DTSSP QB

QP06
Subject Code / Name: EE6403 DISCRETE TIME SYSTEMS

AND SIGNAL PROCESSING
UNIT I INTRODUCTION
PART-A
1. Check if the system described by the difference equation
y ( n )=ay ( n1 )+ x( n) with y (0) =1 is stable.
(APR/MAY 2015)
Any relaxed system is said to be bounded input bounded output (BIBO)

stable if
1 and only if every bounded input yields a bounded output. Mathematically,
their exist
2 some finite numbers, Mx and My such that,
x(n)| Mx < and y(n)| My < .
The given output is depending on the present state of the input. So as long
as the input is finite the output is also finite. Therefore the system is stable.
2. Differentiate between energy and power signals.
(APR/MAY 2015)
Energy Signal
An Energy signal is a signal with
finite energy and zero average
power
Energy signals are time limited
Energy signals are Non periodic
signals
Power Signal
The Power Signals: a power signal is
a signal with infinite energy but
finite average power
Power signals can exist over infinite
time.
Power signals are periodic.
3. Consider the analog signal x(t) = 3cos50t + 10 sin 300t cos 100t.
What is the Nyquist rate for this signal?
(MAY/JUN 2014)
Here, max=300
So,
2fm=300
The Nyquist
Fs
rate,
2fm
fm =300 /2
fm = 150
2 fm = 2 (150) = 300
Hence, Nyquist rate Fs 300
DTSSP | 11
KCE/EEE/QB/II Yr/DTSSP
QP06

4. State Shannons sampling theorem.
(NOV/DEC 2011][APR/MAR 2011)
(MAY/JUN 2014)
A band limited continuous time signal with highest frequency (band width) fm
hertz , can be uniquely recovered from its samples provided that the
sampling rate fs is greater than or equal to 2fm samples per second ,fs2fm
5. Given a continuous time signal x(t)= 2cos500t. What is the Nyquist
rate and fundamental frequency of the signal?

(MAY/JUN 2013)
= 500
2f= 500
f= 250Hz (fundamental frequency of the signal)
Nyquist rate Fs=2fm= 2x250= 500Hz
6. Determine whether x[n]=u[n] is a power signal or an energy signal.
(MAY/JUN 2013)
The energy of a discrete time signal x(n) is defined as

The average power of a discrete time signal x(n) is defined as
Here E= and P= Finite. Therefore the given signal is a power signal.

7. What is the Nyquist rate for the signal xa(t)=3cos
600t+2cos1800t?
(NOV/DEC
2013)
Solution: 1 =600
2 =1800
2f1 = 600
2f2 = 1800
f1 = 300Hz
f2 = 900Hz
Nyquist rate Fs=2fm= 2x900= 1800Hz.
cos
8. Determine fundamental period of the signal
30 n
105
(NOV/DEC 2013)
Solution: Fundamental period, N=
2
m
0
( )
where,
N=
N
periods.
9. Define
Nyquist
0 =
30
105
m
( 2 30105
)
105
m
15
, when m =1 , N = 7
rate.
[MAY/JUN 2012]
DTSSP | 12
QP06

It is the minimum rate at which a signal can be sampled and still

reconstructed from its samples. Nyquist rate (Fs)is always greater than or
equal to twice the maximum frequency(fm) of the signal, Fs2fm.
10.
What
is
an
LTI
system.
[NOV/DEC 2011][NOV/DEC 2012]
If the input-output relation of a system does not vary with time, the system
is said to be time-invariant or shift invariant system.
If the output signal of a system shifts k units of time upon delaying the input
signal by k units, the system under consideration is a time-invariant system.
Example: y(n) = x(n) + x(n-1)
11.
What is aliasing effect?
[NOV/DEC 2010][NOV/DEC 2012]
[MAY/JNU
2011]
Let us consider a band limited signal x(t) having no frequency component

for > m. If we sample the signal x(t) with a sampling frequency F<2f m,
the periodic convolution of x(j) results in spectral overlap. In this case, the
spectrum x(j) cannot be recovered using a low pass filter. This effect is
known as aliasing effect.
What
12.
are
the
different
types
of
signal
representation?
[MAY/JUN 2011]
Graphical representation
Functional representation
Tabular representation
Sequence representation
13.
Define even and odd signals.

(NOV/DEC
2010) discrete time signal x(n) is said to be even / symmetrical
signal if it satisfy the following condition: x(-n)= x(n), for all n.
A discrete time signal x(n) is said to be odd/ anti symmetrical signal if it
satisfy the following condition: x(-n)= -x(n), for all n.
14.
Define Quantization.
The process of converting a discrete-time continuous amplitude signal x(n)
into a discrete-time discrete amplitude signal xq(n) is known as quantization.
15.
Check whether the following system is time-variant
y(n)=nx2(n).
Given:
y(n) = T[x(n)] = nx2(n).
If the input is delayed by k units of time the output y(n,k) = nx12(n-k)
If the output is delayed by k units of time we get y(n-k) = (n-k)x12(n-k)
y(n,k) y(n-k).
DTSSP | 13
QP06

So the above system is time - variant system.

16.
Define DFT pair.
The DFT pair is,
x(k)= x(n) e j2kn/N
for 0 k N-1.
x(n) = 1/N x(k) e j2kn/N
for 0 n N-1.
Where x(n) time domain sequence
X(k) transformed sequence
17.
Define the following (a) System (b) Discrete-time system.
(a)
System
A System is defined as a physical device that performs an operation on a
signal.
(b)
Discrete-time system
A discrete-time system is a device or algorithm that operates on a
discrete-time input signal x(n), to produce another discrete-time signal
y(n) called the output signal.
x(n)
Discrete time system
Input signal
y(n)
Output signal
18.
List the merits and demerits of DSP:
MERITS:
1. The program can be modify easily for getting better results.
2. Better accuracy can be achieved by using adaptive algorithm
3. The digital signals can be easily stored and transported.
DEMERITS:
1. Speed limitations
2. Band width restrictions
3. Finite word length problems
19.
When discrete time signals called as a periodic signals?
A discrete time signals x(n) satisfy then the condition x(n)=x(n+N) then it is
called as periodic signals with periodicity of N samples.
If it does not satisfy the condition then it is called as non-periodic signals.
20.
What is static and dynamic system?

A static or memory less discrete-time system is a system whose output at
any instant depends on the input values at that instant but neither on the
past nor on the future values of the input.
Example : y(n) = ax(n)
A dynamic or a system with memory is one in which the past inputs or
outputs are stored to calculate the present output.
DTSSP | 14
QP06

Example: y(n) = x(n) + 3x(n-1)

21.
What are the classification of discrete-time systems?
The classification of discrete-time systems are,
i)
Static and Dynamic systems
ii)
Time-variant and Time-invariant systems
iii)
Linear and Non-linear systems
iv)
Stable and Unstable systems
v)
Causal and Non-causal systems
vi)
IIR and FIR systems
22.
What is linear and non-linear system?
A linear system is one that satisfies the superposition principle. The principle
of superposition requires that the response of the system to a weighted sum
of input signals be equal to the corresponding weighted sum of the response.
Those systems which are not satisfying the above condition are known as
non-linear systems.
23.
What is an anti-aliasing filter?

To avoid aliasing, we use an analog low pass filter before sampler to
reshape the frequency spectrum of the signal so that the frequency
spectrum of the signal so that the frequency spectrum for < s /2 is
negligible. This filter is known as anti-aliasing filter.
24.
What is a Causal system?

