EC 2302 DIGITAL SIGNAL PROCESSING

UNIT -1
DISCRETE FOURIER TRANSFORM (FFT)

PART-A

1. Determine the DTFT of a sequence x(n) a
n
u(n). Nov/Dec 2006
Solution:
x(n) a
n
u(n)

X(e
j
) _ x(n) e
-jn

n-

X(e
j
) _ a
n
u(n) e
-jn

n-

X(e
j
) _ a
n
e
-jn

n0


X(e
j
) _ (a e
j
)
n

n0



2. What is FFT? Nov/Dec 2006
The Fast Fourier TransIorm is a method or algorithm Ior computing the DFT
with reduced number oI calculations. The computational eIIiciency can be
achieved iI we adopt a divider and conquer approach. This approach is based
on decomposition oI an N-point DFT in to sucessively smaller DFT`s. This
approach leads to a Iamily oI an eIIicient computational algorithm is known
as FFT algorithm.


X(e
j
) 1 / (1-a
-ej
)
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3. The first five DFT coefficients of a sequence x(n) are X(0) 20,
X(1) 5+j2,X(2) 0,X(3) 0.2+j0.4 , X(4) 0 . Determine the
remaining DFT coefficients. May/1une 2007
Solution:
X (K) |20, 5j2, 0, 0.2j 0.4 , 0,X(5),X(6),X(7)|
X (5) 0.2 j0.4
X (6) 0
X (7) 5-j2

4. What are the advantages of FFT algorithm over direct computation of
DFT? May/1une 2007
1. Reduces the computation time required by DFT.
2. Complex multiplication required Ior direct computation is N
2
and Ior
FFT calculation is N/2 log
2
N.
3. Speed calculation.

5. State and prove Parseval`s Theorem. Nov/Dec 2007
Parseval`s theorem states that
II
x(n) X(K) and y(n) Y(K) ,
Then

N-1 N-1
_ x(n) y*(n) 1/N _ X(K) Y*(K)
n0 K 0

When y(n) x(n), the above equation becomes





6. What do you mean by the term ~bit reversal as applied to FFT?
Nov/Dec 2007
Re-ordering oI input sequence is required in decimation in time. When
represented in binary notation sequence index appears as reversed bit order
oI row number.





N-1 N-1
_ x(n)
2
1/N _ X(K)
2

n0 k0

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7. Define the properties of convolution. April/May 2008.
1. Commutative property: x(n)`h(n) h(n) `x(n)
2. Associative Property: x(n)`h
1
(n)] `h
2
(n) x(n)`h
1
(n)`h
2
(n)]
3. Distributive Property: x(n)` h
1
(n)+ h
2
(n)] x(n)` h
1
(n)]+ x(n)` h
1
(n)]

8. Draw the basic butterfly diagram of radix -2 FFT. April/May 2008.

1 1
a A a W
N
nk
b
1
1
W
N
nk

b B a - W
N
nk
b
-1

9. Distinguish between DIT and DIF -FFT algorithm. Nov/Dec 2008

S.No DIT -FFT Algorithm DIF -FFT Algorithm
1. The input is in bit reversed
order; the output will be
normal order.
The input is in normal order;
the output will be bit reversed
order.
2. Each stage oI computation the
phase Iactor are multiplied
beIore add subtract operation.
Each stage oI computation the
phase Iactor are multiplied
aIter add subtract operation.

10.If H(K) is the N-point DFT of a sequence h(n) , Prove that H(K) and
H(N-K) are comples conjugates. Nov/Dec 2008

This property states that, iI h(n) is real , then H(N-K) H*(K) H(-K)
Proof:

By the deIinition oI DFT;
N-1
X(K) _ x(n) e
(j2nk)/N

n0
Replace K` by N-K`
N-1
X(N-K) _ x(n) e
(j2n(N-K))/N

n



X(N-K) X*(K)

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11.State Discrete Fourier Transform.
The DFT is deIined as N-1
X (K) _ x(n) e
(j2nk)/N
; K 0 to N-1
n0
The Inverse Discrete Fourier TransIorm (IDFT) is deIined as
N-1
x (n) _ X(K) e
(j2nk)/N
; n 0 to N-1
K0

12.Distinguish between linear & circular convolution.

s.no Linear convolution circular convolution
1 The length oI the input sequence
can be diIIerent.
The length oI the input
sequence should be same.
2 Zero Padding is not required. Zero padding is required iI
the length oI the sequence is
diIIerent.

13.Define Zero padding? Why it is needed?
Appending zeros to the sequence in order to increase the size or length oI the
sequence is called zero padding.In circular convolution , when the two input
sequence are oI diIIerent size , then they are converted to equal size by zero
padding.

14.State the shifting property of DFT.
Time shiIting property states that
DFT x(n-n
0
)} X(K) e
(j2n0k)/N

15.Why do we go for FFT?
The FFT is needed to compute DFT with reduced number oI
calculations.The DFT is required Ior spectrum analysis on the sinals using
digital computers.

16.What do you mean by radix-2 FFT?
The radix -2 FFT is an eIIicient algorithm Ior coputing N- point DFT oI an
N-point sequence .In radix-2 FFT the n-point is decimated into 2-point
sequence and the 2-point DFT Ior each decimated sequence is computed.
From the results oI 2-point DFT`s, the 4-point DFT`s are computed. From
the results oI 4 point DFT`s ,the 8-point DFT`s are computed and so on
until we get N - point DFT.

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17.Give any two application of DFT?
1. The DFT is used Ior spectral analysis oI signals using a digital
computer.
2. The DFT is used to perIorm Iiltering operations on signals using
digital computer.

18.How many multiplications & addition are involved in radix-2 FFT?
For perIorming radix-2 FFT, the value oI Nshould be such that, N 2
m
. The
total numbers oI complex additions are Nlog
2
N and the total number oI
complex multiplication are (N/2) log
2
N.

19.What is Twiddle factor?
Twiddle Iactor is deIined as W
N
e
j2/N.
It is also called as weight Iactor.

20.What is main advantage of FFT?
FFT reduces the computation time required to compute Discrete Fourier
TransIorm.

PART-B

1. a) i) Calculate the DFT of the sequence x(n) 1,1,-2,-2]
ii) Determine the response of LTI system by radix -2 DIT FFT.
Nov/Dec 2006
Ans:i) X(K) 0, -1-j,6,-1j}
ii) ReI Pg.No 320-328 , DSP by Salivahanan .

2. a) i) Derive the equation for Decimation - in time algorithm for FFT.
ii) How do you linear filtering by FFT using Save -add method?
Nov/Dec 2006 & April /May 2008 & Nov/Dec 2008
Ans:i) ReI Pg.No 320-328 , DSP by Salivahanan .
ii) ReI Pg.No 369, DSP by Salivahanan.

3. a)i) Prove the following properties of DFT when H(k) is the DFT of
an N-point sequence h(n).
1. H(k) is real and even when h(n) is real and even.
2. H(k) is imaginary and odd when h(n) is real and odd.
ii) Compute the DFT of x(n) e
-0.5n
, 0 n 5.
May/1une 2007
Ans: i) ReI Pg.No 309, DSP by Salivahanan.
ii) X(K) 2.414, 0.87-j0.659, 0.627-0.394j, 1.202,
0.62-j0.252, 0.627-j0.252}.
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4. a) i) From first principles obtain the signal flow graph for
Computing 8-point using radix -2 DIF -FFT algorithm.
ii) Using the above signal flow graph compute DFT of
x(n) cos (n/4) ,0 n 7.
May/1une 2007 & Nov/Dec 2007 & Nov/Dec 2008

Ans: i) ReI Pg.No 334-340, DSP by Salivahanan.
ii) X(K) 0, 3, 0, 2.7-j0.7, 0, 1, 0, 1.293-j0.7}

5.a) Two finite duration sequence are given by
x(n) sin (n/2) for n 0,1,2,3
h(n) 2
n
for n 0,1,2,3 Determine circular convolution using
DFT &IDFT method. Nov/Dec 2007
Ans: X(K) 0, -2j, 0, 2j}
H(K) 15, -36j, -5, -3-6j}
y(n) 6, -3, -6, 3}

6.a) i) Discuss in detail the important properties of the DFT.
ii) Find the 4-point DFT of the sequence x(n) cos (n/4)
iii) Compute an 8-point DFT using DIF FFT radix -2 algorithm.
x(n) 1,2,3,4,4,3,2,1]
April /May 2008
Ans: i)ReI Pg.No 308-311, DSP by Salivahanan.
ii) X(K) 1, 1-j1.414, 1, 1j1.414}
iii) X(K) 20,-5.8-j2.4, 0, 0.17-j0.414, 0, -0.17j0.414, 0,
-5.82j2.414}.















