Signals Systems PDF

ACE
Engineering Academy
Hyderabad | Delhi | Pune | Bhubaneswar | Bengaluru | Lucknow | Chennai | Vijayawada | Visakhapatnam | Tirupati | Kukatpally | Kolkata | Ahmedabad
Signals & Systems

(Classroom Practice Questions)
03. Signum function: Sgn(t)

1. Introduction
Z] 1;
]] t>0
01. Unit step function, u(t): is usually employed to Sgn (t) = [ ]] 0; t = 0
switch other signals ON or OFF ] − 1; t < 0
u(t) \
u (t) = (
1; t>0 Sgn (t)

1
0; t<0 1
0
t
0 t
–1
u(t) at t = 0 is discontinuous,
u(0) = (1/2).
04. Unit ramp function: r(t)
Property 1:
k r(t)
u (t − t 0) = [u (t − t 0)] t; t > 0
r (t) = )
with “k” any positive integer 0; t = 0
Property 2: t
t0
u (at − t 0) = u (t − a ), a ! 0, a > 0 0
An example of unit step function is the output r (t) = tu (t)
of 1V dc voltage source in series with a switch An example of a ramp function is the linear -
that is turned on at t = 0 sweep waveform of a cathode - ray tube.
02. Rectangular (or Gate) Pulse: A rect(t/2a)

05. Triangular function tri(t)
A 1
Width = 2
Height = 1
Area = 1
t t
−a 0 a –1 0 1
The rectangular function is the result of an

ON - OFF switching operation of a constant
voltage source in an electrical circuit.
2 Signals & Systems
06. Sampling function: (iii)

3
sin x sin πx # δ (t) dt = 1:

Sa (x) = x Sinc (x) = πx -3
Sa(x) Area under unit impulse function is one.

1
(iv) δ(–t) = δ(t)
(v) Product property

x(t) δ(t – t0) = x(t0) δ(t – t0)
x if x(t) is continuous at t = t0
-2π -π 0 π 2π
Eg: (2t + 1) δ(t – 3) = 7δ(t – 3)
(vi) Sifting property
Sinc(x) t2
x (t 0); t1 # t 0 # t2
1 # x (t) δ (t - t 0) dt = )
t1
0 ; elsewhere
Eg: # (t + t2) δ (t − 3) dt = 3 + 32 = 12
-2
x
–2 –1 0 1 2
Energy and Power signals:

• A signal x(t) (or) x(n) is called an energy
07. Unit impulse function (or) Dirac delta function:
signal if total energy has a non-zero finite
δ(t)
value 0 < Ex < ∞
The impulse function is not a function in the
• A signal is called a power signal if it has
ordinary sense, since it is zero at every point
non-zero finite power i.e, 0 < Px < ∞.
except t0, where it is unbounded. However the
• A signal can’t be both an energy and
area under a unit impulse function is equal to
power signal simultaneously.
unity.
• The term instantaneous power is reserved
Many physical phenomenon such as point
for the true rate of change of energy in
sources, point charges, voltage sources acting
a system. In most cases, when the term
for very short time can be modeled as delta
power is used it refers to average power
functions.
i.e, the average rate of energy utilization,
(i) δ(0) → ∞
a constant quantity independent of time.
(ii) δ(t) = 0, t ≠ 0
T
E x (t) = Lt # x (t) 2 dt
δ (t) T"3
-T
T
1 #
Pavg x (t) = Lt x (t) 2 dt
T " 3 2T
-T
N
E x (n) = Lt / x (n) 2
t N " 3 n =-N
0
1
N
Pavg x (n) = Lt / x (n) 2
N"3
2N + 1 n = - N
3 Signals & Systems
Periodic And Non Periodic

(Or Aperiodic) Signals:
A periodic function is one which has been
repeating an exact pattern for an infinite 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.
x(t)
1
---- ---
t
-2 0 2 4
T=2
y(t)

T
An analog sinusoid or harmonic signals is always
periodic & unique for any choice of period or
frequency.
Hint:
The sum of harmonic signals
y(t) = x1(t) + x2(t) + x3(t) + - - - - - is periodic with
overall period
T = LCM (T1, T2, T3, ….)
• A discrete signal x(n) is periodic
if x ⇒ x[n] = x[n + N] ;
where N → periodic of x[n]
Hint:
For finding fundamental period of discrete
sinusoid (or) Complex sinusoids always use the
equation ω0/2π ratio is a rational number.
4 Signals & Systems
Sampled version of certain continuous Signals

1

0 t
0 0.5 1 1.5 2 2.5 3 3.5 4
–1
(a) Sampling x(t) = sin(3πt) with sample period T = 0.25
0 t
0 1 2 3 4
–1
(b) Sampling x(t) = sin(3πt) with sample period T = 1/π
1 1 1
0 0 0
–1 –1 –1
0 0.5 1 1.5 2 0 0.5 1 1.5 2 0 0.5 1 1.5 2
(a) cos(πnT), T = 0.25 (b) cos(2πnT), T = 0.25 (c) cos(3πnT), T = 0.25
1 1 1
0 0 0
–1 –1 –1
0 0.5 1 1.5 2 0 0.5 1 1.5 2 0 0.5 1 1.5 2
(d) cos(4πnT), T = 0.25 (e) cos(5πnT), T = 0.25 (f) cos(6πnT), T = 0.25
1 1 1
0 0 0
–1 –1 –1
0 0.5 1 1.5 2 0 0.5 1 1.5 2 0 0.5 1 1.5 2
(g) cos(7πnT), T = 0.25 (h) cos(8πnT), T = 0.25 (i) cos(9πnT), T = 0.25
5 Signals & Systems
Classification of Systems: 4. Static (or) Memoryless & Dynamic

1. Linear and Non Linear Systems: (With Memory) System:
A system is static if its output at t = t0 depends
For the system to be linear, it should satisfy
only on the value of the input at t = t0 and no
2 properties other value of the input signal. The response of
(A) x1(t)→ y1(t) and x2(t) → y2(t) a dynamic system depends on past (and/or
future) inputs.
Then x1(t) + x2(t) → y1(t) + y2(t) → Additivity
Ex: y(n) = (n+1) x(n) is static
(B) cx(t) → cy(t) → Scaling (or) Homogenity y(n) = x(n+3) is dynamic
(or)
ax1(t) + bx2(t) → ay1(t) + by2 (t)→ Superposition 5. Stable And Unstable System:
• A system is said to be BIBO stable if and
Example: y(n) = x(n –2) only if every bounded input results in a
y(n) = nx(n) are linear systems bounded output If x (t) # M x 1 3 then
y (t) # M y 1 3
2. Time-Invariant (Shift-Invariant) & Time-Variant • A bounded signal has an amplitude
(Shift-Dependent) Systems: remains finite.
A system is T.I. if the input output characteristic 6. Invertible & Inverse System:
don’t change with time. T.I. implies that the A system is said to be invertible if the input of
shape of the response y(t) depends only on the system can be recovered from the output
the shape of the input x(t) and not on the time x(t) y(t) x(t)
when it is applied. If the input is delayed by “t0” T {-} T-1 {-}
the output is also delayed by the same amount
and is simply shifted replica of the original
T.T-1 = I
output.
In any event, a system is not invertible unless
Ex: y (n) = x (n)
2
are time invariant distinct inputs applied to the system produces
y (n) = x (n − 5) distinct outputs
For T.I system if x(t) → y(t);
Then x(t – t0)  y(t – t0 )
3. Causal & Non Causal System:

A causal (or) non – anticipating 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 (t > t0) systems whose
present response requires knowledge of future
values of the input are termed non causal.
Ex: y(n) = 2n+1x(n) is causal
y(n) = 2nx(n+1) is Non-causal
6 Signals & Systems
Useful Mathematical Relations Practice Questions
x n +1
∫ x dx =
n
01. An aperiodic discrete time signal of length 5 is
n +1 shown in Fig. The maximum value of
- Cos ax
∫ Sin (ax) dx = a
x(n)
1
Sin ax
∫ Cos (ax) dx = a n
−2 −1 0 1 2
1 1  bx 
∫a 2 2 2
+b x
dx =
ab
Tan -1  
 a 
−1
Sin (ax) - ax Cos (ax)
∫ x Sin (ax) dx = a2
(A) x(n) + 2x(–n) (B) 5x(n) x(n–1)
(C) x(n) x(–n–1) (D) 4x(2n)
Cos (ax) + ax Sin (ax) (a) A > D > B > C (b) C > D > A > B
∫ x Cos (ax) dx = a2 (c) B > D > A > C (d) D > A > C > D
N -1
1- α N
∑αn =
K =0 1- α 02. A signal x(t) is nonzero for –1 ≤ t ≤ 2 and zero
elsewhere. Then the signal y (t) = 2x ; 2 t E is

−
+∞
1
∑αk =
K =0 1- α
,| α |< 1
nonzero for duration of
2
(a) 6 (b) 3
+∞
α
∑
K =1
K αk =
(1 - α) 2
,| α | < 1 (c) 1.5 (d) 2
N
N ( N + 1) 03. Find the value of the following integrals
∑K =
K =1 2 2
(a) # (t + t2) δ (t − 4) dt
-1
N
N ( N + 1) (2 N + 1)
∑
K =1
K2 =
6 (b)
3
# (t + cos πt) δ (t − 1) dt
-2
∞
1 π 3
∫e d x =
2
-a x
,a>0 (c)
# cos t u (t − 3) δ (t) dt
0
2 a 0
3
∞ ∞
(d) # e(t - 2) δ (2t − 4) dt
-2
∫ sin cx dx = ∫ sin c
2
x dx = 1
-∞ -∞ 2r
(e) # t sin t δ (π/2 − t) dt
0
7 Signals & Systems
04. Consider the D.T. signal These signals are sampled with a sampling
3
period of T = 0.25 seconds to obtain discrete
x(n) = 1 − / δ [n − 1 − k] time signals x1[n] and x2[n], respectively. Which
k=3
one of the following statements is true?
Find the values of M and n0 so that (a) The energy of x1[n] is greater than the
x(n) = u[Mn – n0 ] energy of x2[n].
(b) The energy of x2[n] is greater than energy of
05. Sketch the wave forms of the following signals? x2[n].
(a) x(t) = u(t+1) – 2 u(t) + u(t–1) (c) x1[n] and x2[n] have equal energies.
(b) x(t) = r(t+2) – r (t+1) – r(t–1) + r(t–2) (d) Neither x1[n] nor x2[n] is a finite-energy signal.
where r(t) is unit ramp function
08. Find the conjugate anti-symmetric part of
x (n) = &1 + j2, 2- , j5 0
Classification of signals
09. Give in figure are the parts of a signal x(t) and
06. Determine whether the following signals are its even part xe(t) for t ≥ 0 only; that is x(t) & xe(t)
energy (or) power signals? for t < 0 are not given. Complete the plots of x(t)
(a) x(t) = e-t u(t) & x0(t).
(b) x(t) = A xe(t)
x(t)
(c) y(t) 2
2
A
–t
Ae
t
t 0 1 0 2 t
0
(d) x(t) = A cos(ωt + θ) 10. Determine which of the following signals are
(e) x(t) = tu(t) periodic, if periodic find the fundamental
(f) x(t) = e3tu(t) period ?
(a) x(t) = cos(18πt) + sin (12πt)
07. (i) Two sequences x1[n] and x2[n] have the same (b) x(t) = sin (2πt/3)cos(4πt/5)
energy. Suppose x1[n] = α(0.5)nu[n], where α is (c) x(t) = cos 3t + sin 5πt
a positive real number and u[n] is the unit step (d) x(t) = jej10t
sequence. Assume (e) x(t) = cos(5t)u(t)
(f) x(t) = Ev[cos(2πt)u(t)]
 1.5 for n = 0,1
x2[n] =  (g) x(n) = sin : 5πn D
3
 0 other wise.
(h) x (n) = 2 cos : 4 D + sin : 8 D − 2 cos : 2 + 6 D
πn πn πn π
Then the value of α is ________.
(ii) Consider the two continuous-time signals (i) x(n) = ej7πn
(j) x(n) = cos(n/4) sin : 8 D
πn
defined below:
t , −1 # t # 1 (k) x(n) = u(n) + u(– n)
x1 (t) = )

0, other wise (l) x(n) = (- 1) n
x2 (t) = )1 − t , − 1 # t # 1 / ^δ (n − 4k) − δ (n − 1 − 4k)h
3
(m) x (n) =
0, other wise k =-3

8 Signals & Systems
11.
d2 y (t) 5dy (t)
A) A signal x(t) = 2 cos (150πt + 300) is sampled at (m) + + 2 = x (t)
dt2 dt
200Hz. Find the fundamental period of discrete
signal? (n) y(n) = ex(n)
B) A periodic discrete time signal x(n) is given by
x(n) = cos (3πn) + sin (7πn) + cos (2.5πn). The 13. Test the following systems for time - invariance?
term sin (7πn) in x(n) corresponds to (a) y(t) = tx(t) + 3
(a) 14th harmonic (b) 7th harmonic (b) y(t) = ex(t)
(c) 6 harmonic
th
(d) 5th harmonic (c) y(t) = x(t) cos3t
C) For each periodic signal shown in figure, (d) y(t) = sin{x(t)}
d
evaluate the average value xav, the energy E (e) y(t) = x (t)
dt
in one period, the signal power P, and the rms (f) y(t) = x2(t)
value xrms. (g) y(t) = x(2t)
(h) y(n) = 2x(n) x(n)
x(t) Signal 1 x(t) Signal 2
4 (i) y(n) = x(n + 2) – x(7 – n)
1 d2 y (t)
7 t (j) + y (t) y (3t) = x (t)
t -5 -2 -2 2 5
dt2
–2 2 d2 y (t)
x(t) Signal 3 (k) + 3y (t) = 2x (t)
dt2
2
1
2 t
5 14. Consider an LTI system whose response to the

input signal x1(t) is y1(t) as shown in figure. Find
Classification of systems the response of the system due to the Input
x2(t) and x3(t)?
12. Determine which of the following systems is y1(t)
x1(t)
linear?
(a) y(t) = x(t) x(t – 2) 1 2
(b) y(t) = sin{x(t)}
d t
(c) y (t) =
dt
(x (t))
0 2 t 0 2
(d) y(t) = 2x(t) + 3
t

(e) y (t) = # x (τ) dτ x2(t) x3(t)
-3
(f) y(t) = x2(t) 1 2

1
(g) y(t) = x(t) cosω0t 0
(h) y(n) = log {x[n]}
2 4 t
-1- -1 0 1 2 t
(i) y(n) = |x[n]|
(j) y(n) = x*(n)
(k) y(n) = Median {x[n]}
x (n)
(l) y (n) =
x (n − 1)
9 Signals & Systems
15. Check whether the following systems are

d2 y (t) 2dy (t)
causal (or) non causal? (c) 4 + + y (t) = 3d x (t)
dt2 dt dt
(a) y(t) = (2t + 3) x(t)
dy (t) 2 4dx (t)
(b) y(t) = x2(t) (d) ; E + 2t y (t) =
dt dt
(c) y(t) = x(t)sin5t
(d) y(t) = x{sin(t)} 18. Match the following
List I (System)
(e) y(t) = |x(t)|
(a) y(n+2)+ y(n+1) + y(n) = 2x(n+1) + x(n)
(f) y(n) = ex(n)
n (b) n2y2(n) + y(n) = x2(n)

(g) y (n) = / x [k], "n0 " is finite (c) y(n + 1) + ny(n) = 4nx(n)
k = n0
n + n0 (d) y(n + 1)y(n) = 4 x(n)

(h) y (n) = / x [k], "n 0 " is finite
k = n - n0 List II (system category)
n (1) Linear, T.V., dynamic
(i) y (n) = / x [k]
k =- 3
(2) Linear, T.I., dynamic
n (3) N.L., T.V., dynamic
(j) y (n) = / x (k)
k=0
(4) N.L., T.I., dynamic
(5) N.L., T.V., memory less.
1
m
(k) y (n) = / x (n − k)
2m + 1 k =- m 19. Check whether the following systems are stable
(or) not?
(l) y(t) = x(αt)
(m) yl (t + 4) + 2y (t) = x (t − 2) (a) y(t) = x2(t)
(n) yl (t) + 2y (t) = x (t + 3)
(b) y(t) = x(t) cos 3t
16. Check whether the following systems are static (c) y(t) = x(t–3)
or dynamic? (d) y (t) = d x (t)

dt
(a) y(t) = 3x(t) + 5 t
(b) y(t) = x2(t)

(e) y (t) = # x (τ) dτ
-3
(c) y(n) = ex(n)

(f) y(t) = sin{x(t)}
(d) y(n) = cos{x(n)}
(g) y(t) = tx(t)
(e) y(n) = x[3n]
(h) y(n) = ex(n)
(f) y (t) = d x (t)
dt (i) y(n) = x[3n]
t n
(g) y (t) = # x (τ) dτ (j) y (n) = / x [k], "n0 " is finite
-3 k = n0
20. Determine which of the following systems are

17. Find whether the following systems are linear,
invertible?
T.I. and dynamic
(a) y(t) = x2(t)
2tdy (t) (b) y(t) = x(t) 
(a) + 4y (t) = 2tx (t)
dt (c) y(t) = x(t – 3)

d2 y (t) d
(b) + 4y (t) = 2x (t) (d) y (t) = dt x (t)
dt2
10 Signals & Systems
(e) y(n) = x[n]x[n – 1] 24. Statement (I): A memory less system is causal
(f) y(n) = nx(n) Statement (II): A system is causal if the output
(g) y(n) = x[n] – x[n – 1] at any time depends only on values of input at
n
(h) y (n) = / x [k] that time and in the past.
k =- 3
21. Consider the feedback system shown in figure, Key for Practice Questions
assume that y[n] = 0 for n < 0
x(n) y(n)
01. Ans: (c)
+ Unit delay
+_
02. Ans: (a)

03. (a). Ans: 0 (b). Ans: 0
Find y(n) when the input is
(i) x(n) = δ[n] (ii) x(n) = u[n] (c). Ans: 0 (d). Ans: 0.5

(e). Ans: π/2
22. Statement (I): A discrete time system with
input x(n) and output y(n) and given by the
04. Ans: M = -1, n0 = -3
relation y(n) = nx(n–2) + 2 is non linear.
08. Ans: 2 %1 + j7, 0- , − 1 + j7 /
Statement (II): If x(n) is delayed by 2 units, 1
y(n) is not delayed by 2 units.

(a) Both Statement (I) and Statement (II) are 10. Ans: a, b, d, f, g, h, i, l, m are periodic
individually true and Statement (II) is the signals.
correct explanation of Statement (I)

12. Ans: c, e, g are linear
(b) Both Statement (I) and Statement (II) are
individually true but Statement (II) is not
13. Ans: b, d, e, f, h, k are time invariant
the correct explanation of Statement (I)
(c) Statement (I) is true but Statement (II) is 15. Ans: a, b, c, e, f, i, m are causal
false
(d) Statement (I) is false but Statement (II) is 16. Ans: a, b, c, d are static
true
18. (a). Ans: -(2), (b). Ans: -(5),
(c). Ans: -(1), (d). Ans: -(4)
23. Statement (I): The discrete time system
described by y[n] = 2 x[n] + 4 x[n - 1] is unstable, 19. Ans: a, b, c, f, h, i are stable
(here y[n] is the output and x[n] the input)
Statement (II): It has an impulse response with 21. (i) y(n) = {0, 1, -1, 1,-------}
a finite number of non-zero samples. (ii) y(n) = {0, 1, 0, 1,-------
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
A linear system's characteristics are completely specified by the system's impulse response, as governed
by the mathematics of convolution. For example: Digital filters are created by designing an appropriate
impulse response. Enemy aircraft are detected with radar by analyzing a measured impulse response.
Echo suppression in long distance telephone calls is accomplished by creating an impulse response that
counteracts the impulse response of the reverberation.
2.1 Continuous & Discrete Convolution
x(t) L.T.I y(t) = x(t)*h(t) x(n) L.T.I y(n) = x(n)*h(n)

System h(t) System h(n)
3 3
y (t) = # x (τ) h (t − τ) dτ y (n) = / x [k] h [n − k]
-3 k =-3
3 3
= # x (t − τ) h (τ) dτ = / x [n − k] h [k]
-3 k =-3
Steps: Steps:

1. x(t) → x(τ), h(t) → h(τ) 1. x[n] → x[k], h[n] → h[k]
x(–τ) x[–k]
2. Folding 2. Folding
h(–τ)

h[–k]
x[n – k]
x(t – τ)
3. Shifting 3. Shifting
h(t – τ) h[n – k]
x(t – τ)h(τ) x[k] h[n – k]

4. Multiplication 4. Multiplication
h(t – τ)x(τ) x[n – k] h[k]
5. Integration
5. Summation
• An L.T.I System is always considered w.r.t

Practice Questions
impulse response denoted as h(t) or h(n).
• If the Input is impulse, then the output is impulse
response. 01.(a) Find the convolution of the signals
• SIFTING property states that any signal can be x(t) = e–3t u(t) & h(t) = u(t – 2)
produced as a combination of impulses.
• Convolution may be regarded as a method of (b)Find the convolution of the signals shown in
finding the zero-state response of a relaxed LTI figure?
x(t) h(t)
system.
• Any DT signal is the sum of scaled and shifted 1 1
unit impulses.
• Convolution may be treated as flip - shift - t
0 1 t 0
multiply - time - area method.
• In the convolution integral time “t” determines
relative location of h(t – τ) w. r. t. x(τ). The 02. The impulse response and the excitation
convolution will yield a non zero result only function of a linear time invariant causal
for those value of “t” over which h(t - τ) & x(τ) system are shown in Fig. a and b respectively.
overlap. The output of the system at t = 2 sec. is equal
to
2.2 Properties of L.T.I System h(t) x(t)

1
Causality: Stability: 1/2

+3
h(t) = 0; t < 0 # h (x) dx < 3 0 6 t (sec) 0 2 6 t (sec)

-3 Fig. a Fig. b
+3
h(n) = 0; n < 0 / h (n) <3
(a) 0 (b) 1/2
n =- 3
(c) 3/2 (d) 1

Memory less : Invertibility & Inverse:
h(t) = 0 for t ≠ 0 h(t) ∗ hinv(t) = δ(t)
h(n) = 0 for n ≠ 0 h(n) ∗ hinv(n) = δ(n) 03. An L.T.I system is having impulse response
h(t) = u(–t–1) for which the input signal
applied is shown in figure. Find the output at
t = 4 & t = 0.5?
x(t)
3
1
t
0 1 2 6
04. The impulse response of a continuous time 10. An Input signal x(t) shown in figure is applied to
system is given by h(t) = δ(t - 1) + δ(t - 3). The the system with impulse response
3
value of the step response at t = 2 is h (t) = / δ (t − kT) .
k =-3
(a) 0 (b) 1
Find the output, for the values of
(c) 2 (d) 3
i) T = 4 x(t)
ii) T = 2
3
05. Suppose z (t) = # x (− τ + a) h (t + τ) dτ

