718 lines
28 KiB
C++
718 lines
28 KiB
C++
/*
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==============================================================================
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This file is part of the JUCE library.
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Copyright (c) 2017 - ROLI Ltd.
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JUCE is an open source library subject to commercial or open-source
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licensing.
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By using JUCE, you agree to the terms of both the JUCE 5 End-User License
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Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
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27th April 2017).
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End User License Agreement: www.juce.com/juce-5-licence
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Privacy Policy: www.juce.com/juce-5-privacy-policy
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Or: You may also use this code under the terms of the GPL v3 (see
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www.gnu.org/licenses).
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JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
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EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
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DISCLAIMED.
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==============================================================================
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*/
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namespace juce
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{
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namespace dsp
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{
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/** Abstract class for the provided oversampling engines used internally in
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the Oversampling class.
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*/
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template <typename SampleType>
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class OversamplingEngine
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{
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public:
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//===============================================================================
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OversamplingEngine (size_t newNumChannels, size_t newFactor)
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{
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numChannels = newNumChannels;
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factor = newFactor;
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}
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virtual ~OversamplingEngine() {}
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//===============================================================================
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virtual SampleType getLatencyInSamples() = 0;
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size_t getFactor() { return factor; }
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virtual void initProcessing (size_t maximumNumberOfSamplesBeforeOversampling)
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{
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buffer.setSize (static_cast<int> (numChannels), static_cast<int> (maximumNumberOfSamplesBeforeOversampling * factor), false, false, true);
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}
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virtual void reset()
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{
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buffer.clear();
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}
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dsp::AudioBlock<SampleType> getProcessedSamples (size_t numSamples)
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{
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return dsp::AudioBlock<SampleType> (buffer).getSubBlock (0, numSamples);
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}
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virtual void processSamplesUp (dsp::AudioBlock<SampleType> &inputBlock) = 0;
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virtual void processSamplesDown (dsp::AudioBlock<SampleType> &outputBlock) = 0;
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protected:
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//===============================================================================
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AudioBuffer<SampleType> buffer;
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size_t factor;
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size_t numChannels;
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};
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//===============================================================================
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/** Dummy oversampling engine class which simply copies and pastes the input
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signal, which could be equivalent to a "one time" oversampling processing.
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*/
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template <typename SampleType>
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class OversamplingDummy : public OversamplingEngine<SampleType>
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{
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public:
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//===============================================================================
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OversamplingDummy (size_t numChannels) : OversamplingEngine<SampleType> (numChannels, 1) {}
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~OversamplingDummy() {}
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//===============================================================================
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SampleType getLatencyInSamples() override
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{
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return 0.f;
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}
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void processSamplesUp (dsp::AudioBlock<SampleType> &inputBlock) override
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{
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jassert (inputBlock.getNumChannels() <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumChannels()));
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jassert (inputBlock.getNumSamples() * OversamplingEngine<SampleType>::factor <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumSamples()));
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for (size_t channel = 0; channel < inputBlock.getNumChannels(); channel++)
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OversamplingEngine<SampleType>::buffer.copyFrom (static_cast<int> (channel), 0,
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inputBlock.getChannelPointer (channel), static_cast<int> (inputBlock.getNumSamples()));
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}
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void processSamplesDown (dsp::AudioBlock<SampleType> &outputBlock) override
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{
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jassert (outputBlock.getNumChannels() <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumChannels()));
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jassert (outputBlock.getNumSamples() * OversamplingEngine<SampleType>::factor <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumSamples()));
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outputBlock.copy (OversamplingEngine<SampleType>::getProcessedSamples (outputBlock.getNumSamples()));
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}
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private:
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//===============================================================================
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JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (OversamplingDummy)
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};
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//===============================================================================
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/** Oversampling engine class performing 2 times oversampling using the Filter
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Design FIR Equiripple method. The resulting filter is linear phase,
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symmetric, and has every two samples but the middle one equal to zero,
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leading to specific processing optimizations.
