/*
*
      
_______
                       
_____
   
_____ _____
  

*
     
|__
   
__|
                     
|
  
__ \ / ____|__ \
 

*
        
| | __ _ _ __ ___
  
______| || | (___ | |__) |
*
        
| |/ _` | '__/ __|/ _ \/ __| |
  
| |\___ \|___/
 

*
        
| | (_| | |
  
\__ \ (_) \__ \ |__| |____) | |
     

*
        
|_|\__,_|_|
  
|___/\___/|___/_____/|_____/|_|
     

*
                                                         

* -------------------------------------------------------------
*
* 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.resample;

import be.tarsos.dsp.AudioDispatcher;
import be.tarsos.dsp.AudioEvent;
import be.tarsos.dsp.AudioProcessor;

/**
 
* Currently not working sample rate transposer, works only for integer factors.
 
* Changes sample rate by using linear interpolation.
 
*
 
* Together with the time stretcher this can be used for pitch shifting.
 
* @author Joren Six
 
* @author Olli Parviainen
 
*/

public class SoundTouchRateTransposer implements AudioProcessor {

	
private double rate;
	
int slopeCount;
    
double prevSample;
    
private AudioDispatcher dispatcher;

    
public void setDispatcher(AudioDispatcher newDispatcher){
		
this.dispatcher = newDispatcher;
	
}
    

	
public SoundTouchRateTransposer(double d){
		
this.rate = d;
	
}
	

	
@Override
	
public boolean process(AudioEvent audioEvent) {
		
int i, used;
		
float[] src = audioEvent.getFloatBuffer();
		
float[] dest = new float[(int) Math.round(audioEvent.getBufferSize() / rate)];
	    
used = 0;
	    
i = 0;

	    
// Process the last sample saved from the previous call first...
	    
while (slopeCount <= 1.0f) {
	        
dest[i] = (float)((1.0f - slopeCount) * prevSample + slopeCount * src[0]);
	        
i++;
	        
slopeCount += rate;
	    
}
	    
slopeCount -= 1.0f;
	    
end:
        
while(true){
            
while (slopeCount > 1.0f) {
                
slopeCount -= 1.0f;
                
used++;
                
if (used >= src.length - 1)
                	
break end;
            
}
            
if(i < dest.length){
            	
dest[i] = (float)((1.0f - slopeCount) * src[used] + slopeCount * src[used + 1]);
            
}
            
i++;
            
slopeCount += rate;
        
}
	    

	    
//Store the last sample for the next round
	    
prevSample = src[src.length - 1];
	    
dispatcher.setStepSizeAndOverlap(dest.length, 0);
	    
audioEvent.setFloatBuffer(dest);
		
return true;
	
}

	
@Override
	
public void processingFinished() {

	
}

}