-3
Express z(t) in terms of y(t) = x(t) ∗ h(t)

t
–1 0 1
06. Find the following terms
(a) x(t + 5) * δ(t – 7) = _________
(b) x(t) * δ(at + b) = __________ 11. (a) Find the convolution of
7u (t + 3) - u (t - 1)A with 7u (t + 1) - u (t - 1)A
x(t)
(c) 1 2
* = (b) The impulse response of a system is
–1 0 1 t –3 3 t h(t) = t u(t). For an input u(t – 1), the output
is
07. Explain the difference between each of the t (t - 1)
t2
following operations? (a)
2
u (t) (b)
2
u (t - 1)
(a) [e-t u(t)]δ(t – 1) (t - 1) 2 t2 - 1
(c)
2
u (t - 1) (d)
2
u (t - 1)
3
(b) # e -t u (t) δ (t − 1) dt
-3
(c) Consider a signal averaging filter whose

(c) e-tu(t) ∗ δ(t – 1)
impulse response is h (t) = T rect b t 0.5T l .
1 −

T
Find the response of the filter due to the
08. Let x(t) = u(t–3) – u(t – 5) & h(t) = e–3tu(t).
input x(t) = u(t)?

d
Find x (t) ) h (t)
dt
1, 0 < t < 1
12. Suppose that x (t) = )
0, elsewhere
09. Signals that replicate under self-convolution
and h (t) = x b αt l, 0<α#1
include the impulse, Sinc, Gaussian, and
dy (t)
Lorentzian. For each of the following known Let y(t) = x(t) ∗ h(t). If has to contain
dt
results, determine the constant A using the only three discontinuities, the value of α is
(a) 1
area property of convolution.
(b) 2
(a) δ(αt) ∗ δ(αt) = Aδ(αt)
(c) 0.5
(b) Sinc(αt) ∗ Sinc(αt) = ASinc(αt) (d) – 1
(c) e - rt ) e - rt = Ae - rt /2
2 2 2
1 1 = A
(d) )
1 + t2 1 + t2 1 + 0.25t2
18. Two discrete -time signals x[n] and h[n] are

13. Given x (t) = 5rect b 2 l and h (t) = / 2δ (t − 2k)
t 3
k =-3 both non-zero only for n = 0,1,2, and are zero

then x (t) ) h (t) is
otherwise. It is given that x[0] = 1, x[1] = 2,
(a) 10 ∀t
x[2]=1, h[0] = 1, Let y[n] be the linear
5rect c t 2 m
3
-
(b) / 2 convolution of x[n] and h[n]. Given that
k =-3
y[1] = 3 and y[2] = 4, the value of the expression
/ 10rectc t 2-k2 m (10y[3] + y[4]) is _______
3
(c)
k =-3
(d) None of these

3
14. The integral # x (τ) h (τ − t) dτ can be

-3
Properties of L.T.I system
interpreted as the Convolution of
(a) x(τ) and h(τ)
(b) x(t) and h(t) 19. The range of ‘a’ and ‘b’ for the impulse
a n; n $ 0
(c) x(-τ) and h(-τ) response h (n) = ) n
b ; n<0
(d) x(t) and h(-t)
to be stable is _______
(a) |a|<1, |b| < 1
(b) |a|>1, |b| < 1
(c) |a|>1, |b| > 1
Discrete Convolution (d) |a|<1, |b| > 1
15. A linear system with Input x(n) & output y(n) 20. Given h(t) = eαt u(t) + eβt u(–t). For what values of
related 3as α and β system is stable?
y (n) = / x (k) g (n - 2k) (a) α < 0, β < 0
k =-3
where g(n) = u(n) – u(n – 4). Find y(n) when (b) α < 0, β > 0
x(n) = δ(n – 2) (c) α > 0, β > 0
(d) α > 0, β < 0
16. The I.R of a D.T LTI system is given by
h(n)=(0.5)n u(n) of the input is x (n) = 2δ (n) + δ (n − 3). 21. Consider the system in figure.
Find the output at n = 1 & n = 4?
x(n) y(n)
h2(n) = αnu(n)
17. Given x = [a, b, c, d] as the Input to an LTI +
system produces an output
y = [x, x, x, x, … repeated N times].
h1(n) = βδ(n – 1)
The impulse response of the system is
(a) /
N-1

δ [n − 4i]
i=0 (a) Find I.R. of overall system.
(b) u(n) – u(n – N) (b) Is this system causal ? Under
(c) u(n) – u(n–N–1) What condition the system is stable.

N-1
(d) / δ [n − i]
i=0
22. Consider a D.T system ‘s1’, with I.R 28. If step responses of 2 L.T.I systems are s1(t) &
h(n) = (1/5) u(n) n
s2(t) respectively, how the cascaded step
(a) Find ‘A’ such that h(n) – Ah(n–1) = δ(n) response sc(t) is related interms of s1(t) &
(b) Using result from part (a), determine the I.R s2(t)?
g(n) of an LTI system s2 which is inverse of s1
23. For the interconnected system shown in Fig.

Key for Practice Questions
find the overall impulse response.
-
01. (a) Ans: 1 e3
-3 (t - 2)
x(n) y(n) u (t - 2) 04. Ans: (b)
y1[n] = x(n) – ½ x(n-1) n
h2 [n] = (½ ) u[n]

05. Ans: z(t) = y(t + a)
24. Determine whether each of the following 1 b +bl

06. a. Ans: x(t – 2), b. Ans: a x t a
statements are TRUE (or) FALSE.
Justify your answer.
07. a. Ans: e -1 δ (t − 1) , b. Ans: e -1 ,
(1) The cascade of a non causal LTI system
c. Ans: e–(t–1). u(t–1)
with casual one is necessarily noncausal
(2) If an LTI system is causal, it is stable 08. Ans: e -3 (t - 3) .u (t - 3) - e -3 (t - 5) .u (t - 5)
(3) If h(t) is the I.R of an LTI system which is
periodic & nonzero, the system is unstable 11. b). Ans: (c) 13. Ans: (a)
(4) The inverse of a causal system is always
14. Ans: (d)
causal
15. Ans: y(n) = g (n-4)
25. If the unit step response of a system is
16. Ans: y(1) = 1, y(4) = 5/8
(1 - e - at) u (t) , then its unit impulse response is
_______ 17. Ans: (a) 19. Ans: (d) 20. Ans: (b)

(a) αe - at
u (t)
21. Ans: h (n) = α n u (n) + βα n - 1 u (n − 1) is causal
(b) α e -1 - at
u (t)
for any value of α, β and stable if |α| < 1,
(c) (1 − α -1) e - at u (t)
and any value of β.
(d) (1 − α) e - at u (t)
22. Ans: A = 15 23. Ans: h (n) = δ (n)
26. Find the step response of the system if the
impulse response is h(n) = (0.5)n u(n) 24. Ans: 1-false, 2-false, 3-true, 4-false.
25. Ans: (a)

27. An LTI system with Input u(n) produces the
26. Ans: s (n) = 2 ;1 − b 1 l E u (n)
n+1
output as δ(n), then find the output due to
2
the input nu(n)?
27. Ans: u(n-1)
However, musical instruments, such as a piano,

3. Fourier Series
are made of many strings all vibrating at once.
The question that intrigued Fourier was: How do
Historical perspective:
you evaluate the waveforms from a number of
Jean Baptiste Joseph Fourier (1768 – 1830)
strings all vibrating at once? As a product of his
Joseph Fourier was born in Auxerre, France
research, Fourier realized that the sound heard
on March 21, 1768 and died in Paris on May 4,
by the ear is actually the arithmetic sum of
1830.
each of the individual waveforms. This is called
Motivation: the principle of superposition.
• Representation of continuous time, periodic
signals in the frequency domain • Representing CT signals as superposition of
• Periodic signals occur frequently - motion complex exponentials leads to frequency –
of planets and their satellites, vibration of domain characterizations. eg :- A human ear is
oscillators, electric power distribution, beating sensitive to audio signals within the frequency
of the heart, vibration of vocal chords, etc. range 20Hz to 20 kHz. Typically, musical note
Introduction: occupies a much wider frequency range.
• The Fourier series is named after the French Therefore the human ear processor frequency
mathematician Joseph Fourier. components within the audible range &
In this chapter we will consider approximating rejects other frequency components. In such
a function by a linear combination of basis applications, frequency-domain analysis
functions, which are simple functions that can provides a convenient means of solving for the
be generated in a laboratory. Joseph Fourier response of L.T.I. systems to arbitrary input.
(1768–1830) developed the mathematical
• By using F.S, a non-sinusoidal periodic function
theory of heat conduction using a set of
can be expressed as an infinite sum of sinusoidal
trigonometric (sine and cosine) series of the
functions.
form we now call Fourier series. He established
that an arbitrary mathematical function can • Sinusoidal signals arise in describing motion of
be represented by its Fourier series. planets & periodic behavior of earth’s climate.
• Fourier series and the Fourier transform are A.C. sources generate sinusoidal voltages &
basics to mathematics and science, especially currents.
to the theory of communications. For example,
• There are 2 reasons for evaluating the F.S.
a phoneme in a speech signal is smooth and
1. To obtain an expression for f(t) that applies
wavy. A linear combination of a few sinusoidal
everywhere, rather than only over a single
functions would approximate a segment of
period.
speech within some error tolerance.
2. To obtain phasors, which indirectly tell how
• Fourier and a number of his contemporaries
much power is available at each harmonic
were interested in the study of vibrating
of the waveform.
strings. In the simple case of just one naturally
vibrating string the analysis is straightforward:
the vibration is described by a sine wave.
3.1 ANALOGY BETWEEN VECTORS & SIGNALS Length of the component vf along X v
• Signals are not just like vectors. = CX v = vf Cosθ
A vector can be represented as a sum of CX v
v Cosθ = vf .X
v 2 = vf . X
its components, depending on the choice
vf .X
v
of coordinate system. A signal can also be C= v 2
X
represented as a sum of its components.
• We know that an arbitrary M-dimensional • 2 vectors vf & X
v are orthogonal if inner (or)
vector can be represented in terms of scalar product vf : X
v =0
M orthogonal co-ordinates.
• If we consider 2 basis vectors vi & vj
A vector is specified by its magnitude &
direction. orthogonality vi=: vj vj=
: vi 0
v 4 orthonormal Property
: vi
unit magnitude i = vj=
: vj 1
f

e
Component of a Signal
θ g(t)
X
CX x(t) = sin t
FIG (a) +1
t
0 π 2π
f
e1
−1
C1 X X
FIG (b) g(t) ≅ C x(t) ; t1 < t < t2
t 2
# g (t) x (t) dt
f e2 C = t
t
1
2
# x2 (t) dt
X
t1
C2 X Note:
FIG (c)
2 signals g(t) & x(t) are said to be orthogonal
Consider 2 vectors vf & X
v as shown in figure.
over the interval (t1, t2)
Let the component of f along X be CX.
t 2 t 2
(Geometrically the component f along X is the # g (t) x* (t) dt =
if = # x (t) g* (t) dt 0
projection of f on X) t1 t1
From Fig (a) vf = CX v +ev They are also said to be orthonormal if they
From Fig (b) & (c) satisfies
vf = C1 X
v +e v +e
v 1 = C2 X v2 t 2 t 2
=# x (t) x* (t) dt =
# g (t) g* (t) dt 1 (unit magnitude)
If we approximate f by CX, vf , CX v
t1 t1
Error in the approximation e = f – CX

Minimize the error vector, such that vf and X
v
are approximated
Examples (b) For orthonormality 2T = 1 ⇒ T = 1/2

(c) x(t) = C1∅1(t) + C2∅2(t) + C3 ∅3(t)
Example 01: t2
1 #
Ci = E x (t) Q*i (t) dt but EQ = 2T
For the three continuous functions shown in Q
t1
figure T
= # A (1) A/2
C1 1=
φ1(t)
0
φ2(t)
1 JK T/2 T NO
KK
C2 1 K # A dt
=
K0
− # A dtOOOO = 0
L T/2
P
Similarly C3 = – A/2
-T 0 T t -T 0 T t ∴ x(t) = A/2 [∅1(t) – ∅3(t)]
φ3(t)
1
3.2 Trigonometric F.S
t • Any practical periodic function of frequency

-T 0 T
ω0 can be expressed as an infinite sum of sine
-1 (or) cosine functions that are integral multiples

of ω0
(a) Show that the functions form an orthogonal
g(t) = ao+ a1cosω0t + a2cos2ω0t + - - - - - -
set
+ b1sinω0t + b2sin2ω0t + - - - - - - - -
(b) Find the value of T that makes 3 functions
orthonormal 3
g (t) = a 0 + / a14444444444444
n cos nω 0 t + b n sin nω 0 t
42444444444444443 ------ (I)
(c) Express the signal . n=1
ac
dc
A for 0 # t # T
x (t) = )
0 elsewhere ω0 → Fundamental frequency
a0, an, bn → T.F.S. coefficients
in terms of orthogonal set determined

in (a)? T
1
a 0 = T # g (t) dt " d.c (or) Average value
Sol: (a) For all the 3 signals

0
T
T T T
2
# Q1 (t) 2 dt = # Q2 (t) 2 dt = # Q3 (t) 2 dt = 2T
a n = T # g (t) cos nω 0 t dt
1-444444444444444444444444444
T -T -T
42444444444444444444444444444
43 0
Unit - Magnitude Property T

2 #
bn = T g (t) sin nω 0 t dt
0
Orthogonality
T T T

# Q1 (t) Q2 (t) dt = # Q2 (t) Q3 (t) dt = # Q1 (t) Q3 (t) dt = 0
-T -T -T
Effect of symmetry on Fourier coefficients:
Symmetry Condition a0 an bn
Even g(t) = g(–t) ? ? 0
Odd g(t) = – g(–t) 0 0 ?
Half-wave (or) g(t) = – g(t ± T/2) 0 = 0 ; n even = 0 ; n even
Rotational = ? ; n odd = ? ; n odd
Notes:
• a0, the DC offset term, can be non-zero even though all the other an’s are zero
• An odd-harmonic function is one where the second half of its period is the negative of the first half.
• An even-harmonic function is one where the second half of its period is exactly the same as the first
half. Therefore, any function that is even-harmonic is actually a regular periodic function whose period
has been labeled twice what it should be. In other words, there is nothing special about even-harmonic
functions.
• Shifting a signal left/right in time does not affect whether or not it is odd-harmonic.
• Shifting a signal up/down (adding a DC offset) does not affect whether it is odd-harmonic, other than
adding a term in the Fourier series at Zero frequency.
• An odd-harmonic function does not have to be odd.
t t
even
odd
t t
even and odd - harmonic
odd-harmonic
t t
odd and odd - harmonic even and odd - harmonic w/ DC offset
3.3 Exponential (or) Complex F.S 5. Differentiation in time:-

Another form of the F.S that involves complex x(t) ↔ Cn
exponential functions can be obtained by dx (t)
* (jnω 0) Cn
dt
substituting the complex exponential forms of
sin ω0t and cos ω0t into (I) 6.
Parseval’s Power theorem:-
The total average power in a periodic signal
| Cn | 2-sided equal to sum of squared amplitude of each
spectrum harmonic
x(t) ↔ Cn
1 T 3
then T #0 x (t) 2 dt = / Cn 2
n =-3

- 2ω0 - ω0 0 ω0 2ω0 nω0
3 T
g (t) = / Cn e jn~Where
t 1 #
Cn = T g (t) e -jn~ t dt
0 0
n = -3
0
Convergence Of F.S: (Dirichlet Conditions)

(1) x(t) is absolutely integrable i.e.,
T
# x (t) dt < 3
0
(2) x(t) has only a finite number of maxima &

minima
(3) The number of discontinuities in x(t) must be
finite.
3.4 Properties of F.S

1. Linearity: x1(t) ↔ Cn
x2(t) ↔ dn
then αx1(t) + βx2(t) ↔ αCn + βdn
2. Time Shift: x (t - t 0) * Cn e -jn~0 t0

When we shift in the time-domain, it changes
the phase of each harmonic in proportion to its
frequency nω0.
3. Frequency Shift:
x(t) ↔ Cn
then x (t) e j~0 Mt * Cn - M
4. Time-Scaling:
x (t) * Cn then x (at) * Cn
Time-Compressing by α changes frequency
from ω0 to αω0
Practice Questions (R) g (t)
+V
01. A periodic signal is given by
x(t) = 3 sin (4t + 30o) – 4 cos (12t – 60o).
Find the amplitude of second harmonic? -2π - π 0 π 2π t
02. Which of the following signal is not the

-V
representation of F.S.?
(a) Cos3t + sin12t (a) Sine terms
(b) 1 + sinπt (b) Sine terms with odd harmonics
(c) ej6t (c) Sine terms with even harmonics
(d) cos2πt + sin6t (d) DC, Cosine and Sine terms
03. For the Periodic waveforms shown in figure (S)

what frequency components are present f(t) 1
in trigonometric series expansion.
(P) t
x(t) –π -π/2 0 π/2 π

2 (a) DC, Cosine terms
(b) DC, Cosine terms with even harmonics
(c) DC, Cosine terms with odd harmonics
t (d) None of these
-2 -1 0 1 2
(T) m(t)

2
(a) DC and cosine terms
1
(b) DC and sine terms
(c) Cosine & sine terms 4 t
–2 –1 1 2
(d) None of these –1
–2
(Q)
(a) DC, Cosine and Sine terms
f(t) (b) DC, Cosine terms with even harmonics
V (c) DC, Cosine terms with odd harmonics
(d) None of these
-π 0 π 2π 3π t
-V 04. Consider the signal
x(t) = 10cos(10πt + π/7) + 4sin(30πt + π/8).
(a) Sine and Cosine terms It’s power lying within the frequency band
(b) Only odd harmonics 10 Hz to 20 Hz is ____
(c) Sine terms with odd harmonics (a) 4W (b) 8W
(d) DC and Sine terms (c) 50W (d) 58W
05. Consider the trigonometric series, which holds 09. f(t), shown in figure is represented by
3
true∀t, given by f (t) = a 0 + / an Cosnt + bn Sinnt
n=1
x(t) = sinω0 t + 1/3 sin 3ω0t + 1/5 sin 5ω0t The value of a0 is
+ 1/7 sin 7ω0 t + - - - - - - - - - -
f(t)
At ω0t = π/2 the series converges to
(a) 0.5 1.5
(b) π/4 +1
(c) π/2 –π 0 π 2π 3π t
(d) 2
-1.5

06. x(t) is a real valued function of a real variable (a) 0 (b) π/2
with period T. Its trigonometric. Fourier Series (c) π (d) 2π
expansion contains no terms of frequency
ω = 2π(2k)/T; k = 1, 2….……. Also, no sine terms are
10. A periodic rectangular signal x(t) has the
present. Then x(t) satisfies the equation
(a) x(t) = –x(t – T) waveform shown in fig frequency of the fifth
(b) x(t) = x(T – t) = –x(–t) harmonic of its spectrum is
(c) x(t) = x(T – t) = –x(t –T/2)
(d) x(t) = x(t – T) = x(t – T/2) x(t)
07. The fundamental frequency of the composite

signal –4 –2 0 2 4 t (msec)
x(t) = 2 + 3cos (0.2t) + cos (0.25t + π/2)

+ 4cos (0.3t – π) is
(a) 0.05 rad/sec (a) 40 Hz
(b) 0.1 rad/sec (b) 200 Hz
(c) 0.2 rad/sec (c) 250 Hz
(d) 0.25 rad/sec (d) 1250 Hz
08. The average value of the periodic signal x(t)

11. A periodic signal in a duration of one period is
shown in figure is
shown. Its Fourier series expansion Includes
x(t)
3
3
1

– 6 – 4 – 3– 2 6 8 9 10 t
0 2 3 4 t 0 3 4 5
−1
(a) 5/6
(a) cosine terms of odd harmonics
(b) 1
(b) sine terms of odd harmonics
(c) 5
(c) sine terms of even harmonics
(d) 6
(d) cosine terms of even harmonics
12. The rms value of the periodic waveform shown Exponential F.S
in figure is
16. For the periodic signal
6A
x (t) = 2 + cos : 3 D + 4 sin : 3 D
2rt 5rt

0 T/2 T t Find the E.F.S. coefficients?
–6A 17. Obtain the E.F.S. representation of periodic

(a) 2 6 A (b) 6 2 A signal shown, hence find the T.F.S?
4 (d)1.5 A
(c) A
3 x(t)
1
13. A half - wave rectified sinusoidal waveform has
a peak voltage of 10 V. Its average value and
the peak value of the fundamental component
are respectively given by : −1 0 1 2 t

20 10 10 10
(a) r V, 2 V (b) r V, 2 V 18. The F.S. coefficient of the signal x(t) shown in
fig (a) are C0 = 1/π, C1 = –j0.25, Cn = 1/π(1 – n2)
10 20
(c) r V, 5V (d) r V, 5V (n even). Find F.S. coefficients of y(t), f(t) and
g(t)?
14. A periodic signal with period T = 2 is defined as
x(t)
t, 0 # t # 1
x (t) = ) 1
2 - t, 1 # t # 2
If A cos (πt ) is one of the terms it contains A is

equal to -1 t
0 1 2 3
4 2
(a) 2 (b) 2 Fig(a)
r r
y(t)
-2 -4
(c) (d) 1
r2 r2
15. A periodic input signal x(t) shown below is

applied to an L.T.I. system with frequency -1 0 1 2 3 t
response f(t)
1; ω < 4π
H (ω) = )
1

0; ω > 4π
Which frequency components are present in 0 2 3
1 t
- -
the output?
2 1 -1
x(t)
10 g(t)
1
-2 -1 0 1 2 3 t t
-1 0 1 2
19. Let x(t) be a periodic signal with period T and

F.S coefficient Cn w1 1 w2
1
Let y(t) = x(t–t0) + x(t + t0). The F.S. coefficient of
y(t) is dn.
0 T/2 T t 0 T/2 T t
If dn= 0 ∀ odd n then t0 can be
(a) T/8 (b) T/4
–1 –1
(c)T/2 (d) 2T
(a) |n-3| & |n-2| (b) |n-2| & |n-3|

20. By using derivative method, find F.S. coefficient
(c) |n-1| & |n-2| (d) |n-4| & |n-2|
of the signal shown in figure?
x(t) 23. The magnitude & phase spectra of a periodic

signal x(t) are shown in figure
1
Magnitude
2
t
-T/2 -d/2 0 d/2 T/2 T -3 -2 -1 0 1 2 3 f(Hz)
900
Phase 600
21. Consider a periodic signal x(t) as shown below
300
x(t)
1
-3 -2 -1 0 1 2 3 f(Hz)
0
–30
–600
−3−2−1 0 1 2 3 4 5 6 t –900

−1
(a) Write the polar form of TFS.
Fig.
(b) Sketch the magnitude & phase spectrum
It has a Fourier series representation of f(t) = x(2t), g(t) = x(t - 1/6) & h(t) = xl (t)
3
x (t) = / ak e j(2r/T)kt
k =- 3
Which one of the following statement is TRUE?

