EE6403 DSP 2m-3

SVS COLLEGE OF ENGINEERING
COIMBATORE-642 109
DEPARTMENT OF ELECTRONICS AND COMMUNICATION
ENGINEERING
Academic Year: 2016-2017
EE6403 DISCRETE TIME SYSTEMS

AND SIGNAL PROCESSING
II EEE – IV SEM
Prepared by,
Mr. K. Manoharan,
Assistant Professor,
ECE Department.
SVS College of Engineering.
SVS College of Engineering / K. Manoharan / II EEE / EE6403 Discrete Time Systems and Signal Processing Page 1
UNIT I - INTRODUCTION
1. Define Nyquist rate. [Dec 2016] [May 2012]

If the sampling frequency is , all the frequency above cause aliasing. This

aliasing can be avoided if the input signal frequencies are below
. This frequency is called
Nyquist frequency / Nyquist rate.
2. What is quantization error? [Dec 2016]

In digital signal processing, the continuous time input signals are converted into digital
using a b-bit ADC. The representation of continuous time signal amplitude by a fixed digit
produces an error, which is known as input quantization error.

3. Determine if the system described the equation (
) = (
) + ()
is causal or non
causal. [May 2016]
1
(
) = (
) +
(
− 1)

=0
1
(0) = (0) +
(−1)

=1
1
(1) = (1) +
(0)

=2
1
(2) = (2) +
(1)
Outputs depend on present and past input therefore the system is causal.
4. What is an anti aliasing filter? [May 2016]

A filter that is used to reject frequency signal before it is sampled to remove the
aliasing of unwanted high frequency signals is called an anti aliasing filter.
5. Given a continuous signal () = 2 300 . What is the nyquist rate and fundamental
frequency of the signal. [Dec 2015]
() = 2 300
2 = 300
= 150
!"# $%!"%
= 2 = 300&'
6. Find whether the signal (

) = "(
) is power signal or energy signal? [Dec 2015][May
2013]
Energy
+∝
( = ) * (
)*
+∝
+∝
( = ) *1* "(
)
+∝
+∝
( = )*1* = ∞
+-
Power
+2
1
. = lim ) * (
)*
2→∞ 2 + 1
+2
+2
1
. = lim ) *1*
2→∞ 2 + 1
+-
+2
1
. = lim 4 ) 1 5
2→∞ 2 + 1
+-
1
. = lim ( + 1)
2→∞ 2 + 1
1
61 + 7
. = lim
2→∞ 1
62 + 7
1
=
2
The energy of the signal is infinite and power is finite. Hence the signal is power
signal.
7. Check if the system described by the difference equation (

) = 8 (
− 1) + (
) with
(0) = 1 is stable. [May 2015]
Given (
) = 8 (
− 1) + (
)
9(') − 8 : 9(') = ;(')
9(')(1 − 8 : ) = ;(')
9 (' ) 1
&(') = =
;(') 1 − 8 :
ℎ(
) = 8 "(
)
For stability
∝
) *ℎ(
)* < ∞
+∝
1
∞ ∞ ∞
) *ℎ(
)* = ) *8 * " (
) = )*8 * = 1 + 8 + 8 … . . 8∞ = <∞
1−8
+∞ +∞ +-
System is stable.
8. Differentiate between Energy and power signal. [May 2015]

Energy signal Power signal
Energy of the signal x(n) is Power of the signal x(n) is
+∞ +2
1
( = ) * (
)* . = lim ) * (
)*
2→∞ 2 + 1
+∞ +2
Signal is called Energy signal if E is finite Signal is called Power signal if E = ∞ and P
and P=0. finite
9. Test whether the system governed by the relation (

) = ?∞
A+∞ ;(@) is linear time invariant
or not? [Dec 2014]
∞
(
) = ) (@)
A+∞
Response due to delayed input,
∞
(
, ) = ) (@ − )
A+∞
Delayed response,
∞
(
− ) = ) (@ − )
A+∞
(
− ) = (
, )
System is time invariant.
10. What is aliasing effect? [Dec 2014] [Dec 2012] [May 2011] [May 2011]
If the signal x(t) sampled at the sampling rate F < 2fm results are in spectral overlap.
This signal cannot be recovered using a low pass filter. This effect is known as aliasing.
11. Consider the analog signal () = 3 50 + 10 #

300 − 100 . What is
the Nyquist rate for this signal? [May 2014]
() = 3 50 + 10 #
300 − 100
2 1 = 50 → 1 = 25
2 2 = 300 → 2 = 150
2 3 = 100 → 3 = 50
= 8 (1, 2, 3)
= 150
Nyquist frequency = 2 = 300&'
12. State Shannon’s sampling theorem. [May 2014] [May 2011] [Dec 2011][May 2011]
A band limited continuous time signal, with higher frequency fm Hertz, can be uniquely
recovered from its samples provided the sampling rate F ≥ 2fm samples per second.
13. What is Nyquist rate for this signal x(t) = 3 cos 600π t + 2cos 1800 π t ? [Dec 2013]
() = 3 600 + 2 1800
2 1 = 600 → 1 = 300
2 2 = 1800 → 1 = 900
= 8 (1, 2, )
= 900
Nyquist frequency = 2 = 1800&'
π G-H
14. Determine the fundamental period of the signal 6 -I 7. [Dec 2013]
π G-H
(
) = 6 -I 7
30π
- = J K
105
π π
The fundamental period = = G-π . 105=7
L
15. Given a continuous time signal () = 2 500 What is the Nyquist rate and
fundamental frequency of the signal? [May 2013]
() = 2 500
2 = 500
= 250
!"# $%!"%
= 2 = 500&'
16. What is an LTI system? [Dec 2012]

A linear time invariant (LTI) system follows two principles
For liner system, the response due to linear combination of inputs is same as linear
combinations of corresponding outputs.
Time invariance means shift of time origin of input does not change the response of the
system.
17. What is a linear time invariant system? [Dec 2011]

If the input output relation of a system does not vary with time, the system is said to be
time invariant or shift invariant.
Example: (
) = (
) + (
− 1)
18. What is continuous and discrete time signal?

