EE8591 Digital Signal Processing Dept of EEE/EIE 2019-2020

UNIT I INTRODUCTION
PART- A
1. What do you understand by the terms: Signal and Signal Processing?
A signal is defined as any physical quantity that varies with time, space, or any other
independent variable. Signal processing is any operation that changes the characteristics of
a signal. These characteristics include the amplitude, shape, phase and frequency content of
a signal.
2. What are the classifications of signals?
There are five methods of classifying signals based on different features:
(a) Based on independent variable.
(i) Continuous time signal (ii) Discrete time signal.
(b) Depending upon the number of independent variable.
(i) One dimensional signal, (ii) Two dimensional signal.
(ii) Multi dimensional signal.
(c) Depending upon the certainty by which the signal can be uniquely described as
(i) Deterministic signal. (ii) Random signal.
(d) Based on repetition nature.
(i) Periodic signal. (ii) Non – Periodic signal.
(e) Based on reflection
(i) Even signal. (ii) Odd signal.
3. Define discrete system.
A discrete time system is a device or algorithm that operates on a discrete time input signal
x(n) , according to some well defined rule , to produce another discrete – time signal y(n)
called the output signal.
4. What are the classifications of discrete – time systems?
1. Static and Dynamic system. 2. Time – variant and time – invariant system.
3. Linear and non – linear system. 4. Stable and Un-stable system.
5. Causal and non-causal system. 6. IIR and FIR system.
5. Differentiate Continuous time and Discrete time signal.
Continuous time signal: It is also referred as analog signal i.e., the signal is represented
continuously in time.Discrete time signal :Signals are represented as sequence at discrete
time intervals .
n
y (n )= ∑ x(k)
6. Test whether the system governed by the relation k =−∞ is time –invariant
or not? (Dec 2014)
If the output is delayed by k units in time
y(n)=x(−∞)+.......... ...+x(0)+... ... .. x(n)
y(n−k)=x(−∞)+.. ...........+x(0)+........ x(n−k)−−−−(1)
If we delay the input by k units in time
y(n)=x(−∞)+.......... ...+x(0)+........ x(n)
y(n,k )=x(−∞)+... ...... ... .+x(0)+........ x(n−k)−−−−(2)
Equation (1)=(2). So, that the system is time invariant.
7. What is Deterministic signal and random signal? Give example.
A signal that can be uniquely determined by a well - defined process such as a mathematical
expression or rule , or look-up table is called a deterministic signal.
Example : A sinusoidal signal v (t )=V m sin ωt
A signal that is generated in a random fashion and cannot be predicted ahead of time is
called a “ randomsignal”.Example : Speech signal , ECG signal and EEG signal.
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8. Define (a) Periodic signal (b) Non – periodic signal.
Periodic signal: A signal x(n) is periodic with period N if and only if x(n+N)=x(n for all n.
Non – periodic signal: If there is no value of N that satisfies the above equation the signal
is called non-periodic or aperiodic.
9. Define symmetric and antisymmetric signals.
Symmetric signal: A real valued signal x(n) is called symmetric if x(-n) = x(n).
Antisymmetric signal: A signal x(n) is called antisymmetric if x(-n) = -x(n).
10. Differentiate energy and power signals? (May 2015) (May 2017)
Energy signal:
2
 The energy of a discrete time signal x(n) is defined as E= ∑|x (n )|
 A signal x(n) is called an energy signal if and only if the energy obeys the relation
0 <E< ∞ and an energy signal P=0.
Power signal :
 The average power of a discrete time signal x(n) is defined as
N
1
∑ |x ( n )|2 ¿
2N +1 n=− N .
 A signal x(n) is called power signal if and only if the average power P satisfies the
condition 0 <P< ∞ and E= ∞ .
11. Check if the system described by the difference equation y(n) = ay(n-1)+x(n) with
y(0) =1is stable. (May 2015)
On taking Z-Tranform we get
Y ( Z )= X ( Z ) +a Z −1 Y ( Z)

Y (Z) 1 Z
= H ( Z )= h ( n )=a n u( n)
X (Z) 1−a Z−1 Z−a
α α α

∑ |h(k )|= ∑ |b k
u (k )|∑ b k= 1 < ∞ 1−|b|
k=−α k=−α k=0

This term is less than infinity and hence the system is stable.
12. What are the different types of operations performed on discrete – time signals?
The different types of operations performed on discrete – time signals are
(1)Delay of a signal (2) Advance of a signal (3)Folding or Reflection of a
signal
(4) Time scaling (5) Amplitude scaling (6)Addition of signals
(7)Multiplication
13. What is a static and dynamic system?
A discrete –time system is called static or “memory less” if its output at any instants ‘n’
depends on the input samples at the same time , but not an past or future samples of the
input.
Ex.,y(n) = ax(n) Y(n)=ax2(n)
In any other case, the system is said to be dynamic or to have memory.
Ex.,y(n) = ax(n-1)+x(n-2) y(n)=x(n)+x(n-1)
14. What is a time – invariant system?
A system is called time – invariant if its input – output characteristics do not change with
time.
Ex.,y(n)=x(n)+x(n-1)
15. What is a causal system?
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A system is said to be causal if the output of the system at any time n depends only on
present and past inputs, but does not depend on future inputs.
This can be expressed mathematically as,
y(n)=F[x(n), x(n-1), x(n-2)………]
16. Define a stable system?
Any relaxed system is said to be bounded input-bounded output (BIBO) stable if and only if
every bounded input yields a bounded output. Mathematically, their exist some finite
numbers, Mx and My such that,
|x (n)|≤Mx< ∞ and|y ( n )|≤My<∞
17. What do you meant by sampling process? (Dec 2012)
Sampling is the conversion of a continuous –time signal (or analog signal) into a discrete –
time signal obtained by taking samples of the continuous time signal (or analog signal) at
discrete time instants.
18. State Shannon’s sampling theorem. (May 2014) (May 2017)
A band limited continuous time signal with highest frequency (band width) f mhertz , can be
uniquely recovered from its samples provided that the sampling rate f s is greater than or
equal to 2fm samples per second.
19. Define Nyquist rate. (June 2012)(Nov 2018)
The Nyquist rate or frequency is the minimum rate at which a finite bandwidth signal needs
to be sampled to retain all of the information. For a bandwidth of span fm Hz, the Nyquist
frequency is 2fm Hz.
20. What is aliasing effect? How can aliasing be avoided? (Dec 2014)(Nov 2017)(May
2018)
The superimposition of high frequency component on the low frequency is known as
“frequency aliasing” or “aliasing effect”.To avoid aliasing the sampling frequency must be
greater than twice the highest frequency present in the signal.
21. What is a linear time invariant system? (Dec 2012)
An LTI system is one which possess both Linearity and Time- invariance.
A system is linear if y1(n) = T[x1(n)] and y2(n) = T[x2(n)]
then T[a1 x1(n)+a2 x2(n)] = a1 y1(n)+ a2 y2(n)
22. What is the Nyquist rate for the signal xa(t)=3cos 600πt+2cos1800πt? (Dec 2013)
Solution: ω1=600π ω2=1800π
2πf1= 600π 2πf2= 1800π
f1= 300Hz f2= 900Hz
Nyquist rate Fs=2fm= 2x900= 1800Hz.
π 30 n
23. Determine fundamental period of the signal Cos (105 ) .(Dec 2013)
2π
Solution: Fundamental period, N=
( )
,ωo
m

30 π 105
ωo= m,
Where 105 = 15
when m=1 & N = 7 periods.
24. Given a continuous time signal x(t)= 2cos500πt. What is the Nyquist rate and
fundamental frequency of the signal? (June2013)
ω=500π , 2πf= 250π
f= 250Hz
hence Nyquist rate Fs=2fm= 2x250= 500Hz.
25. Determine whether x[n]=u[n] is a power signal or an energy signal. (June2013)
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The energy of a discrete time signal x(n) is defined as E= ∑ |x (n )|2 = ∞

The average power of a discrete time signal x(n) is defined as
N
1
∑ |x ( n )|2 ¿
2N +1 n=− N = 0.5
Here E= ∞ and P= Finite. Therefore the given signal is a power signal.
26.Given a Continuous signal x(t) =2 cos300πt.What is the nyquist rate and
fundamental frequency of the signal.(Dec 2015)
ω=300π ,
2πf= 150π
f= 150Hz hence Nyquist rate Fs=2fm= 2x150= 300Hz.
27.What is an Anti –Aliasing filter?(May 2016)
An anti-aliasing filter (AAF) is a filter used before a signal sampler to restrict the
bandwidth of a signal to approximately or completely satisfy the sampling theorem over the
band of interest
28. Distinguish between discrete signal and discrete signal representation.(Dec 2016)
Discrete time signals: Discrete time signals are the signals or quantities that can be defined
and represented at certain time instants of the sequence.

