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EC8352 Signals and Systems Notes

Signal and system (Gujarat Technological University)

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Table of Contents

[Link] Contents [Link]

Unit I- Classification of Signals and Systems

1.1 Introduction 1
1.2 Continuous time and Discrete time signals 1

et
1.3 Elementary signals 2
1.3.1 Unit step signal 2
1.3.2 Ramp signal 2

.n
1.3.3 Unit impulse signal 2
1.3.4 Sinusoidal signal 2
1.3.5 Exponential signal 3

pz
1.4 Classification of CT and DT signals 3
1.5 CT and DT systems 5
1.6 Classification of systems 5
1.6.1 Static and Dynamic 5
ee
1.6.2 System with memory and without memory 5
1.6.3 Time variant and time invariant 5
1.6.4 Linear and Non linear 6
ad

1.6.5 Causal and non causal 7

1.6.6 Stable and Unstable 7
Unit II – Analysis of Continuous time signals
2 Fourier series analysis 14
.p

2.1 Fourier transform 14

2.2 Inverse transform 17
2.3 Parsevals theorem 17
w

2.4 Laplace transform 18

2.5 Properties of ROC of Laplace transform 19
Unit III – Linear Time invariant continuous time system 24
w

3.1 System 24
3.2 LTI System 24
w

3.3 Block diagram representation 25

3.4 Impulse response 25
3.5 Properties of impulse response 26
3.6 Convolution integral 26
3.6.1 Convolution Integral Properties 27
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3.7 Properties of Laplace transform 28

3.8 State variables and Matrix representation 30
Unit IV – Analysis of Discrete Time systems
4.1 Sampling Theory 34
4.2 Aliasing 35
4.3 Sampling of Non-bandlimited Signal: Anti-aliasing Filter 36

et
4.4 Discrete time fourier transform 36
4.5 Inverse transform 37
4.6 Properties 38

.n
4.7 Symmetry properties 39
4.8 Z-transform 40
4.9 Properties 41

pz
Unit V – Linear time invariant discrete time system
5.1 Introduction 47
5.2 Block Diagram representation 47
5.3 Convolution Sum 50
ee
5.4 LTI system analysis using DTFT 51
5.5 LTI system analysis using Z transform 51
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UNIT – I
CLASSIFICATION OF SIGNALS AND SYSTEMS

1.1 INTRODUCTION:
A signal, as stated before is a function of one or more independent variables. A signal
is a quantitative description of a physical phenomenon, event or process. More precisely, a
signal is a function, usually of one variable in time. However, in general, signals can be
functions of more than one variable, e.g., image signals. Signals are functions of one or more
variables.

t
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Systems respond to an input signal by producing an output signal .
Examples of signals include:
1. A voltage signal: voltage across two points varying as a function of time.
2. A force pattern: force varying as a function of 2-dimensional space.
3. A photograph: color and intensity as a function of 2-dimensional space.

z.
4. A video signal: color and intensity as a function of 2-dimensional space and time.
A continuous-time signal is a quantity of interest that depends on an independent
variable, where we usually think of the independent variable as time. Two examples are the
voltage at a particular node in an electrical circuit and the room temperature at a particular
spot, both as functions of time.
ep
A discrete-time signal is a sequence of values of interest, where the integer index can
be thought of as a time index, and the values in the sequence represent some physical
quantity of interest.
de
A signal was defined as a mapping from a set of the independent variable (domain) to
the set of the dependent variable (co-domain). A system is also a mapping, but across signals,
or across mappings. That is, the domain set and the co-domain set for a system are both sets
of signals, and corresponding to each signal in the domain set, there exists a unique signal in
the co-domain set.
a

System description
.p

The system description specifies the transformation of the input signal to the output
signal. In certain cases, a system has a closed form description. E.g. the continuous-time
system with description y (t) = x(t) + x(t-1); where x(t) is the input signal and y(t) is the
output signal.
w

1.2 Continuous-time and discrete-time systems

•
w

Physically, a system is an interconnection of components, devices, etc., such as a

computer or an aircraft or a power plant.
• Conceptually, a system can be viewed as a black box which takes in an input signal
x(t) (or x[n]) and as a result generates an output signal y(t) (or (y[n]).
w

• A system is continuous-time (discrete-time) when its I/O signals are continuous-time

(discrete-time).

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1.3 Elementary Signals:

The elementary signals are used for analysis of systems. Such signals are,
• Step
• Impulse
• Ramp
• Exponential
• Sinusoidal

t
1.3.1 Unit step signal:
• Unit Step Sequence: The unit step signal has amplitude of 1 for positive value and

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amplitude of 0 for negative value of independent variable.
• It have two different parameter such as CT unit step signal u(t) and DT unit step
signal u(n).
• The mathematical representation of CT unit step signal u(t) is given by,

z.
1.3.2 Ramp Signal:
• The amplitude of every sample is linearly increased with the positive value of

•
independent variable.
ep
Mathematical representation of CT unit ramp signal is given by,
de
1.3.3 Unit impulse function:
• Amplitude of unit impulse approaches 1 as the width approaches zero and it has zero
a

value at all other values.

• The mathematical representation of unit impulse signal for CT is given by,
.p
w

• It is used to determine the impulse response of system.

w

1.3.4 Sinusoidal signal:

• A continuous time sinusoidal signal is given by,
w

Where, A – amplitude - phase angle in radians

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1.3.4 Exponential signal:

• It is exponentially growing or decaying signal.
• Mathematical representation for CT exponential signal is,

t
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z.
1.4
ep
Classification of CT and DT signals:
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• Periodic and non-periodic Signals
A periodic function is one which has been repeating an exact pattern for an infinite period of
time and will continue to repeat that exact pattern for aninfinite time. That is, a periodic
function x(t) is one for which
x (t) = x(t+nT)
a

for any integer value of n, where T >0 is the period of the function and −∞ < t <∞ . The
signal repeats itself every T sec. Of course, it also repeats every 2 T,3T and nT. Therefore,
2T, 3T and nT are all periods of the function because the function repeats over any of those
.p

intervals. The minimum positive interval over which a function repeats itself is called the
fundamental period T0.T0 is the smallest value that satisfies the condition x ( t ) = x ( t+T0).
The fundamental frequency f 0 of a periodic function is the reciprocal of the fundamental
period f 0=1/T0. It is measured in Hertz and is the number of cycles (periods) per second.
w

The fundamental angular frequency ω0 measured in radians per second is ω0=2πT0= 2πf0.
A signal that does not satisfy the condition in (2.1) is said to be a periodic or non-periodic.

