//------------------------------------------------------------------------------
void designFirstOrderLowpass(double* dcoefs, double cutoff, double withFs)
// design parametric filter based on input center frequency, gain, Q and sampling rate
{
double b0, b1, b2, a0, a1, a2; //storage for continuous-time filter coefs
double acoefs[6];
//Design lowpass filter here. Filter should be of the form
//
// 2
// b2s + b1s + b0
// ---------------
// 2
// a2s + a1s + a0
//
// where b2 and a2 are zero (1st order, just keeM_PIng them so the same BLT function can be used)
//
// Parameters are cutoff frequency in Hz
b0 = 1; // design analog filter based on cutoff frequency
b1 = 0;
b2 = 0;
a0 = 1;
a1 = 1 / cutoff;
a2 = 0;
acoefs[0] = b0; acoefs[1] = b1; acoefs[2] = b2; // pack the analog coeffs into an array
acoefs[3] = a0; acoefs[4] = a1; acoefs[5] = a2;
// and apply the bilinear tranform
bilinearTransform(acoefs, dcoefs, withFs); // inputs the 6 analog coeffs, output the 5 digital coeffs
}
//------------------------------------------------------------------------------
void Reverb::designParametric(double* dcoefs, double center, double gain, double qval)
// design parametric filter based on input center frequency, gain, Q and sampling rate
{
double b0, b1, b2, a0, a1, a2; //storage for continuous-time filter coefs
double acoefs[6];
//Design parametric filter here. Filter should be of the form
//
// 2
// b2s + b1s + b0
// ---------------
// 2
// a2s + a1s + a0
//
// Parameters are center frequency in Hz, gain in dB, and Q.
// TODO: design analog filter based on input gain, center frequency and Q
// Remeber to handle the two cases: boost and cut!
///////////////START//////////////////
float pi = 3.14159;
float w = 2*pi*center;
//boost
if (gain > 0.0)
{
b2 = 1/(w*w);
b1 = gain/(qval*w);
b0 = 1;
a2 = 1/(w*w);
a1 = 1/(qval*w);
a0 = 1;
}
//cut
else
{
b2 = 1/(w*w);
b1 = 1/(qval*w);
b0 = 1;
a2 = 1/(w*w);
a1 = 1/(gain*qval*w);
a0 = 1;
}
////////////////END/////////////////////
// pack the analog coeffs into an array and apply the bilinear tranform
acoefs[0] = b0; acoefs[1] = b1; acoefs[2] = b2;
acoefs[3] = a0; acoefs[4] = a1; acoefs[5] = a2;
// inputs the 6 analog coeffs, output the 5 digital coeffs
bilinearTransform(acoefs, dcoefs);
}
//------------------------------------------------------------------------------
void designParametric(double* dcoefs, double center, double gain, double qval, double withFs)
// design parametric filter based on input center frequency, gain, Q and sampling rate
{
double b0, b1, b2, a0, a1, a2; //storage for continuous-time filter coefs
double acoefs[6];
//Design parametric filter here. Filter should be of the form
//
// 2
// b2s + b1s + b0
// ---------------
// 2
// a2s + a1s + a0
//
// Parameters are center frequency in Hz, gain in dB, and Q.
// design analog filter based on input gain, center frequency and Q
if (gain > 0.0) { // boost case
b0 = 1;
b1 = gain / (qval * center * 2 * M_PI);
b2 = 1 / (center * center * 4 * M_PI * M_PI);
a0 = 1;
a1 = 1 / (qval * center * 2 * M_PI);
a2 = 1 / (center * center * 4 * M_PI * M_PI);
} else { // cut case
b0 = 1;
b1 = 1 / (qval * center * 2 * M_PI);
b2 = 1 / (center * center * 4 * M_PI * M_PI);
a0 = 1;
a1 = 1 / (qval * gain * center * 2 * M_PI);
a2 = 1 / (center * center * 4 * M_PI * M_PI);
}
acoefs[0] = b0; acoefs[1] = b1; acoefs[2] = b2; // pack the analog coeffs into an array
acoefs[3] = a0; acoefs[4] = a1; acoefs[5] = a2;
// and apply the bilinear tranform
bilinearTransform(acoefs, dcoefs, withFs); // inputs the 6 analog coeffs, output the 5 digital coeffs
}