platform/upstream/gstreamer.git
11 years agoclient: clean some variables
Wim Taymans [Tue, 25 Jun 2013 13:09:13 +0000 (15:09 +0200)]
client: clean some variables

Clean some variables and add some guards to _send_request()

11 years agoAdd gst_rtsp_client_send_request API
Youness Alaoui [Mon, 10 Jun 2013 21:32:12 +0000 (17:32 -0400)]
Add gst_rtsp_client_send_request API

This makes it possible to send arbitrary messages to a client, such as
SET_PARAMETER or GET_PARAMETER

11 years agomedia: add _get_element() method
Wim Taymans [Mon, 24 Jun 2013 21:56:57 +0000 (23:56 +0200)]
media: add _get_element() method

Add method to get the element used when creating the media.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008

11 years agomedia: fix docs
Wim Taymans [Mon, 24 Jun 2013 21:51:38 +0000 (23:51 +0200)]
media: fix docs

11 years agostream: allow access to the rtp session
Aleix Conchillo Flaque [Mon, 24 Jun 2013 18:41:27 +0000 (11:41 -0700)]
stream: allow access to the rtp session

  https://bugzilla.gnome.org/show_bug.cgi?id=703004

11 years agodscp qos support in gst-rtsp-stream
Alexander Schrab [Mon, 24 Jun 2013 08:43:59 +0000 (10:43 +0200)]
dscp qos support in gst-rtsp-stream

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645

11 years agotests: fix test
Wim Taymans [Thu, 20 Jun 2013 15:30:49 +0000 (17:30 +0200)]
tests: fix test

Actually do what the comment says. Also keep the old code around, not sure what
should happen when you get a 454 from a TEARDOWN, does it close the connection?
it currently doesn't.

11 years agoclient: also watch newly created session
Wim Taymans [Thu, 20 Jun 2013 10:20:21 +0000 (12:20 +0200)]
client: also watch newly created session

When we newly created a session, start watching it immediately instead of
on the next request.

11 years agotests: add unit test for new-session
Patricia Muscalu [Thu, 20 Jun 2013 10:18:23 +0000 (12:18 +0200)]
tests: add unit test for new-session

See https://bugzilla.gnome.org/show_bug.cgi?id=701587

11 years agoclient: emit new-session when new session is created
Wim Taymans [Thu, 20 Jun 2013 10:16:07 +0000 (12:16 +0200)]
client: emit new-session when new session is created

Only emit new-session when we created a new session for a client, not when a
client picked up a previous session.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701587

11 years agoclient: handle asterisk as path in requests
Alexander Schrab [Thu, 20 Jun 2013 09:17:29 +0000 (11:17 +0200)]
client: handle asterisk as path in requests

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701266

11 years agomedia: handle segment query format mismatch
Wim Taymans [Thu, 20 Jun 2013 09:14:31 +0000 (11:14 +0200)]
media: handle segment query format mismatch

It's possible that the segment query returns with a different format than what
we asked for, handle this case also.

11 years agomedia: use segment stop in collect_media_stats
David Svensson Fors [Tue, 11 Jun 2013 13:28:32 +0000 (15:28 +0200)]
media: use segment stop in collect_media_stats

Use segment stop instead of duration as range end point.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701185

11 years agortsp-media: Do not leak the element in take_pipeline
Ognyan Tonchev [Mon, 17 Jun 2013 14:47:56 +0000 (16:47 +0200)]
rtsp-media: Do not leak the element in take_pipeline

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702470

11 years agortsp-client: Make configure_client_transport virtual
Ognyan Tonchev [Mon, 17 Jun 2013 14:18:37 +0000 (16:18 +0200)]
rtsp-client: Make configure_client_transport virtual

This patch makes configure_client_transport virtual. The functionality is
needed to handle some weird clients sending multicast transport settings as url
options.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702173

11 years agortsp-client: Make param_set and param_get virtual
Ognyan Tonchev [Wed, 12 Jun 2013 10:23:56 +0000 (12:23 +0200)]
rtsp-client: Make param_set and param_get virtual

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702072

11 years agomedia: convert_range replaces get_range_times
David Svensson Fors [Wed, 5 Jun 2013 13:49:45 +0000 (15:49 +0200)]
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

11 years agosdp: cleanup sdp info
Wim Taymans [Fri, 14 Jun 2013 14:05:59 +0000 (16:05 +0200)]
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

11 years agouse 0.0.0.0 or :: for c= line instead of server address
Alexander Schrab [Wed, 12 Jun 2013 13:22:57 +0000 (15:22 +0200)]
use 0.0.0.0 or :: for c= line instead of server address

