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356 commits

Author SHA1 Message Date
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
Sebastian Dröge
15a0bb0a10 rtspconnection: Use a GSocketAddressNumerator to resolve the addresses 
Instead of just trying the first possible resolution we're trying all
resolutions until one works.
 2013-05-27 14:53:48 +02:00
Thomas Scheuermann
9a78542ded rtsp: Don't use / as path if no path was provided 
RTSP does not mandate that a non-zero-length path is used and
some devices (e.g. IQinVision IQeye 1080p) requires that a
zero-length path is used.
 2013-04-08 09:09:33 +02:00
Wim Taymans
a4e44df6b9 rtsp: make local_ip and remote_ip variables 
Separate local_ip and remote_ip into separate variables for clarity.
 2013-04-04 12:32:24 +02:00
Wim Taymans
4826ec4e4d rtsp: calculate the local ip address in accept 
Calculate the local IP address in the accept call. We need to place this IP
address in the GET reply in the X-Server-IP-Address header so that the client
knows where to send the POST to in case of tunneled RTSP. Before this patch
it used the client IP address, which would make the client send the POST request
to itself and fail.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=697092
 2013-04-04 12:16:47 +02:00
David Svensson Fors
5ef9921bcd rtsprange: use gst_util_gdouble_to_guint64 in get_seconds 
https://bugzilla.gnome.org/show_bug.cgi?id=696818
 2013-04-02 14:33:51 -04:00
Emanuele Aina
f05a95ea3c build: Link libgstrtsp-1.0.so to libm for pow() 
https://bugzilla.gnome.org/show_bug.cgi?id=695658
 2013-03-11 19:30:13 -04:00
Olivier Crête
17d5dbd337 rtsprange: Add function to convert a range between formats 
Also add unit tests.
 2013-03-11 10:41:31 +01:00
Olivier Crête
0353e608f8 rtsprange: Make _to_string() be more in line with RFC 2326 
Fix various nits to make it more in line with the RFC, also add unit tests.
 2013-03-11 10:41:25 +01:00
Olivier Crête
3cfec4de73 rtsprange: Avoid going through fractions for large numbers 
If the number of seconds exceeds 2^31, then it will be truncated if the
conversion is done using fractions, so multiply it directly.
 2013-03-11 10:41:17 +01:00
Olivier Crête
203c27b42b rtsprange: Fix conversion from UTC to GstClockTime 
Do the difference in the right direction.
 2013-03-11 10:41:09 +01:00
Olivier Crête
aef8de337c rtspconnection: Add API to disable session ID caching in the connection 
This is necessary to allow having more than one session in the same connection.

API: gst_rtsp_connection_set_remember_session_id()
API: gst_rtsp_connection_get_remember_session_id()
 2013-03-11 10:41:00 +01:00
Tim-Philipp Müller
664adc6e19 gst-libs: 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.
 2013-01-16 10:16:27 +00:00
Wim Taymans
65c5ecd270 rtspconnection: add limit to queued messages 
Add a limit to the amount of queued bytes or messages we allow on the watch.

API: GstRTSPConnection::gst_rtsp_watch_set_send_backlog()
API: GstRTSPConnection::gst_rtsp_watch_get_send_backlog()
 2012-12-14 11:36:58 +01:00
Sebastian Dröge
3f82e919dd libs: Use foo/foo.h as single-include header consistently everywhere 
https://bugzilla.gnome.org/show_bug.cgi?id=688785
 2012-12-12 17:13:10 +00:00
Sebastian Rasmussen
d4b6f3c1a0 rtspmessage: Add several missing g-i annotations 
https://bugzilla.gnome.org/show_bug.cgi?id=689873
 2012-12-10 10:58:12 +01:00
Wim Taymans
b511f938d4 rtsp: add method to parse options list 2012-11-27 11:15:34 +01:00
Wim Taymans
ce904ec551 rtsprange: add string conversion for new formats 2012-11-21 16:25:24 +01:00
Wim Taymans
fdf904db32 rtsprange: add method to convert ranges to GstClockTime 
Add a method to convert the values of GstRTSPRange to GstClockTime.
Add unit tests for the conversions.

