void FormantFilter::filterout(REALTYPE *smp)
{
int i, j;
for(i = 0; i < SOUND_BUFFER_SIZE; i++) {
inbuffer[i] = smp[i];
smp[i] = 0.0;
}
for(j = 0; j < numformants; j++) {
for(i = 0; i < SOUND_BUFFER_SIZE; i++)
tmpbuf[i] = inbuffer[i] * outgain;
formant[j]->filterout(tmpbuf);
if(ABOVE_AMPLITUDE_THRESHOLD(oldformantamp[j], currentformants[j].amp))
for(i = 0; i < SOUND_BUFFER_SIZE; i++)
smp[i] += tmpbuf[i]
* INTERPOLATE_AMPLITUDE(oldformantamp[j],
currentformants[j].amp,
i,
SOUND_BUFFER_SIZE);
else
for(i = 0; i < SOUND_BUFFER_SIZE; i++)
smp[i] += tmpbuf[i] * currentformants[j].amp;
oldformantamp[j] = currentformants[j].amp;
}
}
void
FormantFilter::filterout (float * smp)
{
int i, j;
for (i = 0; i < PERIOD; i++)
{
inbuffer[i] = smp[i];
smp[i] = 0.0;
};
for (j = 0; j < numformants; j++)
{
for (i = 0; i < PERIOD; i++)
tmpbuf[i] = inbuffer[i] * outgain;
formant[j]->filterout (tmpbuf);
if (ABOVE_AMPLITUDE_THRESHOLD
(oldformantamp[j], currentformants[j].amp))
for (i = 0; i < PERIOD; i++)
smp[i] +=
tmpbuf[i] * INTERPOLATE_AMPLITUDE (oldformantamp[j],
currentformants[j].amp, i,
PERIOD);
else
for (i = 0; i < PERIOD; i++)
smp[i] += tmpbuf[i] * currentformants[j].amp;
oldformantamp[j] = currentformants[j].amp;
};
};
void FormantFilter::filterout(float *smp)
{
float inbuffer[buffersize];
memcpy(inbuffer, smp, bufferbytes);
memset(smp, 0, bufferbytes);
for(int j = 0; j < numformants; ++j) {
float tmpbuf[buffersize];
for(int i = 0; i < buffersize; ++i)
tmpbuf[i] = inbuffer[i] * outgain;
formant[j]->filterout(tmpbuf);
if(ABOVE_AMPLITUDE_THRESHOLD(oldformantamp[j], currentformants[j].amp))
for(int i = 0; i < buffersize; ++i)
smp[i] += tmpbuf[i]
* INTERPOLATE_AMPLITUDE(oldformantamp[j],
currentformants[j].amp,
i,
buffersize);
else
for(int i = 0; i < buffersize; ++i)
smp[i] += tmpbuf[i] * currentformants[j].amp;
oldformantamp[j] = currentformants[j].amp;
}
}
void FormantFilter::filterout(float *smp)
{
int i;
int j;
for (i = 0 ; i < SOUND_BUFFER_SIZE ; i++)
{
m_inbuffer[i] = smp[i];
smp[i] = 0.0;
}
for (j = 0 ; j < m_numformants ; j++)
{
for (i = 0 ; i < SOUND_BUFFER_SIZE ; i++)
{
m_tmpbuf[i] = m_inbuffer[i] * m_outgain;
}
m_formants[j].filterout(m_tmpbuf);
if (ABOVE_AMPLITUDE_THRESHOLD(m_oldformantamp[j], m_currentformants[j].amplitude))
{
for (i = 0 ; i < SOUND_BUFFER_SIZE ; i++)
{
smp[i] += m_tmpbuf[i] * INTERPOLATE_AMPLITUDE(m_oldformantamp[j], m_currentformants[j].amplitude, i, SOUND_BUFFER_SIZE);
}
}
else
{
for (i = 0 ; i < SOUND_BUFFER_SIZE ; i++)
{
smp[i] += m_tmpbuf[i] * m_currentformants[j].amplitude;
}
}
m_oldformantamp[j] = m_currentformants[j].amplitude;
}
}
/*
* Note Output
*/
