void AnalogFilter::setfreq (REALTYPE frequency)
{
if (frequency < 0.1) frequency = 0.1;
REALTYPE rap = freq / frequency;
if (rap < 1.0) rap = 1.0 / rap;
oldabovenq = abovenq;
abovenq = frequency > (SAMPLE_RATE / 2 - 500.0);
int nyquistthresh = (abovenq ^ oldabovenq);
if ((rap > 3.0) || (nyquistthresh != 0))
{
// if the frequency is changed fast, it needs interpolation
// (now, filter and coeficients backup)
for (int i = 0; i < 3; i++)
{
oldc[i] = c[i];
oldd[i] = d[i];
}
for (int i = 0; i < MAX_FILTER_STAGES + 1; i++)
{
oldx[i] = x[i];
oldy[i] = y[i];
}
if (firsttime == 0)
needsinterpolation = 1;
}
freq = frequency;
computefiltercoefs();
firsttime = 0;
}
/*
* Initialise the filters
*/
void SUBnote::initfilter(bpfilter &filter,
float freq,
float bw,
float amp,
float mag)
{
filter.xn1 = 0.0f;
filter.xn2 = 0.0f;
if(start == 0) {
filter.yn1 = 0.0f;
filter.yn2 = 0.0f;
}
else {
float a = 0.1f * mag; //empirically
float p = RND * 2.0f * PI;
if(start == 1)
a *= RND;
filter.yn1 = a * cosf(p);
filter.yn2 = a * cosf(p + freq * 2.0f * PI / synth.samplerate_f);
//correct the error of computation the start amplitude
//at very high frequencies
if(freq > synth.samplerate_f * 0.96f) {
filter.yn1 = 0.0f;
filter.yn2 = 0.0f;
}
}
filter.amp = amp;
filter.freq = freq;
filter.bw = bw;
computefiltercoefs(filter, freq, bw, 1.0f);
}
/*
* Initialise the filters
*/
void SUBnote::initfilter(bpfilter &filter,
REALTYPE freq,
REALTYPE bw,
REALTYPE amp,
REALTYPE mag)
{
filter.xn1=0.0;
filter.xn2=0.0;
if (start==0) {
filter.yn1=0.0;
filter.yn2=0.0;
} else {
REALTYPE a=0.1*mag;//empirically
REALTYPE p=RND*2.0*PI;
if (start==1) a*=RND;
filter.yn1=a*cos(p);
filter.yn2=a*cos(p+freq*2.0*PI/SAMPLE_RATE);
//correct the error of computation the start amplitude
//at very high frequencies
if (freq>SAMPLE_RATE*0.96) {
filter.yn1=0.0;
filter.yn2=0.0;
};
};
filter.amp=amp;
filter.freq=freq;
filter.bw=bw;
computefiltercoefs(filter,freq,bw,1.0);
};
void
RBFilter::settype (int type_)
{
type = type_;
if(!qmode) computefiltercoefs ();
else computefiltercoefs_hiQ ();
};
void
RBFilter::setgain (float dBgain)
{
gain = dB2rap (dBgain);
if(!qmode) computefiltercoefs ();
else computefiltercoefs_hiQ ();
};
void
RBFilter::setq (float q_)
{
q = q_;
if(!qmode) computefiltercoefs ();
else computefiltercoefs_hiQ ();
};
void
RBFilter::setfreq (float frequency)
{
if (frequency > (fSAMPLE_RATE / 2.0f - 500.0f)) frequency = fSAMPLE_RATE / 2.0f - 500.0f;
if (frequency < 0.1)
frequency = 0.1f;
float rap = freq / frequency;
if (rap < 1.0)
rap = 1.0f / rap;
oldabovenq = abovenq;
abovenq = frequency > (fSAMPLE_RATE / 2.0f - 500.0f);
int nyquistthresh = (abovenq ^ oldabovenq);
if ((rap > 3.0) || (nyquistthresh != 0)) {
//if the frequency is changed fast, it needs interpolation (now, filter and coeficients backup)
if (firsttime == 0)
needsinterpolation = 1;
ipar = par;
};
freq = frequency;
