TABLEOFCONTENTS

PAGE
S.NO. DATE EXPERIMENTTITLE SIGN.
NO
Generation of elementary Discrete-Time sequences
1
Linear and Circular convolution

2
Auto correlation and Cross Correlation
3
Frequency Analysis using DFT
4
Design of FIR filters (LPF/HPF/BPF/BSF) and
demonstrates the filteringoperation
5
Design of Butterworth and Chebyshev IIR filters
6 (LPF/HPF/BPF/BSF) anddemonstrate the filtering
operations
Study of architecture of Digital Signal Processor
7
Study of architecture of Digital Signal Process
8
Perform MAC operation using various addressing modes
9
10 Generation of various signals and random noise

11 Design and demonstration of FIR Filter for Low pass,

High pass,Band pass and Band stop filtering

12 Design and demonstration of Butter worth and Chebyshev

IIR Filtersfor Low pass, High pass, Band pass and Band
stop filtering
13 Design and demonstration of Butter worth and
Chebyshev IIR Filtersfor Low pass, High pass, Band
pass and Band stop filtering

14 Implement an Up-sampling and Down-sampling

operation in DSPProcessor
EXP.NO:1 GENERATION OF ELEMENTARY DISCRETE-TIME SEQUENCES

AIM:

To write a program to generate the elementary discrete time sequences using MATLAB.

SOFTWARE REQUIRED:

MATLAB R2014a

ALGORITHM:

 Get the input for the required sequences.

 Generate the sequence.
 Plot the corresponding sequences.

PROGRAM:

%Program for sine wave

Clc;

Clearall;
Closeall;
N = 7;
n = 0:1:N-1;
y = ones(1,N);
subplot(3,2,1);
stem(n,y);
xlabel('time');
ylabel('amplitude');
title('unit step seqence');
N1 = 5;
n1 = 0:1:N-1;
y1 = n1;
subplot(3,2,2);
stem(n1,y1);
xlabel('time');
ylabel('amplitude');
title('unit ramp sequence');
N2 = 6;
n2 = 0:0.1:N-1;

2
y2 = sin(2*pi*n2);
subplot(3,2,3);
stem(n2,y2);
xlabel('time');
ylabel('amplitude');
title('sinusoidal seqeuence');
N3 = 4;
n3 = 0:0.1:N3-1;
y3 = cos(2*pi*n3);
subplot(3,2,4);
stem(n3,y3);
xlabel('time');
ylabel('amplitude');
title('cosine sequence');
N4 = 5;
n4 = 0:0.1:N4-1;
a = 3;
y4 = exp(a*n4);
subplot(3,2,5);
stem(n4,y4);
xlabel('time');
ylabel('amplitude');
title('exponential sequence');
n5 = -3:1:3;
y5 = [zeros(1,3),ones(1,1),zeros(1,3)];
subplot(3,2,6);
stem(n5,y5);
xlabel('time');
ylabel('amplitude');
title('unit impluse');

3
OUTPUT:

unit step seqence unit ramp sequence

1 6
amplitude

amplitude
4
0.5
2

0 0
0 1 2 3 4 5 6 0 1 2 3 4 5 6
time time
sinusoidal seqeuence cosine sequence
1 1

0.5 0.5
amplitude

amplitude
0 0

-0.5 -0.5

-1 -1
0 1 2 3 4 5 6 0 0.5 1 1.5 2 2.5 3
time time
5
x 10 exponential sequence unit impluse
2 1

1.5
amplitude

amplitude
1 0.5

0.5

0 0
0 0.5 1 1.5 2 2.5 3 3.5 4 -3 -2 -1 0 1 2 3
time time

RESULT:

Thus the elementary discrete time sequences are generated and plottedusing MATLAB.

4
EXP.NO: 2 LINEAR AND CIRCULAR CONVOLUTIONS

AIM:

To write a program to performLinear and Circular Convolution of two sequences using

MATLAB.

SOFTWARE REQUIRED:

MATLAB R2014a

ALGORITHM:

 Get the input sequence x (n) and impulse sequence h (n).

 Perform the convolution of two sequences.
 Plot the convoluted sequences.

PROGRAM:

LINEAR CONVOLUTION:

clc;
clearall;
closeall;
x=input('Enter the input sequence');
h=input('Enter the impulse sequence');
y=conv(x,h);
subplot(2,2,1);
stem(x);
xlabel('n');
ylabel('amplitude');
title('input sequence');
subplot(2,2,2);
stem(h);
xlabel('n');
ylabel('amplitude');
title('impulse sequence');
subplot(2,2,3);
stem(y);
xlabel('n');
ylabel('amplitude');
title('convoluted sequence');
disp('Convoluted sequence');y

5
INPUT :

Enter the input sequence[1 2 3 4]

Enter the impulsesequence[5 6 7 8]
Convoluted sequence

y= 5 16 34 60 61 52 32

OUTPUT:

input sequence impulse sequence

4 8

3 6
amplitude

amplitude

2 4

1 2

0 0
1 2 3 4 1 2 3 4
n n
convoluted sequence
80

60
amplitude

40

20

0
0 2 4 6 8
n

6
CIRCULAR CONVOLUTION

clc;
clearall;
closeall;
x=input('Enter the input sequence');
h=input('Enter the impulse sequence');
N1=length(x);
N2=length(h);
N=max(N1,N2);
N3=N1-N2;
if (N3>=0);
h=[h,zeros(1,N3)];
else
x=[x,zeros(1,N3)];
end
for n=1:N;
y(n)=0;
for i=1:N;
j=n-i+1;
if (j<=0)
j=N+j;
end
y(n)=y(n)+[x(i)*h(j)];
end
end
subplot(1,3,1);
stem(y);
xlabel('n');
ylabel('amplitude');
title('convoluted sequence');
disp('Convoluted sequence');y
subplot(1,3,2);
stem(x);
xlabel('n');
ylabel('amplitude');
title('input sequence');
subplot(1,3,3);
stem(h);
xlabel('n');
ylabel('amplitude');
title('impulse sequence');

7
INPUT:

Enter the inputsequence[1 2 3 4]

Enter the impulsesequence[4 3 2 1]
Convoluted sequence

y= 24 22 24 30

OUTPUT :

convoluted sequence input sequence impulse sequence

30 4 4

3.5 3.5
25

3 3

20
2.5 2.5
amplitude

amplitude

amplitude
15 2 2

1.5 1.5
10

1 1

5
0.5 0.5

0 0 0
0 2 4 0 2 4 0 2 4
n n n

RESULT:

Thus the linear and circular convolutions of two sequences wereperformed using
MATLAB.

8
EXP NO: 3 AUTO CORRELATION AND CROSS CORRELATION

AIM:

To write a program to perform the Autocorrelation and cross correlation of two

sequences using MATLAB.

SOFTWARE REQUIRED:

MATLAB R2014a

ALGORITHM:

 Get the required input sequences.

 Perform the correlation of two sequences.
 Plot the correlated sequences.