int SUBnote::noteout(float *outl, float *outr)
{
memcpy(outl, synth.denormalkillbuf, synth.bufferbytes);
memcpy(outr, synth.denormalkillbuf, synth.bufferbytes);
if(NoteEnabled == OFF)
return 0;
float tmprnd[synth.buffersize];
float tmpsmp[synth.buffersize];
//left channel
for(int i = 0; i < synth.buffersize; ++i)
tmprnd[i] = RND * 2.0f - 1.0f;
for(int n = 0; n < numharmonics; ++n) {
float rolloff = overtone_rolloff[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] * rolloff;
}
if(GlobalFilterL != NULL)
GlobalFilterL->filterout(&outl[0]);
//right channel
if(stereo) {
for(int i = 0; i < synth.buffersize; ++i)
tmprnd[i] = RND * 2.0f - 1.0f;
for(int n = 0; n < numharmonics; ++n) {
float rolloff = overtone_rolloff[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] * rolloff;
}
if(GlobalFilterR != NULL)
GlobalFilterR->filterout(&outr[0]);
}
else
memcpy(outr, outl, synth.bufferbytes);
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;
}
int PADnote::noteout(float *outl, float *outr)
{
computecurrentparameters();
float *smps = pars->sample[nsample].smp;
if(smps == NULL) {
for(int i = 0; i < synth->buffersize; ++i) {
outl[i] = 0.0f;
outr[i] = 0.0f;
}
return 1;
}
float smpfreq = pars->sample[nsample].basefreq;
float freqrap = realfreq / smpfreq;
int freqhi = (int) (floor(freqrap));
float freqlo = freqrap - floor(freqrap);
if(config.cfg.Interpolation)
Compute_Cubic(outl, outr, freqhi, freqlo);
else
Compute_Linear(outl, outr, freqhi, freqlo);
if(firsttime) {
fadein(outl);
fadein(outr);
firsttime = false;
}
NoteGlobalPar.GlobalFilterL->filterout(outl);
NoteGlobalPar.GlobalFilterR->filterout(outr);
//Apply the punch
if(NoteGlobalPar.Punch.Enabled != 0)
for(int i = 0; i < synth->buffersize; ++i) {
float punchamp = NoteGlobalPar.Punch.initialvalue
* NoteGlobalPar.Punch.t + 1.0f;
outl[i] *= punchamp;
outr[i] *= punchamp;
NoteGlobalPar.Punch.t -= NoteGlobalPar.Punch.dt;
if(NoteGlobalPar.Punch.t < 0.0f) {
NoteGlobalPar.Punch.Enabled = 0;
break;
}
}
if(ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude, globalnewamplitude))
// Amplitude Interpolation
for(int i = 0; i < synth->buffersize; ++i) {
float tmpvol = INTERPOLATE_AMPLITUDE(globaloldamplitude,
globalnewamplitude,
i,
synth->buffersize);
outl[i] *= tmpvol * NoteGlobalPar.Panning;
outr[i] *= tmpvol * (1.0f - NoteGlobalPar.Panning);
}
else
for(int i = 0; i < synth->buffersize; ++i) {
outl[i] *= globalnewamplitude * NoteGlobalPar.Panning;
outr[i] *= globalnewamplitude * (1.0f - NoteGlobalPar.Panning);
}
// Apply legato-specific sound signal modifications
legato.apply(*this, outl, outr);
// Check if the global amplitude is finished.