A System is said to be causal if the output of the system at any time
depends only on present and past input, but does not depend on future
inputs.
This can be mathematically represented as y(n) = F [ x(n),x(n-1),x(n-2).]
Example: y(n) = x(n) + x(n-1)
25.
Determine whether x[n]=u[n] is a power signal or an energy
signal.
The energy of a discrete time signal x(n) is defined as
E=
|x (n)| =
n=
The average power of a discrete time signal x(n) is defined as

N
2
1
P= lim
| x(n)| =0.5
N 2 N +1 n =N
Here E= and P= Finite. Therefore the given signal is a power signal.
PART B
DTSSP | 15
QP06

1. (i)Find the impulse response of a discrete time invariant system whose

difference equation is given by y ( n )= y ( n1 )+ 0.5 y (n2)+ x ( n )+ x ( n1) . (12)
[MAY/JUN 2015]
(ii)Explain the properties of discrete time system.(4)

2. (i)A discrete time system is represented by the following difference equation in
which x(n) is input and y(n) is output. y ( n )=3 y ( n1 ) nx (n)+ 4 x ( n1 ) +2 x (n+1) ;
and n 0. Is this system linear? Shift invariant? Causal? In each case, justify your
answer.(12)
[MAY/JUN 2015]
(ii)What is meant by quantization and quantization error?(4)
3. (i)Check the causality and stability of the system.
y ( n )=x (n )+ x ( n2)+ 4 x ( n ) + x (2 n1) .(8)
(ii)Check the system for linearity and time invariance.
(MAY/JUN 2014]
y ( n )=( n1 ) x 2 (n)+c .(8)
4. Explain the digital signal processing system with necessary sketches and give its
merits and demerits. (16)
[MAY/JUN 2014]
5. Determine the response of the following systems to the input signal .

(NOV/DEC 2013]
x(n) =
(i)
(ii)
(iii)
(iv)
(v)
n3
{|0n|;;3
otherwise
x 1 ( n ) =x ( n2 ) (n3)
x 2 ( n ) =x ( n+1 ) u(n1)
1
y ( n )= [ x ( n1 ) + x (n)+4 x ( n ) + x (n1) ]
3
y ( n )=max [ x ( n+1 ) , x ( n ) , x (n1) ]
Find the even and odd components of given x(n).(16)
6. A discrete time system can be

(i)
Static or Dynamic
(ii)
Linear or Non-linear
(iii)
Time invariant or Time varying
(iv)
Stable or Unstable
(NOV/DEC 2013]
Examine the following system with respect to the properties above

y ( n )=x ( n ) +nx (n+1) .(16)
7. (i) Given y[n]=x[n2]. Determine whether the system is linear, time invariant,
memory less and causal. (8)
[MAY/JUN 2013]
(ii) Determine whether the following is an energy signal or power signal.(8)

(1) x1[n]=6cos((/2)n)
(2) x2[n]=3(0.5)n u[n].
DTSSP | 16
QP06

8. Starting from the first principles, state and explain sampling theorem both in
time domain and in frequency domain.(16)

[MAY/JUN 2013]
9. Check for following systems are linear, causal, time invariant, stable and
static. (16)
[MAY/JUN 2012]
(i) y(n)=x(1/2n)
(ii) y(n)=sin(x(n))
(iii) y(n)=x(n)cos(x(n))
(iv) y(n)=x(-n+5)
(v) y(n)=x(n)+nx(n+2)
10.
Compute linear and circular convolution of the two sequence

X1(n)= {1,2,2,2} and X2(n)= {1,2,3,4}.(16)
[MAY/JUN 2012]
11.(i) What is causality and stability of a system? Derive the necessary and sufficient
condition on the impulse response of the system for causality and stability. (8)
[NOV/DEC 2012]
(ii) Determine the stability for each of the following linear systems: (8)
(i) y1(n)= (3/4)kx(n-k)
(ii) y2(n)= 2kx(n-k)
12.
(i)What is meant by energy and power signal? Determine whether the
following signals are
energy or power or neither energy nor power
signals. (12)
[NOV/DEC 2012]
(1) x1(n)=(1/2)nu(n)
(2) x2(n)=sin((/6)n)
(3) x3(n)=ej((2/3)+(/6))
(4) x4(n)=e2nu(n)
(ii) What is meant by sampling? Explain sampling theorem. (4)
13.
(i) Check whether following are linear, time invariant, causal and
stable.(8)
y(n ) = x(n ) + nx (n +1) .
[APR/MAY
2011]
(ii)Check whether the following are energy or power signals (8)

(1) x(n )=(1/2)n u(n)
(2) x(n)=Aejw0n.
14. (i)Describe in detail the process of sampling and quantization. Also
determine the expression for quantization linear. (10)

[APR/MAY 2011]
DTSSP | 17
QP06

(ii) Check whether the following are periodic. (6)

(i) x(n)=cos(3n)
(ii) x(n)=sin(3n).
15.(i) What do you mean by Nyquist rate? Give its significance. (6)
2010]
[NOV/DEC
(ii) Explain the classification of discrete signal. (10)

16.(i) Explain in detail the Quantization of digital signals. (8)
2010]
(ii) Describe the different types of sampling methods used. (8)
UNIT II
DISCRETE TIME SYSTEM ANALYSIS

PART A
x ( n )=an .
1. Determine the Z-transform of

2015)
The given x(n) =
[NOV/DEC
an
(APR/MAY
has the limit - to . It is split into two as shown
below,
2. Find the DFT of the sequence
x (n )
= { 1,1,0,0}.
(APR/MAY 2015)
The N-point DFT of x(n) is given by,

N 1
DFT{ x(n)} = X(k) =
2 nk
N
for k = 0,1,2N-1
n=0
Since x(n) is a 4-point sequence,

3
X(k) =
2 nk
4
= x(0) e 0
n=0
+ x(1) e j
=1+
k
2
k
2
+ x(2) e jk
+0 + 0
=1+
+ x(3)
e
3 k
2
k
2
3. What is meant by region of convergence? State its properties.

(MAY/JUN 2014)
The region of convergence (ROC) of X(z) is the set of all values of z for which
X(z) attains
a finite value.
The properties of region of convergence
DTSSP | 18
QP06

The
The
The
The
ROC
ROC
ROC
ROC
is a ring or disk in the Z plane centered at the origin.

cannot contains any poles.
of an LTI stable system contains the unit circle.
must be a connected region.
4. Given a difference equation y(n) = x(n)+3x(n+1)+2y(n-1).
Determine the system function H(z).

(MAY/JUN 2013)
On taking Z- transform, Y(Z) = X(Z)+3Z-1 X(z)+2Z-1Y(z)

Y(Z) [1-2Z-1] = X(Z) [1+3Z-1]
1
Y (Z)
[1+ 3 z ]
The system function, H(Z) =
=
1
X (Z)
[12 z ]
5. Find the stability of the system whose impulse response h(n) =
1 n
u (n) .
2
()
(MAY/JUN 2013)
|h(k )|<
The condition for the system to be stable is,
|h(k )|=2
In the given system ,
|h(k )|=2<
As
Therefore the given system is stable.
6. What are the properties of z- transform ?

