EC2306 DSP Lab Manual PDF

EINSTEIN
COLLEGE OF ENGINEERING
Sir.C.V.Raman Nagar, Tirunelveli-12
Department of Electronics and Communication

Engineering
Subject Code: EC56
Digital Signal Processing Laboratory
Name
Reg No
Branch
Year & Semester
EC 56-Digital Signal Processing Lab
INDEX
S.No
Date
Name of the experiment
Page no
Marks
USING MATLAB
1.
Generation of signals
2.
Convolution of signals
3.
Sampling and effect of aliasing
11
4.
Calculation of FFT
13
Design of FIR filter using Kaiser
15
5(a).
window
5(b).
Design of FIR filter using
18
Rectangular window
5(c)
21
Hamming window
5(d)
24
Hanning window
5(e)
27
Blackmann window
6(a)
Butterworth digital IIR filter
30
6(b)
Chebyshev digital IIR filter
33
Decimation by Polyphase
36
7.
Decomposition
8.
Periodogram based Spectral
38
Estimation
Einstein College of Engineering

Page 2 of 81
Signature
S.No
Date
Name of the experiment
Page no
Marks
USING TMS320C5X
9.
Study of addressing modes
40
10.
Implementation of Linear and
42
Circular Convolution
11.
Sampling of signals
47
12.
Waveform Generation
49
Design of FIR filter-Low Pass
59
13(a)
Filter.
13(b)
Design of FIR filter-High Pass
63
Filter
13(c)
Design of FIR filter-Band Pass
67
Filter
13(d)
Design of FIR filter-Band Reject
71
Filter

Page 3 of 81
Signature
EX. NO. 1
Date:
GENERATION OF SIGNALS
AIM:
To generate the following signals using MATLAB 7.0
1. Unit impulse signal
2. Unit step signal
3. Unit ramp signal
4. Exponential growing signal
5. Exponential decaying signal
6. Sine signal
7. Cosine signal
APPARATUS REQUIRED:
System with MATLAB 7.0.
ALGORITHM:
1. Get the number of samples.
2. Generate the unit impulse, unit step using ones, zeros matrix
command.
3. Generate ramp, sine, cosine and exponential signals using corresponding
general formula.
4. Plot the graph.
PROGRAM:
% 1. UNIT IMPULSE SIGNAL
clc;
clear all;
close all;
N=input('Number of Samples');
n=-N:1:N
x=[zeros(1,N) 1 zeros(1,N)]
stem(n,x);
xlabel('Time');
ylabel('Amplitude');
title('Impulse Response');

Page 4 of 81
% 2. UNIT STEP SIGNAL

clc;
clear all;
close all;
N=input('Number of Samples = ');
n=-N:1:N
x=[zeros(1,N) 1 ones(1,N)]
stem(n,x);
xlabel('Time');
title('Unit Step Response');
% 3. UNIT RAMP SIGNAL

clc;
clear all;
close all;
disp('UNUT RAMP SIGNAL');
a=input('Amplitude = ')
n=-N:1:N
x=a*n
stem(n,x);
xlabel('Time');
title('Unit Ramp Response');
% 4. EXPONENTIAL DECAYING SIGNAL

clc;
clear all;
close all;
disp('EXPONENTIAL DECAYING SIGNAL');
a=0.5
n=0:.1:N
x=a.^n
stem(n,x);
xlabel('Time');
title('Exponential Decaying Signal Response');
% 5. EXPONENTIAL GROWING SIGNAL
clc;
clear all;
close all;
disp('EXPONENTIAL GROWING SIGNAL');
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a=0.5
n=0:.1:N
x=a.^n
stem(n,x);
xlabel('Time');
title('Exponential Growing Signal Response');
% 6. COSINE SIGNAL
clc;
clear all;
close all;
disp('COSINE SIGNAL');
n=0:.1:N
x=cos(n)
stem(n,x);
xlabel('Time');
title('Cosine Signal');
% 7. SINE SIGNAL
clc;
clear all;
close all;
disp('SINE SIGNAL');
n=0:.1:N
x=sin(n)
stem(n,x);
xlabel('Time');
title('sine Signal');
OUTPUT:

Page 6 of 81
RESULT:

Page 7 of 81
EX.NO. 2
Date:
CONVOLUTION OF SIGNALS
AIM:
To write the program for finding the linear and circular convolution of two
signals using MATLAB 7.0.
APPARATUS REQUIRED:
LINEAR CONVOLUTION:
ALGORITHM:
2. Generate the output sequence of the given two input signals using
conv command.
3. Plot the graph.
PROGRAM:
% PROGRAM FOR LINEAR CONVOLUTION

clc;
clear all;
close all;
x=input('Enter the first sample');
N1=length(x);
h=input('Enter the second sample');
N2=length(h);
n=0:1:N1-1;
subplot(2,2,1);
stem(n,x);
xlabel('Time');
title('X Sequence Response');
n=0:1:N2-1;
subplot(2,2,2);
stem(n,h);
xlabel('Time');
title('H Sequence Response');
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y=conv(x,h);
N=N1+N2-1;
n=0:1:N-1;
subplot(2,2,3);
stem(n,y);
xlabel('Time');
title('Convolution of X&H Response');
CIRCULAR CONVOLUTION:
ALGORITHM:
2. Generate the sequence for the given two input signals using zeros and
ones matrix command.
3. Generate the convoluted output sequence of the given two input signals
using conv command.
4. Plot the graph.
PROGRAM:
% PROGRAM FOR CIRCULAR CONVOLUTION
clc;
clear all;
close all;
x=input('Enter the first sequence');
N1=length(x);
h=input('Enter the second sequence');
N2=length(h);
n=0:1:N1-1;
subplot(2,2,1);
stem(n,x);
xlabel('Time');
title('X Sequence Response');
n=0:1:N2-1;
subplot(2,2,2);
stem(n,h);
xlabel('Time');
title('H Sequence Response');
N=max(N1,N2);
x=[x,zeros(1,N-N1)];
h=[h,zeros(1,N-N2)];
for n=1:N
y(n)=0;
Page 9 of 81

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
disp('Convolution of x & h is');
subplot(2,2,3);
stem(y);
xlabel('Time');
title('Convolution of X&H Response');
OUTPUT:
RESULT:

