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:   Auto-Wah: LFO or envelope-operated wah effect
andrewm@0:   See textbook Chapter 4: Filter 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: 
andrewm@0: // The filter will produce a resonant peak of amplitude Q; bring everything
andrewm@0: // down somewhat to compensate, though try to maintain some perceptual balance
andrewm@0: // of being similar loudness. (This factor has been chosen somewhat arbitrarily.)
andrewm@0: const double kWahwahFilterGain = 0.5;
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: AutoWahAudioProcessor::AutoWahAudioProcessor()
andrewm@0: {
andrewm@0:     // Set default values:
andrewm@0:     baseFrequency_ = 350.0;
andrewm@0:     q_ = 5.0;
andrewm@0:     lfoFrequency_ = 2.0;
andrewm@0:     lfoWidth_ = 1000.0;
andrewm@0:     envelopeWidth_ = 0.0;
andrewm@0:     envelopeAttack_ = 0.005;
andrewm@0:     envelopeDecay_ = 0.1;
andrewm@0:     
andrewm@0:     // Initialise the filters later when we know how many channels
andrewm@0:     wahFilters_ = 0;
andrewm@0:     numWahFilters_ = 0;
andrewm@0:     envelopes_ = 0;
andrewm@0:     numEnvelopes_ = 0;
andrewm@0:     attackMultiplier_ = 1.0;
andrewm@0:     decayMultiplier_ = 0.0;
andrewm@0:     
andrewm@0:     inverseSampleRate_ = 1.0/44100.0; // start with a sensible default
andrewm@0:     
andrewm@0:     lastUIWidth_ = 550;
andrewm@0:     lastUIHeight_ = 200;
andrewm@0: }
andrewm@0: 
andrewm@0: AutoWahAudioProcessor::~AutoWahAudioProcessor()
andrewm@0: {
andrewm@0:     deallocateFilters();
andrewm@0: }
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: const String AutoWahAudioProcessor::getName() const
andrewm@0: {
andrewm@0:     return JucePlugin_Name;
andrewm@0: }
andrewm@0: 
andrewm@0: int AutoWahAudioProcessor::getNumParameters()
andrewm@0: {
andrewm@0:     return kNumParameters;
andrewm@0: }
andrewm@0: 
andrewm@0: float AutoWahAudioProcessor::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 kBaseFrequencyParam:  return baseFrequency_;
andrewm@0:         case kQParam:              return q_;
andrewm@0:         case kLFOFrequencyParam:   return lfoFrequency_;
andrewm@0:         case kLFOWidthParam:       return lfoWidth_;
andrewm@0:         case kEnvelopeWidthParam:  return envelopeWidth_;
andrewm@0:         case kEnvelopeAttackParam: return envelopeAttack_;
andrewm@0:         case kEnvelopeDecayParam:  return envelopeDecay_;
andrewm@0:         default:                   return 0.0f;
andrewm@0:     }
andrewm@0: }
andrewm@0: 
andrewm@0: void AutoWahAudioProcessor::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 kBaseFrequencyParam:
andrewm@0:             baseFrequency_ = newValue;
andrewm@0:             break;
andrewm@0:         case kQParam:
andrewm@0:             q_ = newValue;
andrewm@0:             break;
andrewm@0:         case kLFOFrequencyParam:
andrewm@0:             lfoFrequency_ = newValue;
andrewm@0:             break;
andrewm@0:         case kLFOWidthParam:
andrewm@0:             lfoWidth_ = newValue;
andrewm@0:             break;
andrewm@0:         case kEnvelopeWidthParam:
andrewm@0:             envelopeWidth_ = newValue;
andrewm@0:             break;
andrewm@0:         case kEnvelopeAttackParam:
andrewm@0:             envelopeAttack_ = newValue;
andrewm@0:             // See comment below for justification
andrewm@0:             if(envelopeAttack_ == 0.0)
andrewm@0:                 attackMultiplier_ = 0.0;
andrewm@0:             else
andrewm@0:                 attackMultiplier_ = pow(1.0 / M_E, inverseSampleRate_ / envelopeAttack_);
andrewm@0:             break;
andrewm@0:         case kEnvelopeDecayParam:
andrewm@0:             envelopeDecay_ = newValue;
andrewm@0:             // envelopeDecay_ sets the time constant tau. The decay is
andrewm@0:             // given as e^-(t/tau) so after tau seconds, it will have
andrewm@0:             // decayed to 1/e of its original value. tau*sampleRate samples
andrewm@0:             // will have passed by then, each of which multiplies the signal
andrewm@0:             // by decayMultiplier_.
