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Author SHA1 Message Date
Wim Taymans
17b07d1c0e Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute" 
This reverts commit 5fd034ff1a.

We already have a way to place extra attributes in the SDP by using a string
property with prefix x- or a- in the caps.
 2013-05-31 15:43:11 +02:00
Wim Taymans
2a0aaa1019 Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute" 
This reverts commit d6a4dee036.

We already have a way to place extra attributes in the SDP, just make a string
property in the payloader with a- or x- prefix.
 2013-05-31 15:43:11 +02:00
Wim Taymans
cfdf2e6db5 rtsp: place a- and x- properties as attributes 
application/x-rtp has properties with a- and x- prefixes that should be
placed as attributes in the SDP for the media instead of being added to the
fmtp.
 2013-05-31 15:43:10 +02:00
Wim Taymans
0a285290cb server: add support for TLS 
Add methods to set and get a TLS certificate.
Add vmethod to configure a new connection. By default, configure the TLS
certificate in a new connection if needed.
 2013-05-31 11:42:36 +02:00
Wim Taymans
531ffca018 server: remove accept_client vmethod 
This vmethod is not very useful so remove it.
 2013-05-31 11:14:17 +02:00
Wim Taymans
0091339254 server: don't crash on NULL GError 2013-05-30 17:23:51 +02:00
Patricia Muscalu
aa50aac669 rtsp-session-pool: corrected session timeout detection 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253
 2013-05-30 13:13:05 +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
Wim Taymans
7b880231b1 stream: keep the transport object alive 
Keep the transport object alive while we have it as qdata on the
source.
 2013-05-30 06:49:20 +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
Sebastian Rasmussen
d6a4dee036 rtsp-sdp: Parse framerate caps field and set SDP attribute 
The SDP attribute and its format is described in RFC4566.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
 2013-05-23 21:02:58 +02:00
Sebastian Rasmussen
5fd034ff1a rtsp-sdp: Parse width/height from caps and set SDP attribute 
The SDP attribute and its format is described in RFC6064.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=700747
 2013-05-23 21:02:50 +02:00
Patricia Muscalu
0951aa37e1 rtsp-sdp: add bandwidth line 
https://bugzilla.gnome.org/show_bug.cgi?id=699220
 2013-05-15 12:36:32 +02:00
Wim Taymans
573b10bc83 media: release lock when removing fakesink 2013-04-23 10:28:35 +02:00
Wim Taymans
0ddd98bfa6 stream: set elements to NULL before removing 
When removing a stream, set the elements to NULL first. This avoids
element-is-not-in-NULL-state errors when we dispose the elements.
 2013-04-23 10:28:34 +02:00
Wim Taymans
b80b8824be media: listen to pad-removed signals 
Listen to the pad-removed signal and remove the stream associated with the
removed pad.
Add signal to be notified of the removed pad.
Remove the fakesink in unprepare()
Fix signatures of the signal methods
 2013-04-22 17:34:37 +02:00
Ognyan Tonchev
00291e5285 stream: add method to get the srcpad 2013-04-22 17:32:31 +02:00
Ognyan Tonchev
a26b06cc69 media: disconnect from signal handlers in unprepare() 
We connected to the pad-added and no-more-pads signals in prepare() so
we need to disconnect from them in unprepare().

See https://bugzilla.gnome.org/show_bug.cgi?id=698376
 2013-04-22 16:40:48 +02:00
Ognyan Tonchev
9b31fcc7f8 media: don't free streams array 
Don't free the streams array in the unprepare() method, they were not
added in prepare().

See https://bugzilla.gnome.org/show_bug.cgi?id=698376
 2013-04-22 16:25:17 +02:00
Ognyan Tonchev
0bdff0161c media: don't unref the pipeline in unprepare 
Unprepare() should undo what prepare() does. Because the pipeline is
not created in prepare(), we should not unref it in unprepare()
 2013-04-22 16:19:35 +02:00
Ognyan Tonchev
6081f91351 stream: clear session and caps for reuse 
Set the session and caps to NULL after unref otherwise we might unref
them again later.

