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platform-external-webrtc/modules/audio_processing
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Artem Titov 81f5197512 Fix pylint presubmit errors and warnings from untouched modules. 
BUG=None

Change-Id: I619dab14875e19477beb8bfb566ed1f34009c025
Reviewed-on: https://webrtc-review.googlesource.com/85960
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23754}
2018-06-27 09:31:29 +00:00
..
aec
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
aec3
AEC3: Reverb modeling: Including the freq shape of the tails when modeling the reverberation
2018-06-26 16:17:45 +00:00
aec_dump
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
aecm
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
agc
Fix overflow in digital AGC1
2018-06-26 10:31:09 +00:00
agc2
Add RNN-VAD to AGC2.
2018-06-20 15:04:06 +00:00
audio_generator
Add stub draft of audio generator to APM
2018-03-05 09:28:52 +00:00
echo_detector
Replace rtc::Optional with absl::optional in modules/audio processing
2018-06-19 10:38:56 +00:00
include
Remove nonlinear beamformer API from APM
2018-06-21 08:49:52 +00:00
intelligibility
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
logging
Remove stringstream usages from the APM
2018-04-06 14:17:03 +00:00
ns
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
test
Fix pylint presubmit errors and warnings from untouched modules.
2018-06-27 09:31:29 +00:00
transient
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
utility
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
vad
Extract fft4g into separate build target
2018-06-26 13:39:25 +00:00
audio_buffer_unittest.cc
…
audio_buffer.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_buffer.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_frame_view_unittest.cc
Add namespace 'webrtc' to AudioFrameView.
2018-05-14 12:33:49 +00:00
audio_processing_impl_locking_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_processing_impl_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_processing_impl.cc
Remove nonlinear beamformer API from APM
2018-06-21 08:49:52 +00:00
audio_processing_impl.h
Remove nonlinear beamformer API from APM
2018-06-21 08:49:52 +00:00
audio_processing_performance_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
audio_processing_unittest.cc
Remove non-API beamformer references
2018-06-19 08:29:24 +00:00
BUILD.gn
Extract fft4g into separate build target
2018-06-26 13:39:25 +00:00
common.h
…
config_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
debug.proto
Options and settings for the Pre-amplifier.
2018-04-16 12:25:48 +00:00
DEPS
…
echo_cancellation_bit_exact_unittest.cc
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echo_cancellation_impl_unittest.cc
Use AudioProcessingBuilder everywhere AudioProcessing is created.
2018-01-09 13:45:20 +00:00
echo_cancellation_impl.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
echo_cancellation_impl.h
Enforcing a stream delay of 0 to be assumed in the AEC on Chrome OS
2017-12-22 15:42:13 +00:00
echo_control_mobile_impl.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
echo_control_mobile_impl.h
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echo_control_mobile_unittest.cc
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gain_control_for_experimental_agc.cc
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gain_control_for_experimental_agc.h
…
gain_control_impl.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
gain_control_impl.h
Replace rtc::Optional with absl::optional in modules/audio processing
2018-06-19 10:38:56 +00:00
gain_control_unittest.cc
…
gain_controller2_unittest.cc
Set a positive initial gain in the Adaptive Digital GC.
2018-04-27 09:05:25 +00:00
gain_controller2.cc
Make possible to activate adaptive AGC2 in the APM.
2018-03-29 09:42:07 +00:00
gain_controller2.h
Make possible to activate adaptive AGC2 in the APM.
2018-03-29 09:42:07 +00:00
level_estimator_impl.cc
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level_estimator_impl.h
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level_estimator_unittest.cc
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low_cut_filter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
low_cut_filter.cc
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low_cut_filter.h
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noise_suppression_impl.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
noise_suppression_impl.h
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noise_suppression_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
OWNERS
Add ivoc@ and saza@ to audio_processing OWNERS
2018-06-14 12:18:07 +00:00
render_queue_item_verifier.h
…
residual_echo_detector_unittest.cc
Change echo detector to scoped_refptr
2018-06-14 09:51:41 +00:00
residual_echo_detector.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
residual_echo_detector.h
Add more parameters to the Initialize function of the echo detector.
2018-03-15 09:21:56 +00:00
rms_level_unittest.cc
Move some more numeric utility code from rtc_base/ to rtc_base/numerics/
2017-11-22 12:39:39 +00:00
rms_level.cc
Replace rtc::Optional with absl::optional in modules/audio processing
2018-06-19 10:38:56 +00:00
rms_level.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
splitting_filter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
splitting_filter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
splitting_filter.h
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three_band_filter_bank.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
three_band_filter_bank.h
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typing_detection.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
typing_detection.h
Remove dependencies on modules:module_api from AudioProcessing.
2018-04-12 22:05:27 +00:00
voice_detection_impl.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
voice_detection_impl.h
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voice_detection_unittest.cc
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