int AM::run()
{
const int samps = framesToRun() * inputChannels();
rtgetin(in, this, samps);
for (int i = 0; i < samps; i += inputChannels()) {
if (--branch <= 0) {
double p[7];
update(p, 7, kAmp | kFreq | kPan);
amp = p[3];
if (amptable)
amp *= tablei(currentFrame(), amptable, amptabs);
if (freqtable)
modfreq = tablei(currentFrame(), freqtable, freqtabs);
else
modfreq = p[4];
modosc->setfreq(modfreq);
spread = p[6];
branch = skip;
}
float out[2];
out[0] = in[i + inchan] * modosc->next() * amp;
if (outputChannels() == 2) {
out[1] = out[0] * (1.0 - spread);
out[0] *= spread;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int WAVESHAPE::run()
{
for (int i = 0; i < framesToRun(); i++) {
if (--branch <= 0) {
doupdate();
branch = skip;
}
float sig = osc->next();
float wsig = wshape(sig * index, xferfunc, lenxfer);
// dc blocking filter
float osig = a1 * z1;
z1 = b1 * z1 + wsig;
osig += a0 * z1;
float out[2];
out[0] = osig * amp;
if (outputChannels() == 2) {
out[1] = (1.0 - spread) * out[0];
out[0] *= spread;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int MMESH2D :: run()
{
int i;
float out[2];
for (i = 0; i < framesToRun(); i++) {
if (--branch <= 0) {
double p[10];
update (p, 10, kAmp | kPan);
amp = p[2];
if (amptable)
amp *= theEnv->next(currentFrame());
branch = getSkip();
}
out[0] = dcblocker->next(theMesh->tick()) * amp;
if (outputChannels() == 2) {
out[1] = out[0] * (1.0 - pctleft);
out[0] *= pctleft;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int WAVETABLE::run()
{
const int nframes = framesToRun();
for (int i = 0; i < nframes; i++) {
if (--branch <= 0) {
if (fastUpdate) {
if (amptable)
amp = ampmult * tablei(currentFrame(), amptable, amptabs);
}
else
doupdate();
branch = getSkip();
}
float out[2];
out[0] = osc->next() * amp;
if (outputChannels() == 2) {
out[1] = (1.0 - spread) * out[0];
out[0] *= spread;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int STEREO::run()
{
const int inchans = inputChannels();
const int samps = framesToRun() * inchans;
rtgetin(in, this, samps);
for (int i = 0; i < samps; i += inchans) {
if (--branch <= 0) {
if (fastUpdate) {
if (amptable)
amp = ampmult * tablei(currentFrame(), amptable, amptabs);
}
else
doupdate();
branch = getSkip();
}
float out[2];
out[0] = out[1] = 0.0;
for (int j = 0; j < inchans; j++) {
if (outPan[j] >= 0.0) {
out[0] += in[i+j] * outPan[j] * amp;
out[1] += in[i+j] * (1.0 - outPan[j]) * amp;
}
}
rtaddout(out);
increment();
}
return framesToRun();
}
int MULTIWAVE::run()
{
const int samps = framesToRun();
const int chans = outputChannels();
float out[chans];
for (int i = 0; i < samps; i++) {
if (--branch <= 0) {
doupdate();
branch = getSkip();
}
for (int j = 0; j < chans; j++)
out[j] = 0.0f;
for (int j = 0; j < numpartials; j++) {
float sig = oscil[j]->next() * amp[j];
if (chans == 1)
out[0] += sig;
else {
out[0] += sig * pan[j];
out[1] += sig * (1.0 - pan[j]);
}
}
float scale = (1.0 / float(numpartials)) * (overall_amp * chans);
for (int j = 0; j < chans; j++)