API: gst_rtsp_range_get_times()
 2012-11-21 15:35:46 +01:00
Wim Taymans
f1669d7d9c range: don't overwrite unit field 2012-11-21 15:29:05 +01:00
Wim Taymans
0bf50cd3d8 range: add g_return_if check 2012-11-21 15:29:05 +01:00
Evan Nemerson
4d77fba46c libs: Add missing single include headers and use them in GIRs 2012-11-21 11:01:24 +01:00
Wim Taymans
a87cd40f49 rtsprange: improve docs 2012-11-21 10:25:51 +01:00
Wim Taymans
b785c66098 rtsp: avoid ABI break 
Move new fields into structures appended at the end of the GstRTSPRange
to avoid ABI break.
 2012-11-20 11:13:01 +01:00
Wim Taymans
41d36b2584 rtsp: fix format string 2012-11-19 17:08:38 +01:00
Wim Taymans
fe4b415f98 rtsp: parse UTC ranges 2012-11-19 16:59:48 +01:00
Wim Taymans
b113f9697a rtsp: parse SMPTE ranges 2012-11-19 16:15:46 +01:00
Wim Taymans
02a5940a45 range: handle parse errors better 2012-11-19 16:13:56 +01:00
Wim Taymans
84b1ee4987 rtsp: detect npt time parse errors 2012-11-19 16:04:01 +01:00
Wim Taymans
25580430b0 range: a single - is not allowed 2012-11-19 13:56:53 +01:00
Wim Taymans
db7ea32f35 range: handle ranges starting with - 
An RTSP range that starts with a - means that the first value of the range is
the end of the stream.
 2012-11-19 13:56:53 +01:00
Wim Taymans
6313e5f1af rtspconnection: improve docs 2012-11-12 14:18:00 +01:00
Ognyan Tonchev
f67c6a768b rtsp: fix g-i annotation for gst_rtsp_message_set_body(), take_body() and take_header() 
https://bugzilla.gnome.org/show_bug.cgi?id=687620
 2012-11-09 21:24:12 +00:00
Ognyan Tonchev
6318a4602a rtsp: fix GstRTSPMessage g-i annotations for out parameters 
https://bugzilla.gnome.org/show_bug.cgi?id=687620
 2012-11-05 13:21:39 +00:00
Tim-Philipp Müller
5f59b4f7ee Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-03 23:05:09 +00:00
Miguel Angel Cabrera Moya
4b083d608e rtspconnection: remove extra return and fix GError leak 
https://bugzilla.gnome.org/show_bug.cgi?id=687473
 2012-11-02 19:30:23 +00:00
Ognyan Tonchev
ff04a1b4c6 rtspconnection: fix g-i annotations for out parameters 
https://bugzilla.gnome.org/show_bug.cgi?id=687421
 2012-11-02 12:43:52 +00:00
Tim-Philipp Müller
a4f2df6341 Revert "g-i: change g-ir-scanner arg --library=libgstfoo-X.la to --library=gstfoo-X" 
This reverts commit e39fbe6b7e.

Looks like we need to pass the full .la file after all in a setup
with libtool, or it might not find the library, e.g. like

  ERROR: can't resolve libraries to shared libraries: gstfft-1.0

Conflicts:
	gst-libs/gst/audio/Makefile.am
	gst-libs/gst/pbutils/Makefile.am

Also see https://bugzilla.gnome.org/show_bug.cgi?id=603710
 2012-10-29 12:47:05 +00:00
Tim-Philipp Müller
e39fbe6b7e g-i: change g-ir-scanner arg --library=libgstfoo-X.la to --library=gstfoo-X 
As it should be according to the man page.

https://bugzilla.gnome.org/show_bug.cgi?id=679315
 2012-10-28 17:35:57 +00:00
Ognyan Tonchev
6e5ea441e7 rtsp: Don't use invalid sockets 
return false from dispatch () if the read and write sockets have been
unset in tunnel_complete ()

Setting up HTTP tunnels causes segfaults since the watch for the second
connection is not destroyed anymore in tunnel_complete () and the connection
will still be used even though it is not valid anymore.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=686276
 2012-10-25 17:59:47 +02:00
Tim-Philipp Müller
336842d35c rtsprange: fix formatting and parsing of range floating-point values 
Other locales might use a comma instead of a floating point
for floats, which might lead to parsing errors.

