Classification of Signals

and Systems

Syllabus:
Introduction, Continuous Time and discrete time signals, classification of signals,
simple manipulations of discrete time signals, amplitude and phase spectra,
classification of systems, analog to digital conversion of signals.

I-----------------------------------------------;-----------------------------------------------------------------------------Contents
Page No.
1.1 Introduction

1-2

1.2 Concept of Signal and Signal Processing

1 -2

1.3 Block Diagram Representation of DSP System

1-3

1.4 Analog to Digital Conversion of Signals

1 -4

1.5 Classification of Signals

1 -5

1.6 Representation of DT Signals

1-5

1.7 Basic Sequences

1-6

1.8 Simple Manipulation of Discrete Time Signals

1 -6

1.9 Classification of Signals

1 -11

1.10

1 -28

Representation of DT Signals

1.11Basic Sequences

1-30

1.12

Simple Manipulation of Discrete Time Signals

1-36

1.13

Relationship between Unit Step and Unit Impulse

1-56

1.14 1.1

introduction :_______________________________________________

1.15
The world of science and engineering is filled with signals such as images from remote
space probes, voltages generated by the heart and brain and countless other applications.
1.16

1.1.1

What is DSP?

1.17
Digital signal processing is used in a wide variety of applications. It is hard to get exact
definition of DSP.
1.18

Let us first look at the dictionary meanings of these words:

1.19

Digital: Operating by the use of discrete signals to represent data in the form of numbers.

1.20

Signal: A variable parameter by which information is conveyed through an electronic circuit.

1.21 Processing : To perform operations on data according to programmed instructions. This leads to
a simple definition of DSP.
1.22 Definition of DSP : DSP is defined as changing or analysing information which is measured as
discrete sequences of numbers.
1.23

1.2
Concept of Signal and Signal Processing :

In a communication system, the word 'signal' is very commonly used. Therefore we must know
its exact meaning.
Mathematically, signal is described as a function of one or more independent variables.
Basically it is a physical quantity. It varies with some dependent or independent variables.
So the term signal is defined as "A physical quantity which contains some information and which
is function of one or more independent variables."

The signals can be one-dimensional or multidimensional.

1.24

When the function depends on a single variable, the signal is said to be one dimensional.

Example of one dimensional signal is speech signal whose amplitude varies with time.
1.25

One dimensional signals:

Multidimensional signals:

When the function depends on two or more variables, the signal is said to be multidimensional.
The example of a multidimensional signal is an image because it is a two dimensional signal with
horizontal and vertical co-ordinates.

1.2.1 System :

A system is defined as the entity that operates on

one or more signals to accomplish a function, to

produce new signals.

Fig. 1.2.1 demonstrates the interaction between

signals and system.

The types of input and output signals depends on the type of system being used.

1.2.2 Types of Systems :

1.
2.
3.

Signals and systems have several applications. Some of the important types of systems are as

follows :
Communication system.
4. Biomedical signal processing.
Control system.
5. Auditory system.
Remote sensing system.

____________________________________________________________________________________1.3

Block Diagram Representation of DSP System :______________________

Fig. 1.3.1 shows that the basic elements of digital signal processing system

Fig. 1.3.1

The different blocks of this system are as follows :

1.

Input signal:

It is the signal generated from some transducer or from some communication system. It
may be biomedical signal like ECG or EEG. Generally input signal is analog in nature. It is
denoted by x(t).

2.

Anti-aliasing filter:

Anti aliasing filter is basically a low pass filter. It is used for the following purposes :

(a) It removes the high frequency-noise contain in input signal.

(b) As the name indicates; it avoids aliasing effect. That means it is used to band limit the
signal.

3.

Sample and hold circuit:

As the name indicates; this block takes the samples of input signal. It keeps the voltage
level of input signal relatively constant which is the requirement of ADC.

Sometimes amplifiers are used to bring the voltage level of input signal upto the required
voltage level of ADC.

4.

Analog to digital converter (ADC):

As the name indicates; this block is used to convert analog signal into digital form. This is
required because digital signal processor accepts the signal which is digital in nature.

5.

Digital signal processor :

It processes input signal digitally. In a simple languages processing of input signal making
modifying the signal as per requirement. For this purpose DSP processors like ADSP 2100 or
TMS 320 can be used.

 6. Digital to analog converter (DAC):

The output of digital signal processor is digital in nature. But the required final output is analog
in nature. So to convert digital signal into analog signal DAC is used.
7.Reconstruction filter:
Output signal of DAC is analog, that means it is a continuous signal. But it may contain high
frequency components. Such high frequency components are unwanted. To remove these
components; reconstruction filter is used.
____________________________________________________________________________________

1.4 Advantages of Digital over Analog Signal Processing :

___________________________________________________________

1.

1.

Versatility : Digital systems can be reprogrammed for other applications (where programmable
DSP chips are used). Moreover, digital systems can be ported to different hardware.

2.

Repeatability : Digital systems can be easily duplicated. These systems do not depend upon
component tolerances and temperature.

Simplicity : It is easy to built any digital system as compared to an analog one.

3.

Accuracy : To design analog system; analog components like resistors, capacitors and inductors
are used. The tolerance of these components reduce accuracy of analog system. While in case of
DSP ; much better accuracy is obtained.

4.

Remote processing : Analog signals are difficult to store because of problems like noise and
distortion. While digital signal can be easily stored on storage media like magnetic tapes, disks
etc. Thus compared to analog signals; digital signals can be easily transposed. So remote
processing of digital signal can be done easily.

5.

Implementation of algorithms : The mathematical processing algorithms can be easily

implemented in case of digital signal processing. But such algorithms are difficult to implement
in case of analog signals.

6.

Easy upgradations : Because of the use of software; digital signal processing systems can be
easily upgraded compared to analog system.

7.

Compatibility : In case of digital systems; generally all applications needs standard hardware.
Thus operation of dsp system is mainly dependent on software. Hence universal compatibility is
possible compared to analog systems.

8.

Cheaper : In many applications; the digital systems are comparatively cheaper than analog
systems.

9.

1.5

Disadvantages of Digital Over Analog Signal Processing :__________

10.___________________________________________________________The digital signal

processing systems have many advantages. Even though there are certain
disadvantages as follows :_______________________________________________

11.

1.

System complexity : The digital signal processing system, makes use of converters like ADC
and DAC. This increases the system complexity compared to analog systems. Similarly in many
applications; the time required for this conversion is more.

2.

Bandwidth limitation : In case of DSP system; if input signal is having wide bandwidth then it
demands for high speed ADC. This is because, to avoid aliasing effect, the sampling rate should
be atleast twice the bandwidth. Thus such signals require fast digital signal processors. But
always there is a practical limitation in the speed of processors and ADC.

3.

Power consumption : A typical digital signal processing chip contains more than 4 lakh
transistors. Thus power dissipation is more in dsp systems compared to analog systems.

4.

Cost: For small applications digital signal processing systems are expensive compared to analog
systems.

12.__________________________________________________________________________________1

.6

Comparison between Digital and Analog Signal Processing :________

13.

Table 1.6.1 shows comparison between digital and analog signal processing
14.

15.

Table 1.6.1: Comparison between digital and analog signal processing

1.7 Applications of
DSP :

16._____________________________________________________________

1.8 Analog to Digital Conversion of Signals :

_______________________________________________________________

An incoming signal may come from a digital or analog source. If it is coming from a digital
source then it is in the right form for processing digitally.
But input signal can be analog in nature, (e.g. speech signal or video signal).
Then it has to be converted into digital form before it can be processed by a digital system. This
type of conversion is performed using analog to digital converters. (A/ D)
17.

The simple block diagram of ADC is shown in Fig. 1.8.1.

18.

19.
20. 1.8.1

Fig. 1.8.1: Analog to digital conversion

Sampling:

In order to represent the original message signal "faithfully" (without loss of information), it is
necessary to take as many samples of the original signal as possible.
Higher the number of samples, closer is the representation. The number of samples depends on the
"sampling rate" and the maximum frequency of the signal to be sampled.
Sampling theorem was introduced to the communication theory in 1949 by Shannon. Therefore this
theorem is also called as "Shannon's sampling theorem".
The statement of sampling theorem in time domain, for the bandlimited signals of finite energy is as
follows :

21. Statement:
22. (i)

If a finite energy signal x(t) contains no frequencies higher than "W" Hz (i.e. it is a bandlimited

signal) then it is completely determined by specifying its values at the instants of time which are spaced
(1/2W) seconds apart. ii)

If a finite energy signal x (t) contains no frequency components higher than

"W" Hz then it may be completely recovered from its samples which are spaced (1/2W) seconds apart.

23.

24. 1.8.2

Quantization :

Quantization is a process of approximation or rounding off. The sampled signal is applied to the
quantizer block.
Quantizer converts the sampled signal into an approximate quantized signal which consists of only a
finite number of predecided voltage levels.
Each sampled value at the input of the quantizer is approximated or rounded off to the nearest
standard predecide voltage level.
25.

These standard levels are known as the "quantization

levels."
The quantization process takes place as follows :
26.

The input signal x (t) is assumed to have a peak to peak swing of VL to VH volts. This entire
voltage range has been divided into "Q" equal intervals each of size "S".

At the center of these steps, the quantization levels q0, qv ... q7 are
located.

xq (t) represents the quantized version of x (t). We obtain x q (t) at the output of the quantizer.
When x (t) is in the range A0, then corresponding to each value of x (t), the quantizer output will
be equal to "q0". Similarly for all the values of x (t) in the range A l5 the quantizer output is
constant equal to "qj". Thus in each range from A0 to A7 , the signal x (t) is rounded off to the
nearest quantization level and the quantized signal is produced.

The quantized signal x (t) is thus an approximation of x (t). The difference between them is called
as quantization error or quantization noise. This error should be as small as possible. To minimize
the quantization error we need to reduce the step size "S" by increasing the number of
quantization levels Q.
27. Why is quantization required ?

If we do not use the quantizer block, then we will have to convert each and every sampled value
into a unique digital word.
This will need a large number of bits per word (N). This will increase the bit rate and hence the
bandwidth requirement of the channel.
To avoid this, if we use a quantizer with only 256 quantization levels then all the sampled values
will be finally approximated into only 256 distinct voltage levels.

So we need only 8 bits per word to represent each quantized sampled value.
Thus the number of bits per word can be reduced. This will eventually reduce the bit rate and
bandwidth requirement.

Quantization error or quantization noise e :

The difference between the instantaneous values of the quantized signal and input is called as
quantization error or quantization noise.

e =
xq(t)-x(t)
...(1.8.2)
The quantization error is shown by shaded portions of the waveform in Fig. 1.8.2.
The maximum value of quantization error is S/2 where S is step size. Therefore to reduce the
quantization error we have to reduce the step size by increasing the number of quantization levels
i.e. Q.

The mean square value of the quantization is given by,

Mean square value ofquantization error = TX

S2
...(1.8.3)

The relation between the number of quantization levels Q and the number of bits per word (N)
in
the transmitted signal can be found as follows :
Because each quantized level is to be converted into a unique N bit digital word, assuming a
binary coded output signal.

The number of quantization levels Q = Number of combinations of bits/word.

2N

I
...(1.8.4)

/i

Q=

Thus if N = 4 i.e. 4 bits per word then the number of quantization levels will be 2 i.e. 16.

1.8.3 Encoding:

Our final aim is to convert the signal into the binary form. So after quantizing, the signal is
applied to encoder block.
Encoder assigns unique binary number to each quantization level. That means each quantization
level is converted into the binary digits.
The bits in the binary digit are denoted by 'b'. The number of bits in the binary digit depends on
the number of levels (L). This relation is 2 > L
Thus

b > log2L

Solved Problems on Sampling Theorem :

Ex. 1.8.1 :

Two signals x^t) = cos 20 nt and x2 (t) = cos 100 ret are sampled with sampling

frequency 40 Hz. Obtain the associated discrete time signals x,(n) and x2(n) and comment on the result.
Soln. :

Given signal is,

= cos 20nt

xt(t)
...(1)

 Compare Equation (1) with standard equation

x^t)

= cos2rcF1t

...(2)

.-.

2TCF, = 20 re ^> Ft = 10 Hz.

Now discrete signal Xj(n) is obtained by replacing 't' in Equation (2) by J;

here fs = Sampling frequency = 40 Hz. Thus Equation (2) becomes,

Xj(n) = cos 2TCFJ T

x^n) = cos 2 Tt "TT"

Xj(n) = cos 2 Tt (7J n

(ii)

The given signal is,

= coslOOJtt

x2(t)
...(3)

= cos2rcF2t

x2(t)
...(4)

Compare it with,

.-.

2nF2 = 100 7i => F2 = 50 Hz

Now discrete time signal x2(n) is obtained by putting t = T. Thus Equation (4) becomes,

x2(n) = cos 2 Tt -~7fr

x2(n) = cos 2 Tt ( 7 ) n

x2(n)

= cos2n( 1 + TJ n = cos (2rcn + 2TC 7 nJ

(5)1

Now we have cos (2rcn + 0) = cos 9. Thus Equation (5) becomes,

x2(n) = cos 2TC 7 n

Comment:

Given sampling frequency, fs = 40 Hz. Thus the frequency contained in signal should be less than I

fs or equal to ~x; that means < 20 Hz. But this is not the case in this example. So aliasing
takes place. Here I

both the sequences Xj(n) and x2(n) are equal; due to aliasing effect.

 Ex. 1.8.2 :

For an analog signal,

xa(t) = 3 cos 50 Ttt + 10 sin 300 7it - cos 100 7it. Calculate Nyquist rate.

Soln.: The given equation can be

written as,

= 3cos(2nx25t) + 10cos(2rcx 150t)-cos(2n x50t)

xa(t)
...(1)

Now we can write,

= 3 cos (2 7T Ft t) + 10 cos (2 n F21) - cos (2 n F31)

Comparing Equations (1) and (2) wee get,

F1 = 25Hz,
F2=150Hz

xa(t)
...(2)
and

F3 = 50Hz.

