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Digital signal processing question bank

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DEPARTMENT OF C.S.E

QUESTION BANK

CS2403 DIGITAL SIGNAL PROCESSING

UNIT I SIGNALS AND SYSTEMS 9

Basic elements of digital signal Processing Concept of frequency in continuous time and

discrete time signals Sampling theorem Discrete time signals. Discrete time systems Analysis

of Linear time invariant systems Z transform Convolution ( Linear & circular) correlation.

Introduction to DFT DFT Properties of DFT Filtering methods based on DFT- FFT algorithms

Decimation in Time Decimation in Frequency algorithms Use of FFT algorithms in Linear

Filtering DCT.

Structure of IIR Analog filter design - Discrete time IIR filter from continuous time(analog) filter

IIR filter design by Impulse Invariance. Bilinear transformation Approximation of

derivatives(HPF,BPF,BRF) Filter design using frequency translation.

Structure for FIR systems-- Linear phase FIR filter Filter design using Windowing technique

Frequency sampling techniques Finite word length effects.

UNIT V APPLICATIONS 9

Multirate signal processing speech compression-Adaptive filter-Musical sound processing

Image enchancement

TOTAL : 45 periods

TEXT BOOK

1. John G Proakis and Dimtris G Manolakis, Digital Signal Processing Principles, Algorithms

and ApplicationS, PHI/Pearson Education, 2007,Fourth Edition.

2. Emmanuel C.I.D eachor&Feachor7Barrie.W.Jervis, Digital Signal Processing second Edition

,pearson Education /prentice Hall,2002

REFERENCES

1. Alan V Oppenheim, Ronald W Schafer and John R Buck, Discrete Time Signal

Processing, PHI/Pearson Education, 2000, 2nd Edition.2005.

2. Andreas Antoniou, Digital Signal Processing .Tata Mcgraw Hill,2001

UNIVERSITY QUESTION PAPERS

UNIT -1

SIGNALS AND SYSTEMS

PART A

1+0z-1+2z-2+3Z-4

1.More accuracy

2.It is easier to perform mathematical operation

3. Digital signals can be easily stored on magnetic disk without

any loss of information.

Also called as delta function

Represented by S(t)

S(t)=1 for t is equal to 0

=0 for t is not equal to 0

5.Find whether the signal y=n x(n) is linear. April/may 2008

The system is linear.

SOLUTION

= 8/7

2f = 8/7

f= 4/7 ; here K= 4 & N =7

It is periodic and the fundamental period is N =7 samples.

7. What is meant by causal & non causal system?

A system is said be causal if its output at anytime depends upon

present and past input only. A system is said be non causal if its

output at anytime depends upon present and future input only.

8..State the condition for the BIBO stable?

The condition for the BIBO stable is given by

h (t)dt <

9. Distinguish between linear and non linear system.

a1 y1(t) + a2 y2(t) = f[a1x1(t) + a2x2(t)]

If the above equation satisfies then the system is said to be Linear

system. If the above equation does not satisfies then the system is said

to be non Linear system.

10.What are energy and power signals?

The energy signal is one in which has finite energy and zero

average power. The power signal is one in which has finite average

power and infinite energy .

T

E = Lt x(t)2 dt joules .

T -T

P = Lt T

T 1 / 2T x(t)2 dt joules .

PART B

h(n)={0.5,1,2,1,0.5} April/May 2008

Ans. Refer page no 165 DSP by Nagoorkani

Y(n)+3y(n-1)+2y(n-2)=2x(n)-x(n-1) April /May 2008

Ans.Refer page no 18 DSP by Nagoorkani

3. Find the response of the system for the input signal (8)

May/June2007

X(n)={1,2,2,3} and h(n)={1,0,3,2)

Ans. Refer page no 164 DSP by Nagoorkani

1/(1-1/2Z-1)(1-1/4 Z-1)

Ans. Refer page no 461 DSP by Nagoorkani

May/June2007

(i) y(n)=A+Bx(n) Refer page no 31 DSP by Nagoorkani

(ii) Y(n)=ex(n) Refer page no 29 DSP by Nagoorkani

(iii)Y(n)=A.X(n)+B (x(n-1) Refer page no 30 DSP by Nagoorkani

Nov/Dec2009

(ii)y(n)=x(-n-2)(8) Refer page no 41 DSP by Nagoorkani

mathematically sampling of the following continuous time function

Nov/Dec2009

(i)x(t)=sinwt Refer page no 455 DSP by Nagoorkani (8)

(ii)x(t)=coswt Refer page no 456 DSP by Nagoorkani (8)

UNIT -1I

FAST FOURIER TRANSFORM (FFT)

PART-A

Solution:

x(n) = an u(n)

X(e ) = x(n) e -jn

j

n=-

X(e ) = an u(n) e -jn

j

n=-

X(e ) = an e -jn

j

n=0

X(e ) = (a e j)n

j

n=0

X(ej) = 1 / (1-a-ej )

The Fast Fourier Transform is a method or algorithm for computing

the DFT with reduced number of calculations. The computational

efficiency can be achieved if we adopt a divider and conquer

approach. This approach is based on decomposition of an N-point

DFT in to sucessively smaller DFTs. This approach leads to a family

of an efficient computational algorithm is known as FFT algorithm.

