andrewm@0: /*
andrewm@0:   This code accompanies the textbook:
andrewm@0:  
andrewm@0:   Digital Audio Effects: Theory, Implementation and Application
andrewm@0:   Joshua D. Reiss and Andrew P. McPherson
andrewm@0:  
andrewm@0:   ---
andrewm@0:  
andrewm@0:   Flanger: flanging effect using time-varying delay
andrewm@0:   See textbook Chapter 2: Delay Line Effects
andrewm@0:  
andrewm@0:   Code by Andrew McPherson, Brecht De Man and Joshua Reiss
andrewm@0:  
andrewm@0:   ---
andrewm@0: 
andrewm@0:   This program is free software: you can redistribute it and/or modify
andrewm@0:   it under the terms of the GNU General Public License as published by
andrewm@0:   the Free Software Foundation, either version 3 of the License, or
andrewm@0:   (at your option) any later version.
andrewm@0:  
andrewm@0:   This program is distributed in the hope that it will be useful,
andrewm@0:   but WITHOUT ANY WARRANTY; without even the implied warranty of
andrewm@0:   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
andrewm@0:   GNU General Public License for more details.
andrewm@0: 
andrewm@0:   You should have received a copy of the GNU General Public License
andrewm@0:   along with this program.  If not, see <http://www.gnu.org/licenses/>.
andrewm@0: */
andrewm@0: 
andrewm@0: #include "PluginProcessor.h"
andrewm@0: #include "PluginEditor.h"
andrewm@0: #include <math.h>
andrewm@0: 
andrewm@0: const float FlangerAudioProcessor::kMaximumDelay = 0.02;
andrewm@0: const float FlangerAudioProcessor::kMaximumSweepWidth = 0.02;
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: FlangerAudioProcessor::FlangerAudioProcessor() : delayBuffer_ (2, 1)
andrewm@0: {
andrewm@0:     // Set default values:
andrewm@0:     delay_ = .0025;
andrewm@0:     sweepWidth_ = .010;
andrewm@0:     depth_ = 1.0;
andrewm@0:     feedback_ = 0.0;
andrewm@0:     frequency_ = 0.2;
andrewm@0:     waveform_ = kWaveformSine;
andrewm@0:     interpolation_ = kInterpolationLinear;
andrewm@0:     stereo_ = 0;
andrewm@0:     
andrewm@0:     delayBufferLength_ = 1;
andrewm@0:     lfoPhase_ = 0.0;
andrewm@0:     inverseSampleRate_ = 1.0/44100.0;
andrewm@0:     
andrewm@0:     // Start the circular buffer pointer at the beginning
andrewm@0:     delayWritePosition_ = 0;
andrewm@0:     
andrewm@0:     lastUIWidth_ = 550;
andrewm@0:     lastUIHeight_ = 200;
andrewm@0: }
andrewm@0: 
andrewm@0: FlangerAudioProcessor::~FlangerAudioProcessor()
andrewm@0: {
andrewm@0: }
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: const String FlangerAudioProcessor::getName() const
andrewm@0: {
andrewm@0:     return JucePlugin_Name;
andrewm@0: }
andrewm@0: 
andrewm@0: int FlangerAudioProcessor::getNumParameters()
andrewm@0: {
andrewm@0:     return kNumParameters;
andrewm@0: }
andrewm@0: 
andrewm@0: float FlangerAudioProcessor::getParameter (int index)
andrewm@0: {
andrewm@0:     // This method will be called by the host, probably on the audio thread, so
andrewm@0:     // it's absolutely time-critical. Don't use critical sections or anything
andrewm@0:     // UI-related, or anything at all that may block in any way!
andrewm@0:     switch (index)
andrewm@0:     {
andrewm@0:         case kDelayParam:         return delay_;
andrewm@0:         case kSweepWidthParam:    return sweepWidth_;
andrewm@0:         case kDepthParam:         return depth_;
andrewm@0:         case kFeedbackParam:      return feedback_;
andrewm@0:         case kFrequencyParam:     return frequency_;
andrewm@0:         case kWaveformParam:      return (float)waveform_;
andrewm@0:         case kInterpolationParam: return (float)interpolation_;
andrewm@0:         case kStereoParam:        return (float)stereo_;
andrewm@0:         default:                  return 0.0f;
andrewm@0:     }
andrewm@0: }
andrewm@0: 
andrewm@0: void FlangerAudioProcessor::setParameter (int index, float newValue)
andrewm@0: {
andrewm@0:     // This method will be called by the host, probably on the audio thread, so
andrewm@0:     // it's absolutely time-critical. Don't use critical sections or anything
andrewm@0:     // UI-related, or anything at all that may block in any way!
