INTERVIEW

GUIDANCE MATERIAL
ELECTRONICS
ENGINEERING
SIGNALS & SYSTEMS
 Signals & Systems
Interview Guidance 03. Periodic And Non Periodic (Or
1. INTRODUCTION
(A) Continuous - time (CT) or Analog
signals: They are Continuous in time and
continuous in amplitude. Continuous in
amplitude means that the amplitude can
assume any value in the continuous range
from – to + .
(B) Discrete - time (DT) Signals: Discretized
in time and continuous in amplitude. Most
DT signals arise from sampling CT signals.
(C) Digital Signals: Discretized in time and
quantized in amplitude. If the amplitude of a
signal can assume only a value from a finite
set of numbers, the amplitude is said to be
discretized or quantized.
CLASSIFICATION OF SIGNALS
01. Energy and Power signals:
 A signal x (t) (or) x (n) is called an energy
signal if total energy has a non –zero finite
value i.e. 0  Ex  
 A signal is called a power signal if it has
non – zero finite power i.e, 0 Px  .
 A signal can’t be both an energy and a
power signal simultaneously.
02. Even and Odd signals:
 A continuous signal x(t) is even if x(t)= x(-t)
 A discrete signal x(n) is even if x (n) = x(-n)
 A continuous signal x(t) is odd if x (t)= -x(-t)
 A discrete signal x(n) is odd if x (n)= -x(-n)
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Aperiodic) Signals:
 A periodic function is one which has been
repeating an exact pattern for an infinite
time and will continue to repeat that exact
pattern for an infinite time.
 A signal is periodic if g (t) = g (t + nT) for
any integer “n” ; T  period of a function.
 A signal is aperiodic if g(t)  g (t + nT)
Causal & Noncausal Signals:
 A continuous time signal is causal if x (t) =
0 for t  0, otherwise the signal x(t) is non
causal.
 A discrete time signal is causal if x (n) = 0
for n  0, otherwise the signal x(n) is non
causal
Deterministic and Random signals:
 If the value of a signal can be predicted for
all time (t or n) in advance without any
error, the signal is a deterministic signal.
 Signals whose values can’t be predicted
with complete accuracy for all time are
known as random signals.
 Random signals are generally characterized
by mean, mean square value.
Systems and Classification
A System, designated T, is a mathematically
describable operation or transformation that
acts on an input signal to produce an output.
Input Output
Signal Signal
System T { }
01. Linear And Non Linear Systems:
A system is linear if it follows additivity
and homogenity property.
 2 Signals & Systems
For the system to be linear, it should satisfy
2 properties
(A) x1(t) y1(t) and x2(t)  y2(t)
Then x1(t) + x2(t)  y1(t) + y2(t)  Additivity
(B) cx(t)  cy(t)  Scaling (or) Homogenity
(or)
ax1(t) + bx2(t)  ay1(t) + by2 (t) Superposition
02. Time – Invariant (Shift – Invariant) &
Time – Variant (Shift – Dependent)
Systems:
A system is T.I. if the input output
characteristic don’t change with time. T.I.
implies that the shape of the response y(t)
depends only on the shape of the input x(t)
and not on the time when it is applied. If
the input is delayed by “t0” the output is
also delayed by the same amount and is
simply shifted replica of the original output.
For T.I system if x (t)  y (t);
Then x (t – t0)  y(t – t0)
03. Causal & Non Causal System:
A causal (or) non – anticipatory system is
one whose present response does not
depend on future values of the input (or) a
system is causal if its output at t = t0
depends on the values of the input in the
past t  t0 and doesn’t require future value
of input (i.e.(t  t0)). Systems whose present
response requires knowledge of future
values of the input are termed non causal.
04. Static (or) Memoryless & Dynamic (With
Memory) System:
A system is static if its output at t = t0
depends only on the value of the input at
t = t0 and no other value of the input signal.
The response of a dynamic system depends
on past (and /or future) inputs.
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05. Stable and Unstable System:
 A system is said to be BIBO stable if and
only if every bounded input results in a
bounded output If x t   Mx   then
yt   My  
06. Invertible & Inverse System:
 A system is said to be invertible if the input
of the system can be recovered from the
output
 In any event, a system is not invertible
unless distinct inputs applied to the system
produces distinct outputs
2. LTI (LSI) systems
 Convolution is a formal mathematical
operation, just as multiplication, addition,
and integration. Addition takes two numbers
and produces a third number, while
convolution takes two signals and produces
a third signal. Convolution is used in the
mathematics of many fields, such as
probability and statistics
 An L.T.I System is always considered w.r.t
impulse response denoted as h(t) or h(n).
 If the input is impulse, then the output is
impulse response.
 SIFTING property states that any signal can
be produced as a combination of impulses.
 Convolution may be regarded as a method
of finding the zero – state response of a
relaxed LTI system.
 Any DT signal is the sum of scaled and
shifted unit impulses.
 Convolution may be treated as flip – shift –
multiply – time – area method.
 In the convolution integral time “t”
determines relative location of h(t-) w. r.t.
x(). The convolution will yield a non zero
result only for those value of “t” over which
h(t - ) & x() overlap.
 3 ISRO - Interview Guidance Programme

