void processAudio(AudioBuffer &buffer) {
float delayTime, feedback, wetDry;
delayTime = getParameterValue(PARAMETER_A);
feedback = getParameterValue(PARAMETER_B);
wetDry = getParameterValue(PARAMETER_D);
int size = buffer.getSize();
int32_t newDelay;
if(abs(time - delayTime) > 0.01){
newDelay = delayTime * (delayBuffer.getSize()-1);
time = delayTime;
}else{
newDelay = delay;
}
float* x = buffer.getSamples(0);
float y;
for (int n = 0; n < size; n++){
// y = buf[i] + feedback * delayBuffer.read(delay);
// buf[i] = wetDry * y + (1.f - wetDry) * buf[i];
// delayBuffer.write(buf[i]);
if(newDelay - delay > 4){
y = getDelayAverage(delay-5, 5);
delay -= 5;
}else if(delay - newDelay > 4){
y = getDelayAverage(delay+5, 5);
delay += 5;
}else{
y = delayBuffer.read(delay);
}
x[n] = wetDry * y + (1.f - wetDry) * x[n]; // crossfade for wet/dry balance
delayBuffer.write(feedback * x[n]);
}
}
void mixAudioBufferChannelsLogrithmicDRC(AudioBuffer &audioBuffer, std::vector<float> &channelLevels, AudioBuffer &mixBuffer, float threshold)
{
if(audioBuffer.getChannels() == 0)
return;
AudioFormat format=audioBuffer.getFormat();
unsigned int samples=audioBuffer.getSamples();
switch(format)
{
case AudioFormat::UInt8:
case AudioFormat::UInt8P:
mixChannelsLogrithmicDRC<uint8_t>((uint8_t *)audioBuffer.getBuffer(), channelLevels, (uint8_t *)mixBuffer.getBuffer(), samples, threshold);
break;
case AudioFormat::Int16:
case AudioFormat::Int16P:
mixChannelsLogrithmicDRC<int16_t>((int16_t *)audioBuffer.getBuffer(), channelLevels, (int16_t *)mixBuffer.getBuffer(), samples, threshold);
break;
case AudioFormat::Int32:
case AudioFormat::Int32P:
mixChannelsLogrithmicDRC<int32_t>((int32_t *)audioBuffer.getBuffer(), channelLevels, (int32_t *)mixBuffer.getBuffer(), samples, threshold);
break;
case AudioFormat::Float:
case AudioFormat::FloatP:
mixChannelsLogrithmicDRC<float>((float *)audioBuffer.getBuffer(), channelLevels, (float *)mixBuffer.getBuffer(), samples, threshold);
break;
case AudioFormat::Double:
case AudioFormat::DoubleP:
mixChannelsLogrithmicDRC<double>((double *)audioBuffer.getBuffer(), channelLevels, (double *)mixBuffer.getBuffer(), samples, threshold);
break;
}
}
void processAudio(AudioBuffer& buf){
float minf = getParameterValue(PARAMETER_A)*0.1 + 0.001;
float maxf = min(0.4, minf + getParameterValue(PARAMETER_B)*0.2);
// range should be exponentially related to minf
// int tones = getParameterValue(PARAMETER_C)*(TONES-1) + 1;
int tones = 12;
float spread = getParameterValue(PARAMETER_C) + 1.0;
float rate = 1.0 + (getParameterValue(PARAMETER_D) - 0.5)*0.00002;
int size = buf.getSize();
FloatArray out = buf.getSamples(LEFT_CHANNEL);
float amp;
for(int t=1; t<tones; ++t)
inc[t] = inc[t-1]*spread;
for(int i=0; i<size; ++i){
for(int t=0; t<tones; ++t){
amp = getAmplitude((inc[t]-minf)/(maxf-minf));
out[i] += amp * getWave(acc[t]);
acc[t] += inc[t];
if(acc[t] > 1.0)
acc[t] -= 1.0;
else if(acc[t] < 0.0)
