DTSSP 2mark Q&A

EE6403-Discrete Time signals and Signal Processing
Unit 1 TWO MARK QUESTIONS

1. Differentiate analog and digital signal.
Analog signals are defined for every value of time and they take on values in
the continuous interval (a,b) where ‘a’ can be -∞ and ‘b’ can be ∞.
Ex : x(t)=cos t ; -∞<t< ∞.
Discrete time signals are defined only at certain specific values of time.
Ex : x(n)= 0.8n, for n≥0
0 else
2. A DT signal x(n) is defined as x(n)= {2,3,4,5,6}
�
Sketch the signal x(-n+4).
x(n)= {2,3,4,5,6}
�
x(-n)={6,5,4,3, 2}
�
x(-n+4)={6,5,4,3,2}
�
3. Distinguish between continuous valued and discrete valued signals.

If a signal takes on all possible values on a finite or an infinite range, it is said
to be continuous valued signal.
If a signal takes on values from a finite set of possible values, it is said to be
discrete-valued signal.
4. Explain the operation of folding and time delaying.
In a signal x(n), replacing the independent variable ‘n’ by’-n’ results in
x(-n). The result of this operation is a folding or a reflection of the signal about
the time origin n=0.
A signal x(n) may be shifted in time by replacing the independent variable ‘n’
by ‘n-k’, where k is an integer. If ‘k’ is a positive integer, the time shift results in a
delay of the signal by k units in time.
x(n)= {2,1,3,4,2} & x(-n)= {2,4,3,1,2} & x(n-2)={2,1,3,4, 2}
� � �
5. Explain the operation of time advancing.

A signal x(n) may be shifted in time by replacing the independent variable ‘n’
by n-k, where k is an integer. If ‘k’ is a negative integer, the time shift results in an
advance of the signal by k units in time.
x(n)= {2,1,3,4,2} & x(n+2)={2,1,3,4,2}
� �
6.When a discrete time signal is said to be symmetric (or) anti symmetric?

A real valued signal x(n) is called symmetric (even) if x(-n)=x(n).
A real valued signal x(n) is called anti symmetric (odd) if x(-n)= -x(n).
7. Give the mathematical representation of Unit sample sequence & Unit step
sequence.
Prepared by Prof.S.Nagammai, HOD/EIE,KLNCE 242

The unit sample (impulse) sequence is denoted as d (n) and is defined as
�1; for n =0
d ( n) = �
�0; for n �0
The unit step sequence is denoted as u(n) and is defined as
1; for n�0
�
u ( n) = �
0; for n<0
�
8. What is a causal system? Give an example.
A system is said to be causal if the output of the system at any time ‘n’ depends
only on present and past but does not depend on future inputs. In mathematical
terms, the output of a causal system satisfies an equation of the form,
y(n)=F[x(n),x(n-1),x(n-2)…….]
Example : y(n)=x(n)-x(n-1)
9.Test the stability of the system whose impulse response h(n)=(1/2)n u(n).
�
The condition for stability is � h( n ) < �
n =-�
� n 1
�1 2
n=0
=
1 - 0.5
= 2 <�
Hence the system is stable.

10.What is meant by static system?
A discrete time system is called static or memory less if its output at any
instant ‘n’ depends at most on the input sample at the same time, but not on past
or future samples of the input.
11.Test the following system for time in variance.
y(n)=n x2(n)
The response y(n) to delayed excitation is,
y(n-k)=n x2(n- k )…………..(1)
The delayed response is,
y(n- k)=(n- k) x2(n- k)……..(2)
Eqn. (1) is not equal to Eqn. (2). Hence the given system is time varying.
12.Define stability of discrete Time system.
A linear time invariant system is said to be stable, if its impulse response is
�
absolutely sum able. (ie)., � h ( n) < �
n =-�
13.Define BIBO stability.

A system is said to be bounded input bounded output (BIBO) stable, if and only
if every bounded input produces a bounded output.
14.What is meant by recursive and non recursive system?

A causal recursive system is described by an input-output equation of the form

y(n)=F[y(n-1),……..y(n-N), x(n)…….x(n-M)]
A causal non-recursive system does not depend on past values of the output and
described by an input-output equation of the form
y(n)=F[x(n), x(n-1),……x(n-M)]
15.Write the recursive and non-recursive difference equations.
The recursive system is described by the difference equation of the form
N M
y (n) =-�a k y(n-k) + �bk x(n-k)
k =1 k =0
The output at any instant depends upon the present and past values of input
and past values of output. The non-recursive system is described by the
M
difference equation of the form y (n) = �bk x(n-k)
k =0
The output at any instant depends upon the present and past values of input
alone.
16.Determine the stability of the system, whose transfer function
Z2
H (Z ) =
Z 2 - 4Z + 3
Z2
Writing H(Z) in factored form, H (Z ) =
( Z - 3)( Z - 1)
The system function has one pole at z=1 and another pole at z=3, which lies
outside the unit circle. Hence, the system is unstable.
�2 �
17.Find the periodicity of x(n) = cos � �n
�7 �
2
w0 =
7
1
f 0 = ( irrational number )
7
Hence, x(n) is aperiodic signal.
18.Determine the range of values of the parameter ‘a’ for which the LTI system
with impulse response h(n)=an u(n) is stable.
�
For stability, � h( n) < �
n =-�
�
1
�a
n
Hence, = <�
n=0 1- a
The above geometric series converges, if |a|<1.
Therefore, the system is stable if |a|<1.