Page 10 of 81
EX.NO. 3
Date
SAMPLING AND EFFECT OF ALIASING
AIM:
To sample the signal and observe the aliasing effect of that signal using
MATLAB 7.0.
APPARATUS REQUIRED:
ALGORITHM:
1. Get the input frequency and number of samples.
2. The input signal is sampled at the frequencies fs1=2fm, fs2>2fm and
fs3<2fm.
3. Plot the input and sampled signal at various frequencies.
PROGRAM:
% PROGRAM FOR SAMPLING AND EFFECT OF ALIASING
clc;
clear all;
close all;
N=input('Enter the number of samples N= ');
fm=input('Enter the frequency');
t=0:0.01:N;
x=cos(2*pi*fm*t);
subplot(3,2,1);
plot(t,x);
% xlabel('Time');
title('Input Signal');
fs1=fm;
fs2=2*fm;
fs3=5*fm;
n=0:0.08:N;
y=cos(2*pi*fm*n);
subplot(3,2,2);
stem(n,y);
% xlabel('Time');
title('Sampling Signal');
y1=cos(2*pi*n*fm/fs1);
subplot(3,2,3);
stem(n,y1);
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% xlabel('Time');
title('Sampling Signal with fs<2fm');
subplot(3,2,4);
stem(n,y2);
xlabel('Time');
title('Sampling Signal with fs=2fm');
subplot(3,2,5);
stem(n,y3);
xlabel('Time');
title('Sampling Signal with fs>2fm');
OUTPUT:
RESULT:

Page 12 of 81
EX.NO. 4
Date:
CALCULATION OF FFT
AIM:
To write the program for calculating the N-point FFT and IFFT of given input
sequence using MATLAB 7.0.
APPARATUS REQUIRED:
ALGORITHM:
1. Get the input sequence and its length.
2. Calculate the Fast Fourier Transform and Inverse Fast Fourier Transform
of given sequence using user defined function.
3. Plot the magnitude and phase response of FFT sequence.
4. Plot the IFFT response.
PROGRAM:
% CALCULATION OF FFT AND IFFT
clc;
clear all;
close all;
x=input('x(n)');
N=length(x);
n=input('n');
y=fft(x,n);
v=0:1:N-1;
t=0:1:n-1;
subplot(2,2,1);
stem(v,x);
xlabel('Time');
title('Input Signal');
subplot(2,2,2);
stem(t,abs(y));
xlabel('Time');
title('Magnitude Response');
subplot(2,2,3);
stem(t,angle(y));
xlabel('Time');
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title('Phase Response');
disp('fft x(n)=1');
disp(y);
y1=ifft(y,n);
subplot(2,2,4);
stem(t,y1);
xlabel('Time');
title('IFFT Response');
disp('ifft X(K)');
disp(y1);
OUTPUT:
RESULT:

Page 14 of 81
EX.NO. 5 (a)
Date:
DESIGN OF FIR FILTER USING KAISER WINDOW
AIM:
To design the FIR filter using Kaiser Window with MATLAB 7.0.
APPARATUS REQUIRED:
ALGORITHM:
1.
2.
3.
4.
5.
Get the pass band and stop band attenuation and frequency.
Get the sampling frequency.
Find transition width and center frequency (cut-off frequency).
Design the low pass, high pass, band pass and band stop filter.
Plot the magnitude response of all filters.
PROGRAM:
clc;
clear all;
close all;
ap=input('Pass band attenuation=');
as=input('Stop band attenuation=');
fs=input('Stop band frequency in rad/sec=');
fp=input('Pass band frequency in rad/sec=');
wsf=input('Sampling frequency in rad/sec=');
dw=fs-fp;
wc=(fs+fp)/2;
wp1=2*fp/wsf;
ws1=2*fs/wsf;
s1=10^(-0.05*as);
e=10^(0.05*ap);
s2=(e-1)/(e+1);
S=min(s1,s2);
A=-20*log10(S);
if(A<=21)
B=0
elseif(A>21&A<=50)
B=0.5842*((A-21)^0.4)+(0.07886*(A-21));
elseif(A>50)
B=0.1102*(A-8.7);
end
N=((wsf*(A-8))/(dw*14.36))+1;
Page 15 of 81

N=round(N);
y=kaiser(N,B);
%LOW PASS FILTER
x=fir1(N-1,wp1,y);
[h,o]=freqz(x,1,256);
subplot(2,2,1);
plot(o/pi,20*log10(abs(h)));
ylabel('Gain in db---->');
xlabel('(a)Normalized frequency---->');
title('LOW PASS FILTER')
%HIGH PASS FILTER
y1=length(y);
t=y(1:y1-1);
x=fir1(N-2,wp1,'high',t);
[h,o]=freqz(x,1,256);
subplot(2,2,2);
xlabel('(b)Normalized frequency---->');
title('HIGH PASS FILTER')
%BAND PASS FILTER
wn=[wp1,ws1];
x=fir1(N-1,wn,y);
[h,o]=freqz(x,1,256);
subplot(2,2,3);
xlabel('(c)Normalized frequency---->');
title('BAND PASS FILTER')
%BAND STOP FILTER
x=fir1(N-2,wn,'stop',t);
[h,o]=freqz(x,1,256);
subplot(2,2,4);
xlabel('(d)Normalized frequency---->');
title('BAND STOP FILTER')
OUTPUT:

Page 16 of 81
RESULT:

Page 17 of 81
EX.NO. 5 (b)
DESIGN OF FIR FILTER USING RECTANGULAR

WINDOW
Date:
AIM:
To design a FIR filter using Rectangular window with MATLAB 7.0.
APPARATUS REQUIRED:
ALGORITHM:
1.
2.
3.
4.
5.
Get the pass band and stop band ripple and frequency.
Find the order of filter N.
Design the low pass, high pass, band pass and band stop filter.
Plot the magnitude response of all filters.
PROGRAM:
clc;
clear all;
close all;
rp=input('Pass band ripple=');
rs=input('Stop band ripple=');
f=input('Sampling frequency in rad/sec=');
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=boxcar(n1);
%LOW PASS FILTER
b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
Page 18 of 81

m=20*log10(abs(h));
subplot(2,2,1);
plot(o/pi,m,'k');
xlabel('Normalized frequency---->');
%HIGH PASS FILTER
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,'k');
%BAND PASS FILTER
wn=[wp,ws];
x=fir1(n,wn,y);
[h,o]=freqz(b,1,256);
subplot(2,2,3);
plot(o/pi,m,'k');
%BAND STOP FILTER
b=fir1(n,wn,'stop',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,4);
plot(o/pi,m,'k');
OUTPUT:

Page 19 of 81
RESULT:

Page 20 of 81
EX.NO. 5 (c)
Date:
DESIGN OF FIR FILTER USING HAMMING WINDOW
AIM:
To design a FIR filter using Hamming window with MATLAB 7.0.
APPARATUS REQUIRED:
System with MATLAB 7.0
ALGORITHM:
1.
2.
3.
4.
5.
Design the lowpass,High pass,Band pass and Band stop filter.
Plot the magnitude response of all the filters.
PROGRAM:
clc;
clear all;
close all;
wp=2*fp/f;
ws=2*fs/f;
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);
%LOW PASS FILTER
b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
Page 21 of 81

m=20*log10(abs(h));
subplot(2,2,1);
plot(o/pi,m,'k');
%HIGH PASS FILTER
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,2);
plot(o/pi,m,'k');
%BAND PASS FILTER
wn=[wp,ws];
x=fir1(n,wn,y);
[h,o]=freqz(b,1,256);
subplot(2,2,3);
plot(o/pi,m,'k');
%BAND STOP FILTER
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,4);
plot(o/pi,m,'k');
OUTPUT:

Page 22 of 81
RESULT:

Page 23 of 81
EX.NO. 5 (d)
Date:
DESIGN OF FIR FILTER USING HANNING WINDOW
AIM:
To design a FIR filter using Hanning window with MATLAB 7.0.
APPARATUS REQUIRED:
ALGORITHM:
1.
2.
3.
4.
5.
Design the lowpass,High pass,Band pass and Band stop filter.
Plot the magnitude response of all the filters.
PROGRAM:
clc;
clear all;
close all;
wp=2*fp/f;
ws=2*fs/f;
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);

Page 24 of 81
%LOW PASS FILTER

b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1);
plot(o/pi,m,'k');
%HIGH PASS FILTER
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,2);
plot(o/pi,m,'k');
%BAND PASS FILTER
wn=[wp,ws];
x=fir1(n,wn,y);
[h,o]=freqz(b,1,256);
subplot(2,2,3);
plot(o/pi,m,'k');
%BAND STOP FILTER
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,4);
plot(o/pi,m,'k');

Page 25 of 81
OUTPUT:
RESULT:

Page 26 of 81
EX.NO. 5 (e)
Date:
DESIGN OF FIR FILTER USING BLACKMANN WINDOW
AIM:
To design a FIR filter using Blackmann window with MATLAB 7.0.
APPARATUS REQUIRED:
ALGORITHM:
6. Get the pass band and stop band ripple and frequency.
7. Get the sampling frequency.
8. Find the order of filter N.
9. Design the lowpass,High pass,Band pass and Band stop filter.
10. Plot the magnitude response of all the filters.
PROGRAM:
clc;
clear all;
close all;
wp=2*fp/f;
ws=2*fs/f;
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=blackman(n1);

Page 27 of 81
%LOW PASS FILTER

b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1);
plot(o/pi,m,'k');
%HIGH PASS FILTER
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,2);
plot(o/pi,m,'k');
%BAND PASS FILTER
wn=[wp,ws];
x=fir1(n,wn,y);
[h,o]=freqz(b,1,256);
subplot(2,2,3);
plot(o/pi,m,'k');
%BAND STOP FILTER
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,4);
plot(o/pi,m,'k');

Page 28 of 81
OUTPUT:
RESULT:

Page 29 of 81
EX.NO. 6 (a)
Date:
BUTTERWORTH DIGITAL IIR FILTER
AIM:
To design a Butterworth digital IIR filter using MATLAB 7.0.
APPARATUS REQUIRED:
ALGORITHM:
1.
2.
3.
4.
Get the pass band and stop band ripples.

Get the pass band and stop band frequencies.
Calculate the order of the filter using
log [10 kp 1 / 10 kp 1]
N = ------------------------------------log s / p
5.Find the filter co-efficient.

6.Draw the magnitude and phase response.
PROGRAM:
clear all;
clc;
close all;
format long
rp=input(enter the pass band ripple);
rs=input(enter the stop band ripple);
wp=input(enter the pass band frequency );
ws=input(enter the stop band frequency );
fs=input(enter the sampling frequency );
w1=2*wp/fs;
w2=2*ws/fs;
%LOW PASS FILTER
[n,wn]=buttord(w1,w2,rp,rs];
[b,a]=butter(n,wn);
%[b,a]=zp2tf(z,p,k);
[b,a]= butter(n,wn);
Page 30 of 81

W=0:0.01:pi;
[h,om]=freqz(b,a,w);
m=20*log10(abs(h));
an=angle(h);
%figure(1);
Subplot(2,1,1);
Plot(om/pi,m);
Ylabel(gain in db>);
Xlabel((a)normalized frequency>);
%figure(1);
Subplot(2,1,2);
Plot(om/pi,an);
Xlabel((b)normalized frequency>);
Ylabel(phase in radians>);
%HIGH PASS FILTER

[n,wn]=buttord(w1,w2,rp,rs];
[b,a]=butter(n,wn,high);
W=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
figure(2);
Subplot(2,1,1);
Plot(om/pi,m);
figure(2);
Subplot(2,1,2);
Plot(om/pi,an);
%BAND PASS FILTER
[n]=buttord(w1,w2,rp,rs];
Wn=[w1,w2];
[b,a]=butter(n,wn,band pass);
W=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
figure(3);
Subplot(2,1,1);
Plot(om/pi,m);
figure(3);
Subplot(2,1,2);
Plot(om/pi,an);
Page 31 of 81

%BAND STOP FILTER

[n]=buttord(w1,w2,rp,rs];
Wn=[w1,w2];
[b,a]=butter(n,wn,band stop);
W=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
figure(4);
Subplot(2,1,1);
Plot(om/pi,m);
figure(4);
Subplot(2,1,2);
Plot(om/pi,an);
OUTPUT:
RESULT:

Page 32 of 81
EX.NO. 6 (b)
Date:
CHEBYSHEV DIGITAL IIR FILTER
AIM:
To design a digital IIR filter using Chebyshev filter with MATLAB 7.0.
APPARATUS REQUIRED:
ALGORITHM:
1.
2.
3.
4.
Get the pass band and stop band ripples.

