QUESTION BANK
UNIT I
PART A
1. Distinguish between energy and power signal.
2. How can we prevent aliasing?
Aliasing is a effect of violation of the NyquistShannonSamplingTheory. During
the sampling the base band spectrum of the sampled signal ist mirrored to every
multifold of the sampling frequency. These mirrored spectra are called alias. If the
signal spectrum reaches farther than half the sampling frequency base band spectrum
and aliases touch each other and the base band spectrum gets superimposed by the
first alias spectrum. The easiest way to prevent aliasing is the application of a steep
sloped lowpass filter with half the sampling frequency before the conversion.
3. Classify the signals?
Multichannel Signal:
Signals generated by multiple sources or multiple sensors. Ex: EEG signal.
One dimensional Signals:
A function of single independent variable. Ex: v(t) = V
m
sin(t)
Multi Dimensional Signals:
A function of two or more independent variables.
Ex: Photo 2 dimensional signal
Ex: B/W TV Picture intensity 3 dimensional signal, I(x,y,t)
4. What is a multi channel signal?
Signals generated by multiple sources or multiple sensors. Ex: EEG signal.
5. State analog signal.
The analog signal is a continuous function of an indepentent variable such as
time, space, etc.
6. Find whether the given system is static or dynamic. y(n) = n x(n)+5x
3
(n)
The output y(n) depends on the input at that instant only. Therefore the
system is static.
7. Determine z transform and ROC of the signal {1,2,3,4}
Solution:
The Ztransform of a sequence , ( ) ( )
3
0
n
x z x n z
( )
0 1 2 3
(0) (1) (2) (3) x z x z x z x z x z
= + + +
Where x(0) = 1
x(1) = 2
x(2) = 3
x(3) = 4
Substitute x(0),x(1), x(2) and x(3) values in above equation,
( )
0 1 2 3
(1)* (2)* (3)* (4)* x z z z z z
= + + +
Answer : ( )
0 1 2 3
2 3 4 x z z z z z
= + + +
8. List the mathematical operations performed on discrete time signals.
Shifting
Time reversal
Shifting after time reversal
Time scaling
Scalar multiplication
Signal multiplier
Signal addition
9. Find whether the given system is linear or not. Y(n)=n x(n)
Let H be the system represented by the equation , y(u) = n x(n) and the system H
operates on x(n) to produce y(n) = H[x(n)] = n x(n)
Consider two signals x
1
(n) and x
2
(n).
Let y
1
(n) and y
2
(n) be the response of the system H for inputs x
1
(n) and x
2
(n) respectively.
y
1
(n) = H[x
1
(n)] = n x
1
(n) .(1)
y
2
(n) = H[x
1
(n)] = n x
2
(n) .(2)
consider av linear combination of inputs a
1
x
1
(n)+ a
2
x
2
(n). let the response of the system for
this linear combination of inputs be y
3
(n).
y
3
(n) = H[a
1
x
1
(n)+ a
2
x
2
(n)] = n(a
1
x
1
(n)+ a
2
x
2
(n)) = a
1
nx
1
(n)+ a
2
nx
2
(n) (3)
The condition to be satisfied for linearity is, y
3
(n) = a
1
y
1
(n)+ a
2
y
2
(n) .(4)
From equation (1) we get, a
1
y
1
(n) = a
1
nx
1
(n)
From equation (2) we get, a
2
y
2
(n) = a
2
nx
2
(n)
Therefore, a
1
y
1
(n)+ a
2
y
2
(n) = a
1
nx
1
(n)+ a
2
nx
2
(n) (5)
From equations (4) and (5) we can say that the condition for linearity is satisfied.
i.e., y
3
(n) = a
1
y
1
(n)+ a
2
y
2
(n). Hence the system is linear.
10. What is meant by ROC?
The values of z for which the ztransform converges is called region of
convergence (ROC).
11. Define z transform.
The ztransform of the discretetime signal x[n] is defined to be
where is a complex variable. In polar form z can be expressed as z = re
j
,
where r is the radius of the circle.
12. List the various methods of classifying discrete time system.
Static & Dynamic systems
Timevariant & Timeinvariant systems
Causal & Noncausal systems
Linear & Nonlinear systems
Stable & Unstable systems
FIR & IIR systems
13. Determine z transform and ROC of the signal {5,6,7,8}
x (n) = {5,6,7,8}
Solution :
The Ztransform of a sequence , ( ) ( )
0
3
n
x z x n z
( )
3 2 1 0
( 3) ( 2) ( 1) (0) x z x z x z x z x z = + + +
Where x(0) = 8
x(1) = 6
x(2) = 7
x(3) = 8
Substitute x(0), x(1), x(2) and x(3) values in above equation,
( )
3 2 1 0
(8)* (7)* (6)* (5)* x z z z z z = + + +
Answer : ( )
3 2 1 0
8 7 6 5 x z z z z z = + + +
14. What are the various methods of representing discrete time signal?
Graphical representation
Functional representation
Tabular representation
Sequence representation
15. How will you classify the discrete time signal?
Signals can be classified into various types by
i. Nature of the independent variables
ii. Value of the function defining the signals
16. List out some important properties of ROC.
The importance of the ROC cannot be overemphasized. It is part of the Z
transform.
In specifying the Ztransform x( z) of a signal x( n), the ROC must be given 
otherwise, the Ztransform cannot be inverted  in order to reobtain x( n) from
X(z), the ROC must be given.
o Example  Consider two sequences
o
transforms do not even intersect. Not equal! X( z) + ROC unique x( n).
ROC Shape is either Ring or disk in the zplane centered at the origin.
Fourier Transform converges absolutely if and only if the ROC of the Z
transform of x(n) includes unite circle.
ROC cannot contain poles.
ROC Region must be a connected region.
17. Determine the convolution sum of two sequences x(n)={3,2,1,2} and h(n)={1,2,1,2}.
Solution :
H(n) = x(n) * h(n)
h(n)
x(n)
1 2 1 2
3 3 6 3 6
2 2 4 2 4
1 1 2 1 2
2 2 4 2 4
y(0) = 3
y(1) = 2+6 = 8
y(2) = 1+4+3 = 8
y(3) = 2+2+2+6 = 12
y(4) = 4+1+4 = 9
y(5) = 2+2 =4
y(6) = 4
y(n) = {3,8,8,12,9,4,4}
PART B
1. Explain signals and classify the signals with suitable examples. (12 marks)
Signals:
Any physical quantity that varies with time, space or any other independent
variables is called a signal. Contain information about the behavior or nature of some
phenomenon. Signals are the link up between systems. Signal is a record of an activity
containing information.
Ex: x(t) = sin(3t)
x[n] = 10 cos[5n]
Types of Signals:
1. Multichannel Signal:
Signals generated by multiple sources or multiple sensors. Ex: EEG signal.
2. One dimensional Signals:
A function of single independent variable. Ex: v(t) = V
m
sin(t)
3. Multi Dimensional Signals:
A function of two or more independent variables.
Ex: Photo 2 dimensional signal
Ex: B/W TV Picture intensity 3 dimensional signal, I(x,y,t)
Continuous time and Discrete time signals:
Continuoustime Signal x(t):
It is also called analogue signal.
It is continuous in its time axis.
t
Amplitude
Discretetime Signal x[n]:
The time axis is discrete.
Independent variable (usually time) has integer values.
It is sequence of numbers. e.g. x[n]=[,3,8,9,0,1,]
DISCRETE TIME SIGNALS:
 The discrete time signals are obtained by time sampling of continuous time
signals.
 It can be defined at only at sampling instants.
Types of Discrete time Signals
1. Deterministic & Nondeterministic
2. Periodic & Aperiodic
3. Even & Odd
4. Energy & Power
Deterministic & Nondeterministic:
Deterministic:
Functions that are completely specified in time.
The nature and amplitude at any time can be predicted.
The pattern is regular and can be characterized mathematically.
Ex: x[n] = {1,0,3,8}
Nondeterministic (Random):
Signals whose occurrence is random in nature.
Pattern is quite irregular.
The amplitude at any time can not be predicted in advance.
Ex. Noises
nT
Amplitude
T
T
2T
3T 4T
5T 6T 7T
0
Explanation:
Deterministic and Random Signals:
 A deterministic signal is a signal about which there is no uncertainty with
respect to its value at any time.
 A deterministic signal can be completely represented by mathematical equation at
any time.
 For example sine wave, cosine wave, triangular, exponential pulse etc are
deterministic signals, since they have unique mathematical equations.
 An example of deterministic signal is,
i.e x(t) = A Sin et
 A random signal has some degree of uncertainty before it actually occurs.
 That means random signal cannot be defined by some mathematical
expression.
 The value of random signal is not predefined or it cannot be calculated from
previous values of the signal.
 Random signals are broadly called as Noise signals.
 An example of random signal is,
Periodic & Aperiodic:
Periodic:
Should exhibit periodicity.
x[n+N] = x[n],  < n < , where N is the period.
Aperiodic:
Does not satisfy the above condition.
Explanation:
Periodic and Nonperiodic Signals:
 A signal is said to be periodic if it repeats after a fixed time period.
 This can be defined by mathematical expression as,
X(t) = x (t + T
0
)
 An example of periodic signal is sine wave.
 This waveform can be represented by the following mathematical equation
i.e.,
= x (t)
 A signal is said to be nonperiodic if it does not repeat.
 Some times nonperiodic signals are said to have a period T
0
equal to
infinity.
 An example of nonperiodic signal is decaying exponential pulse.
Even & Odd:
Even:
Exhibits symmetry in the time domain.
The signal must be identical to its reflection about the origin.
x[n] = x[n]
Odd:
Exhibits antisymmetry.
The signal must be identical to the negative of its reflection about the origin.
x[n] = x[n]
Explanation:
Symmetric and Nonsymmetric Signals or Even and Odd Signals:
Odd or nonsymmetrical signal:
 A signal x(t) is said to be odd or nonsymmetric if,
x(t) = x(t)
 This means if the sign of time axis is changed, it also changes the sign of amplitude.
 For example consider the sine wave starting at zero. This sine wave is shown.
= x(t)
Even or symmetric signal:
 A signal is said to be even or symmetric if,
x(t) = x(t)
 This means even if the sign of the time axis is changed, the sign of the amplitude is not
changed.
 For example consider the cosine wave as shown in Fig. below.
= A cos (2ft) Since cos (u) = cos u
= x (t)
Energy & Power:
Let x[n] be a discretetime signal
The energy content of the signal x[n] is as follows:
The power content of the signal is:
If E is finite and P = 0, then x[n] is Energy signal.
If P is finite and E = infinite, then x[n] is Power signal.
Singularity Functions
These functions are important classifications of aperiodic signals. They are used to
represent more complicated signals.
Types:
1. Unit impulse function / Delta function
2. Unit step function
3. Unit ramp function
Delta function is the basic singularity function and other singularity functions are
derived by repeated integration or differentiation of the delta function.
=
=
n
n x E
2
] [
+
=
+
=
N
N n
N
n x
N
P
2
] [
1 2
1
lim
Unit impulse function
[n] = 1, n = 0
0, n 0
Unit step function
u[n] = 0, n < 0
1, n 0
Unit ramp function
r[n] = 0, n < 0
n, n 0
Representation of discrete time signals
Graphical representation
Functional representation
Tabular representation
Sequence representation
1
3 2 1 0 1 2 3
1
3 2 1 0 1 2 3
1
3 2 1 0 1 2 3
2
3
Operations on discrete time signals
Shifting:
y[n] = x[nk]
x[n] = input, y[n] = output
k = +ive, delay the sequence  right shift
k = ive, advance the sequence left shift
Time reversal:
y[n] = x[n]
Shifting after time reversal :
x[n+2] is x[n] delayed by 2 units
x[n2] is x[n] advanced by 2 units
Time scaling:
y[n] = x[n]
< 1, up sampling
> 1, down sampling
n
x[n]
X[n] =
1, n = 0,4,7
5, n = 3
4, n = 1,2,5,6
n
x[n]
0 1 2 3 4 5 6 7
1 4 4 5 1 4 4 1
X[n] = {1, 4, 4, 5, 1, 4, 4, 1}
Graphical
Functional
Tabular
Sequence
1
2
3
Scalar multiplication:
y[n] = a x[n]
Signal multiplier:
y[n] = x1[n] x2[n]
Signal addition:
y[n] = x1[n] + x2[n]
X[n] a y[n]
X
x x1 1[ [n n] ]
x x2 2[ [n n] ]
y[n]
+
x x1 1[ [n n] ]
x x2 2[ [n n] ]
y[n]
2. Find the following summations
i) ( ) n n
n
2 sin 2
=
o (3 marks)
ii)  
+ n n n n 2 sin ) 1 ( 2 cos ) 2 ( o o (3 marks)
iii) ( )
n
n
e n
2
0
=
o (3 marks)
iv) ( ) ) ( 1
0
n x n
n
=
+ o
(3 marks)
Solution :
i) ( ) n n
n
2 sin 2
=
o
condition,
(n2) = 1 for n = 2
= 0 for n 2
2
sin 2
n
n
=
=
Sub n = 2 in above equaton,
= sin4 (or) 0.0698
ii)  
+ n n n n 2 sin ) 1 ( 2 cos ) 2 ( o o
solution:
condition,
(n2) = 1 for n = 2
= 0 for n 2
2
cos 2
n
n
=
=
= cos 2(2)
= cos 4
( 2)cos 2 n n o
condition,
(n1) = 1 for n = 1
= 0 for n 1
1
sin 2
n
n
=
=
= sin 2(1)
= sin 2
 