https://bugzilla.gnome.org/show_bug.cgi?id=684411
 2012-10-13 00:19:54 +01:00
Sebastian Pölsterl
e8fed7f04b rtsp: mark url argument of gst_rtsp_url_parse() as out arg 
https://bugzilla.gnome.org/show_bug.cgi?id=685242
 2012-10-01 22:36:06 +01:00
Tim-Philipp Müller
5e0dfec62c Remove -DGST_USE_UNSTABLE_API 2012-09-17 16:05:37 +01:00
Thibault Saunier
91cdd763eb rtsp: port to the new GLib thread API 2012-09-09 20:41:06 -03:00
Tim-Philipp Müller
2079a8c12b Remove glib-compat-private.h stuff we don't need any more 
It's all been ported to the latest GLib API now.
 2012-09-09 18:36:49 +01:00
Marc Leeman
791163aba2 gst-rtsptransports: no warning Transport end with semicolumn 2012-07-24 12:49:29 +02:00
Edward Hervey
2817bdadc9 libs: Remove "Since" markers and minor doc fixups 2012-07-13 12:11:06 +02:00
Ognyan Tonchev
de9aeb0c72 rtsp: Update the initial_buffer when merging RTSP Connections 
See https://bugzilla.gnome.org/show_bug.cgi?id=679337
 2012-07-10 11:34:47 +02:00
Wim Taymans
90b3f525e9 rtspconnection: handle cancellation correctly 2012-06-06 16:41:03 +02:00
David Svensson Fors
0b0dde7ce1 rtsp: don't leak address and socket 
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=677466
 2012-06-06 14:53:43 +02:00
Wim Taymans
b0cc0a31e2 rtsp: unref sockets in _close 
When closing the connection, unref the currently used sockets. This should close
them when not in use. We need to do this because else we cannot reconnect
anymore after a close, the connect function requires that the sockets are NULL.
 2012-05-18 09:47:26 +02:00
Wim Taymans
2cd15bbef8 rtsp: clear the GError for pending connect 
Clear the GError after g_socket_connect tells us that the connection is pending.
If we don't do this, glib complains when we try to reuse the non-NULL GError
variable a little below.
 2012-05-18 09:47:26 +02:00
Sebastian Rasmussen
b7b123964b gst-libs: make pkg-config get path to pkg-config dirs from configure 
When --with-pkg-config-path is supplied to configure this path is now
explicitly propagated to pkg-config.

https://bugzilla.gnome.org/show_bug.cgi?id=673377
 2012-05-05 23:26:20 +01:00
Sebastian Dröge
65307dd132 gst: Update versioning 2012-04-04 14:55:15 +02:00
Wim Taymans
26f63027a6 rtsp: fix connection 2012-02-20 17:44:59 +01:00
Wim Taymans
268d52fd33 Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/rtsp/gstrtspconnection.c
	win32/common/libgstaudio.def
 2012-02-17 23:46:17 +01:00
Ognyan Tonchev
f6e07b65a4 rtspconnection: only send new data immediately if there are no queued messages 
Even if watch->messages->length is 0 there may still be some
data from a message that was only written partially at the
previous attempt stored in watch->write_data, so check for
that as well. We don't want to write data into the middle
of another message, which could happen when there wasn't
enough bandwidth.

https://bugzilla.gnome.org/show_bug.cgi?id=669039
 2012-02-17 14:40:35 +00:00
Tim-Philipp Müller
bd4bf43171 rtsp: make g-ir-scanner include Gio-2.0 to suppress complaints about GSocket etc. 2012-02-07 23:42:48 +00:00
Olivier Crête
e391118125 Use macros to register boxed types thread safely 2012-01-28 14:53:21 +00:00
Sebastian Dröge
aed2666b53 rtsp: Port to GIO 2012-01-17 16:38:45 +01:00
Sebastian Dröge
dc8984d76c Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/app/gstappsrc.c
	gst-libs/gst/audio/multichannel.h
	gst-libs/gst/video/videooverlay.c
	gst/playback/gstplaysink.c
	gst/playback/gststreamsynchronizer.c
	tests/check/Makefile.am
	win32/common/libgstvideo.def
 2012-01-10 13:15:12 +01:00
Tim-Philipp Müller
9f042ae224 rtspconnection: make hostname lookup more thread-safe 
Don't write IP number string to return into a static
array which is shared amongst all threads (note: of
course a copy is returned).

