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Author SHA1 Message Date
Tim-Philipp Müller
9ce663f04d flacenc: audio caps have a *list* of formats, not an array of formats 
A list of things in caps is something where one is picked in the
course of negotiation. An array is always something that only makes
sense as a whole in that order.
 2012-02-22 17:03:42 +00:00
Mark Nauwelaerts
38516ad367 flacenc: remove post-port bogus _unref 2012-02-22 18:03:11 +01:00
Tim-Philipp Müller
b7e96ebe37 flacenc: remove bogus pad locking that causes deadlocks 
It's not clear why the pad object lock is taken here. But
gst_pad_{has,get}_current_caps() will try to take the lock
as well and deadlock, since it's not recursive.
 2012-02-22 17:00:19 +00:00
Tim-Philipp Müller
9c5c33790a flacenc: set right number of channels on caps in get_caps function 2012-02-22 16:59:42 +00:00
Vincent Untz
a32e030faa pulse: Fix a build warning when compiling with asserts disabled 
Return a value even if the code will never be reached, to make compilers
happy.

https://bugzilla.gnome.org/show_bug.cgi?id=670561
 2012-02-21 20:12:06 +00:00
Edward Hervey
98dd495272 id3v2mux: Fix merge error 2012-02-21 10:40:00 +01:00
Mark Nauwelaerts
594867fe37 shout2: use some more boilerplate 2012-02-16 23:35:11 +01:00
Mark Nauwelaerts
834c63f87e shout2: port to 0.11 2012-02-16 23:35:11 +01:00
Tim-Philipp Müller
91a58ccda1 Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	ext/taglib/gstapev2mux.cc
	ext/taglib/gstid3v2mux.cc
	ext/taglib/gsttaglibmux.c
	ext/taglib/gsttaglibmux.h
 2012-02-12 17:03:37 +00:00
Tim-Philipp Müller
736f68e1e6 taglib: port to GstTagMux base class 2012-02-12 16:22:21 +00:00
Vincent Penquerc'h
8620c16b27 taglib: finish off a few missed variable changes 
Local variables are now unused, and the values from the segment copy
are used instead, so remove the now useless local variables and write
to the segment where appropriate.
 2012-02-12 12:24:50 +00:00
Wim Taymans
225e98d623 Merge branch 'master' into 0.11 
Conflicts:
	ext/flac/gstflacenc.c
	ext/jack/gstjackaudioclient.c
	ext/jack/gstjackaudiosink.c
	ext/jack/gstjackaudiosrc.c
	ext/pulse/plugin.c
	ext/shout2/gstshout2.c
	gst/matroska/matroska-mux.c
	gst/rtp/gstrtph264pay.c
 2012-02-10 16:23:14 +01:00
Mark Nauwelaerts
abc8c162ed flacdec: shift in proper direction for audio sample conversion 2012-02-09 22:09:31 +01:00
Wim Taymans
9365f12d6e GST_FLOW_WRONG_STATE -> GST_FLOW_FLUSHING 2012-02-08 16:43:30 +01:00
Ralph Giles
8b2ca70124 shout2send: send video/webm through libshout. 
This requires SHOUT_FORMAT_WEBM, added in libshout 2.3.0,
so video/webm support is contingent on that symbol being
defined.

Also an indentation change required by the pre-commit hook.

https://bugzilla.gnome.org/show_bug.cgi?id=669590
 2012-02-08 10:50:18 +00:00
Stefan Sauer
7a9970fc29 jack: rework transport support 
Move common code to jackclient. There we can also handle the request state
message in a better way, as the element callbacks are only run if the element is
active.
 2012-02-07 21:59:29 +01:00
Wim Taymans
819865ea54 taglib: fix object registration 
We can't use G_DEFINE_TYPE because the class is not set in the class_init and we
need it to get the srcpad template.
Fix a caps leak
 2012-02-07 10:30:28 +01:00
Wim Taymans
88733ea2f8 soup: fix caps 2012-02-07 10:30:28 +01:00
Mark Nauwelaerts
9be8a6f48c dv: port to 0.11 2012-02-06 22:15:45 +01:00
Arun Raghavan
4e2cf393c0 pulseaudiosink: Lower rank to prevent autoplugging 
pulseaudiosink breaks visualisations in its current form, so let's
prevent it from being autoplugged for the time being.

The best we can hope to do in the 0.10 series is query the list of
available sinks and their formats, and expose these as the bin's sinkpad
caps. While this is not a comprehensive solution, it will make sure that
we're only trying to support compressed formats if we're certain that
one exists.

The long-term fix for this will be in the form of proper upstream
renegotiation support in the 0.11/1.0 series.

https://bugzilla.gnome.org/show_bug.cgi?id=666361
 2012-02-03 22:12:06 +05:30
Wim Taymans
78ed46cc4f cmmlenc: fix caps handling 2012-02-03 16:14:08 +01:00
Vincent Penquerc'h
5ff31d446e flacenc: fix event leak when there is no peer on the src pad 2012-02-03 14:53:31 +00:00
Mark Nauwelaerts
da0cdc7713 taglib: port to 0.11 2012-02-01 18:02:15 +01:00
Tim-Philipp Müller
0f3b7b010e build: ignore GValueArray deprecation warnings for the time being 
until this gets sorted out with the GLib folks and we have a
viable alternative.

https://bugzilla.gnome.org/show_bug.cgi?id=667228
 2012-02-01 16:40:51 +00:00
Tim-Philipp Müller
284ee0b84a pulse: disable some unused property probe code 
which was using GValueArray
 2012-02-01 16:36:53 +00:00
Sebastian Dröge
a67bd41d75 speex: Use new audio encoder/decoder base class API for srcpad caps 2012-02-01 16:27:47 +01:00
Sebastian Dröge
161229a384 flac: Use new audio encoder/decoder base class API for srcpad caps 2012-02-01 16:27:47 +01:00
Wim Taymans
bb2bd604e0 update for HEADER flag 2012-01-30 17:16:51 +01:00
Jayakrishnan M
d6821a336d cairo: fix build, make sure libgstvideo can be found 
https://bugzilla.gnome.org/show_bug.cgi?id=668648
 2012-01-25 15:21:44 +00:00
Sebastian Dröge
0b517ce9fb Merge branch '0.11' of ssh://git.freedesktop.org/git/gstreamer/gst-plugins-good into 0.11 2012-01-25 12:49:34 +01:00
Sebastian Dröge
10554b271f Merge branch 'master' into 0.11 
Conflicts:
	ext/flac/gstflacdec.c
	ext/jpeg/gstjpegenc.c
	ext/pulse/pulsesink.c
	sys/v4l2/gstv4l2src.c
 2012-01-25 12:49:11 +01:00
Wim Taymans
f87ab74add png: port to new memory API 2012-01-25 12:41:42 +01:00
Wim Taymans
b4630dd3e0 more memory API porting 2012-01-25 12:30:29 +01:00
Mark Nauwelaerts
a31a9e1f33 pngenc: disably snapshot behaviour by default 
... since such behaviour is not consistent, if allowable at all.
 2012-01-24 18:25:04 +01:00
Mark Nauwelaerts
2fcb5fa05b pngdec: port to 0.11 2012-01-24 18:25:04 +01:00
Mark Nauwelaerts
a1797459cb pngenc: port to 0.11 2012-01-24 18:25:04 +01:00
Tim-Philipp Müller
7cb9b7ab9d Use new GLib API unconditionally 2012-01-22 23:15:19 +00:00
Mark Nauwelaerts
1911812572 flacdec: improve upstream peer duration querying 
... to avoid accepting unhandled duration query result.
 2012-01-20 17:10:19 +01:00
Mark Nauwelaerts
e44d930289 pulsesrc: additional error condition checking 2012-01-20 17:10:17 +01:00
Mark Nauwelaerts
3168b77e04 pulsesink: additional error condition checking 2012-01-20 17:10:14 +01:00
Mark Nauwelaerts
ad11ec4121 jpegenc: check _alloc_buffer result and perform fallback alloc if needed 
... rather than carrying on with NULL buffer.
 2012-01-20 17:10:11 +01:00
Wim Taymans
b22c0dd3f6 update for memory API 2012-01-19 12:44:39 +01:00
Wim Taymans
1584806634 port to new gthread API 2012-01-19 11:33:53 +01:00
Vincent Penquerc'h
f0ac29113c pulsesrc: fix wrong error check 
pa_stream_* functions return negative on error, despite the defines
for error codes being positive.

I only got to repro the error twice, so I'm not sure 100% sure this
fixes the issue (the negative var being uninitialized after returning
from pa_stream_get_latency).
 2012-01-13 18:11:36 +00:00
Tim-Philipp Müller
8580dd86c9 eqMerge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	ext/jack/gstjackaudiosink.c
	ext/jack/gstjackaudiosrc.c
	gst/matroska/matroska-mux.c
	gst/matroska/matroska-read-common.c
	gst/rtpmanager/gstrtpssrcdemux.c
 2012-01-12 23:48:50 +00:00
Vincent Penquerc'h
483514528a flacenc: do not drop the first data buffer on the floor (and leak it either) 2012-01-12 10:30:56 +00:00
Stefan Sauer
bc1fa747a7 jack: add a transport mode enum 
Clients can configure the desired behaviour via "transport" property. The
default behaviour is ignoring the transport state. Other modes are master and
slave.
 2012-01-11 14:52:14 +01:00
Sebastian Dröge
e3c8c4f8b0 souphttpsrc: Fix buffer handling 
souphttpsrc is now usable again and doesn't crash anymore
whenever something is read from a HTTP connection.
 2012-01-11 14:10:46 +01:00
Stefan Sauer
747e63f4e7 jack: deactivate the request_state code 
When qjackctl is started, transport is stopped by default. This would be a
regression for gstreamer apps that before just started to play right away.
 2012-01-10 23:02:45 +01:00
Stefan Sauer
7d4044aa46 jack: add transport control handling 
This feature allows to start and stop playback from other jack applications (e.g. qjackctl).
 2012-01-10 22:35:02 +01:00
Stefan Sauer
0280ab04ed jack: use jack type for the callback 
Jack headers have a typedef for the shutdown callback as well.
 2012-01-10 15:08:16 +01:00
Sebastian Dröge
93e3ed5a86 Merge branch 'master' into 0.11 
Conflicts:
	ext/cairo/gsttextoverlay.c
	ext/pulse/pulseaudiosink.c
	gst/audioparsers/gstaacparse.c
	gst/avi/gstavimux.c
	gst/flv/gstflvmux.c
	gst/interleave/interleave.c
	gst/isomp4/gstqtmux.c
	gst/matroska/matroska-demux.c
	gst/matroska/matroska-mux.c
	gst/matroska/matroska-mux.h
	gst/matroska/matroska-read-common.c
	gst/multifile/gstmultifilesink.c
	gst/multipart/multipartmux.c
	gst/shapewipe/gstshapewipe.c
	gst/smpte/gstsmpte.c
	gst/udp/gstmultiudpsink.c
	gst/videobox/gstvideobox.c
	gst/videocrop/gstaspectratiocrop.c
	gst/videomixer/videomixer.c
	gst/videomixer/videomixer2.c
	gst/wavparse/gstwavparse.c
	po/ja.po
	po/lv.po
	po/sr.po
	tests/check/Makefile.am
	tests/check/elements/qtmux.c
	tests/check/elements/rgvolume.c
 2012-01-10 14:32:32 +01:00
Sebastian Dröge
a22a566c0b flac: Port to the new raw audio caps 2012-01-06 09:40:55 +01:00
Sebastian Dröge
4b6a410be0 speex: Update for the new raw audio caps 2012-01-05 10:36:49 +01:00
Sebastian Dröge
42bdbbcb29 jack: Add the new layout field to the raw audio caps 2012-01-05 10:36:48 +01:00
Sebastian Dröge
531d611f83 jackaudiosrc: Port to the new multichannel audio caps 2012-01-05 10:36:45 +01:00
Sebastian Dröge
dc049d1f1f pulse: Port to the new multichannel caps 2012-01-05 10:30:30 +01:00
Wim Taymans
47a1da9076 GST_FLOW_UNEXPECTED -> GST_FLOW_EOS 2012-01-04 10:01:48 +01:00
Wim Taymans
5fd2b7abe3 GST_FLOW_UNEXPECTED -> GST_FLOW_EOS 2012-01-03 15:26:21 +01:00
Nicola Murino
7202d37c9d jpegdec: fix peer_caps leak 
https://bugzilla.gnome.org/show_bug.cgi?id=666688
 2011-12-26 15:34:47 -03:00
Tim-Philipp Müller
ff74718616 pulse: remove pulseaudiosink helper bin 
This is causing us lots of headaches in 0.10 and needs to be done
differently and properly in 0.11. playbin or decodebin should
reconfigure themselves based on reconfigure events, for example.
 2011-12-25 22:21:36 +00:00
Tim-Philipp Müller
2799bcd32e pulse: update for ring buffer audio format type enum rename 2011-12-25 21:45:45 +00:00
Wim Taymans
4b8975f867 update for removed property probe 2011-12-21 11:59:46 +01:00
Vincent Penquerc'h
cf344d50b1 cairotextoverlay: port to GstCollectPads2 2011-12-14 18:34:25 +00:00
Tim-Philipp Müller
b8b8454bcb Suppress deprecation warnings in selected files, for g_static_rec_mutex_* mostly 
GStaticRecMutex is part of our API/ABI, not much we can do here
in 0.10 for most of these.
 2011-12-12 09:46:27 +00:00
Tim-Philipp Müller
330d984288 Use g_thread_try_new() instead of g_thread_crate() with newer glib versions 2011-12-12 09:46:27 +00:00
Tim-Philipp Müller
66f6e12888 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.
 2011-12-12 09:46:27 +00:00
Tim-Philipp Müller
8aebe194aa gdkpixbufsink: fix inverted pixel-aspect-ratio 
Spotted by Mike Morrison.

