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webrtc/modules/audio_processing
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Jesús de Vicente Peña 2e79d2b398 AEC3: Misadjustment estimator of the linear filter. 
In this work the performance of the linear filter is
estimated. The estimation aims at capture situations when the linear
filter is largely over-estimating the echo. In those circumstances,
the linear filter is scaled with the purpose of accelerating its
convergence.

Change-Id: I05ea3739d82838a6f08673432da92125c47943e0
Bug: webrtc:9466,chromium:857426
Reviewed-on: https://webrtc-review.googlesource.com/86133
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23789}
2018-06-29 15:05:14 +00:00
..
aec Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
aec3 AEC3: Misadjustment estimator of the linear filter. 2018-06-29 15:05:14 +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.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_buffer_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_frame_view_unittest.cc Add namespace 'webrtc' to AudioFrameView. 2018-05-14 12:33:49 +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_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_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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
echo_cancellation_bit_exact_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +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_cancellation_impl_unittest.cc Use AudioProcessingBuilder everywhere AudioProcessing is created. 2018-01-09 13:45:20 +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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
echo_control_mobile_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
gain_control_for_experimental_agc.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
gain_control_for_experimental_agc.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +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
gain_controller2_unittest.cc Set a positive initial gain in the Adaptive Digital GC. 2018-04-27 09:05:25 +00:00
level_estimator_impl.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
level_estimator_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
level_estimator_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
low_cut_filter.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
low_cut_filter.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
low_cut_filter_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
noise_suppression_impl.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
noise_suppression_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +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
residual_echo_detector_unittest.cc Change echo detector to scoped_refptr 2018-06-14 09:51:41 +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
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
splitting_filter.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
splitting_filter.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
splitting_filter_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
three_band_filter_bank.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
three_band_filter_bank.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
voice_detection_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00