bit of code clean-up

modules/juce_dsp/processors/juce_Oversampling.cpp

@@ -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());
 

modules/juce_dsp/processors/juce_Oversampling.h

@@ -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:
     //===============================================================================