Downsample after upsample. Sounds nasty! Too nasty to be the default.…

… Keep for posterity
tote-bag-labs · Jul 24, 2023 · ac51a8d · ac51a8d
1 parent 50200d2
commit ac51a8d
Show file tree

Hide file tree

Showing 3 changed files with 34 additions and 3 deletions.
diff --git a/libs/tote_bag/dsp/DigiDegraders.cpp b/libs/tote_bag/dsp/DigiDegraders.cpp
@@ -55,6 +55,19 @@ void SimpleZOH::processBlock (juce::AudioBuffer<float>& inAudio,
     }
 }
 
+void SimpleZOH::processChannel (float* buffer, size_t samplesPerBlock)
+{
+    auto intDownSampleCoeff = static_cast<int> (downsampleCoeff);
+    auto frac = downsampleCoeff - intDownSampleCoeff;
+
+    for (int sample = 0; sample < samplesPerBlock; ++sample)
+    {
+        auto y1 = getZOHSample (buffer, sample, intDownSampleCoeff);
+        auto y2 = getZOHSample (buffer, sample, intDownSampleCoeff + 1);
+        buffer[sample] = tote_bag::audio_helpers::linearInterp (y1, y2, frac);
+    }
+}
+
 inline float
     SimpleZOH::getZOHSample (const float* channelData, int sampleIndex, int dsCoef)
 {

diff --git a/libs/tote_bag/dsp/DigiDegraders.h b/libs/tote_bag/dsp/DigiDegraders.h
@@ -30,6 +30,7 @@ class SimpleZOH
     void processBlock (juce::AudioBuffer<float>& inAudio,
                        int samplesPerBlock,
                        int numChannels);
+    void processChannel (float* buffer, size_t samplesPerBlock);
     float getZOHSample (const float* channelData, int sampleIndex, int dsCoef);
 
 private:

diff --git a/src/PluginProcessor.cpp b/src/PluginProcessor.cpp
@@ -243,7 +243,9 @@ void ValentineAudioProcessor::prepareToPlay (double sampleRate, int samplesPerBl
         FFCompParameterID()[getParameterIndex (VParameter::attack)]));
     ffCompressor->setRelease (*treeState.getRawParameterValue (
         FFCompParameterID()[getParameterIndex (VParameter::release)]));
-    ffCompressor->setSampleRate (sampleRate * oversampleMultiplier);
+
+    const auto processingSampleRate = sampleRate * oversampleMultiplier;
+    ffCompressor->setSampleRate (processingSampleRate);
     ffCompressor->reset (samplesPerBlock * oversampleMultiplier);
     ffCompressor->setOversampleMultiplier (oversampleMultiplier);
 
@@ -252,7 +254,8 @@ void ValentineAudioProcessor::prepareToPlay (double sampleRate, int samplesPerBl
 
     saturator->reset (sampleRate);
     boundedSaturator->reset (sampleRate);
-    simpleZOH->setParams (static_cast<float> (sampleRate / detail::kDownSampleRate));
+    simpleZOH->setParams (
+        static_cast<float> (processingSampleRate / detail::kDownSampleRate));
 
     updateLatencyCompensation (true);
 
@@ -362,12 +365,26 @@ void ValentineAudioProcessor::processBlock (juce::AudioBuffer<float>& buffer,
     juce::dsp::AudioBlock<float> highSampleRateBlock =
         oversampler->processSamplesUp (processBlock);
 
+    if (crushOn.get())
+    {
+        for (size_t channel = 0; channel < highSampleRateBlock.getNumChannels();
+             ++channel)
+        {
+            simpleZOH->processChannel (highSampleRateBlock.getChannelPointer (channel),
+                                       highSampleRateBlock.getNumSamples());
+        }
+
+        // simpleZOH->processBlock (processBuffer,
+        //                          currentSamplesPerBlock,
+        //                          totalNumOutputChannels);
+    }
+
     ffCompressor->process (highSampleRateBlock);
 
     if (saturateOn.get())
     {
         // Clear the buffers if saturate just got turned back on
-        if(!saturateOnState)
+        if (!saturateOnState)
         {
             saturator->clearBuffers();
             saturateOnState = true;