EC2306 DSP LAB RECORD - Final PDF

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PA R I VA K K A M , P O O N A M A L L E E , C H E N N A I - 5 6
CHENNAI 600 056
DEPARTMENT OF ELECTRONICS AND

COMMUNICATION ENGINEERING
LAB CODE/NAME: EC2306 / DIGITAL SIGNAL PROCESSING LAB
NAME
REG NO
YEAR /SEM :
Third Year / Fifth Semester
www.Vidyarthiplus.com
SRINIVASA INSTITUTE OF ENGINEERING & TECHNOLOGY

PA R I VA K K A M , P O O N A M A L L E E , C H E N N A I - 5 6
REGISTER NUMBER:
BONAFIDE CERTIFICATE
Certified that this is a bonafide record of the work done by

--------------------------------------------------------------------------------
Third Year (Fifth Semester) of
Electronics and Communication Engineering Department in the DIGITAL

SIGNAL PROCESSING LABARTORY (EC2306) during academic year
2013- 2014.
Signature of the Staff
Signature of HOD
Submitted for the Practical Examination held on ---------------------------------------
Internal Examiner
External Examiner
INDEX
S.No
Date
Name Of The Experiment

USING MATLAB
1.
Study of Matlab functions
2.
Generation of signals
3.
Linear and circular convolution
4.
Sampling and effect of aliasing
5.
Design of FIR filter using hamming window
6.
Design of FIR filter using hanning window
7.
Design of FIR filter using blackmann window
8.
Design of butterworth IIR filter
9.
Design of chebyshev IIR filter
10.
Calculation of FFT and IFFT of a signal
11.
Decimation by polyphase decomposition
12.
Computation of DFT and IDFT
13.
Impulse invariant and Bilinear transformation
USING TMS320C5X
Page
No
Staff
Sign
STUDY OF MATLAB FUNCTIONS
AIM:
To study various MATLAB function.
FUNCTIONS:
clc: Clear the command window.
xlabel: Label for X-axis.
ylabel: Label for Y-axis.
title: Title to the graph.
gird: To display grid lines in graph.
abs (x): This command computes the absolute value of the element of
x,when x is complex abs(x) is the magnitude of the element of x
angle (x): This command computer the phase angle in radians of
vector x.
log10(x): This command computes the logarithm to the base 10 if the
element x.
rem(n,m): This command determines the remainder after dividing n
by N.
log (x): This command determines the remainder natural logarithm of
each element in x.
mod (m,N): This command computes m mod N.
ones (N): Its an N-by-N matrix of ones.
ones (m,n): Its an m-by-n matrix of ones.
zeros (n): Its an n-by-n matrix of zeros .
plot (x,y): Plot vector y versus vector x.
subplot (a,b,c): It breaks the figure window in a-by-b matrix if small
axis selects the Cth axis for the current point.
[b,w] freq z (b,a,N): It returns the N-point freq response vector h and
n-point freq vector w in redians/sample of fig filter described by (b,a)
where a and b are numerator and denominator co-efficient vector.
h = freqz (b,a,w): It returns the freq response at frequencies
designated in vector w in radians/samples.
h = filter (b,a,x): It filters the data in vector x with the filter data y by
the vector a and b to create the filtered data y.
y = conv (x,h): It computes the convolution of two sequence x and h.
h = impz (b,a,w): It computes N samples of the impules response of

