Re:stuck with code for FSK(V.21) modem

#include "tonecfg.h"

#include "dsk5510.h"

#include "dsk5510_aic23.h"

#include<stdio.h>

#include<math.h>

/* Codec configuration settings */

DSK5510_AIC23_Config config = {

    0x0017, // 0 DSK5510_AIC23_LEFTINVOL  Left line input channel volume

    0x0017, // 1 DSK5510_AIC23_RIGHTINVOL Right line input channel volume

    0x00FF, // 2 DSK5510_AIC23_LEFTHPVOL  Left channel headphone volume

    0x00FF, // 3 DSK5510_AIC23_RIGHTHPVOL Right channel headphone volume

    0x0011, // 4 DSK5510_AIC23_ANAPATH    Analog audio path control

    0x0000, // 5 DSK5510_AIC23_DIGPATH    Digital audio path control

    0x0000, // 6 DSK5510_AIC23_POWERDOWN  Power down control

    0x0043, // 7 DSK5510_AIC23_DIGIF      Digital audio Int16erface format

    0x0081, // 8 DSK5510_AIC23_SAMPLERATE Sample rate control

    0x0001  // 9 DSK5510_AIC23_DIGACT     Digital Int16erface activation

};

#define FIRLEN 19 

float coefs[FIRLEN] = {-0.008370237662365, 0.003523724616947,  0.01237363596323,  0.02713952454333, 0.04697784958127,  0.06994028726489,  0.09309959888495,   0.1130209563111,  0.1265124921103,    0.131279039604,   0.1265124921103,   0.1130209563111,   0.09309959888495,  0.06994028726489,  0.04697784958127,  0.02713952454333,    0.01237363596323, 0.003523724616947,-0.008370237662365};

//Int16 inputs[30] = {1.13128697433765 ,1.10984462553037 ,1.38344070298819 ,0.626150903279837 ,0.274450597351357 ,0.580994546781292 ,0.237492756839462 ,0.679663383970494 ,0.322571475796019 ,1.21779302841047 ,0.570692318798823 ,1.16763875860796 ,0.653554635345044 ,0.714598072144237 ,0.726021551753316 ,0.905479767550582 ,0.00435839319756108 ,0.479972769744528 ,0.113617074520466 ,1.09094320060140 ,1.12589149609019 ,0.521013993015299 ,0.763457292330922 ,0.108789020498579 ,0.920646227924225 ,1.15623623159461 ,0.0791557324943119 ,1.09534727278226 ,0.148991578063601 ,0.6170647215283747}; 

extern void circbuff();      // defined in asmfunctions.asm

float *inPtr; //poInt16er to input sample

float *outPtr; //poInt16er to output

float *coeff; //poInt16er to the coefficients

//Int16 leftsample, rightsample;

//Int16 input=0;

float output=0;

Int16 sample1;

Int16 j,i; 

float mixerReal[120],mixerImg[120],mixer[120];

float mixerReal_param , mixerImg_param;

void filter_input(float*,float*,Int16);

void main()

{

    DSK5510_AIC23_CodecHandle hCodec;

    /* Initialize the board support library, must be called first */

    DSK5510_init();

    /* Start the codec */

    hCodec = DSK5510_AIC23_openCodec(0, &config);

DSK5510_AIC23_setFreq(hCodec, DSK5510_AIC23_FREQ_16KHZ);

   coeff = &coefs[0]; //coefs is coefficient of impulse response defined in fdacoefs_Int16.h

   inPtr = &mixer[0]; //inPtr is a globally declared poInt16er to a Int16 

   //inPtr = &mixerReal[0];

   outPtr = &output;

   

for ( j = 0; j < 120; j++ ) 

{

filter_input(&mixerReal_param,&mixerImg_param,j);

mixerReal[j]=mixerReal_param;

mixerImg[j]=mixerImg_param;

mixer[j]=mixerReal_param + mixerImg_param;

} 

   

 

   

   

for(i=0;;i++)

{

for(sample1=0;sample1<120;sample1++)

{

 //while (!DSK5510_AIC23_read16(hCodec, mixer[sample1]));

 //while (!DSK5510_AIC23_read16(hCodec, mixer[sample1]));

      circbuff();

      

      

      

      

      

}      

      while (!DSK5510_AIC23_write16(hCodec, output));

      while (!DSK5510_AIC23_write16(hCodec, output));

}

   

    DSK5510_AIC23_closeCodec(hCodec);

}

void filter_input(float*mixerReal_dummy,float*mixerImg_dummy,Int16 j_dummy)

{

    float bitperiod = (1.0 / 300.0);

    float fs = 9000, f1 = 1650, f0 = 1850, fc = 1750;

    float temp  = 0.0;

    Int16 ii,jj,k=0;

    

    float input[4]={1.0,0.0,1.0,0.0};

    float modwave[120];

    float Cos[ 4 ], Sin [ 4 ];

    

    Int16 len = 0;

    float sample[30];

    for ( temp = (1/(fs+0.0)); temp <= bitperiod; temp += (1/(fs+0.0)) )

    {

         sample[len++] = ii;

    }

   

   ////////////////////////////////////////////////////////////////////////////////////

   ////////////////***   Modulation   ***//////////////////////////////////////////////

   ////////////////////////////////////////////////////////////////////////////////////

    

    //Int16 modwave[len * 4];

   // Int16 j=0, jj;

    for( ii = 0; ii < 4; ii++ ) {

         if(input[ii]) {

                for( jj = 0; jj < len; jj++ ) {

                     modwave[k++] = sin ( 2.0 * 3.1416 * (f1 + 0.0) * sample[jj] );

                     //prInt16f("%f ",modwave[j-1]);    

                } 

         } else { 

                for( jj = 0; jj < len; jj++ ) {

                     modwave[k++] = sin ( 2.0 * 3.1416 * (f0 + 0.0) * sample[jj] );

                     //prInt16f("%f ",modwave[j-1]);    

                }

         }

    }

           

   ////////////////////////////////////////////////////////////////////////////////////

   ////////////////***   Mixxing      ***//////////////////////////////////////////////

   ////////////////////////////////////////////////////////////////////////////////////

         

   //Int16 Cos[ 4 ], Sin [ 4 ];

      

   for ( jj = 0; jj < len; jj++ ) {

       Cos[jj] = cos( 2.0 * 3.1416 * (fc+0.0) * sample[jj] );

       Sin[jj] = sin( 2.0 * 3.1416 * (fc+0.0) * sample[jj] );

   }

      

  // Int16 mixerReal_dummy[len * 4], mixerImg_dummy[len * 4];

   

      *mixerReal_dummy = modwave[j_dummy] * Cos[j_dummy%len];

      *mixerImg_dummy = modwave[j_dummy] * Sin[j_dummy%len];

   

  // return (*mixerReal_dummy,*mixerImg_dummy);

   

}    

//ASM code for filtering:

Can anybody sort out my problem?

Thanks in advance.

Suk