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hexefx_audiolib_F32/src/basic_lfo.h

93 lines
2.1 KiB

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