Signal Notes - Siddharth Sir

Chapter
Introduction to
Signals and Systems 1
1.1 Introduction
Anything that carries information can be called a signal. Signals constitute an important part of our
daily life. A signal is defined as a single-valued function of one or more independent variables which contain
some information. A signal may also be defined as any physical quantity that varies with time, space or any other
independent variable. A signal may be represented in time domain or frequency domain. Human speech is a
familiar example of a signal. Electrical current and voltage are also examples of signals. A signal can be a
function of one or more independent variables. A signal may be a function of time, temperature, position,
pressure, distance etc. If a signal depends on only one independent variable, it is called a one-dimensional
signal, and if a signal depends on two independent variables, it is called a two-dimensional signal.
In this chapter we will study:
( i ) Type of signals
( i i ) Basic analog and digital signal
( i i i ) Basic operation as signals
( i v ) Type of systems
1.1 Analog, Discrete and Digital Signals
1. Continuous Time and Continuous Amplitude Signal (Analog Signals)

Continue time means in the signal x(t) ‘t’ is a continuous variable, that is signal x(t) has some finite
value at all values of variable ‘t’.
Continuous amplitude means that signal x(t) can take any value in a given time internal.
The analog signal are continuous time and continuous amplitude signals.
Example of analog signal:
x (t ) x (t )
t t
0 0
Fig 1.1 : Example of analog signal
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2 Signals & Systems
2. Discrete Time and Continuous Amplitude Signals (Discrete Signals)

Discrete time means that the signal is defined for distinct value on time axis. In the previous case signal had
continuous time and amplitude and to generate discrete signal we have to change continuous time axis into discrete
time axis. The time axis is transformed by using a sampler. The sampler takes in a signal with continuous time axis
and produces output which has discrete time axis. Sampler do not effect the amplitude axis of input axis.
The Block diagram of sampler:
Sampler
x (t ) Output
sampling time Ts
Fig. 1.2 : Block diagram of Sampler

In the above system outputs is equal to input at time instants equal to integer multiple of Ts. So if input
is signal shown in Fig. 1.3(a) then output will be signal shown in Figure 1.3(b).
Output
x (t )
Sampler
t t
0 Ts 2T s 3T s
(a) (b)
Fig. 1.3 : Input and Output of sampler
So we can say that output of sampler = Input t = nTs

or, output of sampler = x(t) t = nTs = x(nTs)
REMEMBER So if in any question signal x(t) is input to sampler then output is x(nTs), that is expression of output is
obtained by replacing t by nTs in expression of x(t).
For example:
Sampler
sin(100t) Output
Ts = 0.001 sec.
= sin(100t ) nTs
= sin(100 × 0.001n)
= sin(0.1n )
Fig.1.4 : Example of sampler
Study Note
• Sampler convert continuous time axis into discrete time axis, without affecting amplitude axis.
• Sampler can be visualized as a switch that connect input with output at integer multiple of Ts.
Input t = nT s
Sampler
Input Output Output
Ts

Introduction to Signals and Systems 3
3. Discrete Time and Discrete Amplitude Signals (Digital Signals)

Discrete amplitude means that the signal can take finite number of distinct values on amplitude axis.
Lets learn about discrete amplitude from an example:
The continuous amplitude signal can be changed to discrete amplitude signal by passing the signal through
a quantizer. The quantizer donot effect the time axis of the input signal. Suppose we have a quantizer whose transfer
characteristic is
Output
3.0
2.0
1.0
–2.5 –1.5 –0.5

0.5 1.5 2.5 Input
–1.0
–2.0
–3.0
Fig. 1.5 : Transfer characteristic of quantizer
From the transfer characteristic we can say that:

• When amplitude of input is between –0.5 and 0.5 then output is zero.
• When amplitude of input is between 0.5 and 1.5 then output is 1.0.
• When amplitude of input is between 1.5 and 2.5 then output is 2.0.
• When amplitude of input is between –0.5 and –1.5 then output is –1.0.
• When amplitude of input is between –1.5 and –2.5 then output is –2.0.
• When amplitude of input is between greater than 2.5 then output is 3.0.
• When amplitude of input is between less than –2.5 then output is –3.0.
Note: Transfer characteristic of system is output vs input characteristic.

4 Signals & Systems
Now let input to quantizer is x(t) = 3 sint

x (t )
3
2.5
1.5
0.5
t7 t8 t9 t 10 t 11 t 12
t
0 t1 t2 t 3 t4 t 5 t 6 2
–0.5
–1.0
–2.5
3.0
Output
3
2
1
t
0
–1
–2
–3
Fig. 1.6 : Input output of quantizer
The input x(t) has continuous amplitude as it can have any amplitude between –3 and 3 and when x(t) pass
through quantizer then output has only finite number of distinct amplitude values. That is output has only 7 amplitude
levels (–3, –2, –1, 0, 1, 2, 3).
So, quantizer convert continuous amplitude signal into discrete amplitude signal.
Thus to generate a digital signal from analog signal we use both quantizer and sampler in cascade.
Analog Sampler Quantizer Digital

signal signal
Discrete
signal
Fig. 1.7 : Block diagram of system to produce digital signal
Note: In the above system even if we interchange location of blocks then also output signal will remain same.
1.2 Elementary Analog Signals

There are several elementary, signals which play vital role in the study of signals and systems. These
elementary signals serve as basic building blocks for the construction of more complex signals. Infact, these
elementary signals may be used to model a large number of physical signals which occur in nature. These
elementary signals are also called standard signals.
1.2.1 Unit Step Function

The unit step function u(t) is defined as:
1 t 0
u(t) = ...(1)
0 t 0

Study Note
• The function is similar to that we studied in mathematics, here name of function is ‘u’ and ‘t’ is input to the function.
• From above definition we can see that when input to function ‘u’ is greater than zero then function give value ‘1’ else ‘0’.
From the above analysis we can see that, general definition of step function is
1 inpu t 0
u(input) = ...(2)
0 inpu t 0
u(input)
1.0
input
Fig. 1.8 : General definition of unit step function
Now to draw u(t), here input is ‘t’, so u(t) is

u (t )
1.0
input
Fig. 1.9 : Unit step function
3
1, t
1, 2 t 3 0 2
So, u(2t + 3) =
0, 2 t 3 0 3
0, t
2
1, t 0 1, t 0
and u(– t) =
0, t 0 0, t 0
Remember: Using definition of unit step function given in equation (2) we can find any complicated step function.
1.2.2 The Unit Impulse Function

The function is also called as diac delta function. The function is defined as:
, t 0
(t) = ...(3)
0, t 0
and ( t ) d( t ) = 1
So dirac delta function ‘ ’ with input ‘t’ is equal to infinity when input ‘t’ = 0 and 0 when input ‘t’ 0.
Also, function has area equal to 1.
So, ideal dirac delta function is

6 Signals & Systems
( t)
Fig. 1.10 : Ideal dirac delta function

If we look at ideal unit step function u(t), then it has discontinuity at t = 0 and it is not differentiable but if
we look at practical step function.
u p(t ) u (t )
Limit 0
1.0 1.0
t t
– /2 /2
(a) (b)
Fig. 1.11 : Practical and ideal unit step function
Now let us differentiate the practical unit step function, since practical unit step function has no
discontinuity and it can be differentiated easily, so it’s differentiation p(t) is
u p(t ) p( t)
1/
Differentiation
1.0
t t
– /2 /2 – /2 /2
Lim Lim
0 0
u (t ) (t )
1.0 Differentiation
t t
Fig. 1.12 : Dirac delta function is first derivative of unit step function
So, we can say that (t) is first derivative of u(t).
Study Note
This show that discontinuous signals can be differentiated and differentiation of the signal at discontinuity gives a delta
function.
Now, from the definition of dirac delta function we can say that general definition will be
, in pu t 0
(input) = ...(4)
0, in pu t 0
So, (at) where a is any constant is defined as
, at 0 , t 0
(at) =
0, at 0 0, t 0

but area of (t) was 1 i.e. t ) dt 1 then area of (at) will be
1 1
at ) dt = (t) d
a a
1
so, (at) is located at t = 0 with area
a
1
(at) = (t) ...(5)
a
(–t) = (t) ...(6)
Remember: Thus delta function is an even function.
b 1
so we can say that (at + b) is located at at + b = 0 i.e., t and it’s area is
a a
1 b
(at + b) = t ...(7)
a a
Lets see what happen when an arbitrary signal is multiplied with delta function, to analyse this we multiply
signal x(t) with practical delta function ( is very small).
x (t ) p(t )
1/ x(0)/
t t t
0 – /2 0 /2 – /2 0 /2
Fig.1.15 : Multiplication of signal x(t) with p (t)
From above we can see that,

x(t) . p (t) = x(0) p (t) so,
x(t) . (t) = x(0) (t) ...(8)
x(t) (t) = x(0) (t)
b
Also, x(t) (at + b) = x ( at b) ...(9)
a
REMEMBER x(t) (at + b) = value of x(t) at location of delta function multiplied by delta function
b
Thus, x(t) (at+ b) = x ( at b)
a
Using equation (8) and (9) we can say that,
x (t ) (t ) dt = x(0) (t ) dt = x (0) (t ) dt = x(0) ...(10)

8 Signals & Systems
1 b
x(t ) (at b) dt = x(t ) t dt
|a| a
b 1 b b 1
= x . t dt x ...(1)
a a a a a
b
So, general form of the question asked in GATE is to find the value of integral, I x ( t ) ( ct d ) dt. Then
a
to find the result we follow the following steps.
d
Step-1: Find out location of delta function, here location of delta function is t , and area of delta
c
1
function is .
c
d
S t e p - 2 : Now check that does the delta function lie within integration limits i.e. a b ; if yes then
c
move to step (3) else answer is obviously zero.
S t e p - 3 : The value of integration
I = value of x(t) at location of delta function multiplied by the area of delta
function
1
I = x( t ) t d
c c
Example 1.1
t
If (t) is dirac delta function then draw y ( t ) ( )d .
Solution 1.1
t
We need to find y( t ) ( )d
We have to integrate delta function to find y(t),

t
Since, y(t) = ( )d , for t < 0 y(t) = 0
(t )
and for t > 0 y(t) = 1

1 t 0
so, y(t) =
0 t 0
so, y( t) = u(t) t
t
u(t) = ( )d

Example 1.2
t
Plot y(t) if y ( t ) ( 3) ( 5) d .
Solution 1.2
(t + 3) – (t – 5)
y(t)
1.0
Integrate
5
–3 –3 5
t
So, y(t) = ( 3) ( 5) d
t t
= ( 3) d ( 5) d
y ( t ) = u(t + 3) – u(t – 5)
Example 1.3
Find the value of integration I,

3
I = (( 5 t 6) (t 5) 9t 2 ( t ) 3t 2 ( t 4)) dt
10
Solution 1.3
3
I = ((5 t 6) (t 5) 9t 2 ( t ) 3t 2 ( t 4)) dt
10
So, (t + 5) will be at t = –5, which is within integration limits and it’s area is 1, so
3
(5 t 6) ( t 5) dt = 1 (5 t 6) t 5
19
10
3
Now, 9t 2 ( t ) dt , since (t) is at t = 0 and it come inside integration limits and area of delta function is
10
1 so,
3
9t 2 ( t ) dt = 9t 2 1 0
t 0
10
3
Now, 3t 2 ( t 4) dt , since (t – 4) is at t = 4 that is outside integration limits so this is zero..
10
3
(( 5 t 6) ( t 5) 9 t 2 ( t ) 3 t 2 ( t 4) dt = –19 + 0 + 0 = –19
10

10 Signals & Systems
Now lets look at differentiation of delta function (t). For this we use the p(t) that is practical delta function.
p(t) p(t)
1/
Differentiation
/2
t t
– /2 /2 – /2
The signal p(t) has discontunuity at and thus differentiation will have delta functions and area of
delta function will be equal to amount of discontinuity
1
Thus, p(t) = t t
2 2
If we multiply p(t) with a signal x(t) and integrate the result we get,
(t / 2) (t / 2)
p(t ) x ( t ) dt = x( t ) dt
x( / 2) x ( / 2)
p(t ) x ( t ) dt =
Thus, applying 0 we get p(t) tend to (t) and
( x ( / 2) x ( / 2)
p(t ) x ( t ) dt = Lim
0
dx ( t )
p(t ) x ( t ) dt = 1
dt t 0
Thus, x ( t ) ( t ) dt = ( 1) x ( t ) t 0
Similarly we can prove,
• x( t ) ( t b ) dt ( 1) x ( t ) t b
1 1
• x ( t ) ( at ) dt x( t ) ( t ) dt x (t ) t 0
a a
1 b 1
• x ( t ) ( at b) dt x( t ) t dt x (t ) t b
a a a a
If n(t) denote nth derivative of delta function then,
n
• x( t ) ( t ) dt ( 1) n x n ( t )
t 0
Here xn(t) is nth derivative of x(t).

Thus the general expression is to find the value of integration,
d
n
I = x( t ) ( at b ) dt
c

Here, n(at + b) is nth derivative of delta function. The procedure to find value of I is
Step-1: Find location of delta function and it’s area, here (at + b) is at t = –b/a and area is 1 / a .
S t e p - 2 : Now find that does the delta function lie within the integration limit that is we need to check
that does the condition
b
d c
a
is correct or not. If yes move to step-3 else answer is zero.
1
Step-3: The value of I is = ( 1) n x n ( t ) b . That is I is equal to product of (–1)n with area of delta
t
a
a
function with nth derivative of x(t) at t = location of delta function.

For example:
1. To find value of e t
( t ) dt
Here, x(t) = e–t
t d
Thus, e ( t ) dt = ( 1) (e t ) ( 1) ( e t ) 1
dt t 0
t 0
20
2. To find value of cos( t ) ( t ) dt
10
Here, x(t) = cost and location of delta function is zero and it lie in the integration limits, thus
20
d 2 cos( t )
cos( t ) ( t ) dt = ( 1)2 1
10 dt 2 t 0
10
3. To find value of (3 t 2 ) (3 t 6 ) dt
10
The delta function has area 1/3 and location –2 the value of integration will be
d 1
= ( 1) (3t 2 )
dt t 2 3
1
( 1) (6 t ) t 2 =4
3
1.2.3 Rectangular Function

The rectangular function is defined as
t A, t
A rect = 2 2 ...(12)
0, else wh ere
t
so, A rect is

t
– /2 /2
Fig. 1.14 : Rectangular function
The equation (12) show the basic definition of rectangular function, the name of function is ‘rect’ and
input to function is having some numerator and denominator and using equation (12) we can get a general
definition of rectangular function,
Denominator Denominator
Nu merator A, Numerator <
A rect = 2 2 ...(13)
Den om inator 0, elsewhere
Now using equation (13) we can easily find out any rectangular signal.
For example:
2 t 30
To plot 5 rect we can use equation (13) and no need to do shifting, inversion on scaling. Heree
20
numerator is –2t + 30 and denominator is 20. 5 rect
2t 30
20
20 20
2 t 30 5, 2t 30
So, 5 rect = 2 2
20 0, else where 5
t
5 , 10 t 20 10 20
=
0 , else wh ere Fig. 1.15
Similarly we can find any rectangular function easily.

There are two pattern of question here.
• Pattern 1: A rectangular function is given and we have to find value of function at a time instant To
Nu merator
suppose A rect is given then to find value of function at To we will find value of
Den om inator
Denominator Denominator
numerator at To and if (Numerator ) t To
then value of function is A else
2 2
it is zero.
• Pattern 2: To plot a given rectangular function then it can be done using the procedure shown in
above example.
1.2.4 Triangular Function

The triangular function is defined as
t
t t A 1 , t
A = A tri ...(14)
0, elsewhere
The function is defined as shown in equation (14), this is the basic definition. Here (t/ ) is input to
function and ‘ ’ or ‘tri’ is name of function.

The plot of A (t/ ) will be
t
–
Fig. 1.16 : Triangular function
Num erator
Now let us define the triangular function for general cases, the input to the function will be ,
Denom in ator
using equation (14) we can define.
Numerator
Num erator A 1 , Denominator < Numerator < Denominator
A = Denomirator ...(15)
Den om in ator
0, elsewhere
Now using equation (15) we can define any triangular function.
For example:
2 t 40
To plot 10 , we can use equation (15), and we do not apply shifting, scaling and inversion.
50
The numerator is –2t + 40 and denominator is 50, so
2t 40
2 t 40 10 1 , 50 2t 40 50
10 = 50
50
0, elsewhere
2t 40
10 1 , 5 t 45
= 50
0, elsewhere
2 t 40
The maximum value of triangular function is 10 and it occur when 10 1 is maximum.
50
That is –2t + 40 = 0
t = 20
2 t 40
so, 10 can be drawn as
50
10
t
–5 20 45
Fig. 1.17

For triangular function also there are two pattern of question.

• Pattern 1: A triangular function is given and we have to find value of function at t = To. Suppose the
Nu merator
rectangular function A is given then to find value of function at t = To, first of all wee
Den ominator
need to check whether Denominator < (Numerator) t To

Denominator if yes then value of function is
Numinator
A 1 else value of function is zero..
Denominator t To
• Pattern 2: To plot a given triangular function, this can be done using procedure shown in above
example.
Remember: Maximum value A, of at rectangular function is at that value of t where input to the triangular function is zero.
1.2.5 Unit Ramp Function

The continuous time unit ramp function r(t) is that function which starts at t = 0 and increases linearly with
time and is defined as:
t for t 0
r(t) = ...(16)
0 for t < 0
or, r(t) = t u(t) ...(17)
The unit ramp function has unit slope. It is a signal whose amplitude varies linearly. It can be obtained by
integrating the unit step function. That means, a unit step signal can be obtained by differentiating the unit ramp
signal.
t t
i.e., r(t) = u( ) d 1d t for t 0

0 0
d
u(t) = r( t ) ...(18)
dt
The delayed unit step ramp signal r(t – a) is given by
t a for t a
r(t – a) = ...(19)
0 for t a
or, r(t – a) = (t – a) u(t – a) ...(20)
r( t) r( t – a)
Slope = 1
Slope = 1
t t
0 0
(a) (b)
Fig. 1.18 : (a) Unit ramp signal (b) Delayed unit ramp signal
In equation (16), so we can say that ‘r’ is the ramp function and ‘t’ is input, so general ramp function can be

defined as,
In pu t , In pu t 0
r(input) = ...(21)
0, In pu t > 0
r(input) = input u(input) ...(22)
Let us define r(2t + 3), here input is (2t + 3) so,
r(2t + 3) = (2t + 3) u(2t + 3)
The plot of r(2t + 3)
slope = 2
3.0
t
–1.5
Fig. 1.19
1.3 Representing any Arbitrary Signal In Terms of Elementary Analog

Signals
Sometimes in a question x(t) is given and we have to find expression of x(t) in terms of unit step, unit
impulse and ramp signals. For solving such questions we will use these three concepts:
( i ) The location of discontinuity in x(t) will give the location of step function and the multiplication
factor of step function will be decided by amount of discontinuity.
( i i ) The location of change of slope in x(t) will give the location of ramp function and multiplication
factor of ramp function will be decided by amount of change in slope.
( i i i ) The location of impulse will be visible in the plot of x(t) directly.
For example:
x ( t)
t
–5 5
In the above signal x(t) there is no discontinuity, but change of slope is seen at t = –5, t = 0, t = 5.
So we will use three ramp signals.
x(t)
–2
5
slope = +1
+1 slope = -1
+1
slope = 0 slope = 0
–5 5

So, we have to add slope +1 at t = –5, add slope –2 at t = 0 and add +1 at t = 5. So we can represent,
x(t) = 1 r(t + 5) –2r(t) + r(t – 5)
x(t) = (t + 5) u(t + 5) –2t(u) + (t – 5) u(t – 5)
Now lets see another example to represent x(t) in terms of elementary analog signals,
x (t )
t
–5 –4 –1 1 2 3
Now lets analyse the signal.
x (t )
0
slope = 0 2 0
–1
–1
slope = 0 slope = 0
+1 1 +1
slope = 1 slope = –1
slope = 0 slope = 0
t
–5 –4 –1 1 2 3
Discontinuity Discontinuity
So, discontinuity at t = –1 and 1, discontinuity at t = –1 is of 1 and at t =1 is of –1, slope changes are shown
in figure. So, slope change of +1 at t= –5, slope change of –1 at t = –4, slope change of –1 at t = 2 and slope change
of +1 at t = 3.
So, x(t) = +1 r(t + 5) – r(t + 4) –r(t – 2) + r(t – 3) + u(t + 1) – u(t – 1)
1.4 Differentiation of an Arbitrary Signal

Suppose an arbitrary signal x(t) is given and we have to find expression of differentiation x(t). To find
expression of dx(t)/dt we will remember one concept that is
• The differentiation of discontinuity is given by delta function (differentiation of u(t) is (t)). This
means where x(t) has discontinuity, dx(t)/dt will have delta function and area of delta function
depend on amount of discontinuity.
• And where the signal is continuous there the differentiation is done as we do in maths.
Study Note
The presence of discontinuity and amount of discontinuity can be found by assuming that the signal is a road and
we are riding our bike on the whole signal, all the points where we encounter an accident there we have a
discontinuity and the amount of upward or downward shift we need to get back on the road tells the amount of
discontinuity. If we have to move up to get back on the road then the delta function in differentiation will be
upward and area will be equal to shift, similarly if we have to move down to get back on the road then the delta
function in differentiation will be downward and area will be equal to shift.

dx ( t )
For example: To plot of x(t)
dt
x ( t)
t
–3 3
dx ( t )
To plot , since x(t) has no discontinuity its differentiation can be plotted easily, because the signal has
dt
no discontinuity as if the signal is a road then we can ride bike on whole time axis and we will not encounter any
accident.
d
x( t )
dt
5/3
3
t
–3 0
–5/3
dx ( t ) 5 10 5
So, = u( t 3) u( t ) u( t 3)
dt 3 3 3
Now taking another signal
This signal has two discontinuities, at t = 5 and –5, and for other values of ‘t’ the signal is continuous.
x (t )
Slope = 1
5
–5
t
5
–5
The discontinuity at t = –5 is –5 and at t = 5 is –5. So area of delta at t = 5 and –5 is –5.
Remember: Suppose a signal has discontinuity at t = T0 then amount of discontinuity is equal to x( T0 ) x ( T0 ) .

d
So, x ( t ) is
dt
d
x( t )
dt
1
–5 5 t
d
so, x ( t ) = –5 (t – 5) – 5 (t + 5) + u(t + 5) –u(t – 5)
dt
1.5 Elementary Discrete Signals

A discrete-time signal x[n] may represent a phenomenon for which the independent variable is inherently
discrete. For instant, the daily closing stock market average is by its nature a signal that evolves at discrete
points in time (that is, at the close of each day). On the other hand a discrete-time signal x[n] may be obtained
by sampling a continuous time signal x(t) such as
x(t 0 ), x(t 1 ),....., x(t n )....
or in a shorter form as
x[0], x[1], ......., x[n], .....
or, x0, x1, ......., xn, ......
where we understand that,
x n = x[n] = x(t n )
A discrete-time signal x[n] can be defined in two ways:
1. We can specify a rule for calculating nth value of the sequence. For example
n
1
n 0
x[n] = x n 2
0 n 0
n
1 1 1
or, {x n } = 1, , , ....., ,......
2 4 2
2. We can also explicitly list the values of the sequence. For example the sequence shown in figure can
be written as
{x n } = {...., 0, 0, 1, 2, 2, 1, 0, 1, 0, 2, 0, 0,....}
or, {x n } = {1, 2, 2, 1, 0, 1, 0, 2}
We use the arrow to denote the n = 0 term. We shall use the convention that if no arrow is indicated,
then the first term corresponding to n = 0 and all the values of the sequence are zero for n < 0.

1.5.1 Unit Step Sequence

The unit step sequence is defined as
1, n 0
u[n] = ...(23)
0, n 0
Here ‘u’ is name of function and ‘n’ is input to function, n that is input to function can only be integer.
The plot of u(n) is
u [n ]
1.0
.....
n
0 1 2 3
Fig. 1.20 : Unit step sequence

Using equation (23) we can define general form of step sequence,
1, in pu t 0
u[input] = ...(24)
0 , in pu t 0
so we can easily define any unit step sequence using equation (24).
For example:
u[2n + 7], here input is 2n + 7, so
1, 2n 7 2 1, n 3.5
u[2n + 7] =
0 , 2n 7 0 0, n 3.5 u [2n + 7]
since input or value ‘n’ can be integer so,

1, n 3 .....
u[2n + 7] = n
0, n 3 –3 –2 –1 0 1 2
Study Note
The difference between u(t) and u[n] is that at t =0 the value of u(t) is undefined but value of u[n] is defined and is equal to
1.0.
1.5.2 Unit Impulse Sequence

The unit impulse sequence, [n] is defined as
1, n 0
[n] = ...(25)
0, n 0
[n]
1.0
n
0
Fig. 1.21 : Unit impulse sequence
Here ‘ ’ is name of function and ‘n’ is input to function, and value of function is ‘1’ where input is zero.
Note : Difference between (t) and (n), in case of (t) value is and area is 1 but in case of [n] value is 1.

So, we can define unit impulse sequence in general form also,

1, in pu t 0
[input] = ...(26)
0 , in pu t 0
In the expression of equation (26) we have to see that value of ‘n’ should be integer for which input = 0.
Lets see through example,
• [2n + 5], here input is 2n + 5, so
1, 2n + 5 = 0 1, n 2.5
[2n +5] =
0, elsewh ere 0, n 2.5
Now if we plot [2n + 5] then, [2n + 5] will be 1 at n = –2.5, and in discrete axis n = –2.5 is invalid
or not existing.
So, plot of [2n + 5] is zero always.
1, 2n + 8 = 0 1, n 4
• [2n + 8] =
0, elsewh ere 0, n 4
Since here ‘n’ is integer so plot of [2n + 8] will be
[2n + 8]
1.0
n
–4 0
Study Note
We can see that [2n] will be
1, 2n = 0 1, n 0
[2n] =
0 , elsewh ere 0 , elsewh ere
1
so, [2n] = [n], but (2 t ) (t )
2
The general form will be [an + b] is

b
1, n
1 , an b 0 a
[an + b] =
0 , elsewh ere b
0, n
a
b b
Now if is an integer then [an + b] is ‘1’ at n
a a
Otherwise [an + b] is zero always.
Multiplication of arbitrary discrete signal x[n] with delta function.
Let x[n] is any signal then,
• x[n] [n], here
x[n] is {...x [ 2], x[ 1], x[0], x [1], x[2]...}
and [n] is {...0, 0, 1, 0, 0...}

so, x[n] [n] = x[0] [n],
here x[0] show value and [n] show location n = 0. That is x[n]. [n] is equal to delta function with
value equal to x[0].

• Similarly x[n] [n – n0] = x[n0] [n – n0] ...(27)

That is multiplying signal x[n] with shifted delta signal only changed the amplitude of that delta function
by factor equal to value of x[n] at location of delta function.
The procedure to find multiplication of arbitrary signal with delta function that is x[n] [an + b].
Step-1: Find the information about delta function,
b
1, n
1, an b a
[an + b] =
0 , elsewh ere b
0, n
a
b
If is integer then goto step-2 else if not integer then delta function will be zero always and
a
x[n] [an + b] = 0
b
Step-2: x[n] [an + b] = x [an b]
a
Now lets look at another pattern of questions.
Using definition of [n] we can say that,
• [ n ] 1, because [n] is ‘1’ only at n = 0 and 0 elsewhere
• x[n] [n] x[0] [n] x[0], because x[n] [n] is x[0] at n = 0 and 0 elsewhere..
The general form of question is to find the value of summation,

d
S= x[ n ] [ a n b]
c
To find value of S we follow the procedure.

Step-1: Find the delta function,
b
1, n
a
[an + b] =
b
0, n
a
b
if is not integer then S = 0, else goto step-2.
a
b b
Step-2: The location of delta function is , now check whether lie is summation limits i.e.,
a a
b
c d.
a
If above condition is true then move to step-3 else answer S is zero.
Step-3: Value of S = value of x[n] at location of delta function.
S = x[n ] n b
a

Example 1.4
If y[n] is a discrete signal defined as,
∞
y[n] = δ[n - m] then plot y[n]
m=0
Solution 1.4
We can see that, y[n] = [n m] [n ] [n 1] [n 2]... y [n ]

m 0
[n] is ‘1’ only at n = 0, [n – 1] is ‘1’ at n = 1, 1.0

.....
[n – 2] is ‘1’ only at n = 2. Thus y[n] is n
0 1 2 3
so, y[n] = u [n ] [n m]
m 0
Example 1.5
Find the discrete signal x[n] which is defined as,

n
x[n] = δ[m]
m= -
∞
Solution 1.5
we know that [m] will be 0 if m 0 and 1 if m = 0.
n
So, if n < 0 then, x[n] = [m ] (for n < 0, [m] will be zero always)
m
x[n] = 0
n
Now if n 0 then, x[n] = [m ] for n > 0 x[n] will be
m
= ... [–2] + [–1] + [0] + [1]...

= ...0 + 0 + 1 + 0...
= 1
so, for n 0 x[n] = 1
1, n 0
x[n] = u[n ]
0, n 0
n
u[n] = [n m] = [ m]
REMEMBER m 0 m
n
and u[n – n0] = [n n0 m] = [m no ]
m 0 m

Example 1.6
What is the difference between

( a ) u(t) + u(–t) and u[n] + u[–n]
(b) u(t + 5) – u(t – 5) and u[n + 5] – u[n – 5]
Solution 1.6
(a) We know that,
1, t 0 1, t 0
u(t) = and u( t )
0, t 0 0, t 0
The value of u(t) is undefined at t = 0, but if we recall practical step function then value of u(t) = 0.5 at t =
0, similarly value of u(–t) at t = 0 is 0.5 so u(t) + u(–t) will be
u (t ) u (–t ) u (t ) + u (–t )
1.0
1.0 + 1.0 =
t t t
0
So, u(t) + u(–t) = 1.0
1, n 0 1, n 0
Now, u[n] = and u [ n ]
0, n 0 0, n 0
Here value of u[n] at n = 0 is 1.0 and value of u[–n] at n = 0 is 1.0.

u [n ] + u [–n ]
u( n) u( n) 2.0
1.0
1.0 1.0
..... + ..... = ..... .....
n n n
0 1 2 3 –2 –1 0 –3 –2 –1 0 1 2 3
So, u[n] + u[–n] = 1 + [n]

(b) To plot u(t + 5) –u(t – 5) we have to subtract two signals.
u (t + 5) u( t – 5) u (t + 5) –u ( t – 5)
1.0 – = 1.0
1.0
n n n
–5 5 –5 5
The plot of u(n + 5) – u(n – 5) can be obtained by subtracting these two signals.
u[n + 5] – u [n – 5]
u [n + 5] u [n – 5]
1.0 1.0
1.0
.....
– ..... =
n n n
–5 –4 –3 –2 –1 0 1 2 3 5 6 7 –5 –4 –3 –2 –1 0 1 2 3 4

Study Note
We saw that any signal x[n] can be written as
x[n] = {....., x[–3], x[–2], x[–1], x[0], x[1], x[2]...}
so, we can represent x[n] as a sum of weighted and shifted impulses as
x[n] = {.....+ x[–3] [n + 3] + x[–2] [n + 2] + x[–1] [n + 1] + x[0] [n] +
x[1] [n – 1] + ...}
Here the representation is simple to understand, x[–3] is at n = – 3 and to show that we can multiply it with delta
function at n = –3 and so on for all other terms. Thus we can write whole signal as summation of weighted and
shifted delta function.
x[n] = x[k ] [n k]
k
Similarly in continuous time domain we can write an arbitrary signal as integration of weighted and shifted delta
functions.
x( t) = x( ) ( t )d
1.5.3 Ramp Signal

In discrete domain ramp signal is r[n], the signal is defined as
n, n 0
r[n] = ...(28)
0, n 0
r[n]
3
2
.....
1
0 n
0 1 2 3
Fig. 1.22 : Ramp signal
From equation (28) we can see that ‘r’ is function and ‘n’ is input here r[n] can also be written as
r[n] = nu[n] ...(29)
Thus, r [input] = input u[input] ...(30)
in pu t , in pu t 0
r[input] = ...(31)
0, in pu t 0
Equation (30) and (31) show general definition of ramp function.

Using the general definition easily define.
r[3n + 9]
• r[3n + 9] = [3n + 9] u[3n + 9]
12
3n 9 , 3n 9 0 9
=
0, 3n 9 0 6
.....
3
3n 9, n 3
= n
0, n 3 –3 –2 –1 0 1

• r[2n + 7] = [2n + 7] u[2n + 7] r[2n + 7]
2n 7 , 2n 7 0 2n 7, n 3.5 7
= 5
0, 2 7n 0 0, n 3.5
3
.....
2n 7, n 3 1
=
0, n 3 n
–3 –2 –1
1.6 Basic Operation on Signals

There are three basic operation on signals where a new signal is obtained by manipulating independent
variable of the given signal:
1. Time shifting
2. Time scaling
3. Time inversion
To generate a new signal from given signal we should first apply time shifting then the scaling and then
time inversion. Lets look at these operation.
1. Time Shifting:
Suppose x(t) is given and let us assume,
y ( t ) = x(t – 2)
so shifting of variable has to be done to get y(t), so
y ( t ) = x(t – 2)
y ( 0 ) = x(–2),
y(1) = x(–1),
y ( 2 ) = x( 0) ,
we can see that, x(–2) goto y(0) also x(0) goto y(2), So what was at t = –2 goto t =0 and what was
at t = 0 goto t = 2, which show that to get y(t) we shift x(t) right by 2 units.
Similarly for y(t) = x(t + 2) which implies that,
y ( 0 ) = x( 2) ,
y ( 1 ) = x( 3) ,
y ( 2 ) = x( 4) ,
So, x(2) goto y(0) also x(4) goto y(2), which implies that to get y(t) from x(t) we shift x(t) left by
2 units.
y(t) = x(t – a) Right shift by a
y(t) = x(t + a) Left shift by a
x (t ) x (t – 5) x (t + 2)
1 1.0
1
t t t
–3 –2 4 2 3 9 –5 –4 2
Fig. 1.23 : Time shifting in continuous time domain

Same is the case for discrete signals also,

3 3
x(n ) x ( n – 3)
2 2 2 2
1 1 1 1
–3 –2 –1 0 1 2 3 4 5 6 n –2 –1 0 1 2 3 4 5 6 7 n
(a) (b)
3
x ( n + 2)
2 2
1 1
–4 –3 –2 –1 0 1 2 3 4 n
(c)
Fig. 1.24 : Time shifting in discrete time domain
2. Time Scaling:
Suppose x(t) is given signal and y(t) is defined as y(t) = x(at), here a is positive.
Now there are 2 cases.
Case-1: a > 1, let a = 2
y(t) = x(2t)
so, y(0) = x(0)
y(1) = x(2)
y(2) = x(4)
So, x(2) goes to y(1) and x(4) goes to y(2)
This implies that y(t) is contracted version of x(t) and contraction is by factor of 2. For example,
x (t ) y ( t)
5 5
t t
–4 4 –2 2
Fig. 1.25 : Plot of x(t) and x(2t)
1
Case-2: a < 1, let a
2
t
y(t) = x
2
So, y(t) = x(0)
1
y(1) = x
2
y(2) = x(1)
3
y(3) = x
2
y(4) = x(2)
So, x(1/2) goto y(1) and x(1) goto y(2).

This implies that y(t) is expanded version of x(t) and expansion is by factor of 2, for example
x (t ) y ( t)
5 5
t t
–4 4 –8 8
Fig. 1.26 : Plot of x(t) and x(t/2)
In the discrete-time case, we can write the time scaling as follows:

y(n) = x(an)
Again when a > 1, it is time compression and when a < 1, it is time expansion.
Let x(n) be a sequence as shown in Fig. 1.27 (a). If a = 2, y(n) = x(2n). Then we can plot the time
scaled signal y(n) by substituting different values for n as shown in Fig. 1.27 (b). Here the signal is compressed
by 2.
y ( 0 )= x(0) = 1
y( – 1) = x(–2) = 3
y( – 2) = x(–4) = 0
y ( 1 ) = x(2) = 3
y ( 2 ) = x(4) = 0
and so on.
4 4
3 3 3 3
2 2
1 1
–4 –3 –2 –1 0 1 2 3 4 n –2 –1 0 1 2 n
(a) (b)
4 4
3 3
2 2
1
–7 –6 –5 –4 –3 –2 –1 0 1 2 3 4 5 6 7 n
(c)
Fig. 1.27 : Discrete-time scaling (a) Plot of x(n), (b) Plot of x(2n), (c) Plot of x(n/2)
So to plot x(2n) we have to skip odd numbered samples in x(n).

1 n
If a , y( n ) x , then
2 2
y ( 0 ) = x(0) = 1
y ( 2 ) = x(1) = 2
y ( 4 ) = x(2) = 3
y ( 6 ) = x(3) = 4
y ( 8 ) = x(4) = 0

y(–2) = x(–1) = 2
y(–4) = x(–2) = 3
y(–6) = x(–3) = 4
y(–8) = x(–4) = 0
We can plot y(n) = x(n/2) as shown in Fig. 1.27 (c). Hence the signal is expanded by 2. All odd components
in x(n/2) are zero because x(n) does not have any value in between the sampling instants.
Remember: You don’t have to learn that y(t) = x(t ± a) means right/left shift or y(t) = x(at) mean compression/expansion just
remember the method we used for analysis.
3. Time Reversal:
Suppose x(t) is a given signal let,
y(t) = x(–t)
which means that, y(0) = x(0)
y(1) = x(–1)
y(2) = x(–2)
y(–1) = x(1)
y(–2) = x(2)
so, y(t) is obtained by taking mirror image of x(t) about amplitude axis, for example
x (t ) y (t )
1.0 1.0
t t
–5 –4 5 –5 4 5
Fig. 1.28 : Time reversal, plot of x(t) and x(–t)
Similar for discrete signal, if y[n] = x[–n] then y[n] is obtained by taking mirror image of x[n] about amplitude
axis.
Study Note
• A system that do shifting operation on input signal replaces variable ‘t’ in input by ‘t – a’ or ‘t + a’ to get output. So,
Shifter by
x (t ) y ( t – a)
‘a’
Shifter by
x (2 t) x (2(t – a) = x (2t – 2a)
‘a’
• A system that do scaling operation an input signal replaces variable ‘t’ in input by ‘at’ to get output. So,
Shifter by
x (t ) x (at )
‘a’
Shifter by
x (t – b) x (at – b)
‘a’
• Similarly the system that perform inversion will change the variable ‘t’ in input by ‘–t’. For example if input to system
that perform inversion operation is x(at +b) then output is obtained by replacing ‘t’ by ‘–t’ and we get output as x(–at +
b).

Suppose, in a question we have x(t) and we have to find

(a) x(t + b), then simply apply shifting operation by b units.
(b) x(at + b), here a is positive, then we have to apply shifting by b units to get x(t + b) and then scaling by
a units to get x(at + b).
(c) x(–at + b), here a is negative, first of all we do shifting by b units to get x(t + b), then scaling by a units
to get x(at + b) and then inversion to get x(–at + b).
Example 1.7
Let x(t) is
x (t )
1.0
t
–5 –4 5
Solution 1.7
( a ) To plot x(2t + 5) first of all we plot x(t + 5) [Left shift x(t) by 5]
x (t + 5)
t
–10 –9 0
Now we scale x(t + 5) to get x(2t + 5), here we do compression.
x (2t + 5)
1.0
t
–5 –4.5
( b ) To plot x(–t + 7), so first of all we plot x(t + 7) [Left shift x(t) by 7]
x (t + 7)
1.0
t
–12 –11 –2
Here no scaling required, here we need to do inversion to get x(–t + 7) [Mirror image of x(t + 7)]
x (–t + 7)
1.0
t
2 11 12

REMEMBER To obtain x(–2t + 7) from x(t).
Shifter by Scaling by
x ( t) Inversion x (–2t + 7)
+7 x ( t + 7) 2 x (2t + 7)
(OR)
Scaling by Shifting by
x ( t) Inversion x (–2t + 7)
2 x (2t ) 3.5 x (2(t + 3.5)
= x (2t + 7)
1.7 Energy and Power Signals

Consider v(t) to be the voltage across a resistor R producing a current i(t). The instantaneous power p(t) per
ohm is defined as
v( t ) i( t )
p(t) = i2 ( t ) ...(32)
R
Total energy E and average power P on a per-ohm basis are
E = i 2 ( t ) dt jou les ...(33)
T /2
1
P = Lim i 2 ( t ) dt watts ...(34)
T T T /2
For an arbitrary continuous-time signal x(t), the normalized energy content E of x(t) is defined as
2
E = x ( t ) dt ...(35)
The normalized average power P of x(t) is defined as

T /2
1 2
P = Lim x(t ) dt ...(36)
T T T /2
Similarly, for a discrete-time signal x[n], the normalized energy content E of x[n] is defined as
2
E = x[n ] ...(37)
n
The normalized average P of x[n] is defined as

N
1 2
P = Lim x[n] ...(38)
N 2N 1 n N
Based on definitions (35) to (36), the following classes of signals are defined:
1. x(t) (or x[n]) is said to be an energy signal (or sequence) if and only if 0 < E < , and so P = 0.
2. x(t) (or x[n]) is said to be a power signal (or sequence) if and only if 0 < P < , thus implying that
E= .
3. Signals that satisfy neither property are referred to as neither energy signals nor power signals.
From the above analysis we can see that energy signals are those which have
2
x ( t ) dt or fin ite

or we can say that, x ( t ) dt is finite ...(39)
From equation (39) we can say those signals are energy signals which have area under x ( t ) finite. This is
possible only when
• signal exist only for finite duration of time.
• signal should be a decaying signal if it is of infinite duration.
• signal should not be a periodic signal.
For example:
• x(t) = e–t u(t),
e–t u (t )
1.0
e–t
The signal is of infinite duration and decaying so it is an energy signal with energy,
2 1
E = e t u ( t ) dt e 2t
dt
2
• x(t) = e –t
e–t
1.0
here also e–t is a decaying at one end but at other end it is rising so e–t is not energy signal.
Similarly,
t
x( t ) e x (t )
1.0
Energy signal Energy signal
t t
–5 –4 5
Now lest look at power signal, the power of a signal is defined as

T /2
1 2
P = Lim x ( t ) dt
T T T /2
Always remember that,

1. All periodic signals are power signals. The power of periodic signal is given by
To / 2
1 2
Pavg. = x (t ) dt ...(40)
To To /2
here x(t) is periodic signal with period T0.

2. Signals which remain constant for infinite duration are power signal, for example u(t), 1, all are
power signals.

Study Note
• Signals which decay for infinite duration are energy signal.
• Signals which remain constant for infinite duration are power signal.
• Signals which rises for infinite duration are neither energy nor power signal.
Example 1.8
Determine which of the following signals are energy, power and neither energy non power signals.
(a) x(t) = u(t) (b) x(t) = ejt
t
(c) x(t) = (t 5) (t 7 ) dt (d) x(t) = t u(t)

∞
(E) x[n]= 1- δ[n -1- k]

k =3
Solution 1.8
(a) Power signal as it is constant for infinite duration
T /2 T /2 x ( t)
1 2 1
Pavg. = Lim x ( t ) dt Lim 1 dt
T T T /2
T T 0 1.0
T /2 1 t
Pavg. = Lim 0
T T 2
so, Pavg.= 0.5
(b) This is a signal with x ( t ) is ‘1’ for all ‘t’ so it is constant which give it is a power signal
T /2 T /2
1 2 1
Pavg. = Lim x ( t ) dt Lim 1 dt 1
T T T /2
T T T /2
( c ) The plot of x(t) is

x (t )
–7 5
t
–1
Signal will be energy signal, x ( t ) = [–u(t + 7) +u(t – 5)]

5
2
E = x(t ) dt 1 dt 12
7
( d ) The plot of x(t) is

x ( t)
The signal is rising towards for infinite duration of time and thus it is neither energy non power signal.

(e) x[n] = 1 [n 1 k ]
k 3
so, x[n] = 1 – [ [n – 4] + [n – 5] + [n – 6] + .....]
x[n] can be found by substracting two signal ‘1’ and [n 1 4], that is
k 3
[n 1 k ] x [n ]
k 3
1.0 1.0
1
.....
– ..... = .....
n n n
–3 –2 –1 0 1 2 3 4 5 4 5 6 –3 –2 –1 0 1 2 3 4 5 6
1, n 3
so, x[n] =
0, n 3
x[n] = u[3 – n]
1
so, x[n] is power signal with Pavg. =
2
REMEMBER • If power of x(t) is P then

(i ) Power of Ax(t) will be A2P
(i i ) Power of x(t ± b) will be P, that is even after shifting power of signal remain same.
(i i i ) Power of x(at) will be P, scaling do not affect the power of signal.
( i v ) Power of Ax(at + b) will be A2P
• If energy of x(t) is E then
(i ) Energy of x(t ± b) will be E, shifting do not affect energy.
(i i ) Energy of x(at) will be E/a.
(i i i ) Energy of Ax(t) will be A2 E.
( i v ) Energy of Ax(at ± b) will be A2 E/a.
1.8 Periodic and Non-periodic Signals

A signal which has a definite pattern and which repeats itself at regular intervals of time is called
periodic signal, and a signal which does not repeat at regular intervals of time is called a non-periodic or
aperiodic signal.
x(t + T) = x(t) for all t, i.e. for – < t <
where ‘t’ denotes time and T is a constant representing the period. The smallest value of T which
satisfies the above condition is called the fundamental period or simply period of x(t). A signal is aperiodic if the
above condition is not satisfied even for one value of ‘t’. Sometimes aperiodic signals are said to have a period
equal to infinity.
The reciprocal of fundamental period T is called the fundamental frequency f of x(t).
1
i.e. f =
T
The angular frequency is given by
2
= 2 f
T
2
Fundamental period, T =

Similarly, a discrete time signal x(n) is said to be periodic if it satisfies the condition x(n) = x(n + N) for all
integers n.
The smallest integer value of N which satisfies the above condition is known as fundamental period.If the
above condition is not satisfied even for one value of n, then the discrete-time signal is aperiodic.
2
The angular frequency is given by, =
N
Study Note
The fundamental period T of a signal x(t) can have any value but fundamental period N of signal x[n] has to be an integer.
Also periodic signal exist for – to on time axis.
Now lets look at period of cosine, sine and exponential signals.

Case-1: Continuous Time Signals
• ej 0t : Let the period is T so
ej 0 (t T ) = ej 0t
ej 0T = 1 = e j2n Since fundamental period has to be minimum
2n 2
T =
0 n 1 0
2
• sin( 0t) will also be periodic with period .
0
2
• cos( 0t) will also be periodic with period .
0
Case-2: Discrete-time Signals:

• ej 0n : Let the period is N and N has to be an integer
so, ej 0(n N) = e j 0n
ej 0N = 1 = e j2 m
0N = 2 m
2 m
N = [Here N and m both are integer]
0
N 2
=
m 0
2
Since is irrational and has to be rational as N and m are integers, so e j 0n will periodic if 0 has
0
in it or 0 is equal to ( multiplied by some rational number).

2 m
Thus, Period of e j 0n
is N heree m is minimum integer value to get N as integer.
0
For example:
j10 n
10
1. e 3 is periodic as 0 , 0 has in it, and
3
2 2 3m
N = m m
0 10 3 5

3m
N= and minimum value of ‘m’ to get N as integer is 5 and N = 3.
5
2. e j 5 n is non-periodic as 0 = 5 and have no is 0.
• Period of sin[ 0 n] and cos[ 0n]: Similar to e j 0n , sin[ 0 n] is periodic only if 0 has in if or 0 is (
multiplied by some rational number), the period N will be
2 m
N = [Here, m is smallest integer to ge N as integer]
0
Generally in GATE questions are asked where we have to find the period of signal which is sum of
various other signals. To find out the result we will follow this procedure.
To find out period of:
X(t) = x1(t) + x2(t) + x3(t) + ....
or, X[n] = x1[n] + x2[n] + x3[n] + ....
Step-1: First of all check that, all individual signals are periodic or not i.e.
x1(t), x2(t), x3(t).... are periodic, or
x1[n], x2[n], x3[n].... are periodic
If any one of the individual signal is non periodic then X(t) or X[n] will be non-periodic.
If all individual signals are periodic the move to step-2.
Step-2: Find period of each individual signals, that is
x1(t) T1 x1[n] N1
x2(t) T2 or x2[n] N2
x3(t) T3 x3[n] N3
and so on..
Step-3: Find out these ratio,
T1 T1 T1
, , .... or N1 , N1 , N1 ....
T2 T3 T4 N2 N3 N4
If any of the above ratio is not integer/integer then X(t) or X(n) will be non-period, else goto
step-4.
S t e p - 4 : Find out LCM of denominator of all ratio of step 3. Let LCM is x.
S t e p - 5 : Period of signal X(t) is xT1.
Period of signal X[n] is xN1.
Example 1.9
Find period of these signals:

( a ) X(t) = sin(15t) – 4 cos(7t)
( b ) X( t )
10 sin( 3 t ) 40 cos( 3 t )
( c ) X[n] = cos[13n] + e j5 n
2 3
j n j n
( d ) X[ n ] e 3
e 4
Sampler
( e ) sin(125 t ) find period of output
Ts 0.01 sec .

Solution 1.9
(a) In the given signal both sin(15t) and 4 cos(7t) are periodic.
2
Period of sin(15t) T1
15
2
Period of 4 cos(7t) T2
7
T1 2 /15 7
so, =
T2 2 /17 15
since, ratio is integer/integer so X(t) is periodic. LCM of ratio is 15.
2
x = 15, T0 = xT1 15 2
15
(b) Here, both the individual signals are periodic
2
10 sin(3t) T1
3
2
4 cos( 3 t ) T2
3
T1 2 /3 1 Integer
but ratio of .
T2 2 / 3 3 Integer
So, X(t) is non-periodic.
( c ) Signal cos[13n] is non-periodic so X[n] is non-periodic.
( d ) Here, both the individual signals are periodic.
2
j n 2
Periodic e 3 N1 = m 3m
2 /3
N 1 = 3 m, for m = 1, N = 3
3
j n 2 8
Period of e 4 N2 = m m
3 /4 3
For m = 3, N2 = 8
N1 In teger
The ratio is . So X[n] is periodic.
N2 In teger
N1 3
=
N2 8
So, LCM of denominator is 8.
Thus Period of x[n] = 8 N1 = 24
( e ) The output of the sampler will be x[n ] sin (125 t ) t nTs

sin (1.25 n )
so sin(1.25 n) will be periodic with period
2 m 8
N = m
1.25 5
For m = 5, N = 8
so period of output is 8.

Power of Periodic Signal

The power of periodic signal x(t) is
To / 2
1 2 1 2
Pavg. = x ( t ) dt or x(t ) dt
To To / 2
To t To
Here To is period of the signal x(t). [here t = <To> means integration over period To]
N0 1
1 2 1 2
Pavg. = x [ n] or x[ n ]
N0 n 0 N0 n N0
Here N0 is period of x[n]. [n = <N0> means summation of period N0].

So power of A cos( 0t) will be
T /2
1 2
Pavg. = A cos( 0t ) dt
T T /2
T /2
1 1 cos 2 0t A2
= A2 dt
T T /2
2 2
Similarly power of A sin( 0t) is A2/2.

Now let us find power of signal which is sum of various sinusoidal signals,
x(t) = A1 cos( 1t) + A2 cos( 2t)
T0 / 2
1 2
Pavg. = x ( t ) dt , here T0 is period of x(t)
T T0 / 2
T0 / 2
1
= ( A12 cos 2 ( 1t ) A22 cos 2 ( 2t ) 2 A1 A2 cos( 1t ) cos( 2 t )) dt
T T0 / 2
A12 A22
=
2 2
Pavg. = Pavg 1 Pavg 2
REMEMBER So, general form is, if a signal
x(t) = x1(t) + x2(t) + x3(t) + ....
Here all x1(t), x2(t), x3(t) are sinusoidal signals of different frequencies the
Power of x(t) = Power of x1(t) + Power of x2(t) +......
1.9 Even and Odd Signals

Even (Symmetric) Signal:
A continuous-time signal x(t) is said to be an even (symmetric) signal if it satisfies the condition
x(t) = x(–t) for all t ...(42)
A discrete-time signal x(n) is said to be an even (symmetric) signal if it satisfies the condition
x(n) = x(–n) for all n ...(43)

Even signals are symmetrical about the vertical axis or time origin. Hence they are also called symmetric
signals: cosine wave is an example of an even signal. Since even signals are shown in Fig. 1.29(a).
x (t ) x ( t)
t t
0 0
(a) (c)
x (n ) x (n )
–3 –2 –1
n n
–2 –1 0 1 2 0 1 2 3
(b) (d)
Fig. 1.29 : (a) Even signal (b) Even signal (c) Odd signal (d) Odd signal
Odd (Antisymmetric) Signal

A continuous-time signal x(t) is said to be an odd (antisymmetric) signal if it satisfies the condition,
x( – t) = –x(t) for all t ...(44)
A discrete-time signal x(n) is said to be an odd (antisymmetric) if it satisfies the condition,
x( – n) = –x(n) for all n
Odd signals are anti-symmetrical about the vertical axis. Hence they are called anti-symmetrical signals.
Sine wave is an example of an odd signal. For an odd signal x(t) = 0, x(n) = 0. Some odd signals are shown in
Fig. 1.29 (b).
Any signal x(t) or x[n] can be expressed as a sum of two signals, one of which is even and one of the
which is odd. That is,
x ( t ) = x e (t) +x 0 (t) ...(46)
x[n] = xe[n] + x0[n] ...(47)
1
where, x e( t ) = {x ( t ) x ( t )} even part of x(t) ...(48)
2
1
x e[n] = {x[n ] x[ t ]} even part of x[n] ...(49)
2
1
x0( t ) = {x ( t ) x ( t )} odd part of x(t) ...(50)
2
1
x 0 [n] = {x[n ] x[ n ]} odd part of x[n] ...(51)
2
1.10 Conjugate Symmetric and Conjugate Antisymmetric Signals

Conjugate Symmetric Signal
A signal is conjugate symmetric if
x ( t ) = x(–t) * ...(52)
or, x[n] = x[–n] * ...(53)

Now, lets see few properties of conjugate symmetric signal [Properties valid for both continuous and discrete
time signals].
1. x(t) = x(–t) x(0) = x(0)
This implies that x(t) t=0 will be real if x(t) is conjugate symmetric.
2. Let x(t) which is conjugate symmetric has real and imaginary part a(t) and b(t) respectively
x(t) = a(t) + jb(t)
Now applying conjugate symmetric property
x(t) = x(–t)
a(t) + jb(t) = a(–t) – jb(–t)
so, a(t) = a(–t) and b(t) = –b(–t)
real part of x(t) is even and imaginary part is odd.
Note: For conjugate symmetric signal real part is even and imaginary part is odd.
3. Let x(t) = a(t) + jb(t) and x(t) is conjugate symmetric.

1 b( t )
The magnitude plot M( t ) a( t )2 b( t )2 and phase plot ( t ) tan .
a( t )
M( t) = a( t )2 b( t )2 , M( t ) ( a( t ))2 ( b( t ))2
M(–t) = ( a( t ))2 ( b( t ))2 a( t )2 b( t )2

M(–t) = M( t)
So, magnitude plot is even.
1 b( t )
Now, ( t ) = tan
a( t )
1 b( t )
and (–t) = tan [since imaginary part is odd and real part is even]
a( t )
1 b( t ) 1 b( t )
(–t) = tan tan
a( t ) a( t )
Study Note
Here if a(t) + jb(t) is in first quadrant then a(–t) + jb(–t) will be in fourth quadrant as here imaginary part is
becoming negative and real part remain same. Thus (t) = – (–t).For conjugate symmetric signals phase plot is odd
and magnitude plot is even.
Conjugate Anti-Symmetric
A signal is conjugate antisymmetric if
x ( t ) = – x( t) ...(54)
or, x[n] = –x[–n] ...(55)
Now let us assume x[t] is conjugate antisymmetric and x(t) has real part a(t) and imaginary part b(t).
So, x ( t ) = a(t) + jb(t)
The properties of x(t) will be [valid for discrete signal x[n]]
1. At t = 0 x(0) = –x(0) , so x(0) is imaginary or zero.
2. Since x(t) is conjugate antisymmetric.
So, x ( t ) = –x(–t)

a(t) + jb(t) = –a(–t) + jb(–t)

a(t) = –a(–t), b(t) = b(–t)
So, real part is odd and imaginary part is even.
Note : For conjugate antisymmetric signals real part is odd and imaginary part is even.
3. Now M(t) is magnitude plot then,
M( t) = a( t )2 b( t )2 , M( t ) ( a( t ))2 ( b( t ))2
Using above property, M(–t) = ( a( t ))2 ( b( t ))2 ( a( t ))2 ( b( t ))2

M(–t) = M( t)
So, magnitude plot is even.
Now (t) that is phase plot,
1 b( t )
( t ) = tan
a( t )
1 b( t ) 1 b( t )
and (–t) = tan tan
a( t ) a( t )
There is no relation (t) and (–t). So phase plot is neither even neither odd.
So, for conjugate antisymmetric signals magnitude plot is even and phase plot neither even non-
odd.
Study Note
Here if a(t) + jb(t) is in first quadrant then a(–t) + jb(–t) will be in second quadrant as here imaginary part remain
same and real part is becoming negative.Thus (t) has no relation with (–t).For conjugate antisymmetric signals
magnitude plot is even and phase plot has no relationship.
Any signal x(t) or x[n] can be expressed as sum of two signals, one of which is conjugate symmetric and
other is conjugate antisymmetric.
x ( t ) = x CS (t) +x CAS (t) ...(56)
x[n] = xCS[n] + xCAS[n] ...(57)
x( t ) x( t )
Conjugate symmetric part, xCS [n] = ...(58)
2
x [n ] x[ n ]
xCS [n] = ...(59)
2
x( t ) x( t )
Conjugate antisymmetric part, x CAS (t) = ...(60)
2
x [n ] x [ n ]
xCAS[n] = ...(61)
2
x( t ) x( t )
REMEMBER • Real part of signal is .
2
x ( t ) x( t )
• Imaginary of signal is .
2

Example 1.10
Find conjugate antisymmetric part of x[ n] {1 2 j, 3, 1 j, 4} .
Solution 1.10
x [n ] x [n ]
Conjugate antisymmetric part of x[n] is
2
x[n] = {0, 1 2 j, 3, 1 j, 4}
x[–n] = {4, 1 j , 3, 1 2 j , 0}
x [–n] = {4, 1 j , 3, 1 2 j , 0}
x [n ] x [n ] j j
Now, xCAS[n] = = 2, , 0, , 2
2 2 2
1.11 Causal and Non-Causal Signals

A continuous-time signal x(t) is said to be causal if x(t) = 0 for t < 0, otherwise the signal is non-causal.
A continuous-time signal x(t) is said to be anti-causal if x(t) = 0 for t > 0.
A causal signal does not exist for negative time and an anti-causal signal does not exist for positive time.
A signal which exists in positive as well as negative time is neither causal non anti-causal, it is non-causal. u(t) is a
causal signal and u(–t) is anti-causal signal.
Similarly, a discrete-time signal x(n) is said to be causal if x(n) = 0 for n < 0, anti-causal if x(n) = 0 for n > 0
and if signal is neither causal nor anticausal it is called non causal.
1.12 System and Classification of System

System Representation
A system is a mathematical model of a physical process that relates the input (or excitation) signal to the
output (or response) signal.
Let x and y be the input and output signals, respectively, of a system. Then the system is viewed as a
transformation (or mapping) of x into y. This transformation is represented by the mathematical notation.
y = Tx ...(62)
where, T is the operator representing some well defined rule by which x is transformed into y. Relationship
(62) is depicted as shown in Fig.1.30 (a). Multiple input and/or output signals are possible as shown in
Fig. 1.30 (b). We will restrict our attention for the most part in this text to the single-output, single-output case.
x1 y1
System System
x y
T
xn ym
(a) (b)
Fig. 1.30 : System with single or multiple input and output signals

1.12.1 Classification of System

1. Continuous Time and Discrete Time Systems
If the input and output signals x and y are continuous-time signals, then the system is called continuous-time
system Fig. 1.31 (a). If the input and output signals are discrete-time signals or sequences, then the system is called a
discrete-time system Fig. 1.31 (b)].
System System
x ( t) y (t ) x [n ] y [t ]
T T
(a) (b)
Fig. 1.31 : (a) Continuous-time system, (b) Discrete-time system
2. System with Memory and Without Memory

System is memoryless if output at any time depends on only the input at that same time. These are also
called static systems.
If the output of the system at any time depend on input at any other than that same time then system has
memory and it is called dynamic system.
For example:
• y(t) = x(t) + 5 is static system
• y(t) = x(t – 5) is dynamic system
• y(t) = x(t2) is dynamic system
A relation between output and input is given and we have to find out that the system is
static or dynamic. To find out the type of system we follow this procedure:
In the given relationship between y(t) and x(t) or y[n] and x[n] try to find value of y(t) at t = –2, –1, 0,
1, 2 or value of y[n] at n = –2, –1, 0, 1, 2 if for all these instances the output depend only on x[t] at t = –2, –1, 0, 1, 2
or x[n] at n = –2, –1, 0, 1, 2 respectively. Then system is static else it is dynamic.
For example:
• y(t) = x(t) + cos(t + 1), here
y(0) = x(0) + cos(1)
y(1) = x(1) + cos(2)
y(2) = x(2) + cos(3)
Here present output depend only on present input so it is a static system.
Remember: We have to look only at the input and output terms to find type of system rest all terms should be
neglected as in above example cos(t + 1) can confuse us while finding the type of system.
3. Causal and Non-causal Systems

Causal Systems: Those systems in which present output is dependent only on present and or past
value of input. In other words.
A system is called causal if its output y(t) at an arbitrary time t = t0 depends on only the input x(t) for
t t0. That is, the output of a causal system at the present time depends on only the present and/or past values
of the input, not on its future values. Thus, in a causal system, it is not possible to obtain an output before an
input is applied to the system.
Non-causal Systems: Those systems in which present output depend on future value of inputs.
Note: Causal systems are also called non-anticipatory system and all physically realisable systems are causal systems.

A relation between output and input is given and we have to find out that the system is causal of
non-causal. To find out the type of system we follow this procedure:
A relationship between y(t) and x(t) or y[n] and x[n] is given. Now find out y(t) or y[n] for t = –2, –1, 0, 1, 2
or n = –2, –1, 0, 1, 2 and check if in any of the case the present value of output depend on future value of input then
system is non-causal otherwise it is causal.
Study Note
• Always check for all values of t or n given in above method for correct answer.
• A non-causal system is always dynamic.
• Static is causal but not vice-versa.
4. Linear and Non-linear System

A system is linear if it follow super-position that is if the input to the system is weighted sum of several
signals then the output is equal to weighted sum of response of system to each of the signal.
That is,
If x1(t) is input and y1(t) is output.
If x2(t) is input and y2(t) is output.
Then for ax 1 (t) + by 1 (t) output should be ay 1(t) + by 2 (t).
Then only system is linear.
To find out the given system is linear or non-linear we follow this procedure.
S t e p - 1 : First of all from the given y(t) – x(t) or y[n] – x[n] relation find out relationship between
output-input. This step will clear by going through the given examples.
S t e p - 2 : Now apply input x1(t) or x1[n] and using relationship of step-1 find output y1(t) or y1[n].
S t e p - 3 : Now apply x2(t) or x2[n] and using relationship of step-1 find output y2(t) or y2[n].
S t e p - 4 : Now apply ax 1(t) + bx 2(t) or ax 1[n] + by 1[n] and find output y(t), if y(t) = ay 1 (t) + by 2 (t)
then system is linear else it is non-linear.
For example:
• System is defined as, y(t) = x(t) –x(t – 1), to find linearity. So,
Step-1: We can see that input is x(t) and output is y(t), this means.
Output(t) = [Input(t) – Input(t – 1)]
S t e p - 2 : For x 1(t) as input, output y 1(t) is
y 1 ( t ) = x 1 (t) – x 1(t – 1)
Step-3: For x 2 (t) as input, output y 2 (t) is
y 2 ( t ) = x 2 (t) – x 1(t – 1)
Step-4: Now input is ax1(t) + bx2(t) so output will be Input(t)-Input(t – 1), that is
y ( t ) = ax 1 (t) + bx 2 (t) – (ax 1 (t – 1) + bx 2 (t – 1))
= a(x 1 (t) – x 1 (t – 1)) + b(x 2 (t) – x 2 (t – 1))
= ay 1 (t) + by 2 (t)
So, system is linear.
• System is defined as y(t) = x(t2), to find linearity:
S t e p - 1 : Output (t) = Input(t2), that is system replaces t by t2 in input.
S t e p - 2 : For input x 1 (t), output y 1 (t) is y 1 (t) = x 1 (t2 ).
S t e p - 3 : Similarly y 2 (t) = x 2 (t 2 ).

Step-4: For input ax1(t) + bx2(t), using step-1

Output (t) = Input(t2), so y(t) is
y(t) = ax1(t2) + bx2(t2)
y(t) = ay1(t) + by2(t)
So, system is linear.
• System is defined as y(t) = x(t) .
Step-1: Output (t) = Input(t) , output is modulus of input.
Step-2: y1(t) = x1( t )
Step-3: y2 (t ) x2 (t )
Step-4: Here input is ax1(t) + bx2(t) so,
y(t) = ax1( t ) bx 2 ( t )
ay1(t) + by2(t)
So, system is non-linear.
• System is defined as y[n] = x[n]
S t e p - 1 : Output[n] = (Input[n])*, output is conjugate of input
S t e p - 2 : y 1 [n] = (x 1 [n])*
S t e p - 3 : y 2 [n] = (x 2 [n])*
S t e p - 4 : Here input is ax1[n] + bx2[n] and output will be
y[n] = (ax 1 [n 1 ] + bx 2 [n])*
= a (x 1 [n])* + b (x 2 [n])*
ay 1 [n] + by 2 [n]
System is non-linear.
5. Time Variant and Time Invariant System

A system is time invariant if a time shift in the input signal result is an identical time shift in the output
signal. That is for a time invariant system if x(t) is input and y(t) is output then for x(t – T0) the output should
be y(t – T0).
To determine that the given system is time invariant or not we will check that if output for x(t – T0) is
equal to y(t – T0) if yes then system is time invariant otherwise system is time invariant.
Procedure that should be followed to determine that system is Time Invariant or not is
S t e p - 1 : From the given relation between y(t) – x(t) or y[n] – x[n] find out relation between output and
input.
S t e p - 2 : Now find expression of y(t – T0) or y[n – m0] by replacing t by t – T0 or n by n – m0 in the
expression given in question.
S t e p - 3 : Now apply x(t – T0) or x[n – m0] as input to system and use relation found in step-1 to find
output of this input, let output is y (t) or y [n]. If y (t) = y(t – T0) or y [n] = y[n – m0] then
system is time invariant else system is time invariant.
For example:
• System is defined as y(t) = x(t) + x(t – 5):
S t e p - 1 :Output (t) = Input(t) + Input(t – 5), system add input expression with another expression
obtained by replacing t by t – 5 in input.
S t e p - 2 :y(t – T0) = x(t – T0) + x(t – T0 – 5)

Step-3: Now input x(t) = x(t – T0), using relation of step-1 we can find output for x(t – T0),
y (t) = x(t – T0) + x(t – T0 – 5)
Since, y (t) = y(t – T0), system is time invariant.
• System is defined as y(t) = x(2t), to find that system is time variant or invariant.
Step-1: Output = Input(2t), to get output system replaces t by 2t in input.
Step-2: y(t – T0) = x(2(t – T0))
= x(2t – 2T0)
Step-3: Here input is x(t – T0), now using relation of step-1. We can say that output is
y (t) = x(2t – T0)
y (t) y(t – T0)
System is time invariant.
• System is defined as y(t) = x(–t) then for checking time invariant or not:
Step-1: Output (t) = Input (–t), system replace t by –t in input to get output.
Step-2: y(t – T0) = x(–(t – T0)) = x(–t + T0)
Step-3: Now input is x(t – T0) then output y (t) using step-1 is
y (t) = x(–t – T0)
y (t) y(t – T0)
• System is defined as y(t) = cos(t) x(t) then for checking time invariant or not:
Step-1: Output(t) = Input(t) cos(t), system multiply cos(t) with input to get output.
Step-2: y(t – T0) = cos(t – T0) x(t – T0)
Step-3: Now input is x(t – T0) then using step-1 output is
y (t) = cos(t) x(t – T0)
y (t) y(t – T0)
Shortcut to find that given system is time variant or time invariant is
This is valid for both discrete and continuous time system. There are three form of systems that are time
variant.
1. If the independent variable in input and output is same in the given relation then system is time
invariant else it is variant.
For example:
( a ) y( t ) x( t ) x( t 5) , so time invariant
variable variable variable
t t t
(b) y( t ) x ( t 2 ) , so time variant

variable variable
t t2
(c) y( t ) x ( t 10) , so time variant

variable variable
t t
(d) y( t ) x( 2 t 5) , so time variant

variable variable
t 2t
(e) y( t ) x ( t ) , so time variant

variable variable
t t

2. If the input-output relationship has output equal to input multiplied with any other time dependent
signal, then system is time variant.
For example:
( a ) y(t) = cos(t) x(t) is time variant.
( b ) y[n] = nx[n] is time variant.
3 . The system which is defined such that input-output relation is break relation then system is time
variant.
For example:
x( t ) t 0
( a ) y( t )
x( t ) t 0
In pu t( t ) t 0
Step-1 : Output(t) =
In pu t( t) t < 0
x ( t T0 ), t T0 0
Step-2: y(t – T 0 ) =
x ( ( t T0 )), t T0 0
Step-3: Now input is x(t – T0), so output y (t) will be
x ( t T0 ), t 0
y (t) =
x ( t T0 ), t 0
so, y (t) y(t), system is time variant.
x [n 1] , n 1
( b ) y[ n ] , will also be time variant.
x [n 5] , n 1
so, all break functions are time variant.
Remember: In 2nd point of shortcut the time dependent signal should not be input itself it has to be any other time
dependent signal, because y(t) = x(t).x(t) = x2(t) is time invariant.
6. Stable and Unstable System

The stability of a system is the BIBO stability i.e. bonded input and bounded output stability. The stability
of system means that if bounded input is applied at input then output is also bounded.
Bounded input means magnitude of input at any time that is In pu t ( t ) or Inpu t ( n ) M where M is any
finite value and bounded output means magnitude of output at any time that is Ou tpu t ( t ) or Output [ n] N
where N is any finite value.
To find out that the given system is stable or not we will follow this procedure:
Apply input x(t) = u(t) or x[n] = u[n] and find out output, if output is bounded then system is stable
otherwise it is unstable.
For example:
• y[n] = x[–n], now we apply x[n] = u[n], so y[n] = u[–n] so system is stable as output is bounded as
value of output is always 1.
n
• y[n] x[m], here if x[n] = u[n] then y[n] = r[n]. The output is unbounded so system is unstable.
n

n n0
• y[n] x[k ], here if input x[n] = u[n] then
k n n0
n n0
y[n] = u [k ] r[n r0 ] r[n n0 ]
k n n0
y[n] = (n + n0) u(n + n0) – (n – n0) u(n – n0)

so, value of y[n] is always 2n0, so system is stable.
• y(t) = cos(x(t)), this is simple as y(t) will always have value between –1 and 1 so it is BIBO stable.
• y(t) = tx(t) + 5, if x(t) = u(t);
y(t) = tu(t) + 5, output is not bounded, system in unstable.
7. Invertible and Non-Invertible System

Suppose we have a system which take in input x(t) and produces y(t) output if from the output y(t) the whole
input x(t) can be determined then system is invertible.
The invertible system are those:
(i) Which produces distinct output for distinct input.
(ii) All the input elements must have distinct output that is information of all input elements should exist at
output, then only we can find input from output.
From example:
• y(t) = 2x(t) This system is invertible, simply divide output by 2 to get input.
• y[n] = nx[n] Here we have distinct output for distinct input and all input elements are
present at output except x[0] as y[n] = 0 x[0] = 0. So, no information of x[0]
at output, so whole input cannot be determined. System is non-invertible.
t
• y( t ) x( ) d invertible system, we can get back signal by differentiation.
dx(t )
• y (t ) is non invertible because if x(t) is a constant then output is zero then we
dt
cannot get back x(t) from output.
• y(t) = x(2t) Invertible, output is contracted version of x(t), we can simply get back
input by expanding output.
t
• y( t ) x Invertible, output is expanded version of x(t), we can get back input by
2
contracting output.
• y[n] = x[2n] Non-invertible, here odd components of input x[1], x[3], x[5] are lost and
cannot be obtained from output.
x[n / 2] , n even
• y[ n ] Invertible as output is obtained by adding a zero between two samples of
0, n odd
input. No information lost.
8. Recursive and Non-Recursive System

The system that have feedback or present output depends on input values and several past output
values is recursive system. The system in which present output depends only several input values is non-
recursive system.

REMEMBER Two signals x1(t) and x2(t) can be said to be orthogonal over a time interval (t0, t0 + ) if
t0
x1 ( t ) x 2 ( t ) dt = 0
t0
Sometimes the system is defined by difference equation and differential equation. The continuous time
system are defined using differential equation then such system are always dynamic and we cannot find that
system is causal or non-causal, stable or unstable we will use Laplace transform for this, but we can find that
system is linear or non-linear and time invariant or time variant.
To find that given system defined by differential equation are linear or non-linear and time variant or
time invariant we will follow this process
The system will be linear if the given differential equation is linear differential equation that is there is no
product of input x(t) with itself or with it’s derivative and no product of output y(t) with itself or it’s derivative then
the system is linear and also there should be non constant term in either side of differential equation.
To check whether the system is time variant or time invariant. For time invariant we need that the coefficient
of terms x(t) and y(t) and their derivative should not be time dependent term and the dependent variable in x(t) and
it’s derivative should be same as that of dependent variable in y(t) and it’s derivative. If these conditions are followed
then system is time invariant else time variant.
For example:
d 2 y( t )
1. 5 y( t ) = x(t) + 7
dt 2
d 2 y( t )
The coefficient of x(t), y(t) and are constant and dependent variable is same ‘t’ in all terms
dt 2
so it is a time invariant system.
But in right side of equation we have a constant term 7 thus system is non-linear.
t dy( t ) 3 dx ( t )
2. 7 y( t ) = s x ( t )
dt dt
dy ( t )
The coefficient of is ‘t’ thus system is time variant and their is no constant terms and all
dt
dy( t ) dx ( t )
y(t), , x ( t ) and are sitting alone thus system is linear. We can check linearity by checking
dt dt
superposition also.
2
dy( t ) d 2 y( t )
3. y( t ) 6 y( t ) = 7 x ( t )
dt dt 2
Obviously the system is non-linear and system is time invariant because all dependent variable are
same and coefficient of all terms are not time dependent.
dy (t )
4. y (t 2 ) = 7x(t)
dt
The system is linear but due to, difference in variable the system is time variant. (t2 is variable in
left side of equation).

dy( t )
5. y( t ) = x(t)
dt
System is neither linear nor time invariant.
dy( t )
6. cos( x ( t )) = x(t)
dt
System is non-linear and time invariant.
The discrete time system are defined by difference equations and to find that system is causal or not we will
use Z-transform, but to find that system is time invariant or not and linear or non-linear we can use the same method
we have used in continuous time system defined by differentiation.
For example:
1. x(n) + x 2 (n) y(n) = y(n – 1)
Here, y(n) and x(n) terms are not sitting alone so system is non linear and coefficient in all terms is time
independent term and variable used is same in all terms so it is time invariant.
2. x(n) +x(n + 3) = y(n + 7) + y(n – 8)
Clearly system is linear and time invariant.
3. x(n 2 ) + y(n) y(n – 1) = n x ( n )
Clearly system is non-linear and time variant.
4. x(n) + 6y(n – 1) = 7y(n) + 8y(n)
System is linear and time variant.
Example 1.11
Find the properties system:

( a ) System is defined as y(t) = cos(10t) x(t).
n 3
( b ) System is defined as, y [ n ] x[ m ].

m
Solution 1.11
( a )• Since present value of output depend on present value of input so system is static and causal.
• To check linearity: Output = cos(10t) Input
So, output for x1(t) is y 1 ( t ) = cos(10t) x 1 (t)
output for x2(t) is y 2 ( t ) = cos(10t) x 2 (t)
Now, when input is ax1(t) + bx2(t) then output
y ( t ) = cos(10t) (ax 1 (t) + bx 2 (t))
= ay 1 (t) + by 2 (t)
so system is linear.
• Since output is equal to input multiplied with time dependent signal so it is time invariant.
• y(t) = cos(10t) x(t), if input is u(t) then output is bounded so system is stable.
3
• System is non-invertible because output will be zero at t , ,.... and are will have no
2 2
information of x( t ) t ,
3
....
is output.
2 2
• Clearly system is non-recursive.

3
(b) • Since y[0] x[m ], so y[0] depend in input from – to 3 so system is non-static or dynamic and
m
since present output depend on future inputs so system is non-causal.

• Since summation, differentiation, integration, difference all are linear operation, so system is linear.
• Now for check of time invariance, when input is x[n] then output is
n 3
y[n] = x[m ],
m
n 3
Output[n] = In pu t [m ]
m
n m0 3
so, y[n – m0] = x [m ]
m
Now when input is x[n – m0] then expression of output can be obtained using output-input equation,
n 3 n m0 3
y [n] = x [m m0 ] x[ ]
m
since, y [n] = y[n – m0]

So system is time invariant.
• For checking stability, let input is x[n] = u[n],
n 3
y[n] = u [m ] r [n 3]
Since output is not bounded output. So system is unstable.

• The system will be invertible because output contain information of all component of inputs,
n 3 n 2
y[n] = x [m ], y[n 1] x[m ]
so, y[n] = {x[n + 3] + x[n + 2] +x[n + 1] +.... x[– ]

and y[n – 1] = {x(n + 2] + x[n + 1] + .... x[– ]
so, x[n + 3] = y[n] – y[n – 1]
or, x[n] = y[n – 3] – y[n – 4]
we can find input back from output.
Example 1.12
Evaluate the following integrals:

at 2
(a) e (t 5) dt (b) t 2 δ(t - 6) dt
0
3
(c) ( t ) sin 5 t dt (d) δ(t + 2) e-2t dt

0 -
(e) (t 2 )3 ( t 2 ) dt (f) (t) e j t

dt
(g) [ ( t ) cos 2 t (t 2 ) sin 2 t ] dt

Solution 1.12
(a) Given: at 2
e (t 5) dt
1 for t = 5
We know that, (t – 5) =
0 elsewh ere
at 2 at 2 25 a
e (t 5) dt = e e
t 5
( b ) Given: t 2 ( t 6 ) dt
0
1 for t = 6
0 elsewh ere
t2 (t 6) dt = [t2]t = 6 = 36
0
3
( c ) Given: ( t ) sin 5 t dt
0
1 for t = 0
We know that, ( t) =
0 elsewh ere
3
( t ) sin 5 t dt = [sin 5 t]t = 0 = 0
0
2t
( d ) Given: (t 2) e dt
1 for t = 2
We know that, (t + 2) =
0 elsewh ere
(t 2) e 2t
dt = [e –2t] t = –2 = e4
( e ) Given: (t 2)3 ( t 2) dt
1 for t = 2
0 elsewh ere
(t 2)3 ( t 2) dt = [(t – 2)3]t = 2 = 0
j t
( f ) Given: (t) e dt
1 for t = 0
We know that, ( t) =
0 elsewh ere
(t) e j t
dt = [e–j t]t = 0 = 1

(g) Given: [ ( t ) cos 2 t (t 2) sin 2 t ] dt
1 for t = 0 1 for t =2
We know that, ( t) = and (t 2)
0 elsewh ere 0 elsewhere
[ ( t ) cos 2 t (t 2) sin 2 t ] dt = [cos2t]t = 0 + [sin2t]t = 2 = 1 + sin 4
Example 1.13
Find the following summations:
(a) e3 n ( n 3) (b) (n 2) cos 3 n

n n
(c) n2 ( n 4) (d) (n 2) e n
n n
(e) (n 1) 4 n
n 0
Solution 1.13
(a) Given: e3 n ( n 3)
n
1 for n = 3
We know that, (n – 3) =
0 elsewh ere
e3 n ( n 3) = [e 3n ] n = 3 =e 9
n
( b ) Given: (n 2) cos 3 n
n
1 for n = 2
0 elsewh ere
(n 2) cos 3 n = [cos3n] n = 2 = cos6

n
( c ) Given: n2 ( n 4)
n
1 for n = 4
We know that, (n + 4) =
0 elsewh ere
n2 ( n 4) = [n2]n = –4 = 16
n

2
(d) Given: (n 2) e n
n
1 for n = 2
0 elsewh ere
2 2 2
(n 2) e n = [e n ]n 2 e2 e4
n
( e ) Given: ( n 1) 4 n
n 0
1 for n = 1
We know that, (n + 1) =
0 for n 1
( n 1) 4 n = 0
n 0
Example 1.14
Evaluate the following integral:
2t
(a) e ( t ) dt
4
1 1
(b) (t 2 )2 t dt
4 3 2
1
2t
(c) te (t 1) dt
4
Solution 1.14
( a ) Consider the given integral,
2t d 2 2t
e ( t ) dt = e ( t ) dt 2e 2
dt t 0
( b ) Consider the given integral,

4 4
1 1 1 3
(t 2 )2 t dt = (t 2)2 t dt
4
3 2 4
3 2
4
3
= 3( t 2)2 t dt
4
2
4
d 3
= 3 (t 2 )2 t dt 6(t 2) t 3 3
4
dt 2 2
( c ) Consider the given integral,

1
2t 2 d2 2t 2t
te (t 1) dt = ( 1) (t e ) ( 4t 4) e 0
4
dt 2 t 1 t 1

Example 1.15
Examine whether the following signals are periodic or not? If periodic determine the fundamental period.
(a) sin12 t (b) ej4 t
t
( c ) sin t u(t) (d) e
( e ) cos 2 t sin 3 t (f) 3 sin200 t + 4 cos100t

( g ) sin10 t + cos20 t (h) sin(10t + 1) –2 cos(5t – 2)]
( i ) j e j6t (j ) 3 u(t) + 2 sin2t
( k ) 6 e j[4t + /3)] + 8 e j[3 t + ( /4) ]
Soluton 1.15
( a ) Given, x ( t ) = sin12 t
Comparing it with t, we have
2 2 1
= 12 or T
12 6
Thus, x(t) is periodic with fundamental period T = 1/6 sec.
( b ) Given, x ( t ) = e j4 t
j t
Comparing it with e , we have
2 2 1
= 4 or T
4 2
Thus, x(t) is periodic with fundamental period T = 1/2 sec.
( c ) Given, x ( t ) = sin t u(t)
2 2
sin t is periodic with period, T 2 sec. u (t ) exists only between t = 0 to t = . Hence it is not
periodic.
Therefore, sin t u(t) which is the product of a periodic and a non-periodic signal is not periodic.
t
( d ) Given, x(t) = e
t
The plot of x(t) = e versus t is shown in figure. It does not repeat at all. So it is not periodic.
x (t ) = e– t
t
0
( e ) Given, x ( t ) = cos 2t sin 3t

Let, x ( t ) = x 1 (t) + x 2 (t)
where, x 1 ( t ) = cos2t and x2 (t ) sin 3 t

comparing, x(t) = cos2t with cos 1t, we have
1
1 = 2 i.e., 2 f1 = 2 or f1

1
Period of x1(t) is T1 =
f1
Comparing x 2 ( t ) sin 3 t with sin 2t, we have

3
2 = 3 i.e., 2 f 2 3 or f2
2
1 2
f2 3
T1 3
The ratio of two periods =
T2 2 /3 2
Since, T1/T2 is not a ratio of two integers (i.e. not a rational number), the given signal x(t) is non-
periodic.
( f ) Given, x ( t ) = 3 sin200 t + 4 cos100t
Let, x ( t ) = x 1 (t) + x 2 (t)
where, x 1 ( t ) = 3 sin200 t and x 2 (t) = 4 cos100t
Comparing x1(t) = 3 sin200 t with A sin 1t, we have
1 = 200 , i.e., 2 f1 = 200 or f1 = 100
1 1
f1 100
Comparing x2(t) = 4 cos100t with B cos 2t, we have
50
2 = 100 i.e., 2 f2 = 100 or f 2
T1 1 /100 1
T2 / 50 2
Since, T1/T2 is not a ratio of two integers (i.e. not a rational number), the given signal x(t) is non-
periodic.
( g ) Given, x(t) = sin10 t + cos200 t
Let, x(t) = x 1 (t) + x 2 (t)
where, x1( t ) = sin10 t and x 2 (t) = cos20 t
Comparing x2(t) = cos20 t with cos 2t, we have
2 = 20 i.e., 2 f2 = 20 or f2 = 10
1 1
f2 10
T1 1/5
The ratio of two periods = 2
T2 1 /10
T 1 = 2T 2
Since, T1/T2 is a rational number (ratio of two integers 2 and 1), the given signal x(t) is periodic.
1
The fundamental period, T = T1 2 T2 sec .
5
( h ) Given, x ( t ) = sin(10t + 1) – 2 cos(5t – 2)
Let, x ( t ) = x 1 (t) + x 2 (t)
where, x 1 ( t ) = sin(10t + 1) and x 2(t) = 2 cos(5t – 2)

Comparing x1(t) = sin(10t + 1) with sin( 1t + 1), we have

10 5
1 = 10 i.e., 2 f 1 = 10 or f1
2
The period of x1(t) is, T1 =

5
Comparing x2(t) = 2 cos(5t – 2) with A cos( 2t + 2), we have
5
2 = 5 i.e., 2 f2 = 5 or f 2
2
1 2
The period of x2(t) is, T2 = sec .
f2 5
T1 /5 1
T2 2 /5 2
2 T1 = T2
Since, T1/T2 is a ratio of two integers (i.e. a rational number), the given signal x(t) is periodic,
2
Fundamental period, T = 2 T1 T2 sec .
5
( i ) Given, x(t) = j e j 6 t
Comparing this with the standard form x(t) = e j t, we have
3
= 6 i.e. 2 f = 6 or f
1
Time period, T =
f 3
x(t) is periodic with period /3.
( j ) Given, x ( t ) = 3 u(t) + 2 sin2t
signal 3 u(t) is non periodic. Therefore, x(t), is non periodic.
( k ) Given, x ( t ) = 6ej[4t + /3)] + 8e j[3 t + ( /4)]
Let, x ( t ) = x 1 (t) + x 2 (t)
where, x 1 ( t ) = 6e j[4t + /3)] and x 2(t) = 8e j[3 t + ( /4)]
Comparing x1(t) = 6ej[4t + /3)] with A e j( 1 1) , we have

2
1 = 4 i.e. 2 f1 = 4 or f1
1
Time period of 6ej[4t + /3)] is T1
f1 2
Comparing x2(t) = 8e j[3 t + ( /4)] with B e j( 2)

2 , we have
3
2 = 3 i.e. 2 f2 = 3 or f 2
2
t + ( /4)] 1 2
Time period of 8e j[3 is T2
f2 3
T1 /2 3
T2 2 /3 4
Since the ratio T1/T2 is not rational, the given signal x(t) is not periodic.

Example 1.16
Determine the power and rms value of the following signals:
( a ) x( t ) 7 cos 20 t ( b ) x( t ) 12 cos 20 t 16 sin 30 t

2 3 2
( c ) x(t) = 8 cos4t cos6t ( d ) x(t) = e j2t cos10t
( e ) x(t) = A e j5t
Solution: 1.16
( a ) Given, x ( t ) = 7 cos 20 t
2
It is of the form A cos( 0t + ).
72
Then, power of the signal, P = 24.5 W
2
The rms value of the signal is 24.5 .
( b ) Given, x ( t ) = 12 cos 20 t 16 sin 30 t

3 2
2 2
12 16
Then power of the signal, P = 200 W
2 2
and Rms value of the signal = 200
( c ) Given, x ( t ) = 8 cos4t cos6t
[cos10 t cos 2 t ]
= 8 4 cos10 t 4 cos 2 t
2
42 42
Then power of the signal, P = 16 W
2 2
The rms value of the signal is 16 4.
( d ) Given, x ( t ) = e j2t cos10t
= (cos2t + j sin2t) cos10t
= cos2t cos10t + j sin2t cos10t
(cos12 t cos 8 t ) j (sin 12 t sin 8t )
=
2 2
2 2 2 2
1 1 1 1
2 2 2 2 1
Then, Power of signal =
2 2 2 2 2
1
and Rms value =
2
( e ) Given, x ( t ) = A e j5t
= A cos5t + jA sin5t
A2 A2
Then, Power of the signal = A2
2 2
and Rms value of the signal = A2 A

Example 1.17
Find the even and odd components of x(t) = ejt.

Soluton 1.17
Let xe(t) and x0(t) be the even and odd components of ejt, respectively,
e j t = x e (t) + x 0 (t)
From equation using Euler’s formula, we obtain
1 jt jt
x e( t ) = (e e ) cos t
2
1 jt jt
x0( t ) = (e e ) j sin t
2
Example 1.18
Consider the capacitor shown in figure. Let input x(t) = i(t) and output y(t) = v c (t).
+
i( t) C vc(t )
–
( a ) Find the input-output relationship.

( b ) Determine whether the system is
( i ) memory less ( i i ) c a usa l
( i i i ) lin ear ( i v ) time-invariant
( v ) st a ble
Solution 1.18
( a ) Assume the capacitance C is constant. The output voltage y(t) across the capacitor and the input
current x(t) are related by equation,
t
y ( t ) = T {x ( t )} 1
x ( ) dt
C
( b ) ( i ) From equation it is seen that the output y(t) depends on the past and the present values of the
input. Thus, the system is not memoryless.
( i i ) Since the output y(t) does not depends on the future values of the input, the system is causal.
( i i i )Let x(t) = 1 x 1 (t) + 2 x 2 (t).
Then, y ( t ) = T{x(t)}
t
1
= [ 1 x1( ) 2 x 2 ( )] d
C
t t
1 1
= 1 x1( ) d 2 x2( ) d
C C
= 1 y 1 (t) + 2 y 2 (t)
Thus, the superposition property is satisfied and the system is linear.
( i v ) Let y1(t) be the output produced by the shifted input current x1(t) = x(t – t0).

t
1
Then, y1(t) = T {x ( t t 0 )} x( t0 ) d
C
t t0
1
= x( ) d y( t t0 )
C
Hence, the system is time-invariant.
( v ) Let, x(t) = k1 u(t), with k1 0. r( t) = tu (t )
t
1
Then, y(t) = k1u ( ) d
C
t
k1 k1 k1
= d t u( t ) r( t )
C 0
C C
0
where, r(t) = t u(t) is known as the unit ramp function.
Since y(t) grows linearly in time without bound, the system is not BIBO stable.
Example 1.19
Consider an energy signal x(t) whose even and odd part is x e (t) and x o (t) then prove that,
E x ( t ) = E xe (t) + E x0 (t)
Solution 1.19
We know that, any signal x(t) can be written as
x ( t ) = x o (t) + x e (t)
2
Energy of x(t) is, Ex( t) = x ( t ) dt
2 2
Ex ( t ) x ( t ) dt = xo (t ) xe (t ) dt
2 2 2
= xo (t ) dt xe (t ) dt 2 xo (t ) xe (t ) dt
The signal (xo(t))2, (xe(t))2 will be even but xo(t) xe(t) will be odd and area under odd signal is zero. So,
2 2
Ex( t) = xo (t ) dt xe (t ) dt = E x 0 ( t ) Ex e ( t )

6 . Consider a discrete-time system with input x[n] and

output y[n]. The input-output relationship for this
system is
y[n] = x[n] x[n – 2]
Subjective Practice Problems
(a) Is the system memory less?
(b) Is the system invertible?
1 . Determine whether or not each of the following 7 . A continuous-time linear system S with input x(t)
signals is periodic: and output y(t) yields the following input-output
(a) x1(t) = 2e j(t + /4) u(t) pairs:
(b) x2[n] = u[n] + u[–n]
x( t) = e j 2 t S
y( t ) e j3t
(c) x 3 [n ] { [n 4k ] [ n 1 4 k ]}
k
j2t S j3 t
x( t) = e y( t ) e
2 . Determine whether or not each of the following (a) If x1(t) = cos(2t), determine the corresponding
signals is periodic. If a signal is periodic, specify its
output y1(t) for system S.
fundamental period.
(a) x 1 (t) = je 10t 1
(b) x 2 (t) = e –(1 + j)t (b) If x2(t ) cos 2 t determine the
2
(c) x3[n] = e j7 n
(d) x4[n] = e j3 (n + 1/2)/5 corresponding output y2(t) system S.
(e) x5[n] = 3e j3/5(n + 1/2)
8 . Determine whether or not each of the following
3 . Consider the discrete-time signal, continuous-time signals is periodic. If the signal is
periodic, determine its fundamental period.
x[n] = 1 [n 1 k ]
k 3
(a) x ( t ) 3 cos 4t
Determine the values of the integers M and n0 so 3
that x[n] may be expressed as (b) x(t) = e j( t – 1)
x[n] = u[Mn – n0] 2
4 . Consider the continuous-time signal (c) x ( t ) cos 2 t

3
x( t) = (t + 2) – (t – 2)
(d) x(t) = {cos(4 t) u(t)}
Calculate the value of E for the signal,
(e) x(t) = {sin(4 t) u(t)}
t
y(t) = x( ) d ( 2t n )
(f) x ( t ) e u (2 t n)
n
5 . Consider the periodic signal,

9 . In this chapter, we introduced a number of general
1, 0 t 1 properties of systems. In particular, a system may or
x( t) =
2, 1 t 2 may not be
with period T = 2. The derivative of this signal is (1) Memory less (2) Time invariant
related to the ‘impulse train’ (3) Linear (4) Causal
(5) Stable
g( t ) = (t 2 k ) Determine which of these properties hold and which
k
do not hold for each of the following continuous-
with period T = 2. It can be shown that, time systems. In each example, y(t) denotes the
d x( t ) system output and x(t) is the system input.
= A1 g(t – t1) + A2 g(t – t2) (a) y(t) = x(t – 2) + x(2 – t)
dt
Determine the values of A1, t1, A2 and t2. (b) y(t) = [cos (3t)] x(t)

2t dx ( t )
(c) y( t ) x( ) d (j) y( t )
dt
x[n 1], n 0
0, t 0 (k) y[n ]
(d) y( t ) x[n ], n 1
x( t ) x( t 2), t 0
(l) y(t) = x(2t)
0, x( t ) 0 (m) y[n] =x[2n]
(e) y( t )
x ( t ) x( t 2), x ( t ) 0
x[n / 2], n even
(n) y[n ]
t 0, n odd
(f) y( t ) x
3
1 2 . Let x(t) be a continuous-time signal, and let
dx ( t )
(g) y( t ) t
dt y1 ( t ) x ( 2 t ) an d y 2 ( t ) x
2
1 0 . Determine the properties of the following discrete- The signal y1(t) represent a speeded up version of
time systems. x(t) in the sense that the duration of the signal is
(a) y[n] = x[–n] cut in half. Similarly, y2(t) represents a slowed down
(b) y[n] = x[n – 2] –2x[n – 8] version of x(t) in the sense that the duration of the
(c) y[n] = nx[n] signal is doubled. Consider the following statements:
(d) y[n] = {x[n – 1]} (1) If x(t) is periodic, then y1(t) is periodic.
(2) If y1(t) is periodic, then x(t) is periodic.
x[n ], n 1
(3) If x(t) is periodic, then y2(t) is periodic.
(e) y[n ] 0, n 0
(4) If y2(t) is periodic, then x(t) is periodic.
x [n 1], n 1 For each of these statements, determine whether it
is true, and if so, determine the relationship between
x[n ], n 1
the fundamental periods of the two signals considered
(f) y[ n ] 0, n 0 in the statement. If the statement is not true, produce
x [n ], n 1 a counter example to it.
(g) y[n] = x[4n +1] 1 3 . Let x[n] be a discrete-time signal, and let
1 1 . Determine whether system is invertible or not x[n / 2], n even
(a) y(n) = x(t – 4) y1 [ n ] x[2 n ] an d y 2 [t ]
0, n odd
(b) y(t) = cos[x(t)]
The signals y1[n] and y2[n] respectively in some sence
(c) y[n] = nx[n]
the speeded up slowed down versions of x[n].
t
(d) y( t ) x( ) d However, it should be noted that the discrete-time

notions of speeded up and slowed down have subtle
differences with respect to the continuous-time
x[n 1], n 1
counter parts. Consider the following statements:
(e) y[n ] 0, n 0
(a) If x[n] is periodic, then y1[n] is periodic.
x[n ], n 1 (b) If y1[n] is periodic, then x[n] is periodic.
(f) y[n] = x[n] x[n – 1] (c) If x[n] is periodic, then y2[n] is periodic.
(g) y[n] = x[1 – n] (d) If y2[n] is periodic, then x[n] is periodic.
t For each of these statements, determine whether it
(t )
(h) y( t ) e x( ) d is true, and if so, determine the relationship between
the fundamental periods of the two signals considered
n k
n
1 in the statement. If the statements is not true,
(i) y[ n ] x[k ]
k 2 produce a counter example to it.