Get the pass band and stop band frequencies.
Calculate the order of the filter using
log [10 kp 1 / 10 kp 1]
N = ------------------------------------log s / p
5.Find the filter co-efficient.

6.Draw the magnitude and phase response.
PROGRAM:
clear all;
clc;
close all;
rp=input(enter the pass band ripple);
rs=input(enter the stop band ripple);
wp=input(enter the pass band frequency );
ws=input(enter the stop band frequency );
fs=input(enter the sampling frequency );
w1=2*wp/fs;
w2=2*ws/fs;
%LOW PASS FILTER
[n,wn]=cheb ord(w1,w2,rp,rs];
[b,a]=cheby 1(n,wn);
W=0:0.01:pi;
Page 33 of 81

m=20*log10(abs(h));
an=angle(h);
Subplot(2,1,1);
Plot(om/pi,m);
Subplot(2,1,2);
Plot(om/pi,an);
%HIGH PASS FILTER
[n,wn]= cheb1 ord(w1,w2,rp,rs];
[b,a]=cheby 1(n,rp,wn,high);
W=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
figure(2);
Subplot(2,1,1);
Plot(om/pi,m);
figure(2);
Subplot(2,1,2);
Plot(om/pi,an);
%BAND PASS FILTER
[n]=cheb1 ord(w1,w2,rp,rs];
Wn=[w1,w2];
[b,a]=cheby 1(n,rs,wn,band pass);
W=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
figure(3);
Subplot(2,1,1);
Plot(om/pi,m);
figure(3);
Subplot(2,1,2);
Plot(om/pi,an);
%BAND STOP FILTER
[n]=cheb 1 ord(w1,w2,rp,rs];
Page 34 of 81

Wn=[w1,w2];
[b,a]=cheby 1(n,rp,wn,band stop);
W=0:0.1/pi:pi;
m=20*log10(abs(h));
an=angle(h);
figure(4);
Subplot(2,1,1);
Plot(om/pi,m);
figure(4);
Subplot(2,1,2);
Plot(om/pi,an);
OUTPUT:
RESULT:

Page 35 of 81
EX.NO. 7
Date:
DECIMATION BY POLYPHASE DECOMPOSITION
AIM:
To write a program to compute Convolution and m-fold decimation by
polyphase decomposition.
APPARATUS REQUIRED:
ALGORITHM:
1. Get the input sequence.
2. Get the filter coefficients and also the decimation factor.
3. Find the response by using convolution.
4. Plot the graph.
PROGRAM:
%Program for computing convolution and m-fold decimation by polyphase
decomposition.
clear all;
clc;
close all;
x=input(enter the input sequence);
h=input(enter the FIR filter coefficients);
M=input(enter the decimation factor);
N1=length(x);
N=0:1:N1-1;
Subplot(2,1,1);
Stem(n,x);
Xlabel(time);
Ylabel(amplitude);
Title(X sequence response);
N2=length(h);
n=0:1:N2-1;
Subplot(2,1,2);
Stem(n,h);
Xlabel(time);
Ylabel(amplitude);
Title(H sequence response);
H=length(h);
P=floor((H-1)/M)+1;
Page 36 of 81

r=reshape([reshape(h,1,H),zeros(1,P*M-H)],M,P);
X=length(x);
Y=floor((X+H-2)/M)+1;
U=floor((X+M-2)/M)+1;
R=zeros(1,X+P-1);
for m=1:M
R=R+conv(x(1,:),r(m,:));
End
Disp(R);
N=length(R);
n=0:1:N-1;
figure(2);
stem(n,R);
xlabel(time);
ylabel(amplitude);
title(Decimation of polyphase decomposition sequence response);
OUTPUT:

Page 37 of 81

RESULT:
EX.NO. 8
Date:
PERIODOGRAM BASED SPECTRAL ESTIMATION
AIM:
APPARATUS REQUIRED:
ALGORITHM:
PROGRAM:

Page 38 of 81
OUTPUT:
RESULT:

Page 39 of 81
EX.NO. 9
Date:
STUDY OF ADDRESSING MODES
AIM:
To study about direct, indirect and immediate addressing modes in
TMS320C50 debugger.
APPARATUS REQUIRED:
1.System with TMS 320C50 debugger software
2.TMS 320C50 Kit.
ALGORITHM:
IMMEDIATE ADDRESSING MODE:
1.
2.
3.
4.
Initialize data pointer with 100H data.

Load the accumulator with first data.
Add the second data with accumulator content.
Store the accumulator content in specified address location.
DIRECT ADDRESSING MODE:

1. Initialize data pointer with 100H data.
2. Load the accumulator with first data, whose address is specified in the
instruction.
3. Add the accumulator content with second data, whose address is specified
in the instruction.
4. Store the accumulator content in specified address location.
IN-DIRECT ADDRESSING MODE:
1. Load the auxiliary register with address location of first data.
2. The auxiliary register (AR0) is modified indirectly as # symbol.
3. Load the second data into accumulator and perform addition operation.
4. Store the result.
PROGRAM:
;program for immediate addressing mode
.mmregs
.text
START:
LDP #100H
Page 40 of 81
H:
LACC #1241H
ADD #1200H
SACL 2H
BH
;program for direct addressing mode

.mmregs
.text
START:
H:
LDP #100H
LACC 0H
ADD 1H
SACL 2H
BH
;program for adding two numbers with indirect addressing mode.

