/*
 *  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_DECIMATOR_H_
#define MODULES_AUDIO_PROCESSING_AEC3_DECIMATOR_H_

#include <array>
#include <vector>

#include "api/array_view.h"
#include "modules/audio_processing/aec3/aec3_common.h"
#include "modules/audio_processing/utility/cascaded_biquad_filter.h"

namespace webrtc {

// Provides functionality for decimating a signal.
class Decimator {
 public:
  explicit Decimator(size_t down_sampling_factor);

  Decimator(const Decimator&) = delete;
  Decimator& operator=(const Decimator&) = delete;

  // Downsamples the signal.
  void Decimate(rtc::ArrayView<const float> in, rtc::ArrayView<float> out);

 private:
  const size_t down_sampling_factor_;
  CascadedBiQuadFilter anti_aliasing_filter_;
  CascadedBiQuadFilter noise_reduction_filter_;
};
}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_AEC3_DECIMATOR_H_