out[j] *= scale;
rtaddout(out);
increment();
}
return framesToRun();
}
int BEND1::run()
{
for (int i = 0; i < framesToRun(); i++) {
if (--branch <= 0) {
if (amptable)
aamp = tablei(currentFrame(), amptable, amptabs) * amp;
float freq = diff * tablei(currentFrame(), glissf, tags) + freq0;
sset(SR, freq, tf0, tfN, strumq1);
branch = reset;
}
float a = strum(d, strumq1);
float b = dist(dgain*a);
d = fbgain*delay(b, dq);
float out[2];
out[0] = (cleanlevel*a + distlevel*b) * aamp;
if (outputChannels() == 2) { /* split stereo files between the channels */
out[1] = (1.0 - spread) * out[0];
out[0] *= spread;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int MBOWED :: run()
{
int i;
float out[2];
for (i = 0; i < framesToRun(); i++) {
if (--branch <= 0) {
doupdate();
branch = getSkip();
}
if (--vibupdate < 0) { // reset the vibrato freq after each cycle
float vibfreq = theRand->range(viblo, vibhi);
theVib->setfreq(vibfreq);
vibupdate = (int)(SR/vibfreq);
}
out[0] = theBow->tick(bowvel) * amp;
theBow->setFrequency(freqbase + (freqamp * ( (theVib->next()+2.0) * 0.5 )));
if (outputChannels() == 2) {
out[1] = out[0] * (1.0 - pctleft);
out[0] *= pctleft;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int DISTORT::run()
{
const int insamps = framesToRun() * inputChannels();
rtgetin(in, this, insamps);
for (int i = 0; i < insamps; i += inputChannels()) {
if (--branch <= 0) {
doupdate();
branch = getSkip();
}
float sig = in[i + inchan];
if (!bypass) {
sig *= (gain / 32768.0f); // apply gain, convert range
sig = distort->next(sig, param);
sig *= 32768.0f;
if (usefilt)
sig = filt->tick(sig);
}
sig *= amp;
float out[2];
if (outputChannels() == 2) {
out[0] = sig * pctleft;
out[1] = sig * (1.0f - pctleft);
}
else
out[0] = sig;
rtaddout(out);
increment();
}
return framesToRun();
}
int IINOISE::run()
{
for (int i = 0; i < framesToRun(); i++) {
if (--branch <= 0) {
if (fastUpdate) {
if (amptable)
amp = ampmult * tablei(currentFrame(), amptable, amptabs);
}
else
doupdate();
branch = getSkip();
}
float sig = rrand();
float out[2];
out[0] = 0.0f;
for (int j = 0; j < nresons; j++)
out[0] += resons[j]->next(sig) * resonamp[j];
out[0] *= amp;
if (outputChannels() == 2) {
out[1] = out[0] * (1.0f - pan);
out[0] *= pan;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int FMINST::run()
{
const int nframes = framesToRun();
for (int i = 0; i < nframes; i++) {
if (--branch <= 0) {
if (fastUpdate) {
if (amptable)
amp = ampmult * tablei(currentFrame(), amptable, amptabs);
float guide = tablei(currentFrame(), indexenv, indtabs);
peakdev = modfreq * (minindex + (indexdiff * guide));
}
else
doupdate();
branch = getSkip();
}
float out[2];
float modsig = modosc->next() * peakdev;
carosc->setfreq(carfreq + modsig);
out[0] = carosc->next() * amp;
if (outputChannels() == 2) {
out[1] = (1.0 - pan) * out[0];
out[0] *= pan;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int WAVY::run()
{
const int frames = framesToRun();
const int chans = outputChannels();