if(!qmode) computefiltercoefs ();
else computefiltercoefs_hiQ ();
firsttime = 0;
};
void AnalogFilter::setfreq(float frequency)
{
if(frequency < 0.1f)
frequency = 0.1f;
float rap = freq / frequency;
if(rap < 1.0f)
rap = 1.0f / rap;
oldabovenq = abovenq;
abovenq = frequency > (halfsamplerate_f - 500.0f);
bool nyquistthresh = (abovenq ^ oldabovenq);
//if the frequency is changed fast, it needs interpolation
if((rap > 3.0f) || nyquistthresh) { //(now, filter and coeficients backup)
oldCoeff = coeff;
for(int i = 0; i < MAX_FILTER_STAGES + 1; ++i)
oldHistory[i] = history[i];
if(!firsttime)
needsinterpolation = true;
}
freq = frequency;
computefiltercoefs();
firsttime = false;
}
void SVFilter::setstages(int stages_)
{
if(stages_ >= MAX_FILTER_STAGES)
stages_ = MAX_FILTER_STAGES - 1;
stages = stages_;
cleanup();
computefiltercoefs();
}
void SUBnote::initfilter_rl(bpfilter &filter,REALTYPE freq,REALTYPE bw,REALTYPE amp,REALTYPE mag)
{
filter.amp=amp;
filter.freq=freq;
filter.bw=bw;
computefiltercoefs(filter,freq,bw,1.0);
};
void
AnalogFilter::setSR(unsigned int value)
{
iSAMPLE_RATE = value;
ifSAMPLE_RATE=(float) iSAMPLE_RATE;
computefiltercoefs ();
}
void
setstages (ZEq10ban_t * s, AnalogFilter *filter, int stages_)
{
if (stages_ >= MAX_FILTER_STAGES)
stages_ = MAX_FILTER_STAGES - 1;
filter->stages = stages_;
AnalogFilter_Cleanup (s,filter);
computefiltercoefs (s,filter);
};
void
RBFilter::setstages (int stages_)
{
if (stages_ >= MAX_FILTER_STAGES)
stages_ = MAX_FILTER_STAGES - 1;
stages = stages_;
cleanup ();
if(!qmode) computefiltercoefs ();
else computefiltercoefs_hiQ ();
};
void AnalogFilter::setstages(int stages_)
{
if(stages_ >= MAX_FILTER_STAGES)
stages_ = MAX_FILTER_STAGES - 1;
if(stages_ != stages) {
stages = stages_;
cleanup();
computefiltercoefs();
}
}
void SVFilter::setfreq(REALTYPE frequency) {
if (frequency<0.1) frequency=0.1;
REALTYPE rap=freq/frequency;
if (rap<1.0) rap=1.0/rap;
oldabovenq=abovenq;
abovenq=frequency>(SAMPLE_RATE/2-500.0);
int nyquistthresh=(abovenq^oldabovenq);
if ((rap>3.0)||(nyquistthresh!=0)) { //if the frequency is changed fast, it needs interpolation (now, filter and coeficients backup)
if (firsttime==0) needsinterpolation=1;
ipar=par;
};
freq=frequency;
computefiltercoefs();
firsttime=0;
};
void SVFilter::setfreq(REALTYPE frequency)
{
REALTYPE rap;
int nyquistthresh;
if (frequency < 0.1)
{
frequency = 0.1;
}
rap = freq / frequency;
if (rap < 1.0)
{
rap = 1.0 / rap;
}
oldabovenq = abovenq;
abovenq = frequency > (m_sample_rate / 2 - 500.0);
nyquistthresh = (abovenq ^ oldabovenq);
if (rap > 3.0 || nyquistthresh != 0)
{
// if the frequency is changed fast, it needs interpolation (now, filter and coeficients backup)
if (firsttime == 0)
{
needsinterpolation = 1;
}
ipar = par;
}
freq = frequency;
computefiltercoefs();
firsttime = 0;
}
void SVFilter::setfreq(float frequency)
{
