diff effects/autowah/Source/PluginProcessor.cpp @ 0:e32fe563e124

First commit

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/effects/autowah/Source/PluginProcessor.cpp	Fri Oct 10 15:41:23 2014 +0100
@@ -0,0 +1,459 @@
+/*
+  This code accompanies the textbook:
+ 
+  Digital Audio Effects: Theory, Implementation and Application
+  Joshua D. Reiss and Andrew P. McPherson
+ 
+  ---
+ 
+  Auto-Wah: LFO or envelope-operated wah effect
+  See textbook Chapter 4: Filter 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"
+
+// The filter will produce a resonant peak of amplitude Q; bring everything
+// down somewhat to compensate, though try to maintain some perceptual balance
+// of being similar loudness. (This factor has been chosen somewhat arbitrarily.)
+const double kWahwahFilterGain = 0.5;
+
+//==============================================================================
+AutoWahAudioProcessor::AutoWahAudioProcessor()
+{
+    // Set default values:
+    baseFrequency_ = 350.0;
+    q_ = 5.0;
+    lfoFrequency_ = 2.0;
+    lfoWidth_ = 1000.0;
+    envelopeWidth_ = 0.0;
+    envelopeAttack_ = 0.005;
+    envelopeDecay_ = 0.1;
+    
+    // Initialise the filters later when we know how many channels
+    wahFilters_ = 0;
+    numWahFilters_ = 0;
+    envelopes_ = 0;
+    numEnvelopes_ = 0;
+    attackMultiplier_ = 1.0;
+    decayMultiplier_ = 0.0;
+    
+    inverseSampleRate_ = 1.0/44100.0; // start with a sensible default
+    
+    lastUIWidth_ = 550;
+    lastUIHeight_ = 200;
+}
+
+AutoWahAudioProcessor::~AutoWahAudioProcessor()
+{
+    deallocateFilters();
+}
+
+//==============================================================================
+const String AutoWahAudioProcessor::getName() const
+{
+    return JucePlugin_Name;
+}
+
+int AutoWahAudioProcessor::getNumParameters()
+{
+    return kNumParameters;
+}
+
+float AutoWahAudioProcessor::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 kBaseFrequencyParam:  return baseFrequency_;
+        case kQParam:              return q_;
+        case kLFOFrequencyParam:   return lfoFrequency_;
+        case kLFOWidthParam:       return lfoWidth_;
+        case kEnvelopeWidthParam:  return envelopeWidth_;
+        case kEnvelopeAttackParam: return envelopeAttack_;
+        case kEnvelopeDecayParam:  return envelopeDecay_;
+        default:                   return 0.0f;
+    }
+}
+
+void AutoWahAudioProcessor::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 kBaseFrequencyParam:
+            baseFrequency_ = newValue;
+            break;
+        case kQParam:
+            q_ = newValue;
+            break;
+        case kLFOFrequencyParam:
+            lfoFrequency_ = newValue;
+            break;
+        case kLFOWidthParam:
+            lfoWidth_ = newValue;
+            break;
+        case kEnvelopeWidthParam:
+            envelopeWidth_ = newValue;
+            break;
+        case kEnvelopeAttackParam:
+            envelopeAttack_ = newValue;
+            // See comment below for justification
+            if(envelopeAttack_ == 0.0)
+                attackMultiplier_ = 0.0;
+            else
+                attackMultiplier_ = pow(1.0 / M_E, inverseSampleRate_ / envelopeAttack_);
+            break;
+        case kEnvelopeDecayParam:
+            envelopeDecay_ = newValue;
+            // envelopeDecay_ sets the time constant tau. The decay is
+            // given as e^-(t/tau) so after tau seconds, it will have
+            // decayed to 1/e of its original value. tau*sampleRate samples
+            // will have passed by then, each of which multiplies the signal
+            // by decayMultiplier_.
