comparison effects/phaser/Source/PluginProcessor.h @ 0:e32fe563e124

First commit

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