Discrete signal representation:A discrete time signal can be represented in (i)Tabular

representation (ii) Graphical representation (iii) Sequence representation
29.If x(n) = x(n+1)+x(n-2), is the system causal? (Dec 2016)
When n=0; x(0)= x(1)+x(-2). When n= -1, x(-1) = x(0)+x(-3). The system is non causal,
since the system depends on future.
30. Determine x(n) =u(n) is a power signal or energy signal. (Dec 2015)

1
31.Determine if the system described by the equation x ( n )+ is causal or non
x (n−1)
causal.(May 2016)

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32. List the sampling techniques.(May 2018)

There are three types of sampling techniques,
i) Impulse sampling,
ii) Natural sampling
iii) Flat top sampling.
33. Define spectral density.(Nov 2018)
The spectral density of the signal describes the energy or power present in the signal as a
function of frequency, per unit frequency. The power spectral density has infinite energy
and finite power.
PART– B
1. Determine which of the following signals are periodic and determine the fundamental
( i ) x ( t )=20 sin 25 πt ( ii ) x ( t )= 20sin √ 5 t
π
( iii ) x ( t )=10 cos 10 πt ( iv ) x( t )=3 cos( 5 t + )
period also. 6
2.Explain the digital signal processing system with necessary sketches and give its merits
and
demerits.State the advantages of convolution technique. (Dec 2016)
3. i) Find the impulse response of a Discrete Time LTI system
y (n )= y ( n−1)+0 .5 y (n−2)+ x ( n )+ x( n−1 )
(May 2015)
(ii)What is meant by sampling? Explain sampling theorem? What is spectral density (Dec
2012) (Dec 2014)
ii) Explain the properties of discrete time system?(May 2015)
4)i)A discrete time system is represented by difference equation verify whether it is linear?
Shift Invariant? Causal? In each case justify your answer. (May 2015)
y (n )=3 y (n−1 )−nx (n )+4 x (n−1)+2 x (n+1) and n≥0 .
ii) What is meant by quantization and quantization error ? (Dec2014) (May 2017)
5. Check whether following are linear, time invariant, causal and stable for the following
system,
(i)y(n) = x(n)+nx(n+1) , (ii) y(n) = cos x(n), (iii) y(n)= x(-n=5), (iv) y(n)= x(2n)(Dec 2011)
(June2012) (May 2017)
6. What is causality, stability and dynamic of a system? Derive the necessary and sufficient
condition on the impulse response of the system for causality and stability.(Dec 2012)(Dec
2014)

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7. What is meant by energy and power signal? Determine whether the following signals are
energy or power or neither energy nor power signals.
1n π
x1(n) = 2 ()
u(n )
, x2(n) = sin 6
n ( )
, x3(n) = e
πn π
j( + )
3 6
,
1n
2n
x4(n) = e u(n) , x5(n) =
()
5
u(n )
,x6(n) = e
πn π
j( + )
3 7
(Dec 2012) (Dec 2014)(Jun 2013)
8. A discrete time systems can be (i) Static or Dynamic, (ii) Linear or Non-Linear, (iii)
Time invariant or time varying &(iv) Stable or Unstable. Examine the Following system
with respect to the properties above y(n)= x(n)+ nx(n+1) (Dec 2013)
9. Given y[n]= x[n2]. Determine whether the system is linear, time invariant, memoryless
and causal. (June 2013)
10. i) Check the causality and stability of the systems y (n )=x (−n)+ x (n−2)+ x (2n−1)
ii) Check the system for linearity and time invariance y(n )=(n−1) x 2 (n )+c (May 2014)
11.(i) Check the causality and stability of the systems
y (n )=x (−n)+x (n−2)+x (2n−1)
(ii)Check the system for linearity and time variance
y (n )=(n−1) x (n)+C (Dec 2015)
12.What is meant by energy and power signal ? Determine whether the following signals
are energy or power or neither energy nor power signals
1n
(i)x 1 ( n)= ()
2
u(n )

π
(ii) x 2 (n )=Sin n
6( ) (Dec 2015)
State and prove sampling theorem.
13. (i) Determine if the signals x 1(n) and x2(n) are power , energy or neither energy nor
power signals. (May 2016)
1 n
x 1 (n )=()
3
u(n) 2n
and x 2 (n )=e u (n)
(ii)What is the input signal x(n) that will generate the output sequence
Y(n)={1,5,10,11,8,4,1} for a system with impulse response h(n) {1,2,1}.
14.(i) A signal x(t)= sin c(50πt) is sampled at a rate of (1) 20 Hz (2) 50 Hz and (3) 75 Hz.
For Each of these cases explain if you can recover the signal x(t) from the samples signal.
(ii) Determine whether or not each of the following signals is periodic .If the signal is
periodic specify its fundamental period(May 2016)
j 16 πn
(1) x(n)=e
π 3π
x (n)=cos n+Cos n
3 4
(2)

15.Distinguish the following with examples and formulae (Dec 2016)

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(i) Energy and power signal

(ii) Time variant and time- invariant signal

16. Compute the Nyquist sampling frequency of the signal x(t)= 4 sinc (3t/π). (May 2017)

17. i) Determine the power and energy of the given signal. State the signal is power or

energy (Nov 2017)

nπ
x (n)=sin
4
ii) Determine the given signal is periodic or not
2 πn
x (n)=Cos
3
iii) Discuss the mathematical representation of signal. Write the difference between
continuous and discrete time signal.
18. i) Determine whether the system is linear or not y(n) = ax(n) + bx(n-1) (Nov 2017)
ii) Determine whether the given system is causal or not y(n) = x(n) + x2(n-1)
iii) Determine whether the system is time invariant and stability. y(n) = ex(n)
19. Explain the classification of continuous time signals with its mathematical
representation.(May 2018)
20.Describe the different types of system and write the condition to state the system with its
types. (May 2018)
21.i) Illustrate the condition of the system to be causal and linearity. Check the same for the
given system (Nov 2018)
1
y ( n )=x ( n ) +
x( n−1)
ii) Check the time invariant and stability of the given system y(n) = cosx(n)
22. i) Determine the value of power and energy of the given signal. (Nov 2018)
π
x(n)=sin −1
4 ( )
ii) Explain the types of signals with its mathematical expression and neat diagram.

UNIT II -DISCRETE TIME ANALYSIS

Part A
1. Define Z – transform.
The Z –transform of a discrete time signal x(n) is denoted by X(z) and it is defined as,
∞
X ( z)= ∑ x (n )z −n
n=−∞

Where z is a complex variable and n is the sequence interval. x(n) and X(z) is called z-

transform pair.
2. What is meant by region of convergence? (Dec 2011) (Dec 2012)

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The region of convergence (ROC) of X(z) is the set of all values of z for which X(z) attains
a finite value. ROC for a discrete time signals x(n) is defined as a continuous region in z
plane where the z-transform converges.
3. What are the properties of region of convergence? (Dec 2012) (May 2017)
(i) The ROC is a ring or disk in the Z – plane centered at the origin.
(ii) The ROC cannot contains any poles.
(iii) The ROC of an LTI stable system contains the unit circle.
(iv) The ROC must be a connected region.
4. What are the properties of z- transform ? (Dec 2011) (Dec 2013)
Linearity: z [ a1 x 1( n)+a2 x 2 (n)] =a1 X 1 ( z )+a 2 X 2 ( z )
m−1

Shifting: (a)
−m
[
z [ x ( n+m ) ] =z m X ( z )− ∑ x (i )z m−i
i=0 ]
(b) z [ x ( n−m) ] =z X ( z )
d m
Multiplication:
z [ n x (n )]= −z m
dz
X ( z) ( )
n
Scaling in z- domain: z [ a x ( n) ] =X ( a z )
−1

−1
Time reversal : z [ x (−n) ] =X ( z ) (May 2012)
Conjugation: z [ x ( n ) ] =X ( z )
n

Convolution:
z
[∑
m=0 ]
h ( n−m )r ( m ) =H ( z ) R( z )
(Dec 2013)
Initial value: X ( z)¿
Final value: ( 1−z−1 ) X ( z )¿
5. State Parseval’s relation in z-transform.(Dec 2013,Nov 2017)
If x1(n) and x2(n) are complex valued sequences, then
1 1 −1
∑
n→∞
x1 ( n) x 2 ( n )= ∮
2πj c
X 1( v ) X 2
v()
v dv

6. What is the relationship between z-transform and DTFT?(May 2012)(Nov 2018)

∞
X ( z)= ∑ x (n )z −n
The z-transform of x(n) is given by n=−∞ ……. (1)
jω
Where z=re
Substituting z value in eqn (1) we get,
X (re jω )=∑ x ( n )r−n e− jωn
………………….. (2)
The Fourier transform of x(n) is given by
∞
X (e jω )= ∑ x (n)e− jωn
…………………… . (3)
n=−∞
Eqn (2) and Eqn (3) are identical, when r=1. In the z-plane this corresponds to the
jω
locus of points on the unit circle |z|=1. Hence X (e ) is equal to X(z) evaluated along
unit circle , or
X ( e jω )=X ( z )|z=e jω
jω
For X (e ) to exist, the ROC of X(z) must include the unit circle.
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7. Find the convolution of the following using z- transform. (May 2017)
, 2,1 ¿ ¿ ¿
Z { x ( n)∗h( n ) } =X ( z ) H ( z )
X ( z )=( 1 +z−1 +z−2 )
H ( z )=( 1+ 2z−1 +z−2 )
X ( z ) H ( z )=( 1+ 3z−1 + 4z−2 +3z−3 +z− 4 )
Solution: x ( n)∗h( n )= {1,3,4,3,1 }
8. Define system function.
Let x(n) and y(n) is the input and output sequences of an LTI system with impulse response
h(n). Then the system function of the LTI system is defined as the ratio of Y(z) and X(z),
i.e.,
Y (z )
H ( z )=
X ( z ) where,
Y(z) is the z – transform of the output signal y(n)
X(z) is the z – transform of the input signal x(n)
9. Define Fourier transform of a discrete time signal.
The Fourier transform of a discrete time signal x(n) is defined as
∞
F { x (n ) } =X (ω )= ∑ x (n )e− jωn
n=−∞
10. Why FT of a discrete time signal is called signal spectrum?
By taking Fourier transform of a discrete time signal x(n) , it is decomposed into its
frequency components . Hence the Fourier transform is called signal spectrum.
11. Define inverse Fourier transform.
The inverse Fourier transform of X(ɷ) is defined as
π
1
−1
F { X (ω ) } =x( n )=
2π
∫ X ( ω ) e jωn dω
−π
12. What is the frequency response of LTI system?
The Fourier transform of the impulse response h(n) of the system is called frequency
response of the system. It is denoted by H(ɷ).
13. Write the properties of frequency response of LTI system.
i) The frequency response is periodic function ɷwith a period of 2 π .
ii) If h(n) is real then |H ( ω )| is symmetric and ∠H ( ω ) is antisymmetric.
iii) If h(n) is complex then the real part of H (ω) is antisymmteric over the interval
0≤ω≤2π .
iv) The frequency response is a continuous function of ɷ.
14. Define discrete Fourier series (or) Define discrete fourier series representation for
a periodic sequence. (Dec 2014)
Consider a sequence xp(n) with a period of N samples so that x p(n)=xp(n/N); Then the
discrete Fourier series of the sequence xp(n) is defined as
N−1
X p (k )= ∑ x p (n)e− j2πkn/ N
n= 0

15. What are the two basic differences between the Fourier transform of a discrete
time signal with the Fourier transform of a continuous time signal?
1. For a continuous signal, the frequency range extends from −∞ to+∞ . On the other
hand, the frequency range of a discrete – time signal extends from −πto+π ( or 0 to 2π ) .

