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Added Oversampling processor to DSP module

This commit is contained in:
hogliux 2017-08-23 16:15:58 +01:00
parent 945b3e8a14
commit bd6ca234cb
4 changed files with 809 additions and 0 deletions

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@ -55,6 +55,7 @@ namespace juce
{
#include "processors/juce_FIRFilter.cpp"
#include "processors/juce_IIRFilter.cpp"
#include "processors/juce_Oversampling.cpp"
#include "maths/juce_SpecialFunctions.cpp"
#include "maths/juce_Matrix.cpp"
#include "maths/juce_LookupTable.cpp"

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@ -213,6 +213,7 @@ namespace juce
#include "processors/juce_FIRFilter.h"
#include "processors/juce_Oscillator.h"
#include "processors/juce_StateVariableFilter.h"
#include "processors/juce_Oversampling.h"
#include "frequency/juce_FFT.h"
#include "frequency/juce_Convolution.h"
#include "frequency/juce_Windowing.h"

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@ -0,0 +1,672 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2017 - ROLI Ltd.
JUCE is an open source library subject to commercial or open-source
licensing.
By using JUCE, you agree to the terms of both the JUCE 5 End-User License
Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
27th April 2017).
End User License Agreement: www.juce.com/juce-5-licence
Privacy Policy: www.juce.com/juce-5-privacy-policy
Or: You may also use this code under the terms of the GPL v3 (see
www.gnu.org/licenses).
JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
DISCLAIMED.
==============================================================================
*/
//===============================================================================
/** Abstract class for the provided oversampling engines used internally in
the Oversampling class.
*/
template <typename SampleType>
class OversamplingEngine
{
public:
//===============================================================================
OversamplingEngine (size_t newFactor) { factor = newFactor; }
virtual ~OversamplingEngine() {}
//===============================================================================
virtual SampleType getLatencyInSamples() = 0;
size_t getFactor() { return factor; }
virtual void initProcessing (size_t maximumNumberOfSamplesBeforeOversampling)
{
buffer.setSize (1, static_cast<int> (maximumNumberOfSamplesBeforeOversampling * factor));
}
virtual void reset()
{
buffer.clear();
}
SampleType* getProcessedSamples() { return buffer.getWritePointer (0); }
size_t getNumProcessedSamples() { return static_cast<size_t> (buffer.getNumSamples()); }
virtual void processSamplesUp (SampleType *samples, size_t numSamples) = 0;
virtual void processSamplesDown (SampleType *samples, size_t numSamples) = 0;
protected:
//===============================================================================
AudioBuffer<SampleType> buffer;
size_t factor;
};
//===============================================================================
/** Dummy oversampling engine class which simply copies and pastes the input
signal, which could be equivalent to a "one time" oversampling processing.
*/
template <typename SampleType>
class OversamplingDummy : public OversamplingEngine<SampleType>
{
public:
//===============================================================================
OversamplingDummy() : OversamplingEngine<SampleType>(1) {}
~OversamplingDummy() {}
//===============================================================================
SampleType getLatencyInSamples() override
{
return 0.f;
}
void processSamplesUp (SampleType *samples, size_t numSamples) override
{
auto bufferSamples = this->buffer.getWritePointer (0);
for (size_t i = 0; i < numSamples; i++)
bufferSamples[i] = samples[i];
}
void processSamplesDown (SampleType *samples, size_t numSamples) override
{
auto bufferSamples = OversamplingEngine<SampleType>::buffer.getWritePointer (0);
for (size_t i = 0; i < numSamples; i++)
samples[i] = bufferSamples[i];
}
private:
//===============================================================================
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (OversamplingDummy)
};
//===============================================================================
/**
Oversampling engine class performing 2 times oversampling using the Filter
Design FIR Equiripple method. The resulting filter is linear phase,
symmetric, and has every two samples but the middle one equal to zero,
leading to specific processing optimizations.