 By using F.S, a non-sinusoidal periodic

Properties of LTI systems function can be expressed as an infinite sum
of sinusoidal functions.
Causality: Stability:

Symmetry Condition a0 an bn
h(t) = 0; t < 0  | h(τ ) | dτ  


Even g(t) =g (-t) ? ? 0
h(n) = 0; n < 0  | h (n ) |  
n  Odd g(t) = -g(-t) 0 0 ?
=0; = 0;
Memory less : Invertibility & Inverse: Half-wave (or) g(t) = -g (t ± n even n even
0
Rotational T/2) =?; =?;
h(t) = 0 for t  0 h(t) * hinv(t) = (t) n odd n odd

h(n) = 0 for n  0 h(n) * hinv(n) = (n) Notes:

 a0, the DC offset term, can be non-zero even
Step response of an LTI systems though all the other an’s are zero
 An odd-harmonic function is one where the
Step response is the response when the second half of its period is the negative of
input is unit step function. the first half.
 An even-harmonic function is one where
x(t)=u(t) L.T.I y(t)=s(t) the second half of its period is exactly the
System h(t) same as the first half.
 Shifting a signal left/right in time does not
t affect whether or not it is odd-harmonic.
d
s ( t )   h ( ) d ; h ( t )  s( t )  Shifting a signal up/down (adding a DC

dt offset) does not affect whether it is odd-
harmonic, other than adding a term in the
Fourier series at Zero frequency.
x(n)=u(n) L.T.I y(n)=s(n)
 An odd-harmonic function does not have to
System h(n) be odd.

System with Periodic Inputs

n
s(n )   h (k ) ; h(n) = s(n)s(n–1)
k  
 If we apply an Input of the form x(t)=ejt,
then the output is y(t)=ejtH(), where H()
is known as frequency response of the
system.
3. Fourier Series
 Knowing H(), we can determine whether
the system amplifies or attenuates a given
 Representing CT signals as superposition of sinusoidal component of the input & how
complex exponentials leads to frequency – much of a phase shift adds to that particular
domain characterizations component.
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 4 Signals & Systems

 The response y(t) of an LTI system  In time reversal property only the phase
to the periodic input x(t) is spectrum changes but the magnitude

spectrum remains same.
y( t )   C Hn e
n  
n 0
jn0 t
 Time - delay in a signal causes a linear
phase shift in its spectrum. Shifting in time
doesn’t alter the magnitude spectrum of the
4. Fourier Transform signal.
 Modulation spreads the signal spectrum to
 Fourier Transform (F.T.) provides a higher frequencies.
frequency domain description of time  Differentiation attenuates the HF
domain signals and is extension of F.S to components of the signal. Differentiation
non-periodic signals. destroys any d.c. component of x(t) i.e.,
 CTFT expresses signals as linear F.T. of the differentiated signal at  = 0 is
combination of complex Sinusoids zero.
 Transformation makes the analysis of signal  Convolution in time-domain corresponds to
much easier because certain features multiplication in frequency domain.
which may be obscure in one form may be
obvious in other form Rayleigh’s Energy theorem (or) Parseval’s
 Spectrum of F.T. is continuous whereas Power theorem:-
spectrum of F.S is discrete.
It states that the energy (or) power in the
Convergence of F.T time domain representation of a signal is
equal to the energy (or) power in the
 F.T. is defined for all stable signals i.e., frequency domain.