�2 n �
- j� �
19.Find the fundamental period for the sequence
g (n) = e �3 �
2
w0 =
3
1
f 0 = ( rational number )
3
Hence, the fundamental period is, N=3 secs.
20.Find the signal energy of x(n) = (1/2)n u(n).
Energy = lim E N
N ��
N
where, E N = �|x(n)|
n = -N
2
� � �
1
Energy = �|x(n)|2 = �|1/2|2n =
n=0 n=0
�|1/4|
n=0
n
=
1-0.25
= 1.33 J
�4 n � �2 n �
j� � j� �
21.Determine the fundamental period of the signal x ( n) = 1 + e �8 �
-e �5 �
4 2
w1 = and w2 =
8 5
f1 = 2/8 (rational number)
f2 = 1/5 (rational number)
Therefore, f1 = ¼ = k/N1 implies N1=4
f2 = 1/5 = k/N2 implies N2=5
The ratio of fundamental period is
N1/N2 = 4/5 = 0.8 (rational number)
Hence, x(n) is periodic with fundamental period 20secs .
22. List the properties of discrete time sinusoids.
 A discrete time sinusoid is periodic only if its frequency f is a rational
number.
 Discrete time sinusoids whose frequencies are separated by an integer
multiple of 2  are identical.
 The highest rate of oscillation in a discrete time sinusoid is attained when
w =  (or -  ) or f = 1/2 (or -1/2).
23. Determine the signal y(n) = x(n-1) for the input signal
�n for - 3 �n �3
x ( n) = �
�0 else
Thus, x(n) = {3,2,1, 0,1,2,3}
�
y(n) = x(n-1) = {3,2,1,0,1,2,3}

�

24. Determine the odd & even parts of the DTS x(n)
x(n) = {1,0.5,0.5,1,1,1,1,0.5}
�
The odd signal component is,

1
X 0(n) = [x(n)-x(-n)]
2
x(-n) = {0.5,1,1,1, 1 ,0.5,0.5,1}
�
1
Therefore, x 0 (n) = {-0.5,-0.5,-0.5, 0,0.5,0.5,0,0.5}
2 �
The even signal component is,

1
xe(n) = [x(n)+x(-n)]
2
1
= {0.5,2,1.5,1.5, 2,1.5,1.5,2,0.5}
2 �
25. What is the condition that the signal x (n) = e an u ( n) to be an energy signal?
Energy = lim E N < �
N ��
N
where, E N = �|x(n)|
n=-N
2
� 2an
Energy = � e
n=0
1
Energy = <�
2a
1-e
The above series converges if e2a < 1.
26. Compute the energy of the signal
�n 0 �n �5
�
x ( n) = �10 - n 5 �n �10
�0 else
�
x(n) = {0,1,2,3,4,5,4,3,2,1,0}
10
It is aperiodic, hence energy = �[ x(n)]
n=0
2
= 2+8+18+32+25 = 85 Joules.
27.What is the overall impulse response of h1(n) & h2(n) when they are in
a)series b) parallel?

28. Find the energy and power of the signal x (n) = 2e j 3n for all n.
1 N 2
Power = lim � |x(n)|
N��2 N + 1 n=-N
1 N 4
= lim � 4 = lim ( 2 N + 1) = 4 watts
N��2 N + 1 n=-N N��2 N + 1
29.Determine whether the ramp signal is an energy or power signal.
x(n) = n u(n)
�n n �0
x ( n) = �
�0 else
N 2 N ( N + 1)(2 N + 1)
energy = lim � n = lim =�
N��n=0 N�� 6
1 N 2 1 � N ( N + 1)(2 N + 1) �
Power = lim � n = lim =�
N��2 N + 1 n=0 �
N ��2 N + 1 � 6 �
�
Hence, ramp signal is neither energy nor a power signal.
� n
�1 �
�� n �0
30. Determine the energy of the signal x ( n) = ��2 �
�n
�3 else
� �1 � 2n -1 � �1 � n � �1 � n
2n
Energy = � � � + � ( 3) = � � � + � � �
n = 0 �2 � n = -� n = 0 �4 � n = 1�9 �

1
1 4 9
Energy = + 9 = + = 2.4 Joules
1 1 3 8
1- 1-
4 9
31. Determine the energy of the signal x (n) = 3n u (- n - 1)

�
3n n �-1
x ( n) = �
0
� n �-1
1
-1 � n
�1 � 1
Energy = � (3)2n = � � � = 9 = Joules
n =-� n =1�9 � 1 - 1 8
9
�2 �
32. Determine x(n) = 2 Cos � � n is energy or power signal.
�5 �
2 1 K
w0 = � f0 = =
5 5 N
Thus, x(n) is periodic with fundamental period, N=5sec
1 N-1 2 1 4
Power = � x(n) = � 4= 4 w
N n=0 5 n=0
.Define energy and power signals .33
,The energy E of a signal x(n) is defined as
�
Energy = � |x(n)|2
n = -�
.If E is finite, then x(n) is called an energy signal
,The average power of a discrete time signal x(n) is defined as
1 N 2
Power = lim � |x(n)|
N��2N+1 n = -N
.Define periodic and aperiodic signals .34
.A signal x(n) is periodic with period N if and only if x(n+N) = x(n) for all n
.The smallest value of N for which above equation holds is called the period
If there is no value of N that satisfies the above condition, the signal is called
.aperiodic
Plot the sequence x(n) = u(n+1)-u(n-3).35
1; n �-1
�
u (n + 1) = �
0; n < -1
�
1; n �3
�
u (n - 3) = �
0; n < 3
�
?What is meant by dynamic system.36

A discrete time system is called dynamic if its output at any instant ‘n depends on
the present & past samples of input, & the dynamic system is said to have
.memory
g (n) = x (n) Check for linearity for
Given g (n) = x(n)
The output g(n) for an arbitrary input x1(n) is
g1 ( n) = x1 ( n)
And the output g(n) for input x2(n) is
g 2 ( n ) = x2 ( n )
Thus g (n) = g1 (n) + g 2 (n) = x1 ( n) + x2 (n) -----------(1)

When the inputs are applied simultaneously then,
g (n) = x1 (n) + x2 (n) ------------(2)
As an equation (1) is not equal to the equation (2) the given system is non-linear
37. Is the system y(n) = ln[x(n)] is linear?
Given y(n) = ln[x(n)]
The o/p y(n) for an arbitrary i/p x1(n) is
y1(n) = ln[x1(n)]
And the o/p g(n) for an i/p x2(n) is
y2(n) = ln[x2(n)]
Thus y(n) = y1(n) + y2(n) = ln[x1(n)] + ln[x2(n)]-----------(1)
When the i/ps are applied simultaneously then,
y(n) = ln[x1(n)+x2(n)]------------(2)
As an equation (1) is not equal to the equation (2) the given system is non-
linear.
38.What is quantization& quantization error?
Quantization is the process of conversion of discrete time continuous valued
signal into a discrete time discrete valued signal.
The difference between actual output and quantized output is called quantization
error.
.State the need for antialiasing filter.39
An anti-aliasing filter 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
40.What is meant by aliasing? How can it be avoided?
When the analog signal is sampled at a rate less than the Nyquist rate (fs<2fm),
multiple folding of the frequency axis of the frequency variable ‘F’ for the analog
signal occurs. This effect is known as aliasing. This can be avoided by sampling the
analog signal at a rate fs>2fm.

