path: root/FFT_Test/fft.hcc

/*! \file fft.hcc
 *
 * \section generic	This modules will take care of the actual FFT calculation
 *			on the samples. Besides the FFT this module also will 
 *			equalize the audio signal according to the setting made by the user.
 *
 * \section project Project information.
 * Project Graphic Equalizer\n
 * \author M. Lauwerijssen 
 * \date 20041110
 * \version 0.1
 *
 * \section copyright Copyright
 * Copyright ©2004 Koninklijke Philips Electronics N.V. All rights reserved
 *
 * \section history Change history
 * 20041110: M. Lauwerijssen\n	Initial version
 *
 ********************************************************************/
#define PAL_TARGET_CLOCK_RATE 50000000

/******** System Includes *************/
#include <stdlib.hch>
#include "pal_master.hch"

/******** Application Includes ********/
#include "audio.hch"
#include "weights_256.hch"
#include "configuration.hch"
#include "debug.hch"
#include "xilinxmult.hch"
/* Define two multi-port RAMs for FFT calculation; one for real and one for imaginary values
 * Extra block RAM settings are defined to make sure read and write actions can be performed
 * within one clock-cycle.
 */
#if HARDWARE_MULTIPLY
mpram
{
  ram signed 18 rwrite[256];
  rom signed 18 read[256];
} real with {block = "BlockRAM"};

mpram 
{
  ram signed 18 rwrite[256];
  rom signed 18 read[256];
} imaginary with {block = "BlockRAM"};
#else
mpram
{
  ram signed 24 rwrite[256];
  rom signed 24 read[256];
} real with {block = "BlockRAM"};

mpram 
{
  ram signed 24 rwrite[256];
  rom signed 24 read[256];
} imaginary with {block = "BlockRAM"};
#endif
// multiplication factors for equalizer function
ram signed 7 eq_settings[16] = {0,2,4,7,10,13,16,19,22,26,30,35,41,48,55,63};


/*! \fn		macro proc multiply(result, op_a, op_b);
 * \brief	Procedure used for multiply-ing
 * 
 * \param	result	variable containing the result of the multiply procedure
 * \param	op_a	integer value to be multiplied.
 * \param	op_b	integer value to be multiplied.
 *
 * \return	Procedure returns through variable.
 * \retval	signed 36
 */
macro proc multiply(result, op_a, op_b)
{
#if HARDWARE_MULTIPLY
	xilinxmult(result, op_a, adjs(op_b,18));
#else
	result = (adjs(op_a,38))*(adjs(op_a,38));
#endif	
}

/*
 * Forward declarations
 */
static macro proc FFTRun ();
static macro expr ClockRate = PAL_ACTUAL_CLOCK_RATE;
void calculate_fft(unsigned 1 select_inverse);
void TransferAudio(unsigned 1 import);


chan unsigned 1 AudioOutReady;
extern chan unsigned 1 AudioInReady;

/* Create a dual port RAM */

mpram DualPortInputRam AudioIn with {block = "BlockRAM"};
mpram DualPortOutputRam AudioOut with {block = "BlockRAM"};

#if HARDWARE_MULTIPLY
signed 18 *audioptr_in1,*audioptr_in2,*audioptr_in3,*audioptr_in4;

signed 18 *audioptr_out1,*audioptr_out2;

unsigned 6 *displayptr1,*displayptr2,*displayptr3,*displayptr4;
#else
signed 16 *audioptr_in1,*audioptr_in2,*audioptr_in3,*audioptr_in4;

signed 16 *audioptr_out1,*audioptr_out2;

#endif

/*
 * Main program
 */
void main (void)
{  
        FFTRun ();
}

/*! \fn		macro proc FFTRun();
 * \brief	
 * 
 * \return	Never returns
 * \retval	void
 */
static macro proc FFTRun ()
{
	unsigned 1 select_inverse, sync;
	select_inverse = 0;

