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| 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 { |
| 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_ |
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