11 years agouse local address, not remote, in SDP
Alexander Schrab [Wed, 12 Jun 2013 08:56:16 +0000 (10:56 +0200)]
use local address, not remote, in SDP

See https://bugzilla.gnome.org/show_bug.cgi?id=702063

11 years agoAutomatic update of common submodule
Sebastian Dröge [Wed, 5 Jun 2013 13:18:26 +0000 (15:18 +0200)]
Automatic update of common submodule

From 098c0d7 to 01a7a46

11 years agomedia: possibility to override range time conversion
David Svensson Fors [Wed, 29 May 2013 11:45:00 +0000 (13:45 +0200)]
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

11 years agortsp-server: Expose the use_client_settings API
Ognyan Tonchev [Mon, 3 Jun 2013 10:04:44 +0000 (12:04 +0200)]
rtsp-server: Expose the use_client_settings API

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=699935

11 years agortspstream: handle both ipv4 and ipv6 clients
Alexander Schrab [Thu, 30 May 2013 06:07:48 +0000 (08:07 +0200)]
rtspstream: handle both ipv4 and ipv6 clients

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129

11 years agoRevert "rtsp-sdp: Parse width/height from caps and set SDP attribute"
Wim Taymans [Fri, 31 May 2013 13:28:58 +0000 (15:28 +0200)]
Revert "rtsp-sdp: Parse width/height from caps and set SDP attribute"

This reverts commit 5fd034ff1a517db7f629ffcc3ed16839c61f5c97.

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.

11 years agoRevert "rtsp-sdp: Parse framerate caps field and set SDP attribute"
Wim Taymans [Fri, 31 May 2013 13:27:48 +0000 (15:27 +0200)]
Revert "rtsp-sdp: Parse framerate caps field and set SDP attribute"

This reverts commit d6a4dee03642a2d2c05fec4752dc3ccb60b19494.

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.

11 years agortsp: place a- and x- properties as attributes
Wim Taymans [Fri, 31 May 2013 13:41:55 +0000 (15:41 +0200)]
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.

11 years agoexample: add TLS example
Wim Taymans [Fri, 31 May 2013 10:10:28 +0000 (12:10 +0200)]
example: add TLS example

11 years agoserver: add support for TLS
Wim Taymans [Fri, 31 May 2013 09:42:36 +0000 (11:42 +0200)]
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.

11 years agoserver: remove accept_client vmethod
Wim Taymans [Fri, 31 May 2013 09:14:17 +0000 (11:14 +0200)]
server: remove accept_client vmethod

This vmethod is not very useful so remove it.

11 years agoserver: don't crash on NULL GError
Wim Taymans [Thu, 30 May 2013 15:23:51 +0000 (17:23 +0200)]
server: don't crash on NULL GError

11 years agortsp-session-pool: corrected session timeout detection
Patricia Muscalu [Thu, 30 May 2013 08:46:33 +0000 (10:46 +0200)]
rtsp-session-pool: corrected session timeout detection

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701253

11 years agoclient: improve debug
Wim Taymans [Thu, 30 May 2013 08:52:46 +0000 (10:52 +0200)]
client: improve debug

11 years agoserver: refactor connection setup
Wim Taymans [Thu, 30 May 2013 05:18:22 +0000 (07:18 +0200)]
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.

11 years agostream: keep the transport object alive
Wim Taymans [Thu, 30 May 2013 04:49:20 +0000 (06:49 +0200)]
stream: keep the transport object alive

Keep the transport object alive while we have it as qdata on the
source.

11 years agortsp-server: Do not crash on nmapping of server
Alexander Schrab [Mon, 27 May 2013 10:58:07 +0000 (12:58 +0200)]
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

11 years agortsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6
Alexander Schrab [Fri, 24 May 2013 11:39:50 +0000 (13:39 +0200)]
rtsp-client: ipv4 adress should not be marked ipv6 even if socket is ipv6

https://bugzilla.gnome.org/show_bug.cgi?id=700953

11 years agortsp-sdp: Parse framerate caps field and set SDP attribute
Sebastian Rasmussen [Wed, 22 May 2013 01:29:38 +0000 (03:29 +0200)]
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

11 years agortsp-sdp: Parse width/height from caps and set SDP attribute
Sebastian Rasmussen [Wed, 22 May 2013 01:29:30 +0000 (03:29 +0200)]
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