(a) ak = 0, for k odd integer and T = 3
(b) ak = 0, for k even integer and T = 3 24.
(c) ak = 0, for k even integer and T = 6 (a) Find the power up to second harmonic for the
periodic signal shown in figure?
(d) ak = 0, for k odd integer and T = 6
22. One period (0, T) of 2 periodic waveforms 1

w1 & w2 are shown in Fig. then Fourier series
Coefficient are respectively proportional to
–1 0 1 2 3 t
____
(b) For the periodic signal x(t) shown below with

∠Cn π/2
period T = 8 s, the power in the 10th harmonic is π/4
x(t)
−3 −2 0 2 n
+1 −4 −1 1 3 4
−π/4
4 8 −π/2
–4 0 t
–1 (a) Find the power in the periodic signal

(a) 0 (b) Is this signal odd symmetric, even
1 b 2 l2 symmetric & neither
(b)
2 10π (c) Assume that the fundamental frequency
1 b 4 l2 of 10Hz, find the trigonometric form of the
(c)
2 10π signal x(t)?
(d) If this signal is passed through an ideal LPF
(d) b 10π l
4 2
with cut-off frequency 25Hz, what would
be the steady-state response
27. Match the following

25. The F.S Coefficients, of a periodic signal x(t) is
3
List – I (Periodic function)
expressed as x (t) = / Cn e jn~ t 0
are given by
n =-3 A. Impulse train
C–2 = 2– j1; C–1= 0.5 + j0.2; C0 = j2;
C1 = 0.5 – j0.2; C2 = 2 + j1;
Cn = 0 for |n| >2.
Which of the following is TRUE? -2T -T 0 T t

(a) x(t) has finite energy because only finitely
B. Full-wave rectified sine wave
many coefficients are non zero C. 2sin(2πt/6) cos(4πt/6)
(b) x(t) has zero average value because it is
D.
periodic 1
(c) the imaginary part of x(t) is constant
t
(d) the real part of x(t) is even. 0 T/2 T
-1
26. For the following spectrum shown in figure.
|Cn|
4 List-II (Properties of spectrum)
2 1. Only even harmonics are present
1
2. Impulse train with strength 1/T
0.5
3. C3 = 1/2j C-3 = –1/2j
C1 = –1/2j C-1 = 1/2j
4. Only odd harmonics are present
−4 −3 −2 −1 0 1 2 3 4 n
5. Both even & odd harmonics are present
28. The systems S1 & S2 shown in figure satisfy the

following properties
sin ωt 5 cos (ωt + 30°)

S1

01. Ans: 0
(1/4) sin10t cos (5t)
S2
02. Ans: (d)

(a) Both S1 and S2 are LTI systems
03. P. b, Q. b, R. b, S. c, T. d
(b) S1 is LTI, but S2 is not LTI system
(c) S1 is not LTI system, S2 is LTI system
04. Ans: (b)
(d) Both S1 & S2 are not LTI systems
05. Ans: (b)
29. Consider the following statements related to
Fourier series of a periodic waveform:
07. Ans: (a)
1. It expresses the given periodic waveform
as a combination of d.c. component,
08. Ans: (a)
sine and cosine waveforms of different
harmonic frequencies.
09. Ans: (a)
2. The amplitude spectrum is discrete.
3. The evaluation of Fourier coefficients gets
10. Ans: (d)
simplified if waveform symmetries are used

4. The amplitude spectrum is continuous
11. Ans: (b)
Which of the above statements are

correct?
13. Ans: (c)
(a) 1, 2 and 4 (b) 2, 3 and 4
(c) 1, 3 and 4 (d) 1, 2 and 3
14. Ans: (d)
30. Statement (I): In the exponential Fourier 20 20

representation of a real-valued periodic 15. Ans: y (t) = 5 + r sin rt + sin 3rt
3r
function f(t) of frequency f0, the coefficients of 1 1
16. C 0 = 2, C 2 = ,C , C - 2j, C -5 = 2j
the terms e j 2 π n f0 t and e - j 2 π n f0 t are negatives 2 -2 = 2 5 =
of each other.
19. Ans: (b)
Statement (II): The discrete magnitude
spectrum of f(t) is even and the phase spectrum
22. Ans: (c)
is odd.
24. (a) Ans: 0.45 Watt.
27. A – 2, B. –1, C. –3, D

4. Fourier Transform

4.1 Introduction
• Fourier Transform (F.T.) provides a frequency domain description of time domain signals and is extension
of F.S to non-periodic signals.
• CTFT expresses signals as linear combination of complex Sinusoids
• Transformation makes the analysis of signal much easier because certain features which may be
obscure in one form may be obvious in other form
• Spectrum of F.T. is continuous whereas spectrum of F.S is discrete.
• F.T (or) spectrum of a signal x(t) is
3
= X (ω) = # x (t) e -j~t dt ........(1)

-3
• Inverse F.T. (I.F.T) is

3
1 #
x (t) =
2π
X (ω) e j~t dω ......(2)
-3
• X (f) = # x (t) e-j2rft dt

x (t) = # X (f) e j2rft df
Convergence of F.T.
1. F.T. is defined for all stable signals i.e.,

3
# x (t) dt < 3
-3
2. Periodic signals, which are neither absolutely integrable nor square integrable over an infinite interval,
can be considered to have F.T. if impulse functions are permitted in the transform.
3. x(t) have a finite number of discontinuities and finite number of maxima and minima within any finite
interval.
F.T. of Standard Signals:
1
1. Decaying exponential: e -at u (t) * ;a>0
a + j~
1/a |X(ω)|
1 π/2 1
π/4 a 2
-a 0 a ω
0 t
- π/4
-π/2 ∠X(ω)
2. Increasing exponential
1
1
0
e at u (− t ) ↔
t a − jω

3. C.T. impulse function: δ(t) ↔ 1
t
0
0 ω
4. Rectangular (or) Gate function:-
x (t) = A rect (t/T) (or) Ar (t/T) * ATSa b 2 l (or) ATSinc b 2r l

~T ~T

| X(ω) |
AT
x(t)
A
−T/2 0 T/2 t −
4π
−
2π 0 2π 4π
ω
A A T T T T

∠X(ω)
−
4π 2π 0 2π 4π ω
− T
T T A T
-π

Spectral Width
In the frequency domain, all signals may be classified as follows:
• A broadband signal is the one, which spectrum is distributed over a wide range of frequencies as it is
shown in Fig. 4(a)
• A bandlimited signal is limited in the frequency domain with some maximum frequency as it is shown in
Fig. 4(b).
• A narrowband signal has a spectrum that is localized about a frequency f0 that is illustrated in Fig. 4(c).
• A baseband signal has a spectral contents in a narrow range close to zero (Fig. 4(d)). Accordingly, a
spectrum beginning at 0 Hz and extending contiguously over an increasing frequency range is called a
baseband spectrum.
(c)
(d)
(a)
(b)
0 f0 f
Fig (4): Types of signals: (a) broadband,
(b) bandlimited, (c) narrowband, and
(d) baseband.
Operational Properties of the Fourier Transform

Property x(t) X(f) X(ω)
(1). Similarity X(t) x(-f) 2πx(-ω)
(2). Time Scaling x(αt) 1 bfl 1 bωl

X a X α
a α
(3). Folding x(-t) X(-f) X(-ω)
(4). Time Shift x(t - α) e-j2πfα X(f) e-jωα X(ω)
(5). Frequency shift ej2παt x(t) X(f - α) X(ω - 2πα)
(6). Convolution x(t) ∗ h(t) X(f) H(f) X(ω) H(ω)
(7). Multiplication x(t) h(t) X(f) ∗ H(f) 1

X (ω) ) H (ω)
2π
(8). Modulation x(t) cos(2παt) 0.5[X(f + α) + X(f- α)] 0.5[X(ω + 2πα) + X(ω - 2πα)]
(9). Derivative x′(t) j2πf X(f) jωX(ω)
(10). Times-t -j2πt x(t) X′(f) 2πX′(ω)

t
(11). Integration # x (t) dt 1 1
X (f) + 0.5X (0) δ (f) X (ω) + πX (0) δ (ω)
j2πf jω
-3
(12). Conjugation x∗(t) X∗(-f) X∗(-ω)
(13). Correlation x(t) ∗∗ y(t) X(f) Y∗(f) X(ω) Y∗(ω)
(14). Autocorrelation x(t) ∗∗ x(t) X(f) X∗(f) = |X(f)|2 X(ω) X∗(ω) = |X(ω)|2
3 3 3
# 1 # #
(15). Central ordinates x (0) = X (f) df = X (ω) dω X (0) = x (t) dt
2π
-3 -3 -3
3 3 3
(16). Parseval’s theorem E= # x (t) 2 dt = # X (f) 2 df = 1 # X (ω) 2 dω

2π
-3 -3 -3
3 3 3
# # 1 #
(17). Plancheral’s theorem x (t) y) (t) dt = X (f) Y) (f) df = X (ω) Y) (ω) dω
2π
-3 -3 -3
Some Useful Fourier Transform Pairs

Entry x(t) X(f) X(ω)
1 δ(t) 1 1
sin c b 2π l
ω
2 rect(t) sinc(f)
sin c2 b 2π l
ω
3 tri(t) sinc2(f)
rect b 2π l
4 Sinc(t) rect(f) ω
5 cosω0t [δ(f – f0) + δ(f + f0)]/2 π[δ(ω–ω0) + δ(ω+ω0)]
π
6 sinω0t [δ(f – f0) – δ(f + f0)]/2j [δ(ω–ω0) – δ(ω+ω0)]
j
7 e-αtu(t) 1 1
a + j2rf a + j~
1 1
8 te-αtu(t)
(a + j2rf) 2 (a + j~) 2
9 e- a t 2a 2a
a2 + 4r2 f2 a2 + ~2
10 2 2 2
e- π t e- π f e-ω / 4π
1 2
11 sgn(t) jr f jω
12 u(t) 1 1
0.5δ (f) + πδ (ω) +
j2πf jω
a + j2rf α + jω
13 e-αt cos(2πβt) u(t)
(a + j2rf) 2 + ^2rbh (α + jω) 2 + ^2πβ h
2 2
2rb 2πβ
14 e-αt sin(2πβt) u(t)
(a + j2rf) 2 + ^2rbh (α + jω) 2 + ^2πβ h
2 2
1 / bf − k l 2π / b ω − 2πk l
3 3 3
15 / δ (t − nT) T k =-3 δ T T k =-3 δ T
n =- 3
3 3
/ Cn e j2rf t / 2πCn δ (ω − nω0)
3
16 xp (t) = 0
/ Cn δ (f − nf0) n =-3
n =- 3 n =-3
4.2 DISTORTIONLESS TRANSMISSION • If the gain is not constant over the required
In several application such as signal frequency range, we have amplitude distortion.
If the phase shift is not linear with frequency, we
amplification or message signal transmission
have phase distortion as the signal undergoes
over communication channel, we require that
different delays for different frequencies.
the output waveform be a replica of the Input
waveform. • Ideal filters are non causal, unstable and
physically unrealizable in the sense that their
• Transmission is said to be distortionless if the characteristics can not be achieved with a
input and output have identical waveshapes finite number of elements
within a multiplicative constant (or) a delayed
• For a physically realizable system, h(t) must be
output that retains the input waveform is
causal i.e., h(t) = 0 for t < 0. In the frequency
considered to be distortionless.
domain, this condition is known as Paley –
Wiener Criterion which states that the necessary
• For distortionless transmission, the input x(t) and and sufficient condition for the amplitude
output y(t) satisfies the condition response |H(ω)| to be realizable is
y(t) = k x(t – t0)
Y (ω) +3
ln H (ω)
H (ω) =
X (ω)
= Ke -j~t 0
# 1 + ω2
dω < 3
-3
k • Phase delay tp(ω) is the delay occurring at a

Single frequency. As the signal propagates
ω from source to destination the amount of delay
caused is known as phase delay.
|H(ω)|= k θ (ω )
t p (ω ) = − ω
∠H(ω) = θ(ω) = –ωt0
• Phase delay is not necessarily the true signal
delay. A steady Sinusoidal signal doesn’t carry
information. Information can be transmitted
ω only by applying some appropriate change to
Sinusoidal wave. Suppose that a slowly varying
signal is multiplied by a Sinusoidal wave so that
Slope = –t0 resulting modulated wave consists of a narrow
group of frequencies. When this modulated
wave is transmitted through the channel, we
• For distortion less transmission, magnitude find that there is a delay between envelope
response must be a constant, phase response of Input and received signal. This is known as
must be a linear function of ω with slope –t0, envelope (or) group delay (True signal delay)
where t0 is delay in output with respective to dθ (ω)
input. t g (ω ) = −
dω
Example : The following figure illustrates the effect of ideal filters using the input voltage
v i (t) = 0.8 sin ω 0 t + 0.5 sin 4ω 0 t + 0.2 sin 16ω 0 t V
1.5v 1.5v
0v −1.5v
1.5v 1.5v
Low pass
0v −1.5v
1.5v 1.5v
High pass
0v −1.5v
1.5v 1.5v
Band pass
0v −1.5v
1.5v 1.5v
Band reject
0v −1.5v
Frequency Time
As an example, Shown in figure at the left are the spectra that we would observe with a spectrum
analyzer; shown at the right are the waveforms that we would observe with an oscilloscope. The
spectrum and waveform at the top pertain to the input signal, and those below pertain, respectively,
to the low-pass, high-pass. Band-pass, and band-reject outputs. For instance, if we send Vi(t) through a
low-pass filter with ωc somewhere between 4ω0 and 16ω0, the first two components are multiplied by 1
and thus passed, but the third component is multiplied by 0 and is thus blocked: the result is
v i ( t ) = 0.8 Sinω0 t + 0.5Sin 4ω0 t V
Example :
A channel has the frequency response
]Z]
]]4rect b 40 l e -j 30 for f # 15Hz
f rf
H (f) = ][
]] 4rect b f l e -j r2 for f > 15Hz
] 40
\
Draw the phase delay tp(f) & group delay tg(f). For what values of f does tp(f) = tg(f)?
Z] rf
]] -
Ans: ] 30 ; f # 15
i (f) = [] r
]] - ; f > 15
] 2
\ Z] 1
]]
] 60 ; f # 15
Phase delay t p (f) = [] 1
]] ; f > 15
] 4f
\
1 Properties of H.T:
Group delay t g (f) = * 60
; f # 15
1. H.T. doesn’t change the domain of a signal
0; f > 15
2. H.T. doesn’t alter the amplitude spectrum of a
signal
∴ tp(f) = tg(f) for f # 15
3. If xt (t) is H.T. of x(t), then H.T. of xt (t) is – x(t)
tg(f) 4. x(t) and xt (t) are orthogonal to each other.
tp(f)
1
60
Example : Find H.T. of
(1) x(t) = cosω0t
(2) x(t) = sinω0t
0 −15 0 15
−15 15 f 3
(3) x (t) = / an cos nω0 t + bn sin nω0 t
n=1
4.3 HILBERT TRANSFORM: (H.T) (4) x(t)cos(2πfct), where x(t) represents a

The Hilbert transform is an operation that shifts signal band limited to B, fc > B
the phase of x(t) by – π/2, while the amplitude Example :

spectrum of the signal remains unaltered. For the system shown in figure if x(t) = cost and
h(t) = (1/πt), then the output y(t) is
An ideal H.T. is an all pass 900 phase shifter.
H.T. is used in number of application such as x(t) y(t)
h(t) h(t) h(t)
representation of band pass signals, phase shift
modulators, generation of SSB. (a) cos t (b) – cos t
(c) sin t (d) – sin t
1
x(t) xˆ ( t ) = x( t ) ∗  
1  πt 
πt
4.4 CORRELATION
Frequency response of the H.T.
= –jsgn(ω) • It provides a measure of the similarity between
2 waveforms as the function of search
|H(ω)| 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
0 ω 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
∠H(ω)
absence of the reflected pulse we confirm the
π /2 presence or absence of the target.
• In digital communication, the important
ω thing is that 1s and 0s in the data stream be
distinguishable from each other so the receiver
– π /2 can reproduce the bit pattern that was
transmitted.
• Auto correlation function of an energy signal

Practice Questions
x(t) is
3 3
R x (x) = # x (t) x (t - x) dt = # x (t + x) x (t) dt

-3 -3
01. If x(t) is a voltage waveform, then what are the
units of X(f) ?
• ACF of power signals is
T
1 #
R x (x) = Lt x (t) x (t - x) dt
T"3 2T 02. For the signal x(t) shown in figure, find
-T
1
T
(a) X(0)
= Lt 2T # x (t + x) x (t) dt
(b)
T"3 +3
-T
# X (ω) dω
-3
Properties of ACF:
x(t)
1. ACF is an even function of τ i.e., x(t) 2
Rx(τ) = Rx(-τ)
1
2. ACF at origin indicates either energy (or) power
in the signal t
-1 0 1 2 3
3. Maximum value of ACF occurs at origin i.e.,
|Rx(τ)| ≤ |Rx(0)| ∀ τ
4. Rx(τ) = x(τ) ∗ x(-τ)
5. F.T. of ACF is known as ESD (or) PSD
Rx(τ) ↔ Sx(ω) ESD / PSD Properties of F.T
6. For an LTI system
Y (ω) = X (ω) H (ω) Linearity, duality & scaling
|Y(ω)|2 = |X(ω)|2 |H(ω)|2
SY(ω) = SX(ω) |H(ω)|2 03. Find the F.T of the signals
i) x (t) = e -a t [Two sided exponential]
L .T.I
x(t) y(t) ii) x(t) = Sgn(t) [Signum function]
System h(t)

output S.D = [input S.D] [|H(ω)|2] 04. The F.T. of a function g(t) is given as
ω2 + 21
G (ω) = 2 . Find g(t)?
ω +9
05. Find the F.T of the signal shown in figure Find

Y(f) at f = 1 ?
y(t)
2
t
-2 -1 0 1 2
A
06. The magnitude of F.T. X(ω) of a function 11. The F.T of a triangular pulse f(t) shown in figure
in fig (a) the magnitude of F.T Y(ω) of other e j~ − jωe j~ − 1
is F(ω) = using this find the F.T of
ω2
function y(t) is shown below in fig (b). The phases the signals shown in figure?
X(ω) and Y(ω) are zero for all ω. The magnitude f(t)

and frequency units are identical in both the 1
figures. The function y(t) can be expressed in
terms of x(t) as ______
X(ω) Y(ω) t
3 -1 0
1
f1(t) f2(t)
1
–100 100 ω 1.5
0 -50 0 50 ω

2 btl 3 ]2tg
(a) x (b) x
3 2 2 -1/2 0 1/2 t 0 2 t

2 ]2tg 3 btl
(c) x (d) x 12. If x(t) as shown in Fig. (a) has Fourier transform
3 2 2
X(f), then the Fourier transform of g(t) as shown
07. Find F.T. of the following signals: in Fig.(b) is
(i) x(t) = 1 x(t)

1 1
(ii) x (t) =
(a + jt)
2a
(iii) x (t) = 2 2
(a + t ) t
(iv) x (t) = 1
0 1 Fig. (a)
rt
08. Let x(t) = rect( t – ½ ) g(t)

{where rect(x) = 1 for –1/2 ≤ x ≤ 1/2 } then if 1
sin rx
sinc(x) = = rx
F.T. of x(t) + x(–t) will be given by ____ t
0 1 2 3 4 5

09. What is the I.F.T. of u(ω) ? Fig. (b)
(a) e-j6πfX(f ) – e-j4πfX(–f )

(b) [e-j6πf – e-j4πf]X(f)
(c) [e-j6πf – e-j4πf]X(–f )
Time & frequency shifting (d) e-j6πf X(–f ) – e-j4π f X(f )
13. Find the F.T. of the following signals?

10. Find the F.T. of the signals
i) cosω0t
i) x(t) = e -3t u(t-1)
ii) sinω0t
ii) z(t) = π [(t – 1) / 2] iii) e–at sinωctu(t)
iii) y (t) = e -2 t - 2 iv) A rect(t/T) cosω0t
14. Find the F.T. of y(t) = Sinc(t) cos10πt The output, y(t) is equal to
15. i) The inverse F.T of X(4ω+3) in terms of x(t) x(t) cos(ωct) y(t)
H(ω)
is_______
Filter
ii) I.F.T of X(ω) = 2πδ(ω) + πδ(ω - 4π)
+ πδ(ω + 4π) is
(a) 1+ cos 4πt (b) π(1– cos 4 πt) cos(ωCt +θ)

(c) 2π(1– cos 4 πt) (d) 2π(1+cos 4 πt) 1
(a) x(t) (b) x (t)
2
16. The Fourier spectrum X(f) of a signal x(t) is (c) – x(t) (d) x(t) cos(θ)
shown. The phase angle of x(t) at
t = 1/(8fo) is equal to Time & Frequency Differentiation
X(f)
19. A differentiable non constant even function
1
x(t) has a derivative y(t), and their respective
f Fourier Transforms are X(ω) and Y(ω). Which of
fo
the following statements is TRUE ?
(a) –90° (b) 180°
(a) X(ω) and Y(ω) are both real.
(c) –45° (d) 45°
(b) X(ω) is real Y(ω) is imaginary
17. F.T. of x(t) is 2rect(0.25f). Then F.T. of (c) X(ω) and Y(ω) are both imaginary
x(t)cos(2πt) is (d) X(ω) is imaginary Y(ω) is real
(a)
1
20. For the spectrum X(ω) shown in figure,
d
find x (t) at t = 0 ?
dt
- 4 f
0 4
X(ω)

j π
(b) 2
ω
-1 0 1
1 T
f -j π
-3 -1 0 1 3
(c)
4 21. Find the F.T. of te - t ,hence find the transform
4t
of
f ( 1 + t 2) 2
-3 3
0
(d) None of these 22. Given x(t)↔X(ω), express the F.T. of the following
signals in terms of X(ω) ?
18. A signal x(t) with bandwidth B is put on a carrier (i) x1(t) = x(2 – t) + x(– t – 2 )
cos(ωct) with ωc >> B. The modulated signal x(t) (ii) x2(t) = x( 3t – 6 )
cos(ωct) is then applied to a system shown in d2
(iii) x3 (t) = x (t - 3)
fig with filter frequency response given by dt2
2, − B < ω < B
H (ω) = )
tdx (t)
(iv) x4 (t) =
0, all other ω dt
23. Given sin ω t sin ω t

25. The input signal x (t) = πt 1 + πt 2 ω1< ω2
2 sin ω
x (t) = *1; t < 1 ) ω is applied to the system shown in
0; elsewhere
figure.
Find the F. T of the following signals?
H(ω)
(a) y1(t) (b) y2(t) 1
2
1
0 2 t −2 2 t -ωf 0 ωf ω

y3(t) Sin πt find y(t) if
(c) (d) y4(t)
1 (a) 0 < ωf < ω1
(b) ω1 < ωf < ω2
−1 0 1 t −1 0 1 t (c) ω2 < ωf

