/*
* Envelope Output (dB)
*/
float Envelope::envout_dB()
{
float out;
if(linearenvelope != 0)
return envout();
if((currentpoint == 1) && (!keyreleased || (forcedrelase == 0))) { //first point is always lineary interpolated
float v1 = dB2rap(envval[0]);
float v2 = dB2rap(envval[1]);
out = v1 + (v2 - v1) * t;
t += inct;
if(t >= 1.0f) {
t = 0.0f;
inct = envdt[2];
currentpoint++;
out = v2;
}
if(out > 0.001f)
envoutval = rap2dB(out);
else
envoutval = MIN_ENVELOPE_DB;
}
else
out = dB2rap(envout());
return out;
}
float EQ::getfreqresponse(float freq)
{
float resp = 1.0f;
for(int i = 0; i < MAX_EQ_BANDS; ++i) {
if(filter[i].Ptype == 0)
continue;
resp *= filter[i].l->H(freq);
}
return rap2dB(resp * outvolume);
}
void
out_smb (holharm_t * s, float *smpsl, float *smpsr,
unsigned long sample_count)
{
int i;
float elratiol, elratior;
float i_sum = 1e-12;
float temp_sum, val_i_sum;
float tmp;
for (i = 0; i < sample_count; i++)
{
s->outil[i] = smpsl[i];
s->outir[i] = smpsr[i];
tmp = fabsf (smpsr[i] + smpsl[i]);
if (tmp > i_sum)
i_sum = tmp;
}
temp_sum = CLAMP (rap2dB (i_sum), -48.0, 15.0);
val_i_sum = .6 * s->val_sum + .4 * temp_sum;
s->val_sum = val_i_sum;
if (*(s->PSELECT) == 1)
{
elratiol = s->ratiol;
elratior = s->ratior;
}
else
{
elratiol = powf (2, *(s->intervall) / 12.0);
elratior = powf (2, *(s->intervalr) / 12.0);
}
smbPitchShiftL (s, elratiol, sample_count, 2048, s->hq,
(float) s->SAMPLE_RATE, s->outil, s->outol);
smbPitchShiftR (s, elratior, sample_count, 2048, s->hq,
(float) s->SAMPLE_RATE, s->outir, s->outor);
}
void FormantFilterGraph::draw() {
int maxdB=30;
int ox=x(),oy=y(),lx=w(),ly=h(),i,oiy;
REALTYPE freqx;
fl_color(FL_BLACK);
fl_rectf(ox,oy,lx,ly);
//draw the lines
fl_color(FL_GRAY);
fl_line_style(FL_SOLID);
//fl_line(ox+2,oy+ly/2,ox+lx-2,oy+ly/2);
freqx=pars->getfreqpos(1000.0);
if ((freqx>0.0)&&(freqx<1.0))
fl_line(ox+(int) (freqx*lx),oy,
ox+(int) (freqx*lx),oy+ly);
for (i=1;i<10;i++){
if(i==1){
draw_freq_line(i*100.0,0);
draw_freq_line(i*1000.0,0);
}else
if (i==5){
draw_freq_line(i*100.0,2);
draw_freq_line(i*1000.0,2);
}else{
draw_freq_line(i*100.0,1);
draw_freq_line(i*1000.0,1);
};
};
draw_freq_line(10000.0,0);
draw_freq_line(20000.0,1);
fl_line_style(FL_DOT);
int GY=10;if (ly<GY*3) GY=-1;
for (i=1;i<GY;i++){
int tmp=(int)(ly/(REALTYPE)GY*i);
fl_line(ox+2,oy+tmp,ox+lx-2,oy+tmp);
};
fl_color(FL_YELLOW);
fl_font(FL_HELVETICA,10);
if (*nformant<pars->Pnumformants){
draw_freq_line(pars->getformantfreq(pars->Pvowels[*nvowel].formants[*nformant].freq),2);
//show some information (like current formant frequency,amplitude)
char tmpstr[20];
snprintf(tmpstr,20,"%.2f kHz",pars->getformantfreq(pars->Pvowels[*nvowel].formants[*nformant].freq)*0.001);