• Deterministic and Random Signals

w

Deterministic Signals are signals who are completely defined for any instant of time, there is
no uncertainty with respect to their value at any point of time. They can also be described
mathematically, at least approximately. Let a function be defined as
w

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A random signal is one whose values cannot be predicted exactly and cannot be described by

t
any exact mathematical function, they can be approximately described.

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• Energy and Power Signals:
Consider v(t) to be the voltage across a resistor R producing a current i(t). The
instantaneous power p(t) per ohm is defined as, Total energy E and average power P on a per-
ohm basis are

z.
ep
For an arbitrary continuous-time signal x (t), the normalized energy content E of x(t)
is defined as,
a de

The normalized average power P of x (t) is defined as,

.p

Similarly, for a discrete-time signal x[n], the normalized energy content E of x[n] is defined
w

as,
w

The normalized average power P of x[n] is defined as,

w

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1.5 CT Systems and DT Systems:

A system is defined as a physical device which contains set of elements or functional
blocks and that generates a response or output signal for a given input.

1.6 Classification of system:

The systems are classified as,
• Static & dynamic system
• Time invariant and variant system
• Linear and non linear system
• Causal and non causal system

t
• Stable and unstable system

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1.6.1 Static and dynamic system:
• Static system is said to be a memoryless system.
• The output does not depend the past or future input.
• It only depends the present input for an output.

z.
Eg, y(n) = x(n)
• Dynamic system is said to be as system with memory.
• Its output depend the past values of input for an output.

1.6.2
• ep
Eg. Y(n) = x(n) + x(n - 1)
This static and dynamic systems are otherwise called as memoryless and system with
memory.

Systems with and without memory:

•
de
A system is called memory less if the output at any time t (or n) depends only on the
input at time t (or n); in other words, independent of the input at times before of after t
(or n). Examples of memory less systems:
a
.p

Examples of systems with memory:

w

1.6.3 Time invariant and time variant system:

w

• If the time shifts in the input signals results in corresponding time shift in the output,
then the system is called as time invariant.
•
w

The input and output characteristics do not change with time.

• For a continuous time system,
f[x(t1 – t2)] = y(t1 – t2)
• For a discrete time system,
F[x(n - k)] = y(n - k)

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• If the above relation does not satisfy, then the system is said to be a time variant
system.
• A system is called time-invariant if the way it responds to inputs does not change over
time:

t
ne
Examples of time-invariant systems:
• The RC circuit considered earlier provided the values of R or C are constant.

z.
Examples of time-varying systems:

• The RC circuit considered earlier if the values of R or C change over time.

•
ep
Most physical systems are slowly time-varying due to aging, etc. Hence, they can be
de
considered time-invariant for certain time periods in which its behavior does not
change significantly.

1.6.4 Linear and non linear system:

•
a

A system is said to be linear if it satisfies the superposition principle.

• Superposition principle states that the response to a weighted sum of input signal be
equal to the weighted sum of the output corresponding to each of the individual input
.p

signal
• The continuous system is linear if,
F[a1x1(t) + a2x2(t)] = a1y1(t) + a2y2(t)
• The discrete system is linear if,
w

F[a1x1(n) + a2x2(n)] = a1y1(n) + a2y2(n)

• Otherwise the system is non linear.
• A system is called linear if its I/O behavior satisfies the additivity and homogeneity
w

properties:
w

for any complex constant a.

• Equivalently, a system is called linear if its I/O behavior satisfies the superposition
property:

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where any complex constants a and b.

1.6.5 Causal and non causal system:

t
• A causal system is one whose output depends upon the present and past input values.
•

ne
If the system depends the future input values, the system is said to be non causal.
Eg. for causal system.
Y(t) = x(t) + x(t - 1)
Y(n) = x(n) + x(n - 3)
Eg. For non causal system,

z.
Y(t) = x(t+3) + x2(t)
Y(n) = x(2n)
• A system is called causal or non-anticipative if the output at any time t (or n) depends

•
ep
only on the input at times t or before t (or n or before n); in other words, independent
of the input at times after t (or n). All memory less systems are causal. Physical
systems where the time is the independent variable are causal.
Non-causal systems may arise in applications where the independent variable is not
the time such as in the image processing applications.
de
Examples of causal systems:
a
.p

Examples of non-causal systems:

w

1.6.6 Stable and unstable system:

w

• When every bounded input produces bounded output then the system is called as
stable system or bounded input bounded output (BIBO stable).
• Otherwise the system is unstable.
w

• A system is called stable if it produces bounded outputs for all bounded inputs.
• Stability in a physical system generally results from the presence of mechanisms that
dissipate energy, such as the resistors in a circuit, friction in a mechanical system, etc.

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Sample Problems:
1. Determine whether the following systems are: i) Memoryless, ii) Stable iii)
Causal iv) Linear and v) Time-invariant.
i) y(n)= nx(n)
ii) y(t)= ( )

Solution:-

t
ne
z.
ep
a de
.p

2. Determine whether the following systems are time invariant or not. [

i) Y(t) = tx(t)
w

ii) Y(n) = x(2n)

Solution:
i) Y(t) = tx(t)
w

Y(t) = T[x(t)] = tx(t)

The output due to delayed input is,
Y(t,T) = T[x(t - T)] = tx(t - t)
w

If the output is delayed by T, we get

Y(t -T) = (t - T) x( t - T)
The system does not satisfy the condition, y(t,T) = y(t – T).
Then the system is time invariant.

