FaceAccess/VocieProcess/modules/audio_processing/aec3/downsampled_render_buffer.h

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2024-09-05 09:59:28 +08:00
/*
* 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 <stddef.h>
#include <vector>
#include "rtc_base/checks.h"
namespace webrtc {
// Holds the circular buffer of the downsampled render data.
struct DownsampledRenderBuffer {
explicit DownsampledRenderBuffer(size_t downsampled_buffer_size);
~DownsampledRenderBuffer();
int IncIndex(int index) const {
RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
return index < size - 1 ? index + 1 : 0;
}
int DecIndex(int index) const {
RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
return index > 0 ? index - 1 : size - 1;
}
int OffsetIndex(int index, int offset) const {
RTC_DCHECK_GE(buffer.size(), offset);
RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
return (size + index + offset) % size;
}
void UpdateWriteIndex(int offset) { write = OffsetIndex(write, offset); }
void IncWriteIndex() { write = IncIndex(write); }
void DecWriteIndex() { write = DecIndex(write); }
void UpdateReadIndex(int offset) { read = OffsetIndex(read, offset); }
void IncReadIndex() { read = IncIndex(read); }
void DecReadIndex() { read = DecIndex(read); }
const int size;
std::vector<float> buffer;
int write = 0;
int read = 0;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_