State Shannon’s sampling theorem.41

A band limited continuous time signal, with highest frequency ‘fm’ Hertz, can be
uniquely recovered from its samples provided that the sampling rate Fs≥2fm
.samples per Second
.Distinguish between Deterministic and Random signals .42
A signal is said to be deterministic if its future value can be predicted from the
.knowledge of past and present values of the signal without uncertainty
Analyze whether the system is .43
.static or dynamic
The output at any instant depends on the past value of the input. So the system is
.dynamic
44.Define shift invariant system.
A relaxed system is shift invariant if and only if x(n)→y(n) implies that
x(n-k) →y(n-k) for every input signal x(n) and every time shift ‘k’.
45.Whether the system defined by impulse response h(n) = 2n u (-n) + 2- n u (n) is
causal?
h( n) = 2 n u ( - n ) + 2 - n u ( n)
As the impulse response is defined for negative values of input the system is non
.causal
UNIT – 2 (TWO MARK QUESTIONS)
n
�1 �
1. Determine the Z – transform of � � �u ( n ) - u ( n - 8) �
� �& indicate its ROC.
�2 �
� n
�1 � 0 �n �7
�
��
x (n) = �
�2 �
�0 else
�
8
7 7 �1 �
n n 1- � �
�1 � -n �1 � �2z �
X ( Z ) = � � �Z = � � � =
�2 � �2z � 1
n =0 n =0 1-
2z
( 2z ) - 1
8
1
X (Z) = ROC : <1� Z > 1
( 2z ) ( 2z - 1)
7
2z 2
2. Define ROC in Z – transform

The set of values of Z for which X(z) attains finite value is defined as ROC of X(z)
3. Determine the Z – transform of x ( n ) and indicate its ROC.
x ( n ) = { 1,2,5,7,0,1}
�
X ( n ) = d ( n + 2) + 2d ( n + 1) + 5d ( n ) + 7d ( n - 1) + d ( n - 3 )
X ( z ) = z 2 + 2z + 5 + 7z -1 + z -3

ROC is entire Z plane expect Z=0 and Z= �

4. Define linear convolution & circular convolution
�
The linear convolution sum is given by y ( n ) = x ( n ) * h ( n ) = � x ( k ) h ( n - k )
k = -�
N -1
The circular convolution is y ( m ) = x ( n ) * h ( n ) = � x ( n ) h ( ( m - n ) ) N
n =0
5. Define Z – transform
The Z transform is a complex valued function of a complex variable Z.
�
( ie ) X ( Z ) = ZT �
x (n) �
� �= � x ( n ) Z
-n
n = -�
6. State the initial & final value theorems of Z – transform
Lt
Initial value theorem : x ( 0 ) = X (z)
Z ��
Lt
Final value theorem: x ( �) =
Z �1
( 1 - z -1 ) X ( z ) provided all poles of
( 1 - z ) X ( z ) lies inside the unit circle.
-1
7. Compute Fourier transform of the signal x ( n ) = u ( n ) - u ( n -1)

1
� n =0
x ( n ) = u ( n ) - u ( n - 1) = �
0
� else
N -1
-j
2 kn
K = 0,1.....N - 1
X (K) = � x (n) e N
; where
N = 1, \ K = 0
n =0
�
= 1 e -0 = 1
n =0 8. Write the relationship
between system function and the frequency response of LTI system.
The frequency response H(w)of the system is obtained by evaluating H(z) on the
unit circle.
�
H ( w) = H ( Z ) | = � h ( n ) e - jwn
z =e j w
n = -�
Where H(Z) is the system function.
9. What are the different methods of evaluating inverse Z – transform.
Direct evaluation by contour integration
Expansion into a series of terms in the variables Z & Z-1
Partial fraction expansion method.
Convolution method

10. Find the convolution of the following using Z-transform
x ( n ) = { 1,2,1} h ( n ) = { 1,1,1}
� �
2
X ( z) = � x ( n ) Z -n = 1 + 2z -1 + z -2
n =0
2
H ( z) = � h ( n ) Z -n = 1 + z -1 + z -2
n =0
Y ( z ) = X ( z ) H ( z ) = ( 1 + 2z -1 + z -2 ) ( 1 + z -1 + z -2 )
= 1 + 2z -1 + z -2 + z -1 + 2z -2 + z -3 + z -2 + 2z -3 + z -4 = 1 + 3z -1 + 4z -2 + 3z -3 + z -4
y ( n ) = { 1,3,4,3,1}
�
11. List any two properties of discrete time Fourier transform.
1)linearity:
If x1(n) DTFTX1(w) & X2(n) DTFT X2(w)
Then a1x1(n)+a2x2(n) DTFT a1X1(w)+a2X2(w)
2)time shifting:
If x(n) DTFT X(w) then
X(n-k) DTFT e-j w k X(w)
12. How can you find step response of a system if the impulse response h(n) is
known?
n
The step response of the system is s(n ) = � h k( )
k = -�
13. State the properties of Linear convolution
Commutativity: x(n)*h(n)=h(n)*x(n)
Associativity: x(n)*[y(n)*z(n)]=[x(n)*y(n)]*z(n)
Distributivity: x(n)*[y(n)+z(n)]=[x(n)*y(n)]+[x(n)*z(n)]
14. What is the Z-transform of discrete unit step function?