andrewm@0:             if(envelopeDecay_ == 0.0)
andrewm@0:                 decayMultiplier_ = 0.0;
andrewm@0:             else
andrewm@0:                 decayMultiplier_ = pow(1.0 / M_E, inverseSampleRate_ / envelopeDecay_);
andrewm@0:             break;
andrewm@0:         default:
andrewm@0:             break;
andrewm@0:     }
andrewm@0: }
andrewm@0: 
andrewm@0: const String AutoWahAudioProcessor::getParameterName (int index)
andrewm@0: {
andrewm@0:     switch (index)
andrewm@0:     {
andrewm@0:         case kBaseFrequencyParam:  return "base frequency";
andrewm@0:         case kQParam:              return "Q";
andrewm@0:         case kLFOFrequencyParam:   return "LFO frequency";
andrewm@0:         case kLFOWidthParam:       return "LFO width";
andrewm@0:         case kEnvelopeWidthParam:  return "envelope width";
andrewm@0:         case kEnvelopeAttackParam: return "envelope attack";
andrewm@0:         case kEnvelopeDecayParam:  return "envelope decay";
andrewm@0:         default:                   break;
andrewm@0:     }
andrewm@0:     
andrewm@0:     return String::empty;
andrewm@0: }
andrewm@0: 
andrewm@0: const String AutoWahAudioProcessor::getParameterText (int index)
andrewm@0: {
andrewm@0:     return String (getParameter (index), 2);
andrewm@0: }
andrewm@0: 
andrewm@0: const String AutoWahAudioProcessor::getInputChannelName (int channelIndex) const
andrewm@0: {
andrewm@0:     return String (channelIndex + 1);
andrewm@0: }
andrewm@0: 
andrewm@0: const String AutoWahAudioProcessor::getOutputChannelName (int channelIndex) const
andrewm@0: {
andrewm@0:     return String (channelIndex + 1);
andrewm@0: }
andrewm@0: 
andrewm@0: bool AutoWahAudioProcessor::isInputChannelStereoPair (int index) const
andrewm@0: {
andrewm@0:     return true;
andrewm@0: }
andrewm@0: 
andrewm@0: bool AutoWahAudioProcessor::isOutputChannelStereoPair (int index) const
andrewm@0: {
andrewm@0:     return true;
andrewm@0: }
andrewm@0: 
andrewm@0: bool AutoWahAudioProcessor::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 AutoWahAudioProcessor::getTailLengthSeconds() const
andrewm@0: {
andrewm@0:     return 0.0;
andrewm@0: }
andrewm@0: 
andrewm@0: bool AutoWahAudioProcessor::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 AutoWahAudioProcessor::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 AutoWahAudioProcessor::getNumPrograms()
andrewm@0: {
andrewm@0:     return 0;
andrewm@0: }
andrewm@0: 
andrewm@0: int AutoWahAudioProcessor::getCurrentProgram()
andrewm@0: {
andrewm@0:     return 0;
andrewm@0: }
andrewm@0: 
andrewm@0: void AutoWahAudioProcessor::setCurrentProgram (int index)
andrewm@0: {
andrewm@0: }
andrewm@0: 
andrewm@0: const String AutoWahAudioProcessor::getProgramName (int index)
andrewm@0: {
andrewm@0:     return String::empty;
andrewm@0: }
andrewm@0: 
andrewm@0: void AutoWahAudioProcessor::changeProgramName (int index, const String& newName)
andrewm@0: {
andrewm@0: }
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: void AutoWahAudioProcessor::prepareToPlay (double sampleRate, int samplesPerBlock)
andrewm@0: {
andrewm@0:     // Use this method as the place to do any pre-playback
andrewm@0:     // initialisation that you need..