See https://bugzilla.gnome.org/show_bug.cgi?id=698376
 2013-04-22 16:09:22 +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
Wim Taymans
a64cb68164 media: add method to get the base_time of the pipeline 
Together with a shared clock, this base-time could eventually be sent to
the client so that it can reconstruct the exact running-time of the clock
on the server.
 2013-04-12 11:34:38 +01:00
Wim Taymans
36ff679558 media: add GstNetTimeProvider support 
Add a property to let the media provide a GstNetTimeProvider for its clock.
Make methods to get the clock and nettimeprovider
Add a x-gst-clock property to the SDP with the IP and port number of the nettime
provider and also the current time of the clock. This should make it possible
for (GStreamer) clients to slave their clock to the server clock.
 2013-04-09 22:38:44 +02:00
Wim Taymans
95bf53513f media: wait for buffering to complete 
Wait for buffering to complete before changing the state to the target state.
 2013-04-09 20:39:58 +02:00
Wim Taymans
ec0718d7c9 media: small cleanup 2013-04-09 20:11:35 +02:00
Olivier Crête
91210f40f2 rtsp-media-factory: g_signal_connect_object is not thread safe, can't use it here 
Instead use a GWeakRef which is safe to use

This is a known GLib bug, see:
https://bugzilla.gnome.org/show_bug.cgi?id=667145
 2013-03-22 18:59:50 -04:00
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
5a39e25949 stream: Select unicast address from pool if appropriate 2013-03-11 11:07:20 +01:00
Olivier Crête
a797cbde06 stream: Properties are always there in Gst 1.0 2013-03-11 11:07:20 +01:00
Olivier Crête
27a057962c address-pool: Verify that multicast addresses are used for multicast and vice-versa 2013-03-11 11:07:20 +01:00
Olivier Crête
d06e68abd1 address-pool: Add unicast addresses 2013-03-11 11:07:20 +01:00
Olivier Crête
4c61c6d308 rtsp-server: Limit the number of threads per server instance 
If we exceed the maximum, just round robin the clients over the existing
threads.
 2013-03-11 11:07:20 +01:00
Olivier Crête
4071e1b999 rtsp-server: No need to store the GMainContext in the client context 2013-03-11 11:07:20 +01:00
Olivier Crête
b9d111372e Document locking and its order 2013-03-11 11:07:19 +01:00
Olivier Crête
f0ab7ce1bf docs: Generate docs for GstRTSPAddressPool 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
Olivier Crête
cda75709bb address-pool: Add API to request a specific address from the pool 
Also add relevant unit tests.
 2013-03-11 11:07:19 +01:00
Olivier Crête
456f4367e3 address-pool: Fix off by one error 
When splitting a port range, the port after a skip is not part of range.
 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
Ognyan Tonchev
0844e8afbc rtsp-media: Make the element a constructor parameter 
https://bugzilla.gnome.org/show_bug.cgi?id=689594
 2012-12-10 10:25:57 +01:00
Wim Taymans
6beabf1ed4 media: match prepare with unprepare 
Really unprepare when there were an equal amount of prepare calls.
 2012-11-30 15:03:15 +01:00
Wim Taymans
ca26588c7e media: media has to be unprepared in finalize 
Because unprepare takes away the last ref on the media.
 2012-11-30 14:58:46 +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
Wim Taymans
119674a828 media: let the source unref the last media ref 
the last ref to the media is held by the source so we don't need to add more ref
and unrefs, we simply destroy the media when the source is gone.
 2012-11-30 13:35:05 +01:00
Wim Taymans
339ea9b085 media: improve debug 2012-11-30 12:54:10 +01:00
Wim Taymans
241baba20a media: check state 
Make sure we are in the right state when collecting the position and duration.
Only make ourselves PREPARED when we were previously PREPARING.
 2012-11-30 12:53:02 +01:00
Wim Taymans
edf2ef4f0b media: use g_object_ref/unref for GObjects 2012-11-30 10:05:48 +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
Alessandro Decina
00d9a94e1a Fix compiler warning 2012-11-30 06:17:46 +01:00
Alessandro Decina
e2a7690cb3 Add missing g_type_class_add_private in GstRTSPMediaFactoryURI 2012-11-30 06:14:49 +01:00
Wim Taymans
1abc9be682 small cleanup 2012-11-29 17:21:12 +01:00
Wim Taymans
28fd887547 media: avoid element leak 2012-11-29 17:20:56 +01:00
Wim Taymans