for (int i = 0; i < frames; i++) {
if (--_branch <= 0) {
doupdate();
_branch = getSkip();
}
float sig1 = _oscilA->nexti();
float sig2 = _oscilB->nexti();
float out[chans];
if (_fp)
out[0] = eval(sig1, sig2) * _amp;
else
out[0] = (*_combiner)(sig1, sig2) * _amp;
if (chans == 2) {
out[1] = out[0] * (1.0f - _pan);
out[0] *= _pan;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int REVMIX::run()
{
int samps = framesToRun() * inputChannels();
rtinrepos(this, -framesToRun(), SEEK_CUR);
rtgetin(in, this, samps);
for (int i = samps - inputChannels(); i >= 0; i -= inputChannels()) {
if (--branch <= 0) {
double p[nargs];
update(p, nargs);
amp = p[3];
if (amparray)
amp *= tablei(currentFrame(), amparray, amptabs);
pctleft = nargs > 5 ? p[5] : 0.5; // default is .5
branch = skip;
}
float out[2];
out[0] = in[i + inchan] * amp;
if (outputChannels() == 2) {
out[1] = out[0] * (1.0 - pctleft);
out[0] *= pctleft;
}
rtaddout(out);
increment();
}
rtinrepos(this, -framesToRun(), SEEK_CUR);
return framesToRun();
}
int JFIR :: run()
{
const int samps = framesToRun() * inputChannels();
rtgetin(in, this, samps);
for (int i = 0; i < samps; i += inputChannels()) {
if (--branch <= 0) {
doupdate();
branch = getSkip();
}
float insig;
if (currentFrame() < insamps) // still taking input
insig = in[i + inchan] * amp;
else // in ring-down phase
insig = 0.0;
float out[2];
if (bypass)
out[0] = insig;
else
out[0] = filt->tick(insig);
if (outputchans == 2) {
out[1] = out[0] * (1.0 - pctleft);
out[0] *= pctleft;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int CRACKLE::run()
{
for (int i = 0; i < framesToRun(); i++) {
if (--branch <= 0) {
doupdate();
branch = getSkip();
}
x3 = x2;
x2 = x1;
x1 = x0;
x0 = std::abs(0.03 - x2 + x3 - param * x1);
float out[2];
out[0] = x0 * amp * 5;
if (outputChannels() == 2) {
out[1] = out[0] * (1.0 - pan);
out[0] *= pan;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int BROWN::run()
{
for (int i = 0; i < framesToRun(); i++) {
if (--_branch <= 0) {
doupdate();
_branch = getSkip();
}
float out[2];
while (true) {
float r = rrand();
_brown += r;
if (_brown <- 8.0 || _brown > 8.0)
_brown -= r;
else
break;
}
out[0] = _brown * 0.125 * _amp;
if (outputChannels() == 2) {
out[1] = out[0] * (1.0 - _pan);
out[0] *= _pan;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int MMODALBAR :: run()
{
int i;
float out[2];
for (i = 0; i < framesToRun(); i++) {
if (--branch <=0) {
doupdate();
branch = getSkip();
}
if (currentFrame() < 256) // feed in excitation
out[0] = theBar->tick(exciteamp, excite[currentFrame()]) * amp;
else
out[0] = theBar->tick(exciteamp, 0.0) * amp;
if (outputChannels() == 2) {
out[1] = out[0] * (1.0 - pctleft);
out[0] *= pctleft;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int SHAPE :: run()
{
const int samps = framesToRun() * inputChannels();
rtgetin(in, this, samps);
for (int i = 0; i < samps; i += inputChannels()) {
if (--branch <= 0) {
doupdate();
branch = skip;
}
// NB: WavShape deals with samples in range [-1, 1].