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ii) Y(n) = x(2n)

Y(n) = x(2n)
Y(n) = T[x(n)] = x(2n)
If the input is delayed by K units of time then the output is,
Y(n,k) = T[x(n-k)] = x(2n-k)
The output delayed by k units of time is,
Y(n-k) = x[2(n-k)]
Therefore, y(n,k) is not equal to y(n-k). Then the system is time variant.

t
ne
z.
ep
a de
.p
w
w
w

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2 mark question and answer

1. Define unit impulse and unit step signals. [May 2010]

Unit Impulse signal:
Amplitude of unit impulse is 1 as its width approaches zero. Then it has zero
value at all other values.
Unit Step Signal:
The unit step signal has amplitude of 1 for positive values of independent
variable and amplitude of 0 for negative of independent variable.

t
ne
2. Give the mathematical and graphical representation of CT (continuous time) and
DT (discrete time) impulse function. [Dec 2013]
CT and DT impulse function:

z.
ep
de
3. Define step and impulse function in discrete signals. [May 2012]
a
.p
w
w
w

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4. State the two properties of unit impulse function. [Dec 2014]

i) Shifting property:

ii) Replication property:

t
5. Find the fundamental period of signal [Dec 2010]

ne
= ( + )
Solu:
7
( ) = sin +
3

z.
2
!=
#
7
#=
3
ep =%
$
sec

6. Check e time whether the discrete signal sin 3n is periodic? [June

de
2013]
• The frequency of the discrete time signal is 3, because it is not a multiple of .
• Therefore the signal is aperiodic.

7. Distinguish between deterministic and random signals. [May 2011]

a

(or)
Define random signal and deterministic signal. [May 2013]
.p

Random Signal:
It has some degree of uncertainty before it actually occurs. The random signal
cannot be defined by mathematical expressions.
Deterministic Signal:
w

There is no uncertainty occurrence. It is completely represented by

mathematical expressions.
w

8. Determine the period of the signal [Dec 2011]

= &'()(* /,)
Solu:
2 -. = . /4
w

1
-= × = 1/8
4 2

We know that, f=1/T

So T = 8sec

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9. When is a system said to be memory less? Give an example. [May 2010]

If the system output does not depend the previous input, it only depends the present
input. Then the system is called memory less or static system.
Eg:
3( ) = & ( ) + ( )
3(*) = (*) + √ (*)

t
ne
10. Define energy and power signals. [Dec 2010]
Energy Signal:

• A signal is said to be an energy signal if its normalized energy is non zero and
finite.

z.
• For an energy signal, P = 0.
i.e., 0 < 7 < ∞

Power Signal:

• For a
ep
• A signal is said to be the power signal if it satisfies 0 < 9 < ∞

power signal, 7=∞

de
11. What is the classification of system? [Dec 2009]
The classification of systems is,
(i). Linear and Non-Linear systems
(ii). Time invariant and Time varying systems.
a

(iii). Causal and Non causal systems.

(iv). Stable and unstable systems.
(v). Static and dynamic systems.
.p

(vi). Invertible and non invertible systems.

12. Verify whether the system described by the equation :( ) = ( ); is linear and
w

time invariant.

• The system is linear since output is direct function of input.

w

• The system is time variant since time parameter is squared in the given equation.
w

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13. Draw the signal x(n) = u(n) – u(n-3) [may 2011]

t
ne
z.
14. Check whether the following system is static/dynamic and casual/non casual
y(n) = x(2n).
ep
• If n=1, y(1) = x(2). This means system requires memory. Hence it is dynamic system.
• Since y(1) = x(2), the present output depends upon future input. Hence the system is
non casual.
de
15. Distinguish between static and dynamic system.
Static system:
• Does not require memory
• Impulse response is of the form ℎ( ) = =#( )
a

Dynamic system:
• Requires memory
• Impulse response can be any form except ℎ( ) = =#( )
.p
w
w
w

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Unit – II
Analysis of Continuous Time Signals

2. Fourier series analysis:

Fourier series: a complicated waveform analyzed into a number of harmonically
related sine and cosine functions

t
A two parts tutorial on Fourier series. In the first part an example is used to show how

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Fourier coefficients are calculated and in a second part you may use an applet to further
explore Fouries series of the same function.

Fourier series may be used to represent periodic functions as a linear combination of

sine and cosine functions. If f(t) is a periodic function of period T, then under certain

z.
conditions, its Fourier series is given by:

ep
where n = 1 , 2 , 3 , ... and T is the period of function f(t). an and bn are called Fourier
coefficients and are given by
a de
.p
w
w

2.1 Continuous Time Fourier Transform:

The Fourier expansion coefficient ( in OWN) of a periodic signal is

w

is

and the Fourier expansion of the signal is:

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which can also be written as:

t
ne
where is defined as

z.
When the period of
becomes a non-periodic signal
ep
approaches infinity
and the following will result:
, the periodic signal

Interval between two neighboring frequency components becomes zero:

a de

Discrete frequency becomes continuous frequency:

.p

Summation of the Fourier expansion in equation (a) becomes an integral:

w
w
w

the second equal sign is due to the general fact:

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Time integral over in equation (b) becomes over the entire time axis:

t
In summary, when the signal is non-periodic , the Fourier
expansion becomes Fourier transform. The forward transform (analysis) is:

ne
z.
and the inverse transform (synthesis) is:

ep
Note that is denoted by in OWN.
de
Comparing Fourier coefficient of a periodic signal with with Fourier spectrum of a

non-periodic signal
a
.p
w

we see that the dimension of is different from that of :

w
w

If represents the energy contained in the kth frequency component of a periodic

signal , then represents the energy density of a non-periodic signal

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distributed along the frequency axis. We can only speak of the energy contained in a

particular frequency band :

2.2 Inverse Transforms

If we have the full sequence of Fourier coefficients for a periodic signal, we can reconstruct it

t
by multiplying the complex sinusoids of frequency ω0k by the weights Xk and summing:

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z.
We can perform a similar reconstruction for aperiodic signals

ep
These are called the inverse transforms.
de
2.4 Parseval’s theorem:
a
.p
w
w
w

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2.5 Laplace Transform

►Lapalce transform is a generalization of the Fourier transform in the sense that it allows
“complex frequency” whereas Fourier analysis can only handle “real frequency”. Like
Fourier transform, Lapalce transform allows us to analyze a “linear circuit” problem, no
matter how complicated the circuit is, in the frequency domain in stead of in he time domain.

