| /* |
| * 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 |