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2. The Fourier transform of a continuous signal involves integration , whereas , the Fourier
transform of a discrete – time signal involves summation process.
16.Write the commutative and distributive properties of convolution. (Dec 2011)
Commutative Property: x(n)*h(n)= h(n)* x(n)
Distributive property: x(n)*[h1(n)+h2(n)]=[ x(n)* h1(n)]+[ x(n)* h2(n)]

17. Find the Fourier transform of a sequence x(n) = 1, for −2≤n≤2

= 0, otherwise.
∞ 2
X ( ω )= ∑ x ( n ) e− jωn= ∑ e− jωn
Solution: n=−∞ n=−2

=e +e +1 +e− jω +e− j2ω

j2ω jω

=1+2cos ω+2cos2ω
18. Determine the Z-transform and ROC for the signal x(n)=δ(n-k)+ δ(n+k). (Dec
2013)
−k +k
Solution: X(Z)= Z + Z , X(Z) will converge for all the values of Z, except Z = 0 and
∞ .
19. Given a difference equation y(n)= x[n]+3x[n-1]+2y[n-1]. Determine the system
function H(Z). (May 2013)
Solution: On taking Z- Transform, Y(Z)= X(Z)+3Z-1 X(Z)+2Z-1 Y(Z)
Y(Z)[1-2Z-1]= X(Z)[1+3Z-1]
Y ( Z ) 1+3Z-1
= -1
H(Z)= X (Z ) 1-2Z
1n
20. Find the stability of the system whose impulse response h(n)=
()
2
u(n )
. (May
2013)
h(n) = (1/2)n u(n)
Take z-transform on both sides,.
z
H ( z) = ;|z|<1
1
z−
2
The ROC is |z| < 1. It is within unit circle. Therefore, the system is stable.
21. Define DTFT pair for a discrete sequence. (Dec 2012)
The discrete time fourier transform of a discrete time signal x(n) is defined as
+∞
∑
F{x(n)}=X()= n=−∞ x(n) e-jn
+∞
∑
The discrete time fourier transform exists only if n=−∞ x(n)<

The inverse discrete time fourier transform of X() is defined as

1 π
∫ X (ω) ejnd
F-1{X()}= x(n) = 2 π −π
22.What is the draw back in Discrete Time Fourier transform and how it is overcome?
The drawback in Discrete TimeFourier transform(DTFT) is that it is a continuous function
of ω and so it cannot be processed by digital system. This drawback is overcome by using
Discrete Fourier Transform(DFT). The DFT converts the continuous function of ω to a
discrete functionof ω.
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n
23. Determine the Z-transform of x(n)=a (May 2015)
Solution:
∞
X( z)= ∑ x(n)z −n
n=−∞
α
X ( Z )= ∑ ¿¿
N =−α

1
X ( Z )=
1−az−1

Z
X ( Z )=
Z−a

24. Determine the Fourier transform of the signal x(t)=sinω0t.(May 2015)

Solution:
1−e− jω cos ω 0
X ( ω )= ¿¿

25. Find the Z-transform and ROC of the discrete time signal
x(n)=−an u(−n−1),
a>0(Dec 2014)

X(Z)=
¿∞¿
∞
=− ∑ (a−1 Z )n
n=1
1
¿−
[ −1
1−a Z
−1
]
−Z
¿
Z−a
26.What is ROC of Z Transform? State its properties.(Dec 2015)
The region of convergence (ROC) is the set of points in the complex plane for which
the Z-transform summation converges.
Properties:
 The ROC for a finite duration sequence includes entire z-plane
except z=0z=0 and/or |z|=∞.
 ROC does not contain any poles
 ROC is the ring in the z-plane centered about origin.

27.State initial and final value theorem of Z transform. (Dec 2015)

Initial value theorem for causal signal u(0) = lim z→∞ U(z) if the limit exists.
The final value theorem for z-transforms states that if lim k→∞ x(k) exists, then lim k→∞
x(k) = limz→1 (z − 1)X(z).
28.Determine the Z Transform and ROC of the following finite duration signals( May
2016)

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(i) x(n)= {3,2,2,3,5,0,1}

(ii)x(n)= Ϩ(n-k)

(i) x(n)= {3,2,2,3,5,0,1}

x(0) =3; x(1)=2; x(2)=2; x(3)=3; x(4)=5; x(5)=0; and x(6)=1
By the definition of Z-transform,
∞
Z {x(n)} = X(Z )= ∑ x( n)Z −n
n=−∞

The given sequence is a finite sequence defined in the range n=0 to 6, hence the limits of
summation is changed to n=0 to 6.
6
Z {x (n )} = X(Z )= ∑ x (n )Z−n
n=0

= x(0) z0 + x(1) z-1+ x(2) z-2+ x(3) z-3+ x(4) z-4+ x(5) z-5+ x(6) z-6
= 3+2 z-1+2 z-2 +3 z-3 +5 z-4 +0 z-5+1 z-6
= 3 + 2/Z+2/Z2 + 3/Z3+5/Z4+ 1/Z6
In X(z) , when z=0, except the first terms all other terms will become infinite. Hence X(z)
will be finite for all values of z, except z=0. Therefore , the ROC is entire z-plane except
z=0.

(ii)x(n)= Ϩ(n-k)

The above signal is a left sided or anticausalsequence and its Z- transform can be obtained
with:
0
Z {x (n )} = X(Z )= ∑ x (n )Z−n
n=−k
.

The ROC of this signal is :

29.Compute the convolution of the two sequences ( May 2016)

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X(n)={2,1,0,0,5} and n(n) = {2,2,1,1}

30.What is the relationship between s-plane and z-plane? (Dec 2016)

The s-plane and the z-plane are related by a conformal mapping specified by the analytic
complex function  z = es = eσ+jw = eσ . ejw = r ejw. Where, Re{s}= σ and Im{s}= jw and |z|= r
= eσ and <z= w.
31. Find the system transfer function H(Z)if y(n)= x(n)+y(n-1) (Dec 2016)
Y(Z)=X(Z)+ Z-1Y(Z)
Y(Z)- Z-1Y(Z) = X(Z)
Y(Z)[1- Z-1]=X(Z)
Y(Z)/X(Z)=1/[1- Z-1]
H(Z)= 1/[1- Z-1]
32. List the methods to find inverse z transform.
i) Long division method
ii) Partial fraction method
iii) Residue method
iv) Convolution method
33. Write the conditions to define stability in ROC.
∞

For Stable, −∞
∑|h(n)|<∞
An LTI system is stable if and only if its impulse response is absolutely summable, i.e., the
frequency response function H ( e jω ) exits, i.e. the ROC of its transfer function   H ( z ) includes
the unit circle |z|=1.

34. What is the inverse z transform of

2z
H ( z) =
1 (May 2017)
z−
2

h(n)=2(1/2)n u(n)
n
35. Find the stability of the system whose impulse response h(n)= ( 2 ) u(n ) . (May
2018)
h(n) = 2n u(n)
Take z-transform on both sides,.
z
H ( z) = ;|z|>1
z−2
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The ROC is |z| > 1. It does not contain unit circle. Therefore, the system is unstable.

PART – B
1.Find the Z- Transform of following :(Dec 2013) (May 2017)
(a)x(n )=sin(nω 0 )u(n)
(b)x(n )=cos(ωo n)u(n )
2. Determine the causal signal x(n) having the Z- Transform(Dec 2013)
1+ Z−1
X ( Z )= −1
(a) 1-Z +0 .5Z−2
1
X ( Z )=
(b) 1−Z +0 . 5Z−2
−1

1+2z−1 +z−2
X ( Z )=
(c) 1+4Z−1 +4Z−2
n
3. i) Determine the DTFT of the given sequence x(n)=a (u(n)−u(n−8)),|a|<1
ii) Prove the linearity and frequency shifting theorem of the DTFT (May 2014)
1
4. Using the Z- Transform determine the response y[n] for n≥0 if y[n] = 2 y[n-1]+x[n],
1
( )n
x[n]= 3 u[n], y(-1)=1. (June 2013)
5. Evaluate the frequency response of the system described by the system function
1
−1
H(Z)= 1−0 .5 z . (June 2013)
6. Find the Z- transform and its associated ROC for the following discrete time signal x[n]=
1 1
( )n ( )−n
3 u[n]+5 2 u[-n-1] (June 2013)
n
7. i) Find the Z-transform and ROC of x(n)=r cos(nθ)u(n ). (Nov 2017) (May 2018)
z
X ( Z )= 2
ii) Find the inverse Z-transform of 3 z −4 z+1 , ROC |z|>1 (May 2014)
(May 2015)
2
8. i) Find the Z- Transform of following x(n)=n u (n) (May 2015)
ii) Obtain the linear convolution of x 1 [ n ] =[0,1,4 ,−2] ∧ x 2 [ n] = [ 1,2,2,2] (May 2015)
iii) Find the frequency response of LTI system governed by the equation
y (n ) = a1 y (n−1 )−a 2 y (n−2 )−x (n ) (May 2015)
9.(i) Find the Z transform and its ROC of x(n)=(1/2)n+(-1/2)n(Dec 2014)
(ii) Find the linear convolution of x(n)={1,2,3,4,5,6,7} with h(n)={2,4,6,8}.
10.(i) What is frequency response? Explain its properties. (Dec 2014)
4Z
2
(ii) Find the inverse z –transform of X(Z)= (Z +1) (Z +3 ) for all possible ROCs.
z
X ( z )=
11. Find the inverse Z transform of 3 z 2−4 z +1 ROC |Z| >1(Dec 2015)
12.Using Z Transform determine the response y(n) for n≥0 if

