B.E/B.Tech.

DEGREE EXAMINATION, APRIAL/MAY 2008

Sixth semester
Electrical and Electronics Engineering
EC 1361-DIGITAL SIGNAL PROCESSING
(Common to Instrumentation and Control Engineering and Electronics and instrumentation
Engineering)
(Regulation2004)
PART A-(10*2=20)
1. What is a linear time invariant system?
2. What is known as aliasing?
3. Define ROC in Z-transform.
4.determine the Z-transform of the sequence x(n)={2,1,-1,0,3}
5. Define DFT pair.
6. Draw the basic butterfly structure for DITFFT and DIF FFT Algorithms.
7. State the condition for linear phase in FIR filters for symmetric and anti symmetric response.
8. What is called pre warping?
9. What are the various interrupt types supported by TMS 320 C 54?
10. Mention the function of the program controller of the DSP processor TMS 320 C 54.
PART B-(5*16=80)
11.(a)(i) explain the concept of energy and power signals. Also checks whether the following
signals are energy or power signal.
(1)x(n)=(1/3)n u(n)
(2)x(n)=sin (p*/4)n (12) * see note
(ii)briefly explain Quantization. (4)
Or
(b) check the following system for linearity, time invariance , causality and stability .
(i) y(n) = e^x(n)
(ii)y(n) = x(-n+2). (16)
12.(a)(i) determine the Z-transform of x(n)=cos wn u(n). (6)
(ii) state and prove the following properties of Z-transforms:
(1)time shifting
(2)time reversal
(3)differentiation
(4)scaling in Z domain. (10)
or
(b) (i) determine the inverse Z transform of X(z)=(1+3z ^-1)/(1+3z^ -1 +2z ^-2) for |z| >2. (8)
(ii) compute the response of the system y(n)=0.7y(n-1)-0.12y(n-2)+x(n-1)+x(n-2)
to input x(n)= n u(n).
(8)
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13.(a)(i) derive and draw the flow graph of the Radix-2 DIF FFT algorithm for the computation
of 8-ponit DFT. (10)
(ii) what are differences and similarities between DIT and DIF FFT algorithm? (6)
or
(b)(i) compute the 8-point DFT of the sequence x(n) ={1,2,3,4,4,3,2,1} (10)
(ii) illustrate the concept of circular convolution property of DFT. (6)
14.(a)(i) obtain the cascade and parallel realization of the system described by
y(n)= -0.1y(n-1)+0.2y(n-2)+3x(n)+3.6x(n-1)+0.6x(n-2). (10)
(ii) discuss about any three window function used in the design of FIR filters. (6)
or
(b) (i) design a digital Butterworth filter satisfying the following constraints with T=1sec.using
Bilinear transformation. (12)
0.707=< |H(ej?)|=< 1 for 0 = <W=< 2|H(ej?)|=<0.2 for 3p/4=
(ii) what are the different frequency transformations in analog domain? (4)
15(a) (i) describe the function of on chip peripherals of TMS 320 C 54 DSP processor.
(12)
(ii) what are the different buses of TMS 320 C 54 and their functions? (4)
or
(b)discuss in detail the various quantization effect in the design of digital filters. (16)N
B.E / B.Tech, DEGREE EXAMINATION , MAY / JUNE 2006
Fourth Semester (IT)
Computer Science and EngineeringVII
IT 1252 DSP
(Regulation 2004)
Time: 3 hours Maximum : 100 marks
Answer ALL Questions
PART A (10 X 2 = 20 marks)
1. What is meant by aliasing? How can it be avoided?
2. Is the system y(n) =In{x9n)} is linear and time invariant?
3. Define DFT pair.
4. Differentiate b/w DIT and DIF FFT algorithms.
5. Find the transfer function for normalized Butterworth filter of order 1 by
determing the pole values.
6. What does frequency warping mean?
7. State the advantages of FIR filter over FIR filter.
8. List out the different forms of structural realizations available for realizing a FIR
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system.
9. Bring out the difference between fixed point and floating point arithmetic.
10. How will you avoid cycle oscillations due to overflow in addition?
PART B (5 X 16 = 80 marks)
11. (i) With a neat diagram , explain the analysis and synthesis part of a vocoder
in detail.
11 (ii) The system is characterized by the difference equation
y(n) = 0.75 y(n1)
+ 5x(n) . The input signal x(n) has a range of 6v
to +6v
represented by 8 bits. Find the quantization step signal, variance of the error
signal, variance of the error signal and variance of the quantization noise at the
output.
12.(a) (i)Find the output response of the system given the input signal
12.(a) (ii) Define correlation and bring out the difference between convolution and
correlation.
12.(b).(i) Determine the Z
transform of the signal. ) 1 ( ) ( ) ( - - - = n u b n u a n x n n b>a
and plot the ROC.
12.(b)(ii) Find the steady state value given.
12.(b)(ii) Find the system function of the system described by y(n) = 0.75
y (n1)
+0.125 y (n2)
= x (n) x(n1)
and plot the poles and zeros of H(z).
13.(a)(i) Using DFTIDFT
method, perform circular convolution of the two
sequences x(n) = { 1, 2, 0, 1} and h(n) = {2, 2, 1, 1 }.
13.(a) (ii) State & prove the circular convolution property of DFT.
13.(b)(i) Determine the number of complex Multiplications and additions involved in
Npoint
Radix 2 and Radix4FFTT
Algorithm .
13.(b)(ii) Compute the 8point
DFT of the given data sequence
{}0,0,0,0,2
1,2
1,2
1,2
1 ) ( = n x using radix 2
decimation in Time FFT
Algorithm.
14.(a) (i)Connect the analog Filter with system
function. [ { } 9 ) 1 . 0 ( / ) 1 . 0 ( ) ( 2 + + + = S S S H a into a digital IIR filter using
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impulse invariance method.(Assume T=0.1 sec)

