gstreamer

mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-09-23 20:40:24 +00:00

Author	SHA1	Message	Date
Wim Taymans	7a947e8dfe	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	2013-12-18 15:57:03 +01:00
Tim-Philipp Müller	91fac8eb29	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.	2013-12-12 00:36:07 +00:00
Aleix Conchillo Flaqué	ab3651d339	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	2013-12-09 17:14:26 +01:00
Wim Taymans	9473fa0d2c	stream-transport: free url in finalize	2013-11-29 15:50:52 +01:00
Ognyan Tonchev	3b4894c4f1	media: also do state change in suspended state	2013-11-29 15:50:23 +01:00
Wim Taymans	53859ac34b	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.	2013-11-29 10:53:08 +01:00
Wim Taymans	421499c102	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 17:47:18 +01:00
Wim Taymans	568477d9b5	stream-transport: add method to get/set url	2013-11-28 17:35:45 +01:00
Wim Taymans	b1e8172ef3	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 16:18:40 +01:00
Wim Taymans	2f17369e9d	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 16:18:39 +01:00
Wim Taymans	db771c5167	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 16:18:39 +01:00
Wim Taymans	6ce48c51a2	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 16:18:39 +01:00
Wim Taymans	bdef631218	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	2013-11-28 16:18:39 +01:00
Wim Taymans	b3baa2801d	media: move default implementations to where they are used	2013-11-26 17:23:04 +01:00
Wim Taymans	b8ae2570d9	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 16:25:37 +01:00
Wim Taymans	9da7b5eeb5	media: handle add-added on non-bins too Handle dynamic payloaders that are not bins, as used in the unit-test.	2013-11-26 16:24:35 +01:00
Sebastian Rasmussen	1ebc2c703e	rtsp-media/-factory: Fix request pad name comments These must be escaped for gtk-doc to parse the comments without warnings.	2013-11-22 11:53:04 +01:00
Aleix Conchillo Flaque	b6d4a29d75	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 11:25:15 +01:00
Wim Taymans	7b5763179a	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	2013-11-22 11:19:35 +01:00
Aleix Conchillo Flaque	e5332535a7	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 11:19:35 +01:00
Sebastian Rasmussen	08160e0913	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 09:13:14 +00:00
Sebastian Rasmussen	d1a2853659	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 09:13:08 +00:00
Tim-Philipp Müller	33c4bdfa01	rtsp-server: sprinkle some allow-none annotations for g-i	2013-11-18 10:47:04 +00:00
Wim Taymans	a106950f70	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	2013-11-18 11:18:15 +01:00
Wim Taymans	b0f609ce7f	rtsp: allow NULL func in filters Passing a null function make the filters return a list of refcounted objects.	2013-11-15 16:35:05 +01:00
Wim Taymans	b2bc84cdbf	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 16:54:59 +01:00
Patricia Muscalu	adc02db975	client: allow absolute path in requests Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711689	2013-11-12 12:21:50 +01:00
Patricia Muscalu	8ce453d97d	client: make make_path_from_uri a vmethod	2013-11-12 12:08:06 +01:00
Wim Taymans	8d5ce0d4ee	stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100	2013-11-12 12:04:55 +01:00
Wim Taymans	d443f8546b	context: defing a GType for the context Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710018	2013-11-12 11:21:55 +01:00
Sebastian Pölsterl	e756324490	Fixed several GIR warnings	2013-11-12 11:15:58 +01:00
Wim Taymans	533d237754	auth: small typos	2013-11-12 11:15:46 +01:00
Sebastian Rasmussen	e4509ed92c	token: Validate args for gst_rtsp_token_is_allowed See https://bugzilla.gnome.org/show_bug.cgi?id=710520	2013-11-12 10:38:37 +01:00
Sebastian Rasmussen	c9025a58e1	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 10:37:45 +01:00
Wim Taymans	c4ae13dbb5	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.	2013-11-12 10:28:55 +01:00
Sebastian Rasmussen	4036f210af	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	2013-11-12 09:54:04 +01:00
Ognyan Tonchev	7b34d1e915	rtsp-server: Fix socket leak https://bugzilla.gnome.org/show_bug.cgi?id=710088	2013-11-04 20:08:06 +01:00
Sebastian Dröge	935e8f852d	rtsp-session-pool: Make sure session IDs are properly URI-escaped https://bugzilla.gnome.org/show_bug.cgi?id=643812	2013-10-30 22:16:54 +01:00
Ognyan Tonchev	78e5a9148e	thread-pool: Fix thread leak when reusing threads https://bugzilla.gnome.org/show_bug.cgi?id=709730	2013-10-30 19:01:48 +01:00
Patricia Muscalu	de7be1c9b2	tests: fixed racy behavior in rtspserver tests https://bugzilla.gnome.org/show_bug.cgi?id=710078	2013-10-15 11:28:27 +02:00
Wim Taymans	d4b8a8249c	client: append query string in PAUSE/PLAY/TEARDOWN as well	2013-10-04 06:29:30 +02:00
Jonas Holmberg	1742399e23	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 06:14:32 +02:00
Wim Taymans	59b53c90c3	rtsp-media: remove old line	2013-10-04 05:48:52 +02:00
Jonas Holmberg	fcf51d3485	stream: Correct control comparison https://bugzilla.gnome.org/show_bug.cgi?id=709176	2013-10-02 11:57:06 +02:00
Youness Alaoui	917bbfcc20	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.	2013-10-02 06:00:10 +02:00
Youness Alaoui	33dc78209c	media: Return FALSE if seeking is not supported	2013-10-02 05:57:15 +02:00
Wim Taymans	d138f79da0	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.	2013-10-01 17:16:11 +02:00
Jonas Holmberg	0cd24e2257	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 14:32:43 +02:00
Jonas Holmberg	74b8da9396	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	2013-09-26 14:25:36 +02:00
Jonas Holmberg	eb03b5c172	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 17:36:07 +02:00
Patrick Radizi	7b0ad7c25f	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	2013-09-24 17:30:18 +02:00
Jonas Holmberg	e88b71aee4	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.	2013-09-19 15:44:26 +02:00
Wim Taymans	b41422bad7	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 14:39:44 +02:00
Wim Taymans	4d6c038fc7	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 12:21:02 +02:00
Wim Taymans	fe3f63de7c	Revert "client: map URL to path in requests" This reverts commit `e3fded2cec`. 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.	2013-09-17 11:41:57 +02:00
Wim Taymans	e3fded2cec	client: map URL to path in requests	2013-09-16 17:17:35 +02:00
Wim Taymans	952aa309dc	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.	2013-09-16 17:17:34 +02:00
Ognyan Tonchev	258f63b8ac	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	2013-09-09 14:36:12 +02:00
Jonas Holmberg	23b3f21595	client: free threadpool https://bugzilla.gnome.org/show_bug.cgi?id=707638	2013-09-06 20:44:10 +01:00
Jonas Holmberg	19178a413c	auth, media, media-factory: unref permissions https://bugzilla.gnome.org/show_bug.cgi?id=707638	2013-09-06 18:57:55 +01:00
Patricia Muscalu	ff10d24130	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 12:23:06 +02:00
Wim Taymans	1287b5f772	client: don't check url in response There is no url or method in the response to check	2013-08-23 10:38:43 +02:00
Youness Alaoui	1f84618725	Add handle-response signal for when we receive a GET_PARAMETER response	2013-08-23 10:32:33 +02:00
Youness Alaoui	76cbc7c86c	Fix gst_rtsp_server_client_filter, using wrong variable type	2013-08-23 10:29:21 +02:00