−1
26. Let x(t) be a signal whose F.T. is
X (ω) = δ (ω) + δ (ω − π) + δ (ω − 5) & Let
(e) y5(t) (f) y6(t) h(t) = u(t) – u(t–2)
1 (a) is x(t) periodic?
2 (b) is x(t)∗h(t) periodic?
(c) can the convolution of two aperiodic
−1 0 1 t
−1 1 t signals be periodic?
−1
27. Using convolution property of F.T. find the
(g) y7(t) (h) y8(t)
convolution of following signals.
1 1
(a) y1 (t) = rect (t) ) cos (rt)
(b) y2 (t) = rect (t) ) cos (2rt)
(c) y3 (t) = sin c (t) ) sin c b 2 l
t t
0 −2 −1 0 1 2 t
4
(d) y4 (t) = sin c (t) ) e j3rt sin c (t)
−1
y9(t) 28. (a) Find the output of a system having impulse
response h (t) = 8 sin c 68 ]t - 1g@ when the
(i) 1 (j)
input applied is x(t) = cosπt
y10(t)
1 (b) Let g (t) = e - rt , and h(t) is a filter matched
2
to g(t). If g(t) applied as input to h(t), then

1
t the Fourier transform of the output is
−2 t −2 −1
2 0 2
0 (a) e - rf (b) e - rf /2
Convolution
2 2
(c) e - r f (d) e -2rf

2
24. An L.T.I system is having Impulse response

sin 4t
h (t) = rt for which the input applied is
x(t) = cos2t + sin 6t, find the output?
29. The time function x(t) corresponding to the 2a

36. For the signal g (t) = a2 + t2 , determine the
Fourier spectrum:
essential B.W. B Hz of g(t) such that the energy
X (f) = e - r (f - 1) is
2
contained in the spectral components of g(t) of

(a) real and even symmetric
frequencies below B Hz is 99% of signal energy
(b) conjugate odd symmetric
in g(t)?
(c) conjugate even symmetric
(d) real and odd symmetric 37. The magnitude of the Fourier transform of a
signal x(t) is shown in figure.
30. Find the F.T. of The energy of the signal is
(a) y(t) = x(t)cos ω0t ? |X(f)|
Sint Sin b 2 l
t 10-6
(b) x (t) =
rt 2
t
sin (2rt) -104 104
31. Find the F.T. of # rt dt 0 f(Hz)
-3

(a) 32 # 10-8 (b) 32 # 10-8
Parseval’s Power theorem (c) 13 # 10-8 (d) 10-10
32. Find the energy in the signal Distortionless Transmission

sin at
x (t) = rt
38. Consider a transmission system H(ω) with
magnitude and phase response as shown
33. Find the energy in the spectrum shown in fig.?
in figure. If an input signal
X(ω)
x(t) = 2Cos 10πt + Sin 26πt is given to the system
π the output will be ________
π |H(ω)|
2
2
-1 -0.5 0 0.5 1 ω
- 40π −20π 20π 40π ω (rad/sec)
34. An input signal x(t) = e -2t u(t) is applied to an ideal

L.P.F with frequency response characteristics ∠H(ω)
H (ω) = 1; ω < ω c π/2
= 0; ω > ω c
30π ω
Find ωc, such that energy in the output is half - 30π
that of Input energy? - π/2

(a) 4 cos 10πt + sin 26πt
35. Find the value of the integral
+3 (b) 8 cos 10πt + sin 26πt
# ^ω2 8+ 4h2 dω (c) 4 cos (10πt – π/6) + sin (26πt – [13π/30])
-3
(d) 8 cos (10πt – π/2) + sin (26πt – π/2)
39. For a linear phase channel, what is tp & tg? 43. Consider an LTI system with magnitude
response
40. The system under consideration is an RC LPF ]Z]
] − f
with R = 1 k Ω & C = 1 μF H (f) ][1 20 , f # 20
=
]] 0, f > 20
i) Let H(f) denote the frequency response of RC and phase \ response Arg[H(f)] = –2f.
LPF. Let f1 be the highest frequency component If the input to the system is
x (t) = 8 cos b 20πt + l + 16 sin b 40πt + l + 24 cos b 80πt + l
H (f1) π π π
such that 0≤|f|≤f1, $ 0.95 4 8 16
H (0) Then the average power of the output signal
then f1 (in Hz) is _____
y(t) is ______
(a) 327.8 (b) 163.9
(c) 52.2 (d) 104.4 44. An analog filter has the magnitude and phase
(ii) Let tg(f) denote the group delay of RC LPF and characteristics shown in figure. If the following
f2 = 100 Hz, then tg(f2 ) in msec, is ____ Inputs are applied to this filter, we got the
following steady - state outputs. Tell what kind
(a) 0.717 (b) 7.17
of distortion has occurred?
(c) 71.7 (d) 4.505 |H(ω)|
2
41. The input to a channel is a bandpass signal. It
is obtained by linearly modulating a sinusoidal 1
carrier with a single-tone signal. The output of ω

−200 0 200
the channel due to this input is given by
1 ∠H(ω)
y (t) = cos (100t - 10 -6) cos (106 t - 1.56)
100 π
The group delay (tg) & the phase delay (tp) in 2
seconds, of the channel are
ω
(a) tg = 10−6, tp = 1.56 (b) tg = 1.56, tp = 10−6 −100 100
(c) tg = 10−8, tp = 1.56×10−6 (d) tg = 10−8, tp = 1.56 π

−
2
42. Suppose a transmission system has the
frequency response as shown in figure. For Input S.S. Output
what range of frequency there is no distortion?
y1 (t) = cos b 20t - 10 l + cos b60t - 10 l
r 3r
x1(t)=cos20t+cos 60t
|H(f)|
y2 (t) = cos b 20t - 10 l + cos b140t - 2 l

r r
x2(t)=cos20t+cos140t
1
y3 (t) = cos b 20t - 10 l + 2 cos b 220t - 2 l
r r
x3(t)=cos20t+cos220t
20 50 f, kHz
0 30
Arg H(f)
f
-900
Correlation Sampling Theorem
45. Find the auto correlation and power in the

51. Find the Nyquist rate & Nyquist interval for each
signal x(t) = 6 cos(6πt + π/3)
of the following signals?

(a) x1 (t) = b
46. Find the ACF of x(t) = e-3t u(t) sin 200rt l
rt
(b) x2 (t) = b
1 sin 200rt l2
47. Consider a filter with H (ω) = and input rt
1 + jω
x(t) = e u(t)
-2t
(c) x3(t) = 5cos1000πt cos4000πt
(a) Find the ESD of the output?
(b) Show that total energy in the output is (d) x4 (t) = e -6t u (t) ) sin at
rt
one-third of the input energy?
(e) x5 (t) = sin c (100t) + 3 sin c2 (60t)

48. Let g(t) = exp(–8t)u(t)
sin b 2 l + 3
πt
Define x(t) = convolution of g(t) with it self and (f) x (t) = ) / δ (t − 10n)
y(τ) = correlation of g(t) with it self b πt l n =-3
2
i) The value of x(t) at t = 1/16 is
1 1
(a) (b)
^8 e h ^16 e h
52. Let x(t) be a signal with Nyquist rate ωo.
1 Determine the Nyquist rate for each of the
(c) (d) 1/(16e)
^1 e h following signals.
d
ii) The energy density spectrum at f = 0 is (a) x(t) + x(t–1) (b) x (t)
dt
(a) 8 (b) 1/8 (c) x(3t) (d) x(t) cosωot
(c) 1/64 (d) 64
53. Two signals x1(t) & x2(t) are band limited to 2 kHz
iii) y(τ) value at τ = 0, is & 3 kHz respectively, find the Nyquist rate of the
1 1
(a) (b) following signals?
8 16
1 (a) x1(2t)
(c) (d) 16
4 (b) x2(t – 3)
(c) x1(t) + x2(t)
49. Find the C.C.F of x(t) = e-tu(t) and
(d) x1(t)x2(t)
y(t) = e-3tu(t) ?
(e) x1(t)∗ x2(t)
(f) x1(t) cos(1000πt)
50. The signal x(t) = sinc10t is the input to a system
with frequency response
H (ω) = 3rect b 8π l e -j2~
ω 54. A signal x(t) = 100 cos(24π×103t) is ideally

sampled with a sampling period of 50 µsec
Find the output energy?
and then passed through an ideal low pass
filter with cutoff frequency of 15 kHz. Which of
the following frequencies is/are present at the
filter output?
(a) 12 kHz only
(b) 8 kHz only
(c) 12 kHz and 9 kHz
(d) 12 kHz and 8 kHz
55. A signal represented by x(t) = 5cos(400πt) 57. A signal x(t) = 6cos10πt is sampled at a rate
is sampled at a rate of 300Hz. The resulting of 14 Hz to recover the original signal, cut-off
samples are passed through an ideal LPF frequency of the LPF should be _________
with cut-off frequency of 150 Hz. Which of the (a) 5 < fc < 9 (b) 9
following will be contained in the output of LPF? (c) 10 (d) 14
(a) 100 Hz
(b) 100 Hz, 150 Hz 58. The spectrum of a bandlimited signal after
(c) 50 Hz, 100Hz sampling is shown in figure. The value of
(d) 20,100,150Hz sampling interval is
56. The frequency spectrum of a signal is shown in

figure, if this signal is ideally sampled at intervals
of 1 msec, then the frequency spectrum of the
f(Hz)
−100 0 100 150 350 400 600
sampled signal will be _________
U (f) (a) 1msec (b) 2msec
(c) 4msec (d) 8msec
59. Let x(t) = 2 cos(800πt) + cos(1400πt) and x(t) is

f (kHz)
–1 0 1 sampled with the rectangular pulse train shown
in fig. The only spectral components (in kHz)
(a) present in the sampled signal in the frequency
range 2.5kHz to 3.5kHz.
f p(t)
(b) 3
f t
–T0 –T0/6 0 T0/6 T0
(c) T0 = 10–3sec
(a) 2.7, 3.4

(b) 3.3, 3.6
f (c) 2.6, 2.7, 3.3, 3.4, 3.6
(d) 2.7, 3.3
(d)
f
60. Hilbert Transform & Complex Envelope

(i) The output y(t) of the following system is to be
sampled, so as to reconstruct it from its samples 61. The complex envelope of the bandpass signal
uniquely. The required minimum sampling
x (t) = - 2 c m sin b rt - r l ,
sin (rt/5)
rate is r t/ 5 4
1
X(ω) centered about f = Hz, is
2
(a) c m e j 4
sin (rt/5) r

rt/5
ω
−1000π 1000π
(b) c me
sin(1500πt) sin (rt/5) -j r4
h(t) = y(t) rt/5
x(t)↔ X(ω) πt
(c) 2 c m e j 4
sin (rt/5) r
rt/5
cos(1000πt)
(d) 2 c me
sin (rt/5) -j r4
(a) 1000 samples/s rt/5
(b) 1500 samples/s
(c) 2000 samples/s 62. A modulated signal is given by
(d) 3000 samples/s s(t) = e-atcos[(ωc+∆ω)t]u(t), where a, ωc are
positive constants, and ωc >> ∆ω. The complex
envelope of s(t) is given by
(ii) The signal cos b10rt + 4 l is ideally sampled
r
(a) exp(-at) exp[j(ωc+ ∆ω)t]u(t)
at a sampling frequency of 15 Hz. The (b) exp(-at) exp(j∆ωt)u(t)
sampled signal is passed through a filter (c) exp(j∆ωt)u(t)
with impulse response b sin ]rtg l cos b 40rt - r l . (d) exp[j(ωc + ∆ω)t]
rt 2
The filter output is
63. The input 4sinc(2t) is fed to a Hilbert transformer
cos b 40rt - 4 l
15 r
(a) to obtain y(t), as shown in the figure below:
2
cos b10rt + 4 l
15 c sin (rt) m r
(b) Hilbert
2 rt y(t)
4sinc(2t)
Transform
cos b10rt - 4 l
15 r
(c)
2
cos b 40rt - 2 l
15 c sin (rt) m r sin (πx)
(d) Here sin c (x) = πx . The value (accurate to
2 rt 3
two decimal places) of # y (t) 2 dt is __________

-3
64. A modulated signal is y(t) = m(t)cos(40000πt),

where the baseband signal m(t) has frequency
components less than 5 kHz only. The minimum
required rate (in kHz) at which y(t) should be
sampled to recover m(t) is ______.
65. Statement (I): Aliasing occurs when the

sampling frequency is less than twice the
maximum frequency in the signal.
01. Ans: Volt/Hz
Statement (II): Aliasing is a reversible process.
02. (a). Ans: 7, (b). Ans: 4π

66. Statement (I): Sampling in one domain makes
the signal to be periodic in the other domain.
04. Ans: g (t) = δ (t) + 2e -3 t
Statement (II): Multiplication in one domain is
the convolution in the other domain. 05. Ans: 0 06. Ans: (d) 12.Ans: (a)
1 - 34 jt b t l
15. i. Ans: e x 4 ii. Ans: (a)
4
17. Ans: (b)
18. Ans: (d)
24. Ans: cos(2t)
2 sin ]ω f tg
25. a). Ans: πt ,
sin (ω1 t) sin (ω f t)

b). Ans: πt + πt
c). Output = Input
26. Ans: a is non - periodic, b is periodic, c may

be periodic.
2
27. a). Ans: y1 (t) = r cos rt ,
b). Ans: y2 (t) = 0
c). Ans: Sinc b 2 l

t
d). Ans:0
28. (a). Ans: cos π (t - 1), (b). Ans: d
29. Ans: (c)
a
32. Ans: r
35. Ans: π/2
2.302
36. Ans: B = a rad/ sec
37. Ans: (a) 38. Ans: (c)

40. Ans: (i). c, (ii). a,

5. Laplace Transform
45. Ans: rxx (x) = 18 cos (6rx) , power = 18
e -3 x • L.T expresses signals as linear combination

46. Ans:
6 of complex exponentials, which are eigen
functions of D.E which describe continuous –
48. Ans: (i). b, (ii). c, (iii). b
time L.T.I systems.
51. Ans:
a) 200 Hz, b) 400 Hz, c) 5 KHz, • The primary role of the L.T in engineering is
d) 2a e) 120 Hz. the transient & stability analysis of Causal L.T.I
systems.
52. Ans:
a). ω0, b). ω0, c). 3 ω0, d). 3 ω0, • L.T provides a broader characterization of
systems & their interaction with signals than is
possible with F.T.
55. Ans: (a)
• In addition to its simplicity, many design
56. Ans: (b) techniques in circuits, filters & Control systems
have been developed in L.T. domain.
57. Ans: (a)
• Consider applying an INPUT of the form x(t) = est

58. Ans: (c)
(where s = σ +jω then the output is y(t) = estH(s)
where H(s) is transfer function of the system.
60. (ii) Ans: (a)
• L.T. of a general signal x(t) is

3
X (s) = # x (t) e -st dt --------- (1)

-3
= F.T {x(t)e−σt}
• e-σt may be decaying (or) growing depending

on whether ‘σ’ is +ve (or) –ve.
x (t) ) X (s)

5.1 Region of Convergence (R.O.C) of L.T:

The range of values
3
of ‘S’ for which eq(1) is
satisfied i.e. # |x(t)e-σt|dt < ∞, is known as
R.O.C of L.T. -3
• L.T. calculated on the jω-axis (σ = 0) is F.T.

L.T. of Standard Signals

6. unit step function u (t) ) s ; Re ! s + > 0
1
x (t) = e -at u (t), a > 0 ) s + a ; Re ! s + > - a

1
1. Note: If the L.T. X(s) of x(t) is rational, then if x(t) is
right sided the ROC is the region in the s-plane
jω
to the right of the rightmost pole and if x(t) is left
1
sided, ROC is left of the left most pole.

-a σ

0 t 5.2 Properties of L.T
1. Linearity:
x (t) = - e -at u (- t) ) s + a ; Re ! s + < - a
1
2.
If x1(t) ↔ X1(s) with ROC = R1
jω x2(t) ↔ X2(s) with ROC = R2
0
Then a x1(t) + b x2(t) ↔ aX1(s) + bX2(s)
t with ROC = R1 ∩ R2
-1 -a σ
2. Time-shifting:

x(t) ↔ X(s), ROC = R
then x (t - t 0) ) e -st X (s)

x (t) = e at u (t) ) - ; Re ! s + > a
1
0
3.
s a
with ROC = R
jω

3. Shift in S-domain
1
x(t) ↔ X(s) with ROC = R
a σ
then x (t) e s t ) X (s - s 0)
0
0 t
with ROC = R + Re(s0)
x (t) = - e at u (- t) ) - ; Re ! s + < a
1
4.
s a 4. Time-reversal:
jω x(t) ↔ X(s)
then x(–t) ↔ X(–s), ROC = – R
0
t 5. Differentiation in time:
a σ
x(t) ↔ X(s) with ROC = R
dx (t)
then ) sX (s) with ROC = R
dt
5. unit impulse δ(t)↔1; ROC: entire s-plane

UNILATERAL L.T Ans: (a) 5cos (2t+30°)

3
X (s) = # x (t) e -st dt j2 + 2

ω 0 = 2, H (jω 0) = −
4 + 10j + 4
0
Differentiation in time: j2 + 2 j + 1
= =
d 10j 5j
x (t) * sX (s) - x (0) 1 1
dt H (jω 0) = 5 +
d2 5j
x (t) * s2 X (s) - sx (0) - xl (0)
dt2 1 j
=5-5
Example:
1 1 2
H (ω 0) =
Suppose that an LTI system has the following 25 + 25 = 5
8 −1 5
H (jω 0) = tan -1 e o= π
T.F H (s) = . Compute the system response
−
s+4
due to following inputs. Identify the steady - 1 5 4
state & transient solution? 2
y (t) = 5 # 5 cos (2t + 30 o - 45 o)
(a) x1(t) = u(t)
y1(t) = x1(t)*h(t)
= 2 cos (2t - 15 o)
8 2 2
Y1(s) = X1(s)H(s) = = s -s 4
s (s + 4) +
Take I.L.T. (b) Ans: 2 2 sin (2t)
y1 (t) = 2u (t) - 2e -4t u (t)
Z 1444442444443 Sol: ω0 = 2, |H(jω0)| = 5
2
s.s Transient
(b) x2(t) = tu(t) π

H (jω 0) = −
4
8 1 2 1 2 2
Y2(s) = X2(s)H(s) = = -
s2 (s + 4) s + 4 s + s2 2
y (t) = 10 # 5 sin (2t + 45 o - 45 o)
1 -4t 1
y2 (t) = e u (t) - u (t) + 2tu (t) y (t) = 2 2 sin (2t)
2
1444442444443 12444444442444444443
Transient S.S
(c) x3 (t) = 2 sin 2tu (t)
8 5 8 5 s 16 2
Y3 (s) = : s + 4 D ; s2 + 4 E = s + 4 - s2 + 4 + s ; s2 + 4 E
8 4
8 - 4t - 8 16
y3 (t) = >1244424443 124444444444424444444444
e cos 2t + 5 sin 2tH
43 u (t)
Transient S.S
Note that the form of the transient remains the

same no matter what the input is !
Example:
For an L.T.I system described by the transfer
s+2
function H (s) = 2 , find the response
s + 5s + 4
due to
(a) 5cos(2t+300)
(b) 10sin(2t+450)
06. Find the L.T of the waveform shown in figure?

Practice Questions

x(t)

01. Find the L.T. of the following signals with R.O.C? 10
1. x1(t) = e-t u(t) + e-3t u(t)
2. x2(t) = e-2t u(t) + e4t u(-t)
3. x3(t) = e-t u(-t) + e5t u(t)
0 2 4 t(secs)
4. x4(t) = 1 ∀ t -5
5. x5(t) = sgn(t)
6. x6(t) = e-|t|
07. L.T. of the waveform shown in fig is
1^
02. Consider the signal x(t) = e u(t) + e u(t) & its L.T. 1 + Ae -s + Be -4s + Ce -6s + De -8sh then D
–5t –βt
is
is X(s). What are the constraints placed on the s2
______
real & imaginary parts of β if the R.O.C of X(s) is
Re{s} > - 3 ?
1
03. How many possible ROCs are there for the

pole-zero plot shown in fig ? 6 7 8
0 1 2 3 4 5 t
jω
-1
σ Fig.

-3 -1 1 2
(a) –0.5 (b) –1.5
(c) 0.5 (d) 2
Fig

08. Let x(t) be a signal that has a rational L.T. with
Time shifting and shift in S - domain
exactly 2 poles located at s = −1 and s = −3.
04. Find the I.L.T. of If g(t) = e2tx(t) and G(ω) converges, determine
e -3s
Y (s) = , σ > -1 whether g(t) is
(s + 1) (s + 2)
(a) Left-sided
05. Consider the signal x(t) = e u(t−1) with L.T. X(s)
−5t
(b) right-sided
(c) two-sided
(a) Find X(s) with R.O.C.?
(d) finite-duration.
(b) Find the values of ‘A’ &‘t0’ such that the
L.T. G(s) of g(t) = Ae-5tu(-t-t0) has same 09. Let g(t) = x(t) + αx(–t) where
algebraic form as X(s). What is the R.O.C
x(t) = βe–tu(t) and
corresponding to G(s)?
; - 1 < Re ! s + < 1
s
G (s) = 2
s -1
Find α and β?
Differentiation 17. For the interconnected system shown in figure,

the combined Casual impulse response is
10. Consider 2 right-sided signals x(t) & y(t) related
x(t) v(t) y(t)
through the equation v ( t ) + v ( t ) = x ( t ) y ( t ) + y ( t ) = v ( t )
dx (t) dy (t)
dt =
− 2y (t) + δ (t)&
dt =
2x (t)
Fig.

