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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_GAIN_APPLIER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_
#include <stddef.h>
#include "api/audio/audio_view.h"
#include "modules/audio_processing/include/audio_frame_view.h"
namespace webrtc {
class GainApplier {
public:
GainApplier(bool hard_clip_samples, float initial_gain_factor);
void ApplyGain(DeinterleavedView<float> signal);
void SetGainFactor(float gain_factor);
float GetGainFactor() const { return current_gain_factor_; }
[[deprecated("Use DeinterleavedView<> version")]] void ApplyGain(
AudioFrameView<float> signal) {
ApplyGain(signal.view());
}
private:
void Initialize(int samples_per_channel);
// Whether to clip samples after gain is applied. If 'true', result
// will fit in FloatS16 range.
const bool hard_clip_samples_;
float last_gain_factor_;
// If this value is not equal to 'last_gain_factor', gain will be
// ramped from 'last_gain_factor_' to this value during the next
// 'ApplyGain'.
float current_gain_factor_;
int samples_per_channel_ = -1;
float inverse_samples_per_channel_ = -1.f;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_