https://bugzilla.gnome.org/show_bug.cgi?id=665882
 2011-12-11 18:40:31 +00:00
Tim-Philipp Müller
9c1095f474 pulseaudiosink: don't leak pad template 2011-12-11 17:55:14 +00:00
Tim-Philipp Müller
5bb4dcd89c soup: fix start/stop race in souphttpclientsink 
Fix crash or hang in generic/states unit test when doing stop()
right after start(). Create main loop in the start function already
and not just in the thread function, so that stop() always has a
valid main loop to quit on. Also, calling g_main_loop_quit() before
g_main_loop_run() won't work and result in the stop function waiting
for the thread to join forever. Therefore, wait for the thread to
be ready and get the main loop running in the start() function, to
be sure stop() always works.
 2011-12-11 17:24:20 +00:00
Tim-Philipp Müller
adb15bf34a pulse: rename "client" properties to "client-name" 
Better name, but also matches the property on the jack
elements (where "client" is used for something else).
 2011-12-09 16:04:56 +00:00
Tim-Philipp Müller
2e078fa556 jack: don't leak client name when freeing the element 
And add gtk-doc chunks for the new property.

https://bugzilla.gnome.org/show_bug.cgi?id=665872
 2011-12-09 15:50:28 +00:00
Nicolas Baron
92cfb335cd jack: add "client-name" property to jackaudiosink and jackaudiosrc 
https://bugzilla.gnome.org/show_bug.cgi?id=665872
 2011-12-09 15:45:03 +00:00
Wim Taymans
1538803ac4 update for basesink event handler changes 2011-12-02 22:25:17 +01:00
Wim Taymans
5bfc7b4bfe update for moved audio interfaces 2011-11-30 07:57:40 +01:00
Thiago Santos
1e6bd5ad57 Revert "pulseaudiosink: fix caps leak" 
This reverts commit d6a9de9e2a.

setcaps functions aren't supposed to take ownership of the caps passed
 2011-11-29 17:34:49 -03:00
Vincent Penquerc'h
c0e101e93f various: fix pad template leaks 
https://bugzilla.gnome.org/show_bug.cgi?id=662664
 2011-11-28 13:30:27 +00:00
Tim-Philipp Müller
eeaa9e0bbc pulseaudio: require pulseaudio >= 1.0 2011-11-26 13:54:22 +00:00
Tim-Philipp Müller
be0d6baac5 Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	ext/pulse/pulseaudiosink.c
	ext/pulse/pulsesrc.c
	gst/audioparsers/gstaacparse.c
	gst/audioparsers/gstamrparse.c
	gst/audioparsers/gstdcaparse.c
	gst/audioparsers/gstflacparse.c
	gst/effectv/gstradioac.c
	gst/effectv/gstradioac.h
	gst/effectv/gstripple.c

Some possible FIXMEs remaining in the audio parser getcaps functions.
 2011-11-26 13:34:10 +00:00
Arun Raghavan
1f4bb68794 pulsesrc: Implement GstStreamVolume interface 
PulseAudio 1.0 supports per-source-output volumes, and this exposes the
functionality via the GstStreamVolume interface.

When compiled against pre-1.0 PulseAudio, the interface is not
implemented, and the "volume" or "mute" properties are not available.
This bit of ugliness will go away when we can depend on PulseAudio 1.0
or greater.

https://bugzilla.gnome.org/show_bug.cgi?id=595055
 2011-11-25 22:30:41 +05:30
Arun Raghavan
8c6a548698 pulsesrc: Trivial comment copy-paste-o fix 2011-11-25 22:30:41 +05:30
Arun Raghavan
bdf95eb39b pulseaudiosink: Remove redundant code 2011-11-25 22:30:41 +05:30
Arun Raghavan
f6f1605468 pulseaudiosink: Clean up refcounting in event probe 
Makes sure we don't leak a refcount if the object is disposed before a
NEWSEGMENT turns up.
 2011-11-25 22:30:41 +05:30
Tim-Philipp Müller
2cfb92f253 souphttpsrc: get rid of iradio-* properties, post tags instead 2011-11-24 01:45:43 +00:00
Tim-Philipp Müller
3f7c432869 souphttpsrc: always send icecast request header, drop iradio-mode property 
Server should ignore unknown/unhandled headers..
 2011-11-24 01:41:34 +00:00
Wim Taymans
bb3fbfc18e pulseaudiosink: avoid endless caps loop 
Check if the caps are the same before adding a new probe. Because of reconfigure
events, upstreams sends multiple caps events.
 2011-11-23 09:26:17 +01:00
Tim-Philipp Müller
736a484129 More printf format warning fixes 2011-11-22 01:40:39 +00:00
Wim Taymans
b7aa7bca52 add parent to activate functions 2011-11-18 13:57:20 +01:00
Wim Taymans
07cc855b24 Merge branch 'master' into 0.11 
Conflicts:
	ext/speex/gstspeexenc.c
	gst/rtpmanager/rtpsession.c
 2011-11-17 17:17:11 +01:00
Wim Taymans
105650127e add parent to pad functions 2011-11-17 15:02:55 +01:00
Stefan Sauer
8643d1caaf collectpads: port API changes 2011-11-17 08:44:45 +01:00
Mark Nauwelaerts
7df8122322 speexenc: ensure to free allocated padded data 2011-11-16 19:08:05 +01:00
Mark Nauwelaerts
c0d86fd26f speexenc: reset tag setter interface when appropriate 2011-11-16 19:06:09 +01:00
Mark Nauwelaerts
413f445455 flacenc: reset tag setter interface when appropriate 2011-11-16 19:06:07 +01:00
Wim Taymans
6190312214 add parent to query function 2011-11-16 17:27:13 +01:00
Wim Taymans
e7918a5aba _query_peer_*() -> _peer_query_*() 2011-11-15 18:04:44 +01:00
Wim Taymans
04579335c4 _accept_caps() -> _query_accept_caps() 2011-11-15 18:04:44 +01:00
Wim Taymans
797523efbd _peer_get_caps() -> _peer_query_caps() 2011-11-15 18:04:44 +01:00
Wim Taymans
b2d508ac40 update for _get_caps() -> _query_caps() 2011-11-15 18:04:44 +01:00
Wim Taymans
75dc9634eb change getcaps to query 
Chain up event function in payloaders.
 2011-11-15 18:04:44 +01:00
Vincent Penquerc'h
8548b2c777 flacdec: fix spurious timestamp discontinuity 
We need to tell the base class that we're dropping buffers,
so it drops the input timestamps corresponding to these.
Otherwise, the first actual audio buffers we output will be
stamped with those - GST_CLOCK_TIMESTAMP_NONE. That mismatch
between input buffer count and output buffer count will stay
while playing. With enough headers and long enough buffer
durations, the sink will have played enough before receiving
the first valid timestamp (usually 0), and will trigger an
audible discontinuity.
 2011-11-15 13:36:15 +00:00
Tim-Philipp Müller
c27bbe4be2 Update for GstURIHandler get_protocols() changes 2011-11-13 23:44:44 +00:00
Tim-Philipp Müller
a150d1e734 soup, pushfile, rtsp, udp, v4l2: update for GstURIHandler API changes 2011-11-13 18:50:51 +00:00
Wim Taymans
b0ccc61ed3 Merge branch 'master' into 0.11 
Conflicts:
	ext/pulse/pulseaudiosink.c
 2011-11-11 19:24:27 +01:00
Thiago Santos
d6a9de9e2a pulseaudiosink: fix caps leak 2011-11-11 11:02:22 -03:00
Mark Nauwelaerts
37c8abcdbd pulsesink: do not leak clientname when setting up property 2011-11-11 14:59:04 +01:00
Arun Raghavan
6a8af50111 pulse: Chain up dispose() in pulseaudiosink 2011-11-11 18:05:35 +05:30
Wim Taymans
3d9d2c6c05 update for audiobase* rename 2011-11-11 12:01:17 +01:00
Wim Taymans
86e33bc46b audio: update for base class rename 2011-11-11 11:53:45 +01:00
Wim Taymans
9daea802fa fix for ringbuffer rename 2011-11-11 11:33:44 +01:00
Wim Taymans
1ad11e307a update for ringbuffer change 2011-11-11 11:24:00 +01:00
Stefan Sauer
9ce6c731c3 various: add missing includes 2011-11-10 23:09:23 +02:00
René Stadler
3293b88ea1 pulsesink: fix compilation with pulseaudio 0.9 2011-11-10 21:37:38 +01:00
Wim Taymans
7e12b58e37 update for adapter api changes 2011-11-10 18:32:58 +01:00
Wim Taymans
00d3f3a454 fix for audio clock change 2011-11-10 13:50:34 +01:00
Wim Taymans
88e398b0ea update for removed fixate function 2011-11-10 11:03:18 +01:00
Wim Taymans
aa0b2b7ea7 updates for new acceptcaps query 2011-11-09 17:38:03 +01:00
Wim Taymans
95f3987332 Merge branch 'master' into 0.11 
Conflicts:
	ext/flac/gstflacdec.c
	gst/audioparsers/gstflacparse.c
	gst/isomp4/qtdemux.c
 2011-11-09 12:18:01 +01:00
Wim Taymans
49658dd5b5 remove query types 2011-11-09 11:53:01 +01:00
Wim Taymans
c48df77320 update for probe api changes 2011-11-08 11:18:06 +01:00
Wim Taymans
de020130e6 fix for probe updates 2011-11-07 17:14:17 +01:00
Vincent Penquerc'h
5a73374f2c flacdec: fix off by one between granpos and last_stop 2011-11-07 12:38:10 +00:00
Vincent Penquerc'h
6a25727321 cairotextoverlay: add a 'silent' property to skip rendering 
https://bugzilla.gnome.org/show_bug.cgi?id=662856
 2011-11-07 12:35:26 +00:00
Stefan Sauer
fb162c8eb4 controller: port to new controller location and api 2011-11-04 20:15:48 +01:00
Wim Taymans
7753feb4fd pulseaudiosink: more 0.11 fixing 
Make sure the caps event gets to the sink.
 2011-11-04 16:21:13 +01:00
Wim Taymans
f6f8d9bb17 pulseaudiosink: port some more 
Rename decodebin2 -> decodebin some more
Cleanup up sinkpad event handling
 2011-11-04 15:35:42 +01:00
Wim Taymans
1352a08a71 pulseaudiosink: port some more to 0.11 
We must not forward the caps event. instead we will decide what to do when the
pad block is taken.
Use decodebin instead of decodebin2
 2011-11-04 13:56:06 +01:00
Wim Taymans
e038ab5a0b tags: update for tag API removal 2011-11-02 12:09:20 +01:00
Wim Taymans
22eb0d2300 Merge branch 'master' into 0.11 2011-11-02 10:40:12 +01:00
Wim Taymans
9a8a8e72c8 structure: fix for api update 2011-11-02 09:06:37 +01:00
Tim-Philipp Müller
9f77b02b15 Update for pad API changes 
GstProbeType, GstProbeReturn and GstActivateMode -> GstPad*
 2011-11-01 00:52:28 +00:00
Tim-Philipp Müller
d6e1f53233 flacenc: remove dead code from header 
We require a new-enough libflac that this condition will never apply.
 2011-10-30 19:30:14 +00:00
Tim-Philipp Müller
a49818f876 flacdec: parse stream headers from caps in set_format function 
Not that this seems to be actually needed, libflac happily decodes
stuff even if we just drop all headers and never feed it to the
library.
 2011-10-30 19:12:44 +00:00
Tim-Philipp Müller
ab591b6d53 flacdec: don't extract metadata, leave that to the parser or container 2011-10-30 19:12:44 +00:00
Tim-Philipp Müller
5ab43cdf91 flacdec: we expect framed input now, remove some more code 2011-10-30 19:12:39 +00:00
Tim-Philipp Müller
92361863e6 flacdec: naive port to GstAudioDecoder 
This would probably have been too invasive to do in the 0.10
branch, with all the pull-mode and parser handling code in
there.
 2011-10-30 17:39:40 +00:00
Tim-Philipp Müller
9cd17092d8 ext, gst: update for taglist API changes 2011-10-30 11:44:53 +00:00
Wim Taymans
d40e915449 Merge branch 'master' into 0.11 2011-10-28 16:52:08 +02:00
Wim Taymans
3389e79f38 pulseaudiosink: fix porting errors 
The probes were ported wrongly and caused deadlocks.
 2011-10-28 15:11:10 +02:00
Tim-Philipp Müller
ff40deb139 jpegdec: add sof-marker to template caps, so we don't get plugged for lossless jpeg 
jpegdec (using libjpeg 6.2/8) can't decode some lossless types of JPEG.

