void HRIR::update (double azimuth, double elevation)
{
int nAzimuth = (int) azimuth;
int nElevation = (int) elevation;
int taps = mit_hrtf_availability (nAzimuth, nElevation, sampleRate, diffused);
if (taps)
{
leftEar.setSize (1, taps);
rightEar.setSize (1, taps);
AudioBuffer<short> leftShort (1, taps);
AudioBuffer<short> rightShort (1, taps);
short* const writeLeft = leftShort.getWritePointer (0);
short* const writeRight = rightShort.getWritePointer (0);
taps = mit_hrtf_get (&nAzimuth, &nElevation, sampleRate, diffused, writeLeft, writeRight);
float* const leftEarWrite = leftEar.getWritePointer (0);
float* const rightEarWrite = rightEar.getWritePointer (0);
const short* const readLeft = leftShort.getReadPointer (0);
const short* const readRight = rightShort.getReadPointer (0);
for (int n = 0; n < taps; ++n)
{
leftEarWrite[n] = readLeft[n] / 32768.0f;
rightEarWrite[n] = readRight[n] / 32768.0f;
}
}
}
void VAOscillator::fillBufferSquarePulse(AudioBuffer<float>& buffer, AudioBuffer<float>& phaseModBuffer, AudioBuffer<float>& volumeModBuffer, AudioBuffer<float>& pitchModBuffer)
{
if(isActive)
{
float* const data = buffer.getWritePointer(0);
float const *phaseMod = phaseModBuffer.getReadPointer(0);
float const *volMod = volumeModBuffer.getReadPointer(0);
float const *pitchMod = pitchModBuffer.getReadPointer(0);
//write momentary phase values into the buffer
for(int sampleIndex = 0; sampleIndex < buffer.getNumSamples(); sampleIndex++)
{
double phase = currentPhase + phaseModAmp * phaseMod[sampleIndex];
while (phase < 0)
{
phase += 2 * double_Pi;
}
while (phase > 2 * double_Pi)
{
phase -= 2 * double_Pi;
}
currentPhase += 2 * double_Pi * ((currentFrequency * (pitchMod[sampleIndex] + 2.0))/currentSampleRate);
if(currentPhase > 2 * double_Pi)
{
currentPhase -= 2 * double_Pi;
}
if(phase < double_Pi)
{
data[sampleIndex] = -1;
}
else
{
data[sampleIndex] = 1;
}
data[sampleIndex] += getBlep(phase, currentFrequency);
if(phase < double_Pi)
{
data[sampleIndex] -= getBlep(phase + double_Pi, currentFrequency);
}
if(phase > double_Pi)
{
data[sampleIndex] -= getBlep(phase - double_Pi, currentFrequency);
}
data[sampleIndex] *= (float)std::abs(0.5 * (volMod[sampleIndex] + 1));
}
}
else
{
buffer.clear();
}
}// end square pulse
void VAOscillator::fillBufferTriangle(AudioBuffer<float>& buffer, AudioBuffer<float>& phaseModBuffer, AudioBuffer<float>& volumeModBuffer, AudioBuffer<float>& pitchModBuffer)
{
if(isActive)
{
float* const data = buffer.getWritePointer(0);
float const *phaseMod = phaseModBuffer.getReadPointer(0);
float const *volMod = volumeModBuffer.getReadPointer(0);
float const *pitchMod = pitchModBuffer.getReadPointer(0);
for(int sampleIndex = 0; sampleIndex < buffer.getNumSamples(); sampleIndex++)
{
double phase = currentPhase + phaseModAmp * phaseMod[sampleIndex];
while (phase < 0)
{
phase += 2 * double_Pi;
}
while (phase > 2 * double_Pi)
{
phase -= 2 * double_Pi;
}
currentPhase += 2 * double_Pi * ((currentFrequency * (pitchMod[sampleIndex] + 2.0))/currentSampleRate);
if(currentPhase > 2 * double_Pi)
{
currentPhase -= 2 * double_Pi;
}
if(phase < double_Pi)
{
data[sampleIndex] = (float)((2 * phase)/double_Pi - 1);
}
else
{
data[sampleIndex] = (float)((3 - 2 * phase/double_Pi));
}
//the 0.000026 is kind of a magic number i didnt calculate it just found it by trying out
data[sampleIndex] += (float)(0.000026 * currentFrequency * getTriRes(phase, currentFrequency));
if(phase < double_Pi)
{
data[sampleIndex] -= (float)(0.000026 * currentFrequency * getTriRes(phase + double_Pi, currentFrequency));
}
else if(phase >= double_Pi)
{
data[sampleIndex] -= (float)(0.000026 * currentFrequency * getTriRes(phase - double_Pi, currentFrequency));
}
data[sampleIndex] *= (float)std::abs(0.5 * (volMod[sampleIndex] + 1));
}
}
else
{
buffer.clear();
}
}// end triangle
void VAOscillator::fillBufferSine(AudioBuffer<float>& buffer, AudioBuffer<float>& phaseModBuffer, AudioBuffer<float>& volumeModBuffer, AudioBuffer<float>& pitchModBuffer, Array<int>& midiOns)// add midi buffer and know channel with control Voltage