1
� n >= 0 �
x(n) = u(n) = � �
0 � n <0
� � n
�1 � Z
X(z) = � z = � � � = -n
�Z � Z -1
n =0 n =0
15. state differentiation theorem of Z-transform
z
If x(n) �� X ( z ) then
d
z
nx(n) �� -z X ( z)
dz
x ( n ) = { 1,1,1,1} . h ( n ) = { 1,2,2,1}
16. Find the convolution sum for
� �

1 1 1 1
1 1 1 1
2 2 2 2
2 2 2 2
1 1 1 1
y (n) = { 1,3,5, 6,5,3,1 }

�
17. Define Parsevel’s relation for discrete time signals.
�  2
2 1
X (w ) dw
2 -�
� x (n ) =

n = -�
18. Write down the time reversal property of discrete time Fourier transform
If x(n) DTFT X( w ) then
x(-n) DTFTX(- w )
19. Show the region of convergence for the Z-transform of the sequence
x ( n ) = a n u ( n ) + b n u ( -n - 1) when |b |>|a |.
� n -1 n
�a � �b �
X (Z ) = � � � - � ��
n = 0�Z � n = -��Z�
� n � n
�a � �Z �
X (z ) = � � �- � � �
n = 0�Z � n = 1�b�
a
for convergence , <1 or Z > a
Z
Z
<1 or Z < b
b
ROC of X ( Z ) is a < Z < b
20. What are the four steps to obtain the convolution.

�
Y(n) = � x(k)h(n-k)
k = -�
1) h(k) is folded to get h(-k)
2) h(-k) is time shifted by n0 units
3) h(n0-k) is multiplied with x(k)
4) summation
21. Discuss the stability of the system described by
1
H (z) =
( 1 - 0.5Z ) ( 1 + 0.5Z -1 ) ( 1 - 0.25Z -1 )
-1
16z 3
H (z ) =
( 2z - 1) ( 2z + 1) ( 4z - 1)

z3
H ( z) =
( Z - 0.5 ) ( z + 0.5 ) ( z - 0.25 )
The system function has multiple zeros located at z=0 and Poles located at
z=1/2,-1/2&1/4 all poles and zeros lies inside the unit circle, hence the system
is stable .
22.Define DTFT and IDTFT of a sequence?
The DTFT (Discrete Time Fourier Transform) of a sequence x(n) is defined as
�
X (w ) = �x (n )e
n =-�
- j wn

1
The IDTFT is defined as x(n)= = X (w ) e
�
- j wn
dw
2 -
23.Compute the Z-transform of x(n) = a nu ( n - 1)

a
� � n �
�a � a
X ( z ) = �x( n) z = �a z = �� = z =
-n n -n
n =-� n =1 n =1 �z � 1- a z - a
z
24.Given x(n)=u(n-1).Determine the z transform and ROC.
1
� � n�
�1 � 1
X ( z ) = �x( n) z - n = �z - n = �� = z =
n =-� n =1 n =1 �z � 1 - 1 z -1
z
25.What are the properties of ROC?
i)The ROC cannot contain any poles
ii)The ROC must be a connected region
iii) The ROC of an LTI stable system contains the unit circle
iv) The ROC is a ring in the Z plane centered ay the origin
26.State initial and final value theorem of Z transform.
Lt
Initial value theorem : x ( 0 ) = X (z)
Z ��
Lt
Final value theorem : x ( �) =
Z �1
( 1 - z -1 ) X ( z )
27.Compute the Z-transform and ROC of the following finite duration signal
x ( n ) = { 1,2,5,7,0,1}
�
X ( n ) = d ( n + 2) + 2d ( n + 1) + 5d ( n ) + 7d ( n - 1) + d ( n - 3 )
X ( z ) = z 2 + 2z + 5 + 7z -1 + z -3
ROC is entire Z plane expect Z=0 and Z= �
28. Compute the Z transform and ROC of d ( n - k ) .
-k
X ( Z ) = � z -n = z
n =k
ROC is entire Z plane except z=0

29.Compute the convolution of the two sequences using Z transform

x ( n ) = { 1,2,1} h ( n ) = { 1,1,1}
� �
2
X ( z ) = � x ( n ) Z -n = 1 + 2z -1 + z -2
n =0
2
H (z) = � h ( n ) Z -n = 1 + z -1 + z -2
n =0
Y ( z ) = X ( z ) H ( z ) = ( 1 + 2z -1 + z -2 ) ( 1 + z -1 + z -2 )
= 1 + 2z -1 + z -2 + z -1 + 2z -2 + z -3 + z -2 + 2z -3 + z -4 = 1 + 3z -1 + 4z -2 + 3z -3 + z -4
y ( n ) = { 1,3, 4,3,1}
�
30. Draw the pole-zero plot of Z-transform of x(n) =(a n )u (n) .
� n
�a � Z
X (Z ) = � ��=
n = 0�Z� ( Z -a)
It has a zero at z=0 and pole at z=a
Unit- III (TWO MARK QUESTIONS)
1.Calculate the number of multiplications and addition needed in the

calculation of DFT and FFT with 64 point sequence.
DFT FFT
No of additions needed -- 64x63=4032 64 log 64=384
No of multiplications needed—642= 4096 (64/2) log 64=192
2.What is DIT radix2 algorithm.

The radix 2 DIT FFT is an efficient algorithm for computing DFT.The idea
is to break N point sequence in to two sequences ,the DFT of which can be
combined to give DFT of the original N-point sequence. Initially the N point
sequence is divided in to two N/2 point sequences ,on the basis of odd and even
and the DFTs of them are evaluated and combined to give N-point sequence.
Similarly the N/2 DFT s are divided and expressed in to the combination of
N/4 point DFTs. This process is continued until we left with 2-point DFTs
3. What is DIF radix2 algorithm.
The radix 2 DIF FFT is an efficient algorithm for computing DFT. In this the
output sequence X(k) is divided in to smaller and smaller. The idea is to break N
point sequence in to two sequences ,x 1(n) and x2(n) consisting of the first N/2

points of x(n)and last N/2 points of x(n) respectively. Then we find N/2 point
sequences g1(n) and g2(n).
g1(n) =x1(n)+x2(n)and g2(n)= (x1(n)+x2(n))WNn .Similarly the N/2 DFT s are

divided and expressed in to the combination of N/4 point DFTs. This process is
continued until we left with 2-point DFTs
4.What is FFT?
FFT is a method for computing the DFT with reduced number of
calculations using symmetry and periodicity properties of twiddle factor W kN .
The computational efficiency is achieved by decomposing of an N-point DFT
into successively smaller DFTs to increase the speed of computation.
5.What are the differences between DIT and DIF algorithms?
For DIT the input is bit reversed and the output is in natural order ,and in DIF
the input is in natural order and output is bit reversed. In butterfly the phase
factor is multiplied before the add and subtract operation but in DIF it is
multiplied after add-subtract operation
6.Determine the circular convolution of the sequence

