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73 commits

Author SHA1 Message Date
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
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
748d044b62 client: unref auth in finalize 2011-01-12 12:07:20 +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
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
da35feb1aa rtsp: move network includes where they are needed 2011-01-11 22:42:25 +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
899f624845 client: fix typo 2010-12-28 12:18:41 +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
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
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
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
d749f1e7d5 client: use right size for malloc 2010-03-16 18:33:23 +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
Luca Ognibene
e19c382bbb client: call unlink_streams in client finalize 
Fixes #599027
 2010-03-05 18:37:17 +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
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
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
Luca Ognibene
745900dd48 client: call weak-unref on client->sessions from finalize 
Fixes bug #596305
 2009-10-13 10:57:35 +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
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
9bed89c3b7 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 19:01:10 +02:00
Wim Taymans
461169537b 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 17:25:59 +02:00
Wim Taymans
740d71bd50 client: warn when we can't do RTP-Info 2009-05-23 16:30:55 +02:00
Wim Taymans
8fcbe501dc 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.
 2009-05-23 16:17:02 +02:00
Wim Taymans
3f1f38f479 server: use appsink and appsrc with the API 
Use the appsink/appsrc API instead of the signals for higher
performance.
 2009-04-14 23:38:58 +02:00
Wim Taymans
47c822bdf3 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.
 2009-04-03 19:43:33 +02:00
Tim-Philipp Müller
0b8ffbbb5c 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 01:23:32 +01:00
Wim Taymans
525d639cde 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 20:32:14 +01:00
Wim Taymans
0ae095e825 allow pause requests for now. 
--
 2009-03-12 20:31:22 +01:00
Wim Taymans
d3c404f32f 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 20:03:06 +01:00
Wim Taymans
1be35624da 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 19:38:06 +01:00
Wim Taymans
ebc28a47da Add TCP transports 
Use appsrc and appsink to send and receive RTP/RTCP packets in the TCP
connection.
 2009-03-11 16:45:12 +01:00
Wim Taymans
de1ebbc21b Add support for live streams 
Add support for live streams and ranges
Start on handling TCP data transfer.
 2009-03-06 19:34:14 +01:00
Wim Taymans
d85b34f1b1 Only free the pending tunnel if there is one 
--
 2009-03-04 16:33:21 +01:00
Wim Taymans
2f8025dbdd 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.
 2009-03-04 12:53:07 +01:00