Thus Fmax = 150 Hz. Now Nyquist

rate = 2 Fmax = 2 x 150 Hz

____________________________________________________________________________________

1.9 Classification of Signals :

_______________________________________________________________

There are various types of signals. Every signal is having its own characteristic. The
processing of signal mainly depends on the characteristics of that particular signal. So classification of
signal is necessary. Broadly the signals are classified as follows :

Continuous and discrete time signals

Continuous valued and discrete valued signals
Periodic and non-periodic signals
Even and odd signals
Energy and power signals
Deterministic and random signals
Multichannel and multidimensional signals.
1.9.1

Continuous and Discrete Time

Signals : Continuous signal:

A signal of continuous amplitude or time is known as continuous signal or analog signal.
This signal is having some value at every instant of time.
Examples:
Sinewave, cosinewave, triangular wave etc. Similarly certain electrical signals derived in
proportion with physical quantity such as temperature, pressure, sound etc. are also examples of
continuous signal. Some of the continuous signals are as shown in Fig. 1.9.1.

Mathematical expression :

Mathematically a continuous signal (eg. sinewave) can be expressed as,

+ 6) Here

x ( t ) = Asin(ci)t

A = Amplitude of signal

0) = Angular frequency = 2 rcf

9 = Phase shift

Characteristics:

For every fix value of t, x (t) is periodic in nature.

If the frequency ( ~ J is increased then the rate of oscillation also changes.

Discrete time signal:

In this case the value of signal is specified only at specific time. So the signal represented at
"discrete interval of time" is called as discrete time signal.
The discrete time signal is generated from continuous time signal by using the sampling
operation. This process is shown in Fig. 1.9.2.
Consider a continuous analog signal as shown in Fig. 1.9.2(a). This signal is continuous in nature
from - to + .
The sampling pulses are shown in Fig. 1.9.2(b). These are train of pulses. Here the samples are
taken at Ts, 2 Ts, 3 Ts... and Ts is the sampling time.

Fig. 1.9.2(c) shows discrete time signal. Observe that this signal takes the value, only where the
sampling pulse is present. In between the two sampling pulses the signal is absent. So this is
called as discrete time signal.
In Fig. 1.9.2(a), on X-axis time (t) is plotted. On Y-axis the amplitude is plotted. So continuous
time signal is represented by x (t). Observe Fig. 1.9.2(c). On X-axis index n is plotted. Here n is
the number of corresponding sample. So discrete time signal is denoted by x ( n).
For signal in Fig. 1.9.2(a), the expression is,
x ( t ) = A cos tot and for
signal shown in Fig. 1.9.2(c), the expression is,
x (n) = A cos con

Characteristics:

Discrete time sinusoidal signals are identical when their frequencies are separated by integer
multiple of 2 n.
If the frequency of discrete time sinusoidal is a rational number, then such signal is periodic in
nature.
For the discrete time sinusoidal, the highest oscillation is obtained when angular frequency to =
n.

1.9.2 Continuous Valued or Discrete Valued

Signals : Continuous valued signal:

If the variation in the amplitude of signal is continuous then, it is called as continuous valued
signal. Such signal may be continuous or discrete in nature.

Such signals are as shown in Figs. 1.9.2(a) and (c).

Discrete valued signal:

If the variation in the amplitude of signal is not continuous; but the signal has certain discrete
amplitude levels then such signal is called as discrete valued signal.
Such signal may be again continuous or discrete in nature as shown in Figs. 1.9.3(a) and 1.9.3(b).

(b) Discrete amplitude signal discrete

in nature Fig. 1.9.3

As shown in Fig. 1.9.3(a), the signal is defined at all instants of time.

So it is continuous signal. But it takes only certain discrete amplitude levels.
The amplitude is not continuously changing with time. So it is discrete amplitude signal
continuous in nature.
As shown in Fig. 1.9.3(b), the signal is defined only at discrete intervals of time.
So it is discrete signal. And this signal takes only certain discrete amplitude levels.
So it is discrete amplitude signal discrete in nature.

1.9.3 Periodic and Non-periodic Signals :

Periodic signal:

A signal which repeats itself after a fixed time period or interval is called as periodic signal.
The
periodicity of continuous time signal can be defined mathematically as,

x(t) x ( t + T0)
...(1.9.1)

This is called as condition of periodicity.. Here T0 is called as fundamental period. That

means
after this period the signal repeats itself.

For the discrete time signal, the condition of periodicity is,

x ( n + N)

x(n) =
...(1.9.2)

Here number 'N' is the period of signal. The smallest value of N for which the condition of
periodicity exists is called as fundamental period.

Periodic signals are shown in Figs. 1.9.4(a) and (b).

(b) Discrete
time periodic
signal Fig.

1.9.4

Non-periodic signal:

called as

A signal which does not repeat itself after a fixed time period or does not repeat at all is

non-periodic or aperiodic signal. Thus mathematical expression for non-periodic signal is,

..

.
(1.9.3) ...

This is

(1.9.4)
Sometimes it is said that non-periodic signal has a period T = as shown in Fig. 1.9.4(c).
exponential signal having period, T = .

Fig. 1.9.4(c): A periodic signal having period, T =

Condition for periodicity of a discrete time signal:

A discrete time sinusoidal signal is periodic only if its frequency(f 0) is rational. That means
frequency f0 should be in the form of ratio of two integers.

Proof:
For the discrete signal, the condition of periodicity is,

x(n +

N) = x(n)

...(1.9.5)

Let x(n) be the cosine wave. So it can be expressed as,

x(n) =

A cos(2rcf0 n + 6)

...(1.9.6)

Here A = Amplitude
and 0 = Phase shift

Now the equation of x(n + N) can be obtained by replacing 'n' by 'n + N' in Equation (1.9.6).

x(n + N) = A cos[2nf0 (n + N) + 0 ]

.-.
...(1.9.7)

According to condition of periodicity Equation (1.9.5); we can equate Equations (1.9.5) and
(1.9.7).

cos[27tf0 (n + N) + 0 ] = A cos(27tf0 n + 0)
A cos(2nf0 n + 2nf0 N + 0) = A cos(2rcf0 n + 0)

...(1.9.S

To satisfy this equation,

= 27tk

2rcf0N
...(1.9.9

where k is an integer

....Proved
...(1.9.10*
Here k and N both are integers. Thus discrete time (DT) signal is periodic if its frequency f 0 is
rational.

Periodicity condition for x(n) = x.,(n) + x2(n):

Here input sequence x(n) is expressed as summation of two discrete time sequences. We can
calculate the values of fj and f2 corresponding to Xj(n) and x2(n).

Let Xj(n) and x2(n) both be periodic discrete time signals (sequences).

 So according to condition of periodicity,

f, = ^

kj

and

k2

f2 = -^

N,
The resultant signal x(n) is periodic if "j^~ is ratio of two integers. The period of x(n) will be
least
common multiple of Nj and N2.
Similarly if continuous time signals is,

x(t) = x^O + x^t)

We can calculate the values of T; and T2 corresponding to Xj(t) and x2(t). Then the resultant
T

i x(t) is periodic if ~~zr is ratio of two integers. The fundamental

period of x(t) will be least
common multiple of Tl and T2.

Solved examples:

Ex. 1.9.1 :

Prove that the sinewave shown in Fig. P. 1.9.1 is a periodic signal.

Fig. P. 1.9.1

5-: n. : The sinewave shown in the Fig. P. 1.9.1 can be mathematically represented as,

= A sin 0)o t

x (t)
...(1)

Now, let us test if it satisfies the condition for periodicity i.e. if,

x(t)

= x(t + T0)

...(2)

So, let us find the expression for x (t + T0)

x(t + T0) = Asinco0(t + T0)

Asin[o)0t + co0T0]

=
.-(3)

But 0)o = 27t f0 and T0 = f. Therefore Cfl0 T0 = 2n f0 x T = 2n. Substitute this in Equation (3), to

 x (t + T0) = A sin [oo01 + 2n ]

= A

[ sin(co01) cos 2% + cos(co01) sin 2n ]

:. x ( t + T0) = A sin co01 = x (t)

...(4)

Therefore the sinewave shown in Fig. P. 1.9.1 is a periodic signal.-Ex. 1.9.2 :

Prove that the exponential signal shown in Fig. P. 1.9.2 is non-periodic.

Fig. P. 1.9.2

Soln.: The exponential signal shown in Fig. P. 1.9.2 is expressed mathematically as,

= e-at

x(t)
...(1)

Substitute t = (t + T0) to get,

,T ,
-a(t + T)
x(t + T0) = e
=e

-at -oT

ButT0 = oo

= e

=0 .-.

x(t + T0) = e ~ a t - 0 = 0 .-.

x(t) * x(t

+ T0) Hence the exponential signal shown in Fig. P. 1.9.2 is a non-periodic

signal.

Ex. 1.9.3: What is the fundamental frequency of the waveform shown in Fig. P. 1.9.3, in Hz
and rad/sec ?

 Soln.:

One cycle corresponds to 0.2 sec. Hence T0 = 0.2 sec.

1
.-. Frequency f0 = -j- = TTZ = 5 Hz

1
...Ans.

Frequency in rad/sec. = co0 = 2JI f0 = 2x 3. 14x5 = 31.4 rad/s

Ex. 1.9.4 :

...Ans.

What is the fundamental frequency of the D.T. square wave shown in Fig. P. 1.9.4.

Fig. P. 1.9.4
Soln. :

The fundamental angular frequency or simply fundamental frequency of x (n) is given by

22

" " N
When N = a positive integer indicating number of samples in one cycle.

For the given signal N = 8.

2n it
.. Q = "o" = T radians

...Ans.

Ex. 1.9.5 : State whether the following signals x(t) is periodic or not, giving reasons. If it is periodic,
find the corresponding period, x (t) = 2 cos 100 n t + 5 sin 501 Sofa.: The given signal is,

x(t)

= 2cosl00 7tt + 5sin50t

...(1)

Let

x(t)

= X j W + x-jCt)

...(2)

x, (t) = 2 cos 100 m t

and
= 5 sin 501

Her
...(3)
x2(t)
...(4)

The standard equation can be expressed as,

xl (t) = A cos ! t

...(5)

2TI 6 '
2rc~l

"

t[

T' = | = l

-<*>

Similarly comparing Equations (3) and (4) we get,

co2 = . 18 7t,

.-. 2jtf2 = 18 rc

<> - -

2 7t

'
-(6)

"t2 = 9

The resultant signal x (t) is periodic if Tf is the ratio of two integers. From Equations (5)
and (6)

-eget,

li
T2

1/6_I 9 _ 9
1/9 " 6' 1 ~ 6

It is the ratio of two integers. Thus x (t) is periodic. Now the fundamental period of x (t) is
least

;: smon multiple of Tx and T2. Thus fundamental period is T sec.

Given

x (t) =

3sin4t

...(7)

We have the standard equation,

x(t) =

sin cot

...(8)

Comparing Equations (7) and (8) we get,

'

co =

.-. 2 n f =

-'-
It is not the ratio of integer values. Thus this signal is non-periodic in nature.
x(t) = 3 + t2

Given

...(9)

We know that a continuous time signal is periodic in nature if it

satisfies the equation,
x(t) = x(t + T0)

Where T0 is the fundamental period of repetation.

...(10)
>

From Equation (9) we can write,

x ( t + T0) = 3 + (t + T0)2
i

.-. x(t + T0) = 3 + t2 + 2tT0 + To

...(11)

For any value of 'T0' Equations (9) and (11) cannot be made equal. Thus given signal is
-periodic.

Ex. 1.9.7 :

Few discrete time sequence are given below :

(i)

cos (0.01 n n )

(ii) cos (3 7t n )

(iii) sin (3 n )

Determine whether they are periodic or non periodic. If a sequence is periodic,

determine its fundamental period.

Soln.:
(i)

Given sequence is

= cos (0.017tn)

x(n)
...(1)

We have the standard equation,

= cos con

x (n)
...(2)

Comparing Equations (1) and (2) we get,

co =

0.017i
But

co = 27tf

27tf = 0.0171

"

0.017t 0.01
27t - 2

2Qn cycles per sample

f=
...(3) I

Since frequency 'f is expressed as the ratio of two integers; this sequence is periodic. Now we I
have the condition of periodicity,

Here 'N' indicates, the fundamental period.

Comparing Equations (3) and (4)

samples (ii)

Fundamental period = N = 200

Given equation is

x(n)

= cos(37tn)

...(5

Comparing with Equation (2) we get,

/.

27tf = 3 7T

f = x cycles/sample

Since 'f is ratio of two integers; the given sequence is periodic. Comparing Equations (4) and
(6) we get,

samples (iii)

Fundamental period = N = 2

Given sequence is,

sin 3 n

x (n) =
...(7)

Comparing with Equation (2) we get,

(0
= 3 .-. 2n f =
3

"

f =

Jn

Here 2 7t is not an integer. That means ' f' cannot be expressed as the ratio of two integers. Thus
the given sequence is non-periodic.

1.9.4 Even and Odd Signals :

Even signals:

An even signal is also called as symmetrical signal. A continuous time (C.T.) signal x (t) is said
to be symmetrical or even if it satisfies the following condition :

Cor*

-t)

...forC.T. signal.

Here x ( - t) indicates that the signal is present for negative time period. That means x ( -1) is
the signal which is reflected about vertical (Y) axis. So even signals are symmetric about vertical
axis or at t = 0.

Odd (Antisymmetric) signal:

A continuous time (C.T.) signal x ( t ) is said to be antisymmetric or odd if it satisfies the

following condition
Conditio- ...for C.T.

signal

Here x (-1) indicates that the signal is present for negative time period. While - x (t)

indicates r i.: the amplitude of signal negative. Thus antisymmetric signal is not symmetric about vertical
axis.

