view effects/chorus/Source/PluginProcessor.cpp @ 0:e32fe563e124

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/*
  This code accompanies the textbook:
 
  Digital Audio Effects: Theory, Implementation and Application
  Joshua D. Reiss and Andrew P. McPherson
 
  ---
 
  Chorus: chorus effect based on time-varying delays
  See textbook Chapter 2: Delay Line Effects
 
  Code by Andrew McPherson, Brecht de Man and Joshua Reiss
 
  ---

  This program is free software: you can redistribute it and/or modify
  it under the terms of the GNU General Public License as published by
  the Free Software Foundation, either version 3 of the License, or
  (at your option) any later version.
 
  This program is distributed in the hope that it will be useful,
  but WITHOUT ANY WARRANTY; without even the implied warranty of
  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  GNU General Public License for more details.

  You should have received a copy of the GNU General Public License
  along with this program.  If not, see <http://www.gnu.org/licenses/>.
*/

#include "PluginProcessor.h"
#include "PluginEditor.h"
#include <math.h>

const float ChorusAudioProcessor::kMaximumDelay = 0.05;
const float ChorusAudioProcessor::kMaximumSweepWidth = 0.05;

//==============================================================================
ChorusAudioProcessor::ChorusAudioProcessor() : delayBuffer_ (2, 1)
{
    // Set default values:
    delay_ = .03;
    sweepWidth_ = .02;
    depth_ = 1.0;
    frequency_ = 0.2;
    waveform_ = kWaveformSine;
    interpolation_ = kInterpolationLinear;
    numVoices_ = 2;
    stereo_ = 0;
    
    delayBufferLength_ = 1;
    lfoPhase_ = 0.0;
    inverseSampleRate_ = 1.0/44100.0;
    
    // Start the circular buffer pointer at the beginning
    delayWritePosition_ = 0;
    
    lastUIWidth_ = 550;
    lastUIHeight_ = 200;
}

ChorusAudioProcessor::~ChorusAudioProcessor()
{
}

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

int ChorusAudioProcessor::getNumParameters()
{
    return kNumParameters;
}

float ChorusAudioProcessor::getParameter (int index)
{
    // This method will be called by the host, probably on the audio thread, so
    // it's absolutely time-critical. Don't use critical sections or anything
    // UI-related, or anything at all that may block in any way!
    switch (index)
    {
        case kDelayParam:         return delay_;
        case kSweepWidthParam:    return sweepWidth_;
        case kDepthParam:         return depth_;
        case kFrequencyParam:     return frequency_;
        case kWaveformParam:      return (float)waveform_;
        case kInterpolationParam: return (float)interpolation_;
        case kNumVoicesParam:     return (float)numVoices_;
        case kStereoParam:        return (float)stereo_;
        default:                  return 0.0f;
    }
}

void ChorusAudioProcessor::setParameter (int index, float newValue)
{
    // This method will be called by the host, probably on the audio thread, so
    // it's absolutely time-critical. Don't use critical sections or anything
    // UI-related, or anything at all that may block in any way!

    switch (index)
    {
        case kDelayParam:
            delay_ = newValue;
            break;
        case kSweepWidthParam:
            sweepWidth_ = newValue;
            break;
        case kDepthParam:
            depth_ = newValue;
            break;
        case kFrequencyParam:
            frequency_ = newValue;
            break;
        case kWaveformParam:
            waveform_ = (int)newValue;
            break;
        case kInterpolationParam:
            interpolation_ = (int)newValue;
            break;
        case kNumVoicesParam:
            numVoices_ = (int)newValue;
            break;
        case kStereoParam:
            stereo_ = (int)newValue;
            break;
        default:
            break;
    }
}

const String ChorusAudioProcessor::getParameterName (int index)
{
    switch (index)
    {
        case kDelayParam:         return "delay";
        case kSweepWidthParam:    return "sweep width";
        case kDepthParam:         return "depth";
        case kFrequencyParam:     return "frequency";
        case kWaveformParam:      return "waveform";
        case kInterpolationParam: return "interpolation";
        case kNumVoicesParam:     return "number of voices";
        case kStereoParam:        return "stereo";
        default:                  break;
    }
    
