/*
 *  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/block_buffer.h"

#include <algorithm>

namespace webrtc {

BlockBuffer::BlockBuffer(size_t size,
                         size_t num_bands,
                         size_t num_channels,
                         size_t frame_length)
    : size(static_cast<int>(size)),
      buffer(size,
             std::vector<std::vector<std::vector<float>>>(
                 num_bands,
                 std::vector<std::vector<float>>(
                     num_channels,
                     std::vector<float>(frame_length, 0.f)))) {
  for (auto& block : buffer) {
    for (auto& band : block) {
      for (auto& channel : band) {
        std::fill(channel.begin(), channel.end(), 0.f);
      }
    }
  }
}

BlockBuffer::~BlockBuffer() = default;

}  // namespace webrtc