AnalogTapeModel

commit 6d2a46aa1be134f957422d5283ea7839081ae063
parent 4477d4be55a53af00a4093beaa15b146082335c2
Author: jatinchowdhury18 <jatinchowdhury18@users.noreply.github.com>
Date:   Sun, 24 Feb 2019 00:51:33 -0800

reduce CPU load and fix oversampling

Diffstat:
M
Plugin/Source/PluginProcessor.cpp
 | 
2
+-
M
Plugin/Source/Processors/Hysteresis/HysteresisProcessor.cpp
 | 
92
+++++++++++++++++++++++++++++++++++++++++++++++--------------------------------
M
Plugin/Source/Processors/Hysteresis/HysteresisProcessor.h
 | 
9
++++++---

3 files changed, 62 insertions(+), 41 deletions(-)
diff --git a/Plugin/Source/PluginProcessor.cpp b/Plugin/Source/PluginProcessor.cpp
@@ -22,7 +22,7 @@ ChowtapeModelAudioProcessor::ChowtapeModelAudioProcessor()
     outGain->addListener (this);
 
     addParameter (overSampling = new AudioParameterChoice (String ("overSampling"), String ("Upsample"),
-                                                           StringArray ({ "4x", "8x", "16x" }), 0));
+                                                           StringArray ({ "2x", "4x", "8x" }), 0));
     overSampling->addListener (this);
 
     addParameter (tapeSpeed = new AudioParameterChoice (String ("tapeSpeed"), String ("Speed"),
diff --git a/Plugin/Source/Processors/Hysteresis/HysteresisProcessor.cpp b/Plugin/Source/Processors/Hysteresis/HysteresisProcessor.cpp
@@ -1,14 +1,10 @@
 #include "HysteresisProcessor.h"
 
-namespace
-{
-    constexpr float biasFilterFreq = 24000.0f;
-}
-
 HysteresisProcessor::HysteresisProcessor() : ProcessorBase ("HysteresisProcessor")
 {
-    overSample.reset (new dsp::Oversampling<float> (2, 1, dsp::Oversampling<float>::filterHalfBandFIREquiripple));
-    biasFilter.setFreq (biasFilterFreq);
+    overSample2.reset (new dsp::Oversampling<float> (2, 1, dsp::Oversampling<float>::filterHalfBandFIREquiripple));
+    overSample4.reset (new dsp::Oversampling<float> (2, 2, dsp::Oversampling<float>::filterHalfBandFIREquiripple));
+    overSample8.reset (new dsp::Oversampling<float> (2, 3, dsp::Oversampling<float>::filterHalfBandFIREquiripple));
 }
 
 void HysteresisProcessor::prepareToPlay (double sampleRate, int samplesPerBlock)
@@ -18,25 +14,43 @@ void HysteresisProcessor::prepareToPlay (double sampleRate, int samplesPerBlock)
     hProcs[0].setSampleRate ((float) (sampleRate * overSamplingFactor));
     hProcs[1].setSampleRate ((float) (sampleRate * overSamplingFactor));
 
-    overSample->initProcessing (samplesPerBlock);
-
-    biasFilter.prepareToPlay (sampleRate * overSamplingFactor, samplesPerBlock * overSamplingFactor);    
+    if (overSamplingFactor == 8)
+        overSample8->initProcessing (samplesPerBlock);
+    else if (overSamplingFactor == 4)
+        overSample4->initProcessing (samplesPerBlock);
+    else
+        overSample2->initProcessing (samplesPerBlock);
 
     n[0] = 0;
     n[1] = 0;
+
+    fadeIn = true;
 }
 
 void HysteresisProcessor::releaseResources()
 {
-    overSample->reset();
-
-    biasFilter.releaseResources();
+    overSample2->reset();
+    overSample4->reset();
+    overSample8->reset();
 }
 
-void HysteresisProcessor::processBlock (AudioBuffer<float>& buffer, MidiBuffer& midi)
+void HysteresisProcessor::processBlock (AudioBuffer<float>& buffer, MidiBuffer& /*midi*/)
 {
+    if (fadeIn)
+    {
+        buffer.applyGainRamp (0, buffer.getNumSamples(), 0.0f, 1.0f);
+        fadeIn = false;
+    }
+
     dsp::AudioBlock<float> block (buffer);
-    dsp::AudioBlock<float> osBlock = overSample->processSamplesUp(block);
+    dsp::AudioBlock<float> osBlock;
+
+    if (overSamplingFactor == 8)
+        osBlock = overSample8->processSamplesUp (block);
+    else if (overSamplingFactor == 4)
+        osBlock = overSample4->processSamplesUp (block);
+    else
+        osBlock = overSample2->processSamplesUp (block);
 
