void audioCB(AudioIOData& io){
while(io()){
if(tmr()) burst.reset();
//float s = io.in(0)[i];
float s = burst()*2;
// This short-hand method is convenient for simple delays.
s += delay(s);
// The delay-line can also be read from and written to using separate calls.
//delay.write(s);
//s += delay();
// We can create infinite echoes by feeding back a small amount of the
// output back into the input on each iteration.
//s = delay(s + delay()*0.2);
// We can also create mult-tap delay-lines through multiple calls to
// the read() method.
//s += delay(s) + delay.read(0.15) + delay.read(0.39);
// How about multi-tap feedback?
//s += delay(s + delay.read(0.197)*0.3 + delay.read(0.141)*0.4 + delay.read(0.093)*0.2)*0.5;
io.out(0) = io.out(1) = s;
}
}
void onAudio(AudioIOData& io){
while(io()){
// Increment waveform type
if(tmr()) (++waveform)%=13;
// Modulate modifier parameter with raised cosine
osc.mod(mod.cosU());
float s = 0.f;
switch(waveform){
// non-modifiable generators ordered from dull to bright:
case 0: s = osc.cos(); break; // Cosine approximation: one harmonic
case 1: s = osc.even3(); break; // Even harmonic sine-like wave (3rd order)
case 2: s = osc.even5(); break; // Even harmonic sine-like wave (5th order)
case 3: s = osc.tri(); break; // Triangle wave: 1/f^2 odd harmonics
case 4: s = osc.para(); break; // Parabola train: 1/f^2 all harmonics
case 5: s = osc.sqr()/4; break; // Square wave: 1/f odd harmonics
case 6: s = osc.down()/4; break; // Downward saw wave: 1/f all harmonics
case 7: s = osc.up()/4; break; // Upward saw wave: 1/f all harmonics
case 8: s = osc.imp()/4; break; // Impulse train: flat spectrum all harmonics
// modifiable generators ordered from dull to bright:
case 9: s = osc.stair()/4; break; // Mix between a square and impulse
case 10: s = osc.pulse()/4; break; // Mix between up and down
case 11: s = osc.line2()/4; break; // Mix between a saw and a triangle
case 12: s = osc.up2()/4; break; // Mix between two saws
}
io.out(0) = io.out(1) = s * 0.2f;
}
}
void audioCB(AudioIOData& io){
while(io()){
if(tmr()){
//good way to get random numbers in gamma. Uniform distribution between two numbers
int tempHarmonic = rnd::uni(1,6);
//set the center frequency by calling freq(freqYouWant)
filter.freq(tempHarmonic*fundamentalFrequency);
float tempWidth = (tempHarmonic*fundamentalFrequency)/rnd::uni(2,100);
//set the filter's width (related to Q) by calling width(widthYouWant).
filter.width(tempWidth);
//print the filter's center frequency and width
cout<< "frequency: " << (tempHarmonic*fundamentalFrequency) << endl << "width: " << tempWidth << endl;
}
//this is how we filter our source.
float s = filter(src())*5.0;
io.out(0) = io.out(1) = s;
}
}
void onAudio(AudioIOData& io){
while(io()){
if(tmr()){
// Our sequence of pitches
float pitches[] = {0,0,12,0,0,10,-5,0};
// Map pitch class to a frequency in Hz
float f = 55 * pow(2, pitches[step]/12.);
// Increment step counter
step = (step + 1) % 8;
// Set new target frequence of portamento
freq = f;
// Restart envelope using a soft reset (to avoid clicks)
env.resetSoft();
}
// Set saw frequency from portamento filter
saw.freq(freq());
// Get next envelope value
float e = env();
// Map envelope value to cutoff frequency
lpf.freq(e * (modCutoff.paraU()*6000 + 500) + 40);
// Generate next saw sample
float s = saw() * 0.3;
// Filter saw sample
s = lpf(s) * e;
// Send sample to DAC
io.out(0) = io.out(1) = s;
}
}
void audioCB(AudioIOData& io){
while(io()){
if(tmr()){
switch(feedType){
case 0: printf("Low-pass feedforward\n");
comb.feeds( 1,0); break;
case 1: printf("High-pass feedforward\n");
comb.feeds(-1,0); break;
case 2: printf("Low-pass feedback\n");
comb.feeds(0,0.7); break;
case 3: printf("High-pass feedback\n");
comb.feeds(0,-0.7); break;
case 4: printf("Low-pass dual-feed\n");
comb.feeds(0.7,0.7); break;
case 5: printf("High-pass dual-feed\n");
comb.feeds(-0.7,-0.7); break;
case 6: printf("All-pass 1\n");
comb.feeds(0.9,-0.9); break;
case 7: printf("All-pass 2\n");
comb.feeds(-0.9,0.9); break;
}
(++feedType) %= 8;
}
float s = src()*0.4;
comb.ipolType(ipl::ROUND);
comb.delay(mod.triU() * 1./400 + 1./10000);
s = comb(s);
io.out(0) = io.out(1) = s;
}
}
// Audio callback
virtual void onSound(AudioIOData& io){
// Set the base frequency to 55 Hz
double freq = 55/io.framesPerSecond();
while(io()){
// Update the oscillators' phase
phase += freq;
if(phase > 1) phase -= 1;
// Generate two sine waves at the 5th and 4th harmonics
float out1 = cos(5*phase * 2*M_PI);
float out2 = sin(4*phase * 2*M_PI);
// Write the waveforms to the ring buffer.