https://bugzilla.gnome.org/show_bug.cgi?id=666711
 2012-01-07 20:16:41 +00:00
Tim-Philipp Müller
c3e6e23b85 audio, rtsp: remove private/protected gtk-doc markup for enums 
This confuses glib-mkenums, and is not really useful anyway.

https://bugzilla.gnome.org/show_bug.cgi?id=666618
 2012-01-02 00:19:57 +00:00
Wim Taymans
59d5ad42b0 rtsp: use rtpbin 2011-12-09 19:22:21 +01:00
Tim-Philipp Müller
fb6d09055a Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	ext/alsa/gstalsadeviceprobe.c
	ext/alsa/gstalsamixer.c
	ext/pango/gsttextoverlay.c
	ext/pango/gsttextoverlay.h
	gst-libs/gst/audio/gstaudiobasesink.c
	gst-libs/gst/audio/gstaudioringbuffer.c
	gst-libs/gst/audio/gstaudiosrc.c
	gst-libs/gst/video/Makefile.am
	gst-libs/gst/video/video.c
	gst/encoding/gststreamcombiner.c
	gst/encoding/gststreamsplitter.c
	gst/playback/gstplaybasebin.c
	gst/playback/gststreamsynchronizer.c
	gst/playback/gstsubtitleoverlay.c
	gst/playback/gsturidecodebin.c
	sys/xvimage/xvimagesink.c
	tests/examples/Makefile.am
	win32/common/libgstvideo.def

Video overlay composition disabled for now, needs
porting to buffer meta.
 2011-12-08 01:19:03 +00:00
Tim-Philipp Müller
0d98aa25b8 Work around deprecated thread API in glib master 
Add private replacements for deprecated functions such as
g_mutex_new(), g_mutex_free(), g_cond_new() etc., mostly
to avoid the deprecation warnings. We'll change these
over to the new API once we depend on glib >= 2.32.

