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bit of code clean-up

tags/2021-05-28
jules 6 years ago
parent
38b553da43
commit
397b0f80a2
2 changed files with 63 additions and 69 deletions
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  1. +57
    -66
      modules/juce_dsp/processors/juce_Oversampling.cpp
  2. +6
    -3
      modules/juce_dsp/processors/juce_Oversampling.h

+ 57
- 66
modules/juce_dsp/processors/juce_Oversampling.cpp View File

@@ -63,8 +63,8 @@ public:
return dsp::AudioBlock<SampleType> (buffer).getSubBlock (0, numSamples);
}
virtual void processSamplesUp (dsp::AudioBlock<SampleType> &inputBlock) = 0;
virtual void processSamplesDown (dsp::AudioBlock<SampleType> &outputBlock) = 0;
virtual void processSamplesUp (dsp::AudioBlock<SampleType>&) = 0;
virtual void processSamplesDown (dsp::AudioBlock<SampleType>&) = 0;
protected:
//===============================================================================
@@ -88,10 +88,10 @@ public:
//===============================================================================
SampleType getLatencyInSamples() override
{
return 0.f;
return 0;
}
void processSamplesUp (dsp::AudioBlock<SampleType> &inputBlock) override
void processSamplesUp (dsp::AudioBlock<SampleType>& inputBlock) override
{
jassert (inputBlock.getNumChannels() <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumChannels()));
jassert (inputBlock.getNumSamples() * OversamplingEngine<SampleType>::factor <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumSamples()));
@@ -101,7 +101,7 @@ public:
inputBlock.getChannelPointer (channel), static_cast<int> (inputBlock.getNumSamples()));
}
void processSamplesDown (dsp::AudioBlock<SampleType> &outputBlock) override
void processSamplesDown (dsp::AudioBlock<SampleType>& outputBlock) override
{
jassert (outputBlock.getNumChannels() <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumChannels()));
jassert (outputBlock.getNumSamples() * OversamplingEngine<SampleType>::factor <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumSamples()));
@@ -109,8 +109,6 @@ public:
outputBlock.copy (OversamplingEngine<SampleType>::getProcessedSamples (outputBlock.getNumSamples()));
}
private:
//===============================================================================
JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (OversamplingDummy)
};
@@ -121,10 +119,9 @@ private:
leading to specific processing optimizations.
*/
template <typename SampleType>
class Oversampling2TimesEquirippleFIR : public OversamplingEngine<SampleType>
class Oversampling2TimesEquirippleFIR : public OversamplingEngine<SampleType>
{
public:
//===============================================================================
Oversampling2TimesEquirippleFIR (size_t numChans,
SampleType normalizedTransitionWidthUp,
SampleType stopbandAttenuationdBUp,
@@ -132,7 +129,7 @@ public:
SampleType stopbandAttenuationdBDown)
: OversamplingEngine<SampleType> (numChans, 2)
{
coefficientsUp = *dsp::FilterDesign<SampleType>::designFIRLowpassHalfBandEquirippleMethod (normalizedTransitionWidthUp, stopbandAttenuationdBUp);
coefficientsUp = *dsp::FilterDesign<SampleType>::designFIRLowpassHalfBandEquirippleMethod (normalizedTransitionWidthUp, stopbandAttenuationdBUp);
coefficientsDown = *dsp::FilterDesign<SampleType>::designFIRLowpassHalfBandEquirippleMethod (normalizedTransitionWidthDown, stopbandAttenuationdBDown);
auto N = coefficientsUp.getFilterOrder() + 1;
@@ -167,7 +164,7 @@ public:
position.fill (0);
}
void processSamplesUp (dsp::AudioBlock<SampleType> &inputBlock) override
void processSamplesUp (dsp::AudioBlock<SampleType>& inputBlock) override
{
jassert (inputBlock.getNumChannels() <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumChannels()));
jassert (inputBlock.getNumSamples() * OversamplingEngine<SampleType>::factor <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumSamples()));
@@ -192,6 +189,7 @@ public:
// 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];
@@ -206,7 +204,7 @@ public:
}
}
void processSamplesDown (dsp::AudioBlock<SampleType> &outputBlock) override
void processSamplesDown (dsp::AudioBlock<SampleType>& outputBlock) override
{
jassert (outputBlock.getNumChannels() <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumChannels()));
