SARDAR RAJA COLLEGE OF ENGINEERING

ALANGULAM
DEPARTMENT OF ELECTRONICS & COMMUNICATION ENGINEERING

MICRO LESSON PLAN

SUBJECT NAME
SUBJECT CODE
YEAR/SEM
BRANCH

:
:
:
:

DIGITAL SIGNAL PROCESSING

EC52
III YEAR / V SEM
ECE

STAFF NAME: Mr.M.VELLAPANDIAN

Asst.Prof / ECE

SUBJECT DESCRIPTION AND OBJECTIVES

AIM:
To enable the students to study the signal processing methods and processors.

OBJECTIVES:

To study DFT and its computation

To study the design techniques for digital filters
To study the finite word length effects in signal processing
To study the non-parametric methods of power spectrum estimations
To study the fundamentals of digital signal processors.

UNIVERSITY TEXT BOOKS:

1. John G Proakis and Manolakis, Digital Signal Processing Principles, Algorithms
and Applications, Pearson, Fourth Edition, 2007.
2. S.Salivahanan, A. Vallavaraj, C. Gnanapriya, Digital Signal Processing,
TMH/McGraw Hill International, 2007

COLLEGE REFERRED TEXT BOOKS:

1. John G Proakis and Manolakis, Digital Signal Processing Principles, Algorithms
and Applications, Pearson, Fourth Edition, 2007.
2. S.Salivahanan, A. Vallavaraj, C. Gnanapriya, Digital Signal Processing,
TMH/McGraw Hill International, 2007
3. P.Ramesh Babu, Digital Signal Processing, Scitech Publications (India) Pvt Ltd,
Fourth Edition, 2008

REFERENCES:
1. E.C. Ifeachor and B.W. Jervis, Digital signal processing A practical approach,
Second edition, Pearson, 2002.
2. S.K. Mitra, Digital Signal Processing, A Computer Based approach, Tata McGraw
Hill, 1998.
3. P.P.Vaidyanathan, Multirate Systems & Filter Banks, Prentice Hall, Englewood
cliffs, NJ, 1993.
4. Johny R. Johnson, Introduction to Digital Signal Processing, PHI, 2006.

EC52 DIGITAL SIGNAL PROCESSINGL T P C

3104

AIM
To study the signal processing methods and processors.
OBJECTIVES
To study DFT and its computation
To study the design techniques for digital filters
To study the finite word length effects in signal processing
To study the non-parametric methods of power spectrum estimations
To study the fundamentals of digital signal processors.
UNIT I

DISCRETE FOURIER TRANSFORM

DFT and its properties, Relation between DTFT and DFT, FFT computations using
Decimation in time and Decimation in frequency algorithms, Overlap-add and save
methods
UNIT II

INFINITE IMPULSE RESPONSE DIGITAL FILTERS 9

Review of design of analogue Butterworth and Chebyshev Filters, Frequency

transformation in analogue domain Design of IIR digital filters using impulse
invariance technique Design of digital filters using bilinear transform pre warping
Realization using direct, cascade and parallel forms.
UNIT III

FINITE IMPULSE RESPONSE DIGITAL FILTERS

Symmetric and Antisymmetric FIR filters Linear phase FIR filters Design using
Hamming, Hanning and Blackmann Windows Frequency sampling method
Realization of FIR filters Transversal, Linear phase and Polyphase structures.
UNIT IV

FINITE WORD LENGTH EFFECTS

Fixed point and floating point number representations Comparison Truncation and
Rounding errors - Quantization noise derivation for quantization noise power
coefficient quantization error Product quantization error - Overflow error
Roundoff noise power - limit cycle oscillations due to product roundoff and overflow
errors signal scaling
UNIT V

MULTIRATE SIGNAL PROCESSING

Introduction to Multirate signal processing-Decimation-Interpolation-Polyphase

implementation of FIR filters for interpolator and decimator -Multistage
implementation of sampling rate conversion- Design of narrow band filters Applications of Multirate signal processing.
L: 45, T: 15, Total= 60 Periods

TEXT BOOKS:
1. John G Proakis and Manolakis, Digital Signal Processing Principles, Algorithms
and Applications, Pearson, Fourth Edition, 2007.
2. S.Salivahanan, A. Vallavaraj, C. Gnanapriya, Digital Signal Processing,
TMH/McGraw Hill International, 2007
REFERENCES:
1. E.C. Ifeachor and B.W. Jervis, Digital signal processing A practical approach,
Second edition, Pearson, 2002.
2. S.K. Mitra, Digital Signal Processing, A Computer Based approach, Tata McGraw
Hill, 1998.
3. P.P.Vaidyanathan, Multirate Systems & Filter Banks, Prentice Hall, Englewood
cliffs, NJ,1993.
4. Johny R. Johnson, Introduction to Digital Signal Processing, PHI, 2006.