1 4 . Consider a system S with input x[n] and output y[n] 1 6 . (a) Consider a time-invariant system with input x(t)
related by: and output y(t). Show that if x(t) is periodic with
y[n] = x[n] {g[n] + g[n – 1]} period T, then so is y(t). Show that the analogous
(a) If g[n] = 1 for all n, show that S is time invariant. result also holds in discrete-time.
(b) If g[n] = n, show that S is not time invariant. (b) Give an example of a time-invariant system and
(c) If g[n] = 1 + (–1)n, show that S is time invariant. a non-periodic input signal x(t) such that the
corresponding output y(t) is periodic.
1 5 . (a) Is the following statement true or false?
The series interconnection of two linear time- 1 7 . Find whether the following signal is energy or power
invariant systems is itself a linear time-invariant or neither energy nor power signal
system. (a) x(t ) e 2t u (t )
(b) Is the following statement true of false?
j (2t )t
The series interconnection of two non-linear (b) x(t ) e u(t )
systems is itself non-linear. 2
1
(c) Consider three systems with the following input- (c) x[n] u[n]
2
output relationships:
x [n / 2], n even (d) x[n] 3 n u[n]
System-1: y[ n ]
0, n odd (e) x[n] u[n] u[ n]
t
1 1 (f) x(t ) e u (t )
System-2: y[ n ] x[n ] x [n 1] x[n 2]
2 4
System-3: y[n] = x[2n] 1 8 . Find the value of these signals at the given vaue of t
Suppose that these systems are connected in series 10t 43 3t 68
as depicted in figure. Find the input-output (a) x(t ) rect 20tri at t = –5
50 100
relationship for the overall interconnected system.
Is this system linear? Is it time invariant? 5t 30 8t 40
(b) x(t ) 30rect 10tri at t = 5
System System System 100 160
x [n] y [n ]
1 2 3

Chapter
Linear Time Invariant

System 2
Introduction
Two most important attributes of systems are linearity and time-invariance. In this chapter we will develop
the fundamental input-output relationship for systems having these attributes. It will be shown that the input-output
relationship for LTI system is described in terms of a convolution operation. The importance of the convolution
operation in LTI systems steps from the fact that knowledge of the response of an LTI system to the unit impulse
input allows us to find its output to any input signals.
2.1 Response of Continuous Time LTI System and Convolution Integral

We assume that the system is continuous time and linear time invariant. That is the system follows superposition
and time invariance property. First of all let us try to write any arbitrary continuous time signal in terms of impulse
( (t)) function.
Let us take any signal x(t) and practical delta function p(t),
x (t )
p(t)
1/
t t
– /2 /2
Fig. 2.1 : Arbitrary signal x(t) and practical delta function p(t)
Now, x(t) p(t) is x(0) p(t). Plot of x(0) p(t) is shown in Fig. 2.2(a) and plot of x(0) p(t) is shown in
Fig. 2.2(b). Similarly Fig. 2.2(c) and Fig.2.2 (d) show x(t) p(t – ) i.e. x( ) p(t – ) and x( ) p(t – ).

x(0) p( t) x(0) p(t )
x(0)/ x(0)
t t
– /2 /2
(a) (b)
x( ) p( t – ) x( ) p( t – )
x( )/ x( )
t t
(c) (d)
Fig. 2.2 : Multiplication of x(t) with practical impulse function
Now, we form a new signal x ( t ) :
x( t ) = [ x(k ) p (t k )] ...(1)
k
we can see that x ( t ) is staircase approximation of x(t) of Fig. 2.1.

x~ (t )
t
5 3 0 3 5 7
2 2 2 2 2 2 2
Fig. 2.3 : Plot of x ( t )
Now as 0 the staircase approximation x ( t ) will become x(t),
Lim x ( t ) = x(t) ...(2)

0
Since, 0 will make input practical delta function into ideal delta function and summation of equation (1)
will change to integration. Thus,
x(t) = x( ) ( t )d ...(3)
The above equation show how any time domain signal can be written in terms of impulse function.
Remember: We can intuitively think that x(t ) as a sum of weighted shifted impulses, where the weight on the impulse
(t ) is x( )d

Linear Time Invariant System 65
Response of Continuous Time LTI System for any Arbtrary Input

Suppose any LTI system in given, we first of all find the impulse response of system. That is at input (t) is
applied and output is the impulse response h(t).
LTI
(t ) h( t)
system
Impulse Impulse
input response
Fig. 2.4 : Impulse response of system
Sy stem
(t) h( t )
Since, the system is time invariant so.
System
(t ) h( t )
Since system is linear so if we multiply input with a constant then output also get multiplied by same
constant.
Sy stem
x( ) ( t ) x( ) h( t )
Since, integration is a linear operation applying this operation at input and output side.
Sy stem
x( ) ( t )d x ( ) h( t )d
Sy stem
x( t ) x ( ) h( t )d
So output, y(t) = x ( ) h( t )d ...(4)
Equation (4) is referred as the convolution integral or superposition integral and symbolically
represented as,
y ( t ) = x(t) h(t) ...(5)
Thus, in case of any linear time invariant system we can find output for any arbitrary input x(t) by
convolution of x(t) with impulse response of system h(t).
Procedure that should be followed to find out convolution of two signals:
Suppose, y ( t ) = x ( t ) h( t ) x ( ) h( t )d
Signals x(t) and h(t) are given and we have to find y(t), or we have to compute the above integral. Here
the integration variable is ‘ ’ and ‘t’ is a constant.
S t e p - 1 : Change variable ‘t’ by ‘ ’ in x(t) to get x( ).
S t e p - 2 : From h(t) find h( ) by changing variable ‘t’ by ‘ ’ then plot h(– ) by taking mirror image of h( )
from amplitude axis. To get h(t – ) we have to do the required shifting in h(– ).
S t e p - 3 : To find y(t) multiply x( ) and h(t – ) and find, x ( ) h( t )d .
Lets look an example, x ( t ) = e –t u(t),

h(t) = u(t)
and y ( t ) = x(t) h(t)

First of all we plot x(t) and h(t),
x(t) h(t)
1.0
e –t
1.0
t t
0 0
Now to find x( ) and h( ),we will simply change t by ,
x( ) h( )
1.0
e–
1.0
0 0
Now h(– ) will be inversion of h( )
h( – )
1.0
Now to find h(t – ) we have to shift h(– ). For t > 0 the h(– ) will shift right by t units.
h( t – )
1.0
(for t > 0)
For t < 0 the h(– ) will shift left by t units.

h( t – )
1.0 (for t > 0)
t 0

Now to find y(t) we have to multiply x( ) and h(t – ),

Case-1: For t > 0 the plot of x( ) and h(t – ).
x( ) h(t – )
1
–
–
1.0 e
e
× =
t 0 t
t
For t > 0, y(t) = e d
0
= (1 – e–t)
Case-2: For t < 0, plot of x( ) and h(t – ) are
x( ) h( t – )
1.0
–
e
× = 0
y(t) = 0 for t < 0
(1 e t ), t 0
so, y(t) =
0, t 0
= (1 – e–t) u(t)
Example 2.1
Find the convolution of two continuous time signals,

x ( t ) = e – t and h(t) = e –t u(t – 1)
Solution 2.1
We have to find y(t) = x(t) h(t). First let us drawn x(t) and h(t),
x (t )
h( t)
1.0
–t –t –t
e e e
t t
1.0
The plot of x( ) and h( ) is

x( )
h( )
1.0
– –
e e e
1.0

The plot of h(– ) is

h(– )
–1.0
Plot of h(t – ) will be shifted h(– ) by ‘t’, if t is positive h(– ) shift right by ‘t’ units and if t is negative h(– ) shift
left by ‘t’ units. Another way is to use equation to visualise, h(t) = e t u(t 1)
(t ) (t )
(t ) e t 0 e t
Thus h(t )= e u (t )
0 t 0 0 t
These two figures show two cases when t > 1 and t < 1.
h(t – ) h(t – )
e–(t – )
e–(t – )
( t – 1) ( t – 1)
SInce y(t) is convolution of x(t) and h(t),
y(t) = x ( ) h( t ), putting x( ) we get
= e h(t )d e h(t )d
0
Analysing the plot of h(t – ) we can see that if t < 1 then h(t – ) = 0 for t > 0. So for t < 1
0
y(t) = e h( t )d 0
For t < 1, h(t – ) = e–(t – ) for <t– 1

t 1 t 1
(t ) t
y(t) = e e d e e2 d
e( t 2)
y(t) = for t 1
2
Now for t > 1, h(t – ) = e–(t – ) for < t– 1
0 t 1
(t ) (t )
y(t) = e e dt e e dt
0
0 t 1
t et
= e e2 dt e td = e t ( t 1)
0 2
et
y(t) = e t ( t 1) for t 1
2

Example 2.2
Consider a LTI system with impulse response h(t) and input x(t), the output of system is y(t). The value of
output at t = 5 is
x (t ) h( t)
1.0
2.0
t t
–5 –4 4 5 0 10
Solution 2.2
Since y(t) is output,
y ( t ) = x(t) h(t)
y(t) = x( ) h ( t )d
Now, y( t ) t 5 = x ( ) h(5 )d
Now we plot x(t) and h(5 – ),

x( ) h(5 – )
1.0 2.0
–5 –4 4 5 –5 5
So, y(5) = x ( ) h(5 )d
= Area of x(t) h(5 – t)

x ( ) h(5 – )
2.0
–5 –4 4 5
So, y(5) = 18
Example 2.3
Find the convolution of a rectangular signal x(t) with itself.

x(t) = A rect(t/T)
Solution 2.3
We have to find y(t), y(t) = x(t) x(t)
y(t) = x ( ) x( t )d

x( ) x( – )
–T/2 T/2 –T/2 T/2
The plot of x(t – ) will be, for t > 0 the signal x(– ) will move right side by t units
So, when t > 0
x(t – )
y(t) = x ( ) x( t )d
T T
A,
we can seen that, x( ) = 2 2
t – /2 t + /2
0 else wh ere
T T
A, t t
and x(t – ) = 2 2
0 else wh ere
T /2
y(t) = A2 d
t T /2
y ( t ) = A2 (T – t)
So, y(t) = A2 (T – t) when t > 0 and t < T
because overlap between x( ) and x(t – ) exist for t < T and for t > T the overlap of x( ) and x(t – ) will
be zero and their product will also be zero thus y(t) = 0 for t > T.
Now when t is negative then x(– ) will move to left by t units, so plot of x(t – ) for t < 0 is
x(t – )
t – /2 t + /2
T T
A,
Now, x( ) = 2 2
0, else wh ere
T T
A, t t
and x(t – ) = 2 2
0, else wh ere
t T /2
y(t) = x( ) x( t )d = A2
T /2
y(t) = A2(t
+ T) for t < 0 and t > – T
because for t < –T overlap between x( ) and x(t – ) do not exist and y(t) = 0 for t < –T.

The plot of y(t) is
2
AT
–T T
Thus, A rect (t/T) A rect (t/T) = A2T (t/T)
Study Notes
Generally in GATE exam whenever we have to find convolution of two signals we will not use the method shown above
because this is a lenghty method, it is better to use laplace transform to do the same. We will learn laplace transform in
Chapter 4. The above method will be helpful in those questions where x(t) is input to an LTI system with impulse response
h(t) and we have to find value of output at a particular time instant as shown in Example 2.2.
2.2 Properties of Convolution Integral

Continuous-time convolution satisfied the following properties:
2.2.1 Commutative Property

The convolution operation possesses the commutative property, or
x(t) h(t) = h(t) x(t) ...(6)
This property (see Fig. 2.5), implies that the role of the input signal and impulse response are interchangeable.
It is often used to simplify the evaluation or interpretation of the convolution integral.
x (t ) h( t) y ( t) = h( t ) x (t ) y ( t)
Fig. 2.5
Proof: By definition,
x(t) h(t) = x ( ) h( t )d
If we perform a change of variables by letting (t – ) = , then = (t – ), d = –d , as – , and

– as . Therefore,
x(t) h(t) = x(t ) h( ) d h( ) x(t )d h(t ) x(t )
Study Notes
Suppose, y(t) = x(t) h(t),
to find the result we can perform,
x(t) h(t) = x ( ) h( t )d or h(t) x(t) = h( ) x(t )d
If we have signal x(t) simpler than h(t) then we perform h(t) x(t) and if h(t) is simpler then we perform
x(t) h(t).

For example if h(t) = e–t u(t) and x(t) = u(t)

then to find y(t) it will be easier to perform,
h(t) x(t) = h( ) x(t )d
and if x( t) = e –t u(t) and h(t ) u (t )

then to find y(t) it will be easier to perform,
x(t) h(t) = x ( ) h( t )d
2.2.2 Associative Property

The continuous convolution is associative, i.e.,
x(t) [h1(t) h2(t)] = [x(t) h1(t)] h2(t) ...(7)
Associativity, as shown in Fig. 2.6 implies that a cascade combination of LTI systems can be replaced by a
single system whose impulse response is the convolution of the individual impulse responses.
x ( t) h1 ( t ) h2 ( t ) y (t ) x (t ) h1 ( t ) h2 ( t ) y (t )
Fig. 2.6
Since continuous convolution is associative, there is no need to indicate which convolution is to be

performed first, hence the square brackets can be removed from equation (7). Since continuous convolution is
also commutative, then
x(t) h 1 (t) h 2 (t) = x(t) h 2 (t) h 1 (t) = ... = h 2 (t) h 1 (t) x(t)
2.2.3 Distributive Property

The convolution operation possesses the distribution property, or
x(t) [h 1 (t) + h 2 (t)] = x(t) h 1 (t) + x(t) h 2 (t) ...(8)
Distributivity, as shown in Fig. 2.7 states that a parallel combination of LTI systems is equivalent to a
single system whose impulse response is the sum of the individual impulse responses in the parallel configuration.
h1(t)
x(t) y(t) x(t) h1(t) + h2(t) y(t)
h2(t)
Fig. 2.7
2.2.4 Shift Property

According to the shift property,
if x(t) h(t) = y( t) ,
then, x(t) h(t – t0) = y(t – t0) ...(9)
and x(t – t1) h(t – t2) = y(t – t1 – t2) ...(10)

x(t) h(t) = x ( ) h( t )d y( t )
x(t) h (t – t0) = x ( ) h( t t 0 )d y( t t 0 )
Equation (10) and follows from equation (9).
2.2.5 Convolution with an Impulse

The convolution of a signal x(t) with a unit impulse function results in the signal x(t) itself
x(t) (t) = x(t) ...(11)
x(t) (t) = x( ) ( t )d
From the sampling property of the impulse function, we have

x( ) (t – ) = x(t) (t – )
and therefore, x(t) (t) = x( ) ( t )d x( t ) (t )d x( t )
similarly we can prove that,

x(t) (t – t0) = x(t – t0)
1 x( t )
x(t) (at) = x ( t ) (t)
a a
b
x t
1 b a
x(t) (at + b) = x ( t ) (t )
a a a
Study Notes
To perform the convolution of a signal with delta function, we should follow these steps
1
• First of all find the area and location of delta function and write delta function as ( t -location) .
area
• Now convolution of signal x(t) with given delta fuction is found by replacing variable ‘t’ in x (t) by (t - location) and
divide expression by area of delta function.
2.2.6 Width Property

The width of the non-zero extent (the interval of time between the first and last non-zero values) of the
continuous convolution of two functions equals the sum of the widths of the non-zero extents of the two functions.
In other words, if the durations (widths) of x(t) and h(t) are finite, given by Wx and Wh, respectively, then the
duration (width) of x(t) h(t) is Wx + Wh.

Study Notes
Suppose y(t) = x(t) h(t), signal x(t) is non-zero for TxL < t < TxH and h(t) is non-zero for ThL < t < ThH then y(t) will be non-
zero only for ThL + TxL < t < TxH + ThH ,and width of x(t) and h(t) is Wx = TxH TxL and Wh = ThH ThL.
2.2.7 Differentiation Property

According to the differentiation property,
If, x(t) h(t) = y(t)
d d d
then, x( t ) h( t ) = x ( t ) h( t ) y( t ) ...(12)
dt dt dt
Proof : By definition,
y(t) = x ( t ) h( t ) x ( ) h( t )d
By differentiating both sides of the above equation w.r.t. t, we get
d d d
y( t ) = x( ) h( t ) d x( t ) h( t )
dt dt dt
Similarly we can prove that
d2 d 2 x( t ) d 2 h( t ) dx ( t ) dh( t )
2
y( t ) = h( t ) x( t )
2
dt dt dt 2 dt dt
2.2.8 Time-Scaling Property

According to the time-scaling property,
if, x(t) h(t) = y(t),
1
then, x(at) h(at) = y (at )
a
This property states that if both x(t) and h(t) are time-scaled by a, their convolution is also time-scaled by a
and multiplied by 1/[a].
Proof : Case-I: a> 0
By definition, x(t) h(t) = x ( ) h( t )d y( t )
x(at) h(at) = x ( a ) h( a( t )) d x ( a ) h( at a )d
A change of variables is performed by letting a = , which also yields,

1
d = d , as , and as
a
1 1
Therefore, x(at) h(at) = x ( ) h( at d y( at )
a a

Case-II: a < 0
By definition, x(t) h(t) = x ( ) h( t d y( t )
x(–at) h(–at) = x ( a ) h( a( t d x ( a ) h( at a d
A change of variables is performed by letting –a = , which also yields,

1
d = d , as – and – as
a
1 1
Therefore, x(at) h(at) = x ( ) h( at )d y( t )
a a
From the above two cases it is evident that,
1
x(at) h(at) = y( at )
a
REMEMBER • The convolution of an odd and even function is an odd function.

• The convolution of an even and even function is an even function.
• The convolution of an odd and odd function an even function.
• The convolution has the property that the area of convolution integral is equal to product of
area of two signals which are convolved. That is if y(t) = x(t) h(t), then Ay = Ax Ah
d
• x( t ) (t) x (t)
dt
• u(t) u(t) = r(t)
t
• x( t ) u ( t ) x( ) d
Example 2.4
Determine and sketch the convolution of the following signals:

1, 1 t 1
x(t) =
0 , else where
and h(t) = (t + 1) + 2 (t + 2)
Solution 2.4
Using the result,
y(t) = x(t) h(t) = x(t) [ (t + 1) + 2 (t + 2)]
= [x(t) (t + 1)] + [x(t) 2 (t + 2)]
= x(t + 1) + 2x(t + 2)]
= x(t + 1) + 2x(t + 2)

The signals x(t), h(t), and y(t) are
y ( t)
3
h( t)
x (t ) 2 2
1
1 1
t t t
–1 0 1 –2 –1 0 –3 –2 –1 0
Example 2.5
Consider an LTI system with input and output related through the following equations:
t
y(t) = e (t )
x( 2) d
What is the impulse response h(t) for this system?

Solution 2.5
t
(t )
Given that, y(t) = e x( 2) d
A change of variables is performed by letting = ( – 2), which also yields d =d , – as – ,

and (t – 2) as t. Therefore,
t 2
y(t) = e (t 2 )
x( ) d e (t 2 )
u( t 2 ) x( ) d
Comparing with, y(t) = h( t ) x( ) d
we get, h(t – ) = e–(t – 2 – ) u(t – 2 – )

Therefore, the impulse response of the system is
h ( t ) = e–(t – 2) u(t – 2)
Example 2.6
What is (t – 0.5) (t – 0.5)?

Solution 2.6
Let, y ( t ) = (t – 0.5) (t – 0.5)
If we assume that x ( t ) = (t – 0.5)
Thus y ( t ) = x(t) (t – 0.5)
= x(t – 0.5)
so, y ( t ) = (t – 0.5 – 0.5)
= (t – 1)

Study Notes
The step response s(t) of a continuous-time LTI system is defined as the response of the system when the input is u(t).
In many applications, the step response s(t) is also a useful characterization of the system. The step response s(t) can be
easily determined by equation, that is
t
s(t) = h(t) u(t) = h( ) u ( t )d h( t ) d
Thus, the step response s(t) can be obtained by integrating the impulse response h(t). Differentiating with respect to t, we
get
d s( t )
h(t) = s ( t )
dt
Thus, the impulse response h(t) can be determined by differentiating the step response s(t).
2.3 Interconnection of LTI System
2.3.1 Series or Cascade Connection of LTI System

Two LTI systems are connected in cascade and they have impulse response h1(t) and h2(t).
x ( t) h1 ( t ) h2 ( t ) y (t ) x (t ) h( t ) y (t )
Fig. 2.8 : Series or Cascade connection

y(t) = x(t) h1(t) h2(t) ...(14)
y(t) = x(t) h(t)
So, this cascade can be replaced by a single system with impulse response h(t),
h(t) = h1(t) h2(t) = h2(t) h1(t) ...(15)
That is systems can interchange location also.
2.3.2 Parallel Connection

The two systems having impulse response h1(t) and h2(t) are connected in parallel as shown in figure.
We can see that,
y(t) = x(t) h1(t) ± x(t) h2(t)
= x(t) (h1(t) ± h2(t))
= x(t) h(t) ...(16)
h1( t)
+
x ( t) y (t ) x (t ) h1(t ) ± h2( t) y (t )
–
h2( t)
Fig. 2.9 : Parallel connection of two LTI system
So, parallel connection can be replaced by single system with impulse response,
h(t) = h1(t) ± h2(t)

2.4 Response of Discrete Time LTI System and Convolution Sum

In this section we assume that system is discrete and linear and time invariant. Signal x[n] is any arbitrary
signal and we can write it as a sum of shifted and scaled unit impulses. For example:
x[ 1], n 1
x[–1] [n + 1] =
0, n 1
x[0], n 0
x[0] [n] =
0, n 0
x[1], n 1
x[1] [n – 1] =
0, n 1
More generally, by including additional shifted, scaled impulses, we can write
x[n] = ... + x[–3] [n + 3] + x[–2] [n + 2] + x[–1] [n + 1] + x[0] [n]
+ x[1] [n – 1] + x[2] [n – 2] + x[3] [n – 3] + .... ...(17)
For any value of n, only one of the terms on the right-hand side of equation (17) is non-zero, and the scaling
associated with that term is precisely x[n]. Writing this summation in a more compact form, we have,
x[n] = x[k ] [n k ] ...(18)

k
This corresponds to the representation of an arbitrary sequence as a linear combination of shifted unit
impulses [n – k], where the weights in this linear combination are x[k].
Response of Discrete-time LTI System for any Arbitrary Input:

Suppose a discrete-time LTI system is given and impulse response of system h[n] is also given (that is h[n] is
output when [n] is input)
LTI
[n ] Impulse response
system
h[n ]
Fig. 2.10 : Impulse response of system

Sy stem
[n ] h[n ],
Since system is time invariant, so
System
[n m] h[n m]
Since system is linear so if input is multiplied with a constant then output get multiplied by same constant.
Sy stem
x[m ] [n m] x[m ] h[n m]
Since summation is a linear operation, applying summation on both side we get,
Sy stem
x[ m ] [ n m] x[m ] h[n m]
m m
Sy stem
x[n ] x[m ] h[n m]
m
Thus, output is y[n] = x[m ] h[n m] ...(19)

m

Equation (19) is referred as the convolution sum or superposition sum and symbolically represented as
y[n] = x[n] h[n] = x[m ] h[n m] ...(20)

m
Thus in case of any linear time invariant we can find output for any arbitrary input x[n] by convolution of
x[n] with impulse response of system h[n].
Procedure that should be followed to find out convolution of two signals:
Suppose, y[n] = x[n] h[n] = x[m ] h[n m]

m
Signal x[n] and h[n] are given and we have to find y[n] or we have to compute above summation. In this
summation variable is ‘m’ and ‘n’ is a constant. So we will follow this procedure.
S t e p - 1 : Change variable ‘n’ to ‘m’ in x[n] and h[n] to get x[m] and h[m].
S t e p - 2 : Now we get h[–m] by inversion of h[m]. To get h[n – m] we have to do shifting in h[–m].
S t e p - 3 : To find y[n] we multiply x[m] and h[n – m] and find x[m ] h[n m].
m
Let’s look at an example,

x[n] = (0.5)n u[n] and h[n] = u[n]
we have to find, y[n] = x[n] h[n]
First of all we plot x[n] and h[n]
x [n ] h[n ]
1.0 1.0
0.5 .....
.....
n n
0 1 2
Now to find x[m] we simply change variable n by m.
(0.5) m , m 0
So, x[m] = (0.5) m u [m ]
0, m 0
1, m 0
and for h[m] = u [ m ]
0, m 0
Now, h[–m] = u[–m]
1, n m 0 1, m n
Thus, h[n – m] = u[n – m] = =
0, n m 0 0, m n
For n 0,
h[n – m]
1.0
..... ...
m
–4 –3 –2 –1 0 1 2 n

and for n < 0

h[n – m ]
Now, y[n] = x[m ] h[n m]

m 1.0
.....
For n > 0, using definition of x[m], h[n – m] we get n
m
n
((0.5) n 1 1)
y[n] = (0.5) m 1
0 (0.5) 1
y[n] = 2[1 – (0.5)n + 1] for n 0

For n < 0 we can see that y[n] = 0 because for n < 0 h[n – m] is zero for m > 0 and x[m] h[n – m] will be zero.
Thus, y[n] = 2(1 – (0.5)n + 1)
Study Note
The discrete-time LTI system which have impulse response h[n] non-zero only for finite value of ‘n’ is called finite impulse
response filter on FIR filter. If impulse h[n] is non-zero for infinite values of ‘n’ then it is called infinite impulse response
filter IIR filter.
Example 2.7
Find the convolution of two sequences:
0, n < -5 0, n 3
n n
x[n] = 1 and h[n] = 1
, n -5 , n 3
≥
2 3
Solution 2.7
So we have to find, y[n] = x[n] h[n]
= x[m ] h[ n m]
m
0, m 5
we can define, x[m] = 1
m
, m 5
2
0, n m 3 0, m n 3
and h[n – m] = 1
n m
1
n m
, n m 3 , m n 3
3 3
Now, using above definition we can say that, product x[m] h[n – m] will be non zero for –5 m n – 3,
n 3 m n m
1 1
y[n] = x[m ] h[n m]
m m 5 2 3
and for this it is required that (n – 3) –5 that is n –2.

n 3 n m
1 3
For n –2, y[n] =
5 3 2

n 3 n 3
3 3
n 5
1 1
1 3 2 2
5
1
n 2
= =
3 2 3 3 3 1
1
2 2
5 n n 3
2 1 3
So, for n –2, y[n] = 2 1
3 3 2
and for n –2, y[n] = 0
Example 2.8
Find convolution of rectangular signal with itself.

n
x[ n] = rect
2N
Solution 2.8
That is we have to find out, y[n] = x[n] x[n]
n 1, N n N
x[n] = rect
2N 0, else wh ere
Since, y[n] = x[m ] x[n m]

m
m 1, N m N
Since, x[m] = rect
2N 0, else where
1, N n m N
and x[n – m] =
0, else wh ere
1, N n m N n
=
0, else wh ere
when n 0 that is when n is positive then,
N
y[n] = x[m ] x[n m] 1 (2 N n)
m N n
and for n > 2N, overlap between x[m] and x[n – m] is 0 so,
y[n] = 0 for n > 2N
Now for n < 0, using above definition we get,
N n
y[n] = x[m] x[n m] 1 2N n
m N
and y[n] = 0 for n < –2N as overlap between x[m] and x[n – m] is zero.
So, y[n] = 2N – m 0 n 2N
y[n] = 2N + n –2N n < 0

Example 2.9
Compute the convolution y(n) = x(n) h(n) of the following pairs of signals:
( a ) x(n) = h(n) = u(n)
( b ) x(n) = (0.8) n u(n) and h(n) = (0.4) n u(n)
( c ) x(n) = h(n) = a n u(n)
( d ) x(n) = u(n – 1) and h(n) = n u(n – 1)
( e ) x(n) = r(n) = nu(n) and h(n) = a –n u(n – 1), where a < 1
Solution 2.9
By definition:
(a) y ( n ) = x(n) h(n)
= x ( k ) h( n k) u( k ) u ( n k)
k k
The lower limit on the convolution sum simplifies to k = 0 [because u(k) = 0, k < 0], the upper limit
to k = n [because u(n – k) = 0, k > n], and we get
n
y(n) = 1 ( n 1) u ( n ) r( n 1)
k 0
( b ) By definition, y ( n ) = x(n) h(n)
= x ( k ) h( n k) (0.8)k u ( k ) (0.4) n k
u( n k)
k k
to k = n [because u(n – k) = 0, k < n], and we get
n n
y(n) = (0.8)k (0.4) n k
(0.4) n ( 2)k
k 0 k 0
n
n
1 2n 1
= (0.4) 2k (0.4) n (0.4) n ( 2 n 1
1)
k 0 1 2
= [2(0.8) n – (0.4) n ] u(n)

( c ) By definition, y ( n ) = x(n) h(n)
= x ( k ) h( n k) a k u( k ) a n k
u( n k)
k k
to k = n [because u(n – k) = 0, k > n], and we get
n n
y(n) = ak a n k
an 1 ( n 1) a n u ( n )
k 0 k 0
1, n 1
( d ) Given that, x ( n ) = u ( n 1)
0, n 1
n
n , n 1
and h(n) = u (n 1)
0, n 1

By definition, y(n) = x ( n ) h( n ) x ( m ) h( n m)
m
1, m 1
Here x ( m ) = u ( m 1)
0, m 1
n m n m
n m , n m 1 , m n 1
and h(n – m) = u( n m 1)
0, n m 1 0, m n 1
using the above definitions we can see that, x(m) h(n – m) will be zero for m < 1 and m > n 1, and non
zero for m n 1 and m 1 ,thus n has to be greater than equal to 2
n 1 n 1
and y(n) = x ( m ) h( n m) = an m
m 1 m 1
n 1 n 2 n 2 n
n m n ( m 1) n 1 (m)
= a
m 1 m 0 m 0 1
n
, n 2
Therefore, y(n) = 1
0, n 0
n
or, more compactly, y(n) = u( n 2)
1
n, n 0
( e ) Given that, x ( n ) = n u( n )
0, n 0
n a n, n 1
and h(n) = a u ( n 1)
0, n 1
By definition, y ( n ) = x ( n ) h( n ) x ( k ) h( n k)
k
k, k 0
Since, x ( k ) = k u( k )
0, k 0
(n k) (n k)
(n k) a , n k 1 a , k n 1
and h(n – k) = a u( n k 1)
0, n k 1 0, k n 1
Thus product x(k) h(n – k) will be non zero for k n 1 and k 0 , that is 0 k n – 1 and n 1. Thus
n 1 n 1
(n k)
y(n) = x ( k ) h( n k) = ka ;
k 0 k 0
n 1 n 1
n a
= a k ak [1 n a n 1
( n 1)a n ]
k 0 (1 a )2
n 1
a
[1 n a n 1
( n 1) a n ], n 1
Therefore, y(n) = (1 a ) 2
0; n 1
n 1
a
or, more compactly, y(n) = 2
[1 n a n 1
( n 1) a n ] u ( n 1)
(1 a )

Example 2.10
Consider a LTI system with impulse response h[n] = u[n] and input x[n] then, output y[n] will be _____.
Solution 2.10
Since, y[n] = x[n] h[n]
= x[m ] h[n m]
m
Since, h[n] = u[n]

1, n m 0
h[n – m] = u[n – m] =
0, n m 0
1, m 0
h[n – m] =
0, m 0
n
Thus, y[n] = x [m ]
m
Example 2.11
Fin d:
( a ) x[n] [n – n 0 ]
( b ) x[n] [an]
( c ) x[n] [an + b]
Solution
n
(a) x[n] [n – n0] = x[m ] [n m n0 ]
m
Since, x[n] = x[ m ] [n m]
m
So, we can say that,

x[n] [n – n0] = x[n – n0]
( b ) x[n] [an], since [an] = [n]
so, x[n] [an] = x[n]
( c ) x[n] [an + b], first of all we find [an + b]
b
[an + b] = n if b/a is integer and
a
if b/a is not integer then [an + b] = 0.
b
So, if b/a is integer then x[n] [an + b] = x n .
a
Otherwise, x[n] [an + b] = 0.

2.5 Properties of Convolution Sum

The structure of the convolution sum is similar to that of the convolution integral. Moreover, the properties
of the convolution sum are similar to those of the convolution integral. We shall enumerate these properties
here without proofs. The proofs are similar to those for the convolution integral.
2.5.1 Commulative Property

The convolution sum possesses the commutative property:
x(n) h(n) = h(n) x(n) ...(21)
Study Notes
Suppose, y(n) = x(n) h(n),
to find the result we can perform,
x(n) h(n) = x ( k ) h( n k)
k
or h(n) x(n) = h( k ) x ( n k)
k
If we have signal x(n) simpler than h(n) then we perform h(n) x(n) and if h(n) is simpler then we perform
x(n) h(n).
For example if, h(n) = an u(n) and x(n) = u(n)
then to find y(n) it will be easier to perform,
h(n) x(n) = h( k ) x ( n k)
k
and if, x(n) = an u(n) and h(n) = u(n)

then to find y(n) it will be easier to perform,
x(n) h(n) = x ( k ) h( n k)
k
2.5.2 Associative Property

The convolution sum possesses the associative property:
x(n) [h1(n) h2(n)] = [x(n) h1(n)] h2(n) ...(22)
2.5.3 Distributive Property

The convolution sum possess the distributive property:
x(n) [h1(n) + h2(n)] = [x(n) h1(n)] + [x(n) h2(n)] ...(23)
2.5.4 Shifting Property

According to the shift property:
If, x(n) h(n) = y(n) ...(24)
then, x(n) h(n – n0) = x(n – n0) h(n) = y(n – n0)
and x(n – n1) h(n – n2) = y(n – n1 – n2) ...(25)

2.5.5 Convolution with an Impulse

Convolution of a signal x(n) with a unit impulse function results in the signal x(n) itself:
x(n) (n) = x(n) ...(26)
Similarly we can prove that,
x(n) (n – n0) = x(n no)
Study Notes
To perform the convolution of a signal with delta function, we should follow these steps
• First of all find the location of delta function if location is an integer then move to step 2 else answer is zero.
• Now convolution of signal x(n) with given delta fuction is found by replacing variable ‘n’ in x(n) by
(n – location), where location is location of the delta function.
2.5.6 Width Property

Suppose two signals x[n] and h[n] are convolved to get y[n] then, if the location of first non-zero sequence
value (Left edge) of x[n] and h[n] is NLx and NLh and location of last non-zero sequence value (Right edges) of x[n]
and h[n] is NHx and NHh. Then
Number of samples of x[n], length of x[n] = Lx = NHx – NLx + 1
Number of samples of h[n], length of h[n] = Lh = NHh – NLh + 1
Then length of y[n], Ly = Lx + Lh – 1
Also, first non-zero sequence value (left edge) of y[n] is NLx + NLh and right edge of yh[n] is NHx + NHh. Thus
length of convolution of two signals is equal to sum of length of two signals being convolved.
REMEMBER If h[n] is impulse response of a LTI system then step response will be h[n] u[n]
n
[n] = h[n ] u [n ] h[ m ]
and from [n] we can get h[n] by

h[n] = [n] – [n – 1]
2.5.7 Sum Property

The sum of the impulses in a convolution sum of two discrete-time sequence is the product of the sums of the
impulses in the two individual sequences. Let the sum of all the impulses in the signals y(n), x(n) and h(n) be Sy, Sx,
and Sh, respectively. Then
S y = Sx Sh ...(25)
y(n) = x ( n ) h( n ) x ( k ) h( n k )
k
Sy = y( n ) x ( k ) h( n k )
n n k
Interchanging the order of the summation,
Sy = x( k ) h( n k) S x Sh
k n
Sx Sk

2.6 Relation between LTI System Properties and the Impulse System
The impulse response completely characterizes the input-output behaviour of an LTI system. Hence, the
properties of the system, such as memory, causality, and stability, are related to the system’s impulse response.
2.6.1 LTI System With and Without Memory

The present output of a memoryless LTI system depends only on the present input. Exploiting the commutative
property of convolution, we may express the output of discrete-time LTI system as,
y(n) = h(n) x(n)
= h( k ) x ( n k )
k
= ... + h(–2) x(n + 2) + h(–1) x(n + 1) + h(0) x(n) + h(1) x(n – 1) + h(2) x(n – 2) + ....
For the system to be memoryless, y(n) must depend only on x(n) and therefore cannot depend on x(n – k) for
k 0. Hence, every term in the above equation must be zero, except h(0) x(n). The condition implies that h(k) = 0 for
k 0, thus a discrete-time LTI system is memoryless if and only if
h(n) = K (n) ...(26)
where K = h(0) is a constant and the convolution sum reduces to the relation,
y(n) = K x(n)
If a discrete-time LTI system has an impulse response h(n) that is not identically zero for n 0, then the
system has memory.
From equation (4), we can deduce similar properties for continuous-time LTI systems with an without
memory. In particular, a continuous-time LTI system is memoryless if h(t) = 0 for t 0, and such a memoryless
LTI system has the form,
y ( t ) = K x(t)
for some constant K and has the impulse response,
h ( t ) = K (t) ...(27)
2.6.2 Causality for LTI Systems

The output of a causal system depends only on the present and past values of the input to the system.
Again, we write the convolution sum as,
1
y(n)= h( k ) x ( n k ) h( k ) x ( n k ) h( k ) x ( n k )
k k k 0
y ( n ) =[...+ h(–2) x(n + 2) + h(–1) x(n + 1)] + [h(0) x(n) + h(1) x(n – 1) + h(2) x(n – 2) +....]
We see that past and present values of the input x(n), x(n – 1), x(n – 2), .... are associated with indices
k 0 in the impulse response h(k), while future values of the input, x(n + 1), x(n + 2),... are associated with
indices k < 0. In order, then, for y(n) to depend only on past and present values of the input we require that
h(k) = 0 for k < 0. Hence, for a discrete-time causal LTI system,
h ( n ) = 0, for n < 0 ...(28)
and the discrete-time convolution takes the new form
y(n) = h( k ) x ( n k )
k 0

The alternative equivalent form is
y(n) = x ( k ) h( n k )
k
Similarly, a continuous-time LTI system is causal if,

h(t) = 0, for t < 0 ...(29)
and in this case the continuous convolution is given by
t
y(t) = h( ) x ( t )d x ( ) h( t )d
0
2.6.3 Stability for LTI System

A system is stable if every bounded input produces a bounded output. We shall now derive conditions on
h(t) that guarantee stability of the system by bounding the convolution sum. The magnitude of the output is
given by,
y( n ) = h( n ) x( n )
= h( k ) x ( n k ) h( k ) x ( n k )
k k
If we assume that the input is bounded, or x( n ) Bx , then x(n k ) Bx , and it follows that,
y( n ) Bx h( k )
k
From the above equation, we can conclude that if the impulse response is absolutely summable, i.e. if
h( k ) <
k
they y(n) is bounded in magnitude, and hence, the system is stable.

In continuous time, we obtain an analogous characterization of stability in terms of the impulse response of
an LTI system. Specifically, if x ( t ) Bx for all t, then
y( t ) = h ( ) x (t )d h( ) x (t d Bx h( ) d
Therefore, the system is stable if the impulse response is absolutely integrable, i.e., if
h( ) d <
2.6.4 Invertibility for LTI Systems

Consider a continuous-time LTI system with impulse response h(t). This system is invertible only if an
inverse system exists that, when connected in series with the original system, produces an output equal to the
input to the first system. Further more, if an LTI system is invertible, then it has an LTI inverse.

y ( t)
x (t ) h( t) hI( t) x (t )
y ( n)
x (n ) h( n) hI( n) x (n )
Fig. 2.11 : Concept of a general invertible system
The relationship between the impulse response, h(t), and that of the corresponding inverse system hI(t), is
easily derived. The impulse response of the cascade connection in Fig. 2.11 is the convolution of h(t) and hI(t). We
require the output of the cascade to equal the input, or
x(t) [h(t) hI(t)] = x(t)
This requirement implies that,
h(t) hI(t) = (t) ...(32)
Similarly, the impulse response of a discrete-time LTI system, hI(n), must satisfy
h(n) hI(n) = (n) ...(33)
Remember: Generally all LTI systems are invertible.
Property Continuous-time System Discrete-time System

Memoryless h(t ) = K (t ) h(n) = K (n )
Causality h(t ) = 0, for t < 0 h(n) = 0, for n < 0
Stability h( ) dt h( n )
n
Invertibility h(t ) hI(t ) = ( t ) h(n) hI(n ) = (n)
Table-2.1 : Properties of the impulse response representation for LTI system
REMEMBER In discrete-time LTI system also we can have series/cascade connection and parallel connection
as that of continuous time LTI system.
h[t ]
x [n] h1[n ] h2[n ] y [n ] x [n ] y [n ]
= h1[n ] h2 [n]
h1[n]
+ h[n ]
x [t ] y [n ] x [n ] y [n ]
= h1[n ] h2[n ]
–
h2[n]
2.7 Correlation of Signal

A mathematical operation that closely resembles convolution is correlation. Just as in the case of
convolution two signals are involved in correlation as well. In contrast to convolution, however, our objective in
computing the correlation between the two signals is to measure the degree to which the two signals are similar.

2.7.1 Cross-correlation Function of Energy Signals

Suppose that we have two signals x(t) and y(t) each of which has finite energy. The cross-correlation of x(t)
and y(t) is a signal Rxy(t), which is defined as
Rxy( ) = x( t ) y ( t ) dt x( t ) y ( t ) dt ...(34)
or, if both signals x(t) and y(t) are real, then
Rxy( ) = x ( t ) y( t ) dt x( t ) y( t ) dt ...(36)
The index is the (time) shift (or lag) parameter and the subscripts xy in the cross-correlation function Rxy( )
indicate the signals being correlated. The order of the subscripts, with x proceeding y, indicates the direction in which
one signal is shifted relative to the other.
If we reverse the roles of x(t) and y(t) in equation (36) and therefore reverse the order of the indices xy, we
obtain the cross-correlation function,
Ryx( ) = y( t ) x ( t ) dt y( t ) x ( t ) dt ...(37)
By comparing equation (36) with equation (37), we conclude that,

Rxy( ) = Ryx(– ) ...(38)
Study Note
Therefore, Rxy(t) is simply the folded version of Ryx(t), where the folding is done w.r.t. t = 0. Hence, Rxy(t) provides the same
information as Ryx(t) w.r.t. the similarly of x(t) to y(t).
The similarities between the computation of the cross-correlation of two signals and the convolution integral
of two signals is apparent. For energy signals, there is a simple mathematical relationship between correlation and
convolution integral.
Rxy( ) = x( ) y(– ) ...(39)
Proof:
Rxy( ) = x( ) y(– )
= x ( t ) y( ( t )) dt x ( t ) y( t ) dt
Remember: For real signals cross correlation is defined as Rxy( ) = x( ) y(– ) and for non real signals Rxy( ) = x( ) y*(– ).
Similarly for real signals Ryx( ) = y( ) x(– ) and for non real signals Ryx( ) = y( ) x*(– ).
2.7.2 Cross-correlation Function of Power Signals

The cross-correlation function between two power signals x(t) and y(t) is mathematically defined by,
T /2 T /2
1 1
Rxy( ) = Lim x( t ) y ( t ) dt Lim x( t ) y ( t ) dt ...(40)
T T T /2
T T T /2

If x(t) and y(t) are both real, then

T /2 T /2
1 1
Rxy( ) = Lim x ( t ) y( t ) dt Lim x( t ) y( t ) dt ...(41)
T T T /2
T T T /2
An important special case of correlation of power signals is the correlation between two periodic signals
whose fundamental periods are such that the product of the two signals is also periodic. This will happen any
time the fundamental periods of the two periodic signals have an LCM.
For two signals whose product has a period T, the general for of the correlation function (for real power
signals) is given by,
T /2
1
R xy ( ) = Lim x ( t ) y( t ) dt
T T T /2
T /2
1
can be replaced by R xy ( ) = x ( t ) y( t ) dt ...(42)
T T /2
2.7.3 Auto-correlation Function of Continuous-time Signals

A very important special case of the correlation function is the correlation of a function with itself. This
type of correlation function is called the autocorrelation function. If x(t) is an energy signal, its autocorrelation
is
R xy ( ) = x( t ) x ( t ) dt x( t ) x ( t ) dt ...(43)
Remember: for real signal auto correlation is Rxx( ) = x( ) x(– ) and for non real signal Rxx( ) = x( ) x*(– )
If x(t) is a power signal, its autocorrelation is
T /2 T /2
1 1
R xx ( ) = Lim x( t ) x( t ) dt Lim x( t ) x ( t ) dt ...(44)
T T T /2
T T T /2
2.7.4 Properties of Cross-correlation and Autocorrelation Functions

Following are the properties of cross-correlation and autocorrelation functions:
1. We have already demonstrated that the cross-correlation function satisfies the property
R xy ( ) = R yx (– )
With y(t) = x(t), this relation results in the following important property for the autocorrelation
function:
R xx ( ) = R xx (– ) ...(45)
Note : Hence, the autocorrelation function is an even function.
2. Relation to signal energy and signal power:
Case-1: If x(t) is an energy signal, then
R xx ( 0 ) = Ex x 2 ( t ) dt ...(46)
Proof: R xx ( ) = x ( t ) x( t ) dt

Rxx ( ) 0 = x ( t ) x ( t ) dt
Rxx(0) = x 2 ( t ) dt Ex
Case-2: If x(t) is a power signal, then

T /2
1
Rxx(0) = Px = Lim x 2 ( t ) dt ...(47)
T T T /2
T /2
1
Proof: Rxx( ) = Lim x( t ) x( t ) dt
T T T /2
T /2
Rxx ( ) 1
0 = Lim x ( t ) x ( t ) dt
T T T /2
T /2
1
Rxx(0) = Lim x 2 ( t ) dt Px
T T T /2
3. The cross-correlation satisfies the following condition:
Rxy ( ) R x x (0) R y y (0) E x Ey ...(48)

The autocorrelation satisfies the following condition:
Rxx ( ) Rxx(0) = Ex ...(49)

4. The autocorrelation of a periodic signal with period T is periodic with the same period:
If, x(t) = x(t ± T)
then, Rxx( ) = Rxx( ± T) ...(50)
Proof: Let x(t) be a periodic signal with period T. Then,
x(t) = x(t + T)
x(t – ) = x(t – + T)
T /2
1
Now, Rxx( ) = x( t ) x( t ) dt
T T /2
T /2
1
Rxx( – T) = x ( t ) x( t ( T )) dt
T T /2
T /2 T /2
1 1
= x( t ) x( t T ) dt x( t ) x( t ) dt Rx x ( )
T T /2 T T /2
5. If Rxy(0) = 0, then x(t) and y(t) are orthogonal.

R xy ( ) = x ( t ) y( t ) dt
Rxy ( ) = R x y (0) x ( t ) y( t ) dt 0
0

2.7.5 Cross-correlation Sequence of Discrete-time Energy Signals

Suppose that we have two signals x(n) and y(n) each of which has finite energy. The cross-correlation of x(n)
and y(n) is a signal Rxy(m), which is defined as,
Rxy(m) = x( n ) y ( n m) x( n m)y ( n) ...(51)

n n
or, if both signals x(n) and y(n) are real, then
Rxy(m) = x ( n ) y( n m) x( n m ) y( n ) ...(52)
n n
The index m is the (time) shift (or lag) parameter and the subscripts xy on the cross-correlation function
Rxy(m) indicate the signals being correlated. The order of the subscripts with x preceeding y, indicates the
direction in which one signal is shifted relative to the other.
If we reverse the roles of x(n) and y(n) equation (52) and therefore reverse the order of the indices xy,
we obtain the cross-correlation function,
R yx( m ) = y( n ) x ( n m) y( n m ) x( n ) ...(53)
n n
By comparing equation (52) with equation (53), we conclude that,

R x y ( m ) = R yx (– m) ...(54)
Study Note
Therefore, Rxy(m) is simply the folded version of Ryx(m), where the folding is done w.r.t. m = 0. Hence, Rxy(m)
provides the same information as Ryx(m) w.r.t. the similarity of x(n) to y(n).
The similarities between the computation of the cross-correlation of two signals and the convolution
sum of two sequences is apparent. For energy signals, there is a simple mathematical relationship between
correlation and convolution sum:
R x y ( m ) = x(m) y(–m) ...(55)
Remember: For real signals the crosscorrelation is Rxy(m) = x(m) y(–m) and for non real signals crosscorrelation
is Rxy(m) = x(m) y*(–m). Similarly for real signals the crosscorrelation is Ryx(m) = y(m) x(–m) and for non real
signals crosscorrelation is Ryx(m) = y(m) x*(–m).
Proof:
R x y ( m ) = x(m) y(–m)
= x ( n ) y( ( m n )) x ( n ) y( n m)
n n
2.7.6 Cross-correlation Sequence of Power Signals

The cross-correlation function between two power signals x(n) and y(n) is mathematically defined by
1
R x y ( m ) = Lim x( n ) y ( n m) ...(56)
N 2N 1 n N
1
= Lim x( n m)y (n)
N 2N 1 n N

If x(n) and y(n) are both real,
1
Rxy(m) = Lim x ( n ) y( n m) ...(57)
N 2N 1 n N
1
= Lim x( n m ) y( n )
N 2N 1 n N
An important special case of correlation of power signals is the correlation between two periodic signals
whose fundamental periods are such that the product of the two signals is also periodic. This will happen any
time the fundamental periods of the two periodic signals have an LCM.
For two signals whose product has a period N, the general form of the correlation function (for real
power signals).
1 N
R x y ( m ) = Lim x (n) y (n m)
N 2N 1 n N
can be replaced by,

N
1
R xy( m ) = x ( n ) y ( n m) ...(58)
2N 1 n N
2.7.7 Autocorrelation Sequence of Discrete-Time Signals

In the special case where y(n) = x(n), we have the autocorrelation sequence of x(n). If x(n) is an
energy signal, its autocorrelation is
Rxx( m ) = x( n ) x( n m) x( n m ) x( n ) ...(59)
n n
If x(n) is a power signal, its autocorrelation is
1 N
R x x ( m ) = Lim x( n ) x( n m) ...(60)
N 2N 1 n N
1 N
= Lim x( n m ) x( n )
N 2N 1 n N
Remember: For real signal autocorrelation is Rxx(m) =x(m) x(–m) and for non real signal autocorrelation will
be R xx(m) =x(m) x*(–m)
2.7.8 Properties of Crosscorrelation and Autocorrelation Sequences

Following are the properties of crosscorrelation and autocorrelation sequences:
1. We have already demonstrated that the crosscorrelation sequences satisfies the property,
R x y ( m ) = R yx (– m)
With y(n) = x(n), this relation results in the following important property for the autocorrelation
sequence,
R x x ( m ) = R xx (– m) ...(61)
Hence, the autocorrelation sequences is an even function.

Rxx(m) = x( n ) x ( n m)
n
A change of variables is performed by letting n – m = k, which also yields k as n and k

as n – . Therefore,
Rxx(m) = x(k m) x ( k ) x(k ) x(k m) x( k ) x( k ( m))

k k k
= Rxx(–m)
2. Relation to signal energy and signal power.
Case-1: If x(n) is an energy signal, then
Rxx(0) = Ex x2( n) ...(62)

n
Rxx(m) = x( n ) x ( n m)
n
Rxx ( m ) m 0
R x x (0) = x( n ) x( n ) x2( n) Ex
n n
Case-2: If x(n) is a power signal, then
1 N
Rxx(0) = Px = Lim x2( n) ...(63)
N 2N 1 n N
1 N
Rxx(m) = Lim x( n ) x( n m)
N 2N 1 n N
1 N
Rxx ( m ) m 0
R x x (0) = Lim x ( n ) x( n )
N 2N 1 n N
N
1
= Lim x2( n) Px
N 2N 1 n N
3. The cross-correlation sequence satisfies the following condition:
Rxy ( m ) R x x (0) R y y (0) E x Ey ...(64)

The autocorrelation sequence satisfies the following condition:
Rx x ( m ) Rxx(0) = Ex ...(65)
Example 2.12
Let h(t) be the triangular pulse shown in Fig. 2.12 (a) and let x(t) be the unit impulse train Fig. 2.12 (b)
expressed as:
x( t ) T(t) (t nT )
n

Determine and sketch y(t) = h(t) x(t) for the following values of T:
(a) T = 3, (b) T = 2, (c) T = 1.5.
h( t) T (t )
1 1
t t
–1 0 1 –2T –T 0 T 2T
(a) (b)
Fig. 2.12
Solution 2.12
y ( t ) = h(t) (t)
= h( t ) (t nT ) h( t ) (t nT )
n n
= h(t nT )
n
( a ) For T = 3, y(t) = h(t 3n)

n
which is sketched in Fig. 2.13 (a).

( b ) For T = 2, becomes
y(t) = h(t 2n)

n
which is sketched in Fig. 2.13 (b).
y(t)
T=3
1.0
t
–7 –6 –5 –4 –3 –2 –1 0 1 2 3 4 5 6 7
(a)
y(t)
T=2
1.0
t
–7 –6 –5 –4 –3 –2 –1 0 1 2 3 4 5 6 7
(b)
y(t)
T = 1.5
t
–7 –6 –5 –4 –3 –2 –1 0 1 2 3 4 5 6 7
(c)

(c) For T = 1.5, equations becomes,
y(t) = h ( t 1.5 n )
n
which is sketched in Fig. 2.13 (c). Note that when T < 2, the triangular pulses are no longer separated
and they overlap.
Example 2.13
The system shown in figure formed by connecting two systems in cascade. The impulse
responses of the systems are given by h 1 (t) and h 2 (t) respectively and
h 1 ( t ) = e –2t u(t);
h 2 ( t ) = 2e –t u(t)
( a ) Find the impulse response h(t) of the overall system shown in figure.
w( t )
x (t ) h1 ( t ) h2( t) y ( t)
(a)
x (t ) h( t ) y ( t)
(b)
( b ) Determine if the overall system is BIBO stable.

Solution 2.13
( a ) Let w(t) be the output of the first system,
w ( t ) = x(t) h 1 (t)
Then we have, y ( t ) = w(t) h 2 (t)
= [x(t) h 1 (t)] h 2(t)
But by the associativity property of convolution above equations can be rewritten as,
y ( t ) = x(t) [h 1 (t) h 2 (t)] = x(t) h(t)
Therefore, the impulse response of the overall system is given by
h ( t ) = h 1 (t) h 2 (t)
Thus, with the given h1(t) and h2(t), we have
h(t) = h1 ( ) h2 ( t )d
2 (t )
= e u( ) 2 e u( t )d
t
t t
= 2e e u( ) u( t )d 2e e d u( t )
0
= 2(e –t – e –2t ) u(t)

( b ) Using the above h(t), we have
h( ) d 2
= 2 (e e )d
2 1
= 2 e d e d 2 1 1
0 0
2

Thus, the system is BIBO stable.
Example 2.14
Consider a continuous-time LTI system whose step response is given by

s ( t ) = e –t u(t)
Determine and sketch the output of this system to the input x(t) shown in figure (a).
Solution 2.14
From figure (a) the input x(t) can be expressed as,
x ( t ) = u(t – 1) –u(t – 3)
Since the system is linear and time-invariant, the output y(t) is given by
y ( t ) = s(t – 1) –s(t – 3)
= e–(t – 1) u(t – 1) – e–(t – 3) u(t – 3)
which is sketched in figure (b).
y ( t)
x (t )
1
1
t
0 1 2 3
t
0 1 2 3 4 –1
(a) (b)
Example 2.15
Consider a continuous-time LTI system described by,

t T/2
1
y ( t ) = T { x( t )} x( ) d
T t T/2
( a ) Find and sketch the impulse response h(t) of the system.

( b ) Is this system causal?
Solution 2.15
( a ) Equation can be rewritten as,
t T /2 t T /2
1 1
y(t) = x( ) d x( ) d
T T
This equation can be expressed as,
1 T 1 T
y(t) = x( t ) u t x( t ) u t
T 2 T 2
1 T T
= x( t ) u t u t = x(t) h(t)
T 2 2
1 T T
Thus, we obtain h(t) = u t u t
T 2 2
1 T T
t
= T 2 2
0 oth erwise
h(t) is sketched in figure.

(b) From figure, we see that, h(t) 0 for t < 0. Hence, the system is not causal.
h( t )
1/T
t
–T/2 0 T/2
Example 2.16
Compute y[n] = x[n] h[n], where

( a ) x[n] = n u[n], h[n] = n u[n]
( b ) x[n] = n u[n], h[n] = –n u[–n], 0 < < 1
Solution 2.16
( a ) We have,
k n k
y[n] = x[k ] h[n k] u[k ] u[n k]
k k
k n k
= u[k ] u [n k]
k
1 0 k n
Since, u[k] u[n – k] =
0 oth erwise
n n k
k n k n
we have, y[n] = ,n 0
k 0 k 0
n 1 ( / )n 1
u [n ]
Thus , y[n] = 1 ( / )
n
( n 1) u[n ]
1 n 1 n 1
( ) u[n ]
or, y[n] =
n
( n 1) u[n ]
k (n k)
(b) y[n] = x k ] h[n k] u[k ] u[ ( n k )]
k k
n 2k
= u[k ] u[k n]
k
For n 0, we have
1 n k
u[k] u[k – n] =
0 oth erwise
n
n 2k
Thus, we have y[n] = 2
,n 0
k 0 1
For n greater than zero we have
1 n k
u[k] u[k – n] =
0 oth erwise

2n
n 2k n
y[n] = 2
k n 1
n
= 2
; n 0
1
n
we obtain, y[n] = 2
; ( All n )
1
which is sketched in figure.
y [n]
n
–2 –1 0 1 2 3
Example 2.17
Consider a discrete-time LTI system whose input x[n] and output y[n] are related by,
y[n] = 2k n
x[k + 1]
k=
Is the system causal?
Solution: 2.17
By the definition the impulse response of system is given by
n n n
h[n] = 2k n
[k 1] 2 ( n 1)
[k 1] = 2 ( n 1)
[k 1]
k k k
By changing the variable k + 1 = m, we obtain

n 1
( n 1) ( n 1)
h[n] = 2 [m ] 2 u[n 1]
m
From equation we have h[–1] = u[0] = 1 0. Thus, the system is not causal.
Example 2.18
Consider a discrete-time LTI system with impulse response h[n] given by,
h[n] = n u[n]
( a ) Is this system causal?
( b ) Is this system BIBO stable?
Solution: 2.18
( a ) Since h[n] = 0 for n < 0, the system is causal.
( b ) For BIBO stability,
k
h[k ] = u[n ]
k k
k 1
= ( 1)
k 1
Therefore, the system is BIBO stable if 1 and unstable if 1.

Example 2.19
Find the impulse response h[n] for each of the causal LTI discrete-time systems satisfying the following
difference equations and indicate whether each system is an FIR or an IIR system.
(a) y[n] = x[n] – 2x[n – 2] + x[n – 3]
( b ) y[n] + 2y[n – 1] = x[n] + x[n – 1]
1
( c ) y[ n] y[ n 2] 2 x[ n ] x[ n 2]
2
Solution 2.19
( a ) By definition,
h[n] = [n] – 2 [n – 2] + [n – 3]
or, h[n] = {1, 0, –2, 1}
Since, h[n] has only four terms, the system is an FIR system.
(b) h[n] = –2h[n – 1] + [n] + [n – 1]
Since, the system is causal, h[–1] = 0.
Then, h[0] = –2h[–1] + [0] + [–1]
= [0] = 1
h[1] = –2h[0] + [1] + [0]
= –2 + 1 = –1
h[2] = –2h[1] + [2] + [1]
= –2(2) = –2 2
h[n] = –2h[n – 1] + [n] + [n – 1] = (–1)n 2n – 1

Hence, h[n] = [n] + (–1)n 2n – 1 u[n – 1]
Since, h[n] has infinite terms, the system is an IIR system.
1
(c) h[n] = h[n 2] 2 [n ] [n 2]
2
Since, the system is causal,
h[–2] = h[–1] = 0, Then
1
h[0] = h[ 2] 2 [0] [ 2] 2 [0] 2
2
1
h[1] = h[ 1] 2 [1] [ 1] 0
2
1 1
h[2] = h[0] 2 [2] [0] (2) 1 0
2 2
1
h[3] = h[1] 2 [3] [1] 0
2
Hence, h[n] = 2 [n]

Since, h[n] has only one term, the system is an FIR system.

5 . Let, y(t) = e t u ( t ) (t 3k)

k
Subjective Practice Problems Show that, y(t) = Ae–t for 0 t < 3,
and determine the value of A.
1 . Consider the signal, 6 . Which of the following impulse response

n 1 correspond(s) to stable LTI systems?
1
h[n ] {u[n 3] u[n 10]} (a) h 1(t) = e –(1 – 2j)t u(t)
2 (b) h2(t) = e–t cos(2t) u(t)
Express A and B in terms of n so that the following
7 . Which of the following impulse response
equation holds:
correspond(s) to stable LTI systems?
n k 1
1
; A k B (a) h1[n ] n cos n u [n ]
h[n k] 2 4
0; elsewh ere (b) h2[n] = 3n u[–n + 10]
8 . For each of the following statements, determine

1; 0 n 9 whether it is true or false:
2 . Let, x[n] =
0 ; elsewh ere (a) If x[n] = 0 for n < N1 and h[n] = 0 for n < N2,
then x[n] h[n] = 0 for n < N1 + N2.
1; 0 n N
and h[n] = (b) If y[n] = x[n] h[n], then
0 ; elsewh ere
y[n – 1] = x[n – 1] h[n – 1].
where, N 9 is an integer. Determine the value of (c) If y(t) = x(t) h(t), then y(–t) = x(–t) h(–t).
N, given that, (d) If x(t) = 0 for t > T1 and h(t) = 0 for t > T2,
y[n] = x[n] h[n] then x(t) h(t) = 0 for t > T1 + T2.
y[4] = 5, y[14] = 0
9 . Consider the cascade of the following two systems S1
3 . A linear system S has the relationship, and S2, as depicted in figure.
w[n]
x [n] S1 S2 y [n ]
y[n] = x[k ] g[n 2k ]
k
S1 : Causal LTI,
between its input x[n] and its output y[n],
1
where g[n] = u[n] –u[n – 4]. w[n] = w[n 1] x[n ]
2
(a) Determine y[n] when x[n] = [n – 1]. S2 : Causal LTI,
(b) Determine y[n] when x[n] = [n – 2]. y[n] = y[n – 1] + w[n]
(c) Is S LTI? The difference equation relating x[n] and y[n] is,
(d) Determine y[n] when x[n] = u[n].
1 3
y[n] =y[n 2] y[n 1] x[n ]
4 . Suppose that, 8 4
(a) Determine and .
1; 0 t 1
x( t) = (b) Determine the impulse response of the cascade
0 ; elsewh ere connection of S1 and S2.
t 1 0 . Consider the cascade interconnection of three causal

and h(t) = x where, 0 < 1
LTI systems, illustrated in figure (a). The impulse
(a) Determine and sketch y(t) = x(t) h(t). response h2[n] is
(b) If dy(t)/dt contains only three discontinuities, h2[n] = u[n] – u[n – 2]
what is the value of ? and the overall impulse response is as shown in
figure (b).

t
x [n ] h1[n] h2[n ] h2[n] y [n ] (t )
y(t) = e x( 2) d
(a)
11
What is the impulse response h(t) for this system?
10 (b) Determine the response of the system when the
8 input x(t) is as shown in figure.
5
4 x ( t)
1 1
n
–1 0 1 2 3 4 5 6 7 1
(b)
t
(a) Find the impulse response h1[n]. –1 2
(b) Find the response of the overall system to the
input, 1 5 . Suppose that the signal,
x[n] = [n] – [n – 1] x( t) = u(t + 0.5) – u(t – 0.5)
is convolved with the signal,
1 1 . The following are the impulse responses of discrete-
time LTI systems. Determine whether each system h(t) = e j 0 t
is causal and/or stable. (a) Determine a value of 0 which ensures that,
n
y(0) = 0
1 where, y(t) = x(t) h(t)
(a) h [n ] u[n ]
5
(b) Is your answer to the previous part unique?
(b) h[n] = (0.8)n u[n + 2]
n
1 6 . Determine whether each of the following statements
1
(c) h [n ] u[ n ] concerning LTI systems is true or false. Justify your
2
answers.
(d) h[n] = (5)n u[3 – n] (a) If h(t) is the impulse response of an LTI system
1
n and h(t) is periodic and non-zero, the system is
(e) h [n ] u [n ] (1.01) n u [n 1] unstable.
2
(b) The inverse of a causal LTI system is always
n
1 causal.
(f) h [n ] u [n ] (1.01) n u [1 n ]
2
(c) If h[n ] K for each n, where K is a given
n
1 number, then the LTI system with h[n] as its
(g) h [n ] n u[n 1]
3 impulse response is stable.
(d) If a discrete-time LTI system has an impulse
1 3 . The following are the impulse response of continuous
response h[n] of finite duration, the system is
time LTI systems. Determine whether each system
stable.
is causal and/or stable.
(e) If an LTI system is causal, it is stable.
(a) h(t) = e–4t u(t – 2)
(b) h(t) = e–6t u(3 – t) (f) The cascade of a non-causal LTI system with a
(c) h(t) = e–2t u(t + 50) causal one is necessarily non-causal.
(d) h(t) = e2t u(–1 – t) (g) A continuous-time LTI system is stable if and
6t
only if its step response s(t) is absolutely
(e) h( t ) e integrable – that is, if and only if
(f) h(t) = t e –t u(t)
(g) h(t) = (2e–t – e(t – 100)/100) u(t) s( t ) dt
1 4 . (a) Consider an LTI system with input and output
(h) A discrete-time LTI system is causal if and only
related through the equation,
if its step response s[n] is zero for n < 0.

1 7 . Consider the cascade of two systems shown in figure. Show that the commutativity property does not
The first system, A is known to be LTI. The second hold for these two systems by computing the
system, B is known to be the inverse of system A. Let impulse response of the cascade combination in
y1(t) denote the response of system A to x1(t), and figure (a) and (b), respectively.
let y2(t) denote the response of system A to x2(t).
System System
x [n] y [n ]
LTI y ( t) System A B
x ( t) x (t )
system A B
(a)
(a) What is the response of system B to the input
ay1(t) + by2(t), where a and b are constants? System System
x [n] y [n ]
B A
(b) What is the response of system B to the input
y1(t – )? (b)
1 8 . In the text, we saw that the overall input-output (b) Suppose that we replace system B in each of the
relationship of the cascade of two LTI systems does interconnected systems of figure by the system
not depend on the order in which they are cascaded. with the following relationship between its input
This fact, known as the commutativity property, w[n] and output z[n];
depends on both the linearity and the time z[n] = w[n] + 2
invariance of both systems. In this problem, we Repeat the calculations of part (a) in this case.
illustrate the point.
(a) Consider two discrete-time systems A and B,
where system A is an LTI system with unit sample
response h[n] = (1/2)n u[n]. System B, on the
other hand, is linear but time varying.
Specifically, if the input to system B is w[n], its
output is
z[n] = n w[n]

Chapter
Continuous-Time
Fourier Series 3
Introduction
As we have seen in the previous chapter, we can obtain the response of a linear system to an arbitrary input
by representing it in terms of basic signals. The specific signals used were the shifted -functions. Often, it is convenient
to choose a set of orthogonal waveforms as the basic signals. There are several reasons for doing this. First, it is
mathematically convenient to represent an arbitrary signal as a weighted sum of orthogonal waveforms, since many
of the calculations involving signals are simplified by using such a representation. Second, it is possible to visualize
the signal as a vector in orthogonal coordinate system, with the orthogonal waveforms being coordinates. Finally,
representations in terms of orthogonal basis functions provides a convenient means of solving for the response of
linear systems to arbitrary inputs.
For periodic signals, a convenient choice for an orthogonal basis is the set of harmonically related complex
exponential. The choice of these waveforms is appropriate since such complex exponentials are periodic, are relatively
easy to manipulate mathematically, and yield results that the have a meaningful physical interpretation. The
representation of a non-sinusoidal periodic signal in terms of complex exponentials, or equivalently; in terms of sine
and cosine waveforms, leads to the Fourier series. Or, in other words, the Fourier series as a mathematical tool that
allows the representation of any periodic signal as the sum of harmonically related sinusoids.
So we saw that, it is mathematically convenient to represent a signal in terms of orthogonal signals. Another
reason for doing this is:
when we represent any signal x(t) as a weighted sum of a basic signal then it is easier to find output of that
signal when it is applied as input to an LTI system. The basic signal is eC1t where s1 is any constant. Suppose eC1t is
input to the LTI system.
C1t
Input =e LTI System y(t)
Fig. 3.1 : e s1t is input to system
So, y(t) = input h(t) ...(1)

= eC1t h(t)
= h( ) eC1 ( t )
d

y(t) = eC1 t h( ) e C1
d {Here C1 is a constant} ...(2)
Now if we define a new function H(s) where ‘s’ is variable,
st
H(s) = h(t ) e d ...(3)
In the above equation ‘s’ is variable, and
H (s ) s C1 = h(t ) e C1 t
dt
So, equation (2) can be modified as,

y(t) = eC1 t H (C1 ) ...(4)
Thus if we define x(t) as a weighted sum of eCk t for k = 1, 2, 3... then output of LTI system for x(t) as input
can be easily determined.
That is if, x(t) = a k eCk t
System
Then, output of e Ck t H ( s) s Ck
System
Output of ak eCk t ak eCk t H ( s)
s Ck
System
ak eCk t ak eCk t H ( s)
s Ck
k k
Study Note
• Thus when input to an LTI system is of the form eCt where C is a constant, then output is eCt H ( s) .
s C
• eCt is eigen function of the continuous time LTI system and H ( s ) s C

is eigen value of this function
3.1 Fourier Series Representation of Continuous Time Periodic Signal

Let x(t) is a periodic signal with period T, if Ck represent exponential Fourier series coefficient then,
x(t) = Ck exp ( j o kt ) ...(5)

k
1 j o kt
Ck = x (t ) e dt ...(6)
T t T
2
In above equations o , and t = <T> means integration over one period of x(t).
T
If we analyze equation (5) we can see that,
x(t) = ....+ C–2 exp(–j2 ot) + C–1 exp(–j ot) + C0 + C1 exp(j ot) + C2 exp(2j ot) + ....

Continuous-Time Fourier Series 107
So, C0 : Shows the dc content in signal x(t).

C1/C–1 : Show content of o/– o that is fundamental frequency in signal x(t).
C2/C–2 : Show content of 2 o/–2 o that is content of 2nd harmonic in signal x(t).
CN/C–N : Show content of N o/–N o, that is content of Nth harmonic in signal x(t).
Again analyzing equation (5) we can see that,
x(t) = ...+ C–2 exp(–2j ot) + C–1 exp(–j ot) + C0
+ C1 exp(j ot) + C2 exp(2j ot) + ....
Since we know that, exp(j t) = cos( t) + j sin( t) ...(6)
Now using equation (6), x(t) can be written as,
x(t) = ...+ C–2[cos(2 ot) – j sin(2 ot)] + C–1[cos ot – j sin ot]
+ C0 + C1[cos( ot) + j sin( ot)] + C2[cos(2 ot) + j sin(2 ot)] +....
x(t) = C0 + cos( ot) [C1 + C–1] + cos(2 ot) [C2 + C–2] + ....
+ sin( ot) [j(C1 – C–1)] + sin(2 ot) [j(C2 – C–2)] + ....
x(t) = C0 cos( k o t ) [C k C k] sin( k ot ) [ j (C k C k )] ...(7)

k 1 k 1
The above equation is written in simplified form as,
x(t) = C0 ak cos( k ot ) bk sin( k ot ) ...(8)

k 1 k 1
The equation (8) represented x(t) in terms of sine and cosine terms. We can see that, ak are coefficient of
cosine terms and bk are coefficient of sine terms.
Also, ak = Ck + C–k ...(9)
bk = j[Ck – C–k] ...(10)
Study Note
• C0 is the average value of signal x(t),
1
C0 = x(t ) dt
T t T
• If in any question if we have to find ak or bk then best method is to find Ck using equations (6) and use (9) and (10) to find
answer.
• To find ak or bk we have direct relations also,
2
ak = x (t ) cos( k ot) dt
T t T
2
bk = x (t ) sin( k ot ) dt
T t T
Let us assume that signal x(t) is real.

Since, x(t) is a real signal.
x(t) = x (t) ...(11)
Now let us assume exponential Fourier series coefficient of x(t) are Ck, thus
1 j o kt
Ck = x (t ) e dt ...(12)
T T

Let, the exponential Fourier series coefficient of x (t) are Bk, thus
1 j o kt
Bk = x * (t ) e dt ...(13)
T T
Taking conjugate both side in equation (12) we get

1
Ck = x (t ) e j o kt dt ...(14)
T T
Putting k = –k in equation (14) we get,

1 j o kt
C k = x * (t ) e dt ...(15)
T T
By comparing equation (15) and (12) we get

Bk = C k
Thus if Ck are exponential Fourier coefficients of x(t) then exponential Fourier coefficient of x (t) will be C k .
And if x(t) is real i.e. x(t) = x (t) then
Ck = C k ...(16)
REMEMBER If x(t) is real then,

Ck = C k and C k Ck
That is exponential Fourier series coefficient are conjugate symmetric.
Now, for real signal x(t) the trigonometric Fourier series coefficient will be
ak = Ck C k Ck Ck 2 Real{Ck } ...(17)
bk = {Ck C k} j{Ck Ck } 2 Img {Ck } ...(18)
REMEMBER For real valued signal x(t)

ak = 2 times Real part of Ck
bk = –2 times Imaginary part of Ck
Since any signal real periodic signal x(t) can be written as sum of it’s even and odd parts that is xe(t) and xo(t).
If exponential Fourier series coefficients of x(t) are Ck then lets find out contribution due to xe(t) and xo(t).
x(t) = xe(t) + xo(t) [Since x(t) is real xe(t) and x0(t) will also be real]
1
and Ck = x (t ) e j o kt
dt
T T
1 j o kt
= ( xe (t ) xo (t )) e dt
T T
1 j o kt 1 j o kt
= xe (t ) e dt xo ( t ) e dt
T T
T T
= Exponential Fourier series coefficient of xe(t) +

Exponential Fourier series coefficient of xo(t)
The exponential Fourier series coefficient of xe(t) are Ek

1 j o kt
Ek = xe (t ) e dt
T T
1
= xe (t ) [cos( o kt ) j (sin o kt )] dt
T T
In above equation xe(t) is real and even.

So, xe(t) cos( okt) will also be real and even and xe(t) sin( okt) will be real and odd.
1
Ek = xe (t ) cos( o kt ) dt j xe (t ) sin( o kt ) dt
T T T
Since xe(t) sin( okt) is odd it’s integration over the period will be zero.
1
Ek = x e(t ) cos( 0 kt ) dt,
T T
since xe(t) and cos( okt) both are real so Ek will also be real.
The exponential Fourier series coefficient of xo(t) are Ok,
1 j o kt
Ok = xo (t ) e dt
T T
1
= xo (t ) (cos( o kt ) j sin( o kt )) dt
T T
Here, xo(t) cos( okt) will be odd and xo(t) sin( okt) will be even.
1
Ok = j xo (t ) sin( o kt ) dt
T T
Since xo(t) and sin( okt) both are real so Ok is imaginary.

Thus, Ck = [Ek + Ok]
Here, Ek is real and Ok is imaginary.
Real {Ck} = Exponential Fourier series coefficient of xe(t) ...(19)
and jImaginary{Ck} = Exponential Fourier series coefficient of x0(t) ...(20)
REMEMBER If x(t) is real and Ck are its exponential Fourier series coefficients then Real {Ck} i.e. real part of Ck are
exponential Fourier series coefficients of xe(t) and j Img{Ck} are exponential Fourier series coefficient
of x0(t).
3.2 Convergence of Fourier Series

Any periodic signal, i.e., one for which x(t) = x(t + T), can be expressed by a Fourier series provided that,
1. if it is discontinuous, there are a finite number of discontinuities in the period T;
2. it has a finite average value over the period T;
3. it has a finite number of positive and negative maxima in the period T.
When these Dirichlet condition are satisfied, the Fourier series exist. These conditions are sufficient but not
necessary.

Example 3.1
Find the exponential Fourier series coefficient of the signal x(t).

x(t)
1.0
t
–8 –6 –4 –2 2 4 6 8
Solution 3.1
We can see that x(t) is periodic with period and T = 8. To find Ck
4
1 j o kt
Ck = x (t ) e dt
8 4
2
o =
8 4
4 2
1 jk / 4t 1 jk / 4t
Ck = x (t ) e dt 1 e dt
8 4
8 2
k
k k sin
1 j j 2
Ck = [e 2 e 2 ]
j2 k k
We can find C0 using above expression and applying L-Hospital rule, or C0 is average value of signal x(t).
4 2
1 1 1
So, C0 = x (t ) dt 1 dt
8 4
8 2
2
The plot of Ck is
Ck
0.5
1/
–3
k
–2 –1 0 1 2 3
–1/3 –1/3
Study Note
We can see that using Fourier series we find Ck of a signal, Ck are Fourier series coefficient which has value only at integer
value of k, so Ck is a discrete signal.Generally Ck is non-periodic, thus we say that Fourier series convert a continuous
periodic signal x(t) into discrete non-periodic signal Ck.

Example 3.2
The periodic signal x(t) is real and Ck are it’s exponential Fourier series coefficient. If value of C1 = 1 + 3j,
C2 = 0.5 + 0.5j then the value of C–1 and C–2 will be _______ and ______.
Solution 3.2
Since, x(t) is real so Ck will be conjugate symmetric.
Ck = C k or C k Ck
C–1 = C1 1 3j
C–2 = C2 0.5 0.5 j
Example 3.3
The real periodic signal x(t) has exponential Fourier series coeffici ent, Ck 0.5 3j
. Then what is
k k
exponential Fourier series coefficient of even and odd part of x(t).
Solution 3.3
We know that real part of Ck correspond to exponential Fourier series coefficient of even part of x(t). That is 0.5/
k.
3
And for odd part of x(t) the exponential Fourier series coefficient are j Imaginary part of Ck. That is j .
k
3.3 Polar Form Representation of Fourier Series

We have already seen exponential and trigonometric representation of Fourier series, the trigonometric
representation can be modified into polar form.
Case-1: From equation (8),
x(t) = C0 ak cos( k ot) bk sin( k ot)

k 1
If we assume that, dk = ak2 bk2 ...(21)
ak = dk cos k ...(22)
bk = –dk sin k ...(23)
So, x(t) = C0 d k cos k cos( k ot ) d k sin k sin( k ot)

k 1
= C0 d k cos( k ot k) ...(25)
k 1
The equation (25) represent polar form of Fourier series of signal x(t). In this expression C0 show dc content
in x(t), dk show the amplitude of kth harmonic in x(t).
Case-2: The polar form can also be written in another form.

Let, dk = a k2 bk2
ak = dk sin k ...(26)
bk = dk cos k ...(27)
1 ak
k = tan ...(28)
bk
So, equation (8) can be written as,
x(t) = C0 ( d k sin k cos( k ot) d k cos k sin k ot )

k 1
= C0 d k sin( k ot k ) ...(29)
k 1
So equation (25) and (29) both show polar form representation of Fourier series.
3.4 Symmetry Condition

1. x(t) is an even signal i.e. x(t) = x(–t)
Let us assume that Ck are exponential Fourier series coefficient of signal x(t). Thus,
1 j o kt
Ck = x (t ) e dt ...(30)
T T
Now, let Bk are exponential Fourier series coefficient of x(–t), Thus

1 j o kt
Bk = x( t ) e dt ...(31)
T T
Now replacing t by –t in equation (30) we get

1 j 0 kt
Ck = x( t ) e dt
T T
and replacing k by –k in above equation we get

1 j 0 kt
C–k = x( t ) e dt ...(32)
T T
So comparing (32) and (31) we get

Bk = C–k
That is if Ck are exponential Fourier series coefficient of x(t) then C–k will be exponential Fourier series
coefficient of x(–t).
Since, x(t) = x(–t) (signal is even)
Ck = C–k ...(33)
In the above case we took x(t) as even and we found Ck are also even. Now using equation (9) and (10)
ak = Ck + C–k = 2Ck
bk = j[Ck – C–k] = 0
Thus for even signal in trignometric fourier series representation only cosine terms will be present and sine

terms will be absent.
Note: If x(t) is even then Ck is also even and trignometric fourier series representation has only cosine terms.
Now, let us assume x(t) is real and even and Ck are it’s exponential Fourier series coefficient, then Ck is even
and conjugate symmetric that is
Ck = C–k and Ck C k
C k = C–k
Thus if x(t) is real and even then Ck is real and even.
Note: If x(t) is real and even then Ck is also real and even.
When x(t) is real and even then using equation (17) and (18) we get
ak = 2 Real {Ck} = 2 Ck
bk = 2Img {Ck} = 0
Thus when x(t) is real and even then ak = 2 Ck and bk = 0, that is trigonometric representation of Fourier
series of x(t) only cosine terms exist. Also the coefficient of cosine terms ak are also real and even.
Remember: If x(t) is real and even then it’s trigonometric representation has only cosine terms and coefficients of cosine
term is also real and even.
2. x(t) is odd signal, i.e., x(t)

If x(t) is an odd signal and Ck is exponential Fourier series coefficient of x(t).
Since, x(t) = –x(–t), thus
Ck = –C–k ...(34)
Thus if x(t) odd then exponential Fourier series coefficients are also odd.
Note: For odd signals C0 = –C0, that is C0 = 0. That is for odd signals average value of signal = 0 always.
In the above case we took x(t) as odd and we found Ck are also odd. Now using equation (9) and (10)
ak = Ck + C–k = 0
bk = j[Ck – C–k] = j2Ck
Thus for odd signal in trignometric fourier series representation only sine terms will be present and cosine
terms will be absent.
Note: If x(t) is odd then Ck is also odd and trignometric fourier series representation has only sine terms.
Now let us assume that signal is real and odd

Ck = –C–k and Ck C k
C k = –C–k
Thus C–k is purely imaginary, that is Ck is also imaginary.
Thus when x(t) is real and odd then Ck is odd and imaginary.
Now if we use equation (17) and (18) for a real and odd signal then,
ak = 2 real{Ck} = 0

bk = –2 Img {Ck} = –2 Img{Ck]

So in trigonometric Fourier series representation of real and odd x(t) there are no cosine terms and only sine
terms are present.
REMEMBER • For odd signal, Ck is also odd.

• For real and odd signal, Ck is imaginary and odd.
• For real and odd signal, only sine terms exist in trigonometric representation of x(t).
T
3. Half wave symmetric, i.e., x( t ) x t
2
That is if we shift the signal right or left by period/2, then we get –x(t). Some example of signals having such
symmetry are shown in Fig. 3.2.
x (t) x (t)
t t
0 T 0 T
Fig. 3.2 : Waveforms with half-wave symmetry

Let x(t) has exponential Fourier series coefficient Ck,
1 j o kt
Ck = x (t ) e dt ...(35)
T T
Now if x(t + T/2) is created by shifting x(t) then it will also be periodic with period T and let Bk are its
exponential Fourier series coefficient. Thus,
1 T j o kt
Bk = x t e dt ...(36)
T T
2
T
Let, t =
2
T
1 j ok
Bk = x( ) e 2
d
T T
1 j ok
= x( ) e ej o kT /2 d
T T
1 j ok
= x( ) e d ej o kT /2
T T
2
Using equation (35) and o
T
Bk = Ck ej k

T T
So, exponential Fourier series of x t are Ck ej k and since x(t ) x t , Thus
2 2
Ck = –Ck ej k
Ck + Ck ej k = 0
Ck(1 + ej k) = 0
• If k is even then 2 Ck = 0, thus for half wave symmetric signal x(t), Ck = 0 for even value of k.
• If k is odd then 0 = 0, thus Ck can be non-zero for odd value of k.
Study Note
• For half wave symmetric signal, that is x(t) = –x(t ± T/2) the Fourier series coefficient Ck = 0 for even values of k and
non-zero for odd value of k.
• Thus half wave symmetric signal has only odd harmonics and even harmonics are zero. The trigonometric representation
will also have only odd harmonics of sine and cosine terms.
3.4 Quarter Wave Symmetry

A periodic function possesses a quarter-wave symmetry if
(i) it has either odd or even symmetry and
(ii) it has half-wave symmetry.
A waveform that satisfies the above conditions and if it is shifted towards right or left by T/4 period gives a
new waveform with a half-wave symmetry, but the even (odd) symmetry of the original unshifted waveform has
changed into odd (even) symmetry, is said to be having a quarter-wave symmetry.
If x(t) is real and even and half-wave symmetric:
Then Ck is real and even, for even values of k it is zero.
So, ak = 2 Ck and it will be zero for even value of k
and bk = 0
Thus x(t) will have only odd harmonics of cosine terms.
If x(t) is real and odd and half wave symmetric:
Then Ck is imaginary and odd, for even values of k it is zero.
So, ak = 0
bk = –2 Img{Ck} and it will be zero for even values of k
Thus x(t) will have only odd harmonics of sine terms.
3.5 Properties of Continuous Time Fourier Series

3.5.1 Linearity
If x(t) and y(t) denote periodic signals with period T and
Exponential Fourier
x(t ) Series coefficient
Ck

Exponential Fourier
y (t ) Series coefficient
Bk
Exponential Fourier
Then, ax(t ) by(t ) Series coefficient
aCk bBk
Example 3.4
Find the exponential Fourier series coefficient of signal x(t).

x (t)
5.0
t
–2 – – /2 0 /2 2
Solution 3.4
So we can see that signal x(t) is periodic with period 2 ,
2
1 j kt 1 jkt
Ck = x (t ) e 2 dt x(t ) e dt
2 2
/2
1 jkt 1 j k j k
= 5e dt = [5( e 2 e 2 )]
2 /2
2 ( jk )
5
Ck = sin k
k 2
Example 3.5
Find the exponential Fourier series coefficient of y(t).

y (t)
5/2
t
– 0 +
–5/2
Solution 3.5
5
We can see that, y(t) = x(t ) {x(t) is signal in example 3.4}
2

5
So, y(t) = x(t )
2
The Fourier series coefficient of y(t) (Bk) = The Fourier series coefficient of x(t) (Ck) –
The Fourier series coefficient of (5/2)
Bk = Ck – Fourier series coefficient of (5/2)
Since, signal 5/2 is a dc signal let it’s Fourier series coefficient are Dk
5/2 , k 0
so, Dk =
0, k 0
C0 5/ 2, k 0
So, Bk =
Ck , k 0
5 k
Since, Ck = sin
k 2
C0 can be obtained by L-Hospital rule.
5
C0 =
2
0, k 0
Bk = 5 k
sin , k 0
k 2
So, y(t) has Fourier series coefficient equal to Bk.

In this example we used linearity property.
REMEMBER • Analyzing Example 3.4 and 3.5 we can see that x(t) is not looking like half-wave symmetric but if
we creat a signal y(t) = x(t) –5/2 then y(t) is a half-wave symmetric signal. Thus y(t) will have only
odd harmonics and using linearity property we can also say that x(t) will have only odd harmonics
because x(t) and y(t) have only difference in their dc values as:
{Bk = Ck – Fourier series coefficient of dc signal (5/2)}
• Thus if in any question we have a signal x(t) and we have to find harmonics of the signal then we can
shift the signal on amplitude axis i.e. we can form signal (x(t) ± a) to check that signal is half-wave
symmetric or not. If (x(t) ± a) is half-wave symmetric then x(t) will also have only odd harmonics.
3.5.2 Time Shifting

When a time shift is applied to a period signal x(t), the period ‘T’ of the signal is preserved. If
Exponential Fourier
Ck
Exponential Fourier
then, y(t) = x(t to ) Series coefficient
Bk Ck exp ( j o kto )
Thus the magnitude of coefficient remain same but the phase changes.
Proof:
1 j o kt
Since, Ck = x(t ) e dt
T T
Let, t – t0 =

1 j o kt
and Bk = x(t to ) e dt
T T
Let, t – to =
1
So, Bk = x( ) e j ok ( to )
d
T T
Bk = 1 j ok j o kto
x( ) e d e
T T
Bk = Ck exp(–j okto)
3.5.3 Frequency Shifting

Exponential Fourier
If x(t) is a periodic signal, and x(t ) Series coefficient
Ck ,
j 0mt Exponential Fourier

then, y(t) = x(t ) e Series coefficient
Bk Ck m
x(t) = Ck exp( j o kt )
k
and y(t) = Bk exp( j o kt ) = Ck m exp( j o kt )

k k
Let, k–m =
y(t) = C exp( j o( m) t )
= C exp( j o t ) exp( j o mt )
= x(t) exp (j omt)
Thus, y(t) = ( x(t ) e j o mt )
REMEMBER In case of time shifting,

j 0 kt0
x(t ) Ck , x(t t0 ) Ck e
The shifting in time goes with same sign in fourier series.
In case of frequency shifting,
j 0 mt
x(t ) Ck , x(t ) e Ck m
The shifting in frequency goes with apposite sign in time domain.
3.5.4 Time Reversal

Exponential Fourier
If x(t) is periodic and x(t ) Series coefficient
Ck .
Exponential Fourier
Then, x( t ) Series coefficient
C k
This we have already produced.
3.5.5 Time Scaling

Exponential Fourier
If x(t) is a periodic signal, x(t ) Series coefficient
Ck .
Exponential Fourier
Then, y(t) = x(at ) Series coefficient
Bk Ck
Proof: x(t) is periodic with period T, then period of x(at) will be T/a. So,
T
1 j o kt 2
Ck = x(t ) e dt , o
T
T 0
T /a
1 j o kt 2
and Bk = x(t ) e dt , o
T /a 0
T /a
T /a
1 ja o kt
= x(t ) e dt
T /a 0
Let, at =
T
1 j ok
Bk = x( ) e d
T 0
Bk = Ck
The Fourier series coefficient has not changed but the Fourier series representation will change. We can see
that,
x(t) = Ck exp( j 0 kt )
k
the representation will be

Ck Ck
C0 C0
C–1 C–1
C–2 C1 C–2 C1
C–3 C2 C–3 C2
C3 C3
k
–3 –2 –1 0 1 2 3 –3 0 –2 0
– 0 0 2 0 3 0
So we can see that, x(t) has C0 at = 0, C1 at =2 0, C–1 at =– 0, C2 at = 2 0 and so on.

Now, for x(at) the value of coefficient are same. Thus,
x(at) = Bk exp( j 0 kt ) Ck exp( j 0 kt )

k k

Bk Bk
C0 C0
C–1 C–1
C–2 C1 C–2 C1
C–3 C2 C–3 C2
C3 C3
k
0 1 2 3 –3 0 –2 0
– 0 0 2 0 3 0
3.5.6 Periodic Convolution

If x(t) and y(t) are two periodic signal with period T and
Exponential Fourier
Ck
Exponential Fourier
Bk
then if z(t) is periodic convolution of x(t) and y(t), that is
1
z(t) = x( ) y (t )d x(t ) * y (t ) ...(36)
T T
If Zk are fourier series coefficient of z(t) then
Exponential Fourier
z(t ) Series coefficient
Zk Ck Bk
Proof:
1 j o kt
Since, Zk = z (t ) e dt
T T
1 1 j o kt
= x( ) y (t )d e dt
T T
T T
1 1 j o kt
Zk = x( ) y (t )e dt d
T T
T T
Bk e j 0 k
We know that, y (t ) Bk , y (t ) Bk e j ok
1 j ok
Zk = x( ) Bk e d
T T
Zk = Bk Ck ...(37)
Study Note
Periodic convolution in time domain lead to multiplication in frequency domain,
Exponential Fourier
x(t ) y(t ) Series coefficient
Bk Ck
3.5.7 Multiplication in Time Domain

If x(t) and y(t) are periodic with period T and

Exponential Fourier
Ck
Exponential Fourier
Bk
Exponential Fourier
then, Z(t) = x(t ) y (t ) Series coefficient
Zk Cn Bk n
n
Thus, Zk is discrete time convolution of Ck and Bk.

Proof:
1 j o kt
Zk = z (t ) e dt
T T
z(t) = x(t ) y (t ) Z k exp ( j o kt )

k
Thus, z(t) = Cn exp ( j o nt ) Bm exp ( j o mt )

n m
Let, k = n + m, m = k – n
z(t) = Cn Bk n exp ( j o kt )
k n
Zk = Cn Bk n ...(38)
n
Study Note
Multiplication in time domain lead to convolution in frequency domain
Exponential Fourier
x(t ) y(t ) Series coefficient
Cn Bk n Ck Bk
n
3.5.8 Differentiation
Exponential Fourier
If x(t) is a periodic signal and x(t ) Series coefficient
Ck .
dx(t ) Exponential Fourier

Then, y(t) = Series coefficient
Bk j o k Ck
dt
Proof:
From definition, x(t) = Ck exp( j o kt )

k
Now, if we differentiate both side in above equation with respect to ‘t’ we get
dx(t ) j
y (t ) = ok Ck exp ( j o kt )
dt k
= [j ok Ck ] exp ( j o kt )
k

So, exponential Fourier series coefficient of y(t) are

Bk = [j ok Ck]
Study Note
dx(t )
If y (t ) then average value of y(t) will always be zero because differentiation removes dc component.
dt
3.5.9 Integration
Exponential Fourier
If x(t) is any periodic signal, and x(t ) Series Coefficient
Ck ,
t
then, y(t) = x( ) d Exponential Fourier
Series Coefficient
Bk Ck
Ck
Bk =
j ok
Proof: By definition, x(t ) = Ck exp( j o kt )

k
then, integrating above equation we get
t
Ck
y(t) = x( )d exp( j o kt )
k j ok
Study Note
t Ck
• So we can see that if signal x(t) has Fourier series coefficient Ck then x( ) d will have Fourier series .
j ok
Thus integration attenuates the magnitude of the high frequency component of the signal.
• Since high frequency components are main cause of sharp details thus integration make signal smooth.
Example 3.6
Find the Fourier series coefficient of signal z ( t ).

z(t)
– /2
t
– /2 3 /2 2
Solution 3.6

1
It will be very difficult to find out integration of z (t ) exp( j o kt ) dt . Now let us define a new signal w(t).
T T
d (t)
w(t) = z(t )
dt
4 10/
w(t) = y(t ),
t
–
[y(t) is signal in example (3.5)]

–10/
4
So Fourier series, Wk = Bk
0, k 0
= 20 k
2
sin , k 0
k 2
Wk
Using the differentiation property, Zk =
j ok
since period = 2 , so 0=1
Wk
Zk =
jk
Wk 20 k
So, Zk = 2 2
sin ; when k 0
jk j k 2
and Z0 is average value of Z(t) = 0
0, k 0
Zk = 20 sin(k / 2)
, k 0
j 2k2
Study Note
d
Suppose we have to find Fourier series coefficient of any signal z(t) and we do this by finding w (t ) z (t ), then average
dt
Wk
value of w(t) will be zero always even if average value of z(t) 0. So if we define Zk then value of Zk will be correct
j 0k
for all values of k except k = 0. So to find Z0 we simply find average value of Z(t).
Example 3.7
Find the exponential Fourier series coefficient of signal a(t).

a (t)
10
t
– /2 /2 3 /2 5 /2 7 /2
Solution 3.7
d
Again we find, b(t) = a(t )
dt
b(t) = w(t) (of example 3.6)
So, Bk = Wk
0, k 0
= 20 k
2
sin , k 0
k 2
Bk
and Ak = , k 0
j 0k
20 k
= sin , k 0
j 2k2 2
and A0 = Average value of a(t)
1
= a(t ) dt
2 2
A0 = 5
5, k 0
Ak = 20 k
sin , k 0
j 2k2 2
3.5.10 Conjugation in Time Domain

If x(t) is a periodic signal with period T, if
Exponential Fourier
Ck
Exponential Fourier
then, y( t ) x (t ) Series coefficient
Bk C k
1 j o kt
Ck = x(t ) e dt ...(39)
T T
1 j o kt
and Bk = x (t ) e dt ...(40)
T T

Conjugating both side in equation (39),

1 j o kt
Ck = x (t ) e dt
T T
Now replacing k by –k we get
1 j o kt
Ck = x (t ) e dt ...(41)
T T
so using (40) and (41) we get
Bk = C k
Thus if x(t) is real, that is

x(t) = x (t)
Ck = C k [Fourier series coefficient are conjugate symmetric]
if x(t) is imaginary, that is
x(t) = –x (t)
Ck = C k [Fourier series coefficient are conjugate anti-symmetric]
3.5.11 Parseval’s Theorem for Power Signals

Since all periodic signals are power signal, if x(t) is a periodic signal and
Exponential Fourier
Ck
T
1 2 2
then, Power of x(t) = x(t ) dt Ck
T 0 k
The Parseval’s relation state that total average power in a periodic signal equals the sum of average power in
all of it’s harmonics.
x(t) = Ck exp ( j o kt )
k
2
The power of Ck exp(j okt) is Ck and exp(j 0kt) are orthogonal for different values of k. So
1 2 2
x(t ) dt = Ck
T T k
Example 3.8
Suppose we are given the following information about a signal x(t):

1. x(t) is real and odd.
2. x(t) is periodic with period T = 2 and has Fourier coefficients Xn.
3. Xn = 0 for n 1.

2
1 2
4. x( t ) dt 1.
2 0
Solution 3.8
Since, x(t) is real and odd (clue 1), its Fourier series coefficients are purely imaginary and odd. Therefore,
Xn = –X–n and X0 = 0
Also, since it is given that, Xn = 0 for n 1, the only unknown Fourier series coefficients are X1 and X–1. Using
Parseval’s relation,
T
1 2 2
x(t ) dt = Xn
T 0 n
for the given signal we have,

2
1 2
Xn
2
1
x(t ) dt =
2 0 n 1
2 2
X1 X 1 =1
2
2 X1 =1
2 1
X1 =
2
1 1
Therefore, X1 = X1 j or X1 = X1 j
2 2
two possible signals which satisfy the given condition are,
1 1
x1(t) = j ej t
j e j t
2 2
1 ej t
e j t
= j 2j 2 sin( t )
2 2j
1 1
x2(t) = j ej t
j e j t
2 2
1 ej t
e j t
= j 2j 2 sin( t )
2 2j
3.6 Finding Signal from It’s Fourier Series Coefficient

If exponential Fourier series coefficients Ck are know then signal can be obtained using formulae,
x(t) = Ck exp( j 0 kt )
k
Here, Ck = Ck e j k
where Ck is magnitude of Ck and k is phase.