3. The first five DFT coefficients of a sequence x(n) are X(0) = 20,

X(1) = 5+j2,X(2) = 0,X(3) = 0.2+j0.4 , X(4) = 0 . Determine the

remaining DFT coefficients. May/June 2007

Solution:

X (K) = [20, 5+j2, 0, 0.2+j 0.4 , 0,X(5),X(6),X(7)]

X (5) = 0.2 j0.4

X (6) = 0

X (7) = 5-j2

of DFT? May/June 2007

1. Reduces the computation time required by DFT.

2. Complex multiplication required for direct computation is N2 and for

FFT calculation is N/2 log 2 N.

3. Speed calculation.

Parsevals theorem states that

If

x(n) X(K) and y(n) Y(K) ,

Then

N-1 N-1

x(n) y*(n) = 1/N X(K) Y*(K)

n=0 K =0

N-1 N-1

x(n) = 1/N X(K)2

2

n=0 k=0

Nov/Dec 2007

Re-ordering of input sequence is required in decimation in time.

When represented in binary notation sequence index appears as

reversed bit order of row number.

1. Commutative property: x(n)*h(n) = h(n) *x(n)

2. Associative Property: [x(n)*h1(n)] *h2(n) = x(n)*[h1(n)*h2(n)]

3. Distributive Property: x(n)*[ h1(n)+ h2(n)] = [x(n)* h1(n)]+ [x(n)* h1(n)]

1 1

a A = a+ WNnk b

1

1

nk

WN

b B = a - WNnk b

-1

1. The input is in bit reversed The input is in normal order;

order; the output will be the output will be bit reversed

normal order. order.

2. Each stage of computation the Each stage of computation the

phase factor are multiplied phase factor are multiplied

before add subtract operation. after add subtract operation.

10. If H(K) is the N-point DFT of a sequence h(n) , Prove that H(K) and

H(N-K) are comples conjugates. Nov/Dec 2008

This property states that, if h(n) is real , then H(N-K) = H*(K) = H(-

K)

Proof:

N-1

X(K) = x(n) e (j2nk)/N

n=0

Replace K by N-K

N-1

X(N-K) = x(n) e (j2n(N-K))/N

X(N-K) n= = X*(K)

The DFT is defined as N-1

X (K) = x(n) e (j2nk)/N ; K = 0 to N-1

n=0

The Inverse Discrete Fourier Transform (IDFT) is defined as

N-1

x (n) = X(K) e (j2nk)/N ; n = 0 to N-1

K=0

1 The length of the input sequence The length of the input

can be different. sequence should be same.

2 Zero Padding is not required. Zero padding is required if

the length of the sequence is

different.

Appending zeros to the sequence in order to increase the size or length

of the sequence is called zero padding.In circular convolution , when

the two input sequence are of different size , then they are converted

to equal size by zero padding.

Time shifting property states that

DFT {x(n-n0)} = X(K) e (j2n0k)/N

The FFT is needed to compute DFT with reduced number of

calculations.The DFT is required for spectrum analysis on the sinals

using digital computers.

The radix -2 FFT is an efficient algorithm for coputing N- point DFT

of an N-point sequence .In radix-2 FFT the n-point is decimated into

2-point sequence and the 2-point DFT for each decimated sequence is

computed. From the results of 2-point DFTs, the 4-point DFTs are

computed. From the results of 4 point DFTs ,the 8-point DFTs are

computed and so on until we get N - point DFT.

1. The DFT is used for spectral analysis of signals using a digital

computer.

2. The DFT is used to perform filtering operations on signals

using digital computer.

18. How many multiplications & addition are involved in radix-2 FFT?

For performing radix-2 FFT, the value of Nshould be such that, N=

2m. The total numbers of complex additions are Nlog 2 N and the total

number of complex multiplication are (N/2) log 2 N.

Twiddle factor is defined as WN = e j2/N. It is also called as weight

factor.

FFT reduces the computation time required to compute Discrete

Fourier Transform.

PART-B

1. a) i) Calculate the DFT of the sequence x(n) = {1,1,-2,-2}

ii) Determine the response of LTI system by radix -2 DIT FFT.

