DSP Lab Manual EC8562 (R 2017)

SEC - ECE EC8562 – DIGITAL SIGNAL PROCESSING
FAMILIARISATION WITH MATLAB
Aim:
To familiarize with MATLAB software, general functions and signal processing toolbox
functions.
The name MATLAB stands for MATrix LABoratory produced by Mathworks Inc., USA.
It is a matrix-based powerful software package for scientific and engineering computation
and visualization. Complex numerical problems can be solved in a fraction of the time that
required with other high level languages. It provides an interactive environment with
hundreds of built -in –functions for technical computation, graphics and animation. In
addition to built-in-functions, user can create his own functions.
MATLAB offers several optional toolboxes, such as signal processing, control systems,
neural networks etc. It is command driven software and has online help facility.
MATLAB has three basic windows normally; command window, graphics window and edit
window.
Command window is characterized by the prompt ‘>>’.
All commands and the ready to run program filename can be typed here. Graphic window
gives the display of the figures as the result of the program. Edit window is to create
program files with an extension .m.
Some important commands in MATLAB
Help - List topics on which help is available
Help command name Provides - help on the topic selected
Demo - Runs the demo program
Who - Lists variables currently in the workspace
Whos - Lists variables currently in the workspace with their size
Clear - Clears the workspace, all the variables are removed
Clear x,y,z - Clears only variables x,y,z
Quit - Quits MATLAB
Some of the frequently used built-in-functions in Signal Processing Toolbox
filter(b.a.x) - Syntax of this function is Y = filter(b.a.x)
It filters the data in vector x with the filter described by vectors
a and b to create the filtered data y.
fft (x) It is the DFT of vector x
ifft (x) It is the DFT of vector x
conv (a,b) Syntax of this function is C = conv (a,b)
It convolves vectors a and b. The resulting vector is of
Length, Length (a) + Length (b)-1
deconv(b,a) Syntax of this function is [q,r] = deconv(b,a)
It deconvolves vector q and the remainder in vector r such that
b = conv(a,q)+r
butter(N,Wn) designs an Nth order lowpass digital
Butterworth filter and returns the filter coefficients in length
N+1 vectors B (numerator) and A (denominator). The coefficients
1
are listed in descending powers of z. The cutoff frequency Wn must be 0.0 <
Wn < 1.0, with 1.0 corresponding to half the sample rate.
buttord(Wp, Ws, Rp, Rs) returns the order N of the lowest order digita Butterworth filter
that loses no more than Rp dB in the passband and has at least Rs dB of
attenuation in the stopband. Wp and Ws are the passband and stopband
edge frequencies, Normalized from 0 to 1 ,(where 1 corresponds to pi
rad/sec)
Cheby1(N,R,Wn) designs an Nth order lowpass digital Chebyshev filter with R decibels of
peak-to-peak ripple in the passband. CHEBY1 returns the filter
coefficients in length N+1 vectors B (numerator) and A (denominator).
The cutoff frequency Wn must be 0.0 < Wn < 1.0, with 1.0
corresponding to half the
sample rate.
Cheby1(N,R,Wn,'high') designs a highpass filter.
Cheb1ord(Wp, Ws, Rp, Rs) returns the order N of the lowest order digital Chebyshev Type
I filter that loses no more than Rp dB in the passband and has at
least Rs dB of attenuation in the stopband. Wp and Ws are the
passband and stopband edge frequencies, normalized from 0 to 1
(where 1 corresponds to pi radians/sample)
cheby2(N,R,Wn) designs an Nth order lowpass digital Chebyshev filter with the
stopband ripple R decibels down and stopband edge frequency Wn.
CHEBY2 returns the filter coefficients in length N+1 vectors B
(numerator) and A . The cutoff frequency Wn must be 0.0 < Wn < 1.0,
with 1.0 corresponding to half the sample rate.
cheb2ord(Wp, Ws, Rp, Rs) returns the order N of the lowest order digital Chebyshev Type II
filter that loses no more than Rp dB in the passband and has at least Rs dB of attenuation
in the stopband. Wp and Ws are the passband and stopband edge frequencies,
abs(x) - It gives the absolute value of the elements of x. When x is complex abs(x) is the
complex modulus (magnitude) of the elements of x.
angle(H) - It returns the phase angles of a matrix with complex elements in radians.
freqz(b,a,N) -Syntax of this function is [h,w] = freqz(b,a,N) returns the N-
point frequency vector w in radians and the N-point complex
frequency response vector h of the filter b/a.
stem(y) It plots the data sequence y aa stems from the x axis terminated
with circles for the data value.
stem(x,y) It plots the data sequence y at the values specified in x.
ploy(x,y) It plots vector y versus vector x. If x or y is a matrix, then the
vector is plotted versus the rows or columns of the matrix,
whichever line up.
title(‘text’) It adds text at the top of the current axis.
xlabel(‘text’) It adds text beside the x-axis on the current axis.
ylabel(‘text’) It adds text beside the y-axis on the current axis.
2
Expt. No: 1
Date: GENERATION OF SIGNALS
AIM:
To generate the following signals using MATLAB 9.0
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 9.0.
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.
3
UNIT IMPULSE
clc;
clear all;
close all;
t=-2:1:2;
y=[zeros(1,2),ones(1,1),zeros(1,2)];
subplot(2,2,1);
stem(t,y);
xlabel('time-->.');
ylabel('amplitude-->');
OUTPUT:
4
Expt. No: 1(b) UNIT RAMP
Date:
clc;
close all;
clear all;
n=input ('Enter the length of ramp sequence:');
t=0:1:n;
subplot(2,2,3);
stem(t,t);
xlabel('time-->.');
INPUT: Enter the length of ramp sequence: 6
OUTPUT:
5
Expt. No: 1(c) UNIT SINE
Date:
clc;
clear all;
t=0:.05:pi;
y=sin(2*pi*t);
figure(2);
subplot(2,1,2);
plot(t,y);
xlabel('time-->.');
OUTPUT:
6
Expt. No: 1(d) UNIT COSINE
Date:
clc;
clear all;
t=0:0.05:pi;
y=cos(2*pi*t);
subplot(2,2,2);
plot(t,y);
xlabel('time-->.');
OUTPUT:
7
Expt. No: 1(e) UNIT EXPONENTIAL
Date:
clear all;
close all;
n=input('Enter the length of the exponential sequence');
t=0:1:n;
a=input('Enter value');
y=exp(a*t);
subplot(2,2,4);
stem(t,y);
xlabel('time-->.');
INPUT: Enter the length of the exponential sequence 3
