Mercurial > hg > audio_effects_textbook_code
view effects/robotisation/Source/PluginProcessor.h @ 0:e32fe563e124
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author | Andrew McPherson <andrewm@eecs.qmul.ac.uk> |
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date | Fri, 10 Oct 2014 15:41:23 +0100 |
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/* This code accompanies the textbook: Digital Audio Effects: Theory, Implementation and Application Joshua D. Reiss and Andrew P. McPherson --- Robotisation: robot effect using phase vocoder See textbook Chapter 8: The Phase Vocoder Code by Andrew McPherson, Brecht De Man and Joshua Reiss This code requires the fftw library version 3 to compile: http://fftw.org --- 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__ #include "../JuceLibraryCode/JuceHeader.h" #include <fftw3.h> //============================================================================== /** */ class RobotisationAudioProcessor : public AudioProcessor { public: //============================================================================== RobotisationAudioProcessor(); ~RobotisationAudioProcessor(); //============================================================================== void prepareToPlay (double sampleRate, int samplesPerBlock); void releaseResources(); 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 { kFFTSizeParam = 0, kHopSizeParam, kWindowTypeParam, kNumParameters }; enum Window { kWindowRectangular = 1, kWindowBartlett, kWindowHann, kWindowHamming }; // This parameter indicates the FFT size for phase vocoder computation. It is selected // by the GUI and may temporarily differ from the actual size used in calculations. int fftSelectedSize_; int hopSelectedSize_; // Hop size, as chosen by user int windowType_; // Type of window used private: // Methods to initialise and de-initialise the FFT machinery void initFFT(int length); void deinitFFT(); // Methods to initialise and de-initialise the window void initWindow(int length, int windowType); void deinitWindow(); // Methods to update the buffering for the given hop size and the output scaling void updateHopSize(); void updateScaleFactor(); // Whether the FFT has been initialised and is therefore ready to go bool fftInitialised_; // Variables for calculating the FFT and IFFT: complex data structures and the // "plan" used by the fftw library to calculate the transforms. fftw_complex *fftTimeDomain_, *fftFrequencyDomain_; fftw_plan fftForwardPlan_, fftBackwardPlan_; // Size of the FFT (generally a power of two) and the hop size (in samples, generally a fraction of FFT size) int fftActualTransformSize_; int hopActualSize_; // Amount by which to scale the inverse FFT to return to original amplitude: depends on the // transform size (because of fftw implementation) and the hop size (because of inherent overlap) double fftScaleFactor_; // Circular buffer gathers audio samples from the input until enough are available // for the FFT calculation AudioSampleBuffer inputBuffer_; int inputBufferLength_; int inputBufferWritePosition_; // Circular buffer that collects output samples from the FFT overlap-add process // before they are ready to be sent to the output stream AudioSampleBuffer outputBuffer_; int outputBufferLength_; int outputBufferReadPosition_, outputBufferWritePosition_; // How many samples since the last FFT? int samplesSinceLastFFT_; // Stored window function for pre-processing input frames double *windowBuffer_; int windowBufferLength_; // Whether or not prepareToPlay() has been called, i.e. that resources are in use bool preparedToPlay_; // Spin lock that prevents the FFT settings from changing in the middle of the audio // thread. SpinLock fftSpinLock_; //============================================================================== JUCE_DECLARE_NON_COPYABLE_WITH_LEAK_DETECTOR (RobotisationAudioProcessor); }; #endif // __PLUGINPROCESSOR_H_4693CB6E__