Continuous time signal:
A signal is said to be a continuous time signal if the amplitude and the time interval are
called continuous time signal. It is denoted by x(t).
Discrete time signal:
A signal is said to be a discrete time signal if the amplitude is continuous but discrete
in time are called discrete time signal. It is denoted by x(n).
19. Teat whether the system y(n)=0.5 x(n)+9 is linear and time invariant system.
Linear
For input x1(n), y1(n)=0.5 x1(n)+9
For input x2(n), y2(n)=0.5 x2(n)+9
a1 y1(n)+ a2 y2(n) = a1[0.5 x1(n)+9]+ a2 [0.5 x2(n)+9]
=0.5[a1 x1(n)+ a2 x2(n)]+9[a1 + a2]
Considering linear combination of inputs
x3(n)=a1 x1(n)+ a2 x2(n)
y3(n)=0.5[a1 x1(n)+ a2 x2(n)]+9
y3(n) ≠ a1 y1(n)+ a2 y2(n) hence system is non linear,
Time invariant
y(n)=0.5 x(n)+9
y(n,k) = y(n-k)
0.5 x(n - k)+9 = 0.5 x(n - k)+9
LHS =RHS
system is time invariant system.
20. Find whether the signal (

) = 6G
+ M7 is power signal or energy signal? (
) =
π π
6G
+ M7
π π
Energy
+∞
( = ) * (
)*
+∞
+∞
( = ) NO 6
+ 7PN
π π
3 6
+∞
+∞
( = ) O 6
+ 7P
π π
3 6
+∞
2π
1 + 6
+ 67
+∞ π
(= ) 3
2
+∞
+∞ +∝
1 2π
( = 4 ) 1 5 + ) J
+ K
π
2 3 6
+∞ +∝
1
E= (∞ − 0) = ∞
2
Power
+2
1
R = lim ) * (
)*
2→∞ 2 + 1
+2
+2
1
R = lim ) NO 6
+ 7PN
π π
2→∞ 2 + 1 3 6
+2
+2
1 1
R = lim 4 ) 1 5 + 0
2→∞ 2 + 1 2
+2
1 1 1
R = lim (2 + 1) =
2→∞ 2 + 1 2 2
The energy of the signal is infinite and power is finite. Hence the signal is power
signal.
21. A discrete time signal x(n)={0,0,1,1,2,0,0...} Sketch the x(n) and x(-n+2) signals.
x(n)
x(-n)
x(-n+2)
22. Determine whether the following signals are periodic, if periodic then compute the
fundamental period.
SM
a. cos(0.01 n) b. #
6 - 7
For periodicity (
) = (
+ )
(
+ ) = [0.01 (
+ )]
= [0.01
+ 0.01 ]

Fundamental period V- = = = 200
W -.-
For periodicity (
) = (
+ )
(
+ ) = #
J 10 (
+ )K
62
62
62
= #
6 10 + 10
7
Fundamental period
2 2 20
V- = = =
X 62 62
10
23. What is Energy and power signal.
Energy of the signal x(n) is
+∞
( = ) * (
)*
+∞
Power of the signal x(n) is
+2
1
. = lim ) * (
)*
2→∞ 2 + 1
+2
Signal is called Energy signal if Energy is finite and Power is zero.
Signal is called Power signal if E = infinite and P finite value.
24. What is correlation? What are its types?

The correlation measures the similarity between two signals.
Cross correlation - similarity between different signals
Auto correlation - similarity between time shifted version of same signal.
25. Determine the odd and even components of the signal, (

) = % R 6Y
+ Y 7 where

Z
Y = [−1.

(
) = % R 6Y
+ Y 7
4 2

(−
) = % R 6−Y
+ Y 7
4 2
Odd component
1 1
- (
) = [ (
) − (−
) ] = ]% R 6Y
− Y 7 − % R 6−Y
− Y 7 ^
2 2 4 2 4 2

= [ 6
− 7 + Y sin 6
− 7 − 6
− 7 + Y sin 6
− 7]
Z Z Z Z

= Y sin 6
− 7
4 2
Even component
1 1
a (
) = [ (
) − (−
) ] = ]% R 6Y
− Y 7 + % R 6−Y
− Y 7 ^
2 2 4 2 4 2

= [ 6Z
− 7 + Y sin 6Z
− 7 + 6Z
− 7 − Y sin 6Z
− 7]

= cos 6
− 7
4 2
UNIT II - DISCRETE TIME SYSTEM ANALYSIS
1. Given a difference equation (
) = (
) + 3 (
− 1) + 2 (
− 1). Determine the system
function H (z). [Dec 2016] [May 2013]
(
) − 2 (
− 1) = (
) + 3 (
− 1)
9(') − 2 : 9(') = ;(') + 3 : ;(')
9(')(1 − 2 : ) = ;(') (1 + 3 : )
9(') 1 + 3 :
& (' ) = =
;(') 1 − 2 :

2. Find the stability of the system whose impulse response ℎ(
) = 67 "(
) [Dec 2016] [May
2013]
For stability
∝
) *ℎ(
)* < ∞
+∝

Given ℎ(
) = 67 "(
)
1 1 1 1 1 1
∞ ∞ ∞
) *ℎ(
)* = ) NJ K "(
) N = ) NJ K N = 1 + + … . . ∞ = =2<∞
2 2 2 4 2 1
+∞ +∞ +- 1−2
System is stable.
3. What is the relation between DFT and Z-Transform? [Dec 2016]

Let N point DFT of x (n) be X (K) and z transform of x (n) be X (Z). The N point
sequence X(K) can be obtained from X(Z) by evaluating X(Z) at N equally spaced points
around the unit circle.
4. Determine the Z-transform and ROC of the following finite duration signals. [May 2016]
i) x(n)={3,2,2,3,5,0,1}
ii) x(n)=δ(n-k)
x(n)={3,2,2,3,5,0,1}
X(Z)=3+2z-1 +2z-2+3z-3+5z-4+z-6
ROC is entire z plane except z=0;
x(n)=δ(n-k)
Z {δ (n-k)} =' A X (z)
ROC is all z except 0 if k >0
5. Compute the convolution of the two sequence x(n)={2,1,0,0,5} and y(n)={2,2,1,1} [May
2016]
2 2 0 0 5
2 4 4 0 0 10
2 4 4 0 0 10
1 2 2 0 0 5
1 2 2 0 0 5
y(n)={4,8,6,4,12,10,5,5}
6. What is ROC of Z transform? State its properties. [Dec 2015] [May 2014][Dec 2012][Dec
2011][May 2011]
The region of convergence (ROC) is defined as the set of all values of z for which X(z)
converges.
The ROC is ring in the z plane centered at the origin.
The ROC cannot contain any pole.
The ROC of a LTI stable system contains the unit circle.
The ROC must be a connected region.
7. State initial and final value theorem of Z transform. [Dec 2015] [May 2014]
Initial value theorem
If X(z)=Z{x(n)}
(0) = d# X(z)
e→∞
Final value theorem
If X(z)=Z{x(n)}
(∞) = d#(1 − Z )X(z)
e→
8. Determine the Z transform of (

