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

/*
 *  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.
 */
#include "modules/audio_processing/capture_levels_adjuster/audio_samples_scaler.h"

#include <algorithm>

#include "api/array_view.h"
#include "modules/audio_processing/audio_buffer.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_minmax.h"

namespace webrtc {

AudioSamplesScaler::AudioSamplesScaler(float initial_gain)
    : … { … }

void AudioSamplesScaler::Process(AudioBuffer& audio_buffer) { … }

}  // namespace webrtc