Linearity
: z [a1 x1 (n) + a2 x2 (n)]= a1 X
(z) + a2 X
(z)
m1
Shifting
:
(i)
z[x ( n +
m
m)]= z
X (z) x(i)z
mi
i=0
(ii)
z[x(n m)]= z
m
: z [ n x (n) ]= z
Multiplication
d
dz
X(z)
) X ( z)
Scaling in z- domain : z[a n x(n)]= X (a 1 z )

: z[x (n )]= X ( z
Time reversal
Conjugation
: z[x (n)]= X
Convolution
Initial value
Final value
DTSSP | 19
z
z
)
(z )
h(nm)r ( m)
m=0
= H (z)R(z)
[ x( 0) ] =lim X (z)
z
: z[x( )]= Lt (1 z
( z)
z 1
)X
QP06

7. Find the convolution for x(n)={0,1,2} and h(n)={2,0,1}.

[MAY/JUNE 2012]
x(n)=>
0 1 2
h(n)=>2 0 1
____________________________________
0 1
2
0
0 0
0
2
4
____________________________________
0
2
4 1 2
y(n)={0,2,4,1,2}
8. Write the commutative and distributive properties of convolution.
[NOV/DEC 2011]
The commutative and distributive properties of convolution are,

(i)
Commutative property: x(n)*h(n)=h(n)*x(n)
(ii)
Distributive property: x(n)* [ h1(n)+h2(n) ] = [ x(n) * h 1(n) ] +[ x(n) *
h2(n) ]
9. What is zero padding? What are its uses?.
[NOV/DEC 2010]
Let the sequence x(n) has a length L. If we want to find the N-point DFT
(N>L) of the sequence x(n), we have to add (N-L) zeros to the sequence
x(n). This is known as zero padding .The uses of padding a sequence with
zeros are
(i)
We can get better display of the frequency spectrum.
(ii)
With zero padding, the DFT can be used in linear filtering.
10.
Give any two properties of linear convolution.
[NOV/DEC 2010]
The properties of linear convolution are,

(i)
Commutative property: x(n)*h(n)=h(n)*x(n)
(ii)
Associative property : [x(n)*h1(n)]*h2(n)=x(n)*[ h1(n)]*h2(n) ]
(iii) Distributive property: x(n)* [ h1(n)+h2(n) ] = [ x(n) * h1(n) ] +[ x(n) *
h2(n) ]
11.
Distinguish
between
Linear
convolution
and
circular
convolution.[NOV/DEC2010]
S.No
1
2
DTSSP | 20
Linear Convolution
Linear Convolution can be used
to find the response of a linear
filter
Zero padding is not necessary
to find the response of a linear
filter.
Circular Convolution
Circular convolution cannot be
used to find the response of a
filter.
Zero padding is necessary to
find the response of a filter.
QP06

If x(n) is a sequence of L no. of

samples and h(n) with m
number
of
samples,
after
convolution y(n) will contain
N=L+M-1 samples.
If x(n) is a sequence of L no. of

samples and h(n) with M
samples, after convolution y(n)
will
contain
N=Max(L,M)
samples.
12.
What are the different methods of evaluating inverse ztransform?
The different methods of evaluating inverse z-transform are,
(i)
Long division method
(ii)
Partial fraction method expansion method
(iii) Residue method
(iv) Convolution method.
13.
What is the relationship between z-transform and DTFT?
The z-transform of x(n) is given by
X ( z )= x (n) zn ---------(1)
n=
where z=r e j
Substituting z value in equation (1) we get,
X ( r e j )= x (n) rn e jwn ----(2)
The Fourier transform of x(n)is given by
X ( e j )
jn
x (n)e
------(3)
n=
Equation(2) and Equation (3) are identical, when r =1. In the z-plan this
corresponds to
the locus of points on the unit circle |z|=1. Hence X (e j ) is equal to X(z)
evaluated along
unit circle, or X ( e j ) =X(z) | z=e j
For X (e j ) to exist, the ROC of X(z) must include the unit circle.
14.
What are the properties of frequency response H(e i) of an LTI
system?
The properties of frequency response H(ei) of an LTI system are,
i) H(ei) is a continuous function of .
ii) The frequency response H(ei) is periodic with period 2.
iii) The magnitude function of H(ei) is even symmetric with respect to = .
iv) The phase function of H(ei) is anti-symmetric with respect to = .
15.
What is the necessary and sufficient condition on the impulse
response of stability?
The necessary and sufficient condition on the impulse response of stability is
given by
DTSSP | 21
QP06

+
|h (n)|<
n=-
where the h(n) impulse response
16.
How will you obtain linear convolution from circular
convolution?
Consider two finite duration sequences x(n) and h(n) of duration L samples
and M Samples respectively. The linear convolution of these two sequences
produces an output sequence of duration L+M-1 samples, whereas, the
circular convolution of x(n) and h(n) give N samples where N=Max(L,M). In
order to obtain the number of samples in circular convolution equal to L+M1, both x(n) and h(n) must be appended with appropriate number of zero
valued samples.
17.
What is meant by sectioned convolution?
If the data sequence x(n) is of long duration, it si very difficult to obtain the
output sequence y(n) due to limited memory of a digital computer.
Therefore, the data sequence is divided up into smaller sections. These
sections are processed separately one at a time and combined later to get
the output.
18.
Determine the Z-transform and ROC for the signal

x ( n )= ( nk )+ ( n+k ) .
The Z-transform and ROC for the given signal,

X(Z)= Z-k + Z+k
X(Z) will converge for all the values of Z, except Z = 0 and .
19.
Distinguish between Overlap add and Overlap save method.
S.No
1
2
3
4
DTSSP | 22
Overlap-save method
In this method the size of the
input data block is N=L+M-1
Each data block consists of the
last M-1 data points of the
previous data block followed
by L new data points
In each output block M-1
points are corrupted due to
aliasing,
as
circular
convolution is employed
To form the output sequence
the first M-1 data points are
discarded in each output block
and the remaining data are
Overlap-add method
In this method the size of the
input data block is L
Each data block is L points and
we append M-1 zeros to
compute N-point DFT
In this no corruption due to
aliasing, as linear convolution is
performed
using
circular
convolution
To form the output sequence,
the last M-1 points from each
output block is added to the
first (M-1) points of the
QP06

fitted together
20.
succeeding block
Distinguish between DFT and DTFT.

S.No
1
2
DFT
DTFT
Obtained
by
performing
Sampling is performed only in
sampling operation in both the
time domain
time and frequency domains
Discrete frequency spectrum
Continuous function of
PART B
1. (i) Find the Z-transform of x(n) = n2u(n).(8)
(APR/MAY
2015)
(ii)Find the inverse Z-transform of X(Z)=
Z
3 z 4 z+ 1
2
for Region of
convergence.(8)
1.
|Z|> 1,
2.
|Z|<
1
3
3.
1
<|Z|<1.
3
2. (i)Convolute the following two sequences x1(n)= {0,1,4,-2} & dx2(n)=

{1,2,2,2}.(8)
(ii)Find the frequency response of the LTI system governed by the equation,
(APR/MAY 2015)
y ( n )=a 1 y ( n1 ) a2 y ( n2 )x ( n ) . (8)
3. (i)Find the Z-transform and ROC of x(n)= r n cos(n )u(n).(8)

(MAY/JUN 2014)
(ii)Find Inverse Z-transform of X(z)= z / [ 3 x2 4 z +1 ] , ROC |z|>1. (8)
4. (i) Determine the DTFT of the given sequence x(n)= a n (u(n)-u(n-8)), |a| <1.(8)
(ii)Prove the linearity and frequency shifting theorems of the DTFT.(8) (MAY/JUN
2014)
5. (i) Find the Z-transform and its associated ROC for the following discrete time signal
x[n]=(-1/5)n u[n]+5(1/2)-n u[-n-1]. (8)

[MAY/JUN 2013]
(ii) Evaluate the frequency response of the system described by system

function
H(z)= 1/(1-0.5z-1). (8)
6.
Using z-transform determine the response y[n] for n>=0 if
y[n]=(1/2) y[n-1]+ x[n], x[n]=(1/3) n u(n)y(-1)=1. (16)

[MAY/JUN 2013]
DTSSP | 23
QP06
7.