) = 8 [May 2015]
Let us take (
) = 8 u(n)
∞
;(') = ) (
)'
+∞
1 '
∞ ∞ ∞
;(') = ) 8 u(n)'
= )8 '
= )(8' ) = =
1 − 8' '−8
+∞ +- +-
9. Determine the Fourier transform of the signal x(t)=sin w0t [May 2015]
;() = j ()% kl m
∞
∞
;() = j sin w0t % kl m

∞
∞
% kL l − % kLl
;() = j O P % kl m
∞
∞ 2j
% k(L)l % k(pL )l
( )
; =j O − P m
∞
∞ 2j 2j
k
= [q ( − - ) − q ( + - )]

10. Find the Z transform and its ROC of the discrete time signals (
) = − 8 "(−
− 1), a>0.
[Dec 2014]
∞
;(') = ) (
)'
+∞

1 8 '
∞ ∞
;(') = ) −8 '
= )8 ' = )(8 ') =

− 1 =
1 − 8 ' 1 − 8 '
+∞ + +-
11. Determine the Z transform of for the signal (

) = q (
− @) + q (
+ @).[Dec 2013]
Z {δ (n)} =1
Z {δ (n - k)} =' A X (z)
Z {δ (n + k)} =' A X (z)
Z{ q (
− @) + q (
+ @)}=( ' A + ' A ) X (z)
12. Prove the convolution property of z – transform. [Dec 2013]
: { (
)} = X (z)
: { (
)} = X (z)
: { (
) ∗ (
)} = X (z) X (z)
13. Define discrete time Fourier transform pair for a discrete sequence. [Dec 2012]
∞
;u% k
v = ) (
)% k
+∞
14. Give relation between DTFT and Z transform. [May 2012]

The Z transform of x(n) is given by
∞
;(') = ) (
)'
+∞
where ' = $% k
The DTFT of x(n) is given by
∞
;u% k
v = ) (
)% k
+∞
15. Write the commutative and distributive properties of convolution. [Dec 2011]
Commutative property
(
) ∗ ℎ (
) = ℎ (
) ∗ (
)
Distributive property
(
) ∗ [ℎ (
) + ℎ (
)] = (
) ∗ ℎ (
) + (
) ∗ ℎ (
)
16. Write the DTFT for (a) (

) = 8 "(
), (b) (
) = 4q (
) − 3q (
− 1).
(
) = 8 "(
)
∞ ∞ ∞
;u% k
v = ) (
)% k
= ) 8 "(
)% k
= ) 8 % k
+∞ +∞ +-
1
∞
= )(8 % k ) =
1 − 8% k
+-
(
) = 4q (
) − 3q (
− 1) .
∞ ∞
;u% k
v = ) (
)% k
= )(4q (
) − 3q (
− 1) ) % k
+∞ +-
∞ ∞
= 4 ) q (
)% k − 3 ) q (
− 1) ) % k = 4 − 3% k
+- +-
17. Perform linear convolution for the following sequence x1(n)={1,2,3,4}, x2(n)={1,2,2,1}.
1 2 2 1
1 1 2 2 1
2 2 4 4 2
3 3 6 6 3
4 4 8 8 4
y(n)={1,4,9,15,16,11,4}
18. Perform circular convolution of x1(n)={1,2,3,4}, x2(n)={1,1,2,2} using matrix method.
1 2 2 1 1 15
1 1 2 2 2 = 17
2 1 1 2 3 15
2 2 1 1 4 13
y(n)={15,17,15,13}
19. List the properties of discrete time sinusoidal signals.

A discrete time sinusoidal is periodic only if its frequency is a rational number.
The highest rate of oscillation in a discrete time sinusoidal is attained when w= .
20. Distinguish between DFT and DTFT.

DFT DTFT
Obtained by performing sampling
Sampling is performed only in time
operation in both the time and frequency
domain
domains.
Discrete frequency spectrum Continuous function of w.
21. Find the DTFT of (

) = −w "(−
− 1).
∝
;(% k
) = ) (
). % k
+∝
∝
;(% k ) = ) −w "(−
− 1). % k
+∝
1
=
1 − % k
UNIT III - DISCRETE FOURIER TRANSFORM
1. Draw the graph of a 4 point DIT FFT butterfly structure for DFT. [May 2016][May
2016] 2015][Dec
2013]
XZ- =1; XZ =e-j2π/4 = -j
2. What are the applications of FFT algorithms? [May 2016]

Linear filtering
Correlation analysis
Power spectrum analysis
Frequency analysis
3. Calculate % saving in computing through radix 2 DFT algorithm of DFT coefficients. Assume
N=256. [Dec 2015]
DFT
The number of complex multiplications required using direct computation is
N2=2562=65536
The number of complex addition required using direct computation is
N(N-1)=256(256-1)=65280
1)=65280
FFT
The number of complex multiplication required using FFT is
(N/2) log2 N=(256/2) log2 256=1024
The number of complex addition required in FFT is N log2 N=256 log 2 256=2048
MIIGM
% Saving in Multiplication is ∗ 100 = 6400x
-Z
MIy-
% Saving in Addition is ∗ 100 = 318z%
-Zy
4. State the circular frequency shifting properties of DFT [Dec 2015][May 2014]
Let DFT{x (n)} =X (K)
Circular frequency shifting : DFT {{ ((n − m))2 } = |}π~
~
X()
5. Compute the DFT of the sequence x(n)={1, 1, 0, 0}. [May 2015]

DFT of x (n) is given by
2

kSA
;(@) = ) {(n). % 2 || = 0, 1, 2 … . . − 1
+-
-
XZ- XZ -j2π/4

Input S1 Output
1 1+0(1)=1 1+1(1)=2
0 1-0(1)=1 1+1(-j)=1-j
1 1+0(1)=1 1-1(1)=0
0 1-0(1)=1 1-(-j)=1+j
X(K)={2, 1-j , 0, 1+j}
6. What is zero padding? What are its uses? [Dec 2014]

Let the sequence x(n) has the 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 zero padding’s are
We can get better display of the frequency spectrum.
With zero padding, the DFT can be used in linear filtering.
7. State parsavel’s relation for DFT. [Dec 2014]