Example 3.9
A continous-time periodic signal x(t) is real-valued and has a fundamental period T0 = 8. The non zero
Fourier series coefficients(Xn) for x(t) are
X 1 = X–1 = 2, X 3 = X *-3 = 4j
Find x(t).
Solution 3.9
The signal x(t) is periodic with fundamental frequency o=2 /8 = /4, we have
x(t) = X n e jn ot
= X1 e j ot X 1 e j ot
X 3 e j3 ot X 3 e 3 j ot
= 2 ej / 4t
2e j / 4t
4 j e j3 / 4t
4je j 3 / 4t
j t j t j3 t j3 t
e 4 e 4 e 4 e 4 3
= 4 8 4cos t 8sin t
2 2j 4 4
3
= 4 cos t 8 cos t
4 4 2
Example 3.10
k
If signal x(t) is periodic with period 2 and it’s Fourier series coefficient Xk (1 / 3) . Then the expression
of x(t) is _______ .
Solution 3.10
x(t) = X k exp( j o kt )
k
2
Here, = 1
o 2
k
1
x(t) = exp ( jkt )
k 3
k 1 k
1 1
= exp( jkt ) exp( jkt )
k 0 3 k 3
1 (3 exp( jt )) 1
=
exp( jk ) 1
1 1
3 3 exp( jt )

3 1 9exp( jt ) 3 3 exp( jt )
= =
3 exp( jt ) 3 exp( jt ) 1 (3 exp( jt )) (3exp( jt ) 1)
8 exp( jt )
x(t) =
10 exp( jt ) 3 exp(2 jt ) 3
3.7 To find Fourier Series Coefficient of Signal x(t)

Case-1: If signal x(t) is a sum of sine, cosine or complex exponentials then to find the exponential Fourier
series coefficients we should follow this procedure.
Step-1: First of all find the period of signal x(t), if in case x(t) is non-periodic then Fourier series coefficient
cannot be found. If signal is periodic then move to step-2.
Step-2: Since, period of x(t) is T and o = 2 /T. Now expand each sine and cosine in terms of complex
exponentials,
ej e j
ej e j
sin = and cos
2j 2
Step-3: Now x(t) has only complex exponential terms, Since,
x(t) = Ck exp( j o kt )
k
Compare each term in x(t) with Ck exp(j okt) to find value of k and Ck.
Case-2: For rest of the cases we can easily find Fourier series coefficient Ck using formulae,
1
Ck = x(t ) exp( j o kt ) dt
T T
Example 3.11
For the continuous-time periodic signal,

2 5
x(t) = 2 cos t 4sin t
3 3
Determine the fundamental frequency o and the Fourier series coefficients Xn such that,
x(t) = X n e jn ot
Solution 3.11
2 5
Consider the given signal, x(t) = 2 cos t 4sin t
3 3
2 2 5 5
j t j t j t j t
e 3 e 3 e 3 e 3
= 2 4
2 2j

1 j2 3t 1 j2 t 2 j5 t 2 j5 t
= 2 e e 3 e 3 e 3
2 2 j j
1 j2 3t 1 j2 t 5 t 5j t
x(t) = 2 e e 3 2je 3 2je 3
2 2
Comparing with, x(t) = X0 X2 e j 2 0t X 2e j 2 0t

X5 e j 5 0t X 5e j 5 0t
The non zero Fourier series coefficients are

1
X 0 = 2, X2 X 2 , X5 X 5 2j
2
Example 3.12
If Fourier series coefficients of x(t) are Ck then find Fourier series coefficient of
(a) x(t – to) + x(t + to) (b) Even {x(t)}
d2 x( t )
(c) Real {x(t)} (d)
dt 2
(e) x(3t – 1)
Solution 3.12
(a) y(t) = x(t – to) + x(t + to)
If y(t ) Bk
Using time shifting property,
Bk = Ck exp(–j okto) + Ck exp(j okto)
Bk = 2 cos( o kto) Ck
x (t ) x ( t )
(b) y(t) = Even {x(t )}
2
If y(t ) Bk
Ck C k
then, Bk =
2
x(t ) x (t )
(c) y(t) = Real {x(t)} =
2
If y(t ) Bk
Ck C k
then, Bk =
2
d2
(d) y(t) = x(t )
dt 2
If y(t ) Bk
then, Bk = (j ok) ( j 0k) Ck

2
= o k 2 Ck
(e) y(t) = x(3t – 1)
If y(t ) Bk
If period of x(t) is T then that of y(t) is T/3, so if fundamental frequency of x(t) is o then that of y(t) is
o =3 o
Now let, z(t) = x(3t)
so, Zk = Ck because scaling do not effect Fourier series coefficient
1
Now, y(t) = z t
3
2
So, Bk =
T 0
1
= Z k exp j3 ok
3
Bk = Ck exp(–j ok)
Example 3.13
Suppose we are given the following information about a signal x(t).

1. x(t) is real.
2. x(t) is periodic with period T = 6 and has Fourier coefficients Xn.
3. Xn = 0 for n = 0 and n > 2.
4. x(t) = –x(t – 3).
3
1 2 1
5. x( t ) dt .
6 3
2
6. X1 is a positive real number.

Show that x(t) = A cos(Bt + C), and determine the values of the constants A, B and C.
Solution 3.13
The given signal x(t) is periodic with period T = 6 and fundamental frequency o=2 /6 = /3. Since x(t) is real,
Xn X n . Since Xn = 0 for n = 0 and n > 2, the only unknown Fourier series coefficients are X1, X–1 ,X2, and X–
2. Also x(t) is half wave symmetric so X2, and X–2 will be zero
x(t) = X 1 e j ot X 1 e j ot
Since, X1 is a positive real number, X1 = X–1,
ej ot e j ot
x(t) = 2 X1
2
x(t) = 2 X1 cos( o t )
t
x(t) = 2 X1 cos
3
Now using Parseval’s relation on clue 5, we get

T/2
1 2 Xn
2
x(t ) dt =
T T/2
n
3
1 2 1
x(t ) dt = X1 2 X 1
2
6 3 2
2 1
2 X1 =
2
1
X1 =
2
Since, X1 is positive, we have X1 = X–1 = 1/2.
t
Therefore, x(t) = cos
3
Example 3.14
Consider the periodic impulse train T0 ( t ) shown in figure and defined by
T0 ( t ) (t kT0 )
k
(a) Determine the complex exponential Fourier series of T0 ( t ).
(b) Determine the trigonometric Fourier series of T0 ( t ).
T0 (t)
(t) (t – T0)
t
–T0 0 T0 2T0
Solution 3.14
2
T0 (t ) Ck e jk ot
(a) Let, = , o
k T0
Since (t) is involved, we obtain

T0 / 2
1 jk o t 1
Ck = (t ) e dt
T0 T0 / 2
T0
Hence, we get T0 (t ) = (t kT0 )

k
1 2
= e jk ot , o
T0 k T0
a0
(b) Let, T0 (t ) = (ak cos k ot bk sin k ot)
2 k 1

2
o =
T0
Since T0 (t ) is even bk = 0 and ak is equal to 2Ck
2
ak = 2Ck = T
0
1 2 2
Thus, we get T0 (t ) = cos k ot, o
T0 T0 k 1
T0
Study Note
• We can see that in case of impulse train the Fourier series coefficient are constant Ck = 1/T0, in case of square wave
1
Fourier series coefficient are C k inversely proportional to k, in case of triangular wave Fourier series coefficients
k
1
are Ck inversely proportional to k2.
k2
• Thus in impulse train all harmonics have equal contribution, in square wave higher harmonics have less contribution
because value of Ck decrease with increase in k, in triangular wave decrease Ck is more with increase in k so higher
harmonics have much less contribution.
Example 3.15
Find the exponential Fourier series coefficient of signal x(t).

x(t)
t
– 0 2 3
Solution 3.15
dx(t )
Let, y(t) =
dt
y (t)
A (t – )
A (t)
t
– 0 2 3
–A/

A
So, y(t) = A (t k )
k
The Fourier series coefficient of y(t) = The Fourier series coefficient of dc signal (–A/ ) +
The Fourier series coefficient of impulse train
A
The Fourier series coefficient of impulse train is Ik
and Fourier series coefficient of dc signal is DCk,

A
, k 0
DCk =
0, k 0
So, Y k = Ik + DCk
0, k 0
Yk = A
, k 0
d
Since, y(t) = x(t )
dt
Yk
So, Xk = for k 0
j ok
A
= , k 0
j ok
Since time period = , o=2

A
Xk = , k 0
j2 k
A
X 0 = Average value of x(t) =
2
A
, k 0
2
So, Xk = A
, k 0
j2 k
Example 3.16
Find the exponential Fourier series coefficient of x(t)

m 2m
x(t) = t t
m 3 3
Solution 3.16
Let us plot signal x(t),

x (t)
2 (t –2/3)
2 (t)
(t –1/3) (t –1)
t
0 1/3 2/3 1
So we can see that period is 2/3 and in one period there are two impulses.
2/3
1 j o kt
Ck = x (t ) e dt
2/3 0
2
o = 3
2/3
2/3
3 1 j 3 kt 3 j 3 k(1 / 3)
Ck = 2 (t ) t e dt = 2 e
2 0
3 2
3 j k 3 j k
Ck = 2 e 3 e
2 2
Example 3.17
Consider the signal x(t), the power of x(t) till its second harmonic is _____ W.
x (t)
15
t
–10 –5 –3 0 3 5 10
Solution 3.17
First of all lets find exponential Fourier series coefficient of x(t), Ck
1 j o kt
Ck = x (t ) e dt
T T
5 2 5 2
1 j kt 1 j kt
= x(t ) e 10 dt 15 e 10 dt
10 5
10 5
2 2 3 3
15 j 3k j 3k 15 j k j k
= [e 10 e 10 ] = [e 5 e 5 ]
j2 k 2j k
15 3
Ck = sin k
k 5
2
2
The power in x(t) till 2nd harmonic is Ck .
2

C0 = 9,
15 3
C1 = C–1 = sin
5
15 6
C2 = C 2 sin
2 5
2
2
Power fill 2nd harmonic = Ck 126.16 Watt
2
Example 3.18
Find the magnitude of 2nd harmonic in the signal x(t) = 3 sin(4t + 30°) – 4 cos(12t + 60°).
Solution 3.18
Since, x(t) = 3 sin(4t + 30°) – 4 cos(12t + 60°)
Let, time period of x(t) is T.
2
Period of 3 sin(4t + 30°) is T1 = .
4 2
2
Period of 4 cos(12t + 60°) is T2 .
4 6
T1 3
So, is rational , LCM of denominator or ratio =1.
T2 1
T = T1
2
o =4
Thus if we expand x(t) we get
3 j 4t j 30 3 j 4t j 30 4 j12t j 60 4 j12 t
x(t) = e e e e e e e e j 60
2j 2j 2 2
Now if we compare each term with Ck e j o kt i.e. Ck e j 4kt we can see that x(t) has terms with k = ±1 and k = ±3
and not for k = ±2. Thus x(t) has no 2nd harmonic. So answer is 0.
Example 3.19
Which harmonics will be present in these signals.

x (t)
1
A
(a) t
–T 0 T 2T
–A

x 2(t)
A
(b)
t
–T/2 0 T/2 T
d
(c) y 1( t ) x (t )
dt 1
d
(d) y 2 ( t ) x (t )
dt 2
Solution 3.19
(a) Signal x1(t) is odd, thus it will have only sine terms in trigonometric representation of Fourier series.
(b) If we shift x2(t) on amplitude axis by –A/2 then we can see that it is half wave symmetric also. So x2(t) is
even and it will have only odd harmonics except C0 as it has non zero average value of signal. Thus x2(t)
will have odd cosine terms.
dx1 (t )
(c) Since x1(t) was odd so y1(t) = , which will be even thus y1(t) will have only cosine terms in trigonometric
dt
Fourier series representation.
d
(d) y 2 (t ) x (t ), since x2(t) was even so y2(t) will be odd. Thus y2(t) will have only odd harmonics of sine
dt 2
terms and zero dc value.
Example 3.20
Determine the time signal corresponding to the magnitude and phase spectra shown in figure with 0 = .
2 2
Cn
1 1
n
–4 –3 –2 –1 0 1 2 3 4
/4
/8
Cn
–4
n
–3 –2 –1 0 1 2 4
– /8
– /4
Solution 3.20
From figure, we have
C3 = 1 1e j /4
4

C–3 = 1 1 ej /4
4
C4 = 2 2 ej /8
8
j /8
C–4 = 2 2 e
8
By definition, the exponential Fourier series is
x(t) = Cn e jn 0t
= 2e j /8
e j 4 ot
1e j /4
e j3 ot
e j /4
e j3 ot 2e j /8 j 4 o t
e
j[( /8) 4 t ]
= 2{ e e j[( /8) 4 t ]
} { e j[( /4) 3 t ]
e j[( /4) 3 t ]
}
= 4cos 4 t 2cos 3 t
8 4
Example 3.21
Find the time domain signal whose Fourier series coefficient is given by,
Cn = j (n – 1) – j (n + 1) + (n – 3) + (n + 3), o =
Solution 3.21
We have, x(t) = Cn e jn ot Cn e jn t
n n
= jej t – je–j t + ej3 t + e–j3 t
= 2 cos3 t – 2 sin t
Example 3.22
For the signal shown in figure obtain the Fourier series representation using time differentiation property.
x (t)
–T/2 T/2
t
–T – /2 0 /2 T
Solution 3.22
Differentiating x(t) with respect to time, we get the signal y(t) as shown in figure.

y (t) = [dx (t)/ dt]
/2
t
–2T –3T/2 –T –T/2 – /2 0 T/2 T 3T/2 2T
–1
2
Here, o =
T
Fourier series of y(t) is given by,
T /2
1 jn ot
Dn = y (t ) e dt
T T /2
T /2
1 jn ot
= t t e dt
T T /2
2 2
2 2
1 jn T 2 jn
= e e T2
T
2j n
Dn = sin
T T
d
we have, y(t) = x(t )
dt
Let, Cn be the Fourier series of x(t). Using time differentiation property, we get
Dn = jn 0Cn
1
Cn = Dn
jn 0
T 2j n 1 n
= sin sin
jn 2 T T n T
sin( n / T ) n
= sinc
T (n / T ) T T
Example 3.23
The waveform shown in figure (a) is applied to the circuit shown in figure (b). Determine the rms value of
third harmonic of current in the circuit.

v(t)
25 25 0.5 H i (t)
– t
0 2
–25
v (t)
(a) (b)
Solution 3.23
First of all lets find Ck of v(t),
2
1 j o kt
Ck = v(t ) e dt
2 0
Since, T0 = 2 , o=1
2
1 jkt
Ck = v(t ) e dt
2 0
2
1 jkt jkt
= 25 e dt 25 e dt
2 0
1 jk j2 k j k
= 25( e 1) 25( e e )
2 ( jk )
25 jk j k 25 jk
= (e 1) (1 e ) = (e 1)
2 ( jk ) ( jk )
25
Ck = [1 ( 1)k ]
j k
50
, k is odd
So, Ck = j k
0, k is even
So, v(t) = Ck exp ( j o kt ) Ck exp ( jkt )

k k
For third harmonic k = 3 and –3

So, 3rd harmonic component in v(t) is
= C3 exp(3jt) + C–3 exp(–3jt)
50 3 jt 50 3 jt
= e e
j3 j3
100
3rd harmonic = sin 3t
3

100
sin 3t
v (t ) 3
So, 3rd harmonic component of i(t) =
R j L 25 j (3) (0.5)
= 0.423sin(3t 3.4 )
0.423
So, rms value is 0.299 A .
2
3.7 Summary
Property Section Periodic signal Fourier series coefficients
x (t ) Periodic with period T and ak
y (t ) Fundamental freqeuncy 0 = 2 /T bk
Linearity 3.5.1 A x( t ) B y (t ) Aak Bbk
jk 0 t0 ak jk(2 / T )t0
Time shifting 3.5.2 x(t t0 ) ak e e
Frequency shifting e jM 0t
x(t ) e jM(2 / T )t
x(t ) ak M
Conjugation 3.5.6 x(t ) a k

Time reversal 3.5.3 x( t ) ak
Time scaling 3.5.4 x( t ), 0 (periodic with period T/ ) Tak bk
Periodic convolution x( ) y (t )d a l bk l
T l
2
Multiplication 3.5.5 x( t ) y( t ) jk 0ak jk ak
T
dx(t ) 1 1
Differentiation ak ak
dt jk 0 jk 2 / T
ak a*k
t Re{ak } Re{a k }
(finite valued and
Integration x( t ) dt Im{a k } Im{a k }
periodic only if a0 = 0)
ak a k
ak a k
Conjugate symmetry for real signals 3.5.6 x( t ) real a k real and even
Real and Even signals 3.5.6 x( t ) real and even a k purely imaginary and odd
Real and Odd signals 3.5.6 x(t ) real and o dd Re{a k }
x e( t ) ev {x(t )} [ x( t ) real]
Even-Odd decomposition j Im{ak }
x0 (t ) = od{x( t )} [ x( t ) real]
Parseval's relation for periodic signals
1 2 2
x( t ) dt ak
T T
k
Table-3.1

Time domain signal x(t) Fourier series coefficient C k
Real Conjugate symmetric

Imaginary Conjugate anti-symmetric
Even Even
Odd Odd
Real and Even Real and Even
Real and Odd Imaginary and Odd
Imaginary and Even Imaginary and Even
Imaginary and Odd Real and Odd
Table-3.2
Study Note
In the Table 3.2 the arrow is both sided that is if x(t) is real then Ck is conjugate symmetric and if Ck is real then x(t) is
conjugate symmetric and if Ck is real then x(t) conjugate symmetric. This apply to all in Table 3.2.

fundamental period T = /6 and Fourier series

coefficients ak, it is found that,
S
x(t ) y(t ) x(t )
1 Subjective Practice Problems For what values of k is it guaranteed that ak = 0?
t, 0 t 1
1. A continuous-time periodic signal x(t) is real valued and 5. Let, x(t) =
2 t, 1 t 2
has a fundamental period T = 8. The non zero Fourier
be a periodic signal with fundamental period T = 2 and
series coefficients for x(t) are
Fourier coefficients ak:
a1 = a–1 = 2, a3 a 3 4j (a) Determine the value of a0.
Express x(t) in the form (b) Determine the Fourier series representation of
dx(t)/dt.
x(t) = Ak cos( k k) (c) Use the result of part (b) and the differentiation
k 0
property of the continuous-time Fourier series to
help determine the Fourier series coefficients of x(t).
2. Consider three continuous-time periodic signals whose
Fourier series representations are as follows: 6. Consider the following three continuous-time signals
100 k 2 with a fundamental period of T = 1/2.
1 jk t
x1(t) = e 50 x(t) = cos(4 t)
k 0 2
y(t) = sin(4 t)
100 2 z(t) = x(t) y(t)
jk t
x2(t) = cos(k ) e 50 (a) Determine the Fourier series coefficient of x(t).
k 100 (b) Determine the Fourier series coefficients of y(t).
2
(c) Use the results of parts (a) and (b), along with the
100
k jk t
multiplication property of the continuous-time
x3(t) = j sin e 50
k 100 2 Fourier series, to determine the Fourier series
Use Fourier series properties to help answer the following coefficients of z(t) = x(t) y(t).
questions: (d) Determine the Fourier series coefficients of z(t)
(a) Which of the three signals is/are real valued? through direct expansion of z(t) in trigonometric
(b) Which of the three signals is/are even? form, and compare your result with that of part (c).
3. Suppose we are given the following information about a 7. Let x(t) be a periodic signals whose Fourier series
signal x[n]: coefficients are:
(i) x[n] is a real and even signal. 2, k 0
(ii) x[n] has period N = 10 and Fourier coefficient ak. ak = k
1
(iii) a11 = 5 j , otherwise
2
9
1 2 Use Fourier series properties to answer the following
(iv) x[ n] 50
10 n 0 questions:
Show that, x[n] = A cos(Bn + C), and specify numerical (a) Is x(t) real?
values for the constants A, B and C. (b) Is x(t) even?
(c) Is dx(t)/dt even?
4. Consider a continuous-time ideal low-pass filter S whose
frequency response is 8. Suppose we are given the following information about a
1, 100 continuous-time periodic signal with period 3 and
H( j ) = Fourier coefficients ak:
0, 100
1. ak = ak + 2 2. ak = a–k
When the input to this filter is a signal x(t) with
0.5 2
3. x(t ) dt 1 4. x(t ) dt 2
0.5 1

Determine x(t). (a) constant coefficients.

(b) only cosine functions.
9. If the Fourier series coefficients of a signal are periodic,
(c) only sine functions.
then the signal must be
(d) orthogonal functions.
(a) continuous time, periodic
(b) discrete time, periodic 14. What is the magnitude of the exponential Fourier series
(c) continuous time, non-periodic coefficient of the fundamental term of the periodic signal
(d) discrete time, non-periodic x(t) = 3 + 2 cos(3t) + sin(3t)
10. If a periodic function f( t) of period T satisfies 3 5

(a) (b)
2 2
T
f (t ) f t , then in its Fourier series expansion, 1
2 (c) (d) 1
2
(a) the constant term will be zero.
(b) there will be no cosine terms.
(c) there will be no sine terms.
(d) there will be no even harmonics.
11. A periodic signal which can be expanded in Fourier series

(a) is a power signal.
(b) is an energy signal.
(c) is neither a power signal or an energy signal.
(d) can be either a power signal or an energy signal
depending on the nature.
12. The Fourier series of a real, even periodic signal will

contain only
(a) cosine terms (b) sine terms
(c) even terms (d) odd harmonic
13. The Fourier series representations are based on using

Chapter
Laplace Transform
4
Introduction
In the next chapter we will study Fourier transform X( ) of a time domain signal x(t) where we can get
information of frequency component of signal x(t). Before studying Fourier transform we will study Laplace
transform to make understanding of fourier transform simpler.
Lets see what is Fourier transform of signal x(t).
j t
X( ) = x(t ) e dt ...(1)
The Fourier transform X( ) is valid only when

• Signal is absolutely integrable.
• Signal has finite number of discontinuities in a finite interval of time and each discontinuity should
be finite.
• Signal should have finite number of maxima and minima in a finite interval of time.
These conditions are Dirichlet conditions (these conditions are sufficient but not necessary). So if a
signal is absolutely integrable then we can do Fourier analysis for getting it’s frequency analysis. If signal is not
absolutely integrable then we use Laplace transform to get frequency analysis. Examples are eat u(t), a > 0; e–
at, – < t < ; tu(t); and other signals that are not absolutely integrable. The Fourier transform does not exist
for signals that are not absolutely integrable, so the Fourier transform based methods cannot be employed in this
class of problems. The difficulty could be resolved by generalizing the Fourier transform so that the signal x(t) is
expressed as a sum of complex exponential est, where s = + j and thus is not restricted to the imaginary axis
only. This is equivalent to multiplying the signal by an exponential convergent factor. For example, e– t eat u(t)
satisfies Dirichlet’s conditions for > .
Another importance of Laplace transform is: generally a linear time invariant (LTI) system is described
by differential equations. The response of a system for a given input is obtained by solving the differential
equations relating its output and input signals. We know that the solution of higher order differential equations
is quite tedious and time consuming compared to the solution of algebraic equations. So the Laplace transform
is used to solve the differential equations. Laplace transform is a powerful mathematical tools used to convert
the differential equations into algebraic equations. It is a simple and systematic method which provides the
complete solution in one stroke by taking into account the initial conditions in a natural way at the beginning of
the process itself. For solving the differential equations using Laplace transform, we take the Laplace transform
of the differential equations (i.e. convert the differential equations in time domain into algebraic equations in

laplace Transform 145
frequency domain), insert the initial conditions, solve the resultant algebraic equations (i.e. get the solution in
s-domain) and take the inverse Laplace transform of the solution (to get the solution in time domain).
The Laplace transform comes in two varieties: Bilateral, or two-sided Laplace transform, and unilateral,
or one-sided Laplace transform.
Study Note
The bilateral Laplace transform offers insight into the nature of system characteristics such as stability, causality,
and frequency response. The unilateral Laplace transform is a convenient tool for solving differential equations
with initial conditions.
4.1 The Bilateral Laplace Transform

Consider a case when eCt is input to LTI system.
LTI
Ct
Input = e System Output
Impulse
response = h(t )
Fig. 4.1 : LTI system with eCt as input
Ct
If H( s) h(t ) e st
dt then output is e H (s )
s C
. This we have proved in previous chapter. The variable
s in H(s) is +j ,
s = +j ...(2)
H(s) is called Laplace transform of h(t). Thus for any arbitrary signal x(t), Laplace transform can be
defined as,
st
L(x(t)) = X (s ) x (t ) e dt ...(3)
( j )t
X (s ) = x(t ) e dt
t j t
= x(t ) e e dt
Thus we can see that X(s) is Fourier transform of x(t) e– t. Thus when = 0 we can say that,
L[x(t)] = f [x(t)] ...(4)
Since X(s) is Fourier transform of x(t) e– t, thus x(t) e– t should follow all Dirichlet conditions. That is
x(t) e– t should be absolutely integrable.
Study Note
This means Laplace transform X(s) is defined when x(t) e– t is absolutely integrable.
That is, x(t ) e t

dt ...(5)

146 Signals and Systems
In equation(5) we can see that (Real part of s) is the variable . The values of s for which equation (5)
is valid for a signal x(t) is called Region of convergence (ROC) of X(s). Since X(s) is defined only when
equation (5) is satisfied. This means X(s) is defined for those values of where equation (5) is satisfied.
Remember: Laplace transform is valid only when value of lie within ROC.
For example: Let’s take Laplace transform of e–at u(t). Thus,
at st
X (s ) = e u(t ) e dt {we assume a is real.}
( a s)t (a j )
= e dt e dt
0 0
1 (a j )
= [e 1]
(a j )
Now for finite X(s) we want e–(a + +j < .
Since e–j will always have magnitude equal to 1, the condition will be satisfied when, e–(a + ) < .
This will be satisfied when (a + ) > 0 because e–(a + ) will be 0 when (a + ) > 0.
Thus X(s) is finite when, a + >0 a
1 (a j )
X (s ) = [e 1]
(a j )
1 1
=
a j a s
1
X (s ) = , a
a s
In above example Laplace transform is valid any when > –a .
Example 4.1
Find Laplace transform of x ( t ) = e –(3 + 5j) u ( t ).

Solution 4.1
Now applying formulae, X (s ) = st (3 5 j ) t ( j )t

x(t ) e dt e e dt
0
[3 5 j j ]t
= e dt
0
1 (3 5j j )
= [e 1]
(3 5j j )
Now for finite X(s) we want (3 + ) > 0 > –3
1
X(s ) = , > –3
(3 5 j s)

Poles and Zeros of Laplace Transform

The most commonly encounted form of the Laplace transform in engineering is a ratio of two polynomials
in s, i.e.,
bms m bm 1s
m 1
... b1s b0
X (s ) = n n 1
...(6)
a ns an 1s ... a1s a0
It is often convenient to factor the polynomials in the numerator and denominator, and to write the
transfer function in terms of those factors.
N( s) ( s z1 ) ( s z2 )...( s zm 1 ) ( s zm )
X (s ) = K ...(7)
D( s) ( s p1 )( s p2 )...( s pn 1 ) ( s pn )
where the numerator and denominator polynomials, N(s) and D(s), have real coefficients and K = bm/an.
As written in the above equation, the zi’s are the roots of the equation N(s) = 0, and are defined as zeros, and
the pi’s are the roots of the equation D(s) = 0, and are defined as poles. In equation (7) the factors in the
numerator and denominator are written so that, when s = zi the numerator N(s) = 0 and the Laplace transform
i.e.,
Lim X( s) = 0
s zi
and similarly when s = pi the denominator polynomial D(s) = 0 and the value of the Laplace transform
becomes unbounded, that is
Lim X( s) =
s pi
Remember: The pole of a Laplace transform is where X(s) tends to and Zero of a Laplace transform is where
X(s) tends to 0.
Example 4.2
For each of the following integrals, specify the values of the real parameter which ensures
that the integral converges:
0
5t ( j )t 5t ( j )t
1. e e dt 2. e e dt
0
5
5t ( j )t 5t ( j )t
3. e e dt 4. e e dt
5
5t ( j )t
5. e e dt
Solution 4.2
1 . Consider the given integral,
5t ( j )t ( 5) t j t
e e dt = e e dt
0 0
1 ( 5) t j t
= e e
(5 j ) 0

j t
Note that, e 1 regardless of the value of t. Therefore, as t , e–( + 5)t e–j t 0 only if
( + 5) > 0 or, < –5. Thus,
( 5)t j t 0, ( 5) 0 5
Lim e e =
t , ( 5) 0 5
( 5)t j t 1 1
Therefore, e e dt = [0 1] for > –5
0
(5 j ) (5 j )
2 . Consider the given integral,

0 0
5t ( j )t ( 5) t j t
e e dt = e e dt
1 ( 5) t j t 0
= e e
(5 j )
j t
Note that, e 1 regardless of the value of t. Therefore, t – , e–( + 5)t e–j t 0 only if ( + 5) < 0
or < –5, and e–( + 5)t e–j t only if ( + 5) > 0 or, > –5. Thus,
0
( 5)t j t 1 1
Therefore, e e dt = [1 0] for < –5
(5 j ) (5 j )
5
3 . The given integral may be written as, e (5 )t
e j t
dt .
5
Since this integral has a finite range of limit, it has a finite value for all finite values of .
4 . The given integral may be written as,
0
(5 )t j t (5 )t j t (5 )t j t
e e dt = e e dt e e dt
0
The first integral diverges for < –5. The second integral diverges for > –5. If = 5, then both the
integrals still diverge. Therefore, the integral does not converge for any value of .
5 . The given integral may be written as,
0
(5 )t j t (5 )t j t
e e dt e e dt
0
The first integral converges for < 5. The second integral converges for > –5. Therefore, the
integral converges for 5.
Example 4.3
Find Laplace transform of – e – at u (– t ).

Solution 4.3
st at st
X(s ) = x(t ) e dt e u( t ) e dt

0 0
at st (a j )
= e e dt e dt
1 a j )
= [1 e( ]
(a j )
For finite X(s) we need a + <0 < –a
1
X (s ) = , <–a
s a
4.2 Properties of ROC

Note that the Laplace transforms for the signals e–at u(t) and –e–at u(–t) are identical except for their
ROCs. Therefore, for a given X(s), there may be more than one inverse transform, depending on the ROC. In
other words, unless the ROC is specified, there is no one-to-one correspondence between X(s) and x(t). In this
section, we explain properties of the ROC of various classes of signals.
1. The ROC of X(s) consists of strips parallel to the j -axis in the s-plane.
2. If X(s) is rational, then the ROC must not contain any poles.
3. If x(t) is of finite duration and is absolutely integrable (i.e. x(t ) dt ), then the ROC is the
entire s-plane.
4. If x(t) is right-sided and of infinite duration (i.e., x(t) = 0 for all t < T1 for some finite T1), then the
ROC is the region in the s-plane to the right of the right most pole. That is ROC, > real part of
right most pole.
5. If x(t) is left-sided and of infinite duration (i.e., x(t) = 0 for all t > T2 for some finite T2), then the
ROC is the region in the s-plane to the left of the left most pole, that is ROC, < real part of
smallest pole.
6. If x(t) is two-sided and infinite duration (i.e. the signal is of infinite extent for both t < 0 and t > 0),
then the ROC will consist of a strip in the s-plane.
Example 4.4
Find Laplace transform of

( a ) x ( t ) = e –2 t u ( t ) – e –3 t u ( t )
( b ) x ( t ) = – e –2 t u (– t ) + e –3 t u (– t )
at
e , 0 t T
( c ) x( t )
0, elsewhere
Solution 4.4
(a) x(t ) = e–2t u(t) – e–3t u(t)
Signal x(t) has two components.
1
1. e–2t u(t) has Laplace transform = .
s 2
So, pole is at –2, signal is right sided.

ROC is > –2,

2t Laplace transform 1
e u(t ) , 2
s 2
1
2. e–3t u(t) has Laplace transform = .
s 3
So, pole is at –3, signal is right sided so ROC is > –3.
Laplace 1 1 1
For x (t ) , ( 2) ( 3) = 2
, ( 2)
Transform s 2 s 3 s 5s 6
(b) x ( t ) = e–2t u(–t) + e–3t u(–t)
Signal x(t) has two signals.
1
1. e–2t u(–t) has Laplace transform , and pole is at –2, signal is left sided so ROC will be < –2.
s 2
1
2. e–3t u(–t) has Laplace transform and pole is at –3, signal is left sided so ROC will be < –3.
s 3
Laplace 1 1 1
x (t ) , ( 2) ( 3) , ( 3)
Transform s 2 s 3 s 2
5s 6
at
e , 0 t T
(c) x(t ) =
0, elsewhere
T
X (s ) = x(t ) e st dt e at
e st
dt
0
1 (a s )T
= (1 e )
(s a)
Now if we will look at X(s) we can see that at s = –a, X(s) is not so X(s) has no pole at s = –a.
d
(1 e ( a s )T )
Lim X( s) = Lim dt
a a d
(s a )
dt
X(–a) = T
So, ROC can be entire s-plane because no pole exist for X(s) at any location. Also at s = 0 and s =
X(s) is finite. So ROC is entire is s-plane.
Example 4.5
Determine the Laplace transform of

( a ) a unit impulse function x ( t ) = ( t )
( b ) a unit step function x ( t ) = u ( t )
( c ) a unit ramp signal x ( t ) = r ( t )
Solution 4.5
( a ) Given that, x( t ) = (t)
st st st
By definition, X(s ) = x(t ) e dt (t ) e dt e 1
t 0
Therefore, (t) 1 , ROC is the entire s-plane

( b ) Given that, x ( t ) = u( t )
st st st
By definition, X (s ) = x(t ) e dt u(t ) e dt e dt
0
1 st 1
= e [0 1], 0
s 0 s
1
X (s ) = , 0
s
1
Therefore, u(t ) , 0
s
( c ) Given that, x ( t ) = r(t) = t u(t), by definition
st st st
X (s ) = x(t ) e dt t u(t ) e dt te dt
0
1 st 1 st 1 st 1
= te e dt 0 e = [0 1], 0
s 0 s 0 s2 0 s2
1
X (s ) = 0
s2
1
Therefore, r(t ) , 0
s2
Example 4.6
Find Laplace transform of signal x ( t ) = e – a t when a > 0 and a < 0.

Solution 4.6
Case-1: When a > 0
a t
So, x(t ) = e e at
u(t ) e at
u( t )
at Laplace 1
e u(t ) , a
Transform s a
1
eat u( t ) Laplace
, a
Transform s a
Laplace 1 1
So for x(t ) ,( a) ( a)
Transform s a s a
1 1
, ( a = a)
s a s a
So, when a > 0 then Laplace transform exist as ROC is –a < < a.
x (t)
1.0
eat e –at
Figure: x(t) when a is positive

Case-2: When a is negative then also we get same expressions,
Laplace 1 1
x (t ) , ( a a)
Transform s a s a
Here if we look at ROC since a is negative ROC will be null, ROC = thus Laplace do not exist.
x (t)
eat e–at
Figure: x(t) when a is negative
Study Note
Laplace transform of a signal x(t) will not exist when ROC is a null, ROC = because in that case for no value of
Laplace transform will be finite.
Example 4.7
Find Laplace transform of x ( t ) = 1.

Solution 4.7
x ( t ) = 1 = u(t) + u(–t)
Laplace transform of u(t) is 1/s, since it is right sided signal and pole is at 0 so ROC is > 0.
Now, the Laplace transform of u(–t) is –1/s and it is left sided signal. Since pole is at 0 so ROC is < 0.
The ROC of x(t) is ( < 0) ( > 0) = .
So, Laplace transform of x(t) = 1 do not exist.
x (t)
1.0
REMEMBER We can see that why Laplace transform of x(t) =1 do not exist because x(t) is not absolutely
integrable and when we try to find x(t) e– t we can see that it will also be not integrable for any
value of .
And clearly we can see that when x(t) = u(t) for > 0 we can find x(t) e– t which is absolutely
integrable. Similarly in example 4.6 when a < 0, then x(t) is rising in both directions of t. So
here also x(t) e– t is not absolute integrable for any value of .
Thus we can say that if signal is rising both side or signal is constant from – < t < then
Laplace transform do not exist.

1.0
1.0
t t
Fig. : e– t when >0 Fig. : e– t when <0
4.3 Properties of Laplace Transform

4.3.1 Linearity
Laplace transform
If x1(t ) X1( s), ROC = R1
Laplace transform
and x 2 (t ) X 2 (s ), ROC = R2
then, ax1(t) + bx2(t) aX1(s) + bX2(s) with ROC containing R1 R2. ...(6)
As indicated, the ROC of X(s) is at least the intersection of R1 and R2, which could be empty, in which
case X(s) has no ROC, i.e., x(t) has no Laplace transform.
Proof : The Laplace transform of ax1(t) + bx2(t) is given by,
st
L[ax1(t) + bx2(t)] = [ax1(t ) bx2 (t )] e dt
st st
= a x1(t ) e dt b x2 ( t ) e dt
= aX1(s) + bX2(s)
Note: In case if R1 R2 = then Laplace transform of x1(t) + x2(t) do not exist.
Example 4.8
In this example, we illustrate the fact that the ROC for the Laplace transform of a linear
combination of signals can sometimes extend beyond the intersection of the ROCs for the
individual terms.
Solution 4.8
Consider, x ( t ) = x1(t) – x2(t)
where the Laplace transform of x1(t) and x2(t) are, respectively,
1
X1( s ) = , Re( s) 1
s 1
1
and X2( s ) = , Re( s) 1
( s 1) ( s 2)
The pole-zero plot, including the ROCs X1(s) and X2(s), is shown in figure (a) and (b).
1 1 s 1 1
X(s ) =
s 1 ( s 1) ( s 2) ( s 1) ( s 2) s 2

Thus, in the linear combination of x1(t) and x2(t), the pole at s = –1 is canceled by a zero at s = –1. The
pole-zero plot of X(s) = X1(s) – X2(s) is shown in figure (c). The intersection of the ROCs for X1(s) and
X2(s) is Re{s} > –1. However, since the ROC is always bounded by a pole or infinity, for this example the
ROC for X(s) can be extended to the left to be bounded by the pole at s = –2, as a result of the pole-zero
cancellation at s = –1.
Im Im Im
s-plane s-plane s-plane

R1 R2 R3
Re Re Re
–1 –2 –1 –2
(a) (b) (c)
Figure : Pole-zero poles and ROCs for (a) X1(s); (b) X2(s); (c) X1(s) –X2(s)
REMEMBER When even we are applying any property to find Laplace transform of a signal then find the
expression of Laplace transform using properties and do not find ROC using properties because
we can easily get ROC from expression of Laplace transform using properties of ROC
(Section 4.2).So learn how property effects the expression of laplace transform and no need to
learn the effect on ROC.
4.3.2 Time Shifting

If, x(t ) X(s) with ROC = R
then, x(t – t0) X(s) e–st0 with ROC = R ...(7)
Proof : The Laplace transform of x(t – t0) is given by
L[x(t – t0)] = st
x(t t0 ) e dt
A change of variables is performed by letting = t – t0, which also yields d = dt, – as t – ,

and – , and as t . Therefore,
s( t0 ) st0 s
L[x(t – t0)] = x( ) e d e x( ) e d
st0
L[x(t – t0)] = X( s) e
as
1 e
The laplace transform of u(t) is then laplace transform of u(t a) will be .
s s
1 e as
The laplace transform of r(t) is then laplace transform of r(t a) will be .
s2 s2
t T T A sT / 2 sT / 2
The laplace transform of A rect = u t u t (e e ).
T 2 2 s

Study Note
We can see that shifting operation by a in time domain lead to multiplication of e–as in laplace domain. Since
multiplication of e–as do not change the location of poles in laplace transform, also the shifting operation donot
change the format of the signal that is left sided signal remain left sided, right sided signal remain left sided signal,
finite duration signal remain finite duration. Thus ROC will not change by shifting operation.
4.3.3 Shifting in the s-domain

If, x(t ) X(s) with ROC = R
then, x ( t ) e so t X(s – so) with ROC = R + Re{s0} ...(8)
Proof : The Laplace transform of x(t) e so t is given by
L[x(t) e so t ] = [ x (t ) e s 0t ] e st
dt x (t ) e ( s so )t
dt = X(s – so)
REMEMBER • If X(s) has a pole or zero at a then X(s – so) will have pole or zero at a + so. So we can see
that shifting in s-domain by s0 leads to multiplication of e so t in time domain. So format of

signal do not change.
• If ROC of X(s) is R then X(s – s0) will have ROC = R + Re{s0}, because pole of X(s – s0) will
shift by s0 as seen in above point.
For example:
Laplace transform 1
u(t ) , ROC is >0
s
at Laplace transform 1
e u(t ) , ROC is > – Re{a}
s a
1
e j u(t ) Laplace transform
, ROC is >0
s j
j Laplace transform 1
e u(t ) , ROC is >0
s j
Laplace transform 1 1 1 s
cos( 0t ) u(t ) , 0 2
, 0
2 s j s j s
Laplace transform 1 1 1
sin( 0t) u(t ) , 0 , 0
2j s j s j s2 2
Study Note
We can see the in time shifting,
x (t ) X(s), x (t – a ) X(s) e–as

The shifting goes with same sign, but in frequency shifting
s0t
x(t) X(s), x (t ) e X(s – s0)
The shifting goes with opposite sign.

4.3.4 Time Scaling

If x(t ) X(s) with ROC = R
1 s
then, x ( at ) X with ROC = aR ...(9)
a a
Aside from the amplitude factor of 1/a, scaling in time domain by a factor of a corresponds to scaling in
the s-domain by a factor of 1/a. Also, for any value of s in the ROC of x(t), the value s/a will be in the ROC of
x(at); that is, the ROC associated with x(at) is a compressed (a > 1) or expanded (a < 1)version of the ROC of
x(t).
Proof : The Laplace transform of x(at) is
at
L[x(at)] = x( at ) e dt
Case-I: For a positive real constant ‘a’
st
L[x(at)] = x( at ) e dt
A change of variables is performed by letting = at, which also yields d = a dt, – as t – , and
as t . Therefore,
1 s
L[x(at)] = 1 x( )e ( s /a )
d = X
a a a
Case-II: For a negative real constant ‘a’
st
L[x( at)] = x ( at ) e dt
A change of variables is performed by letting = at, which also yields d = a dt, as t – , and
– as t . Therefore,
1 1 1 s
L[x(–at)] = x ( ) e (s /a ) d x( )e ( s /a )
d = X
a a a a
Combining the two cases, we obtain,
1 s
L[x(at)] = X , with ROC R1 = aR
a a
That is, for any value s in R [which is illustrated in Fig. 4.2 (a)], the value as will be in R1, as illustrated
in Fig. 4.2 (b) for a positive value of a < 1. Note that, for 0 < a < 1, there is a compression in the size of the
ROC of X(s) by a factor of a, as depicted in Fig. 4.2 (b), while for a > 1, the ROC is expanded by a factor of a.
Also, equation (9) implies that if a is negative, the ROC undergoes a reversal plus a scaling. In particular, as
1 s
depicted in Fig. 4.2 (c), the ROC of X for 0 > a > –1 involves a reversal about the j -axis, together with
a a
a change in the size of the ROC by a factor of a . Thus, time reversal of x(t) results in a reversal of the ROC.
That is,
L
x( t ) X( s); with ROC = –R ...(10)

Im Im Im
s-plane s-plane s-plane

R
r2 r1 Re Re Re
ar2 ar1 ar1 ar2
(a) (b) (c)
Fig. 4.2 : Effect on the ROC of time scaling:
(a) ROC of X(s); (b) ROC of (1 / a ) X( s / a ) for 0 < a < 1; (c) ROC of (1 / a ) X( s / a ) 0 > a > –1
REMEMBER If x(t) has poles and zeros of it’s laplace transform at pi and zi then x(at) will have pole and zeros
of it’s laplace transform at api and azi .
Example 4.9
Find Laplace transform of

( a ) u (–2 t – 7)
( b ) e3t – 7 u (3 t – 9)
Solution 4.9
( a ) Since we know that,
1
u(t ) , 0
First shifting,
s
1 7s
u(t 7) e
s
Then scaling,
1 7s / 2 1 7s / 2
u(2t 7) e e
s s
2
Then inversion, 2
1 7s / 2
u( 2t 7) e
s
1
So we get expression of Laplace transform e7s / 2 .
s
Since, Laplace transform has pole at origin and signal is left sided so ROC is < 0.
( b ) To find Laplace transform of e3t – 7 u(3t – 9)
1
u(t )
s
1
et u(t )
s 1
1
et 9
u(t 9) e 9s
s 1
1 1 1
e3t 9
u(3t 9) e 3s
e 3s
3 s 1 s 3
3

1
e2 e3t 9
u(3t 9) e2 e 3s
s 3
1
e3t 7
u(3t 9) e2 3s
s 3
Since signal is right sided and Laplace transform has a pole at 3 so ROC is > 3.
Study Note
Equation (9) show that expansion in time domain that is if a 1 then Laplace domain contract and vice-versa. So
scaling in time domain lead to scaling in Laplace domain and vice-versa.
4.3.5 Differentiation in Time Domain

If x(t) X(s) with ROC = R
dx(t )
then, sX( s) with ROC containing R ...(11)
dt
Proof : If x(t) has it’s Laplace transform X(s), that is
st
X (s ) = x(t ) e dt
d
and let Y(s) is Laplace transform of x(t ) that is
dt
d st
Y(s) = x(t ) e dt
dt
st st
= x(t ) e ( s) x(t ) e dt
st
= 0 s x(t ) e dt
= sX ( s )
Similarly differentiation property can be extended to yield,
d n x(t )
sn X ( s ) ...(12)
dt n
Consequently, dx(t)/dt is the inverse Laplace transform of sX(s). The ROC of sX(s) includes the ROC of
X(s) and may be larger if X(s) has a first-order pole at s = 0 that is canceled to the multiplication by s. For
example, if x(t) = u(t), then X(s) = 1/s, with an ROC that is Re{s} > 0. The derivative of x(t) is an impulse with
an associated Laplace transform that is unity and an ROC that is the entire s-plane.

Example 4.10
Using the bilateral time-shift and differentiation properties find the Laplace transform of
d2 3(t 2)
x(t) = (e u (t 2))
dt 2
Solution 4.10
We know that,
3t 1
e u(t ) , 3
s 3
Using the time-shifting property, we obtain
3( t 2) 1
e u(t 2) e 2s , 3
s 3
Now, using differentiation in the time-domain property, we obtain
d2 3( t 2) s2 2s
e u(t 2) e , 3
dt 2 s 3
4.3.6 Differentiation in the s-Domain
If x(t ) X( s) with ROC = R

dX( s)
then, t x(t ) with ROC = R ...(13)
ds
st
X (s ) = x(t ) e dt
Differentiating both sides w.r.t. s, we get
dX( s) st
= [ t x(t )] e dt
ds
dX( s)
= L[–t x(t)]
ds
dX( s)
Therefore, t x(t )
ds
Similarly, differentiation in the s-domain property can be extended to yield,
d n X( s)
t n x(t ) ( 1)n ...(14)
dsn
Example 4.11
Determine the L aplace transform ROC for following signals:

( a ) t e – at u ( t )
( b ) t sin( 0 t ) u ( t )
( c ) t cos( 0 t ) u ( t )

Solution 4.11
at Laplace transform 1
( a ) We know that, e u(t )
s a
Now if we apply differentiation property, we get
at Laplace transform d 1 1
te u (t ) 1
ds s a (s a )2
Since, Laplace transform has pole at –a and signal is right sided signal so ROC is > – a.
Laplace Transform 0
( b ) Since, sin( 0t ) u (t ) , 0
s2 2
0
To find Laplace transform of t sin( 0t) u(t), we apply differentiation in s-domain property
d 0 2 0s
t sin( 0t ) u(t ) 1
ds s2 2
0 s 2 2
0
Since, poles of Laplace transform are at ±j 0, and signal is right sided so ROC is > 0.
Laplace s
( c ) Since, cos( 0t ) u(t ) Transform 2 2
, 0
s 0
To find Laplace transform of t cos( 0t) u(t), we apply differentiation in s-domain property.
d s s2 2
0
t cos( 0t ) u (t ) 1
ds s 2 2
0 (s 2 2
0)
Since, Laplace transform has poles at ±j 0 and signal is right sided so ROC is > 0.
Study Note
We can find Laplace transform of sin( 0t) u(t) and cos( 0t) u(t) but we cannot find Laplace transform of cos( 0t)
and sin( 0t), because they exist for – < t < and when we multiply signal with e– t we cannot find any value of
for which x(t) e– t is absolutely summable. Thus there is no valid ROC for these signals.
For example:
sin( 0t) = sin( 0t) [u(t) + u(–t)]
= sin( 0t) u(t) + sin( 0t) u(–t)
Laplace 0
Now, sin( 0t) u(t ) , 0
Transform
s2 2
0
2
Laplace 0
and sin( 0 t ) u( t ) , 0
Transform
s2 2
0
Since, ROC = ( > 0) ( < 0) =

so Laplace transform of sin( 0t) do not exist.
4.3.7 Convolution Property

If x 1( t ) X1(s) with ROC = R1
and x 2(t) X2(s) with ROC = R2
x1(t) x2 ( t ) X2(s) X2(s) with ROC containing R1 R2 ...(15)

The Laplace transform maps the convolution of two signals into the product of their Laplace transforms.
The ROC of X1(s) X2(s) includes the intersection of the ROCs of X1(s) and X2(s) and may be larger if pole-zero
cancellation occurs in the product.
Proof : The Laplace transform of x1(t) x2(t) is given by
st
L[x1(t) x2(t)] = [x1( t ) x2 (t )] e dt
st
= x1( ) x2 (t )d e dt
Interchanging the order of integration and noting that x1( ) does not depend on t, we obtain
st
L[x1(t) x2(t)] = x 1( ) x 2 (t )e dt d
Using the time-shifting property the bracketed term is X2(s) e–st. Substituting this into the above equation
yields,
s
L[x1(t) x2(t)] = x1( ) ( X2 ( s) e )d
= X2 ( s) s
x1( ) e d
L[x1(t) x2(t)] = X2(s) X1(s) = X1(s) X2(s)

Therefore, x1(t) x2(t ) X1(s) X2(s)
The convolution property states that convolution in the time domain corresponds to multiplication in
the s-domain.
Example 4.12
Consider a signal y ( t ) which is related to two signals x 1 ( t ) and x 2 ( t ) by,

y ( t ) = x 1 ( t – 2) x 2 (– t + 3)
where, x 1 ( t ) = e –2 t u ( t ) and x 2 ( t ) = e –3 t u ( t )
Use properties of the Laplace transform to determine the Laplace transform Y ( s ) of y ( t ).
Solution 4.12
2t 1
We know that, x 1( t ) = e u(t ) , 2
s 2
2( t 2) e 2s
x1(t – 2) = e u(t 2) , 2
s 2
Similarly, we obtain
3t 1
x 2( t ) = e u (t ) , 3
s 3
Applying time shifting property we get
3(t 3) e 3s
x2(t + 3) = e u (t 3) , 3
s 3

Now, using the time-reversal property, we obtain
3( t 3) e 3s
x2(–t + 3) = e u( t 3) , 3
s 3
Now, using the convolution property, we obtain
L[y(t)] = L[x1(t – 2)] L[x2(–t + 3)]
e 2s e 3s
Y(s) = , 2 3
s 2 s 3
5s
e
Y(s) = , 2 3
( s 2) (3 s)
REMEMBER Suppose in any question we have to find convolution of two signal x1(t) and x2(t). Then first of
all find X1(s) and X2(s) and inverse Laplace transform of X1(s) X2(s). This will be much easier
than what we have learnt in Chapter-2.
In equation (15) we saw that ROC of Laplace transform of convolution of two signals will be R1 R2 but
this can become in valid in some cases where pole-zero cancellation take place.
For example:
( s 5)
1. X 1( s ) = , 6 7
( s 6) ( s 7)
( s 6)
and X2( s ) = , 8 5
( s 5) ( s 8)
Here we can see that R1 R2 = then also we can find x1(t) x2(t) and its Laplace transform will
exist because here pole-zero cancellation take place.
Laplace transform 1
x 1(t ) x 2 (t ) , 8 7
(s 7) (s 8)
s 5
2. X1( s ) = , 6 7
( s 6) ( s 7)
1
and X 2( s ) = , 8 5
( s 5) ( s 8)
Laplace transform 1
So, x1( t ) x2 (t ) , 6 7
( s 6) ( s 7) ( s 8)
1
3. X1( s ) = , 6 7
( s 6) ( s 7)
1
and X2( s ) = , 8 5
( s 5) ( s 8)
Here no pole-zero cancellation, so Laplace transform of x1(t) x2(t) will not exist.

4.3.8 Integration in Time-Domain

If x(t ) X(s) with ROC = R
t
1
then, x( ) d X( s) with ROC containing R { > 0} ...(16)
s
Proof: If a signal x(t) is convolved with a unit step function u(t), we get
x(t) u(t) = x( ) u(t )d
1, t
Since, u(t – ) =
0, t
t
we have, x(t) u(t) = x( ) d
The convolution of a signal with a unit step function is the same as the cumulative integral of the signal.
Now we can prove the integration property of the Laplace transform,
t
x( ) d = x(t) u(t)
t
L x( ) d = L[x(t) u(t)]
Using the convolution property, we obtain

t
L x( ) d 1
= X( s ) U( s) X( s)
s
t
1
Therefore, x( ) d X( s)
s
4.3.9 Conjugation Property

If x(t ) X (s )
then, x (t) X (s ) ...(17)
Proof : The Laplace transform of x (t) is
st s t
L[x (t)] = x (t ) e dt x (t )e dt [ X (s )] X (s )
Therefore, if x(t) is real, that is, x(t) = x*(t), then

X(s ) = X ( s ) ...(18)
X (s) = X(s ) ...(19)
Thus, when signal is real then X(s) = [X(s )] . That is if X(s) has a pole or zero at a then it must have
pole or zero at a . That is
Thus in rational Laplace transform of real signal the coefficients of s will be real. If pole or zero in
Laplace transform of real signal is complex then there will be a pole or zero at it’s conjugate location also that
is complex pole or zero always occur in conjugate pairs, it pole or zero is real then they can sit alone without any
pair.

4.3.10 Reversal in Time

If, x(t ) X(s), ROC is R
then, x (– t ) X(–s), ROC is –R
We have already proved this property. If signal is even then,
x ( t ) = x (– t )
X(s ) = X(–s)
Thus, X(s) will have only even power of s. The poles and zeros will be at mirror image with respect to
imaginary axis. That is if pole or zero is at a then it will be at –a also.
If signal is odd then, x ( t ) = –x (– t )
X(s) = –X(–s)
Thus X(s) will have only odd power of s. For odd signals also if pole or zero is at a then it will be
at –a also.
st
REMEMBER Since Laplace transform of signal x(t) is X(s) = x (t )e dt , and for odd signals we know that
x (t )dt 0 thus we can say that X(0) =0. That is Laplace transform of odd signal always
have zero at s = 0.
Property Signal Transform ROC
x (t) X(s) R
x 1(t) X1(s) R1
x 2(t) X2(s) R2
Linearity a 1x 1(t) + a 2x 2(t) a 1X1(s) + a 2X2(s) R R1 R2

– s0 t
Time shifting x (t – t0) e X(s) R =R
Shifting in s es0t x (t) X(s – s0) R = R + Re(s0)
1 s
Time scaling x (at) X( ) R = aR
a a
Time reversal x (–t) X(–s) R = –R

dx(t ) sX(s)
Differentiation in t R R
dt
dX( s)
Differentiation in s –tx (t) R =R
ds
t 1
Integration x( ) d X( s) R R1 {Re(s) > 0}
s
Convolution x 1(t) x 2(t) X1(s) X2(s) R R1 R2
4.4 Inverse Laplace Transform

Inverse Laplace transform means to find out signal x(t) from it’s Laplace transform X(s). Symbolically
it is represented by,
x ( t ) = L –1{X(s)} ...(20)

Method-1: Inverse Formulae

This is the procedure that can be used for all type of Laplace transform, this include evaluation of a line
integral in complex s-plane; that is
C j
1
x(t ) = X( s) est ds ...(21)
2 j C j
In the above integral the integration is performed for any value of C within ROC.
Note: Generally this method is not used.
2. Partial Fraction Expansion

If X(s) is a rational function, that is, of the form
N( s)
X (s ) = ...(22)
D( s)
For partial fraction the X(s) should be a proper function that is degree of numerator < degree of
denominator. If in any question this condition is not satisfied then divide numerator with denominator and
convert X(s) into
R( s)
X(s) = Q( s) ...(23)
D( s)
R( s)
Now solve using partial fraction.
D( s)
Case-1: When denominator has only single order roots or X(s) has single order poles.
N (s ) (s Z 1 ) (s Z 2 ) .... (s Zm)
So, X (s ) = K ...(24)
D (s ) (s P1 ) (s P2 ) .... (s Pn )
In this case when all poles are of single order,
C1 C2 Cn
X (s ) = .... ...(25)
(s P1 ) (s P2 ) (s Pn )
Here, Ck = X (s ) (s Pk ) s Pk
...(26)
Each pole in X(s) can give a right sided or left sided signal
Left sided Pk t
–Ck e u(–t), < Re{Pk }
Signal
Ck
s – Pk
Right sided Pk t
Ck e u(t), > Re{Pk}
Signal
If in a question ROC of X(s) is known then

• If ROC is in right side of pole then pole correspond to right sided signal.
• If ROC is in left side of pole then pole correspond to left sided signal.
• If in a question ROC of X(s) is not given then we have to find all possible x(t) that can correspond
to X(s) as Laplace transform.

REMEMBER If ROC of X(s) is not given and we have to find number of time domain signals that will correspond
to X(s), the method to find the number of time domain signals the procedure is as follows.
First of all mark all poles of X(s) on s-plane and draw lines parallel to imaginary axis and find
out into how many regions the s-plane is divided. Each region will correspond to a different x(t).
So number of time domain signal is equal to number of regions in which s-plane is divided.
Example 4.13
( s 10) ( s 15)
If, X( s ) 2
then find number of time domain signal that correspond
(s 16) ( s2 5s 6) ( s2 7)
to X ( s ) as Laplace transform.
Solution 4.13
The poles of X(s) are at 4 j, 7 j , 3, 2 let us mark all of these on s-plane.
Im
4j
7j
Re
–3 –2
7j
–4j
R1 R2 R3 R4
So, 4 regions are created in the plane > 0, < 3, 3 < < 2, 2 < < 0. So, 4 time domain signals
can be created.
Example 4.14
2s 4
If, X( s ) , 3 1 then find the time domain signal corresponding to it.
s2 4s 3
Solution 4.14
2s 4 C1 C2
Using partial fraction, X (s ) = 2
s 4s 3 ( s 3) ( s 1)
C 1 = ( s 3) X( s) s 3
1
C 2 = ( s 1) X( s) s 1
1
1 1
So, X(s ) =
( s 3) ( s 1)

If we plot the ROC and mark poles on s-plane, we get

Im
Re
–3 –1
Since ROC is left of –1 so pole at –1 correspond to left sided signal and ROC is right of –3 so pole at –3
correspond to right sided signal.
1 1
Since, X (s ) =
( s 3) ( s 1)
x(t ) = e–3t u(t) – e–t u(–t)
We can see that ROC is a strip and the corresponding time domain signal is both sided.
Example 4.15
Find inverse Laplace transform of

s2 6s 7
(a) X(s) = 2 , 1
s 3s 2
s3 2 s2 6
(b) X(s) = , 0
s2 3s
Solution 4.15
( a ) Since X(s) is not proper function so first of all we divide numerator by denominator.
3s 5
X (s ) = 1
( s 1) ( s 2)
3s 5
Applying partial fraction as ,
( s 1) ( s 2)
3s 5 C1 C2
X 1( s ) =
( s 1) ( s 2) s 1 s 2
C 1 = X1( s) ( s 1) s 1 2
and C 2 = X1( s) ( s 2) s 2 1
2 1
X1( s ) =
s 1 s 2
2 1
X (s ) = 1
( s 1) ( s 2)

Now plotting ROC and marking poles of X(s) on s-plane we get

Im
Re
–2 –1
Since ROC is right of both poles, so both correspond to right sided signal.
2 1
Since, X (s ) = 1
s 1 s 2
x(t ) = (t) + 2e–t u(t) + e–2t u(t)
( b ) Here also X(s) is not proper, so
s3 2s2 6 3s 6
X (s ) = 2
s 1
s 3s s( s 3)
3s 6
Let, X 1( s ) = ,
s( s 3)
Now applying partial fraction at X1(s) we get
C1 C2
X 1( s ) =
s s 3
C 1 = X1( s) s s 0
2
and C 2 = X1( s) ( s 3) s 3
1
2 1
so, X 1( s ) =
s s 3
2 1
X (s ) = s 1
s s 3
Since ROC [ > 0] is right to both poles [0, –3].
Both pole given right sided signals.
2 1
Since, X(s ) = s 1
s s 3
We know that laplace transform of (t) is 1 thus laplace transform of (t) will be s (using differentiation
in time domain property)
So, x ( t ) = (t) – (t) + 2u(t) + e–3t u(t)

Example 4.16
(3 s 7)
Find time domain signal corresponding to X ( s ) = 2
.
(s 5s 6)
Solution 4.16
Since ROC is not given so first to all we apply partial fraction on X(s),
3s 7 C1 C2
X (s ) =
( s 3) ( s 2) s 3 s 2
C1 = ( s 3) X( s) s 3
2
and C 2 = ( s 2) X( s) s 2
1
1 2
So, X (s ) =
s 2 s 3
There are two poles –2 and –3, there are three regions in which s-plane is divided.
Im
R1 R2 R3
Re
–3 –2
• So signal corresponding to ROC = R1 i.e. < –3. Since ROC is left of both poles so both poles
correspond to left sided signal.
So, x(t ) = –e–2t u(–t) –2 e–3t u(–t)
• Now, when ROC = R2 i.e. –3 < < –2 then pole –2 given left sided signal and pole –3 given right sided
signal.
So, x(t ) = e–2t u(–t) + 2 e–3t u(t)
• Now, when ROC = R3 i.e. > –2 then both pole given right sided signal.
So, x(t ) = e–2t u(t) + 2 e–3t u(t)
In previous case we analysed the Laplace transform where poles were of single order
Case-2 : Laplace Transform with non single order poles.

Consider a case where D(s) or denominator of X(s) has roots which are not of single order. Let us
assume one root of D(s) has order r, that is
N (s ) (s Z 1 ) (s Z 2 )....( s Z m )
X(s)= K
D (s ) (s P1 ) (s P2 )....(s Pi )r ....(s Pn )
The partial fraction for all poles will be same as in case-1 except for pole Pi,
C1 C2 1 2 r Cn
X(s) = .... .... r
....
(s P1 ) (s P2 ) (s Pi ) (s Pi )2
(s Pi ) s Pn

Here, Ck = (s Pk ) X (s ) s Pk
...(27)
1 dk
and r–k = [(s Pi )r X (s )] ...(28)
k ! ds k s Pi
It depend on ROC of X(s) that pole will correspond to right side signal or left side signal.
REMEMBER While finding inverse Laplace transform in case-2, this formulae can be helpful.
When pole correspond to right sided signal, when > Real {Pi}
A For right A t m 1 Pi t
e u (t )
(s Pi )m Sided signal ( m 1)!
When pole correspond to left sided signal, when < Real {Pi]
A For ledt t m 1 Pi t
A e u( t )
(s Pi )m Sided signal ( m 1)!
Example 4.17
s2 2s 5
Find the inverse Laplace transform of X( s ) , 3.
( s 3) ( s 5)2
Solution 4.17
We see that, X(s) has one simple pole at s = –3 and one multiple pole at s = –5 with multiplicity 2. Then by
we have,
c1 1 2
X (s ) =
s 3 s 5 (s 5)2
we have, c 1 = ( s 3) X( s) s 3
s2 2s 5
= 2
( s 5)2 s 3
2
2 = ( s 5) X( s)
s 5
s2 2s 5
= 10
s 3 s 5
d
1 = [( s 5)2 X( s)]
ds s 5
d s2 2 s 5 s2 6s 1
= 1
ds s 3 s 5
( s 3)2 s 5
2 1 10
Hence, X (s ) =
s 3 s 5 ( s 5)2
The ROC of X(s) is Re(s) > –3. Thus, x(t) is a right-sided signal and we obtain
x ( t ) = 2e–3t u(t) – e–5t u(t) – 10t e–5t u(t)
= [2e–3t – e–5t – 10t e–5] u(t)

Example 4.18
Find the inverse Laplace transform of

2s 4s
2 2s e 4e
X(s) = , 1
s2 4s 3
Solution 4.18
We see that, X(s) is a sum
X(s ) = X1(s) + X2(s) e–2s + X3(s) e–4s
2
where, X 1( s ) = 2
s 4s 3
2s
X 2( s ) = 2
s 4s 3
4
X 3( s ) =
s24s 3
If x1(t) X1(s) x2(t) X2(s) x3(t) X3(s)
then by the linearity property and the time-shifting property we obtain,
x(t ) = x1(t) + x2(t – 2) + x3(t – 4)
Next, using partial fraction expansions and from table, we obtain
1 1 t 3t
X 1( s ) = x 1(t ) (e e ) u (t )
s 1 s 3
1 3 t 3t
X 2( s ) = x 2 (t ) ( e 3e ) u (t )
s 1 s 3
2 2 t 3t
X 3( s ) = x3 (t ) 2( e
) u(t ) e
s 1 s 3
Thus, x(t ) = (e–t – e–3t) u(t) + [–e–(t – 2) – 3 e–3(t – 2)] u(t – 2) +
2[e–(t – 4) – e–3(t – 4)] u(t – 4)
4.5 Analysis and of LTI System Using Laplace Transform

Suppose we have an LTI system with impulse response h(t) and input x(t), the output will be
y(t) = x(t) h(t).
Output = Input Impulse response ...(29)
If we take Laplace transform of above equation we get
Y(s) = X(s) H(s) ...(30)
h(t)
x (t) y (t) = x (t) h(t)
X(s) Y(s) = X(s) H(s)

H(s)
Fig. 4.3 : Impulse response and System function

In equation (30) H(s) is Laplace transform of h(t), here

Y( s) Laplace transform of output
H (s ) = ...(31)
X( s) Laplace transform of input
Study Note
• H(s) is commonly referred as the system function or transfer function.
• For s = j , H(s) is frequency response of LTI system.
• Equation (31) is valid only when initial condition are zero.
The Transfer Function and Differential Equation System Description:

The transfer function may be related directly to the differential-equation description of an LTI system by
using the bilateral Laplace transform. The relationship between the input and output of an Nth order LTI system
is described by the differential equation,
N
dk y(t ) M
dk x(t )
ak = bk
k 0 dtk k 0 dt k
Taking the Laplace transform of the above equation, we obtain
N N
ak sk Y( s) = bk sk X( s)
k 0 k 0
M
bk sk
Y( s) k 0
H( s) = N
X( s)
ak sk
k 0
H(s) is a ratio of polynomials in s and is thus termed a rational transfer function.
4.5.1 Step Response and Impulse Response

The h(t) is impulse response of system, that is output when (t) is input to the system.
Laplace transform
h(t ) H( s)
The step response s(t) is output when input is u(t), so
s( t ) = u(t) h(t)
H( s)
S(s ) = ...(32)
s
or, H(s) = s(S(s)) ...(33)
4.5.2 Causality of LTI System

For a causal LTI system the impulse response h(t) = 0 for t < 0, thus the signal should be right sided.
In case of right sided signal which is of infinite duration, then ROC is the region in the s-plane to the right of
right most pole.
If H(s) is rational, then we can determine whether the system is causal simply by checking to see if its
ROC is a right-half plane, that is for a system with a rational system function, causality of the system is equivalent
to the ROC being the right-half plane to the right of the right most pole.

In an exactly analogous manner, we can deal with the concept of anticausality. For an anticausal LTI
system, the impulse response h(t) = 0 for t > 0 and is thus left-sided. We know that if a signal is left-sided and
of infinite duration, then the ROC is the region in the s-plane to the left of the left most pole. Consequently, the
ROC associated with the system function for an anticausal system is a left-half plane. However, the converse of
this statement is not necessarily true.
Study Note
• In the above explanation we were talking about rational H(s), but if H(s) is not rational that is for example
e5s/(s + 5), > –5 is irrational function and has right sided ROC but it will be non-causal as here h(t) will
be e–5(t + 5) u(t + 5) that is non zero for t < 0.
• In case of rational H(s) the system will be causal if ROC is > Real part of right most pole.
4.5.3 Stability of LTI System

An LTI system is stable if the impulse response h(t) is absolutely integrable, that is
h(t ) dt < ...(34)
Since, H(s) is transfer function of the system, that is Laplace transform of h(t). So ROC of H(s) is
defined as values of where h(t) e– t is absolutely integrable. That is
t
h(t ) e dt < ...(35)
If we compare above two equations then we get that if signal h(t) is absolutely integrable then ROC of
it’s laplace transform will contain the j -axis. Now if H(s) is given and we know it’s ROC then to find that h(t)
is absolutely integrable the ROC of H(s) must contain = 0 or j -axis or imaginary axis. Thus to have stable LTI
system the ROC of transfer function must contain j -axis.
REMEMBER Suppose in any question for example it is given that x(t) e–4t is absolutely integrable. If X(s) is
Laplace transform of x(t) then it’s ROC is values of where x(t) e– t is absolutely integrable. So
if we compare the given information with definition of ROC we can say that = 4 is present in
ROC.
4.5.4 Stability of a Causal LTI System

If system is causal as well as stable and H(s) is a rational function then it must satisfy two conditions
together:
(i ) ROC should be > Real part of right most pole.
( ii ) ROC should certain j -axis.
If we want both conditions then it means that right most pole of H(s) lie left of j -axis. That is all poles
of H(s) should lie in left of j -axis.
Study Note
• System with rational H(s) will be causal and stable if and only if all poles lie in left of j -axis.
• Causal and stable system will have ROC, > Real part of right most pole.

Example 4.19
The unit step response of an LTI system is s ( t ) = 2 e –t u ( t ). Determine its system function
and the impulse response.
Solution 4.19
Given that the unit step response,
s (t ) = 2 e–t u(t)
L[s(t)] = L[2 e–t u(t)]
2
S (s ) =
s 1
The relationship between L[h(t)] = H(s) and L[s(t)] = S(s) is given by
2s 2s 2 2 2( s 1) 2
h(s) = s S(s) =
s 1 s 1 s 1 s 1
2
H (s ) = 2
s 1
The inverse transform of this equation yields,
h(t) = 2 (t) – 2 e–t u(t)
Example 4.20
For the following check whether corresponding system is causal and stable.
1
( a ) H1( s ) 2
, 3
s s 6
1
( b ) H 2 (s ) , 2
s2 s 6
1
( c ) H3 ( s) 2
, 2 3
s s 6
Solution 4.20
( a ) Since transfer function is rational and ROC is > Real part of right most pole. So it is causal but since
ROC do not contain j -axis, so unstable.
H1(s) Causal, unstable system
( b ) In this system ROC is < –2, ROC is < Real part of left most pole. So system is anticausal and
unstable as ROC do not contain j -axis.
( c ) In this system, ROC is –2 < < 3, since ROC is a strip in s-plane thus time domain signal will be both
sided signal this it is non-causal, ROC contain j -axis so it is stable.
Example 4.21
For the following system function, check whether the corresponding LTI system is causal
and stable,
1
H (s ) 2
, 2
s 5s 6

Solution 4.21
Since H(s) has all poles in left side of j and > –2 so it is stable and causal.
1 1 1
H (s ) = 2
s 5s 6 s 2 s 3
Since both poles are in left of ROC so both give right sided signal.
h(t) = e–2t u(t) – e–3t u(t)
Example 4.22
4s2 15s 8
Find the inverse Laplace transform of H( s) , assuming that (a) h ( t ) is causal
( s 2)2 ( s 1)
and (b) the Fourier transform of h ( t ) exists, i.e., h ( t ) is absolutely integrable.

Solution 4.22
4s2 15s 8
Given that, H (s ) =
( s 2)2 ( s 1)
Using partial fraction expansion, we obtain
1 2 3
H (s ) = 2
s 2 ( s 2) s 1
The system has a double pole at –2 and a single pole at 1.
( a ) For this system to be causal, the ROC lies to the right of the right most pole, i.e. > 1. Since the ROC,
> 1, is to the right of the right most pole, all the poles correspond to causal (right-sided) signals.
Therefore,
2t 1
e u(t )
s 2
2t 1
2t e u(t )
( s 2)2
3
3 et u(t )
s 1
and hence we obtain, h(t) = e–2t u(t) + 2 t e–2t u(t) + 3 et u(t)
( b ) For h(t) to be Fourier transformable, i.e., for h(t) to be absolutely summable (stable system), the ROC
must include the j -axis. To include the j -axis, its ROC is in the region –2 < < 1. The pole of the
first and second term is at –2. The ROC lies left of pole s =1 thus it correspond to left sided signal and
ROC lie right of the pole s= 2 thus this pole give right sided signals. Therefore,
2t 1
e u(t )
s 2
2t 1
2t e u(t )
( s 2)2
3
3 et u( t )
s 1
and hence we obtain, h( t ) = e–2t u(t) + 2 t e–2t u(t) –3 et u(–t)

Example 4.23
Consider a LTI system for which input is x ( t ) and output is y ( t ) are related by differential
equation,
d2 y ( t ) dy ( t )
2y (t ) x( t )
dt 2 dt
Determine h ( t ) for each of the following case:
1 . The system is stable.
2 . The system is causal.
3 . The system is neither stable nor causal.
Solution 4.23
Taking the Laplace transform of differential equation we get,
s2 Y(s) – sY(s) – 2Y(s) = X(s )
Y( s) 1 1/ 3 1/ 3
H( s) =
X( s) s2 s 2 ( s 2) ( s 1)
So, the system has two poles, –1 and 2.
( a ) For stable system, ROC must contain j -axis, so ROC will be –1 < < 2. So pole –1 will give right
sided signal and 2 will give left sided signal.
1 2t 1 t
h( t ) =
e u( t ) e u(t )
3 3
( b ) For causal system, ROC must be right of right most pole i.e. > 2, so both poles give right sided signal,
1 2t 1 t
e u(t )
h( t ) = e u(t )
3 3
( c ) For neither stable or causal, ROC must not be right of right most pole and must not certain j -axis, so
ROC is < –1. So both poles will give left sided signal. So
1 2t 1 t
h( t ) = e u( t ) e u( t )
3 3
Example 4.24
Find the response y ( t ) of a non-causal system with the transfer function,

1
H(s) = , 1
s 1
to the input x ( t ) = e –2 t u ( t ).
Solution 4.24
Given that, x ( t ) = e–2t u(t)
1
L[x(t)] = X( s) , 2
s 2
1
we knot that, Y(s) = X( s) H( s)
( s 1) ( s 2)
The ROC of Y(s) is in the region –2 < < 1. Using partial fraction expansion, we obtain
1/3 1/3
Y(s) = , 2 1
s 1 s 2

Taking the inverse Laplace transform, we obtain

1 t 2t
y(t) = [e u( t ) e u(t )]
3
4.5.5 System Interconnection
Cascade Connection
For two LTI systems [with h1(t) and h2(t), respectively] in cascade [Fig. 4.4(a)], the overall impulse
response h(t) is given by
h(t) = h1(t) h2(t) ...(36)
x(t) h1(t) h2(t) y(t) x(t) h(t) y(t)
h(t) = h1(t)* h2(t)....

(a)
X(s) H1(s) H2(s) Y(s) X(s) H(s) Y(s)
H(s) = H1(s) H2(s)....

(b)
Fig. 4.4 : Two systems in cascade (a) Time-domain representation (b) s-domain representation
Thus, the corresponding system functions are related by the product,
H(s) = H1(s) H2(s) ...(37)
This relationship is illustrated in Fig. 4.4 (b).
Parallel Connection
The impulse response of a parallel combination of two LTI systems [Fig. 4.5(a)] is given by,
h(t) = h1(t) + h2(t)
Thus, H(s) = H1(s) + H2(s) ...(38)
This relationship is illustrated in Fig. 4.5 (b).
h1(t)
x(t) y(t) x(t) h(t) y(t)
h(t) = h1(t) + h2(t)

h2(t)
(a)
H1(s)
X(s) Y(s) X(s) H(s) Y(s)
H(s) = H1(s) + H2(s)

H2(s)
(b)
Fig. 4.5 : Two systems in parallel

(a) Time-domain representation (b) s-domain representation

Feedback Connection
When the output is feedback to the input, as shown in Fig. 4.6, the overall transfer function
H(s) = Y(s)/X(s) can be computed as follows.
+ E(s)
X(s) H1(s)
–
H1( s)
= X(s) H( s) Y(s)
1 H1( s) H2 (s)
H2(s)
Fig. 4.6 : Elementary connection of block diagram
The input to the adder are X(s) and –H2(s) Y(s). Therefore, E(s), the output of the adder, is
E(s) = X(s) – H2(s) Y(s)
and the output is given by,
Y(s) = H1(s) E(s) = H1(s) [X(s) – H2(s) Y(s)]
Y(s) = H1(s) X(s) – H1(s) H2(s) Y(s)
Y(s) [1 + H1(s) H2(s)] = H1(s) X(s)
Y( s) H1( s)
H( s) =
X( s) 1 H1( s) H2 ( s)
Therefore, the feedback loop can be replaced by a single block with the transfer function shown above.
The transfer function H1(s) is the forward path transfer function and H2(s) is the feedback path transfer function.
4.5.6 Response of Input (eCt ) from LTI Continuous Time System

When input is of format eCt where C is any constant value, we can see that Laplace transform of eCt do
not exist. [eCt = eCt u(t) + eCt u(–t) and then ROC will be for this case]. So we cannot use Y(s) = H(s) X(s).
LTI
eCt System
h(t ) Laplace
H( s) Output
Transform
y ( t ) = h(t) eCt ...(39)
st
and H(s ) = h(t ) e dt ...(40)
Now, if we look at equation (39), we get
y(t) = h( ) eC( t )
d
= eCt h( ) eC d
Ct
y ( t ) = e H (s ) ...(41)
s C

Study Note
So whenever input is eCt for – < t < then output is equal to
y ( t )= eCt. Value of Laplace transform at s = C
Using above result we get,
ej 0t
H(s) ej 0t H( s)
s j 0
e–j 0 H(s) e j 0t
H( s)
s j 0
When h(t) is real then H(s) is rational function and H (s ) s j 0

(H ( s ) s j 0
) , that is for real h(t) we
have H ( j o ) H( j o )* . This will be proved in Fourier transform chapter. When nothing is given about h(t)
then we can assume that h(t) is real.
Now, consider a case that LTI system with impulse response h(t) has input cos( 0t) [we cannot find
Laplace transform of cos( 0t)] then to find output,
ej 0t e j 0t
cos( 0t ) =
2
so, if H(s) is transfer function of the system then output of cos( 0t) will be
ej 0t H (s ) s j 0
e j 0t
H (s ) s j 0
cos( 0t )
2
ej 0t H( j 0) e j 0t
H( j 0)
cos( 0t)
2
Assuming that h(t) is real then
Let, H( j 0) = H(j o ) e j , where H(j 0)
H(–j 0) = H(j o )*
j
Thus, H(-j 0) = H(j o )e
ej 0t H( j 0) ej e j 0t
H( j 0) e j
Thus, output =
2
= H( j 0) cos( 0t )
Thus, cos( 0t) H(s) H(j

cos( 0t) 0)
or
H( j 0 ) cos( 0 t H( j 0 ))
Similarly, sin( 0t ) H(s) H(j

sin( 0) 0 t)
or
H( j 0 ) sin( 0 t H( j 0 ))

4.6 The Unilateral Laplace Transform

The unilateral (or one-sided) Laplace transform X1(s) of a signal x(t) is defined as
st
X 1( s ) = x (t ) e dt ...(42)
0
The lower limit of integration is chosen to be 0– (rather than 0 or 0+) to permit x(t) to include (t) or its
derivatives. Thus, we note immediately that the integration from 0– to 0+ is zero except when there is an
impulse function of its derivative at the origin. The unilateral Laplace transform ignores x(t) for t < 0. Since
x(t) in equation (42) is a right-sided signal, the ROC of X1(s), is always of the form > max, that is, a right half-
plane in the s-plane.
In case of unilateral Laplace transform it is not necessary to define ROC because it is understood that
signal will be right sided.
If we want to find unilateral Laplace transform of a signal x(t) then it simply means to find bilateral
Laplace transform of x(t) u(t).
For example:
If, x ( t ) = e–a(t + 1) u(t + 1) then,
it’s bilateral Laplace transform is,
1
X (s ) = es , > Real{–a}
(s a)
but to find unilateral Laplace transform we multiply x(t) with u(t) i.e. now,
xu ( t ) = e–a(t + 1) u(t + 1) u(t)
That is now, xu ( t ) = e–a(t + 1) u(t) = e–a e–at u(t)
Now, bilateral Laplace of xu(t) will be unilateral Laplace transform of x(t)
1
e ae at
u(t ) e a
s a
ea
Thus, unilateral Laplace transform is .
s a
Example 4.25
Find unilateral transform of

( a ) x ( t ) = e –2 t u ( t + 1)
( b ) x ( t ) = ( t + 1) + ( t ) + e –2( t + 3) u ( t + 1)
Solution 4.25
We can use equation (42) or find bilateral Laplace transform of x(t) u(t). That is for
(a) x(t ) = e– t u(t + 1)
t
Since, X (s ) = x(t ) e dt
0
1, t 1
= e 2t
u(t 1) e st
dt u(t 1)
0, t 1
0

2t st
= e e dt
0
1
= , 2
s 2
(b) x ( t ) = (t + 1) + (t) + e–(2t+ 3) u(t + 1)
st
0
(2 t 3) st
= [ (t 1) (t ) e u(t 1)] e dt
0
Here, (t + 1) is at t = –1, (t) is at t = 0, u(t + 1) starts from-1
(2t 3) st 1
X (s ) = ( (t ) e )e =1
s 2
0
s 3
= , 2
s 2
4.6.1 Properties of Unilateral Laplace Transform

Most of the properties of the unilateral Laplace transform are the same as for the bilateral transform.
The unilateral Laplace transform is useful for calculating the response of a causal system to a causal input when
the system is described by a linear constant-coefficient differential equation with nonzero initial conditions. The
basic properties of the unilateral Laplace transform that are useful in this application are the time-differential
and time-integration properties which are different from those of the bilateral transform. They are presented in
the following.
1. Linearity:
Unilateral Laplace transform
If x 1(t ) X 1(s )
and x 2 (t ) X 2 (s )
then, ax 1(t ) bx 2 (t ) aX 1(s )bX 2 (s ) ...(43)
2. Time Scaling
If x (t ) X (s )
Unilateral Laplace transform 1 s

then, x (at ) X a 0 ...(44)
a a
Note: This will be invalid if a is negative, because for negative value of a signal will become left sided.
3. Shifting in the s-domain

If x (t ) X (s )
then, e s0t x (t ) Unilateral Laplace transform

X (s s0 ) ...(45)

4. Conjugation
If x (t ) X (s )

then, x (t ) X (s ) ...(46)
5. Differential in s-domain
If x (t ) X (s )
Unilateral Laplace transform d X (s )

then, t x (t ) ...(47)
ds
6. Convolution
If x 1( t ) = x2(t) = 0, for all t < 0, and if
x 1(t ) X 1(s )
and x 2 (t ) X 2 (s )
then, x1(t ) x2(t ) X1(s ) X2(s ) ...(48)
It is important to note that the convolution property for unilateral transform applies only if the signals
x1(t) and x2(t) are both zero for t < 0. That is, while we have see that the bilateral Laplace transform of
x1(t) x2(t) always equals to the product of the bilateral transform of x1(t) and x2(t), the unilateral transform
of x1(t) x2(t) in general does not equal to the product of the unilateral transform if either x1(t) or x2(t) is
nonzero for t < 0.
7. Integration in the s-domain
If x(t) satisfies the condition of existence x(t ) e t

dt , and the limit of x(t)/t, as t approaches 0
0
from the right, exists, and

x (t ) X (s )
then,
x (t ) Unilateral Laplace transform
...(49)
X (s ) ds
t s
8. Time Shifting
If x (t ) X (s )
Unilateral Laplace transform st0

then, x (t t 0 ) X (s ) e ...(50)
Observe that x(t) starts at t = 0, and therefore, x(t – t0) starts at t = t0. So, this property may be
restated as follows.
If x (t ) u (t ) X (s )
Unilateral Laplace transform st 0
then, x (t t 0 ) u (t t 0 ) X (s ) e ...(51)
Note: This property is valid if signal is shifted right, that is t0 > 0.

9. Differentiation in Time-domain
If x (t ) X (s )
then,
d Unilateral Laplace transform
...(52)
x (t ) s X (s ) x (0 )
dt
Proof : From definition,
st
X (s ) = x(t ) e dt
0
d d
Let Y(s) is unilateral Laplace transform of x(t ), and y(t ) x(t ).
dt dt
d st
Y(s) = x(t ) e dt Applying integration by parts,
dt
0
st st
= x(t ) e ( s) x(t ) e dt
0
0
st
= [0 x(0 )] s x(t ) e dt
0
= s X(s) – x(0–)
Thus, d 2 x (t ) Unilateral Laplace transform

s (Y (s )) y (0 )
dt 2
2 d
= s X( s) s x(0 ) x(t )
dt t 0
General formulae will be
d n x (t ) n
s n X (s ) sn k
x (k 1)
(0 ) ...(53)
dt n k 1
10. Integration in the time domain

If x (t ) X (s )
0
t
x( )d
then, Unilateral Laplace transform X (s ) ...(54)
x( )d
S S
t
Unilateral Laplace transform X (s )
and x( )d ...(55)
0
s
Proof:
t
Let, y(t) = x( ) d
d
that is, x(t ) = y(t )
dt

Let X(s) and Y(s) are unilateral Laplace transform of x(t) and y(t). So using equation (52) we get
X(s ) = s Y(s) – y(0–)
X (s ) y (0 )
Y(s) =
s s
t
when, y(t) = x( ) d then,
0
y (0 – ) = x( ) d
t
x ( )d
x( )d
s s
t
and when, y(t) = x( ) d then,
0
y (0 – ) = 0
t
x( )d 0
s
0
11. Initial value theorem

The initial value theorem allows us to determine the initial value x(0+) of x(t) directly from X(s). The
theorem states that if x(t) = 0, for t < 0, and if it contains no impulses or higher order singularities at the
origin, then
x(0+ ) = Lim s X( s) ...(55)
s
The initial value theorem does not apply to rational functions X(s) in which the order of the numerator
polynomial is greater than or equal to that of the denominator polynomial.
Proof : We know that,
d Unilateral Laplace transform
x (t ) s X (s ) x (0 )
dt
d
That is, s X(s) – x(0–) = x(t ) e st dt
dt
0
Now we put Lim in above equation. As s then RHS will be zero because of e–st term.
s
So, Lim [s X( s) x(0 )] = 0

s
Lim s X( s) = x(0 – )
s
Remember: x(0–) = x(0+) because no impulses exist in x(t) because X(s) is proper function.

Example 4.26
Use the initial value theorem to find the initial value of the signal corresponding to the
L aplace transform,
s 1
X(s) =
s( s 2)
Solution 4.26
From the initial value theorem, we have
s 1 s 1
x(0 ) Lim s X( s) = Lim s Lim
s s s( s 2) s s 2
1 (1 s)
x(0+ ) = Lim 1
s 1 (2 / s)
12. Final value theorem

The final value theorem allows us to determine the final value x( ) of x(t) directly from X(s). The
theorem states that if x(t) = 0, for t < 0, and if it contains no impulses or higher order singularities at the origin,
then
Lim x(t ) = x( ) Lim s X( s)
t t
The final value theorem is applicable only if the poles of s X(s) are in the left half of the s-plane, with at
most a single pole at s = 0. If X(s) has a pole in the right half of the s-plane, x(t) contains an exponentially
growing term and x( ) does not exist. If there is a pole on the imaginary axis, then x(t) contains an oscillating
term and x( ) does not exist. However, if there is a pole at the origin, then x(t) contains a constant term, and
hence x( ) exists and is a constant.
Proof : Taking the unilateral Laplace transform of dx(t)/dt, we obtain
dx(t ) dx(t ) st
Lu = e dt
dt dt
0
Using differentiation in the time-domain property, we obtain
dx(t ) st
sX(s) – x(0–) = e dt
dt
0
Lim[sX( s) x(0 )] = Lim dx(t ) st

e dt x(t ) 0
s 0 s 0 dt
0
Lim[sX( s) x(0 )] = x( ) – x(0–)

s 0
Lim s X( s) = x ( )
s 0

Example 4.27
Determine the initial and final values of x ( t ) if its Laplace transform is given by:
10(2 s 3)
X(s) =
s( s2 2 s 5)
Solution 4.27
From the initial value theorem, we have
10(2s 3)
x(0+ ) = Lim s X( s) Lim s
s s s( s2 2s 5)
= Lim 10(2s 3) 6
s s2 2s 5
From the final value theorem, we have
10(2s 3) 10(2s 3)
Lim x (t ) x ( ) = Lim sX (s ) Lim s Lim 6
2 2
t s 0 s 0 s (s 2s 5) s 0 s 2s 5
Example 4.28
Find the value of output at steady state when input to system is x ( t ) and transfer function is
H(s),
s 3
H(s) =
s2 4s 5
When,
( a ) x(t) = u(t)
( b ) x(t) = (t)
Solution 4.28
Now, steady state value is y( ), thus
(a) Y(s) = H(s) X(s)
s 3 1 ( s 3)
= = 2
2
s 4s 5 s s( s 4s 5)
[Since all poles of sY(s) are in left of j -axis, so we can apply final value theorem]
3
Y( ) = Lim s Y( s)
s 0 5
(b) Y(s) = H(s) X(s)
s 3
= 2
1
s 4s 5
Here also poles of sY(s) lie in left of j -axis so we can apply final value theorem,
Y( ) = Lim s Y( s) 0
s 0
Remember: If in any question system transfer function H(s) is given but it’s ROC is not given then assume that
system is causal and assume ROC accordingly.

4.7 Solution of Differential and Integer Differential Equations

One of the most common applications of the Laplace transform is to solve linear constant coefficient
differential equations. Such equations are used to model continuous-time LTI systems. Solving of these equations
depends on the differentiation property of the Laplace transform. Procedure that should be followed to solve is
1. For a given set of initial conditions, take the Laplace transform of both sides of the differential
equation to obtain an algebraic equation in Y(s).
2. Solve the algebraic equation for Y(s).
3. Take the inverse Laplace transform to obtain y(t).
Example 4.29
Solve the second-order linear differential equations,

d2 y ( t ) dy ( t ) dx( t )
2
5 6y(t ) x( t )
dt dt dt
dy ( t )
for the initial conditions y (0 – ) = 2 and y (0 ) 1 and the input x ( t ) = e –4 t u ( t ).
dt t 0
Solution 4.29
If x(t) = e–4t u(t), then x(0–) = 0 and
1
X (s ) =
s 4
Now, consider the given differential equation,
d2 y ( t ) dy(t ) dx(t )
5 6y(t ) = x(t )
dt 2 dt dt
[s2 Y(s) – sy(0–) – y (0 ) ] + 5[sY(s) – y(0–)] + 6Y(s) = [sX(s) – x(0–)] + X(s)

[s2 Y(s) – 2s – 1] + 5[sY(s) – 2] + 6Y(s) = [sX(s) – 0] + X(s)
(s2 + 5s + 6) Y(s) – (2s + 11) = X(s) (s + 1)
s 1
(s2 + 5s + 6) Y(s) = (2s 11)
s 4
2s2 20s 45
Y(s) =
( s 2) ( s 3) ( s 4)
Expanding the RHS into partial fractions yields,
13 / 2 3 3/2
Y(s) =
s 2 s 3 s 4
Taking the inverse Laplace transform of the above equation, we obtain
13 2t 3t 3 4t
y(t) = e 3e e u(t )
2 2
Study Note
In the above example we can see that output Y(s) has no ROC but since system is causal and input is also causal so
output is always right sided.

4.7.1 Zero-Input Response and Zero-State Response

The Laplace transform method for solving differential equations offers a clear separation between the
zero-input response of the system to initial conditions and the zero-state response of the system associated with
the input. The zero-input response represents the system output when the input is zero, i.e., x(t) = 0, and thus
is the result of internal system conditions (such as energy storage, initial conditions) alone. In contrast, the
zero-state response represents the system response to the external input x(t) when the initial conditions are
zero. The total response is defined as,
Total response = Zero-input response + Zero-state response ...(58)
Natural response and Force response: In total response, the response due to the poles of the
system is called natural response of the system. The response due to the poles of the input is called the forced
response of the system.
Example 4.30
Use the unilateral Laplace transform to determine the output of a system represented by the
differential equation,
d2 y ( t ) dy ( t ) dx( t )
2
5 6 y (t ) 6 x( t )
dt dt dt
in response to the input x ( t ) = u ( t ). Assume that the initial conditions on the system are
.
y (0 – ) = 1 and y (0 ) = 2. Identify the zero-state response y zs ( t ), of the system, and the zero
input response y zi ( t ).
Solution 4.30
If x(t) = u(t), then x(0–) = 0 and X(s) = 1/s. Now, consider the given differential equation,
d2 y(t ) dy(t ) dx(t )
2
5 6 y(t ) 6 x(t )
dt dt dt
.
[s2 Y(s) – sy(0–) – y (0 ) ] + 5[sY(s) – y(0–)] + 6Y(s) = [sX(s) – x(0–)] + 6X(s)
[s2 Y(s) – s – 2] + 5[sY(s) – 1] + 6Y(s) = [sX(s) – 0] + 6X(s)
(s2 + 5s + 6) Y(s) – (s + 7) = X(s) (s + 6)
(s2 + 5s + 6) Y(s) = ( s 7) X( s) ( s 6)
Initial condition terms Input terms
s 7 s 6
Y(s) = 2 2
s 5s 6 s( s 5s 6)
Zero-input component Zero-state component
s 7 s 6
Y(s) =
( s 2) ( s 3) s( s 2) ( s 3)
5 4 1 2 1
Y(s) =
s 2 s 3 s s 2 s 3

The inverse transform of this equation yields,

y(t) = [5e–2t u(t) – 4e–3t u(t)] + [u(t) – 2e–2t u(t) + e–3t u(t)]
2t 3t 2t 3t
y(t) = [5 e 4e ] u(t ) [1 2 e e ] u(t )
Zero-input response or natual response Zero-state response or forced response
y(t) = yzi(t) + yzs(t)

where the zero-input response is,
yzi(t ) = [5e–2t – 4e–3t] u(t)
and the zero-state response is
yzs(t) = [1 – 2e–2t + e–3t] u(t)
4.8 System Realization

We now develope a systematic method for realization (or implementation) of an arbitrary Nth-order
transfer function. The most general transfer function with M = N is given by,
b0 sN b1sN 1 ... bN 1 s bN
H(s ) =
sN a1 sN 1 ... a N 1 s aN
Since realization is basically a synthesis problem, there is no unique way of realizing a system. A given
transfer function can be realized in many different ways. A transfer function H(s) can be conveniently represented
in block diagram from using the basic building blocks representing the integrators of differentiators, the multiplier,
the adder, and the pick-off node as shown in Fig. 4.7.
X( s) A
X(s) 1/s Y( s) X(s) Y(s) = AX(s)
s
(a) (b)
+ Y(s) = X1(s) + X2(s)

X1(s) X(s) X(s)
X(s)
X2(s)
(c) (d)
Fig. 4.7 : Basic building block representing

(a) Integrator, (b) Multiplier, (c) Adder and (d) Pick-off
4.8.1 Direct Form I Realization

An Nth order transfer function is characterized by 2N + 1 unique coefficients and, in general, requires
2N + 1 multipliers and 2N two input adders for implementation. Structures in which the multiplier coefficients
are precisely the coefficients of the transfer functions are called direct form structures.
b0 s3 b1s2 b2 s b3
H(s ) =
s3 a1s2 a2 s a3
b0 ( b1 / s) ( b2 / s2 ) ( b3 / s3 )
H(s ) =
1 ( a1 / s) ( a2 / s2 ) ( a3 / s3 )
b1 b2 b3 1
We can express H(s) as, H(s ) = b0
s s s 1 a1 / s ( a2 / s2 ) ( a3 / s3 )
H1( s ) H2 ( s )

We can realize H(s) as a cascade of transfer function H1(s) followed by H2(s), as depicted in Fig. 4.8 (a),
where
W( s) b1 b2 b3
H 1( s ) = b0 ...(59)
X( s) s s2 s3
b1 b2 b3
W(s) = b0 X( s) ...(60)
s s2 s3
Y( s ) 1
and H 2( s ) = ...(61)
W( s ) 1 ( a1 / s ) ( a 2 / s2 ) ( a3 / s3 )
a1 a2 a3
Y(s) = W( s) Y( s) ...(62)
s s2 s3
We shall first realize H1(s) given by equation (59). Equation (60) shows that the output W(s) can be
synthesized by adding the input b0 X(s) to b1X(s)/s, b2X(s)/s2, and b3X(s)/s3. Because the transfer function of an
integrator is 1/s, the signals X(s)/s, X(s)/s2, and X(s)/s3 can be obtained by successive integration of the input
x(t). The left-half section of Fig. 4.7 (b) shows the realization of H1(s).
W(s)
X(s) H1(s) H2(s) Y(s)
(a)
X(s) b0 W(s)
Y(s)
1 1
s s
b1 –a 1
1 1
s s
b2 –a 2
1 1
s s
b3 –a 3
(b)
Fig. 4.8 : (a) Realization of a function in two steps and
(b) Direct form I realization of a third- order continuous-time LTI system
We next consider the realization of H2(s) given by, equation (61). Equation (62) shows that the output
Y(s) can be synthesized by subtracting a1Y(s)/s, a2Y(s)/s2, and a3Y(s)/s2 from W(s). To obtain signals Y(s)/s,
Y(s)/s2, and Y(s)/s2, we assume that we already have the desired output Y(s). Successive integration of Y(s)
yields the needed signals Y(s)/s, Y(s)/s2, and Y(s)/s3. The right-half section of Fig. 4.8 (b) shows the realization
of H2(s).
We can generalize this procedure, known as the direct form I realization, for any value of N. This
procedure requires 2N integrators to realize an Nth-order transfer function. This realization is noncanonic since
it employs six integration to implement a third-order transfer function. A realization is canonic if the number of
integrators used in the realization is equal to the order of the transfer function realized. Thus, canonic realization
has no redundant integrators.