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UNIT -2
INFINITE IMPULSE RESPONSE DIGITAL FILTERS

PART-A


1. Give any two properties of Butterworth and Chebyshev filter.
Nov/Dec 2006
Properties of Butterworth:
1. The butterworth Iilters are all pole design.
2. The Iilter order N completely speciIies the Iilter
3. The magnitude is maximally Ilat at the origin.
4. The magnitude is monotomically decreasing Iunction oI ohm.
Properties of Chebyshev:
1. The magnitude reponse oI the Iilter exhibits ripples in the pass band
or stop band
2. The pole oI the Iilter lies on an ellipse.

2. Find the digital transfer function H(Z) by using impulse invariant method
for the analog transfer function H(S) 1/ (S+2).Assume T0.5sec
May /1une 2007 &Nov/Dec 2007
Solution:
H(S) 1/ (S2).
H(Z) 1/|1-e
-1
Z
-1
|
H(Z) 1/ |1-0.368Z
-1
|

3. What is the relationship between analog and digital frequency in impulse
invariant transformation? April/May 2008

Digital Frequency: T
analog Irequency
T Sampling interval

4. What is Prewarping? Why is it needed? Nov/Dec 2008
In IIR design using bilinear transIormation the conversion oI speciIied
digital Irequencies to analog Irequencies is called Pre-warping. The Pre-
Warping is necessary to eliminate the eIIect oI warping on amplitude
response.



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5. Compare FIR & IIR filter.

S.No FIR filter IIR filter
1. Only N samples oI
impulse response are
considered.
All the inIinite samples oI
impulse response are
considered.
2. Linear phase
characteristics can be
achieved
Linear phase characteristics
can not be achieved

6. Define Frequency warping.
The non linear relation ship between analog and digital Irequencies
introduced Irequency distortion which is called as Irequency warping.


PART-B

1. a) Design a digital Butterworth filter satisfying the constraints
0.707 [ H()[ 1.0 ; 0 /2
[ H()[ 0.2 ; 3/4 .
Nov/Dec 2006
Ans: ReI Pg.No 435-437, DSP by Salivahanan.

2. a) Design a digital Butterworth filter satisfying the constraints
0.8 [ H()[ 1.0 ; 0 /4
[ H()[ 0.2 ; /2 .
Apply Bilinear transformation method.
May/1une2007 & Nov/Dec 2008
Ans: ReI: Pg.No: 359-362, DSP by Nagoorkani.

3. a)i) Design a digital BUTTERWORTH filter that satisfies the
following constraint using BILINEAR Transformation.
Assume T 1 sec.
0.9 [ H()[ 1 ; 0 /2

[ H()[ 0.2 ; (3 /4)
ii) Determine the magnitude response of the FIR filter (M11)
and show that phase and group delay are conatant.
iii) The desired frequency response of a low pass filter is given by
H
d
() e
-j3

;

-3/4 3/4
0 ; other wise.
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Determine H(e
j
) for M 7using HAMMING window.
iv) For the analog transfer function H(S) 1/ (S+1)(S+2) .
Determine H(Z) using impulse invariant technique.

April /May 2008

Ans: a) i) ReI Pg.No 437-439, DSP by Salivahanan.
ii) ReI Pg.No 383-384, DSP by Salivahanan.
iii) ReI Pg.No 400-401, DSP by Salivahanan.
iv) ReI Pg.No 426, DSP by Salivahanan.

UNIT -3
FINITE IMPULSE RESPONSE DIGITAL FILTERS

PART-A

1. Obtain the block diagram representation of a FIR system?
Nov/Dec 2006
X(Z)

h(N-1)
h(1)
h(0) h(2) h(N-2)


Y(Z)




2. Show that the filter with h(n) -1,0,1] is a linear phase filter.
May /1une 2007 & Nov/Dec 2008
Solution:
h(n) | -1,0,1|
h(0) -1 -h(N-1-n) -h(3-1-0) -h(2)
h(1) 0 -h(N-1-n) -h(3-1-1) -h(1)
h(2) 1 -h(N-1-n) -h(3-1-2) -h(0)
It is a linear phase Iilter.




Z
-1
Z
-1


Z
-1


Z
-1



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3. In the design of FIR digital filter, how is Kaiser Window different from
other windows? Nov/Dec 2007
In all other windows a trade oII exists between ripple ratio and main lobe
width. In Kaiser Window both ripple ratio and main lobe width can be
varied independently.

4. What are the merits and demerits of FIR filter? April/May 2008
Merits :
1. Linear phase Iilter.
2. Always Stable
Demerits:
1. The duration oI the impulse response should be large
2. Non integral delay.

5.What are the advantages of FIR filter?
1. They can have an exact linear phase.
2. They are always stable
3. They can be realised eIIiciently in hardware
4. The design methods are generally stable.



6.What is the necessary & sufficient condition of linear phase FIR filter?
The condition Ior a linear phase Iilter is
1. (N-1)/2
2. h(n) h(N-1-n)


7. What is Gibb`s phenomenon?
In Fir Iilter design using Fourier analysis method Ior rectangular window
method, the inIinite duration impulse response is truncated to Iinite
duration impulse response.The abrupt truncation oI impulse response
introduce a oscillation in the pass band and stop band .This eIIect is known
as Gibb`s phenomenon.

8. Compare Rectangular & Hamming window.

S.No Rectangular Window Hamming window.
1. The width oI the main
lobe in window spectrum
is 4/N
The width oI the main lobe
in window spectrum is 8/N
2. The maximun side lobe The maximun side lobe
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magnitude in window
spectrum is -13 dB
magnitude in window
spectrum is -41 dB

9. Compare Hamming window & Kaiser Window.

S.No Kaiser Window Hamming window.
1. The width oI the main
lobe in window spectrum
depends on the value oI
and N.
The width oI the main lobe
in window spectrum is 8/N
2. The maximun side lobe
magnitude with respect to
peak oI main lobe is
variable using the
parameter .
The maximun side lobe
magnitude in window
spectrum is -41 dB

10.Compare FIR & IIR filter.

S.No FIR filter IIR filter
1. Only N samples oI
impulse response are
considered.
All the inIinite samples oI
impulse response are
considered.
2. Linear phase
characteristics can be
achieved
Linear phase characteristics
can not be achieved




11.Compare Rectangular Window& Hanning Window.

S.No Rectangular Window Hanning Window
1. The width oI the main
lobe in window spectrum
is 4/N
The width oI the main
lobe in window spectrum
is 8/N
2. The maximun side lobe
magnitude in window
spectrum is -13 dB
The maximun side lobe
magnitude in window
spectrum is -31 dB



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12.Compare Hamming Window& Hanning Window.

S.No Hamming window. Hanning Window
1. The width oI the main
lobe in window spectrum
is 8/N
The width oI the main
lobe in window spectrum
is 8/N
2. The maximun side lobe
magnitude in window
spectrum is -41 dB
The maximun side lobe
magnitude in window
spectrum is -31 dB

13.Compare Hamming Window& BlackmanWindow.

S.No Hamming window. BlackmanWindow
1. The width oI the main
lobe in window spectrum
is 8/N
The width oI the main
lobe in window spectrum
is 12/N
2. The maximun side lobe
magnitude in window
spectrum is -41 dB
The maximun side lobe
magnitude in window
spectrum is -58 dB

PART-B

1.a) Design a high pass filter hamming window by taking 9 samples
of w(n) and with a cutoff frequency of 1.2 radians/sec.
Nov/Dec 2006
Ans: ReI: Pg.No: 298-301, DSP by Nagoorkani.

2.a) Design a digital Butterworth filter satisfying the constraints
0.707 [ H()[ 1.0 ; 0 /2
[ H()[ 0.2 ; 3/4 .
Nov/Dec 2006
Ans: ReI Pg.No 435-437, DSP by Salivahanan.

3.a) Design a digital Butterworth filter satisfying the constraints
0.8 [ H()[ 1.0 ; 0 /4
[ H()[ 0.2 ; /2 .
Apply Bilinear transformation method.
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May/1une2007 & Nov/Dec 2008
Ans: ReI: Pg.No: 359-362, DSP by Nagoorkani.

4. a) Describe the design of FIR filter using frequency sampling
technique.
b) The desired frequency response of a low pass filter is given by
H
d
() e
-j2

;

-/4 /4
0 ; other wise.
Obtain the filter coefficient, h(n) using RECTANGUAR window
define by W(n) 1; 0 n 4
0; otherwise.
Nov/Dec 2007
Ans: a) ReI Pg.No 389-391, DSP by Salivahanan.
b) ReI Pg.No 399, DSP by Salivahanan.


