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core.be.tarsos.dsp.synthesis.PitchResyntheziser Maven / Gradle / Ivy

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/*
*      _______                       _____   _____ _____  
*     |__   __|                     |  __ \ / ____|  __ \ 
*        | | __ _ _ __ ___  ___  ___| |  | | (___ | |__) |
*        | |/ _` | '__/ __|/ _ \/ __| |  | |\___ \|  ___/ 
*        | | (_| | |  \__ \ (_) \__ \ |__| |____) | |     
*        |_|\__,_|_|  |___/\___/|___/_____/|_____/|_|     
*                                                         
* -------------------------------------------------------------
*
* TarsosDSP is developed by Joren Six at IPEM, University Ghent
*  
* -------------------------------------------------------------
*
*  Info: http://0110.be/tag/TarsosDSP
*  Github: https://github.com/JorenSix/TarsosDSP
*  Releases: http://0110.be/releases/TarsosDSP/
*  
*  TarsosDSP includes modified source code by various authors,
*  for credits and info, see README.
* 
*/

package be.tarsos.dsp.synthesis;

import java.util.Arrays;

import be.tarsos.dsp.AudioEvent;
import be.tarsos.dsp.EnvelopeFollower;
import be.tarsos.dsp.pitch.PitchDetectionHandler;
import be.tarsos.dsp.pitch.PitchDetectionResult;

/**
 * This pitch detection handler replaces the audio buffer in the pipeline with a
 * synthesized wave. It either follows the envelope of the original signal or
 * not. Use it wisely. The following demonstrates how it can be used.
 * 
 * 
 * 
 * PitchEstimationAlgorithm algo = PitchEstimationAlgorithm.FFT_YIN;
 * PitchResyntheziser prs = new PitchResyntheziser(samplerate);
 * AudioDispatcher dispatcher = AudioDispatcher.fromFile(new File("in.wav"),1024, 0);
 * //Handle pitch detection
 * dispatcher.addAudioProcessor(new PitchProcessor(algo, samplerate, size, prs));
 * //Write the synthesized pitch to an output file.
 * dispatcher.addAudioProcessor(new WaveformWriter(format, "out.wav"));//
 * dispatcher.run();
 * 
 * 
* * @author Joren Six */ public class PitchResyntheziser implements PitchDetectionHandler { private double phase = 0; private double phaseFirst = 0; private double phaseSecond = 0; private double prevFrequency = 0; private float samplerate; private final EnvelopeFollower envelopeFollower; private boolean usePureSine; private boolean followEnvelope; private final double[] previousFrequencies; private int previousFrequencyIndex; public PitchResyntheziser(float samplerate){ this(samplerate,true,false); } public PitchResyntheziser(float samplerate,boolean followEnvelope,boolean pureSine){ this(samplerate,followEnvelope,pureSine,5); } public PitchResyntheziser(float samplerate,boolean followEnvelope,boolean pureSine,int filterSize){ envelopeFollower = new EnvelopeFollower(samplerate,0.005,0.01); this.followEnvelope=followEnvelope; this.usePureSine = pureSine; this.samplerate = samplerate; previousFrequencies = new double[filterSize]; previousFrequencyIndex = 0; } @Override public void handlePitch(PitchDetectionResult pitchDetectionResult, AudioEvent audioEvent) { double frequency = pitchDetectionResult.getPitch(); if(frequency==-1){ frequency=prevFrequency; }else{ if(previousFrequencies.length!=0){ //median filter //store and adjust pointer previousFrequencies[previousFrequencyIndex] = frequency; previousFrequencyIndex++; previousFrequencyIndex %= previousFrequencies.length; //sort to get median frequency double[] frequenciesCopy = previousFrequencies.clone(); Arrays.sort(frequenciesCopy); //use the median as frequency frequency = frequenciesCopy[frequenciesCopy.length/2]; } prevFrequency = frequency; } final double twoPiF = 2 * Math.PI * frequency; float[] audioBuffer = audioEvent.getFloatBuffer(); float[] envelope = null; if(followEnvelope){ envelope = audioBuffer.clone(); envelopeFollower.calculateEnvelope(envelope); } for (int sample = 0; sample < audioBuffer.length; sample++) { double time = sample / samplerate; double wave = Math.sin(twoPiF * time + phase); if(!usePureSine){ wave += 0.05 * Math.sin(twoPiF * 4 * time + phaseFirst); wave += 0.01 * Math.sin(twoPiF * 8 * time + phaseSecond); } audioBuffer[sample] = (float) wave; if(followEnvelope){ audioBuffer[sample] = audioBuffer[sample] * envelope[sample]; } } double timefactor = twoPiF * audioBuffer.length / samplerate; phase = timefactor + phase; if(!usePureSine){ phaseFirst = 4 * timefactor + phaseFirst; phaseSecond = 8 * timefactor + phaseSecond; } } }




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