https://bugzilla.gnome.org/show_bug.cgi?id=556648
 2011-10-28 12:10:34 +01:00
Wim Taymans
65c71b0717 pulse: fix check for empty caps 2011-10-28 12:51:31 +02:00
Wim Taymans
4b6a226263 Merge branch 'master' into 0.11 
Conflicts:
	ext/pulse/pulseaudiosink.c
	ext/pulse/pulsesink.c
 2011-10-27 16:08:22 +02:00
Wim Taymans
fc4684f4c6 fix compilation 2011-10-27 16:03:17 +02:00
Stefan Sauer
2468492f02 interfaces: clean up the use of iface and class/klass 2011-10-21 14:58:41 +02:00
René Stadler
41116224c8 pngenc: increase arbitrary resolution limits 
Apparently libpng can technically do up to 2^31-1 rows and columns. However it
imposes an (arbitrary) default limit of 1 million (that could theoretically be
lifted by using some additional API).

Moved array allocation to the heap now.
 2011-10-21 10:27:04 +02:00
René Stadler
db1f10adc8 pngenc: don't unconditionally allocate 4096 pointers on the stack 
Instead allocate as many as needed (on the stack still).
 2011-10-21 10:26:48 +02:00
René Stadler
65f9354803 pngenc: ensure setcaps was called before chain function 
This is needed to properly error out for e.g. "fakesrc ! pngenc ! fakesink".
 2011-10-21 10:26:20 +02:00
René Stadler
7e390c4635 pngenc: validate input buffer size 
Just for safety; of course such mismatch represents a bug in another element.
 2011-10-21 10:25:51 +02:00
René Stadler
9eb55c3d8f pngenc: make setcaps more robust, use gstvideo functions 
A setcaps function needs to actually verify the caps carefully. In this case,
it was possible to e.g. link a video decoder with YUV+RGB template caps to
pngenc.  That would cause a crash when the decoder pushes a YUV buffer. Same
thing when pushing a valid buffer that exceeds the resolution limits.

Also, missing framerate caps field would cause a glib critical warning due to
invalid GValue. This fails hard now.
 2011-10-21 10:25:08 +02:00
Arun Raghavan
a7790efd04 pulse: Get caps correctly on pad block 
Instead of always going upstream, we should first see if already got
caps from a setcaps() call.

https://bugzilla.gnome.org/show_bug.cgi?id=661262
 2011-10-18 20:02:55 +05:30
Tim-Philipp Müller
e9ad06e202 wavpackenc: don't unref buffer with gst_object_unref() 2011-10-18 12:25:14 +01:00
Wim Taymans
6de67bb014 pulsesink: only use is_pcm for 1.0 of pulseaudio 2011-10-18 12:05:01 +02:00
Wim Taymans
0ade1a5822 pulsesink: only disable trickmodes for !pcm 
Only disable trickmodes when we are not dealing with raw PCM samples.
 2011-10-18 11:58:57 +02:00
Edward Hervey
1a10116bbe flacenc: Properly register type 
It's a subclass of GstAudioEncoder and not of GstElement
 2011-10-13 17:12:23 +02:00
Wim Taymans
16649b2508 fix compile 2011-10-13 09:02:47 +02:00
Wim Taymans
a5cc912140 Merge branch 'master' into 0.11 
Conflicts:
	ext/jpeg/gstjpegdec.c
	gst/rtp/gstrtpvrawpay.c
 2011-10-13 08:58:06 +02:00
Thiago Santos
0e167e59d4 pulseaudiosink: Use new GstIterator API correctly 
GstIterator now uses GValue, use it correctly.
 2011-10-12 07:36:09 -03:00
Sjoerd Simons
95db648516 jpegdec: Implement upstream negotiation 
Add upstream negotiation for jpegdec. Fixes #660275
 2011-10-10 21:37:10 +01:00
Wim Taymans
03ca12d974 annodex: port to 0.11 2011-10-10 12:27:06 +02:00
Wim Taymans
aea9b5e8c8 Merge branch 'master' into 0.11 
Conflicts:
	ext/speex/gstspeexenc.c
 2011-10-10 11:48:20 +02:00
Thiago Santos
b09704020c pulse: port pulseutil to 0.11 2011-10-10 00:18:56 -03:00
Thiago Santos
4517eb28c0 pulseaudiosink: port to 0.11 2011-10-09 21:19:30 -03:00
Thiago Santos
358767e217 pulsesink: Fixing getcaps function 
Update getcaps function to 0.11 API
 2011-10-09 21:19:27 -03:00
Mark Nauwelaerts
00a91fc061 speexenc: only push header buffers following initial events 2011-10-09 21:32:32 +02:00
Wim Taymans
586ef0babd Merge branch 'master' into 0.11 
Conflicts:
	ext/speex/gstspeexdec.c
	ext/speex/gstspeexenc.c
	gst/isomp4/atoms.c
	gst/isomp4/gstqtmux.c
 2011-10-06 12:23:39 +02:00
Tim-Philipp Müller
ca77c96c51 speexenc: initialise variable before adding to it 2011-09-29 23:21:46 +01:00
Mark Nauwelaerts
c5354bee04 speexdec: port to audiodecoder 2011-09-29 17:33:25 +02:00
Mark Nauwelaerts
53476c1580 speexenc: clean up some unused remnants 2011-09-29 17:33:23 +02:00
Mark Nauwelaerts
c1909c32c5 speexenc: port to audioencoder 2011-09-29 17:33:21 +02:00
Tim-Philipp Müller
3d01b9f398 flacdec: get rid of granulepos handling 
Leave that to the parser or demuxer. There's still some
code for operating in DEFAULT (samples) format, but that
will be removed later.
 2011-09-28 19:10:27 +01:00
Tim-Philipp Müller
5c28f426d7 flacdec: get rid of pull-mode support and focus on being a decoder 
Leave all the other stuff to flacparse.
 2011-09-28 19:03:13 +01:00
Tim-Philipp Müller
e0d994c9e1 flac, jpeg: fix compiler warning 2011-09-28 17:39:06 +01:00
Wim Taymans
b4524858be flac: port to 0.11 2011-09-28 17:40:01 +02:00
Wim Taymans
762602d56a Merge branch 'master' into 0.11 
Conflicts:
	ext/flac/gstflacenc.c
 2011-09-28 17:39:12 +02:00
Mark Nauwelaerts
e8bcd41d73 flacenc: port to audioencoder 2011-09-28 16:14:46 +02:00
Wim Taymans
87fbd1e784 Merge branch 'master' into 0.11 
Conflicts:
	common
	ext/pulse/pulsesink.c
	ext/soup/gstsouphttpclientsink.c
	gst/audioparsers/gstaacparse.c
	gst/audioparsers/gstac3parse.c
	gst/rtp/gstrtph264depay.c
	gst/rtpmanager/gstrtpjitterbuffer.c
	gst/rtpmanager/rtpjitterbuffer.c
	gst/rtsp/gstrtspsrc.c
	sys/ximage/gstximagesrc.c
 2011-09-28 12:44:59 +02:00
Tim-Philipp Müller
3828537857 soup: rename souphttpsink to souphttpclientsink 
To avoid confusion, and because we might want a server
sink at some point too.

https://bugzilla.gnome.org/show_bug.cgi?id=659947
 2011-09-25 15:13:39 +01:00
Tim-Philipp Müller
be7cbd4c21 souphttpsink: don't create unused second sink pad object 
The base class will create the sink pad.
 2011-09-23 16:39:46 +01:00
Vincent Penquerc'h
7e4574e968 speexenc: do not use invalid buffer timestamps 2011-09-19 09:37:58 +02:00
Arun Raghavan
8ca420f547 pulse: New pulseaudiosink element to handle format changes 
This introduces a new bin which wraps around pulsesink and depending on
the formats supported by the sink, plugs in/out a decodebin2 as
required. This allows users to switch sinks on the stream and adapts
accordingly (for example, you could watch a movie in passthrough mode on
your receiver which supports AC3 decode, then plug out and switch to a
non-digital profile to continue uninterrupted on analog output).

