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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/fft_buffer.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
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