if(frequency < 0.1f)
frequency = 0.1f;
float rap = freq / frequency;
if(rap < 1.0f)
rap = 1.0f / rap;
oldabovenq = abovenq;
abovenq = frequency > (synth->samplerate_f / 2 - 500.0f);
bool nyquistthresh = (abovenq ^ oldabovenq);
//if the frequency is changed fast, it needs interpolation
if((rap > 3.0f) || nyquistthresh) { //(now, filter and coeficients backup)
if(!firsttime)
needsinterpolation = true;
ipar = par;
}
freq = frequency;
computefiltercoefs();
firsttime = false;
}
void
setfreq (ZEq10ban_t * s, AnalogFilter *filter, float frequency)
{
int i;
if (frequency < 0.1)
frequency = 0.1;
float rap = filter->freq / frequency;
if (rap < 1.0)
rap = 1.0 / rap;
filter->oldabovenq = filter->abovenq;
filter->abovenq = frequency > (s->SAMPLE_RATE / 2 - 500.0);
int nyquistthresh = (filter->abovenq ^ filter->oldabovenq);
if ((rap > 3.0) || (nyquistthresh != 0))
{ //if the frequency is changed fast, it needs interpolation (now, filter and coeficients backup)
for (i = 0; i < 3; i++)
{
filter->oldc[i] = filter->c[i];
filter->oldd[i] = filter->d[i];
};
for (i = 0; i < MAX_FILTER_STAGES + 1; i++)
{
filter->oldx[i] = filter->x[i];
filter->oldy[i] = filter->y[i];
};
if (filter->firsttime == 0)
filter->needsinterpolation = 1;
};
filter->freq = frequency;
computefiltercoefs (s,filter);
filter->firsttime = 0;
};
void
setq (ZEq10ban_t * s, AnalogFilter *filter, float q_)
{
filter->q = q_;
computefiltercoefs (s,filter);
};
void SVFilter::setq(float q_)
{
q = q_;
computefiltercoefs();
}
void SVFilter::setgain(float dBgain)
{
gain = dB2rap(dBgain);
computefiltercoefs();
}
void SVFilter::settype(int type_)
{
type = type_;
computefiltercoefs();
}
void
setgain (goomf_synth_t * s, AnalogFilter * filter, float dBgain)
{
filter->gain = dB2rap (dBgain);
computefiltercoefs (s, filter);
};
void
settype (ZEq10ban_t * s, AnalogFilter *filter, int type_)
{
filter->type = type_;
computefiltercoefs (s,filter);
};
void
setgain (ZEq10ban_t * s, AnalogFilter *filter, float dBgain)
{
filter->gain = dB2rap (dBgain);
computefiltercoefs (s,filter);
};
void AnalogFilter::computefiltercoefs(void)
{
float tmp;
bool zerocoefs = false; //this is used if the freq is too high
//do not allow frequencies bigger than samplerate/2
float freq = this->freq;
if(freq > (halfsamplerate_f - 500.0f)) {
freq = halfsamplerate_f - 500.0f;
zerocoefs = true;
}
if(freq < 0.1f)
freq = 0.1f;
//do not allow bogus Q
if(q < 0.0f)
q = 0.0f;
float tmpq, tmpgain;
if(stages == 0) {
tmpq = q;
tmpgain = gain;
}
else {
tmpq = (q > 1.0f) ? powf(q, 1.0f / (stages + 1)) : q;
tmpgain = powf(gain, 1.0f / (stages + 1));
}
//Alias Terms
float *c = coeff.c;
float *d = coeff.d;
//General Constants
const float omega = 2 * PI * freq / samplerate_f;
const float sn = sinf(omega), cs = cosf(omega);
float alpha, beta;
//most of theese are implementations of
//the "Cookbook formulae for audio EQ" by Robert Bristow-Johnson
