Wim Taymans [Tue, 20 Nov 2012 10:30:09 +0000 (11:30 +0100)]
test: add test for server reuse
See https://bugzilla.gnome.org/show_bug.cgi?id=688395
David Svensson Fors [Thu, 15 Nov 2012 13:02:37 +0000 (14:02 +0100)]
server: start and stop multiple times
Stop listening on the RTSP port when the GSource is removed, so clients
can't connect and the server can be started again.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688395
Wim Taymans [Tue, 20 Nov 2012 10:24:35 +0000 (11:24 +0100)]
server: fix small leak
Wim Taymans [Tue, 20 Nov 2012 08:42:51 +0000 (09:42 +0100)]
media: unref source in finish_unprepare
The source is created in prepare, unref it in finish_unprepare.
See https://bugzilla.gnome.org/show_bug.cgi?id=688707
David Svensson Fors [Mon, 19 Nov 2012 14:47:08 +0000 (15:47 +0100)]
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
Alessandro Decina [Sat, 17 Nov 2012 13:51:52 +0000 (14:51 +0100)]
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
David Svensson Fors [Mon, 19 Nov 2012 14:44:27 +0000 (15:44 +0100)]
rtsp-stream: plug socket leak
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
Tim-Philipp Müller [Mon, 19 Nov 2012 11:31:12 +0000 (11:31 +0000)]
Automatic update of common submodule
From 6bb6951 to a72faea
Tim-Philipp Müller [Sat, 17 Nov 2012 00:11:27 +0000 (00:11 +0000)]
rtsp-server: don't use deprecated API
Tim-Philipp Müller [Sat, 17 Nov 2012 00:03:42 +0000 (00:03 +0000)]
rtsp-client: fix unused-but-set-variable compiler warning
rtsp-client.c:1260:21: error: variable 'protocols' set but not used
Wim Taymans [Thu, 15 Nov 2012 16:11:16 +0000 (17:11 +0100)]
rtsp: cleanups
Wim Taymans [Thu, 15 Nov 2012 15:52:42 +0000 (16:52 +0100)]
examples: add another multicast example
Add an example for how to configure separate multicast ranges for each media
stream.
Wim Taymans [Thu, 15 Nov 2012 15:21:51 +0000 (16:21 +0100)]
test: set shared
Wim Taymans [Thu, 15 Nov 2012 15:18:29 +0000 (16:18 +0100)]
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.
Wim Taymans [Thu, 15 Nov 2012 15:15:20 +0000 (16:15 +0100)]
rtsp: improve debug
Wim Taymans [Thu, 15 Nov 2012 14:41:42 +0000 (15:41 +0100)]
media: add signal for new streams
This allows applications to listen for new streams and configure properties on
them, like the address pool.
Wim Taymans [Thu, 15 Nov 2012 14:41:19 +0000 (15:41 +0100)]
media: configure address pool in new streams
Wim Taymans [Thu, 15 Nov 2012 14:36:21 +0000 (15:36 +0100)]
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.
Wim Taymans [Thu, 15 Nov 2012 14:32:43 +0000 (15:32 +0100)]
media: remove MTU property
It is a stream property
Wim Taymans [Thu, 15 Nov 2012 14:29:35 +0000 (15:29 +0100)]
client: set blocksize only on stream
Set the blocksize only on the current stream.
Wim Taymans [Thu, 15 Nov 2012 12:52:07 +0000 (13:52 +0100)]
stream: share src and sink sockets
the allocated socket is in the used-socket property, not socket.
Wim Taymans [Thu, 15 Nov 2012 12:25:14 +0000 (13:25 +0100)]
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.
Wim Taymans [Thu, 15 Nov 2012 12:22:54 +0000 (13:22 +0100)]
examples: add multicast example
Show how to set up the multicast address pool so that media can be
server with multicast.
Wim Taymans [Wed, 14 Nov 2012 16:23:59 +0000 (17:23 +0100)]
rtsp: use AddressPool
Remove the multicast_group property.
Use the configured addresspool to allocate multicast addresses.