11 years agortsp-sdp: add bandwidth line
Patricia Muscalu [Mon, 29 Apr 2013 12:46:30 +0000 (14:46 +0200)]
rtsp-sdp: add bandwidth line

https://bugzilla.gnome.org/show_bug.cgi?id=699220

11 years agoAutomatic update of common submodule
Sebastian Dröge [Wed, 15 May 2013 08:55:09 +0000 (10:55 +0200)]
Automatic update of common submodule

From 5edcd85 to 098c0d7

11 years agotests: add dynamic payloader prepare/unprepare check
Ognyan Tonchev [Tue, 23 Apr 2013 09:28:39 +0000 (11:28 +0200)]
tests: add dynamic payloader prepare/unprepare check

11 years agomedia: release lock when removing fakesink
Wim Taymans [Tue, 23 Apr 2013 08:27:35 +0000 (10:27 +0200)]
media: release lock when removing fakesink

11 years agostream: set elements to NULL before removing
Wim Taymans [Tue, 23 Apr 2013 08:16:17 +0000 (10:16 +0200)]
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.

11 years agoAutomatic update of common submodule
Tim-Philipp Müller [Mon, 22 Apr 2013 22:55:48 +0000 (23:55 +0100)]
Automatic update of common submodule

From 3cb3d3c to 5edcd85

11 years agomedia: listen to pad-removed signals
Wim Taymans [Mon, 22 Apr 2013 15:34:37 +0000 (17:34 +0200)]
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

11 years agotests: add example of reusable pipelines
Wim Taymans [Mon, 22 Apr 2013 15:33:30 +0000 (17:33 +0200)]
tests: add example of reusable pipelines

11 years agostream: add method to get the srcpad
Ognyan Tonchev [Mon, 22 Apr 2013 15:32:31 +0000 (17:32 +0200)]
stream: add method to get the srcpad

11 years agocheck: add media prepare/unprepare test
Ognyan Tonchev [Mon, 22 Apr 2013 14:49:39 +0000 (16:49 +0200)]
check: add media prepare/unprepare test

See https://bugzilla.gnome.org/show_bug.cgi?id=698376

11 years agomedia: disconnect from signal handlers in unprepare()
Ognyan Tonchev [Mon, 22 Apr 2013 14:40:48 +0000 (16:40 +0200)]
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

11 years agomedia: don't free streams array
Ognyan Tonchev [Mon, 22 Apr 2013 14:25:17 +0000 (16:25 +0200)]
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

11 years agomedia: don't unref the pipeline in unprepare
Ognyan Tonchev [Mon, 22 Apr 2013 14:19:35 +0000 (16:19 +0200)]
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()

11 years agostream: clear session and caps for reuse
Ognyan Tonchev [Mon, 22 Apr 2013 14:09:22 +0000 (16:09 +0200)]
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

11 years agoclient: send out teardown signal before tearing down
David Svensson Fors [Mon, 15 Apr 2013 10:21:54 +0000 (12:21 +0200)]
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

11 years agoclient: expose connection
David Svensson Fors [Mon, 15 Apr 2013 10:17:34 +0000 (12:17 +0200)]
client: expose connection

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697546

11 years agoAutomatic update of common submodule
Tim-Philipp Müller [Sun, 14 Apr 2013 16:58:22 +0000 (17:58 +0100)]
Automatic update of common submodule

From aed87ae to 3cb3d3c

11 years agomedia: add method to get the base_time of the pipeline
Wim Taymans [Fri, 12 Apr 2013 10:34:38 +0000 (11:34 +0100)]
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.

11 years agomedia: add GstNetTimeProvider support
Wim Taymans [Tue, 9 Apr 2013 20:35:28 +0000 (22:35 +0200)]
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.

11 years agoAutomatic update of common submodule
Stefan Sauer [Tue, 9 Apr 2013 19:02:47 +0000 (21:02 +0200)]
Automatic update of common submodule

From 04c7a1e to aed87ae

11 years agomedia: wait for buffering to complete
Wim Taymans [Tue, 9 Apr 2013 18:39:58 +0000 (20:39 +0200)]
media: wait for buffering to complete

Wait for buffering to complete before changing the state to the target state.

