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117 commits

Author SHA1 Message Date
Wim Taymans
bd89f2430b audiotestsrc: increase freq limit 
Raise the frequency limit and try to negotiate to a samplerate of 4*freq
when larger then the default samplerate.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=754450
 2015-11-02 15:54:19 +01:00
Wim Taymans
c688eb0d88 audiotestsrc: add support for unlimited number of channels 
Raise the channel limit and set the channel-mask for > 2 channels.
 2015-11-02 15:46:22 +01:00
Wim Taymans
b0bf294a62 audiotestsrc: add support for all formats 
Use the pack functions to also support the other audio formats we
have.
 2015-11-02 13:22:18 +01:00
Tim-Philipp Müller
ec5c93f169 docs: update element example pipelines 
- gst-launch -> gst-launch-1.0
- use autoaudiosink and audiovideosink more often
- review pipeline examples and descriptions
 2015-05-10 11:38:19 +01:00
Tim-Philipp Müller
c680e324bc Remove obsolete Android build cruft 
This is not needed any longer.
 2015-04-26 18:42:34 +01:00
Luis de Bethencourt
df08f5eabe remove unused enum items PROP_LAST 
This were probably added to the enums due to cargo cult programming and are
unused. Removing them.
 2015-04-24 17:11:01 +01:00
Sebastian Dröge
631d356845 audiotestsrc: Report our latency properly in live mode 
While we have no latency at all in theory, any other live source has the
duration of one buffer as minimum latency. Do the same in audiotestsrc.

https://bugzilla.gnome.org/show_bug.cgi?id=741879
 2014-12-24 12:59:37 +01:00
Sebastian Dröge
948a4a3632 gst: Add better support for static plugins 2013-04-15 15:52:58 +02:00
Stefan Sauer
fbf2647f3e audiotestsrc: fix a comment typo from previous commit 2013-03-29 17:16:17 +01:00
Stefan Sauer
f68c95ebaa audiotestssrc: truncate the seek pos to the sample and round the time 
Before it was done the other way around and that can trigger the assert that
already is in place. This also makes more sense; when seeking to time x, we want
then sample that is <= that pos.
 2013-03-29 16:46:14 +01:00
Stefan Sauer
8c390fe80a audiotestsrc: simplify the caps 
Drop channel-mask as we only do mon/stereo and channel-mask is optional in these
cases.
 2013-03-25 16:47:02 +01:00
Simon Berg
f18d2a5a9a audiotestsrc: fix rounding errors that might cause segments to be one sample too short 
https://bugzilla.gnome.org/show_bug.cgi?id=676884
 2013-03-24 20:53:05 +00:00
Simon Berg
d8b42e993b audiotestsrc: fix buffer size of last buffer 
The last buffer before EOS may be smaller than the maximum
size. The current code doesn't adjust for this, it only sets
the duration and offsets.