The bin is required because doing the same with playbin2/playsink will
require API changes that cannot be made in 0.10. With 0.11/1.0, we
should be able to ask for upstream caps renegotiation to deal with all
this.

https://bugzilla.gnome.org/show_bug.cgi?id=657179
 2011-09-19 07:43:04 +05:30
Konstantin Miller
24d002e04d souphttpsrc: Don't handle HTTP response 407 as error if proxy authentication data is available 
Fixes bug #657422.
 2011-09-07 13:28:45 +02:00
Wim Taymans
33f18b8ea4 Merge branch 'master' into 0.11 
Conflicts:
	gst/audioparsers/gstamrparse.c
	gst/isomp4/qtdemux.c
 2011-09-06 16:06:25 +02:00
Wim Taymans
e204c5934c -good: port to new audio caps 2011-09-06 13:16:27 +02:00
Sebastian Dröge
7b592ff126 souphttpsrc: Allow positive, non-1.0 segment rates 
Only negative rates are not supported. Fixes bug #658305.
 2011-09-06 10:34:35 +02:00
Wim Taymans
85d7fe14b2 soup: port soup elements to 0.11 2011-08-29 18:02:15 +02:00
Wim Taymans
34ea60526d pulse: add some more channels 2011-08-24 18:44:01 +02:00
Wim Taymans
e9df54819c Merge branch 'master' into 0.11 2011-08-24 14:16:44 +02:00
Arun Raghavan
bd604175c5 pulsesink: Trivial indentation fix 2011-08-23 22:48:34 +05:30
Monty Montgomery
799c8e3d04 flacdec: Correct sample number rounding resulting in timestamp jitter 
flacdec converts the src timestamp to a sample number, uses that internally, then reconverts the sample number to a timestamp for the output buffer.  Unfortunately, sample numbers can't be represented in an integer number of nanoseconds, and the conversion process was truncating rather than rounding, resulting in sample numbers and output timestamps that were often off by a full sample.

This corrects the time->sample convesion
 2011-08-23 10:09:41 +02:00
Wim Taymans
0eeffef222 pulsesink: port after merge 2011-08-19 16:13:23 +02:00
Wim Taymans
e1b795ac13 Merge branch 'master' into 0.11 2011-08-19 16:12:01 +02:00
Wim Taymans
77ad0a1363 port more elements to new audio caps and API 2011-08-19 14:01:45 +02:00
David Henningsson
e70020b456 pulsesink: Allow writes in bigger chunks 
There's no use in splitting the incoming data down to the segsize
limit - by writing as much as possible in one chunk, we increase
performance and avoid PulseAudio unnecessary rewinds.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
 2011-08-19 09:48:27 +02:00
Wim Taymans
09b15d7dfe port to new audio caps. 2011-08-18 19:21:07 +02:00
Wim Taymans
ce1e7cb108 Merge branch 'master' into 0.11 
Conflicts:
	ext/flac/gstflacdec.c
 2011-08-17 15:52:18 +02:00
Wim Taymans
be4f60b062 jpeg: port to 0.11 
Also disable smoke for now.
 2011-08-17 15:39:27 +02:00
Vincent Penquerc'h
3e0134f51f flacdec: avoid timestamp/offset tracking going out of sync 
The libFLAC API is callback based, and we must only call it to
output data when we know we have enough input data. For this
reason, a single processing step is done when receiving a buffer.
However, if there were metadata buffers still pending, a step
intended for the first audio frame might end up writing that
leftover metadata. Since a single step is done per buffer, this
will cause every buffer to be written one step late.

This would add some latency (a bufferfull's worth), possibly
lose a buffer when seeking or the like, and also cause timestamp
and offset to be applied to the wrong buffer, as updates to
the "current" segment last_stop (from incoming buffer timestamp)
will be applied to an output buffer originating from the previous
incoming buffer.

This fixes the issue by ensuring that, upon receiving the first
audio frame, processing is done till all metadata is processed,
so the next "single step" done will be for the audio frame. After
this, we should keep to 1 input buffer -> 1 output buffer and so
avoid getting out of sync.

https://bugzilla.gnome.org/show_bug.cgi?id=650960
 2011-08-17 13:40:59 +01:00
Vincent Penquerc'h
e09eb95a5f flacdec: bail on reserved value 
Now that we look at the right bits, we can test against the reserved
value as we do for other fields.

https://bugzilla.gnome.org/show_bug.cgi?id=650960
 2011-08-17 00:02:38 +01:00
Vincent Penquerc'h
64beef4610 flacdec: fix bit twiddling 
Right shifting a 8 bit value by 8 bits is twice too much
to get the high 4 bits.

https://bugzilla.gnome.org/show_bug.cgi?id=650960
 2011-08-17 00:01:37 +01:00
Vincent Penquerc'h
1549aaba27 flacdec: warn if we see a variable block size where unsupported 
https://bugzilla.gnome.org/show_bug.cgi?id=650960
 2011-08-17 00:01:07 +01:00
Wim Taymans
4bb2b140e9 Merge branch 'master' into 0.11 
Conflicts:
	sys/v4l2/v4l2src_calls.c
 2011-08-16 18:35:53 +02:00
Tim-Philipp Müller
26a3a12513 jackaudiosrc: fix error message code 
And also post 'not found' error if jackd is not even installed.
 2011-08-13 16:52:53 +01:00
Edward Hervey
145f6da5bb aasink: Remove unused variables 2011-08-10 11:28:26 +02:00
Tim-Philipp Müller
9f904ac438 aalib: make sure -DGST_USE_UNSTABLE_API is defined 
So we don't get warnings.
 2011-08-08 15:26:00 +01:00
Wim Taymans
71346020d5 pulsesrc: avoid race in starting 
Sine the base class now does the negotiation from the streaming thread we have
to be careful and check if the stream is ready before changing its corked state.
 2011-08-07 11:17:41 +02:00
Wim Taymans
d9750387c1 pulse: more cleanups 2011-08-04 18:41:29 +02:00
Wim Taymans
9ae85cb662 pulsesrc: small cleanups 2011-08-04 18:15:55 +02:00
Wim Taymans
fcbe26cd6f pulsesrc: small cleanups 2011-08-04 16:32:39 +02:00
Wim Taymans
ee2aa25e04 port to new API 2011-08-03 18:37:27 +02:00
Wim Taymans
4121021bb2 Merge branch 'master' into 0.11 
Conflicts:
	ext/pulse/pulsesink.c
	ext/pulse/pulsesrc.c
	gst/audioparsers/gstac3parse.c
	gst/rtp/gstrtph264depay.c
	gst/rtp/gstrtph264pay.c
	gst/rtpmanager/gstrtpssrcdemux.c
 2011-08-03 18:25:30 +02:00
Sebastian Dröge
f18eccd286 hal: Remove hal plugin 
hal is not developed anymore and nobody is using the plugin nowadays.
 2011-08-03 10:59:56 +02:00
Tristan Matthews
c26442a3ba jackaudiosink: Don't call g_alloca() in process_cb 
g_alloca() is not RT-safe, so instead we should allocate the
memory needed in advance. Fixes #655866
 2011-08-03 09:44:05 +02:00
Tim-Philipp Müller
25ace0e524 pulsesink: fix variable-set-but-not-used compiler warning with older pulse versions 2011-07-29 13:05:42 +01:00
Arun Raghavan
ac7cad431c pulsesink: Add support for compressed formats 
This adds support for various compressed formats (AC3, E-AC3, DTS and
MP3) payloaded in IEC 61937 format (used for transmission over S/PDIF,
HDMI and Bluetooth).

The acceptcaps() function allows bins to probe for what formats the sink
being connected to support. This only works after the element is set to
at least READY.

If the underlying sink changes and the format we are streaming is not
available, we emit a message that will allow upstream elements/bins to
block and renegotiate a new format.
 2011-07-29 01:25:15 +05:30
Arun Raghavan
a67b536741 pulsesink: Use the extended stream API if available 
This uses the new extended API for creating streams. This will allow us
to support compressed formats natively in pulsesink as well.
 2011-07-29 01:25:15 +05:30
Arun Raghavan
379049809c pulsesrc: Add a source-output-index property 
This exposes the source output index of the record stream that we open
so that clients can use this with the introspection if they want (to
move the stream, for example).
 2011-07-29 00:07:52 +05:30
Tim-Philipp Müller
ab62599832 speex: update for position/query/convert API changes 2011-07-28 11:38:31 +01:00
Thiago Santos
14b9fb7be6 pulsesrc: Fix default value leaking 
Remember to free the default value of client name, avoiding a
leak
 2011-07-18 15:16:01 -03:00
Wim Taymans
da28ebfbe3 aasink: port to new video API 2011-07-06 17:50:54 +02:00
Wim Taymans
1a0a6f54bb cacasink: port to 0.11 2011-07-06 17:40:20 +02:00
Wim Taymans
f70da0a542 jpeg: beginnings of porting to 0.11 2011-07-06 16:51:36 +02:00
Wim Taymans
fdf5a49422 speex: port speex elements 2011-07-06 15:57:23 +02:00
Wim Taymans
3fd1106b7e Merge branch 'master' into 0.11 2011-07-06 12:05:12 +02:00
René Stadler
ae87731de5 pulsesink: prevent race condition causing ref leak 
Since commit 8bfd80, gst_pulseringbuffer_stop doesn't wait for the
deferred call to be run before returning. This causes a race when
READY->NULL is executed shortly after, which stops the mainloop. This
leaks the element reference which is passed as userdata for the callback
(introduced in commit 7cf996, bug #614765).

The correct fix is to wait in READY->NULL for all outstanding calls to
be fired (since libpulse doesn't provide a DestroyNotify for the
userdata). We get rid of the reference passing from 7cf996 altogether,
since finalization from the callback would anyways lead to a deadlock.

Re-fixes bug #614765.
 2011-07-05 16:36:17 +02:00
René Stadler
f8456e2a1a pulsesink: small cleanup of copy-paste code 2011-07-05 16:36:17 +02:00
René Stadler
3589cee762 pulsesink: remove unused member variable and misleading log message 
Wim changed it in commit 8bfd80 so that pa_defer_ran is not read
anywhere.

The log message used to annotate a mainloop_wait call which is gone.
 2011-07-05 16:36:17 +02:00
Wim Taymans
8b040cfae2 pulse: remove implementsinterface 2011-07-04 18:12:56 +02:00
Mark Nauwelaerts
0c25863253 jpegdec: avoid crashing on invalid input without components 2011-07-04 14:32:27 +02:00
Mark Nauwelaerts
d59a00aa1c Merge branch 'master' into 0.11 
Conflicts:
	ext/pulse/pulsesink.c
 2011-07-04 11:48:13 +02:00
David Schleef
f69dcaab58 pulse: Increase ranks to PRIMARY + 10 
So that pulsesrc/pulsesink get chosen over other possible PRIMARY
src/sinks by autoaudiosink.  Presumably, if pulse is available, it
is always preferred over another src/sink.