MICRO-LESSON PLAN
Hour
No.

Week
No.

Topic

T/R
Book
No.

Page
No.

A /V
class

UNIT I
1
2
3
4
5

6
7
8
9
10
11
12

II

III

13
14
III
15
16
17
18

IV

19
20
21
22
23
24

25
26
27
28

V
VI

DISCRETE FOURIER TRANSFORM

Introduction
DFT and its properties
Relation between DTFT and DFT
FFT computations using Decimation in time
FFT computations using Decimation in time
FFT computations using Decimation in
frequency
FFT computations using Decimation in
frequency
Overlap-add and save methods
Overlap-add and save methods
Tutorial - Problems
Tutorial - Problems
Tutorial - Problems
UNIT II

T2
T2
T2
T2
T2

305
305-311
306-308
319-334
319-334

T2

334-344

T2

334-344

T2
T2
T2
T2
T2

368-373
368-373
377-379
377-379
377-379

INFINITE IMPULSE RESPONSE DIGITAL FILTERS

T2
Introduction
417-418
Review of design of analogue Butterworth
T2
432-439
Filters
Review of design of analogue Chebyshev
T2
439-446
Filters
T2
Frequency transformation in analogue domain
446-450
Design of IIR digital filters using impulse
T2
423-427
invariance technique
Design of digital filters using bilinear
T2
427-432
transform
T2
pre warping
427-432
Realization using direct, cascade and parallel
T2
455-489
forms.
Realization using direct, cascade and parallel
T2
455-489
forms.
T2
Tutorial - Problems
489-495
T2
Tutorial - Problems
489-495
T2
Tutorial - Problems
489-495
UNIT III
FINITE IMPULSE RESPONSE DIGITAL FILTERS
T2
Introduction
380
T2
Symmetric and Antisymmetric FIR filters
381-383
T2
Linear phase FIR filters
384-391
Design using Hamming, Hanning and
T2
392-402
Blackmann Windows

Yes

Yes

Yes

29
VI
30
31
32
33
34
35
36

VII

VIII

37
38
39

VIII

40
41
42
43

IX

44
45
46
47
48

49
50
51
52
53

XI

54
55
56
57
58
59
60

XII

Design using Hamming, Hanning and

Blackmann Windows
Design using Hamming, Hanning and
Blackmann Windows
Frequency sampling method
Realization of FIR filters
Transversal, Linear phase and Polyphase
structures
Tutorial - Problems
Tutorial - Problems
Tutorial - Problems
UNIT IV
FINITE WORD LENGTH EFFECTS
Introduction
Fixed point and floating point number
representations
Comparison Truncation and Rounding
errors
Quantization noise derivation for
quantization noise power
coefficient quantization error Product
quantization error
Overflow error
Round off noise power
limit cycle oscillations due to product round
off and overflow errors
signal scaling
Tutorial - Problems
Tutorial - Problems
Tutorial - Problems
UNIT V
MULTIRATE SIGNAL PROCESSING
Introduction
Multirate signal processing
Decimation
Interpolation
Polyphase implementation of FIR filters for
interpolator and decimator
Multistage implementation of sampling rate
conversion
Design of narrow band filters
Design of narrow band filters
Applications of Multirate signal processing
Tutorial - Problems
Tutorial - Problems
Tutorial - Problems

T2

392-402

T2

392-402

T2
T2

389-392
482-485

T2

485-489

T2
T2
T2

414-416
414-416

489-495

T2

496

T2

496-499

T2

497-499

T2

499-504

T2

505-507

T2
T2

512
502-505

T2

510-512

T2
T2
T2
T2

518-519
521-522
521-522
521-522

T2
T2
T2
T2

523
523-524
524-530
530-533

T2

541-551

T2

533-535

T2
T2
T2
T2
T2
T2

565-581
565-581
578
581-583
581583
581583

Yes

Yes

T1 - John G Proakis and Manolakis, Digital Signal Processing Principles,

Algorithms and Applications, Pearson, Fourth Edition, 2007.
T2 - S.Salivahanan, A. Vallavaraj, C. Gnanapriya, Digital Signal Processing,
TMH/McGraw Hill International, 2007

ASSIGNMENT- I
1. i) Calculate the DFT of the sequence x(n) = {1,1,-2,-2}
ii) Derive the equation for Decimation-in-time algorithm for FFT.
ASSIGNMENT- II
1. An IIR digital low-pass filter is required to meet the following specifications:
Passband ripple (or peak to peak ripple) 0.5 dB
Passband edge: 1.2 KHz
Stopband attenuation: 40dB
Stopband edge : 2.0 KHz
Sample rate: 8.0 KHz
Determine the required filter order for
i)
Digital Butterworth filter
ii)
Digital Chebyshev filter
ASSIGNMENT- III
1.

