst_rtsp_stream_get_ssrc</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-has-control" title="gst_rtsp_stream_has_control ()">gst_rtsp_stream_has_control</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-is-blocking" title="gst_rtsp_stream_is_blocking ()">gst_rtsp_stream_is_blocking</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-is-transport-supported" title="gst_rtsp_stream_is_transport_supported ()">gst_rtsp_stream_is_transport_supported</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-join-bin" title="gst_rtsp_stream_join_bin ()">gst_rtsp_stream_join_bin</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-leave-bin" title="gst_rtsp_stream_leave_bin ()">gst_rtsp_stream_leave_bin</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-new" title="gst_rtsp_stream_new ()">gst_rtsp_stream_new</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-recv-rtcp" title="gst_rtsp_stream_recv_rtcp ()">gst_rtsp_stream_recv_rtcp</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-recv-rtp" title="gst_rtsp_stream_recv_rtp ()">gst_rtsp_stream_recv_rtp</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-remove-transport" title="gst_rtsp_stream_remove_transport ()">gst_rtsp_stream_remove_transport</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-reserve-address" title="gst_rtsp_stream_reserve_address ()">gst_rtsp_stream_reserve_address</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-address-pool" title="gst_rtsp_stream_set_address_pool ()">gst_rtsp_stream_set_address_pool</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-blocked" title="gst_rtsp_stream_set_blocked ()">gst_rtsp_stream_set_blocked</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-control" title="gst_rtsp_stream_set_control ()">gst_rtsp_stream_set_control</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-dscp-qos" title="gst_rtsp_stream_set_dscp_qos ()">gst_rtsp_stream_set_dscp_qos</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-mtu" title="gst_rtsp_stream_set_mtu ()">gst_rtsp_stream_set_mtu</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-profiles" title="gst_rtsp_stream_set_profiles ()">gst_rtsp_stream_set_profiles</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-set-protocols" title="gst_rtsp_stream_set_protocols ()">gst_rtsp_stream_set_protocols</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-transport-filter" title="gst_rtsp_stream_transport_filter ()">gst_rtsp_stream_transport_filter</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-rtpinfo" title="gst_rtsp_stream_transport_get_rtpinfo ()">gst_rtsp_stream_transport_get_rtpinfo</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-stream" title="gst_rtsp_stream_transport_get_stream ()">gst_rtsp_stream_transport_get_stream</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-transport" title="gst_rtsp_stream_transport_get_transport ()">gst_rtsp_stream_transport_get_transport</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-url" title="gst_rtsp_stream_transport_get_url ()">gst_rtsp_stream_transport_get_url</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-is-timed-out" title="gst_rtsp_stream_transport_is_timed_out ()">gst_rtsp_stream_transport_is_timed_out</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-keep-alive" title="gst_rtsp_stream_transport_keep_alive ()">gst_rtsp_stream_transport_keep_alive</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-new" title="gst_rtsp_stream_transport_new ()">gst_rtsp_stream_transport_new</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-send-rtcp" title="gst_rtsp_stream_transport_send_rtcp ()">gst_rtsp_stream_transport_send_rtcp</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-send-rtp" title="gst_rtsp_stream_transport_send_rtp ()">gst_rtsp_stream_transport_send_rtp</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-active" title="gst_rtsp_stream_transport_set_active ()">gst_rtsp_stream_transport_set_active</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-callbacks" title="gst_rtsp_stream_transport_set_callbacks ()">gst_rtsp_stream_transport_set_callbacks</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-keepalive" title="gst_rtsp_stream_transport_set_keepalive ()">gst_rtsp_stream_transport_set_keepalive</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-timed-out" title="gst_rtsp_stream_transport_set_timed_out ()">gst_rtsp_stream_transport_set_timed_out</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-transport" title="gst_rtsp_stream_transport_set_transport ()">gst_rtsp_stream_transport_set_transport</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-url" title="gst_rtsp_stream_transport_set_url ()">gst_rtsp_stream_transport_set_url</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html" title="GstRTSPStreamTransport">GstRTSPStreamTransport</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPStream.html#gst-rtsp-stream-update-crypto" title="gst_rtsp_stream_update_crypto ()">gst_rtsp_stream_update_crypto</a>, function in <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-new" title="gst_rtsp_thread_new ()">gst_rtsp_thread_new</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-cleanup" title="gst_rtsp_thread_pool_cleanup ()">gst_rtsp_thread_pool_cleanup</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-get-max-threads" title="gst_rtsp_thread_pool_get_max_threads ()">gst_rtsp_thread_pool_get_max_threads</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-get-thread" title="gst_rtsp_thread_pool_get_thread ()">gst_rtsp_thread_pool_get_thread</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-new" title="gst_rtsp_thread_pool_new ()">gst_rtsp_thread_pool_new</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-set-max-threads" title="gst_rtsp_thread_pool_set_max_threads ()">gst_rtsp_thread_pool_set_max_threads</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-ref" title="gst_rtsp_thread_ref ()">gst_rtsp_thread_ref</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-reuse" title="gst_rtsp_thread_reuse ()">gst_rtsp_thread_reuse</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-stop" title="gst_rtsp_thread_stop ()">gst_rtsp_thread_stop</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-unref" title="gst_rtsp_thread_unref ()">gst_rtsp_thread_unref</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html" title="GstRTSPThreadPool">GstRTSPThreadPool</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-get-string" title="gst_rtsp_token_get_string ()">gst_rtsp_token_get_string</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-get-structure" title="gst_rtsp_token_get_structure ()">gst_rtsp_token_get_structure</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-is-allowed" title="gst_rtsp_token_is_allowed ()">gst_rtsp_token_is_allowed</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#GST-RTSP-TOKEN-MEDIA-FACTORY-ROLE:CAPS" title="GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE">GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new" title="gst_rtsp_token_new ()">gst_rtsp_token_new</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new-empty" title="gst_rtsp_token_new_empty ()">gst_rtsp_token_new_empty</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new-valist" title="gst_rtsp_token_new_valist ()">gst_rtsp_token_new_valist</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-ref" title="gst_rtsp_token_ref ()">gst_rtsp_token_ref</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="GstRTSPAuth.html#GST-RTSP-TOKEN-TRANSPORT-CLIENT-SETTINGS:CAPS" title="GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS">GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS</a>, macro in <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth">GstRTSPAuth</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-unref" title="gst_rtsp_token_unref ()">gst_rtsp_token_unref</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
-</dt>
-<dd></dd>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-writable-structure" title="gst_rtsp_token_writable_structure ()">gst_rtsp_token_writable_structure</a>, function in <a class="link" href="gst-rtsp-server-GstRTSPToken.html" title="GstRTSPToken">GstRTSPToken</a>
-</dt>
-<dd></dd>
-<a name="idxS"></a><h3 class="title">S</h3>
-<dt>
-<a class="link" href="gst-rtsp-server-GstRTSPSdp.html#GstSDPInfo" title="GstSDPInfo">GstSDPInfo</a>, struct in <a class="link" href="gst-rtsp-server-GstRTSPSdp.html" title="GstRTSPSdp">GstRTSPSdp</a>
-</dt>
-<dd></dd>
-</div>
-<div class="footer">
-<hr>
- Generated by GTK-Doc V1.21</div>
-</body>
-</html>
\ No newline at end of file
+++ /dev/null
-<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
-<html>
-<head>
-<meta http-equiv="Content-Type" content="text/html; charset=UTF-8">
-<title>GStreamer RTSP Server Reference Manual: </title>
-<meta name="generator" content="DocBook XSL Stylesheets V1.78.1">
-<link rel="home" href="index.html" title="GStreamer RTSP Server Reference Manual">
-<link rel="up" href="index.html" title="GStreamer RTSP Server Reference Manual">
-<link rel="prev" href="index.html" title="GStreamer RTSP Server Reference Manual">
-<link rel="next" href="GstRTSPServer.html" title="GstRTSPServer">
-<meta name="generator" content="GTK-Doc V1.21 (XML mode)">
-<link rel="stylesheet" href="style.css" type="text/css">
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-<body bgcolor="white" text="black" link="#0000FF" vlink="#840084" alink="#0000FF">
-<table class="navigation" id="top" width="100%" summary="Navigation header" cellpadding="2" cellspacing="5"><tr valign="middle">
-<td width="100%" align="left" class="shortcuts"></td>
-<td><a accesskey="h" href="index.html"><img src="home.png" width="16" height="16" border="0" alt="Home"></a></td>
-<td><img src="up-insensitive.png" width="16" height="16" border="0"></td>
-<td><a accesskey="p" href="index.html"><img src="left.png" width="16" height="16" border="0" alt="Prev"></a></td>
-<td><a accesskey="n" href="GstRTSPServer.html"><img src="right.png" width="16" height="16" border="0" alt="Next"></a></td>
-</tr></table>
-<div class="chapter">
-<div class="titlepage"></div>
-<div class="toc"><dl class="toc">
-<dt>
-<span class="refentrytitle"><a href="GstRTSPServer.html">GstRTSPServer</a></span><span class="refpurpose"> — The main server object</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPClient.html">GstRTSPClient</a></span><span class="refpurpose"> — A client connection state</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPContext.html">GstRTSPContext</a></span><span class="refpurpose"> — A client request context</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPMountPoints.html">GstRTSPMountPoints</a></span><span class="refpurpose"> — Map a path to media</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPMediaFactory.html">GstRTSPMediaFactory</a></span><span class="refpurpose"> — A factory for media pipelines</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPMediaFactoryURI.html">GstRTSPMediaFactoryURI</a></span><span class="refpurpose"> — A factory for URI sources</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPMedia.html">GstRTSPMedia</a></span><span class="refpurpose"> — The media pipeline</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPStream.html">GstRTSPStream</a></span><span class="refpurpose"> — A media stream</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPSessionPool.html">GstRTSPSessionPool</a></span><span class="refpurpose"> — An object for managing sessions</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPSession.html">GstRTSPSession</a></span><span class="refpurpose"> — An object to manage media</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPSessionMedia.html">GstRTSPSessionMedia</a></span><span class="refpurpose"> — Media managed in a session</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPStreamTransport.html">GstRTSPStreamTransport</a></span><span class="refpurpose"> — A media stream transport configuration</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPSdp.html">GstRTSPSdp</a></span><span class="refpurpose"> — Make SDP messages</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPAddressPool.html">GstRTSPAddressPool</a></span><span class="refpurpose"> — A pool of network addresses</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPThreadPool.html">GstRTSPThreadPool</a></span><span class="refpurpose"> — A pool of threads</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPAuth.html">GstRTSPAuth</a></span><span class="refpurpose"> — Authentication and authorization</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPToken.html">GstRTSPToken</a></span><span class="refpurpose"> — Roles and permissions for a client</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPPermissions.html">GstRTSPPermissions</a></span><span class="refpurpose"> — Roles and associated permissions</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPParams.html">GstRTSPParams</a></span><span class="refpurpose"> — Param get and set implementation</span>
-</dt>
-</dl></div>
-</div>
-<div class="footer">
-<hr>
- Generated by GTK-Doc V1.21</div>
-</body>
-</html>
\ No newline at end of file
+++ /dev/null
-<?xml version="1.0" encoding="utf-8" standalone="no"?>
-<!DOCTYPE book PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN" "">
-<book xmlns="http://www.devhelp.net/book" title="GStreamer RTSP Server Reference Manual" link="index.html" author="" name="gst-rtsp-server-1.0" version="2" language="c">
- <chapters>
- <sub name="" link="ch01.html">
- <sub name="GstRTSPServer" link="GstRTSPServer.html"/>
- <sub name="GstRTSPClient" link="GstRTSPClient.html"/>
- <sub name="GstRTSPContext" link="gst-rtsp-server-GstRTSPContext.html"/>
- <sub name="GstRTSPMountPoints" link="GstRTSPMountPoints.html"/>
- <sub name="GstRTSPMediaFactory" link="GstRTSPMediaFactory.html"/>
- <sub name="GstRTSPMediaFactoryURI" link="gst-rtsp-server-GstRTSPMediaFactoryURI.html"/>
- <sub name="GstRTSPMedia" link="GstRTSPMedia.html"/>
- <sub name="GstRTSPStream" link="GstRTSPStream.html"/>
- <sub name="GstRTSPSessionPool" link="GstRTSPSessionPool.html"/>
- <sub name="GstRTSPSession" link="GstRTSPSession.html"/>
- <sub name="GstRTSPSessionMedia" link="gst-rtsp-server-GstRTSPSessionMedia.html"/>
- <sub name="GstRTSPStreamTransport" link="gst-rtsp-server-GstRTSPStreamTransport.html"/>
- <sub name="GstRTSPSdp" link="gst-rtsp-server-GstRTSPSdp.html"/>
- <sub name="GstRTSPAddressPool" link="gst-rtsp-server-GstRTSPAddressPool.html"/>
- <sub name="GstRTSPThreadPool" link="gst-rtsp-server-GstRTSPThreadPool.html"/>
- <sub name="GstRTSPAuth" link="GstRTSPAuth.html"/>
- <sub name="GstRTSPToken" link="gst-rtsp-server-GstRTSPToken.html"/>
- <sub name="GstRTSPPermissions" link="gst-rtsp-server-GstRTSPPermissions.html"/>
- <sub name="GstRTSPParams" link="gst-rtsp-server-GstRTSPParams.html"/>
- </sub>
- <sub name="Object Hierarchy" link="rtsp-server-hierarchy.html"/>
- <sub name="API Index" link="api-index-full.html"/>
- <sub name="Annotation Glossary" link="annotation-glossary.html"/>
- </chapters>
- <functions>
- <keyword type="function" name="gst_rtsp_server_new ()" link="GstRTSPServer.html#gst-rtsp-server-new"/>
- <keyword type="function" name="gst_rtsp_server_get_address ()" link="GstRTSPServer.html#gst-rtsp-server-get-address"/>
- <keyword type="function" name="gst_rtsp_server_set_address ()" link="GstRTSPServer.html#gst-rtsp-server-set-address"/>
- <keyword type="function" name="gst_rtsp_server_get_service ()" link="GstRTSPServer.html#gst-rtsp-server-get-service"/>
- <keyword type="function" name="gst_rtsp_server_set_service ()" link="GstRTSPServer.html#gst-rtsp-server-set-service"/>
- <keyword type="function" name="gst_rtsp_server_get_backlog ()" link="GstRTSPServer.html#gst-rtsp-server-get-backlog"/>
- <keyword type="function" name="gst_rtsp_server_set_backlog ()" link="GstRTSPServer.html#gst-rtsp-server-set-backlog"/>
- <keyword type="function" name="gst_rtsp_server_get_bound_port ()" link="GstRTSPServer.html#gst-rtsp-server-get-bound-port"/>
- <keyword type="function" name="gst_rtsp_server_get_mount_points ()" link="GstRTSPServer.html#gst-rtsp-server-get-mount-points"/>
- <keyword type="function" name="gst_rtsp_server_set_mount_points ()" link="GstRTSPServer.html#gst-rtsp-server-set-mount-points"/>
- <keyword type="function" name="gst_rtsp_server_get_session_pool ()" link="GstRTSPServer.html#gst-rtsp-server-get-session-pool"/>
- <keyword type="function" name="gst_rtsp_server_set_session_pool ()" link="GstRTSPServer.html#gst-rtsp-server-set-session-pool"/>
- <keyword type="function" name="gst_rtsp_server_get_thread_pool ()" link="GstRTSPServer.html#gst-rtsp-server-get-thread-pool"/>
- <keyword type="function" name="gst_rtsp_server_set_thread_pool ()" link="GstRTSPServer.html#gst-rtsp-server-set-thread-pool"/>
- <keyword type="function" name="gst_rtsp_server_get_auth ()" link="GstRTSPServer.html#gst-rtsp-server-get-auth"/>
- <keyword type="function" name="gst_rtsp_server_set_auth ()" link="GstRTSPServer.html#gst-rtsp-server-set-auth"/>
- <keyword type="function" name="gst_rtsp_server_transfer_connection ()" link="GstRTSPServer.html#gst-rtsp-server-transfer-connection"/>
- <keyword type="function" name="gst_rtsp_server_io_func ()" link="GstRTSPServer.html#gst-rtsp-server-io-func"/>
- <keyword type="function" name="gst_rtsp_server_create_socket ()" link="GstRTSPServer.html#gst-rtsp-server-create-socket"/>
- <keyword type="function" name="gst_rtsp_server_create_source ()" link="GstRTSPServer.html#gst-rtsp-server-create-source"/>
- <keyword type="function" name="gst_rtsp_server_attach ()" link="GstRTSPServer.html#gst-rtsp-server-attach"/>
- <keyword type="function" name="GstRTSPServerClientFilterFunc ()" link="GstRTSPServer.html#GstRTSPServerClientFilterFunc"/>
- <keyword type="function" name="gst_rtsp_server_client_filter ()" link="GstRTSPServer.html#gst-rtsp-server-client-filter"/>
- <keyword type="struct" name="struct GstRTSPServer" link="GstRTSPServer.html#GstRTSPServer-struct"/>
- <keyword type="struct" name="struct GstRTSPServerClass" link="GstRTSPServer.html#GstRTSPServerClass"/>
- <keyword type="property" name="The “address” property" link="GstRTSPServer.html#GstRTSPServer--address"/>
- <keyword type="property" name="The “backlog” property" link="GstRTSPServer.html#GstRTSPServer--backlog"/>
- <keyword type="property" name="The “bound-port” property" link="GstRTSPServer.html#GstRTSPServer--bound-port"/>
- <keyword type="property" name="The “mount-points” property" link="GstRTSPServer.html#GstRTSPServer--mount-points"/>
- <keyword type="property" name="The “service” property" link="GstRTSPServer.html#GstRTSPServer--service"/>
- <keyword type="property" name="The “session-pool” property" link="GstRTSPServer.html#GstRTSPServer--session-pool"/>
- <keyword type="signal" name="The “client-connected” signal" link="GstRTSPServer.html#GstRTSPServer-client-connected"/>
- <keyword type="function" name="gst_rtsp_client_new ()" link="GstRTSPClient.html#gst-rtsp-client-new"/>
- <keyword type="function" name="gst_rtsp_client_close ()" link="GstRTSPClient.html#gst-rtsp-client-close" since="1.4"/>
- <keyword type="function" name="gst_rtsp_client_get_session_pool ()" link="GstRTSPClient.html#gst-rtsp-client-get-session-pool"/>
- <keyword type="function" name="gst_rtsp_client_set_session_pool ()" link="GstRTSPClient.html#gst-rtsp-client-set-session-pool"/>
- <keyword type="function" name="gst_rtsp_client_get_mount_points ()" link="GstRTSPClient.html#gst-rtsp-client-get-mount-points"/>
- <keyword type="function" name="gst_rtsp_client_set_mount_points ()" link="GstRTSPClient.html#gst-rtsp-client-set-mount-points"/>
- <keyword type="function" name="gst_rtsp_client_get_auth ()" link="GstRTSPClient.html#gst-rtsp-client-get-auth"/>
- <keyword type="function" name="gst_rtsp_client_set_auth ()" link="GstRTSPClient.html#gst-rtsp-client-set-auth"/>
- <keyword type="function" name="gst_rtsp_client_get_thread_pool ()" link="GstRTSPClient.html#gst-rtsp-client-get-thread-pool"/>
- <keyword type="function" name="gst_rtsp_client_set_thread_pool ()" link="GstRTSPClient.html#gst-rtsp-client-set-thread-pool"/>
- <keyword type="function" name="gst_rtsp_client_get_connection ()" link="GstRTSPClient.html#gst-rtsp-client-get-connection"/>
- <keyword type="function" name="gst_rtsp_client_set_connection ()" link="GstRTSPClient.html#gst-rtsp-client-set-connection"/>
- <keyword type="function" name="gst_rtsp_client_attach ()" link="GstRTSPClient.html#gst-rtsp-client-attach"/>
- <keyword type="function" name="GstRTSPClientSendFunc ()" link="GstRTSPClient.html#GstRTSPClientSendFunc"/>
- <keyword type="function" name="gst_rtsp_client_set_send_func ()" link="GstRTSPClient.html#gst-rtsp-client-set-send-func"/>
- <keyword type="function" name="gst_rtsp_client_handle_message ()" link="GstRTSPClient.html#gst-rtsp-client-handle-message"/>
- <keyword type="function" name="gst_rtsp_client_send_message ()" link="GstRTSPClient.html#gst-rtsp-client-send-message"/>
- <keyword type="function" name="GstRTSPClientSessionFilterFunc ()" link="GstRTSPClient.html#GstRTSPClientSessionFilterFunc"/>
- <keyword type="function" name="gst_rtsp_client_session_filter ()" link="GstRTSPClient.html#gst-rtsp-client-session-filter"/>
- <keyword type="struct" name="struct GstRTSPClient" link="GstRTSPClient.html#GstRTSPClient-struct"/>
- <keyword type="struct" name="struct GstRTSPClientClass" link="GstRTSPClient.html#GstRTSPClientClass"/>
- <keyword type="property" name="The “drop-backlog” property" link="GstRTSPClient.html#GstRTSPClient--drop-backlog"/>
- <keyword type="property" name="The “mount-points” property" link="GstRTSPClient.html#GstRTSPClient--mount-points"/>
- <keyword type="property" name="The “session-pool” property" link="GstRTSPClient.html#GstRTSPClient--session-pool"/>
- <keyword type="signal" name="The “closed” signal" link="GstRTSPClient.html#GstRTSPClient-closed"/>
- <keyword type="signal" name="The “describe-request” signal" link="GstRTSPClient.html#GstRTSPClient-describe-request"/>
- <keyword type="signal" name="The “get-parameter-request” signal" link="GstRTSPClient.html#GstRTSPClient-get-parameter-request"/>
- <keyword type="signal" name="The “handle-response” signal" link="GstRTSPClient.html#GstRTSPClient-handle-response"/>
- <keyword type="signal" name="The “new-session” signal" link="GstRTSPClient.html#GstRTSPClient-new-session"/>
- <keyword type="signal" name="The “options-request” signal" link="GstRTSPClient.html#GstRTSPClient-options-request"/>
