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VALLIAMMAIENGINEERINGCOLLEGE
SRM Nagar,Kattankulathur 603 203.

DEPARTMENTOFINFORMATION TECHNOLOGY
2015-2016 ODD SEMESTER
IT 6502 DIGITAL SIGNAL PROCESSING
QUESTION BANK

Handled By,
Mr.K.Sanjay, A.P(O.G)& Mr.M.Selvaraj, A.P(O.G)
UNIT 1
PART A
1. Calculate the minimum sampling frequency required for x(t) = 0.5 sin 50t+ 0.25 sin 25t, so as to
avoid aliasing.
2. State any two properties of Auto correlation function.
3. State sampling theorem.
4. Distinguish between power and energy signal with an example.
5. Define and express the transfer function of Nth order LTI system.
6. Compare linear convolution and circular convolution.
7. State low pass sampling theorem.
8. What is correlation?What are its types?
9. Find the energy and power of x(n) = Aejnu(n).
10. Determine which of the following sequences is periodic, and compute their fundamental period. (a)
Aej7n
(b) sin(3n)
11. Is the system y (n) = ln [x (n)] is linear and time invariant?
12. Determine Z transform of x(n) = 5nu(n)
13. Find the signal energy of (1/2)nu(n)
14. Determine whether the following sinusoids is periodic, if periodic then compute their fundamental
period.
(a) cos 0.01n
(b) sin(62n/10)
x (n)
15. Check whether the system y(n) = e is linear.
16. Determine Z transform of x(n) = anu(n)
17. Define the auto correlation and cross correlation.
18. List any four properties of Z-Transform.
19. What are the properties of convolution?
20. List the properties of discrete time sinusoidal signals?

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PART B

1. (i) Find the convolutionx(n) * h(n) , wherex(n) = an u(n) h(n) = n u(n)


(ii) Find the Z-transform of the following sequences :
n
x(n) = (0.5) u(n) + u(n 1)
x(n) = (n 5) .
2. (i) State and explain sampling theorems.
(ii) Find the Z-transform auto correlation function.
3. (i) Suppose a LTI system with input x(n ) and output y(n ) is characterized by its unit sample
response h(n ) = (0.8)nu(n ). Find the response y(n ) of such a system to the input signalx(n )
= u(n ).
(ii) A causal system is represented by the following differenceEquation

Compute the system function H (z )and find the unit sample response of the system in
analytical form.
4. (i) Compute the normalized autocorrelation of the signal x(n ) = an u(n ),0 < a <1
(ii) Determine the impulse response for the cascade of two LTI system having impulse
responses
h1(n)=(0.5)nu(n)andh2(n)=(0.2)nu(n)
,

5. (i) Find the inverse Z-Transform of

using
(1) Residue method and
(2) Convolution method.
(ii) State and prove circular convolution.
6. Determine the causual signal x(n)for the following Z-transform

7.

(i)

X(z) = (z2+z) / ((z-0.5) (z-0.25))

(ii)

X(z) = (1+z-1) / (1-z-1+0.5z-2)

(i) Find inverse Z transfer of X(Z) =

ifROC : |Z| > 1, (2) ROC : |Z| < 0.5, (3) ROC :

0.5 < |Z| <1


(iii)
Derive expressions to relate Z transfer and DFT

8.

(i)Determine the transfer function, and impulse response of the system y(n)

y(n 1) + y(n

2) = x(n) + x(n 1).


(ii) Find the convolution sum of

1
2
0

2, 0, 1
1

andh(n) = (n) (n 1) + (n 2) (n 3).


9. (i) Find the Z transform of
x(n) = 2nu(n 2)
x(n) = n2u(n)

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(ii) State and explainscaling, linearity and time delay properties of Z transform

10. (i)Derive the equation for convolution sum and summarize the steps involved in computing
convolution.
(ii) State, prove and explain the sampling theorem
11. Determinethecasualsignalx(n)forthefollowingZtransform

12. (i) Explain the different types of digital signal representation with examples
(ii) What is Nyquist rate? Explain its significance while sampling analog signals
(ii) List the properties of ROC.

13. Checkwhetherthefollowingsystemsarelinearnonlinear,timevariantorinvariant,causalnoncausal,
stableandunstable
1.y(n)=cos[x(n)]
2.y(n)=x(n+2)
3.y(n)=x(2n)
4.y(n)=x(n)cos(n)

14. Find the convolution of the signal x(n) = {1,2,-3,4} and h(n) = {-5,-6,7,8,9} using tabulation method
Compute the normalized autocorrelation of the signal.

UNIT 2
PART A
1. Write down DFT pair of equations.
2. Calculate % saving in computing through radix 2, DFT algorithm of DFTcoefficients. Assume N =
512.
3. State and prove Parseval's theorem.
4. Compute the DFT of the four point sequence x(n ) = {0,1,2,3}.
5. What is the relation between DFT and Z-Transform?
6. What is phase factor or twiddle factor?
7. List the uses of FFT in linear filtering?
8. Find the DTFT of x(n)=-bn.u(-n-1).
9. Compute the IDFT of Y(k)={1,0,1,0}.
10. How DFT is differ from DTFT.
11. Find DFT of sequence x(n) = {1, 1, -2, -2}
12. What are the computational saving (both complex multiplication and complex addition) in using N
point FFT algorithm.
13. What do you mean by in place computation?
14. Differentiate between DIT and DIF FFT algorithm.

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15. What is meant by radix 4 FFT?
16. List any four properties of DFT.
17. Compute DFT of x(n) = {1, -1, 1, -1}
18. Find the value of WNK when N = 8 and K = 2 and also k = 3
19. How many multiplication and additions are required to compute N point DFT using
Radix 2 FFT?
20. Give transform pair equation of DCT?