PROGRAM:

AUTOCORRELATION:
clc;
clearall;
closeall;
x=input('Enter the input sequence');
y=xcorr(x,x);
subplot(2,1,1);
stem(x);
ylabel('amplitude');
xlabel('x(n)');
subplot(2,1,2);
stem(y);
ylabel('amplitude');
xlabel('y(n)');
disp('The resultant signal is');y

9
INPUT:

Enter the inputsequence[1 2 3 4]

The resultant signal is

y= 4.0000 11.0000 20.0000 30.0000 20.0000 11.0000 4.0000

OUTPUT:

3
amplitude

0
1 1.5 2 2.5 3 3.5 4
x(n)

30

25

20
amplitude

15

10

0
1 2 3 4 5 6 7
y(n)

10
CROSS CORRELATION:

clc;
clearall;
closeall;
x=input('Enter the first sequence');
h=input('Enter the second sequence');
y=xcorr(x,h);
subplot(3,1,1);
stem(x);
ylabel('amplitude');
xlabel('x(n)');
subplot(3,1,2);
stem(h);
ylabel('amplitude');
xlabel('h(n)');
subplot(3,1,3);
stem(y);
ylabel('amplitude');
xlabel('y(n)');
disp('The resultant signal is:');y

11
INPUT:

Enter the first sequence [1 2 3 4]

Enter the second sequence[4 5 6 7]
The resultant signal is:

y= 7.0000 20.0000 38.0000 60.0000 47.0000 32.0000 16.0000

OUTPUT:

4
amplitude

0
1 1.5 2 2.5 3 3.5 4
x(n)
10
amplitude

0
1 1.5 2 2.5 3 3.5 4
h(n)
100
amplitude

50

0
1 2 3 4 5 6 7
y(n)

RESULT:

Thus the autocorrelation and cross correlation of two sequences were performed
using MATLAB.

12
EXP.NO:4 FREQUENCY ANALYSIS USING DFT

AIM:

To writea MATLAB program for frequency analysis using DFT.

SOFTWARE REQUIRED:

MATLAB R2014a

ALGORITHM:

 Get the input sequence x (n).

 Obtain DFT of the input sequence(resultant sequence) using FFT.
 Plot the resultant sequence.
 Calculate the magnitude and phase values of resultant signal.
 Plot the magnitude and phase plots.

PROGRAM:

clc;
clearall;
closeall;
xn=input('enter the input sequence');
XK = fft (xn);
subplot(1,4,1);
stem(xn);
xlabel('n');
ylabel('amplitude');
title('input sequence');
subplot(1,4,2);
stem(XK)
xlabel('n');
ylabel('amplitude');
title('output sequence');
disp('resultant sequence XK');XK
subplot(1,4,3);
stem(abs(XK));
xlabel('k');
ylabel('magnitude of x(K)');
title('magnitude plot');
subplot(1,4,4);
stem(angle(XK));
xlabel('k');
ylabel('angle of x(K)');
title('phase plot');

13
INPUT:

Enter the input sequence[1 1 1 1]

Resultant sequence XK

XK = 4 0 0 0

OUTPUT:

input sequence output sequence magnitude plot phase plot

1 4 4 1

0.9 0.8
3.5 3.5

0.8 0.6
3 3
0.7 0.4

2.5 2.5
magnitude of x(K)

0.6 0.2

angle of x(K)
amplitude

amplitude

0.5 2 2 0

0.4 -0.2
1.5 1.5

0.3 -0.4
1 1
0.2 -0.6

0.5 0.5
0.1 -0.8

0 0 0 -1
0 2 4 0 2 4 0 2 4 0 2 4
n n k k

RESULT:

Thus the frequency analysis using DFT was performed using MATLAB.

14
EXP.NO:5(a) DESIGN OF FIR FILTER USING HAMMING WINDOW

AIM:

To write a MATLAB program to design a FIR filters(LPF/HPF/BPF/BSF) and

demonstrates the filtering operation using hamming window.

SOFTWARE REQUIRED

MATLAB R2014a

ALGORITHM:

 Get the FIR filter specifications.

 Obtain the filter coefficients using window function.
 Plot the frequency response of the filters.

PROGRAM:

clc;
clearall;
closeall;
rp=input('Enter the passband ripple');
rs=input('Enter the stopband ripple');
fp=input('Enter the passband frequency');
fs=input('Enter the stopband frequency');
f=input('Enter the sampling frequency');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(rp*rs))-13;
dem=14.6*(fs-fp)/f;
n=ceil(num/dem);
n1=n+1;
if(rem(n,2)~=0)
n1=n;
n=n-1;
end
y=hamming(n1);
disp('The window coefficient are as follows');y
b= fir1(n,wp,y);
disp('unit sample response of fir filter is h(n)=');b
disp(b);b
[h,o]=freqz(b,1,256);

15
m=20*log(abs(h));
subplot(2,2,1);
plot(o/pi,m);
ylabel('gain in dB');
xlabel('(a)normalized frequency');
title('LPF');
b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,2);
plot(o/pi,m);
ylabel('gain in dB');
xlabel('(b)normalized frequency');
title('HPF');
wn=[wpws];
b=fir1(n,wn,y);
[h,o]=freqz(b,1,256);
m=20*log(abs(h));
subplot(2,2,3);
plot(o/pi,m);
ylabel('gain in dB');
xlabel('(c)normalized frequency');
title('BPF');
wn=[wpws];
b=fir1(n,wn,'stop',y);
[h,o]=freqz(b,1,256);
m=20*log(abs(h));
subplot(2,2,4);
plot(o/pi,m);
ylabel('gain in dB');
xlabel('(d)normalized frequency');
title('BSF');

16
INPUT:

Enter the passband ripple0.01

Enter the stopband ripple0.02
Enter the passband frequency1000
Enter the stopband frequency2000
Enter the sampling frequency5000
The window coefficient are as follows

y=
0.0800
0.2147
0.5400
0.8653
1.0000
0.8653
0.5400
0.2147
0.0800
unit sample response of fir filter is h(n)=
b=

Columns 1 through 8

-0.0061 -0.0136 0.0512 0.2657 0.4057 0.2657 0.0512 -0.0136

Column 9

-0.0061

Columns 1 through 8

-0.0061 -0.0136 0.0512 0.2657 0.4057 0.2657 0.0512 -0.0136

Column 9

-0.0061

b=

Columns 1 through 8

-0.0061 -0.0136 0.0512 0.2657 0.4057 0.2657 0.0512 -0.0136

Column 9

-0.0061

17
OUTPUT:

LPF HPF
0 0

-50 -10
gain in dB

gain in dB
-100 -20

-150 -30

-200 -40
0 0.5 1 0 0.5 1
(a) normalized frequency (b) normalized frequency
BPF BSF
50 0

0 -10
gain in dB

gain in dB

-50 -20

-100 -30
0 0.5 1 0 0.5 1
(c) normalized frequency (d) normalized frequency

RESULT:

Thus the FIR filter using hamming window is designed using MATLAB.

18
EXP.NO:5(b) DESIGN OF FIR FILTER USING HANNING WINDOW

AIM:

To write a MATLAB program for design a FIR filters(LPF/HPF/BPF/BSF) and

demonstrates the filtering operation usinghanning window.

SOFTWARE REQUIRED:

MATLAB R2014a

ALGORITHM:

 Get the FIR filter specifications.

 Obtain the filter coefficients using window function.
 Plot the frequency response of the filters.

PROGRAM:

clc;
clearall;
closeall;
rp=input('Enter the passband ripple');
rs=input('Enter the stopband ripple');
fp=input('Enter the passband frequency');
fs=input('Enter the stopband frequency');
f=input('Enter the sampling frequency');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(rp*rs))-13;
dem=14.6*(fs-fp)/f;
n=ceil(num/dem);
n1=n+1;
if(rem(n,2)~=0)
n1=n;
n=n-1;
end
y=hanning(n1);
disp('the window coefficient are as follows');y
b= fir1(n,wp,y);
disp('unit sample response of fir filter is h(n)=');b
disp(b);b
19
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1);
plot(o/pi,m);
ylabel('gain in dB');
xlabel('(a)normalized frequency');
title('LPF');
b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log(abs(h));
subplot(2,2,2);
plot(o/pi,m);
ylabel('gain in dB');
xlabel('(b)normalized frequency');
title('HPF');
wn=[wpws];
b=fir1(n,wn,y);
[h,o]=freqz(b,1,256);
m=20*log(abs(h));
subplot(2,2,3);
plot(o/pi,m);
ylabel('gain in dB');
xlabel('(c)normalized frequency');
title('BPF');
wn=[wpws];
b=fir1(n,wn,'stop',y);
[h,o]=freqz(b,1,256);
m=20*log(abs(h));
subplot(2,2,4);
plot(o/pi,m);
ylabel('gain in dB');
xlabel('(d)normalized frequency');
title('BSF');