Nov/Dec 2006

Ans:i) X(K) = { 0, -1-j,6,-1+j}

ii) Ref Pg.No 320-328 , DSP by Salivahanan .

FFT.

ii) How do you linear filtering by FFT using Save add method?

Nov/Dec 2006 & April /May 2008 & Nov/Dec 2008

Ans:i) Ref Pg.No 320-328 , DSP by Salivahanan .

ii) Ref Pg.No 369, DSP by Salivahanan.

3. a)i) Prove the following properties of DFT when H(k) is the DFT

of an N-point sequence h(n).

1. H(k) is real and even when h(n) is real and even.

2. H(k) is imaginary and odd when h(n) is real and odd.

ii) Compute the DFT of x(n) = e-0.5n , 0 n 5.

May/June 2007

Ans: i) Ref Pg.No 309, DSP by Salivahanan.

ii) X(K) = { 2.414, 0.87-j0.659, 0.627-0.394j, 1.202,

0.62-j0.252, 0.627-j0.252}.

Computing 8-point using radix -2 DIF FFT algorithm.

ii) Using the above signal flow graph compute DFT of

x(n) = cos (n/4) ,0 n 7.

May/June 2007 & Nov/Dec 2007 & Nov/Dec 2008

ii) X(K) = {0, 3, 0, 2.7-j0.7, 0, 1, 0, 1.293-j0.7}

x(n) = sin (n/2) for n = 0,1,2,3

h(n) = 2 n for n = 0,1,2,3 Determine circular convolution using

DFT &IDFT method. Nov/Dec 2007

Ans: X(K) = {0, -2j, 0, 2j}

H(K) = {15, -3+6j, -5, -3-6j}

y(n) = {6, -3, -6, 3}

6. a) i) Discuss in detail the important properties of the DFT.

ii) Find the 4-point DFT of the sequence x(n) = cos (n/4)

iii) Compute an 8-point DFT using DIF FFT radix -2 algorithm.

x(n) = { 1,2,3,4,4,3,2,1}

April /May 2008

Ans: i)Ref Pg.No 308-311, DSP by Salivahanan.

ii) X(K) = {1, 1-j1.414, 1, 1+j1.414}

iii) X(K) = {20,-5.8-j2.4, 0, 0.17-j0.414, 0, -0.17+j0.414, 0,

-5.82+j2.414}.

DIGITAL FILTERS(IIR & FIR ) DESIGN

PART-A

Nov/Dec 2006

X(Z)

Z-1 Z-1 Z-1 Z-1

h(N-1)

h(1)

h(0) h(2) h(N-2)

Y(Z)

+ + + +

Nov/Dec 2006

Properties of Butterworth:

1. The butterworth filters are all pole design.

2. The filter order N completely specifies the filter

3. The magnitude is maximally flat at the origin.

4. The magnitude is monotomically decreasing function of ohm.

Properties of Chebyshev:

1. The magnitude reponse of the filter exhibits ripples in the pass

band or stop band

2. The pole of the filter lies on an ellipse.

3. Show that the filter with h(n) = [-1,0,1] is a linear phase filter.

May /June 2007 & Nov/Dec

2008

Solution:

h(n) = [ -1,0,1]

h(0) = -1 = -h(N-1-n) = -h(3-1-0) = -h(2)

h(1) = 0 = -h(N-1-n) = -h(3-1-1) = -h(1)

h(2) = 1 = -h(N-1-n) = -h(3-1-2) = -h(0)

It is a linear phase filter.

invariant method for the analog transfer function H(S) = 1/

(S+2).Assume T=0.5sec May /June

2007 &Nov/Dec 2007

Solution:

H(S) = 1/ (S+2).

H(Z) = 1/[1-e-1 Z-1]

H(Z) = 1/ [1-0.368Z-1]

different from other windows? Nov/Dec 2007

In all other windows a trade off exists between ripple ratio and main

lobe width. In Kaiser Window both ripple ratio and main lobe width

can be varied independently.

6. What are the merits and demerits of FIR filter? April/May 2008

Merits :

1. Linear phase filter.

2. Always Stable

Demerits:

1. The duration of the impulse response should be large

2. Non integral delay.

impulse invariant transformation? April/May 2008

Digital Frequency: = T

= analog frequency

T= Sampling interval

In IIR design using bilinear transformation the conversion of

specified digital frequencies to analog frequencies is called Pre-

warping. The Pre-Warping is necessary to eliminate the effect of

warping on amplitude response.

1. They can have an exact linear phase.

2. They are always stable

3. They can be realised efficiently in hardware

4. The design methods are generally stable.

filter?