 x (t ) dt   Distortion less Transmission


 Periodic signals, which are neither  For distortion less transmission, magnitude
absolutely integrable nor square integrable response must be a constant, phase response
over an infinite interval, can be considered must be a linear function of  with slope –t0,
to have F.T. if impulse functions are where t0 is delay in output with respective to
permitted in the transform. input.
 x(t) have a finite number of discontinuities k
| H() | = k
and finite number of maxima and minima
within any finite interval. 

Properties of Fourier Transform

H() = () = -t0
 Expansion in one domain corresponds to 
compression in other domain. Linear
scaling in time by a factor of “a”
corresponds to linear scaling in frequency
by a factor of “1 / a”.
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Hilbert Transform
The Hilbert transform is an operation that
shifts the phase of x(t) by - /2, while the
amplitude spectrum of the signal remains
unaltered. An ideal H.T. is an all pass 900
phase shifter. H.T. is used in number of
application such as representation of band
pass signals, phase shift modulators,
generation of SSB.
F.T of Periodic signals
 F.T. of a periodic signal consists of a
sequence of equidistant impulses located at
harmonic frequencies of the signal
 
  (t  nT )  2 / T  (  n )
n  
s s
n  
s
 spectral overlap, aliasing & irreversible loss
x p t   C n e jn0 t ;
n  

X p   2  C n   n0 
n  
CORRELATION
 It provides a measure of the similarity
between 2 waveforms as the function of
search parameter.
 An application of correlation to signal
detection in a radar, where a signal pulse is
transmitted in order to detect a suspect
target. If a target is present, the pulse will be
reflected by it. If the target is not present,
there will be no reflection pulse, just noise.
By detecting the presence or absence of the
reflected pulse we confirm the presence or
absence of the target.
 In digital communication, the important
thing is that 1s and 0s in the data stream be
distinguishable from each other so the
receiver can reproduce the bit pattern that
was transmitted.
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SAMPLING THEOREM
Ideal sampling:- describes a sampled
signal as a weighted sum of impulses, the
weights being equal to the values of the
analog signal at the impulse locations.
 An analog signal band-limited to a
frequency m can be sampled without loss
of information if the sampling rate  2m
 To reconstruct the original signal, it is
necessary that the B.W. of the original
continuous signal being sampled is smaller
than half the sampling frequency [‘Nyquist
frequency’]
 Critical sampling rate is known as Nyquist
rate N.R = 2m
 If s < 2m (under sampling) causes
of information
5. Laplace Transform
 L.T expresses signals as linear combination
of complex exponentials, which are eigen
functions of D.E which describe continuous
– time L.T.I systems.
 The primary role of the L.T in engineering
is the transient & stability analysis of
Causal L.T.I systems.
 L.T provides a broader characterization of
systems & their interaction with signals
than is possible with F.T.
 In addition to its simplicity, many design
techniques in circuits, filters & Control
systems have been developed in L.T.
domain.
 Consider applying an input of the form
x(t) = est (where s =  +j then the output is
y(t) = estH(s) where H(s) is transfer function
of the system.
 6 Signals & Systems

