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 BiasedDelay::processBlock(AudioSampleBuffer& buffer, int numInputChannels,
int numOutputChannels, MidiBuffer& midiMessages){
// Atm we're assuming matching input/output channel counts
jassert(numInputChannels==numOutputChannels);
for (int channel=0; channel<numInputChannels; channel++)
{
processChannelBlock(buffer.getNumSamples(),
buffer.getSampleData(channel),
delayBuffer.getSampleData(channel),
delayBufferIdx);
}
delayBufferIdx = (delayBufferIdx + buffer.getNumSamples()) % getSampleDelay(getParameterValue(PARAMETER_TIME));
}
void BiasedDelay::processChannelBlock(int size, float* buf, float* delayBuf, int delayBufIdx){
unsigned int sampleDelay = getSampleDelay(getParameterValue(PARAMETER_TIME));
float feedback = getParameterValue(PARAMETER_FEEDBACK);
float bias = getBiasExponent(1 - getParameterValue(PARAMETER_BIAS));
float dryWetMix = getParameterValue(PARAMETER_DRYWET);
for (int i=0; i<size; i++)
{
float delaySample = delayBuf[delayBufIdx];
float v = buf[i] + delaySample * feedback;
v = applyBias(v, bias);
delayBuf[delayBufIdx] = softLimit(v); // Guard: range limit.
buf[i] = sigmoidXFade(buf[i], delaySample, dryWetMix);
delayBufIdx = (++delayBufIdx) % sampleDelay;
}
}
uint32 ADLPlayer::pollMusic(bool first) {
if (_timbres.empty() || !_songData || !_playPos || (_playPos >= (_songData + _songDataSize))) {
end();
return 0;
}
// We'll ignore the first delay
if (first)
_playPos += (*_playPos & 0x80) ? 2 : 1;
byte cmd = *_playPos++;
// Song end marker
if (cmd == 0xFF) {
end();
return 0;
}
// Set the instrument that should be modified
if (cmd == 0xFE)
_modifyInstrument = *_playPos++;
if (cmd >= 0xD0) {
// Modify an instrument
if (_modifyInstrument == 0xFF)
warning("ADLPlayer: No instrument to modify");
else if (_modifyInstrument >= _timbres.size())
warning("ADLPlayer: Can't modify invalid instrument %d (%d)", _modifyInstrument, _timbres.size());
else
_timbres[_modifyInstrument].params[_playPos[0]] = _playPos[1];
_playPos += 2;
// If we currently have that instrument loaded, reload it
for (int i = 0; i < kMaxVoiceCount; i++)
if (_currentInstruments[i] == _modifyInstrument)
setInstrument(i, _modifyInstrument);
} else {
// Voice command
uint8 voice = cmd & 0x0F;
uint8 note, volume;
switch (cmd & 0xF0) {
case 0x00: // Note on with volume
note = *_playPos++;
volume = *_playPos++;
setVoiceVolume(voice, volume);
noteOn(voice, note);
break;
case 0xA0: // Pitch bend
bendVoicePitch(voice, ((uint16)*_playPos++) << 7);
break;
case 0xB0: // Set volume
setVoiceVolume(voice, *_playPos++);
break;
case 0xC0: // Set instrument
setInstrument(voice, *_playPos++);
break;
case 0x90: // Note on
noteOn(voice, *_playPos++);
break;
case 0x80: // Note off
noteOff(voice);
break;
default:
warning("ADLPlayer: Unsupported command: 0x%02X. Stopping playback.", cmd);
end(true);
return 0;
}
}
uint16 delay = *_playPos++;
if (delay & 0x80)
delay = ((delay & 3) << 8) | *_playPos++;
return getSampleDelay(delay);
}