jassert (outputBlock.getNumSamples() * OversamplingEngine<SampleType>::factor <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumSamples()));
@@ -234,6 +232,7 @@ public:
// 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];
@@ -273,10 +272,9 @@ private:
phase, and provided with a method to get the exact resulting latency.
*/
template <typename SampleType>
class Oversampling2TimesPolyphaseIIR : public OversamplingEngine<SampleType>
class Oversampling2TimesPolyphaseIIR : public OversamplingEngine<SampleType>
{
public:
//===============================================================================
Oversampling2TimesPolyphaseIIR (size_t numChans,
SampleType normalizedTransitionWidthUp,
SampleType stopbandAttenuationdBUp,
@@ -292,14 +290,14 @@ public:
dsp::IIR::Coefficients<SampleType> coeffsDown = getCoefficients (structureDown);
latency += static_cast<SampleType> (-(coeffsDown.getPhaseForFrequency (0.0001, 1.0)) / (0.0001 * MathConstants<double>::twoPi));
for (auto i = 0; i < structureUp.directPath.size(); ++i)
coefficientsUp.add (structureUp.directPath[i].coefficients[0]);
for (auto& i : structureUp.directPath)
coefficientsUp.add (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 : structureDown.directPath)
coefficientsDown.add (i.coefficients[0]);
for (auto i = 1; i < structureDown.delayedPath.size(); ++i)
coefficientsDown.add (structureDown.delayedPath[i].coefficients[0]);
@@ -320,13 +318,12 @@ public:
void reset() override
{
OversamplingEngine<SampleType>::reset();
v1Up.clear();
v1Down.clear();
delayDown.fill (0);
}
void processSamplesUp (dsp::AudioBlock<SampleType> &inputBlock) override
void processSamplesUp (dsp::AudioBlock<SampleType>& inputBlock) override
{
jassert (inputBlock.getNumChannels() <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumChannels()));
jassert (inputBlock.getNumSamples() * OversamplingEngine<SampleType>::factor <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumSamples()));
@@ -349,6 +346,7 @@ public:
{
// Direct path cascaded allpass filters
auto input = samples[i];
for (auto n = 0; n < directStages; ++n)
{
auto alpha = coeffs[n];
@@ -362,6 +360,7 @@ public:
// Delayed path cascaded allpass filters
input = samples[i];
for (auto n = directStages; n < numStages; ++n)
{
auto alpha = coeffs[n];
@@ -379,7 +378,7 @@ public:
snapToZero (true);
}
void processSamplesDown (dsp::AudioBlock<SampleType> &outputBlock) override
void processSamplesDown (dsp::AudioBlock<SampleType>& outputBlock) override
{
jassert (outputBlock.getNumChannels() <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumChannels()));
jassert (outputBlock.getNumSamples() * OversamplingEngine<SampleType>::factor <= static_cast<size_t> (OversamplingEngine<SampleType>::buffer.getNumSamples()));
@@ -403,6 +402,7 @@ public:
{
// Direct path cascaded allpass filters
auto input = bufferSamples[i << 1];
for (auto n = 0; n < directStages; ++n)
{
auto alpha = coeffs[n];
@@ -410,10 +410,12 @@ public:
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];
@@ -467,12 +469,12 @@ private:
*/
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) });
constexpr auto one = static_cast<SampleType> (1.0);
dsp::Polynomial<SampleType> temp;
dsp::Polynomial<SampleType> numerator1 ({ one });
dsp::Polynomial<SampleType> denominator1 ({ one });
dsp::Polynomial<SampleType> numerator2 ({ one });
dsp::Polynomial<SampleType> denominator2 ({ one });
for (auto n = 0; n < structure.directPath.size(); ++n)
{
@@ -480,19 +482,13 @@ private:
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);
numerator1 = numerator1.getProductWith (dsp::Polynomial<SampleType> ({ coeffs[0], coeffs[1] }));
denominator1 = denominator1.getProductWith (dsp::Polynomial<SampleType> ({ one, coeffs[2] }));
}
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);
numerator1 = numerator1.getProductWith (dsp::Polynomial<SampleType> ({ coeffs[0], coeffs[1], coeffs[2] }));
denominator1 = denominator1.getProductWith (dsp::Polynomial<SampleType> ({ one, coeffs[3], coeffs[4] }));
}
}
@@ -502,31 +498,25 @@ private:
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);
numerator2 = numerator2.getProductWith (dsp::Polynomial<SampleType> ({ coeffs[0], coeffs[1] }));
denominator2 = denominator2.getProductWith (dsp::Polynomial<SampleType> ({ one, coeffs[2] }));
}
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);