.mmregs
.text
START:
LAR AR0,#8000H
MAR *,AR0
LACC *+,0 ;WITH ZERO SHIFT
ADD *+
SACL *+
H:
BH
OUTPUT:
RESULT:

Page 41 of 81
EX.NO.10 IMPLEMENTATION OF LINEAR AND CIRCULAR

CONVOLUTION
Date:
AIM:
To write a program to find the Convolution of two sequences X(n) and H(n)
using TMS320C50 debugger.
APPARATUS REQUIRED:
2.TMS 320C50 Kit.
LINEAR CONVOLUTION
ALGORITHM:
1. Start the program.
2. LPD will load the 9LSBs of address data memory.
Location or immediate value into the data pointer (DP) register.
3. Load auxiliary register (LAR) loads the content of the address data
memory location into the designated auxilary register.
4. Then ZC clears the content of accumulator too.
5. MAR modify current AR and ARP as specified.
6. The RPT load the 8 bit LSBs of the addressed value of the RPT.
7. The SACL store the 16 LSBs of the accumulator into the addressed data
memory.
8. MUL is used to multiply the content of T register.
9. The LTP would load the contents of the addressed data memory location
into T register.
10. ADD will add the content of the memory.
11. LACC is used to load the content of the addressed data memory into
accumulator.
PROGRAM:
;Program for Linear Convolution
.MMREGS
.TEXT
START:
LDP #100H
LAR AR1,#(8100H+4H)
ZAC
MAR *,AR1
Page 42 of 81

RPT#2H
SACL*+
LAR AR1,#(8200H+4H)
ZAC
MAR *,AR1
RPT#2H
SACL*+
LAR AR3,#(8300H+6H)
LAR AR4,#06H
;Multiply & Accumulate:
Next_YN:
LAR AR1,#8100H
LAR AR2,#(8200H+6H)
LAR AR0,#06H
ZAC
SACL 0H
MAR *,AR1
LT *+,AR2
Loop_MAC:
MPY *-,AR1
LTP *+,AR0
ADD 0H
SACL 0H
BANZ Loop_MAC,*-,AR2
LACC 0H
MAR *,AR3
SACL *;Shiftx(n) Values:
LAR AR1,#(8100H+6H)
LAR AR2,#5H
LAR AR0,#2H
MAR *,AR1
LACC *SACL 1H
Shift_XN:
LACC *+
SACL *0-,AR2
BANZ Shift_XN,*-,AR1
MAR *+
LACC 1H
SACL *,AR4
BANZ Next_YN,*NOP
Page 43 of 81

H1:
B H1
OUTPUT:
CIRCULAR CONVOLUTION
ALGORITHM:
1. Start the Program.
2. Load the LSBs of the address data memory location.
3. Load the contents of the address data memory location into auxiliary
register.
4. Modify the current AR as specified.
5. Load the content of addressed data memory location.
6. Store the 16 LSBs of the accumulator into the address data memory
location.
7. Load the content in ARL Z.0
8. Branch the contents of entire auxiliary register.
9. Load the content AR.1,AR.2,AR.0
10. Store the content of the accumulator
12. Stop the program.
PROGRAM:
;Program for Circular Convolution
.MMREGS
.TEXT
LDP #100H
LACC 0H
SUB #1H
SACL 1H
LAR AR0,1H
LAR AR2,#8010H
LOOP3: LAR AR1,#8060H
LAR AR3,#8050H
LAR AR4,1H
ZAP
LOOP:MAR *,AR3
Page 44 of 81

LT
*+,AR1
MPY *+
SPL 5H
ADD 5H
MAR *,AR4
BANZ LOOP,*-,AR2
SACL *+
CALL ROTATE
LOOP2:MAR *,AR0
BANZ LOOP3,*H:
B
H
ROTATE:
LDP #100H
LACC 1H
SUB #1H
SACL 2H
LACC 0050H
SACB
LAR AR3,#8051H
LAR AR5,#8070H
LAR AR6,2H
LOOP1:MAR *,AR3
LACC *+,0,AR5
SACL *+,0,AR6
BANZ LOOP1,*LACB
MAR *,AR5
SACL *+
LACC #8070H
SAMM
BMAR
LAR AR3,#8050H
MAR *,AR3
RPT 0H
BLDD BMAR,*+
RET
OUTPUT:

Page 45 of 81
RESULT:

Page 46 of 81
EX.NO. 11
Date:
SAMPLING OF SIGNALS
AIM:
To write a program to convert analog signals into digital signals using
TMS320C50 debugger.
APPARATUS REQUIRED:
2.TMS 320C50 Kit.
3.CR0
4.Function Generator.
ALGORITHM:
1.
2.
3.
4.
5.
Initialize data pointer with 100H data

give the analog signal as input
Introduce the time delay as per required.
Observe the discrete signal as output
Plot the graph.
PROGRAM:
.mmregs
.text
LDP #100H
START:
IN 0,06H
NOP
IN 0,04H
RPT #4FFH
NOP
OUT 0,04H
RPT #4FFH
NOP
B START
MODEL GRAPH:

Page 47 of 81
OUTPUT:
WAVEFORM
AMPLITUDE (V)
TIME PERIOD (ms)
INPUT
OUTPUT
RESULT:

Page 48 of 81

Thus the analog signal was converted into digital signal using TMS320C50
debugger.
EX.NO. 12
WAVEFORM GENERATION
Date:
AIM:
To write a program to generate Triangle Wave,Sine Wave,Square Wave and
SawTooth Wave using TMS320C50 debugger.
APPARATUS REQUIRED:
2.TMS 320C50 Kit.
3.CR0
4.Function Generator.
ALGORITHM:
TRIANGLE WAVE GENERATION:
2. Load the LSBs of the address data memory location.
3. Write full 16 bit pattern into address data memory location.
4. Load the content of address data memory location.
5. Add the content of address data memory location.
6. Store the data of accumulator into address data memory location.
8. Repeat the step 2,3.
9. Subtract the contents of address data memory location.
10. Repeat the steps 5,6,7.
SINE WAVE GENERATION:

1. Start the program
2.
Load the LSBs of the address data memory location
3. Write full 16 bit pattern into address data memory location.