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*/
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template <typename SampleType>
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class Oversampling2TimesEquirippleFIR : public OversamplingEngine<SampleType>
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{
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public:
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//===============================================================================
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Oversampling2TimesEquirippleFIR (size_t numChannels,
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SampleType normalizedTransitionWidthUp,
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SampleType stopbandAttenuationdBUp,
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SampleType normalizedTransitionWidthDown,
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SampleType stopbandAttenuationdBDown) : OversamplingEngine<SampleType> (numChannels, 2)
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{
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coefficientsUp = *dsp::FilterDesign<SampleType>::designFIRLowpassHalfBandEquirippleMethod (normalizedTransitionWidthUp, stopbandAttenuationdBUp);
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coefficientsDown = *dsp::FilterDesign<SampleType>::designFIRLowpassHalfBandEquirippleMethod (normalizedTransitionWidthDown, stopbandAttenuationdBDown);
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auto N = coefficientsUp.getFilterOrder() + 1;
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stateUp.setSize (static_cast<int> (numChannels), static_cast<int> (N));
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N = coefficientsDown.getFilterOrder() + 1;
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auto Ndiv2 = N / 2;
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auto Ndiv4 = Ndiv2 / 2;
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stateDown.setSize (static_cast<int> (numChannels), static_cast<int> (N));
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stateDown2.setSize (static_cast<int> (numChannels), static_cast<int> (Ndiv4 + 1));
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position.resize (static_cast<int> (numChannels));
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}
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~Oversampling2TimesEquirippleFIR() {}
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//===============================================================================
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SampleType getLatencyInSamples() override
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{
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return static_cast<SampleType> (coefficientsUp.getFilterOrder() + coefficientsDown.getFilterOrder()) * 0.5f;
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}
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void reset() override
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{
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OversamplingEngine<SampleType>::reset();
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stateUp.clear();
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stateDown.clear();
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stateDown2.clear();
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position.fill (0);
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}
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void processSamplesUp (dsp::AudioBlock<SampleType> &inputBlock) override
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{
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jassert (inputBlock.getNumChannels() <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumChannels()));
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jassert (inputBlock.getNumSamples() * OversamplingEngine<SampleType>::factor <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumSamples()));
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// Initialization
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auto fir = coefficientsUp.getRawCoefficients();
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auto N = coefficientsUp.getFilterOrder() + 1;
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auto Ndiv2 = N / 2;
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auto numSamples = inputBlock.getNumSamples();
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// Processing
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for (size_t channel = 0; channel < inputBlock.getNumChannels(); channel++)
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{
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auto bufferSamples = OversamplingEngine<SampleType>::buffer.getWritePointer (static_cast<int> (channel));
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auto buf = stateUp.getWritePointer (static_cast<int> (channel));
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auto samples = inputBlock.getChannelPointer (channel);
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for (size_t i = 0; i < numSamples; i++)
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{
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// Input
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buf[N - 1] = 2 * samples[i];
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// Convolution
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auto out = static_cast<SampleType> (0.0);
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for (size_t k = 0; k < Ndiv2; k += 2)
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out += (buf[k] + buf[N - k - 1]) * fir[k];
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// Outputs
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bufferSamples[i << 1] = out;
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bufferSamples[(i << 1) + 1] = buf[Ndiv2 + 1] * fir[Ndiv2];
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// Shift data
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for (size_t k = 0; k < N - 2; k += 2)
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buf[k] = buf[k + 2];
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}
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}
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}
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void processSamplesDown (dsp::AudioBlock<SampleType> &outputBlock) override
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{
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jassert (outputBlock.getNumChannels() <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumChannels()));
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jassert (outputBlock.getNumSamples() * OversamplingEngine<SampleType>::factor <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumSamples()));
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// Initialization
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auto fir = coefficientsDown.getRawCoefficients();
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auto N = coefficientsDown.getFilterOrder() + 1;
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auto Ndiv2 = N / 2;
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auto Ndiv4 = Ndiv2 / 2;
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auto numSamples = outputBlock.getNumSamples();
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// Processing
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for (size_t channel = 0; channel < outputBlock.getNumChannels(); channel++)
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{
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auto bufferSamples = OversamplingEngine<SampleType>::buffer.getWritePointer (static_cast<int> (channel));
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auto buf = stateDown.getWritePointer (static_cast<int> (channel));
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auto buf2 = stateDown2.getWritePointer (static_cast<int> (channel));
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auto samples = outputBlock.getChannelPointer (channel);
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auto pos = position.getUnchecked (static_cast<int> (channel));
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for (size_t i = 0; i < numSamples; i++)
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{
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// Input
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buf[N - 1] = bufferSamples[i << 1];