Tim-Philipp Müller	3766914a18	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	2013-08-22 18:39:59 +01:00
Wim Taymans	d74cbf2911	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 17:05:24 +02:00
Wim Taymans	f094256add	media: set protocols on streams	2013-08-16 16:34:56 +02:00
Wim Taymans	1a838d6179	client: use protocols supported by stream	2013-08-16 16:20:00 +02:00
Wim Taymans	04d2da4d03	media-factory: allow all protocols	2013-08-16 16:19:27 +02:00
Wim Taymans	cf96774e6d	media: configure protocols in new streams	2013-08-16 16:10:43 +02:00
Wim Taymans	a84b71c0f0	stream: add protocols property	2013-08-16 16:08:43 +02:00
Youness Alaoui	cdbb6bcc15	rtsp-media: send state in "new-state" signal https://bugzilla.gnome.org/show_bug.cgi?id=705110	2013-08-13 16:41:53 +01:00
Wim Taymans	f124d11298	server: add method to iterate clients of server	2013-08-02 17:19:42 +02:00
Youness Alaoui	a95ab4b29e	Add vmethod for rtsp-media subclass to access rtpbin	2013-08-02 16:59:04 +02:00
Youness Alaoui	081e6d3204	small documentation fix	2013-08-02 16:58:24 +02:00
Youness Alaoui	7618800088	Do not take range header if range is invalid	2013-08-02 16:58:20 +02:00
Wim Taymans	6ac547cc34	media: add docs for new method	2013-08-02 16:57:26 +02:00
Youness Alaoui	050b16ad84	Add API to rtsp-media set the pipeline's state	2013-08-02 16:53:07 +02:00
Youness Alaoui	5e642c7ef1	Update current position/duration when gst_rtsp_media_get_range_string is called	2013-08-02 16:51:15 +02:00
Wim Taymans	f78a65379c	ClientState -> Context Rename the clientstate to context and put the code in a separate file.	2013-07-22 14:25:04 +02:00
Wim Taymans	25547176be	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 12:27:33 +02:00
Wim Taymans	1a307c707d	auth: use defines when possible	2013-07-18 12:27:33 +02:00
Wim Taymans	3dc34af5aa	address-pool: improve docs	2013-07-18 12:27:33 +02:00
Wim Taymans	472010666c	permissions: add the role to the copy	2013-07-18 12:27:33 +02:00
Olivier Crête	db74d5c559	permissions: Also copy the roles	2013-07-17 19:35:33 -04:00
Olivier Crête	91a32754e3	permissions: Make it build	2013-07-17 19:32:09 -04:00
Wim Taymans	81745b43b4	docs: small fixes	2013-07-16 12:36:56 +02:00
Wim Taymans	041b1b79a1	docs: improve docs	2013-07-16 12:32:51 +02:00
Wim Taymans	d3d7df5a1e	address-pool: cleanups Remove redundant method, improve docs.	2013-07-16 12:32:00 +02:00
Wim Taymans	0a8f5c8892	docs: improve docs	2013-07-15 17:31:35 +02:00
Wim Taymans	fbe0cefae1	permissions: implement _remove_role	2013-07-15 17:12:57 +02:00
Wim Taymans	5e297ea093	permissions: update docs	2013-07-15 17:12:43 +02:00
Wim Taymans	f18f2619e1	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 16:47:07 +02:00
Wim Taymans	7064b9fda7	thread-pool: add more docs	2013-07-15 15:25:00 +02:00
Wim Taymans	0ce4d4d5c7	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 14:50:38 +02:00
Wim Taymans	3fe1096fd1	server: add small debug	2013-07-15 14:50:26 +02:00
Wim Taymans	38d91a2bf8	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 11:57:49 +02:00
Wim Taymans	7db2f9f3cf	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 11:56:06 +02:00
Wim Taymans	692cbc1364	mount-points: add some debug	2013-07-15 11:51:34 +02:00
Wim Taymans	9fe107a96a	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 17:07:53 +02:00
Wim Taymans	c4db302559	token: add method to check boolean permission	2013-07-12 17:06:37 +02:00
Wim Taymans	b8c5aa3a6b	token: simplify token constructor Use variable arguments to make easier API.	2013-07-12 16:36:05 +02:00
Wim Taymans	67d0fbc048	media-factory: add convenience API for factory	2013-07-12 16:17:57 +02:00
Wim Taymans	facc91a942	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 16:17:15 +02:00
Wim Taymans	a6a8293595	auth: fix typo	2013-07-12 16:01:14 +02:00
Wim Taymans	5cf75e64af	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 15:19:29 +02:00
Wim Taymans	7532de687a	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 15:13:48 +02:00
Wim Taymans	9a09d98e6d	docs: fix docs	2013-07-12 14:10:13 +02:00
Wim Taymans	4b2e6d88b3	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 12:41:52 +02:00
Wim Taymans	a1e96c2269	client: add state push/pop	2013-07-12 12:38:54 +02:00
Wim Taymans	e1628a0515	client: add connection to state	2013-07-12 12:37:25 +02:00
Wim Taymans	f6674d5c10	mount-points: fix debug	2013-07-11 20:45:11 +02:00
Wim Taymans	7f8fdbc453	thread-pool: we don't require a state	2013-07-11 17:28:04 +02:00
Wim Taymans	c2d4b79b69	server: let context ref the server So that we don't risk losing the server object early anc crash.	2013-07-11 17:18:58 +02:00
Wim Taymans	0b3644a21b	docs: improve docs	2013-07-11 16:57:14 +02:00
Wim Taymans	8b4c9570fa	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 16:28:09 +02:00
Wim Taymans	d357fc55af	docs: more updates	2013-07-11 12:24:33 +02:00
Wim Taymans	ccceb1de11	docs: update docs	2013-07-11 12:18:26 +02:00
Wim Taymans	6f5a82aed3	thread-pool: fix vmethod invocation	2013-07-10 20:48:47 +02:00
Wim Taymans	8cec0f8a46	thread-pool: store thread type in thread	2013-07-10 20:48:18 +02:00
Wim Taymans	4e9c4d8bb7	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 17:09:27 +02:00
Wim Taymans	d1e4baab6c	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 17:08:14 +02:00
Wim Taymans	01b921e8a6	server: configure client thread pool	2013-07-10 17:07:13 +02:00
Wim Taymans	00997d956f	client: add method to configure thread pool	2013-07-10 17:06:36 +02:00
Wim Taymans	27917f4ef3	server: use thread pool Use the thread pool instead of doing our own thing.	2013-07-10 17:02:58 +02:00
Wim Taymans	25269c7b1a	thread-pool: add object to manage threads Add an object to manage the client and media threads.	2013-07-10 16:47:43 +02:00
Wim Taymans	1a0c7051aa	auth: debug authorization check	2013-07-10 15:28:35 +02:00
Wim Taymans	c4c9c873b8	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 20:44:51 +02:00
Wim Taymans	ca28a46600	factory: pass permissions to media by default	2013-07-09 16:38:39 +02:00
Wim Taymans	d7dec33328	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 16:04:35 +02:00
Wim Taymans	c9d6455ad3	client: add state to current thread Add the client to the ClientState object. Place the ClientState on the current thread.	2013-07-09 16:01:29 +02:00
Wim Taymans	0499a1ec7d	media: make it possible to set permissions Make it possible to set permissions on media and media factory objects	2013-07-09 14:33:43 +02:00
Wim Taymans	8f008807ad	permissions: add permissions object Add a mini object to store permissions based on a role.	2013-07-09 14:31:15 +02:00
Wim Taymans	a63f4a2a4c	auth: add auth checks Add an enum with auth checks and implement the checks in the auth object. Perform the checks from the client.	2013-07-08 16:29:01 +02:00
Wim Taymans	fb7c9b8122	auth: use the token after authentication After we authenticated a user, keep the Token around in the state.	2013-07-08 11:10:20 +02:00
Wim Taymans	12583e819c	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.	2013-07-08 11:10:20 +02:00
Wim Taymans	48ff096a25	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 20:53:19 +02:00
Wim Taymans	19cffc7999	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.	2013-07-05 20:53:19 +02:00
Wim Taymans	78bc979690	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.	2013-07-04 14:33:59 +02:00
Wim Taymans	a1e5bde58d	client: error out on non-aggregate control We require aggregate control (for now) for PLAY, PAUSE and TEARDOWN.	2013-07-03 16:15:04 +02:00
