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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.
*/
#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
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