andrewm@0: 
andrewm@0:     switch (index)
andrewm@0:     {
andrewm@0:         case kDelayParam:
andrewm@0:             delay_ = newValue;
andrewm@0:             break;
andrewm@0:         case kSweepWidthParam:
andrewm@0:             sweepWidth_ = newValue;
andrewm@0:             break;
andrewm@0:         case kDepthParam:
andrewm@0:             depth_ = newValue;
andrewm@0:             break;
andrewm@0:         case kFeedbackParam:
andrewm@0:             feedback_ = newValue;
andrewm@0:             break;
andrewm@0:         case kFrequencyParam:
andrewm@0:             frequency_ = newValue;
andrewm@0:             break;
andrewm@0:         case kWaveformParam:
andrewm@0:             waveform_ = (int)newValue;
andrewm@0:             break;
andrewm@0:         case kInterpolationParam:
andrewm@0:             interpolation_ = (int)newValue;
andrewm@0:             break;
andrewm@0:         case kStereoParam:
andrewm@0:             stereo_ = (int)newValue;
andrewm@0:             break;
andrewm@0:         default:
andrewm@0:             break;
andrewm@0:     }
andrewm@0: }
andrewm@0: 
andrewm@0: const String FlangerAudioProcessor::getParameterName (int index)
andrewm@0: {
andrewm@0:     switch (index)
andrewm@0:     {
andrewm@0:         case kDelayParam:         return "delay";
andrewm@0:         case kSweepWidthParam:    return "sweep width";
andrewm@0:         case kDepthParam:         return "depth";
andrewm@0:         case kFeedbackParam:      return "feedback";
andrewm@0:         case kFrequencyParam:     return "frequency";
andrewm@0:         case kWaveformParam:      return "waveform";
andrewm@0:         case kInterpolationParam: return "interpolation";
andrewm@0:         case kStereoParam:        return "stereo";
andrewm@0:         default:                  break;
andrewm@0:     }
andrewm@0:     
andrewm@0:     return String::empty;
andrewm@0: }
andrewm@0: 
andrewm@0: const String FlangerAudioProcessor::getParameterText (int index)
andrewm@0: {
andrewm@0:     return String (getParameter (index), 2);
andrewm@0: }
andrewm@0: 
andrewm@0: const String FlangerAudioProcessor::getInputChannelName (int channelIndex) const
andrewm@0: {
andrewm@0:     return String (channelIndex + 1);
andrewm@0: }
andrewm@0: 
andrewm@0: const String FlangerAudioProcessor::getOutputChannelName (int channelIndex) const
andrewm@0: {
andrewm@0:     return String (channelIndex + 1);
andrewm@0: }
andrewm@0: 
andrewm@0: bool FlangerAudioProcessor::isInputChannelStereoPair (int index) const
andrewm@0: {
andrewm@0:     return true;
andrewm@0: }
andrewm@0: 
andrewm@0: bool FlangerAudioProcessor::isOutputChannelStereoPair (int index) const
andrewm@0: {
andrewm@0:     return true;
andrewm@0: }
andrewm@0: 
andrewm@0: bool FlangerAudioProcessor::silenceInProducesSilenceOut() const
andrewm@0: {
andrewm@0: #if JucePlugin_SilenceInProducesSilenceOut
andrewm@0:     return true;
andrewm@0: #else
andrewm@0:     return false;
andrewm@0: #endif
andrewm@0: }
andrewm@0: 
andrewm@0: double FlangerAudioProcessor::getTailLengthSeconds() const
andrewm@0: {
andrewm@0:     return 0.0;
andrewm@0: }
andrewm@0: 
andrewm@0: bool FlangerAudioProcessor::acceptsMidi() const
andrewm@0: {
andrewm@0: #if JucePlugin_WantsMidiInput
andrewm@0:     return true;
andrewm@0: #else