numerator2 = numerator2.getProductWith (dsp::Polynomial<SampleType> ({ coeffs[0], coeffs[1], coeffs[2] }));
denominator2 = denominator2.getProductWith (dsp::Polynomial<SampleType> ({ one, coeffs[3], coeffs[4] }));
}
}
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);
auto numeratorf1 = numerator1.getProductWith (denominator2);
auto numeratorf2 = numerator2.getProductWith (denominator1);
auto numerator = numeratorf1.getSumWith (numeratorf2);
auto denominator = denominator1.getProductWith (denominator2);
dsp::IIR::Coefficients<SampleType> coeffs;
coeffs.coefficients.clear();
auto inversion = static_cast<SampleType> (1.0) / denominator[0];
auto inversion = one / denominator[0];
for (auto i = 0; i <= numerator.getOrder(); ++i)
coeffs.coefficients.add (numerator[i] * inversion);
@@ -551,7 +541,8 @@ private:
//===============================================================================
template <typename SampleType>
Oversampling<SampleType>::Oversampling (size_t newNumChannels, size_t newFactor, FilterType newType, bool newMaxQuality)
Oversampling<SampleType>::Oversampling (size_t newNumChannels, size_t newFactor,
FilterType newType, bool newMaxQuality)
{
jassert (newFactor >= 0 && newFactor <= 4 && newNumChannels > 0);
@@ -571,13 +562,13 @@ Oversampling<SampleType>::Oversampling (size_t newNumChannels, size_t 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 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 ? -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);
auto gaindBStartUp = (isMaximumQuality ? -75.0f : -65.0f);
auto gaindBStartDown = (isMaximumQuality ? -70.0f : -60.0f);
auto gaindBFactorUp = (isMaximumQuality ? 10.0f : 8.0f);
auto gaindBFactorDown = (isMaximumQuality ? 10.0f : 8.0f);
engines.add (new Oversampling2TimesPolyphaseIIR<SampleType> (numChannels,
twUp, gaindBStartUp + gaindBFactorUp * n,
@@ -590,13 +581,13 @@ Oversampling<SampleType>::Oversampling (size_t newNumChannels, size_t 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 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.f : -70.f);
auto gaindBStartDown = (isMaximumQuality ? -70.f : -60.f);
auto gaindBFactorUp = (isMaximumQuality ? 10.f : 8.f);
auto gaindBFactorDown = (isMaximumQuality ? 10.f : 8.f);
auto gaindBStartUp = (isMaximumQuality ? -90.0f : -70.0f);
auto gaindBStartDown = (isMaximumQuality ? -70.0f : -60.0f);
auto gaindBFactorUp = (isMaximumQuality ? 10.0f : 8.0f);
auto gaindBFactorDown = (isMaximumQuality ? 10.0f : 8.0f);
engines.add (new Oversampling2TimesEquirippleFIR<SampleType> (numChannels,
twUp, gaindBStartUp + gaindBFactorUp * n,
@@ -665,7 +656,7 @@ void Oversampling<SampleType>::reset() noexcept
}
template <typename SampleType>
typename dsp::AudioBlock<SampleType> Oversampling<SampleType>::processSamplesUp (const dsp::AudioBlock<SampleType> &inputBlock) noexcept
typename dsp::AudioBlock<SampleType> Oversampling<SampleType>::processSamplesUp (const dsp::AudioBlock<SampleType>& inputBlock) noexcept
{
jassert (! engines.isEmpty());
@@ -685,7 +676,7 @@ typename dsp::AudioBlock<SampleType> Oversampling<SampleType>::processSamplesUp
}
template <typename SampleType>
void Oversampling<SampleType>::processSamplesDown (dsp::AudioBlock<SampleType> &outputBlock) noexcept
void Oversampling<SampleType>::processSamplesDown (dsp::AudioBlock<SampleType>& outputBlock) noexcept
{
jassert (! engines.isEmpty());


+ 6
- 3
modules/juce_dsp/processors/juce_Oversampling.h View File

@@ -86,7 +86,10 @@ public:
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);
Oversampling (size_t numChannels,
size_t factor,
FilterType type,
bool isMaxQuality = true);
/** Destructor. */
~Oversampling();
@@ -121,14 +124,14 @@ public:
Don't forget to set the sample rate of that processing to N times the original
sample rate.
*/
dsp::AudioBlock<SampleType> processSamplesUp (const dsp::AudioBlock<SampleType> &inputBlock) noexcept;
dsp::AudioBlock<SampleType> processSamplesUp (const dsp::AudioBlock<SampleType>& inputBlock) noexcept;
/** 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> &outputBlock) noexcept;
void processSamplesDown (dsp::AudioBlock<SampleType>& outputBlock) noexcept;
private:
//===============================================================================


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