2 1
( 2) cos 2 ( 1)sin 2 cos 2 sin 2
n n
n n n n n n o o
= =
+ = +
= cos4 + sin2
= 1.0325
iii) ( )
n
n
e n
2
0
=
o
condition,
(n) = 1 for n = 0
= 0 for n 0
=
2
0
n
n
e
=
=
( ) 2 0
e
=
0
e
= 1
iv) ( ) ) ( 1
0
n x n
n
=
+ o
condition,
(n) = 1 for n = 1
= 0 for n 1
=
0
( )
n
x n
=
= (0) x
( 1)sin2 n n o
3. Determine the values of power and energy of the following signals and Find whether
the signals are power or energy signals.
i) ( )

.

\

= n n x
4
sin
t
(4 marks)
ii) ( ) ( ) n u n x
n

.

\

=
3
1
(4 marks)
iii) ( )
=
=
N
n
N
n u E
0
2
lim
(4 marks)
Solution:
i) ( )

.

\

= n n x
4
sin
t
To find E values,
2
( )
n
E x n
=
=
2
sin
4
n
n
t
=
 
=

\ .
where
2
1 cos 2
4
sin
4 2
t
t
 

 
\ .
=

\ .
2
sin
4
n
n
t
=
 
=

\ .
1 cos 2
4
2
n
n
t
=
1 cos
2
2
n
t
=
E (Energy) =
To find P values,
( )
2 1
2 1
N
N
n N
P Lim x n
N
=
=
=
+
2
1
sin
2 1 4
N
N
n N
P Lim n
N
t
=
=
=
+
2
1
sin
2 1 4
N
N
n N
P Lim n
N
t
=
=
=
+
1 cos
1 2
2 1 2
N
N
n N
n
P Lim
N
t
=
=
 

\ .
=
+
1
1
2 1
N
N
n N
P Lim
N
=
=
+
where
1 2 1
N
n N
N
=
= +
1
2 1
2 1
N
P Lim N
N
= +
+
P (Power) =
The energy is infinite and the power is finite. Therefore, the signal is Power signal.
ii) ( ) ( ) n u n x
n

.

\

=
3
1
To find E values,
2
( )
n
E x n
=
=
Therefore u(n) = 1 for n 0
= 0 for n < 0
Put u(n) = 1
2
1
3
n
n
E
=
 
=

\ .
0
1
9
n
n
E
=
 
=

\ .
where
2
1
1 ..............
1
a a
a
+ + + + =
1 9
1
8
1
9
E = =
9
8
E =
To find P values,
( )
2 1
2 1
N
N
n N
P Lim x n
N
=
=
=
+
1
1
1
1 9
1
2 1
1
9
N
N
P Lim
N
+
=
(
 
(

\ .
(
=
( +
(
P = 0.
The energy is finite and the power is zero. Therefore, the signal is an energy signal.
iii) ( )
=
=
N
n
N
n u E
0
2
lim
To find E values,
2
lim ( )
N
N
n N
E x n
=
=
2
lim ( )
N
N
n N
E u n
=
=
where ,
( )
2
0
1
N
n
u n N
=
= +
2
lim ( )
N
N
n N
E u n
=
=
lim 1
N
N
n N
E N
=
= +
E =
To find P values,
( )
2 1
2 1
N
N
n N
P Lim x n
N
=
=
+
2
0
1
( )
2 1
N
N
n
P Lim u n
N
=
=
+
2
0
1
( )
2 1
N
N
n
P Lim u n
N
=
=
+
0
1
1
2 1
N
N
n
P Lim N
N
=
= +
+
1
1
2 1
N
P Lim N
N
= +
+
1
1
1
2
N
N
N
P Lim
N
N
 