►Mathematically, it produces the benefit of converting a set of differential equations into a

corresponding set of algebraic equations, which are much easier to solve. Physically, it
produces more insight of the circuit and allows us to know the bandwidth, phase, and transfer

t
characteristics important for circuit analysis and design.

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►Most importantly, Laplace transform lifts the limit of Fourier analysis to allow us to find
both the steady-state and “transient” responses of a linear circuit. Using Fourier transform,
one can only deal with he steady state behavior (i.e. circuit response under indefinite
sinusoidal excitation).

z.
►Using Laplace transform, one can find the response under any types of excitation (e.g.
switching on and off at any given time(s), sinusoidal, impulse, square wave excitations, etc.

ep
a de
.p
w
w
w

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2.6 Properties of ROC of Laplace Transform:

1. The ROC of X(s) consists of strips parallel to the ># axis in the s-plane.
2. The ROC does not contain any poles.
3. If x(t) is of finite duration and is absolutely integrable, then the ROC is the entire s-
plane.
4. It x(t) is a right sided signal, that is x(t) = 0 for t<t0<∞ then the ROC is of the form
Re(s)> max , where max equals the maximum real part of any of the poles of X(s).
5. If x(t) is a left sided, that is x(t) = 0 for t>t1> -∞, then the ROC is of the form Re(s)<
min , where min equals the minimum real part of any of the poles of X(s).

t
6. If x(t) is a two sided signal, than the ROC is of the form 1<Re(s)< 2.

ne
z.
ep
a de
.p
w
w
w

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(2 mark questions)

1. What are the Dirichlet’s conditions of Fourier series? [June 2014, Dec 2009,2013]
(i).The function x(t) should be single valued within the interval T0
(ii). The function x(t) should have atmost a finite number of discontinuities
in the interval T0
(iii). The function x(t) should have finite number of maxima and minima
in the interval T0
(iv). The function should have absolutely integrable.

t
ne
2. State any two properties of continuous time Fourier transform.
[May 2010, Dec 2009]
Convolution (Time) Property:
It states that,
?@
x (t)*y(t) ↔X (jΩ)Y(jΩ)

z.
Modulation Property (or) Frequency Shifting:
It states that,
?@
x (t) BCD E
↔X(jΩ-jFG ) ep
3. Find the laplace transform of the signal ( ) = HI
J( ) [may2010]
de
Given x (t) = HKE u(t)
We know that,
O O O
X (Ω) = N ( ) HBCE
=N HKE
u(t) HBCE
=N HKE HBCE
HO HO G
a

O T UV TD X
= RG H(KSBC)E
=[H(KSBC) − H(KSBC)] = KSBC
.p

X
X (Ω)= KSBC
w

4. What are the difference between Fourier series and Fourier transform? [OCT/NOV
2002,NOV/DEC 2004,DEC 2009,MAY/JUNE 2010]
w

[Link] Fourier Series Fourier Transform

1 Fourier series is calculated Fourier Transform is calculated for
for periodic signals. non-periodic as well as periodic
w

signals.
2 Expands the signals in time Represents the signal in frequency
domain. domain
3 Three types of Fourier series Fourier transform has no such types.
such as trigonometric, Polar
and Complex Exponential

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5. State initial and final value theorem of Laplace transform.[DEC 2009,MAY-11]

Initial value theorem: (0) = lim ]^(])

[→O
Final value theorem: ( ) = lim]^(])
[→G

6. Define the Fourier transform pair for continuous time signal. (Or) Give synthesis

t
and analysis equations of CT Fourier Transform. [ NOV/DEC 2012]

ne
O
Fourier Transform: X (Ω) = RHO ( ) HBCE
X O
Inverse Fourier Transform:x (t) = ;` RHO ^(F) BCE
F

z.
7. Find inverse Fourier transform of X(ω)=2πδ (ω). [MAY/JUNE 2010]
X O
Inverse Fourier Transform: x (t) = RHO ^(F) BCE
F
;`
X

= 1.
O
x (t) = ;` RHO 2πδ(Ω)
ep
Since δ(Ω) = c
FBCE

1 for Ω = 0h
0 for Ω ≠ 0
[Note: ω=Ω]
de
8. State the time scaling property of Laplace Transform. [MAY/JUNE 2013]

It states that,
X [
L[x (at)] = |K| ^(K)
a

If L[x (t)] = X(S) then,

.p

9. What is the Fourier Transform of a DC signal of amplitude 1? [MAY/JUNE 2013]

O
W.K.T X (jΩ) = RHO ( ) HBCE
w

Here x(t) = 1 then, F [1] = 2 δ ( )

w

10. Define region of convergence of the Laplace Transform. [NOV/DEC 2012]

O
For a given signal the range of values of s, for which the integral RHO | ( )| converges
w

is called the region of convergence.