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n
y (n )= ( 12 ) y (n−1)+x (n) ,
( 13 ) u(n) y(−1) (Dec 2015)
x (n)=
1
1+ Z −1
2
x (n)ifX ( z)=
1
1− Z−1
13.(i) Find 2
(ii) Find the response of the causal system y(v) – y(n-1) =x(n) + x(n-1) to the input x(n) =
u(n) Test its stability
14. What is the need for frequency response analysis? Find the impulse response , frequency
response ,magnitude response and phase response of the second order system

(i) y(n) = 2x(n)+x(n-1)+y(n-2)

15.Evaluate the following: (Dec 2016)

(i) The impulse response h(n) for y(n)= x(n)+2x(n-1)-4x(n-2)+x(n-3)

(ii) The ROC of a finite duration signal x(n)= {2,-1,-2,-3,0,-1}

(iii)Inverse Z-Transform for X(Z)= 1/(z-1.5)4; ROC: |z|>1/4

15. State and prove convolution and parseval’s theorem using Z transform. (May 2017)

(May 2018)

16. Find the inverse Z transform of

( x+1)
X(Z)= ( x+0.2)( x−1 )
|z|>1 using residue theorem. (May 2017)

17. State and prove any three properties of Z transform. (Nov 2017)

18. i) A discrete system has a unit sample response (Nov 2017)

h(n)=1/ 2 δ (n)+δ (n−1)+1/2 δ (n−2)

Find the system frequency response.

ii) Find the convolution of the two sequence x(n) = {1,2,-1,1} and h(n) = {1,0,1,1} using
graphical method.

19. i) Find the circular convolution of the two sequences x1(n) = {1,2,2,1} and x2(n) =
{1,2,3,1} (May 2018)
ii) How do you obtain the magnitude and phase response of DTFT.
20. Determine the frequency response H(e jw) for the given system and plot magnitude and
1
phase response , y ( n ) + y ( n−1 )=x ( n ) + x (n−1) (May 2018)
4
21. Determine the impulse response of the given difference equation (May 2018)

y ( n )= y ( n−1 )+ 0.25 y ( n−2 ) + x ( n )+ x( n−1)

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Plot the pole zero pattern and check its stability.

z3 + z 2
22. Find the inverse z transform of X ( z )= . ROC |z|>3 (Nov 2018)
( z−1 )( z−3 )

23. Find the frequency response for the given sequence and plot the magnitude and phase
response (Nov 2018)

H d (ω)=¿ {1 ,for n=−2,−1,0,1,2¿¿¿¿

24. Compute the reponse of the system y ( n )=0.7 y ( n−1 )−0.12 y ( n−2 ) + x ( n−1 ) + x (n−2) to
input x(n) = nu(n). Is the system stable. (Nov 2018)

UNIT –III-- DISCRETE FOURIER TRANSFORMATION & COMPUTATION

PART-A
1. Define the Discrete Fourier transformation of a given sequence x(n).
The N- point DFT of a sequence x(n) is
N−1
X (k )= ∑ x(n )e− j2πkn /N
n= 0 k= 0, 1, 2……..N-1.
2. Write the formula for N- point IDFT of a sequence X(k).
The N-point IDFT of a sequence X(k) is
N−1
1
x ( n)=
N
∑ X (k ) e j2πk /N
K=0 n= 0, 1 , 2 …..N-1 .
3. List any four properties of DFT.
(a) Periodicity
If X(k) is N- point DFT of a finite duration sequence x(n)then
x ( n+N ) =x( n ) foralln
X ( k+N )=X ( k ) forallk .
(b)Linearity
If X1(k)=DFT[x1(n)] and
X2(k)=DFT[x2(n)]
then DFT[a1x1(n)+a2x2(n)]=a1X1(k)+a2x2(k)
( c ) Time reversal of a sequence
If DFT {x(n)}=X(k),
ThenDFT{x((-n))N}=DFT{x(N-n)}=X((-k))N=X(N-k)
(d)Circular time shifting of a sequence
If DFT {x(n)}=X(k),
− j2πkl / N
ThenDFT{x((n-l))N}=X(k) e
4. If N-point sequence x(n) has N- point DFT X(k) then what is the DFT of the
following?
j2π ln/ N
(i) x (n )( ii) x ( N −n )(iii ) x ((n−l))N (iv ) x(n ) e
( i ) DFT { x ( n) } =X ( N−k )
( ii) DFT { x ( N −n) } =X ( k )
( iii ) DFT { x (( n−l ))N } =X ( k ) e− j2πkl/ N

Solution: ( iv ) DFT { x ( n) e j2π ln/ N } =X( ( k−l )) N

n
1
5. Calculate the DFT of the sequence x(n) =
( ) forN=16
4
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Solution:
N−1
X (k )= ∑ x(n )e− j2πkn /N
n= 0 K=0, 1 , 2…..N-1
15 n
1
¿∑
n=0
( )e
4
− j2πkn / 16

15 n
∑ (14 e− jπk / 8)
n= 0
1 16 − j2πk
1−
4
e()
1
1− e− jπk /8
4

6. Find the DFT of the sequence x(n)={1,1,0,0} (May 2015)(May 2018)(Nov 2018)
Solution:
N−1
X (k )= ∑ x(n )e− j2πkn /N
n= 0 K=0, 1 , 2…..N-1

3 3 3
−jπn/2 −jπn −j3πn/2
3X(1)=∑x(n)e ={1−j+0+ }=1−jX(2)=∑x(n)e ={1− +0 }=0X(3)=∑x(n)e ={1+j 0+ }=1+jX(k)={2,1−j,01+j}
n=0 n=0 n=0
7. State Circular frequency shifting property of DFT. (May 2014) (Dec2015)(Nov 2018)
If DFT[x(n)]=X(k),
e j2π ln/ N ]=X (( k−l ))N
Then DFT[x(n)
Thus shifting the frequency components of DFT circularly is equivalent to multiplying the
j2π ln/ N
time domain sequence by e
8. What is zero padding? What are its uses? (Dec 2014) (Dec 2016)(May 2018)
The process of lengthening the sequence by adding zero – valued samples is called
appending with zeros or 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.
9. What are the steps involved in circular convolution.
The circular convolution involve basically four steps as the ordinary linear
convolution. These are 1.Folding the sequence 2. Circular time shifting the folded
sequence3.Multiplying the two sequences to obtain the product sequence. 4. Summing the
values of product sequence
10. Obtain the circular convolution the following sequences.(June 2012)
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x(n)={0,1,0,2 }; h(n)={ 2,0,1 }
Solution:
The circular convolution of the above sequences can be obtained by using matrix method.
h( 0 ) h( 2) h( 1) x ( 0 ) y(0 )

[
1 2 −2 1 3
][ ] [ ]
h( 1) h( 0 ) h( 2) x ( 1) = y ( 1)
h( 2) h( 1) h( 0) x ( 2) y ( 2)

[
−2 1 2
2 2 1 ][ ] [ ]
y ( n )= {3,2,−1 }
2 =2
1 −1

11. State the difference between (i) overlap-add method (ii) overlap – save method.
Sl.No Overlap – add method Overlap – save method
1. In this method the size of the In this method the size of the
input data block is N=L+M-1 input data block is L.
2. Each data block consists of the Each data block is L points and
last M-1 data points of the we appended M-1 zeros to
previous data followed by the L compute N-point DFT.
new data points.
3. In each output block M-1 In this no corruption due to
points are corrupted due to aliasing as linear convolution is
aliasing , as circular performed using circular
convolution is employed. convolution.
4. To form the output sequence To form the output sequence, the
the first M-1 data points are last M-1 points from each output
discarded in each output block block is added to the first (M-1)
and the remaining datas are points of the succeeding block.
fitted together.
12. What is FFT?(Dec 2012)
The term Fast Fourier Transform (FFT) usually refers to a class of algorithms for
efficiently computing the DFT.It makes use of the symmetry and periodicity properties of
K
twiddle factor W N to effectively reduce the DFT computation time.
It is based on the fundamental principle of decomposing the computation of DFT of a
sequence of length N into successively smaller discrete Fourier transforms. The FFT
algorithm provides speed increase factors , when compared with direct computation of the
DFT, of approximately 64 and 205 for 256 points and 1024 – point transforms respectively.
13. How many multiplications and additions are required to compute N-point DFT
using radix-2 FFT?(May2011)
The number of multiplications and additions required to compute N-point DFT using radix-
N
N log 2 Nand log 2 N
2 FFT are 2respectively.
14. What is the speed improvement factor in calculating 64 – point DFT of a sequence
using direct computation and FFT algorithms?(OR)Calculate the number of
multiplications needed in the calculation of DFT and FFT with 64-point sequence.
The number of complex multiplications required using direct computation is
N 2 =642 =4096
The number of complex multiplications required using FFT is
N 64
log 2 N= log 2 64=192
2 2

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4096
=21 .33
192
Speed improvement factor=
15. What is decimation – in – time algorithm? (Dec 2013)
The computation of 8 – point DFT using radix-2 FFT , involves three stages of
computations. Here N=8=23, therefore r=2 and m=3.
The given 8 – point sequence is decimated to 2- point sequences . For each 2 – point
sequence, the 2-popint DFT is computed . From the result of 2 – point DFT the 4 – point
DFT can be computed. From the result of 4-point DFT , the 8 – point DFT can be
computed.
16.What are the difference between and similarities between DIT and DIF algorithms?
(May 2014)
Difference between DIT and DIF:
1. In DIT , the input is bit-reversed while the output is in natural order. For DIF , the reverse
is true, i.e., input is normal order, while the output bit is reversed. However , both DIT and
DIF can go from normal to shuffled data or vice versa.
2.Considering the butterfly diagram, in DIF , the complex multiplication takes place after
the add – subtract operation.
Similarities:
1. Both algorithms require same number of operations to compute DFT.
2. Both algorithm require bit – reversal at some place during computation.
17. Draw the basic butterfly diagram for Radix 2 DITFFT. (Dec 2011)(Dec 2013)(May
2015)(Nov 2017)