4eb010824e media: require an element in media constructor 2012-11-29 17:20:26 +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
e11287eb7c media: check if prepared for some methods 
Check that the media object is prepared before doing seek and getting the
current position etc.
Add some g_return checks.
 2012-11-28 14:45:30 +01:00
Wim Taymans
d3d74ab77b stream: improve debug 2012-11-28 12:40:18 +01:00
Wim Taymans
fe71114a7d media: unref pipeline in finalize to avoid leaking it 2012-11-28 12:39:37 +01:00
Wim Taymans
d43a31055e rtsp: use gst_object_unref on GstObjects 2012-11-28 12:10:47 +01:00
Wim Taymans
6b36241816 media-factory: require an url 2012-11-28 12:10:14 +01:00
Wim Taymans
20f09bf3e7 server: remove unused include 2012-11-28 11:17:27 +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
b4c168c71d mounts: add g_return_if guards 2012-11-28 10:40:14 +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
fc0f176a17 client: remove unused include 2012-11-26 17:34:35 +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
Wim Taymans
0f93879b2c media: fix seeking 2012-11-21 17:21:28 +01:00
Wim Taymans
5eb5fd45f3 media: support more Range formats 
Use the new -base methods to convert the Range string into a seek start and stop
value.
 2012-11-21 16:41:56 +01:00
Wim Taymans
37a7ec8033 factory: keep ref to factory while media active 
While the media from a factory is alive, keep a ref to the factory.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=663555
 2012-11-20 12:29:55 +01:00
Wim Taymans
8fcdca987d factory-uri: add some debug 2012-11-20 12:29:26 +01:00
Wim Taymans
1826844ee4 stream: set udp sources to PLAYING 
Set the UDP sources to PLAYING and locked state before we add it to the pipeline
so that it doesn't cause our pipeline to produce ASYNC-DONE.
 2012-11-20 12:24:13 +01:00
Wim Taymans
8211cdfdc2 factory-uri: take ref to factory 
Take a ref to the factory that we place in our list.
 2012-11-20 12:10:16 +01:00
David Svensson Fors
0eeb4a5c73 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
 2012-11-20 11:30:37 +01:00
Wim Taymans
8a7197f078 server: fix small leak 2012-11-20 11:24:35 +01:00
Wim Taymans
989f004e24 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
 2012-11-20 09:46:40 +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
David Svensson Fors
0996266342 rtsp-stream: plug socket leak 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688703
 2012-11-20 09:26:28 +01:00
Tim-Philipp Müller
0006ca6d60 rtsp-server: don't use deprecated API 2012-11-17 00:11:27 +00:00
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
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
Wim Taymans
8684fc5c69 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 15:03:06 +02:00
Wim Taymans
56a16f9f5a client: use media multicast group 2011-08-16 14:50:21 +02:00
Wim Taymans
2c9701bd73 retab some .h 2011-08-16 14:50:18 +02:00
Robert Krakora
a5e028ba72 sdp: copy and free the server ip address 
Copy and free the server ip address to make memory management easier later.
 2011-08-16 14:50:15 +02:00
Wim Taymans
647e8c7af8 media-factory: configure multicast in media 2011-08-16 14:50:12 +02:00
Wim Taymans
c079325169 media: add property for multicast group 
Add a property to configure the multicast group in the media.

Based on patches from Marc Leeman and Robert Krakora.
 2011-08-16 14:50:05 +02:00
Wim Taymans
514728864a media-factory: add property for multicast group 
Add a property to configure the multicast group in the media factory.

Based on patches from Marc Leeman and Robert Krakora.
 2011-08-16 14:49:59 +02:00
Wim Taymans
b881dc6669 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 14:49:55 +02:00
Wim Taymans
9573058f54 client: use media multicast group 2011-08-16 13:43:44 +02:00
Wim Taymans
26c8898e79 retab some .h 2011-08-16 13:37:50 +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
ccfb99f852 media-factory: configure multicast in media 2011-08-16 13:27:39 +02:00
Wim Taymans
5b53335873 media: add property for multicast group 
Add a property to configure the multicast group in the media.

Based on patches from Marc Leeman and Robert Krakora.
 2011-08-16 13:25:16 +02:00
Wim Taymans
1f8b97d940 media-factory: add property for multicast group 
Add a property to configure the multicast group in the media factory.