x1 ( n ) = { 1,2,3,1} & x 2 ( n ) = { 4,3,2,2}
x1 ( n ) = { 1,2,3,1} & x 2 ( n ) = { 4,3,2,2}
x3 ( 0) �
� 1
� 1 3 2��
4
� � � ��
x 3 ( 1) �
� 2 1 1 3��
3
= �
x 3 ( 2) �
� �
3 2 1 1 ��
2
� � � ��
x 3 ( 3) �
� 1
� 3 2 1 ��
2
x 3 ( n ) = x1 ( n ) e x 2 ( n ) = ( 17,19,22,19 )
7.Write briefly about over lap – add method.
i)The longer sequence x(n) is partitioned into non-over lapping sub sequences of
length L
ii)M-1 zeros are appended to each data block of size L to make the data of size
L+M-1.
iii)L – 1 zeros are added to the impulse response h(n) to make the data of size
L+M – 1 & circular convolution is performed.
iv)The last M – 1 points from each output block is overlapped & added to the
first M – 1 points of the succeeding block.
8.Draw the basic butterfly structure of DITFFT and DIF FFT algorithms.
The basic radix – 2 DIT butterfly diagram is,

The basic radix – 2 DIF butterfly diagram is,
9. Compare linear and circular convolution.

linear convolution circular convolution
1.If x(n)is a sequence of length L &h(n) 1.If x(n)is a sequence of length L
of length M, then y(n) is of length &h(n) of length M, then y(n) is of
N=L+M-1 length N=L=M
2.Linear convolution can be used to find 2.Linear convolution cannot be used

the response of the linear filter. to find the response of the linear
filter.
3.zero padding is not necessary to find 3.zero padding is necessary to find
the response of a linear filter. the response of a linear filter.
10) Find the DFT of a sequence x ( n ) = { 1,1,0,0}
N -1 j 2 kn 3 j 2 kn
-( ) -( )
X (k) = � x (n) e N = � x (n) e 4 where N = 4 and k = 0,1, 2, 3
n =0 n =0
- j k j 3 k
- j k -
X ( k ) = x (0) + x (1)e 2 + x (2)e + x (3)e 2
X (0) = x (0) + x (1) + x (2) + x (3) = 2

j
-
2 =1- j & - j
X (1) = x (0) + x (1)e X (2) = x (0) + x (1)e =0
3
-j
X (3) = x (0) + x (1)e 2 =1+ j X ( k ) = { 2,1 - j ,0,1 + j }
11)List any two properties of DFT
Time shifting:
DFT
If x ( n) � X ( k )
DFT - j 2 kn0
Then,
x(n - n0 ) � X (k ) e N

Periodicity:
DFT
If x ( n) � X ( k )
Then, x ( n + N ) = x (n) for all ' n '
X (k + N ) = X (k ) for all ' k '

12)Define N point DFT pair
N -1
j 2 kn
DFT: X(k) = � x(n) - N ,k=0,1,2……..,N-1
e
n =0
N -1
1 j 2 kn
IDFT: x(n) = � X(k) , n=0,1,2,………N-1
N e N
n =0
13)Briefly discuss the relationship of the discrete Fourier transform to the Z-
transform.
The Z-transform of a sequence x(n) is
�
( ie ) X ( Z ) = � x ( n ) Z -n
n = -�
where Z = re jw
�
X (Z ) Z = re jw = � [ x (n )r -n ] e - jwn
n = -�
if X (Z )converges for Z =1, then
�
X (w ) = � [ x (n )] e - jwn
n = -�
The Fourier transform is the Z transform of the sequence evaluated on the unit
circle.
14) Give the advantages and applications of FFT algorithm.
Advantages:
i) Computation of DFT with reduced no. of calculations.
ii)As the computations are performed in phase, the number of memory

locations required is 2N.

Applications:
1) linear filtering
2) Correlation
3) Spectrum analysis
15) What is radix-2FFT?
In an N point sequence, N is expressed as r m and the given sequence is

decimated into r point sequences. In radix 2 FFT, the N point sequence is
decimated into 2 point sequences and then 2 point DFT is computed.
16) what is meant by in place computations in DIT and DIF algorithms?
An algorithm in which same memory devices are used to store input and output
data’s is known as in place computations in DIT and DIF algorithms.
17.What is zero padding? What are its uses?
Appending zeros to a sequence in order to increase the size or length of the
sequence is called Zero padding. In circular convolution the length of the
sequences must be same. If the length of the sequences are different they can be
made equal by zero padding.
18.How will you perform linear convolution using circular convolution?
i)Pad the sequences h(n) & x(n) with zeros so that they are of length N = N 1+N2-
1.
ii)Find the N-point DFT of h(n) & x(n)
iii)Multiply the DFT’s to form the product Y(K) = X(k) H(k).
iv)Find the inverse DFT of Y(k) that results in Y(n).
19)What are twiddle factors of the DFT?
The N point DFT is given by,
N -1 - j 2 kn
X (K ) = � x (n )e N , k=0,1,2……..,N-1
n =0
- j 2
Let which is called as twiddle factor.
wN = e N
N -1
X (K ) = � x (n )wN kn
n =0
20.How many multiplication and additions are required to compute N point
DFT using radix 2 FFT?
No of additions required to compute N point DFT using radix 2 FFT is,
N log 2 N
No of multiplications required to compute N point DFT using radix 2 FFT is,
N
log 2 N
2

21.Compute the IDFT of using DIF FFT algorithm.
x(n) = [1,0, 0,1]

22. Compute the IDFT of using DIT FFT algorithm.
x(n) = [1,0, 0,1]

23.State the differences between DIT and DIF algorithms.
i. In DIT the input is in bit reversed order while the output is in normal order.
In DIF the input is in normal order while the output is in bit reversed order.
ii. Considering the butterfly diagram, in DIT the complex multiplication takes
place before the add subtract operation. While in DIF the complex multiplication
takes place after the add subtract operation
24. Draw the flow graph of the first stage DIT FFT algorithm for N=8.