    return String::empty;
}

const String ChorusAudioProcessor::getParameterText (int index)
{
    return String (getParameter (index), 2);
}

const String ChorusAudioProcessor::getInputChannelName (int channelIndex) const
{
    return String (channelIndex + 1);
}

const String ChorusAudioProcessor::getOutputChannelName (int channelIndex) const
{
    return String (channelIndex + 1);
}

bool ChorusAudioProcessor::isInputChannelStereoPair (int index) const
{
    return true;
}

bool ChorusAudioProcessor::isOutputChannelStereoPair (int index) const
{
    return true;
}

bool ChorusAudioProcessor::silenceInProducesSilenceOut() const
{
#if JucePlugin_SilenceInProducesSilenceOut
    return true;
#else
    return false;
#endif
}

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

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

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

int ChorusAudioProcessor::getNumPrograms()
{
    return 0;
}

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

void ChorusAudioProcessor::setCurrentProgram (int index)
{
}

const String ChorusAudioProcessor::getProgramName (int index)
{
    return String::empty;
}

void ChorusAudioProcessor::changeProgramName (int index, const String& newName)
{
}

//==============================================================================
void ChorusAudioProcessor::prepareToPlay (double sampleRate, int samplesPerBlock)
{
    // Allocate and zero the delay buffer (size will depend on current sample rate)
    // Add 3 extra samples to allow cubic interpolation even at maximum delay
    delayBufferLength_ = (int)((kMaximumDelay + kMaximumSweepWidth)*sampleRate) + 3;
    delayBuffer_.setSize(2, delayBufferLength_);
    delayBuffer_.clear();
    lfoPhase_ = 0.0;
    
    inverseSampleRate_ = 1.0/sampleRate;
}

void ChorusAudioProcessor::releaseResources()
{
    // When playback stops, you can use this as an opportunity to free up any
    // spare memory, etc.
    
    // The delay buffer will stay in memory until the effect is unloaded.
}

void ChorusAudioProcessor::reset()
{
    // Use this method as the place to clear any delay lines, buffers, etc, as it
    // means there's been a break in the audio's continuity.
    
    delayBuffer_.clear();
}


void ChorusAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
{
    // Helpful information about this block of samples:
    const int numInputChannels = getNumInputChannels();     // How many input channels for our effect?
    const int numOutputChannels = getNumOutputChannels();   // How many output channels for our effect?
    const int numSamples = buffer.getNumSamples();          // How many samples in the buffer for this block?
    
    int channel, dpw; // dpr = delay read pointer; dpw = delay write pointer
    float dpr, currentDelay, ph;
    
    // Go through each channel of audio that's passed in. In this example we apply identical
    // effects to each channel, regardless of how many input channels there are. For some effects, like
    // a stereo chorus or panner, you might do something different for each channel.
    
    for (channel = 0; channel < numInputChannels; ++channel)
    {
        // channelData is an array of length numSamples which contains the audio for one channel
        float* channelData = buffer.getSampleData(channel);
        
        // delayData is the circular buffer for implementing delay on this channel
        float* delayData = delayBuffer_.getSampleData (jmin (channel, delayBuffer_.getNumChannels() - 1));
        
        // Make a temporary copy of any state variables declared in PluginProcessor.h which need to be
        // maintained between calls to processBlock(). Each channel needs to be processed identically
        // which means that the activity of processing one channel can't affect the state variable for
        // the next channel.
        
        dpw = delayWritePosition_;
        ph = lfoPhase_;
        
        for (int i = 0; i < numSamples; ++i)
        {
            const float in = channelData[i];
            float interpolatedSample = 0.0;
            float phaseOffset = 0.0;
            float weight;
            
            // Chorus can have more than 2 voices (where the original, undelayed signal counts as a voice).
            // In this implementation, all voices use the same LFO, but with different phase offsets. It
            // is also possible to use different waveforms and different frequencies for each voice.
            