+            if(envelopeDecay_ == 0.0)
+                decayMultiplier_ = 0.0;
+            else
+                decayMultiplier_ = pow(1.0 / M_E, inverseSampleRate_ / envelopeDecay_);
+            break;
+        default:
+            break;
+    }
+}
+
+const String AutoWahAudioProcessor::getParameterName (int index)
+{
+    switch (index)
+    {
+        case kBaseFrequencyParam:  return "base frequency";
+        case kQParam:              return "Q";
+        case kLFOFrequencyParam:   return "LFO frequency";
+        case kLFOWidthParam:       return "LFO width";
+        case kEnvelopeWidthParam:  return "envelope width";
+        case kEnvelopeAttackParam: return "envelope attack";
+        case kEnvelopeDecayParam:  return "envelope decay";
+        default:                   break;
+    }
+    
+    return String::empty;
+}
+
+const String AutoWahAudioProcessor::getParameterText (int index)
+{
+    return String (getParameter (index), 2);
+}
+
+const String AutoWahAudioProcessor::getInputChannelName (int channelIndex) const
+{
+    return String (channelIndex + 1);
+}
+
+const String AutoWahAudioProcessor::getOutputChannelName (int channelIndex) const
+{
+    return String (channelIndex + 1);
+}
+
+bool AutoWahAudioProcessor::isInputChannelStereoPair (int index) const
+{
+    return true;
+}
+
+bool AutoWahAudioProcessor::isOutputChannelStereoPair (int index) const
+{
+    return true;
+}
+
+bool AutoWahAudioProcessor::silenceInProducesSilenceOut() const
+{
+#if JucePlugin_SilenceInProducesSilenceOut
+    return true;
+#else
+    return false;
+#endif
+}
+
+double AutoWahAudioProcessor::getTailLengthSeconds() const
+{
+    return 0.0;
+}
+
+bool AutoWahAudioProcessor::acceptsMidi() const
+{
+#if JucePlugin_WantsMidiInput
+    return true;
+#else
+    return false;
+#endif
+}
+
+bool AutoWahAudioProcessor::producesMidi() const
+{
+#if JucePlugin_ProducesMidiOutput
+    return true;
+#else
+    return false;
+#endif
+}
+
+int AutoWahAudioProcessor::getNumPrograms()
+{
+    return 0;
+}
+
+int AutoWahAudioProcessor::getCurrentProgram()
+{
+    return 0;
+}
+
+void AutoWahAudioProcessor::setCurrentProgram (int index)
+{
+}
+
+const String AutoWahAudioProcessor::getProgramName (int index)
+{
+    return String::empty;
+}
+
+void AutoWahAudioProcessor::changeProgramName (int index, const String& newName)
+{
+}
+
+//==============================================================================
+void AutoWahAudioProcessor::prepareToPlay (double sampleRate, int samplesPerBlock)
+{
+    // Use this method as the place to do any pre-playback
+    // initialisation that you need..
+    
+    allocateFilters();
+    inverseSampleRate_ = 1.0 / sampleRate;
+    if(envelopeDecay_ == 0.0)
+        decayMultiplier_ = 0.0;
+    else
+        decayMultiplier_ = pow(1.0 / M_E, inverseSampleRate_ / envelopeDecay_);
+    if(envelopeAttack_ == 0.0)
+        attackMultiplier_ = 0.0;
+    else
+        attackMultiplier_ = pow(1.0 / M_E, inverseSampleRate_ / envelopeAttack_);
+}
+
+void AutoWahAudioProcessor::releaseResources()
+{
+    // When playback stops, you can use this as an opportunity to free up any
+    // spare memory, etc.
+
+    deallocateFilters();
+}
+
+void AutoWahAudioProcessor::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;
+    float ph;
+    
+    // Go through each channel and put it through the resonant lowpass filter, updating
+    // the coefficients as we go along. Each channel is processed identically in this effect.
+    
+    for(channel = 0; channel < jmin(numInputChannels, numWahFilters_); ++channel)
+    {
+        // channelData is an array of length numSamples which contains the audio for one channel
+        float* channelData = buffer.getSampleData(channel);
+        ph = lfoPhase_;
+        
+        for (int sample = 0; sample < numSamples; ++sample)
+        {
+            const float in = channelData[sample];
+            float centreFrequency = baseFrequency_;
+            
+            // Calculate the envelope of the signal. Do this even if we're not currently
+            // changing the frequeny based on it, since it involves maintaining a history
+            // of the signal's behaviour.
+            
+            if(channel < numEnvelopes_) {   // Safety check
+                if(fabs(in) > envelopes_[channel]) {
+                    envelopes_[channel] += (1.0 - attackMultiplier_) * (fabs(in) - (double)envelopes_[channel]);
+                }
+                else
+                    envelopes_[channel] *= decayMultiplier_;
+            }
+            
+            // Calculate the centre frequency of the filter based on the LFO and the
+            // signal envelope
+            if(lfoWidth_ > 0.0) {
+                centreFrequency += lfoWidth_ * (0.5f + 0.5f*sinf(2.0 * M_PI * ph));
+            }
+            if(envelopeWidth_ > 0.0 && channel < numEnvelopes_) {
+                centreFrequency += envelopeWidth_ * envelopes_[channel];
+            }
+            
+            // Update filter coefficients (see ResonantLowpassFilter.cpp for calculation)
+            wahFilters_[channel]->makeResonantLowpass(inverseSampleRate_,
+                                                      centreFrequency,
+                                                      q_,
+                                                      kWahwahFilterGain);
+            
+            // Process one sample and store it back in place. See juce_IIRFilter.cpp for the
+            // application of the IIR filter.