https://bugzilla.gnome.org/show_bug.cgi?id=696411
 2013-03-24 20:53:05 +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
Sebastian Dröge
3c1041d5eb Revert "gst: Add better support for static plugins" 
This reverts commit d2d79e3bc2,
which was accidentially pushed.
 2012-10-24 13:26:26 +02:00
Sebastian Dröge
d2d79e3bc2 gst: Add better support for static plugins 2012-10-24 12:10:44 +02:00
Mark Nauwelaerts
c629a44162 replace gst_tag_list_free with gst_tag_list_unref 2012-09-14 17:53:21 +02:00
Sebastian Dröge
99d73c94e9 tag: Update for taglist/tag event API changes 2012-07-28 00:35:02 +02:00
Wim Taymans
a2172bdb4b update for tag event change 2012-06-06 13:05:47 +02:00
Tim-Philipp Müller
3c6a3ad629 Use new gst_element_class_set_static_metadata() 2012-04-10 00:45:16 +01:00
Sebastian Dröge
ad42b16375 gst: Update for GST_PLUGIN_DEFINE() API change 2012-04-05 15:11:05 +02:00
Sebastian Dröge
65307dd132 gst: Update versioning 2012-04-04 14:55:15 +02:00
Wim Taymans
25137962ad fix for caps API changes 2012-03-11 19:04:41 +01:00
Wim Taymans
fcdc385aa1 port to new map API 2012-01-25 12:30:53 +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
Havard Graff
95be60de15 Fix various unlikely, but still potential memoryleaks in error code paths 
https://bugzilla.gnome.org/show_bug.cgi?id=667311
 2012-01-05 13:27:23 +00:00
Sebastian Dröge
2db0238450 audiotestsrc: Fix channel-mask handling 2012-01-05 10:34:25 +01:00
Sebastian Dröge
5bdf6b3383 gst: Add new layout field to the raw audio caps 2012-01-05 10:34:25 +01:00
Vincent Penquerc'h
96374054ac various: fix pad template leaks 
https://bugzilla.gnome.org/show_bug.cgi?id=662664
 2011-11-28 13:09:02 +00:00
Wim Taymans
d0bd5f04c0 update for new scheduling query 2011-11-18 17:58:58 +01:00
Stefan Sauer
0019bcaa47 controller: port to new location and api changes 2011-11-04 20:14:54 +01:00
Tim-Philipp Müller
5ee51e47a1 ext, gst, gst-libs, tests: update for tag list API changes 2011-10-31 14:22:39 +00:00
Tim-Philipp Müller
a586547b0c audiotestsrc: fix crash when setting the wave property before having negotiated a format 
https://bugzilla.gnome.org/show_bug.cgi?id=661911
 2011-10-17 15:47:31 +01:00
Thiago Santos
6eb5f5b13e audiotestsrc: update blocksize when caps or samples-per-buffer change 
Blocksize needs to be updated so we get a correct size buffer on
_fill function.
 2011-10-10 12:31:46 -03:00
Wim Taymans
f1088ed647 update for UNEXPECTED -> EOS flowreturn 2011-10-10 11:39:52 +02:00
Wim Taymans
73b894107a Merge branch 'master' into 0.11 
Conflicts:
	ext/vorbis/gstvorbisdec.c
	ext/vorbis/gstvorbisenc.c
	ext/vorbis/gstvorbisenc.h
	gst/audiotestsrc/gstaudiotestsrc.c
 2011-10-08 10:19:06 +02:00
Vincent Penquerc'h
70239887e8 audiotestsrc: add missing break 
And make violet noise usable

https://bugzilla.gnome.org/show_bug.cgi?id=661105
 2011-10-06 20:45:09 +02:00
Stefan Sauer
7ce811f1ed auditestsrc: indent fix 2011-10-04 23:10:05 +02:00
Sebastian Dröge
0f654f3feb Merge branch 'master' into 0.11 
Conflicts:
	docs/libs/Makefile.am
	tests/check/elements/decodebin2.c
 2011-09-08 14:42:00 +02:00
Stefan Sauer
abc96efb2a docs: add two mising enum docs 2011-09-07 14:14:02 +02:00
Wim Taymans
33196cdd2c audio: change audio format syntax a little 
Remove the _ in front of the endianness prefix.
Remove the _3 postfix for the 24 bits formats.
Add a _32 postfix after the formats that occupy extra space beyond their
natural size.
The result is that the GST_AUDIO_NE() macro can simply append the endianness
after all formats and that we only specify a different sample width when it is
different from the natural size of the sample. This makes things more consistent
and follows the pulseaudio conventions instead of the alsa ones.
 2011-09-06 12:06:39 +02:00
Wim Taymans
81457756f0 audiotestsrc: use base class fill method 2011-08-25 13:21:14 +02:00
Wim Taymans
b0b6d9124d audiotestsrc: fix build 2011-08-24 11:05:05 +02:00
Wim Taymans
2ce5c8b8be audio: use convert audio helper 2011-08-22 16:21:02 +02:00
Wim Taymans
dae848818d audio: rework audio caps. 
Rework the audio caps similar to the video caps. Remove
width/depth/endianness/signed fields and replace with a simple string
format and media type audio/x-raw.
Create a GstAudioInfo and some helper methods to parse caps.
Remove duplicate code from the ringbuffer and replace with audio info.
Use AudioInfo in the base audio filter class.
Port elements to new API.
 2011-08-18 19:15:03 +02:00
Wim Taymans
0290df6fc5 audiotestsrc: properly override fixate 2011-08-17 17:22:03 +02:00
Tim-Philipp Müller
dd56714b14 ffmpegcolorspace -> videoconvert 2011-07-07 23:59:59 +01:00
Wim Taymans
40d567153a Merge branch 'master' into 0.11 2011-06-13 19:09:05 +02:00
David Schleef
4db89c82bb convert M_PI to G_PI, for msvc 2011-06-10 23:56:34 -07:00
Sebastian Dröge
bf08ca7020 Merge branch 'master' into 0.11 2011-05-26 13:54:09 +02:00