  	//pointers for double and quadruple buffering:
  	audioptr_in1 = &AudioIn.fft[0];
  	audioptr_in2 = &AudioIn.fft[64];
  	audioptr_in3 = &AudioIn.fft[128];
  	audioptr_in4 = &AudioIn.fft[192];

  	audioptr_out1 = &AudioOut.fft[0];
  	audioptr_out2 = &AudioOut.fft[64];

	displayptr1 = &audiodata.ifft_info.write[0];
	displayptr2 = &audiodata.ifft_info.write[64];
	displayptr3 = &audiodata.ifft_info.write[128];
	displayptr4 = &audiodata.ifft_info.write[192];

	for(;;)
	{
		//wait for the audiodata to become available
		AudioInReady ? sync;
		par
		{
			// switch pointers 
	    		audioptr_in1 = audioptr_in2;
			audioptr_in2 = audioptr_in3;
			audioptr_in3 = audioptr_in4;
			audioptr_in4 = audioptr_in1;
				
			audioptr_out1 = audioptr_out2;
			audioptr_out2 = audioptr_out1;

			displayptr1=displayptr2;
			displayptr2=displayptr3;
			displayptr3=displayptr4;
			displayptr4=displayptr1;
		}
		
		//perform FFT
		perform_fft(audioptr_in1)
		select_inverse  != select_inverse;
	
		//perform equalize
//		equalize_audio(&audiodata);

		//perform IFFT
		perform_ifft(audioptr_out1,displayptr1);
		select_inverse != select_inverse;	
		
		//Notify the audio I/O module of the completion of the FFT/IFFT
//		AudioOutReady ! 1;
	}
}


/*! \fn		void calculate_fft(unsigned 1 select_inverse)
 * \brief	This routine performs the Fast Fourier Transform for calculation of the frequency spectrum
 * 
 * \param	select_inverse determines if a FFT or iFFT has to be calculated
 *
 * \return	nothing
 * \retval	void
 *
 * cost	12391 cycles
 */
void calculate_fft(unsigned 1 select_inverse)
{
  unsigned 4 level;
  unsigned 8 point1,point2,j,f,k;
  unsigned 9 e,i;
  signed 16 weight1,weight2;
#if HARDWARE_MULTIPLY
  signed 18 p,q,r,t;
#else
  signed 24 p,q,r,t;	
#endif
  signed a,b;

#if HARDWARE_MULTIPLY
  // Macro to provide rescaling of 36-bit result of fixed point multiply
  // down to an 18-bit result. The range of bits selected depends on the 
  // number that represents the value of "1" in the trig function lookup
  // tables. (Eg. for 16384 == 1, the lowest bit selected should be [14]).
  macro expr rescale (x) = (x[35] @ x[30:14]);
#else
  //Macro to rescale the multiply result down to a 24-bit value.
  macro expr rescale (x) = ((x>>FRACBITS)<-24);
#endif

  for(level=1;level<=NUMBER_OF_COLUMNS;level++) // count all the columns
  {  
	e=1<<(NUMBER_OF_COLUMNS-level+1); // number of points in each block in this column
    	f=(e>>1)<-8; 			  // number of butterflies in each block in this column

    	for(j=1;j<=f;j++) 	// count all the butterflies in each block
    	{
		par
		{
      			// Weight factors for real (the same for FFT and iFFT)
	  		weight1 = weight_re[((j-1)<<(level-1))<-7]; 

			
			// Weight factors for imaginary (opposite for FFT and iFFT)
	  		weight2 = (!select_inverse) ? (weight_im[((j-1)<<(level-1))<-7]) : -(weight_im[((j-1)<<(level-1))<-7]); 

			/* ORIGINAL CODE BELOW, MODIFIED BECAUSE OF MISMATCHING OUTPUT WITH BORLAND TESTAPP
			weight2 = (!select_inverse) ? -(weight_im[((j-1)<<(level-1))<-7]) : weight_im[((j-1)<<(level-1))<-7]; 
			*/
			