Wim Taymans	9182263532	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 15:55:38 +02:00
Wim Taymans	3999bd4e4e	media: add method to find a stream by control url	2013-07-03 15:14:39 +02:00
Wim Taymans	d4e8d800c9	stream: add method to check control url of stream	2013-07-03 15:13:45 +02:00
Wim Taymans	5a833f503e	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 12:37:48 +02:00
Wim Taymans	8f79daef5e	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 11:10:27 +02:00
Wim Taymans	df08a2dd9e	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 10:45:51 +02:00
Sebastian Dröge	a22889ac08	rtsp-server: Allow building of static library	2013-07-03 10:40:48 +02:00
Wim Taymans	714e84d891	sdp: get control string from stream Use the control string as configured in the stream.	2013-07-02 15:54:43 +02:00
Wim Taymans	2ffb0f69d2	stream: add methods and property to set control string	2013-07-02 14:50:30 +02:00
Wim Taymans	0248775c74	client: cleanups Rename variables for clarity Keep media in state when we can	2013-07-02 11:58:02 +02:00
Wim Taymans	a7fe63298c	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 16:46:39 +02:00
Wim Taymans	13016309b1	client: fix comment	2013-07-01 15:18:43 +02:00
Wim Taymans	82812988a6	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 14:47:33 +02:00
Wim Taymans	284a0a5cd1	stream: improve docs	2013-07-01 12:20:50 +02:00
Wim Taymans	5b6cbb4ede	stream-transport: remove old if 0 block	2013-07-01 12:04:45 +02:00
Wim Taymans	ffd4b1aaf1	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 17:19:11 +02:00
Wim Taymans	27a786aa4a	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 16:32:06 +02:00
Wim Taymans	13ab4905e4	media-factory: fix typo	2013-06-26 16:31:39 +02:00
Ognyan Tonchev	cd4120ef26	rtsp-media: Do not leak the query in default_query_stop Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703120	2013-06-26 15:42:01 +02:00
Wim Taymans	81c3843ad1	media: don't unlock when conversion fails Don't unlock the state lock when conversion fails because it was not locked.	2013-06-25 15:46:41 +02:00
Youness Alaoui	0b94f50eab	Add query_position and query_stop vmethods to rtsp-media	2013-06-25 15:23:36 +02:00
Youness Alaoui	842f5ad9c4	Fix typo in property install for rtsp-media's time-provider	2013-06-25 15:12:36 +02:00
Wim Taymans	55214d0d52	client: clean some variables Clean some variables and add some guards to _send_request()	2013-06-25 15:09:13 +02:00
Youness Alaoui	d2dab47085	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	2013-06-25 14:58:17 +02:00
Wim Taymans	aab1198516	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 23:56:57 +02:00
Wim Taymans	6d69a4ae80	media: fix docs	2013-06-24 23:51:38 +02:00
Aleix Conchillo Flaque	aeaadf0e5e	stream: allow access to the rtp session https://bugzilla.gnome.org/show_bug.cgi?id=703004	2013-06-24 23:42:58 +02:00
Alexander Schrab	c3f8673174	dscp qos support in gst-rtsp-stream Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645	2013-06-24 14:51:44 +02:00
Wim Taymans	fa1d3354c0	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 12:20:21 +02:00
Wim Taymans	949f11c643	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 12:16:07 +02:00
Alexander Schrab	a5490e323b	client: handle asterisk as path in requests Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266	2013-06-20 11:17:29 +02:00
Wim Taymans	23ec78faea	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.	2013-06-20 11:14:31 +02:00
David Svensson Fors	52eb796bec	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	2013-06-20 10:17:32 +02:00
Ognyan Tonchev	d9e245e62e	rtsp-media: Do not leak the element in take_pipeline Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470	2013-06-17 17:18:40 +02:00
Ognyan Tonchev	7e9df0e112	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	2013-06-17 16:18:37 +02:00
Ognyan Tonchev	b5f8ff8232	rtsp-client: Make param_set and param_get virtual Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072	2013-06-17 16:11:40 +02:00
David Svensson Fors	6151072a2e	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 16:11:34 +02:00
Wim Taymans	3dbe0e17d4	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	2013-06-14 16:06:46 +02:00
Alexander Schrab	f9f994e33d	use 0.0.0.0 or :: for c= line instead of server address	2013-06-14 15:58:52 +02:00
Alexander Schrab	275e2d52a4	use local address, not remote, in SDP See https://bugzilla.gnome.org/show_bug.cgi?id=702063	2013-06-14 15:52:14 +02:00
David Svensson Fors	7efa871c1f	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 14:29:05 +02:00
Ognyan Tonchev	c5b3066c33	rtsp-server: Expose the use_client_settings API Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935	2013-06-03 12:04:44 +02:00
Alexander Schrab	3e119be829	rtspstream: handle both ipv4 and ipv6 clients Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129	2013-06-03 11:23:40 +02:00
Wim Taymans	17b07d1c0e	Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute" This reverts commit `5fd034ff1a`. 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 15:43:11 +02:00
Wim Taymans	2a0aaa1019	Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute" This reverts commit `d6a4dee036`. 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 15:43:11 +02:00
Wim Taymans	cfdf2e6db5	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 15:43:10 +02:00
Wim Taymans	0a285290cb	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 11:42:36 +02:00
Wim Taymans	531ffca018	server: remove accept_client vmethod This vmethod is not very useful so remove it.	2013-05-31 11:14:17 +02:00
Wim Taymans	0091339254	server: don't crash on NULL GError	2013-05-30 17:23:51 +02:00
Patricia Muscalu	aa50aac669	rtsp-session-pool: corrected session timeout detection Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253	2013-05-30 13:13:05 +02:00
Wim Taymans	7526178a09	client: improve debug	2013-05-30 10:52:46 +02:00
Wim Taymans	d638b03ff9	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 07:18:22 +02:00
Wim Taymans	7b880231b1	stream: keep the transport object alive Keep the transport object alive while we have it as qdata on the source.	2013-05-30 06:49:20 +02:00
Alexander Schrab	c75e1c6b47	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	2013-05-27 13:20:36 +02:00
Alexander Schrab	e047c9fec1	rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6 https://bugzilla.gnome.org/show_bug.cgi?id=700953	2013-05-27 11:15:50 +02:00
Sebastian Rasmussen	d6a4dee036	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	2013-05-23 21:02:58 +02:00
Sebastian Rasmussen	5fd034ff1a	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	2013-05-23 21:02:50 +02:00
Patricia Muscalu	0951aa37e1	rtsp-sdp: add bandwidth line https://bugzilla.gnome.org/show_bug.cgi?id=699220	2013-05-15 12:36:32 +02:00
Wim Taymans	573b10bc83	media: release lock when removing fakesink	2013-04-23 10:28:35 +02:00
Wim Taymans	0ddd98bfa6	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.	2013-04-23 10:28:34 +02:00
Wim Taymans	b80b8824be	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 17:34:37 +02:00
Ognyan Tonchev	00291e5285	stream: add method to get the srcpad	2013-04-22 17:32:31 +02:00
Ognyan Tonchev	a26b06cc69	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 16:40:48 +02:00
Ognyan Tonchev	9b31fcc7f8	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 16:25:17 +02:00
Ognyan Tonchev	0bdff0161c	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 16:19:35 +02:00