{
if(isActive)
{
float* const data = buffer.getWritePointer(0);
float const *phaseMod = phaseModBuffer.getReadPointer(0);
float const *volMod = volumeModBuffer.getReadPointer(0);
float const *pitchMod = pitchModBuffer.getReadPointer(0);
for(int sampleIndex = 0; sampleIndex < buffer.getNumSamples(); sampleIndex++)
{
//check if this is the best i can do
if(midiOns.size() != 0)
{
if(sampleIndex == midiOns[0])
{
resetPhase();
midiOns.remove(0);
}
}
double phase = currentPhase + phaseModAmp * phaseMod[sampleIndex] + currentPhaseOffset;
while (phase < 0)
{
phase += 2 * double_Pi;
}
while (phase > 2 * double_Pi)
{
phase -= 2 * double_Pi;
}
data[sampleIndex] = static_cast<float>(sin(phase) * std::abs(0.5 * (volMod[sampleIndex] + 1)));
currentPhase += 2 * double_Pi * ((currentFrequency * (pitchMod[sampleIndex] + 2.0))/currentSampleRate);
if(currentPhase > 2 * double_Pi)
{
currentPhase -= 2 * double_Pi;
}
}
}
else
{
buffer.clear();
}
}// end sine
void DaalDelAudioProcessor::processBlock (AudioBuffer<float>& buffer, MidiBuffer& midiMessages)
{
ScopedNoDenormals noDenormals;
auto totalNumInputChannels = getTotalNumInputChannels();
auto totalNumOutputChannels = getTotalNumOutputChannels();
// In case we have more outputs than inputs, this code clears any output
// channels that didn't contain input data, (because these aren't
// guaranteed to be empty - they may contain garbage).
// This is here to avoid people getting screaming feedback
// when they first compile a plugin, but obviously you don't need to keep
// this code if your algorithm always overwrites all the output channels.
for (auto i = totalNumInputChannels; i < totalNumOutputChannels; ++i)
buffer.clear (i, 0, buffer.getNumSamples());
// ====
// Lengths for circular buffer
const int bufferLength = buffer.getNumSamples();
const int delayBufferLength = _delayBuffer.getNumSamples();
// This is the place where you'd normally do the guts of your plugin's
// audio processing...
// Make sure to reset the state if your inner loop is processing
// the samples and the outer loop is handling the channels.
// Alternatively, you can process the samples with the channels
// interleaved by keeping the same state.
for (int channel = 0; channel < totalNumInputChannels; ++channel)
{
//auto* channelData = buffer.getWritePointer (channel);
// ..do something to the data...
// Set up circular buffer
const float* bufferData = buffer.getReadPointer(channel);
const float* delayBufferData = _delayBuffer.getReadPointer(channel);
float* dryBuffer = buffer.getWritePointer(channel);
// Apply gains (now do this before getting from delay)
applyDryWetToBuffer(buffer, channel, bufferLength, dryBuffer);
// Copy data from main to delay buffer
fillDelayBuffer(channel, bufferLength, delayBufferLength, bufferData, delayBufferData);
// Copy data from delay buffer to output buffer
getFromDelayBuffer(buffer, channel, bufferLength, delayBufferLength, bufferData, delayBufferData);
// Feedback
feedbackDelay(channel, bufferLength, delayBufferLength, dryBuffer);
}
_writePosition += bufferLength; // Increment
_writePosition %= delayBufferLength; // Wrap around position index
// Update values from tree
updateTreeParams();
}
void VAOscillator::fillBufferRisingSaw(AudioBuffer<float>& buffer, AudioBuffer<float>& phaseModBuffer, AudioBuffer<float>& volumeModBuffer, AudioBuffer<float>& pitchModBuffer)
{
if(isActive)
{
float* const data = buffer.getWritePointer(0);
float const *phaseMod = phaseModBuffer.getReadPointer(0);
float const *volMod = volumeModBuffer.getReadPointer(0);
float const *pitchMod = pitchModBuffer.getReadPointer(0);
//write momentary phase values into the buffer
for(int sampleIndex = 0; sampleIndex < buffer.getNumSamples(); sampleIndex++)
{
double phase = currentPhase + phaseModAmp * phaseMod[sampleIndex];
while (phase < 0)
{
phase += 2 * double_Pi;
}
while (phase > 2 * double_Pi)
{
