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SECTION NO.

1 (Software)
EXPERIMENT NO. 1:
Signals in Matlab (Continuous time & Discrete time)
EXPERIMENT NO. 2:
Discrete Time Systems and its Properties
1) Linearity
2) Time invariant
EXPERIMENT NO. 3:
Impulse Response of a system
1) IIR
2) FIR
EXPERIMENT NO. 4:
DTFT and Properties
EXPERIMENT NO. 5:
Sampling, A/D Conversion and D/A Conversion
EXPERIMENT NO. 6:
System Frequency Transform
Introduction to Digital Filter
EXPERIMENT NO. 7:
Simple Digital Filter
1) IIR
2) FIR
EXPERIMENT NO. 8:
Digital Filter Design
1 FIR
2 Introduction to Widows method
EXPERIMENT NO. 9:
FIR Filter Design using Rectangular Windows method
1 LPF
2 HPF
3 BPF
4 BSF
EXPERIMENT NO. 10:
DIGITAL FILTER STRUCTURE
EXPERIMENT NO. 11:
Design of FIR filters using Matlab commands .
EXPERIMENT NO. 12:
Design of IIR filters using Matlab commands
EXPERIMENT NO. 13:
DFT
EXPERIMENT NO. 14:
Interpolation & Decimation

EXPERIMENT NO. 15:

Filter designing by Matlab tools


EXPERIMENT NO. 16:

Design an IIR filter to suppress frequencies of 5 Hz and 30 Hz from given signal

EXPERIMENT NO. 1
SIGNALS IN MATLAB (CONTINUOUS TIME &
DISCRETE TIME)

Signals are broadly classified into two classifications:


a. Continuous Time Signals
b. Discrete Time Signals

A continuous time signal will be denoted by x(t), in which the variable t can
represent any physical quantity.
A discrete time signal will be denoted x[n], in which the variable n is integer value.
In this lab we will learn to represent and operate on signals in MATLAB.

1. Continuous Time Signals

For the following: Run the following lines and explain why the plots are
different.
Provide the snapshots of the plots for each step given below.
close all,
clear all
t = 0:2*pi;
plot(t,sin(t))
figure
t = 0:0.2:2*pi;
plot(t,sin(t))
figuret = 0:0.02:2*pi;
plot(t,sin(t))

For the last graph, add a title and axis labels with:
title('Continuous time signal plot')
xlabel('t (Seconds)')
ylabel('y(t)')

Change the axis with:


axis([0 2*pi -1.2 1.2])

Put two plots on the same axis


t = 0:0.2:2*pi;
plot(t,sin(t),t,sin(2*t))

Produce a plot without connecting the points


t = 0:0.2:2*pi;
plot(t,sin(t),'.')

Try the following command


t = 0:0.2:2*pi;
plot(t,sin(t),t,sin(t),'g.')

Question 1: What does g do?

2. Discrete Time Signals

Use stem to plot the discrete time signals. Provide the snapshots of the step
below.
close all
clear all
n = -10: 10;
f = n >= 0;
stem(n,f)

3. Elementary sequences in digital signal processing


For each part below, provide an example using any input and also provide the
plots of input and output sequences using subplot.
a. Unit sample sequence

( nn 0 )=

1,n=n0
0, n n0

% Generate a vector from -10 to 20


n = -10:20;
n0=0;
% Generate the unit sample sequence
u =n-n0==0;
% Plot the unit sample sequence
stem(n,u);
xlabel('Time index n');
ylabel('Amplitude');
title('Unit Sample Sequence');
axis([-10 20 0 2]);

b. Unit step sequence


u(nn0 )= 1, n n 0
0,n< n0

n=-10:30
n0=0;
y=n-n0>=0;
figure
stem(n,y);
title('UNIT STEP FUNCTION');

c. Real Valued Exponential sequence

x ( n )=an , a R

n = 0:35;
a = 1.2; K = 0.2;
x = K*a.^+n;
stem(n,x);
xlabel('Time index n');
ylabel('Amplitude');

d. Complex valued exponential sequence

x ( n )=e (

+ j 0) n

c = -(1/12)+(pi/6)*i;
K = 2;
n = 0:40;
x = K*exp(c*n);
subplot(2,1,1);
stem(n,real(x));
xlabel('Time index n');
ylabel('Amplitude');
title('Real part');
subplot(2,1,2);
stem(n,imag(x));
xlabel('Time index n');
ylabel('Amplitude');
title('Imaginary part');

e. Sinusoidal sequence
n = 0:40;
f = 0.1;
phase = 0;
A = 1.5;
arg = 2*pi*f*n - phase;
x = A*cos(arg);
stem(n,x); % Plot the generated sequence
axis([0 40 -2 2]);
grid;
title('Sinusoidal Sequence');
xlabel('Time index n');
ylabel('Amplitude');
axis;

f. Operations on sequence:

Shifting

In this operation each sample of x(n) is shifted by an amount k to


obtain a shifted sequence y(n)

y ( n )= { x (nk ) }
If we let m = n-k, then n = m+k the above operation is given by
y ( m+ k )= { x (m) }

For this we can use the following code


s=1:100;
subplot(2,1,1)
stem(s)
title('Sequence');
axis([0 100 0 100]);
s_new=[zeros(1,10) s];
subplot(2,1,2)
stem(s_new);
axis([0 100 0 100]);
title('Delayed sequence');

Folding
In this operation each sample of x(n) is flipped around its axis to obtain a
folded sequence y(n)

y ( n )= { x (n) }

For this the following code is shown.


Exercise:
Generate and plot each of the following sequences over the indicated
interval. Provide the scripts used to generate the plots.
Ex.1:
a.

z ( n )=2 ( n+2 ) ( n4 ) ,5 5

b.

x (n)=n [u(n)u(n10)]+10 e

(0.3(n10))

[u(n10)u (n20)], 0 n 20

Ex.2:
Let x(n) = {1,2,3,4,5,6,7,6,5,4,3,2,1}, Determine and plot the following
sequences.

a.

x 1 ( n ) =2 x ( n5 )3 x ( n+ 4 )

b.

x 2 ( n ) =x ( 3n ) + x (n) x (n2)

EXPERIMENT NO. 2
(Discrete Time Systems)

DSP LAB # 3: Discrete Time Systems


To provide an overview of discrete time signals and systems on MATLAB, to analyze
various properties of discrete time systems and verify them on MATLAB.
Introduction:
Mathematically, a discrete-time system is described as an operator T[.] that takes a
sequencex(n) called excitation and transforms it into another sequence y(n) (called
response). Discretetime systems can be classified into two categories i) LTI systems
ii) NON-LTI systems. Adiscrete system T[.] is a linear operator L[.] if and only if L[.]
satisfies the principle ofsuperposition, namely

L [ a1 x 1 ( n ) +a2 x2 ( n ) ] =a1 L [ x 1 ( n ) ]+ a1 L [x 2 ( n ) ]
A discrete system is time-invariant if shifting the input only causes the same shift in
the output.
A system is said to be bounded-input bounded-output (BIBO) stable if every
bounded input produces a bounded output.

|x (n)|< | y(n)|< , x , y
An LTI system is BIBO stable if and only if its impulse response is absolutely sum
able.

BIBO Stability |h(n)|<


A system is said to be causal if the output at any instant depends only on the
present & past values only.
An LTI system is causal if and only if the impulse response is

h ( n )=0,n< 0

1. Linearity and Non-Linearity:


We now investigate the linearity property of a causal system of described by
the following equation.

y [ n ] 0.4 y [ n1 ] +0.75 y [ n2 ] =2.2 x [ n ] +2.3 x [ n1 ] + 2.4 x [n2]

Following program simulates the above mentioned equation.

clear all, close all


n = 0:40;
a = 2; b = -3;
x1 = cos(2*pi*0.1*n);
x2 = cos(2*pi*0.4*n);
x = a*x1 + b*x2;
num = [2.2 2.3 2.4];
den = [1 -0.4 0.75];
ic = [0 0]; % Set zero initial conditions
y1 = filter(num,den,x1,ic); % Compute the output y1[n]
y2 = filter(num,den,x2,ic); % Compute the output y2[n]
y = filter(num,den,x,ic); % Compute the output y[n]
yt = a*y1 + b*y2;
d = y - yt; % Compute the difference output d[n]
% Plot the outputs and the difference signal
subplot (3,1,1)
stem(n ,y);
ylabel('Amplitude');
title('Output Due to Weighted Input');
subplot(3,1,2)
stem(n,yt);
ylabel('Amplitude');
title('Weighted Output');
subplot(3,1,3)
stem(n,d);
xlabel('Time index n');
ylabel('Amplitude');
Question 1:
Run above program
and compare y[n] obtained with weighted input
title('Difference
Signal');

with y[n] obtained by combining the two outputs y1[n] and y2[n] with the same
weights. Are these two sequences equal? Is this system linear?
Exercise 1: Consider another system described by

y [ n ] y [ n2 ] +0.75 y [ n4 ] =2.2 x [ n ] +2.3 x [ n3 ] + 2.4 x [n5]


y [ n ] 0 y [ n2 ] +1 y [ n4 ] =x [ n ] +0 x [ n3 ] + x [n4]
y [ n ] y [ n2 ] +0.75 y [ n4 ] + 3 y [n5]=x [ n ] +0.3 x [ n1 ] + 2.4 x [n3]

Modify given program to compute the output sequences y1[n], y2[n], and y[n] of
the above system.
Compare y[n] with y[n]. Are these two sequences equal? Is this system linear?

2. Time-Invariant and Time-Varying Systems:

We next investigate the time-invariance property. Following program


simulates following difference equation

y [ n ] 0.4 y [ n1 ] +0.75 y [ n2 ] =2.2 x [ n ] +2.3 x [ n1 ] + 2.4 x [n2]


Two input sequence x[n] and x[n-D], are generated and corresponding output
sequences
y1[n],y2[n]
areall
plotted.
close
all, clear
n = 0:40; D = 10;a = 3.0;b = -2;
x = a*cos(2*pi*0.1*n) + b*cos(2*pi*0.4*n);
xd = [zeros(1,D) x];
num = [2.2 2.3 2.4];
den = [1 -0.4 0.75];
ic = [0 0];% Set initial conditions
% Compute the output y[n]
y = filter(num,den,x,ic);
% Compute the output yd[n]
yd = filter(num,den,xd,ic);
% Compute the difference output d[n]
d = y - yd(1+D:41+D);
% Plot the outputs
subplot(3,1,1)
stem(n,y);
ylabel('mplitude');
title('Output y[n]');grid;
subplot(3,1,2)
stem(n,yd(1:41));
ylabel('Amplitude');
title(['Output Due to Delayed Input x[n , num2str(D),]']);grid;
subplot(3,1,3)
stem(n,d);
xlabel('Time index n'); ylabel('Amplitude');
title('Difference Signal');grid;

Exercise 2: Consider another system described by:

y [ n ] y [ n1 ] + y [ n2 ] + 0.01 y [n4 ]=x [ n ] +2.3 x [ n1 ] +2.4 x [n5]


y [ n ] y [ n2 ] + x [ n ] +0.02 x [ n2 ] + x [ n4 ]
y [ n ] =x [ n ] +2.3 x [ n3 ] +2.4 x [n5]

Modify program to simulate the above system and determine whether this
system is time-invariant or not.

