OLD | NEW |
---|---|
(Empty) | |
1 /* | |
2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. | |
3 * | |
4 * Use of this source code is governed by a BSD-style license | |
5 * that can be found in the LICENSE file in the root of the source | |
6 * tree. An additional intellectual property rights grant can be found | |
7 * in the file PATENTS. All contributing project authors may | |
8 * be found in the AUTHORS file in the root of the source tree. | |
9 */ | |
10 | |
11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_ | |
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_ | |
13 | |
14 #include <array> | |
15 | |
16 #include "webrtc/modules/audio_processing/aec3/aec3_common.h" | |
17 | |
18 namespace webrtc { | |
19 | |
20 // Holds the circular buffer of the downsampled render data. | |
21 struct DownsampledRenderBuffer { | |
peah-webrtc
2017/03/30 05:36:55
This is a bit too simple to keep in a separate fil
| |
22 DownsampledRenderBuffer(); | |
23 ~DownsampledRenderBuffer(); | |
24 std::array<float, kDownsampledRenderBufferSize> buffer; | |
25 int position = 0; | |
26 }; | |
27 | |
28 } // namespace webrtc | |
29 | |
30 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_ | |
OLD | NEW |