The condition for a linear phase filter is

1. = (N-1)/2

2. h(n) = h(N-1-n)

In Fir filter design using Fourier analysis method for rectangular

window method, the infinite duration impulse response is truncated

to finite duration impulse response.The abrupt truncation of impulse

response introduce a oscillation in the pass band and stop band .This

effect is known as Gibbs phenomenon.

1. The width of the main The width of the main lobe

lobe in window spectrum in window spectrum is 8/N

is 4/N

2. The maximun side lobe The maximun side lobe

magnitude in window magnitude in window

spectrum is -13 dB spectrum is -41 dB

S.No Kaiser Window Hamming window.

1. The width of the main The width of the main lobe

lobe in window spectrum in window spectrum is 8/N

depends on the value of

and N.

2. The maximun side lobe The maximun side lobe

magnitude with respect to magnitude in window

peak of main lobe is spectrum is -41 dB

variable using the

parameter .

1. Only N samples of All the infinite samples of

impulse response are impulse response are

considered. considered.

2. Linear phase Linear phase characteristics

characteristics can be can not be achieved

achieved

The non linear relation ship between analog and digital frequencies

introduced frequency distortion which is called as frequency

warping.

PART-B

samples

of w(n) and with a cutoff frequency of 1.2 radians/sec.

Nov/Dec 2006

Ans: Ref: Pg.No: 298-301, DSP by Nagoorkani.

0.707 | H()| 1.0 ; 0 /2

| H()| 0.2 ; 3/4 .

Nov/Dec 2006

Ans: Ref Pg.No 435-437, DSP by Salivahanan.

0.8 | H()| 1.0 ; 0 /4

| H()| 0.2 ; /2 .

Apply Bilinear transformation method.

May/June2007 & Nov/Dec 2008

Ans: Ref: Pg.No: 359-362, DSP by Nagoorkani.

technique.

b) The desired frequency response of a low pass filter is given by

Hd() ={ e j2 ; -/4 /4

0 ; other wise.

Obtain the filter coefficient, h(n) using RECTANGUAR

window

define by W(n) = { 1; 0 n 4

0; otherwise.

Nov/Dec 2007

Ans: a) Ref Pg.No 389-391, DSP by Salivahanan.

b) Ref Pg.No 399, DSP by Salivahanan.

following constraint using BILINEAR Transformation.

Assume T = 1 sec.

0.9 | H()| 1 ; 0 /2

ii) Determine the magnitude response of the FIR filter (M=11)

and show that phase and group delay are conatant.

iii) The desired frequency response of a low pass filter is given

by

Hd() ={ e j3 ; -3/4 3/4

0 ; other wise.

j

Determine H(e ) for M= 7using HAMMING window.

iv) For the analog transfer function H(S) = 1/ (S+1)(S+2) .

Determine H(Z) using impulse invariant technique.

ii) Ref Pg.No 383-384, DSP by Salivahanan.

iii) Ref Pg.No 400-401, DSP by Salivahanan.

iv) Ref Pg.No 426, DSP by Salivahanan.

PART-A

complement and 1s complement. Nov/Dec 2006

Solution:

7/8 = 0.875 = (0.111)2 is sign magnitude

1s Complement = (0.111)2

2s Complement = (0.111)2

- 7/8 = -0.875

Sign magnitude: (1.111)2

1s Complement = (1.000)2

2s Complement = (1.001)2

2. a) What are the quantization error due to finite word length

register

in digital filter.

b) What are the different quantization methods? Nov/Dec 2006

Quantization Error :

1. Input quantization error

2. Coefficient quantization error

3. Product quantization error

Quantization methods

1. Truncation

2. Rounding

the digital filter implementation when finite word length is used.

May /June 2007 & April/May 2008 & Nov/Dec 2008

1. Input quantization error

2. Coefficient quantization error

3. Product quantization error

May /June 2007 & Nov/Dec 2007 &April/May 2008

In recursive system when the input is zero or same non-zero constant

value the non linearities due to finite precision arithmetic operation

may cause periodic oscillation in theoutput. Thus the oscillation is

called as Limit cycle.

complement notations using 6 bits. Nov/Dec 2007 &Nov/Dec 2008

In Signed Magnitude: 1.001110

In 2s complement: 1.110010

1. The position of the binary The position of the binary

Point is fixed. Point is variable.

2. The resolution is uniform The resolution is variable.

throughout the range

system?

1. Rounding

2. Truncation

Truncation is the process of discarding all bits less significant than

least significant bit that is retained.

Rounding of a b bit is accomplished by choosing the rounded result

as the b bit number closed to the original number unrounded.

In the limit cycle the amplitude of the output are confined to a range

of value which is called as dead band of the filter.

PART-B

1 1

H1(Z) = and H2(Z) =

-1

1 0.5 Z 1 0.6 Z-1

Find the output round off noise power.