acc[t] += 1.0;
inc[t] *= rate;
}
}
if(inc[0] > maxf)
inc[0] = minf;
// while(inc[0] > minf)
// inc[0] *= 0.5;
else if(inc[0] < minf)
inc[0] = maxf;
// while(inc[0] < maxf)
// inc[0] *= 2.0;
}
void processAudio(AudioBuffer &buffer){
double rate = getSampleRate();
unsigned int sampleDelay = getSampleDelay(getRampedParameterValue(PARAMETER_A), rate);
sampleDelay = min(sampleDelay, bufferSize);
float feedback = getRampedParameterValue(PARAMETER_B);
float bias = getBiasExponent(1 - getRampedParameterValue(PARAMETER_C));
float dryWetMix = getRampedParameterValue(PARAMETER_D);
int size = buffer.getSize();
for(int ch = 0; ch<buffer.getChannels(); ++ch)
{
float* buf = buffer.getSamples(ch);
for (int i=0; i<size; ++i)
{
float delaySample = circularBuffer[writeIdx];
float v = buf[i] + circularBuffer[writeIdx] * feedback;
v = applyBias(v, bias);
circularBuffer[writeIdx] = min(1, max(-1, v)); // Guard: hard range limits.
buf[i] = linearBlend(buf[i], delaySample, dryWetMix);
writeIdx = (++writeIdx) % sampleDelay;
}
}
}
void processAudio(AudioBuffer &buffer){
FloatArray l1 = buffer.getSamples(LEFT_CHANNEL);
FloatArray r1 = buffer.getSamples(RIGHT_CHANNEL);
FloatArray l2 = buf->getSamples(LEFT_CHANNEL);
FloatArray r2 = buf->getSamples(RIGHT_CHANNEL);
float morph = getParameterValue(MORPH_PARAMETER);
l2.copyFrom(l1);
r2.copyFrom(r1);
green.processAudio(*buf);
red.processAudio(buffer);
int size = buffer.getSize();
for(int i=0; i<size; ++i){
l1[i] = l1[i]*(1-morph) + l2[i]*morph;
r1[i] = r1[i]*(1-morph) + r2[i]*morph;
}
}
void processAudio(AudioBuffer &buffer) {
double rate = getSampleRate();
float p1 = getRampedParameterValue(PARAMETER_A);
float freq1 = p1*p1 * (MAX_FREQ-MIN_FREQ) + MIN_FREQ;
double step1 = freq1 / rate;
float amt1 = getRampedParameterValue(PARAMETER_B);
float p2 = getRampedParameterValue(PARAMETER_C);
float freq2 = p2*p2 * (MAX_FREQ-MIN_FREQ) + MIN_FREQ;
float amt2 = getRampedParameterValue(PARAMETER_D);
double step2 = freq2 / rate;
int size = buffer.getSize();
for(int ch = 0; ch<buffer.getChannels(); ++ch)
{
float* buf = buffer.getSamples(ch);
for (int i=0; i<size; ++i)
{
float mod1 = sin(2 * M_PI * phase1) / 2 + .5; // 0..1
float mod2 = sin(2 * M_PI * phase2) / 2 + .5; // 0..1
float gain1 = (amt1 * mod1) + (1 - amt1);
float gain2 = (amt2 * mod2) + (1 - amt2);
buf[i] = (gain1 * gain2) * buf[i];
phase1 += step1;
phase2 += step2;
}
}
}
void processAudio(AudioBuffer &buffer){
float y[getBlockSize()];
setCoeffs(getLpFreq(), 0.8f);
float delayTime = getParameterValue(PARAMETER_A); // get delay time value
float feedback = getParameterValue(PARAMETER_B); // get feedback value
float wetDry = getParameterValue(PARAMETER_D); // get gain value
if(abs(time - delayTime) < 0.01)
delayTime = time;
else
time = delayTime;
float delaySamples = delayTime * (delayBuffer.getSize()-1);
int size = buffer.getSize();
float* x = buffer.getSamples(0);
process(size, x, y); // low pass filter for delay buffer