Energy and Power Signals :

In Equation (1.9.11), it is expected that N 1.

The power signal is as shown in Fig. 1.9.5(a).

Energy signal:

The total normalized energy for a "real" signal x (t) is given by,

CO

E= f
...(1.9.12)

x (t)dt

CO

But if the signal x (t) is complex then Equation (1.9.12) is modified as,

CO

E= f|
...(1.9.13)

x( t ) | dt

oo

The energy signal is as shown in Fig. 1.9.5(b).

Note:

Ex. 1.9.8 :

What is the total energy of the rectangular pulse shown in Fig. P. 1.9.8 ?

Fig. P. 1.9.9

Sotn.: Given signal is periodic. So consider one cycle from 0 to T.

1.9.5 Deterministic and Random Signals :

Deterministic Signal:

A signal which can be described by a mathematical expression, loop-up table or some

well
defined rule is called as the deterministic signal.

Examples:

Sine wave, cosine wave, square wave etc.

Fig. 1.9.6(a) shows C.T. sine wave signal, which is deterministic signal. Because it can be
represented mathematically as,
x ( t ) = Asin(2rcft) Here A =
Amplitude of signal f = Frequency of
signal.

Similarly for D.T. wave we have,

x (n) = A sin (2jtfn)

Random signal:

A signal which cannot be described by any mathematical expression is called as random

signal
Due to this it is not possible to predict about the amplitude of such signals at a given instant of I
time.

Example:

A good example of random signal is "noise" in the communication system. Such a

random signal I
is as shown in Fig. 1.9.6(b).

1.9.6 Multichannel and Multidimensional Signals :

Multichannel signals :

As the name indicates, multichannel signals are generated by multiple sources or multiple I
sensors.
The resultant signal is the vector sum of signals from all channels.

Example:

A common example of multichannel signal is ECG waveform. To generate ECG waveform;

different leads are connected to the body of a patient. Each lead is acting as individual channel. Since
there are 'n' number of leads; the final ECG waveform is a result of multichannel signal.

Mathematically final wave is expressed as,

Multidimensional signals:

If a signal is a function of single independent variable, the signal is called as one-dimensional

signal. On the other hand, if the signal is a function of multi (many) independent variables then it is
called as multidimensional signal.

A good example of multidimensional signal is the picture displayed on the TV screen. To locate a
pixel (a point) on the TV screen two co-ordinates namely X and Y are required. Similarly this point is
a function of time also. So to display a pixel, minimum three dimensions are required; namely x, y and
t. Thus this is multidimensional signal. Mathematically it can be written as, P (
(x, y, t). -"parison of Multichannel and Multidimensional

Signal:

1.10 Representation of DT Signals :

The discrete time sequence is denoted by x (n ). Consider such a discrete time signal as

shown in Fig. 1.10.1.

On the X-axis index 'n' is plotted. Here 'n' is corresponding number of the sample. In the
given I diagram value of n varies from - 3 to + 3. On the Y-axis, amplitude of signal is plotted. The signal
a having some amplitude at each value of n. Now the different methods used to represent the signal x (n)
are as follows :

1. Functional Representation
3. Sequence Representation

2. Tabular Representation

1.
Functional Representation : For the signal shown in Fig. 1.10.1, the functional
representation of signal is as follows :

3.

Here the amplitude of signal is written below the corresponding value of n.

Sequence Representation : The sequence representation of given signal is as follows :

x ( n ) = {1,2,-1, 1,2,0,1}

Here all the amplitudes of signal are written sequentially starting from
the leftmost amplitude.
ARROW ALWAYS INDICATES THE AMPLITUDE OF SAMPLE AT n = 0. If arrow is no:
shown in the sequence then by default it is at first position.
e.g.:
x ( n ) = {1,2,3,4,5}

Here arrow is not shown; so by default it is at first sample.

That means we can write,

x (n ) = {1 , 2, 3,4, 5}

Number of samples contained in the given sequence is called as the length of sample,
To adjust the length of sequence we can add any number of zeros at the beginning or at the end
sequence. This is called as ZERO PADDING.

e.g.:
I f x ( n ) = {1 ,2 ,0 ,1 ,2 }

Then we can write,

x ( n ) = {0,0,1,2,0,1,2}

or x ( n ) = {1, 2, 0 , 1, 2, 0, 0}

t T

Remember that the position of pointer (arrow) does not change.

Ex. 1.10.1 : Represent the following signals graphically :

(i)

x(n) = {1, 2, 0 , - 1 , 1 }

(ii)

x(n) = { 0 , 0 , - 1 , 2 , 3 }

(iii) x(n) = {0,1,-1,1,-1)}

loin. : These signals are as shown in Fig. P. 1.10.1(a), (b) and (c) respectively.

____________________________________________________________________________________1

.11

Basic Sequences :____________________________________________

In the analysis of communication systems, standard test signals play a vital role. Such
signals are used to check the performance of a system. Applying such signals at the system; the output is
checked. Now depending on the input-output characteristics of that particular system; study of different
properties of a system can be done. Some standard test signals are as follows :

Delta or unit impulse function.

Unit step signal

Unit Ramp signal

Exponential signal

Sinusoidal signal

1.11.1 Delta or Unit Impulse Function :

for all

A discrete time unit impulse function is denoted by 8 ( n ). Its amplitude is 1 at n = 0 and

other values of n; its amplitude is zero.

The graphical representation of delta function is as shown in Fig. 1.11.1(a).

A continuous time delta function is denoted by 8 (t).

Mathematically it is expressed as,

Jlfort = 0
B" N
5(t) = lOfort^O
It is as shown in Fig. 1.11.1(b).

Fig. 1.11.1(b): Unit impulse function 8 (t)

1.11.2 Unit Step Signal:

A discrete time unit step signal is denoted by u(n). Its value is unity (1) for all positive values of
n. That means its value is one for n > 0. While for other values of n; its value is zero.

flforn>0

l0forn<0

'

u(n)

 In the form of sequence it can be written as,

.........................................u ( n ) = {1,1,1,1,

 Here arrow is absent; so by default it is at first position.

Graphically, it is represented as shown in Fig. 1.11.2(a).
A continuous time unit step is denoted by u( t).
Mathematically it can be expressed as,

, ,
J l. f o r t > 0 U(t)
10fort<0

It is as shown in Fig. 1.11.2(b).

"3 Unit Ramp Signal:

A discrete time unit ramp signal is denoted by ur ( n ). Its value increases linearly with sample
number n. Mathematically it is defined as,

~t>\
f n for n > 0

r < n ) = lOforn<0

Graphically it is represented as shown in Fig. 1.11.3(a).

A continuous time ramp signal is denoted by r( t). Mathematically it is expressed as,

ort>0 r ( t ) ~
L0fort<0

Fig. 1.11.3(b).

flf

It is as shown in
\

1.11.4 Exponential Signal:

A discrete time exponential signal is expressed as,

x(n) =

an

...(1.11.1)

Here 'a' is some real constant.

If 'a' is the complex number then x (n) is written as,

reJe
Here 9 denotes the phase.

Now depending upon value of 6 we have different

cases.
Case (i): When a > 1
Let a = 3. Thus we have,
x ( n ) = a" = 3".
Graphically such signal is represented as shown in Fig. 1.11.4(a).

Since the signal is exponentially growing; it is called as rising exponential

signal.

x(n) =
...(1.11.2)

 Case(ii) :WhenO<a<l

In this case we will get decaying exponential sequence. Let a = r

x ( n ) = a=(Dn

Graphically such signal is represented as shown in Fig. 1.11.4(b).

Fig. 1.11.4(b): Decaying exponential signal

*se i iii) : When a < -1 In this case we will get double sided
rising exponential signal. Let

a = -3 .-.

x ( n ) = (-3)n

Graphically such signal is represented as shown in Fig. 1.11.4(c).

Case (iv) : When -1 < a < 0

In this case we will get double sided decaying exponential signal.

Let

a = -T

Graphically such signal is as shown in Fig. 1.11.4(d).

Fig. 1.11.4(d): Double sided decaying exponential signal

A continuous time exponential signals for various values of a are as shown in Fig.
1.11.5.

(a)
(b)
Fig. 1.11.5 : Continuous time exponential signals

(c)
(d)
Fig. 1.11.5 : Continuous time exponential signals

1.11.5 Sinusoidal Waveform :

A discrete time sinusoidal waveform is denoted by,
x (n) = A sin con
Here,

A = Amplitude

co = Angular frequency

= 2 Ttf This waveform is as shown in Fig. 1.11.6.

Fig. 1.11.6 : Discrete time sinusoidal waveform

___________________________________________________________________________________1

.12 Simple Manipulation of Discrete Time Signals :______________________

Many times it is necessary to modify the original signal. This modification is achieved bjj
performing different operations on given discrete time signal. These operations are divided i following
types:
Time shifting operations
Time scaling operations
Amplitude scaling operations

* 12.1 Time Shifting Operations :

The different time

shifting operations are as follows :

Time delay

Folding and advance

1.

Time advance
Folding
Folding and delay

Time Delay (T.D.):

As the name indicates, delay means signals are not appearing instantly at the receiver. Almost :
- signal used in communication provides time delay at the output. Some common examples are TV
-s. telephone signals, radar signals etc.
In case of discrete time signals, the given sequence can be delayed by few samples. We know

:-_: ;.:crete time signal is denoted by x ( n ). Suppose we want to delay this sequence by 'k'
samples. It tZ be denoted by x (n - k)

and

x (n) > Original sequence

x (n - k) > Original sequence delayed by k samples.

Here k is an integer.
E.g.: Let the given signal be,

x ( n ) = {1,2,3,4,5}

This sequence is as shown in Fig. 1.12.1(a).

Suppose we want to delay this sequence by 2 samples. Then here k = 2. Thus delayed sequence
mE be denoted by x (n - 2 ).
This sequence will be shifted towards right by 2 samples. This delayed sequence is shown in
Kg. 1.12.1(b)..
From Fig. 1.12.1(b), we can write the delayed sequence as,

x ( n - k ) = x ( n - 2 ) = {0,0,1,2,3,4,5}

We know that, arrow always indicate the n = 0 sample. Since delayed sequence is shifted
towards Tginr so the first sample starts at n = 2. In the original sequence it is starting at n = 0. Thus to
delay the

eccr.;e.
Shift the diagram towards right by 'k' samples.
Shift the arrow towards left by 'k'.
To adjust the length of sequence, add zeros.

version x (n - k)

(a) x (n) = {1,2,3,4, 5 }

2.

(b) Delayed

Fig. 1.12.1

Time Advance (T.A.):

Time advance operation means the signals are present before they are generated. In a real time
operation, this is practically impossible. But if the signals are stored in the memory of computer before
starting a particular operation then it is possible to have time advanced signal.

Time advance operation is opposite to the time delay operation. Consider the same sequence x
( n ) shown in Fig. 1.12.1(a).

.-. x(n) = {1,2,3,4,5}

If we want to advance this sequence by two samples then it is denoted by x ( n + 2 ). In this case
the diagram is shifted towards left by two samples. This sequence is shown in Fig. 1.12.1(c).
Fig. 1.12.1(c): Advanced version, x ( n + k)
From Fig. 1.12.1(c), we can write advanced sequence as,
x ( n + k) = x ( n + 2 ) = {1, 2, 3, 4, 5}

3.

Folding (FD):

Folding is also called as reflection. So folding means taking the mirror image of signal. Tha
means the signal is folded about time origin n = 0. Here independent variable 'n' is replaced by - n.
Thus if x (n) represents input signal then x (- n) represents folded input signal.
Consider the same input signal x (n) as shown in Fig. 1.12.1(a)

.-. x ( n ) =

{1,2,3,4,5}

 Then folded signal x (- n ) is as shown in Fig. 1.12.1(d).

So the sequence x (- n) = {5, 4, 3, 2, 1}

Thus to obtain folded version of sequence,

Take the mirror image of sequence about n = 0.
Write the sequence in reverse order.
Keep the position of arrow as it is. In the sequence x ( n ) arrow is at 1 and in the sequence x (n) arrow is still at 1.

4.

Folding and Delay :

This operation is a combination of folding and delaying operation. Now let us take a quick
view it the modifications used for different operations.

I A)

For drawing diagrams :

To obtain delayed sequence, shift the original diagram towards right by 'k'
samples. To obtain advanced sequence, shift the original diagram towards left by 'k'
samples. To obtain the folded version, take the mirror image of the diagram at n = 0.

B)

For writing the sequence :

The delayed version of x (n) is denoted by x (n - k)

Negative sign always indicates delay operation of x (n).

The advanced version of x (n) is denoted by x (n + k)

Positive sign always indicates advanced operation of x (n).

To obtain the folded version of x (n); replace n by (- n)

Now folding and delay operation means : First fold the
sequence x (n); that means obtain x (- n ) Then delay the
folded sequence by k samples.
One major difference between x ( n ) and x ( - n ) x ( - n ) denotes the mirror image of x ( n ).
tfc know that in case of mirror image; left and right sides are reversed. Now if x ( n ) is original .:. -r-.;e
then its delayed version is denoted by x ( n - k ). The folded sequence is denoted by x ( - n ). irate it is a
mirror image the delayed version of folded signal is denoted by x (- n + k).
------------x (n ) > x (n - k)

Folding !

---------------x(-n)

> x [ - ( n - k ) ] = x ( - n + k)

 But stick to the basic concepts. Delay means shift the diagram towards right by k samples.
Consider the same original sequence x ( n ) = {1 , 2, 3, 4, 5} as shown in Fig. 1.12.1(a). T

folded sequence x ( - n) is shown in

Fig. 1.12.1(d). Suppose we want to delay this folded sequecct x ( - n ) by '2' samples then it will be
denoted by x ( - n + 2 ). This sequence is as she Fig. 1.12.1(e).

sequence x (- n + 2)

(d) Folded version x (- n) (e) Delay of folded

Fig. 1.12.1

From Fig. 1.12.1(e), the sequence x (- n + 2 ) can be written as,

x ( - n + 2 ) = {5,4,3,2,1}

5.

