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Author SHA1 Message Date
David Svensson Fors
6151072a2e media: convert_range replaces get_range_times 
get_range_times worked for handling UTC ranges for seeks, but we also
need to convert back from NPT to the requested unit in
get_range_string. convert_range is now used for both.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
 2013-06-14 16:11:34 +02:00
Wim Taymans
3dbe0e17d4 sdp: cleanup sdp info 
We don't need to pass the proto, we can more easily check a boolean.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702063
 2013-06-14 16:06:46 +02:00
Alexander Schrab
f9f994e33d use 0.0.0.0 or :: for c= line instead of server address 2013-06-14 15:58:52 +02:00
Alexander Schrab
275e2d52a4 use local address, not remote, in SDP 
See https://bugzilla.gnome.org/show_bug.cgi?id=702063
 2013-06-14 15:52:14 +02:00
David Svensson Fors
7efa871c1f media: possibility to override range time conversion 
Make it possible to override the conversion from GstRTSPTimeRange to
GstClockTimes, that is done before seeking on the media
pipeline. Overriding can be useful for UTC ranges, where the default
conversion gives nanoseconds since 1900.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701191
 2013-06-03 14:29:05 +02:00
Ognyan Tonchev
c5b3066c33 rtsp-server: Expose the use_client_settings API 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935
 2013-06-03 12:04:44 +02:00
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
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