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// Convolution
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auto out = static_cast<SampleType> (0.0);
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for (size_t k = 0; k < Ndiv2; k += 2)
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out += (buf[k] + buf[N - k - 1]) * fir[k];
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// Output
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out += buf2[pos] * fir[Ndiv2];
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buf2[pos] = bufferSamples[(i << 1) + 1];
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samples[i] = out;
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// Shift data
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for (size_t k = 0; k < N - 2; k++)
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buf[k] = buf[k + 2];
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// Circular buffer
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pos = (pos == 0 ? Ndiv4 : pos - 1);
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}
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position.setUnchecked (static_cast<int> (channel), pos);
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}
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}
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private:
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//===============================================================================
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dsp::FIR::Coefficients<SampleType> coefficientsUp, coefficientsDown;
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AudioBuffer<SampleType> stateUp, stateDown, stateDown2;
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Array<size_t> position;
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//===============================================================================
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JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (Oversampling2TimesEquirippleFIR)
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};
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//===============================================================================
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/** Oversampling engine class performing 2 times oversampling using the Filter
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Design IIR Polyphase Allpass Cascaded method. The resulting filter is minimum
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phase, and provided with a method to get the exact resulting latency.
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*/
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template <typename SampleType>
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class Oversampling2TimesPolyphaseIIR : public OversamplingEngine<SampleType>
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{
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public:
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//===============================================================================
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Oversampling2TimesPolyphaseIIR (size_t numChannels,
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SampleType normalizedTransitionWidthUp,
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SampleType stopbandAttenuationdBUp,
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SampleType normalizedTransitionWidthDown,
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SampleType stopbandAttenuationdBDown) : OversamplingEngine<SampleType> (numChannels, 2)
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{
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auto structureUp = dsp::FilterDesign<SampleType>::designIIRLowpassHalfBandPolyphaseAllpassMethod (normalizedTransitionWidthUp, stopbandAttenuationdBUp);
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dsp::IIR::Coefficients<SampleType> coeffsUp = getCoefficients (structureUp);
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latency = static_cast<SampleType> (-(coeffsUp.getPhaseForFrequency (0.0001, 1.0)) / (0.0001 * MathConstants<double>::twoPi));
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auto structureDown = dsp::FilterDesign<SampleType>::designIIRLowpassHalfBandPolyphaseAllpassMethod (normalizedTransitionWidthDown, stopbandAttenuationdBDown);
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dsp::IIR::Coefficients<SampleType> coeffsDown = getCoefficients (structureDown);
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latency += static_cast<SampleType> (-(coeffsDown.getPhaseForFrequency (0.0001, 1.0)) / (0.0001 * MathConstants<double>::twoPi));
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for (auto i = 0; i < structureUp.directPath.size(); i++)
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coefficientsUp.add (structureUp.directPath[i].coefficients[0]);
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for (auto i = 1; i < structureUp.delayedPath.size(); i++)
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coefficientsUp.add (structureUp.delayedPath[i].coefficients[0]);
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for (auto i = 0; i < structureDown.directPath.size(); i++)
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coefficientsDown.add (structureDown.directPath[i].coefficients[0]);
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for (auto i = 1; i < structureDown.delayedPath.size(); i++)
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coefficientsDown.add (structureDown.delayedPath[i].coefficients[0]);
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v1Up.setSize (static_cast<int> (numChannels), coefficientsUp.size());
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v1Down.setSize (static_cast<int> (numChannels), coefficientsDown.size());
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delayDown.resize (static_cast<int> (numChannels));
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}
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~Oversampling2TimesPolyphaseIIR() {}
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//===============================================================================
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SampleType getLatencyInSamples() override
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{
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return latency;
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}
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void reset() override
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{
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OversamplingEngine<SampleType>::reset();
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v1Up.clear();
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v1Down.clear();
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delayDown.fill (0);
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}
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void processSamplesUp (dsp::AudioBlock<SampleType> &inputBlock) override
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{
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jassert (inputBlock.getNumChannels() <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumChannels()));
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jassert (inputBlock.getNumSamples() * OversamplingEngine<SampleType>::factor <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumSamples()));
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// Initialization
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auto coeffs = coefficientsUp.getRawDataPointer();
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auto numStages = coefficientsUp.size();
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auto delayedStages = numStages / 2;
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auto directStages = numStages - delayedStages;
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auto numSamples = inputBlock.getNumSamples();
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// Processing
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for (size_t channel = 0; channel < inputBlock.getNumChannels(); channel++)
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{
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auto bufferSamples = OversamplingEngine<SampleType>::buffer.getWritePointer (static_cast<int> (channel));
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auto lv1 = v1Up.getWritePointer (static_cast<int> (channel));