*/
template <typename SampleType>
class Oversampling2TimesEquirippleFIR : public OversamplingEngine<SampleType>
{
public:
//===============================================================================
Oversampling2TimesEquirippleFIR (SampleType normalizedTransitionWidthUp,
SampleType stopbandAttenuationdBUp,
SampleType normalizedTransitionWidthDown,
SampleType stopbandAttenuationdBDown) : OversamplingEngine<SampleType> (2)
{
coefficientsUp = *dsp::FilterDesign<SampleType>::designFIRLowpassHalfBandEquirippleMethod (normalizedTransitionWidthUp, stopbandAttenuationdBUp);
coefficientsDown = *dsp::FilterDesign<SampleType>::designFIRLowpassHalfBandEquirippleMethod (normalizedTransitionWidthDown, stopbandAttenuationdBDown);
auto N = coefficientsDown.getFilterOrder() + 1;
auto Ndiv2 = N / 2;
auto Ndiv4 = Ndiv2 / 2;
stateUp.setSize (1, static_cast<int> (coefficientsUp.getFilterOrder() + 1));
stateDown.setSize (1, static_cast<int> (N));
stateDown2.setSize (1, static_cast<int> (Ndiv4));
}
~Oversampling2TimesEquirippleFIR() {}
//===============================================================================
SampleType getLatencyInSamples() override
{
return static_cast<SampleType> (coefficientsUp.getFilterOrder() + coefficientsDown.getFilterOrder());
}
void reset() override
{
OversamplingEngine<SampleType>::reset();
stateUp.clear();
stateDown.clear();
stateDown2.clear();
position = 0;
}
void processSamplesUp (SampleType *samples, size_t numSamples) override
{
// Initialization
auto bufferSamples = OversamplingEngine<SampleType>::buffer.getWritePointer (0);
auto fir = coefficientsUp.getRawCoefficients();
auto buf = stateUp.getWritePointer (0);
auto N = coefficientsUp.getFilterOrder() + 1;
auto Ndiv2 = N / 2;
// Processing
for (size_t i = 0; i < numSamples; i++)
{
// Input
buf[N - 1] = 2 * samples[i];
// Convolution
auto out = static_cast<SampleType> (0.0);
for (size_t k = 0; k < Ndiv2; k += 2)
out += (buf[k] + buf[N - k - 1]) * fir[k];
// Outputs
bufferSamples[i << 1] = out;
bufferSamples[(i << 1) + 1] = buf[Ndiv2 + 1] * fir[Ndiv2];
// Shift data
for (size_t k = 0; k < N - 2; k+=2)
buf[k] = buf[k + 2];
}
}
void processSamplesDown (SampleType *samples, size_t numSamples) override
{
// Initialization
auto bufferSamples = OversamplingEngine<SampleType>::buffer.getWritePointer (0);
auto fir = coefficientsDown.getRawCoefficients();
auto buf = stateDown.getWritePointer (0);
auto buf2 = stateDown2.getWritePointer (0);
auto N = coefficientsDown.getFilterOrder() + 1;
auto Ndiv2 = N / 2;
auto Ndiv4 = Ndiv2 / 2;
// Processing
for (size_t i = 0; i < numSamples; i++)
{
// Input
buf[N - 1] = bufferSamples[2 * i];
// Convolution
auto out = static_cast<SampleType> (0.0);
for (size_t k = 0; k < Ndiv2; k += 2)
out += (buf[k] + buf[N - k - 1]) * fir[k];
// Output
out += buf2[position] * fir[Ndiv2];
buf2[position] = bufferSamples[2 * i + 1];
samples[i] = out;
// Shift data
for (size_t k = 0; k < N - 2; k++)
buf[k] = buf[k + 2];
// Circular buffer
position = (position == 0 ? Ndiv4 - 1 : position - 1);
}
}
private:
//===============================================================================
dsp::FIR::Coefficients<SampleType> coefficientsUp, coefficientsDown;
AudioBuffer<SampleType> stateUp, stateDown, stateDown2;
size_t position;
//===============================================================================
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (Oversampling2TimesEquirippleFIR)
};
//===============================================================================
/** Oversampling engine class performing 2 times oversampling using the Filter
Design IIR Polyphase Allpass Cascaded method. The resulting filter is minimum
phase, and provided with a method to get the exact resulting latency.