(i) Determine the system function and the unit sample response of the system
described by
the difference equation y(n)+(1/2)y(n-1)+2x(n).(8)

[MAY/JUN 2012]
(ii) Determine the step response of the system y(n)-y(n-1)+x(n), -1<<1,

when the
initial condition is y(-1)=1. (8)
8. An filter system is described by the difference equation y(n)=x(n)+x(n-10).
(16)
(i) Compute and sketch its magnitude phase response.
[MAY/JUN 2012]
(ii) Determine its response to the input x(n)=cos(/10)n+3sin((/3)n+

(/10)n).
9. (i) Find the Z transform and its ROC of x(n)=(-1/5)nu(n)+5(1/2)-n u(-n-1).(6)
(ii) A system is described by the difference equation y(n)-(1/2)y(n-1)=5x(n).
Determine the
solution, when the input x(n)=(1/5 )n u(n) and the initial
condition is given by
y(-1)=1,using Z-transform.(10)
[NOV/DEC 2012]
10.
(i) Determine the impulse response of the system described by the
difference equation
y(n)=y(n-1)-(1/2)y(n-2)+x(n)+x(n-1) using Z transform and discuss its
stability. (10)
(ii) Find the linear convolution of x(n)={2,4,6,8,10} with h(n)={1,3,5,7,9}.
(6)
[NOV/DEC 2012]
11.(i) Determine the Z transform of
[APR/MAY 2011]
(1) x(n)=an cos 0n u(n) (5)

(2) x(n)=3n u(n).
(3)
(ii) Obtain x(n) for the following:
X(z)=(1/(1-1.52-1+0.52-2)) for ROC: |z|>1, |z|<0.5, 0.5<|z|<1.(8)
12.
(i) Determine the linear convolution of the following sequences

x1(n)={1,2,3,1} , x2(n)={1,2,1,-1}.(6)
[APR/MAY 2011]
(ii) Obtain the system function and impulse response of the

following system y(n )-5 y(n -1) = x(n )+x(n -1).(10)
13.(i)Obtain the linear convolution of
2011]
[NOV/DEC
x(n ) = {3,2,1,2}, h(n ) = {1,2,1,2}. (6)

(ii) A discrete time system is described by the following equation :
DTSSP | 24
QP06

y(n)+(1/4)y(n-1)=x(n)+(1/2)x(n-1)
Determine its impulse response. (10)
14.
(i) Obtain the discrete Fourier series coefficients of x (n)=cos 0n .(4)
[NOV/DEC 2011]
(ii) Determine x(n ) for the given X(Z) with following ROC, (12)
(1) | z| > 2
(2) | z| < 2
X(Z)=(1+3z-1 )/(1+3z-1+2z-2)
15. (i) Explain the properties of Z-transform.(8)
2010]
16.
[NOV/DEC
(ii) Find the impulse response given by difference equation. (8)

y(n)-3y(n -1)-4y(n - 2) = x(n) + 2x(n -1)
Test the stability of given systems. (8)
[NOV/DEC 2010]
(1) y(n) = cos(x (n))

(2) y(n) = x(-n - 2)
(3) y(n) = n x(n)
(ii) Find the convolution. (8)
x(n)={-1,1,2,-2}, h(n)={0.5,1,-1,2,0.75}
UNIT III
DISCRETE FOURIER TRANSFORM AND

COMPUTATION
PART A
1. Determine the Fourier Transform of the signal
x ( t )=sin 0 t .
(APR/MAY 2015)
The Fourier Transform the given signal x(t) is,

F{ x(t)} = F{ sin 0 t }
=
[ ( 0 ) ( +0) ]
j
2. Draw the basic butterfly flow graph for the computation in the DIT FFT
algorithm.
(NOV/DEC2013)[NOV/DEC2011][APR/MAY2011] (APR/MAY
2015)
Butterfly flow graph for the computation in the DIT FFT algorithm is given
below.
DTSSP | 25
QP06

3. Calculate DFT of x(n)={1,1,-2,-2}.
[NOV/DEC
2010]
Given x(n)={1,1,-2,-2}
X(k)= x(n) e-j2kn/N
Where K=0,1,2,. N-1
and n=0,1,2,3
X(0)= x(0) e-j2kn/N
={1+1-2-2}
X(1)= x(n) e-jn/2
={1+1(-j)+(-2)(-1)-2(j)} =(3-j3)
X(2)= x(n) e-jn

X(4)= x(n) e-j3n/2
=-2
=1+1(-1)-2(1)-2(-1)
=0
=1+1(j)-2(-1)-2(-j)
=3+j3
X(k)={-2,3-j3,0,3+j3}
4. State circular frequency shift property of DFT.
(MAY/JUN
2014)
Circular frequency shift, if

x ( n ) DFT X (k )
x (n )e
then
jln / N
DFT X ( kl ) N
5. What are the differences and similarities between DIF and

DIT algorithms?
Differences:
[NOV/DEC2010](MAY/JUN
2014)
For DIT, the input is bit reversal while the output is in natural order,
whereas for DIF, the input is in natural order while the output is bit
reversed.
The DIF butterfly is slightly different from the DIT butterfly, the difference
being that the complex multiplication takes place after the add-subtract
operation in DIF.
Similarities: Both algorithms require same number of operations to
compute the DFT. Bot algorithms can be done in place and both need to
perform bit reversal at some place during the computation.
6. Find the Discrete Fourier Transform of (n).
[MAY/JUN 2013]
DFT{x(n)} =X(k)= x(n) e-j2kn/N

DFT{ (n)} =X(k)= (n)e-j2kn/N =1
(n)=1 for n=0

=0 for n is equal to 0
7. Draw the basic butterfly diagram of DIF algorithm.

[MAY/JUN 2013]
DTSSP | 26
QP06

The basic butterfly diagram of DIF algorithm is,
8. In eight point decimation in time(DIT), what is the gain of the signal path
that goes
from x(7) to X(2)?
2013]
[NOV/DEC
In eight point DIT the gain of the signal the path that goes from x(7) to X(2)
is
W80W82 .
9. Give relationship between DTFT and Z-Transform.
[MAY/JUN 2009][MAY/JUN 2012]
The Z-transform of x(n) is given by

Z{x(n)}=X(z)= x(n)z
j
where z= r e
Sub (2) in (1) we get
-n
------(1)
--------(2)
j
The Fourier transform of x(n) is given by

X(e ) = x(n)e-j
When r=|1| the z-plane corresponds to the locus of points on the unit circle. |
z|=1.
Hence
X(ej)=X(z)
when z= ej
10.
What is FFT? What are its advantages?
[NOV/DEC 2012]
The fast fourier transform is an algorithm used to compute the DFT. The
direct evaluation of DFT using the formula X(k) = x(n) e j2nk/N
n=0
requires N2 complex multiplication and N(N-1) complex additions. Thus for

large values of N(in the order of 1000) the DFT requires an inordinate
amount of computation. By using FFT algorithms the number of
computations can be reduced.
Main advantage of FFT: Computation time is reduced.
11.
What are the applications of FFT algorithm?
The applications of FFT algorithm are,

i)
Linear filtering
ii)
Correlation
iii)
Spectrum analysis
DTSSP | 27
QP06

Calculate the number of multiplications needed

computation of DFT and FFT with 64-point sequence.
12.
in
the
The number of complex multiplications required using direct computation is