20
INPUT:

Enter the passband ripple 0.01

Enter the stopband ripple0.02
Enter the passband frequency1000
Enter the stopband frequency2000
Enter the sampling frequency5000
Ehe window coefficient are as follows
y=
0.0955
0.3455
0.6545
0.9045
1.0000
0.9045
0.6545
0.3455
0.0955
unit sample response of fir filter is h(n)=

b=
Columns 1 through 8

-0.0071 -0.0213 0.0605 0.2704 0.3950 0.2704 0.0605 -0.0213

Column 9

-0.0071

Columns 1 through 8

-0.0071 -0.0213 0.0605 0.2704 0.3950 0.2704 0.0605 -0.0213

Column 9

-0.0071

b=

Columns 1 through 8

-0.0071 -0.0213 0.0605 0.2704 0.3950 0.2704 0.0605 -0.0213

Column 9

-0.0071

21
OUTPUT:

LPF HPF
0 50

-20 0
gain in dB

gain in dB
-40 -50

-60 -100

-80 -150
0 0.5 1 0 0.5 1
(a) normalized frequency (b) normalized frequency
BPF BSF
0 0

-50 -10
gain in dB

gain in dB

-100 -20

-150 -30

-200 -40
0 0.5 1 0 0.5 1
(c) normalized frequency (d) normalized frequency

RESULT:

Thus the FIR filter using hanning window is designed using MATLAB.

22
EXP.NO:5(c) DESIGN OF FIR FILTER USING KAISER WINDOW

AIM:

To write a MATLAB program for design a FIR filters (LPF/HPF/BPF/BSF) and

demonstrate the filtering operation using Kaiser Window.

SOFTWARE REQUIRED:

MATLAB R2014a

ALGORITHM:

 Get the FIR filter specifications.

 Obtain the filter coefficients using window function.
 Plot the frequency response of the filters.

PROGRAM:

clc;
clearall;
closeall;
rp=input('Enter the passband ripple');
rs=input('Enter the stopband ripple');
fp=input('Enter the passband frequency');
fs=input('Enter the stopband frequency');
f=input('Enter the sampling frequency');
beta=input(‘enter the beta value’);
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(rp*rs))-13;
dem=14.6*(fs-fp)/f;
n=ceil(num/dem);
n1=n+1;
if(rem(n,2)~=0)
n1=n;
n=n-1;
end
y=kaiser(n1,beta);
disp('The window coefficient are as follows');y
b= fir1(n,wp,y);
disp('unit sample response of fir filter is h(n)=');b
disp(b);b

23
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1);
plot(o/pi,m);
ylabel('gain in dB');
xlabel('(a)normalized frequency');
title('LPF');
b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log(abs(h));
subplot(2,2,2);
plot(o/pi,m);
ylabel('gain in dB');
xlabel('(b)normalized frequency');
title('HPF');
wn=[wpws];
b=fir1(n,wn,y);
[h,o]=freqz(b,1,256);
m=20*log(abs(h));
subplot(2,2,3);
plot(o/pi,m);
ylabel('gain in dB');
xlabel('(c)normalized frequency');
title('BPF');
wn=[wpws];
b=fir1(n,wn,'stop',y);
[h,o]=freqz(b,1,256);
m=20*log(abs(h));
subplot(2,2,4);
plot(o/pi,m);
ylabel('gain in dB');
xlabel('(d)normalized frequency');
title('BSF');

24
INPUT:

Enter the passband ripple 0.02

Enter the stopband ripple0.04
Enter the passband frequency1000
Enter the stopband frequency2000
Enter the sampling frequency8000
Enter the beta value 2
The window coefficient are as follows
y=
0.9403
0.9616
0.9783
0.9903
0.9976
1.0000
0.9976
0.9903
0.9783
0.9616
0.9403

unit sample response of fir filter is h(n)=

b = Columns 1 through 8

-0.0393 0.0000 0.0682 0.1464 0.2086 0.2322 0.2086 0.1464

Columns 9 through 11

0.0682 0.0000 -0.0393

Columns 1 through 8

-0.0393 0.0000 0.0682 0.1464 0.2086 0.2322 0.2086 0.1464

Columns 9 through 11

0.0682 0.0000 -0.0393

b = Columns 1 through 8

-0.0393 0.0000 0.0682 0.1464 0.2086 0.2322 0.2086 0.1464

Columns 9 through 11

0.0682 0.0000 -0.0393

25
OUTPUT:
LPF HPF
0 50

gain in dB -20 0

gain in dB
-40 -50

-60 -100

-80 -150
0 0.5 1 0 0.5 1
(a) normalized frequency (b) normalized frequency
BPF BSF
100 50

0
0

gain in dB
gain in dB

-50
-100
-100

-200 -150
0 0.5 1 0 0.5 1
(c) normalized frequency (d) normalized frequency

RESULT:

Thus the FIR filter using Kaiser Window was designed using MATLAB.

26
EXP.NO:5(d) DESIGN OF FIR FILTER USING RECTANGULAR WINDOW

AIM:

To write a MATLAB program for design a FIR filters (LPF/HPF/BPF/BSF) and

demonstrates the filtering operation using Rectangular Window.

SOFTWARE REQUIRED:

MATLAB R2014a

ALGORITHM:

 Get the FIR filter specifications.

 Obtain the filter coefficients using window function.
 Plot the frequency response of the filters.

PROGRAM:

clc;
clear all;
close all;
rp=input('enter the pass band ripple');
rs=input('enter the stop band ripple');
fp=input('enter the pass band frequency');
fs=input('enter the stop band frequency');
f=input('enter the sampling frequency');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(rp*rs))-13;
dem=14.6*(fs-fp)/f;
n=ceil(num/dem);
% computation for odd or even
n1=n+1;
if(rem(n,2)~=0)
n1=n;
n=n-1;
end
% window function
y=boxcar(n1);
disp('the window coefficient are as follows');

27
% low pass filter design
b=fir1(n,wp,y);
disp('unit sample response of fir filter is h(n)=');b
% frequency response
[h,o]=freqz(b,1,256);
% to find gain
m=20*log10(abs(h));
subplot(2,2,1);
plot(o/pi,m);
ylabel('gain in db');
xlabel('(a)normalised frequency');
title('LPF');

% high pass filter design

% fir filter design
b=fir1(n,wp,'high',y);
% frequency response
[h,o]=freqz(b,1,256);
% to find gain
m=20*log10(abs(h));
subplot(2,2,2);
plot(o/pi,m);
ylabel('gain in db');
xlabel('(b)normalised frequency');
title('HPF');

% band pass filter design

% fir filter design
wn=[wpws];
b=fir1(n,wn,y);
% frequency response
[h,o]=freqz(b,1,256);
% to find gain
m=20*log10(abs(h));
subplot(2,2,3);
plot(o/pi,m);
ylabel('gain in db');
xlabel('(c)normalised frequency');
title('BPF');

28
% band stop filter design
% fir filter design
wn=[wpws];
b=fir1(n,wn,'stop',y);
% frequency response
[h,o]=freqz(b,1,256);
% to find gain
m=20*log10(abs(h));
subplot(2,2,4);
plot(o/pi,m);
ylabel('gain in db');
xlabel('(d)normalised frequency');
title('BSF');

29
INPUT:

Enter the pass band ripple: 0.07

Enter the stop band ripple: 0.05

Enter the pass band freq: 1300

Enter the stop band freq: 2000

Enter the sampling freq: 7000

The window co-efficient are follows

Y=1

Unit sample response of fir filter is h (n)=

Columns 1 through 7

-0.0834 -0.0391 0.1207 0.3070 0.3896 0.3070 0.1207

Columns 8 through 9

-0.0391 -0.0834

30
Output Waveform:

LPF HPF
20 20

0 0
gain in db

gain in db
-20 -20

-40 -40

-60 -60
0 0.5 1 0 0.5 1
(a) normalised frequency (b) normalised freq
BPF BSF
20 10

0
0
gain in db

gain in db

-20
-10
-40

-60 -20
0 0.5 1 0 0.5 1
(c) normalised freq (d) normalised freq

RESULT:

Thus the FIR filter using Rectangular Window is designed using MATLAB.