Find X(s) & Y(s) with ROCs?
(a) e-tu(t) (b) te-tu(t)
(c) δ(t ) (d) u(t)
11. Find the ILT of
4
(a) X (s) = 18. The running integrator, given by
(s + 2) (s + 1) 3
t
dd 1 n # x (t') dt'
(b) X (s) = e-2s ds (s + 1) 2
y (t) =
-3
Convolution (a) has no finite singularities in its double sided

Laplace Transform Y(s)
12. Solve the following equation. (b) produces a bounded output for every
3
y (t) + # y (τ) x (t − τ) dτ = x (t) + δ (t) ? causal bounded input

0 (c) produces a bounded output for every
13. Consider a signal y(t) = x1(t–2) ∗ x2(–t + 3) where anticausal bounded input
x1(t) = e-2t u(t) & x2(t) = e-3tu(t). (d) has no finite zeros in its double sided
Find Y(s) with ROC? Laplace Transform Y(s)
19. A causal LTI system has zero initial conditions

14. Find the impulse response of a linear causal
and impulse response h(t). Its input x(t) and
system describe by the equation.
t output y(t) are related through the linear
dy (t)
+ 4y (t) + 3 # y (x) dx = x (t) constant - coefficient differential equation
dt
-3
d2 y (t) dy (t)
Also determine response to an excitation + a dt + a2 y (t) = x (t)
dt 2
x(t) = u(t) + δ(t) ? Let another signal g(t) be defined as

t
15. An Input x(t) = exp(–2t) u(t) + δ(t – 6) is applied to g (t) = a2 # h (x) dx + dhd(t t) + ah (t) .
an L.T.I system with impulse response 0
h(t)= u(t). The output is If G(s) is the Laplace transform of g(t), then the
(a) [1 – exp(–2t)]u(t) + u(t + 6) number of poles of G(s) is _______.
(b) [1 – exp(–2t)]u(t) + u(t – 6)
(c) 0.5[1 – exp(–2t)] u(t) + u(t + 6)
(d) 0.5[1 – exp(–2t)] u(t) + u(t – 6)
16. Consider an LTI system with impulse response

h(t) = e−5t u(t). If the output of the system is
y(t) = e−3t u(t) − e−5t u(t) then the input, x(t), is given
by
(a) e−3t u(t) (b) 2e−3t u(t)
(c) e−5t u(t) (d) 2e−5t u(t)
Unilateral L.T 24. An LTI system is having transfer function

s2 + 1
and input x(t) = sin(t+1) is in steady
s + 2s + 1
2
20. A system described by a linear, constant state. The output is sampled at ωs rad/sec to
coefficient, ordinary, first order differential obtain final output {y(k)} which of the following
equation has an exact solution given by y(t) for
is true?
t > 0, when the forcing function is x(t) and the
(a) y(•) = 0 for all ωs
initial condition is y(0). If one wishes to modify
the system so that solution becomes –2y(t) for (b) y(•) ≠ 0 for all ωs
t > 0, we need to (c) y(•) ≠ 0 for ωs > 2 but zero for ωs < 2
(a) change the initial condition to –y(0) and (d) y(•) = 0 for ωs >2 but nonzero for ωs < 2
the forcing function to 2x(t)
(b) change the initial condition to 2y(0) and 25.
the forcing function to –x(t) (i) What is the output as t → ∞ for a system that
(c) change the initial condition to j 2 y (0) 2
has T.F., G (s) = 2 when subjected to a
s -s-2
and the forcing function to j 2 x (t) step input?
(d) change the initial condition to –2y(0) and (a) –1 (b) 1
the forcing function to –2x(t) (c) 2 (d) unbounded
(ii) The transfer function of a causal LTI system
21. Find the initial & final values for the following is H(s) = 1/s. If the input to the system is
L.T.? x(t) = [sin(t)/πt]u(t); where u(t) is a unit step
2s + 5 function. The system output y(t) as t→∞ is ____
(a) X (s) =
s2 + 5s + 6
4s + 5 26. Let a signal a1sin(ω1t + φ1) be applied to a stable
(b) X (s) =
2s + 1
LTI system. Let the corresponding steady state
12 (s + 2)
(c) X (s) = output be represented as a2F(ω2t + φ2). Then
s (s2 + 4)
which of the following statement is TRUE?

(d) X (s) = e -s < F
-2

s (s + 2) (a) F is not necessarily a “sine” or “cosine”
function but must be peroidic & ω1 = ω2
22. A LTI, Causal continuous time system has a
(b) F must be “sine” or “cosine” with a1= a2
rational transfer function with simple poles at (c) F must be “sine”, ω1 = ω2, a1 ≠ a2
s = -2 and s = -4 and one of the simple zero at (d) F must be “sine” (or) “cosine” functions
s = -1. A unit step u(t) is applied as the input of with ω1 = ω2.
the system. At steady state, the output has a
constant value of 1. Find the impulse response?
23. Consider a system with transfer function

s-2
H (s) = 2 . Find the steady-state
s + 4s + 4
response when the input applied is 8cos2t?
Causality & Stability 31. The Laplace transform of a causal signal

s+2
Y (s) = + . The value of the signal y(t) at
s 6
27. Consider an LTI system for which we are given t = 0.1 sec is _______
the following information
s+2
X (s) = − and x(t) = 0, t > 0 and output is
s 2
2 2t − + 1 -t
y (t) = − e u ( t) e u (t)
3 3
1 1
(a) Find T.F & R.O.C.? 01. (1) = + + + , σ > − 1
s 1 s 3
(b) Find the output if the input is 1 − 1 −
(2) = 2<σ<4
s+2 s−4
x(t) = e ∀ t using part (a)?
3t
(3) No ROC, No L.T

(4) No ROC, No L.T
28. Consider an LTI system with input x(t) and
(5) No ROC, No L.T
output y(t) related as
dy (t) d2 x (t) dx (t)
dt + 3y (t) = + dt - 2x (t) 02. Ans: Re[β] = 3, Img[β] any value
dt2
Find the T.F. of inverse system. Does a stable &
Causal inverse system exist? 04. Ans: y (t) = e - (t - 3) u (t - 3) - e -2 (t - 3) u (t - 3)
e - ( s + 5)
29. Which one of the following statements is NOT 05. Ans: s + 5 , σ > −5
TRUE for a continuous time causal and stable A = −1, to = −1
LTI system?
5 - 5e -2s - e -2s 5e -4s
(a) All the poles of the system must lie on the 06. Ans: X (s) = 15. s + s
s 2
s 2
left side of the jω axis.

(b) Zeros of the system can lie anywhere in 07. Ans: (a) 08. Ans: (c)
the s-plane.
1
(c) All the poles must lie within |s| = 1. 09. Ans: a = - 1, b =
2
(d) All the roots of the characteristic equation
s
must be located on the left side of the jω 10. Ans: X (s) = ,v > 0
s2 + 4
axis. 2
Y (s) = ,v > 0
s2 + 4
30. The output of a causal all-pass system is
y(t) = e-2tu(t), for which the system function is 12. Ans: y(t) = δ(t)
s-1
H (s) = s + 1 . Then the corresponding stable
e -5s
input signal is 13. Ans: Y (s) = −2 < σ < 3
(s + 2) (3 - s)
2 1
(a) − e t u (− t) + e -2t u (t) - 1 -t 3
3 3 14. Ans: h (t) = e u (t) - e -3t u (t)
2 2
2 t 1 - 2t
(b) e u (t) + e u (t)
3 3
1 2 15. Ans: (d) 17. Ans: (b)
(c) − e u (t) + e t u (− t)
-2t
3 3
1 2 22. Ans: h(t) = −4e-2tu(t) + 12e-4tu(t)
(d) e t u (- t) - e -2t u (t)
3 3
23. Ans: y(t) = 2.8288 cos(2t + 45°)

6. DTFT
24. Ans: (a) • The DTFT describes the spectrum of discrete
signals & formalizes that discrete signals have
27. Ans:
s periodic spectra. The frequency range for a
(a) H (s) = , σ > −1
(s + 1) (s + 2)
discrete signal is unique over ( –π, +π) (or)(0, 2π)
(b) y (t) = 3 e3t 3
20 X (e j~) = / x (n) e-j~n
n =-3
28. Ans:
s+3 1 #
H inv (s) = , it can’t be both stable x (n) = X (e j~) e j~n dω
(s + 2) (s - 1) 2π
and causal. < 2r >
• X(ejω) is decomposition of x(n) into its frequency

components.

6.1 Convergence of DTFT
A sufficient condition for existence of DTFT is

+3
/ x (n) < 3
n =-3
• Some sequences are not absolutely summable,

but they are square summable.
• There are signals that are neither absolutely

summable nor have finite energy, but still have
DTFT.
1
anu(n), |a| < 1 ↔ -
1 ae -j~

δ(n) ↔ 1

Periodicity property:
X(e j(ω+2π)) = X(ejω)
Time – Shift:
x(n) ↔ X(ejω)
then x(n-n0) ↔ e -j~n X(ejω) 0
Frequency – shift:
x(n) ↔ X(ejω)
then x (n) e j~ n * X (e j (~ - ~ ))
0 0
6.2 Oversampling and Sampling Rate Conversion the same signal sampled at NS Hz extends only to
In practice, different parts of a DSP system are often B/NS Hz, and the spectrum of the signal sampled
designed to operate at different sampling rates at S/M Hz extends farther out to BM/S Hz. After an
because of the advantages it offers. For example, analog signal is first sampled, all subsequent sampling
oversampling the analog signal prior to digital rate changes are typically made by manipulating
processing can reduce the errors (Sinc distortion) the signal samples (and not re-sampling the analog
caused by zero-order-hold sampling devices. Since signal). The key to the process lies in interpolation
real-time digital filters must complete all algorithmic and decimation.
operations in one sampling interval, oversampling
can impose an added computational burden. Zero Interpolation and Spectrum Compression:
It is for this reason that the sampling rate is often A property of the DTFT that forms the basis for signal
reduced (by decimation) before performing interpolation and sampling rate conversion is that
DSP operations and increased (by interpolation) M fold zero interpolation of a discrete-time signal
before reconstruction. Oversampling prior to signal x[n] leads to an M-fold spectrum compression and
reconstruction allows us to relax the stringent replication, as illustrated in Figure (ii).
requirements for the design of reconstruction filters.
Figure (i) shows the spectrum of sampled The zero-interpolated signal y[n] = x[n/N] is
signals obtained from a band-limited analog signal nonzero only if n = kN, k = 0, ±1,± 2, ......(i.e., if n is an
whose spectrum is X(f) at a sampling rate S, a higher integer multiple of N). The DTFT Yp(F) of y[n] = y[kN]
rate NS, and a lower rate S/M. may be expressed as
X(f) Sampling rate =S

Yp ]F g y 5n? e / y]kNge
SX(f) 3 3
= /
= -j2rnF -j2rkNF
f f n =- 3 k =- 3
B B S
/ x]k ge = X p ]NFg
3
Sampling rate =NS Sampling rate =S/M = -j2rkNF
NSX(f) SX(f)M k =- 3
f f
B S B S
x[n] Discrete-time signal
Figure (i) Spectra of a signal X[F] Spectrum of
sampled at three sampling rates discrete-time signal
n
F
1234 0.5 1
y[n] Fourfold zero interpolation
The spectrum of the oversampled signal shows
Y[F] Fourfold spectrum compression
a gain of N but covers a smaller fraction of the
principal period. The spectrum of a signal sampled n F
1234 0.5/4 0.5 1
at the lower rate S/M is a stretched version with
a gain of 1/M In terms of the digital frequency F, Figure (ii) Zero interpolation of a signal
leads to spectrum compression
the period of all three sampled versions is unity.
One period of the spectrum of the signal sampled
at S Hz extends to B/S, whereas the spectrum of
This describes Yp(F) as a scaled (compressed)

Practice Questions
version of the periodic spectrum Xp(F) and leads to
N-folds spectrum replication. The spectrum of the
01. Determine whether or not the DT systems with
interpolated signal shows N compressed images in
these frequency response are causal?
the principal period |F| ≤ 0.5 with the central image
sin b 2 l
7ω
occupying the frequency range |F| ≤ 0.5/N. This is (a) H (ω) =
sin b 2 l
ω
exactly analogous to the Fourier series result where
sin b 2 l
3ω
spectrum zero interpolation produced replication
(b) H (ω) = e -j~
sin b 2 l
(compression). The spectrum of the interpolated ω
signal y(n) shows N compressed images per period

(c) H(ω) = e –j3ω + e +j2ω
1
centered at F = N
x (Mn) * Yp ]F g = M X p b M l
1 F 02. (a) Let x(n) = (1/2)n u(n), y(n) = x2(n) and
Y(e jω) be the F.T of y(n). Then Y(ej0) is
(b) Given X(ejω) = cos3(3ω), then find the sum
The factor 1/M ensures that we satisfy Parseval’s
3
relation. If the spectrum of the original signal x[n] S= / (- 1) n x (n)

n =-3
extends to |F|≤ 0.5/M in the central period, the (c) What is the d.c & high - frequency gain of
spectrum of the decimated signal extends over the filter described by h(n) = {1, 2, 3, 4}
|F| ≤ 0.5, and there is no aliasing or overlap
between the spectral images. 03. (i) Find the inverse DTFT of
X(ejω) = 1 + 2 cosω + 3 cos2ω
(ii) The DTFT of x(n) = 2δ[n+3] − 3δ[n−3],

can be expressed in the form
X(ejω) = asin(bω) + cejdω Find a, b, c & d.
04. Find the signal corresponding to the spectrum

shown in figure?
Y (ejω)
1
-3π/4 -π/4 π/4 3π/4 ω

05. (a) Let h(n) is the impulse response of ideal L.P.F

with cutoff frequency ωc, what type of filter
has unit impulse response as
( g n) = ]- 1gn h (n)
(b) A discrete system with input x(n) & output Convolution

y(n) are related as
10. Consider x (n) = sin b 8 l - 2 cos b 4 l

y(n) = x(n) + (−1)n x(n) rn rn
If the input spectrum X(ejω) is shown below Find the output if the impulse response is
sin b 6 l
the o/p spectrum values at ω = 0 & ω = π rn
are h (n) = rn
X(ejω) 11. An L.T.I system is having impulse response

1 ]Z] 4 2 ; n = 2, - 2
]]
h (n) = []] - 2 2 ; n = 1, - 1
]] 0 ; elsewhere
\
ω Find the output when input applied is
−π −π/2 0 π/2 π
x(n) = ejnπ/4 ?
Scaling & Frequency Differentiation 12. For each of the following pair of signals
determine whether or not the system is LTI if
1 such system exists find the frequency response
06. Find the I.F.T of Y (e j~) = 1 ?
1 - e -j10~
(a) x1 (n) b=
2
= 1 ln b 1 l u (n)
n
2 u (n ), y1 (n)
4
= (b) x2 (n) e= jnr/4
, y2 (n) 0.5e jnr/4
07. Given the signal x (n) = %1, 2, 3- , 2, 1, 0 / .
sin (nπ/4) sin (nπ/2)
Then Fourier transform of x[2n] is (c) x3 (n) = nπ , y3 (n) = nπ
(a) 3 + 2cos 2ω + 4 cosω
(b) 3 + 2cosω 13. Design a 3 point FIR filter with impulse response
h (n) = {a, b, a} and the amplitude response
(c) 3 + 2cos2ω -
(d) 3 + 2cosω + 2cos(ω/2) blocks the frequency f = 1/3 & passes the
frequency f = 1/8 with unity gain. What is the
D.C gain of the filter?
08. The spectrum of signal x[n] is X(F) = 2tri(5F).
Sketch X(F) and the spectra of the following
14. Consider the system described by the
signals and explain how they are related to
equation y(n) = ay(n–1) + bx(n) + x(n–1), where
X(F).
‘a’ & ‘b’ are real, find the relation between
i. y[n] = x[n/2]
‘a’ & ‘b’ such that
ii. d[n] = x[2n]
|H(ejω)| = 1∀ω ?
x [n]; even "n"
iii. g (n) = )
0; odd "n"
15. A filter is described by
09. Find the F.T of x (n) = ne

jnr
8 a n - 3 u (n - 3) y(n) = x(n) + αx(n–1)+ βx(n–2). Then the values of
α & β, such that the input x(n) = 1 + 4 cos (nπ)
results in the output y(n) = 4, are
(a) α = 2, β = 1 (b) α = 1, β = 2
(c) α = β = 1 (d) α = 1, β = ± 2
16. An input x(n) with length 3 is applied to a LTI 21. Let h[n] be the impulse response of a discrete-
system having an impulse response h(n) of time linear time invariant (LTI) filter. The impulse
response is given by h (0) = ; h 51? = ;
length 5, and Y(ω) is the DTFT of the output y(n) 1 1
3 3
h 52? = ; and h[n] = 0 for n < 0 and n > 2.
of the system. If |h(n)| ≤ L & |x(n)|≤ B∀n, the 1
3
maximum value of Y(0) can be …. Let H (ω) be the Discrete-Time Fourier transform
(a) 15 LB (b) 12 LB (DTFT) of h[n], where ω is the normalized angular
(c) 8 LB (d) 7 LB frequency in radians. Given that H(ω0)= 0 and
0 < ω0 < π, the value of ω0 (in radians) is equal to
17. An L.T.I filter is described by the difference ______.
equation y(n) = x(n) + 2x(n-1) + x(n-2)
(a) Obtain the magnitude & phase response? 22. Find the energy in the signal
(b) Find the o/p when the input is sin ω n
x (n) = πnc
x (n) = 10 + 4 cos : 2 + 4 D ?
rn r
sin b 4 l sin b 3 l
nr nr
+3
18. The impulse response of a causal linear phase 23. Find the value of / 2rn 5rn
n =-3
discrete time system of five samples is given
by h(0) = 2, h(1)= -3, h(2) = 0, h(3) = 3 and
h(4)= k. The value of k and slope of the phase 24. For the signal shown in fig., find the following
curve are respectively quantities without calculating DTFT?
(a) 2, –2 (b) –2, –2
x(n)
(c) 0, 2 (d) –3, 2
2
19. The frequency response of a discrete LTI system
1
is H(ejω) = e -jω/4, -π<ω≤ π.
Then the output of the system due to the input,
x 5n? = cos : 2 D is
5rn -1 0 n
-3 -2 1 2 3 4 5 6 7
-1
(a) cos : r2n + r4 D

(b) cos : 2 - 8 D
rn r Fig.
(c) cos : 2 + 6 D
3rn r (a) X(ej0) (b) X(ejπ)
r r
(c) # X (e j~) dω (d) # X (e j~) e j2~ dω

(d) 2 cos : r2n + 3r D -r -r
20. A signal x(n) is defined by (e) # X (e j~) dω

2
x(n) = 1, for n = ±1, 3 & 5 -r
= 2, for n = 0 & 4
r
= 0, elsewhere d 2
(f) # dω
X (e j~) dω
Its phase spectrum at ω = 0.25π is -r
(a) 0.25π (b) – 0.5π

(g) X (e j~)
(c) 0.5π (d) π
25. Given h(n) = [1, 2, 2], f(n) is obtained by

convolving h(n) with itself and g(n) by
correlating h(n) with itself. Which one of the
01. (a). Ans: Non causal
following statements is TRUE?
(b). Ans: causal
(a) f(n) is causal and its maximum value is 9
(c). Ans: Non causal
(b) f(n) is non-causal
(c) g(n) is causal and its maximum value is 9 4
02. (a). Ans: 3 (b).Ans: -1
(d) g(n) is non causal and maximum value is 9
(c). Ans: DC gain = 10, HF gain = -2
sin b 4 l
26. A continuous time signal x(t) is to be filtered to nr
04. Ans: y (n) = 2 cos b 2 l .
nr
remove frequency component in the range nr
5kHz ≤ f ≤ 10 kHz. The maximum frequency
present in x(t) is 20 kHz. Find the minimum 05. (a). Ans: Ideal High pass filter
sampling frequency & find frequency response (b). Ans: 1
of ideal digital filter that will remove the desired
07. Ans: (b)
frequencies from x(t)?
10. Ans: sin b 8 l

nr
27. The sequence x 5n? = cos : 4 D was obtained
rn
by sampling a continuous signal, x(t) = cos(Ω0t) jnr

11. Ans: y (n) = - 4e 4
at a sampling rate of 1000 Hz. The two possible
values of Ω0 that have resulted in the sequence 3
13. Ans: DC gain =
x[n] are respectively 1+ 2
(a) 250π and 2250π
14. Ans: b = -a
(b) 125π and 2250π
(c) 250π and 1125π 15. Ans: (a)
(d) 125π and 1125π
16. Ans: (a)
28. A continuous time signal x(t) is sampled at
fs = 9kHz is passed through a digital filter with 17. Ans: (b)
impulse response h(n) = 0.5[δ[n] + δ[n-1]].
Find the 3 dB cut-off frequency of the analog 18. Ans: (b)
filter?
19. Ans: (b)
1
23. Ans:
40
25. Ans: (d)

π π
26. Ans: fs = 40 kHz, #ω#
4 2
27. Ans: (a)
Z.T. of standard Signals:

7. Z-transform
1 z
a n u (n), 0 < a < 1 * or - ; z > a
1 - az -1 z a
7.1 INTRODUCTION TO Z-TRANSFORM
• Discrete-time counterpart of L.T. is Z.T.

• For a D.T.L.T.I system with impulse response h(n),
the response y(n) of the system to a complex
exponential Input of the form zn is y(n) = znH(z) n
0 1 2 3 4
where H(z) is known as transfer function of the
system. Im{z}
Z.T. of a general D.T. signal x(n) is
3
X (z) = / x (n) z-n ------------ (1)
n =- 3
where z = rejω → Complex Variable
x (n)
z
X (z) a 1 Re{z}
X(z) = F{x(n)r-n}
• Z.T. calculated on the unit circle is D.T.F.T.
- a n u (- n - 1) * 1 z
or - ; z < a
Im{z} 1 - az -1 z a
z-plane
−4 −3 −2 −1
n
1 Re{z}
Unit circle

• The range of values of ‘z’ for which eq (1) is Im
defined

= x (n) r -n < 3G is R.O.C. of Z.T. 0
3
/ Re
n =- 3 a
• DTFT is defined only for stable signals where as

Z.T. is defined for unstable signals also.
δ(n)↔1; ROC : entire z-plane
• The primary role of Z.T. in engineering are the u(n) ↔ 1/1-z-1 ; |z| > 1
study of system characteristics & the derivation
of computational structures for implementing
discrete systems on computers.
Note:
Practice Questions
1. For a finite length signal, ROC is entire
z-plane, except perhaps for z = 0 and/or
z=∞ 01. Let x(n) = (-1)n u(n) + αn u(-n-n0). Determine the
2. For a right-sided signal ROC is outside constraints on “α” & “n0”,
a circle whose radius is largest pole in given that the ROC of X(z) is 1<|z|<2
magnitude.
02. Find the ROC of the following signals without
3. For a left sided signal, ROC is inside a
finding Z.T.
circle whose radius is smallest pole in
magnitude. (a) x1 (n) = %1, 2, 3- , - 1 /
4. For a two-sided signal, ROC is annulus (b) x2(n) = (1/2)n[u(n) - u(n-10)]
bounded by largest & smallest pole (c) x3(n) = {(1/2)n + (3/4)n}u(n-10)
radius. (d) x4 5n? = (1 3) n - (1 2) n u 5n?
03. Given x(n) = anu(n) – b2nu(–n– 1). What condition

must hold on ‘a’ and ‘b’ for the
z – transform to exist?
(a) |a| < |b2|
(b)|a| > |b2|
(c) |a2| < |b|
(d) |a2| > |b|
04. i) The R.O.C for Z – transform of the signal

x(n) = an u(n) + bn u(n) + cn u(-n - 1)
fora<b<c is
(a)a<z<c (b)b<z<c
(c)z<c (d)z>a
ii) Consider the signal
x[n] = αn u[n] + βn u[–n–1]. Find its
Z-transform X(z). Will X(z) represent a valid
transform for the following cases?
(a) α > β (b) α < β (c) α = β
05. The Z transform of a sequence x(n) is given by
, z >b2l
1 - z -1 1
X (z) =
- b
1 4 z
1 l -2
The value of x(n) at n = 2 is given by

1 1
(a) (b) -
2 2
1 1
(c) (d) -
4 4
06. The Z.T. X(z) of a real & right-sided sequence Im(z)

x(n) has exactly 2 poles and one of them is at
z = ejπ/2 and there are 2 zeros at the origin.
× 0.5 ×
If X(1)=1, which one of the following is TRUE ?
2z2 1 Re(z)
(a) X (z) = ; < z <1 –0.5 0.5
(z - 1) 2 + 2 2
× 0.5 ×
2z2 z 1
(b) X (z) = ; >
z +1
2
2
2z2
(c) X (z) = ; z > 1 (a) h(n) is real for all n
(z - 1) 2 + 2
2z2 (b) h(n) is purely imaginary for all n
(d) X (z) = ; z >1
z +1
2
(c) h(n) is real for only even n
(d) h(n) is purely imaginary for odd n
07. The discrete-time signal
Time shifting, scaling & differentiation
x 5n? * X (z) =
3 n 2n
3
/ 2+n
z , where ↔ denotes a
n=0
10. Find the Z.T. of y (n) = b - 3 l u 6- n - 2@

transform-pair relationship, is orthogonal to the 1 n
signal
(a) y1 5n? * Y1 (z) = / 3n = 0 b 32 l z-n
n

11. Consider the signal
(b) y2 5n? * Y2 (z) = / ^ n
5 - nh z x (n) = )
1; 0 # n # 5
3 - (2n + 1)
n=0
0; elsewhere
(c) y3 5n? * Y3 (z) = / 3n =- 3 2- n z-n Let g(n) = x(n) - x(n-1). Find G(z) with ROC?
(d) y4[n] ↔ Y4(z) = 2z-4 + 3 z-2 + 1

12. Given X ] z g =
z3 − 2z
and x(n) is left-sided, find
z−2
x(n)?
08. Let H1(z) = (1–pz–1)–1, H2(z) = (1–qz–1)–1, 13. Consider a causal and stable LTI system
H(z) = H1(z) + rH2(z). The quantities p, q, r are with rational transfer function H(z), whose
1 1
real numbers. Consider, p = , q = - , r < 1. corresponding impulse response begins at
2 4
If the zero H(z) lies on the unit circle, then 5
n = 0. Further more, H(1) = .
4
The poles of H(z) are Pk = 1 exp c j (2k 1) π m
r = ______ −

2 4
for k = 1, 2, 3, 4. The zeros of H(z) are all at
z = 0. Let g(n) = jnh(n). The value of g(8) equals
09. The pole-zero diagram of a causal and stable ________ . (Give the answer up to three decimal
discrete-time system is shown in the figure. The places).
zero at the origin has multiplicity 4. The impulse
14. Find the Z-transform & ROC of
response of the system is h[n]. If h[0] = 1, we
x (n) = b 4 l u (n) + b 7 l u (- n)
5 n 10 n

can conclude.