andrewm@0:     
andrewm@0:     allocateFilters();
andrewm@0:     inverseSampleRate_ = 1.0 / sampleRate;
andrewm@0:     if(envelopeDecay_ == 0.0)
andrewm@0:         decayMultiplier_ = 0.0;
andrewm@0:     else
andrewm@0:         decayMultiplier_ = pow(1.0 / M_E, inverseSampleRate_ / envelopeDecay_);
andrewm@0:     if(envelopeAttack_ == 0.0)
andrewm@0:         attackMultiplier_ = 0.0;
andrewm@0:     else
andrewm@0:         attackMultiplier_ = pow(1.0 / M_E, inverseSampleRate_ / envelopeAttack_);
andrewm@0: }
andrewm@0: 
andrewm@0: void AutoWahAudioProcessor::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:     deallocateFilters();
andrewm@0: }
andrewm@0: 
andrewm@0: void AutoWahAudioProcessor::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:     int channel;
andrewm@0:     float ph;
andrewm@0:     
andrewm@0:     // Go through each channel and put it through the resonant lowpass filter, updating
andrewm@0:     // the coefficients as we go along. Each channel is processed identically in this effect.
andrewm@0:     
andrewm@0:     for(channel = 0; channel < jmin(numInputChannels, numWahFilters_); ++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:         ph = lfoPhase_;
andrewm@0:         
andrewm@0:         for (int sample = 0; sample < numSamples; ++sample)
andrewm@0:         {
andrewm@0:             const float in = channelData[sample];
andrewm@0:             float centreFrequency = baseFrequency_;
andrewm@0:             
andrewm@0:             // Calculate the envelope of the signal. Do this even if we're not currently
andrewm@0:             // changing the frequeny based on it, since it involves maintaining a history
andrewm@0:             // of the signal's behaviour.
andrewm@0:             
andrewm@0:             if(channel < numEnvelopes_) {   // Safety check
andrewm@0:                 if(fabs(in) > envelopes_[channel]) {
andrewm@0:                     envelopes_[channel] += (1.0 - attackMultiplier_) * (fabs(in) - (double)envelopes_[channel]);
andrewm@0:                 }
andrewm@0:                 else
andrewm@0:                     envelopes_[channel] *= decayMultiplier_;
andrewm@0:             }
andrewm@0:             
andrewm@0:             // Calculate the centre frequency of the filter based on the LFO and the
andrewm@0:             // signal envelope
andrewm@0:             if(lfoWidth_ > 0.0) {
andrewm@0:                 centreFrequency += lfoWidth_ * (0.5f + 0.5f*sinf(2.0 * M_PI * ph));
andrewm@0:             }
andrewm@0:             if(envelopeWidth_ > 0.0 && channel < numEnvelopes_) {
andrewm@0:                 centreFrequency += envelopeWidth_ * envelopes_[channel];
andrewm@0:             }
andrewm@0:             
andrewm@0:             // Update filter coefficients (see ResonantLowpassFilter.cpp for calculation)
andrewm@0:             wahFilters_[channel]->makeResonantLowpass(inverseSampleRate_,
andrewm@0:                                                       centreFrequency,
andrewm@0:                                                       q_,
andrewm@0:                                                       kWahwahFilterGain);
andrewm@0:             
andrewm@0:             // Process one sample and store it back in place. See juce_IIRFilter.cpp for the
andrewm@0:             // application of the IIR filter.
