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DSP Am08

This document appears to be an exam for a course on digital signal processing. It contains 15 questions across two parts: - Part A contains 10 short answer questions testing key concepts like the z-transform, FFT algorithms, filtering techniques, and sampling theory. - Part B contains 5 longer form questions asking the student to complete tasks like analyzing signals and systems, designing digital filters using different techniques, and discussing topics like multirate signal processing, quantization noise, and speech compression.

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Senthil Kumar
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© Attribution Non-Commercial (BY-NC)
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45 views3 pages

DSP Am08

This document appears to be an exam for a course on digital signal processing. It contains 15 questions across two parts: - Part A contains 10 short answer questions testing key concepts like the z-transform, FFT algorithms, filtering techniques, and sampling theory. - Part B contains 5 longer form questions asking the student to complete tasks like analyzing signals and systems, designing digital filters using different techniques, and discussing topics like multirate signal processing, quantization noise, and speech compression.

Uploaded by

Senthil Kumar
Copyright
© Attribution Non-Commercial (BY-NC)
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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I Reg. No. : C n

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v 42L6
DEGREEEXAMINATION,APRIUMAY 2008. B.E.lB.Tech.
Seventh Semester (Regulation 2004) Computer Scienceand Engineering IT L252- DIGITAL SIGNAL PROCESSING (Common to B.E. (Part-Time) Sixth SemesterRegulation 2005) Time : Three hours

Answer ALL questions.

PARTA-(10 x2=20 marks)

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N.

co m

Maximum : 100 marks

1.

2. 3. 4. 5. 6. 7. 8. g.

State and prove convolution property of z transform. Draw Radix 4 butterfly structure for (DIT) FFT algorithm. Find DFT for [, 0, 0, 1]. Write frequency translation for BPF from LPF. Compare Butterworth, Chebyshevfilters. What is Gribb's phenomenon. Write procedure for designing FIR filter using windows. Draw block diagram to convert a 500 m/s signal to 2500 m/s signal and state the problem due to this conversion. List errors due to finite word length in filter design.

10.

ww

w.

aa na

State Sampling theorem, and n x(t) = 5 sin 250n t + 6 cos300 t

find

Nyquist

rate

of

the

signal

PARTB-(5x16=80marks) 11. (a) (i) Find the response of the system if the input i s { r , 4 , 6 , 2 } a n d (8) of impulse response the systemis [, 2, 3, 1]. F i n d r " u a n d r r , , f o r x = $ , 0 , 2 , 3 ] a n d y = { 4 , 0 , 7 ,2 \ . Or (b) (i) Check whether the system y(")=ay(n-l)+r(n)
shift invariant. and stable.

(ii )

(8)

is linear, casual,

(10) (6)

( i i ) F i n d c o n v o l u t i o nS , and [, O,+rr. { of ?r,r}

co m
bilinear

1 2 . , ( a ) F i n d D F T f o r [ , L , 2 , 0 , I , 2 , 0 , 1 ] u s i n g F F T DIT butterfly algorithm


and plot the spectrum. Or (16)

N.

(h)

(i) (ii)

va

IDFT for [, 4,3,I] using FFT-DIF method. _Find FindDFT for [, 2,3,4,L].

(8)

(s)
(8)

aa na

13. (a)

(i)

Design a digital filter with I/(r)=

-:usins ? = 1sec. -""o s2 + 7s -I2 transform

(ii)

Design

digtal

filter

using

for

w.

ww

with ^F/(s)= =r----(s+1)(s+2J

cutoff frequency as 100 rad./sec and (8)

sampling time = 1.2 ms. Or

(b)

(i)

Realize the following filter using cascade and parallel form with direct form-I structure.
- - r - 9 - -3

I+z

+z

+Cz

(t n r-' )\t * zr-' + +r-' )'


(ii) Find H(s) for a 3'd order lowpass butter worth filter.

( 1 0) (6)

v 42t6

14.

(a)

Designa linear HPF using Hanning windolv u.'ith N = 9 H@)=1 -0


-nLow, otherwise Or and ru. to n

(16)

(b)

Design a LPF for the following response. Use Hamming window rvith (16) N =7.

co m

15. (a)

(i)

Show deadbandeffecton y(n)= .95 y(n-t)+x(n) t o 4 b i t s .A s s u m ex ( O ) =0 . 7 5 a n d y ( - 1 ) = O .

systemrestricted

(10) (6)

va

Or

N.

(ii)

processing. Discussabout Multirate sig"nal

(ii)

Discuss about quantization noise and derive the equation for (8) finding quantization noise power.

ww

w.

aa na

ft)

(i)

Explain how the speechcompressionis achieved'

(8)

v 42t6

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