 The range of values of ‘s’ for which the  Some sequences are not absolutely
 summable, but they are square summable.
equation X(s) =  xt e dt is satisfied i.e.,
st
 There are signals that are neither absolutely


summable nor have finite energy, but still
t
dt   is known as R.O.C of L.T. have DTFT.
 x ( t )e
  Parseval’s relation is known as conservation
 L.T. calculated on the j-axis ( = 0) is F.T. of energy theorem, because DTFT operator
 If the L.T. X(s) of x(t) is rational, then if preserves energy when going from time
x(t) is right sided the ROC is the region in domain to frequency domain.
the s-plane to the right of the rightmost pole
and if x(t) is left sided, ROC is left of the 7. Z‐Transform
left most pole.
 L.T. of impulse response is known as  Discrete-time counterpart of L.T. is Z.T.
system (or) transfer function.  For a D.T.L.T.I system with impulse
response h(n), the response y(n) of the
Stability system to a complex exponential Input of
 An LTI system is stable if and only if the the form Zn is y(n) = ZnH(Z) where H(Z) is
ROC of the system function H(s) includes known as transfer function of the system.
j axis.  Z.T. calculated on the unit circle is D.T.F.T.
 The range of values of ‘Z’ for which
L.T. of switched periodic signals:- 
equation Xz    xn z n
is defined i.e.
n  
T
  
 x(t) e
st
dt   x (n )r   
n is R.O.C. of Z.T.
X p (s)  0 n   
1 e  s T
 DTFT is defined only for stable signals
where as Z.T. is defined for unstable signals
6. DTFT also.
 The primary role of Z.T. in engineering
are the study of system characteristics &
 The DTFT describes the spectrum of the derivation of computational structures
discrete signals & formalizes that discrete for implementing discrete systems on
signals have periodic spectra. The computers.
frequency range for a discrete signal is
unique over ( -, +) (or) (0,2)
 X(ej) is decomposition of x(n) into its Causality & Stability
frequency components.
 A D.T. LTI system with rational system
Convergence of DTFT function H(Z) is causal if and only if (a)
ROC is exterior of a circle outside the
A sufficient condition for existence of outermost pole (b) with H(Z) expressed as a
 ratio of polynomials in Z, order of Nr. can’t
DTFT is  x (n )  be greater than order of Dr.
n  
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Summary of Transfer function stability

Previous Interview Questions with Solutions
conditions
Pole location Pole location Stable 01. Classify signals on the basis of their
of H(s) in the of H(z) in the
s plane z-plane
bandwidths.
Left-half plane Inside the unit Yes Ans: Base Band Signals: signal with
<0 circle significantly low frequency or 0 frequencies
|Z| < 1 where most of the signal lies near to f = 0 is
Right – half Outside the No a base band signal.
plane unit circle
>0 |Z| > 1 Ex: A pulse or gate function
j axis On the unit Marginally
( = 0) circle Band limited signals:
|Z| = 1 A signal is said to be band limited signal
 An LTI system is stable if and only if the if |X(f)|  0 ; for |f| < 
ROC of H(Z) includes the unit circle. And |X(f)| = 0 ; else where
Ex: sinc function

Relation between S-plane & Z-plane x(t) X(f)

The j  axis in the S-plane should map onto –B 0 B

the unit circle in the Z-plane & the left-half
plane of the S-plane should map into the Band pass signals:
inside of the unit circle in the Z-plane. Thus A signal is said to be bandpass signal whose
a stable analog filter will be converted to a spectrum |X(f)| is not equal to zero for fa < |f|
stable digital filter. < fb and is zero else where
Z = eST   =  T i.e., |X(f)|  0 ; fa < f < fb
= 0; else where
Broad band signals:
8. DFT & FFT A signal which is distributed over a wide
range of frequencies is called as broad band
signal.
Narrow band:
 The Fourier series describes periodic signals It has spectrum that is centered/localized
by discrete spectra, whereas the DTFT around a frequency
describes discrete signals by periodic
X(f)
spectra. As a result, signals that are both d c
discrete and periodic in one domain are also a
periodic & discrete in the other. This is the b
basis for the formulation of the DFT.
 Sampled version of D.T.F.T. spectrum is 0 f0 f
D.F.T.
 The DFT & its IDFT are also periodic with (a) Broad band signals
period N, and it is sufficient to compute the (b) Band limited
results for only one period (0 to N-1). (c) Narrow band
(d) Base band
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 8 Signals & Systems