*/
template <typename SampleType>
class Oversampling2TimesPolyphaseIIR : public OversamplingEngine<SampleType>
{
public:
//===============================================================================
Oversampling2TimesPolyphaseIIR (SampleType normalizedTransitionWidthUp,
SampleType stopbandAttenuationdBUp,
SampleType normalizedTransitionWidthDown,
SampleType stopbandAttenuationdBDown) : OversamplingEngine<SampleType> (2)
{
auto structureUp = dsp::FilterDesign<SampleType>::designIIRLowpassHalfBandPolyphaseAllpassMethod (normalizedTransitionWidthUp, stopbandAttenuationdBUp);
dsp::IIR::Coefficients<SampleType> coeffsUp = getCoefficients (structureUp);
latency = static_cast<SampleType> (-(coeffsUp.getPhaseForFrequency (0.0001, 1.0)) / (0.0001 * 2 * double_Pi));
auto structureDown = dsp::FilterDesign<SampleType>::designIIRLowpassHalfBandPolyphaseAllpassMethod (normalizedTransitionWidthDown, stopbandAttenuationdBDown);
dsp::IIR::Coefficients<SampleType> coeffsDown = getCoefficients (structureDown);
latency += static_cast<SampleType> (-(coeffsDown.getPhaseForFrequency (0.0001, 1.0)) / (0.0001 * 2 * double_Pi));
for (auto i = 0; i < structureUp.directPath.size(); i++)
coefficientsUp.add (structureUp.directPath[i].coefficients[0]);
for (auto i = 1; i < structureUp.delayedPath.size(); i++)
coefficientsUp.add (structureUp.delayedPath[i].coefficients[0]);
for (auto i = 0; i < structureDown.directPath.size(); i++)
coefficientsDown.add (structureDown.directPath[i].coefficients[0]);
for (auto i = 1; i < structureDown.delayedPath.size(); i++)
coefficientsDown.add (structureDown.delayedPath[i].coefficients[0]);
v1Up.resize (coefficientsUp.size());
v1Down.resize (coefficientsDown.size());
}
~Oversampling2TimesPolyphaseIIR() {}
//===============================================================================
SampleType getLatencyInSamples() override
{
return latency;
}
void reset() override
{
OversamplingEngine<SampleType>::reset();
v1Up.fill (0);
v1Down.fill (0);
delayDown = 0;
}
void processSamplesUp (SampleType *samples, size_t numSamples) override
{
// Initialization
auto bufferSamples = OversamplingEngine<SampleType>::buffer.getWritePointer (0);
auto coeffs = coefficientsUp.getRawDataPointer();
auto lv1 = v1Up.getRawDataPointer();
auto numStages = coefficientsUp.size();
auto delayedStages = numStages / 2;
auto directStages = numStages - delayedStages;
// Processing
for (size_t i = 0; i < numSamples; i++)
{
// Direct path cascaded allpass filters
auto input = samples[i];
for (auto n = 0; n < directStages; n++)
{
auto alpha = coeffs[n];
auto output = alpha * input + lv1[n];
lv1[n] = input - alpha * output;
input = output;
}
// Output
bufferSamples[i << 1] = input;
// Delayed path cascaded allpass filters
input = samples[i];
for (auto n = directStages; n < numStages; n++)
{
auto alpha = coeffs[n];
auto output = alpha * input + lv1[n];
lv1[n] = input - alpha * output;
input = output;
}
// Output
bufferSamples[(i << 1) + 1] = input;
}
// Snap To Zero
snapToZero (true);
}
void processSamplesDown (SampleType *samples, size_t numSamples) override
{
// Initialization
auto bufferSamples = OversamplingEngine<SampleType>::buffer.getWritePointer (0);
auto coeffs = coefficientsDown.getRawDataPointer();
auto lv1 = v1Down.getRawDataPointer();
auto numStages = coefficientsDown.size();
auto delayedStages = numStages / 2;
auto directStages = numStages - delayedStages;
// Processing
for (size_t i = 0; i < numSamples; i++)
{
// Direct path cascaded allpass filters
