Chromium Code Reviews| Index: webrtc/modules/audio_processing/aec3/downsampled_render_buffer.cc |
| diff --git a/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.cc b/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.cc |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..597b869e57dbde27af34a6b566a2d95cdd078af1 |
| --- /dev/null |
| +++ b/webrtc/modules/audio_processing/aec3/downsampled_render_buffer.cc |
| @@ -0,0 +1,21 @@ |
| +/* |
| + * 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 "webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h" |
| + |
| +namespace webrtc { |
| + |
| +DownsampledRenderBuffer::DownsampledRenderBuffer() { |
| + buffer.fill(0.f); |
| +} |
| + |
| +DownsampledRenderBuffer::~DownsampledRenderBuffer() {} |
|
ivoc
2017/03/31 13:58:31
= default?
peah-webrtc
2017/04/03 08:02:32
Done.
|
| + |
| +} // namespace webrtc |