Nov/Dec 2006

-2b

Ans: 2 /12(5.4315)

system described by the difference equation

y(n) = 0.95y(n-1)+x(n).Determine the dead band of the filter.

IIR filter.

Nov/Dec 2006 & Nov/Dec 2008

Ans: a) i) Dead band = [-10,10]

ii) Ref Pg.No 513-514, DSP by Salivahanan.

represented in(+1) bit fixed point binary form including sign bit

. Let (-b) bits be truncated .Obtain the range of truncation

errors for signed magnitude ,2s complement and 1s

complement representation of negative numbers.

Nov/Dec 2007

1

H(Z) =

(1-0.4Z-1)(1-0.55Z-1)

are represented in anumber with a sign bit and 3 data bits.

Cascade realization of first order systems.Compare the

movements of the new pole away from the original ones in

both the cases.

expression for signal to quantization noise ratio .

May /June 2007& Nov/Dec 2007&April/May2008 & Nov/Dec 2008

ii) Direct form: 1/ [1-0.875z-1+0.125Z-2]

Cascade form:1/[1-0.375Z-1][1-0.5Z-1]

iii) Ref Pg.No 499-503, DSP by Salivahanan.

UNIT V

APPLICATIONS

PART-A

1. Define multirate digital signal processing.

The process of converting a signal from a given rate to a

different rate is called sampling rate conversion. The system that employs

multiple sampling rates in the processing of digital signals are called

digital signal processing systems.

2. Give the advantages of multirate digital signal processing.

Computational requirements are less

Storage for filter coefficients is less

Finite arithmetic effects are less

Sensitivity to filter coefficients lengths are less

Communication systems

Speech and audio processing systems

Antenna systems

Radar systems

4. Define Decimation.

The process of reducing the sampling rate of the signal is called

decimation (sampling rate compression).

5. Define Interpolation

The process of increasing the sampling rate of the signal is

called interpolation (sampling rate Expansion).

PART-B

1. Explain briefly: Multi rate signal processing May/June 2007

Ref Pg.No 751, DSP by Proakis

2. Explain briefly: Vocoder May/June 2007

Ref Pg.No 754, DSP by Proakis

3. Explain decimation of sampling rate by an integer factor D and

derive spectra for decimated signal May/June 2006

Ref Pg.No 755, DSP by Proakis

4. Explain interpolation of sampling rate by an integer factor I and

derive spectra for decimated signal May/June 2006

Ref Pg.No 760, DSP by Proakis

5. Explain about adaptive filters

Ref Pg.No 880, DSP by Proakis

B.E B.TECH DEGREE EXAMINATION MAY/JUNE 2007

SEVENTH SEMESTER

(REGULATION 2004)

TIME:3 HOURS

MAX:100 MARKS

2.Check whether the system y(n)=ex(n) is linear.

3.Draw the radix 4 FFT DIF butterfly diagram.

4.Find the values of WN K when N=8 and K=2 also for K=3.

5.Draw the response curve for butterworth , chebyshev,and Eliptic filters.

6.Write the equation for frequency transformation from low pass to band

pass filters.

7.find digital filter equivalent for H(s)=1/S+8

8.Explain Gibbs phenomenon.

9State sampling theorem.

10.Explain briefly the musical sound processing

11.(a) (i) Find the response of the system for the input signal (8)

X(n)={1,2,2,3} and h(n)={1,0,3,2)

(ii) Find the inverse Z transform of (8)

1/(1-1/2Z-1)(1-1/4 Z-1)

Or

(b)check whether the following systems are linear time invarient.

(ii)y(n)=A+Bx(n)

Y(n)=ex(n)

Y(n)=A.X(n)+B (x(n-1)2

12.(a) Derive and draw radix 2 DIT algorithms for FFT of 8 points.

Or

(b) compute the DFT for the sequence {1,2,0,0,0,2,1,1} .Using radix 2

DIF FFT algorithms.

13.(a) (i)Design a digital filter using H(s)=1/s2+9S +18 with T=0.2 sec (8)

(ii) Design a second order band reject filter with W1 and W2 as cut off

frequency and sampling interval as T.(8)

Or

b(i) Realize the given transfer function using direct form 1 and parallel

methods.

H(Z)=4Z2+11Z-2/(Z+1)(Z-3) (8)

(ii)If H(S)=1/(S+1)(S+2) find H(z) using impulse invariance method

for sampling frequency of 5 samples/sec( 8).

14.(a)Design a linear phase FIR digital filter for the given specifications

using Hamming

window of length M=7

Hd(W)=e-j3w,for w<ii/6

0; for ii/6<w<ii

Or

(b) Design and implement linear phase FIR filter of length N=15 which

has following unit sample sequence

H(K)=1 K=0,1,2,3

=0 K=4,5,6,7

15.(a)Explain in detail about finite word length effect in the filter design.