Based on patches from Marc Leeman and Robert Krakora.
 2011-08-16 13:13:36 +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
1aefff4959 Merge branch 'master' into 0.11 2011-08-16 11:53:37 +02:00
Emmanuel Pacaud
5dc9e76125 media-factory: add a "media-constructed" signal to GstRTSPMediaFactory 
For example, it can be used to retrieve source elements like appsrc, in a more
convenient way than subclassing get_element.
 2011-08-16 11:22:55 +02:00
Wim Taymans
b5aa7628bf Merge branch 'master' into 0.11 2011-08-16 11:12:33 +02:00
David Schleef
041b62db8b rtsp-server: hold on to reference while using object 2011-08-11 18:07:08 -07:00
Wim Taymans
bbab01747d media: use new api 2011-08-04 08:59:17 +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
Edward Hervey
b93f046708 Makefile.am: 0.10 => @GST_MAJORMINOR@ 2011-06-07 11:04:10 +02:00
Edward Hervey
597a99e9b9 rtsp-media-factory-uri: GST_PLUGIN_FEATURE_NAME is no longer 2011-06-07 10:59:16 +02:00
Edward Hervey
14f8ed65b4 .gitignore: 0.10 => 0.11 2011-06-07 10:59:03 +02:00
Edward Hervey
c94416d486 Makefile.am: 0.10 => @GST_MAJORMINOR@ 2011-06-07 10:54:26 +02:00
Wim Taymans
80e0b0b19a media: port to new caps API 2011-05-17 09:48:13 +02:00
Wim Taymans
debbea1008 Merge branch 'master' into 0.11 2011-05-17 09:45:04 +02:00
Fabian Deutsch
6ef7c966ae Add a signal for newly connected clients. 
Signed-off-by: Fabian Deutsch <fabian.deutsch@gmx.de>
 2011-05-17 09:44:14 +02:00
Wim Taymans
914b481e42 rtsp-server: port to 0.11 2011-04-26 19:22:50 +02:00
Wim Taymans
6959ebd8e8 Merge branch 'master' into 0.11 
Conflicts:
	common
	configure.ac
 2011-04-26 19:07:13 +02:00
Miguel Angel Cabrera Moya
17ce0df09a session: use full charset for RTSP session ID 
As specified in RFC 2326 section 3.4 use full valid charset to make guessing
session ID more difficult.