02. Explain sampling theorem. (c) Under sampling i.e. fs<2W

Ans: A band limited signal of finite energy,
which has no frequency components higher 1/Ts
than W Hertz, is completely described
by specifying the values of the signals
at instants of time separated by –2fs –fs 0 fs–W fs 2fs f
1 In under sampling case where fs <2W the
seconds.
2W spectral nodes overlap and thus signal
A band limited signal of finite energy, distortion takes place leading to aliasing
which has no frequency components higher effect.
than W Hertz, may be completely recovered
from a knowledge of its samples taken at 05. What is the reason for the occurrence of
the rate of 2W samples per second. Aliasing?
03. What is Nyquist rate and Nyquist interval? Ans: Generally the signal x(t) is strictly band
Ans: The sampling rate of 2 W samples per limited, but in practice, an information
second, for a signal bandwidth of W Hertz, bearing signal is not strictly band limited,
1 with the result that some degree of under
is called Nyquist rate; its reciprocal sampling is encountered. Consequently,
2W some aliasing is produced by the sampling
(measured in seconds) is called Nyquist process.
interval.
04. What is Aliasing? 06. What are the steps to be taken to overcome
Ans: Aliasing refers to the phenomenon of a high the effect of Aliasing?
frequency component in the spectrum of the Ans: The following steps are to be taken to
signal seemingly taking on the identity of a 1. Prior to sampling, a low pass anti-
lower frequency in the spectrum of its aliasing filter is used to attenuate those
sampled version. X(f) high frequency components of the signal
1 that are not essential to the information,
x(t) being conveyed by the signal.
2. The filtered signal is sampled at a rate
–W 0 W slightly higher than the Nyquist rate.
(a) Over sampling i.e. fs>2W
07. What is Fourier Transform?
1/Ts Sol: The Fourier Transform is a generalization
of the Fourier Series. Strictly speaking it
applies only to continuous and aperiodic
W fs-W
functions, but the use of the impulse
fs f
function allows the use for discrete signals
(b) Sampling at the Nyquist rate i.e. fs=2W also. Physical significance of Fourier
Transform:
1/Ts
The Fourier Transform converts a set of
time domain data vectors into a set of
–2fs –fs W fs 2fs f frequency (or per time) domain vectors.
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is a unique transformation and the 'Inverse

08. What are the applications of Fourier Laplace Transform' also exists.
Transform? Note that est is also an eigen function of the
Sol:  Designing and using antennas LSI system only if H(s) converges. The
 Image Processing and filters range of values for which the expression
 Transformation, representation, and described above is finite is called as the
encoding Region of Convergence (ROC). In this case,
 Smoothing and sharpening the region of convergence is Re(s) > 0.
 Restoration, blur removal, and Wiener
filter Thus, the Laplace transform has two parts
 Data Processing and Analysis which are , the expression and region of
 Seismic arrays and streamers convergence respectively. The region of
 Multibeam echo sounder and side scan convergence of the Laplace transform is
sonar essentially determined by Re(s).
 Interferometers — VLBI — GPS
 Synthetic Aperture Radar (SAR) and 10. What are the properties of ROC for Laplace
Interferometric SAR (InSAR) transform
 High-pass, low-pass, and band-pass filters Ans:  The ROC of Laplace transform of
 Cross correlation — transfer functions –– function f(t), F(s) consists of strips
Coherence parallel to j  axis in complex variable
 Signal and noise estimation — encoding plane (s-plane).
time series.  The ROC does not contain any poles.
09. What is Laplace transform? What are Eigen  If h(t) is a time limited signal and is
functions ? Laplace Transformable, then its ROC
Ans: The response of a Linear Time Invariant will be the entire s-plane.
system with impulse response h(t) to a  The region of convergence is always
complex exponential input of the form est between two vertical lines in s-plane.
can be represented in the following way : These vertical lines need not be in finite
region.
Continuous Time 
est  h ( )e
s ( t  )
d  If x(t) is right sided signal, ROC is right
LTI System
h(t)

side of right most pole.

