SRI VENKATESWARA

ENGINEERINGCOLLEGE
KARAKAMBADIROAD, TIRUPATI - 517507

DEPARTMENT OF ELECTRONICS AND COMMUNICATIONENGINEERING

LECTURENOTES

20A04301T-SIGNALS & SYSTEMS

Regulation : R20

AcademicYear : 2021-2022

Year/Semester : II/I

Prepared by
M.Vasudeva Reddy
Associate Professor

Head of the Department

 CONTENTS
1 Unit-I : Signals and Systems Page No
1.1 Introduction 1
1.2 Unit-I notes 1
1.3 Solved Problems 45
1.4 Part A-2 Marks Questions and Answers 50
1.5 Part B-10 Marks Questions and Answers 51
2 Unit-II : Fourier Series and Fourier Transform
2.1 Introduction 53
2.2 Unit-II notes 53
2.3 Solved Problems 74
2.4 Part A-2 Marks Questions and Answers 101
2.5 Part B-10 Marks Questions and Answers 102
3 Unit-III : Laplace Transform
3.1 Introduction 103
3.2 Unit-III notes 103
3.3 Solved Problems 113
3.4 Part A-2 Marks Questions and Answers 130
3.5 Part B-10 Marks Questions and Answers 130
4 Unit-IV : Signal Transmission Through LTI systems
4.1 Introduction 131
4.2 Unit-IV notes 131
4.3 Solved Problems 146
4.4 Part A-2 Marks Questions and Answers 154
4.5 Part B-10 Marks Questions and Answers 154
5 Unit-V : DTFT & Z-Transform
5.1 Introduction 155
5.2 Unit-V notes 155
5.3 Solved Problems 174
5.4 Part A-2 Marks Questions and Answers 182
5.5 Part B-10 Marks Questions and Answers 182
 Course Code SIGNALS AND SYSTEMS L T P C
20A04301T 3 0 0 3
Pre-requisite Mathematics - I Semester III
Course Objectives:
 To introduce students to the basic idea of signal and system analysis and its characterization
in time and frequency domains.
 To present Fourier tools through the analogy between vectors and signals.
 To teach concept of sampling and reconstruction of signals.
 To analyze characteristics of linear systems in time and frequency domains.
 To understand Laplace and z-transforms as mathematical tool to analyze continuous and
discrete-time signals and systems.
Course Outcomes (CO):
CO1: Understand the mathematical description and representation of continuous-time and discrete-
time signals and systems. Also understand the concepts of various transform techniques.
CO2: Apply sampling theorem to convert continuous-time signals to discrete-time signals and
reconstruct back, different transform techniques to solve signals and system related problems.
CO3: Analyze the frequency spectra of various continuous-time and discrete-time signals using
different transform methods.
CO4: Classify the systems based on their properties and determine the response of them.
UNIT - I Signals and Systems
Signals & Systems: Basic definitions and classification of Signals and Systems (Continuous time and
discrete time), operations on signals, Concepts of Convolution and Correlation of signals, Analogy
between vectors and signals-Orthogonality, mean square error.
UNIT - II Fourier Series and Fourier Transform
Fourier series: Trigonometric & Exponential, Properties of Fourier series, concept of discrete
spectrum, Illustrative Problems.
Continuous Time Fourier Transform: Definition, Computation and properties of Fourier transform
for different types of signals and systems, Inverse Fourier transform. Statement and proof of sampling
theorem of low pass signals, Illustrative Problems.
UNIT - III Laplace Transform
Laplace Transform: Definition, ROC, Properties, Inverse Laplace transforms, the S-plane and BIBO
stability, Transfer functions, System Response to standard signals, Solution of differential equations
with initial conditions.
UNIT - IV Signal Transmission through LTI systems
Signal Transmission through Linear Systems: Linear system, impulse response, Response of a
linear system for different input signals, linear time-invariant (LTI) system, linear time variant (LTV)
system, Transfer function of a LTI system. Filter characteristics of linear systems. Distortion less
transmission through a system, Signal bandwidth, System bandwidth, Ideal LPF, HPF and BPF
characteristics, Causality and Paley-Wiener criterion for physical realization, Relationship between
bandwidth and rise time, Energy and Power spectral densities, Illustrative Problems.
UNIT - V DTFT & Z-Transform
Discrete Time Fourier Transform: Definition, Computation and properties of Discrete Time Fourier
transform for different types of signals and systems.
Z–Transform: Definition, ROC, Properties, Poles and Zeros in Z-plane, The inverse Z-Transform,
System analysis, Transfer function, BIBO stability, System Response to standard signals, Solution of
difference equations with initial conditions. Illustrative Problems.
Textbooks:
1. A.V. Oppenheim, A.S. Willsky and S.H. Nawab, “Signals and Systems”, 2nd Edition, PHI,
2009.
2. Simon Haykin and Van Veen, “Signals & Systems”, 2nd Edition, Wiley, 2005.
Reference Books:
1. BP Lathi, “Principles of Linear Systems and Signals”, 2nd Edition, Oxford University Press,
2015.
2. Matthew Sadiku and Warsame H. Ali, “Signals & Systems A primer with MATLAB”, 2016.
 UNIT-I
SIGNALS & SYSTEMS
Introduction:

In a communication system, the word ‘signal’ is commonly used. Therefore we must

know itsexact 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 a function of one or more independent
variables.”
Examples: speech signal, ECG signal, radio signal, TV

signal. These signals can be one dimensional or

multidimensional.

One dimensional signal: when the function depends on a single variable, the signal is
said tobe one dimensional.

Example: speech signal

Multidimensional signal: When the function depends on two or more variables, the
signal issaid to be multi dimensional.

Example: image

Definition of system:

A system is a physical device (or an algorithm) which performs required operation on a

signal.

Fig. 1.1 Functional block diagram of system

A system is a set of elements or functional blocks that are connected together

& produces an output in response to an input signal. The response of the system
depends ontransfer function of the system.

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The functional relationship between input & output is y(t)= T [

x(t)]Where x (t) is the input to the system or excitation.

Y (t) is the output or response of the

system.T is the transfer function of the

system.

Examples: communication filters, amplifiers, TV, audio amplifiers, transmitters, receivers etc.

Classification of signals

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

 Continuous and discrete time signals

 Periodic and non-periodic signals
 Even and odd signals
 Energy and power signals
 Deterministic and random signals
Continuous Time and Discrete Time Signals
Continuous Time signal: A signal is said to be continuous when it is defined for all
instants oftime. Or a signal of continuous amplitude & time is known as Continuous Time
signal.

Examples: electrical signals derived in proportion with the physical quantities such as
temperature, pressure, sound etc.

Fig. 1.2 Continuous time sin signal

Discrete Time Signals: A signal is said to be discrete when it is defined at only

discrete instants of time.

Example: if we take blood pressure readings of a patient after every one hour & plot the
graph then it is discrete signal.

2
 Fig. 1.3. Discrete time signal

Periodic and Aperiodic Signals

Periodic signal: A signal which repeats itself after a fixed time period is called as
Periodicsignal. Alternatively a signal is said to be periodic if it satisfies the condition

x(t) = x(t + T0) for CT signal

x(n) = x(n + N) for DT signal

Where T0 = fundamental time period. The minimum possible interval over which a
functionrepeats is called fundamental period T0.

1/ T0= F= fundamental frequency.

Fig 1.4 Periodic signal

The above signal will repeat for every time interval T0 hence it is periodic with period T0.

Examples: sine, cosine waves, square waves etc..

Aperiodic Signals: A signal which does not repeat at regular interval is called as
AperiodicSignals. Alternatively a signal is said to be non periodic if it satisfies the condition

x(t) ≠ x(t + T0) for CT signal

x(n) ≠ x(n + N) for DT signal

Examples : exponential signal, rectangular pulse, noise etc.

3
 Fig 1.4 Aperiodic Signal

The above signal is a exponential signal which does not repeat for a particular time interval.
So it is a non periodic signal.

Condition for periodicity of DT signal

For DT signal, the condition of periodicity is x(n) = x(n + N)

Let x(n)= A cos[ 2 π f0 n +θ]

So x(n + N) = A cos[ 2 π f0 (n+N) +θ]

According to the condition of periodicity A cos[ 2 π f0 n +θ]= A cos[ 2 π f0 (n+N) +θ]

= A cos[ 2 π f0 n + 2 π f0 N +θ]

To satisfy this condition, 2 π f0 N = 2 π k

f0 = k/N

Here k,N are integers. Thus DT signal is periodic if its frequency f0 is rational.

Periodicity condition for x(n) = x1(n)+x2(n)

According to the Periodicity f 1 = k1/ N1 f2 = k2 / N2

The resultant signal is periodic if N1/ N2 is the ratio of two integers. The period of x(n) will
be least common multiple of N1, N2.

Deterministic and random Signals

Deterministic signal: A signal is said to be deterministic if there is no uncertainty with

respect to its value at any instant of time. Or, signals which can be defined exactly by a
mathematical formula, look up table or some well defined rule are known as deterministic
signals. Example: sine, cosine wave.

Random Signals: A signal is said to be non-deterministic if there is uncertainty with respect

to its value at some instant of time. Non-deterministic signals are random in nature
hence they are called random signals. Random signals cannot be described by a
mathematical equation. They are modeled in probabilistic terms.

4
 Example: noise generated in electronic components, transmission channels etc

Fig 1.5 Deterministic signal Fig 1.6 Random signal

Deterministic signal Random Signal

1. A deterministic signal is one which can A random signal is one which cannot be
be completely represented by represented by any mathematical
Mathematical equation at any time. equation.

2.eg. Sine wave, cosine wave. Noise generated in electronic

components, transmission channels,
cables.
3. These signals can be periodic or non These signals are non periodic.
periodic.
Even and odd signal
Even or symmetrical signal
A signal x(t) is said to be symmetrical (or) even if it satisfies the following
condition:X(t) = x(-t) for CT signal
X(n) = x(-n) for DT signal
Example: cosine waveform, rectangular pulse.

As shown in the following diagram, rectangular function satisfies the condition x(t) = x(-t) so
it isalso even function.

Fig 1.7(a) Example of Even signal Fig 1.7(b) Example of odd signal

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Odd (or) Anti symmetrical signal
A signal x(t) is said to be anti symmetrical (or) odd if it satisfies the following
condition:X(t) = -x(-t) for CT signal
X(n)= -x(-n) for DT signal
Example: sine waveform
Decomposing a CT signal into even & odd parts:
Any CT signal can be expressed as the summation of even part & odd part.
X(t)= xe(t)+ xo(t)

Even part of x(t) is expressed as,

Odd part of x(t) is expressed as,

The even part of a signal is an

even signal, since

Similarly any DT signal can be expressed as the summation of even part & odd part.
X(n)= xe(n)+ xo(n)
Even part of x(t) is expressed as, xe(n) = [ x(n)+ x(-
n] / 2Odd part of x(t) is expressed as, xo(n) = [ x(n)-
x(-n)] / 2.
Steps to be followed to find even and odd component of the given signal is:
1) Draw the signal x (t).
2) Draw its folded version x (-t)
3) Add x (t) and x (-t) or subtract x (-t) from x (t)
4) Divide the addition by 2 to get xe(t)& subtraction by 2 to get xo(t).
Energy and power signal

Power signal: A signal x (t) is said to be power signal, if and only if the normalized
averagepower p is finite and non-zero. (0 < P < , E  ).

Almost all the periodic signals are power

signals. The average normalized power is

given by

For periodic signals, the power P can be computed using a simpler form based on
theperiodicity of the signal as

where T is the period of the signal.