14.(a) (ii) Obtain the direct form I, canonic form and parallel form realization
structures for the system given by the difference equation.
14.(b) Design and realize a digital Butterworth filter using bilinear transformation to meet
the following requirements.
1 ) ( 707 . 0 jw e H 2 0 p w
2 . 0 ) ( jw e H p w p 2 3
15. (a) (i) Determine the filter coefficient h(n) of length M=15 obtained by sampling its
frequency response as .
15.(a)(ii) Obtain the transversal and linear phase relazation for a filter given by
h(n) ={0.5, 2.88, 3.404, 2.88, 0.5}
H(z) = 0.5+2.88Z 1
15.(b) Design a digital filter with p p = w e H jw
d2
1)(
o otherwise
using Hamming Window with N=7. Draw the frequency response
ANNA UNIVERSITY :: CHENNAI 600 025
MODEL QUESTION PAPER
B.E. Computer Science and Engineering
V SEMESTER
CS 331 - DIGITAL SIGNAL PROCESSING
Part-A (10 x 2 = 20 Marks)
1. Represent the following discrete time signal in Z domain.
X(n) = {.6n 0 < n < 5
0 otherwise
2. Draw the 8 point Radix DIT signal flow graph.
3. Write down the procedure for designing IIR filter.
4. Write the relationship between S domain and Z domain .
5. What is Gibbs Phenomenon?
6. Write the expression for Kaiser window function.
7. Demonstrate the two types of cascade realization in implementation of filters.
8. Realize the filter
9. Give the relationship between autocorrelation sequence and power spectrum density of a
discrete sequence and list out their use.
10. What is meant by upsampling show its implementation?
Part-B (5 x 16 = 80 Marks)
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11. i. Find h(n) for

ii. Given x1(n) = {2,1,1,2}, x2(n) = {1, -1, -1, 1}
Find the convolution of x1(n) and x2(n)
12. (a) Design a LPF with following specifications. Use Hamming window
and at least 8 points.

(b) Design and implement a linear phase FIR filter of length M = 15 which has the following unit
sample response
H(2 k/15) = { 1 k = 0, 1, 2, 3
0 k = 4, 5, 6, 7}
13. (a) i. Obtain H(z) from H(s) when T = 1 sec.
ii. Design a digital BPF using w1 & w2 as cutoff frequencies
(OR)
(b) i. Find H(s) for 3rd order filter.
ii. Design a Chebyshev LPF filter for the following specification
Maximum pass band ripple = 1.2dB.
At W = 2.5 r/s, the loss is 30 dB at 1r/s.
14. (a) i. Discuss errors due to finite word length effect.
ii. Realize the following H(z) given by
using cascade and Parallel form with Direct form-I.
(OR)
(b) i. What is meant by quantization error? Explain briefly.
ii. Realize the following filter using cascade technique with
DF-I and DF-II.