5. a) Design a bandpass filter to pass frequencies in the range
1to2radians/sec using hanning window,with N5.
Nov/Dec 2006
Ans: ReI: Pg.No: 301, DSP by Nagoorkani.

UNIT -4
FINITE WORD LENGTH EFFECTS

PART-A

1. Express the fraction 7/8 and - 7/8 in sign magnitude, 2`s complement
and 1`s complement. Nov/Dec 2006
Solution:
7/8 0.875 (0.111)
2
is sign magnitude
1`s Complement (0.111)
2
2`s Complement (0.111)
2
- 7/8 -0.875
Sign magnitude: (1.111)
2

1`s Complement (1.000)
2
2`s Complement (1.001)
2

2. a) What are the quantization error due to finite word length register
in digital filter.
b) What are the different quantization methods? Nov/Dec 2006
Quantization Error :
1. Input quantization error
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2. CoeIIicient quantization error
3. Product quantization error
Quantization methods
1. Truncation
2. Rounding

3. Identify the various factors which degrade the performance of the
digital filter implementation when finite word length is used.
May /1une 2007 & April/May 2008 & Nov/Dec 2008
1. Input quantization error
2. CoeIIicient quantization error
3. Product quantization error

4. What is meant by limit cycle oscillation in digital filter?
May /1une 2007 & Nov/Dec 2007 &April/May 2008
In recursive system when the input is zero or same non-zero constant value
the non linearities due to Iinite precision arithmetic operation may cause
periodic oscillation in theoutput. Thus the oscillation is called as Limit cycle.


5. Express the fraction (-7/32) in signed magnitude and 2`s complement
notations using 6 bits. Nov/Dec 2007 &Nov/Dec 2008
In Signed Magnitude: 1.001110
In 2`s complement: 1.110010

6. Compare fixed & floating point number representation.

S.no Fixed point number Floating point number
1. The position oI the binary
Point is Iixed.
The position oI the binary
Point is variable.
2. The resolution is uniIorm
throughout the range
The resolution is variable.

7. What are the two types of quantization employed in digital system?
1. Rounding
2. Truncation

8. Define Rounding & truncation.
Truncation is the process oI discarding all bits less signiIicant than least
signiIicant bit that is retained.
Rounding oI a b bit is accomplished by choosing the rounded result as the b
bit number closed to the original number unrounded.
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9. What is dead band?
In the limit cycle the amplitude oI the output are conIined to a range oI
value which is called as dead band oI the Iilter.

PART-B

1. For the given transfer function H(Z) H
1
(Z) .H
2
(Z) ,where
1 1
H
1
(Z) and H
2
(Z)
1 - 0.5 Z
-1
1 - 0.6 Z
-1

Find the output round off noise power.
Nov/Dec 2006
Ans: 2
-2b
/12(5.4315)

2. a)i) Explain the characteristics of a limit cycle oscillation w.r.t the
system described by the difference equation
y(n) 0.95y(n-1)+x(n).Determine the dead band of the filter.

ii) Draw the product quantisation noise model of second order
IIR filter.
Nov/Dec 2006 & Nov/Dec 2008
Ans: a) i) Dead band |-10,10|
ii) ReI Pg.No 513-514, DSP by Salivahanan.

3.a)i) Consider the truncation of negative fraction number represented
in(+1) bit fixed point binary form including sign bit . Let (-b)
bits be truncated .Obtain the range of truncation errors for signed
magnitude ,2`s complement and 1`s complement representation of
negative numbers. Nov/Dec 2007

ii) The coefficients of a system defined by
1
H(Z)
(1-0.4Z
-1
)(1-0.55Z
-1
)
are represented in anumber with a sign bit and 3 data bits.
Determine the new pole location for 1) Direct realization and 2)
Cascade realization of first order systems.Compare the
movements of the new pole away from the original ones in both
the cases.

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iii) Consider the (b+1) bit bipolar A/D converter.Obtain an
expression for signal to quantization noise ratio .
May /1une 2007& Nov/Dec 2007&April/May2008 & Nov/Dec 2008

Ans: a) i) ReI Pg.No 496-499, DSP by Salivahanan.
ii) Direct Iorm: 1/ |1-0.875z
-1
0.125Z
-2
|
Cascade Iorm:1/|1-0.375Z
-1
||1-0.5Z
-1
|
iii) ReI Pg.No 499-503, DSP by Salivahanan.

UNIT -5
MULTIRATE SIGNAL PROCESSING

PART-A

1. Define multirate signal processing.
The process oI converting a signal Irom a given rate to a diIIerent rate is
called sampling rate conversion. The system that employs multiple sampling
rates in the processing oI digital signals are called multirate digital signal
processing systems.

2. What are the advantages of multirate digital signal processing.
Computational requirements are less
Storage Ior Iilter coeIIicients is less
Finite arithmetic eIIects are less
Filter order required in multirate application are low and
Sensitivity to Iilter coeIIicients lengths are less


3. Give the applications of multirate digital signal processing. p
Communication systems
Speech and audio processing systems
Antenna systems
Radar systems

4. Define Decimation
The process oI reducing the sampling rate oI the signal is called decimation
(sampling rate compression).

5. Define Interpolation
The process oI increasing the sampling rate oI the signal is called interpolation
(sampling rate Expansion).
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6. Define signal flow graph.
A signal Ilow graph is a graphical representation oI the relationships between
the variables oI a set oI linear diIIerence equations.

PART-B
1. Explain Multi rate Digital signal processing
ReI Pg.No.523, DSP by Salivahanan

2. Explain the Decimation process with an example.
ReI Pg.No.526, DSP by Salivahanan
3. Explain the Interpolation process with an example.
ReI Pg.No.531, DSP by Salivahanan
4. Explain the polyphase decomposition for FIR filter.
ReI Pg.No.541, DSP by Salivahanan
5. Explain Multistage Decimators and interpolators.
ReI Pg.No.555, DSP by Salivahanan


B.E/B.TECH DEGREE EXAMINATION, NOV/DEC 2006
FIFTH SEMESTER (REGULATION - 2004)
ELECTRONICS AND COMMUNICATION ENGINEERING
EC 1302 - DIGITAL SIGNAL PROCESSING
TIME: 3 hrs Max. Mark: 100


PART A (10`2 20 MARKS)
Answer all Questions

21.Determine the DTFT of a sequence x(n) a
n
u(n).
Solution:
x(n) a
n
u(n)

X(e
j
) _ x(n) e
-jn

n-

X(e
j
) _ a
n
u(n) e
-jn

n-

X(e
j
) _ a
n
e
-jn

n0

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X(e
j
) _ (a e
j
)
n

n0



22.What is FFT?
The Fast Fourier TransIorm is a method or algorithm Ior computing the DFT
with reduced number oI calculations. The computational eIIiciency can be
achieved iI we adopt a divider and conquer approach. This approach is based
on decomposition oI an N-point DFT in to successively smaller DFT`s. This
approach leads to a Iamily oI an eIIicient computational algorithm is known
as FFT algorithm.

23.Obtain the block diagram representation of a FIR system?

X(Z)

h(N-1)
h(1)
h(0) h(2) h(N-2)


Y(Z)




24.Give any two properties of Butterworth and Chebyshev filter.
Properties of Butterworth:
5. The butterworth Iilters are all pole design.
6. The Iilter order N completely speciIies the Iilter
7. The magnitude is maximally Ilat at the origin.
8. The magnitude is monotomically decreasing Iunction oI ohm.
Properties of Chebyshev:
3. The magnitude response oI the Iilter exhibits ripples in the pass band
or stop band
4. The pole oI the Iilter lies on an ellipse.

9. Express the fraction 7/8 and - 7/8 in sign magnitude, 2`s complement
and 1`s complement.
Solution:
7/8 0.875 (0.111)
2
is sign magnitude
1`s Complement (0.111)
2
X(e
j
) 1 / (1-a
-ej
)
Z
-1
Z
-1


Z
-1


Z
-1



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2`s Complement (0.111)
2
- 7/8 -0.875
Sign magnitude: (1.111)
2

1`s Complement (1.000)
2
2`s Complement (1.001)
2

10.a) What are the quantization errors due to finite word length
register in digital filter.
b) What are the different quantization methods?


Quantization Error :
4. Input quantization error
5. CoeIIicient quantization error
6. Product quantization error
Quantization methods
4. Truncation
5. Rounding

11.What is zero padding? Does Zero padding improve the frequency
resolution in the spectral estimate?
Let the sequence x(n) has a length L.II we want to Iind the N-point DFT
(N~L) oI the sequence x(n) we have to add (N-L) zeros to the sequence
x(n).This is known as Zero padding. Yes , it will improve the Irequency
resolution in the spectral estimate.