https://bugzilla.gnome.org/show_bug.cgi?id=643812
 2011-03-07 18:39:43 +00:00
Sebastian Dröge
63744dfece rtsp-server: Don't install the funnel header 2011-03-07 10:23:06 +01:00
Wim Taymans
a924e90c79 media: remove more unused code 2011-02-02 15:37:03 +01:00
Wim Taymans
ec2201a3a8 media: remove duplicate filtering 
Remove the duplicate filtering code now that we have a released -good version.
Give a warning instead.
 2011-02-02 15:30:45 +01:00
Wim Taymans
8477fdbf43 media: fix default buffer size 2011-01-31 17:38:47 +01:00
Wim Taymans
e86b7c4b15 media-factory: add property to configure the buffer-size 
Add a property to configure the kernel UDP buffer size.
 2011-01-31 17:37:45 +01:00
Wim Taymans
88b4c02dff media: add property to configure kernel buffer sizes 
Add a property to configure the kernel UDP buffer size.
 2011-01-31 17:37:41 +01:00
Wim Taymans
325b2cf8a2 rtsp-server: clarify docs a little 2011-01-19 15:29:55 +01:00
Wim Taymans
44b418b346 media: init debug category before starting thread 2011-01-13 18:57:15 +01:00
Wim Taymans
cd8382674d auth: add realm to make it more spec compliant 2011-01-13 18:40:48 +01:00
Wim Taymans
b076933f5e server: add locking 2011-01-12 18:57:41 +01:00
Wim Taymans
94c9999715 server: ensure the watch has a ref to the server 2011-01-12 18:26:57 +01:00
Wim Taymans
3315031bf6 server: simpify channel function 2011-01-12 18:24:44 +01:00
Wim Taymans
ba4d65a673 server: simplify management of channel and source 
We don't need to keep around the channel and source objects. Let the mainloop
and the source manage the source and channel respectively.
 2011-01-12 18:18:13 +01:00
Wim Taymans
9e97faf2db server: improve debugging in various objects 2011-01-12 18:14:48 +01:00
Wim Taymans
0ef53a2d4f server: chain up to the parent finalize 2011-01-12 16:38:34 +01:00
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
318b3a1df4 server: use signal to keep track of clients 
Keep track of all the clients that the server creates and remove them when they
fire the 'closed' signal.
 2011-01-12 15:36:22 +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
9ea0346d97 media-factory: add methods to configure authorisation 2011-01-12 12:07:40 +01:00
Wim Taymans
748d044b62 client: unref auth in finalize 2011-01-12 12:07:20 +01:00
Wim Taymans
6915572695 server: unref auth in finalize 2011-01-12 12:07:04 +01:00
Wim Taymans
6d6ba1ee61 server: separate create and accept 
Create separate create and accept methods so that subclasses can create custom
client object.
Configure the server in the client object and prepare for keeping track of
connected clients.
 2011-01-12 10:57:08 +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
9f52f281ba auth: fix memleak and add some docs 
Fix a memleak of the basic auth token.
Add docs for the helper function
 2011-01-12 10:41:42 +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
61bee9985a server: move includes back 
the includes are needed for sockaddr_in.
 2011-01-12 00:20:36 +01:00
Wim Taymans
da35feb1aa rtsp: move network includes where they are needed 2011-01-11 22:42:25 +01:00
Sreerenj Balachandran
28597c913d rtsp-media.h: Minor corrections in comments. 
Fixes #638944
 2011-01-11 21:32:45 +01:00
Edward Hervey
2cc9eee3e6 gitignore: updates 2011-01-11 13:04:31 +01:00
Wim Taymans
e1787e0776 funnel: rename fsfunnel to rtspfunnel 
Rename the funnel to avoid conflicts with the farsight one.
 2011-01-10 15:10:53 +01:00
Wim Taymans
7b3cbfde1b rtsp-media: add and use fsfunnel 
Add a copy of fsfunnel to the build because input-selector removed the (broken)
select-all property that we need.
 2011-01-10 13:43:10 +01:00
Tim-Philipp Müller
c19eb8fb4e gobject-introspection: use PKG_CONFIG_PATH specified at configure time 
Use PKG_CONFIG_PATH specified at configure time (if any) as well
for the g-ir-compiler, rather than just assuming the env var has
been set.
 2011-01-08 02:00:12 +00:00
Tim-Philipp Müller
8b1ec41d08 gobject-introspection: fix g-i build for uninstalled setup 
Requires gst-plugins-base git (> 0.10.31.2).
 2011-01-08 01:15:35 +00:00
Wim Taymans
186089ff1e factory-uri: use right property type 2011-01-07 11:24:39 +01:00