2
N =64 =4096
The number of complex multiplications required using FFT is
(N/2)log2N =(64/2) log264=192
Speed improvement = 4096/192 =21.33
13.
Distinguish between DFT and DTFT
DFT
DTFT
Obtained
by
performing
sampling Sampling is performed only in time
operation in both the time and domain.
frequency domains
Discrete frequency spectrum.
Continuous function of
14.
Distinguish between FIR and IIR.
S.No
.
FIR Filter
IIR Filter
These filters can be easily designed to have

perfectly linear phase.
These filters do not

have
linear phase.
FIR filters can be realized recursively and IIR
filters
can
be
non-recursively.
realized recursively.
Greater flexibility to control the shape of Less flexibility, usually
their magnitude response.
Errors due to roundoff noise are less severe The roundoff noise in IIR
in FIR filters, mainly because feedback is filters are more.
not used.
1.
2.
3.
4.
15.
Define the Discrete Fourier transformation of a given sequence
x(n).
The N- point DFT of a sequence x(n) is
N 1
n =0
X (k ) = x(n)e j 2kn / N
k= 0, 1, 2..N-1.
16.
Obtain the
x(n)={0,1,0,2 };
circular
convolution
the
following
sequences
h(n)={ 2,0,1 }.
The circular convolution of the above sequences can be obtained by using
matrix method.
DTSSP | 28
QP06

17.
IF N-point sequence x(n) has N- point DFT X(k) then what is
18.the DFT of the following?
(i)x*(n)
(ii)x*(N n)
(iii)x((n l))N
(iv)x(n)e
j2 ln/N
(i) DFT { x (n )} = X ( N k )
(ii ) DFT { x ( N n )} = X (k )
(iii ) DFT { x (( n l )) N } = X (k )e j 2kl / N
(iv ) DFT { x (n )e j 2 ln/ N } = X (( k l )) N
18. List any four proerties of DFT.
(a) Periodicity
If X(k) is N- point DFT of a finite duration sequence
x(n) then
x(n + N ) =
for
x(n)
all n
X (k + N ) =
X(k)
for all k.
(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), then
DFT{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), then
DFT{x((n-l))N}=X(k)
19.
j 2 kl / N
What is decimation in time algorithm?
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-point 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.
DTSSP | 29
QP06

20.
Find the discrete Fourier Transform for [n].
The discrete Fourier transform of [n]
N 1
X(k) = x(n)e
n=0
j 2nk / N
= (n)e() =1.
n=0
PART B
1. (i)Determine the DFT of the sequence x(n) =
1
, for 0 n 2
4
0 , otherwise
(8) (APR/MAY
2015)
(ii)Draw the flow graph of an 8-point DIF FFT algorithm and explain. (8)
2. (i)Given x(n) = n+1, and N=8, find X(K) using DIT, FFT algorithm.(8)
(APR/MAY 2015)
(ii)Use 4-point inverse FFT for the DFT result {6,-2+j2,-2,-2-j2} and
determine the input sequence.(8)
3. An 8-point sequence is given by x(n)= {2,2,2,2,1,1,1,1} compute DFT of x(n)
using radix 2 DIT FFT.(16)
(MAY/JUN 2014)
4. (i) Determine 8-point DFT of the sequence x(n)= {1,1,1,1,1,1,10,0}.(12)
(MAY/JUN 2014)
(ii)Find circular convolution of the sequences using concentric circle method

x1={1,1,2,1} and x2 = {1,2,3,4}(4)
5. Find the output y[n] of a filter whose impulse response is h[n]={1,1,1} and
input signal
x[n]={3,-1,0,1,3,2,1,2,1} using overlap save method.(16)
[MAY/JUN 2013]
6. Find the DFT of a sequence x[n]={1,2,3,4,4,3,2,1}.Using DIT algorithm.(16)

[MAY/JUN 2013]
7. ( i) Derive the computational equation for 8 point FFT DIT. (8)
[MAY/JUN2012]
(ii) State and prove any five properties of DFT.(8)

DTSSP | 30
QP06

8. Find the X(K) for the given sequence x(n)={1,2,3,4,1,2,3,4}.

[MAY/JUN2012]
(16)
9. (i) State and prove convolution property of DFT. (6)

(ii) Find the IDFT of X(k)={7,-2-j2,-j,2-j2,1,2+j2,j,-2+j2}.(10)
[NOV/DEC 2012]
10.
(i) Derive decimation in time radix-2 FFT algorithm and draw signal
flow graph for 8-point sequence.(8)
[NOV/DEC 2012]
(ii) Using FFT algorithm compute the DFT of x(n)={2,2,2,2,1,1,1,1} . (8)
11.
(i) Explain the following properties of DFT.
[APR/MAY 2011]
(1) Convolution.
(2) Time shifting
(3) Conjugate Symmetry.
(10)
(ii) Compute the 4 point DFT of x(n ) = {0,1, 2,3}.(6)
12.
(i) Explain the Radix 2 DIFFFT algorithm for 8 point DFT. (8)
[APR/MAY 2011]
(ii) Obtain the 8 point DFT using DITFFT algorithm for

x(n ) = {1,1,1,1,1,1,1,1}.(8)
13.(i) Explain 8 pt DIFFFT algorithm with signal flow diagram. (10)
2011]
[NOV/DEC
(ii) Compute the DFT of x(n ) = {1, 1, 0, 0}. (6)

14.
(i) Describe the following properties of DFT.

(1) Time reversal
(2) Circular convolution. (10)
(ii) Obtain the circular convolution of
X1(n)= {1, 2, 2, 1} , x2(n ) = {1, 2, 3, 1} (6)
[NOV/DEC 2011]
15.
An 8-point sequence is given by x(n)= {2, 2, 2, 2, 1,1,1,1,}. Compute
8-point DFT of x(n) by radix DIT--FFT method also sketch the magnitude and
phase. (16) [NOV/DEC 2010]
16.
Determine the response of LTI system when the input sequence is x(n)
= {1,1,2,1,1} using radix 2 DIF FFT. The impulse response is h(n) =
{1,1,1,1}.(16)[NOV/DEC 2010]
UNIT IV
DTSSP | 31
DESIGN OF DIGITAL FILTERS

QP06

PART A (2 MARKS)
1. Comment on the pass band and stop band characteristics of butter
worth filter.
(APR/MAY 2015)
Compared with a Chebyshev Type I/Type II filter or an elliptic filter, the

Butterworth filter has a slower roll-off, and thus will require a higher order to
implement a particular stop band specification, but Butterworth filters have a
more linear phase response in the pass-band than Chebyshev Type I/Type II
and elliptic filters.
2. Realize the following causal linear phase FIR system function
2 1 2 2
H ( z )= + z + z
.
(APR/MAY 2015)
3
3
X(n)
z-1
z-1
2/3
2/3
Y(n)
3. Define pre-warping effect.

Why it+ is employed?
+
2014)
(MAY/JUN
In IIR filter design using bilinear transformation the specified digital

frequencies are converted to analog equivalent frequencies, which are called
pre-warp frequencies.
The pre-warping is necessary to eliminate The effect of the non-linear
compression at high frequencies can be compensated.
4. What is window and why it is necessary?
One possible way of finding an FIR filter that approximates
H ( e j )
[ ]
N 1
. The
2
abrupt truncation of the series will lead to oscillation both in pass band
and in stop band. 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.
would be to truncate the infinite Fourier series at n=
What are the advantages of FIR filter.

[NOV/DEC 2010]
The advantages of FIR filter are,

a. FIR filters have linear phase
b. FIR filters are always stable.
DTSSP | 32
QP06

c. FIR filters can be realized in both recursive and non-recursive

structure.
d. Filters with arbitrary magnitude response can be tackled using FIR
sequence.
5. Mention the significance of Chebychevs approximation.
[NOV/DEC 2010]
From the Chebychevs approximation we can obtain the parameters like

the order of the filter N,
transition ratio k, and
the poles of the filter.
6. Name two methods for digitizing the transfer function of an analog
filter.
[MAY/JUNE 2013]
The two important procedures for digitizing the transfer function of an analog
filter are
(i) Bilinear transformation method.
(ii) Impulse invariant method
7. List the properties of chebychew filter.
[May/June
2013]
i) The magnitude response of the chebychew filter exhibits ripple either
in pass band or in
stop band according to type.
ii) The poles of the chebychew filter lie on an ellipse.