+

\ .
=
 
+

\ .
Sub N=,
1
1
1
2
P
 
+

\ .
=
 
+

\ .
1
2
P =
The energy is infinite and the power is finite. Therefore, the signal is Power signal.
4. Test the causality of the following systems
i) y(n) = x(n)x(n1) (3marks)
ii) y(n) = ax(n)+bx(n1) (3marks)
iii) y(n) = x(n
2
) (3marks)
iv) y(n) = nx(n) (3marks)
Solution:
i) y(n) = x(n)x(n1)
for n = 1 ; y(1) = x(1) x(11) = x(1) x(2)
for n = 0 ; y(0) = x(0) x(01) = x(0) x(1)
for n = +1 ; y(1) = x(1) x(11) = x(1) x(0)
For all values of n the otput y(n) depends on present and past inputs. Therefore,
the system is causal.
ii) y(n) = ax(n)+bx(n1)
for n = 1 ; y(1) = ax(1) bx(11) = ax(1) bx(2)
for n = 0 ; y(0) = ax(0) b x(01) = ax(0) bx(1)
for n = +1 ; y(1) =a x(1) bx(11) = ax(1) bx(0)
For all values of n the otput y(n) depends on present and past inputs. Therefore,
the system is causal.
iii) y(n) = x(n
2
)
for n = 1 ; y(1) = x(1
2
) = x(1)
for n = 0 ; y(0) = x(0)
for n = +1 ; y(1) = x(1
2
) = x(1)
For all values of n, expect for n=0 and n=1, the output y(n) depends on future
inputs.Therefore, the system is noncausal.
iv) y(n) = nx(n)
for n = 1 ; y(1) = 1x(1) = x(1)
for n = 0 ; y(0) = x(0)
for n = +1 ; y(1) = x(1)
For all values of n the otput y(n) depends on present and past inputs. Therefore,
the system is causal.
5. Explain the properties of Z  transform. (12marks)
Properties of ztransform:
6. Test the Time invariance of the following systems.
i).y(n) = x(n)+c (3marks)
ii).y(n) = x(n)x(n1) (3marks)
iii).y(n) = x(n) (3marks)
iv).y(n) = x(n)bx(n1) (3marks)
Solution :
i).y(n) = x(n)+c
unshifed input = H[x(n)] = y(n) = x(n)+c
Delayed input = y(nk) = H[x(nk)] = x(nk)+c
Delayed Response = y(nk) = z
k
H[x(n)] = z
k
x(n)+c
= x((nk))+c
= x(n+k)+c
Here , y(nk) y(nk)
Hence this system is a time varient
ii).y(n) = x(n)x(n1)
unshifed input = H[x(n)] = y(n) = x(n)x(n1)
Delayed input = y(nk) = H[x(nk)] = x(nk)x(nk1)
Delayed Response = y(nk) = z
k
H[x(n)] = z
k
[x(n)x(n1)]
= z
k
x(n)  z
k
x(n1)
= x(nk)x(nk1)
Here , y(nk) = y(nk)
Hence this system is a time invarient
iii).y(n) = x(n)
unshifed input = H[x(n)] = y(n) = x(n)
Delayed input = y(nk) = H[x(nk)] = x(nk)
Delayed Response = y(nk) = z
k
H[x(n)] = z
k
x(n)
= x((nk))
= x(n+k)
Here , y(nk) y(nk)
Hence this system is a time varient
iv).y(n) = x(n)bx(n1)
unshifed input = H[x(n)] = y(n) = x(n)bx(n1)
Delayed input = y(nk) = H[x(nk)] = x(nk)bx(nk1)
Delayed Response = y(nk) = z
k
H[x(n)] = z
k
[x(n)bx(n1)]
= z
k
x(n)  z
k
bx(n1)
= x(nk)bx(nk1)
Here , y(nk) = y(nk)
Hence this system is a time invarient
7. Determine the Ztransform and ROC of the causal and non causal sequence
i).x (n) = {1, 0, 3,1, 2} (4marks)
ii).x (n) = {1,2, 1, 3, 4} (4marks)
iii).x (n) = {1, 2, 5,4, 1, 3,1, 2, 1} (4marks)
Solution:
7 i).x (n) = {1, 0, 3,1, 2}
The Ztransform of a sequence , ( ) ( )
4
0
n
x z x n z
( )
0 1 2 3 4
(0) (1) (2) (3) (4) x z x z x z x z x z x z
= + + + +
Where x(0) = 1
x(1) = 0
x(2) = 3
x(3) = 1
x(4) = 2
Substitute x(0),x(1), x(2),x(3) and x(4) values in above equation,
( )
0 1 2 3 4
(1)* (0)* (3)* ( 1)* (2)* x z z z z z z
= + + + +
Answer : ( )
0 2 3 4
3 1 2 x z z z z z
= + +
7. ii) x (n) = {1,2, 1, 3,4}
Solution :
The Ztransform of a sequence , ( ) ( )
0
4
n
x z x n z
( )
4 3 2 1 0
( 4) ( 3) ( 2) ( 1) (0) x z x z x z x z x z x z = + + + +
Where x(0) = 4
x(1) = 3
x(2) = 1
x(3) = 2
x(4) = 1
Substitute x(0),x(1),x(2),x(3) and x(4) values in above equation,
( )
4 3 2 1 0
(1)* ( 2)* (1)* (3)* (4)* x z z z z z z = + + + +
Answer : ( )
4 3 2 1 0
2 3 4 x z z z z z z = + + +
( )
4 3 2 1
2 3 4 x z z z z z = + + +
7.iii) x (n) = {1, 2, 5,4, 1, 3,1, 2, 1}
Solution :The Ztransform of a sequence , ( ) ( )
4
4
n
x z x n z
( )
4 3 2 1 0 1 2 3 4
( 4) ( 3) ( 2) ( 1) (0) (1) (2) (3) (4) x z x z x z x z x z x z x z x z x z x z
= + + + + + + + +
Where x(0) = 1 x(1) = 3
x(2) = 1 x(3) = 2
x(4) = 1 x(1) = 4
x(2) = 5 x(3) = 2
x(4) = 1
Substitute x(1),x(2),x(3),x(4),x(0),x(1),x(2),x(3) and x(4) values in
above equation,
( )
4 3 2 1 0 1 2 3 4
(1)* (2)* (5)* ( 4)* (1)* (3)* ( 1)* (2)* (1)* x z z z z z z z z z z
= + + + + + + + +
Answer : ( )
4 3 2 1 0 1 2 3 4
2 5 4 3 1 2 x z z z z z z z z z z
= + + + + + +
( )
4 3 2 1 1 2 3 4
2 5 4 1 3 2 x z z z z z z z z z
= + + + + + +
8. Define discrete time system and classify the discrete time system with suitable examples.
(12marks)
Discrete Time Systems
Definition:
It is defined as a system in which the associated signals are also discrete time
signals. This is means that in a discrete time system, the output will be depends on values
of input x(t) for t t
0
.
Types:
 Static & Dynamic systems
 Timevariant & Timeinvariant systems
 Causal & Noncausal systems
 Linear & Nonlinear systems
 Stable & Unstable systems
 FIR & IIR systems
Static & Dynamic:
Static:
The output at any instant depends on the input samples at the same time, but not
on the past or future samples of the input.
Memoryless system.
Ex: y[n] = a x[n] and y[n] = a x2[n]
Dynamic:
System with memory.
Ex: y[n] = x[n1] + x[n2]
Time variant & Time invariant:
Time invariant:
o The input output characteristics do not change with time.
o If y[n] = H{x[n]} then, y[nk] for an input of x[nk] will be y[n], time shifted by k
units.
o The response y[nk] to a shifted version of the input x[nk] is identical to a shifted
version of the response y[n] for the unshifted input x[n].
Time invariant: y[nk] = y[nk]
Time variant: y[nk] y[nk]
Linear & Nonlinear:
Linear:
Should satisfy superposition principle.
The response of the system to a weighted sum of signals should be equal to the
corresponding weighted sum of the outputs to each of the individual input signals.
H{a1x1[n] + a2x2[n]} = a1H{x1[n]} + a2H{x2[n]}
Nonlinear: It does not satisfy the superposition principle.
Delay H
x[n]
x[nk]
y[nk]
H Delay
x[n]
y[n] y[nk]
x1[n]
x2[n]
x1[n]
x2[n]
a1
a2
a1
a2
+
+
H
H
H
y[n]
y[n]
Causal & Noncausal:
Causal:
The output at any time depends only on present and past inputs, but does not
depend on future inputs.
Ex: y[n] = 7x[n] 9x[n3]
Noncausal:
The output at any time depends not only on present and past inputs but also on
future inputs unrealistic.
Ex: y[n] = 3x[n] + 6x[n+3]
Stable & Unstable:
Stable:
A system is said to be Bounded Input Bounded Output (BIBO) stable if and only
if every bounded input produces a bounded output
Bounded means finite
System is stable if the impulse response is absolutely summable. (i.e.)
<
= k
k h ] [
9. Perform the circular convolution of the following sequences x
1
(n) = {4, 3, 2, 1} and
x
2
(n) ={5,2,3,4}. (12marks)
10. Find the linear convolution for the given sequence x(n)= { 1,2,3,4} and
h (n) = {1, 1,1,1}. (12marks)
Solution:
The convolution sum formula is used to find the response of the system
( ) ( ) ( )
k
y n x k h n k
=
=
( ) ( ) ( ) * y n x n h n =
Input signal x(n) has the values from n = 0 to n = 3 and hence x(k) extends from 0 to 3.
( ) ( ) ( )
3
0 k
y n x k h n k
=
=
The inputs signal x(n) varies from 0 to 6 and the impulse response varies fronm 0 to 3.
Hence y(n) varies from [ 0 + (0) ] to [ 3 + 3 = 6]. y(n) values to be determined are y(0), y(1), y(2),
y(3), y(4), y(5), y(6).
The length of the output sequence is determined by M
1
+ M
2
1, where M
1
is the length of input
signal and M
2
is the length of impulse response.
Here , M
1
+ M
2
1 = 4 + 4 1 = 7
To find y(0) values :
( ) ( ) ( )
3
0
0 0
k
y x k h k
=
=
( ) ( ) ( )
3
0
0
k
y x k h k
=
=
( ) 0 (0) (0) (1) ( 1) (2) ( 2) (3) ( 3) y x h x h x h x h = + + +
( )
( )
0 (1*1) (2*0) (3*0) (4*0)
0 1
y
y
= + + +
=
To find y(1) values :
( ) ( ) ( )
3
0
1 1
k
y x k h k
=
=
( ) 1 (0) (1) (1) (0) (2) ( 1) (3) ( 2) y x h x h x h x h = + + +
( )
( )
( )
1 (1*1) (2*1) (3*0) (4*0)
1 1 2
1 3
y
y
y
= + + +
= +
=
To find y(2) values :
( ) ( ) ( )
3
0
2 2
k
y x k h k
=
=
( ) 2 (0) (2) (1) (1) (2) (0) (3) ( 1) y x h x h x h x h = + + +
( )
( )
( )
2 (1*1) (2*1) (3*1) (4*0)
2 1 2 3
2 6
y
y
y
= + + +
= + +
=
To find y(3) values :
( ) ( ) ( )
3
0
3 3
k
y x k h k
=
=
( ) 3 (0) (3) (1) (2) (2) (1) (3) (0) y x h x h x h x h = + + +
( )
( )
( )
3 (1*1) (2*1) (3*1) (4*1)
3 1 2 3 4
3 10
y
y
y
= + + +
= + + +
=
To find y(4) values :
( ) ( ) ( )
3
0
4 4
k
y x k h k
=
=
( ) 4 (0) (4) (1) (3) (2) (2) (3) (1) y x h x h x h x h = + + +
( )
( )
( )
4 (1*0) (2*1) (3*1) (4*1)
4 2 3 4
4 9
y
y
y
= + + +
= + +
=
To find y(5) values :
( ) ( ) ( )
3
0
5 5
k
y x k h k
=
=
( ) 5 (0) (5) (1) (4) (2) (3) (3) (2) y x h x h x h x h = + + +
( )
( )
( )
5 (1*0) (2*0) (3*1) (4*1)
5 3 4
5 7
y
y
y
= + + +
= +
=
To find y(6) values :
( ) ( ) ( )
3
0
6 6
k
y x k h k
=
=
( ) 6 (0) (6) (1) (5) (2) (4) (3) (3) y x h x h x h x h = + + +
( )
( )
( )
6 (1*0) (2*0) (3*0) (4*1)
6 4
6 4
y
y
y
= + + +
=
=
The linear convolution for the output sequence, y(n) = { 1, 3, 6, 10, 9, 7, 4 }
QUESTION BANK
UNIT II
PART A
1. List down any four properties of DTFT.
 Linearity
 Periodicity
 Time shifting
 Time reversal
 Convolution
 Frequency shift
2. Define DFT.
integer an is k where ), ( ) 2 ( e t e X k X = +
) ( ]} [ { e
e
X e k n x F
k j
=
) ( ]} [ { e = X n x F
) ( ) ( ]} [ ] [ { e e H X n h n x F = 
) ( ]} [ { 0 e e
e
= X n x e F
n j o
) ( ) ( ]} [ ] [ { 2 2 1 1 2 2 1 1 e e X a X a n x a n x a F + = +
The discrete fourier transform compute the values of the ztransform for
evenly spaced points around the unit circle for a given sequence.
3. State shifting property of DFT.
Let x
p
(n) is a periodic sequence with period N, which is obtained by x(n)
( ) ( )
p
t
x n x n lN
=
=
( ) ( )
p
t
x n k x n k lN
=
=
4. What is zero padding?
Let the sequence x(n) has a length L. If we want to find the Npoint DFT
(NL) of the sequence x(n). we have to add (NL) zeros to the sequences
x(n). this is known as zero padding.
Uses :
1. We can get better display of the frequency spectrum.
2. With zero padding, the DFT can be used in linear filtering.
5. Find Fourier transform of a sequence
s s
otherwise
n for
0
2 2 1
Solution :
( )
1 2 2
0
for n
x n
otherwise
s s
( ) ( )
j n
n
X x n e
e
e
=
=
( )
2
2
j n
n
X e
e
e
=
=
( ) ( ) ( ) ( ) ( ) ( )
2 0 2
2 1 0 1 2
j j j j
X x e x e x e x e x e
e e e e
e
= + + + +
( )
2 0 2 j j j j
X e xe e e e
e e e e
e
= + + + +
6. What is DIT radix2 FFT?
The DIT radix2 FFT is an efficient algorithm for computing DFT. The time
domain Npoint sequence is decimated into 2point sequences. The result of 2
point DFTs are used to compute 4points DFTs. Two numbers of 2point
DFTs are combined to get a 4point DFT. The results of 4point DFTs are
used to compute 8point DFTs. Two numbers of 4point DFTs are combined
to get an 8point DFT.
7. Find the computation efficiency of 1024 point FFT over 1024 point DFT.
The direct computation of 1024 point DFT requires N
2
= (1024)
2
=
1048576 multiplications.
The 1024 points requires
2
log
2
N
N
2
1024
log 1024 5120
2
multiplications = =
8. Why FFT is needed?
FFT is algorithms are required for the coefficient computation of DFT.
The direct computation of DFT requires N
2
complex multiplication and N
2
N complex
additions where as FFT requires only
2
log
2
N
N complex multiplication and
2
log
2
N
N
complex additions. If N increases then the processing speed increases in FFT algorithms.
9. What is the sufficient condition for the existence of DTFT?
The DTFT equation will converge, if x(n) is absolutely summable.
( )
n
x n
=
<
This is the sufficient and stronger dirichlet condition , the first two dirichlet
conditions of continuous time signals are not applicable in discrete time signal. Some
signals are not absolutly summable, but they are square summable.
10. Distinguish between DTFT and DFT.
Te spectrum of DTFT x() is continuous and if is not convenient to
calculate x() in digital signal processor.
11. How many multiplications and additions are required to compute Npoint
DFT using radix2 FFT.
Number of multiplications
2
log
2
N
N =
Number of additions
2
log N N =
12. What are the applications of FFT algorithms?
 Linear filtering
 Correlation
 Spectrum analysis
13. Give relationship between DTFT, DFT and Z transform.
The DTFT X() and ztransform X(z) are related by X() = H(z)
z=e
j
The DTFT X(k) and DTFT X() are related by X(k) = H()
=e
j2k/N
14. Write the application of Fourier transforms.
 The frequency response of LTI system is given by the fourier transform of the
impulse response of the system.
 The ratio of the fourier transform of output to fourier transform of input is the
transfer function of the system is frequency domain.
 The response of an LTI system can be easily computed using convolution
property of fourier transform.
15. Determine the DTFT of the sequence x(n) = {1,1,1,1).
( ) ( )
j n
n
X x n e
e
e
=
=
( )
3
0
( )
j n
n
X x n e
e
e
=
=
( ) ( ) ( ) ( ) ( )
0 2 3
0 1 2 3
j j j
X x e x e x e x e
e e e
e
= + + +
( )
2 3
1
j j j
X e e e
e e e
e
= +
16. Draw the radix2 butterfly diagram for DIT and DIF algorithms.
Basic butterfly of DIT radix 2
Basic butterfly of DIF radix 2
17. Arrange the 8 point sequence x(n)={1,2,3,4,1,2,3,4} in bit reversed order.
In General order Bit Reverse Bit Reverse order
x(0) 0 0 0 0 0 0 x(0)
x(1) 0 0 1 1 0 0 x(4)
x(2) 0 1 0 0 1 0 x(2)
x(3) 0 1 1 1 1 0 x(6)
x(4) 1 0 0 0 0 1 x(1)
x(5) 1 0 1 1 0 1 x(5)
x(6) 1 1 0 0 1 1 x(3)
x(7) 1 1 1 1 1 1 x(7)
x(n) = { 1, 1, 3, 3, 2, 2, 4, 4 }
18. What are the differences between DIT and DIF algorithms?