//The original location of the Cookbook is:
//http://www.harmony-central.com/Computer/Programming/Audio-EQ-Cookbook.txt
switch(type) {
case 0: //LPF 1 pole
if(!zerocoefs)
tmp = expf(-2.0f * PI * freq / samplerate_f);
else
tmp = 0.0f;
c[0] = 1.0f - tmp;
c[1] = 0.0f;
c[2] = 0.0f;
d[1] = tmp;
d[2] = 0.0f;
order = 1;
break;
case 1: //HPF 1 pole
if(!zerocoefs)
tmp = expf(-2.0f * PI * freq / samplerate_f);
else
tmp = 0.0f;
c[0] = (1.0f + tmp) / 2.0f;
c[1] = -(1.0f + tmp) / 2.0f;
c[2] = 0.0f;
d[1] = tmp;
d[2] = 0.0f;
order = 1;
break;
case 2: //LPF 2 poles
if(!zerocoefs) {
alpha = sn / (2.0f * tmpq);
tmp = 1 + alpha;
c[1] = (1.0f - cs) / tmp;
c[0] = c[2] = c[1] / 2.0f;
d[1] = -2.0f * cs / tmp * -1.0f;
d[2] = (1.0f - alpha) / tmp * -1.0f;
}
else {
c[0] = 1.0f;
c[1] = c[2] = d[1] = d[2] = 0.0f;
}
order = 2;
break;
case 3: //HPF 2 poles
if(!zerocoefs) {
alpha = sn / (2.0f * tmpq);
tmp = 1 + alpha;
c[0] = (1.0f + cs) / 2.0f / tmp;
c[1] = -(1.0f + cs) / tmp;
c[2] = (1.0f + cs) / 2.0f / tmp;
d[1] = -2.0f * cs / tmp * -1.0f;
d[2] = (1.0f - alpha) / tmp * -1.0f;
}
else
c[0] = c[1] = c[2] = d[1] = d[2] = 0.0f;
order = 2;
break;
case 4: //BPF 2 poles
if(!zerocoefs) {
alpha = sn / (2.0f * tmpq);
tmp = 1.0f + alpha;
c[0] = alpha / tmp *sqrtf(tmpq + 1.0f);
c[1] = 0.0f;
c[2] = -alpha / tmp *sqrtf(tmpq + 1.0f);
d[1] = -2.0f * cs / tmp * -1.0f;
d[2] = (1.0f - alpha) / tmp * -1.0f;
}
else
c[0] = c[1] = c[2] = d[1] = d[2] = 0.0f;
order = 2;
break;
case 5: //NOTCH 2 poles
if(!zerocoefs) {
alpha = sn / (2.0f * sqrtf(tmpq));
tmp = 1.0f + alpha;
c[0] = 1.0f / tmp;
c[1] = -2.0f * cs / tmp;
c[2] = 1.0f / tmp;
d[1] = -2.0f * cs / tmp * -1.0f;
d[2] = (1.0f - alpha) / tmp * -1.0f;
}
else {
c[0] = 1.0f;
c[1] = c[2] = d[1] = d[2] = 0.0f;
}
order = 2;
break;
case 6: //PEAK (2 poles)
if(!zerocoefs) {
tmpq *= 3.0f;
alpha = sn / (2.0f * tmpq);
tmp = 1.0f + alpha / tmpgain;
c[0] = (1.0f + alpha * tmpgain) / tmp;
c[1] = (-2.0f * cs) / tmp;
c[2] = (1.0f - alpha * tmpgain) / tmp;
d[1] = -2.0f * cs / tmp * -1.0f;
d[2] = (1.0f - alpha / tmpgain) / tmp * -1.0f;
}
else {
c[0] = 1.0f;
c[1] = c[2] = d[1] = d[2] = 0.0f;
}
order = 2;
break;
case 7: //Low Shelf - 2 poles
if(!zerocoefs) {
tmpq = sqrtf(tmpq);
alpha = sn / (2.0f * tmpq);
beta = sqrtf(tmpgain) / tmpq;
tmp = (tmpgain + 1.0f) + (tmpgain - 1.0f) * cs + beta * sn;
c[0] = tmpgain
* ((tmpgain
+ 1.0f) - (tmpgain - 1.0f) * cs + beta * sn) / tmp;
c[1] = 2.0f * tmpgain
* ((tmpgain - 1.0f) - (tmpgain + 1.0f) * cs) / tmp;
c[2] = tmpgain
* ((tmpgain
+ 1.0f) - (tmpgain - 1.0f) * cs - beta * sn) / tmp;
d[1] = -2.0f * ((tmpgain - 1.0f) + (tmpgain + 1.0f) * cs)
/ tmp * -1.0f;
d[2] = ((tmpgain + 1.0f) + (tmpgain - 1.0f) * cs - beta * sn)
/ tmp * -1.0f;
}
else {
c[0] = tmpgain;
c[1] = c[2] = d[1] = d[2] = 0.0f;
}
order = 2;
break;
case 8: //High Shelf - 2 poles
if(!zerocoefs) {
tmpq = sqrtf(tmpq);