6
Energy signal

A signal x (t) is said to be energy signal if and only if the total normalized energy is
finiteand non-zero. (0 ≤ E < , P = 0)

The total energy contained in and average power provided by a signal x(t) (which is a
function of time) are defined as

Comments:

1. The square root of the average power P of a power signal is what is usually
defined as the RMS value of that signal.

2. If a signal approaches zero as t approaches  then the signal is an energy

signal. This is in most cases true but not always.

3. All periodic signals are power signals (but not all non–periodic signals are
energysignals).

4. Any signal f that has limited amplitude (| f | < ) and is time

limited (f = 0 for | t | > t0 for some t0 > 0) is an energy signal .

Energy and Power: The total energy of a discrete-time signal is defined by

The time-average power is

7
Comparison of energy and power signals
Power signals Energy signals

The normalized average power is finite Total normalized energy is finite and
and non zero. non- zero.
Almost all periodic signals are power Almost all non periodic signals are
signals. energy signals.

Energy of the power signal is infinite. Power of the energy signal is zero.

Energy signals exist over a short period

of time. They are time limited.
Power signals can exist over an infinite
time. They are not time limited.

PROBLEMS BASED ON CLASSIFICATION OF SIGNALS

1. Prove that the sine wave is a periodic signal.

The sinewave is mathematically expressed as, x (t) = A sin w0 t

If it is a periodic signal it should satisfy the condition x(t) = x(t+T 0)Now let us calculate x (t+T0) = A sin

w0 ( t + T0)

= A sin (w0t + w0T0)

= A sin (w0t + 2 πT0 / T0)

= A sin (w0t + 2 π)

= A [sin w0t cos 2 π + cos w0t sin 2 π)

= A [sin w0t .1+ cos w0t .0]

= A sin w0t

= x(t)

Therefore x (t+T0) = x (t).The sine wave is a periodic signal.

2. Prove that the exponential signal is non periodic.

The exponential signal is expressed as, x(t) = e-αt

8
Put t= t+T0 in above equation, we get

x(t+T0) = e-α(t+T )
0

= e-αt e-

αT
0
But for exponential signal

T0=∞.

So e-αT0 = 0

x (t+T0) = e-α.t .0=0

x(t) ≠ x(t+T0).

Hence the exponential signal is non periodic signal.

3. State whether the following signals x(t) is periodic or not. If it is periodic find
the corresponding period.

(a) x(t)= 3 sin 4t

The given equation is a sine wave. So it is a periodic signal.

To find time period, compare the given equation with standard sine wave x(t)= A

sin wt w =4

f = 4 / 2π = 2 / π.

So time period T= π / 2.

(b) x(t)= 2 cos 100πt + 5 sin 50t

The given signal x(t) is the addition of two signals x 1(t) &

x2(t).Let x1(t) = 2 cos 100πt & x2(t)= 5 sin 50t

If x(t) is a periodic, the ratio T1/T2 is the ratio of two integers.(rational

no)So calculate T1 & T2 .

Compare x1(t)= 2 cos 100πt with the standard cosine wave equation x 1(t)= A cos

w1tw1 = 100π

9
 2 π f1 = 100π 2 π/T 1 = 100πT 1 = 1/50

Similarly Compare x2(t)= 5 sin 50t with the standard sine wave equation x 2(t)= A sin

w2t,W 2 = 50

2 π f2 = 50

2 π/T 2= 50

T 2 = 2π /50

Therefore the ratio T1/ T 2 = 1/2π.

It is not the ratio of two integers. Thus x (t) is non periodic signal.

4. Find the fundamental period and frequency of the following signals

(a)

Let T1 &T2 be the fundamental periods of x1(t) x2(t) respectively

10
The composite signal is periodic signal. Since T 1 &T2 are rational,x(t) is periodic. The
fundamental period is the LCM of T1 &T2.

In this case, T1 &T2 are fractions; they are made integers by multiplying by a least number.
for T1&T2 thus obtained,LCM is found.T0 is obtained by dividing by the same number which
was chosen to make T1 &T2 as integers.

(1)

(2) By multiplying T1 &T2 by 36, T1 = 3 T2 = 2.

(3) The LCM for the new T1 &T2 is easily obtained

as 6. (4)T0 is obtained by dividing LCM by 36.

(b)

Where

11
Let T1 &T2 be the fundamental periods of x1(t) x2(t) respectively.

The composite signal is periodic with fundamental period is

5. Determine whether the following signal is periodic. If periodic find the

fundamentaltime period.

Solution:

12
The signal is periodic with fundamental period N=8.

6. Find whether the following signals are even or odd. Find the even and
oddcomponents.

13
7. Sketch the even and odd components of exponential signal x(t)=10 e-2t

1) Draw the signal x (t).

2) Draw its folded version x (-t).
3) Add x (t) and x (-t) or subtract x (-t) from x (t).
4) Even & odd component is calculated by,

14
8. Determine whether the following signals are even or odd.

15
9. Determine the following signals are energy or power signal?

(a) x(t)= u(t)

The unit-step function, defined by

is a power signal, since

P=1/2 watts.

16
10.Determine whether the following signals are power or energy signal.

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18
(d) X (n) = (1/3)n n>0

Elementary signals

In the analysis of communication system, standard test signals play very important
role.Such signals are used to check the performance of the system. Applying such

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signals at thesystem; the output is checked. Now depending on the input-output
characteristic of that particular system study of different properties of a system can be
done. Some standard test signals are as follows:

1) DC signal 2) Unit step signal 3) Delta or unit impulse function

4) Unit ramp signal 5) Sinusoidal signal 6) Rectangular pulse

7) Exponential signal 8) Signum function 9) Sinc function

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 1. DC signal

The amplitude of DC signal remains constant & is independent of time.

Mathematically CT is expressed as x(t) = A -< t < 


DT is expressed as x(n) = A -< n < 


2. Unit delta function δ(t)

CT unit impulse δ(t) : A continuous time delta function is denoted by δ(t). Mathematically it
isexpressed as follows:

δ(t)=1 for

t=0δ(t)=0

for t≠0

The graphical representation of delta function for C.T. signal is shown in figure

CT Unit impulse

The delta function is an extremely important function used for the analysis of
communication systems.The unit impulse, δ(t), is a function that is zero for all t ≠ 0 and for
which

21
The area under the pulse is unity. Due to its unity area it is called as a unit impulse function.

Importance of impulse function:

1. By applying impulse signal to a system one can get the impulse response of the
system. From the impulse response it is possible to get the transfer function of the system.

2. For a LTI system if the area under the impulse response curve is finite, then the
system issaid to be stable.

3. From the impulse response of the system, one can easily get the step response by
integrating it once &twice respectively.

4. It is easy to generate &apply to any system.

Properties of impulse response:

1. δ(at)= 1/a δ(t)

2. δ(-t)= δ(t)

3.
4. X(t)* δ(t) = X(t)
5. X(t) )* δ(t-t0) = X(t0)
DT unit impulse δ(n) :

A discrete time unit impulse function is denoted by δ(n). Its amplitude is 1 at n=0 and for
allother values of n; its amplitude is zero.

In the sequence form it can be represented as,

In the above sequence the arrow represents 0 th sample. The above sequence can
also bewritten as δ(n) = {1}

The graphical representation of delta function for D.T. signal is as shown in figure below:

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 DT unit impulse

3. Unit step function

CT unit step : A continuous time unit step signal is denoted by u(t). Its value is unity (1) for
all positive values of t. that means its value is one for t ≥ 0. While for other values of t, its
value is zero. Unit step function is defined as U(t )= 1 for t ≥ 0

0 Otherwise
The graphical representation of CT unit step function is as shown in figure below:

Unit step function

DT unit step sequence

A discrete time unit step signal is denoted by u(n) and all its samples have value of 1 for n
≥ 0.While for other values of n, its value is zero.

DT Unit step function is defined as

In the form of sequence it can written as,

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Graphically it can be represented as follows:

Graphical representation of DT unit step signal

The unit step and the impulse functions are related to one another by

4. Unit Ramp signal

CT unit Ramp signal: A continuous time unit ramp signal is denoted by r(t). Its value is t for
allpositive values of t. While for other values of t, its value is zero.

Unit ramp function is defined as r(t) = t for t ≥ 0

0 otherwise

The graphical representation of CT unit ramp signal is as shown in figure below:

CT unit Ramp signal

DT unit Ramp:

A discrete time unit ramp signal is denoted by r(n). Its value increases linearly with
samplenumber n. mathematically it is defined as,

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 From above equation, it is clear that the value of signal at a particular interval is equal to
thenumber of interval at that instant.

Graphically it is represented in figure below:

DT unit Ramp sequence

Relationship between unit impulse, step and ramp signal

1. Integrating the unit step signal we get,

By integrating the unit step function, unit ramp function is obtained. In the reverse
process, bydifferentiating unit ramp function, the unit step function is obtained.

2. The continuous time unit step function is the running integral of unit impulse function
which isexpressed as,

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3. By differentiating the ramp function twice, the impulse function is obtained.

Thus impulse function is obtained by differentiating the ramp function twice. In reverse
process, By integrating unit impulse function twice, the ramp function is obtained which is
mathematically expressed as follows:

The relationship between unit step, impulse and ramp signals are represented below

5. Signum function (sgn(t))

The CT signum function is defined as

The graphical representation of CT signum function is as shown in figure below:

26
 Representation of unit Signum function

Signum function is an odd function.

The relation between sgn(t) and u(t) as follows: Sgn(t) = -1+ 2 u(t)

Sgn(t) = u(t)- u(-t)

DT Signum function

The DT signum function is defined as Sgn(n) = 1 for n > 0

-1 for n < 0

6. Sinusoidal signal

Continuous-Time Sinusoidal Signals: A simple harmonic oscillation is

mathematically described by the following continuous-time sinusoidal signal as
shown in figure.

The subscript a used with x (t) denotes an analog signal. This signal is
completelycharacterized by three parameters:
A is the amplitude of the sinusoid.
Ω is the frequency in radian s per second
(rad/s),Ө is the phase in radians.
Instead of Ω, we often use the frequency F in cycles per second or hertz (Hz), where

Xa(t) can be written as

We will use both forms for representing sinusoidal signals.

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 Analog sinusoidal
signalDiscrete-Time Sinusoidal Signals
A discrete-time sinusoidal signal may be ex pressed as

Where n is an integer variable, called the sample number.

A is the amplitude of the sinusoid,

ώ is the frequency in radians per sample,

and Ө is the phase in radians.

If instead of ώ we use the frequency variable f defined by

The x(n) relation becomes

Discrete-time sinusoidal signal

In contrast to continuous time sinusoids, the discrete time sinusoids are characterized by
thefollowing properties:

A discrete-time sinusoid is periodic only if its frequency f is a rational number.

By definition, a discrete time signal x(n) is periodic with period N ( N > 0) if and only if

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The smallest value of N for which the above is true is called the fundamental period.
The proof of the periodicity property is simple. For a sinusoid with frequency to be periodic,
we should have

This relation is true if an d only if there exists an integer k such that

According to the above discrete-time sinusoidal signal is periodic only if its frequency f
can beexpressed as the ratio of two integers (i.e.f0 is rational).

7. Rectangular pulse

The Rectangular signal is having constant amplitude A for the time interval between –
T/2 to T/2. Mathematically it is defined as , A rect(t/T) = A for –T/2 <t< T/2

= 0 Otherwise

The graphical representation of CT Rectangular function is as shown in figure below:

CT rectangular pulse

DT Rectangular signal is defined as

The above equation can also be written as,

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 DT rectangular sequence

8. SINC Function

Sinc function is defined as follows:

Sinc(t) = 1 at t=0 and Sinc(t) = 0 at t =1,-1,2,-2,3,-3 …..

