Commit graph

165 commits

Author SHA1 Message Date
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
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
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
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
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
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
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
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
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
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
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
6fa73b2552 client: use method to access property 2011-08-16 16:07:04 +02:00
Wim Taymans
9573058f54 client: use media multicast group 2011-08-16 13:43:44 +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
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
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