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auto samples = inputBlock.getChannelPointer (channel);
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for (size_t i = 0; i < numSamples; i++)
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{
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// Direct path cascaded allpass filters
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auto input = samples[i];
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for (auto n = 0; n < directStages; n++)
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{
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auto alpha = coeffs[n];
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auto output = alpha * input + lv1[n];
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lv1[n] = input - alpha * output;
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input = output;
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}
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// Output
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bufferSamples[i << 1] = input;
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// Delayed path cascaded allpass filters
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input = samples[i];
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for (auto n = directStages; n < numStages; n++)
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{
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auto alpha = coeffs[n];
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auto output = alpha * input + lv1[n];
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lv1[n] = input - alpha * output;
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input = output;
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}
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// Output
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bufferSamples[(i << 1) + 1] = input;
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}
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}
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// Snap To Zero
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snapToZero (true);
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}
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void processSamplesDown (dsp::AudioBlock<SampleType> &outputBlock) override
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{
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jassert (outputBlock.getNumChannels() <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumChannels()));
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jassert (outputBlock.getNumSamples() * OversamplingEngine<SampleType>::factor <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumSamples()));
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// Initialization
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auto coeffs = coefficientsDown.getRawDataPointer();
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auto numStages = coefficientsDown.size();
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auto delayedStages = numStages / 2;
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auto directStages = numStages - delayedStages;
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auto numSamples = outputBlock.getNumSamples();
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// Processing
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for (size_t channel = 0; channel < outputBlock.getNumChannels(); channel++)
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{
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auto bufferSamples = OversamplingEngine<SampleType>::buffer.getWritePointer (static_cast<int> (channel));
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auto lv1 = v1Down.getWritePointer (static_cast<int> (channel));
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auto samples = outputBlock.getChannelPointer (channel);
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auto delay = delayDown.getUnchecked (static_cast<int> (channel));
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for (size_t i = 0; i < numSamples; i++)
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{
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// Direct path cascaded allpass filters
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auto input = bufferSamples[i << 1];
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for (auto n = 0; n < directStages; n++)
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{
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auto alpha = coeffs[n];
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auto output = alpha * input + lv1[n];
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lv1[n] = input - alpha * output;
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input = output;
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}
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auto directOut = input;
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// Delayed path cascaded allpass filters
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input = bufferSamples[(i << 1) + 1];
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for (auto n = directStages; n < numStages; n++)
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{
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auto alpha = coeffs[n];
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auto output = alpha * input + lv1[n];
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lv1[n] = input - alpha * output;
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input = output;
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}
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// Output
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samples[i] = (delay + directOut) * static_cast<SampleType> (0.5);
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delay = input;
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}
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delayDown.setUnchecked (static_cast<int> (channel), delay);
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}
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// Snap To Zero
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snapToZero (false);
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}
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void snapToZero (bool snapUpProcessing)
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{
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if (snapUpProcessing)
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{
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for (auto channel = 0; channel < OversamplingEngine<SampleType>::buffer.getNumChannels(); channel++)
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{
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auto lv1 = v1Up.getWritePointer (channel);
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auto numStages = coefficientsUp.size();
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for (auto n = 0; n < numStages; n++)
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util::snapToZero (lv1[n]);
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}
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}
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else
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{
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for (auto channel = 0; channel < OversamplingEngine<SampleType>::buffer.getNumChannels(); channel++)
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{
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auto lv1 = v1Down.getWritePointer (channel);
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auto numStages = coefficientsDown.size();
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for (auto n = 0; n < numStages; n++)
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util::snapToZero (lv1[n]);
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}
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}
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}
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private:
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//===============================================================================
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/** This function calculates the equivalent high order IIR filter of a given
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polyphase cascaded allpass filters structure.