Ognyan Tonchev	6081f91351	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	2013-04-22 16:09:22 +02:00
David Svensson Fors	bba7c4042d	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 12:21:54 +02:00
David Svensson Fors	825d6f0b51	client: expose connection Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546	2013-04-15 12:17:34 +02:00
Wim Taymans	a64cb68164	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.	2013-04-12 11:34:38 +01:00
Wim Taymans	36ff679558	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 22:38:44 +02:00
Wim Taymans	95bf53513f	media: wait for buffering to complete Wait for buffering to complete before changing the state to the target state.	2013-04-09 20:39:58 +02:00
Wim Taymans	ec0718d7c9	media: small cleanup	2013-04-09 20:11:35 +02:00
Olivier Crête	91210f40f2	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	2013-03-22 18:59:50 -04:00
Olivier Crête	c18eafbb24	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.	2013-03-22 15:53:06 +01:00
Patricia Muscalu	8a08fddb41	rtsp-client: expose uri	2013-03-18 23:44:38 +00:00
Olivier Crête	5a39e25949	stream: Select unicast address from pool if appropriate	2013-03-11 11:07:20 +01:00
Olivier Crête	a797cbde06	stream: Properties are always there in Gst 1.0	2013-03-11 11:07:20 +01:00
Olivier Crête	27a057962c	address-pool: Verify that multicast addresses are used for multicast and vice-versa	2013-03-11 11:07:20 +01:00
Olivier Crête	d06e68abd1	address-pool: Add unicast addresses	2013-03-11 11:07:20 +01:00
Olivier Crête	4c61c6d308	rtsp-server: Limit the number of threads per server instance If we exceed the maximum, just round robin the clients over the existing threads.	2013-03-11 11:07:20 +01:00
Olivier Crête	4071e1b999	rtsp-server: No need to store the GMainContext in the client context	2013-03-11 11:07:20 +01:00
Olivier Crête	b9d111372e	Document locking and its order	2013-03-11 11:07:19 +01:00
Olivier Crête	f0ab7ce1bf	docs: Generate docs for GstRTSPAddressPool	2013-03-11 11:07:19 +01:00
Olivier Crête	773c48e22f	client: Check client provided addresses against the address pool	2013-03-11 11:07:19 +01:00
Olivier Crête	cda75709bb	address-pool: Add API to request a specific address from the pool Also add relevant unit tests.	2013-03-11 11:07:19 +01:00
Olivier Crête	456f4367e3	address-pool: Fix off by one error When splitting a port range, the port after a skip is not part of range.	2013-03-11 11:07:19 +01:00
Wim Taymans	6db0dbc76c	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	2013-01-28 11:11:46 +01:00
Wim Taymans	4100b20b0a	rtsp-client: set the client backlog Set the client backlog to a reasonable default	2012-12-14 11:58:29 +01:00
Ognyan Tonchev	0844e8afbc	rtsp-media: Make the element a constructor parameter https://bugzilla.gnome.org/show_bug.cgi?id=689594	2012-12-10 10:25:57 +01:00
Wim Taymans	6beabf1ed4	media: match prepare with unprepare Really unprepare when there were an equal amount of prepare calls.	2012-11-30 15:03:15 +01:00
Wim Taymans	ca26588c7e	media: media has to be unprepared in finalize Because unprepare takes away the last ref on the media.	2012-11-30 14:58:46 +01:00
Wim Taymans	38addd7822	Revert "client: never call gst_rtsp_media_unprepare, let gst_rtsp_media_finalize do it" This reverts commit `ba5b78ff2f`. 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 14:36:30 +01:00
Wim Taymans	119674a828	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 13:35:05 +01:00
Wim Taymans	339ea9b085	media: improve debug	2012-11-30 12:54:10 +01:00
Wim Taymans	241baba20a	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 12:53:02 +01:00
Wim Taymans	edf2ef4f0b	media: use g_object_ref/unref for GObjects	2012-11-30 10:05:48 +01:00
Alessandro Decina	ba5b78ff2f	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 07:06:17 +01:00
Alessandro Decina	00d9a94e1a	Fix compiler warning	2012-11-30 06:17:46 +01:00
Alessandro Decina	e2a7690cb3	Add missing g_type_class_add_private in GstRTSPMediaFactoryURI	2012-11-30 06:14:49 +01:00
Wim Taymans	1abc9be682	small cleanup	2012-11-29 17:21:12 +01:00
Wim Taymans	28fd887547	media: avoid element leak	2012-11-29 17:20:56 +01:00
Wim Taymans	4eb010824e	media: require an element in media constructor	2012-11-29 17:20:26 +01:00
Wim Taymans	865c9a6b30	Revert "client: TEARDOWN brings that state to Init again" This reverts commit `4b61fdad85`. The object is already disposed, there is no point in setting the state.	2012-11-29 17:07:30 +01:00
Wim Taymans	4b61fdad85	client: TEARDOWN brings that state to Init again	2012-11-29 12:30:20 +01:00
Wim Taymans	ad00c5e792	rtsp: make object details private Make all object details private Add methods to access private bits	2012-11-29 11:11:05 +01:00
Wim Taymans	e11287eb7c	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 14:45:30 +01:00
Wim Taymans	d3d74ab77b	stream: improve debug	2012-11-28 12:40:18 +01:00
Wim Taymans	fe71114a7d	media: unref pipeline in finalize to avoid leaking it	2012-11-28 12:39:37 +01:00
Wim Taymans	d43a31055e	rtsp: use gst_object_unref on GstObjects	2012-11-28 12:10:47 +01:00
Wim Taymans	6b36241816	media-factory: require an url	2012-11-28 12:10:14 +01:00
Wim Taymans	20f09bf3e7	server: remove unused include	2012-11-28 11:17:27 +01:00
Wim Taymans	e5ba372808	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 11:05:08 +01:00
Wim Taymans	b4c168c71d	mounts: add g_return_if guards	2012-11-28 10:40:14 +01:00
Wim Taymans	b3fe3357ab	client: improve debug	2012-11-27 12:33:02 +01:00
Wim Taymans	d5389c940d	client: improve debug and fix leaks Cleanup the uri and session when there is a bad request.	2012-11-27 12:24:21 +01:00
Wim Taymans	a26e9b621e	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 12:11:41 +01:00
Wim Taymans	4782d08bdc	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 11:17:45 +01:00
Wim Taymans	18bb9ffa6b	client: small cleanup	2012-11-26 17:35:51 +01:00
Wim Taymans	fc0f176a17	client: remove unused include	2012-11-26 17:34:35 +01:00
Wim Taymans	9f8e8bc02d	client: fix compilation	2012-11-26 17:34:24 +01:00
Wim Taymans	eb88fa9e76	client: call destroy without the lock	2012-11-26 17:28:29 +01:00
Wim Taymans	33da3af265	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 17:20:39 +01:00
Wim Taymans	26a4b98ab0	client: small cleanup	2012-11-26 16:45:04 +01:00
Wim Taymans	8da4171055	client: remove reference to server We don't need to keep a ref to the server	2012-11-26 16:39:26 +01:00
Wim Taymans	4fa7502fd9	client: add locking Also add some g_return_if()	2012-11-26 16:31:43 +01:00
Wim Taymans	b21b46ec4d	client: log more errors	2012-11-26 13:37:20 +01:00
Wim Taymans	f460e7360e	client: fix compilation	2012-11-26 13:36:19 +01:00
Wim Taymans	84e72262d0	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 13:19:06 +01:00
Wim Taymans	1d53c46d23	MediaMapping -> MountPoints Describes better what the object manages.	2012-11-26 12:37:55 +01:00
Wim Taymans	0f93879b2c	media: fix seeking	2012-11-21 17:21:28 +01:00
Wim Taymans	5eb5fd45f3	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 16:41:56 +01:00
Wim Taymans	37a7ec8033	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 12:29:55 +01:00
Wim Taymans	8fcdca987d	factory-uri: add some debug	2012-11-20 12:29:26 +01:00
Wim Taymans	1826844ee4	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 12:24:13 +01:00
Wim Taymans	8211cdfdc2	factory-uri: take ref to factory Take a ref to the factory that we place in our list.	2012-11-20 12:10:16 +01:00