     float* ptrArray[] = { osBlock.getChannelPointer(0), osBlock.getChannelPointer(1) };
     AudioBuffer<float> osBuffer (ptrArray, 2, static_cast<int> (osBlock.getNumSamples()));
@@ -48,44 +62,38 @@ void HysteresisProcessor::processBlock (AudioBuffer<float>& buffer, MidiBuffer& 
         auto* x = osBuffer.getWritePointer (channel);
         for (int samp = 0; samp < osBuffer.getNumSamples(); samp++)
         {
-            x[samp] = hProcs[channel].process ((float) 1e5 * (x[samp]));// + biasGain * sinf (sineTerm * (float) n[channel])));
+            x[samp] = hProcs[channel].process ((float) 5e4 * (x[samp]));// + biasGain * sinf (sineTerm * (float) n[channel])));
             
             n[channel]++;
             if ((float) (n[channel] / (getSampleRate() * overSamplingFactor)) >= 1.0f / biasFreq)
                 n[channel] = 0;
         }
     }
-    biasFilter.processBlock (osBuffer, midi);
 
-    overSample->processSamplesDown(block);
+    if (overSamplingFactor == 8)
+        overSample8->processSamplesDown (block);
+    else if (overSamplingFactor == 4)
+        overSample4->processSamplesDown (block);
+    else
+        overSample2->processSamplesDown (block);
 }
 
 void HysteresisProcessor::setOverSamplingFactor (String osFactor)
 {
-    osFactor = "4x";
-    //@TODO: figure out how to change oversampling factor without breaking everything
-
-    int factor = overSamplingFactor;
-    
-    if (osFactor == "4x")
-    {
+    auto factor = overSamplingFactor;   
+    if (osFactor == "2x")
+        factor = 2; // set overSample factor 2 = 2^1
+    else if (osFactor == "4x")
         factor = 4; // set overSample factor 4 = 2^2
-        overSample.reset (new dsp::Oversampling<float> (2, 2, dsp::Oversampling<float>::filterHalfBandFIREquiripple));
-    }
     else if (osFactor == "8x")
-    {
         factor = 8; // set overSample factor 8 = 2^3
-        overSample.reset (new dsp::Oversampling<float> (2, 3, dsp::Oversampling<float>::filterHalfBandFIREquiripple));
-    }
-    else if (osFactor == "16x")
+
+    if (factor != overSamplingFactor)
     {
-        factor = 16; // set overSample factor 16 = 2^4
-        overSample.reset (new dsp::Oversampling<float> (2, 4, dsp::Oversampling<float>::filterHalfBandFIREquiripple));
+        overSamplingFactor = factor;
+        releaseResources();
+        prepareToPlay (getSampleRate(), getBlockSize());
     }
-
-    overSamplingFactor = factor;
-    overSample->reset();
-    overSample->initProcessing (getBlockSize());
 }
 
 void HysteresisProcessor::setBiasFreq (float newFreqKHz)
@@ -97,3 +105,13 @@ void HysteresisProcessor::setBiasGain (float newGainDB)
 {
     biasGain = Decibels::decibelsToGain (newGainDB);
 }
+
+double HysteresisProcessor::getTailLengthSeconds() const
+{
+    if (overSamplingFactor == 8)
+        return overSample8->getLatencyInSamples() * getSampleRate();
+    else if (overSamplingFactor == 4)
+        return overSample4->getLatencyInSamples() * getSampleRate();
+    else
+        return overSample2->getLatencyInSamples() * getSampleRate();
+}
diff --git a/Plugin/Source/Processors/Hysteresis/HysteresisProcessor.h b/Plugin/Source/Processors/Hysteresis/HysteresisProcessor.h
@@ -14,7 +14,7 @@ public:
     void releaseResources() override;
     void processBlock (AudioBuffer<float>& buffer, MidiBuffer& midiBuffer) override;
 
-    double getTailLengthSeconds() const override { return overSample->getLatencyInSamples() * getSampleRate(); }
+    double getTailLengthSeconds() const override;
 
     void setOverSamplingFactor (String osFactor);
     void setBiasFreq (float newFreqKHz);
@@ -22,8 +22,9 @@ public:
 
 private:
     HysteresisProcessing hProcs[2];
-    std::unique_ptr<dsp::Oversampling<float>> overSample;
-    BiasFilter biasFilter;
+    std::unique_ptr<dsp::Oversampling<float>> overSample2;
+    std::unique_ptr<dsp::Oversampling<float>> overSample4;
+    std::unique_ptr<dsp::Oversampling<float>> overSample8;
 
     int overSamplingFactor = 8;
 
@@ -31,6 +32,8 @@ private:
     float biasGain = 5.0f;
     int n[2] = { 0, 0 };
 
+    bool fadeIn = false;
+
     JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (HysteresisProcessor)
 };