andrewm@0:             channelData[sample] = wahFilters_[channel]->processSingleSampleRaw(in);
andrewm@0:             
andrewm@0:             // Update the LFO phase, keeping it in the range 0-1
andrewm@0:             ph += lfoFrequency_*inverseSampleRate_;
andrewm@0:             if(ph >= 1.0)
andrewm@0:                 ph -= 1.0;
andrewm@0:         }
andrewm@0:     }
andrewm@0:     
andrewm@0:     lfoPhase_ = ph;
andrewm@0:     
andrewm@0:     // Go through the remaining channels. In case we have more outputs
andrewm@0:     // than inputs, or there aren't enough filters, we'll clear any
andrewm@0:     // remaining output channels (which could otherwise contain garbage)
andrewm@0:     while(channel < numOutputChannels)
andrewm@0:     {
andrewm@0:         buffer.clear (channel++, 0, buffer.getNumSamples());
andrewm@0:     }
andrewm@0: }
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: bool AutoWahAudioProcessor::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* AutoWahAudioProcessor::createEditor()
andrewm@0: {
andrewm@0:     return new AutoWahAudioProcessorEditor (this);
andrewm@0: }
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: void AutoWahAudioProcessor::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("baseFrequency", baseFrequency_);
andrewm@0:     xml.setAttribute("q", q_);
andrewm@0:     xml.setAttribute("lfoFrequency", lfoFrequency_);
andrewm@0:     xml.setAttribute("lfoWidth", lfoWidth_);
andrewm@0:     xml.setAttribute("envelopeWidth", envelopeWidth_);
andrewm@0:     xml.setAttribute("envelopeAttack", envelopeAttack_);
andrewm@0:     xml.setAttribute("envelopeDecay", envelopeDecay_);
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 AutoWahAudioProcessor::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:             q_ = (float)xmlState->getDoubleAttribute("q", q_);
andrewm@0:             baseFrequency_ = (float)xmlState->getDoubleAttribute("baseFrequency", baseFrequency_);
andrewm@0:             lfoFrequency_ = (float)xmlState->getDoubleAttribute("lfoFrequency", lfoFrequency_);
andrewm@0:             lfoWidth_ = (float)xmlState->getDoubleAttribute("lfoWidth", lfoWidth_);
andrewm@0:             envelopeWidth_ = (float)xmlState->getDoubleAttribute("envelopeWidth", envelopeWidth_);
andrewm@0:             envelopeAttack_ = (float)xmlState->getDoubleAttribute("envelopeAttack", envelopeAttack_);
andrewm@0:             envelopeDecay_ = (float)xmlState->getDoubleAttribute("envelopeDecay", envelopeDecay_);
andrewm@0:             inverseSampleRate_ = 1.0 / getSampleRate();
andrewm@0:             if(envelopeDecay_ == 0.0)
andrewm@0:                 decayMultiplier_ = 0.0;
andrewm@0:             else
andrewm@0:                 decayMultiplier_ = pow(1.0 / M_E, inverseSampleRate_ / envelopeDecay_);
andrewm@0:             if(envelopeAttack_ == 0.0)
andrewm@0:                 attackMultiplier_ = 0.0;
andrewm@0:             else
andrewm@0:                 attackMultiplier_ = pow(1.0 / M_E, inverseSampleRate_ / envelopeAttack_);
andrewm@0:         }
andrewm@0:     }
andrewm@0: }
andrewm@0: 
andrewm@0: void AutoWahAudioProcessor::allocateFilters()
andrewm@0: {
andrewm@0:     // Prevent leaks from reallocation
andrewm@0:     if(wahFilters_ != 0 || envelopes_ != 0)
andrewm@0:         deallocateFilters();
andrewm@0:     
andrewm@0:     // Create as many filters as we have input channels
andrewm@0:     numWahFilters_ = getNumInputChannels();
andrewm@0:     wahFilters_ = (ResonantLowpassFilter**)malloc(numWahFilters_ * sizeof(ResonantLowpassFilter*));
andrewm@0:     if(wahFilters_ == 0)
andrewm@0:         numWahFilters_ = 0;
andrewm@0:     else {
andrewm@0:         for(int i = 0; i < numWahFilters_; i++)
andrewm@0:             wahFilters_[i] = new ResonantLowpassFilter;
andrewm@0:     }
andrewm@0:     
andrewm@0:     numEnvelopes_ = getNumInputChannels();
andrewm@0:     envelopes_ = (double *)malloc(numEnvelopes_ * sizeof(double));
andrewm@0:     if(envelopes_ == 0)
andrewm@0:         numEnvelopes_ = 0;
andrewm@0:     else {
andrewm@0:         for(int i = 0; i < numEnvelopes_; i++)
andrewm@0:             envelopes_[i] = 0.0;
andrewm@0:     }
andrewm@0: }
andrewm@0: 
andrewm@0: void AutoWahAudioProcessor::deallocateFilters()
andrewm@0: {
andrewm@0:     for(int i = 0; i < numWahFilters_; i++)
andrewm@0:         delete wahFilters_[i];
andrewm@0:     if(numWahFilters_ != 0)
andrewm@0:         free(wahFilters_);
andrewm@0:     numWahFilters_ = 0;
andrewm@0:     wahFilters_ = 0;
andrewm@0:     if(envelopes_ != 0)
andrewm@0:         free(envelopes_);
andrewm@0:     envelopes_ = 0;
andrewm@0:     numEnvelopes_ = 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 AutoWahAudioProcessor();
andrewm@0: }