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:   PVOC Passthrough: phase vocoder structure which passes input
andrewm@0:   to output without performing any processing
andrewm@0:  
andrewm@0:   See textbook Chapter 8: The Phase Vocoder
andrewm@0:  
andrewm@0:   Code by Andrew McPherson, Brecht De Man and Joshua Reiss
andrewm@0:  
andrewm@0:   This code requires the fftw library version 3 to compile:
andrewm@0:   http://fftw.org
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: #include "../JuceLibraryCode/JuceHeader.h"
andrewm@0: #include <fftw3.h>
andrewm@0: 
andrewm@0: //==============================================================================
andrewm@0: /**
andrewm@0: */
andrewm@0: class PVOCPassthroughAudioProcessor  : public AudioProcessor
andrewm@0: {
andrewm@0: public:
andrewm@0:     //==============================================================================
andrewm@0:     PVOCPassthroughAudioProcessor();
andrewm@0:     ~PVOCPassthroughAudioProcessor();
andrewm@0: 
andrewm@0:     //==============================================================================
andrewm@0:     void prepareToPlay (double sampleRate, int samplesPerBlock);
andrewm@0:     void releaseResources();
andrewm@0: 
andrewm@0:     void processBlock (AudioSampleBuffer& buffer, MidiBuffer& midiMessages);
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:         kFFTSizeParam = 0,
andrewm@0:         kHopSizeParam,
andrewm@0:         kWindowTypeParam,
andrewm@0:         kNumParameters
andrewm@0:     };
andrewm@0:     
andrewm@0:     enum Window
andrewm@0:     {
andrewm@0:         kWindowRectangular = 1,
andrewm@0:         kWindowBartlett,
andrewm@0:         kWindowHann,
andrewm@0:         kWindowHamming
andrewm@0:     };
andrewm@0:     
andrewm@0:     enum HopSize
andrewm@0:     {
andrewm@0:         kHopSize1Window = 1,
andrewm@0:         kHopSize1_2Window,
andrewm@0:         kHopSize1_4Window,
andrewm@0:         kHopSize1_8Window
andrewm@0:     };
andrewm@0:     
andrewm@0:     // This parameter indicates the FFT size for phase vocoder computation. It is selected
andrewm@0:     // by the GUI and may temporarily differ from the actual size used in calculations.
andrewm@0:     int fftSelectedSize_;
andrewm@0:     int hopSelectedSize_;       // Hop size, chosen from one of the options above
andrewm@0:     int windowType_;            // Type of window used
andrewm@0:     
andrewm@0: private:
andrewm@0:     // Methods to initialise and de-initialise the FFT machinery
andrewm@0:     void initFFT(int length);
andrewm@0:     void deinitFFT();
andrewm@0:     
andrewm@0:     // Methods to initialise and de-initialise the window
andrewm@0:     void initWindow(int length, int windowType);
andrewm@0:     void deinitWindow();
andrewm@0:     
andrewm@0:     // Methods to update the buffering for the given hop size and the output scaling
andrewm@0:     void updateHopSize();
andrewm@0:     void updateScaleFactor();
andrewm@0:     
andrewm@0:     // Whether the FFT has been initialised and is therefore ready to go
andrewm@0:     bool fftInitialised_;
andrewm@0:     
andrewm@0:     // Variables for calculating the FFT and IFFT: complex data structures and the
andrewm@0:     // "plan" used by the fftw library to calculate the transforms.
andrewm@0:     fftw_complex *fftTimeDomain_, *fftFrequencyDomain_;
andrewm@0:     fftw_plan fftForwardPlan_, fftBackwardPlan_;
andrewm@0:     
andrewm@0:     // Size of the FFT (generally a power of two) and the hop size (in samples, generally a fraction of FFT size)
andrewm@0:     int fftActualTransformSize_;
andrewm@0:     int hopActualSize_;
andrewm@0:     
andrewm@0:     // Amount by which to scale the inverse FFT to return to original amplitude: depends on the
andrewm@0:     // transform size (because of fftw implementation) and the hop size (because of inherent overlap)
andrewm@0:     double fftScaleFactor_;
andrewm@0:     
andrewm@0:     // Circular buffer gathers audio samples from the input until enough are available
andrewm@0:     // for the FFT calculation
andrewm@0:     AudioSampleBuffer inputBuffer_;
andrewm@0:     int inputBufferLength_;
andrewm@0:     int inputBufferWritePosition_;
andrewm@0:     
andrewm@0:     // Circular buffer that collects output samples from the FFT overlap-add process
andrewm@0:     // before they are ready to be sent to the output stream
andrewm@0:     AudioSampleBuffer outputBuffer_;
andrewm@0:     int outputBufferLength_;
andrewm@0:     int outputBufferReadPosition_, outputBufferWritePosition_;
andrewm@0:     
andrewm@0:     // How many samples since the last FFT?
andrewm@0:     int samplesSinceLastFFT_;
andrewm@0:     
andrewm@0:     // Stored window function for pre-processing input frames
andrewm@0:     double *windowBuffer_;
andrewm@0:     int windowBufferLength_;
andrewm@0:     
andrewm@0:     // Whether or not prepareToPlay() has been called, i.e. that resources are in use
andrewm@0:     bool preparedToPlay_;
andrewm@0:     
andrewm@0:     // Spin lock that prevents the FFT settings from changing in the middle of the audio
andrewm@0:     // thread.
andrewm@0:     SpinLock fftSpinLock_;
andrewm@0:     
andrewm@0:     //==============================================================================
andrewm@0:     JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (PVOCPassthroughAudioProcessor);
andrewm@0: };
andrewm@0: 
andrewm@0: #endif  // __PLUGINPROCESSOR_H_4693CB6E__