12.Explain Deterministic and random signal with examples.
Deterministic signals are signals that are completely speciIied in time.
Deterministic signals can be predicted. Ex: Sin n
Non deterministic signals are deIined as the signal that takes on random
value at any given instant and it cannot be predicted. It is also called as
Random signal. Ex: ECG signal.

13.Give the digital signal processing application with the TMS 320 family.

Communication system like voice coder, Speech recognization, Audio signal
processing, Control and data acquisition, Biometric inIormation processing
and image /video processing.

14.What is the advantage of Harvard architecture of TMS 320 series?
Harvard architecture has two or three memory buses allowing
access to Iilter coeIIicients and input signals in the same cycle. Since it
possesses two independent bus system. The Harvard architecture is
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capable oI Simultaneous reading an instruction code and reading or
writing a memory or peripheral as part oI the execution oI the
previous instruction.


PART B (5`16 80MARKS)

11. a) i) Calculate the DFT of the sequence x(n) 1,1,-2,-2]
ii) Determine the response of LTI system by radix -2 DIT FFT.
Ans:i) X(K) 0, -1-j,6,-1j}
ii) ReI Pg.No 320-328 , DSP by Salivahanan .

(OR)

b) i) Derive the equation for Decimation - in time algorithm for
FFT.
ii) How do you linear filtering by FFT using Save -add method?
Nov/Dec 2006 & April /May 2008 & Nov/Dec 2008
Ans:i) ReI Pg.No 320-328 , DSP by Salivahanan .
ii) ReI Pg.No 369, DSP by Salivahanan.

a. a) Design a high pass filter hamming window by taking 9 samples
of w(n) and with a cutoff frequency of 1.2 radians/sec.
Ans: ReI: Pg.No: 298-301, DSP by Nagoorkani.

(OR)

b) Design a digital Butterworth filter satisfying the constraints
0.707 [ H()[ 1.0 ; 0 /2
[ H()[ 0.2 ; 3/4 .
Ans: ReI Pg.No 435-437, DSP by Salivahanan.

b.a) For the given transfer function H(Z) H
1
(Z) .H
2
(Z) ,where
1 1
H
1
(Z) and H
2
(Z)
1 - 0.5 Z
-1
1 - 0.6 Z
-1

Find the output round off noise power.
Ans: 2
-2b
/12(5.4315)
(OR)

b) i) Explain the characteristics of a limit cycle oscillation w.r.t the
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system described by the difference equation
y(n) 0.95y(n-1)+x(n).Determine the dead band of the filter.
ii) Draw the product quantisation noise model of second order
IIR filter.
Ans: a) i) Dead band |-10,10|
ii) ReI Pg.No 513-514, DSP by Salivahanan
c. a) Determine the performance characteristics of Non -parametric
Power spectrum estimator Welch, Bartlett and Blackman and
Tukey.
Ans: a) ReI Pg.No 594-603, DSP by Salivahanan.

(OR)

b) i) Give the key features of the digital signal processor
ii) Write Short notes on:
a) 32- bit accumulator
b) 16`16 bit parallel multiplier
c) Shifter
Ans: a) i) ReI Pg.No 8-9, Digital Signal Processor by
B.Venkataramani &M.Bhaskar
ii) ReI Pg.No 40, 59,266, Digital Signal Processor by
B.Venkataramani &M.Bhaskar

15. a) i) Explain the function of auxillary register in the indirect
Addressing mode to point the data memory location.
ii) Write a program to use the auxillary register in memory
pointing and looping.
Ans: ReI Pg.No 48-49, Digital Signal Processor by
B.Venkataramani &M.Bhaskar

(OR)

b) i) Write a program to compute the following equation .
Y A`X1 + B` X2 +C`X3
ii) Write a program to perform addition of two 64 bit numbers
Ans: ReI Pg.No 265, Digital Signal Processor by
B.Venkataramani &M.Bhaskar





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B.E/B.TECH DEGREE EXAMINATION, MAY/1UNE 2007
FIFTH SEMESTER (REGULATION - 2004)
ELECTRONICS AND COMMUNICATION ENGINEERING
EC 1302 - DIGITAL SIGNAL PROCESSING
TIME: 3 hrs Max. Mark: 100


PART A (10`2 20 MARKS)
Answer all Question

1. The first five DFT coefficients of a sequence x(n) are X(0) 20,
X(1) 5+j2,X(2) 0,X(3) 0.2+j0.4 , X(4) 0 . Determine the remaining
DFT coefficients
Solution:
X (K) |20, 5j2, 0, 0.2j 0.4 , 0,X(5),X(6),X(7)|
X (5) 0.2 j0.4
X (6) 0
X (7) 5-j2

2. What are the advantages of FFT algorithm over direct computation of
DFT?
1. Reduces the computation time required by DFT.
2. Complex multiplication required Ior direct computation is N
2
and Ior
FFT calculation is N/2 log
2
N.
3. Speed calculation.

3. Show that the filter with h(n) -1,0,1] is a linear phase filter.
Solution:
h(n) | -1,0,1|
h(0) -1 -h(N-1-n) -h(3-1-0) -h(2)
h(1) 0 -h(N-1-n) -h(3-1-1) -h(1)
h(2) 1 -h(N-1-n) -h(3-1-2) -h(0)
It is a linear phase Iilter.

4. Find the digital transfer function H(Z) by using impulse invariant
method for the analog transfer function H(S) 1/ (S+2).Assume
T0.5sec


Solution:
H(S) 1/ (S2).
H(Z) 1/|1-e
-1
Z
-1
|
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H(Z) 1/ |1-0.368Z
-1
|

5. Identify the various factors which degrade the performance of the
digital filter implementation when finite word length is used.

10.Input quantization error
11.CoeIIicient quantization error
12.Product quantization error

6. What is meant by limit cycle oscillation in digital filter?
In recursive system when the input is zero or same non-zero
constant value the non linearities due to Iinite precision arithmetic operation
may cause periodic oscillation in the output. Thus the oscillation is called as
Limit cycle.

7. Define Power spectral density and cross power spectral density?

The spectral characteristics oI a random process is obtained according to
wiener khinchine theorem, by computing the Fourier transIorm oI the
autocorrelation Iunction. The power spectral density is given by


xx
(F) }
xx
() e
j2It
d
-

The power density spectrum Ior two jointly stationary random process X(t)
and Y(t) gives the cross correlation Iunction
xy
() .The Fourier transIorm
oI
xy
(F) is


xy
(F) }
xx
() e
j2It
d.
-

8. What are the disadvantages of non-parametric methods of power
spectral estimation?
1. It requires long data sequence to obtain the necessary Irequency
resolution.
2. Spectral leakage eIIects because oI windowing.
9.Differentiate between Von Neumann and Harvard architecture.
Von Neumann architecture possesses one bus system. The same bus carries
all the inIormation exchanged between the CPU and the peripherals
including the instruction codes as well as data processed by the CPU.
Harvard architecture possesses two independent bus systems. It is capable
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oI simultaneous reading an instruction code and reading or writing a
memory or peripheral as part oI the execution oI precious instruction.

10.State the merits and demerits of multi ported memories.
A Mutiported memory operates as on chip memory and oII chip memory.
1. Higher perIormance because no wait states are required.
2. Lower cost than external memory.
3. Lower Power than external memory.
4. Higher perIormance because better Ilow with in the pipeline at lential
arithmetic logic unit.
5. Ability to access large memory space.

PART B (5`16 80MARKS)

11.a) i) Prove the following properties of DFT when H(k) is the DFT of
an N-point sequence h(n).
3. H(k) is real and even when h(n) is real and even.
4. H(k) is imaginary and odd when h(n) is real and odd.
ii) Compute the DFT of x(n) e
-0.5n
, 0 n 5.

Ans: i) ReI Pg.No 309, DSP by Salivahanan.
ii) X(K) 2.414, 0.87-j0.659, 0.627-0.394j, 1.202,
0.62-j0.252, 0.627-j0.252}.

(OR)

b) i) From first principles obtain the signal flow graph for
Computing 8-point using radix -2 DIF -FFT algorithm.
ii) Using the above signal flow graph compute DFT of
x(n) cos (n/4) ,0 n 7.
Ans: i) ReI Pg.No 334-340, DSP by Salivahanan.
ii) X(K) 0, 3, 0, 2.7-j0.7, 0, 1, 0, 1.293-j0.7}
12.a) Design a digital Butterworth filter satisfying the constraints
0.8 [ H()[ 1.0 ; 0 /4
[ H()[ 0.2 ; /2 .

