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:   Ring Modulator: modulation using a carrier oscillator
andrewm@0:   See textbook Chapter 5: Amplitude Modulation
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: //==============================================================================
andrewm@0: RingModulatorAudioProcessor::RingModulatorAudioProcessor()
andrewm@0: {
andrewm@0:     // Set default values:
andrewm@0:     carrierFrequency_ = 100.0;
andrewm@0:     sweepWidth_ = 0.0;
andrewm@0:     lfoFrequency_ = 1.0;
andrewm@0:     waveform_ = kWaveformSine;
andrewm@0:     
andrewm@0:     lfoPhase_ = carrierPhase_ = 0.0;
andrewm@0:     inverseSampleRate_ = 1.0/44100.0;
andrewm@0:     
andrewm@0:     lastUIWidth_ = 370;
andrewm@0:     lastUIHeight_ = 160;
andrewm@0: }
andrewm@0: 
andrewm@0: RingModulatorAudioProcessor::~RingModulatorAudioProcessor()
andrewm@0: {
andrewm@0: }
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: const String RingModulatorAudioProcessor::getName() const
andrewm@0: {
andrewm@0:     return JucePlugin_Name;
andrewm@0: }
andrewm@0: 
andrewm@0: int RingModulatorAudioProcessor::getNumParameters()
andrewm@0: {
andrewm@0:     return kNumParameters;
andrewm@0: }
andrewm@0: 
andrewm@0: float RingModulatorAudioProcessor::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 kCarrierFrequencyParam: return carrierFrequency_;
andrewm@0:         case kSweepWidthParam:       return sweepWidth_;
andrewm@0:         case kLFOFrequencyParam:     return lfoFrequency_;
andrewm@0:         case kWaveformParam:         return (float)waveform_;
andrewm@0:         default:                     return 0.0f;
andrewm@0:     }
andrewm@0: }
andrewm@0: 
andrewm@0: void RingModulatorAudioProcessor::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 kCarrierFrequencyParam:
andrewm@0:             carrierFrequency_ = newValue;
andrewm@0:             break;
andrewm@0:         case kSweepWidthParam:
andrewm@0:             sweepWidth_ = newValue;
andrewm@0:             break;
andrewm@0:         case kLFOFrequencyParam:
andrewm@0:             lfoFrequency_ = newValue;
andrewm@0:             break;
andrewm@0:         case kWaveformParam:
andrewm@0:             waveform_ = (int)newValue;
andrewm@0:             break;
andrewm@0:         default:
andrewm@0:             break;
andrewm@0:     }
andrewm@0: }
andrewm@0: 
andrewm@0: const String RingModulatorAudioProcessor::getParameterName (int index)
andrewm@0: {
andrewm@0:     switch (index)
andrewm@0:     {
andrewm@0:         case kCarrierFrequencyParam: return "carrier frequency";
andrewm@0:         case kSweepWidthParam:       return "LFO sweep width";
andrewm@0:         case kLFOFrequencyParam:     return "LFO frequency";
andrewm@0:         case kWaveformParam:         return "LFO waveform";
andrewm@0:         default:                     break;
andrewm@0:     }
andrewm@0:     
andrewm@0:     return String::empty;
andrewm@0: }
andrewm@0: 
andrewm@0: const String RingModulatorAudioProcessor::getParameterText (int index)
andrewm@0: {
andrewm@0:     return String (getParameter (index), 2);
andrewm@0: }
andrewm@0: 
andrewm@0: const String RingModulatorAudioProcessor::getInputChannelName (int channelIndex) const
andrewm@0: {
andrewm@0:     return String (channelIndex + 1);
andrewm@0: }
andrewm@0: 
andrewm@0: const String RingModulatorAudioProcessor::getOutputChannelName (int channelIndex) const
andrewm@0: {
andrewm@0:     return String (channelIndex + 1);
andrewm@0: }
andrewm@0: 
andrewm@0: bool RingModulatorAudioProcessor::isInputChannelStereoPair (int index) const
andrewm@0: {
andrewm@0:     return true;
andrewm@0: }
andrewm@0: 
andrewm@0: bool RingModulatorAudioProcessor::isOutputChannelStereoPair (int index) const
andrewm@0: {
andrewm@0:     return true;
andrewm@0: }
andrewm@0: 
andrewm@0: bool RingModulatorAudioProcessor::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 RingModulatorAudioProcessor::getTailLengthSeconds() const
andrewm@0: {
andrewm@0:     return 0.0;
andrewm@0: }
andrewm@0: 
andrewm@0: bool RingModulatorAudioProcessor::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 RingModulatorAudioProcessor::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 RingModulatorAudioProcessor::getNumPrograms()
andrewm@0: {
andrewm@0:     return 0;
andrewm@0: }
andrewm@0: 
andrewm@0: int RingModulatorAudioProcessor::getCurrentProgram()
andrewm@0: {
andrewm@0:     return 0;
andrewm@0: }
andrewm@0: 
andrewm@0: void RingModulatorAudioProcessor::setCurrentProgram (int index)
andrewm@0: {
andrewm@0: }
andrewm@0: 
andrewm@0: const String RingModulatorAudioProcessor::getProgramName (int index)
andrewm@0: {
andrewm@0:     return String::empty;
andrewm@0: }
andrewm@0: 
andrewm@0: void RingModulatorAudioProcessor::changeProgramName (int index, const String& newName)
andrewm@0: {
andrewm@0: }
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: void RingModulatorAudioProcessor::prepareToPlay (double sampleRate, int samplesPerBlock)
andrewm@0: {
andrewm@0:     // Calculations that happen before play begins. Pretty simple in this effect, just
andrewm@0:     // reset the previous state.