4.8.2 Direct Form II (or Canonic) Realization

In Direct form I, we realize H(s) by implementing H1(s) followed by H2(s), as shown in Fig. 4.9 (a). We
can also realize H(s), as shown in Fig. 4.9, where H2(s) is followed by H1(s). This procedure is known as the
direct form II realization.
V(s)
X(s) H2(s) H1(s) Y(s)
(a)
Study Note
The direct form II realization Fig. 4.9 (b) implements zeros first [the left-half section represented by H1(s)] followed
by realization of poles [the right-half section represented by H2(s)] of H(s). In contrast, the direct form II realization
implements poles first followed by zeros.
Again consider for simplicity a third-order system characterized by a transfer function,
b0 s3 b1s2 b2 s b3
H(s ) =
s3 a1s2 a2 s a3
b0 ( b1 / s) ( b2 / s2 ) ( b3 / s3 )
H(s ) =
1 ( a1 / s) ( a2 / s2 ) ( a3 / s3 )
1 b1 b2 b3
We can express H(s) as, H(s ) = 2 3
b0
1 ( a1 / s) ( a2 / s ) ( a3 / s ) s s2 s3
H2 ( s ) H1( s )
We can realize H(s) as a cascade of transfer function H2(s) followed by H1(s), as depicted in Fig. 4.8(a),
where,
V( s) 1
H2(s) = ...(63)
X( s ) 1 ( a1 / s ) ( a 2 / s2 ) ( a3 / s3 )
a1 a2 a3
V(s) = X( s) V( s) ...(64)
s s2 s3
Y (s ) b1 b2 b3
and H 1( s ) = b0 ...(65)
V (s ) s s2 s3
b1 b2 b3
Y(s) = b0 V( s) ...(66)
s s2 s3
The left-half section of Fig. 4.9(b) shows the realization of H2(s) and the right-half section shows the
realization of H1(s).

X(s) V(s) b0 Y(s)
1 1
s s
–a1 b1
1 1
1 1
s s
–a2 b2
2 2
1 1
s s
–a3 b3
3 3
(b)
X(s) V(s) b0
Y(s)
1
s
–a 1 b1
1
s
–a 2 b2
1
s
–a 3 b3
(c)
Fig. 4.9 : Direct form II realization of a third-order LTI system
We observe that in Fig. 4.9 (b), the signal variables at nodes 1 and 1’ are the same, and hence the two
top integrators can be shared. Likewise, the signal variables at nodes 2 and 2’ are the same, which permits the
sharing of the two middle integrators. Following the same argument, we can share the integrators, leading to
the final structure shown in Fig. 4.9 (c).
Study Note
This implementation halves the number of integrators to N, and is thus more efficient in hardware utilization. This
is the direct form II realization. This realization is canonic since it employs N integrators to implement an Nth order
transfer function.

Example 4.31
4s 28
Find the canonic realization of the transfer function, H( s ) 2
.
s 6s 5
Solution 4.31
Y( s) V( s) Y( s)
Given that, H (s ) =
X( s) X( s) V( s)
4 28
4s 28 s s2 1 4 28
= 2 2
s 6s 5 1
6 5 1 (6 / s) (5 / s ) s s2
s 2
s H2 ( s ) H1( s )
V( s) 1
where, H 2( s ) =
X( s) 6 5
1
s s2
V(s) = X( s) 6 V( s) 5 V(2s)
s s X(s)
Y( s) 4 28
and H 1( s ) =
V( s) s s2 1
s
Y(s) = 4 V (s ) 28 V (s ) –6 4
Y(s)
s s2
4s 28 1
The given transfer function H( s) is of the second s
2
s 6s 5
–5 28
order therefore, we need only two integrators for its realization.
(d)
4.8 Laplace Transform of Causal Period Signal

The Laplace transform of a causal periodic signal can be determined from the knowledge of the Laplace
transform of its first cycle (period). Consider a causal periodic signal x(t) with period T0 as shown in Fig. 4.10.
x(t)
x 1(t) x 2(t) x 3(t)
t
0 T0 2T0 3T0 4T0
Fig. 4.10 : Causal periodic signal
Let x1(t), x2(t), x3(t)..., be the signals representing the 1st, 2nd, 3rd, ...., cycles of a causal periodic
signal x(t). Therefore, the causal periodic signal x(t) can be written as,
x(t ) = x1(t) + x2(t) + x3(t) + x4(t) +....
= x1(t) + x1(t – T0) + x1(t – 2T0) + x1(t – 3T0) +....
Assume x1(t) X1(s). Using the time-shifting property, the Laplace transform of the above equation
becomes,

sT0 s2T0 s3T0

X(s ) = X1( s) X1( s) e X1( s) e X1( s) e ....
sT0 s2T0 s3T0
= X1( s) [1 e e e ....]
snT0 sT0 n
= X1( s) e X1( s) (e )
n 0 n 0
1
X(s ) = X1( s) sT0
1 e
Transform, the Laplace transform of a periodic signal x(t) is given by,
X1( s)
X (s ) = ...(67)
1 e sT0
Example 4.32
Find the Laplace transform of the square wave shown in Fig. (a).
Solution 4.32
We know that the Laplace transform of a causal periodic signal is given by equation,
X1( s)
X (s ) =
1 e sT0
x(t) x 1(t)
A A
t t
0 1 2 3 4 5 0 1 2
–A –A
(a) (b)
Figure : (a) Causal periodic sequence wave (b) first cycle of x(t)
where X1(s) is the Laplace transform of the first cycle x1(t) of the causal periodic signal x(t). The given
square wave is a causal periodic signal with period T0 = 2. The first cycle x1(t) of x(t) is as shown in
Fig. (b). Using the step function u(t), x1(t) can be written as,
x 1( t ) = Au(t) – 2 Au(t – 1) + Au(t – 2)
L[x1(t)] = AL[u(t)] – 2 AL[u(t – 1)] + AL[u(t – 2)]
A 2 A s A 2s
X 1( s ) =
e e
s s s
A A
= [1 2 e s e 2 s ] (1 e s )2
s s
Substituting X1(s) and T0 = 2 in the expression of X(s), we get
X 1(s ) A (1 e s )2 A (1 e s )2
X (s ) sT0 =
1 e s 1 e 2s s (1 e s ) (1 e s )
s
A1 e A e s / 2 ( es / 2 e s / 2 )
= s
s 1 e s e s / 2 (e s / 2 e s / 2 )
A es / 2 e s / 2 A s
X(s ) = tan h
s es / 2 es / 2 s 2

Example 4.33
1
An LTI system is described by H( s ) 2
. Find the system response for the input:
s 3s 1
( i ) x ( t ) = 2 e –2 t
( i i ) x ( t ) = 2 cos(2 t + 20°)
Solution 4.33
Ct
( i ) When the input is exponential eCt , the response of system with transfer function is e H (s ) s C
Given, x(t ) = 2e–2t

(This is real exponential with C = –2)
1
H(–2) = 1
4 6 1
Hence, ycs(t) = –2e–2t
( i i ) x(t) = 2 cos(2t + 20°), we know that output for A cos( ot ) is A cos( ot )H ( j o ).
Here o = 2, thus response will be

yss ( t ) = 2 H( j 2) cos(2t 20 H( j 2) )
= 0.4 cos(2t + 20° + 53.1°)
= 0.2 cos(2t + 73.1°)
Example 4.34
An absolutely integrable signal is known to have a pole at s = 2. Answer the following

questions:
( i ) Could x ( t ) be of finite duration?
( i i ) Could x ( t ) be left sided?
( i i i ) Could x ( t ) be right sided?
( i v ) Could x ( t ) be two sided?
Solution 4.34
(i) No we know that for a finite duration signal, ROC is the entire s-plane. Therefore, there can no poles
in finite s-plane for a finite duration signal. Since, the given signal x(t) has a pole at s = 2, this is not
the case.
( i i ) Yes since the signal is absolutely integrable, the ROC must include the j -axis. Furthermore, X(s) has
a pole at s = 2. Therefore, one valid ROC for signal would be < 2. Signal it would correspond to a
left-sided.
( i i i ) Since the signal is absolutely integrable, the ROC must include the j -axis. Furthermore, X(s) has a
pole at a s = 2. For a right-sided signal ROC is of the form > a for some constant a and ROC does
not contain any pole. Therefore, x(t) cannot be right-sided signal.
( i v ) Yes since the signal is absolutely integrable, the ROC must include the j -axis. Furthermore, X(s) has
a pole at s = 2. Therefore, a valid ROC for the signal could be a strip in the region a < R{s} < 2 such
that a < 0 and we know that this would correspond to a two-sided signal.

Example 4.35
How many signals have a Laplace transform that my be expressed as,

1
X( s )
(s 2) ( s 3) ( s2 s 1)
in its region of convergence?
Solution 4.35
We may find different signals with the given Laplace transform by choosing different region of convergence.
Consider the given Laplace transform,
1
X (s ) =
( s 2) ( s 3) ( s2 s 1)
1
=
1 j 3 1 j 3
( s 2) ( s 3) s s
s 2 2 2
The poles of the given Laplace transform are

s 1 = –2, s2 = –3
1 3 1 3
s3 = j ; s4 j
2 2 2 2
The pole-zero diagram is shown in figure. Based on the location of these poles, we may choose from the
following ROCs:
1 1
(i ) ( ii ) 2
2 2
(iii) –3 < < –2 (iv ) < –3
Therefore, we may find four different signals with the given Laplace transform.
Example 4.36
Let x ( t ) be a signal that has a rational Laplace transform with exactly two poles located at
s = –1 and s = –3. If g ( t ) = e 2t x ( t ) and G ( ) = F [ g ( t )] converges, determine whether x ( t )
is left sided, right sided, or two sided.
Solution 4.36
It is given that g(t) = x(t) e2t is absolutely integrable, since Laplace transform X(s) has two poles –1 and –3.
So X(s) can be have three possibilities of ROC.
1. < –3 2. –1 < < –3 3. > –1
From the definition of ROC, ROC is value of where x(t) e is absolutely integrable. Since x(t) e2t is
– t
absolutely integrable so ROC must contain = –2. So ROC of X(s) is –3 < < –1. So, x(t) is double-sided
signal.
Example 4.37
Suppose the following facts are given about the signal x ( t ) with Laplace transform X ( s ):
1 . x ( t ) is real and even.
2 . X ( s ) has four poles and no zeros in the finite s-plane.

1 j /4
3 . X ( s ) has a pole at s e .
2
4. x( t ) dt 4.
Determine X ( s ) and it’s ROC.

Solution 4.37
Given that, x(t) is real and even. So poles will be in conjugate
pairs and will be at mirror image with respect to j -axis also.
Im
1 j /4
1st : Pole at P1 e , 1
j
2 2 2
1 j /4
2nd : Pole will be P2 P1 e ,
2 Re
1 1
1 j 2 2 2 2
3rd : Pole will be mirror image of P1 i.e. P3 P1 e /4
and
2
1 1
j /4 j
4th : Pole will be at mirror image of P2 i.e. P4 P2 e 2 2
2
Thus, there are three ROC is
1 1 1 1
1. 2. 3.
2 2 2 2 2 2 2 2
and x(t) is absolutely integrable so ROC must certain j -axis so ROC is
A
X (s ) =
( s P1 ) ( s P2 ) ( s P3 ) ( s P4 )
st
X(0) = x(t ) dt 4
So, X(0) = 4 will give value of A

1
so we can find, A =
4
1/ 4 1 1
X (s ) = ,
( s P1 ) ( s P2 ) ( s P3 ) ( s P4 ) 2 2 2 2
Example 4.38
Determine the impulse response h ( t ) of the system H ( s ) of an LTI system from the following
facts:
1 . When the input to the system is x ( t ) = e 2 t and the output is y ( t ) = 1/6 e 2t .
dh( t )
2 . When h ( t ) satisfies the differential equation, 2h( t ) e 4t
u( t ) b u( t ) where b is an
dt
unknown constant. Your answer must not contain any unknown constant.

Solution 4.38
We know that if we apply a complex exponential input x(t) = eCt to an LTI system with impulse response
h(t), the system output will be y(t) = H(C) eCt, etc.,
x(t ) y(t)
eCt H(C) eCt
1 2t
Given that, e2t e = H(2) e2t
6
Taking the Laplace transform of the equation given in question, we obtain
1 b
sH(s) + 2H(s) =
s 4 s
s b( s 4)
(s + 2) H(s) =
s( s 4)
H( s) s 2 6b 1
Also, given that, 2 = H(2)
2 4 6 6
Thus, b = –1
s ( s 4) 2( s 2)
Therefore, H (s ) =
s( s 2) ( s 4) s( s 2) ( s 4)
2
H (s ) =
s( s 4)
1/ 2
1/ 2
H(s ) =
s s 4
The inverse Laplace transform of the above equation yields,
1 4t
h( t ) = [1 e ] u(t )
2
Example 4.39
The input x ( t ) and output y ( t ) of an LTI system are related through above representation
shown in figure. Find differential equation that represent the system and find all properties
of system.
x(t) y (t)
1
s
–2 –1
1
s
–1 –6

Solution 4.39
Let us draw the system again, we can see that,
V(s)
x(t) y (t)
1 1
s s
–2 –1
1 1
s s
–1 –6
V( s) V( s) X( s)
X (s ) = 2 V( s) V(s) =
s s2 2 1
1
s s2
V( s) 6 V( s)
and Y ( s ) = V( s )
s s2
1 6
X( s) 1
1 6 s s2
Y(s) = V( s) 1 Y(s) = 2 1
s 2
s 1
s s2
Y( s) s2 s 6
= H( s)
X( s) s2 2 s 1
Now, in such systems since system is realizable so it is causal and since pole of H(s) are at –1, –1 so ROC is
> –1 so system is stable.
Now we can easily differential equation.
(s2 + 2s + 1) Y(s) = X(s) (s2 – s – 6)
d2 y(t ) dy(t ) d2 x(t ) dx(t )
2 y (t ) = 6 x(t )
dt2 dt dt 2 dt
Study Note
• A finite duration signal has ROC entire s-plane thus it will never have a finite location pole. If pole is at s =
in that case ROC is entire s-plane except s = . Example (t) which has Laplce transform ‘s’.
(t ) s , ROC entire s-plane except s =
• The signal which is both sided must have minimum of 2 poles with different real values.
• If in any question transfer function of a system, H(s) is given and no information about it then assume it
to be causal.
• If an LTI system has transfer function H(s) then it is always invertible and invertible system will have
1
transfer function HI ( s) .
H( s)

1. x(t) e–3t is absolutely integrable.

2. x(t) (e–t u(t)) is absolutely integrable.
3. x(t) = 0, t > 1.
1 Subjective Practice Problems

4. x(t) = 0, t < –1.
Im
2j
Im
2j
1 . Consider the signal,

x(t) = e–5t u(t) + e– t u(t) –2 2
Re
–2 2
Re
and denote its Laplace transform by X(s). What are

the constraints placed on the real and imaginary parts –2j –2j
if the region of convergence of X(s) is Re{s} > –3?

Im Im
2 . Suppose we are given the following three facts about
2j 2j
the signal x(t):
1. x(t) = 0 for t < 0
Re Re
2. x(k/80) = 0 for k = 1, 2, 3.... –2 2 –2 2
3. x(1/160) = e–120
–2j –2j
Let X(s) denote the Laplace transform of x(t), and
determine which of the following statements is
consistent with the given information about x(t):
6 . We are given the following five facts about a real
(a) X(s) has only one pole in the finite s-plane.
signal x(t) with Laplace transform X(s):
(b) X(s) has only two poles in the finite s-plane.
1. X(s) has exactly two poles.
(c) X(s) has more than two poles in the finite s-plane.
2. X(s) has no zeros in the finite s-plane.
3 . Consider two right-sided signals x(t) and y(t) related 3. X(s) has a pole at s = –1 + j.
through the differential equations, 4. e2t x(t) is not absolutely integrable.
dx(t ) 5. X(0) = 8.
= –2 y(t) + (t) Determine X(s) and specify its region of convergence.
dt
dy(t ) 7 . Consider an LTI system for which the system function
and = 2 x(t)
dt H(s) has the pole-zero pattern shown in figure.
Determine Y(s) and X(s), along with their regions of (a) Indicate all possible ROCs that can be associated
convergence. with this pole-zero pattern.
4 . A causal LTI system S has the block diagram (b) For each ROC identified in part (a), specify
representation shown in figure. Determine a whether the associated system is stable and/or
differential equation relating the input x(t) to the causal.
output y(t) of this system. Im
x(t) 2/s y (t)
–4 Re
–2 –1 +1 +2
1/s
–2 8 . Determine whether each of the following statements

is true or false. If a statement is true, construct a
5 . For each of the following statements about x(t), and
convincing argument for it. If it false, give a counter
for each of the four pole-zero plots in figure,
example:
determine the corresponding constraint on the ROC:

(a) The Laplace transform of t 2 u ( t ) does not 12. Determine whether or not each of the following
converge anywhere on the s-plane. statements about LTI system is true. If a statement is
2
(b) The Laplace transform of e t u (t ) does not true, construct a convincing argument for it. If it
false, give a counter example.
converge anywhere on the s-plane.
(a) A stable continuous-time system must have all
(c) The Laplace transform of e j 0t does not converge
its poles in the left half of the s-plane
anywhere on the s-plane.
(d) The Laplace transform of e j 0t u(t) does not [i.e. Re{s} < 0].
converge anywhere on the s-plane. (b) If a system function has more poles than zeros,
(e) The Laplace transform of t does not converge and the system is causal, the step response will
anywhere on the s-plane. be continuous at t = 0.
(c) If a system function has more poles than zeros,
9 . Let h(t) be the impulse response of a causal and and the system is not restricted to be causal, the
stable LTI system with a rational system function.
step response can be discontinuous at t = 0.
(a) Is the system with impulse response dh(t)/dt
(d) A stable, causal system must have all its poles
guaranteed to be causal and stable?
and zeros in the left half of the s-plane.
t
(b) Is the system with impulse response h( ) d 13. Suppose that we are given the following information
about an LTI system:
guaranteed to be causal and unstable?
1. The system is causal.
10. Let x(t) be the sampled signal specified as, 2. The system function is rational and has only two
nT poles, at s = –2 and s = 4.
x(t) = e (t nT )
n 0
3. If x(t) = 1, then y(t) = 0.
where, T > 0. 4. The value of the impulse response at t = 0+ is 4.
(a) Determine X ( s ), including its region of Determine the system function.
convergence. 14. Consider a stable and causal system with impulse
(b) Sketch the pole-zero plot for X(s). response h(t) and system function H(s) is rational,
(c) Use geometric interpretation of the pole-zero plot
contains a pole at s = –2, and does not have a zero
to argue that X(j ) is periodic.
at the origin. The location of all other poles and
11. Consider a stable and causal system with a real zeros is unknown. For each of the following
impulse response h(t) and system function H(s). It is statements let us determine whether we can
known that H(s) is rational, one of its poles is at definitely say that is true, whether we can definitely
–1 + j, one of is zeros is at 3 + j, and it has exactly say that it is true, whether we can definitely say that
two zeros at infinity. For each of the following it is false, or whether there is insufficient information
statements, determine whether it is true, whether is to as certain the statement’s truth:
false, or whether there is insufficient information to (a) L[h(t) e3t} converges.
determine the statement’s truth.
(a) h(t) e–3t is absolutely integrable. (b) h(t ) dt 0.
(b) The ROC for H(s) is Re{s} > –1.
(c) The differential equation relating inputs x(t) and (c) t h(t) is the impulse response of a causal and
outputs y(t) for system may be written in a form stable system.
having only real coefficients. (d) dh(t)/dt contains at least one pole in its Laplace
(d) Lim H( s) 1. transform.
s
(e) H(s) does not have fewer than four poles. (e) h(t) has finite duration.
(f) H(j ) = 0 for at least one finite value of . (f) H(s) = H(–s).
(g) If the input to system is e3t sint, the output is (g) Lim H( s) 2.
s
e3t cost.

15. Find the inverse Laplace transform of the following 17. The Laplace transform of a signal x(t) that is zero
functions: for t < 0 is
2s 5 s3 2 s2 3s 2
(a) X( s) ; 3 R{s} 2 X( s)
( s 2) ( s 3) s
2s 2s 2s 24 3 2
2s 5 Determine the Laplace transform of the following

(b) X( s) ; 2 R{s} 3 signals:
( s 2) ( s 3)
t
2s 3 (a) y(t ) 3x
(c) X( s) ; R{s} 1 3
( s 1) ( s 2)
(b) y(t) = tx(t)
2s 3 (c) y(t) = t x(t – 1)
(d) X( s) ; R{s} 2
( s 1) ( s 2) dx(t )
(d) y(t )
dt
16. An absolutely integrable signal x(t) has a pole at
s = . It is possible that other poles may be present. dx(t )
(e) y(t ) (t 1) x(t 1)
(a) Can x(t) be left-sided? Explain. dt
(b) Can x(t) be right-sided? Explain.
(c) Can x(t) be Two-sided? Explain.
(d) Can x(t) be finite duration? Explain.

Chapter
Continuous Time
Fourier Transform 5
5.1 Introduction:
In chapter 3 we have seen how Fourier series can be used to represent any periodic signal in terms of
linear combination of harmonically related Complex exponential of form e j o kt . In this chapter we will try to
express any periodic or non-periodic signal in terms of ejwt, that is we will find Fourier Transform of signals
which specifies the spectral content of a signal.
We already know Laplace Transform, the Fourier transform is simply Laplace transform with s = 0 that
is Laplace transform at jw axis is Fourier Transform.
5.2 Fourier Transform Representation of non-Periodic signal

In case of Fourier series the periodic signal is represented as
x(t) = Cke j o kt
... (1)
k
In frequency domain this can be represented as discrete signal shown in Fig(5.1)
Ck
C–1 C0 C1
C–2 C2
C3
–2 0 – 0 0 0 2 0 3 0
Fig. 5.1 (Magnitude plot of Ck with respect to )
As the time period (T) of the periodic signal is increased the o will decrease and if we view a non-periodic
signal as periodic with period T ,then as the period increase the fundamental frequency decreases and the
coefficients Ck will come closer on the plot of Ck versus . When T o tends to zero and we can see that the
coefficients Ck come so close to each other that disrete plot become a continuous plot. Thus the magnitude

spectrum of non periodic signal is not a line spectrum (as with periodic signal) but a continuous plot The same
is true with phase spectrum.
To clarify how the change from discrete to continuous spectra takes place, consider the periodic signal
x(t ) shown in Fig 5.2. Now think of keeping the waveform of one period of x(t ) unchanged, but carefully and
intentianally increase T. In the limit as T , only single pulse remains because the nearest neighbours have
been moved to infinity.
x(t) T
x(t)
0 T 0
Fig. 5.2: Allowing the period T of periodic signal to increase
to obtain non-periodic signal x(t)
2
As explained earlier that as the period T increases the fundamental frequency 0 decreases and
T
the harmonically related components become closer in frequency. As T ® ¥, the spacing between lines approaches
zero. This means that the spectral lines move closer, eventually becoming a continuum. The overall shapes of
the magnitude and phase spectra are determined by the shape of the single pulse that remains in the new signal
x(t), which is aperiodic.
To investigate what happens mathematically, we use the exponential form of the Fouriers series
representation for x(t ) i.e.,
x(t ) = X ne jn 0t
... (2)
n
T /2
1 jn 0t
where, Xn = x(t)e .... (3)
T T /2
2
In the limit as T , we see that 0 becomes an infinitesimally small quantity, d , so that
T
1 d
T 2
We argue that in the limit, n 0 should be a continuous variable. Then, from equation (3), the Fourier
coefficients per unit frequency interval are
Xn 1
= x (t)e j t
dt ... (4)
d 2
Substituting equation (4) into equation (2), and recognizing that in the limit the sum becomes an

Discrete Time Fourier Transform 205
integral and x(t ) approaches x(t), we obtain
d
x(t) = x(t)e j t
dt e j t
... (5)
2
The inner integral, in brackets, is a function of only and not t. Denoting the integral by X( ), we can
write Equation (5) as
1
x(t) = X( )e j td ... (6)
2
j t
where X( ) = x (t )e dt ... (7)
Equations (6) and (7) are referred to as the Fouriers transform pair, with the function X( ) referred to
as the Fourier transform or Fourier integral of x(t) and equation (6) as the inverse Fouriers transform equation.
X( ) plays the same role for non-periodic signals that Xn plays for periodic signals. Thus, X( ) is the spectrum
of x(t) and is a continuous function defined for all values of , whereas Xn is defined only for discrete frequencies.
Therefore, a non-periodic signal has a continuous spectrum rather than a line spectrum.
Sometime we use X(f) as Fourier transform of x(t). This X(f) is written.
X( f ) = [ x (t)] x (t)e j2 ft
dt ... (8)
The inverse Fourier transform in terms of cyclic frequency f is written as
x(t) = ƒ 1[X(f )] X(f )e j2 ftdt ... (9)
5.3 Dirichlet Conditions For Fourier Transform

Dirichlet showed that if x(t) satisfies certain conditions (Dirichlet conditions), its Fourier transform is
guaranteed to converge pointwise at all point where x(t) is continuous. These conditions are as follows:
1. x(t) be absolutely integrable, i.e.,
x(t) dt < ... (10)
This guarantees that X( ) will be finite since
X( ) = x(t)e j t
dt
X( ) x(t)e j t
dt x(t) dt

So, if x(t) dt < ,
then, X( )
2.In any finite interval of time, x(t) is of bounded variation, i.e., x(t) have a finite number of maxima and
minima.
3 .In any finite interval of time, there are only a finite number of discontinuity. Furthermore, each of
these discontinuities if finite.
The condition just given for the existence of the Fourier transform of x(t) are sufficient conditions. This
means that there are signals that violate either one or both conditions and yet posses a Fourier transform.
Examples are power signals (unit step signal, periodic signals, etc.) that are neither absolutely integrable nor
square integrable over an infinite interval, but still have Fourier transforms.
Remember: Dirichlet condition of Fourier transform are sufficient but not necessary conditions
5.4 Finding Fourier transform of a signal

1. For energy signals:-
For all energy signals, that is that signals which are absolutely integrable the best method is
Step 1: Find the Laplace Transform of the signal x(t)
Step 2: If ROC of X(s) contain j axis i.e s = 0 then we can find it’s Fourier transform because then
only the signal x(t) will be absolutely integrable. If ROC don’t contain j axis then we can’t find Fourier
Transform of signal using this method
Step 3: If ROC of X(s) contain j axis then
X(j ) = X ( s ) σ 0 X ( s ) s=jω
that is X(j ) or X( ) is simply found by replacing s by j in expression of X(s)
Study Note
1.If in a question we have to find magnitude spectrum of any signal x(t) then find fourier transform of signal X( )
and magnitude spectrum of signal is magnitude of X( ).
2. Similarly phase spectrum of x(t) is phase of X( ).
Example 5.1
Find the Fourier transform of continuous time signal x(t),

x(t) = e–at u(t), a > 0
Also find it’s magnitude and phase spectrum
Solution 5.1
We know that
at Laplace 1
x(t) = e u (t ) Transform s+a

Since pole of X(s) is at s = – a thus ROC will be > a because x(t) is right sided signal thus ROC contain
j axis. Thus
1
X( ) = X(s) s=j =
j a
Since this Fourier transform is complex valued, to plot it as a function of , we express X( ) in terms of its
magnitude and phase:
1
X( ) =
a2 2
1
X( ) = tan
a
This magnitude spectrum X( ) and phase spectrum X( ) are depicted in Fig (a) and (b). Observe that
X( ) is an even function of and X( ) is an odd function of as expected.
X( ) X( )
/2
1/a
/4
1 a
a2 –a
– /4
–a 0 a – /2
(a) (b)
Fig. (a) Magnitude spectrum and (b) Phase spectrum
Example 5.2
Determine the fouriers transform of signal

at
x(t) = e ,a 0
Solution 5.2
We know that Laplace transform of e–a|t| is
at Laplace 2a
e Transform 2
a s2
e–a|t| = e–atu(t) + eatu(–t)
1
e atu (t ) Laplace
,ROC( a)
Transform
s a
1
e atu ( t ) Laplace
,ROC( a)
Transform
a s

Thus e–a|t| will have Laplace transform
1 1 2a
=
s a a s a2 s2
and ROC is –a < < a, since a > 0 thus ROC is defined and ROC contains j axis
2a
X ( ) = X (s ) s j 2 2
.
a
The plot of X( ) is
X( )
2/a
1/a
–a 0 a
Example 5.3
Determine the Fourier transform of the unit impulse

x(t) = (t)
Solution 5.3
By definition,
j t j t j t
X( ) = x (t )e dt (t )e dt e 1
t 0
(t) 1
That is, the unit impulse has a Fourier transform consisting of equal contributions at all frequencies. The
unit impulse function and its Fourier transform are depicted in Fig. (a) and (b).
x(t) x(t)
1
1
t
0 0
(a) (b)
Fig. (a) Unit impulse function and (b) its Fourier transform
Example 5.4
Find the inverse Fourier transform of X(w) = d(w).
Solution 5.4

1
We have, x(t) = ƒ 1
X( ) X ( )e j t d
2
1
= ( )e j t d
2
1 j t 1
ƒ 1[ ( )] = e
2 0 2
1
( ) = ƒ
2
1
Therefore, ( )
2
or, 1 2 ( )
This result shows that the spectrum of a constant signal x(t) = 1 is an impulse 2 ( ). as illustrated in Fig
(a) and (b).
x(t) x(t)
........
1
2
........
t
0 0
(a) (b)
Fig. (a) Constant signal and (b) its Fourier transform
Example 5.5
Find the inverse Fourier transform of X( ) = ( – 0).

Solution 5.5
1
We have, x(t) = ƒ 1
X( ) X ( )e j t d
2
ƒ 1[ (
1
o )] = ( 0 )e j t d
2
ƒ 1[ (
1 j t 1 j 0t
o )] = e e
2 0
2

1 j 0t
( – 0) = ƒ e
2
1 j 0t
Therefore e ( 0 )
2
or, ej 0t
2 ( 0 )
j 0t
Similarly, we have e 2 ( 0 )
Study Note
1.We can see that x(t) = 1 is a signal which is not absolutely but still we are able to find its Fourier transform. This
show that dirichlet conditions are not necessary.
2.Another important point is that we cannot find Laplace transform of x(t) = 1, x(t) but we can find their Fourier
transform.
2. For Power Signals
For signals like x(t) = 1, e j o t , cos( ot), sin( ot) we cannot find the Laplace transform and we cannot
apply the formulae equation (2) to find X( ) because equation (2) can be used to find fourier trasform only for
those signals which are absolutely integrable. To find Fourier transform of power signals we will use formulae of
inverse Fourier transform (as we did in example 5.4) or duality theorem (will come later in chapter).
1 2 ( ) .. (9)
and from example 5.5 we get
ej 0t 2 ( – 0) ... (10)
j 0t
and e 2 ( + 0) ... (11)
j 0t j 0t
e e
Since cos( t) = ,
2
thus Fourier transform of cos ( ot) is
2 ( 0) 2 ( 0)
cos ( ot)
2
( – 0) + ( + 0)
Example: 5.6
Find the Fourier transform of signal

(a) x(t) = sin( 0t)
t A, t t0
(b) x(t) = Arect
2T0 0, t t0
Solution 5.6

( a ) Since x(t) is a power signal whose Laplace transform don’t exist we will use equation 9, 10, 11 for
finding result.
j 0t j 0t
e e
Since sin( 0t ) =
2j
( 0) ( 0)
sin( 0t ) j
( b ) The signal is energy signal with Laplace transform X(s).
Since x(t ) = Arect t = A[u(t + T0) – u(t – T0)]

2T0
A T0s
X (s ) = [e e -T0s ]
s
Since x(t) is a finite duration signal thus ROC is entire s plane, Thus
X( ) = X(s) s = j
j T0 j T0
A[e e ] 2A
X( ) = sin( T0 )
j
2A
X( ) = sin( T0 )
T0
X( ) = 2AT0 sin c
T0
Since sinc 0 when sin( T0) = 0, where it appears to be indeterminate. This means that
T0
sin c 0 for T0 = ± n , n = 1, 2, 3,.... The Fourier transform X( ) shown in Fig(a) exhibits
positive and negative values. A negative amplitude can be considered to be a positive amplitude with a
T0
phase of – or . We use this observation to plot the magnitude spectrum X( ) = 2AT0 sinc
[Fig (b)] and the phase spectrum X( ) [Fig (c)].
X( )
2AT0
0
– 3 –2 –
T0 T0
2 3
T0 T0 T0 T0
(a)
Fourier transform of a gate pulse

X( )
X( )
2AT0
2 3 4
T0 T0 T0 T0
–4 –3 –2 –
T0
T0 T0 T0
0
–3 –2 –
T0 T0
2 3
T0 T0 T0 T0
(b) (c)
Magnitude spectrum Phase spectrum
Remember: sinc x = sin( x)/ x
Example: 5.7
Consider the Gaussian pulse signal depicted in Fig (a) and defined as
x(t) = e t2
Find the Fourier transform of x(t).
x( t) X( )
1 1
.
0 0
(a) Gaussian pulse (b) Fourier transform
-
Solution 5.7
2
( t2 j )
X( ) = x(t)e j t
dt ete j t
dt e dt
2 2
j
Substituting, t2 +j t = t gives
2 4
2 2
j j
t 2 2 t
X( ) = e 2
e /4
dt e /4
e 2
dt

j du
A change of variables is performed by letting u t , which also yields dt ,u as
2
variable t and u as t . Therefore
2
/4 u2 du 2
/4 2 u2
X( ) = e e e e du
Since u2
e du , we have
0 2
2
/4 2 2
/4
X( ) = e e
2
t2 2
/4 ... (12)
e e
t2 f2
or e e ... (13)
The Fouriers transform X( ) of a Gaussian pulse is a Gaussian pulse [as shown in Fig (b)]
Example: 5.8
Find the Fourier Transform of signum function x(t)

x(t) = sgn(t)
1, t 0
sgn(t) = 0, t 0
1, t 0
Solution 5.8
The signal sgn(t) is not absolutely integrable thus are cannot find X( ), using the formulae of equation (2).
We also cannot find Laplace transform of the signal
As sgn(t) = u(t) – u(–t)
If we calculate try to find X(s) of the signal then we will get x(t)
that its ROC will be null and thus Laplace transform does
1
not exit. So, we approach this problem by considering sgn(t)
e–at u(t)
to be a sum of exponential e–at u(t) – eat u(–t) in the limit as
a 0 (Fig (a)). t
–eatu (–t)
at –1
Thus, sgn(t) = lim[e u (t ) e atu ( t )]
a 0
Fig. (a) Signum function

and, X( ) = ƒ[sgn(t )] lim ƒ[e atu (t ) e atu ( t )]
a 0
1 1
= lim ƒ[e atu (t )] ƒ[e atu ( t )] lim
a 0 a 0 a j a j

2j 2j
= lim
a 0 a 2 2 2
2
X( ) = ƒ[sgn(t )]
j
2
sgn(t ) ... (14)
j
2
The magnitude spectrum is X( ) =
, 0
and the phase spectrum is X( ) = tan 1 2
0
, 0
2
Fig b) and (c) shows the magnitude and phase spectrum.
X( ) X( )
/2
– /2
0
(a) (b)
Fig. (a) Magnitude spectrum and (b) Phase spectrum of the signum function
Example: 5.9
Find the Fourier Transform of the unit step function u(t).

Solution 5.9
Step function u(t) is a power signal and the Laplace transform of u(t) don’t contain j axis so we cannot use
method of finding fourier transform using X(s). The unit step function can be expressed as
2u(I) = 1 + sgn(t)
1 1
u( t ) = sgn(t )
2 2
1 1
ƒ [u(t)] = U( ) = ƒ sgn(t )
2 2
1 1 1 1 2 1
U( ) = ƒ[1] ƒ[sgn(t )] 2 ( ) ( )
2 2 2 2 j j

1
u (t ) ( ) ... (15)
j
Study Note
1.From the above example we can see that we can find Fourier transform of energy signals using Laplace transform
and that of power signals using some change in approach or using equation (9, 10, 11).
2.For signals which are neither energy nor power we can find Laplace transform but cannot find Fourier transform
examples eatu(t), a > 0.
5.5 Properties of Fourier Transform

The Fourier Transform of x(t) is X( ),
x(t) X( )
Note: The properties of Fourier transform will be similar of that of Laplace transform and Fourier series.
5.5.1 Linearity
If
x 1(t ) X 1( ) and x 2 (t ) X 2( )
then, ax 1(t ) bx 2 (t ) aX 1( ) bX 2 ( ) ... (16)

Proof: The Fourier transform of ax1(t) + bx2(t) is given by
j t
ƒ [ax1(t) + bx2(t)] = [ax 1(t ) bx 2 (t )]e dt
j t j t
ƒ [ax1(t) + bx2(t)] = a x 1(t )e dt b x 2 (t )e dt aX 1( ) bX 2 ( )
X 1( ) X2( )
5.5.2 Time Shifting

If
x(t) X( )
j t0
then, x (t t0 ) X ( )e ... (17)
Proof: The Fourier transform of x(t – t0) is given by
j t
ƒ [x(t – t0)] = x (t t 0 )e dt

A change of variables is performed by letting = (t – t0), which also yields dt = d , as t ,

and as t . Therefore
j ( t0 )
ƒ [x(t – t0)] = x ( )e d e-j t0
x ( )e - j t d
= X ( )e - j t0
REMEMBER One consequence of this time shifting property is that when a signal is shifted in time, the
magnitudes of its Fourier transform remain unaltered. That is if we express X( ) in polar form
as
ƒ [x(t)] = X( ) = X( ) e j X( )
j t0
then ƒ [x(t – t0)] = X ( )e X ( )e j( X( ) t0 )
Thus, the effect of a time shift on a signal is to introduce into its Fourier transform a phase shift,
t0, which is a linear function of .
5.5.3 Frequency Shifting

If
x(t) X( )
then, x (t )e j t0
X( 0 ) ... (18)
Proof: The Fourier transform of x (t )e j 0t

is given by
ƒ[x (t )e j 0t ] = [ x (t )e j 0t
]e j t
dt [ x (t )e j( 0 )t
]dt
= X( – 0)
Remember: Hence a frequency shift corresponds to multiplication in time domain by a complex sinusoid whose
frequency is equal to the frequency shift. We can also see that in time shifting the effect go with same sign and in
frequency shift effect goes with opposite sign
5.5.4 Time and Frequency Scaling

If
x(t) X( )
1
then, x (at ) X ... (19)
a a
Proof: The Fourier transform of x(at) is given by

j t
ƒ [x(at)] = x (at )e dt
Case 1: For a positive constant a,
j t
ƒ [x(at)] = x (at )e dt
A change of variables is performed by letting = at, which also yields d = a dt, as t , and
as t . Therefore
1 j ( /a ) 1
ƒ [x(at)] = x ( )e d X
a a a
Case 2: For a negative real constant – a,
j t
ƒ [x(–at)] = x ( at )e dt
A change of variables is performed by letting = –at, which also yields d = –a dt, as t , and
as t . Therefore
1 j ( /a ) 1 j ( /a ) 1
ƒ [x(–at)] = x ( )e d x ( )e d X
a a a a
Combining the two cases, we have
1
ƒ [x(at)] = X
a a
REMEMBER The scaling property states that time compression of a signal results in its spectral expansion,
and time expansion of the signal results in its spectral compression. The relation between
compression in one domain and expansion in the order is the basis for an idea called the
uncertainty principle of Fourier analysis.
5.5.5 Time Reversal

If
x(t) X( )
then, x( t) X( ) ... (20)
Proof: Substituting a = –1 in equation (19), we have
1
ƒ [x(– t)] = X
1 1

ƒ [x(–t)] = X(– )
Study Note
The time-reversal property states that reversing a signal in time also reverses its Fourier transform. An interesting
consequence of the time-reversal property is that if x(t) is even, then its Fourier transform is also even, i.e.,
if x(–t) = x(t),
then, X(– ) = X( )
Similarly, if x(t) is odd, then so is its Fourier transform, i.e.,
If x(–t) = –x(t),
then, X(– ) = –X( )
Thus, for even signal x(t), X( ) is even and for odd signal x(t), X( ) is odd
5.5.6 Differentiation in Time Domain

If
x(t) X( )
dx (t )
then, j X( ) ... (21)
dt
Proof: We know that
1
x(t) = ƒ 1[ X ( )] [ X ( )]e j t d
2
Differentiating both sides gives
dx (t ) 1
= [ j X ( )]e j t d
dt 2
dx (t )
= ƒ –1[j X( )]
dt
dx (t )
Therefore, j X( )
dt
Similarly, the differentiation property can be extended to yield.
d 2 x (t )
( j )2 X ( ) ... (22)
dt 2
Example 5.10
Find the Fourier transform of signal using the differentiation in Time domain
t
(a) x (t ) Arec
T

t 0 1,
(b) x (t ) sgn(t ) 0, t 0
1, t 0
t
(c) x (t ) A
T
Solution 5.10
(a) Here x(t) = A rect(t /T) is shown in Fig (a) and it’s differentiations is shown in Fig(b)
x(t) x (t )
A A
T/2 t
–T /2 T/2 –T /2
(a) –A
(b)
dx (t ) T T
Thus = A t A t
dt 2 2
Thus Fourier Transform of x (t) is
Y( ) = Ae+j T/2 – Ae j T/2
Y( )
And Fourier Transform of x(t) is
j
Ae j T /2
Ae j T /2
sin( T / 2)
X( ) = 2A
j
(b) Here x(t) is sgn(t), it is shown in Fig (c) and it’s differentiation is shown in Fig(d)
x(t) x (t)
1 2
t t
–1
(c) (d)
We can see that x’(t) is 2 (t) and fourier transform of x’(t) is

Y( ) = 2

Y( )
Thus Fourier transform of x(t) is
j
2
X( ) =
j
(c) Here x(t) = A ( t/T), is shown in Fig(e) and x’(t), x”(t) are shown in Fig(f) and Fig(g).
x(t)
A dx(t) d 2x(t)
dt dt2
A
A/T
T
t t t
–T 0 T –T 0 –T 0 T
(e)
–A/T – 2A
(f) (g) T
The Fourier transform of the second derivative is
d 2 x (t ) A 2A A
ƒ = ƒ (t T ) (t ) (t T )
dt 2
T T T
A 2A A
= ƒ[ (t+T)] ƒ[ (t)]+ ƒ[ (t T)]
T T T
Now, using the differentiation and the time-shifting property, we get
A j T 2A A j T 2A e j T
e j T
2A
(j )2X( ) = e e
T T T T 2 T
2A
= [cos( T ) 1]
T
2A
= [1 cos( T )]
T
2
2
4A T sin( T / 2)
X( ) = sin 2
AT
T
2
2 ( T / 2)
2
sin( T / 2 ) T
= AT AT sinc 2
( T /2 ) 2


If
x1(t) X1( ) and x2(t) X2( )
then, x1(t) x2(t) X1( ) X2( ) ... (23)
The Fourier transform maps the convolution of two signals into the product of the Fourier transforms.
Proof: The Fourier transform of x1(t) * x2(t) is given by
j t j t
ƒ [x1(t) * x2(t)] = [ x 1(t ) x 2 (t )]e dt x 1( )x 2 (t )d e dt
Interchanging the order of integration and noting that x1( ) does not depend on t gives
ƒ [x1(t) * x2(t)] = [ x 1( ) x 2 (t )e j t dt d
By the time-shifting property, the bracketed term is X2( ) e–j , substituting this into the above equation
yields
j j
ƒ [x1(t) * x2(t)] = x 1( )( X 2 ( )e )d X 2 ( ) x 1( )e d
= X2( )X1( ) = X1( ) X2( )

Therefore x1(t) * x2(t) X1( ) X2( )
the frequency domain.
5.5.8 Multiplication Property

If
x1(t) X1( ) and x2(t) X2( )
1
then, x 1(t )x 2 (t ) [X 1( ) X 2 ( )] ... (24)
2
The Fourier transform maps the multiplication of two signals into the convolution of their Fourier
transforms.
Proof: The Fourier transform of x1(t) x2(t) is
j t 1
ƒ [x1(t)x2(t)] = [ x 1(t )x 2 (t )]e dt X 1( )e j td x 2 (t )e j t
dt
2
Interchanging the order of integration and noting that X1( ) does not depend on t yields
1
ƒ [x1(t)x2(t)] = X 1( ) [x 2 (t )e j t ]e j t
dt d
2

By the frequency-shifting property, the bracketed term is X2( – ). Substituting this into the above
equation yields
1 1
ƒ [x1(t)x2(t)] = X 1( )X 2 ( )d [ X 1( ) X 1 ( )
2 2
1
Therefore, x 1(t )x 2 (t ) [ X 1( ) X 2 ( )]
2
Example 5.11
Find the Fourier Transform of signals

(a) x(t) = y(t)* cos( 0t)
(b) x(t) = y(t). cos( 0t)
(c) x(t) = A rect(t/T) .cos( 0t)
(d) x(t) = cos( 0t) u(i)
(e) x(t) = e–t cos( 0t)
Solution 5.11
(a) Given that
x(t ) = y(t) * cos( 0t)
Let y(t) Y( )
and cos( 0t) ( ( – 0) + ( + 0))
We can apply property that convolution in time domain is multiplication in Fourier domain.
y(t)* cos( 0t) Y( ) [ ( ( – 0) + ( + 0)]
[Y( ) ( – 0) + Y( ) ( + 0)]
X( ) = [Y( 0) ( – 0) + Y(– 0) ( + 0)]
(b) Here x(t ) = y(t) cos ( 0t)

If y(t) Y( )
and cos( 0t) X( ) [ ( – 0) + ( + 0)]
By applying multiplication property we get
1
y(t) cos( 0t)
Y( ) [ ( ( 0 ) ( 0 )]
2
X( ) [Y ( ) ( 0 ) Y( ) ( 0 )]
2
1
[Y ( 0 ) Y( 0 )]
2
(c) Here, x(t ) = A rect(t/T) cos( 0t)

Using the result derived in part (b) and Y( ) in Fourier transform of A rect(t/T)

2 A sin( T / 2)
Y( ) =
1
XX ( ) = [Y ( 0 ) Y( 0 )]
2
( )T ( )T
A sin 0
A sin 0
2 2
X( ) =
0 0
(d) Here
ej 0t
e j 0t
x(t ) = cos( u(t )

0t) u(t) =
2
1
Since, u( t ) ( )
j
1
e j tu (t )
0
( 0 )
j( 0 )
1
and e j 0t
u (t ) ( 0 )
j( 0 )
j
cos( ot) u(t) ( 0 ) ( 0 ) 2 2
2 0
(e) Here
x(t ) = e–t cos( ot) u(t)
We can say that
x(t ) = (e–t u(t))cos( ot)
By applying the multiplication property we get
1 1
X( ) = [ ( 0 ) ( 0 )]
2 j 1
1 1 1
=
2 j( 0 ) 1 j( 0 ) 1
1 j (1 j )
=
1 j( 0 ) j( 0 ) ( 2 2
0 ) (1 j )2 2
0

5.5.9 Differentiation in frequency domain

If
x(t) X( )
dX ( )
then, –jtx(t) ... (25)
d
dX ( )
or tx(t ) j ... (26)
d
Proof: We know that
j t
X( ) = x (t )e dt
Differentiating both sides w.r.t. gives
dX ( ) j t
= [ jtx (t )]e dt ƒ[ jtx (t )]
d
dX ( )
Therefore, –jtx(t)
d
dX ( )
tx(t ) j ... (27)
d
5.5.10 Integration in time domain

If
x(t) X( )
t
1
then, x ( )d X( ) X (0) ( ) ... (28)
j
Proof: To prove this property we use convolution property
We can write
t
x(t)*u(t) = x ( )d
t
Thus Fourier Transform of x( )d is
t
1
x( )d X( ) ( )
j

X( )
X (0) ( )
j
5.5.11 Duality
If
x(t ) X( )
then, X(t) 2 x( )
Proof: By definition
1
x(t ) = X ( )e j td
2
2 x(t) = X ( )e j td
Replacing t with –t in the above equation gives
j t
2 x(–t) = X ( )e d
Now interchanging the variables t and yields
j t
2 x(– ) = X (t )e dt
2 x(– ) = ƒ [X(t)]
Therefore, X(t) 2 x(– )
Example 5.12
Using duality property find Fourier transform of
1
(a) x(t) = 1 (b) x(t) =
1 jt
1 1
(c) x(t) = (d) x(t) =
t 1 t2
Solution 5.12
(a) We know that
(t) 1
Thus by applying duality property we get
(t) 1

1 2 (– )
(Since delta function is even)
1 2 ( )
(b) Here
1
x(t ) =
1 jt
It would be difficult to use formulae to find X( ) thus we use duality. We know that
1
e–atu(t)
j a
1
e–tu(t)
j 1
Applying duality we get
1
2 e u(– )
jt 1
1
Fourier transform of is 2 e u(– )
jt 1
1
(c) Here x(t) = , the signal is not absolutely integrable and formulae cannot be used to find X( ). We
e
t
will use duality to find the result
The Fourier transform of
2
sgn(t)
j
Now applying duality we get
2
2 sgn(– )
jt
–jsgn( )
Since signum function is odd
j
2
–2 sgn( )
jt
–j
1
–j sgn(– )
t
The Fourier Transform of is shown in figure

j, 0
So we can see that –j sgn( ) is
j, 0
1
(d) Here x(t ) =
1 t2
2a
We know that e–a|t|
a 2 2
Thus by applying duality theorem we get

2a
2 e–a| |
a 2
t2
2a
2 e–a| |
a 2
t2
2
2 e–| |
1 t2
1
e–| |
1 t2
The signal x(t) and it’s Fourier transform are shown in Fig (a) and (b)
x(t) X( )
1/a
0 0
(a) (b)
Example 5.13
Find the Fourier transform G(w) of the signal [Fig (a)]
t
g ( t ) = A sin c
2
Solution 5.13
Define X( ) = A sinc by replacing t with in the expression of g(t). We have

2
t T0
A rect 2AT0 sin c
2T0

Substituting 2T0 = 1 yields
Arect(t ) A sinc
2
x(t ) X( )
x(t)=A rect(t) and X( )=A sinc

2
x(– ) = A rect(– ) = A rect( )
t
and X(t) = A sinc
2
Applying duality property of the Fourier transform
t
A sinc 2 A rect( )
2
1 1
t 2 A,
Therefore, ƒ A sinc( ) = 2 2 ... (30)
2
0, otherwise
The Fourier transform G( ) in shown in figure (b).

g(t)
A G( )
2 A
–4
t –½
–6 –2 0 2 4 ½
(a) (b)
Remember: From the above example we can see that ideal Low pass filter will have sinc function as impulse
response
5.5.12 Conjugation and conjugate symmetry

If x(t ) X( )
then, x*(t) X *( ) ... (31)
Proof: The Fourier transform of x *(t ) is given by
* *
j t j t j( )t
ƒ [x*(t)]= x (t )e dt x (t )e dt x (t )e dt

*
= X( )
Let us consider following cases

1. When x (t) is real
Thus x(t ) = x*(t)
If Fourier transform of x(t) is X( ) then Fourier transform of x*(t) is X*( )
If x(t) is real
X( ) = X*(– ) ... (32)
or X(– ) = X*( ) ... (33)
Study Note
1.When x(t) is real then it’s Fourier transform is conjugate symmetric.
Real conjugate symmetric
2. That is magnitude of X( ) is even and phase plot of X( ) is odd
2. When x ( t ) is real and even

Thus x(t ) = x*(t) and x(t) = x(–t)
If Fourier transform of x(t) is X( ) then Fourier transform of x*(t) will be X*(– ) and that of x(–t) will
be X*(– )
Thus, X( ) = X*(– ) and X( ) = X(– )
X(– ) = X*( ) = X( )
This show that X( ) is real and even.
Remember: Thus when x(t) is real and even then Fourier transform of x(t) will be real and even
real and even real and even
3. When x (t) is real and odd

Thus x(t ) = x*(t) and x(t) = –x(–t)
Thus X( ) = X*(– ) and X( ) = –X(– )
X(– ) = –X( ) = X*( )
Thus X( ) is odd and imaginary
Remember: When x(t) is real and odd the Fourier transform of x(t) will be imaginary and odd
real and odd imaginary and odd
4 .When x ( t ) is real then it’s even and odd decomposition:

If x(t) is real and x(t) X( )
x( t ) x( t )
Then Even {x(t)}
2
X( ) X( )
Even {x(t)}
2
Since x(t) is real, X( ) will be conjugate symmetric [X( ) = X*(– )]

X( ) X *( )
Even {x(t)}
2
Real {X( )}
Similarly
x( t ) x( t )
Odd {x(t)}
2
X( ) X( )
Odd{x(t)}
2
X( ) X *( )
2
j Img {X( )}
Study Note
1.We can see that if x(t) is real then X( ) is conjugate symmetric, it has some real part and imaginary part. The
real part of X( ) is due to even part of x(t) and imaginary part of X( ) is due to odd part of x(t)
2.In a question if Fourier transform of x(t) is given then Fourier transform of even part of x(t) is real part of X(w)
and odd part of x(t) is jImg part of X( ).
We have seen properties of few special cases here, we can extend them :
Time Domain signal Fourier transform
x(t ) X( )
Imaginary Conjugate anti symmetric
Real and even Real and even
Real and odd Imaginary and odd
Imaginary and even Imaginary and even
Imaginary and odd Real and odd
REMEMBER Here the arrow used is double sided means if x(t) is real then X( ) is conjugate symmetric and
if x(t) is conjugate symmetric then X( ) is real. Similarly for all others.
5.5.13 Area under x(t)

If x(t ) X( )
j t
then X( ) x (t )e dt
X(0) x (t )dt
jct
Thus X(0) is area under x(t). Similarly if in any question we have to find value of x (t )e dt then it is
equal to X(c).

5.5.14 Area under X( )

If x(t ) X( )
1 j t
then x(t) X( )e d
2
1
Thus x(0) X( )d
2
The area under X( ) = 2 x(0)
1
If in a question we have to find value of X( )e j d ,thus we have to find x (t ) t 1 =x(1)
2
1
If in a question we have to find value of X( )Y( )d ,then it correspond to value of time domain
2
in signal x(t)*y(t) at t = 0, Because x(t) * y(t) have fourier transform X( )Y( ).
1
If in a question we have to find value of X( )Y( )e jct ,then we have to find value of x(t) * y(t) at
2
t = C.
Example: 5.14
Find the total area under the function
t 4
x(t) = 10sinc
7
Solution 5.14
t
We have, ƒ A sinc = 2 A rect( )
2
t
A sin c 2 A rect( )
2
Using the time-scaling property (choose time-scaling constant a = 2 ). We get
Asinc(t) A rect
2
Substitute A = 10 in the above equation:
10 sinc(t) 10 rect
2
Using the time-shifting property, we get
4
4 j
10 sinc t 10 rect e 7
7 2

1
Again using the time-scalling property a= , we get
7
1 4 7
10 sin c t 70rect e j4
7 7 2
t 4 7
10 sin c 70rect e j4
7 2
t 4 7
We have, 10sinc dt = 70rect e j4 70
7 2 0
5.5.15 Parseval’s Theorem

Let x(t) be an energy signal and if
x(t) X( )
2 1 2
then Ex = x(t) dt X( ) d ... (34)
2
Parseval’s theorem states that the signal energies of an energy signal and its Fourier transform are
equal.
Proof: Consider the LHS of equation (34)
2
Ex = x(t) dt x(t)x *(t)dt
*
1
= x (t) X ( )e d j t
dt
2
1
= x(t) X *( )e j t
d dt
2
1 1
= X *( ) x(t)e j t
dt d = X *( )X( )d
2 2
2
x(t) dt =

Properties of the Fourier Transform
Property Signal Fourier transform
x(t) X(t)
x1(t) X1(t)
x2(t) X2(t)
Linearity a1x1(t) + a 2x 2(t) a1X1( ) + a2X2( )
Time shifting x(t – t0) e–j tX( )
Frequency shifting ej tx(t) X( – 0)

1
Time scaling x(at) X a
a
Time reversal x(– t) X(– )
Duality X(t) 2 x(– )
dx(t) j X( )
Time differentiation
dt
dX( )
Frequency differentiation (–jt)x(t)
d
t
Integration x(t )dt X( ) ( )+ X( )
Convolution x1(t) * x2(t) X1( )X2( )
Multiplication x1(t) x2(t) 1 X ( )*X ( )

1 2
2
Real signal x(t) = xe(t) + x0(t) X( ) = A( ) + jB( )
X(– ) = X*( )
Even component xe(t) Re{X( )} = A( )
Odd component x0(t) jIm{X( )} = jB ( )
Parseval’s relations
2 1 2
x(t ) dt X 1 (ω) dω
2π
Example - 5.15
If a time domain signal x(t) that is shown in Fig(a) has Fourier transform X( ) then find the value of
1 sin
X( ) e j 5t dt
2
x(t)
1.0
0.5
t
–10 –9 9 10 4 6
(a) (b) 5

Solution 5.15
sin( ) sin
We have to find value of signal with Fourier transform X( ) at t = 5. Signal corresponding to
1 t
is rect .
2 2
1 t
Thus we have to find value of x(t )* rect at t = 5
2 2
1 sin 1 5
Thus X( ) e j 5t dt = x( ) rect d
2 2 2
5
The figure of x(t) will be same as x( ) and rect is shown in figure (b)
2
1 5
Thus x ( ) rect d =1
2 2
Thus answer is 1
Example - 5.16
Find the phase response of x(t), if x(t) is as shown in figure (a)
x(t) y(t)
5 5
t t
–1 2 5 –3 3
(a) (b)
Solution 5.16
Since x(t) is real and x(t) can be made real and even by shifting it. If we create a new signal y(t) such that
y(t) = x(t + 2)
Then y(t) will be real and even. Since y(t) is real and even and Y( ) will also be real and even thus it will
have zero phase
Y( ) = |Y( )| e j Y( )
and Y( ) = 0
Using shifting property we can say that
Since y(t) = x(t + 2)
Y( )= X( )e j2
Y( ) = X( )e j2
|Y( )| e j Y( )
= |X( )| e j X( )
e j2
Since Y( ) = 0
X ( ) = –2
Thus phase response of X( ) is –2 .

5.6 Fourier Transform of periodic signals

We can construct the Fourier transform of any periodic signal using Fourier series. We cannot find
Fourier transform of periodic signal using formulae because periodic time domain signal is not absolutely
integrable.
If Fourier series coefficients of signal periodic signal x(t) are Ck then
x(t) Cke j o kt
k
The Fourier transform of
ej o kt 2 ( –k o)
and Ck e j o kt 2 Ck ( – k o)
Cke j o kt
2 Ck ( k o)
k k
Remember: To find Fourier Transform of any periodic signal first of all find the fourier series coefficients and then
use the above analysis.
Example - 5.17
Find and sketch the Fourier transform of the impulse train x(t) = To(t) shown in Fig (a)
Solution 5.17
The impulse train x(t) is periodic with period To.
Thus
x(t ) (t kTo )
k
We can find Fourier series coefficient of x(t)
To /2
1 j o kt
Ck = x (t )e dt
To To /2
To To
Here x(t) is (t), t .
2 2
To /2
1 j o Kt 1
Ck = (t )e dt Ck =
To To /2 To
X( ) = 2 Ck ( k o)
k
2
Here o =
To
2
X( ) = ( k o)
To k
The Fourier transform of periodic impulse train which period To in time domain is a periodic impulse train
2
in frequency domain with period o shown in figure (b)
To

x(t) X( )
1 0
........ ........ ........ ........
t
–3T0 –2T0 –T0 0 T0 2T0 3T0 –3 0 –2 0 – 0 0 0 2 0 3 0
(a) An impulse train (b)

(b) Fourier transform
5.7 Inverse Fourier Transform

To find inverse Fourier there are two methods that should be followed
Method 1: If expression of rational function X( ) is given in the question and we want to find
corresponding time domain signal x(t). The method is to replace by s/j or s = j then we get function
X(s) and we can find x(t) by applying inverse Laplace transform taking that ROC which contain imaginary
axis.
Method 2: If simpler form of X( ) is given in question.
X( ) = |X( )| e j X( )
Where |X( )| is magnitude of X( ) and X ( ) is phase of X( ).To find x(t) we can apply inverse
Fourier transform
1
x(t ) = X( )e j t d
2
Example - 5.18
If X( ) is Fourier transform of x(t)

X( ) = [u( + 3) – u( – 3)]ej3
Find x(t)
Solution 5.18
We can easily find x(t) using inverse Fourier transform formulae
1
x(t ) = [u( 3) u( 3)]e j 3 e j t d
2
1
= [u( 3) u( 3)]e j ( t 3)
d
2
3
1 1 e j 3(t 3) e j 3( t 3)
x(t ) = ej ( t 3)
d =
2 3 2 j(t 3)

sin(t 3)
x(t ) =
(t 3)
5.8 Signal Transmission Through LTI System

If x(t) and y(t) are the input and output of an LTI system with impulse response h(t), then
y(t) = x(t)*h(t)
Application of the time convolution property to the above equation yields[Assuming that both x(t) and
h(t) are Fourier transformable]
ƒ [y(t)] = ƒ [x(t)] ƒ [h(t)]
Y( ) = X( ) H( ) ... (35)
Y( )
or H( ) = ... (36)
X( )
Remember: If H(s) is transfer function of a system then H( ) = H(s) s=j
LTI y(t) = x(t) * h(t)

x(t)
h(t)
X( ) H( ) Y( ) = X ( )H( )
Fig 5.3 Convolution property of LTI system response
The use of the convolution property for LTI systems is demostrated in Fig 5.23 equation (35) can be
expressed in polar form as
Y( ) e j Y( ) = ( X( ) e j X ( )) H( ) e j H( )
= X( ) H( ) e j[ X( ) + H( )]
Therefore, Y( ) = X( ) H( ) ... (37)

and Y( ) = X( ) + H( ) ... (38)
During transmission, the input signal magnitude spectrum X( ) is change to X( ) H( ) . Similarly,
the input signal phase spectrum X( ) is changed to X( ) + H( ). An input signal spectral component of
frequency is modified in amplitude by a factor H( ) and is shifted in phase by an angle H( ). Clearly,
H( ) is the magnitude response, and H( ) is the phase response of the system. H( ) is called the frequency
Example - 5.19
Consider an LTI system with impulse response

h(t) = e–atu(t)
Find the response y(t) of this system when the input is the unit step function, i.e., x(t) = u(t).
Solution 5.19
1
We have H(s ) =
s a

1
and X (s ) =
s
Since Y(s) = H(s) X(s)
1
Y(s) =
s( s a )
1 1 1 1 1
Y(s) = =
a s sta as a( s a )
Since input and impulse response are casual thus output will be causal thus both poles correspond to right
sided signal
u(t ) e at u(t )
y(t) .
a a
REMEMBER We can use the knowledge of Laplace transform to solve question in fourier tranform for system
analysis. The Fourier transform can also be used but Laplace transform can make analysis
easily
Example - 5.20
The output of an LTI system in response to an input x(t) = e–2tu(t) is y(t) = e–tu(t). Find the frequency
response and impulse response of this system.
Solution 5.20
1 1
Here X (s ) = Y(s) =
s 2 s 1
Y( s) s 2 1
Since H(s ) = 1
X( s) s 1 s 1
Since input is causal and output is also causal so h(t) also causal thus
h(t) = (t) + e–tu(t)
Example - 5.21
Consider a stable LTI system characterized by the differential equation
d2 y(t ) dy(t ) dx(t )

4 3y (t ) 2 x(t )
dt 2
dt dt
(a) Find the frequency response H( ) and the impulse response h(t) of the system.
(b) What is the response of this system if the input x(t) = e–tu(t)?
Solution 5.21
(a) Given that
d2 y ( t ) dy(t ) dx(t )
4 3y ( t ) = 2 x(t )
dt 2
dt dt
Laplace
If y(t) Transform
Y(s)
Laplace
and x(t ) Transform
X(s)
Taking Laplace transform of above equation

s2Y(s) + 4sY(s) + 3Y(s) = sX(s) + 2X(s)
Y( s) ( s 2)
X( s) = 2
( s 4s 3)
Y( s) 1 1 1
H (s ) =
X( s) 2 s 1 s 3
Since poles are at –1 and –3 and since Fourier transform exist so system is stable and ROC must contain
= 0. Thus both poles are right sided
h(t) = 0 5e–tu(t) + 0 5e–3tu(t)
(b) Here x(t ) = e–tu(t)
1
X (s ) =
(s 1)
s 2
Since H (s ) = 2
s 4s 3
( s 2)
Y(s) = X(s) H(s)
( s 1)( s 3)( s 1)
( s 2)
=
( s 1)2 ( s 3)
1 1 1 1 1 1
=
4 ( s 1) 2 ( s 1)2 4s 3
Since Fourier transform Y( ) exist so ROC of Y(s) must have = 0, since Y(s) has poles at –1 and –3 thus
both poles give right sided signal
1 t 1 t 1 3t
y(t) = e te e u(t )
4 2 4
5.8.1 System Properties using H(w)

We have already analyzed system using Laplace transform. If in a question Fourier transform is given
then also we will use Laplace transform as we did in previous examples. If in any question system is defined and
it is given that fourier transform of system transfer function exist the first of all find H(s) then to find h(t) we
should apply inverse Laplace transform such that ROC of H(s) contain j ox is that is h(t) is absolutely integrable
because then only H( ) exist.
•If H( ) of system exist then system is already stable.
•For causal system if H( ) exist then it must be causal and stable. For causal and stable all poles of H(s)
or H( ) should lie in left side of j axis.
Remember: If in any question system is defined and it is given that fourier transform of system transfer function
exist than it means system is stable
5.8.2 Distortionless System

A system is called distortionless when the output signal has exactly same shape as that of input signal.
Thus output can be amplified and delayed version of input. Therefore, if x(t) is the input signal, the required

output is
y(t) = Kx(t – td) ... (39)
where td is the time delay and K (> 0) is a gain constant. Taking the Fourier transform of both side of
equation (39), we get
j td
Y( ) = Ke X( ) ... (40)
Thus, we see that for distortionless transmission the system must have
j H( ) j td
H( ) = H ( ) e Ke ... (41)
Thus, H( ) = K ... (42)

H( ) = –j td ... (43)
Remember: That is for distortionless transmission , the magnitude response of system must be constant over the
entire frequency range, and the phase response of system must be linear with the frequency
These two conditions of distortionless system are shown in figure 5.4(a) and (b) respectively.
H( ) H( )
0
Slope = –td
0
(a) (b)
Fig 5.4 . Frequency response for distortionless transmission through an
LTI system (a) Magnitude Response (b) Phase Response
Amplitude Distortion and Phase Distortion: When the amplitude spectrum H( ) of the system
is not constant within the frequency band of interest, the frequency components of the input signal are transmitted
with a different amount of gain or attenuation. This effect is called amplitude distortion. When the phase
spectrum H( ) of the system is not linear with the frequency, the output signal has a different waveform than
the input signal because of different delays in passing through the system for different frequency components of
the input signal. This form of distortion is called phase distortion.
5.8.3 System Output for Complex Exponential

Similar to that of Laplace transform (assuming h(t) is real then H( ) will be conjugate symmetric)
LTI System
ej ot
H( o) e
j ot
H( )
LTI System
cos( ot ) H( o) cos( ot )
H( )
LTI System
sin( ot) H( o) sin( ot)
H( )

If input to a LTI system is a periodic signal x(t)
That is x(t) Cke j o kt
k
Then using the above properties we can say that
y(t) Ck H ( ok )ej o kt
k Fourier Series
Coefficient of y (t )
Very important illustration is that if input to LTI system is periodic signal with fundamental frequency
o then output will also be periodic with same fundamental frequency and Fourier series coefficient of output
will be CkH( ok).
REMEMBER e j ot is eigen function for LTI system and H( o) is called it’s eigen value
5.8.3 Phase Delay and Group Delay

The two important parameters that characterize the form of the response y(t) of an LTI system excited
by an input signal x(t) composed of a weighted sum of sinusoidal signals are phase delay and group delay. These
two parameters are associated with the frequency response H( ) of the system. For a linear phase system
(whose phase varies linearly with frequency); both the phase delay and the group delay are constant.
Phase Delay: The time delay experienced by a single-frequency signal (i.e., a sinusoidal signal) when
the signal passes through a system is referred to as the system phase delay. Now, let us assume that the input
signal is the single-frequency signal
x(t ) = Acos( 0t + )
A A
x(t) = e j( 0t )
e j( 0t )
2 2
and the system frequency response is
j H( )
H( ) = H( ) e
Then, the system output signal is
A A
y(t) = H( 0 ) e j( 0t H( 0 ))
H( 0 )e j( 0t H( 0 ))
2 2
For real h(t) we have, H( 0) = H(– 0) and H( 0) = H(– 0), we have
A A
y(t) = H( 0 ) e j( 0t H( 0 ))
H( 0 )e j( 0t H( 0 ))
2 2
= A H( 0) cos( 0t + H( 0) + )
H( )
= A H( 0 ) cos 0 t 0

y(t) = A H( 0) cos( 0t – p ( 0)) + )
H( 0 )
Where, p ( 0) = ... (44)
0
is the time delay experienced by the single-frequency signal with frequency 0 when it passes through
the system. Thus, p( 0) is the system phase delay for a signal with frequency 0. A negative system phase
response at positive frequencies indicates that a signal is delayed in time when it passes through the system,
whereas a positive system phase response at positive frequencies indicates that a signal is advanced in time
when it passes through the system.
Group Delay: When the input signal contains many sinusoidal components with different frequencies
that are not harmonically related, each component will go through different phase delays when processed by a
frequency-selective LTI system, and the signal delay is determined using a different parameter called the group
delay:
d H( )
g( ) = ... (45)
d
We now derive equation (45) that defines the group delay by using a single-frequency modulating and
carrier signals with zero phase for simplicity.
The assumed input signal (double side band-suppressed carrier, i.e, DSB-SC modulated signal) is
s(t) = Acos( mt)cos( ct) ... (46)
where c is the carrier frequency and m is the frequency of the modulating signal.
We use the cosine-product trigonometric identity to rewrite the input signal as
A A
s(t) = cos ( c m )t cos ( c m )t
2 2
A A
= cos( 1t ) cos( 2 t)
2 2
where, 1 = c + m ... (47)
and 2 = c + m ... (48)
Now let the signal s(t) passed through the system with frequency response H( ). We assume that the
system magnitude response and phase response are H( ) = 1 and H( ) = ( ), respectively. The system
output signal is
A A
y(t) = H( 1 ) cos[ 1t H( 1 )] H( 2 ) cos[ 2 t H( 2 )]
2 2
A A
y(t) = cos[ 1t ( 1 )] cos[ 2 t ( 2 )]
2 2
where ( 1) and ( 2) are the phase shifts produced by the system at frequencies 1 and 2, respectively.
Equivalently, we may express y(t) as
A ( ) ( ) ( ) ( )
y(t) = cos c t mt 1 2 1 2
2 2 2
A ( ) ( ) ( ) ( )
cos c t m t 1 2 1 2
2 2 2
=
A ( ) ( ) ( ) ( )
= cos t
c
1 2
t
m
1 2
2 2 2
A ( ) ( ) ( ) ( )
cos t
c
1 2
m t 1 2
2 2 2
( ) ( ) ( ) ( )
y(t)= A cos t
c
1 2
cos m t 1 2
2 2
( ) ( ) ( ) ( )
= A cos c t 1 2
cos m t 1 2
2 c 2 m
y(t)=Acos( c[t – p( c)]cos( m[t – g( c)]) ... (49)

Comparing the output signal y(t) in equation (49) with the input signal s(t) in equation (46), we make
the following two observations:
( ) ( )
1.The carrier components at frequency c in y(t) lags its counterpart in s(t) by 1 2
,
2
which represents a time delay
( 1 ) ( 2 ) ( 1 ) ( 2 )
p( c ) = ... (50)
2 c 1 2
2.The modulating signal component at frequency m in y(t) lags its counterpart in s(t) by
( ) ( )
1 2
, which represents a time delay
2
( ) ( ) ( ) ( )
g( c) = ... (51)
1 2 1 2
2 m 1 2
Suppose that the modulating signal frequency m is small compared with the carrier frequency c,
which implies that the frequencies 1 and 2 are close together, with c between them. Such a modulated signal
is known as narrowband signal. Then we may approximate the phase response ( ) in the vicinity of = c by
the two-term Taylor expansion
d ( )
( ) = ( c ) ( c ) ... (52)
d c
Evaluating ( 1) and ( 2) using equation (52), we get
d ( )
( 1) = ( c ) ( 1 c ) ... (53)
d c
d ( )
and ( 2) = ( c ) ( 2 c ) ... (54)
d c

( c )
Thus, we get p( c ) = ... (55)
c
Similarly, using the above equations we get
d( ) d H( )
p( c )= ... (56)
d c
d c
The time delay g( ) is called the group delay or envelope delay. Thus, the group delay at each frequency
equals the negative of the slope of the phase at that frequency. If g( ) is constant, all the components are
delayed by the same interval.
Example - 5.22
Consider the following frequency response for a causal and stable LTI system:
1 j
H( )
1 j
(a) Show that H( ) = A, and determine the value of A.
(b) Determine which of the following statements is true about g( ), the group delay.
(i) g( ) = 0, for > 0
(ii) g( ) > 0, for > 0
(iii) g( ) < 0, for > 0
Solution 5.22
1 j
(a) Given that, H( ) =
1 j
2
1
H( ) = 2
1
1
Therefore, A = 1,
(b) The phase response of the system is
H( ) = tan–1 (– ) – tan–1( )
= –2 tan–1( )
Therefore, the group delay is
d ( H( ) d 2
g( ) = 2 tan 1( )
d d 1 2
Since g( ) > 0 for > 0, statement (ii) is true
Example - 5.23
1
An LTI system has impulse response h(t) . Find out the output for input x(t) = cos( ot + ). comment
t
on application of system.

Solution 5.23 H( )
1
Since h(t) the Fourier transform of h(t) is
t
H( ) = –j sgn( )
When input is x(t) then Fourier transform of output will be
–j
Y( ) = X( ) H( )
The system simply work as a –90° phase shifter because positive frequencies of X( ) get multiplied by
–j and negative frequencies of X( ) get multiplied by j.
Thus when input is cos( ot + ) output will be cos( ot + – 90°)
1
REMEMBER The LTI system whose impulse response is h(t) t
is called Hilbert transform, it provide –90°
phase shift to input signal.
Example - 5.24
A causal LTI system has the frequency response H( ) shown in figure. For each of the input signals given
below, determine the filtered output signal y(t).
(a) x(t) = ejt
(b) x(t) = sin( 0t) u(t)
1
(c) X( ) =
j (6 j )
1
(d) X( ) =
2 j
H( )
2j
–1 0
–2j
Fig. Frequency response

Solution 5.24
From figure the frequency response of the filter can be expressed as
H( ) = –2j
We know that the filtered output signal is given by
Y( ) = H( ) X( )
Y( ) = –2j X( )
Taking the inverse Fourier transform gives
dx(t )
y(t) = 2
dt
(a) Given that x(t) = ejt

dx(t ) d jt
Therefore, y(t ) = 2 2 e 2 je jt
dt dt
(b) Given that x(t ) = sin( 0t) u(t)
dx (t )
y(t) = 2 =–2[ 0 cos( 0t) u(t)+ sin( 0t) (t)]
dt
= –2 0 cos( 0t) u(t)
2j 2
(c) We know that Y( ) = H( ) X( ) =
j (6 j ) 6 j
Taking the inverse Fourier transform, we obtain
y(t) = –2e–6t u(t)
1
(d) Given that X( ) =
2 j
Taking the inverse Fourier transform, we obtain,
x(t ) = e–2t u(t)
The filtered output signal is given by
dx(t ) d 2t
y(t) = 2 e u(t ) 4e 2t u(t ) 2e 2t (t )
dt dt
= 4e–2t u(t) – 2 (t)
Example - 5.25
Consider a continuous -time low-pass filter whose impulse response h(t) is known to be real and whose
frequency response magnitude is given as
1, 200
H( ) =
0, otherwise
Determine and sketch the real-valued impulse response h(t) for this filter when the corresponding group
delay function is specified as
(a) g( ) = 5,
(b) g( ) = 5/2,
Solution 5.25
We know that,
d H( )
g( ) =
d
If g( ) = , where is a constant, then,
=
d H( )
d
H( ) = – +
Where is another constant.

We know that if h(t) is real, then H( ) has to be an odd function. Therefore, the value of = 0. Let us
define H0( ) = H( ) . then
1, 200
H( ) =
0, otherwise
Taking the inverse Fourier transform, we have
200
1 j t 1 sin(200 t )
h0(t ) = H0 ( )e d ej td
2 2 200 t
(i) Given that g( ) = = 5.
Hence, H( ) = –5 ,
Then H( ) = H( ) ej H( ) = H0( ) e–5j
Taking the inverse Fourier transform, we get
sin(200 (t 5)
h(t) = h0 t 5
(t 5)
(ii) Given that g( ) = = 5/2.
Hence, H( ) = (5/2)
Then H( ) = H( ) ej H( ) = H0( ) e (5/2)j
5 sin(200 (t 5 / 2)
h(t) = h0 t
2 (t 5 / 2)
Example - 5.26
Consider on LTI system with impulse response

sin(400 t )
h( t )
t
If input to the system x(t), is a periodic signal shown in Fig then find the bandwidth of output
x(t)
5
........ ........
t (ms)
–10 0 10 20 30
Solution 5.26
We can see that signal x(t) is periodic with period To = 10 ms
2
o 3
200
10 10
Since x(t) can be made half wave symmetric by shifting it on amplitude axis by –2.5, if Ck are exponential

Fourier series coefficient of x(t) then Co 0 and Ck will be zero for all even values of k. Thus if we write X( )
then
X( ) 2 Ck ( k o )
k
The plot of X( ) is
X( )
2 C0
2 C–1
2 C –3 2 C1
2 C3
........ ........
–600 –400 –200 0 200 400 600
1, | | 400
and H( )
0, | | 0
Since Y( ) = X( ) H( )
Thus Y( ) will be nonzero only for = 0, ± 200
thus band width of y(t) is 200 rad/sec
Example - 5.27
Consider a periodic signal x(t) with time period To = 100 m sec and Ck are it’s Fourier series coefficients.
If x(t) is applied as input to an ideal low pass filter with cut off ( c) equal to 500 rad/sec and output is equal
to input. For the given condition maximum value of k for which Ck is non zero is _______.
Solution 5.27
Given that x(t) is periodic with period To = 100 m sec
2
o 20
1
The Fourier transform of x(t) will be
X( ) 2 Ck ( 20 k )
k
Thus X( ) will be non zero only at integer multiple of 20 . Since x(t) is applied as input to a low pass filter
with c = 500 rad/sec
thus Y( ) = H( ) X( )
for output equal to input we need
Y( ) = X( )
Thus X( ) should not have any frequency component above 500 rad/sec that is maximum value of k for
which Ck should be non zero is given by
20 k < 500
500
k<
20
< 7.96

k < 7.0
Thus maximum value of k is 7.
5.9 Ideal and Practical filters

Filtering is a process by which the essential and useful part of a signal is separated from the extraneous
and undesirable components. The idea of filtering using LTI system is based on the convolution property of the
Fourier transform, which states that for LTI systems, the Fourier transform of the output Y( ) is the product of
the Fourier transform of the input X( ) and the frequency response H( ), i.e. Y( ) = X( )H( ). In a sense,
H( ) acts as a weighting function or spectral-shaping function to the different frequency components in the
input signal. Ideal filters allow distortionless transmission of a certain band of frequencies and completely
suppress the remaining frequencies. The passband of a filter is the band of frequencies that are passed by the
system, while the stopband refers to the range of frequencies that are attenuated by the system. In the ideal case
H( ) = 1 in a passband, while H( ) = 0 in a stopband. Filters are usually classified according to their
frequency - domain characteristics as low-pass, high-pass, bandpass, bandstop or band elimination, and all-pass
filters. The following lists definitions for each of these:
1. Low pass filters are those characterized by a passband that extends from = 0 to = c, where c is
called the cutoff frequency of the filter [Fig. 5.5(a)]. A low pass filter attenuates high-frequency
components of the input and passes the lower frequency components.
2. High pass filters are those characterized by a stopband that extends from = 0 to = c, and a
passband that extend from = c to infinity [Fig. 5.5(b)]. A high pass filter attenuates high-frequency
components of the input and passes the lower frequency components.
H( ) H( )
Passband Stopband Stopband Passband
(a) LPF (b) HPF
H( ) H( )
Stopband Passband Stopband Passband Stopband Stopband
1 2 1 2
(c) BPF (d) BSF
Fig 5.5. Most common classes of filters:

3. Bandpass filters are characterized by a passband that extends from = 1 to = 2 and all other
frequencies are stopped [Fig. 5.5(c)]. A bandpass filter passes signals within a certain frequency band
attenuates signals outside that band.
4. Bandstop filters stop frequencies extending from = 1 to = 2 and pass all other frequencies [Fig.
5.5(d)]. A bandstop filter attenuates signals within a certain frequencies band and passes signals outside
that band.
5. All pass filter is characterized by a magnitude response that is constant for all frequencies. The
characteristics of an allpass filter are completely determined by its phase-shift characteristics. An

allpass filter has nonconstant group delay and different frequencies in the input are delayed by different
amounts.
If in a question a transfer function of filter is given ( H ( s ) or H ( )) and we need to find the
type of filter then we should follow these steps to find result:-
Step 1: If H( ) is given then it’s fine and if H(s) is given then find H( ) by replacing s by j in H(s).
Step 2: Find the values
A lim| H( )|
0
and B lim | H( )|
Step 3: If • A > B and B is close to zero then it means that filter alternate the high frequency component
of input and amplify low frequency component of input. Thus it is low pass filter.
• B > A and A is close to zero then it means that filter alternate the low frequency component of input
and amplify high frequency component of input. Thus it is high pass filter.
• A, B both are small close to zero then it means filter alternate low and high frequency component of
input but will pass signal of some specific frequency. Thus it is a band pass filter.
• A and B both are greater than zero then it might be an all pass filter or band stop filter. Now to find that
given filter in all pass filter or Band stop filter then the Numerator of H( ) should go to zero at a particular
frequency for band stop filter and for all pass filter the property is that it’s magnitude response remain
constant for all frequencies.
REMEMBER For an all pass filter

• The magnitude plot (|H( |) remain constant for all
• The pole-zero of transfer function (H( ) or H(s)) are at mirror image with respect to imaginary
axis, for example
Img
s–a
Real H(s) = s + a
Img
jb
s2 – 2as + (a 2 – b2)
Real H(s) =
–a a s2 + 2as + (a2 – b2)
–jb
• Thus order of numberator and denominator of transfer function is same, number of poles and
zeros is same.
The above condition is a must to satisfy that (H( )) is a constant

Example - 5.26
Find the type of filter if the transfer is

s 10
(a) H(s) 2
s 5s 15
s
(b) H(s)
s 10
s2
(c) H(s)
s2 10s 100
s 10
(d) H(s)
s 10
Solution 5.26
j 10
(a) H( ) 2
5j 15
10
|H( )| =0
15
|H( )| = = 0
So we can see that high frequency component of input will be completely alternated and low frequency pass
with less attenuation. It will be low pass filter
j
(b) H( )
j 10
|H( )| =0 =0
|H( )| = =1
Clearly it is high pass filter
2
(c) H( ) 2
10 j 100
|H( )| =0 =0
|H( )| = =1
Clearly it is high pass filter
j 10
(d) H( )
j 10
|H( )| =0 =1
and |H( )| = =1
the magnitude of H( ) is
2
100
2
=1
100
Thus it is an all pass filter as magnitude response is constant always.

5.10 Sampling
Let x(t) be any continuous signal periodically sampled at equal sample time period Ts to get discrete time
signal represented by x[n]
x(t) x[n]
Sampling
Fig. 5.6 Sampling
i.e., x[n] = x(nTs) ... (57)

Where, n = 0, ± 1, ± 2 ...
Ts = Sampling time period
In figure 5.7, the process of converting x[n] from x(t) is shown,
x(t) x [n]
Xa(Ts)
Ts
x(t) x[t] = xa(nT s)
t –2Ts –Ts 0 Ts 2Ts n
0
Fig. 5.7 Process of sampling Fig. 5.8 Sampling operation
Sampling operation is illustrated in Fig 5.6 and 5.7 may also be conveniently represented as simple switch
which is opened and closed at equal time intervals Ts (fig. 5.8). There are many ways of sampling, here we
will discuss only ideal sampling.
5.10.1 Impulse train sampling

Impulse train sampling is simply the switching operation shown in figure 5.8. Sampling can be represented
by simply multiplying the signal x(t) with impulse train.
x(t) x(t)P(t)
P(t)
Fig. 5.9 Mathematical representation of sampling.
Here p(t) = (t kTs )

k
1
That is impulse train with period Ts (Sampling period) and frequency s (fs = 1/Ts) (sampling
2 Ts
frequency). The Fourier transform of x(t).p(t) will be
1
x(t).p(t ) (X( )*P( ))
2
Where X( ) is Fourier transform of x(t) and P( ) is Fourier transform of p(t)
2
P( ) = ( k s) ... (58)
Ts k

1 2 1
x(t ) p(t ) X( )* ( k s) = X( k s)
Sample x( t ) 2 Ts k Ts k
fourier transform of sampled signal
X( )
A xP( )
A/Ts
........ ........
– B B – s – B B s
(a) (b)
Fig. 5.10 (a) Fourier transform of x(t) (b) Fourier transform of sampled x(t)
Study Note
1. We can see that frequency axis become periodic with period s after sampling
2.Also s should be greater than 2 B then only the X( ) can be recovered back from Xp( ) otherwise over lapping
or aliasing occur. So Nyquist sampling rate is 2 B and for correct results
s >2 B
Impulse train sampling is also called ideal sampling. To get back the signal x(t) from sampled signal
(that is we need X( ) from Xp( ) then we need an ideal low pass filter with pass band gain Ts and c = s
Ts , | | s /2
Thus H( ) = ... (59)
0, | | s /2
We can see that if s >2 , then

X( ) = Xp( ) . H( ) ... (60)
x(t ) = xp(t) * h(t) ... (61)
Where xp(t) is sampled x(t) and h(t) is impulse response of system with transfer function H( ).
From our learning
Since H( ) = Ts rect ( / s)
st
Ts sin
2
We get h( t ) = ... (62)
t
H( )
h(t)
1
T
t
s 0 s –Ts 0 Ts 3Ts
– –3Ts
2 2
(a) (b)
Fig. 5.11 (a) Graphical representation of H ( ) (b) plot of h(t)

Remember: Since h(t) exist for – <t< thus practically it is not possible.
Example - 5.27
If the continuous time signal xa(t) = cos(1250 t) is sampled at sampling frequency fs = 10 Hz, then find
the discrete time sequence x(n).
Solution 5.27
1250 n
Given, x(t) = x( nTs ) cos
fs
t
Where fs = ;
Ts fs = 10 Hz (Given)
1250 t
x(n) = cos cos[125 n]
10
5.11 Energy spectral density

Parseval’s theorem relates the total signal energy in a signal x(t) to its Fourier transform through
2 1 2 2
Ex = x(t ) dt X( ) d X( ) df
2
Parseval’s theorem states that the total energy Ex may be determined either by computing the energy
per unit time ( x(t) 2) and intergrating over all time or by computing the energy per unit frequency ( X( ) 2)
and integrating over all frequencies. For this reason, X( ) 2 represents energy per unit bandwidth and is often
referred to as the energy spectral density(per unit bandwidth in hertz) or energy density spectrum of the signal
x(t) and is denoted by x( ). Hence,
) = X( ) 2
x( ... (65)
The units of ESD depend on the units of the underlying signal x(t). For example, if the signal unit is
volts (V), its Fourier transform has units of V/Hz and its ESD has units of (V/Hz)2 or (Vs)2.
5.11.1 Relationship Between Input and Output Energy Spectral Densities of an LTI System
Consider an LTI system with frequency response H( ), input x(t), and output y(t). If x(t) and y(t) are
energy signals, then their energy spectral densities are x( = H( ) 2 and , y( = Y( ) 2 respectively.
Since, we know that
Y( ) = H( )Y( )
it follows that Y( ) 2 = H( )X( ) 2
Y( ) 2 = H( ) 2 X( ) 2
) = H( ) 2
y( x( ) ... (66)

5.8.2 Relation of ESD to Autocorrelation

The autocorrelation function Rxx( ) of an energy signal is defined as
Rxx( ) = x(t )x(t )dt
Rxx( ) = x(t) * x(– )

Taking the Fourier transform of the above equation gives
ƒ [Rxx( )] = X( )X(– ) = X( )X*( )
ƒ [Rxx( )] = X( ) 2
ƒ [Rxx( )] = x( )
Rxx( )] x( ) ... (67)
Thus, the autocorrelation function Rxx( ) and ESD makes a Fourier transform pair.
5.13 Relationship Between Fourier Transform and Fourier Series

We have seen in starting of the chapter that we can assume a non-periodic signal as periodic with period
T due to which the 0 0 and the discrete plot of exponential fourier series coefficient become continuous
and we get X( ) a continuous plot. Let us consider a square wave periodic with period To and a square pulse
x1(t) x2(t)
A A
........ ........
t
–T0 –TP/2 0 TP/2 T0 –TP/2 0 TP/2
(a) (b)
Fig. 5.21 (a) Periodic square wave (b) square pulse

The fourier series coefficients of x1(t) are
A o Tp k
Ck sin
k 2
2
where o
To
The fourier transform of x2(t) is
sin Tp /2
X 2( ) 2A
We can see that the form of fourier series coefficient and fourier transform is same and a closer analysis
will give:
Fourier series coefficients of periodic signal with period To are samples of fourier transform of same
signal with period tending to infinity at = integer multiple of o and with a scaling factor
X( )
Ck ... (68)
To k o

Remember: Fourier series coefficients are samples of fourier transform at =k o and scaling factor of To
5.14 Fourier transform in f – Domain

We have analyzed the fourier transform in domain but sometimes we have to find fourier transform in
f – Domain.
That is
j 2 ft
X( f ) x(t )e dt ... (69)
and x(t ) X ( f )e j 2 f df ... (70)
The properties remain same in f domain as that of -doamain, the only difference is of factor of 2 .
Table 5.2 Properties of fourier transform ( f domain)
Pr op er ty Signal Fourier transform
x(t ) x(f)
x1 ( t ) x1 ( f )
x2 ( t ) x2 ( f )
Linearity a1x1(t) + a2x2(t) a1x1(f) + a2X2(f)
j 2 ƒto
Time shifting x(t – to) X( f) e
Frequency shifting e j2 fot x(t) X(f – fo)
1
Time scaling x(at) X( f / a )
|a|
Time reversal x(–t) x(–f)
Duality X(t) x(–f)
d
Time differentiation x(t ) j2 fX ( f )
dt
t
1
Integration x ( )d X( o) ( f ) X( f )
j2 f
dX ( f )
Frequency differentiation (–j2 t)x(t)
df
Convolution x1(t) * x2(t) X1( f ).X2( f )
Multiplication x1( t).x2(t ) X1(f ) * X2( f )
Conjugation x*(t ) X*( f )
Parsevals Theorem | x(t )|2 dt | X ( f )|2 df
Example - 5.28
Given that x(t) has the Fourier transform X( ), express the Fourier transforms of the signal listed below in
terms of X( ).
(a) x1(t) = x(1 – t) + x(–1 – t)
(b) x2(t) = x(3t – 6)

d2
(c) x2(t) = x(t 1)
dt 2
Solution 5.28
Given that, x(t ) X( )
(a) Using the time shifting property, we have
x(t + 1) X( )ej
and x(t – 1) X( )e–j
Now, using the time-reversal property, we have
x(–t + 1) X(– )e–j
and x(–t – 1) X(– )ej
Therefore, x1(t) X 1( )
x(1 – t) + x(–1 – t) X(– )e–j + X(– )ej
x(1 – t) + x(–1 – t) 2X(– )cos
ƒ (x1(t) = 2X(– )cos
(b) Using the time-shifting property, we have
x(t – 6) X ( ) –j 6
Now, using the time-scalling property, we get
1 j 6 /3
x2(t ) = x(3t – 6) X2( ) = X e
3 3
1 j2
X 2( ) = X e
3 3
(c) Using the differentiation in time-domain property, we get
dx(t )
j X( )
dt
Applying this property again, we get
d2 x(t )
(j )2X( )
dt 2
Now applying the time-shifting property, we get
d2 x(t 1) j
x3(t) = X2 ( ) ( j )2 X( )e
dt 2
X 3( ) = – 2 X( ) e–j
Example - 5.29
For each of the following Fourier transform, use Fourier transform properties to determine whether the
corresponding time-domain signal is (i) real, imaginary, or neither and (ii) even, odd and neither. Do this
without evaluating the inverse of any of the given transforms.
(a) X1( ) = u( ) – u( – 2)
(b) X2( ) = cos(2 ) sin( /2)
(c) X3( ) = A( )ejB( ), where A( ) = sin(2 )/ and B( ) = + /2
k
(d) X4( ) = k (1 / 2) ( k / 4)

Solution 5.29
(a) We know that the Fourier transform of a real, signal is conjugate symmetric. Since X1( ) is not conjugate
symmetric, the corresponding signal x1(t) is not real. Since X1( ) is neither even nor odd, the corresponding
signal x1(t) is neither even nor odd.
(b) We know that the Fourier transform of a real and odd signal is purely imaginary and odd. Therefore, we
may conclude that the Fourier transform of a purely imaginary and odd signal is real and odd. The given
transform X2( ) is real and odd, we may therefor conclude that the corresponding signal x1(t) is purely
imaginary and odd.
(c) We know that if x(t) is real, then its X( ) = X(– ) and X( ) = – X( ). Here
sin(2 )
X( ) = A( ) X( )
and X( ) = B( ) = 2 = – X(– )
So, we may conclude that the signal x(t) is real. Now, consider the signal x3(t) with fourier transform
X3( )= X( )ej( /2) = jX( ). We may conclude that x3(t) is imaginary. Since, the Fourier transform X3( ) is
neither purely imaginary nor purely real, the signal x3(t) is neither even nor odd.
(d) Since X4( ) is real and even, the corresponding signal x3(t) is real and even.
Example - 5.30
(a) Use the Fourier transform properties to determine the Fourier transform of the following signal:
2
sin(t )
x (t ) t tg 2 (t )
t
Where g(t) = sin(t)/ t.
(b) Use parseval’s relation and the result of the previous part to determine the numerical value of
4
sin(t )
A t 2
dt
t
Solution 5.30
(a) We know that
t
Asinc 2 A rect( )
2
sin(t / 2)
2 rect( )
t/2
Using the time-scalling property, we have
sin(t )
rect
t 2
sin(t )
g(t) = G( ) rect
t 2

G( ) is shown in figre(a), which is a rectangle property, now
1
g 2( t ) = g(t )g(t ) [G( )* G( )]
2
2
sin(t ) 1
rect * rect
t 2 2 2
g1( t ) G1 ( )
We know that the convolution of a rectangle function with itself is a triangle [G2( ) is shown in figure (b)].
Therefore, we have
2
sin(t ) 1
t 2
1
1 , 2 0
2
1 1
where G1( ) = 1 , 0 2
2 2
0, Otherwise
Using the differentiation in frequency domain property, we have
dG1( )
t g 1( t ) j
d
j
, 2 0
2
2
sin(t ) j
t X( ) , 0 2
t 2
x( t ) 0, Otherwise
X( ) is shown in figure (c).