 The DIT input is bit reversal while the output is in natural order, where as for DIF
input is in natural order while the output is bit reversal.
 The DIF butterfly is slightly different from the DIT butterfly. The difference
being that the complex multiplications takes place after the addition , subtraction
operation in DIF.
PART B
1. Find the DFT of the sequence of = {1, 1, 0, 0} (12marks)
Solution:
By the definition of Npoint DFT, the k
th
complex coefficient of X(k), is given by
2
1
0
( ) ( ) , 0,1,........., 1
j nk
N
N
n
X k x n e k N
t
=
= =
here N=4 , therefore the 4point DFT is
2
4 1
4
0
( ) ( )
j nk
n
X k x n e
t
=
=
2
3
4
0
( ) ( )
j nk
n
X k x n e
t
=
=
The values of X (k) can be varied for k=0, 1, 2, 3
When k=0
2 (0)
3
4
0
(0) ( )
j n
n
X x n e
t
=
=
3
0
0
(0) ( )
n
X x n e
=
=
(0) (0) (1) (2) (3) X x x x x = + + +
Where
x(0) =1
x(1) =1
x(2) =0
x(3) =0
(0) 1 1 0 0
(0) 2
X
X
= + + +
=
When k=1
2 (1)
3
4
0
(1) ( )
j n
n
X x n e
t
=
=
3
2
0
(1) ( )
j n
n
X x n e
t
=
=
(0) (1) (2) (3)
(1) (0) (1) (2) (3)
2 2 2 2
j j j j
X x e x e x e x e
t t t t
= + + +
0
3
(1) (0) (1) (2) (3)
2 2
j
j j
X x e x e x e x e
t
t t
= + + +
Where
x(0) =1
x(1) =1
x(2) =0
x(3) =0
0
3
(1) (1)* (1)* (0)* (0)*
2 2
j
j j
X e e e e
t
t t
= + + +
(1) 1 (cos sin ) 0 0
2 2
X j
t t
= + + +
X(1) = 1+0j
X(1) = 1j
When k=2
2 (2)
3
4
0
(2) ( )
j n
n
X x n e
t
=
=
3
0
(2) ( )
j n
n
X x n e
t t
=
=
(0) (1) (2) (3)
(2) (0) (1) (2) (3)
j j j j
X x e x e x e x e
t t t t
= + + +
0 2 3
(2) (0) (1) (2) (3)
j j j
X x e x e x e x e
t t t
= + + +
Where
x(0) =1
x(1) =1
x(2) =0
x(3) =0
0 2 3
(2) (1)* (1)* (0)* (0)*
j j j
X e e e e
t t t
= + + +
(2) 1 (cos sin ) 0 0 X j t t = + + +
X(2) = 11+0
X(2) = 0
When k=3
2 (3)
3
4
0
(3) ( )
j n
n
X x n e
t
=
=
3
0
3
(3) ( )
2
n
j n
X x n e
t
=
3 (0) 3 (1) 3 (2) 3 (3)
2 2 2 2
(3) (0) (1) (2) (3)
j j j j
X x e x e x e x e
t t t t
= + + +
3 9
0 3
(3) (0) (1) (2) (3)
2 2
j j
j
X x e x e x e x e
t t
t
= + + +
Where
x(0) =1
x(1) =1
x(2) =0
x(3) =0
3 9
0 3
(3) (1)* (1)* (0)* (0)*
2 2
j j
j
X e e e e
t t
t
= + + +
3 3
(3) 1 (cos sin ) 0 0
2 2
X j
t t
= + + +
X(3) = 1+0(j)
X(3) = 1+j
The 4point DFT sequence of x(n) is given by, X(k) = {2,1j,0,1+j}
2. Find the 8 point DFT of the sequence =
1, 0 < < 7
0,
(12marks)
Solution:
By the definition of Npoint DFT, the k
th
complex coefficient of X(k), is given by
2
1
0
( ) ( ) , 0,1,........., 1
j nk
N
N
n
X k x n e k N
t
=
= =
here N=8 , therefore the 8point DFT is
2
8 1
8
0
( ) ( )
j nk
n
X k x n e
t
=
=
2
7
8
0
( ) ( )
j nk
n
X k x n e
t
=
=
The values of X (k) can be varied for k=0, 1, 2, 3,4,5,6,7
When k=0
2 (0)
7
8
0
(0) ( )
j n
n
X x n e
t
=
=
7
0
0
(0) ( )
n
X x n e
=
=
(0) (0) (1) (2) (3) (4) (5) (6) (7) X x x x x x x x x = + + + + + + +
Where
x(0) =0
x(1) =1
x(2) =1
x(3) =1
x(4) = 1
x(5) = 1
x(6) = 1
x(7) = 0
(0) 0 1 1 1 1 1 1 0
(0) 6
X
X
= + + + + + + +
=
When k=1
2 (1)
7
8
0
(1) ( )
j n
n
X x n e
t
=
=
7
4
0
(1) ( )
j n
n
X x n e
t
=
=
(0) (1) (2) (3) (4) (5) (6) (7)
(1) (0) (1) (2) (3) (4) (5) (6) (7)
4 4 4 4 4 4 4 4
j j j j j j j j
X x e x e x e x e x e x e x e x e
t t t t t t t t
= + + + + + + +
0
3 5 3 7
(1) (0) (1) (2) (3) (4) (5) (6) (7)
4 2 4 4 2 4
j
j j j j j j
X x e x e x e x e x e x e x e x e
t
t t t t t t
= + + + + + + +
Where
x(0) =0
x(1) =1
x(2) =1
x(3) =1
x(4) = 1
x(5) = 1
x(6) = 1
x(7) = 0
0
3 5 3 7
(1) 0* 1* 1* 1* 1* 1* 1* 0*
4 2 4 4 2 4
j
j j j j j j
X e e e e e e e e
t
t t t t t t
= + + + + + + +
3 5 3
(1)
4 2 4 4 2
j
j j j j j
X e e e e e e
t
t t t t t
= + + + + + +
When k=2
2 (2)
7
8
0
(2) ( )
j n
n
X x n e
t
=
=
7
0
(2) ( )
2
n
j n
X x n e
t
=
(0) (1) (2) (3) (4) (5) (6) (7)
(2) (0) (1) (2) (3) (4) (5) (6) (7)
2 2 2 2 2 2 2 2
j j j j j j j j
X x e x e x e x e x e x e x e x e
t t t t t t t t
= + + + + + + +
(0) (1) (2) (3)
(2) (0) (1) (2) (3)
j j j j
X x e x e x e x e
t t t t
= + + +
0 2 3
(2) (0) (1) (2) (3)
j j j
X x e x e x e x e
t t t
= + + +
Where
x(0) =1
x(1) =1
x(2) =0
x(3) =0
0 2 3
(2) (1)* (1)* (0)* (0)*
j j j
X e e e e
t t t
= + + +
(2) 1 (cos sin ) 0 0 X j t t = + + +
X(2) = 11+0
X(2) = 0
When k=3
2 (3)
3
4
0
(3) ( )
j n
n
X x n e
t
=
=
3
0
3
(3) ( )
2
n
j n
X x n e
t
=
3 (0) 3 (1) 3 (2) 3 (3)
2 2 2 2
(3) (0) (1) (2) (3)
j j j j
X x e x e x e x e
t t t t
= + + +
3 9
0 3
(3) (0) (1) (2) (3)
2 2
j j
j
X x e x e x e x e
t t
t
= + + +
Where
x(0) =1
x(1) =1
x(2) =0
x(3) =0
3 9
0 3
(3) (1)* (1)* (0)* (0)*
2 2
j j
j
X e e e e
t t
t
= + + +
3 3
(3) 1 (cos sin ) 0 0
2 2
X j
t t
= + + +
X(3) = 1+0(j)
X(3) = 1+j
The 4point DFT sequence of x(n) is given by, X(k) = {2,1j,0,1+j}
3. Find the DFT of the sequence =
1, 0 2
0,
for N = 3 and N = 5. (12marks)
4. An 8point sequence is given by = 2, 2, 2, 2, 1, 1, 1, 1 Compute 8point
DFT of by radix2 DIFFFT. (12marks)
solution
i) Second stage of computation:
Phase factor (first stage)
0
8
1
8
2
8
3
8
1
0.707 0.707
0.707 0.707
W
W j
W j
W j
=
=
=
=
Output sequence of first stage of computation
{ } 3, 2, 2, 3,1, 0, 0, 0.707 0.707 j
ii) Second stage of computation:
Phase factor (Second stage)
0
4
1
4
1 W
W j
=
=
Output sequence of first stage of computation
{ } 5, 5,1, ,1, 0.707 0.707,1, 0.707 0.707 j j j
iii) Third stage of computation:
Phase factor (Third stage)
0
2
1 W =
Output sequence of first stage of computation
{ } 10, 0,1 ,1 , 0.293 0.707,1.707 0.707,1.707 0.707, 0.293 0.707 j j j j j j + + +
The sequence X (k) in bit reversed order
{ } 10, 0,1 ,1 , 0.293 0.707,1.707 0.707,1.707 0.707, 0.293 0.707 j j j j j j + + +
iv) The sequence X (k) in normal order
{ } 10, 0.293 0.707,1 ,1.707 0.707, 0,1.707 0.707,1 , 0.293 0.707 j j j j j j + + +
v) Therefore DFT,
{x (n)} = X (k) =
{ } 10, 0.293 0.707,1 ,1.707 0.707, 0,1.707 0.707,1 , 0.293 0.707 j j j j j j + + +
5. In 8 point sequence is given by x(n)=(2,1,1,2,1,1,1,1) compute 8 point DFT of x(n) by radix2
DITFFT (12marks)
6. i) ) Find 4 point DFT of the following signal x(n)=sin(n/2) (8marks)
ii) Compare the DIT and DIF radix2 FFT. (4marks)
UNITIII
PART A
1. Draw the general realization structure in direct formI of IIR system.
2. State the condition for the stability of digital filter.
The analog system function H(s) is stable if all its poles lies in the
left half of the splane similarly the digital system function H(z) is stable . if it
stratifies the following properties.
1. The j axis in the splane map into the unit circle in the zplane
2. The lest half plane of the splane should map into the inside of the unit circle in
the zplane thus a stable analog filter will be converted into a stable digital
filter.
3. Define IIR.
If the system is designed by choosing all the infinite sample of
impulse response then it is called IIR system.
+
+
z