3. Linearity and Time-Invariant: (LOOPS)


Using loops to solve the given equation
y [ n ] =x [ n ] + x [n1]

x1=[1 1 2];
x2=[2 2 0];
%for y1
for i=1:3
if(i==1)
y1(i)=x1(i);
else if(i>1)
y1(i)=x1(i)+x1(i-1);
end
end
end
%for y2
for i=1:3
if(i==1)
y2(i)=x2(i);
else if(i>1)
y2(i)=x2(i)+x2(i-1);
end
end
end
x3=x1+x2;
%for y3
for i=1:3
if(i==1)
y3(i)=x3(i);

else if(i>1)
y3(i)=x3(i)+x3(i-1);
end
end
end
y3
y1+y2
if(y3==(y1+y2))
display('System is linear')
end

Exercise 1: Consider another system described by:

y [ n ] + y [n1]=x [ n ] + x [n3]
y [ n ] =x [ n ] +nx [n1]
Modify program to simulate the above system and determine whether this
system is time-invariant or not.

x1=[1 1 2];
%for y1
for i=1:3
if(i==1)
y1(i)=x1(i);
else if(i>1)
y1(i)=x1(i)+x1(i-1);
end
end
end
y1
x2=[0 0 x1]
%for y2
for i=1:5
if(i==1)
y3(i)=x2(i);
else if(i>1)
y3(i)=x2(i)+x2(i-1);
end
end
end
y3
if(y3(3:5)==y1)
display('System is Time invariant')
end

EXPERIMENT NO. 3
Impulse Response of
System

Impulse Response Computation:


Response of a system to an impulse is called impulse response, two types
FIR( Finite impulse response)
IIR (infinite impulse response)
FIR:
In signal processing, a finite impulse response (FIR) filter is a filter whose
impulse response (or response to any finite length input) is of finite duration,
because it settles to zero in finite time.
For a discrete-time FIR filter, the output is a weighted sum of the current and a
finite number of previous values of the input. The operation is described by the
following equation, which defines the output sequence y[n] in terms of its input
sequence x[n]:
y[n]=b0x(n)+ b1x(n-1)+ b2x(n-2) .+ bNx(n-N)
y [n]=bi x(n-i)
where:

is the input signal,

is the output signal,

are the filter coefficients, also known as tap weights, that make up the
impulse response,
is the filter order; an th-order filter has
terms on the right-hand side.
The
in these terms are commonly referred to as taps, based on the
structure of a tapped delay line that in many implementations or block
diagrams provides the delayed inputs to the multiplication operations. One may
speak of a 5th order/6-tap filter, for instance.

Properties:
An FIR filter has a number of useful properties which sometimes make it
preferable to an infinite impulse response (IIR) filter. FIR filters:
Require no feedback. This means that any rounding errors are not
compounded by summed iterations. The same relative error occurs in each
calculation. This also makes implementation simpler.

Are inherently stable. This is due to the fact that, because there is no
required feedback, all the poles are located at the origin and thus are located
within the unit circle (the required condition for stability in a Z transformed
system).
They can easily be designed to be linear phase by making the coefficient
sequence symmetric; linear phase, or phase change proportional to
frequency, corresponds to equal delay at all frequencies. This property is
sometimes desired for phase-sensitive applications, for example data
communications, crossover filters, and mastering.

Impulse response:
The impulse response
can be calculated if we set
in the above
relation, where
is impulse. The impulse response for an FIR filter then
becomes the set of coefficients , as follows

for
to .
FIR filters are clearly bounded-input bounded-output (BIBO) stable, since the
output is a sum of a finite number of finite multiples of the input values, so can
be no greater than

times the largest value appearing in the input.

y [ n ] =2.2 x [ n ] + x [ n1 ] + x [n2]
% Compute the impulse response y
close all, clear all
N = 40;
num = [2.2 1 1];
den = [1];
y = impz(num,den,N);
% Plot the impulse response
stem(y);
xlabel('Time index n'); ylabel('Amplitude');
title('Impulse Response'); grid;

Exercise 1: Compute Impulse Response

y [ n ] =2.2 x [ n ] + x [ n1 ] + x [ n2 ] +0.1 x [n6]


IIR:
IIR systems have an impulse response function that is non-zero over an infinite
length of time. This is in contrast to finite impulse response (FIR) filters, which
have fixed-duration impulse responses. The simplest analog IIR filter is an RC
filter made up of a single resistor (R) feeding into a node shared with a single
capacitor (C).
Digital filters are often described and implemented in terms of the difference
equation that defines how the output signal is related to the input signal:

where:
Is the feed forward filter order
are the feed forward filter coefficients
is the feedback filter order
are the feedback filter coefficients
Is the input signal
Is the output signal.
A more condensed form of the difference equation is:

which, when rearranged, becomes:

To find the transfer function of the filter, we first take the Z-transform of each
side of the above equation, where we use the time-shift property to obtain:

We define the transfer function to be:

Considering that in most IIR filter designs coefficient


function takes the more traditional form:

is 1, the IIR filter transfer

Properties:
The main advantage IIR filters have over FIR filters is that through recursion
they use fewer taps. Therefore in digital signal processing applications IIR filters
use fewer computing resources than an equivalent FIR filter. A disadvantage of
IIR filters is they can be unstable and subject to limit cycle behavior.

y [ n ] 2 y [n1]=2.2 x [ n ] + x [ n1 ] + x [n2]
% Compute the impulse response y
close all, clear all
N = 40;
num = [2.2 1 1];
den = [1 2];
y = impz(num,den,N);
% Plot the impulse response
stem(y);
xlabel('Time index n'); ylabel('Amplitude');
title('Impulse Response'); grid;

Know For N=100

Following equation computes impulse response of following difference equation

y [ n ] 0.4 y [ n1 ] +0.75 y [ n2 ] =2.2 x [ n ] +2.3 x [ n1 ] + 2.4 x [n2]


% Compute the impulse response y
close all, clear all
N = 40;
num = [2.2 2.3 2.4];
den = [1 -0.4 0.75];
y = impz(num,den,N);
% Plot the impulse response
stem(y);
xlabel('Time index n'); ylabel('Amplitude');
title('Impulse Response'); grid;

Exercise 3: Write MATLAB program to generate and plot the step response of
a causal LTI system.
Exercise 2:

y [ n ] 0.4 y [ n1 ] +0.75 y [ n2 ] =2.2 x [ n ] +2.3 x [ n1 ] + 2.4 x [n2]


Using this program compute and plot the first 40 samples of the step
response above mentioned LTI system.

EXPERIMENT NO. 4
Z-Transform

Study of Z-Transform in MATLAB:


THEORY:

Z-Transform technique is an important tool in the analysis of characterization of


discrete time signals and LTI systems; Z-Transform gives the response of various
signals by its pole zero locations.
Z-Transform is an important tool in DSP that gives the solution of difference
equation in one go.
The Z-Transform of a discrete time system x (n) is defined as power series;

And the inverse Z-Transform is denoted by;

Since, Z-Transform is the infinite power series; it exists only for the region for
which this series converges (region of convergence). Inverse Z-Transform is the
method for inverting the Z-Transform of a signal so as to obtain the time domain
representation of signal. The features of Z-Transform which are explained are as
fellows;
Z-Transform of a Discrete time function
EXAMPLE:

MATLAB CODE:
% Z transform
close all, clear all

b=[1 -1.6 180 1];


a=[1 -1.5 161 0.878];
A=roots(a)
B=roots(b)
zplane(b,a)
A=
0.7527 +12.6666i
0.7527 -12.6666i
-0.0055
B=
0.8028 +13.3927i
0.8028 -13.3927i
-0.0056

Z-TRRANSFORM OF A DISCRETE TIME FUNCTION:


Z-transform is defined as
X(z)

Or
X(z)=Z(x(n))
Let the function equation be

u [ n ] =(1 /4)n

% Z-transform
close all, clear all
syms z n
u =1/(4^n);
U=ztrans(u)
U=
4*z/(4*z-1)

x [ n ]=sin (an)
% Z-transform
close all, clear all
Exercise 1: syms z n a
x=sin(a*n)
Consider that
the equation is
X=ztrans(x)
X=
4 z*sin(a)/(z^2-2*z*cos(a)+1)
x [ n ]=n

Find Z-transform

INVERSE Z-TRANSFORM:

The inverse Z-Transform is denoted by,

Let the Z-domain is:

% Z-transform
close all, clear all
syms z n
X=2*z/(2*z-1)
x=iztrans(X)
x=
(1/2)^n

EXPERIMENT NO. 5
Fourier Transform and
Properties

Discrete-Time Fourier Transform:

The discrete-time Fourier transform (DTFT) X(ej ) of a sequence x[n] is a


continuous function of . Since the data in MATLAB is in vector form, X(ej )
can only be evaluated at a prescribed set of discrete frequencies. Moreover,
only a class of the DTFT that is expressed as a rational function in ej in the
form
j

X ( e j )=

j2

jM

p0 + p1 e + p2 e
++ p M e
j
j2
jN
d 0+ d 1 e +d 2 e
++ d N e

(5.1)
can be evaluated. In the following two projects you will learn how to evaluate
and plot the DTFT and study certain properties of the DTFT using MATLAB.

DTFT Computation
The DTFT X(ej ) of a sequence x[n] of the form of Eq. (5.1) can be computed
easily at a prescribed set of L discrete frequency points = using the
MATLAB function freqz. Since X(ej ) is a continuous function of , it is
necessary to make L as large as possible so that the plot generated using the
command plot provides a reasonable replica of the actual plot of the DTFT. In
MATLAB, freqz computes the L-point DFT of the sequences {p0 p1 . . . PM } and
{d0 d1 . . . dM }, and then forms their ratio to arrive at X(ejl ), l = 1, 2, . . . , L.
For faster computation, L should be chosen as a power of 2, such as 256 or 512.