Convolution 21. Let y(n) denote the convolution of h(n) &

g(n) where h (n) = b 2 l u (n) & g(n) is a causal
1 n
15. A sequence x(n)↔X(z) = z4 + z2 - 2z + 2 - 3z-4 sequence. If y(0) = 1 & y (1) = 1 , then g(1)
2
is applied as an input to a LTI system with equals
impulse response h(n) = 2δ(n-3). The output at 1
(a) 0 (b)
2
n = 4 is__________ 3
(c) 1 (d)
2
16. Consider a signal 22. An analog signal is sampled at 9 kHz. The

y(n) = x1(n + 3) ∗ x2( - n + 1) sequence so obtained is filtered by an FIR
Where x1(n) = (1/2)nu(n) and filter with T.F. H(z) = 1 – z-6. One of the analog
x2(n) = (1/3)nu(n) frequencies for which the magnitude response
Find Y(z) with ROC? of the filter is zero is
(a) 0.75 kHz (b) 1 kHz
(c) 1.5 kHz (d) 2 kHz
17. Find the response of the system
y(n) = 5/6 y(n-1) - 1/6 y(n-2) + x(n) to the Input
Initial & final value
signal, x(n) = δ(n) - 1/3δ(n-1)
0.5
23. Given X (z) = . It is given that the ROC
18. G(z) = αz-1 + βz-3 represents a digital LPF with 1 - 2z -1
linear phase if and only if of X(z) includes the unit circle. Then x(0) is
(a) α = β (b) α = β1/3 _________
(c) α = -β (d) α = -β1/3
1; n even
24. Apply F.V.T. for x (n) = ) .
0; n odd
19. A system with T.F. H(Z) has I.R. h(n) defined as
Assume the signal is Causal?
h(2) = 1, h(3) = –1 and h(k) = 0
otherwise, consider the following statements
25. A digital filter has an impulse response
S1: H(Z) is a LPF δ (n) + δ (n − 1) + δ (n − 2)
h (n) =
S2: H(Z) is a FIR filter 10
(a) only S2 is true (a) How many finite poles & zeros are there
(b) both are false in its transfer function?
(c) both are true and S2 is the reason for S1 (b) If the excitation is unit step sequence,
(d) both are true but S2 is not a reason for S1 what is final value of output?

20. The following is known about a D.T.L.T.I system
with Input x(n) & output y(n) 26. An input signal x(t) = 2 + 5sin(100πt)u(t) is
1. If x(n) = (-2)n ∀n, then y(n) = 0∀n sampled with a sampling frequency of
2. If x(n) = (1/2)nu(n)∀n, 400 Hz and applied to the system
then y(n) = δ(n) + a(1/4)nu(n) ∀n whose transfer function is represented by
Where ‘a’ is a constant. H (z) = N ; − -1 E
1 1 − z -N
1 z
where N represents the
(a) Find the value of ‘a’
number of samples per cycle. The output y(n)
(b) Find the response y(n) if the Input is
of the system under steady state is ________.
x(n) = 1 ∀n.
(a) 0 (b) 1 (c) 2 (d) 5
27. If h 5n? = Aδ 5n? + b 3 l u (n) is the unit sample

1 n 1 - 2z -1
31. The discrete-time transfer function is
1 - 0.5z -1
response of a LSI system and S[n] is the step (a) non-minimum phase and unstable
response, then the value of A that will make (b) minimum phase and unstable
steady-state step response as zero is (c) minimum phase and stable
(a) – 1 (d) non-minimum phase and stable
(b) 0
32. Consider the system with transfer function,
(c) – 3/2
z -1
(d) – 2/3 H (z) = Then the corresponding stable
1 - 2z -1
impulse response is
Causality & Stability
(a) -0.5 δ(n) - 0.5 (2)n u[-n - 1]
28. A casual LTI system is described by the
(b) 2n-1 u[n - 1]
D.E 2y(n) = αy[n–2] – 2x[n] + βx[n–1]
(c) 0.5 δ(n) + 0.5 (2)n-1 u[n - 1]
The system is stable only if
(d) 0.5 δ(n) + 2n u[-n - 1]
(a) |α|= 2, |β| < 2 (b) |α|> 2, |β| > 2
(c)|α|< 2, any β (d) |β|< 2, any α
33. Suppose x[n] is an absolutely summable
29. Consider an LTI system whose pole-zero pattern discrete-time signal. Its z-transform is a rational
is shown in figure function with two poles and two zeros. The
poles are at z = ! 2j . Which one of the following
Im {z} statements is TRUE for the signal x[n]?
(a) It is a finite duration signal.
X X 0 X (b) It is a causal signal.
−3 −0.5 1 2 Re {z} (c) It is a non-causal signal.
(d) It is a periodic signal.

(a) Find the ROC of system function, if it is
34. Assertion (A): A linear time-invariant discrete-
known to be stable?
time system having the system function
(b) Is it possible for the given pole-zero plot to
H]zg =
z
is a stable system.
be a causal & stable system? z +1
2
(c) How many possible ROC’s are there?
Reason (R): The pole of H(z) is in the left-half

plane for a stable system.
30. The impulse response h(n) of a LTI system is real.
The transfer function H(z) of the system has only
one pole and it is at z = 4/3. The zeros of H(z)
are non-real and located at |z| = 3/4. The
system is
(a) stable & causal
(b) unstable & anticausal
(c) unstable & causal
(d) stable & anticausal
35. Consider the following statements regarding a Realization Structures

linear discrete - time system
z2 + 1 P0 + P1 z -1 + P3 z -3
H(z) = 39. A DF having T.F. H (z) = using
(z + 0.5) (z - 0.5) 1 + d 3 z -3
DFI and DFII realizations of IIR the number
1. The system is stable. of delay units required in DFI and DFII are,
2. The initial value h(0) of the impulse respectively …….
response is - 4. (a) 6 & 6 (b) 6 & 3
3. The steady-state output is zero for a (c) 3 & 3 (d) 3 & 2
sinusoidal discrete time input of frequency
equal to one-fourth the sampling 40. 2 systems H1(z) & H2(z) are connected in
cascaded as shown below. The overall output
frequency.
y(n) is the same as the input x(n) with a one
Which of these statements are correct?
unit delay. The transfer function of the second
(a) 1, 2 and 3 (b) 1 and 2 system H2(z) is
(c) 1 and 3 (d) 2 and 3
x(n)
y(n)
1 − 0.4 z −1
36. Assertion (A): The stability of the system is H1 ( z ) = H2(z)
1 − 0.6 z −1
assured if the Region of Convergence (ROC)
includes the unit circle in the Z-plane.
Reason (R): For a causal stable system all the
poles should be outside the unit circle in the 41. A direct form implementation of an LTI system
Z-plane. 1
with H (z) = is shown in fig. the
1 - 0.7z -1 + 0.13z -2
37. Statement (I): The system function values of a0, a1 & a2 are respectively _________
H]zg =
z3 − 2z2 + z
1 1 is not causal. Input Output
z2 + z +
4 8
Statement (II): If the numerator of H(z) is of a0
z−1
lower order than the denominator, the system a1
may be causal. z−1
a2
38. Statement (I):
(a) 1.0, 0.7 and - 0.13
Z-transform approach is used to analyze the
(b) -0.13, 0.7 and 1.0
discrete time systems and is also called as pulse
(c) 1.0, -0.7 and 0.13
transfer function approach.
(d) 0.13, -0.7 and 1.0
Statement (II):
The sampled signal is assumed to be a train of
impulses whose strengths, or areas, are equal
to the continuous time signal at the sampling
instants.
42. In the IIR filter shown below, a is a variable gain. 45. The nature of the above filter is
For which of the following cases, the system will (a) Band pass (b) All pass
transit from stable to unstable condition? (c) Low pass (d) High pass
x(n)
Σ 46. For the causal filter structure shown in figure,
y(n)
the range of k for system stability is ____
a x(n) + y(n)
+ Z-1+ Z-2
z -1
+
Fig. k

(a) :- 1, 2 D (b) : 2 , 1D
(a) 0.1 < a < 0.5 1 1

(b) 0.5 < a < 1.5
(c) 6- 1, 1@ (d) : 2 , 2D
(c) 1.5 < a < 2.5 1

(d) 2 < a < ∞
47. Consider an All-pass system shown in figure.

43. Consider the causal digital filter structure shown
H (z) = z − 0.54-1
-1
in fig. For what values of k the system is stable? 1 − 0.54z

d
x[n] y[n] x(n) y(n)
+ +
∑ z −1 c
b
z–1
Find b, c & d such that SFG is a realisation
of H(z)
− k/3 − k/4
Common Data for Questions 44 & 45

A digital filter realization is shown:
-1
X 3 z -1 Y
-1
Y (z)
44. The transfer function H (z) = of this filter is
X (z)
0.2 + z -1 3 + z -1
(a) (b)
1 + 2z -1 1 + 3z -1
2z + 1 2z - 1
(c) (d)
z+2 z+2
Key for Practice Questions 8. Digital Filter Design
01. Ans: α = ±2, n0 is any value

In the design of frequency-selective filters, the
desired filter characteristic are specified in the
02. (a). Ans: 0<|z|<∞
frequency domain in terms of the desired magnitude
(b). Ans: |z|>0
and phase response of the filter. In the filter design
(c). Ans: |z| > 3/4
process, we determine the coefficients of a causal
1 FIR or IIR filter that closely approximates the desired
(d). Ans: < z <3
2
frequency response specifications.
03. Ans: (a)
In practice FIR filters are employed where there is a
05. Ans: (c) 06. Ans: (d) requirement for linear phase characteristic with the
pass band of the filter.
3z z <1
10. Ans: Y (z) = 1 -1 ,
1+ z 3
3 An IIR filter has lower side lobes in the stop band than
an FIR filter having same number of parameters.
11. Ans: G(z) = 1-z-6, |z|>0
Design of IIR Filters from Analog Filters:
12. Ans: x(n) = δ(n+2) + 2δ(n+1) - 2(2)n u(-n-1) To convert analog filter to digital filter the following
properties are desirable.
15. Ans: 0 1. The jΩ axis in the s-plane should map into the
unit circle in the z-plane. Thus there will be a
z -2
16. Ans: Y (z) = direct relationship between the two frequency
b1 - 1 z -1 l b1 - 1 z l
2 3 variables in the two domains
2. The left-half plane (LHP) of the s-plane should
9 -1 map into the inside of the unit circle in the
20. (a). Ans: a = - (b). Ans: y (n) =
8 4
z-plane. Thus a stable analog filter will be
21. Ans: (a) converted to a stable digital filter.

1 Converting an analog filter to a digital filter
23. Ans: 0 24. Ans:
2
Low-pass Analog s →z
29. (a). Ans: 0.5 <|z|< 2 analog transformation Analog mapping Digital
prototype HP(s) Filter H(s) filter H(z)
(b). Ans: No ΩC = 1 rad/s
1
(c). |z|< 0.5,|z|>3, < z < 2, 2 < z < 3
2
z -1 (1 - 0.6z -1) Indirect conversion of an analog filter to a digital

40. Ans: 41. Ans: (a)
(1 - 0.4z -1) filter.
Low-pass s→z D2D
43. Ans: |k|<3 44. Ans: (c) mapping Low-pass
analog transformation Digital
digital
prototype HP(s) filter H(z)
prototype HP(z)
ΩC = 1 rad/s
ωC = 1 rad/s
45. Ans: (b)
Steps in the design of IIR filters:

1. Convert digital filter specifications to analog jΩ
filter specifications. Unit circle
2. Find the order (n) & cut-off frequency (ΩC) of z plane
the prototype L.P.F. s plane
3. Find the normalized analog transfer function
using frequency transformations. σ
1
4. Convert analog transfer function to digital 2
transfer function.
5. Draw the filter structure.
Therefore this technique is limited to the design of

low pass and band pass filter only.
8.1 IIR filter design by approximation of Derivatives
One of the simplest methods for converting an
8.2 IIR filter Design by Impulse Invariance
analog filter into a digital filter is to approximate the
In this technique impulse response of digital system
differential equation by an equivalent difference
is sampled version of impulse response of analog
equation.
system so the frequency response of digital system
For the derivative dy(t)/dt at time t = nT, we substitute
the backward difference is aliased version of frequency response of analog
system.
y (nT) - y (nT - T)
h 5nT? $ H (z)
I.L.T t = nT
T H (s) h (t)
ana log T.F digital T.F
dy (t) y (nT) - y (nT - T)
= T
dt t = nT
y (n) - y (n - 1) Example:
= T 1
H (s) = s + a
Where T represents the sampling interval and . I.L.T

h (t) = e -at u (t)
y(n) ≡ y(nT). The analog differentiator with
output dy(t)/dt has the system function . t = nT
h 5nT? = e -anT u (nT)
Z.T 1
H(s) = s, while the digital system that produces H (z) =
1 - e -aT z -1
y (nT) - y (nT - T)
the output has the system function
T 1 1
(1 - z )
-1 I .T →
.I
H (z) = T . s+a 1 - e - aT z -1
Therefore s =
(1 - z -1)
……. (1)
1
→(- 1)m-1 . d m-1 . 1
T (s + a )m (m - 1)! d a m-1 1 - e -aT z -1
For kth derivative s k = b 1 z l …… (2)
- -1 k
→ 1 - 2e1 -(ecos (bT

cos bT )z
T s+a - aT -1
1 (s + a )2 + b 2 -aT
)z + e -1 - 2 aT - 2
z
From equation (1) z = - . If we substitute s = jΩ,
1 sT
we can observe that as Ω varies from –∞ to ∞, the
b
→ 1 - 2e e (sin bT )z
- aT -1
corresponding locus of points in the z-plane is a (s + a ) 2

+b 2 -aT
(cos bT )z -1 + e -2aT z -2
circle of radius 1/2 and center at (1/2, 0)
Let us consider the mapping of points from the 8.3 IIR Filter Design by Bilinear Transformation
s-plane to the z-plane implied by the relation
The bilinear transformation is a conformal mapping
z = esT
that transforms the jΩ-axis into the unit circle in
If we substitute s = σ + jΩ and express the complex the z-plane only once, thus avoiding aliasing of
variable z in polar form as z = rejω frequency components. Furthermore, all points
rejω = eσT ejΩT in the LHP of s-plane are mapped inside the unit
clearly, we must have circle in the z-plane and all points in the RHP of s are
r = eσT mapped into corresponding points outside the unit
ω = ΩT circle in the z-plane.
s= Tc
2 1 - z -1 m
Consequently, σ < 0 implies that 0 < r < 1 and …………… (1)
1 + z -1
σ > 0 implies that r > 1. When σ = 0, we have
r = 1. Therefore, the LHP in s is mapped inside the From equation (1) we note that if
unit circle in z and the RHP in s is mapped outside r < 1, then σ < 0,
the unit circle in z. r > 1, then σ > 0,
However, the mapping of the jΩ-axis into the unit When r = 1, then σ = 0,

circle is not one-to-one. Since ω is unique over the 2 ω
X = T tan ………………..(2)
range (–π, π) the mapping ω = ΩT implies that the 2
interval –π /T ≤ Ω ≤ π/T maps into the corresponding Which show that
values of –π ≤ ω ≤ π. Even the frequency interval Ω=0 ⇒ω=0
π/T ≤ Ω ≤ 3π/T also maps into the interval –π ≤ ω ≤ π.
Ω→∞ ⇒ω→π …….. (3)
Thus the mapping from analog frequency Ω to the Ω → –∞ ⇒ ω → –π
frequency variable ω in the digital domain is many
to one, which reflect the affect of aliasing due to The non-linear relationship between ω and Ω
sampling. in eq (2) is known as frequency warping. The
jΩ bilinear transformation converts H(jΩ) to H(ejω) by
compressing the continuous time frequency axis
according to eq (3).
z - plane s-plane
π  ΩT 
z=e sT ω 2 tan −1  
T  2 
π
σ
0 Ω
π
–
Unit T -π
circle
3π
–
T 1. We note that for ω less than about 0.3π, the
relation between Ω and ω is approximately
Due to the presence of aliasing, the impulse linear (recall that tanφ ≈ φ for small φ). Thus, any
invariance method is appropriate for the design of shape of magnitude response in this range is
low-pass and band-pass filters only. preserved.
2. Frequency warping does not distort Example:

“flat” magnitude responses because any Design a single-pole lowpass digital filter with
“rearrangement” of equal values will preserve a 3-dB bandwidth of 0.2π, using the bilinear
transformation applied to the analog filter
the flatness of the shape. Thus, equiripple
X
bands are mapped into equiripple bands. H (s) =
s + Xc
3. Frequency warping distorts “non flat” Where Ωc is the 3-dB bandwidth of the analog
magnitude response. Thus, a continuous- filter.
time differentiator cannot be converted Sol: The digital filter is specified to have its –3dB gain
to a discrete-time one using the bilinear at ωc = 0.2π. In the frequency domain of the
transformation. analog filter ωc = 0.2π corresponds to
2 0.65
X c = T tan 0.1r = T
To avoid warping we are using pre-warping that is
Thus the analog filter has the system function
converting digital filter specification to analog filter
0.65/T
H (s) =
specification. The bilinear transformation maps the s + 0.65/T
point s = ∞ into the point z = –1. This represents our filter design in the analog
We conclude form the previous discussion that the domain. By applying bilinear transformation
bilinear transformation is most appropriate for filters 0.245 (1 + z -1)

we will get H (z) = .
1 - 0.509z -1
with piecewise-constant magnitude responses,

such as lowpass, highpass and bandpass filter.
Types of Analog Filters:
1. Butterworth Filter (Maximally flat)
Example:
2. chebyshev
Convert the analog filter with system function 3. Elliptical (cauer)
s + 0.1
H a (s) =
(s + 0.1) 2 + 16
8.4 Butterworth Filter
Into a digital IIR filter by means of the bilinear
Lowpass Butterworth filters are all-pole filters
transformation. The digital filter is to have a
characterized by the magnitude-squared
resonant frequency of ωr = π/2
frequency response
Sol: 1
1 + ]X/X cg2n
H (X) 2 = ………… (1)
Analog filter frequency Ω = 4. This frequency
is to be mapped into ω = π/2 by selecting the
Where n is the order of filter Ωc is its –3 dB frequency.
value of T.
Eq (1) can be adjusted as
Tan a 22 k & T 4

= 2 r 2
4 T=
1
H (s) H (- s) = …….. (2)
1 + (- s2 /X2c) n
s = 4 ; + -1 E
1 − z -1

1 z
The poles of H(s) H(–s) occur on a circle of radius Ωc
The resulting digital filter has the system function at equally spaced points.
From eq (2) poles can be obtained by using
0.128 + 0.006z -1 - 0.122z -1
H (z) = S k = X c e jr/2 e j (2k + 1) r/2n , k = 0, 1, ….., n –1
1 + 0.0006z -1 + 0.975z -1
From eq (1) we can verify Pole positions for Butterworth filters