Or

(b) Explain briefly

(i)Multi rate signal processing

(ii)Vocoder

B.E/B.Tech DEGREE EXAMINATION,NOV/DEC 2009

SEVENTH SEMESTER

COMPUTER SCIENCE AND ENGINEERING

IT-1252-DIGITAL SIGNAL PROCESSING

PART A

2. Define impulse signal

3. Calculate the DFT sequence x(n)={1,1,-2,-2}

4. List any four properties of DFT

5. Compare digital and analog filter

6. Sketch the mapping of s-plane and z-plane in bilinear

transformation

7. Write the steps involved in FIR filter design

8. Write the expression for Kaiser window function

9. What are the different formats of fixed point representation?

10. How overflow limit cycles can be eliminated?

PART B

(i) y(n)=cosx(n) (8)

(ii) y(n)=x(-n-2) (8)

OR

by mathematically sampling of the following continuous time

function

(i)x(t)=sin wt (8)

(ii)x(t)=cos wt (8)

computing a 512-point using radix-2 FFT when compared to

discrete DFT (8)

(ii) draw and explain the basic butterfly diagram of DIF radix-2

FFT (8)

OR

b) An 8-point sequence is given by x(n)={2,2,2,2,1,1,1,1} compute 8

point DFT of x(n) by

(i) Radix-2 DIT-FFT (8)

(ii) Radix-2 DIF-FFT (8)

also sketch the magnitude and phase spectrum

(i)Ha(s)=2/(s+1)(s+2) with T=1 sec find out H(z) (8)

(i)Ha(s)=2s/s2+0.2s+1 with T=1 sec find out H(z) (8)

OR

(6)

(ii) explain the design procedure for lowpass digital butterworth

IIR filter(10)

samples of w(n) and with cutoff frequency of 1.2 radians/sec (16)

OR

90 rad/sec using frequency sampling techniques.(Take N=17) (16)

representation and operations. (6)

(ii) What is meant by product quantization error? Draw and

explain the product quantization noise model of IIR system with

two first order section in cascade. (10)

OR

b)(i) What are zero I/P and overflow limit cycle? (6)

(ii) Explain the characteristics of the limit cycle in the filter

y(n)=0.95y(n-1)+x(n). determine the dead band of filter. (10)

B.E./B.Tech. DEGREE EXAMINATION, NOVEMBER/DECEMBER

2010

Fifth Semester

hesworld.com

CS 2403 DIGITAL SIGNAL PROCESSING

(Regulation 2008)

Time : Three hours Maximum : 100 Marks

Answer ALL questions

PART A (10 2 = 20 Marks)

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1. Calculate the minimum sampling frequency required for x(t) = 0.5 sin 50? t

+ 0.25 sin 25? t , so as to avoid aliasing.

2. State any two properties of Auto correlation function.

3. Write down DFT pair of equations.

4. Calculate % saving in computing through radix 2, DFT algorithm of DFT

coefficients. Assume N = 512.

5. What are the limitations of Impulse invariant method of designing digital filters?

6. Draw the ideal gain Vs frequency characteristics of :

(a) HPF and(b) BPF.

7. Compare FIR filters and FIR filters with regard to :

(a) Stability and(b) Complexity

8. Represent decimal number 0.69 in fixed point representation of lengthN = 6.

9. Prove that up sampling by a factor M is time varying system.

10. State a few applications of adaptive filter.

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PART B (5 16 = 80 Marks)

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x(n) a u(n) n =h(n) u(n) n = ?

(ii) Find the Z-transform of the following sequences :

x(n) = (0.5) u(n) + u(n ?1) n

x(n) = ? (n ? 5) .

Or

(b) (i) State and explain sampling theorems.

(ii) Find the Z-transform auto correlation function.

12. (a) (i) Explain, how linear convolution of two finite sequences are obtained via DFT.

(ii) Compute the DFT of the following sequences :

(1) x = [1,0,?1,0]

(2) x = [ j,0, j,1] when j = ?1 .

Or

(b) Draw the flow chart for N = 8 using tadix-2, DIF algorithm for finding DFT

coefficients.

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13. (a) Design digital low pass filter using Bilinear transformation, Given that ( 1)( 1.732

1)

1

()

+++

=

sss

Ha s .

Assume sampling frequency of 100 rad/sec.

Or

(b) Design FIR filter using impulse invariance technique. Given that

( 5 6)

1

()

++

=

ss

Ha s

and implement the resulting digital filter by adder, multipliers and

delays Assume sampling period T = 1 sec.