Fixes: #647540.
 2011-07-03 19:53:42 -07:00
David Schleef
2f94df8032 jpegenc: Don't round up size of encoded buffers 
For some reason, in code dating to 2001, encoded jpeg buffers were
rounded up to multiples of 4 bytes.  With the added bonus that the
extra bytes are unwritten, causing valgrind issues.  Oops.  I can't
think of any reason why JPEG buffers need to be multiples of 4 bytes,
so I removed the padding.  There might be some code somewhere that
depends on this behavior, so if this needs to be reverted, please fix
the valgrind issues.
 2011-06-29 23:55:33 -07:00
Andoni Morales Alastruey
d9f4c59c49 dv1394src: make the internal clock thread safe 
Fixes: #653091.
 2011-06-24 12:01:39 -07:00
Arun Raghavan
04786a6d31 pulse: Drop support for PA versions before 0.9.16 
This drops support fof PulseAudio versions prior to 0.9.16, which was
released about 1.5 years ago. Testing with very old versions is not
feasible and we don't want to maintain 2 independent code-paths.
 2011-06-21 16:47:11 -07:00
Wim Taymans
adee56222b pulsesink: fix for header cleanups 2011-06-21 18:19:02 +02:00
Tim-Philipp Müller
3d6955d686 Remove esound/esdsink plugin 2011-06-15 15:09:18 +01:00
Tim-Philipp Müller
728732438b Remove gconf elements and plugin 
GConf was deprecated in favour of GSettings etc.
 2011-06-15 14:37:29 +01:00
Wim Taymans
c2358404c1 jack: port some more 2011-06-09 17:56:18 +02:00
Wim Taymans
3446ca801e aasink: register template and klass correctly 2011-06-09 11:37:47 +02:00
Wim Taymans
710fa239d5 Merge branch 'master' into 0.11 2011-06-08 18:06:56 +02:00
Wim Taymans
0af32751ae assink: port aasink to 0.11 2011-06-08 18:05:20 +02:00
David Schleef
6d8d08eac7 souphttpsink: code cleanup 2011-06-04 13:43:28 -07:00
Wim Taymans
0b1bdcf7cb Merge branch 'master' into 0.11 
Conflicts:
	sys/ximage/ximageutil.c
 2011-06-02 18:51:29 +02:00
Sebastian Dröge
252c1afcc2 dvdemux: First query the peer duration in the requested format before converting to BYTES 
Fixes usage of dvdemux after another demuxer, e.g. mxfdemux.
Fixes bug #650503.
 2011-06-02 13:38:30 +02:00
Sebastian Dröge
e376913fc3 souphttpsink: Fix refcounting of the "session" property 
Properties should never take ownership of the values
passed to them.
 2011-06-02 10:41:52 +02:00
David Schleef
3937abf999 soup: Add souphttpsink 2011-06-01 17:19:02 -07:00
Vincent Penquerc'h
38b596337f cairotextoverlay: forward new segment events from the sink to the source 
Not doing so will cause buffers to be received by downstream without
a time base set.
We use the same method avimux uses to get access to the event when
collectpads got the sink event function.

https://bugzilla.gnome.org/show_bug.cgi?id=640323
 2011-05-26 11:35:01 +02:00
Vincent Penquerc'h
b99f4be3db textoverlay: forward source events to sinks 
Events are passed to the video sink, and to the text sink if it is
linked.
This will allow seeking, for instance.

https://bugzilla.gnome.org/show_bug.cgi?id=586450
 2011-05-26 11:33:10 +02:00
Wim Taymans
d89790d545 Merge branch 'master' into 0.11 
Conflicts:
	gst/avi/gstavidemux.c
	gst/rtp/gstrtpac3depay.c
	gst/rtp/gstrtpg726depay.c
	gst/rtp/gstrtpmpvdepay.c
	gst/videofilter/gstgamma.c
 2011-05-24 17:34:19 +02:00
Tim-Philipp Müller
900b2fb63d flacdec: also try upstream first for duration query in DEFAULT format 
https://bugzilla.gnome.org/show_bug.cgi?id=650785
 2011-05-23 13:25:44 +01:00
Wim Taymans
cf1248e566 Merge branch 'master' into 0.11 
Conflicts:
	configure.ac
 2011-05-16 17:52:11 +02:00
Wim Taymans
a0e301ab1a -good: fix for new API 2011-05-16 17:50:15 +02:00
Pino Toscano
b52c7ebd19 pulse: Define PATH_MAX if it isn't defined 
GNU Hurd for example doesn't define it.
 2011-05-16 09:04:31 +02:00
Sebastian Dröge
d4b9cbc1ae flacdec: Send EOS when seeking after the end of file instead of failing 
Fixes bug #649780.
 2011-05-14 11:52:13 +02:00
Wim Taymans
992e3f7c55 Merge branch 'master' into 0.11 
Conflicts:
	configure.ac
	gst/rtp/gstrtpgstpay.c
 2011-04-29 15:49:41 +02:00
Wim Taymans
5ed18ad7b9 jack: port jack elements 2011-04-26 15:42:47 +02:00
Philip Jägenstedt
660987c6d7 jpegdec: documentation typo "jpegddec" 
https://bugzilla.gnome.org/show_bug.cgi?id=648589
 2011-04-25 21:36:28 +01:00
Wim Taymans
a1894ed363 Merge branch 'master' into 0.11 2011-04-25 11:38:28 +02:00
David Schleef
24d57ac961 pngdec: Remove temporary code 
Now that we depend on (what will be) -base-0.10.33.
 2011-04-24 18:53:50 -07:00
Wim Taymans
2f9a7b1a1b flac: port to 0.11 2011-04-19 18:55:31 +02:00
Wim Taymans
dea50d0b98 use G_DEFINE_TYPE 2011-04-19 17:07:18 +02:00
Wim Taymans
7555d0949f Merge branch 'master' into 0.11 
Conflicts:
	android/apetag.mk
	android/avi.mk
	android/flv.mk
	android/icydemux.mk
	android/id3demux.mk
	android/qtdemux.mk
	android/rtp.mk
	android/rtpmanager.mk
	android/rtsp.mk
	android/soup.mk
	android/udp.mk
	android/wavenc.mk
	android/wavparse.mk
	configure.ac
 2011-04-18 10:23:45 +02:00
Tim-Philipp Müller
f325935314 pulse, speexenc, rtpgsmpay: don't use g_assert() for error handling 
Don't use g_assert() for error handling, even if they're highly unlikely.
Either we *know* that something can't happen, in which case we
should just not handle it, or we think something can happen, but it is
very very unlikely that it will ever happen, in which case we should
handle it like any other error instead of asserting.

g_assert() is best left for conditions we have control of, like checking
internal consistency of our code, not checking return values of external
code.

Fixes a bunch of warnings when compiling with -DG_DISABLE_ASSERT:
gstrtpgsmpay.c: In function 'gst_rtp_gsm_pay_handle_buffer':
gstrtpgsmpay.c:130:17: warning: variable 'rtpgsmpay' set but not used
gstspeexenc.c: In function 'gst_speex_enc_encode':
gstspeexenc.c:904:19: warning: variable 'written' set but not used
pulsesink.c: In function 'gst_pulsesink_change_state':
pulsesink.c:2725:9: warning: variable 'res' set but not used
pulsesrc.c: In function 'gst_pulsesrc_change_state':
pulsesrc.c:1253:7: warning: variable 'e' set but not used
 2011-04-16 18:15:43 +01:00
Tim-Philipp Müller
a068133ee7 jack: fix unused-but-set-variable warnings with gcc-4.6 2011-04-16 13:33:45 +01:00
Robert Swain
098cf89e34 jpegenc: pulsesink: raw1394: Address unused but set variables 
GCC 4.6.x spits warnings about such usage of variables. The variables in
raw1394 were marked with G_GNUC_UNUSED as this seemed omre appropriate.
The others were removed.
 2011-04-16 13:13:04 +01:00
Sebastian Dröge
18eee7719d flacenc: Properly interprete the result of strcmp() 2011-04-13 16:17:41 +02:00
Sebastian Dröge
16f82e46f5 flacenc: Don't store image tags inside the vorbiscomments and the flac metadata 
Instead only store them inside the flac metadata. There's
no point in storing them twice and the flac metadata is
still the official way to store image tags inside flac.
 2011-04-13 16:11:44 +02:00
Stefan Kost
21a688ce25 jack: also handle deprecations for jack 1.9.7 
Jack 1.9.7 was released 20.Mar.2011, need to handle the deprecated api for this
version too.
 2011-04-11 12:44:19 +03:00
Sebastian Dröge
b784173e4a jpegdec: Unref event if the parent element disappeared 2011-04-08 15:22:47 +02:00
Havard Graff
9386448649 jpegdec: Make upstream events MT-safe 2011-04-08 15:21:52 +02:00
Sebastian Dröge
11bcac7c90 Revert "Pulsesink: Allow chunks up to bufsize instead of segsize" 
This reverts commit 1e2c1467ae.

The commit causes pulsesink to ignore the latency-time baseaudiosink property.
 2011-04-08 14:35:04 +02:00
Alexey Fisher
0016ceaa2b speexenc: Use speex intern silence detection 
Speex has build in silence detection. If speex_encode_int returns 0,
than there is silence and sample do not need to be transmitted.
This work only if vbr=1 and dtx=1 optionas are enabled.
So if we get 0, we add GAP flag to the sample.
 2011-04-08 13:54:49 +02:00
Wim Taymans
c124ba1489 Merge branch 'master' into 0.11 
Conflicts:
	gst/rtsp/gstrtspsrc.c
 2011-04-05 17:20:08 +02:00
Jon Nordby
d68dd46084 jack: Fix build with jack 0.120.1 
9544622674 checked
for 0.120.2 and later, but the deprecation was introduced in
0.120.1
 2011-04-05 13:12:28 +03:00
Arun Raghavan
dc48eaac13 pulsesink: Always call pa_stream_new_with_proplist() 
pa_stream_new_with_proplist() can take a NULL proplist, so we don't need
to concern ourselves with whether it's NULL or not.
 2011-04-04 17:23:21 +05:30
Sebastian Dröge
6fd1546bce flacenc: Add support for writing METADATA_BLOCK_PICTURE blocks for GST_TAG_IMAGE and GST_TAG_PREVIEW_IMAGE 2011-04-01 13:18:55 +02:00
Mark Nauwelaerts
176b8ffbff pulsesink: also uncork during EOS waiting (and after EOS is rendered) 
Pulsesink was recently changed to defer uncorking until there is data
to write. This condition will however never occur when EOS in being
rendered (since that marks the end of data). Changing to PAUSED state
while EOS is being waited on results in a hang: pausing corks the
stream, which will never be undone since there is no more data when
going back to PLAYING. If pulsesink is the clock provider, deadlock
ensues since time doesn't continue in corked state and the clock id
for EOS wait never fires.

Fixes #645961.
 2011-03-31 13:25:19 +02:00
Tim-Philipp Müller
c365fbddba flacdec: fix issues with large metadata blocks when streaming unframed flac 
Parse metadata blocks when handling unparsed flac in push mode. This
works around a bunch of issues with the flac decoder when handling
metadata blocks that are larger than the max. flac framesize, which
coverart blocks often are. We need to have all the data for these
blocks available when we pass data to libflac.

http://gstreamer-devel.966125.n4.nabble.com/Flac-files-that-will-playback-but-not-stream-td3338198.html#a3395276

https://bugzilla.gnome.org/show_bug.cgi?id=566769
 2011-03-28 23:46:47 +01:00
Wim Taymans
8f22a09dc4 Merge branch 'master' into 0.11-fdo 2011-03-28 20:50:59 +02:00
Stefan Kost
ed77b14aa0 cairo: fix the name of the *-marshall.list file to unbreak make distcheck 2011-03-25 12:53:43 +02:00
Sebastian Dröge
85ace6d413 speexdec: Get and use streamheader from the caps if possible 
This allows playback of streams where the streamheader buffers
were dropped from the stream for some reason.
 2011-03-24 09:00:32 +01:00
Robert Swain
a430042f41 jackaudiosink: Fix typo from 9544622674 2011-03-16 10:43:47 +01:00
Vincent Penquerc'h
56edbd0fa3 speexdec: silence warning message when appropriate 
If we did not know how many frames to expect, then we get an unexpected
end of stream when trying to decode more frames that are there, if there
are leftover bits to pad to the next byte
 2011-03-15 19:44:03 +01:00
Sebastian Dröge
4d7b4ca2ae speexdec: Always process the number of frames per packet as specified in the header 
Looking at the remaining bits in the bitstream after decoding a
single frame can't be used as loop condition. The remaining
bits might not give a complete frame and the speex decoder will
then output nothing but access uninitialized memory, which leads
to valgrind warnings.