andrewm@0:     return false;
andrewm@0: #endif
andrewm@0: }
andrewm@0: 
andrewm@0: bool FlangerAudioProcessor::producesMidi() const
andrewm@0: {
andrewm@0: #if JucePlugin_ProducesMidiOutput
andrewm@0:     return true;
andrewm@0: #else
andrewm@0:     return false;
andrewm@0: #endif
andrewm@0: }
andrewm@0: 
andrewm@0: int FlangerAudioProcessor::getNumPrograms()
andrewm@0: {
andrewm@0:     return 0;
andrewm@0: }
andrewm@0: 
andrewm@0: int FlangerAudioProcessor::getCurrentProgram()
andrewm@0: {
andrewm@0:     return 0;
andrewm@0: }
andrewm@0: 
andrewm@0: void FlangerAudioProcessor::setCurrentProgram (int index)
andrewm@0: {
andrewm@0: }
andrewm@0: 
andrewm@0: const String FlangerAudioProcessor::getProgramName (int index)
andrewm@0: {
andrewm@0:     return String::empty;
andrewm@0: }
andrewm@0: 
andrewm@0: void FlangerAudioProcessor::changeProgramName (int index, const String& newName)
andrewm@0: {
andrewm@0: }
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: void FlangerAudioProcessor::prepareToPlay (double sampleRate, int samplesPerBlock)
andrewm@0: {
andrewm@0:     // Allocate and zero the delay buffer (size will depend on current sample rate)
andrewm@0:     // Add 3 extra samples to allow cubic interpolation even at maximum delay
andrewm@0:     delayBufferLength_ = (int)((kMaximumDelay + kMaximumSweepWidth)*sampleRate) + 3;
andrewm@0:     delayBuffer_.setSize(2, delayBufferLength_);
andrewm@0:     delayBuffer_.clear();
andrewm@0:     lfoPhase_ = 0.0;
andrewm@0:     
andrewm@0:     inverseSampleRate_ = 1.0/sampleRate;
andrewm@0: }
andrewm@0: 
andrewm@0: void FlangerAudioProcessor::releaseResources()
andrewm@0: {
andrewm@0:     // When playback stops, you can use this as an opportunity to free up any
andrewm@0:     // spare memory, etc.
andrewm@0:     
andrewm@0:     // The delay buffer will stay in memory until the effect is unloaded.
andrewm@0: }
andrewm@0: 
andrewm@0: void FlangerAudioProcessor::reset()
andrewm@0: {
andrewm@0:     // Use this method as the place to clear any delay lines, buffers, etc, as it
andrewm@0:     // means there's been a break in the audio's continuity.
andrewm@0:     
andrewm@0:     delayBuffer_.clear();
andrewm@0: }
andrewm@0: 
andrewm@0: 
andrewm@0: void FlangerAudioProcessor::processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages)
andrewm@0: {
andrewm@0:     // Helpful information about this block of samples:
andrewm@0:     const int numInputChannels = getNumInputChannels();     // How many input channels for our effect?
andrewm@0:     const int numOutputChannels = getNumOutputChannels();   // How many output channels for our effect?
andrewm@0:     const int numSamples = buffer.getNumSamples();          // How many samples in the buffer for this block?
andrewm@0:     
andrewm@0:     int channel, dpw; // dpr = delay read pointer; dpw = delay write pointer
andrewm@0:     float dpr, currentDelay, ph;
andrewm@0:     float channel0EndPhase = lfoPhase_;
andrewm@0:     
andrewm@0:     // Go through each channel of audio that's passed in. In this example we apply identical
andrewm@0:     // effects to each channel, regardless of how many input channels there are. For some effects, like
andrewm@0:     // a stereo chorus or panner, you might do something different for each channel.
