blob: d9ad1d935a5b3f42eb44ded7f47283bb4e09629b [file] [log] [blame]
/*
* Copyright (c) 2017 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 WEBRTC_MODULES_AUDIO_PROCESSING_AGC2_DIGITAL_GAIN_APPLIER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC2_DIGITAL_GAIN_APPLIER_H_
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/rtc_base/array_view.h"
namespace webrtc {
class DigitalGainApplier {
public:
DigitalGainApplier();
// Applies the specified gain to an array of samples.
void Process(float gain, rtc::ArrayView<float> samples);
private:
void LimitToAllowedRange(rtc::ArrayView<float> x);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC2_DIGITAL_GAIN_APPLIER_H_