David Svensson Fors	0eeb4a5c73	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 11:30:37 +01:00
Wim Taymans	8a7197f078	server: fix small leak	2012-11-20 11:24:35 +01:00
Wim Taymans	989f004e24	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	2012-11-20 09:46:40 +01:00
David Svensson Fors	01973c924d	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	2012-11-20 09:46:00 +01:00
Alessandro Decina	65042a9551	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	2012-11-20 09:32:19 +01:00
David Svensson Fors	0996266342	rtsp-stream: plug socket leak Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703	2012-11-20 09:26:28 +01:00
Tim-Philipp Müller	0006ca6d60	rtsp-server: don't use deprecated API	2012-11-17 00:11:27 +00:00
Tim-Philipp Müller	290968eb8c	rtsp-client: fix unused-but-set-variable compiler warning rtsp-client.c:1260:21: error: variable 'protocols' set but not used	2012-11-17 00:03:42 +00:00
Wim Taymans	26ff5fc073	rtsp: cleanups	2012-11-15 17:11:16 +01:00
Wim Taymans	e4ea72ccdf	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 16:18:29 +01:00
Wim Taymans	ba21661ce4	rtsp: improve debug	2012-11-15 16:15:20 +01:00
Wim Taymans	c34f5d1c1a	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 15:41:42 +01:00
Wim Taymans	4168a67992	media: configure address pool in new streams	2012-11-15 15:41:19 +01:00
Wim Taymans	44a2855eb3	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 15:36:21 +01:00
Wim Taymans	1b4ac6e5b0	media: remove MTU property It is a stream property	2012-11-15 15:32:43 +01:00
Wim Taymans	2160d6dbd3	client: set blocksize only on stream Set the blocksize only on the current stream.	2012-11-15 15:29:35 +01:00
Wim Taymans	6c2947e68b	stream: share src and sink sockets the allocated socket is in the used-socket property, not socket.	2012-11-15 13:52:07 +01:00
Wim Taymans	45b6693b39	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 13:25:14 +01:00
Wim Taymans	f15ffb521c	rtsp: use AddressPool Remove the multicast_group property. Use the configured addresspool to allocate multicast addresses.	2012-11-14 17:23:59 +01:00
Wim Taymans	d0ffc8e679	address-pool: add clear method	2012-11-14 16:20:36 +01:00
Wim Taymans	6085b1fcc1	address-pool: small cleanups	2012-11-14 16:10:45 +01:00
Wim Taymans	b30202b174	address-pool: add object to manage multicast addresses Make an object that can manage a rage of multicast addresses and ports.	2012-11-14 15:49:06 +01:00
Wim Taymans	7d6e4606fa	server: set default max-threads property	2012-11-13 12:05:42 +01:00
Wim Taymans	dfe3efef74	media: wait for concurrent _prepare If a prepare is busy, wait for the result.	2012-11-13 11:54:17 +01:00
Wim Taymans	47127bd270	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 11:49:08 +01:00
Wim Taymans	9a97de88ea	media: add lock to protect state changes	2012-11-13 11:15:35 +01:00
Wim Taymans	4753588b09	stream: add locking	2012-11-13 11:14:49 +01:00
Wim Taymans	c7d20e5603	stream-transport: add keep-alive method	2012-11-12 17:11:18 +01:00
Wim Taymans	75473fc88d	stream-transport: add method to handle RTP/RTCP Call new methods instead of poking into the structures directly.	2012-11-12 17:06:42 +01:00
Wim Taymans	883cf794e4	session-media: add locking	2012-11-12 16:51:03 +01:00
Wim Taymans	11cf3f3ccb	session: add locking	2012-11-12 16:42:37 +01:00
Wim Taymans	65fa516677	server: free old socket	2012-11-12 16:30:16 +01:00
Wim Taymans	04881bd632	mapping: add locking	2012-11-12 16:18:57 +01:00
Wim Taymans	b8cba7719c	media-factory: add locking	2012-11-12 16:14:19 +01:00
Wim Taymans	e61c84c9bb	auth: add locking	2012-11-12 16:03:21 +01:00
Wim Taymans	06cadebe71	server: add max-thread property	2012-11-12 15:53:28 +01:00
Wim Taymans	8523c9ca92	server: use a threadpool for the mainloops	2012-11-12 15:29:39 +01:00
Wim Taymans	c431592976	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 15:01:13 +01:00
Wim Taymans	a58d404e1f	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 15:01:09 +01:00
Wim Taymans	5b4340067a	session: move session header code in session object	2012-11-12 12:40:34 +01:00
Tim-Philipp Müller	4dba434f16	Fix FSF address	2012-11-04 00:14:25 +00:00
Sebastian Pölsterl	75598337a9	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 15:39:04 +00:00
Wim Taymans	543aa383e7	rtsp: only create transport when needed Only create the StreamTransport when configured.	2012-10-28 00:23:57 +02:00
Wim Taymans	66a29c7ed9	client: small cleanup	2012-10-27 23:53:35 +02:00
Wim Taymans	fb117a4f75	rtsp: refactor configuration of transport Move the configuration of the transport to a place where it makes more sense.	2012-10-27 23:49:24 +02:00
Wim Taymans	8c30d050fa	client: refactor transport parsing	2012-10-27 21:26:55 +02:00
Wim Taymans	fee8216513	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 21:05:03 +02:00
Wim Taymans	0bb0e3733c	small fixes to docs and debug	2012-10-27 11:53:51 +02:00
Wim Taymans	6a838fd5c8	stream: transports must already have been removed	2012-10-26 17:29:30 +02:00
Wim Taymans	6f7d755894	stream: improve join and leave of the pipeline simplify code Do the cleanup properly Add some docs	2012-10-26 17:28:10 +02:00
Wim Taymans	693dd3cfc4	media: move unprepare below default implementation Makes it easier to find the default implementation	2012-10-26 15:23:16 +02:00
Wim Taymans	0d55e1f50c	media: signal unprepared when we actually finish	2012-10-26 15:21:50 +02:00
Wim Taymans	84b7cf1590	media: no need to unlock, unprepare does that when needed	2012-10-26 15:19:23 +02:00
Wim Taymans	348b7f9c21	docs: update docs	2012-10-26 12:35:20 +02:00
Wim Taymans	6b7ff45ca6	rtsp: fix MTU setting Fix setting of the MTU. There is no need for a vmethod.	2012-10-26 12:35:20 +02:00
Wim Taymans	de7c72dec2	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.	2012-10-25 21:29:58 +02:00
Sebastian Rasmussen	0de6262dc4	rtsp-client: Unref server address clients connected to Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725	2012-10-23 23:05:45 +01:00
Ognyan Tonchev	78bde6fa3e	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 18:11:28 +01:00
Sebastian Pölsterl	5cec59737b	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	2012-10-18 19:31:23 +01:00
Sebastian Pölsterl	e11e855ac8	rtsp-server: fixed comments and GIR annotations https://bugzilla.gnome.org/show_bug.cgi?id=680777	2012-10-18 19:17:01 +01:00
Alessandro Decina	bc474a5b26	media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory	2012-10-15 10:50:27 +02:00
Alessandro Decina	1e954a1a5e	rtsp-server: allow binding on port 0 (binds on a random port)	2012-10-15 10:50:27 +02:00
Alessandro Decina	6a49744088	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.	2012-10-15 10:50:27 +02:00
Alessandro Decina	8f507e4512	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.	2012-10-15 10:50:26 +02:00
Alessandro Decina	3a49b8e783	rtsp-media-factory-uri: don't autoplug parsers in a loop Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging h264parse forever.	2012-10-15 10:50:26 +02:00
Alessandro Decina	8da18a85ef	Explicitly link against gio. Fix link error on mac.	2012-10-15 10:50:26 +02:00
Ognyan Tonchev	4f0ef292f0	session: add ttl to the transport header in SETUP See https://bugzilla.gnome.org/show_bug.cgi?id=685561	2012-10-10 11:13:58 +02:00
Ognyan Tonchev	d581b7bd4e	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	2012-10-10 11:07:59 +02:00
Patricia Muscalu	870b8db279	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	2012-10-05 11:44:32 +02:00