- <keyword type="signal" name="The “pause-request” signal" link="GstRTSPClient.html#GstRTSPClient-pause-request"/>
- <keyword type="signal" name="The “play-request” signal" link="GstRTSPClient.html#GstRTSPClient-play-request"/>
- <keyword type="signal" name="The “send-message” signal" link="GstRTSPClient.html#GstRTSPClient-send-message"/>
- <keyword type="signal" name="The “set-parameter-request” signal" link="GstRTSPClient.html#GstRTSPClient-set-parameter-request"/>
- <keyword type="signal" name="The “setup-request” signal" link="GstRTSPClient.html#GstRTSPClient-setup-request"/>
- <keyword type="signal" name="The “teardown-request” signal" link="GstRTSPClient.html#GstRTSPClient-teardown-request"/>
- <keyword type="function" name="gst_rtsp_context_get_current ()" link="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-get-current"/>
- <keyword type="function" name="gst_rtsp_context_push_current ()" link="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-push-current"/>
- <keyword type="function" name="gst_rtsp_context_pop_current ()" link="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-pop-current"/>
- <keyword type="struct" name="struct GstRTSPContext" link="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext"/>
- <keyword type="function" name="gst_rtsp_mount_points_new ()" link="GstRTSPMountPoints.html#gst-rtsp-mount-points-new"/>
- <keyword type="function" name="gst_rtsp_mount_points_add_factory ()" link="GstRTSPMountPoints.html#gst-rtsp-mount-points-add-factory"/>
- <keyword type="function" name="gst_rtsp_mount_points_remove_factory ()" link="GstRTSPMountPoints.html#gst-rtsp-mount-points-remove-factory"/>
- <keyword type="function" name="gst_rtsp_mount_points_match ()" link="GstRTSPMountPoints.html#gst-rtsp-mount-points-match"/>
- <keyword type="function" name="gst_rtsp_mount_points_make_path ()" link="GstRTSPMountPoints.html#gst-rtsp-mount-points-make-path"/>
- <keyword type="struct" name="struct GstRTSPMountPoints" link="GstRTSPMountPoints.html#GstRTSPMountPoints-struct"/>
- <keyword type="struct" name="struct GstRTSPMountPointsClass" link="GstRTSPMountPoints.html#GstRTSPMountPointsClass"/>
- <keyword type="function" name="gst_rtsp_media_factory_new ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-new"/>
- <keyword type="function" name="gst_rtsp_media_factory_get_launch ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-launch"/>
- <keyword type="function" name="gst_rtsp_media_factory_set_launch ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-launch"/>
- <keyword type="function" name="gst_rtsp_media_factory_get_permissions ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-permissions"/>
- <keyword type="function" name="gst_rtsp_media_factory_set_permissions ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-permissions"/>
- <keyword type="function" name="gst_rtsp_media_factory_add_role ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-add-role"/>
- <keyword type="function" name="gst_rtsp_media_factory_set_shared ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-shared"/>
- <keyword type="function" name="gst_rtsp_media_factory_is_shared ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-is-shared"/>
- <keyword type="function" name="gst_rtsp_media_factory_is_eos_shutdown ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-is-eos-shutdown"/>
- <keyword type="function" name="gst_rtsp_media_factory_set_eos_shutdown ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-eos-shutdown"/>
- <keyword type="function" name="gst_rtsp_media_factory_get_protocols ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-protocols"/>
- <keyword type="function" name="gst_rtsp_media_factory_set_protocols ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-protocols"/>
- <keyword type="function" name="gst_rtsp_media_factory_set_profiles ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-profiles"/>
- <keyword type="function" name="gst_rtsp_media_factory_get_profiles ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-profiles"/>
- <keyword type="function" name="gst_rtsp_media_factory_get_address_pool ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-address-pool"/>
- <keyword type="function" name="gst_rtsp_media_factory_set_address_pool ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-address-pool"/>
- <keyword type="function" name="gst_rtsp_media_factory_get_buffer_size ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-buffer-size"/>
- <keyword type="function" name="gst_rtsp_media_factory_set_buffer_size ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-buffer-size"/>
- <keyword type="function" name="gst_rtsp_media_factory_get_suspend_mode ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-get-suspend-mode"/>
- <keyword type="function" name="gst_rtsp_media_factory_set_suspend_mode ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-set-suspend-mode"/>
- <keyword type="function" name="gst_rtsp_media_factory_construct ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-construct"/>
- <keyword type="function" name="gst_rtsp_media_factory_create_element ()" link="GstRTSPMediaFactory.html#gst-rtsp-media-factory-create-element"/>
- <keyword type="struct" name="struct GstRTSPMediaFactory" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory-struct"/>
- <keyword type="struct" name="struct GstRTSPMediaFactoryClass" link="GstRTSPMediaFactory.html#GstRTSPMediaFactoryClass"/>
- <keyword type="property" name="The “buffer-size” property" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory--buffer-size"/>
- <keyword type="property" name="The “eos-shutdown” property" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory--eos-shutdown"/>
- <keyword type="property" name="The “launch” property" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory--launch"/>
- <keyword type="property" name="The “profiles” property" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory--profiles"/>
- <keyword type="property" name="The “protocols” property" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory--protocols"/>
- <keyword type="property" name="The “shared” property" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory--shared"/>
- <keyword type="property" name="The “suspend-mode” property" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory--suspend-mode"/>
- <keyword type="signal" name="The “media-configure” signal" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory-media-configure"/>
- <keyword type="signal" name="The “media-constructed” signal" link="GstRTSPMediaFactory.html#GstRTSPMediaFactory-media-constructed"/>
- <keyword type="function" name="gst_rtsp_media_factory_uri_new ()" link="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-new"/>
- <keyword type="function" name="gst_rtsp_media_factory_uri_set_uri ()" link="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-set-uri"/>
- <keyword type="function" name="gst_rtsp_media_factory_uri_get_uri ()" link="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-get-uri"/>
- <keyword type="struct" name="struct GstRTSPMediaFactoryURI" link="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI"/>
- <keyword type="struct" name="struct GstRTSPMediaFactoryURIClass" link="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURIClass"/>
- <keyword type="function" name="gst_rtsp_media_new ()" link="GstRTSPMedia.html#gst-rtsp-media-new"/>
- <keyword type="function" name="gst_rtsp_media_get_element ()" link="GstRTSPMedia.html#gst-rtsp-media-get-element"/>
- <keyword type="function" name="gst_rtsp_media_take_pipeline ()" link="GstRTSPMedia.html#gst-rtsp-media-take-pipeline"/>
- <keyword type="function" name="gst_rtsp_media_set_permissions ()" link="GstRTSPMedia.html#gst-rtsp-media-set-permissions"/>
- <keyword type="function" name="gst_rtsp_media_get_permissions ()" link="GstRTSPMedia.html#gst-rtsp-media-get-permissions"/>
- <keyword type="function" name="gst_rtsp_media_set_shared ()" link="GstRTSPMedia.html#gst-rtsp-media-set-shared"/>
- <keyword type="function" name="gst_rtsp_media_is_shared ()" link="GstRTSPMedia.html#gst-rtsp-media-is-shared"/>
- <keyword type="function" name="gst_rtsp_media_set_reusable ()" link="GstRTSPMedia.html#gst-rtsp-media-set-reusable"/>
- <keyword type="function" name="gst_rtsp_media_is_reusable ()" link="GstRTSPMedia.html#gst-rtsp-media-is-reusable"/>
- <keyword type="function" name="gst_rtsp_media_set_profiles ()" link="GstRTSPMedia.html#gst-rtsp-media-set-profiles"/>
- <keyword type="function" name="gst_rtsp_media_get_profiles ()" link="GstRTSPMedia.html#gst-rtsp-media-get-profiles"/>
- <keyword type="function" name="gst_rtsp_media_set_protocols ()" link="GstRTSPMedia.html#gst-rtsp-media-set-protocols"/>
- <keyword type="function" name="gst_rtsp_media_get_protocols ()" link="GstRTSPMedia.html#gst-rtsp-media-get-protocols"/>
- <keyword type="function" name="gst_rtsp_media_set_eos_shutdown ()" link="GstRTSPMedia.html#gst-rtsp-media-set-eos-shutdown"/>
- <keyword type="function" name="gst_rtsp_media_is_eos_shutdown ()" link="GstRTSPMedia.html#gst-rtsp-media-is-eos-shutdown"/>
- <keyword type="function" name="gst_rtsp_media_set_address_pool ()" link="GstRTSPMedia.html#gst-rtsp-media-set-address-pool"/>
- <keyword type="function" name="gst_rtsp_media_get_address_pool ()" link="GstRTSPMedia.html#gst-rtsp-media-get-address-pool"/>
- <keyword type="function" name="gst_rtsp_media_set_buffer_size ()" link="GstRTSPMedia.html#gst-rtsp-media-set-buffer-size"/>
- <keyword type="function" name="gst_rtsp_media_get_buffer_size ()" link="GstRTSPMedia.html#gst-rtsp-media-get-buffer-size"/>
- <keyword type="function" name="gst_rtsp_media_setup_sdp ()" link="GstRTSPMedia.html#gst-rtsp-media-setup-sdp"/>
- <keyword type="function" name="gst_rtsp_media_prepare ()" link="GstRTSPMedia.html#gst-rtsp-media-prepare"/>
- <keyword type="function" name="gst_rtsp_media_unprepare ()" link="GstRTSPMedia.html#gst-rtsp-media-unprepare"/>
- <keyword type="function" name="gst_rtsp_media_get_status ()" link="GstRTSPMedia.html#gst-rtsp-media-get-status"/>
- <keyword type="function" name="gst_rtsp_media_set_suspend_mode ()" link="GstRTSPMedia.html#gst-rtsp-media-set-suspend-mode"/>
- <keyword type="function" name="gst_rtsp_media_get_suspend_mode ()" link="GstRTSPMedia.html#gst-rtsp-media-get-suspend-mode"/>
- <keyword type="function" name="gst_rtsp_media_suspend ()" link="GstRTSPMedia.html#gst-rtsp-media-suspend"/>
- <keyword type="function" name="gst_rtsp_media_unsuspend ()" link="GstRTSPMedia.html#gst-rtsp-media-unsuspend"/>
- <keyword type="function" name="gst_rtsp_media_collect_streams ()" link="GstRTSPMedia.html#gst-rtsp-media-collect-streams"/>
- <keyword type="function" name="gst_rtsp_media_create_stream ()" link="GstRTSPMedia.html#gst-rtsp-media-create-stream"/>
- <keyword type="function" name="gst_rtsp_media_n_streams ()" link="GstRTSPMedia.html#gst-rtsp-media-n-streams"/>
- <keyword type="function" name="gst_rtsp_media_get_stream ()" link="GstRTSPMedia.html#gst-rtsp-media-get-stream"/>
- <keyword type="function" name="gst_rtsp_media_find_stream ()" link="GstRTSPMedia.html#gst-rtsp-media-find-stream"/>
- <keyword type="function" name="gst_rtsp_media_seek ()" link="GstRTSPMedia.html#gst-rtsp-media-seek"/>
- <keyword type="function" name="gst_rtsp_media_get_range_string ()" link="GstRTSPMedia.html#gst-rtsp-media-get-range-string"/>
- <keyword type="function" name="gst_rtsp_media_set_state ()" link="GstRTSPMedia.html#gst-rtsp-media-set-state"/>
- <keyword type="function" name="gst_rtsp_media_set_pipeline_state ()" link="GstRTSPMedia.html#gst-rtsp-media-set-pipeline-state"/>
- <keyword type="function" name="gst_rtsp_media_get_clock ()" link="GstRTSPMedia.html#gst-rtsp-media-get-clock"/>
- <keyword type="function" name="gst_rtsp_media_get_base_time ()" link="GstRTSPMedia.html#gst-rtsp-media-get-base-time"/>
- <keyword type="function" name="gst_rtsp_media_use_time_provider ()" link="GstRTSPMedia.html#gst-rtsp-media-use-time-provider"/>
- <keyword type="function" name="gst_rtsp_media_is_time_provider ()" link="GstRTSPMedia.html#gst-rtsp-media-is-time-provider"/>
- <keyword type="function" name="gst_rtsp_media_get_time_provider ()" link="GstRTSPMedia.html#gst-rtsp-media-get-time-provider"/>
- <keyword type="struct" name="struct GstRTSPMedia" link="GstRTSPMedia.html#GstRTSPMedia-struct"/>
- <keyword type="struct" name="struct GstRTSPMediaClass" link="GstRTSPMedia.html#GstRTSPMediaClass"/>
- <keyword type="enum" name="enum GstRTSPMediaStatus" link="GstRTSPMedia.html#GstRTSPMediaStatus"/>
- <keyword type="enum" name="enum GstRTSPSuspendMode" link="GstRTSPMedia.html#GstRTSPSuspendMode"/>
- <keyword type="property" name="The “buffer-size” property" link="GstRTSPMedia.html#GstRTSPMedia--buffer-size"/>
- <keyword type="property" name="The “element” property" link="GstRTSPMedia.html#GstRTSPMedia--element"/>
- <keyword type="property" name="The “eos-shutdown” property" link="GstRTSPMedia.html#GstRTSPMedia--eos-shutdown"/>
- <keyword type="property" name="The “profiles” property" link="GstRTSPMedia.html#GstRTSPMedia--profiles"/>
- <keyword type="property" name="The “protocols” property" link="GstRTSPMedia.html#GstRTSPMedia--protocols"/>
- <keyword type="property" name="The “reusable” property" link="GstRTSPMedia.html#GstRTSPMedia--reusable"/>
- <keyword type="property" name="The “shared” property" link="GstRTSPMedia.html#GstRTSPMedia--shared"/>
- <keyword type="property" name="The “suspend-mode” property" link="GstRTSPMedia.html#GstRTSPMedia--suspend-mode"/>
- <keyword type="property" name="The “time-provider” property" link="GstRTSPMedia.html#GstRTSPMedia--time-provider"/>
- <keyword type="signal" name="The “new-state” signal" link="GstRTSPMedia.html#GstRTSPMedia-new-state"/>
- <keyword type="signal" name="The “new-stream” signal" link="GstRTSPMedia.html#GstRTSPMedia-new-stream"/>
- <keyword type="signal" name="The “prepared” signal" link="GstRTSPMedia.html#GstRTSPMedia-prepared"/>
- <keyword type="signal" name="The “removed-stream” signal" link="GstRTSPMedia.html#GstRTSPMedia-removed-stream"/>
- <keyword type="signal" name="The “target-state” signal" link="GstRTSPMedia.html#GstRTSPMedia-target-state"/>
- <keyword type="signal" name="The “unprepared” signal" link="GstRTSPMedia.html#GstRTSPMedia-unprepared"/>
- <keyword type="function" name="gst_rtsp_stream_new ()" link="GstRTSPStream.html#gst-rtsp-stream-new"/>
- <keyword type="function" name="gst_rtsp_stream_get_index ()" link="GstRTSPStream.html#gst-rtsp-stream-get-index"/>
- <keyword type="function" name="gst_rtsp_stream_get_srcpad ()" link="GstRTSPStream.html#gst-rtsp-stream-get-srcpad"/>
- <keyword type="function" name="gst_rtsp_stream_get_control ()" link="GstRTSPStream.html#gst-rtsp-stream-get-control"/>
- <keyword type="function" name="gst_rtsp_stream_set_control ()" link="GstRTSPStream.html#gst-rtsp-stream-set-control"/>
- <keyword type="function" name="gst_rtsp_stream_has_control ()" link="GstRTSPStream.html#gst-rtsp-stream-has-control"/>
- <keyword type="function" name="gst_rtsp_stream_get_mtu ()" link="GstRTSPStream.html#gst-rtsp-stream-get-mtu"/>
- <keyword type="function" name="gst_rtsp_stream_set_mtu ()" link="GstRTSPStream.html#gst-rtsp-stream-set-mtu"/>
- <keyword type="function" name="gst_rtsp_stream_get_dscp_qos ()" link="GstRTSPStream.html#gst-rtsp-stream-get-dscp-qos"/>
- <keyword type="function" name="gst_rtsp_stream_set_dscp_qos ()" link="GstRTSPStream.html#gst-rtsp-stream-set-dscp-qos"/>
- <keyword type="function" name="gst_rtsp_stream_set_profiles ()" link="GstRTSPStream.html#gst-rtsp-stream-set-profiles"/>
- <keyword type="function" name="gst_rtsp_stream_get_profiles ()" link="GstRTSPStream.html#gst-rtsp-stream-get-profiles"/>
- <keyword type="function" name="gst_rtsp_stream_get_protocols ()" link="GstRTSPStream.html#gst-rtsp-stream-get-protocols"/>
- <keyword type="function" name="gst_rtsp_stream_set_protocols ()" link="GstRTSPStream.html#gst-rtsp-stream-set-protocols"/>
- <keyword type="function" name="gst_rtsp_stream_is_transport_supported ()" link="GstRTSPStream.html#gst-rtsp-stream-is-transport-supported"/>
- <keyword type="function" name="gst_rtsp_stream_get_address_pool ()" link="GstRTSPStream.html#gst-rtsp-stream-get-address-pool"/>
- <keyword type="function" name="gst_rtsp_stream_set_address_pool ()" link="GstRTSPStream.html#gst-rtsp-stream-set-address-pool"/>
- <keyword type="function" name="gst_rtsp_stream_reserve_address ()" link="GstRTSPStream.html#gst-rtsp-stream-reserve-address"/>
- <keyword type="function" name="gst_rtsp_stream_join_bin ()" link="GstRTSPStream.html#gst-rtsp-stream-join-bin"/>
- <keyword type="function" name="gst_rtsp_stream_leave_bin ()" link="GstRTSPStream.html#gst-rtsp-stream-leave-bin"/>
- <keyword type="function" name="gst_rtsp_stream_get_server_port ()" link="GstRTSPStream.html#gst-rtsp-stream-get-server-port"/>
- <keyword type="function" name="gst_rtsp_stream_get_multicast_address ()" link="GstRTSPStream.html#gst-rtsp-stream-get-multicast-address"/>
- <keyword type="function" name="gst_rtsp_stream_get_rtpsession ()" link="GstRTSPStream.html#gst-rtsp-stream-get-rtpsession"/>
- <keyword type="function" name="gst_rtsp_stream_get_ssrc ()" link="GstRTSPStream.html#gst-rtsp-stream-get-ssrc"/>
- <keyword type="function" name="gst_rtsp_stream_get_rtpinfo ()" link="GstRTSPStream.html#gst-rtsp-stream-get-rtpinfo"/>
- <keyword type="function" name="gst_rtsp_stream_get_caps ()" link="GstRTSPStream.html#gst-rtsp-stream-get-caps"/>
- <keyword type="function" name="gst_rtsp_stream_get_pt ()" link="GstRTSPStream.html#gst-rtsp-stream-get-pt"/>
- <keyword type="function" name="gst_rtsp_stream_recv_rtcp ()" link="GstRTSPStream.html#gst-rtsp-stream-recv-rtcp"/>
- <keyword type="function" name="gst_rtsp_stream_recv_rtp ()" link="GstRTSPStream.html#gst-rtsp-stream-recv-rtp"/>
- <keyword type="function" name="gst_rtsp_stream_add_transport ()" link="GstRTSPStream.html#gst-rtsp-stream-add-transport"/>
- <keyword type="function" name="gst_rtsp_stream_remove_transport ()" link="GstRTSPStream.html#gst-rtsp-stream-remove-transport"/>
- <keyword type="function" name="gst_rtsp_stream_get_rtp_socket ()" link="GstRTSPStream.html#gst-rtsp-stream-get-rtp-socket"/>
- <keyword type="function" name="gst_rtsp_stream_get_rtcp_socket ()" link="GstRTSPStream.html#gst-rtsp-stream-get-rtcp-socket"/>
- <keyword type="function" name="gst_rtsp_stream_set_blocked ()" link="GstRTSPStream.html#gst-rtsp-stream-set-blocked"/>
- <keyword type="function" name="gst_rtsp_stream_is_blocking ()" link="GstRTSPStream.html#gst-rtsp-stream-is-blocking"/>
- <keyword type="function" name="gst_rtsp_stream_update_crypto ()" link="GstRTSPStream.html#gst-rtsp-stream-update-crypto"/>
- <keyword type="function" name="GstRTSPStreamTransportFilterFunc ()" link="GstRTSPStream.html#GstRTSPStreamTransportFilterFunc"/>
- <keyword type="function" name="gst_rtsp_stream_transport_filter ()" link="GstRTSPStream.html#gst-rtsp-stream-transport-filter"/>
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- <keyword type="function" name="gst_rtsp_session_pool_get_n_sessions ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-get-n-sessions"/>
- <keyword type="function" name="gst_rtsp_session_pool_create ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-create"/>
- <keyword type="function" name="gst_rtsp_session_pool_find ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-find"/>
- <keyword type="function" name="gst_rtsp_session_pool_remove ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-remove"/>
- <keyword type="function" name="gst_rtsp_session_pool_cleanup ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-cleanup"/>
- <keyword type="function" name="GstRTSPSessionPoolFunc ()" link="GstRTSPSessionPool.html#GstRTSPSessionPoolFunc"/>
- <keyword type="function" name="gst_rtsp_session_pool_create_watch ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-create-watch"/>
- <keyword type="function" name="GstRTSPSessionPoolFilterFunc ()" link="GstRTSPSessionPool.html#GstRTSPSessionPoolFilterFunc"/>
- <keyword type="function" name="gst_rtsp_session_pool_filter ()" link="GstRTSPSessionPool.html#gst-rtsp-session-pool-filter"/>
- <keyword type="struct" name="struct GstRTSPSessionPool" link="GstRTSPSessionPool.html#GstRTSPSessionPool-struct"/>
- <keyword type="struct" name="struct GstRTSPSessionPoolClass" link="GstRTSPSessionPool.html#GstRTSPSessionPoolClass"/>
- <keyword type="property" name="The “max-sessions” property" link="GstRTSPSessionPool.html#GstRTSPSessionPool--max-sessions"/>
- <keyword type="signal" name="The “session-removed” signal" link="GstRTSPSessionPool.html#GstRTSPSessionPool-session-removed"/>
- <keyword type="function" name="gst_rtsp_session_new ()" link="GstRTSPSession.html#gst-rtsp-session-new"/>
- <keyword type="function" name="gst_rtsp_session_get_sessionid ()" link="GstRTSPSession.html#gst-rtsp-session-get-sessionid"/>
- <keyword type="function" name="gst_rtsp_session_get_header ()" link="GstRTSPSession.html#gst-rtsp-session-get-header"/>
- <keyword type="function" name="gst_rtsp_session_set_timeout ()" link="GstRTSPSession.html#gst-rtsp-session-set-timeout"/>
- <keyword type="function" name="gst_rtsp_session_get_timeout ()" link="GstRTSPSession.html#gst-rtsp-session-get-timeout"/>
- <keyword type="function" name="gst_rtsp_session_touch ()" link="GstRTSPSession.html#gst-rtsp-session-touch"/>
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- <keyword type="function" name="gst_rtsp_session_allow_expire ()" link="GstRTSPSession.html#gst-rtsp-session-allow-expire"/>
- <keyword type="function" name="gst_rtsp_session_next_timeout ()" link="GstRTSPSession.html#gst-rtsp-session-next-timeout"/>
- <keyword type="function" name="gst_rtsp_session_is_expired ()" link="GstRTSPSession.html#gst-rtsp-session-is-expired"/>
- <keyword type="function" name="gst_rtsp_session_manage_media ()" link="GstRTSPSession.html#gst-rtsp-session-manage-media"/>
- <keyword type="function" name="gst_rtsp_session_release_media ()" link="GstRTSPSession.html#gst-rtsp-session-release-media"/>
- <keyword type="function" name="gst_rtsp_session_get_media ()" link="GstRTSPSession.html#gst-rtsp-session-get-media"/>
- <keyword type="function" name="GstRTSPSessionFilterFunc ()" link="GstRTSPSession.html#GstRTSPSessionFilterFunc"/>
- <keyword type="function" name="gst_rtsp_session_filter ()" link="GstRTSPSession.html#gst-rtsp-session-filter"/>
- <keyword type="struct" name="struct GstRTSPSession" link="GstRTSPSession.html#GstRTSPSession-struct"/>