the filter describe b and a.
y = fft (x): It computes the FFT of the vector and stores it in vector Y.
y = fft (x,n): It computes N-point FFT of the vector x by padding the
sequence x with zeros if the sequence length is less than the N-point
truncated if it has more and stores it in vector Y.
ifft (x): It computes the IFFT of the vector x.
ifft (x,N): It computes N-point IFFT of the vector x.
z plane (z,p): Plot the poles and zeros supplied by the column vector
z and p.
z plane (num,den): plots the poles and zeros of the transfer function
represented by the rows in num and dem.
h = impz (num,den,N): It computes the impulse response of the
digital filter transfer function represented by num and dem.
[N,Wn] = buttord (wp,ws,rp,rs): Find the lowest order N of digital
butterworth filter that loses no more than rp db in pass band and has
atleast rs db of attenuation in stop band, wp and ws are the passband
and stopband edge frequencies normalized from 0 to 1.
[B,A] = butter {N,Wn,high}: Design an Nth order high pass digital
butterworth filter.
[N,Wn] = cheb1ord (wp,ws,rp,rs): Finds the lowest order N of the
digital chebyshev type I filter that loses no more than rp db in pass
band and has atleast rs db of attenuation in stop band wp and ws are the
passband and stopband edge frequencies mormalised from 0 to 1.
[N,Wn] = cheb2ord (wp,ws,rp,rs): Finds the lowest order N of the
digital chebyshev type II filter that loses no more than rp db in pass
band and has atleast rs db of attenuation in stop band wp and ws are the
passband and stopband edge frequencies mormalised from 0 to 1.
[B,A] = cheby2 {N,R,Wn}: Design an Nth order low pass digital
chebyshev filter with the stop band ripple R describe down and stop
band edge frequency Wn.
[B,A] = cheby2 {N,R,Wn,high}: Design an Nth order high pass
digital chebyshev filter
[B,A] = cheby2 {N,R,Wn,stop}: Design an Nth order band stop
digital chebyshev filter if [w1 and w2].
[B,A] = cheby1 {N,R,Wn}: Design an Nth order low pass digital

chebyshev type I filter with R db of ripples in the stopband and R db
of ripple ine pass band, cheby1 returns the filter co-efficient N+1
vector with B (num) and A (den).
[B,A] = cheby1 {N,R,Wn,high}: Design an Nth order high pass
digital chebyshev type I filter.
[numd,dend] = bilinear(num,den,fs): Converts the s domain transfer
function specified by num and den containing numerator and
denominator transfer function coefficient into z domain transfer
function are store in the vetor numd, dend.
W = boxcar (N): Return the N point rectangular window.
W = Bartlett (N): Return the N point bartlett triangular window.
W = hamming (N): Return N point, symmetric hamming window in
coluwn vector.
hann (N): Return N point window symmetric hamming window in
coluwn vector.
hanning (N): Return N point symmetric hamming window in coluwn
vector. In this case the first and cast zero-weighted window samples
are not included.
blackman (N): Return N point symmetric blackman window in
coluwn vector.
B = firl (N,Wn): Design a N order low pass FIR digital filter and
return the filter, the filter coefficient in length N+1 vector B. The
cut-off freq Wn must be between 0 <Wn<10, with 1.0 corresponds to
half the sample rate.
B = firl (N,Wn,high): Design a N order FIR digital high pass filter
B = firl (N,Wn,stop): It is a bandstop filter if wn=[w,wn].
y = decimate (x,m): It resamples the sequence in the vector x at m
times the original sampling rate.
RESULT:
Thus, the MATLAB function used in the program are studied.
PROGRAM:
% program for Unit Step Signal
clc;
t=0:1:10;
y=ones(1,11);
subplot(3,2,1);
plot(t,y,'k');
xlabel('Time');
ylabel('Amplitude');
title('Unit step Signal');
% program for Unit ramp Signal
clc;
t1=0:1:10;
y1=t1;
subplot(3,2,2);
plot(t1,y1,'k');
xlabel('Time');
title('Unit Ramp Signal');
% program for Sine wave
clc;
t2=0:0.1:10;
y2=sin(t2);
subplot(3,2,3);
plot(t2,y2,'k');
xlabel('Time');
title('Sine Wave');
% program for Cosine Wave
clc;
t3=0:0.1:10;
y3=cos(t3);
subplot(3,2,4);
plot(t3,y3,'k');
xlabel('Time');
title('Cosine Wave');
% program for Square wave
clc;
t4=0:0.001:10;
y4=square(t4);
subplot(3,2,5);
plot(t4,y4,'k');
xlabel('Time');
title('Square Wave');
% program for Sawtooth Wave
clc;
t5=0:0.1:10;
y5=sawtooth(t5);
subplot(3,2,6);
plot(t5,y5,'k');
xlabel('Time');
title('Sawtoth Wave');
% program for Unit Step Sequence