float insig = in[i + inchan] * (1.0 / 32768.0);
float outsig = shaper->tick(insig * index);
if (outsig) {
if (ampnorm)
outsig = dcblocker->tick(outsig) * ampnorm->tick(norm_index);
else
outsig = dcblocker->tick(outsig);
}
float out[2];
out[0] = outsig * amp * 32768.0;
if (outputChannels() == 2) {
out[1] = out[0] * (1.0 - pctleft);
out[0] *= pctleft;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int NOISE :: run()
{
for (int i = 0; i < framesToRun(); i++) {
if (--branch <= 0) {
double p[nargs];
update(p, nargs);
amp = p[2];
if (amparray)
amp *= tablei(currentFrame(), amparray, amptabs);
pctleft = nargs > 3 ? p[3] : 0.5; // default is .5
branch = skip;
}
float out[2];
out[0] = rrand() * amp;
if (outputChannels() == 2) {
out[1] = out[0] * (1.0 - pctleft);
out[0] *= pctleft;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int TRANS3::run()
{
const int outframes = framesToRun();
const int inchans = inputChannels();
float *outp = outbuf; // point to inst private out buffer
double frac;
for (int i = 0; i < outframes; i++) {
if (--branch <= 0) {
doupdate();
branch = getSkip();
}
while (getframe) {
if (inframe >= RTBUFSAMPS) {
rtgetin(in, this, RTBUFSAMPS * inchans);
inframe = 0;
}
oldersig = oldsig;
oldsig = newsig;
newsig = newestsig;
newestsig = in[(inframe * inchans) + inchan];
inframe++;
incount++;
if (counter - (double) incount < 0.0)
getframe = false;
}
// const double frac = (counter - (double) incount) + 2.0;
const double frac = (counter - (double) incount) + 1.0;
outp[0] = interp3rdOrder(oldersig, oldsig, newsig, newestsig, frac) * amp;
#ifdef DEBUG_FULL
printf("i: %d counter: %g incount: %d frac: %g inframe: %d cursamp: %d\n",
i, counter, incount, frac, inframe, currentFrame());
printf("interping %g, %g, %g, %g => %g\n", oldersig, oldsig, newsig, newestsig, outp[0]);
#endif
if (outputChannels() == 2) {
outp[1] = outp[0] * (1.0 - pctleft);
outp[0] *= pctleft;
}
outp += outputChannels();
increment();
counter += _increment; // keeps track of interp pointer
if (counter - (double) incount >= 0.0)
getframe = true;
}
#ifdef DEBUG
printf("OUT %d frames\n\n", i);
#endif
return framesToRun();
}
int MYINST::run()
{
// framesToRun() gives the number of sample frames -- 1 sample for each
// channel -- that we have to write during this scheduler time slice.
const int samps = framesToRun() * inputChannels();
// Read <samps> samples from the input file (or audio input device).
rtgetin(_in, this, samps);
// Each loop iteration processes 1 sample frame. */
for (int i = 0; i < samps; i += inputChannels()) {
// This block updates certain parameters at the control rate -- the
// rate set by the user with the control_rate() or reset() script
// functions. The Instrument base class holds this value as a number
// of sample frames to skip between updates. Get this value using
// getSkip() to reset the <_branch> counter.
if (--_branch <= 0) {
doupdate();
_branch = getSkip();
}
// Grab the current input sample, scaled by the amplitude multiplier.
float insig = _in[i + _inchan] * _amp;
float out[2]; // Space for only 2 output chans!
// Just copy it to the output array with no processing.
out[0] = insig;
// If we have stereo output, use the pan pfield.
if (outputChannels() == 2) {
out[1] = out[0] * (1.0f - _pan);
out[0] *= _pan;
}
// Write this sample frame to the output buffer.
rtaddout(out);
// Increment the count of sample frames this instrument has written.
increment();
}
// Return the number of frames we processed.
