void Source::getNextAudioBlock (const AudioSourceChannelInfo& bufferToFill)
{
// calculations
float trackCutoff = parameters->getUnchecked(Controller::getParameterId(Controller::params::cutoff, trackId))->getScaledValue();
float trackDistort = parameters->getUnchecked(Controller::getParameterId(Controller::params::distort, trackId))->getScaledValue();
float trackLevel = parameters->getUnchecked(Controller::getParameterId(Controller::params::level, trackId))->getScaledValue();
float trackMute = parameters->getUnchecked(Controller::getParameterId(Controller::params::mute, trackId))->getScaledValue();
float globalCutoff = parameters->getUnchecked(Controller::getParameterId(Controller::params::cutoff))->getScaledValue();
float globalDistort = parameters->getUnchecked(Controller::getParameterId(Controller::params::distort))->getScaledValue();
float modulationCutoff = sampler.currentModulations != nullptr
? Parameter::scale(Controller::params::cutoff, true, sampler.currentModulations->getUnchecked(Mixer::mods::cutoff))
: 0.0;
float cutoff = fmax(0, fmin(1, trackCutoff + globalCutoff + modulationCutoff));
float distort = 1 - fmax(0, fmin(0.93, trackDistort + globalDistort));
float level = trackLevel * !trackMute;
// setup
bufferToFill.clearActiveBufferRegion();
midiCollector.removeNextBlockOfMessages (incomingMidi, bufferToFill.numSamples);
// render sampler
sampler.renderNextBlock(*bufferToFill.buffer, incomingMidi, 0, bufferToFill.numSamples);
// dsp: distortion
float* outL = bufferToFill.buffer->getWritePointer (0, 0);
float* outR = bufferToFill.buffer->getWritePointer (1, 0);
for (int i=bufferToFill.numSamples; i>=0; --i) {
outL[i] = foldback(outL[i], distort);
outR[i] = foldback(outR[i], distort);
}
// dsp: filter
filterL.setCoefficients(IIRCoefficients::makeLowPass(sampleRate, MidiMessage::getMidiNoteInHertz(cutoff * 128)));
filterR.setCoefficients(IIRCoefficients::makeLowPass(sampleRate, MidiMessage::getMidiNoteInHertz(cutoff * 128)));
filterL.processSamples(bufferToFill.buffer->getWritePointer(0), bufferToFill.buffer->getNumSamples());
filterR.processSamples(bufferToFill.buffer->getWritePointer(1), bufferToFill.buffer->getNumSamples());
// dsp: level
bufferToFill.buffer->applyGainRamp(0, bufferToFill.numSamples, lastLevel, level);
lastLevel = level;
}
TEST test_manual_foldback() {
// Lower foldback: wrap tonewheels < 13 back into the 13..92 range.
for (int i = 1; i < 13; i++) {
ASSERT_EQ_FMT(i + 12, foldback(i), "%d");
}
// No foldback applied up through tonewheel 92.
for (int i = 13; i < 92; i++) {
ASSERT_EQ_FMT(i, foldback(i), "%d");
}
// Upper foldback: the first octave above 91 is mapped down one octave.
for (int i = 92; i < 104; i++) {
ASSERT_EQ_FMT(i - 12, foldback(i), "%d");
}
// Upper foldback: the next octave is mapped down two octaves.
for (int i = 104; i < 116; i++) {
ASSERT_EQ_FMT(i - 24, foldback(i), "%d");
}
PASS();
}