Representation of SINC Function

9. CT complex exponential signal

The CT complex exponential signal is of the following form x(t) =c eαt

Where c,α are parameters.

Types of CT complex exponential signal:

Depending on the type of c,α, the complex exponential signal can be classified as,

1. Real exponential signal: If c,α are real.

a) Decaying exponential x(t)= e-αt b) Rising exponential x(t)= eαt

Decaying exponential
signal Rising

30
2. Periodic complex exponential signal.

The complex exponential signal is described as X(t)=ejw t0

The most important property of this signal is that it is a periodic signal.

Discrete time exponential signal

A discrete time exponential signal is expressed as,

Here ‘a’ is a real constant. Depending upon the value of ‘a’ we have four different cases:

Case 1: if a > 1, then x (n) becomes the rising exponential sequence.

Case 2: if 0 < a < 1, then x (n) becomes decaying exponential sequence.

Case 3: if a < -1, then x (n) becomes double sided rising exponential sequence.

Case 4: if -1 < a < 0, then x (n) becomes double sided decaying exponential sequence

Rising exponential sequence Decaying exponential sequence

Double sided rising exponential sequence Double sided decaying exponential sequence

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The exponential sequence can be real or complex valued. If ‘a’ is complex valued then it
can berepresented as,

Here r is the magnitude of ‘a’ & θ is the phase of ‘a’. Hence the sequence x(n) becomes

Using Euler’s identity, the above equation becomes,

Thus each sample of sequence x(n) has real and imaginary part.ie

Similarly we can write magnitude and phase of x (n) as,

1.Prove the following:

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The right side of above equation is unit impulse sequence u(n). Hence the given equation is
proved.

The right side of above equation is unit impulse sequence u (n). Hence the given equation is
proved.
Basic operations on the signal
The basic operations performed on the signal are

1) Amplitude scaling
Amplitude scaling means changing an amplitude of given continuous time signal. We will
denote continuous time signal by x(t). If it is multiplied by some constant ‘B’ then resulting
signal is,
y(t)= B x(t)
Example: Sketch y(t) =
5u(t)
Solution: we know that u(t) is unit step function. So if we multiply it with 5, its amplitude
willbecome 5 and it shown as follows:

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2) Sum and difference of two signals:
Consider two signals x1(t) and x2(t). Then addition of these signals is denoted by
y(t)=x1(t)+x2(t).Similarly subtraction is given by y(t)=x1(t)-x2(t).
Example: Sketch y(t) = u(t) – u(t – 2)
Solution: First, plot each of the portions of this signal
separatelyx1(t) = u(t) …….Simply a step signal
x2(t) = –u(t-2) ……. Delayed step signal by 2 units and multiplied by
-1.Then, move from one side to the other, and add their instantaneous
values:

3) Product of two signals:

If x1(t) and x2(t) are two continuous signals then the product of x1(t) and x2(t) is,Y(t) = x1(t) x2(t).
Example: Sketch y(t) = u(t)·u(t – 2)
Solution: First, plot each of the portions of this signal
separatelyx1(t) = u(t) Simply a step signal
x2(t) = u(t-2) Delayed step signal
Then, move from one side to the other, and multiply instantaneous values:

Operations performed on independent variables:

1) Time shifting:
A signal x(t) is said to be ‘shifted in time’ if we replace t by (t-T). thus x(t-T) represents the
time shifted version of x(t) and the amount of time shift is ‘T’ sec. if T is positive then the
shift is to right (delay) and if T is negative then the shift is to the left (advance).
Example: Sketch y(t) = u(t – 2)

34
2) Time scaling:
The compression or expression of a signal in time is known as the time scaling. If x(t) is
theoriginal signal then x(at) represents its time scaled version. Where a is constant.
 If a> 1 then x(at) will be a compressed version of x(t) and
 if a< 1 then it will be a expanded version of x(t).
Example: Let x(t) = u(t) – u(t – 2). Sketch y(t) = x(t/2)

3) Time reversal (Time inversion):

Flips the signal about the y axis. y (t) = x(-t) .
Consider the signal x (t) shown in figure (a). The signal x (-t) is obtained by putting a
mirror along the vertical axis. The signal to the right of the vertical axis gets reflected to
the left andvice versa. Alternatively, if we make a folding across the vertical axis, the
signal in the right of the axis is printed in left and vice versa. The signal so obtained is x (-

t).
Original signal folded version x(-t)

Q. Consider the triangular waveform x (t) shown in figure. Sketch the

followingwaveforms. (a) x(2t+3) (b) x [(t+3)/2] (c) x (t/2-3) (d) x(-2t+3) (e) x
(-2t-3)

35
 (a) To sketch x(2t+3)
Figure (a) shows x (t) = tri(t).by shifting by t=3 towards left,x(t+3) is obtained and this is
sketched in figure (b).x(t+3) is time compressed by a factor to get x(2t+3).this is sketched in
figure(c).

(b)To sketch x [(t+3)/2]

The signal x [(t+3)/2] is written as x(t/2+1.5). The signal x (t) is time shifted to the left by
1.5unit to get x(t+1.5) which is nothing but x[(t+3)/2].This is sketched in figure(e).

36
(c) To sketch x (t/2-3)

X(t-3) is obtained from x(t) by shifting the signal x(t) to the right by 3 unit and is shown
in figure(f).by time expansion of x(t-3) by a factor 2, x(t/2-3) is obtained and sketched as
shown infigure (g).

(d) To sketch x(-2t+3):

Signal x(-t) is obtained by folding and it is shown in figure (h).x(-t) is time shifted to the
rightby 3 unit to get x(-t+3).this is shown in figure(i).the signal x(-t+3) is time compressed by
a factor 2 to get x(-2t+3).

(e) To sketch x (-2t-3)

X(-t) is time shifted towards left by 3 units to get x(-t-3).this is shown in figure(k).x(t-
3) istime compressed by a factor 2 to get x(-2t-3).this is sketched in figure (l)

37
Systems Definition:

A system is a physical device (or an algorithm) which performs required operation on

a discrete time signal. Alternatively a system is defined as a set of elements or functional
blocksthat are connected together and produces an output in response to an input signal.
Eg: An audio amplifier, attenuator, TV set etc. The block diagram of CT/DT system is
shown infigure below.

CT/DT system

Here x(t) (or) x(n) is the input signal applied to the system. It is also called as excitation.

Output of the system is denoted by y(t) (or) y(n). It is also called as response of the system.

Example: A filter is good example of a system. A signal containing noise is applied to the
input of the filter. This is an input signal to the system. The filter cancels or attenuates noise
signal. This is the processing of the signal. A noise-free signal obtained at the output of the
filter is called as response of the system.

Classification of systems

Generally systems are broadly classified into two categories, such as continuous
time system(CT) and discrete time system(DT), depending upon the type of given input
to the system.

CT system: if the input and output signals x(t) & y(t) are continuous time signals, then
thesystem is continuous time system. The output y(t)= T[ x(t)]

38
 DT system: if the input and output signals x(n) & y(n) are discrete time signals, then
thesystem is discrete time system. The output y(n)= T[ x(n)]

Both CT System & DT system are classified into the following categories:

 Linear and Non-linear Systems

 Time Variant and Time Invariant Systems
 Linear Time variant and Linear Time invariant systems
 Static and Dynamic Systems
 Causal and Non-causal Systems
 Invertible and Non-Invertible Systems
 Stable and Unstable Systems
(a) Linear and Non-linear Systems
Linear Systems: A system is said to be linear when it satisfies superposition and
homogenate principles. Consider two systems with inputs as x 1(t), x2(t), and outputs as
y1(t), y2(t) respectively. Then, according to the superposition and homogenate principles,

T [a1 x1(t) + a2 x2(t)] = a1 T[x1(t)] + a2 T[x2(t)]

∴, T [a1 x1(t) + a2 x2(t)] = a1 y1(t) + a2 y2(t)

From the above expression, is clear that response of overall system is equal to
response ofindividual system.

Communication channels and filters are example of linear systems.

Non-linear Systems: If the system does not satisfy the superposition theorem, then it is
saidto be a nonlinear system.
How to determine whether the given system is Linear or not?
To determine whether the given system is Linear or not, we have to follow the following
steps: Step 1: Apply zero input and check the output. If the output is zero then the system
is linear. If this step is satisfied then follow the remaining steps.
Step 2: Apply individual inputs to the system and determine corresponding outputs. Then
addall outputs. Denote this addition by y’(n). This is the R.H.S. of the 1st equation.
Step 3: Combine all inputs. Apply it to the system and find out y”(n). This is L.H.S. of
equation(1).
Step 4: if y’(n) = y”(n) then the system is linear otherwise it is non-linear system.

1. Determine whether the following system is linear or not?

(a) Y(n)= n x(n)

Solution:
Step 1: When input x(n) is zero then output is also zero. Here first step is satisfied so we
willcheck remaining steps for linearity.
Step 2: Let us consider two inputs x1(n) and x2(n) be the two inputs which produces
outputsy1(t) and y2(t) respectively. It is given as follows:

39
Now add these two output to get y’(n)
Therefore y’(n) = y1(n) + y2(n) = n x1(n) + n
x2(n)Therefore y’(n) = n [x1(n) + x2(n)]
Step 3: Now add x1(n) and x2(n) and apply this input to the system.

Therefore We know that the function of system is to multiply input by ‘n’.

Here [x1(n) + x2(n)] acts as one input to the system. So the corresponding output is,
y”(n) = n [x1(n) + x2(n)]
Step 4: Compare y’(n) and y”(n).
Here y’(n) = y”(n). Hence the given system is linear.
(b) y(t) = x2(t)

Solution: y1 (t) = T[x1(t)] = x12(t)

y2 (t) = T[x2(t)] = x22(t)

T [a1 x1(t) + a2 x2(t)] = [ a1 x1(t) + a2 x2(t)]2

Which is not equal to a1 y1(t) + a2 y2(t). Hence the system is said to be non linear.
(b) Time Variant and Time Invariant Systems
A system is said to be Time Invariant if its input output characteristics do not change
with time. Otherwise it is said to be Time Variant system.
Explanation:
As already mentioned time invariant systems are those systems whose input output
characteristics do not change with time shifting. Let us consider x(n) be the input to the
systemwhich produces output y(n). Now delay input by k samples, it means our new input
will become x(n-k). Now apply this delayed input x(n-k) to the same system as shown in
figure below.
Now if the output of this system also delayed by k samples (i.e. if output is equal to y(n-k))
thenthis system is said to be Time invariant (or shift invariant) system.
If we observe carefully, x(n) is the initial input to the system which gives output y(n), if we
delayed input by k samples output is also delayed by same (k) samples. Thus we can say
that input output characteristics of the system do not change with time. Hence it is Time
invariant system.
Now let us discuss about How to determine that the given system is Time invariant or
not?To determine whether the given system is Time Invariant or Time Variant, we have to
follow the following steps:

40
Step 1: Delay the input x(n) by k samples i.e. x(n-k). Denote the corresponding
output byy(n,k).That means x(n-k) → y(n,k)
Step 2: In the given equation of system y(n) replace ‘n’ by ‘n-k’ throughout. Thus the
output isy(n-k).
Step 3: If y(n,k) = y(n-k) then the system is time invariant (TIV) and if y(n,k) ≠ y(n-k)
then system is time variant (TV).
Same steps are applicable for the continuous time systems.

