06-05-2011, 04:23 PM
Code:
#include "xyzcfg.h"
#include "dsk6713.h"
#include "dsk6713_aic23.h"
#define beta 1E-12 //rate of convergence
#define N 30
short int adaptive_filter(short int ,short int );
float delay[N];
float w[N];
//union{unsigned int uint; short channel[2];} AIC23_data;
DSK6713_AIC23_Config config = {\
0x0017, /* 0 DSK6713_AIC23_LEFTINVOL Left line input channel volume */ \
0x0017, /* 1 DSK6713_AIC23_RIGHTINVOL Right line input channel volume */\
0x00d8, /* 2 DSK6713_AIC23_LEFTHPVOL Left channel headphone volume */ \
0x00d8, /* 3 DSK6713_AIC23_RIGHTHPVOL Right channel headphone volume */ \
0x0011, /* 4 DSK6713_AIC23_ANAPATH Analog audio path control */ \
0x0000, /* 5 DSK6713_AIC23_DIGPATH Digital audio path control */ \
0x0000, /* 6 DSK6713_AIC23_POWERDOWN Power down control */ \
0x0043, /* 7 DSK6713_AIC23_DIGIF Digital audio interface format */ \
0x0081, /* 8 DSK6713_AIC23_SAMPLERATE Sample rate control */ \
0x0001 /* 9 DSK6713_AIC23_DIGACT Digital interface activation */ \
};
/*
* main() - Main code routine, initializes BSL and generates tone
*/
void main()
{
DSK6713_AIC23_CodecHandle hCodec;
int l_input, r_input;
int l_output, r_output, T;
/* Initialize the board support library, must be called first */
DSK6713_init();
/* Start the codec */
hCodec = DSK6713_AIC23_openCodec(0, &config);
DSK6713_AIC23_setFreq(hCodec, 1);
for (T = 0; T < 30; T++)
{
w[T] = 0; //init buffer for weights
delay[T] = 0; //init buffer for delay samples
}
while(1)
{
/* Read a sample to the left channel */
while (!DSK6713_AIC23_read(hCodec,&l_input));
/* Read a sample to the right channel */
while (!DSK6713_AIC23_read(hCodec, &r_input));
l_output=(short int)adaptive_filter(l_input,r_input);
r_output=l_output;
/* Send a sample to the left channel */
while (!DSK6713_AIC23_write(hCodec, l_output));
/* Send a sample to the right channel */
while (!DSK6713_AIC23_write(hCodec, r_output));
}
/* Close the codec */
DSK6713_AIC23_closeCodec(hCodec);
}
short int adaptive_filter(short l_input1,short r_input1) //ISR
{
short i,output,T;
float yn=0, E=0, dplusn=0, desired=0, noise=0;
desired = l_input1;
noise = r_input1;
dplusn = desired + noise; //desired+noise
delay[0] = noise; //noise as input to adapt FIR
for (i = 0; i < N; i++) //to calculate out of adapt FIR
yn += (w[i] * delay[i]); //output of adaptive filter
E = (desired + noise) - yn; //"error" signal=(d+n)-yn
for (i = N-1; i >= 0; i--) //to update weights and delays
{
w[i] = w[i] + beta*E*delay[i]; //update weights
delay[i] = delay[i-1]; //update delay samples
}
output=((short)E); //error signal as overall output
//output=((short)dplusn);//output (desired+noise)
//overall output result
return(output);
}PROCEDURE :
Switch on the DSP board.
Open the Code Composer Studio.
Create a new project
Project New (File Name. pjt , Eg: noisecancellation.pjt)
Initialize on board codec.
Note: “Kindly refer the Topic Configuration of 6713 Codec using BSL”
Add the above ‘C’ source file to the current project (remove codec.c source file from the project if you have already added).
Desired Signal 400 Hz
Noise 3.0 KHz
Input a desired sinusoidal signal into the Left channel and Noise signal of 3KHz into the Right channel
Build the project.
Load the generated object file(*.out) on to Target board.
Run the program.
Observe the waveform that appears on the CRO screen. Verify that the 3 KHz noise signal is being cancelled gradually
