chromium/third_party/webrtc/modules/audio_processing/agc2/limiter.h

/*
 *  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_LIMITER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_

#include <vector>

#include "absl/strings/string_view.h"
#include "api/audio/audio_frame.h"
#include "modules/audio_processing/agc2/fixed_digital_level_estimator.h"
#include "modules/audio_processing/agc2/interpolated_gain_curve.h"
#include "modules/audio_processing/include/audio_frame_view.h"

namespace webrtc {
class ApmDataDumper;

class Limiter { … };

}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_AGC2_LIMITER_H_