/*
 *  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.
 */

#include "modules/audio_processing/aec3/fft_buffer.h"

#include <cstddef>
#include <vector>

#include "modules/audio_processing/aec3/fft_data.h"

namespace webrtc {

FftBuffer::FftBuffer(size_t size, size_t num_channels)
    : size(static_cast<int>(size)),
      buffer(size, std::vector<FftData>(num_channels)) {
  for (auto& block : buffer) {
    for (auto& channel_fft_data : block) {
      channel_fft_data.Clear();
    }
  }
}

FftBuffer::~FftBuffer() = default;

}  // namespace webrtc