chromium/third_party/webrtc/modules/audio_processing/aec3/refined_filter_update_gain.h

/*
 *  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 MODULES_AUDIO_PROCESSING_AEC3_REFINED_FILTER_UPDATE_GAIN_H_
#define MODULES_AUDIO_PROCESSING_AEC3_REFINED_FILTER_UPDATE_GAIN_H_

#include <stddef.h>

#include <array>
#include <atomic>
#include <memory>

#include "api/array_view.h"
#include "api/audio/echo_canceller3_config.h"
#include "modules/audio_processing/aec3/aec3_common.h"

namespace webrtc {

class AdaptiveFirFilter;
class ApmDataDumper;
struct EchoPathVariability;
struct FftData;
class RenderSignalAnalyzer;
struct SubtractorOutput;

// Provides functionality for  computing the adaptive gain for the refined
// filter.
class RefinedFilterUpdateGain { … };

}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_AEC3_REFINED_FILTER_UPDATE_GAIN_H_