Explain the design procedures of FIR digital filters using Hamming & Hanning
window.
ASSIGNMENT- IV

1. i) What is Quantization? Explain the Quantization methods and Quantization

errors in detail.
ii) Derive an expression quantization noise power.
ASSIGNMENT- V
1. i) Write short notes on Sampling & Sampling Rate alteration.
ii) Explain the different interpolation techniques with suitable examples.

QUESTION BANK
Unit I - Discrete Fourier Transform
Part A
1.
2.
3.
4.

5.
6.
7.
8.
9.
10.
11.
12.
13.
14.
15.

16.
17.
18.
19.
20.
21.
22.
23.
24.
25.
26.

27.
28.
29.
30.

State the relation between DTFT and DFT.

What is Twiddle factor?
Prove the time reversal property of Discrete Fourier Transform.
Calculate the number of complex multiplications required for direct
computation of DFT and FFT for 64 and 128 point sequences.
State and prove the circular frequency shift property of DFT.
Relate z-transform and DFT.
Compute the DFT of x(n) = (n no).
State and prove the Parsevals relation of DFT.
What do you mean by the term bit reversal as applied to DFT?
Define discrete Fourier series.
Draw the basic butterfly diagram of DIF FFT algorithm.
Compute the DFT of x(n) = an
State the time shifting and frequency shifting properties of DFT.
Draw the basic butterfly diagram of DIT FFT algorithm.
Determine the 3 point circular convolution of x(n) = {1,2,3} and h(n) =
{0.5,0,1}
What is FFT and what are its advantages?
Distinguish between DFT and DTFT (Fourier transform)
What is the basic operation of DIT FFT algorithm?
What is zero padding? What are its uses?
State and prove Parsevals relation for DFT.
Draw the flow graph of radix 2 DIF - FFT algorithm for N= 4
What do you mean by bit reversal in DFT?
Write the periodicity and symmetry property of twiddle factor.
Give the relationship between z-domain and frequency domain.
Distinguish between discrete Fourier series and discrete Fourier transform.
What is the relationship between Fourier series co-efficient of a periodic
sequence and DFT?
Establish the relation between DFT and z-transform.
Define DFT pair.
Define Gibbs phenomenon.
How many multiplications and additions are required to compute N-point
DFT using radix 2 FFT?

Part B
1. In a LTI system, the input X(n) ={1, 2, 1, 2} and the impulse response
h(n) = {-1, 1, -1, 1}. Determine the impulse response of the LTI system by
Radix 2 DIF FFT algorithm.
2. The unit impulse response of an FIR digital filter is {1, 2, 1}. Illustrate the
Overlap save and Add method to determine its response to the input sequence
{1, 0, 2, 0, -1, 1, 0, 0, 1, 2, -1}
3. Determine the FFT of the following sequence using DIT and DIF Radix-2
algorithm. { 1, -1, 1, -1, 1, -1, 1, -1} Also draw the butterfly structure and
mention the intermediate values.
4. i) Perform Linear Convolution of the following sequences by Overlap and
Save method. X(n) = { -1, 1, 2, -1, 1, 2, -1, 1, -1} and h(n) = {2, 1, -2} (10)
ii) Find the N-Point DFT of {1, 2, 3, 4}
(6)
5. i) Perform circular convolution of the sequence using DFT and IDFT
technique. x1(n) = {2, 1,2,1} x2 (n) = {0,1,2,3}
(8)
ii) Compute the DFT of the sequence x(n) = {1,1,1,1,1,1,0,0}
(8)
6. From the first principles obtain the signal flow graph for computing 8 point
DFT using radix-2 DIT FFT algorithm. Using the above compute the DFT of
sequence x(n) = {0.5,0.5,0.5,0.5,0,0,0,0}
7. Determine the 8-point DFT of the sequence x(n) = {0,0,1,1,1,0,0,0}
8. Compute the output using 8 point DIT FFT algorithm for the sequence
x(n) = {1,2,3,4,5,6,7,8}
9. Derive DIF FFT algorithm. Draw its basic butterfly structure and compute
the DFT x(n) = (-1)n using radix 2 DIF FFT algorithm.
10. From the first principles obtain the signal flow graph for computing 8 point
DFT using radix-2 DIF-FFT algorithm. An 8 point sequence is given by
x(n)={2,2,2,2,1,1,1,1} compute its 8 point DFT of x(n) by radix-2 DIF-FFT