Program P5_1 can be used to evaluate and plot the DTFT of the form of Eq.
(5.1).
% Program P5_1
% Evaluation of the DTFT
clear all; close all; clc
% Compute the frequency samples of the DTFT
w = -4*pi:8*pi/511:4*pi;
num = [2 1];den = [1 -0.6];
h = freqz(num, den, w);
% Plot the DTFT
subplot(2,1,1)
plot(w/pi,real(h));grid
title('Real part of H(e^{j\omega})')
xlabel('\omega /\pi');
ylabel('Amplitude');
subplot(2,1,2)
% plot the imaginary part in the same way your self and give it title
pause
subplot(2,1,1)
% In this subplot plot magnitude of FFT your self
subplot(2,1,2)
% In this subplot plot magnitude of FFT your self

DTFT Properties
Most of the properties of the DTFT can be verified using MATLAB. Since all data
in MATLAB have to be finite-length vectors, the sequences being used to verify
the properties are thus restricted to be of finite length.
Program P5_2 can be used to verify the time-shifting property of the DTFT.
% Program P5_2
% Time-Shifting Properties of DTFT
close all; clear all; clc
w = -pi:2*pi/255:pi; wo = 0.4*pi;D=10;
num=[1 2 3 4 5 6 7 8 9];
h1 = freqz(num, 1, w);
h2 = freqz([zeros(1,D) num], 1, w);
subplot(2,2,1)
plot(w/pi,abs(h1));grid
title('Magnitude Spectrum of Original Sequence')
subplot(2,2,2)
plot(w/pi,abs(h2));grid
title('Magnitude Spectrum of Time-Shifted Sequence')
subplot(2,2,3)
plot(w/pi,angle(h1));grid
title('Phase Spectrum of Original Sequence')
subplot(2,2,4)
plot(w/pi,angle(h2));grid
title('Phase Spectrum of Time-Shifted Sequence')

Magnitude Spectrum of Original Sequence


Magnitude Spectrum of Time-Shifted Sequence
60
60
40

40

20

20

0
-1

-0.5

0.5

0
-1

-0.5

0.5

Phase Spectrum of Original Sequence Phase Spectrum of Time-Shifted Sequence


4
4
2

-2

-2

-4
-1

-0.5

0.5

-4
-1

-0.5

0.5

Program P5_3 can be used to verify the frequency-shifting property of the DTFT.
% Program P5_3
% Frequency-Shifting Properties of DTFT
clear all; close all; clc
w = -pi:2*pi/255:pi; wo = 0.4*pi;
num1=[13579111315 17];
L = length(num1);
h1 = freqz(num1, 1, w);
n = 0:L-1;
num2 = exp(wo*i*n).*num1;
h2 = freqz(num2, 1, w);
subplot(2,2,1)
plot(w/pi,abs(h1));grid
title('Magnitude Spectrum of Original Sequence')
subplot(2,2,2)
plot(w/pi,abs(h2));grid
title('Magnitude Spectrum of Frequency-Shifted Sequence')
subplot(2,2,3)
plot(w/pi,angle(h1));grid
title('Phase Spectrum of Original Sequence')
subplot(2,2,4)
plot(w/pi,angle(h2));grid
title('Phase Spectrum of Frequency-Shifted Sequence')

10

1.3579

x 10

10

Magnitude Spectrum of Original Sequence

1.3579

1.3579

1.3579

1.3579

1.3579

1.3579

1.3579

1.3579

1.3579

1.3579

1.3579

1.3579

1.3579

1.3579

1.3579

1.3579
-1

-0.8

-0.6

-9

1.5

-0.4

-0.2

0.2

0.4

0.6

0.8

1.3579
-1

1.5

0.5

0.5

-0.5

-0.5

-1

-1

-1.5
-1

-0.8

-0.6

-0.4

-0.2

0.2

0.4

-0.8

-9

Phase Spectrum of Original Sequence

x 10

x 10

0.6

0.8

x 10

-1.5
-1

-0.8

Magnitude Spectrum of Frequency-Shifted Sequence

-0.6

-0.4

-0.2

0.2

0.4

0.6

0.8

0.8

Phase Spectrum of Frequency-Shifted Sequence

-0.6

-0.4

-0.2

0.2

Program P5_4 can be used to verify the convolution property of the DTFT.
% Program P5_4
% Convolution Property of DTFT
close all; clear all; clc
w = -pi:2*pi/255:pi;
x1=[1 3 5 7 9 11 13 15 17];
x2=[1 -23 -21];
% convolve x1 and x2 yourself
h1 = freqz(x1, 1, w);
% compute freqz of x2 at w points and save it in h2 yourself
hp = h1.*h2;
%
P5_5
h3Program
= freqz(y,1,w);
%
Modulation Property of DTFT
subplot(2,2,1)
close
all; clear all; clc
plot(w/pi,abs(hp));grid
w
=
-pi:2*pi/255:pi;
title('Product of Magnitude Spectra')
x1=[1
3 5 7 9 11 13 15 17];
subplot(2,2,2)
x2=[1
-1 1 -1 1 -1 1 -1 1];
plot(w/pi,abs(h3));grid
y
=
x1.*x2;
title('Magnitude Spectrum of Convolved Sequence')
h1
= freqz(x1, 1, w);
subplot(2,2,3)
h2
= freqz(x2, 1, w);
plot(w/pi,angle(hp));grid
Program
P5_5 can be used to verify the modulation property of the DTFT.
% similarly compute freqz of y and save it in h3 yourself
subplot(3,1,1)
plot(w/pi,abs(h1));grid
title('Magnitude Spectrum of First Sequence')
subplot(3,1,2)
plot(w/pi,abs(h2));grid

0.4

0.6

Program P5_6 can be used to verify the time-reversal property of the DTFT.
% Program P5_6
% Time-Reversal Property of DTFT
close all; clear all; clc
w = -pi:2*pi/255:pi;
num=[1 2 3 4];
L = length(num)-1;
h1 = freqz(num, 1, w);
h2 = freqz(fliplr(num), 1, w);
h3 = exp(w*L*1i).*h2;
subplot(2,2,1)
plot(w/pi,abs(h1));grid
title('Magnitude Spectrum of Original Sequence')
subplot(2,2,2)
plot(w/pi,abs(h3));grid
title('Magnitude Spectrum of Time-Reversed Sequence')
subplot(2,2,3)
plot(w/pi,angle(h1));grid
title('Phase Spectrum of Original Sequence')
subplot(2,2,4)
plot(w/pi,angle(h3));grid
title('Phase Spectrum of Time-Reversed Sequence')

Magnitude Spectrum of Original Sequence Magnitude Spectrum of Time-Reversed Sequence


10
10
8

2
-1

-0.5

0.5

Phase Spectrum of Original Sequence


4

2
-1

-2

-2
-0.5

0.5

0.5

Phase Spectrum of Time-Reversed Sequence


4

-4
-1

-0.5

-4
-1

-0.5

0.5

EXPERIMENT NO. 6

Sampling, A/D Conversion


and D/A Conversion

The Sampling Process in the Time Domain:


The purpose of this section is to study the relation in the time domain between
a continuous-time signal xa (t) and the discrete-time signal x[1] generated by a
periodic sampling of xa (t).
1. Sampling of a Sinusoidal Signal
% Program 6_1

In this
project you will
investigate
sampling of a continuous-time sinusoidal
% Illustration
of the
Samplingthe
Process
signal
xa
(t)
at
various
sampling
rates.
Since MATLAB cannot strictly generate a
% in the Time Domain
clear all; signal,
close all;
continuous-time
you willclc;
generate a sequence {xa (nTH )} from xa (t) by
t = 0:0.0005:1;
sampling it at a very high rate, 1/TH , such that the samples are very close to
f = 13;
each
A plot of xa (nTH ) using the plot command will then look like a
xa other.
= cos(2*pi*f*t);
subplot(2,1,1)
continuous-time
signal.
plot(t,xa,'LineWidth',1.5);
xlabel('Time, msec');ylabel('Amplitude');
title('Continuous-time signal x_{a}(t)');
axis([0 1 -1.2 1.2])
subplot(2,1,2);
T = 0.1;
n = 0:T:1;
xs = cos(2*pi*f*n);
k = 0:length(n)-1;
stem(k,xs,'r');
xlabel('Time index n');ylabel('Amplitude');
title('Discrete-time signal x[n]');
axis([0 (length(n)-1) -1.2 1.2])

2. Aliasing Effect in the Time Domain


In this project you will generate a continuous-time equivalent ya (t) of the
discrete-time signal x[1] generated in Program P4_1 to investigate the relation
between the frequency of the sinusoidal signal xa (t) and the sampling period.
To generate the reconstructed signal ya (t) from x[1], we pass x[1] through an
ideal low pass filter that in turn can be implemented according to Eq. (4.1). If
Eq. (4.1) is computed at closely spaced values of t, a plot of ya (t) will resemble
a continuous-time signal. In order to implement this equation on MATLAB, the
summation in Eq. (4.1) needs to be replaced with a finite sum, and hence we
can generate only an approximation to the desired reconstructed continuoustime signal ya (t).

1
hr ( t )=
H r ( j ) e jt dt

2
c

sin (c t)
T

e jt d=
, t

2
T t /2
c

(4.1)
% Program P6_2
% Illustration of Aliasing Effect in the Time Domain
clear all;
close all;
clc;
T = 0.1;
f = 13;
n = (0:T:1)';
xs = cos(2*pi*f*n);
t = linspace(-0.5,1.5,500)';
ya = sinc((1/T)*t(:,ones(size(n))) - (1/T)*n(:,ones(size(t)))')*xs;
plot(n,xs,'bo',t,ya, 'r','Linewidth',1.5);grid;
xlabel('Time, msec');ylabel('Amplitude');
title('Reconstructed continuous-time signal y_{a}(t)');
axis([0 1 -1.2 1.2]);

3. Effect of Sampling in the Frequency Domain


Aliasing Effect in the Frequency Domain
% Program P6_3
% Illustration of the Aliasing Effect
% in the Frequency Domain
clear all;
close all;
clc;
t = 0:0.005:10;
xa = 2*t.*exp(-t);
subplot(2,2,1)
plot(t,xa);grid
xlabel('Time, msec');ylabel('Amplitude');
title('Continuous-time signal x_{a}(t)');
subplot(2,2,2)
wa = 0:10/511:10;
ha = freqs(2,[1 2 1],wa);
plot(wa/(2*pi),abs(ha));grid;
xlabel('Frequency, kHz');ylabel('Amplitude');
title('|X_{a}(j\Omega)|');
axis([0 5/pi 0 2]);
subplot(2,2,3)
T=1 ;
n = 0:T:10;
xs = 2*n.*exp(-n);
k = 0:length(n)-1;
stem(k,xs);grid;
xlabel('Time index n');ylabel('Amplitude');
title('Discrete-time signal x[n]');
subplot(2,2,4)
wd = 0:pi/255:pi;
hd = freqz(xs,1,wd);
4. Analog
Low pass Filters
plot(wd/(T*pi),
T*abs(hd));grid;
xlabel('Frequency, kHz');ylabel('Amplitude');
Analog title('|X(e^{j\omega})|');
low pass filters are employed as anti-aliasing filters and as anti-imaging
axis([0 1/T 0 2])

filters in the digital processing of continuous-time signals.

