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webrtc_m130/modules/audio_processing
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Alex Loiko 9d2788f745 Make possible to activate adaptive AGC2 in the APM. 
We update the configuration settings for AGC2. We also update their
effects. Now, 'gain_controller2.enable=true' means 'first run Adaptive
AGC2; then run AGC2 limiter'.

Previously, only the AGC2 limiter was implemented. To run that, one
had to set both 'gain_controller2.enable=true' and
'gain_controller2.enable_limiter=true'.

This setting also enables adaptive AGC2 in the test tool 'audioproc_f'.

Bug: webrtc:7494
Change-Id: I0d5dfe443f2cdc0ecf3aa4054442dab6276d284d
Reviewed-on: https://webrtc-review.googlesource.com/64990
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22669}
2018-03-29 09:42:07 +00:00
..
aec
Delete unneeded includes of wav_file.h and file_wrapper.h.
2018-03-20 15:59:27 +00:00
aec3
Allow the headset mode to be entered after the call has started
2018-03-28 17:28:46 +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
Make possible to activate adaptive AGC2 in the APM.
2018-03-29 09:42:07 +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_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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_frame_view_unittest.cc
Comments in FixedDigitalLevelEstimator.
2018-02-16 14:17:08 +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_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_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
Level estimation and saturation protection stub.
2018-03-28 08:41:45 +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_unittest.cc
Use AudioProcessingBuilder everywhere AudioProcessing is created.
2018-01-09 13:45:20 +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_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_unittest.cc
Make possible to activate adaptive AGC2 in the APM.
2018-03-29 09:42:07 +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
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_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
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_unittest.cc
Make the echo detector injectable.
2018-01-11 15:43:01 +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
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
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
splitting_filter_unittest.cc
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +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
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