4. Load the content of address data memory location.Specify the address of a
subroutine.
5. Multiply the content of the register.
6. Load the content of the register into accumulator.
location.
Page 49 of 81

8. Modify the current AR & ARP as specified.
9. Branch the contents of entire auxiliary register.
11. Repeat the Step.
SQUARE WAVE GENERATION:

1. Start the program
location.
5. Complement the contents of the accumulator.
SAWTOOTH WAVE GENERATION:

2. Load the LSBs of the address data memory location
3. Load the content of address data memory location into
accumulator.
4. Store the 16 LSBs of the accumulator into the address data
memory location
5. Write 16 bit value from from a data memory location to the
specified I/O Port.
6. Add the content of address data memory location.
7. Subtract the content of the register.
8. Branch the program memory address if the specified conditions are
met.
PROGRAM:
;Program for Triangle Wave Generation
AMPLITUDE .SET 4
FREQ .SET 350
TEMP .SET 0
Page 50 of 81

.mmregs
.text
START:LDP #100H
SPLK #0,TEMP
CONT1:LAR AR2,#FREQ
CONT:OUT TEMP ,4
LACC TEMP
ADD #AMPLITUDE
SACL TEMP
MAR *,AR2
BANZ CONT,*LAR
AR2,#FREQ
CONTx:OUT TEMP,4
LACC TEMP
SUB
#AMPLITUDE
SACL TEMP
MAR *,AR2
BANZ CONTx
B
CONT1
;Program for Sine Wave Generation

INCFREQ
.SET 10
DECFREQ .SET 0
LENGTH
.SET 360
AMPLITUDE .SET 5
DELAY1
.SET 7
TEMP
.SET 0
TEMP1
.SET 1
.mmregs
.text
START:LDP #100H
SPLK #B500,TEMP
LAR AR2,#((LENGTH/INCFREQ)+(LENGTH*DECFREQ))-1
LAR AR3,#DECFREQ
CONT:CALL DELAY
LACC TEMP
TBLR TEMP1
LT
TEMP1
MPY #AMPLITUDE
Page 51 of 81

PAC
SACL TEMP1
MAR *,AR3
BANZ REPEAT,*LAR AR3,#DECFREQ
LACC TEMP
ADD #INCFREQ
SACL TEMP
OUT TEMP1,4
MAR *,AR2
BANZ CONT,*B
START
DELAY:LAR AR7,#DELAY1
MAR *,AR7
BACK:BANZ BACK,*RET
;
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Page 56 of 81
;Program for Square Wave Generation

.MMREGS
.TEXT
START:
LDP #100H
LACC #0FFFH
LOOP:
SACL 0
RPT #0FFH
OUT 0,04
CMPL
B LOOP
END
;Program for SawTooth Wave Generation

.MMREGS
.TEXT
START:
LDP #120H
LACC #0H
SACL 0
LOOP:
LACC 0
OUT 0,04H
ADD #05H
SACL 0
SUB #0FFFH
BCND LOOP,LEQ
B START
END
MODEL GRAPH:

Page 57 of 81
OUTPUT:
WAVEFORM
AMPLITUDE (V)
TIME PERIOD (ms)
RESULT:

Page 58 of 81
EX.NO. 13 (a)
Date:
DESIGN OF FIR FILTER-LOW PASS FILTER
AIM:
To design a FIR low pass filter in serial mode.
APPARATUS REQUIRED:
1.System with VI debugger software
2.TMS 320C50 Kit.
3.CR0
ALGORITHM:
2. Initialize the C table value.
3. Load the auxillary register with 0200H.
4. Modify the auxillary register zero.
5. Block the C table to the program.
6. Set configuration control bit.
7. Load the data pointer with 0AH.
8. Initialize the analog to digital conversion.
9. Load the auxillary register 1 with 0300 content.
10. Load the accumulator in 8000H.
11. AND the accumulator with 0FFFH.
12. Subtract the accumulator content with data 800H.
13. Modify the auxillary register 1.
14. Store the accumulator data in 8000H.
15. Load the auxillary register 1 with content 0333H.
16. Zero the accumulator register.
17. Multiply the accumulator with data.
18. Load the program register, with PM bits to accumulator.
20. Add accumulator content with 800H.
21. Shift the accumulator right 1 bit.
22. Store the accumulator content in 8200 location.
23. Branch the program to step 7.
PROGRAM:
*Approximation type:Window design- Blackmann Window
*Filter type:Low Pass Filter
*Filter Order:52
*Cutoff frequency in KHZ=3.000000
.mmregs
.text
Page 59 of 81

B START
CTABLE:
.word 0FFE7H
.word 0FFD3H
.word 0FFC6H
.word 0FFC4H
.word 0FFD0H
.word 0FFECH
.word 018H
.word 051H
.word 08CH
.word 0B9H
.word 0C2H
.word 092H
.word 01AH
.word 0FF5FH
.word 0FE78H
.word 0FD9AH
.word 0FD10H
.word 0FD30H
.word 0FE42H
.word 071H
.word 03B5H
.word 07CAH
.word 0C34H
.word 01054H
.word 01387H
.word 01547H
.word 01547H
.word 01387H
.word 01054H
.word 0C34H
.word 07CAH
.word 03B5H
.word 071H
.word 0FE42H
.word 0FD30H
.word 0FD10H
.word 0FD9AH
.word 0FE78H
.word 0FF5FH
.word 01AH
.word 092H
.word 0C2H
.word 0B9H
.word 08CH
.word 051H
.word 018H
.word 0FFECH
.word 0FFD0H
Page 60 of 81
*
*
.word 0FFC4H
.word 0FFC6H
.word 0FFD3H
.word 0FFE7H
Move the Filter coefficients
from program memory to data memory
START:
LAR
MAR
RPT
BLKP
SETC
AR0,#0200H
*,AR0
#33H
CTABLE,*+
CNF
* Input data and perform convolution

ISR: LDP #0AH
IN
0,06H
IN
0,04H
NOP
NOP
NOP
NOP
LAR AR1,#0300H
LACC 0
AND #0FFFH
SUB #800H
MAR *,AR1
SACL *
LAR AR1,#0333H
ZAP
RPT #33H
MACD 0FF00H,*APAC
LAR AR1,#0300H
SACH *
; give as SACH *, 1 in case of overflow
LACC *
ADD #800H
SFR
; remove if output is less amplitude
SACL *
OUT *,4
;pulse to find sampling frequency
NOP
B ISR
.end