- <keyword type="struct" name="struct GstRTSPSessionClass" link="GstRTSPSession.html#GstRTSPSessionClass"/>
- <keyword type="enum" name="enum GstRTSPFilterResult" link="GstRTSPSession.html#GstRTSPFilterResult"/>
- <keyword type="property" name="The “sessionid” property" link="GstRTSPSession.html#GstRTSPSession--sessionid"/>
- <keyword type="property" name="The “timeout” property" link="GstRTSPSession.html#GstRTSPSession--timeout"/>
- <keyword type="property" name="The “timeout-always-visible” property" link="GstRTSPSession.html#GstRTSPSession--timeout-always-visible"/>
- <keyword type="function" name="gst_rtsp_session_media_new ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-new"/>
- <keyword type="function" name="gst_rtsp_session_media_matches ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-matches"/>
- <keyword type="function" name="gst_rtsp_session_media_get_media ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-media"/>
- <keyword type="function" name="gst_rtsp_session_media_get_base_time ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-base-time"/>
- <keyword type="function" name="gst_rtsp_session_media_get_rtpinfo ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-rtpinfo"/>
- <keyword type="function" name="gst_rtsp_session_media_set_state ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-state"/>
- <keyword type="function" name="gst_rtsp_session_media_get_rtsp_state ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-rtsp-state"/>
- <keyword type="function" name="gst_rtsp_session_media_set_rtsp_state ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-rtsp-state"/>
- <keyword type="function" name="gst_rtsp_session_media_get_transport ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-transport"/>
- <keyword type="function" name="gst_rtsp_session_media_set_transport ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-transport"/>
- <keyword type="function" name="gst_rtsp_session_media_alloc_channels ()" link="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-alloc-channels"/>
- <keyword type="struct" name="struct GstRTSPSessionMedia" link="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia"/>
- <keyword type="struct" name="struct GstRTSPSessionMediaClass" link="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMediaClass"/>
- <keyword type="function" name="gst_rtsp_stream_transport_new ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-new"/>
- <keyword type="function" name="gst_rtsp_stream_transport_get_stream ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-stream"/>
- <keyword type="function" name="gst_rtsp_stream_transport_get_transport ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-transport"/>
- <keyword type="function" name="gst_rtsp_stream_transport_set_transport ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-transport"/>
- <keyword type="function" name="gst_rtsp_stream_transport_get_url ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-url"/>
- <keyword type="function" name="gst_rtsp_stream_transport_set_url ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-url"/>
- <keyword type="function" name="gst_rtsp_stream_transport_get_rtpinfo ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-rtpinfo"/>
- <keyword type="function" name="GstRTSPSendFunc ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPSendFunc"/>
- <keyword type="function" name="gst_rtsp_stream_transport_set_callbacks ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-callbacks"/>
- <keyword type="function" name="GstRTSPKeepAliveFunc ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPKeepAliveFunc"/>
- <keyword type="function" name="gst_rtsp_stream_transport_set_keepalive ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-keepalive"/>
- <keyword type="function" name="gst_rtsp_stream_transport_keep_alive ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-keep-alive"/>
- <keyword type="function" name="gst_rtsp_stream_transport_set_active ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-active"/>
- <keyword type="function" name="gst_rtsp_stream_transport_set_timed_out ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-timed-out"/>
- <keyword type="function" name="gst_rtsp_stream_transport_is_timed_out ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-is-timed-out"/>
- <keyword type="function" name="gst_rtsp_stream_transport_send_rtcp ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-send-rtcp"/>
- <keyword type="function" name="gst_rtsp_stream_transport_send_rtp ()" link="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-send-rtp"/>
- <keyword type="struct" name="struct GstRTSPStreamTransport" link="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport"/>
- <keyword type="struct" name="struct GstRTSPStreamTransportClass" link="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransportClass"/>
- <keyword type="function" name="gst_rtsp_sdp_from_media ()" link="gst-rtsp-server-GstRTSPSdp.html#gst-rtsp-sdp-from-media"/>
- <keyword type="struct" name="GstSDPInfo" link="gst-rtsp-server-GstRTSPSdp.html#GstSDPInfo"/>
- <keyword type="function" name="gst_rtsp_address_copy ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-copy"/>
- <keyword type="function" name="gst_rtsp_address_free ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-free"/>
- <keyword type="function" name="gst_rtsp_address_pool_new ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-new"/>
- <keyword type="function" name="gst_rtsp_address_pool_clear ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-clear"/>
- <keyword type="function" name="gst_rtsp_address_pool_dump ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-dump"/>
- <keyword type="function" name="gst_rtsp_address_pool_add_range ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-add-range"/>
- <keyword type="function" name="gst_rtsp_address_pool_has_unicast_addresses ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-has-unicast-addresses"/>
- <keyword type="function" name="gst_rtsp_address_pool_acquire_address ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-acquire-address"/>
- <keyword type="function" name="gst_rtsp_address_pool_reserve_address ()" link="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-reserve-address"/>
- <keyword type="macro" name="GST_RTSP_ADDRESS_POOL_ANY_IPV4" link="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-ANY-IPV4:CAPS"/>
- <keyword type="macro" name="GST_RTSP_ADDRESS_POOL_ANY_IPV6" link="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-ANY-IPV6:CAPS"/>
- <keyword type="struct" name="struct GstRTSPAddress" link="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress"/>
- <keyword type="enum" name="enum GstRTSPAddressFlags" link="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressFlags"/>
- <keyword type="struct" name="struct GstRTSPAddressPool" link="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool"/>
- <keyword type="struct" name="struct GstRTSPAddressPoolClass" link="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPoolClass"/>
- <keyword type="enum" name="enum GstRTSPAddressPoolResult" link="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPoolResult"/>
- <keyword type="function" name="gst_rtsp_thread_new ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-new"/>
- <keyword type="function" name="gst_rtsp_thread_ref ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-ref"/>
- <keyword type="function" name="gst_rtsp_thread_unref ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-unref"/>
- <keyword type="function" name="gst_rtsp_thread_reuse ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-reuse"/>
- <keyword type="function" name="gst_rtsp_thread_stop ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-stop"/>
- <keyword type="function" name="gst_rtsp_thread_pool_new ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-new"/>
- <keyword type="function" name="gst_rtsp_thread_pool_get_max_threads ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-get-max-threads"/>
- <keyword type="function" name="gst_rtsp_thread_pool_set_max_threads ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-set-max-threads"/>
- <keyword type="function" name="gst_rtsp_thread_pool_get_thread ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-get-thread"/>
- <keyword type="function" name="gst_rtsp_thread_pool_cleanup ()" link="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-cleanup"/>
- <keyword type="enum" name="enum GstRTSPThreadType" link="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadType"/>
- <keyword type="struct" name="struct GstRTSPThread" link="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread"/>
- <keyword type="struct" name="struct GstRTSPThreadPool" link="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool"/>
- <keyword type="struct" name="struct GstRTSPThreadPoolClass" link="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPoolClass"/>
- <keyword type="function" name="gst_rtsp_auth_new ()" link="GstRTSPAuth.html#gst-rtsp-auth-new"/>
- <keyword type="function" name="gst_rtsp_auth_get_tls_certificate ()" link="GstRTSPAuth.html#gst-rtsp-auth-get-tls-certificate"/>
- <keyword type="function" name="gst_rtsp_auth_set_tls_certificate ()" link="GstRTSPAuth.html#gst-rtsp-auth-set-tls-certificate"/>
- <keyword type="function" name="gst_rtsp_auth_make_basic ()" link="GstRTSPAuth.html#gst-rtsp-auth-make-basic"/>
- <keyword type="function" name="gst_rtsp_auth_add_basic ()" link="GstRTSPAuth.html#gst-rtsp-auth-add-basic"/>
- <keyword type="function" name="gst_rtsp_auth_remove_basic ()" link="GstRTSPAuth.html#gst-rtsp-auth-remove-basic"/>
- <keyword type="function" name="gst_rtsp_auth_check ()" link="GstRTSPAuth.html#gst-rtsp-auth-check"/>
- <keyword type="function" name="gst_rtsp_auth_get_default_token ()" link="GstRTSPAuth.html#gst-rtsp-auth-get-default-token"/>
- <keyword type="function" name="gst_rtsp_auth_set_default_token ()" link="GstRTSPAuth.html#gst-rtsp-auth-set-default-token"/>
- <keyword type="struct" name="struct GstRTSPAuth" link="GstRTSPAuth.html#GstRTSPAuth-struct"/>
- <keyword type="struct" name="struct GstRTSPAuthClass" link="GstRTSPAuth.html#GstRTSPAuthClass"/>
- <keyword type="macro" name="GST_RTSP_AUTH_CHECK_CONNECT" link="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-CONNECT:CAPS"/>
- <keyword type="macro" name="GST_RTSP_AUTH_CHECK_URL" link="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-URL:CAPS"/>
- <keyword type="macro" name="GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_ACCESS" link="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-MEDIA-FACTORY-ACCESS:CAPS"/>
- <keyword type="macro" name="GST_RTSP_AUTH_CHECK_MEDIA_FACTORY_CONSTRUCT" link="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-MEDIA-FACTORY-CONSTRUCT:CAPS"/>
- <keyword type="macro" name="GST_RTSP_AUTH_CHECK_TRANSPORT_CLIENT_SETTINGS" link="GstRTSPAuth.html#GST-RTSP-AUTH-CHECK-TRANSPORT-CLIENT-SETTINGS:CAPS"/>
- <keyword type="macro" name="GST_RTSP_TOKEN_MEDIA_FACTORY_ROLE" link="GstRTSPAuth.html#GST-RTSP-TOKEN-MEDIA-FACTORY-ROLE:CAPS"/>
- <keyword type="macro" name="GST_RTSP_TOKEN_TRANSPORT_CLIENT_SETTINGS" link="GstRTSPAuth.html#GST-RTSP-TOKEN-TRANSPORT-CLIENT-SETTINGS:CAPS"/>
- <keyword type="macro" name="GST_RTSP_PERM_MEDIA_FACTORY_ACCESS" link="GstRTSPAuth.html#GST-RTSP-PERM-MEDIA-FACTORY-ACCESS:CAPS"/>
- <keyword type="macro" name="GST_RTSP_PERM_MEDIA_FACTORY_CONSTRUCT" link="GstRTSPAuth.html#GST-RTSP-PERM-MEDIA-FACTORY-CONSTRUCT:CAPS"/>
- <keyword type="function" name="gst_rtsp_token_new_empty ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new-empty"/>
- <keyword type="function" name="gst_rtsp_token_new ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new"/>
- <keyword type="function" name="gst_rtsp_token_new_valist ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new-valist"/>
- <keyword type="function" name="gst_rtsp_token_ref ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-ref"/>
- <keyword type="function" name="gst_rtsp_token_unref ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-unref"/>
- <keyword type="function" name="gst_rtsp_token_get_structure ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-get-structure"/>
- <keyword type="function" name="gst_rtsp_token_writable_structure ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-writable-structure"/>
- <keyword type="function" name="gst_rtsp_token_get_string ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-get-string"/>
- <keyword type="function" name="gst_rtsp_token_is_allowed ()" link="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-is-allowed"/>
- <keyword type="struct" name="struct GstRTSPToken" link="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken"/>
- <keyword type="function" name="gst_rtsp_permissions_new ()" link="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-new"/>
- <keyword type="function" name="gst_rtsp_permissions_ref ()" link="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-ref"/>
- <keyword type="function" name="gst_rtsp_permissions_unref ()" link="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-unref"/>
- <keyword type="function" name="gst_rtsp_permissions_add_role ()" link="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-add-role"/>
- <keyword type="function" name="gst_rtsp_permissions_add_role_valist ()" link="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-add-role-valist"/>
- <keyword type="function" name="gst_rtsp_permissions_remove_role ()" link="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-remove-role"/>
- <keyword type="function" name="gst_rtsp_permissions_get_role ()" link="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-get-role"/>
- <keyword type="function" name="gst_rtsp_permissions_is_allowed ()" link="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-is-allowed"/>
- <keyword type="struct" name="struct GstRTSPPermissions" link="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions"/>
- <keyword type="function" name="gst_rtsp_params_get ()" link="gst-rtsp-server-GstRTSPParams.html#gst-rtsp-params-get"/>
- <keyword type="function" name="gst_rtsp_params_set ()" link="gst-rtsp-server-GstRTSPParams.html#gst-rtsp-params-set"/>
- </functions>
-</book>
+++ /dev/null
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-<a name="gst-rtsp-server-GstRTSPAddressPool"></a><div class="titlepage"></div>
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-<td valign="top">
-<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPAddressPool.top_of_page"></a>GstRTSPAddressPool</span></h2>
-<p>GstRTSPAddressPool — A pool of network addresses</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPAddressPool.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="returnvalue">GstRTSPAddress</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-copy" title="gst_rtsp_address_copy ()">gst_rtsp_address_copy</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-free" title="gst_rtsp_address_free ()">gst_rtsp_address_free</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="returnvalue">GstRTSPAddressPool</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-new" title="gst_rtsp_address_pool_new ()">gst_rtsp_address_pool_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-clear" title="gst_rtsp_address_pool_clear ()">gst_rtsp_address_pool_clear</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-dump" title="gst_rtsp_address_pool_dump ()">gst_rtsp_address_pool_dump</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-add-range" title="gst_rtsp_address_pool_add_range ()">gst_rtsp_address_pool_add_range</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-has-unicast-addresses" title="gst_rtsp_address_pool_has_unicast_addresses ()">gst_rtsp_address_pool_has_unicast_addresses</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="returnvalue">GstRTSPAddress</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-acquire-address" title="gst_rtsp_address_pool_acquire_address ()">gst_rtsp_address_pool_acquire_address</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPoolResult" title="enum GstRTSPAddressPoolResult"><span class="returnvalue">GstRTSPAddressPoolResult</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-reserve-address" title="gst_rtsp_address_pool_reserve_address ()">gst_rtsp_address_pool_reserve_address</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPAddressPool.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody>
-<tr>
-<td class="define_keyword">#define</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-ANY-IPV4:CAPS" title="GST_RTSP_ADDRESS_POOL_ANY_IPV4">GST_RTSP_ADDRESS_POOL_ANY_IPV4</a></td>
-</tr>
-<tr>
-<td class="define_keyword">#define</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-ANY-IPV6:CAPS" title="GST_RTSP_ADDRESS_POOL_ANY_IPV6">GST_RTSP_ADDRESS_POOL_ANY_IPV6</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress">GstRTSPAddress</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">enum</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressFlags" title="enum GstRTSPAddressFlags">GstRTSPAddressFlags</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool">GstRTSPAddressPool</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPoolClass" title="struct GstRTSPAddressPoolClass">GstRTSPAddressPoolClass</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">enum</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPoolResult" title="enum GstRTSPAddressPoolResult">GstRTSPAddressPoolResult</a></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPAddressPool.description"></a><h2>Description</h2>
-<p>The <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> 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.</p>
-<p>A range of addresses can be added with <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-add-range" title="gst_rtsp_address_pool_add_range ()"><code class="function">gst_rtsp_address_pool_add_range()</code></a>.
-Both multicast and unicast addresses can be added.</p>
-<p>With <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-acquire-address" title="gst_rtsp_address_pool_acquire_address ()"><code class="function">gst_rtsp_address_pool_acquire_address()</code></a> an unused address and port range
-can be acquired from the pool. With <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-reserve-address" title="gst_rtsp_address_pool_reserve_address ()"><code class="function">gst_rtsp_address_pool_reserve_address()</code></a> a
-specific address can be retrieved. Both methods return a boxed
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a> that should be freed with <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-free" title="gst_rtsp_address_free ()"><code class="function">gst_rtsp_address_free()</code></a> after
-usage, which brings the address back into the pool.</p>
-<p>Last reviewed on 2013-07-16 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPAddressPool.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-address-copy"></a><h3>gst_rtsp_address_copy ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="returnvalue">GstRTSPAddress</span></a> *
-gst_rtsp_address_copy (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a> *addr</code></em>);</pre>
-<p>Make a copy of <em class="parameter"><code>addr</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.14.6.2.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>addr</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.14.6.2.6"></a><h4>Returns</h4>
-<p> a copy of <em class="parameter"><code>addr</code></em>
-.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-address-free"></a><h3>gst_rtsp_address_free ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_address_free (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a> *addr</code></em>);</pre>
-<p>Free <em class="parameter"><code>addr</code></em>
- and releasing it back into the pool when owned by a
-pool.</p>
-<div class="refsect3">
-<a name="id-1.2.14.6.3.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>addr</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-address-pool-new"></a><h3>gst_rtsp_address_pool_new ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="returnvalue">GstRTSPAddressPool</span></a> *
-gst_rtsp_address_pool_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
-<p>Make a new <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.14.6.4.5"></a><h4>Returns</h4>
-<p> a new <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a>. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-address-pool-clear"></a><h3>gst_rtsp_address_pool_clear ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_address_pool_clear (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>);</pre>
-<p>Clear all addresses in <em class="parameter"><code>pool</code></em>
-. There should be no outstanding
-allocations.</p>
-<div class="refsect3">
-<a name="id-1.2.14.6.5.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-address-pool-dump"></a><h3>gst_rtsp_address_pool_dump ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_address_pool_dump (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>);</pre>
-<p>Dump the free and allocated addresses to stdout.</p>
-<div class="refsect3">
-<a name="id-1.2.14.6.6.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-address-pool-add-range"></a><h3>gst_rtsp_address_pool_add_range ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_address_pool_add_range (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *min_address</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *max_address</code></em>,
- <em class="parameter"><code><span class="type">guint16</span> min_port</code></em>,
- <em class="parameter"><code><span class="type">guint16</span> max_port</code></em>,
- <em class="parameter"><code><span class="type">guint8</span> ttl</code></em>);</pre>
-<p>Adds the addresses from <em class="parameter"><code>min_addess</code></em>
- to <em class="parameter"><code>max_address</code></em>
- (inclusive)
-to <em class="parameter"><code>pool</code></em>
-. The valid port range for the addresses will be from <em class="parameter"><code>min_port</code></em>
- to
-<em class="parameter"><code>max_port</code></em>
- inclusive.</p>
-<p>When <em class="parameter"><code>ttl</code></em>
- is 0, <em class="parameter"><code>min_address</code></em>
- and <em class="parameter"><code>max_address</code></em>
- should be unicast addresses.