// If it does, disable the note
if(NoteGlobalPar.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;
}
finished_ = 1;
}
return 1;
}
/*
* Note Output
*/
int SUBnote::noteout(REALTYPE *outl,REALTYPE *outr)
{
int i;
memcpy(outl, denormalkillbuf, SOUND_BUFFER_SIZE * sizeof(REALTYPE));
memcpy(outr, denormalkillbuf, SOUND_BUFFER_SIZE * sizeof(REALTYPE));
if (NoteEnabled==OFF) return(0);
//left channel
for (i=0;i<SOUND_BUFFER_SIZE;i++) tmprnd[i]=RND*2.0-1.0;
for (int n=0;n<numharmonics;n++) {
memcpy(tmpsmp, tmprnd, SOUND_BUFFER_SIZE * sizeof(REALTYPE));
for (int nph=0;nph<numstages;nph++)
filter(lfilter[nph+n*numstages],tmpsmp);
for (i=0;i<SOUND_BUFFER_SIZE;i++) outl[i]+=tmpsmp[i];
};
if (GlobalFilterL!=NULL) GlobalFilterL->filterout(&outl[0]);
//right channel
if (stereo!=0) {
for (i=0;i<SOUND_BUFFER_SIZE;i++) tmprnd[i]=RND*2.0-1.0;
for (int n=0;n<numharmonics;n++) {
memcpy(tmpsmp, tmprnd, SOUND_BUFFER_SIZE * sizeof(REALTYPE));
for (int nph=0;nph<numstages;nph++)
filter(rfilter[nph+n*numstages],tmpsmp);
for (i=0;i<SOUND_BUFFER_SIZE;i++) outr[i]+=tmpsmp[i];
};
if(GlobalFilterR != NULL)
GlobalFilterR->filterout(&outr[0]);
}
else
memcpy(outr, outl, SOUND_BUFFER_SIZE * sizeof(REALTYPE));
if (firsttick!=0) {
int n=10;
if (n>SOUND_BUFFER_SIZE) n=SOUND_BUFFER_SIZE;
for (i=0;i<n;i++) {
REALTYPE ampfadein=0.5-0.5*cos((REALTYPE) i/(REALTYPE) n*PI);
outl[i]*=ampfadein;
outr[i]*=ampfadein;
};
firsttick=0;
};
if (ABOVE_AMPLITUDE_THRESHOLD(oldamplitude,newamplitude)) {
// Amplitude interpolation
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
REALTYPE tmpvol=INTERPOLATE_AMPLITUDE(oldamplitude
,newamplitude,i,SOUND_BUFFER_SIZE);
outl[i]*=tmpvol*panning;
outr[i]*=tmpvol*(1.0-panning);
};
} else {
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
outl[i]*=newamplitude*panning;
outr[i]*=newamplitude*(1.0-panning);
};
};
oldamplitude=newamplitude;
computecurrentparameters();
// Apply legato-specific sound signal modifications
if (Legato.silent) { // Silencer
if (Legato.msg!=LM_FadeIn) {
memset(outl, 0, SOUND_BUFFER_SIZE * sizeof(REALTYPE));
memset(outr, 0, SOUND_BUFFER_SIZE * sizeof(REALTYPE));
};
};
switch (Legato.msg) {
case LM_CatchUp : // Continue the catch-up...
if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
for (i=0;i<SOUND_BUFFER_SIZE;i++) {//Yea, could be done without the loop...
Legato.decounter--;
if (Legato.decounter<1) {
// Catching-up done, we can finally set
// the note to the actual parameters.
Legato.decounter=-10;
Legato.msg=LM_ToNorm;
SUBlegatonote(Legato.param.freq, Legato.param.vel, Legato.param.portamento, Legato.param.midinote, false);
break;
}
}
break;
case LM_FadeIn : // Fade-in
if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
Legato.silent=false;
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
Legato.decounter--;
if (Legato.decounter<1) {
Legato.decounter=-10;
Legato.msg=LM_Norm;
break;
}
Legato.fade.m+=Legato.fade.step;
outl[i]*=Legato.fade.m;
outr[i]*=Legato.fade.m;
}
break;
case LM_FadeOut : // Fade-out, then set the catch-up
if (Legato.decounter==-10) Legato.decounter=Legato.fade.length;
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
Legato.decounter--;
if (Legato.decounter<1) {
for (int j=i;j<SOUND_BUFFER_SIZE;j++) {
outl[j]=0.0;
outr[j]=0.0;
}
Legato.decounter=-10;
Legato.silent=true;
// Fading-out done, now set the catch-up :
Legato.decounter=Legato.fade.length;
Legato.msg=LM_CatchUp;
REALTYPE catchupfreq=Legato.param.freq*(Legato.param.freq/Legato.lastfreq);//This freq should make this now silent note to catch-up (or should I say resync ?) with the heard note for the same length it stayed at the previous freq during the fadeout.