Page 61 of 81
MODEL GRAPH:
TABULATION:
S.No.
Frequency(Hz)
Vout(v)
Vout/Vin
Gain in db=
20log Vout/Vin
RESULT:
Thus the program for designing a FIR low pass filter in serial mode was
performed

Page 62 of 81
EX.NO. 13 (b)
Date:
DESIGN OF FIR FILTER-HIGH PASS FILTER
AIM:
To design a FIR high pass filter in serial mode.
APPARATUS REQUIRED:
2.TMS 320C50 Kit.
3.CR0
ALGORITHM:
PROGRAM:
*Filter type:high Pass Filter
*Filter Order:52
*Cutoff frequency in KHZ=3.000000
Page 63 of 81

.mmregs
.text
B START
CTABLE:
.word 0FCD3H
.word 05H
.word 0FCB9H
.word 087H
.word 0FD3FH
.word 01ADH
.word 0FE3DH
.word 0333H
.word 0FF52H
.word 04ABH
.word 0FFF8H
.word 0595H
.word 0FFACH
.word 0590H
.word 0FE11H
.word 047CH
.word 0FB0BH
.word 029DH
.word 0F6BAH
.word 0AEH
.word 0F147H
.word 01CH
.word 0E9FDH
.word 04C5H
.word 0D882H
.word 044BCH
.word 0D882H
.word 04C5H
.word 0E9FDH
.word 01CH
.word 0F147H
.word 0AEH
.word 0F6BAH
.word 029DH
.word 0FB0BH
.word 047CH
.word 0FE11H
.word 0590H
.word 0FFACH
.word 0595H
.word 0FF8H
.word 04ABH
.word 0FF52H
.word 0333H
.word 0FE3DH
.word 01ADH
Page 64 of 81
.word 0FD3FH
.word 087H
.word 0FCB9H
.word 05H
.word 0FCD3H
*
START:
LAR
MAR
RPT
BLKP
SETC
AR0,#0200H
*,AR0
#33H
CTABLE,*+
CNF

ISR: LDP #0AH
IN
0,06H
IN
0,04H
NOP
NOP
NOP
NOP
LAR AR1,#0300H
LACC 0
AND #0FFFH
SUB #800H
MAR *,AR1
SACL *
LAR AR1,#0333H
ZAP
RPT #33H
MACD 0FF00H,*APAC
LAR AR1,#0300H
SACH *
LACC *
ADD #800H
SFR
SACL *
OUT *,4
NOP
B ISR
.end

Page 65 of 81
MODEL GRAPH:
TABULATION:
S.No.
Frequency(Hz)
Vout(v)
Vout/Vin
Gain in db=
20log Vout/Vin
RESULT:
Thus the program for designing a FIR high pass filter in serial mode was
performed.

Page 66 of 81
EX.NO. 13 (c)
Date:
DESIGN OF FIR FILTER-BAND PASS FILTER
AIM:
To design a FIR band pass filter in serial mode.
APPARATUS REQUIRED:
2.TMS 320C50 Kit.
3.CR0
ALGORITHM:
23. Branch the program to step
PROGRAM:
*Filter type:band Pass Filter
*Filter Order:52
*lower Cutoff frequency in KHZ=3.000000
*upper Cutoff frequency in KHZ=3.000000
Page 67 of 81
.mmregs
.text
B START
CTABLE:
.word 024AH
.word 010FH
.word 0FH
.word 0FFECH
.word 0C6H
.word 0220H
.word 0312H
.word 02D3H
.word 012FH
.word 0FEBDH
.word 0FC97H
.word 0FBCBH
.word 0FCB0H
.word 0FE9EH
.word 029H
.word 0FFDCH
.word 0FD11H
.word 0F884H
.word 0F436H
.word 0F2A0H
.word 0F58AH
.word 0FD12H
.word 075FH
.word 01135H
.word 01732H
.word 01732H
.word 01135H
.word 075FH
.word 0FD12H
.word 0F58AH
.word 0F2A0H
.word 0F436H
.word 0F884H
.word 0FD11H
.word 0FFDCH
.word 029H
.word 0FE9EH
.word 0FCB0H
.word 0FBCBH
.word 0FC97H
.word 0FEBDH
.word 012FH
.word 02D3H
.word 0312H
.word 0220H
Page 68 of 81

.word
.word
.word
.word
.word
0C6H
0FFECH
0FH
010FH
024AH
*
START:
LAR
MAR
RPT
BLKP
SETC
AR0,#0200H
*,AR0
#33H
CTABLE,*+
CNF

ISR: LDP #0AH
IN
0,06H
IN
0,04H
NOP
NOP
NOP
NOP
LAR AR1,#0300H
LACC 0
AND #0FFFH
SUB #800H
MAR *,AR1
SACL *
LAR AR1,#0333H
ZAP
RPT #33H
MACD 0FF00H,*APAC
LAR AR1,#0300H
SACH *
LACC *
ADD #800H
SFR
SACL *
OUT *,4
NOP
B ISR
.end

Page 69 of 81
MODEL GRAPH:
TABULATION:
S.No.
Frequency(Hz)
Vout(v)
Vout/Vin
Gain in db=
20log Vout/Vin
RESULT:
Thus the program for designing a FIR band pass filter in serial mode was
performed.

Page 70 of 81
EX.NO. 13 (d)
Date:
DESIGN OF FIR FILTER-BAND REJECT FILTER
AIM:
To design a FIR band reject filter in serial mode.
APPARATUS REQUIRED:
2.TMS 320C50 Kit.
3.CR0
ALGORITHM:
PROGRAM:
*Filter type:band Pass Filter
*Filter Order:52
*lower Cutoff frequency in KHZ=3.000000
*upper Cutoff frequency in KHZ=5.000000

Page 71 of 81

.mmregs
.text
B START
CTABLE:
.word 0FEB9H
.word 014EH
.word 0FDA1H
.word 155H
.word 0FE1BH
.word 282H
.word 0FEAFH
.word 2ACH
.word 0FD35H
.word 8DH
.word 0F9D9H
.word 0FE07H
.word 0F7CCH
.word 0FEE2H
.word 0FA2FH
.word 4BAH
.word 1AH
.word 25CH
.word 420H
.word 1008H
.word 89H
.word 0D61H
.word 0F3F2H
.word 0AF9H
.word 0DB7EH
.word 045DFH
.word 0DB7EH
.word 0AF9H
.word 0F3F2H
.word 0D61H
.word 81H
.word 89H
.word 1008H
.word 420H
.word 25CH
.word 1AH
.word 4BAH
.word 0FA2FH
.word 0FEE2H
.word 0F7CCH
.word 0FE07H
.word 0F9D9H
.word 8DH
.word 0FD35H
.word 2ACH
.word 0FEAFH
Page 72 of 81