			

			for(i=0@j;i<=NUMBER_OF_POINTS;i+=e)   // count all the blocks in this column
			{	
				// Butterfly calculation
        			par
				{
					point1 = ((i<-8)-1);
	      				point2 = (((i<-8)+f)-1);
	    			}
				
				par
				{
					p = (real.read[point1] >> 1) + (real.rwrite[point2] >> 1);
		  			q = (imaginary.read[point1] >> 1) + (imaginary.rwrite[point2] >> 1);
				}
				
				par
				{
		      			r = (real.read[point1] >> 1) - (real.rwrite[point2] >> 1);
					t = (imaginary.read[point1] >> 1) - (imaginary.rwrite[point2] >> 1);
				}		

				multiply(a,r,weight1);
				multiply(b,t,weight2);

       		        	par
 				{
        	  			real.rwrite[point2] = (rescale(a-b));
          				imaginary.rwrite[point1] = q;
    				}

				multiply(a,t,weight1);
				multiply(b,r,weight2);

				par
				{	
					real.rwrite[point1] = p;
		  			imaginary.rwrite[point2] = (rescale(a+b));
		  		}

		  	}
		}
	}
  }
  j=1;
  for(i=1;i<NUMBER_OF_POINTS;i++)
  {
  	if(i<(0@j))
   	{
   		par
		{
			point1=j-1;
   			point2=(i-1)<-8;
		}
		/*
		  COPYING ARRAY VALUES FROM ONE PLACE TO ANOTHER IN THE ARRAT MUST BE DONE IN 
		  2 STEPS. FIRSTLY THE VALUES ARE COPIED TO SEPARATE VARIABLES AFTER THAT THEY
		  ARE COPIED BACK TO THEIR NEW POSITION IN THE ARRAY. THIS MUST BE DONE TO 
		  PREVENT TIMING ISSUES FROM OCCURING.
		*/
   		par
		{
			p = real.read[point1];
   			q = imaginary.read[point1];
		}
		par
		{
			r = real.read[point2];
   			t = imaginary.read[point2];
		}
		par
		{
			real.rwrite[point1] = r;   	
   			imaginary.rwrite[point1] = t;
		}
  		par
		{
			real.rwrite[point2] = p;
   			imaginary.rwrite[point2] = q;
		}
   	}

  	k = NUMBER_OF_POINTS>>1;


  	while(k<j)
   	{
   		j = j-k;
   		k = k>>1;
   	}

  	j+=k;
  }
}

/*! \fn		void perform_fft(signed 18 *pcm_audio)
 * \brief	This routine obtains the audio data from the audio I/O component and copies this  	   
 * 		data to local arrays for calculating purposes, and calls the FFT algorithm.
 * 
 * \param	*pcm_audio pointer to array containg the audio data
 *
 * \return	nothing
 * \retval	void
 *
 * cost	258 cycles (excl. the calculate FFT function)
 */
#if HARDWARE_MULTIPLY
void perform_fft(signed 18 *pcm_audio)
#else
void perform_fft(signed 16 *pcm_audio)
#endif
{
  	unsigned 8 k;
#if HARDWARE_MULTIPLY
	signed 18 sample;
	k=0;
	sample = adjs(pcm_audio[k],18);
#else
	signed 24 sample;
	k=0;
	sample = adjs(pcm_audio[k],24);
#endif
    	
