mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-05 14:16:06 +00:00
Code Issues 1 Projects Releases Wiki Activity
357 commits 44 branches 1505 tags 476 MiB
Commit graph

82 commits

Author SHA1 Message Date
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 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 b5aa7628bf Merge branch 'master' into 0.11 2011-08-16 11:12:33 +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
Wim Taymans 914b481e42 rtsp-server: port to 0.11 2011-04-26 19:22:50 +02:00
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 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