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