Vineeth TM
44b70ca3a1
base: use new gst_element_class_add_static_pad_template()
...
https://bugzilla.gnome.org/show_bug.cgi?id=763075
2016-03-24 14:25:41 +02:00
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
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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
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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
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- 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
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This is not needed any longer.
2015-04-26 18:42:34 +01:00
Luis de Bethencourt
df08f5eabe
remove unused enum items PROP_LAST
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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
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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
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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
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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
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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
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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
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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"
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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
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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
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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
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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
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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
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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
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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
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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
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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.
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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