Let DFT{x1 (n)} =X1 (K), DFT{x2 (n)} =X 2(K)
2 2
1
) { (n) {∗ (n) = ) X () X ∗ ()
+- +-
8. Compare DIT radix – 2 FFT and DIF radix – 2 FFT [May 2014]
DIT radix – 2 FFT DIF radix – 2 FFT
The time domain sequence is decimated. The frequency domain sequence is decimated.
When the input is in bit reversed order, the When the input is in bit normal order, the
output will be in normal order and vice versa. output will be in bit reversed order and vice
versa.
In each stage of computations, the phase In each stage of computations, the phase
factors are multiplied before add and subtract factors are multiplied after add and subtract
operations. operations.
The value of N should be expressed such that The value of N should be expressed such that
N = 2m and this algorithm consists of m N = 2m and this algorithm consists of m
stages of computations. stages of computations.
Total number of arithmetic operations is N Total number of arithmetic operations is N
log N complex additions and (N/2) log N log N complex additions and (N/2) log N
complex multiplications. complex multiplications.
9. In eight point DIT what is the gain of the signal path that goes from x(7) to X(2)?[Dec 2013]
Gain = y- . y- (−1). (−Y) = Y
10. Find the discrete Fourier transform for x(n)=δ(n) [May 2013]
2
kSA
;(@) = ) (
). % 2 || K = 0, 1, 2 … . . − 1
+-
δ(n)=1 for n=0;
δ(n)=0 for n≠0;
Then, the DFT of the sequence δ (n) is given by,
2
kSA kSA.-
;(@) = ) δ(n). % 2 = δ(0). % 2 =1
+-
11. Draw the basic butterfly flow graph for the computation in the DIF FFT algorithm. [May
2013]
XZ- =1; XZ =e-j2π/4 = -j
12. Define discrete Fourier transform pair for a discrete sequence. [Dec 2012]
2
kSA
;(@) = ) {(n). % 2 || = 0, 1, 2 … . . − 1
+-
2
1 kSA
(
) = ) X(). % 2 || = 0, 1, 2 … . . − 1
+-
13. Find the 4 point DFT of the sequence xx(n)={1, 1}. [Dec 2012]
After zero padding x(n)={1,
(n)={1, 1,0,0}.
2

kSA
;(@ ) = ) {(n). % 2 || = 0, 1, 2 … . . − 1
+-
-
XZ- =1; XZ =e-j2π/4 = -j
Input S1 Output
1 1+0(1)=1 1+1(1)=2
0 1-0(1)=1 1+1(-j)=1-j
1 1+0(1)=1 1-1(1)=0
0 1-0(1)=1 1-(-j)=1+j
X(K)={2, 1-j , 0, 1+j}
14. The first five DFT value for N=8 is as follows X(k)={28, -4+j9.656, -4+4j, -4+j1.656,
- -4, ...}
compute rest of three DFT values
X(n)=X*(N-n)
X(5)=X*(8-5)=X*(3)= -4 - j1.656
X(6)=X*(8-6)=X*(2)= -4 - 4j
X(7)=X*(8-7)=X*(1)= -4 - j9.656
15. Compute 4 point IDFT for X(k)={2, 3+j, -4, 3-j)
XZ- =1; XZ =e-j2π/4 = -j
Input S1 Output
2 2+(-4)=-2 (-2)+( 6)=4
-4 2-(-4)=6 6 + (-2j)(-j)=4
3-j 3-j +(3+j)=6 (-2)-( 6)=-8
3+j 3-j -(3+j)=-2j 6 - (-2j)(-j)=8
x(n)={1,1,-2,2}
16. What is meant by radix-2 FFT?

The FFT algorithm is most efficient algorithm for calculating N-point DFT. If the
number of output points N can be expressed as a power of 2, that is N=2 , where M is an
integer, then this algorithm is known as radix-2 FFT algorithm.
17. In direct computation of N-point DFT of a sequence, how many multiplications and additions
are required? (Or) How is FFT faster? How many multiplications and additions are required to
compute N point DFT using radix-2 FFT?
FFT is faster because it requires less number of complex multiplications and Complex
additions compared to direct computation of DFT.
Operation FFT DFT
d
2
Complex multiplications
Complex additions d N ( N - 1)
18. List any two properties of DFT.

Let DFT{x (n)} =X (K), DFT{x1 (n)} =X1 (K), DFT{x2 (n)} =X 2(K)
Periodicity: X (K+N) =X (K) for all K.
Linearity: DFT[a1 x1 (n)+a2 x2(n)]=a1 X1 (K)+a2 X2 (K)
DFT of time reversed sequence: DFT[ x(N-n)]=X(N-K)
Circular convolution :DFT[x1(n)*x2(n)]=X1(K) X2(K)
19. Compute the IDFT of Y(k)={1, 0, 1, 0)
XZ- =1; XZ =e-j2π/4 = -j
Input S1 Output
1 1+1=2 (2)+( 0)=2
1 1-1=0 0 + (0)(-j)=0
0 0+0=0 (2)-( 0)=2
0 0-0=0 0 - (0)(-j)=0
x(n)={0.5, 0, 0.5, 0}
20. State the circular time shifting and circular frequency shifting properties of DFT
Circular time reversal : DFT {{ ((−n))2 } = X((−))2 =X (N-k)
Circular frequency shifting : DFT {{ ((n − m))2 } = |}π~ X()
21. Distinguish between linear convolution and Circular Convolution.

S. No Linear Convolution Circular Convolution
1. If x(n) is a sequence of L number of If x(n) is a sequence of L number of
samples and h(n) with m number of samples and h(n) with m number of
samples, after convolution y(n) will samples, after convolution y(n) will
contain N = L + M – 1 samples. contain N = Max(L,M) samples
2. Linear convolution can be used to find Circular convolution can be used to find
the response of a linear filter. the response of a linear filter
3. Zero padding is not necessary to find the Zero padding is necessary to find the
response of a linear filter. response of a linear filter.
22. What is bit reversal?

When the binary representation of one number is the mirror image of the binary
representation of the other, then both the numbers are said to be in bit reversal order. For
example, in a three-bit system, binary equivalent of one and four are bit-reversed values of
each other, since the three-bit binary representation of one, 001, is the mirror image of the
three-bit binary representation of four, 100.
UNIT IV DESIGN OF DIGITAL FILTERS
1. Distinguish between FIR and IIR filters. [Dec 2016] [May 2012]
S. No FIR filter IIR filter
These filters can be easily designed to
1 These filters do not have linear phase.
have perfectly linear phase
FIR filters can be realized recursively
2 IIR filters can be realized recursively
and non-recursively
Greater flexibility to control the shape Less flexibility, usually limited to
3
of their magnitude response kind of filters
Errors due to round-off
off noise are less
The round-offoff noise in IIR filters are
4 severe in FIR filters, mainly because
more
feedback is not used
2. What is the need for employing window for designing FIR filter?[Dec 2016][Dec 2012]
The windows are finite duration sequence used to modify the impulse response of the
FIR filters in order to reduce the ripples in the pass band and stop band and also to achieve the
desired transition from pass band to stop band.
3. Obtain the cascade realization for the system function.

