Wednesday, October 28, 2009

DIGITAL SIGNAL PROCESSING

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 bydeterming the

pole values.

6. What does ‘frequency warping’ mean?

7. State the advantages of FIR filter over FIRfilter?

8. List out the different forms of structural realizations available for realizing a FIRsystem

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 vocoderin detail.

11 (ii) The system is characterized by the difference equationy(n) = 0.75 y(n1)+ 5x(n) .

The input signal x(n) has a range of 6vto +6vrepresented by 8 bits. Find the quantization

step signal, variance of the errorsignal, variance of the error signal and variance of the

quantization noise at theoutput.

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

andcorrelation.

12.(b).(i) Determine the Ztransform of the signal. ) 1 ( ) ( ) ( - - - = n u b n u a n x n n

b>aand plot the ROC.

12.(b)(ii) Find the steady state value given.

12.(b)(iii) Find the system function of the system described by y(n) = 0.75y (n1)+0.125 y

(n2)= x (n) –x(n1)and plot the poles and zeros of H(z).

13.(a)(i) Using DFTIDFTmethod, perform circular convolution of the twosequences 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

inNpointRadix 2 and Radix4FFTTAlgorithm .

13.(b)(ii) Compute the 8pointDFT of the given data sequence{ } 0 , 0 , 0 , 0 , 21 , 21 , 21 ,

21 ) ( = n x using radix 2decimation in Time FFTAlgorithm.

14.(a) (i)Connect the analog Filter with systemfunction. [ { } 9 ) 1 . 0 ( / ) 1 . 0 ( ) ( 2 + +

+ = S S S H a into a digital IIR filter usingimpulse invariance method.(Assume T=0.1 sec)

14.(a) (ii) Obtain the direct form I, canonic form and parallel form realizationstructures

for the system given by the difference equation.

14.(b) Design and realize a digital Butterworth filter using bilinear transformation to

meetthe 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

itsfrequency response as .

15.(a)(ii) Obtain the transversal and linear phase relazation for a filter given byh(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 jwd 21 ) (o otherwiseusing Hamming

Window with N=7. Draw the frequency response


Click the following link to download:
http://www.ziddu.com/download/7201563/DIGITALSIGNALPROCESSING.pdf.html

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