Niels De Graef
a1ef6a1179
audioresample: Use G_DECLARE_FINAL_TYPE
2020-03-16 15:47:58 +00:00
Wim Taymans
524ea147cc
audio-resampler: improve filter construction
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Remove some unused variables from the inner product functions.
Make filter coefficients by interpolating if required.
Rename some fields.
Try hard to not recalculate filters when just chaging the rate.
Add more proprties to audioresample.
2016-03-28 13:25:52 +02:00
Wim Taymans
1d9a793545
audio-converter: more work on resampling
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- Fix the resampler in the audio converter
- fix memory leaks
2016-03-28 13:13:59 +02:00
Wim Taymans
75d668e152
audio-converter: add resampler
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Add a resampler to the processing chain when needed.
port the audio resampler to the new audioconverter library
2016-03-28 13:13:59 +02:00
Peter G. Baum
0b4abc267e
audioresample: remove unused variables
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https://bugzilla.gnome.org/show_bug.cgi?id=738026
2014-10-07 14:59:10 +03:00
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
Carlos Rafael Giani
c41faa3d8e
audioresample: sinc filter performance improvements
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Original idea comes from Jyri Sarha
( http://lists.xiph.org/pipermail/speex-dev/2011-September/008243.html ).
Patch was discovered by Branislav Katreniak
( branislav.katreniak@streamunlimited.com ) for StreamUnlimited
( http://streamunlimited.com/ ). Tests showed up to 5x speed increase in
the resampler in the 44.1<->48kHz case.
I added the sinc-filter-mode and sinc-filter-auto-threshold properties
and the auto mode threshold tests, and adapted the code to GStreamer 1.0.
Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
2012-10-25 14:03:52 +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
Leo Singer
25a154be5f
audioresample: changed num_gap_samples, num_nongap_samples from guint32 to guint64 so that gaps of greater than or equal to 2^32 samples do not cause integer overflow
2010-12-17 19:34:42 +01:00
Leo Singer
aac8b21678
audioresample: renamed count_gap, count_nongap to more descriptive num_gap_samples, num_nongap_samples
2010-12-17 19:34:42 +01:00
Leo Singer
b4cd3329a9
Revert "Revert "audioresample: Add GAP flag support""
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This reverts commit 35c76b3409
.
Conflicts:
gst/audioresample/gstaudioresample.c
gst/audioresample/gstaudioresample.h
2010-12-17 19:34:41 +01:00
Mark Nauwelaerts
a7cf165289
audioresample: provide as much valid output ts and offset as valid input
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... by independently tracking time and offset, rather than having no offset
leading to no output ts.
2010-12-13 10:10:15 +01:00
Sebastian Dröge
35c76b3409
Revert "audioresample: Add GAP flag support"
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This reverts commit 129af0d8e6
.
This shouldn't be committed at all, it isn't ready and apparently
was in the wrong branch locally.
2010-09-15 11:28:29 +02:00
Leo Singer
129af0d8e6
audioresample: Add GAP flag support
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Fixes bug #586570 .
2010-09-15 11:01:45 +02:00
Kipp Cannon
a69068d70d
audioresample: Fix timestamp drift
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Fixes bug #591934 .
2009-08-26 09:10:17 +02:00
Sebastian Dröge
5dfcb63252
Rename files and types from speexresample to audioresample
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Rename files and types from speexresample to audioresample
to finish the move and to prevent any confusion.
2009-01-23 12:33:41 +01:00
Julien Moutte
6940042ecf
gst/audioresample/gstaudioresample.c: Handle discontinuous streams.
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Original commit message from CVS:
2007-03-14 Julien MOUTTE <julien@moutte.net>
* gst/audioresample/gstaudioresample.c: (gst_audioresample_init),
(audioresample_transform_size), (audioresample_do_output),
(audioresample_transform), (audioresample_pushthrough): Handle
discontinuous streams.
* gst/audioresample/gstaudioresample.h:
* tests/check/elements/audioresample.c:
(test_discont_stream_instance), (GST_START_TEST),
(audioresample_suite): Add a test for discontinuous streams.
* win32/common/config.h: Updated.