11 years agomedia: small cleanup
Wim Taymans [Tue, 9 Apr 2013 18:11:35 +0000 (20:11 +0200)]
media: small cleanup

11 years agotests: remove extra unref in test_setup_non_existing_stream
David Svensson Fors [Wed, 20 Mar 2013 11:33:54 +0000 (12:33 +0100)]
tests: remove extra unref in test_setup_non_existing_stream

The unref is not needed anymore, teardown runs without it.

https://bugzilla.gnome.org/show_bug.cgi?id=696542

11 years agotests: GSocketService cleanup in test_bind_already_in_use
David Svensson Fors [Wed, 20 Mar 2013 10:28:11 +0000 (11:28 +0100)]
tests: GSocketService cleanup in test_bind_already_in_use

Use g_socket_service_stop so the rtspserver test stops listening for
incoming connections in test_bind_already_in_use.

https://bugzilla.gnome.org/show_bug.cgi?id=696541

11 years agortsp-media-factory: g_signal_connect_object is not thread safe, can't use it here
Olivier Crête [Fri, 22 Mar 2013 22:25:07 +0000 (18:25 -0400)]
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

11 years agortsp-media/client: Reply to PLAY request with same type of Range
Olivier Crête [Fri, 22 Feb 2013 19:17:29 +0000 (14:17 -0500)]
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.

11 years agortsp-client: expose uri
Patricia Muscalu [Mon, 18 Mar 2013 08:25:54 +0000 (09:25 +0100)]
rtsp-client: expose uri

11 years agotests: Hold ref while creating second media
Olivier Crête [Wed, 13 Mar 2013 21:46:58 +0000 (17:46 -0400)]
tests: Hold ref while creating second media

To test if the media aren't shared, make sure we keep the first one while creating a second
otherwise the same memory address may be reused.

11 years agoconfigure: remove out-of-date comment
Tim-Philipp Müller [Tue, 12 Mar 2013 00:10:18 +0000 (00:10 +0000)]
configure: remove out-of-date comment

11 years ago.gitignore: ignore more build files
Tim-Philipp Müller [Tue, 12 Mar 2013 00:05:49 +0000 (00:05 +0000)]
.gitignore: ignore more build files

11 years agotests: use right _LIBS variable for gst-plugins-base libs
Tim-Philipp Müller [Tue, 12 Mar 2013 00:03:36 +0000 (00:03 +0000)]
tests: use right _LIBS variable for gst-plugins-base libs

11 years agocheck: add librtp to libs
Wim Taymans [Mon, 11 Mar 2013 10:35:14 +0000 (11:35 +0100)]
check: add librtp to libs

11 years agotests: Add test to check selecting a port the server will send from
Olivier Crête [Thu, 21 Feb 2013 00:37:51 +0000 (19:37 -0500)]
tests: Add test to check selecting a port the server will send from

11 years agotests: Make sure packets are actually received
Olivier Crête [Wed, 20 Feb 2013 23:30:01 +0000 (18:30 -0500)]
tests: Make sure packets are actually received

11 years agostream: Select unicast address from pool if appropriate
Olivier Crête [Tue, 19 Feb 2013 23:27:20 +0000 (18:27 -0500)]
stream: Select unicast address from pool if appropriate

11 years agostream: Properties are always there in Gst 1.0
Olivier Crête [Tue, 19 Feb 2013 21:43:08 +0000 (16:43 -0500)]
stream: Properties are always there in Gst 1.0

11 years agotests: Add tests for unicast addresses in pool
Olivier Crête [Tue, 19 Feb 2013 21:36:20 +0000 (16:36 -0500)]
tests: Add tests for unicast addresses in pool

11 years agoaddress-pool: Verify that multicast addresses are used for multicast and vice-versa
Olivier Crête [Wed, 20 Feb 2013 19:26:03 +0000 (14:26 -0500)]
address-pool: Verify that multicast addresses are used for multicast and vice-versa

11 years agoaddress-pool: Add unicast addresses
Olivier Crête [Tue, 19 Feb 2013 21:34:16 +0000 (16:34 -0500)]
address-pool: Add unicast addresses

11 years agortsp-server: Limit the number of threads per server instance
Olivier Crête [Tue, 19 Feb 2013 18:19:41 +0000 (13:19 -0500)]
rtsp-server: Limit the number of threads per server instance

If we exceed the maximum, just round robin the clients over the existing
threads.