Enter value 1
OUTPUT:
RESULTS:-
Thus the generation of continues time signals using matlab was verified
8
Expt. No: 2(a)
Date: (a). LINEAR CONVOLUTION
AIM:
To write the program for finding the linear convolution of two signals using
MATLAB 9.0.
APPARATUS REQUIRED:
LINEAR CONVOLUTION:
ALGORITHM:
2. Generate the output sequence of the given two input signals using
‘conv’ command.
3. Plot the graph.
9
clc;
clear all;
close all;
x=input('enter the 1st sequences');
h=input('enter the 2nd sequences');
y=conv(x,h);
subplot(3,1,1);
stem(x);
ylabel('amplitude x(n)-->');
xlabel('n-->');
subplot(3,1,2);
stem(h);
ylabel('amplitude h(n)-->');
xlabel('n-->');
subplot(3,1,3);
stem(y);
ylabel('amplitude y(n)-->');
xlabel('n-->');
disp('the result b');y
10
INPUT:
Enter the 1st sequences [1 2 3 4]
Enter the 2nd sequences [4 3 2 1]
the resultant b
y = 4 11 20 30 20 11 4
OUTPUT:
RESULTS :-
Thus the program for linear convolution is written using MATLAB and
verified.
11
Expt. No: 2(b)
Date: (b)CIRCULAR CONVOLUTION
AIM:
To write the program for finding the circular convolution of two signals using
MATLAB 9.0.
APPARATUS REQUIRED:
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.
12
clc;
clear all;
close all;
a=input('x1(n)');
b=input('x2(n)');
N=length(b);
M=length(a);
c=fft(a,N);
d=fft(b);
e=c.*d;
f=ifft(e);
disp('circular convolution output');
disp(f);
subplot(1,3,1);
stem(a);
xlabel('time-->');
ylabel('amplitude');
title('x1(n)');
subplot(1,3,2);
stem(b);
xlabel('time');
title('x2(n)');
subplot(1,3,3);
stem(f);
xlabel('time');
13
title('output');
INPUT:
x1(n) [1 2 3 4]
x2(n) [4 3 2 1]
Circular convolution output:
24 22 24 30
RESULTS:-
Thus the program for circular convolution is written using MATLAB and
verified.
14
Expt. No: 3
Date: CORRELATION
%Cross-CorrelationandAuto-Correlationoffinitesequences
AIM:
To write the program for finding the correlation of two signals using MATLAB 9.0.
APPARATUS REQUIRED:
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.
15
clc;clearall;closeall;
x=input('EntertheSignalx=');
y=input('EntertheSignaly=');
rxy=xcorr(x,y);%cross-correlation of x and y
rxx=xcorr(x);%Auto-Correlation of x
subplot(221);
stem(x);
xlabel(Time');
ylabel('Amplitude');
title('x(n)');
subplot(222);
stem(y);
xlabel('Time');
title('y(n)');
subplot(223);
stem(rxy);
xlabel('Lag');
title('rxy(k)');
subplot(224);
stem(rxx);
xlabel('Lag');
ylabel('Amplitude'); title('rxx(k)');
16
Enter the Signal x=[1 -1 2 3 1]
Enter the Signal y=[2 1 -2 1 4 4 1]
RESULTS:-
Thus the program for correlation of two signals is written using MATLAB and
verified.
Expt. No: 4
17
Date: CALCULATION OF FFT
AIM:
To write the program for calculating the N-point FFT and IFFT of given input
sequence using MATLAB 9.0.
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.
clc;
18
clear all;
close all;
a=input('enter the sequence');
b=fft(a);
disp('fft of a sequence');
disp(b);
c=abs(b);
disp('magnitude plot');
disp(c);
d=angle(b);
disp('phase plot');
disp(d);
subplot(3,3,1);
stem(a);
xlabel('time');
title('input');
subplot(3,3,2);
stem(b);
xlabel('time');
title('fft');
subplot(3,3,3);
stem(c);
xlabel('time');
title('magnitude plot');
19
subplot(3,3,4);
stem(d);
xlabel('time');
title('phaseplot');
Enter the sequence [2 2 2 2 1 1 1 1]
OUTPUT:
fft of a sequence
Columns 1 through 8
12.0000, 1.0000 - 2.4142i , 0 ,1.0000 - 0.4142i ,0 , 1.0000 + 0.4142i , 0 , 1.0000 + 2.4142i
Magnitude plot
12.0000 2.6131 0 1.0824 0 1.0824 0 2.6131
Phase plot
0 -1.1781 0 -0.3927 0 0.3927 0 1.1781
RESULTS:-
Thus the program for fft is written using MATLAB and verified.
Expt. No: 5
20
Date: CALCULATION OF DFT
AIM:
To write the program for calculating the N-point DFT and IDFT of given input
sequence using MATLAB 9.0.
APPARATUS REQUIRED:
ALGORITHM:
1. Get the input sequence and its length.
2. Calculate the DFT and Inverse DFT
of given sequence using user defined function.
3. Plot the magnitude and phase response of DFT sequence.
4. Plot the IDFT response.
clear all;
21
clc;
n=8;
x=[1,1,1,1,1,1,0,0];
subplot(3,1,1);
stem(x);
xlabel('n-->');
ylabel('amp-->');
title('input');
A=dftmtx(n);
y=A*x';
subplot(3,1,2);
stem(y);
xlabel('n-->');
ylabel('amp-->');
title('dft');
disp(y);
c=abs(y);
disp('magnitude plot');
disp(c);
d=angle(y);
disp('phase plot');
disp(d);
subplot(3,3,1);
stem(x);
xlabel('time');
ylabel('amp');
22
title('input');
subplot(3,3,2);
stem(y);
xlabel('time');
ylabel('amp');
title('dft');
subplot(3,3,3);
stem(c);
xlabel('time');
title('magnitude plot');
subplot(3,3,4);
stem(d);
xlabel('time');
title('phaseplot');
OUTPUT:
23
6.0000
-0.7071 - 1.7071i
1.0000 - 1.0000i
0.7071 + 0.2929i
0
0.7071 - 0.2929i
1.0000 + 1.0000i
-0.7071 + 1.7071i
magnitude plot:
6.0000, 1.8478 , 1.4142, 0.7654, 0, 0.7654, 1.4142, 1.8478
phase plot:
0, -1.9635, -0.7854, 0.3927, 0, -0.3927, 0.7854, 1.9635
RESULTS:-
Thus the program for dft is written using MATLAB and verified.
Expt. No: 6
24
Date: SAMPLING AND ALIASING OF EFFECT
AIM
To sample the signal and observe the aliasing effect of that signal using
MATLAB 9.0.
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.
clear all;
25
clc;
close all;
t=-5:0.1:5;
T=1;
fm=1/T;
x=cos(2*pi*fm*t);
fs1=input('enter the 1st frequency if input frequency is Hz');
fs2=input('enter the 2nd frequency if input frequency is Hz');
fs3=input('enter the 3rd frequency if input frequency is Hz');
n=-5:0.1:5;
y1=cos(2*pi*fm*n/fs1);
subplot(4,1,1);
plot(t,x);
xlabel('times-->');
ylabel('x(t)-->');
title('input');
subplot(4,1,2);
stem(n,y1);
xlabel('n-->');
ylabel('x(n)==>');
if fs1<2*fm
title('sampled signal for fs1 aliasing');
26
else
title('sampled for fs1');
end
subplot(4,1,3);
stem(n,y2);
xlabel('n-->');
ylabel('x(n)-->');
if fs2<2*fm
else
title('sampled signal for fs2');
end
subplot(4,1,4);
stem(n,y3);
xlabel('n-->');
ylabel('x(n)-->');
if fs3<2*fm
else
title('sampled signal for fs3');
end
INPUT:
27
Enter the 1st frequency if input frequency is hz 2
Enter the 2nd frequency if input frequency is hz 3
Enter the 3rd frequency if input frequency is hz 1
RESULTS:-
Thus the sample the signal and observe the aliasing effect of that signal using matlab
was verified
Expt. No: 7(a)

