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webrtc_m130/modules/audio_processing/agc2
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Tommi 51ad7c1277 Update FrameCombiner et al to use DeinterleavedView 
* FrameCombiner is simpler. No additional channel pointers for buffers.
* Improve consistency in using views in downstream classes.
* Deprecate older methods (some have upstream dependencies).
* Use samples per channel instead of sample rate where the former is
  really what's needed.

Bug: chromium:335805780
Change-Id: I0dde8ed7a5a187bbddd18d3b6c649aa0865e6d4a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352582
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42575}
2024-07-02 15:58:20 +00:00
..
rnn_vad
Mass removal of absl_deps in all BUILD.gn files
2024-05-23 15:09:46 +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: retuning and large refactoring
2021-04-14 19:01:01 +00:00
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
AGC2 limiter: faster recovery
2021-11-19 10:00:21 +00:00
biquad_filter_unittest.cc
BiQuadFilter: API improvements
2021-10-28 14:04:09 +00:00
biquad_filter.cc
Optimize the AGC2 Biquad filter.
2022-06-20 16:05:51 +00:00
biquad_filter.h
BiQuadFilter: API improvements
2021-10-28 14:04:09 +00:00
BUILD.gn
Updates to AudioFrameView and VectorFloatFrame
2024-06-17 12:13:40 +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
Use backticks not vertical bars to denote variables in comments for /modules/audio_processing
2021-08-09 21:49:02 +00:00
compute_interpolated_gain_curve.h
Use backticks not vertical bars to denote variables in comments for /modules/audio_processing
2021-08-09 21:49:02 +00:00
cpu_features.cc
RNN VAD: FC layer simplified
2020-12-09 00:04:50 +00:00
cpu_features.h
RNN VAD: FC layer simplified
2020-12-09 00:04:50 +00:00
fixed_digital_level_estimator_unittest.cc
Update FrameCombiner et al to use DeinterleavedView
2024-07-02 15:58:20 +00:00
fixed_digital_level_estimator.cc
Update FrameCombiner et al to use DeinterleavedView
2024-07-02 15:58:20 +00:00
fixed_digital_level_estimator.h
Update FrameCombiner et al to use DeinterleavedView
2024-07-02 15:58:20 +00:00
gain_applier_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
gain_applier.cc
AGC2: GainController::ApplyConfig removed
2021-10-14 12:58:25 +00:00
gain_applier.h
AudioFrameView: size_t -> int
2021-10-07 14:41:03 +00:00
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
AGC2: retuning and large refactoring
2021-04-14 19:01:01 +00:00
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
AGC2 Limiter class renamed.
2018-10-23 15:20:52 +00:00
limiter_db_gain_curve.cc
Use backticks not vertical bars to denote variables in comments for /modules/audio_processing
2021-08-09 21:49:02 +00:00
limiter_db_gain_curve.h
AGC2 Limiter class renamed.
2018-10-23 15:20:52 +00:00
limiter_unittest.cc
Update FrameCombiner et al to use DeinterleavedView
2024-07-02 15:58:20 +00:00
limiter.cc
Update FrameCombiner et al to use DeinterleavedView
2024-07-02 15:58:20 +00:00
limiter.h
Update FrameCombiner et al to use DeinterleavedView
2024-07-02 15:58:20 +00:00
noise_level_estimator_unittest.cc
AGC2: removed unused noise estimator implementation
2021-09-01 12:45:20 +00:00
noise_level_estimator.cc
AGC2 NoiseLevelEstimator: DCHECK pointer
2022-12-06 15:14:04 +00:00
noise_level_estimator.h
AGC2: removed unused noise estimator implementation
2021-09-01 12:45:20 +00:00
saturation_protector_buffer_unittest.cc
AGC2: retuning and large refactoring
2021-04-14 19:01:01 +00:00
saturation_protector_buffer.cc
AGC2: retuning and large refactoring
2021-04-14 19:01:01 +00:00
saturation_protector_buffer.h
AGC2: retuning and large refactoring
2021-04-14 19:01:01 +00:00
saturation_protector_unittest.cc
AGC2: use only one headroom parameter
2021-09-28 16:52:16 +00:00
saturation_protector.cc
AGC2: use only one headroom parameter
2021-09-28 16:52:16 +00:00
saturation_protector.h
AGC2: use only one headroom parameter
2021-09-28 16:52:16 +00:00
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
Replace "(const override)" with "(const, override)" in GMOCKs
2022-01-26 10:59:40 +00:00
vad_wrapper.cc
Use MonoView for deinterleaved channels in AudioFrameView
2024-06-05 12:39:27 +00:00
vad_wrapper.h
AGC2 adaptive digital controller config clean-up
2022-12-09 13:07:34 +00:00
vector_float_frame.cc
Updates to AudioFrameView and VectorFloatFrame
2024-06-17 12:13:40 +00:00
vector_float_frame.h
Updates to AudioFrameView and VectorFloatFrame
2024-06-17 12:13:40 +00:00