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webrtc_m130/modules/audio_processing/agc2
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Florent Castelli 0afde7614d Move webrtc::AudioProcessing include to api/ folder 
Bug: webrtc:15874
Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42128}
2024-04-20 07:02:50 +00:00
..
rnn_vad
Format /modules
2023-04-20 02:02:45 +00:00
adaptive_digital_gain_controller_unittest.cc
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
adaptive_digital_gain_controller.cc
Fix APM AGC2 initialization: pass the correct number of channels
2022-12-09 15:02:20 +00:00
adaptive_digital_gain_controller.h
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
agc2_common.h
AGC2 adaptive digital controller config clean-up
2022-12-09 13:07:34 +00:00
agc2_testing_common_unittest.cc
…
agc2_testing_common.cc
Use sinf instead of std::sinf to improve libstdc++ compatibility
2023-05-30 20:16:31 +00:00
agc2_testing_common.h
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biquad_filter_unittest.cc
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biquad_filter.cc
Optimize the AGC2 Biquad filter.
2022-06-20 16:05:51 +00:00
biquad_filter.h
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BUILD.gn
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
clipping_predictor_level_buffer_unittest.cc
ClippingPredictor/Evaluator/LevelBuffer and GainMap: Move to agc2
2022-10-05 08:35:42 +00:00
clipping_predictor_level_buffer.cc
ClippingPredictor/Evaluator/LevelBuffer and GainMap: Move to agc2
2022-10-05 08:35:42 +00:00
clipping_predictor_level_buffer.h
ClippingPredictor/Evaluator/LevelBuffer and GainMap: Move to agc2
2022-10-05 08:35:42 +00:00
clipping_predictor_unittest.cc
ClippingPredictor/Evaluator/LevelBuffer and GainMap: Move to agc2
2022-10-05 08:35:42 +00:00
clipping_predictor.cc
InputVolumeController: readability improvements
2022-12-02 12:49:04 +00:00
clipping_predictor.h
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
compute_interpolated_gain_curve.cc
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compute_interpolated_gain_curve.h
…
cpu_features.cc
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cpu_features.h
…
fixed_digital_level_estimator_unittest.cc
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fixed_digital_level_estimator.cc
…
fixed_digital_level_estimator.h
…
gain_applier_unittest.cc
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gain_applier.cc
…
gain_applier.h
…
gain_map_internal.h
InputVolumeController: readability improvements
2022-12-02 12:49:04 +00:00
input_volume_controller_unittest.cc
Add generic input volume controller test for both AGC1 and AGC2
2022-12-20 14:41:31 +00:00
input_volume_controller.cc
Unify AGC2 experiment field trials into one
2022-12-14 16:40:41 +00:00
input_volume_controller.h
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
input_volume_stats_reporter_unittest.cc
InputVolumeStatsReporter: replace WebRTC.Audio.AgcSetLevel
2022-12-08 15:44:43 +00:00
input_volume_stats_reporter.cc
Fix WebRTC.Audio.Apm.RecommendedInputVolume.OnChangeToMatchTarget tests
2022-12-12 19:26:01 +00:00
input_volume_stats_reporter.h
Fix WebRTC.Audio.Apm.RecommendedInputVolume.OnChangeToMatchTarget tests
2022-12-12 19:26:01 +00:00
interpolated_gain_curve_unittest.cc
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interpolated_gain_curve.cc
Adopt absl::string_view in modules/audio_processing/
2022-08-16 13:49:14 +00:00
interpolated_gain_curve.h
Adopt absl::string_view in modules/audio_processing/
2022-08-16 13:49:14 +00:00
limiter_db_gain_curve_unittest.cc
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limiter_db_gain_curve.cc
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limiter_db_gain_curve.h
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limiter_unittest.cc
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limiter.cc
Adopt absl::string_view in modules/audio_processing/
2022-08-16 13:49:14 +00:00
limiter.h
Adopt absl::string_view in modules/audio_processing/
2022-08-16 13:49:14 +00:00
noise_level_estimator_unittest.cc
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noise_level_estimator.cc
AGC2 NoiseLevelEstimator: DCHECK pointer
2022-12-06 15:14:04 +00:00
noise_level_estimator.h
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saturation_protector_buffer_unittest.cc
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saturation_protector_buffer.cc
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saturation_protector_buffer.h
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saturation_protector_unittest.cc
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saturation_protector.cc
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saturation_protector.h
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speech_level_estimator_unittest.cc
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
speech_level_estimator.cc
AGC2 adaptive digital controller config clean-up
2022-12-09 13:07:34 +00:00
speech_level_estimator.h
Move webrtc::AudioProcessing include to api/ folder
2024-04-20 07:02:50 +00:00
speech_probability_buffer_unittest.cc
Add SpeechProbabilityBuffer
2022-10-06 11:23:03 +00:00
speech_probability_buffer.cc
Add SpeechProbabilityBuffer
2022-10-06 11:23:03 +00:00
speech_probability_buffer.h
Add SpeechProbabilityBuffer
2022-10-06 11:23:03 +00:00
vad_wrapper_unittest.cc
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vad_wrapper.cc
AGC2 adaptive digital controller config clean-up
2022-12-09 13:07:34 +00:00
vad_wrapper.h
AGC2 adaptive digital controller config clean-up
2022-12-09 13:07:34 +00:00
vector_float_frame.cc
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vector_float_frame.h
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