25.Calculate the multiplication reduction factor in the calculation of DFT and

FFT with 1024 point sequence.
DFT FFT
No of multiplications needed—10242= 1048576 (1024/2) log 1024=5120
Speed improvement factor= 1048576/5120=204.8
26.Distinguish linear convolution & circular convolution.
linear convolution circular convolution

1.If x(n)is a sequence of length L &h(n) 1.If x(n)is a sequence of length L &h(n)
of length M, then y(n) is of length of length M, then y(n) is of length
N=L+M-1 N=L=M
2.Linear convolution can be used to 2.Linear convolution cannot be used to

find the response of the linear filter. find the response of the linear filter.
3.zero padding is not necessary to find 3.zero padding is necessary to find the
the response of a linear filter. response of a linear filter.
27.Draw the flow graph of a 4 point radix- 2 DIT-FFT butterfly structure for
DFT.
28.Calculate the percentage saving in calculation in a 256 point radix-2 FFT

when Compared to direct FFT.
DFT FFT
No of additions needed 256x255=65280 256 log 256= 2048
No of multiplications needed 2562=65536 (256/2) log 256=

1024
Hence the percentage saving in addition is, 100 – (2048/ 65280 ) = 97%
Hence the percentage saving in multiplication is,100 – (1024/65536) = 98.5%
29.State circular frequency shift property of DFT.
Circular frequency shifting:
DFT
If x ( n) � X ( k )

j 2 ln DFT
Then, x( n) e N
� X ( (k - l ) ) N
30.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.
X (k ) = X * ( N - k )
With N=8,
X (5) = X * (3) = 0
X (6) = X * (2) = 0.125 + j 0.3018
X (7) = X * (1) = 0
31.What is meant by in place computation in FFT algorithm?
An algorithm in which same memory devices are used to store input and output
data’s
is known as in place computations in DIT and DIF algorithms.
UNIT-4 TWO MARK QUESTIONS
1. List the properties of Butterworth low pass filters?
The magnitude response of the Butterworth filter decreases monotonically as

the frequency W increases from zero to infinity.
The magnitude response of the Butterworth filter closely approximates the ideal
response as the order N decreases . The poles of the Butterworth filter lies on a
unit circle.
2. What are properties of Chebyshev filter?
The magnitude response of the Chebyshev filter exhibits ripple either in pass
band or in stop band according to the type.
The poles of the Chebyshev filter lies on an ellipse.
3. Distinguish between the frequency response of chebyshev type 1 filter for

N odd and N even?
The frequency response curve starts from unity odd values of N, and starts
1
from for even values of N.
1+ e2

4. Distinguish between the frequency response of chebyshev type -1 and type

-2 filters?
Type 1 chebyshev filters are all pole filters that exhibits equiripple behavior in
the pass band and monotonically characteristics in the stop band. On the other
hand, the family of type-2 chebyshev filter contains both poles and zeros and
exhibits a monotonic behavior in the pass band and an equiripple behavior in
the stop band
5. What are properties that are maintained same in the transfer of analog filter
into a digital filter?
i). The j W axis in the s-plane should map into the unit circle in the z-plane.
Thus there will be a direct relationship between the two frequency variables in
the two domains.
ii). The left half of the s-plane should map into the inside of the unit circle in the
z-plane .Thus a stable analog filter will be converted to a stable digital filter.
6. What is the mapping procedure between s-plane and z-plane in the method
of mapping using IIT? What are its characteristics?
The mapping procedure between s-plane and z-plane in the method of

mapping of using IIT is given by
1 1
�
s - pi 1 - e pi T Z -1
The above mapping has the following characteristics
i). The left half of s-plane maps inside the unit circle in the z-plane
ii). The right half of s-plane maps outside the unit circle in the z-plane

iii). The j W axis maps on to the perimeter of the unit circle in the z-plane
7. By impulse invariance method obtain the digital filter transfer function of

1
analog filter H ( S ) =
S +1
1
H (Z) =
1 - e -T Z -1
8. Why impulse invariant method is not preferred in the design of IIR filter
other than low pass filters?
In impulse invariance method, the mapping from s-plane to z-plane is

(2k - 1)
many to one, i.e., all the poles in the s-plane between the intervals
T
(2k + 1)
to for (k=0, 1, 2,…) maps into the entire z-plane. Thus, there are an
T
infinite number of poles that map to the same location in the z-plane, producing
aliasing effect. Due to spectrum aliasing the impulse invariance method is
inappropriate for designing high pass filters. That is why the impulse invariance
method is not preferred in the design of IIR filter other than low pass filters.
9. What is bilinear transformation?
The bilinear transformation is a mapping that transforms the left half of s-

plane into the unit circle in the z-plane only once thus avoiding aliasing of
frequency components.
The mapping from the s-plane to the z-plane is in bilinear transformation

is
2�z - 1�
S=
T �
� + 1�
z �
10. What are the properties of the bilinear transformation?
 The mapping for the bilinear transformation is a one-to-one mapping that is

for every point z, there is exactly one corresponding point s,and vice versa
 The j W axis maps on to the unit circle z = 1 ,the left half of the s-plane maps
to the interior of the circle z = 1 and the right half of the s-plane maps on to
the exterior of the unit circle z = 1
11. What is warping effect? What is its effect on magnitude and phase
response?
The relation between the analog and digital frequencies in bilinear

2 w
transformation is given by W = tan . For smaller values of w there exist linear
T 2

relationship between w and W . But for large values of w the relationship are
non-linear, this non-linearity introduces distortion in the frequency axis. This is
known as warping effect. This effect compresses the magnitude and phase
response at high frequencies.
12. Write a short note on prewarping?
The effect of the non-linear compression at high frequencies can be

compensated. When the desired magnitude response is piece-wise constant over
frequency, this compression can be compensated by introducing a suitable pre
2 w
scaling or pre warping the critical frequencies by using the formula W = tan .
T 2
13. What are the advantages and disadvantages of bilinear transformation?

ADVANTAGES:
 The bilinear transformation provides one-to-one mapping

 Stable continuous systems can be mapped into realizable, stable digital
systems
 There is no aliasing
DISADVANTAGES:
 The mapping is highly non-linear producing frequency compression at

high frequencies.
 Neither the impulse response nor the phase response of the analog filter is
preserved in a digital filter obtained by bilinear transformation.
14. What is main advantage of direct-form II realization when compared to
direct-form I realization?
In direct-form II realization, the number of memory locations required is

less than of direct-form I realization
15. What are the advantages and disadvantages of FIR filters?
ADVANTAGES:
FIR Filters have exact linear phase

FIR filters are always stable
FIR filters can be realized in both recursive and non recursive structure
Filter with any arbitrary magnitude response can be tackled using FIR
sequence.
DISADVANTAGES:
 For the same filter specification the order of FIR filter can be as high as 5
to 10 times that in a IIR filter design
 Large storage requirements needed
 Powerful computational facilities required for the implementation
16. Distinguish between FIR and IIR filters?