Ognyan Tonchev	f4a0a371b7	media: fix check for seekability	2012-09-10 16:29:35 +02:00
Wim Taymans	3e55e0e467	client: use more GIO Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593	2012-09-07 17:14:30 +02:00
Wim Taymans	87c73c06fb	server: remove obsolete includes	2012-09-07 17:14:10 +02:00
Aleix Conchillo Flaque	c6cce4a6b9	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	2012-09-07 16:45:17 +02:00
Aleix Conchillo Flaque	bef57648b8	rtsp-client: add signals for rtsp requests (fixes #683287 )	2012-09-07 16:41:29 +02:00
Aleix Conchillo Flaque	ebc4ce4de1	add new-session signal to rtsp-client (fixes #683058 )	2012-08-30 22:00:30 +02:00
Patricia Muscalu	50e4c7e8c4	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	2012-08-20 11:49:27 +02:00
Patricia Muscalu	228e2ccc2d	rtsp-client: make create_sdp virtual method Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173	2012-07-24 12:52:53 +02:00
Wim Taymans	f305020636	client: fix docs	2012-07-10 11:39:58 +02:00
Ognyan Tonchev	ed66f974dd	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.	2012-07-10 11:38:05 +02:00
Ognyan Tonchev	86e53af34a	rtsp: Handle the blocksize parameter Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325	2012-07-10 11:26:01 +02:00
Tim-Philipp Müller	217a46e4c1	rtsp-media: update for gst_element_make_from_uri() changes	2012-06-23 15:06:11 +01:00
David Svensson Fors	36df0dd8be	rtsp-media: don't collect media stats when going to NULL Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015	2012-06-14 10:14:06 +02:00
Wim Taymans	853128e1c7	client: don't leak transports	2012-06-14 10:14:06 +02:00
David Svensson Fors	3f49c2d8f4	rtsp-client: free transport on no_stream in SETUP handler	2012-06-14 10:14:06 +02:00
David Svensson Fors	8f5d82be6d	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.	2012-06-14 10:14:06 +02:00
David Svensson Fors	dc796bf075	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.	2012-06-14 10:14:06 +02:00
David Svensson Fors	aa158fa738	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.	2012-06-14 10:14:06 +02:00
David Svensson Fors	7b145aeeab	client: fix GSocketAddress leak in gst_rtsp_client_accept Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677463	2012-06-06 14:49:40 +02:00
David Svensson Fors	ffa3166fbd	rtsp: fix compiler warnings Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676500	2012-05-22 15:37:25 +02:00
Wim Taymans	6cc2fb9bfc	rtsp-server: port to new thread API	2012-05-11 09:42:47 +02:00
Sebastian Dröge	e2f10f5ba5	rtsp-server: Fix compilation and compiler warnings	2012-04-13 15:27:22 +02:00
Sebastian Dröge	7df1696713	configure: Modernize autotools setup a bit Also we now only create tar.bz2 and tar.xz tarballs.	2012-04-13 14:02:15 +02:00
Sebastian Dröge	fb0718a036	rtsp-server: Update versioning	2012-04-04 14:48:44 +02:00
Sebastian Dröge	e9ef6f6254	Merge remote-tracking branch 'origin/0.10' Conflicts: gst/rtsp-server/rtsp-session-pool.c	2012-03-29 15:12:21 +02:00
Sebastian Dröge	1f442d45b6	rtsp-server: Don't use deprecated GLib API	2012-03-27 10:13:20 +02:00
Wim Taymans	e0be150e91	media: fix state of the appqueue	2012-03-13 18:10:53 +01:00
Wim Taymans	6403227471	factory: use videoconvert	2012-03-13 16:07:16 +01:00
Wim Taymans	377f6d9156	factory: change to new style caps	2012-03-13 16:02:47 +01:00
Wim Taymans	4c59e211e2	rtsp-server: port to GIO Port to GIO	2012-03-07 15:04:29 +01:00
Tim-Philipp Müller	e67a1c664c	rtsp-client: update for new map API	2012-02-13 11:06:33 +00:00
Wim Taymans	fde25cd9c3	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.	2011-12-09 10:53:30 +01:00
Wim Taymans	bace3995d5	Merge branch 'master' into 0.11	2011-11-03 12:58:42 +01:00
Wim Taymans	a701e8595e	media: add a seekable boolean Maintain the seekable state with a new variable instead of reusing the is_live variable.	2011-11-03 12:55:24 +01:00
Victor Gottardi	526bbb5a8f	Disallow seek in live media	2011-11-03 12:45:18 +01:00
Wim Taymans	05c3928b11	Merge branch 'master' into 0.11	2011-11-03 11:58:42 +01:00
mat	20b6be3852	#ifdef statements for windows socket creation were missing	2011-11-03 11:56:51 +01:00
Wim Taymans	6759a4b9b0	client: use method to access property	2011-08-16 16:39:11 +02:00
Wim Taymans	4c8f3696d0	media-factory: add protocols property Add a property to configure the allowed protocols in the media created from the factory.	2011-08-16 16:39:07 +02:00
Wim Taymans	85e2013ca4	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 16:39:04 +02:00
Wim Taymans	6fa73b2552	client: use method to access property	2011-08-16 16:07:04 +02:00
Wim Taymans	0e9ce1caf3	media-factory: add protocols property Add a property to configure the allowed protocols in the media created from the factory.	2011-08-16 15:15:19 +02:00
Wim Taymans	8684fc5c69	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 15:03:06 +02:00
Wim Taymans	56a16f9f5a	client: use media multicast group	2011-08-16 14:50:21 +02:00
Wim Taymans	2c9701bd73	retab some .h	2011-08-16 14:50:18 +02:00
Robert Krakora	a5e028ba72	sdp: copy and free the server ip address Copy and free the server ip address to make memory management easier later.	2011-08-16 14:50:15 +02:00
Wim Taymans	647e8c7af8	media-factory: configure multicast in media	2011-08-16 14:50:12 +02:00
Wim Taymans	c079325169	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 14:50:05 +02:00
Wim Taymans	514728864a	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 14:49:59 +02:00
Wim Taymans	b881dc6669	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 14:49:55 +02:00
Wim Taymans	9573058f54	client: use media multicast group	2011-08-16 13:43:44 +02:00
Wim Taymans	26c8898e79	retab some .h	2011-08-16 13:37:50 +02:00
Robert Krakora	ae67971cde	sdp: copy and free the server ip address Copy and free the server ip address to make memory management easier later.	2011-08-16 13:31:52 +02:00
Wim Taymans	ccfb99f852	media-factory: configure multicast in media	2011-08-16 13:27:39 +02:00
Wim Taymans	5b53335873	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 13:25:16 +02:00
Wim Taymans	1f8b97d940	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 13:13:36 +02:00
Wim Taymans	b0e22d6861	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 12:51:44 +02:00
Wim Taymans	8749b1e08f	Merge branch 'master' into 0.11	2011-08-16 12:11:59 +02:00
Robert Krakora	f7223cfdab	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 12:09:48 +02:00
Wim Taymans	1aefff4959	Merge branch 'master' into 0.11	2011-08-16 11:53:37 +02:00
Emmanuel Pacaud	5dc9e76125	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 11:22:55 +02:00
Wim Taymans	b5aa7628bf	Merge branch 'master' into 0.11	2011-08-16 11:12:33 +02:00
David Schleef	041b62db8b	rtsp-server: hold on to reference while using object	2011-08-11 18:07:08 -07:00
Wim Taymans	bbab01747d	media: use new api	2011-08-04 08:59:17 +02:00
David Schleef	aa128813fe	client: fix reference counting	2011-07-27 15:02:08 -07:00
Thijs Vermeir	93fb73b46f	fix compiler warnings about unused variables	2011-07-20 17:16:42 +02:00
Wim Taymans	bd8eb8f3d9	client: update for buffer API change	2011-06-13 19:05:57 +02:00
Edward Hervey	b93f046708	Makefile.am: 0.10 => @GST_MAJORMINOR@	2011-06-07 11:04:10 +02:00
Edward Hervey	597a99e9b9	rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer	2011-06-07 10:59:16 +02:00
Edward Hervey	14f8ed65b4	.gitignore: 0.10 => 0.11	2011-06-07 10:59:03 +02:00