Ans: ReI: Pg.No: 359-362, DSP by Nagoorkani.

(OR)
b) A band pass FIR filter of length 7 is required. It is to have lower
and upper cutt frequencies of 3 KHZ and 6 Khz respectively
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and indended to be used with a sampling frequency of 24 KHZ
Determine the filter coefficient using HANNING window
Consider the filter to be causal. (16)


13.a) i) Consider the truncation of negative fraction number
represented in(+1) bit fixed point binary form including sign
bit . Let (-b) bits be truncated .Obtain the range of truncation
errors for signed magnitude ,2`s complement and 1`s
complement representation of negative numbers.

ii) The coefficients of a system defined by
1
H(Z)
(1-0.4Z
-1
)(1-0.55Z
-1
)
are represented in anumber with a sign bit and 3 data bits.
Determine the new pole location for 1) Direct realization and 2)
Cascade realization of first order systems. Compare the
movements of the new pole away from the original ones in
both the cases.
(OR)



b) Consider the (b+1) bit bipolar A/D converter.Obtain an
expression for signal to quantization noise ratio .

Ans: a) i) ReI Pg.No 496-499, DSP by Salivahanan.
ii) Direct Iorm: 1/ |1-0.875z
-1
0.125Z
-2
|
Cascade Iorm:1/|1-0.375Z
-1
||1-0.5Z
-1
|
b) ReI Pg.No 499-503, DSP by Salivahanan.

14.a)i) Determine the performance characteristics of Non -parametric
Power spectrum estimator Welch, Bartlett and Blackman and
Tukey.
ii) Determine the frequency resolution of the Bartlett, Welch &
Blackman Tukey methods of power spectrum estimates for a
Quality factor Q 20. Assume that overlap in Welch`s method
is 50 and the length of the sample sequences is 2000.

Ans: a) i) ReI Pg.No 594-603, DSP by Salivahanan.
ii)ReI Pg.No 606, DSP by Salivahanan.
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(OR)

b) i) Compute the auto correlation & power spectral density for
the signal x(t) K cos (2ft + ) .
ii) Explain briefly the periodogram method of power spectral
Estimation.
Ans: b) i) ReI Pg.No 590-591, DSP by Salivahanan.
ii) ReI Pg.No 589, DSP by Salivahanan.

15.a) i) Explain what is meant by instruction pipelining. Explain with
an example ,how pipelining increase the through put
efficiency.
ii) Explain the operation of TDM serial port in P-DSP`s.
Ans: a) i) ReI Pg.No 45-46, Digital Signal Processor by
B.Venkataramani &M.Bhaskar
ii) ReI Pg.No 50-51, Digital Signal Processor by
B.Venkataramani &M.Bhaskar
(OR)
b)i) With a suitable diagram describe the function of
Multiplier/adder units of TMS 320 C54X.
ii) Explain the operation of CSSU of TMS 320 C54X and explain
its use considering the viteri operator.
Ans: b) i) ReI Pg.No 267-268, Digital Signal Processor by
B.Venkataramani &M.Bhaskar
ii) ReI Pg.No 269-270, Digital Signal Processor by
B.Venkataramani &M.Bhaskar















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B.E/B.TECH DEGREE EXAMINATION, NOV/DEC 2007
FIFTH SEMESTER (REGULATION - 2004)
ELECTRONICS AND COMMUNICATION ENGINEERING
EC 1302 - DIGITAL SIGNAL PROCESSING
TIME: 3 hrs Max. Mark: 100

PART A (10`2 20 MARKS)
Answer all Question

1. State and prove Parseval`s Theorem.
Parseval`s theorem states that
II
x(n) X(K) and y(n) Y(K) ,
Then

N-1 N-1
_ x(n) y*(n) 1/N _ X(K) Y*(K)
n0 K 0

When y(n) x(n), the above equation becomes





2. What do you mean by the term ~bit reversal as applied to FFT?
Re-ordering oI input sequence is required in decimation in time. When
represented in binary notation sequence index appears as reversed bit order
oI row number.

3. Find the digital transfer function H(Z) by using impulse invariant
method for the analog transfer function H(S) 1/ (S+2).Assume
T0.5sec
Solution:
H(S) 1/ (S2).
H(Z) 1/|1-e
-1
Z
-1
|
H(Z) 1/ |1-0.368Z
-1
|



4. In the design of FIR digital filter, how is Kaiser Window different from
other windows?
N-1 N-1
_ x(n)
2
1/N _ X(K)
2

n0 k0

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In all other windows a trade oII exists between ripple ratio and main lobe
width. In Kaiser Window both ripple ratio and main lobe width can be
varied independently.

5. What is meant by limit cycle oscillation in digital filter?

In recursive system when the input is zero or same non-zero constant value
the non linearities due to Iinite precision arithmetic operation may cause
periodic oscillation in the output. Thus the oscillation is called as Limit
cycle.

6. Express the fraction (-7/32) in signed magnitude and 2`s complement
notations using 6 bits.
In Signed Magnitude: 1.001110
In 2`s complement: 1.110010

7. Define the terms: autocorrelation sequence and power spectral density.

Autocorrelation sequence:
R
xx
(n,nm) E|X(n) X(nm)|
Power spectral density:
S
xx
(e
j
) _ R
xx
(m) e
-jm

m-
8. Define unbiased estimate and consistent estimate.

Expected value oI the estimate is equal to actual value is called as unbiased
estimate.
Consistent estimate: Lim Varience (estimate) 0.
N
9.What are the factors that may be considered when selecting a DSP
processor for an application?
1. Architectural Ieatures
2. Exection speed
3. Type oI arithmetic
4. Word length
10.What is pipelining?
A task is broken down in to a number oI distinct subtasks which are then
overlapped during execution. It is used in digital signal processors to
increase speed.



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PART -B (5`16 80 MARKS)

11.a) Two finite duration sequence are given by
x(n) sin (n/2) for n 0,1,2,3
h(n) 2
n
for n 0,1,2,3 Determine circular convolution using
DFT &IDFT method.
Ans: X(K) 0, -2j, 0, 2j}
H(K) 15, -36j, -5, -3-6j}
y(n) 6, -3, -6, 3}
(OR)
b) i) From first principles obtain the signal flow graph for
Computing 8-point using radix -2 DIF -FFT algorithm.
ii) Using the above signal flow graph compute DFT of
x(n) cos (n/4) ,0 n 7.

Ans: i) ReI Pg.No 334-340, DSP by Salivahanan.
ii) X(K) 0, 3, 0, 2.7-j0.7, 0, 1, 0, 1.293-j0.7}


12.a) i) Describe the design of FIR filter using frequency sampling
technique.
ii) The desired frequency response of a low pass filter is given by
H
d
() e
-j2

;

-/4 /4
1 ; other wise.
Obtain the filter coefficient, h(n) using RECTANGUAR
Window define by W(n) 1; 0 n 4
0; otherwise.

Ans: a) i) ReI Pg.No 389-391, DSP by Salivahanan.
ii) ReI Pg.No 399, DSP by Salivahanan.
(OR)

b) Design a digital Butterworth filter satisfying the constraints
0.7 [ H()[ 1.0 ; 0 0.2
[ H()[ 0.004 ; 0.6 .
Apply impulse invariant transformation method.

Ans: ReI: Pg.No: 359-362, DSP by Nagoorkani.

13. a) i) Consider the truncation of negative fraction number represented
in(+1) bit fixed point binary form including sign bit . Let (-b)
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bits be truncated .Obtain the range of truncation errors for
signed magnitude ,2`s complement and 1`s complement
representation of negative numbers.
ii) A 8-bit ADC feeds a DSP system characterised by the following
Transfer function H(Z) 1/(Z+0.5) .Estimate the steady state
Quantisation noise power at the output of the system.
Ans: a) i) ReI Pg.No 496-499, DSP by Salivahanan.
ii) ReI Pg.No 504-505, DSP by Salivahanan.

(OR)

b) i) The coefficients of a system defined by
1
H(Z)
(1-0.4Z
-1
)(1-0.55Z
-1
)
are represented in anumber with a sign bit and 3 data bits.
Determine the new pole location for 1) Direct realization and 2)
Cascade realization of first order systems.Compare the
movements of the new pole away from the original ones in both
the cases.
ii) An IIR causal filter has the system function
H(Z)Z/Z-0.97,Determine the dead band of the filter.

Ans: b) i) Direct Iorm: 1/ |1-0.875z
-1
0.125Z
-2
|
Cascade Iorm:1/|1-0.375Z
-1
||1-0.5Z
-1
|
ii) ReI Pg.No 510, DSP by Salivahanan.