alpha = sn / (2.0f * tmpq);
beta = sqrtf(tmpgain) / tmpq;
tmp = (tmpgain + 1.0f) - (tmpgain - 1.0f) * cs + beta * sn;
c[0] = tmpgain
* ((tmpgain
+ 1.0f) + (tmpgain - 1.0f) * cs + beta * sn) / tmp;
c[1] = -2.0f * tmpgain
* ((tmpgain - 1.0f) + (tmpgain + 1.0f) * cs) / tmp;
c[2] = tmpgain
* ((tmpgain
+ 1.0f) + (tmpgain - 1.0f) * cs - beta * sn) / tmp;
d[1] = 2.0f * ((tmpgain - 1.0f) - (tmpgain + 1.0f) * cs)
/ tmp * -1.0f;
d[2] = ((tmpgain + 1.0f) - (tmpgain - 1.0f) * cs - beta * sn)
/ tmp * -1.0f;
}
else {
c[0] = 1.0f;
c[1] = c[2] = d[1] = d[2] = 0.0f;
}
order = 2;
break;
default: //wrong type
type = 0;
computefiltercoefs();
break;
}
}
void
computefiltercoefs (ZEq10ban_t * s, AnalogFilter *filter)
{
float tmp;
float omega, sn, cs, alpha, beta;
int zerocoefs = 0; //this is used if the freq is too high
//do not allow frequencies bigger than samplerate/2
float freq = filter->freq;
if (freq > (s->SAMPLE_RATE / 2 - 500.0))
{
freq = s->SAMPLE_RATE / 2 - 500.0;
zerocoefs = 1;
};
if (freq < 0.1)
freq = 0.1;
//do not allow bogus Q
if (filter->q < 0.0)
filter->q = 0.0;
float tmpq, tmpgain;
if (filter->stages == 0)
{
tmpq = filter->q;
tmpgain = filter->gain;
}
else
{
tmpq = (filter->q > 1.0 ? powf (filter->q, 1.0 / (filter->stages + 1)) : filter->q);
tmpgain = powf (filter->gain, 1.0 / (filter->stages + 1));
};
//most of theese are implementations of
//the "Cookbook formulae for audio EQ" by Robert Bristow-Johnson
//The original location of the Cookbook is:
//http://www.harmony-central.com/Computer/Programming/Audio-EQ-Cookbook.txt
switch (filter->type)
{
case 0: //LPF 1 pole
if (zerocoefs == 0)
tmp = expf (-2.0 * M_PI * freq / s->SAMPLE_RATE);
else
tmp = 0.0;
filter->c[0] = 1.0 - tmp;
filter->c[1] = 0.0;
filter->c[2] = 0.0;
filter->d[1] = tmp;
filter->d[2] = 0.0;
filter->order = 1;
break;
case 1: //HPF 1 pole
if (zerocoefs == 0)
tmp = expf (-2.0 * M_PI * freq / s->SAMPLE_RATE);
else
tmp = 0.0;
filter->c[0] = (1.0 + tmp) / 2.0;
filter->c[1] = -(1.0 + tmp) / 2.0;
filter->c[2] = 0.0;
filter->d[1] = tmp;
filter->d[2] = 0.0;
filter->order = 1;
break;
case 2: //LPF 2 poles
if (zerocoefs == 0)
{
omega = 2 * M_PI * freq / s->SAMPLE_RATE;
sn = sinf (omega);
cs = cosf (omega);
alpha = sn / (2 * tmpq);
tmp = 1 + alpha;
filter->c[0] = (1.0 - cs) / 2.0 / tmp;
filter->c[1] = (1.0 - cs) / tmp;
filter->c[2] = (1.0 - cs) / 2.0 / tmp;
filter->d[1] = -2 * cs / tmp * (-1);
filter->d[2] = (1 - alpha) / tmp * (-1);
}
else
{
filter->c[0] = 1.0;
filter->c[1] = 0.0;
filter->c[2] = 0.0;
filter->d[1] = 0.0;
filter->d[2] = 0.0;
};
filter->order = 2;
break;
case 3: //HPF 2 poles
if (zerocoefs == 0)
{
omega = 2 * M_PI * freq / s->SAMPLE_RATE;
sn = sinf (omega);
cs = cosf (omega);
alpha = sn / (2 * tmpq);
tmp = 1 + alpha;
filter->c[0] = (1.0 + cs) / 2.0 / tmp;
filter->c[1] = -(1.0 + cs) / tmp;
filter->c[2] = (1.0 + cs) / 2.0 / tmp;
filter->d[1] = -2 * cs / tmp * (-1);