S.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
shape of their magnitude specific kind of filters
response
4. Errors due to round off noise The round off noise in IIR filters are
are less severe in filters, mainly more
because feedback is not used.
17. What are the design techniques of designing FIR filters?
There are three well-known methods for designing FIR filters with linear
phase. These are 1. Window method 2. Frequency sampling method 3. Optimal
or minimax design method.
18. What do you understand by linear phase response?
For a linear phase filter q (w )a w . 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 characteristic is
required throughout the pass band of the filter to preserve the shape of a given
signal with in 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.
19. What is the condition for the impulse response of FIR filter to satisfy for
constant phase delay or linear phase?
For linear phase FIR filter to have constant phase delay q (w )a w
For satisfying above condition, h(n)=h(N-1-n)
i.e., the impulse response must be symmetrical about n=(N-1)/2
20. What are the properties of FIR filter?
 FIR filter is always stable

 A reliable filter can always be obtained
 FIR filter has a linear phase response
21. Give the expression for location of poles of normalized Butterworth filter.
The poles of Butterworth filter is given by,

j 2 k
sk = e 2N
, k = 1,2,....2N for N odd
j (2k -1)
sk = e 2N
, k = 1,2,....2N for N even
22. What are the disadvantages of Fourier series method?
In designing FIR filter using Fourier series method the infinite duration
impulse response is truncated at n= + (N-1)/2. Direct truncation of the series
will lead to fixed percentage overshoots and undershoots before and after an
approximated discontinuity in the frequency response
23. What is Gibbs phenomenon? Or what are Gibbs oscillations?
One possible way of finding an FIR filter that approximates H(e jw)
would be to truncate the infinite Fourier series at n= + (N-1)/2. Abrupt
truncation of the series will lead to oscillation both in pass band and in stop
band. This phenomenon is known as Gibbs phenomenon.
24. Where FIR filters employed in practice?
FIR filters are used in applications where there is a need for a linear phase filter.
In telecommunications there is a requirement to separate signals such as data

that have been frequency division multiplexed, with out distorting these signals
in the process of demultiplexing.
25. Write equation of bartlet window and hamming window.
The N – point triangular window function is given by,
� 2n �N - 1 �
�1 - for n ��
wT (n) = � N - 1 �2 �
�0 else
�
The Hamming window sequence is
� �2 n � �N - 1 �
�0.5 + 0.46 cos � � for n ��
w H ( n) = � �N - 1 � �2 �
�0 else
�
26. If the impulse response of the symmetric linear phase FIR filter of length 5
is h[n] = {2,3,0,x,y}, then find the values of x and y.

For symmetric condition h(n)=h(N-1-n)

Hence, h(0)=h(5-1-0)=h(4) & h(1)=h(5-1-1)=h(3)
h[n] = {2,3,0,3,2}
27.The impulse response of analog filter is given in figure 1. Let h(n) = h a(nT)
where T=1. Determine the system function H(Z).
28 . Comment on the passband and stopband characteristics of Butter worth

filter.
The magnitude response of the Butterworth filter decreases monotonically as
the frequency W increases from zero to infinity. It has a monotonous pass band
and stop band.
29. Obtain the cascade realization for the system function,
X ( z) =
( 1 + 0.25 z -1 )
( 1 + 0.5z -1 ) ( 1 + 0.5 z -1 + 0.25 z -2 )
H ( z) =
( 1 + 0.25z )
-1
& H 2 ( z) =
1
1
( 1 + 0.5 z )
-1
( 1 + 0.5z -1 + 0.25z -2 )
30.Realize the following causal linear phase FIR system function
2 -1 2 -2
H ( z) = +z + z
3 3

UNIT-5 TWO MARK QUESTIONS
1. What are the features of TMS 320 c54 processor?
 Advanced multibus architecture with three separate 16-bit data buses

and one program memory bus
 40-bit arithmetic logic unit (ALU), including a 40-bit barrel shifter and
two independent 40-bit accumulators
 17×17-bit parallel multiplier coupled to a 40-bit dedicated adder for
nonpipelined single-cycle multiply/accumulate (MAC) operation
 Compare, select, and store unit (CSSU) for the add/compareselection of
the Viterbi operator
 Exponent encoder to compute an exponent value of a 40-bit accumulator
value in a single cycle
2. List the elements in program controller of TMS320c54 processor.
 Program Counter, The

 Status And Control Register
 Stack
 Address-Generation Logic.
3. What are various interrupts supported by TMS 320 c54 processor?
The C54x DSP supports both software and hardware interrupts:
 A software interruptis requested by a program instruction (INTR,

TRAP,or RESET)
 A hardware interrupt is requested by a signal from a physical device.
Two types exist:
 External hardware interrupts are triggered by signals at external
interrupt ports.
 Internal hardware interrupts are triggered by signals from the on-chip
peripherals.
4. Mention the function of the program controller of TMS320c54 processor.
The program controller decodes instructions, manages the pipeline,

stores the status of operations, and decodes conditional operations. Some of the

hardware elements included in the program controller are the program counter,
the status and control register, the stack, and the address-generation logic.
5. Mention the different buses of TMS 320 C54 processor.
The ’54x device architecture is built around eight major 16-bit buses:
 One program-read bus (PB) which carries the instruction code and
immediate operands from program memory
 Two data-read buses (CB, DB) and one data-write bus (EB), which
interconnect to various elements, such as the CPU, data-address
generation logic (DAGEN), program-address generation logic
(PAGEN),on-chip peripherals, and data memory
 The CB and DB carry the operands read from data memory.The EB
carries the data to be written to memory.
 Four address buses (PAB, CAB, DAB, and EAB), which carry the
addresses needed for instruction execution
6. Give the merits and demerits of VLIW architecture.
Merits:
 The instruction issue logic is very simple.
 Clock cycles required for implementing the instructions are shorter than
super scalar processor.
 The controller has complete control for detecting ILP
 VLIW can fit more execution unit on a given chip space because of simple
instruction issue logic.
Demerits:
i) Compiler complexity
ii) Track of instruction scheduling
iii) Increased memory usage.
iv) High power consumption.
7. List the factors that influence the selection of DSP processor for an
application.
Arithmetic Format
Data Width
Speed
Memory Organization
Ease of Development
Cost
Power Consumption & Management
8. Define pipelining in DSPs.
An instruction pipeline consists of a sequence of operations that occur