1
z

1
z

1
z

1
z

1
z

1
+
+
+
+
+
X(z)
Y(z)
Z
1
X(z)
b
0
b
1
b
M1
b
M
a
1
a
N1
a
N
Z
2
X(z)
Z
M
X(z)
+
b
2
Z
(M1)
X(z)
Z
1
Y(z)
Z
2
Y(z)
+
a
2
Z
(N1)
Y(z)
Z
N
Y(z)
+
+
+
+
+
+
+
+
4. Mention any two procedures for digitizing transfer function of an analog
filter.
The two important procedures for digitizing the transfer functions
of analog filter are
1.impulse invariant method
2.bilinear transformation method
5. Compare the digital and analog filter.
Sl.no Digital filter Analog filter
1 Operates on digital samples of
the signal
Operates on analog
signals
2 It is defined by the linear
difference equation
It is defined by linear
difference
3 It consists of adder, multiplier
and delays implemented in
digital logic
It consists of electrical
components like resistors,
capacitors, inductors
4 The filter co efficient are
designed to statisfy the desired
frequency response
The approximation
problem is solved to
statisfy the desired
frequency response
6. Mention the important features of IIR filters.
1. The physically realizable IIR filter does not have linear phase.
2. The IIR filter specification includes the desired characteristics for the
magnitude response only.
7. How bilinear transformation is performed?
The bilinear transformation is performed by letting
1
1
2 1
1
s
T
=
+
Z
Z
In the analog filter transfer function
i.e H(Z) =H
a
(s)
1
1
2 1
1
s
T
=
+
Z
Z
8. What are the advantages of bilinear transformation?
1.The bilinear transformation provides one to one mapping.
2.Stable continuous systems can be mapped into realizable, stable digital systems
3.There is no aliasing.
9. List out the basic properties of IIR filters.
1.The imaginary axis in the splane should map into the unit circle in the zplane .
this is will be direct relationship between two frequency variables in the two
domain
2.The left half splane should map into the inter unit circle in the zplane . the
stable analog filter will be converted to a stable digital filter.
10. Classify the filters based on frequency response.
Based on the frequency response, the filters can be classified into the four types.
1.Low pass
2.High pass
3.Band pass
4.Band stop filters
11. How will you determine the order N of chebyshev filter?
Calculate a parameter N, using the following equation and correct it to nearest
integer
(
O
O
)
1
2 1
2
1
2
2
1
1
cosh
1
1 1
cosh
A
N
c
12. What are the steps involved in the design of digital IIR filter.
1.The specification of a digital filters are converted into the specification of
analog filter.
2.The analog filters is designed for the specification
3.The analog filter is converted to digital filters by transformation techniques.
13. Give the transformation used in the approximation of derivates.
The analog filter H(S) is transformed to digital filter H(Z) by
H(Z) = H(S)
1
1
s
T
=
Z
14. Write the properties of Butterworth filter.
1.The magnitude response of a Butterworth filter has a monotonic techniques flat
pass band and stop band
2.The poles of the Butterworth filter lies on a circle .
15. Distinguish between recursive and non recursive realization.
For recursive realization the present output y(n) is a function of
past output and present input . this form corresponds to an infinite impulse
response IIR digital filter
For non recursive realization the current output y(n) is a function of
only past and present input . this form corresponds to an finite impulse response
FIR digital filter.
16. What do you understand by backward difference?
One of the simplest method for converting an analog filter into a digital
filter is to approximate the differential equation by an equivalent differential
equation.
i.e d/dt .y(t) t =nT = y(nT) y(nTT)
T
= y(n) y(n1)
T
The above equation is known as backward difference
17. Give the magnitude function of Butterworth filter.
The magnitude function of Butterworth filter
1/ 2
2
1
( )
1
N
C
H jO =
 
  O
 +


O
\ .
\ .
N = 1,2,3
18. Sketch the mapping of Splan to Zplan in bilinear transformation.
Jv
J1
U
1 +1
J1
PART B
1. Convert the analog filter with transfer function H(s) into digital filter using
bilinear
transformation.
i) H(s) =
1 2 . 0
2
2
+ + s s
s
Solution:
Given that : H(s) =
1 2 . 0
2
2
+ + s s
s
Put
1
1
2 1
1
s
T
=
+
Z
Z
in H(s) to get H(z)
H(z) =
3
1
1
2
1 1 1
1 1 1
2 1
2
1
2 1 2 1 2 1
1 1
1 1 1
T
T T T
 

+
\ .
(
   
+ + + (
 
+ + +
( \ . \ .
Z
Z
Z Z Z
Z Z Z
=
( ) ( ) ( ) ( )
( )
1
1
2 2
1 1 1 1
2
1
4 1
1
4 1 0.4 1 1 1
1
T
T T
T
+
(
+ + + +
(
+
Z
Z
Z Z Z Z
Z
Put T=1 sec
Therefore
H(z) =
( )( )
( ) ( )( ) ( )
1 1
2 2
1 1 1 1
4 1 1
4 1 0.4 1 1 1
Z Z
Z Z Z Z
+
+ + + +
=
( )
( ) ( ) ( )
2
2 1 2 2 1
4 1
4 1 2 0.4 1 1 2
Z
z z Z Z z
+ + + + +
=
( )
2
2 1 2 2 1
4 1
4 4 8 0.4 0.4 1 2
Z
z z z z z
+ + + + +
=
( )
2
2 1
4 1
5.4 4.6 6
Z
z z
+
=
( )
2
2 1
4 1
6 5.4
4.6
4.6 4.6
Z
z z
 
+

\ .
H(z) =
( )
2
2 1
0.87 1
1.3 1.17
z
z z
+
ii) H(s) =
) 1 )( 1 (
2
3
+ + + s s s
s
Solution:
Given that H(s) =
) 1 )( 1 (
2
3
+ + + s s s
s
Put
1
1
2 1
1
s
T
=
+
Z
Z
in H(s) to get H(z)
Therefore
H(z) =
1
2 1 1
2.67( 1)
( 2 2.33)( 3)
Z
Z Z Z
+
=
3
1
3 1
1 1 1 2 1 1 1 2
1 1 2
8 1
1
2(1 ) (1 ) 4(1 ) 2(1 ) (1 ) ( (1 ))
(1 ) ( (1 ))
T
Z T Z Z Z T Z T Z
T Z T Z
 