-<em class="parameter"><code>min_address</code></em>
- and <em class="parameter"><code>max_address</code></em>
- can be set to
-<a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-ANY-IPV4:CAPS" title="GST_RTSP_ADDRESS_POOL_ANY_IPV4"><span class="type">GST_RTSP_ADDRESS_POOL_ANY_IPV4</span></a> or <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-ANY-IPV6:CAPS" title="GST_RTSP_ADDRESS_POOL_ANY_IPV6"><span class="type">GST_RTSP_ADDRESS_POOL_ANY_IPV6</span></a> to bind
-to all available IPv4 or IPv6 addresses.</p>
-<p>When <em class="parameter"><code>ttl</code></em>
- > 0, <em class="parameter"><code>min_address</code></em>
- and <em class="parameter"><code>max_address</code></em>
- should be multicast addresses.</p>
-<div class="refsect3">
-<a name="id-1.2.14.6.7.7"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>min_address</p></td>
-<td class="parameter_description"><p>a minimum address to add</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>max_address</p></td>
-<td class="parameter_description"><p>a maximum address to add</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>min_port</p></td>
-<td class="parameter_description"><p>the minimum port</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>max_port</p></td>
-<td class="parameter_description"><p>the maximum port</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>ttl</p></td>
-<td class="parameter_description"><p>a TTL or 0 for unicast addresses</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.14.6.7.8"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the addresses could be added.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-address-pool-has-unicast-addresses"></a><h3>gst_rtsp_address_pool_has_unicast_addresses ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_address_pool_has_unicast_addresses
- (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>);</pre>
-<p>Used to know if the pool includes any unicast addresses.</p>
-<div class="refsect3">
-<a name="id-1.2.14.6.8.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.14.6.8.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the pool includes any unicast addresses, <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#FALSE:CAPS"><code class="literal">FALSE</code></a> otherwise</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-address-pool-acquire-address"></a><h3>gst_rtsp_address_pool_acquire_address ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="returnvalue">GstRTSPAddress</span></a> *
-gst_rtsp_address_pool_acquire_address (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressFlags" title="enum GstRTSPAddressFlags"><span class="type">GstRTSPAddressFlags</span></a> flags</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> n_ports</code></em>);</pre>
-<p>Take an address and ports from <em class="parameter"><code>pool</code></em>
-. <em class="parameter"><code>flags</code></em>
- can be used to control the
-allocation. <em class="parameter"><code>n_ports</code></em>
- consecutive ports will be allocated of which the first
-one can be found in <em class="parameter"><code>port</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.14.6.9.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>flags</p></td>
-<td class="parameter_description"><p>flags</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>n_ports</p></td>
-<td class="parameter_description"><p>the amount of ports</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.14.6.9.6"></a><h4>Returns</h4>
-<p> a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a> that should be freed with
-gst_rtsp_address_free after use or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when no address could be
-acquired. </p>
-<p><span class="annotation">[<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-address-pool-reserve-address"></a><h3>gst_rtsp_address_pool_reserve_address ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPoolResult" title="enum GstRTSPAddressPoolResult"><span class="returnvalue">GstRTSPAddressPoolResult</span></a>
-gst_rtsp_address_pool_reserve_address (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *pool</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *ip_address</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> port</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> n_ports</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> ttl</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a> **address</code></em>);</pre>
-<p>Take a specific address and ports from <em class="parameter"><code>pool</code></em>
-. <em class="parameter"><code>n_ports</code></em>
- consecutive
-ports will be allocated of which the first one can be found in
-<em class="parameter"><code>port</code></em>
-.</p>
-<p>If <em class="parameter"><code>ttl</code></em>
- is 0, <em class="parameter"><code>address</code></em>
- should be a unicast address. If <em class="parameter"><code>ttl</code></em>
- > 0, <em class="parameter"><code>address</code></em>
-
-should be a valid multicast address.</p>
-<div class="refsect3">
-<a name="id-1.2.14.6.10.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>ip_address</p></td>
-<td class="parameter_description"><p>The IP address to reserve</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>port</p></td>
-<td class="parameter_description"><p>The first port to reserve</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>n_ports</p></td>
-<td class="parameter_description"><p>The number of ports</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>ttl</p></td>
-<td class="parameter_description"><p>The requested ttl</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>address</p></td>
-<td class="parameter_description"><p> storage for a <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddress" title="struct GstRTSPAddress"><span class="type">GstRTSPAddress</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.14.6.10.7"></a><h4>Returns</h4>
-<p> <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GST-RTSP-ADDRESS-POOL-OK:CAPS"><span class="type">GST_RTSP_ADDRESS_POOL_OK</span></a> if an address was reserved. The address
-is returned in <em class="parameter"><code>address</code></em>
-and should be freed with gst_rtsp_address_free
-after use.</p>
-<p></p>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPAddressPool.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GST-RTSP-ADDRESS-POOL-ANY-IPV4:CAPS"></a><h3>GST_RTSP_ADDRESS_POOL_ANY_IPV4</h3>
-<pre class="programlisting">#define GST_RTSP_ADDRESS_POOL_ANY_IPV4 "0.0.0.0"
-</pre>
-<p>Used with <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-add-range" title="gst_rtsp_address_pool_add_range ()"><code class="function">gst_rtsp_address_pool_add_range()</code></a> to bind to all
-IPv4 addresses</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GST-RTSP-ADDRESS-POOL-ANY-IPV6:CAPS"></a><h3>GST_RTSP_ADDRESS_POOL_ANY_IPV6</h3>
-<pre class="programlisting">#define GST_RTSP_ADDRESS_POOL_ANY_IPV6 "::"
-</pre>
-<p>Used with <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#gst-rtsp-address-pool-add-range" title="gst_rtsp_address_pool_add_range ()"><code class="function">gst_rtsp_address_pool_add_range()</code></a> to bind to all
-IPv6 addresses</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPAddress"></a><h3>struct GstRTSPAddress</h3>
-<pre class="programlisting">struct GstRTSPAddress {
- GstRTSPAddressPool *pool;
-
- gchar *address;
- guint16 port;
- gint n_ports;
- guint8 ttl;
-};
-</pre>
-<p>An address</p>
-<div class="refsect3">
-<a name="id-1.2.14.7.4.5"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="struct_members_name">
-<col class="struct_members_description">
-<col width="200px" class="struct_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="struct_member_name"><p><a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> *<em class="structfield"><code><a name="GstRTSPAddress.pool"></a>pool</code></em>;</p></td>
-<td class="struct_member_description"><p>the <a class="link" href="gst-rtsp-server-GstRTSPAddressPool.html#GstRTSPAddressPool" title="struct GstRTSPAddressPool"><span class="type">GstRTSPAddressPool</span></a> owner of this address</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *<em class="structfield"><code><a name="GstRTSPAddress.address"></a>address</code></em>;</p></td>
-<td class="struct_member_description"><p>the address</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><span class="type">guint16</span> <em class="structfield"><code><a name="GstRTSPAddress.port"></a>port</code></em>;</p></td>
-<td class="struct_member_description"><p>the port number</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> <em class="structfield"><code><a name="GstRTSPAddress.n-ports"></a>n_ports</code></em>;</p></td>
-<td class="struct_member_description"><p>number of ports</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><span class="type">guint8</span> <em class="structfield"><code><a name="GstRTSPAddress.ttl"></a>ttl</code></em>;</p></td>
-<td class="struct_member_description"><p>TTL or 0 for unicast addresses</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPAddressFlags"></a><h3>enum GstRTSPAddressFlags</h3>
-<p>Flags used to control allocation of addresses</p>
-<div class="refsect3">
-<a name="id-1.2.14.7.5.4"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="enum_members_name">
-<col class="enum_members_description">
-<col width="200px" class="enum_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-FLAG-NONE:CAPS"></a>GST_RTSP_ADDRESS_FLAG_NONE</p></td>
-<td class="enum_member_description">
-<p>no flags</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-FLAG-IPV4:CAPS"></a>GST_RTSP_ADDRESS_FLAG_IPV4</p></td>
-<td class="enum_member_description">
-<p>an IPv4 address</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-FLAG-IPV6:CAPS"></a>GST_RTSP_ADDRESS_FLAG_IPV6</p></td>
-<td class="enum_member_description">
-<p>and IPv6 address</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-FLAG-EVEN-PORT:CAPS"></a>GST_RTSP_ADDRESS_FLAG_EVEN_PORT</p></td>
-<td class="enum_member_description">
-<p>address with an even port</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-FLAG-MULTICAST:CAPS"></a>GST_RTSP_ADDRESS_FLAG_MULTICAST</p></td>
-<td class="enum_member_description">
-<p>a multicast address</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-FLAG-UNICAST:CAPS"></a>GST_RTSP_ADDRESS_FLAG_UNICAST</p></td>
-<td class="enum_member_description">
-<p>a unicast address</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPAddressPool"></a><h3>struct GstRTSPAddressPool</h3>
-<pre class="programlisting">struct GstRTSPAddressPool {
- GObject parent;
-};
-</pre>
-<p>An address pool, all member are private</p>
-<div class="refsect3">
-<a name="id-1.2.14.7.6.5"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="struct_members_name">
-<col class="struct_members_description">
-<col width="200px" class="struct_members_annotations">
-</colgroup>
-<tbody><tr>
-<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject"><span class="type">GObject</span></a> <em class="structfield"><code><a name="GstRTSPAddressPool.parent"></a>parent</code></em>;</p></td>
-<td class="struct_member_description"><p>the parent GObject</p></td>
-<td class="struct_member_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPAddressPoolClass"></a><h3>struct GstRTSPAddressPoolClass</h3>
-<pre class="programlisting">struct GstRTSPAddressPoolClass {
- GObjectClass parent_class;
-};
-</pre>
-<p>Opaque Address pool class.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPAddressPoolResult"></a><h3>enum GstRTSPAddressPoolResult</h3>
-<p>Result codes from RTSP address pool functions.</p>
-<div class="refsect3">
-<a name="id-1.2.14.7.8.4"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="enum_members_name">
-<col class="enum_members_description">
-<col width="200px" class="enum_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-POOL-OK:CAPS"></a>GST_RTSP_ADDRESS_POOL_OK</p></td>
-<td class="enum_member_description">
-<p>no error</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-POOL-EINVAL:CAPS"></a>GST_RTSP_ADDRESS_POOL_EINVAL</p></td>
-<td class="enum_member_description">
-<p>invalid arguments were provided to a function</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-POOL-ERESERVED:CAPS"></a>GST_RTSP_ADDRESS_POOL_ERESERVED</p></td>
-<td class="enum_member_description">
-<p>the addres has already been reserved</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-POOL-ERANGE:CAPS"></a>GST_RTSP_ADDRESS_POOL_ERANGE</p></td>
-<td class="enum_member_description">
-<p>the address is not in the pool</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-ADDRESS-POOL-ELAST:CAPS"></a>GST_RTSP_ADDRESS_POOL_ELAST</p></td>
-<td class="enum_member_description">
-<p>last error</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPAddressPool.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p>
-</div>
-</div>
-<div class="footer">
-<hr>
- Generated by GTK-Doc V1.21</div>
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-<a name="gst-rtsp-server-GstRTSPContext"></a><div class="titlepage"></div>
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-<td valign="top">
-<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPContext.top_of_page"></a>GstRTSPContext</span></h2>
-<p>GstRTSPContext — A client request context</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPContext.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="returnvalue">GstRTSPContext</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-get-current" title="gst_rtsp_context_get_current ()">gst_rtsp_context_get_current</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-push-current" title="gst_rtsp_context_push_current ()">gst_rtsp_context_push_current</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-pop-current" title="gst_rtsp_context_pop_current ()">gst_rtsp_context_pop_current</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPContext.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody><tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext">GstRTSPContext</a></td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPContext.description"></a><h2>Description</h2>
-<p>Last reviewed on 2013-07-11 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPContext.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-context-get-current"></a><h3>gst_rtsp_context_get_current ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="returnvalue">GstRTSPContext</span></a> *
-gst_rtsp_context_get_current (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
-<p>Get the current <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a>. This object is retrieved from the
-current thread that is handling the request for a client.</p>
-<div class="refsect3">
-<a name="id-1.2.3.6.2.5"></a><h4>Returns</h4>
-<p> a <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a></p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-context-push-current"></a><h3>gst_rtsp_context_push_current ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_context_push_current (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *ctx</code></em>);</pre>
-<p>Pushes <em class="parameter"><code>ctx</code></em>
- onto the context stack. The current
-context can then be received using <a class="link" href="gst-rtsp-server-GstRTSPContext.html#gst-rtsp-context-get-current" title="gst_rtsp_context_get_current ()"><code class="function">gst_rtsp_context_get_current()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.3.6.3.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>ctx</p></td>
-<td class="parameter_description"><p>a #<a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-context-pop-current"></a><h3>gst_rtsp_context_pop_current ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_context_pop_current (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *ctx</code></em>);</pre>
-<p>Pops <em class="parameter"><code>ctx</code></em>
- off the context stack (verifying that <em class="parameter"><code>ctx</code></em>
-
-is on the top of the stack).</p>
-<div class="refsect3">
-<a name="id-1.2.3.6.4.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>ctx</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPContext.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstRTSPContext"></a><h3>struct GstRTSPContext</h3>
-<pre class="programlisting">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;
-};
-</pre>
-<p>Information passed around containing the context of a request.</p>
-<div class="refsect3">
-<a name="id-1.2.3.7.2.5"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="struct_members_name">
-<col class="struct_members_description">
-<col width="200px" class="struct_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="struct_member_name"><p><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a> *<em class="structfield"><code><a name="GstRTSPContext.server"></a>server</code></em>;</p></td>
-<td class="struct_member_description"><p>the server</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspconnection.html#GstRTSPConnection"><span class="type">GstRTSPConnection</span></a> *<em class="structfield"><code><a name="GstRTSPContext.conn"></a>conn</code></em>;</p></td>
-<td class="struct_member_description"><p>the connection</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *<em class="structfield"><code><a name="GstRTSPContext.client"></a>client</code></em>;</p></td>
-<td class="struct_member_description"><p>the client</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#GstRTSPMessage"><span class="type">GstRTSPMessage</span></a> *<em class="structfield"><code><a name="GstRTSPContext.request"></a>request</code></em>;</p></td>
-<td class="struct_member_description"><p>the complete request</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="type">GstRTSPUrl</span></a> *<em class="structfield"><code><a name="GstRTSPContext.uri"></a>uri</code></em>;</p></td>
-<td class="struct_member_description"><p>the complete url parsed from <em class="parameter"><code>request</code></em>
-</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPMethod"><span class="type">GstRTSPMethod</span></a> <em class="structfield"><code><a name="GstRTSPContext.method"></a>method</code></em>;</p></td>
-<td class="struct_member_description"><p>the parsed method of <em class="parameter"><code>uri</code></em>
-</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> *<em class="structfield"><code><a name="GstRTSPContext.auth"></a>auth</code></em>;</p></td>
-<td class="struct_member_description"><p>the current auth object or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a></p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *<em class="structfield"><code><a name="GstRTSPContext.token"></a>token</code></em>;</p></td>
-<td class="struct_member_description"><p>authorisation token</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a> *<em class="structfield"><code><a name="GstRTSPContext.session"></a>session</code></em>;</p></td>
-<td class="struct_member_description"><p>the session, can be <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a></p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *<em class="structfield"><code><a name="GstRTSPContext.sessmedia"></a>sessmedia</code></em>;</p></td>
-<td class="struct_member_description"><p>the session media for the url can be <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a></p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> *<em class="structfield"><code><a name="GstRTSPContext.factory"></a>factory</code></em>;</p></td>
-<td class="struct_member_description"><p>the media factory for the url, can be <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a></p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *<em class="structfield"><code><a name="GstRTSPContext.media"></a>media</code></em>;</p></td>
-<td class="struct_member_description"><p>the media for the url can be <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a></p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *<em class="structfield"><code><a name="GstRTSPContext.stream"></a>stream</code></em>;</p></td>
-<td class="struct_member_description"><p>the stream for the url can be <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a></p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspmessage.html#GstRTSPMessage"><span class="type">GstRTSPMessage</span></a> *<em class="structfield"><code><a name="GstRTSPContext.response"></a>response</code></em>;</p></td>
-<td class="struct_member_description"><p>the response</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPContext.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPServer.html" title="GstRTSPServer"><span class="type">GstRTSPServer</span></a>, <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p>
-</div>
-</div>
-<div class="footer">
-<hr>
- Generated by GTK-Doc V1.21</div>
-</body>
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- <a href="#gst-rtsp-server-GstRTSPMediaFactoryURI.description" class="shortcut">Description</a></span>
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-<div class="refentry">
-<a name="gst-rtsp-server-GstRTSPMediaFactoryURI"></a><div class="titlepage"></div>
-<div class="refnamediv"><table width="100%"><tr>
-<td valign="top">
-<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPMediaFactoryURI.top_of_page"></a>GstRTSPMediaFactoryURI</span></h2>
-<p>GstRTSPMediaFactoryURI — A factory for URI sources</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPMediaFactoryURI.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI"><span class="returnvalue">GstRTSPMediaFactoryURI</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-new" title="gst_rtsp_media_factory_uri_new ()">gst_rtsp_media_factory_uri_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-set-uri" title="gst_rtsp_media_factory_uri_set_uri ()">gst_rtsp_media_factory_uri_set_uri</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-get-uri" title="gst_rtsp_media_factory_uri_get_uri ()">gst_rtsp_media_factory_uri_get_uri</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPMediaFactoryURI.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI">GstRTSPMediaFactoryURI</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURIClass" title="struct GstRTSPMediaFactoryURIClass">GstRTSPMediaFactoryURIClass</a></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPMediaFactoryURI.description"></a><h2>Description</h2>
-<p>This specialized <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a> constructs media pipelines from a URI,
-given with <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#gst-rtsp-media-factory-uri-set-uri" title="gst_rtsp_media_factory_uri_set_uri ()"><code class="function">gst_rtsp_media_factory_uri_set_uri()</code></a>.</p>
-<p>It will automatically demux and payload the different streams found in the
-media at URL.</p>
-<p>Last reviewed on 2013-07-11 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPMediaFactoryURI.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-uri-new"></a><h3>gst_rtsp_media_factory_uri_new ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI"><span class="returnvalue">GstRTSPMediaFactoryURI</span></a> *
-gst_rtsp_media_factory_uri_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
-<p>Create a new <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI"><span class="type">GstRTSPMediaFactoryURI</span></a> instance.</p>
-<div class="refsect3">
-<a name="id-1.2.6.6.2.5"></a><h4>Returns</h4>
-<p> a new <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI"><span class="type">GstRTSPMediaFactoryURI</span></a> object. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-uri-set-uri"></a><h3>gst_rtsp_media_factory_uri_set_uri ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_media_factory_uri_set_uri (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI"><span class="type">GstRTSPMediaFactoryURI</span></a> *factory</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *uri</code></em>);</pre>
-<p>Set the URI of the resource that will be streamed by this factory.</p>
-<div class="refsect3">
-<a name="id-1.2.6.6.3.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>uri</p></td>
-<td class="parameter_description"><p>the uri the stream</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-media-factory-uri-get-uri"></a><h3>gst_rtsp_media_factory_uri_get_uri ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-gst_rtsp_media_factory_uri_get_uri (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI"><span class="type">GstRTSPMediaFactoryURI</span></a> *factory</code></em>);</pre>
-<p>Get the URI that will provide media for this factory.</p>
-<div class="refsect3">
-<a name="id-1.2.6.6.4.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>factory</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.6.6.4.6"></a><h4>Returns</h4>
-<p> the configured URI. <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPMediaFactoryURI.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstRTSPMediaFactoryURI"></a><h3>struct GstRTSPMediaFactoryURI</h3>
-<pre class="programlisting">struct GstRTSPMediaFactoryURI {
- GstRTSPMediaFactory parent;
-};
-</pre>
-<p>A media factory that creates a pipeline to play and uri.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPMediaFactoryURIClass"></a><h3>struct GstRTSPMediaFactoryURIClass</h3>
-<pre class="programlisting">struct GstRTSPMediaFactoryURIClass {
- GstRTSPMediaFactoryClass parent_class;
-};
-</pre>
-<p>The <a class="link" href="gst-rtsp-server-GstRTSPMediaFactoryURI.html#GstRTSPMediaFactoryURI" title="struct GstRTSPMediaFactoryURI"><span class="type">GstRTSPMediaFactoryURI</span></a> class structure.</p>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPMediaFactoryURI.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory"><span class="type">GstRTSPMediaFactory</span></a>, <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p>
-</div>
-</div>
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-<td valign="top">
-<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPParams.top_of_page"></a>GstRTSPParams</span></h2>
-<p>GstRTSPParams — Param get and set implementation</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPParams.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="returnvalue">GstRTSPResult</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPParams.html#gst-rtsp-params-get" title="gst_rtsp_params_get ()">gst_rtsp_params_get</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="returnvalue">GstRTSPResult</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPParams.html#gst-rtsp-params-set" title="gst_rtsp_params_set ()">gst_rtsp_params_set</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPParams.description"></a><h2>Description</h2>
-<p>Last reviewed on 2013-07-11 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPParams.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-params-get"></a><h3>gst_rtsp_params_get ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="returnvalue">GstRTSPResult</span></a>
-gst_rtsp_params_get (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *ctx</code></em>);</pre>
-<p>Get parameters (not implemented yet)</p>
-<div class="refsect3">
-<a name="id-1.2.19.5.2.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>ctx</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.19.5.2.6"></a><h4>Returns</h4>
-<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="type">GstRTSPResult</span></a></p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-params-set"></a><h3>gst_rtsp_params_set ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="returnvalue">GstRTSPResult</span></a>
-gst_rtsp_params_set (<em class="parameter"><code><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a> *client</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *ctx</code></em>);</pre>
-<p>Set parameters (not implemented yet)</p>
-<div class="refsect3">
-<a name="id-1.2.19.5.3.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>client</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>ctx</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.19.5.3.6"></a><h4>Returns</h4>
-<p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPResult"><span class="type">GstRTSPResult</span></a></p>
-<p></p>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPParams.other_details"></a><h2>Types and Values</h2>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPParams.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p>
-</div>
-</div>
-<div class="footer">
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-<div class="refentry">
-<a name="gst-rtsp-server-GstRTSPPermissions"></a><div class="titlepage"></div>
-<div class="refnamediv"><table width="100%"><tr>
-<td valign="top">
-<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPPermissions.top_of_page"></a>GstRTSPPermissions</span></h2>
-<p>GstRTSPPermissions — Roles and associated permissions</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPPermissions.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="returnvalue">GstRTSPPermissions</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-new" title="gst_rtsp_permissions_new ()">gst_rtsp_permissions_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="returnvalue">GstRTSPPermissions</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-ref" title="gst_rtsp_permissions_ref ()">gst_rtsp_permissions_ref</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-unref" title="gst_rtsp_permissions_unref ()">gst_rtsp_permissions_unref</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-add-role" title="gst_rtsp_permissions_add_role ()">gst_rtsp_permissions_add_role</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-add-role-valist" title="gst_rtsp_permissions_add_role_valist ()">gst_rtsp_permissions_add_role_valist</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-remove-role" title="gst_rtsp_permissions_remove_role ()">gst_rtsp_permissions_remove_role</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="returnvalue">GstStructure</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-get-role" title="gst_rtsp_permissions_get_role ()">gst_rtsp_permissions_get_role</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-is-allowed" title="gst_rtsp_permissions_is_allowed ()">gst_rtsp_permissions_is_allowed</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPPermissions.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody><tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions">GstRTSPPermissions</a></td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPPermissions.description"></a><h2>Description</h2>
-<p>The <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> object contains an array of roles and associated
-permissions. The roles are represented with a string and the permissions with
-a generic <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="type">GstStructure</span></a>.</p>
-<p>The permissions are deliberately kept generic. The possible values of the
-roles and <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="type">GstStructure</span></a> keys and values are only determined by the <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a>
-object that performs the checks on the permissions and the current
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a>.</p>
-<p>As a convenience function, <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#gst-rtsp-permissions-is-allowed" title="gst_rtsp_permissions_is_allowed ()"><code class="function">gst_rtsp_permissions_is_allowed()</code></a> can be used to
-check if the permissions contains a role that contains the boolean value
-<a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> for the the given key.</p>
-<p>Last reviewed on 2013-07-15 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPPermissions.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-permissions-new"></a><h3>gst_rtsp_permissions_new ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="returnvalue">GstRTSPPermissions</span></a> *
-gst_rtsp_permissions_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
-<p>Create a new empty Authorization permissions.</p>
-<div class="refsect3">
-<a name="id-1.2.18.6.2.5"></a><h4>Returns</h4>
-<p> a new empty authorization permissions. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-permissions-ref"></a><h3>gst_rtsp_permissions_ref ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="returnvalue">GstRTSPPermissions</span></a> *
-gst_rtsp_permissions_ref (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>);</pre>
-<p>Increase the refcount of this permissions.</p>
-<div class="refsect3">
-<a name="id-1.2.18.6.3.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>permissions</p></td>
-<td class="parameter_description"><p>The permissions to refcount</p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.18.6.3.6"></a><h4>Returns</h4>
-<p> <em class="parameter"><code>permissions</code></em>
-(for convenience when doing assignments). </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-permissions-unref"></a><h3>gst_rtsp_permissions_unref ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_permissions_unref (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>);</pre>
-<p>Decrease the refcount of an permissions, freeing it if the refcount reaches 0.</p>
-<div class="refsect3">
-<a name="id-1.2.18.6.4.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>permissions</p></td>
-<td class="parameter_description"><p> the permissions to refcount. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-permissions-add-role"></a><h3>gst_rtsp_permissions_add_role ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_permissions_add_role (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *role</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *fieldname</code></em>,
- <em class="parameter"><code>...</code></em>);</pre>
-<p>Add a new <em class="parameter"><code>role</code></em>
- to <em class="parameter"><code>permissions</code></em>
- with the given variables. The fields
-are the same layout as <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html#gst-structure-new"><code class="function">gst_structure_new()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.18.6.5.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>permissions</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>role</p></td>
-<td class="parameter_description"><p>a role</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>fieldname</p></td>
-<td class="parameter_description"><p>the first field name</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>...</p></td>
-<td class="parameter_description"><p>additional arguments</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-permissions-add-role-valist"></a><h3>gst_rtsp_permissions_add_role_valist ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_permissions_add_role_valist (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *role</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *fieldname</code></em>,
- <em class="parameter"><code><span class="type">va_list</span> var_args</code></em>);</pre>
-<p>Add a new <em class="parameter"><code>role</code></em>
- to <em class="parameter"><code>permissions</code></em>
- with the given variables. Structure fields
-are set according to the varargs in a manner similar to <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html#gst-structure-new"><code class="function">gst_structure_new()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.18.6.6.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>permissions</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>role</p></td>
-<td class="parameter_description"><p>a role</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>fieldname</p></td>
-<td class="parameter_description"><p>the first field name</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>var_args</p></td>
-<td class="parameter_description"><p>additional fields to add</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-permissions-remove-role"></a><h3>gst_rtsp_permissions_remove_role ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_permissions_remove_role (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *role</code></em>);</pre>
-<p>Remove all permissions for <em class="parameter"><code>role</code></em>
- in <em class="parameter"><code>permissions</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.18.6.7.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>permissions</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>role</p></td>
-<td class="parameter_description"><p>a role</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-permissions-get-role"></a><h3>gst_rtsp_permissions_get_role ()</h3>
-<pre class="programlisting">const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="returnvalue">GstStructure</span></a> *
-gst_rtsp_permissions_get_role (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *role</code></em>);</pre>
-<p>Get all permissions for <em class="parameter"><code>role</code></em>
- in <em class="parameter"><code>permissions</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.18.6.8.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>permissions</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>role</p></td>
-<td class="parameter_description"><p>a role</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.18.6.8.6"></a><h4>Returns</h4>
-<p> the structure with permissions for <em class="parameter"><code>role</code></em>
-. It
-remains valid for as long as <em class="parameter"><code>permissions</code></em>
-is valid. </p>
-<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-permissions-is-allowed"></a><h3>gst_rtsp_permissions_is_allowed ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_permissions_is_allowed (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *permissions</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *role</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *permission</code></em>);</pre>
-<p>Check if <em class="parameter"><code>role</code></em>
- in <em class="parameter"><code>permissions</code></em>
- is given permission for <em class="parameter"><code>permission</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.18.6.9.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>permissions</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>role</p></td>
-<td class="parameter_description"><p>a role</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>permission</p></td>
-<td class="parameter_description"><p>a permission</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.18.6.9.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>role</code></em>
-is allowed <em class="parameter"><code>permission</code></em>
-.</p>
-<p></p>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPPermissions.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstRTSPPermissions"></a><h3>struct GstRTSPPermissions</h3>
-<pre class="programlisting">struct GstRTSPPermissions {
- GstMiniObject mini_object;
-};
-</pre>
-<p>The opaque permissions structure. It is used to define the permissions
-of objects in different roles.</p>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPPermissions.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a>, <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a></p>
-</div>
-</div>
-<div class="footer">
-<hr>
- Generated by GTK-Doc V1.21</div>
-</body>
-</html>
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-<a href="#" class="shortcut">Top</a><span id="nav_description"> <span class="dim">|</span>
- <a href="#gst-rtsp-server-GstRTSPSdp.description" class="shortcut">Description</a></span>
-</td>
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-</tr></table>
-<div class="refentry">
-<a name="gst-rtsp-server-GstRTSPSdp"></a><div class="titlepage"></div>
-<div class="refnamediv"><table width="100%"><tr>
-<td valign="top">
-<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPSdp.top_of_page"></a>GstRTSPSdp</span></h2>
-<p>GstRTSPSdp — Make SDP messages</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPSdp.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody><tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPSdp.html#gst-rtsp-sdp-from-media" title="gst_rtsp_sdp_from_media ()">gst_rtsp_sdp_from_media</a> <span class="c_punctuation">()</span>
-</td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPSdp.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody><tr>
-<td class="datatype_keyword"> </td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPSdp.html#GstSDPInfo" title="GstSDPInfo">GstSDPInfo</a></td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPSdp.description"></a><h2>Description</h2>
-<p>Last reviewed on 2013-07-11 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPSdp.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-sdp-from-media"></a><h3>gst_rtsp_sdp_from_media ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_sdp_from_media (<em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstsdpmessage.html#GstSDPMessage"><span class="type">GstSDPMessage</span></a> *sdp</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSdp.html#GstSDPInfo" title="GstSDPInfo"><span class="type">GstSDPInfo</span></a> *info</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Add <em class="parameter"><code>media</code></em>