Wim Taymans [Wed, 14 Nov 2012 15:17:33 +0000 (16:17 +0100)]
address-pool: add clear method
Wim Taymans [Wed, 14 Nov 2012 15:10:45 +0000 (16:10 +0100)]
address-pool: small cleanups
Wim Taymans [Wed, 14 Nov 2012 14:50:42 +0000 (15:50 +0100)]
tests: add addresspool unit test
Wim Taymans [Wed, 14 Nov 2012 14:49:06 +0000 (15:49 +0100)]
address-pool: add object to manage multicast addresses
Make an object that can manage a rage of multicast addresses and ports.
Wim Taymans [Tue, 13 Nov 2012 11:05:42 +0000 (12:05 +0100)]
server: set default max-threads property
Wim Taymans [Tue, 13 Nov 2012 10:54:17 +0000 (11:54 +0100)]
media: wait for concurrent _prepare
If a prepare is busy, wait for the result.
Wim Taymans [Tue, 13 Nov 2012 10:49:08 +0000 (11:49 +0100)]
media: add lock around message handler
We don't want to dispatch messages while we are still processing the result of
the state change.
Wim Taymans [Tue, 13 Nov 2012 10:15:35 +0000 (11:15 +0100)]
media: add lock to protect state changes
Wim Taymans [Tue, 13 Nov 2012 10:14:49 +0000 (11:14 +0100)]
stream: add locking
Wim Taymans [Mon, 12 Nov 2012 16:11:18 +0000 (17:11 +0100)]
stream-transport: add keep-alive method
Wim Taymans [Mon, 12 Nov 2012 16:06:42 +0000 (17:06 +0100)]
stream-transport: add method to handle RTP/RTCP
Call new methods instead of poking into the structures directly.
Wim Taymans [Mon, 12 Nov 2012 15:51:03 +0000 (16:51 +0100)]
session-media: add locking
Wim Taymans [Mon, 12 Nov 2012 15:42:37 +0000 (16:42 +0100)]
session: add locking
Wim Taymans [Mon, 12 Nov 2012 15:30:16 +0000 (16:30 +0100)]
server: free old socket
Wim Taymans [Mon, 12 Nov 2012 15:18:57 +0000 (16:18 +0100)]
mapping: add locking
Wim Taymans [Mon, 12 Nov 2012 15:14:19 +0000 (16:14 +0100)]
media-factory: add locking
Wim Taymans [Mon, 12 Nov 2012 15:03:21 +0000 (16:03 +0100)]
auth: add locking
Wim Taymans [Mon, 12 Nov 2012 14:53:28 +0000 (15:53 +0100)]
server: add max-thread property
Wim Taymans [Mon, 12 Nov 2012 14:29:39 +0000 (15:29 +0100)]
server: use a threadpool for the mainloops
Wim Taymans [Mon, 12 Nov 2012 13:30:43 +0000 (14:30 +0100)]
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.
Wim Taymans [Mon, 12 Nov 2012 13:09:09 +0000 (14:09 +0100)]
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.
Wim Taymans [Mon, 12 Nov 2012 11:40:34 +0000 (12:40 +0100)]
session: move session header code in session object
Tim-Philipp Müller [Sun, 4 Nov 2012 00:14:25 +0000 (00:14 +0000)]
Fix FSF address
Sebastian Pölsterl [Sun, 28 Oct 2012 12:48:44 +0000 (13:48 +0100)]
rtsp-server: added annotations to indicate type of ownership transfer of return values
https://bugzilla.gnome.org/show_bug.cgi?id=680777
Tim-Philipp Müller [Sun, 28 Oct 2012 15:37:51 +0000 (15:37 +0000)]
No need to define GST_USE_UNSTABLE_API any more, 1.0 is stable now
Tim-Philipp Müller [Sun, 28 Oct 2012 15:09:04 +0000 (15:09 +0000)]
bindings: remove vala bindings
They'll be reunited with the other GStreamer bindings
https://bugzilla.gnome.org/show_bug.cgi?id=680777
Wim Taymans [Sat, 27 Oct 2012 22:23:57 +0000 (00:23 +0200)]
rtsp: only create transport when needed
Only create the StreamTransport when configured.