8. Define the condition for stability of digital filters.
[May/June 2012]
A digital is stable if its impulse response is absolutely summable.
|h(n)|<
9. What is the need for employing window for designing FIR filters?
[Nov/Dec 2012]
One possible way of finding an FIR filter that approximates H(e
jw
) would
truncate the
infinite Fourier series at n= (N-1)/2. Abrupt truncation of the series will lead to
oscillation in the pass band and stop band. The oscillation can be reduced
through the use of less abrupt truncation of the Fourier series. This can be
DTSSP | 33
QP06

achieved by multiplying the infinite impulse response with a finite weighing

sequence w(n), called a window
10.
What is meant by linear phase response of a filter?

[Nov/Dec2011]
For a linear phase filter ()= . The linear phase filter did not alter the
shape of the original signal. If the phase response of the filter is non linear
the output signal may be distorted one. In many cases a linear phase
characteristics is required throughout the passband of the filter to preserve
the shape of a given signal within the pass band. IIR filter cannot produce a
linear phase. The FIR filter can give linear phase, when the impulse response
of the filter is symmetric about its mid point.
The necessary and sufficient condition for linear phase characteristic in FIR
filter is, the impulse response h(n) of the system should have the symmetry
property, i.e.,
h(n) = h(N-1-n)
11.
Compare bilinear transformation and impulse invariant method
of IIR filter design.

[Nov/Dec2011]
S.NO
1.
IMPULSE INVARIANT
Many to one mapping
BILINEAR TRANSFORMATION
One to one mapping
Linear frequency relationship

Non-Linear
frequency
2.
between
analog
and
its
relationship.
12.
transformed digital frequency
Aliasing will be eliminated by
Aliasing as a main drawback
3.
the
help
of
one-to-one
of the systems.
mapping.
are the features of FIR filter?
[APR/MAY2011]
W
ha
t
These filters can be easily designed to have perfectly Linear phase

It can be realized both recursive and non-recursively.
FIR filter realized non recursively are always stable.
Errors due to round off noise are severe in FIR filters, mainly because of
feedback not used.
DTSSP | 34
QP06
13.

Write the procedure for designing FIR filters

1. Choose the desired frequency response.
2. Take inverse Fourier Transform of Hd(ejw) to get hd(n).
3. Convert the infinite duration hd(n) to finite duration sequence h(n).
4. Take Z transform of h(n) to get the transfer function.
14.
What are the design techniques available for the designing FIR
filter?
1. Fourier series method
2. Window methods
3. Frequency sampling method.
15.
What are the possible types of impulse response for linear
phase FIR filters?

1. Symmetrical impulse response when N is odd.
2. Symmetrical impulse response when N is even.
3. Anti-symmetrical impulse response when N is odd.
4. Anti-symmetrical impulse response when N is even.
16.
What is GIBBs phenomenon?

[MAY/JUNE 2012]
In FIR filter design by fourier series method the infinite duration impulse
response is truncated to finite duration impulse response. The abrupt
truncation of the impulse response introduces some oscillations in pass band
and stop band. This effect is known as Gibbs oscillations.
17.
Write the desirable characteristics of frequency response of
window functions.
a. The width of the main lobe should be small and it should contain as
much of the total energy as possible.
b. The side lobe should decrease in energy rapidly as tends to .
18.
Write the characteristics features of rectangular window.

a. The main lobe width is equal to 4 /N.
b. The maximum side lobe magnitude is -13db.
c. The side lobe magnitude does not decrease significantly with
increasing .
19.
List merits and demerits of rectangular window.
DTSSP | 35
QP06

a. Number of side lobe is low compared to other windows.

b. The Width of the main-lobe is low compared to other windows mainlobe width for the same filter length.
20.
What are the two types of filter based on the impulse
response?
Based on the impulse response the filters are of two types.
1. IIR
2. FIR
The IIR filters are of recursive type, whereby the present output sample
depends on the present input, past input samples and output samples.
The FIR filters are of non recursive type whereby the present output samples
depends on the present input sample and previous input samples.
21.
What are the properties of FIR filter?

a. FIR filter is always stable
b. A realizable filter can always be obtained
c. FIR filter has a linear phase response.
22.
Define IIR filter.
The filter considering all the infinite samples of infinite response are called
IIR filter. The impulse response is obtained by taking inverse transform of
ideal frequency response.
23.
What are the methods available for designing analog IIR filter?
There are four methods:

a. Approximation derivation method.
b. Impulse invariant method.
c. Bilinear transformation method.
d. The matched Z- Transformation method.
24.
Mention the two properties of Butterworth low pass filter.
a. The magnitude response of the Butterworth filter decreases monotonically

as the frequency increases from 0.
b. The magnitude response of the Butterworth filter closely approximates
the ideal response as the order N increases.
25.
Write the properties of chebyshev type-I filter:
a. The magnitude response of the chebyshev type-I filter exhibits ripple in

the pass band.
DTSSP | 36
QP06

b. The poles of the chebyshev type-I filter lies on an ellipse.

26.
.What is aliasing? Why it is absent in bilinear transformation?
The phenomenon of the high frequency sinusoidal components acquiring the

identity of low frequency sinusoidal components after sampling is called
aliasing.
This can be avoided in band limited filters by choosing very small values of
sampling time. The bilinear transformation is one-to-one mapping and there
is no effect of aliasing.
27.
How one can design digital filter from analog filter?
a. Map the desired digital filter specification into those for an equivalent
analog filter.
b. Derive the analog transfer function for the analog prototype.
c. Transform the transfer function of the analog prototype into an equivalent
digital filter transfer function.
28.
What is bilinear transformation?
The bilinear transformation method overcomes the effect of aliasing that is

causal due to the analog frequency response containing components at or
beyond the Nyquist frequency. The bilinear transform is a method of
compressing the infinite, straight analog frequency axis to a finite one long
enough to wrap around the unit circle only once.
S = (2/T) (Z-1)/(Z+1)
29.
Write merits and demerits of bilinear transformation.
Merits:
1. One to one mapping
2. Due to one to one mapping, there are no possibilities of aliasing.
3. The effect of wrapping will be eliminated by the pre-wrapping
technique
Demerits:
1. The nonlinear frequency effect will introduce frequency distortion,
which is called as warping technique.
2. Using bi-linear transformation a linear phase analog filter cannot be
transformed to a linear phase digital filter.
DTSSP | 37
QP06

30.
What is the main advantage of direct-form II realization when
compared to direct-form I realization?

In direct-form II realization, the number of memory location required is less
than that of direct-form I realization.
31.
What is the main disadvantage of direct-form realization?
The direct-form realization is extremely sensitive to parameter quantization.