Edward Hervey	c94416d486	Makefile.am: 0.10 => @GST_MAJORMINOR@	2011-06-07 10:54:26 +02:00
Wim Taymans	80e0b0b19a	media: port to new caps API	2011-05-17 09:48:13 +02:00
Wim Taymans	debbea1008	Merge branch 'master' into 0.11	2011-05-17 09:45:04 +02:00
Fabian Deutsch	6ef7c966ae	Add a signal for newly connected clients. Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>	2011-05-17 09:44:14 +02:00
Wim Taymans	914b481e42	rtsp-server: port to 0.11	2011-04-26 19:22:50 +02:00
Wim Taymans	6959ebd8e8	Merge branch 'master' into 0.11 Conflicts: common configure.ac	2011-04-26 19:07:13 +02:00
Miguel Angel Cabrera Moya	17ce0df09a	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 18:39:43 +00:00
Sebastian Dröge	63744dfece	rtsp-server: Don't install the funnel header	2011-03-07 10:23:06 +01:00
Wim Taymans	a924e90c79	media: remove more unused code	2011-02-02 15:37:03 +01:00
Wim Taymans	ec2201a3a8	media: remove duplicate filtering Remove the duplicate filtering code now that we have a released -good version. Give a warning instead.	2011-02-02 15:30:45 +01:00
Wim Taymans	8477fdbf43	media: fix default buffer size	2011-01-31 17:38:47 +01:00
Wim Taymans	e86b7c4b15	media-factory: add property to configure the buffer-size Add a property to configure the kernel UDP buffer size.	2011-01-31 17:37:45 +01:00
Wim Taymans	88b4c02dff	media: add property to configure kernel buffer sizes Add a property to configure the kernel UDP buffer size.	2011-01-31 17:37:41 +01:00
Wim Taymans	325b2cf8a2	rtsp-server: clarify docs a little	2011-01-19 15:29:55 +01:00
Wim Taymans	44b418b346	media: init debug category before starting thread	2011-01-13 18:57:15 +01:00
Wim Taymans	cd8382674d	auth: add realm to make it more spec compliant	2011-01-13 18:40:48 +01:00
Wim Taymans	b076933f5e	server: add locking	2011-01-12 18:57:41 +01:00
Wim Taymans	94c9999715	server: ensure the watch has a ref to the server	2011-01-12 18:26:57 +01:00
Wim Taymans	3315031bf6	server: simpify channel function	2011-01-12 18:24:44 +01:00
Wim Taymans	ba4d65a673	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 18:18:13 +01:00
Wim Taymans	9e97faf2db	server: improve debugging in various objects	2011-01-12 18:14:48 +01:00
Wim Taymans	0ef53a2d4f	server: chain up to the parent finalize	2011-01-12 16:38:34 +01:00
Wim Taymans	df0e2c2859	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 15:37:39 +01:00
Wim Taymans	318b3a1df4	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 15:36:22 +01:00
Wim Taymans	4a4a15077b	client: emit signal when closing	2011-01-12 15:35:51 +01:00
Wim Taymans	7797023fda	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 13:57:09 +01:00
Wim Taymans	5773df1d52	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 13:16:08 +01:00
Wim Taymans	9ea0346d97	media-factory: add methods to configure authorisation	2011-01-12 12:07:40 +01:00
Wim Taymans	748d044b62	client: unref auth in finalize	2011-01-12 12:07:20 +01:00
Wim Taymans	6915572695	server: unref auth in finalize	2011-01-12 12:07:04 +01:00
Wim Taymans	6d6ba1ee61	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 10:57:08 +01:00
Wim Taymans	8ccebd90b4	client: add support for setting the server. Add support for keeping a ref to the server that started this client connection.	2011-01-12 10:42:52 +01:00
Wim Taymans	9f52f281ba	auth: fix memleak and add some docs Fix a memleak of the basic auth token. Add docs for the helper function	2011-01-12 10:41:42 +01:00
Wim Taymans	c59d9e2970	client: delegate setup of auth to the manager Delegate the configuration of the authentication tokens to the manager object when configured.	2011-01-12 00:35:28 +01:00
Wim Taymans	5fb5f75020	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 00:22:27 +01:00
Wim Taymans	61bee9985a	server: move includes back the includes are needed for sockaddr_in.	2011-01-12 00:20:36 +01:00
Wim Taymans	da35feb1aa	rtsp: move network includes where they are needed	2011-01-11 22:42:25 +01:00
Sreerenj Balachandran	28597c913d	rtsp-media.h: Minor corrections in comments. Fixes #638944	2011-01-11 21:32:45 +01:00
Edward Hervey	2cc9eee3e6	gitignore: updates	2011-01-11 13:04:31 +01:00
Wim Taymans	e1787e0776	funnel: rename fsfunnel to rtspfunnel Rename the funnel to avoid conflicts with the farsight one.	2011-01-10 15:10:53 +01:00
Wim Taymans	7b3cbfde1b	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.	2011-01-10 13:43:10 +01:00
Tim-Philipp Müller	c19eb8fb4e	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 02:00:12 +00:00
Tim-Philipp Müller	8b1ec41d08	gobject-introspection: fix g-i build for uninstalled setup Requires gst-plugins-base git (> 0.10.31.2).	2011-01-08 01:15:35 +00:00
Wim Taymans	186089ff1e	factory-uri: use right property type	2011-01-07 11:24:39 +01:00
Wim Taymans	257bac1bab	factory-uri: attempt to configure buffer-lists Attempt to configure buffer lists in the payloader for improved performance.	2011-01-05 12:07:42 +01:00
Wim Taymans	790c067919	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 12:06:23 +01:00
Jonas Larsson	b5a1719e89	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	2011-01-05 11:26:30 +01:00
Wim Taymans	160fc25867	docs: improve docs	2010-12-30 12:41:31 +01:00
Wim Taymans	50b4c8de98	rtsp-server: add support for buffer lists Add support for sending bufferlists received from appsink. Fixes #635832	2010-12-29 16:26:41 +01:00
Wim Taymans	4234d96314	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 18:35:01 +01:00
Wim Taymans	915cd708ea	media: add signal to notify of state changes	2010-12-28 18:34:10 +01:00
Wim Taymans	43f4696f78	client: cleanup headers	2010-12-28 18:31:26 +01:00
Wim Taymans	899f624845	client: fix typo	2010-12-28 12:18:41 +01:00
Wim Taymans	50a71b9d86	factory-uri: add support for gstpay Add an option to prefer gstpay over decoder + raw payloader.	2010-12-23 18:53:01 +01:00
Wim Taymans	9ce4ea165b	factory-uri: rework the autoplugger. Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers before payloaders.	2010-12-23 15:58:14 +01:00
Wim Taymans	1053860112	factory-uri: use better factory filter Make better payloader filter based on autoplug rank and RTP use case.	2010-12-21 17:39:09 +01:00
Wim Taymans	ad2e0edee5	server: set SO_REUSEADDR before bind Set the SO_REUSEADDR _before_ bind() to make it actually work.	2010-12-18 11:24:48 +01:00
Wim Taymans	1ea450179e	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.	2010-12-17 18:45:10 +01:00
Wim Taymans	fc12ade012	Merge branch 'master' into 0.11 Conflicts: common configure.ac	2010-12-13 11:43:13 +01:00
Wim Taymans	ca76a73ca0	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 15:48:47 +01:00
Wim Taymans	d99a448f79	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.	2010-12-12 04:06:41 +01:00
Wim Taymans	7ef0bf98da	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 18:04:34 +01:00
Wim Taymans	34f0973831	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 18:04:34 +01:00
Wim Taymans	75a7cda97d	media-factory: make lock macro	2010-12-11 18:04:29 +01:00
Edward Hervey	a6556551e3	rtsp-server: Remove unused variable and dead assignment	2010-12-11 10:53:28 +01:00
Edward Hervey	eb83fc6318	rtsp-server: Run gst-indent Since it wasn't using the upstream common previously, there was no indentation check before commiting.	2010-12-11 10:48:42 +01:00
Edward Hervey	b95165fcff	rtsp-server: Some more doc fixups	2010-12-11 10:48:25 +01:00
Edward Hervey	bdd477f2bf	Makefile.am: Use standard GIR make behaviour	2010-12-07 18:14:39 +01:00