1) Determine whether the following system is time invariant or not. y(n) = x(n) – x(n-2)
Solution:
Step 1: Delay the input by ‘k’ samples and denote the output by
y(n,k)Therefore y(n,k) = x(n-k) – x(n-2-k)
Step 2: Replace ‘n’ by ‘n-k’ throughout the given equation.
Therefore y(n-k) = x(n-k) – x(n-k-2)
Step 3: Compare above two equations. Here y(n,k) = y(n-k).
Thus the system is Time Invariant.
2) Determine whether the following systems are time invariant or not?y(n) = x(n) + n x(n-2)
Solution:
Step 1: Delay the input by ‘k’ samples and denote the output by
y(n,k)Therefore y(n,k) = x(n-k) + n x(n-k-2)
Step 2: Replace ‘n’ by ‘n-k’ throughout the given equation.
Therefore y(n-k) = x(n-k) + (n-k) x(n-k-2)
Step 3: Compare above two equations. Here y(n,k) ≠ y(n-k).
Thus the system is Time Variant.
3) Determine whether the following systems are time invariant or not? y(n) = x(-n)

Step 1: Delay the input by ‘k’ samples and denote the output by
y(n,k)y(n, k) = T[x(n-k)] = x(-n-k)

Step 2: Replace ‘n’ by ‘n-k’ throughout the given

equationy(n-k) = x(-(n-k)) = x(-n + k)

Step 3: Compare above two equations. Here y(n,k) ≠ y(n-k).

Thus the system is Time Variant.

(c) Linear Time variant (LTV) and Linear Time Invariant (LTI) Systems
If the system satisfies both linearity and time variant property, then it is called linear
time variant (LTV) system.

If the system satisfies both linearity and time Invariant then that system is called liner
time invariant (LTI) system.

(d) Static (memory less) and Dynamic system(system with memory )

Static system: A system is said to be static or memory less if its output depends upon the
present input only.

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Why static systems are memory less systems?
Observe the input output relations of static system. Output does not depend on
delayed [x(n-k)] or advanced [x (n+k)] input signals. It only depends on present input (nth)
input signal. Ifoutput depends upon delayed input signals then such signals should be
stored in memory to calculate the output at nth instant. This is not required in static systems.
Thus for static systems, memory is not required. Therefore static systems are memory less
systems.
Dynamic systems:
Definition: It is a system in which output at any instant of time depends on input sample at
thesame time as well as at other times.
Here other time means, other than the present time instant. It may be past time or future
time. Note that if x (n) represents input signal at present instant then,
1) x (n-k); that means delayed input signal is called as past signal.
2) x (n+k); that means advanced input signal is called as future signal.
Thus in dynamic systems, output depends on present input as well as past or future inputs.
Why dynamic system has a memory?
Observe input output relations of dynamic system. Since output depends on past or future
input sample; we need a memory to store such samples. Thus dynamic system has a
memory.
1) Determine whether the following systems are static or dynamic?

a) y(t) = 2 x(t)

For present value t=0, the system output is y(0) = 2x(0). Here, the output is only
dependentupon present input. Hence the system is memory less or static.

b) y(t) = 2 x(t) + 3 x(t-3)

For present value t=0, the system output is y(0) = 2x(0) + 3x(-3).

Here x(-3) is past value for the present input for which the system requires memory to get
thisoutput. Hence, the system is a dynamic system.

b) y (n) = 9x(n)
In this example 9 is constant which multiplies input x(n). But output at nth instant that
means y (n) depends on the input at the same (nth) time instant x(n). So this is static
system.d) y (n) = x2(n) + 8x(n) + 17
Here also output at nth instant, y (n) depends on the input at nth instant. So this is
staticsystem.
e) y (n) = x(n) + 6x(n-2)
Here output at nth instant depends on input at n th instant, x(n) as well as (n-2)th instant x(n-
2)is previous sample. So the system is dynamic.

f) y (n) = 4x(n+7) + x(n)

Here x (n+7) indicates advanced version of input sample that means it is future
sampletherefore this is dynamic system.

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(e) Causal and non-Causal systems
A system is said to be causal if its output depends upon present and past inputs, and
doesnot depend upon future input.

For non causal system, the output depends upon future inputs also.

Example 1: y (n) = 2 x (t) + 3 x(t-3)

For present value t=1, the system output is y (1) = 2x(1) + 3x(-2).Here, the system
output only depends upon present and past inputs. Hence, the system is causal.

Example 2: y(n) = 2 x(t) + 3 x(t-3) + 6x(t + 3)

For present value t=1, the system output is y(1) = 2x(1) + 3x(-2) + 6x(4) Here, the
systemoutput depends upon future input. Hence the system is non-causal system. .

(f) Stable and unstable systems

The system is said to be stable only when the output is bounded for bounded input.
For abounded input, if the output is unbounded in the system then it is said to be unstable.

BIBO stable system: Any relaxed systems is said to be bounded input – output (BIBO)
stable if and only if every bounded input yields a bounded outputs.

Here we will see how to determine whether the system is stable or unstable i.e. stability
property. To define stability of a system we will use the term ‘BIBO’. It stands for Bounded
InputBounded Output. The meaning of word ‘bounded’ is some finite value. So bounded
input meansinput signal is having some finite value. i.e. input signal is not infinite. Similarly
bounded output means, the output signal attains some finite value i.e. the output is not
reaching to infinite level.

Mathematical representation:

Let us consider some finite number Mx whose value is less than infinite. That means Mx < ∞,
so it’s a finite value. Then if input is bounded, we can write,
|x(n)| ≤ Mx < ∞
Similarly for C.T. system
|x(t)| ≤ Mx < ∞
Similarly consider some finite number My whose value is less than infinity. That means M y
< ∞,so it’s a finite value. Then if output is bounded, we can write,

|y(n)| ≤ My < ∞
Similarly for continuous time system

|y(t)| ≤ My < ∞

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Definition of Unstable system:

An initially system is said to be unstable if bounded input produces unbounded (infinite) output.

Significance: Unstable system shows erratic and extreme behavior.

When unstable system is practically implemented then it causes overflow.

Determine whether the following discrete time functions are stable or not.

1) y(n) = x(-n)

Solution: we have to check the stability of the system by applying bounded input. That
means the value of x(-n) should be finite. So when input is bounded output will be
bounded. Thus thegiven function is Stable system.

2) y (t) = x2(t)

Let the input is u (t) (unit step bounded input) then the output y(t) = u2(t) = u(t) =
bounded output. Hence, the system is stable.

3) y (t) = ∫ x(t) dt
Let the input is u (t) (unit step bounded input) then the output y(t) = ∫u(t)dt = ramp signal
(unbounded because amplitude of ramp is not finite it goes to infinite when t → infinite).
Hence, the system is unstable.
(g) Invertible and noninvertible system

A system is said to invertible if the input of the system appears at the output.

Y(S) = X(S) H1(S) H2(S)

= X(S) H1(S) · 1(H1(S)) Since H2(S) = 1/( H1(S) )

∴ Y(S) = X(S)
→ y(t) = x(t)
Hence, the system is
invertible.

If y(t) ≠ x(t), then the system is said to be non-invertible.

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Solved Problems
1. Determine whether the system described by the following input output equations are linear,
time invariant, stable, memory less and casual or not.
(1) y(t)=cos[x(t)]
(i) Memory less / system with memory
A system is said to be static or memory less if its output depends upon the present input

only.at time t=0; y(0)= cos[x(0)] ; the present output y(0) depends only on present input

x(0).

at time t=1; y(1)= cos[x(1)] ; the present output y(1) depends only on present input x(1).

From the above equations, we can say that the present output is only dependent upon
presentinput. Hence the system is memory less or static system.

(ii) Casual / non casual system

A system is said to be causal if its output depends upon present and past inputs, and
doesnot depend upon future input.

For present value t=1, the system output is y(1)= cos[x(1)] ;.Here, the system output only
depends upon present inputs. So the system is a causal system.

(iii) Time variant/ time invariant

A system is said to be time variant if its input and output characteristics vary with time.
Otherwise, the system is considered as time invariant.

The condition for time invariant system is : y (t , T) = y(t-T)The condition for time variant

system is : y (t , T) ≠ y(t-T)

y (t , T) = cos[x(t-T)]

y(t-T) = cos[x(t-T)]

∴ y (t , T) = y(t-T) . Hence, the system is time invariant.

(iv) Stable
For the bounded input the system produces bounded output. So the system is said to
be stable.

(v) linear / nonlinear

A system is said to be linear when it satisfies superposition and homogenate principles.

Consider two systems with inputs as x 1(n), x2(n), and outputs as y1(n), y2(n) respectively.
Then according to the superposition and homogenate principles,

45
 T [a1 x1(t) + a2 x2(t)] = a1 T[x1(t)] + a2 T[x2(t)]

y1 (t) = T[x1(t)] = cos[x1(t)]

y2(t) = T[x2(t)] = cos[x2(t)]
a1 T[x1(t)] + a2 T[x2(t)] = a1 cos[x1(t)] + a2 cos[x2(t)] .................. (1)

T [a1 x1(t) + a2 x2(t)] = cos[a1x1(t)+ a2x2(t)]....................... (2)

(1) ≠ (2)
Comparing the above two equations it does not satisfy the superposition property. So
thesystem is said to be non linear.
(2) y(n)= x(-n)
(i) Memory less / system with memory
A system is said to be static or memory less if its output depends upon the present input

only.at time n=0; y(0)=x(0); the present output y(0) depends only on present input x(0).

at time n=1; y(1)=x(-1); the present output y(1) depends only on past input x(-1).

at time n=-1; y(-1)=x(1); the present output y(-1) depends only on future input x(1).

From the above equations, we can say that the present output is not only depends
uponpresent input. Hence the system is said to be dynamic system.
(ii) Casual / non casual system
A system is said to be causal if its output depends upon present and past inputs, and does
notdepend upon future input. Otherwise it is non casual.

at time n=0; y(0)=x(0); the present output y(0) depends only on present input

x(0).at time n=1; y(1)=x(-1); the present output y(1) depends only on past

input x(-1). at time n=-1; y(-1)=x(1); the present output y(-1) depends only on

future input x(1).

Here, the system output depends upon present input, past and future inputs. So the
system issaid to be non causal system.

(iii) Time variant/ Time invariant

A system is said to be time variant if its input and output characteristics vary with
time. Otherwise, the system is considered as time invariant.

The condition for time invariant system is : y (n , k) = y(n-k)The condition for time variant

system is : y (n , k) ≠ y(n-k)

y(n, k) = T[x(n-k)] = x(-n-k)

y(n-k) = x(-(n-k))=x(-n+k)

∴ y(n, k) ≠ y(n-k). Hence, the system is time variant.

46
(iv) Stable/unstable
For the bounded input this system produces bounded output. So the system is
saidto be stable.
(v) Linear / nonlinear

A system is said to be linear when it satisfies superposition and homogenate principles.

Consider two systems with inputs as x 1(n), x2(n), and outputs as y1(n), y2(n) respectively.
Then according to the superposition and homogenate principles,
T [a1 x1(n) + a2 x2(n] = a1 T[x1(n)] + a2 T[x2(n)]
y1 (n) = T[x1(n)] = x 1(-n)
y2(n) = T[x2(n)] = x 2(-n)
a1 T[x1(n)] + a2 T[x2(n)] = a1 x 1(-n)+ a2 x 2(-n) .............. (1)

T [a1 x1(n) + a2 x2(n] = [a1 x 1(-n)+ a2 x 2(-n)]............. (2)

Comparing the above two equations it satisfy the superposition property. So the system is
saidto be linear.
(3) y(t)= t x(t)
(i) Memory less / system with memory

A system is said to be static or memory less if its output depends upon the present input only.At

time t=0; y(0)= 0 x(0) ; the present output y(0) depends only on present input x(0).