18. In eight point decimation in time(DIT), what is the gain of the signal path that goes
from x(7) to X(2)? (Dec 2013)
From the signal flow diagram of Radix -2 eight point DIT FFT, the signal path from x(7) to
X(2) have gain as follows,
0 0 2
Gain from x(7) to X(2) = −W 8 W 8 W 8 =− j

19. Define circular convolution.

The convolution property of DFT says that , the multiplication of the DFTs of the two
sequence is equivalent to the DFT of the circular convolution of the two sequences.
X 1 (k ) X 2 (k )=DFT { x1 (n)⊗ x 2 (n ) }
N−1
x 3 (n )= ∑ x 1 (m )x 2 ((n−m ))N
m=0
20. Find the discrete Fourier Transform for δ[n]. (May 2013)
N−1
X (k )= ∑ x(n )e− j 2π nk/ N
n= 0
δ ( n )=1 for n=1 ,

δ ( n )=0 Otherwise
N−1
X (k )= ∑ δ(n )e− j 2π nk/ N
n= 0
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21.Draw the basic butterfly diagram for Radix 2 DIF FFT.(June2013)

22. What is the relation between Z-transform & DFT? (May 2011)
Let N-point DFT of x(n) be X(k) and the Z-transform of x(n) be X(z).The N-point
sequence X(k) can be obtained from X(z) by evaluating X(z) at N equally spaced points
around the unit circle.
i.e., X(k )=X(z)| j
2πk ;for k=0,1,2,......(N−1)
N
z=e
23. What are the steps involved in computing IDFT through FFT?
1. Take conjugate of x(k)
2. Compute N point DFT of complex conjugate x*(k) using FFT.
3. Take again the conjugate of the output sequence.
4. Then the resultant sequence is divided by N.
24. State parsavel’s relation of DFT? (Dec 2014)
If x[k]x[k] and X[r]X[r] are the pair of discrete time Fourier sequences, where x[k]x[k] is
the discrete time sequence and X[r]X[r] is its corresponding DFT. Prove that the energy of
the aperiodic sequence x[k]x[k]of length NN can be expressed in terms of its N-point DFT
as follows:
Ex=∑k=0N−1|x[k]|2=1N∑rN−1|X[r]|2.
25.Draw the flow graph of a 4 point radix-2 DIT FFT butterfly structure for DFT .
(May 2016)

26.What are the applications of FFT algorithm? (May 2016)

 Some of the important applications of FFT includes
 Fast large integer and polynomial multiplication
 Efficient matrix-vector multiplication for Toeplitz, circulant and other structured
matrices
 Filtering algorithms
 Fast algorithms for discrete cosine or sine transforms (example, Fast DCT used for
JPEG, MP3/MPEG encoding)
 Fast Chebyshev approximation
 Fast discrete Hartley transform
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 Solving difference equations
27.Calculate the percentage saving in calculation in a 256 point radix – 2FFT when
compared to direct FFT. ( Dec 2015) (May 2017)

28. Define twiddle factor. Write its magnitude and phase angle. (May 2017)
The twiddle factor, WN = e-j2π/N, describes a "rotating vector", which rotates in increments
according to the number of samples,N.The magnitude of twiddle factor is 1. The phase
angle is given by -2π/N.It lies on the unit circle in the complex plane from 0 to 2π and it
gets repeated for every cycle.
29. Find the DFT of the signal x(n) = an. (Nov 2017)

x(n) = an
N −1 − j 2 πkn
X ( K ) = ∑ an e N
for 0 ≤ k ≤ N −1
n=0

1−a N e− j 2 πk
X ( k )= − j 2 πk
N
1−a e

PART-B
1. State and prove any four properties of DFT.(May 2012) (Dec 2012)(Dec 2014)(Nov
2017)(May 2018)
2.(i)Develop a Radix-2, 8-point DIF FFT algorithm with neat flow chart
(ii) Determine the DFT of the sequencex(n)={1/4,for 0≤n≤2. (May 2015)(Dec 2012)(Dec
2014).
0,otherwise
3. A finite duration sequence of length L is given as
x(n)=¿ (1, 0≤n≤L−1 (
(0, otherwise Determine N-point DFT of this sequence for N ≥ L 
4. By means of the DFT & IDFT, determine the sequence x3 (n) corresponding to the
circular convolution of the sequence x1 (n) and x2(n).
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x1(n) = {2,1,2,1}, x2 (n) = {1,2,3,4}
5. (i) Find 4 – point DFT of the following sequences
(i )x (n )= {1,0,−1,2 }( ii )x (n )= {2,4,3,2 }
nπ
(iii ) x ( n )=2n ( iv) x ( n)=cos
4
(ii) Determine the IDFT of the following:
( i ) X ( k )={ 1,1,− j2,−1,1+j2 } ( ii) X ( k )= {1,0,1,0 }
( iii ) X ( k )={ 1,−2− j, 0,−2+j }
6.(i)Develop a Radix-2, 8-point DIT- FFT algorithm with neat flow chart

(ii)Compute the DFT for the sequence for N=4 if x(n) = sin 2 ( nπ )
using decimation – in –
time algorithm.
7. An 8-point sequence is given by x(n); x(n) = (1, 1, 1, 1, 1, 1, 1, 1).Compute 8 point DFT
of x(n) by radix -2 DIF-FFT.(Dec 13)(Dec 2014)
8. Find the output y[n] of a filter whose impulse response is h[n] = {1,1,1} and input signal
x[n] = {3, -1, 0, 1, 3, 2, 0, 1, 2, 1} using overlap save method. (May 2013)
9. i) Find the X(K) for x(n) = n+1 ,for N = 8 using DIT FFT algorithm.(May 2015)
ii) Use four point inverse FFT for the DFT result { 6 , −2+ j 2 , −2 , −2− j2 } and determine
the
input sequence.
10.Obtain 8-point DFT of the sequence x(n) = (1, 1, 1, 1).(Dec 2014)
11.(i)The first five points of the eight point DFT of a real valued sequence
are{0.25,0,0.125- j0.3018,0,0.125-j0.0518}.Determine the remaining three points.
(ii)Compute the eight point DFT of the sequence x= {0,1,2,3,4,5,6,7} using DIF FFT
algorithm. (Dec2015)
12.(i) Find the inverse DFT of X(K)={7,-√2-j√2,-j,√2-j√2,1,√2+j√2,j,-√2+j√2}.
(ii)Using FFT algorithm compute the DFT of x(n)={2,2,2,2}(Dec 2015)
13.(i)Summarize the steps of radix -2 DIT FFT algorithm.
(ii)Compute the 4 point DFT of the sequence x(n)={0,1,2,3} using DIT and DIF algorithm. (May
2016)
14.Find the IDFT of the sequence
X(K)={4,1-j2.414,0,1-j0.414,0,1+j0.414,0,1+j2.414} using DIF algorithm (May 2016)
15.Describe the need for bit reversal and butterfly structure. For a sequence x(n)={4,3,2,1,-
1,2,3,4} obtain 8-point FFT by using DIT method.(Dec 2016)
16.Compute 8 point DFT of the sequence {1,1,1,1,0,0}. (May 2017)
17. Compute 8 point DFT of the given sequence using DIT algorithm. (May 2017)

x(n)=¿ {n ,n≤7¿¿¿¿
18. The analog signal has a bandwidth of 4 KHz. If we use N point DFT with N=2m (m is
an integer) to compute the spectrum of the signal with resolution less than or equal to 25
Hz. Determine the minimum sampling rate, minimum number of required samples and
minimum length of the analog signal. What is the step size required for quantize the signal
(May 2017)
19. Determine the DFT of the following sequence x(n) = {5,-1,1,-1,2} (Nov 2017)
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20. Find the DFT of the sequence x(n) = {1,2,3,4,4,3,2,1} using DIT-FFT algorithm. (Nov
2017)
21. Determine the DFT of the given sequence x(n) = {1,-1,-1,-1,1,1,1,-1} using DIT-FFT
algorithm.(May 2018)
22. Determine the DFT of the given sequence x(n) = {1,1,1,1,1,1,1,0} using DIT-FFT
algorithm.(Nov 2018)
23. Find the DFT of the sequence x(n) = {1,2,3,4,4,3,2,1} using DIF-FFT algorithm.(Nov
2018)
UNIT IV--DESIGN OF DIGITAL FILTERS
PART-A
1. What are the different types of structures for realization of IIR systems?
The different types of structures for realization of IIR system are
(i) Direct form I structure (ii) Direct form II structure
(iii) Cascade form structure (iv) Parallel form structure
(v) Lattice – ladder form structure.
2. What are the different types of filters based on impulse response?
Based on impulse response the filters are of two types
1. IIR filter
2. FIR filter
The IIR filters are of recursive type, whereby the present output sample depends on
the present input, past inputs samples and output samples.
The FIR filters are of non recursive type whereby the present output sample is
depends on the present input sample and previous input samples.
3. What is the general form of IIR filter?
The most general form of IIR filter can be written as
M

b z k
k

H(z)  k 0 N
1 ak
k 1

4. Give the magnitude of Butterworth filter. What is the effect of varying order of N
on magnitude and phase response?
The magnitude function of the Butterworth filter is given by
1
|H( jΩ)|= N=1,2,3..............
2N 1