andrewm@0:     
andrewm@0:     for (channel = 0; channel < numInputChannels; ++channel)
andrewm@0:     {
andrewm@0:         // channelData is an array of length numSamples which contains the audio for one channel
b@1:         float* channelData = buffer.getWritePointer(channel);
andrewm@0:         
andrewm@0:         // delayData is the circular buffer for implementing delay on this channel
b@1:         float* delayData = delayBuffer_.getWritePointer (jmin (channel, delayBuffer_.getNumChannels() - 1));
andrewm@0:         
andrewm@0:         // Make a temporary copy of any state variables declared in PluginProcessor.h which need to be
andrewm@0:         // maintained between calls to processBlock(). Each channel needs to be processed identically
andrewm@0:         // which means that the activity of processing one channel can't affect the state variable for
andrewm@0:         // the next channel.
andrewm@0:         
andrewm@0:         dpw = delayWritePosition_;
andrewm@0:         ph = lfoPhase_;
andrewm@0:         
andrewm@0:         // For stereo flanging, keep the channels 90 degrees out of phase with each other
andrewm@0:         if(stereo_ != 0 && channel != 0)
andrewm@0:             ph = fmodf(ph + 0.25, 1.0);
andrewm@0:         
andrewm@0:         for (int i = 0; i < numSamples; ++i)
andrewm@0:         {
andrewm@0:             const float in = channelData[i];
andrewm@0:             float interpolatedSample = 0.0;
andrewm@0:             
andrewm@0:             // Recalculate the read pointer position with respect to the write pointer. A more efficient
andrewm@0:             // implementation might increment the read pointer based on the derivative of the LFO without
andrewm@0:             // running the whole equation again, but this format makes the operation clearer.
andrewm@0:             
andrewm@0:             currentDelay = delay_ + sweepWidth_*lfo(ph, waveform_);
andrewm@0:             dpr = fmodf((float)dpw - (float)(currentDelay * getSampleRate()) + (float)delayBufferLength_,
andrewm@0:                         (float)delayBufferLength_);
andrewm@0:             
andrewm@0:             // In this example, the output is the input plus the contents of the delay buffer (weighted by delayMix)
andrewm@0:             // The last term implements a tremolo (variable amplitude) on the whole thing.
andrewm@0:   
andrewm@0:             if(interpolation_ == kInterpolationLinear)
andrewm@0:             {
andrewm@0:                 // Find the fraction by which the read pointer sits between two
andrewm@0:                 // samples and use this to adjust weights of the samples
andrewm@0:                 float fraction = dpr - floorf(dpr);
andrewm@0:                 int previousSample = (int)floorf(dpr);
andrewm@0:                 int nextSample = (previousSample + 1) % delayBufferLength_;
andrewm@0:                 interpolatedSample = fraction*delayData[nextSample]
andrewm@0:                 + (1.0f-fraction)*delayData[previousSample];
andrewm@0:             }
andrewm@0:             else if(interpolation_ == kInterpolationCubic)
andrewm@0:             {
andrewm@0:                 // Cubic interpolation will produce cleaner results at the expense
andrewm@0:                 // of more computation. This code uses the Catmull-Rom variant of
andrewm@0:                 // cubic interpolation. To reduce the load, calculate a few quantities
andrewm@0:                 // in advance that will be used several times in the equation:
andrewm@0:                 
andrewm@0:                 int sample1 = (int)floorf(dpr);
andrewm@0:                 int sample2 = (sample1 + 1) % delayBufferLength_;
andrewm@0:                 int sample3 = (sample2 + 1) % delayBufferLength_;
andrewm@0:                 int sample0 = (sample1 - 1 + delayBufferLength_) % delayBufferLength_;
andrewm@0:                 
andrewm@0:                 float fraction = dpr - floorf(dpr);
andrewm@0:                 float frsq = fraction*fraction;
andrewm@0:                 
andrewm@0:                 float a0 = -0.5f*delayData[sample0] + 1.5f*delayData[sample1]
andrewm@0:                             - 1.5f*delayData[sample2] + 0.5f*delayData[sample3];
andrewm@0:                 float a1 = delayData[sample0] - 2.5f*delayData[sample1]
andrewm@0:                             + 2.0f*delayData[sample2] - 0.5f*delayData[sample3];
andrewm@0:                 float a2 = -0.5f*delayData[sample0] + 0.5f*delayData[sample2];
andrewm@0:                 float a3 = delayData[sample1];
andrewm@0:                 
andrewm@0:                 interpolatedSample = a0*fraction*frsq + a1*frsq + a2*fraction + a3;
andrewm@0:             }
andrewm@0:             else // Nearest neighbour interpolation
andrewm@0:             {
andrewm@0:                 // Find the nearest input sample by rounding the fractional index to the
andrewm@0:                 // nearest integer. It's possible this will round it to the end of the buffer,
andrewm@0:                 // in which case we need to roll it back to the beginning.