fl_draw(tmpstr,ox+1,oy+1,40,12,FL_ALIGN_LEFT,NULL,0);
snprintf(tmpstr,20,"%d dB",(int)( rap2dB(1e-9 + pars->getformantamp(pars->Pvowels[*nvowel].formants[*nformant].amp)) + pars->getgain() ));
fl_draw(tmpstr,ox+1,oy+15,40,12,FL_ALIGN_LEFT,NULL,0);
};
//draw the data
fl_color(FL_RED);
fl_line_style(FL_SOLID);
pars->formantfilterH(*nvowel,lx,graphpoints);
oiy=(int) ((graphpoints[0]/maxdB+1.0)*ly/2.0);
for (i=1;i<lx;i++){
int iy=(int) ((graphpoints[i]/maxdB+1.0)*ly/2.0);
if ((iy>=0)&&(oiy>=0)&&(iy<ly)&&(oiy<lx))
fl_line(ox+i-1,oy+ly-oiy,ox+i,oy+ly-iy);
oiy=iy;
};
}
/*
* Get the freq. response of the formant filter
*/
void FilterParams::formantfilterH(int nvowel, int nfreqs, REALTYPE *freqs)
{
REALTYPE c[3], d[3];
REALTYPE filter_freq, filter_q, filter_amp;
REALTYPE omega, sn, cs, alpha;
for(int i = 0; i < nfreqs; i++)
freqs[i] = 0.0;
//for each formant...
for(int nformant = 0; nformant < Pnumformants; nformant++) {
//compute formant parameters(frequency,amplitude,etc.)
filter_freq = getformantfreq(Pvowels[nvowel].formants[nformant].freq);
filter_q = getformantq(Pvowels[nvowel].formants[nformant].q) * getq();
if(Pstages > 0)
filter_q =
(filter_q > 1.0 ? POW(filter_q, 1.0 / (Pstages + 1)) : filter_q);
filter_amp = getformantamp(Pvowels[nvowel].formants[nformant].amp);
if(filter_freq <= (SAMPLE_RATE / 2 - 100.0)) {
omega = 2 * PI * filter_freq / SAMPLE_RATE;
sn = sin(omega);
cs = cos(omega);
alpha = sn / (2 * filter_q);
REALTYPE tmp = 1 + alpha;
c[0] = alpha / tmp *sqrt(filter_q + 1);
c[1] = 0;
c[2] = -alpha / tmp *sqrt(filter_q + 1);
d[1] = -2 * cs / tmp * (-1);
d[2] = (1 - alpha) / tmp * (-1);
}
else
continue;
for(int i = 0; i < nfreqs; i++) {
REALTYPE freq = getfreqx(i / (REALTYPE) nfreqs);
if(freq > SAMPLE_RATE / 2) {
for(int tmp = i; tmp < nfreqs; tmp++)
freqs[tmp] = 0.0;
break;
}
REALTYPE fr = freq / SAMPLE_RATE * PI * 2.0;
REALTYPE x = c[0], y = 0.0;
for(int n = 1; n < 3; n++) {
x += cos(n * fr) * c[n];
y -= sin(n * fr) * c[n];
}
REALTYPE h = x * x + y * y;
x = 1.0;
y = 0.0;
for(int n = 1; n < 3; n++) {
x -= cos(n * fr) * d[n];
y += sin(n * fr) * d[n];
}
h = h / (x * x + y * y);
freqs[i] += POW(h, (Pstages + 1.0) / 2.0) * filter_amp;
}
}
for(int i = 0; i < nfreqs; i++) {
if(freqs[i] > 0.000000001)
freqs[i] = rap2dB(freqs[i]) + getgain();
else
freqs[i] = -90.0;
}
}
void
Compressor::out (float *efxoutl, float *efxoutr)
{
int i;
for (i = 0; i < PERIOD; i++) {
float rdelta = 0.0f;
float ldelta = 0.0f;
//Right Channel
if(peak) {
if (rtimer > hold) {
rpeak *= 0.9998f; //The magic number corresponds to ~0.1s based on T/(RC + T),
rtimer--;
}
if (ltimer > hold) {
lpeak *= 0.9998f; //leaky peak detector.