28
Date:
(a) BUTTERWORTH LOW PASS IIR FIFTER
AIM:
To design a Butterworth low pass digital IIR filter using MATLAB 9.0.
APPARATUS REQUIRED:
ALGORITHM:
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band frequencies.
3. Get the sampling frequency.
4. Calculate the order of the filter using
5.Find the filter co-efficient.
6.Draw the magnitude and phase response.
clc;
29
clear all;
clf;
alphap=input('Enter the passband attenuation:');
alphas=input('Enter the stopband attenuation:');
fp=input('Enter the passband frequency:');
fs=input('Enter the stopband frequency:');
f=input('Enter the sampling frequency:');
sip=2*fp/f;
sis=2*fs/f;
[n,sic]=buttord(sip,sis,alphap,alphas,'s');
[b a]=butter(n,sic,'low','s');
t=0:0.01:pi;
[h,p]=freqs(b,a,t);
magnitude=20*log(abs(h));
subplot(2,2,1);
plot(p/pi,magnitude);
xlabel('Normailsed frequency-->');
ylabel('Gain in db-->');
title('Magnitude response of Analog LPF');
ang=angle(h);
subplot(2,2,2);
plot(p/pi,ang);
xlabel('Normalised frequency-->');
ylabel('Phase in radian-->');
30
title('Phase response of analog LPF');
disp('Order of filter:');
disp(n);
disp('cut of frequency:');
disp(sic);
disp('TRANSFER FUNCTION FOR ANALOG LPF FILTER');
disp('coefficient of numerator');
disp(b);
disp('coefficient of denominator');
disp(a);
[bz,az]=bilinear(b,a,f);
w=0:0.01:pi;
[hz,pz]=freqz(bz,az,w);
magnitude=20*log(abs(hz));
subplot(2,2,3);
plot(pz/pi,magnitude);
title('Magnitdue response of Digital LPF');
ang=angle(hz);
subplot(2,2,4);
plot(pz/pi,ang);
ylabel('phase in radian-->');
31
title('Phase response of Digital LPF');
disp('TRANSFER FUNCTION FOR DIGITAL; LPF FILTER');
disp(bz);
disp(az);
INPUT:
Enter the passband attenuation: 2
Enter the stopband attenuation: 4
Enter the passband frequency: 50
Enter the stopband frequency: 70
Enter the sampling frequency: 50
OUTPUT:
Order of filter:
cut of frequency:
2.5251
TRANSFER FUNCTION FOR ANALOG LPF FILTER
coefficient of numerator
0 0 6.3761
coefficient of denominator
1.0000 3.5710 6.3761
TRANSFER FUNCTION FOR DIGITAL; LPF FILTER
32
0.0006 0.0012 0.0006
1.0000 -1.9286 0.9311
RESULTS:-
Thus the magnitude response and phase response of Digital Butter

