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#ifndef _BASIC_LFO_H_
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#define _BASIC_LFO_H_
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#include <Arduino.h>
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#include "AudioStream_F32.h"
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#define BASIC_LFO_PHASE_0 (0)
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#define BASIC_LFO_PHASE_45 (32)
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#define BASIC_LFO_PHASE_60 (43)
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#define BASIC_LFO_PHASE_90 (64)
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#define BASIC_LFO_PHASE_120 (85)
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#define BASIC_LFO_PHASE_180 (128)
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extern "C" {
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extern const int16_t AudioWaveformSine[257];
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}
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/*
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* @brief Basic sin LFO with float output
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*
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*/
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class AudioBasicLfo
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{
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public:
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AudioBasicLfo(float rateHz, uint32_t ampl)
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{
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acc = 0;
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if (!ampl)
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{
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state = false;
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divider = 0x8000;
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}
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else divider = (0x7FFF + (ampl>>1)) / ampl;
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state = true;
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adder = (uint32_t)(rateHz * rate_mult);
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}
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inline void update()
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{
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acc += adder; // update the phase acc
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}
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/**
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* @brief LFO output split in two parts
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*
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* @param phase8bit 0-360deg scaled to 8bit value - phase shift (ie 64=90deg)
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* @param intOffset pointer top integer value used as address offset
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* @param fractOffset pointer to fractional part used for interpolation
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*/
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inline void get(uint8_t phase8bit, uint32_t *intOffset, float *fractOffset)
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{
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if (!state) // lfo off
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{
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*intOffset = 0;
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*fractOffset = 0.0f;
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return;
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}
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uint32_t idx;
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uint32_t y0, y1;
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uint64_t y;
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float intOff;
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idx = ((acc >> 24) + phase8bit) & 0xFF;
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y0 = AudioWaveformSine[idx] + 32767;
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y1 = AudioWaveformSine[idx+1] + 32767;
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idx = acc & 0x00FFFFFF; // lower 24 bit = fractional part
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y = (int64_t)y0 * (0x00FFFFFF - idx);
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y += (int64_t)y1 * idx;
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y0 = (int32_t) (y >> 24); // 16bit output
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*fractOffset = modff((float)y0 / (float)divider, &intOff);
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*intOffset = (uint32_t)intOff;
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}
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inline void setRate(float rateHz)
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{
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adder = (uint32_t)(rateHz * rate_mult);
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}
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inline void setDepth(uint32_t ampl)
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{
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if (!ampl)
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{
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state = false;
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return; // do not update the divider
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}
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state = true;
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divider = (0x7FFF + (ampl>>1)) / ampl;
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}
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private:
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bool state = true;
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uint32_t acc;
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uint32_t adder;
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int32_t divider = 1;
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const uint32_t rate_mult = 4294967295.0f / AUDIO_SAMPLE_RATE_EXACT;
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};
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#endif // _BASIC_LFO_H_
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