+            channelData[sample] = wahFilters_[channel]->processSingleSampleRaw(in);
+            
+            // Update the LFO phase, keeping it in the range 0-1
+            ph += lfoFrequency_*inverseSampleRate_;
+            if(ph >= 1.0)
+                ph -= 1.0;
+        }
+    }
+    
+    lfoPhase_ = ph;
+    
+    // Go through the remaining channels. In case we have more outputs
+    // than inputs, or there aren't enough filters, we'll clear any
+    // remaining output channels (which could otherwise contain garbage)
+    while(channel < numOutputChannels)
+    {
+        buffer.clear (channel++, 0, buffer.getNumSamples());
+    }
+}
+
+//==============================================================================
+bool AutoWahAudioProcessor::hasEditor() const
+{
+    return true; // (change this to false if you choose to not supply an editor)
+}
+
+AudioProcessorEditor* AutoWahAudioProcessor::createEditor()
+{
+    return new AutoWahAudioProcessorEditor (this);
+}
+
+//==============================================================================
+void AutoWahAudioProcessor::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("baseFrequency", baseFrequency_);
+    xml.setAttribute("q", q_);
+    xml.setAttribute("lfoFrequency", lfoFrequency_);
+    xml.setAttribute("lfoWidth", lfoWidth_);
+    xml.setAttribute("envelopeWidth", envelopeWidth_);
+    xml.setAttribute("envelopeAttack", envelopeAttack_);
+    xml.setAttribute("envelopeDecay", envelopeDecay_);
+    
+    // then use this helper function to stuff it into the binary blob and return it..
+    copyXmlToBinary(xml, destData);
+}
+
+void AutoWahAudioProcessor::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_);
+            
+            q_ = (float)xmlState->getDoubleAttribute("q", q_);
+            baseFrequency_ = (float)xmlState->getDoubleAttribute("baseFrequency", baseFrequency_);
+            lfoFrequency_ = (float)xmlState->getDoubleAttribute("lfoFrequency", lfoFrequency_);
+            lfoWidth_ = (float)xmlState->getDoubleAttribute("lfoWidth", lfoWidth_);
+            envelopeWidth_ = (float)xmlState->getDoubleAttribute("envelopeWidth", envelopeWidth_);
+            envelopeAttack_ = (float)xmlState->getDoubleAttribute("envelopeAttack", envelopeAttack_);
+            envelopeDecay_ = (float)xmlState->getDoubleAttribute("envelopeDecay", envelopeDecay_);
+            inverseSampleRate_ = 1.0 / getSampleRate();
+            if(envelopeDecay_ == 0.0)
+                decayMultiplier_ = 0.0;
+            else
+                decayMultiplier_ = pow(1.0 / M_E, inverseSampleRate_ / envelopeDecay_);
+            if(envelopeAttack_ == 0.0)
+                attackMultiplier_ = 0.0;
+            else
+                attackMultiplier_ = pow(1.0 / M_E, inverseSampleRate_ / envelopeAttack_);
+        }
+    }
+}
+
+void AutoWahAudioProcessor::allocateFilters()
+{
+    // Prevent leaks from reallocation
+    if(wahFilters_ != 0 || envelopes_ != 0)
+        deallocateFilters();
+    
+    // Create as many filters as we have input channels
+    numWahFilters_ = getNumInputChannels();
+    wahFilters_ = (ResonantLowpassFilter**)malloc(numWahFilters_ * sizeof(ResonantLowpassFilter*));
+    if(wahFilters_ == 0)
+        numWahFilters_ = 0;
+    else {
+        for(int i = 0; i < numWahFilters_; i++)
+            wahFilters_[i] = new ResonantLowpassFilter;
+    }
+    
+    numEnvelopes_ = getNumInputChannels();
+    envelopes_ = (double *)malloc(numEnvelopes_ * sizeof(double));
+    if(envelopes_ == 0)
+        numEnvelopes_ = 0;
+    else {
+        for(int i = 0; i < numEnvelopes_; i++)
+            envelopes_[i] = 0.0;
+    }
+}
+
+void AutoWahAudioProcessor::deallocateFilters()
+{
+    for(int i = 0; i < numWahFilters_; i++)
+        delete wahFilters_[i];
+    if(numWahFilters_ != 0)
+        free(wahFilters_);
+    numWahFilters_ = 0;
+    wahFilters_ = 0;
+    if(envelopes_ != 0)
+        free(envelopes_);
+    envelopes_ = 0;
+    numEnvelopes_ = 0;
+}
+
+//==============================================================================
+// This creates new instances of the plugin..
+AudioProcessor* JUCE_CALLTYPE createPluginFilter()
+{
+    return new AutoWahAudioProcessor();
+}