during the execution of an instruction. The C54xDSP pipeline has six levels:
prefetch, fetch, decode, access, read, and execute At each of the levels, an
independent operation occurs. Because these operations are independent, from

one to six instructions can be active in any given cycle, each instruction at a
different stage of completion. Typically, the pipeline is full with a sequential set
of instructions, each at one of the six stages. When a PC discontinuity occurs,
such as during a branch, call, or return, one or more stages of the pipeline may
be temporarily unused.
9. List the various addressing modes of DSPs.
Immediate Addressing
Absolute Addressing
Accumulator Addressing
Direct Addressing
Indirect Addressing
Memory-Mapped Register Addressing
Stack Addressing
10. State how a DS processor is applicable for motor control applications.
Speed of the AC induction motor can be controlled by varying the input

voltage to the motor. This can be achieved by using a DSP kit to produce
variable voltage. Variable voltage is produced by triggering the IGBT’s in the
power module at different angles. DSP processor is producing the necessary
PWM gating signals for the IGBT triggering. Speed sensor will sense the speed
and gives equivalent voltage signals to the power module. From power module,
control, signals are fed to the DSP through ADC. PWM signals from the DSP are
given back to the power module which controls the speed of the motor.
11. Define DMA in a DSP system.
The ’54x direct memory access (DMA) controller transfers data between
points in the memory map without intervention by the CPU. The DMA allows
movements of data to and from internal program/data memory, internal
peripherals (such as the McBSPs), or external memory devices to occur in the
background of CPU operation. The DMA has six independent programmable
channels, allowing six different contexts for DMA operation.
12. Give the advantage of Harvard architecture in a DS processor.
 Harvard architecture has separate data and instruction busses, allowing

transfers to be performed simultaneously on both busses.
 It is possible to have two separate memory systems.
 High Performance System
13. State the differences between Von Neumann and Harvard architecture.
Von Neumann
 Same memory holds data, instructions.
 A single set of address/data buses between CPU and memory

Harvard
 Separate memories for data and instructions.
 Two sets of address/data buses between CPU and memory
14. List the various applications of DSPs.
 Analogue emulation systems, Multirate processing and non-linear DSP
 Audio/visual/multimedia, Networking, Biomedical,Noise cancellation

systems, Control
 Nondestructive testing, Control systems engineering,Pattern recognition

and matching
 Digital waveform synthesis, Radar,Earth-based telecommunications,

Remote sensing
 Image processing, Robotics,Image processing in all its representations
 Satellite telemetry, Seismology,Mechatronics, Speech

recognition/synthesis
 Military/surveillance, Scientific Instrumentation; signal analysis
15. How do a digital signal processor differ from other processors.
S.No DSP GPP
1 Specialized, complex instructions General-purpose instructions

Multiple operations per instruction Typically only one ,operation per
Poor orthogonality instruction, Good orthogonality
2 Hardware looping, Interrupts Software looping
disabled during certain operations Interrupts rarely disabled
Limited or no register shadowing Register shadowing common
3 Limited bit-manipulation superior bit manipulation
capabilities capabilities
4 Harvard architecture Von Neumann architecture

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EE6403-Discrete Time signals and Signal Processing

Unit 1 TWO MARK QUESTIONS

3. Distinguish between continuous valued and discrete valued signals.

5. Explain the operation of time advancing.

6.When a discrete time signal is said to be symmetric (or) anti symmetric?

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The unit sample (impulse) sequence is denoted as d (n) and is defined as

Hence the system is stable.

13.Define BIBO stability.

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A causal recursive system is described by an input-output equation of the form

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y(n) = x(n-1) = {3,2,1,0,1,2,3}

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The odd signal component is,

The even signal component is,

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31. Determine the energy of the signal x (n) = 3n u (- n - 1)

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Thus g (n) = g1 (n) + g 2 (n) = x1 ( n) + x2 (n) -----------(1)

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State Shannon’s sampling theorem.41

2. Define ROC in Z – transform

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ROC is entire Z plane expect Z=0 and Z= �

7. Compute Fourier transform of the signal x ( n ) = u ( n ) - u ( n -1)

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10. Find the convolution of the following using Z-transform

14. What is the Z-transform of discrete unit step function?

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y (n) = { 1,3,5, 6,5,3,1 }

20. What are the four steps to obtain the convolution.

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23.Compute the Z-transform of x(n) = a nu ( n - 1)

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29.Compute the convolution of the two sequences using Z transform

1.Calculate the number of multiplications and addition needed in the

No of additions needed -- 64x63=4032 64 log 64=384

No of multiplications needed—642= 4096 (64/2) log 64=192

2.What is DIT radix2 algorithm.

3. What is DIF radix2 algorithm.

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g1(n) =x1(n)+x2(n)and g2(n)= (x1(n)+x2(n))WNn .Similarly the N/2 DFT s are

5.What are the differences between DIT and DIF algorithms?

6.Determine the circular convolution of the sequence

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The basic radix – 2 DIF butterfly diagram is,

9. Compare linear and circular convolution.

2.Linear convolution can be used to find 2.Linear convolution cannot be used

X (0) = x (0) + x (1) + x (2) + x (3) = 2

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Then, x ( n + N ) = x (n) for all ' n '

X (k + N ) = X (k ) for all ' k '

The Z-transform of a sequence x(n) is

14) Give the advantages and applications of FFT algorithm.

i) Computation of DFT with reduced no. of calculations.

ii)As the computations are performed in phase, the number of memory

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In an N point sequence, N is expressed as r m and the given sequence is

16) what is meant by in place computations in DIT and DIF algorithms?

The N point DFT is given by,

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21.Compute the IDFT of using DIF FFT algorithm.

DTSSP 2mark Q&A

DTSSP 2mark Q&A