andrewm@0:                 int closestSample = (int)floorf(dpr + 0.5f);
andrewm@0:                 if(closestSample == delayBufferLength_)
andrewm@0:                     closestSample = 0;
andrewm@0:                 interpolatedSample = delayData[closestSample];
andrewm@0:             }
andrewm@0:             
andrewm@0:             // Store the current information in the delay buffer. With feedback, what we read is
andrewm@0:             // included in what gets stored in the buffer, otherwise it's just a simple delay line
andrewm@0:             // of the input signal.
andrewm@0:             
andrewm@0:             delayData[dpw] = in + (interpolatedSample * feedback_);
andrewm@0:             
andrewm@0:             // Increment the write pointer at a constant rate. The read pointer will move at different
andrewm@0:             // rates depending on the settings of the LFO, the delay and the sweep width.
andrewm@0:             
andrewm@0:             if (++dpw >= delayBufferLength_)
andrewm@0:                 dpw = 0;
andrewm@0: 
andrewm@0:             // Store the output sample in the buffer, replacing the input
andrewm@0:             channelData[i] = in + depth_ * interpolatedSample;
andrewm@0: 
andrewm@0:             // Update the LFO phase, keeping it in the range 0-1
andrewm@0:             ph += frequency_*inverseSampleRate_;
andrewm@0:             if(ph >= 1.0)
andrewm@0:                 ph -= 1.0;
andrewm@0:         }
andrewm@0:         
andrewm@0:         // Use channel 0 only to keep the phase in sync between calls to processBlock()
andrewm@0:         // Otherwise quadrature phase on multiple channels will create problems.
andrewm@0:         if(channel == 0)
andrewm@0:             channel0EndPhase = ph;
andrewm@0:     }
andrewm@0:     
andrewm@0:     // Having made a local copy of the state variables for each channel, now transfer the result
andrewm@0:     // back to the main state variable so they will be preserved for the next call of processBlock()
andrewm@0:     
andrewm@0:     delayWritePosition_ = dpw;
andrewm@0:     lfoPhase_ = channel0EndPhase;
andrewm@0:     
andrewm@0:     // In case we have more outputs than inputs, we'll clear any output
andrewm@0:     // channels that didn't contain input data, (because these aren't
andrewm@0:     // guaranteed to be empty - they may contain garbage).
andrewm@0:     for (int i = numInputChannels; i < numOutputChannels; ++i)
andrewm@0:     {
andrewm@0:         buffer.clear (i, 0, buffer.getNumSamples());
andrewm@0:     }
andrewm@0: }
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: bool FlangerAudioProcessor::hasEditor() const
andrewm@0: {
andrewm@0:     return true; // (change this to false if you choose to not supply an editor)
andrewm@0: }
andrewm@0: 
andrewm@0: AudioProcessorEditor* FlangerAudioProcessor::createEditor()
andrewm@0: {
andrewm@0:     return new FlangerAudioProcessorEditor (this);
andrewm@0: }
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: void FlangerAudioProcessor::getStateInformation (MemoryBlock& destData)
andrewm@0: {
andrewm@0:     // You should use this method to store your parameters in the memory block.
andrewm@0:     // You could do that either as raw data, or use the XML or ValueTree classes
andrewm@0:     // as intermediaries to make it easy to save and load complex data.
andrewm@0:     
andrewm@0:     // Create an outer XML element..
andrewm@0:     XmlElement xml("C4DMPLUGINSETTINGS");
andrewm@0:     
andrewm@0:     // add some attributes to it..