31
EXP.NO:6 DESIGN OF BUTTERWORTHIIR FILTER

AIM:

To design a Butterworth IIR filters (LPF/HPF/BPF/BSF) and demonstrates the filtering

operation using MATLAB program.

SOFTWARE REQUIRED:

MATLAB R2014a

ALGORITHM:

 Get the IIR filter specifications.

 Obtain the filter coefficients
 Plot the frequency response of the filters.

PROGRAM:

clc;
clearall;
closeall;
rp=input('Enter the passband ripple');
rs=input('Enter the stopband ripple');
wp=input('Enter the passband frequency');
ws=input('Enter the stopband frequency');
f=input('Enter the sampling frequency');
w1=2*wp/f;
w2=2*ws/f;
[n,wn]=buttord(w1,w2,rp,rs);
[b,a]=butter(n,wn,'low');
[h,w]=freqz(b,a,512);
subplot(2,2,1);
plot(w/pi,abs(h));
ylabel('gain in dB');
xlabel('(a)normalized frequency');
title('LPF');
[b,a]=butter(n,wn,'high');
[h,w]=freqz(b,a,512);
subplot(2,2,2);
plot(w/pi,abs(h));
ylabel('gain in dB');
xlabel('(b)normalized frequency');

32
title('HPF');
wn1=[w1 w2];
[b,a]=butter(n,wn1);
[h,w]=freqz(b,a,512);
subplot(2,2,3);
plot(w/pi,abs(h));
ylabel('gain in dB');
xlabel('(c)normalized frequency');
title('BPF');
wn2=[w1 w2];
[b,a]=butter(n,wn2,'stop');
[h,w]=freqz(b,a,512);
subplot(2,2,4);
plot(w/pi,abs(h));
ylabel('gain in dB');
xlabel('(d)normalized frequency');
title('BSF');

33
INPUT
Enter the passbandripple6
Enter the stopbandripple20
Enter the passband frequency1000
Enter the stopband frequency2000
Enter the sampling frequency7000

OUTPUT:

LPF HPF
1 1
gain in dB

gain in dB
0.5 0.5

0 0
0 0.5 1 0 0.5 1
(a) normalized frequency (b) normalized frequency
BPF BSF
1 1.5

1
gain in dB

gain in dB

0.5
0.5

0 0
0 0.5 1 0 0.5 1
(c) normalized frequency (d) normalized frequency

RESULT:

Thus the Butterworth IIR filter was designed using MATLAB.

34
EXP.NO:6(b) DESIGN OF CHEBYSHEV-I IIR FILTER

AIM:

To design a Chebyshev-I IIR filters (LPF/HPF/BPF/BSF) and demonstrates the filtering

operation using MATLAB program.

SOFTWARE REQUIRED:

MATLAB R2014a

ALGORITHM:

 Get the IIR filter specifications.

 Obtain the filter coefficients.
 Plot the frequency response of the filters.

PROGRAM:

clc;
clearall;
closeall;
rp=input('Enter the passband ripple');
rs=input('Enter the stopband ripple');
wp=input('Enter the passband frequency');
ws=input('Enter the stopband frequency');
f=input('Enter the sampling frequency');
w1=2*wp/f;
w2=2*ws/f;
[n,wn]=cheb1ord(w1,w2,rp,rs);
[b,a]=cheby1(n,rp,wn,'low');
[h,w]=freqz(b,a,512);
subplot(2,2,1);
plot(w/pi,abs(h));
ylabel('gain in dB');
xlabel('(a)normalized frequency');
title('LPF');
[b,a]=cheby1(n,rp,wn,'high');
[h,w]=freqz(b,a,512);
subplot(2,2,2);
plot(w/pi,abs(h));
ylabel('gain in dB');
xlabel('(b)normalized frequency');

35
title('HPF');
wn1=[w1 w2];
[b,a]=cheby1(n,rp,wn1);
[h,w]=freqz(b,a,512);
subplot(2,2,3);
plot(w/pi,abs(h));
ylabel('gain in dB');
xlabel('(c)normalized frequency');
title('BPF');
wn2=[w1 w2];
[b,a]=cheby1(n,rp,wn2,'stop');
[h,w]=freqz(b,a,512);
subplot(2,2,4);
plot(w/pi,abs(h));
ylabel('gain in dB');
xlabel('(d)normalized frequency');
title('BSF');

36
INPUT

Enter the passband ripple 6

Enter the stopband ripple 20
Enter the passband frequency 1000
Enter the stopband frequency 2000
Enter the sampling frequency 7000

OUTPUT
LPF HPF
1 1
gain in dB

0.5 gain in dB 0.5

0 0
0 0.5 1 0 0.5 1
(a) normalized frequency (b) normalized frequency
BPF BSF
1 1
gain in dB

gain in dB

0.5 0.5

0 0
0 0.5 1 0 0.5 1
(c) normalized frequency (d) normalized frequency

RESULT:

Thus the Chebyshev-I IIR filter was designed using MATLAB.

37
EXP.NO:6(c) DESIGN OF CHEBYSHEV-II IIR FILTER

AIM:

To design a Chebyshev-I IIR filters (LPF/HPF/BPF/BSF) and demonstrates the filtering

operation using MATLAB program.

SOFTWARE REQUIRED:

MATLAB R2014a

ALGORITHM:

 Get the IIR filter specifications.

 Obtain the filter coefficients using .
 Plot the frequency response of the filters.

PROGRAM:

clc;
clearall;
closeall;
rp=input('Enter the passband ripple');
rs=input('Enter the stopband ripple');
wp=input('Enter the passband frequency');
ws=input('Enter the stopband frequency');
f=input('Enter the sampling frequency');
w1=2*wp/f;
w2=2*ws/f;
[n,wn]=cheb2ord(w1,w2,rp,rs);
[b,a]=cheby2(n,rp,wn,'low');
[h,w]=freqz(b,a,512);
subplot(2,2,1);
plot(w/pi,abs(h));
ylabel('gain in dB');
xlabel('(a)normalized frequency');
title('LPF');
[b,a]=cheby2(n,rp,wn,'high');
[h,w]=freqz(b,a,512);
subplot(2,2,2);
plot(w/pi,abs(h));
ylabel('gain in dB');

38
xlabel('(b)normalized frequency');
title('HPF');
wn1=[w1 w2];
[b,a]=cheby2(n,rp,wn1);
[h,w]=freqz(b,a,512);
subplot(2,2,3);
plot(w/pi,abs(h));
ylabel('gain in dB');
xlabel('(c)normalized frequency');
title('BPF');
wn2=[w1 w2];
[b,a]=cheby2(n,rp,wn2,'stop');
[h,w]=freqz(b,a,512);
subplot(2,2,4);
plot(w/pi,abs(h));
ylabel('gain in dB');
xlabel('(d)normalized frequency');
title('BSF');

39
INPUT

Enter the passband ripple 6

Enter the stopband ripple 20
Enter the passband frequency 1000
Enter the stopband frequency 2000
Enter the sampling frequency 7000

OUTPUT

LPF HPF
1 1
gain in dB

gain in dB
0.5 0.5

0 0
0 0.5 1 0 0.5 1
(a) normalized frequency (b) normalized frequency
BPF BSF
1 1.5

1
gain in dB

gain in dB

0.5
0.5

0 0
0 0.5 1 0 0.5 1
(c) normalized frequency (d) normalized frequency

RESULT:

Thus the Chebyshev-II IIR filter was designed using MATLAB

40
EXP.NO: 7 STUDY OF ARCHITECTURE OF DIGITAL SIGNAL PROCESSOR

ARCHITECTURE:

The 54x DSP use an advanced, modified Harvard architecture that maximizes processing
power by maintaining one program memory bus and three data memory buses. These processors
also provide an arithmetic logic unit (ALU) that has a high degree of parallelism, application-
specific hardware logic, on chip memory and additional on –chip peripherals. These DSPs
families also provide a highly specialized instruction set which is the basis of the operational
flexibility and the speed of these DSPs. Separate program and the data spaces allow
simultaneous access to program instructions and data, providing the high degree of parallelism.
Two reads and one write operation can be performed in a single cycle. Instructions with parallel
store and application- specific instructions can fully utilize this architecture. In addition, data can
be transferred between data and program spaces. Such parallelism supports a powerful set of
arithmetic, logic and bit manipulation operations that can be performed in a single machine
cycle. Also included are the control mechanisms to manage interrupts, repeated operations and
function calls.

