void DRowAudioFilter::processBlock (AudioSampleBuffer& buffer,
MidiBuffer& midiMessages)
{
smoothParameters();
const int numInputChannels = getNumInputChannels();
int numSamples = buffer.getNumSamples();
// set up the parameters to be used
float preDelay = (float)params[PREDELAY].getSmoothedValue();
float earlyDecay = (float)params[EARLYDECAY].getSmoothedNormalisedValue();
earlyDecay = sqrt(sqrt(earlyDecay));
float late = (float)params[EARLYLATEMIX].getSmoothedNormalisedValue();
float early = 1.0f - late;
float fbCoeff = (float)params[FBCOEFF].getSmoothedNormalisedValue();
fbCoeff = -sqrt(sqrt(fbCoeff));
float delayTime = (float)params[DELTIME].getSmoothedValue();
float filterCf = (float)params[FILTERCF].getSmoothedValue();
float allpassCoeff = (float)params[DIFFUSION].getSmoothedNormalisedValue();
float spread1 = (float)params[SPREAD].getSmoothedNormalisedValue();
float spread2 = 1.0f - spread1;
float lowEQGain = (float)decibelsToAbsolute(params[LOWEQ].getSmoothedValue());
float highEQGain = (float)decibelsToAbsolute(params[HIGHEQ].getSmoothedValue());
float wet = (float)params[WETDRYMIX].getSmoothedNormalisedValue();
float dry = 1.0f - wet;
float width = (spread1-0.5f) * 0.1f * delayTime;
// we can only deal with 2-in, 2-out at the moment
if (numInputChannels == 2)
{
// pre-delay section
preDelayFilterL.setDelayTime(currentSampleRate, preDelay);
preDelayFilterR.setDelayTime(currentSampleRate, preDelay);
// early reflections section
int roomShape = roundFloatToInt(params[ROOMSHAPE].getValue());
float delayCoeff = 0.08f * delayTime;
if (roomShape != prevRoomShape)
{
delayLineL.removeAllTaps();
delayLineR.removeAllTaps();
for(int i = 0; i < 5; i++) {
delayLineL.addTapAtTime(earlyReflectionCoeffs[roomShape-3][i], currentSampleRate);
delayLineR.addTapAtTime(earlyReflectionCoeffs[roomShape-3][i], currentSampleRate);
}
// we have to set this here incase the delay time has not changed
delayLineL.setTapSpacingExplicitly(delayCoeff);
delayLineR.setTapSpacingExplicitly(delayCoeff + spread1);
prevRoomShape = roomShape;
}
delayLineL.setTapSpacing(delayCoeff);
delayLineL.scaleFeedbacks(earlyDecay);
delayLineR.setTapSpacing(delayCoeff + spread1);
delayLineR.scaleFeedbacks(earlyDecay);
// comb filter section
for (int i = 0; i < 8; ++i)
{
delayTime *= filterMultCoeffs[i];
delayTime += Random::getSystemRandom().nextInt(100)*0.0001;
setupFilter(combFilterL[i], fbCoeff, delayTime, filterCf);
setupFilter(combFilterR[i], fbCoeff, delayTime + width, filterCf);
}
// allpass section
for (int i = 0; i < 4; ++i)
{
delayTime *= allpassMultCoeffs[i];
delayTime -= Random::getSystemRandom().nextInt(100)*0.0001;
allpassFilterL[i].setGain(allpassCoeff);
allpassFilterL[i].setDelayTime(currentSampleRate, delayTime);
allpassFilterR[i].setGain(allpassCoeff);
allpassFilterR[i].setDelayTime(currentSampleRate, delayTime + width);
}
// final EQ section
lowEQL.makeLowShelf(currentSampleRate, 500, 1, lowEQGain);
lowEQR.makeLowShelf(currentSampleRate, 500, 1, lowEQGain);
highEQL.makeHighShelf(currentSampleRate, 3000, 1, highEQGain);
highEQR.makeHighShelf(currentSampleRate, 3000, 1, highEQGain);
//========================================================================
// Processing
//========================================================================
int noSamples = buffer.getNumSamples();
int noChannels = buffer.getNumChannels();
// create a copy of the input buffer so we can apply a wet/dry mix later
AudioSampleBuffer wetBuffer(noChannels, noSamples);
wetBuffer.copyFrom(0, 0, buffer, 0, 0, noSamples);
wetBuffer.copyFrom(1, 0, buffer, 1, 0, noSamples);
// mono mix wet buffer (used for stereo spread later)
float *pfWetL = wetBuffer.getSampleData(0);
float *pfWetR = wetBuffer.getSampleData(1);
while (--numSamples >= 0)
{
*pfWetL = *pfWetR = (0.5f * (*pfWetL + *pfWetR));
pfWetL++;
pfWetR++;
}
numSamples = buffer.getNumSamples();
// apply the pre-delay to the wet buffer
preDelayFilterL.processSamples(wetBuffer.getSampleData(0), noSamples);
preDelayFilterR.processSamples(wetBuffer.getSampleData(1), noSamples);
// create a buffer to hold the early reflections
AudioSampleBuffer earlyReflections(noChannels, noSamples);
earlyReflections.copyFrom(0, 0, wetBuffer, 0, 0, noSamples);
earlyReflections.copyFrom(1, 0, wetBuffer, 1, 0, noSamples);
// and process the early reflections
delayLineL.processSamples(earlyReflections.getSampleData(0), noSamples);
delayLineR.processSamples(earlyReflections.getSampleData(1), noSamples);
// create a buffer to hold the late reverb
AudioSampleBuffer lateReverb(noChannels, noSamples);
lateReverb.clear();
float *pfLateL = lateReverb.getSampleData(0);
float *pfLateR = lateReverb.getSampleData(1);
pfWetL = wetBuffer.getSampleData(0);
