25bd5d8adb
subrepo: subdir: "deps/juce" merged: "b13f9084e" upstream: origin: "https://github.com/essej/JUCE.git" branch: "sono6good" commit: "b13f9084e" git-subrepo: version: "0.4.3" origin: "https://github.com/ingydotnet/git-subrepo.git" commit: "2f68596"
771 lines
29 KiB
C++
771 lines
29 KiB
C++
/*
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==============================================================================
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This file is part of the JUCE library.
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Copyright (c) 2020 - Raw Material Software Limited
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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 6 End-User License
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Agreement and JUCE Privacy Policy (both effective as of the 16th June 2020).
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End User License Agreement: www.juce.com/juce-6-licence
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Privacy Policy: www.juce.com/juce-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 stages 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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struct Oversampling<SampleType>::OversamplingStage
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{
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OversamplingStage (size_t numChans, size_t newFactor) : numChannels (numChans), factor (newFactor) {}
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virtual ~OversamplingStage() {}
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//==============================================================================
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virtual SampleType getLatencyInSamples() const = 0;
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virtual void initProcessing (size_t maximumNumberOfSamplesBeforeOversampling)
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{
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buffer.setSize (static_cast<int> (numChannels),
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static_cast<int> (maximumNumberOfSamplesBeforeOversampling * factor),
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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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AudioBlock<SampleType> getProcessedSamples (size_t numSamples)
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{
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return AudioBlock<SampleType> (buffer).getSubBlock (0, numSamples);
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}
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virtual void processSamplesUp (const AudioBlock<const SampleType>&) = 0;
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virtual void processSamplesDown (AudioBlock<SampleType>&) = 0;
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AudioBuffer<SampleType> buffer;
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size_t numChannels, factor;
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};
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//==============================================================================
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/** Dummy oversampling stage 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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struct OversamplingDummy : public Oversampling<SampleType>::OversamplingStage
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{
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using ParentType = typename Oversampling<SampleType>::OversamplingStage;
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OversamplingDummy (size_t numChans) : ParentType (numChans, 1) {}
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//==============================================================================
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SampleType getLatencyInSamples() const override
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{
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return 0;
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}
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void processSamplesUp (const AudioBlock<const SampleType>& inputBlock) override
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{
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jassert (inputBlock.getNumChannels() <= static_cast<size_t> (ParentType::buffer.getNumChannels()));
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jassert (inputBlock.getNumSamples() * ParentType::factor <= static_cast<size_t> (ParentType::buffer.getNumSamples()));
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for (size_t channel = 0; channel < inputBlock.getNumChannels(); ++channel)