O
i.e.,RHO | ( ) lE
| <

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11. State the relationship between fourier transform and laplace transform.[may 2015]
• The laplace transform is given by,
O

^(m) = N ( ) HnE

HO

• The fourier transform is given by,

O

^(>#) = N ( ) HBoE

HO

t
• The laplace transform is same as fourier transform when m = >#

ne
12. State any two properties of ROC of laplace transform X(s) of a signal x(t).
[jun 2014]
Properties of ROC:

z.
• No poles lie in ROC.
• ROC of the causal signal is right hand sided. It is of the form Re(s)>a.
• ROC of the non causal signal is left hand sided. It is of the form Re(s) < a.
• The system is stable if its ROC includes ># axis of s-plane.
ep
13. Determine fourier series coefficients for signal cos
B`E
+ HB`E
[may 2012]
cos =
2
de
Fourier series is given as,
O
( ) = q ^(r) Bs`E

stHO
a
.p

14. Give analysis and synthesis equations of fourier transform. [dec 2012]

• Fourier transform,
O
Analysis equation => ^(#) = RHO ( ) HBoE
w

• Inverse fourier transform,

O
Synthesis equation => ^(-) = 1/2 RHO (#) BoE
#
w

HKE
15. Obtain the fourier transform of X(f) = u( ) , a > 0
w

^(-) = N ( ) HB;`vE

HO

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= N HKE HB;`vE

HO

= 1/(w + >2 -)

16. What is the condition to be satisfied for the existence of fourier transform for CT
periodic signals? [dec 2011]

t
The function x(t) should be absolutely integrable for the existence of fourier

ne
transform.
O
i.e.,RHO| ( )| < ∞

z.
ep
a de
.p
w
w
w

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UNIT III

LINEAR TIME INVARIANT –CONTINUOUS TIME SYSTEMS

3.1 System:
A system is an operation that transforms input signal x into output signal y.

t
ne
z.
3.2 LTI Systems
• Time Invariant

• Linearity
– X(t) ep
y(t) & x(t-to)

– a1x1(t)+ a2x2(t) a1y1(t)+ a2y2(t)

– a1y1(t)+ a2y2(t)= T[a1x1(t)+a2x2(t)]
y(t-to)

• Meet the description of many physical systems

de
• They can be modeled systematically
– Non-LTI systems typically have no general mathematical procedure to obtain solution

Differential equation:
• This is a linear first order differential equation with constant coefficients (assuming a and b
a

are constants)
.p

The general nth order linear DE with constant equations is

w
w
w

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Linear constant-coefficient differential equations In RC circuit

- To introduce some of the important ideas concerning systems specified by linear
constant-coefficient differential equations ,let us consider a first-order differential
equations:

t
ne
z.
3.3 Block diagram representations
Block diagram representations of first-order systems described by differential and
difference equations
ep
a de
.p
w
w

3.4 Impulse Response

w

This impulse response signal can be used to infer properties about the system’s
structure (LHS of difference equation or unforced solution). The system impulse response,
h(t) completely characterises a linear, time invariant system

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3.5 Properties of System Impulse Response

Stable
A system is stable if the impulse response is absolutely summable

Causal
A system is causal if h(t)=0 when t<0

Finite/infinite impulse response

The system has a finite impulse response and hence no dynamics in y(t) if there exists

t
T>0, such that: h(t)=0 when t>T

ne
Linear
ad(t) ah(t)

z.
Time invariant
d(t-T) h(t-T)

3.6 Convolution Integral

ep
• An approach (available tool or operation) to describe the input-output relationship for LTI
Systems
de
• In a LTI system
– d(t) h(t)
– Remember h(t) is T[d(t)]
– Unit impulse function the impulse response
a

• It is possible to use h(t) to solve for any input-output relationship

• Any input can be expressed using the unit impulse function
.p

3.6.1 Convolution Integral - Properties

w
w
w

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ne
z.
ep
ade
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w
w
w

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3.7 Properties of Laplace Transform:

t
ne
z.
ep
a de
.p
w
w
w

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ne
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ep
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w
w

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3.8 State variables and Matrix representation

• State variables represent a way to describe ALL linear systems in terms of a common set of
equations involving matrix algebra.
• Many familiar properties, such as stability, can be derived from this common
representation. It forms the basis for the theoretical analysis of linear systems.
• State variables are used extensively in a wide range of engineering problems, particularly
mechanical engineering, and are the foundation of control theory.
• The state variables often represent internal elements of the system such as voltages across
capacitors and currents across inductors.
• They account for observable elements of the circuit, such as voltages, and also account for

t
the initial conditions of the circuit, such as energy stored in capacitors. This is critical to

ne
computing the overall response of the system.
• Matrix transformations can be used to convert from one state variable representation to the
other, so the initial choice of variables is not critical.
• Software tools such as MATLAB can be used to perform the matrix manipulations required.
• Let us define the state of the system by an N-element column vector, x(t):

z.
for the state variables.
ep
Note that in this development, v(t) will be the input, y(t) will be the output, and x(t) is used

• Any system can be modeled by the following state equations:

de
• This system model can handle single input/single output systems, or multiple inputs and
outputs.
• The equations above can be implemented using the signal flow graph shown to the below
a
.p
w
w
w

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t
ne
z.
ep
ade
.p
w
w
w

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(2 mark questions)

1. What is the overall impulse response h(t) when two systems with impulse response
xy ( ) and x& ( ) are in parallel and in series? [MAY-10].
(or)
State the properties needed for interconnecting LTI systems. [MAY-09]

For parallel connection, h(t)=ℎX ( )+ℎ; ( )

For series connection, h(t)= ℎX ( ) ∗ ℎ; ( ).

t
ne
2. Write convolution integral of x(t) (or) Define convolution integral of continuous time
systems. [DEC-10,MAY-10,MAY-11]
O
The convolution integral is given as,y(t)=RHO ({)ℎ( − {)d{.

z.
3. Check whether the causal system with transfer function H(s) = 1/(s-2) is stable
[dec 2013]

of ># axis, the system is not stable. ep

Here the pole lies at s = 2. Since the pole of causal system does not lie on the left side

4. The impulse response of the LTI – CT system is given as ℎ( ) = HE u( ). Determine

de
transfer function and check whether the system is causal and stable.
ℎ( ) = HE u( )

Taking laplace transform,

H(s) = 1/(s+1)
a

Here the pole lies at s = -1, i.e. located in left half of s-plane. Hence this system is
causal and stable.
.p

5. What are the conditions for a system to be LTI system? [dec 2013]
Input and output of an LTI system are related by,
O
:( ) = RHO ({)ℎ( − {) { i.e. convolution
w
w

6. What is the impulse response of two LTI systems connected in parallel? [may 2010]
If the system are connected in parallel, having responses ℎX ( ) and ℎ; ( ), then their
overall response is given as,
ℎ( ) = ℎX ( ) + ℎ; ( )
w

7. Write Nth order differential equation. [DEC-10]

The Nth order differential equation can be written as,

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| ~
s
:( ) s
( )
q ws s
= q }s s
stG stG
Here N≥ €.