auto input = bufferSamples[i << 1];
for (auto n = 0; n < directStages; n++)
{
auto alpha = coeffs[n];
auto output = alpha * input + lv1[n];
lv1[n] = input - alpha * output;
input = output;
}
auto directOut = input;
// Delayed path cascaded allpass filters
input = bufferSamples[(i << 1) + 1];
for (auto n = directStages; n < numStages; n++)
{
auto alpha = coeffs[n];
auto output = alpha * input + lv1[n];
lv1[n] = input - alpha * output;
input = output;
}
// Output
samples[i] = (delayDown + directOut) * static_cast<SampleType> (0.5);
delayDown = input;
}
// Snap To Zero
snapToZero (false);
}
void snapToZero (bool snapUpProcessing)
{
if (snapUpProcessing)
{
auto lv1 = v1Up.getRawDataPointer();
auto numStages = coefficientsUp.size();
for (auto n = 0; n < numStages; n++)
JUCE_SNAP_TO_ZERO (lv1[n]);
}
else
{
auto lv1 = v1Down.getRawDataPointer();
auto numStages = coefficientsDown.size();
for (auto n = 0; n < numStages; n++)
JUCE_SNAP_TO_ZERO (lv1[n]);
}
}
private:
//===============================================================================
/** This function calculates the equivalent high order IIR filter of a given
polyphase cascaded allpass filters structure.
*/
const dsp::IIR::Coefficients<SampleType> getCoefficients (typename dsp::FilterDesign<SampleType>::IIRPolyphaseAllpassStructure &structure) const
{
dsp::Polynomial<SampleType> numerator1 ({ static_cast<SampleType> (1.0) });
dsp::Polynomial<SampleType> denominator1 ({ static_cast<SampleType> (1.0) });
dsp::Polynomial<SampleType> numerator2 ({ static_cast<SampleType> (1.0) });
dsp::Polynomial<SampleType> denominator2 ({ static_cast<SampleType> (1.0) });
dsp::Polynomial<SampleType> temp;
for (auto n = 0; n < structure.directPath.size(); n++)
{
auto *coeffs = structure.directPath.getReference (n).getRawCoefficients();
if (structure.directPath[n].getFilterOrder() == 1)
{
temp = dsp::Polynomial<SampleType> ({ coeffs[0], coeffs[1] });
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;
Array<SampleType> v1Up, v1Down;
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 = 1 << newFactor;
isMaximumQuality = newMaxQuality;
type = newType;
numChannels = newNumChannels;
if (newFactor == 0)
{
for (size_t channel = 0; channel < numChannels; channel++)
engines.add (new OversamplingDummy<SampleType>());
numStages = 1;
}
else if (type == FilterType::filterHalfBandPolyphaseIIR)
{
numStages = newFactor;
for (size_t channel = 0; channel < numChannels; channel++)
for (size_t n = 0; n < numStages; n++)
{
auto tw1 = (isMaximumQuality ? 0.10f : 0.12f);
auto tw2 = (isMaximumQuality ? 0.12f : 0.15f);
engines.add (new Oversampling2TimesPolyphaseIIR<SampleType> (tw1, -75.f + 10.f * n, tw2, -70.f + 10.f * n));
}
}
else if (type == FilterType::filterHalfBandFIREquiripple)
{
numStages = newFactor;
for (size_t channel = 0; channel < numChannels; channel++)
for (size_t n = 0; n < numStages; n++)
{
auto tw1 = (isMaximumQuality ? 0.10f : 0.12f);
auto tw2 = (isMaximumQuality ? 0.12f : 0.15f);
engines.add (new Oversampling2TimesEquirippleFIR<SampleType> (tw1, -90.f + 10.f * n, tw2, -70.f + 10.f * n));
}
}
}
template <typename SampleType>
Oversampling<SampleType>::~Oversampling()
{
engines.clear();
}
//===============================================================================
template <typename SampleType>
SampleType Oversampling<SampleType>::getLatencyInSamples() noexcept
{
auto latency = static_cast<SampleType> (0);
auto order = 1;
for (size_t n = 0; n < numStages; n++)
{