Folding and Advance :

The advanced version of original sequence x ( n ) is denoted by x ( n + k). The folded version a

x ( n ) is denoted by x ( - n ). Since folding means mirror image of sequence; the advanced

version a

folded sequence is denoted by x (- n - k).

---------Advance
x (n)--------------> x ( n + k)
Folding 4--------Advance
x(-n):-----------> x [ - ( n + k ) ] = x ( - n - k )
Consider the same folded sequence x ( - n ) shown in Fig. 1.12.1(d). Suppose we want s

advance this sequence by '2' samples, then the advanced version is denoted by x ( - n - 2 ). Such i
sequence is as shown in Fig. 1.12.1(f).

Remember the basic rule. Advancing the sequence means shifting the diagram towards left by 'k'
dimples. From Fig. 1.12.1(f), we can write the sequence x (- n - 2) as,

x ( - n - 2 ) = {5,4,3,2,1,0,0}

I 12.2 Time Scaling Operations :

As the name indicates, time scaling operations are related to the change in time scale. There are '': n,pes of
time scaling operations. Down scaling (Compression) Up scaling (Expansion)

Down Scaling or Compression :

Consider the same sequence x ( n ) = {1 , 2, 3, 4, 5}. It is shown in Fig. 1.12.1(a). Let this

T
n i be input to some device which produces output y (n ). Also say,
y(n)

= x ( 2 n ) Now

from given sequence x (n) we can write,

x ( 0 ) =1

x(l) =2x (2 )

= 3

x( 3) = 4

x(4) = 5

x(5) = 0...

This gives the value of x (n) for different values of n.

Vow we have,

y(n) = x ( 2 n )

.-. y(0) = x ( 0 ) = l
y(l) = x ( 2 ) = 3
y(2) = x ( 4 ) = 5
y(3) = x ( 6 ) = 0
.-. y(n) = x ( 2 n ) = {l,3,5,0,..........................}

This sequence y(n) = x ( 2 n ) i s represented as shown in Fig.

 Compare this diagram with original signal x ( n )

shown in Fig. P. 1.12.1(a). Since the scale ii compressed this
is called as down scaling operation.

2.

Up Scaling or Expansion :

Consider same input sequence x ( n ) = {1 , 2, 3, 4, 5} is applied to certain device whin

produces output y ( n ) = x ( x ) - Thus in this case,

y(0)

= x() =x(0)=l

y(l)

= x ( x J > No sample

y(2)

= x(f) =x(l)=2

y(3)

= x ( y J = x (1.5 ) > No sample

y(4)

= x(|) =x(2) = 3

y(5)

= x ( T J = x ( 2.5 ) -> No sample

y(6)

= x() =x(3) = 4

y(7)

= x ( x J = x ( 3.5 ) No sample

y(8)

= x() = x ( 4 ) = 5

Thus

y(n)

= xf|j = {1 , 0, 2, 0, 3, 0,4, 0, 5}

This sequence is shown in Fig. 1.12.2(b).

Compare this diagram with Fig. 1.12.1(a). Since the

upscaling operation.

scale is expanded, this is called as

Fig. 1.12.2(b): Upsampling

1.12.3 Amplitude Scaling Operation :

As the name indicates, in case of amplitude scaling operations, amplitude of signal is char,;:
Different amplitude scaling operations are as follows :

(Attenuation)

Upscaling (Amplification) Downscaling

Addition Multiplication

1.

Upscaling or Amplification : Consider same input signal x ( n) shown in Fig.

1.12.3(a). .-. x(n) = { 1 , 2 , 3 , 4 , 5 }

Here

Now let,

x(0) = 1,

x(l) = 2,

x(3) = 4,

x(4) = 5.

x (2) = 3,

y(n) = 2 x ( n )

This indicates multiplication of every amplitude by

2.

y(n) = 2 x ( n ) = { 2,4,6,8, 10}

T

This sequence is shown in Fig. 1.12.3(a).

2.
Downscaling or Attenuation : Downscaling
or attenuation means reducing the amplitude of
signal. Consider same sequence x (n) = { 1 , 2, 3,4,
5}
1

L e t y ( n ) = ^f

This operation indicates the division of every

amplitude by 2.

.-. y(n) = ^r1 = {0.5,1,1.5,2,2.5}

2
t

This sequence is shown in Fig. 1.12.3(b).

3.

Addition:

 Consider two sequences,

Xl(n) = {1,1, 0,1,1}

t
and x2(n) = {2,2, 0,2,2}

t
Let y (n ) = Xj (n) + x2 (n )

.-. y(n) = {3,3, 0,3,3}

Addition operation is shown in Fig.

1.12.4(a). 4.

Multiplication:

Consider

X j ( n ) = {1,1,0,1,1}

and x2(n) = {2,2,0,2,2}

Let y(n) =

x 1 ( n ) x x 2 ( n ) .-. y(n) =
{2,2,0,2,2}

Multiplication operation is as shown in Fig.

1.12.4(

Ex. 1.12.1 :
A
di
s
cr
et
e
ti
m
e
si
g
n
al
is
gi
v
e
n
b
y,
x(
n)

=
{1
,1
,
1,
1,
2
}

Sketch the following signals :

(a) x ( n - 2 )
(b) x ( n + 1 )
(c) x ( 3 - n )

(d) x ( n ) u

(n-1)

(e) x ( n - 1 )

5(n-1)

(f) Even

samples of x ( n )

(g) Odd

samples of x ( n )

Soln.: The given sequence is,

x(n) = {1, 1,1,1,2}

We know that arrow indicates the position of

sample at n = 0. Thus the value of x ( n ) for different
values of 'n' are as follows :
x(-l)=l, x ( 0 ) = l , x(l)=l,
x ( 2 ) = 1 and x ( 3 ) = 2
This sequence is shown in Fig. P. 1.12.1(a).

(a)

x(n-2):

Here negative sign indicates delay operation. Delay :

Deration means shift the diagram towards right. So we have to
shift given sequence x ( n ) towards right by 2 samples. This
sequence is shown in Fig. P. 1.12.1(b).

ib) x(n + 1):

Here positive sign indicates advance operation. Advance

: Deration means shift the diagram towards left. So we have to
ifaift given sequence x ( n ) towards left by 1 sample. This
*xjuence is shown in Fig. P. 1.12.1(c).
:

x(3-n):

This sequence can be written a s x ( - n + 3).

Here ' - n' => folding operation.

Remember that in case of folded signal, positive sign
=> delay operation. Delay means shift the diagram
towards right.
So fold the sequence x ( n ) to obtain x ( - n ) and then
shift this diagram towards right by 3 samples.
This sequence is shown in Fig. P. 1.12.1(d).

:x(n)u(n-1):

 This operation indicates multiplication of x ( n ) and

u (n - 1 ).

Fig. P. 1.12.1(e): Multiplication of x (n) and u (n -1)

x(n) is shown in Fig. P. 1.12.1(a).

u (n) means unit step and u (n - 1 )=> unit step delayed by 1 sample.

Draw u ( n - 1) and then take the multiplication, x (n ) u ( n - 1 ). The multiplication takes

place sample to sample.

This operation is shown in Fig. P. 1.12.1 (e). (e)

x(n-1)8(n-1):

This operation indicates multiplication of x (n - 1) and S (n - 1).

x ( n - 1) => delay of x (n ) by 1 sample.

8 (n) is unit impulse and 8 (n - 1) is delayed unit impulse.

The multiplication takes place sample to sample.

This operation is shown in Fig. P. 1.12.1(f).

Fig. P. 1.12.1(f): Multiplication of x (n -1) and 8 (n -1)

Even samples of x ( n ):
I means we have to find out x (2n).
Given sequence is, x ( n ) = { 1, 1 , 1, 1, 2 }

ex(-l)=l, x ( 0 ) = l , x ( l ) = l , x ( 2 ) = l , x(3) = 2

r
At

want

x
n

2n
=

).
-1

Putting

different

x(2n)

At n = 0,

x (2n)

= x (0) = 1

At n = 1, x ( 2 x l ) = x ( 2 )

At n = 2, x ( 2 x 2 ) = x ( 4 )

After this every value of x ( 2n ) will be zero. This is

<r :: - : -,-e given sequence is upto x ( 3 ).

=1
=0

values

of
=

we

x(-2)=

get,
0

2n) = { 0 , 1 , 1 , 0 , 0 . . . . }

T
This sequence is shown in Fig. P. 1.12.1(g).
Fig. P. 1.12.1(g): Even samples of x (n )

(g)

Odd samples of x ( n):

It means we have to find out x ( 2n + 1).

We want x ( 2n + 1). Putting different values of n we get,

at

n = -1,

x(2n+l)=x(-

2+l)=x(-l)=l
at n=0,

x(2n+l)=x(0+l)=x(l)=l

at

n k 1, x ( 2 n + l ) = x ( 2 + l )

at

n = 2, x ( 2 n + l ) = x ( 4 + l ) = x ( 5 ) = 0

=x(3)

=2

After this every value of x ( 2n + 1) will be zero.

.-. x ( 2 n + l ) = {1, 1 , 2 , 0 . . . }

This sequence is shown in Fig. P. 1.12.1(h).

Ex. 1.12.2 : Sketch a discrete time signal x (n) = 2

obtain : (i)

y,(n)

= 2 x( n) + 5(n) (ii)

y2 (n)

for - 2 < n < 2 and

= x (n) u (2 - n)

Soln. :The given discrete time signal is,

x(n) = 2""

for-2<n< 2

That means the range of 'n' is from -2 to + 2. Putting these values of 'n' in the

equation of x (n) we get,

-2

"4

at

n=-2,

x(n) = 2

at

n = -1,

x(n) = 2~" = 2+

=2

at

n = 0,

x(n) = 2~" = 2 = 1

-n

-1_ 1

=2 _ ^

at

n = 1,

x (n) = 2

at

n = 2,

x(n) = 2_" = 2~'

=|
Fig. P. 1.12.2(a):
Discrete time sigj
x(n) = 2_n

Thus the sermenr.e ran he

This sequence is shown in Fig. P. 1.12.2(a).

lil

yi

(n) = 2x(n) + 8(n):

Here 8 (n) is unit impulse. It is expressed

as,

5(n) = {1}
T

That means at n = 0, 8 (n) = 1; while for all

her values of n, 8 (n) = 0. Now we have to
:a.culate,

y (n) = 2x(n) + 8(n)

2x (n) means, multiply each sample of x

(n) 1 To obtain yi (n) add 2x (n) and 8 (n). The

: ?n takes place sample to sample.

This cerition is shown in Fig. P. 1.12.2(b).

(')

y2 (n) = x (n) u (2 - n):

We know that u (n) is the unit step. It is as shown in Fig. P. 1.12.2(c). We want u (2 - n)

that means u ( - n + 2). Now u (- n) indicates folded unit step as shown in Fig. P. 1.12.2(d). While u (- n
+ 2). Indicates that we have to delay u (- n) by two samples. That means shift u (- n) towards right by
two samples. This is shown in Fig. P. 1.12.2(e). x (n> u ( - n + 2) indicates multiplication of x (n) and u
(- n + 2). This operation is shown in Fig. P. 1.12.2(f).

Fig. P. 1.12.2(f): y2 (n) = x (n) u (- n + 2)

. *X3 : Generate a pulse signal (one sided) starting from n = 0 of duration

N using : (i)

Unit sample response, (ii)

Unit step sequences.

Soln.:

(i)

We know that unit sample response is,

5(n) = 1

for n= 0

= 0 otherwise

Let the signal to be generated be x(n). The duration of this signal is from n = 0 to n = N and
the amplitude of every sample is 1. That means the required sequence is,

x(n) = {1,1,1,1,.... 1}

n=0

T
n=N-l

Thus it can be expressed in terms of unit sample response as,

(ii) Unit step starts from n = 0 and every sample is having magnitude equals to ltill n = . But
w want the sequence having samples of magnitude 1 till n = N. This is obtained as shown i Fig.
P. 1.12.3.

Ex. 1.12.4 : A discrete time signal is as shown in Fig. P. 1.12.4(a).

Sketch the following :
(i)

x(n-3)

(ii)

x(3-n)

(iii) x (2n) (iv)

x(n)u(3-n)

Fig. P. 1.12.4(a)

The given sequence can be expressed in sequential form as :

Here x (n - 3) indicates delay operation. Delay means shifting the diagram towards right.
Thus I (n - 3) is obtained by shifting x(n) towards right by '3' positions. It is shown in Fig. P.
1.12.4(b).

Observe that every sample is shifted towards right by three positions.

It can be also written as. x (- n + 3). Brace x (- n) indicates folding operation. Thus x (- n +

3) indicates delay of folded signal, sy 3 positions. This operation is shown in Fig. P. 1.12.4(c).

(ii)

x(2n)

This signal is calculated by putting different values of n.

The range of 'n' in x (n) is n = - 3 to + 3. The values of sequence x (2n) are as

follows :
For

n = -2

=>x(-4) = 0

For n = -1

=> x(-2)= j

For
For
For

n = 0 => x(0) = 1
n = 1
=> x(2)
n = 2
=> x (4)

= 1
= 0

The plot of sequence x (2n) is shown in Fig. P.

1.12.4(d).

(iv)

Fig. P. 1.12.4(d)

x (n) x(3 - n)

This signal is obtained by taking multiplication of x(n) and x(3 - n)

1.12.4(e).

as shown in Fig. P.