Wim Taymans [Sat, 27 Oct 2012 21:53:35 +0000 (23:53 +0200)]
client: small cleanup
Wim Taymans [Sat, 27 Oct 2012 21:49:24 +0000 (23:49 +0200)]
rtsp: refactor configuration of transport
Move the configuration of the transport to a place where it makes
more sense.
Wim Taymans [Sat, 27 Oct 2012 19:26:55 +0000 (21:26 +0200)]
client: refactor transport parsing
Wim Taymans [Sat, 27 Oct 2012 19:05:03 +0000 (21:05 +0200)]
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.
Wim Taymans [Sat, 27 Oct 2012 09:53:51 +0000 (11:53 +0200)]
small fixes to docs and debug
Wim Taymans [Fri, 26 Oct 2012 15:29:30 +0000 (17:29 +0200)]
stream: transports must already have been removed
Wim Taymans [Fri, 26 Oct 2012 15:28:10 +0000 (17:28 +0200)]
stream: improve join and leave of the pipeline
simplify code
Do the cleanup properly
Add some docs
Wim Taymans [Fri, 26 Oct 2012 13:23:16 +0000 (15:23 +0200)]
media: move unprepare below default implementation
Makes it easier to find the default implementation
Wim Taymans [Fri, 26 Oct 2012 13:21:50 +0000 (15:21 +0200)]
media: signal unprepared when we actually finish
Wim Taymans [Fri, 26 Oct 2012 13:19:23 +0000 (15:19 +0200)]
media: no need to unlock, unprepare does that when needed
Wim Taymans [Fri, 26 Oct 2012 10:33:21 +0000 (12:33 +0200)]
docs: update docs
Wim Taymans [Fri, 26 Oct 2012 10:04:02 +0000 (12:04 +0200)]
rtsp: fix MTU setting
Fix setting of the MTU. There is no need for a vmethod.
Wim Taymans [Fri, 26 Oct 2012 09:02:43 +0000 (11:02 +0200)]
docs: update docs
Tim-Philipp Müller [Fri, 26 Oct 2012 10:24:55 +0000 (11:24 +0100)]
configure: bump version number after refactoring
Wim Taymans [Thu, 25 Oct 2012 19:29:58 +0000 (21:29 +0200)]
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.
Sebastian Rasmussen [Tue, 23 Oct 2012 20:11:17 +0000 (22:11 +0200)]
rtsp-client: Unref server address clients connected to
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686725
Ognyan Tonchev [Mon, 22 Oct 2012 14:09:24 +0000 (16:09 +0200)]
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
Sebastian Rasmussen [Mon, 22 Oct 2012 14:29:09 +0000 (16:29 +0200)]
tests: Add libgio link dependency
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686647
Sebastian Pölsterl [Mon, 1 Oct 2012 18:03:43 +0000 (20:03 +0200)]
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
Sebastian Pölsterl [Mon, 1 Oct 2012 17:46:15 +0000 (19:46 +0200)]
rtsp-server: fixed comments and GIR annotations
https://bugzilla.gnome.org/show_bug.cgi?id=680777
Alessandro Decina [Fri, 12 Oct 2012 05:18:19 +0000 (07:18 +0200)]
media-mapping: fix transfer mode for gst_rtsp_media_mapping_add_factory
Alessandro Decina [Fri, 12 Oct 2012 05:08:57 +0000 (07:08 +0200)]
rtsp-server: allow binding on port 0 (binds on a random port)
Alessandro Decina [Fri, 12 Oct 2012 04:21:24 +0000 (06:21 +0200)]
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.
Alessandro Decina [Fri, 12 Oct 2012 04:11:36 +0000 (06:11 +0200)]
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.
Alessandro Decina [Fri, 12 Oct 2012 04:07:07 +0000 (06:07 +0200)]
rtsp-media-factory-uri: don't autoplug parsers in a loop
Stop autoplugging parsers if caps have parsed=true set. Fixes autoplugging
h264parse forever.