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ParentType::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 (AudioBlock<SampleType>& outputBlock) override
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{
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jassert (outputBlock.getNumChannels() <= static_cast<size_t> (ParentType::buffer.getNumChannels()));
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jassert (outputBlock.getNumSamples() * ParentType::factor <= static_cast<size_t> (ParentType::buffer.getNumSamples()));
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outputBlock.copyFrom (ParentType::getProcessedSamples (outputBlock.getNumSamples()));
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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 stage 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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struct Oversampling2TimesEquirippleFIR : public Oversampling<SampleType>::OversamplingStage
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{
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using ParentType = typename Oversampling<SampleType>::OversamplingStage;
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Oversampling2TimesEquirippleFIR (size_t numChans,
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SampleType normalisedTransitionWidthUp,
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SampleType stopbandAmplitudedBUp,
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SampleType normalisedTransitionWidthDown,
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SampleType stopbandAmplitudedBDown)
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: ParentType (numChans, 2)
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{
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coefficientsUp = *FilterDesign<SampleType>::designFIRLowpassHalfBandEquirippleMethod (normalisedTransitionWidthUp, stopbandAmplitudedBUp);
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coefficientsDown = *FilterDesign<SampleType>::designFIRLowpassHalfBandEquirippleMethod (normalisedTransitionWidthDown, stopbandAmplitudedBDown);
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auto N = coefficientsUp.getFilterOrder() + 1;
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stateUp.setSize (static_cast<int> (this->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> (this->numChannels), static_cast<int> (N));
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stateDown2.setSize (static_cast<int> (this->numChannels), static_cast<int> (Ndiv4 + 1));
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position.resize (static_cast<int> (this->numChannels));
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}
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//==============================================================================
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SampleType getLatencyInSamples() const 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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ParentType::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 (const AudioBlock<const SampleType>& inputBlock) override
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{
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jassert (inputBlock.getNumChannels() <= static_cast<size_t> (ParentType::buffer.getNumChannels()));
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jassert (inputBlock.getNumSamples() * ParentType::factor <= static_cast<size_t> (ParentType::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 = ParentType::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 (AudioBlock<SampleType>& outputBlock) override
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{
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jassert (outputBlock.getNumChannels() <= static_cast<size_t> (ParentType::buffer.getNumChannels()));
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jassert (outputBlock.getNumSamples() * ParentType::factor <= static_cast<size_t> (ParentType::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 = ParentType::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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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 stage 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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struct Oversampling2TimesPolyphaseIIR : public Oversampling<SampleType>::OversamplingStage
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{
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using ParentType = typename Oversampling<SampleType>::OversamplingStage;
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Oversampling2TimesPolyphaseIIR (size_t numChans,
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SampleType normalisedTransitionWidthUp,
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SampleType stopbandAmplitudedBUp,
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SampleType normalisedTransitionWidthDown,