1. CENTRAL PROCESSING UNIT (CPU):

The CPU of the 54x devises contains:

 40-bit arithmetic logic unit (ALU)

 Two 40 bit accumulator
 Barrel shifter
 17-bit multiplier /adder.
 A compare, select and store unit (CSSU)

2. ARITHMETIC LOGIC UNIT (ALU):

The 54x devises perform 2’s complement arithmetic using 40-bit ALU and two 40-bit
accumulators (ASSU and ACCB). The ALU also can perform Boolean operations. The ALU can
function as a two 16-bit ALUs and perform two 16-bit operations simultaneously when the C16
bit in status register 1 (ST1) is set.

41
3. ACCUMULATORS:
The accumulators, ACCA and ACCB store the output from the ALU or the
Multiplier/adder block. The accumulators can provide a second input to the ALU or the
multiplier /adder. The bit in each accumulator is grouped as follows:

42
  Guard bits (bits 32-39)
 A high – order word (bits 16-31)
 A low order word (bits 0-15)
 4 barrel Shifter
The 54x’s barrel shifter has a 40-bit input connected to the accumulator or data memory
(CB, DB) and a 40-bit output connected to the ALU or data memory (EB). The barrel shifter
produces a left shift of 0 to 31 bits and a right shift of 0 to 16 bits on the input data. The shift
requirements are defined in the shift-count field (ASM) of ST1 or defined in the temporary
register (TREG), which is designed as a shift-count register. This shifter and the exponent
detector normalize the values in the accumulator in a single cycle. The least significant bits
(LSBs) of the output are filled with 0s and the most significant bits (MSBs) can neither be zero
filled or sign extended, depending on the state of the sign-extended mode bit (SXM) of
ST1.addtional shift capabilities enable the processor to perform numerical scaling, bit extraction,
extended arithmetic and overflow prevention operation.

5. MULTIPLIER/ADDER:
The multiplier /adder perform 17- bit 2’s complement multiplication with the 40-bit
accumulation in a single instruction cycle. The multiplier /adder block consists of several
elements such as multiplier, adder, signed /unsigned input control, fractional control, a zero
detector, a rounder (2’s complement), overflow/saturation logic and TREG. The multiplier has
two inputs: one input is selected from the TREG, a data memory operand or an accumulator; the
other is selected from the program memory, the data memory, an accumulator or an immediate
value. The fast on-chip multiplier allows the 54x to perform operations such as convolution,
correlation and filtering efficiently. In addition, the multiplier and ALU together execute
multiply/accumulate (MAC) computations and ALU operations in parallel in a single instruction
cycle. This function is used in determining the Euclid distance and in implementing symmetrical
and least mean square (LMS) filters which are required for complex DSP algorithms.

6. COMPARE, SELECT AND STORE UNIT (CSSU):

The compare, select and store unit (CSSU) performs maximum comparisons between the
accumulator, high and low words allows the test/control (TC) flag bit of status register 0 (ST0)

43
and the transition (TRN) register to keep their transition histories and selects the larger word in
the accumulator to be stored in data memory. The CSSU also accelerates Veterbi type butterfly
computation with optimized on – chip hardware.

7. PROGRAM CONTROL IS PROVIDED BY SEVERAL HARDWARE AND

SOFTWARE MECHANISMS:
The program controller decodes the instructions, manages the pipeline, stores the status
of operations and decides the conditional operations. Some of the hardware elements included in
the program controller are the programcounter, the status and the control register, the stack and
the address- generation logic.
The 54x supports both the use of hardware and software interrupts for the program
control. The interrupts service routine is vectored through a reloadable interrupt vector table. The
interrupts can be globally enabled/disable and can be individually masked through the interrupt,
mask register (IMR).

8. STATUS REGISTER (ST0, ST1):

The status register ST0, ST1 contain the status of the various conditions and the modes
for the 54x devises. The ST0 contains the flags (OV,C, and TC) produced by the arithmetic
operations and bit manipulations in addition to the data pointer (DP) and the auxiliary register
pointer (ARP fields). ST1 contains the various modes and the instructions that the processor
operates on and executes.

9. AUXILLARY REGISTERS (AR0-AR7):

The eight 16- bit auxiliary registers (AR0-AR7) can be accessed by the central arithmetic
logic unit (CALU) and modified by the auxiliary register arithmetic units (ARAUs). The primary
function of the auxiliary registers is generating 16-bit addresses for data space. However, these
registers also can act as general purpose registers or counters.

10. TEMPORAY REGISTERS(TREG):

The TREG is used to hold one of the multiplicands for multiply and multiply/accumulate
instructions. It can hold a dynamic (execution – time programmable) shift count for instructions

44
with the shift operation such as ADD, LD and SUB. It also can hold a dynamic bit address for
the BITT instruction. The EXP instruction stores the exponent value computed into the TREG
while the NORM instruction uses the TREG value to normalize the number. For ACS operation
of Viterbi decoding, TREG holds branch metrics used by the DADST and DSADT instructions.

11. TRANSITION REGISTER (TRN):

The TRN is a 16-bit register that is used to hold the transition decision for the path to new
metrics to perform the Viterbi algorithm. The CMPS (Compare, select, max and store)
instruction updates the contents of the TRN based on the comparison between the accumulator
high word and the accumulator low word.

12. STACK-POINTER REGISTER (SP):

The SP is a 16-bit register that contains the address at the top of the system stack. The SP
always points to the last element pushed onto the stack. The stack is manipulated by interrupts,
traps, calls, returns and the PUSHD, PSHM, POPD and POPM instructions. Pushes and pops of
the stack pre- decrement and post increment respectively all 16 bits of the SP.

13. CIRCULAR-BUFFER-SIZE REGISTER (BK):

The 16- bit BK is used by the ARAUs in circular addressing to specify the data block
size.

14. BLOCK-REPEAT REGISTERS:

The block-repeat counter (BRC) is a 16-bit register used to specify the number of times a
block of code is to be repeated when performing a block repeat. The block-repeat start address
(RSA) is a 16-bit register containing the starting address of the block of the program memory to
be repeated when operating in the repeat mode.

15. INTERRUPT REGISTERS (IMR, IFR):

The interrupt- mask register (IMR) is used to mask off specific interrupts individually at
required times. The interrupt-flag register (IFR) indicated the current status of the interrupts.

45
16. PROCESSOR-MODE STATUS REGISTER:
The processor – mode status register (PMST) controls memory configurations of the 54x
devices.

17. POWER-DOWN MODES:

There are three power-down modes, activated by the IDLE1, IDLE2 and IDLE3
instructions. In these modes, the 54x devices enter a dormant state and dissipate considerably
less power than in normal operation. The IDLE1 instruction is used to shut down the CPU. The
IDLE2 instruction is used to shut down the CPU and on- chip peripherals. The IDLE3 instruction
is used to shut down the 54x processor completely. This instruction stops the PLL circuitry as
well as the CPU and peripherals.