	//initialize variables for the copying pipeline

	
	// copy audio data to real-array before starting FFT calculation
	// and set imaginary values to zero
	do
	{
		//Copying the array values has been pipelined to prevent parallel access to the
		//pcm_audio array. This copying procedure must be finished before another 
		//sample is read from the audio input. The time available for this loop is 
		//determined by the sampling rate of 44,1 Khz
		par
		{
			//COPYING NEEDS TO BE DONE IN 2 STEPS, BECAUSE THE VALUE THAT NEEDS TO WRITTEN
			//TO THE REAL-RAM NEEDS TO BE AVAILABLE ON THE START OFF THE CLOCKCYCLE.
#if HARDWARE_MULTIPLY
			sample = adjs(pcm_audio[k+1],18);
#else
			sample = adjs(pcm_audio[k+1],24);
#endif
			real.rwrite[k] = sample;
      			imaginary.rwrite[k] = 0;
			k++;
		}		
    	}  while (k);

	

#if PERFORM_FFT_CALCULATION
	calculate_fft(0);
#endif


}

/*! \fn		void perform_ifft(signed 18 *modified_audio, unsigned 6 *ifft_info)
 * \brief	This routine calls the ifft algorithm and after completing that	it obtains the 
 * 		modified audio data and copies that to the output arrays of the audio I/O component. 
 * 		Besides that it also fills the array used by the display routine for displaying the waveform.
 * 
 * \param	*modified_audio pointer to array containg the audio data
 * \param	*ifft_info Pointer to the ifft_info array containing the modified waveform data for display purposes
 *
 * \return	nothing
 * \retval	void
 *
 * cost	258 cycles (excl. the calculate iFFT function)
 */
#if HARDWARE_MULTIPLY
void perform_ifft(signed 18 *modified_audio, unsigned 6 *ifft_info)
#else
void perform_ifft(signed 16 *modified_audio, unsigned 6 *ifft_info)
#endif
{
  	unsigned 6 k;
#if HARDWARE_MULTIPLY 
  	signed 18 p;
#else
	signed 24 p;
#endif
#if PERFORM_FFT_CALCULATION	
	calculate_fft(1);
#endif

    	k=0;
//initialize variables for the copying pipeline
#if PERFORM_FFT_CALCULATION	
	#if HARDWARE_MULTIPLY 
		p = (real.read[(0@k)+95] << NUMBER_OF_COLUMNS);
	#else
		p = (real.read[(0@k)+95] >> NUMBER_OF_COLUMNS);
	#endif
#else
		p = (real.read[(0@k)+95]);
#endif

	do
	{
		//Copying the array values has been pipelined to prevent parallel access to the
		//pcm_audio array. This copying procedure must be finished before another 
		//sample is read from the audio input. The time available for this loop is 
		//determined by the sampling rate of 44,1 Khz
	   	par
		{
			/*
			*	Before copying the modified audio from the local real-array 
			*	to the output array of the audio I/O component, compensate
			*	for the FFT calculation by shifting the values. 
			*	95 is added to start the output from the middle of the sliding
			*	window, this is done to get a better sound quality.
			*/
#if PERFORM_FFT_CALCULATION	
	#if HARDWARE_MULTIPLY 
			p = (real.read[(0@k)+95] << NUMBER_OF_COLUMNS);
	#else
			p = (real.read[(0@k)+95] >> NUMBER_OF_COLUMNS);
	#endif
#else
			p = (real.read[(0@k)+95]);
#endif
			//Copy the modified audio from the local real array to the output array of the audio I/O component.
#if HARDWARE_MULTIPLY
      			modified_audio[k] = p ;
#else
			modified_audio[k] = (p<-16);
#endif
			//Fill the array for displaying the waveform, only the 6 MSB are needed.
			ifft_info[k] = (unsigned 6)(32+(p[17:12]));

			k++;
		}
    	} while(k);
}

/*! \fn		void equalize_audio(audiodata_t *audiodata)
 * \brief	This routine equalizes the frequencies derived by the FFT calculation, 
 *		according to the settings of the equalizer bars.
 * 
 * \param	*audiodata pointer to the audiodata struct, containing the eq_info, etc.
 *
 * \return	nothing
 * \retval	void
 *
 * cost		3844 cycles (Maximum)
 *
 */
void equalize_audio(audiodata_t *audiodata)
{
#if HARDWARE_MULTIPLY
  signed 18 p,q;
#else
  signed 24 p,q;
#endif
  signed 18 a;
  unsigned 8 i, mirror_i, bit, m, n;
  unsigned 7 old_value;
  unsigned 9 tmp;
  
