150 lines
3.9 KiB
C++
150 lines
3.9 KiB
C++
#include "PluginProcessor.h"
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#include "PluginEditor.h"
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#include <math.h>
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LampshadeAudioProcessor::LampshadeAudioProcessor()
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#ifndef JucePlugin_PreferredChannelConfigurations
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: AudioProcessor (BusesProperties()
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#if ! JucePlugin_IsMidiEffect
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#if ! JucePlugin_IsSynth
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.withInput ("Input", AudioChannelSet::stereo(), true)
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#endif
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.withOutput ("Output", AudioChannelSet::stereo(), true)
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#endif
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)
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#endif
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{
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}
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LampshadeAudioProcessor::~LampshadeAudioProcessor()
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{
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}
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const String LampshadeAudioProcessor::getName() const
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{
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return JucePlugin_Name;
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}
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bool LampshadeAudioProcessor::acceptsMidi() const
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{
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#if JucePlugin_WantsMidiInput
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return true;
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#else
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return false;
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#endif
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}
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bool LampshadeAudioProcessor::producesMidi() const
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{
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return false;
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}
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double LampshadeAudioProcessor::getTailLengthSeconds() const
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{
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return 0.0;
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}
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int LampshadeAudioProcessor::getNumPrograms()
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{
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return 1;
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}
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int LampshadeAudioProcessor::getCurrentProgram()
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{
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return 0;
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}
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void LampshadeAudioProcessor::setCurrentProgram (int index)
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{
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}
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const String LampshadeAudioProcessor::getProgramName (int index)
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{
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return {};
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}
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void LampshadeAudioProcessor::changeProgramName (int index, const String& newName)
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{
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}
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void LampshadeAudioProcessor::prepareToPlay (double sampleRate, int samplesPerBlock)
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{
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myYin = Yin(sampleRate, samplesPerBlock);
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aFilter = (filter_svf_t*)malloc(sizeof(filter_svf_t));
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svf_init(aFilter, 0, sampleRate);
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}
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void LampshadeAudioProcessor::releaseResources()
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{
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}
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#ifndef JucePlugin_PreferredChannelConfigurations
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bool LampshadeAudioProcessor::isBusesLayoutSupported (const BusesLayout& layouts) const
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{
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if (layouts.getMainOutputChannelSet() != AudioChannelSet::mono()
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&& layouts.getMainOutputChannelSet() != AudioChannelSet::stereo())
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return false;
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#if ! JucePlugin_IsSynth
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if (layouts.getMainOutputChannelSet() != layouts.getMainInputChannelSet())
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return false;
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#endif
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return true;
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}
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#endif
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void LampshadeAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
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{
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static float incrementer;
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const int totalNumInputChannels = getTotalNumInputChannels();
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const int totalNumOutputChannels = getTotalNumOutputChannels();
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// In case we have more outputs than inputs, this code clears any output
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// channels that didn't contain input data, (because these aren't
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// guaranteed to be empty - they may contain garbage).
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// This is here to avoid people getting screaming feedback
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// when they first compile a plugin, but obviously you don't need to keep
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// this code if your algorithm always overwrites all the output channels.
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for (int i = totalNumInputChannels; i < totalNumOutputChannels; ++i)
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buffer.clear(i, 0, buffer.getNumSamples());
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float pitch = myYin.getPitch(buffer.getWritePointer (0)); // returns Pitch in Hertz
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static float myPitch;
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if(pitch > 0) { myPitch = pitch; }
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svf_set_freq(aFilter, myPitch);
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float* channel1Data = buffer.getWritePointer (0);
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float* channel2Data = buffer.getWritePointer (1);
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for(int i=0; i < buffer.getNumSamples(); i++){
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incrementer += (myPitch / 4000);
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channel1Data[i] = svf_step(aFilter, channel1Data[i]);
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channel2Data[i] = channel1Data[i];
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}
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printf("%f\n",channel1Data[0]);
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}
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bool LampshadeAudioProcessor::hasEditor() const
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{
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return true;
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}
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AudioProcessorEditor* LampshadeAudioProcessor::createEditor()
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{
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return new LampshadeAudioProcessorEditor(*this);
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}
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void LampshadeAudioProcessor::getStateInformation (MemoryBlock& destData)
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{
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}
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void LampshadeAudioProcessor::setStateInformation (const void* data, int sizeInBytes)
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{
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}
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AudioProcessor* JUCE_CALLTYPE createPluginFilter()
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{
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return new LampshadeAudioProcessor();
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}
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