Design of Analog Low pass Filters


The first step in the design of any of these filters is the determination of the
filter order N and the appropriate cutoff frequency C. These parameters can
be determined using the MATLAB commands buttord for the Butterworth filter,
cheb1ord for the Type 1 Chebyshev filter, cheb2ord for the Type 2 Chebyshev
filter, and ellipord for the elliptic filter. C is the 3-dB cutoff frequency for the
Butterworth filter, the passband edge for the Type 1 Chebyshev filter, the
stopband edge for the Type 2 Chebyshev filter, and the pass band edge for the
elliptic filter. For the design of filters MATLAB commands are butter for the

Butterworth filter, cheby1 for the Type 1 Chebyshev filter, cheby2 for the Type 2
Chebyshev filter, and ellip for the elliptic filter.
Program P4 4 can be used for the design of the Butterworth lowpass filter.
% Program P6_4
% Design of Analog Lowpass Filter
clear all;
close all;
clc;
Fp = 3500;Fs = 4500;
Wp = 2*pi*Fp; Ws = 2*pi*Fs;
[N, Wn] = buttord(Wp, Ws, 0.5, 30,'s');
[b,a] = butter(N, Wn, 's');
wa = 0:(3*Ws)/511:3*Ws;
h = freqs(b,a,wa);
plot(wa/(2*pi), 20*log10(abs(h)),'r', 'LineWidth',1.5);grid
xlabel('Frequency, Hz');ylabel('Gain, dB');
Gain response
title('Gain response');
axis([0 3*Fs -60 5]);
0

-10

Gain, dB

-20

-30

-40

-50

-60

2000

4000

6000
8000
Frequency, Hz

10000

12000

EXPERIMENT NO. 7:
System Frequency
Transform
Introduction to Digital
Filter

Frequency Transform:
Is to represent mathematical function of time in frequency domain.

y [ n ] 2 y [ n1 ] =2.2 x [ n ] + 2 x [ n1 ] +2.2 x [ n2]

% Program P7_1
% Illustration of Aliasing Effect in the Time Domain
clear all; close all; clc;
num=[1 2 2.2];
den=[1 2];
freqz(num,den)

% Program P7_2
% Illustration of Aliasing Effect in the Time Domain
clear all; close all; clc;
num=[1 2 2.2];
den=[1 2];
Y=freqz(num,den)
Plot(abs(Y))

Can also view the frequency response of the system by defining w


% Program P7_3
% Illustration of Aliasing Effect in the Time Domain
clear all; close all; clc;
w=-pi:pi/100:pi
num=[1 2 2.2];
den=[1 2];
freqz(num,den)

Another way to do this is by computing the frequency response of the system


then define w and compute the response
% Program P7_4
% Illustration of Aliasing Effect in the Time Domain
clear all; close all; clc;
w=-pi:pi/100:pi
Y=(1+2*exp(-j*w)+2.2*exp(-2*j*w))./(1-2*exp(-j*w))
plot(w,abs(Y))

Exercise 1:

y [ n ] 2 y [ n2 ] =2.2 x [ n ] +2 x [ n1 ] +2.2 x [n2]


y [ n ] 0.1 y [ n1 ] + y [ n3 ] =x [ n ] +0.5 x [ n1 ] + x [n2]
y [ n ] 0.1 y [ n1 ] + y [ n3 ] + y [n5]=x [ n ] + x [ n4 ]
Plot the magnitude and phase response of the given difference
equation.

Filter:
Filters are usually used to discriminate a frequency or a band of frequency from
a given signal which is normally a mixture of both desired and undesired
signals. The undesired portion of the signal commonly comes from noise
sources such as power line hum etc. Or other signals which are not required for
the current application. Analog filters were being used successfully for decades
to serve this purpose. Although analog filters are excellent in some aspects,
especially in cost, they do have some serious demerits. One of the drawbacks of
analog filters is there non-linear phase characteristics. This is not a serious
problem in many of the applications, but it become serious in applications like
telecommunication, voice processing etc... Another drawback is the less sharp
cut-off frequency. It is possible to increase roll-off rate by cascading filter
stages, but this would increase system cost and complexity. On the other hand,
it is possible to achieve all these characteristics fairly by using a digital filter
Two types of Filter

FIR (Finite impulse Response)


IIR (Infinite impulse Response)
Filters can be designed by using different techniques but the most simple type
of filters are simple digital filters. In case of simple digital filters these can be
implemented simply by the difference equation,
Like FIR filters are
Low Pass:

1
1
y [ n ] = x [ n ] + x [ n1 ]
2
2
High Pass:

1
1
y [ n ] = x [ n ] x [ n1 ]
2
2
Band Pass:

1
1
y [ n ] = x [ n ] x [ n2 ]
2
2
Band Stop:

1
1
y [ n ] = x [ n ] + x [ n2 ]
2
2

EXPERIMENT NO. 8:
Simple Digital Filter
1) FIR

2) IIR

Low Pass:

1
1
y [ n ] = x [ n ] + x [ n1 ]
2
2
Find the frequency transform of the given difference equation
% Program
% Low pass FIR filter
clear all; close all; clc;
w=-pi:pi/100:pi;
num=[1/2 1/2];
den=[1];
H=freqz(num,den,w);
plot(w,abs(H))
xlabel('w frequency')
ylabel('abs(H)')

High Pass:

1
1
y [ n ] = x [ n ] x [ n1 ]
2
2
Find the frequency transform of the given difference equation

% Program
% Low pass FIR filter
clear all; close all; clc;
w=-pi:pi/100:pi;
num=[1/2 -1/2];
den=[1];
H=freqz(num,den,w);
plot(w,abs(H))
xlabel('w frequency')
ylabel('abs(H)')

Band Pass:

1
1
y [ n ] = x [ n ] x [ n2 ]
2
2
Find the frequency transform of the given difference equation
% Program
% Low pass FIR filter
clear all; close all; clc;
w=-pi:pi/100:pi;
num=[1/2 0 -1/2];
den=[1];
H=freqz(num,den,w);
plot(w,abs(H))
xlabel('w frequency')
ylabel('abs(H)')

Band Stop:

1
1
y [ n ] = x [ n ] + x [ n2 ]
2
2
Find the frequency transform of the given difference equation
% Program
% Low pass FIR filter
clear all; close all; clc;
w=-pi:pi/100:pi;
num=[1/2 0 1/2];
den=[1];
H=freqz(num,den,w);
plot(w,abs(H))
xlabel('w frequency')
ylabel('abs(H)')

EXPERIMENT NO.9
Design of FIR filters using
Rectangular window

Low Pass Filter:


Compute the impulse response of the Low pass filter. After computing impulse
response computes the frequency response of the response to visualize the
frequency
response.
% Program
Pass the
signal
ofFIR
anyfilter
frequency
from the
filter and observe the output
% Low pass
(Rectangular
Window)
clc
clear all
w=-pi:pi/200:pi;
n=0:40;
%Impulse Responce
h=(1/pi)*sinc((1*(n-20))/pi);
subplot(2,3,1)
plot(n,h)
xlabel('Impulse Responce for n=0:40')
%Frequency Responce
y=freqz(h,1,w);
subplot(2,3,2)
plot(w,abs(y))
xlabel('freqz Responce for n=0:40')
n1=0:50;
x=cos(1.5*n1);
subplot(2,3,3)
plot(n1,x)
xlabel('Input Signal')
subplot(2,3,4)
y=conv(x,h);
plot(y)
axis([0 90 -1 1])
xlabel('Output Signal')
subplot(2,3,5)
X=freqz(x,1,w);
plot(w,abs(X))
xlabel('X(e^j^w)')
subplot(2,3,6)

High Pass Filter:

Compute the impulse response of the Low pass filter. After computing impulse
response computes the frequency response of the response to visualize the
frequency response.
Pass the signal of any frequency from the filter and observe the output
% Program
% High pass FIR filter (Rectangular Window)
clc
clear all
w=-pi:pi/200:pi;
n=0:40;
%Impulse Responce
h=sinc((n-20))-(1.5/pi*sinc(1.5*(n-20)/pi))
subplot(2,3,1)
plot(n,h)
xlabel('Impulse Responce for n=0:40')
%Frequency Responce
y=freqz(h,1,w);
subplot(2,3,2)
plot(w,abs(y))
xlabel('freqz Responce for n=0:40')
n1=0:50;
x=cos(2*n1)+cos(0.5*n1);
subplot(2,3,3)
plot(n1,x)
xlabel('Input Signal')
subplot(2,3,4)
y=conv(x,h);
plot(y)
axis([0 90 -1 1])
xlabel('Output Signal')
subplot(2,3,5)
X=freqz(x,1,w);
plot(w,abs(X))
xlabel('X(e^j^w)')
subplot(2,3,6)

Band Pass Filter:

Compute the impulse response of the Low pass filter. After computing impulse
response computes the frequency response of the response to visualize the
frequency response.
Pass the signal of any frequency from the filter and observe the output
% Program
% Band pass FIR filter (Rectangular Window)
clc
clear all
w=-pi:pi/200:pi;
wL=1.5;wP=2.3;
n=0:40;
%Impulse Responce
h=(wL/pi*sinc(wL*(n-20)/pi))-(wP/pi*sinc(wP*(n-20)/pi))
subplot(2,3,1)
plot(n,h)
xlabel('Impulse Responce for n=0:40')
%Frequency Responce
y=freqz(h,1,w);
subplot(2,3,2)
plot(w,abs(y))
xlabel('freqz Responce for n=0:40')
n1=0:50;
x=cos(2*n1)+cos(0.7*n1);
subplot(2,3,3)
plot(n1,x)
xlabel('Input Signal')
subplot(2,3,4)
y=conv(x,h);
plot(y)
axis([0 90 -1 1])
xlabel('Output Signal')
subplot(2,3,5)
X=freqz(x,1,w);
plot(w,abs(X))
xlabel('X(e^j^w)')
subplot(2,3,6)
Y=freqz(y,1,w);
plot(w,abs(Y))
xlabel('Y(e^j^w)')

Band Pass Filter:

Compute the impulse response of the Band stop filter. After computing impulse
response computes the frequency response of the response to visualize the
frequency response.
Pass the signal of any frequency from the filter and observe the output
% Program
% Band Stop FIR filter (Rectangular Window)
clc
clear all
w=-pi:pi/200:pi;
wL=0.5;wP=2.3;
n=0:100;
%Impulse Responce
h=(wL/pi*sinc(wL*(n-50)/pi))-(wP/pi*sinc(wP*(n-50)/pi))+sinc(n-50)
subplot(2,3,1)
plot(n,h)
xlabel('Impulse Responce for n=0:40')
%Frequency Responce
y=freqz(h,1,w);
subplot(2,3,2)
plot(w,abs(y))
xlabel('freqz Responce for n=0:100')
n1=0:50;
x=cos(1*n1)+cos(1.5*n1);
subplot(2,3,3)
plot(n1,x)
xlabel('Input Signal')
subplot(2,3,4)
y=conv(x,h);
plot(y)
axis([0 90 -1 1])
xlabel('Output Signal')
subplot(2,3,5)
X=freqz(x,1,w);
plot(w,abs(X))
xlabel('X(e^j^w)')
subplot(2,3,6)
Y=freqz(y,1,w);
plot(w,abs(Y))
xlabel('Y(e^j^w)')

EXPERIMENT NO. 10
DIGITAL FILTER STRUCTURE

FIR Filter Design:


Conceptually the simplest approach to FIR filter design is to simply truncate to a
finite number of terms the doubly infinite-length impulse response coefficients
obtained by computing the inverse discrete-time Fourier transform of the
desired ideal frequency response. However, a simple truncation results in an

oscillatory behavior in the respective magnitude response of the FIR filter, which
is more commonly referred to as the Gibbs phenomenon.
The Gibbs phenomenon can be reduced by windowing the doubly infinite-length
impulse response coefficients by an appropriate finite-length window function.
The functions fir1 and fir2 can be employed to design windowed FIR digital
filters in MATLAB. Both functions yield a linear-phase design.
The function fir1 can be used to design conventional lowpass, highpass,
bandpass, and bandstop linear-phase FIR filters. The command
b