At time t=1; y(1)= x(1) ; the present output y(1) depends only on present input x(1).

From the above equations, we can say that the present output is only dependent upon
presentinput. Hence the system is memory less or static system.

(ii) Casual / non casual system

A system is said to be causal if its output depends upon present and past inputs, and does
notdepend upon future input.

For present value t=1, the system output is y(1)= x(1) ;.Here, the system output only
dependsupon present inputs. So the system is a causal system.

(iii) Time variant/ time invariant

A system is said to be time variant if its input and output characteristics vary with
time. Otherwise, the system is considered as time invariant.
The condition for time invariant system is : y (t , T) = y(t-T)

The condition for time variant system is : y (t , T) ≠ y(t-T)

y (t , T) = t x(t-T) y(t-T) = (t-T) x(t-T)

∴ y (t , T) ≠ y(t-T) . Hence, the system is time variant.

47
(iv) Stable
For the bounded input the system produces bounded output. So the system is said
to be stable.
(v) Linear / nonlinear

A system is said to be linear when it satisfies superposition and homogenate principles.

Consider two systems with inputs as x1(t), x2(t), and outputs as y1(t), y2(t) respectively. Then
according to the superposition and homogenate principles,

T [ x1(t) + x2(t)] = T[x1(t)] + T[x2(t)]

y1 (t) = T[x1(t)] = t x1(t)

y2(t) = T[x2(t)] = t x2(t)

T[x1(t)] + T[x2(t)]= t x1(t)+ t x2(t)............... (1)

T [ x1(t) + x2(t)] = t [x1(t)+ x2(t)] ………..(2) (1) = (2)

Comparing the above two equations it satisfy the superposition property. So the system is
saidto be linear.

2. Determine whether the following systems are static, casual, time invariant, linear,
andstable.

48
 4. The input is shifted and time compressed signal. As long as the input is bounded the
output isalso bounded.

58

The representation of signal with respect to time is called as its time domain representation. The time
domain representation is not sufficient for its analysis. Hence we haveto use the frequency domain

representation of the signal. The signal represented in frequency domain is called as the line spectrum.
The line spectrum consists of two graphs namely:

1) Frequency spectrum: A graph of amplitude VS frequency.

2) Phase spectrum: A graph of phase VS frequency.
The time domain representation gives us the following information:
 Shape of the signal
 Its frequency
 Type of signal
 One cycle period
But we cannot know anything about what frequency components are present. All this
information can be obtained from the line spectrum of a signal. Line spectrum can be
obtained by using either Fourier series or Fourier transform. Line spectrum enables us to
analyze and synthesis a signal.
49
 How to plot line spectrum?

Line spectrum is useful in understanding the existence and amplitude/ phases of

variousfrequency components present in a waveform.

1) In all spectral drawing the independent variable plotted on the x axis is frequency f.

2) Phase angle is always measured with respect to cosine wave’s .hence if it

necessary toconvert sine waves to cosine waves using the following standard identity

Sin wt = cos (wt-90)

3) The amplitude is always regarded as positive quantity. So if negative sign appears
theyshould be absorbed in the phase change to keep amplitude positive.

-A cos wt = A cos (wt+-180)

Thus additional phase change of 180 converts the negative amplitude –A to positive amplitude
+A.

PART-A QUESTIONS

1. What are the classifications of signals?

 Continuous and discrete time signals
 Periodic and non-periodic signals
 Even and odd signals
 Energy and power signals
 Deterministic and random signals
2. Give examples of con continuous time signals.
Ans. sine wave, cosine wave, triangular wave etc.
3. Define Even and Odd Signals.
Ans. A signal x(t) is said to be symmetrical (or) even if it satisfies the following
condition:X(t) = x(-t) for CT signal
X(n) = x(-n) for DT signal

A signal x(t) is said to be anti symmetrical (or) odd if it satisfies the following
condition:X(t) = -x(-t) for CT signal
4. Differentiate between Signal & System.
Ans. A signal is a description of how one parameter varies with another parameter. For
instance, voltage changing over time in an electronic circuit, or brightness varying with distance
in an image. A system is any process that produces an output signal in response to an input
signal.

50
PART-B QUESTIONS

(1) With regard to Fourier series representation justify the following statements:
Odd functions have only sine terms.

Functions with half wave symmetry have only odd harmonics.

Even functions have no sine terms

(2) Approximate the function described below by a wave form over the interval
(0, 2).The function is:

Also sketch the original function and approximated function.

(3) Find Convolution integral for x1(t) = t2 u(t) & x2(t) = e-tu(t) ?

(4) Determine complex exponential Fourier series representation for the following signals

(i) x(t) = cos(2t+π/4) (ii) x(t) = cos4t+sin6t (iii) x(t) = sin2(t)

(5) Derive the relations between exponential and trigonometric fourier series

coefficients.(6)Explain about Dirichlet’s conditions.

(7) Determine the trigonometric Fourier series of the half rectified

waveshown below: f (t)

T
T/2 t
(8)Write short notes on exponential Fourier spectrum.

(9) Verify the following signals t and sin m t are the orthogonal or not over
theinterval
(t0, t0 + 2/ ).

(10) Explain why mean square error is preferred in signal approximations. (11)Express the Impulse
function in terms of sampling function and explain.
(12) Discuss the analogy between vectors and signals. Explain orthogonal vector space
andorthogonal signal spaces.

51
(13)Explain the condition of Orthogonality between two signal f1(t) & f2(t).

(14) Show that the functions Sin(nwot) and Sin(mwot) are orthogonal to each other for
allinteger values of m and n.

(15)Find the exponential Fourier series and plot the magnitude and phase spectrum for
thetriangular waveform shown in figure.

f(t)

-4 -3 -2 - 1 0 1 2 3 4
(16)Expand following function f(t) by exponential Fourier series over the interval (0,1). In
thisinterval f(t) is expressed as f(t) = At.
(17) Define a system. How the systems are classified? Define any four systems with examples?

(18) Sketch the following signals

(a)Y(t) = π(0.5t-1)+π(2t-3.5) (b)Y(t) = 3u(t)+2sin2t (c)Y(t)=u(t)+u(t-2)-3u(t-5)+u(t-7)

(19) (a) Define orthogonal subspace.

(b) What is orthonormal vector and orthonormal set of vectors.
(c) Prove that the complex exponential functions are orthogonal
functions.(20)Define and sketch the following signals:
(i) Impulse function. (ii) Unit step function. (iii) Ramp function. (iv) Signum function.
(21) Express discrete impulse function in terms of unit step function. Also show
graphicalillusion.
(22) Describe BIBO stability of a system.(2M)
(23) Find which of the following signals are causal or non – causal:
(i) x(t) = e2t u(t – 1). (ii) x(t) = cos 2t. (iii) x(t) = 2 u(-t). (iv) x(n) =
u(-n).(v) x(n) = u(n + 4) – u(n – 2).

(24) Check whether the following systems are:

(i) Static or dynamic. (ii) Linear on non-linear. (iii) Causal or non-causal.
(iv) Time invariant or time variant. (v) Stable on not stable. The given system is y(n) = an u(n).

52
 UNIT II
FOURIER SERIES & FOURIER TRANSFORM
Continuous time Fourier series
Fourier series represents a periodic waveform in the form of sum of infinite
number of sine and cosine terms. It is a representation of a signal in a time domain series
form. Fourier series is a tool used to analyze any periodic signal. After the analysis we
obtain the followinginformation about the signal.
1) What are the freq. components are present in the signal.
2) Their amplitudes.
3) The relative phase difference between these frequency
components. Types of Fourier series:
1. Trigonometric (or) Quadrature Fourier series
2. Polar Fourier series(or) cosine Fourier series
3. Exponential Fourier series
Trigonometric (or) Quadrature Fourier series
Consider any arbitrary continuous time signal x(t).This arbitrary signal can be split up
as sine and cosines of fundamental frequency w0 and all of its harmonics and expressed as
givenbelow.

In above equation, a0, corresponds to the zeroth harmonic or DC. The expression for the
constant term a0 and the amplitudes of the harmonic can be derived as,

T0 - fundamental period of x(t) in seconds

F0 - fundamental frequency in Hz W 0 - radian frequency in rad/sec
53
Exponential Fourier series
By using Euler’s identity, the complex sinusoids can always be expressed in
terms ofexponentials. Thus the trigonometric Fourier series can be represented as

where

The above equation represents exponential Fourier series and Dn is the coefficient of the
exponential Fourier series. The coefficient Dn is related to trigonometric Fourier series
coefficients an,bn as

Polar Fourier series (or) Cosine Fourier series

In cosine polar series any periodic signal x(t) can be expressed as follows,

The coefficients of compact form Fourier series and exponential form Fourier series are
relatedas

54
1. Find the trigonometric Fourier series of the periodic signal as shown in figure below.

1. From the figure, it is evident that the waveform is symmetrical with respect to the axis t.

So a0=0.

2. By folding x(t) across the vertical axis, it is observed that x(t)=x(-t) which shows that
thefunction of the signal is even. Hence bn=0.

3. From figure it is obtained that the fundamental period T 0 = 4 sec and the fundamental
radianfrequency w0= π/2 rad/sec.

The trigonometric Fourier series is written as

55
The given signal is expressed as

The signal is symmetrical with respect to time axis and hence a 0=0.Also from figure,
it isevident that x(t) = x(-t) and therefore the signal is an even signal so bn=0.

Substituting a0=0 & bn=0 in x(t) equation, we get

The signal x(t) can be expanded as follows

56
2. For the periodic signal shown in figure, determine the trigonometric Fourier series.

1. From figure T0 = 2 sec and the fundamental radian frequency w0 = π rad/sec. The signal
is symmetrical with respect to time axis and hence a 0 = 0.Also from figure, it is evident that
x(t)=-x(-t) and therefore the signal is an odd signal and a n=0.The Fourier series for such a
signal is therefore

2. The coefficient bn is determined as follows:

The above integral is solved using the infinite integral

57
3. Find the trigonometric Fourier series of the periodic signal.

Solution:

1. From figure T0 = 2π sec and the fundamental radian frequency w 0= 1 rad/sec.The signal
isneither odd nor even. Further it is not symmetrical with respect to the time axis. So the
coefficients, a0,an & bn are to be evaluated.

58
59
4. Find the exponential Fourier series of half wave rectified sine wave as shown in
figure. Also plot magnitude and phase spectrum.

Solution: The given signal can be expressed as,

The Fourier series coefficient can be calculated as,

60
With these consideration, cn becomes

61
 5. Find the exponential Fourier series of unit impulse train as shown in figure. Also
plotmagnitude and phase spectrum.

Solution: The unit impulse train can be written as,

Cn is calculated by,

Assuming that only one impulse is present and T0 tends to infinity.

The above equation is arranged for use of sifting property of delta

function. The shift property is given as,

62
Parseval’s power theorem

It states that the total average power of the periodic signal x (t), is equal to the sum of
theaverage powers of its phasor components.

Proof:

The total average normalized power of x(t) is given as,

Exponential Fourier series is given as,

63
Therefore Fourier series for x*(t) will be,

Rearranging the above equation in terms of the order of summation and integration

In the above equation

Therefore above equation will be,

Thus Parseval’s theorem is proved.

64
 Properties of Fourier series:

Note: In Trigonometric Fourier series

 Odd functions[x(-t) =-x(t)] contains only Sine terms i.e. bn Coefficients.