46
EXP.NO: 8(a)PERFORM MAC OPERATION USING VARIOUS ADDRESSING MODES

AIM:

To write an assembly language program for the study of direct, indirect and immediate
addressing modes using TMS320C5X.

TOOLS REQUIRED:

DSP HARDWARE:

 TMS320C5X- Starter Kit

 RS 232 Cable
 Power Supply unit

DSP SOFTWARE:
 Assembler
 Loader
 Debugger

ALGORITHM:

 Initialize all memory mapped register.

 Initialize the processor.
 Initialize the analog interface chip.
 Enable receiver interrupt.
 Store the sample length and buffer starting address.
 Initialize analog interface chip register.

PROGRAM:

DIRECT ADDRESSING MODE:

.mmregs ; includes memory mapped registers

.ds 0f00h ; set data segment to 0f00ah
.ps 0a00h ; origin of the program 0a00h
rint b getdata ; receive interrupt
xint b xint ; transmit interrupt
.ps 0a00h ; program entry point
.entry ; initialize the program counter
.include “c:/c5xinz.asm”

47
 lap #20h ; the data page number 20h(32) is loaded into
accumulator
lacc 10h ; content of 20h(32) page 10h location
lac 5h,2
lar Aro,#15h ; AR0 loaded with content of dma 1115h
sacl 15h
sacl20h,3 ; accumulator low byte is left shifted by 3 bits and
stored in into dma 1120h

getdata samm ART ;accumulator low byte stored into ART in page0 DP
remains unaffected
ldp #12h ;the data page number 12h is loaded in DP
add 25h
add 7h,2
sub 10h ;the content of dma 0910h is subtracted from the
content of accumulator
sub 12h,2
splk #10h,TREGO ;constant 10h is stored into TREGO
mpy 15h
REG1 .set 010ch
REG2 .set 020ah
REG4 .set 0415h
REG5 .set 0505h
.include “c:/ac0 1inz.asm”
.end ; program end

INDIRECT ADDRESSING MODE

.mmregs ; includes memory mapped registers

.ds 0f00h ; set data segment to 0f00ah
.ps 0a00h ; origin of the program 0a00h
rint b getdata ; receive interrupt
xint b xint ; transmit interrupt
.ps 0a00h ; program entry point
.entry ; initialize the program counter
.include “c:/c5xinz.asm”
lar ARO,#1000h

lacc * ;content of dma pointed by ARO is loaded in

accumulator

48
 lacc *,4,AR1 ;content of dma 1000h left shifted by 4 bits and loaded
into accumulator. ARP points to auxiliary register 1
lar AR1,#,1010h
sacl * ;accumulator low byte is stored into the dma pointed by
AR1
sacl*+,2,AR0 ;accumulator low byte is shifted by 2 bits and stored
into the dma pointed in AR1
lacc*-2,AR1
getdata lacc*0+
lacc *BR0+ ;accumulator loaded with content of dma pointed by
AR1 and the content of INDEX register added to AR1
with the reverse carry propagation
add #+.0.ar0
sub *.-2 ;content of dma pointed by AR1 is added from the
content of accumulator. The result is stored into the
accumulator AR0is decremented by 1.
splk #10h,TREGO ;constant 10h is stored into TREGO
mpy * ;content of 0915h is multiplied with the content of
TREGO and the result is stored into PREG
REG1 .set 010ch
REG2 .set 020ah
REG4 .set 0415h
REG5 .set 0505h
.include “c:/ac0 1inz.asm”
.end ; program end

IMMEDIATE ADDRESSING MODE

.mmregs ; includes memory mapped registers

.ds 0f00h ; set data segment to 0f00af
.ps 0a00h ; origin of the program 0a00h
rint b getdata ;receive interrupt
xint b xint ;transmit interrupt
.ps 0a00h ; program entry point
.entry ; initialize the program counter
.include “c:/c5xinz.asm”
lacc#1000h ;value 1000h is loaded into accumulator
lacc#1111h,3 ;constant 1111h is left shifted by 3 bits and loaded in
accumulator. The accumulator after execution is 8888h
getdata lar AR0,#1000h ;AR0 is loaded the content of 1000h
lar AR1,#1100h ;1100f is loaded in AR1
add#00ffh ;ffhis added to the content of accumulator
sp1k #10h,TREGO

49
 mpy#0010h ;0010h is multiplied with the content of TREGO
sub #0022h
sub #0011h,3 ;0011h is left shifted by 3 bits subtracted from the
content of accumulator
REG1 .set 010ch
REG2 .set 020ah
REG4 .set 0415h
REG5 .set 0505h
.include “c:/ac0 1inz.asm”
.end ; end of program

RESULT:

Thus the assembly language program for the study of direct, indirect and immediate
addressing modes was written and executed successfullyusing TMS320C5X.

50
EXP.NO:9 GENERATION OF VARIOUS SIGNALS AND RANDOM NOISE
AIM:

To write an assembly language program for the generation of sine wave using
TMS320C5X.

TOOLS REQUIRED:

DSP HARDWARE:

 TMS320C5X- Starter Kit

 RS 232 Cable
 Power Supply unit

DSP SOFTWARE:
 Assembler
 Loader
 Debugger

ALGORITHM:

 Initialize all memory mapped register.

 Initialize the processor.
 Initialize the analog interface chip.
 Enable receiver interrupt.
 Store the sample length and buffer starting address.
 Initialize analog interface chip register.

PROGRAM:

.mmregs ;initialise all registers.

.ds 1000h
.include "sinetbl.dat" ;load 800 point wavetable
;f1= fs/D = 8000/800 = 10hz.
.ds 0f00h
temp .word 0
mod .word 100 ;Required freq. = mod * f1 = 100*10 = 1000hz.
scale .q15 0.5
;
;Interrupt vectors
;
.ps 080ah

51
rint b getdata ;receive interrupt
xint b xint ;transmit interrupt

.ps 0a00h ;program entry point

.entry

;
;Processor initialization
;

.include "c5xinz.asm"

splk #012h,IMR ;enable RINT & INT2.

call ac01_init ;call to initialize serial port & AC01.
clrc INTM ;enable all interrupts.

wait: nop ;wait for interrupt.

b wait

;
;Receive interrupt handler
;
getdata splk #1,gotflag ;set a flag to indicate data available.
lamm DRR
ldp #mod ;set data page pointer.
lacc mod ;load modifier
samm INDX ;store modifier in INDX register.
callwavgen ;calculate current sample output from
wavetable.
and #0FFFCh,0 ;only 14 MSBs are used in ADC &DAC,So
; mask unused two LSBs.
samm DXR ;send digital data to DAC to produce analog
o/p.
clrc INTM ;enable interrupt.
rete ;return back to main from interrupt routine.

offset .set 1320h ;table length = 800 + table start address.

.include "wavgen.asm" ;includewavgen module.

52
;
;AC01 register initialization.
;
REG1 .set 0124h
REG2 .set 0212h
REG4 .set 0417h
REG5 .set 0505h
;
;Serial port and AC01 initialization
;
.include "ac01inz.asm"

.END ;end of program.

53
TABULATION:

Amplitude (v) Time (ms)

OUTPUT:

RESULT:

Thus the assembly language program for the generation of sine waveform was written
and executed successfullyusing TMS320C5X.

54
EXP.NO:10(a) IMPLEMENTATION OF FIR LOW PASS FILTER

AIM:

To write an assemble language program for the implementation of FIR low pass filter
using TMS320C5X.

TOOLS REQUIRED:

DSP HARDWARE:

 TMS320C5X- Starter Kit

 RS 232 Cable
 Power Supply unit

DSP SOFTWARE:
 Assembler
 Loader
 Debugger

ALGORITHM:

 Initialize all memory mapped register.