fu [May 2016]
4. What is warping effect? Or Define prewarping effect. Or Why it is employed? [Dec 2016]
[Dec 2015][May 2014][Dec 2012][May 2012]
In bilinear transformation, the relation between analog and digital frequencies is
nonlinear. This non-linear
linear relationship introduces distortion in frequency axis, when the ‘s’
plane is mapped into ‘z’ plane using bilinear transformation. This effect is known as frequency
warping. The pre-warping
warping is performed as follows:

Ω = tn 6 7,
ω

In above equation, Ω and ω are analog and digital frequencies respectively. T is
nothing but a sampling rate. Pre-warping
Pre warping is necessary to eliminate the effect of warping on
amplitude response.
5. Mention the advantages of FIR filter over IIR filter. [May 2016]
1. FIR filter have exact linear phase.
2. FIR filters are always stable.
3. FIR filter can be realized in both recursive and recursive structures.
5. FIR filter are free of limit cycle oscillation, when implemented on a finite word
length digital system.
6. The impulse response of analog filter is given in Figure. Let h(n)=h(nT), where T=1.
Determine the system function. [Dec 2015][Dec 2013]
h(n) = {0,1,2,3,4,5,4,3,2,1,0}
-
&(') = ) ℎ(
) '
+-
= ' +2' + 3' G + 4' Z + 5' I + 4' M + 3' +2' y + 1'
7. Comment on pass band and stop band characteristics of butter worth filter. [May 2015]
In butter worth filter the transfer function is monotonic in both pass band and stop
band.

8. Realize the following causal linear phase FIR filter function & (') = + ' + ' [May
G G
2015]

&(') = + ' + '
G G
2
&(') = (1 + ' ) + '

3
9. What are the properties of Chebyshev filter. [Dee 2014][May 2013]

The magnitude response is equi-ripple in the pass band and monotonic in the stop band or
vive-versa.
The Chebyshev type – I filters are all pole designs.
All poles lie on the ellipse.
10. What are the advantages of FIR filter? [Dee 2014]

1. FIR filter have exact linear phase.
2. FIR filters are always stable.
3. FIR filter can be realized in both recursive and recursive structures.
4. Excellent design methods are available for various kinds of FIR filter.
5. FIR filter are free of limit cycle oscillation, when implemented on a finite word
length digital system.
11. Give hamming window function. [May 2014]
The weighing function for the hamming window is given by
S H 2 2
WH (n) = {0.54+0.46cos 6 7 -6 7≤
≤6 7

0 otherwise
p-.y
12. Is the given transfer function & (') = represents low pass filter or high pass filter? [
-.
Dec 2013]
1 + 0.8'
& (' ) =
1 − 0.9'
Pole =-0.9, zero=0.8
Zero close to point at (1,0) and the pole is close to point at (-1,0). Therefore the given
transfer function belongs to high pass filter.
13. Name the two methods for digitizing the transfer function of an analog filter. [May 2013]
1. Impulse invariant method
2. Bilinear transformation
14. What is Gibbs phenomenon?[May 2012]

One possible way of finding an FIR filter that approximates H d(e jω)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 is stop band .This phenomenon is known as Gibbs
phenomenon.
15. Define condition for stability. [May 2012]

The left half plane of 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.
16. Compare the impulse invariant and bilinear transformations. [Dec 2011]
S.
Impulse Invariant Transformation Bilinear transformation
No
1 It is many-to-one mapping It is one-to-one mapping
The relation between analog and The relation between analog and digital
2
digital frequency is linear frequency is non-linear
To prevent the problem of aliasing the No problem of aliasing and so the analog
3
analog filters should be band limited filters need not be band limited
The magnitude and phase response of Due to the effect of warping, the phase
analog filter can be preserved by response of analog filters cannot be
4
choosing low sampling time or high preserved. But the magnitude response
sampling frequency can be preserved by pre-warping
17. What is linear phase characteristic of an FIR filter? [Dec 2011]

The linear phase characteristic of an FIR filter is that the phase function should be a
linear function of w, which in turn requires constant phase and group delay.
18. “IIR filter does not have linear phase” – Justify

A physically realizable and stable IIR filter cannot have linear phase. A linear phase filter
must have a transfer function that satisfies the condition.
H(z)=±z-N H(z-1)
-N
where z represents a delay. But above equation tells us that for every pole inside the unit
circle there is a pole outside the unit circle. Hence the filter would be unstable. Therefore, a
causal and stable IIR filter cannot have linear phase.
19. What is meant by bilinear
ilinear transformation method of designing IIR filter?
Bilinear transformation is a one to one mapping from the s-domain
s domain to the z-domain.
z
That is, the bilinear transformation is a conformal mapping that transforms the j Ω axis into
the unit circle in the z plane only once, thus avoiding the aliasing of frequency components.
Also the transformation of a stable analog filter result in a stable digital filter as all the poles in
the left half of the s plane are mapped inside the unit circle of the z plane. The bilinear
mapping is a one to one mapping and it is accomplished when
2 1 − :
= O P
V 1 + :
20. Compare analog and digital filters.
Analog filter Digital filter
Constructed using active or passive Consists of elements like adder,
components and it is described by a Multiplier and delay units and it is
differential equation described by a difference equation
Frequency response can be changed by Frequency response can be changed by
changing the components changing the filter coefficients
It processes and generates analog output Processes and generates digital output
Output varies due to external conditions Not influenced by external conditions
21. Sketch the mapping of s-plane

plane and z-plane
z in approximation of derivatives.
The mapping procedure between S-plane & Z-plane
plane in the method of mapping of differentials
is given by H(Z) =H(S)|S=(1-ZZ-1)/T
The above mapping has the following characteristics
The left half of S-plane
plane maps inside a circle of radius ½ centered at Z= ½ in the ZZ-
plane.
The right half of S-plane
plane maps into the region outside the circle of radius ½ in the Z
Z-
plane.
22. The j Ω-axis

axis maps onto the perimeter of the circle of radius ½ in the Z Z-plane.
What is meant by aliasing?
When the sampling frequency is less than twice of the highest frequency content of the
signal, then the aliasing is frequency domain takes place. In aliasing, the high frequencies of
the signal mix with lower frequencies and create distortion in frequency spectrum.
23. Mention the properties of Butterworth filter.