return framesToRun();
}
int REVERBIT::run()
{
int samps = framesToRun() * inputChannels();
if (currentFrame() < insamps)
rtgetin(in, this, samps);
for (int i = 0; i < samps; i += inputChannels()) {
if (--branch <= 0) {
doupdate();
branch = getSkip();
}
float insig[2], out[2];
if (currentFrame() < insamps) { // still taking input from file
insig[0] = in[i] * amp;
insig[1] = (inputChannels() == 2) ? in[i + 1] * amp : insig[0];
}
else // in ring-down phase
insig[0] = insig[1] = 0.0;
float rvbsig = -reverbpct * reverb(insig[0] + insig[1], rvbarray);
if (usefilt)
rvbsig = tone(rvbsig, tonedata);
delput(rvbsig, delarray, deltabs);
float delsig = delget(delarray, rtchan_delaytime, deltabs);
out[0] = insig[0] + rvbsig;
out[1] = insig[1] + delsig;
if (dcblock) {
float tmp_in[2];
tmp_in[0] = out[0];
tmp_in[1] = out[1];
out[0] = tmp_in[0] - prev_in[0] + (0.99 * prev_out[0]);
prev_in[0] = tmp_in[0];
prev_out[0] = out[0];
out[1] = tmp_in[1] - prev_in[1] + (0.99 * prev_out[1]);
prev_in[1] = tmp_in[1];
prev_out[1] = out[1];
}
rtaddout(out);
increment();
}
return framesToRun();
}
int ROOM::run()
{
const int samps = framesToRun() * inputChannels();
rtgetin(in, this, samps);
for (int i = 0; i < samps; i += inputChannels()) {
if (--branch <= 0) {
if (amparray)
aamp = tablei(currentFrame(), amparray, amptabs) * amp;
branch = skip;
}
float insig;
if (currentFrame() < insamps) { /* still taking input */
if (inchan == AVERAGE_CHANS) {
insig = 0.0;
for (int n = 0; n < inputChannels(); n++)
insig += in[i + n];
insig /= (float) inputChannels();
}
else
insig = in[i + inchan];
}
else /* in ring-down phase */
insig = 0.0;
echo[jpoint++] = insig;
if (jpoint >= nmax)
jpoint -= nmax;
float out[2];
out[0] = out[1] = 0.0;
for (int j = 0; j < NTAPS; j++) {
float e = echo[ipoint[j]];
out[0] += e * lamp[j];
out[1] += e * ramp[j];
ipoint[j]++;
if (ipoint[j] >= nmax)
ipoint[j] -= nmax;
}
if (aamp != 1.0) {
out[0] *= aamp;
out[1] *= aamp;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int COMBIT::run()
{
int samps = framesToRun() * inputChannels();
if (currentFrame() < insamps)
rtgetin(in, this, samps);
for (int i = 0; i < samps; i += inputChannels()) {
if (--branch <= 0) {
double p[8];
update(p, 8);
amp = p[3];
if (amptable) {
#ifdef EMBEDDED
amp *= rtcmix_table(currentFrame(), amptable, tabs);
#else
amp *= table(currentFrame(), amptable, tabs);
#endif
}
if (p[4] != frequency) {
frequency = p[4];
delsamps = (int) ((1.0 / frequency) * SR + 0.5);
}
if (p[5] != rvbtime) {
rvbtime = p[5];
comb->setReverbTime(rvbtime);
}
pctleft = p[7];
branch = skip;
}
float insig, out[2];
if (currentFrame() < insamps)
insig = in[i + inchan];
else
insig = 0.0;
out[0] = comb->next(insig, delsamps) * amp;
if (outputChannels() == 2) {
out[1] = out[0] * (1.0 - pctleft);
out[0] *= pctleft;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int DUMP::run()
{
float out[2] = {0.0, 0.0};
for (int i = 0; i < framesToRun(); i++) {
if (--branch <= 0) {
doupdate();
branch = skip;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int GRANSYNTH::run()
{
// NOTE: Without a lot more code, we can't guarantee that doupdate will
// be called exactly every getSkip() samples, the way we can when not doing
// block I/O. But this seems worth sacrificing for the clear performance
// improvement that block I/O offers.
