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
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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
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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.
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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
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
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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
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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"
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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
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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
6e160bed3d
Merge branch 'master' into 0.11
...
Conflicts:
android/alsa.mk
android/app.mk
android/app_plugin.mk
android/audio.mk
android/audioconvert.mk
android/decodebin.mk
android/decodebin2.mk
android/gdp.mk
android/interfaces.mk
android/netbuffer.mk
android/pbutils.mk
android/playbin.mk
android/queue2.mk
android/riff.mk
android/rtp.mk
android/rtsp.mk
android/sdp.mk
android/tag.mk
android/tcp.mk
android/typefindfunctions.mk
android/video.mk
2011-04-11 11:37:51 +02:00
Alessandro Decina
030f639a8e
android: make it ready for androgenizer
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Remove the android/ top dir
Fixe the Makefile.am to be androgenized
To build gstreamer for android we are now using androgenizer which generates the
needed Android.mk files.
Androgenizer can be found here:
http://git.collabora.co.uk/?p=user/derek/androgenizer.git
2011-04-11 07:23:21 +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
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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.
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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
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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
Benjamin Otte
5e21fa5e0e
gst_element_class_set_details => gst_element_class_set_details_simple
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Also change my email from the old university one to the current one.
2010-03-16 17:41:50 +01:00
Wim Taymans
0bb9b75a75
audiotestsrc: call send_event directly
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We can't call gst_element_send_event() from a streaming thread as it gets the
state lock. Instead call the send_event method directly until we have a nice API
for this in basesrc.
Fixes #588746
2009-07-20 13:15:32 +02:00
Edward Hervey
196b38d4ef
audiotestsrc: Make sure tags are properly serialized. Fixes #588746
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We do this by letting the basesrc base class handle the tags.
2009-07-20 08:47:50 +02:00
Sebastian Dröge
dc706f7f2f
audiotestsrc: Name gaussian noise "gaussian-noise" instead of just "gaussian"
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Also make all the function arrays constant.
2009-06-21 19:43:18 +02:00
Kipp Cannon
620391b300
audiotestsrc: Add support for generating gaussian white noise
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This patch adds support for stationary white Gaussian noise.
The Box-Muller algorithm is used to generate pairs of independent
normally-distributed random numbers.
Fixes bug #586519 .
2009-06-21 12:29:03 +02:00
Tim-Philipp Müller
8d326479a5
audiotestsrc: fix broken enum nick - it should have a hyphen
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The enum nick should be 'sine-table', not 'sine table'. Technically this is
an API/ABI change I guess, but anyone who was using this and didn't report
it deserves this.
2009-05-12 17:18:37 +01:00
Tim-Philipp Müller
21228a6934
audiotestsrc: seek to the requested byte offset, not the expected byte offset
2009-05-12 15:32:02 +01:00