11 years agortsp-server: No need to store the GMainContext in the client context
Olivier Crête [Tue, 19 Feb 2013 17:31:23 +0000 (12:31 -0500)]
rtsp-server: No need to store the GMainContext in the client context

11 years agotests: Add test for client disconnection
Olivier Crête [Tue, 19 Feb 2013 01:22:18 +0000 (20:22 -0500)]
tests: Add test for client disconnection

11 years agotests: Test client and session timeouts with multiple threads
Olivier Crête [Tue, 19 Feb 2013 01:15:41 +0000 (20:15 -0500)]
tests: Test client and session timeouts with multiple threads

11 years agoDocument locking and its order
Olivier Crête [Mon, 18 Feb 2013 19:59:58 +0000 (14:59 -0500)]
Document locking and its order

11 years agotests: Test that slow DESCRIBE don't block other clients
Olivier Crête [Sat, 16 Feb 2013 01:02:31 +0000 (20:02 -0500)]
tests: Test that slow DESCRIBE don't block other clients

11 years agotests: Add tests for client-requested multicast address
Olivier Crête [Fri, 15 Feb 2013 00:52:09 +0000 (19:52 -0500)]
tests: Add tests for client-requested multicast address

11 years agodocs: Put the various functions in the right sections
Olivier Crête [Thu, 14 Feb 2013 18:44:54 +0000 (13:44 -0500)]
docs: Put the various functions in the right sections

11 years agodocs: Generate docs for GstRTSPAddressPool
Olivier Crête [Thu, 14 Feb 2013 18:38:07 +0000 (13:38 -0500)]
docs: Generate docs for GstRTSPAddressPool

11 years agoclient: Check client provided addresses against the address pool
Olivier Crête [Wed, 13 Feb 2013 23:32:20 +0000 (18:32 -0500)]
client: Check client provided addresses against the address pool

11 years agoaddress-pool: Add API to request a specific address from the pool
Olivier Crête [Wed, 13 Feb 2013 23:01:43 +0000 (18:01 -0500)]
address-pool: Add API to request a specific address from the pool

Also add relevant unit tests.

11 years agotests: Check the passing around of a RTSPAddressPool
Olivier Crête [Wed, 13 Feb 2013 00:34:24 +0000 (19:34 -0500)]
tests: Check the passing around of a RTSPAddressPool

Make sure the RTSPAddressPool is propagated from the MediaFactory all the
way down to the stream.

11 years agotests: Add more tests for the address pool
Olivier Crête [Tue, 12 Feb 2013 21:34:37 +0000 (16:34 -0500)]
tests: Add more tests for the address pool

11 years agoaddress-pool: Fix off by one error
Olivier Crête [Tue, 12 Feb 2013 21:29:25 +0000 (16:29 -0500)]
address-pool: Fix off by one error

When splitting a port range, the port after a skip is not part of range.

11 years agoAutomatic update of common submodule
Tim-Philipp Müller [Thu, 7 Mar 2013 00:04:19 +0000 (00:04 +0000)]
Automatic update of common submodule

From 2de221c to 04c7a1e

11 years agoconfigure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS
George McCollister [Thu, 7 Feb 2013 22:18:08 +0000 (16:18 -0600)]
configure: replace deprecated AM_CONFIG_HEADER with AC_CONFIG_HEADERS

AM_CONFIG_HEADER was removed in automake 1.13

https://bugzilla.gnome.org/show_bug.cgi?id=693368

11 years agoAutomatic update of common submodule
Stefan Sauer [Mon, 28 Jan 2013 19:45:44 +0000 (20:45 +0100)]
Automatic update of common submodule

From a942293 to 2de221c

11 years agoclient: make sure the watch exists while sending data
Wim Taymans [Mon, 28 Jan 2013 09:31:50 +0000 (10:31 +0100)]
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

11 years agotests: use GST_*_1_0 environment variables everywhere
Tim-Philipp Müller [Wed, 16 Jan 2013 12:16:32 +0000 (12:16 +0000)]
tests: use GST_*_1_0 environment variables everywhere

The _1_0 suffixed environment variables override the
non-suffixed ones, so if we're in an environment that
sets the _1_0 suffixed ones, such as jhbuild, we need
to set those to make sure ours actually always get
used.

11 years agoAutomatic update of common submodule
Tim-Philipp Müller [Tue, 15 Jan 2013 15:09:24 +0000 (15:09 +0000)]
Automatic update of common submodule

From acb04d9 to a942293

11 years agortsp-client: set the client backlog
Wim Taymans [Fri, 14 Dec 2012 10:58:29 +0000 (11:58 +0100)]
rtsp-client: set the client backlog

Set the client backlog to a reasonable default

11 years agortsp-media: Make the element a constructor parameter
Ognyan Tonchev [Tue, 4 Dec 2012 08:47:35 +0000 (09:47 +0100)]
rtsp-media: Make the element a constructor parameter

https://bugzilla.gnome.org/show_bug.cgi?id=689594