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14. (a) Design the first 15 coefficients of FIR filters of magnitude specification is

given below :

( ) = 1, jw H e /w/ < ? / 2

= 0, otherwise.

Or

(b) Draw THREE different FIR structures for the H(z) given below:

( ) (1 5 6 )(1 ) ?1 ?2 ?1 H z = + z + z + z .

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15. (a) (i) A signal x(n) = {6,1,5,7,2,1}

Find :

(1) x(n / 2)

(2) x(2n) .

(ii) Explain any one application using multirate processing of signals.

Or

(b) Write short notes on the following :

(i) Adaptive filter

B.E / B.Tech, DEGREE EXAMINATION , NOV / DEC 2006

Computer Science and EngineeringVII SEM

IT1252 DIGITAL SIGNAL PROCESSING

( Common to B.E.(PartTime)

(Regulation 2004)

Time: 3 hours Maximum : 100 marks

PART A (10 X 2 = 20 marks)

2. Find the Poles of the system.

3. Find the DFT of the sequence x(n) ={1, 1, 0, 0 } DFT is obtained by FFT.

4. Calculate the number of multiplications needed in the calculation of 512 point radix

2FFT when compared to Direct DFT?

5. What are the properties that are maintained same in the transfer of analog filter into

a digital filter?

6. What is warping effect?

7.Draw the direct from realization of FIR system?

8.What are the describe features of a window function? Name the different types of

windowing function?

9. What is truncation?

10. Draw a sample/ hold circuit and explain its operations?

PART B (5 X 16 = 80 marks)

11. (a) (i) For each of the following discrete time system, determine whether or not the

system is Linear Time, Variant, Causal and Stable?

11.(a) (ii) Determine the transfer function, magnitude & phase response, impulse

response for the system.

11. (b) (i) Find the Ztransform

of

1) x(n) = 2 n u (n2)

2) x(n) = n 2 u (n)

11.(b)(ii) Use convolution to find x(n), given

11.(b) (iii) Determine the cross correlation values of the sequence x1(n) ={ 1, 2, 3, 4 }

x2(n) = {4, 3, 2, 1}

12. (a) (i) Compute linear and circular convolution of the two sequence

x1(n) ={ 1, 2, 2, 2 } and x2(n) = {1, 2, 3, 4}

12.(a) (ii) Compute the FFT using DIT algorithm for the sequence x(n) = {1, 2, 3, 4, 4,

3,

2, 1 } and draw the corresponding flow diagram.

12.(b) (i)Prove that multiplication DFTs of 2 sequence is equivalent to the DFT of the

circular convolution of the 2 sequence in time domain?

12.(b)(ii) Discuss in detail the use of FFT algorithm , in linear filtering?

13.(a) Find H(z) using impulse invariant technique for the analog system function.

13 (b) (i) Obtain the direct form II, Cascade form parallel form structures for the

system?

13.(b) (ii) Design a butterworth filter using linear transformation that satisfies the

following constraint?

14(a) The desired response of a low pass filter is?

14.(b) Explain the Type I & Type 2 design of FIR filter using frequency sampling

Technique?

15.(a) The output of A/D converter is applied to a digital filter with

system function find the o/p noise power for the digital filter when the input

signal is quantized to 8Hz.

15.(b)(i) A digital system is characterized by the difference equation y(n)=0.95 y(n

1)+x(n). Determine the dead band the system when x(n) =0 and y(n) = 13

15.(b) (ii) With neat diagram explain the analysis and synthesis part of a recorder in

detail?

DSP IMPORTANT QUESTIONS

PART A

1.UNIT I

The energy signal is one in which has finite energy and zero

average power. The power signal is one in which has finite average

power and infinite energy .

T

E = Lt x(t)2 dt joules .

T -T

P = Lt T

T 1 / 2T x(t)2 dt joules .

2. .What is meant by causal & non causal system?

A system is said be causal if its output at anytime depends upon

present and past input only. A system is said be non causal if its

output at anytime depends upon present and future input only.

3..State the condition for the BIBO stable?

The condition for the BIBO stable is given by

h (t)dt <

4.State sampling theorem.

5.What is Nyquest rate.

Fs>2 fa

Fs=sampling frequency

Fa=analog frequency

PART B

1.Problems based on linear,nonlinear causal,stable

2.linear,circular convolution

3.Z transform properties

4.inverse Z transform long division method,partial fraction method

5. Sampling theorem

UNIT II

PART A

Parsevals theorem states that

If

x(n) X(K) and y(n) Y(K) ,

Then

N-1 N-1

x(n) y*(n) = 1/N X(K) Y*(K)

n=0 K =0

N-1 N-1

x(n) = 1/N X(K)2

2

n=0 k=0

2.. What do you mean by the term bit reversal as applied to FFT?