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*/
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const dsp::IIR::Coefficients<SampleType> getCoefficients (typename dsp::FilterDesign<SampleType>::IIRPolyphaseAllpassStructure& structure) const
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{
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dsp::Polynomial<SampleType> numerator1 ({ static_cast<SampleType> (1.0) });
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dsp::Polynomial<SampleType> denominator1 ({ static_cast<SampleType> (1.0) });
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dsp::Polynomial<SampleType> numerator2 ({ static_cast<SampleType> (1.0) });
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dsp::Polynomial<SampleType> denominator2 ({ static_cast<SampleType> (1.0) });
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dsp::Polynomial<SampleType> temp;
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for (auto n = 0; n < structure.directPath.size(); n++)
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{
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auto* coeffs = structure.directPath.getReference (n).getRawCoefficients();
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if (structure.directPath[n].getFilterOrder() == 1)
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{
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temp = dsp::Polynomial<SampleType> ({ coeffs[0], coeffs[1] });
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numerator1 = numerator1.getProductWith (temp);
|
|
|
|
temp = dsp::Polynomial<SampleType> ({ static_cast<SampleType> (1.0), coeffs[2] });
|
|
denominator1 = denominator1.getProductWith (temp);
|
|
}
|
|
else
|
|
{
|
|
temp = dsp::Polynomial<SampleType> ({ coeffs[0], coeffs[1], coeffs[2] });
|
|
numerator1 = numerator1.getProductWith (temp);
|
|
|
|
temp = dsp::Polynomial<SampleType> ({ static_cast<SampleType> (1.0), coeffs[3], coeffs[4] });
|
|
denominator1 = denominator1.getProductWith (temp);
|
|
}
|
|
}
|
|
|
|
for (auto n = 0; n < structure.delayedPath.size(); n++)
|
|
{
|
|
auto* coeffs = structure.delayedPath.getReference (n).getRawCoefficients();
|
|
|
|
if (structure.delayedPath[n].getFilterOrder() == 1)
|
|
{
|
|
temp = dsp::Polynomial<SampleType> ({ coeffs[0], coeffs[1] });
|
|
numerator2 = numerator2.getProductWith (temp);
|
|
|
|
temp = dsp::Polynomial<SampleType> ({ static_cast<SampleType> (1.0), coeffs[2] });
|
|
denominator2 = denominator2.getProductWith (temp);
|
|
}
|
|
else
|
|
{
|
|
temp = dsp::Polynomial<SampleType> ({ coeffs[0], coeffs[1], coeffs[2] });
|
|
numerator2 = numerator2.getProductWith (temp);
|
|
|
|
temp = dsp::Polynomial<SampleType> ({ static_cast<SampleType> (1.0), coeffs[3], coeffs[4] });
|
|
denominator2 = denominator2.getProductWith (temp);
|
|
}
|
|
}
|
|
|
|
dsp::Polynomial<SampleType> numeratorf1 = numerator1.getProductWith (denominator2);
|
|
dsp::Polynomial<SampleType> numeratorf2 = numerator2.getProductWith (denominator1);
|
|
dsp::Polynomial<SampleType> numerator = numeratorf1.getSumWith (numeratorf2);
|
