blob: 499eac7f99512b58b81f0e2d985a8c3b1ccbef57 [file] [log] [blame]
peahcf02cf12017-04-05 14:18:07 -07001/*
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
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020011#ifndef MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
12#define MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
peahcf02cf12017-04-05 14:18:07 -070013
14#include <array>
15
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020016#include "modules/audio_processing/aec3/aec3_common.h"
peahcf02cf12017-04-05 14:18:07 -070017
18namespace webrtc {
19
20// Holds the circular buffer of the downsampled render data.
21struct DownsampledRenderBuffer {
22 DownsampledRenderBuffer();
23 ~DownsampledRenderBuffer();
24 std::array<float, kDownsampledRenderBufferSize> buffer = {};
25 int position = 0;
26};
27
28} // namespace webrtc
29
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020030#endif // MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_