- specific info to <em class="parameter"><code>sdp</code></em>
-. <em class="parameter"><code>info</code></em>
- is used to configure the connection
-information in the SDP.</p>
-<div class="refsect3">
-<a name="id-1.2.13.6.2.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>sdp</p></td>
-<td class="parameter_description"><p>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstsdpmessage.html#GstSDPMessage"><span class="type">GstSDPMessage</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>info</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPSdp.html#GstSDPInfo" title="GstSDPInfo"><span class="type">GstSDPInfo</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p> a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.13.6.2.6"></a><h4>Returns</h4>
-<p> TRUE on success.</p>
-<p></p>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPSdp.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstSDPInfo"></a><h3>GstSDPInfo</h3>
-<pre class="programlisting">typedef struct {
- gboolean is_ipv6;
- const gchar *server_ip;
-} GstSDPInfo;
-</pre>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPSdp.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a></p>
-</div>
-</div>
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- <a href="#gst-rtsp-server-GstRTSPSessionMedia.description" class="shortcut">Description</a></span>
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-<div class="refentry">
-<a name="gst-rtsp-server-GstRTSPSessionMedia"></a><div class="titlepage"></div>
-<div class="refnamediv"><table width="100%"><tr>
-<td valign="top">
-<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPSessionMedia.top_of_page"></a>GstRTSPSessionMedia</span></h2>
-<p>GstRTSPSessionMedia — Media managed in a session</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPSessionMedia.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="returnvalue">GstRTSPSessionMedia</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-new" title="gst_rtsp_session_media_new ()">gst_rtsp_session_media_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-matches" title="gst_rtsp_session_media_matches ()">gst_rtsp_session_media_matches</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="returnvalue">GstRTSPMedia</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-media" title="gst_rtsp_session_media_get_media ()">gst_rtsp_session_media_get_media</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html#GstClockTime"><span class="returnvalue">GstClockTime</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-base-time" title="gst_rtsp_session_media_get_base_time ()">gst_rtsp_session_media_get_base_time</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-rtpinfo" title="gst_rtsp_session_media_get_rtpinfo ()">gst_rtsp_session_media_get_rtpinfo</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-state" title="gst_rtsp_session_media_set_state ()">gst_rtsp_session_media_set_state</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPState"><span class="returnvalue">GstRTSPState</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-rtsp-state" title="gst_rtsp_session_media_get_rtsp_state ()">gst_rtsp_session_media_get_rtsp_state</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-rtsp-state" title="gst_rtsp_session_media_set_rtsp_state ()">gst_rtsp_session_media_set_rtsp_state</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="returnvalue">GstRTSPStreamTransport</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-transport" title="gst_rtsp_session_media_get_transport ()">gst_rtsp_session_media_get_transport</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="returnvalue">GstRTSPStreamTransport</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-transport" title="gst_rtsp_session_media_set_transport ()">gst_rtsp_session_media_set_transport</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-alloc-channels" title="gst_rtsp_session_media_alloc_channels ()">gst_rtsp_session_media_alloc_channels</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPSessionMedia.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia">GstRTSPSessionMedia</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMediaClass" title="struct GstRTSPSessionMediaClass">GstRTSPSessionMediaClass</a></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPSessionMedia.description"></a><h2>Description</h2>
-<p>The <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> object manages a <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> with a given path.</p>
-<p>With <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-get-transport" title="gst_rtsp_session_media_get_transport ()"><code class="function">gst_rtsp_session_media_get_transport()</code></a> and
-<a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-transport" title="gst_rtsp_session_media_set_transport ()"><code class="function">gst_rtsp_session_media_set_transport()</code></a> the transports of a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> of
-the managed <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> can be retrieved and configured.</p>
-<p>Use <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#gst-rtsp-session-media-set-state" title="gst_rtsp_session_media_set_state ()"><code class="function">gst_rtsp_session_media_set_state()</code></a> to control the media state and
-transports.</p>
-<p>Last reviewed on 2013-07-16 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPSessionMedia.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-session-media-new"></a><h3>gst_rtsp_session_media_new ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="returnvalue">GstRTSPSessionMedia</span></a> *
-gst_rtsp_session_media_new (<em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *path</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> *media</code></em>);</pre>
-<p>Create a new <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> that manages the streams
-in <em class="parameter"><code>media</code></em>
- for <em class="parameter"><code>path</code></em>
-. <em class="parameter"><code>media</code></em>
- should be prepared.</p>
-<p>Ownership is taken of <em class="parameter"><code>media</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.11.6.2.6"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>path</p></td>
-<td class="parameter_description"><p>the path</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p> the <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.11.6.2.7"></a><h4>Returns</h4>
-<p> a new <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a>. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-media-matches"></a><h3>gst_rtsp_session_media_matches ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_session_media_matches (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *path</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> *matched</code></em>);</pre>
-<p>Check if the path of <em class="parameter"><code>media</code></em>
- matches <em class="parameter"><code>path</code></em>
-. It <em class="parameter"><code>path</code></em>
- matches, the amount of
-matched characters is returned in <em class="parameter"><code>matched</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.11.6.3.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>path</p></td>
-<td class="parameter_description"><p>a path</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>matched</p></td>
-<td class="parameter_description"><p> the amount of matched characters of <em class="parameter"><code>path</code></em>
-. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.11.6.3.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> when <em class="parameter"><code>path</code></em>
-matches the path of <em class="parameter"><code>media</code></em>
-.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-media-get-media"></a><h3>gst_rtsp_session_media_get_media ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="returnvalue">GstRTSPMedia</span></a> *
-gst_rtsp_session_media_get_media (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>);</pre>
-<p>Get the <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> that was used when constructing <em class="parameter"><code>media</code></em>
-</p>
-<div class="refsect3">
-<a name="id-1.2.11.6.4.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.11.6.4.6"></a><h4>Returns</h4>
-<p> the <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> of <em class="parameter"><code>media</code></em>
-. Remains valid as long
-as <em class="parameter"><code>media</code></em>
-is valid. </p>
-<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-media-get-base-time"></a><h3>gst_rtsp_session_media_get_base_time ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html#GstClockTime"><span class="returnvalue">GstClockTime</span></a>
-gst_rtsp_session_media_get_base_time (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>);</pre>
-<p>Get the base_time of the <a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a> in <em class="parameter"><code>media</code></em>
-</p>
-<div class="refsect3">
-<a name="id-1.2.11.6.5.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.11.6.5.6"></a><h4>Returns</h4>
-<p> the base_time of the media.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-media-get-rtpinfo"></a><h3>gst_rtsp_session_media_get_rtpinfo ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-gst_rtsp_session_media_get_rtpinfo (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>);</pre>
-<p>Retrieve the RTP-Info header string for all streams in <em class="parameter"><code>media</code></em>
-
-with configured transports.</p>
-<div class="refsect3">
-<a name="id-1.2.11.6.6.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.11.6.6.6"></a><h4>Returns</h4>
-<p> The RTP-Info as a string or
-<a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when no RTP-Info could be generated, <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>][<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-media-set-state"></a><h3>gst_rtsp_session_media_set_state ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_session_media_set_state (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#GstState"><span class="type">GstState</span></a> state</code></em>);</pre>
-<p>Tell the media object <em class="parameter"><code>media</code></em>
- to change to <em class="parameter"><code>state</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.11.6.7.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>state</p></td>
-<td class="parameter_description"><p>the new state</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.11.6.7.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-media-get-rtsp-state"></a><h3>gst_rtsp_session_media_get_rtsp_state ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPState"><span class="returnvalue">GstRTSPState</span></a>
-gst_rtsp_session_media_get_rtsp_state (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>);</pre>
-<p>Get the current RTSP state of <em class="parameter"><code>media</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.11.6.8.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.11.6.8.6"></a><h4>Returns</h4>
-<p> the current RTSP state of <em class="parameter"><code>media</code></em>
-.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-media-set-rtsp-state"></a><h3>gst_rtsp_session_media_set_rtsp_state ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_session_media_set_rtsp_state (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPState"><span class="type">GstRTSPState</span></a> state</code></em>);</pre>
-<p>Set the RTSP state of <em class="parameter"><code>media</code></em>
- to <em class="parameter"><code>state</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.11.6.9.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>state</p></td>
-<td class="parameter_description"><p>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspdefs.html#GstRTSPState"><span class="type">GstRTSPState</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-media-get-transport"></a><h3>gst_rtsp_session_media_get_transport ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="returnvalue">GstRTSPStreamTransport</span></a> *
-gst_rtsp_session_media_get_transport (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#guint"><span class="type">guint</span></a> idx</code></em>);</pre>
-<p>Get a previously created <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> for the stream at <em class="parameter"><code>idx</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.11.6.10.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>idx</p></td>
-<td class="parameter_description"><p>the stream index</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.11.6.10.6"></a><h4>Returns</h4>
-<p> a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> that is valid until the
-session of <em class="parameter"><code>media</code></em>
-is unreffed. </p>
-<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-media-set-transport"></a><h3>gst_rtsp_session_media_set_transport ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="returnvalue">GstRTSPStreamTransport</span></a> *
-gst_rtsp_session_media_set_transport (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a> *tr</code></em>);</pre>
-<p>Configure the transport for <em class="parameter"><code>stream</code></em>
- to <em class="parameter"><code>tr</code></em>
- in <em class="parameter"><code>media</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.11.6.11.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>tr</p></td>
-<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.11.6.11.6"></a><h4>Returns</h4>
-<p> the new or updated <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> for <em class="parameter"><code>stream</code></em>
-. </p>
-<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-session-media-alloc-channels"></a><h3>gst_rtsp_session_media_alloc_channels ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_session_media_alloc_channels (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> *media</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPRange"><span class="type">GstRTSPRange</span></a> *range</code></em>);</pre>
-<p>Fill <em class="parameter"><code>range</code></em>
- with the next available min and max channels for
-interleaved transport.</p>
-<div class="refsect3">
-<a name="id-1.2.11.6.12.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>media</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>range</p></td>
-<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPRange"><span class="type">GstRTSPRange</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Parameter for returning results. Default is transfer full."><span class="acronym">out</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.11.6.12.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success.</p>
-<p></p>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPSessionMedia.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstRTSPSessionMedia"></a><h3>struct GstRTSPSessionMedia</h3>
-<pre class="programlisting">struct GstRTSPSessionMedia {
- GObject parent;
-};
-</pre>
-<p>State of a client session regarding a specific media identified by path.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPSessionMediaClass"></a><h3>struct GstRTSPSessionMediaClass</h3>
-<pre class="programlisting">struct GstRTSPSessionMediaClass {
- GObjectClass parent_class;
-};
-</pre>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPSessionMedia.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a>, <a class="link" href="GstRTSPSession.html" title="GstRTSPSession"><span class="type">GstRTSPSession</span></a></p>
-</div>
-</div>
-<div class="footer">
-<hr>
- Generated by GTK-Doc V1.21</div>
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- <a href="#gst-rtsp-server-GstRTSPStreamTransport.description" class="shortcut">Description</a></span>
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-</tr></table>
-<div class="refentry">
-<a name="gst-rtsp-server-GstRTSPStreamTransport"></a><div class="titlepage"></div>
-<div class="refnamediv"><table width="100%"><tr>
-<td valign="top">
-<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPStreamTransport.top_of_page"></a>GstRTSPStreamTransport</span></h2>
-<p>GstRTSPStreamTransport — A media stream transport configuration</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPStreamTransport.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="returnvalue">GstRTSPStreamTransport</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-new" title="gst_rtsp_stream_transport_new ()">gst_rtsp_stream_transport_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-stream" title="gst_rtsp_stream_transport_get_stream ()">gst_rtsp_stream_transport_get_stream</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="returnvalue">GstRTSPTransport</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-transport" title="gst_rtsp_stream_transport_get_transport ()">gst_rtsp_stream_transport_get_transport</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-transport" title="gst_rtsp_stream_transport_set_transport ()">gst_rtsp_stream_transport_set_transport</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="returnvalue">GstRTSPUrl</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-url" title="gst_rtsp_stream_transport_get_url ()">gst_rtsp_stream_transport_get_url</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-url" title="gst_rtsp_stream_transport_set_url ()">gst_rtsp_stream_transport_set_url</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-get-rtpinfo" title="gst_rtsp_stream_transport_get_rtpinfo ()">gst_rtsp_stream_transport_get_rtpinfo</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<span class="c_punctuation">(</span><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPSendFunc" title="GstRTSPSendFunc ()">*GstRTSPSendFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-callbacks" title="gst_rtsp_stream_transport_set_callbacks ()">gst_rtsp_stream_transport_set_callbacks</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<span class="c_punctuation">(</span><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPKeepAliveFunc" title="GstRTSPKeepAliveFunc ()">*GstRTSPKeepAliveFunc</a><span class="c_punctuation">)</span> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-keepalive" title="gst_rtsp_stream_transport_set_keepalive ()">gst_rtsp_stream_transport_set_keepalive</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-keep-alive" title="gst_rtsp_stream_transport_keep_alive ()">gst_rtsp_stream_transport_keep_alive</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-active" title="gst_rtsp_stream_transport_set_active ()">gst_rtsp_stream_transport_set_active</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-timed-out" title="gst_rtsp_stream_transport_set_timed_out ()">gst_rtsp_stream_transport_set_timed_out</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-is-timed-out" title="gst_rtsp_stream_transport_is_timed_out ()">gst_rtsp_stream_transport_is_timed_out</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-send-rtcp" title="gst_rtsp_stream_transport_send_rtcp ()">gst_rtsp_stream_transport_send_rtcp</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-send-rtp" title="gst_rtsp_stream_transport_send_rtp ()">gst_rtsp_stream_transport_send_rtp</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPStreamTransport.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport">GstRTSPStreamTransport</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransportClass" title="struct GstRTSPStreamTransportClass">GstRTSPStreamTransportClass</a></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPStreamTransport.description"></a><h2>Description</h2>
-<p>The <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> configures the transport used by a
-<a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a>. It is usually manages by a <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a> object.</p>
-<p>With <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-callbacks" title="gst_rtsp_stream_transport_set_callbacks ()"><code class="function">gst_rtsp_stream_transport_set_callbacks()</code></a>, callbacks can be configured
-to handle the RTP and RTCP packets from the stream, for example when they
-need to be sent over TCP.</p>
-<p>With <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-active" title="gst_rtsp_stream_transport_set_active ()"><code class="function">gst_rtsp_stream_transport_set_active()</code></a> the transports are added and
-removed from the stream.</p>
-<p>A <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> will call <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-keep-alive" title="gst_rtsp_stream_transport_keep_alive ()"><code class="function">gst_rtsp_stream_transport_keep_alive()</code></a> when RTCP
-is received from the client. It will also call
-<a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-timed-out" title="gst_rtsp_stream_transport_set_timed_out ()"><code class="function">gst_rtsp_stream_transport_set_timed_out()</code></a> when a receiver has timed out.</p>
-<p>Last reviewed on 2013-07-16 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPStreamTransport.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-stream-transport-new"></a><h3>gst_rtsp_stream_transport_new ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="returnvalue">GstRTSPStreamTransport</span></a> *
-gst_rtsp_stream_transport_new (<em class="parameter"><code><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> *stream</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a> *tr</code></em>);</pre>
-<p>Create a new <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> that can be used to manage
-<em class="parameter"><code>stream</code></em>
- with transport <em class="parameter"><code>tr</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.2.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>stream</p></td>
-<td class="parameter_description"><p>a <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>tr</p></td>
-<td class="parameter_description"><p> a GstRTSPTransport. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.12.6.2.6"></a><h4>Returns</h4>
-<p> a new <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a>. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-transport-get-stream"></a><h3>gst_rtsp_stream_transport_get_stream ()</h3>
-<pre class="programlisting"><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="returnvalue">GstRTSPStream</span></a> *
-gst_rtsp_stream_transport_get_stream (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>);</pre>
-<p>Get the <a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a> used when constructing <em class="parameter"><code>trans</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.3.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.12.6.3.6"></a><h4>Returns</h4>
-<p> the stream used when constructing <em class="parameter"><code>trans</code></em>
-. </p>
-<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-transport-get-transport"></a><h3>gst_rtsp_stream_transport_get_transport ()</h3>
-<pre class="programlisting">const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="returnvalue">GstRTSPTransport</span></a> *
-gst_rtsp_stream_transport_get_transport
- (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>);</pre>
-<p>Get the transport configured in <em class="parameter"><code>trans</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.4.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.12.6.4.6"></a><h4>Returns</h4>
-<p> the transport configured in <em class="parameter"><code>trans</code></em>
-. It remains
-valid for as long as <em class="parameter"><code>trans</code></em>
-is valid. </p>
-<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-transport-set-transport"></a><h3>gst_rtsp_stream_transport_set_transport ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_stream_transport_set_transport
- (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a> *tr</code></em>);</pre>
-<p>Set <em class="parameter"><code>tr</code></em>
- as the client transport. This function takes ownership of the
-passed <em class="parameter"><code>tr</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.5.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>tr</p></td>
-<td class="parameter_description"><p> a client <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtsptransport.html#GstRTSPTransport"><span class="type">GstRTSPTransport</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-transport-get-url"></a><h3>gst_rtsp_stream_transport_get_url ()</h3>
-<pre class="programlisting">const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="returnvalue">GstRTSPUrl</span></a> *
-gst_rtsp_stream_transport_get_url (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>);</pre>
-<p>Get the url configured in <em class="parameter"><code>trans</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.6.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.12.6.6.6"></a><h4>Returns</h4>
-<p> the url configured in <em class="parameter"><code>trans</code></em>
-. It remains
-valid for as long as <em class="parameter"><code>trans</code></em>
-is valid. </p>
-<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-transport-set-url"></a><h3>gst_rtsp_stream_transport_set_url ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_stream_transport_set_url (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
- <em class="parameter"><code>const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="type">GstRTSPUrl</span></a> *url</code></em>);</pre>
-<p>Set <em class="parameter"><code>url</code></em>
- as the client url.</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.7.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>url</p></td>
-<td class="parameter_description"><p> a client <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstrtspurl.html#GstRTSPUrl"><span class="type">GstRTSPUrl</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-transport-get-rtpinfo"></a><h3>gst_rtsp_stream_transport_get_rtpinfo ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-gst_rtsp_stream_transport_get_rtpinfo (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstClock.html#GstClockTime"><span class="type">GstClockTime</span></a> start_time</code></em>);</pre>
-<p>Get the RTP-Info string for <em class="parameter"><code>trans</code></em>
- and <em class="parameter"><code>start_time</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.8.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>start_time</p></td>
-<td class="parameter_description"><p>a star time</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.12.6.8.6"></a><h4>Returns</h4>
-<p> the RTPInfo string for <em class="parameter"><code>trans</code></em>
-and <em class="parameter"><code>start_time</code></em>
-or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when the RTP-Info could not be
-determined. <a href="https://developer.gnome.org/glib/unstable/glib-Memory-Allocation.html#g-free"><code class="function">g_free()</code></a> after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>][<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPSendFunc"></a><h3>GstRTSPSendFunc ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-<span class="c_punctuation">(</span>*GstRTSPSendFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a> *buffer</code></em>,
- <em class="parameter"><code><span class="type">guint8</span> channel</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
-<p>Function registered with <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-callbacks" title="gst_rtsp_stream_transport_set_callbacks ()"><code class="function">gst_rtsp_stream_transport_set_callbacks()</code></a> and
-called when <em class="parameter"><code>buffer</code></em>
- must be sent on <em class="parameter"><code>channel</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.9.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>buffer</p></td>
-<td class="parameter_description"><p>a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>channel</p></td>
-<td class="parameter_description"><p>a channel</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p>user data</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.12.6.9.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-transport-set-callbacks"></a><h3>gst_rtsp_stream_transport_set_callbacks ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_stream_transport_set_callbacks
- (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPSendFunc" title="GstRTSPSendFunc ()"><span class="type">GstRTSPSendFunc</span></a> send_rtp</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPSendFunc" title="GstRTSPSendFunc ()"><span class="type">GstRTSPSendFunc</span></a> send_rtcp</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Datasets.html#GDestroyNotify"><span class="type">GDestroyNotify</span></a> notify</code></em>);</pre>
-<p>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.</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.10.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>send_rtp</p></td>
-<td class="parameter_description"><p> a callback called when RTP should be sent. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid until the GDestroyNotify argument is called."><span class="acronym">scope notified</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>send_rtcp</p></td>
-<td class="parameter_description"><p> a callback called when RTCP should be sent. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid until the GDestroyNotify argument is called."><span class="acronym">scope notified</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p> user data passed to callbacks. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="This parameter is a 'user_data', for callbacks; many bindings can pass NULL here."><span class="acronym">closure</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>notify</p></td>
-<td class="parameter_description"><p> called with the user_data when no longer needed. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPKeepAliveFunc"></a><h3>GstRTSPKeepAliveFunc ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-<span class="c_punctuation">(</span>*GstRTSPKeepAliveFunc<span class="c_punctuation">)</span> (<em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>);</pre>
-<p>Function registered with <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#gst-rtsp-stream-transport-set-keepalive" title="gst_rtsp_stream_transport_set_keepalive ()"><code class="function">gst_rtsp_stream_transport_set_keepalive()</code></a> and called
-when the stream is active.</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.11.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p>user data</p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-transport-set-keepalive"></a><h3>gst_rtsp_stream_transport_set_keepalive ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_stream_transport_set_keepalive
- (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPKeepAliveFunc" title="GstRTSPKeepAliveFunc ()"><span class="type">GstRTSPKeepAliveFunc</span></a> keep_alive</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gpointer"><span class="type">gpointer</span></a> user_data</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Datasets.html#GDestroyNotify"><span class="type">GDestroyNotify</span></a> notify</code></em>);</pre>
-<p>Install callbacks that will be called when RTCP packets are received from the
-receiver of <em class="parameter"><code>trans</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.12.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>keep_alive</p></td>
-<td class="parameter_description"><p> a callback called when the receiver is active. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="The callback is valid until the GDestroyNotify argument is called."><span class="acronym">scope notified</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>user_data</p></td>
-<td class="parameter_description"><p> user data passed to callback. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="This parameter is a 'user_data', for callbacks; many bindings can pass NULL here."><span class="acronym">closure</span></acronym>]</span></td>
-</tr>
-<tr>
-<td class="parameter_name"><p>notify</p></td>
-<td class="parameter_description"><p> called with the user_data when no longer needed. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="NULL is OK, both for passing and for returning."><span class="acronym">allow-none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-transport-keep-alive"></a><h3>gst_rtsp_stream_transport_keep_alive ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_stream_transport_keep_alive (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>);</pre>
-<p>Signal the installed keep_alive callback for <em class="parameter"><code>trans</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.13.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-transport-set-active"></a><h3>gst_rtsp_stream_transport_set_active ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_stream_transport_set_active (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> active</code></em>);</pre>
-<p>Activate or deactivate datatransfer configured in <em class="parameter"><code>trans</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.14.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>active</p></td>
-<td class="parameter_description"><p>new state of <em class="parameter"><code>trans</code></em>
-</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.12.6.14.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> when the state was changed.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-transport-set-timed-out"></a><h3>gst_rtsp_stream_transport_set_timed_out ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_stream_transport_set_timed_out
- (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="type">gboolean</span></a> timedout</code></em>);</pre>
-<p>Set the timed out state of <em class="parameter"><code>trans</code></em>
- to <em class="parameter"><code>timedout</code></em>
-</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.15.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>timedout</p></td>
-<td class="parameter_description"><p>timed out value</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-transport-is-timed-out"></a><h3>gst_rtsp_stream_transport_is_timed_out ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_stream_transport_is_timed_out
- (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>);</pre>
-<p>Check if <em class="parameter"><code>trans</code></em>
- is timed out.</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.16.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.12.6.16.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>trans</code></em>
-timed out.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-transport-send-rtcp"></a><h3>gst_rtsp_stream_transport_send_rtcp ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_stream_transport_send_rtcp (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a> *buffer</code></em>);</pre>
-<p>Send <em class="parameter"><code>buffer</code></em>
- to the installed RTCP callback for <em class="parameter"><code>trans</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.17.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>buffer</p></td>
-<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.12.6.17.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-stream-transport-send-rtp"></a><h3>gst_rtsp_stream_transport_send_rtp ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_stream_transport_send_rtp (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a> *trans</code></em>,
- <em class="parameter"><code><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a> *buffer</code></em>);</pre>
-<p>Send <em class="parameter"><code>buffer</code></em>
- to the installed RTP callback for <em class="parameter"><code>trans</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.12.6.18.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>trans</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPStreamTransport.html#GstRTSPStreamTransport" title="struct GstRTSPStreamTransport"><span class="type">GstRTSPStreamTransport</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>buffer</p></td>
-<td class="parameter_description"><p> a <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstBuffer.html"><span class="type">GstBuffer</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.12.6.18.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> on success</p>
-<p></p>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPStreamTransport.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstRTSPStreamTransport"></a><h3>struct GstRTSPStreamTransport</h3>
-<pre class="programlisting">struct GstRTSPStreamTransport {
- GObject parent;
-};
-</pre>
-<p>A Transport description for a stream</p>
-<div class="refsect3">
-<a name="id-1.2.12.7.2.5"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="struct_members_name">
-<col class="struct_members_description">
-<col width="200px" class="struct_members_annotations">
-</colgroup>
-<tbody><tr>
-<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObject"><span class="type">GObject</span></a> <em class="structfield"><code><a name="GstRTSPStreamTransport.parent"></a>parent</code></em>;</p></td>
-<td class="struct_member_description"><p>parent instance</p></td>
-<td class="struct_member_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPStreamTransportClass"></a><h3>struct GstRTSPStreamTransportClass</h3>
-<pre class="programlisting">struct GstRTSPStreamTransportClass {
- GObjectClass parent_class;
-};
-</pre>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPStreamTransport.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPStream.html" title="GstRTSPStream"><span class="type">GstRTSPStream</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPSessionMedia.html#GstRTSPSessionMedia" title="struct GstRTSPSessionMedia"><span class="type">GstRTSPSessionMedia</span></a></p>
-</div>
-</div>
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-<div class="refentry">
-<a name="gst-rtsp-server-GstRTSPThreadPool"></a><div class="titlepage"></div>
-<div class="refnamediv"><table width="100%"><tr>
-<td valign="top">
-<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPThreadPool.top_of_page"></a>GstRTSPThreadPool</span></h2>
-<p>GstRTSPThreadPool — A pool of threads</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPThreadPool.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="returnvalue">GstRTSPThread</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-new" title="gst_rtsp_thread_new ()">gst_rtsp_thread_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="returnvalue">GstRTSPThread</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-ref" title="gst_rtsp_thread_ref ()">gst_rtsp_thread_ref</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-unref" title="gst_rtsp_thread_unref ()">gst_rtsp_thread_unref</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-reuse" title="gst_rtsp_thread_reuse ()">gst_rtsp_thread_reuse</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-stop" title="gst_rtsp_thread_stop ()">gst_rtsp_thread_stop</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="returnvalue">GstRTSPThreadPool</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-new" title="gst_rtsp_thread_pool_new ()">gst_rtsp_thread_pool_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="returnvalue">gint</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-get-max-threads" title="gst_rtsp_thread_pool_get_max_threads ()">gst_rtsp_thread_pool_get_max_threads</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-set-max-threads" title="gst_rtsp_thread_pool_set_max_threads ()">gst_rtsp_thread_pool_set_max_threads</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="returnvalue">GstRTSPThread</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-get-thread" title="gst_rtsp_thread_pool_get_thread ()">gst_rtsp_thread_pool_get_thread</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-cleanup" title="gst_rtsp_thread_pool_cleanup ()">gst_rtsp_thread_pool_cleanup</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPThreadPool.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody>
-<tr>
-<td class="datatype_keyword">enum</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadType" title="enum GstRTSPThreadType">GstRTSPThreadType</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread">GstRTSPThread</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool">GstRTSPThreadPool</a></td>
-</tr>
-<tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPoolClass" title="struct GstRTSPThreadPoolClass">GstRTSPThreadPoolClass</a></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPThreadPool.description"></a><h2>Description</h2>
-<p>A <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> manages reusable threads for various server tasks.
