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/*
* 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 <memory>
#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/saturation_protector.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;
// Adaptive digital gain controller.
// TODO(crbug.com/webrtc/7494): Unify with `AdaptiveDigitalGainApplier`.
class AdaptiveAgc {
public:
explicit AdaptiveAgc(ApmDataDumper* apm_data_dumper);
// TODO(crbug.com/webrtc/7494): Remove ctor above.
AdaptiveAgc(
ApmDataDumper* apm_data_dumper,
const AudioProcessing::Config::GainController2::AdaptiveDigital& config);
~AdaptiveAgc();
// Analyzes `frame` and applies a digital adaptive gain to it. Takes into
// account the envelope measured by the limiter.
// TODO(crbug.com/webrtc/7494): Make the class depend on the limiter.
void Process(AudioFrameView<float> frame, float limiter_envelope);
// Handles a gain change applied to the input signal (e.g., analog gain).
void HandleInputGainChange();
private:
AdaptiveModeLevelEstimator speech_level_estimator_;
VadLevelAnalyzer vad_;
AdaptiveDigitalGainApplier gain_controller_;
ApmDataDumper* const apm_data_dumper_;
std::unique_ptr<NoiseLevelEstimator> noise_level_estimator_;
std::unique_ptr<SaturationProtector> saturation_protector_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
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