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62 commits

Author SHA1 Message Date
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
Wim Taymans
daf27d2704 Fix for channel -> watch rename in gstreamer 
Rename the RTSPChannel to RTSPWatch and remove an unused variable.
 2009-02-19 15:53:50 +01:00
Wim Taymans
39c2e31e65 Use ASYNC RTSP io 
Use the async RTSP channels instead of spawning a new thread for each client.
If a sessionid is specified in a request, fail if we don't have the session.
 2009-02-18 18:57:31 +01:00
Wim Taymans
308ad6f6d0 Add support for session keepalive 
Get and update the session timeout for all requests. get the session as early as
possible.
 2009-02-13 19:52:05 +01:00
Wim Taymans
e1154c92d6 Some more session timeout handling 
Move the session header setting code to a central place so that we always add
the timeout parameter too.

Handle timeouts by running the session cleanup code.

Stop media before cleaning up.
 2009-02-13 12:57:45 +01:00
Wim Taymans
34152ec840 Add timeout property 
Add a timeout property ot the client and make the other properties into GObject
properties.
 2009-02-10 16:24:13 +01:00
Wim Taymans
aedd4652f3 Add beginnings of session timeouts and limits 
Add the timeout value to the Session header for unusual timeout values.

Allow us to configure a limit to the amount of active sessions in a pool. Set a
limit on the amount of retry we do after a sessionid collision.

Add properties to the sessionid and the timeout of a session. Keep track of
creation time and last access time for sessions.
 2009-02-04 19:52:50 +01:00
Wim Taymans
e789a8fdf3 Cleanup of sessions and more 
Fix the refcounting of media and sessions in the client. Properly clean up the
session data when the client performs a teardown.

Add Server header to responses.

Allow for multiple uri setups in one session.

Add Range header to the PLAY response and add the range attribute to the SDP
message.

Fix the session pool remove method, it used the wrong key in the hashtable. Also
give the ownership of the sessionid to the session object.
 2009-02-04 17:00:42 +01:00
Wim Taymans
d5a00f1f23 Rework the way we handle transports for streams 
Make the media accept an array of transports for the streams that we have
configured for the play/pause requests.

Implement server states for a client and its media.

Require 0.10.22.1 (git HEAD) of gstreamer.
 2009-02-03 19:32:38 +01:00
Wim Taymans
f303eef9bb Drop const from functions dealing with urls 
Drop const from GstRTSPUrl stuff because the .h files in gst-plugins-base don't
have the right const in them.
 2009-01-31 19:50:33 +01:00
Wim Taymans
ae2521096a Fix various leaks 
Fix some leaks.
 2009-01-30 17:06:26 +01:00
Wim Taymans
27f069b43c More cleanups 
Don't keep a reference to the GstRTSPMedia in the stream.
Free more things when freeing the GstRTSPMedia.
 2009-01-30 16:24:10 +01:00