chromium/third_party/webrtc/modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.h

/*
 *  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 { … };
}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_CAPTURE_LEVELS_ADJUSTER_AUDIO_SAMPLES_SCALER_H_