-Currently the defined thread types can be found in <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadType" title="enum GstRTSPThreadType"><span class="type">GstRTSPThreadType</span></a>.</p>
-<p>Threads of type <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GST-RTSP-THREAD-TYPE-CLIENT:CAPS"><span class="type">GST_RTSP_THREAD_TYPE_CLIENT</span></a> are used to handle requests from
-a connected client. With <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-get-max-threads" title="gst_rtsp_thread_pool_get_max_threads ()"><code class="function">gst_rtsp_thread_pool_get_max_threads()</code></a> a maximum
-number of threads can be set after which the pool will start to reuse the
-same thread for multiple clients.</p>
-<p>Threads of type <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GST-RTSP-THREAD-TYPE-MEDIA:CAPS"><span class="type">GST_RTSP_THREAD_TYPE_MEDIA</span></a> will be used to perform the state
-changes of the media pipelines and handle its bus messages.</p>
-<p>gst_rtsp_thread_pool_get_thread() can be used to create a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a>
-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.</p>
-<p>gst_rtsp_thread_reuse() can be used to reuse a thread for multiple purposes.
-If all <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-reuse" title="gst_rtsp_thread_reuse ()"><code class="function">gst_rtsp_thread_reuse()</code></a> calls are matched with a
-<a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-stop" title="gst_rtsp_thread_stop ()"><code class="function">gst_rtsp_thread_stop()</code></a> call, the mainloop will be quit and the thread will
-stop.</p>
-<p>To configure the threads, a subclass of this object should be made and the
-virtual methods should be overriden to implement the desired functionality.</p>
-<p>Last reviewed on 2013-07-11 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPThreadPool.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-thread-new"></a><h3>gst_rtsp_thread_new ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="returnvalue">GstRTSPThread</span></a> *
-gst_rtsp_thread_new (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadType" title="enum GstRTSPThreadType"><span class="type">GstRTSPThreadType</span></a> type</code></em>);</pre>
-<p>Create a new thread object that can run a mainloop.</p>
-<div class="refsect3">
-<a name="id-1.2.15.6.2.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>type</p></td>
-<td class="parameter_description"><p>the thread type</p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.15.6.2.6"></a><h4>Returns</h4>
-<p> a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a>. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-thread-ref"></a><h3>gst_rtsp_thread_ref ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="returnvalue">GstRTSPThread</span></a> *
-gst_rtsp_thread_ref (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a> *thread</code></em>);</pre>
-<p>Increase the refcount of this thread.</p>
-<div class="refsect3">
-<a name="id-1.2.15.6.3.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>thread</p></td>
-<td class="parameter_description"><p>The thread to refcount</p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.15.6.3.6"></a><h4>Returns</h4>
-<p> <em class="parameter"><code>thread</code></em>
-(for convenience when doing assignments). </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-thread-unref"></a><h3>gst_rtsp_thread_unref ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_thread_unref (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> *thread</code></em>);</pre>
-<p>Decrease the refcount of an thread, freeing it if the refcount reaches 0.</p>
-<div class="refsect3">
-<a name="id-1.2.15.6.4.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>thread</p></td>
-<td class="parameter_description"><p> the thread to refcount. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-thread-reuse"></a><h3>gst_rtsp_thread_reuse ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_thread_reuse (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a> *thread</code></em>);</pre>
-<p>Reuse the mainloop of <em class="parameter"><code>thread</code></em>
-</p>
-<div class="refsect3">
-<a name="id-1.2.15.6.5.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>thread</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.15.6.5.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if the mainloop could be reused</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-thread-stop"></a><h3>gst_rtsp_thread_stop ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_thread_stop (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a> *thread</code></em>);</pre>
-<p>Stop and unref <em class="parameter"><code>thread</code></em>
-. When no threads are using the mainloop, the thread
-will be stopped and the final ref to <em class="parameter"><code>thread</code></em>
- will be released.</p>
-<div class="refsect3">
-<a name="id-1.2.15.6.6.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>thread</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-thread-pool-new"></a><h3>gst_rtsp_thread_pool_new ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="returnvalue">GstRTSPThreadPool</span></a> *
-gst_rtsp_thread_pool_new (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
-<p>Create a new <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> instance.</p>
-<div class="refsect3">
-<a name="id-1.2.15.6.7.5"></a><h4>Returns</h4>
-<p> a new <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a>. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-thread-pool-get-max-threads"></a><h3>gst_rtsp_thread_pool_get_max_threads ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="returnvalue">gint</span></a>
-gst_rtsp_thread_pool_get_max_threads (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> *pool</code></em>);</pre>
-<p>Get the maximum number of threads used for client connections.
-See <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-pool-set-max-threads" title="gst_rtsp_thread_pool_set_max_threads ()"><code class="function">gst_rtsp_thread_pool_set_max_threads()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.15.6.8.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.15.6.8.6"></a><h4>Returns</h4>
-<p> the maximum number of threads.</p>
-<p></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-thread-pool-set-max-threads"></a><h3>gst_rtsp_thread_pool_set_max_threads ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_thread_pool_set_max_threads (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> *pool</code></em>,
- <em class="parameter"><code><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gint"><span class="type">gint</span></a> max_threads</code></em>);</pre>
-<p>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.</p>
-<div class="refsect3">
-<a name="id-1.2.15.6.9.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>max_threads</p></td>
-<td class="parameter_description"><p>maximum threads</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-thread-pool-get-thread"></a><h3>gst_rtsp_thread_pool_get_thread ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="returnvalue">GstRTSPThread</span></a> *
-gst_rtsp_thread_pool_get_thread (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a> *pool</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadType" title="enum GstRTSPThreadType"><span class="type">GstRTSPThreadType</span></a> type</code></em>,
- <em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a> *ctx</code></em>);</pre>
-<p>Get a new <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a> for <em class="parameter"><code>type</code></em>
- and <em class="parameter"><code>ctx</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.15.6.10.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>pool</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadPool" title="struct GstRTSPThreadPool"><span class="type">GstRTSPThreadPool</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>type</p></td>
-<td class="parameter_description"><p>the <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadType" title="enum GstRTSPThreadType"><span class="type">GstRTSPThreadType</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>ctx</p></td>
-<td class="parameter_description"><p> a <a class="link" href="gst-rtsp-server-GstRTSPContext.html#GstRTSPContext" title="struct GstRTSPContext"><span class="type">GstRTSPContext</span></a>. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.15.6.10.6"></a><h4>Returns</h4>
-<p> a new <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThread" title="struct GstRTSPThread"><span class="type">GstRTSPThread</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#gst-rtsp-thread-stop" title="gst_rtsp_thread_stop ()"><code class="function">gst_rtsp_thread_stop()</code></a> after usage. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-thread-pool-cleanup"></a><h3>gst_rtsp_thread_pool_cleanup ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_thread_pool_cleanup (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
-<p>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.</p>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPThreadPool.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstRTSPThreadType"></a><h3>enum GstRTSPThreadType</h3>
-<p>Different thread types</p>
-<div class="refsect3">
-<a name="id-1.2.15.7.2.4"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="enum_members_name">
-<col class="enum_members_description">
-<col width="200px" class="enum_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-THREAD-TYPE-CLIENT:CAPS"></a>GST_RTSP_THREAD_TYPE_CLIENT</p></td>
-<td class="enum_member_description">
-<p>a thread to handle the client communication</p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-<tr>
-<td class="enum_member_name"><p><a name="GST-RTSP-THREAD-TYPE-MEDIA:CAPS"></a>GST_RTSP_THREAD_TYPE_MEDIA</p></td>
-<td class="enum_member_description">
-<p>a thread to handle media </p>
-</td>
-<td class="enum_member_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPThread"></a><h3>struct GstRTSPThread</h3>
-<pre class="programlisting">struct GstRTSPThread {
- GstMiniObject mini_object;
-
- GstRTSPThreadType type;
- GMainContext *context;
- GMainLoop *loop;
-};
-</pre>
-<p>Structure holding info about a mainloop running in a thread</p>
-<div class="refsect3">
-<a name="id-1.2.15.7.3.5"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="struct_members_name">
-<col class="struct_members_description">
-<col width="200px" class="struct_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="struct_member_name"><p><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstMiniObject.html#GstMiniObject"><span class="type">GstMiniObject</span></a> <em class="structfield"><code><a name="GstRTSPThread.mini-object"></a>mini_object</code></em>;</p></td>
-<td class="struct_member_description"><p>parent <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstMiniObject.html#GstMiniObject"><span class="type">GstMiniObject</span></a></p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a class="link" href="gst-rtsp-server-GstRTSPThreadPool.html#GstRTSPThreadType" title="enum GstRTSPThreadType"><span class="type">GstRTSPThreadType</span></a> <em class="structfield"><code><a name="GstRTSPThread.type"></a>type</code></em>;</p></td>
-<td class="struct_member_description"><p>the thread type</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainContext"><span class="type">GMainContext</span></a> *<em class="structfield"><code><a name="GstRTSPThread.context"></a>context</code></em>;</p></td>
-<td class="struct_member_description"><p>a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainContext"><span class="type">GMainContext</span></a></p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainLoop"><span class="type">GMainLoop</span></a> *<em class="structfield"><code><a name="GstRTSPThread.loop"></a>loop</code></em>;</p></td>
-<td class="struct_member_description"><p>a <a href="https://developer.gnome.org/glib/unstable/glib-The-Main-Event-Loop.html#GMainLoop"><span class="type">GMainLoop</span></a></p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPThreadPool"></a><h3>struct GstRTSPThreadPool</h3>
-<pre class="programlisting">struct GstRTSPThreadPool {
- GObject parent;
-};
-</pre>
-<p>The thread pool structure.</p>
-</div>
-<hr>
-<div class="refsect2">
-<a name="GstRTSPThreadPoolClass"></a><h3>struct GstRTSPThreadPoolClass</h3>
-<pre class="programlisting">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);
-};
-</pre>
-<p>Class for managing threads.</p>
-<div class="refsect3">
-<a name="id-1.2.15.7.5.5"></a><h4>Members</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="300px" class="struct_members_name">
-<col class="struct_members_description">
-<col width="200px" class="struct_members_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="struct_member_name"><p><a href="https://developer.gnome.org/gobject/unstable/gobject-The-Base-Object-Type.html#GObjectClass"><span class="type">GObjectClass</span></a> <em class="structfield"><code><a name="GstRTSPThreadPoolClass.parent-class"></a>parent_class</code></em>;</p></td>
-<td> </td>
-<td> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><a href="https://developer.gnome.org/glib/unstable/glib-Thread-Pools.html#GThreadPool"><span class="type">GThreadPool</span></a> *<em class="structfield"><code><a name="GstRTSPThreadPoolClass.pool"></a>pool</code></em>;</p></td>
-<td class="struct_member_description"><p>a <a href="https://developer.gnome.org/glib/unstable/glib-Thread-Pools.html#GThreadPool"><span class="type">GThreadPool</span></a> used internally</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPThreadPoolClass.get-thread"></a>get_thread</code></em> ()</p></td>
-<td class="struct_member_description"><p>this function should make or reuse an existing thread that runs
-a mainloop.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPThreadPoolClass.configure-thread"></a>configure_thread</code></em> ()</p></td>
-<td class="struct_member_description"><p>configure a thread object. this vmethod is called when
-a new thread has been created and should be configured.</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPThreadPoolClass.thread-enter"></a>thread_enter</code></em> ()</p></td>
-<td class="struct_member_description"><p>called from the thread when it is entered</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-<tr>
-<td class="struct_member_name"><p><em class="structfield"><code><a name="GstRTSPThreadPoolClass.thread-leave"></a>thread_leave</code></em> ()</p></td>
-<td class="struct_member_description"><p>called from the thread when it is left</p></td>
-<td class="struct_member_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPThreadPool.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPMedia.html" title="GstRTSPMedia"><span class="type">GstRTSPMedia</span></a>, <a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a></p>
-</div>
-</div>
-<div class="footer">
-<hr>
- Generated by GTK-Doc V1.21</div>
-</body>
-</html>
\ No newline at end of file
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-<html>
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-<title>GStreamer RTSP Server Reference Manual: GstRTSPToken</title>
-<meta name="generator" content="DocBook XSL Stylesheets V1.78.1">
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- <a href="#gst-rtsp-server-GstRTSPToken.description" class="shortcut">Description</a></span>
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-<a name="gst-rtsp-server-GstRTSPToken"></a><div class="titlepage"></div>
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-<td valign="top">
-<h2><span class="refentrytitle"><a name="gst-rtsp-server-GstRTSPToken.top_of_page"></a>GstRTSPToken</span></h2>
-<p>GstRTSPToken — Roles and permissions for a client</p>
-</td>
-<td class="gallery_image" valign="top" align="right"></td>
-</tr></table></div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPToken.functions"></a><h2>Functions</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="functions_return">
-<col class="functions_name">
-</colgroup>
-<tbody>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new-empty" title="gst_rtsp_token_new_empty ()">gst_rtsp_token_new_empty</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new" title="gst_rtsp_token_new ()">gst_rtsp_token_new</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-new-valist" title="gst_rtsp_token_new_valist ()">gst_rtsp_token_new_valist</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-ref" title="gst_rtsp_token_ref ()">gst_rtsp_token_ref</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<span class="returnvalue">void</span>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-unref" title="gst_rtsp_token_unref ()">gst_rtsp_token_unref</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="returnvalue">GstStructure</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-get-structure" title="gst_rtsp_token_get_structure ()">gst_rtsp_token_get_structure</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="returnvalue">GstStructure</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-writable-structure" title="gst_rtsp_token_writable_structure ()">gst_rtsp_token_writable_structure</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-get-string" title="gst_rtsp_token_get_string ()">gst_rtsp_token_get_string</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-<tr>
-<td class="function_type">
-<a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-</td>
-<td class="function_name">
-<a class="link" href="gst-rtsp-server-GstRTSPToken.html#gst-rtsp-token-is-allowed" title="gst_rtsp_token_is_allowed ()">gst_rtsp_token_is_allowed</a> <span class="c_punctuation">()</span>
-</td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPToken.other"></a><h2>Types and Values</h2>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="name">
-<col class="description">
-</colgroup>
-<tbody><tr>
-<td class="datatype_keyword">struct</td>
-<td class="function_name"><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken">GstRTSPToken</a></td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPToken.description"></a><h2>Description</h2>
-<p>A <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> contains the permissions and roles of the user
-performing the current request. A token is usually created when a user is
-authenticated by the <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> object and is then placed as the current
-token for the current request.</p>
-<p><a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a> can use the token and its contents to check authorization for
-various operations by comparing the token to the <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a> of the
-object.</p>
-<p>The accepted values of the token are entirely defined by the <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a>
-object that implements the security policy.</p>
-<p>Last reviewed on 2013-07-15 (1.0.0)</p>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPToken.functions_details"></a><h2>Functions</h2>
-<div class="refsect2">
-<a name="gst-rtsp-token-new-empty"></a><h3>gst_rtsp_token_new_empty ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
-gst_rtsp_token_new_empty (<em class="parameter"><code><span class="type">void</span></code></em>);</pre>
-<p>Create a new empty Authorization token.</p>
-<div class="refsect3">
-<a name="id-1.2.17.6.2.5"></a><h4>Returns</h4>
-<p> a new empty authorization token. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-token-new"></a><h3>gst_rtsp_token_new ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
-gst_rtsp_token_new (<em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *firstfield</code></em>,
- <em class="parameter"><code>...</code></em>);</pre>
-<p>Create a new Authorization token with the given fieldnames and values.
-Arguments are given similar to <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html#gst-structure-new"><code class="function">gst_structure_new()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.17.6.3.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>firstfield</p></td>
-<td class="parameter_description"><p>the first fieldname</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>...</p></td>
-<td class="parameter_description"><p>additional arguments</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.17.6.3.6"></a><h4>Returns</h4>
-<p> a new authorization token. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-token-new-valist"></a><h3>gst_rtsp_token_new_valist ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
-gst_rtsp_token_new_valist (<em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *firstfield</code></em>,
- <em class="parameter"><code><span class="type">va_list</span> var_args</code></em>);</pre>
-<p>Create a new Authorization token with the given fieldnames and values.