[ ( )]
1+ Ω
Ωc
2

Where N is the order of the filter and Ωc is the cut off frequency. The magnitude
response of the Butterworth filter closely approximates the ideal response as the order
N increases . The phase response becomes more non-linear as N increases.
5. What is Type –1 and Type –2 Chebyshev approximation?
(i)In type –1 Chebyshev approximation, the error function is selected such that, the
magnitude response is equiripple in the pass band and monotonic in the stop band.
(ii)In type –2 Chebyshev approximation, the error function is selected such that, the
magnitude response is monotonic in pass band and equiripple in the stop band. The Type -2
magnitude response is called inverse Chebyshev response.
6. Write the magnitude function of Chebyshevlowpass filter?
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The magnitude response of Type -1 lowpassChebyshev filter is given by
1
|H a ( Ω )|=
Ω

where ε
√ 1 +ε 2 C2N Ω
is attenuation constant and
( )c

CN Ω
Ωc( )
is the Chebyshev polynomial of the first kind of degree N.
7. What are the properties of Chebyshev filter? (May 2013)(Dec 2014)
1. The magnitude response of the Chebyshev filter exhibits in ripple either in pass
band or in the stop band according to the type.
2. The magnitude response approaches the ideal response as the value of N
increases.
3. The Chebyshev type – 1 filters are all pole designs.
4. The poles of Chebyshev filter lies on an ellipse.
8. Compare the Butterworth and Chebyshev Type -1 filter.(Dec 2016)
Sl.N Butterworth filter Chebyshev filter
o
1. All pole design All pole design
2. The poles lie on a circle in s- The poles lie on a ellipse in s-plane
plane
3. The magnitude response is The magnitude response is equiripple in
maximally flat at the origin and pass band and monotonically decreasing in
monotonically decreasing the stop band.
function of Ω .
4. The normalized magnitude The normalized magnitude response has a
1 1
response has a value of value of √2 √1 +ε 2
at the cut off frequency
at the cut off frequency Ω c . Ωc .
5. Only few parameters has to be A large number of parameter has to be
calculated to determine the calculated to determine the transfer
transfer function. function.
9. Distinguish between FIR and IIR filter. (May 2012)
Sl.No FIR filter IIR filter
1. These filters can be easily These filters do not have linear phase.
designed to have perfectly
linear phase.
2. FIR filters can be realized IIR filters are easily realized recursively.
recursively and non –
recursively.
3. Greater flexibility to control the Less flexibility, usually limited to specific
shape of their magnitude kind of filters.
response.
4. Error due to round off noise are The rounds off noise in IIR filters are more.
less severe in FIR filters, mainly
because feedback is not used.

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10. What are the properties of FIR filter?(or) What are the advantages of FIR filter.
(Dec 2012)
FIR filter is always stable because all its poles are at the origin.
1. A realizable filter can always be obtained.
2. FIR filter has a linear phase response.
11. What are the steps involved in the FIR filter design?
1. Choose the desired (ideal ) frequency response H d (ω ) .
2. Take inverse Fourier transform of H d (ω ) to get hd (n ) .
3. Convert the infinite duration hd (n ) to a finite duration sequence h( n ) .
4. Take Z – transform of h( n ) to get the transfer function H ( z ) of the
FIR filter.
12. How the constant group delay and phase delay is achieved in linear phase FIR
filters.
Frequency response of FIR filter with constant group and phase delay
H (ω )=±|H ( ω)|e j ( β−αω )
The following conditions have to be satisfied to achieve constant group and phase delay.
N −1
α=
Phase delay, 2 ( i.e., phase delay is constant )
π
β=±
Group delay, 2
( i.e., group delay is constant)
Impulse response, h(n) = - h( N -1 – n ) (i.e., impulse response is anti symmetric )
13. Write the condition for stability of digital filter.(May 2012)
1. Choose the desired (ideal ) frequency response H d (ω ) of the filter.
2. Evaluate the Fourier series co-efficient of H d (ω ) which gives the desired impulse
π
1
hd ( n )=
2π
∫ H d (ω )e jωn dω
response hd (n ) . Where −π

Truncate the infinite sequence hd (n ) to a finite duration sequence h( n ) .

3.Take Z – transform of h( n ) to get a non causal filter transfer function H ( z ) of the
FIR filter.
− ( N 2−1 )
4.Multiply H ( z ) by z to convert noncausal transfer function to a realizable
causal FIR filter transfer function.
N−1

H ( z )=
z
−
N −1
( 2
[
) h ( 0)+ 2
∑ h( n )( z n +z−n )
n= 1
14. What is Gibbs phenomenon? (May 2012) (Dec 2012) (Dec 2016)
]
jω
One possible way of finding an FIR filter that approximates H (e ) would be to
N−1
truncate the infinite Fourier series at n=
±
2 [ ]
.The abrupt truncation of the series will
lead to oscillation both in passband and in stopband. This phenomenon is known as Gibbs
phenomenon.
15. What is window and why it is necessary?(Dec 2012)(Dec 2016)
jω
One possible way of finding an FIR filter that approximates H (e ) would be to truncate
N−1
the infinite Fourier series at n=
±[ ] 2 . The abrupt truncation of the series will lead to
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oscillation both in passband and in stopband. These oscillations can be reduced through the
use of less abrupt truncation of the Fourier series. This can be achieved by multiplying the
infinite impulse response with a finite weighing w ( n) , called a window.
16.Give the equation specifying Hamming windows. (May 2014)
The equation for Hamming window is given by
2πn ( N−1 ) ( N −1)
w H ( n )=0 .54 +0 . 46cos − ≤n≤
N−1 for 2 2
=0 Otherwise.

17. Compare the Rectangular and Hanning window.

Sl.No Rectangular window Hanning window
1. The width of mainlobe in The width of mainlobe in window spectrum
4π 8π
window spectrum is N
is N
2. The maximum sidelobe The maximum sidelobe magnitude in
magnitude in window spectrum window spectrum is -31dB.
is -13dB.
3. In window spectrum the In window spectrum the sidelobe magnitude
sidelobe magnitude slightly decreases with increasing ω
decreases with increasing ω
4. In FIR filter designed using In FIR filter designed using Hanning
rectangular window the window the minimum stopband attenuation
minimum stopband attenuation is 44dB.
is 22dB.
18. Compare the Hamming and Blackman window.
Sl.N Hamming window Blackman window
o
1. The width of mainlobe in The width of mainlobe in window spectrum
8π 12 π
window spectrum is N
is N
2. The maximum sidelobe The maximum sidelobe magnitude in
magnitude in window spectrum window spectrum is -58dB.
is -41dB.
3. In window spectrum the In window spectrum the sidelobe magnitude
sidelobe magnitude remains decreases rapidly with increasing ω
constant with increasing ω
4. In FIR filter designed using In FIR filter designed using Blackman
Hamming window the window the minimum stopband attenuation
minimum stopband attenuation is 78dB.
is 51dB.
5. The higher value of sidelobe The higher value of sidelobe attenuation is
attenuation is achieved at the achieved at the expense of
expense of constant attenuation increasedmainlobe width.
at high frequencies.
19 .Compare the Hamming and Kaiser window.
Sl.No Hamming window Kaiser window
1. The width of mainlobe in The width of mainlobe in window spectrum
depends on the values of ‘a’ and N. Here ‘a’
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8π is a variable parameter introduced to modify
window spectrum is N the characteristics of window and N is the
order of the filter.
2. The maximum sidelobe The maximum sidelobe magnitude with
magnitude in window spectrum respect to peak of mainlobe is variable using
is -41dB. the parameter ‘a’.
3. In window spectrum the sidelobe In window spectrum the sidelobe magnitude
magnitude remains constant with decreases with increasing ω
increasing ω
4. In FIR filter designed using In FIR filter designed using Kaiser window
Hamming window the minimum the minimum stopband attenuation is
stopband attenuation is 51dB. variable and depends on the value of ‘a’.
20.Mention any two procedures for digitizing the transfer function of an analog filter.
(May 2013) (Dec 2016)
The two important procedures for digitizing the transfer function of an analog filter
are
1. Impulse invariance method.2. Bilinear transformation method.
21. What is frequency warping?(May 2012) (Dec 2012)(Nov 2018)
In bilinear transformation the relation between analog and digital frequencies is
nonlinear. When the s-plane is mapped into z-plane using bilinear transformation , this
nonlinear relationship introduces distortion in frequency axis, which is called frequency
warping.
22. Explain the technique of prewarping. (May 2012)
In IIR filter design using bilinear transformation the specified digital frequencies are
converted to analog equivalent frequencies, which are called prewarp frequencies. Using the
prewarpfrequencies, the analog filter transfer function is designed and then it is transformed
to digital filter transfer function.