worth Low pass filter was verified.
Expt. No: 7(b)
33
Date: (b) BUTTERWORTH HIGH PASS IIR FIFTER
AIM:
To design a Butterworth high pass digital IIR filter using MATLAB 9.0.
APPARATUS REQUIRED:
ALGORITHM:
1. Get the pass band and stop band ripples.
2. Get the pass band and stop band frequencies.
4. Calculate the order of the filter using
5.Find the filter co-efficient.
6.Draw the magnitude and phase response.
clc;
34
clear all;
clf;
alphap=input('Enter the passband attenuation:');
alphas=input('Enter the stopband attenuation:');
fp=input('Enter the passband frequency:');
fs=input('Enter the stopband frequency:');
f=input('Enter the sampling frequency:');
sip=2*fp/f;
sis=2*fs/f;
[n,sic]=buttord(sip,sis,alphap,alphas,'s');
[b a]=butter(n,sic,'high','s');
t=0:0.01:pi;
[h,p]=freqs(b,a,t);
magnitude=20*log(abs(h));
subplot(2,2,1);
plot(p/pi,magnitude);
xlabel('Normailsed frequency-->');
title('Magnitude response of Analog HPF');
ang=angle(h);
subplot(2,2,2);
plot(p/pi,ang);
ylabel('Phase in radian-->');
35
title('Phase response of analog HPF');
disp('Order of filter:');
disp(n);
disp('cut of frequency:');
disp(sic);
disp('TRANSFER FUNCTION FOR ANALOG HPF FILTER');
disp(b);
disp(a);
[bz,az]=bilinear(b,a,f);
w=0:0.01:pi/10;
[hz,pz]=freqz(bz,az,w);
magnitude=20*log(abs(hz));
subplot(2,2,3);
plot(pz/pi,magnitude);
title('Magnitdue response of Digital HPF');
ang=angle(hz);
subplot(2,2,4);
plot(pz/pi,ang);
ylabel('phase in radian-->');
36
title('Phase response of Digital HPF');
disp('TRANSFER FUNCTION FOR DIGITAL; HPF FILTER');
disp(bz);
disp(az);
INPUT:
Enter the passband attenuation: 2
Enter the stopband attenuation: 4
Enter the passband frequency: 70
Enter the stopband frequency:50
Enter the sampling frequency:50
OUTPUT:
Order of filter:
cut of frequency:
2.2177
TRANSFER FUNCTION FOR ANALOG HPF FILTER
1 0 0
coefficient of denominator 1.0000 3.1364 4.9184
TRANSFER FUNCTION FOR DIGITAL; HPF FILTER
37
0.9691 -1.9383 0.9691
1.0000 -1.9373 0.9392
RESULTS:-
Thus the magnitude response and phase response of Digital
Butter worth High pass filter was verified
Expt. No: 8
38
Date: CHEBSHEV FILTER
AIM: -
TO write a MATLAB program to plot magnitude response and phase response of

digital Chebyshev type-1 Low pass filter
ALGORITHM:-
1. Get the passband and stopband ripples

2. Get the passband and stopband edge frequencies
3. Calculate the order of the filter using ‘ cheb1ord ’ function
4. Find the filter coefficients, using ‘cheby1’ function
5. Draw the magnitude and phase response
39
clear all;
alphap=2;
alphas=20;
wp=[0.2*pi,0.4*pi];
ws=[0.1*pi,0.5*pi];
[n,wn]=cheb1ord(wp/pi,ws/pi,alphap,alphas);
[b,a]=cheby1(n,alphap,wn);
w=0.1:0.01:pi;
[h,ph]=freqz(b,a,w);
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);
plot((ph/pi),m);
grid;
ylabel('gain in db');
xlabel('Normalised Frequency');
title('chebshev type I band pass filter');
subplot(2,1,2);
plot((ph/pi),an);
grid;
xlabel('Normalised Frequency');
ylabel('Phase in radians');
40
RESULTS:-
Thus the Amplitude response and phase response of chebyshev type 1 Low
pass filter was verified
Expt. No: 9(a)
41
Date: (a) FIR FILTER USING HANNING WINDOW
AIM:
To design a FIR filter using Hanning window with MATLAB 9.0.
APPARATUS REQUIRED:
System with MATLAB 9.0
ALGORITHM:
1. Get the pass band and stop band ripple and frequency.
3. Find the order of filter N.
4. Design the lowpass,High pass,Band pass and Band stop filter.
5. Plot the magnitude response of all the filters.
clc;
clf;
42
clear all;
wp=input('enter the passband frequency:');
ws=input('enter the stopband frequency:');
n=input('enter the samples:');
m=max(wp,ws);
fs=3*m;
fp=2*wp/fs;
fstop=2*ws/fs;
w=0:.01:pi;
%LPF:
b=fir1(n,fp,'low',hanning(n+1));
h=freqz(b,1,w);
subplot(2,2,1);
plot(w/pi,abs(h));
xlabel('normalised frequency');
ylabel('magnitude');
title('low pass filter');
%HPF:
b=fir1(n,fp,'high',hanning(n+1));
h=freqz(b,1,w);
subplot(2,2,2);
plot(w/pi,abs(h));
title('high pass filter');
%BPF:
b=fir1(n,[fp fstop],'bandpass',hanning(n+1));
h=freqz(b,1,w);
subplot(2,2,3);
plot(w/pi,abs(h));
title('band pass filter');
%BSF:
b=fir1(n,[fp fstop],'stop',hanning(n+1));
h=freqz(b,1,w);
subplot(2,2,4);
plot(w/pi,abs(h));
43
title('band stop filter');
INPUT: Enter the passband frequency:3.15