// Note that this call CANNOT block or wait on a lock from somewhere else. The audio
// thread must keep writing to the buffer without regard for what other threads
// might be reading from it.
ring.write(Vec2f(out1, out2));
// Send scaled waveforms to output...
io.out(0) = out1*0.2;
io.out(1) = out2*0.2;
}
}
void onAudio(AudioIOData& io){
while(io()){
float s = src()*4;
if(stft(s)){
// Set period of pluck envelopes, in seconds.
// This is negative to get a downward ramp going from 1 to 0.
pluckEnvs.period(-60);
// Create a moving ramp function in the aux buffer
float amp = pluckEnvs();
for(unsigned k=0; k<stft.numBins(); ++k){
float A = scl::wrap(amp + float(k)/stft.numBins());
// This gives the envelopes an exponential-like decay
for(int j=0; j<5; ++j) A*=A;
stft.aux(0)[k] = A;
}
// Do a seeded random shuffle of the envelopes so we don't hear
// a pattern.
rnd::push(14122);
rnd::permute(stft.aux(0), stft.numBins());
rnd::pop();
for(unsigned k=0; k<stft.numBins(); ++k){
stft.bin(k) *= stft.aux(0)[k];
}
}
io.out(0) = io.out(1) = stft();
}
}
void audioCB(AudioIOData& io){
while(io()){
if(tmr()){
switch(flangeType){
case 0: printf("Low-pass feedforward\n");
flanger.feeds(0.7, 0); break;
case 1: printf("High-pass feedforward\n");
flanger.feeds(-0.7,0); break;
case 2: printf("Low-pass feedback\n");
flanger.feeds(0,0.7); break;
case 3: printf("High-pass feedback\n");
flanger.feeds(0,-0.7); break;
case 4: printf("Low-pass dual-feed\n");
flanger.feeds(0.7,0.7); break;
case 5: printf("High-pass dual-feed\n");
flanger.feeds(-0.7,-0.7); break;
case 6: printf("All-pass 1\n");
flanger.feeds(0.9,-0.9); break;
case 7: printf("All-pass 2\n");
flanger.feeds(-0.9,0.9); break;
}
(++flangeType) %= 8;
}
float s = src.up();
s = flanger(s)*0.1;
io.out(0) = io.out(1) = s;
}
}
void onAudio(AudioIOData& io) {
while(io()) {
// Our signal is a saw wave plus a little noise
float s = saw()*0.3 + noise()*0.04;
if(stft(s)) {
for(unsigned k=0; k<stft.numBins(); ++k) {
// Get the bin magnitude (the first bin element)
float mag = stft.bin(k)[0];
// If the bin magnitude is less than our threshold, then we
// zero its magnitude. The assumption here is that noisy bins
// will have a relatively small magnitude.
if(mag < 0.0004) stft.bin(k)[0] = 0;
// If we flip the comparison, then we keep only the noise.
//if(mag > 0.0001) stft.bin(k)[0] = 0;
}
}
// Get next resynthesized sample
// (Comment this out to hear the original signal with noise.)
s = stft();
io.out(0) = io.out(1) = s;
}
}
void onAudio(AudioIOData& io){
while(io()){
if(syncMode()){
softSync^=true;
printf("%s sync\n", softSync?"soft":"hard");
}
// When the sync osc completes cycle, reset the phase of the formant osc
float w = sync.up();
if(sync.cycled()) formant.phase(0);
// Set the pitch of the formants
formant.freq(mod.triU()*2200+200);
// The formant oscillator determines the spectral envelope
float s = formant.tri();
// Drop amp to zero at sync points?