The Butterworth filters are all pole designs.
At the cut-off
off frequency Ωc the magnitude of normalized Butterworth filter is 1/√2.
The filter order ‘n’ completely specifies the filter and as the value of N increases the
magnitude response approaches the ideal response.
r
24. Draw the direct form realization of IIR system. [May 2016] [May 2015] [May 2014]
Direct form- I
Direct form- II
25. What are the limitations of impulse invariant method of designing IIR filter?
In this method the mapping from s plane to z plane is many to one. i.e. ,all the poles in
the s plane between the intervals (2k
(2k-1)π/T
1)π/T to (2k+1) π /T .Thus there are an infinite number
of poles that map to the same location in the z plane, producing an aliasing effect. Due to
spectrum aliasing the impulse
pulse invariant method is inappropriate in designing high pass filters.
That is why the impulse method is not much preferred in the design of IIR filters other than
low pass filter.
26. What are the properties of FIR filter?

FIR filters are particularly useful for applications where exact linear phase response is
required.
The FIR filter is generally implemented in a non-recursive
non recursive way which guarantees a stable
filter.
The output depends on the present input and previous inputs
inputs only. It does not depend on
previous output.
27. What are the desirable characteristics of the windows?

The central lobe of the frequency response of the window should contain most of the
energy and should be narrow.
The highest side lobe level of the frequency response should be small.
The side lobes of the frequency response should decrease in energy rapidly as w tends to π
28. What is the necessary and sufficient condition for linear phase characteristics in FIR filter?
(Or) What is linear phase characteristic
charact of an FIR filter?
The linear phase characteristic of an FIR filter is that the phase function should be a
linear function of w, which in turn requires constant phase and group delay.
Impulse response, h (n) = ±hh (N-1-n)
(N
29. State Gibb’s phenomenon

In Fourier series method of FIR filter design, the infinite duration impulse response is
truncated to finite duration impulse response. This abrupt truncation of impulse response
introduces oscillations in the pass band and stop band. This effect is known as Gibb’s
phenomenon (or Gibb’s Oscillation).
30. What are the characteristics of FIR filters designed using windows.
The width of the transition band depends on the type of window.
The width of the transition band can be made narrow by increasing the value
value of N where N
is the length of the window sequence.
The attenuation in the stop band is fixed for a given window, except in case of Kaiser
window where it is variable.
UNIT V - DIGITAL SIGNAL PROCESSORS
1. What is meant by bit reversed addressing mode? What is the application for which this
addressing mode is preferred? [Dec 2016]
Bit reversed addressing mode start with index 0. The present index can be calculated
by adding half the FFT length to the previous index in a bit reversed manner, carry being
propagated from MSB to LSB.
Current index= Previous index+ B (1/2(FFT Size))
To implement FFT algorithms we need to access the data in a bit reversed manner.
Hence a special addressing mode called bit reversed addressing mode is used to calculate the
index of the next data to be fetched.
2. Compare RISC and CISC architecture. [Dec 2016]

CSIC RISC
Multiple instruction sizes and formats Instructions of same set with few formats
Less registers More registers are used
More addressing mode Fewer addressing mode
Extensive use of microprogramming Complexity in compiler
Instruction take varying amount of cycle time Instruction take one cycle time
Pipelining is difficult Pipelining is easy
3. What are the advantages of Harvard architecture in a DSP processor? [Dec 2015]
Harvard architecture is capable of simultaneously reading and instruction code and
reading or writing a memory or peripheral as port of the execution of the previous instruction.
4. How is a DSP processor applicable for motor control applications? [Dec 2015]
The motor control applications are controlled by the DSP processor by connection
the appliance through relay or opto-couplers.
5. How do digital signal processors differ from other processor? [May 2015]
The digital signal processors are microprocessor specially designed for efficient
implementation of digital signal processing system.
6. State any two applications of DSP. [May 2015]

Telecommunication – echo cancellation in telephone networks.
Instrumentation and control – Spectrum analysis, Digital filters etc.,
Speech processing – speech analysis methods used in automatic speech recognition.
7. What are the different stages in pipelining? [Dec 2014]

Fetch
Decode
Read
Execute
8. List the various registers used with ARAU of DSP processor? [Dec 2014][May 2014]
The ARAU contains eight 16 bit auxiliary registers AR0- AR7, A 3 bit Auxiliary
register Pointer (ARP), a 16 bit index register and a 16 bit auxiliary register compare
register(ARCR).
9. What are the different buses of TMS 320C54x processor and list their functions? [May 2014]
PB : Program bus and PAM: Program address bus
Program memory bus to read opcode and immediate operant.
CB : C bus and CAB : C address bus
DB : D bus and DAB : D address bus
Two independent data memory buses to read two data simultaneously from memory.
EB : E bus and EAB : E address bus
Data memory buses to write data in data memory.
10. Mention one important feature of Harvard architecture. [May 2013]

The Harvard architecture has two memory blocks to store code and data separately and
the two memory blocks are connected to CPU by separate buses for simultaneous access of
code and data.
11. What is the advantage of pipelining? [May 2013]

Number of instructions can be executed in parallel, hence speed is high.
12. What is pipelining? What are the different stages in pipelining? [Dec 2012][Dec 2011]
Pipelining is a process by breaking down its instructions into a series of discrete
pipeline stages which can be completed in sequence by specialized hardware.
Different stages in pipelining are
Fetch, Decode, Read and Execute.
13. What is the function of parallel logic unit in DSP processor? [Dec 2012]
The parallel logic unit is an additional logic unit that permits logic operations without
affecting accumulator or product register. It performs Boolean operation or bit manipulations.
It can set, clear or toggle bits in the status register, control register and in any data memory
location.
14. Give the special features of DSP processors. [Dec 2011]

Harvard architecture
VLIW Architecture
Multiplier accumulate unit
Pipelining
15. Write short notes on general purpose DSP processors.

General-purpose digital signal processors are basically high speed microprocessors
with hard ware architecture and instruction set optimized for DSP operations. These
processors make extensive use of parallelism, Harvard architecture, pipelining and dedicated
hardware whenever possible to perform time consuming operations
16. Write notes on special purpose DSP processors.

There are two types of special; purpose hardware.
1. Hardware designed for efficient execution of specific DSP algorithms such as digital filter,
FFT.
2. Hardware designed for specific applications, for example telecommunication, digital
audio.
17. What about of Harvard architecture?

The principal feature of Harvard architecture is that the program and the data
memories lie in t o separate spaces, permitting full overlap of instruction fetch and execution.
18. What are the types of MAC is available?

There are two types MAC’S available
Dedicated & integrated
Separate multiplier and integrated unit
19. What is meant by pipeline technique?

The pipeline technique is used to allow overall instruction executions to overlap. That
is where all four phases operate in parallel. By adapting this technique, execution speed is
increased.
20. What are four phases available in pipeline technique?
The four phases are
1. Fetch
2. Decode
3. Read
4. Execution
21. Write down the name of the addressing modes.