 Initialize the processor.
 Initialize the analog interface chip.
 Enable receiver interrupt.
 Store the sample length and buffer starting address.
 Initialize analog interface chip register.

PROGRAM:

.mmregs ;initialize all memory mapped registers

.ds 0f00h ;set data segment to 0f00h.

;
;201 coefficients table.
;
.include "firlpf.cof"

rbuf .word 0 ;Temp buffer allocation.

;
;Interrupt vectors
;

55
 .ps 080ah

rint b getdata ;receive interrupt

xint b xint ;transmit interrupt

.ps 0a00h ;program entry point

.entry

;
;Processor initialization
;

.include "c5xinz.asm"

;
;Internal memory initialization
;

mar *,AR7 ;ARP = AR7

lacl #0 ;ACC = 0
lar AR7,#300h ;clear 300 to 3ffh(data array).
rpt #255
sacl *+

mar *,AR0
lar AR0,#0200h ;copy 201 co-efficients
rpt #200 ;to address 200h-2C8h(B0).
bldd #FIR_COEFFS,*+,AR0

splk #012h,IMR ;enable RINT & INT2.

call ac01_init ;call to initialize serial port & AC01.
clrc INTM ;enable all interrupts.

wait: nop ;wait for interrupt.

b wait

;
;Receive interrupt handler
;

56
getdata splk #1,gotflag ;set a flag to indicate data available.
lamm DRR ;read ADC data from DRR register.
and #0fffch ;mask LSB two bits.
ldp #rbuf
saclrbuf
lacc rbuf,13 ;load accu-high with ADC data.
ldp #06h ;set page pointer = 6.
sach 0 ;store ADC data(address=300h).
mar *,AR1
lar AR1,#3C8h ;load AR1 with data buffer end addr.
;(data memory).
;
;FILTERING.
;
setc CNF ;convert B0 to program memory.
mpy #0 ;clear product reg.
lacl #0 ;clear accumulator.
rpt #200 ;repeat MACD insru. 201 times.
macd #0ff00h,*- ;convolution process.
apac ;get result in accumulator.
sach 0 ;store result in data buffer.
lacc 0,4
ldp 0
samm DXR ;send digital ADC data to DAC.
clrc CNF ;convert B0 to data memory.
rete ;return from interrupt.
;
;AC01 register initialization.
;
REG1 .set 0124h
REG2 .set 0212h
REG4 .set 0415h
REG5 .set 0505h
;
;Serial port and AC01 initialization
;
.include "ac01inz.asm"

.END ;end of program.

57
TABULATION:

Amplitude (v) Time (ms)

OUTPUT

RESULT:

Thus the assembly language program for the implementation of FIR low pass filter was
written and executed successfullyusing TMS320C5X.

58
EXP.NO:10 (b) IMPLEMENTATION OF FIR HIGH PASS FILTER

AIM:

To write an assembly language program for the implementation of FIR high pass filter
using TMS320C5X.

TOOLS REQUIRED:

DSP HARDWARE:

 TMS320C5X- Starter Kit

 RS 232 Cable
 Power Supply unit

DSP SOFTWARE:
 Assembler
 Loader
 Debugger

ALGORITHM:

 Initialize all memory mapped register.

 Initialize the processor.
 Initialize the analog interface chip.
 Enable receiver interrupt.
 Store the sample length and buffer starting address.
 Initialize analog interface chip register.

PROGRAM:

.mmregs ;initialize all memory mapped registers

.ds 0f00h ;set data segment to 0f00h.

;
;201 coefficients table.
;
.include "firhpf.cof"

rbuf .word 0 ;Temp buffer allocation.

;
;Interrupt vectors
;

59
 .ps 080ah

rint b getdata ;receive interrupt

xint b xint ;transmit interrupt

.ps 0a00h ;program entry point

.entry
;
;Processor initialization
;

.include "c:\fepl\c5xinz.asm"
;
;Internal memory initialization
;

mar *,AR7
lacl #0
lar AR7,#300h ;clear 300 to 3ffh(data array).
rpt #255
sacl *+

mar *,AR0
lar AR0,#0200h ;copy 201 co-efficients
rpt #200 ;to address 200h-2C8h(B0).
blkd COEFFS,*+,AR0

splk #012h,IMR
call ac01_init ;call to initialize serial port & AC01.
clrc INTM

wait: nop ;wait for interrupt.

b wait
;
;Receive interrupt handler
;
Getdata splk #1,gotflag ;set a flag to indicate data available.
lamm DRR
and #0fffch
ldp #rbuf

60
 saclrbuf
lacc rbuf,13
ldp #06h
sach 0
mar *,AR1
lar AR1,#3C8h ;load AR1 with data buffer end addr.

setc CNF
mpy #0
lacl #0
rpt #200
macd #0ff00h,*- ;convolution process.
apac
sach 0
lacc 0,4
ldp 0
samm DXR
clrc CNF
rete
;
;AC01 register initialization.
;
REG1 .set 0124h
REG2 .set 0212h
REG4 .set 0415h
REG5 .set 0505h
;
;Serial port and AC01 initialization
;
.include "c:\fepl\ac01inz.asm"

.end ;end of program.

61
TABULATION:

Amplitude (v) Time (ms)

OUTPUT:

RESULT:

Thus the assembly language program for the implementation of FIR high low pass filter
was written and executed successfully using TMS320C5X.

62
EXP.NO:11 IMPLEMENTATION OF IIR FILTER

AIM:

To write an assembly language program for the implementation of IIR filter using
TMS320C5X.

TOOLS REQUIRED:

DSP HARDWARE:

 TMS320C5X- Starter Kit

 RS 232 Cable
 Power Supply unit

DSP SOFTWARE:
 Assembler
 Loader
 Debugger

ALGORITHM:

 Initialize all memory mapped register.

 Initialize the processor.
 Initialize the analog interface chip.
 Enable receiver interrupt.
 Store the sample length and buffer starting address.
 Initialize analog interface chip register.

PROGRAM:

.mmregs ;initialize all memory mapped registers

.ds 0f00h ;set data segment to 0f00h.

DN .word 0 ;Input data delay line

DNM1 .word 0
DNM2 .word 0

YN .word 0 ;output buffer

XN .word 0 ;input buffer

63
 .include "bilinear.cof" ;bilinear IIR filter coefficients
.include "invarian.cof" ;invariance IIR filter coefficients

.ps 080ah

rint b getdata ;receive interrupt

xint b xint ;transmit interrupt
.ps 0a00h ;program entry point
.entry
.include "c5xinz.asm"

splk #012h,IMR
call ac01_init ;call to initialize serial port & AC01.
lacl #0
ldp #DN
sacl DN ;clear input data delay line.
sacl DNM1
sacl DNM2
clrc INTM

wait: nop ;wait for interrupt.

b wait

getdata splk #1,gotflag ;set a flag to indicate data available.

lamm DRR ;read input from AIC
and #0FFFCh,0 ;mask unwanted bits
ldp #XN
sacl XN ;store input sample

lacc XN,15
lt DNM1
mpy A1
lta DNM2
mpy A2
apac ;DN = x(n) + d(n-1)*a1 + d(n-2)*a2
sach DN,0 ;store pole result
lacl #0
mpy B2
ltd DNM1
mpy B1

64
 ltd DN
mpy B0
apac
sach YN,3 ;Y(n) = d(n)*b0 + d(n-1)*b1 + d(n-2)*b2
lacc YN,0 ;store y(n) result
and #0FFFCh,0
samm DXR ;output the filter response y(n) to AIC.
rete

REG1 .set 010ch

REG2 .set 0212h
REG4 .set 0415h
REG5 .set 0505h
.include "ac01inz.asm"

.END ;end of program.

65
TABULATION:

Amplitude (v) Time (ms)

OUTPUT:

RESULT:

Thus the assembly language program for the implementation of IIR filter was written and
executedsuccessfully using TMS320C5X.