Unit II Infinite Impulse Response Digital Filters

Part A
1. Mention the need for impulse invariant method.
2. Draw the direct form structure for the following difference equation.
Y(n) = Y(n-1) + 5Y(n-2) + X(n) + 2X(n-1)
3. What is prewarping?
4. Write the advantages of bilinear transformation method.
5. Compare the impulse invariance and bilinear transformation methods.
6. Sketch the magnitude response of Type - I and II Chebyshev filter.
7. An analog filter has a transfer function H(s) = 1 / s+2. Using impulse
invariance method, obtain pole location for the corresponding digital filter
with T = 0.1s.
8. What is frequency warping in bilinear transformation?
9. Find the digital transfer function H (z) by using impulse invariance method for
the analog transfer function H(s) = 1 / s+2.
10. What are the different structures of realization of IIR filters?
11. What are the methods used to transform analog to digital filters?
12. Draw the direct form I structure of IIR filter.
13. Draw the direct form II structure of IIR filter.
14. Draw the cascade form realization structure of IIR filter.
15. Draw the parallel form realization structure of IIR filter.
16. When cascade form realization structure is preferred in filters?
17. Give the magnitude function Butterworth filter. What is the effect of varying
the order of N on magnitude and phase response?
18. List out the properties of Butterworth lowpass filter.
19. List out the properties of Chebyshev filter.
20. Give the Chebyshev filters transfer function and draw its magnitude response.
21. Give the equation for the order N and cut off frequency c of Butterworth
filter.
22. Why impulse invariance method is not preferred in the design of IIR filter
other than low pass filters?
23. What are the advantages and disadvantages IIR filters?
24. Compare Butterworth and Chebyshev filters.
25. What are the parameters that can be obtained from the Chebyshev filter
specification?
26. Give the bilinear transform equation between s-plane and z-plane.
27. What are the properties of bilinear transformation?
28. What is the main disadvantage of direct-form realization?
29. Realize y(n) + y(n+1) + y(n-2) = x(n) in cascade form network.
30. What is the advantage of cascade realization?
Part B
1. i) Explain in detail the design procedures of IIR digital filters using Impulse

Invariant method.
ii) State the need for realization of filters in different forms.

(12)
(4)

2. Design a digital Butterworth filter satisfying the following constraints with

T = 1 sec using Bilinear Transformation.
0.707 H(ej) 1
for 0 / 2
j
H(e ) 0.2
for 3 / 4
3. Design a Chebyshev IIR Digital filter to meet the following desired frequency
response H() by using Impulse Invariant techniques with sampling period
T = 1 sec.
0.9 H() 1
;0/4
0 H() 0.25
;/2
4. 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)

ii) Write short notes on frequency transformation in analog domain.

(10)
(6)

5. An IIR digital low-pass filter is required to meet the following specifications:

Passband ripple (or peak to peak ripple) 0.5 dB
Passband edge: 1.2 KHz
Stopband attenuation: 40dB
Stopband edge : 2.0 KHz
Sample rate: 8.0 KHz
Determine the required filter order for
iii)
Digital Butterworth filter
iv)
Digital Chebyshev filter
6. Convert the following analog transfer function H(s) = (s+0.2) / [(s+0.2) 2 + 4]
into equivalent digital transfer function H (z) by using impulse invariance
method assuming T= 1 sec.
7. Convert the following analog transfer function H(s) = 1 / (s+2) (s+4) into
equivalent digital transfer function H (z) by using bilinear transformation with
T = 0.5 sec.
8. Convert the following analog transfer function H(s) = 2/ (s+1) (s+3) into
equivalent digital transfer function H (z) by using bilinear transformation with
T = 0.1sec. Draw the direct form II realization of digital filter.
9. Explain impulse invariance method of digital filter design.
10. Derive an expression between s- domain and z- domain using bilinear
transformation. Explain frequency warping.
Unit III Finite Impulse Response Digital Filters

Part A
1.
2.
3.
4.
5.