Wim Taymans
257bac1bab factory-uri: attempt to configure buffer-lists 
Attempt to configure buffer lists in the payloader for improved performance.
 2011-01-05 12:07:42 +01:00
Wim Taymans
790c067919 media: attempt to configure bigger UDP buffers 
Attempt to configure bigger udp kernel send buffers to avoid overflowing the
send buffers with high bitrate streams.
 2011-01-05 12:06:23 +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
915cd708ea media: add signal to notify of state changes 2010-12-28 18:34:10 +01:00
Wim Taymans
43f4696f78 client: cleanup headers 2010-12-28 18:31:26 +01:00
Wim Taymans
899f624845 client: fix typo 2010-12-28 12:18:41 +01:00
Wim Taymans
50a71b9d86 factory-uri: add support for gstpay 
Add an option to prefer gstpay over decoder + raw payloader.
 2010-12-23 18:53:01 +01:00
Wim Taymans
9ce4ea165b factory-uri: rework the autoplugger. 
Rewrite the autoplugger a little so that it prefers to plug demuxers and parsers
before payloaders.
 2010-12-23 15:58:14 +01:00
Wim Taymans
1053860112 factory-uri: use better factory filter 
Make better payloader filter based on autoplug rank and RTP use case.
 2010-12-21 17:39:09 +01:00
Wim Taymans
ad2e0edee5 server: set SO_REUSEADDR before bind 
Set the SO_REUSEADDR _before_ bind() to make it actually work.
 2010-12-18 11:24:48 +01:00
Wim Taymans
1ea450179e media: emit prepared signal when prepared 
Make a 'prepared' signal and emit it when we successfully prepared the element.
This signal can be used to configure the media object after it has been prepared
for streaming.
 2010-12-17 18:45:10 +01:00
Wim Taymans
fc12ade012 Merge branch 'master' into 0.11 
Conflicts:
	common
	configure.ac
 2010-12-13 11:43:13 +01:00
Wim Taymans
ca76a73ca0 media: update range when active clients changed 
When we changed the number of active clients, update the current range
information because we want the second client connecting to a shared resource
continue from where the stream currently.
 2010-12-12 15:48:47 +01:00
Wim Taymans
d99a448f79 factory-uri: add colorspace and fix pt 
Rework the way we pass data to the autoplugger.
When we have raw caps, plug a converter element to make pluggin to raw
payloaders more successful.
Make sure all dynamically plugged payloaders have a unique payload types.
 2010-12-12 04:06:41 +01:00
Wim Taymans
7ef0bf98da factory-uri: add a factory to stream any URI 
Make a factory that uses uridecodebin to decode any uri and autoplug a payloader
when we have one.
 2010-12-11 18:04:34 +01:00
Wim Taymans
34f0973831 media: ignore spurious ASYNC_DONE messages 
When we are dynamically adding pads, the addition of the udpsrc elements will
trigger an ASYNC_DONE. We have to ignore this because we only want to react to
the real ASYNC_DONE when everything is prerolled.
 2010-12-11 18:04:34 +01:00
Wim Taymans
75a7cda97d media-factory: make lock macro 2010-12-11 18:04:29 +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
Edward Hervey
b95165fcff rtsp-server: Some more doc fixups 2010-12-11 10:48:25 +01:00
Edward Hervey
bdd477f2bf Makefile.am: Use standard GIR make behaviour 2010-12-07 18:14:39 +01:00
Wim Taymans
422fea478c media: warn and fail when gstrtpbin is not found 2010-12-06 19:29:53 +01:00
Sebastian Pölsterl
347e10e1f9 Added initial gobject-introspection support 2010-09-23 13:39:42 +02:00
Wim Taymans
c310f0032c media-factory: don't use host for shared hash key 
When we generate the key to share made between connections, don't include the
host used to connect so that we can share media even if between clients that
connected with localhost and ones with the ip address.
 2010-09-23 11:35:40 +02:00
Wim Taymans
ed473f6f26 media: help the compiler a little 2010-09-22 16:15:56 +02:00
Wim Taymans
450b68252f media: cleanup media transport before freeing 
Cleanup the media transport data before freeing. In particular, remove the qdata
from the rtpsource object.
 2010-08-24 16:47:30 +02:00
Wim Taymans
dc33070da3 media-factory: add eos-shutdown property 
Add an eos-shutdown property that will send an EOS to the pipeline before
shutting it down. This allows for nice cleanup in case of a muxer.