 Even functions[x(-t) =x(t)] contains only Constant and Cos terms i.e.
a0 ,an Coefficients.
 Half wave Symmetry[ x(t)= -x(t ± T)] contains only odd harmonics i.e.
an, bn are defined for n is odd and a0,an,bn are zero for n is even

Fourier transform

The Fourier transform of any signal x(t) is given by

The inverse Fourier transform is calculated using the formula

Fourier transform and inverse Fourier transform can be shown as below

Fourier transform X(f) is the complex function of frequency f. Therefore it can be

expressed in the complex exponential form as follows,
65
 Here ІX(f)І is amplitude spectrum θ(f) is the phase spectrum. We know that for a real
valued signal we can write,

The following conditions should be satisfied by the signal to obtain its Fourier transform.

(i) The function x (t) should be single valued in any finite time interval T.
(ii) The function x (t) should have at the most finite number of discontinuities in any
finitetime interval T.
(iii) The function x (t) should have finite number of maxima and minima in any finite
timeinterval T.
(iv) The function x(t) should be absolutely integrable i.e.

The above conditions are applied to periodic as well as non periodic signals. The
same conditions are also called Dirichlet’s conditions. These conditions are sufficient but
not necessary for the signal to be transformable. Physically realizable signal is always
Fourier transformable. Thus physical realizability is the sufficient condition for the
existence of Fouriertransform. We know that for all energy signals,

Properties of Fourier transform

The Fourier transform possesses the following properties and using the same results
areeasily obtained .these properties are:

66
1. Linearity

2. Time scaling

3. Duality (or) symmetry

4. Time shifting

5. Frequency shifting

6. Area under x(t)

7. Area under X(f)

8. Differentiation in time domain

9. Integration in time domain

10. Conjugate

11. Multiplication in time domain

12. Convolution theorem

67
68
69
70
71
72
73
2.3.3 Fourier Transform Properties

SOLVED PROBLEMS

1. Find the Fourier transform of following and sketch their amplitude and phase spectra.

(a) X(t)= sgn(t)

74
(b) x(t) = 1 for all values of t

75
 The above result shows that a constant signal x(t)=1 for all t,.x (t) and X(jw)
are represented in figure(a) & (b) respectively.
(c) X (t) = u(t)

To find the Fourier transform of unit step u(t) by integration yields an indeterminate value as
isevident from the following equation because it has a jump discontinuity at t=0.

So the problem is approached by considering

76
2. Find the Fourier transform of decaying exponential signal.
The decaying exponential signal can be expressed as, x (t) = e-at

u(t)By the definition of Fourier transform

The lower limit is taken ‘0’ since x(t) = 0, for t < 0.and u(t) =1 for t ≥ 0

Thus the Fourier transform pair becomes,

To calculate magnitude and phase spectrum:

The function X (f) is expressed as,

Here A (f) is real part of X (f) and B (f) is the imaginary part of X(f).

Therefore magnitude spectrum of X (f) is given as,

And phase spectrum is given as,

77
 Multiply and divide RHS by a-j2πf

……. (1)

From the equation (1), The magnitude spectrum of X(f) will be

The phase spectrum will be

78
 Find the Fourier transform of double sided exponential signal.

Solution: The double exponential pulse of above figure can be represented as,

From the definition of Fourier transform

79
 Or in other words integration at single point with upper and lower limits same is zero only.

Find the Fourier transform of Sinc pulse as shown in figure below

Solution:

By applying duality and time scaling properties of Fourier transform we have,

Figure shows the spectrum of Sinc pulse.

Spectrum of Sinc pulse

80
3. Find the Fourier transform of following signal

By frequency shifting property

By using frequency shifting property the FT of x (t) is obtained

81
82
83
84
85
86
4. For the Fourier transforms shown in figure, find the energy of the signals
usingParseval’s theorem.

Solution:

Using Parseval’s theorem, energy is calculated as,

(a)

(b)

87
7. Find the Fourier transform of following using convolution theorem.

88
89
8. Find the magnitude spectrum for H(jw) and plot it. Where

Using the properties of continuous time Fourier transform determine the time domain
signalx (t), if the frequency domain signal

90
 Solution: From inspection of X(jw), the given problem can be solved using differentiation in
frequency, time shifting and scaling in the proper order.

First the time scaling property is applied. Let

According to time shifting property

According to differentiating property,

Applying the above property we have

91
 9. Find the inverse Fourier transform of following:

The above result can also got from first principle of inverse Fourier transform

Using the sampling property of the impulse function which exists only at w = w0 we get

By applying

By applying time differentiation property,

92
Using the definition of inverse FT, we get

93
94
 Compare the coefficients of jw on both sides,

SAMPLING THEOREM FOR LOW-PASS SIGNALS:

 A low pass signal contains frequencies from 1 Hz to some higher value.

 Statement of the sampling theorem

1) A band limited signal of finite energy , which has no frequency components

higher than W hertz , is completely described by specifying the values of the
signal at instants of time separated by1/2W seconds and
2) A band limited signal of finite energy, which has no frequency components
higher than W hertz , may be completely recovered from the knowledge of its
samples taken at the rate of 2W samples per second.

The first part of above statement tells about sampling of the signal and second
part tells about reconstruction of the signal. Above statement can be combined
and stated alternately as follows :

A continuous time signal can be completely represented ints samples and

recovered back ifthe sampling frequency is twice of the highest frequency
content ofthe signal i.e.,

fs≥2W Here fs is the sampling frequencyAnd W is the higher frequency content

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Proof of sampling theorem

There are two parts :

I) Representation of x(t) in terms of its samples II)Reconstruction of x(t) from its samples

PART I: Representation of x(t) in its samples x(nTs)Step 1 : Define xδ(t)

Step 2 : Fourier transform of xδ(t) i.e. Xδ(f)Step 3: Relation between X(f) and Xδ(f) Step 4 :
Relation between x(t) and x(nTs) Step 1 : Define xδ(t)
The sampled signal xδ(t) is given as ,

Here, observe that xδ(t) is the product of x(t) and impulse train δ(t) as shown in
figure.
In the above equation δ(t-nTs) indicates the samples placed at
±Ts,±2Ts,±3Ts…and so on
Step 2 : Fourier transform of xδ(t)i.e.
Xδ(f)Taking FT of equation (1)

96
We know that FT of product in time domain becomes convolution in frequency
domain i.e.,

Conclusions:
i) The RHS of above equation shows that X(f) is placed
at±fs,±2fs,±3fs,.. ii)This means X(f) is periodic in fs.
iii) If sampling frequency is fs=2W , then the spectrums X(f) just touch each other .

97
Step 3:Relation between X(f) and Xδ (f)
Important assumption

Let us assume that fs=2W ,then as per above diagram .

In above equation ‘f' is frequency of CT signal and fs = Frequency of DT signal in

equation (4) .Since x(n)=x(nTs),i.e. samples of x(t), then we have,

Since 1fs=Ts1fs=Ts
Putting above expression in equation (3),

Conclusions:

1) Here x(t) is represented completely in terms of x(nTs).

2) Above equation holds for fs=2W.This means if the samples are taken at the
98
rateof 2W or higher, x(t) is completely represented by its samples.
3) First part of the sampling theorem is proved by above two conclusions.
II) Reconstruction of x(t)from its samples

Step 1 : Take inverse Fourier transform of X(f) which is in terms of Xδ(f)

Step 2 : Show that x(t) is obtained back with the help of interpolation

function.Step 1 :Take inverse Fourier transform of equation (5) becomes ,

Interchanging the order of summation and integration,

Simplifying above equation,

99
Conclusions:

The samples x(nTs) are weighted by sinc functions.

The sinc function is the interpolating function .fig 4.1.3 shows, how x(t)is
interpolated.

100
Step 3:

Reconstruction of x(t) by low pass filter

When the interpolated signal of equation (6) is passed through the low pass filter
of bandwidth -W≤f≤W , then the reconstructed waveform shown in fig.4.1.3(b) is
obtained. The individual sinc functions are interpolated to get smooth x(t).

When high frequency interferes with low frequency and appears as low
frequency , then the phenomenon is called aliasing.

Effects of aliasing:

i) since high and low frequencies interfere with each other , distortion is

generated.ii)The data is lost and it cannot be recovered.

Different ways to avoid aliasing :

Aliasing can be avoided by two

methodsi)sampling rate fs≥2W

ii) Strictly band limit the signal to ‘W’

PART-A QUESTIONS
1. Write Dirichlet’s conditions.
• The function x (t) should be single valued in any finite time interval T.
• The function x (t) should have at the most finite number of discontinuities in any
finitetime interval T.
• The function x (t) should have finite number of maxima and minima in any
finite timeinterval T.
• The function x(t) should be absolutely integrable i.e.

2. Define Sampling Theorem.

The sampling theorem specifies the minimum-sampling rate at which a continuous-time
signal needs to be uniformly sampled so that the original signal can be completely recovered
or reconstructed by these samples alone.
101
3. What is an aliasing Effect?
Aliasing occurs when you sample a signal (anything which repeats a cycle over time) too slowly (at a
frequency comparable to or smaller than the signal being measured), and obtain an incorrect
frequency and/or amplitude as a result.
4. What is the condition for existence of Fourier transform?
 Signal must be absolutely integrable over a period.
 Signal must be of bounded variation in any given bounded interval.
 Signal must have a finite number of discontinuities in any given bounded interval, and the
discontinuities cannot be infinite

PART-B QUESTIONS

(1) Find IDFT X (e jw) =(1+coswt-2sin2wt)

(2) State and prove time convolution and time differentiation properties of Fourier

transform.(3)Obtain the Fourier transform of the following functions: (a)Unit step function

(b) DC signal

(4) Find the fourier transform of symmetrical triangular pulse and sketch the spectrum
usingproperties.

(5)State and prove following properties of fourier transform:(1)Time shifting, (2)Time scaling.

(6) Show that the Fourier transform of a functions; is a

gatefunction. Also sketch both the functions.

(7)Obtain the Fourier transform of the following functions:

(8) Explain the following functions:

(i) Impulse function (ii) DC signal with a value ‘A’. (iii) Unit step function, u(t).

(9) State and prove time convolution property of Fourier transform.

(10) Find the correlation of symmetrical gate pulse with amplitude and time duration ‘ ’ with
itself. Evaluate

(11) Find Fourier transform of sin .

(12)State and prove following properties of Fourier transform: (i) Time shifting. (ii) Scaling.

(13) Determine the Fourier transform of a two sided exponential pulse x (t) = e–||.
102
 UNIT III
Laplace Transform

Introduction

We know that 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.
ThusFT does not exist for such signals. By the use of Laplace transformation this limitation
can be avoided.WKT in FT the variable s = jw.

But in Laplace transform variable s can be expressed as s = σ+jώ

σ - real part of which represents the attenuation factor.

Jώ- imaginary part, ώ- angular frequency.

Laplace transform exists for almost all signals of practical interest. Some of the advantages
ofLaplace transform are as follows: Laplace transform can be used for the analysis of
unstable systems.
There are two types of Laplace transform:
1) Bilateral (or) two sided Laplace transform
2) Unilateral (or) one sided Laplace transform

1) Bilateral Laplace Transform

The Laplace transform can be alternatively defined as the bilateral Laplace

transform ortwo-sided Laplace transform by extending the limits of integration to be the
entire real axis.

The bilateral Laplace transform is defined as follows:

The f(t) and F(s) are Laplace transform pairs. It is written as,

Where‘s’ is complex frequency, it is given as

103
Here σ is the attenuation constant or damping factor and w is the angular frequency. With
abovevalue of ‘s’ we can write F(s) equation as,

The above equation shows that F(s) is basically a Fourier transform of f(t) e -σt.This is
the relationship between Fourier transform and Laplace transform. The Fourier transform
given by the above equation must exist, which is actually Laplace transform. Hence
sufficient condition off(t) to be Laplace transformable is that

For real and positive values of σ.