Direct addressing.
Indirect addressing.
Bit-reversed addressing.
Immediate addressing.
Short immediate addressing.
Long immediate addressing.
Circular addressing
22. What are the instructions used for block transfer in C5X Processors?
The BLDD, BLDP and BLPD instructions use the BMAR to point at the source or
destination space of a block move. The MADD and MADS also use the BMAR to address an
operand in program memory for a multiply accumulator operation
23. 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.
24. Write the name of various part of C5X hardware.

Central arithmetic logic unit (CALU)
Parallel logic unit (PLU)
Auxiliary register arithmetic unit (ARAU)
Memory-mapped registers.
Program controller.
25. What are the merits of VLIM architecture?

Advantages of VLIW architecture
Increased performance
Better compiler targets
Potentially easier to program
Potentially scalable
Can add more execution units; allow more instructions to be packed into the VLIW
instruction.
26. What are the demerits of VLIM architecture?

Disadvantages of VLIW architecture
New kind of programmer I compiler complexity
Program must keep track of instruction scheduling
Increased memory use
High power consumption
27. A DSP has a circular buffer with the start and the end addresses as 0200h and 020Fh
respectively.
What would be the new values of the address pointer of the buffer if, in the course of address
computation, it gets updated to
a. 0212h b.01FCh
Buffer Length= (EAR - SAR+1)= 020F - 0200+1=10h
a. New Address Pointer= Updated Pointer - buffer length = 0212 -10=0202h
b. New Address Pointer= Updated Pointer + buffer length = 01FC+10=020Ch
27. Identify the addressing modes of the operands in each of the following instructions
a. ADD #1234h
b. ADD 1234h
c. ADD *AR+
d. ADD offsetreg - ,*AR
ADD #1234h Immediate Addressing Mode

ADD 1234h Direct Addressing Mode
ADD *AR+ Post Increment Indirect Addressing Mode
ADD offsetreg-,*AR Pre-Sub_Offset Indirect Addressing Mode
UNIT – I INTRODUCTION
1. Explain continuous time signal and discrete time signals.
2. Explain classification of discrete time system.
3. Explain the process of analog to digital conversion of signal in terms of sampling, quantization
and coding.
4. What is energy and power signal? Determine the power and energy of the signal

(
= 6 7 "(
)
G
(
) = #
6 7
π
Z
(
) = %
k6p7
π π
(
) = % "(
)
5. Determine whether the signals is periodic, if periodic find fundamental period.
(
) = % kMπ
Gπ
(
) = cos
+ cos
π
G Z
6. Explain sampling theorem and reconstruction of the analog signal from its sample.
7. Determine whether the following system is
a. Casual b. Linear c. Dynamic d. Time invariant e. Stable.
(
) = 10 (
) (0.25
+ )
(
) = (
− 1)
(
) = (−
)
8. Find the convolution sum of two sequence (
) = {1,2,1,1} and (
) = {1,1,2,1}.
9. Find the linear convolution of two sequence (
) = {1,2,1,1} and (
) = {1,1,2,1}.using
circular convoluction.
10. Find the cross correlation of (
) = {1,2,1,1} and (
) = {1,1,2,1}
11. What is the input signal x(n) that will generate output sequence y(n)={1,5,1,11,8,4,1} for
impulse response h(n={1,2,1}
UNIT – II DISCRETE TIME SYSTEM ANALYSIS

1. Determine the z transform and ROC of the signal

(
) = 6 7 "(
).
G
(
) = 8 " (
) + w "(−
− 1)
(
) = 8 cos -
" (
)
p-.
2. Find inverse z transform of ;(') = (p-.I)(), *'* > 1

3. Find inverse z transform of ;(') = ,for ROC *'* > 1, *'* < , < *'* < 1
G Zp G G

4. A casual system is represented by the difference equation (
) + Z (
− 1) = (
) +

(
− 1) find system transfer function H(z).

5. Compute the linear convolution of the following sequence
h(n)={1,2,1,3} and x(n)={1,2,-1,2, }.
6. Explain the properties of Z-transform
7. Obtain the system function and impulse response of the following system (
) =
5 (
− 1) − (
) + (
– 1)
8. Test the stability of the system
(
) = cos (
)
(
) = (−
− 2)
9. Computer circular convolution using DFT – IDFT method x(n)={0,1,2,3}, h(n)={1,1,1,1}
10. Perform the circular convolution of the following two sequences.
x1(n) = {2 1 2 1}, x2(n) = {1 2 3 4}
UNIT – III DISCRETE FOURIER TRANSFORM
1. Discuss in detail the important properties of the Discrete Fourier Transform
2. Compute 4-point DFT of the sequence x(n)={0,1,2,3}
3. Compute the eight-point DFT of the sequence
1 0 ≤
≤ z
(
) =
0 ℎ%$#%
4. Compute 8-point DFT of the sequence x(n)={0,1,2,3,4,5,6,7} using radix-2 DIF algorithm
5. Compute an 8 point DFT using DIT FFT radix 2 algorithm x(n) = {1,2,3,4,4,3,2,1}
6. Compute IDFT of the sequence X(k)={1,1+j,2,1-2j,,1+2j,0,1-j}using DIT FFT radix 2
algorithm.
7. Compute IDFT of the sequence X(k)= {2,2,2,2,1,1,1,1}using DIF FFT radix 2 algorithm
8. Compute the linear convolution of finite duration sequences
h(n)={1,2} and x(n)={1,2,-1,2,3,-2,-3,-1,1,1,2,-1} by overlap add method.
9. Compute the linear convolution of finite duration sequences
h(n)={1,2,-1} and x(n)={1,2,-1,2,3,-2,-3,-1,1,1,2,-1} by overlap save method.
UNIT VI DESIGN OF DIGITAL FILTERS