for(int n = 0; n < size; n++){
//linear interpolation for delayBuffer index
dSamples = olddelaySamples + (delaySamples - olddelaySamples) * n / size;
y[n] = y[n] + feedback * delayBuffer.read(dSamples);
x[n] = (1.f - wetDry) * x[n] + wetDry * y[n]; //crossfade for wet/dry balance
delayBuffer.write(x[n]);
}
olddelaySamples = delaySamples;
}
void processAudio(AudioBuffer &buffer){
setCoeffs(getLpFreq(), 0.8f);
float delayTime = getParameterValue(PARAMETER_A); // get delay time value
float feedback = getParameterValue(PARAMETER_B); // get feedback value
float wetDry = getParameterValue(PARAMETER_D); // get gain value
float delaySamples = delayTime * (DELAY_BUFFER_LENGTH-1);
int size = buffer.getSize();
for (int ch = 0; ch<buffer.getChannels(); ++ch) {
float* buf = buffer.getSamples(ch);
process(size, buf, outBuf); // low pass filter for delay buffer
for(int i = 0; i < size; i++){
outBuf[i] = outBuf[i] + feedback * delayBuffer.read(delaySamples);
buf[i] = (1.f - wetDry) * buf[i] + wetDry * outBuf[i]; //crossfade for wet/dry balance
delayBuffer.write(buf[i]);
}
}
}
void processAudio(AudioBuffer &buffer) {
int size = buffer.getSize();
float y;
rate = Rate(getParameterValue(PARAMETER_A));
depth = getParameterValue(PARAMETER_B);
feedback = getParameterValue(PARAMETER_C);
//calculate and update phaser sweep lfo...
float d = _dmin + (_dmax-_dmin) * ((sin( _lfoPhase ) + 1.f)/2.f);
_lfoPhase += rate;
if( _lfoPhase >= M_PI * 2.f )
_lfoPhase -= M_PI * 2.f;
//update filter coeffs
for( int i=0; i<6; i++ )
_alps[i].Delay( d );
// for (int ch = 0; ch<buffer.getChannels(); ++ch) {
float* buf = buffer.getSamples(0);
for (int i = 0; i < size; i++) {
//calculate output
y = _alps[0].Update(_alps[1].Update(_alps[2].Update(_alps[3].Update(_alps[4].Update(
_alps[5].Update( buf[i] + _zm1 * feedback ))))));
_zm1 = y;
buf[i] = buf[i] + y * depth;
// }
}
}
void processAudio(AudioBuffer &buffer){
// assert_param(buffer.getChannels() > 1);
float gainLL = getParameterValue(PARAMETER_A);
float gainLR = getParameterValue(PARAMETER_B);
float gainRL = getParameterValue(PARAMETER_C);
float gainRR = getParameterValue(PARAMETER_D);
int size = buffer.getSize();
float* left = buffer.getSamples(0);
float* right = buffer.getChannels() > 1 ? buffer.getSamples(1) : left;
float l, r;
for(int i=0; i<size; ++i){
l = gainLL*left[i] + gainLR*right[i];
r = gainRL*left[i] + gainRR*right[i];
left[i] = l;
right[i] = r;
}
}
FixedDelayPatch() {
AudioBuffer* buffer = createMemoryBuffer(1, REQUEST_BUFFER_SIZE);
delayBuffer.initialise(buffer->getSamples(0), buffer->getSize());
registerParameter(PARAMETER_A, "Feedback");
registerParameter(PARAMETER_B, "Mix");
registerParameter(PARAMETER_C, "");
registerParameter(PARAMETER_D, "");
}
void processAudio(AudioBuffer &buffer) {
float fundamental = getParameterValue(PARAMETER_A)*5.0 - 1.0;
FloatArray left = buffer.getSamples(LEFT_CHANNEL);
hz.setTune(fundamental);
