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399 commits

Author SHA1 Message Date
Wim Taymans
c34f5d1c1a media: add signal for new streams 
This allows applications to listen for new streams and configure properties on
them, like the address pool.
 2012-11-15 15:41:42 +01:00
Wim Taymans
4168a67992 media: configure address pool in new streams 2012-11-15 15:41:19 +01:00
Wim Taymans
44a2855eb3 stream: add methods to deal with address pool 
Add methods to get and set the address pool for the stream
Add method to allocate and get the multicast addresses for this stream.
 2012-11-15 15:36:21 +01:00
Wim Taymans
1b4ac6e5b0 media: remove MTU property 
It is a stream property
 2012-11-15 15:32:43 +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
6c2947e68b stream: share src and sink sockets 
the allocated socket is in the used-socket property, not socket.
 2012-11-15 13:52:07 +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
d0ffc8e679 address-pool: add clear method 2012-11-14 16:20:36 +01:00
Wim Taymans
6085b1fcc1 address-pool: small cleanups 2012-11-14 16:10:45 +01:00
Wim Taymans
b30202b174 address-pool: add object to manage multicast addresses 
Make an object that can manage a rage of multicast addresses and ports.
 2012-11-14 15:49:06 +01:00
Wim Taymans
7d6e4606fa server: set default max-threads property 2012-11-13 12:05:42 +01:00
Wim Taymans
dfe3efef74 media: wait for concurrent _prepare 
If a prepare is busy, wait for the result.
 2012-11-13 11:54:17 +01:00
Wim Taymans
47127bd270 media: add lock around message handler 
We don't want to dispatch messages while we are still processing the result of
the state change.
 2012-11-13 11:49:08 +01:00
Wim Taymans
9a97de88ea media: add lock to protect state changes 2012-11-13 11:15:35 +01:00
Wim Taymans
4753588b09 stream: add locking 2012-11-13 11:14:49 +01:00
Wim Taymans
c7d20e5603 stream-transport: add keep-alive method 2012-11-12 17:11:18 +01:00
Wim Taymans
75473fc88d stream-transport: add method to handle RTP/RTCP 
Call new methods instead of poking into the structures directly.
 2012-11-12 17:06:42 +01:00
Wim Taymans
883cf794e4 session-media: add locking 2012-11-12 16:51:03 +01:00
Wim Taymans
11cf3f3ccb session: add locking 2012-11-12 16:42:37 +01:00
Wim Taymans
65fa516677 server: free old socket 2012-11-12 16:30:16 +01:00
Wim Taymans
04881bd632 mapping: add locking 2012-11-12 16:18:57 +01:00
Wim Taymans
b8cba7719c media-factory: add locking 2012-11-12 16:14:19 +01:00
Wim Taymans
e61c84c9bb auth: add locking 2012-11-12 16:03:21 +01:00
Wim Taymans
06cadebe71 server: add max-thread property 2012-11-12 15:53:28 +01:00
Wim Taymans
8523c9ca92 server: use a threadpool for the mainloops 2012-11-12 15:29:39 +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
Sebastian Pölsterl
75598337a9 rtsp-server: added annotations to indicate type of ownership transfer of return values 
https://bugzilla.gnome.org/show_bug.cgi?id=680777
 2012-10-28 15:39:04 +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
0bb0e3733c small fixes to docs and debug 2012-10-27 11:53:51 +02:00
Wim Taymans
6a838fd5c8 stream: transports must already have been removed 2012-10-26 17:29:30 +02:00
Wim Taymans
6f7d755894 stream: improve join and leave of the pipeline 
simplify code
Do the cleanup properly
Add some docs
 2012-10-26 17:28:10 +02:00
Wim Taymans
693dd3cfc4 media: move unprepare below default implementation 
Makes it easier to find the default implementation
 2012-10-26 15:23:16 +02:00
Wim Taymans
0d55e1f50c media: signal unprepared when we actually finish 2012-10-26 15:21:50 +02:00
Wim Taymans
84b7cf1590 media: no need to unlock, unprepare does that when needed 2012-10-26 15:19:23 +02:00
Wim Taymans
348b7f9c21 docs: update docs 2012-10-26 12:35:20 +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
Ognyan Tonchev
78bde6fa3e rtsp-server: don't ref server socket if it is NULL 
Fixes test_bind_already_in_use unit test again after commit 6a497440.

