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Reg. No. :

Question Paper Code :

20091

M.E./M.Tech. DEGREE EXAMINATION, JANUARY 2011.

First Semester
Applied Electronics

(Common to Communication Systems and Computer and Communication)

248101 ADVANCED DIGITAL SIGNAL PROCESSING
(Regulation 2010)

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Time : Three hours

Maximum : 100 marks

Answer ALL questions.

PART A (10 2 = 20 marks)

Define: Ensemble Average with an example.

2.

State Parsevals theorem.

3.

Define Periodogram.

4.

Write the Yule-Walker equation for a ARMA (p, q) process.

5.

What is the use of a Wiener smoothening filter?

6.

Write the minimum error equation obtained in Pronys method.

7.

What is need for an adaptive filter?

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1.

Compare the adaptive noise cancellation and adaptive echo cancellation.

9.

List out some applications of multirate signal processing.

10.

Multistage implementation of Multirate system will reduce the computational

complexity at each stage- Justify.

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8.

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PART B (5 16 = 80 marks)
11.

(a)

A linear shift invariant system is described by the transfer function

H(z ) = 1 0.5 z 1

) (10.33 z ) which is excited by zero mean exponentially

1

correlated noise x(n) with an auto correlation sequence rx (k ) = (0.5) .

Let y(n) be the output process, y(n) = x(n)*h(n). Determine the

(i)

Power spectrum Py (z ) of y(n)

(ii)

Auto correlation sequence of y(n)

(iii) Cross correlation sequence between x(n) and y(n)

(iv)

Cross power spectral density Pxy (z ) .

(16)

Or
(i)
(ii)

Explain briefly about ergodic process with necessary equations and

also state the mean ergodic theorems.
(10)

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(b)

The power spectrum of a wide sense stationary process x(n) is given

as Px e j = (25 24 cos )/ (2610 cos ) . Find the whitening filter
H(z) that produces unit variance white noise when the input is x(n).
(6)

( )

12.

(a)

With necessary derivation, explain the periodogram averaging using

Bartletts method and compare the same with the Welch method of
periodogram averaging.
(16)
Or

(i)

Consider that Bartletts method is used to estimate the power

spectrum of a process from a sequence of N = 2000 samples.
(1)

What is the minimum length L that may be used for each

sequence to have a resolution of f = 0.005?

(2)

Explain why it would not be advantageous to increase

L beyond the value found in (I)

(3)

The quality factor of a spectrum estimate is defined to be the

inverse of the variability Q =1 / v . Using Bartletts method,
what is the minimum number of data samples, N that are
necessary to achieve a resolution of f = 0.005 and a quality
factor that is 5 times larger than that of the periodogram? (6)

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(b)

(ii)

Given
the
autocorrelation
sequence,
rx (0 ) =1, rx (1) = 0.8,
rx (2) = 0.5, rx (3) = 0.1 . Determine the reflection coefficients j , the

model parameters a j (k ) and the modeling errors j for j = 1, 2 ,3

using Levinson Durbin recursion.

(10)

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(a)

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13.

(i)

With necessary equations and neat sketches, explain the pole zero
modeling using the Pronys method.
(8)

(ii)

Consider the signal x(n) consisting of a single pulse of length N,

x (n ) =1; n = 0, 1,L N 1 ;

0 ; elsewhere

Use Pronys method to model x(n) as a unit sample response of a

linear shift invariant filter having one pole and one zero.
(8)
Or

14.

(a)

(i)

Starting from the basic principles, derive the expression for

minimum error for a FIR Wiener filter in terms of autocorrelation
(8)
matrix Rx and cross correlation vector rdx.

(ii)

With necessary expressions, briefly explain about the discrete

Kalman filter.
(8)

(i)

(ii)

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(b)

With a neat block diagram, explain the operation of an adaptive

channel equalizer.
(8)
Show that the normalized LMS algorithm is equivalent for using
the update equation w n +1 = w n + e' (n ) x (n ) where e(n) is the error
at time n that is based on the new filter coefficients
w n +1 , e'(n ) = d(n ) w T n+1x(n). Discuss the relationship between
(8)
and the parameter in the normalized LMS algorithm.
Or

15.

(b)

Derive the expression for minimizing the weighted least squares error
using Recursive Least squares algorithm and compare the RLS algorithm
with sliding window RLS algorithm.
(16)

(a)

(i)

Write short notes on Wavelet transformation.

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(ii)

With neat sketches and necessary equations, briefly explain the

time domain and frequency domain characterization of a down
sampler with an integer factor of M.
(12)

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(b)

(4)

Or

(i)

Obtain the polyphase decomposition of factor of 3 decimator using a

9-tap FIR lowpass filter with symmetric impulse response.
(8)

(ii)

With a neat block diagram, briefly explain about the subband

coding of speech signals.
(8)

20091