| /* |
| * 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/downsampled_render_buffer.h" |
| |
| namespace webrtc { |
| |
| DownsampledRenderBuffer::DownsampledRenderBuffer(size_t downsampled_buffer_size) |
| : size(downsampled_buffer_size), buffer(downsampled_buffer_size, 0.f) { |
| std::fill(buffer.begin(), buffer.end(), 0.f); |
| } |
| |
| DownsampledRenderBuffer::~DownsampledRenderBuffer() = default; |
| |
| } // namespace webrtc |
