webrtc/modules/audio_processing/agc2/adaptive_agc.h
Alessio Bazzica 1e2542f593 AGC2: adding level estimation option (RMS or peak-based).
This CL makes possible to choose the level estimation for the adaptive
digital GC of AGC2. The options are RMS (default and currently used
estimator) and peak-based (already computed, but not used).

Besides adding the new AGC2 config param for the level estimator, this CL
also refactors the config class by making it more structured.

Bug: webrtc:7494
Change-Id: I20eb558ca50f13536aa7bdea08d21de3b630f8bc
Reviewed-on: https://webrtc-review.googlesource.com/c/110144
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25620}
2018-11-13 14:32:13 +00:00

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1.5 KiB
C++

/*
* Copyright (c) 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
#include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h"
#include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
#include "modules/audio_processing/agc2/noise_level_estimator.h"
#include "modules/audio_processing/agc2/vad_with_level.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "modules/audio_processing/include/audio_processing.h"
namespace webrtc {
class ApmDataDumper;
class AdaptiveAgc {
public:
explicit AdaptiveAgc(ApmDataDumper* apm_data_dumper);
AdaptiveAgc(ApmDataDumper* apm_data_dumper,
const AudioProcessing::Config::GainController2& config);
~AdaptiveAgc();
void Process(AudioFrameView<float> float_frame, float last_audio_level);
void Reset();
private:
AdaptiveModeLevelEstimator speech_level_estimator_;
VadWithLevel vad_;
AdaptiveDigitalGainApplier gain_applier_;
ApmDataDumper* const apm_data_dumper_;
NoiseLevelEstimator noise_level_estimator_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_