66
EX.No: 12 IMPLEMENT AN UP-SAMPLING AND DOWN-SAMPLING
OPERATION IN DSP PROCESSOR
AIM:

To write an assembly language program for sampling the given input signal using
TMS320C5X.

TOOLS REQUIRED:

DSP HARDWARE:

 TMS320C5X- Starter Kit

 RS 232 Cable
 Power Supply unit

DSP SOFTWARE:
 Assembler
 Loader
 Debugger

ALGORITHM:

 Initialize all memory mapped register.

 Initialize the processor.
 Initialize the analog interface chip.
 Enable receiver interrupt.
 Store the sample length and buffer starting address.
 Initialize analog interface chip register.

PROGRAMM

.mmregs ;initialize all memory mapped registers

.ps 080ah
rint b getdata ;receive interrupt
xint b xint ;transmit interrupt
.ps 0a00h ;program entry point

.entry

.include "c5xinz.asm"
lar AR2,#1000h
splk #012h,IMR ;enable RINT & INT2.

67
 call ac01_init ;call to initialize serial port & AC01.

clrc INTM ;enable all interrupts.

lar AR1,#2048
lar AR2,#1000h

wait: idle ;wait for interrupt.

mar *,AR1
banz wait,*-,AR2
nop
nop
setc INTM
splk #02h,IMR
clrc INTM

hlt: b hlt
getdatasplk #1,gotflag ;set a flag to indicate data available.
lamm DRR ;read ADC data from DRR register.
and #0fffch ;mask LSB two bits.

samm DXR ;send digital ADC data to DAC.

mar *,AR2
sacl *+
rete ;return from interrupt.

REG1 .set 010ch

REG2 .set 020ah

REG4 .set 0415h
REG5 .set 0505h
.include "ac01inz.asm"
.END ;end of program.

68
TABULATION:

Amplitude (v) Time (ms)

OUTPUT:

RESULT:

Thus the assembly language program for the sampling operation was written and
executed successfully using TMS320C5X.

69
EXP.NO: LINEAR CONVOLUTION

AIM:

To write an assembly language program for linear convolution using TMS320C5X.

70
 TOOLS REQUIRED:

DSP HARDWARE:

 TMS320C5X- Starter Kit

 RS 232 Cable
 Power Supply unit

DSP SOFTWARE:
 Assembler
 Loader
 Debugger

ALGORITHM:

 Include memory mapped register and set pointer program memory and data memory.
 Append 0’s buffer and after impulse response no of zero in length of input sequence.
 Zero accumulator and product register.
 Multiply accumulator program memory with data memory.
 Each time program memory is incremented by one and data memory decremented by one.
 Repeat step 4 for n+1 timer where n is length of largest sequence.
 Decrement count value ARZ if ARZ≠0 go to step3.

PROGRAM:

.mmregs ;initialize all registers

.ps 0a00h
.word 1h,2h,3h,2h,1h ;x(n) stored form pma 0a00h
.ds 1000h
.word0h,0h,0h,0h
.word 3h,4h,5h,0h
.word oh,0h,0h,0h
.entry
LAR AR0.#1004H ;actual data starts only at 1004h
LAR AR1,#1020H ;starting address for result
LAR AR2,#07H ;length for output sequence
Loop ZAP ;zero accumulator and product reg
MAR*,AR0
RPT#5H ; execute instructions followed by RPT
instructions 5 times
MAC 0a00h,*-
MAR *.AR1

71
SACL*+.0.AR0 ; one result is stored
ADRK#7H
MAR*,AR2
MAR*
BANZ loop
.end ; end of program

72
INPUT: OUTPUT:
1000 01 1020 03
1001 02
1021 10
1002 03
1022 22
1003 02
1023 28
1004 01
1024 26
1005 03
1006 04 1025 10

1007 05 1026 05

1008 00 1027 00

RESULT:

Thus the assembly language program for linear convolution was written and executed
successfully using TMS320C5X.

73
 EXP.NO: 13 STUDY OF ANTI- ALIASING FILTER

Antialiasing filters:

Anti-aliasing filters are always analog filters as they process the signal before it is
sampled. In most cases, they are also low-pass filters unless band-pass sampling techniques are
used.The sampling process incorporating an ideal low-pass filter as the anti-alias filter is shown
below. The ideal filter has a flat passband and the cut-off is very sharp. Since the cut-off
frequency of this filter is half of that of the sampling frequency, the resulting replicated spectrum
of the sampled signal do not overlap each other. Thus no aliasing occurs.

Analog to Digital conversion process using Anti – aliasing filter

If the sampling frequency does not satisfy the sampling theorem (i.e., the sampled signal
has frequency components greater than half the sampling frequency), then the sampling process
creates new frequency components .This phenomenon is called aliasing and must obviously be
avoided in a digital control system. Hence, the continuous signal to be sampled must not include
significant frequency components greater than the Nyquist frequency ωs/2.

74
For this purpose, it is recommended to low-pass filter the continuous signal before sampling,
especially in the presence of high-frequency noise. The analog low-pass filter used for this
purpose is known as the antialiasing filter. The antialiasing filter is typically a simple first-order
RC filter, but some applications require a higher-order filter such as a Butterworth or a Bessel
filter. The overall control scheme is shown below.

Control scheme with an antialiasing filter.

Because a low-pass filter can slow down the system by attenuating high-frequency dynamics,
the cutoff frequency of the low-pass filter must be higher than the bandwidth of the closed-loop
system so as not to degrade the transient response. A rule of thumb is to choose the filter
bandwidth equal to a constant time the bandwidth of the closed-loop system. The value of the
constant varies depending on economic and practical considerations. For a conservative but more
expensive design, the cutoff frequency of the low-pass filter can be chosen as 10 times the
bandwidth of the closed-loop system to minimize its effect on the control system dynamics, and
then the sampling frequency can be chosen 10 times higher than the filter cutoff frequency so
there is a sufficient attenuation above the Nyquist frequency. Thus, the sampling frequency is
100 times the bandwidth of the closed-loop system. To reduce the sampling frequency, and the
associated hardware costs, it is possible to reduce the antialiasing filter cutoff frequency. In the
extreme case, we select the cutoff frequency slightly higher than the closed-loop bandwidth. For
a low-pass filter with a high roll-off (i.e., a high-order filter), the sampling frequency is chosen as
five times the closed-loop bandwidth. In summary, the sampling period T can be chosen in
general as 5ωb≤2πT≤100ωb where ωb is the bandwidth of the closed-loop system.

75
EX.No: 14 CONVERSION OF ANALOG TO DIGITAL FILTERS

AIM:

To write a program for the conversion of analog to digital filters using MATLAB.

SOFTWARE REQUIRED:

MATLAB R2014a

ALGORITHM:

 Get the required analog input specifications.

 Convert the analog specifications to digital specifications .
 Plot the digital filter specifications.

PROGRAM:

alpha = 0.2;
fs = 200; % Sample Frequency [Hz]
% Laplace Domain
B = 1;
A = [1, alpha];
w = 0:0.2:(fs / 2);
h = freqs(B, A, w);
figure;
plot(w, abs(h .* conj(h)));
% Digital Filter
[b, a] = bilinear(B, A, fs);
figure;
freqz(b, a, 1000);
% Frequency Response of the filter
f = 2;
fs = 10;
[b,a] = butter(6,2*pi*f,'s');
[bz,az] = impinvar(b,a,fs);
freqz(bz,az,1024,fs)
% Impulse Response of the Digital filter
fs = 10;
[b,a] = ellip(3,1,60,2*pi*2.5,'s');
[bz,az] = impinvar(b,a,fs);
impz(bz,az,[],fs)

76
OUTPUT:

Frequency Response of the filter

Impulse Response of the Digital filter

77
RESULT:

Thus the analog filter was converted to digital filter using MATLAB.

78