= fir1(N,Wn)

= fir1(N,Wn,high)

returns in vector b the impulse response coefficients, arranged in ascending


powers of z1 , of a lowpass or a bandpass filter of order N for an assumed
sampling frequency of 2 Hz. For lowpass design, the normalized cutoff
frequency is specified by a scalar Wn, a number between 0 and 1. For bandpass
design, Wn is a two-element vector [Wn1, Wn2] containing the specified
passband edges where 0 < Wn1 < Wn2 < 1. The command

with N an even integer, is used for designing a highpass filter. The command
b

= fir1(N,Wn,stop)

with Wn a two-element vector, is employed for designing a bandstop FIR filter. If


none is specified, the Hamming window is employed as a default. The command
b

= fir1(N, Wn, taper)

makes use of the specified window coefficients of length N+1 in the vector
taper. However, the window coefficients must be generated a priori using an
appropriate MATLAB function such as blackman, hamming, hanning, chebwin,
or kaiser. The commands to use are of the following forms:
taper = blackman(N) taper
taper = chebwin(N) taper

= hamming(N) taper = hanning(N)


= kaiser(N, beta)

The function fir2 can be used to design linear-phase FIR filters with arbitrarily
shaped magnitude responses. In its basic form, the command is
b

= fir2(N, fpts,

mval)

which returns in the vector b of length N+1 the impulse response coefficients,
arranged in ascending powers of z 1 . fpts is the vector of specified
frequency points, arranged in an increasing order, in the range 0 to 1 with the

first frequency point being 0 and the last frequency point being 1. As before,
the sampling frequency is assumed to be 2 Hz. mval is a vector of specified
magnitude values at the specified frequency points and therefore must also be
of the same length as fpts. The Hamming window is used as a default. To
make use of other windows, the command to use is
b

= fir2(N, fpts,

mval,taper)

where the vector taper contains the specified window coefficients.


A more widely used linear-phase FIR filter design is based on the Parks
McClellan algorithm, which results in an optimal FIR filter with an equiripple
weighted error (w) defined in Eq.

E ( ) =P ( ) [|H ( e j )|D ( ) ]
It makes use of the Remez optimization algorithm and is available in MATLAB as
the function firpm. This function can be used to design any type of single-band
or multiband filter, the differentiator, and the Hilbert transformer. In its basic
form, the command
b

= firpm(N,fpts,mval)

returns a vector b of length N+1 containing the impulse response coefficients of


the desired FIR filter in ascending powers of z 1 . fpts is the vector of
specified frequency points, arranged in increasing order, in the range 0 to 1
with the first frequency point being 0 and the last frequency point being 1. As
before, the sampling frequency is assumed to be 2 Hz. The desired magnitudes
of the FIR filter frequency response at the specified band edges are given by
the vector mval, with the elements given in equal-valued pairs. The desired
magnitudes between two specified consecutive frequency points f(k) and
f(k+1) are determined according to the following rules. For k odd, the
magnitude is a line segment joining the points {mval(k), fpts(k)} and
{mval(k+1), fpts(k+1)}, whereas, for k even, it is unspecified with the
frequency range [fpts(k), fpts(k+1)] being a transition or dont care
region. The vectors fpts and mval must be of the same length with the length
being even. Figure 7.4 illustrates the relationship between the vectors fpts
and mval given by
fpts = [0 0.2 0.4 0.7 0.8 1.0]
mval = [0.5 0.5 1.0 1.0 0.3 0.3]

FIGURE 9.1: ILLUSTRATION OF RELATIONSHIP BETWEEN VECTORS FPTS AND MVAL

The desired magnitude responses in the passband(s) and the stopband(s) can
be weighted by an additional vector wgts included as the argument of the
function firpm. The function can be used to design equiripple Types 1, 2, 3,
and 4 linear-phase FIR filters. Types 1 and
2 are the default designs for order N even and odd, respectively. Types 3 (N
even) and 4 (N odd) are used for specialized filter designs, the Hilbert
transformer and the differentiator. To design these two types of FIR filters the
fiags hilbert and differentiator are used for ftype in the last two
versions of firpm. The command
b = firpm(N,fpts,mval,wgts)

is used to design an FIR filter weighted in each band by the elements of the
weight vector wgts whose length is thus half that of fpts. The elements of
the vector wgts can be determined from the specified passband and stopband
ripples by dividing the maximum ripple value by the ripple values. To design a
Hilbert transformer or a differentiator, use the forms
firpm(N,fpts,mval,ftype)
firpm(N,fpts,mval,wgts,ftype)

where ftype is the string hilbert or differentiator. In the case of a


Hilbert transformer design, the smallest element in fpts should not be a 0.
The order N of the FIR filter to meet the given specifications can be estimated
using either Kaisers formula of Eq. (9.2).

20 log 10 ( p s ) 13
14.6 ( ) /2

The MATLAB function kaiord given below implements Kaisers formula:

(9.2)

function N = kaiord(Fp, Fs, dp, ds, FT)


% Computation of the length of a linear-phase
% FIR multiband filter using Kaisers formula
% dp is the passband ripple
% ds is the stopband ripple
% Fp is the passband edge in Hz
% Fs is the stopband edge in Hz
% FT is the sampling frequency in Hz.
% If none specified default value is 2
% N is the estimated FIR filter order
if nargin == 4,
F T=2 ;
end
if length(Fp) > 1,
TBW = min(abs(Fp(1) - Fs(1)), abs(Fp(2) - Fs(2)));
else
TBW = abs(Fp - Fs);
end
num = -20*log10(sqrt(dp*ds)) - 13;
den = 14.6*TBW/FT;
N = ceil(num/den);

The function kaiserord in the Signal Processing Toolbox can also be used for
estimating the filter order using Kaisers formula. It can be used in one of the
following forms:
[N,
Wn, beta, ftype] = kaiserord(fedge, aval,
[N,
Wn, beta, ftype] = kaiserord(fedge, aval,
c = kaiserord(fedge, aval,
dev, FT, cell)

dev)
dev, FT)

where FT is the sampling frequency in Hz whose default value is 2 Hz if not


specified; fedge is a vector of bandedge frequencies in Hz, in increasing order
between 0 and FT/2; and aval is a vector specifying the desired values of the
magnitude response at the specified bandedges given by fedge. The length of
fedge is 2 less than twice the length of aval and therefore must be even. dev
is a vector of maximum deviations or ripples in dB allowable for each band. If
the deviations specified are unequal, the smallest one is used for all bands.
The output data are in the desired format for use in fir1, with normalized
bandedges Wn and the parameter beta used for computing the window
coefficients as given in Eq. (7.36). The string ftype specifies the filter type for
fir1. It is high for highpass filter design, and stop for bandstop filter design.
The last form of kaiserord specifies a cell array whose elements are
parameters to fir1.

The MATLAB function firpmord implements the formula of Eq. (7.8). It can be
used in one of the following forms:

[N,fts,mval,wgts] = firpmord(fedge,aval,dev)
[N,fts,mval,wgts] = firpmord(fedge,aval,dev,FT)

where FT is the sampling frequency in Hz whose default value is 2 Hz if not


specified, fedge is a vector of bandedge frequencies in Hz, in increasing order
between 0 and FT/2; and aval is a vector specifying the desired values of the
magnitude response at the specified bandedges given by fedge. The length of
fedge is 2 less than twice the length of aval and therefore must be even. dev
is a vector of maximum deviations or ripples in dB allowable for each band. A
third form of firpmord is given by
c

= firpmord(fedge,aval,dev,FT, cell)

and specifies a cell array whose elements are the parameters to firpm.
In some cases, the order N determined using either method may not result in an
FIR filter meeting the original specifications. If it does not, the order should
either be increased or decreased by 1 gradually until the specifications are met.
Moreover, the order estimates may be highly inaccurate for very narrowband or
very wideband FIR filters.

Lab task
1.1..1

Using MATLAB determine the lowest order of a digital IIR lowpass


filter of all four types. The specifications are as follows: sampling rate
of 40 kHz, passband edge frequency of 4 kHz, stopband edge
frequency of 8 kHz, passband ripple of 0.5 dB, and a minimum
stopband attenuation of 40 dB. Comment on your results.

1.1..2

Using MATLAB determine the lowest order of a digital IIR highpass


filter of all four types. The specifications are as follows: sampling
rate of 3,500 Hz, passband edge frequency of 1,050 Hz, stopband
edge frequency of 600 Hz, passband ripple of 1 dB, and a minimum
stopband attenuation of 50 dB. Comment on your results.

1.1..3

Using MATLAB determine the lowest order of a digital IIR bandpass


filter of all four types. The specifications are as follows: sampling rate
of 7 kHz, passband edge frequencies at 1.4 kHz and 2.1 kHz,
stopband edge frequencies at 1.05 kHz and 2.45 kHz, passband

ripple of 0.4 dB, and a minimum stopband attenuation of 50 dB.


Comment on your results.
1.1..4

Using MATLAB determine the lowest order of a digital IIR bandstop


filter of all four types. The specifications are as follows: sampling
rate of 12 kHz, passband edge frequencies at 2.1 kHz and 4.5 kHz,
stopband edge frequencies at 2.7 kHz and 3.9 kHz, passband ripple
of 0.6 dB, and a minimum stopband attenuation of 45 dB. Comment
on your results.

1.1..5

Design the Butterworth bandstop filter by running Program P9_1.


Write down the exact expression for the transfer function generated.
What are the filter specifications? Does your design meet the
specifications? Using MATLAB, compute and plot the filters
unwrapped phase response and the group delay response.

1.1..6

Modify Program P10_1 to design a Type 1 Chebyshev lowpass filter


meeting the given specifications of Question Q 1. Write down the
exact expression for the transfer function generated. Does your
design meet the specifications? Using MATLAB, compute and plot the
filters unwrapped phase response and the group delay response.

1.1..7

Modify Program P10_1 to design a Type 2 Chebyshev highpass filter


meeting the specifications given in Question Q 2. Write down the
exact expression for the transfer function generated. Does your
design meet the specifications? Using MATLAB, compute and plot the
filters unwrapped phase response and the group delay response.

1.1..8

Modify Program P10_1 to design an elliptic bandpass filter meeting


the specifications given in Question Q 3. Write down the exact
expression for the transfer function generated. Does your design
meet the specifications? Using MATLAB, compute and plot the filters
unwrapped phase response and the group delay response.

1.1..9

Repeat the above question for the following cases: (a) sampling rate
of 20 kHz, (b) p = 0.002 and S = 0.002, and (c) stopband edge
= 2.3 kHz. Compare the filter length obtained in each case with that
obtained in the above question. Comment on the effect of the
sampling rate, ripples, and the transition bandwidth on the filter
order.

EXPERIMENT NO.11
Design of FIR filters using
Matlab commands.