clc;
N=input('Enter the length of unit step sequence(N)= ');
n=0:1:N-1;
y=ones(1,N);
subplot(3,2,1);
stem(n,y,'k');
xlabel('Time');
title('Unit step Sequence');
% program for Unit ramp Sequence

N1=input('Enter the length of the unit ramp sequence(N)= ');
n1=0:1:N1-1;
y1=n1;
subplot(3,2,2);
stem(n1,y1,'k');
xlabel('Time');
title('Unit Ramp Sequence');
% program for Sine wave
N2=input('Enter the length of sinosoidal sequence(N)= ');
n2=0:0.1:N2-1;
y2=sin(n2);
subplot(3,2,3);
stem(n2,y2,'k');
xlabel('Time');
title('Sinosoidal Sequence');
% program for Cosine Wave
N3=input('Enter the length of cosine sequence(N)= ');
n3=0:0.1:N3-1;
y3=cos(n3);
subplot(3,2,4);
stem(n3,y3,'k');
xlabel('Time');
title('Cosine Sequence');
% program for exp sequence
N4=input('Enter the length of Exponential sequence(N)= ');
n4=0:1:N4-1;
a=input('Enter the value of Exponential sequence(a)= ');
y4=exp(a*n4);
subplot(3,2,5);
stem(n4,y4,'k');
xlabel('Time');
title('Exponential Sequence');
% program for Unit Impulse
n5=-3:1:3;
y5=[zeros(1,3),ones(1,1),zeros(1,3)];
subplot(3,2,6);
stem(n5,y5,'k');
xlabel('Time');
title('Unit Impulse');
OUTPUT FOR GENERATION OF CONTINUOUS TIME SIGNAL:
OUTPUT FOR GENERATION OF DISCRETE TIME SIGNAL:

Enter the length of unit step sequence(N)= 10
Enter the length of the unit ramp sequence(N)= 10
Enter the length of sinosoidal sequence(N)= 10
Enter the length of cosine sequence(N)= 5
Enter the length of Exponential sequence(N)= 5
Enter the value of Exponential sequence(a)= 1
RESULT:
PROGRAM:
% PROGRAM FOR LINEAR CONVOLUTION
%input seq
clc;clear all;close all;
x=input('Enter the input sequence(n)= ');
N1=length(x);
n=0:1:(N1-1);
subplot(3,1,1);
stem(n,x,'k');
xlabel('n----->');
title('input sequence X(n)');
%impulse seq.
h=input('Enter the impulse sequence h(n)= ');
N2=length(h);
n1=0:1:N2-1;
subplot(3,1,2);
stem(n1,h,'k');
xlabel('n--->');
title('impulse sequence h(n)');
%Output conv.
y=conv(x,h);
N=N1+N2-1;
n2=0:1:N-1;
subplot(3,1,3);
stem(n2,y,'k');
xlabel('Time');
title('Linear Convolution of two sequence');
% PROGRAM FOR CIRCULAR CONVOLUTION

clc;
clear all;
close all;
x=input('Enter the impluse sequence x1(n)= ');
N1=length(x);
h=input('Enter the impulse sequence x2(n)= ');
N2=length(h);
n=0:1:N1-1;
subplot(3,1,1);
stem(n,x);
xlabel('n--->');
title('x1(n) Sequence Response');
n=0:1:N2-1;
subplot(3,1,2);
stem(n,h);
xlabel('n--->');
title('x2(n) 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;
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 x1(n) & x2(n) is');
disp(y);
subplot(3,1,3);
stem(y);
xlabel('n--->');
title('Convolution of x1(n)&x2(n) Response');
OUTPUT FOR LINEAR CONVOLUTION:

Enter the input sequence(n)= [1 2 3 4]
Enter the impulse sequence h(n)= [4 3 2 1]
OUTPUT FOR CIRCULAR CONVOLUTION:

Enter the impluse sequence x1(n)= [1 2 3 4 5 6]
Enter the impulse sequence x2(n)= [2 2 3 4]
Convolution of x1(n) & x2(n) is
45 44 37 24 35 46
RESULT:
PROGRAM:
%prog for sampling and effect of alliasing
clc;close all; clear all;
% Time period of 50 Hz signal with 0.1s duration at 1000Hz
t=0:0.001:0.1;
fm = 15;
x=sin(2*pi*fm*t);
figure(1)
plot(t,x);
xlabel('Time');
title('Original Analog signal');
%Analog sig sampled at fs<<2fm
fs=10;
n=0:1/fs:0.1;
xn=sin(2*pi*fm*n);
figure(2)
subplot(2,1,1);
stem(n,xn);
xlabel('Time');
title('Undersampled fs<<2fm signal');
subplot(2,1,2);
plot(n,xn);
xlabel('Time');
title('Reconstructed undersampled fs<<2fm signal');
%Analog sig sampled at fs=2fm

fs=30;
n=0:1/fs:0.1;
xn=sin(2*pi*fm*n);
figure(3)
subplot(2,1,1);
stem(n,xn);
xlabel('Time');
title('Sampled at Nyquist rate fs=2fm signal');
subplot(2,1,2);
plot(n,xn);
xlabel('Time');
title('Reconstructed Nyquist rate fs=2fm signal');
%Analog sig sampled at fs>>2fm

fs=500;
n=0:1/fs:0.1;
xn=sin(2*pi*fm*n);
figure(4)
subplot(2,1,1);
stem(n,xn);
xlabel('Time');
title('Oversampled fs>>2fm signal');
subplot(2,1,2);
plot(n,xn);
xlabel('Time');
title('Reconstructed oversampled fs>>2fm signal');
OUTPUT FOR SAMPING AND EFFECT OF ALIASING:

%Original Analog signal
%Undersampled fs<<2fm signal
%Sampled at Nyquist rate fs=2fm signal
%Oversampled fs>>2fm signal
RESULT:
PROGRAM:
%FIR filter using HAMMING window
clc;
clear all;
close all;
wc=.5*pi;
N=input('Enter the length of sequence = ');
alpha=(N-1)/2;
eps=.001;
n=0:1:N-1;
%low pass filter
hd=sin(wc*(n-alpha+eps))./(pi*(n-alpha+eps));
wh=hamming(N);
hn=hd.*wh';
w=0:.01:pi;
h=freqz(hn,1,w);
figure(1);
plot(w/pi,abs(h),'-');
title('low pass filter');
xlabel('Normalised frequency \omega/\pi');
ylabel('Magnitude');
disp('Low pass filter co-efficient');
disp(hn);
%high pass filter
hd=(sin(pi*(n-alpha+eps))-sin(wc*(n-alpha+eps)))./(pi*(nalpha+eps));
wh=hamming(N);
hn=hd.*wh';
w=0:.01:pi;
h=freqz(hn,1,w);
figure(2);
title('high pass filter');
disp('high pass filter co-efficient');
disp(hn);
%band pass filter
wc1=.25*pi;wc2=.75*pi;
hd=(sin(wc2*(n-alpha+eps))-sin(wc1*(n-alpha+eps)))./(pi*(nalpha+eps));
wh=hamming(N);
hn=hd.*wh';
w=0:.01:pi;
h=freqz(hn,1,w);
figure(3);
title('Band pass filter');
disp('band pass filter co-efficient');
disp(hn);
%band stop filter
hd=(sin(wc1*(n-alpha+eps))-sin(wc2*(n-alpha+eps))+sin(pi*(nalpha+eps)))./(pi*(n-alpha+eps));
wh=hamming(N);
hn=hd.*wh';
w=0:.01:pi;
h=freqz(hn,1,w);
figure(4);
title('Band reject filter');
disp('band reject filter co-efficient');
disp(hn);
OUTPUT FOR FIR DIGITAL FILTER USING HAMMING WINDOW:

Enter the length of the sequence = 7
Low pass filter co-efficient
-0.0155 0.0001 0.2720 0.5000
0.2714 -0.0001 -0.0155
high pass filter co-efficient

0.0156 -0.0004 -0.2711
0.5000 -0.2723
0.0004
0.0155
band pass filter co-efficient

-0.0000 -0.1592 0.0006
0.5000 -0.0006 -0.1591
0.0000
band reject filter co-efficient

0.0001 0.1590 0.0003 0.5000 -0.0002
0.1593 -0.0001
RESULT:
PROGRAM:
%FIR filter using HANNING window
clc;
clear all;
close all;
wc=.5*pi;
alpha=(N-1)/2;
eps=.001;
n=0:1:N-1;
%low pass filter
wh=hanning(N);
hn=hd.*wh';
w=0:.01:pi;
h=freqz(hn,1,w);
figure(1);
disp(hn);
%high pass filter
wh=hanning(N);
hn=hd.*wh';
w=0:.01:pi;
h=freqz(hn,1,w);
figure(2);
disp(hn);
%band pass filter
wc1=.25*pi;wc2=.75*pi;
wh=hanning(N);
hn=hd.*wh';
w=0:.01:pi;
h=freqz(hn,1,w);
figure(3);
disp(hn);
%band stop filter
wh=hanning(N);
hn=hd.*wh';
w=0:.01:pi;
h=freqz(hn,1,w);
figure(4);
disp(hn);
OUTPUT FOR FIR DIGITAL FILTER USING HANNING WINDOW:

Enter the length of sequence = 7
-0.0085 0.0001 0.2453 0.5000 0.2449 -0.0001 -0.0085
0.0085 -0.0002 -0.2446
0.5000 -0.2456
0.0002
0.0085

-0.0000 -0.0987 0.0005
0.5000 -0.0005 -0.0986
0.0000

0.0000 0.0986 0.0002
0.5000 -0.0002
0.0988 -0.0000
RESULT:
PROGRAM:
%FIR filter using BLACKMAN window
clc;
clear all;
close all;
wc=.5*pi;
alpha=(N-1)/2;
eps=.001;
n=0:1:N-1;
%low pass filter
wh=blackman(N);
hn=hd.*wh';
w=0:.01:pi;
h=freqz(hn,1,w);
figure(1);
disp(hn);
%high pass filter
wh=blackman(N);
hn=hd.*wh';
w=0:.01:pi;
h=freqz(hn,1,w);
figure(2);
disp(hn);
%band pass filter
wc1=.25*pi;wc2=.75*pi;
wh=blackman(N);
hn=hd.*wh';
w=0:.01:pi;
h=freqz(hn,1,w);
figure(3);
disp(hn);
%band stop filter
wh=blackman(N);
hn=hd.*wh';
w=0:.01:pi;
h=freqz(hn,1,w);
figure(4);
disp(hn);
OUTPUT FOR FIR DIGITAL FILTER USING BLACKMANN WINDOW:

Enter the length of sequence = 7
0 0.0000 0.2007 0.5000 0.2003 -0.0000
0
0 -0.0001 -0.2001 0.5000 -0.2010
0.0001