if(softSync){
w = 1.f - scl::pow8(w); // flattop window
s *= w; // apply window
}
io.out(0) = io.out(1) = s*0.2;
}
}
void onAudio(AudioIOData& io){
while(io()){
float s = (player1() + player2()) * 0.5;
player1.loop();
player2.loop();
io.out(0) = io.out(1) = s;
}
}
void onProcess(AudioIOData& io){
while(io()){
float2 s = float2(io.out(0), io.out(1));
s = chorus(s)*0.5;
io.out(0) += s[0];
io.out(1) += s[1];
}
}
void audioCB(AudioIOData& io){
while(io()){
float s = (osc1() + osc2()) * 0.1;
io.out(0) = io.out(1) = s;
}
}
void audioCB(AudioIOData& io){
while(io()){
float s = sweep(); // upward ramp between 0 and 1
s = table[int(s*N)];
io.out(0) = io.out(1) = s*0.2;
}
}
static void audioCB(AudioIOData& io){
MyApp * self = (MyApp *)io.user();
while(io()){
self->phase += 0.001;
float s0 = rng.uniformS() * sin(self->phase);
float s1 = sin(self->phase * 100.);
io.out(0) = s0;
io.out(1) = s1;
}
}
void audioCB(AudioIOData& io){
while(io()){
float s = osc();
io.out(0) = io.out(1) = s * 0.2f;
osc.phaseAdd(s*0.01*mod.hann());
//osc.freq((s*0.5+0.5)*mod.hann()*800 + 400);
}
}
void audioCB(AudioIOData& io){
while(io()){
oscM.freq(modFreq.hann() * 110 + 1); // change modulator frequency
float s = oscC() * (oscM() * 0.5 + 0.5) * 0.2;
io.out(0) = io.out(1) = s;
}
}
void audioCB(AudioIOData& io){
while(io()){
int nh = mod.triU() * 64;
buzz.harmonics(nh);
float s = buzz();
io.out(0) = io.out(1) = s*0.2;
}
}
void audioCB(AudioIOData& io){
while(io()){
oscM.freq(modFreq.hann() * 110 + 1); // change modulator frequency
oscC.freq(ff + oscM()*100); // modulate frequency of carrier
float s = oscC() * 0.2;
io.out(0) = io.out(1) = s;
}
}
void audioCB(AudioIOData& io){
while(io()){
if(tmr()){
env.value(0.2); // reset amplitude of envelope
}
float s = src() * env();
io.out(0) = io.out(1) = s;
}
}
void audioCB(AudioIOData& io){
while(io()){
float cutoff = scl::pow3(mod.triU()) * 10000;
filt.freq(cutoff);
float s = filt(src());
io.out(0) = io.out(1) = s * 0.2f;
}
}
void audioCB(AudioIOData& io){
while(io()){
float s = 0;
if(tmr()){
tmr.period(rnd::pick(0.4, 0.2));
s = 0.2;
}
io.out(0) = io.out(1) = s;
}
}
void audioCB(AudioIOData& io){
while(io()){
if(tmr()){
env = fund*2 + rnd::uni(10.0); // set new target value of envelope
}
osc2.freq(env()); // modulate frequency of 2nd harmonic
float s = (osc1() + osc2()) * 0.1;
io.out(0) = io.out(1) = s;
}
}
void audioCB(AudioIOData& io){
while(io()){
if(tmr()){
freq = pow(2, rnd::uniS(1.))*440;
}
src.freq(freq());
float s = src();
io.out(0) = io.out(1) = s * 0.2f;
}
}
// Our main audio callback
void audioCB(AudioIOData& io){
UserData& user = *(UserData *)io.user();
float ampL = user.ampL;
float ampR = user.ampR;
// loop through the number of samples in the block
while(io()){
float s = io.in(0); // get the line-in or microphone sample
io.out(0) = s * ampL; // set left and right output channel samples
io.out(1) = s * ampR;
}
}
// create a callback for generating a block of samples
void audioCB(AudioIOData& io){
UserData& user = *(UserData *)io.user();
float ampL = user.ampL;
float ampR = user.ampR;
// loop through the number of samples in the block
for(int i=0; i<io.framesPerBuffer(); ++i){
float s = io.in(0,i); // get the line-in or microphone sample
io.out(0,i) = s * ampL; // set left and right output channel samples
io.out(1,i) = s * ampR;
}
}
void audioCB(AudioIOData& io){
while(io()){
float s = osc() * 0.2;
io.out(0) = s;
io.out(1) = s;
}
}
void audioCB(AudioIOData& io){
while(io()){
float s = src();
// Input next sample for analysis
// When this returns true, then we have a new spectral frame
if(stft(s)){
float frac = scl::pow3( edge.triU() );
int N = stft.numBins();
for(int k=0; k<N; ++k){
int indKnee = frac * N;
stft.bin(k) *= k < indKnee ? 1:0;
}
}
// Get next resynthesized sample
s = stft() * 0.2;
io.out(0) = s;
io.out(1) = s;
}
}
void onAudio(AudioIOData& io){
while(io()){
float s = src();
if(stft(s)){
// Define the band edges, in Hz
float freqLo = 400;
float freqHi = 1800;
for(unsigned k=0; k<stft.numBins(); ++k){
// Compute the frequency, in Hz, of this bin
float freq = k*stft.binFreq();
// If the bin frequency is outside of our band, then zero
// the bin.
if(freq < freqLo || freq > freqHi){
stft.bin(k) = 0;
}
}
}
s = stft();
io.out(0) = s;
io.out(1) = s;
}
}
void audioCB(AudioIOData& io){
while(io()){
if(tmr()){
if(rnd::prob(0.1/1)) pluck1.reset();
if(rnd::prob(0.1/2)) pluck2.reset();
if(rnd::prob(0.1/3)) pluck3.reset();
if(rnd::prob(0.1/4)) pluck4.reset();
}
float s = pluck1() + pluck2() + pluck3() + pluck4();
io.out(0) = io.out(1) = s * 0.2f;
}
}