2) Unilateral Laplace Transform

The unilateral Laplace transform is the special case of Laplace transform and is defined as,

The unilateral Laplace transform has the following features:

1. The unilateral Laplace transform simplifies the system analysis considerably.

2. The signals are restricted to casual signals.
3. There is one to one correspondence between LT and ILT.
4. In view of above advantages Laplace transform means unilateral Laplace transform.

104
3) Inverse Laplace Transform

The inverse Laplace transform is given as

The above formula of inverse Laplace transform involves a complex integration. In this
chapter we will use a partial fraction expansion method to evaluate inverse Laplace
transforms.

Region of convergence

The range variation of σ for which the Laplace transform converges is called region of
convergence.

Properties of ROC
 ROC contains strip lines parallel to jω axis in s-plane.

 If x (t) is absolutely integral and it is of finite duration, then ROC is entire s-plane.

 If x (t) is a right sided sequence then ROC: Re{s} > σo.

 If x (t) is a left sided sequence then ROC: Re{s} < σo.

 If x (t) is a two sided sequence then ROC is the combination of two regions.

2.4.3 Causality and Stability

1. For a system to be causal, all poles of its transfer function must be right half of s-plane.

2. A system is said to be stable when all poles of its transfer function lay on the left half
of s-plane.

105
3. A system is said to be unstable when at least one pole of its transfer function is shifted to
theright half of s-plane.

4. A system is said to be marginally stable when at least one pole of its transfer function lies
onthe jω axis of s-plane.

Properties of Laplace transform

1. Linearity

Let x1(t) be the two Laplace transform pairs. Then linearity property states that,

Here a1 and a2 are constants.

Proof: let us find the Laplace transform of a1f1(t) + a2 f2(t) by applying definition.ie

106
2. Time scaling property

It states that

3. Scaling in s domain

It states that

Proof: According to time scaling property,

Let

Replacing b by a we get

3. Time differentiation

It states that

107
 Proof: According to the definition of Laplace transform

The above integral is evaluated by parts using

The time differentiation twice is proved as follows:

Using the property

We get

108
 In general

5. Time integration
The time integration property states that if

Proof: We define

Differentiating the above equation we get

109
6. Time convolution

The time convolution property states that if

Proof:

The inner integral is the LT of x2(t-ς) with a time delay ς. substituting

In the above equation, we get

110
7. Complex frequency differentiation

According to this property,

Proof: By definition of LT,

Differentiating both sided with respect to s,

8. Frequency shifting

According to this property,

111
9. Conjugation property

According to this property,

Proof: By definition of LT

10. Initial value theorem

According to this theorem,

Proof:

112
 11. Final value theorem
According to this theorem,

Proof: The LT of d/dt(x (t)) could be written as

Taking s-> 0 on both sides of the above equation, we get

The above theorem is valid if X(s) has no poles in RHP of s-plane.

Solved Problems
1. Find the Laplace transform of ramp signal.
The ramp signal is given as,

By the definition of Laplace transform,

113
 2. Find the Laplace transform of delayed ramp signal.
Solution:
If unit ramp function is delayed by time t0, it is given as,

By the shifting property of Laplace transform

Similarly

Find out the Laplace transform of impulse function.

We have evaluated the relationship between unit impulse function and step function.
Thedifferentiation of unit step function gives unit impulse function i.e.,

Taking Laplace transform on both sides,

By differentiation property, the RHS of above equation will be,

In the above equation

Therefore

If the impulse function is delayed by t 0, then its Laplace transform can be obtained by
shiftingproperty as,

114
3. Determine the LT of an exponential decay which is shown in figure.

Solution: The exponential decay is represented by

Taking LT for the above function we get

4. Find out the Laplace transform of sine wave.

Solution:
A sine wave is given as,

We know that sin w0t can be represented using Euler’s identity as,

115
 So f(t) becomes

Taking Laplace transform on both sides,

Putting these values in L{f(t)} we get,

5.Determine the LT of a sine function which is shown in figure

Solution: A sinusoidal function shown in figure is mathematically expressed as follows:

116
The given sinusoidal function is written as follows using Euler’s identity.

6. Determine the LT of a cosine function which is shown in figure

Solution: A cosine function shown in figure is mathematically expressed as follows:

The given sinusoidal function is written as follows using Euler’s identity

Taking LT for x (t), the following equation is written

117
7. Find out the Laplace transform of sine wave.
Solution: A sine wave is given as,

We know that sin w0t can be represented using Euler’s identity as,

So f(t) becomes

Taking Laplace transform on both sides,

118
8. Find the Laplace transform of damped sine wave.

We know that,

Using the complex shifting property,

Applying the above property we get,

(Or)

Taking Laplace transform on both sides,

119
9. Find the Laplace transform of damped cosine wave.

Solution: With the help of Euler’s identity,

Taking Laplace transform on both sides,

10. By applying the complex differentiation property, determine the LT of

Solution: We know that

According to the complex differentiation property,

120
11. Determine the LT of

Solution: The given signal x(t) is written in the following form.

12. Find the Laplace transform of x(t) = tn u(t).

Solution: Using the definition of LT for the given function we get

121
13. Determine the Laplace transform and sketch the ROC in the s- plane.

Solution:
1. X (t) is completely a right sided signal and hence the limit of the integration is from
t=0 tot=∞.Thus the following equation is written for X(s).

2. The poles are at s= -2 and s= -3 and a zero is at s= -2.5 and are marked in figure.

3. For the pole 1/(s+2), the ROC is right sided to the vertical line passing through σ = -2.For
the pole 1/(s+3), the ROC is also right sided passing through σ = -3.If ROC where σ > -2 is
satisfied the ROC where σ > -3 is automatically satisfied. Further no pole of X(s) will be inside
the ROC.

4. A strip to the right of σ =-2 is created and shaded. The strip is enlarged to ∞ in the direction
of real and imaginary axis.

5. Thus, the ROC of a casual signal is to the right of the right most pole of X(s).

122
15. Determine the Laplace transform and locate the poles and zeros of X(s) and also
the ROC in the s- plane.

Solution:
The given signal is fully a left sided signal and hence the limit of LT integration is from -∞
to 0. The LT of x (t) is obtained as follows:

123
16. Determine the Laplace transform and locate the poles and zeros and ROC in

the s- Plane for the following signal.

Solution:

1. The given signal is right- sided signal. And hence the limit of LT integration is from 0
to ∞. The LT of x (t) is obtained as follows

2. For the given signal, a pole at the origin exists and it is marked in figure (b).

3. The LT converges only σ >0.Thus the ROC is the entire right half of s-plane.

124
17. Find the Laplace transform and ROC of

Solution:

18. Find the Laplace transform and ROC of

Solution:

125
 19. Find the Laplace transform and ROC of

Solution:

Referring to the above diagram, combination region lies from –a to a. Hence,

Laplace transform of periodic signal

If a signal x (t) is a periodic signal with period T, then the LT of X(s) is given as

Here x1(t) is the signal which is repeated for every T.

126
 Inverse Laplace transform

The time signal x (t) is the Inverse LT of X(s).This is represented by the following
mathematical equation.

…… (1)
Use of the above equation to obtain x (t) from X(s) is really a tedious process. The
alternative is to express X(s) in polynomial form both in the numerator and the
denominator.Both these polynomials are factorized as

……(2)

The points in the s plane at which X(s) = 0 are called zeros. Thus (s+z1), (s+z2),
(s+z3)…..(s+zm) are the zeros of X(s). Similarly, the points in the s-plane at which X(s) = ∞
are called poles of X(s).

The zeros are identified by a small circle 0 and the poles by a small cross x in the s- plane.
For m<n the degree of the numerator polynomial is less than the degree of the
denominator polynomial. Under this condition X(s) in equation (2) is written in the
following partial fractionform.

…… (3)

In equation (3) A1,A2,…. An are called the residues and are determined by any
convenientmethod. Once the residues are determined, one can easily obtain x(t) which is
the required inverse LT of X(s).

1. Find the inverse LT of

Solution: Consider the function

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Putting this into partial fraction we get

Taking inverse LT we get

According to time shifting property of LT

2. Find the inverse LT of

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Solution: The given function is written in the following form:

Now consider X2(s) without delay as X3(s)

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PART-A QUESTIONS.
1. Define ROC.
The set of signals that cause the system's output to converge lie in the region of convergence (ROC).
2. Write any 2 properties of Laplace Transform.
(a) Linearity

(b) Time Scaling

3. What are the properties of ROC?

4. What is the condition for stability of a system in LT?

A system is said to be stable when all poles of its transfer function lay on the left half of
s-plane.
PART-B QUESTIONS
(1) Determine the LT and associated ROC and pole-zero plot for the following function of
timex(t) =
(2) Find the signal corresponding to X(s) = .

(3) Find Laplace transforms and sketch their ROC of:x(t) = u(t - 5)

𝑑2𝑦(𝑡) 3𝑑(𝑡)
(4) A causal LTI system described by the differential equation + + 2𝑦(𝑡) = 𝑥(𝑡) with input

𝑑
x (t) =2u (t) and with initial conditions 𝑦(0−) = 3;𝑦(0−) = −5 & 𝑥(𝑡) = 2𝑢(𝑡) find system y(t)
𝑑𝑡

(5) State and prove Initial and Final value theorems in LT

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 UNIT IV
Signal Transmission Through Linear Systems

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PART-A QUESTIONS
1. What are the characteristics of a filter?

An ideal low-pass filter completely eliminates all frequencies above the cutoff
frequency while passing those below unchanged; its frequency response is a
rectangular function and is a brick-wall filter. The transition region present in practical filters
does not exist in an ideal filter.
2. Define LTI system?
A linear time-invariant system (LTI system) is a system that produces an output signal from
any input signal subject to the constraints of linearity and time-invariance
3. State Poly-Wiener criterion for physical realization.

4. Give the relation between Rise Time and Bandwidth.

5. Differentiate between Group Delay and Phase Delay.

The phase delay of the filter is the amount of time delay each frequency component of the signal
suffers in going through the filter. 2. The group delay is the average time delay the composite signal
suffers at each frequency.
6. What is a distortion less Transmission?
Transmission is said to be distortion-less if the input and output have identical wave shapes. i.e., in
distortion-less transmission, the input x(t) and output y(t) satisfy the condition: y (t) = Kx(t - td)

PART-B QUESTIONS

(1) State and prove sampling theorem for band limited signals using graphical approach.
(2)What is aliasing effect? How it can be eliminated? Explain with neat diagram.
(3) Explain the conditions required for distortion less transmission.

(4) Define system bandwidth and signal bandwidth, compare them with the help of examples.

(5) Derive an expression for the transfer function of an LTI system.

(6) Prove that the transmission of a pulse through a low pass filter causes the dispersion of pulse.

(7)Derive the expression for transfer function of flat top sampled signal.

(8) Define and derive how rise time and system bandwidth are related?

(9) Mention and compare different types of sampling techniques with neat waveforms.

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 UNIT V
Discrete Time Fourier Transform AND Z-Transform

Introduction
The discrete time Fourier transform (DTFT) is the member of the Fourier transform
family that operates on Aperiodic, discrete signals. The best way to understand the DTFT is
how it relates to the DFT. To start, imagine that you acquire an N sample signal, and want to
find its frequency spectrum. By using the DFT, the signal can be decomposed into sine and
cosine waves, with frequencies equally spaced between zero and one-half of the sampling
rate. As padding in the time domain signal with zeros makes the period of the time
domain longer, as well as making the spacing between samples in the frequency domain
narrower. As N approaches infinity, the time domain becomes Aperiodic, and the frequency
domain becomes a continuous signal. This is DTFT, the Fourier transform that relates an
Aperiodic, discrete signal, with a periodic, continuous frequency spectrum.