______________________________________________________________________________1.

13 Relationship between Unit Step and Unit Impulse :___________________I

Unit step u (t) and unit impulse 8 (t) of CT can be related as follows :
Mathematically

8 (t) is derivative of u (t) or

u (t) is integral of 8 (t)

Unit step is not differentiable because there is discontinuity at t = 0 as shown in Fig. 1.13.1(a).
Hence u (t) has limiting case of uA (t) as shown in Fig. 1.13.1(b).

Fig. 1.13.1

We know that uA (t) is continuous function and hence it is differentiable. Derivative of uA (t) as I
shown in Fig. 1.13.1(c).
As A - 0 Fig. 1.13.1(c) becomes unit impulse.

lim d [uA (t) ]

->0~dT"

"

5t

du(t) .'. 8 (t) =

same can be represented as

Or

Importance of impulse:

To convert CT-signal into DT-signal samples of CT signals are required to take at regula interval of
time.
For sampling of CT signal a train of unit impulses is required.

1.13.1 Precedence Rule for Time Shifting and Time Scaling :

Let x (t) and y (t) be two continuous time signals with y (t) derived from x (t) through i
combination of time shifting and time scaling operations.

Let the relation between x (t) and y (t) be given mathematically as follows :

x(at-b)
_

y(t) =
...(1

Let the relation between x (t) and y (t) satisfy the following two
conditions :
Condition -1
y (0) = x (- b)
and
Condition-2 y (b/a) = x
(0)

In order to obtain y (t) correctly from x (t), it is necessary to perform the time shifting and
time scaling operations in the correct order.

The proper order is decided on the basis of the fact that in the scaling operation "t" is
replaced by "at" and in the time shifting operation "t" is replaced by (t - b). So the time shifting
operation is carried out on x (t) to get an intermediate signal y' (t) as,

y'(t) =
x(t-b)
...(1.13.2)

Then the time scaling operation is performed on y' (t) by replacing t by "at". The result is,
I
desired signal y (t).

y(t)
= y'(t)|t=at
.-.
y(t) = x(at-b)
...(1.13.3)

:: -ammatic representation of precedence rule :

The precedence rule can be represented diagrammatically as shown in Fig.

1.13.2.
inclusion:

Fig. 1.13.2 : Precedence rule

.' -

. <- * the precedence ruie is not foilowed ?

If the rule is not followed, then we do not obtain the desired signal y (t). This is
demonstrated as

First perform the time scaling operation on x (t) to get y' (t)
y'(0 = x(at) Tben perform the time
shifting operation on y' (t) by replacing t by (t - b).

.-. y(t) = x [ a ( t - b ) ] = x [ a t a b ] "* 'e have not obtained the desired signal y (t) = x (at - ab). T:
understand the precedence rule, solve the following example :

:"

If x (t) = rect(t/3) then obtain y (t) = x (2t - 3) first by following the precedence rule

and then by violating the rule.

. .

J.

Draw x
(t): is
given by,

 So it is a rectangular signal of width 3 time units and an

amplitude equal to 1. It is plotted as shown in Fig. P. 1.13.1(a).

Part I: Solution by following the precedence rule :

Step 2 : Delay x (t) by 3 units :

As shown in Fig. P. 1.13.1(b), x (t) is delayed by 3 time units to obtain the intermediate I signal y'
(t) = x (t - 3).

Step 3 : Apply time scaling to y' (t):

Now apply time scaling to y' (t) with a = 2 to get y (t) = x (2t - 3) as shown in Fig. P. 1.13.1(c).
Note that we have obtained the correct signal.

Fig. P. 1.13.1: The proper order of operations

Part II: Solution by violating the precedence rule :

Step 4 : Draw the original signal:

Refer Fig. P. 1.13.1(d).

Step 5 : Time-scaling :
Refer Fig. P. 1.13.1(e). The time scaled signal is y' (t) = x (2t).
Step 6 : Time-shifting :
Refer Fig. P. 1.13.1(f). The intermediate signal y'(t) has been delayed by 3 time units. I obtain y
(t).
But y (t) = 2y' (t) = x [ 2 (t - 3) ] = x [ 2t - 6 ] which is not the desired signal.

Questions

Define the term signal and classify it. With the help of block

diagram explain the working of DSP System. Explain how analog to

digital conversion of signal is done. With examples, explain the
following signals : (a) Continuous and discrete (b)
valued and discrete valued (c)

Continuous

Even and odd. Compare energy

and power signals. Compare multidimensional and multichannel

signals. Explain the concept of sampling.

With the help of neat diagram, explain the quantization process. Explain
the different methods of representing D.T. signals. Z. ' I
different standard signals used in DSP.

What are the

o_____
\T^/

Fourier Analysis of
^*^^H|
Periodic and Aperiodic Continuous 1
Time Signalsand Systems
|

syllabus:
Introduction, trigonometric Fourier series, complex or exponential form of
Fourier series, Parsevals identify for Fourier series, Power spectrum of a periodic
function. Fourier transform and its properties, Fourier transforms of some important
signals, Fourier transforms of power and energy signals.

Contents
Page No.

2.1

Amplitude

and Phase Spectra (Line Spectra)

2-2

2.2

CT Fourier

Series

2-5

2,3

Transform

CT. Fourier
2-27

____________________________________________________________________________________2

.1

Amplitude and Phase Spectra (Line Spectra):____________________

All the signals till now were drawn with respect to time. That means time "t" was considered as a
variable. The representation of signal with respect to time is called as its time domain
representation.
The time domain representation of the signal is not sufficient for its analysis. Hence we have to
use the frequency domain representation of the signal for the sake of analysis.
In the frequency domain representation, the variable plotted on the X-axis is frequency "f' rathe:
than "t".
The signal represented in the frequency domain is called as the line spectrum. The line spectrum
consists of two graphs namely,

Amplitude spectrum : A graph of amplitude versus frequency.

Phase spectrum : A graph of phase versus frequency.

The signal x(t) and its line spectrum are shown in Fig. 2.1.1.

The graph of instantaneous signal voltage versus time is called as the time domain representation]
It is as shown in Fig. 2.1.1(a). The time domain representation gives us the follow* information:
Shape of the signal
Its frequency
Type of the signal (periodic or nonperiodic).
One cycle period.

But we can not know anything about what frequency components are present and in proportion
they have been mixed in order to obtain the particular shape of the signal.
All this information can be obtained from the line spectrum of a signal.
Line spectrum [Fig. 2.1.1(b)] is the representation of the same signal x(t), now in the frequea
domain.
It can be obtained by using either Fourier series or Fourier transform. It consists of the amplai and
phase spectrums of the signal.

The line spectrum indicates the amplitude and phase of various frequency components present in
the given signal. The line spectrum enables us to analyze and synthesize a signal.

2.1.1

How to Plot Line Spectrum ?

The line spectrum is useful in understanding the existence and amplitudes/phases of various
frequency components present in a waveform.
The important conventions about the line spectra are as follows :

1.
2.

In all the spectral drawings, the independent variable plotted on the x-axis is frequency f
in Hz and not co.
Phase angle is always measured with respect to the cosine waves. That means it is
measured with respect to the positive real axis of the phasor diagram. Hence it is
necessary to convert sinewaves to cosines using the following standard identity :

cos (tot -90)

sin cot =
...(2.1.1)

3. The amplitude is always regarded as "positive" quantity. So if negative signs appear,

they
should be absorbed in the phase change to keep amplitude positive. This is explained in
the following expression,

-A cos tot
= A cos (tot 180)
...(2.1.2)
The additional phase change of 180 converts the negative amplitude "- A" to positive
amplitude "+ A". We can choose either + 180 or - 180, as the effect is going to be the same.

2.1.1:

Sketch the line spectrum of

the following signal:

m (t) = 3 - 5 cos (40 n t - 30) + 4 sin 120 n t
Scin.:

-.

In the given signal, the first term represents a dc term which has a zero frequency. The other two
ens can be written as follows :

First term :
3 = 3 cos 2m 01
as f = 0
Second term : - 5 cos (40 rc t - 30 ) = 5 cos ( 2 n 201 - 30 + 180) = 5 cos ( 2 n 201 + 150 )
Thus the negative amplitude has been made positive by adding a phase angle of 180.
Third term : 4 sin 120 n t = 4 sin 2 n 601 = 4 cos ( 2 n 601 - 90 ).
Thus the sine term has been converted to the cosine term by adding a phase shift of - 90.
The amplitudes, frequencies and phase angles of the three terms are listed in the Table P. 2.1.1.

Table P. 2.1.1

spectrum

(a) Amplitude spectrum(b) Phase

Fig. P. 2.1.1

2.1.2 Double Sided Line Spectrum :

The line spectra drawn in section 2.1 are called as "one sided" or "positive frequen: spectra.
But the other spectral representation proves to be more valuable even though it imn "negative
frequency".
It is called as the double sided line spectrum. The double sided line spectra can be obtains: i
the single sided spectrum as shown in Fig. 2.1.2.

Conclusions:

The conclusions drawn from Fig. 2.1.2 are as follows :

1.Looking at Fig. 2.1.2(a) i.e. the amplitude spectrum, we conclude that in the single a
spectrum there is only one frequency component present at f = f0 with an amplitude A.
Whereas in double sided line spectra, two frequency components f0 and - f0 are present i
amplitude (A/2) but no change in polarity.

2.The single sided phase spectrum contains only one component at f0 with phase <|>. But tkJ
sided spectrum contains two components at f0 and - f0 with phases equal to <J> anij
respectively. Thus phase shift remains unchanged but they have opposite signs.

3.
From Fig. 2.1.2(a) it is clear that the double sided amplitude spectrum has an even
symmeo^
Fig. 2.1.2(b) shows that the double sided phase spectrum has an odd symmetry.

The double sided line spectrum representation is very useful in mathematical anal negative
frequency components present in the double sided spectrums are not practically present.

2.2.2 Polar Fourier Series :

The polar fourier series is derived from the trigonometric fourier series by combining the sua
and cosine terms of same frequency. The polar fourier series representation of x( t) is as follows :

E spectrum :

The line spectrum of x( t) can be plotted using Equation (2.2.8). A line spectrum of x( t) with
arbitrary values of amplitudes and phases is shown in Fig. 2.2.1.

Line spectrum of x( t)

a i Time domain representation of x( t)

Fig. 2.2.1: Line spectrum using polar fourier series

-.5 seen from Fig. 2.2.1(b) the frequency spectrum of a continuous signal is discrete in nature. The
frequency components f0, 2 f0, 3 f0.... etc. are called as the "spectral components".
The adjacent spectral components are spaced by "f0" from each other. As the spectrum is nsisting
of vertical lines, (Cv C2, ....) this spectrum is called as the line spectrum.

2-1.3 Exponential Fourier Series [or Complex Exponential Fourier Series]:

The sine and cosine terms can be expressed in terms of the exponential terms using Euler's e
juations as follows :

(b)

Substituting the sine and cosine functions in terms of exponential function in the expression for
the quadrature fourier series, Equation (2.2.1), we can obtain another type of fourier series called
the exponential fourier series.

A periodic signal x (t) is expressed in the exponential fourier series form as follows

Exponential fourier series:

...
(2.2.11)

...
(2.2.12)
As seen from the Equation (2.2.11), this fourier series consists of exponential terms. Therefore it I

2.2.4 Parseval's Identity for Fourier Series (Power Spectrum of

Periodic Function):

This theorem relates the average power P of a periodic signal to its "fourier series" coefficients.

The Parseval's theorem states that the total average power of a periodic signal x (t) is equal to the I
sum of the average powers of the individual fourier coefficients i.e. Cn.

Average power of x (t) = (power of Ct) + (power of C2)+ ...

..
...(2.2.13)1

...(2.2.14

Proof:

The total average normalized power of signal x (t) is given by the following equatio

The term inside the square bracket in Equation (2.2.18) is nothing but "Cn ".

The interpretation of the Parseval s theorem is that the total average power of the signal x (t)
can found by squaring and adding the heights | C | of the amplitude lines in the spectrum of the
periodic signal x (t).
Thus the Parseval's theorem implies "superposition" of the average powers.

Dirichlet Conditions for the Existence of Fourier Series :

The fourier series stated in the Equations (2.2.20) and (2.2.21) will exist if and only if the
I periodic signal x( t) satisfies the following conditions. These are known as the "Dirichlet" conditions. I
They are as follows :
1. The periodic signal x(t) and its integrals are finite and single valued in the interval I (t to t + T0),
i.e. over a period of one cycle T0.
2. x( t) must have only finite number of discontinuities in the given interval of time.
3. x( t) should have only finite number of maxima and minima in the given interval of time.
4. The function x( t) is absolutely integrable, that is

2.2.6 Examples on Fourier Series :

Ex. 2.2.1 :
Obtain the quadrature fourier series for the rectangular pulse
train shown r I Fig. P. 2.2.1.

Fig. P. 2.2.1

Soln.: Let us obtain the quadrature fourier series for the given rectangular pulse. The
quadrature fo series is given by the following expression :

To find the fourier coefficients, we must consider one complete cycle of x (t) for
integration. I

T0
T0
Here we will consider one cycle from t = - ~z~ to t = ~x~. Let us obtain the fourier coefficients now.

Ex. 2.2.2:
Obtain the exponential fourier series for the rectangular pulse train
shown in Fig. P. 2.2.2(a) and sketch the spectrum.

Fig. P. 2.2.2(a): A rectangular pulse train

Soln.: The exponential fourier series is given as,

(ii)

To obtain the fourier series :

Substitute the value of Cn into Equation (1) to obtain the exponential fourier series as,

ni

Spectrum of the signal x (t): From the value of Cn in Equation (4), it is clear that Cn does not

have any imaginary part, "berefore the amplitude spectrum of x (t) is given as,

Thus the amplitude spectrum of a rectangular pulse of duration X is a sine function. The spectrum
. -own in Fig. P. 2.2.2(b).