Enter the stopband frequency:7.5
Enter the samples:38
OUTPUT:
RESULTS:-
Thus the magnitude response and phase response of fir LPF, HPF,BPF, BSF
using hanning window was verified.
Expt. No: 9(b)

44
Date:
(b) FIR FILTER USING HAMMING WINDOW
AIM:
To design a FIR filter using Hamming window with MATLAB 9.0.
APPARATUS REQUIRED:
System with MATLAB 9.0
ALGORITHM:
1. Get the pass band and stop band ripple and frequency.
3. Find the order of filter N.
4. Design the lowpass,High pass,Band pass and Band stop filter.
5. Plot the magnitude response of all the filters.
45
clc;
clf;
clear all;
wp=input('enter the passband frequency:');
ws=input('enter the stopband frequency:');
n=input('enter the samples:');
m=max(wp,ws);
fs=3*m;
fp=2*wp/fs;
fe=2*ws/fs;
w=0:.01:pi;
%LPF:
b=fir1(n,fp,'low',hamming(n+1));
h=freqz(b,1,w);
subplot(2,2,1);
plot(w/pi,abs(h));
title('low pass filter');
%HPF:
b=fir1(n,fp,'high',hamming(n+1));
h=freqz(b,1,w);
subplot(2,2,2);
plot(w/pi,abs(h));
title('high pass filter');
%BPF:
b=fir1(n,[fp fe],'bandpass',hamming(n+1));
h=freqz(b,1,w);
subplot(2,2,3);
plot(w/pi,abs(h));
title('band pass filter');
%BSF:
b=fir1(n,[fp fe],'stop',hamming(n+1));
h=freqz(b,1,w);
subplot(2,2,4);
46
plot(w/pi,abs(h));
title('band stop filter');
INPUT:
Enter the passband frequency:3.15

Enter the stopband frequency:7.5
Enter the samples:38
OUTPUT:
RESULTS:-
Thus the magnitude response and phase response of fir LPF, HPF,BPF, BSF
using hamming window was verified.
47
Expt. No: 10 (a)
Date: INTERPOLATION BY POLYPHASE DECOMPOSITION
(a) UP SAMPLING
AIM:
To write a program to compute Convolution and m-fold interpolation by
polyphase decomposition.
APPARATUS REQUIRED:
ALGORITHM:
1. Get the input sequence.
2. Get the filter coefficients and also the interpolation factor.
3. Find the response by using convolution.
4. Plot the graph.
clc;
48
N=10;
n=0:1:N-1;
x=sin(2*pi*n/10)+sin(2*pi*n/5);
L=3;
x1=[zeros(1,L*N)];
n1=1:1:L*N;
j=1:L:L*N;
x1(j)=x;
subplot(2,1,1);
stem(n,x);
xlabel('n');
ylabel('x');
title('input sequence');
subplot(2,1,2);
stem(n1,x1);
xlabel('n');
ylabel('x1');
title('upsampled sequence');
OUTPUT :
RESULT:
Thus we compute Convolution and m-fold interpolation by Polyphase

decomposition in MATLA
49
Expt. No: 10 (b)
Date:
DECIMATION BY POLYPHASE DECOMPOSITION
(b)DOWN SAMPLING
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.
50
clc;
N=50;
n=0:1:N-1;
x=sin(2*pi*n/20)+sin(2*pi*n/15);
M=2;
x1=x(1:M:N);
n1=1:1:N/M;
subplot(2,1,1);
stem(n,x);
xlabel('n');
ylabel('x');
title('input sequence');
subplot(2,1,2);
stem(n1-1,x1);
xlabel('n');
ylabel('x1');
title('downsampled sequence');
OUTPUT:
RESULT:
Thus we compute Convolution and m-fold decimation by Polyphase

decomposition in MATLAB
51
TMS320C5XProcessor Based
Experiments
1. STUDY OF VARIOUS ADDRESSING MODES
2. (a) CIRCULAR CONVOLUTION
(b) LINEAR CONVALUTION
3. SINE WAVE GENERATION
4. SQUAREWAVE GENERRATION
5. SAWTOOTH WAVE GENERRATION
6. TRIANGULAR WAVEFORM GENERRATION
7. IMPLEMENTATION OF IIR FILTER
8. IMPLEMENTATION OF FIR FILTER
52
Expt. No: 1
STUDY OF TMS320C50 PROCESSOR

Expt. No: 1
STUDY OF ARCHITECTURE OF DIGITAL SIGNAL PROCESSOR
AIM:
To study the Architecture of TMS320VC67XX DSP processor. And TMS320C50