phase -= 2 * double_Pi;
}
currentPhase += 2 * double_Pi * ((currentFrequency * (pitchMod[sampleIndex] + 2.0))/currentSampleRate);
if(currentPhase > 2 * double_Pi)
{
currentPhase -= 2 * double_Pi;
}
data[sampleIndex] = (float)((2 * phase)/(2 * double_Pi) - 1);
data[sampleIndex] += getBlep(phase , currentFrequency);// i have to watch out here because actually currentFrequency is not valid for this calculation
data[sampleIndex] *= std::abs(0.5f * (volMod[sampleIndex] + 1));
}
}
else
{
buffer.clear();
}
}// end rising Saw
bool AudioFormatWriter::writeFromAudioSampleBuffer (const AudioBuffer<float>& source, int startSample, int numSamples)
{
auto numSourceChannels = source.getNumChannels();
jassert (startSample >= 0 && startSample + numSamples <= source.getNumSamples() && numSourceChannels > 0);
if (startSample == 0)
return writeFromFloatArrays (source.getArrayOfReadPointers(), numSourceChannels, numSamples);
const float* chans[256];
jassert ((int) numChannels < numElementsInArray (chans));
for (int i = 0; i < numSourceChannels; ++i)
chans[i] = source.getReadPointer (i, startSample);
chans[numSourceChannels] = nullptr;
return writeFromFloatArrays (chans, numSourceChannels, numSamples);
}
Result Upsampler::upsample(AudioBuffer<float> const &inputBuffer, int const inputChannel, AudioBuffer<float> &outputBuffer_, int const outputChannel, int &outputSampleCount_)
{
int outputSamplesNeeded = roundDoubleToInt( (outputSampleRate * inputBuffer.getNumSamples())/ inputSampleRate);
if (outputBuffer_.getNumSamples() < outputSamplesNeeded)
return Result::fail("Output buffer too small");
//DBG("upsample inputSampleCount:" << inputSampleCount);
if (nullptr == resampler)
return Result::fail("No resampler object");
//
// Clear the resample and pump some initial zeros into it
//
resampler->clear();
#if 0
int preloadSamples = resampler->getInLenBeforeOutStart(MAX_RESAMPLER_INPUT_SAMPLES);
inputBlockBuffer.clear(MAX_RESAMPLER_INPUT_SAMPLES);
while (preloadSamples > 0)
{
int count = jmin((int)MAX_RESAMPLER_INPUT_SAMPLES, preloadSamples);
double* outputBlock = nullptr;
resampler->process(inputBlockBuffer, count, outputBlock);
preloadSamples -= count;
}
#endif
//
// Flush the output buffer
//
outputBuffer_.clear();
//
// Do the actual upsample
//
outputSampleCount_ = 0;
int inputSampleCount = inputBuffer.getNumSamples();
const float * source = inputBuffer.getReadPointer(inputChannel);
while (inputSampleCount > 0)
{
//
// Convert float to double
//
int inputConvertCount = jmin(inputSampleCount, (int)MAX_RESAMPLER_INPUT_SAMPLES);
for (int i = 0; i < inputConvertCount; ++i)
{
inputBlockBuffer[i] = *source;
source++;
}
inputSampleCount -= inputConvertCount;
//
// Run the SRC
//
double* outputBlock = nullptr;
int outputBlockSampleCount = resampler->process(inputBlockBuffer, inputConvertCount, outputBlock);
int outputSpaceRemaining = outputBuffer_.getNumSamples() - outputSampleCount_;
int outputCopyCount = jmin( outputSpaceRemaining, outputBlockSampleCount);
float *destination = outputBuffer_.getWritePointer(outputChannel, outputSampleCount_);
for (int i = 0; i < outputCopyCount; ++i)
{
*destination = (float)outputBlock[i];
destination++;
}
outputSampleCount_ += outputCopyCount;
}
//
// Keep filling the output buffer
//
inputBlockBuffer.clear(MAX_RESAMPLER_INPUT_SAMPLES);
while (outputSampleCount_ < outputBuffer_.getNumSamples())
{
//
// Run the SRC
//
double* outputBlock = nullptr;
int outputBlockSampleCount = resampler->process(inputBlockBuffer, MAX_RESAMPLER_INPUT_SAMPLES, outputBlock);
int outputSpaceRemaining = outputBuffer_.getNumSamples() - outputSampleCount_;
int outputCopyCount = jmin( outputSpaceRemaining, outputBlockSampleCount);
float *destination = outputBuffer_.getWritePointer(outputChannel, outputSampleCount_);
for (int i = 0; i < outputCopyCount; ++i)
{
*destination = (float)outputBlock[i];
destination++;
}
outputSampleCount_ += outputCopyCount;
}
//DBG(" outputSampleCount:" << outputSampleCount);
return Result::ok();
}