14. a)i) With Suitable relation ,explain briefly the periodogram method of
power spectral estimation.Examine the consistency and bias of the
periodogram.
ii) Explain power spectrum estimation using the bartlett method.

Ans: a) i) ReI Pg.No 588-589, DSP by Salivahanan.
ii) ReI Pg.No 594, DSP by Salivahanan.

(OR)
b) i) Explain how the Black man and Tukey is used in smoothing the
periodogram? Derive the mean and variance of the power
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spectral
estimate of the blackman and Tukey method.
ii) Determine the frequency resolution of the Bartlett, Welch and
Blackman and Tukey methods of the power spectral estimation
for a quality factor Q 15 and sample is 1500.
Ans: a) i) ReI Pg.No 596-599, DSP by Salivahanan.
ii) ReI Pg.No 606, DSP by Salivahanan.


15.a) i) Explain how Harvard architecture as used by the TMS 320
family differs from the Strict Harvard architecture .Compare
this with the architecture of a satndard Von Neumann
Processor.
ii) Explain the operation of MAC unit.
Ans: ReI Pg.No 40-44, Digital Signal Processor by
B.Venkataramani &M.Bhaskar

(OR)
b) i) In relation to DSP processor, explain the following technique:
SMID, VLIW.
ii) Explain the operation of CSSU of TMS 320 C54X and explain
its use considering the viterbi operator.

Ans: ReI Pg.No 40-44 &269-270 , Digital Signal Processor by
B.Venkataramani &M.Bhaskar















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B.E/B.TECH DEGREE EXAMINATION, APRIL /MAY 2008
FIFTH SEMESTER (REGULATION - 2004)
ELECTRONICS AND COMMUNICATION ENGINEERING
EC 1302 - DIGITAL SIGNAL PROCESSING
TIME: 3 hrs Max. Mark: 100

PART A (10`2 20 MARKS)
Answer all Question


1. Define the properties of convolution.
1. Commutative property: x(n)`h(n) h(n) `x(n)
2. Associative Property: x(n)`h
1
(n)] `h
2
(n) x(n)`h
1
(n)`h
2
(n)]
3. Distributive Property: x(n)` h
1
(n)+ h
2
(n)] x(n)` h
1
(n)]+ x(n)` h
1
(n)]

2. Draw the basic butterfly diagram of radix -2 FFT.

1 1
a A a W
N
nk
b
1
1
W
N
nk

b B a - W
N
nk
b
-1
3. What are the merits and demerits of FIR filter?
Merits :
3. Linear phase Iilter.
4. Always Stable
Demerits:
3. The duration oI the impulse response should be large
4. Non integral delay.

4. What is the relationship between analog and digital frequency in
impulse invariant transformation?

Digital Frequency: T
analog Irequency
T Sampling interval


5. Identify the various factors which degrade the performance of the
digital filter implementation when finite word length is used.
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i. Input quantization error
ii. CoeIIicient quantization error
iii. Product quantization error

6. What is meant by limit cycle oscillation in digital filter?
In recursive system when the input is zero or same non-zero constant value
the non linearities due to Iinite precision arithmetic operation may cause
periodic oscillation in the output. Thus the oscillation is called as Limit
cycle.

7. Define Periodogram.
F r
xx
(m)} 1/N X (I)
2

The above equation gives the estimation oI power spectral density
(PSD). This equation is also called as Periodogram.

8. Determine the frequency resolution of the Bartlett method of power
spectrum estimates for a quality factor Q 15 .Assume that the length of
the sample sequence is 1500.
Solution:
Q Bart 1.11N I
Frequency resolution: I 15 /(1.11 *1500)
I 0.0009

9.What is pipelining?
A task is broken down in to a number oI distinct subtasks which are then
overlapped during execution. It is used in digital signal processors to
increase speed.

10.What is the principal feature of the Harvard architecture?
The program and data memory lies in two separate spaces, permitting Iull
overlap oI instruction, Fetch and execute.

PART -B (5`16 80 MARKS)

11.a) i) Disuss in detail the imporatnt properties of the DFT.
ii) Find the 4 -point DFT of the sequence x(n) cos n/4.
Ans: i)ReI Pg.No 308-311, DSP by Salivahanan.
ii) X(K) 1, 1-j1.414, 1, 1j1.414}
iii) X(K) 20,-5.8-j2.4, 0, 0.17-j0.414, 0, -0.17j0.414, 0,
-5.82j2.414}.
(OR)
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b)i) Using DIT draw the butterfly line diagram for 8-point FFT
calculation and explain.
ii)Compute an 8-point DFT using DIF FFT algorithm.
X(n) 1,2,3,4,4,3,2,1]
Ans: i) X(K) 20,-5.8-j2.4, 0, 0.17-j0.414, 0, -0.17j0.414, 0,
-5.82j2.414}.
ii) ReI Pg.No 334-340, DSP by Salivahanan.

12. a) i) Determine the magnitude response of an FIR filter (M11) and
show that the phase and group delays are constant.
ii) The desired frequency response of a low pass filter is given by
H
d
() e
-j3

;

-3/4 3/4
0 ; other wise.
Determine H(e
j
) for M 7using HAMMING window.

(OR)

b)i) For the analog transfer function H(S) 1/ (S+1)(S+2) .
Determine H(Z) using impulse invariant technique.
ii) Design a digital Butterworth filter satisfying the constraints
0.9 [ H()[ 1.0 ; 0 /2
[ H()[ 0.2 ; 3/2 .
Apply Bilinear transformation method. (16)

Ans: a) i) ReI Pg.No 437-439, DSP by Salivahanan.
ii) ReI Pg.No 383-384, DSP by Salivahanan.
b)i) ReI Pg.No 400-401, DSP by Salivahanan.
ii) ReI Pg.No 410-413, DSP by Salivahanan.

13.a) i) Discuss in detail the truncation error and round off error for sign
Magnitude and 2`s complement.
ii) Explain the quantization effects in converting analog signal in to
digital signal
Ans: a) i) ReI Pg.No 496-499, DSP by Salivahanan.
ii) ReI Pg.No 495, DSP by Salivahanan.

(OR)

b) i) A digital system, is characterized by the difference equation
y(n) 0.9y(n-1) +x(n) .Determine the dead band of the filter.
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ii) What is meant by coefficient quantization? Explain
Ans: b) ReI Pg.No 510, DSP by Salivahanan.

14.a) i) Explain the Bartlett method of averaging Periodogram.
ii) What is the relation ship between autocorrelation and power
Spectrum? Prove it.
Ans: a) i) ReI Pg.No 588-589, DSP by Salivahanan.
ii) ReI Pg.No 594, DSP by Salivahanan.
(OR)
b)i) Derive the mean and variance of the power spectral estimate of
the Blackman and Tukey method.
ii) Obtain the expression for mean and variance of the
autocorrelation function of random signals.
Ans: b) ReI Pg.No 596-599, DSP by Salivahanan.

15.a) i) Describe the MAC unit in DSP Processor.
ii) Explain the architecture of TMS 320 C5X DSP Processor.
Ans: ReI Pg.No 40-44, &256 Digital Signal Processor by
B.Venkataramani &M.Bhaskar

(OR)

b) i) Discuss in detail the four phases of the pipeline techniques.
ii) Write short notes on:
1. Parallel Logic
2. Circular register.
Ans: ReI Pg.No 40-41,56-65,60,48, Digital Signal Processor by
B.Venkataramani &M.Bhaskar












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B.E/B.TECH DEGREE EXAMINATION, NOV /DEC 2008
FIFTH SEMESTER (REGULATION - 2004)
ELECTRONICS AND COMMUNICATION ENGINEERING
EC 1302 - DIGITAL SIGNAL PROCESSING
TIME: 3 hrs Max. Mark: 100

PART A (10`2 20 MARKS)
Answer all Questions

1. Distinguish between DIT and DIF -FFT algorithm.

S.No DIT -FFT Algorithm DIF -FFT Algorithm
1. The input is in bit reversed
order; the output will be
normal order.
The input is in normal order;
the output will be bit reversed
order.
2. Each stage oI computation the
phase Iactor are multiplied
beIore add subtract operation.
Each stage oI computation the
phase Iactor are multiplied
aIter add subtract operation.