(b) Using Parseval’s theorem, we have
2 1 2
x(t ) dt = X( ) d
2
2 2
0 2
sin(t ) 1 1 1
t dt = d d
t 2 2
4 2 0
4 2
2 2
sin(t ) 1
t dt =
t 2 3

G( ) G1( )
1/
1
–1 0 1 –2 –1 0 1 2
(a) (b)
X( )
j/2
1 2
–2 –1 0
–j/2
(c)
Example - 5.31
Given the relationship

y(t) = x(t)*h(t)
and g (t ) = x(3t)*h(3t)
and given that x(t) has Fourier transform X( ) and h(t) has Fourier transform H( ), use Fourier transform
properties to show that g(t) has the form
g (t ) = Ay(Bt)
Determine the values of A and B.
Solution 5.31
Given that, y(t) = x(t)*h(t)
Taking Fourier transform, we get
Y( ) = X( )H( )
Using the time-scalling property, we have
1
x(3t) X
3 3
1
and h(3t) H
3 3
Taking the Fourier transform of g(t) = x(3t) * h(3t), we get,
G( ) = ƒ [x(3t)] ƒ [h(3t)]]
1 1
= X H Y
9 3 3 9 3

1
g (t) = y(3t ) Ay( Bt )
3
Therefore, A = 1/3 and B = 3.
Example - 5.32
Let x(t) be a signal whose Fourier transform is

X( ) = ( ) + ( – ) + ( – 5)
and let h(t) = u(t) – u(t – 2)
(a) Is x(t) periodic?
(b) Is y(t) = x(t) * h(t) periodic?
(c) Can the convolution of two aperiodic signals be periodic?
Solution 5.32
Given that, X( ) = ( ) + ( – ) + ( – 5)
Taking the inverse Fourier transform of X( ), we have,
1 1 j t 1 j 5t
x(t) = e e
2 2 2
The signal x(t) is therefore a constant summed with two complex exponential whose fundamental frequencies
are
1 = rad/s
and 2 = 5 rad/s
2 2
T1 = 2
1
2 2
and T2 =
2 5
The ratio
T1 5
is not a rational number and so x(t) is not periodic.
T2
(b) Given that, h(t) = u(t) – u(t – 2)
Taking the Fourier transform, we have
H( ) = U( ) – U( ) e–j2
= 2 jU( )e j ej e j
2j
j 1
= 2je–j sin( ) U( ) = 2 je sin( ) ( )
j
2 sin( ) j
H( ) = e
Consider the signal y(t) = x(t) * h(t). From the convolution property, we have
Y( ) = X( )H( ) = [ ( ) + ( – ) + ( – 5) H( )]
= H(0) ( ) + H( ) ( – ) + H(5) ( – 5)
Since H( ) = 0 and H(0) = 1, we get
Y( ) = ( ) + H(5) ( – 5)

1 1
y(t) = H(5)e j 5t
2 2
Therefore, y(t) is a constant summed with a complex exponential whose fundamental frequency is = 5
rad/s and fundamental period t = 2 /5. We know that adding a constant to a complex exponential does not
affect its periodic. Therefor, y(t) = x(t) * h(t) is periodic.
(c) Yes, it is obvious from parts (a) and (b) that the convolution of two aperiodic signals can be periodic.
Example - 5.33
Let x(t), be a signal with Fourier transform X( ). Suppose we are given the following facts:
1. x(t) is real.
2. x(t) = 0 for t 0
1
3. R{X ( )}e j t d t et
2
Determine the closed-form expression for x(t).
Solution 5.33
From fact no. 3, we have
1 t
R{X ( )}e j t d = te
2
t
ƒ 1[RX ( )] = t e
We know that {x(t)} = xe(t) R {X( )}, therefore, we have

t
xe(t) = t e
x(t ) x( t ) t
= te
2
Given that x(t) = 0 for t < 0. This implies that x(–t) = 0 for t > 0. So, we may conclude that
t
x(t) = 2 t e , t 0
Therefore, x(t) = 2t e–tu(t)
Example - 5.34
The output y(t) of a causal LTI system is related to the input x(t) by the equation
dy(t )
10y(t ) = x( )z(t )d x(t )
dt
Where z(t) = e–tu(t) + 3 (t)
(a) Find the frequency response H( ) of this system.
(b) Determine the impulse response of the system.

Solution 5.34
(a) Consider the equation
dy(t ) x( )z(t )d x(t )

10y(t ) =
dt
dy(t )
10y(t ) = [x(t)*z(t)] – x(t)
dt
Taking the Fourier transform of the above equation, we have
j Y( ) + 10Y( ) = X( )Z( ) – X( )
Y( )[10 + j ] = X( )[Z( ) – 1]
Given that z(t) = e–tu(t) + 3 (t),
1
This implies that Z( ) 3. Substituging Z( ) in the above equation, we get
1 j
1
Y( )[10 + j ] = X ( ) 3 1
1 j
Y( ) 3 2j
= H( )
X( ) (1 j )(10 j )
(b) Finding the partial fraction expansion of H( ), we obtain
1 1 17 1
H( ) =
91 j 9 10 j
1 t 17 10 t
h( t ) = e u(t ) e u(t )
9 9
Example - 5.35
A signal x(t) = 2e–tu(t) is passed through an RC LPF [shown in figure (a)] having a cutoff frequency c =
1 rad/s. Find
(a) the transfer function of the LPF.
(b) the spectrum of the input signal x(t).
(c) ESD at the input of the LPF.
(d) spectrum of the output signal
(e) ESD at the output of the LPF.
(f) energy contained in the input and output signals. Are these two energies equal?
Solution 5.35
(a) The frequency response of the RC LPF is given
Y( ) 1/ j C 1
H( ) =
X( ) 1 1 j RC
R
j C

1
=
1 j
c
where c = 1/RC = 1 rad/s. The magnitude and phase response are [as shown in figure (b) and (c)] given
by
H( ) = 1 2
H( ) = –tan–1( )
H( )
R 1
+ +
1/ 2
x(t) C y(t)
– +
(a) –1 (b) 1
H( )
/2
/4
1
–1 0
– /4
– /2
(c)
(b) Consider the given signal, x(t ) = 2e–tu(t)
2
X( ) =
1 j
The magnitude and phase spectrum are [as shown in figures (d) and (e)] given by
2
X( ) =
2
1
X( ) = tan–1( )
(c) The ESD at the input of the LPF is given
2
2 2
x( ) = X( ) =
1 j
4
= 2
1

X( )
X( )
/2
2
/4
2 1
–1 0
– /4
–1 0 1 – /2
(d) (e)
(d) We know that Y( ) = X( )H( )
2 1 2
=
1 j 1 j (1 j )2
(e) The ESD at the output of the LPF is given by
2
2
2 4
y( ) = Y( ) =
(1 j ) 2
(1 2 2
)
The energy contained in the input signal is given by
1 1 4
Ex = x ( )d 2
d
2 2 1
2 4 4
= 2
d tan 1( )
2 0
1 0
4
= 0 2
2
The energy contained in the output signal is given by
1 1 4
Ey = y ( )d 2 2
d
2 2 (1 )
A change of variables is performed by letting = tan( ), which also yields d = sec2( ) d , – /2
as – and /2 as . Therefore,
/2 /2
4 sec 2 ( ) 2
Ey = 4
d cos 2 ( )d
2 /2 sec ( ) /2
/2 /2
2 1 1 sin(2 )
= [1 cos 2( )]d
/2
2 2 /2
1
= 0 1
2 2

1. x(t) is real and nonnegative.
2. 1 {(1 + j )X(j )} = Ae–2t u(t), where A is

ƒ
1
independent of t.
Subjective Practice Problems 2
3. X( j ) d 2 .
1
. Use the Fourier transform synthesis equation to Determine a closed- form expression for x(t).
determine the inverse Fourier transforms of: 6
. Consider the signal
(a) X1( ) = 2 ( ) + ( – 4 ) + ( + 4 )
2, 0 2 sin k
4
(b) X2( ) = 2, 2 0 x(t) = t k
4
k
0, 2 4
(a) Determine g(t) such that
2
. Use the Fourier transform synthesis equation to
determine the inverse Fouriers transform of X(j ) sin t
x(t) = g(t ).
where, t
X( j ) = 2{u( + 3} – u( – 3)}, (b) Use the multiplication property of the Fourier
transform to argue that X(j ) is periodic. Specific
3
X( j ) = X(j ) over one period.
2
Use your answer to determine the values of t for 7
. Find the impulse response of a system with the
which x(t) = 0. frequency response
3
. Consider the signal (sin2 (3 ))cos
H(j ) = 2
0, t 1
x(t) =
(t 1) / 2, 1 t 1 8
. Consider a causal LTI system with frequency response
Find the fourier transform of x(t), even part of x(t) 1

H(j ) =
and odd part of x(t). j 3
For a particular input x(t) this system is observed to
4
. Consider the Fourier transform pair
produce the output
y(t) = e–3tu(t) – e–4tu(t).
t F 2
e 2
Determine x(t).
1
9
. Determine the continuous-time signal corresponding
(a) Use the appropriate Fourier transform
to the following transforms.
t
properties to find the Fourier transform of te X( )
X(j )
(b) Use the result from part(a), along with the
duality property, to determine the Fourier 1
transform of
4t –1 1
(1 t ) 2 2
–3
5
. Consider the signal x(t) with Fourier transform X(j ).
Suppose we are given following facts:

10. Determine which, if any, of the real signals depicted

t 2 /2
in figure has Fourier transform that satisfy each of x(t) = e
the following conditions:
1. Re{X(j )} = 0
2. Im{X(j )} = 0 t
3. There exists a real such that ej X(j ) is real
(e)
4. X( j )d 0 x(t) = t2e– t
5. X( j )d 0
6. X(j ) is periodic
t
x(t) (f)
1 11. Let X(j ) denote the Fourier transform of the signal

–5 –3 1 2 3 4 5 6 7
t x(t) depicted in figure.
–6 –4 –2 –1
–1 (a) X(j ) can be expressed as A(j )ej (j ), where A(j )
and (j ) are both real-values. Find (j ).
(a)
(b) Find X(j0)
x(t)
2 (c) Find X( j )d
2 sin
t
(d) Evaluate X( j ) ej2 d
1
(b) 2
(e) Evaluate X( j ) d
x ( t)
(f) Sketch the inverse Fourier transform of
Re{X(j )}.
8 x(t)
t
2 3
2
(c) 1
x(t)
t
1 0 1 2 3
1
–2
12. Let x(t) have the Fourier transform X(j ), and the
t
2 let p(t) be periodic with fundamental frequency 0
1 and Fourier series representation.
(d) p(t) = a ne jn 0t
n
Determine an expression for the Fourier transform
of
y(t) = x(t)p(t).

(b) Suppose that X(j ) is as depicted in figure (a).

Sketch the spectrum of y(t) for each of the following (a) x1(t ) cos 6t
choices of p(t):
2
1. p(t) = cos(t/2) k
1
2. p(t) = cos2t – cos t (b) x2 (t ) sin(3kt )
k 0
2
3. p(t) = (t n)
n
sin(4(t 1))
(c) x3 (t )
4. p(t) =
n
( t 2 n) (t 1)
2
5. p(t) = ( t 4 n) sin 2t
n (d) x 4(t )
1 t
6. p(t)= ( t 2 n) (t n)
n
2 n
sin t
7. p(t)= the periodic square wave(amplitude =1) 15. Let g(t ) x(t )cos 2 t
shown in figure (b). t
X (j ) Assuming that x(t) is real and X(j ) = 0 for 1,
show that there exists an LTI system S such that
S
1 x(t ) g(t )
16. Consider an ideal low pass filter with pass band gain
= 1 and cut off frequency as c. If input to filter is
–1 0 1 e–tu(t) and energy at output is 25% as that of input
(a) then value of c is ________ rad/sec.
p (t )
17. Let X( ) = rect( – 1/2). Using the Fourier transform
properties, find the Fourier transform of the following
functions:
........ ........ (a) x(–t) (b) tx(t)
t (c) x(t + 1) (d) x(–2t + 4)
–3 –2 – – 2 3 4 5
6 6
(b)
dx(t )
(e) (t – 1)x(t + 1) (f)
dt
13. Suppose g(t) = x(t) cos t and the Fourier transform
dx(t )
of the g(t) is (g) t (h) x(2t – 1)e–j2t
dt
1, 2 (i) x(t)e–j2t
G( j )
0, Otherwise 18. The signal x(t) = e–atu(t) is input into a system with
(a) Determine x(t). sin(2t ) .
(b) Specify the Fourier transform X1(j ) of a signal impulse response h(t) =
t
x1(t) such that
(a) Find the Fourier transform Y( ) of the output.
2
g(t ) x1(t )cos t . (b) For what value of a does the energy in the output
3 equal one-half the input signal energy.
14. Consider an LTI system S with impulse response 19. What is the total area under the function,
sin(4(t 1)) t 8
h(t )
(t 1) x(t) = 100sinc .
30
Determine the output of S for each of the following
inputs:

Chapter
DISCRETE TIME
FOURIER SERIES 6
6.1 Introduction
In the previous chapters we have analyzed frequency of continuous time signals from now we will start
analyzing the frequency domain of discrete signals. The discrete signals are obtained by sampling the continuous
the signal
Continuous Discrete
Sampling
Time signal Time signal
Figure 6.1 Sampling of continuous time signal

In chapter 5 we have seen that when sampling is done that is x(t) is multiplied by impulse train then the
frequency domain become periodic with period . That is sampled signal will always have periodic frequency
s
domain. Discrete time fourier series and discrete time fourier transform are used for frequency analysis of
perodic and non-perodic discrete signal. In both the transforms the frequency domain is periodic. In we take
fourier transform of x(t) as X( ) and sampled signal xp(t) and it is fourier transfom as Xp( ). If sampling
frequency is s
then Xp( ) will be periodic with period s
.
Xp( )
X( )
A
Sampling
B 0 B s
B
0
B
s
Figure 6.2 Frequency analysis of sampling process.

is the continuous frequency and is discrete frequency and to convert to we use scaling.
s 2 ... (1)
That is s of continuous frequency will become 2 in discrete frequency.
Thus s 2

2
Or TS 2
2
For any .
s
That is TS ... (2)

Thus in continous frequency domain can be changed to dicrete frequency whose value will be TS.
REMEMBER of continuous frequency domain can be change to dicrete frequency domain by simply multiplying
by TS.
= TS
And to convert discrete frequency to continuous frequency can be done by division of TS.
6.2 Discrete Time Fourier Series(DTFS)

Using DTFS any discrete time periodic signal with period N can be represented as a linear combination of
N orthogonal peridic complex exponential signals. The fundmental frquency is wo = 2 /N. The complex
exponential function is
jk on
k(n ) = e ... (3)
2
jk n
= e N
... (4)
From equation (4) we can see that
0(n) = N(n)
That is the function is same for k = 0 and k = N, similarly function will be same for k = 1 and k = N + 1.
Thus we can see that k(n) function will be distinct only when k is between O to N 1. Only N distnict function
can be generated for k(n).
Study Note
A very important difference is seen between discrete and continuous complex exponential function. In case of
jk ot
continuous complex exponential function e we have infinite functions for <k< but discrete complex
jk on
function e will give only N distinct function.
Thus using DTFS we can express any periodic signal x[n] periodic with period N in terms of linear combination
of N complex exponentials
x[n] = Xk k ( n) ... (5)

k N
x(n) = X k e jk on
... (6)
k N
We can see in above equation (5) and (6) that summation is over N, thus DTFS coefficients are finite and
only N terms exist in DTFS of a periodic signal with period N.
Study Note
k(n) = k+N(n) that is complex exponential function is periodic with period N that is function is same
for values k and k+N.
X0 is coefficient of 0(n) and XN is coefficient of N(n) thus XN = X0. similarly Xk is coefficient of k(n) and
Xk+N is coefficient of k+N(n) thus Xk+N = Xk. Thus Xk is also periodic with period N.

Discrete Time Fourier Series 271
The DTFS of dicrete periodic signal of period N generate coefficients Xk which are also discrete and
periodic with period N.
To evaluvate DTFS coefficients the formulae is
j o kn
Xk = x ne ... (7)
N n N
Study Note
There are no dicrichlet condition for DTFS, the only condition is that signal should be periodic. The summation
will always give finite result.
6.2.1 Magnitude and phase spectrum of discrete time periodic signal

The fourier series consists of N components
X0 , X 1 e j , X2 e j 2 o n ,...,XN-1 e j ( N 1) o n
on
The frequencies of these components are 0, o, 2 o,...,(N 1) o where o = 2 /N. The amount of the kth
harmonic is Xk (the fourier coefficient). We can plot this amount Xk as a function of index k or . Such a plot is
called the fourier spectrum of x(n). In general, the fourier coefficients Xk, are complex and be represnted in the
polar form as
Xk = X k e j Xk
The plot of |Xk| versus is called the magnitude spectrum and that of Xk versus is called the angle (or
phase) spectrum. These two plots together are the frequency spectra of x(n). Knowing these spectra, we can
reconstruct or synthesize x(n) according Eq.(6).
Study Note
The results are very similar to the representation of a continuous-time periodic signal by an exponential fourier
series except that, the bandwidth of the continuous-time perodic signal is infinite and consists of an infinite
number of exponential components (harmonics). The spectrum of the discrete-time periodic signal, in contrast, is
bandlimited and has at most N components.
6.2.2 Finding DTFS coefficients of a signal

Generally we can use Eq.(7) to find Xk of any given signal x[n], but if x[n] has sinusoidal or complex
exponentials then we cannot use formulae to find Xk. To find Xk in such cases we will follow these steps
Steps 1. Find the period of the signal, if the signal is non periodic then we cannot find DTFS coefficients.
If it is periodic then move to step 2
Steps 2. Let the signal is periodic with period N. Now express all the sine and cosine signals in terms of
complex exponentials
ej e j
ej e j
sin = cos =
2j 2
jk o n
Steps 3. Now compare each term with Xk e and find value of k and Xk. By comparing each term we
can easily get Xk.

Example 6.1
Determine the fourier series coefficients of the signal x ( n ) and plot its magnitude and phase
spectrum.
2π 2π 4π π
x ( n ) = 1+sin n +3cos n +cos b+
N N N 2
Solution 6.1
The given signal is periodic with period N and frequency o = 2 /N and it can be written as
x(n) = 1 sin on 3cos on cos 2 on

2
ej on e j on
ej on e j on
e j2 on /2
e j 2 on /2
x(n) = 1+ 3
2j 2 2
3 1 3 1 1 j 1
= 1 ej on e j on
e /2
e j2 on e j /2
e j 2 on
2 2j 2 2j 2 2
Comparing the above equation with.

x(n) = X 0 + X 1e j on + X 1e j on
+ X 2e j2 on X -2e j2 o n
We get
X0 = 1
3 1 3 1 10 1 1
X1 = j tan
2 2j 2 2 2 3
3 1 3 1 10 1 1
X = j tan
3
1 2 2j 2 2 2
1 1
X2 = j 90
2 2
1 1
X 2 = j 90
2 2
The magnitude spectrum and the phase spectrum are depcted in figure (a) and (b), respectively.
|Xk|
10 / 2
1
0.5
N 1 1 01 2 N k

Xk
/2
N 10 1 2 N k
/2
Figure (a) Magnitude spectrum, (b) Phase spectrum

Example 6.2
Evaluate the fourier series for the discrete-time periodic square wave x ( n ) shown in figure
x(n)
N N1 0 N1 N n
Figure Discrete-time periodic square wave

Solution 6.2
The signal is periodic with period N
1 j o kn
Xk = x ne
N n N
Here o = 2 /N
N1
1 j . kn
Xk = N 1.e N
N1
2
j k (2 N1 1)
e N
1
1 j .kN1
= .e N
2
N j k
e N
1

2 N1 1
j 2 k
Taking e N 2 common from numerator and e
j .
N 2 common from denomination
2 N 1 2 N 1
j k 1 j k 1
N 2 N 2
1 e e
Xk = 2 k 2 k
N j . j .
e N 2
e N 2
2 N 1
sin .k 1
1 N 2
Xk ,0 k N 1
N 2 k
sin .
N 2
Example 6.3
Find the DTFS coeficients of the N-periodic impulse train
x(n) = n- mN
m=
As shown in figure
x n)
N 0 N 2N n
A discrete-time impulse train with time period N.

Solution 6.3
The signal is periodic with period N and frequency 0 = 2 /N. The given signal for one period may be
wrietten as
1, n 0
x(n) =
0,1 n N 1
= (n )
Therefore, the fourier series coefficients are
N 1
1 jk o n
Xk = x ne
N n 0
N 1
1 jk 2 / Nn 1 jk 2 / N n 1
= ne e
N n 0 N n 0 N

Example 6.4
Find the frequency domain representation of the signal depicted in figure
x n)
6 5 0 1 2 3 45 6 78
n
1/2
Solution 6.4
2
The signal has period N = 5, so o = . Also, the has odd symmetery, so we sum over n = – 2 to
5
n = 2. Thus
1 jk o n 1 2 jk 2 /5 n
Xk = x ne = x ne
N n N 5n 2
jk 4 /5
1 x 2 e x 1 e jk 2 /5
=
5 x 0 x 1 e jk 2 /5 x(2)e jk4 / 5
Using the values of x(n), we get
1 1 jk 2 /5 1 jk 2 /5 1 2
Xk = e e 1 j sin k
5 2 2 5 5
The one period of the DTFS coefficients Xk, k = -2 to k = 2 are
1 sin 4 / 5 j 0.531
X-2 = j 0.232e
5 5
1 sin 2 / 5 j 0.760
X-1 = j 0.276e
5 5
1
X0 = 0.2e j 0
5
1 sin 2 / 5
X1 = j 0.276 e j 0.760
5 5
1 sin 4 / 5
X2 = j 0.232 e j 0.531
5 5
Example 6.5
Determine the time-domain signal x ( n ) form the DTFS coefficients depicted in figure

Xk
Xk|
12 3 4 5
7 6 5 3 6 7 8 9 k 7 6 5 4 1 0 12 3 4 5 6 7 8 9 10 k
0
Figure (a) Phase spectrum, (b) Magnitude spectrum
Solution 6.5
The DTFS coefficients have period N = 9, so o = 2 /9. From figure, the DTFS coefficients over the
interval k = 4 to k = 4 are
Xk = | X k | e j Xk
, 4 k 4
X-4 = 0, X 3 1e j 2 /3 , X 2 2e j /3 , X 1 0,
j /3 j 2 /3
X0 = 1ej , = X 1 0, X 2 2e , X3 1e , X4 0
Thus we apply equation to get x(n)
x(n) = X k e jk on
k N
= X k e jk 2 /9 n
k 4
j 8 /9 n j 6 /9 n j 4 /9 n j 2 /9 n
= X 4e X 3e X 2e X 1e
X0 X 1e j 2 /9 n
X 2e j 4 /9 n
X 3e j 6 /9 n
X 4e j8 /9n
= e j2 /3
e j 6 /9n
2e j /3e j 4 /9 n
ej 2e j /3 j 4 /9 n
e e j 2 /3 j 6 /9 n
e
ej 6 /9 n 2 /3
e j 6 /9 n 2 /3
ej 4 /9 n /3
e j 4 /9 n /3
= 2 4 1
2 2
6 2 4
x(n) = 2 cos n 4 cos n 1
9 3 9 3
6.3 Properties of DTFS

The properties of DTFS and continuous time fourier series are similar. Let Xk are DTFS coefficients of x[n]
that is
x[n] Xk

6.3.1 Linearity
It x(n) and y(n) denote two periodic signals with period N, and
x(n) Xk y(n) Yk
Then z(n) = ax(n) + by(n) Zk = aXk + bYk ... (9)
Proof -The fourier series coefficients of z(n) is given by
1 jk on 1 jk on
Zk = xne ax n by n e
Nn N Nn N
1 jk on 1 jk on
= a x n e b y ne
Nn N Nn N
Xk Yk
= aXl+bYk
6.3.2 Time Shifting

When a time shift is applied to a periodic signal x(n), the period N of the signal is preserved.
If x(n) Xk
Then y(n) = x(n n0) Yk = Xk e jk o n0 ... (10)
One consequence of this property is that, when a signal is shifted in time, the magnitudes of its fourier series
coefficients remain unaltered. That is, |Yk|=|Xk|
Proof - by definition,
N 1
1 jk o n 1 jk o n
Yk = y ne y n e
N n N N n 0
N 1
1 jk o n
= x n n0 e
N n 0
A change of variables is perfomed by letting m = ( n n 0), Which also yields ( m n 0) as( n 0),
and(m (N 1 n0)) as (n (N 1)). Therefore,
N 1 n0
1 jk o ( m+n0 )
Yk = x m e
N m n0
N 1 n0
1 jk o m jk o n0
Yk = x m e e
N m n0
jk o n0
Yk = Xk e
Study Note
We can see that in time shifting the effect is see going with same sign
x[n] Xk
x[n] Xk
same sign
jn0 o k
x n n0 X ke

Time shifting only change the phase of DTFS coefficients
6.3.3 Frequency shifting

If x(n) Xk
Then y(n) = e jM on x(n) Yk = Xk M ... (11)
Proof - by definition,
1 jk on 1N1 jk on
Yk = y n e y ne
Nn N Nn0
N 1
1
= e jM on
x n e jk o n
N n 0
N 1
1 j k M on
= x n e Xk M
N n 0
Hence, a frequency shift corresponds to multiplication in time domain by a complex exponential whose
frequency is equal to the time shift.
Study Note
The frequency shifting property is easy to remember, here the effect goes with opposite sign.
x[n] Xk
Then
x[n] Xk
apposite sign
j o Mn
e x (n ) Xk M
6.3.4 Time Reversal

If x(n) Xk
Then y(n) = x( n) Yk = X k
In other words, the time reversal applied to a discrete-time signal results in a time reversal of the corre-
sponding sequence of fourier series coefficients.
Proof
N 1
1 jk o n 1 jk o n
Yk = y ne y n e
N n N N n 0
N 1 0
1 jk o n 1 j k om
= x ne x m e X k
N n 0 N m N 1
An interesting consequence-of the time reversal property is that if x(n) is even then its fourier series coef-
ficients are also even, i.e.,

if x( n) = x(n)
Then X k = Xk ... (13)
Similarly, if x(n) is odd, then so are its fourier series confficients, i.e.,
if x( n ) = x(n )
Then X k = Xk ... (14)
6.3.5 Time scaling

Because of the discrete-time nature of the time index for discrete-time signals, the relation between time
scaling and fourier series coefficients in discrete-time takes on a different form its continuous-time counterpart.
If we try to define the signal x(an), we run into difficulties if a is not an integer. Therefore, we cannot slow down
the signal by choosing a < 1. On the other hand, if we let a be an integer other then ±1, we do not merely speed
up the original signal. For example, since n can take on only integer values, the signal x(2n) consists of the even
samples of x(n) alone. Let m be a positive integer, and the signal is defined as
x n / m , if n multiple of m
x(m)(n) = ... (15)
0, if n is not a multiple of m
xm(n) can be obtained from x(n) by placing (m-1) zeros between successive value of the original signal.
Intuitively, we can think of x(m)(n) as a slowed-down version of x(n). If x(n) is periodic with period N, then
y(n) = x(m)(n) is also periodic with period mN.
Now, if x(n) Xk
1
Then y(n) = x(m)(n) Yk Xk ... (16)
m
The fourier series coefficients Yk = (1/m) Xk are also periodic with period mN.
Proof - the fourier series coefficients of y(n) = x(m)(n) [y(n) is periodic with period mN. and fundamental
frequency o/m] are given by
mN 1
1 jk ( o / m ) n
Yk = y (n)e
mN n 0
mN 1
1 n jk ( o / m ) n
= x e
mN n 0 m
A change of variables is performed by letting r = n/m, which also yilds r = 0 as n = 0, and r = N 1 as n =
m(N 1).
N 1
1 1 jk o r 1
Therefore, Yk = x ( r )e Xk
mN n 0 m
Xk
6.3.6 Periodic convolution

If x(n) Xk ,
y(n) Yk
Then z(n) = x(n) y(n) = x(r ) y (n r ) Zk NX k Yk ... (17)

r N

Where denotes the periodic convolution.

Proof - for periodic signals with the same period, a special form of convolution, known as periodic convolu-
tion, is defined as
z(n) = x(r ) y (n r )
r N
It is easy to show that z(n) is periodic with period N and the periodic convolution is comutative and associa-
tive. Thus, we can write z(n) in a fourier series representation with coefficient
1 jk on 1 jk o n
Zk z(n)e = x (r ) y ( n r ) e
Nn N N n N r N
1 jk o n
Zk = x(r ) y (n r ) e
n N N r N
From the time shifting property, i.e.,

if y(n) Yk,
Then y(n r) = Yk e jr o n
We have
jr o n
Zk = N N x ( r )e Yk NX k Yk
r N
Xk
Study Note
Convolution in time domain is multiplication in freqency domain.
6.3.7 Multiplication
If x(n) and y(n) are periodic signals with the same period N, and
x(n) Xk ,
y(n) Yk,
Then z(n) = x(n)y(n) = Z k X r Yk r ... (18)

r N
Zk may be interpreted as the periodic convolution between the two periodic sequences of fourier coefficients
Xk and Yk.
Proof - consider the signal z(n).
z(n) = x(n)y(n) = X r e jr on
Ym e jm on
r N m N
= Xr Ym e j ( m r ) on
r N m N
A change of variables is perfomed by letting k = m + r, which also yileds m = (k r), (k r) as (m 0), and
[k (r+N 1)] as [m (N 1)]. Therefore,

r N 1
z(n) = Xr Yk r e jk on
r N k r
= Xr Yk r e jk on
r N k N
z(n) = X r Yk r e jk on
Z k e jk on
k N r N k N
Thus, Zk = X r Yk r
r N
Study Note
Multiplication in time domain is convolution in frequency domain.
6.3.8 First differece

If x[n] is periodic signal with period N and it is DTFS coeffients are Xk. A new signal y[n] is related to x[n]
y[n] = x[n] x[n 1] ... (19)
Signal g[n] will also be periodic with period N and
j ok
Yk = Xk Xk e ... (20)
6.3.9 Running Sum or Accumulation

If x(n) Xk
y(n) Yk
n
1
Then y(n)= x(k ) Yk Xk , k 0 ... (21)
k 1 e jk o
Proof - consider the running sum

n
y(n) = x(k )
k
n 1
y(n) = x ( n) x(k )
k
y(n) = x(n)+y(n 1)
y(n) y(n ) = x(n)
Using the time-shifting and linearity properties, we get
j ok
Yk Yk e = Xk
1
Yk = Xk
1 e jk o

Study Note
The discrete-time fourier series coefficient Yk of the running sum y(n) = x(k ) is finite-valued and periodic
k
only if X0 = 0, that is average value of signal should be zero then only the signal y(n) will be finite.
6.3.10 Conjugation and conjugate symmetery

If x(n) Xk
Then y(n) = x*(n) Yk = X* k ... (22)
Taking the complex conjugate of a periodic signal x(n) has the effect of complex conjugation and time
reversal on the corresponding fourier series coefficients.
Proof
1 jk o n 1 jk o n
Yk = y ( n )e x * ( n) e
N n N N n N
*
*
1 1
= x(n)e jk on
x (n)e j ( k ) on
N n N N n N
X k
= (X k)*= X* k
1. When the signal is real

That is x[n] = x*[n]
If x[n] Xk
and x*[n] X* k
Using all properties we get
Xk = X* k
Thus for a real time domain signal Xk values are conjugate symmetric. Thus Xk has even magnitude plot and
odd phase plot.
Study Note
Similary if x[n] is imaginary then Xk the DTFS coefficients will be conjugate anti symmetric.
2 . When the signal is real and even

Here x[n] = x*[n] and x[n] = x[ n]
Xk = X* k and Xk = X k
Using these conditions we get
X* k = X k = Xk
Thus Xk values are also real and even
3 . when the signal is real and odd
Here x[n] = x*[n] and x[n] = x[ n]
Xk = X* k and Xk = X k
Using these conditions we get Xk * = Xk

Thus Xk are imaginary and odd.

4 . If signal is real with DTFS coefficients X k then DTFS coefficients of even and odd part of
signal will be ( X k = Re( X k )+ j Img( X k ))
Here x[n] Xk
x[n] x[ n]
And xe[n] =
2
x[n] x[ n]
And xo[n] =
2
Xk X k
xe[n]
2
Since X k = Xk*
Xk X k*
xe[n] Real X k
2
Xk X k*
And xo[n] j Im g X k
2
REMEMBER Thus we can club all properties of DTFS coefficients for time doman signal
Time domain DTFS coefficients
signal x[n] Xk
Imaginary Conjugate anti symmetric
Even Even
Odd Odd
Real and even Real and even
Real and odd Imaginary and odd
Imaginary and even Imaginary and even
Imaginary and odd Real and odd
The arrow used here is double sided which means for example when time domain signal is
conjugate symmetric then DTFS coefficients will be real and similary for rest all properties
it is both sided

If x(n) Xk
1 | X k |2
Then | x(n) |2 = ... (23)
N n N k N
As in the continuous-time case, the left-hand side of Parseval’s relation is the average power in one period
for the periodic signal x(n). Similarly |Xk|2 is the average power in the kth harmonic component of x(n). Thus,
Parseval’s relation states that the average power in a periodic signal equals the sum of the average powers in all
of its harmonic components. In discrete-time, of course, there are only N distinct harmonic components, and
since the Xk are periodic with period N, the sum on the right-hand side of Eq(23) can be taken over any N

consecutive values of k.
Proof - consider the LHS of Eq. (23),we have
1 1
| x(n) |2 = x ( n ) x * ( n)
N n N N n N
1
= x(n) X k e jk on
N n N k N
1 1 jk o n
= x ( n) X k*e
N n N Nk N
jk o n
= X k* x (n)
k N n N
X k* X k
= k N
| X k |2
= k N
6.3.12 Duality Theorem

We have seen that using DTFS we transform a discrete periodic x[n] into discrete periodic signal Xk.
x[n] Xk
1 j o kn
Xk = x n e ... (24)
N n N
j o kn
And x[n] = Xke ... (25)
k N
Both signals Xk and x[n] are periodic with period N. The format of formulae to find DTFS coefficient is
j o kn
DTFS coefficient = Time domain signal e
n N
If we manipulate eq(25) we get
x[n] 1 j o kn
= X ke
N N k N
Inter changing variable n by k
x[ k] 1 j o kn
= X ne
N N n N
x[ k ]
Xn
N

x[ k ]
Thus are DTFS coefficients of signal Xn , thus the duality theorem is
N
If Xk are DTFS coefficients of x[n], x[n] xk
x[ k ] x[ k ]
Then DTFS coefficients of Xn are Xn
N N
6.4 Output of discrete system with periodic input

Let us consider a LTI system whose impulse response is h[n] and input is x[n] = Cn where C is constant.
Thus output is
y[n] = x[n]*h[n]
= h[m]x n m
m
= h[m]C n m
= C
n
h[m]C m
m m
The z-Transform of h[n] is
m
H(z) = h[n]z
m
y[n] = H(z)|z=C . Cn
Thus when input is Cn then output of LTI system is Cn. H(z)|z=C. From the above analysis we get that
LTI system
ej on
h[n] H(z) ej on
.H ( z ) z ej o
LTI system
X ke j o kn
h[n] H(z)
X ke j on
.H ( z ) j o
z e
x n X ke j o kn LTI system X ke j on
.H ( z ) z ej o
k N h[n] H (z) k N
When x[n] is input then output y(n) is
y[n] = Xk H e j ok
ej o kn
k N
From the above analysis we get that if input to a LTI system is periodic then output is also periodic with
same period. If DTFS coefficient of x[n] are Xk then output will have DTFS coefficient of Xk H( e j ok ).
6.5 Insights on formulae of DTFS

From eq(6) and (7) we can say that

1
X0 = x[n] , average value of periodic signal x[n]
Nk N
x[o] = X k , sum of all DTFS coefficients

k N
If in a question a singal x[n] is given and we have to find phase of Xk then qenerally we can make x[n]
even by shifting it right or left. Thus y[n] = x[n±a] and Yk will have zero phase as y[n] is real and even.
ja o k
Since Yk = Xk e and we can easily find phase of Xk.
If a signal x[n] is periodic with period N and N is even,
N
And x[n] = x n
2
That is signal is half wave symmetric then Xk will be zero for even value of k, because (using time shifting
property)
N
j o k
Xk = X ke 2
Xk = Xke±j k
Xk + Xke±j k = 0
Xk = 0 For even values of k
Similary if N is divisible by 4
N
And x[n] = x n
4
Then Xk = 0 For all where k is multiple of 4.
If x[n] is real and N is even then Xk for k =N/2 will also be real.
1 jn o k
Since, Xk = x[n]e
N n N
jn o k
Since x[n] is real and e will become real when k = N/2.
If a signal x[n] is not looking like half wave symmetric then it can be shifted on amplitude axis. If after
shifting the signal y[n] = x[n]±a we get it is half wave symmetric then Yk will have property of half wave
symmtric and Yk = Xk for all values except for k = 0 because X0 Y0 as by shifting the signal we have
changed it is average value.
As we have done in CTFS we can write the discrete periodic signal in terms of harmonics of cosine and
sine also. We can also prove that odd signals will have only sine terms and even have only cosine terms.
If any signal is made odd by shifting on amplitude axis then it will show all properties of odd signal except
there will be change in value of DTFS coefficient at k = 0.
Example 6.6
A discrete-time periodic signal x ( n ) is real valued and has a fundamental period N = 5.
Tge nonzero fourier series coefficients for x ( n ) are
X 0 = 1, X 2 = X *2e jπ / 4 , X4 = X *4 = 2e jπ / 3

Express x ( n ) in the form
x ( n ) = A0 + Ak sin k n+ k
k=1
Solution 6.6
The singal x(n) is periodic with period N = 5 and fundamental frequency 0 = 2 /N = 2 /5.
Using the fourier synthesis , we have
x(n) =
X k e jk on
k N
= X0 X 2e j 2 on
X 2e j 2 on
X 4e j 4 on
X 4e j 4 on
= X0 X 2 e j 22 / Nn
X 2e j 22 / Nn
X 4 e j 42 / Nn
X 4e j 42 / Nn
= 1 ej /4
e j 22 /5 n
e j /4
e j 22 /5 n
2e j /3e j 42 5/ n
2e j /3
e 42 /5 n
ej 4 /5 n /4
e j 4 /5 n /4
ej 8 /5 n /3
e j 8 /5 n /3
= 1 2 4
2 2
4 8 5
= 1 2 cos n 4 cos n
5 4 5 3
4 3 8 5
= 1 2sin n 4sin n
5 4 5 6
= A0 Ak sin wk n k
k 1
Example 6.7
Let x ( n ) be a real and odd periodic signal with period N = 7 and fourier series coefficients
X k . Given that
X 15 = j, X 16 = 2 j , X 17 = 3 j
Determine the values of X 0 , X -1 , X -2 , X -3 .
Solution 6.7
The discrete-time fourier series coefficients are periodic with period N, i.e.,
Xk = Xk+N = Xk+2N
Therefor, for N = 7, we have
X 1 = X8 = X15 = j
X 2 = X9 = X16 = 2j
X 3 = X10 = X17 = 3j
Since the given signal x(n) is real and odd, the fourier series coefficients Xk will be purely
imaginary and odd (Xk = X k). Therefore,
X0 = 0
X1 = X 1
X2 = X 2

X3 = X3
Finally, we have X1 = X1 = j
X2 = X2 = 2j
X3 = X3 = 3j
Example 6.8
Suppose we are given the following information about a signal x ( n ) :
1 . x ( n ) is a real and even signal.
2 . x ( n ) has a period N = 10 and fourier coefficients X k .
3 . X 11 = 5.
1 9
4. | x(n)|2 = 50.
10 n=0
Show that x ( n ) = A cos ( Bn + C ), and specify unmerical values for the constants A, B, and
C.
Solution 6.8
The fourier series coefficients are periodic with period N = 10, we have
Xk = Xk+N = Xk = Xk+10
X 1 = X11 = 5
X 1 = X9
Since, the given signal x(n) is real and even, the fourier coefficients Xk are also real and even. Therefore,
Xk = X k
X 1 = X 1 = X9 = 5
Using Parseval’s relation,
N 1 N 1
1
| x( n) |2 = | X k |2
N n 0 k 0
1 9 9
| x(n) |2 = | X k |2 50
10 n 0 n 0
8
2 2
| X0 | | X1 | | X9 | | X k |2 = 50
k 2
8
| X 0 |2 25 25 | X k |2 = 50
k 2
8
| X 0 |2 | X k |2 = 0
k 2
Therefor, for one period the fourier coefficients are
5, k 1,9
Xk =
0, k 0 and 2 k 8
Now using the synthesis equation, we have

jk o n
x(n) = Xke
k N
N 1 9
= X k e jk 2 / Nn
X k e jk 2 /10 n
k 0 k 0
= X 1e j 2 /10 n
X 9 e j18 /10 n
Substituting X9 = X 1, we get
x(n) = X 1e j 2 /10 n
X 1e j 2 /10 n
= 5e j /5 n
5e j /5 n
ej /5 n
e j /5 n
= 10
2
n
= 10cos
5
Example 6.9
Each of the two sequences x ( n ) and g ( n ) has a period N = 4, and the corresponding fourier
series coefficients are specified as
x(n) y(n) Yk
1 1
Where X 0 = X3 = X1 = X2 =1
2 2
Y 0 = Y1 = Y2 = Y 3 = 1
Using the multiplication property, determine the fourier series Z k for the signal z ( n ) =
x ( n ) y ( n ).
Solution 6.9
The given signal z(n) = x(n)y(n) is periodic with period N = 4. Using the multiplication property,
we have
z(n) = x(n)y(n)
3
Zk = X rYk r = Zk = X rYk r
r N r 0
Zk = X0Yk+X1Yk 1+X2Yk 2+X2Yk 3

Zk = Yk+2Yk 1+2Yk 2+Yk 3
Since Yk = Yk+N = 1 for all values of k, therefore
Zk = 1 + 2 + 2 + 1 = 6 for all values of k
Example 6.10
When the impulse train
x(n) = (n- 4k )
r=-

Is the inputs to a particular LTI system with frequency response H ( e jw ), the output of the
system is found to be
5π π
y ( n ) = cos( n+ )
2 4
Determine the values of H ( e jk /2) for k = 0, 1, 2 and 3.

Solution 6.10
The given input signal x(n) is periodic with period N = 4 and frequency 0 = 2 /N = 2 /4. The
given signal for one period may be written as
1, n 0
x(n) =
0, n 1, 2,3
= (n )
Therefore the fourier series coefficients are
N 1
1 jk o n
Xk = x ( n) e
N n 0
1 3 jk 2 /4 n 1 3 jk 2 /4 n
= x ( n )e = ( n )e
4n 0 4n 0
1
Xk = , for all k
4
we know that the output y(n) of an LTI system to a periodic input is given by,
y(n) = X k H (e jk o )e jk on
k N
= X k H (e jk 2 /N
)e jk 2 /N
k 0
= X k H (e jk 2 /4 )e jk 2 /4 n
k 0
y(n) = X0H(ej0)ej0n+X1H(ej /2)ej /2n+X

2H(e
j )ej n+X3H(ej3 /2)ej3 /2n
1 1 1 1
y(n) = H e j 0 e j 0n H ej /2
ej /2 n
H ej ej n H e j3 /2
e j3 /2n
4 4 4 4
Given that
5
y(n) = cos n = cos 2 n n = cos n
2 4 2 4 2 4
1 j /2 n /4 1 j /2 n /4
y(n) = e e
2 2

1 j /2 n /4 1 j3 /2 n /4
y(n) = e e
2 2
1 j /4 j /2 n 1
y(n) = e e e /4
ej 3 /2 n
2 2
Comparing the above equation with Eq.(1), we get
H(ej0) = H(ej ) = 0,
H(ej /2) = 2ej /4, H(ej3 /2) = 2e j /4
Example 6.11
If the signal is of period N = 8 and it is DTFS coefficients X k are
kπ 2kπ
X k = Cos +sin
4 4
Then find the signal x [ n ].
Solution 6.11
Since Xk is given in sine and cosine, we cannot apply eq(6) to find x[n]. we will do comparison to find x[n].
1 j o kn
Since Xk = x[ n]e
N n 8
2
Since periodic N = 8 , O =
8
e jk /4 e jk /4
e j 3 k /4
e j 3 k /4
Xk =
2 2 2j 2j
And we will compare above equation with
2
1 4 j .kn
Xk = x n e 8
8n 3
We get x[1] = 4
x[ 1] = 4
x[3] = 4j
x[ ] = 4j
x[n] = 4j [n+3] 4j [n 3]+4 [n 1]+4 [n+1]
for one period 3 to 4
Example 6.12
Suppose we are given the following information about a periodic signal x ( n ) with period
N = 8 and fourier coefficients X k .
1. Xk = Xk 4
2. x (2 n +1) = ( 1) n
Sketch one period of x ( n )
Solution 6.12
We known that if x(n) Xk

Then, from the frequency shifting property, we have

2 N
j n
( 1)nx(n) = ej nx(n) = e N 2 x(n) Xk N/2
Since, it is given that Xk = Xk 4, we have
( 1)nx(n) Xk
n
( 1) x(n) x(n )
x(n)[1+( 1)n] 0
This implies that x(n) = 0, For n = 0,±2, ±4, ±6,
From the given information x(2n+1) = ( 1)n, we get
x(n) = 1, x(3) = 1, x(5) = 1, x(7) = 1
Therefore, over one period 0 n 7,x(n) is defined as
0, n 0, 2,4,6
x(n) = 1 n 1,5
1 n 3,7
One period of x(n) is shown in figure
x n)
3 7
0 1 2 4 5 6 n
Example 6.13
Let x ( n ) be a periodic signal with period N = 8 and fourier series coefficients X k = Xk 4.

A signal
n
1+ -1
y(n) = x n -1
2
With period N = 8 is generated. Denoting the fourier series coefficients of y ( n ) by Y k , find

a function such that
Y k = f(k)Xk
Solution 6.13
We known that if x(n) Xk
Then from the frequency shifting property, we have
2 N
j n
( 1)nx(n) = ej nx(n) = e N 2 x(n) Xk N/2
In this case N = 8, therefore
( 1)nx(n) Xk 4
Since, it is given that Xk = Xk 4, we have

( 1)nx(n) = Xk

( 1)nx(n) = x(n )
( 1)nx(n) = x(n)
( 1)n+1x(n) = x(n)
( 1)nx(n 1) = x(n 1)
Now consider
n
1
y(n) = x n 1
2
1 1 n
= x n 1 1 x n 1
2 2
We have already proved that ( 1)nx(n 1) = x(n 1), Therefore
y(n) = x(n 1)
Yk = Xke jk2 /8 = f(k)Xk
Thus f(k) = e jk2 /8

j n /2
S1 :e ej n /2
u n ,
j n/ 2
S2 :e e j3 n /2
Subjective Practice Problems

j n/ 2
S3 :e e j5 n /2
1. Each if the two sequences x1[n]and x2[n] has a For each system, determine whether the given
period N = 4, and the corresponding fourier se- information is sufficient to concrete that the sys-
ries coefficients are specified as tem is difinitely not LTI.
x1[n] ak,
x2[n] bk, 4. Let
Where
1, 0 n 7
x[n] =
1 1 0, 8 n 9
a0 a3 a1 a2 =1
2 2
be a periodic signal with fundmental period N=
And b0 = b1 = b2= b3= 1. 10 and Fourier series coefficients ak. Also, let
Using the multilication property, determine the g[n] = x[n] x[n 1]

fourier series coefficients ck for the signal g[n] =
(a) Show that g[n] has a fundmental period of 10.
x1[n]x2[n].
(b) Determine the Fourier series coefficients of
2. Determine the output of the filter shown in figure
g[n].
for the following periodic inputs:
5. Consider a causal discrete-time LTI system whose
(a) x1[n] = ( 1)n
input x[n] and output y[n] related by the following
difference equation:
3
(b) x2 n 1 sin n
8 4 1
y n y n 1 =x[n]
4
n 4k
1 Find the fourier series representation of the
(c) x3 n u n 4k
k
2 output y[n] for each of following inputs:
H (e j ) 3
(a) x[n] = sin n
4
1
(b) x[n] = cos n 2 cos n

5 0
4 2
2 5 11
3 12 12 3 3 12 12 3
6. Let x[n] be a periodic sequece with period N and
fourier series representation
3. Consider three discrete-time systems S1,S2 and Ss
whose respective response a complex input ej n/2 ak e
jk 2 / N n .
x[n]=
are specified as k N

Discrete Time Fourier Seriess 295
The fourier series coefficients for each of the NIM 1

following signals can be expressed in terms of ak. N
x n r = 0 for all n,
Derive the expressions. r 0 M
(a) x[n-n0]
Then ak = 0 for every value of k that is a mul-
(b) x[n] x[n 1] tiple of M.
N
(c) x[ n] x n (assume that N is even) 8. Suppose we are given the following information
2
about a periodic signal x[n] period 8 and fourier
coefficients ak:
3
(d) sin n (assume that N is even; note that 1. ak = ak 4.
4
2. x[2n + 1] = ( 1)n.
this signal is periodic with period N/2 )
Sketch one period of x[n].
(e) x*[ n]
9. Let x[n] be a real periodic signal with period N
(f) ( 1)n x[n] (Assume that N is even)
and Fourier coefficients ak.
(g) ( 1)n x[n] (Assume that is odd; note that
(a) Show that if N is even, at least two of the fourier
this signal is periodic with period 2N)
coefficients within one period of ak are real.
x n , n even (b) Show that if N is odd, at least one of the fourier

(h) y[n] coefficients within one period of ak is real.
0, n odd
10. Evaluate the time-domain signals represented by
7. Let x[n] be a periodic sequence with period N and the following DTFS coefficients:
fourier series representation
8
(a) X k cos k
x[n]= ak e
jk 2 / N n .
21
k N
(b) X k
10 4
cos k j 2 sin k
19 19
(a) Suppose that N is even and
N (c) X k m
x[n]= x n for all n, 1 k 2m 2 k 3m
2 m
Show that ak = 0 for all even integers k. (d) Xk as depicted in Figure.
(b) Suppose that N is divisible by 4. Show that if

|Xk|
N
x[n]= x n
4
Then ak = 0 for every value of k that is a mul- 1
tiple of 4.
10 11 4 3 0 34 1011 k
(c)More generally, suppose that N is divisible by
an integer M. Show that if

6
Xk (a) x(n) = cos n
17 3
2 14 10
4 11 (b) x(n) = 2sin n cos n
10
19 19
4 0 3 k
2
m
(c) x(n) ( 1) n 2m n 3m
m
11. One period of the DTFS coefficients of a signal is

13. Consider a dicrete-time LTI system whose fre-
given by
quency response is
k
1 1, | | /8
Xk= ,0 k 9
2 H(ej )=
0, /8 | |
Find the time-domain signal x(n) assuming N = 10
Show that if the input x(n) to this system has a
period N = 3, the output y(n) has only nonzero
12.Evaluate the DTFS coefficients of the following fourier series coefficient per period.
signals:

Chapter
z-Transform
7
7.1 Introduction
For frequency analysis of a discrete periodic signal we have DTFS where we expressed any signal x[n] in
terms of linear combination of complex exponentials. These complex exponentials are orthogonal to each other.
For frequency analysis of a discrete nonperiodic signal we use Discrete Time Fourier Transform (DTFT). We
will look at DTFT in next chapter. The formulae to calculate DTFT of any signal x[n] is
j n
X(e j ) = x( n) e ...(1)
n
The condition to find DTFT of any signal is that the signal x[n] should be absolutely summable, that is
x [ n] < ...(2)
n
For signals which are not absolutely summable we use z-Transform for frequency analysis. The z-Transform
of a signal is defined as,
n
X (z ) = x [ n] z ...(3)
n
Here z is available re j . The equation (3) define the bilateral z-transform of a signal x[n]. Another way to
define z-transform is that suppose we have an LTI system with impulse response h[n] and input x[n] = Cn where
C is any constant.
n LTI system
x[n] = C y [n]
h[n]
Fig. 7.1 : LTI system with input Cn
Thus, y[n] = x[n] h[n] = h[k ] x[ n k] h[k] Cn k
n k
y[n] = Cn h[k ] C k
...(4)
k
n
If we define a new function, H(z) = h[ n] z ...(5)
n

Thus, we can say that, y[n] = C n H(z) ...(6)

z C
Thus output of the system = Input ( H( z) Z C

).
Study Note
n
• Whenever input to system is Cn for < n< where C is any constant then output is C H (z ) z C .
• Cn is called eigen function for discrete time LTI system and H( z) is called eigen value.
Z C
7.2 Bilateral z-Transform

The bilateral z-transform of any discrete signal x[n] is defined as,
n
X (z ) = x[ n] z ...(7)
n
here, z = re j
Thus r is the magnitude of z and is phase of z.
z-Plane
It is convenient to represent complex z as a location an complex plane with distance r from origin and
as the angle from positive real axis. The complex plane also called z-plane here.
Relationship between z-transform and discrete time Fourier transform:

Im{z}
j
re
Re{z}
Fig. 7.2 : The z-plane with point re j shown at a distance

r from origin and angle from real {z} axis
From equation (7) we can see that,
n
X (z ) = x[ n] z x[ n] ( r e j ) n
n n
n j n
= ( x[ n] r )e
n
Thus we can say that, z-transform of x[n] is DTFT of x[n] r–n and z-transform of any signal x[n] at r = 1
will be equal to it’s DTFT.
That is, z{x[n]} = F{x[n] r–n} ...(8)
and for r = 1, z{x[n] = F{x[n]} ...(9)
Thus we can say that DTFT of any signal x[n] = X( z) z ej
but the condition is that z-transform should

z-Transform 299
be valid for r = 1 that is ROC of z-transform must contain r = 1. We will look at ROC in section 7.3.
Poles and Zeros of z-Transform

The z-transform of any discrete signal X(z) can be written as,
bm z m b m 1 z m 1 ... b1z b0
X (z ) = an z n a n 1 z n 1 ... a1z a0
The values of z where X(z) = 0 are called zeros. That is the roots of numerator are zeros, the values of
z where denominator of X(z) tends of zero or X(z) tends to infinity are called poles. Thus roots of denominator
are called poles.
( z z1 ) ( z z2 ) .... ( z zm )
X (z ) = k
( z p1 ) ( z p2 ) .... ( z pn )
The zeros (z1, z2 ...zm) are values of z where numerator of X(z) goes to zero and X(z) = 0. Thus,
LimX( z) = 0
z zi
and poles (p1, p2,...pn) are values of z where denominator of X(z) goes to zero and X(z) = . Thus,
LimX( z) =
z pi
7.3 Region of Convergence (ROC) for z-Transform

z-Transform of x[n] (X(z)) is DTFT of x[n] r–n, to find DTFT of any signal the condition is that it should
be absolutely summable. Thus X(z) is valid any when x[n]r–n is absolutely summable. That is
n
x[ n] r < ...(10)
n
Thus for valid X(z) equation (10) should be satisfied. The region of values of r where equation (10) is
satisfied is called ROC of X(z), region of convergence of z-transform.
Study Note
• ROC define the region of values of r i.e. z for which X(z) is valid or for which X(z) exist.
• DTFT of x[ n] X( z) z ej only when z-transform is valid at r = 1 or ROC contain r = 1.
• In z-plane r = 1 means a unit circle.
Example 7.1
Determine the constraint on r z for each of the following sums to converge:

n 1
1
n 1
1 1 ( 1)n n
(a) z n
(b) zn (c) z
n 1 2 n 1 2 n 0 2
Solution 7.1
( a ) Consider the given summation,

n 1 n n
1 n 1 1 1 n 1 1 1 1
z = z z z z
n 1 2 2 2 n 0 2 2 2 n 0 2
1 1 1
The summation of the second term on the RHS will converge if z 1 or z . Therefore, the
2 2
1
given summation will converge if z .
2
( b ) Consider the given summation,
n 1 n
1 1 1 1
zn = zn (2 z)n
n 1 2 2 n 1 2 2 n 1
1
Therefore, the given summation will converge if 2 z 1, or z .
2
( c ) Consider the given summation,
1 ( 1)n n 1 n 1
z = z ( 1)n z n
n 0 2 2 n 0 2 n 0
1 1
= ( z 1 )n ( z 1 )n
2 n 0 2 n 0
1
The first summation will converge if z 1, or z 1. Therefore, the given summation will converge
if z 1.
Example 7.2
Determine the z -transform of the causal signal,

x ( n ) = an u(n )
and depict the ROC and the locations of poles and zeros in the z-plane.
Solution 7.2
n
By definition, X (z ) = x( n)z a n u( n)z n
n n
1, n 0
Since, u(n) =
0, n 0
we obtain, X (z ) = an z n
( az 1 )n
n 0 n 0
1 1
X (z ) = 1
, for az 1 or equivalently z a
1 az

z-Transform 301
z
X (z ) = , z a
z a
1
Therefore, an u(n) 1
, z a
1 az
Example 7.3
Determine the z-transform of the anticausal signal,

x ( n ) = – a n u (– n – 1)
and depict the ROC and the locations of poles and zeros in the z -plane.
Solution 7.3
n
By definition, X (z ) = x( n) z a n u( n 1) z n
n n
1, ( n 1) 0 n 1
Since, u(–n – 1) =
0, ( n 1) 0 n 1
1
we obtain, X (z ) = a nz n
a n zn ( a 1z)n ( a 1z)n
n n 1 n 1 n 1 0
A change of variables is performed by letting m = n – 1, which also yields n = m + 1, m = 0 as n = 1, and

m = as n = . Therefore,
X (z ) = (a 1
z )m 1
(a 1
z) (a 1
z )m
m 0 m 0
1 1
= (a 1
z) 1
, for a z 1 or equivalent z a
1 a z
1
= 1
z a
1 az
z
X (z ) = z a
z a
1
Therefore, –an u(–n – 1) 1
, z a
1 az
7.3.1 Properties of ROC

In this section, we explain properties of the ROC of various classes of signals:
1. The ROC of X(z) consists of a ring in the z-plane ‘centered’ about the origin.
2. If X(z) is rational, then the ROC must not contain any poles.
3. If x(n) is of finite duration and is absolutely summable (i.e. n

x( n) ) then the ROC is the
entire z-plane, except possibly z = 0 and/or z = .

4. If x(n) is right-sided and of infinite duration (i.e., x(n) = 0 for all n < N1 for some finite N1), then
the ROC is the region in the z-plane outside of the outermost pole, i.e., outside the circle of radius
equal to the largest magnitude of the pole of X(z). Furthermore, if x(n) is causal (i.e., if it is right-
sided and equal to 0 for n < 0), then the ROC also includes z = .

5. If x(n) is left-sided and of infinite duration (i.e., x(n) = 0 for all n > N2 for some finite N2), then the
ROC is the region in the z-plane inside the innermost pole, i.e., inside the circle of radius equal to
the smallest magnitude of the poles X(z) other than any at z = 0 and extending inward to and
possibly including z = 0. In particular, if x(n) is anticausal (i.e., if it is left-sided and equal to 0 for
n > 0), then the ROC also includes z = 0.
6. If x(n) is two-sided and of infinite duration (i.e., the signal is of infinite extent for both n < 0 and
n > 0), then the ROC will consist of a ring in the z-plane.
7. For a finite duration right-sided signal (i.e., x(n) = 0 for n < 0 and n > N1 for some finite N1), the
ROC will be the entire z-plane except z = 0.
8. For a finite duration two-sided signal (i.e., the signal is of finite extent for both n < 0 and n > 0), the
ROC will be the entire z-plane except z = 0 and z = .
9. For a finite duration left-sided signal (i.e., x(n) = 0 for n > 0 and n < –N1 for some finite positive
N1), the ROC will be the entire z-plane except z = .
Study Note
• Defining ROC is very important because expression of z-transform of an u[n] and –an u[–n – 1] is same.
•In z-transform ROC has foramt z > magnitude of pole for right-sided signal, but in Laplace transform ROC has
format s > real part of pole for right-sided signal.
Now lets look at few examples to prove and observe all properties of ROC.
Example 7.4
Find the z-Transform of

( a ) [n] ( b ) [ n – 1]
( c ) [ n + 1] ( d ) u[n]
n n
1 1
( e ) 3n u [n ] u[n ] ( f ) 3 n u[ n ] u [ n 1]
2 3
Solution 7.4
( a ) For, x[n] = [n], z-Transform is
n n
X (z ) = x[ n] z [ n] z
n n
1, n 0
we know that, [n] =
0, n 0
That is signal is finite duration signal.
X (z ) = 1
We can see that, X(z) will never goto or undefined for any value of z. Thus ROC of X(z) is entire
z-plane.
1, n 1
( b ) For, x[n] = [n 1] =
0, n 1
The z-Transform will be

z-Transform 303
n n 1
X (z ) = x [n ] z [n 1] z z
n n
The signal is finite duration and right sided.

1
Thus, X (z ) =
z
The ROC will be entire z-plane except z = 0 because X(z) has a pole at z = 0 and ROC donot contain
pole. Also we can see that when z = 0 then X(z) is undefined and tends to .
1, n 1
( c ) For, x[n] = [n + 1] =
0, n 1
The z-Transform will be

n n
X (z ) = x[ n] z [ n 1] z z1 z
n n
The signal is of finite duration and left-sided. Thus ROC will be entire z-plane except z = because at
z = the X(z) = and thus undefined so ROC is entire z-plane except z = .
1, n 0
( d ) For, x[n] = u[n] =
0, n 0
Thus z-transform will be
n
n n 1
X (z ) = x [n ] z z
n n 0 n 0 z
1
Thus, X (z ) = 1
1 z
Since, the signal is right sided signal and pole of X(z) is at 1, so ROC will be r > 1.
1
X (z ) = 1
, z 1
1 z
n
1
( e ) For, x[n] = 3n u[ n] u[ n]
2
n
1
So, x[n] has two signals 3n u[ n] u[ n] .
2
Let, X1(z) is z-transform of 3n u[n].
n
3 1
X 1( z ) = 3n u [n ] z n
n 0 z 1 3z 1
Since, signal is right sided and pole of X1(z) is at z = 3, so ROC is z 3.
n
1
Let X2(z) is z-transform of u[ n], thus
2
n n
1 n 1 n 1
X 2( z ) = u[ n] z z
2 2 1 1
n n 0 1 z
2

1 1
Since signal is right sided and pole of X2(z) is at z , thus ROC is z .
2 2
1 1 1
Thus z-transform of x[n] is with ROC z (z 3)
1 3z 1 1 1 2
1 z
2
ROC is ( z 3) .
n
1
( f ) For, x[n] = 3n u[ n] u[ n 1]
3
n
1
x[n] has two signals 3n u[n]
and u[ n 1].
3
Let, X1(z) is z-transform of 3n u[n], thus
1
X 1( z ) = 1
, ROC z 3
1 3z
n
1
and let X2(z) is z-transform of u[ n 1], thus
3
n n
1 n 1 n
X 2( z ) = u[ n 1] z z
n 3 n 3
= (3z)k {Let k = –n}

k 1
3z
X 2( z ) =
1 3z
1 1
Thus, pole of X2(z) is at z and signal is left sided signal, so ROC is z .
3 3
1
We can see that, x[n] is made up of 2 signals which have ROC of z-transform z 3 and z . Thus
3
1
X1(z) is valid for z 3 and X2(z) is valid for z . Thus no valid ROC for z-transform of x[n].
3
Thus z-transform for x[n] cannot be found.
Study Note
• Thus we can see that for right sided signal which is of finite duration the z-transform will have negative power
of z and the ROC will be entire z-plane except z = 0.
• For left sided finite duration signal the z-transform will have positive power of z and ROC will be entire z-plane
except z = .
Example 7.5
Determine z -transform of signals:

n
(a) a , 0 a 1
n
(b) a , 1 a

z-Transform 305
(c) 1
Solution 7.5
n
( a ) Here, x[n] = a , 0 a 1
x[n] can be written as, x[n] = an u[n] + a–n u[–n – 1]
n Im{z}
x (n) = a
0<a<1 Re{z}
a 1 1/a
n
0
(a) (b)
z-Transform 1
a n u[ n] 1
1 az
The pole is at z = a and signal is right sided.
Thus ROC will be z a.
n
1
Similarly, z-transform of a n u[ n 1] u[ n 1] is
a
n
1 z-Transform 1
u[ n 1]
a 1 1
1 z
a
1
Since pole of z-transform is at 1/a and signal is left sided so ROC is z .
a
1 1 1
The z-transform of x[n] will be with ROC z a z .
1 az 1 1 1 a
1 z
a
1
Since, a 1 thus ROC will be a z and
a
a2 1 z
X (z ) =
a 1
(z a) z
a
n
( b ) Here, x[n] = a , a 1
Here, also we can break, a

n = an u[n] + a–n u[–n – 1]
z-Transform 1
a n u[ n] 1
, z a
1 az

n
1 z -Transform 1 1
and u[ n 1] , z
a 1 az 1 a
The ROC of these two signals will have no intersections when a 1.
Thus when a 1 then we cannot find z-transform of signal x[n].
x(n) = a n
a>1
n
0
( c ) Here, x[n] = 1, we can write,

x[n] = 1 = u[n] + u[–n – 1]
z-Transform 1
u[ n] 1
, z 1
1 z
z-Transform 1
and u[ n 1] 1
, z 1
1 z
so we can see that, ROC of there two signals donot intersect. Thus ROC of x[n] will be null. Thus for no
value of z z-transform of x[n] will exist.
Thus, x[n] = 1 will have no z-transform.
Study Note
n
• From above example we can see that signal x[n] which is rising both side as a for 1 < a < or any signal
which is constant for – < n < as x[n] = 1 will have no z-transform.
• Similarly we can prove that Cn for – < n < will have no z-transform.
Let us understand why z -transform of some signals cannot be calculated:

Signals which are constant for – < n < , rises both direction a n for > a > 1, Cn for – < n <
are those for which z-transform cannot be obtained. The condition for existence of z-transform is that x[n] r–n
should be absolutely summable.
1 1.0
n n n
r >1 r =0 r <1
Fig. 7.3 : Plot of r wrt n

z-Transform 307
For any value of r the signals of above category cannot form x[n] r–n absolutely summable.
Example 7.6
Determine the z-transform and the pole-zero plot for the signal,
an , 0 n N 1
x( n )
0, elsewhere
where a > 0.
Solution 7.6
N 1 N 1 1 N
n (1 az )
By definition, X (z ) = x ( n )z an z n
(az 1 n
) 1
n n 0 n 0 1 az
1 N
(1 az )
X (z ) = 1
1 az
1 zN a N
X (z ) =
zN 1 z a
since a > 0, the equation zN – aN = 0, or zN = aN, has N roots at
zk = ae j2 k/N k = 0, 1, 2,...., N – 1
The zero z0 = a (i.e., root for k = 0) cancles the pole at z = a. Thus,
which has N – 1 zeros and N – 1 poles, located as shown in figure. Since x(n) is of finite duration, the ROC
is the entire z-plane except at z = 0 because of the N – 1 poles are located at the origin.
Im{z}
st
(N – 1) order
pole Unit circle
a 1 Re{z}
Pole-zero pattern for N = 16 and 0 < a < 1
Remember: In a finite duration signal the pole will never be at any finite location. The poles will always be at
origin or at infinity.
7.4 s-To-z-Plane Mapping

Let us consider a continuous time domain signal x(t), the signal is sampled by multiplying the signal
with an impulse train.

x(t) x s (t ) x( nT ) (t nT )
n
(t nT )
n
Fig. 7.4 : Sampling of a continous time domain signal
Here T is sampling interval, the Laplace transform of xs(t) is
st st
Xs ( s ) = xs ( t ) e dt x( nT ) (t nT ) e dt
n
which, upon interchanging integration and summation, is
st st
Xs ( s ) = x( nT ) (t nT ) e dt x( nT ) e
n n t nT
snT
= x(t ) e x( n) ( esT ) n
n n
By introducing a new variable, z = esT, this equation can be expressed as,
X s (s ) z n
e sT = x( n) z X( z)
n
It is cleared from the above discussion that the z-transform can be considered to be the Laplace transform
with a change of variable,
1
z = esT or equivalent, sln( z) ...(10)
T
Recall from next chapter that the Laplace variables s was given by
s = +j
where was a constant used to ensure convergence of the integral defining the Laplace transform and
thus the existence of the Laplace transform itself. From the equation,
z = e( + j )t = e t ej t ...(11)
so that, the magnitude of z is given by
z = e t ...(12)
Now, consider the following three cases:
Case-1: = 0. The imaginary axis (s = j ) in the s-plane, = 0 corresponds to z 1 in the z-plane.
That is, the transformation z = esT transforms the imaginary axis (s = j ) in the s-plane into a unit circle ( z 1)
in the z-plane.
Case-2: < 0. The left-half of the s-plane, < 0 corresponds to z 1 in the z-plane. That is, the
transformation z = esT maps the left-half of the s-plane into the inside of the unit circle in the z-plane.
Case-3: > 0. The right-half of the s-plane, > 0, corresponds to z 1 in the z-plane. That is, the
transformation z =esT maps the right-half of the s-plane into the outside of the unit circle in the z-plane.
This mapping of the Laplace variable s into the z-plane through z = esT is illustrated in Fig. 7.5.

z-Transform 309
j Im{z}
s-plane
z-plane
Re{z}
0 1
Fig. 7.5 : Mapping from s-plane to z-plane
Example 7.7
Determine the z -transform of the signal, x ( n ), obtained by sampling the continuous-time

signal,
x ( t ) = e –at u ( t )
every T seconds.
Solution 7.7
The signal, x(n), is obtained by sampling the continuous-time signal,
x(t ) = e–at u(t)
every T seconds therefore,
x(t ) t nT
x( nT ) = x(n) = e–anT u(nT)
x(n) = e–anT u(n)

n anT n
By definition, X (z ) = x( n) z e u( n) z
n n
anT n anT
= e z (e z n)
n 0 n 0
1 aT 1 aT
X (z ) = aT
for e z 1 z e
1 e z 1
7.5 Properties of z-Transform

7.5.1 Linearity
The conversion of a signal from time domain to z-domain is a linear operation.
If, x1 ( n ) X1(z) with ROC = R1
and x2 ( n ) X2(z) with ROC = R2
ax1(n) + bx2(z) aX1(z) + bX2(z) with ROC containing R1 R2 ...(13)
Study Note
If the linear combination is such that there is no pole-zero cancellation, then the ROC will be exactly equal to the
overlap of the individual regions of convergence. But, if the linear combination is such that some zeros are introduced

that cancels poles, then the ROC may be larger. A simple example of this occurs when x1(n) and x2(n) are both of
infinite duration, but the linear combination is of finite duration. In this case the ROC of the linear combination
is the entire z-plane with the possible exception of zero and/or infinity.
Example 7.8
Determine the z -transform and the ROC of the signal,

x ( n ) = a n u (n ) – a n u (n – 1)
Solution 7.8
n
By definition, X (z ) = x( n) z [a n u( n) a n u( n 1)] z n
n n
= a n u( n) z n
a n u( n 1) z n
n n
= ( a z 1 )n ( az 1 )n
n 0 n 1
1 az 1 1
= 1 for az 1 z a
1 az 1 az 1
1
1 az
X (z ) = 1
1 ROC is the entire z-plane
1 az
7.5.2 Time Shifting

If, x(n) X(z) with ROC = R
n0
then, x(n – n0) z X( z) ...(14)
with ROC = R, except for the possible
addition and deletion of z = 0 or z = .
Proof: The z-transform of x(n – n0) is given by
n
Z[x(n – n0)] = x( n n0 ) z
n
A change of variables is performed by letting m = n – n0, which also yields n = m + n0, m = – as

n = – , and m = as n = . Therefore,
( m n0 ) n0 m
Z[x(n – n0)] = x( m) z = z x( m) z
m m
n0
Z[x(n – n0)] = X( z) z
Example 7.9
Using the properties find z-transform of

z-Transform 311
( a ) [ n + 5]
( b ) 2 [3 n + 6] – 5 [2n + 5] + 3 [ n – 2]
( c ) u[ n] – u[ n – 10]
Solution 7.9
( a ) The z-transform of [n] 1.
The [n + 5] 5
z , ROC will be entire z-plane except z .
( b ) The z-transform of [n] is 1. Thus,
2 [3n + 6] – 5 [2n + 5] + 3 [n – 2] can be written as, 2 [n + 2] + 3 [n – 2].
The term 5 [2n + 5] will not exist, it will be zero everywhere.
2 [n + 2] + 3 [n – 2] (2z2 + 3z–2)
Here ROC will be entire z-plane except z = 0 and z = .
1
( c ) The z-transform of u[n] is 1
, so
1 z
10 10
1 z 1 z
u[n] – u[n – 10] 1 1 1
1 z 1 z 1 z
z10 1
The X( z) .
( z 1) z9
Thus it has 9 poles at origin and 9 zeros at circle of radius 1.
ROC of this z-transform will be entire z-plane except origin.
7.5.3 Scaling in z-Domain

then, z0n x( n) z with ROC = z0 R ...(15)

X
z0
The notation z0 R implies that the ROC R is scaled by z0 . If R is a z b, then the new ROC is
z0 a z b z0 . Also, if X(z) has a pole (or zero) at z = a, then X(z/z0) has a pole (or zero) at z = z0a. This
indicates that the poles and zeros of X(z) have their radii changed by z0 , and their angles are changed by
{z0}. If z0 has unit magnitude, then the radius is unchanged; if z0 is positive real number, then the angle is
unchanged.
Proof: The z-transform of z0n x( n) is given by
n
z
n
Z[ z0 x( n)] = [z0n x( n)] z n
x( n)
n n z0
z
Z[ z0n x( n)] = X
z0
Example 7.10
Find the z-transform of

( a ) ej 0n ( b ) ej 0n u[ n]

( c ) cos( 0 n ) u [ n ] ( d ) sin( 0 n ) u [ n ]
( e ) a n cos( 0 n ) u [ n ] ( f ) a n sin( 0 n ) u [ n ]
Solution 7.10
(a) x[n] = e j 0 n
we can see that magnitude of x[n] will remain constant 1 for – < n < , thus we cannot find
z-transform of this signal. Thus we cannot find z-transform of sin( 0n) and cos( 0n).
( b ) Here, x2[n] = e j 0n u[n],

the z-transform of step function is
1 1
u[n] 1
1 z 1 1/ z
1
ej 0n u[n] j 0 1
1 e z
Similarly
j 0n
1
e u[n] j 0 1
1 e z
In both the above z-transform, poles are at e j 0 and e j 0

and signals are right sided signal, so
z magnitude of pole z 1.
( c ) Here, x3[n] = cos( 0n) u[n]

ej 0n u[ n] e j 0n
u[ n]
x3[n] =
2
1 1 1 1 z 1 cos( 0)
Thus, X 3( z ) = j 1 j 0 =
2 1 e 0z 1 e z 1 1 2z 1 cos( 0) z
2
and ROC is z 1.
( d ) Here, x4[n] = sin( 0n) u[n]
1 j 0n j 0n
we can write x4[n] as, x4[n] = [e e ] u[ n]
2j
1 1 1 z 1 sin( 0)
X 4( z ) = =
2 j 1 ej 0 z 1
1 e j 0 1
z 1 2 z 1 cos 0 z 2
and ROC is z 1.
( e ) Here, x5[n] = an cos( 0n) u[n]
z
The z-transform, X5(z) = X3
a
1 az 1 cos( 0)
X 5( z ) = 1
1 2az cos( 0) a2z 2
and ROC will be z a.

z-Transform 313
( f ) Here, x6[n] = an sin( 0n) u[n]

z
The z-transform, X6(z) = X4
a
az 1 sin( 0 )
X 6( z ) =
1 2az 1 cos( 0 ) a 2 z 2
and ROC will be z a.
Remember: No need to learn how change in signal or z-transform changes ROC, just remember how expression
of z-transform changes and then find ROC using its properties studied in section 7.3.
In this property we have see that,
if, x[n] X (z )
then, an x[n] z
X
a
Thus, if y[n] = an x[n] and if X(z) has poles at P1, P2 and zeros at Z1, Z2 then Y(z) will have pole and zeros
at aP1, aP2, and aZ1, aZ2 respectively.
(z Z1 ) (z Z2)
That is if, X (z ) =
( z P1 ) ( z P2 )
z z
Z1 Z2
z a a
then, Y( z) X =
a z z
( P1 ) P2
a a
j n
For example if y[ n] e 2 x[ n] then poles and zeros of Y(z) can be obtained by multiplying ej /2 with poles
and zeros of X(z). That is 90° phase added to poles and zeros.
7.5.4 Time Reversal

1 1
then, x(–n) X X( z 1 ) with ROC = ...(16)
z R
The time-reversal, or reflection corresponds to replacing z by z–1. Hence if R is of the form a z b,
1 1 1
ROC of the reflected signal is a b or z .
z b a
Proof: The z-transform of x(–n) is given by
n
Z[(x(–n)] = x( n) z
n
A change of variables is perform by letting m = –n, which also yields m = as n = – , and m = – as

n = . Therefore,

m
1
Z[x(–n)] = x( m) zm x( m)
m m z
1
Z[x(–n)] = X X( z 1 )
z
An interesting consequence of the time-reversal property is that, if x(n) is real and even, i.e.,
if x(n) = x(–n)
1
then, X(z) = X( z 1 ) X ...(17)
z
In time reversal property we have seen that if z-transform of x(n) is X(z) then,
x[–n] X(z–1)
Thus if a signal x[n] is even that is
x[n] = x[–n]
1
then, X(z) = X(z–1) or X( z) X
z
This is only possible if X(z) has pole at a then it must have pole at 1/a also. Similar condition exist for
zeros also.
Note: Thus if signal is even then its z-transform will have poles and zeros in pairs of (P1, 1/P1), (Z1/1/Z1) respectively.
If a signal x[n] is odd that is

x[n] = –x[–n]
thus, X(z) = –X(z–1)
1
or X (z ) = X
z
Since signal is odd
X(1)= x (n) =0
n
Thus for odd signals the z-Transform will have zero at unit circle. Thus rest of the poles and zeros will
satisfy the condition that if pole is at P1 then there will be pole at 1/P1 and zero at Z1 then zero will be at 1/Z1.
7.5.5 Differentition in the z-Domain

dX( z)
then, nx(n) with ROC = R
z ...(18)
dz
Multiplication by n in the time domain corresponds to differentiation with respect to z and multiplication
of the result by –z in the z-domain. This operation does not change the location of the poles and hence, the ROC
remains same.
Proof: The z-transform of x(n) is given by
n
X (z ) = x( n) z
n
Differentiating both sides of the above equation, we obtain

z-Transform 315
dX( z) dz n n 1
= x( n) [ nx( n)] z
dz n dz n
1 n
= [ nx( n)] z
zn
dX( z) [ nx( n)] z n

Z[ nx( n)]
z =
dz n
Example 7.11
Find the z-transform of g [ n ] = na n u [ n ].

Solution 7.11
1
We know that, z-transform of an u[n] is 1
.
1 az
1
Thus, an u[n] 1
1 az
d 1
na n u[n] z
dz 1 az 1
z( a ) ( 1 z 2 ) az 1
=
(1 az 1 ) (1 az 1 )2
az 1
Thus, nan u[n]
(1 az 1 )2
The poles are at z = a and signal is right sided. Thus ROC is z a.

We can manipulate to get,
a
(n + 1) an + 1 u[n + 1]
(1 az 1 )2
1
(n + 1) an u[n + 1]
(1 az 1 )2
Since, (n + 1) an u[n + 1] will be zero at n = –1, thus we can say
1
(n + 1) an u[n] ; z a
(1 az 1 )2
Example 7.12
Find the z-transform of g [ n ] = n a n .
Solution 7.12
The signal can be written as,

na n , n 0
g[n] = n
na , n 0
g[n] = nan u[n] na n u[–n – 1]
az 1
The z-transform of nan u[n] ; z a
(1 az 1 )2
We can write –na–n u[–n – 1] as –na–n u[–n] because at n = 0 this will be zero.
az 1
Thus if x[n] = nan u[n] and X( z)
(1 az 1 )
Then, x[–n] X(z–1)
az
Thus, –na–n u[–n]
(1 az )2
1
So, pole is at 1/a and signal is left sided. So ROC is z .
a
z-Transform of g[n] is
1
G(z) = X(z) + X(z–1), a z
a
The condition for existence of G(z) is that a 1.
7.5.6 Time Expansion

The continuous-time concept of time scaling does not directly extend to discrete-time, since the discrete-
time index is defined only for integer values. However, the discrete-time concept of time expansion can be
defined and does play an important role in discrete-time signal and system analysis. Let ‘m’ be a positive
integer, and define the signal,
n
x , if n is a multiple of m
x(m) (n) = m ...(19)
0, if m is not a multiple of m
xm(n) can be obtained from x(n) by placing m – 1 zeros between successive values of the original signal.
Intuitively, we can think of x(m)(n) as a slowed-down version of x(n). Now, if
x(n) X(z) with ROC = R
then, x(m) (n) X(zm) with ROC = R1/m ...(20)
m
That is, R is a z b, then the new ROC is a z b, or a1 / m z b1 / m. Also, if X(z) has a pole
(or zero) at z = a, then X(zm) has a pole (or zero) at z = a1/m.
Proof: The z-transform of x(m) (n) is given by,
n n n
Z[x(m)(n)] = x ( m )( n )z x z
n n m
A change of variables is performed by letting r = n/m, which also yields r = – as n = – and r = as

z-Transform 317
n = . Therefore,
mr
Z[x(m)(n)] = x( r ) z x( r ) ( zm ) r
X( zm )
r r
Example 7.13
Find the z -transform and ROC of the signal,

n an / 3, n 0, 3, 6....
g(n) = an / 3
3 0, elsewhere
where, a 1.
Solution 7.13
n n
Consider the given signal, g (n ) = a n / 3 u x x(3) ( n)
3 3
where, x(n) = an u(n), and its z-transform is given by
1
an u(n) 1
, z a
1 az
Now, using the time-expansion property, we obtain
x(3) (n) X( z3 )
n 1 1/3
an / 3 3
, z a
3 1 az
Example 7.14
Find the z -transform and ROC of the signal,
n an , n 0, 10, 20....
g ( n ) = a nu
10 0, elsewhere
where, a 1.
Solution 7.14
n n n
Consider the given signal, g (n ) = a n u a10 n /10 u x x(10)( n)
10 10 10
where, x(n) = (a10)n u(n)
and its z-transform is given by
1 10
(a10)n u(n) 10 1
, z a
1 a z
Now, using the time-expansion property, we obtain
x(10)(n) X(z10)
n 1 10 /10
a10 n /10 u 10 10
, z a
10 1 a z

n 1
an u 10 10
, z a
10 1 a z
In case of time expansion we have see that if X(z) is z-transform of x[n] then,
n
y[ n] x X (z m )
m
Thus if X(z) has poles at P1, P2 and zeros at Z1, Z2 then Y(z) will have poles at P11 / m , P21 / m and zeros at
Z11 / m, Z12 / m .
(z Z1 ) (z Z2)
If, X (z ) =
( z P1 ) ( z P2 )
(z m Z 1 ) (z m Z2 )
then, Y(z) = X ( z m ) m
(z P1 ) ( z m P2 )

If, x1( n )
X1(z) with ROC = R1
and x2( n )
X2(z) with ROC = R2
then, x1(n) x2(n)
X1(z) X2(z) with ROC containing R1 R2 ...(21)
The z-transform maps the convolution of two signals into the product of their z-transform.
Proof: The z-transform of x1(n) x2(n) is given by
n
Z[x1(n) x2(n)] = [x1( n) x2 ( n)] z
n
n
= x1( m) x2 ( n m) z
n m
By interchanging the order of summation, we have
n
Z[x1(n) x2(n)] = x1( m) x2 ( n m) z
m n
Applying the time-shifting property, the bracketed term is X2(z) z–m.

Substituting this into the above equation yields,
m m
Z[x1(n) x2(n)] = x1( m) ( X2 ( z)z ) X2 ( z) x1( m) z
m m
Z[x1(n) x2(n)] = X2(z) X1(z) = X1(z) X2(z)

the z-domain. Convolution of the two signals, using z-transform, requires the following steps.
1. Compute the z-transforms of the signal to be convolved, i.e.,
X1(z) = Z[x1(n)]

z-Transform 319
X2(z) = Z[x2(n)]
2. Multiply the two z-transform, i.e.,
X(z) = X1(z) X2(z)
3. Find the inverse z-transform of X(z), i.e.,
x(n) = Z–1 X(z)]
Here also ROC of X1(z) X2(z) will be R1 R2 only when there is no pole zero cancellation. If pole-zero
cancellation take place then ROC can be greater than this range.
Example 7.15
Evaluate the convolution of a unit step function u ( n ) with itself using z-transform.
Solution 7.15
The convolution of a unit step function u(n) with itself can be expressed as
x(n) = u(n) u(n)
Using the convolution property, we obtain
Z[x(n)] = Z[u(n) Z[u(n)]
2
1 1 1
X (z ) = 1 1 1
1 z 1 z 1 z
The inverse z-transform yields equation,
x(n) = (n + 1) u(n)
Example 7.16
If y [ n ] = x 1[ n ] x2[ n ] then find Y (z ) and it’s ROC if z -transform of x1[ n ] and x 2 [ n ] are
( z 5)
X1(z ) = , 7 z 9
( z 7) ( z 9)
( z 7)
and X2(z ) = , 4 z 5
( z 4) ( z 5)
Solution 7.16
Here, Y(z) = X1(z) X2(z)
and pole zero cancellation take place thus ROC will not be equal to R1 R2.
1
Y(z) = , 4 z 9
( z 4) ( z 9)
7.5.8 Correlation Property

If, x1 ( n ) X1(z) with ROC = R1
and x2 ( n ) X2(z) with ROC = R2
then, rx1 x2 (n) = x1(n)*x2( n)
= x 1(m) x 2 (m n ) R x1 x 2 ( z )
m

1 1
= X1( z) X2 with ROC containing R1 ...(22)
z R2
Proof: We know that, rx1 x2 (n) = x 1( m ) x 2 ( m n ) x 1( n ) x 2 ( n )

m
Taking the z-transform of the above equation and using the convolution property, we obtain
Z[ rx1 x2 ( m)] = Z[x1(m) x2(–m)
1
Rx1 x2 ( z) = X1( z) X2
z
Example 7.17
Determine the autocorrelation sequence of the signal.

x [ n ] = a n u [ n ], a 1
Solution 7.17
Autocorrelation of x[n] is
rxx = x[n] x[–n]
Thus, Rxx = X(z) X(z–1)
1
X (z ) = 1
, z a
1 az
1 1
and X(z–1) = , z
1 az a
Here X(z) has pole at a and signal as right sided, and X(z–1) has pole at 1/a and signal is left sided. So we
have chosen ROC accordings.
Rxx = X(z) X(z–1)
1 1 1
= , a z
1 az 1 az a
ROC is only valid when a 1, because if a 1 then ROC will be null.
7.5.9 Accumulation Property

If, x[n] X(z) with ROC = R
n
X( z)
then, y[n] = x( k ) with ROC = { R z 1}
k 1 z1
Proof of this property is done using convolution property. Since,

n 1
y[ n] x(k ) = x[ n] u[ n] X( z) 1
k 1 z
Since ROC of X(z) is R and ROC of u[n] is z 1.
1
ROC of X( z) 1
is { R z 1} .
1 z

z-Transform 321
Example 7.18
Find the z -transform of the signal w[ n ] where,

n
w[ n ] = ak, n 0
k n
and w[ n ] = 0 for n < 0. Assume that a 1.
Solution 7.18
From the given data we can say that,
n
w[n] = ak, n 0
k n
1 n 0 n
k
w[n] = a ak a k
ak 1
k n 0 k n 0
n n
= 2 ak 1 2 a k u[k ] [ n]
k 0 k
If an u[n] is x[n] then w[n] is

n
w[n] = 2 x[k ] [ n]
k
1 2
W(z) = 2 X( z) 1
1 1
1 z (1 az ) (1 z 1 )1
7.5.10 First Difference

then, x(n) – x(n – 1) (1 – z–1) X(z) with ROC at least = R {z 0} ...(23)
A common use of this property is in situations where evaluation of the z-transform is easier for the first
difference than for the original sequence.
Proof:
Given that, x(n) X (z )
x(n – 1) X(z) z–1
Now, using the linearity property, we get
x(n) – x(n – 1) X(z) – X(z) z–1
x(n) – x(n – 1) (1 – z–1) X(z)
7.5.11 Conjugation and Conjugate Symmetry
If, x(n)X(z) with ROC = R

then, x (n )
X (z ) with ROC = R ...(24)
Proof: By definition, the z-transform of x (n) is

*
n n
Z[x (n)] = x (n )z x ( n ) (z )
n n
= [X(z*)]* = X*(z*)
Case-1: If x [ n ] is real i.e. if,
x [n] = x[n]
Thus, X *( z* ) = X ( z )
That is if X(z) has a pole or zero at Z = Z0 then it will have pole at zero at Z Z0 also.
Study Note
Thus when x[n] is real then its z-transform will have complex poles and zeros in conjugate pairs and real poles and
zeros will be single and not in pairs.
Case-2: If x[n] is real and even i.e. if,

x*[n] = x[n] and x[n] = x[–n]
Thus, X*(z*) = X(z) and X(z) = X(z–1)
This means that complex poles and zeros will be in conjugate pairs and also at inverse location that is if
1 1 j
pole (or zero) at Z Z0 re j , , then there will be pole (or zero) at Z Z0 re j
, at Z e , at
Z0 r
1 1
Z e j . Thus a complex pole (or zero) will form a pole constellation (or zero constellation).
Z0 r
1 1 1 1
Z0 re j , Z0 re j
, e j
, ej
Z0 r Z0 r
We can see that if the pole (or zero) is real then if will be at Z = Z0 and Z = 1/Z0 only.
Case-3: If x[n] is real and odd i.e. if,

x [n] = x[n] and x[n] = –x[–n]
Thus, X ( z ) = X(z) and X(z) = –X(z–1)
j
Thus, if X(z) has a pole (or zero) at Z = Z0 = rej , then there must be a pole (or zero) at Z Z0 re ,
1 1 j 1 1 j
at Z e ,Z e . Thus here also we get a pole (or zero) constellation.
Z0 r Z0 r
7.6 z-Transform of Causal Periodic Signals

The z-transform of a causal periodic signal can be determined from the knowledge of the z-transform of
its first cycle (period). Consider a causal periodic signal x(n) with period N0 as shown in Fig. 7.5. Let x1(n),
x2(n), x3(n),...., be this signals representing the 1st, 2nd, 3rd,...., cycles of a causal periodic signal x(n). Therefore,
the causal periodic signal x(n) can be written as
x(n) = x1(n) + x2(n) + x3(n) + x2(n) +.....

z-Transform 323
x(n)
x 1(n) x 2(n) x 3(n)
n
0 N0 – 1 N0 2N0 – 1
Fig. 7.5 : Causal periodic signal
= x1(n) + x1(n – N0) + x1(n – 2N0) + x1(n – 3N0) +....
= x1( n kN0 )
k 0
Assume x1(n) X1(z). Using the time-shifting property, the z-transform of the above equation becomes,
N0 2 N0 3 N0
X(z) = X1( z) Z1( z) z X1( z) z X1( z) z ....
N0 2 N0 3N0
= X1( z) [1 z z z ....]
mN0 N0 m
= X1( z) z X1( z) (z )
m 0 m 0
1 N0
X(z) = X1( z) N0
, z 1 z 1
1 z
X1( z)
= , z 1 ...(25)/
1 z N0
Example 7.19
Consider the signal shown in figure (a). The signal repeats periodically it a period N 0 = 4
for n 0 and is zero for n < 0. Find the z-transform of this signal along with the ROC.
x(n) x 1(n)
2 2
1 1
n n
0 1 2 3 4 5 6 7 8 9 10 11 12 0 1 2 3
(a) (b)
(a) Causal periodic signal (b) its first cycle

Solution 7.19
The given signal x(n) can be expressed as,
x(n) = x1( n kN0 )

k 0
where N0 = 4, and x1(n) is shown in figure (b), which can be expressed as,
x1(n) = (n – 1) + 2 (n – 2) + (n – 3)
Taking the z-transform of the above equation, we obtain

X1(z) = z–1 + 2z–2 + z–3

= z–1 [1 + 2z–1 + z–2]
We know that the z-transform of a causal periodic signal x(n) is defined as,
X1( z) z 1[1 2z 1 z 2 ]
X (z ) = , z 1
1 z N0 1 z4
Property Sequence Transform ROC

x [n] X(z) R
x 1[n] X1(z) R1
x 2[n] X2(z) R2
Linearity a 1x 1[n] + a2 x2[n] a 1X1(z) + a 2X2(z) R R1 R2
Time shifting x [n – n0 z–n 0 X(z) R R {0 < z < }
z
Multiplication by z n0 z n0 x [n] X R = z R
z0
j 0n j 0n –j 0
Multiplication by e e x [n] X(e z) R R
1 1
Time reversal x [–n] X R
z R
dX( z)
Multiplication by n nx [n] z R R
dz
n
x[ n] 1
Accumulation 1
X( z) R R { z > 1}
k 1 z
Convolution x 1[n] x 2[n] X1[z] X2(z) R R1 R
Table-7.1 : Some properties of the z-transform
7.7 Inverse of z-Transform

There are three methods that are used for evaluation of inverse z-transform:
1. Contour integration methods (or residue method): This method is not used and will not be discussed.
2. Long division method.
3. Partial fraction method.
7.7.1 Contour Integration Method

The inverse z-transform is formally given by,
1
x(n) = X (z ) z n 1 dz ...(26)
2 jc
where the integral is a contour integral over a closed path C that encloses the origin and lies within the
region of convergence of X(z). The contour is any circle of radius R in the region of convergence.
7.7.2 Long Division Method

From the formulae of z-transform we have,

z-Transform 325
n
X (z ) = x[ n] z
n
That is if we expand the expression we will get,

X(z) = .... x[–2] z2 + x[–1] z1 + x[0]
+ x[1] z–1 + x[2] z–2 ..... ...(27)
Thus z-transform can be written as power series expansion where signal value x[n] is coefficient of z–n in
expansion. Here we can see that if the signal is right sided that is it exist for n 0 then it will have negative
power of z and if x[n] is left sided signal that is it exist for n < 0 then it will have positive of z. There are three
cases.
Case-1: When ROC of X ( z ) is entire z -plane except z = 0 or z = .
In this case we can directly see that X(z) can be compared with equation (27) and we can find
x[n]. Here x[n] will be non-zero only for finite number of n.
Case-2: When ROC of X(z) is z a.
This means that X(z) has a pole at finite location and since ROC is of the form z a, x[ n] is
right sided signal and X(z) will have negative power of z2 the X(z) will be of the form.
Numerator N( z)
X (z ) = ...(28)
Denominator D( z)
The N(z) will be divided by D(z) to get consecutive negative powers of z.
Case-3: When ROC of X ( z ) is z a.
This means that X(z) has a pole at finite location and since ROC of the form z a, x[ n] is left
sided signal and X(z) will have positive power of z, the X(z) will be of the form,
Numerator N( z)
X (z ) = ...(29)
Denominator D( z)
The N(z) will be divided by D(z) to get consecutive positive power of z.
REMEMBER • The long division method will not given the exact expression of x[n]. We can see that this
method can be helpful in finding value of x[n] at a particular value of n.
• For example we have X(z) and it’s ROC is given and for any right sided signal we have to find
value of x[n] at n = 3 then we will do long division in X(z) and our answer will be coefficient
of z–3. Similarly for any left sided signal we have to find value of x[n] at n = –3 then we have
to find coefficient of z+3.
Example 7.20
Find the inverse z -transform of

1 z 1
X( z )
1 1
1 z
3
when,
1
( a ) ROC: z .
3

1
( b ) ROC: z , using a power series expansion.
3
Solution 7.20
1 z1
Given that, X (z ) =
1 1
1 z
3
( a ) Since the ROC is z 1/ 3, we express X(z) as a power series in z–1, so that we obtain a right sided
signal. We divided the numerator by the denominator to obtain,
4 1 4 2 4 3
1 z z z ...
3 9 27
1 1
1 z 1 + z–1
3
1 1
1 z
3
4 1
z
3
4 1 4 2
z z
3 9
4 2
z
9
4 2 4 3
z z
9 27
4 3
z
27
4 1 4 2 4 3
We can write, therefore, X (z ) = 1 z z z ....
3 9 27
Comparing with, X(z) = x(0) + x(1) z–1 + x(2) z–2 + x(3) z–3 +....
4 4 4
we obtain, x(0) = 1, x(1) , x(2) , x(3) ,....
3 9 27
or equivalently, we may express x(n) as,
4 4 4
x(n) = 0, , , ,.....
3 9 27
( b ) Since the ROC is z 1/ 3, we express X(z) as a power series in z, so that we obtain a right sided signal.
We divided the numerator by the denominator to obtain,
2
–3 – 12z – 36z +.....
1 1 –1
z 1 z +1
3
z–1 – 3
4
4 – 12z
12z
2
12z – 36z
2
36z
We can write, therefore, X(z) = –3 – 12z – 36z2 + .....
= +(–36) z2 + (–12)z + (–3)
Comparing the above equation with,
X(z) = ....+ x(–2)z2 + x(–1)z + x(0)

z-Transform 327
we obtain, x(–2) = –36, x(–1) = –12, x(0) = –3

x(n) = {...., 36, 12, 3}
Example 7.21
Determine the inverse z-transform of

X ( z ) = log(1 + a z –1 ), z a
using the power series expansion.
Solution 7.21
According to the logarithmic series expansion,
1 x2 x3
log(1 + x) = ( x )n x ....
n 1 n 2 3
1 ( az 1 )2 ( az 1 )3
Therefore, X(z) log(1 + az–1) = az ....
2 3
1
= ( az 1 )n , z a
n 1 n
1
= ( a )n z n , z a
n 1 n
1
= ( 1)n 1
a n z n, z a
n 1 n
1
X (z ) = ( 1)n 1
a n u( n 1) z n , z a
n n
n
X (z ) = x( n) z
n
1
we get, x(n) = ( 1)n 1
a n u( n 1)
n
Example 7.22
Determine the inverse z -transform of

X ( z ) = e z with ROC all z except z
Solution 7.22
According to the power series expansion of ex, we have
xn
ex =
n 0 n!

zn 0
z n
Therefore, X(z) = ez =
n 0 n! n ( n)!
1 n
= u( n) z
n ( n)!
n
X (z ) = x( n) z
n
1
we get, x(n) = u( n)
( n)!
Example 7.23
Find the inverse z -transform of X ( z ) = sin( z ).