.word
.word
.word
.word
.word
.word
282H
0FE1BH
155H
0FDA1H
14EH
0FEB9H
*
START:
LAR
MAR
RPT
BLKP
SETC
AR0,#0200H
*,AR0
#33H
CTABLE,*+
CNF

ISR: LDP #0AH
IN
0,06H
IN
0,04H
NOP
NOP
NOP
NOP
LAR AR1,#0300H
LACC 0
AND #0FFFH
SUB #800H
MAR *,AR1
SACL *
LAR AR1,#0333H
ZAP
RPT #33H
MACD 0FF00H,*APAC
LAR AR1,#0300H
SACH *
LACC *
ADD #800H
SFR
SACL *
OUT *,4
NOP
B ISR
.end

Page 73 of 81

MODEL GRAPH:
TABULATION:
S.No.
Frequency(Hz)
Vout(v)
Vout/Vin
Gain in db=
20log Vout/Vin
RESULT:
Thus the program for designing a FIR band reject filter in serial mode was
performed using TMS320C50 debugger.

Page 74 of 81
Viva questions:
1.What is a continous and discrete time signal?
2.Give the classification of signals.
3.What are the types of system?
4.What are even and odd signal?
5.What are energy and power signal?
6.What are elementary signals and name them?

Page 75 of 81
7.What is time invariant system?
8.What do you mean by periodic and aperiodic signals?
9.Determine the convolution sum of two sequences x(n) = {3, 2, 1, 2} and

h(n) = {1, 2, 1, 2}
10.Find the convolution of the signals

x(n) = 1 n=-2,0,1
= 2 n=-1
= 0 elsewhere.
11. Differentiate DTFT and DFT.
12..Differentiate between DIT and DIF algorithm

Page 76 of 81
13.How many stages are there for 8 point DFT
14. How many multiplication terms are required for doing DFT by
expressionalmethod and FFT method
15.Distinguish IIR and FIR filters
16.Write the expression for order State the steps to design digital IIR filter using
bilinear method of Butterworth filter?
17.Write the expression for the order of chebyshev filter?
18.What is warping effect?
19.Write a note on pre warping.
20.What is meant by impulse invariant method?

Page 77 of 81

21. Give the Butterworth filter transfer function and its magnitude characteristics
for different orders of filter.
22. Give the magnitude function of Butterworth filter.
23. How can you design a digital filter from analog filter?
24.write down bilinear transformation.
25.What is filter?
26. What are the types of digital filter according to their impulse response?
27.what is mean by FIR filter?
28.Write the steps involved in FIR filter design.
29.List the well known design technique for linear phase FIR filter design?
30.Define IIR filter?

Page 78 of 81
31.What is the reason that FIR filter is always stable?
32.What are the properties of FIR filter?
33.What is the principle of designing FIR filter using frequency sampling method?
34.What is meant by autocorrelation?
35.Define white noise?
36.List out the addressing modes supported by C5X processors?
37.What do you understand by input quantization error?
38.List the on-chip peripherals in 5X.

Page 79 of 81
39.what is meant rounding?
40.what is meant by A/D conversion noise?
41.what is meant by quantization step size?
42.what is overflow oscillation?
43.what are the methods used to prevent overflow?
44.what are the two kinds of limit cycle behavior in DSP?
45.What is meant by "dead band" of the filter?

Page 80 of 81

Page 81 of 81

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EINSTEIN

Department of Electronics and Communication

Year & Semester

EC 56-Digital Signal Processing Lab

Name of the experiment

Sampling and effect of aliasing

Design of FIR filter using Kaiser

Design of FIR filter using

Design of FIR filter using

Design of FIR filter using

Design of FIR filter using

Butterworth digital IIR filter

Chebyshev digital IIR filter

Periodogram based Spectral

Einstein College of Engineering

EC 56-Digital Signal Processing Lab

Name of the experiment

Study of addressing modes

Implementation of Linear and

Design of FIR filter-Low Pass

Design of FIR filter-High Pass

Design of FIR filter-Band Pass

Design of FIR filter-Band Reject

Einstein College of Engineering

EC 56-Digital Signal Processing Lab

Einstein College of Engineering

EC 56-Digital Signal Processing Lab

% 2. UNIT STEP SIGNAL

% 3. UNIT RAMP SIGNAL

% 4. EXPONENTIAL DECAYING SIGNAL

EC 56-Digital Signal Processing Lab

Einstein College of Engineering

EC 56-Digital Signal Processing Lab

Einstein College of Engineering

EC 56-Digital Signal Processing Lab

% PROGRAM FOR LINEAR CONVOLUTION

EC 56-Digital Signal Processing Lab

EC 56-Digital Signal Processing Lab

Einstein College of Engineering

EC 56-Digital Signal Processing Lab

SAMPLING AND EFFECT OF ALIASING

EC 56-Digital Signal Processing Lab

Einstein College of Engineering

EC 56-Digital Signal Processing Lab

EC 56-Digital Signal Processing Lab

Einstein College of Engineering

EC 56-Digital Signal Processing Lab

DESIGN OF FIR FILTER USING KAISER WINDOW

EC 56-Digital Signal Processing Lab

Einstein College of Engineering

EC 56-Digital Signal Processing Lab

Einstein College of Engineering

EC 56-Digital Signal Processing Lab

DESIGN OF FIR FILTER USING RECTANGULAR

EC 56-Digital Signal Processing Lab

Einstein College of Engineering

EC 56-Digital Signal Processing Lab

Einstein College of Engineering

EC 56-Digital Signal Processing Lab

DESIGN OF FIR FILTER USING HAMMING WINDOW

EC 56-Digital Signal Processing Lab

Einstein College of Engineering

EC 56-Digital Signal Processing Lab