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341 commits

Author SHA1 Message Date
Руслан Ижбулатов 151d156126 rtsp: Link to ws2_32 on Windows 
Needed for getsockname and setsockopt

https://bugzilla.gnome.org/show_bug.cgi?id=729514
 2014-05-05 09:04:28 +02:00
Tim-Philipp Müller b163f111c8 rtspdefs: remove outdated comments 2014-05-02 19:36:34 +01:00
Göran Jönsson 9685e7a583 rtspconnection: Empty queue when flush. 
Empty the watchs queue when calling
gst_rtsp_watch_set_flushing with flushing variabel is TRUE.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728772
 2014-04-30 16:37:17 +02:00
Tim-Philipp Müller bcb8068e27 docs: remove outdated and pointless 'Last reviewed' lines from docs 
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
 2014-04-26 23:28:57 +01:00
Wim Taymans 8d439edd7a rtspconnection: add flush method 
Add a method to set/unset the flushing state that makes _wait_backlog()
unlock.

See https://bugzilla.gnome.org/show_bug.cgi?id=725898
 2014-03-28 09:34:33 +01:00
Wim Taymans 183e441d88 rtsptransport: UDP is also default for SAVP and AVPF 2014-03-25 11:07:34 +01:00
Ognyan Tonchev d7857325c5 rtspconnection: Fix minor memory leaks in error handling 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726642
 2014-03-24 12:45:14 +01:00
Ognyan Tonchev e0af857445 rtspconnection: Fix connection_poll() 
* Only check for conditions we are interested in.
* Makes no sense to specify G_IO_ERR and G_IO_HUP in condition, they
  will always be reported if they are true.
* Do not create timed source if timeout is NULL.
* Correctly wait for sources to be dispatched, context_iteration() is
  not guaranteed to always block even if set to do so.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726641
 2014-03-24 12:43:38 +01:00
Руслан Ижбулатов d6bd37460a rtspconnection: Silence a compiler warning 
Cast the argument into (const char *) on W32, as winsock2 expects it.

https://bugzilla.gnome.org/show_bug.cgi?id=726433
 2014-03-16 11:22:04 +01:00
Göran Jönsson 0b30fdbfbe rtspconnection: gst_rtsp_watch_wait_backlog 
New method that wait until there is room in backlog queue.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725898
 2014-03-10 17:28:40 +01:00
David Svensson Fors 6cd0d10d30 rtspconnection: GstRTSPWatch func for tunnel GET response 
Add a callback in GstRTSPWatch where the response to HTTP GET for
tunneled connections can be modified.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725878
 2014-03-10 10:43:03 +01:00
Wim Taymans 4898c30537 rtspdefs: add RFC 4567 headers and status code 
This new Header and status code is used for SRTP
 2014-03-10 10:33:28 +01:00
Ognyan Tonchev 4220442441 rtspconnection: Call closed() when GET is closed in tunneled mode 
This patch adds read source on the write socket in tunneled
mode and we get a callback when client disconnects the GET
channel.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725313
 2014-03-03 10:34:56 +01:00
Sebastian Rasmussen 35bb1b3328 docs: Add annotations for return values 
Rephrase and clarify some return value descriptions

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521
 2014-03-02 23:41:18 +00:00
Sebastian Rasmussen 5b4f2ba20b docs: Fix argument and annotation typos 
* colorbalance: Fix misspelled annotation
 * rtsp: Replace incorrectly documented function argument
 * sdp: Escape @ character to avoid gtk-doc warning
 * video-*: Add missing annotation colon
 * videodecoder/video-color: Fix function argument typos
 * videoutils: Remove unknown annotation field