23. Compare the impulse invariant and bilinear transformations. (Dec 2011)
Sl.No Impulse Invariant transformation Bilinear transformation
1. It is many – to – one mapping It is one – to – one mapping.
2. The relation between analog and The relation between analog and digital
digital frequency is linear. frequency is nonlinear.
3. To prevent the problem of aliasing There is no problem of aliasing and so the
the analog filters should be band analog filter need not be band limited.
limited.
The magnitude and phase response Due to the effect of warping, the phase
4. of analog filter can be preserved by response of analog filter cannot be
choosing low sampling time or preserved .But the magnitude response can
high sampling frequency. be preserved by prewarping.
24. The impulse response of an analog filter is shown in below figure. Let h(n)= h a(nT).
Where T=1sec. Determine the System function. (Dec 2013)

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nt , 0<t <5
Solution :ha(nT)= { 10−nT , 5<t<10
5 10
∑ nZ−n +∑ (10−n )Z−n
H(Z) = n=0 n=5

25.Comment on the pass band and stop band characteristics of butter worth filter.
(May 2015)
(i)The frequency response of the Butterworth filter is maximally flat (i.e. has no ripples) in
the passband and rolls off towards zero in the stopband.
(ii)When viewed on a logarithmic Bode plot, the response slopes off linearly towards
negative infinity.
(iii)Butterworth filters have a monotonically changing magnitude function with ω, unlike
other filter types that have non-monotonic ripple in the passband and/or the stopband.
26.Define pre-wraping effect? Why it is employed?(Dec 2015) (May 2017)(Nov 2017)
The effect of the non linear compression at high frequencies can be compensated. When the
desired magnitude response is piecewise constant over frequency, this compression can be
compensated by introducing a suitable rescaling or prewar ping the critical frequencies.
27.Obtain the cascade realization for the system function (May 2016)

H ( z )=
(1+ 14 z )
−1

(1+ 12 z−1
)(1+ 1a z + 14 z
−1 −2
)

28. Mention the advantages of FIR filters over IIR filters. (May 2016)
FIR filters are more powerful than IIR filters, but also require more processing power and
more work to set up the filters. They are also less easy to change "on the fly" as you can by
tweaking (say) the frequency setting of a parametric (IIR) filter. However, their greater
power means more flexibility and ability to finely adjust the response of your active
loudspeaker.
29.Why are digital filters more useful than analog filters?(Dec 2016)
Digital filters have the following advantages compared to analog filters:
 Digital filters are software programmable, which makes them easy to build and test.
 Digital filters require only the arithmetic operations of addition, subtraction, and
multiplication.
 Digital filters do not drift with temperature or humidity or require precision
components.
 Digital filters have a superior performance-to-cost ratio.
 Digital filters do not suffer from manufacturing variations or aging.
30. Write the advantages and disadvantages of digital filters. (May 2017)

Advantages:
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 Digital filters are software programmable, which makes them easy to build and test.
 Digital filters require only the arithmetic operations of addition, subtraction, and
multiplication.
 Digital filters do not drift with temperature or humidity or require precision
components.
 Digital filters have a superior performance-to-cost ratio.
 Digital filters do not suffer from manufacturing variations or aging.
Disadvantages
 In reality, the signal bandwidth of the digital sequence is much lower than the analog
sequence.
 Finite word-length effect, which results quantizing noise and round-off noise, is
another major drawback during computation.
 It needs much longer time to design and develop the digital sequence.
31. Draw the direct form I structure for 3rd order system. (Nov 2017)
Consider a 3rd order system transfer function,
p0 + p1 z−1+ p 2 z−2 + p3 z −3
H ( z) =
1+ d 1 z−1 +d 2 z−2+ p3 z−3

The direct form I structure of IIR filter is given by,

32. What is meant by radix-4 FFT? (May 2018)

For a radix-4 FFT, the value of sequence N should be such that, N=2 m, so that the N-point
sequence is decimated into 4-point sequences and 4-point DFT of each decimated sequence
is computed. Here ‘m’ is the number of stages.
33. Obtain the direct form-I realization for the given difference equation (May 2018)
y(n) = 0.5y(n-1)-0.25y(n-2)+x(n)+0.4x(n-1)

Y (z ) 1+0.4 z −1
H ( z) = =
X (z) 1−0.5 z−1 +0.25 z−2

34. Obtain the direct form-I realization for the given difference equation (Nov 2018)
y(n )=−0 . 1 y(n−1)+0 .2 y( n−2 )+3 x (n )+3.6 x(n−1 )+0 .6 x (n−2)

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EE8591 Digital Signal Processing Dept of EEE/EIE 2019-2020
Y (z ) 3+3.6 z + 0.6 z
−1 −2
H ( z) = =
X ( z) 1+0.1 z −1 −0.2 z −2

PART – B
1. Obtain the Direct form –I, Direct form – II, Cascade form and Parallel form structure for
the system described by (Dec 2012)

(a ) y(n )=−0 .1 y(n−1)+0.2 y(n−2 )+3 x(n )+3 .6 x(n−1)+0. 6 x( n−2 )

(b ) y(n )=0 .5 y(n−1 )+0 .25 y(n−2)+x(n)+x(n−1)
(c) y(n) - 2y(n-1) + y(n+2) = x(n) +0.5 x(n-1) (Dec 2016)
2. Design a chebyshev filter for the following specification using (a) bilinear transformation
(b) Impulse invariance method. (May 2018)
0.8≤ |H(ejω| ≤ 1 0≤ ω≤ 0.2π
jω
|H(e | ≤ 1 0.6π≤ ω≤ π
3. Design (a) a Butterworth and (b) a Chebyshev analog high pass filter that will pass all
signals of radian frequencies greater than200rad/sec with no more than 2 dB attenuation
and have a stop band attenuation of greater than 20 dB for all Ω less than 100rad/sec.
4. Realize the following system functions using a minimum number of multipliers
1 3 1
(a ) H ( z ) =1+ z−1 + z−2 + z−3 + z−4
2 4 2
1 1
(b )H ( z ) =1+ z−1 + z−2 + z−3
2 2
1 1
(c ) H ( z ) =(1+ z−1 + z−2 )(1+ z −1 + z−2 )
2 4
5. Design an FIR linear phase, digital filter approximating the ideal frequency response

π
{
H d (ω)=¿ 1 ,for|ω|≤ ¿ ¿¿¿
6
Determine the coefficients of a 11- tap filter based on the window method with a Hamming
window. (Dec 2013)
6. A low pass filter is to be designed with the following desired frequency response. (May
2015)

Determine the
jw
H d ( e )= ¿ e
filter
{ −j 2w

coefficients
,for |ω|≤ ¿ ¿¿¿
hd(n)
4
if the window function is defined as

w ( n )=¿ {1,for 0≤n≤4¿¿¿¿

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1
H a( S )=
7.Convert the analog filter with system function (s+0.2)2 +2 into digital IIR filter
by means of Impulse invariant transformation and Bilinear Transformation method. (Dec
2016)
2
H a( S )=
8. Convert analog filter (s+1)(s+2)
into digital filter by means of bilinear
transformation when T = 1 sec (Dec 2012) (Nov 2017)
9. Determine H(Z) for a butterworth filter Satisfying following constraints
π
√ 0. 5≤|H ( j Ω)|≤1 for 0≤w≤
2
3π
|H ( j Ω)|≤0 . 2 for ≤w≤π
4
With T=1 s .Apply Impulse invariant transformation(May 2015)
10. Convert the following analog transfer function into digital using impulse invariant
s+1
H (s)=
technique with sampling period T=1sec. (s+3)( s+5 )
(iii) Design a low pass FIR filter for the following specifications using rectangular window
function: Cut-off frequency = 500Hz, Sampling frequency =2000Hz, Order of the filter=10.
(Dec 2014)
11.Design a butterworth filter using the impulse invariance method for the following
specifications
0.8≤|H(ejw)|≤1 0≤w≤0.2π
jw
|H(e )|≤0.2 0.6π≤w≤π (Dec 2015) (Nov 2017)
Realize the designed filter using direct form II structure.
12.Design a filter with desired frequency response
−3 π 3π
for ≤w≤
Hd(ejw) = e-j3w 4 4
3π
for ≤|w|≤π
=0 4
Using a hanning window for N=7 (Dec 2015)
13.Design an ideal low pass filter with a frequency response
−π π
for ≤w≤
2 2
Hd(ejw) = 1
π
for ≤|w|≤π
=0 2
Find the values of h(n) for N=11. Find H(z) and the filter coefficients. Assume rectangular
window
(May 2016) (Nov 2017)
14. (i) Given the specifications αp=3dB.αs=10dB.fp=1 kHz and fs=2 kHz. Determine the
order of the filter using chebyshev approximation Find H(s).(Nov 2018)
(ii) Apply bilinear transformation to

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2
H (s)=
(s+1)(s+2)) with T=1sec and find H(z).(May 2016)
15.(i) Explain the role of windowing to realize a FIR filter.(Dec 2016)
(ii) Compare and explain on the choice and type of windows selection for signal analysis.
(iii)Compute numerically the effect of Hamming windows and design the filter if cut –off
frequency=100Hz, Sampling frequency=1000Hz, Order of the filter =2, Filter length
required=5.
16. Using bilinear transformation, design a high pass filter, monotonic in passband with
cutoff frequency of 1000 Hz and down 10 dB at 330Hz. The sampling frequency is 5000Hz.
(May 2017)
17. Design a 15 tap linear phase filter using frequency sampling method to the following
discrete frequency response
H(2πK/15) = 1, K = 0,1,2,3
0.4, K=4
0, K=5,6,7
0 . 5 ( 1+z −1 )
H ( z )=
18. Convert the signal pole low pass filter with system function ( 1−0. 302 z ) into band
−3

pass filter with upper and lower cutoff frequencies ωu and ωl respectively. The lowpass
filter has 3dB bandwidth and ωp =π/6 and ωu=3π/4 ,ωl=π/4 and draw its realization in
direct form II. (May 2017)
19. Obtain the analog transfer Chebyshev filter transfer fuction that satisfies the given
constraints
(Nov 2017)
1
≤|H ( jΩ)|≤1 for 0≤Ω≤2
√2
|H ( j Ω)|≤0 . 1 for Ω≥4
19. Design a filter using hamming window with the specifications N=7 of the system(May
2018)
−π π
for ≤w≤
Hd(ejw) = e-j3w 4 4
−π
for ≤|w|≤π
4
=0

20. Determine an ideal high pass filter using hanning window with the specification N=11

of the system (Nov 2018)