SUBlegatonote(catchupfreq, Legato.param.vel, Legato.param.portamento, Legato.param.midinote, false);
break;
}
Legato.fade.m-=Legato.fade.step;
outl[i]*=Legato.fade.m;
outr[i]*=Legato.fade.m;
}
break;
default :
break;
}
// Check if the note needs to be computed more
if (AmpEnvelope->finished()!=0) {
for (i=0;i<SOUND_BUFFER_SIZE;i++) {//fade-out
REALTYPE tmp=1.0-(REALTYPE)i/(REALTYPE)SOUND_BUFFER_SIZE;
outl[i]*=tmp;
outr[i]*=tmp;
};
KillNote();
};
return(1);
};
int PADnote::noteout(REALTYPE *outl, REALTYPE *outr)
{
computecurrentparameters();
REALTYPE *smps = pars->sample[nsample].smp;
if(smps == NULL) {
for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
outl[i] = 0.0;
outr[i] = 0.0;
}
return 1;
}
REALTYPE smpfreq = pars->sample[nsample].basefreq;
REALTYPE freqrap = realfreq / smpfreq;
int freqhi = (int) (floor(freqrap));
REALTYPE freqlo = freqrap - floor(freqrap);
if(config.cfg.Interpolation)
Compute_Cubic(outl, outr, freqhi, freqlo);
else
Compute_Linear(outl, outr, freqhi, freqlo);
if(firsttime) {
fadein(outl);
fadein(outr);
firsttime = false;
}
NoteGlobalPar.GlobalFilterL->filterout(outl);
NoteGlobalPar.GlobalFilterR->filterout(outr);
//Apply the punch
if(NoteGlobalPar.Punch.Enabled != 0) {
for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
REALTYPE punchamp = NoteGlobalPar.Punch.initialvalue
* NoteGlobalPar.Punch.t + 1.0;
outl[i] *= punchamp;
outr[i] *= punchamp;
NoteGlobalPar.Punch.t -= NoteGlobalPar.Punch.dt;
if(NoteGlobalPar.Punch.t < 0.0) {
NoteGlobalPar.Punch.Enabled = 0;
break;
}
}
}
if(ABOVE_AMPLITUDE_THRESHOLD(globaloldamplitude, globalnewamplitude)) {
// Amplitude Interpolation
for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
REALTYPE tmpvol = INTERPOLATE_AMPLITUDE(globaloldamplitude,
globalnewamplitude,
i,
SOUND_BUFFER_SIZE);
outl[i] *= tmpvol * NoteGlobalPar.Panning;
outr[i] *= tmpvol * (1.0 - NoteGlobalPar.Panning);
}
}
else {
for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
outl[i] *= globalnewamplitude * NoteGlobalPar.Panning;
outr[i] *= globalnewamplitude * (1.0 - NoteGlobalPar.Panning);
}
}
// Apply legato-specific sound signal modifications
if(Legato.silent) // Silencer
if(Legato.msg != LM_FadeIn)
for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
outl[i] = 0.0;
outr[i] = 0.0;
}
switch(Legato.msg) {
case LM_CatchUp: // Continue the catch-up...
if(Legato.decounter == -10)
Legato.decounter = Legato.fade.length;
for(int i = 0; i < SOUND_BUFFER_SIZE; i++) { //Yea, could be done without the loop...
Legato.decounter--;
if(Legato.decounter < 1) {
// Catching-up done, we can finally set
// the note to the actual parameters.
Legato.decounter = -10;
Legato.msg = LM_ToNorm;
PADlegatonote(Legato.param.freq,
Legato.param.vel,
Legato.param.portamento,
Legato.param.midinote,
false);
break;
}
}
break;
case LM_FadeIn: // Fade-in
if(Legato.decounter == -10)
Legato.decounter = Legato.fade.length;
Legato.silent = false;
for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
Legato.decounter--;
if(Legato.decounter < 1) {
Legato.decounter = -10;
Legato.msg = LM_Norm;
break;
}
Legato.fade.m += Legato.fade.step;
outl[i] *= Legato.fade.m;
outr[i] *= Legato.fade.m;
}
break;
case LM_FadeOut: // Fade-out, then set the catch-up
if(Legato.decounter == -10)
Legato.decounter = Legato.fade.length;
for(int i = 0; i < SOUND_BUFFER_SIZE; i++) {
Legato.decounter--;
if(Legato.decounter < 1) {
for(int j = i; j < SOUND_BUFFER_SIZE; j++) {
outl[j] = 0.0;
outr[j] = 0.0;
}
Legato.decounter = -10;
Legato.silent = true;
// Fading-out done, now set the catch-up :
Legato.decounter = Legato.fade.length;
Legato.msg = LM_CatchUp;
REALTYPE catchupfreq = Legato.param.freq
* (Legato.param.freq / Legato.lastfreq); //This freq should make this now silent note to catch-up (or should I say resync ?) with the heard note for the same length it stayed at the previous freq during the fadeout.