+
\ .
( ( + + + + + +
( (
+ +
Z
Z
=
1 3
1 1 1 2 1 1 2 1 2
8(1 )
2(1 ) (1 ) 4(1 ) 2 (1 )(1 ) (1 )
Z
Z T Z Z T Z Z T Z
( ( + + + + + +
Put T=1 sec
Therefore
H(z) =
1 3
1 1 1 2 1 1 1 2
8(1 )
2(1 ) (1 ) 4(1 ) 2(1 )(1 ) (1 )
Z
Z Z Z Z Z Z
( ( + + + + + +
=
1 3
1 1 1 2 2 2 1
8(1 )
(2 2 ) (1 ) (4 8 4 2 2 1 2
Z
Z Z Z Z Z Z Z
( ( + + + + + + +
=
1 3
1 1 2
8(1 )
(3 ) (7 6 3
Z
Z Z Z
( ( +
=
3 1 3
2 1 1
8( 1) ( 1)
6 7
3( )( 1)( 3)
3 3
Z
Z Z Z
+
H(Z) =
1
2 1 1
2.67( 1)
( 2 2.33)( 3)
Z
Z Z Z
+
2. Convert the analog filter with system transfer function
(S+0.1)
i) H(s) = 
(S+0.1)
2
+9
SOLUTION:
The given system response of the analog filter is of the standard form
H(s) =
2 2
( )
s a
s a b
+
+ +
Where a = 0.1 ; b = 3
We can obtain the system response of the digital filter as under
H(z) =
1
1 2 2
1 (cos )
1 2 (cos )
at
at aT
e bT z
e bT z e z
+
..(1)
Substitute the values a and b in equation 1
H(z) =
1
1 2
1 (0.667)
1 1.326 (0.449)
z
z z
+
Taking T= 1s we have
H(z) =
0.1 1
0.1 1 0.2 2
1 (cos3)
1 2 (cos3)
e z
e z e z
+
=
1
1 2
1 (0.3678)(0.9986)
1 2(0.3678)(0.9986) (0.8187)
z
z z
+
H(z) =
1
1 2
1 0.3672
1 0.7345 0.8187
z
z z
+
(S+0.4)
ii) H(s) = 
(S+0.4)
2
+16
into a digital IIR filter by means of the impulse invariance technique.
.
Solution:
Where
a= 0.4; b = 4
H(z) =
( )
0.4 1
0.4 1 0.8 2
1 (cos 4 )
1 2 cos 4
T
T T
e T z
e T z e z
+
Where
t=1s
=
( )
0.4 1
0.4 1 0.8 2
1 (cos 4)
1 2 cos 4
e z
e z e z
+
H(z) =
1
1 2
1 (0.67)(0.99)
1 2(0.67)(0.99) (0.449)
z
z z
+
H(z) =
1
1 2
1 (0.667)
1 1.326 (0.449)
z
z z
+
3. Convert the analog filter with system transfer function H(s) =
) 2 )( 1 (
2
+ + s s
into digital filter by
means of the impulse invariance technique. If a) T= 1 sec and b) T= 0.1 sec
solution:
i) Formula :
Impulse invariant transformation
1
1
i
i i
p T
i
A A
s p e z
ii) A = 2
iii) B = 2
iv) To find H(z) [ General Form]
1 2 1
2 2
( )
1 1
T T
H z
e z e z
=
v) a) when T= 1 sec [ To find H(z)]
1 1 2 1
2 2
( )
1 1
H z
e z e z
=
1
1 1
0.465
( )
(1 0.3678 )(1 0.1353 )
z
H z
z z
=
vi) b) when T=0.1 sec [ To find H(z)]
0.1 1 0.2 1
2 2
( )
1 1
H z
e z e z
=
1
1 1
0.1722
( )
(1 0.9048 )(1 0.8187 )
z
H z
z z
=
4. Obtain the Direct formI realizations of the LTI system governed by the equation
y(n)= 0.5y(n1)0.25 y(n2)+x(n)+3x(n1) (12 marks)
5. Determine the direct form II realizations for the following system
y(n)=  0.1y(n1)+0.72y(n2)+0.7x(n)0.252x(n2) (12 marks)
Solution :
i) General form
0 1 2 1 2 3
( ) ( ) ( 1) ( 2) ................ ( 1) ( 2) ( 3) ........ y n b x n b x n b x n a y n a y n a y n = + + +
( ) 0.7 ( ) 0.252 ( 1) 0.1 ( 1) 0.72 ( ) y n x n x n y n y n = + +
ii) Formula
1 2
1 2
1 2
1 2
......
1 ........
o
b b z b z
a z a z
+ + +
+ + +
iii) To find y(n)
iv) To find w(n)
v) Draw direct form II structure
6. Convert the analog filter with system transfer function H(s) =
2 2
) 0325 . 0 ( ) 25 . 0 (
0325 . 0
+ + s
into a digital filter by means of the impulse invariance technique.
If a) T= 1 sec and b) T= 2 sec. (12 marks)
UNIT IV
PART A
1. List the well known design technique for linear phase FIR filter design.
There are three well known method of design techniques for linear phase FIR
filter.
They are:
1. Fourier series method and window method
2. Frequency sampling methods
3. Optional filter design methods.
2. What are the types of digital filter according to their impulse response?
Based on impulse response filter are of two types
IIR filters:
The IIR filter are of recursive type, where by the present output depends on
the present input , past input samples and output samples.
FIR filters:
The FIR filters are of non recursive type, where by the present output sample
depends on the present input , sample and previous input sample.
3. How phase distortion and delay distortion are introduced?
The phase distortion is introduced when the phase characteristics of a filter is not
linear within the desired frequency band .
The delay distortion is introduced when the delay is not a constant within the
desired frequency range.
4. Write the steps involved in FIR filter design.
1.Choose the desired (idea) frequency response H
d
()
2.Take inverse fourier transform of H
d
() to get h
d
(n)
3.Convert the infinite duration h
d
(n)to finite duration sequence h(n).
4.Take Z transform of h(n) to get the transfer function H(Z) of the FIR
filters.
5. Compare hamming window with Kaiser window.
Sl.no Hamming window Kaiser window
1 The width of main lobe in
window spectrum is 8/N
The width of main lobe in
window spectrum depends on the
values of and N
2 The max side lobe
magnitude in window
spectrum is fixed at 41dB
The maximum side lobe
magnitude with respect to peak
of main lobe is variable using the
parameter
3 In window spectrum the
side lobe magnitude
remain constant with
increasing
In window spectrum the side
lone magnitude decrease with
increasing
6. Draw the impulse response of an ideal low pass filter.
h
d
(n)
7. What are the advantages of FIR filter?
1. Linear phase FIR filters can be easily designed
2. Efficient realizations of FIR filter exist as both recursive and non recursive
structure.
3. FIR filters realized non recursively are always stable
4.The round off noise can be made small in non recursive realization of FIR
filters.
8. Draw the direct form realization of FIR system.
9. What is Gibbs phenomenon?
In FIR filter design by Fourier series method the infinite duration impulse
response is truncated to finite duration impulse response. The abrupt truncation of
impulse response introduces oscillators in the pass band and stop band . this effect is
known as Gibbs phenomenon.
10. Write the characteristic features of rectangular window.
1. The main lobe width is equal to 4/N
2. The maximum side lobe magnitude is 13dB
3. The side lobe magnitude does not decrease significantly with increasing
11. State the condition for a digital filter to be causal and stable.
1. The digital filter transfer function H(Z) should be a rational function of Z
and the co efficient of Z should be real.
2. The poles should lies inside the unit circle in Zplane.
3. The number of zeros should be less than or equal to number of poles.
12. When a cascade form realization is preferred in FIR filters?
X[z]
Y
[z]
b
o
b
1
b
2 b
N2
b
N1
Z
1
Z
1
Z
1
+ + + +
The cascade from realization is preferred when complex zeros with absolute
magnitude are less than one .
13. What are the properties of FIR filters?
1. FIR filter is always stable
2. A realizable filter can always be obtained
3. FIR filter has a linear phase response.
14. For what kind of applications, the anti symmetric impulse response can be used?
The ant symmetric impulse response can be used to design Hilbert transformers
and differences.
15. Draw the frequency response of Npoint rectangular window.
16. What is the response that FIR filter is always stable?
17. List the features of hamming window spectrum.
1. The mainlobe width is equal to 8/N.
2. The maximum side lobe magnitude is 41dB
3. The side lobe magnitude remains constant for increasing
18. What are the techniques of designing FIR filters?
1. Choose the desired frequency response of the filter H
d
()
2. Take inverse fourier transform of H
d
() to obtain the desired impulse
response h
d
(n).
3. Choose a window sequence W(n) and multiply h
d
(n) by W(n) to convert the
duration impulse response to finite duration impulse response h(n).
4.The transfer function H(z) of the filter is obtained by taking Ztransform of h(n).
PART B
1. With neat diagram explain the structure of FIR filter (12 marks)
2. Design a low pass filter using rectangular window by taking 9 samples of w (n) and with a
cutoff frequency of 1.2 radians/sec. (12 marks)
3. Determine the coefficients of a linear phase FIR filter of length M=15 has a symmetric
unit sample response and a frequency response that satisfies the condition.
H (2tk/15) = {1 ; for k=0, 1, 2, 3
0 ; for k=4, 5, 6, 7} (12 marks)
4. Draw the direct form structure of the FIR system described by the transfer function
H (Z) =1+1/2z
1
+3/4z
2
+1/4z
3
+1/2z
4
+1/8z
5
(12 marks)
5. Realize the following system with minimum number of multipliers
H (Z) =1/4+1/2z
1
+3/4z
2
+1/2z
3
+1/4z
4
(12 marks)
1. i) Explain the advantages and disadvantages of FIR filters.
ADVANTAGES: (6 marks)
1.the linear phase fir filters can be easily designed
2.FIR filter efficient realization exists as both recursive and non recursive structures
3.FIR filters realized non recursive are always stable .
4.the round of noise can be made in non recursive realization of FIR filters
DISADVANTAGES:
1. the duration of impulse response should be large to realize sharp cut of filters
2. the non integral delay can lead to problems in some signal processing applications
ii) Compare FIR and IIR filters. (6marks)
SL.NO IIR FIR
1 All the infinite samples of
impulse response are considered
Only n samples of impulse r
Response are considered
2 The impulse response cannot be dire
Directly converted to digital
Filter transfer function
The impulse response can be
Directly converted to digital
Filter transfer function
3 The specification include desire
Character for magnitude response
only
The specifications include the
Desired characteristics for both
And phase response
UNIT V
PART A
1. Define multi rate DSP.
In multi rate dsp system . the samples rates are changed or are different within
the system
2. Define sub band coding
Subband coding (SBC) is any form of transform coding that breaks a signal into
a number of different frequency bands and encodes each one independently. This
decomposition is often the first step in data compression for audio and video signals.
.
3. What is a QMF filter?
In digital signal processing, a quadrature mirror filter is a filter most commonly
used to implement a filter bank that splits an input signal into two bands. The resulting
highpass and lowpass signals are often decimated by a factor of 2, giving a critically
sampled twochannel representation of the original signal.
4. Define decimator.
In digital signal processing, decimation is a technique for reducing the number of
samples in a discretetime signal.
Decimation is a twostep process:
1. Lowpass antialiasing filter
2. Down sampling
5. Define interpolator.
Interpolation is a method of constructing new data points within the range of a
discrete set of known data points
6. What is down sampling?
In signal processing, down sampling (or "sub sampling") is the process of
reducing the sampling rate of a signal. This is usually done to reduce the data rate or the
size of the data.
7. What is sampling rate conversion?
Sample rate conversion is the process of converting a (usually digital) signal
from one sampling rate to another, while changing the information carried by the signal