  //macro expr equalize_bar = multiply(q,a)[29:6];
  
  macro proc equalize_bar(retval)
  {
  	 signed result;
 	 multiply(result, q,a);
#if HARDWARE_MULTIPLY
	 retval = result[23:6]; //drop last 6 bit to compensate the maximum multiplication with 64 from the eq_settings array
#else
	 retval = result[29:6]; //drop last 6 bit to compensate the maximum multiplication with 64 from the eq_settings array
#endif
  } 

  p = real.read[0] - DC_COMPONENT; // remove DC component for calculations
  real.rwrite[0] = p;
  
  for(i=0;i!=NUMBER_OF_FREQUENCIES;i++)	  
  {  
	
	par
	{	
		// set multiplication factor (0..64) for current frequency bar, The first frequency band must be equalized at 100% (63) since there is no DC-component taken into account.
		a = adjs(eq_settings[audiodata->equalizer_levels_ptr[i <- 7]],18);


		// multiply frequency with this factor and divide by 64 (drop 6 LSB's)
		q = real.read[i];
	}
	equalize_bar(p);
	real.rwrite[i] = p;

    	q = imaginary.read[i];
	equalize_bar(p);
	imaginary.rwrite[i] = p;

	// the upper part(128..255) of the spectrum is mirrored to the lower part; 
	// these values need to be adjusted too
	if ((i<-7)!=0) // if not in DC component bar
	{
		mirror_i = (NUMBER_OF_POINTS-1)-i+1;
		q = real.read[mirror_i];
		equalize_bar(p);
		real.rwrite[mirror_i] = p;

		q = imaginary.read[mirror_i];
		equalize_bar(p);
		imaginary.rwrite[mirror_i] = p;
	}
  }
  
  //write data to fft_info for display purposes
  for(i=0;i<NUMBER_OF_FREQUENCIES;i++)
  {
   	par
	{
		p = real.read[i];
    		q = imaginary.read[i];
#if HARDWARE_MULTIPLY
    		if (p[17] == 1) p = -p; else delay;
		if (q[17] == 1) q = -q; else delay;
#else
	    	if (p[23] == 1) p = -p; else delay;
		if (q[23] == 1) q = -q; else delay;
#endif
	}
	p = (p<q) ? q : p; // This is done to get the best visual frequency result
	 
	if (!audiodata->display_log)
	{

		bit=126;
#if HARDWARE_MULTIPLY
		while ((p[15] == 0) && (bit != 0))
#else
		while ((p[21] == 0) && (bit != 0))
#endif
			par
			{
				p = p<<1;
				bit = bit - 18;
			}
		old_value = audiodata->fft_info.write[0 @ (i <- 7)];
		tmp = ((0@old_value) + (0@bit))>>1;
		audiodata->fft_info.write[0 @ (i <- 7)] = (old_value <= (tmp<-7)) ? (tmp<-7) : old_value-1;
	} 
	else 
	{
		old_value = audiodata->fft_info.write[0 @ (i <- 7)];
#if HARDWARE_MULTIPLY
		audiodata->fft_info.write[0 @ (i <- 7)] = (old_value<=(unsigned)(p[15:9])) ? (unsigned)(p[15:9]) : old_value-1;
#else
		audiodata->fft_info.write[0 @ (i <- 7)] = (old_value<=(unsigned)(p[21:15])) ? (unsigned)(p[21:15]) : old_value-1;
#endif
	}
  }

  // add DC component again before inverse FFT calculation is performed

  p = real.read[0] + DC_COMPONENT; 
  real.rwrite[0] = p;
}