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webrtc/modules/audio_processing
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Sam Zackrisson a955849901 Add APM config flag for legacy moderate suppression level in AEC2 
This will be hooked up in clients who need to keep using the moderate
suppression level in AEC2 until other tuning options are available.

Bug: webrtc:9535
Change-Id: I6c40898954d9c856f58bcea87271f4b98fa124de
Reviewed-on: https://webrtc-review.googlesource.com/94148
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24292}
2018-08-15 14:56:06 +00:00
..
aec Explicitly add -mfpu=neon to all targets that use NEON 2018-08-01 13:15:42 +00:00
aec3 AEC3: Enforcing nonlinear mode when transparent mode is active 2018-08-12 20:40:04 +00:00
aec_dump Add UTC time to init event in AEC debug dump. 2018-08-11 20:29:07 +00:00
aecm Explicitly add -mfpu=neon to all targets that use NEON 2018-08-01 13:15:42 +00:00
agc Gain metrics for digital adaptive AGC. 2018-08-15 13:44:46 +00:00
agc2 Gain metrics for digital adaptive AGC. 2018-08-15 13:44:46 +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 Add APM config flag for legacy moderate suppression level in AEC2 2018-08-15 14:56:06 +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 Move fft4g to proper third_party directory 2018-07-25 15:44:53 +00:00
test Optionally disable digital gain control in ExperimentalAgc. 2018-08-09 13:37:30 +00:00
transient Fix MovingMoments::CalculateMoments. 2018-07-31 15:08:12 +00:00
utility Explicitly add -mfpu=neon to all targets that use NEON 2018-08-01 13:15:42 +00:00
vad Move fft4g to proper third_party directory 2018-07-25 15:44:53 +00:00
audio_buffer.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_buffer.h Delete root header file typedef.h. 2018-07-25 14:59:26 +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 Add UTC time to init event in AEC debug dump. 2018-08-11 20:29:07 +00:00
audio_processing_impl.h Revert "Add one-stop-shop for built-in AEC toggling in APM" 2018-07-23 14:48:17 +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 APM: render pre-processor moved before echo detector queuing. 2018-08-09 14:40:31 +00:00
audio_processing_performance_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
audio_processing_unittest.cc Add UTC time to init event in AEC debug dump. 2018-08-11 20:29:07 +00:00
BUILD.gn Add UTC time to init event in AEC debug dump. 2018-08-11 20:29:07 +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 Add UTC time to init event in AEC debug dump. 2018-08-11 20:29:07 +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 Turn off comfort noise generation by default in AECM 2018-07-24 08:52:36 +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 Atomically increment GainControl instance counter. 2018-08-15 07:44:00 +00:00
gain_control_for_experimental_agc.h Reset Agc2 on analog gain changes. 2018-08-08 14:36:37 +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 Reset Agc2 on analog gain changes. 2018-08-08 14:36:37 +00:00
gain_controller2.h Reset Agc2 on analog gain changes. 2018-08-08 14:36:37 +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 Adding alessiob@ and minyue@ as owners of APM. 2018-07-02 07:45:31 +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 Delete root header file typedef.h. 2018-07-25 14:59:26 +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 Delete root header file typedef.h. 2018-07-25 14:59:26 +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