Solution 7.23
According to the power series expansion of sin(x), we have
x3 x5 x7
sin(x) = x .....
3! 5! 7!
z3 z5 z7
Therefore, X(z) = sin(z) = z .....
3! 5! 7!
1 7 1 5 1 3
= .... z z z z
7! 5! 3!
Comparing the above equation with
X(z) = ....+x(–7)z2 + x(–5)z5 + x(–3)z3 + z
1 1
we obtain, x(–7) = , x( 5) ,
7! 5!
1
x(–3) = , x( 1) 1
3!
1 1 1
x(n) = ..., , 0, , 0, , 0, 1, 0
7! 5! 3!
Study Note
While performing long division method on X(z) there are two different ways in which it will be done. For right sided
signal the long division can be performed by writing N(z) and D(z) of X(z) in decreasing power of z and for left sided
signal N(z) D(z) should be written in increasing power of z.
7.7.3 Partial Fraction Expansion Method

If X(z) in rational function, that is

z-Transform 329
N (z ) bM z M bM 1 z M 1....b0
X (z ) = ...(30)
D (z ) aN z N aN 1 z N 1....a0
Here we have to find the inverse z-transform of X(z), ROC of X(z) is given and according to that x[n] will
be found. First of all we will check that X(z) is proper or not, that is (M < N) or not. If M < N then we can
proceed else we will divide N(z) by D(z) and will write X(z) as,
Remainder
X(z) = Quotient ...(31)
D (z )
Remainder N( z)
Quotient of X(z) can be handled easily and we will proceed with as to get result.
D(z) D( z)
Now we have two cases:
Case-1: Roots of D ( z ) are distinct.
Since roots of D(z) are poles of X(z), so this is the case of distinct poles of X(z).
N( z)
Now we have, X (z ) =
D( z)
X( z) N (z )
we will solve for = z D (z )
z
let the poles of D(z) are P1, P2, P3.....PN thus,
X( z) N (z )
= ...(32)
z z (z P1 ) ( z P2 ) ( z P3 )......( z PN )
we can write above equation as,
X( z) C C1 C2 CN
= 0 ..... ...(33)
z z z P1 (z P2 ) ( z PN )
The value of Ck can be found by
X (z )
C0 = z X (z ) z 0
...(34)
z z 0
X (z ) (z Pk )
and Ck = ...(35)
z z Pk
z z CN z
Then we can get, X (z ) = C 0 C 1 C2 ....
z P1 z P2 z PN
C1 C2 CN
= C0 1 1
.... 1
...(36)
1 P1 z 1 P2 z 1 PN z
Each pole in expression of X(z) will correspond to two signals. That is pole at z = Pk:
Ck
• 1
can represent Ck(Pk)n u[n] if ROC of X(z) is z Pk in z-plane.
1 Pk z
Ck
• 1
can represent –Ck(Pk)n u[–n – 1] if ROC of X(z) is z Pk in z-plane.
1 Pk z

REMEMBER Each pole in X(z) will correspond to two signals, if ROC of X(z) is z Pk then it will correspond
to right sided signal. Similarly if ROC of X(z) is z Pk then it will correspond to left sided
signal.
Study Note
X( z)
In above analysis we have X(z) in terms of z and we solved for because then only we can get general
z
Ck
expression of 1
that can be easily converted to n-domain.
1 Pk z
If in any question the z-transform is

M
N( z) b0 b1 z 1 ....bM z
X (z ) = N ...(37)
D( z) a0 a1 z 1 ....a N z
Then for M < N the function will be proper, and for M N the function will be improper. For proper
function we can directly move to second step but for improper function we have to perform first step.
The first step is convert improper function into proper.
Remainder
X(z) = Quotient
D (z )
Remainder
are we move to second step with . Now we have a case where D(z) has distinct roots and we
D(z)
will solve X(z) only here.
N (z ) N (z )
X (z ) =
D (z ) (1 P1z 1
) (1 P2 z 1 ).....(1 PN z 1 )
C1 C2 CN
X(z) = 1 1
.... 1
...(38)
1 P1z 1 P2 z 1 PN z
and Ck = X ( z ) (1 Pk z 1 ) ...(39)
z Pk
Study Note
We can solve for inverse z-transform using any method discussed above. We can change rational function X(z)
written with negative power of z into positive power of z and solve similarly we can change rational function X(z)
written in positive power of z into negative power of z and solve.
Example 7.24

1 2
1 z z
X(z) = with ROC 1 z 2
1 1 1 1
1 z (1 2 z ) (1 z )
2
Solution 7.24

z-Transform 331
1 2
1 z z
1 1
1 z (1 2z 1 ) (1 z 1 )
2
For simplification, we eliminate negative powers of z by multiplying both the numerator and denominator of
the above equation by z3. This result is
z3 z2 z
X (z ) =
1
z ( z 2) ( z 1)
2
X( z) z2
z 1
=
z 1
z ( z 2) ( z 1)
2
We use partial fraction expansion to write,
X( z) A1 A2 A3
=
z 1 z 2 z 1
z
2
1 X( z)
where, A1 = z 1
2 z z 1/ 2
X( z)
A 2 = ( z 2) 2
z z 2
X( z)
A 3 = ( z 1) 2
z z 1
X( z) 1 2 2
Therefore, =
z 1 z 2 z 1
z
2
z 2z 2z 1 2 1
X (z ) = 1 1 1 1
z z 2 z 1 1 z 1 1 2z 1 z
2 2
The ROC and the locations of the poles are depicted in figure. The ROC, 1 z 2, is a ring in the z-plane.
The pole of the first term is at z = 1/2. The ROC has a radius greater than this pole, so this pole corresponds
to causal (right sided) signal. Therefore,
n
1 1
u( n)
2 1 1
1 z
2
The second term has a pole at z = 2. Here the ROC has a radius less than this pole, so this pole corresponds
to the anticausal (left sided) signal. Therefore,
2
2(2 n ) u( n 1) 1
1 2z
The third term has a pole at z = 1. Here the ROC has a radius greater than this pole, so this pole corresponds
to causal (right sided) signal. Therefore,

2
2 u ( n) 1
1 z
Im{z}
0 1/2 1 2 Re{z}
Figure : ROC and location of poles
n
1
and hence, we obtain, x(n) = u( n) 2(2)n u( n 1) 2u ( n)
2
Example 7.25
1
Determine the inverse z -transform of X( z) 1 2
.
1 1.5z 0.5 z
Solution 7.25
Here ROC of X(z) is not given, the poles of X(z) are z = 1 and z = 0.5 that is
1
X (z ) = 1 1 2
1 0.5z z 0.5z
1 2 1
= 1 1 1 1
(1 0.5z ) (1 z ) 1 z 1 0.5z
Since ROC is not given and there are two poles in X(z), to find out all possibilities of ROC we draw circle
of z 0.5 and z 1.0 in z-plane. So we can see that whole plane in divided into three regions. For
or
• R1 = z 0.5 both the poles z = 0.5 and z = 1.0 have radius greater than ROC R1 thus both poles give
left sided signal.
2 Inverse
Thus, 1 z-Transform
2 u[ n 1]
1 z
1
1
(0.5)n u[ n 1]
1 0.5z
Thus, x[n] = –2 u[–n – 1] + (0.5)n u[–n – 1]
• Similarly for R2 0.5 z 1.0
Pole z = 0.5 has radius less than ROC and Pole z = 1.0 has radius greater than ROC. Thus z= 0.5
correspond to right sided and z = 1.0 correspond to right sided and
2
1
2u [ n 1]
1 z

z-Transform 333
1
and 1
(0.5)n u [ n]
1 0.5z
x[n] = –2u[n – 1] –(0.5)n u[n]
• Similarly for R3 z 1.0 both the poles have radius less than ROC, so both give right sided signals.
2
Thus, 1
2u [ n]
1 z
1
1
(0.5)n u [ n]
1 0.5z
x[n] = 2u[n] – (0.5)n u[n]
Case-2: When roots of D ( z ) are not of single order.

That is the poles of X(z) are not of single order, then whole solution is same as that of previous case. That
is we will find X(z)/z for rational function X(z) written in positive of z and we will solve for X(z) only when X(z)
is written negative power of z. The only change will be in finding partial fraction. That is if
X (z ) N (z ) N (z )
= ...(40)
z zD ( z ) (z P1 ) ( z P2 )....( z Pi )r ....( z PN )
we assume Pi is rth order pole then,
X (z ) C 0 C1 C2 1 2 1 r CN
= ...... ..... r
.....
z z z P1 z P2 z Pi (z Pi ) 2
(z Pi ) 3
(z Pi ) ( z PN )
we can see that for all single order poles the expression remain same only for non-single order pole we
have some changes. Here,
C 0 = X( z) Z 0
...(41)
X (z )
Cn = (z Pk ) ...(42)
z z Pk
1 dk X (z )
r–k=
(z Pi )r ...(43)
k ! dz k z z Pi
For converting this X(z) we will follow same procedure that each pole will correspond to a right sided or
left sided signal depending as ROC.
Example 7.26
Determine the causal signal x ( n ) having the z -transform,

1
X( z )
(1 z ) (1 z 1 )2
1
Solution 7.26

1
(1 z ) (1 z 1 )2
1
For simplification, we eliminate negative powers of z by multiplying both the numerator and denominator of
the above equation by z3. This results in
z3
X (z ) =
(1 z) ( z 1)2
X( z) z2
=
z ( z 1) ( z 1)2
We use partial fraction expansion to write,
X( z) A1 A2 A3
=
z z 1 z 1 ( z 1)2
X( z) 1
where, A 1 = ( z 1)
z z 1 4
d X( z) 3
A2 = ( z 1)2
dz z z 1 4
X( z) 1
A 3 = ( z 1)2
z z 1 2
X( z) 1/ 4 3 / 4 1/ 2
Therefore, =
z z 1 z 1 ( z 1)2
1 z 3 z 1 z
X (z ) =
4z 1 4z 1 2 ( z 1)2
1 z 3 z 1 z1
= 1 1
41 z 41 z 2 (1 z 1 )2
Taking the inverse z-transform of the above equation yields,

1 3 1
x(n) = ( 1)n u( n) u( n) nu( n)
4 4 2
1 3 1
= ( 1)n n u( n)
4 4 2
Example 7.27
z3 10 z2 4 z 4
Find the inverse z-transform of X( z) with ROC z 1.
2 z2 2 z 4
Solution 7.27
Since the order of the numerator polynomials is greater than the order of the denominator polynomial, the
given rational z-transform is improper. We use long division to express X(z) as the sum of a proper rational
function and a polynomial in z,

z-Transform 335
0.5z – 4.5
2 2
2z – 2z – 4 z3 – 10z – 4z + 4
2
z3 – z – 2z
–9z2 – 2z + 4
–9z2 + 9z + 18
–11z – 14
11z 14 5.5 z 7
Thus, we may write, X(z) = 0.5 z 4.5 = 0.5z 4.5
2z 2
2z 4 z2 z 2
5.5z 7
X(z) = 0.5z 4.5
( z 1) ( z 2)
Using the partial fraction expansion to expand the rational function, we obtain
0.5 6
X(z) = 0.5z 4.5
z 1 z 2
1 1
0.5z 6z
= 0.5z 4.5 1 1
1 z 1 2z
1 1
0.5z 6z
Also, because X(z) = 0.5z 4.5 1 1
1 z 1 2z
The inverse z-transform yields, x(n) = 0.5 (n + 1) – 4.5 (n) – 0.5(–1)n – 1 u(–n) + 6(2)n – 1 u(–n)
Study Note
If in an equation X(z) is given and we have to find number of time domain signals x[n] that will correspond to X(z).
Then simply find poles of X(z) and draw circles for all poles z Pk and find out the number of regions z-plane is
divided. The number of region in which z-plane is divided is answer.
7.8 The System Function of Discrete Time LTI Systems

The LTI systems of discrete time domain are expressed in difference equation format,
N N
a k y[ n k ] = bk x[ n k ] ...(44)
k 0 k 0
The z-transform of above equation will be

N M
k k
ak Y (z ) z = bk X (z )z ...(45)
k 0 k 0
M
k
bk z
Y( z) k 0
= N ...(46)
X( z) k
ak z
k 0
The ration of z-transform of output and input is called transfer function of the system.

7.8.1 Impulse Response, Transfer Function and Step Response

The impulse response h(n) is defined as the output of an LTI system due to a unit impulse signal input
(n) applied at time n = 0. Also,
h(n) H(z) ...(47)
We know that, the output y[n] of a discrete-time LTI system equals the convolution of the input x[n] with
the impulse response h[n],
y[n] = x[n] h[n] ...(48)
Applying the convolution property of the z-transform, we obtain
Y(z) = X(z) H(z) ...(49)
where Y(z), X(z) and H(z) are the z-transform of y[n], x[n], and h[n], respectively.
Y( z)
Thus, H(z) = ...(50)
X( z)
The z-transform H(z) of h[n] is referred to as the system function (or the transfer function) of the
system.
h[n]
x [n] y [n] = x [n] h[n]
X(z) Y(z) = X(z) H(z)

H(z)
Fig. 7.6 : Impulse response and System function
The step response s(n) is defined as the output of an LTI system due to a unit step input signal, i.e.
x(n) = u(n). The step response of an LTI system with impulse response h(n) is given by
s(n) = h(n) x(n) = h(n) u(n)
= h( m) u( n m)
m
n
s(n ) = h( m )
m
n
Therefore, Z[s(n)] = Z h(m)
m
Using the accumulation property, we obtain

H( z)
S(z ) = ...(51)
1 z1
n n 1
Conversely, s(n ) = h( m ) h( n ) h( m )
m m
s( n ) = h(n) + s(n – 1)
h(n) = s(n) – s(n – 1)
Z[h(n)] = Z[s(n) – s(n – 1)]
H(z) = [1 – z–1] S(z) ...(52)

z-Transform 337
REMEMBER • The output of system that is calculated using transfer function H(z) is always output due to
applied input, or forced response.
• The while finding the transfer function H(z) we can assume that all initial conditions are
zero.
Example 7.28
Determine the transfer function and impulse response for the causal LTI system described
by the difference equation,
1
y ( n) y ( n 1) x( n) 2 x( n 1)
2
Solution 7.28
1
Given that, y ( n) y( n 1) = x(n) + 2x(n – 1)
2
Taking the z-transform of the difference equation, we obtain
1 1
Y( z) 1 z = X(z) [1 – 2z–1]
2
Hence the system function is
Y( z) 1 2z 1 1 2z 1
= H( z)
X( z) 1 1 1 1 1 1
1 z 1 z 1 z
2 2 2
The inverse z-transform of the above equation yields,
n n 1
1 1
h(n) = u( n) 2 u( n 1)
2 2
This is the unit impulse response (or unit sample response) of the system.
7.8.2 Characteristics of Discrete-Time LTI System

The characteristics of discrete-time LTI system can be obtained from H(z) and it’s ROC.
1. Causality
For causal system the condition as impulse response is
h[n] = 0 for n < 0 ...(53)
That is h[n] should be a right sided signal, thus ROC of H(z) should be format, ROC z a. Thus
condition for causal system is that H(z) must be a proper function with ROC of form z a.
Study Note
• Here we should have proper function for causality because for example if,
z4
H(z) = , z 1
z 1

z3
then, H(z) =
1 z1
thus, h[n] = u[n + 3]
Thus even after ROC is z 1 the system will be non-causal.
• Thus whenever H(z) is improper then even when ROC is of format z 1 the system will be non-causal.
• Thus if any question H(z) is given and we have to find h[n] which is causal, then h[n] should be right sided thus
each pole of H(z) should correspond to right sided signal.
2. Stability
For stable system the condition or impulse response is
h[ n] < ...(54)
n
Since H(z) is z-transform of h[n] then ROC of H(z) is values of r where the condition
n
h[ n] r < ...(55)
n
is satisfied. If we compare the above two conditions then we can say that if r = 1 is present in ROC of
H(z) then stability condition will be satisfied. Thus system will be stable only when ROC of H(z) contain
the unit circle.
Study Note
Thus if in any question H(z) is given and we have to find h[n] such that system is stable then we should first of all
draw circles corresponding to each pole (Pk) in z-plane ( z Pk ), and see the option of ROC in z-plane. Then
choose the ROC which contain unit circle. Now write expression of h[n] accordingly.
3. Causal and Stable Systems

For causality and stability we need ROC must contain unit circle and all poles should correspond to
right sided signal. Thus for both the condition to be satisfied together the requirement is that all poles of
H(z) should lie inside unit circle.
Study Note
For causal and stable system first of all the requirement is that all poles of H(z) should remain inside unit circle if
this condition is not satisfied then system cannot be causal and stable. If condition is satisfied then take right sided
signal for each pole and find h(n).
From the above analysis we can see that:

• For anticausal systems i.e. for which h[n] = 0 for n > 0 the ROC of H(z) should be of format z a
and signal should be left sided and H(z) should be proper function.
• For non-causal the system must not follow condition of causal and anticausal system.
• For unstable system ROC of H(z) must not contain unit circle.

z-Transform 339
Example 7.29
For the following system functions, check whether the corresponding LTI system is causal,
anticausal, or noncausal.
1
3 4z
( a ) H1( z) 1 2
, z 3
1 3.5 z 1.5z
1
3 4z
( b ) H2 ( z ) 1 2
, z 0.5
1 3.5 z 1.5 z
1
3 4z
( c ) H3 ( z ) 1 2
, 0.5 z 3
1 3.5z 1.5 z
Solution 7.29
( a ) Consider the given system function,
1
3 4z
H1(z) = 1 2
, z 3
1 3.5z 1.5z
1
3 4z 1 2
= =
(1 0.5z ) (1 3z 1 )
1
1 0.5z 1
1 3z 1
Since the given system function H1(z) is rational and its ROC is the region in the z-plane outside the
outermost pole, the corresponding LTI system is causal. Also it can be verified by its impulse response.
Given that,
1 2
H1(z) = 1
1 0.5z 1 3z 1
Taking the inverse z-transform, we obtain
h1(n) = (0.5)n u(n) + 2(3)n u(n)
Since, h1(n) = 0 for n < 0, this system is causal.
( b ) Consider the given system function,
1
3 4z
H2(z) = 1 2
, z 0.5
1 3.5z 1.5z
1
3 4z 1 2
= =
(1 0.5z 1 ) (1 3z 1 ) 1 0.5z 1
1 3z 1
Since, the given system function H2(z) is rational and its ROC is the region in the z-plane inside the
innermost pole, the corresponding LTI system is anticausal. Also it can be verified by its impulse
response. Given that,
1 2
H2(z) = 1 1
1 0.5z 1 3z
h2(n) = –(0.5)n u(–n – 1) – 2(3)n u(–n – 1)
Since, h2(n) = 0 for n > 0, this system is anticausal.
( c ) Consider the given system function,
1
3 4z
H3(z) = 1 2
, 0.5 z 3
1 3.5z 1.5z

1
3 4z 1 2
= =
(1 0.5z 1 ) (1 3z 1 ) 1 0.5z 1
1 3z 1
Since the given system function H3(z) is rational and its ROC is a ring in the z-plane, the corresponding
LTI system is noncausal. In can also be verified by its impulse response. Given that,
1 2
H3(z) = 1
1 0.5z 1 3z 1
h3(n) = (0.5)n u(n) – 2(3)n u(–n – 1)
Since, h3(t) is two sided, this system is noncausal.
Example 7.30
An LTI system is characterized by the system function,

1
3 4z
H(z) = 1 2
1 3.5z 1.5 z
Specify the ROC of H ( z ) and determine h ( n ) for the following conditions:
( a ) The system is causal and unstable.
( b ) The system is noncausal and stable.
( c ) The system is anticausal and unstable.
Solution 7.30
1 2
3 4z 3z 4z
Given that, H(z) = 1 2 2
1 3.5z 1.5z z 3.5z 1.5
H( z) 3z 4 3z 4
= 2
z z 3.5z 1.5 (z 0.5)( z 3)
Using the partial fraction expansion, we obtain
H( z) 1 2
=
z z 0.5 z 3
z 2z 1 2
H(z) = 1 1
z 0.5 z 3 1 0.5z 1 3z
This system has poles at z = 0.5 and z = 3.
( a ) For this sytem to be causal and unstable, the ROC of H(z) is the region in the z-plane outside the
outermost pole and it must not include the unit circle. Therefore, the ROC is the region, z 3. Since
the ROC, z 3, is the region in the z-plane outside the outermost pole, all the poles correspond to
causal to causal (right sided) signals. Now, consider
1 2
H(z) = 1 1
, z 3
1 0.5z 1 3z
The inverse z-transform yields,
h(n) = (0.5)n u(n) + 2(3)n u(n)
( b ) For this system to be noncausal and stable, the ROC of H(z) is a ring in the z-plane and it must include
the unit circle. Therefore, the ROC is the region, 0.5 z 3.

z-Transform 341
The pole of the first term is at 0.5. The ROC has a radius greater than the pole at z = 0.5, so this pole
corresponds to causal (right sided) signal and pole at z = 3.0 correspond to ledt sided signal. Therefore,
we obtain
h(n) = (0.5)n u(n) – 2(3)n u(–n – 1)
( c ) For this system to be anticausal and untable, the ROC of H(z) is the region in the z-plane inside the
innermost pole and it must not include the unit circle. Therefore, the ROC is the region z 0.5. Since
the ROC, z 0.5, is the region in the z-plane inside the innermost pole, all the poles correspond to
anticausal (right sided) signals. Now, consider
1 2
H(z) = 1 1
, z 0.5
1 0.5z 1 3z
h(n) = (0.5)n u(–n – 1) – 2(3)n u(–n – 1)
Example 7.31
3(1 z 1 )
Find the inverse z-transform of X( z ) 1 2
, assuming that (a) the signal is causal
1 2.5 z z
and (b) the signal has a DTFT, i.e., x ( n ) is absolutely summable.
Solution 7.31
3(1 z 1 ) 3( z2 z)
Given that, X (z ) = 1 2 2
1 2.5z z z 2.5z 1
X( z) 3(1 z) 3( z 1)
=
z z 2 2.5z 1 ( z 0.5) ( z 2)
Using the partial fraction expansion, we obtain
X( z) 1 2
=
z z 0.5 z 2
z 2z 1 2
X (z ) = 1 1
z 0.5 z 2 1 0.5 z 1 2z
The system has poles at z = 0.5 and z = 2.
( a ) For x(n) to be causal, the ROC of X(z) is the region in the z-plane outside the outermost pole, i.e.,
z 2. Since the ROC, z 2, is the region in the z-plane outside the outermost pole, all the poles
correspond to causal (right sided) signals.
Now, consider
1 2
X (z ) = 1 1
, z 2
1 0.5z 1 2z
x(n) = (0.5)n u(n) + 2(2)n u(n)
( b ) For x(n) to be Fourier transformable, i.e. for x(n) to be absolutely summable, the ROC of X(z) must
include the unit circle. Therefore, its ROC is the region 0.5 z 2. The pole of the first term is at 0.5.
The ROC has a radius greater than the pole at z = 0.5, so this pole corresponds to causal (right sided)

signal. Therefore,
1
(0.5)n u(n) 1
1 0.5z
The second term has a pole at z = 2. The ROC has a radius less than the pole at z = 2, so this pole
corresponds to the anticausal (left sided) signal. Therefore,
2
–2(2)n u(–n – 1) 1
1 2z
and hence, we obtain
x(n) = (0.5)n u(n) + 2(2)n u(–n – 1)
7.8.3 Response of Discrete Time LTI System to Complex Exponential Signals
We have seen that z-transform of e j 0n , sin 0n, cos 0n cannot be found because these signals cannot be
made absolutely summable for any value of r, or these signal never satisfy the condition,
n
x[ n] r
n
Similarly we can say that, a signal Cn for – < n < have no z-transform (where C is any constant).
Now consider case where Cn is applied as input to discrete LTI system.
n LTI system
C y [n]
h[n]
Fig. 7.7 : Discrete LTI system with Cn as input signal
Here, we cannot apply, Y(z) = H(z) X(z)

because we cannot find X(z). So we apply the general rule formulae of discrete LTI system,
y[n] = h[n] x[n]
That is, y[n] = h[ m] x[ n m]

m
y[n] = h[ m] C( n m)
y[n] = Cn h[ m] C m
...(56)
m
n
We know that, H(z) = h[n] z ...(57)
n
Thus if we compare equation (56) and (57) we get that,
y[n] = C n H (z ) ...(58)
z C
Thus when input is Cn then output is input multiplied by value of transfer function H (z ) z C
.

z-Transform 343
Note: Cn is called eigen function and H ( z ) z C is called eigen value for discrete time LTI system.
That is when input is e j 0n .
j 0n LTI system
e ej 0n
H( z) ...(59)
H(z) z ej 0
j 0n
Similarly if input is e then,
–j 0 n LTI system j 0n
e
H(z)
e H( z) ...(60)
z e j 0
If h[n] is real then H(z) = H (z ) i.e. if h[n] is real then,

H( z) z j
ej 0 = H( e 0 )
H( z) z j 0
and e j 0 = H( e )
will have relation, H( e j 0 ) = H*( e j 0

)
Thus if, H( e j 0 ) = H( e j 0 ) e j ,
j
that is it has magnitude H( e 0 ) and phase ( ) then,
j 0 j 0 j 0 j
H( e ) = H (e ) H( e )e
using above equation (60) and (59) and assuming h[n] real we get
ej 0n e j 0n
LTI system
cos[ 0 n] H[z]
2
ej 0n H( z) z ej 0
e j 0n
H( z) z e j 0
Here the output will be .
2
ej 0n H( e j 0
) e j 0n
H( e j 0 )
y[n] =
2
ej 0n H( e j 0 ) e j e j 0n
H( e j 0 ) e j
=
2
j
y[n] = H( e 0 ) cos( 0n )
= H( e j 0 ) cos( 0 n)
REMEMBER • Similarly we can say that if h[n] is real then

j
1. for sin( 0n + ) as input, output will be H( e 0 ) sin( 0n ).
2. for cos( 0n + ) as input, output will be H( e j 0 ) cos( 0n ).

• If it is not specified in question that h[n] is real or not and H(z) is given in question and we
have to find output of e j 0 , cos( 0n ), sin( 0n ) then assume h[n] is real.

7.8.4 Type of System

There are two types of systems:
1 . Finite Impulse Response (FIR): If impulse response h[n] of any filter or system is non-zero only
for finite values of n then it is called FIR filter. Since FIR filters have h[n] a finite duration signal
thus H(z) the z-transform of h[n]) that is transfer function of system will have no poles at finite
obtain.
2 . Infinite Impulse Response (IIR): These are the systems or filters which have h[n] or impulse
response of infinite duration that is H(z) (the z-transform of h[n]) transfer function of system will
have pole at finite location.
7.9 Unilateral (One-sided) z-Transform

The unilateral z-transform of a signal x(n) is defined as,
n
Zu[x(n)] = X( z) x( n) z ...(61)
n 0
We shall denote the relationship between x(n) and X(z) as,

Zu
x( n) X( z) ...(62)
The difference in the definitions of the unilateral and bilateral z-transform lies in the lower limit of the
summation. The bilateral z-transform involves a summation from – to + , while the unilateral z-transform
involves a summation from 0 to + . Due to this choice of lower limit, the unilateral z-transform has the following
characteristics:
1. Two signals that differ for n < 0, but that are identical for n 0, will have different bilateral z-transforms,
but identical unilateral z-transform.
2. Any signal x(n) that is identically zero for n < 0 [i.e., the signal x(n) u(n)] has identical bilateral and
unilateral z-transform.
3. The unilateral z-transform is used in analysing causal systems and, particularly, systems specified
by linear constant-coefficient difference equations with non-zero initial conditions (i.e., systems
that are not initially at rest).
4. For a unilateral z-transform, there is a unique inverse transform of X(z); consequently, there is no
need to specify the ROC explicitly (ROC for a unilateral z-transform must always the exterior of a
circle). For this reason, we shall generally ignore any mention of the ROC for unilateral z-transform.
Method of find unilateral z-transform of any signal is
• to use equation (61) directly on the signal to get result.
• or another method is to find unilateral z-transform of x[n] first of all find a new signal x [n] =
x[n] u[n] and then find bilateral z-transform of x [n] to get unilateral z-transform of x[n].
Example 7.32
Find unilateral z-transform of the signal, x ( n ) = a n + 1 u ( n + 1).

Solution 7.32
n
By definition, Zu[x(n)] = X(z) = x( n) z an 1
u( n 1) z n
n 0 n 0

z-Transform 345
= an 1
z n
a an z n
= a ( az 1 )n
n 0 n 0 n 0
a
Zu[x(n)] = X(z) = 1
1 az
Example 7.33
Determine the unilateral z -transform of each of the following signals:

( a ) x ( n ) = a n u ( n + 1)
( b ) x ( n ) = ( n + 1) + ( n ) + a n + 3 u ( n + 1)
Solution 7.33
n
( a ) We have, Zu[x(n)] = X(z) = x( n) z a n u( n 1) z n
n 0 n 0
a nz n
( az 1 )n = 1
= 1
n 0 n 0 1 az
1
Zu[x(n)] = X(z) = 1
1 az
n
( b ) We have, Zu[x(n)] = X(z) = x( n) z [ ( n 1) ( n) a n 3
u( n 1)] z n
n 0 n 0
n
= ( n 1) z 1
( n) z an 3
u( n 1) z n
n 0 n 0 n 0
n
= 0 z an 3
z n
n 0
n 0
= 1 a
3
( az 1 )n
n 0
a3
X (z ) = 1 1
1 az
7.9.1 Properties of Unilateral z-Transform

Mostly all the properties of bilateral z-transform are valid in unilateral z-transform except few like Time
reversal and Time shifting if shifting take the signal towards n < 0.
1. Linearity
Zu Zu
If, x1( n) X1( z) and x2 ( n) X2 ( z)
Zu
then, ax1( n) bx2 aX1( z) bX2 ( z) ...(63)

2. Scaling in The z-Domain

Zu
If, x( n) X( z)
Zu z
then, z0n x( n) X ...(64)
z0
3. Differentiation in z-Domain
Zu
If, x( n) X( z)
Zu dX( z)
then, nx( n) z ...(64)
dz
4. Time Expansion
Let m be a positive integer, and define the signal,
n
x(m)(n) = m ...(66)
0, if n is not a multiple of m
xm(n) can be obtained from x(n) by placing m – 1 zeros between successive values of the original signal.
Intuitively, we can think of x(m)(n) as a slowed down version of x(n). Now, if
Zu
x( n) X( z)
Zu
then, x( m)( n) X( zm ) ...(67)
5. Conjugation Property
Zu
If, x( n) X( z)
Zu
then, x ( n) X (z ) ...(68)
6. Convolution Property
If, x1(n) = x2(n) = 0 for all n < 0, and if
Zu
x1( n) X1( z)
Zu
and x 2 ( n) X2 ( z)
Zu
then, x1( n) x2 ( n) X1( z) X2 ( z) ...(69)
Study Note
It is important to note that the convolution property for unilateral z-transform applies only if the signals x1(n) and
x2(n) are both zero for n < 0. That is, while we have seen that the bilateral z-transform of x1(n) * x2(n) always
equals to the product of the bilateral z-transform of x1(n) and x2(n), the unilateral z-transform of x1(n) * x2(n) in
general does not equal to the product of the unilateral transforms if either x1(n) or x2(n) is non zero for n < 0.

z-Transform 347
7. Accumulation Property
If, x(n) = 0 for n < 0 and if
Zu X( z)
then, x(k ) ...(70)
k 0 1 z1
Proof: We know that if x(n) = 0 for n < 0, then

n
x ( k ) = x(n) u(n)
k 0
Taking the unilateral z-transform on both sides of the above equation, we obtain
Zu x( k ) = Zu[x(n) u(n)]
k 0
X( z)
=
1 z1
n
Zu X (z )
Therefore, x (k ) 1
k 0 1 z
8. Time Delay Property or Right Shift of Signal

If x[n] is a signal whose unilateral z-transform is X(z) that is
Zu
x[ n] X( z)
n
Thus, X (z ) = x[ n] z
n 0
Here we assumed that x[n] = 0 for n < 0 and while calculating X(z) the unilateral z-transform of x[n] we
have neglected all values of x[n] for n < 0.
Suppose we give a right shift to x[n] by 1 and then find its unilateral z-transform that is,
n n
Zu[x[n – 1] = x[ n 1] z = x [ 1] z 0 x [n 1] z
n 0 n 1
m 1
= x [ 1] ( x [m ] z )z
m 0
= x[–1] + X(z) z–1

Now let us understand what is happening: To find X(z) we have considered x[n] for n 0 only, when x[n]
is shifted by 1 in right side then all value of x[n] shift by 1 and at n = 0 we get x[–1] and when we find
Zu[x[n – 1] then we get a new extra term x[–1] at n = 0 and rest all previous terms shifted by 1 whose Zu
will be X(z) z–1.
General expression will be
n
Z[x[n – n0]] = x[ n n0 ] z
n 0
By changing variable we get,

( m n0 )
Z[x[n – n0]] = x[ m] z
m n0
1
n0 m m
= z x[ m] z x[ m] z
m n0 m 0
1
n0 n0
= z x[ n] zn z X( z) ...(71)
n 1
Zu
Thus, x[ n 2] z 2 X( z) x[ 1] z 1
x[ 2]
Zu
and x[ n 3] z 3 X( z) x[ 1] z 2
x[ 2] z 1
x[ 3]
Pictorially if we look at this then it is easier to understand,
x[n] [..., x [ 2], x [ 1], x [0], x [1], x [2]...]
for Zu [x(n)]
Similarly, x[n – 1] [..., x [ 3], x [ 2], x [ 1], x [0], x [1]...]
for Zu [x(n 1)]
so we get whole x[n] shift by 1 right and new term at n = 0.

So, Zu[x[n – 1] = x[–1] z0 + z–1 X(z)
9. Time Advance Property or Left Shift Property

Zu
If, x[ n] X( z)
n
then, X (z ) = x[ n] z
n 0
If, x[n] = [..., x [ 2], x [ 1], x [0], x [1], x [2]...]

for Zu [x(n)]
then, x[n + 1] = [..., x [ 1], x [0], x [1], x [2], x [3]...]

for Zu [x(n 1)]
So, we can see that, Zu[x[n + 1] has one less than term than X(z) because x[0] has gone to n = –1. Thus
we can say that,
Zu[x[n + 1]] = z X (z ) z x (0)

Because whole Because x [0] which is at
signal shift left by 1 n = 1 has to be removed
Similarly, x[n + 2] = [..., x [0], x [1], x [2], x [3], x [4]...]

for Z u [x(n 2)]
This two terms has to be removed,

Zu[x[n + 2] = {z2 X(z) – z2 x[0] – z–1 x[1]}

z-Transform 349
General formulae is,

n0 1
Zu
x( n n0 ) zn0 X( z) zn0 x( n) z n , n0 0 ...(72)
n 0
By definition, we have
n
Zu[x(n + n0)] = x( n n0 ) z
n 0
A change of variables is performed by letting m = n + n0, which also yields n = m – n0, m = n0 as n =

0, and m = as n = . Therefore,
( m n0 )
Zu[x(n + n0)] = x (m )z z n0 x (m )z m
m n0 m n0
n0 1 n0 1
n m m m
= z0 x( m) z x( m) z x( m) z
m n0 m 0 m 0
n0 1
n m m
= z0 x( m) z x( m) z
m 0 m 0
n0 1
n n
= z 0 X( z) x( n) z
n 0
n0 1
Zu[x(n + n0)] = zn0 X( z) zn0 x( n) z n
n 0
10. First Difference

If x(n) is causal [i.e., x(n) = 0 for n < 0], and if
Zu
x( n) X( z)
Zu
then, x( n) x( n 1) (1 z 1 ) X( z) ...(73)
Proof: Given that,

Zu
x( n) X( z)
Zu 1
x( n 1) z X( z)
Now, using the linearity property, we get
Zu Zu
x( n) x( n 1) X( z) z 1 X( z) (1 z 1 ) X( z)
11. Initial Value Theorem

If x(n) is causal [i.e., x(n) = 0 for n < 0]
then, x(0) = Lim X( z) ...(74)
z
Proof: By definition, we have

n
X (z ) = x( n) z
n
Since, x(n) = 0 for n < 0, we obtain

n
X (z ) = x( n) z
n 0
X(z) = x(0) + x(1) z–1 + x(2) z–2 + ....

1 2
Lim X( z) = Lim [ x(0) x(1) z x(2) z ....] x(0)
z z
Study Note
• Initial value theorem is only applicable for causal sequence.
• In equation (74) X(z) is bilateral z-transform of x[n]. Since sequence is causal bilateral and unilateral z-transform
same.
• For a causal sequence, if x(0) is finite, then Lim X( z) is finite. Consequently, with X(z) expressed as a ratio of
z
polynomials in z, the order of the numerator polynomial cannot be greater than the order of the denominator
polynomial; or, equivalently, the number of finite zeros X(z) cannot be greater than the number of finite poles.
12. Final Value Theorem

The final value theorem states that if Lim x( n) exists and if
n
Zu
x( n) X( z)
then, Lim x( n) = x( ) Lim (1 z 1 ) X( n) ...(75)

n z 1
Remember: The limit in equation (75) exists if all the poles of (1 – z–1) X(z) lie inside the unit circle, i.e., all the
poles of (1 – z–1) X(z) have magnitude less than one.
Proof: Let’s take the unilateral z-transform of the signal x(n) – x(n – 1).
n
Zu[x(n) – x(n – 1)] = [ x( n) x( n 1)] z
n 0
N
n
Zu[x(n)] – Zu[x(n – 1)] = Lim [ x( n) x( n 1)] z
N n 0
N
n
X(z) – [z–1 X(z) + x(–1)] = Lim [ x( n) x( n 1)] z
N n 0
N
n
Lim [(1 z 1 ) X( z) x( 1)] = Lim Lim [x( n) x( n 1)] z
z 1 z 1 N
n 0
N
Lim (1 z 1 ) X( z) x( 1) = Lim [x( n) x( n 1)]
z 1 N n 0
= Lim [ x(0) x( 1) [ x(1) x(0) [x(2) x(1) ....

N

z-Transform 351
+ [x(N – 2) –x(N – 3)] + [x(N – 1) – x(N – 2) + [x(N) – x(N – 1)

Lim (1 z 1 ) X( z) x( 1) = Lim [ x( 1) x( N )]
z 1 N
Lim (1 z 1 ) X( z) x( 1) = x( 1) Lim x( N )
z 1 N
Lim(1 z 1 ) X( z) = Lim x( N ) x( )
z 1 N
Example 7.34
Find the final value of the signals corresponding to the following z -transforms:
1
1 z
( a ) X 1( z ) 2
1 0.25 z
1
2z
( b ) X2 ( z ) 1 2
1 1.8 z 0.8 z
1
( c ) X3 ( z ) 1 2
1 2z 3z
Solution 7.34
1
1 z
( a ) Given that, X 1( z ) = 1
1 0.25z
(1 z 1 ) (1 z 1 )
(1 – z–1) X1(z) =
(1 0.5z 1 ) (1 0.5z 1 )
It has two poles at z = 0.5 and z = –0.5. Note that both the poles of (1 – z–1) X1(z) lie inside the unit
circle the unit circle, therefore,
x1( ) = Lim (1 z 1 ) X1( z)
z 1
(1 z 1 ) (1 z 1 ) (0) (2)
= Lim 1 1
0
z 1 (1 0.5z ) (1 0.5z ) (0.5) (1.5)
1
2z
( b ) Given that, X 2( z ) = 1 2
1 1.8z 0.8z
1
2z (1 z 1 ) 2z 1
(1 – z–1) X2(z) = 1 1 1
(1 z ) (1 1.8z ) 1 0.8z
It has one pole at z = 0.8, which lies inside the unit circle, therefore,
x2( ) = Lim (1 z 1 ) X2 ( z)
z 1
1
2z 2
= Lim 1
10
z 1 1 0.8z 0.2
1
( c ) Given that, X 3( z ) = 1 2
1 2z 3z
(1 z 1 ) 1
(1 – z–1) X3(z) =
(1 z 1 ) (1 3z 1 ) 1 3z 1

It has one pole at z = –3, which lies outside the unit circle therefore, the final value theorem cannot be
used to find x3( ).
Example 7.35
Use the initial value theorem to find the initial value of the signals corresponding to the
following z-transforms:
1
2 z
( a ) X1( z)
(1 z 1 ) (1 0.5 z 1 )
1
1 3z
( b ) X2 ( z )
(1 0.1 z 1 ) (1 0.6 z 1 )
1
0.5 0.25 z
( c ) X3 ( z) 1 2 3
1 1.3 z 0.2 z 0.1 z
Solution 7.35
1
2 z 2 1/ z
( a ) Given that, X 1( z ) = 1 1
=
(1 z ) (1 0.5z ) 1 0.5
1 1
z z
Using the initial value theorem, we have
2 1/ z 2 0
x1(0) Lim = Lim
z z 1 0.5 (1 0) (1 0)
1 1
z z
x1(0) = 2
1
1 3z 1 3/z
( b ) Given that, X 2( z ) = 1 1
(1 0.1 z ) (1 0.6 z ) 0.1 0.6
1 1
z z
Using the initial value theorem, we have

1 3/z 1 0
x2 (0) Lim = Lim = (1 0) (1 0)
z z 0.1 0.6
1 1
z z
x2(0) = 1
( c ) Using the initial value theorem, we have
0.5(0.25 / z) 0.5 0
x 3 (0) = Lim
z 1.3 0.2 0.1 1 0 0 0
1
z z2 z3
x3(0) = 0.5
REMEMBER If in a question we have to find steady state value of output then find Y(z) = H(z) X(z) and then
apply final value theorem on Y(z) and find y( ). The value of y( ) will be valid only when all
poles of (1 – z–1) Y(z) be inside unit circle.

z-Transform 353
7.10 Solving Difference Equation using Unilateral z-Transform

One of the most common applications of the unilateral z-transform is to solve linear constant-coefficient
difference equations subject to nonzero initial conditions. Such equations are used to model discrete-time LTI
systems. Solving of these equations depends on the time-delay and time advance properties of the unilateral
z-transform. As in the case of the Laplace transform with differential equations, the z-transform converts difference
equations into algebraic equations that are radially solved to find the solution in the z-domain. The procedure is
straight forward and systematic, and we summarizes it in the following three steps:
1. For a given set of initial conditions, take the z-transform of both sides of the difference equation to
obtain an algebraic equation in Y(z).
2. Solve the algebraic equation for Y(z).
3. Take the inverse z-transform to obtain y(n).
Example 7.36
A discrete-time system is characterized by the difference equation:

1 1
y( n)
y ( n 1) x( n ) x( n 1)
2 2
Find y ( n ) for n 0 when x ( n ) = u ( n ) and the initial conditions is given as y (–1) = 1.
Or, determine the step response of the system,
1 1
y( n)
y ( n 1) x( n ) x( n 1)
2 2
when the initial condition is y (–1) = 1.
Solution 7.36
For x(n) = u(n), we have
1
X (z ) = 1
1 z
Now, consider the given difference equation,
1 1 1
y ( n) y( n 1) = x( n) x( n 1) u( n) u( n 1)
2 2 2
Taking the unilateral z-transform of the above equation, we obtain
1 1
Zu y( n) y ( n 1) = Zu u( n) u ( n 1)
2 2
1
1 1 1 1/ 2 z
Y( z) [ z Y( z) y( 1)] = 1 1
2 1 z 1 z
1 1
1 z
1 1 1 2
Y( z) 1 z y( 1) = 1
2 2 1 z
1
1
1 2
Y(z) =
1 z 1 1 1
1 z
2

n n 1
1 1 1
y(n) = u( n) u( n) 1 u( n)
2 2 2
Example 7.37
A second-order discrete-time system is characterized by the difference equation,

y (n ) – 3y (n – 1) + 2y (n – 2) = x (n ) – 2x ( n – 1)
Find y ( n ) for n 0 when x ( n ) = u ( n ) and the initial conditions are given as
y (–1) = y (–2) = 1.
Solution 7.37
1
For x(n) = u(n), we have X (z ) = 1
1 z
y(n) – 3y(n – 1) + 2y(n – 2) = x(n) – 2x(n – 1) = 2u(n) – u(n – 1)
Zu[y(n) – 3y(n – 1) – 2y(n – 2)] = Zu[u(n) – 2u(n – 1)]
1
1 2z
Y(z) – 3[z–1 Y(z) + y(–1) + 2[z–2 Y(z) + z–1 y(–1) + y(–2)] = 1 1
1 z 1 z
1
1 2z
Y(z) [1 – 3z–1 + 2z–2] –3y(–1) + 2z–1 y(–1) + 2y(–2)] = 1
1 z
1
2z
Y(z) [1 – z–1) (1 – 2z–1)] – 3 + 2z–1 + 2 = 1
1 z
1 1
1 2z 1 2z
Y(z) =
(1 z 1 )2 (1 2 z 1 ) (1 z 1 ) (1 2 z 1 )
1 1
= 1 2 1
(1 z ) 1 z
y(n) = (n + 1) u(n) + u(n) = (n + 2) u(n)
7.10.1 Zero Input Response and Zero State Response

The z-transform method for solving difference equations offers a clear separation between the zero
input response of the system to initial conditions and the zero state response of the system associated with the
input. The zero input response represents the system output when the input is zero, i.e., x(n) = 0, and thus it
is the result of internal system conditions (such as energy storage, initial conditions) alone. In contrast, the zero
state response represents the system response to the external input x(n) when the initial conditions are zero.
The total response is defined as,
Total response = Zero-input response + Zero-state response ...(76)
(natural response) (forced response)
Natural Response and Forced Response: In total response, the response due to the poles of the
system is called natural response of the system. The response due to the poles of the input is called the forced

z-Transform 355
Example 7.38
Use the unilateral z-transform to determine the output of a system represented by the
difference equation,
5 1
y ( n 2)
y ( n 1) y ( n) 5x ( n 1) x( n)
6 6
in response to the input x ( n ) = u ( n ). Assume that the initial conditions on the system are
y (–1) = 2 and y (–2) = 0. Identify the forced response (zero-state response) y f ( n ) of the
system and the natural response (zero-input response) y n ( n ).
Solution 7.38
For, x(n) = u(n), x(–1) = x(–2) = 0
1
and X (z ) = 1
1 z
5 1
y( n 2) y( n 1) y( n) = 5x(n + 1) – x(n)
6 6
Because the given difference equation is in time advance operator form, the use of the left-shift property
may seem appropriate for its solution. This property require a knowledge of auxiliary conditions y(0),
y(1),....y(N – 1) rather than of the initial conditions y(–1), y(–2),....y(–n), which are generally given. This
difficulty can be overcome by expressing the given difference equation in delay operator form (obtained by
replacing n with (n – 2) and then using the right-shift property. The given difference equation in delay
operator form is
5 1
y ( n)y( n 1) y( n 2) = 5x(n – 1) – x(n – 2)
6 6
5 1 1 2
Y( z) [z Y( z) y( 1)] [ z Y( z) z 1 y( 1) y( 2) ]
6 6
= 5[z–1 X(z) + x(–1)] – [z–2 X(z) + z–1 x(–1) + x(–2)]
5 1 1 2 1
Y( z) [z Y( z) 2] [ z Y( z) 2 z 0]
6 6
= 5[z–1 X(z) + 0] – [z–2 X(z) + 0 + 0]
5 1 1 2 5 1 1
Y( z) 1 z z z = 5z–1 X(z) – z2 X(z)
6 6 3 3
5 1 5 1 1 1
Y( z) 1 z 1
z 2
= z X( z) [5z z 2]
6 6 3 3
Input terms
Initial condition terms
1/ 3(5 z 1 ) z 1 (5 z 1 )
Y(z) = 5 1 1 2 5 1
1 z z (1 z 1 ) 1 z 1
z 2
6 6 6 6
zero-input component zero-state response

1 / 3(5 z 1 ) z 1(5 z 1 )
Y(z) =
1 1 1 1 1 1 1 1 1
1 z 1 z (1 z ) 1 z 1 z
3 2 3 2
Using partial fraction, expansion, we obtain
3 4/3 12 18 6
Y(z) = 1 1 1 1 1
1 z 1
1 z 1 1 z 1 z 1
1 z 1
2 3 2 3
The inverse z-transform of this equation yields,

n n n n
1 4 1 1 1
y(n) = 3 u( n) u( n) 12 u( n) 18 u( n) 6 u( n)
2 3 3 2 3
n n n n
1 4 1 1 1
y(n) = 3 u( n) 12 18 6 u( n)
2 3 3 2 3
zero-input response or natural response zero-state response or forced response
y(n) = yzi(n) + yzs(n)

where the zero-input response is
n n
1 4 1
yzi(n) = 3 u( n)
2 3 3
and the zero-state response is

n n
1 1
yzs(n) = 12 18 6 u( n)
2 3
The poles of the given system are located at p1 = 1/2 and p2 = 1/3. The response y(n) can be separated into
two parts: the natural response yn(n), and the forced response yf(n).
n n n n
1 4 1 1 1
y(n) = 3 u( n) u( n) 12u ( n) 18 u( n) 6 u( n)
2 3 3 2 3
n n n n
1 4 1 1 1
= 3 u( n) u( n) 18 u( n) 6 u( n) 12 u( n)
2 3 3 2 3
n n
1 14 1
y(n) = 15 u( n) 12 u( n)
2 3 3
Forced response = y f (n)
Natural response = y n (n)
7.11 Block Diagram Representation

A linear system can be characterized by its transfer function H(z). Fig. 7.8 (a) shows a block diagram of

z-Transform 357
a system with a transfer function H(z) and its input and output X(z) and Y(z), respectively. Subsystems may be
interconnected by using cascade (series), parallel, and feedback interconnections, the three elementary types.
7.11.1 Cascade Interconnection

When the systems are connected in series, as depicted in Fig. 7.8 (b), then the transfer function of the
overall system is the product of the individual transfer functions. This result can also be proved by observing
that in Fig. 7.8 (b).
Y( z) W( z) Y( z)
H(z) = H1( z) H2 ( z)
X( z) X( z) W( z)
We can extend this result to any number of systems connected in cascaded. It follows from this discussion
that the systems connected in cascade can be interchanged without affecting the overall transfer function. This
commutation property of LTI systems follows from the commutative and associative property of convolution.
X(z) H(z) Y(z)
(a)
W(z) Y(z)
X(z) H1(z) H2(z) = X(z) H(z) = H1(z) H2(z) Y(z)
(b)
X(z) H1(z)
H1(z)
X(z) = X(z) H(z) = H1(z) + H2(z) Y(z)

Y(z)
H2(z)
X(z) H2(z)
(c)
+ E(z)
X(z) H1(z) Y(z)
–
H1( z)
= X(z) H1( z) Y(z)
1 H1( z) H2 ( z)
H2(z)
(d)
Fig. 7.8 : Elementary connection of block diagram
7.11.2 Parallel Interconnection

When the systems are connected in parallel, as depicted in Fig. 7.8 (c), then the transfer function of the
overall system is the sum of the individual transfer functions. This result can also be proved by observing that in
Fig. 7.8 (c).
Y(z) = X(z) H1(z) + X(z) H2(z)
Y(z) = X(z) [H1(z) + H2(z)]

Y( z)
= H(z) = H1(z) + H2(z)
X( z)
We can extend this result to any number of systems connected in parallel.
7.11.3 Feedback Interconnection

When the output is fed back to the input, as shown in Fig. 7.8 (d), the overall transfer function H(z) =
Y(z)/X(z) can be computer as follows. The input to the adder are X(z) and –H2(z) Y(z). Therefore, E(z), the
output of the adder, is
E(z) = X(z) – H2(z) Y(z)
but, Y(z) = H1(z) E(z)
= H1(z) [X(z) – H2(z) Y(z)]
Y(z) = H1(z) X(z) – H1(z) H2(z) Y(z)
H(z) [1 + H1(z) H2(z)] = H1(z) X(z)
Y( z) H1( z)
= H( z) ...(76)
X( z) 1 H1( z) H2 ( z)
Therefore the feedback loop can be replaced by a single block with the transfer function shown in
equation (76). The transfer function H1(z) is the forward path transfer function, and H2(z) is the feedback path
transfer function.
7.12 System Realization

We now develop a systematic method for realization (or implementation) of an arbitrary Nth order
transfer function. The most general Nth order transfer function is given by
b0 zN b1 zN 1 ... bN 1 z bN
H(z) =
zN a1 zN 1 ... a N 1 z a N
7.12.1 Direct Form 1 Realization

Consider for simplicity a third-order system characterized by a transfer function:
b0 z3 b1 z2 ... b2 z b3 b0 (b1 / z ) (b2 / z 2 ) (b3 / z 3 )

H(z) = =
z3 a1 z2 a2 z a3 a1 a2 a3
1 2
z z z3
We can express H(z) as,
b1 b2 b3 1
H(z) = b0
z z2 z3 (1 ( a1 / z) ( a2 / z2 ) ( a3 / z3 )
H1( z ) H2 ( z )
We can realize H(z) as a cascade of transfer function H1(z) followed by H2(z), as depicted in Fig. 7.9 (a),
where
W( z) b1 b2 b3
H1(z) = b0 ...(77)
X( z) z z2 z3

z-Transform 359
b1 b2 b3
W (z ) = b0 X( z) ...(78)
z z2 z3
Y( z) 1
and H2(z) = ...(79)
W( z) 1 ( a1 / z) ( a 2 / z2 ) ( a3 / z3 )
a1 a2 a3
Y(z) = W( z) Y( z) ...(80)
z z2 z3
We shall first realize H1(z) given in equation (71). Equation (72) shows that the output W(z) can be
synthesized by adding by adding the input b0X(z) to b1X(z)/z, b2(X(z)/z2), and b3(X(z)/(z3). Because the transfer
function of a unit delay is z–1 = 1/z, the signals X(z)/z, X(z)/z2, and X(z)/z3 can be obtained by a successive delay
of the input x(n). The left-half section of Fig. 7.9 (b) shows the realization of H1(z).
W(z)
X(z) H1(z) H2(z) Y(z)
(a)
b0 W(z)
X(z) Y(z)
z–1 z–1
b1 –a 1
z–1 z–1
b2 –a 2
z–1 z–1
b3 –a 3
(b)
Fig. 7.9 : (a) Realization of a transfer function in two steps

and (b) direct form-I realization of a third-order discrete-time LTI system
We next consider the realization of H2(z) given by equation (79). Equation (80) shows that the output
Y(z) can be synthesized by subtracting a1(Y(z)/z), a2(Y(z)/z2), and a3(Y(z)/z3) from W(z). To obtain signals
(Y(z)/z2), and (Y(z)/z2), and (Y(z)/z3), we assume that the already have the desired output Y(z). Successive
delay of Y(z) yields the needed signals Y(z)/z, Y(z)/z2 and Y(z)/z3. the right-half section of Fig. 7.9 (b) shows the
realization of H2(z). We can generalize this procedure, known as the direct form I realization, for any value of N.
This procedure requires 2N delay units to realize and Nth-order transfer function. This realization is noncanonic
since it employs six delay units to implement a third-order transfer function. A realization is canonic if the
number of delay units used in the realization is equal to the order of the transfer function realized. Thus canonic
realization has no redundant delay units.
7.12.2 Direct Form-2 Realization (Canonic Form Realization)

In the direct form I, we realize H(z) by implementing H1(z) followed by H2(z), as shown in Fig. 7.9 (a).

We can also realize H(z), as shown in Fig. 7.10 (a), where H2(z) is followed by H1(z). This procedure is known
as the direct form II realization.
The direct form I realization [Fig. 7.9 (b)] implements zeros first [the left-half section represented by
H1(z)] followed by realization of poles [the right-half section represented by H2(z)] of H(z). In contrast, the
direct form II realization implements poles first followed by zeros.
Again consider for simplicity a third-order system characterized by a transfer function:
b0 z3 b1 z2 b2 z b3
H(z) =
z3 a1 z2 a2 z a3
b0 (b1 / z ) (b2 / z 2 ) (b3 / z 3 )

H(z) =
a1 a2 a3
1 2
z z z3
We can express H(z) as,
1 b1 b2 b3
H(z) = 2 3
b0
(1 ( a1 / z) ( a2 / z ) ( a3 / z ) z z2 z3
H2 ( z ) H1( z )
We can realize H(z) as a cascade of transfer function H2(z) followed by H1(z), as depicted in Fig. 7.10 (a),
where
V( z ) 1
H2(z) = ...(81)
X( z) (1 ( a1 / z) ( a2 / z2 ) ( a3 / z3 )
a1 a2 a3
V(z) = X( z) V( z) ...(82)
z z2 z3
Y( z) b1 b2 b3
and H1(z) = b0 ...(83)
V( z) z z2 z3
b1 b2 b3
Y(z) = b0 V( z) ...(84)
z z2 z3
The left-half section of Fig. 7.10 (b) shows the realization of H2(z) and the right-half section shows the
realization of H1(z).
We observe that in Fig. 7.10 (b) the signal variables at nodes 1 and 1 are the same, and hence, the two
top delay units can be shared. Likewise, the signal variables at nodes 2 and 2 are the same, which permits the
sharing of the two middle delay units. Following the same argument, we can share the delay units leading to the
final structure shown in Fig. 7.10 (c)
This implementation halves the number of delay units to N, and thus more efficient in hardware utilization.
This is the direct form II realization. This realization is canonic since it employs N delay units to implement a
Nth-order transfer function.

z-Transform 361
V(z) b0
X(z) Y(z)
z–1 z–1
–a 1 b1
1 1
W(z)
X(z) H2(z) H1(z) Y(z)
z–1 z–1
(a)
–a 2 b2
2 2
z–1 z–1
–a 3 3 3 b3
(b)
V(z) b0
X(z) Y(z)
z–1
–a 1 b1
z–1
–a 2 b2
z–1
–a 3 b3
(c)
Fig. 7.10 : Direct form II realization of a third-order LTI system
Example 7.39
Let x ( n ) be an absolutely summable signal with rational z -transform X ( z ). If X ( z ) is known to

have a pole at z = 1/2, could x ( n ) be
( a ) a finite-duration signal? ( b ) a left-sided signal?
( c ) a right-sided signal? ( d ) a two-sided signal?
Solution 7.39
( a ) No. we know that for a finite-length signal, the ROC is the entire z-plane. Therefore, there can be no
poles in the finite z-plane for a finite-length signal.
( b ) No. since the signal is absolutely summable, the ROC must include the unit circle. Since it is given that
the signal has a pole at z = 1/2, the ROC can never be the region z 1/ 2. Therefore, the signal cannot
be left-sided.
( c ) Yes. since the signal is absolutely summable, the ROC must include the unit circle. Since it is given

that the signal has a pole at z = 1/2, a valid ROC would be the region z 1/ 2. Therefore, the signal
could be right-sided.
( d ) Yes. since the signal is absolutely summable, the ROC must include the unit circle. Clearly, we may
define the ROC which is a ring in the z-plane add includes the unit circle. Therefore, the signal could
be two-sided.
Example 7.41
Suppose we are given the following five facts about a particular LTI system with impulse
response h ( n ) and z -transform H ( z ):
1 . h(n) is real.
2 . h(n ) is right-sided.
3. Lim H( z) 1
z
4 . H ( z ) has two zeros.

5 . H ( z ) has one of its poles at a non-real location on the circle defined by z 3 / 4.
Answer the following two questions:
( a ) Is this system causal?
( b ) Is this system stable.
Solution 7.41
( a ) Since Lim H( z) 1 , H(z) has no poles in infinity. Furthermore, since h(n) is given to be right-sided,
z
h(n) has to be causal.

( b ) Since Lim H( z) 1 ,and h(n) is causal, the numerator and denominator polynomials of H(z) have the
z
same order. Since H(z) has two zeros, we may conclude that it also has two poles. Since h(n) is real,
the poles must occur in conjugate pairs. Also, it is given that one of the poles lies on the circle defined
by z 3 / 4. Therefore, the other pole also lies on the same circle. Clearly, the ROC for H(z) is z 3 / 4,
and will include the unit circle. Therefore, we may conclude that the system is stable.
Example 7.41
A causal signal x ( n ) has the z -transform,

X ( z ) = sin( z –1 ) (1 + 3 z –2 + 2 z –4 )
Find: x (11).
Solution 7.41
Given that, X(z) = sin(z–1) (1 + 3z–2 + 2z–4)
3 5 7 9 11
1 z z z z z 2 n
= z (1 3 z 2z 4 ) = x( n) z
3! 5! 7! 9! 11! n 0
Since x(11) is simply the coefficient in front of z–11 in this power series expansion of X(z), therefore
1 3 2
x(11) =
11! 9! 7!

z-Transform 363
Example 7.42

2 4
2 z 3z
X( z) 2
, z 0
z 4z 3
Solution 7.42
We see that, X(z) can be written as,
X(z) = (2z–1 + z–3 + 3z–5) X1(z)
z
where, X 1( z ) =
z2 4 z 3
Thus, if x1[n] X 1( z )
then by the linearity property and the time-shifting property, we get
x[n] = 2x1[n – 1] + x1[n – 3] + 3x1[n – 5]
X1( z) 1 1 c1 c2
Now, = 2
z z 4z 3 ( z 1) ( z 3) z 1 z 3
1 1
where, c1 =
z 3z 1 2
1 1
c2 =
z 1z 3 2
1 z 1 z
Then, X 1( z ) =
2 z 1 2 z 3
Since the ROC of X1(z) is z 0, z 0 x1[n] is a right sided sequence, and we get
1
x1[n] = [( 1)n ( 3)n u[ n]
2
Thus, we get
x[n] = [(–1)n – 1 –(–3)n – 1] u[n – 1]
1 3
[( 1)n 3
( 3)n 3
] u[ n 3] [( 1)n 5
] ( 3)n 5
] u[ n 5]
2 2
Example 7.43
n
1
The output y [n ] of a discrete-time LTI system is found to be 2 u[ n] when the input x[ n] is
3
u [ n ].
( a ) Find the impulse response h[n] of the system.
n
1
( b ) Find the output y [ n ] when the input x[ n ] is u[ n].
2
Solution 7.43
z
(a) x[n] = u[ n] X( z) ; z 1
z 1

n
1 2z 1
y[n] = 2 u[ n] Y( z) , z
3 1 3
z
3
Hence, the system function H(z) is
Y( z) 2( z 1) 1
H(z) = ; z
X( z) 1 3
z
3
Using partial-function expansion, we have
H( z) 2( z 1) c1 c2
=
z 1 z 1
z z z
3 3
2( z 1)
where, c1 = 6
1
z
3 z 1/ 2
2( z 1)
c2 = 4
z z 1/ 3
z 1
Thus, H(z) = 6 4 , z
1 3
z
3
Taking the inverse z-transform of H(z), we obtain
n
1
h[n] = 6 [ n] 4 u[ n]
3
n
1 z 1
(b) x[n] = u[ n] X( z) ; z
2 1 2
z
2
Then, Y(z) = X(z) H(z)
2 z ( z 1) 1
= , z
1 1 2
z z
2 3
Again by partial fraction expansion we have,
Y( z) 2 z ( z 1) c1 c2
=
z 1 1 1 1
z z z z
2 3 z 3
2( z 1)
where, c1 = 6
1
z
3 z 1/2
2( z 1)
c2 = 8
1
z
3 z 1/ 3

z-Transform 365
z z 1
Thus, Y(z) = 6 8 ; z
1 1 2
z z
2 3
Taking the inverse z-transform of Y(z) we obtain,
n n
1 1
y(n) = 6 8 u[ n]
2 3
Example 7.46
Find the steady state response of the discrete time system described by
0.1 z
H( z)
z 0.5
When excited by x [ n ] = cos(0.2 n + 10°) with discrete frequency = 0.2.
Solution 7.46
For the given input, x[n] = cos 0.2n
j
Response, y[n] = H (e ) cos( 1n )
0.1 e j 0.2
H( e j ) =
2
e j 0.2 0.5
0.1 11.3 0.1 11.3
= 0.098 j 0.0196 0.5 0.402 j 0.0196
0.1 11.3
= 0.25 14.1
0.40 2.8
Therefore, y[n] = 0.25 cos[0.2n + 10° + 14.1°]
= 0.25 cos[0.2n + 24.1°]

1 8
(c) X ( z ) , z
64 2 9
1 z
81
1 Subjective Practice Problems

6 . Let, y[n] =
1
9
n
u[ n]
1 . Let, x[n] = (–1)n u[n] + an u[–n – n0] Determine two distinct signals such that each has
Determine the constraints on the complex number z-transform X(z) which satisfies both of the following
a and the integer n0, given that the ROC of X(z) is conditions:
1 z 2. [ X ( z ) X ( z )]
1. Y (z 2 )
2
2 . Consider the signal, 2. X(z) has only one pole and only one zero in the
z-plane.
n
1
cos , n 0 7 . Consider the following system functions for stable
x[n] = 3 4 LTI systems. Without utilizing the inverse
0, n 0 z-transform, determine in each case whether or not
Determine the poles and ROC for X(z). the corresponding system is causal.
4 1 1 2
3 . Suppose that the algebraic expression for the 1 z z
(a) 3 2
z-transform of x[n] is 1 1
1 1 1
z 1 z 1 z
1 2 2 3
1 z
X( z) 4
1 5 3 1
1 z 2
1 z 1
z 2 z
4 4 4 (b) 2
1 3
How many different regions of convergence could z2 z
2 16
correspond to X(z)?
z 1
4 . Let x[n] be a signal whose rational z-transform X(z) (c)
4 1 2 2 3
contains a pole at z = 1/2. Given that, z z z
3 2 3
n
1
x1[n] = x[ n] 8 . Consider a causal LTI system whose input x[n] and
4
output y[n] are related through the block diagram
is absolutely summable and
representation shown in figure.
n
1
x2[n] = x[ n] x[n] y [n]
8
is not absolutely summable, determine whether x[n]
is left sided, right sided, or two sided.
z–1
5 . Determine, for each of the following z-transforms,
whether the corresponding signal has an
2/3 –6
approximately lowpass, bandpass, or highpass
characteristic:
1 z–1
z 8
(a) X( z) , z
8 1 9
1 z –1/9 8
9
8 1 (a) Determine a difference equation relating y[n]
1 z 8
(b) X( z) 9 , z and x[n].
16 1 64 2 9 (b) Is this system stable?
1 z z
9 91