float freq = hz.getFrequency(0);
algo.setFrequency(freq);
algo.getSamples(left);
}
void PatchController::process(AudioBuffer& buffer){
if(activeSlot == GREEN && green.index != settings.patch_green){
memset(buffer.getSamples(0), 0, buffer.getChannels()*buffer.getSize()*sizeof(float));
// green must be active slot when patch constructor is called
green.setPatch(settings.patch_green);
codec.softMute(false);
debugClear();
return;
}else if(activeSlot == RED && red.index != settings.patch_red){
memset(buffer.getSamples(0), 0, buffer.getChannels()*buffer.getSize()*sizeof(float));
// red must be active slot when constructor is called
red.setPatch(settings.patch_red);
codec.softMute(false);
debugClear();
return;
}
switch(mode){
case SINGLE_MODE:
case DUAL_GREEN_MODE:
green.setParameterValues(getAnalogValues());
green.patch->processAudio(buffer);
break;
case DUAL_RED_MODE:
red.setParameterValues(getAnalogValues());
red.patch->processAudio(buffer);
break;
case SERIES_GREEN_MODE:
green.setParameterValues(getAnalogValues());
green.patch->processAudio(buffer);
red.patch->processAudio(buffer);
break;
case SERIES_RED_MODE:
red.setParameterValues(getAnalogValues());
green.patch->processAudio(buffer);
red.patch->processAudio(buffer);
break;
case PARALLEL_GREEN_MODE:
green.setParameterValues(getAnalogValues());
processParallel(buffer);
break;
case PARALLEL_RED_MODE:
red.setParameterValues(getAnalogValues());
processParallel(buffer);
break;
}
}
void processAudio(AudioBuffer &buffer) {
float* x = buffer.getSamples(0);
float feedback = getParameterValue(PARAMETER_A);
float mix = getParameterValue(PARAMETER_B);
for(int n = 0; n < buffer.getSize(); n++){
x[n] = delayBuffer.tail()*mix + x[n]*(1.0f-mix);
delayBuffer.write(feedback * x[n]);
}
}
FlangerPatch(){
AudioBuffer* buffer = createMemoryBuffer(1, FLANGER_BUFFER_SIZE);
delayBuffer.initialise(buffer->getSamples(0), buffer->getSize());
registerParameter(PARAMETER_A, "Rate");
registerParameter(PARAMETER_B, "Depth");
registerParameter(PARAMETER_C, "Feedback");
registerParameter(PARAMETER_D, "");
phase = 0;
}
DubDelayPatch() {
registerParameter(PARAMETER_A, "Time");
registerParameter(PARAMETER_B, "Feedback");
registerParameter(PARAMETER_C, "Tone");
registerParameter(PARAMETER_D, "Wet");
AudioBuffer* buffer = createMemoryBuffer(1, REQUEST_BUFFER_SIZE);
delayBuffer.initialise(buffer->getSamples(0), REQUEST_BUFFER_SIZE);
}
void processAudio(AudioBuffer& buffer){
float tone = 120*powf(2, getParameterValue(PARAMETER_A)*4);
float decay = getParameterValue(PARAMETER_B);
float accent = getParameterValue(PARAMETER_E);
hat->setFrequency(tone);
hat->setFilter(getParameterValue(PARAMETER_A)*0.3 + 0.5);
hat->setDecay(decay);
hat->setAccent(accent);
tone = 20*powf(2, getParameterValue(PARAMETER_C)*4);
decay = getParameterValue(PARAMETER_D);
kick->setFrequency(tone);
kick->setDecay(decay);
kick->setAccent(accent);
FloatArray left = buffer.getSamples(LEFT_CHANNEL);