ltimer --; //keeps the timer from eventually exceeding max int & rolling over
}
ltimer++;
rtimer++;
if(rpeak<fabs(efxoutr[i])) {
rpeak = fabs(efxoutr[i]);
rtimer = 0;
}
if(lpeak<fabs(efxoutl[i])) {
lpeak = fabs(efxoutl[i]);
ltimer = 0;
}
if(lpeak>20.0f) lpeak = 20.0f;
if(rpeak>20.0f) rpeak = 20.0f; //keeps limiter from getting locked up when signal levels go way out of bounds (like hundreds)
} else {
rpeak = efxoutr[i];
lpeak = efxoutl[i];
}
if(stereo) {
rdelta = fabsf (rpeak);
if(rvolume < 0.9f) {
attr = att;
relr = rel;
} else if (rvolume < 1.0f) {
attr = att + ((1.0f - att)*(rvolume - 0.9f)*10.0f); //dynamically change attack time for limiting mode
relr = rel/(1.0f + (rvolume - 0.9f)*9.0f); //release time gets longer when signal is above limiting
} else {
attr = 1.0f;
relr = rel*0.1f;
}
if (rdelta > rvolume)
rvolume = attr * rdelta + (1.0f - attr)*rvolume;
else
rvolume = relr * rdelta + (1.0f - relr)*rvolume;
rvolume_db = rap2dB (rvolume);
if (rvolume_db < thres_db) {
rgain = outlevel;
} else if (rvolume_db < thres_mx) {
//Dynamic ratio that depends on volume. As can be seen, ratio starts
//at something negligibly larger than 1 once volume exceeds thres, and increases toward selected
// ratio by the time it has reached thres_mx. --Transmogrifox
eratio = 1.0f + (kratio-1.0f)*(rvolume_db-thres_db)* coeff_knee;
rgain = outlevel*dB2rap(thres_db + (rvolume_db-thres_db)/eratio - rvolume_db);
} else {
rgain = outlevel*dB2rap(thres_db + coeff_kk + (rvolume_db-thres_mx)*coeff_ratio - rvolume_db);
limit = 1;
}
if ( rgain < MIN_GAIN) rgain = MIN_GAIN;
rgain_t = .4f * rgain + .6f * rgain_old;
};
//Left Channel
if(stereo) {
ldelta = fabsf (lpeak);
} else {
ldelta = 0.5f*(fabsf (lpeak) + fabsf (rpeak));
}; //It's not as efficient to check twice, but it's small expense worth code clarity
if(lvolume < 0.9f) {
attl = att;
rell = rel;
} else if (lvolume < 1.0f) {
attl = att + ((1.0f - att)*(lvolume - 0.9f)*10.0f); //dynamically change attack time for limiting mode
rell = rel/(1.0f + (lvolume - 0.9f)*9.0f); //release time gets longer when signal is above limiting
} else {
attl = 1.0f;
rell = rel*0.1f;
}
if (ldelta > lvolume)
lvolume = attl * ldelta + (1.0f - attl)*lvolume;
else
lvolume = rell*ldelta + (1.0f - rell)*lvolume;
lvolume_db = rap2dB (lvolume);
if (lvolume_db < thres_db) {
lgain = outlevel;
} else if (lvolume_db < thres_mx) { //knee region
eratio = 1.0f + (kratio-1.0f)*(lvolume_db-thres_db)* coeff_knee;
lgain = outlevel*dB2rap(thres_db + (lvolume_db-thres_db)/eratio - lvolume_db);
} else {
lgain = outlevel*dB2rap(thres_db + coeff_kk + (lvolume_db-thres_mx)*coeff_ratio - lvolume_db);
limit = 1;
}
if ( lgain < MIN_GAIN) lgain = MIN_GAIN;
lgain_t = .4f * lgain + .6f * lgain_old;
if (stereo) {
efxoutl[i] *= lgain_t;
efxoutr[i] *= rgain_t;
rgain_old = rgain;
lgain_old = lgain;
} else {
efxoutl[i] *= lgain_t;
efxoutr[i] *= lgain_t;
lgain_old = lgain;
}
if(peak) {
if(efxoutl[i]>0.999f) { //output hard limiting
efxoutl[i] = 0.999f;
clipping = 1;
}
if(efxoutl[i]<-0.999f) {
efxoutl[i] = -0.999f;
clipping = 1;
}
if(efxoutr[i]>0.999f) {
efxoutr[i] = 0.999f;
clipping = 1;
}
if(efxoutr[i]<-0.999f) {
efxoutr[i] = -0.999f;
clipping = 1;
}
//highly probably there is a more elegant way to do that, but what the hey...