Fixes #625597
 2010-08-20 18:17:08 +02:00
Wim Taymans
a900866570 media: use multiudpsink send-duplicates when we can 
If we have a new enough multiudpsink with the send-duplicates property, use this
instead of doing our own filtering. Our custom filtering code should eventually
be removed when we can depend on a released -good.
 2010-08-20 15:58:39 +02:00
Wim Taymans
c89d17ca26 media: don't leak destinations 
Refactor and cleanup the destinations array when the stream is destroyed.
 2010-08-20 13:19:56 +02:00
Wim Taymans
7c0f8a77ec media: don't add udp addresses multiple times 
Keep track of the udp addresses we added to udpsink and never add the same udp
destination twice. This avoids duplicate packets when using multicast.
 2010-08-20 13:09:12 +02:00
Wim Taymans
af732fa749 server: disable use of SO_LINGER 
SO_LINGER cause the client to fail to receive a TEARDOWN message because the
server close()s the connection.
 2010-08-20 10:18:34 +02:00
Wim Taymans
2607ff079d server: use 5 second linger period in SO_LINGER 
Wait 5 seconds before clearing the send buffers and reseting the connection with
the client when we do a close. This should be enough time to get the message to
the client.

See #622757
 2010-08-19 18:52:47 +02:00
Robert Krakora
8f6fd32065 server: use SO_LINGER 
SO_LINGER on the socket will make sure that any pending data on the socket is
flushed ASAP and that the socket connection is reset. This makes sure that the
socket can be reused immediately.

Fixes 622757
 2010-08-16 12:45:24 +02:00
David Schleef
6a880e53df Add stdlib.h for atoi() 2010-08-09 12:56:23 -07:00
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
4fdd2bf4d1 session: add support for prevent session timeouts 
Add an atomix counter to prevent session timeouts when we are, for example,
streaming over TCP.
 2010-04-06 17:07:27 +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
558c7fddd2 session: small cleanups 2010-04-06 15:44:45 +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
ac8343ea62 media: allow configuration of allowed lower transport 2010-03-19 15:15:29 +01:00
Wim Taymans
e866345f15 rtsp: keep track of server ip and ipv6 
Keep track of how the client connected to the server and setup the udp ports
with the same protocol.
Copy the server ip address in the SDP so that clients can send RTCP back to
us.
 2010-03-16 18:37:18 +01:00
Wim Taymans
4eccdd9dd7 session: indent 2010-03-16 18:34:43 +01:00
Wim Taymans
d749f1e7d5 client: use right size for malloc 2010-03-16 18:33:23 +01:00
Wim Taymans
0509aa1cbf server: comment ipv6 server listening address 2010-03-10 11:45:30 +01:00
Wim Taymans
6afa5be799 media: allow for ipv6 sockets 2010-03-10 11:45:06 +01:00
Wim Taymans
17bb89f1fc server: rework server part 
Allow setting a bind address, make sure we can deal with ipv6.
Remove the port property and change with the service property.
 2010-03-09 13:49:00 +01:00
Wim Taymans
1b0dc41534 media: update comments a little 2010-03-09 13:44:20 +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
Alessandro Decina
5f535ecf87 rtspmedia: emit "unprepared" if _prepare fails. 
Emit the unprepared signal if gst_rtsp_media_prepare fails so that the
media object is removed from its factory's cache.
 2010-03-09 10:27:38 +01:00
Wim Taymans
2997806d43 media: collect media position when seek completes 2010-03-05 19:08:08 +01:00
Luca Ognibene
e19c382bbb client: call unlink_streams in client finalize 
Fixes #599027
 2010-03-05 18:37:17 +01:00
Wim Taymans
83ed258684 media: limit the time to wait to something huge 
Avoid waiting forever but limit the timeout to 20 seconds.
 2010-03-05 18:23:18 +01:00
Wim Taymans
f90c422e62 sdp: reindent and check for prepared status 2010-03-05 17:57:08 +01:00
Wim Taymans
c7ca9b74eb media: avoid doing _get_state() for state changes 
When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait
until the media is prerolled or in error. This avoids doing a blocking call of
gst_element_get_state() that can cause lockups when there is an error.

Fixes #611899
 2010-03-05 17:54:09 +01:00
Wim Taymans
d45eae2edd media: reindent 2010-03-05 16:20:08 +01:00
Wim Taymans
851e8aa744 media-factory: better error handling 
Improve the error handling a bit.
 2010-03-05 13:34:15 +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
53f8350b36 media: set multicast sink parameters 
Disable loop and automatic multicast join on the udpsink elements.
Add some more debug info.
Reset some state variables in the right place.
Use the right port numbers for multicast.
 2010-03-05 13:28:58 +01:00
Wim Taymans
63addbc278 session: handle transport setup correctly 
Handle UDP, MCAST and TCP transport negotiation more correctly.
Store the server session SSRC in the transport.
 2010-03-05 13:27:18 +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
Nikolay Ivanov
92eb244215 sdp: make server work better when behind a proxy 2009-12-25 15:22:23 +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
Sebastian Pölsterl
87fbfa54a0 server: Set ghost pad active in gst_rtsp_media_factory_collect_streams 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
Sebastian Pölsterl
f8630c6c81 media: Fixed crasher where caps got unref'ed too often 2009-10-13 10:57:31 +02:00
Wim Taymans
297b6a755a media: add some docs 2009-09-11 13:52:27 +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