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SampleType stopbandAmplitudedBDown)
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: ParentType (numChans, 2)
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{
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auto structureUp = FilterDesign<SampleType>::designIIRLowpassHalfBandPolyphaseAllpassMethod (normalisedTransitionWidthUp, stopbandAmplitudedBUp);
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auto 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 = FilterDesign<SampleType>::designIIRLowpassHalfBandPolyphaseAllpassMethod (normalisedTransitionWidthDown, stopbandAmplitudedBDown);
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auto 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.getObjectPointer (i)->coefficients[0]);
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for (auto i = 1; i < structureUp.delayedPath.size(); ++i)
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coefficientsUp.add (structureUp.delayedPath.getObjectPointer (i)->coefficients[0]);
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for (auto i = 0; i < structureDown.directPath.size(); ++i)
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coefficientsDown.add (structureDown.directPath.getObjectPointer (i)->coefficients[0]);
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for (auto i = 1; i < structureDown.delayedPath.size(); ++i)
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coefficientsDown.add (structureDown.delayedPath.getObjectPointer (i)->coefficients[0]);
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v1Up.setSize (static_cast<int> (this->numChannels), coefficientsUp.size());
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v1Down.setSize (static_cast<int> (this->numChannels), coefficientsDown.size());
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delayDown.resize (static_cast<int> (this->numChannels));
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}
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//==============================================================================
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SampleType getLatencyInSamples() const 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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ParentType::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 (const AudioBlock<const SampleType>& inputBlock) override
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{
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jassert (inputBlock.getNumChannels() <= static_cast<size_t> (ParentType::buffer.getNumChannels()));
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jassert (inputBlock.getNumSamples() * ParentType::factor <= static_cast<size_t> (ParentType::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 = ParentType::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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#if JUCE_DSP_ENABLE_SNAP_TO_ZERO
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snapToZero (true);
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#endif
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}
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void processSamplesDown (AudioBlock<SampleType>& outputBlock) override
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{
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jassert (outputBlock.getNumChannels() <= static_cast<size_t> (ParentType::buffer.getNumChannels()));
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jassert (outputBlock.getNumSamples() * ParentType::factor <= static_cast<size_t> (ParentType::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 = ParentType::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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#if JUCE_DSP_ENABLE_SNAP_TO_ZERO
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snapToZero (false);
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#endif
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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 < ParentType::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 < ParentType::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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IIR::Coefficients<SampleType> getCoefficients (typename FilterDesign<SampleType>::IIRPolyphaseAllpassStructure& structure) const
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{
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constexpr auto one = static_cast<SampleType> (1.0);
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Polynomial<SampleType> numerator1 ({ one }), denominator1 ({ one }),
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numerator2 ({ one }), denominator2 ({ one });
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for (auto* i : structure.directPath)
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{
