void pdsp::Sampler::process_audio( const float* pitchModBuffer, const float* triggerBuffer, const float* &selectBuffer, int &selectState, const float* &startBuffer, int& startState, int bufferSize)noexcept{
float* outputBuffer = getOutputBufferToFill(output);
if(pitchModAR){
//
vect_calculateIncrement(outputBuffer, pitchModBuffer, incBase * sample->fileSampleRate, bufferSize);
//in this way is always correct even with oversample
}
for(int n=0; n<bufferSize; ++n){
if(triggerAR){
if(checkTrigger(triggerBuffer[n])){
selectSample(selectBuffer, selectState, startBuffer, startState, n, bufferSize);
}
}
if(pitchModAR){
inc = outputBuffer[n]; //we have the calculated pitchs inside outputbuffer
}
int readIndex_int = static_cast<int>(readIndex);
if(readIndex_int>=0 && readIndex_int < sample->length){
outputBuffer[n] = interpolatorShell.interpolator->interpolate(sample->buffer[sample->mono], readIndex, sample->length);
}else{
outputBuffer[n] = 0.0f;
}
readIndex += inc*direction;
}
positionMeter.store(readIndex*positionDivider);
}
void pdsp::Sampler::process_once( const float* pitchModBuffer)noexcept{
if(pitchModChange){
vect_calculateIncrement(&inc, pitchModBuffer, incBase * sample->fileSampleRate, 1);
//in this way is always correct even with oversample
}
}
float pdsp::PitchToFreq::formula(const float &x) noexcept {
float fc;
vect_calculateIncrement(&fc, &x, freqCalculationMultiplier , 1);
//vect_pitchToFreq(&fc, &x, 1);
return fc;
}
float pdsp::PitchToFreq::eval(float value){
float fc;
vect_calculateIncrement(&fc, &value, freqCalculationMultiplier , 1);
//vect_pitchToFreq(&fc, &x, 1);
return fc;
}
void pdsp::PitchToFreq::formulaAudioRate(float* &output, const float* &input, const int &bufferSize) noexcept {
vect_calculateIncrement(output, input, freqCalculationMultiplier, bufferSize);
//vect_pitchToFreq(output, input, bufferSize);
}
float pdsp::PitchToFreq::eval(float value){
//return powf(2.0f, value) * freqCalculationMultiplier;
float fc;
vect_calculateIncrement(&fc, &value, freqCalculationMultiplier , 1);
return fc;
}