Alessandro Decina [Sat, 6 Oct 2012 13:49:07 +0000 (15:49 +0200)]
Explicitly link against gio. Fix link error on mac.
Ognyan Tonchev [Wed, 10 Oct 2012 09:13:10 +0000 (11:13 +0200)]
session: add ttl to the transport header in SETUP
See https://bugzilla.gnome.org/show_bug.cgi?id=685561
Ognyan Tonchev [Wed, 10 Oct 2012 09:06:02 +0000 (11:06 +0200)]
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
Tim-Philipp Müller [Sat, 6 Oct 2012 14:02:27 +0000 (15:02 +0100)]
Automatic update of common submodule
From 6c0b52c to 6bb6951
Patricia Muscalu [Mon, 1 Oct 2012 14:13:50 +0000 (16:13 +0200)]
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
Tim-Philipp Müller [Sat, 22 Sep 2012 15:11:48 +0000 (16:11 +0100)]
Automatic update of common submodule
From 4f962f7 to 6c0b52c
Ognyan Tonchev [Mon, 10 Sep 2012 14:25:57 +0000 (16:25 +0200)]
media: fix check for seekability
Wim Taymans [Fri, 7 Sep 2012 15:14:30 +0000 (17:14 +0200)]
client: use more GIO
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=681593
Wim Taymans [Fri, 7 Sep 2012 15:14:10 +0000 (17:14 +0200)]
server: remove obsolete includes
Aleix Conchillo Flaque [Tue, 4 Sep 2012 00:33:17 +0000 (17:33 -0700)]
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
Aleix Conchillo Flaque [Mon, 3 Sep 2012 17:48:14 +0000 (10:48 -0700)]
rtsp-client: add signals for rtsp requests (fixes #683287)
Aleix Conchillo Flaque [Thu, 30 Aug 2012 19:03:27 +0000 (12:03 -0700)]
add new-session signal to rtsp-client (fixes #683058)
Stefan Sauer [Wed, 22 Aug 2012 11:34:55 +0000 (13:34 +0200)]
Automatic update of common submodule
From 668acee to 4f962f7
Patricia Muscalu [Wed, 15 Aug 2012 13:54:32 +0000 (15:54 +0200)]
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
Tim-Philipp Müller [Sun, 5 Aug 2012 15:43:53 +0000 (16:43 +0100)]
Automatic update of common submodule
From 94ccf4c to 668acee
Patricia Muscalu [Wed, 18 Jul 2012 13:54:49 +0000 (15:54 +0200)]
rtsp-client: make create_sdp virtual method
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=680173
Sebastian Dröge [Mon, 23 Jul 2012 06:48:25 +0000 (08:48 +0200)]
Automatic update of common submodule
From 98e386f to 94ccf4c
Wim Taymans [Tue, 10 Jul 2012 09:39:58 +0000 (11:39 +0200)]
client: fix docs
Ognyan Tonchev [Tue, 3 Jul 2012 16:06:00 +0000 (18:06 +0200)]
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.
Ognyan Tonchev [Tue, 3 Jul 2012 11:26:30 +0000 (13:26 +0200)]
rtsp: Handle the blocksize parameter
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=679325
Sebastian Rasmussen [Mon, 25 Jun 2012 12:28:10 +0000 (14:28 +0200)]
Have unit test get header from source dir, not installed dir
This makes compilation of unit tests work in a build directory other
than the source directory.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678789
Tim-Philipp Müller [Sat, 23 Jun 2012 14:06:11 +0000 (15:06 +0100)]
rtsp-media: update for gst_element_make_from_uri() changes
David Svensson Fors [Tue, 19 Jun 2012 13:25:36 +0000 (15:25 +0200)]
rtsp: add unit test
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678076
David Svensson Fors [Wed, 13 Jun 2012 09:43:17 +0000 (11:43 +0200)]
rtsp-media: don't collect media stats when going to NULL
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=678015