const int frames = framesToRun();
int blockframes = min(frames, getSkip());
int framesdone = 0;
while (1) {
if (_branch <= 0) {
doupdate();
_branch = getSkip();
}
_branch -= blockframes;
_stream->processBlock(_block, blockframes, _amp);
rtbaddout(_block, blockframes);
increment(blockframes);
framesdone += blockframes;
if (framesdone == frames)
break;
assert(framesdone < frames);
const int remaining = frames - framesdone;
if (remaining < blockframes)
blockframes = remaining;
}
return frames;
}
int GRANSYNTH::run()
{
const int frames = framesToRun();
const int outchans = outputChannels();
int i;
for (i = 0; i < frames; i++) {
if (--_branch <= 0) {
doupdate();
_branch = getSkip();
}
_stream->prepare();
float out[outchans];
if (outchans == 2) {
out[0] = _stream->lastL() * _amp;
out[1] = _stream->lastR() * _amp;
}
else
out[0] = _stream->lastL() * _amp;
rtaddout(out);
increment();
}
return i;
}
int GVERB::run()
{
const int samps = framesToRun() * inputChannels();
int i;
float out[2];
rtgetin(in, this, samps);
for (i = 0; i < samps; i += inputChannels()) {
if (--branch <= 0) {
doupdate();
branch = getSkip();
}
if (currentFrame() > inputframes) in[i+inputchan] = 0.0;
gverb_do(p, in[i+inputchan], out, out+1);
out[0] = (out[0] * amp) + (in[i+inputchan] * p->drylevel);
out[1] = (out[1] * amp) + (in[i+inputchan] * p->drylevel);
rtaddout(out);
increment();
}
return i;
}
int LSFLUTE::run()
{
for (int i = 0; i < framesToRun(); i++) {
if (olength1 < length1) olength1++;
if (olength1 > length1) olength1--;
if (olength2 < length2) olength2++;
if (olength2 > length2) olength2--;
if (--branch <= 0) {
aamp = tablei(currentFrame(), amparr, amptabs);
oamp = tablei(currentFrame(), oamparr, oamptabs);
branch = skip;
}
float sig = (rrand() * namp * aamp) + aamp;
float del1sig = mdelget(del1ptr,olength1,dl1ptr);
sig = sig + (del1sig * -0.35);
#ifdef MAXMSP
delput(sig,del2ptr,dl2ptr);
#else
mdelput(sig,del2ptr,dl2ptr);
#endif
sig = mdelget(del2ptr,olength2,dl2ptr);
sig = (sig * sig * sig) - sig;
sig = (0.4 * sig) + (0.9 * del1sig);
float out[2];
out[0] = sig * amp * oamp;
sig = (dampcoef * sig) + ((1.0 - dampcoef) * oldsig);
oldsig = sig;
#ifdef MAXMSP
delput(sig,del1ptr,dl1ptr);
#else
mdelput(sig,del1ptr,dl1ptr);
#endif
if (outputChannels() == 2) {
out[1] = (1.0 - spread) * out[0];
out[0] *= spread;
}
rtaddout(out);
increment();
}
return framesToRun();
}
int FREEVERB :: run()
{
float *inL, *inR, *outL, *outR;
inL = in;
inR = inputChannels() > 1 ? in + 1 : in;
outL = outbuf;
outR = outputChannels() > 1 ? outbuf + 1 : outbuf;
int samps = framesToRun() * inputChannels();
if (currentFrame() < insamps)
rtgetin(in, this, samps);
// Scale input signal by amplitude multiplier and setline curve.
for (int i = 0; i < samps; i += inputChannels()) {
if (--branch <= 0) {
double p[11];
update(p, 11, kRoomSize | kPreDelay | kDamp | kDry | kWet | kWidth);
if (currentFrame() < insamps) { // amp is pre-effect
amp = update(3, insamps);
if (amparray)
amp *= tablei(currentFrame(), amparray, amptabs);
}
updateRvb(p);
branch = getSkip();
}
if (currentFrame() < insamps) { // still taking input from file
in[i] *= amp;
if (inputChannels() == 2)
in[i + 1] *= amp;
}
else { // in ringdown phase
in[i] = 0.0;
if (inputChannels() == 2)
in[i + 1] = 0.0;
}
increment();
}
// Hand off to Freeverb to do the actual work.
rvb->processreplace(inL, inR, outL, outR, framesToRun(), inputChannels(),
outputChannels());
return framesToRun();
}