FloatArray right = buffer.getSamples(RIGHT_CHANNEL);
drum[0]->getSamples(left);
drum[1]->getSamples(right);
}
LpfDelayPatch() : x1(0.0f), x2(0.0f), y1(0.0f), y2(0.0f), olddelaySamples(0.0f) {
AudioBuffer* buffer = createMemoryBuffer(1, REQUEST_BUFFER_SIZE);
delayBuffer.initialise(buffer->getSamples(0), buffer->getSize());
registerParameter(PARAMETER_A, "Delay", "Delay time");
registerParameter(PARAMETER_B, "Feedback", "Delay loop feedback");
registerParameter(PARAMETER_C, "Fc", "Filter cutoff frequency");
registerParameter(PARAMETER_D, "Dry/Wet", "Dry/wet mix");
setCoeffs(getLpFreq()/getSampleRate(), 0.6f);
}
SimpleDelayPatch() : delay(0), alpha(0.04), dryWet(0.f)
{
registerParameter(PARAMETER_A, "Delay");
registerParameter(PARAMETER_B, "Feedback");
registerParameter(PARAMETER_C, "");
registerParameter(PARAMETER_D, "Dry/Wet");
AudioBuffer* buffer = createMemoryBuffer(1, SIMPLE_DELAY_REQUEST_BUFFER_SIZE);
delayBuffer.initialise(buffer->getSamples(0), buffer->getSize());
}
void processAudio(AudioBuffer &buffer){
prepare();
int size = buffer.getSize();
for (int ch = 0; ch<buffer.getChannels(); ++ch) {
float* buf = buffer.getSamples(ch);
for(int i = 0; i < size; ++i) buf[i] = processSample(buf[i]);
}
}
void processAudio(AudioBuffer &buffer) {
float paramA = getParameterValue(PARAMETER_A);
float paramB = getParameterValue(PARAMETER_B);
float paramC = getParameterValue(PARAMETER_C);
float paramD = getParameterValue(PARAMETER_D);
float paramE = getParameterValue(PARAMETER_E);
// Note: The third parameter is the timestamp at which to execute the message,
// but in this case it simply means to execute it immediately. "f" says that
// the message contains one element and its type is float. paramA is then the
// value.
hv_vscheduleMessageForReceiver(context, "Channel-A", 0.0, "f", paramA);
hv_vscheduleMessageForReceiver(context, "Channel-B", 0.0, "f", paramB);
hv_vscheduleMessageForReceiver(context, "Channel-C", 0.0, "f", paramC);
hv_vscheduleMessageForReceiver(context, "Channel-D", 0.0, "f", paramD);
hv_vscheduleMessageForReceiver(context, "Channel-E", 0.0, "f", paramE);
float* outputs[] = {buffer.getSamples(0), buffer.getSamples(1) };
hv_owl_process(context, outputs, outputs, getBlockSize());
}
void processAudio(AudioBuffer &buffer){
float gain = getParameterValue(PARAMETER_A)*2;
int size = buffer.getSize();
for(int ch=0; ch<buffer.getChannels(); ++ch){
float* buf = buffer.getSamples(ch);
for(int i=0; i<size; ++i)
buf[i] = gain*buf[i];
}
}
void processAudio(AudioBuffer &buffer)
{
int size = buffer.getSize();
float samp_float = getParameterValue(PARAMETER_A);
int samp_freq = ceil(samp_float*63+0.1);
float mayhem_rate = getParameterValue(PARAMETER_B);
mayhem_rate *= 0.03;
float mayhem = 1;
if(abs(getParameterValue(PARAMETER_C)*2+1-prev_freq)>0.01) //if the knob was turned
{
mayhem_freq = getParameterValue(PARAMETER_C); //update center frequency
mayhem_freq *= 2;