0 -0.0414 0.0004 0.5000 -0.0004 -0.0414

0 0.0413 0.0002 0.5000 -0.0002
0.0414
RESULT:
PROGRAM:
%PROGRAM FOR BUTTERWORTH FILTER
clear all;
clc;
close all;
format long
rp=input('Enter the ripple of pass band (rp)= ');
rs=input('Enter the ripple of stop band (rs)= ');
wp=input('Enter the frequency of pass band (fp)= ');
ws=input('Enter the frequency of stop band (fs)= ');
fs=input('Enter the sampling frequency (F)= ');
w1=2*wp/fs;
w2=2*ws/fs;
%LOW PASS FILTER
[n,wn]=buttord(w1,w2,rp,rs);
[b,a]=butter(n,wn);
w=0:0.01:pi;
[h,om]=freqz(b,a,w);
m=20*log10(abs(h));
an=angle(h);
disp('Filter order');
disp(n);
disp('Cut-off frequency');
disp(wn);
figure(1);
Subplot(2,1,1);
plot(om/pi,m);
ylabel('Gain in deciBel');
xlabel('Normalized frequency in rad/sec');

title('Magnitude Plot (LPF)');
figure(1);
Subplot(2,1,2);
plot(om/pi,an);
ylabel('Phase Angle in rad');
title('Phase Plot (LPF)');
%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);
title('Magnitude Plot (HPF)');
figure(2);
subplot(2,1,2);
plot(om/pi,an);
title('Phase Plot (HPF)');
%BAND PASS FILTER
[n]=buttord(w1,w2,rp,rs);
wn=[w1,w2];
[b,a]=butter(n,wn,'bandpass');
w=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
figure(3);
subplot(2,1,1);
plot(om/pi,m);
title('Magnitude Plot (BPF)');
figure(3);
subplot(2,1,2);
plot(om/pi,an);
title('Phase Plot (BPF)');
%BAND STOP FILTER
[n]=buttord(w1,w2,rp,rs);
wn=[w1,w2];
[b,a]=butter(n,wn,'stop');
w=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
figure(4);
subplot(2,1,1);
plot(om/pi,m);
title('Magnitude Plot (BSF)');
figure(4);
subplot(2,1,2);
plot(om/pi,an);
title('Phase Plot (BSF)');
OUTPUT FOR BUTTERWORTH IIR DIGITAL FILTER :

Enter the ripple of pass band (rp)= .5
Enter the ripple of stop band (rs)= 50
Enter the frequency of pass band (fp)= 1200
Enter the frequency of stop band (fs)= 2400
Enter the sampling frequency (F)= 10000
Filter order
8
Cut-off frequency
0.273047748609120
RESULT:
PROGRAM:
%chebyshev filter
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]=cheb1ord(w1,w2,rp,rs);
[b,a]=cheby1(n,rp,wn);
w=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
disp('Filter order');
disp(n);
disp('Cut-off frequency');
disp(wn);
subplot(2,1,1);
plot(om/pi,m);
ylabel('Gain in db--->');
xlabel('(a)Normalized frequency--->');
title('Low pass filter');
subplot(2,1,2);
plot(om/pi,an);
xlabel('(b)Normalized frequency--->');
ylabel('Phase in radians--->');
%HIGH PASS FILTER
[n,wn]= cheb1ord(w1,w2,rp,rs);
[b,a]=cheby1(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);
title('High pass filter');
figure(2);
subplot(2,1,2);
plot(om/pi,an);
%BAND PASS FILTER
[n]=cheb1ord(w1,w2,rp,rs);
wn=[w1,w2];
[b,a]=cheby1(n,rs,wn,'bandpass');
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]=cheb1ord(w1,w2,rp,rs);
wn=[w1,w2];
[b,a]=cheby1(n,rp,wn,'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);
title('Bandstop filter');
figure(4);
subplot(2,1,2);
plot(om/pi,an);
OUTPUT FOR CHEBYSHEV IIR DIGITAL FILTER:

Enter the pass band ripple:0.02
Enter the stop band ripple:.03
Enter the pass band frequency:1200
Enter the stop band frequency:2400
Enter the sampling frequency:8000
Filter order
1
Cut-off frequency
0.300000000000000
RESULT:
PROGRAM:
% CALCULATION OF FFT AND IFFT
clc;
clear all;
close all;
N=input('Enter the length of sequence(N)= ');
x=input('Enter the input sequence(x)= ');
n=0:1:(N-1);
subplot(2,2,1);
stem(n,x,'k');
xlabel('N--->');
title('Input sequence');
k=0:1:(N-1);
X=fft(x,N);
subplot(2,2,2);
stem(k,abs(X),'k');
xlabel('k--->');
title('Magnitude Plot(FFT)');
subplot(2,2,3);
stem(k,angle(X),'k');
xlabel('k--->');
ylabel('Angle');
title('Phase Plot(FFT)');
n=0:1:(N-1);
x=ifft(X,N);
subplot(2,2,4);
stem(n,abs(x),'k');
xlabel('n--->');
title('Magnitude Plot');
OUTPUT FOR FFT AND IFFT:

Enter the length of sequence(N)= 8
Enter the input sequence(x)= [1 2 3 4 5 6 7 8]
RESULT:
PROGRAM:
%program for decimation
clc;
close all;
clear all;
N=input('Enter the length of the samples(N)=');
t1=input('Enter the first time period(t1)=');
t2=input('Enter the second time period(t2)=');
D=input('Enter the Down sampling factor(D)=');
n=0:N-1;
X=sin(2*pi*n*t1) + sin(2*pi*n*t2);
Y=decimate(X,D);
subplot(2,1,1);
stem(n,X(1:N),'k');
title('Sample Signal');
xlabel('sample');
d=0:(N/D)-1;
subplot(2,1,2);
stem(d,X(1:N/D),'k');
title('Down sampled signal');
xlabel('Sample');
OUTPUT FOR DECIMATION BY POLYPHASE DECOMPOSTION:

Enter the length of the samples(N)=100
Enter the first time period(t1)=.03
Enter the second time period(t2)=.04
Enter the Down sampling factor(D)=3
RESULT:
PROGRAM:
%program for DFT & IDFT
clc;
x1=input('Enter the input sequence ');
N=length(x1);
n=0:1:N-1;
k=0:1:N-1;
WN=exp(-1i*2*pi/N);
nk=n'*k;
WnNK=WN.^nk;
Xk=x1*WnNK;
Xkmag=abs(Xk);
WNnk=WN.^(-nk);
xn=Xk*WNnk/N;
subplot(3,1,1);
stem(n,x1);
xlabel('time period');
title('input sequence');
subplot(3,1,2);
stem(n,Xkmag);
title('DFT');
subplot(3,1,3);
stem(n,xn);
title('IDFT');
OUTPUT FOR DFT & IDFT:

Enter the input sequence [1 1 0 0]
RESULT:
PROGRAM:
%Impluse invariant transformation
clc;
close all;
clear all;
b=input('Enter the numerator co-efficient of analog filter = ');
a=input('Enter the denominator co-efficient of analog filter = ');
fs=input('Sampling frequency = ');
[bz,az]=impinvar(b,a,fs);
disp('bz = ');
disp(bz);
disp('az = ');
disp(az);
Hz=tf(bz,az)
%Bilnear transformation transformation
b=input('Enter the numerator co-efficient of analog filter = ');

a=input('Enter the denominator co-efficient of analog filter = ');
fs=input('Sampling frequency = ');
[bz,az]=bilinear(b,a,fs);
disp('bz = ');
disp(bz);
disp('az = ');
disp(az);
Hz=tf(bz,az)
OUTPUT FOR IMPULSE INVARIANT AND BILINEAR