Fixes bug #644669.
 2011-03-14 19:31:33 +01:00
Philip Jägenstedt
84decd731f pulsesink: Better fix for deadlock on failed connect 
This reverts the previous fix that would cause a double-unlock when the
stream connect failed.

https://bugzilla.gnome.org/show_bug.cgi?id=644510
 2011-03-12 00:48:03 +05:30
Arun Raghavan
387f533a07 pulsesink: Fix deadlock if connecting to PA fails 
Commit dd4ec22e introduced a deadlock in the failure path while trying
to connect to PulseAudio. This makes sure we drop the lock on the
resource mutex to avoid this.

https://bugzilla.gnome.org/show_bug.cgi?id=644510
 2011-03-11 23:19:34 +05:30
Sebastian Dröge
9e44fff3db dvdemux: Chain up to the parent class' ::send_event for non-seek events 2011-03-11 13:48:33 +01:00
Sebastian Dröge
bc6c1bbbab dvdemux: Fix refcount issues with the seek event 
Fixes bug #642963.
 2011-03-11 13:48:32 +01:00
Tim-Philipp Müller
14e1d9de3d docs: fix pulsesink gtk-doc markup 2011-03-11 09:54:36 +00:00
Philippe Normand
9544622674 jack: fix build against jack 0.120.2 
jack_port_get_total_latency() has been deprecated in favor of
jack_port_get_latency_range().

https://bugzilla.gnome.org/show_bug.cgi?id=644477
 2011-03-11 09:48:36 +00:00
Wim Taymans
338c300f5f Merge branch 'master' into 0.11 2011-03-04 16:30:36 +01:00
Stefan Kost
89914439b7 pulsesink: add a doc example for setting stream-properties 2011-03-04 15:50:49 +02:00
Stefan Kost
edfa99ef31 pulsesink: fix the xml in the docs 2011-03-04 15:50:49 +02:00
Sebastian Dröge
fdbdbfefd2 cairooverlay: Some minor cleanup 2011-03-02 23:14:36 +01:00
Jon Nordby
32dff9df75 cairooverlay: Add generic Cairo overlay video element. 
Allows applications to connect to the "draw" signal of
the element and do their custom drawing there.
Includes an example application demonstrating usage.

Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=595520
 2011-03-02 22:55:52 +01:00
Wim Taymans
db5aa9067b Merge branch 'master' into 0.11 2011-02-28 13:29:47 +01:00
Wim Taymans
59c825a938 id3: use boxed type instead of miniobject 2011-02-28 11:58:48 +01:00
Wim Taymans
c899229c48 jpegdec: add duration when extimating QoS time 
When we need to decide on the next QoS time, take into account the duration of
the buffers.
 2011-02-28 11:58:48 +01:00
Philip Jägenstedt
dd4ec22e7e pulsesink: release pa_shared_resource_mutex before pa_threaded_mainloop_wait 
Not doing so can result in a deadlock when two threads enter
gst_pulseringbuffer_open_device at the same time, as
pa_threaded_mainloop_wait releases the mainloop lock while waiting,
allowing another thread to take it, resulting in a deadlock as two
threads waits for the lock the other is holding.

https://bugzilla.gnome.org/show_bug.cgi?id=643087
 2011-02-28 16:20:22 +05:30
Philip Jägenstedt
683c96b37a pulsesink: s/ressource/resource/ 
https://bugzilla.gnome.org/show_bug.cgi?id=643087
 2011-02-28 16:20:20 +05:30
Carsten Kroll
d4eaff5e88 dvdemux: First try if upstream handles TIME seeks before handling them here 
Fixes bug #642963.
 2011-02-24 13:57:01 +01:00
Edward Hervey
6e1c701502 Revert "Check that collectpads exists before removing pad" 
This reverts commit 8e6b876e76.

Depends on a core commit that was reverted
 2011-02-21 12:25:19 +01:00
David Schleef
8e6b876e76 Check that collectpads exists before removing pad 
The core now calls release pad from finalize, at which point
the collectpads might have already been freed.
 2011-02-20 23:46:01 -08:00
David Schleef
13d441315f pngdec: Handle 16-bit-per-channel images 2011-02-20 12:12:51 -08:00
Andoni Morales Alastruey
b2fb4edcb7 souphttpsrc: Add URI query handler 
Fixes bug #642337.
 2011-02-15 09:44:01 +01:00
David Henningsson
1e2c1467ae Pulsesink: Allow chunks up to bufsize instead of segsize 
By allowing larger chunks to be sent, PulseAudio will have a
lower CPU usage. This is especially important on low-end machines,
where PulseAudio can crash if packets are coming in at a higher
rate than PulseAudio can process them.

Signed-off-by: David Henningsson <david.henningsson@canonical.com>
 2011-01-31 17:15:05 +01:00
Arun Raghavan
e18e25a324 pulsesink: Uncork stream while flushing the ringbuffer 
After starting the ringbuffer, we wait for enough data to arrive before
uncorking the stream. This will cause the pipeline to stall if we get an
EOS (or otherwise need to flush the stream) before sufficient data
becomes available. This patch makes sure that the stream is uncorked
while flushing to avoid this problem.

Fixes issue with a webkit unit test testing reverse playback of
an MP4 H.264/AAC file.

https://bugzilla.gnome.org/show_bug.cgi?id=639740
 2011-01-17 15:40:12 +00:00
Arun Raghavan
7280ea1f68 pulsesink: Make corking during pause synchronous 
This makes the call to pa_stream_cork() during ringbuffer pause()
synchronous, which makes sure that the clock does not advance after we
take a snapshot for start_time.

https://bugzilla.gnome.org/show_bug.cgi?id=639240
 2011-01-11 20:43:54 +00:00
Wim Taymans
c17f7a943d cacasink: fix masks and strides 
Use the right endianness to read the masks.
Use the right strides for the bitmap.

Fixes #638569
 2011-01-03 18:06:48 +01:00
Tim-Philipp Müller
57e35559c8 jack: fix up element details and some other minor clean-ups 2011-01-02 19:51:54 +00:00
Tim-Philipp Müller
79a8e0ee42 jack: template caps had lists with one value, just use value directly 2011-01-02 17:40:41 +00:00
Tim-Philipp Müller
1f5083a17c jack: make get_type functions thread-safe 
Because we can (shouldn't be needed with other workarounds still there).
 2011-01-02 17:27:35 +00:00
Tim-Philipp Müller
67f4489cd0 jack: new jackaudiosrc and jackaudiosink elements, moved from gst-plugins-bad 
https://bugzilla.gnome.org/show_bug.cgi?id=621929
 2011-01-02 15:26:30 +00:00
Stefan Kost
3c43a9e102 various (ext): add missing G_PARAM_STATIC_STRINGS flags 
Canonicalize property names as needed.
 2011-01-02 14:30:09 +00:00
Tristan Matthews
11c4157af4 jack: added translatable text for server not found error 2011-01-02 14:30:08 +00:00
Tristan Matthews
c070cfa7c8 jack: added client property 2011-01-02 14:30:08 +00:00
Tristan Matthews
a110fe3343 jack: removed unused file gstjackbin.c 
This is a 0.8 leftover.
 2011-01-02 14:30:08 +00:00
Wim Taymans
71b06e4957 jacksrc: make sure we always read nframes 
Error out when we are asked to read a different size that what was configured as
the jack period size because that would mean something else is wrong.

Fixes #618409
 2011-01-02 14:30:08 +00:00
Tristan Matthews
e05b723274 jack: improve process_cb 2011-01-02 14:30:08 +00:00
Tristan Matthews
55f1ea09b0 jack: implement multichannel support correctly for jackaudiosrc 
Fixes parts of bug #616541.
 2011-01-02 14:30:08 +00:00
Stefan Kost
3153aa995f jack: remove empty dispose and finalize methods 2011-01-02 14:30:08 +00:00
Stefan Kost
2b7ca1413a jack: don't leak caps 
Add dispose methods to clear caps.
 2011-01-02 14:30:08 +00:00
Stefan Kost
084e8269c2 jack: don't use GST_DEBUG_FUNCPTR for gobject vmethods 2011-01-02 14:30:08 +00:00
Stefan Kost
0a22a42268 jack: fix element name in section doc blob 2011-01-02 14:30:08 +00:00
Benjamin Otte
de91f84dfb Add -Wold-style-definition 
and fix the warnings
 2011-01-02 14:30:08 +00:00
Benjamin Otte
71366f80d9 Add -Wmissing-declarations -Wmissing-prototypes to configure flags 
And fix all warnings
 2011-01-02 14:30:07 +00:00
Benjamin Otte
4ccda26327 gst_element_class_set_details => gst_element_class_set_details_simple 2011-01-02 14:30:07 +00:00
Stefan Kost
f20c73684b jack: ensure segtotal is at least 2 
Not only adjust buffer-time and avoid segtotal=0, but instead ensure segtotal is
atleast 2. Do same change on jacksrc. We could also check the latency and buffer
time configured by the client and adjust buffer-time so that we get to the same
number of segments.
 2011-01-02 14:30:07 +00:00
Stefan Kost
45fb781f9b jack: don't crash in ringbuffer with SIGFPE on small buffer-times 
Jack overrides user-specified latency-time with the one it gets from jack
itself. It also needs to adjust buffer-time somewhat to avoid segtotal being 0
 2011-01-02 14:30:07 +00:00
Stefan Kost
8a002a1121 jack: when stopping playback, do one more cycle to flush the port. Fixes #582167 
The gst_jack_audio_client_set_active() flags the port as deactivating and uses
a GCond to wait until the jack_process_cb() has run once more and cleared the
flag. This way the client zero's the buffer. This happens if one manyally go
to PAUSED and then to READY, while leting the mainloop run inbetween.
 2011-01-02 14:30:07 +00:00
Wim Taymans
7dde7eca28 jack: Add new connection mode 
Add a new connection mode to jacksrc and jacksink. In this new auto-force
connection mode jack will create as many ports as requested/needed in the
pipeline and will then connect as many physical ports as possible, possibly
leaving some ports unconnected.

Also get rid of some leftover g_print.