Nov/Dec 2007

Re-ordering of input sequence is required in decimation in time.

When represented in binary notation sequence index appears as

reversed bit order of row number.

3.Define the properties of convolution. April/May 2008.

1. Commutative property: x(n)*h(n) = h(n) *x(n)

2. Associative Property: [x(n)*h1(n)] *h2(n) = x(n)*[h1(n)*h2(n)]

3. Distributive Property: x(n)*[ h1(n)+ h2(n)] = [x(n)* h1(n)]+ [x(n)* h1(n)]

1 1

a A = a+ WNnk b

1

1

nk

WN

b B = a - WNnk b

-1

5.Distinguish between DIT and DIF FFT algorithm. Nov/Dec 2008

1. The input is in bit reversed The input is in normal order;

order; the output will be the output will be bit reversed

normal order. order.

2. Each stage of computation the Each stage of computation the

phase factor are multiplied phase factor are multiplied

before add subtract operation. after add subtract operation.

8. Butterfly structure for radix 4 algorithms.

PART B

2. . Derive the equation for Decimation in frequency algorithm for

FFT.

3. DIT FFT and DIF FFT butterfly (signal flow graph) diagrams.

4.DFT problems.

UNIT III

Advantages:

1. FIR filters have exact linear phase.

2. FIR filters are always stable.

3. FIR filters can be realized in both recursive and non recursive

structure.

4. Filters with any arbitrary magnitude response can be tackled using

FIR sequence.

Disadvantages:

1. For the same filter specifications the order of FIR filter design can

be as high as 5 to 10 times that in an IIR design.

2. Large storage requirement is requirement

3. Powerful computational facilities required for the implementation.

FIR filter IIR filter

1. These filters can be easily designed tohave perfectly linear phase.These

filters do not have linear phase.

2. FIR filters can be realized recursive and non-recursively.

3. Greater flexibility to control the shapeof their magnitude response.

4. Errors due to round off noise are less severe in FIR filters, mainly because

feedback is not used.IIR filters are easily realized recursively.Less

flexibility, usually limited to specific kind of filters.The round off noise in

IIR filters is more.

There are three well known methods for designing FIR filters with linear

phase .They are (1.)Window method (2.)Frequency sampling method

(3.)Optimal or minimax design.

One possible way of finding an FIR filter that approximates H(ejw) would

be to truncate the infinite Fourier series at n=(N-1/2).Direct truncation of

the series will lead to fixed percentage overshoots and undershoots before

and after an approximated discontinuity in the frequency response.

characteristic in FIR filter?

The necessary and sufficient condition for linear phase characteristic

in FIR filter is, the impulse response h(n) of the system should have the

symmetry property i.e.,H(n) = h(N-1-n)where N is the duration of the

sequence.

PART B

1.. Draw the structures of FIR filters

2. Butterworth Filter design using impulse invariant method,Bilinear

transformation method.

3.Chebhevshev filter design design using impulse invariant

method,Bilinear transformation method.

UNIT IV

PART A

1. 1.Give any two properties of Butterworth and Chebyshev filter.

Nov/Dec 2006

Properties of Butterworth:

5. The butterworth filters are all pole design.

6. The filter order N completely specifies the filter

7. The magnitude is maximally flat at the origin.

8. The magnitude is monotomically decreasing function of ohm.

Properties of Chebyshev:

3. The magnitude reponse of the filter exhibits ripples in the pass

band or stop band

4. The pole of the filter lies on an ellipse.

2. What are the merits and demerits of FIR filter? April/May 2008

Merits :

3. Linear phase filter.

4. Always Stable

Demerits:

3. The duration of the impulse response should be large

4. Non integral delay.

In IIR design using bilinear transformation the conversion of

specified digital frequencies to analog frequencies is called Pre-

warping. The Pre-Warping is necessary to eliminate the effect of

warping on amplitude response.

5. They can have an exact linear phase.

6. They are always stable

7. They can be realised efficiently in hardware

8. The design methods are generally stable.

filter?

The condition for a linear phase filter is

3. = (N-1)/2

4. h(n) = h(N-1-n)

6.What is Gibbs phenomenon?

In Fir filter design using Fourier analysis method for rectangular

window method, the infinite duration impulse response is truncated

to finite duration impulse response.The abrupt truncation of impulse

response introduce a oscillation in the pass band and stop band .This

effect is known as Gibbs phenomenon.

PART B

1.Design of filter using windowing techniques.

2.Design of Filter using frequency sampling techniques.

3.structure of filter.

4.Finite word length effects. Dead band problems,signal scaling

Unit 5

PART B

1.voice coder(vocoder)

2.applications of DSP

3.speech synthesis

4.adaptive filter.

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