2. If H(K) is the N-point DFT of a sequence h(n) , Prove that H(K) and
H(N-K) are complex conjugates.
This property states that, iI h(n) is real , then H(N-K) H*(K) H(-K)
Proof:

By the deIinition oI DFT;
N-1
X(K) _ x(n) e
(j2nk)/N

n0
Replace K` by N-K`
N-1
X(N-K) _ x(n) e
(j2n(N-K))/N

n


3. Show that the filter with h(n) -1,0,1] is a linear phase filter.

Solution:
h(n) | -1,0,1|
h(0) -1 -h(N-1-n) -h(3-1-0) -h(2)
h(1) 0 -h(N-1-n) -h(3-1-1) -h(1)
h(2) 1 -h(N-1-n) -h(3-1-2) -h(0)
It is a linear phase Iilter.
X(N-K) X*(K)

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4. What is Prewarping? Why is it needed?
In IIR design using bilinear transIormation the conversion oI speciIied
digital Irequencies to analog Irequencies is called Pre-warping. The Pre-
Warping is necessary to eliminate the eIIect oI warping on amplitude
response.

5. Identify the various factors which degrade the performance of the
digital filter implementation when finite word length is used.

1. Input quantization error
2. CoeIIicient quantization error
3. Product quantization error

6. Express the fraction (-7/32) in signed magnitude and 2`s complement
notations using 6 bits.
In Signed Magnitude: 1.001110
In 2`s complement: 1.110010

7. What are the disadvantages of non-parametric methods of power
spectral estimation?
1. It requires long data sequence to obtain the necessary Irequency
resolution.
2. Spectral leakage eIIects because oI windowing.

8. Define unbiased estimate and consistent estimate.
Expected value oI the estimate is equal to actual value is called as unbiased
estimate.
Consistent estimate: Lim Varience (estimate) 0.
N
9.State the merits and demerits of multi ported memories.

A Mutiported memory operates as on chip memory and oII chip memory.
1. Higher perIormance because no wait states are required.
2. Lower cost than external memory.
3. Lower Power than external memory.
4. Higher perIormance because better Ilow with in the pipeline at lential
arithmetic logic unit.
5. Ability to access large memory space.



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10.What are the factors that may be considered when selecting a DSP
processor for an application?
1. Architectural Ieatures
2. Exection speed
3. Type oI arithmetic
4. Word length


PART -B ( 5`16 80 MARKS)

11. a) Two finite duration sequence are given by
x(n) Cos (n/2) for n 0,1,2,3
h(n) 0.5
n
for n 0,1,2,3 Determine circular convolution using
DFT &IDFT method.
Ans: X(K) 0, -2j, 0, 2j}
H(K) 15, -36j, -5, -3-6j}
y(n) 6, -3, -6, 3}

(OR)
b) i) From first principles obtain the signal flow graph for
Computing 8-point using radix -2 DIT -FFT algorithm.
ii) Using the above signal flow graph compute DFT of
x(n) cos (n/4) ,0 n 7.
Ans: i) ReI Pg.No 334-340, DSP by Salivahanan.
ii) X(K) 0, 3, 0, 2.7-j0.7, 0, 1, 0, 1.293-j0.7}


12. a) A band pass FIR filter of length 7 is required. It is to have lower
and upper cutt frequencies of 3 KHZ and 5 Khz respectively .
and indended to be used with a sampling frequency of 20 KHZ
Determine the filter coefficient using HANNING window
Consider the filter to be causal.
(OR)

b) Design a digital Butterworth filter satisfying the constraints
0.8 [ H()[ 1.0 ; 0 0.2
[ H()[ 0.2 ; 0.6 .
Apply Bilinear transformation method.

Ans: ReI: Pg.No: 359-362, DSP by Nagoorkani.


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13. a) i) Consider the (b+1) bit bipolar A/D converter. Obtain an
expression for signal to quantization noise ratio .
ii) Consider the truncation of negative fraction number represented
in(+1) bit fixed point binary form including sign bit . Let (-b)
bits be truncated .Obtain the range of truncation errors for
signed magnitude ,2`s complement and 1`s complement
representation of negative numbers.

Ans: a) i) ReI Pg.No 496-499, DSP by Salivahanan.
ii) ReI Pg.No 499-503, DSP by Salivahanan.

(OR)
b) The coefficients of a system defined by
1
H(Z)
(1-0.3Z
-1
)(1-0.65Z
-1
)
are represented in a number with a sign bit and 3 data bits.
Determine the new pole location for 1) Direct realization and 2)
Cascade realization of first order systems. Compare the
movements of the new pole away from the original ones in
both the cases.
Ans: b) Direct Iorm: 1/ |1-0.875z
-1
0.125Z
-2
|
Cascade Iorm:1/|1-0.375Z
-1
||1-0.5Z
-1
|


14. a)i) With Suitable relation ,explain briefly the periodogram
method of power spectral estimation. Examine the consistency
and bias of the periodogram.
ii) Explain power spectrum estimation using the bartlett method.

Ans: a) i) ReI Pg.No 588-589, DSP by Salivahanan.
ii) ReI Pg.No 594, DSP by Salivahanan.

(OR)

b) i) Explain how the Black man and Tukey is used in smoothing the
periodogram? Derive the mean and variance of the power
spectral
estimate of the blackman and Tukey method.
ii) Determine the frequency resolution of the Bartlett, Welch and
Blackman and Tukey methods of the power spectral estimation
for a quality factor Q 15 and sample is 1500.
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Ans: a) i) ReI Pg.No 596-599, DSP by Salivahanan.
ii) ReI Pg.No 606, DSP by Salivahanan.

15. a) i) Explain what is meant by instruction pipelining. Explain with
an example ,how pipelining increase the through put
efficiency.
ii) Describe the MAC unit in DSP Processor.

Ans: a) i) ReI Pg.No 45-46, Digital Signal Processor by
B.Venkataramani &M.Bhaskar
ii) ReI Pg.No 40-44, &256 Digital Signal Processor by
B.Venkataramani &M.Bhaskar

(OR)

b)i) With a suitable diagram describe the function of
Multiplier/adder units of TMS 320 C54X.
ii) In relation to DSP processor, explain the following technique:
SMID, VLIW.
Ans: b) i) ReI Pg.No 267-268, Digital Signal Processor by
B.Venkataramani &M.Bhaskar
ii) ReI Pg.No 40-44, Digital Signal Processor by
B.Venkataramani &M.Bhaskar



















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B.E/B.TECH DEGREE EXAMINATION, NOV /DEC 2009
FIFTH SEMESTER (REGULATION - 2004)
ELECTRONICS AND COMMUNICATION ENGINEERING
EC 1302 - DIGITAL SIGNAL PROCESSING
TIME: 3 Hrs Max. Mark: 100

PART A (10`2 20 MARKS)
Answer all Questions


1. What do you mean by radix-2 FFT?
The radix -2 FFT is an eIIicient algorithm Ior coputing N- point DFT oI an
N-point sequence .In radix-2 FFT the n-point is decimated into 2-point
sequence and the 2-point DFT Ior each decimated sequence is computed.
From the results oI 2-point DFT`s, the 4-point DFT`s are computed. From
the results oI 4 point DFT`s ,the 8-point DFT`s are computed and so on
until we get N - point DFT.

2. What is the relationship between z transform and DFT?
DFT oI x(n), x(k)x(z)/ze
j2k/N

X(Z) is the ZtransIorm.

3. What are the properties of FIR Filters?
FIR Iilters are stable.
FIR Iilters have linear phase.
They need higher order Iilters Ior the same magnitude response
compared to IIR Filters.

4. State the limitations of Impulse Invariance mapping technique.
In Impulse Invariance method, the mapping Irom S plane to Z plane is many
to one. This produces aliasing eIIect. Hence it is not suitable Ior designing
high pass Iilters.

5. Define zero input limit cycle oscillation.
For an IIR Iilter, implanted with inIinite precision arithmetic, the output
should approach zero in the steady state iI the input is zero,and it should
approach a constant value, iI the input is a constant. Due to Iinite word length
eIIect, the output may oscillate between positive and negative values.This is
called zero input limit cycle.


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6. What is called dead band?
The amplitude oI the output during a limit cycle is conIined to a range oI
values called the dead band oI the Iilter.

7. What is the need for spectrum estimation?
To detect the narrow band signals in noise, spectrum estimation is required.

8. Define Periodogram.
F r
xx
(m)} 1/N X (I)
2

The above equation gives the estimation oI power spectral density
(PSD). This equation is also called as Periodogram.

9. What is meant by pipelining?
A task is broken down in to a number oI distinct subtasks which are then
overlapped during execution. It is used in digital signal processors to increase
speed.

10. What are the desirable features of DSP Processors?
(i) DSP processors should have multiple registers.
(ii) Multiple operand Ietch capacity.
(iii) On-chip memory.
(iv) Multiprocessing capability.

PART -B (5`16 80 MARKS)

11. (a) (i) Describe the following properties of DFT.
(1)Convolution
(2)Time Reversal
(3)Time Shift
(4)Perioddicity. (12)
Ans: (1) DFT |x
I
(n) *x
2
(n| X
1
(k)X
2
(k)
(2) II DFTx(n)X(K) then DFT|x (-n) | X(N-K)
(3) DFT|x (n-m) N| e
-j2km/N
X(k)
(4)II x(nN)x(n) Ior all n,
Then X(KN)X(K) Ior all k. and Explanation.