The imaginary part of Cn is zero therefore the phase spectrum is zero for all the
values of f. The _>e spectrum is shown in Fig. P. 2.2.2(c).

Fig. P. 2.2.2(c): Amplitude and phase spectrum of a periodic pulse train

Conclusions:

Important points related to the amplitude and phase spectrums of Fig. P. 2.2.2(c) are :

1. The spacing between the adjacent spectral components is f0.

2.

The shape of the envelope of amplitude spectrum | Cn | is determined by A, x and ir envelope is

"sine" shaped.

3. Zero crossings occur in the envelope of the amplitude spectrum at frequencies of f = 1/T. ~ 1
4.

The phase spectrum takes on values of 0 corresponding to the positive values of | Cn |. Hon it
takes on values of 180 corresponding to the negative values of | Cn | e.g. bet* f = 1/T,
2/x. The negative values in the amplitude spectrum are made positive by assigl phase shifts of
180.

5.

Note that the choice of phase shift + 180 or - 180 is arbitrary. However we have used : :<:
them to preserve the "antisymmetry" of the phase spectrum.

Ex. 2.2.3 :

A train of rectangular pulses making excursions from 0 to 10 volts has a

durand 50 msec and are separated by intervals of 500 msec. Assuming that the centre
ol is located at t = 0, obtain the fourier series of the above signal and sketch the specrjJ

Soln.:

It is given that:
50 msec.

T0 = 500 msec.

A = 10 volts, ;jr=0.1

x=

(i)

Fig. P. 2.2.3(a): Train of rectangular pulses

To obtain the value of Cn :

We have already obtained the expression of Cn in the Ex. 2.2.2. It is as follows :

and the amplitude spectrum is given as,

How to plot the amplitude spectrum ?

Substitute n = 0 in Equation (2) to get,

x But we know that 7f= 0.1

and A = 10 and sine (0 ) = 1

The other values of I Cn I are found using Equation (1), as shown in the Table P. 2.2.3.

Using the Equation (5), we can calculate the amplitude response as shown in Table P.
2.2.3. Table P. 2.2.3 : Amplitude and phase spectrums for positive value of "n"

To obtain the values of | Cn | for negative values of "n", substitute negative "n" in
Equation (5).

Thus the values of amplitude spectrum for various negative values of "n" will be same as that
fori the corresponding positive values. The amplitude and phase spectrums are as shown in Fig. P.
2.2.3(b).
Ex. 2.2.4 :
Obtain the fourier series of the unit impulse train shown in Fig. P. 2.2.4(a).
Also plot thai amplitude and phase spectrums for the same.

Soln.: (i) Value of Cn for a train of rectangular pulses :

In the Ex. 2.2.2, we have already obtained the exponential fourier series for a train of rectangulJ
pulses of duration x and period T0. Its value is,

We will use it to find the value of Cn for an impulse train.

Equivalence between a rectangular pulse and an impulse :

The rectangular pulse of amplitude A and width x is as shown in Fig. P. 2.2.4(b).

.-.

Area under a rectangular pulse = Ax

...(2)

As

we know, area under an impulse = T

...(3)

Fig. P. 2.2.4(b): Equivalence between rectangular pulse and unit impulse

Henc

e a rectangular pulse will be equivalent to a unit impulse as its area i.e. "AT" approaches
or." and as its width x approaches zero.
Ax-*landx->0

Ji

...(4)

To obtain Cn for the unit impulse train : We can obtain the value of "Cn" for the unit impulse

train by applying Equation (4) to Soti- 3n (1) as follows :

This is the value of Cn for the unit impulse train. To obtain the exponential fourier series :
Substitute the value of "Cn" from Equation (5) into the standard expression of exponential
fourier


This is the required fourier series for the unit impulse train

Amplitude spectrum :

Amplitude spectrum:
This means that for every value of "n" the value of Cn is going to be the same, equal to ( IT The
amplitude spectrum also is train of impulses each having amplitude of ( 1/T0 ), as showx 1 Fig. P.
2.2.4(c).

Phase spectrum :

The phase spectrum <|)n = arg ( Cn ) = 0 as Cn is constant. The phase spectrum is as shown in
Fig. P. 2.2.4(c).

Ex. 2.2.5 :
Obtain the fourier series of the sawtooth waveform shown in Fig. P. 2.2.5(a)
and plot si spectrum.

Sain.:

Representation of the signal:

The sawtooth signal x (t) shown in Fig. P. 2.2.5(a) can be represented over one cycle as,

and one cycle period T0 = T.

.
To find Cn:
In order to represent the signal in the form of an exponential fourier series, let us find "Cn" using IK
following equation :
.

At this point, let us use a standard identity which states that,

Using Equation (4) we get,

r " ""

Ex. 2.2.6 :

Fig. P. 2.2.5(b): Amplitude and phase spectrums

Let x (t) denote the periodic signal represented in Fig. P. 2.2.6(a).

Fig. P. 2.2.6(a)

(i)
Describe analytically that the above pulse is a rectangular pulse using follow
equation.

(ii)

Determine the fourier series expansion for the signal x( t). (iii)

Find the fourier series coefficients of x( t).

:r :

7i f - J represents a rectangular pulse centered about t = 0. The width of this rectangular pulse

is as shown in Fig. P. 2.2.6(b).

The given waveform of Fig. P. 2.2.6(a) can be obtained by adding infinite number of
shifted rectangular pulses which are of width x and centered around t = T0, 2 T0 .... as shown
in Fig. P. 2.2.6(c). The mathematical expressions for such shifted rectangular pulse are written in
Fig. P. 2.2.6(c) itself.

Fig. P. 2.2.6

Addition of the waveforms in Fig. P. 2.2.6(b) and (c) will give us the waveform in Fig. P. 2.2.6(a).
I

(ii)

For the fourier series and fourier coefficients refer Ex. 2.2.1.

Ex. 2.2.7:
Find the quadrature fourier series for the full wave rectified sine wave shown
in I Fig. P. 2.2.7.

Fig. P. 2.2.7 : Full wave rectified sinewave

Soln.:

The quadrature fourier series for a periodic signal x (t) is given by :

So, we will have to find the values of the fourier coefficients aQ, an
and bn.

Phase spectrum:

Phase spectrum

<)>n = arg (Cn)

= 0 The phase spectrum is as shown in Fig. P. 2.2.8.

Exponential fourier series:

Substitute "Cn" in Equation (1) to get,

12.3
C.T.
Fourier
Transform
:
_____________________________________________________________

Till now we have seen how to represent the periodic signals extended over the interval (- , ),

using the fourier series. Non-periodic time limited signal can also be represented by the fourier
series.

1(a) Line spectrum showing vertical spectral

(b) Continuous spectrum as f > 0

.................................................lines at f0 2f0,
Fig. 2.3.1
However the non-periodic signals which extend from - to can be represented more
conveniently using the "Fourier Transform" in the frequency domain.
It is possible to find the fourier transform of periodic signal as well. For the periodic signals
T0 > . Hence the frequency f0 = 7f> 0. Therefore the difference between the spectral
components which is f0 (as seen in the line spectrum) becomes extremely small and they come
very close to each other. Due to this the frequency spectrum appears to be continuous as shown
in Fig. 2.3.1(a) and (b).

I Necessity of Fourier Transform :

Any signal is build up by addition of elementary signals which are at different frequencies, have
different amplitudes and relative phases.
Using the fourier transform we can plot the amplitude and phase spectrums which provide us all
die information about amplitudes and relative phases of such elementary signals.
i

Thus fourier transform can be used for the "analysis" of a signal. It is used for transform a 3t I
from the time domain to frequency domain.
The F.T. can also be used for analysis of LTI systems.

2.3.2 Definition of Fourier Transform :

The fourier transform of a signal x (t) is defined as follows :

These equations are known as "analysis" equations.

2.3.3 Definition of Inverse Fourier Transform :

The signal x (t) can be obtained back from fourier transform X (f) by using the inverse few
transform. The inverse fourier transform (IFT) is defined as follows :

Representation :

The signal x(t) and its fourier transform X(f) form a fourier transform pair which car
represented as,

- X(f)

x(t)
:.

The other way to represent is as follows :

= F[x(t)]

X(t)
J

or it can be represented as,

= F_1|X(f)|

X(t)
U

The fourier transform is a complex function of frequency f. Therefore it is possible to expw in

the complex exponential form as follows :

X(f)|-e

X(f)

j e ( f)

In this expression:

| X (f) | =

The amplitude spectrum of x (t) and

9 (f)

The phase spectrum.


The amplitude spectrum is a graph of amplitude versus frequency. Whereas the phase
spectrum is 4)h of phase angle versus frequency.

4 Conditions for the Existence of Fourier Transform :

These conditions should be satisfied by a signal x (t), then only it is possible to obtain
the fourier

transform of x (t).

For the periodic signals the integration is obtained over one period however for the
periodic

signals, it will be obtained over a range - to .

The signal x(t) will have to satisfy the following conditions so that it's fourier transform
can be

obtained:

1. The function x (t) should be single valued in any finite time interval T.

2.It should have a finite number of discontinuities in any finite interval T.

3.The function x (t) should have a finite number of maxima and minima in any finite
interval of time T.

4.The function x (t) should be an absolutely integrable function.

oo

means J |x(t)|dt<

That
...(2.3.7)

oo

The conditions stated above are sufficient conditions, but they are not the necessary
conditions.

15 5 Amplitude and Phase Spectrums :

The amplitude and phase spectrums are continuous rather than being discrete in nature.
Out of them, the amplitude spectrum of a real valued function x (t) exhibits an even symmetry.

X(f) = X(-f)

.-.
...(2.3.8)

And the phase spectrum has an odd symmetry. That means,

6(f) =

-0(-f)

...(2.3.9)

Courier transform of some Important Signals (Energy and power signals):

We will obtain fourier transform of some important signals by solving following examples.

h. 2.3.1 :

Find the fourier transform of the decaying exponential pulse shown in Fig. P. 2.3.1.

Soln.:

The exponential pulse shown in the Fig. P. 2.3.1 can be represented mathematically as
follows:
x(t) = e~at
= 0 fort<0

fort>0

It can be represented in an alternate way as,

x(t) = e~atu(t)

The meaning of both the Equations (1) and (2) is the same. This is because u (t) = 1 for t > 1
multiplying by u (t) does not affect the original function.

To find the fourier transform :

Ex. 2.3.2 :
Find the fourier transform of the exponential pulse shown in Fig. P.
2.3.2(a). Asel amplitude and phase spectrums for the same.

Fig. P. 2.3.2(a)

Soln.:

The pulse shown in Fig. P. 2.3.2(a) can be represented as,

x(ft = e* fox \.<Q

= 0

for t > 0

(ii) The phase spectrum

0(f) = 0.

The amplitude and phase spectrums are plotted as shown in Fig. P. 2.3.3(b).

------------------------------------------------------------------------------------------------------------Ex 2.3.4 :
Obtain the fourier transform of the delta function shown in Fig. P.
2.3.4(a).

By the definition of fourier transform,

X(f) =
...(1)

J x(t)e-j2,rftdt= J 8(t)e*"*<k

OO

OO

We cannot substitute the value of 8 (t) directly in the Equation (1) because it is infinitely large at =
Therefore let us use the sifting property of the delta function.
Sifting property of delta function :
The sifting property states that

oo

j f (t) 8 (t

- td) dt = f(td)

...(2)

OO

Let us use this property in Equation (1) as follows :

_____________________________________________________1J3-J Digital Signals and Systems

(B.Sc. IT - Mil) 2-34_____________________________________Fourier Analysis of Periodic & ACTS&S

(iii)

In Equation (2) assume that td = 0 and f (t) = e~J

.-.

X(f) =

oo

J e"j2rft-5(t-0) .-.

by using Equation (2).

OO

X(f) = e"j2nft<i,but td = 0

.-.

X(f) = e'i2m'=l

...(3)

Thus (Sit) - I
__________________i
I
The amplitude spectrum of the delta function is a
shown in the Fig. P. 2.3.4(b) This shows that the delt
function contains all the frequencies from - to wit]
equal amplitudes. The fourier transform of a delt
function is a dc signal.
Ex. 2.3.5 :

Obtain the FT. of the antisymmetrical pulse shown in the Fig. P. 2.3.5.

Fig. P. 2.3.5

Soln.:

The antisymmetric pulse can be

represented as,
x(t) = e~"

t>0

= |1|

t=0

= -eat

t <0

Therefore the fourier transform is given by,

Ex. 2.3.7 :

Obtain the fourier transform of a unit step function.

Soln.:

A unit step function is mathematically defined as,

=u

Using the definition of the fourier transform we get,

But unit step function is present only for t > 0

eisewnei

Ex. 2.3.8:
Obtain the fourier transform of a rectangular pulse of duration T and
amplitude A as shown in Fig. P. 2.3.8(a).

Fig. P. 2.3.8(a): Rectangular pulse

Soln.:

The rectangular pulse shown in Fig. P. 2.5.8(a) can be expressed mathematically as,

rect(t/T) = A for-T/2< + <T/2

This is also known as the gate function.

Therefore the fourier transform will be,

= 0

elsewhere

...by definition of FT

As per the Euler's theorem,

je -jo
eJ -e J
sine =

Applying this to Equation (1),

we get,

F[x (t)] = [sin (TtfT)]

...(2)3

Multiply and divide the RHS of Equation (2) by T to get,

In the above equation,

Thus the rectangular pulse transforms into a sine

function. *iccrjde spectrum :

The amplitude spectrum of the rectangular function is as shown in Fig. P. 2.3.8(b).

t
sinc(fT) =0

Fig. P. 2.3.8(b): Amplitude spectrum of a rectangular

pulse
- e already know, sine (0) = 1

.\ AT sine (0)

= AT
The sine function will have zero value for the following
values of "fT" :
forfT = 1,2,3,........................

i.e. for f = j,j,^.................................................

o_____

p-^
Application of Laplace
^^^| Transform to System Analysis
I

abus:
Introduction, definition, region of convergence (ROC) LT of some important
functions, Initial and final value theorems, convolution integral, Table of Laplace
transforms, partial fraction expansions, network transfer function. S-plane Poles
and zeros. LT of periodic functions. Application of LT in analysing networks.