π
for ≤|w|≤π
Hd(ejw) = 1 4
π
for|w|≤
0 4
=
21. Design a ideal bandpass filter with a frequency response (Nov 2018)
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π 3π
{
H d (ω)=¿ 1,for |ω|≤ ¿ ¿¿¿
4 4
Find the value of h(n) for N=11 and plot the frequency response.
UNIT –V DIGITAL SIGNAL PROCESSOR
PART-A
1. What are the classification digital signal processors?
1. The digital signal processors are classified as
2. General purpose digital signal processors.
3. Special purpose digital signal processors.
2. Give some examples for fixed point and floating point DSPs.
Fixed point DSPs are
TMS320C50, TM 320C54, TM 320C55, ADSP-219x, ADSP-219xx.
Floating point DSPs are
TMS320C3x, TMS320C67x, ADSP-21xxx.
3. What are the factors that influence selection of DSPs?
The main factors that influence selection of DSPs are,
1. Architectural features 2.Execution speed 3.Type of arithmetic 4.Word length
4. What are the applications of PDSPs?(May 2015) (May 2018)
Digital cell phones, automated inspection, voicemail, motor control, video conferencing,
Noise cancellation, Medical imaging, speech synthesis, satellite communication etc.
5. What is pipelining?(Dec 2011) (Dec 2012)
Pipelining a processor means breaking down its instruction into a series of discrete pipeline
stages which can be completed in sequence by specialized hardware.
6. What is the pipeline depth of TMS320C50, TM 320C54x?
TMS320C50 – 4 TM 320C54x – 6
7.What are the different stages in pipelining?(Dec 2014)(May 2018)
i. The Fetch phase ii. The Decode phase iii. Memory read phase iv. The Execute phase
8.What are the different buses of TM 320C5x and their functions?(Nov 2018)
The ‘C5x architecture has four buses
1. Program bus (PB) 2.Program address bus (PAB) 3.Data read bus (DB) 4.Data read
address bus (DAB)
 The program bus carriers the instruction code and immediate operands from program
memory to the CPU.
 The program address bus provides address to program memory space for both read and
write.
 The data read bus interconnects various elements of the CPU to data memory spaces.
 The data read address bus provides the address to access the data memory spaces.
9. List the various registers used with ARAU of DSP processor? (Dec 2014)
i) Eight auxiliary registers (AR0 –AR7). They are used for indirect addressing. ii) Index
(INDX) and auxiliary register compare register (ARCR) are used to calculate indirect
address.
10. What are the elements that the control processing unit of ‘c5X consists of
1. Central arithmetic logic unit (CALU).2 Parallel logic units (PLU)
3. Auxiliary register arithmetic unit (ARAU) 4. Memory mapped registers
5. Program controller
11. What is the function of parallel logic unit?

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The parallel logic unit is second logic units, which execute logic operations on data without
affecting the contents of accumulator.
12. List the on chip peripherals in ‘c5x.
The on-chip peripherals interfaces connected to the ‘c5x CPU include
1. Clock generator 2. Hardware timer 3. Software programmable wait state
generators, 4.General purpose I/O pins 5. Parallel I/O ports, 6. Serial port interface, 7.
Buffered serial port
8. Time-divisions multiplexed (TDM) serial port, 9. Host port interface10. User unmask
able interrupts
13.Give the special features of DSP processors. (Dec 2011) (May 2015)(Nov 2017)(Nov
2018)
1. Harmonics can be analyzed using Fourier analysis.2. Generation of pulses 3. Discretizing
the waveform
14. What is the function of parallel logic unit DSP processor? (Dec 2012)
The parallel logic unit is a second logic unit, that execute logic operations on data
Without affecting contents of accumulator.
15. Define Period gram. (May 2012)
Periodic analysis of the waveform can be analyzed.
16. What is meant by bit reversed addressing mode? What is the application for which
this addressing mode is preferred? (Dec 2013)
Bit-reverse addressing is a special type of indirect addressing. It uses one of the auxiliary
registers (AR0−AR7) as a base pointer of an array and uses temporary register 0 (T0) as an
index register. When you add T0 to the auxiliary register using bit-reverse addressing, the
address is generated in a bit-reversed fashion, with the carry propagating from left to right
instead of from right to left.
Application: Bit-reversed addressing, a special addressing mode useful for calculating FFTs
17. Compare the RISC and CISC processors. (Dec 2013)
CISC RISC
Emphasis on hardware Emphasis on software
Memory-to-memory: Register to register:
"LOAD" and "STORE" "LOAD" and "STORE"
incorporated in instructions are independent instructions
Small code sizes, Low cycles per second,
high cycles per second large code sizes
Transistors used for storing Spends more transistors
complex instructions on memory registers
If it reads as above (i.e. as CISC
computer), If it reads as above (i.e. as RISC computer),
it means a computer that has it means a computer that has
a Complex Instruction Set Chip as a Reduced Instruction Set Chip as its cpu
its cpu.
18. Mention one important feature of Harvard architecture. (May 2013)
(i) Harvard architecture has separate memories for program and data. It also has separate
address and data buses for program and data. Because of these separate on chip memories
and internal buses, the speed of execution in Harvard architecture is high.
19. What is the advantage of Pipelining? (May 2013)
It provides sequential flow of execution with one after the other process without any
interruption. This concept of pipelining increases computational efficiency of the processor.
20.How many buses does C54X processor have & what are they? (May 2014)
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i) There are eight major 16 bit buses (four program/data bus and four address buses).
ii) Program bus (PB) carries instruction code and immediate operands from program
memory.
iii) Three address buses (CB, DB and EB) interconnect CPU, data address generation
logic, program address generation logic, on chip peripherals and data memory.
iv) Four address buses (PAB, CAB, DAB and EAB) carry the addresses needed for
instruction execution.
21.What are the elements present in CPU of 54X processor?
i. 40-Bit Arithmetic Logic Unit (ALU), ii.Two 40- Bit Accumulator Registers,
iii.Barrel Shifter, iv. Multiply / Accumulate Block,v. 16-Bit Temporary Register (T), vi.
16-Bit Transition Register, vii.Compare, select and store unit, viii. Exponent Encoder
22.What is IMR and IFR? (May 2014)
IMR (Interrupt mask register) individually masks off specific interrupts at required
times.
IFR (Interrupt flag register) indicates the current status of the interrupts.
23.State the intended applications of DaVinci Digital Media processors(May 2011)
Image compression, Image coding ,speech compression, multirate signal filters
24.What is the advantage of Harvard architecture in a DS processor? (Dec 2015)
In Harvard architecture, memory of data and memory of instruction are separated. Its
advantages includes, faster execution time and it allows concurrent access of data and
instruction.
25.How is a DS processor applicable for motor control applications (Dec 2015)

26. What are the merits and demerits of VLIW architecture?(May 2016)
Advantages:Increased performance , Better compiler targets, Potentially easier to program ,
Disadvantages:Increased memory use, High program memory bandwidth requirements
,High power consumption,Misleading MIPS ratings
27. What are the factors that influence the selection of DSP processor for an
application
(May 2016)
The right DSP processor for a job depends heavily on the application. One processor may
perform well for some applications, but be a poor choice for others. With this in mind, one
can consider a number of features that vary from one DSP to another in selecting a
processor.
These features are: Ease of Development,Multiprocessor Support,Power Consumption and
Management, Cost,Memory Organization.
28.State how spectrum meter application can be designed with DS processor (Dec
2016)
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EE8591 Digital Signal Processing Dept of EEE/EIE 2019-2020
The FFT or Fast Fourier Transform spectrum analyser uses digital signal processing
techniques to analyse a waveform with Fourier transforms to provide in depth analysis of
signal waveform spectra.With the FFT analyse, its able to provide facilities that cannot be
provided by swept frequency analyzers, enabling fast capture and forms of analysis that are
not possible with sweep / superheterodyne techniques alone.
29. What is pipelining and how do define its depth? (May 2017)
Pipelining a processor means breaking down its instruction into a series of discrete pipeline
stages which can be completed in sequence by specialized hardware. The number of
pipeline stages is referred to as the pipeline depth.
30. Write some commercial DSP Processors. (May 2017)(Nov 2017).
TMS320C50, TM 320C54, TM 320C55, ADSP-219x, ADSP-219xx, TMS320C3x,
TMS320C67x, ADSP-21xxx.
PART – B
1. Describe in detail the architecture of TMS 320C 54 DSP processor and state the main
features of this processor.(Dec 2011) (May 2012) (Dec 2012) (May 2013)(Dec 2014) (May
2015)
2. Explain the following with reference to DSP processors.
(i) MAC (ii) Pipelining (May 2014)
3. Explain the addressing modes of a DSP processor with suitable examples (OR) What are
the special addressing modes of TMS 320C54 chip (Dec 2012)(Dec 2013) (May 2013)(Dec
2014)(Nov 2018)
4. Explain Von Neumann, Harvard architecture and modified Harvard architecture for the
computer (Dec 2012) (Dec 2013)
5. Explain the advantages and disadvantages of VLIW architecture (Dec 2012)
6.(i)Write short notes on memory mapped register addressing(Dec 2012)
(ii)Explain about pipelining in DSP(May 2012)
7.(i)Write short notes on circular addressing mode(Dec 2012)
(ii)Write short notes on auxiliary registers(Dec 2012)
8. Explain how convolution is performed using a single MAC unit. Discuss the addressing
modes used in programmable DSPs. (Dec 2013)
10. Explain the types of operations performed by L functional mode. (Dec 2014)
13.(i)Discuss on the addressing modes supported by a DSP processor (May 2018)
(ii)Design a DSP based system for the process of audio signals in an audio recorder system.
(May 2016)
14. (i) Explain the data path architecture and the bus structure in a DSP processor with
suitable diagram
(ii) Elaborate on radar signal processing using a DSP processor (May 2016)
15.Compute the following if x1 =[-1,-1,-1,2]; x= [-2,-1,-1,-2] (Dec 2016)
(i) Linear and circular convolution of a sequence
(ii)x1:x2 subject to addition and multiplication
16.Write briefly on any two of the following: (Dec 2016)
(i) Quantisation and errors in DS processor
(ii) With neat figure explain the architecture of any one type of a DS processor
(iii)The addressing modes of one type of DS processor.
17.Elaborate one application of digital signal processing with a DS processor.(Dec 2016)
18. Discuss the features and architecture of TMS 32050 processor.(May 2017)(Nov 2017)
19. Explain the addressing mode and registers of DSP Processors. (May 2017)
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20. Explain the classification of instructions in DSP processor with suitable examples. (Nov
2017)
21. Draw the structure of central processing unit and explain the each units with its
functions. (May 2018)(Nov 2018)

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