Commit graph

95 commits

Author SHA1 Message Date
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
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
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
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
Stefan Kost
5cd0e0f666 audiotestsrc: add blue and violet noise by using spectral inversion
Add blue and violet noise by spectral inversion of pink and red noise.
Fixes #649969
2011-05-26 00:18:55 +03:00
Stefan Kost
1cf831e74e audiotestsrc: add red (brownian) noise generator
Add another noise generator which produces a quite dark noise color.

Fixes parts of #649969.
2011-05-25 23:43:56 +03:00
Wim Taymans
010add200a scheduling: port to new scheduling query 2011-05-24 17:37:45 +02:00
Sebastian Dröge
318ed07598 Revert "-base_port to new query API"
This reverts commit c9f4e0676b.
2011-05-17 11:25:31 +02:00
Wim Taymans
94dfe80f71 -base: port to new SEGMENT API 2011-05-16 13:48:11 +02:00
Wim Taymans
c9f4e0676b -base_port to new query API 2011-05-10 18:39:07 +02:00
Wim Taymans
ec57868488 -base: don't use buffer caps
Port to newest 0.11 core API, remove GST_PAD_CAPS and GST_BUFFER_CAPS.
2011-05-09 13:05:12 +02:00
Wim Taymans
86a4771f8e remove buffer_alloc 2011-04-29 13:28:17 +02:00
Sebastian Dröge
f10a8f0986 gst: Use G_DEFINE_TYPE instead of GST_BOILERPLATE 2011-04-19 11:35:53 +02:00
Wim Taymans
3b03e23559 plugins: port some plugins to the new memory API 2011-03-27 16:35:28 +02:00
Leo Singer
82199c5815 audiotestsrc: each element gets its own instance of GRand, if needed
As a result, pipelines that contain multiple instances of audiotestsrc
with the 'wave' property set to 'white-noise', 'pink-noise', or
'gaussian-noise' will run much faster, since they won't be competing
for access to the global, lock-protected instance of GRand.

Fixes bug #642720.
2011-02-19 08:37:46 +01:00
Stefan Kost
45b39fcfc1 audiotestsrc: swap timestamps in forward and reverse mode.
In reverse mode we want use the next next timestamp (and not the other way
around). Fixes the tests again. Also readd a log line that was dropped with
previous commit.
2010-04-03 22:52:01 +03:00
Stefan Kost
718edb5c14 audiotestsrc: implement reverse playback
Support playback at negative rates. When having a GstController assigned, the
element will produce time dependend output.
2010-04-02 21:04:37 +03:00