15. (a) Briefly explain

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i. Interpolator
ii. Decimator
iii. Effects due to sampling rate conversion
iv. Applications of multirate signal processing.
(OR)
(b) i. Briefly explain energy density spectrum, periodogram.
ii. Find periodogram for the following sequence using DFT.
{1,0,2,0,3,1,0,2}
******

B.E/B.Tech. Degree Examination, November/December 2007

Seventh Semester
(Regulation 2004)
Computer Science and Engineering
IT 1252-Digital Signal Processing
Part-A(10*2=20 marks)
1. Determine which of the following signals are periodic and compute their fundamental period.
(a) sin v2 ?t
(b) sin 20 ? t + sin 5 ? t
2. State the convolution property of Z-transform.
3. Determine the circular convolution of the sequence x1(n)={1,2,3,1} and x2(n)={4,3,2,1}.
4. Draw the basic butterfly diagram for radix 2 DIT-FFT and DIF-FFT.
5. Determine the order of the analog Butterworth filter that has a -2 db pass band attenuation at a
frequency of 20 rad/sec and atleast -10 db stop band attenuation at 30 rad/sec.
6. By Impulse Invariant method, obtain the digital filter transfer function and differential
equation of the analog filter H(s)=1 / (s+1)
7. Distinguish between FIR and IIR filters.
8. What are Gibbs oscillations?
9. Explain briefly the need for scaling in the digital filter realization.
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10. What do you mean by limit cycle oscillations in digital filter ?

Part-B(5*16=80 marks)
11. (a) (i) Compute the convolution y(n) of the signals
x(n)= an, -3=n=5
0 , elsewhere
and
h(n)= 1, 0=n=4
0, elsewhere [8]
(ii) A discrete-time system can be static or dynamic, linear or nonlinear, Time invariant or time
varying, causal or non causal, stable or unstable. Examine the following system with respect to
the properties also.
(1) y(n) = cos [x(n)]
(2) y(n)=x(-n+2)
(3) y(n)=x(2n)
(4) y(n)=x(n).cos?0(n) [8]
(OR)
(b) (i) Determine the response of the casual system.
y(n)-y(n-1)=x(n)+x(n-1) to inputs x(n)=u(n) and x(n)=2-n u(n). Test its stability. [8]
(ii) Determine the IZT of X(z)=1 / [(1-z-1)(1-z-1)2] [8]
12. (a ) Compute the eight point DFT of the sequence x(n)={ ,,,,0,0,0,0} using radix2
decimation in time and radix2 decimation in frequency algorithm. Follow exactly the
corresponding signal flow graph and keep track of all the intermediate quantities by putting them
on the diagram. [16]
(OR)
(b) (i) Discuss the properties of DFT. [10]
(ii)Discuss the use of FFT algorithm in linear filtering. [6]
13. (a) (i) Derive bilinear transformation for an analog filter with system function
H(s)=b / (s+a) [8]
(ii)Design a single pole low pass digital IIR filter with -3 db bandwidth of 0.2? by use of bilinear
transformation. [8]
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(OR)
(b) (i) Obtain the Direct Form I, Direct Form II, cascade and parallel realization for the following
system
Y(n)= -0.1y(n-1)+0.2y(n-2)+3x(n)+3.6x(n-1)+0.6x(n-2) [11]
(ii) Discuss the limitation of designing an IIR filter using impulse invariant method. [5]
14. (a) (i) Prove that an FIR filter has linear phase if the unit sample response satisfies the
condition h(n)= h(M-1-n), n=0,1,.M-1. Also discuss symmetric and antisymmetric cases of
FIR filter. [10]
(ii) Explain the need for the use of window sequences in the design of FIR filter. Describe the
window sequences generally used and compare their properties. [6]
(OR)
(b) (i) Explain the type 1 design of FIR filter using frequency sampling
technique. [8]
(ii)A low pass filter has the desired response as given below
Hd(ei?) = e-i3? , 0 = ? = ?/2
0, ?/2=?=?
Determine the filter coefficients h(n) for M=7 using frequency sampling method. [8]
15. (a) Two first order low pass filter whose system functions are given below are connected in
cascade. Determine the overall output noise power.
H1(z) = 1/ (1-0.9z-1) and H2(z) = 1/ (1-0.8z-1) [16]
(OR)
(b) (i) Describe the quantization errors that occur in rounding and truncation in twos
complement. [6]
(ii) Draw a sample/hold circuit and explain its operation. [4]
(iii)What is a Vocoder ? Explain with a block diagram. [6]