as little as possible. When applied to an image, this process is sometimes called image
scaling
8. Give advantages of multirate DSP.
The advantages of multi rate DSP are
1.speech coding
2.image compression
3.adaptive equalization
4.echo cancellation
9. What is up sampling?
Up sampling is the process of increasing the sampling rate of a signal. For
instance, up sampling raster images such as photographs means increasing the resolution
of the image.
10. Define Periodiogram.
The periodiogram is an estimate of the spectral density of a signal
11. What is the need for antiimaging filter?
In a mixedsignal system (analog and digital), a reconstruction filter (or anti
imaging filter) is used to construct a smooth analogue signal from the output of a digital
to analogue converter (DAC) or other sampled data output device.
PART B
1. Briefly explain
(i)Interpolator
Interpolation is the exact opposite of decimation. It is an information preserving
operation, in that all samples of x[n] are present in the expanded signal y[n]. The
mathematical definition of Lfold interpolation is defined by Equation (1), and the block
diagram notation is depicted in Figure (1). Interpolation works by inserting (L1) zero
valued samples for each input sample. The sampling rate therefore increases from Fs to
LFs. With reference to Figure (1), the expansion process is followed by a unique digital
lowpass filter called an antiimaging filter. Although the expansion process does not
cause aliasing in the interpolated signal, it does however yield undesirable replicas in the
signals frequency spectrum. We shall see how this special filter, is necessary to remove
these replicas from the frequency
spectrum.
(1)
Figure (1): Block diagram notation of interpolation, by a factor of L.
(ii)Decimator
Decimation can be regarded as the discretetime counterpart of sampling.
Whereas in sampling we start with a continuoustime signal x(t) and convert it into a
sequence of samples x[n], in decimation we start with a discretetime signal x[n] and
convert it into another discretetime signal y[n], which consists of subsamples of x[n].
Thus, the formal definition of Mfold decimation, or downsampling, is defined by
Equation (1). In decimation, the sampling rate is reduced from Fs to Fs/M by discarding
M 1 samples for every M samples in the original sequence.
Figure 1: Block diagram notation of decimation, by a factor of M.
The block diagram notation of the decimation process is depicted in Figure (1).
An antialiasing digital filter precedes the downsampler to prevent aliasing from
occurring, due to the lower sampling rate. In Figure (2) below, it illustrates the concept of
3fold decimation i.e. M = 3. Here, the samples of x[n] corresponding to n = , 2, 1,
4, and n = , 1, 2, 5, are lost in the decimation process. In general, the samples of
x[n] corresponding to nkM, where k is an integer, are discarded in Mfold decimation. In
Figure 2 (b), it shows samples of the decimated signal y[n] spaced three times wider than
the samples of x[n]. This is not a coincidence. In real time, the decimated signal appears
at a slower rate than that of the original signal by a factor of M. If the sampling frequency
of x[n] is Fs, then that of y[n] is Fs/M.
.(1)
Figure 2: Decimation of a discretetime signal by a factor of 3.
(iii)Effects due to sampling rate conversion
A common use of multirate signal processing is for samplingrate conversion.
Suppose a digital signal x[n] is sampled at an interval T1, and we wish to obtain a signal
y[n] sampled at an interval T2. Then the techniques of decimation and interpolation
enable this operation, providing the ratio T1/T2 is a rational number i.e. L/M.
Samplingrate conversion can be accomplished by Lfold expansion, followed by
lowpass filtering and then Mfold decimation, as depicted in Figure (1). It is important to
emphasis that the interpolation should be performed first and decimation second, to
preserve the desired spectral characteristics of x[n]. Furthermore by cascading the two in
this manner, both of the filters can be combined into one single lowpass filter.
Figure (1): Samplingrate conversion by expansion, filtering, and decimation.
An example of samplingrate conversion would take place when data from a CD
is transferred onto a DAT. Here the samplingrate is increased from 44.1 kHz to 48 kHz.
To enable this process the noninteger factor has to be approximated by a rational
number:
Hence, the samplingrate conversion is achieved by interpolating by L i.e. from
44.1 kHz to [44.1x160] = 7056 kHz. Then decimating by M i.e. from 7056 kHz to
[7056/147] = 48 kHz.
2. Explain sub band coding with neat sketch.
SUB BAND CODING :
Subband coding (SBC) is any form of transform coding that breaks a signal into
a number of different frequency bands and encodes each one independently. This
decomposition is often the first step in data compression for audio and video signals.
Basic Principles
The utility of SBC is perhaps best illustrated with a specific example. When used
for audio compression, SBC exploits what might be considered a deficiency of the human
auditory system. Human ears are normally sensitive to a wide range of frequencies, but
when a sufficiently loud signal is present at one frequency, the ear will not hear weaker
signals at nearby frequencies. We say that the louder signal masks the softer ones. The
louder signal is called the masker, and the point at which masking occurs is known,
appropriately enough, as the masking threshold.
The basic idea of SBC is to enable a data reduction by discarding information
about frequencies which are masked. The result necessarily differs from the original
signal, but if the discarded information is chosen carefully, the difference will not be
noticeable
A basic SBC scheme
To enable higher quality compression, one may use subband coding. First, a
digital filter bank divides the input signal spectrum into some number (e.g., 32) of
subbands. The psychoacoustic model looks at the energy in each of these subbands, as
well as in the original signal, and computes masking thresholds using psychoacoustic
information. Each of the subband samples is quantized and encoded so as to keep the
quantization noise below the dynamically computed masking threshold. The final step is
to format all these quantized samples into groups of data called frames, to facilitate
eventual playback by a decoder.
Decoding is much easier than encoding, since no psychoacoustic model is
involved. The frames are unpacked, subband samples are decoded, and a frequencytime
mapping reconstructs an output audio signal.
Over the last five to ten years, SBC systems have been developed by many of the
key companies and laboratories in the audio industry. Beginning in the late 1980s, a
standardization body called the Motion Picture Experts Group (MPEG) developed
generic standards for coding of both audio and video. Subband coding resides at the heart
of the popular MP3 format (more properly known as MPEG 1 audio layer III), for
example.
3. Explain sampling rate conversion by the integer factor I. (12 marks)
4. Explain sampling rate conversion by the integer factor D. (12 marks)
5. Explain poly phase rectification.
POLYPHASE RECTIFICATION:
A polyphase quadrature filter, or PQF, is a filter bank which splits an input signal into
a given number N (mostly a power of 2) of equidistant subbands. These subbands are
subsampled by a factor of N, so they are critically sampled.
This critical sampling introduces aliasing. Similar to the MDCT time domain alias
cancellation the aliasing of polyphase quadrature filters is canceled by neighbouring sub
bands, i.e. signals are typically stored in two subbands.
Note that signal in odd subbands is stored frequency inverted.
PQF filters are used in MPEG1 Layer I and II, Muse pack (which was based on MPEG1
layer II), in MPEG1 Layer III with an additional MDCT, in MPEG4 AACSSR for the
4 band PQF bank, in MPEG4 V3 SBR for the analysis of the upper spectral replicated
band, and in DTS.
PQF has an advantage over the very similar stacked quadrature mirror filter (QMF).
Delay is much lower, computational effort is much lower.
A PQF filter bank is constructed using a base filter, which is a lowpass at fs/4N. This
lowpass is modulated by a N cosine functions and converted to N bandpasses with a
bandwidth of fs/2N.
The base lowpass is typically a FIR filter with a length of 10*N ... 24*N taps. Note that it
is also possible to build PQF filters using recursive IIR filters.
6. Briefly explain QMF filter with neat diagram. (12 marks)
QuadratureMirror Filter Bank
In many applications, a discretetime signal x[n] is split into a number of subband
signals by means of an analysis filter bank
The subband signals are then processed
Finally, the processed subband signals are combined by a synthesis filter bank
resulting in an output signal y[n]
If the subband signals are bandlimited to frequency ranges much smaller than
that of the original input signal x[n], they can be downsampled before processing
Because of the lower sampling rate, the processing of the downsampled signals
can be carried out more efficiently
After processing, these signals are then upsampled before being combined by the
synthesis filter bank into a higherrate signal
The combined structure is called a quadraturemirror filter (QMF) bank
If the upsampling and downsampling factors are equal to the number of bands,
then the structure is called a critically sampled filter bank
The most common application of this scheme is in the efficient coding of a signal
x[n]
TwoChannel QMF Bank
Figure below shows the basic twochannel QMF bankbased subband codec
(coder/decoder)
The analysis filters and have typically a lowpass and highpass frequency responses,
respectively, with a cutoff at /2
Each downsampled subband signal is encoded by exploiting the special spectral
properties of the signal, such as energy levels and perceptual importance
It follows from the figure that the sampling rates of the output y[n] and the input
x[n] are the same
The analysis and the synthesis filters are chosen so as to ensure that the
reconstructed output y[n] is a reasonably close replica of the input x[n]
Moreover, they are also designed to provide good frequency selectivity in order to
ensure that the sum of the power of the subband signals is reasonably close to the
input signal power
In practice, various errors are generated in this scheme
In addition to the coding error and errors caused by transmission of the coded
signals through the channel, the QMF bank itself introduces several errors due to
the sampling rate alterations and imperfect filters
We ignore the coding and the channel errors