What are linear phase characteristics in FIR filters?

Draw the shapes of any four window functions.
Why do we use window functions in FIR filter design?
Draw the block diagram of poly phase structure.
What is the effect of having abrupt discontinuity in frequency response of FIR
filters?
6. What are the characteristic features of FIR filters?
7. What are the different structures of realization of FIR filters?
8. Write the equation of Hamming window.
9. Write the equation of Hanning window.
10. Write the equation of Blackmann window.
11. Draw the basic FIR filter structure.
12. Distinguish between FIR and IIR filters.
13. Compare analog and digital filters.
14. Why FIR filters are always stable?
15. State the condition for a digital filter to be causal and stable.
16. What are the desirable characteristics of windows?
17. What are the advantages and disadvantages FIR filters?
18. What are the desirable and undesirable features of FIR filters?
19. What are the design techniques of designing FIR filters?
20. Draw the direct form realization of FIR system.
21. What do you understand by linear phase response?
22. For what kind of applications, the antisymmetrical impulse response can be
used?
23. For what kind of applications, the symmetrical impulse response can be used?
24. Draw the direct form realization of a linear phase FIR system for N even.
25. Draw the direct form realization of a linear phase FIR system for N odd.
26. What is Gibbs phenomenon?
27. What are the desirable characteristics of the window?
28. What is the principle of designing FIR filter using frequency sampling
method?
29. For what type of filters frequency sampling method is suitable?
30. Give the equation of the frequency sampling realization of FIR filter.
31. State the condition for linear phase in FIR filters for symmetric and anti
symmetric response.

Part B
1. i) Explain Symmetric and Antisymmetric methods of FIR filter design.

(8)

2.

3.

ii) Explain the design of Linear phase FIR filters by the frequency sampling
method.
(8)
Explain the design procedures of FIR digital filters using Hamming & Hanning
window.
Design a LPF for the following purpose. Use Hamming Window N=7. Fig.1
H[ej]
1

-/2

/2

Fig.1
4.

i) What are linear phase characteristics?

(4)
ii) Explain the design procedures of FIR filter using Blackman window. (12)

5.

With suitable examples, describe the realization of linear phase FIR filters.

6.

Design a low pass filter using Hamming window for N=7 for the desired
frequency Response
D () = ej3 for -3 / 4 3 / 4
=0
for 3 / 4

7.

Design a LPF with

Hd() = e-j3 for - 3/4 3/4
=0
for 3/4
Using a Hamming window with N= 7
Using frequency sampling method, design a bandpass filter with the following
specifications.
Sampling frequency, F = 8000 Hz
Cut off frequencies, fc1= 1000 Hz and fc2 = 3000 Hz.
Determine the filter coefficients for N = 7.

8.

9.

Design an ideal low pass filter with a frequency response

Hd(ej ) = 1 for
- /2 /2
= 0 for
/2
Find the values of h(n) for N=11 and H(z)

10.

Consider an FIR filter with system function

H(z) = 1 + 2.88 z - 1 + 3.4048 z - 2 + 1.74 z - 3 + 0.4 z - 4

Sketch the direct form I , II and determine in detail the corresponding input
output equations. Is the system minimum phase?
Unit IV Finite Word Length Effects

Part A
1. Compare fixed point and floating point number system.
2. What is the effect of quantization on pole locations?
3. Define limit cycle.
4. What are Scaling & Overflow?
5. What is round-off noise error?
6. Why rounding is preferred over truncation in realizing digital filter?
7. What are the different types of arithmetic in digital systems?
8. What do you understand by a fixed-point number?
9. What are the different types in fixed-point number representation?
10. What do you understand by sign-magnitude representation?
11. What is meant by block floating point representation? What are its
advantages?
12. What are advantages of floating point arithmetic?
13. What are the three quantization errors due to finite word length registers in
digital filters?
14. What is coefficient quantization error? What is its effect?
15. What is product quantization error in digital signal processing?
16. What do you understand by input quantization error?
17. What are the different quantization methods?
18. What is truncation? What is the error that arises due to truncation in floating
point numbers?
19. What is meant by rounding? Discuss its effect on all types of number
representations.
20. What is meant by A/D conversion noise?
21. Which realization is less sensitive to the process of quantization?
22. What is meant by quantization step size?
23. Draw the Quantizer characteristic with rounding.
24. Draw the Quantizer characteristic with truncation.
25. Draw the probability density function for rounding.
26. How do you relate the steady-state noise power due to quantization to the b
bits representing the binary sequence?
27. Draw the quantization noise model for a first order system.
28. Draw the quantization noise model for a second order system.
29. What are the methods used to prevent overflow?
30. Define dead band of the filter.