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auto coeffs = i->getRawCoefficients();
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if (i->getFilterOrder() == 1)
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{
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numerator1 = numerator1 .getProductWith (Polynomial<SampleType> ({ coeffs[0], coeffs[1] }));
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denominator1 = denominator1.getProductWith (Polynomial<SampleType> ({ one, coeffs[2] }));
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}
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else
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{
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numerator1 = numerator1 .getProductWith (Polynomial<SampleType> ({ coeffs[0], coeffs[1], coeffs[2] }));
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denominator1 = denominator1.getProductWith (Polynomial<SampleType> ({ one, coeffs[3], coeffs[4] }));
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}
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}
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for (auto* i : structure.delayedPath)
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{
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auto coeffs = i->getRawCoefficients();
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if (i->getFilterOrder() == 1)
|
|
{
|
|
numerator2 = numerator2 .getProductWith (Polynomial<SampleType> ({ coeffs[0], coeffs[1] }));
|
|
denominator2 = denominator2.getProductWith (Polynomial<SampleType> ({ one, coeffs[2] }));
|
|
}
|
|
else
|
|
{
|
|
numerator2 = numerator2 .getProductWith (Polynomial<SampleType> ({ coeffs[0], coeffs[1], coeffs[2] }));
|
|
denominator2 = denominator2.getProductWith (Polynomial<SampleType> ({ one, coeffs[3], coeffs[4] }));
|
|
}
|
|
}
|
|
|
|
auto numeratorf1 = numerator1.getProductWith (denominator2);
|
|
auto numeratorf2 = numerator2.getProductWith (denominator1);
|
|
auto numerator = numeratorf1.getSumWith (numeratorf2);
|
|
auto denominator = denominator1.getProductWith (denominator2);
|
|
|
|
IIR::Coefficients<SampleType> coeffs;
|
|
|
|
coeffs.coefficients.clear();
|
|
auto inversion = one / denominator[0];
|
|
|
|
for (int i = 0; i <= numerator.getOrder(); ++i)
|
|
coeffs.coefficients.add (numerator[i] * inversion);
|
|
|
|
for (int 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)
|
|
: numChannels (newNumChannels)
|
|
{
|
|
jassert (numChannels > 0);
|
|
|
|
addDummyOversamplingStage();
|
|
}
|
|
|
|
template <typename SampleType>
|
|
Oversampling<SampleType>::Oversampling (size_t newNumChannels, size_t newFactor,
|
|
FilterType newType, bool isMaximumQuality,
|
|
bool useIntegerLatency)
|
|
: numChannels (newNumChannels), shouldUseIntegerLatency (useIntegerLatency)
|
|
{
|
|
jassert (isPositiveAndBelow (newFactor, 5) && numChannels > 0);
|
|
|
|
if (newFactor == 0)
|
|
{
|
|
addDummyOversamplingStage();
|
|
}
|
|
else if (newType == FilterType::filterHalfBandPolyphaseIIR)
|
|
{
|
|
for (size_t n = 0; n < newFactor; ++n)
|
|
{
|
|
auto twUp = (isMaximumQuality ? 0.10f : 0.12f) * (n == 0 ? 0.5f : 1.0f);
|
|
auto twDown = (isMaximumQuality ? 0.12f : 0.15f) * (n == 0 ? 0.5f : 1.0f);
|
|
|
|
auto gaindBStartUp = (isMaximumQuality ? -90.0f : -70.0f);
|
|
auto gaindBStartDown = (isMaximumQuality ? -75.0f : -60.0f);
|
|
auto gaindBFactorUp = (isMaximumQuality ? 10.0f : 8.0f);
|
|
auto gaindBFactorDown = (isMaximumQuality ? 10.0f : 8.0f);
|
|
|
|
addOversamplingStage (FilterType::filterHalfBandPolyphaseIIR,
|
|
twUp, gaindBStartUp + gaindBFactorUp * (float) n,
|
|
twDown, gaindBStartDown + gaindBFactorDown * (float) n);
|
|
}
|
|
}
|
|
else if (newType == FilterType::filterHalfBandFIREquiripple)
|
|
{
|
|
for (size_t n = 0; n < newFactor; ++n)
|
|
{
|
|
auto twUp = (isMaximumQuality ? 0.10f : 0.12f) * (n == 0 ? 0.5f : 1.0f);
|
|
auto twDown = (isMaximumQuality ? 0.12f : 0.15f) * (n == 0 ? 0.5f : 1.0f);
|
|
|
|
auto gaindBStartUp = (isMaximumQuality ? -90.0f : -70.0f);
|
|
auto gaindBStartDown = (isMaximumQuality ? -75.0f : -60.0f);
|
|
auto gaindBFactorUp = (isMaximumQuality ? 10.0f : 8.0f);
|
|
auto gaindBFactorDown = (isMaximumQuality ? 10.0f : 8.0f);
|
|
|
|
addOversamplingStage (FilterType::filterHalfBandFIREquiripple,
|
|
twUp, gaindBStartUp + gaindBFactorUp * (float) n,
|
|
twDown, gaindBStartDown + gaindBFactorDown * (float) n);
|
|
}
|
|
}
|
|
}
|
|
|
|
template <typename SampleType>
|
|
Oversampling<SampleType>::~Oversampling()
|
|
{
|
|
stages.clear();
|
|
}
|
|
|
|
//==============================================================================
|
|
template <typename SampleType>
|
|
void Oversampling<SampleType>::addDummyOversamplingStage()
|
|
{
|
|
stages.add (new OversamplingDummy<SampleType> (numChannels));
|
|
}
|
|
|
|
template <typename SampleType>
|
|
void Oversampling<SampleType>::addOversamplingStage (FilterType type,
|
|
float normalisedTransitionWidthUp,
|
|