Definition

The Discrete time Fourier transform of the discrete signal x[n] is given by,

The above equation is called analysis equation.

The time domain signal x[n] is obtained from X(e jw) by taking inverse Discrete Time
FourierTransform which is given by,

The above equation is referred to as synthesis equation (or) Inverse DTFT.

Fourier transform of a signal in general is a complex valued function, we can write

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Where XR(ejw) is the real part of X(ejw) and XI (ejw) is imaginary part of the function X(ejw).we
canalso use a polar form

Where I X(ejw)I is magnitude and < X(ejw) is the phase of X(ejw).we also use the term Fourier
transform or simply, the spectrum to refer to X(ejw).thus I X(ejw)I is called magnitude
spectrumand < X(ejw) is called the phase spectrum.

For simplicity ejw is considered as Ω.so the Fourier transform pair can be represented as,

Existence of DTFT

From the definition of DTFT observe that there is summation over infinite range of
n.Hencefor DTFT to exist, the convergence of this summation is necessary. The above
equation will converge if x(n) is absolutely summable.i.e.,

…….(3)

If x(n) is not absolutely summable,then it should have finite energy for DTFT to exist.i.e.,

……(4)

Note that inverse DTFT does not have convergence problem since the integration is over
limited range (-π to π).most of the physical signals satisfy above conditions.

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Properties of DTFT

1) Periodicity

It states that

Proof: By definition of DTFT,

In the above equation k and n are integers. Hence cos (2πkn)=1 always and
sin(2πkn)=0always. Hence e-j2πkn =1 and the above equation becomes,

2) Linearity

This property states that,

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Proof: By definition of DTFT,

Putting for z(n)=a x(n)+b y(n) i.e. linear combination of two inputs in above equation

Thus the outputs are linearly related. This is superposition principle.

3. Time shifting

This property states that,

Proof: By definition of DTFT,

put m=n-n0 .since n varies from -∞ to ∞,m will also have the same range.
The above equationbecomes,

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Thus delaying sequence in time domain is equivalent to multiplying its spectrum by

e-j Ωn 0 .

4. Frequency shifting

This property states that,

Proof: By definition of DTFT,

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5. Differentiation in frequency domain

This property states that,

Proof: By definition of DTFT,

Changing the summation and differentiation,

Comparing above equation with the definition of DTFT, we find that –jnx(n) has

DTFT ofd/dΩ [X(Ω)] .i.e,

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6. Time reversal

This property states that

Proof: By definition of DTFT,

Put m=-n, the above equation becomes,

Thus the sequence is folded in time, then its spectrum is also folded.

7. Convolution in time domain

This property states that

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Proof: By definition of DTFT,

Putting for

Changing the order of summation,

Put n-k=m, the above equation becomes,

Thus convolution of the two sequences is equivalent to multiplication of their spectrums.

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8. Multiplication in time domain

This property states that

Proof: By definition of DTFT,

Putting for z(n)= x(n) y(n) in above equation,

From the inverse DTFT, we know that,

Here we have used separate frequency variable λ.putting the above expression of x(n) in z(Ω).

Interchanging the order of summation and integration,

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The above equation represents the convolution of X(Ω) and Y(Ω)

Thus multiplication of two sequences in time domain is equivalent to convolution of their

spectrums.

9. Parseval’s theorem

Parseval’s theorem states that

Then energy of the signal is given as,

Proof: we know that energy of the signal is given as,,

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We can write the inverse DTFT of x*(n) as,

Putting the above value in energy equation,

Changing the order of summation and integration,

The energy of the discrete signal is given by

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Z-transform

In mathematics and signal processing, the Z-transform converts a discrete-time

signal, which is a sequence of real or complex numbers, into a complex frequency domain
representation.
Definition

The Z-transform can be defined as either a one-sided or two-sided transform.

(i) Bilateral Z-transform

The bilateral or two-sided Z-transform of a discrete-time signal x[n] is the formal

powerseries X(z) defined as

where n is an integer and z is, in general, a complex number:

where A is the magnitude of z, j is the imaginary unit, and ɸ is the complex argument (also
referred to as angle or phase) in radians.

(ii) Unilateral Z-transform

Alternatively, in cases where x[n] is defined only for n ≥ 0, the single-sided or

unilateralZ-transform is defined as

Inverse Z-transform

The inverse Z-transform is calculated by using the formula,

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where C is a counter clockwise closed path encircling the origin and entirely in the
region of convergence (ROC).

A special case of this contour integral occurs when C is the unit circle (and can be
used when the ROC includes the unit circle which is always guaranteed when X(z) is stable,
(i.e. all the poles are within the unit circle). The inverse Z-transform simplifies to the inverse
discrete- time Fourier transform:

The Z-transform with a finite range of n and a finite number of uniformly spaced z values
can becomputed efficiently via Bluestein's FFT algorithm.

Region of convergence

The region of convergence (ROC) is the set of points in the complex plane for which
theZ-transform summation converges.

The range of variation of z for which z-transform converges is called region of

convergence ofz-transform.

The stability of a system can also be determined by knowing the ROC alone. If the
ROCcontains the unit circle (i.e., |z| = 1) then the system is stable.

If you are provided a Z-transform of a system without an ROC .you can determine a unique
x[n]provided you desire the following:

 Stability
 Causality

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If you need stability then the ROC must contain the unit circle.

If you need a causal system then the ROC must contain infinity and the system function
will bea right-sided sequence.

If you need an anti-causal system then the ROC must contain the origin and the system
functionwill be a left-sided sequence.

If you need both, stability and causality, all the poles of the system function must be inside
theunit circle.

Properties of ROC of Z-Transform

 ROC of z-transform is indicated with circle in z-plane.

 Roc is a ring, whose center is at origin.

 ROC does not contain any poles.

 If x(n) is a finite duration causal sequence or right sided sequence, then the
ROC isentire z-plane except at z = 0.

 If x(n) is a finite duration anti-causal sequence or left sided sequence, then the
ROC isentire z-plane except at z = ∞.

 If x(n) is a infinite duration causal sequence, ROC is exterior of the circle with radius a.
i.e. |z| > a.

 If x(n) is a infinite duration anti-causal sequence, ROC is interior of the circle with radius
a. i.e. |z| < a.

 If x(n) is a finite duration two sided sequence, then the ROC is entire z-plane except at z
= 0 & z = ∞.

Causality and Stability

Causality condition for discrete time LTI systems is as follows:
A discrete time LTI system is causal when

 ROC is outside the outermost pole.

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  In The transfer function H[Z], the order of numerator cannot be greater than the
order ofdenominator.

Stability Condition for Discrete Time LTI Systems

A discrete time LTI system is stable when

 Its system function H[Z] include unit circle |z|=1.

 All poles of the transfer function lay inside the unit circle |z|=1.

Properties of the z-Transform

The z-transform has a few very useful properties, and its definition extends to infinite
signals/impulse responses.

(1) Linearity

This property states that

While it is obvious that the ROC of the linear combination of x(n) and y(n) should be the
intersection of the their individual ROCs in which both X(Z) and Y(Z) exist.

(2) Time Shifting

This property states that

Proof:

Let , we have and

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 The new ROC is the same as the old one except the possible addition/deletion of the
origin orinfinity as the shift may change the duration of the signal.

(3) Time Expansion (Scaling)

This property states that

Proof: The z-transform of such an expanded signal is

Note that the change of the summation index from n to m has no effect as the terms skipped
areall zeros.

(4) Convolution

This property states that

The ROC of the convolution could be larger than the intersection of RX and RY, due to the
possible pole-zero cancellation caused by the convolution.

(5) Time Difference

This property states that

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Proof:

Note that due to the additional zero z=1 and pole z=0, the resulting ROC is the same as R X
except the possible deletion of z=0 caused by the added pole and/or addition of z = 1
caused bythe added zero which may cancel an existing pole.

(6) Time Accumulation

This property states that

Proof: The accumulation of x[n] can be written as its convolution with u[n]:

Applying the convolution property, we get

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 .

(7) Time Reversal

This property states that

Proof:

where m=-n.

(8) Scaling in Z-domain

This property states that,

Proof:

In particular, if a=ejw0, the above becomes

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This property is essentially the same as the frequency shifting property of discrete Fourier
transform.

(9) Conjugation

Proof: Complex conjugate of the z-transform of is

Replacing z by , we get the desired result.

(10) Differentiation in z-Domain

Proof:

i.e.,

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SOLVED PROBLEMS

1. Determine the Fourier transform of the unit sample sequence x(n) = δ(n).

Solution: The unit sample sequence is defined as,

By the definition of Fourier transform,

Thus the Fourier transform has the value 1 for all values of Ω.

2. Determine the Fourier transform of the unit step sequence x(n) =

u(n)Solution: The unit step sequence is defined as,

By the definition of Fourier transform,

Now let us use the relation,

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Hence X(Ω) becomes

This relation is not convergent for Ω=0.This is because x(n) is not absolutely summable
sequence. However X(Ω) can be evaluated for other values of Ω. Let us rearrange equation

By Euler’s identity we can write,

3. Determine the Fourier transform of

Solution: Let us check whether the Fourier transform is convergent i.e.

Hence Fourier transform is convergent. By definition of Fourier transform we have

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Hence we can apply geometric summation formula. i.e.,

This is the required Fourier transform.

4. Determine the Fourier transform of the discrete time rectangular pulse of

amplitude Aand length L.ie

Solution: Let us check whether the Fourier transform is convergent. i.e.

Thus x(n) is absolutely summable and Fourier transform will exist. By definition of

Fouriertransform

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Here let us use the standard relation,

The above equation can be further simplified using Euler’s identity

5. Determine the discrete time sequence where DTFT is given as,

Solution: The inverse DTFT is given by,

By Euler’s identity we can write above equation as,

When n=0 in above equation, we get

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Thus the sequence x(n) is,

SOLVED PROBLEMS

1) Find the z transform of following:

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The pole zero diagram is shown in above figure. The ROC is the interior of the circle.

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2. Find the inverse z transform of following:

For ROC: IzI > 2, x[n] is a right-sided sequence where n ≥ 0.Hence , the long
division isdone in such a way that X[z] is expressed in power of z-1.

For ROC: IzI < 1, x[n] sequence is negative where n≤0.Hence , the long division is
done insuch a way that X[z] is expressed in power of z.

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3. Find the inverse z transform of following:

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 PART-A Questions

1. Describe the ROC of the signal.

The range of variation of z for which z-transform converges is called region of
convergence ofz-transform.
2. Write any two properties of ZT.

(a) Linearity

(b) Time Shifting

3. Find the Z-Transform of x[n] = {1, 3, 2, 1}.

X(Z)=1 + 3Z-1 + 2Z-2 + 1 Z-3
4. What is the relation between LT &ZT.
The Laplace transform evaluated at s=jω is equal to the Fourier transform if its region of
convergence (ROC) contains the imaginary axis.
5. Write Initial and final value theorems in ZT.
For a causal signal x(n), the initial value theorem states that. x(0)=limz→∞X(z)

PART-B Questions

1. Find z transform, ROC and pole zero locations of: X(n) = u(n).

2. State and prove transform time reversal property.

3. Find the inverse transform of X(z) = (1/1 + z) (2z/z – 0.2).

4.Find the Laplace transform of: x(t) = e-(t – 2) (t – 2) u(t – 2) (13)Find the inverse z – transform of:

X(z) = (1/1 + 2z) + (2z/z – 0.25)

5. Prove Initial and Final value theorems of ZT.

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