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725521
 2014-03-02 23:22:51 +00:00
Tim-Philipp Müller 14b82bbc9a rtsp: fix build with older GLib versions 
The gio/gnetworking.h header is only available since glib 2.36

https://bugzilla.gnome.org/show_bug.cgi?id=725206
 2014-02-26 11:44:18 +00:00
Ognyan Tonchev 5445682c6a rtspconnection: Add missing include 
https://bugzilla.gnome.org/show_bug.cgi?id=725206
 2014-02-26 11:25:13 +00:00
Ognyan Tonchev ebe3530f51 rtspconnection: Remove read child source when POST is disconnected 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=724720
 2014-02-21 16:21:45 +01:00
Aleix Conchillo Flaqué 0a115bd31f rtspconnection: allow specifying a certificate database 
Two new functions have been added,
gst_rtsp_connection_set_tls_database() and
gst_rtsp_connection_get_tls_database(). The certificate database will be
used when a certificate can't be verified with the default database.

https://bugzilla.gnome.org/show_bug.cgi?id=724393
 2014-02-19 21:48:13 +01:00
Aleix Conchillo Flaqué 9121b16aa0 rtspconnection: get rid of superfluous whitespaces 2014-02-19 21:22:30 +01:00
Wim Taymans 594dd4287b rtsptransport: calculate default lower transport 
Add an internal method to calculate the default lower transport whan it
is missing.
 2014-01-07 14:51:46 +01:00
Wim Taymans 124cf22d5d rtsptransport: add method to get media-type from transport 
Add a method to make a media-type from the transport. Deprecate the old
method that only used the mode.

Based on patch from Aleix Conchillo Flaqué <aleix@oblong.com>

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720219
 2014-01-07 14:51:37 +01:00
Wim Taymans 5b13c5b464 rtsptransport: add GType for Profile 
See https://bugzilla.gnome.org/show_bug.cgi?id=720696
 2014-01-07 11:52:27 +01:00
Wim Taymans 01c7fb11ba rtsptransport: add more profiles 
Add support for Feedback profiles
 2013-12-26 17:41:00 +01:00
Tim-Philipp Müller 4af1e064fe docs: cosmetic since marker fixes 2013-11-16 16:10:06 +00:00
Sebastian Dröge b0aad9dd84 rtspconnection: Fix indention in header 2013-11-01 16:43:56 +01:00
Aleix Conchillo Flaque 53c7ad0c87 rtspconnection: allow setting tls certificate validation 
Added new functions gst_rtsp_connection_set_tls_validation_flags() to
allow setting the TLS certificate validation flags when establishing a
TLS connection.
A getter is also available, gst_rtsp_connection_get_tls_validation_flags().

https://bugzilla.gnome.org/show_bug.cgi?id=711231
 2013-11-01 16:42:34 +01:00
Hans Månsson 6bb58eec8a rtspconnection: Connect to proxy if specified 
Reference: https://bugzilla.gnome.org/show_bug.cgi?id=708880
 2013-10-04 07:27:12 +02:00
Ognyan Tonchev 02ac18b699 rtspconnection: Unset input/output_stream after freeing the GIOStream 
watch->input_stream and watch->output_stream are owned by the GIOStream
and should be unset after freeing the stream.

https://bugzilla.gnome.org/show_bug.cgi?id=708689
 2013-09-24 18:35:14 +02:00
Ognyan Tonchev 8ba90931ae rtspconnection: Only create writesrc when it is actually needed 
Creating a GSource and not attaching it to a context will cause
a leak of it's child sources. That is why we create writesrc right
before attaching it to a context.

https://bugzilla.gnome.org/show_bug.cgi?id=708667
 2013-09-24 12:10:00 +02:00
Tim-Philipp Müller c449ae6343 rtsp: fix direct includes 
https://bugzilla.gnome.org/show_bug.cgi?id=695889
 2013-08-16 14:14:22 +01:00
Sebastian Dröge c6f8220920 rtspconnection: Create a new write GSource after removing it 
After removal, a GSource is destroyed and can never be attached
again to a main context. We need to create a new one instead.