PROCESSOR
Architecture of TMS320VC67XX DSP processor
he hardware experiments in the DSP lab are carried out on the Texas
Instruments TMS320C6713 DSP Starter Kit (DSK), based on the TMS320C6713
floating point DSP running at 225MHz. The basic clock cycle instruction time is
1/(225 MHz)= 4.44 nanoseconds. During each clock cycle, up to eight instructions
can be carried out in parallel, achieving up to 8×225 = 1800 million instructions per
second (MIPS). The DSK board includes a 16MB SDRAM memory and a 512KB
Flash ROM. It has an on-board 16-bit audio stereo codec (the Texas Instruments
AIC23B) that serves both as an A/D and a D/A converter. There are four 3.5 mm
audio jacks for microphone and stereo line input, and speaker and headphone
outputs. The AIC23 codec can be programmed to sample audio inputs at the
following sampling rates: fs = 8, 16, 24, 32, 44.1, 48, 96 kHz The ADC part of the
codec is implemented as a multi-bit third-order noise-shaping delta-sigma
converter) that allows a variety of oversampling ratios that can realize the above
choices of fs. The corresponding oversampling decimation filters act as anti-
aliasing pre-filters that limit the spectrum of the input analog signals effectively to
the Nyquist interval [−fs/2,fs/2]. The DAC part is similarly implemented as a multi-
53
bit second-order noise-shaping delta-sigma converter whose oversampling

interpolation filters act as almost ideal reconstruction filters with the Nyquist
interval as their pass band. The DSK also has four user-programmable DIP
switches and four LEDs that can be used to control and monitor programs running
on the DSP. All features of the DSK are managed by the Code Composer Studio
(CCS). The CCS is a complete integrated development environment (IDE) that
includes an optimizing C/C++ compiler, assembler, linker, debugger, and program
loader. The CCS communicates with the DSK via a USB connection to a PC. In
addition to facilitating all programming aspects of the C6713 DSP, the CCS can
also read signals stored on the DSP‟s memory, or the SDRAM, and plot them in
the time or frequency domains. The following block diagram depicts the overall
operations involved in all of the hardware experiments in the DSP lab. Processing
is interrupt-driven at the sampling rate fs, as explained below.
Architecture généraledu starter kit TMS320C50

Architecture generated starter kit TMS320C50
Schéma de la carte
La carte starter kit TMS320C50 que vous avez à disposition contient (Cf. Figure 1)
- Un DSP TMS320C50 (16 bits, virgule fixe, cycle 50ns)
- Une mémoire PROM de 32 ko
- Un circuit analogique d'interface (AIC) TLC32040
54
- Des connecteurs RCA entrée/sortie audio

- Un circuit d'horloge
- Un port-série qui permet le chargement de codes via un PC
Figure 1 : Schéma général de la carte starterkit TMS320C5x
Organisation de la mémoire
La carte starter kit TMS320C50 comprend une mémoire PROM-8bits de 32 K
réservée pour le noyau (kernel) qui est parcourue au démarrage du DSP et une
mémoire RAM de 10 K pour les programmes application. Au niveau CPU (Voir
Figure 3) du TMS320C50, on trouve une Single-Access RAM (SARAM) et trois
(B0,B1,B2) Dual-Access RAM (DARAM). La DARAM B2 est une mémoire tampon
des registres de status.La plan général d'adressage mémoire de la carte TMS320C50
est donnée Figure 2.
55
Figure 2 : Plan d'adressagemémoire de la carte starterkit TMS320C5x

Dans le fichier pass.asm (Cf. Annexe 6.2), on peut relever les instructions .ds
0F00h, .ps 080Ah et .ps 0A00h qui permettent d'implanter à différentes adresses
mémoire les mots de données, les routines d'interruption et le code d'initialisation.
Schéma fonctionnel de l'unité centrale de calcul (CPU)

Le schéma fonctionnel du CPU du TMS320C50 est donné Figure 2. Vous en
trouverez une description complète p. 3-4 de la référence [1].
On peut noter sur ce schéma :
- la présence de deux bus (data, Program),
- le type d'opérande du multiplieur et leur taille,
- le nombre et la position des 'scaler',

- la taille de l'accumulateur.
56
57
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. Initialize data pointer with 100H data.
2. Load the accumulator with first data.
3. Add the second data with accumulator content.
4. 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.
58
PROGRAM:
;program for immediate addressing mode
.mmregs
.text
START:
LDP #100H
LACC #1241H
ADD #1200H
SACL 2H
H: B H
;program for direct addressing mode
.mmregs
.text
START:
LDP #100H
LACC 0H
ADD 1H
SACL 2H
H: B H
;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: B H
RESULT:
Thus, the arithmetic operations using TMS320 kit was performed
successfully.
59
Expt. No: 2(a)
Date: CIRCULAR CONVOLUTION
AIM:
To write a program to find the Convolution of two sequences X(n) and