auto& engine = *engines[static_cast<int> (n)];
order *= engine.getFactor();
latency += engine.getLatencyInSamples() / std::pow (static_cast<SampleType> (2), 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.size() > 0);
for (size_t channel = 0; channel < numChannels; channel++)
{
auto currentNumSamples = maximumNumberOfSamplesBeforeOversampling;
auto offset = numStages * channel;
for (size_t n = 0; n < numStages; n++)
{
auto& engine = *engines[static_cast<int> (n + offset)];
engine.initProcessing (currentNumSamples);
currentNumSamples *= engine.getFactor();
}
}
isReady = true;
reset();
}
template <typename SampleType>
void Oversampling<SampleType>::reset() noexcept
{
jassert (engines.size() > 0);
if (isReady)
for (auto n = 0; n < engines.size(); n++)
engines[n]->reset();
}
template <typename SampleType>
typename dsp::AudioBlock<SampleType> Oversampling<SampleType>::getProcessedSamples()
{
jassert (engines.size() > 0);
Array<SampleType*> arrayChannels;
for (size_t channel = 0; channel < numChannels; channel++)
arrayChannels.add (engines[static_cast<int> (((channel + 1) * numStages) - 1)]->getProcessedSamples());
auto numSamples = engines[static_cast<int> (numStages - 1)]->getNumProcessedSamples();
auto block = dsp::AudioBlock<SampleType> (arrayChannels.getRawDataPointer(), numChannels, numSamples);
return block;
}
template <typename SampleType>
void Oversampling<SampleType>::processSamplesUp (dsp::AudioBlock<SampleType> &block) noexcept
{
jassert (engines.size() > 0 && block.getNumChannels() <= numChannels);
if (! isReady)
return;
for (size_t channel = 0; channel < jmin (numChannels, block.getNumChannels()); channel++)
{
SampleType* dataSamples = block.getChannelPointer (channel);
auto currentNumSamples = block.getNumSamples();
auto offset = numStages * channel;
for (size_t n = 0; n < numStages; n++)
{
auto& engine = *engines[static_cast<int> (n + offset)];
engine.processSamplesUp (dataSamples, currentNumSamples);
currentNumSamples *= engine.getFactor();
dataSamples = engine.getProcessedSamples();
}
}
}
template <typename SampleType>
void Oversampling<SampleType>::processSamplesDown (dsp::AudioBlock<SampleType> &block) noexcept
{
jassert (engines.size() > 0 && block.getNumChannels() <= numChannels);
if (! isReady)
return;
for (size_t channel = 0; channel < jmin (numChannels, block.getNumChannels()); channel++)
{
auto currentNumSamples = block.getNumSamples();
auto offset = numStages * channel;
for (size_t n = 0; n < numStages - 1; n++)
currentNumSamples *= engines[static_cast<int> (n + offset)]->getFactor();
for (size_t n = numStages - 1; n > 0; n--)
{
auto& engine = *engines[static_cast<int> (n + offset)];
auto dataSamples = engines[static_cast<int> (n + offset - 1)]->getProcessedSamples();
engine.processSamplesDown (dataSamples, currentNumSamples);
currentNumSamples /= engine.getFactor();
}
engines[static_cast<int> (offset)]->processSamplesDown (block.getChannelPointer (channel), currentNumSamples);
}
}
template class Oversampling<float>;
template class Oversampling<double>;

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@ -0,0 +1,135 @@
/*
==============================================================================
This file is part of the JUCE library.
Copyright (c) 2017 - ROLI Ltd.
JUCE is an open source library subject to commercial or open-source
licensing.
By using JUCE, you agree to the terms of both the JUCE 5 End-User License
Agreement and JUCE 5 Privacy Policy (both updated and effective as of the
27th April 2017).
