admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/modules/audio_processing
History
Gustaf Ullberg 51f4014acd AEC3: Slower adaptation of main filter 
The main filter is adapted at a lower rate which reduces the risk of
diverging during double talk. The change yields notable transparency
improvements.

Bug: webrtc:9497
Change-Id: Ib23b7a4055d313dede535d2b65dc7e023a2db042
Reviewed-on: https://webrtc-review.googlesource.com/87300
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23858}
2018-07-05 14:37:27 +00:00
..
aec
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
aec3
AEC3: Slower adaptation of main filter
2018-07-05 14:37:27 +00:00
aec_dump
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
aecm
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
agc
Fix fuzzer-found overflow in AGC1
2018-07-03 09:56:34 +00:00
agc2
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
Flags and settings for AGC2 in AgcManagerDirect.
2018-07-02 13:20:39 +00:00
audio_processing_impl.h
Flags and settings for AGC2 in AgcManagerDirect.
2018-07-02 13:20:39 +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
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +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
…
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
…
echo_control_mobile_unittest.cc
…
gain_control_for_experimental_agc.cc
…
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
…
level_estimator_impl.h
…
level_estimator_unittest.cc
…
low_cut_filter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
low_cut_filter.cc
…
low_cut_filter.h
…
noise_suppression_impl.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
noise_suppression_impl.h
…
noise_suppression_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
OWNERS
Adding alessiob@ and minyue@ as owners of APM.
2018-07-02 07:45:31 +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
…
three_band_filter_bank.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
three_band_filter_bank.h
…
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
…
voice_detection_unittest.cc
…