1
1. At X = X c , H (X) = for all n
2
2
π π
+
2. From eq (1) it is clear that |H(Ω)|2 decreases 2 8
monotonically as Ω increases.
π π
Poles of − − Poles of
2 8
|H(Ω)|2 H(s) H(–s)
Ideal characteristic
N=4
1.0
n =18 π π
+
n =14 2 10
0.5 n =10
Poles of
n =6 Poles of
H(–s)
n =2 H(s)
N=5
0 Ωc Ω
Example:
3. From the above figure as the order of the filter Determine the order of a lowpass Butterworth
increases the Butterworth filter characteristic is
filter that has a –3 dB bandwidth of 500 Hz and
more close to ideal characteristics.
an attenuation of 40 dB at 1000 Hz.
The order & cut off frequency of the Butterworth
Sol: The critical frequencies are the –3 dB
prototype filter is obtained using
frequency Ωc and the stop-band frequency
RS V
SS 12 − 1 WWW Ωs which are
1 Sδ WW
log SS 1s Ωc = 1000π
2 SS − 1 WWW 1 log <10 -1
F
0.1d dB s
S δp
2
W 2 10 .1d dB - 1
0 p
Ωs = 2000π
n= T X= …… (3)
log c Ω s m log c Ω s m
Ω Ω
log10 (10 4 - 1)
p p
n=
2 log10 2
Xp
Xc = ……… (4) = 6.64 , 7
1
d 2 - 1n
1 2n
dp
8.5 Chebyshev Filter
Butterworth polynomials There are two types of chebyshev filters.
Type-I chebyshev filter are all-pole filters that
Order Factors exhibit equiripple behavior in the pass-band and
1 s+1
a monotonic characteristic in the stop-band. On
2 s2 + 2 s + 1
the other hand, the family of type-II chebyshev
3 (s2 + s + 1) (s + 1)
filters contains both poles and zeros and exhibits
4 (s2 + 0.76536s + 1) (s2 + 1.848s + 1)
a monotonic behavior in the pass-band and an
5 (s + 1) (s2 + 0.6180s + 1) (s2 + 1.6180s + 1)
equiripple behavior in the stop-band. The zeros of
6 (s2 + 0.5176s + 1) (s2 + 2 s + 1)
this class of filters lie on the imaginary axis in the
(s2 + 1.9318s + 1)
s-plane.
Few observations can be made from the above

|H(Ω)|2 |H(Ω)|2
function:
1 i. For |Ω| ≤ 1, |H (jΩ)|2 varies between 1 and
1
1 1 1
1+ ∈2 1+ ∈2 . This is the pass-band description of
1 + f2
1
|H(jΩ)|. i.e., 1 # H (jΩ) # for Ω # 1
1 + ε2
1
and, ripple in pass-band = 1 -
1 + f2
Ωp Ω Ωp Ω ii. At Ω = 1, C2n (1) = 1 always. This can be easily
n odd type I n even type I
verified for the polynomials for Ω = 1 eq (2) can
be as follows:
The chebyshev approximation is implemented
with the help of chebyshev polynomials. They are 1 1
H (jX) = or H (jX) =
2
defined as follows. 1 + f2 1 + f2
C n (X) = cos (n cos -1 (X)) X # 1 Thus at cutoff frequency Ω = 1, magnitude is
= cosh (n cosh -1 (X)) X > 1 1
=
1 + f2
For n = 0 ⇒ C0(Ω) = cos(0) = 1
iii. In the stopband |Ω| > 1, and f2 C2n (X >> 1) .
For n = 1 ⇒ C1(Ω) = cos(1cos–1 (Ω)) = Ω
Hence eq (2) can be written as,
The higher order chebyshev polynomials are
obtained by following recursive formula. H (jX) 2 = 2 12
f C n (X)
Cn(Ω) = 2Ω Cn–1 (Ω) – Cn-2 (Ω)
H (jX) , 1
or
fC n (X)
n Chebyshev polynomials The above equation can be written in decibels
0 1 also i.e.,
H (jX) in dB = 20 log10 ; E
1 Ω 1
fC n (X)
2 2Ω2 – 1
3 4Ω3 – 3Ω = –20 log10 [εCn (Ω)]
4 8Ω – 8Ω + 1
4 2
As Ω is large Cn(Ω) is mainly represented by its
5 16Ω5 – 20Ω3 + 5Ω
first term. Hence Cn ≅ 2n-1 Ωn for large Ω
6 32Ω6 – 48Ω4 + 18Ω2 – 1 Hence the above equation can be written as
|H(j(Ω)| in dB = –20 log10 [ε.2n-1 Ωn]
Some of the properties of these polynomials are = –20 log ε – 20 log 2n-1 – 20log Ωn
1. |Cn(Ω)| ≤ 1 for all |Ω| ≤ 1 = –20 log ε – 20(n – 1) log 2 – 20n log Ω
2. Cn(1) = 1 for all n = –20 log ε – 6(n –1) – 20n log Ω (norm)
f = 610 0.1d dB - 1@
p 1 2
3. All the roots of the polynomials Cn(Ω) occurs in
the interval –1≤ Ω ≤ 1
The following points can be noted about transfer
The squared magnitude function of the chebyshev function of the chebyshev filter:
i. The transfer function of the chebyshev filter is
filter is given as,
an all pole function like Butterworth filter.
H (jX) 2 = 1 ……….. (2)
1 + f2 C2n (X) ii. The numerator is constant and there are no
finite zeros.
iii. Poles of the transfer function lie on an ellipse |H(Ω)|2

[for Butterworth filter they lie on the circle]. The
1
major axis of the ellipse lies along imaginary
axis of the s – plane and minor axis lies along
real axis of the s-plane.
iv. The narrower the ellipse, the closer will be the δ 22
poles to the imaginary axis and hence each
Ωp Ωs Ω
individual pole will have stronger impact. This
n even
means that ripples will be more pronounced.
v. The ripple magnitude will have a strong effect
Frequency Transformations for Analog Filters:
on the locations of the poles of the transfer
(Prototype low-pass filter)
function.
Prototype Required
The transfer function of the chebyshev filter is
LPF s s/Ωc
given as
K HPF s Ωc/s
H (s) = n
s + b n - 1 s n - 1 + f + b1 s + b 0 s s2 + X20
The constant K in the numerator in the above BPF Xc s
equation is adjusted to provide a loss of 0 dB at s Xc s

BSF s2 + X20
the pass band minima.
Z] b for odd "n" X20 = X P X P
]] 0 1 2
K = [] b 0 for even "n" Xc = XP - XP

]]
2 1
1 +! 2
\
Poles location for a chebyshev filter
r2
r1
Type II chebyshev filter

|H(Ω)|2
δ 22
Ωp Ωs Ω
n odd
Frequency Transformations for Digital Filters

(Prototype LPF has band-edge frequency ωP)
Type of transformation Transformation Parameters
ω'p = band edge frequency of new

z -1 - a
sin 7^ω p − ωl p h /2A
Low-pass z "
-1
1 - az -1 filter a =
sin 6^ω p + ωl ph /2@
ω'p = band edge frequency of new

filter a = − cos 6^ω p + ω p h /2@
High-pass l
z -1 + a
z -1 "
1 + az -1 cos 7^ω p − ωl p h /2A
ω , = lower band edge frequency

ωu = upper band edge frequency
a1 = 2αK / (K + 1)
Band-pass z - a1 z + a2
-2 -1 a2 = (K – 1) / (K + 1)
z -1 " -
a2 z -2 - a1 z -1 + 1 cos 6]ω u + ω ,g /2@
cos 6^ω u − ω ,h /2@
α=
^ω u − ω ,h ωp
K = cot tan
2 2
ω , = lower band edge frequency

ωu = upper band edge frequency
a1 = 2α / (K + 1)
Band-stop z -2 - a1 z -1 + a2
z "
-1
a2 = (1 – K) / (1 + K)
a2 z -2 - a1 z -1 + 1
cos 6]ω u + ω ,g /2@
cos 6^ω u − ω ,h /2@
α=
^ω u − ω ,h ωp
K = tan tan
2 2
8.6 FIR Filter Design

Type 1
n F
An FIR filter of length N with input x(n) and output –1 –0.5 0.5 1
y(n) is described by the difference equation Center of symmetry Even symmetry about
F = 0 and F = 0.5
y(n) = b0x(n) + b1x(n–1) +…+ bN–1x(n–N+1)

N-1
= / b k x (n - k ) Type 2
k=0
–1
n –0.5 1F
0.5
Center of symmetry
Where {bk} is the set of filter coefficients. Alternatively, Odd symmetry about F = 0.5
we can express the output sequence as the

convolution of the unit sample response h(n) of the
system with the input signal. Thus we have Type 3
N-1
y (n) = / h (k) x (n - k) ………. (1)

n –1
–0.5
0.5
1
F
k=0
Odd symmetry about F = 0.5

Center of symmetry
Where the lower and upper limits on the convolution
sum reflect the causality and finite-duration
characteristics of the filter.
N-1 Type 4
H (z) = / h (k) z-k ………….. (2)
k=0 n –1 –0.5 0.5 F
1
Odd symmetry about F = 0.5

Center of symmetry
The roots of this polynomial constitute the zero of
the filter.
An FIR filter has linear phase if its unit sample

response satisfies the condition
h(n) = ± h(N–1–n), n = 0, 1, …., N –1
Type Symmetry Length

1 Even Odd
2 Even Even
3 Odd Odd
4 Odd Even
Window Functions
Name Time-domain sequence

of window h(n), 0 ≤ n ≤ N –1
Bartlett -
2 n- N 1
(triangular) 2
1-
N-1
Blackman 2rn 4rn

0.42 - 0.5 cos 0.08 cos -
N-1 + N 1
Hamming 2rn
0.54 - 0.46 cos
N-1
Hanning 1 b1 - cos 2rn l

2 N-1
I 0 <a c N 1 m - cn - N 1 m F
- 2 - 2
Kaiser 2 2
I 0 ;a c N 1 mE
-
2
Comparison of window characteristics
Type of window Approximate Peak Transition Minimum S.B

transition sidelobe Width Attenuation (dB)
width of level (dB)
main lobe
Rectangular 4π/N – 13 0.9π –21
N
Bartlett 8π/N – 25 3π –25
N
Hanning 8π/N – 31 3.1π –44
N
Hamming 8π/N – 41 3.3π –53
N
Blackman 12π/N – 57 5.5π –74
N
The following example illustrates low-pass filter Location of zeros of a linear phase FIR filter:
design using rectangular (a), Hamming (b), 1
Blackman (c) and Kaiser window with α = 4 for a z1*
1
length of 61 samples.
z*3
z3 z1
1
z2 z2
z*3
z1*
1
z3
Unit 1
circle z1
(a)
If “z0” is a zero of H(z) in eq (2)
Then “ z 0-1 ” is a zero of H(z)
If z1 = –1 then z1-1 = 1
If z2 is a real zero then z2-1 = 1/z2
If z3 is a complex zero 6 z3 = 1@ then z3-1 = z*3
If z4 is a complex zero [|z4| ≠ 1] then other zeros are
z 4-1, z*4 ^z*4h
-1
8.7 FIR Filter Design using windowing method

(b) 1. Choose the desired frequency response
Hd(ejω)
r
2. Obtain hd 5n? =
1 #
Hd (e j~n) e j~n dω → non
2π
-r
causal impulse response.

3. h 5n? = hd 5n? w 5n? is the causal Impulse
response of FIR filter.
/ h5n? z-n
N-1
4. Find H (z) =
(c) n=0
5. Draw filter structure.
Design of FIR Differentiators:

Differentiators are used in many analog and digital
systems to take the derivative of a signal. An ideal
differentiator has a frequency response that is
linearly proportional to frequency. Similarly, an ideal
digital differentiator is defined as one that has the
frequency response
(d) Hd(ω) = jω –π ≤ ω ≤ π
The unit sample response corresponding to Hd(ω) is Putting the above result in eq (1)
r
1 # Hd (ω) e j~n dω 1
N-1
1 - z -N
hd (n) =
2π H (z) = N / H (k) .
-r k=0 1 - e j2rk/N z -1
r
1 1 - z -N /
N-1
H (k)
= 2π # jω e j~n dω = N k = 0 1 - e j2rk/N z -1
-r
cos rn -
= n , 3 < n < 3, n ! 0
Design of Hilbert Transformers

An ideal Hilbert transformer is an all-pass filter that
imparts a 90o phase shift on the signal at its input.
Hence the frequency response of the ideal Hilbert
transformer is specified as
Hd (ω) = ) − j 0 < ω # π
j −π < ω < 0
Hilbert transformers are frequently used in
communication systems and signal processing, as,
for example, in the generation of single-side band
modulated signals, radar signal processing, and
speech signal processing.
The unit sample response of an ideal Hilbert
transformer is
r
1 #
hd (n) = H d (ω) e j~n dω
2π
-r
0 r
1 # #
= 2π j e j~n dω − j e j~n dω
-r 0
2 sin (rn/2)
= *r
2
n , n!0
0 , n=0
Frequency Sampling Method

The transfer function of FIR filter is
N-1
(
H z) = / h (n) z-n
n=0
In the method we specify the desired frequency
response at a set of equally spaced frequencies
and solve for that unit impulse response from these
equally spaced frequency specifications.
1
H (z) = / ) N / H (k) e 2 3 z -n ….. (1)
N-1 N-1
j rkn/N
k=0 k=0
H (z) = N / H (k) ) / e j2rkn/N z n 3

N-1
1
N-1
-
k=0 n=0
/ ^e j2rk/N z-1hn = 1 --^e j2rk/Nz -1h

N-1 j2rk/N -1 N
n=0 1 e z
Normalized Chebyshev Polynomials

(a) Ripple = 0.5 dB i.e ε = 0.349
N b0 b1 b2 b3 b4 b5 b6 b7 b8 b9
1 2.8627752
2 1.5162026 1.4265245
3 0.7156938 1.5348954 1.2529130
4 0.3790506 1.0254553 1.7168662 1.1973856
5 0.1789234 0.7525181 1.3095747 1.9373675 1.1724909
6 0.0947626 0.4323669 1.1718613 1.5897635 2.1718446 1.1591761
7 0.0447309 0.2820722 0.7556511 1.6479029 1.8694079 2.4126510 1.1512176
8 0.0236907 0.1525444 0.5735604 1.1485894 2.1840154 2.1492173 2.6567498 1.1460801
9 0.0111827 0.0941198 0.3408193 0.9836199 1.6113880 2.7814990 2.4293297 2.9027337 1.425705
10 0.0059227 0.2372688 0.2372688 0.6269689 1.5274307 2.1442372 3.4409268 2.7097415 3.1498757 1.1400664
(b) Ripple = 1 dB i.e ε = 0.508
N b0 b1 b2 b3 b4 b5 b6 b7 b8 b9
1 1.9652267
2 1.1025103 1.0977343
3 0.4913067 1.2384092 0.9883412
4 0.2756276 0.7426194 1.4539248 0.9528114
5 0.1228267 0.5805342 0.9743961 1.6888160 0.9368201
6 0.0689069 0.3070808 0.9393461 1.2021409 1.9308256 0.9282510
7 0.0307066 0.2136715 0.5486192 1.3575440 1.4287930 2.1760778 0.9231228
8 0.0172267 0.1073447 0.4478257 0.8468243 1.8369024 1.6551557 2.4230264 0.9198113
9 0.0067767 0.0706048 0.2441864 0.7863109 1.2016071 2.3781188 1.8814798 2.6709468 0.9175474
10 0.0043067 0.0344971 0.1824512 0.4553892 1.2444914 1.6129856 2.9815094 2.1078524 2.9194657 0.9159320
(c) Ripple = 2 dB i.e ε = 0.764
N b0 b1 b2 b3 b4 b5 b6 b7 b8 b9
1 1.3075603
2 0.6367681 0.8038164
3 0.3268901 1.0221903 0.7378216
4 0.2057651 0.5167981 1.2564819 0.7162150
5 0.0817225 0.4593491 0.6934770 1.4995433 0.7064606
6 0.0514413 0.2102706 0.7714618 0.8670149 1.7458587 0.7012257
7 0.0204228 0.1660920 0.3825056 1.1444390 1.0392203 1.9935272 0.6978929
8 0.0128603 0.0729373 0.3587043 0.5982214 1.5795807 1.2117121 2.2422529 0.6960646
9 0.0051076 0.0543756 0.1684473 0.6444677 0.8568648 2.0767479 1.3837474 2.4912897 0.6946793
10 0.0032151 0.0233347 0.1440057 0.3177560 1.0389104 1.1585287 2.6362507 1.5557424 2.7406032 0.6936904
(d) Ripple = 3dB i.e ε = 0.997
N b0 b1 b2 b3 b4 b5 b6 b7 b8 b9
1 1.0023773
2 0.7079478 0.6448996
3 0.2505943 0.9283480 0.5972404
4 0.1769869 0.4047679 1.1691176 0.5815799
5 0.626391 0.4079421 0.5488626 1.4149847 0.5744296
6 0.0442497 0.1634299 0.6990977 0.6906098 1.6628481 0.5706979
7 0.0156621 0.1461530 0.3000167 1.0518448 0.8314411 1.6628481 0.5684201
8 0.0110617 0.0564813 0.3207646 0.4718990 1.4666990 0.9719473 2.1607148 0.5669476
9 0.0039154 0.0475900 0.1313851 0.5834984 0.6789075 1.9438443 1.1122863 2.4101346 0.5659234
10 0.0027654 0.0180313 0.1277560 0.2492043 0.9499208 0.9210659 2.4834205 1.2526467 2.6597378 0.5652218
04. We are given the analog low-pass filter

Practice Questions 1
H (s) = s + 1 whose cutoff frequency is known
to be 1 rad/s. It is required to use this filter as the
1 basis for designing a digital filter by the impulse
01. Consider the analog filter H (s) =
s+2
invariant transformation. The digital filter is to
(a) Convert H(s) to a digital filter H(z) using
have a cutoff frequency of 50 Hz and operate
impulse invariance. Assume that the at a sampling frequency of 200 Hz.
sampling frequency is 2 Hz. What is the transfer function H(z) of the digital
(b) Will the impulse response h[n] match the filter if no gain matching is used?
impulse response h(t) of the analog filter at 05. Consider the low-pass analog filter
3
the sampling instants? Should it? Explain. H (s) = 2 .
s + 3s + 3
(c) Will the step response s[n] match the step (a) Use the bilinear transformation to convert
response s(t) of the analog filter at the this analog filter H(s) to a digital filter H(z) at
sampling instants? Should it? Explain. a sampling rate of 2 Hz.
(b) Use H(s) and the bilinear transformation to
design a digital lowpass filter H(z) whose
1
02. Consider the analog filter H (s) = . gain at f = 20 kHz matches the gain of H(s)
s+2
(a) Convert H(s) to a digital filter H(z) using step at Ω = 3 rad/s. The sampling frequency is
invariance at a sampling frequency of 2Hz. 80 kHz.
(b) Will the impulse response h[n] match the s

06. Consider the analog filter H (s) = .
impulse response h(t) of the analog filter at s2 + s + 1
(a) What type of filter does H(s) describe?
the sampling instants? Should it? Explain.
(b) Use H(s) and the bilinear transformation
(c) Will the step response s[n] match the step
to design a digital filter H(z) operating at
response s(t) of the analog filter at the 1 kHz such that its gain at f0 = 250 Hz
sampling instants? Should it? Explain. matches the gain of H(s) at ωa = 1 rad/s.
What type of filter does H(z) describe?
1
03. Consider the analog filter H (s) = .
s+2 07. A digital low-pass filter is required to meet the
(a) Convert H(s) to a digital filter H(z) using following specifications:
ramp invariance at a sampling frequency Passband ripple: ≤ 1 dB
of 2Hz. Passband edge: 4 kHz
Stopband edge: ≥ 40 dB
(b) Will the impulse response h[n] match the
Stopband edge: 6 kHz
impulse response h(t) of the analog filter at
Sample rate: 24 kHz
the sampling instants? Should it? Explain. The filter is to be designed by performing a
(c) Will the step response s[n] match the step bilinear transformation on an analog system
response s(t) of the analog filter at the function. Determine what order Butterworth,
Chebyshev analog design must be used
sampling instants? Should it? Explain.
to meet the specifications in the digital
implementation.
08. An IIR digital lowpass filter is required to meet 13. Find H(z) and H(F) for each sequence and
the following specifications: establish the type of FIR filter it describes by
Passband ripple checking values of H(F) at F = 0 and F = 0.5
(or peak-to-peak ripple): ≤ 0.5 dB
(a) h 5n? = %1, 0, 1 /
0
Passband edge: 1.2 kHz
(b) h 5n? = %1, 2, 2, 1 /

Stopband attenuation: ≥ 40 dB 0
Stopband edge: 2.0 kHz

(c) h 5n? = %1, 0, - 1 /
0
Sample rate: 8.0 kHz
(d) h 5n? = %1, - 2, 2, - 1 /

Determine the required filter order for

0
(a) A digital Butterworth filter

(b) A digital chebyshev filter
14. The first few values of the impulse response

09. Determine the system function H(z) of the sequence of a linear-phase filter are
lowest-order chebyshev digital filter than meets h[n] = {2, –3, 4, 1, ….. }. Determine the complete
the following specifications: sequence (assuming the smallest length for
(a) 1-dB ripple in the pass-band 0 ≤|ω|≤ 0.3π h[n] if the sequence is to be:
(b) At least 60 dB attenuation in the (a) type 1
stop-band 0.35π ≤ |ω| ≤ π. Use the bilinear (b) type 2
transformation.
(c) type 3
10. Determine the system function H(z) of the (d) type 4

lowest-order chebyshev digital filter that meets
the following specification. 15. Partial details of various filters are listed. Zero
1 - locations are in keeping with linear-phase and
(a) dB ripple in the pass-band
2
0 ≤ |ω| ≤ 0.24π real coefficients. Assuming the smallest length,
(b) At least 50 dB attenuation in the stop-band identify the sequence type and find the transfer
0.35π ≤ |ω| ≤ π. function of each filter.
Use the bilinear transformation. (a) zero location: z = 0.5ej0.25π
(b) zero location: z = ej0.25π
11. A linear phase FIR filter has one of the zero at (c) zero location: z = 1, z = ej0.25π
1 (d) zero locations: z = 0.5, z = –1; odd symmetry
z = e jr/3 . Find the remaining zeros?
2
(e) zero locations: z = 0.5, z = 1, z = –1; even

symmetry
12. Transfer function of a IVth order linear phase
F.I.R filter is given by H(z) = (1 + 2z–1 + 3z–2)G(z)
16. Design the symmetric FIR LPF whose desired
then G(z) is _____
frequency response is given as
(a) 3 + 2z–1 + z–2 π
Hd (e j~) = *
e -j3~ for ω #
4
1 0 else where
(b) 1 + z -1 + z -2
2
1 use Hamming window.
(c) 2z -1 + z -2
3+
(d) 1 + 2z–1 + 3z–2
Properties of DFT
9. DFT & FFT
01. Circular shift:
9.1 Introduction to DFT x 6n - n 0@N * e X 5k?
-j2r
N kn0
• The Fourier series describes periodic signals by (or) W Nkn X (K)

0
x :n - 2 D * (- 1) k X 5k?
N
discrete spectra, whereas the DTFT describes
discrete signals by periodic spectra. As a result,

02. Modulation:
signals that are both discrete and periodic in * X 6k - k 0@
j2rk 0 n
x (n) e N
one domain are also periodic & discrete in the ]- 1gn x (n) * X :k - N
2
D
other. This is the basis for the formulation of the
03. Circular Convolution:
DFT.
x(n) Ⓝ h(n) ↔ X(k) H(k)
• Sampled version of D.T.F.T. spectrum is D.F.T.
• The N point DFT of a signal x(n) is 04. Central ordinates:
(a) X 50? =
N-1
N-1
/ x (n) e
j2rkn / x (n)
X (k) = - N k = 0, 1 ……..(N – 1) n=0
n=0
(b) x 50? = N / X (k)

1
N-1
IDFT of X(k) is
k=0
1
N-1
x (n) = N / X (K) e N n = 0, 1 ………. (N – 1)
j2rkn
(c) X : 2 D = / ]- 1gn x (n), N even

N N-1
k=0
n=0
(d) x : 2 D = N / ]- 1gk X (k)

N 1
N-1
• The DFT & its IDFT are also periodic with period
k=0
N, and it is sufficient to compute the results for

05. Parseval’s relation:
only one period (0 to N-1).
x (n) 2 = 1 / X (k)
N-1 N-1
/ N k=0
2
n=0

N-1
X (k) = / x (n) WNkn
n=0
1
N-1
x (n) = N / X (k) W N-kn
k=0
Where WN= e -j2r/N is the phase factor.
Periodicity: W NK + N = W NK
N
Symmetry: W N = - W NK
K+ 2

Note: For direct calculation of N point DFT, we

require N2complex multiplications and
N(N-1) additions.
9.2 Linear convolution using DFT
The circular convolution of 2 sequences, of lengths N1 & N2, respectively, can be made equal to the
linear convolution of the 2 sequences by zero padding both sequences, so that they both consist of (N1
+ N2–1) samples.
x1(n) Zero padding x e1 (n) X1(k)

