void next_k(int inNumSamples)
{
float newFreq = in0(1);
float newQ = in0(2);
float newHPCutoff = in0(3);
if (q != newQ)
set_q(newQ);
if (newHPCutoff != hpCutoff)
setFeedbackHPF(newHPCutoff * sampleDur());
double a, a_inv, a2, b, b2, c, g, g0;
calcFilterCoefficients(newFreq, a, a2, a_inv, b, b2, c, g, g0);
double z0 = z[0];
double z1 = z[1];
double z2 = z[2];
double z3 = z[3];
double z4 = z[4];
const float * inSig = zin(0);
float * outSig = zout(0);
for (int i = 0; i != inNumSamples; ++i) {
double x = ZXP(inSig);
ZXP(outSig) = tick(x, a, a2, a_inv, b, b2, c, g, g0, z0, z1, z2, z3, z4);
}
z[0] = z0;
z[1] = z1;
z[2] = z2;
z[3] = z3;
z[4] = z4;
}
void calcFilterCoefficients(const double newFreq, double & a, double & a2, double & a_inv,
double & b, double & b2, double & c, double & g, double & g0)
{
double fc = sc_max(10, newFreq) * sampleDur();
fc = sc_min(fc, 0.25);
a = M_PI * fc; // PI is Nyquist frequency
// a = 2 * tan(0.5*a); // dewarping, not required with 2x oversampling
a_inv = 1/a;
a2 = a*a;
b = 2*a + 1;
b2 = b*b;
c = 1 / (2*a2*a2 - 4*a2*b2 + b2*b2);
g0 = 2*a2*a2*c;
g = g0 * bh;
}
MySaw2() {
// 1. set the calculation function.
if (isAudioRateIn(0)) {
// if the frequency argument is audio rate
set_calc_function<MySaw2,&MySaw2::next_a>();
} else {
// if thene frequency argument is control rate (or a scalar).
set_calc_function<MySaw2,&MySaw2::next_k>();
}
// 2. initialize the unit generator state variables.
// initialize a constant for multiplying the frequency
mFreqMul = 2.0 * sampleDur();
// get initial phase of oscillator
mPhase = in0(1);
// 3. calculate one sample of output.
if (isAudioRateIn(0)) {
next_a(1);
} else {
next_k(1);
}
}