Contents
Page No.

3.1 Introduction

3-2

3.2 Definition of Laplace Transform

3-2

3.3 Laplace Transform used for Waveform Synthesis

3-16

3.4 Partial Fraction Expansion Method (P.F.E.)

3-26

3.5 Network Transfer Function

3-50

3.6 Application of Laplace Transform in Analysing Networks

3-51

3.7

3.1

Introduction :___________________________________________ I

3.8
We have studied the fourier analysis of continuous time signals and systems. We know
that I fourier transform exists if the signals have finite energy. But for the signals such as ramp, rising
exponents etc.; this condition of finite energy is not satisfied. Thus fourier transform does not exist for I
such signals.
3.9
By the use of laplace transformation; this limitation can be avoided. We know that in
fourier transform the variable 's' = jco. But in laplace transformation variable 's' can be expressed as,
3.10
Here a is real part which represents the attenuation factor and jco is imaginary part, in
which co is angular frequency.
3.11
Laplace transform exists for almost all signals of practical interest. Some of the
advantages of I Laplace transform are as follows :
1. Solution of integrodifferential equations of continuous time systems can be easily obtained.
2. Initial conditions are automatically incorporated.
3. Both complementary and particular solution can be obtained in one operation. Thus it gnJ
complete solution.

3.2
Definition of Laplace Transform :____________________________I

3.12

3.13
3.14
3.15

The laplace transform of continuous time signal x (t) is denoted by X (s).

Now, recall the definition of fourier transform.
A fourier transform of continuous function x (t) is given by,
3.16
3.17
J x (t) e~"j<ot dt

oo

X(jco) =
...(3.2-1

3.18 oo
3.19
We know that fourier transform exists only if x (t) has a finite energy. If the signal x (t) is
hz infinite energy then we will introduce a convergence factor, e~ ; to convert x (t) into finite enesJ
signal. Then we will denote the resulting transformation by X (a + jco). Thus Equation (3.2.1) becomoJ

3.20

Notation :

3.21

The laplace transform of signal x (t) is X (s) and it is denoted by,

3.22 L
3.23x(t) X(s)
3.24

3.25

or L{x(t)} = X(s)

Here bidirectional arrow indicates that we can obtain original signal x (t) from its

laplace : ?rm by using inverse laplace transformation (ILT).

3.26 .-. x(t) =
L-'{X(s)} Here L~ denotes laplace inverse operation.
3.27

In Equation (3.2.3), the limits of integration are from - to + ; so it is called as

double sided lateral laplace transform.

3.28 Region of Convergence (Existence of Laplace Transform):
3.29

According to the definition of laplace transform.

3.30

oo

X(s) =
...(3.2.4)

3.31

Jx(t)e"stdt
3.32
3.33

oo

Thus laplace transform will exist only if,

3.34

J|

x(t )e~ s , |dt

3.35

<

...(3.2.5)

Now we have,
3.36
3.37
-jrot

3.38
= e

s = a+jto
-st
/0 ~ ,x

-at

.-. e
...(3.2.6)

3.39 c have e~JC0t = cos cot j sin cot. Thus the value of e~JC0t is always in the range + 1 to - 1. So
we ami -. c:fy the condition of existence of laplace transform as,
3.40 oo
3.41
x(t)e~ot|dt
3.42
3.43

3.44

< oo

I|
...(3.2.7)

oo

Equation (3.2.7) gives sufficient condition for the existence of laplace transform.

rta;

3.45 . -e range of values of a for which Equation (3.2.7) attains some finite value is called as
region ::;ence(ROC).

3.46 3.2.2 S-plane Poles and Zeros :

3.47
The laplace transform of any time domain signal can be expressed in terms of ratio of
numerator I polynomial and denominator polynomial.

3.48

Here N (s) = Numerator polynomial

3.49 D (s) = Denominator polynomial

3.50
We can factorize the numerator and denominator. Thus after obtaining roots of
numerator and I denominator; Equation (3.2.8) can be written as,
3.51
-(s-z 1 )(s-z 2 )....(s-z n )
3.52
3.53

a0
1

b 0 (s-p 1 )(s-p 2 )....(s-p m )

Here ag and b0 are constants.

Zeros:

3.54

3.55
If 's' takes values zx, z2 .... zn then according to Equation (3.2.9); laplace transform X i
vanishes. Thus such complex frequencies are called as zeros of laplace transform. Zeros are denoted i
the mark '0' in the pole zero diagram.

Poles:

3.56

3.57
When V takes the values pj, p2 .... pm then according to Equation (3.2.9); laplace
transform Xm becomes infinity. Such complex frequencies are called as the poles of laplace transform.
Poles m denoted by the mark 'X' in the pole-zero diagram.

Example:

3.58

3.59

Consider that the laplace transform of signal f (t) is given by,

3.60
s2 - s - 2 s
-s-6

3.61

By obtaining the roots of numerator and denominator we can write,

3.62

X(s)

(s + 2)(s-3)

Zeros:
3.63

Zeros are obtained by equating each bracket of numerator term to zero.

3.64

s+l

= 0

=>

s = -l that means Zj - - 1 and

s-2 = 0

=>

s = 2 that means z2 = 2 Poles:

3.65
Poles are obtained by equating each bracket of denominator term to zero.
3.66

.. s + 2 = 0 => s = - 2, that means pl=-2

3.67 s - 3 = 0 => s = 3, that means p2 = 3

3.68
The pole-zero plot is a graph drawn in s-plane. That means it is graph of real part of V
verses imaginary part of V. So it is the graph of 'a' verses 'jco'. The pole-zero plot is shown in Fig. 3.2.1.
3.69

II 2.3 Laplace Transform of Some Important Functions

In this section we will obtain the laplace transform of roe basic

signals. Such signals include delta function unit E?. exponential signal
etc. Delta

3.70I

function :

The delta function is shown in Fig. 3.2.2

3.71

Fig. 3.2.2

It is defined as,
8(t) =

3.72

fort = 0
= 0

otherwise

...

(3.2.10)
3.73

According to the definition of laplace transform,

CO

X(s) =

|x(t)-e~stdt

3.74

oo

...(3.2.11)

Here x (t) = 8 (t) and its value is 1 only at t = 0; it is

not necessary to take integration. Thus L. -_::on (3.2.11) becomes,

IX(s) =

l-e=l

L.T.
.-. o(t) <-----------------------------------------------*1
Since 's- term is absent in Equation (3.2.12);
ROC is entire s-plane.

Unit step:

Unit step is as shown in Fig. 3.2.3.

3.75

3.76

v________________________1

-------

3.77 ROC:
3.78

The laplace transform of unit step is - and it is for the range Re {s} > 0, that

for a > 0. Thus ROC is o>0. It has a pole at s = 0 that means at origin. The sketch of ROC is start
3.79
Fig. 3.2.4.

3.80

3.81

It is defined as, r(t) = t

fort>0 = 0

otherwise

3.82

3.83
3.84

3.86
3.85
3.87

Positive sided growing exponential pulse :

i is also called as right handed growing exponential pulse I a

pulse is represented as F (t) = ea u (t). Here a is arbitrary

F"

3.88
/x(t)e~stdt

We have, X(s) =
...(3.2.19)

3.89
oo
3.90
Here
x(t) = e a t - u (t ) ...
(3.2.20)
3.91 v'..:tiplication by unit step u (t) indicates that the exponential pulse ea' is present in the range
3.92 "": -5 limits of integration in Equation (3.2.19) will be from t = 0 to t = .

ROC:

3.93

3.94
The laplace transform is _ . It has
pole as + 'a'.
3.95
Thus ROC is Re {s - a} > 0, that means Re {s} >
a. But Re {s} means a.
3.96

.-. ROC is a > a

3.97

This ROC is shown in Fig. 3.2.8.

3.98 5.

Positive sided decaying

exponential signal:

3.99
This is also called as right handed
decaying exponential signal. It is given by x (t) = e~ a u
(t). Here 'a' is some positive arbitrary constant. Such a
function is shown in Fig. 3.2.9.

3.100
According to definition of laplace
transform,
3.101
We have x (t) = e at u (t). Multiplication
by unit step u (t) indicates that the signal is onl; I range t
= 0 to t = . Thus Equation (3.2.22) becomes,

3.102

3.103

3.104

3.105

The ROC is,

3.106

Re {s + a}> 0 ; that means Re {s} > - a

3.107

.-. ROCisa>-a

3.108

This ROC is shown in Fig. 3.2.10.

3.109

Negative sided (left handed) exponential signal: As the name indicates;

such exponential signal is in the range t = - to t = 0. This signal is given x(t) = -e-atu(-t)
Here 'a' is some arbitrary constant. For a > 0 the signal is shown in Fig. 3.2.11(a) and for a <
0, lie

:-_" is shown in Fiff. 3.2.1 \(b\

3.110
3.111

Since the eiven signal is in the ranee t = - to t = 0 we get,

3.112 In Equation (3.2.24) if the power of exponent of second term is negative then we will get
tha I means this term becomes zero. Thus we can write the laplace transform.

3.113
3.114

The pole is at s = - a

ROC:

3.115
To obtain this laplace transform, the condition
is Re {(s + a)} < 0. That means Re {s} < - a.
3.116
3.117

.-. ROC is o <-a

This ROC is shown in Fig. 3.2.12.

3.118

3.2.4

Convolution Integral:

Statement:
3.119
If x(t) i--------->X(s),

LT
ROC:^

3.120
and h (t) <-------> H (s)
3.121 LT
then x; (t) * h (t) <-------> X (s) H (s)
3.122

3.2.5

Initial Value Theorem :

3.123LT
If x (t) <------- X (s)

LT
ROC : R2
ROC : Intersection of R[ and R-

3.124

1.2.6 Final Value

Theorem
:

3.125 2.7 Laplace Transform of Periodic Signal:

Satement:

3.126

3.127
If x (t) is the periodic function with fundamental period T0' that means x (t) = x (t
+ T0) then

3.128

3.129

tanf : From the definition for unilateral Laplace transform

We know signal x (t) is

3.130
3.131

3.2.8 Table of Laplace transform :

Table 3.2.1 shows the standard laplace transform pairs.
3.132 Table 3.2.1

3.133

3.134

3.135

Summary of laplace transform properties :

Table 3.2.2 shows the summary of laplace transform

properties. Table 3.2.2

3.136

3.137

3.138

Solved Problems:

3.139

iii

According to differentiation in s domain property

in I

Given y (t) = x (t) cos 7t

3.140

3.141

3.142
that

One of the important properties of Laplace transform is Time cos property. It states

3.143

3.144

EL. 3.2.2 :

Find Lapface transform of following signafs. Draw ROC in each case

3.145

3.146

Here ROC remains unchanged.

Thus ROC is 6 > 0. It is shown in Fig.

3.147

ROC of X (s) is the combination of two ROCs. Thus combined ROC is a > 0. It is

same as ROC shown in Fig. P. 3.2.2.

3.148___________________________________________________________________________3

.3

Laplace Transform used for Waveform Synthesis :__________

3.149
Any time domain function can be expressed in terms of singular functions. There are
additions a subtractions of a singular function to an existing function. There are three possibilities.
(1) Step Step
(2) Step Ramp
(3) Ramp Ramp Case 1: Step
Step :
3.150
The addition or subtraction of a step to a step results in a step function. The
magnitude of n resultant is the algebraic addition or subtraction respectively of the two steps. The
change in ll magnitude occurs at the instant of addition.

3.151

DEMO
3.152

Consider x (t) = u
(t) + 3u (t - 2) = Xj
(t) + x2 (t) say

3.153

3.154 Fig. 3.3.1

3.155
For t < 0, X! (t) and x2 (t) have value zero. For 0 < t < 2, x2 (t) is zero. Therefore
resultant will ha t only value of x, (t) i.e. 1. At t = 2, we have addition of case 1 - step plus step. The
resultant is a 'kn The magnitude is the addition of two steps, i.e. 1 + 3 = 4. The change in the resultant
occurs at -.stant of addition i.e. at t = 2 as shown in Fig. 3.3.1(c).

3.156

3.157

lf instead we add a wave - 3u (t - 2) the resultant would have magnitude 1 + (- 3) = - 2,

at . as shown in Fig. 3.3.1(d). The student should note that in any term such as M u (t N), M refers to the magnitude (either or negative) and N to the instant in time. I : Step
Ramp : en the addition of a step is done to a ramp function, the result is a ramp
function, shifted by an 1IIL-: equal to the step.

3.158 DEMO
3.159

Consider x (t)

3.160

= u(t) + r(t-2)

For t

< 0

x(t)

For 0

< t < 2

x(t)

=0
3.161
=l

3.162

3.163

Fig. 3.3.2

3.164
At t = 2, case 2 addition occurs. The resultant is a ramp. It is shifted by the magnitude
of >JM i.e. + 1. This is done at t = 2. This gives the wave as shown in Fig. 3.3.2(c).

3.165

3.166

Case 3 : Ramp Ramp :

3.167

The addition of a ramp to a ramp gives a ramp. The slope of this ramp is the algebraic

addition of two slopes. The change in slope occurs at the instant of addition.
3.168

A special case is an addition of two equal and opposite slopes. This results in a ramp

of zero slope i.e. a horizontal line.