mayhem_freq += 1; //mayhem_freq range = 1 to 3 --> 375 -- 1125 Hz
prev_freq = mayhem_freq; //store value to compare next time
}
float mayhem_depth = getParameterValue(PARAMETER_D);
mayhem_depth *= depth;
//for(int ch=0; ch<buffer.getChannels(); ++ch){
float* buf = buffer.getSamples(0);
for(int i=0; i<size; ++i)
{
if(i%samp_freq==0)
{
buf[i] = buf[i]*((1-mayhem)+mayhem*abs(cos(2*M_PI*mayhem_freq*(i+update_freq_cnt*size)/size)));
samp = buf[i];
}
else
buf[i] = samp;
// buf[i] = samp*(1-mayhem)+buf[i]*mayhem*abs(cos(2*M_PI*mayhem_freq*(i+update_freq_cnt*size)/size));
}
// update_freq_cnt++;
// if(update_freq_cnt == 10)
{
update_freq_cnt = 0;
if(mayhem_freq>=prev_freq+mayhem_depth || mayhem_freq>=3) inc_flag = 0; //sets maximum freq 3*fs/size = 1125 Hz
if(mayhem_freq<=prev_freq-mayhem_depth || mayhem_freq<=1)inc_flag = 1; //minimum freq that can be achieved in 128 samples is 375 Hz
if(inc_flag == 0)
{
mayhem_freq /= 1+mayhem_rate*mayhem_depth/depth;
// freq = floor(fs/size*mayhem_freq); //only integer frequencies
}
if(inc_flag == 1)
{
mayhem_freq *= 1+mayhem_rate*mayhem_depth/depth;
// freq = ceil(fs/size*mayhem_freq); //only integer frequencies
}
// mayhem_freq = freq*size/fs; //only integer frequencies
}
}
SimpleDriveDelayPatch() : delay(0)
{
registerParameter(PARAMETER_A, "Delay");
registerParameter(PARAMETER_B, "Feedback");
registerParameter(PARAMETER_C, "Drive");
registerParameter(PARAMETER_D, "Wet/Dry ");
AudioBuffer* buffer = createMemoryBuffer(1, REQUEST_BUFFER_SIZE);
delayBuffer.initialise(buffer->getSamples(0), buffer->getSize());
}
void processAudio(AudioBuffer& _buf){
uint32_t sample_count = _buf.getSize();
float s_rate = getSampleRate();
bits = getParameterValue(PARAMETER_A)*23 - 1;
fs = getParameterValue(PARAMETER_B)*0.999 - 0.001;
input = _buf.getSamples(0);
output = _buf.getSamples(0);
DecimatorPatch* plugin_data = this;
unsigned long pos;
float step, stepr, delta, ratio;
double dummy;
if (bits >= 31.0f || bits < 1.0f) {
step = 0.0f;
stepr = 1.0f;
} else {
step = pow(0.5f, bits - 0.999f);
stepr = 1/step;
}
if (fs >= sample_rate) {
ratio = 1.0f;
} else {
ratio = fs/sample_rate;
}
for (pos = 0; pos < sample_count; pos++) {
count += ratio;
if (count >= 1.0f) {
count -= 1.0f;
delta = modf((input[pos] + (input[pos]<0?-1.0:1.0)*step*0.5) * stepr, &dummy) * step;
last_out = input[pos] - delta;
buffer_write(output[pos], last_out);
} else {
buffer_write(output[pos], last_out);
}
}
plugin_data->last_out = last_out;
plugin_data->count = count;
}
void processAudio(AudioBuffer &buffer){
float rate = getParameterValue(PARAMETER_A);
rate = rate*rate*0.005;
float a = getParameterValue(PARAMETER_B) - 0.5;
if(!triggered && isButtonPressed(PUSHBUTTON)){
x = -6+getParameterValue(PARAMETER_C)*6;
triggered = true;
}else{
triggered = false;
}
int size = buffer.getSize();
FloatArray left = buffer.getSamples(LEFT_CHANNEL);
FloatArray right = buffer.getSamples(RIGHT_CHANNEL);
for(int i=0; i<size; ++i){