(N2 – 1) samples DFT
N1 samples
Zero padding x e 2 (n) X2(k) Y(k)

x2(n) y(n)
(N1 – 1) samples DFT I DFT
N2 samples
x1(n) ∗ x2(n)
Let us consider the computational efficiency of calculating a convolution using the DFT rather than
the direct method. In calculating the convolution of two N - element sequences using DFT method, we
required 3N log22N + 2N Complex multiplications where as direct convolution of 2 sequences requires N2
complex multiplication. ∴ DFT method is more efficient for N ≥ 32.
(a) Picket - fence Effect: at frequencies other than ±f0 because the
The picket - fence effect is caused by the periodic extension of the sampled portion of
approximation of the continuous frequency the periodic signal doesn’t match the original
signal but describes a different signal altogether
spectrum of the DTFT using a finite number of
as shown in fig. suppose we sample the signal
frequency points. The spectrum is observed
x(t) = sin(2πt) at fs=16 Hz. Then the sampled
very much like looking through a picket fence signal frequency (i.e, digital frequency)
with the exact value of the spectrum known ana log frequency
1
F0 = = 16
only as integer multiples of the frequency sampling rate
resolution. The peak of a particular frequency
component in a signal could be hidden from If we choose N = 8 the DFT spectral spacing
view because it is located between 2 adjacent = fs/N = 2 Hz. In other words there is no DFT
frequency points in the spectrum. To reduce component at 1 Hz, the frequency of the sine
this effect, the number of frequency points wave ! where should we expect to see the DFT
must be increased, since this enables more components?
If we express
frequency components of a signal to coincide
1 K
with the more closely spaced frequency points. F0 = as F0 = Nf
16
(b) Spectral leakage: We obtain kf = NF0 = 0.5. Thus F0 corresponds

Leakage is present in the DFT results if a periodic to the fractional index kf = 0.5 & the largest
signal x(t) is not sampled for an integer number DFT components should appear at the integer
of periods. The DFT shows nonzero components indices closest to kf at k = 0 (d .c), k = 1(2Hz)
Sinusoid sampled for one Example:

full period &its periodic Let x(n) = {1, 2, 3, 4, 5} and h(n) = {1,1,1} using
extension Overlap-add method find y(n)?
Sol: L = 6 & N = 3. x0(n) = {1, 2, 3}
h(n) = {1, 1, 1} x1(n) = {3, 4, 5}
t y0(n) = x0(n) * h(n) = {1, 3, 6, 5, 3}
y1(n) = x1(n) * h(n) = {3, 7, 12, 9, 5}
Shifting & superposition results in the required
Sinusoid sampled for half convolution
period & its periodic y(n) = y0(n) + y1[n – 3]
extension
y0(n) → 1 3 6 5 3

y1[n–3] → 3 7 12 9 5
t

To minimize leakage sample for the longest y(n) = {1, 3, 6, 8, 10, 12, 9, 5}
duration possible or for integer periods.
Convolution of Long Sequences 2. Overlap-Save method:
1. Overlap-Add method If L > N and we zero - pad the second sequence
2. Overlap-Save method to length L, their periodic convolution has
(2L – 1) samples. Its first (N -1) samples are
Some times we have to process a long stream contaminated by wraparound, and the rest
of incoming data by a filter whose impulse correspond to the regular convolution. eg. Let
response is much shorter than that of incoming L = 16 & N = 7.
data. The convolution of a short sequence h(n) If we pad N by 9 zeros, their regular convolution
of length N with a very long sequence x(n) of has 31(or 2L-1) samples with 9 trailing zeros
length L > > N can involve large amount of (L-N = 9). For periodic convolution 15 samples
computation & memory. (L-1 = 15) are wrapped around. Since the last
nine [or L-N] are zeros, only the first 6 samples
1. Overlap - Add method: of the periodic convolution are contaminated
Suppose h(n) is of length N, and the length of by wraparound, which is the basis idea of this
x(n) is L = mN (if not, we can always zero pad it method.
to this length). We partition x[n] into m segments First, we add (N – 1) leading zeros to the longer
x0(n), x1(n) …xm-1(n), each of length N. We find sequence x(n) & section it into k overlapping
the regular convolution of each section with {by N – 1} segments of length M. Typically, we
h(n) to give partial results y0(n), y1(n), … ym-1(n). choose M ≈ 2N.
y(n) = y0[n] + y1[n–N] + ym-1[n–(m–1)N] Next, we zero-pad h(n) { with trailing zeros} to
Since each regular convolution contains length M, and find the periodic convolution
(2N – 1) samples, we zero – pad h(n) and each of h(n) with each section of x(n). Finally, we
section xk[n] with [N – 1] zeros before finding discard the first (N–1) [contaminated] samples
yk[n] using the FFT. Splitting x(n) into equal – from each convolution & glue(concatenate)
length segments is not a strict requirement. the results to give the required convolution.
Example:
Find convolution of x(n) = {1, 2, 3, 4, 5} and h(n) = {1, 1, 1} using Overlap Save method?
Sol: First add (N – 1) = 2 zeros to
x(n) = {0, 0, 1, 2, 3, 3, 4, 5}
Take M = 2N – 1 = 5, section into K overlapping segments of length M(5)
x0(n) = {0, 0, 1, 2, 3}
Zero pad h(n) to length M = 5 samples
x1(n) = {2, 3, 3, 4, 5}
h(n) = {1, 1, 1, 0, 0}
x2(n) = {4, 5, 0, 0, 0}
x0(n)⊛h(n) = {5, 3, 1, 3, 6}
x1(n)⊛h(n) = {11, 10, 8, 10, 12}
x2(n)⊛ h(n) = {4, 9, 9, 5, 0}
We discard the first 2 samples from each convolution & give the results to obtain
y(n) = {1, 3, 6, 8, 10, 12, 9, 5, 0}
9.3 FFT
Fast algorithm reduce the problem of calculating an N-point DFT to that of calculating many smaller-
size DFTs. The computation is carried out separately on even - indexed and odd-indexed samples to
reduce the computational effort. All algorithms allocate for computed results. The less the storage
required, the more efficient is the algorithm.
Many FFT algorithms reduce storage requirements by performing computations in place by storing
results in the same memory locations that previously held the data.
3 stages in an 8-point DIT-FFT:

x[0] 2 point
x[4] X[0]
DFT Combine
X[1]
2 point
x[2] 2 point DFTs

x[6] DFT Combine
4 point
DFTs
x[1] 2 point
x[5] DFT Combine

2 point
DFTs
x[3] 2 point X[7]
DFT
x[7]
Stage 1 Stage 2 Stage 3

Typical butterfly for DIT – FFT algorithm:
A + BWr
A
Wr
B ⊗ A - BWr
–1
DIT algorithms for N = 2, 4, 8:
DIF algorithm for N = 2, 4, 8:
Feature N-point DFT N-point FFT

Algorithm Solution of N N/2 butterflies/stage
equations in N m stages
unknowns Total butterflies = m(N/2)
Multiplication N per equation 1 per butterfly

Addition N–1 per equation 2 per butterfly
Total multiplication N2
N
log2N
2
Total additions N(N–1) N log2N

Bit - reversal using Bruneman’s algorithm:

Practice Questions
(1) Start with {0, 1} multiply by 2 to get {0, 2}
(2) Add 1 to the list of numbers obtained above.
Now it is {1, 3}. 01. Analog data to be spectrum analyzed are
(3) Append the list in step2 to that in step1 to get sampled at 10kHz & DFT of 1024 samples
{0, 2, 1, 3} are computed. Find the frequency spacing
(4) The list obtained in step3 now becomes the between spectral samples?
starting list in step1. The steps are repeated until
the desired length of list is obtained. 02. Find the 4 point DFT of x(n) = {0, 1, 2, 3}?

0 1 X 5k? , then prove the following
N pt
  03. Let x (n)
statements.
↓×2
0 +1
2 → 13 (i) If x(n) = – x[N – 1 – n], then X(0) = 0
 
(ii) If x(n) = x[N – 1 – n], with N even,
↓×2
then X : 2 D = 0
N
0 4 2 6 +1
→1 5 3 7

 
↓×2
04. Fig shows a finite length sequence x(n).
+1
0 8 4 12 2 10 6 14 → 1 9 5 13 3 11 7 15 6
5 x(n)
2 4
2 3
2
2
n
0 1 2 3
2 2
Fig
Sketch the signals
(a) x([n – 2])4
(b) x([n + 1])4
(c) x([– n])4
05. The first five points of an 8 point DFT of a

real-valued sequence are
{0.25, 0.125 - j0.3018, 0, 0.125 - j0.0518, 0}.
Find other 3 points?
06. The DFT of a vector [a b c d] is the vector

[α β γ δ]. Consider the product
RS V
SSa b c dWWW
Sd a b cWW
6p q r s@ = 6a b c d@SSS W
SSc d a bWWW
SSb c d aWW
T X
The DFT of the vector [p q r s] is a scaled
version of
(a) 7α β γ δ A 10. Find the 10-point inverse DFT of

2 2 2 2
3, k = 0
(b) 7 α β γ δ A
X (k) = )
1, 1 # k # 9
(c) 6α + β β + δ δ + γ γ + α@
(d) [α β γ δ] 11. A signal x(t) is bandlimited to 10 kHz is sampled
with a sampling frequency of 20 kHz. The DFT of
07. Given x(n) = {1, –2, 3, –4, 5, –6}, without N = 1000 samples x(n) is then calculated.
calculating DFT, find the following quantities (a) What is the spacing between the spectral
5 samples?
(a) X(0) (b) / X (k) (c) X(3) (d) (b) To what analog frequency does the index
k=0
5 5 k = 150 correspond? What about k = 800?
/ X (k) 2
(e) / (- 1) k X (k)
k=0 k=0
12. Consider the real finite-length sequence
08. X(k) is the discrete Fourier Transform of a 6-point x[n] shown in Fig., whose six-point DFT is X[k].
real sequence x(n). If Q[k] = X[2k], k = 0, 1, 2 represents the 3-point
If X(0) = 9 + j0, X(2) = 2 + j2, DFT, then q [n] is
4 x[n]
X(3) = 3 – j0, X(5) = 1 – j1, x(0) is
3
(a) 3 (b) 9
2
(c) 15 (d) 18 1
09.
(i) The two 8-point sequences x1(n) & x2(n) shown 0 1 2 3 4 5 n
Fig.
in fig. have DFTs X1(k) and X2(k) respectively.
x2[n] (a) {5, 3, 2}
c x1[n] c (b) {4, 3, 2}
b d d b

a e e a (c) {2, 3, 4}
(d) {1, 2, 3}
0 1 2 3 4 5 6 7 n 0 1 2 3 4 5 6 7 n 13. Given x[n] = {A, 2, 3, 4, 5, 6, 7, B} is having 8
point DFT X[k]. If X(0) = 20 & X(4) = 0, find A & B?
Find the relation between X1(k) & X2(k)?
(ii). Consider the sequence x(n) shown in fig. Find 14. Given X[K] = K+1; 0 ≤ K ≤ 7 is 8-point DFT of x[n].
y(n) whose six-point DFT is
3
Then the value of / x [2n] is ____
Y (k) = W64k X (k) , n=0
Where X(k) is the six-point DFT of x(n). 15. Let x[n] be a real 8 point sequence and let X(k)
be its 8 point DFT
4
2 3 2
(A) Evaluate
x(n)
2 2 1 1 /
7
X (k) e j (2r/8) kn n = 9 in terms of x (n)
2 8 k=0
0 1 2 34 5 n (B) Let w(n) be a 4 point sequence for

2 22 2 0 ≤ n ≤ 3 and W(k) be its 4 point DFT.
Fig.
If W(k) = X(k) + X(k+4), express w(n) in
terms of x(n)
20. Let X 5k? = %1, - 2, 1 - j, j2, 0f / be the 8-point DFT

(C) Let y(n) be an 8 point sequence for 0
0 ≤ n ≤ 7 and Y(k) be 8 point DFT.

of a real signal x[n]
2X (k) k = 0, 2, 4, 6
If Y (k) = ) (a) Determine X[k] in its entirety.
0 for k = 1, 3, 5, 7
(b) What is the DFT Y[k] of the signal
express y(n) in terms of x(n). y[n] = (–1)n x[n]?
(c) What is the DFT G[k] of the zero-interpolated
16. The four point DFTS of x(n) and y(n) is
signal g[n] = x[n/2]
X(K) = {22, – 4 + j2, – 6, – 4 – j2} and
Y(K) = {8, – 2 – j2, 0, – 2 + j2}. The circular
convolution of x(n) and y(n) is w(n), then w(2)
21. The Discrete Fourier Transform (DFT) of the
is
4-point sequence
(a) 38 (b) 30
x[n] = {x[0], x[2], x[3]} = { 3, 2, 3, 4} is
(c) 12 (d) 60
X[k] = {X[0], X[1], X[2], X[3]}
= {12,2j, 0, -2j}.
17. Consider x[n] = {1, 2, 3, 4} with DFT
If X1[k] is the DFT of the 12-point sequence
X(K) = {10, -2+2j, -2, -2-2j}. If the sampling rate
x1[n]= {3, 0, 0, 2, 0, 0, 3, 0, 0, 4, 0, 0} the value
X1 58?
is 10Hz
X1 511?
(i) Determine the sampling period, time index of is
and the sampling instant for a digital sample
x(3) in time domain 22. Assume that a complex multiply takes 1µs and
(ii) Determine the frequency resolution, that the amount of time to compute a DFT is
frequency bin number and frequency for determined by the amount of time it takes to
each of the DFT coefficients X(1) and X(3). perform all of the multiplications.
(a) How much time does it take to compute a
18. We wish to sample a signal of 1-s duration, and 1024-point DFT directly?
band-limited to 100Hz, in order to compute (b) How much time is required if an FFT is used
its spectrum. The spectral spacing should not
exceed 0.5 Hz. Find the minimum number N
of samples needed and the actual spectral 23. Speech data is sampled at a rate of 10 kHz is
spacing ∆f if we use to be processed in real time using FFT. Part of
(a) The DFT (b) The radix-2 FFT the computations required involve collecting
blocks of 1024 speech values and computing
a 1024-point DFT and a 1024 point inverse DFT.
19. We wish to sample the signal,
If it takes 1 µs for each real multiply, how much
x(t) = cos(50πt) + sin(200πt) at 800 Hz and
time remains for processing the data after the
compute the N-Point DFT of the sampled
DFT and inverse DFT are computed?
signal x[n]
(a) Let N = 100. At what indices would you
expect to see the spectral peaks? Will the
peaks occur at the frequencies of x(t)?
(b) Let N = 128. At what indices would you
expect to see the spectral peaks? Will the
peaks occur at the frequencies of x(t)?
10 # 103
01. Ans:
1024

02. Ans: { 6, -2+2j, -2, -2 -2j}
04. (a). Ans: [4, 3, 6, 5]

(b). Ans: [5, 4,3,6]
(c). Ans: [6, 3,4,5]
05. Ans: X (5) = 0.125 + j0.0518

X(6) = 0, X(7) = 0.125 + j0.3018
06. Ans: (a)
09. (i) Ans: X2(k) = (-1)kX1(k)
(ii) Ans: y(n) = x((n-4))6

= {2, 1, 0, 0, 4, 3}
12. Ans: (a)

16. Ans: (a)
10. Tables
Operational Properties of the Laplace Transform
Note: x(t) is to be regarded as the causal signal x(t)u(t).
Entry Property x(t) X(s)
1 Superposition αx1(t) + βx2(t) αX1(s) + βX2(s)
2 Times-exp e-αt x(t) X(s + α)
3 Times-cos cos(αt)x(t) 0.5[X(s + jα) + X(s - jα)]
4 Times-sin sin(αt)x(t) j0.5[X(s + jα) - X(s - jα)]
5 Time Scaling x(αt), α > 0 1 bsl

aX a
6 Time Shift x(t - α) u(t - α), α > 0 e-αs X(s)
7 Times-t t x(t) dX (s)

-
ds
]- 1gn ds n
d n X (s)
8 tn x(t)
9 Derivative x′(t) sX(s) - x(0-)
10 x′′(t) s2X(s) – sx(0–) – x′(0–)
11 x(n)(t) snX(s) – sn–1 x(0–) –….– xn-1(0–)

t
12 Integral # x (t) dt X (s)

0-
s
13 Convolution x(t)∗h(t) X(s)H(s)
Switched periodic, X1 (s)

14 xp(t)u(t) , T = time period of x (t)
x1(t) = first period 1 - e -sT
Laplace Transform Theorems
15 Initial value x (0 +) = lim 6sX (s)@ (if X (s) is strictly proper)
s"3
16 Final value x (t) t"3 = lim 6sX (s)@ (if poles of X (s) lie in LHP)
s"0
A Short Table of Laplace Transform

Entry x(t) X(s) Entry x(t) X(s)
s2 - b2
^s2 + b2h2
1 δ(t) 1 10 tcos(βt)u(t)
2sb
1
^s2 + b2h2
2 u(t) 11 tsin(βt)u(t)
s
1 s2 + 2b2
3 r(t) = tu(t) 12 cos2(βt)u(t)
s2 s ^s2 + 4b2 h
2 2b2
4 t2u(t) 13 sin2(βt)u(t)
s3 s ^s + 4b2 h
2
n! s+a
]s + ag2 + b2
5 tnu(t) 14 e-αtcos(βt)u(t)
sn + 1
6 e-αtu(t) 1 15 e-αtsin(βt)u(t) b
s+a ]s + ag2 + b2
1 (s + a) 2 - b2
7 te-αtu(t) 16 te-αtcos(βt)u(t)
(s + a) 2 6]s + ag2 + b2@2
n! 2b (s + a)
6]s + ag2 + b2@2
8 tn e-αtu(t) 17 te-αtsin(βt)u(t)
(s + a) n + 1
s b
9 cos(βt)u(t) 18 sin(βt)u(t)
s2 + b2 s2 + b2
SFG for 8-Point DIT-FFT algorithm:
x(0) X(0)
W 80
x(4) X(1)
–1
W 80
x(2) X(2)
–1
0
W8 W8 2
x(6) X(3)
–1 –1 W8 0
x(1) X(4)
0
–1
W8 W 81
x(5) X(5)
–1 –1
0 2
W8 W8
x(3) X(6)
–1 –1
W 80 W 82 W 83
x(7) X(7)
–1 –1 –1
SFG for 8-Point DIF-FFT algorithm:
x(0) X(0)
0
W8
x(1) X(4)
–1
0
W8
x(2) X(2)
–1
0
W8 2 W8
x(3) X(6)
W8 0 –1 –1
x(4) X(1)
–1 0
W 81 W8
x(5) X(5)
–1 –1
2 0
W8 W8
x(6) X(3)
–1 –1
2 0
W 83 W8 W8
x(7) X(7)
–1 –1 –1
SFG for computation of IDFT using inverse Radix-2 DIT-FFT:

1/8
X(0) x(0)
-
W 80 1/8
X(4) x(1)
–1
W 80
- 1/8
X(2) x(2)
–1 -0
W 82
-
W 8 1/8
X(6) x(3)
W 80
- –1 –1
1/8
X(1) x(4)
–1 -0
W 81
- W 8
1/8
X(5) x(5)
–1 –1
-2 -
W W 80
8
1/8
X(3) x(6)
–1 –1
-3 -2 -0
W 8 W 8 W 8 1/8
X(7) x(7)
–1 –1 –1
SFG for computation of IDFT using inverse Radix-2 DIF-FFT:

1/8
X(0) x(0)
-
W 80 1/8
X(1) x(4)
–1
W 80
- 1/8
X(2) x(2)
-0
–1
W 8 W
-2
8 1/8
X(3) x(6)
–1 –1 W
-0
8
1/8
X(4) x(1)
-0
–1
W 8 W 81
-
1/8
X(5) x(5)
–1 –1
-0 -2
W 8 W 8
1/8
X(6) x(3)
–1 –1
-0 -2 -
W 8 W 8 W 83 1/8
X(7) x(7)
–1 –1 –1

Signals Systems PDF

Uploaded by

Document Information

Original Title

Copyright

Available Formats

Share this document

Share or Embed Document

Sharing Options

Did you find this document useful?

Is this content inappropriate?

Copyright:

Available Formats

Signals Systems PDF

Uploaded by

Copyright:

Available Formats

ACE

Signals & Systems

03. Signum function: Sgn(t)

02. Rectangular (or Gate) Pulse: A rect(t/2a)

The rectangular function is the result of an

06. Sampling function: (iii)

sin x sin πx # δ (t) dt = 1:

Sa(x) Area under unit impulse function is one.

(v) Product property

Energy and Power signals:

Periodic And Non Periodic

Sampled version of certain continuous Signals

Classification of Systems: 4. Static (or) Memoryless & Dynamic

3. Causal & Non Causal System:

Useful Mathematical Relations Practice Questions

elsewhere. Then the signal y (t) = 2x ; 2 t E is

5 14. Consider an LTI system whose response to the

(f) y(t) = x2(t) 1 2

15. Check whether the following systems are

n + n0 (d) y(n + 1)y(n) = 4 x(n)

or dynamic? (d) y (t) = d x (t)

(b) y(t) = x2(t)

(c) y(n) = ex(n)

20. Determine which of the following systems are

y(n) is not delayed by 2 units.

correct explanation of Statement (I)

2. LTI (LSI Systems)

2.1 Continuous & Discrete Convolution

x(t) L.T.I y(t) = x(t)*h(t) x(n) L.T.I y(n) = x(n)*h(n)

x(t – τ)h(τ) x[k] h[n – k]

• An L.T.I System is always considered w.r.t

2.2 Properties of L.T.I System h(t) x(t)

Causality: Stability: 1/2

h(t) = 0; t < 0 # h (x) dx < 3 0 6 t (sec) 0 2 6 t (sec)

(c) 3/2 (d) 1

05. Suppose z (t) = # x (− τ + a) h (t + τ) dτ

Express z(t) in terms of y(t) = x(t) ∗ h(t)

(c) Consider a signal averaging filter whose

18. Two discrete -time signals x[n] and h[n] are

k =-3 both non-zero only for n = 0,1,2, and are zero

(d) None of these

14. The integral # x (τ) h (τ − t) dτ can be

Discrete Convolution (d) |a|<1, |b| > 1

(c) u(n) – u(n–N–1) What condition the system is stable.

23. For the interconnected system shown in Fig.

24. Determine whether each of the following 1 b +bl

25. Ans: (a)

However, musical instruments, such as a piano,

Error in the approximation e = f – CX

Examples (b) For orthonormality 2T = 1 ⇒ T = 1/2

t • Any practical periodic function of frequency

-1 (or) cosine functions that are integral multiples

Unit - Magnitude Property T

Effect of symmetry on Fourier coefficients:

3.3 Exponential (or) Complex F.S 5. Differentiation in time:-

Convergence Of F.S: (Dirichlet Conditions)

(2) x(t) has only a finite number of maxima &

3.4 Properties of F.S

2. Time Shift: x (t - t 0) * Cn e -jn~0 t0

Practice Questions (R) g (t)

02. Which of the following signal is not the

03. For the Periodic waveforms shown in figure (S)

07. The fundamental frequency of the composite

x(t) L.T.I y(t) = x(t)h(t) x(n) L.T.I y(n) = x(n)h(n)