Wim Taymans	422fea478c	media: warn and fail when gstrtpbin is not found	2010-12-06 19:29:53 +01:00
Sebastian Pölsterl	347e10e1f9	Added initial gobject-introspection support	2010-09-23 13:39:42 +02:00
Wim Taymans	c310f0032c	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.	2010-09-23 11:35:40 +02:00
Wim Taymans	ed473f6f26	media: help the compiler a little	2010-09-22 16:15:56 +02:00
Wim Taymans	450b68252f	media: cleanup media transport before freeing Cleanup the media transport data before freeing. In particular, remove the qdata from the rtpsource object.	2010-08-24 16:47:30 +02:00
Wim Taymans	dc33070da3	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 18:17:08 +02:00
Wim Taymans	a900866570	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 15:58:39 +02:00
Wim Taymans	c89d17ca26	media: don't leak destinations Refactor and cleanup the destinations array when the stream is destroyed.	2010-08-20 13:19:56 +02:00
Wim Taymans	7c0f8a77ec	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 13:09:12 +02:00
Wim Taymans	af732fa749	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.	2010-08-20 10:18:34 +02:00
Wim Taymans	2607ff079d	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	2010-08-19 18:52:47 +02:00
Robert Krakora	8f6fd32065	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 12:45:24 +02:00
David Schleef	6a880e53df	Add stdlib.h for atoi()	2010-08-09 12:56:23 -07:00
Wim Taymans	336ffc0941	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 17:08:40 +02:00
Wim Taymans	4fdd2bf4d1	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 17:07:27 +02:00
Wim Taymans	48a54054e7	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 15:45:56 +02:00
Wim Taymans	558c7fddd2	session: small cleanups	2010-04-06 15:44:45 +02:00
Wim Taymans	30c31a65eb	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	2010-04-06 11:13:51 +02:00
Wim Taymans	09b97dd4ac	rtsp-server: add more support for multicast	2010-03-19 18:03:40 +01:00
Wim Taymans	ac8343ea62	media: allow configuration of allowed lower transport	2010-03-19 15:15:29 +01:00
Wim Taymans	e866345f15	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 18:37:18 +01:00
Wim Taymans	4eccdd9dd7	session: indent	2010-03-16 18:34:43 +01:00
Wim Taymans	d749f1e7d5	client: use right size for malloc	2010-03-16 18:33:23 +01:00
Wim Taymans	0509aa1cbf	server: comment ipv6 server listening address	2010-03-10 11:45:30 +01:00
Wim Taymans	6afa5be799	media: allow for ipv6 sockets	2010-03-10 11:45:06 +01:00
Wim Taymans	17bb89f1fc	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 13:49:00 +01:00
Wim Taymans	1b0dc41534	media: update comments a little	2010-03-09 13:44:20 +01:00
Wim Taymans	b3814d4646	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 13:43:29 +01:00
Wim Taymans	171e89c63a	client: guard against invalid paths	2010-03-09 13:42:50 +01:00
Alessandro Decina	5f535ecf87	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.	2010-03-09 10:27:38 +01:00
Wim Taymans	2997806d43	media: collect media position when seek completes	2010-03-05 19:08:08 +01:00
Luca Ognibene	e19c382bbb	client: call unlink_streams in client finalize Fixes #599027	2010-03-05 18:37:17 +01:00
Wim Taymans	83ed258684	media: limit the time to wait to something huge Avoid waiting forever but limit the timeout to 20 seconds.	2010-03-05 18:23:18 +01:00
Wim Taymans	f90c422e62	sdp: reindent and check for prepared status	2010-03-05 17:57:08 +01:00
Wim Taymans	c7ca9b74eb	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 17:54:09 +01:00
Wim Taymans	d45eae2edd	media: reindent	2010-03-05 16:20:08 +01:00
Wim Taymans	851e8aa744	media-factory: better error handling Improve the error handling a bit.	2010-03-05 13:34:15 +01:00
Wim Taymans	73e8d6c69a	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 13:31:37 +01:00
Wim Taymans	53f8350b36	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 13:28:58 +01:00
Wim Taymans	63addbc278	session: handle transport setup correctly Handle UDP, MCAST and TCP transport negotiation more correctly. Store the server session SSRC in the transport.	2010-03-05 13:27:18 +01:00
Wim Taymans	ce6724f788	rtsp-client: implement error_full Implement error_full to avoid some segfaults when the rtspconnection calls it. See #608245	2010-01-27 18:38:27 +01:00
Wim Taymans	996112db95	docs: update docs and comments	2009-12-25 18:24:10 +01:00
Nikolay Ivanov	92eb244215	sdp: make server work better when behind a proxy	2009-12-25 15:22:23 +01:00
Sebastian Pölsterl	3d7610b033	client: dump rtsp message only if debug threshold is higher than GST_LEVEL_LOG	2009-11-21 19:20:39 +01:00
Sebastian Pölsterl	6d227be7a9	Use GStreamer's debugging subsystem	2009-11-21 19:20:23 +01:00
Sebastian Pölsterl	87fbfa54a0	server: Set ghost pad active in gst_rtsp_media_factory_collect_streams	2009-11-21 19:20:23 +01:00
Luca Ognibene	745900dd48	client: call weak-unref on client->sessions from finalize Fixes bug #596305	2009-10-13 10:57:35 +02:00
Sebastian Pölsterl	f8630c6c81	media: Fixed crasher where caps got unref'ed too often	2009-10-13 10:57:31 +02:00
Wim Taymans	297b6a755a	media: add some docs	2009-09-11 13:52:27 +02:00
Peter Kjellerstedt	309f53a12b	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.	2009-08-24 13:27:00 +02:00
Wim Taymans	7338ab81e1	rtsp: allocate channels in TCP mode When the client does not provide us with channels in TCP mode, allocate channels ourselves.	2009-07-27 19:42:44 +02:00
Wim Taymans	daccf6bc99	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	2009-07-24 12:49:41 +02:00
Wim Taymans	a4c90c28c7	sessionpool: add function to filter sessions Add generic function to retrieve/remove sessions.	2009-06-30 21:27:53 +02:00
Wim Taymans	5d4c0e20c0	media: fix indentation	2009-06-18 16:05:18 +02:00
Sebastian Pölsterl	f384231ca3	Unref pipeline and set it to NULL. Set stream's caps to NULL, otherwise we unref it too often.	2009-06-18 15:54:15 +02:00
Sebastian Pölsterl	036550bf60	set state and remove elements of media in for loop	2009-06-18 15:54:11 +02:00
Sebastian	3bd2d36b1b	Added gst_rtsp_media_remove_elements function	2009-06-18 15:54:04 +02:00
Sebastian	1a3e5b369c	Don't use name for gstrtpbin so we can add multiple instances to the pipeline	2009-06-18 15:54:01 +02:00
Sebastian Pölsterl	749765b921	Added vmethod unprepare to GstRTSPMedia The default implementation sets the state of the pipeline to GST_STATE_NULL	2009-06-18 15:53:49 +02:00
Sebastian Pölsterl	045875ecbe	Made collect_streams function public	2009-06-18 15:53:42 +02:00
Sebastian Pölsterl	e417d83dce	Added vmethod create_pipeline to GstRTSPMediaFactory The pipeline is created in this method and the GstRTSPMedia's element is added to it	2009-06-18 15:53:34 +02:00
Wim Taymans	a697d16c75	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.	2009-06-11 11:27:47 +02:00
Wim Taymans	5e4757eff6	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.	2009-06-10 00:01:07 +02:00
Wim Taymans	94b6da045a	media: don't leak session pads	2009-06-04 19:20:26 +02:00
Wim Taymans	9a38f95417	media: clean up the messages a bit	2009-06-04 18:32:15 +02:00
Wim Taymans	e1765dec13	sdp: warn and skip streams without media	2009-06-03 12:13:21 +02:00
Wim Taymans	03ae66062b	media: fix message Fix a debug message Make dumping RTCP stats configurable	2009-05-27 11:15:22 +02:00
Wim Taymans	3fc1439965	media: be less verbose and leak less	2009-05-26 19:20:07 +02:00

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1135 commits