void SVFilter::filterout(float *smp)
{
for(int i = 0; i < stages + 1; ++i)
singlefilterout(smp, st[i], par);
if(needsinterpolation) {
float *ismp = getTmpBuffer();
memcpy(ismp, smp, synth->bufferbytes);
for(int i = 0; i < stages + 1; ++i)
singlefilterout(ismp, st[i], ipar);
for(int i = 0; i < synth->buffersize; ++i) {
float x = i / synth->buffersize_f;
smp[i] = ismp[i] * (1.0f - x) + smp[i] * x;
}
returnTmpBuffer(ismp);
needsinterpolation = false;
}
for(int i = 0; i < synth->buffersize; ++i)
smp[i] *= outgain;
}
/*
* Note Output
*/
int SUBnote::noteout(float *outl, float *outr)
{
memcpy(outl, denormalkillbuf, synth->bufferbytes);
memcpy(outr, denormalkillbuf, synth->bufferbytes);
if(NoteEnabled == OFF)
return 0;
float *tmprnd = getTmpBuffer();
float *tmpsmp = getTmpBuffer();
//left channel
for(int i = 0; i < synth->buffersize; ++i)
tmprnd[i] = RND * 2.0f - 1.0f;
for(int n = 0; n < numharmonics; ++n) {
memcpy(tmpsmp, tmprnd, synth->bufferbytes);
for(int nph = 0; nph < numstages; ++nph)
filter(lfilter[nph + n * numstages], tmpsmp);
for(int i = 0; i < synth->buffersize; ++i)
outl[i] += tmpsmp[i];
}
if(GlobalFilterL != NULL)
GlobalFilterL->filterout(&outl[0]);
//right channel
if(stereo != 0) {
for(int i = 0; i < synth->buffersize; ++i)
tmprnd[i] = RND * 2.0f - 1.0f;
for(int n = 0; n < numharmonics; ++n) {
memcpy(tmpsmp, tmprnd, synth->bufferbytes);
for(int nph = 0; nph < numstages; ++nph)
filter(rfilter[nph + n * numstages], tmpsmp);
for(int i = 0; i < synth->buffersize; ++i)
outr[i] += tmpsmp[i];
}
if(GlobalFilterR != NULL)
GlobalFilterR->filterout(&outr[0]);
}
else
memcpy(outr, outl, synth->bufferbytes);
returnTmpBuffer(tmprnd);
returnTmpBuffer(tmpsmp);
if(firsttick != 0) {
int n = 10;
if(n > synth->buffersize)
n = synth->buffersize;
for(int i = 0; i < n; ++i) {
float ampfadein = 0.5f - 0.5f * cosf(
(float) i / (float) n * PI);
outl[i] *= ampfadein;
outr[i] *= ampfadein;
}
firsttick = 0;
}
if(ABOVE_AMPLITUDE_THRESHOLD(oldamplitude, newamplitude))
// Amplitude interpolation
for(int i = 0; i < synth->buffersize; ++i) {
float tmpvol = INTERPOLATE_AMPLITUDE(oldamplitude,
newamplitude,
i,
synth->buffersize);
outl[i] *= tmpvol * panning;
outr[i] *= tmpvol * (1.0f - panning);
}
else
for(int i = 0; i < synth->buffersize; ++i) {
outl[i] *= newamplitude * panning;
outr[i] *= newamplitude * (1.0f - panning);
}
oldamplitude = newamplitude;
computecurrentparameters();
// Apply legato-specific sound signal modifications
legato.apply(*this, outl, outr);
// Check if the note needs to be computed more
if(AmpEnvelope->finished() != 0) {
for(int i = 0; i < synth->buffersize; ++i) { //fade-out
float tmp = 1.0f - (float)i / synth->buffersize_f;
outl[i] *= tmp;
outr[i] *= tmp;
}
KillNote();
}
return 1;
}
