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webrtc/modules/audio_processing
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Alex Loiko cab48c391d Adaptive digital gain applier 
AGC2 component that computes and applies the digital gain.
The gain is computed from an estimated speech and noise level.
This component decides how fast the gain can change and what it
should be.

Bug: webrtc:7494
Change-Id: If55b6e5c765f958e433730cd9e3b2b93c14a7910
Reviewed-on: https://webrtc-review.googlesource.com/64985
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22741}
2018-04-05 06:40:02 +00:00
..
aec Delete unneeded includes of wav_file.h and file_wrapper.h. 2018-03-20 15:59:27 +00:00
aec3 Corrected the threshold for determining filter convergence in AEC3 2018-03-29 11:31:57 +00:00
aec_dump Move file_wrapper.h to rtc_base/system/ 2018-03-23 11:17:15 +00:00
aecm Use of unititialized value in AECM. 2018-01-25 15:09:14 +00:00
agc Fixing -Wstrict-prototypes warnings. 2018-03-19 16:57:21 +00:00
agc2 Adaptive digital gain applier 2018-04-05 06:40:02 +00:00
audio_generator Add stub draft of audio generator to APM 2018-03-05 09:28:52 +00:00
beamformer Fix macro clash with _USE_MATH_DEFINES. 2017-12-13 09:39:20 +00:00
echo_detector Optional: Use nullopt and implicit construction in /modules/audio_processing 2017-11-20 10:19:30 +00:00
include Make possible to activate adaptive AGC2 in the APM. 2018-03-29 09:42:07 +00:00
intelligibility Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
logging Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
ns Fixing -Wstrict-prototypes warnings. 2018-03-19 16:57:21 +00:00
test Add a specific AEC3 behavior for setups with known clock-drift 2018-03-28 16:51:57 +00:00
transient Move file_wrapper.h to rtc_base/system/ 2018-03-23 11:17:15 +00:00
utility Fix typo in the include path of ooura_fft.h 2018-01-11 07:57:40 +00:00
vad Adaptive Digital gain control structure. 2018-03-27 14:12:50 +00:00
audio_buffer.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_buffer.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +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 Comments in FixedDigitalLevelEstimator. 2018-02-16 14:17:08 +00:00
audio_processing_impl.cc Move file_wrapper.h to rtc_base/system/ 2018-03-23 11:17:15 +00:00
audio_processing_impl.h Move file_wrapper.h to rtc_base/system/ 2018-03-23 11:17:15 +00:00
audio_processing_impl_locking_unittest.cc Use AudioProcessingBuilder everywhere AudioProcessing is created. 2018-01-09 13:45:20 +00:00
audio_processing_impl_unittest.cc Change return types of refcount methods. 2017-10-20 07:46:03 +00:00
audio_processing_performance_unittest.cc Use AudioProcessingBuilder everywhere AudioProcessing is created. 2018-01-09 13:45:20 +00:00
audio_processing_unittest.cc Reland "Deprecate the adaptive level controller" 2018-03-09 09:42:13 +00:00
BUILD.gn Noise level estimation for AGC2. 2018-04-04 18:23:55 +00:00
common.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
config_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
debug.proto Moving src/webrtc into src/. 2017-09-15 04:25:06 +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 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.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 Streamline error handling and logging in the audio processing module 2018-02-15 15:06:26 +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 Optional: Use nullopt and implicit construction in /modules/audio_processing 2017-11-20 10:19:30 +00:00
gain_control_impl.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +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 Make possible to activate adaptive AGC2 in the APM. 2018-03-29 09:42:07 +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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
noise_suppression_impl.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Add gustaf to audio_processing OWNERS 2018-02-06 10:54:29 +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 Add more parameters to the Initialize function of the echo detector. 2018-03-15 09:21:56 +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 Make the echo detector injectable. 2018-01-11 15:43:01 +00:00
rms_level.cc Optional: Use nullopt and implicit construction in /modules/audio_processing 2017-11-20 10:19:30 +00:00
rms_level.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
three_band_filter_bank.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +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 Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
typing_detection.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
voice_detection_impl.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +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