Description:
Digital filters refers to the hard ware and software implementation of the
mathematical Algorithm which accepts a digital signal as input and produces
another digital signal as output whose wave shape, amplitude and phase
response has been modified in a specified manner.
Digital filter play very important role in DSP. Compare with analog filters they
are preferred in number of application due to following advantages.

Truly linear phase response


Better frequency response
Filtered and unfiltered data remains saved for further use.

There are two types of digital filters.


FIR (finite impulse response) filter
IIR (infinite impulse response) filter
Description of the Commands Used In FIR Filter Design:
FIR1:
FIR filters design using the window method. B = FIR1(N,Wn) designs an N'th
order low
pass FIR digital filter and returns the filter coefficients in length N+1 vector B.
The cutoff
frequency Wn must be between 0 < Wn < 1.0, with 1.0 corresponding to half
the
sample rate. The filter B is real and has linear phase. The normalized gain of the
filter
at Wn is -6 dB.
B = FIR1(N,Wn,'high') designs an N'th order high pass filter. You can also use B
=
FIR1 (N,Wn,'low') to design a low pass filter. If Wn is a two-element vector, Wn =
[W1

W2], FIR1 returns an order N band pass filter with pass band W1 < W < W2.
B = FIR1 (N,Wn,'stop') is a band stop filter if Wn = [W1 W2]. You can also
specify If Wn
is a multi-element vector, Wn = [W1 W2 W3 W4 W5 ... WN], FIR1 returns an
order N
Multiband filter with bands 0 < W < W1, W1 < W < W2, ..., WN < W < 1.
B = FIR1 (N,Wn,'DC-1') makes the first band a pass band.
B = FIR1(N,Wn,'DC-0') makes the first band a stop band.
By default FIR1 uses a Hamming window. Other available windows, including
Boxcar,
Hann, Bartlett, Blackman, Kaiser and Chebwin can be specified with an optional
trailing
argument. For example, B = FIR1(N,Wn,kaiser(N+1,4)) uses a Kaiser window
with
beta=4. B = FIR1(N,Wn,'high',chebwin(N+1,R)) uses a Chebyshev window.
For filters with a gain other than zero at Fs/2, e.g., high pass and band stop
filters, N must
be even. Otherwise, N will be incremented by one. In this case the window
length
should be specified as N+2.
By default, the filter is scaled so the center of the first pass band has magnitude
exactly
one after windowing. Use a trailing 'noscale' argument to prevent this scaling,
e.g.
B = FIR1(N,Wn,'noscale')
B = FIR1(N,Wn,'high','noscale')
B = FIR1(N,Wn,wind,'noscale').
You can also specify the scaling explicitly, e.g. FIR1(N,Wn,'scale'), etc.
FREQZ Digital Filter Frequency Response.
[H,W] = FREQZ(B,A,N) returns the N-point complex frequency response vector H
and
the N-point frequency vector W in radians/sample of the filter: given numerator
and
denominator coefficients in vectors B and A. The frequency response is
evaluated at N
points equally spaced around the upper half of the unit circle. If N isn't
specified, it
defaults to 512.
[H,W] = FREQZ(B,A,N,'whole') uses N points around the whole unit circle.
H = FREQZ(B,A,W) returns the frequency response at frequencies designated in
vector
W, in radians/sample (normally between 0 and pi).

[H,F] = FREQZ(B,A,N,Fs) and [H,F] = FREQZ(B,A,N,'whole',Fs) return frequency


vector F (in Hz), where Fs is the sampling frequency (in Hz).
H = FREQZ(B,A,F,Fs) returns the complex frequency response at the frequencies
designated in vector F (in Hz), where Fs is the sampling frequency (in Hz).
[H,W,S] = FREQZ(...) or [H,F,S] = FREQZ(...) returns plotting information to be
used
with FREQZPLOT. S is a structure whose fields can be altered to obtain different
frequency response plots. For more information see the help for FREQZPLOT.
FREQZ(B,A,...) with no output arguments plots the magnitude and unwrapped
phase of
the filter in the current figure window.
Designing A Low Pass Filter:
Suppose out target is to pass all frequencies below 1200 Hz

fs=8000; % sampling frequency


n=50; % order of the filter
w=1200/ (fs/2);
b=fir1(n,w,'low'); % Zeros of the filter
freqz(b,1,128,8000); % Magnitude and Phase Plot of the filter
figure(2)
[h,w]=freqz(b,1,128,8000);
plot(w,abs(h)); % Normalized Magnitude Plot
grid
figure(3)
zplane(b,1);
Designing High Pass Filter:
Now our target is to pass all frequencies above 1200 Hz
fs=8000;
n=50;
w=1200/ (fs/2); b=fir1(n,w,'high');
freqz(b,1,128,8000);
figure(2)
[h,w]=freqz(b,1,128,8000);
plot(w,abs(h)); % Normalized Magnitude Plot
grid
figure(3)
zplane(b,1);
Designing High Pass Filter:

fs=8000;
n=50;
w=1200/ (fs/2);
b=fir1(n,w,'high');
freqz(b,1,128,8000);
figure(2)
[h,w]=freqz(b,1,128,8000);
plot(w,abs(h)); % Normalized Magnitude Plot
grid
figure(3)
zplane(b,1);
Designing Band Pass Filter:
fs=8000;
n=40;
b=fir1(n,[1200/4000 1800/4000],bandpass);
freqz(b,1,128,8000)
figure(2)
[h,w]=freqz(b,1,128,8000);
plot(w,abs(h)); % Normalized Magnitude Plot
grid
figure(3)
zplane(b,1);
Designing Multi Band Filter
n=50;
w=[0.2 0.4 0.6];
b=fir1(n,w);
freqz(b,1,128,8000)
figure(2)
[h,w]=freqz(b,1,128,8000);
plot(w,abs(h)); % Normalized Magnitude Plot
grid
figure(3)
zplane(b,1);

Problems:
Design a band pass filter and band stop filter with the help of LPF and HPF
The filter has following specifications.
Band pass = 1200 2800 Hz

Band stop =1200-2800 Hz

Design a Multi band filter using HPF and LPF


The filter has following specifications
Pass band=1200 Hz 1800 Hz
Stop band = 1900 Hz 2200 Hz
Pass band = 2300 Hz 2700 Hz

EXPERIMENT NO.12
Design of IIR filters using
Matlab commands.

Description:
Matlab contains various routines for design and analyzing digital filter IIR. Most
of these are part of the signal processing tool box. A selection of these filters is
listed below.
Buttord ( for calculating the order of filter)
Butter ( creates an IIR filter)
Ellipord ( for calculating the order of filter)
Ellip (creates an IIR filter)
Cheb1ord (for calculating the order of filter)
Cheyb1 (creates an IIR filter)
Explanation Of The Commands For Filter Design:
Buttord:
Butterworth filter order selection.
[N, Wn] = BUTTORD(Wp, Ws, Rp, Rs) returns the order N of the lowest order
digital Butterworth filter that loses no more than Rp dB in the pass band and
has at least Rs dB of attenuation in the stop band. Wp and Ws are the pass
band and stop band edge frequencies, normalized from 0 to 1

(where 1 corresponds to pi radians/sample). For example


Low pass: Wp = .1, Ws = .2
High pass: Wp = .2, Ws = .1
Band pass: Wp = [.2 .7], Ws = [.1 .8]
Band stop: Wp = [.1 .8], Ws = [.2 .7]
BUTTORD also returns Wn, the Butterworth natural frequency (or, the "3 dB
frequency") to use with BUTTER to achieve the specifications. [N, Wn] =
BUTTORD(Wp, Ws, Rp, Rs, 's') does the computation for an analog filter, in
which case Wp and Ws are in radians/second. When Rp is chosen as 3 dB, the
Wn in BUTTER is equal to Wp in BUTTORD.
Ellipord:
Elliptic filter order selection.
[N, Wn] = ELLIPORD(Wp, Ws, Rp, Rs) returns the order N of the lowest order
digital elliptic filter that loses no more than Rp dB in the pass band and has at
least Rs dB of attenuation in the stop band Wp and Ws are the pass band and
stop band edge frequencies, normalized from 0 to 1 (where 1 corresponds to pi
radians/sample). For example,
Low pass: Wp = .1, Ws = .2
High pass: Wp = .2, Ws = .1
Band pass: Wp = [.2 .7], Ws = [.1 .8]
Band stop: Wp = [.1 .8], Ws = [.2 .7]
ELLIPORD also returns Wn, the elliptic natural frequency to use with ELLIP to
achieve the specifications. [N, Wn] = ELLIPORD(Wp, Ws, Rp, Rs, 's') does the
computation for an analog filter, in which case Wp and Ws are in
radians/second. NOTE: If Rs is much greater than Rp, or Wp and Ws are very
close, the estimated order can be infinite due to limitations of numerical
precision.
Cheb1ord:
Chebyshev Type I filter order selection.
[N, Wn] = 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 pass band
and has at least Rs dB of attenuation in the stop band. Wp and Ws are the pass
band and stop band edge frequencies, normalized from 0 to 1 (where 1
corresponds to pi radians/sample). For example,
Low pass: Wp = .1, Ws = .2
High pass: Wp = .2, Ws = .1
Band pass: Wp = [.2 .7], Ws = [.1 .8]
Band stop: Wp = [.1 .8], Ws = [.2 .7]

CHEB1ORD also returns Wn, the Chebyshev natural frequency to use with
CHEBY1 to achieve the specifications. [N, Wn] = CHEB1ORD(Wp, Ws, Rp, Rs, 's')
does the computation for an analog filter, in which case Wp and Ws are in
radians/second.
Butter:
Butterworth digital and analog filter design.
[B,A] = 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 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. If Wn is a two-element vector, Wn = [W1 W2], BUTTER returns an
order 2N bandpass filter with passband W1 < W < W2.
[B,A] = BUTTER(N,Wn,'high') designs a highpass filter. [B,A] =
BUTTER(N,Wn,'stop') is a bandstop filter if Wn = [W1 W2]. When used with three
left-hand arguments, as in [Z,P,K] = BUTTER(...), the zeros and poles are
returned in length N column vectors Z and P, and the gain in scalar K. When
used with four left-hand arguments, as in [A,B,C,D] = BUTTER(...), state-space
matrices are returned. BUTTER(N,Wn,'s'), BUTTER(N,Wn,'high','s') and
BUTTER(N,Wn,'stop','s') design analog Butterworth filters. In this case, Wn is in
[rad/s] and it can be greater than 1.0.