8. What is the condition for LTI system to be stable? [may 2013]

An LTI system is stable if the impulse response is absolutely integrable.
O

N |ℎ( )| < ∞

t
HO

ne
9. What is meant by impulse response of any system? [MAY-11]
When the unit impulse function is applied as input to the system, the output is
nothing but impulse response h(t). The impulse response is used to study various

z.
properties of the system such as causality, stability, dynamicity etc.

ep
10. Determine the response of the system with impulse response h(t)= t u(t) for the
input x(t)= u(t) [DEC-11]
de
The response is given as,
O
y(t)=RHO ℎ({)u( − {)d{.
O
y(t)=RHO {u({)u( − {)d{.
Here u({)u( − {)=1 for 0 to t. hence above equation will be,
E X
y(t)=RG {d{ = ; ; .
a
.p

11. State the properties of convolution. (DEC 2009).

1) Commutative property:x(t)*h(t) = h(t)*x(t)
2) Associative property: [x(t)*h1(t)]*h2(t) = x(t)*[h1(t)*h2(t)]
w

3) Distributive property:x(t)*h1(t)+x(t)*h2(t) = x(t)*[h1(t)+h2(t)]

12. What are the three elementary operations in block diagram representation of
w

continuous time system? [dec 2012,2013]

• Scalar multiplication
w

X(t) y(t) = ax(t)

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UNIT IV

ANALYSIS OF DISCRETE TIME SIGNALS

4.1 Sampling theory

Let x(t) be a continuous signal which is to be sampled, and that sampling is performed
by measuring the value of the continuous signal every T seconds, which is called the
sampling interval. Thus, the sampled signal x[n] given by: x[n] = x(nT), with n = 0, 1, 2, 3, ...

t
The sampling frequency or sampling rate fs is defined as the number of samples

ne
obtained in one second, or fs = 1/T. The sampling rate is measured in hertz or in samples per
second.

The frequency equal to one-half of the sampling rate is therefore a bound on the
highest frequency that can be unambiguously represented by the sampled signal. This

z.
frequency (half the sampling rate) is called the Nyquist frequency of the sampling system.
Frequencies above the Nyquist frequency fN can be observed in the sampled signal, but their
frequency is ambiguous. That is, a frequency component with frequency f cannot be
distinguished from other components with frequencies NfN + f and NfN – f for nonzero
ep
integers N. This ambiguity is called aliasing. To handle this problem as gracefully as
possible, most analog signals are filtered with an anti-aliasing filter (usually a low-pass filter
with cutoff near the Nyquist frequency) before conversion to the sampled discrete
representation.
de
► The theory of taking discrete sample values (grid of color pixels) from functions defined
over continuous domains (incident radiance defined over the film plane) and then using those
samples to reconstruct new functions that are similar to the original (reconstruction).

► Sampler: selects sample points on the image plane

► Filter: blends multiple samples together
a
.p
w
w

► Sampling theory
w

Sampling Theorem: bandlimited signal can be reconstructed exactly if it is sampled at

a rate atleast twice the maximum frequencycomponent in it."

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► Consider a signal g(t) that is bandlimited.

t
ne
The maximum frequency component of g(t) is fm. To recover the signal g(t) exactly
from its samples it has to be sampled ata rate fs _ 2fm. The minimum required sampling rate

z.
fs = 2fm is called nyquist rate

A continuous time signal can be processed by processing its samples through a

discrete time system. For reconstructing the continuous time signal from its discrete time

by the sampling theorem.

4.2 Aliasing
ep
samples without any error, the signal should be sampled at a sufficient rate that is determined
de
Aliasing is a phenomenon where the high frequency components of the sampled signal
interfere with each other because of inadequate sampling ωs < 2ωm. Aliasing
a
.p
w
w
w

Aliasing leads to distortion in recovered signal. This is the reason why sampling frequency
should be atleast twice the bandwidth of the signal.

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4.3 Sampling of Non-bandlimited Signal: Anti-aliasing Filter

Anti aliasing filter is a filter which is used before a signal sampler, to restrict the
bandwidth of a signal to approximately satisfy the sampling theorem. The potential defectors
are all the frequency components beyond fs/2 Hz. We should have to eliminate these
components from x(t) before sampling x(t). As a result of this we lose only the components
beyond the folding frequency fs/2 Hz. These frequency components cannot reappear to
corrupt the components with frequencies below the folding frequency. This suppression of
higher frequencies can be accomplished by an ideal filter of bandwidth fs/2 Hz. This filter is
called the anti-aliasing filter. The anti aliasing operation must be performed before the
signal is sampled. The anti aliasing filter, being an ideal filter is unrealizable. In practice, we

t
use a steep cutoff filter, which leaves a sharply attenuated residual spectrum beyond the

ne
folding frequency fs/2.

4.4 DISCRETE TIME FOURIER TRANSFORM

In mathematics, the discrete-time Fourier transform (DTFT) is one of the specific
forms of Fourier analysis. As such, it transforms one function into another, which is called

z.
the frequency domain representation, or simply the "DTFT", of the original function (which
is often a function in the time-domain). But the DTFT requires an input function that is
discrete. Such inputs are often created by sampling a continuous function, like a person's
voice.
ep
Given a discrete set of real or complex numbers:
discrete-time Fourier transform (or DTFT) of is usually written:
(integers), the
a de
.p
w
w
w

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t
ne
z.
ep
a de
.p
w
w

4.5 Inverse transform

w

The following inverse transforms recover the discrete-time sequence:

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The integrals span one full period of the DTFT, which means that the x[n] samples are
also the coefficients of a Fourier series expansion of the DTFT.

t
ne
Infinite limits of integration change the transform into a continuous-time Fourier
transform [inverse], which produces a sequence of Dirac impulses. That is:

z.
ep
de

4.6 Properties
a
.p
w
w
w

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t
ne
z.
ep
a de
.p

4.7 SYMMETRY PROPERTIES

w

The Fourier Transform can be decomposed into a real and imaginary part or into an
even and odd part.
w
w

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4.8 Z-transforms

t
ne
► Definition: The Z – transform of a discrete-time signal x(n) is defined as the power series:

z.
where z is a complex variable. The above given relations are sometimes called the direct Z -

ep
transform because they transform the time-domain signal x(n) into its complex-plane
representation X(z). Since Z – transform is an infinite power series, it exists only for those
values of z for which this series converges.