TRANFORMATION:
Enter the numerator co-efficient of analog filter = [1 2]
Enter the denominator co-efficient of analog filter = [1 5 11 15]
Sampling frequency = 5
bz =
0.0000 0.0290 -0.0195
az =
1.0000 -2.0570
1.4980 -0.3679
Transfer function:
5.551e-018 s^2 + 0.02904 s - 0.01953
-----------------------------------s^3 - 2.057 s^2 + 1.498 s - 0.3679
Enter the numerator co-efficient of analog filter = [ 2 ]
Enter the denominator co-efficient of analog filter = [1 3 2]
Sampling frequency = 1
bz =
0.1667 0.3333 0.1667
az =
1.0000 -0.3333
0.0000
Transfer function:
0.1667 s^2 + 0.3333 s + 0.1667
-----------------------------s^2 - 0.3333 s + 7.401e-017
RESULT:

EC2306 DSP LAB RECORD - Final PDF

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DEPARTMENT OF ELECTRONICS AND

LAB CODE/NAME: EC2306 / DIGITAL SIGNAL PROCESSING LAB

Third Year / Fifth Semester

SRINIVASA INSTITUTE OF ENGINEERING & TECHNOLOGY

Certified that this is a bonafide record of the work done by

Third Year (Fifth Semester) of

Electronics and Communication Engineering Department in the DIGITAL

Signature of the Staff

Submitted for the Practical Examination held on ---------------------------------------

Name Of The Experiment

Study of Matlab functions

Linear and circular convolution

Sampling and effect of aliasing

Design of FIR filter using hamming window

Design of FIR filter using hanning window

Design of FIR filter using blackmann window

Design of butterworth IIR filter

Design of chebyshev IIR filter

Calculation of FFT and IFFT of a signal

Decimation by polyphase decomposition

Computation of DFT and IDFT

Impulse invariant and Bilinear transformation

STUDY OF MATLAB FUNCTIONS

h = impz (b,a,w): It computes N samples of the impules response of

[B,A] = cheby1 {N,R,Wn}: Design an Nth order low pass digital

% program for Unit Step Sequence

% program for Unit ramp Sequence

OUTPUT FOR GENERATION OF CONTINUOUS TIME SIGNAL:

OUTPUT FOR GENERATION OF DISCRETE TIME SIGNAL:

% PROGRAM FOR CIRCULAR CONVOLUTION

OUTPUT FOR LINEAR CONVOLUTION:

OUTPUT FOR CIRCULAR CONVOLUTION:

%Analog sig sampled at fs=2fm

%Analog sig sampled at fs>>2fm

OUTPUT FOR SAMPING AND EFFECT OF ALIASING:

%Undersampled fs<<2fm signal

%Sampled at Nyquist rate fs=2fm signal

%Oversampled fs>>2fm signal

OUTPUT FOR FIR DIGITAL FILTER USING HAMMING WINDOW:

0.2714 -0.0001 -0.0155

high pass filter co-efficient

band pass filter co-efficient

0.5000 -0.0006 -0.1591

band reject filter co-efficient

OUTPUT FOR FIR DIGITAL FILTER USING HANNING WINDOW:

band pass filter co-efficient

0.5000 -0.0005 -0.0986

band reject filter co-efficient

OUTPUT FOR FIR DIGITAL FILTER USING BLACKMANN WINDOW:

band pass filter co-efficient

band reject filter co-efficient

xlabel('Normalized frequency in rad/sec');

OUTPUT FOR BUTTERWORTH IIR DIGITAL FILTER :

OUTPUT FOR CHEBYSHEV IIR DIGITAL FILTER:

OUTPUT FOR FFT AND IFFT:

OUTPUT FOR DECIMATION BY POLYPHASE DECOMPOSTION:

OUTPUT FOR DFT & IDFT:

b=input('Enter the numerator co-efficient of analog filter = ');

OUTPUT FOR IMPULSE INVARIANT AND BILINEAR

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