Part B

1. i) What is Quantization? Explain the Quantization methods and Quantization

errors in detail.
(10)
ii) Derive an expression quantization noise power.
(6)
2. i) Explain the limit cycle oscillations due to product round off.
(6)
ii) Study the limit cycle behavior of the system described by
(10)
W(n) = Q[ aW(n-1) + x(n) ] where W(n) is the output of the filter and Q[.]
is quantization. Assume that a = 7 / 8 , x(0) = 3 / 4 and x(n) = 0 for n > 0
3. Find the output roundoff noise power for the system having transfer function
H(z) = 1 / (1- 0.8z -1) ( 1- 0.4z -1) which is realized in cascade form.
Assume word length is 4 bits.
4. Write short notes on the following.
i) Quantization methods, Errors and its effects.
ii) Limit Cycle Oscillations and Overflow errors.

(8)
(8)

5. i) Explain the effect of input scaling on signal to quantization noise ratio of A/D
converter.
(8)
ii) Explain the statistical model for analysis of round off error multiplication. (8)
6. Write short notes on the following:
i)
Dynamic range scaling
ii)
Low sensitivity digital filters
iii)
Limit cycles in IIR filters
iv)
Finite Precision effects.

(4 x 4 = 16)

7. The output signal of an A/D converter is passed through a first order low pass
(1 a ) z
filter, with transfer function given by H(z) =
for 0 < a < 1. Find the
za
steady state output noise power due to quantization at the output of the digital
filter.
8. Find the steady state variance of the noise in the output due to quantization of
input for the first order filter. y(n) = ay(n-1) + x(n)
9. Consider the transfer function H(z) = H1(z) H2(z) where

H1(z) = ( 1 / (1-a1z-1))

&

H2(z) = ( 1 / (1-a2z-1))

Assume a1 = 0.5 and a2 = 0.6 and find the output round-off noise power.
10. Realize the first order transfer function H(z) = ( 1 / (1-az-1)) and draw its
quantization noise model. Find the steady state noise power due to product
roundoff.
Unit V Multirate Signal Processing

Part A
1. List the applications of multirate signal processing.
2. State the Nyquist sampling theorem.
3. Distinguish Upsampling & Downsampling.
4. What is aliasing? Mention the cause producing aliasing effect.
5. Define decimation and interpolation.
6. What is the need for multirate signal processing?
7. What is multirate signal processing?
8. Define downsampling.
9. What is meant by upsampling?
10. If the spectrum of a sequence x(n) is X(ej), the what is the spectrum of a
signal down sampled by a factor of 2?
11. If the z-transform of a sequence x(n) is X(z) then what is the z-tranform of a
sequence down sampled by a factor M?
12. If the z-transform of a sequence x(n) is X(z) then what is the z-tranform of a
sequence upsampled by a factor L?
13. What is the need for anti aliasing filter prior to downsampling?
14. What is the need for anti-aliasing filter after upsamling a signal?
15. What are the characteristics of a narrow band lowpass filter?
16. What are types of filter banks?
17. Draw the structure of an analysis filter bank.
18. Draw the structure of a synthesis filter bank.
19. Draw the structure of a two-channel quadrature mirror filter bank.
20. What are the sections of a two channel QMF bank?

Part B

1. i) Explain the design procedures of narrow band filters.

(8)
ii) What is aliasing? Explain the aliasing effect in digital filters with suitable
waveforms. Mention the methods of eliminating the aliasing effect. (8)
2. i) Write short notes on Sampling & Sampling Rate alteration.
(8)
ii) Explain the different interpolation techniques with suitable examples. (8)
3. i) Explain the principle of sampling rate conversion for multirate signal
processing.
(6)
ii) What is Nyquist sampling theorem? Explain Decimation by a factor of D
and Interpolation by a factor of L with suitable examples.
(10)
4. Explain in detail the poly phase structures for decimation and interpolation
filters.
5. Explain the design of narrow band filters in detail.
6. With necessary equations and diagrams, discuss about the interpolation and
decimation in multirate signal processing.
7. Design one-stage and two-stage interpolators to meet the following
specification:
I = 20
Input sampling rate : 10000 Hz
Pass band
: 0 F 90
Transition ban
: 90 F 100
Ripple
: 1 = 10-2 and 2 = 10-3