When the order of the system N is large, a small change in a filter coefficient
due to the parameter quantization, results in a large change in the location
of the poles and zeros of the system.
32.
What is the advantage of cascade realization?
Quantization errors can be minimized if we realize an LTI system in cascade

form.
33.
Compare analog and digital filter.
34.S.NO
ANALOG FILTER
DIGITAL FILTER
1.
In analog filter both input In analog filter both input and
and output are continues output are discrete time signals.
signal thus it process it
continues signal.
2.
It can be construct by using It can be construct by using
both active and passive adder, multiplier and delay units.
component
3.
It is governed by linear It
is
governed
by
linear
differential equations.
difference equations
4.
These filters can directly It needs some supports from the
interact with analog real A/D, D/A converters.
world.
5.
The frequency response may Here it may change by changing
be changed by changing the its filter co-efficient.
component.
Butterworth and Chebyshev Filter:
S.N
O
BUTTERWORTH FILTER
Magnitude response
decreases monotonically as
1.
frequency increases from 0
to .
2.
Transition band is high.
The
poles
of
the
3.
Butterworth
filter
form
circle.
The
order
of
the
4.
Butterworth filter is more
DTSSP | 38for given specifications.
Co
mp
ar
e
CHEBYSHEV FILTER
Magnitude response decreases
monotonically with certain amount
of ripples, as frequency increases
from 0 to .
Transition band is poor.
The poles of the Chebyshev Filter
forms the ellipse.
The order of the Chebyshev Filter is
less for the given specifications.
QP06

i)
PART B
1. Design a low pass filter using rectangular window by taking I samples of (n)
with cut off
Sequence of 1.2 radians/sec also draws the filter. (16)
[NOV/DEC 2010]
2. The specification of defined low pass filter is :
0.8 |H()| 1.0
; 0 0.2
|H()| 0.2
; 0.32
Design Chebyshevs digital filter using bilinear transformation. (16)

[NOV/DEC 2010]
3. (i) Realize the following using cascade and parallel form.
H(z)= (3+3.6z-1+0.6z-2)/(1+0.1z-1 -0.2z-2)
(12)
(ii) Explain how an analog filter maps into a digital filter in Impulse Invariant
transformation. (4)
[APR/MAY
2011]
4. (i) Using Hanning window, design a filter with
Hd(e-j) = e-j2
-/4 || /4
/4 ||
0
Assume N=7.
(12)
(ii) Write a note on need and choice on windows. (4)
[APR/MAY
2011]
5. Design a filter with desired frequency response
Hd(e-j3) = e-j3
for -3/4 || 3/4
=0
for
3/4 ||
Using Hanning window for N=7
(16)
[NOV/DEC 2011]
6. Design a butter worth filter using the Impulse invariance method for the
following specifications.
0 0.2
0.8 |H(ej)| 1
|H(ej)| 0.2
0.6
(16)
[NOV/DEC 2011]
DTSSP | 39
QP06

7. For the analog transfer function H(s)=2/(s+1)(s+3) determine H(z) using

bilinear transformation. With T=0.1 sec.
(16)
[MAY/JUN 2012]
8. Design an ideal high pass filter with
/4 || <
Hd(ej)={ 1
0
/4
||
using Hamming window with Hamming window with N=11.
(16)
[MAY/JUN 2012]
9. (i) Obtain cascade and parallel realization for the system having difference
equation
y(n)+0.1y(n-1)-0.2y(n-2)=3x(n)+3.6x(n-1)+0.6x(n-2). (8)
[NOV/DEC 2012]
(ii) Design a length-5 FIR band reject filter with a lower cutoff frequency of
2KHz,an upper
cutoff frequency of 2.4 KHz, and a sampling rate of
8000Hz using Hamming window.(8)

[NOV/DEC
2012]
10.
(i) Explain impulse invariant method of designing IIR filter. (6)
(ii) Design a second order digital low pass Butterworth filter with a cut-off
frequency of 3.4KHz at a sampling frequency of 8 KHz using bilinear
transformation. (10)
[NOV/DEC 2012]
11.
Design and realize a digital filter using bilinear transformation for the
following
specifications. Monotonic pass band and stop band -3.01
dB cut off at 0.5 rad magnitude down at least 15 dB
at =0.75 rad.
(16)
[MAY/JUN 2013]
12.
(i) Consider the causal linear shift invariant filter with system function
H(z)= 1+0.875z-1/(1+0.2z-1+0.9z-2)(1-0.7z-1). Draw the structure using a
parallel interconnection of first and second order systems. (8)

[MAY/JUN 2013]
(ii) Consider the following interconnection of a linear shift invariant system.
DTSSP | 40
QP06

X[n]
w[n]
H2(ej)
y[n]
h1[n]
Where x[n]=[n]
h2[n]=[n-1]
H2(ej)={1 ||/2
0 /2<||.
Find the overall impulse response h[n] of the
system.
UNIT V
DIGITAL SIGNAL PROCESSORS
PART A (2 MARKS)
1. How does a digital signal processor differ from other processor?
[APR/MAY 2010] (APR/MAY 2015)
Digital signal processors have the same level of integration, clock

frequencies as general purpose microprocessors .But on signal processing
tasks DSPs overtakes them from 2 to 3 order in speed. This is because of
architectural differences.
The specialized architecture of Digital Signal Processor is:
Arithmetic Unit Architecture,
Specialized units(multipliers, etc)
Regular instruction cycle (RSIC-like architecture)
Parallel processing
Harvard Bus architecture
Internal memory organization
Multiprocessing organization
Local links (TMS320C40)
Memory banks interconnection (TMS320C80)
2. State any two application of DSP. (AU AM 2015)
Applications of DSP are listed below,
Dual-Tone Multi frequency Signal Detection
Spectral Analysis of Sinusoidal Signals
Analysis of Speech Signals
Spectral Analysis of Random Signals
Musical Sound Processing
3. What are the different buses of TMS320C54 processor and list their
functions? [May/Jun 2014]
The C5X architecture has four buses and their functions are as
follows:
a. Program bus (PB):
DTSSP | 41
QP06

It carries the instruction code and immediate operands from program

memory Space to the CPU.
b. Program address bus (PAB):
It provides addresses to program memory space for both reads and
writes.
c. Data read bus (DB):
It interconnects various elements of the CPU to data memory space.
d. Data read address bus (DAB):
It provides the address to access the data memory space.
4. List the various registers used with ARAU. [May/Jun 2014]
The various registers used with ARAU are,
Eight auxiliary registers (AR0 AR7)
Auxiliary register pointer (ARP)
5. What is meant by bit reversed addressing mode? What is the application
for which this addressing mode is preferred? (Nov/Dec 2013)
Bit-reverse addressing is a special type of indirect addressing. It uses one of

the auxiliary registers (AR0AR7) 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.
6. Compare the RISC and CISC processors. (Nov/Dec13)
7. What is BSAR instruction? Give an example.
[Nov/Dec 2010]
BSAR- Barrel Shift Accumulator Right.
This instruction shift the content of Accumulator to the right in single clock input.
Ex: BSAR 12H
8. What are the advantages and disadvantages of VLIW architecture?
[May.June 2013]
Advantages of VLIW architecture
Increased performance
Better compiler targets
Potentially easier to program
Potentially scalable
DTSSP | 42
QP06

Can add more execution units; allow more instructions to be packed

into the VLIW
instruction.
Disadvantages of VLIW architecture?
New kindof programmer/compiler complexity
Program must keep track of instruction scheduling
Increased memory use and high power consumption
9. What is meant by auxiliary register file?
[May/June
2013]
The auxiliary register file contains eight memory-mapped auxiliary registers
(AR0-AR7), which can be used for indirect addressing of the data memory or
for temporary data storage. Indirect auxiliary register addressing allows
placement of the data memory address of an instruction operand into one of
the AR. The ARs are pointed to by a 3-bit auxiliary register pointer (ARP) that
is loaded with a value from 0-7, designating AR0-AR7, respectively.
10.
Define periodogram.
[May/June
2012]
The Peridogram,a non parametric method of power spectrum estimation to

study about the
hidden peridocities in sunspotdata.The average value of peridiogram is
fourier transform of the windowed autocorrelation function.
11.
What is pipelining? What are the different stages in pipelining.

[Nov/Dec 2012]
[Nov/Dec2011]
Pipelining a processor means breaking down its instruction into a series of

discrete pipeline stages which can be completed in sequence by specialized
hardware.
(i)The fetch phase
(ii) The decode phase
(iii)Memory read phase
(iv) The execute phase
12.
What is the function of parallel logic unit in DSP processor.