 If the signal x(t) is left sided signal,
e st  h ( )e s d ROC is left side of left most pole.


11. What is Gibbs phenomenon?
 h ( )e
s
H (s)  d

Ans: We can approximate a signal having a
Where H(s) is known as the Laplace Fourier Series expansion by taking a finite
Transform of h(t). We notice that the limits number of terms of the expansion.
N
are from [-∞ to +∞] and hence this i.e: ~x N ( t )  C e k
j2 f 0 kt
is an approximation
transform is also referred to as Bilateral or k  N

Double sided Laplace Transform. There to the periodic signal x(t).

exists a one-to-one correspondence between ~
x N ( t ) is also called a Partial Sum. We
the h(t) and H(s) i.e. the original domain would obviously expect that as the number
and the transformed domain. Therefore L.T. of terms taken is increased, this summation
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 10 Signals & Systems

would become a better and better 3) The signal x(t) should have only a finite
approximation to x(t), i.e ~ x N ( t ) would number of maxima and minima in one
approach x(t) uniformly. period. An example of a function which
Indeed this happens in regions of continuity has infinite number of maxima
of the original signal. However, at the and minima is: a periodic signal with
points of discontinuity in the original signal, period 1, defined on
an interesting phenomenon is observed. The 1
x(t) = sin  
partial sum oscillates near the point of t
discontinuity. We might expect these
If the signal satisfies the above conditions,
oscillations to decrease as the number of
then at all points where the signal is
terms taken is increased. But surprisingly,
continuous, the Fourier Series converges to
as the number of terms taken is increased,
the signal. However, at points where the
although these oscillations get closer and
signal is discontinuous (Dirichlet conditions
closer to the point of discontinuity, their
allow finite number of discontinuities in a
amplitude does not decrease to zero, but
period), the Fourier Series converges to the
tends to a non zero limit. This phenomenon
average of the left and the right hand limits
is known as the Gibb's Phenomenon
of the signal. Mathematically, at a point of
Mathematically, this means if the periodic
lim x ( t )  lim x ( t )
signal has discontinuities, its Fourier Series
discontinuity t0, ~
x t  x t 0
x (t 0 )  0
does not converge uniformly. 2
In practice, the restrictions imposed on
12. What are Dirichlets conditions of Fourier signals by the Dirichlet conditions are not
series? very severe, and most of the signals we will
Ans: Dirichlet conditions for pointwise deal with satisfy these conditions.
convergence are:
1) x(t) should be absolutely integrable over 13. What are applications of DSP?
a period. A signal that does not satisfy Ans: Some selected applications of digital signal
this condition is x(t) = tan(t) as:- processing that are often encountered in
/2
daily life are listed as follows:
 tanx  dx
 / 2
does not exist.
1. Telecommunication: Echo cancellation
2) x(t) should have only a finite number of in telephone networks.
discontinuities over one period. 2. Military Radar signal processing
Furthermore, each of these 3. Consumer electronics Digital Audio/TV
discontinuities must be finite. An
4. Instrumentation and control
example of a function which has infinite
number of discontinuities is illustrated 5. Image processing image representation,
below. The function is shown over one image compression
period. 6. Speech processing speech analysis
methods are used in automatic speech
recognition
7. Medicine Medical diagnostic
instrumentation such as computerised
tomography (CT)
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8. Seismology DSP techniques are The Laplace transform of x(t) is

employed in geophysical exploration 

 x ( t )e
 st
for oil and gas. X(s) = dt

9. Signal Filtering Removal of unwanted
background noise. X(s) = FT {x(t)e–t}
X(  ) = X(s) s j
14. What are the properties of ROC for
Z - Transform? i.e Laplace transform obtained on the j
Ans: 1. For a finite length signal, ROC is entire axis is called Fourier transform.
Z-plane, except perhaps for Z=0 and/or Note:If ROC include the j axis then only it is
Z= possible to find Fourier transform from
Laplace transform
2. For a right-sided signal ROC is outside
a circle whose radius is largest pole in
magnitude. 17. What is the relation between DTFT &
Z-transform?
3. For a left sided signal, ROC is inside a 

) =  x ( n )e
 j n
j
circle whose radius is smallest pole in Ans: The DTFT of x(n) is X(e
n  
magnitude.

