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/*
* 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_