Commit graph

310 commits

Author SHA1 Message Date
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
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
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
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
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
b9d111372e Document locking and its order 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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
Wim Taymans
b19b1fbe6b Cleanups and reuse media from DESCRIBE
Handle thread create errors.
Rename some internal methods to better match what they actually do.
Handle misconfiguration of session_pool and media_mapping gracefully.
Cache the DESCRIBE media and uri in the client connection and reuse them when
we receive a SETUP request in the same connection for the same uri.
Cleanup the client connection object.
2009-01-30 12:17:51 +01:00
Wim Taymans
082099005d Add a little comment
Add some comment about the content-base header.
2009-01-30 12:17:38 +01:00
Wim Taymans
41dd6399a6 Reorganize things, prepare for media sharing
Added various other test server examples
Move the SDP message generation to a separate helper.
Refactor common code for finding the session.
Add content-base for realplayer compatibility
Clean up request uris before processing for better vlc compatibility.
Move prerolling and pipeline construction to the RTSPMedia object.
Use multiudpsink for future pipeline reuse.
2009-01-30 12:17:28 +01:00
Wim Taymans
b0fcbfd290 Cleanups and doc updates
Add some more documentation and do some minor cleanups here and there.
2009-01-22 18:35:17 +01:00
Wim Taymans
cf18709634 More improvements
Rename GstRTSPMediaBin to GstRTSPMedia
Parse the request url into a GstRTSPUri object and pass this object to the
various handlers and methods that require the uri.
2009-01-22 17:58:19 +01:00
Wim Taymans
6f9b659b1d Handle state change failures better
Handle state change failures better when changing the state of the pipeline to
determine the SDP.
2009-01-22 16:53:16 +01:00
Wim Taymans
4b1c190a5f Make the server handle arbitrary pipelines
Make GstMediaFactory an object that can instantiate GstMediaBin objects.
The GstMediaBin object has a handle to a bin with elements and to a list of
GstMediaStream objects that this bin produces.

Add GstMediaMapper that can map url mountpoints to GstMediaFactory objects along
with methods to register and remove those mappings.

Add methods and a property to GstRTSPServer to manage the GstMediaMapper object
used by the server instance.

Modify the example application so that it shows how to create custom pipelines
attached to a specific mount point.

Various misc cleanps.
2009-01-22 15:33:29 +01:00
Wim Taymans
94d60a8611 Allow setting a custom media factory for a client. 2009-01-20 19:46:21 +01:00
Wim Taymans
b312f98627 Move the connection code in one place
Add some comments
2009-01-20 13:57:47 +01:00
Wim Taymans
8d2ace0026 Name the parameters more appropriately. 2009-01-19 19:34:29 +01:00
Wim Taymans
a76656ad8d Check if return value of gst_rtsp_session_get_media is not NULL 2009-01-08 16:28:24 +01:00
Wim Taymans
7889395787 Split in library and example program 2009-01-08 13:18:55 +01:00
Renamed from src/rtsp-client.c (Browse further)