[Nov/Dec 2012]
DTSSP | 43
QP06

The parallel logic unit (PLU) can directly set, clear, test, or toggle multiple
bits in control/status register pr any data memory location. The PLU provides
a direct logic operation path to data memory values without affecting the
contents of the ACC or the PREG.
13.
Mention the features of DSP processor.
[Apr/May
2011] [Nov/Dec2011]
Architectural features
Execution speed
Type of arithmetic
Word length
14.
Mention one important feature of Harvard architecture.

[MAY/JUNE 2013]
The Harvard architecture has two separate memories for their instructions
and data. It is
capable of simultaneous reading an instruction code and
reading or writing a memory or peripheral.

15.
Briefly explain about multiplier accumulator.
[APR/MAY 2009]
The way to implement the correlation and convolution is array multiplication
Method.
For getting down these operations need the help of adders and multipliers.
The combination of these accumulator and multiplier is called as multiplier
accumulator.
16.
In a non-pipeline machine, the instruction fetch, decode and
execute take 30 ns, 45 ns and 25 ns respectively. Determine the

increase in throughput if the instruction were pipelined. Assume a
5ns pipeline overhead in each stage and ignore other delays.
The average instruction time is = 30 ns+45 ns + 25 ns = 100 ns
Each instruction has been completed in three cycles = 45 ns * 3 = 135ns
Throughput of the machine = The average instruction time/Number of M/C
per instruction
= 100/135
= 0.7407
DTSSP | 44
QP06

But in the case of pipeline machine, the clock speed is determined by the
speed of the slowest stage plus overheads.
In our case is
= 45 ns + 5 ns
=50 ns
The respective throughput is = 100/50

= 2.00
The amount of speed up the operation is = 135/50
= 2.7 times
17.
Assume a memory access time of 150 ns, multiplication time of
100 ns, addition time of 100 ns and overhead of 10 ns at each pipe

stage. Determine the throughput of MAC
After getting
successive addition and multiplications .

The total time delay is 150 + 100 + 100 + 5 = 355 ns
System throughput is
18.
= 1/355 ns.
Write down the name of the addressing modes.
Direct addressing.
Indirect addressing.
Bit-reversed addressing.
Immediate addressing.
i. Short immediate addressing.
ii. Long immediate addressing.
iii.Circular addressing.
19.
Briefly explain about the dedicated register addressing modes.
The dedicated-registered addressing mode operates like the long immediate

addressing modes, except that the address comes from one of two specialpurpose memory-mapped registers in the CPU: the block move address
register (BMAR) and the dynamic bit manipulation register (DBMR). The
advantage of this addressing mode is that the address of the block of
memory to be acted upon can be changed during execution of the program.
20.
What is meant by auxiliary register file?
The auxiliary register file contains eight memory-mapped auxiliary registers

(AR0-AR7), which can be used for indirect addressing of the data memory or
for temporary data storage. Indirect auxiliary register addressing allows
DTSSP | 45
QP06

placement of the data memory address of an instruction operand into one of

the AR. The ARs are pointed to by a 3-bit auxiliary register pointer (ARP) that
is loaded with a value from 0-7, designating AR0-AR7, respectively.
PART B
1. (i) Explain the addressing formats in the DSP processor. (8)
[MAY/JUNE 2012]
(ii)Draw the architecture of DSP processor and explain(8)
[MAY/JUN2012]
2. (i) Explain the functional modes present in the DSP processor. (8)
[MAY/JUN2012]
(ii)Explain about pipelining in DSP (8)
[MAY/JUN2012]
3. Explain various addressing modes of a digital signal processor (16)
[MAY/JUNE 2013]
4. Draw the functional block diagram of a digital signal processor and explain.
(16)
[MAY/JUNE 2013]
5. Explain the addressing modes of a DSP processor. (16)
[NOV/DEC 2011]
6. Describe the Architectural details of a DSP processor.(16)
[NOV/DEC 2011]
7. (i) With a neat diagram explain Von-Neumann architecture. (8)
[NOV/DEC 2011]
(ii) What is MAC unit? Explain its functions. (8)
[NOV/DEC
2011]
8. Explain the architecture of TMS320C50 with a neat diagram. (16)
[NOV/DEC 2011]
9. (i)Draw the block diagram of Harvard architecture and explain (8)
[NOV/DEC2012]
(ii)Explain the advantages and disadvantages of VLIW architecture. (8)
[NOV/DEC2012]
DTSSP | 46
QP06
10.

Write short notes on: (16)
[NOV/DEC2012]
(i)Memory mapped register addressing
(ii)Circular addressing
(iii) Auxiliary registers
11.
Explain in detail the architectural features of a DSP processor.(16)
[APR/MAY 2011]
12.
Explain the addressing formats and functional modes of a DSP
processor.(16)
[APR/MAY 2011]
DTSSP | 47

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QP06

Subject Code / Name: EE6403 DISCRETE TIME SYSTEMS

Any relaxed system is said to be bounded input bounded output (BIBO)

Subject Code / Name: EE6403 DISCRETE TIME SYSTEMS

4. State Shannons sampling theorem.

(NOV/DEC 2011][APR/MAR 2011)

rate and fundamental frequency of the signal?

The energy of a discrete time signal x(n) is defined as

Here E= and P= Finite. Therefore the given signal is a power signal.

8. Determine fundamental period of the signal

Solution: Fundamental period, N=

Subject Code / Name: EE6403 DISCRETE TIME SYSTEMS

It is the minimum rate at which a signal can be sampled and still

Let us consider a band limited signal x(t) having no frequency component

Define even and odd signals.

Subject Code / Name: EE6403 DISCRETE TIME SYSTEMS

So the above system is time - variant system.

Discrete time system

What is static and dynamic system?

Subject Code / Name: EE6403 DISCRETE TIME SYSTEMS

Example: y(n) = x(n) + 3x(n-1)

What is an anti-aliasing filter?

What is a Causal system?

The average power of a discrete time signal x(n) is defined as

Subject Code / Name: EE6403 DISCRETE TIME SYSTEMS

1. (i)Find the impulse response of a discrete time invariant system whose

(ii)Explain the properties of discrete time system.(4)

5. Determine the response of the following systems to the input signal .

Find the even and odd components of given x(n).(16)

6. A discrete time system can be

Examine the following system with respect to the properties above

(ii) Determine whether the following is an energy signal or power signal.(8)

Subject Code / Name: EE6403 DISCRETE TIME SYSTEMS

time domain and in frequency domain.(16)

Compute linear and circular convolution of the two sequence

(ii)Check whether the following are energy or power signals (8)

determine the expression for quantization linear. (10)

Subject Code / Name: EE6403 DISCRETE TIME SYSTEMS

(ii) Check whether the following are periodic. (6)

(ii) Explain the classification of discrete signal. (10)

DISCRETE TIME SYSTEM ANALYSIS

1. Determine the Z-transform of

The given x(n) =

has the limit - to . It is split into two as shown

2. Find the DFT of the sequence

The N-point DFT of x(n) is given by,

DFT{ x(n)} = X(k) =

Since x(n) is a 4-point sequence,

3. What is meant by region of convergence? State its properties.

Subject Code / Name: EE6403 DISCRETE TIME SYSTEMS

is a ring or disk in the Z plane centered at the origin.

4. Given a difference equation y(n) = x(n)+3x(n+1)+2y(n-1).

Determine the system function H(z).

On taking Z- transform, Y(Z) = X(Z)+3Z-1 X(z)+2Z-1Y(z)

The condition for the system to be stable is,

In the given system ,

Therefore the given system is stable.

6. What are the properties of z- transform ?

Scaling in z- domain : z[a n x(n)]= X (a 1 z )

Subject Code / Name: EE6403 DISCRETE TIME SYSTEMS

7. Find the convolution for x(n)={0,1,2} and h(n)={2,0,1}.

The commutative and distributive properties of convolution are,