| /* | |
| * 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 <cstddef> | |
| #include "modules/audio_processing/aec3/block.h" | |
| namespace webrtc { | |
| BlockBuffer::BlockBuffer(size_t size, size_t num_bands, size_t num_channels) | |
| : size(static_cast<int>(size)), | |
| buffer(size, Block(num_bands, num_channels)) {} | |
| BlockBuffer::~BlockBuffer() = default; | |
| } // namespace webrtc |