andrewm@0:     xml.setAttribute("uiWidth", lastUIWidth_);
andrewm@0:     xml.setAttribute("uiHeight", lastUIHeight_);
andrewm@0:     xml.setAttribute("delay", delay_);
andrewm@0:     xml.setAttribute("feedback", feedback_);
andrewm@0:     xml.setAttribute("sweepWidth", sweepWidth_);
andrewm@0:     xml.setAttribute("depth", depth_);
andrewm@0:     xml.setAttribute("frequency", frequency_);
andrewm@0:     xml.setAttribute("waveform", waveform_);
andrewm@0:     xml.setAttribute("interpolation", interpolation_);
andrewm@0:     xml.setAttribute("stereo", stereo_);
andrewm@0:     
andrewm@0:     // then use this helper function to stuff it into the binary blob and return it..
andrewm@0:     copyXmlToBinary(xml, destData);
andrewm@0: }
andrewm@0: 
andrewm@0: void FlangerAudioProcessor::setStateInformation (const void* data, int sizeInBytes)
andrewm@0: {
andrewm@0:     // You should use this method to restore your parameters from this memory block,
andrewm@0:     // whose contents will have been created by the getStateInformation() call.
andrewm@0:     
andrewm@0:     // This getXmlFromBinary() helper function retrieves our XML from the binary blob..
andrewm@0:     ScopedPointer<XmlElement> xmlState (getXmlFromBinary (data, sizeInBytes));
andrewm@0:     
andrewm@0:     if(xmlState != 0)
andrewm@0:     {
andrewm@0:         // make sure that it's actually our type of XML object..
andrewm@0:         if(xmlState->hasTagName("C4DMPLUGINSETTINGS"))
andrewm@0:         {
andrewm@0:             // ok, now pull out our parameters..
andrewm@0:             lastUIWidth_  = xmlState->getIntAttribute("uiWidth", lastUIWidth_);
andrewm@0:             lastUIHeight_ = xmlState->getIntAttribute("uiHeight", lastUIHeight_);
andrewm@0:             
andrewm@0:             delay_      = (float)xmlState->getDoubleAttribute("delay", delay_);
andrewm@0:             feedback_   = (float)xmlState->getDoubleAttribute("feedback", feedback_);
andrewm@0:             sweepWidth_ = (float)xmlState->getDoubleAttribute("sweepWidth", sweepWidth_);
andrewm@0:             depth_      = (float)xmlState->getDoubleAttribute("depth", depth_);
andrewm@0:             frequency_  = (float)xmlState->getDoubleAttribute("frequency", frequency_);
andrewm@0:             waveform_   = xmlState->getIntAttribute("waveform", waveform_);
andrewm@0:             interpolation_ = xmlState->getIntAttribute("interpolation", interpolation_);
andrewm@0:             stereo_     = xmlState->getIntAttribute("stereo", stereo_);
andrewm@0:         }
andrewm@0:     }
andrewm@0: }
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: // Function for calculating LFO waveforms. Phase runs from 0-1, output is scaled
andrewm@0: // from 0 to 1 (note: not -1 to 1 as would be typical of sine).
andrewm@0: float FlangerAudioProcessor::lfo(float phase, int waveform)
andrewm@0: {
andrewm@0:     switch(waveform)
andrewm@0:     {
andrewm@0:         case kWaveformTriangle:
andrewm@0:             if(phase < 0.25f)
andrewm@0:                 return 0.5f + 2.0f*phase;
andrewm@0:             else if(phase < 0.75f)
andrewm@0:                 return 1.0f - 2.0f*(phase - 0.25f);
andrewm@0:             else
andrewm@0:                 return 2.0f*(phase-0.75f);
andrewm@0:         case kWaveformSquare:
andrewm@0:             if(phase < 0.5f)
andrewm@0:                 return 1.0f;
andrewm@0:             else
andrewm@0:                 return 0.0f;
andrewm@0:         case kWaveformSawtooth:
andrewm@0:             if(phase < 0.5f)
andrewm@0:                 return 0.5f + phase;
andrewm@0:             else
andrewm@0:                 return phase - 0.5f;
andrewm@0:         case kWaveformSine:
andrewm@0:         default:
andrewm@0:             return 0.5f + 0.5f*sinf(2.0 * M_PI * phase);
andrewm@0:     }
andrewm@0: }
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: // This creates new instances of the plugin..
andrewm@0: AudioProcessor* JUCE_CALLTYPE createPluginFilter()
andrewm@0: {
andrewm@0:     return new FlangerAudioProcessor();
andrewm@0: }