PADlegatonote(catchupfreq,
Legato.param.vel,
Legato.param.portamento,
Legato.param.midinote,
false);
break;
}
Legato.fade.m -= Legato.fade.step;
outl[i] *= Legato.fade.m;
outr[i] *= Legato.fade.m;
}
break;
default:
break;
}
// Check if the global amplitude is finished.
// If it does, disable the note
if(NoteGlobalPar.AmpEnvelope->finished() != 0) {
for(int i = 0; i < SOUND_BUFFER_SIZE; i++) { //fade-out
REALTYPE tmp = 1.0 - (REALTYPE)i / (REALTYPE)SOUND_BUFFER_SIZE;
outl[i] *= tmp;
outr[i] *= tmp;
}
finished_ = 1;
}
return 1;
}
/*
* Master audio out (the final sound)
*/
void Master::AudioOut(REALTYPE *outl,REALTYPE *outr){
int i,npart,nefx;
/* //test!!!!!!!!!!!!! se poate bloca aici (mutex)
if (seq.play){
int type,par1,par2,again,midichan;
int ntrack=1;
// do{
again=seq.getevent(ntrack,&midichan,&type,&par1,&par2);
if (type>0) {
// printf("aaa\n");
if (type==1){//note_on or note_off
if (par2!=0) NoteOn(midichan,par1,par2);
else NoteOff(midichan,par1);
};
};
// } while (again);
};
*/
// printf("zzzz\n");
//Swaps the Left channel with Right Channel (if it is asked for)
if (swaplr!=0){
REALTYPE *tmp=outl;
outl=outr;
outr=tmp;
};
//clean up the output samples
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
outl[i]=0.0;
outr[i]=0.0;
};
//Compute part samples and store them part[npart]->partoutl,partoutr
for (npart=0;npart<NUM_MIDI_PARTS;npart++)
if (part[npart]->Penabled!=0) part[npart]->ComputePartSmps();
//Insertion effects
for (nefx=0;nefx<NUM_INS_EFX;nefx++){
if (Pinsparts[nefx]>=0) {
int efxpart=Pinsparts[nefx];
if (part[efxpart]->Penabled!=0)
insefx[nefx]->out(part[efxpart]->partoutl,part[efxpart]->partoutr);
};
};
//Apply the part volumes and pannings (after insertion effects)
for (npart=0;npart<NUM_MIDI_PARTS;npart++){
if (part[npart]->Penabled==0) continue;
REALTYPE newvol_l=part[npart]->volume;
REALTYPE newvol_r=part[npart]->volume;
REALTYPE oldvol_l=part[npart]->oldvolumel;
REALTYPE oldvol_r=part[npart]->oldvolumer;
REALTYPE pan=part[npart]->panning;
if (pan<0.5) newvol_l*=pan*2.0;
else newvol_r*=(1.0-pan)*2.0;
if (ABOVE_AMPLITUDE_THRESHOLD(oldvol_l,newvol_l)||
ABOVE_AMPLITUDE_THRESHOLD(oldvol_r,newvol_r)){//the volume or the panning has changed and needs interpolation
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
REALTYPE vol_l=INTERPOLATE_AMPLITUDE(oldvol_l,newvol_l,i,SOUND_BUFFER_SIZE);
REALTYPE vol_r=INTERPOLATE_AMPLITUDE(oldvol_r,newvol_r,i,SOUND_BUFFER_SIZE);
part[npart]->partoutl[i]*=vol_l;
part[npart]->partoutr[i]*=vol_r;
};
part[npart]->oldvolumel=newvol_l;
part[npart]->oldvolumer=newvol_r;
} else {
for (i=0;i<SOUND_BUFFER_SIZE;i++) {//the volume did not changed
part[npart]->partoutl[i]*=newvol_l;
part[npart]->partoutr[i]*=newvol_r;
};
};
};
//System effects
for (nefx=0;nefx<NUM_SYS_EFX;nefx++){
if (sysefx[nefx]->geteffect()==0) continue;//the effect is disabled
//Clean up the samples used by the system effects
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
tmpmixl[i]=0.0;
tmpmixr[i]=0.0;
};