}
}
}
/*
* Effect output
*/
void
Vocoder::out (float * smpsl, float * smpsr)
{
int i, j;
float tempgain;
float maxgain=0.0f;
float auxtemp, tmpgain;
if(DS_state != 0)
{
A_Resample->mono_out(auxresampled,tmpaux,PERIOD,u_up,nPERIOD);
}
else
memcpy(tmpaux,auxresampled,sizeof(float)*nPERIOD);
for (i = 0; i<nPERIOD; i++) //apply compression to auxresampled
{
auxtemp = input * tmpaux[i];
if(fabs(auxtemp > compeak)) compeak = fabs(auxtemp); //First do peak detection on the signal
compeak *= prls;
compenv = cbeta * oldcompenv + calpha * compeak; //Next average into envelope follower
oldcompenv = compenv;
if(compenv > cpthresh) //if envelope of signal exceeds thresh, then compress
{
compg = cpthresh + cpthresh*(compenv - cpthresh)/compenv;
cpthresh = cthresh + cratio*(compg - cpthresh); //cpthresh changes dynamically
tmpgain = compg/compenv;
}
else
{
tmpgain = 1.0f;
}
if(compenv < cpthresh) cpthresh = compenv;
if(cpthresh < cthresh) cpthresh = cthresh;
tmpaux[i] = auxtemp * tmpgain;
tmpaux[i]=vlp->filterout_s(tmpaux[i]);
tmpaux[i]=vhp->filterout_s(tmpaux[i]);
};
//End compression
auxtemp = 0.0f;
if(DS_state != 0)
{
U_Resample->out(smpsl,smpsr,tsmpsl,tsmpsr,PERIOD,u_up);
}
else
{
memcpy(tsmpsl,smpsl,sizeof(float)*nPERIOD);
memcpy(tsmpsr,smpsr,sizeof(float)*nPERIOD);
}
memset(tmpl,0,sizeof(float)*nPERIOD);
memset(tmpr,0,sizeof(float)*nPERIOD);
for (j = 0; j < VOC_BANDS; j++)
{
for (i = 0; i<nPERIOD; i++)
{
auxtemp = tmpaux[i];
if(filterbank[j].speak < gate) filterbank[j].speak = 0.0f; //gate
if(auxtemp>maxgain) maxgain = auxtemp; //vu meter level.
auxtemp = filterbank[j].aux->filterout_s(auxtemp);
if(fabs(auxtemp) > filterbank[j].speak) filterbank[j].speak = fabs(auxtemp); //Leaky Peak detector
filterbank[j].speak*=prls;
filterbank[j].gain = beta * filterbank[j].oldgain + alpha * filterbank[j].speak;
filterbank[j].oldgain = filterbank[j].gain;
tempgain = (1.0f-ringworm)*filterbank[j].oldgain+ringworm*auxtemp;
tmpl[i] +=filterbank[j].l->filterout_s(tsmpsl[i])*tempgain;
tmpr[i] +=filterbank[j].r->filterout_s(tsmpsr[i])*tempgain;
};
};
for (i = 0; i<nPERIOD; i++)
{
tmpl[i]*=lpanning*level;
tmpr[i]*=rpanning*level;
};
if(DS_state != 0)
{
D_Resample->out(tmpl,tmpr,efxoutl,efxoutr,nPERIOD,u_down);
}
else
{
memcpy(efxoutl,tmpl,sizeof(float)*nPERIOD);
memcpy(efxoutr,tmpr,sizeof(float)*nPERIOD);
}
vulevel = (float)CLAMP(rap2dB(maxgain), -48.0, 15.0);
};