-Arguments are given similar to <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html#gst-structure-new-valist"><code class="function">gst_structure_new_valist()</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.17.6.4.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>firstfield</p></td>
-<td class="parameter_description"><p>the first fieldname</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>var_args</p></td>
-<td class="parameter_description"><p>additional arguments</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.17.6.4.6"></a><h4>Returns</h4>
-<p> a new authorization token. </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-token-ref"></a><h3>gst_rtsp_token_ref ()</h3>
-<pre class="programlisting"><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="returnvalue">GstRTSPToken</span></a> *
-gst_rtsp_token_ref (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *token</code></em>);</pre>
-<p>Increase the refcount of this token.</p>
-<div class="refsect3">
-<a name="id-1.2.17.6.5.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>token</p></td>
-<td class="parameter_description"><p>The token to refcount</p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.17.6.5.6"></a><h4>Returns</h4>
-<p> <em class="parameter"><code>token</code></em>
-(for convenience when doing assignments). </p>
-<p><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-token-unref"></a><h3>gst_rtsp_token_unref ()</h3>
-<pre class="programlisting"><span class="returnvalue">void</span>
-gst_rtsp_token_unref (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *token</code></em>);</pre>
-<p>Decrease the refcount of an token, freeing it if the refcount reaches 0.</p>
-<div class="refsect3">
-<a name="id-1.2.17.6.6.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>token</p></td>
-<td class="parameter_description"><p> the token to refcount. </p></td>
-<td class="parameter_annotations"><span class="annotation">[<acronym title="Free data after the code is done."><span class="acronym">transfer full</span></acronym>]</span></td>
-</tr></tbody>
-</table></div>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-token-get-structure"></a><h3>gst_rtsp_token_get_structure ()</h3>
-<pre class="programlisting">const <a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="returnvalue">GstStructure</span></a> *
-gst_rtsp_token_get_structure (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *token</code></em>);</pre>
-<p>Access the structure of the token.</p>
-<div class="refsect3">
-<a name="id-1.2.17.6.7.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>token</p></td>
-<td class="parameter_description"><p>The <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a>.</p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.17.6.7.6"></a><h4>Returns</h4>
-<p> 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.</p>
-<p>MT safe. </p>
-<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-token-writable-structure"></a><h3>gst_rtsp_token_writable_structure ()</h3>
-<pre class="programlisting"><a href="http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstStructure.html"><span class="returnvalue">GstStructure</span></a> *
-gst_rtsp_token_writable_structure (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *token</code></em>);</pre>
-<p>Get a writable version of the structure.</p>
-<div class="refsect3">
-<a name="id-1.2.17.6.8.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody><tr>
-<td class="parameter_name"><p>token</p></td>
-<td class="parameter_description"><p>The <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a>.</p></td>
-<td class="parameter_annotations"> </td>
-</tr></tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.17.6.8.6"></a><h4>Returns</h4>
-<p> 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
-<em class="parameter"><code>token</code></em>
-is writable and will never return <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a>.</p>
-<p>MT safe. </p>
-<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-token-get-string"></a><h3>gst_rtsp_token_get_string ()</h3>
-<pre class="programlisting">const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="returnvalue">gchar</span></a> *
-gst_rtsp_token_get_string (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *token</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *field</code></em>);</pre>
-<p>Get the string value of <em class="parameter"><code>field</code></em>
- in <em class="parameter"><code>token</code></em>
-.</p>
-<div class="refsect3">
-<a name="id-1.2.17.6.9.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>token</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>field</p></td>
-<td class="parameter_description"><p>a field name</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.17.6.9.6"></a><h4>Returns</h4>
-<p> the string value of <em class="parameter"><code>field</code></em>
-in
-<em class="parameter"><code>token</code></em>
-or <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#NULL:CAPS"><code class="literal">NULL</code></a> when <em class="parameter"><code>field</code></em>
-is not defined in <em class="parameter"><code>token</code></em>
-. The string
-becomes invalid when you free <em class="parameter"><code>token</code></em>
-. </p>
-<p><span class="annotation">[<acronym title="Don't free data after the code is done."><span class="acronym">transfer none</span></acronym>][<acronym title="NULL may be passed as the value in, out, in-out; or as a return value."><span class="acronym">nullable</span></acronym>]</span></p>
-</div>
-</div>
-<hr>
-<div class="refsect2">
-<a name="gst-rtsp-token-is-allowed"></a><h3>gst_rtsp_token_is_allowed ()</h3>
-<pre class="programlisting"><a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gboolean"><span class="returnvalue">gboolean</span></a>
-gst_rtsp_token_is_allowed (<em class="parameter"><code><a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a> *token</code></em>,
- <em class="parameter"><code>const <a href="https://developer.gnome.org/glib/unstable/glib-Basic-Types.html#gchar"><span class="type">gchar</span></a> *field</code></em>);</pre>
-<p>Check if <em class="parameter"><code>token</code></em>
- has a boolean <em class="parameter"><code>field</code></em>
- and if it is set to <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a>.</p>
-<div class="refsect3">
-<a name="id-1.2.17.6.10.5"></a><h4>Parameters</h4>
-<div class="informaltable"><table width="100%" border="0">
-<colgroup>
-<col width="150px" class="parameters_name">
-<col class="parameters_description">
-<col width="200px" class="parameters_annotations">
-</colgroup>
-<tbody>
-<tr>
-<td class="parameter_name"><p>token</p></td>
-<td class="parameter_description"><p>a <a class="link" href="gst-rtsp-server-GstRTSPToken.html#GstRTSPToken" title="struct GstRTSPToken"><span class="type">GstRTSPToken</span></a></p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-<tr>
-<td class="parameter_name"><p>field</p></td>
-<td class="parameter_description"><p>a field name</p></td>
-<td class="parameter_annotations"> </td>
-</tr>
-</tbody>
-</table></div>
-</div>
-<div class="refsect3">
-<a name="id-1.2.17.6.10.6"></a><h4>Returns</h4>
-<p> <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a> if <em class="parameter"><code>token</code></em>
-has a boolean field named <em class="parameter"><code>field</code></em>
-set to <a href="https://developer.gnome.org/glib/unstable/glib-Standard-Macros.html#TRUE:CAPS"><code class="literal">TRUE</code></a>.</p>
-<p></p>
-</div>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPToken.other_details"></a><h2>Types and Values</h2>
-<div class="refsect2">
-<a name="GstRTSPToken"></a><h3>struct GstRTSPToken</h3>
-<pre class="programlisting">struct GstRTSPToken {
- GstMiniObject mini_object;
-};
-</pre>
-<p>An opaque object used for checking authorisations.
-It is generated after successful authentication.</p>
-</div>
-</div>
-<div class="refsect1">
-<a name="gst-rtsp-server-GstRTSPToken.see-also"></a><h2>See Also</h2>
-<p><a class="link" href="GstRTSPClient.html" title="GstRTSPClient"><span class="type">GstRTSPClient</span></a>, <a class="link" href="gst-rtsp-server-GstRTSPPermissions.html#GstRTSPPermissions" title="struct GstRTSPPermissions"><span class="type">GstRTSPPermissions</span></a>, <a class="link" href="GstRTSPAuth.html" title="GstRTSPAuth"><span class="type">GstRTSPAuth</span></a></p>
-</div>
-</div>
-<div class="footer">
-<hr>
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-<div><table class="navigation" id="top" width="100%" cellpadding="2" cellspacing="0"><tr><th valign="middle"><p class="title">GStreamer RTSP Server Reference Manual</p></th></tr></table></div>
-<div><p class="releaseinfo">
- for GStreamer RTSP Server 1.4.5
- </p></div>
-</div>
-<hr>
-</div>
-<div class="toc"><dl class="toc">
-<dt><span class="chapter"><a href="ch01.html"></a></span></dt>
-<dd><dl>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPServer.html">GstRTSPServer</a></span><span class="refpurpose"> — The main server object</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPClient.html">GstRTSPClient</a></span><span class="refpurpose"> — A client connection state</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPContext.html">GstRTSPContext</a></span><span class="refpurpose"> — A client request context</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPMountPoints.html">GstRTSPMountPoints</a></span><span class="refpurpose"> — Map a path to media</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPMediaFactory.html">GstRTSPMediaFactory</a></span><span class="refpurpose"> — A factory for media pipelines</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPMediaFactoryURI.html">GstRTSPMediaFactoryURI</a></span><span class="refpurpose"> — A factory for URI sources</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPMedia.html">GstRTSPMedia</a></span><span class="refpurpose"> — The media pipeline</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPStream.html">GstRTSPStream</a></span><span class="refpurpose"> — A media stream</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPSessionPool.html">GstRTSPSessionPool</a></span><span class="refpurpose"> — An object for managing sessions</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPSession.html">GstRTSPSession</a></span><span class="refpurpose"> — An object to manage media</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPSessionMedia.html">GstRTSPSessionMedia</a></span><span class="refpurpose"> — Media managed in a session</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPStreamTransport.html">GstRTSPStreamTransport</a></span><span class="refpurpose"> — A media stream transport configuration</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPSdp.html">GstRTSPSdp</a></span><span class="refpurpose"> — Make SDP messages</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPAddressPool.html">GstRTSPAddressPool</a></span><span class="refpurpose"> — A pool of network addresses</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPThreadPool.html">GstRTSPThreadPool</a></span><span class="refpurpose"> — A pool of threads</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="GstRTSPAuth.html">GstRTSPAuth</a></span><span class="refpurpose"> — Authentication and authorization</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPToken.html">GstRTSPToken</a></span><span class="refpurpose"> — Roles and permissions for a client</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPPermissions.html">GstRTSPPermissions</a></span><span class="refpurpose"> — Roles and associated permissions</span>
-</dt>
-<dt>
-<span class="refentrytitle"><a href="gst-rtsp-server-GstRTSPParams.html">GstRTSPParams</a></span><span class="refpurpose"> — Param get and set implementation</span>
-</dt>
-</dl></dd>
-<dt><span class="chapter"><a href="rtsp-server-hierarchy.html">Object Hierarchy</a></span></dt>
-<dt><span class="index"><a href="api-index-full.html">API Index</a></span></dt>
-<dt><span class="glossary"><a href="annotation-glossary.html">Annotation Glossary</a></span></dt>
-</dl></div>
-</div>
-<div class="footer">
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- Generated by GTK-Doc V1.21</div>
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- <span class="lineart">├──</span> <a class="link" href="GstRTSPMediaFactory.html" title="GstRTSPMediaFactory">GstRTSPMediaFactory</a>
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- <span class="lineart">├──</span> <a class="link" href="GstRTSPClient.html" title="GstRTSPClient">GstRTSPClient</a>
- <span class="lineart">╰──</span> <a class="link" href="GstRTSPStream.html" title="GstRTSPStream">GstRTSPStream</a>
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-# Otherwise a system limit (for SysV at least) may be exceeded.
-.NOEXPORT:
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#include <gst/rtsp-server/rtsp-server.h>
-
-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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#include <gst/rtsp-server/rtsp-server.h>
-
-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;
- }
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2013 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <libcgroup.h>
-
-#include <gst/gst.h>
-#include <gst/rtsp-server/rtsp-server.h>
-
-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;
- }
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#include <gst/rtsp-server/rtsp-server.h>
-
-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 <launch line> \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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#include <gst/rtsp-server/rtsp-server.h>
-
-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 <filename.mp4>", 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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#include <gst/rtsp-server/rtsp-server.h>
-
-
-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;
- }
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#include <gst/rtsp-server/rtsp-server.h>
-
-
-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;
- }
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#include <gst/rtsp-server/rtsp-server.h>
-
-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 <filename.ogg>", 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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#include <gst/rtsp-server/rtsp-server.h>
-
-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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2009 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#include <gst/rtsp-server/rtsp-server.h>
-
-
-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 <filename.sdp>", 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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#include <gst/rtsp-server/rtsp-server.h>
-#include <gst/rtsp-server/rtsp-media-factory-uri.h>
-
-
-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 <uri>", 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;
- }
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#include <gst/rtsp-server/rtsp-server.h>
-
-/* 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;
- }
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <glib-unix.h>
-#include <gst/gst.h>
-
-#include <gst/rtsp-server/rtsp-server-wfd.h>
-#include <gst/rtsp-server/rtsp-media-factory-wfd.h>
-
-#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;
- }
-}
+++ /dev/null
-<Project
- xmlns:rdf="http://www.w3.org/1999/02/22-rdf-syntax-ns#"
- xmlns:rdfs="http://www.w3.org/2000/01/rdf-schema#"
- xmlns="http://usefulinc.com/ns/doap#"
- xmlns:foaf="http://xmlns.com/foaf/0.1/"
- xmlns:admin="http://webns.net/mvcb/">
-
- <name>GStreamer RTSP Server</name>
- <shortname>gst-rtsp-server</shortname>
- <homepage rdf:resource="http://gstreamer.freedesktop.org/modules/gst-rtsp-server.html" />
- <created>1999-10-31</created>
- <shortdesc xml:lang="en">
-RTSP server library based on GStreamer
-</shortdesc>
- <description xml:lang="en">
-RTSP server library based on GStreamer
- </description>
- <category></category>
- <bug-database rdf:resource="http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-rtsp-server" />
- <screenshots></screenshots>
- <mailing-list rdf:resource="http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel" />
- <programming-language>C</programming-language>
- <license rdf:resource="http://usefulinc.com/doap/licenses/lgpl" />
- <download-page rdf:resource="http://gstreamer.freedesktop.org/download/" />
-
- <repository>
- <GitRepository>
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- fi; \
- if test -d $(srcdir)/$$file && test $$d != $(srcdir); then \
- cp -fpR $(srcdir)/$$file "$(distdir)$$dir" || exit 1; \
- find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
- fi; \
- cp -fpR $$d/$$file "$(distdir)$$dir" || exit 1; \
- else \
- test -f "$(distdir)/$$file" \
- || cp -p $$d/$$file "$(distdir)/$$file" \
- || exit 1; \
- fi; \
- done
-check-am: all-am
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-installdirs:
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- test -z "$$dir" || $(MKDIR_P) "$$dir"; \
- done
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- install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
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- install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
- "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'" install; \
- fi
-mostlyclean-generic:
-
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- -test -z "$(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
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- mostlyclean-am
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-distclean: distclean-am
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-
-
-@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) \
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-@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:
+++ /dev/null
-/* GStreamer
- * Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.com>
- *
- * 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
- *
- * <refsect2>
- * <para>
- * The GstWFDMessage helper functions makes it easy to parse and create WFD
- * messages.
- * </para>
- * </refsect2>
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include <gio/gio.h>
-
-#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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.com>
- *
- * 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 <glib.h>
-#include <gst/gst.h>
-
-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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2012 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <string.h>
-#include <gio/gio.h>
-
-#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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2012 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#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;
-
- /*<private >*/
- 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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <string.h>
-
-#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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <stdio.h>
-#include <string.h>
-
-#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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-#include <gst/rtsp/gstrtspconnection.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <stdio.h>
-#include <string.h>
-
-#include <gst/sdp/gstmikey.h>
-
-#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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-#include <gst/rtsp/gstrtspconnection.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2013 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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);
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-#include <gst/rtsp/gstrtspconnection.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <string.h>
-
-#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;
- }
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <stdio.h>
-#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;
- }
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-#include <gst/rtsp/gstrtspurl.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <string.h>
-#include <stdlib.h>
-
-#include <gst/app/gstappsrc.h>
-#include <gst/app/gstappsink.h>
-
-#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;
- }
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-#include <gst/rtsp/gstrtsprange.h>
-#include <gst/rtsp/gstrtspurl.h>
-#include <gst/net/gstnet.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <string.h>
-
-#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);
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <string.h>
-
-#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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#include <gst/rtsp/gstrtspurl.h>
-#include <gst/rtsp/gstrtspmessage.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2013 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <string.h>
-
-#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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <string.h>
-
-#include <gst/sdp/gstmikey.h>
-
-#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;
- }
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-#include <gst/sdp/gstsdpmessage.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <stdlib.h>
-#include <string.h>
-
-#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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <stdlib.h>
-#include <string.h>
-
-#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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <string.h>
-
-#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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#include <gst/rtsp/gstrtsptransport.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <string.h>
-
-#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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#include <gst/rtsp/gstrtsptransport.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <string.h>
-#include <stdlib.h>
-
-#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);
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-#include <gst/rtsp/gstrtsprange.h>
-#include <gst/rtsp/gstrtspurl.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <stdlib.h>
-#include <stdio.h>
-#include <string.h>
-
-#include <gio/gio.h>
-
-#include <gst/app/gstappsrc.h>
-#include <gst/app/gstappsink.h>
-
-#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;
-
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-#include <gst/rtsp/gstrtsprange.h>
-#include <gst/rtsp/gstrtspurl.h>
-#include <gio/gio.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2013 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <string.h>
-
-#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;
- }
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#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__ */
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <string.h>
-
-#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;
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2010 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#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__ */
+++ /dev/null
-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
-
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-Summary: Multimedia Framework RTSP server library (DEV)
-Group: Development/Libraries
-Requires: %{name} = %{version}-%{release}
-
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-Summary: Multimedia Framework RTSP server Library (Factory)
-Group: Development/Libraries
-Requires: %{name} = %{version}-%{release}
-
-%description factory
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-%prep
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-make %{?jobs:-j%jobs}
-
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-cp -rf %{_builddir}/%{name}-%{version}/COPYING %{buildroot}%{_datadir}/license/%{name}
-
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-rm -rf %{buildroot}
-
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-
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-%{_includedir}/gstreamer-1.0/gst/rtsp-server/gstwfd*.h
-%{_libdir}/pkgconfig/*
+++ /dev/null
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- if (line ~ /$(am__global_test_result_rx)/) \
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- rst_section(global_test_result ": " $$0); \
- while ((rc = (getline line < ($$0 ".log"))) != 0) \
- { \
- if (rc < 0) \
- fatal("failed to read from " $$0 ".log"); \
- print line; \
- }; \
- printf "\n"; \
- }; \
- close ($$0 ".trs"); \
- close ($$0 ".log"); \
-}'
-# Restructured Text title.