4. For a two-sided signal, ROC is annulus The Z-transform of x(n) is X(z)   x (n )z n

bounded by largest & smallest pole n  

radius.
z = r.ej
15. Why are ideal filters non realizable? If r = 1, z = ej
Ans: For a physically realizable system, h(t) must X(z) = DTFT [x(n)r–n]
be causal i.e., h(t) = 0 for t < 0. In the X(ej) = X(z) ze j
frequency domain, this condition is known i.e Z-transform obtained on the unit circle is
as Paley – Wiener Criterion which states called DTFT.
that the necessary and sufficient condition
for the amplitude response | H ( )| to be Note: DTFT can be obtained from Z-transform
realizable is only if ROC includes the unit circle.

n H
 1  2 d   18. What is the Fourier transform of (t) ?

Ideal filters are non causal, unstable and
 x ( t )e
 j t
Sol: X() = dt
physically unrealizable in the sense that 
their characteristics can not be achieved
with a finite number of elements. From sifting property


 ( t ) e
 jt
X() = dt = e  jt =1
16. What is the relation between FT and LT ? 
t 0

Ans: The Fourier transform of x(t) is


19. What is the FT of unit step function ?
 x ( t )e
 j t
X() = dt

Ans: sgn(t) = 2u(t) – 1
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 12 Signals & Systems

1 1 Time invariant Property:

u(t) =  sgn( t ) y(t) = T[x(t)]
2 2
FT 1 2  y1(t) = T[x(t – t0)]
u(t)  () +   If, y1(t) = y(t – t0) then the system is
2  j  called TIV.
1
u(t)  () +
j 23. Why sampling is used?
Ans: Sampling is used to convert continuous time
signal to discrete time signal and
20. When two signals are called orthogonal ?
multiplexing.
Ans: Two signals x1(t) & x2(t) are orthogonal if


 x (t)x
1 2 ( t ) dt = 0 24. What is the relation between LT & ZT ?
 Ans: x(t) = e–at u(t)
ex: cos m0t, cos n0t [mn] [ta t ta+To] 1
Xs  
sa
21. Laplace transform of some basic functions x t  t nT  x nTs   e  anTs u nTs 
like unit step. s


Ans: (t)  1 Xz  
1  xnT z s
n

u(t)  n  
s 

1  e  anTs
u nTs z  n
r(t)  2 n  
s
z
1 
e at u ( t )  z  e aTs
sa
Pole of X(s) is S = – a
Pole of X(z) is z = e–aTs
22. What are the properties to be satisfied by
So, z = esTs
LTI system.?
Ans: Linear system: A system is called linear i.e left half of s-plane mapped to inside the
system is if it satisfies superposition and unit circle
homogenity property. right half of s-plane mapped to outside the
unit circle and j axis mapped to on the unit
circle.
Superposition: y(t) = T[x(t)]
y1(t) = T[x1(t)]
y2(t) = T[x2(t)] 25. What is the difference between DFT &
y3(t) = T[x1(t) + x2(t)] DTFT?
If y3(t) = y1(t) + y2(t) then it satisfies Ans: The DFT of x(n) is
superposition. N 1 j
2 nk
X(k) =  x ( n )e
n 0
N
k = 0 to N–1

Homogenity: y(t) = T[x(t)]

y1(t) = T[c x(t)] The DTFT of x(n) is
If y1(t) = c y(t) then it satisfies homogeneity 

property. X(ej) =  x ( n )e
n 
 jn

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 13 ISRO - Interview Guidance Programme

