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webrtc/common_audio
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Artem Titov df3bcdbe88 Extract fft4g into separate build target 
common_audio/fft4g.c is third party codem that have to be moved into
third_party folder, so to be able to do it at first we have to extract
it into separate target. It is extracted with corresponding header file
and will be moved in futher CL.

Bug: webrtc:8366
Change-Id: I586ca94d4e9242c23163b987fa334dfa705020ed
Reviewed-on: https://webrtc-review.googlesource.com/85372
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23742}
2018-06-26 13:39:25 +00:00
..
include Add decibel conversion functions to //common_audio:common_audio 2018-02-16 10:46:48 +00:00
mocks Replace rtc::Optional with absl::optional 2018-06-21 09:32:56 +00:00
resampler Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
signal_processing Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
vad Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_converter.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_converter.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_converter_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_ring_buffer.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_ring_buffer.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_ring_buffer_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_util.cc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
audio_util_unittest.cc Add decibel conversion functions to //common_audio:common_audio 2018-02-16 10:46:48 +00:00
blocker.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
blocker.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
blocker_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
BUILD.gn Extract fft4g into separate build target 2018-06-26 13:39:25 +00:00
channel_buffer.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
channel_buffer.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
channel_buffer_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
DEPS Removing openmax_dl 2018-05-07 12:12:04 +00:00
fft4g.c Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
fft4g.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
fir_filter.h Fix circular deps in common_audio. 2017-11-17 11:20:17 +00:00
fir_filter_c.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
fir_filter_c.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
fir_filter_factory.cc Fix circular deps in common_audio. 2017-11-17 11:20:17 +00:00
fir_filter_factory.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
fir_filter_neon.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
fir_filter_neon.h Move aligned memory utilities to rtc_base/memory/ 2018-03-22 14:13:24 +00:00
fir_filter_sse.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
fir_filter_sse.h Move aligned memory utilities to rtc_base/memory/ 2018-03-22 14:13:24 +00:00
fir_filter_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
lapped_transform.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
lapped_transform.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
lapped_transform_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
real_fourier.cc Disabling openmax_dl 2018-04-24 13:08:56 +00:00
real_fourier.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
real_fourier_ooura.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
real_fourier_ooura.h FFT-based auto correlation. 2018-05-08 12:07:42 +00:00
real_fourier_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
ring_buffer.c Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
ring_buffer.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
ring_buffer_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
smoothing_filter.cc Replace rtc::Optional with absl::optional 2018-06-21 09:32:56 +00:00
smoothing_filter.h Replace rtc::Optional with absl::optional 2018-06-21 09:32:56 +00:00
smoothing_filter_unittest.cc Replacing rtc::TimeDelta with webrtc::TimeDelta. 2018-05-08 13:22:53 +00:00
sparse_fir_filter.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
sparse_fir_filter.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
sparse_fir_filter_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
wav_file.cc Remove unused WavFile::FormatAsString method. 2018-06-14 09:05:20 +00:00
wav_file.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
wav_file_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
wav_header.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
wav_header.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
wav_header_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
window_generator.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
window_generator.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
window_generator_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00