(ii) Compare the computational complexity of Direct DFT computation
and FFT computation of a sequence with N64. (4)
Ans: Direct DFT computation: N
2
(64)
2
4096
FFT computation: N/2log
2
N64/2log
2
64192

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(OR)
(b) (i) Explain Decimation in time FFT algorithm for N8. (8)
Ans: (i) ReI Pg.No 320-328 , DSP by Salivahanan .

(ii) Determine the 4 point DFT of x(n) (0,1,2,3). (8)

Ans: w
4
0
x
4
1
2
3

X
4
W
4
x
4
6
-22j
-2
-2-2j

12. (a) (i) Describe the Frequency sampling method of designing FIR
Filters. (8)
Ans: (i) In Irequency sampling method the desired magnitude response
is sampled and a linear phase response is speciIied.The samples oI
the desired Irequency response are identiIied as DFT coeIIients.The
Iilter coeIIicients are determined as the IDFT oI this set oI samples.
Frequency sampling method is attractive Ior narrowband Irequency
selective Iilters.
(ii)And Explanation.
(ii) Derive the condition for linear phase in FIR filters. (8)
Ans: When h(n) is symmetric about nN-1/2
h(n)h(N-1-n)
When h(n) is antisymmetric about nN-1/2
h(n)-h(N-1-n)
(ii)And Derivation.
(OR)
(b) (i) The desired response of a low pass filter is
H
d
(e
jw
) e
-j3w
-3 /4 w 3/4
0 3/4 w
Design the filter for M7 using Hamming Window. (10)
(i) ReI Pg.No 400-401, DSP by Salivahanan.

(ii) Using Impulse Invariant mapping, convert the analog transfer
Function into digital.
Assume T0.1 sec .
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2
H(S) (6)
(s+1)(s+2)


Ans: H(Z)



13. (a) (i) Draw the Quantization noise model for a second order system

1
H(Z)
1-2rcosz
-1
+r
2
z
-2

and find the steady state output noise variance. (8)
(ii) What are called overflow oscillations? How can it be prevented?(8)
Ans: steady state output noise variance

v
2
v
1
2
v
2
2

v
1
2
v
2
2

h(n) r
n
sin(n1) u(n)

(OR)

(b) (i) Describe the effect of quantization on pole location with an
example. (6)
(ii) Explain the characteristics of a limit cycle oscillation with respect
to the system described by the difference equation
y(n) 0.95y(n-1)+x(n)
Determine the dead band of the filter. (10)
Ans: b) ReI Pg.No 510, DSP by Salivahanan.
Deadband (-0.625,0.625)
14. (a) (i) Derive the relationship between autocorrelation and power
spectral density of a signal. (8)
(ii) Explain the Bartlet method of spectrum estimation. (8)

Ans: i) ReI Pg.No 588-589, DSP by Salivahanan.
ii) ReI Pg.No 594, DSP by Salivahanan.


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(OR)
(b) (i) Describe the use of DFT in spectrum estimation. (8)
(ii) Explain the Blackman and Tukey method of spectrum estimation.
(8)
Ans: (i) ReI Pg.No 596-599, DSP by Salivahanan.


15. (a) Write note on:
(i) Harvard Arahitecture
(ii) Dedicated MAC unit
(iii) Multiple ALUs. (16)
Ans: a) i) ReI Pg.No 45-46, Digital Signal Processor by
B.Venkataramani &M.Bhaskar

ii) ReI Pg.No 40-44, &256 Digital Signal Processor by
B.Venkataramani &M.Bhaskar

(OR)
(b) Describe the Advanced modes of DSP processors in detail. (16)
Ans: ii) ReI Pg.No 40-44, Digital Signal Processor by
B.Venkataramani &M.Bhaskar





















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B.E./B.Tech. DEGREE EXAMINATION, NOVEMBER/DECEMBER 2010
EC 2302 - DIGITAL SIGNAL PROCESSING
Maximum : 100 Marks
Answer ALL questions
http://2freshesworld.com
PART A - (10 X 2 20 Marks)
1. Obtain the circular convolution of the following sequences x(n) 1, 2,1]
& h(n) 1, 1 2, 2].
Using matrix method, Appending zero to the sequence x(n),
x(n)1,2,1.0}
h(n) 1,1,2.2}
Answer X(n) 7,5,5,7}
2. How many multiplications and additions are required to compute N-point
DFT using radix-2 FFT?
Complex Addition - Nlog
2
N
Complex Multiplications (N/2)log
2
N
3. What is prewarping?
In IIR design using bilinear transIormation the conversion oI speciIied
digital Irequencies to analog Irequencies is called Pre-warping. The Pre-
Warping is necessary to eliminate the eIIect oI warping on amplitude response.

4. What is the advantage of direct form II realization when compared to
direct form I realization?
In direct Iorm II structure the number oI delay elements required is exactly
halI that Ior direct Iorm I structure .Hence it requires less amount oI
memory.
5. Give the equations for Hamming window and Blackman window.
W
H
(n) 0.54 - 0.46cos(2n[/N-1) ; 0 _ n _ N-1
W
B
(n) 0.42 0.5 cos(2n[/N-1) 0.08 cos (2n[/N-1) ; 0 _ n _ N-1

6. Determine the transversal structure of the system function ( ) 1 2 3 1 2 3 4
H z + z ? ? z ? ? z ?


7. What is truncation?
Truncation is the process oI discarding all bits less signiIicant than least
signiIicant bit that is retained.
8. What is product quantization error?
In digital computations the output oI multipliers are quantized to Iinite word
length inorder to store the result in the registers
9. What is decimation?
The process oI reducing the sampling rate oI the signal is called decimation
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(sampling rate compression).
10.What is sub band coding?
It is application oI multirate signal processing
http://2freshesworld.comrld.com
PART B (5 ; 16 80 Marks)
http://2freshesworld.
11. (a) (i) Compute the eight-point DFT of the sequence Using the radix-2
decimation-in-time algorithm. (10)
ReI Pg.No 207-209, DSP by A.Nagoor Kani.
(ii) Explain overlap-add method for linear FIR filtering of a long sequence. (6)
ReI Pg.No 191, DSP by A.Nagoor Kani.

(Or)

(b) (i) Compute the eight-point DFT of the sequence By using the decimation-
in-frequency FFT algorithm. (10)
ReI Pg.No 215-219, DSP by A.Nagoor Kani.
(ii) Summarize the properties of DFT. (6)
ReI Pg.No 158-161, DSP by A.Nagoor Kani.

12. (a) Determine the system function H(z) of the Chebyshevs low pass digital
filter with the specifications 1 dB p ? ripple in the pass band 0 ?? ? 0.2 ? s
? 15 dB ripple in the stop band 0.3? ?? ? ? using bilinear transformation
(assume T 1 sec). (16)
ReI Pg.No 365-367, DSP by A.Nagoor Kani.
(Or)

(b) Obtain the direct form I, direct form II, cascade and parallel form
realization for the system
y(n) 0.1 y(n -1) + 0.2 y(n - 2) + 3x(n) + 3.6x(n -1) + 0.6 x(n - 2) (16)
ReI Pg.No 54-57, DSP by A.Nagoor Kani.



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13. (a) Design an ideal high pass filter with a frequency response Find the
values of h(n) for N 11 using hamming window. Find H(z) and determine the
magnitude response. (16)
ReI Pg.No 298-301, DSP by A.Nagoor Kani.

(Or)

(b) (i) Determine the coefficients h(n)] of a linear phase FIR filter of length M
15 which has a symmetric unit sample response and a frequency response
that satisfies the condition (10)
ReI Pg.No 308-311, DSP by A.Nagoor Kani.

(ii) Obtain the linear phase realization of the system function
H (z ) + z + z + z + z + z z (6)
ReI Pg.No 207-209, DSP by A.Nagoor Kani.

14. (a) Discuss in detail the errors resulting from rounding and truncation.
ReI Pg.No 496-499, DSP by Salivahanan
Or
(b) Explain the limit cycle oscillations due to product round off and overflow
errors. (16)
ReI Pg.No420-423, DSP by A.Nagoor Kani.

15. (a) Explain the polyphase structure of decimator and interpolator. (16)
ReI Pg.No 541-551, DSP by Salivahanan.

Or
(b) Discuss the procedure to implement digital filter bank using multirate
signal processing. (16)
ReI Pg.No 578-581, DSP by Salivahanan.


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