PluginProcessor.cpp

#include "PluginProcessor.h"
#include "PluginEditor.h"



float defaultGain = -60.0f;
float defaultFrequency = 440.0f;
float defaultVirtualFilter = 1.0f;
bool defaultUseMidiMode = false;

juce::AudioProcessorValueTreeState::ParameterLayout parameters() 
{
    std::vector<std::unique_ptr<juce::RangedAudioParameter>> parameter_list;

    // gain parameter from -60 dB to 0 dB
    //// idk what's the parameters here tho
    parameter_list.push_back(std::make_unique<juce::AudioParameterFloat>(
        juce::ParameterID{"gain", 1}, "Gain", -60.0, 0.0, defaultGain));
    parameter_list.push_back(std::make_unique<juce::AudioParameterFloat>(
        juce::ParameterID{"frequency", 2}, "Frequency", 20.0, 20000.0, defaultFrequency));
    parameter_list.push_back(std::make_unique<juce::AudioParameterFloat>(
        juce::ParameterID{"virtualFilter", 3}, "Virtual Filter", 0.0, 1.0, defaultVirtualFilter));
    parameter_list.push_back(std::make_unique<juce::AudioParameterBool>(
        juce::ParameterID{"useMidiMode", 4}, "Use MIDI Input", defaultUseMidiMode));
    return {parameter_list.begin(), parameter_list.end()};
}



//==============================================================================
AudioPluginAudioProcessor::AudioPluginAudioProcessor()
    : AudioProcessor (BusesProperties()
                    #if ! JucePlugin_IsMidiEffect
                        #if ! JucePlugin_IsSynth
                            .withInput ("Input",  juce::AudioChannelSet::stereo(), true)
                        #endif
                        .withOutput ("Output", juce::AudioChannelSet::stereo(), true)
                    #endif 
                    ),
                    // initialize APVTS
                    apvts(*this, nullptr, "Parameters", parameters())
{
}

AudioPluginAudioProcessor::~AudioPluginAudioProcessor()
{
}

//==============================================================================
const juce::String AudioPluginAudioProcessor::getName() const
{
    return JucePlugin_Name;
}

bool AudioPluginAudioProcessor::acceptsMidi() const
{
   #if JucePlugin_WantsMidiInput
    return true;
   #else
    return false;
   #endif
}

bool AudioPluginAudioProcessor::producesMidi() const
{
   #if JucePlugin_ProducesMidiOutput
    return true;
   #else
    return false;
   #endif
}

bool AudioPluginAudioProcessor::isMidiEffect() const
{
   #if JucePlugin_IsMidiEffect
    return true;
   #else
    return false;
   #endif
}

double AudioPluginAudioProcessor::getTailLengthSeconds() const
{
    return 0.0;
}

int AudioPluginAudioProcessor::getNumPrograms()
{
    return 1;   // NB: some hosts don't cope very well if you tell them there are 0 programs,
                // so this should be at least 1, even if you're not really implementing programs.
}

int AudioPluginAudioProcessor::getCurrentProgram()
{
    return 0;
}

void AudioPluginAudioProcessor::setCurrentProgram (int index)
{
    juce::ignoreUnused (index);
}

const juce::String AudioPluginAudioProcessor::getProgramName (int index)
{
    juce::ignoreUnused (index);
    return {};
}

void AudioPluginAudioProcessor::changeProgramName (int index, const juce::String& newName)
{
    juce::ignoreUnused (index, newName);
}

//==============================================================================
void AudioPluginAudioProcessor::prepareToPlay (double sampleRate, int samplesPerBlock)
{
    // Use this method as the place to do any pre-playback
    // initialisation that you need..
    juce::ignoreUnused (sampleRate, samplesPerBlock);
    // _phasor.setFrequency(defaultFrequency, sampleRate);
    _soundWave = new fx::SchoffhauzerSawtooth(defaultFrequency, sampleRate);
}

void AudioPluginAudioProcessor::releaseResources()
{
    // When playback stops, you can use this as an opportunity to free up any
    // spare memory, etc.
    delete _soundWave;
    _soundWave = nullptr;
}

bool AudioPluginAudioProcessor::isBusesLayoutSupported (const BusesLayout& layouts) const
{
  #if JucePlugin_IsMidiEffect
    juce::ignoreUnused (layouts);
    return true;
  #else
    // This is the place where you check if the layout is supported.
    // In this template code we only support mono or stereo.
    // Some plugin hosts, such as certain GarageBand versions, will only
    // load plugins that support stereo bus layouts.
    if (layouts.getMainOutputChannelSet() != juce::AudioChannelSet::mono()
     && layouts.getMainOutputChannelSet() != juce::AudioChannelSet::stereo())
        return false;

    // This checks if the input layout matches the output layout
   #if ! JucePlugin_IsSynth
    if (layouts.getMainOutputChannelSet() != layouts.getMainInputChannelSet())
        return false;
   #endif

    return true;
  #endif
}

void AudioPluginAudioProcessor::processBlock (juce::AudioBuffer<float>& buffer,
                                              juce::MidiBuffer& midiMessages)
{
    // juce::ignoreUnused (midiMessages);

    juce::ScopedNoDenormals noDenormals;
    auto totalNumInputChannels  = getTotalNumInputChannels();
    auto totalNumOutputChannels = getTotalNumOutputChannels();