Ellip:
Elliptic or Cauer digital and analog filter design .
[B,A] = ELLIP(N,Rp,Rs,Wn) designs an Nth order low pass digital elliptic filter
with Rp decibels of peak-to-peak ripple and a minimum stop band attenuation
of Rs decibels. ELLIP 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. Use Rp = 0.5 and Rs = 20
as starting points, if you are unsure about choosing them.
If Wn is a two-element vector, Wn = [W1 W2], ELLIP returns an order 2N band
pass filter with pass band W1 < W < W2. [B,A] = ELLIP(N,Rp,Rs,Wn,'high')
designs a high pass filter. [B,A] = ELLIP(N,Rp,Rs,Wn,'stop') is a band stop filter if
Wn = [W1 W2]. When used with three left-hand arguments, as in [Z,P,K] =
ELLIP(...), the zeros and poles are returned in length N column vectors Z and P,

and the gain in scalar K. When used with four left-hand arguments, as in
[A,B,C,D] = ELLIP(...), state-space matrices are returned.
ELLIP(N,Rp,Rs,Wn,'s'),
ELLIP(N,Rp,Rs,Wn,'high','s')
and
ELLIP(N,Rp,Rs,Wn,'stop','s')
design analog elliptic filters. In this case, Wn is in [rad/s] and it can be greater
than 1.0.
Cheby1:
Chebyshev Type I digital and analog filter design.
[B,A] = 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. Use R=0.5 as a starting point, if you are unsure about choosing R.
If Wn is a two-element vector, Wn = [W1 W2], CHEBY1 returns an order 2N
bandpass filter with passband W1 < W < W2. [B,A] = CHEBY1(N,R,Wn,'high')
designs a highpass filter. [B,A] = CHEBY1(N,R,Wn,'stop') is a bandstop filter if
Wn = [W1 W2]. When used with three left-hand arguments, as in [Z,P,K] =
CHEBY1(...), the zeros and poles are returned in length N column vectors Z and
P, and the gain in scalar K.
When used with four left-hand arguments, as in [A,B,C,D] = CHEBY1(...), statespace
matrices are returned.
CHEBY1(N,R,Wn,'s'), CHEBY1(N,R,Wn,'high','s') and CHEBY1(N,R,Wn,'stop','s')
design analog Chebyshev Type I filters.In this case, Wn is in [rad/s] and it can be
greater
than 1.0.
Buttord and Butter Filter:
Designing IIR Low Pass Filter:
Suppose our target is to design a filter to pass all frequencies below 1200 Hz
with pass
band ripples = 1 dB and minimum stop band attenuation of 50 dB at 1500 Hz.
fs=8000;
The
[n,w]=buttord(1200/4000,1500/4000,1,50); % finding the order of the filter
sampling frequency for the filter is 8000 Hz;
[b,a]=butter(n,w); % finding zeros and poles for filter
figure(1)
freqz(b,a,512,8000);
figure(2)
[h,q] = freqz(b,a,512,8000);
plot(q,abs(h)); % Normalized Magnitude plot
grid
figure(3)
f=1200:2:1500;
freqz(b,a,f,8000) % plotting the Transition band
figure(4)
zplane(b,a) % pole zero constellation diagram

Designing IIR High Pass Filter:

We will consider same filter but our target now is to pass all frequencies above 1200 Hz

[n,w]=buttord(1200/5000,1500/5000,1,50);
[b,a]=butter(n,w,'high');
figure(1)
freqz(b,a,512,10000);
figure(2)
[h,q] = freqz(b,a,512,8000);
plot(q,abs(h)); % Normalized Magnitude plot
grid
figure(3)
f=1200:2:1500;
freqz(b,a,f,10000)
figure(4)
zplane(b,a)

Designing IIR Band Pass Filter:

[n,w]=buttord([1200/4000,2800/4000],[400/4000, 3200/4000],1,50);
Now we
wish to design a filter to pass all frequencies between 1200 Hz and 2800 Hz
[b,a]=butter(n,w,'bandpass');
with pass
band ripples = 1 dB and minimum stop band attenuation of 50 dB. The
figure(1)
sampling
frequency for the filter is 8000 Hz;
freqz(b,a,128,8000)
figure(2)
[h,w]=freqz(b,a,128,8000);
plot(w,abs(h))
grid
figure(3)
f=600:2:1200;
freqz(b,a,f,8000); % Transition Band
figure(4)
f=2800:2:3200;
freqz(b,a,f,8000); % Transition Band
figure(5)
zplane(b,a)

Designing IIR Band Stop Filter:

[n,w]=buttord([1200/4000,2800/4000],[400/4000, 3200/4000],1,50);
[b,a]=butter(n,w,'stop');
figure(1)
freqz(b,a,128,8000)
[h,w]=freqz(b,a,128,8000);
figure(2)
plot(w,abs(h));
grid
figure(3)
f=600:2:1200;
freqz(b,a,f,8000); % Transition Band
figure(4)
f=2800:2:3200;
freqz(b,a,f,8000); % Transition Band
figure(5)
zplane(b,a);

Problems:
Design all above filter using following commands

Ellipord( )

Ellip( )
Cheb1ord( )
Cheby1( )

EXPERIMENT NO.13
DFT

The discrete Fourier transform (DFT) X[k] of a finite-length sequence x[n] can be
easily computed in MATLAB using the function fft. There are two versions of this
function. fft(x) computes the DFT X[k] of the sequence x[n] where the length of
X[k] is the same as that of x[n]. fft(x,L) computes the L-point DFT of a sequence
x[n] of length N where L N . If L > N , x[n] is zero-padded with L N trailing
zero-valued samples before the DFT is computed. The inverse discrete Fourier
transform (IDFT) x[n] of a DFT sequence X[k] can likewise be computed using
the function ifft, which also has two versions.
Project 13.1

DFT Properties

Two important concepts used in the application of the DFT are the circular-shift
of a sequence and the circular convolution of two sequences of the same
length. As these operations are needed in verifying certain properties of the
DFT, we implement them as MATLAB functions circshift1 and circonv as
indicated below:
function y = circshift1(x,M)
% Develops a sequence y obtained by
% circularly shifting a finite-length
% sequence x by M samples
if
end

abs(M) > length(x)


M = rem(M,length(x));

if
end
y =
end

M
M

<
=

0
M

length(x);

[x(M+1:length(x))

x(1:M)];

function y = circonv(x1,x2)
L1 = length(x1);
L2 = length(x2);
if L1 ~= L2,
error('Sequences of unequal lengths'),
end
y = zeros(1,L1);
x2tr = [x2(1) x2(L2:-1:2)];
for k = 1:L1
sh = circshift1(x2tr,1-k); h = x1.*sh;
y(k) = sum(h);
end

Program P13_1 can be used to illustrate the concept of circular shift of a finitelength sequence. It employs the function circshift1
% Program P13_1
% Illustration of Circular Shift of a Sequence
clear all; close all; clc
M=6;
a=[0 1 2 3 4 5 6 7 8 9];
b = circshift1(a,M);
L = length(a)-1;
n = 0:L;
subplot(2,1,1);
stem(n,a);axis([0,L,min(a),max(a)]);
title('Original Sequence');
subplot(2,1,2);
stem(n,b);axis([0,L,min(a),max(a)]);
title(['Sequence Obtained by Circularly Shifting by
',num2str(M),'Samples']);

Program P13_2 can be used to illustrate the circular time-shifting property of the
DFT. It employs the function circshift1.

% Program P13_2
% Circular Time-Shifting Property of DFT
close all; clear all; clc
x=[0 2 4 6 8 10 12 14 16];
N = length(x)-1; n = 0:N;
y = circshift1(x,5);
XF = fft(x);
YF = fft(y);
subplot(2,2,1)
stem(n,abs(XF)); grid
title('Magnitude of DFT of Original Sequence');
subplot(2,2,2)
stem(n,abs(YF)); grid
title('Magnitude of DFT of Circularly Shifted Sequence');
subplot(2,2,3)
stem(n,angle(XF)); grid
title('Phase of DFT of Original Sequence');
subplot(2,2,4)
stem(n,angle(YF)); grid
title('Phase of DFT of Circularly Shifted Sequence');
Magnitude of DFT of Original Sequence
Magnitude of DFT of Circularly Shifted Sequence
80
80
60

60

40

40

20

20

Phase of DFT of Original Sequence


4

Program
be used
the

-2

-2
8

Phase of DFT of Circularly Shifted Sequence


4

-4
property
0
2
4
6
It employs the function circonv.

-4

% Program P13_3
% Circular Convolution Property of DFT
clear all; close all; clc
g1=[1 2 3 4 5 6];
g2=[1 -2 3 3 -2 1];
ycir = circonv(g1,g2);
disp('Result of circular convolution = ');
disp(ycir)
G1 = fft(g1);
% similarly compute fft of g2 and save in G2
yc = real(ifft(G1.*G2));

P13_3 can
to illustrate
circular
convolution
of the DFT.

disp('Result of IDFT of the DFT products = ');


disp(yc)

Program P13_4 can be used to illustrate the relation between circular and linear
convolutions
% Program P13_4
% Linear Convolution via Circular Convolution
close all; clear all; clc
g1=[1 2 3 4 5];
g2 = [2 2 0 1 1];
g1e = [g1
g2e = [g2

zeros(1,length(g2)-1)];
zeros(1,length(g1)-1)];

%Do circular convolution of g1e and g2e and save in ylin yourself
disp('Linear convolution via circular convolution = ');
disp(ylin);
y = conv(g1, g2);
disp('Direct linear convolution = ');disp(y)

Program P6_5 can be used to verify the relation between the DFT of a real
sequence, and the DFTs of its periodic even and the periodic odd parts.
% Program P13_5
% Relations between the DFTs of the Periodic Even
% and Odd Parts of a Real Sequence
close all; clear all; clc
x=[1 2 4 2 6 32 6 4 2 zeros(1,247)];
x1 = [x(1) x(256:-1:2)];
xe = 0.5 *(x + x1);
XF = fft(x);
XEF = fft(xe);
k = 0:255;
subplot(2,2,1);
plot(k/128,real(XF)); grid
ylabel('Amplitude');
title('Re(DFT\{x[n]\})');
subplot(2,2,2);
plot(k/128,imag(XF));
grid
ylabel('Amplitude');
title('Im(DFT\{x[n]\})');
subplot(2,2,3);
plot(k/128,real(XEF)); grid
xlabel('Time index n'); ylabel('Amplitude');
title('Re(DFT\{x_{e}[n]\} )');
subplot(2,2,4);
plot(k/128,imag(XEF)); grid
xlabel('Time index n');ylabel('Amplitude');
title('Im(DFT\{x_{e}[n]\})');

Re(DFT{x[n]})

50
0
-50

1.5

0
-50
-100

Amplitude
0.5
1
1.5
Time index n

0.5
-14

0
x 10

50

-50

50

Re(DFT{x e[n]} )

100
Amplitude

0.5

Im(DFT{x[n]})

100
Amplitude

Amplitude

100

1.5

Im(DFT{x e[n]})

0.5
0
-0.5
0

0.5
1
1.5
Time index n

Parsevals relation can be verified using the following program.


% Program P13_6
% Parseval's Relation
x = [(1:128) (128:-1:1)];
XF = fft(x);
% Take square of vector x and then add all its
% entries and save in a . Do yourself
b = round(sum(abs(XF).^2)/256)

EXPERIMENT NO. 14
(DECIMATION &
INTERPOLATION)

The digital signal processing structures discussed so far belong to the class of
single-rate systems as the sampling rates at the input and the output and all
internal nodes are the same. There are applications where it is necessary and
often convenient to have unequal rates of sampling at various parts of the
system including the input and the output. In this laboratory exercise you will
investigate first using MATLAB the properties of the up-sampler and the downsampler, the two basic components of a multi-rate system. You will then
investigate their use in designing more complex systems, such as interpolators
and decimators, and filter banks.