|
dsp::Polynomial<SampleType> denominator = denominator1.getProductWith (denominator2);
|
|
|
|
dsp::IIR::Coefficients<SampleType> coeffs;
|
|
|
|
coeffs.coefficients.clear();
|
|
auto inversion = static_cast<SampleType> (1.0) / denominator[0];
|
|
|
|
for (auto i = 0; i <= numerator.getOrder(); i++)
|
|
coeffs.coefficients.add (numerator[i] * inversion);
|
|
|
|
for (auto i = 1; i <= denominator.getOrder(); i++)
|
|
coeffs.coefficients.add (denominator[i] * inversion);
|
|
|
|
return coeffs;
|
|
}
|
|
|
|
//===============================================================================
|
|
Array<SampleType> coefficientsUp, coefficientsDown;
|
|
SampleType latency;
|
|
|
|
AudioBuffer<SampleType> v1Up, v1Down;
|
|
Array<SampleType> delayDown;
|
|
|
|
//===============================================================================
|
|
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (Oversampling2TimesPolyphaseIIR)
|
|
};
|
|
|
|
|
|
//===============================================================================
|
|
template <typename SampleType>
|
|
Oversampling<SampleType>::Oversampling (size_t newNumChannels, size_t newFactor, FilterType newType, bool newMaxQuality)
|
|
{
|
|
jassert (newFactor >= 0 && newFactor <= 4 && newNumChannels > 0);
|
|
|
|
factorOversampling = static_cast<size_t> (1) << newFactor;
|
|
isMaximumQuality = newMaxQuality;
|
|
type = newType;
|
|
numChannels = newNumChannels;
|
|
|
|
if (newFactor == 0)
|
|
{
|
|
numStages = 1;
|
|
engines.add (new OversamplingDummy<SampleType> (numChannels));
|
|
}
|
|
else if (type == FilterType::filterHalfBandPolyphaseIIR)
|
|
{
|
|
numStages = newFactor;
|
|
|
|
for (size_t n = 0; n < numStages; n++)
|
|
{
|
|
auto twUp = (isMaximumQuality ? 0.10f : 0.12f) * (n == 0 ? 0.5f : 1.f);
|
|
auto twDown = (isMaximumQuality ? 0.12f : 0.15f) * (n == 0 ? 0.5f : 1.f);
|
|
|
|
auto gaindBStartUp = (isMaximumQuality ? -75.f : -65.f);
|
|
auto gaindBStartDown = (isMaximumQuality ? -70.f : -60.f);
|
|
auto gaindBFactorUp = (isMaximumQuality ? 10.f : 8.f);
|
|
auto gaindBFactorDown = (isMaximumQuality ? 10.f : 8.f);
|
|
|
|
engines.add (new Oversampling2TimesPolyphaseIIR<SampleType> (numChannels,
|
|
twUp, gaindBStartUp + gaindBFactorUp * n,
|
|
twDown, gaindBStartDown + gaindBFactorDown * n));
|
|
}
|
|
}
|
|
else if (type == FilterType::filterHalfBandFIREquiripple)
|
|
{
|
|
numStages = newFactor;
|
|
|
|
for (size_t n = 0; n < numStages; n++)
|
|
{
|
|
auto twUp = (isMaximumQuality ? 0.10f : 0.12f) * (n == 0 ? 0.5f : 1.f);
|
|
auto twDown = (isMaximumQuality ? 0.12f : 0.15f) * (n == 0 ? 0.5f : 1.f);
|
|
|
|
auto gaindBStartUp = (isMaximumQuality ? -90.f : -70.f);
|
|
auto gaindBStartDown = (isMaximumQuality ? -70.f : -60.f);
|
|
auto gaindBFactorUp = (isMaximumQuality ? 10.f : 8.f);
|
|
auto gaindBFactorDown = (isMaximumQuality ? 10.f : 8.f);
|
|
|
|
engines.add (new Oversampling2TimesEquirippleFIR<SampleType> (numChannels,
|
|
twUp, gaindBStartUp + gaindBFactorUp * n,
|
|
twDown, gaindBStartDown + gaindBFactorDown * n));
|
|
}
|
|
}
|
|
}
|
|
|
|
template <typename SampleType>
|
|