            for(int j = 0; j < numVoices_ - 1; ++j)
            {
                if(stereo_ != 0 && numVoices_ > 2)
                {
                    // A stereo chorus pans each voice to a different location in the stereo field.
                    // How this is done depends on the number of voices:
                    // -- 2 voices: N/A (need at least 2 delayed voices for stereo chorus)
                    // -- 3 voices: 1 voice left, 1 voice right (0, 1)
                    // -- 4 voices: 1 voice left, 1 voice centre, 1 voice right (0, 0.5, 1)
                    // -- 5 voices: 1 voice left, 1 voice left-centre,
                    //              1 voice right-centre, 1 voice right (0, 0.33, 0.66, 1)
                    
                    weight = (float)j/(float)(numVoices_ - 2);
                    
                    // Left and right channels are mirrors of each other in weight
                    if(channel != 0)
                        weight = 1.0 - weight;
                }
                else
                    weight = 1.0;

                // Add the voice to the mix if it has nonzero weight
                if(weight != 0.0)
                {
                    // Recalculate the read pointer position with respect to the write pointer. A more efficient
                    // implementation might increment the read pointer based on the derivative of the LFO without
                    // running the whole equation again, but this format makes the operation clearer.
                    
                    currentDelay = delay_ + sweepWidth_*lfo(fmodf(ph + phaseOffset, 1.0f), waveform_);
                    dpr = fmodf((float)dpw - (float)(currentDelay * getSampleRate()) + (float)delayBufferLength_,
                                (float)delayBufferLength_);
                    
                    // In this example, the output is the input plus the contents of the delay buffer (weighted by delayMix)
                    // The last term implements a tremolo (variable amplitude) on the whole thing.
          
                    if(interpolation_ == kInterpolationLinear)
                    {
                        // Find the fraction by which the read pointer sits between two
                        // samples and use this to adjust weights of the samples
                        float fraction = dpr - floorf(dpr);
                        int previousSample = (int)floorf(dpr);
                        int nextSample = (previousSample + 1) % delayBufferLength_;
                        interpolatedSample = fraction*delayData[nextSample]
                            + (1.0f-fraction)*delayData[previousSample];
                    }
                    else if(interpolation_ == kInterpolationCubic)
                    {
                        // Cubic interpolation will produce cleaner results at the expense
                        // of more computation. This code uses the Catmull-Rom variant of
                        // cubic interpolation. To reduce the load, calculate a few quantities
                        // in advance that will be used several times in the equation:
                        
                        int sample1 = (int)floorf(dpr);
                        int sample2 = (sample1 + 1) % delayBufferLength_;
                        int sample3 = (sample2 + 1) % delayBufferLength_;
                        int sample0 = (sample1 - 1 + delayBufferLength_) % delayBufferLength_;
                        
                        float fraction = dpr - floorf(dpr);
                        float frsq = fraction*fraction;
                        
                        float a0 = -0.5f*delayData[sample0] + 1.5f*delayData[sample1]
                                    - 1.5f*delayData[sample2] + 0.5f*delayData[sample3];
                        float a1 = delayData[sample0] - 2.5f*delayData[sample1]
                                    + 2.0f*delayData[sample2] - 0.5f*delayData[sample3];
                        float a2 = -0.5f*delayData[sample0] + 0.5f*delayData[sample2];
                        float a3 = delayData[sample1];
                        
                        interpolatedSample = a0*fraction*frsq + a1*frsq + a2*fraction + a3;
                    }
                    else // Nearest neighbour interpolation
                    {
                        // Find the nearest input sample by rounding the fractional index to the
                        // nearest integer. It's possible this will round it to the end of the buffer,
                        // in which case we need to roll it back to the beginning.
                        int closestSample = (int)floorf(dpr + 0.5f);
                        if(closestSample == delayBufferLength_)
                            closestSample = 0;
                        interpolatedSample = delayData[closestSample];
                    }

                    // Store the output sample in the buffer, which starts by containing the input sample
                    channelData[i] += depth_ * weight * interpolatedSample;
                }
                
                // 3-voice chorus uses two voices in quadrature phase (90 degrees apart). Otherwise,
                // spread the voice phases evenly around the unit circle. (For 2-voice chorus, this
                // code doesn't matter since the loop only runs once.)
                if(numVoices_ < 3)
                    phaseOffset += 0.25f;
                else
                    phaseOffset += 1.0f / (float)(numVoices_ - 1);
            }
            
            // Store the current input in the delay buffer (no feedback in a chorus, unlike a flanger).
            delayData[dpw] = in;
            
