| /* |
| * 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. |
| */ |
| |
| #ifndef MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_ |
| #define MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_ |
| |
| #include <array> |
| |
| #include "modules/audio_processing/aec3/aec3_common.h" |
| |
| namespace webrtc { |
| |
| // Holds the circular buffer of the downsampled render data. |
| struct DownsampledRenderBuffer { |
| DownsampledRenderBuffer(); |
| ~DownsampledRenderBuffer(); |
| std::array<float, kDownsampledRenderBufferSize> buffer = {}; |
| int position = 0; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_ |