2007-03-14 17:16:30 +00:00
Stefan Kost
131fb86b4b
Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
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Original commit message from CVS:
* ext/alsa/gstalsamixerelement.h:
* ext/alsa/gstalsamixeroptions.h:
* ext/alsa/gstalsamixertrack.h:
* ext/gnomevfs/gstgnomevfssink.h:
* ext/gnomevfs/gstgnomevfssrc.h:
* ext/theora/gsttheoradec.h:
* ext/theora/gsttheoraenc.h:
* ext/theora/gsttheoraparse.h:
* ext/vorbis/vorbisparse.h:
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstaudiofilter.h:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
* gst/audioconvert/gstaudioconvert.h:
* gst/audioresample/gstaudioresample.h:
* gst/audiotestsrc/gstaudiotestsrc.h:
* gst/ffmpegcolorspace/gstffmpegcolorspace.h:
* gst/playback/gststreamselector.h:
* gst/tcp/gstmultifdsink.h:
* gst/tcp/gsttcpclientsink.h:
* gst/tcp/gsttcpclientsrc.h:
* gst/tcp/gsttcpserversink.h:
* gst/tcp/gsttcpserversrc.h:
* gst/videorate/gstvideorate.h:
* gst/videoscale/gstvideoscale.h:
* gst/videotestsrc/gstvideotestsrc.h:
* gst/volume/gstvolume.h:
* sys/v4l/gstv4ljpegsrc.h:
* sys/v4l/gstv4lmjpegsink.h:
* sys/v4l/gstv4lmjpegsrc.h:
* sys/v4l/gstv4lsrc.h:
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.h:
* tests/old/testsuite/alsa/sinesrc.h:
Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
2006-06-01 19:19:51 +00:00
Wim Taymans
af09257fd0
docs/plugins/: Add audioresample to docs.
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Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
Add audioresample to docs.
* gst/audioconvert/gstaudioconvert.c:
Add revision date.
* gst/audioresample/gstaudioresample.c:
(gst_audioresample_base_init), (gst_audioresample_class_init),
(gst_audioresample_init), (gst_audioresample_dispose),
(audioresample_get_unit_size), (audioresample_transform_caps),
(resample_set_state_from_caps), (audioresample_transform_size),
(audioresample_set_caps), (audioresample_event),
(audioresample_do_output), (audioresample_transform),
(audioresample_pushthrough), (gst_audioresample_set_property),
(gst_audioresample_get_property), (plugin_init):
* gst/audioresample/gstaudioresample.h:
Added docs.
Small code cleanups.
2006-03-02 18:23:55 +00:00
Wim Taymans
e3a77670f0
gst/audioresample/: Fix audioresample, seek torture, new segments, reverse negotiation etc.. work fine.
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Original commit message from CVS:
* gst/audioresample/buffer.c: (audioresample_buffer_queue_flush):
* gst/audioresample/buffer.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c: (resample_input_flush),
(resample_input_pushthrough), (resample_input_eos),
(resample_get_output_size_for_input),
(resample_get_input_size_for_output), (resample_get_output_size),
(resample_get_output_data):
* gst/audioresample/resample.h:
* gst/audioresample/resample_ref.c: (resample_scale_ref):
Fix audioresample, seek torture, new segments, reverse negotiation
etc.. work fine.
2005-12-02 11:34:50 +00:00
Thomas Vander Stichele
6dff9c2cbd
check/: add a test for audioconvert
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Original commit message from CVS:
* check/Makefile.am:
* check/elements/audioconvert.c: (setup_audioconvert),
(cleanup_audioconvert), (get_int_caps), (verify_convert),
(GST_START_TEST), (audioconvert_suite), (main):
add a test for audioconvert
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b);
note that for buffers of 1/3 sec this means DURATION(c) is
one nanosecond more than for a and b
2005-08-25 17:20:02 +00:00
Thomas Vander Stichele
752a59192c
port audioresample to basetransform
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Original commit message from CVS:
port audioresample to basetransform
2005-08-24 14:08:58 +00:00
David Schleef
ae8f41b658
gst/audioresample/Makefile.am: Leet audioresampling code
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Original commit message from CVS:
* gst/audioresample/Makefile.am: Leet audioresampling code
* gst/audioresample/buffer.c:
* gst/audioresample/buffer.h:
* gst/audioresample/debug.c:
* gst/audioresample/debug.h:
* gst/audioresample/functable.c:
* gst/audioresample/functable.h:
* gst/audioresample/gstaudioresample.c:
* gst/audioresample/gstaudioresample.h:
* gst/audioresample/resample.c:
* gst/audioresample/resample.h:
* gst/audioresample/resample_chunk.c:
* gst/audioresample/resample_functable.c:
* gst/audioresample/resample_ref.c:
2005-08-23 19:29:38 +00:00