left[i] = agnesi(x, a);
right[i] = serpentine(x, a);
x += rate;
}
}
void processAudio(AudioBuffer &buffer) {
float tune = getParameterValue(PARAMETER_A)*2.0 - 1.0;
int octave = round(getParameterValue(PARAMETER_B)*8)-6;
float gain = getParameterValue(PARAMETER_D);
gain = gain*gain*0.8;
FloatArray left = buffer.getSamples(LEFT_CHANNEL);
FloatArray right = buffer.getSamples(RIGHT_CHANNEL);
hz.setTune(tune+octave);
for(int n = 0; n<buffer.getSize(); n++){
float frequency = hz.getFrequency(left[n]);
float amp = hz.sampleToVolts(right[n]);
amp = gain + amp*amp*0.5;
left[n] = amp*wave(pos);
right[n] = left[n];
pos += frequency*mul;
if(pos > 1.0)
pos -= 2.0;
}
}
void processAudio(AudioBuffer &buffer){
data.numSamps = getParameterValue(PARAMETER_A) * (KP_NUM_SAMPLES-8)+8;
data.amp = getParameterValue(PARAMETER_B);
data.g = getParameterValue(PARAMETER_C)*(0.5-0.48)+0.48;
data.duration = getParameterValue(PARAMETER_D) * KP_NUM_BUFFER;
if(isButtonPressed(PUSHBUTTON) && !data.noteOn){
data.noteOn = true;
pressButton(RED_BUTTON);
}
int size = buffer.getSize();
float* left = buffer.getSamples(0);
float* right = buffer.getChannels() > 1 ? buffer.getSamples(1) : left;
for(int i=0; i<size; ++i){
if(data.noteOn){
if(data.phase > (data.numSamps + 1)){
// if we have filled up our delay line, y(n) = g * (y(n-N) + y( n-(N+1) ))
data.pluck[data.phase] = data.g * ( data.pluck[data.phase-data.numSamps]
+ data.pluck[data.phase - (data.numSamps + 1)] );
}else{
// computing the first N samples, y(n) = x(n)
if(data.noiseType == KP_NOISETYPE_GAUSSIAN)
data.pluck[data.phase] = data.noise[data.phase]; // use gaussian white noise
if(data.noiseType == KP_NOISETYPE_RANDOM)
data.pluck[data.phase] = rand()%100000/100000.; // use random noise
}
left[i] = data.amp * data.pluck[data.phase]; // left channel
right[i] = data.amp * data.pluck[data.phase]; // right channel
if(data.phase >= data.duration){
// if we have reached the end of our duration
data.phase = 0;
data.noteOn = false;
pressButton(GREEN_BUTTON);
}else{
data.phase++;
}
}else{
left[i] = right[i] = 0;
}
}
}
BiasedDelayPatch() : MIN_DELAY(0.01), MAX_DELAY(4), MIN_BIAS(0.5), MED_BIAS(1), MAX_BIAS(3), ramp(0.1), circularBuffer(NULL) {
registerParameter(PARAMETER_A, "Delay");
registerParameter(PARAMETER_B, "Feedback");
registerParameter(PARAMETER_C, "Bias");
registerParameter(PARAMETER_D, "Dry/Wet");
memset(oldVal, 0, sizeof(oldVal));
AudioBuffer* buffer = createMemoryBuffer(1, MAX_DELAY * getSampleRate());
bufferSize = buffer->getSize();
circularBuffer = buffer->getSamples(0);
}
void processAudio(AudioBuffer &buffer) {
float frequency = getParameterValue(PARAMETER_A) * 10000;
float amplitude = getParameterValue(PARAMETER_B);
float* left = buffer.getSamples(LEFT_CHANNEL);
float linc = frequency/getSampleRate();
int size = buffer.getSize();
for(int n = 0; n<size; n++){
left[n] = sinf(2*M_PI*pos) * amplitude;
if((pos += linc) > 1.0f)
pos -= 1.0f;
}
}