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DSP: Fixed multiple issues with the DSP Oversampling class and updated DSP module plug-in demo code accordingly

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This commit is contained in:
hogliux 2017-09-06 10:17:09 +01:00
parent 7a34790388
commit f1d6298206
10 changed files with 352 additions and 239 deletions
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44
examples/DSP module plugin demo/Source/PluginProcessor.cpp
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@ -38,6 +38,9 @@ DspModulePluginDemoAudioProcessor::DspModulePluginDemoAudioProcessor()
waveShapers { {std::tanh}, {dsp::FastMathApproximations::tanh} },
clipping { clip }
{
// Oversampling 2 times with IIR filtering
oversampling = new dsp::Oversampling<float> (2, 1, dsp::Oversampling<float>::filterHalfBandPolyphaseIIR, false);
addParameter (inputVolumeParam = new AudioParameterFloat ("INPUT", "Input Volume", { 0.f, 60.f, 0.f, 1.0f }, 0.f, "dB"));
addParameter (highPassFilterFreqParam = new AudioParameterFloat ("HPFREQ", "Pre Highpass Freq.", { 20.f, 20000.f, 0.f, 0.5f }, 20.f, "Hz"));
addParameter (lowPassFilterFreqParam = new AudioParameterFloat ("LPFREQ", "Post Lowpass Freq.", { 20.f, 20000.f, 0.f, 0.5f }, 20000.f, "Hz"));
@ -50,12 +53,11 @@ DspModulePluginDemoAudioProcessor::DspModulePluginDemoAudioProcessor()
"Cassette recorder cabinet" }, 0));
addParameter (cabinetSimParam = new AudioParameterBool ("CABSIM", "Cabinet Sim", false));
addParameter (oversamplingParam = new AudioParameterBool ("OVERS", "Oversampling", false));
addParameter (outputVolumeParam = new AudioParameterFloat ("OUTPUT", "Output Volume", { -40.f, 40.f, 0.f, 1.0f }, 0.f, "dB"));
cabinetType.set (0);
}
DspModulePluginDemoAudioProcessor::~DspModulePluginDemoAudioProcessor()
@ -82,8 +84,6 @@ void DspModulePluginDemoAudioProcessor::prepareToPlay (double sampleRate, int sa
auto channels = static_cast<uint32> (jmin (getMainBusNumInputChannels(), getMainBusNumOutputChannels()));
dsp::ProcessSpec spec { sampleRate, static_cast<uint32> (samplesPerBlock), channels };
updateParameters();
lowPassFilter.prepare (spec);
highPassFilter.prepare (spec);
@ -92,6 +92,11 @@ void DspModulePluginDemoAudioProcessor::prepareToPlay (double sampleRate, int sa
convolution.prepare (spec);
cabinetType.set (-1);
oversampling->initProcessing (static_cast<size_t> (samplesPerBlock));
updateParameters();
reset();
}
void DspModulePluginDemoAudioProcessor::reset()
@ -99,6 +104,7 @@ void DspModulePluginDemoAudioProcessor::reset()
lowPassFilter.reset();
highPassFilter.reset();
convolution.reset();
oversampling->reset();
}
void DspModulePluginDemoAudioProcessor::releaseResources()
@ -115,20 +121,34 @@ void DspModulePluginDemoAudioProcessor::process (dsp::ProcessContextReplacing<fl
// Note : try frequencies around 700 Hz
highPassFilter.process (context);
// Upsampling
dsp::AudioBlock<float> oversampledBlock;
setLatencySamples (audioCurrentlyOversampled ? roundFloatToInt (oversampling->getLatencyInSamples()) : 0);
if (audioCurrentlyOversampled)
oversampledBlock = oversampling->processSamplesUp (context.getInputBlock());
dsp::ProcessContextReplacing<float> waveshaperContext = audioCurrentlyOversampled ? dsp::ProcessContextReplacing<float> (oversampledBlock) : context;
// Waveshaper processing, for distortion generation, thanks to the input gain
// The fast tanh can be used instead of std::tanh to reduce the CPU load
auto waveshaperIndex = waveshaperParam->getIndex();
if (isPositiveAndBelow (waveshaperIndex, (int) numWaveShapers) )
{
waveShapers[waveshaperIndex].process (context);
waveShapers[waveshaperIndex].process (waveshaperContext);
if (waveshaperIndex == 1)
clipping.process(context);
clipping.process (waveshaperContext);
context.getOutputBlock() *= 0.7f;
waveshaperContext.getOutputBlock() *= 0.7f;
}
// Downsampling
if (audioCurrentlyOversampled)
oversampling->processSamplesDown (context.getOutputBlock());
// Post-lowpass filtering
lowPassFilter.process (context);
@ -209,13 +229,20 @@ bool DspModulePluginDemoAudioProcessor::producesMidi() const
//==============================================================================
void DspModulePluginDemoAudioProcessor::updateParameters()
{
auto newOversampling = oversamplingParam->get();
if (newOversampling != audioCurrentlyOversampled)
{
audioCurrentlyOversampled = newOversampling;
oversampling->reset();
}
//==============================================================================
auto inputdB = Decibels::decibelsToGain (inputVolumeParam->get());
auto outputdB = Decibels::decibelsToGain (outputVolumeParam->get());
if (inputVolume.getGainLinear() != inputdB) inputVolume.setGainLinear (inputdB);
if (outputVolume.getGainLinear() != outputdB) outputVolume.setGainLinear (outputdB);
dsp::IIR::Coefficients<float>::Ptr newHighPassCoeffs, newLowPassCoeffs;
auto newSlopeType = slopeParam->getIndex();
if (newSlopeType == 0)
@ -246,6 +273,7 @@ void DspModulePluginDemoAudioProcessor::updateParameters()
}
cabinetIsBypassed = ! cabinetSimParam->get();
}
//==============================================================================