filter->d[2] = (1 - alpha) / tmp * (-1);
}
else
{
filter->c[0] = 0.0;
filter->c[1] = 0.0;
filter->c[2] = 0.0;
filter->d[1] = 0.0;
filter->d[2] = 0.0;
};
filter->order = 2;
break;
case 4: //BPF 2 poles
if (zerocoefs == 0)
{
omega = 2 * M_PI * freq / s->SAMPLE_RATE;
sn = sinf (omega);
cs = cosf (omega);
alpha = sn / (2 * tmpq);
tmp = 1 + alpha;
filter->c[0] = alpha / tmp * sqrtf (tmpq + 1);
filter->c[1] = 0;
filter->c[2] = -alpha / tmp * sqrtf (tmpq + 1);
filter->d[1] = -2 * cs / tmp * (-1);
filter->d[2] = (1 - alpha) / tmp * (-1);
}
else
{
filter->c[0] = 0.0;
filter->c[1] = 0.0;
filter->c[2] = 0.0;
filter->d[1] = 0.0;
filter->d[2] = 0.0;
};
filter->order = 2;
break;
case 5: //NOTCH 2 poles
if (zerocoefs == 0)
{
omega = 2 * M_PI * freq / s->SAMPLE_RATE;
sn = sinf (omega);
cs = cosf (omega);
alpha = sn / (2 * sqrtf (tmpq));
tmp = 1 + alpha;
filter->c[0] = 1 / tmp;
filter->c[1] = -2 * cs / tmp;
filter->c[2] = 1 / tmp;
filter->d[1] = -2 * cs / tmp * (-1);
filter->d[2] = (1 - alpha) / tmp * (-1);
}
else
{
filter->c[0] = 1.0;
filter->c[1] = 0.0;
filter->c[2] = 0.0;
filter->d[1] = 0.0;
filter->d[2] = 0.0;
};
filter->order = 2;
break;
case 6: //PEAK (2 poles)
if (zerocoefs == 0)
{
omega = 2 * M_PI * freq / s->SAMPLE_RATE;
sn = sinf (omega);
cs = cosf (omega);
tmpq *= 3.0;
alpha = sn / (2 * tmpq);
tmp = 1 + alpha / tmpgain;
filter->c[0] = (1.0 + alpha * tmpgain) / tmp;
filter->c[1] = (-2.0 * cs) / tmp;
filter->c[2] = (1.0 - alpha * tmpgain) / tmp;
filter->d[1] = -2 * cs / tmp * (-1);
filter->d[2] = (1 - alpha / tmpgain) / tmp * (-1);
}
else
{
filter->c[0] = 1.0;
filter->c[1] = 0.0;
filter->c[2] = 0.0;
filter->d[1] = 0.0;
filter->d[2] = 0.0;
};
filter->order = 2;
break;
case 7: //Low Shelf - 2 poles
if (zerocoefs == 0)
{
omega = 2 * M_PI * freq / s->SAMPLE_RATE;
sn = sinf (omega);
cs = cosf (omega);
tmpq = sqrtf (tmpq);
alpha = sn / (2 * tmpq);
beta = sqrtf (tmpgain) / tmpq;
tmp = (tmpgain + 1.0) + (tmpgain - 1.0) * cs + beta * sn;
filter->c[0] =
tmpgain * ((tmpgain + 1.0) - (tmpgain - 1.0) * cs +
beta * sn) / tmp;
filter->c[1] =
2.0 * tmpgain * ((tmpgain - 1.0) - (tmpgain + 1.0) * cs) / tmp;
filter->c[2] =
tmpgain * ((tmpgain + 1.0) - (tmpgain - 1.0) * cs -
beta * sn) / tmp;
filter->d[1] = -2.0 * ((tmpgain - 1.0) + (tmpgain + 1.0) * cs) / tmp * (-1);
filter->d[2] =
((tmpgain + 1.0) + (tmpgain - 1.0) * cs - beta * sn) / tmp * (-1);
}
else
{
filter->c[0] = tmpgain;
filter->c[1] = 0.0;
filter->c[2] = 0.0;
filter->d[1] = 0.0;
filter->d[2] = 0.0;
};
filter->order = 2;
break;
case 8: //High Shelf - 2 poles
if (zerocoefs == 0)
{
omega = 2 * M_PI * freq / s->SAMPLE_RATE;
sn = sinf (omega);
cs = cosf (omega);
tmpq = sqrtf (tmpq);
alpha = sn / (2 * tmpq);
beta = sqrtf (tmpgain) / tmpq;
tmp = (tmpgain + 1.0) - (tmpgain - 1.0) * cs + beta * sn;
filter->c[0] =
tmpgain * ((tmpgain + 1.0) + (tmpgain - 1.0) * cs +
beta * sn) / tmp;