The region of convergence of X(z) is the set of all values of z for which X(z) attains a
finite value.
de
► For discrete-time systems, z-transforms play the same role of Laplace transforms do in
continuous-time systems
a

Bilateral forward Z transform

.p
w
w
w

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t
ne
z.
ep
► Using the definition requires a contour integration in the complex z-plane.
► Fortunately, we tend to be interested in only a few basic signals (pulse, step, etc.) Virtually
de
all of the signals we’ll see can be built up from these basic signals.

4.9 Z transform properties

a
.p
w
w
w

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z.
ep
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.p
w
w
w

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Sample Problem:

1. Obtain the z transform of,

t
ne
z.
ep
de
2. Find the inverse z transform of,
a
.p
w
w
w

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2 mark questions

1. What is the relation between Z transform and fourier transform of discrete time
signal. (APR/MAY 2010).
‚ƒ
X(•)=X(Z)|z= .This means Z transform is same as fourier transform when
evaluated on unit circle.

2. Define region of convergence with respect to Z transform. [MAY-11, 2015].

t
Region of convergence (ROC) is the area in Z plane where Z transform convergence

ne
.In other word, it is possible to calculate the X(z) in ROC.

3. State the initial value theorem of Z transforms. (APR/MAY 2010).

The initial value of the sequence is given as, X(0)=„ …†→y ‡(†).

z.
4. What is meant by aliasing? (MAY/JUN 2010).
ep
When the high frequency interferes with low frequency and appears as low
then the phenomenon is called aliasing.

5. Define Nyquist rate and Nyquist interval. (MAY/JUN 2010).

de
When the sampling rate becomes exactly equal to ‘2W’ samples/sec,for a give
bandwidth of W hertz, then it is called Nyquist rate .’
Nyquist interval is the time interval between any two adjacent samples.
Nyquist rate =2W hz&Nyquist interval=1/2W seconds.
a
.p

6. Define unilateral Z-Transform or one sided Z-transform [MAY-10]

The unilateral Z-Transform of signal x(t) is given as,
O
w

^(ˆ) = q (.)ˆ H‰
‰tG
The unilateral and bilateral Z-Transforms are same for causal signals.
w

7. State the final value theorem for z-transform. [may 2012]

The final value of a sequence is given as,
w

(∞) = Š‹Œ(y − •Hy )‡(†)

†→y

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8. Define DTFT pair. [dec 2012]

DTFT,
‡ • = ∑O *tHO * H‚•*
(analysis equation)

* = y/& RH ‡ • ‚•*
•• (Synthesis equation)

9. State the sampling theorem.

• A bandwidth signal of finite energy, which has no frequency components higher than

t
W hertz, is completely described by specifying the values of the signal at instants of

ne
time separated by 1/2W seconds.
• 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.

z.
10. Define two sided Z transform. [may 2010,2013]
The z- transform of the DT signal is given by,
O

ep
‡ † = q
*tHO

Here ‘z’ is the complex variable. The z- transform pair is denoted by,
* †H*

x(n) X(Z)
de
E

: = N {
HO
a

11. State the convolution property of z transform. [dec 2012]

The convolution states that,
.p

If x1(n) X1(z)
x2(n) X2(z)
then x1(n)* x2(n) X1(z)X2(z)
That is the convolution of two sequences in time domain is equivalent to
w

multiplication of their z-transforms.

w

12. State parseval‟s theorem.

Consider the complex valued sequences x(n) and y(n).If
x(n) ----_X(k)
y(n) ----_Y(k)
w

then x(n)y*(n)=1/N X(k)Y*(k)

13. Find Z transform of x(n)={1,2,3,4}

x(n)= {1,2,3,4}
X(z)= x(n)z-n

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= 1+2z-1+3z-2+4z-3.
= 1+2/z+3/z2+4/z3.

14. What z transform of (n-m)?

By time shifting property
Z[A (n-m)]=AZ-m sinZ[ (n)] =1

15. Obtain the inverse z transform of X(z)=1/z-a,|z|>|a|

t
Given X(z)=z-1/1-az-1

ne
By time shifting property
X(n)=an.u(n-1)

z.
ep
a de
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w
w
w

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UNIT V

LINEAR TIME INVARIANT DISCRETE TIME SYSTEMS

5.1 Introduction
A discrete-time system is anything that takes a discrete-time signal as input and
generates a discrete-time signal as output.1 The concept of a system is very general. It may be
used to model the response of an audio equalizer . In electrical engineering, continuous-time
signals are usually processed by electrical circuits described by differential equations.

t
For example, any circuit of resistors, capacitors and inductors can be analyzed using

ne
mesh analysis to yield a system of differential equations. The voltages and currents in the
circuit may then be computed by solving the equations. The processing of discrete-time
signals is performed by discrete-time systems. Similar to the continuous-time case, we may
represent a discrete-time system either by a set of difference equations or by a block diagram
of its implementation.

z.
For example, consider the following difference equation. y(n) = y(n-1)+x(n)+x(n-
1)+x(n-2) This equation represents a discrete-time system. It operates on the input signal
x(n)x(n) to produce the output signal y(n).