Fixes #575284.
 2011-01-02 14:30:07 +00:00
Stefan Kost
fb03eb6bac ext/jack/: Query port latencies for sink/src delays. 
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c:
* ext/jack/gstjackaudiosrc.c:
Query port latencies for sink/src delays.
* ext/jack/gstjackbin.c:
No printf please.
 2011-01-02 14:30:07 +00:00
Stefan Kost
7ba1bcb6aa Don't install static libs for plugins. Fixes #550851 for -bad. 
Original commit message from CVS:
* ext/alsaspdif/Makefile.am:
* ext/amrwb/Makefile.am:
* ext/apexsink/Makefile.am:
* ext/arts/Makefile.am:
* ext/artsd/Makefile.am:
* ext/audiofile/Makefile.am:
* ext/audioresample/Makefile.am:
* ext/bz2/Makefile.am:
* ext/cdaudio/Makefile.am:
* ext/celt/Makefile.am:
* ext/dc1394/Makefile.am:
* ext/dirac/Makefile.am:
* ext/directfb/Makefile.am:
* ext/divx/Makefile.am:
* ext/dts/Makefile.am:
* ext/faac/Makefile.am:
* ext/faad/Makefile.am:
* ext/gsm/Makefile.am:
* ext/hermes/Makefile.am:
* ext/ivorbis/Makefile.am:
* ext/jack/Makefile.am:
* ext/jp2k/Makefile.am:
* ext/ladspa/Makefile.am:
* ext/lcs/Makefile.am:
* ext/libfame/Makefile.am:
* ext/libmms/Makefile.am:
* ext/metadata/Makefile.am:
* ext/mpeg2enc/Makefile.am:
* ext/mplex/Makefile.am:
* ext/musepack/Makefile.am:
* ext/musicbrainz/Makefile.am:
* ext/mythtv/Makefile.am:
* ext/nas/Makefile.am:
* ext/neon/Makefile.am:
* ext/ofa/Makefile.am:
* ext/polyp/Makefile.am:
* ext/resindvd/Makefile.am:
* ext/sdl/Makefile.am:
* ext/shout/Makefile.am:
* ext/snapshot/Makefile.am:
* ext/sndfile/Makefile.am:
* ext/soundtouch/Makefile.am:
* ext/spc/Makefile.am:
* ext/swfdec/Makefile.am:
* ext/tarkin/Makefile.am:
* ext/theora/Makefile.am:
* ext/timidity/Makefile.am:
* ext/twolame/Makefile.am:
* ext/x264/Makefile.am:
* ext/xine/Makefile.am:
* ext/xvid/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/dshow/Makefile.am:
* gst/aiffparse/Makefile.am:
* gst/app/Makefile.am:
* gst/audiobuffer/Makefile.am:
* gst/bayer/Makefile.am:
* gst/cdxaparse/Makefile.am:
* gst/chart/Makefile.am:
* gst/colorspace/Makefile.am:
* gst/dccp/Makefile.am:
* gst/deinterlace/Makefile.am:
* gst/deinterlace2/Makefile.am:
* gst/dvdspu/Makefile.am:
* gst/festival/Makefile.am:
* gst/filter/Makefile.am:
* gst/flacparse/Makefile.am:
* gst/flv/Makefile.am:
* gst/games/Makefile.am:
* gst/h264parse/Makefile.am:
* gst/librfb/Makefile.am:
* gst/mixmatrix/Makefile.am:
* gst/modplug/Makefile.am:
* gst/mpeg1sys/Makefile.am:
* gst/mpeg4videoparse/Makefile.am:
* gst/mpegdemux/Makefile.am:
* gst/mpegtsmux/Makefile.am:
* gst/mpegvideoparse/Makefile.am:
* gst/mve/Makefile.am:
* gst/nsf/Makefile.am:
* gst/nuvdemux/Makefile.am:
* gst/overlay/Makefile.am:
* gst/passthrough/Makefile.am:
* gst/pcapparse/Makefile.am:
* gst/playondemand/Makefile.am:
* gst/rawparse/Makefile.am:
* gst/real/Makefile.am:
* gst/rtjpeg/Makefile.am:
* gst/rtpmanager/Makefile.am:
* gst/scaletempo/Makefile.am:
* gst/sdp/Makefile.am:
* gst/selector/Makefile.am:
* gst/smooth/Makefile.am:
* gst/smoothwave/Makefile.am:
* gst/speed/Makefile.am:
* gst/speexresample/Makefile.am:
* gst/stereo/Makefile.am:
* gst/subenc/Makefile.am:
* gst/tta/Makefile.am:
* gst/vbidec/Makefile.am:
* gst/videodrop/Makefile.am:
* gst/videosignal/Makefile.am:
* gst/virtualdub/Makefile.am:
* gst/vmnc/Makefile.am:
* gst/y4m/Makefile.am:
* sys/acmenc/Makefile.am:
* sys/cdrom/Makefile.am:
* sys/dshowdecwrapper/Makefile.am:
* sys/dshowsrcwrapper/Makefile.am:
* sys/dvb/Makefile.am:
* sys/dxr3/Makefile.am:
* sys/fbdev/Makefile.am:
* sys/oss4/Makefile.am:
* sys/qcam/Makefile.am:
* sys/qtwrapper/Makefile.am:
* sys/vcd/Makefile.am:
* sys/wininet/Makefile.am:
* win32/common/config.h:
Don't install static libs for plugins. Fixes #550851 for -bad.
 2011-01-02 14:30:07 +00:00
Jan Schmidt
051dba6c01 Fix compiler warnings on OS/X 
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (jack_process_cb):
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
Fix compiler warnings on OS/X
 2011-01-02 14:30:07 +00:00
Stefan Kost
e0d25d42db ext/jack/gstjackaudiosrc.c: Try committing this once again. Now properly renamed. 
Original commit message from CVS:
* ext/jack/gstjackaudiosrc.c:
Try committing this once again. Now properly renamed.
 2011-01-02 14:30:07 +00:00
Stefan Kost
37f51bb59d docs/plugins/: docs/plugins/inspect/plugin-jack.xml 
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.prerequisites:
* docs/plugins/inspect/plugin-jack.xml
Add new element to docs.
* ext/jack/gstjack.h
Add missing file.
* ext/jack/gstjackaudiosrc.c:
* ext/jack/gstjackaudiosrc.h:
Rename jackaudiosrc to jack_audio_src.
 2011-01-02 14:30:06 +00:00
Tristan Matthews
b34bd1425c ext/jack/: Add a jackaudiosrc. Refactor sink slightly for better code reuse. 
Original commit message from CVS:
patch by: Tristan Matthews <tristan@sat.qc.ca>
* ext/jack/Makefile.am:
* ext/jack/gstjack.c:
* ext/jack/gstjackaudioclient.c:
* ext/jack/gstjackaudiosink.c:
* ext/jack/gstjackaudiosink.h:
* ext/jack/gstjackaudiosrc.c:
* ext/jack/gstjackaudiosrc.h:
* ext/jack/gstjackringbuffer.h:
Add a jackaudiosrc. Refactor sink slightly for better code reuse.
Fixes #545197.
 2011-01-02 14:30:06 +00:00
Stefan Kost
d32bf24142 docs/plugins/: docs/plugins/inspect/plugin-mythtv.xml 
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.prerequisites:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-alsaspdif.xml:
* docs/plugins/inspect/plugin-amrwb.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-bayer.xml:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdaudio.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-dvb.xml:
* docs/plugins/inspect/plugin-dvdspu.xml:
* docs/plugins/inspect/plugin-faac.xml:
* docs/plugins/inspect/plugin-faad.xml:
* docs/plugins/inspect/plugin-fbdevsink.xml:
* docs/plugins/inspect/plugin-festival.xml:
* docs/plugins/inspect/plugin-filter.xml:
* docs/plugins/inspect/plugin-flvdemux.xml:
* docs/plugins/inspect/plugin-freeze.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-gstinterlace.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-interleave.xml:
* docs/plugins/inspect/plugin-jack.xml:
* docs/plugins/inspect/plugin-ladspa.xml:
* docs/plugins/inspect/plugin-metadata.xml:
* docs/plugins/inspect/plugin-mms.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-mpeg2enc.xml:
* docs/plugins/inspect/plugin-mpeg4videoparse.xml:
* docs/plugins/inspect/plugin-mpegtsparse.xml:
* docs/plugins/inspect/plugin-mpegvideoparse.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-mve.xml:
* docs/plugins/inspect/plugin-mythtv.xml
* docs/plugins/inspect/plugin-nas.xml:
* docs/plugins/inspect/plugin-neon.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-nuvdemux.xml:
* docs/plugins/inspect/plugin-oss4.xml
* docs/plugins/inspect/plugin-rawparse.xml:
* docs/plugins/inspect/plugin-real.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-rfbsrc.xml:
* docs/plugins/inspect/plugin-sdl.xml:
* docs/plugins/inspect/plugin-sdp.xml:
* docs/plugins/inspect/plugin-selector.xml:
* docs/plugins/inspect/plugin-sndfile.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-speexresample.xml:
* docs/plugins/inspect/plugin-stereo.xml:
* docs/plugins/inspect/plugin-subenc.xml
* docs/plugins/inspect/plugin-timidity.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-vcdsrc.xml:
* docs/plugins/inspect/plugin-videosignal.xml:
* docs/plugins/inspect/plugin-vmnc.xml:
* docs/plugins/inspect/plugin-wildmidi.xml:
* docs/plugins/inspect/plugin-x264.xml:
* docs/plugins/inspect/plugin-xvid.xml:
* docs/plugins/inspect/plugin-y4menc.xml:
* ext/amrwb/gstamrwbdec.c:
* ext/amrwb/gstamrwbenc.c:
* ext/amrwb/gstamrwbparse.c:
* ext/dc1394/gstdc1394.c:
* ext/directfb/dfbvideosink.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/mpeg2enc/gstmpeg2enc.cc:
* ext/mplex/gstmplex.cc:
* ext/musicbrainz/gsttrm.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* ext/timidity/gsttimidity.c:
* ext/timidity/gstwildmidi.c:
* gst-libs/gst/app/gstappsink.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/dvdspu/gstdvdspu.c:
* gst/festival/gstfestival.c:
* gst/freeze/gstfreeze.c:
* gst/interleave/deinterleave.c:
* gst/interleave/interleave.c:
* gst/modplug/gstmodplug.cc:
* gst/nuvdemux/gstnuvdemux.c:
Add missing elements to docs. Fix doc-markup: use convinience syntax
for examples (produces valid docbook), add several refsec2 when we
have several titles. Fix some types.
 2011-01-02 14:30:06 +00:00
Stefan Kost
8909b106f0 Do not use short_description in section docs for elements. We extract them from element details and there will be war... 
Original commit message from CVS:
* ext/dc1394/gstdc1394.c:
* ext/ivorbis/vorbisdec.c:
* ext/jack/gstjackaudiosink.c:
* ext/metadata/gstmetadatademux.c:
* ext/mythtv/gstmythtvsrc.c:
* ext/theora/theoradec.c:
* gst-libs/gst/app/gstappsink.c:
* gst/bayer/gstbayer2rgb.c:
* gst/deinterlace/gstdeinterlace.c:
* gst/rawparse/gstaudioparse.c:
* gst/rawparse/gstvideoparse.c:
* gst/rtpmanager/gstrtpbin.c:
* gst/rtpmanager/gstrtpclient.c:
* gst/rtpmanager/gstrtpjitterbuffer.c:
* gst/rtpmanager/gstrtpptdemux.c:
* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/gstrtpssrcdemux.c:
* gst/selector/gstinputselector.c:
* gst/selector/gstoutputselector.c:
* gst/videosignal/gstvideoanalyse.c:
* gst/videosignal/gstvideodetect.c:
* gst/videosignal/gstvideomark.c:
* sys/oss4/oss4-mixer.c:
* sys/oss4/oss4-sink.c:
* sys/oss4/oss4-source.c:
Do not use short_description in section docs for elements. We extract
them from element details and there will be warnings if they differ.
Also fixing up the ChangeLog order.
 2011-01-02 14:30:06 +00:00
Wim Taymans