andrewm@0:     lfoPhase_ = 0.0;
andrewm@0:     carrierPhase_ = 0.0;
andrewm@0:     inverseSampleRate_ = 1.0/sampleRate;
andrewm@0: }
andrewm@0: 
andrewm@0: void RingModulatorAudioProcessor::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: 
andrewm@0: void RingModulatorAudioProcessor::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:     lfoPhase_ = 0.0;
andrewm@0:     carrierPhase_ = 0.0;
andrewm@0: }
andrewm@0: 
andrewm@0: 
andrewm@0: void RingModulatorAudioProcessor::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;
andrewm@0:     float cph, lph;
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:         // 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:         cph = carrierPhase_;
andrewm@0:         lph = lfoPhase_;
andrewm@0:         
andrewm@0:         for (int i = 0; i < numSamples; ++i)
andrewm@0:         {
andrewm@0:             const float in = channelData[i];
andrewm@0: 
andrewm@0:             // Ring modulation is easy! Just multiply the waveform by a periodic carrier
andrewm@0:             channelData[i] = in * sinf(2.0 * M_PI * cph);
andrewm@0: 
andrewm@0:             // Update the carrier and LFO phases, keeping them in the range 0-1
andrewm@0:             lph += lfoFrequency_*inverseSampleRate_;
andrewm@0:             if(lph >= 1.0)
andrewm@0:                 lph -= 1.0;
andrewm@0:             cph += (carrierFrequency_ + sweepWidth_*lfo(lfoPhase_, waveform_))*inverseSampleRate_;
andrewm@0:             if(cph >= 1.0)
andrewm@0:                 cph -= 1.0;
andrewm@0:         }
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:     carrierPhase_ = cph;
andrewm@0:     lfoPhase_ = lph;
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 RingModulatorAudioProcessor::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* RingModulatorAudioProcessor::createEditor()
andrewm@0: {
andrewm@0:     return new RingModulatorAudioProcessorEditor (this);
andrewm@0: }
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: void RingModulatorAudioProcessor::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("carrierFrequency", carrierFrequency_);
andrewm@0:     xml.setAttribute("sweepWidth", sweepWidth_);
andrewm@0:     xml.setAttribute("lfoFrequency", lfoFrequency_);
andrewm@0:     xml.setAttribute("waveform", waveform_);
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 RingModulatorAudioProcessor::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:             carrierFrequency_  = (float)xmlState->getDoubleAttribute("carrierFrequency", carrierFrequency_);
andrewm@0:             sweepWidth_ = (float)xmlState->getDoubleAttribute("sweepWidth", sweepWidth_);
andrewm@0:             lfoFrequency_  = (float)xmlState->getDoubleAttribute("lfoFrequency", lfoFrequency_);
andrewm@0:             waveform_   = xmlState->getIntAttribute("waveform", waveform_);
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 -1 to 1 (note: not 0 to 1 as in delay-based effects)
andrewm@0: float RingModulatorAudioProcessor::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 4.0f*phase;
andrewm@0:             else if(phase < 0.75f)
andrewm@0:                 return 1.0f - 4.0f*(phase - 0.25f);
andrewm@0:             else
andrewm@0:                 return -1.0f + 4.0f*(phase - 0.75f);
andrewm@0:         case kWaveformSquare:
andrewm@0:             if(phase < 0.5f)
andrewm@0:                 return 1.0f;
andrewm@0:             else
andrewm@0:                 return -1.0f;
andrewm@0:         case kWaveformSawtooth:
andrewm@0:             if(phase < 0.5f)
andrewm@0:                 return 2.0f*phase;
andrewm@0:             else
andrewm@0:                 return 2.0f*phase - 2.0f;
andrewm@0:         case kWaveformInverseSawtooth:
andrewm@0:             if(phase < 0.5f)
andrewm@0:                 return -2.0f*phase;
andrewm@0:             else
andrewm@0:                 return 2.0f - 2.0f*phase;
andrewm@0:         case kWaveformSine:
andrewm@0:         default:
andrewm@0:             return 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 RingModulatorAudioProcessor();
andrewm@0: }