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:   Phaser: phasing effect using time-varying allpass filters
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: #ifndef __PLUGINPROCESSOR_H_4693CB6E__
andrewm@0: #define __PLUGINPROCESSOR_H_4693CB6E__
andrewm@0: 
andrewm@0: #define _USE_MATH_DEFINES
andrewm@0: #include "../JuceLibraryCode/JuceHeader.h"
andrewm@0: #include "OnePoleAllpassFilter.h"
andrewm@0: #include "TwoPoleAllpassFilter.h"
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: /**
andrewm@0: */
andrewm@0: class PhaserAudioProcessor  : public AudioProcessor
andrewm@0: {
andrewm@0: public:
andrewm@0:     //==============================================================================
andrewm@0:     PhaserAudioProcessor();
andrewm@0:     ~PhaserAudioProcessor();
andrewm@0: 
andrewm@0:     //==============================================================================
andrewm@0:     void prepareToPlay (double sampleRate, int samplesPerBlock);
andrewm@0:     void releaseResources();
andrewm@0:     void reset();
andrewm@0:     void processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages);
andrewm@0:     
andrewm@0: 
andrewm@0:     //==============================================================================
andrewm@0:     AudioProcessorEditor* createEditor();
andrewm@0:     bool hasEditor() const;
andrewm@0: 
andrewm@0:     //==============================================================================
andrewm@0:     const String getName() const;
andrewm@0: 
andrewm@0:     int getNumParameters();
andrewm@0: 
andrewm@0:     float getParameter (int index);
andrewm@0:     void setParameter (int index, float newValue);
andrewm@0: 
andrewm@0:     const String getParameterName (int index);
andrewm@0:     const String getParameterText (int index);
andrewm@0: 
andrewm@0:     const String getInputChannelName (int channelIndex) const;
andrewm@0:     const String getOutputChannelName (int channelIndex) const;
andrewm@0:     bool isInputChannelStereoPair (int index) const;
andrewm@0:     bool isOutputChannelStereoPair (int index) const;
andrewm@0: 
andrewm@0:     bool silenceInProducesSilenceOut() const;
andrewm@0:     double getTailLengthSeconds() const;    
andrewm@0:     bool acceptsMidi() const;
andrewm@0:     bool producesMidi() const;
andrewm@0: 
andrewm@0:     //==============================================================================
andrewm@0:     int getNumPrograms();
andrewm@0:     int getCurrentProgram();
andrewm@0:     void setCurrentProgram (int index);
andrewm@0:     const String getProgramName (int index);
andrewm@0:     void changeProgramName (int index, const String& newName);
andrewm@0: 
andrewm@0:     //==============================================================================
andrewm@0:     void getStateInformation (MemoryBlock& destData);
andrewm@0:     void setStateInformation (const void* data, int sizeInBytes);
andrewm@0:     
andrewm@0:     //==============================================================================
andrewm@0:     
andrewm@0:     // these are used to persist the UI's size - the values are stored along with the
andrewm@0:     // filter's other parameters, and the UI component will update them when it gets
andrewm@0:     // resized.
andrewm@0:     int lastUIWidth_, lastUIHeight_;
andrewm@0:     
andrewm@0:     enum Parameters
andrewm@0:     {
andrewm@0:         kBaseFrequencyParam = 0,
andrewm@0:         kSweepWidthParam,
andrewm@0:         kDepthParam,
andrewm@0:         kFeedbackParam,
andrewm@0:         kLFOFrequencyParam,
andrewm@0:         kFiltersParam,
andrewm@0:         kWaveformParam,
andrewm@0:         kStereoParam,
andrewm@0:         kNumParameters
andrewm@0:     };
andrewm@0:     
andrewm@0:     enum Waveforms
andrewm@0:     {
andrewm@0:         kWaveformSine = 1,
andrewm@0:         kWaveformTriangle,
andrewm@0:         kWaveformSquare,
andrewm@0:         kWaveformSawtooth,
andrewm@0:         kNumWaveforms
andrewm@0:     };
andrewm@0:     
andrewm@0:     // Adjustable parameters:
andrewm@0:     float baseFrequency_; // Lowest frequency of allpass filters
andrewm@0:     float sweepWidth_;    // Amount of change from min to max delay
andrewm@0:     float depth_;         // Mix level for phase-shifted signal (0-1)
andrewm@0:     float feedback_;      // Feedback level for feedback phaser (0-<1)
andrewm@0:     float lfoFrequency_;  // LFO frequency (Hz)
andrewm@0:     int   filtersPerChannel_; // How many allpass filters to use
andrewm@0:     int   waveform_;      // What shape should be used for the LFO
andrewm@0:     int   stereo_;        // Whether to use stereo phasing
andrewm@0:     
andrewm@0: private:
andrewm@0:     float lfo(float phase, int waveform);
andrewm@0:     void allocateFilters();   // Create the filter objects...
andrewm@0:     void deallocateFilters(); // Delete them...
andrewm@0:     void reallocateFilters(); // Delete and rebuild in one combined operation
andrewm@0:     
andrewm@0:     CriticalSection lock_; // Lock for updating number of filters
andrewm@0:     
andrewm@0:     float lfoPhase_;   // Phase of the low-frequency oscillator
andrewm@0:     double inverseSampleRate_; // It's more efficient to multiply than divide, so
andrewm@0:                                // cache the inverse of the sample rate
andrewm@0:     unsigned int sampleCount_; // Counter to keep track of samples elapsed, to
andrewm@0:                                // keep track of filter updates
andrewm@0:     unsigned int filterUpdateInterval_; // How often to update filter coefficients
andrewm@0:     
andrewm@0:     // Bank of allpass filters that do the phasing; N filters x M channels
andrewm@0:     OnePoleAllpassFilter **allpassFilters_;
andrewm@0:     
andrewm@0:     // Storage of the last output sample from each bank of filters, for use in
andrewm@0:     // feedback loop
andrewm@0:     float *lastFilterOutputs_;
andrewm@0:     int numLastFilterOutputs_;
andrewm@0: 
andrewm@0:     int totalNumFilters_;
andrewm@0:     
andrewm@0:     //==============================================================================
andrewm@0:     JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (PhaserAudioProcessor);
andrewm@0: };
andrewm@0: 
andrewm@0: #endif  // __PLUGINPROCESSOR_H_4693CB6E__