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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
60 lines
1.8 KiB
C++
60 lines
1.8 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_AEC3_FFT_BUFFER_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_FFT_BUFFER_H_
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#include <stddef.h>
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#include <vector>
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#include "modules/audio_processing/aec3/fft_data.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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// Struct for bundling a circular buffer of FftData objects together with the
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// read and write indices.
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struct FftBuffer {
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explicit FftBuffer(size_t size);
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~FftBuffer();
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int IncIndex(int index) const {
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RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
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return index < size - 1 ? index + 1 : 0;
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}
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int DecIndex(int index) const {
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RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
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return index > 0 ? index - 1 : size - 1;
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}
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int OffsetIndex(int index, int offset) const {
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RTC_DCHECK_GE(buffer.size(), offset);
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RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
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return (size + index + offset) % size;
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}
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void UpdateWriteIndex(int offset) { write = OffsetIndex(write, offset); }
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void IncWriteIndex() { write = IncIndex(write); }
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void DecWriteIndex() { write = DecIndex(write); }
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void UpdateReadIndex(int offset) { read = OffsetIndex(read, offset); }
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void IncReadIndex() { read = IncIndex(read); }
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void DecReadIndex() { read = DecIndex(read); }
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const int size;
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std::vector<FftData> buffer;
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int write = 0;
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int read = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_FFT_BUFFER_H_
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