3.169 Consider, x (t) = r ( t ) - 2 r ( t - 3 )
3.170
nesaj

r (t) is shown in Fig. 3.3.3(b). At t = 3, there is a case 3 addition. The new ramp has

3.171

3.172

r^nitude [1 + (- 2) = - 1]. So the resultant ramp has a slope of- 1 units. This change in slope
occurs at x(t) = r ( t ) - 2 r ( t - 3 ) If x (t) = r (t) - r (t - 3), then the resultant would be a ramp of
slope zero [1 + (- 1) = 0] and in r: - case the line would be horizontal as shown in Fig. 3.3.3(c).

3.173
In waveforms synthesis, we have to usually decode a waveform into component
parts. The ir - :~s will illustrate this.
3.174

fc^ped Problems:

3.175

ML 13.1 :

Find the Laplace transform of Fig. P. 3.3.1.

3.176

3.177

^Fig. P. 3.3.1 :;= .-.

x(t) = u(t) + u ( t - l ) - 2 u ( t - 2 )

3.178

Step 2 : We have,
3.179 L x (t)
3.180
3.181

.-.

X (s) then

Lx(t-a)

For x(t)=

e_asX(s)

u(t) + u ( t - l ) - 2 u ( t - 2 )

3.182

3.183
3.184

Ex. 3.3.2 :

Find the Laplace transform of Fig. P. 3.3.2.

Soln.:

3.185

.-.

x(t) = u(t) + u ( t - l ) - u ( t - 2 ) - u ( t -

3 ) Step 1:
3.186

The Laplace of u (t) = ~

3.187
3.188
e
Laplace of u (t - 1) = ~~~

-s

3.189
3.190
e
Laplace of - u (t - 2) = - -

-2s

3.191
-3s
3.192
e
Laplace of-u(t-3) = - -"~
3.193 X (s)vy =\ I ~~+
, I -s eI -2s
i -3s
3.194
3.195 v '
s -~e
s
s-~e
s

3.196

Ex. 3.3.3 :

Find the Laplace transform x (t) of Fig. P. 3.3.3.

3.197

3.198

fltopl:
u(t-3)

3.199
3.200

uz 2 :

x(t) = r ( t ) - 2 r ( t - l ) + r(t-2) + u ( t - 2 ) -

The Laplace transform is,

...Ans. 3.5 :

transform of wave shown in Fig. P. 3.3.5.

Find the iaplace

3.201

Soln.:

3.202 Step 1: The given waveform is,

periodic. For a single saw tooth function.
3.203
x(t) = r ( t ) - r ( t a ) - u ( t - a ) Step 2 : Its laplace is

3.204

3.205
is.

Step 3 : Laolace of oeriodic waveform

3.206

Ex. 3.3.6 :
Find LT of
periodic wave of Fig. P. 3.3.6.

3.207

3.208
t lg.
F.
3.3.
6
3.209

Soln.:

3.210
Step 1: Xj (t) for one period
of given waveform is as shown in Fig. P.
3.3.6(a).
3.211

Aj \i) u yij z, u vi
i) -r u ^i t.)

3.212

3.213

SteD 2 : Its Laolace is.

3.214

^^3.7 :

Find LT of periodic wave of Fig. P. 3.3.7.

3.215

3.216 Fig. P. 3.3.7

3.217

^M1:
x (t) is given as,
3.218
x(t) = u(t) + u ( t - l ) - 2 u ( t - 2 )
3.219 for one wave only. Call it xt (t)
3.220 The period, T
- 3 She ". :
Laplace of x: (t)

is,

Laplace of given periodic wry? X

(s) is, 3.221

3.222 Find the L.T of function shown in Fig. P. 3.3.8.

3.223
3.224 Fig. P. 3.3.8
3.225 tea.

rvm Fig. P. 3.3.8.

3.226

x (t) = A sin t
3.227

= 0 for 7t<t<2rc

for 0 < t < n

3.228

3.229

3.230

3.231

This method is suitable for the laplace transforms which are rational in nature. That

means they are expressed as the ratio of two polynomials.

3.232

X(s) "

D(s) Here N (s) = Numerator polynomial D

(s) = Denominator polynomial

3.233 Steps to follow in

partial fraction expansion :
3.234 1.

Factorize the denominator and obtain the roots. Then the denominator will be in the form.
3.235

(s-P,)(s-

P 2 )....(s-P N ) Here P^ P2... PN are

called as poles.
3.236 2.

Write down the equation in partial fraction expansion form as,

3.237
3.238
3.239

Aj

A2 A^

x (s) = 7T + fr+ ....+ ?r

W

S - P[

S - P 2 S - PN

3.240 Here Al5 A2.... AN are the coefficients of

P.F.E. The coefficient AK is calculated as,
3.241

= (s - PK) x (s)

3.242
3.243

s = pK

After calculating Aj, A2.... AN use the standard laplace transform pairs.
3.244 LT
1
3.245-------------------(i)
eatu(t)<> ROC:rj>a
3.246
s a
3.247
LT
1
3.248-----------------------(ii) -eatu(-t)< >
ROC:a<a
3.249 s a

3.250 The nature of time domain signal; that means whether it is right handed (causal) or lefthanded
3.251 (anticausal), is decided using following rules.
3.252(i)
When the poles lie to the L.H.S. of ROC then corresponding signal is right handed
3.253 (causal). That means it is multiplied by u (t). (ii) When the poles lie to the
R.H.S. of ROC then corresponding signal is left handed
3.254 (anticausal). That means it is multiplied by u (- n).
(iii) For bilateral ILT, the poles must lie on both sides of ROC
3.255

Soived Problems:
3.256While solving the numericals we will consider different cases depending on the nature of
poles.

3.257

Case t: V

3.258

3.259 IfeV 1: First we will obtain the roots (poles) of denominator

term. To obtain poles Pl and P2 use following equation,

3.260

3.261

3.262

3.263

Sfiep 3 : The pole-zero plot and given ROC is shown in Fig. P. 3.4.1. Here both the poles are
to the left side of ROC. Hence both terms corresponds to right handed i rial (multiplied by u
(t)).

3.264

3.265

Now we have standard laplace transform pair.

3.266

3.267
3.268

3.270
3.269

3.271

3.272

3.273

3.274

1K (iii): When ROC is Re (s) < - 2 i.e. a < - 2. The ICC

shown

and

position

of

poles

are

in -z P. 3.4.3(b). Both the poles lie at the R.H.S. of given ROC. nee the corresponding

time domain term are left tanoed. That means multiplied by u (-1). We have standard laplace
transform pair.

3.275

3.276

Now we will consider all possible ROC conditions. The poles are at - 1

and - 2. Thus possible ROC conditions are (i)

(iii)
3.277

ROC : - 2 < a < - 1

ROC : a < - 2
Step 3 : We will calculate inverse laplace for each ROC condition

separately. Condition (i): ROC : a > - 1.

3.278

ROC : CT > - 1 (ii)

3.279

The poles and ROC are shown in Fig. P. 3.4.4.

3.280 Here both the poles lie at the L.H.S. of

given ROC. So the corresponding time domain terms are
right handed. That means, multiplied by u (t).

: ~ we
have,

i.\ x(t) = e_tu(t) + e"?,u(t)

y

(t)

: ladition (ii): ROC : - 2 < a < - 1 This ROC

is shown in Fig. P. 3.4.4(a).

3.281

We have standard laplace transform pair,

3.282
3.283 t

Consider the first term L 1 j

A . Here the pole is at Pl = - 1. This pole lies at R.H.S. of given I So

the corresponding time domain sequence is left handed, that means multiplied by u (-1). We have
standard laplace transform pair,
3.284

S'ow consider the second term. It is L \s + 9| ^ere ^e P^e *s at ^2 = ~~ 2. This pole lies at
of given ROC. So the corresponding time domain sequence is right handed; that means

________J
:edbyu(t).

We have standard laplace transform pair.

3.285

3.286

Condition (iii):

3.287

When ROC a < - 2.

3.288

This ROC is shown in Fig. P. 3.4.4(b).

3.289
In this case both the poles lie at R.H.S. of given ROC. Hence corresponding time domain
terms are left handed that means multiplied by u (- t). We have standard laplace transform pair,
3.290Ex. 3.4.5 :
each cam

3.291

Specify all possible ROCS for the function x(s) given below. Also find x(t) in

3.292

3.293

3.295

Pole-zero plot:

3.294
Ail possible conditions of ROC : The poles are at - 2 and - 4. Thus

possible ROC conditions are : ROC : a > - 2 ROC:-4<cr<-2 ROC:a<-4

Ii_culating inverse laplace for each ROC condition separately. loodition
(i): ROC : a > - 2 The poles and ROC are shown in Fig. P. 3.4.5(a)

Here both poles lie at the L.H.S. of given ROC. So the corresponding time
3.296

domain terms are right mcoed. That means multiplied by u(t).

3.297

Condition (ii): ROC : - 4 < o < - 2

3.298
The poles and ROC are
shown in Fig. P. 3.4.5(b)
3.299 if-41
For the first term L i~Tf Here
3.300 U + 2J
3.301 the pole is at P, = - 2. This pole
lies at R.H.S. of given ROC. So the
corresponding time domain sequence is
left handed, that means multiplied by u(t).

3.302

3.303

Here the pole is at P2 = - 4. This pole lies at L.H.S. of given ROC. So corresponding

time domain sequence is right handed, that means multiplied by u(t).

3.304

Condition (iii): ROC : a < - 4

3.305

The poles and ROC are shown in Fig. P. 3.4.5(c).

3.306
In this case both the poles lie at R.H.S.
of given ROC. Hence corresponding time domain terms
are left handed that means multiplied by u (-1).

3.307

3.308 Ex. 3.4.6 :

Find the inverse Laplace Transform of:

3.309

3.310

(a) s > 3

(b) s < -1

(c) -1 < s < 3

3.311

3.312

3.313

3.314

3.315

3.316 4
4
3.317 Step iii : ROC is not mentioned in the numerical. Here the poles are at -r and + T . We will
consider
3.318
3.319
3.320
3.321

fleeing conditions of ROC to calculate inverse Laplace.

10Ca>4/3

(ii) -T < <7 <T (iii) G<--T

: :ion (i)ROCa>g :
The poles and ROC are shown in Fig. P. 3.4.8

3.322
Here both poles lie at the L.H.S. of given ROC. Hence corresponding time domain
signals are

3.323

' 25 1
4
1 36
- 25 -31
u
L
=36"e

s+x

-i (51361
and L I
J

3l /x
^1 =35-6

u(t)

l -3j

25 ~3l
+

36TT

5 e4
3.324

(t)

|_36'e

4
4

Condition (ii)
ROC -3 < o < + :

3.325

3.326
The ROC and pole
plot is shown in Fig. P. 3.4.8(a).
3.327
4
Here the pole at -T is to the
left of
3.328 ROC. Hence
corresponding time domain term
is right handed.
3.329
4

3.330
r-i
f25 / 361
25
~3l ^ ' L \
u(t) S
4~l = 36e
+

3j

3.331
4 And
the pole at + T is to the
right side
3.332 of ROC. Hence
corresponding time domain term
is left handed.
3.333
f5/361
3.334
u( l)
~

_i
_5_ 3l
= "36e
T

' L i_4

3.335
4
4
3.336
25 "3'
5 3l
3.337
x(t) =
^e u(t) + 3ge u(-t)
3.338
3:

Condition (iii): ROC a < -

3.339
The ROC and plot of
poles is shown in Fig. P.
3.4.8(b).
3.340
Here both poles lie
towards R.H.S. of given ROC.
Hence corresponding time
domain tern

3.341

3.342

3.343
Here the poles are at P, = a + j|3 and P2 = a - jp\ That means the poles are complex conjugate of
h other. But the procedure of obtaining the time domain signal is same as that of case-I.
3.344

?! 3.4.Q ;

Finrl im/firse lanlarfi transform of

3.345

3.346ep 1: First we will obtain the

roots of denominator. The
denominator term is s + s + 1.

3.347

3.349

3.348
Here the poles are complex
conjugate of each other. tea 2 : In the P.F.E. form
the equation of X (s) can be written a

3.350

3.351

3.352

Let us rearrange the second term as follows,

3.354

Thus complete time domain expression is,

3.353

3.355

3.356_________________________________i___________,

________i

I Case III: When there are repeated poles.

3.357

3.358
When there are repeated poles then we have to add some extra terms in the equation of
X (s). For such extra terms the coefficients AK are calculated by taking derivative of X (s). Let as solve
some numericals related to repeated poles.
3.359

Ex. 3.4.10 :

Find inverse laplace transform of

3.360

3.361

for the causal time

domain signal. Soln.: Step 1: The given

equation is,

3.362

3.363

There is a single pole at + 1; while there are repeated poles at + ^. Step 2 : For the

repeated poles; in partial fraction expansion form equation of X (s) can be written as.

3.364

3.365
Note that the second term in the equation of X (s) is an added term. The coefficients
A, aaz are calculated using earlier method. But extra coefficient A2 is calculated by taking the derivalh
fnllows

3.366

3.367

3.368

3.369___________________________________________________________________________3.

Network Transfer Function :_______________________________j

3.370

We know that LTI system can be completely characterised by its impulse response h

3.371

3.372

3.373

3.374

Thus laplace inverse of Equation (5) can be

written as,

3.375

This is the required solution.

3.376

Q. 1
transform.

3.377

Explain the conditions of existence of laplace

Q. 2
transform.

Define the terms poles and zeros with reference to iapiace

3.378

Obtain laplace transform of:

Q. 3

3.379

(i)

8(t)

(ii) u(t)

(iii)r(t) Q. 4

of positive sided exponential signal. Q. 5

negative sided exponential signal. Q. 6

Obtain laplace transform

Obtain laplace transform of

Explain the various steps of

obtaining inverse laplace transform.

+ 1) Q. 7
3)

3.380 (s
Obtain inverse laplace transform of x (s) = (s + 2) (s +
3.381

3.382 Q. 8

,2.

Solve the differential equation -p y (t) + 2 ^ y (t) = x (t)