X(k) = X(e j ) 2 k = 0 to N–1 Digital Signals: Discretised in time and

 k
N quantized in amplitude. If the amplitude of
DTFT is a continuous and periodic a signal can assume only a value from a
DFT is a discrete and periodic finite set of numbers, the amplitude is said
to be discretised or quantized.
Sampled spectrum of DTFT is called DFT.
Engineers must model 2 distinct physical
phenomena. First physical systems are
26. What is the physical sense of negative modeled by mathematical equations & the
frequency in Fourier transform ? second one physical signals are modeled by
Ans: Positive and negative frequencies denote mathematical functions.
opposite direction of rotation i.e, opposite
phase angle. Negative frequency is included 30. What was the output of a bandwidth LPF
in the spectrum to recover original real with bandwidth ‘W’ when an arbitrary
component. signal is passé through it.
Ans: When an arbitrary signal is passed through a
27. What is homogenous system? LPF of cut-off frequency ‘W’, the
Ans: The output for the scaled input is equal to frequencies lower than ‘W’ in the signal are
the scaled output with the same scaled allowed, frequencies greater than ‘W’ are
factor then system called the homogenous rejected.
system.
y(t) = T[x(t)] 31. DFT and its significance
y1(t) = T[cx(t)] Ans: Discrete form of DTFT is called DFT.
The DFT is one of the most powerfull tools
y1(t) called output for scaled input
in DSP which enables us to find the
If y1(t) = cy(t) then system called spectrum of a finite duration signal.
homogenous system.
32. FFT and its advantages along with the order
28. What is a signal and a system? of complexity in both DFT & FFT
Ans: Signal: signal is an integration about which Ans: FFT is a algorithm used to calculate DFT.
some amount of information is conveyed. FFT algorithm takes less computations than
System: An operator that maps relation direct method of DFT.
between input and output is called system. Number of complex multiplications in
direct method DFT = N2
29. What is analog signal, discrete signal and Number of complex additions in direct
digital signal. method of DFT = N(N – 1)
Ans: Continuous – time (CT) or Analog Number of complex multiplications using
signals: Continuous in time and N
FFT = log 2 N
continuous in amplitude. Continuous in 2
amplitude means that the amplitude can Number of complex additions using
assume any value in the continuous range FFT = N log 2 N
from - to + .
33. What are IIR & FIR filter
Discrete – time (DT) Signals: Discretised Ans: IIR & FIR filters are two classes of digital
in time and continuous in amplitude. Most filter IIR filters are recursive filters and
DT signals arise from sampling CT signals. contain both poles & zeros.
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 14 Signals & Systems

FIR filters are non-recursive filters and 12. What are disadvantage of Fourier transform
contains only zeros, all poles lies inside the over Laplace transform?
unit circle.
13. What is the Fourier transform of auto
34. Linear phase criteria for FIR filter correlation of a signal?
Ans: The condition for linear phase FIR filter is
h(n) =  h(N – 1 – n). 14. What are the applications of auto
correlation and cross correlation?

15. What is Hilbert Transform? How does it

Interview Questions differ from Fourier transform? What are
applications of Hilbert transform?

01. Why are Fourier series, and Fourier 16. Which of the following can be a suitable
transform required? Why is Laplace autocorrelation function?
transform required when we have Fourier 1. Atri(/0) ; where tri(x) is unit triangular
transform? function
2. Asin(0)
02. What is ‘s’ in Laplace transform? Give 3. Acos(0)
examples where Laplace transform exist 4. A rect (/0)
and Fourier transform does not exist
17. Why are sinusoidal signals preferred over
03. Why is one sided Laplace transform needed. other signals for signal analysis?
What is discrete time counter part of
Laplace transform and what is the relation
between the two of them.

04. How is the transform used in the design of

the filter?

05. What are the applications of FFT?

06. Explain convolution practically?

07. What is causality and stability of a system?

How is a system checked for causality and
stability?

08. What is interpolation and decimation?

09. What is difference between DTFT and

DFT?

10. What are the application of Laplace

transform?

11. Compare Laplace transform and Fourier

transform?
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