102b2e3fc4 ext/jack/gstjackaudiosink.c: Include the element name in the port name to avoid duplicate port names. 
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c:
(gst_jack_audio_sink_allocate_channels):
Include the element name in the port name to avoid duplicate port names.
 2011-01-02 14:30:06 +00:00
Tim-Philipp Müller
d418db9aed ext/jack/gstjackaudiosink.c: Work around missing bits of thread-safety on older GLibs some more to avoid assertions w... 
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (gst_jack_audio_sink_class_init):
Work around missing bits of thread-safety on older GLibs some
more to avoid assertions when starting up multiple playbin
objects concurrently (see #512382).
 2011-01-02 14:30:06 +00:00
Sebastian Dröge
459484bc4d Use GST_LICENSE, GST_PACKAGE_NAME and GST_PACKAGE_ORIGIN instead of hardcoding values where possible. Fixes bug #522212. 
Original commit message from CVS:
* ext/alsaspdif/alsaspdifsink.c:
* ext/gsm/gstgsm.c:
* ext/jack/gstjack.c:
* ext/libmms/gstmms.c:
* ext/neon/gstneonhttpsrc.c:
* ext/shout/gstshout.c:
* ext/timidity/gsttimidity.c:
* ext/timidity/gstwildmidi.c:
* gst/nuvdemux/gstnuvdemux.c:
* gst/tta/gsttta.c:
Use GST_LICENSE, GST_PACKAGE_NAME and GST_PACKAGE_ORIGIN instead
of hardcoding values where possible. Fixes bug #522212.
 2011-01-02 14:30:06 +00:00
Stefan Kost
be5f69be58 ext/jack/gstjackaudiosink.c: Add stdlib include here too. 
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_acquire):
Add stdlib include here too.
 2011-01-02 14:30:06 +00:00
Stefan Kost
5b2d689774 ext/jack/gstjackaudiosink.c: Try t better name clients. properly handle return codes when re- establishing links. 
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_acquire):
Try t better name clients. properly handle return codes when re-
establishing links.
 2011-01-02 14:30:06 +00:00
Paul Davis
a70ffeed7c ext/jack/gstjackaudioclient.c: Don't need to take the connection lock, it will not be used and could cause deadlocks. 
Original commit message from CVS:
Based on patch by: Paul Davis <paul at linuxaudiosystems dot com>
* ext/jack/gstjackaudioclient.c: (gst_jack_audio_unref_connection):
Don't need to take the connection lock, it will not be used and could
cause deadlocks.
 2011-01-02 14:30:06 +00:00
Paul Davis
60bcffa5ef ext/jack/: Make an object to manage client connections to the jack server which we will use in the future to run sele... 
Original commit message from CVS:
Includes patch by: Paul Davis <paul at linuxaudiosystems dot com>
* ext/jack/Makefile.am:
* ext/jack/gstjackaudioclient.c: (gst_jack_audio_client_init),
(jack_process_cb), (jack_sample_rate_cb), (jack_buffer_size_cb),
(jack_shutdown_cb), (connection_find),
(gst_jack_audio_make_connection), (gst_jack_audio_get_connection),
(gst_jack_audio_unref_connection),
(gst_jack_audio_connection_add_client),
(gst_jack_audio_connection_remove_client),
(gst_jack_audio_client_new), (gst_jack_audio_client_free),
(gst_jack_audio_client_get_client),
(gst_jack_audio_client_set_active):
* ext/jack/gstjackaudioclient.h:
Make an object to manage client connections to the jack server which we
will use in the future to run selected jack elements with the same jack
connection.
Make some stuff a bit more threadsafe.
Activate the jack client ASAP.
* ext/jack/gstjackaudiosink.c:
(gst_jack_audio_sink_allocate_channels),
(gst_jack_audio_sink_free_channels), (jack_process_cb),
(gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_close_device),
(gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release),
(gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init),
(gst_jack_audio_sink_getcaps):
* ext/jack/gstjackaudiosink.h:
Use new client object to manage connections.
Don't remove and recreate all ports, try to reuse them.
 2011-01-02 14:30:06 +00:00
Wim Taymans
b0bfe6fcdd ext/jack/gstjackaudiosink.*: Improve docs. 
Original commit message from CVS:
* ext/jack/gstjackaudiosink.c: (jack_sample_rate_cb),
(jack_buffer_size_cb), (jack_shutdown_cb),
(gst_jack_ring_buffer_acquire):
* ext/jack/gstjackaudiosink.h:
Improve docs.
 2011-01-02 14:30:06 +00:00
Jan Schmidt
8beff1f58c ext/jack/.cvsignore: Ignore old files as requested by the build slave. 
Original commit message from CVS:
* ext/jack/.cvsignore:
Ignore old files as requested by the build slave.
 2011-01-02 14:30:06 +00:00
Wim Taymans
e2546f8275 ext/Makefile.am: Fix build. 
Original commit message from CVS:
* ext/Makefile.am:
Fix build.
* ext/jack/gstjackaudiosink.c: (jack_process_cb),
(jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb),
(gst_jack_ring_buffer_acquire):
Small cleanups.
 2011-01-02 14:30:06 +00:00
Wim Taymans
32aaf1e95d Added fully functional jackaudiosink. 
Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/jack/Makefile.am:
* ext/jack/gstjack.c: (plugin_init):
* ext/jack/gstjack.h:
* ext/jack/gstjackaudiosink.c: (gst_jack_ring_buffer_get_type),
(gst_jack_ring_buffer_class_init), (jack_process_cb),
(jack_sample_rate_cb), (jack_buffer_size_cb), (jack_shutdown_cb),
(gst_jack_ring_buffer_init), (gst_jack_ring_buffer_dispose),
(gst_jack_ring_buffer_finalize),
(gst_jack_ring_buffer_open_device),
(gst_jack_ring_buffer_close_device),
(gst_jack_ring_buffer_acquire), (gst_jack_ring_buffer_release),
(gst_jack_ring_buffer_start), (gst_jack_ring_buffer_pause),
(gst_jack_ring_buffer_stop), (gst_jack_ring_buffer_delay),
(gst_jack_connect_get_type), (gst_jack_audio_sink_base_init),
(gst_jack_audio_sink_class_init), (gst_jack_audio_sink_init),
(gst_jack_audio_sink_set_property),
(gst_jack_audio_sink_get_property), (gst_jack_audio_sink_getcaps),
(gst_jack_audio_sink_create_ringbuffer):
* ext/jack/gstjackaudiosink.h:
Added fully functional jackaudiosink.
 2011-01-02 14:30:06 +00:00
Stefan Kost
8fa70e3622 Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) 
Original commit message from CVS:
* ext/amrwb/gstamrwbdec.c: (gst_amrwbdec_class_init):
* ext/amrwb/gstamrwbenc.c: (gst_amrwbenc_class_init):
* ext/amrwb/gstamrwbparse.c: (gst_amrwbparse_class_init):
* ext/arts/gst_arts.c: (gst_arts_class_init):
* ext/artsd/gstartsdsink.c: (gst_artsdsink_class_init):
* ext/audiofile/gstafsink.c: (gst_afsink_class_init):
* ext/audiofile/gstafsrc.c: (gst_afsrc_class_init):
* ext/audioresample/gstaudioresample.c:
* ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init):
* ext/directfb/dfbvideosink.c: (gst_dfbvideosink_class_init):
* ext/divx/gstdivxdec.c: (gst_divxdec_class_init):
* ext/hermes/gsthermescolorspace.c:
(gst_hermes_colorspace_class_init):
* ext/ivorbis/vorbisfile.c: (gst_ivorbisfile_class_init):
* ext/jack/gstjack.c: (gst_jack_class_init):
* ext/jack/gstjackbin.c: (gst_jack_bin_class_init):
* ext/lcs/gstcolorspace.c: (gst_colorspace_class_init):
* ext/libfame/gstlibfame.c: (gst_fameenc_class_init):
* ext/musicbrainz/gsttrm.c: (gst_musicbrainz_class_init):
* ext/nas/nassink.c: (gst_nassink_class_init):
* ext/shout/gstshout.c: (gst_icecastsend_class_init):
* ext/snapshot/gstsnapshot.c: (gst_snapshot_class_init):
* ext/sndfile/gstsf.c: (gst_sf_class_init):
* ext/swfdec/gstswfdec.c: (gst_swfdecbuffer_class_init),
(gst_swfdec_class_init):
* ext/tarkin/gsttarkindec.c: (gst_tarkindec_class_init):
* ext/tarkin/gsttarkinenc.c: (gst_tarkinenc_class_init):
* gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_class_init):
* gst/chart/gstchart.c: (gst_chart_class_init):
* gst/colorspace/gstcolorspace.c: (gst_colorspace_class_init):
* gst/deinterlace/gstdeinterlace.c: (gst_deinterlace_class_init):
* gst/festival/gstfestival.c: (gst_festival_class_init):
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init):
* gst/filter/gstiir.c: (gst_iir_class_init):
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init):
* gst/librfb/gstrfbsrc.c: (gst_rfbsrc_class_init):
* gst/mixmatrix/mixmatrix.c: (gst_mixmatrix_class_init):
* gst/mpeg1sys/gstmpeg1systemencode.c:
(gst_system_encode_class_init):
* gst/mpeg1videoparse/gstmp1videoparse.c:
(gst_mp1videoparse_class_init):
* gst/mpeg2sub/gstmpeg2subt.c: (gst_mpeg2subt_class_init):
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_class_init):
* gst/overlay/gstoverlay.c: (gst_overlay_class_init):
* gst/passthrough/gstpassthrough.c: (passthrough_class_init):
* gst/playondemand/gstplayondemand.c: (play_on_demand_class_init):
* gst/rtjpeg/gstrtjpegdec.c: (gst_rtjpegdec_class_init):
* gst/rtjpeg/gstrtjpegenc.c: (gst_rtjpegenc_class_init):
* gst/smooth/gstsmooth.c: (gst_smooth_class_init):
* gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init):
* gst/spectrum/gstspectrum.c: (gst_spectrum_class_init):
* gst/stereo/gststereo.c: (gst_stereo_class_init):
* gst/switch/gstswitch.c: (gst_switch_class_init):
* gst/tta/gstttadec.c: (gst_tta_dec_class_init):
* gst/tta/gstttaparse.c: (gst_tta_parse_class_init):
* gst/vbidec/gstvbidec.c: (gst_vbidec_class_init):
* gst/videocrop/gstvideocrop.c: (gst_video_crop_class_init):
* gst/virtualdub/gstxsharpen.c: (gst_xsharpen_class_init):
* gst/y4m/gsty4mencode.c: (gst_y4mencode_class_init):
* sys/cdrom/gstcdplayer.c: (cdplayer_class_init):
* sys/directsound/gstdirectsoundsink.c:
(gst_directsoundsink_class_init):
* sys/dxr3/dxr3audiosink.c: (dxr3audiosink_class_init):
* sys/dxr3/dxr3spusink.c: (dxr3spusink_class_init):
* sys/dxr3/dxr3videosink.c: (dxr3videosink_class_init):
* sys/qcam/gstqcamsrc.c: (gst_qcamsrc_class_init):
* sys/v4l2/gstv4l2colorbalance.c:
(gst_v4l2_color_balance_channel_class_init):
* sys/v4l2/gstv4l2tuner.c: (gst_v4l2_tuner_channel_class_init),
(gst_v4l2_tuner_norm_class_init):
* sys/ximagesrc/ximagesrc.c: (gst_ximagesrc_class_init):
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
 2011-01-02 14:30:05 +00:00