z-Transform 367
9 . Determine the unilateral z-transform of each of the 5 6
following signals, and specify the corresponding 1 1
z z
regions of convergence. 4 4
(c) 6
(d) 5
1 1
n z z
1 2 2
```(a) x1[ n] u[ n 5]
4
14. The following is known about a discrete-time LTI
(b) x2[n] = [n + 3] + [n] + 2n u[–n]
system with input x[n] and output y[n]:
n
1 1. If x[n] = (–2)n for all n, then y[n]= 0 for all n.
(c) x3 [ n]
2 2. If x[n] = (1/2)n u[n] for all n, then y[n] for all n
is of the form
10. A right sided sequence x[n] has z-transform,
n
1
3z 10
z 7
5z 2
4z 1
1 y[n] = [ n] a u[ n],
X( z) 4
z 10 5z 7
z 3
where a is a constant.
Determine x[n] for n < 0. (a) Determine the value of the constant a.
11. We are given the following five facts about a discrete (b) Determine the response y[n] if the input x[n] is
time signal x[n] with z-transform X(z): x[n] = 1, for all n
1. x[n] is real and right sided. 15. When the input to an LTI system is
2. X(z) has exactly two poles. n
3. X(z) has two zeros at the origin. 1
x( n) u( n) (2)n u( n 1)
3
1 j /3
4. X(z) has a pole at z e . and the corresponding output is
2
n n
8 1 2
5. X(1) . y ( n) 5 u( n) 5 u( n)
3 3 3
Determine X(z) and specify its region of convergence. (a) Find the system function H(z) of the system. Plot
the poles and zeros of H(z) and indicate the ROC.
12. A causal LTI system is described by the difference
(b) Find the impulse response h(n) of the system.
equation,
(c) Write a difference equation that is satisfied by
y[n] = y[n – 1] + y[n – 2] + x[n – 1]
the given input and output.
(a) Find the system function H(z) = Y(z)/X(z) for
(d) Is the system stable? Is it causal?
this system. Plot the poles and zeros of H(z) and
indicate the region of convergence. 2
(b) Find the unit sample response of the system. 16. Determine the inverse transform of X( z) ez with
(c) You should have found the system to be unstable.
ROC all z except z .
Find a stable (noncausal) unit sample response
that satisfies the difference equation.
17. Find the z-transform of the digital signal obtained by
13. Determine which of the following z-transforms could sampling the analog signal x(t) = e–4t sin(4t) u(t) at
be the transfer function of a discrete-time linear intervals of T = 0.1 seconds.
system that is not necessarily stable, but for which
18. Using long division, determine the inverse z-
the unit sample response is zero for n < 0. State
transform of
your reasons clearly.
1
1 2z
(1 z 1 )2 ( z 1)2 X( z)
(a) (b) 1 2z 1 z 2
1 1 1
1 z z if (a) x(n) is causal and (b) x(n) is anticausal.
2 2

n
1
20. (a ) When input x( n) u( n) u( n) is
2 21. Computer the unit step response s(n) of the system
with impulse response
applied to a linear causal time invariant system, the
output is 3n , n 0
n n h( n) 2
n
1 1 , n 0
y( n) 6 u( n) 6 u( n) 5
4 3
Find the transfer function of the system.
(b) What is the difference equation representation
of the system?

Chapter
Discrete Time
Fourier Transform 8
8.1 Introduction
Discrete time Fourier Transform (DTFT) is used to perform frequency analysis of periodic and non-
periodic discrete time signals. We have already learned about the z-Transform and we know that discrete
frequency axis is always periodic with period 2 . In this chapter we will study all properties of DTFT of discrete
time signals
8.2 DTFT representation of non periodic discreate-time signals

The periodic discrete time signals are represented using DTFS. For periodic signal with period N has
DTFS coefficients Xn which are discrete and periodic with period N and
j o nk
x[n] Xk e
k N
We can assume a non periodic signal with N i.e. Infinite period and o 0 thus the discrete
components will come close to each other and will form a continuous plot. Thus DTFT is a continuous plot
which is periodic with period 2 . We can write
1
x[n] X( e j )e j n d ...(1)
2 2
j n
and X(e j ) x[ n]e ...(2)
n
REMEMBER
• Thus DTFT convert a non periodic discrete signal into continuous periodic signal with period 2
x[n] X(e j )
• From eq.(2) we can see that DTFT of a signal is z-transform at unit circle
DTFT = X(e j ) X ( z )|z ej
Thus if a signal is absolutely summable or ROC of it’s z-transform contain unit circle then best method is to
find X(z) and replace z by e j

8.3 Periodicity of DTFT

The DTFT is a periodic function in w with a period 2 , i.e.,
X( e j( 2 )
) = X(e j ) …(3)
Proof We know that
j n
X(e j ) x ( n )e
n
For any integer k, we have
X( e j( 2 k)
) x( n)e j( 2 k )n
j n j 2 kn
x( n)e e
n
X( e j( 2 k)
) x( n)e j n
X( e j( 2 k)
) = X(e j )
where we have used the fact that e–j2 kn = 1. Hence X(e j ) is periodic with period 2 . But this property
is just a consequence of the fact that frequency range for any discrete-time signal is unique over the frequency
interval of (– , ) or , equivalently, (0, 2 ), and any frequency outside this interval is equivalent to a frequency
within this interval.
8.4 Convergence of DTFT

There is one sufficient but not necessary condition for existence of DTFT of signal x[n]. The condition is
that signal should be absolutely summable that is
x( n) ...(4)
n
Thus if ROC of z-transform of x[n] has unit circle then we can directly find DTFT by replacing z by e j .
Study Note
Only one condition exist for existence of DTFT of x[n]. There is no condition on number of discontinuity in x[n]
or finite number of maxima and minima because x[n] is discrete.
Example 8.1
Find the Fourier transform of the unit sample squence x ( n ) = (n)

Solution 8.1
The Fourier transform X(e j ) of the sequence x(n) is given by
j n
X(e j ) x( n)e
n
j n j n
( n)e e 1
n n 0

(n ) 1 …(5)
x (n ) X(ej )
1
1
–4 –3 –2 –1 1 2 3 4 n –
(a) (b)
Fig (a) Unit impulse sequence and (b) its Fourier transform
Note: The unit sample sequence has a Fourier transform representation consisting of equal contributions at all
frequencies. The unit sample sequence and its Fourier transform are shown in Fig. (a) and (b), respectively.
Example 8.2
Find the Fourier transform of the causal sequence [shown in Fig. (a)]
x ( n ) = a n u (n ), | a | < 1.
x(n) x(n)
1 0 a<1 1 a 0
1/a
2
1/a
1 3 5 7 9
0 1 2 3 4 n 0 2 4 6 8 n
–1/a
(a)
Fig 8.2 : (a) Signal x(n)
Solution 8.2
We can see that the signals are absolutely summable and the z-transform is
1
X (z ) ,
1 az 1
pole of X(z) is at z = a and signal is right sided, thus ROC is |z| > |a|
1
X (z ) 1
, | z| | a |
1 az
Since |a| < 1 thus ROC of X(z) has unit circle and DTFT will be
1
X( z)|z ej j
1 ae
To find magnitude and phase response of X(e j ) we will replace
e j = (cos + j sin )
1
Thus X(e j )
1 a(cos j sin )

1
Magnitude of X(e j )
(1 a cos )2 ( a sin )2
1 a sin
and phase of X(e j ) is tan
1 a cos
The phase and magnitude plot both will be periodic with period 2 .
Example 8.3
Find the Fourier transform of the sequence

x ( n ) = a n u( –n – 1), | a | > 1.
x (n)
… –4 –3 –2 –1 0 n
Solution 8.3
The z-transform of x[n] is
X (z ) a nz n
n
n
z
n 1 a
z/a
X (z )
1 z/a
The pole of X(z) is z = a, since signal is left sided signal the ROC will be |z| < |a|.
Now |a| > 1 thus ROC contain unit circle
Thus X(e j ) X( z)|z ej
z
a zz ej
ej
a ej
cos j sin
X(e j )
a cos j sin
1
The magnitude of X(e j ) is
( a cos )2 (sin )2

1 sin 1 sin
and phase of X(e j ) is tan tan
cos a cos
Example 8.4
Find the fourier transform of signal

x ( n ) = a |n | , | a | < 1.
Solution 8.4
When x[n] = a|n| and |a| < 1 then signal x[n] is absolutely summable and
x[n] = anu[n] + a–nu[–n – 1]
1
a n u[ n] z transform ,ROC| z | | a |
1 az 1
z transform az
and a nu [ n 1] ,ROC| z | |1 / a |
1 az
1 az
a n u[ n] a n u[ n 1] 1
1 az 1 az
and ROC of X(z) is |a| < |z| < |1/a|
Since |a| < 1 thus ROC contain unit circle
1 ae j
X (e j ) X( z)|z ej j
1 ae 1 ae j
1 ae j ae j a2
1 ae j ae j a2
1 a2
X(e j ) =
1 2a cos a2
Example 8.5
Find the fourier transform of rectangular pulse

1, | n| N1
x ( n ) = u[n + N1] – u[n – N1 – 1] =
0, | n|> N1
Solution 8.5
Since x[n] is non zero only for finite number of n thus it is absolutely summable.
Since x[n] = u[n + N1] – u[n – N1 – 1]
1 zN1
X (z ) 1
zN1 z ( N1 1)
(1 z 2 N1 1
)
1 z (1 z 1 )
Thus the pole at z = 1 will be cancelled by zero at z = 1 because 1 z 2N1 1 will give 2N1 + 1 zeros on cicle
of unit radius.
Thus ROC of X(z) is entire z plane except z = and ROC contain unit circle.
(1 e (2N1 1) j )
X(e j ) X( z)|z ej
ej N1
(1 e j )
Taking e j ( N1 1/2)
comman from numerator and e–j /2 from denominator..

sin( ( N1 1 / 2))
X(e j )
sin( / 2)
Example 8.6
Find the inverse fourier transform of
X(e j ) = 2 ( 2 m)
m=
Solution 8.6
The plot of X(e j ) is
X (e j )
2
… …
–2 0 2 4 6
The DTFT is periodic with period 2 . The signal x[n] is
1
x[n] X (e j )e j n
d
2
For one period between – and X(e j ) = 2 ( )
1
x[n] 2 ( )e j n d 1
2
1 2 ( ) ( to )
or 1 2 ( 2 m)
m
REMEMBER
• We can see that the DTFT of 1 exist, z-transform of 1 do not exist
1 2 ( 2 m)
m
• Signal x[n] = 1 is not absolutely summable thus we cannot find result by applying the eq.(2)
Example 8.7
Find the DTFT of unit step function u [ n ]

Solution 8.7
Since signal x[n] = u[n] is not absolutely summable we cannot apply eq.(2) to find X(e j ) and neither
X( z)|z ej because ROC of z-transform of u[n] do not contain unit circle. We will use another method to
find X(e j ) of u[n].
1
Let f [n] for n
2

1
for n 0
2
and g[n]
1
for n 0
2
So g[n] – g[n – 1] = [n]
If g[ n] G( e j )
g[ n 1] G( e j )e j
[ n] 1
1
Thus G(ej )= j
1 e
and f [ n] ( 2 m)
m
Since u[n] = f [n] + g[n]

j j
u[ n] F( e ) G( e )
1
( 2 m)
1 e j m
Thus for – < < we get

1
u[ n] j
( )
1 e
Example 8.8
Find the inverse DTFT of the rectangular pulse spectrum defined only for –
1, | | c
X(ej ) rect
2 c 0, c | |
which is depicted in Fig. (a). Alternatively, we can define X ( e j ) over all by writing it as an
infinite sum of rectangle functions shifted by integer multiple of 2
2 m
X(ej ) rect
m 2 c
Solution 8.8
To find x[n] we can apply equation
1
x[n] X( e j )e j n d
2
for – < < we have
X(e j ) rect
2 c

1
x[n] ej nd
2
c
1
(e j cn e j cn
)
2 jn
sin( c n)
x[n]
n
sin( c n)
rect ,
n 2 c
The signal x(n) is depicted in Fig. (b)

x (n )
X(ej )
c
–2 – – c 0 c 2 0
(a) (b)
Fig : (a) Rectangular pulse spectrum and (b) its inverse Fourier transform
REMEMBER • Thus a discrete low pass filter with cut off frequency c is H(e j ) rect ,
2 c
sin( c n)
and it’s time domain signal will be h[n]
n
• Thus we have learned two method to find DTFT of a signal x[n]. If x[n] is absolutely summable
then X(e j ) X( z)|z ej
. But if signal is power signal then we cannot apply eq.(2) to find
X(e j ) and we have to apply some manipulations to find result
• We cannot find DTFT of signals which are neither energy and nor power signals.
8.5 Properties of DTFT

We assume that X(e j ) is DTFT of x[n], that is
x[ n] X( e j )
8.5.1 Linearity
If
x1( n) X1( e j ) and x2 ( n) X2 ( e j )
then
ax1( n) bx2 ( n) aX1( e j ) bX2 ( e j ) …(6)
Proof The Fourier transform of ax1(n) + bx2(n) is

j n
[ax1(n)+bx 2 (n)] [ax1( n) bx2 ( n)]e
n
j n j n
a x 1( n )e b x 2 ( n )e
n n
X 1 (e j ) X 2 (e j )
= aX1(e j ) + bX2(e j )
8.5.2 Time shifting

If
x( n) X( e j )
then
j n0
y(n)
x( n n0 ) Y( e j ) X( e j )e …(7)
Proof The Fourier transform of x(n – n0) is
j n
[x (n n0 )] x( n n0 )e
n
A change of variables is performed by letting m = (n – n0), which also yields n = (m + n0), m = – as
n = – and m = as n = . Therefore,
j ( m n0 )
[x (n n0 )] x( m)e
n
j n0 j m
e x( m)e
m
j j n0
[x (n n0 )] X( e )e
REMEMBER
• One consequence of this property is that, when a signal is shifted in time, the magnitude of its Fourier transform
remain unaltered. That is, if we express X(e j ) in polar form as
X( e j )
[ x ( n )] X( e j ) | X( e j )| e j
then
j n0 X( e j )
[x (n n0 )] X( e j )e | X( e j )| e j( n0 )
Thus, the effect of a time shift on a signal is to introduce into its Fourier transform a phase shift, n0, which is a
linear function of .
• In time shifting the effect goes with some sign
If x[n] X(e j )
j n0
then x[n – n0] X( e j )e
Example 8.9
Find DTFT of sequence

1 1
x[n ] sin c (n 2)
4 4

Solution 8.9
It will be difficult to solve for X(e j ) using eq.(2) because summation will be difficult
sin n
Since 4 rect ,
n /2
sin ( n 2)
4 rect e j2
,
( n 2) /2
1 1 j2
sin c ( n 2) rect e ,
4 4 /2
8.5.3 Frequency shifting

If
x( n) X( e j )
then
y(n) x( n)e j 0n Y( e j ) X( e j( 0) ) …(8)
Proof The Fourier transform of x( n)e j 0 n is given by
[ x ( n )e j 0 ] [x( n)e j 0n
]e j n
n
j( 0 )n
x( n)e
n
[ x ( n )e j 0 ] X( e j( 0) )
Hence, a frequency shift corresponds to multiplication in time domain by a complex sinusoid whose
frequency is equal to the frequency shift.
Note: In case of frequency shifting the effect in seen with apposite sign, that is if
x[n] X(e j )
then e+j n
0 x[n] X(e j( – 0) )
Example 8.10
If DTFT of x [ n ] is X ( e j ) then DTFT of (–1) n x [ n ] is __________.

Solution 8.10
Given that
x[n] X(ej – )
then (–1)nx[n] = e j n x[n] X(e j( – ))
Thus shifting X(e j ) on axis by left side
8.5.4 Time Reversal

If x[n] X(ej )
then
y(n) = x(–n) Y(e j ) = X(e–j ) …(9)

Proof The Fourier transform of y(n) is given by

j n
Y( e j ) y( n)e
n
j n
x( n)e
n
Substituting m = –n into the above equation gives
Y( e j ) x( m)e j m
j( )m
x( m)e
n
Y( e j ) = X(e–j )
An interesting consequence of the time-reversal property is that if x(n) is even, then its Fourier transform
is also even, i.e.,
if x(–n) = x(n),
then X(e–j ) = X(ej ) …(10)
Similarly, if x(n) is odd, then so is its Fourier transform, i.e.,
if x(–n) = – x(n),
then X(e–j ) = – X(ej ) …(11)
Note: Thus even signal has even DTFT and odd signal has odd DTFT
8.5.5 Time Expansion

Suppose x[n] is a discrete time signal and X(e j ) is DTFT of x[n], that is
j n
X(e j ) x[ n]e
n
if we generate a new signal y[n] = x[an] then DTFT of y[n] will be
j n
Y( e j ) x [an ]e
n
There will be no exact relation between Y(e j ) and X(e j ) because X(e j ) has all values of x[n] and Y(e j )
has x[a], x[2a] … and not all value of x[n]. But if we expand signal x[n] that is
Let m be a positive integer, and define the signal
n
x(m)(n) m …(12)
0, if n is not a multiple of m.
xm(n) can be obtained from x(n) by placing (m – 1) zeros between successive values of the original
signal. Intuitively, we can think of x(m)(n) as a slowed-down version of x(n).
Now, if
x(n) X(ej )
then

y(n) = x(m)(n) Y(e j ) = X(e jm ) …(13)

Proof The Fourier transform of y(n) = x(m)(n) is given by
j n
Y( e j ) y( n)e
n
j n n j n
x( m)( n)e x e
n n m
n
A change of variables is performed by letting r , which also yields r = – as n = – and r = as
m
n = . Therefore,
j mr
Y( e j ) x ( r )e
r
Y(e j ) = X(ejm )
Note that as the signal spread out and slowed down in time by taking m > 1, its Fourier transform is
compressed.
8.5.6 Differencing in Time Domain

If a signal y[n] = x[n] – x[n – 1]
and DTFT of x[n] is X(e j ) then DTFT of y[n] will be
Y(e j ) = X(e j ) – X(ej )e–j
Y( e j ) = (1 – e –j )X(e j ) …(14)
8.5.7 Differentiation in Frequency Domain

If
x(n) X(ej )
then
dX( e j )
–jnx(n) …(15)
d
or
dX( e j )
nx(n) j …(16)
d
Proof We have
j n
X(e j ) x( n)e
n
By differentiating both sides w.r.t. , we obtain
dX( e j ) [ jnx( n)]e j n
d n
dX( e j )
= [ jnx ( n )]
d
Therefore,
dX( e j )
–jnx(n)
d

Multiplying both sides by j yields

dX( e j )
nx(n) j
d
Example 8.11
Determine the Fourier transform of the sequence

y ( n ) = na n u ( n ), | a | < 1
Solution 8.11
Define a squence
x(n) = anu(n), |a| < 1
Its Fourier transform is give by
1
X(e j ) j
1 ae
Now consider the given sequence
y(n) = nanu(n) = nx(n)
Now apply the differentiation in frequency domain property to get the Fourier transform of y(n):
dX( e j ) d 1
Y( e j ) j j j
d d 1 ae
ae j
(1 ae j )2
ae j
nanu(n) , |a| 1
(1 ae j )2
Example 8.12
Determine the inverse DTFT of
1
X(ej ) j
, |a| 1
(1 ae )2
Solution 8.12
We have
ae j
nanu(n) , |a| 1
(1 ae j )2

ae j
(n + 1)an+1u(n + 1) ej
(1 ae j )2
a
(n + 1)an+1u(n + 1) j
(1 ae )2
1
(n + 1)anu(n + 1) j
(1 ae )2
Therefore,

x(n) = (n + 1)anu(n + 1)
It is worth noting that although the RHS is multiplied by a step function that begins at n = –1, the sequence
(n + 1)anu(n + 1) is still zero prior to n = 0, since the factor n + 1 is zero at n = –1. Thus, we can
alternatively express x(n) as
x(n) = (n + 1)anu(n)
and hence,
1
(n + 1)anu(n) j
(1 ae )2

If
x1(n ) X1(ej ) and x2(n) X2(ej )
then
x 1( n )* x 2 ( n ) X1(e j )X2(ej ) …(17)
The fourier transform maps the convolution of two signals into the product of their Fourier transforms.
Proof The Fourier transform of x1(n) * x2(n) is
j n
x 1( n )* x 2 ( n ) [x1( n)* x2 ( n)]e
n
j n
x1( m)x2 ( n m) e
n m
By interchangin the order of summation, we have
j n
x 1( n )* x 2 ( n ) x1( m) x2 ( n m)e
m n
By applying the time-shifting property the bracketed term is X2(e j )e–j m. Substituting this into the
above equation yields
x 1( n )* x 2 ( n ) x1( m)( X2 ( e j )e j m
)
m
X2 ( e j ) x1( m)e j m
x 1( n )* x 2 ( n ) = X2(e j )X1(e j ) = X1(e j )X2(e j )

Therefore,
x1(n) * x2(n) X1(e j )X2(ej )
the frequency domain.
8.5.9 Accumulation Property

If
x[n] X(ej )

then
n
X (e j )
x[k] X (e j 0 ) ( 2 m) ...(18)
k 1 e j m
This can be simply proved by convolution property

n 1
Since x[k] x[ n]* u[ n] X( e j ) j
( 2 m)
k 1 e m
X (e j )
x[n] * u[n] X (e j 0 ) ( 2 m)
1 e j m
8.5.10 Multiplication Property

If
x1( n ) X1(ej ) and x2(n) X2(ej )
then
1 1
x1(n ) x2(n ) [X1(e j ) X 1(e j )] X1(e j )X 2(e j ( )
)d …(19)
2 2 2
where denotes the periodic convolution. The Fourier transform maps the multiplication of two signals
into the convolution of their Fourier transforms.
Proof The Fourier transform of x1(n)x2(n) is given by
j n
x 1( n )x 2 ( n ) [x1( n)x2 ( n)]e
n
1
X1( e j )e j n d x2 ( n)e j n
n 2 2
Interchanging the order of intergration and summation and noting that X1(ej ) does not depend on n,
we obtain
1
x (n 2) X1( e j ) [ x2 ( n)e j n ]e j n
d
2 2 n
By applying the frequency-shifting property the bracketed term is X2(e j( – )). Substituting this into the
above equation yields
1
x 1( n )x 2 ( n ) X1( e j )X2 ( e j( )
)d
2 2
1
[X 1(e j ) X 2 (e j )]
2
Therefore,
1
x1 ( n ) x2 ( n ) [ X 1(e j ) X 2 (e j )]
2
The convolution integral in Eq. (19) is known as the periodic convolution of X1(e j ) and X2(ej ) because
it is the convolution of two periodic functions having the same period. We note that the limits of integration

extend over a single period.
8.5.11 Conjugation and conjugate symmetry
If
x(n) X(ej )
then
x*(n) X*(e–j ) …(20)
Proof The Fourier transform of x*(n) is
*
j n j n
x *(n ) x *( n)e x( n)e
n n
*
j( )n
x( n)e
n
= [X(e –j )]*
F[x*(n)] = X *(e–j )
1. When the signal is real
that is x[n] = x*[n]
If DTFT of x[n] is X(e j ) then that of x*[n] is X *(e–j )
Thus for a real signal
X(e j ) = X *(e–j )
Thus DTFT will be conjugate symmetric
Note: Similarly if x[n] is imaginary then X(e j ) that is DTFT will be conjugate anti symmetric
2. When the signal is real and even

Here x[n] = x*[n] and x[n] = x[–n]
Thus X(e j ) = X *(e–j ) and X(ej ) = X(e–j )
X( e j ) = X *(ej ) = X(e–j )
X(e j ) is real and even
Thus for real and even time domain signal DTFT is also real and even
3. When the signal is real and odd

That is x[n] = x*[n] and x[n] = –x[–n]
Thus X( e j ) = X *(e –j ) and X(ej ) = –X(e–j )
X(e j ) = –X *(ej ) = –X(e–j )
Thus X(e j ) is imaginary and odd
for a real and odd signal DTFT is imaginary and odd
4. If signal x[n] is real then DTFT of even and odd part of x[n] is
Given that x[n] X(ej )
and x[n] = x*[n]
X( e j ) = X *(e–j )

x[ n] x[ n] X( e j ) X( e j
)
Even part of x[n]
2 2
X (e j ) X *(e j )
2
xe[n] Real {X(e j )}
Similarly
x[ n] x[ n] X( e j ) X( e j
)
odd part of x[n]
2 2
X (e j ) X *(e j )
2
xo[n] jImg{X(ej )}
Note: Thus for a real signal the real part of DTFT is due to even part of signal and imaginary part of DTFT is due
to odd part of signal.

Let x(n) be an energy signal, and if
x(n) X(ej )
then
1
Ex | x( n)|2 | X( e j )|2 d ...(21)
n 2 2
Proof Consider the LHS of above equation, we have
Ex | x( n)|2 x( n)x *( n)
n n
*
1 j j n
x( n) X( e )e d
n 2 2
1
x (n ) X *(e j )e j n
d
n 2 2
1
X *(e j ) x (n )e j n
d
2 2 n
1
X *(e j )X (e j )d
2 2
1
| x( n)|2 | X( e j )|2 d
n 2 2
Example 8.13
Find the energy of the sequence
x ( n ) = sinc cn

assuming c < .
Solution 8.13
We have
c cn 2 m
sinc rect
m 2 c
cn 2 m
x(n) = sinc X( e j ) rect
c m 2 c
Hence, from the Parseval’s relation we have

1
Ex |x( n)|2 | X( e j )|2 d
n 2 2
2
2
1
Ex 2
rect d
2 c 2 c
We know that
1, | | c
rect
2 c 0, c | |
Substituting the value of rect into Eq.(6.61) yields

2 c
2 c
1
Ex 2
1d 2
2 c
2 c 2 c c
c
8.5.13 Duality Theorem

In DTFT the time domain signal (x[n]) is discrete and nonperiodic but fourier domain signal (X(e j )) is
continuous and periodic with period 2 .
DTFT
discrete non periodic signal continuous periodic signal (period 2 )
j n
X(e j ) = x[ n]e ...(22)
n
1
and x[n] X( e j )e j n d ...(23)
2 2
If we think about apposite transform then continuous time fourier series (CTFS) convert continuous
periodic signal (x(t)) into discrete non periodic signal (Ck) and we if we take period of x(t) as 2 then
CTFS
continuous periodic signal with period (2 ) discrete non periodic (Ck)
Since period is 2 0 =1
x(t ) Ck e j 0tk Ck e jtk …(24)

k k

1 j 0 kt 1 jkt
and Ck x(t )e dt x(t )e …(25)
2 2
2 2
If we compare equation 22, 23, 24 and 25 we can see that variable and t are continuous and k and n
are discrete
If we manipulate equation (22) and (23) by replacing n = –k and = t we get
X(ejt) x[ k]e jtk

k
1
and x[–k] X( e jt )e jtk
dt
2 2
The time domain signal X(e jt) formed by replacing by t in expression of X(ej ) will be periodic with
period 2 and it’s continuous time fourier series (CTFS) coefficients will be x[–k]
Thus if
DTFT
x [n] X( ej )
CTFS
X(ejt) Ck = x (–k )
Thus we can say that duality exist between CTFS and DTFT and it can be used to find DTFT of power
signals for which we cannot apply equation (2) to find DTFT.
8.6 Insight in some equations

Let X(e j ) is DTFT of x[n]
1. X(ejo) x[ n]
n
j n
Since X(e j ) x[ n]e
n
X(e jo) x[ n]
n
2. X( e j ) ( 1)n x[ n]
n
j n
Since X( e j ) x[ n]e
n
X( e j ) ( 1)n x[ n]
n
1
3. x[0] X( e j )d
2
1
Since x[n] X( e j )e j n d
2

1
x[0] X( e j )d
2
The average value of X(e j ) is x[0].

1 dX( e j )
4. d =0
2 d
dX( e j )
Since nx[n] j
d
n dX( e j )
x[ n]
j d
n 1 dX( e j ) j n
x[ n] e d
j 2 d
Putting n = 0 both side of the equation we get
1 dX (e j )
0 d
2 d
2
dX( e j )
5. d 2 | nx[ n]|2
d n
dX( e j )
Since nx[n] j
d
Now applying parsevals theorem we get
2
1 dX( e j )
d | nx[ n]|2
2 d n
REMEMBER whenever we have to find angle of X(ej ) then the time domain signal x(n) which is already real
can be made even by shifting. Now the new signal(y(n)) obtained by shifting of x(n) is real and
even then it’s DTFT will also be real and even thus will have zero phase. Using the time shifting
property we can easily find the phase of X(ej ). The following example illustrate this procedure.
Example 8.14
L et X ( e j ) denote the Fourier transform of the signal x ( n ) depicted in Fig. Perform the
following calculations without explicity evaluating X ( e j ):
( a ) Evaluate X ( e j 0 )
( b ) Find X ( e j )
( c ) Evaluate X( e j )d
( d ) Find X ( e j )

( e ) Determine and sketch the signal whose Fourier transform is Real{ X ( e j )}
( f ) Evaluate | X( e j )|2 d
dX( e j )
( g ) Evaluate d
d
Solution 8.14
From Fig. we can observe that
x(n) { 1 0 1 2 1 0 1 2 1 0 1}, 3 n 7
x (n)
2
–3 0 7
–2 –1 1 2 3 4 5 6 n
–1
Fig. : Signal x(n) for Example

(a) we have
7
X( e j0 ) x( n) x( n) 6
n n 3
(b) We can observe that y(n) = x(n + 2) is real and even. Therefore (using the properties of the
DTFT) the DTFT Y(ej ) = y ( n ) is real and even. Hence, Y(ej ) has zero phase. We also know
that
y(n) = x(n + 2)
y (n ) = x ( n 2)
Y(e j ) = X(ej )ej2
or, equivalently,
X(e j ) = Y(e j )e –j
Since Y(ej ) has zero phase, we have
X(e j ) = –2
(c) we have
X( e j )d = 2 x(0) = 4
(d) we have
X (e j ) ( 1)n x( n)
n
7
( 1)n x( n)
n 3

= –x(–3) + x(–2) – x(–1) + x(0) – x(1) + x(2) – x(3) + x(4)

– x(5) + x(6) – x(7)
=1+0–1+2–1+0–1+2–1+0+1=2
(e) we have
xe(n) Real{X(e j )}
Therefore, the desired signal is
x( n) x( n)
xe(n)
2
The signal is as shown in Fig.
x e( n)
2
0 7
–4 –3 –2 –1 1 2 3 4 5 6
n
(f) We have
| X( e j )|2 d 2 | x( n)|2
n
7
2 | x( n)|2 28
n 3
(g) we have
2
dX( e j )
d 2 | nx( n)|2
d n
2 | n|2| x( n)|2
n
7
2 | n|2| x( n)|2 316
n 3
8.7 DTFT of Periodic Signals

If a signal x[n] is periodic then to find DTFT we cannot apply eq.(2) because x[n] is not absolutely
summable. To find DTFT of periodic signals we use DTFS. Any discrete time periodic signal can be written as
x[n] Xk e j 0 nk
k N
2
Here 0 N
The DTFT of
1 2 ( ), 0 <2
Using frequency shifting property e j 0 n 2 ( – 0), 0 <2
ej 0 kn 2 ( – 0k), 0 <2

Xk e j 0 kn 2 Xk – 0k), 0 <2
x[n] Xk e j 0 kn
2 Xk ( k 0) ...(22)
k N k N
Since the DTFT is periodic with period 2 , it follows that X(e j ) consists a set of N impulses of strength
2 Xk, k = 0, 1, 2, ..., N – 1, repeated at intervals of N 0 = 2 . Thus, X(e j ) can be compactly written as
X(e j ) 2 Xk ( k 0) …(23)
k
Thus, the Fourier transform of a periodic signal is simply an impulse train with impulses located at
= k 0, each of which has a strength 2 Xk, and all impulses are separated from each other by 0.
Example 8.15
Find and sketch the Fourier transform of the discrete-time impulse train
x(n) (n mN )
m
depicted in Fig.
x ( n)
–2N –N 0 N 2N n
Fig. : Discrete-time impulse train
Solution 8.15
2
The signal is periodic with period N and frequency 0 . The given signal for one period may be written
N
as
1, n 0
x(n)
0, 1 n ( N 1)
= (n)
Therefore, the Fourier series coefficients are
N 1
1 jk 0 n
Xk x( n)e
Nn 0
N 1
1 jk(2 / N )n
( n)e
Nn 0
1 jk (2 / N )n 1
e
N n 0 N
The DTFT of a periodic signal is given by
X( e j ) 2 Xk ( k 0)
k
1
Substituting Xk into the above equation gives
N

1
X(e j ) 2 ( k 0)
k N
2 2 k
N k N
which is illustrated in Fig.
X (e j )
… …
0
2
N
Fig. : DTFT of a discrete-time impulse train
Example 8.16
Find the Fourier transform of the periodic signal

x ( n ) = cos( 0 n )
2
with 0 .
5
Solution 8.16
Consider the given signal
1 j 0n
1 j 0n
x(n) = cos( 0n) e e
2 2
We know that
ej 0n 2 ( 0 2 m)
m
Therefore,
1 1
X(e j ) [e j 0n ] [e j 0n
]
2 2
1 1
X(e j ) 2 ( 0 2 m) 2 ( 0 2 m)
2m 2m
1 2 1 2
X( e j ) 2 2 m 2 2 m
2m 5 2m 5
The DTFT X(e j ) is illustrated in fig.
X (e j )
… …
2
–2 – 0
0 0 2 – 0 2 + 0
Fig. : DTFT of x(n) = cos( 0n)

Example 8.17
Find the Fourier transform of the periodic signal

x ( n ) = sin( 0n )
2
with 0 .
5
Solution 8.17
Consider the given signal
1 j 0n
1 j 0n
x(n) = sin( 0n) e e
2j 2j
We know that
ej 0n 2 ( 0 2 m)
m
Therefore,
1 1
X(e j ) [e j 0n
] [e j 0n
]
2j 2j
1 1
X(e j ) 2 ( 0 2 m) 2 ( 0 2 m)
2j m 2j m
1 2 1 2
X(e j ) 2 2 m 2 2 m
2j m 5 2j m 5
The DTFT X(e j ) us illustrated in Fig.

X (e j )
j
…
2 – 0 – 0
2
… –2 0 2 +
0 0
–
j
Fig. : DTFT of x(n) = sin( 0n)
8.8 Inverse Discrete Time Fourier Transform

Case 1:
If in a question an expression of X(e j ) is given, and expression is rational function in ej with some
numerator and denominator for example
2 e j
X( e j )
e 2 j 5e j 7
Then in such case replace e j by z and we will get a function X(z), Now simply apply inverse
z-transform with ROC of X(z) such that it contain unit circle as DTFT of signal exist it must be
absolutely summable.

Case 2:
In a question X(e j ) is a simple function of , for example
2sin( / 10) j10

X(e j ) e
10
or X(e j ) = (u[ + 3] – u[ – 3])e j
Then in such cases to find x[n] we will apply equation
1
x[n] X( e j )e j n
d
2
X (e j )
Here X(e j ) = X (e j ) e j
Example 8.18
Find the DTFT time domain signal for these

1 e j
(a) X ( e j )
1 2e j
(b) X ( e j ) rect ,
2 c
(c) X ( e j ) = 1
(d) X ( e j ) = e – j 10
Solution 8.18
1 e j
(a) X (ej )
1 2e j
1 z1 1 z 1
X (z )
1 2z 1 1 2z 1
1 2z 1
The pole of X(z) is at –2, since ROC must contain unit circle ROC is |z| < 2. Thus signal is left
sided signal
1
1
–(–2)nu[–n – 1]
1 2z
1
z
and (–2)n–1 u[–n]
1
1 2z
x[n] = –(–2)nu[–n – 1] + (–2)n–1u[–n]
(b) X( e j ) rect ,
2 c
1, | | c
We know that rect
2 c 0, elsewhere
1
x[n] X( e j )e j n
d
2

c
1 j n
e d
2
c
1
(e j cn e j cn
)
2 jn
sin( c n)
x[n]
n
(c) X(e j ) = 1
We will not answer by applying eq.(1) but we know that
[n] 1
Thus x[n] = [n]
(d) X(e j ) = e –j10
Here also we will not get x[n] by applying eq.(1) but we know that
[n – 10] e–j10
x[n] = [n – 10]
8.9 Signal Transmission through LTI System

If x(n) and y(n) are the input and output of an LTI system with impulse response h(n), then
y(n) = x(n) * h(n)
Application of the time convolution property to the above equation yields [assuming that both x(n) and
h(n) are Fourier transformable]
[ x (1 n ) x ( 1 n )] = [ x ( n )]* [h( n )]
Y( e j ) = X(ej )H(ej ) …(26)
or
Y( e j )
H(e j ) …(27)
X( e j )
x(n) h(n) y(n)

j
X (e j ) H (e ) Y (e j ) X (e j )H (e j )
Fig.8.1 : LTI system analysis using DTFT

The use of the convolution property for LTI systems is demonstrated in Fig.8.1 Equation (26) can be
expressed in the polar form as
Y( e j ) X (e j ) H (e j )
| Y( e j )| e j | X (e j )|e j |H (e j )|e j
Y( e j ) X( e j ) H( e j )
| Y( e j )| e j | X( e j )|| H( e j )| e j
Therefore,
|Y(e j )| = |X(e j )||H(e j )| …(28)
and
Y( e j ) = X(e j ) + H(ej ) …(29)

REMEMBER During transmission, the input signal magnitude spectrum | X ( e j )| is changed to

j j j
|X( e )||H( e )|. Similarly, the input signal phase spectrum X ( e ) is changed to
X(e j ) + H(e j ). An input signal spectral component of frequency is modified in amplitude
by a factor |H(e j )| and is shifted in phase by angle H(e j ). Clearly |H(e j )| is magnitude
responce and H(ej ) is the phase response of the system. H(ej ) is called the frequency response
of the system.
When H(e j ) exist then system is stable, since the system will be causal only when all poles give
right sided signal. Thus here system will be stable and causal and cannot be causal alone.
8.9.1 Response of LTI system to Complex Exponentials

We have seen in z-transform that
LTI System
ej on ej on H (z ) z ej o
h[n] H(z)
Similarly if H(e j ) is given in the question then output for e j 0n will be H(ej 0).ej 0n .
If h[n] is real and it’s DTFT (H(e j 0 )) will be conjugate symmetric then
ej 0n
e j 0n
LTI System H( e j 0 )e j 0n H( e j 0
)e j 0n
cos[ 0n]
2 h( n ) H (e j ) 2
Since h[n] is real thus H(e j ) will be conjugate symmetric. Thus H(e j 0) = H *(e –j 0)
If H(e j 0) = |H(ej 0)|
then H(e –j 0) = |H(ej 0)| –
Thus output will be
ej 0n ej e j 0n j
e
| H( e j 0 )| = cos( 0n + )|H(e j 0)|
2
LTI System
cos[ 0n] H(ej 0).cos[ 0n] = |H(ej 0)|cos( 0n + )
j
h( n ) H (e )
Similarly
LTI System
sin[ 0n] H(ej 0).sin[ 0n] = |H(e j 0)|sin( 0n + )
j
h( n ) H (e )
REMEMBER In above analysis we have taken h[n] as real, generally h[n] is real and if not specified in
question then assume h[n] is real.
8.9.2 Solving Difference Equation of System

If in the question a difference equation of system in given then even if the terms are given in DTFT
format then change the entire equation into z-transform and then apply z-transform analysis to find result.

Example 8.19
Consider a discrete time LTI system with input x(n) and output y(n) defined by difference
equation
3 1
y(n) - y(n - 1) + y(n - 2) = 2x(n)
4 8
(a) Find the impulse response of the system if DTFT of transfer function exist.
n
1
(b) Find the output of the system for input x(n) = u(n)
4
Solution 8.19
(a) We will use z-transform for analysis:
3 1
Y( z) Y( z)z 1 Y( z)z 2 = 2X(z)
4 8
Y( z) 2
H(z)
X( z) 1 3 z 1 1 z 2
4 8
16 16
H(z) 1 2
8 6z z (2 z )(4 z 1 )
1
Now to find h[n] we will apply inverse z-transform such that ROC of H(z) has unit circle
1 1 1
Since poles of H(z) are at and , since ROC must contain unit circle so ROC is | z | .
2 4 2
Thus both poles give right sided signal
8 8 4 2
H(z)
2 z 1
4 z 1 1 1 1 1
1 z 1 z
2 4
n n
1 1
h[n] 4 u[ n] 2 u[ n]
2 4
n
1
(b) To find output for input x[n] u[ n]
4
Since Y(z) = X(z) H(z)
1 16
1 1 (2 z )(4 z 1 ) 1
1 z
4
4 2 8
1 1 1
2 1 1
1 z 1 z 1 1 z
4 4 2
n n n
1 1 1
y[n] 4 u[ n] 2( n 1) u[ n] 8 u[ n]
4 4 2

Example 8.20
Determine the response of the system with impulse response

n
1
h(n) u( n )
2
when the input is the complex exponential sequence
x ( n ) = Ae j ( /2)n , – < n <
Solution 8.20
Given that
n
1
h(n) u( n)
2
Taking its DTFT, we obtain
1
H(e j )
1 j
1 e
2
At , the above equation yields
2
1
H(e j /2)
1 j /2
1 e
2
1 2 j 26.6
e
1 5
1 j
2
The response of the system to the complex exponential is given by
y(n) = AH(ej 0)ej 0n = AH(e j /2)e j /2n
2 j 26.6
A e ej /2 n
5
2
y(n) A e j( n /2 26.6 )
5
Example 8.21
Determine the response of the system with impulse response

n
1
h(n) u( n )
2
to the input signal
x(n) 5 5 sin n 10 cos( n ), n

2

Solution 8.21
Given that
n
1
h(n) u( n)
2
Taking its DTFT, we obtain
1
H(e j )
1 j
1 e
2
The first term in the input signal x(n) is a constant signal corresponding to = 0, thus
1
H( e j 0 ) 2
1
1
2
The second term in x(n) has a frequency . At this frequency, the frequency response of the system is
2
1 1 2 j 26.6
H(e j /2) e
1 j /2 1 5
1 e 1 j
2 2
Finally, the third term in x(n) has a frequency = . At this frequency, the frequency response of the system
is
1 1 1 2
H (e j )
1 j 1 3 3
1 e 1
2 2 2
Hence the response of the system to x(n) is
y(n) 5H( e j 0 ) H( e j /2
)5sin n H( e j )10cos( n)
2
10 20
y(n) 10 sin n 26.6 cos( n)
5 2 3
Example 8.22
If DTFT of a signal x [ n ] is X ( e j ) then find the phase of x [ n ] if X ( e j ) is

j
X(e )
A
… …
–2 0 2
4

Solution 8.22
The plot of X(e j ) show that it is real and X(e j( + /4)) is real and even thus time domain signal corresponding
to it will be real and even. Thus X(e j( + /4)) is real and even
If x[n] X(ej )
j n
x[ n]e 4 X(ej( + /4))
j n
Thus x[ n]e 4 will be real and even thus phase of x[n] will be n.
4
8.9.3 Distortionless System or Distortion less Transmission

Consider a discrete-time LTI system with impulse response h(n) and frequency response H(ej ). Let a
signal x(n) with Fourier transform X(e j ) be applied to the input of the system. Let a signal y(n) with fourier
transform Y(e j ) denote the output of the system. In several applications, such as signal amplification or message
signal transmission over a communication channel, we require that the output waveform be a replica of the
input waveform, i.e., distortionless transmission. As in the continuous-time case, transmission is said to be
distortionless if the input and the output have identical waveshapes within a multiplicative constant and a
constant time delay. Thus, in distortionless transmission, the input x(n) and the output y(n) satisfy the condition
(Fig. 8.2)
Input signal LTI Output signal
x (n ) System y (n ) = Gx (n – n d)
Fig 8.2 : Distortionless transmission from system

y(n) = Gx(n – nd) …(30)
where the constant G accounts for a change in amplitude and the constant nd accounts for a delay in
transmission. Taking the Fourier transform of the above equation yields
Y(e j ) = G X (e j )e j nd
Y( e j )
= H(e j ) = G e j nd
X( e j )
This is the frequency response required of a system for distortionless transmission. From this equation
it follows that
1. the magnitude response |H(e j )| must be a constant, i.e., we must have
|H(e j )| = G …(31)
for some constant G.
2. the phase response H(e j ) must be a linear function of with slope –nd and intercept zero, i.e.,
we must have
H(ej ) = – nd …(32)

|H(ej )| H(ej )
G
0 0
Slope = nd
(a) (b)
Fig 8.3 : (a) Magnitude response and (b) phase response of a system for distortionless transmission
Note: For distortionless transmission we need magnitude response of LTI system to be constant and phase response
to be linearly related to
8.9.4 Phase Delay and Group Delay
The two important parameters that characterize the form of the response y(n) of an LTI system excited
by an input signal x(n) composed of a weighted sum of sinusoidal signals are phase delay and group delay. These
two parameters are associated with the frequency response H(e j ) of the system. For a linear phase system
(whose phase varies linearly with frequency), both the phase delay and the group delay are constant.
Phase Delay
The time delay experienced by a single-frequency signal (i.e. sinusoidal signal) when it passes through
a system is referred to as system phase delay. Suppose input to system is cos( 0n + ) then output will be
LTI System
cos[ 0n + ] H(ej 0) cos[ 0n + ]
h[n] H(ej )
y[n] = |H(e j 0)|cos[ 0n + + H(ej 0)]
H( e j 0 )
| H( e j 0
)|cos 0 n
0
Thus we can see that delay in time is

H (e j 0
)
p( 0) …(33)
0
Group Delay
When the input signal constains many sinusoidal components with different frequencies that are not
harmonically related, each component will go through different phase delays when processed by a frequency
selective LTI system, and the signal delay is determined using a different parameter called the group delay, as
defined below:
d H (e j )
g( ) …(34)
d 0

The time delay g( ) is called the group delay or enveolope delay. Thus, the group delay at each frequency
equals the negative of the phase at that frequency. If g( ) is constant, all the components are delayed by the
same interval.
Example 8.23
Consider a discrete-time LTI system with frequency response H (e j ) and real impulse response
h ( n ). The group-delay function for such a system is defined as
d H( e j )
g( )
d
where H ( e j ) has no discontinuities. Suppose that, for this system,
|H(e j /2 ) | = 2, H ( e j0) = 0, g 2
2
Determine the output of the system for each of the following inputs:
7
( a ) cos n (b) sin n
2 2 4
Solution 8.23
(a) The input signal x(n) cos has a frequency 0

. Hence, the response of the system to x(n) is
2n 2
y(n) | H( e j 0 )|cos( 0(n g ( 0 )))
| H (e j /2
)|cos n g
2 2
y(n) 2cos ( n 2)
2
y(n) 2cos n
2
(b) Consider the given input signal
7
x(n) sin n
2 4
x(n) sin n
2 4
The input signal x(n) has a frequency 0

. Hence, the response of the system to x(n) is
2
y(n) = |H(ej 0)|sin( 0(n – g( 0)) + )
7
| H( e j /2
)|sin n g
2 2 4
7
y(n) 2sin ( n 2)
2 4

7
y(n) 2sin n 7
2 4
7
2sin n
2 4
7 3
y(n) 2sin n
2 4
Example 8.24
Consider a discrete-time low-pass filter whose impulse response h ( n ) is known to be real

and whose frequency response magnitude in the region – is given as
1, | |
|H(e j )| 4
0, otherwise
Determine the real-valued impuse response h(n) for this filter when the corresponding
group-delay function is specified as
5 5
(a) g( ) = 5, (b) g( ) , (c) g( )
2 2
Solution 8.24
Let H0(e j ) = |H(e j )|. Then the inverse DTFT of H0(e j ) is given by
1
h0(n) H0 ( e j )e j n d
2
/4 /4
1 j n 1 ej n
e d
2 /4
2 jn /4
1 ej /4n
e j /4n
n 2j
sin( / 4n)
h0(n)
n
d H (e j )
If g( ) (where is a constant), then H(ej ) =– + , where is a constant. Given
d
that h(n) is real, therefore H(e j ) is an odd function, and hence for H(e j ) is an odd function, and hence
for H(e j ) to be an odd function, the constant = 0. Therefore,
H( e j )
H(e j ) | H( e j )| e j H0 ( e j )e j
Taking the inverse DTFT, we obtain

h(n) = h0(n – )

sin (n )
4
h(n)
(n )
(a) Given that g( ) = 5 = . Substituting = 5 in the expression of h(n) yields
sin ( n 5)
4
h(n)
( n 5)
5 5
(b) Given that g( ) . Substituting in the expression of h(n) yields
2 2
5
sin n
4 2
h(n)
5
n
2
5 5
(c) given that g( ) . Substituting in the expression of h(n) yields
2 2
5
sin n
4 2
h(n)
5
n
2
8.10 Discrete Domain filters

We have seen that discrete frequency axis is periodic with period 2 , thus in discrete frequency we can
say that
0 DC
+ (highest frequency)
– – (highest negative frequency)
Thus if we have any filter whose impulse response is h[n] and DTFT of h[n] is H(e j ) then to find type of
filter we should follow these steps
Step 1 : Find the magnitude of H(e j )| =0 i.e. |H(e j0)|
let A = |H(e j0)|
Step 2 : Find the magnitude of H(e j )| = and –
Let B = |H(e j )| or |H(e –j )|
generally h[n] is real so |H(e j )| and |H(e –j )| will be same
Step 3 :
Case-1: If A > B and B is close to zero then we can say that filter passes low frequency component
of input and attenuates the high frequency component thus it is a low pass filter
Case 2. If B > A and A is close to zero then the system will attenuate low frequency component in
input and will pass high frequency component in input. Thus it act as a high pass filter.
Case-3: If A and B both are small and close to zero then system will alternate both low and high
frequency component in input but surely |H(ej )| will be greater than 1 at some frequency thus
system act as a band pass filter.

Case-4: If A and B both are greater than 0 then system may be band stop filter that is the filter will
attenuate some frequency of input or an all pass filter which pass all frequency of input without
alternation.
Following are the characteristics of these filters:
1. Low-pass filters are those characterized by a passband that extends from = 0 to = c, where c
is called the cutoff frequency of the filter [Fig. 8.4(a)]. A low-pass filter attenuates high-frequency
components of the input and passes the low-frequency components.
|H(e j )|
– – c
0 c
(a)
Fig 8.4 : (a) Low-pass filter
2. High-pass filters are characterized by a stopband that extends from = 0 to = c and a passband
that extends from = c to [Fig. 8.4(b)]. A high-pass filter attenuates low frequencies and passes
the high frequencies.
|H(e j )|
– – c
0 c
(b)
Fig 8.4 : (b) High-pass filter
3. Bandpass filters are characterized by a passband that extends from = 1 to = 2 and all other
frequencies are stopped [Fig. 8.4(c)]. A bandpass filter passes signals within a certian frequency
band and attenuates signals outside that band.
|H(e j )|
– – 2 – 0 – 1
0 – 2 – 0 – 1
(c)
Fig 8.4 : (c) Bandpass filter
4. Bandstop filters stop frequencies extending from = 1 to = 2 and pass all other frequencies
[Fig.8.4 (d)]. A bandstop filter attenuates signals within a certain frequency band and passes signals
outside that band.

|H(e j )|
– – 2 – 0 – 1
0 – 2 – 0 – 1
(d)
Fig 8.4 : (d) Bandstop filter

5. An all-pass filter is characterized by a magnitude response that is constant for all frequencies
[Fig.8.4(e)]. The characteristics of an all-pass filter are completely determined by its phase-shift
characteristics. An all-pass filter has nonconstant group delay and different frequencies in the
input are delayed by different amounts.
|H(e j )|
– 0
(e)
Fig 8.4 : (e) All-pass filter
Note: If in a question H(z) is given and we have find the type of filter then simply replace z by e j and follow the
procedure
Properties of Ideal all pass filter :

General transfer function is
N
i
bi z
i 0
H(z) N
i
ci z
i 0
For all pass filter we need magnitude response to be constant for all , here order of numerator will be
same as denominator. Thus for all pass filter (and generally c0 = 1).
N
i
c N* i z
i 0
H(z) N
1
ci z
i 0
That is for first order N = 1
c1* c 0* z 1
c1* z 1
H(z) 1 1
c 0 c1z 1 c1z
1
The pole is at (–c1) and zero at .
c1*

1
Tor all pass filter if pole is at pi then zero will be at zi .
pi*
8.11 Compiling Properties of DTFT

Table 8.1 Properties of the Discrete Time Fourier Transform
Property Signal Fourier transform
x[ n] X( e j ) periodic with
y[ n] Y( e j ) period 2
Linearity ax[n] + by[n] aX(e j ) + bY(e j )
j n0
Time Shifting x[n – n0] X (e j )e
Frequency shifting x ( n )e j 0n
X (e j 0
)
Conjugation x*[ n ] X *(e –j )
Time Reversal x [– n ] X( e – j )
n
x , if n multiple of k
Time Expansion x(k) [n ] k X(e jk )
0, if n multiple of k
Convolution x[n] * y[n] X(e j )Y(e j )
1
Multiplication x[ n ] y [ n ] X( e j )Y( e j( )
)d
2 2
Differencing in Time x[n] – x[n – 1] (1 – e –j )X(e j )

n
1
Accumulation x{k}
j
X( e j )
k 1 e
X( e j 0 ) ( 2 k)
k
dX( e j )
Differentiation in Frequency nx [n ] j
d
X (e j ) X *(e j
)
j j
e{X (e )} e{X (e )}
j j
Conjugate Symmetry for Real x[n] real m{X (e )} m{X (e )}
Signals | X (e j )| | X (e j
)|
j j
X (e ) X (e )
Symmetry for Real, Even x[n] real an even X(e j ) real and even
Signals
Symmetry for Real, odd x[n] real an odd X(e j ) purely imaginary and
Signals odd
Even-odd Decomposition of xe[n] = v{x[n]} [x[n] real] e{X(e j )}
Real Signals xe[n] = Od{x[n]} [x[n] real] j m{X(e j )}
Parseval’s Relation for Aperiodic Signals
1
| x[ n]|2 | X( e j )|2 d
n 2 2

If x[n] is discrete time signal and it’s DTFT is X(e j ) then for
The domain signal DT FT
real Conugate symmetric
imaginary Conjugate anti symmetric
odd odd
even even
real and odd imaginary and odd
imaginary and even imaginary and even
imaginary and odd real and odd
REMEMBER In above expression we have double sided arrow that is if time domain signal is real then DTFT
is conjugate symmetric and if time domain signal is conjugate symmetric then DTFT will be real
Table 8.2 Sumary of Fourier Series and Transform Expressions
Continuous time Continuous time

Time domain Frequency domain Time domain Frequency domain
ak e jk 0t 1 1
x(t ) ak x(t )e jk 0 t
x[n] ak e jk(2 / N )n
ak x[ n]e jk(2 / N ) n
Fourier k T0 T k ( N) Nk (N )
0
Series
continuous time discrete frequency discrete time duality discrete frequency
periodic in time aperiodic in frequency du periodic in time periodic in frequency
al
ity
1 1
x(t ) X( j )e j t d X( j ) x(t)e j t
dt x[ n] X( e j )e j nd X( e j ) x[n]e j n
2 2 n
Fourier 2
Transform
continuous time duality continuous frequency discrete time continuous frequency

aperiodic in time aperiodic in frequency aperiodic in time periodic in frequency
8.12 Energy Spectral Density

Parseval’s theorem relates the total signal energy in a signal x(n) to its Fourier transform through
1/2
1
Ex | x( n)|2 | X( e j )|2 d | X( e j )|2 df
n 2 1/2
Parseval’s theorem states that the total energy Ex may be determined either by computing the energy
| X( e j )|2
per unit time (|x(n)|2) and summing over all time or by intergrating the energy per unit frequency
2
over a full 2 interval of distinct discrete-time frequencies. For this reason |X(e j )|2 represents energy per unit
bandwidth and is often referred to as the energy spectral density or energy density spectrum of the signal x(n)
and is denoted by x(e j ). Hence,
j ) = |X(e j )|2
x (e …(35)
For discrete-time signals, the unit of ESD is simply the square of the signal unit (whatever that may be)

8.12.1 Relationship between input and outpud ESDs of an LTI System

Consider an LTI system with frequency response H(e j ), input x(n), and output y(n). If x(n) and y(n)
are energy signals, then their ESDs are x(e j ) = |X(e j )|2 and y(e j ) = |Y(e j )|2, respectively. Since, we
know that
Y(e j ) = H(e j )X(e j )
it follows that
|Y(e j )|2 = |H(e j )X(e j )|2
|Y(e j )|2 = |H(e j )|2|X(e j )|2
j ) = |H(e j )|2 j
y(e x(e ) …(36)
8.12.2 Relation of ESD to Autocorrelation

The autocorrelation function Rxx(m) of a real energy signal is defined as
Rxx(m) x( n)x( n m)
n
Rxx(m) = x(m) * x(–m)

taking the Fourier transform of the above equation, we have
[R xx ( m )] = X(e j )X(e –j )
= X( e j ) X * ( e j )
[R xx ( m )] = |X(e j )|2
j
[R xx ( m )] = x(e )
Rxx(m) j
x(e ) …(37)
Thus, the autocorrelation function Rxx(m) and ESD makes a Fourier transform pair.
8.13 Power Spectral Density (PSD)

PSD has the same relation to power signals as ESD has to energy signals. The expression of PSD may be
obtained by assuming the power signal as a limiting case of an energy signal.
Consider a power signal of infinite duration. Since the total energy of a power signal cannot be found, let
us first find the ESD of a truncated version xN(n) of the infinite duration signal x(n). The signal xN(n) may be
expressed as
x( n), N n N
xN(n) …(38)
0, otherwise
The truncated signal xN(n) is of finite duration, therefore, it is an energy signal. Now, if xN(n) XN(e j ),
then using the Parseval’s theorem, we have
1
ExN | xN ( n)|2 | XN ( e j )|2 d
n 2
Substituting the value of xN(n) from Eq. (38) into the above equation gives

N
1
| x( n)|2 | XN ( e j )|2 d
n N 2
N
1 1 1
lim | x( n)|2 lim | XN ( e j )|2 d
N 2N 1 n N N 2N 1 2
N
1 1 | XN ( e j )|2
lim | x( n)|2 lim d
N 2N 1 n N 2 N 2N 1
The LHS of the above equation represents the average power Px of the signal x(n). Therefore,
N
1
Px lim | x( n)|2
N 2N 1 n N
1 | XN ( e j )|2
lim d
2 N 2N 1
1
Px Gx ( e j )d …(39)
2
where
| XN ( e j )|2
Gx (e j ) lim …(40)
N 2N 1
is the PSD. The units of PSD depend on the units of the underlying signal x(n).
8.13.1 Relationship Between input and output PSDs of an LTI System

Consider an LTI system with frequency response H(e j ), input xN(n), and output yN(n). If x(n) and y(n)
are power signals, then their PSDs are
| XN ( e j )|2
Gx (e j ) lim
N 2N 1
| YN ( e j )|2
and Gy (e j ) lim
N 2N 1
respectively. Since, we know that
YN(e j ) = H(e j )XN(e j )
it follows that
|YN(e j )|2 = |H(e j )XN(e j )|2
|YN(e j )|2 = |H(e j )|2|XN(e j )|2
| YN ( e j )|2 | XN ( e j )|2
lim | H( e j )|2 lim
N 2N 1 N 2N 1
Gy(e j ) = |H(e j )|2Gx(e j ) …(41)
8.13.2 Relation of PSD to Autocorrelation

The autocorrelation function Rxx(m) of a power signal is defined as
N
1
Rxx(m) lim x( n)x( n m)
N 2N 1 n N

1
lim x N ( n)xN ( n m)
N 2N 1 n
1
Rxx(m) lim [ x N ( m)* x N ( m)]
N 2N 1
taking the Fourier transform of the above equation, we obtain
1
[R xx ( m )] lim XN ( e j )XN ( e j )
N 2N 1
1
lim XN ( e j )XN* ( e j )
N 2N 1
1
[R xx ( m )] lim | XN ( e j )|2
N 2N 1
[R xx ( m )] = Gx(e j )
Rxx(m) Gx(e j ) …(42)
Thus, the autocorrelation function Rxx(m) and PSD makes a Fourier transform pair.
8.14 Relationship between DTFT and DTFS

We know that DTFS give discrete periodic Xk and DTFT give continuous periodic X(e j ). DTFS is
2
calculated for periodic signals with period N and 0 , and
N
x[n] Xk e j 0 kn
k N
for DTFT
1
x[n] X( e j )e j n d
2 2
from here we can see that Xk i.e. DTFS coefficients of periodic signal are samples of X(e j ) taken at
integral multiple of 0.
1
Xk X( e j )| k 0
2
In period of 0 to 2 we will get N samples only and Xk will also become periodic.
Example 8.25
Determine the Fourier transform for – < in the case of each of the following periodic
signals:
( a ) x1(n ) sin n
3 4
( b ) x 2( n ) 2 cos n
6 8
Solution 8.25

(a) x1[n] sin n

3 4
j n j j n j
e 3 e 4 e 3 e 4
2j
Since 1 2 ( ) – < <
j n
e 3 2 – < <
3
j n
and e 3 2 – < <
3
j j
e 42 e 42
3 3
sin n
3 4 2j 2j
3
j j
X1 (e j ) e 4 e 4
– < <
3 3
(b) Here
x[n] 2 cos n
6 8
j n j j j
e 6 e 8 e 6e 8
2
2 2
We know that
Since 2 4 ( ), – < <
j n
e 6 2 , – < <
6
j n
and e 6 2 – < <
6
j j
x[n] 4 ( ) e 8 e 8 – < <
6 6
Example 8.26
Determine the inverse Fourier transform of
( a ) X1( e j ) 2 ( 2 m) 2 m 2 m
m 2 2
2 j, 0
( b ) X2 ( e j )
2 j, 0
Solution 8.26

(a) The inverse DTFT x1(n) of X1(e j ) is given by

1
x1(n ) X1( e j )e j n d
2
1
2 ( ) ej nd
2 2 2
1 j 1 j
ej n
e n
e n
0 2 2
2 2
1 j 1
e j0 e /2 n
e j /2 n
2 2
x1(n ) 1 cos n
2
(b) The inverse DTFT x2(n) of X2(e j ) is given by
1
x2(n ) X2 ( e j )e j n d
2
1 1
2 je j n d 2 je j n d
2 2 0
j n 0 j n
je je
jn jn 0
1 jn 1
[1 e ] [e jn 1]
n n
1
[e jn 1 1 e jn
]
n
1
[e jn e jn
2]
n
1
[2 cos( n ) 2]
n
2
[cos( n ) 1]
n
4 n
x2 ( n ) sin2
n 2
Example 8.27
Given that x ( n ) has Fourier transform X ( e j ), express the Fourier transform of the following
signals in terms of X ( e j ):
( a ) x1( n ) = x (1 – n ) + x (–1 – n )
x * ( n ) x( n )
( b ) x2 ( n)
2
( c ) x 3 ( n ) = ( n – 1) 2 x ( n )

Solution 8.27
Given that
x(n) X(e j )
(a) Using the time-shifting property, we have
x(n + 1) X(e j )e j
and x(n – 1) X(e j )e –j
Using the time-reversal property on this, we have
x(–n + 1) = x(1 – n) X(e –j )e –j
and x(–n – 1) = x(–1 – n) X(e –j )e j
Therefore,
[ x 1( n )] = [ x (1 n ) x ( 1 n )]
X1(e j ) = X(e –j )e –j + X(e –j )e j
X1(e j ) = 2X(e –j )cos( )
(b) Using the conjugation property, we have
x*(n) X *(e –j )
Using the time reversal-property on this, we have
x*(–n) X *(e j )
Therefore,
x *( n ) x ( n )
[x2(n)]
2
X *( e j ) X( e j )
X 2( e j )
2
= {X(e j )} = X (e j )
X2 (e j
) = X (e j )
(c) Consider the given signal x3(n) = (n – 1)2 x(n) = n2x(n) – 2nx(n) + x(n).
Using the differentiation in frequency-domain property, we have
dX( e j )
nx(n) j
d
Using the differentiation in frequency-domain property again, we have
d dX( e j )
n[nx(n)] j j
d d
d2 X( e j )
n2x(n)
d
Therefore,
[x3(n)] = [(n – 1)2x(n)]
= [n2x(n) – 2nx(n) + x(n)]
d2 X( e j ) dX( e j )
X3( e j ) 2j X( e j )
d d
d2 X( e j ) dX( e j )
X 3( e j ) 2j X( e j )
d d

Example 8.28
Find the discrete-time Fourier transform of

( a ) x 1 ( n ) = cos( 0 n ) u ( n )
( b ) x 2 ( n ) = sin( 0 n ) u ( n )
Solution 8.28
(a) Consider the given signal
x1(n) = cos( 0n)u(n)
Using the multiplication property, we have
X 1( e j ) = [cos( 0n)]* [u(n)], – <
1
= [ ( – 0) + ( + 0)] * j
( )
1 e
1 1 2
j
[ ( 0) ( 0 )]
1 e 1 ej 0
X 1( e j ) = + 2[ ( – 0) + ( + 0)]
(b) Consider the given signal
x2(n) = sin( 0n)u(n)
Using the multiplication property, we have
X 2( e j ) = [sin( 0n)]* [u(n)], – <
1
[ ( 0) ( 0 )]* j
( )
j 1 e
2
1 1
j 0
[ ( 0) ( 0 )]
j 1 e 1 ej 0 j
sin( 0 ) 2
[ ( 0) ( 0 )]
1 cos( 0 )
0 0
2sin cos
2 2 2
[ ( 0) ( 0 )]
2 0
2sin
2
0 2
X 2( e j ) cot [ ( 0) ( 0 )]
2
Example 8.29
A linear time-invariant system is described by the following difference equation:

y ( n ) = ay (n – 1) + bx( n), 0 < a < 1
( a ) Determine the magnitude response | H ( e j )| and phase response H ( e j ) of the system
( b ) Choose the parameter b such that the maximum value of | H ( e j )| is unity.
Solution 8.29
(a) Consider the given difference equation

y(n) = ay(n – 1) + bx(n), 0 < a < 1

Taking the Fourier transform of the above equation we obtain
Y(e j ) = aY(e j )e –j + bX(e j )
Y( e j ) b
H( e j ) j
X( e j ) 1 ae
b
H( e j )
1 a cos( ) ja sin( )
Therefore,
| b|
|H(e j )|
(1 a cos( ))2 ( a sin( ))2
| b|
|H(e j )| 2
1 a 2a cos( )
1 a sin( )
and H(e j ) = angle of b tan
1 a cos( )
(b) Since the parameter a is positive, the denominator of |H(e j )| attains a minimum at = 0. Therefore,
|H(e j )| attains its maximum value at = 0. At this frequency, we have
| b|
|H(e j0)| 1
1 a
|b| = 1 – a
b = (1 – a)
Example 8.30
Consider a discrete-time LTI system with frequency response
j2 j4
1 e 4e j
4
H(e j ) e
1 j2
1 e
2
Determine the Fourier transform of the output if the input is
n
x(n) cos
2
Solution 8.30
Consider the given input
n
x(n) cos
2
Taking the Fourier transform of the above equation, we obtain
X(e j )
2 2
Fourier transform of the output is given by
Y(e j ) = H(e j )X(e j )

H( e j )
2 2
Y(e j ) H( e j /2
) H( e j /2
)
2 2
Given that
j2 j4
1 e 4e j
4
H( e j ) e
1 j2
1 e
2
Therefore,
2 4
j j
1 e 2 4e 2 j
2 4
H(e j /2) 2
e
1 j
1 e 2
2
j j2
1 e 4e j
e 4
1 j
1 e
2
1 1 4 j
H(e j /2) e 4
1
1
2
j /2
H(e ) = 8e–j /4
Similarly, we can obtain

H(e –j /2) = 8e j3 /4
Substituting the values of H(e j /2) and H(e –j /2) in he expression of Y(e j ), we obtain
3
j j
Y(e j ) 8 e 4 e 4
2 2

where * denotes convolution and | c| . Determine

a stricter constraint on c which ensures that
2
1 Subjective Practice Problems y[n]

sin n
4
n
1 . Use the Fourier transform synthesis equation to 6 . Suppose we are given the following facts about an
determine the inverse Fourier transform of LTI system S with impulse response h [ n ] and
X( e j ) , frequency response H(e j ):
X(e j ) | X( e j )| e j where
n
1
1. u[ n] g[ n], where g[n] = 0 for n 2
1, 0 | | 4
4 3 and n < 0
|X(e j )| and X( e j ) .
0,| | 2 2. H(e j /2) = 1.
4 3. H(e j ) = H(e j( – )).
Use your answer to determine the values of n for Determin h[n].
which x[n] = 0.
7 . Let the inverse Fourier transform of Y(e j ) be
2 . For each of the following Fourier transforms, use 2
Fourier transform properties to determine whether sin cn
y[n] .
the corresponding time-domain signal is (i) real, n
imaginary, or neither and (ii) even, odd, or neither. where 0 < c < . Determine the value of c which
10
ensures that
j
(a) X1(e j ) e (sin k ) 1
k 1 Y(e j ) .
2
(b) X2(e j ) = jsin( ) cos(5 )
(c) X3(e j ) = A( ) + e jB( ) where 8 . Given the fact that
1 a2
1, 0 | | a|n| , | a | 1,
8 3 1 2a cos a2
A( ) and B( ) .
0, | | 2 use duality to determine the Fourier series coefficients
8 of the following continuous time signal with period
3 . Determine x[n] when its Fourier transform is T = 1.
3 1
sin x(t)
1 2 5 4cos(2 t )
X(e j ) j
5 ( ),
1 e sin 9 . The following are the Fourier transforms of discrete-
2
time signals. Determine the signal corresponding to
4 . The following four facts are given about a real signal
each transform.
x[n] with Fourier transform X(e j ):
1. x[n] = 0 for n > 0. 3
1, | |
2. x[0] > 0 4 4
(a) X(e j )
3. m{X(e j )} = sin – sin2 . 3
0, | | ,0 | |
4 4
1
4. | X( e j )|2 d 3. (b) X(e j ) = 1 + 3e–j + 2e–j2 – 4e–j3 + e–j10
2
(c) X(e j ) = e–j /2 for –
Determine x[n] (d) X(e j ) = cos2 + sin23
5 . Let 10. Determine which, if any, of the following signals have

2 Fourier transforms that satisfy each of the following
sin n
4 sin cn conditions:
y[n] * .
n n 1. e{X(e j )} = 0.

2. m{X(e j )} = 0 11. Consider the signal

3. There exists an integer a such that e j X(e j ) is n n
real. x[n] sin 2cos
8 4
Suppose that this signal is the input to LTI systems
4. X( e j )d 0 with the following impulse responses. Determine the
output in each case.
5. X(e j ) periodic.
sin( n / 6)
6. X(e j0) = 0. (i) h[n]
n
(a) x[n] as in Figure (a) sin( n / 6) sin( n / 2)
(b) x[n] as in Figure (b) (ii) h[n]
n n
n
1 sin( n / 6)sin( n / 3)
(c) x[n] u[ n] (iii) h[n]
2 n
|n|
1 sin( n / 6)sin( n / 3)
(d) x[n] (iv) h[n] 2 2
2 n
(e) x[n] = [n – 1] + [n + 2]
12. Consider an LTI system with unit sample response
(f) x[n] = [n – 1] + [n + 3]
sin( n / 3)
(g) x[n] as in Figure (c) h[n]
n
(h) x[n] as in Figure (d)
Determine the output for each of the following inputs:
(i) x[n] = [n – 1] – [n + 1]
(i) x[n] = the square wave(amplitude =1) depicted
x [n]
in figure
2
3
2
(ii) x[n] [ n 8k ]
1 1 k
2
(iii) x[n] = (–1)n times the square wave depicted in
–1 0 1 2 3 4 5 6 n
Figure
(a) (iv) x[n] = [n + 1] + [n – 1]
x [ n] x [ n]
1
1
–8 0 8 16 n
–2 –1 0 2 n
–1
Fig.:
(b)
x [n] 13. Consider a system consisting of the cascade of two
LTI systems with frequency responses
2
j
2 e
1 H1( e j ) j
1 1 / 2e
0 n and
–1
1
(c) H2 ( e j )
1 j 1 j2
1 e e
x [ n] 2 4
(a) Find the difference equation describing the
2 overall system.
1 (b) Determine the impulse response of the overall
system
0 n
–1
14. A causal LTI system is described by the difference
(d) equation

y[n] – ay[n – 1] = bx[n] + x[n – 1], n

where a is real and less than 1 in magnitude. (ii) p[ n] cos
2
Find a value of b such that the frequency response n
of the system satisfies (iii) p[ n] sin
2
|H(e j )| = 1, for all .
This kind of system is called an all-pass system, as it (iv) p[ n] [ n 2k ]
k
does not attenuate the input e j n for any value of .
(v) p[ n] [ n 4k ]
15. Let k
n n X (e j )
x1[n] cos sin
3 2
be a signal, and let X 1( e j ) denote the Fourier 1
transform of x1[n]. Sketch x1[n], together with the
signals with the following Fourier transforms;
–2 – – 0 2 4
(i) X2(e j ) = X1(e j )e j , | | < 2 2
(ii) X3(e j ) = X1(e j )e –j3 /2, | |<
19. X(e j ) denotes the DTFT of a length-9 sequence x(n)
16. Determine whether each of the following statements given by
is true or false. Justify your answers. In each x( n) {2, 3, 1, 0, 4, 3, 1, 2, 4},
statement, the Fourier transform of x[n] is denoted
by X(e j ). –2 n 6
Evaluate the following functions of X(ej ) without
(a) If X(e j ) = X(e j( – 1)), then x[n] = 0 for |n| > 0 computing the transform itself:
(b) If X(e j ) = X(e j( – )), then x[n] = 0 for |n| > 0 (a) X(e j0)
(b) X(e j )
(c) If X(e j ) = X(e j /2)), then x[n] = 0 for |n| > 0
(c) X( e j )d
(d) If X(e j ) = X(e j2 ), then x[n] = 0 for |n| > 0
17. A particular discrete-time system has input x[n] and

(d) | X( e j )|2 d
output y[n] and output y[n]. The Fourier transforms
of these signals are related by the equation 2
dX( e j )
(e) d
j j j dX( e j )
j
d
Y( e ) 2 X( e ) e X( e )
d 20. Consider the filter
(i) Is the system linear? Clearly justify your answer. y(n) = 0.9y(n – 1) + bx(n)
(a) Determine b so that |H(e j0)| = 1.
(ii) Is the system time invariant? Clearly justify your
(b) Determine the frequency at which |X(e j )|
answer. 1
.
(iii) What is y[n] if x[n] = [n]? 2
(c) Is this filter low-pass, high-pass, or bandpass?
18. Let x [n] be a discrete-time signal with Fourier
transform X(e j ), which is illustrated in Figure.
Sketch the Fourier transform of
w[n] = x[n]p[n]
for each of the following signals p[n]:
(i) p[n] = cos n

Chapter
SAMPLING AND
HILBERT TRANSFORM
9
9.1 Introduction
Sampling is the process of converting any continuous time signal into discreate time signal. Sometimes
sampling is done on discrete signals. In this chapter we will study sampling of continuous and discrete time
signals. Hilbert transform as we have already studied in previous chapter is a system that provide 90° phase
shift to input signal.
9.2 Sampling for a low pass signal and sampling theorem

Low pass signal are those which have no frequency component higher than H rad/s and it is low pass signal
because it has non zero X( ) at = 0. Let us consider ideal sampling which is performed by multiplying time
domain signal with impulse train
x(t) x ( nTs ) (t nTs )

n
(t nTs )
n
Figure 9.1 Sampling process is simply multiplying signal x ( t ) with impulse train .
sampler
x(t)
TS
x ( nTs ) (t nTs )
n
x(t) x(t)
x(3TS)
x(2TS)
x(TS)
0 t 0 TS 2TS 3TS 4TS 5TS t
Figure 9.2 Sampling continuous time signal and sampled signal

Now let us look frequency analysis of the samplying process
x(t)
X( )
0 t h 0 h
s(t) S( )
2TS TS 0 TS 2TS t 2 S S
0 S 2 S
x (t)
x(3TS) X( )
x(2TS)
fs
x(TS)
0 TS 2TS 3TS 4TS t

s 0 s s
Figure 9.3 (a) signal x(t), spectrum of x(t), (c) sampling signal, (d) spectrum of sampling signal, (e)
sampled signal and (f) spectrum of the sampled signal
S(t) = (t nTs )
n
2π
S(t) CTFT
S( ) = T ( n s)
s n
CTFT 1
Since x (t) = x (t ) S (t ) S( )* X ( )
2π
1
X ( )* ( n s)
TS n
fs X( n s )
n

Sampling and Hilbert Transform 423
X ( ) = fs X( n s ) ... (1)
n
Thus we can see that X ( ) will not be distorted until and unless s 2 h because if this condition is not
satisfied then shifted X( ) will overlap over each other and we cannot recover X( ) from X ( ) or x(t) from
x (t).
REMEMBER Sampling theorem states that, a low pass signal should be samples at s 2 h . s 2 h is
called Nyquist sampling rate
Study Note
Now when s 2 h then the shifted version of X( ) will overlap over each other and then we cannot recover X( )
from Xd(w) and this is called aliasing
For recovery of signal

C ase1.To get back x(t) from x (t) or X( ) from X ( ) we need to pass signal from low pass filter with cut off
frequency C such that h < C< s - h. The pass gain of low pass filter should be TS.
X( )
x(t) x (t) LPF x(t)

X 1
H( ) X( )=X ( )H( )
X( )
s(t) h 0 h
X( )
fs
s h 0 s s
X( )=X ( ) H( ) H( )
1
TS
h 0 h c 0 c
Figure 9.4 Exact recpvery of a continuous-time signal form its samples by using an ideal LPF
Case 2. s = 2 h.
When s = 2 h, there is no overlap between the shifted replicas of X( ). Consequently x(t) can be recov-
ered exactly from x (t) by means of an ideal low-pass filter with gain Ts and a cutoff frequency C = h, as

depicted in figure 9.5
X( )
fs
s 0 h s s s
H( ) X( )=X ( )H( )
1
TS
h 0 h h 0 h
Figure 9.5 Exact recovery of x ( t ) from its samples by using an ideal LPF of cut-off frequency c = h
REMEMBER When S< 2 then aliasing take place. Overlap between the replicas of X( ) centred at = 0
h
and at = s occurs of frequencies between s h and h. These replicas add, and thus the
basic shape of the spectrum changes from portions of a triangle to a constant,as shown in figure.
Note that the spectra cross at frequency s/2. This frequency is called the folding frequency. The
spectrum, therefore onto itself at the folding frequency. The components of frequencies above
s/2 reappear as components of frequencies below s/2. This tail inversion is known as spectral
folding or aliasing.
X( )
fs
S S 0 S S S
S h
h
Aliasing
In question where we have to find Nyquist sampling of a signal we should follow these rules
We know that Nyquist sampling rate s is 2 Bandwidth of the signal. So we have to find bandwidth of
the signal.
Bandwidth of (x1 (t) + x2(t) + x3(t)...) is max{B1, B2, ...} where B1, B2, B3.... are bandwidth of x1(t),
x2(t)....
Bandwidth of x1(t) * x2(t) is max(B1, B2) because convolution in time domain give multiplication in
fourier domain
Bandwidth of x1(t). x2(t) is (B1 + B2) because multiplication in time domain give convolution in
frequency domain
Cos( ot), sin( ot) have bandwidth o rad/s.

sin( xt )
s in c( xt ) = has bandwidth x rad/s.
xt
sin( xt )
Sa( xt )= has bandwidth x rad/s.
xt
If band width of x(t) is B then that of x(at) will be aB and shifting of signal donot effect the bandwidth.
Example - 9.1
Determine the Nyquist rate corresponding to each of the following signals:

( a ) x ( t ) = sin(200 t )
( b ) x ( t ) = sin 2 (200 t )
( c ) x ( t ) = 1 + cos(200 t ) + sin(400 t )
( d ) x ( t ) = cos(150 t ) sin(100 t )
( e ) x ( t ) = cos 3 (200 t )
Solution 9.1
(a) Here x(t ) = sin (200 t) O = 200 rad/s
Bandwidth is 200 rad/s = 100Hz
fS = 2 fB = 200 Hz
200 sample/sec is nyquist rate
(b) Here x(t ) = sin2(200 t)

Signal sin (200 t) has bandwidth 100 Hz thus band width of sin2 (200 t) will 2 100 Hz.
Thus fS = 2 200 Hz = 400 Hz
400 sample/sec is nyquist rate
(c) Here x(t ) = 1 + cos(200 t) + sin (400 t)

Bandwidth of cos(200 t) is 100 Hz and that of sin(400 t) is 200 Hz. Thus bandwidth of x(t) is
200 Hz.
Nyquist rate is 400 Hz.
(d) Here x(t ) = cos(150 t) sin(100 t)

Band width of cos(150 t) is 75 Hz and that of sin(100 t) is 50 Hz thus band width of x(t) will be (75
+ 50) = 125 Hz. Thus nyquist rate is 250 samples/sec
(e) Here x( t ) = cos3(200 t)
The band width of cos(200 t) is 100 Hz and that of x(t) will be (3 100) = 300 Hz. Thus sampling
rate will be 600 samples/sec.
Example - 9.2
Specify the Nyquist rate and Nyquist interval for each of the following signals :
( a ) x(t) = sinc(200 t )
( b ) x(t) = sinc 2 (200 t )
( c ) x(t) = sinc(200 t ) + sin 2 (200 t )
Solution 9.2
sin(π200t )
(a) Here x(t ) = sinc(200t) =
200πt
The band width of x(t) is 200 rad/s that is 100 Hz. The nyquiist sampling rate is 200 Hz that is 200

samples/sec
(b) Here x(t ) = sinc2(200t)

The band width of sinc(200t) is 200 rad/sec that is 100 Hz, this band width of sinc2(200t) will be
200 Hz. Thus nyquist rate is 400 Hz that is 400 samples/sec
(c) Here x(t ) = sinc(200t) + sinc2(200t)

The band width of x(t) is max(Band width of sinc(200t), Band width of sinc2(200t))
Thus band width of x(t) will be 200 Hz, nyquist rate will be 400 Hz that is 400 samples/sec
Example - 9.3
L et x ( t ) be a signal with Nyquist rate o . Determine the Nyquist rate for each of the
following signals :
(a) x ( t + 1) (b) x ( t ) + x ( t + 1)
dx(t)
(c) (d) x 2 ( t )
dt
t
(e) x ( t ) cos( ot ) (f) x( )d
Solution 9.3
Here we will take X( ) as fourier transform of x(t).
(a) Shifting operator don’t change bandwidth thus will not change sampling frequency also
Bandwidth of x(t + 1) = Bandwidth of x(t)
Nyquist rate of x(t + 1) is o
(b) Here x(t) + x(t 1) X( ) + X( ) e

The band width of X( ) and X( ) (1 + e ) remain same so Nyquist rate of x(t) + x(t 1) will also be
same. Thus Nyquist rate x(t) + x(t 1) is
dx(t)
(c) Here j X( )
dt
dx(t)
The band width of X( ) and j X( ) will be same so Nyquist rate of remain same as o
dt
1
(d) Here x2 ( t ) X ( )* X ( )
2π
X ( )* X ( )
The bandwidth of will be double that of bandwidth of X( ). Thus nyquist rate of x2(t) will
2π
be 2 o.
(e) Here x(t) cos( ot) [X( o) + X( + o)]

Since Bandwidth of X( ) is ( o/2) as nyquist rate is o, the bandwidth of [X( o) + X( + o)]
nyquist rate is 2 o
(f) Here x(τ)dτ = x (t ) * u (t ) X ( ).U ( )

1
U( ) = π ( )+
j
The band width of U( ) is infinity, X( ) is ( o/2) thus bandwidth of X( ). U( ) will be ( o/2) and
nyquist rate will be o.
9.4 Sampling Techniques.

We have already discussed instantaneous sampling, the ideal sampling where impulse train is used for
sampling. Here we will discuss two more practical sampling techniques
1. Natural sampling
2. Flat tap sampling
9.3.1 Natural sampling

Although instantaneous sampling is a convenient model, a more practical way of accomplishing the sam-
pling of a band-limited continuos-time signal x(t) is to multiply x(t), shown in figure (a), by the sampling signal
(pulse train or switching waveform) s(t), shown in figure (c). Each pulse in the sampling signal s(t) has width T
and amplitude 1/T. Multiplication by s(t) can be viewed as the opening and closing of a switch as shown
x(t)
X( )
0 t h 0 h
a b
|S( )|
s(t)
S S S
S 2
1
S
1/T S 3 S
T T t
2TS TS TS 2TS S S S
0 S S
2 2 S
(c) (d)
|XS( )|
xS(t)
0 TS 2TS 3TS 4TS t

S S 0 S S
(e)
(f)
Figure 9.6 Natural sampling

In figure 9.7. When s(t) goes high, a switch ‘S’ is closed and the sampled signal xs(t) = x(t), and when s(t)
goes low, the switch ‘S’ is open and the sampled signal xs(t) = 0. As before, the sampling frequency is
designated s(or fs), and Ts is the sampling interval. The resulting sampled signal xs(t) is illustrated in
figure 9.6(e) and is expressed as
xs ( t ) = x ( t ) s ( t ) ... (2)
Study Note
The sampling here is termed natural sampling, since the top of each pulse in hte sampled signal xs(t) retains the
shape of the signal x(t) during the pulse interval.
x(t) s(t) xS(t)
Figure 9.7 Natural sampler

To perform frequency analysis of natural sampling, we first of all find fourier transform of s(t). Since s(t) is
periodic signal thus using fourier series
s(t) = Sk e j o kt
...(3)
k
2π
Where Sk are CTFS coefficients of s(t) and o = we can calculate Sk.
Ts
1 kT
Sk = sin c ... (4)
Ts Ts
The fourier transform of s(t) will be
S( ) = 2π Sk ( k s ) ....(5)
k
The fourier transform of sampled signal

1
xs(t) = x(t). s(t) X ( )* S( )
2π
X ( )* Sk ( k s )
k
Xs ( ) = Sk X ( k s ) ....(6)
k

Thus Xs( ) is Xs( )= ...S 1 X( + s) + So X( ) + S1 X( s)...

That is Xs( ) is formed by repeating X( ) by shifting it by integer multiple of s and multiplying with factor
Sk. We can easily recover X( ) from Xs( ) by passing Xs( ) through a low pass filter with pass band gain 1/So.
Thus in natural sampling the sampled signal is not distorted and we can easily recover the signal x(t).
REMEMBER The natural sampling is difficult to generate as the sampled signal has pulse with varying amplitude
by. The noise effect is also significant on signal produced by natural
9.3.2 Flat Top Sampling

Flat-top sampling is somewhat similar to natural sampling, where the continuous-time signal x(t) is multi-
plied by a periodic rectangular pulse train. However, in natural samping the top of each rectangular pulse in the
sampled signal xs(t) varies with the signal x(t), whereas in flat top sampled signal the top of each rectangular
pulse remains constant and equal to instantaneous value of the signal x(t) at the start of sampling. There are
two operations involved in the generation of a flat-top sampled signal:
1. Instantaneous sampling of the continuous-time signal x(t) every Ts seconds, where the sampling rate
fs = 1/Ts is chosen in accordance with the sampling theorem.
2. Lengthening the duration of each sample so obtained to some constant value T.
These two operations are jointly referred to as ‘sample and hold’. One important reason for intentionally
lengthening the duration of each pulse is to reduce the bandwidth. Figure 9.8(a) shows the functional diagram
of the sample and hold circuit generating flat-top samples and figure (c) shows the flat-top sampled signal xs(t)
. We can see from figure (c) that only the starting edge of the pulse represents instantaneous value of the signal
x(t).
MOSFET
x(t) s(t) c xS(t)
(a)
x(t)
xS(t)
x(4 Ts)
x(3Ts)
x(2 Ts)
x(Ts)
0 t 0 TS 2TS 3TS 4TS 5TS t

T
(b) (c)
Figure 9.8 (a) sample and hold circuit, (b) continuous-time signal, and (c) flat-top sampled signal xs( t )
Mathe matically we can represent flat top sampling as first of all we perform instantaneous sampling

x(t ) (t nTs ) x ( nT s ) (t nT s ) ....(7)

n n
Then we increase thickness of each pulse by convolution of sampled signal with pulse h(t)
xs(t) = x (t) * h(t) ... (8)
Xs( ) = X ( ) H( ) ...(9)
h(t) is the pulse of duration T shown in figure 9.9(a) and |H( )| is shown in figure 9.9(b)
h(t) |H( )|
1 T
0 T t 6 4 2 2 4 6
T T T T T T
Figure 9.9 (a) Rectangular pulse h ( t ) and (b) its magnitude spectrum
1
Xs ( ) = H ( ) X( n s) ...(10)
Ts n
We can see that the multiplication of H( ) with Xs( ) will distort it because magnitude of H( ) changes
with . Generally T is very small thus is figure 9.9(b) the [H(w)] will have large width of main lobe and band
width of X( ) will much less than width of main lobe. But for reconstruction we need a low pass filter with C =
h and shape should be such that it counter effect of multiplication of H( ). Thus the filter should have.
1 e j T /2
Heq( ) = ....(11)
H( ) T sin c( T / 2π)
9.4 Reconstruction of sampled signal

1. Ideal Reconstruction
We have already see that if the signal is ideally sampled then to get back the signal we pass it through an
ideal low pass filter with cut off frequency s /2 and pass band gain Ts(figure 9.10(a))
The impulse response of the system will be inverse fourier transform of H( ) that is
Ts
Since H( ) = Ts rect ...(12)
2π
tπ
Ts sin
Ts
h( t ) = ...(13)
πt
The impulse response is shown in figure 9.10 (c).
Thus when x (t) = x(nTs ) (t nTs )

n
is applied as input to LTI system then output is

y(t) = x (t)*h(t) = x(nTs )h(t nTs )

n
π
sin (t nTs )
Ts
= x(nTs ) ...(14)
π
n
(t nTs )
Ts
Equation (14) is the interpolation formula, which yields value of x(t) between samples as a weighted sum of
all the sample values as shown in figure 9.10(a).
H( ) x(t)
TS
0 TS 2TS 3TS 4TS t

TS TS
h(t)
2Ts Ts Ts 2Ts t
Figure 9.10 (a) Frequency response of an ideal low-pass filter and (b) the reconstructed signal (c)
impulse response of the interpolating (low-pass) filter
2 . Zero order hold interpolation
The zero-order hold can be viewed as a form of interpolation between sample values in which the impulse
response h(t) of the interpolating filter is a rectangular pulse as depicted in figure 9.11(a)
t Ts / 2 1, 0 t Ts
h(t) = u(t) u(t Ts) = rect ...(15)
Ts 0, otherwise
When the sampled signal x (t) is applied to this filter, the output is xr(t). Each sampled in x (t), being an
impulse, generates a rectangular pulse of height equal to the strength
Of the sample, as illustrated in figure 9.11(b). The output of the filter xr(t) is obtained by the convolution of
the input x (t) with the impulse response h(t):
xr(t) = x (t) * h(t) = x(nTs )δ(t nTs )*h(t )

n

xr ( t ) = x(nTs )h(t nTs ) ...(16)

n
Substitution of h(t) from Eq. (15) in Eq. (16) yields
t nTs (Ts / 2)
xr ( t ) = x(nTs )rect ...(17)
n Ts
The filter output is staircase approximation of x(t). Figure 9.11(c) shows the magnitude of the transfer
function of zero-order hold interpolatingfilter superimposed on the desired transfer function of the exact inter-
polating filter.
xr(t)
h(t)
|H( )|
1 Ideal interpolating tilter
Zero-order hold
0 TS t 0 TS 2TS 3TS 4TS 5TS t 0

TS TS TS TS
(a) (b) (c)
Figure 9.11(a) Impulse response of a zero-order hold interpolating filter, (b) output of the interpolating
filter (c) frequency response for the zero-order hold and for the ideal interpolating filter.
3 . First order hold interpolation

The first-order hold can be viewed as a form of interpolation between sample values in which the impulse
response h(t) of the interpolating filter is a triangular pulse, as depicted figure 9.12(c).
t
1 , Ts t 0
Ts
t t
h(t) = r(t + Ts) 2r(t) + r(t Ts) = 1 , 0 t Ts ...(18)
Ts Ts
0, otherwise
When the sampled signal x (t) is applied to this filter, the output is xr(t). Each sample in x (t), being an
impulse, generates a triangular pulse of height equal to the strength of the sample, as illustrated in figure 9.12(d).
The output of the filter xr(t) is obtained by the convolution of the input x (t) with the impuse response h(t):
xr(t) = x (t) * h(t) = x(nTs )δ(t nTs )*h(t )

n
xr ( t ) = x(nTs )h(t nTs ) ...(19)

n

xr(t)
x(t) x (t)
x h(t)
xr(t)
s(t) 0 TS 2TS 3TS 4TS t

(a) (d)
h(t)
xr(t)
|H( )|
1
Ideal interpolation
TS
First-order hold
t -TS TS t
TS TS TS TS
(b) (c) (e)
Figure 9.12(a) System for sampling and interpolation, (b) input signal x ( t ), (c) impuse response of first-
order interpolating filter, (d) output of the interpolating filter and (e) frequency response for the first-order
hold and for ideal interpolating filter
Substitution of h(t) from Eq. (18) in Eq. (19) yields
t nTs
xr ( t ) = x(nTs ) ...(20)
n Ts
The filter output is a linear approximation of x(t). Figure 9.12(e) show the magnitude of the transfer
function of the first-order hold interpolating filter superimposed on the desired transfer function of the exact
interpolating filter.
Example - 9.4
Consider the two continuous-time sinusoidal signals

x 1 ( t ) = cos( 20 t )
x 2 ( t ) = cos(100 t )
Which are sampled at a rate fs = 40 Hz. Find the corresponding discrete-time signals.
Solution 9.4
To convert a continuous-time sinusoidal signal into a discrete-time signal, substitute t = nTs. Therefore,
10 π
x1(t)|t=nTs = x1(nTs) = x1(n) = cos(20 nTs) = cos 2π n cos n
40 2
50 5π
x2(t)|t=nTs = x2(nTs) = x2(n) = cos(100 nTs) = cos 2π n cos n
40 2
5π πn πn
However, x2(n) = cos n = cos 2πn cos x1 (n)
2 2 2

Hence, x2(n) = x1(n). Thus, the sinusoidal signals are identical and, consequently, indistinguishable. If
we are given the sampled values generated by cos( n/2), there is some ambiguity as to whether these
sampled values correspond to x1(t). Since x2(t) yields exactly the same values are x1(t) when the two are
sampled at fs = 40 samples/s, we say that the frequency f2 = 50 Hz is an alias of the frequency f1 = 10 Hz
at the sampling frequency fs = 40 Hz.
Study Note
We can see that in above example when we sample sinusoidal signal below fS we get sampled signal same. That is
we get sampled signal cos(100 t) same as that of cos(20 t) this happenes because fs used was not satisfying nyquist
rate for cos(100 t). When sampling frequency is fs = 1/TS then
sampled 2πf o
A cos(2 f ot ) A cos n A cos(2 f d n)
fs
we can get same result for any other sinusoidal signal A cos(2πf k t ) where fk = fo + kfs, k is an integer
sampled f o +kf s
A cos(2 f k t ) A cos 2π n A cos(2πf d n 2πk ) .
fs
We get same result for sampled Acos(2 fot) and Acos(2 fkt) when fk = (fo + kfs).
Example - 9.5
Consider the continuous-time signal

x ( t ) = cos(100 t )
( a ) Determine the minimum sampling rate required to avoid aliasing.
( b ) Suppose that the signal is sampled at the rate f s = 200 Hz. What is the discrete-time
signal obtained after sampling?
( c ) Suppose that the signal is sampled as the rate f s = 75 Hz. What is the discrete-time
signal obtained after sampling?
( d )What is the frequency 0 < f < f s /2 of a sinusoid that yelds samples identical to those
obtained in part (c)?
Solution 9.5
(a) Here x(t) = cos(100 t) here band width is 50 Hz thus nyquist rate will be 100 samples/sec
(b) Here fs = 200 Hz.
f s 200 Hz π
cos(100 t) cos n
2
(c) Here fs = 75 Hz
4π 2π
cos(100 t)
fs 75Hz
cos n cos n
3 3
2π
Thus aliasing took place here because fs < Nyquist rate. and if we keep fs = 75 Hz then cos n
3

2π
will correspond to cos . f s .t cos(50πt )
3
2π
(d)cos(50 t) with fs = 75 Hz correspond to cos n
3
9.5 Sampling theorem of real valued band pass signal

In previous sections, we have discussed the sampling theorem for low-pass signals. However, when the
signal is a bandpass signal, then a different criterion must be used to sample the signal. The spectrum of
baseband (low pass) signal includes the origin, whereas the spectrum of band pass signals occupies a range of
frequencies between fL and fH, where fL > 0. The quantity B = fH fL is a measure of the band width of the band
pass signal.
The bandpass sampling theorem states that if a real valued bandpass signal x(t) has a spectrum of band-
width B = 2 B= 2 (fH fL) and an upper frequency limit H = 2 fH, then the original signal x(t) can be
recorvered from the sampled signal xs(t) by bandpass filtering if fs = 2fH/k, where k = [fH/B] is the integer part
of fH/B.
Study Note
In real valued signal fourier transform X( ) of signal exist in both side on axis, non zero for fL < |f| < fH.
9.6 Sampling theorem of complex valued band pass signal

A complex bandpass continuous-time signal x(t) has a Fourier transform X(f) that is nonzero over the
frequency range [fL, fH]. The bandpass sampling theorem for complex signals states that if a complex bandpass
signal x(t) has a spectrum of bandpass B = 2 B = 2 (fH fL) and an upper frequency limit H = 2 fH, then the
original signal x(t) can be recovered from the sampled signal xs(t) by bandpass filtering if the sampling fre-
quency.
fs fH fL
Study Note
In complex valued signal there is no need of both sided fourier transform we have only one side transform, that is
non zero for fL < f < fH.
Example - 9.6
Determine the minimum sampling frequency fs for each of the following bandpass signals:
( a ) x ( t ) is real with X ( t ) nonzero only for 9 kHz < | f | < 12 kHz.
( b ) x ( t ) is real with X ( f ) nonzero only for 18 KHz < | f | < 22 kHz.
( c ) x ( t ) is complex with X ( f ) nonzero only for 30 kHz f 35 kHz.
Solution 9.6
(a) x(t) is real with X(f) nonzero only for 9 kHz< |f| < 12 kHz. For this signal , the bandwidth B = fH fL
= 3 kHz and fH = 12 kHz = 4B is an integer multiple of B. Therefore, the minimum sampling fre-
quency is fs = 2B = 6 kHz.
(b) x(t) is real with X(f) nonzero only for 18 kHz< |f| < 22 kHz. For this signal, the bandwidth

B = fH fL = 4 kHz and fH = 22 kHz is not an integer multiple of B. Therefore, the sampling frequency
is
2 fH 2 22k 2 22k 2 22k

fs = = 8.8kHz
[ f H / B] [22 / 4] [5.5] 5
(c) For a complex bandpass signal with a spectrum that is nonzero for fL f fH, the minimum sampling
frequency is fs = (fH fL). Thus, for given signal x(t) the sampling frequency fs = 35 30 = 5 kHz.
Example - 9.7
Given the signal x ( t ) = 10cos(2000 t ) cos(8000 t )

( a ) What is the minimum sampling rate based on the low-pass uniform sampling theorem?
( b ) Repeat (a) based on the bandpass sampling theorem.
Solution 9.7
(a) Consider the given signal
x(t ) = 10cos(2000 t) cos(8000 t)
= 5cos(6000 t) + 5cos(10000 t)
Comparing with x(t ) = A1 cos(2 f1t) + A2 cos(2 f2t)
We have f1 = 3000 = 3kHz and f2 = 5000 = 5 kHz
The maximum frequency present in the signal x(t) is
fmax = f2 = 5000 = 5 kHz
There, the Nyquist rate is given by
fs = 2fmax = 10000 = 10 kHz
(b) For the given signal x(t), we have fL = 3 kHz, fH = 5 kHz, and B = fH fL = 2 kHz. Therefore, the
sampling frequency is
2 fH 2 5k 2 5k 2 5k
fs = [ f / B] 5 kHz
H [5 / 2] = [2.5] 2
9.7 Hilbert Transform

A LTI system with impulse response h(t) = 1/ t act as system to produce hilbert transform of input signal
LTI system
x(t)
h(t) =
x (t )
t
Figure 9.13 Hilbert Transform system
CTFT of h(t) is H( ) = jsgn( ) ...(21)

H( )
|H( )| 2
1
0
2
0
Figure 9.14 (a) Magnitude spectrum and (b) phase response of the HIlbert transformer

Thus the system act as the 90° phase shifter. The properties of Hilbert transform are
1. A signal g(t) and its Hilbert transform g(t ) have the same magnitude spectrum.
Gˆ ( ) = | G ( ) | ...(22)
As corollaries to this property, we may state that
(a) if a signal g(t) is band limited, then its Hilbert transform g(t ) is also band limited.
(b) the signal g(t) and its Hilbert transform g(t ) have the same energy spectral density if g(t) is an energy
signal, and the same power spectral density if it is a power signal.
(c) the signal g(t) and its Hilbert transform g(t ) have the same autocorrelation function.
2. if g(t ) is the Hilbert transform of g(t), then the Hilbert transform of g(t ) is g(t), i.e., if
g(t ) = HT[g(t)]
Then -g (t ) = HT g(t ) ...(23)
3. A signal g(t) and its Hilbert transform g(t ) are orthogonal, i.e., if g(t) is an energy signal,
Then g(t )g(t )dt = 0 ...(24)
4. An energy signal g(t) and its Hilbert transform g (t ) have the same energy:
Eg = | g( t )|2 dt | g(t )|2 dt ...(25)
Example - 9.8
Determine the Hilbert transform of g ( t ) = cos( c t ).

Solution 9.8
Consider the given signal g(t) = cos( ct). The Fourier transform of g(t) is given by
G( ) = [ ( c) + ( + c)]
We know that G ( ) = j sgn( )G( )

= j sgn( ) [ ( c) + ( + c)]
π
G( ) = ( c ) ( c ) sgn( )
j
It is clear that the function
[ ( c) + ( + c)]sgn( ) = [ ( c) ( + c)]. Therefore,
π
G( ) = ( c ) ( c )
j
Taking the inverse fourier transform of the above equation, we obtain
g (t ) = sin( ct )
Therefore, if g(t) = cos( ct),
Then
GATE MASTERS PUBLICATION g (t ) = sin( ct)
Then g (t ) = sin( ct )
Example - 9.9
Determine the Hilbert transform of g ( t ) = sin( c t ).

Solution 9.9
Consider the given signal g(t) = sin( ct). The Fourier transform of g(t) is given by
π
G( ) = ( c ) ( c )
j
π
We know that G( ) = j sgn( )G( ) = j sgn( ) ( c ) ( c )
j
G( ) = [ ( c) ( + c)sgn( )] = [ ( c) ( + c)]
Taking the inverse fourier transform of the above equation, we obtain
G( ) = cos( ct )
9.8 Pre-Envelop of continuous time signal

Let g(t) be a real-valued signal. The pre-envelope of the signal g(t) is a complex-valued function defined by
g+(t) = g (t ) j g (t ) ... (26)
Where g(t ) is the Hilbert transform of the signal g(t). The pre-envelope is useful in handling bandpass
signals and system. One of the important features of the pre-envelope g+(t) is the behaviour of its Fourier
transform. Let G+( ) denote the Fourier transform of g+(t). Then, by taking the Fourier transform of the Eq.
(26), we may write
G+( ) = G( ) +j[ j sgn( )]G( )
= G( ) +sgn( )G( )
G+( ) = G( )[1+sgn( )]
2G ( ), 0
G+( ) = G (0), 0 ... (27)
0, 0
|G( )| |G+( )|
2G(0)
G(0)
B B B
Fi gur e 9 .15 (a ) M agnitu de spe ctr um of low -pa ss sig nal g ( t ) and (b ) m agnitu de spe ctr um of
p r e - e n v e l o p e g + (t)

Where G(0) is the value of G( ) at = 0. This means that the fourier transform of pre-envelope vanishes
for < 0 as illustrated in figure 9.15 for the case of a low-pass signal. For the purpose of illustration in figure
9.15 we have used a low-pass signal with its spectrum limited to the band B B and centred at the origin.
Nevertheless, it should be emphasized that the pre-envelope can be defined for any signal, be it low-pass or
bandpass, so long as it possesses a spectrum. Equation (26) defines the pre-envelope g+(t) for positive fre-
quency; in a similar way, we may difine the pre-envelope for negative frequencies as
g (t) = g(t ) j g(t ) ... (28)
The two pre-envelopes g+(t) and g (t) are simply the complex conjugate of each other, as shown by
*
g (t) = g (t ) ...(29)
The spectrum of the pre-envelope g+(t) is nonzero only for positive frequencies, hence the use of a plus sign
as the subscript. In contrast, the spectrum of the other preenvelope g (t) is nonzero only for negative frequen-
cies, as shown by the fourier transform
0, 0
G( ) = G(0) 0 ...(30)
2G( ), 0
9.9 Complex envelope and band pass signal

We say that a signal g(t) is a bandpass signal if its Fourier transform G( ) is non-negligible only in a band of
frequencies of total extent 2B, say, centred about some frequency ± c. This is illustrated in figure 9.16(a). We
refer to c as a carrier frequency. In the majority of communication signals (modulated signals), we find that the
bandwidth 2B is small compared with c, and so we refer to such a signal as a narrowband signal.
Let the pre-envenlope of a narrowband signal g(t), with Fourier transform G( ) centred about some fre-
quency ± c, be expressed in the form
g+(t ) = g (t )e j ct ...(31)
Or, enqivalently, j ct
g(t ) = g (t )e ...(32)
Where g(t ) is the complex envelope of the narrowband signal g(t). Form Eq.(27), we have
2G ( ), 0
G+( ) = G (0) 0
0, 0
The spectrum G+( ) is limited to the frequency band ( c B) ( c+B), as illustrated in figure 9.16(b).
Taking the fourier trasform of Eq.(32) and applying the frequency-shifting property, we obtain
G( ) = G+( + c) ...(33)
The spectrum G ( ) of the complex envelope g(t ) is limited to the band B B and centred at the origin
as illustrated in figure 9.16(c). The complex envelope g(t ) of a bandpass signal g(t) is a low-pass signal. From
Eqs(26) and (32), we obtain
g ( t ) = Re[g+(t)] ...(34)
j ct
g ( t ) = Re g (t )e ...(35)

|G( )|
|G( )|
C
C B C C B 0 C B C C B
2B
(a)
|G +( )| | G( C)|
| 2G ( C )| |2G ( C )|
C B C C B B B
2B
(b) (c)
Figure 9.16 Magnitude spectrum of (a) bandpass signal g (t) pre-envelope q+(t), and (c) complex envelope g(t )
In general, g(t ) is a complex-valued quantity; thus, we may express it in the fom
g(t ) = gI(t) +jgQ(t) ...(36)
Where gI(t) and gQ(t) are both real-valued low-pass functions. Substituting g(t ) = gI(t) + jgQ(t) into
Eq. (35), we obtain
g(t) = Re g I (t ) jg Q (t ) cos( c t ) j sin( c t )

g ( t ) = gI(t) cos( ct) -j gQ(t) sin( ct) ...(37)
Eqution (36) represents the bandpass signal g(t) in the cononical form. We refer to gI(t) as the in-phase
component of the bandpass signal g(t) and gQ(t) as the quadrature component of the signal. Both gI(t) and gQ(t)
are low-pass signals limited to the band B B. Hence, they may be derived from the bandpass singnal g(t).
Using Eqs. (33) and (36), we obtain
gI ( t ) = Re g(t )
j ct
gI ( t ) = Re g (t ) e
Substituting g+(t) = g(t) jg(t ) into the above equation, we obtain

gI (t ) = Re g (t ) j g (t ) cos( c t ) j sin( c t )
gI (t ) = g (t )cos( ct ) g (t )sin( ct ) ...(38)

Equation (38) represents the in-phase component gI(t) in terms of the bandpass signal g(t) and its HIlbert
transform g(t ) . Similary, we may obtain the quadrature component gQ(t) in terms of the bandpass signal g(t)
and its Hilbert transform g(t ) . Using Eqs (32) and (36), we obtain
gQ (t ) = Im g(t )
j ct
gQ(t) = Im g (t )e
Substituting g+(t) = g(t) j g(t ) into the above equation, we obtain
gQ(t) = Re g (t ) j g (t ) cos( c t ) j sin( c t )
gQ (t ) = g (t )cos( c t ) g (t )sin( c t ) ...(39)

5. It is given that the fourier transform of the pulse

shown in figure. is
sin(πx)
Subjective Practice Problems X(f ) = sinc2( f ) where sinc(x)
πx
1. Determine the conditions on the sampling interval

The acoustic pulse received by the sound re-
Ts so that x(t) is uniquely represented by the discrete- ceiver on a submaarine is of the form
time sequence x(n) = x(nTs) y(t)=30 sinc2(100t)
(a) x(t) = cos( t) +3 sin(2 t) +sin(4 t) (a) Sketch the spectrum of the pulse.
(b) x(t) = cos(2 t) sinc(t) +3 sin(6 t) sin(2t) (b) If thei pulse is to be processed in discrete-
(c) The signal x(t) with FT given in figure time, what is the mean sampling frequency, fs, that
can be used.
X( ) (c) Also sketch the spectrum of the sampled

signal ys(t) if sampling is done at the following
rates:
(i) f ’s = fs, (ii) f ’s = 2fs, and (iii) f ’s = fs/2.

6. Let x(t) be a continuous-time signal whose fourier
0 transform has the property that X( ) = 0 for
| | 200 . A discrete-time signal
2. A signal x(t) = cos(200 t) +2cos(320 t) is ide-
x(n)=x(n 0.5 10 3)
ally sampled at fs = 300 kHz. If the sampled sig-
nal is passed through an ideal low-pass filter with is obtained. For each of the following constraints
a gain 1/fs and cutoff frequency of 250 Hz, what on the fourier transform X(ej ) of x(n), determine
frequency components will appear in the output the corresponding constraints on X( ):
3.Consider the continuous-time signal. (a) X(ej ) is real.
x(t)= 3 cos(2000 t) +5sin(6000 t) (b) The maximum value of X(ejw) over all is 1.
+10cos(12000 t) (c) X(ej ) = 0 for 3 /4 | | .
(a) What is the Nyquist rate for this signal (d) X(ej ) = X(ej( )).
(b) Assume now that we sample this signal us- 7. The complex envelope. of the signal x(t) =
ing a sampling rate fs = 5000 samples/s. What is
the discrete-time signal obtained after sampling? Re A(t )e j ct (t )
equals.
(c) What is the continuous-time signal y(t) we (a) A ( t ) (b)A(t) cos[ (t)]
can reconstruct form the sample
(c) A(t) sin[ (t)] (d) A(t) e f(t)
(fs = 5000 samples/s) if we use ideal interpola-
tion?
4. A signal x(t) = sin( t)/ t is sampled by x(t) =
n
(t n / 2) . Determine and sketch the
sampled signal and its fourier transform.

Chapter
Discrete Fourier
Transform (DFT) and
Fast Fourier Transform 10
10.1 Introduction
We have seen that the frequency analysis of discrete signal is done by DTFT and DTFT is continuous
and periodic with period 2 . Computation and storage of continuous signal is difficult and we convert it into
discrete to store them. The DFT is sampled version of DTFT and FFT is a method to calculate DFT will less
time required
10.2 Design of IIR filters

The DFT of a discrete-time signal x(n) is a finite duration discrete frequency sequence. The DFT
sequence is denoted by X(k). The DFT is obtained by sampling one period of the DTFT X(ej ) of the signal
x(n) at a finite number of frequency points. This sampling is conventionally perfomed at N equally spaced
points in the period 0 2 or at k = 2 k/N ; 0 N 1 . We can say that DFT is used for transforming
discrete-time sequence x(n) of finite length into discrete freqency sequence X(k) of finite length.
The DFT is important for two reasons. First it allows us to determine the frequency content of a
signal, that is to perform spectral analysis. The second application of the DFT is to perform filtering opera-
tion in the frequency domain.
Let x(n) be a discrete-time sequence with fourier transform X (ej ), then the DFT of x(n) denoted by
X(k) is defined as
X(k ) = X e j 2 k/N ; for k = 0,1,2,.., N 1
The DFT of x(n) is a sequency consisting of N samples of X(k). The DFT sequence starts at k = 0,
corresponding to = 0, but does not include k = N corresponding to = 2 (since the sample at = 0 is same
as the sample at = 2 . Generally, the DFT is defnined along with number of samples and is called N-point
DFT. The number N for a finite duration sequence x(n) of length L should be such that N L.
The DTFT is nothing but the z-transform evaluated along the unit circle centred at the origin of the
z-plane. The DFT is nothing but the z-transform evaluated at a finite number of equally spaced points on
the unit circle centred at the origin of the z-plane.
To calculate the DFT of a sequence, it is not necessary to compute its fourier tranform, since the DFT
can be directly computed.
DFT - The N-point of a finite duration sequence x(n) of length L, where N L is defined as

N 1 N 1
j 2 nk / N
DFT {x(n)} = X k x n e x n WNnk ; for k = 0,1,2,..,N 1 ...(1)
n 0 n 0
IDFT - The inverse discrete fourier transform (IDFT) of the sequence X(k) of length N is defined
as
N 1 N 1
1 1
IDFT {X(k)} = X k e j2 nk / N
X k WN nk ; for k = 0,1,2,..,N 1 ...(2)
N n 0 N n 0
Where WN = e j(2 N)
is called the twiddle factor.
The N-point DFT pair x(n) and X(k) is denoted as
DFT
x(n) N
X(k )
Thus N point DFT of signal x[n] means we take N samples of DTFT. Taking N samples of DTFT
(X(e )) of x[n] means we multiply X(ej ) with P(ej ) where P(ej ) is impulse train in frequency domain.
j
2
P(ej ) =
k N
The signal p[n] which has DTFT P(ej ), p[n] will be impulse train with period N, that is