float stopbandAmplitudedBUp,
|
|
float normalisedTransitionWidthDown,
|
|
float stopbandAmplitudedBDown)
|
|
{
|
|
if (type == FilterType::filterHalfBandPolyphaseIIR)
|
|
{
|
|
stages.add (new Oversampling2TimesPolyphaseIIR<SampleType> (numChannels,
|
|
normalisedTransitionWidthUp, stopbandAmplitudedBUp,
|
|
normalisedTransitionWidthDown, stopbandAmplitudedBDown));
|
|
}
|
|
else
|
|
{
|
|
stages.add (new Oversampling2TimesEquirippleFIR<SampleType> (numChannels,
|
|
normalisedTransitionWidthUp, stopbandAmplitudedBUp,
|
|
normalisedTransitionWidthDown, stopbandAmplitudedBDown));
|
|
}
|
|
|
|
factorOversampling *= 2;
|
|
}
|
|
|
|
template <typename SampleType>
|
|
void Oversampling<SampleType>::clearOversamplingStages()
|
|
{
|
|
stages.clear();
|
|
factorOversampling = 1u;
|
|
}
|
|
|
|
//==============================================================================
|
|
template <typename SampleType>
|
|
void Oversampling<SampleType>::setUsingIntegerLatency (bool useIntegerLatency) noexcept
|
|
{
|
|
shouldUseIntegerLatency = useIntegerLatency;
|
|
}
|
|
|
|
template <typename SampleType>
|
|
SampleType Oversampling<SampleType>::getLatencyInSamples() const noexcept
|
|
{
|
|
auto latency = getUncompensatedLatency();
|
|
return shouldUseIntegerLatency ? latency + fractionalDelay : latency;
|
|
}
|
|
|
|
template <typename SampleType>
|
|
SampleType Oversampling<SampleType>::getUncompensatedLatency() const noexcept
|
|
{
|
|
auto latency = static_cast<SampleType> (0);
|
|
size_t order = 1;
|
|
|
|
for (auto* stage : stages)
|
|
{
|
|
order *= stage->factor;
|
|
latency += stage->getLatencyInSamples() / static_cast<SampleType> (order);
|
|
}
|
|
|
|
return latency;
|
|
}
|
|
|
|
template <typename SampleType>
|
|
size_t Oversampling<SampleType>::getOversamplingFactor() const noexcept
|
|
{
|
|
return factorOversampling;
|
|
}
|
|
|
|
//==============================================================================
|
|
template <typename SampleType>
|
|
void Oversampling<SampleType>::initProcessing (size_t maximumNumberOfSamplesBeforeOversampling)
|
|
{
|
|
jassert (! stages.isEmpty());
|
|
auto currentNumSamples = maximumNumberOfSamplesBeforeOversampling;
|
|
|
|
for (auto* stage : stages)
|
|
{
|
|
stage->initProcessing (currentNumSamples);
|
|
currentNumSamples *= stage->factor;
|
|
}
|
|
|
|
ProcessSpec spec = { 0.0, (uint32) maximumNumberOfSamplesBeforeOversampling, (uint32) numChannels };
|
|
delay.prepare (spec);
|
|
updateDelayLine();
|
|
|
|
isReady = true;
|
|
reset();
|
|
}
|
|
|
|
template <typename SampleType>
|
|
void Oversampling<SampleType>::reset() noexcept
|
|
{
|
|
jassert (! stages.isEmpty());
|
|
|
|
if (isReady)
|
|
for (auto* stage : stages)
|
|
stage->reset();
|
|
|
|
delay.reset();
|
|
}
|
|
|
|
template <typename SampleType>
|
|
AudioBlock<SampleType> Oversampling<SampleType>::processSamplesUp (const AudioBlock<const SampleType>& inputBlock) noexcept
|
|
{
|
|
jassert (! stages.isEmpty());
|
|
|
|
if (! isReady)
|
|
return {};
|
|
|
|
auto* firstStage = stages.getUnchecked (0);
|
|
firstStage->processSamplesUp (inputBlock);
|
|
auto block = firstStage->getProcessedSamples (inputBlock.getNumSamples() * firstStage->factor);
|
|
|
|
for (int i = 1; i < stages.size(); ++i)
|
|
{
|
|
stages[i]->processSamplesUp (block);
|
|
block = stages[i]->getProcessedSamples (block.getNumSamples() * stages[i]->factor);
|
|
}
|
|
|
|
return block;
|
|
}
|
|
|
|
template <typename SampleType>
|
|
void Oversampling<SampleType>::processSamplesDown (AudioBlock<SampleType>& outputBlock) noexcept
|
|
{
|
|
jassert (! stages.isEmpty());
|
|
|
|
if (! isReady)
|
|
return;
|
|
|
|
auto currentNumSamples = outputBlock.getNumSamples();
|
|
|
|
for (int n = 0; n < stages.size() - 1; ++n)
|
|
currentNumSamples *= stages.getUnchecked(n)->factor;
|
|
|
|
for (int n = stages.size() - 1; n > 0; --n)
|
|
{
|
|
auto& stage = *stages.getUnchecked(n);
|
|
auto audioBlock = stages.getUnchecked (n - 1)->getProcessedSamples (currentNumSamples);
|
|
stage.processSamplesDown (audioBlock);
|
|
|
|
currentNumSamples /= stage.factor;
|
|
}
|
|
|
|
stages.getFirst()->processSamplesDown (outputBlock);
|
|
|
|
if (shouldUseIntegerLatency && fractionalDelay > static_cast<SampleType> (0.0))
|
|
{
|
|
auto context = ProcessContextReplacing<SampleType> (outputBlock);
|
|
delay.process (context);
|
|
}
|
|
}
|
|
|
|
template <typename SampleType>
|
|
void Oversampling<SampleType>::updateDelayLine()
|
|
{
|
|
auto latency = getUncompensatedLatency();
|
|
fractionalDelay = static_cast<SampleType> (1.0) - (latency - std::floor (latency));
|
|
|
|
if (fractionalDelay == static_cast<SampleType> (1.0))
|
|
fractionalDelay = static_cast<SampleType> (0.0);
|
|
else if (fractionalDelay < static_cast<SampleType> (0.618))
|
|
fractionalDelay += static_cast<SampleType> (1.0);
|
|
|
|
delay.setDelay (fractionalDelay);
|
|
}
|
|
|
|
template class Oversampling<float>;
|
|
template class Oversampling<double>;
|
|
|
|
} // namespace dsp
|
|
} // namespace juce
|