https://bugzilla.gnome.org/show_bug.cgi?id=686644
 2012-10-22 18:11:28 +01:00
Sebastian Pölsterl
5cec59737b rtsp-media-mapping: rename find_media vfunc to find_factory 
The virtual method and class method should have the same name
so it is correctly represented in GIR file

https://bugzilla.gnome.org/show_bug.cgi?id=680777
 2012-10-18 19:31:23 +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
Alessandro Decina
bc474a5b26 media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory 2012-10-15 10:50:27 +02:00
Alessandro Decina
1e954a1a5e rtsp-server: allow binding on port 0 (binds on a random port) 2012-10-15 10:50:27 +02:00
Alessandro Decina
6a49744088 rtsp-server: add bound-port property 
bound-port can be used to retrieve the port number when the server is bound on
port 0, which binds on a random port.
 2012-10-15 10:50:27 +02:00
Alessandro Decina
8f507e4512 rtsp-media-factory: make ::get_element overridable by GI bindings 
The way to annotate vfuncs with GI seems to be to create an invoker (GI term)
for them and to annotate the invoker. Add gst_rtsp_media_factory_get_element()
as the invoker for ::get_element(), making it overridable by GI generated
bindings.
 2012-10-15 10:50:26 +02:00
Alessandro Decina
3a49b8e783 rtsp-media-factory-uri: don't autoplug parsers in a loop 
Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
h264parse forever.
 2012-10-15 10:50:26 +02:00
Alessandro Decina
8da18a85ef Explicitly link against gio. Fix link error on mac. 2012-10-15 10:50:26 +02:00
Ognyan Tonchev
4f0ef292f0 session: add ttl to the transport header in SETUP 
See https://bugzilla.gnome.org/show_bug.cgi?id=685561
 2012-10-10 11:13:58 +02:00
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
Ognyan Tonchev
f4a0a371b7 media: fix check for seekability 2012-09-10 16:29:35 +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
Wim Taymans
87c73c06fb server: remove obsolete includes 2012-09-07 17:14:10 +02:00
Aleix Conchillo Flaque
c6cce4a6b9 rtsp-media: also initialize transports in on_ssrc_active (bug #683304) 
* gst/rtsp-server/rtsp-media.c: GstRTSPMediaStream transports might not
  be available in "on_new_ssrc". The transports are added in
  gst_rtsp_media_set_state when going to PLAYING state. However,
  "on_new_ssrc" might be called before this happens.

  https://bugzilla.gnome.org/show_bug.cgi?id=683304
 2012-09-07 16:45:17 +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
50e4c7e8c4 rtsp-server: fixed segfault in gst_rtsp_server_create_socket 
Do not assume that *error is set in g_socket_address_enumerator_next.
Added test_bind_already_in_use unit-test.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681914
 2012-08-20 11:49:27 +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
Tim-Philipp Müller
217a46e4c1 rtsp-media: update for gst_element_make_from_uri() changes 2012-06-23 15:06:11 +01:00
David Svensson Fors
36df0dd8be rtsp-media: don't collect media stats when going to NULL 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015
 2012-06-14 10:14:06 +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
aa158fa738 factory: plug pad leak in collect_streams 
In gst_rtsp_media_factory_collect_streams: unref the srcpad that
was retrieved using gst_element_get_static_pad. gst_ghost_pad_new
will take one reference, and the other reference will otherwise
give a memory leak.
 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
Sebastian Dröge
7df1696713 configure: Modernize autotools setup a bit 
Also we now only create tar.bz2 and tar.xz tarballs.
 2012-04-13 14:02:15 +02:00
Sebastian Dröge
fb0718a036 rtsp-server: Update versioning 2012-04-04 14:48:44 +02:00
Sebastian Dröge
e9ef6f6254 Merge remote-tracking branch 'origin/0.10' 
Conflicts:
	gst/rtsp-server/rtsp-session-pool.c
 2012-03-29 15:12:21 +02:00
Sebastian Dröge
1f442d45b6 rtsp-server: Don't use deprecated GLib API 2012-03-27 10:13:20 +02:00
Wim Taymans
e0be150e91 media: fix state of the appqueue 2012-03-13 18:10:53 +01:00
Wim Taymans
6403227471 factory: use videoconvert 2012-03-13 16:07:16 +01:00
Wim Taymans
377f6d9156 factory: change to new style caps 2012-03-13 16:02:47 +01:00
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
bace3995d5 Merge branch 'master' into 0.11 2011-11-03 12:58:42 +01:00
Wim Taymans
a701e8595e media: add a seekable boolean 
Maintain the seekable state with a new variable instead of reusing the
is_live variable.
 2011-11-03 12:55:24 +01:00
Victor Gottardi
526bbb5a8f Disallow seek in live media 2011-11-03 12:45:18 +01:00
Wim Taymans
05c3928b11 Merge branch 'master' into 0.11 2011-11-03 11:58:42 +01:00
mat
20b6be3852 #ifdef statements for windows socket creation were missing 2011-11-03 11:56:51 +01:00
Wim Taymans
6759a4b9b0 client: use method to access property 2011-08-16 16:39:11 +02:00
Wim Taymans
4c8f3696d0 media-factory: add protocols property 
Add a property to configure the allowed protocols in the media created from the
factory.
 2011-08-16 16:39:07 +02:00
Wim Taymans
85e2013ca4 media-factory: add media-configure signal 
Add signal to allow the application to configure the media after it was created
from the factory.
 2011-08-16 16:39:04 +02:00
Wim Taymans
6fa73b2552 client: use method to access property 2011-08-16 16:07:04 +02:00
Wim Taymans
0e9ce1caf3 media-factory: add protocols property 
Add a property to configure the allowed protocols in the media created from the
factory.
 2011-08-16 15:15:19 +02:00