H(n) Using TMS320C50 debugger.
APPARATUS REQUIRED:
1. System with TMS 320C50 debugger software
2. TMS 320C50 Kit.
3. RS232 cable
PROGRAM LOGIC:
1. Start the Program.
2. Load the LSB’s 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.
.mmregs
.text
60
START:
LDP #100H
LACC 0H ;length of the input is given in 8000
SUB #1H
SACL 1H
LAR AR0,1H
LAR AR1,#8060H;
LAR AR2,#8100H
COPYX2:
MAR *,AR1
LACC *+
MAR *,AR2
SACL *+
MAR *,AR0
BANZ COPYX2,*-
LAR AR0,1H
LAR AR2,#8010H
LOOP3:
LAR AR1,#8060H ;give the inputs x1[n] & h2[n] in AR1 & AR3
LAR AR3,#8050H
LAR AR4,1H
ZAP
LOOP:
MAR *,AR3 ;multiply x1[n] & X2[n] and add the
;multiplication
LT *+
MAR *,AR1 ;output
MPY *+
SPL 5H ;store the partial result at location 8005
ADD 5H ; add the partial result with accumulator
MAR *,AR4
BANZ LOOP,*-,
MAR *,AR2
;outputs of correlation are stored in AR2
SACL *+
CALL ROTATE
LOOP2:
MAR *,AR0
BANZ LOOP3,*-
61
H: B H
ROTATE:
LDP #100H ;rotate the values of X1[n]
LACC 1H
SUB #1H
SACL 2H
LACC 0050H
SACB ; 8050 DATA MOVED TO THE ACCUMULATOR TO THE
ACCUMULATOR BUFFER
LAR AR3,#8051H
LAR AR5,#8070H
LAR AR6,2H
LOOP1:
MAR *,AR3
LACC *+
MAR *,AR5
SACL *+,
MAR *,AR6
BANZ LOOP1,*-; DATA FROM 8051-8053 TO 8070-8072
LACB ;MOVE THE DATA ACCUMULATOR BUFFER TO
ACCUMULATOR AS LAST DATA
MAR *,AR5
SACL *+
LACC #8070H
SAMM BMAR
LAR AR3,#8050H
MAR *,AR3
RPT #3H ;ROTATE 4 TIMES
BLDD BMAR,*+ ;BMAR AUTOMATICALLY INCREMENTED ,TO
COPY SHIFTED DATA TO 8050
RET
62
INPUT:
; 8000-0004
;X1(n) = 8050 - 0002
; 8051 - 0001
; 8052 - 0002
; 8053 - 0001
;H2(n ) = 8060 - 0001
; 8061 - 0002
; 8062 - 0003
; 8063 - 0004
;OUTPUT:
; 8010-000E
; 8011-0010
; 8012-000E
; 8013-0010
RESULT:
Thus the circular convolution of two sequences is done successfully by

using TMS320C50.
63
Expt. No: 2 (b)
Date: LINEAR CONVALUTION
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.
3. RS232 cable
LINEAR CONVOLUTION
1. Start the program.
2. LPD will load the 9LSB’s of address data memory. Location or immediate value
into the data pointer (DP) registers.
3. Load auxiliary register (LAR) loads the content of the address data
Memory location into the designated auxilary registers.
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 acc.
64
.mmregs
.text
START:
LDP #02H
LAR AR1,#8100H ; x(n) datas
lar ar0,#08200H ;h(n) datas
LAR AR3,#8300H ;y(n) starting
LAR AR4,#0007 ;N1+N2-1
;to fold the h(n) values
;************************
lar ar0,#8203H ; data mem 8200 to program mem c100(tblw)
lacc #0c100h
mar *,ar0
rpt #3
tblw *- ;to move 8203- 8200 to c100- c103
;padding of zerros for x(n) values
;**********************************
lar ar6,#8104h
mar *,ar6
lacc #0h
rpt #3h
sacl *+
;convalution operation starts
;******************************
LOP: MAR *,AR1
LACC *+
SACL 050H ;starting of the scope of multiplication
LAR AR2,#0153H ; end of the array, to be multiplied with h(n) {150+N1-1}
MAR *,AR2
ZAP
RPT #03H ;N1-1 times so that N1 times
MACD 0C100H,*-
APAC ;to accmulate the final product sample
MAR *,AR3
SACL *+
MAR *,AR4
BANZ LOP,*-
H: B H
INPUT ( x(n) )
65
;8100 - 1
;8101 - 3
;8102 - 1
;8103 - 3
;INPUT ( h(n) )
; 8200 - 0
; 8201 - 1
; 8202 - 2
; 8203 - 1
;OUTPUT ( y(n) )
; 8300 - 1
; 8301 - 5
; 8302 - 8
; 8303 - 8
; 8304 - 7
; 8305 - 3
; 8306 - 0
RESULT:
Thus the linear convolution of two sequences is done successfully by using

TMS320C50.
66
Expt. No: 3
Date: SINE WAVE GENERATION
AIM:
To write a program to generate Sine Wave using TMS320C50 debugger.
APPARATUS REQUIRED:
1.System with TMS 320C50 debugger software
2.TMS 320C50 Kit.
3.CR0
4.Function Generator.
ALGORITHM:
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.
7. Store the 16 LSBs of the accumulator into the address data memory
location.
8. Modify the current AR & ARP as specified.
9. Branch the contents of entire auxiliary register.
10. Load the content of address data memory location.
11. Repeat the Step.
67
SINE WAVE GENERATION ( 1 KHZ)

INCFREQ .set 10 ;minimum value 1 - change for increasing frequency
;every count will increase frequency in steps of 100hz
DECFREQ .set 0 ;minimu value 0 - change for decreasing frequency
;every count will decrease frequency by half
LENGTH .set 360
AMPLITUDE .set 5
delay .set 7 ;change to increase/decrease frequency in
;different steps;
TEMP .set 0
TEMP1 .set 1
.mmregs
.text
START:
LDP #100H
SPLK #TABLE,TEMP ;load start address of table
lar AR2,#( (LENGTH/INCFREQ)+(LENGTH*DECFREQ) )-1
lar ar3,#DECFREQ ;repeat counter for reducing frequency
CONT:
CALL DELAY
LACC TEMP ;load address of sine value
TBLR TEMP1 ;read sine data to TEMP1
LT TEMP1
MPY #AMPLITUDE
PAC
SACL TEMP1
mar *,ar3
banz repeat,*-
lar ar3,#DECFREQ ;repeat counter for reducing frequency
LACC TEMP
ADD #INCFREQ ;increase table value
repeat:
SACL TEMP ;store it
OUT TEMP1,4 ;send sine data to DAC
MAR *,AR2
BANZ CONT,*-
b START
DELAY:
lar ar7,#delay
mar *,ar7
back: banz back,*-
ret ;
TABLE
.word 100
.word 101
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.end
RESULT:
Thus the sine waveform was generated and executed using