//Mix the channels according to the part settings about System Effect
for (npart=0;npart<NUM_MIDI_PARTS;npart++){
//skip if the part has no output to effect
if (Psysefxvol[nefx][npart]==0) continue;
//skip if the part is disabled
if (part[npart]->Penabled==0) continue;
//the output volume of each part to system effect
REALTYPE vol=sysefxvol[nefx][npart];
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
tmpmixl[i]+=part[npart]->partoutl[i]*vol;
tmpmixr[i]+=part[npart]->partoutr[i]*vol;
};
};
// system effect send to next ones
for (int nefxfrom=0;nefxfrom<nefx;nefxfrom++){
if (Psysefxsend[nefxfrom][nefx]!=0){
REALTYPE v=sysefxsend[nefxfrom][nefx];
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
tmpmixl[i]+=sysefx[nefxfrom]->efxoutl[i]*v;
tmpmixr[i]+=sysefx[nefxfrom]->efxoutr[i]*v;
};
};
};
sysefx[nefx]->out(tmpmixl,tmpmixr);
//Add the System Effect to sound output
REALTYPE outvol=sysefx[nefx]->sysefxgetvolume();
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
outl[i]+=tmpmixl[i]*outvol;
outr[i]+=tmpmixr[i]*outvol;
};
};
//Mix all parts
for (npart=0;npart<NUM_MIDI_PARTS;npart++){
for (i=0;i<SOUND_BUFFER_SIZE;i++) {//the volume did not changed
outl[i]+=part[npart]->partoutl[i];
outr[i]+=part[npart]->partoutr[i];
};
};
//Insertion effects for Master Out
for (nefx=0;nefx<NUM_INS_EFX;nefx++){
if (Pinsparts[nefx] == -2)
insefx[nefx]->out(outl,outr);
};
//Master Volume
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
outl[i]*=volume;
outr[i]*=volume;
};
//Peak computation (for vumeters)
vuoutpeakl=1e-12;vuoutpeakr=1e-12;
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
if (fabs(outl[i])>vuoutpeakl) vuoutpeakl=fabs(outl[i]);
if (fabs(outr[i])>vuoutpeakr) vuoutpeakr=fabs(outr[i]);
};
if ((vuoutpeakl>1.0)||(vuoutpeakr>1.0)) vuclipped=1;
if (vumaxoutpeakl<vuoutpeakl) vumaxoutpeakl=vuoutpeakl;
if (vumaxoutpeakr<vuoutpeakr) vumaxoutpeakr=vuoutpeakr;
//RMS Peak computation (for vumeters)
vurmspeakl=1e-12;vurmspeakr=1e-12;
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
vurmspeakl+=outl[i]*outl[i];
vurmspeakr+=outr[i]*outr[i];
};
vurmspeakl=sqrt(vurmspeakl/SOUND_BUFFER_SIZE);
vurmspeakr=sqrt(vurmspeakr/SOUND_BUFFER_SIZE);
//Part Peak computation (for Part vumeters or fake part vumeters)
for (npart=0;npart<NUM_MIDI_PARTS;npart++){
vuoutpeakpart[npart]=1.0e-12;
if (part[npart]->Penabled!=0) {
REALTYPE *outl=part[npart]->partoutl,
*outr=part[npart]->partoutr;
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
REALTYPE tmp=fabs(outl[i]+outr[i]);
if (tmp>vuoutpeakpart[npart]) vuoutpeakpart[npart]=tmp;
};
vuoutpeakpart[npart]*=volume;
} else {
if (fakepeakpart[npart]>1) fakepeakpart[npart]--;
};
};
//Shutup if it is asked (with fade-out)
if (shutup!=0){
for (i=0;i<SOUND_BUFFER_SIZE;i++) {
REALTYPE tmp=(SOUND_BUFFER_SIZE-i)/(REALTYPE) SOUND_BUFFER_SIZE;
outl[i]*=tmp;
outr[i]*=tmp;
};
ShutUp();
};
//update the LFO's time
LFOParams::time++;
if (HDDRecorder.recording()) HDDRecorder.recordbuffer(outl,outr);
dump.inctick();
};