Replace g_thread_create() with g_thread_try_new().
 2011-12-04 17:16:30 +00:00
Tim-Philipp Müller
177525f89f Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	gst-libs/gst/netbuffer/gstnetbuffer.c
	gst/ffmpegcolorspace/avcodec.h
	gst/ffmpegcolorspace/gstffmpegcodecmap.c
	gst/ffmpegcolorspace/imgconvert.c
	gst/ffmpegcolorspace/imgconvert_template.h
	gst/ffmpegcolorspace/mem.c
	gst/playback/README
	gst/playback/gstplaybasebin.c
	gst/playback/gstplaybasebin.h
	gst/playback/gstplaybin.c
	sys/v4l/v4lmjpegsrc_calls.c
	sys/v4l/videodev_mjpeg.h
	tests/check/elements/gnomevfssink.c
 2011-12-02 11:10:17 +00:00
Piotr Fusik
14644457b0 various: typo fixes 
Fix typos in code and docs. Fixes. #658984
 2011-12-02 12:03:27 +01:00
Edward Hervey
d94535832b gst-libs: Add --warn-all to introspection scanner 
And let's get fixing those docs :)
 2011-11-25 10:31:38 +01:00
Wim Taymans
fc04bcecbe fix docs 2011-11-14 10:46:56 +01:00
Wim Taymans
bdf3845498 rtsp: cleanup headers 
Add padding, fix indentation, remove deprecated stuff
 2011-11-11 19:35:33 +01:00
Wim Taymans
ad8f694ec6 remove bogus files 
They got somehow commited in 7012e88090
 2011-11-11 10:39:52 +01:00
Wim Taymans
ace51b689f rtsp: remove deprecated base64 library 2011-11-10 17:39:10 +01:00
Stefan Sauer
53d7d2e966 interfaces: clean up the use of iface and class/klass 2011-10-21 14:46:48 +02:00
Edward Hervey
17bfba09f1 Merge branch 'master' into 0.11 
Conflicts:
	ext/ogg/gstoggdemux.c
	ext/pango/gsttextoverlay.c
	gst-libs/gst/audio/gstaudioencoder.c
	gst-libs/gst/audio/gstbaseaudiosrc.c
	gst/playback/gstsubtitleoverlay.c
	gst/videorate/gstvideorate.c
 2011-09-23 18:27:11 +02:00
Mark Nauwelaerts
e574f58e71 rtspdefs: add RTCP-Interval header 2011-09-19 11:32:23 +02:00
Wim Taymans
7012e88090 Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/audio/audio.h
	gst-libs/gst/audio/gstaudiodecoder.c
	gst-libs/gst/audio/gstaudiodecoder.h
	gst-libs/gst/audio/gstaudioencoder.c
	gst-libs/gst/audio/gstbaseaudioencoder.h
	gst/playback/Makefile.am
	gst/playback/gstplaybin.c
	gst/playback/gstplaysink.c
	gst/playback/gstplaysinkvideoconvert.c
	gst/playback/gstsubtitleoverlay.c
	gst/videorate/gstvideorate.c
	gst/videoscale/gstvideoscale.c
	win32/common/libgstaudio.def
 2011-09-06 15:24:32 +02:00
Wim Taymans
3fab57b5cf Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/interfaces/videooverlay.c
	gst-libs/gst/rtp/gstrtpbuffer.c
	po/af.po
	po/az.po
	po/bg.po
	po/ca.po
	po/cs.po
	po/da.po
	po/de.po
	po/el.po
	po/en_GB.po
	po/es.po
	po/eu.po
	po/fi.po
	po/fr.po
	po/gl.po
	po/hu.po
	po/id.po
	po/it.po
	po/ja.po
	po/lt.po
	po/lv.po
	po/nb.po
	po/nl.po
	po/or.po
	po/pl.po
	po/pt_BR.po
	po/ro.po
	po/ru.po
	po/sk.po
	po/sl.po
	po/sq.po
	po/sr.po
	po/sv.po
	po/tr.po
	po/uk.po
	po/vi.po
	po/zh_CN.po
 2011-08-22 13:06:27 +02:00
Stefan Kost
01bbdd6bdf docs: handle warnings emitted by gtk-doc 
This is useful and in most cases someone had put arbitrary markup into the docs,
misspelled xref'ed symbols, forgot to add stuff to the docs etc..
 2011-08-20 19:16:42 +02:00
Wim Taymans
33467d9629 Merge branch 'master' into 0.11 
Conflicts:
	configure.ac
	ext/pango/gsttextoverlay.c
	ext/theora/gsttheoradec.c
	gst/adder/gstadder.c
	gst/adder/gstadder.h
	gst/audioresample/gstaudioresample.c
	gst/encoding/gstencodebin.c
	gst/playback/gstdecodebin.c
	gst/playback/gstdecodebin2.c
	tests/check/elements/decodebin2.c
	tests/check/elements/playbin-compressed.c
	win32/common/libgsttag.def
 2011-08-16 18:01:14 +02:00
Alessandro Decina
22cc529409 rtspconnection: add OSX specific hack to detect when a connection is refused 
Unlike linux, OSX wakes up select with POLLOUT (instead of POLLERR) when
connect() is done async and the connection is refused. Therefore always check
for the socket error state using getsockopt (..., SO_ERROR, ...) after a
connection attempt.
 2011-08-15 23:46:53 +02:00
Tim-Philipp Müller
4bf26ba5d2 Add -DGST_USE_UNSTABLE_API to the compiler flags to avoid warnings 2011-07-05 10:07:08 +01:00
Tim-Philipp Müller
d77991106b rtsp: GstRTSPExtension isn't wrapped by GstImplementsInterface 
Fix copy'n'paste error in headers, GstRTSPExtension isn't
something that's per-instance.
 2011-06-26 21:07:52 +01:00
Stefan Kost
8ca5d1274b docs: add minimal docblobs for status code and headers 
Use a trick to avoid documenting all 100 enums.
 2011-05-23 23:56:09 +03:00
Edward Hervey
66016eedc7 rtsp: Fix typo which broke the build 2011-05-17 10:20:36 +02:00
Miguel Angel Cabrera Moya
30b2abaddd rtspconnection: not enter in not controllable state unless it is necessary 
When closing rtspsrc the state change blocks until the polling in the
connection timeouts. This is because the second time we loop to read a
full message controllable is set to FALSE in the poll group, even though no
message is half read.
This can be avoided by not setting controllable to FALSE the poll group
unless we had begin to read a message.

Fixes #610916
 2011-05-17 09:29:47 +02:00