filter->c[1] =
-2.0 * tmpgain * ((tmpgain - 1.0) + (tmpgain + 1.0) * cs) / tmp;
filter->c[2] =
tmpgain * ((tmpgain + 1.0) + (tmpgain - 1.0) * cs -
beta * sn) / tmp;
filter->d[1] = 2.0 * ((tmpgain - 1.0) - (tmpgain + 1.0) * cs) / tmp * (-1);
filter->d[2] =
((tmpgain + 1.0) - (tmpgain - 1.0) * cs - beta * sn) / tmp * (-1);
}
else
{
filter->c[0] = 1.0;
filter->c[1] = 0.0;
filter->c[2] = 0.0;
filter->d[1] = 0.0;
filter->d[2] = 0.0;
};
filter->order = 2;
break;
default: //wrong type
filter->type = 0;
computefiltercoefs (s,filter);
break;
};
};
void
setq (goomf_synth_t * s, AnalogFilter * filter, float q_)
{
filter->q = q_;
computefiltercoefs (s, filter);
};
/*
* Compute Parameters of SUBnote for each tick
*/
void SUBnote::computecurrentparameters()
{
if(FreqEnvelope || BandWidthEnvelope
|| (oldpitchwheel != ctl.pitchwheel.data)
|| (oldbandwidth != ctl.bandwidth.data)
|| portamento) {
float envfreq = 1.0f;
float envbw = 1.0f;
float gain = 1.0f;
if(FreqEnvelope) {
envfreq = FreqEnvelope->envout() / 1200;
envfreq = powf(2.0f, envfreq);
}
envfreq *=
powf(ctl.pitchwheel.relfreq, BendAdjust); //pitch wheel
if(portamento) { //portamento is used
envfreq *= ctl.portamento.freqrap;
if(!ctl.portamento.used) //the portamento has finished
portamento = false; //this note is no longer "portamented"
}
if(BandWidthEnvelope) {
envbw = BandWidthEnvelope->envout();
envbw = powf(2, envbw);
}
envbw *= ctl.bandwidth.relbw; //bandwidth controller
float tmpgain = 1.0f / sqrt(envbw * envfreq);
for(int n = 0; n < numharmonics; ++n) {
overtone_rolloff[n] = computerolloff(overtone_freq[n] * envfreq);
}
for(int n = 0; n < numharmonics; ++n)
for(int nph = 0; nph < numstages; ++nph) {
if(nph == 0)
gain = tmpgain;
else
gain = 1.0f;
computefiltercoefs(lfilter[nph + n * numstages],
lfilter[nph + n * numstages].freq * envfreq,
lfilter[nph + n * numstages].bw * envbw,
gain);
}
if(stereo)
for(int n = 0; n < numharmonics; ++n)
for(int nph = 0; nph < numstages; ++nph) {
if(nph == 0)
gain = tmpgain;
else
gain = 1.0f;
computefiltercoefs(
rfilter[nph + n * numstages],
rfilter[nph + n * numstages].freq * envfreq,
rfilter[nph + n * numstages].bw * envbw,
gain);
}
oldbandwidth = ctl.bandwidth.data;
oldpitchwheel = ctl.pitchwheel.data;
}
newamplitude = volume * AmpEnvelope->envout_dB() * 2.0f;
//Filter
if(GlobalFilterL != NULL) {
float globalfilterpitch = GlobalFilterCenterPitch
+ GlobalFilterEnvelope->envout();
float filterfreq = globalfilterpitch + ctl.filtercutoff.relfreq
+ GlobalFilterFreqTracking;
filterfreq = Filter::getrealfreq(filterfreq);
GlobalFilterL->setfreq_and_q(filterfreq,
globalfiltercenterq * ctl.filterq.relq);
if(GlobalFilterR != NULL)
GlobalFilterR->setfreq_and_q(
filterfreq,
globalfiltercenterq
* ctl.filterq.relq);
}
}
void
settype (goomf_synth_t * s, AnalogFilter * filter, int type_)
{
filter->type = type_;
computefiltercoefs (s, filter);
};