https://bugzilla.gnome.org/show_bug.cgi?id=704198
 2013-07-14 18:11:59 +02:00
Wim Taymans 32a1deb404 rtsp: make read uncancelable when reading a message 
When we start to read a message, we need to continue reading until the end of
the message or else we lose track and cause parse errors. Use a variable
may_cancel to avoid cancelation after we read the first byte until we have
the complete message.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703088
 2013-06-26 15:06:00 +02:00
Wim Taymans bcc5ac5298 rtsp: dispatch when initial buffer has data 
When we have data in the inital buffer, dispath the read function to read it
even if the socket has no data to read.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702652
 2013-06-21 11:50:33 +02:00
Wim Taymans ad6c16fdfc rtsp: manage writer child source better 
Only add the write child source when we have something to write or else
we will dispatch forever without doing anything.
 2013-06-20 17:28:46 +02:00
Sebastian Dröge 567be29db2 rtspconnection: Make sure to set a sensible default port for the GSocketConnection 
Otherwise it will connect to port 0 if no port is given in the URI.

https://bugzilla.gnome.org/show_bug.cgi?id=701798
 2013-06-10 15:31:38 +02:00
Brendan Long 63961242df rtspconnection: remove functions added in GLib 2.34 
g_pollable_stream_read and g_pollable_stream_write were added in GLib 2.34,
but Ubuntu 12.04 and Debian Wheezy still use GLib 2.32.

Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=701316
 2013-05-31 14:12:10 +02:00
Wim Taymans 0b933ff87b rtsp: add method to get the TLS connection 2013-05-30 17:31:13 +02:00
Wim Taymans c0f13c2513 rtsp: let the sockets be reffed by the connection 
Don't add an extra ref to the sockets but use that of the connection.
Keep the connection around as an IOStream.
 2013-05-30 13:14:46 +02:00
Wim Taymans 2fc85d3980 rtsp: Cleanup the error path 
Make sure the watch is removed when we close the read socket because of
an error.
 2013-05-30 10:50:42 +02:00
Wim Taymans ad5632586a rtsp: cleanup the watch reset function 2013-05-30 10:45:42 +02:00
Wim Taymans 07babdd68a rtsp: check if the streams are still active 
Don't try to read/write from an inactive stream. When we, for example,
transfer the second connection in tunneling mode, we are not interested anymore
on read/write activity on the old connection.
 2013-05-30 10:30:09 +02:00
Wim Taymans d09028b4c3 rtsp: use child sources instead of using the sockets 
Use the source of the pollable input/output streams instead of
accessing the sockets directly.
 2013-05-30 07:36:52 +02:00
Wim Taymans 4ada677095 rtsp: fix input/output streams for tunneling 2013-05-30 07:35:18 +02:00
Wim Taymans 4f660c388c rtsp: don't use sockets for blocking 
Use the blocking and non-blocking API of the input/output streams instead
of polling the sockets directly. This also allows us to simplify some
code.
 2013-05-30 07:35:18 +02:00
Wim Taymans 909e119a23 rtsp: add TLS support 
Add flag to select TLS in the transport.
Enable TLS on the socketclient when we use a TLS uri.
 2013-05-30 07:35:14 +02:00
Wim Taymans 057bbae6c5 rtspconnection: use the input/output stream of clientconnection 
Don't use the raw sockets for RTSP communication but use the IOStream.
This is needed if we are going to use TLS later.
 2013-05-30 07:20:51 +02:00
Wim Taymans 2d41ee370c rtsp: set sockets non-blocking 2013-05-30 07:20:51 +02:00
Wim Taymans a42a7be5df rtsp: use GSocketClient for making connections 
Use the GSocketClient API for making connections with the server. This removes a
bit of code and gives us the ability to do TLS later.
 2013-05-30 07:20:51 +02:00
Wim Taymans 15f3c995aa Revert "rtspconnection: Use a GSocketAddressNumerator to resolve the addresses" 
This reverts commit 15a0bb0a10.

We should be using GSocketClient
 2013-05-30 07:20:51 +02:00