    // In case we have more outputs than inputs, this code clears any output
    // channels that didn't contain input data, (because these aren't
    // guaranteed to be empty - they may contain garbage).
    // This is here to avoid people getting screaming feedback
    // when they first compile a plugin, but obviously you don't need to keep
    // this code if your algorithm always overwrites all the output channels.
    for (auto i = totalNumInputChannels; i < totalNumOutputChannels; ++i)
        buffer.clear (i, 0, buffer.getNumSamples());

    // This is the place where you'd normally do the guts of your plugin's
    // audio processing...
    // Make sure to reset the state if your inner loop is processing
    // the samples and the outer loop is handling the channels.
    // Alternatively, you can process the samples with the channels
    // interleaved by keeping the same state.




    float _gain = apvts.getParameter("gain")->getValue();
    float _frequency = apvts.getParameter("frequency")->getValue();
    useMidiInput = apvts.getParameter("useMidiMode")->getValue() > 0.5f;

    // 0.0 to 1.0
    // _gain parameter was normalized, so we need to convert it back to actual gain value
    // 0-1 parameter to -60 dB to 0 dB
    // then dB to amplitude 0-1 (logarithmically)
    _gain = fx::dbtoa(fx::map(_gain, 0.0f, 1.0f, -60.0f, 0.0f)); // -60 dB to 0 dB

    (*_soundWave).setCutoff(apvts.getParameter("virtualFilter")->getValue());

    // Process MIDI messages if MIDI mode is active
    if (useMidiInput)
    {
        keyboardState.processNextMidiBuffer(midiMessages, 0, buffer.getNumSamples(), true);
        
        for (auto metadata : midiMessages)
        {
            auto message = metadata.getMessage();
            if (message.isNoteOn())
            {
                int noteNumber = message.getNoteNumber();
                // Convert MIDI note number to frequency using mtof
                float midiFrequency = fx::mtof(static_cast<float>(noteNumber));
                midiNotes[noteNumber] = true;
                isMidiPlaying = true;
                (*_soundWave).setFrequency(midiFrequency, getSampleRate());
                (*_soundWave).setCutoff(apvts.getParameter("virtualFilter")->getValue());
            }
            else if (message.isNoteOff())
            {
                int noteNumber = message.getNoteNumber();
                midiNotes[noteNumber] = false;
                isMidiPlaying = false;
            }
        }
    }
    else 
    {
        // set frequency
        float mappedFrequency = fx::map(_frequency, 0.0f, 1.0f, 20.0f, 20000.0f);
        (*_soundWave).setFrequency(mappedFrequency, getSampleRate());

    }

    // allocate array
    float *_buffer = new float[buffer.getNumSamples()];

    if (useMidiInput)
    {
        if (isMidiPlaying)
        {
            // Play the last active MIDI note
            for (int sample = 0; sample < buffer.getNumSamples(); ++sample)
            {
                _buffer[sample] = (*_soundWave)();
            }
        }
        else
            // No active notes in MIDI mode
            std::fill(_buffer, _buffer + buffer.getNumSamples(), 0.0f);
    }
    else
    {
        // Use slider frequency
        for (int sample = 0; sample < buffer.getNumSamples(); ++sample)
        {
            _buffer[sample] = (*_soundWave)();
        }
    }

    // for all channels
    for (int channel = 0; channel < totalNumInputChannels; ++channel)
    {
        auto* channelData = buffer.getWritePointer (channel);
        juce::ignoreUnused (channelData);
        
        for (int sample = 0; sample < buffer.getNumSamples(); ++sample)
        {
            // assign sine waves sample value to channel buffer
            // apply gain
            channelData[sample] = _buffer[sample] * _gain;
        }
    }

    delete[] _buffer;
}



//==============================================================================
bool AudioPluginAudioProcessor::hasEditor() const
{
    return true; // (change this to false if you choose to not supply an editor)
}

juce::AudioProcessorEditor* AudioPluginAudioProcessor::createEditor()
{
    return new AudioPluginAudioProcessorEditor (*this);
}

//==============================================================================
void AudioPluginAudioProcessor::getStateInformation (juce::MemoryBlock& destData)
{
    // You should use this method to store your parameters in the memory block.
    // You could do that either as raw data, or use the XML or ValueTree classes
    // as intermediaries to make it easy to save and load complex data.
    juce::ignoreUnused (destData);
}

void AudioPluginAudioProcessor::setStateInformation (const void* data, int sizeInBytes)
{
    // You should use this method to restore your parameters from this memory block,
    // whose contents will have been created by the getStateInformation() call.
    juce::ignoreUnused (data, sizeInBytes);
}

//==============================================================================
// This creates new instances of the plugin..
juce::AudioProcessor* JUCE_CALLTYPE createPluginFilter()
{
    return new AudioPluginAudioProcessor();
}