-am__rst_title = { sed 's/.*/ & /;h;s/./=/g;p;x;s/ *$$//;p;g' && echo; }
-# Solaris 10 'make', and several other traditional 'make' implementations,
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- *) am__odir=.;; \
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-AMTAR = @AMTAR@
-AM_DEFAULT_VERBOSITY = @AM_DEFAULT_VERBOSITY@
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-DEFAULT_AUDIOSRC = @DEFAULT_AUDIOSRC@
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-DLLTOOL = @DLLTOOL@
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-DSYMUTIL = @DSYMUTIL@
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-GIO_CFLAGS = @GIO_CFLAGS@
-GIO_LDFLAGS = @GIO_LDFLAGS@
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-GLIB_GENMARSHAL = @GLIB_GENMARSHAL@
-GLIB_LIBS = @GLIB_LIBS@
-GLIB_MKENUMS = @GLIB_MKENUMS@
-GLIB_REQ = @GLIB_REQ@
-GREP = @GREP@
-GSTPB_PLUGINS_DIR = @GSTPB_PLUGINS_DIR@
-GSTPD_PLUGINS_DIR = @GSTPD_PLUGINS_DIR@
-GSTPG_PLUGINS_DIR = @GSTPG_PLUGINS_DIR@
-GST_AGE = @GST_AGE@
-GST_ALL_CFLAGS = @GST_ALL_CFLAGS@
-GST_ALL_LDFLAGS = @GST_ALL_LDFLAGS@
-GST_ALL_LIBS = @GST_ALL_LIBS@
-GST_API_VERSION = @GST_API_VERSION@
-GST_BASE_CFLAGS = @GST_BASE_CFLAGS@
-GST_BASE_LIBS = @GST_BASE_LIBS@
-GST_CFLAGS = @GST_CFLAGS@
-GST_CHECK_CFLAGS = @GST_CHECK_CFLAGS@
-GST_CHECK_LIBS = @GST_CHECK_LIBS@
-GST_CURRENT = @GST_CURRENT@
-GST_LEVEL_DEFAULT = @GST_LEVEL_DEFAULT@
-GST_LIBS = @GST_LIBS@
-GST_LIBVERSION = @GST_LIBVERSION@
-GST_LIB_LDFLAGS = @GST_LIB_LDFLAGS@
-GST_LICENSE = @GST_LICENSE@
-GST_LT_LDFLAGS = @GST_LT_LDFLAGS@
-GST_OBJ_CFLAGS = @GST_OBJ_CFLAGS@
-GST_OBJ_LIBS = @GST_OBJ_LIBS@
-GST_OPTION_CFLAGS = @GST_OPTION_CFLAGS@
-GST_PACKAGE_NAME = @GST_PACKAGE_NAME@
-GST_PACKAGE_ORIGIN = @GST_PACKAGE_ORIGIN@
-GST_PKG_CONFIG_PATH = @GST_PKG_CONFIG_PATH@
-GST_PLUGINS_BAD_CFLAGS = @GST_PLUGINS_BAD_CFLAGS@
-GST_PLUGINS_BAD_DIR = @GST_PLUGINS_BAD_DIR@
-GST_PLUGINS_BAD_LIBS = @GST_PLUGINS_BAD_LIBS@
-GST_PLUGINS_BASE_CFLAGS = @GST_PLUGINS_BASE_CFLAGS@
-GST_PLUGINS_BASE_DIR = @GST_PLUGINS_BASE_DIR@
-GST_PLUGINS_BASE_LIBS = @GST_PLUGINS_BASE_LIBS@
-GST_PLUGINS_DIR = @GST_PLUGINS_DIR@
-GST_PLUGINS_GOOD_CFLAGS = @GST_PLUGINS_GOOD_CFLAGS@
-GST_PLUGINS_GOOD_DIR = @GST_PLUGINS_GOOD_DIR@
-GST_PLUGINS_GOOD_LIBS = @GST_PLUGINS_GOOD_LIBS@
-GST_REVISION = @GST_REVISION@
-GST_TOOLS_DIR = @GST_TOOLS_DIR@
-GTKDOC_CHECK = @GTKDOC_CHECK@
-GTKDOC_DEPS_CFLAGS = @GTKDOC_DEPS_CFLAGS@
-GTKDOC_DEPS_LIBS = @GTKDOC_DEPS_LIBS@
-GTKDOC_MKPDF = @GTKDOC_MKPDF@
-GTKDOC_REBASE = @GTKDOC_REBASE@
-HAVE_DOCBOOK2PS = @HAVE_DOCBOOK2PS@
-HAVE_DVIPS = @HAVE_DVIPS@
-HAVE_EPSTOPDF = @HAVE_EPSTOPDF@
-HAVE_JADETEX = @HAVE_JADETEX@
-HAVE_PNGTOPNM = @HAVE_PNGTOPNM@
-HAVE_PNMTOPS = @HAVE_PNMTOPS@
-HAVE_PS2PDF = @HAVE_PS2PDF@
-HAVE_XMLLINT = @HAVE_XMLLINT@
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-HTML_DIR = @HTML_DIR@
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-INTROSPECTION_LIBS = @INTROSPECTION_LIBS@
-INTROSPECTION_MAKEFILE = @INTROSPECTION_MAKEFILE@
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-INTROSPECTION_TYPELIBDIR = @INTROSPECTION_TYPELIBDIR@
-LD = @LD@
-LDFLAGS = @LDFLAGS@
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-LIBCGROUP_LIBS = @LIBCGROUP_LIBS@
-LIBOBJS = @LIBOBJS@
-LIBS = @LIBS@
-LIBTOOL = @LIBTOOL@
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-MAINT = @MAINT@
-MAKEINFO = @MAKEINFO@
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-MKDIR_P = @MKDIR_P@
-NM = @NM@
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-OBJDUMP = @OBJDUMP@
-OBJEXT = @OBJEXT@
-OTOOL = @OTOOL@
-OTOOL64 = @OTOOL64@
-PACKAGE = @PACKAGE@
-PACKAGE_BUGREPORT = @PACKAGE_BUGREPORT@
-PACKAGE_NAME = @PACKAGE_NAME@
-PACKAGE_STRING = @PACKAGE_STRING@
-PACKAGE_TARNAME = @PACKAGE_TARNAME@
-PACKAGE_URL = @PACKAGE_URL@
-PACKAGE_VERSION = @PACKAGE_VERSION@
-PACKAGE_VERSION_MAJOR = @PACKAGE_VERSION_MAJOR@
-PACKAGE_VERSION_MICRO = @PACKAGE_VERSION_MICRO@
-PACKAGE_VERSION_MINOR = @PACKAGE_VERSION_MINOR@
-PACKAGE_VERSION_NANO = @PACKAGE_VERSION_NANO@
-PACKAGE_VERSION_RELEASE = @PACKAGE_VERSION_RELEASE@
-PATH_SEPARATOR = @PATH_SEPARATOR@
-PKG_CONFIG = @PKG_CONFIG@
-PLUGINDIR = @PLUGINDIR@
-PROFILE_CFLAGS = @PROFILE_CFLAGS@
-RANLIB = @RANLIB@
-SED = @SED@
-SET_MAKE = @SET_MAKE@
-SHELL = @SHELL@
-STRIP = @STRIP@
-VALGRIND_CFLAGS = @VALGRIND_CFLAGS@
-VALGRIND_LIBS = @VALGRIND_LIBS@
-VALGRIND_PATH = @VALGRIND_PATH@
-VERSION = @VERSION@
-WARNING_CFLAGS = @WARNING_CFLAGS@
-XML_CATALOG = @XML_CATALOG@
-XSLTPROC = @XSLTPROC@
-XSLTPROC_FLAGS = @XSLTPROC_FLAGS@
-abs_builddir = @abs_builddir@
-abs_srcdir = @abs_srcdir@
-abs_top_builddir = @abs_top_builddir@
-abs_top_srcdir = @abs_top_srcdir@
-ac_ct_AR = @ac_ct_AR@
-ac_ct_CC = @ac_ct_CC@
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-am__leading_dot = @am__leading_dot@
-am__quote = @am__quote@
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-build = @build@
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-build_vendor = @build_vendor@
-builddir = @builddir@
-datadir = @datadir@
-datarootdir = @datarootdir@
-docdir = @docdir@
-dvidir = @dvidir@
-exec_prefix = @exec_prefix@
-host = @host@
-host_alias = @host_alias@
-host_cpu = @host_cpu@
-host_os = @host_os@
-host_vendor = @host_vendor@
-htmldir = @htmldir@
-includedir = @includedir@
-infodir = @infodir@
-install_sh = @install_sh@
-libdir = @libdir@
-libexecdir = @libexecdir@
-localedir = @localedir@
-localstatedir = @localstatedir@
-mandir = @mandir@
-mkdir_p = @mkdir_p@
-oldincludedir = @oldincludedir@
-pdfdir = @pdfdir@
-plugindir = @plugindir@
-prefix = @prefix@
-program_transform_name = @program_transform_name@
-psdir = @psdir@
-sbindir = @sbindir@
-sharedstatedir = @sharedstatedir@
-srcdir = @srcdir@
-sysconfdir = @sysconfdir@
-target = @target@
-target_alias = @target_alias@
-target_cpu = @target_cpu@
-target_os = @target_os@
-target_vendor = @target_vendor@
-top_build_prefix = @top_build_prefix@
-top_builddir = @top_builddir@
-top_srcdir = @top_srcdir@
-
-# inspect every plugin feature
-GST_INSPECT = $(GST_TOOLS_DIR)/gst-inspect-$(GST_API_VERSION)
-CHECK_REGISTRY = $(top_builddir)/tests/check/test-registry.reg
-TEST_FILES_DIRECTORY = $(top_srcdir)/tests/files
-REGISTRY_ENVIRONMENT = \
- GST_REGISTRY_1_0=$(CHECK_REGISTRY)
-
-TESTS_ENVIRONMENT = \
- CK_DEFAULT_TIMEOUT=120 \
- GST_STATE_IGNORE_ELEMENTS="$(STATE_IGNORE_ELEMENTS)" \
- $(REGISTRY_ENVIRONMENT) \
- GST_PLUGIN_SYSTEM_PATH_1_0= \
- GST_PLUGIN_PATH_1_0=$(GST_PLUGINS_DIR):$(GSTPB_PLUGINS_DIR):$(GSTPG_PLUGINS_DIR):$(GSTPD_PLUGINS_DIR) \
- GST_PLUGIN_LOADING_WHITELIST="gstreamer:gst-plugins-base:gst-plugins-good:gst-plugins-bad"
-
-
-# ths core dumps of some machines have PIDs appended
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-TESTS = $(check_PROGRAMS)
-AM_CFLAGS = -I$(top_srcdir)/gst/rtsp-server \
- $(GST_PLUGINS_BASE_CFLAGS) \
- $(GST_BASE_CFLAGS) \
- $(GIO_CFLAGS) \
- $(GST_CFLAGS) \
- $(GST_CHECK_CFLAGS) \
- -DGST_TEST_FILES_PATH="\"$(TEST_FILES_DIRECTORY)\"" \
- -UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS
-
-AM_CXXFLAGS = $(GST_CXXFLAGS) $(GST_CHECK_CFLAGS) \
- -DGST_TEST_FILES_PATH="\"$(TEST_FILES_DIRECTORY)\"" \
- -UG_DISABLE_ASSERT -UG_DISABLE_CAST_CHECKS
-
-LDADD = $(top_builddir)/gst/rtsp-server/libgstrtspserver-@GST_API_VERSION@.la \
- $(GST_PLUGINS_BASE_LIBS) -lgstrtp-@GST_API_VERSION@ \
- -lgstrtsp-@GST_API_VERSION@ -lgstsdp-@GST_API_VERSION@ \
- $(GST_BASE_LIBS) $(GIO_LIBS) \
- $(GST_LIBS) $(GST_CHECK_LIBS) $(GST_RTSP_SERVER_LIBS)
-
-SUPPRESSIONS = $(top_srcdir)/common/gst.supp
-all: all-am
-
-.SUFFIXES:
-.SUFFIXES: .c .lo .log .o .obj .test .test$(EXEEXT) .trs
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- && { if test -f $@; then exit 0; else break; fi; }; \
- exit 1;; \
- esac; \
- done; \
- echo ' cd $(top_srcdir) && $(AUTOMAKE) --gnu tests/check/Makefile'; \
- $(am__cd) $(top_srcdir) && \
- $(AUTOMAKE) --gnu tests/check/Makefile
-.PRECIOUS: Makefile
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- @case '$?' in \
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- echo ' cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@ $(am__depfiles_maybe)'; \
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-$(top_builddir)/config.status: $(top_srcdir)/configure $(CONFIG_STATUS_DEPENDENCIES)
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-$(top_srcdir)/configure: @MAINTAINER_MODE_TRUE@ $(am__configure_deps)
- cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
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- cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
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- rm -f $$list || exit $$?; \
- test -n "$(EXEEXT)" || exit 0; \
- list=`for p in $$list; do echo "$$p"; done | sed 's/$(EXEEXT)$$//'`; \
- echo " rm -f" $$list; \
- rm -f $$list
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- @list='$(noinst_PROGRAMS)'; test -n "$$list" || exit 0; \
- echo " rm -f" $$list; \
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- test -n "$(EXEEXT)" || exit 0; \
- list=`for p in $$list; do echo "$$p"; done | sed 's/$(EXEEXT)$$//'`; \
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- rm -f $$list
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- @$(MKDIR_P) gst
- @: > gst/$(am__dirstamp)
-gst/$(DEPDIR)/$(am__dirstamp):
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- @: > gst/$(DEPDIR)/$(am__dirstamp)
-gst/addresspool.$(OBJEXT): gst/$(am__dirstamp) \
- gst/$(DEPDIR)/$(am__dirstamp)
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- $(AM_V_CCLD)$(LINK) $(gst_addresspool_OBJECTS) $(gst_addresspool_LDADD) $(LIBS)
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- @rm -f gst/client$(EXEEXT)
- $(AM_V_CCLD)$(LINK) $(gst_client_OBJECTS) $(gst_client_LDADD) $(LIBS)
-gst/media.$(OBJEXT): gst/$(am__dirstamp) gst/$(DEPDIR)/$(am__dirstamp)
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- $(AM_V_CCLD)$(LINK) $(gst_mediafactory_OBJECTS) $(gst_mediafactory_LDADD) $(LIBS)
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-gst/mountpoints$(EXEEXT): $(gst_mountpoints_OBJECTS) $(gst_mountpoints_DEPENDENCIES) $(EXTRA_gst_mountpoints_DEPENDENCIES) gst/$(am__dirstamp)
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- $(AM_V_CCLD)$(LINK) $(gst_mountpoints_OBJECTS) $(gst_mountpoints_LDADD) $(LIBS)
-gst/permissions.$(OBJEXT): gst/$(am__dirstamp) \
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-gst/permissions$(EXEEXT): $(gst_permissions_OBJECTS) $(gst_permissions_DEPENDENCIES) $(EXTRA_gst_permissions_DEPENDENCIES) gst/$(am__dirstamp)
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- $(AM_V_CCLD)$(LINK) $(gst_permissions_OBJECTS) $(gst_permissions_LDADD) $(LIBS)
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- $(AM_V_CCLD)$(LINK) $(gst_rtspserver_OBJECTS) $(gst_rtspserver_LDADD) $(LIBS)
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- $(AM_V_CCLD)$(LINK) $(gst_sessionmedia_OBJECTS) $(gst_sessionmedia_LDADD) $(LIBS)
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- -test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
- -test . = "$(srcdir)" || test -z "$(CONFIG_CLEAN_VPATH_FILES)" || rm -f $(CONFIG_CLEAN_VPATH_FILES)
- -rm -f gst/$(DEPDIR)/$(am__dirstamp)
- -rm -f gst/$(am__dirstamp)
-
-maintainer-clean-generic:
- @echo "This command is intended for maintainers to use"
- @echo "it deletes files that may require special tools to rebuild."
-clean: clean-am
-
-clean-am: clean-checkPROGRAMS clean-generic clean-libtool clean-local \
- clean-noinstPROGRAMS mostlyclean-am
-
-distclean: distclean-am
- -rm -rf gst/$(DEPDIR)
- -rm -f Makefile
-distclean-am: clean-am distclean-compile distclean-generic \
- distclean-tags
-
-dvi: dvi-am
-
-dvi-am:
-
-html: html-am
-
-html-am:
-
-info: info-am
-
-info-am:
-
-install-data-am:
-
-install-dvi: install-dvi-am
-
-install-dvi-am:
-
-install-exec-am:
-
-install-html: install-html-am
-
-install-html-am:
-
-install-info: install-info-am
-
-install-info-am:
-
-install-man:
-
-install-pdf: install-pdf-am
-
-install-pdf-am:
-
-install-ps: install-ps-am
-
-install-ps-am:
-
-installcheck-am:
-
-maintainer-clean: maintainer-clean-am
- -rm -rf gst/$(DEPDIR)
- -rm -f Makefile
-maintainer-clean-am: distclean-am maintainer-clean-generic
-
-mostlyclean: mostlyclean-am
-
-mostlyclean-am: mostlyclean-compile mostlyclean-generic \
- mostlyclean-libtool
-
-pdf: pdf-am
-
-pdf-am:
-
-ps: ps-am
-
-ps-am:
-
-uninstall-am:
-
-.MAKE: check-am install-am install-strip
-
-.PHONY: CTAGS GTAGS TAGS all all-am check check-TESTS check-am clean \
- clean-checkPROGRAMS clean-generic clean-libtool clean-local \
- clean-noinstPROGRAMS cscopelist-am ctags ctags-am distclean \
- distclean-compile distclean-generic distclean-libtool \
- distclean-tags distdir dvi dvi-am html html-am info info-am \
- install install-am install-data install-data-am install-dvi \
- install-dvi-am install-exec install-exec-am install-html \
- install-html-am install-info install-info-am install-man \
- install-pdf install-pdf-am install-ps install-ps-am \
- install-strip installcheck installcheck-am installdirs \
- maintainer-clean maintainer-clean-generic mostlyclean \
- mostlyclean-compile mostlyclean-generic mostlyclean-libtool \
- pdf pdf-am ps ps-am recheck tags tags-am uninstall \
- uninstall-am
-
-
-# keep target around, since it's referenced in the modules' Makefiles
-clean-local-check:
- @echo
-
-# hangs spectacularly on some machines, so let's not do this by default yet
-@HAVE_VALGRIND_TRUE@check-valgrind:
-@HAVE_VALGRIND_TRUE@ $(MAKE) valgrind
-@HAVE_VALGRIND_FALSE@check-valgrind:
-@HAVE_VALGRIND_FALSE@ @true
-
-LOOPS ?= 10
-
-# run any given test by running make test.check
-# if the test fails, run it again at at least debug level 2
-%.check: %
- @$(TESTS_ENVIRONMENT) \
- CK_DEFAULT_TIMEOUT=20 \
- $* || \
- $(TESTS_ENVIRONMENT) \
- GST_DEBUG=$$GST_DEBUG,*:2 \
- CK_DEFAULT_TIMEOUT=20 \
- $*
-
-# just like 'check', but don't run it again if it fails (useful for debugging)
-%.check-norepeat: %
- @$(TESTS_ENVIRONMENT) \
- CK_DEFAULT_TIMEOUT=20 \
- $*
-
-# run any given test in a loop
-%.torture: %
- @for i in `seq 1 $(LOOPS)`; do \
- $(TESTS_ENVIRONMENT) \
- CK_DEFAULT_TIMEOUT=20 \
- $*; done
-
-# run any given test in an infinite loop
-%.forever: %
- @while true; do \
- $(TESTS_ENVIRONMENT) \
- CK_DEFAULT_TIMEOUT=20 \
- $* || break; done
-
-# valgrind any given test by running make test.valgrind
-%.valgrind: %
- @$(TESTS_ENVIRONMENT) \
- CK_DEFAULT_TIMEOUT=360 \
- G_SLICE=always-malloc \
- $(LIBTOOL) --mode=execute \
- $(VALGRIND_PATH) -q \
- $(foreach s,$(SUPPRESSIONS),--suppressions=$(s)) \
- --tool=memcheck --leak-check=full --trace-children=yes \
- --show-possibly-lost=no \
- --leak-resolution=high --num-callers=20 \
- ./$* 2>&1 | tee valgrind.log
- @if grep "==" valgrind.log > /dev/null 2>&1; then \
- rm valgrind.log; \
- exit 1; \
- fi
- @rm valgrind.log
-
-# valgrind any given test and generate suppressions for it
-%.valgrind.gen-suppressions: %
- @$(TESTS_ENVIRONMENT) \
- CK_DEFAULT_TIMEOUT=360 \
- G_SLICE=always-malloc \
- $(LIBTOOL) --mode=execute \
- $(VALGRIND_PATH) -q \
- $(foreach s,$(SUPPRESSIONS),--suppressions=$(s)) \
- --tool=memcheck --leak-check=full --trace-children=yes \
- --show-possibly-lost=no \
- --leak-resolution=high --num-callers=20 \
- --gen-suppressions=all \
- ./$* 2>&1 | tee suppressions.log
-
-# valgrind torture any given test
-%.valgrind-torture: %
- @for i in `seq 1 $(LOOPS)`; do \
- $(MAKE) $*.valgrind || \
- (echo "Failure after $$i runs"; exit 1) || \
- exit 1; \
- done
- @banner="All $(LOOPS) loops passed"; \
- dashes=`echo "$$banner" | sed s/./=/g`; \
- echo $$dashes; echo $$banner; echo $$dashes
-
-# valgrind any given test until failure by running make test.valgrind-forever
-%.valgrind-forever: %
- @while $(MAKE) $*.valgrind; do \
- true; done
-
-# gdb any given test by running make test.gdb
-%.gdb: %
- @$(TESTS_ENVIRONMENT) \
- CK_FORK=no \
- $(LIBTOOL) --mode=execute \
- gdb $*
-
-%.lcov-reset:
- $(MAKE) $*.lcov-run
- $(MAKE) $*.lcov-report
-
-%.lcov: %
- $(MAKE) $*.lcov-reset
-
-@GST_GCOV_ENABLED_TRUE@%.lcov-clean:
-@GST_GCOV_ENABLED_TRUE@ $(MAKE) -C $(top_builddir) lcov-clean
-
-@GST_GCOV_ENABLED_TRUE@%.lcov-run:
-@GST_GCOV_ENABLED_TRUE@ $(MAKE) $*.lcov-clean
-@GST_GCOV_ENABLED_TRUE@ $(MAKE) $*.check
-
-@GST_GCOV_ENABLED_TRUE@%.lcov-report:
-@GST_GCOV_ENABLED_TRUE@ $(MAKE) -C $(top_builddir) lcov-report
-@GST_GCOV_ENABLED_FALSE@%.lcov-run:
-@GST_GCOV_ENABLED_FALSE@ echo "Need to reconfigure with --enable-gcov"
-
-@GST_GCOV_ENABLED_FALSE@%.lcov-report:
-@GST_GCOV_ENABLED_FALSE@ echo "Need to reconfigure with --enable-gcov"
-
-# torture tests
-torture: $(TESTS)
- -rm test-registry.*
- @echo "Torturing tests ..."
- @for i in `seq 1 $(LOOPS)`; do \
- $(MAKE) check || \
- (echo "Failure after $$i runs"; exit 1) || \
- exit 1; \
- done
- @banner="All $(LOOPS) loops passed"; \
- dashes=`echo "$$banner" | sed s/./=/g`; \
- echo $$dashes; echo $$banner; echo $$dashes
-
-# forever tests
-forever: $(TESTS)
- -rm test-registry.*
- @echo "Forever tests ..."
- @while true; do \
- $(MAKE) check || \
- (echo "Failure"; exit 1) || \
- exit 1; \
- done
-
-# valgrind all tests
-valgrind: $(TESTS)
- @echo "Valgrinding tests ..."
- @failed=0; \
- for t in $(filter-out $(VALGRIND_TESTS_DISABLE),$(TESTS)); do \
- $(MAKE) $$t.valgrind; \
- if test "$$?" -ne 0; then \
- echo "Valgrind error for test $$t"; \
- failed=`expr $$failed + 1`; \
- whicht="$$whicht $$t"; \
- fi; \
- done; \
- if test "$$failed" -ne 0; then \
- echo "$$failed tests had leaks or errors under valgrind:"; \
- echo "$$whicht"; \
- false; \
- fi
-
-# valgrind all tests until failure
-valgrind-forever: $(TESTS)
- -rm test-registry.*
- @echo "Forever valgrinding tests ..."
- @while true; do \
- $(MAKE) valgrind || \
- (echo "Failure"; exit 1) || \
- exit 1; \
- done
-
-# valgrind torture all tests
-valgrind-torture: $(TESTS)
- -rm test-registry.*
- @echo "Torturing and valgrinding tests ..."
- @for i in `seq 1 $(LOOPS)`; do \
- $(MAKE) valgrind || \
- (echo "Failure after $$i runs"; exit 1) || \
- exit 1; \
- done
- @banner="All $(LOOPS) loops passed"; \
- dashes=`echo "$$banner" | sed s/./=/g`; \
- echo $$dashes; echo $$banner; echo $$dashes
-
-# valgrind all tests and generate suppressions
-valgrind.gen-suppressions: $(TESTS)
- @echo "Valgrinding tests ..."
- @failed=0; \
- for t in $(filter-out $(VALGRIND_TESTS_DISABLE),$(TESTS)); do \
- $(MAKE) $$t.valgrind.gen-suppressions; \
- if test "$$?" -ne 0; then \
- echo "Valgrind error for test $$t"; \
- failed=`expr $$failed + 1`; \
- whicht="$$whicht $$t"; \
- fi; \
- done; \
- if test "$$failed" -ne 0; then \
- echo "$$failed tests had leaks or errors under valgrind:"; \
- echo "$$whicht"; \
- false; \
- fi
-inspect:
- @echo "Inspecting features ..."
- @for e in `$(TESTS_ENVIRONMENT) $(GST_INSPECT) | head -n -2 \
- | cut -d: -f2`; \
- do echo Inspecting $$e; \
- $(GST_INSPECT) $$e > /dev/null 2>&1; done
-
-help:
- @echo
- @echo "make check -- run all checks"
- @echo "make torture -- run all checks $(LOOPS) times"
- @echo "make (dir)/(test).check -- run the given check once, repeat with GST_DEBUG=*:2 if it fails"
- @echo "make (dir)/(test).check-norepeat -- run the given check once, but don't run it again if it fails"
- @echo "make (dir)/(test).forever -- run the given check forever"
- @echo "make (dir)/(test).torture -- run the given check $(LOOPS) times"
- @echo
- @echo "make (dir)/(test).gdb -- start up gdb for the given test"
- @echo
- @echo "make valgrind -- valgrind all tests"
- @echo "make valgrind-forever -- valgrind all tests forever"
- @echo "make valgrind-torture -- valgrind all tests $(LOOPS) times"
- @echo "make valgrind.gen-suppressions -- generate suppressions for all tests"
- @echo " and save to suppressions.log"
- @echo "make (dir)/(test).valgrind -- valgrind the given test"
- @echo "make (dir)/(test).valgrind-forever -- valgrind the given test forever"
- @echo "make (dir)/(test).valgrind-torture -- valgrind the given test $(LOOPS) times"
- @echo "make (dir)/(test).valgrind.gen-suppressions -- generate suppressions"
- @echo " and save to suppressions.log"
- @echo "make inspect -- inspect all plugin features"
- @echo
- @echo
- @echo "Additionally, you can use the GST_CHECKS environment variable to"
- @echo "specify which test(s) should be run. This is useful if you are"
- @echo "debugging a failure in one particular test, or want to reproduce"
- @echo "a race condition in a single test."
- @echo
- @echo "Examples:"
- @echo
- @echo " GST_CHECKS=test_this,test_that make element/foobar.check"
- @echo " GST_CHECKS=test_many_threads make element/foobar.forever"
- @echo
-
-clean-local: clean-local-check
-
-$(CHECK_REGISTRY):
- $(TESTS_ENVIRONMENT)
-
-# Tell versions [3.59,3.63) of GNU make to not export all variables.
-# Otherwise a system limit (for SysV at least) may be exceeded.
-.NOEXPORT:
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2012 Wim Taymans <wim.taymans@gmail.com>
- *
- * 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 <gst/check/gstcheck.h>
-
-#include <rtsp-address-pool.h>
-
-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);
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2012 Wim Taymans <wim.taymans@gmail.com>
- *
- * 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 <gst/check/gstcheck.h>
-
-#include <rtsp-client.h>
-
-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);
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2012 Wim Taymans <wim.taymans@gmail.com>
- *
- * 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 <gst/check/gstcheck.h>
-
-#include <rtsp-media-factory.h>
-
-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);
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2012 Wim Taymans <wim.taymans@gmail.com>
- *
- * 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 <gst/check/gstcheck.h>
-
-#include <rtsp-media-factory.h>
-
-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);
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2012 Wim Taymans <wim.taymans@gmail.com>
- *
- * 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 <gst/check/gstcheck.h>
-
-#include <rtsp-mount-points.h>
-
-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);
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2013 Sebastian Rasmussen <sebras@hotmail.com>
- *
- * 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 <gst/check/gstcheck.h>
-
-#include <rtsp-permissions.h>
-
-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);
+++ /dev/null
-/* GStreamer
- *
- * unit test for GstRTSPServer
- *
- * Copyright (C) 2012 Axis Communications <dev-gstreamer at axis dot com>
- * @author David Svensson Fors <davidsf at axis dot com>
- *
- * 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 <gst/check/gstcheck.h>
-#include <gst/sdp/gstsdpmessage.h>
-#include <gst/rtp/gstrtpbuffer.h>
-#include <gst/rtp/gstrtcpbuffer.h>
-
-#include <stdio.h>
-#include <netinet/in.h>
-
-#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);
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2013 Branko Subasic <branko.subasic@axis.com>
- *
- * 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 <gst/check/gstcheck.h>
-
-#include <rtsp-media-factory.h>
-#include <rtsp-session-media.h>
-
-#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);
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2014 Sebastian Rasmussen <sebras@hotmail.com>
- *
- * 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 <gst/check/gstcheck.h>
-#include <rtsp-session-pool.h>
-
-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);
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2013 Axis Communications AB <dev-gstreamer at axis dot com>
- *
- * 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 <gst/check/gstcheck.h>
-
-#include <rtsp-stream.h>
-#include <rtsp-address-pool.h>
-
-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);
+++ /dev/null
-/* GStreamer
- * unit tests for GstRTSPThreadPool
- * Copyright (C) 2013 Axis Communications <dev-gstreamer at axis dot com>
- * @author Ognyan Tonchev <ognyan at axis dot com>
- *
- * 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 <gst/check/gstcheck.h>
-
-#include <rtsp-thread-pool.h>
-
-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);
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2013 Sebastian Rasmussen <sebras@hotmail.com>
- *
- * 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 <gst/check/gstcheck.h>
-
-#include <rtsp-token.h>
-
-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);
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#include <gst/rtsp-server/rtsp-server.h>
-
-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;
- }
-}
+++ /dev/null
-/* GStreamer
- * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
- *
- * 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 <gst/gst.h>
-
-#include <gst/rtsp-server/rtsp-server.h>
-
-#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;
- }
-}
projects / platform / upstream / gst-rtsp-server.git / commitdiff