IIR FILTER
1. Design a analog Butterworth filter with a maximum pass band attenuation of -2Db pass band
attenuation at a frequency of 20 rad/sec and at least -10db stop band attenuation at 50 rad/sec.
2. Design a Chebyshev filter with a maximum pass band attenuation of 2.5 db; at Ώp = 20 rad/sec
and the stop band attenuation of 30 db at ΏS = 50 rad/sec.
3. Design a Butterworth filter using impulse invariance method satisfying the constraints.
Assume T=1sec.
0.8 ≤|H(ejw)| ≤ 1 ; 0 ≤ w ≤0.2π
jw
|H(e )| ≤0.2 ; 0.6π ≤ w ≤ π
4. Design a digital Butterworth filter satisfying the constraints
0.707≤|H (ejω)| ≤1 for 0≤ω≤π/2
|H (ejω)| ≤0.2 for 3π/4≤ ω≤ π with T=1 sec using Bilinear Transformation. Realize the filter in
each case using the most convenient realization form.
5. Design a chebyshev low pass filter with the following specification = 3mw,ripple in the
passband 0 ≤ w ≤0.2π, = 15mw, ripple in the stop band 0.3 π ≤ w ≤π, using Bilinear
Transformation.
6. Design a chebyshev filter for the following specifications using impulse invariance method.
Assume T=1sec.
0.8 ≤|H(ejw)| ≤ 1 ; 0 ≤ w ≤0.2π
jw
|H(e )| ≤0.2 ; 0.6π ≤ w ≤ π
7. Using the Bilinear transform design a high pass filter, monotonic in Pass band with cut off
frequency of 1000Hz and down 10db at 350Hz.The sampling frequency is 5000Hz
8. For the analog transfer function H(s) = 1/ (S+1) (S+2) determine H(z) using impulse invariant
technique. Assume T=1sec.
9. Obtain the direct form I, direct form II and cascade form realization of the following system
functions. Y(n)=0.1 y(n-1)+0.2 y(n-2)+3x(n)+3.6x(n-1)+0.6x(n-2)
FIR FILTER
10. Using a rectangular window technique design a low pass filter with pass band gain of unity,
cut off frequency of 1000 Hz and working at a sampling frequency of 5 KHz. The length of
the impulse response should be 7.
11. Explain the designing of FIR filters using windows
12. Design an ideal low pass filter using Fourier series method with a frequency response
Hd(ejw) = 1 for -π/2 ≤ w ≤ π/2
= 0 for π/2 ≤ w ≤π .
Assume N=11. Find H(z) plot the magnitude response.
13. Design an ideal high pass filter using hanning window with a frequency response
Hd(ejw) = 1; π/4 ≤ |w| ≤ π
= 0; |w| ≤ π /4 .
Find value of h(n) for N=11.Find H(z) plot the magnitude response.
14. Design an ideal band pass filter using hamming window with a frequency response
Hd(ejw) = 1; π/4 ≤ |w| ≤ 3π/4
= 0; otherwise
15. Design an ideal band reject filter using rectangular window with a frequency response
Hd(ejw) = 1; |w|≤ π/3 and |w|≥ 2π/3
= 0; otherwise
16. Realize the system function H(z)=[ ]z+1+[ ]' 1 by linear phase FIR structure.
2 2
3 3
17. Design a 15 tap linear phase filter to the following discrete frequency response.
H(k)= 1 0≤ @ ≤ 4
0.5 k=5
0.25 k=6
0.1 k=7
0 elsewhere
UNIT – V DIGITAL SIGNAL PROCESSORS

1. Explain the addressing modes of a DSP processor.
2. Describe the Architectural details of a DSP processor.(or)
Explain the architecture of TMS320C50 with a neat diagram.
3. With a neat diagram explain Von-Neumann architecture
4. What is MAC unit? Explain its functions.
5. Write a note on commercial DSP processor.
6. Explain in detail about pipelining.
7. Explain the architecture of TMS320C54 with a neat diagram.
8. Write short note on ALU and barrel shifter.
9. Explain various instruction groups available in digital signal processor with an example.

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SVS COLLEGE OF ENGINEERING

Academic Year: 2016-2017

EE6403 DISCRETE TIME SYSTEMS

2. What is quantization error? [Dec 2016]

4. What is an anti aliasing filter? [May 2016]

6. Find whether the signal (

7. Check if the system described by the difference equation (

8. Differentiate between Energy and power signal. [May 2015]

9. Test whether the system governed by the relation (

11. Consider the analog signal () = 3  50  + 10 #

16. What is an LTI system? [Dec 2012]

17. What is a linear time invariant system? [Dec 2011]

18. What is continuous and discrete time signal?

20. Find whether the signal (

24. What is correlation? What are its types?

25. Determine the odd and even components of the signal, (

3. What is the relation between DFT and Z-Transform? [Dec 2016]

8. Determine the Z transform of (

;() = j sin w0t % kl m

11. Determine the Z transform of for the signal (

14. Give relation between DTFT and Z transform. [May 2012]

16. Write the DTFT for (a) (

19. List the properties of discrete time sinusoidal signals.

20. Distinguish between DFT and DTFT.

Discrete frequency spectrum Continuous function of w.

21. Find the DTFT of (

XZ- =1; XZ =e-j2π/4 = -j

2. What are the applications of FFT algorithms? [May 2016]

5. Compute the DFT of the sequence x(n)={1, 1, 0, 0}. [May 2015]

XZ- XZ -j2π/4

6. What is zero padding? What are its uses? [Dec 2014]

7. State parsavel’s relation for DFT. [Dec 2014]

XZ- =1; XZ =e-j2π/4 = -j

XZ- =1; XZ =e-j2π/4 = -j

XZ- =1; XZ =e-j2π/4 = -j

16. What is meant by radix-2 FFT?

Operation FFT DFT

18. List any two properties of DFT.

19. Compute the IDFT of Y(k)={1, 0, 1, 0)

XZ- =1; XZ =e-j2π/4 = -j

21. Distinguish between linear convolution and Circular Convolution.

22. What is bit reversal?

3. Obtain the cascade realization for the system function.

9. What are the properties of Chebyshev filter. [Dee 2014][May 2013]

10. What are the advantages of FIR filter? [Dee 2014]

14. What is Gibbs phenomenon?[May 2012]

15. Define condition for stability. [May 2012]

17. What is linear phase characteristic of an FIR filter? [Dec 2011]

18. “IIR filter does not have linear phase” – Justify

21. Sketch the mapping of s-plane

22. The j Ω-axis

23. Mention the properties of Butterworth filter.

26. What are the properties of FIR filter?

27. What are the desirable characteristics of the windows?

29. State Gibb’s phenomenon

2. Compare RISC and CISC architecture. [Dec 2016]

6. State any two applications of DSP. [May 2015]

7. What are the different stages in pipelining? [Dec 2014]

10. Mention one important feature of Harvard architecture. [May 2013]

11. What is the advantage of pipelining? [May 2013]

14. Give the special features of DSP processors. [Dec 2011]

15. Write short notes on general purpose DSP processors.

16. Write notes on special purpose DSP processors.

7. Check if the system described by the difference equation (

9. Test whether the system governed by the relation (

11. Consider the analog signal () = 3 50 + 10 #

;() = j sin w0t % kl m

XZ- =1; XZ =e-j2π/4 = -j

XZ- XZ -j2π/4

XZ- =1; XZ =e-j2π/4 = -j

XZ- =1; XZ =e-j2π/4 = -j

XZ- =1; XZ =e-j2π/4 = -j

XZ- =1; XZ =e-j2π/4 = -j