I.1 Basic Sampling Rate Alteration Devices


The objective of this section is to investigate using MATLAB the
operations of the up- sampler and the down-sampler both in the time
domain and in the frequency domain.
Project 10.1
Relations in the Time-Domain

Input-Output

Program P10_1 can be used to study the operation of a up-sampler.


%Program14_1
%Illustration of Up-Sampling by an Integer Factor
%
close all; clear all; clc
n=0:50;
x=sin(2*pi*0.12*n);
y=zeros(1,3*length(x));
y([1:3:length(y)])=x;
subplot(2,1,1)
stem(n,x);
title('InputSequence');
xlabel('Timeindexn');ylabel('Amplitude');
subplot(2,1,2)
stem(n,y(1:length(x)));
title('OutputSequence');
xlabel('Timeindexn');ylabel('Amplitude');
InputSequence

Amplitude

1
0.5
0
-0.5
-1

10

15

20

25
30
Timeindexn
OutputSequence

35

40

45

50

10

15

20

35

40

45

50

Amplitude

1
0.5
0
-0.5
-1

25
30
Timeindexn

%ProgramP14_2
%Illustration of Down-Sampling by an Integer Factor
%
close all; clear all; clc
n = 0:49;
m = 0:50*3-1;
x = sin(2*pi*0.042*m);
y = x([1:3:length(x)]);
subplot(2,1,1)
stem(n,x(1:50));axis([0 50 -1.2 1.2]);
title('InputSequence');
xlabel('Timeindexn');
ylabel('Amplitude');
subplot(2,1,2)
stem(n,y);axis([0 50 -1.2 1.2]);
title('OutputSequence');
xlabel('Timeindexn');
ylabel('Amplitude');

The Signal Processing Toolbox includes three M-functions which can be


employed to design and implement an interpolator or a decimator. The three Mfunctions are decimate, interp, and resample. Each function is available
with several options. In this section you will study the decimation and
interpolation operation using these functions.

Project 14.2

Decimator Design and Implementation

Program P10_3 illustrates the use of the M-function decimate in the design
and implementation of a decimator with an integer-valued decimation factor M.
In the option utilized in this program, decimate designs and uses a lowpass
decimation filter with a stopband edge.
%ProgramP14_3
%Illustration of Decimation Process
%
clear all; close all; clc
M=input('Down-samplingfactor=');
n=0:99;
x=sin(2*pi*0.043*n)+sin(2*pi*0.031*n);
y=decimate(x,M,'fir');
subplot(2,1,1);
stem(n,x(1:100));
title('InputSequence');
xlabel('Timeindexn');ylabel('Amplitude');
subplot(2,1,2);
m=0:(100/M)-1;
stem(m,y(1:100/M));
title('OutputSequence');
xlabel('Timeindexn');ylabel('Amplitude');

InputSequence

Amplitude

2
1
0
-1
-2

10

20

30

40

50
60
Timeindexn
OutputSequence

70

80

90

100

10

15

20

35

40

45

50

Amplitude

2
1
0
-1
-2

I.2 Decimator
and
Implementation

25
30
Timeindexn

Interpolator

Project 10.3
and Implementation

Design

and

Interpolator Design

Program P10_4 illustrates the use of the M-function interp in the design and
implementation of an interpolator with an integer-valued interpolation factor L.
interp designs and uses a lowpass interpolation filter with a stopband edge
satisfying Eq. (10.1).

|H ( e j )|=

L ,|| c / L
0, /L ||
(10.1)

%ProgramP14_4
%Illustration of Interpolation Process
%
clear all; close all; clc
L=input('Up-samplingfactor=');
% Generate the input sequence

n=0:49;
x=sin(2*pi*0.043*n)+sin(2*pi*0.031*n);
% Generate the interpolated output sequence
y=interp(x,L);
% Plot the input and the output sequences
subplot(2,1,1);
stem(n,x(1:50));
title('InputSequence');
xlabel('Timeindexn');ylabel('Amplitude');
subplot(2,1,2);
m=0:(50*L)-1;
stem(m,y(1:50*L));
title('OutputSequence');
xlabel('Timeindexn');ylabel('Amplitude');
InputSequence

Amplitude

2
1
0
-1
-2

10

15

20

25
30
Timeindexn
OutputSequence

35

40

45

50

10

20

30

40

70

80

90

100

Amplitude

2
1
0
-1
-2

Project 10.4

50
60
Timeindexn

Fractional-Rate Sampling Rate Alteration

Program P10_5 illustrates the use of the M-function resample in the design
and implementation of an interpolator with a fractional-rate interpolation factor
L/M. Resample designs and uses a lowpass interpolation filter with a stopband
edge.
% Program14_5
% Illustration of Sampling Rate Alteration by
% a Ratio of Two Integers

%
close all; clear all; clc
L=input('Up-samplingfactor=');
M=input('Down-samplingfactor=');
n=0:29;
x=sin(2*pi*0.43*n)+sin(2*pi*0.31*n);
y=resample(x,L,M);
subplot(2,1,1);
stem(n,x(1:30));axis([0 29 -2.2 2.2]);
title('InputSequence');
xlabel('Timeindexn');ylabel('Amplitude');
subplot(2,1,2);
m=0:(30*L/M)-1;
stem(m,y(1:30*L/M));axis([0 (30*L/M)-1 -2.2 2.2]);
title('OutputSequence');
xlabel('Timeindexn');ylabel('Amplitude');

I.3 Lab Task


1. What is the angular frequency in radians of the sinusoidal sequence in Program
P10_1? What is its length? What is the up-sampling factor L?
2. How is the up-sampling operation implemented in Program P10_1?
3. Modify Program P10_1 to study the operation of an up-sampler on a ramp
sequence.
4. Program P10_2 can be used to study the operation of a down-sampler
5. What is the angular frequency in radians of the sinusoidal sequence Program
P10_2? What is its length? What is the down-sampling factor M?
6. How is the down-sampling operation implemented in Program P10_2?
7. What are the frequencies of the two sinusoidal sequences forming the input
sequence in Program P10_3? What is the length of the input?
8. What are the type and order of the decimation filter?
9. Run Program P10_3 for M = 4 and comment on the results.
10.Change the frequencies of the two sinusoidal sequences in Program P10_3 in
the input signal to 0.045 and 0.029, and the length of the input to 120. Run the
modified Program P10 5 for M = 3. Comment on your results.

EXPERIMENT NO.15
Filter designing by Matlab
tools

Description:
There are two tool boxes available for designing, analyzing and for viewing
different responses (Impulse & Step) of FIR and IIR filters.
fvtool
fdatool

Filter Visualization Tool:


FVTOOL is a Graphical User Interface (GUI) that allows you to analyze digital
filters. FVTOOL (B,A) launches the Filter Visualization Tool and computes the
magnitude Response for the filter defined in B and A. FVTOOL(B,A,B1,A1,...) will
perform an analysis on multiple filters. The real advantage of this visualization
tool is that we can view the magnitude response and phase response
simultaneously, the impulse response, step response the coefficients of the
filter etc Let us consider a Low Pass FIR filter of order 30 which passes all
frequencies below 2000 Hz with sampling rate of 8000 Hz.
b=fir1(30,2000/4000,low);
fvtool(b,1)

Filter Design & Analysis Tool.


FDATOOL launches the Filter Design & Analysis Tool (FDATool). FDATool is a
Graphical User Interface (GUI) that allows you to design or import, and analyze
digital FIR and IIR filters. If the Filter Design Toolbox is installed, FDATool
seamlessly integrates advanced filter design methods and the ability to
quantize filters.

Now we will design a LPF on fdatool, the specifications for the filter are shown in
respective columns of FDA tool

Problems:
Design IIR butter worth filter with following specifications
-50 dB or more for 0 to 1200 Hz ( Stop Band Attenuation )
-1 dB or less from 2000 Hz to 4000 Hz ( Pass Band Characteristics )
-50 dB or more above 6000 Hz ( Stop Band Attenuation )
Sampling frequency 16000 Hz

EXPERIMENT No.16
Design an IIR filter to
suppress frequencies of 5
Hz and 30 Hz from given
signal.

Description:
We know from Fourier analysis that signals can be described by a summation of
frequency components. Typically, a filter is used to enhance signals by
attenuating unwanted frequency components and retaining desired frequency
components. In this practical we begin by creating a signals s with three
sinusoidal components ( at 5,15,30 Hz) and a time vector t of 100 samples
with a sampling rate of 100 Hz, and displaying it in the time domain. The Matlab
commands are shown below.
fs=100;
t=(1:100)/fs;
s=sin(2*pi*t*5)+sin(2*pi*t*15)+sin(2*pi*t*30);
plot(t,s)
grid

Now we design a filter to keep the 15 Hz sinusoid and eliminate the 5 and 30 Hz
sinusoids. We use the functions ellipord and ellip to create an infinite impulse
response (IIR) filter with a pass band from 10 to 20 Hz. The ellipord function
requires the specification of pass band corner frequencies, minimum transition
band frequencies near the pass band corner frequencies, the maximum pass
band ripple in decibels (dB), and the minimum stop band attenuation in dB. In
this example, we choose a transition frequency to be 5 Hz near the pass band
corners, with a maximum of 0.1 dB ripple in the pass band, and a minimum of
40 dB attenuation in the stop bands. We start by determining the minimum

order (pass band and stop band frequencies are normalized to the Nyquist
frequency):
wp1 = 10/50;
wp2 = 20/50;
ws1 = 5/50;
ws2 = 25/50;
wp = [Wp1 Wp2];
ws = [Ws1 Ws2];
rp = 0.1;
rs = 40;
[n,wn] = ellipord(wp,ws,rp,rs);
ellipord returns an order of 5, the minimum possible order for a low pass
prototype that will meet the constraints upon transformation to a band pass
filter. When we apply this order to the ellip function, internally we transform the
low pass prototype to a band pass filter using the function lp2bp. This doubles
the order, making n = 10. Next we use n, the order, and Wn, the pass band
corner frequencies, to actually design the filter. We also use freqz, a tool for
computing and displaying the frequency response of the descriptive transfer
function. When called with no left-hand-side arguments (i.e., return values),
freqz displays the magnitude and phase response of the filter normalized to the
Nyquist frequency.
[b,a] = ellip(n,.1,40,w);
freqz(b,a,128,100)
[h,w]=freqz(b,a,128,100);
plot(w,abs(h));
grid
title(Normalized Magnitude Response);
axis([0 50 0 1.2]);

figure(4)
sf=filter(b,a,s); % Time domain Response of the Filter
plot(t,sf)
grid
xlabel('Time (seconds)');
ylabel('Signal Amplitude');
title('Filtered Signal only 15 Hz frequency');

Problem:
Design an IIR filter to remove 100 and 150 frequencies from above signal.
x=1+sin (2*pi*50*t) + sin (2*pi*100*t) + 0.5 sin (2*pi*125*t) + 0.25 sin (2*pi*150*t);