TMS320C5X.
75
EX.NO:4
Date:
SQUAREWAVE GENERATION
AIM:
To write a program to generate square Wave using TMS320C50 debugger.
APPARATUS REQUIRED:
2. TMS320C50 Kit.
3. CR0
4. Function Generator.
ALGORITHM:
1. Start the program
3. Store the 16 LSBs of the accumulator into the address data memory
location.
5. Complement the contents of the accumulator.
76
.mmregs
.text
start:
ldp #100h
lacc #0fffh ;change this value for amplitude.
loop: sacl 0
rpt #0ffh ;change this value for frequency.
out 0,04 ;address for dac.
cmpl
b loop
.end
RESULT:
Thus the square waveform was generated and executed using

TMS320C5X.
77
Expt. No: 5
Date: SAWTOOTH WAVEFORM GENERRATION
AIM:
To write a program to generate saw tooth Wave using TMS320C50 debugger.
APPARATUS REQUIRED:
2. TMS320C50 Kit.
3. CR0
ALGORITHM:
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.
78
.MMREGS
.TEXT
START:
LDP #120H
LACC #0H ;change lower amplitude
SACL 0
LOOP: LACC 0
OUT 0,04H
ADD #05h ;change frequency
SACL 0
SUB #0FFFh ;change upper amplitude
BCND LOOP,LEQ
B START
.END
RESULT:
Thus the saw tooth waveform was generated and executed using
TMS320C5X.
79
Expt. No: 6
Date: TRIANGULAR WAVEFORM GENERRATION
AIM:
To write a program to generate Triangular Wave using TMS320C50 debugger.
APPARATUS REQUIRED:
2. TMS320C50 Kit.
3. CR0
TRIANGLE WAVE GENERATION:
2. Load the LSBs of the address data memory location.
3. Write full 16 bit pattern into 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.
80
AMPLITUDE .SET 15
FREQ .SET 43
TEMP .SET 0;
.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
.end
RESULT:
Thus the Triangular waveform was generated and executed using

TMS320C5X.
81
Expt. No: 7
Date: FIR FILTER
AIM:
To design a FIR low pass filter in serial mode.
82

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SEC - ECE EC8562 – DIGITAL SIGNAL PROCESSING

FAMILIARISATION WITH MATLAB

ylabel(‘text’) It adds text beside the y-axis on the current axis.

Date: GENERATION OF SIGNALS

Expt. No: 1(b) UNIT RAMP

INPUT: Enter the length of ramp sequence: 6

Expt. No: 1(c) UNIT SINE

Expt. No: 1(d) UNIT COSINE

Expt. No: 1(e) UNIT EXPONENTIAL

INPUT: Enter the length of the exponential sequence 3

Expt. No: 2(a)

Date: (a). LINEAR CONVOLUTION

x=input('enter the 1st sequences');

h=input('enter the 2nd sequences');

disp('the result b');y

Enter the 1st sequences [1 2 3 4]

Enter the 2nd sequences [4 3 2 1]

Expt. No: 2(b)

Date: (b)CIRCULAR CONVOLUTION

disp('circular convolution output');

Circular convolution output:

Enter the Signal x=[1 -1 2 3 1]

Enter the Signal y=[2 1 -2 1 4 4 1]

Date: CALCULATION OF FFT

a=input('enter the sequence');

Date: CALCULATION OF DFT

Date: SAMPLING AND ALIASING OF EFFECT

fs1=input('enter the 1st frequency if input frequency is Hz');

fs2=input('enter the 2nd frequency if input frequency is Hz');

fs3=input('enter the 3rd frequency if input frequency is Hz');

title('sampled signal for fs1 aliasing');

title('sampled for fs1');

title('sampled signal for fs2 aliasing');

title('sampled signal for fs2');

title('sampled signal for fs3 aliasing');

title('sampled signal for fs3');

Enter the 1st frequency if input frequency is hz 2

Enter the 2nd frequency if input frequency is hz 3

Enter the 3rd frequency if input frequency is hz 1

Expt. No: 7(a)

(a) BUTTERWORTH LOW PASS IIR FIFTER

alphap=input('Enter the passband attenuation:');

alphas=input('Enter the stopband attenuation:');

fp=input('Enter the passband frequency:');

fs=input('Enter the stopband frequency:');

f=input('Enter the sampling frequency:');

title('Magnitude response of Analog LPF');

title('Phase response of analog LPF');

disp('TRANSFER FUNCTION FOR ANALOG LPF FILTER');

title('Magnitdue response of Digital LPF');

title('Phase response of Digital LPF');

disp('TRANSFER FUNCTION FOR DIGITAL; LPF FILTER');

Enter the passband attenuation: 2

Enter the stopband attenuation: 4

Enter the passband frequency: 50

Enter the stopband frequency: 70

Enter the sampling frequency: 50

TRANSFER FUNCTION FOR ANALOG LPF FILTER

1.0000 3.5710 6.3761