Oversampling<SampleType>::~Oversampling()
|
|
{
|
|
engines.clear();
|
|
}
|
|
|
|
//===============================================================================
|
|
template <typename SampleType>
|
|
SampleType Oversampling<SampleType>::getLatencyInSamples() noexcept
|
|
{
|
|
auto latency = static_cast<SampleType> (0);
|
|
size_t order = 1;
|
|
|
|
for (size_t n = 0; n < numStages; n++)
|
|
{
|
|
auto& engine = *engines[static_cast<int> (n)];
|
|
|
|
order *= engine.getFactor();
|
|
latency += engine.getLatencyInSamples() / static_cast<SampleType> (order);
|
|
}
|
|
|
|
return latency;
|
|
}
|
|
|
|
template <typename SampleType>
|
|
size_t Oversampling<SampleType>::getOversamplingFactor() noexcept
|
|
{
|
|
return factorOversampling;
|
|
}
|
|
|
|
//===============================================================================
|
|
template <typename SampleType>
|
|
void Oversampling<SampleType>::initProcessing (size_t maximumNumberOfSamplesBeforeOversampling)
|
|
{
|
|
jassert (! engines.isEmpty());
|
|
auto currentNumSamples = maximumNumberOfSamplesBeforeOversampling;
|
|
|
|
for (size_t n = 0; n < numStages; n++)
|
|
{
|
|
auto& engine = *engines[static_cast<int> (n)];
|
|
|
|
engine.initProcessing (currentNumSamples);
|
|
currentNumSamples *= engine.getFactor();
|
|
}
|
|
|
|
isReady = true;
|
|
reset();
|
|
}
|
|
|
|
template <typename SampleType>
|
|
void Oversampling<SampleType>::reset() noexcept
|
|
{
|
|
jassert (! engines.isEmpty());
|
|
|
|
if (isReady)
|
|
for (auto n = 0; n < engines.size(); n++)
|
|
engines[n]->reset();
|
|
}
|
|
|
|
template <typename SampleType>
|
|
typename dsp::AudioBlock<SampleType> Oversampling<SampleType>::processSamplesUp (const dsp::AudioBlock<SampleType> &inputBlock) noexcept
|
|
{
|
|
jassert (! engines.isEmpty());
|
|
|
|
if (! isReady)
|
|
return dsp::AudioBlock<SampleType>();
|
|
|
|
dsp::AudioBlock<SampleType> audioBlock = inputBlock;
|
|
|
|
for (size_t n = 0; n < numStages; n++)
|
|
{
|
|
auto& engine = *engines[static_cast<int> (n)];
|
|
engine.processSamplesUp (audioBlock);
|
|
audioBlock = engine.getProcessedSamples (audioBlock.getNumSamples() * engine.getFactor());
|
|
}
|
|
|
|
return audioBlock;
|
|
}
|
|
|
|
template <typename SampleType>
|
|
void Oversampling<SampleType>::processSamplesDown (dsp::AudioBlock<SampleType> &outputBlock) noexcept
|
|
{
|
|
jassert (! engines.isEmpty());
|
|
|
|
if (! isReady)
|
|
return;
|
|
|
|
auto currentNumSamples = outputBlock.getNumSamples();
|
|
|
|
for (size_t n = 0; n < numStages - 1; n++)
|
|
currentNumSamples *= engines[static_cast<int> (n)]->getFactor();
|
|
|
|
for (size_t n = numStages - 1; n > 0; n--)
|
|
{
|
|
auto& engine = *engines[static_cast<int> (n)];
|
|
|
|
auto audioBlock = engines[static_cast<int> (n - 1)]->getProcessedSamples (currentNumSamples);
|
|
engine.processSamplesDown (audioBlock);
|
|
|
|
currentNumSamples /= engine.getFactor();
|
|
}
|
|
|
|
engines[static_cast<int> (0)]->processSamplesDown (outputBlock);
|
|
}
|
|
|
|
template class Oversampling<float>;
|
|
template class Oversampling<double>;
|
|
|
|
} // namespace dsp
|
|
} // namespace juce
|