5.2 BLOCK DIAGRAM REPRESENTATION

ep
Block diagram representation of
a de
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w
w
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LTI systems with rational system function can be represented as constant-coefficient

difference equation

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• The implementation of difference equations requires delayed values of the

– input
– output
– intermediate results
• The requirement of delayed elements implies need for storage

t
ne
z.
• We also need means of
– addition
ep
– multiplication
de
Direct Form I

General form of difference equation

a
.p

Alternative equivalent form

w
w
w

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t
ne
z.
ep
a de
.p
w

• Cascade form
General form for cascade implementation
w
w

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t
Parallel form

ne
► Represent system function using partial fraction expansion

5.3 CONVOLUTIO N SUM

z.
The convolution sum provides a concise, mathematical way to express the output of
an LTI system based on an arbitrary discrete-time input signal and the system's response. The
convolution sum is expressed as,

ep
a de
.p
w
w

LTI systems are causal if

w

h[n] = 0 n < 0

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5.4 LTI System analysis using DTFT

t
ne
z.
5.5 ep
LTI SYSTEMS ANALYSIS USING Z-TRANSFORM
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t
ne
z.
Sample Problems:
1. Consider the system described by the difference equation.

ep
a de
.p
w
w
w

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2. Given y[-1]=1 and y[-2]=0. Compute recursively a few terms of the following 2nd
order DE:

t
ne
3. Compute the impulse response of the system described by,

z.
ep
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4. Obtain the structures realization of LTI system

t
ne
z.
ep
a de
.p
w
w
w

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t
ne
z.
ep
ade
.p
w
w
w

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5. Find the convolution of x(n)=[1,1,1,1,2,2,2,2] with h(n)=[3,3,0,0,0,0,3,3] by using

matrix method.

Solution: By using matrix method, N=8

t
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z.
ep
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w
w

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2 mark questions and answers

1. States the properties of convolution.(DEC 2009).
i).Commutative property of convolution
∗ ∗
x(n) h(n)=h(n) x(n)=y(n)
ii).Associative property of convolution

[x(n) ∗
xy * ] ∗
x& * =x(n )∗ [xy (*) ∗
x& (*)]

t
iii).Distributive property of convolution

ne
x(n) ∗
[ xy (*)+x& (*)] =x(n) ∗
xy (*)+x(n) ∗
x& (*).

2. Define non recursive and recursive of the following system.(MAY/JUN 2010).

z.
When the output y(n) of the system depends upon present and past inputs then it is
called non-recursive system. When the output y(n) of the system depends upon present
and past inputs as well as past outputs, then it is called recursive system.

3. Define convolution sum?

ep
If x(n) and h(n) are discrete variable functions, then its convolution sum
y(n) is given by,
de
y(n)=_ x(k) h(n-k)

4. If x(n) and y(n) are discrete variable functions, what is its convolution sum.
[dec 2013]
a

The convolution sum is,

∑O
stHO (‘):(. − ‘)
.p

5. Determine the system function of the discrete time system described by the
difference equation.
w

Y(n) = 0.5y(n-1)+x(n) [may 2012]

Taking z-transform of both sides,

Y(z) = 0.5z-1Y(z)+X(z)
w

H(z) = Y(z)/X(z) = 1/(1 – 0.5z-1)

w

6. A causal LTI system has impulse response h(n), for which the z-transform is
H(z) = (1+z-1)/(1-0.5z-1)(1+0.25z-1). Is the system stable? Explain.

H(z) can be written in terms of positive powers of z as follows:

H(z) = z(z+1)/(z-0.5)(z+0.25)

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Poles are at p1 =0.5 and p2 = -0.25. Since both the poles are inside unit circle. This system
is stable.

7. Check whether the system with system function H(Z) = (1/1-0.5z-1)+(1/1-2z-1) with
ROC |z| < 0.5 is causal and stable? [dec 2013]
H(z) = z/(z – 0.5) + z/(z - 2). Poles of this system are located at z = 0.5 and z = 2. This
system is not causal and stable, since all poles are not located inside unit circle.

t
8. Is the discrete time system described by the difference equation y(n) = x(-n) is

ne
causal? [may 2013]
Here y(-2) = x(-(-2)) = x(2). This means output at n=-2 depends upon future inputs.
Hence this system is not causal.

z.
9. Consider a system whose impulse is h(t) = e-|t|. Is this system is causal or non causal?
[dec 2011]
Here h(t) = e-|t|
= e-t for t>=0
= et for t<0
ep
Since h(t) is not equal to zero for t<0, the system is non causal.

10. Find the step response of the system if the impulse response [may 2011]
ℎ . =’ .−2 −’ .−1
de
Solution:
Y(n) = h(n)*u(n), since x(n) = u(n), step input.

= ’ .−2 ∗u . −’ .−1 ∗u .
a

= u(n - 2) – u(n - 1)
.p

11. Obtain the convolution of

a) X(n) * ’ .
b) X(n) * [h1(n)+h2(n)]
w

Solution:
x(n)* ’ . = ’ .
w

x(n)*[h1(n)+h2(2)] = x(n)*h1(n) + x(n)*h2(n)

w

12. List the steps involved in finding convolution sum?

o folding
o Shifting
o Multiplication
o Summation

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13. Consider an LTI system with impulse response h(n)= δ(n-*“ ) for an input x(n), find the
Y( ‚• .( NOV/DEC 2003).

Here is the spectrum of output. By convolution theorem we can write,

Y( ‚• )=H( ‚• )X( ‚•
Here H( ‚• )=DTFT{ δ(n-*“ )}= H‚•*“
Y( ‚• )= H‚•*“ X( ‚• .

t
14. List the properties of convolution?
o Commutative property of convolution

ne
x(n) * h(n) = h(n) * x(n) = y(n)
o Associative property of convolution
[ x(n) * h1(n)] * h2(n) = x(n) * [h1(n) * h2(n)]
o Distributive property of convolution
x(n) * [h1(n) + h2(n)] = x(n) * h1(n) + x(n) * h2(n)

z.
15. Define system function?
H(z)= Y(z) is called system [Link] is the z transform of the unit sample X(Z)
response h(n) of the system. ep
a de
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w
w
w

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