End User License Agreement: www.juce.com/juce-5-licence
Privacy Policy: www.juce.com/juce-5-privacy-policy
Or: You may also use this code under the terms of the GPL v3 (see
www.gnu.org/licenses).
JUCE IS PROVIDED "AS IS" WITHOUT ANY WARRANTY, AND ALL WARRANTIES, WHETHER
EXPRESSED OR IMPLIED, INCLUDING MERCHANTABILITY AND FITNESS FOR PURPOSE, ARE
DISCLAIMED.
==============================================================================
*/
template <typename NumericType>
class OversamplingEngine;
//===============================================================================
/**
A processing class performing multi-channel oversampling.
It can be configured to do 2 times, 4 times, 8 times or 16 times oversampling
using a multi-stage approach, either polyphase allpass IIR filters or FIR
filters for the filtering, and reports successfully the latency added by the
filter stages.
The principle of oversampling is to increase the sample rate of a given
non-linear process, to prevent it from creating aliasing. Oversampling works
by upsampling N times the input signal, processing the upsampling signal
with the increased internal sample rate, and downsample the result to get
back the original processing sample rate.
Choose between FIR or IIR filtering depending on your needs in term of
latency and phase distortion. With FIR filters, the phase is linear but the
latency is maximum. With IIR filtering, the phase is compromised around the
Nyquist frequency but the phase is minimum.
@see FilterDesign.
*/
template <typename SampleType>
class JUCE_API Oversampling
{
public:
/** The type of filter that can be used for the oversampling processing. */
enum FilterType
{
filterHalfBandFIREquiripple = 0,
filterHalfBandPolyphaseIIR,
numFilterTypes
};
//===============================================================================
/**
Constructor of the oversampling class. All the processing parameters must be
provided at the creation of the oversampling object.
Note : you might want to create a class heriting from Oversampling with a
different constructor if you need more control on what happens in the process.
@param numChannels the number of channels to process with this object
@param factor the processing will perform 2 ^ factor times oversampling
@param type the type of filter design employed for filtering during
oversampling
@param isMaxQuality if the oversampling is done using the maximum quality,
the filters will be more efficient, but the CPU load will
increase as well
*/
Oversampling (size_t numChannels, size_t factor, FilterType type, bool isMaxQuality = true);
/** Destructor. */
~Oversampling();
//===============================================================================
/** Returns the latency in samples of the whole processing. Use this information
in your main processor to compensate the additional latency involved with
the oversampling, for example with a dry / wet functionality, and to report
the latency to the DAW.
*/
SampleType getLatencyInSamples() noexcept;
/** Returns the current oversampling factor. */
size_t getOversamplingFactor() noexcept;
//===============================================================================
/** Must be called before any processing, to set the buffer sizes of the internal
buffers of the oversampling processing.
*/
void initProcessing (size_t maximumNumberOfSamplesBeforeOversampling);
/** Resets the processing pipeline, ready to oversample a new stream of data. */
void reset() noexcept;
/** Must be called to perform the upsampling, prior to any oversampled processing. */
void processSamplesUp (dsp::AudioBlock<SampleType> &block) noexcept;
/** Can be called to access to the oversampled input signal, to perform any non-
linear processing which needs the higher sample rate. Don't forget to set
the sample rate of that processing to N times the original sample rate.
*/
dsp::AudioBlock<SampleType> getProcessedSamples();
/** Must be called to perform the downsampling, after the upsampling and the
non-linear processing. The output signal is probably delayed by the internal
latency of the whole oversampling behaviour, so don't forget to take this
into account.
*/
void processSamplesDown (dsp::AudioBlock<SampleType> &block) noexcept;
private:
//===============================================================================
bool isMaximumQuality;
size_t factorOversampling, numStages;
FilterType type;
size_t numChannels;
//===============================================================================
bool isReady = false;
OwnedArray<OversamplingEngine<SampleType>> engines;
//===============================================================================
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (Oversampling)
};