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/*
* Copyright (c) 2021 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_CAPTURE_LEVELS_ADJUSTER_AUDIO_SAMPLES_SCALER_H_
#define MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_AUDIO_SAMPLES_SCALER_H_
#include <stddef.h>
#include "modules/audio_processing/audio_buffer.h"
namespace webrtc {
// Handles and applies a gain to the samples in an audio buffer.
// The gain is applied for each sample and any changes in the gain take effect
// gradually (in a linear manner) over one frame.
class AudioSamplesScaler {
public:
// C-tor. The supplied `initial_gain` is used immediately at the first call to
// Process(), i.e., in contrast to the gain supplied by SetGain(...) there is
// no gradual change to the `initial_gain`.
explicit AudioSamplesScaler(float initial_gain);
AudioSamplesScaler(const AudioSamplesScaler&) = delete;
AudioSamplesScaler& operator=(const AudioSamplesScaler&) = delete;
// Applies the specified gain to the audio in `audio_buffer`.
void Process(AudioBuffer& audio_buffer);
// Sets the gain to apply to each sample.
void SetGain(float gain) { target_gain_ = gain; }
private:
float previous_gain_ = 1.f;
float target_gain_ = 1.f;
int samples_per_channel_ = -1;
float one_by_samples_per_channel_ = -1.f;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_AUDIO_SAMPLES_SCALER_H_
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