pfWetR = wetBuffer.getSampleData(1);
// comb filter section
for (int i = 0; i < 8; ++i)
{
combFilterL[i].processSamplesAdding(pfWetL, pfLateL, noSamples);
combFilterR[i].processSamplesAdding(pfWetR, pfLateR, noSamples);
}
// allpass filter section
for (int i = 0; i < 4; ++i)
{
allpassFilterL[i].processSamples(lateReverb.getSampleData(0), noSamples);
allpassFilterR[i].processSamples(lateReverb.getSampleData(1), noSamples);
}
// clear wet buffer
wetBuffer.clear();
// add early reflections to wet buffer
wetBuffer.addFrom(0, 0, earlyReflections, 0, 0, noSamples, early);
wetBuffer.addFrom(1, 0, earlyReflections, 1, 0, noSamples, early);
// add late reverb to wet buffer
lateReverb.applyGain(0, noSamples, 0.1f);
wetBuffer.addFrom(0, 0, lateReverb, 0, 0, noSamples, late);
wetBuffer.addFrom(1, 0, lateReverb, 1, 0, noSamples, late);
// final EQ
lowEQL.processSamples(pfWetL, noSamples);
lowEQR.processSamples(pfWetR, noSamples);
highEQL.processSamples(pfWetL, noSamples);
highEQR.processSamples(pfWetR, noSamples);
// create stereo spread
while (--numSamples >= 0)
{
float fLeft = *pfWetL;
float fRight = *pfWetR;
*pfWetL = (fLeft * spread1) + (fRight * spread2);
*pfWetR = (fRight * spread1) + (fLeft * spread2);
pfWetL++;
pfWetR++;
}
numSamples = buffer.getNumSamples();
// apply wet/dry mix gains
wetBuffer.applyGain(0, noSamples, wet);
buffer.applyGain(0, noSamples, dry);
// add wet buffer to output buffer
buffer.addFrom(0, 0, wetBuffer, 0, 0, noSamples);
buffer.addFrom(1, 0, wetBuffer, 1, 0, noSamples);
}
//========================================================================
// in case we have more outputs than inputs, we'll clear any output
// channels that didn't contain input data, (because these aren't
// guaranteed to be empty - they may contain garbage).
for (int i = getNumInputChannels(); i < getNumOutputChannels(); ++i)
{
buffer.clear (i, 0, buffer.getNumSamples());
}
}
void DRowAudioFilter::processBlock (AudioSampleBuffer& buffer,
MidiBuffer& midiMessages)
{
smoothParameters();
const int numInputChannels = getNumInputChannels();
int numSamples = buffer.getNumSamples();
// set up the parameters to be used
float inGain = decibelsToAbsolute(params[INGAIN].getSmoothedValue());
float outGain = decibelsToAbsolute(params[OUTGAIN].getSmoothedValue());
buffer.applyGain(0, buffer.getNumSamples(), inGain);
if (numInputChannels == 2)
{
// get sample pointers
float* channelL = buffer.getSampleData(0);
float* channelR = buffer.getSampleData(1);
// pre-filter
inFilterL->processSamples(buffer.getSampleData(0), numSamples);
inFilterR->processSamples(buffer.getSampleData(1), numSamples);
while (--numSamples >= 0)
{
float sampleL = *channelL;
float sampleR = *channelR;
// clip samples
sampleL = jlimit(-1.0f, 1.0f, sampleL);
sampleR = jlimit(-1.0f, 1.0f, sampleR);
if (sampleL < 0.0f) {
sampleL *= -1.0f;
sampleL = linearInterpolate(distortionBuffer, distortionBufferSize, sampleL*distortionBufferMax);
sampleL *= -1.0f;
}
else {
sampleL = linearInterpolate(distortionBuffer, distortionBufferSize, sampleL*distortionBufferMax);
}
if (sampleR < 0.0f) {
sampleR *= -1.0f;
sampleR = linearInterpolate(distortionBuffer, distortionBufferSize, sampleR*distortionBufferMax);
sampleR *= -1.0f;
}
else {
sampleR = linearInterpolate(distortionBuffer, distortionBufferSize, sampleR*distortionBufferMax);
}
*channelL++ = sampleL;
*channelR++ = sampleR;
}
// post-filter
outFilterL->processSamples(buffer.getSampleData(0), buffer.getNumSamples());
outFilterR->processSamples(buffer.getSampleData(1), buffer.getNumSamples());
buffer.applyGain(0, buffer.getNumSamples(), outGain);
}
else if (numInputChannels == 1)
{
// get sample pointers
float* channelL = buffer.getSampleData(0);
// pre-filter
inFilterL->processSamples(buffer.getSampleData(0), numSamples);
while (--numSamples >= 0)
{
float sampleL = *channelL;
// clip samples
sampleL = jlimit(-1.0f, 1.0f, sampleL);
if (sampleL < 0.0f) {
sampleL *= -1.0f;
sampleL = linearInterpolate(distortionBuffer, distortionBufferSize, sampleL*distortionBufferMax);
sampleL *= -1.0f;
}
else {
sampleL = linearInterpolate(distortionBuffer, distortionBufferSize, sampleL*distortionBufferMax);
}
*channelL++ = sampleL;
}
// post-filter
outFilterL->processSamples(buffer.getSampleData(0), buffer.getNumSamples());
buffer.applyGain(0, buffer.getNumSamples(), outGain);
}
//========================================================================
// in case we have more outputs than inputs, we'll clear any output
// channels that didn't contain input data, (because these aren't
// guaranteed to be empty - they may contain garbage).
for (int i = getNumInputChannels(); i < getNumOutputChannels(); ++i)
{
buffer.clear (i, 0, buffer.getNumSamples());
}
}