            // Increment the write pointer at a constant rate. The read pointer will move at different
            // rates depending on the settings of the LFO, the delay and the sweep width.
            
            if (++dpw >= delayBufferLength_)
                dpw = 0;

            // Update the LFO phase, keeping it in the range 0-1
            ph += frequency_*inverseSampleRate_;
            if(ph >= 1.0)
                ph -= 1.0;
        }
    }
    
    // Having made a local copy of the state variables for each channel, now transfer the result
    // back to the main state variable so they will be preserved for the next call of processBlock()
    
    delayWritePosition_ = dpw;
    lfoPhase_ = ph;
    
    // In case we have more outputs than inputs, we'll clear any output
    // channels that didn't contain input data, (because these aren't
    // guaranteed to be empty - they may contain garbage).
    for (int i = numInputChannels; i < numOutputChannels; ++i)
    {
        buffer.clear (i, 0, buffer.getNumSamples());
    }
}

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

AudioProcessorEditor* ChorusAudioProcessor::createEditor()
{
    return new ChorusAudioProcessorEditor (this);
}

//==============================================================================
void ChorusAudioProcessor::getStateInformation (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.
    
    // Create an outer XML element..
    XmlElement xml("C4DMPLUGINSETTINGS");
    
    // add some attributes to it..
    xml.setAttribute("uiWidth", lastUIWidth_);
    xml.setAttribute("uiHeight", lastUIHeight_);
    xml.setAttribute("delay", delay_);
    xml.setAttribute("sweepWidth", sweepWidth_);
    xml.setAttribute("depth", depth_);
    xml.setAttribute("frequency", frequency_);
    xml.setAttribute("waveform", waveform_);
    xml.setAttribute("interpolation", interpolation_);
    xml.setAttribute("numVoices", numVoices_);
    xml.setAttribute("stereo", stereo_);
    
    // then use this helper function to stuff it into the binary blob and return it..
    copyXmlToBinary(xml, destData);
}

void ChorusAudioProcessor::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.
    
    // This getXmlFromBinary() helper function retrieves our XML from the binary blob..
    ScopedPointer<XmlElement> xmlState (getXmlFromBinary (data, sizeInBytes));
    
    if(xmlState != 0)
    {
        // make sure that it's actually our type of XML object..
        if(xmlState->hasTagName("C4DMPLUGINSETTINGS"))
        {
            // ok, now pull out our parameters..
            lastUIWidth_  = xmlState->getIntAttribute("uiWidth", lastUIWidth_);
            lastUIHeight_ = xmlState->getIntAttribute("uiHeight", lastUIHeight_);
            
            delay_      = (float)xmlState->getDoubleAttribute("delay", delay_);
            sweepWidth_ = (float)xmlState->getDoubleAttribute("sweepWidth", sweepWidth_);
            depth_      = (float)xmlState->getDoubleAttribute("depth", depth_);
            frequency_  = (float)xmlState->getDoubleAttribute("frequency", frequency_);
            waveform_   = xmlState->getIntAttribute("waveform", waveform_);
            interpolation_ = xmlState->getIntAttribute("interpolation", interpolation_);
            numVoices_  = xmlState->getIntAttribute("numVoices", numVoices_);
            stereo_     = xmlState->getIntAttribute("stereo", stereo_);
        }
    }
}

//==============================================================================
// Function for calculating LFO waveforms. Phase runs from 0-1, output is scaled
// from 0 to 1 (note: not -1 to 1 as would be typical of sine).
float ChorusAudioProcessor::lfo(float phase, int waveform)
{
    switch(waveform)
    {
        case kWaveformTriangle:
            if(phase < 0.25f)
                return 0.5f + 2.0f*phase;
            else if(phase < 0.75f)
                return 1.0f - 2.0f*(phase - 0.25f);
            else
                return 2.0f*(phase-0.75f);
        case kWaveformSquare:
            if(phase < 0.5f)
                return 1.0f;
            else
                return 0.0f;
        case kWaveformSawtooth:
            if(phase < 0.5f)
                return 0.5f + phase;
            else
                return phase - 0.5f;
        case kWaveformSine:
        default:
            return 0.5f + 0.5f*sinf(2.0 * M_PI * phase);
    }
}

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