N
p[ n ] = n kN
2 k
x[n]* p[n ] = P(ej ). X(ej )
N
Thus sampling in DTFT leads to convolution in time domain if we remove scaling factor for
2
understanding then in time domain signal becomes
x n * n kN = x n kN x n
k k
Thus we can say that N point DFT of a nonperiodic signal x[n] actually correspond to periodic
signal x n with period N made periodic by repeating x[n] at integer multiple of N.
Since DTFS is used for fourier or frequency analysis of periodic signal suppose a discrete
non periodic signal x[n] is repeated to make periodic signal x n with period then it is DTFS coefficients are
N 1 2
1 j
N .nK
Ck = x n e ...(3)
N n 0
N 1 2
j
N .nK
and x n = Ck e ...(4)
k 0

Discrete Fourier Transform and Fast Fourier Transform 445
Study Note
N 1 2
1 j nk
Equation(3) is equal to Ck = x n e N
because for 0 n N 1, x n x n
N n 0
If we compare equation 1, 2, 3, and 4 then we get relationship between DFT and DTFS
X(k) = N Ck ...(5)
In other words if x[n] is a non periodic signal with length L, we generate a new signal
x n = x n kN
k
And N point DFT of x[n] is N time DTFS fo x n .
Method to find N point DFT of a sequence x n
To find N point DFT of sequence x n of length (L) where L < N. We will simply apply eq(1)
and for values of n for which x[n] is not known is to be taken as zero.
Example 10.1
Find N point DFT of

(a) x [ n ] = [ n ]
(b) x [ n ] = a n , 0 n 10
Solution 10.1
(a) Here
1, n 0
x[n] = [n] =
0, n 0
To find N point DFT
N 1 2 N 1 2
j .nk j .nk
X(k ) = x ne N
ne N
n 0 n 0
X(k ) = 1
(b) Here x[n] = a , 0
n
n 10 for N point DFT, obviously N > 10
N 1 2 10 2
j .nk j .nk
X( k ) = x n e N
ane N
n 0 n 0
11
a
1 2
j k
e N
X( k ) =
a
1 2
j k
e N

Study Note
Since N point DFT of x[n] is N time DTFS coefficients of x n thus DFT is also periodic with period
N. X(k) is periodic with period N.
Example 10.2
(a) Find the 4-point DFT of x ( n ) = {1, 1, 2, 2} directly.

( b ) Find the IDFT of X ( k ) = {4, 2, 0, 4} directly.
Solution 10.2
(a) Given sequence is x(n) = {1, 1, 2, 2}. Here the DFT X(k) to be found is N = 4 point and length of
the sequence L = 4. So no padding of zeros is required.
We know that the DFT {x(n)} is given by

N 1 N 1 3
j 2 / N nk j /2 nk
X(k ) = x n WNnk x n e x n e , k = 0,1,2,3
n 0 n 0 n 0
N 1
X(0) = x n e0 x 0 x 1 x 2 x 3 = 1 1+2 2= 0
n 0
3
j /2 n j /2 j j 3 /2
X(1) = x n e x 0 x1e x 2 e x(3)e
n 0
= 1+( 1)(0 j ) +2( 1 j 0) 2(0 +j )

= 1 j
3
X(2) = x n e j n
x 0 x 1 e j
x 2 e j2
x(3)e j3
n 0
= 1 1( 1 j 0) +2(1 j 0) 2( 1 j 0) = 6
3
j 3 /2 n j 3 /2 j3 j 9 /2
X(3) = x n e x 0 x1 e x 2 e x(3)e
n 0
=1 1( j ) +2( 1 j 0) 2( j 0) = 1+j
X(k) ={0, 1 j, 6, 1 +j }
(b) Given DFT is X(k) = {4, 2, 0, 4}. The IDFT of X(k), i.e. x(n) is given by
N 1 N 1
1 1 j 2 / N nk
x(n) = X k WN nk X k e
N k 0 N k 0
3
1
i.e. x(n) = X k ej / N nk
N k 0

1 3 1
x(0) = X k e0 X 0 X 1 X 2 X 3
4k 0 4
1
= 4 2 0 4 = 2.5
4
1 3 1
x(1) = X k ej /2 k
X 0 X 1 ej /2
X 2 e j
X 3 ej 3 /2
4k 0 4
1
= 4 2 0 j 0 4 0 j = 1 j 0.5
4
1 3 1 j 3 /2
x(2) = X k ej k
X 0 X 1 ej X 2 e j2 X 3 e
4k 0 4
1
= 4 2 1 j0 0 4 1 j0 = 0.5
4
1 3 j 3 /2 k 1 j 3 /2 j 9 /2
x(3) = X k e X 0 X 1e X 2 e j3 X 3 e
4k 0 4
1
= 4 2 0 j 0 4 0 j = 1+ j 0.5
4
x(n) ={2.5, 1 j 0.5, 0.5, 1+j 0.5}
Example 10.3
Compute the DFT of the 3-point sequence x ( n ) = {1, 2, 3}. Using the ssame sequence,
compute the 6-point DFT and compare the two DFTs.
Solution 10.3
The given 3-point sequence is x(n) = {2, 1, 2}, N = 3.
N 1 2
j 2 /3 nk
DFT x(n) = x n WNnk x n e , k = 0, 1, 2
n 0 n 0
j 2 /3 k j 4 /3 k
= x 0 x 1e x 2 e
2 2 4 4
= 2 cos k j sin k 2 cos k j sin k
3 3 3 3
When k = 0, X(k) = X(0) = 2 + 1 + 2 = 5
2 2 4 4
When k = 1, X( k ) = X 1 2 cos j sin 2 cos j sin
3 3 3 3
= 2 + ( 0.5 j0.866) + 2( 0.5 + j0.866)
= 0.5 + j0.866
4 4 8 8
When k = 2, X( k ) = X 2 2 cos j sin 2 cos j sin
3 3 3 3

= 2 + ( 0.5 + j0.866) + 2( 0.5 j0.866)

= 0.5 j0.866
3-point DFT of x(n) = X(k) = {5,0.5 + j0.866,0.5 j0.866}
To compute the 6-point DFT, convert the 3-point sequence x(n) into 6-point sequence by padding
with zeros.
x ( n ) = {2, 1, 2, 0, 0, 0}, N = 6
N 1 5
j 2 / N nk
DFT {x(n)} = X k x n WNnk x n e , k = 0,1,2,3,4,5
n 0 n 0
j 2 /6 k j 4 /6 k j 6 /6 k j 8 /6 j 10 /6 k
= x 0 x 1e x 2 e x 3 e x 4 e x 5 e
j /3 k j 2 /3 k
= 2 e 2e
When k = 0, X(0) = 2 + 1 + 2 = 5
j /3 j 2 /3
When k = 1, X(1) = 2 e 2e
= 2+(0.5 j0.866)+2( 0.5 j0.866) = 1.5 j2.598
j 2 /3 j 4 /3
When k = 2, X(2) = 2 e 2e
= 2+( 0.5 j0.866)+2( 0.5 j0.866) = 0.5 j0.866
j 3 /3 j 6 /3
When k = 3, X(3) = x 0 x1e x 2 e
= 2 cos j sin 2 cos 2 j sin 2

=2 1+2=3
j 4 /3 j 8 /3
When k = 4, X(4) = x 0 x1e x 2 e
4 4 8 8
= 2 cos j sin 2 cos j sin
3 3 3 3
= 2+( 0.5 + j0.866) +2( 0.5 j0.866)
= 0.5 j0.866
j 5 /3 j 10 /3
When k = 5, X(5) = x 0 x1e x 2 e
5 5 10 10
= 2 cos j sin 2 cos j sin
3 3 3 3
= 2 + (0.5 j0.866) + 2( 0.5 + j0.866) = 1.5 j0.866
Tabulating the above 3-point and 6-point DFTs, we have
DFT X(0) X(1) X(2) X(3) X(4) X(5)
3-point 5 j j
6-point 5 j j 0.5 j0.866 1.5 j0.866
Since 6-point = 3 2-point
X(k) of 3-point sequence = X(2k) of 6-point sequence
10.3 Matrix Formulation of the DFT and IDFT

If we let WN = e j (2 /N), the defining relations for the DFT and IDFT may be written as

N 1
X(k ) = x n WNnk , k = 0,1,...., N 1
n 0
N 1
1
x(n) = X k WN nk , n = 0,1,2,.., N 1
N n 0
The first set of N point DFT equations in N unknowns may be expressed in matrix as
X = WN x ...(6)
Here X and x are N 1 matrices, and WN is an N N square matrix called the DFT matrix.
The full matrix form is described by
X 0 WN0 WN0 WN0 WN0 x 0

N 1
X 1 WN0 WN1 WN2 WN x 1
X 2 WN0 WN2 WN4 2 N 1
WN x 2
= ...(7)
X N 1 WN0 WN0 WN2 N 1
WN
N 1 N 1 x N 1
The matrix x may be expressed in terms of the inverse of WN as:
x = WN 1 X ...(8)
The matrix WN 1 X is called the IDFT matrix. We may also obtain x directly from the IDFT relation
in mat r ix for m wher e t he change of index for m n to k and the change in the sign of the exponent in ej(2 /N)nk
lead to the conjugate transpose of WN. We than have
1 T
x = WN* X ...(9)
N no transpose
We can see that

REMEMBER
1
WN 1 = WN*
N
Example 10.4
Find the DFT of the sequence

x ( n ) = { 1, 2, 1, 0}
Solution 10.4
The DFT X(k) of the given sequence x(n) = {1,2,1,0} may be obtained by solving the matrix product
as follows. Here N = 4.

W40 W40 W40 W40 x(0) 1 1 1 1 1 0

W40 W41 W42 W43 x(1) 1 j 1 j 2 j2
X(k ) = W40 W42 W44 W46 x(2) 1 1 1 1 1 0
W40 W43 W46 W49 x(3) 1 j 1 j 0 j2
The result is DFT {x(n)} = X(k) = {4, j2, 0, j2}
10.4 Representation of discrete periodic signal on a circle

Looking at a discrete periodic signal of period N on a circle, that is let x[n] is discrete periodc signal
with period N. Now suppose we want to find value of x[k] then we move k steps on this circle anticlockwise
and get the vlaue. That is to get value of x[1] we move 1 step anticlockwise for x[10] we move 10 stpes.
x[2]
x[4]
x[3]
x[3] x[1]
Start point
x[2] Start point
x[N 2]
x[N 1] x[1]
x[0]
x[0]
Figure 10.1circular representation Figure 10.2 circular representation of
of signal x[n] signal with period 4
For example Let x[n] is periodic with period 4 then to find x[138] we move 138 steps
anticlockwise direction
x[136+2] = x [2]
To right shift a signal x [ n ] periodic with period N by k units.

That is we need x[n-k, mod N] then we move all point on the circle of figure (10.1) anticlockwise
by k steps to get circular representation of x[n-k, mod N]
x[2]
x[1]
x[0] x[N 1]
x[N 1] x[0] x[k 2]

x[3] x[N 3]
Start point Start point
x[1] Start point
x[2] x[k 1] x[N 2]
x[2]
x[N 1]
x[N k 1] x[k] x[1]
x[N k] x[0]
Figure 10.3 Right shift of signal Figure 10.4 Left shift of signal Figure 10.5 Inversion of signal x[n]
by k units by k units

Similarly to give left shift of k units to periodic signal x[n] with period N, that is we need
x[n+k, mod N] then we move all points in figure (10.1) clockwise by k steps to get circular representation
of x[n+k], that is shown in figure 10.4. Now for time reversal, suppose we want circular representation of
x[ n] or x[ n, mod N] then we can get by taking mirror image of figure (10.1) about starting point that is
shown in figure 10.5.
Thus x[ n, mod N] = x[N n], 0 n N 1
10.5 Properties of DFT

In all the properties the assumption is that : x[n] which was non periodic and is made periodic with
period N corresponsing to N point DFT.
10.4.1 Perodicity
If a sequence x[n] is periodic with period N then DFT X(k) is also periodic with same period.
Then is both x[n] and X(k) have N samples in one period.
Proof : By definition of DFT, the (k +N )th coefficient of X(k) is given by
N 1 N 1
j2 n k N / N j 2 nk / N j 2n N / N
X(k +N ) = x n e x n e e
n 0 n 0
But e j2 n = 1 for all n (Here n is an integer)
N 1
j 2 nk / N
X(k +N ) = x n e X k
n 0
10.4.2 Linearity
If x1(n) and x2(n) are two finite duration sequences and if
DFT {x1(n)} = X1(k )
and DFT {x2(n)} = X2(k )
Then for any real valued or complex valued constants a and b,
DFT {ax1(n) + bx2(n)} = aX1(k) + bX2(k)
N 1
j 2 nk / N
Proof : DFT {ax1(n)+bx2(n)} = ax1 n bx2 n e
n 0
N 1 N 1
j 2 nk / N j 2 nk / N
= a x1 n e b x2 n e
n 0 n 0
= aX1(k) + bX2(k)
10.4.2 Time reversal of the sequence

Since time domain signal x[n] is periodic with period N and when then reversal (x[-n]) is dare then
x[n] = x N 1 , x 0 , x 1 , x 2 ....x N 1 , x 0 , x 1 ...
To generate x[ n] we take mirror image of x[n] about n = 0

x[ n] = x 2 ,x 1 ,x 0 ,x N 1 ,x N 2 ,x N 3 ...x 1 , x 0 ....
The time reversal of an N point sequence x(n) is obtained by wrapping the sequence x(n) over the
circle in a clockwise direction. It is denoted by x[( n),mod N].
If DFT{x(n)} = X(k) then
DFT{x[( n),mod N]} = DFT{x(N n)}
= X[( k),mod N] = X(N k)
DFT
The property to learn is x[ n] X( k)
REMEMBER
Thus for even signal x[n] then X(k) = x( k) that is DFT will also be even similarly for odd
signal x[n] we have X(k) = X( k), that is DFT is also odd
10.5.4 Circular frequency shift

If DFT {x(n)} = X(k)
Then, DFT {x(n)ej2 mn/N} = X{(k m)}, (mod N )
N 1
Proof : DFT {x(n)e j2 mn/N
} = x n e j2 mn / N
e j 2 kn / N
n 0
N 1 N 1
j2 n k m / N j2 n N k m / N
= x n e x n e
n 0 n 0
= X(N +k m) = X[(k m), (mod N)]
The basic property to learn is

REMEMBER
DFT
If x[n] X(k)
DFT
Then e j 2 mn / N
x(n) X(k m)
10.5.5 Complex Conjugation Property

Then DFT {x*(n)}= X *(N k) = X *[( k), mod N]
N 1
Proof DFT {x*(n)}= x* n e j 2 kn / N
n 0
N 1 *
j 2 kn / N
= x n e
n 0
N 1 *
j2 n N k / N
= x n e X* N k
n 0
DFT {x*(N n)} =X *(k)

N 1
1
Proof IDFT {X *(k)} = X * k e j2 kn / N
N k 0
N 1 * N 1 *
1 j 2 kn / N 1 j 2 kn / N n / N
= X k e X k e
N k 0 N k 0
= x*(N n)

REMEMBER
DFT
If x[n] X(k)
Then x*[n] DFT X *( k)
Considering few cases :-

Case : 1. Signal x[n] is real
That is x [ n ] = x* [ n ]
then X ( k ) = X *[ k ]
Thus X(k) that is DFT will be conjugate symmetric
Case : 2. Signal x[n] is real and even

That is x [ n ] = x*[n] and x[n] = x[ n]
Then X(k) = X *(k) and X(k) = X( k)
X(k) = X( k) = X*( k)
Thus X(k) that is DFT will be real and even
Case : 3. Signal x[n] is real and odd
That is x [ n ] = x*[n] and x[n] = x[ n]
X(k) = X *( k) and X(k) = X( k)
X(k) = X *( k) = X( k)
Thus X(k) that is DFT is imaginary and odd
Case : 4. If x[n] is real and it’s DFT is X(k) [X(k) = Re(X(k) + j Img (X(k)))] then DFT of even and
odd part of x[n] wil be
Since x [ n ] = xe[n]+xo[n]
Here x[n] is real
X(k) = X *( k) or X(k)* = X( k)
x n x n
Since xe [ n ] =
2
X k X k
Then DFT of xe[n] will be
2
X k X* k
xe [ n ] Re X k
2
x n x n X k X k X k X* k
and xo[n] =
2 2 2

j Img {X(k)}
Study Note
If we recollect all then these properties are exactly similar to that of DTFS.
Time domain signal DFT of the signal
real conjugate symmetric
imaginary conjugate anti symmetric
even even
odd odd
real and odd real and odd
imaginary and even imaginary and even
imaginary and odd imaginary and odd
imaginary and odd real and odd
Here all the properties have both sided arrows thus for example if signal is real then DFT is conjugate
symmmetric and if time domain signal is conjugate symmetric then DFT will be real
10.5.6 DFT of Delayed Sequence -Time shifting property

Let x(n) be a discrete sequence, and x(n no) be a delayed or shifted sequence of x(n) by no units of
time.
j 2 no k / N
Then DFT {x[(n no)]} = X(k) e
Proof By the definition of IDFT
N 1 2
1 j nk
IDFT{X(k)} = x n X k e N
N k 0
Replacing n by n no, we have

N 1 2 N 1 2 2
1 j n n0 k 1 j no k j nk
x(n no) = X k e N
= X k e N e N
N k 0 N k 0
2
j no k
= IDFT X k e N
On taking DFT of both sides, we get

2
j kno
DFT [x(n no)] = X k e N

REMEMBER
DFT
If x[n] X(k )
go with some sign 2

Then j , no k
x n no DFT
X k e N

10.5.7 Multiplication of two sequence

If DFT [ x1(n)] = X1(k)
and DFT [x2(n)] = X2(k)
1
Then DFT [x1(n)]x2(n) = X1 k X2 k
N
Here X1 k X 2 k is circular convolution of two periodic signals.
10.5.8 Circular convolution of two sequence

The convolution property of DFT says that, the multiplication of DFTs of two sequences if eqivalent
to the DFT of the circular convolution of the two sequences.
Let DFT [x1(n)] = X1(k) and DFT [x2(n)] = X2(k), then by the convolution property
X1(k)X2(k) = DFT {x1(n) x2(n)}.
10.5.9 Parseval’s theorem

Parseval’s theorem says that the DFT is an energy-conserving transformation and allows us to
find the signal energy either from the signal or its spectrum. This implies that the sum of squares of the
signal is related to the sum of squares of the magnitude of the DFT samples.
If DFT {x1(n)} = X1(k)
and DFT {x2(n)} = X2(k)
N 1 N 1
x1 n x2* n 1
Then = X1 k X 2* k
n 0 N k 0
If we take same signal then
N 1 N 1
x1 n x1* n 1
= X1 k X1* k
n 0 N k 0
N 1 N 1
x1 n
2 1 2
= X1 k
n 0 N k 0
We can see that LHS of his equation is energy of signal x(n).
10.5.10 Circular Correlation

For complex valued sequences x(n) and y(n),
and DFT {y(n)} = Y(k)
Then
N 1
DFT {rxy(l)} = DFT x n y* n l , mod N X k Y* k
n 0
here, rxy(l) is the circular cross correlation sequence.

10.5.11 Duality
In DFT the time domain signal x[n] is also discrete and periodic and X(k) is also discrete and periodic
thus periodic and can be easily seen by observing these equations
N 1 2 N 1 2
j nk 1 j nk
X(k ) = x n e N
and x[n] = X n e N
n 0 N k 0
Manipulating the first equation we get
N 1 2
j nk
X(n) = x k e N
k 0
N 1 2
X n 1 j nk
= x k e N
N N k 0
X n
The DFT of is x(k)
N
X n DFT
x (k )
N
DFT
Or X(n) Nx( k)
DFT
If x[n] X( k )
DFT
Then X[n] Nx(-k)
TABLE (10.1) Properties of the DFT
Property Time domain Frequency domain

Periodicity x(n) = x(n + N) X(k) = X(k + N )
Linearity ax1(n) +bx2( n) aX1(k) + bX 2(k)
Time reversal x(( n), mod N ) = x(N + n) X(N k)
Circular time shift (delayed sequence) x(( n l ), mod N ) X(k)e j2 kl/N
Circular frequency shift x(n)e

j2 ln/N
X((k l), mod N )
Circular convolution x1(n) x2(n) X1(k)X2(k)
1
Multiplication of two sequences x1(n)x2(n) X1 k X2 k
N
Complex conjugate *
x (n)
*
X (N k)
*
Circular correlation x1(n) y ( n) X(k)Y*(k)
N 1 N 1
1
Parseval’s theorem x n y* n X k Y* k
n 0 N k 0
10.6 Circular Convolution

Suppose x1[n] and x2[n] are two non periodic sequence, their N point DFT is X1(k) and X2(k). Since N
point DFT X1(k) will correspond to x1 n and x1 n where x1 n and x1 n are two periodic signals made
periodic with period N by repeating x1[n] and x2[n]. If Y(k) = X1(k) X2(k) then the sequence that will Correspond
to Y(k) will be y n where y n is made periodic with period N by repeating y[n] where y[n] is circular
convolution of x1[n] and x2[n] that is
N 1
y [ n ] = x1 n x2 n x1 i x2 n i mod N
i 0
The result of circular convolution of two sequence x1[n] and x2[n] of length N will be of length N
only.
Method To perform circular convolution
Suppose x1[n] and x2[n] have N = 4.
3
Then y[n] = x1 i x2 n i
i 0
y [0] = x1[0]x2[0] + x1[1]x2[ 1] +x1[2]x2[ 2] +x1[3]x2[ 3]

Applying mod 4, y [0] = x1[0]x2[0] + x1[1]x2[ ] +x1[2]x2[2] +x1[3]x2[1]
Similarly we can write for y[1], y[2], y[3] but this is a lengthy method, this can be simplified as
First of all we write the terms of x1[n] and multiply then as we do in simple method.
x2 3 x2 2 x2 1 x2 0
x1 3 x1 2 x1 1 x1 0
x1[0]x2[3] x1[0]x2[2] x1[0]x2[1] x1[0]x2[0]
x1[1]x2[3] x1[1]x2[2] x1[1]x2[1] x1[1]x2[0]
x1[2]x2[3] x1[2]x2[2] x1[2]x2[1] x1[2]x2[0]
x1[3]x2[3] x1[3]x2[2] x1[3]x2[1] x1[3]x2[0]
But result will not be calculated by adding directly we first of all fill all X spaces as shown and then
we add. Thus we get
x1[0]x2[3] x1[0]x2[2] x1[0]x2[1] x1[0]x2[0]
x1[1]x2[2] x1[1]x2[1] x1[1]x2[0] x1[1]x2[3]
x1[2]x2[1] x1[2]x2[0] x1[2]x2[3] x1[2]x2[2]
x1[3]x2[0] x1[3]x2[3] x1[3]x2[2] x1[3]x2[1]
y[3] y[2] y[1] y[0]
Now perform addition and get result.

For example : To perform circular convolution for [1,2,1,0] and [3,2,5,1] with N = 4
Thus
0 1 2 1
1 5 2 3
0 3 6 3
0 2 4 2
0 5 10 5
01 2 1
0 3 6 3
2 4 2 0
10 5 0 5
1 0 1 2
13 12 9 10
y [ n ] = [10,9,12,13]
Example 10.5
Find the circular convolution of these two signals [1,2,1] and [1,0,1,0,1] with N=5.
Solution 10.5
For circular convolution we need both the sequence of same length
x1[n ] = [1,2,1,0,0] and x2[n] = [1,0,1,0,1]
0 0 1 2 1
1 0 1 0 1
0 0 1 2 1
0 0 0 0 0
00 1 2 1
000 0 0
0 011 1
0 0 1 2 1
0 0 0 0 0
1 2 1 0 0
0 0 0 0 0
1 0 0 1 2
2 1 2 3 3
y [ n ] = [3,3,2,1,2]
Example 10.6
Let X ( k ) be a 12-point DFT of length 12 real sequence x ( n ). The first 7 samples of X ( k )

are given by X (0) = 8, X (1) = 1+ j 2, X (2) = 2 + j 3, X (3) = 1 j 4, X (4) = 2+ j 2, X (5) =
3+ j 1, X (6) = 1 j 3. Determine the remaining samples of X ( k ).
Solution 10.6

Given N = 12, we have X(k) = X *(N k) = X *(12 k). The first 7 samples of X(k) are given, the
remaining samples are:
X(7) = X *(12 7) = X *(5) = 3 j1
X(8) = X *(12 8) = X *(4) = 2 j2
X(9) = X *(12 9) = X *(3) = 1 j4
X(10) = X *(12 10) = X *(2) = 2 j3
X(11) = X *(12 11) = X *(1) = 1 j2
Example 10.7
The DFT of a real signal is {1, A , 1, B , 0, j 2, C , 1+ j }. Find A , B and C .
Solution 10.7
Given X(k) = {1, A, 1, B, 0, j2, C, 1, j }, using conjugate symmetry, we have
X(k) = {X(0),X(1),X(2),X(3),X(4),X(5),X(6),X(7)}
= {X(0),X*(8 1),X(2),X*(8 3),X(4),X(5),X*(8 6),X(7)}
= {1, 1 j, 1, j2, 0, j2, 1, 1+j }
A = 1 j, B = j2 and C = 1
Example 10.8
Let x ( n ) = { A ,2,3,4,5,6,7, B }. If X (0) = 20 and X (4) = 0, find A and B .
Solution 10.8
Using central ordinates, we have
7
X(0) = x n A 2 3 4 5 6 7 B A B 27 20
n 0
7
N n
And X(4) = X 1 x n A 2 3 4 5 6 7 B 0
2 n 0
i.e A B+3 = 0
Therefore, A + B + 27 = 20 and A B + 3 =0
Solving the above two equations, we have A = 5 and B = 2
Example 10.9
The DFT of a real signal is X ( k ) = {1, A , 1, B , 7, j 2, C , 1+ j }. What is its signal

ener gy?
Solution 10.9
Using the conjugate symmetry property, we have
X(k) = {1, A, 1, B, 7, j2, C, 1+ j }
= {1,X*(8 1), 1,X*(8 3), 7, j2, X*(8 6), 1+j }
= {1, 1 j, 1, j2, 7, j2, 1, 1+j }

Using Parseval’s theorem, we have signal energy
N 1
2 1N 1
2
= x n X k
n 0 Nn 0
1 2 2 2 2 2 2 2 2
= 1 1 j 1 j2 7 j2 1 1 j
8
1
= 1 2 1 4 49 4 1 4 8.25
8
Example 10.10
Consider the length-6 sequence defined for 0 n < 6

x(n) = {1, 2, 3, 0, 1, 1}
with a 8-point DFT X ( k ). Evaluate the following functions of X ( k ) without computing
DFT:
(a) X (0), ( b )X(3),
5 5
2
(c) X k , (d) X k
k=0 k=0
Solution 10.10
N 1 2
j nk
(a) We know that X(k ) = x ne N , k = 0, 1,..., N 1
n 0
5 5
X(0) = x n e0 x n 1 2 3 0 1 1 2
n 0 n 0
5 2 5
j 3n n
(b) X(3) = x ne 6 x n 1
n 0 n 0
= x(0)(1)+x(1)( 1)+x(2)(1)+x(3)( 1)+x(1)(1)+x(5)( 1)

= (1)(1) 2( 1)+3(1)+(0)( 1) 1(1)+1( 1)
= 4
N 1 2
1 j nk
(c) We have x(n) = X k e N , n = 0,1,..., N 1
N k 0
N 1
X k = Nx(0) = 6(1) = 6
k 0
(d) From Parseval’s theorem

5 5
2 2
X k N x n = 6 1
2
2
2
3
2
0
2
1
2
1
2
k 0 n 0
= 6(1+4+9+0+1+1) = 96

Example 10.11
If the DFT { x ( n )} = X ( k ) = {4, - j 2, 0, j 2}, using properties of DFT, find

(a) DFT of x ( n - 2) ( b ) DFT of x (- n )
(c) DFT of x * ( n ) ( d ) DFT of x 2 ( n )
(e) DFT of x ( n ) x( n ) ( f ) Signal energy
Solution 10.11
(a) Using the time shift property of DFT, we have for N = 4
2
j 2k
DFT {x(n 2)} = e 4 X k e j k
X k
= {X(0)e0,X(1)e j ,X(2)e j2
,X(3)e j3
}
= {4(1), j2( 1), 0(1), j2( 1)} = {4, j2, 0, j2}
(b) Using the flipping (time reversal) property of DFT, we have
DFT {x( n)} = X( k) = X*(k) = {4, j2, 0, j2}* = {4, j2, 0, j2}
(c) Using the conjugate property of DFT, we have
DFT {x*(n)} = X*( k) = {4, j2, 0, 2j }*= {4, j2, 0, j2}
Since DFT {x*(n)} = DFT {x(n)}, we can say that x(n) is real valued.
(d) Using the property of convolution of product of two signals, we have
1 1
DFT {x(n)x(n)} = X k X k 4, j 2,0, j 2 4, j 2, 0, j2
N 4
= {6, j4, 0, 4}
(e) Using the circular convolution property of DFT, we have
DFT {x(n) x(n)} = [X(k)X(k)] = {4, j2, 0, j2} {4, j2, 0, j2}
= {16, 4, 0, 4}
(f) Using Parseval’s theorem, we have
1 2 1 2
Signal energy = X k 4, j 2, 0, j 2
4 4
1
= 16 4 0 4 6
4
Example 10.12
If IDFT { X ( k )} = x ( n ) = {1,2,1,0}, using properties of DFT, find

(a) IDFT { X ( k 1)} ( b ) IDFT { X ( k ) X ( k )}
(c) IDFT { X ( k ) X ( k )} (d) Signal energy
Solution 10.12
Given IDFT {X(k)} = x(n) = {1,2,1,0}

(a) Using modulation property, we have

n
j
IDFT {X(k 1)} = x(n)ej2 n/4
= x n e 2
j 3
j
= x 0 e0 , x 1 e 2 , x 2 e j , x 3 e 2
= 1(1), 2( j ), 1( 1), 0( j ) = {1, j2, 1, 0}

(b) Using periodic convolution property, we have
IDFT{X(k) X(k)} = Nx2(n) = 4{1,2,1,0}2 = {4,16,4,0}
(c) Using the convolution in time domain property, we have
IDFT {X(k)X(k)} = {1,2,1,0} {1,2,1,0} = {2,4,6,4}
N 1
2 2 2 2 2
(d) Signal energy x n = x 0 x 1 x 2 x 3
n 0
=(1)2+(2)2+(1)2+(0)2 = 6
10.7 Insight into formulae of DFT

Formulae are
N 1 2
j kn
X(k ) = x n e N
n 0
N 1 2
1 j kn
And x[n] = X k e N
N k 0
1 . Average value of periodic signal X(k) is
N 1
1
x [0] = X k
N k 0
2. If N even then
N 1 N 1
1 N 1
n
x = X k ej k
1 X k
2 N k 0 N k 0
3. The sun of all values of sequence x[n]

N 1
j k
Since X(0) = x n e
n 0
Similary if N is even
N 1 N 1
n
X N
j n
Then = x n e 1 x n
2 n 0 n 0
4 . Using DFT to find IDFT

1 *
IDFT{X(k)} = DFT X * k
N
N 1 2
j .mk
Let Y(m) = DFT{X (k)} = * X* k e N
k 0
2
1 * 1N1 j .mk
Then DFT X * k = X k eN
N Nk 0
If we compare this with IDFT equation we get that this will be equal to x[m], x[m] = x[n] only
there is change of varialbe
5 . Another method to solve for circular convolutions
Let y [ n ] = x 1[ n ] x 2[ n ]
And N = 4 then
y[n] = x1 (0) x1[1] x1[2] x1[3]

6 . If x[n] is a discrete time signal and x1(k), x2(k), x3(k) are N point, 2N point 3N point DFT of x[n]
N 1 2
j . n. k
Then X1(k) = x n e N
n 0
2N 1 2
j .n.k
X2(k) = x n e 2N
n 0
3N 1 2
j nk
And X3(k) = x n e 3N
n 0
The signal x[n] will be zero for n N because then only we are able to find N point DFT of x[n].
from above equation we can see that
X1(k) = X2(2k) = X3(3k)
Similar we get in example 10.3
10.8 Linear Convolution Using The DFT

The DFT support only circular convolution . When two numbers of N-point sequence are circularly
convolved, it produce another N-point sequence. For circular convolution, one of the sequence should be
periodically exteneded. Also the resultant sequence is periodic with period N. The linear convolution of two
sequence of length N1 and N2 produces output sequence of length N1 + N2 1. To perform linear convolution
using DFT, both the sequence should be converted to N1 + N2 1 sequences by padding with zeros. Then
take N1 + N2 1 point DFT of both the sequences and determine the product of their DFTs. The resultant
sequence is given by the IDFT of the product of DFTs. [Actually the response is resultant sequence is given
by the circular convolution to the N1 + N2 1 sequences].
Let x(n) be an N1-point sequence and h(n) be an N2-point sequence. The linear convolution of
x(n) and h(n) produces a sequence h(n) of length N1 + N2 1. So pad x(n) with N2 1 zeros and h(n) with N1 1

zeros and make both of them of length N1 + N2 1. Let X(k) be an N1 + N2 1 point DFT of x(n), and H(k) be
an N1 + N2 1 point DFT of h(n). Now, the sequence y(n) is gives the inverse DFT of the product X(k) H(k).
y ( n ) = IDFT{X(k)H(k)}
10.9 Convolution of Long Sequence

The response of an LTI system for any arbitrary inputis given by linear convolution of the input and
the impulse response of the system. While implementing linear convolution in FIR filters, the input signal
sequence x(n) is much longer than the impulse response h(n) of a DSP system. If one of the sequence is very
much longer than the other, then it is very difficults to compute linear convolution using DFT for the
following reasons:
1. The entire sequence must be available before convolution can be carried out. This makes long
delay in getting in the output.
2. Large amouts of memory is required and computation of DFT becomes cumbersome. To overcome
these problems, we go to sectioned convolutions. In this technique, the longer sequence is sectioned (or
splitted) into the size of smaller sequence. If required, the longer sequence may be padded with zeros. Then
the linear convolution of each section (block) of longer sequence and the smaller sequence is performed. The
output sequence obtained from the convolutions of all the sections are combined to get the overall ouput
sequence. There are two of sectioned convolutions. They are overlap-add method and overlap-save method.
Example 10.13
Two finite-length sequences, x 1( n ) and x 2( n ), that are zero outside the interval [0, 99]
are circularly convolved to form a new sequence y ( n ),
y ( n ) = x 1( n ) x 2( n )
Where N = 100. If x 1( n ) is nonzero only for 10 n 39. determine the values of n for
which y ( n ) is guranteed to be equal the linear convolution of x 1 ( n ) and x 2( n ).
Solution 10.13
Because
99
y(n) = x2 k x1 n k 100
k 0
the values of n for which y(n) is equal to the linear convolution of x1(n) are those values of n in
the interval [0,99] for which the circular shift x1((n k))100 is equal to the linear shift x1(n k). With x1(n)
nonzero only over the interval [10,39] we see that x1((n k)) for n in the interval [39,99]. Therefore, the
circular convolution and the linear convolution are equal for 39 n 99.
Example 10.14
We would like to linearly convolve a 3000-point sequence with a linear shift-invariant

filter whose unit sample response is 60 point long. To utilize the computational efficiency of the
fast fourier transform algorithm, the filter is to implemented using 128-point discrete fourier
tranforms and inverse discrete fourier tranforms. If the overlap-add method is used, how many
DFTs are needed to complete the filtering operation?

Solution 10.14
With overlap-add, x(n) is partitioned into nonoverlapping sequence lf length M. If h(n) is of length
L, x1(n) * h(n) is of length L + M 1. Therefore, we must use a DFT of length N L + M 1. Here, we have
set N = 128, and h(n) is of length L = 60. Therefore, x(n) must be partitioned into sequence of length
M = N L 1= 69
Because x(n) is 3000 point long, we will have 44 sequences (with the last sequence containing only
33 nonzero vlaues). Thus, to perfrom convolution we need:
1. One DFT to compute H(k)
2. 44 DFTs for X1(k)
3. 44 inverse DFTs for Y(k) = H(k)Xi(k)
For a total of 45 DFTs and 44 inverse DFTs.
Example 10.15
Find the 10-point inverse DFT of
3 k=0
X(k) =
1 1 k 9
Solution 10.15
To find the inverse DFT, note that X(k) may be expressed as follows:
X(k) = 1+2 (k) 0 n 9
Written in this way, the inverse DFT may be easily determined. Specifically, note that the inverse
DFT of a constant is a unit sample:
DFT
x 1( n ) = n X1 k 1
Similarly DFT of constant is a unit sample

DFT
x 2( n ) = 1 X2 k N (k)
Therefore, it follows that
1
x(n) = n
5
Example 10.16
Consider the finite-length sequence

x(n) = (n) + (n 5)
( a ) Find the 10-point discrete fourier transfrom of x ( n ).
( b ) Find the sequence that has a discrete fourier transform
2π
j2k
Y(k) = e 10 X k

where X ( k ) is the 10-point DFT of x ( n ).

( c ) Find the 10-point sequence y ( n ) that has a discrete fourier transform
Y(k ) = X(k)W(k)
where X ( k ) is the 10-point DFT of x ( n ), and W ( k ) is the 10-point DFT of the
sequence
1 0 n 6
w(n) =
0 otherwise
Solution 10.16
(a) The DFT of x(n) is easily seen to be
2
j 5k k
5k
X(k) = 1 2WN 1 2e 10 1 2 1
(b) Multiplying X(k) by a complex exponential of the from WNnk0 corresponds to a circular shift of
x(n) by n0. In this case, because n0 = 2, x(n) is circularly shifted to the left by 2, and we have
y ( n ) = x((n+2))10= 2 (n 3)+ (n 8)
(c) Multiplying X(k) by W(k) corresponds to the circular convolution of x(n) with w(n). The 10-point
circular convolution is
y ( n ) = [3, 3, 1, 1, 1, 3, 3, 2, 2, 2,]
Example 10.17
Consider the sequence

x ( n ) = 4 ( n )+ 3 ( n 1) + n ( n 3)
Let X ( k ) be the six-point DFT of x( n ).
( a ) Find the finite-length sequence y ( n ) that has a six point DFT
Y ( k ) = W64k X k
( b ) Find the finite-length sequence w ( n ) that has six point DFT that is equal to the
real part of X( k )
W ( k ) = Re{ X ( k ) }
Solution 10.17
(a) The sequence y(n) is formed by multiplying the DFT of x(n) by the complex exponential W64 k .
Because this corresponds to a circular shift of x(n) by 4.
y ( n ) = x((n 4))6
It follows that
y( n ) = 4 (n 4)+3 (n 5)+2 (n)+ (n 1)
(b) The real part f X(k) is

1
Re{X(k)} = X k X* k
2
To find the inverse DFT of Re{X(k)}, we need to evaluate the inverse DFT of X *(k). Because
N 1 * N 1
X*(k) = x n WNnk x* n WN nk
n 0 n 0
N 1 N 1
N n k
= x* n WN x* N n N
WNnk
n 0 n 0
X*(k) is the DFT of x*((N n))N. Therefore, the inverse DFT of Re {X(k)}is
1
w(n) = x n x* N n N
2
With N = 6, this becomes
3 3
w(n) = 4, ,1,1,1,
2 2
10.10 Fast Fourier Transform (FFT)

The FFT is a method to compule DFT of any signal x[n] will less number of complex multiplication
and complex addition. Since N point DFT is
N 1 2 2
j nk j
X( k ) = x n e N
, if WN e N
n 0
N 1K
X(k) = x 0 WN0 x 1 WNK x 2 WN2 K x N 1 WN
Thus to calculate one terms of DFT (N 1) complex addition and N complex multiplication are needed.
For N terms we need N2 complex multiplication and N(N 1) complex additions. Here we are taking
multiplication with 1 and -1 also as complex. The FFT is a method to compute DFT with less number of
computation. FFT uses the concept of divide and rule. There are two FFT algorithms : Decimation in time
and Decimation in frequency.
The properties of factor WN are
N
k K
Symmetry property W 2 = WN
N
k N
Periodic property W N = WNK
FFT exploits the above properties of WN and divide the N point DFT into smaller size DFTs
.
10.11 Decimation in Time (DIT) Radix -2 FFT
In decimation in time (DIT) algorithm, the time domain sequence x(n) is decimated and smaller

point DFTs are computed and they are combined to get result of N-point DFT.
In general, we can say that, in DIT algorithm the N-point DFT can be realized from two numbers of
N/2-point DFTs, the N/2-point can be realized form two numbers of N/4-point DFTs, and so on.
In DIT radix-2 FFT, the N-point time domain sequence is decimated into 2-point Sequences and the
2-point DFT for each decimated sequence is computed. From the results of 2-point DFTs, the 4-point DFTs,
from the results of 4-point DFTs, the 8-point DFTs and so on are computed until we get N-point DFT.
For performing radix-2 FFT, the value of r shold be such that, N =2m. Here, the decimation can be
performed m times, where m = log2N. In direct computation of N-point DFT, the total number of complex
additions are N(N 1) and the total number of complex multiplications are N2. In radix-2 FFT, the total
number of complex additions are reduced to N log2N and the total number of complex multiplitcations are
reduced to (N/2) log2N.
Let x(n) be a sequence of length N = 2 , and suppose that x(n) is split (decimated) into two
subsequences, each of length N/2. The first sequence, g(n), is formed form the even-index terms,
N
g ( n ) = x(2n) = n = 0,1,.... 1
2
and the second, h(n), is formed the odd-index terms,
N
h ( n ) = x(2n+1) = n = 0,1,.... 1
2
In terms of these sequences, the N-point DFT of x(n) is
N 1
X(k ) = x n WNnk x n WNnk x n WNnk
n 0 n even n odd
N N
1 1
2 2
X(k) = x 2n WNk 2 n x 2n 1 WN2 n 1k
n 0 n 0
Since WNk 2 n = WNkn and WN2 n 1k

WNkWNnk
2 2
N N
1 1
2 2
X( k ) = g n WNnk WNk h n WNnk
n 0 2 n 0 2
N N
1 1
2 N 2 N
We can see that g n WNnk is point DFT of g(n) and h n WNnk is point DFT of h(n).
n 0 2 2 n 0 2 2
Thus X( k ) = G k WNk H k
N
Thus N point DFT can be written as sum of two point DFT wih a factor of WNk . G(k) and H(k)
2

N k N
WNk
2, and WN
2
will be periodic wih period =
N
Thus X(k) = G (k ) +WNk H ( k ) for 0 k 1
2
N N N
And X(k ) = G k WNk H k for k N 1
2 2 2
Let N = 8 then FFT after decimation will be
G(0)
x(0) X(0)
x(2) G(1) W80
4-point X(1)
DFT G(2) W81
x(4) X(2)
G(3) W82
x(6) X(3)
W83
x(1) X(4)
H(0) W84
x(3) 4-point X(5)
DFT
H(1) W85
x(5) X(6)
H(2) W86
x(7) X(7)
H(3) W87
Figure 10.6: An eight-point decimation-in-time FFT algorithm after the first decimation
Now next stage of decimation we decimate to find G(k) and H(k). As before this lead to
G (k ) = g n WNnk g n WNnk
n even 2 n odd 2
N N
1 1
4 4
2n 1 k
= g 2n WN2 nk g 2n 1 WN
n 0 2 n 0 2
N N
1 1
4 4
G(k ) = g 2n WNnk WNk g 2n 1 WNnk
n 0 4 2 n 0 4
N
Let f1[n] = g(2n) for n = 0 to 1
4
N
And f2[n] = g(2n+1) for n = 0 1
4

N 1 N 1
4
nk N 4
N
Then g 2n W N is point DFT of f1[n] and g 2n 1 WNnk is point DFT of f2[n]
n 0 4 4 n 0 4 4
G(k) = F1 k WNk F2 k
2
F1(k) and F2(k) are periodic with period N 4
k N
and WN 4
= WNk ,
2 2
N
Thus, G(k) = F1 k WNk F2 k for 0 k 1
2 4
N N N N
and G(k) = F1 k WNk F2 k for k 1
4 2 4 4 2
N
Similarly we can decompose H(k) into two point DFT F3(k) and F4(k), taking N = 8 as in previous
4
case we get the figure as
F1(0)
x(0) G(0)
N/4-point
x(4) DFT
F1(1) W40
G(1)
F2(0) W41
x(2) G(2)
N/4-point F2(1) W42
x(6) DFT G(3)
F3(0) W43
x(1) H(0)
N/4-point
DFT
F3(1) W40
x(5) H(1)
F4(0) W41
x(3) H(2)
N/4-point
DFT
F4(1) W42
x(7) H(3)
W43
F1(0)
x(0) G(0)
N/4-point
F1(1)
x(4) DFT G(1)
0
F2(0) W4
x(2) G(2)
N/4-point F2(1) W41
x(6) DFT G(3)
F3(0)
x(1) H(0)
N/4-point
F3(1)
x(5) DFT H(1)
F4(0) W40
x(3) H(2)
N/4-point
F4(1) W41
x(7) DFT H(3)
Figure 10.7: Flow graph for second stage FFT algorithm for N = 8.
The above decimation is done till we are left with 2 point DFT. Two point DFT for example for

N= 8 the 3rd stage of FFT will be to calculate F1(k), F2(k), F3(k) and F4(k).
Here for N = 8 f1(k) = 2 point DFT of points x(0) and x(4) that is g(2n), Let f1(n) = g(2n) and N = 8
1
F1(k) = f1 n W2nk f1 0 W20 f 1 W2k
n 0
F1(0) = f1 0 f1 1 WN0
and F1(1) = f1 0 f1 1 W21 f1 0 f1 1

Thus 3rd stage of FFT will be
x(0) F1(0)
x(4) W20
F1(1)
x(2) F2(0)
W20
x(6) F2(1)
x(1) F3(0)
W20
x(5) F3(1)
x(3) F4(0)
W20
x(7) F4(1)
Figure 10.8: Third stage of FFT algorithm for N = 8.
The 8 point DFT using radix -2 FFT

The computation of 8-point DFT using radix-2 FFT involves three stages of computation. Here N =
8 = 23, therefore, r = 2 and m = 3. The given 8-point sequence is decimated into four 2-point sequence. For
each 2-point sequence, the two point DFT is computed. From the results of four 2-point DFTs, two 4-point
DFTs are obtained and from the result of two 4-point DFTs, the 8-point DFT is obtained
Let the given 8-sample sequence x(n) be {x(0),x(1),x(2),x(3),x(4),x(5),x(6),x(7)}. The 8-samples should
be decimated into sequences of two samples. Before decimation they are arranged in bit reversed order as
shown in Table (10.2)
Table 10.2 Normal and bit reversed order for N - 8
Normal order Bit reversed order

x(0) x(000) x(0) x(000)
x(1) x(001) x(100)
x(2) x(010) x(2) x(010)
x(3) x(011) x(6) x(110)
x(4) x(100) x(1) x(001)
x(5) x(101) x(5) x(101)
x(6) x(110) x(3) x(011)
x(7) x(111) x(7) x(111)

The x(n) in bit reversed order is decimated into 4 numbers of 2-point sequence as shown below.
(ii) x(0) and x(4)
(ii) x(2) and x(6)
(iii) x(1) and x(5)
(iv) x(3) and x(7)
Using the decimated sequences as input, the 8-point DFT is computed. Figure (10.11) shows the
three stages of computation of an 8-point DFT.
x(0) Compute X(0)

x(4) 2-point DFT Combine 2-point X(1)
DFTs to get
x(2) Compute 4-point DFT X(2)
Combine
x(6) 2-point DFT X(3)
4-point
DFTs to
x(1) Compute get 8-point X(4)
x(5) 2-point DFT DFT
Combine 2-point X(5)
DFTs to get
x(3) Compute 4-point DFT X(6)
x(7) 2-point DFT X(7)
Figure 10.9 : Three stage of computation in 8-point DFT.
The basic computational unit of the FFT, shown in figure (10.10) (a), is called a butterfly. This
structure may be simplified by factoring out a term WNr from the lower branch as illustrated in figure
(10.10) (b). The factor that remains is WNN / 2 = 1. A complete eight-point radix-2 decimation-in-time FFT is
shown in figure (10.11)
WNr
WNr N /2
WNr
Figure 10.10 : (a) The butterfly, which is the basic computational element of the FFT
algorithm (b) A simplified butterfly, with only one complex multiplication .
Computing an N-point DFT using a radix-2 decimation-in-time FFT is much more efficient than
calculating the DFT directly. For example, if N = 2v, there are log2N = v stages of computation. Because each
stage requires N/2 complex multiplies by the twiddle factors WNr and N complex additions, there are a total
N
of log2N complex multiplications and N log2N complex additions.
2
From the structure of the decimation-in-time FFT algorithm, note that once a butterfly operations

has been perfomed on a pair of complex numbers, there is no need to save the input pair. Therefore, the
output pair may be stored in the same registers as the input. Thus, only one array of size N is required, and
it is said that the computations may be performed in place. To perform the computations in place, however,
the input sequence x(n) must be stored (or accessed) in nonsequential order as seen in figure 10.11.
To calculate number of complex multiplication and complex addition in FFT
Case 1 : Radix 2 FFT If N = 2v then these are log2 N = v number of stages and each stage has N / 2
number of butterflies. Each butterfly need one multiplication of twiddle factor WNr and 2 complex addition
thus
x(0) X(0)
W80
x(4) X(1)
W80
x(2) X(2)
W80 W82
x(6) X(3)
W80
x(1) X(4)
W80 W81
x(5) X(5)
W80 W82
x(3) X(6)
W80 W82 W83

x(7) X(7)
Figure 10.11 : A complete eight-point radix-2 decimation-in-time FFT.
Study Note
Here we have taken multiplication with ± 1
Total number of complex multiplication =Number of stages butterflies in each stage Number
of complex multiplication in each butterfly
N
=v 1
2
N
=v
2

N
= Log 2 N . .
2
Total Number of complex addition =Number of stage Number of butterflies in each
stage number of complex addition each stage
N
= Log 2 N 2
2
=N log2 N
Case 2 : Radix 3 FFT
If N = 3v then we use radix 3 FFT then we divide N point sequence into 3 sequence x(3n),
N
x(3n+1) and x(3n+2), n = 0, 1, --- 1 .Let F1(k) is N point DFT of x(3n), F2(k) is N DFT
3 3 3
N
of x(3n+1) and F3(k) is point DFT of x(3n+2) and
3
N 1
X(k ) = x n WNnk
n 0
N N N
1 1 1
3 3 3
3n 1 k 3n 2 k
= x 3n WN3nk x 3n 1 WN x 3n 2 WN
n 0 n 0 n 0
N N N
1 1 1
3 3 3
= x 3n WNnk WNk x 3n 1 WNnk WN2 k x 3n 2 WNnk
n 0 3 n 0 3 n 0 3
X(k) = F1 k WNk F2 k WN2 k F3 k

N
Here for 0 k 1, let m = k
3
= F1 m WNm F2 m WN2 m F3 m
Since F1(k), F2(k) and F3(k) are

N point DFT so they will be priodic with period N
3 3
N 2N
Thus for k 1, let m = k N
3 3 3
N N N
3 3 3
= F1 m WNmWNN /3 F2 m WN2 mWN2 N /3 F3 m

2N
and for k N 1, let m = k 2N/3
3

2N 2N 2N
3 3 3
= F1 m WNmWN2 N /3 F2 m WN2 mWN4 N /3 F3 m
Since there is no relation between each twiddle factor and WN2 N /3 , WNN /3 are complex thus for each
butterfly we need 6 complex multiplication
F1(0) X(0)
F1(1) X(1)
W91
W92
F1(2) X(2)
F2(0) X(3)
W94
F2(1) X(4)
W98
F2(2) X(5)
F3(0) X(6)
W97
F3(1) X(7)
W914
F3(2) X(8)
Figure 10.12 : First stage of 9 point DIT FFT(only one butterfly)
We can see that in one butterfly we need 6 complex multiplication (if we include ±1 multiplication)
and there is no relation between WNk and WN2k here so we cannot reduce number of complex
N
multiplication. Also each butterfly need 6 complex addition. There are butterfly in each stage
3
and log3N stages thus
Total number of complex multiplication are
N
Log3N 6 = 2 N log3N
3
Total number of complex addition are
N
Log3N 6 = 2N Log3N
3
Case 3: Radix 4 FFT
When N = 4v we use radix 4 FFT where in decimation in time FFT we divide N point input
N
sequence into four sequence x(4n), x(4n+1), x(4n+2), x(4n+3), n = 0, 1,2,---- 1 .Let F1(k)
4
N N point DFT of x(4n +1), F (k) is N point DFT of x(4n +2),
is point DFT of x(4n), F2(k) is 3
4 4 4
N
F4(k) is point DFt of x(4n+3).
4

N 1
Since, X(k ) = x n WNnk

n 0
N N N
1 1 1
4 4 4
4n 1 k
= x 4n WN4 nk x 4 n 1 WN x 4n 2 WN4 n 2 k
n 0 n 0 n 0
N
1
4
4n 3 k
x 4n 3 WN
n 0
N N N
4 1 4 1 4 1
nk k nk 2k
X(k ) = x(4n)W N W
N x(4n 1)W N W N x 4n 2 WNnk
n 0 4 n 0 4 n 0 4
N
4 1
3k
W N x 4n 3 WNnk
n 0 4
X(k) = F1 k WNk F2 k WN2 k F3 k WN3k F4 k

Here F1(k), F2(k), F3(k), F4(k) will be periodic with period N/4.
N
Thus for 0 k 1 , let m = k
4
X(k) = F1 k WNk F2 k WN2 k F3 k WN3k F4 k
= F1 m WNm F2 m WN2m F3 m WN3m F4 m

N N
For k 1, let m = k N/4
4 2
N N N N
X(k) = F1 k WNk F2 k WN2 k F3 k WN3 k F4 k
4 4 4 4
= F1 m WNmWNN /4 F2 m WN2 mWNN /2 F3 m WN3mWN3 N /2 F4 m
= F1 m j WNm F2 m 1 WN2 m F3 m j WN3m F4 m

N 3N
For k 1, let m = k N/2
2 4
Using same method we get
X(k) = F1 m WNm F2 m WN2 m F3 m WN3m F4 m
3N 3N
and for k N 1, let m = k
4 4
We get
X(k) = F1 m jWNm F2 m 1 WN2 m F3 m j WN3m F4 m

To find X(k) we need to multiply F2(m) with WN2m , F3(m) with WN3m and F4(m) and then for each X(k)
seperately we multiply with 1, j, j, thus we say that only 3 complex multplication needed in each 4 point
N
butterfly multiplication with 1, j, j are not taken as complex multiplication. Thus each stage has
4
butterflies and log4N number of stages, Thus total number of complex multiplication is
N 3N
3 Log 4 N = Log 4 N
4 4
In each butterfly we need 12 complex addition. Thus total number of complex addition will be
N
12 Log 4 N = 3N log4N
4
10.12 Decimation in Frequency (DIF) Radix 2 FFT

In decimation in frequency algorithm, the frequency domain sequence X(k) is decimated. In this
algorithm, the N-point time domain sequence is converted to two numbers of N/2-point sequences. Then
each N/2-point is converted to two numbers of N/4-point sequences. This process is continued until we get
N/2 numbers of 2-point sequences. Finally, the 2-point DFT of each 2-point sequence is computed. The
2-point DFTs of N/2 numbers of 2-point sequence for N/2 will give N-samples, which is the N-point DFT of
the time domain sequence. Here the equaions for N/2-point sequence, N/4-point sequence, etc., are obtained
by decimation of frequency domain sequences. Hence this method is called DIF. The number of complex
multiplication and addition are same in DIT and DIF FFT. No need of analysis of DIF thats why we have
skipped it from the book.
Study Note
Comparison of DIT (Decimation-in-time) and DIF (Decimation-in-frequency) algorithms
Diffrence between DIT and DIF
1. In DIT, the input is bit reversed while the output is in normal order. For DIF, the reverse is true, i.e. the
input is normal order, while the output is it reversed. However both DIT and DIF can go from normal to
shuffled data or vice versa.
2. Consider the butterfly operation, while in DIF, the complex multiplication takes place before the add
subtract operations, while in DIF, the complex multiplication takes place after the add subtract opera-
tions.
Similarities
1. Both algorithms reqire the same number of operations to compute DFT.
2. Both algorithms require bit reversal at some place during computation.
10.13 Computation of IDFT through FFT

The IDFT of an N-point sequence {X(k)}; k = 0, 1, ...., N 1 is defined as
N 1 2 N 1
1 j nk 1
x(n) = X k e N
X k WN nk
N k 0 N k 0
Taking the conjugate of the above eqation for x(n), we get

N 1 * N 1
1 nk 1
x (n) =
* X k WN X * k WNnk
N k 0 N k 0
Taking the conjugate of the above equation for x*(n), we get
N 1 *
1 * nk
x(n) = X k WN
N k 0
The term inside the square brackets in the above equation for x(n) is same as the DFT computa-
tion of a sequence X*(k) and may be computed using any FFT algorithm. So we can say that the IDFT of
X(k) can be obtained by finding the DFT of X*(k), taking the conjugate of that DFT and dividing by N.
Hence, to compute the IDFT of X(k) the following procedure can be followed
1. Take conjugate of X(k), i.e. determine X*(k).
2 . Compute the N-point DFT of X*(k) using radix-2 FFT.
3 . Take conjugate of the output sequence of FFT.
4 . Divide the sequence obtained in step-3 by N.
The resultant sequence is x(n)
Thus,a signal FFT algorithm serves the evaluation of both direct and inverse DFTs.
Example 10.18
Calculate the percentage saving in calculations in a 256-point radix-2 FFT, when

compared to direct DFT?
Solution 10.18
Direct computation of DFT
Number of complex multiplications = N2 = (256)2 = 65,356.
Number of complex addit ion = N(N 1) = 256(256 1) = 65,280
Radix 2 FFT
N 256
Number of complex multiplications = log 2 N log 2 256 128 log 2 28
2 2
= 1280
Number of complex additions = N log2N = 256 log2 256 = 256 log228 = 256 8 = 2048.
Percentage saving
Number of multiplications in radix-2 FFT

Percentage saving in multications = 100- ×100
Number of multiplications in direct FFT
1024
= 100 100 98.43%
65536
Number of multiplications in radix-2 FFT

Percentage saving in additions = 100- ×100
Number of multiplications in direct FFT

2048
= 100 100 96.86%
65280
Example 10.19
Speech that is sampled at a rate of 10 kHz is to be processed in real time. Part of the
computations required involve collecting blocks of 1024 speech values and computing a
1024-point DFT and a 1024 point inverse DFT. If it takes 1 s for each real multiply. how
much time remains for processing the data after the DFT and the inverse DFT are
computed?
Solution 10.19
With a 10-kHz sampling rate. a block of 1024 samples is collected every 102.4 ms. With a radix-2
FFT, the number of complex multiplications for a 1024-point DFT is approximately 512 log2 1024 =
5120. With a complex multiply consisting of four real multiplies. this means that we have 5.120. 4 =
20,480 real multiplies for the DFT and the same number for the inverse DFT. With 1 s per multiply.
this will take
t = 2 . 20.48 = 40.96 ms
which leaves 61.44 ms for anly additional processing
Example 10.20
If FFT processor take 20 sec to compute complex multiplication and negligible time to
comput complex addtion then calculate time to compute FFT when
1 . N = 4096 using radix 2 FFT (including ±1 multiplication as complex multiplication)
2 . N = 729 using radix 3 FFT (including ±1 multiplication as complex multiplications)
Solution 10.20
1. Here total number of complex multiplication in radix 2 FFT are
N N
Log 2 N 1 = Log 2 N
2 2
N
Thus total time = Log 2 N 20 sec
2
= 49.1520 10 2
sec
N
2. Here total number of complex multiplication in radix 3 FFT are 6 Log3 N
3
= 2 N Log3N
Total time = 2 N Log3N 20 sec
= 17.496 10 2 sec

Cha pter
Digital Filter Design

11
11.1 Introduction
There are two type of filters Finite Impulse Response (FIR) and Infinite Impulse Response (IIR). The
type of filters that use feedback connection for realisation are called recursive filters and such filters have
impulse response of infinite duration and are called IIR filters. The type of filters that don’t use feedback
connection for realisation are called nonrecursive filter and such filters have finite duration of impulse response
and are called FIR filters. In this chapter we will study about design of IIR and FIR filters. The most popular
method is to design an analog filter and then transform it into equivalent digital filter.
11.2 Design of IIR filters

We first of all design an analog filter and then convert it into digital filter. The system describing analog
filter may be written as
M
bk sk
Y( s) k 0
Ha ( s ) = N
...(1)
X( s)
ak sk
k 0
Where {ak} and {bh} are filter coefficients, the impulse response of the system is h(t) then
Ha ( s ) = h(t ) e st dt ...(2)
The differential equation of system will be

N
dk y(t )
M
dk x(t )
ak = bk ...(3)
k 0 dtk k 0 dt k
Where x(t) is input to system and y(t) is output of system. For stable and causal analog filter the
conditions are
1. Ha(s) should have real coefficients ak and bk.
2. The poles of Ha(s) should lie on the left half of s-plane.
3. The number of zeros in Ha(s) should be less than or equal to number of poles of Ha(s).
Digital Filter Design 481
For stable and causal digital filter the conditions are:

1. The H(z) should be rational function of z and coefficients of ‘z’ should be real.
2. The poles of H(z) should lie within the unit circle in z-plane.
3. The number of zeros in H(z) should be less than or equal to number of poles in H(z).
REMEMBER • From the above analysis we can see that the left half of s-plane should map into unit circle
of z-plane and imaginary axis of s-plane should map into unit circle of z-plane.
• The physically realizable and stable IIR filter cannot have a linear phase. Only FIR filters
can have linear phase.
11.2.1 Design of IIR filter by approximation of derivative

Since the differential equation of analog system is
N
dk y(t )
M
dk x(t )
ak = bk
k 0 dtk k 0 dt k
In this method of IIR filter design the derivative is approximated and analog filter is converted into
dy(t )
digital filter. The derivative at t = nT is (this is called backward difference method),
dt
dy(t ) y nT y n 1T
=
dt t nT T
dy(t) y n y n 1
or =
dt t nT T
Where T is the sampling interval, y[n] = y[nT]. Thus in analog domain a differentiator system is H(s) = s,
but in z-Transform or in digital system corresponding to it will be
1
1 z
H(z) =
T
dx(t )
That in analog domain if input to system is x(t) and output is then system transfer function
dt
x n x n 1
H(s) = s, converting this into digital system, if input to system is x[n] then output is .
T
1
1 z
Thus, H(z) =
T
From the above analysis we get that
1
1 z
s = ....(4)
T
The equation (4) show the transformation from analog to digital domain. Similarly we can prove that
1 2
1 z
second derivative in analog domain s2 can be written as in digital domain. Thus ith derivative in analog
T

1 i
1 z
domain (si) can be written as in digital domain.
T
1 i
1 z
That is, si = ...(5)
T
Thus we can convert any analog filter Ha(s) into digital filter H(z) by using following approximation
H(z) = Ha ( s) s 1 z 1 ...(6)
T
The outcomes of the mapping of the z -plane from the s-plane are discussed below.
We have,
1
1 z
s = ...(7)
T
1
and z=
1 sT
Substituting, s = j in the expression for z, we have
1
z= ...(8)
1 j T
1 T
= j ...(9)
1 T2 1 2 2
T
Varying from – to the corresponding locus of points in the z-plane is a circle wth radius 1/2 and
with centre at z = 1/2,as shown in Fig. 11.1.
It can be observed that the mapping of the equation (7), takes the left half plane of s-domain into the
corresponding points inside the circle of radius 0.5 and centre at z = 0.5. Also the right half of the s-plane is
mapped into the outside of the unit circle. Because of this, this mapping results in a stable analog filter transformed
into a stable digital filter. However, since the location of poles in the z-domain are confined to smaller frequencies,
this design method can be used only for tranforming analog low-pass filters and band pass filters which are
having smaller resonant frequencies. This means that neither a high-pass filter nor a band-reject filter can be
realized using this technique.
j Im(z)
z-plane
s-plane
s Re(z)
0 0 0.5 1
Unit circle
Fig. 11.1 : Mapping of s-plane into z-plane by the backward differene method.
The forward difference can be substituted for the derivative instead of the backward difference.

This provides,
dy(t ) y( nT T ) y( nT )
= ...(10)
dt T
y( n 1) y( n)
= ...(11)
T
The tranformation formula would be
z 1
s = ...(12)
T
or, s = 1 + sT ...(13)
The mapping of the equation z = 1 + sT is shown in Fig. 11.2. This results in a worse situation than the
backward difference substitution for the derivative. When s = j , the mapping of these points in the s-domain
results in a straight line in the z-domain with coordinates (zreal, zimg) = (1, T). As a result of this, stable analog
filters do not always map into stable digital filters.
Im(z)
j
z-plane
s-plane
Re(z)
0 0 1
Unit circle
Fig. 11.2 : Mapping of s-plane into z-plane by the forward difference method.
Example 11.1
Convert the analog low-pass filter specified by

2
Ha(s) =
s 3
Into a digital filter making use of the backward difference for the derivative.
Solution : 11.1
We know that the mapping formula for the backward difference for the derivative is given by
1
1 z
s =
T
2
For the given analog filter function Ha(s) = , the corresponding digital filter function is
s 3
2 2T
H(z) = Ha (s) 1 z1 = 1
s 1 1 z 3T
T (1 z )
3
T
2 2
If T = 1s, H(z) = 1 1
1 z 3 4 z

Example 11.2
Making use of the backward difference for the derivative, convert the analog filter function
given below to a digital filter function.
4
Ha(s) =
s2 + 9
Solution : 11.2
The mapping formula for the backward difference by the derivative is :
1
1 z
s =
T
Therefore, for the given Ha(s), the corresponding digital filter function is :
4 4T 2
H(z) = Ha ( s) s 1 z 1 =
1 2 1 2z 1
z 2
9T 2
T 1 z
9
T
4 4
If T = 1s, then = 1 2 2 1 2
1 2z z 9T 10 2 z z
Example 11.3
Convert the analog filter given below into a digital filter using the backward difference for
the derivative:
3
Ha(s) =
( s 0.5)2 16
Solution : 11.3
For the given Ha(s), the system function of the corresponding digital filter is:
3
H(z) = Ha ( s) 1 z 1
s ( s 0.5)2 16 1 z 1
T s
T
3 3T 2
= 2
=
1 z 1 [(1 0.5T ) z 1 ]2 16 T 2
0.5 16
T
3T 2
= 2 2 1
1 0.5T z 2 1 0.5T z 16T 2
3 3
T = 1 s, then H(z) = 2 1 1 2
2.25 z 3z 16 18.25 3z z

11.2.2 Design of IIR filter by Impulse Invariant Transformation

In this technique the desired impulse response of digital filter is obtained by sampling the impulse
response ofequivalent analog filter. The sampling period T should be very small such that aliasing don’t take
place.
Let ha(t) is impulse response of analog filter T is sampling period h(n) is impulse response of digital
filter.
For Impulse Invariant Transformation,
h(n) = ha (t ) t nT
ha (nT ) ...(14)
The laplace transform of h(t) is transfer function of system,
Ha (s) = L[ha(t)] ...(15)
N
Ai
If, Ha ( s ) = ...(16)
i 1 s pi
Thus analog filter has simple poles at pi then taking inverse laplace transform of Ha(s) to get impulse
response of stable and causal system (ha(t)), that is
N
ha(t ) = Ai epi t u(t ) ...(17)
i 1
N
Thus, h(n) = ha (t ) t nT
Ai epi nT u( nT ) ...(18)
i 1
Now taking z-Transform of h(n) we get

N
H(z) = h( n) z n
= Ai epi nT u( nt ) z n
n 0 i 1n 0
N
Ai
H(z) = ...(19)
i 1 1 epi T z 1
Thus the transformation from analog to digital domain is
1 Transformation 1
s pi 1 e pi T
z 1 ...(20)
The above mapping shows that the analog pole at s = pi is mapped into a digital pole at z = epiT.
Therefore, the analog poles and the digital poles are related by the reation.
z = esT
The general characteristic of the mapping s = esT can be obtained by substituting, s = +j and expressing
the complex variable z in polar form as z = re j .
re j = e( + j )T
That means, |z| = r = esT
and z = = T
Study Note
So the relationship between analog frequency and digital frequency is = T or = /T.

As a result of this < 0 implies that 0 < r < 1 and > 0 implies that r > 1 and = 0 implies that r = 1.
Therefore, the left half of s-plane is mapped into the interior of the unit circle in the z-plane. The right half of the
s-plane is mapped into the exterior of the unit circle in the z-plane. This is one of the desirable properties for
stability. The j -axis is mapped into the unit circle in z-plane. However, the mapping or j -axis is not one-to-one.
The mapping = T implies that the trip of width 2 /T in the s-plane for values of in the range
– /T /T maps into the corresponding values of – , i.e., into the entire z-plane. Similarly, the strip
of width 2 /T in the s-plane for values of s in the range /T 3 /T also maps into the interval – , i.e.,
into the entire z-plane. Similarly , the strip of width 2 /T in the s-plane for values of s in the range – /T –3 /T
also maps into the interval – , i.e., into the entire z-plane. Hence the mapping from the analog frequency
to the digital frequency w by impulse invariant transformation is many-to-one which simply reflects the
effects of aliasing due to sampling of the impulse response. Fig. 11.3 illustrates the mapping from the s-plane to
z-plane.
j j
3 /T j1
LHP RHP
/T
0 –1 1
– /T
–j 1
–3 /T
Fig. 11.3 : Mapping of (a) s-plane into (b) z-plane by impulse invariant transformation.
The stability of a filter (or system) is related to the location of the poles. For a stable analog filter the
poles should lie on the left half of the s-plane. That mean for a stable digital filter the poles should lie inside the
unit circle in the z-plane.
REMEMBER Some useful invariant transformation:
1. 1 ( 1)m 1 d m 1 1
m
; s pi
( s pi ) ( m 1)! dsm 1 1 e sT z 1
aT 1
s a 1 e (cos bT ) z
2. aT 2 aT
( s a )2 b2 1 2e (cos bT ) z 1
e z 2
aT 1
b e (sin bT ) z
3. aT 2 aT
( s a )2 b2 1 2e (cos bT ) z 1
e z 2
Example 11.4
2
For the analog transfer function: H a ( s ) = .
( s 1) ( s 3)
Determine: H ( z ) if (a) T = 1 s and (b) T = 0.5 s using impulse invariant method.
Solution : 11.4
2
Given, Ha ( s ) =
( s 1) ( s 3)

Using partial fractions, Ha(s) can be expressed as:

A B
Ha ( s ) =
s 1 s 3
2
A = ( s 1) Ha ( s) s 1
1
s 3s 1
2
B = ( s 3) Ha ( s) s 3
1
s 1s 3
1 1 1
Ha ( s ) =
s 1 s 3 s ( 1) z ( 3)
By impulse invariant transformation, we know that
Ai Ai
s pi (is transformed to)
1 ePi T z 1
Here, Ha(s) has two poles and p1 = –1 and p2 = –3.

Therefore, the system function of the digital filter is:
1 1 1 1
H(z) = p1T 1 p2 T 1
= T 1 3T 1
1 e z 1 e z 1 e z 1 e z
1 1 1 1
( a ) When T = 1s, H(z) =
1 e 1z 1
1 e 3
z 1
1 0.3678 z 1
1 0.0497 z 1
(1 0.0497 z 1 ) (1 0.3678 z 1 ) 0.3181 z 1

=
(1 0.3678 z 1 ) (1 0.0497 z 1 ) 1 0.4175 z 1
0.0182 z 1
1 1 (1 0.223z 1 ) (1 0.0606 z 1 )
( b ) When T = 0.5s, H(z) = 0.5 1 3 0.5 2
1 e z 1 e z (1 0.606z 1 ) (1 0.223z 1 )
1
0.383 z
= 1 2
1 0.829 z 0.135 z
Example 11.5
The system function of an analog filter is expressed as:

s 0.5
Ha(s) =
( s 0.5)2 4
Convert this analog filter into a digital filter suing the impulse invariant tranformation.
(Assume T = 1 s ).
Solution : 11.5
Observe that the given system function of the analog filter is of the standard form,
s a
Ha(s) =
(s a )2 b2
where we are given a = 0.5 and b = 2.
By the impulse invariant transformation,we know that,
aT 1
s a 1 e (cos bT ) z
aT 2 aT
( s a )2 b2 (is transformed to)
1 2e (cos bT ) z 1
e z 2

Therefore, for the given Ha(s), we can write the system function of the digital filter
0.5 T 1
1 e (cos 2) z
H(z) = 0.5 T 1 2(0.5) T 2
1 2e (cos 2T ) z e z
Given, T = 1s, we have
0.5 1
1 e (cos 2) z
H(z) = 0.5 1
1 2e (cos 2) z e 1z 2
1 1
1 0.606( 0.416) z 1 0.252 z
= 1 2 1 2
1 2(0.606) ( 0.416) z 0.3678 z 1 0.504 z 0.3678 z
11.2.3 Design of IIR Filter by the Bilinear Transformation Method

The Impulse Invariant method and approximation of derivative method can only be used for design of
low pass filter and band pass filter and not for high pass and band reject filter. The limitation can be overcome
by using a mapping which is from s-domain to z-domain and is one to one. The transformation which has this
property is called Bilinear Transformation.
That is, the Bilinear Tranformation is a conformal mapping that transforms the imaginary axis of s-plane
into the unit circle in the z-plane only once, thus avoiding aliasing of frequency components. In this mapping, all
points in the left half of s-plane are mapped inside the unit circle in the z-plane, and all point in the right half
of s-plane are mapped outside the unit circle in the z-plane. So the transformation of a stable analog filter
results in a stable digital filter. The Bilinear Transformation can be obtained by using the trapezoidal formula for
the numerical integration. The bilinear transformation is performed by
1
2 1 z 2 z 1
s = 1
...(21)
T 1 z T z 1
T
1 s
2
and z = T ...(22)
1 s
2
1
2 1 z
Thus to convert analog filter into digital filter we need to replace ‘s’ by 1
, That is
T 1 z
H(z) = Ha ( s) s 2 1 z 1
,
...(23)
T 1 z 1
The general characteristic of the mapping z = esT may be obtained by putting s = +j and the
complex variable z in the polar form as z = rej in the above equation for s.
2 z 1 2 re j 1
Thus, s = j
T z 1 T re 1
2 ( re j 1) ( re j
1)
s =
T ( re j 1) ( re j
1)
2 r2 1 2 r sin
= j
T 1 r 2 2 r cos 1 r2 2 r cos

Since, s = +j
2 r2 1
we get, s =
T 1 r 2 2 r cos
2 2 r sin
=
T 1 r2 2 r cos
In the above equation for s, we observe that if r < 1 then s < 0 and if r > 1, then s > 0, if r = 1, then
s = 0. Hence the left half of the s-plane maps into points inside the unit circle in the z-plane, the right half of the
s-plane maps into points outside the unit circle in the z-plane and the imaginary axis of s-plane maps into the
unit circle in the z-plane. This transformation results in a stable digital system.
Relation between Analog and Digital Frequencies

On the imaginary axis of s-plane that is = 0 and correspondingly in the z-plane r = 1.
2 2sin 2 sin
=
T 1 1 2cos T 1 cos
2sin cos
2 2 2 2
= tan
T 1 2cos2 /2 1 T 2
the relation between analog and digital frequencies is,

2
= tan ...(24)
T 2
1 T
or, equivalently, we have = 2 tan ...(25)
2
The above relation between analog and digital frequencies shows that the entire range in is mapped
only once into the range – . The entire negative imaginary axis in the s-plane (from = – to 0) is
mapped into the lower half of the unit circle in z-plane (from = – to 0) and the entire positive imaginary axis
in the s-plane (from = to 0) is mapped into the upper half of unit circle in z-plane (from = 0 to + ). But
as seen in Fig. (11.4) the mapping is non-linear and the lower frequencies in analog domain are expanded in the
digital domain, whereas the higher frequencies are compressed. This is due to the nonlinearity of the arctangent
function and usually known as frequency warping.
1 T
2tan
2
Fig. 11.4 : Mapping between and in Bilinear Ttranformation.

The effect of warping on the magnitude response can be explained by considering an analog filter with
a number of passbands. The corresponding digital filter will have same number of passbands, but with
disproportionate bandwidth, as shown in figure 11.5(a).
In designing digital filter using Bilinear Transformation, the effect of warping on amplitude response
can be eliminated by prewarping the analog filter. In this method, the specified digital frequencies are converted
2
to analog equivalent using the equation tan . This analog frequencies are called prewarp frequencies.
T 2
Using the prewarp frequencies, the anaog filter transfer function is designed, and then it is transformed to
digital filter transfer function.
This effect of warping on the phase response can be explained by considering an analog filter with linear
phase response as shown in Fig. 11.5(b). The phase response of responding digital filter will be nonlinear.
Transformation
Transformation
Resulting
digital filter Resulting
digital filter
|H( )| |H( )|
Analog filter Analog filter
|Ha( )| |Ha( )|
Fig. 11.5 : The warping effect (a) Magnitude response and (b) Phase response
Example 11.6
Convert the following analog filter with transfer function:

s 0.1
Ha(s) =
( s 0.1)2 9
Into a digital IIR filter by using bilinear transformation. The digital IIR filter is having a
r esonant fr equency of r = /2.
Solution : 11.6
From the transfer function, we observe that C = 3. The sampling period T can be determined using the
equation:
2 r
C = tan
T 2
2 r 2 /2
T= tan tan 0.6666 s
C 2 3 2

Using the bilinear transformation, the digital filter system function is,
H(z) = Ha ( s) 21 z 1
Ha ( s)
s
T1 z 1 1 z 1
s 3
1 z 1
1
1 z
3 0.1
s 0.1 1 z 1
H(z) = = 2
( s 0.1)2 9s 3
1 z 1
1 z 1
1 z 1 3 1
0.1 9
1 z
[3(1 z 1 ) 0.1(1 z 1 )] [1 z 1 ]
=
[3(1 z 1 ) 0.1(1 z 1 )]2 9(1 z 1 )2
1 2
3.1 0.2 z 2.9 z
= 1 2
18.61 0.02 z 17.41 z
Example 11.7
4
Apply the bilinear transformation to Ha ( s) , T = 0.5 s and find H ( z ).
( s 3)( s 4)
Solution : 11.7
4
Given that, Ha ( s ) = and T = 0.5s
( s 3)( s 4)
1
2 1 z
To obtain H(z) using the bilinear transformation, replace s by 1
in Ha(s),
T 1 z
4 4
H(z) =
( s 3)( s 4) s 21 z 1 ( s 3)( s 4) s 21 z 1
T1 z 1 T1 z 1
4
= 1 1
1 z 1 z
4 1
3 4 1
4
1 z 1 z
4 1 (1 z 1 )2
= = 2
4 4z 1 3 3 z 1
4 4z 1 4 4z 1 (7 z 1 )
1 z1 1 z1
Example 11.8
A digital filter with a 3dB band width of 0.25 is to be designed from the analog filter whose
response is
C
H( s )
s C
Use bilinear transformation and find H ( z ).

Solution : 11.8
The bandwidth of analog system is C and that of digital filter is 0.25
2
C = tan
T 2
C = 0.808/T
C 0.8028( z 1)
H(z) = H( s) s 2 z 1 =
T z 1 2 z 1 2( z 1) 0.828( z 1)
C
T z 1
Example 11.9
The normalized transfer function of an analog filter is given by

1
H(sn) =
s2n 1.6 sn 1
Convert the analog filter to a digital filter with a cutoff frequency of 0.6 , using the bilinear
transformation.
Solution : 11.9
The prewarping of analog filter has to be preserve the magnitude response. For this analog cutoff frequency
is determined using the bilinear transformation, and the analog transfer function is unnormalized using
this analog cutoff frequency. Then analog transfer function is converted to digital transfer function using
the blinear transformation.
Given that, digital cutoff frequency,
C = 0.6 rad/s.
Let, T = 1s
In the blinear transformation,
2 C 0.6
Analog cutoff frequency, = tan 2 tan 2.753 rad/s.
C T 2 2
Normalized analog transfer function,
1
Ha ( s ) = 2
sn 1.6sn 1
The analog transfer function is unnormalized by replacing sn by s/ C.
Therefore, unnormalized analog filter transfer function is given by
1 1
Ha ( s ) = 2 2
s s s s
1.6 1 1.6 1
C C
2.753 2.753
2.7532 7.579
= 2 2 2
s 1.6 2.753s 2.753 s 4.404s 7.579
The digital filter system function H(z) is obtained by substituting in H(s).
1
2 1 z
s = 1
T 1 z

Here, T = 1. Therefore, the digital filter transfer function is,

7.579
H(z) = 2
1 1
1 z 1 z
2 1
4.404 2 1
7.579
1 z 1 z
7.579(1 z 1 )2
= 1
4(1 2 z z 2 ) 4.404(1 z 1 )2 (1 z 1 ) 7.579(1 z 1 )2
1 1 2
7.579[1 2z z 2] 0.371 0.74 z 0.371 z
= 1 1 = 1 2
20.387 7.158z 2.77 z 1 0.351 z 0.136 z
So in bilinear transform only 4 format of question arise:
1. H(s) is given and resonant frequency can be found from given H(s), thus we find r and resonant
frequency of digital filter r is given in question, using the information and equation (24) we find T and
apply transformation on H(s) to find H(z).
2. H(s) is given and cut-off frequency (3-db bandwidth) can be calculated fron given H(s) and cut-off
frequency (3-dB band width) of digital filter is given in question then we apply equation (24) to find T
and apply transformation on H(s) to find H(z).
3. H(s) is given and T is also given, we can find H(z) by directly applying transformation.
4. Normalised H(s) is given and we have cut off frequency of digital filter then we find C using equation (24)
and replace s by s/ in H(s) and apply transformation in H(s) to get digital filter H(z).
Study Note
In gate direct questions cannot be asked from butterworth and chebyshev because process is lengthy, but can ask
theoretical question so go through the whole process for knowledge and to answer theoretical questions and we are
providing small introduction only on these topics.
11.2.4 Design of Low pass Digital Filter : Batter Worth Filter

Specification of the low pass filter:
Generally specification of digital low pass filter to be designed is given as shown in Fig. 11.6.
|H( )|
1
Digital low
A1 pass filter
A2
1 2
Pass Transition Stop
Band Band band
Fig. 11.6 : Magnitude response of digital low pass filter
We assume that, A1 H( ) 1.0
0 1 is pass band of low pass filter and stop band of low pass filter is
H( ) < A2 and 2

Low-pass Butterworth Filter

First of all an equivalent analog filter is designed and then that analog filter is transformed into digital
filter using any method learned earlier. The magnitude response of analog butterworth filter is given by
2 1
Ha ( ) = 2N
...(26)
1
C
Where C is 3-dB cut-off frequency and N is order of the filter. The frequency response of the butterworth
filter will be
|Ha( )|
1.0 N = 4 N = 10
N=2
N=1
1 2 0.707
Ideal N=1
response
N=2
N=4
N = 10
0 C
Fig. 11.7 : Magnitude response of butterworth low-pass filter various values of N
We can see that more the order of filter the sharper is transition of the filter thus the order depend on
1, 2, A1 and A2.
A1, A2, 1, 2 are specification of analog low pass filter corresponding to digital low pass filter to be
designed. A1and A2 are same and 1, 2 are analog frequency corresponding to 1 and 2. Relation between
and w will depend on the type of transformation used.
The order of the filter will be calculated as
1 1
log 1 1
1 A22 A12
N ...(27)
2 2
log
1
The butterworth lowpass filter transfer function:

The unnormalized transfer function of the butterworth filter is usually written in factored form as:
N /2 2
C
Ha ( s ) = (when N is even) ...(28)
k 1 s2 bk Cs
2
C
N 1
2 2
C C
Ha ( s ) = 2 2
(when N is odd) ...(29)
s C k 1 s bk Cs C

(2k 1)
Here, b k = 2sin
2N
If s/ C (where C is the 3-dB cutoff frequency of the low-pass filter) is replaced by sn, then the normalized
butterworth filter transfer function is given by
N/2
1
Ha ( s ) = 2
sn bk sn 1
(when N is even) ...(30)
k 1
N 1
1 2 1
Ha ( s ) = 2
(when N is odd) ...(31)
sn 1 k 1 sn bk sn 1
(2k 1)
Here, b k = 2sin
2N
Procedure for Designing Low-pass Digital Butterworth Filter

The digital low pass filter has 1 and 2 as pass band and stopband frequencies and A1 is gain at 1 and
A2 is gain at 2.
St ep -1: Find value of 1 and 2 analog frequencies corresponding to 1 and 2. For bilinear
transformation,
2 1 2 2
1 = tan , 2 tan
T 2 T 2
2 tan 2 /2
=
1 tan 2 /2
For impulse invariant tranformation,
1 2
1 = , 2
T T
2 2
=
1 1
St ep -2: Decide the order N of the filter. The order N should be such that
1 1
log 1 1
1 A22 A12
N
2 2
log
1
Choose N such that it is an integer just greater than or equal to the value obtained above.
St ep -3: Calculate the analog cut-off frequency,
1
C = 1 / 2N
1
1
A12
For bilinear transformation,
2
tan 1 /2
= T
C 1 / 2N
1
1
A12

For impulse invariant transformation,

1 /T
C = 1 / 2N
1
1
A12
St ep -4: Determine the transfer function of the analog filter. Let Ha(s) be the transfer function of the
analog filter. When the order N is given, for unity dc gain filter, Ha(s) is given by
N/2 2
C
Ha ( s ) =
k 1 s2 bk C
s 2
C
When the order N is odd, for unity dc gain filter, Ha(s) is given by
N /1
2 2
C C
Ha ( s ) =
s C k 1 s2 bk C
s 2
C
The coefficient bk is given by

(2k 1)
b k = 2sin
2N
St ep -5: Using the chosen transformation, transform the analog filter transfer function Ha(s) to digital
filter transfer function H(z).
Properties of Butterworth Filter:
1. The butterworth filters are all pole designs (i.e., the zeros of the filters exist at ).
2. The filter order N completely specifies the filter.
3. The magnitude response approaches the ideal response as the value of N increases.
4. The magnitude is maximally flat at the origin.
5. The magnitude is monotonically decreasing function of .
6. At the cutoff frequency C,
the magnitude of normalized butterworth filter 1 / 2 . Hence the dB
magnitude at the cutoff frequency will be 3dB less than the maximum value.
Example 11.10
Design a butterworth digital filter using the bilinear transformation. The specifications of
the desired low-pass filter are:
0.9 H 1; 0
2
3
= H( ) 0.2;
4
with T = 1 s
Solution : 11.10
The butterworth digital filter is designed as per the following steps. For the given specification, we have
A 1 = 0.9 and 1
2
3
A 2 = 0.9 and 2 and T = 1s
4
Here the bilinear transformation is already specified.

Step-1: Determination of the ratio of the analog filter’s edge frequencies, 2/ 1

2 2 2 (3 / 4) 3
2 = tan tan 2 tan 4.828
T 2 1 2 8
2 1 2 ( / 2)
1 = tan tan 2 tan 2
T 2 1 2 4
2
= 2.414
1
Step-2: Determination of the order of the filter N,
1 1 1 1
log 1 1 log 1 2
1
1 A22 A12 1 0.2 0.9
N
2 2 2 log1.207
log
1
1 log 24 / 0.2345
2.636
2 log 2.414
Since, N 2.636, choose N = 3
Step-3: Determination of the analog cutoff C (i.e., –3-dB frequency)
1 2
C = 1 / 2N 1/ 2 3
2.5467
1 1
1 1
A12 0.92
Step-4: Determination of the transfer function of the analog Butterworth filter Ha(s).
For odd N, we have
N 1
2 2
C 1
Ha ( s ) =
s C k 1 s2 bk Cs
2
C
(2k 1)
where, b k = 2sin
2N
For N = 3, we have
2
C C
Ha ( s ) =
s C s2 bk Cs
2
C
(2 1 1)
Where, b1 = 2sin 1
2 3 6
2.5467 (2.5467)2
Therefore, Ha ( s ) =
s 2.5467 s2 1(2.5467) s (2.5467)2
Step-5: Conversion of Ha(s) into H(z).
Since bilinear transformation is to be used, the digital filter transfer function is:
H(z) = Ha ( s) s 2 1 z 1 Ha ( s) s 2
1 z 1
T 1 z 1 1 z 1

2.5467 (2.5467)2
H(z) = 1 2
1 z 1 z 1
1 z 1
2 1
2.5467 2 2.5467 2 2.5467
2
1 z 1 z 1
1 z 1
0.2332(1 z 1 )3
= 1 2 3
1 0.4394z 0.3845z 0.0416z
11.2.5 Design of Digital Low-pass Chebyshev Filter

Here also the method is same, that is we first of all design the analog filter and then transform it into
digital domain.For designing a chebyshev IIR digital filter, first an analog filter is designed using. the given
specifications. Then the analog filter transfer function is transformed to digital filter transfer function by using
either Impulse Invariant Transformation or Bilinear Transformation.
The analog Chebshev filter is designed by approximating the ideal frequency response using an error
function. There are two type of Chebyshev approximations. In type-1 approximation, the error function is selected
such that the magnitude response is equiripple in the passband and monotonic in the stopband. In type-2
approximation, the error function is selected such that the magnitude function is monotonic in the passband
and equiripple in the stopband. The type-2 magnitude response is also called inverse Chebyshev response.
The magnitude response of type-1 Chebyshev low-pass filter is given by
2 1
Ha ( ) = ...(32)
2 2
1 cN
C
Where e is attenuation constant given by

1/ 2
1
= 1 ...(33)
A12
A1 is the gain at the passband edge frequency 1 and CN is the Chebyshev polynomial of the first
C
kind of degree N given by

CN(x) = {cos(N cos–1 x)}, for |x| 1 ...(34)
= {cos(N cosh–1 x)}, for |x| 1 ...(35)
and C is the 3-dB cut-off frequency.
The frequency response of Chebyshev filter depends on order N. The approximated response approaches
the ideal response as the order N increases. The phase response of the Chebyshev filter is more nonlinear than
of the Butterworth filter for a given filter length N. The magnitude response of type-1 Chebyshev filter is shown
in figure.

|Ha ( )|
|Ha ( )|
1 1
1 1
2
1 1 2
A2 A2
2 2
Fig. 11.8 : Magnitude response of type-1 Chebyshev filter
11.2.6 Designing Other Filter using Low-pass Filter

Basically there are four types of frequency selective filters, viz. low-pass, high-pass, band pass and band
stop. In Fig. 11.9, the frequency of the ideal case is shown in solid lines and practical case in dotted lines.
|Ha ( )| |Ha ( )|
1 1
C 2 C 2
|Ha ( )| |Ha( )|
Fig. 11.9 : Frequency response of (a) Low-pass filter (b) High-pass filter,
(c) Band pass and (d) Band stop filter

In the design techniques so far, we have considered only low-pass filters. This low-pass filter can be
considered as a prototype filter and its system function Hp(s) can be determined. This high-pass or band pass or
band stop filters are designed by designing a low-pass filter and then transforming that low-pass transfer function
into the required filter function by frequency transformation. Frequency transformation can be accomplished in
two ways.
1. Analog frequency transformation
2. Digital frequency transformation
Note: Digital frequency transformation is lengthy cannot be asked in GATE.
Analog Frequency Transformation

In the analog frequency transformation, the analog system function Hp(s) of the prototype filter is converted
into another system function H(s) of the desired filter (a low-pass filter with another cutoff frequency or a high
pass filter or a band pass filter or a band stop filter). Then using any of the mapping techniques (Impulse
Invariant Transformation or Bilinear Transformation) this analog filter is converted into the digital filter with a
system function H(z).
The frequency transformation formulae used to convert a prototype low-pass filter into low-pass (with
a different cutoff frequency), high-pass band pass or band stop are given in Table 11.1. Here C is the cut-off
*
frequency of the low-pass prototype filter. C cutoff frequency of new low-pass filter or high pass filter and 1
and 2 are the cutoff frequencies of band pass or band stop filter.
Type Transformation
s
Low-pass s C *
C
*
High-pass s C
C
s
s2 1 2
Band-pass s C
s( 2 1)
s( 2 1)
Bandstop s C
s2 1 2
Table 11.1 : Analog Frequency Transformation
0 is the centre frequency, 0 = 1 2
0
Quality factor, Q=
2 1
Example 11.11
A prototype low-pass filter has the system function,

1
Hp ( s )
s2 3 s 2
obtain a band pass filter with 0 = 3 rad/s and Q = 12.

Solutions : 11.11
We know that the centre frequency,
0 = 1 2
0
and quality factor, Q=
2 1
From Table 11.2, we have the low-pass to band pass transformation

s2 1 2 s2 2
s C = C
0
s( 2 1) s( 0 / Q)
s2 32 s2 9
s C = 4 C
s(3 /12) s
Therefore, the transfer function of band pass filter is,
H (s ) = H p ( s ) s 4 C
s2 9
s
1
= 2
s2 9 s2 9
4 C 3 4 C 2
s s
1 s2
= 2 4 3 3
16 Cs 0.75 Cs (18 c 0.125) s2 6.75 Cs 81 2
C
11.3 FIR Filter

The FIR filters are those which have impulse response non zero only for finite values of n. The present
output of FIR filter depend only on the present and past input samples where as for IIR filter the present ouput
depend on present and past input and past values of reponse.
The following are the main advantages of FIR filters over IIR filter:
1. FIR filter are always stable.
2. FIR filters with exactly linear phase can easily be designed.
3. FIR filters can be realized in both recursive and non-recursive structures.
4. FIR filter are free of limit cycle oscillations, when implemented on a finite word length digital
system.
5. Excellent design methods are available for various kinds of FIR filters.
The disadvantages of FIR filters are as follows:
1. The implementation of narrow transition band FIR filters is very costly, as it requires considerably
more arithmetic operations and hardware components such as multipliers, adders and delay
elements.
2. Memory requirement and execution time are very high.
Note: FIR filter will have linear phase only when the it’s impulse response has centre of symmetry.
Frequency response of any system is the fourier transform of it is impulse response. The fourier transform
of h(n) is H(e j ), thus magnitude response of system is |H(ej )| and phase response is H(ej ).

Since h(n) is impulse response of FIR filter and N is it’s length. Depending on the value of N (odd or
even) and the type of symmetry of the filter impulse response sequence (symmetric or antisymmetric), there
are following four possible types of impulse response for linear phase FIR filters.
1. Symmetrical impulse response when N is odd.
2. Symmetrical impluse response when N is even.
3. Antisymmetric impulse response when N is odd.
4. Antisymmetric impulse response when N is even.
11.3.1 Frequency Response of Linear Phase FIR Filter When Impulse Response is
Symmetrical and N is Odd
When N is odd then h(n) is nonzero only for 0 n N – 1,
( N 1)
j n
Thus, H( ej ) = h( n) e ...(36)
n 0
Since, h(n) is symmetric about n = (N – 1)/2

Thus, h(n) = h(N – 1 – n)
Let, N= 9
8
j n
Then, H( ej ) = h( n) e
n 0
= h(0) + h(1) e–j + h(2) e–j2 + h(3) e–j3 + h(4) e–j4

+ h(5) e–j5 + h(6) e–j6 + h(7) e–j7 + h(8) e–j8
Now taking common e–j4 and using h(n) = h(8 – n) we get
H(ej ) = e–j4 [2h(0) cos4 + h(1) cos3 + 2h(2) cos2
+ 2h(3) cos + h(4)]
j
Thus phase of H(e ) = –4 and [2h(0) cos4 + h(1) cos3 + 2h(2) cos2 + 2h(3) cos + h(4)] is
magnitude. Also we can see that magnitude is even function of .
Thus in general if length of sequence is N and impulse response is symmetric then phase of H(ej )
will be
N 1
H (e j ) =
2
and magnitude will be even function of .
Centre of symmetry
j
|H(e )| Centre of symmetry
h(n) N=9
1 7 n n
0 2 3 4 5 6 8 0 2
Fig. 11.10 : (a) Symmetrical impuse response, N = 9, (b) Magnitude function of H( )

Study Note
N 1 N 1
If h(n) is real and symmetric about n then if we generate a new signal g[n] by left shifting h[n] by
2 2
then we will get g[n] a real and even signal.
If g(n) G(e j )
h(n) H(e j )
n N 1
And since g(n) is real and even then phase of G(ej ) wil be zero and g(n) = h then ,
2
N 1
j
j 2
G(e j ) = H( e ) e
N 1
Since, phase of G(e j ) = 0 then phase of H(e j ) will be
2
Symmetrical and N is Even
Here Length is even then h(n) is nonzero for 0 n N – 1.
N 1
j
Thus, H( ej ) = h( n) e
n 0
N 1
and h(n) = h(N – 1 – n), centre of symmetry is .
2
N 1
Since real h(n) can be made real and even by shifting it left by we get phase of H(ej ) is again
2
N 1
, and magnitude response will also be even in this case.
2
For example: N = 8, thus h(n) = h(7 – n)

7
j n
and H( ej ) = h( n) e
n 0
= h(0) + h(1) e–j + h(2) e–j2 + h(3) e–j3 + h(4) e–j4

+ h(5) e–j5 + h(6) e–j6 + h(7)–j7
Taking, e–j3.5 common and h(n) = h(7 – n) we get
H(ej ) = e–j3.5 [2h(0) cos(3.5 ) + 2h(1) cos(2.5 )
+ 2h(2) cos(1.5 ) + 2h(3) cos(0.5 )]

Centre of symmetry
j
|H(e )| Centre of symmetry
h(n)
1 6 n n
0 2 3 4 5 7 0
Fig. 11.11 : (a) Symmetrical impulse response, N = 8 (b) Magnitude function of H( )
Anti symmetric and N is Odd
Here h(n) is antisymmetric, and length is N.
Thus, h(n) = –h(N – 1 – n)
N 1
j n
n 0
N 1 N 1
Since h(n) is real and antisymmetric about n = thus if we give left shift of to h(n) then
2 2
it will become real and odd and it’s Fourier transform will be imaginary and odd.
N 1
Let, g (n ) = h n
2
N 1
j
G(ej ) = H( e ) e j 2
Since phase of G(ej ) is ± thus phase of H(ej ) will be

2
N 1
H (e j ) =
2 2
For example: N= 9
Thus, h(n)= –h(8 – n)
8
j n
n 0
= h(0) + h(1) e–j + h(2) e–j2 + h(3) e–j3 + h(4) e–j4

+h(5) e–j5 + h(6) e–j6 +h(7) e–j7 + h(8) e–j8
Now taking e–j4 commom and using h(n) = –h(8 – n) we get
h(4) = –h(4)
Thus, h(4) = 0
and H(ej ) = e–j4 2j[h(0) sin(4 ) + h(1) sin(3 ) + h(2) sin(2 ) + h(3) sin( )]
Thus phase of H(ej ) is –4 + 90° and magnitude of H(ej ) is odd.

In general if FIR filter has antisymmetric impulse response h(n) with length N which is odd then
N 1
h = 0
2
N 1
H (e j ) =
2 2
and magnitude response will be odd.
Centre of symmetry
|Ha ( )|
Centre of symmetry
1 3 6 8 2
n
0 2 4 5 7 0
Fig. 11.12 : (a) Antisymmetric impulse response for n = 9 (b) Magnitude function of H( )
Antisymmetrical and N is Even
Here, h(n) is antisymmetric and length N is even.
h(n) = –h(N – 1 – n)
N 1
j n
n 0
N 1 N 1
h(n) is antisymmetric about n , thus if we give left shift of to h(n) we get a real and odd
2 2
signal which will have imaginary and odd Fourier transform, thus phase of H(ejw) will be
N 1
H (e j ) =
2 2
For example: N = 8, h(n) = –h(7 – n)
7
j n
H(ejw) = h( n) e
n 0
= h(0) + h(1) e–j + h(2) e–j2 + h(3) e–j3 + h(4) e–j4

+h(5) e–j5 + h(6) e–j6 + h(7) e–j7
Taking, e–j3.5 common and using h(n) = –h(7 – n) we get
H(ej ) = 2j e–j3.5 [h(0) sin(3.5 ) + h(1) sin(2.5 ) + h(2) sin(1.5 )
+ h(3) sin(0.5 )]
Thus phase of H(ej ) is (–3.5 + /2) and magnitude is odd. Thus for a general FIR filter with
antisymmetric impulse response with odd length.

N 1
The phase response will be H(ejw)=
2 2
and the magnitude response will be odd.
Centre of symmetry
|Ha ( )| Centre of symmetry
1 3 5 7 n
0 2 4 6 0 2
Fig. 11.13 : (a) Antismmetrical impulse response for N = 8 (b) Magnitude function of H( )
11.4 Design Techniques of FIR Filter

The well known methods of designing FIR filters are as follows:
1. Fourier series method
2. Window method
3. Frequency sampling method
4. Optimum filter design method
11.4.1 Fourier Series Method of FIR Filter Design

The frequency response of a digital filter is periodic with period 2 , and
j n
Hd ( e j ) = hd ( n) e
n
So, Hd(ej ) is periodic signal and
1
hd(n) = Hd ( e j ) e j n
d
2
Thus in this method we first of all find Hd(ej ) and then find hd(n), these are desired fourier transform
and impulse response of the digital filter. Thus hd(n) is the impulse response of ideal filter that is to be designed,
but we convert this ideal filter into FIR filter by truncating the impulse response. If h(n) is impulse response of
FIR filter then it is formed by truncating hd(n) into finite duration signal of length N, where N is odd.
N 1 N 1
hd n , for n
Thus, h[n] = 2 2
0, else where
The z-transform of the filter with impulse response h[n] will be
N 1
2
n
H(z) = h( n) z
N 1
n
2

N 1
Since, H(z) will be a non-causal system as h(n) is nonzero for n < 0. Thus we shift h(n) right by
2
N 1
and get causal system with z-transform z 2 H( z) .
Summarizing the procedure for designing FIR filters by fourier series method:
Step-1: Find the desined frequency response Hd(ej ) of filter.
Step-2: Find hd(n) signal corresponding to Hd(ej ).
Step-3: Now truncate hd(n) to finite sequence h(n) of odd length N.
Step-4: Take z-transform of h(n) and find H(z).
N 1
Step 5. Multiply H(z) by z 2 to convert noncausal system into causal FIR system.
Example 11.12
Design a low-pass FIR filter with fibe stage. [Given: Sampling time 1ms , f c = 200 Hz].
Also find the frequency response of the filter.
Solution : 11.12
1
Given that, fc = 200 Hz and fs = 1 kHz
1 ms
The normalized cutoff frequency,

2 fc 200
c = 2 0.4 rad/sec.
fs 100
The given filter can be expressed by the following specifications:
1, for c c
Hd (ej ) = 0, for c
0, for c
The desired impulse response of the filter is given by

0.4
1 1
hd(n) = Hd ( e j ) e j n
d ej n
d
2 2 0.4
0.4
1 ej n 1 e j 0.4 e j 0.4 n
1
= sin 0.4 n
2 jn 0.4
2 jn n
1
When n 0, hd(n) = sin 0.4 n
n
1
when n = 0, the factor sin 0.4 n becomes 0/0, which is interminate.
n
Hence using L’hospital rule, when n = 0,
1
hd(n) = hd (0) Lt sin 0.4 n 0.4
n 0 n

The impulse response of FIR filter is obtained by truncating hd(n) to 5 samples.

So, N = 5, N – 1 = 5 – 1 = 4]
( n 1)
and = 2
2
Therefore, h(n) = hd(n)
( N 1) ( N 1)
For n , i.e., for –2 n 2
2 2
h(n) = hd(n) = 0.4; for n = 0
sin0.4 n
and h(n) = ; for n 0, for 2 n 2
n
when n = 0, h(n) = h(0) = 0.4
when n = 1, h(n) = h(1) = h(–1) = 0.3027
sin 2(0.4 )
when n = 2, h(n) = h(2) 0.0935 h( 2)
2
when coefficients of the designed filter are:
h(0) = 0.4,
h(1) = h(–1) = 0.3027,
h(–2) = 0.0935
The above coefficients correspond to a non-causal filter which is not realizable. The transfer function of the
N 1
realizable digital filter is given by z 2 H( z) , where H(z) is z-transform of h(n).
z-2H(z) = z–2[0.4 + 0.3027[z + z–1] + 0.935[z2 + z–2]]
= 0.0935 + 0.3027 z–2 + 0.4 z–2 + 0.3027 z–3 + 0.0935 z–4
Therefore, the coefficients of the realizable digital filter are:
h(0) = 0.0935, h(1) = 0.3027,
h(2) = 0.4, h(3) = 0.3027, h(4) = 0.0935
The frequency response of the filter is,
H(ej ) = e–j2 [0.4 + 2(0.3027) cos + 2h(0.0935) cos2 ]
= e–j2 [0.4 + 0.6054 cos + 0.1870 cos2 ]
The magnitude function of the filter is,
H( e j ) = 0.4 + 0.6054 cos + 0.1870 cos2
11.4.2 Design of FIR Filter using Windows

The procedure to design FIR filter using window is:
1. Find Hd(ej ) the desired frequency response of system.
2. Find hd(n) the impulse response corresponding to Hd(ej ).
3. Since hd(n) is infinite sequence so it is truncated by multiplying it with a window h(n) = hd(n) (n).
4. The transfer function H(z) corresponding to z-transform of h(n).
There are type of window function:
1 . Rectangular Window:
The weighting function (window function) for an N-point rectangular window is given by
N 1 N 1
1, n
wR (n ) = 2 2
0, elsewhere

If we plot the magnitude response of window function we will get that first side lobe is 13 dB down
the peak of main lobe and roll off is of 20 dB/decade. Also the width of main lobe is 4 /N. Thus decay
is slow and response is poor.
2. Triangular or Barlett Window:
The triangular window has been chosen such that it has tapered sequences from the middle on
either side. The window function wT(n) is defined as
2| n | N 1 N 1
1 , for n
wT(n) = N 1 2 2
0 otherwise
2n N 1 /2
or, wT(n) = 1 ,0 n N 1
N 1
0 , otherwise
In magnitude response of triangular window, the side lobe level is smaller than the of the rectangular
window being reduced from –13 dB to –25 dB. However, the main lobe width is now 8 /N or twice
that ot the rectangular window. The triangular window produces a smooth magnitude response in
both pass band and stop band, but it has the following disadvantages when compared to magnitude
response obtained by using rectangular window:
( i ) The transition region is more.
( ii ) The attenuation stop band is less.
Because of these characteristics, the triangular window is not usually a good choice.
3. Raised Consine Window:
The raised consine window multiplies the central fourier coefficients by approximately unity and
smoothly truncates the fourier coefficients toward the ends of the filter. The smoother ends and
broader middle section produces less distortion of hd(n) aroud n = 0. It is also called generalized
Hamming window. The window sequence is of the form:
2 n N 1 N 1
(1 ) cos , for n
wH ( n ) = N 1 2 2
0 elsewhere
4. Hanning Window
The hanning window function is given by
2 n N 1 N 1
0.5 0.5cos , for n
wH ( n ) = N 1 2 2
0 otherwise
2n
0.5 0.5cos , 0 n N 1
or, wH ( n ) = N 1
0 otherwise
The width of main lobe is 8 /N, i.e., twice that of rectangular window results in doubling of the
transition region of the filter. The peak of the first side lobe is –32 dB relative to the maximum
value. This results in smaller ripples in both pass band and stop band of the low-pass filter designed
using Hanning window. The minimum stop band attenuation of the filter is 77 dB. At higher
frequencies the stop band attenuationis even greater. When compared to triangular window, the
main lobe width is same, but the magnitude of the side lobe is reduced, hence the Hanning window
is preferable to triangular window.

5. Hamming Window
The Hamming window function is given by
2 n N 1 N 1
0.54 0.46 cos , for n
wH ( n ) = N 1 2 2
0 otherwise
2n
0.54 0.46 cos , 0 n N 1
or, wH ( n ) = N 1
0 otherwise
In the magnitude response for N = 31, the magnitude of the first side lobe is down by about 41 dB
from the main lobe peak, an improvement of 10 dB relative to the Hanning window. But this
improvement is achieved at the expense of the side lobe magnitudes at higher frequencies, which
are almost constant with frequency. The width of the main lobe is 8 /N. In the magnitude response
of low-pass filter designed using Hamming window, the first side lobe peak is –51 dB, which is –7 db
lesser with respect to the Hanning window filter. However, at higher frequencies, the stop band
attenuation is low when compared to that of Hanning window. Because the Hamming window
generates lesser oscillations in the side lobes than the Hanning window for the same main lobe
width, the Hamming window is gerally preferred.
6. Black Man Window:
The blackman window function is another type of cosine window and given by the equation
2 n 4 n N 1 N 1
0.42 0.5cos 0.08 cos , for n
wB(n) = N 1 N 1 1 2
0 otherwise
2 n 4n
0.42 0.5cos 0.08cos , 0 n N 1
wB = N 1 N 1
0 otherwise
In the magnitude response, the width of the main lobe is 12 /N, which is highest among windows.
The peak of the first lobe is at –58 dB and the side lobe magnitude decreases with frequency. This
desirable feature is achieved at the expense of increased main lobe width. However, the main lobe
width can be reduced by increasing the value of N. The side lobe attenuation of a lowpass filter
Blackman window is –78 dB.
Table 11.2 frequency domain characteristics of some window function.
Approximate Minimum stop Peak of first

Type of band attenuation side lobe
transition
window (dB) (dB)
width of main lobe
Rectangular 4 /N –21 –13
Barlett 8 /N –25 –25
Hanning 8 /N –44 –31
Hamming 8 /N –51 –41
Blackman 12 / N –78 –58

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Chapter

1.1 Analog, Discrete and Digital Signals

1. Continuous Time and Continuous Amplitude Signal (Analog Signals)

Fig 1.1 : Example of analog signal

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2. Discrete Time and Continuous Amplitude Signals (Discrete Signals)

Fig. 1.2 : Block diagram of Sampler

Fig. 1.3 : Input and Output of sampler

So we can say that output of sampler = Input t = nTs

Fig.1.4 : Example of sampler

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3. Discrete Time and Discrete Amplitude Signals (Digital Signals)

–2.5 –1.5 –0.5

Fig. 1.5 : Transfer characteristic of quantizer

From the transfer characteristic we can say that:

Note: Transfer characteristic of system is output vs input characteristic.

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Now let input to quantizer is x(t) = 3 sint

Fig. 1.6 : Input output of quantizer

Analog Sampler Quantizer Digital

1.2 Elementary Analog Signals

1.2.1 Unit Step Function

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Fig. 1.8 : General definition of unit step function

Now to draw u(t), here input is ‘t’, so u(t) is

Fig. 1.9 : Unit step function

1.2.2 The Unit Impulse Function

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Fig. 1.10 : Ideal dirac delta function

Fig. 1.11 : Practical and ideal unit step function

So, we can say that (t) is first derivative of u(t).

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but area of (t) was 1 i.e. t ) dt 1 then area of (at) will be

Remember: Thus delta function is an even function.

Fig.1.15 : Multiplication of signal x(t) with p (t)

From above we can see that,

Using equation (8) and (9) we can say that,

x (t ) (t ) dt = x(0) (t ) dt = x (0) (t ) dt = x(0) ...(10)

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to find the result we follow the following steps.

We have to integrate delta function to find y(t),

and for t > 0 y(t) = 1

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Find the value of integration I,

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Thus, applying 0 we get p(t) tend to (t) and

Similarly we can prove,

If n(t) denote nth derivative of delta function then,

Here xn(t) is nth derivative of x(t).

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function with nth derivative of x(t) at t = location of delta function.

Here, x(t) = e–t

1.2.3 Rectangular Function

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Similarly we can find any rectangular function easily.

1.2.4 Triangular Function

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The plot of A (t/ ) will be

Fig. 1.16 : Triangular function

The numerator is –2t + 40 and denominator is 50, so

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For triangular function also there are two pattern of question.