sample rate stuff for chorus
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@ -21,24 +21,21 @@
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#include <math.h>
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#include <string.h>
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#include <cstdio>
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Chorus::Chorus() {
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lpfOut1 = new float[bufferSize];
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lpfOut2 = new float[bufferSize];
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ram = new float[DELAYSIZE]; // probably needs to be calculated based on sample rate
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ram = new float[DELAYSIZE];
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fastPhase = 0;
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lfoPhase = 1;
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lfoSpeed = 6.283 * 10.7 / sampleRate; // plainly silly value to show if it hasn't been set
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// not quite Butterworth but you'd never hear the difference
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// these are calculated from the real-world component values
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postFilter1l = new SVF(9688, .549);
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postFilter2l = new SVF(10377, 1.291);
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postFilter1r = new SVF(9688, .549);
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postFilter2r = new SVF(10377, 1.291);
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// lfo values taken from a rough simulation
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fastOmega = 6.283 * 0.7 / sampleRate; // approximate, can be adjusted
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// zero out the delay buffer
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memset(ram, 0, sizeof(float) * DELAYSIZE);
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memset(lpfOut1, 0, sizeof(float) * bufferSize);
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@ -56,32 +53,33 @@ Chorus::~Chorus() {
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}
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void Chorus::run(float* input, float** outputs, uint32_t frames) {
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// actual effects here
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// run highpass / bass boost and stereo chorus effect for one full block
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// FIXME add highpass filter
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// now run the DSP
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float s0 = 0, s1 = 0;
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float lfoMod, dly1, frac, flt;
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uint16_t tap, delay;
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for (uint32_t i = 0; i < frames; i++) {
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// run a step of LFO
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fastPhase += fastOmega;
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if (fastPhase > 6.283) fastPhase -= 6.283;
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lfoPhase += (lfoState & 0x01) ? lfoSpeed : -lfoSpeed;
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flt = ((hpdelay - input[i]) * hpcut) + input[i];
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hpdelay = flt;
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input[i] += (flt * hpgain);
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if (abs(lfoPhase) > 1) {
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// lfoPhase -= 2;
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lfoState++;
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}
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// highpass/bass boost
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flt = ((input[i] - hpDelay) * hpCut) + hpDelay;
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hpDelay = flt;
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input[i] += (flt * hpGain);
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ram[delayptr] = input[i];
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// delays in milliseconds
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#define BASE 0.005
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#define AMT 0.00175
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// FIXME make this triangle not sine, and move arithmetic out of mainloop
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lfoMod = sin(fastPhase);
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dly1 = (BASE + (AMT * lfoMod)) * sampleRate;
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dly1 = (BASE + (AMT * lfoPhase)) * sampleRate;
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delay = (int)dly1;
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frac = dly1 - delay;
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@ -90,7 +88,7 @@ void Chorus::run(float* input, float** outputs, uint32_t frames) {
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s0 = ram[tap & 0x3ff];
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lpfOut1[i] = ((s1 - s0) * frac) + s0;
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dly1 = (BASE - (AMT * lfoMod)) * sampleRate;
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dly1 = (BASE - (AMT * lfoPhase)) * sampleRate;
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delay = (int)dly1;
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frac = dly1 - delay;
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@ -99,9 +97,6 @@ void Chorus::run(float* input, float** outputs, uint32_t frames) {
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s0 = ram[tap & 0x3ff];
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lpfOut2[i] = ((s1 - s0) * frac) + s0;
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// lpfOut1[i] = input[i] + 1.2 * out0; //(out0 + (out120 * 0.66) + (out240 * 0.33));
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// lpfOut2[i] = input[i] + 1.2 * out120; //(out0 + (out120 * 0.33) + (out240 * 0.66));
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delayptr++;
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delayptr &= 0x3ff;
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}
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@ -119,22 +114,27 @@ void Chorus::run(float* input, float** outputs, uint32_t frames) {
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}
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void Chorus::setHpf(uint8_t mode) {
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// the simple 1-pole lowpass has its output
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// subtracted from input for the highpass
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// or added for bass boost
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// cutoff is calculated as
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// k = 1-exp(-2pi * Fc * sampleRate)
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switch (mode) {
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case 0x00:
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hpcut = 0.91245;
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hpgain = -1;
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hpCut = 1 - exp(-6.283 * 720 / sampleRate);
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hpGain = -1;
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break;
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case 0x08:
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hpcut = 0.97097;
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hpgain = -1;
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hpCut = 1 - exp(-6.283 * 225 / sampleRate);
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hpGain = -1;
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break;
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case 0x10:
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hpcut = 1;
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hpgain = 0;
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hpCut = 1;
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hpGain = 0;
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break;
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case 0x18:
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hpcut = 0.99088;
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hpgain = 1;
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hpCut = 1 - exp(-6.283 * 85 / sampleRate);
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hpGain = 1.707;
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break;
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}
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}
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@ -147,11 +147,11 @@ void Chorus::setChorus(uint8_t mode) {
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break;
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case 0x40:
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gain = 1.2;
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fastOmega = 6.283 * 0.5 / sampleRate;
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break; // approximate, can be adjusted
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lfoSpeed = 6.283 * 0.5 / sampleRate;
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break;
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case 0x00:
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gain = 1.2;
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fastOmega = 6.283 * 0.9 / sampleRate;
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break; // approximate, can be adjusted
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lfoSpeed = 6.283 * 0.9 / sampleRate;
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break;
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}
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}
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@ -38,16 +38,18 @@ class Chorus {
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void setChorus(uint8_t mode);
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private:
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double fastPhase = 0, fastOmega = 0;
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double lfoPhase = 0, lfoSpeed = 0;
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uint8_t lfoState=0;
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float gain = 1.2;
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uint16_t delayptr = 0;
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float* ram;
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float* lpfIn;
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float *lpfOut1, *lpfOut2;
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float hpdelay=0, hpcut=0, hpgain=1;
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float hpDelay=0, hpCut=0, hpGain=0;
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SVF *postFilter1l, *postFilter2l, *postFilter1r, *postFilter2r;
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};
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