PART B
1. (i) Explain, how linear convolution of two finite sequences are obtained via DFT.
(ii) Compute the DFT of the following sequences :
x = [1,0,1,0]

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

2. Draw the flow chart for N = 8 using tadix-2, DIF algorithm for finding DFT coefficients.
3. By means of the DFT and IDFT, determine the response at the FIR filter with the impulse response
h(n ) = [1,2,3] and the input sequencex(n ) = [1,2,2,1].
4. Compute the DFT of the following sequence x(n ) using the decimation in time FFT algorithmx(n )
= [1,1,-1,-1,1,1,1,-1].

5. (i) Evaluate the 8-point for the following sequences using DIT-FFT algorithm

(ii) Calculate the percentage of saving in calculations in a 1024-point radix -2 FFT, when compared
to direct DFT.
6. Determine the response of LTI system when the input sequence x (n ) = {1, 1, 2, 1, 1 } by radix 2
DIT FFT. The impulse response of the system is h(n) = {1, 1, 1, 1}.

7. (i)Find 8-point DFT for the following sequence using direct method
{1,1,1,1,1,1,0,0}
(ii)state any six properties of DFT.
8. Compute 8 point DFT of the following sequenceusing radix 2 DIT FFT algorithm
x(n) = {1,-1,-1,-1,1,1,1,-1}
9. (i)Discuss the properties of DFT.
(ii)Discuss the use of FFT algorithm in linear filtering and correlation
10. Find DFT for {1,1,2,0,1,2,0,1} using FFT DIT butterfly algorithm and plot the spectrum.
11. Compute the eight point DFT of the given sequence x(n) = { , , , , 0, 0, 0, 0} using radix 2
DIT - DFT algorithm.
12. (i)Find 8-point DFT for the following sequence using direct method {1,1,1,1,1,1,0,0}
(ii)StateanysixpropertiesofDFT.
13. a) Draw the flow chart for N = 8 using tadix-2, DIF algorithm for finding DFT coefficients.

b) Determine the following system for static, linear, time variance, causal. i) y(n) = x(n+2);ii) y(n) =
x(n2); ) y(n) = x(-n);
14. Determine the response of LTI system when the input sequence x (n ) = {1, 1, 2, 1, 1 } by radix 2
DIT FFT. The impulse response of the system is h(n) = {1, 1, 1, 1}.

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UNIT 3
PART A
1. What are the limitations of Impulse invariant method of designing digitalfilters?
2. Draw the ideal gain Vs frequency characteristics of :HPF and BPF.
3. What is meant by warping?
4. What are the limitations of impulse invariance method?
5. Sketch the various tolerance limits to approximate an ideal lowpassandhighpass filter.
6. What is the importance of poles in filter design?
7. Compare bilinear and impulse invariant transformation.
8. What is aliasing?
9. Define Bilinear transformation with expressions.
10. Mention the properties of Butterworth filter.
11. What are the characteristics of Chebyshev filter?
12. Write the transformation equation to convert low pass filter into band stop filter.
13. Define Phase Delay and Group Delay.
14. Why IIR filters do not have linear phase?
15. Use the backward difference for the derivative and convert the analog filter to digital filter given
H(s)=1/(s2 +16)
16. State the relationship between the analog and digital frequencies when converting an analog filter
using bilinear transformation.
17. Explain the advantage and drawback of Bilinear transformation.
18. Compare the Butterworth and Chebyshev Type-1 filters.
19. What is the main drawback of impulse invariant mapping?
20. Compare the digital and analog filter.

PART B
1. Design digital low pass filter using Bilinear transformation, Given that

Assume sampling frequency of 100 rad/sec.


2. Design FIR filter using impulse invariance technique. Given that

and implement the resulting digital filter by adder, multipliers and delays Assume sampling period T
= 1 sec.
3. (i) Find the H (z ) corresponding to the impulse invariance design using a sample rate of 1/T
samples/sec for an analog filter H (s) specified as follows :

(ii) Design a digital low pass filter using the bilinear transform to satisfy the following
characteristics
(1) Monotonic stop band and pass band
(2) 3 dBcutoff frequency of 0.5 rad
(3) magnitude down at least 15 dB at 0.75rad.

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4. Design
D
an IIR
R filter using impulse in
nvariance te
echnique fo
or the given

A
Assume
T = 1 sec. Realize this filter using dire
ect form I an
nd direct fo
orm II.

5. The
T specificaation of the desired
d
lowpass filter is

Design a Buttterworth diggital filter usiing bilinear transformatiion.


D
6. The
T specificaation of the desired
d
low pass
p filter is

Design a Cheebyshev digittal filter usinng impulse innvariant trannsformation..


D
7. Design
D
an IIR
R digital low
w pass butterw
worth filter to
t meet the following
f
requirements: Pass band
riipple (peak to
t peak): 0.5dB,
0
Pass band
b
edge: 1.2kHz,
1
Stopp band attenuuation: 40ddB, Stop bannd
eddge: 2.0 kHzz, Sampling rate: 8.0 kH
Hz. Use bilineear transform
mation technnique.
8. (ii) Discuss th
he limitation of designingg an IIR filteer using impuulse invariannt method
(iii)convert thee analog filter with systeem transfer function
f
usinng bilinear trransformatioon
Ha(S)=(S+0.33) / ((S+0.3)2+16)
9. The
T specificaation of the desired
d
low pass
p filter is
0.8 H() 1.0; 0 0.2
H() 0.2; 0.322
D
Design
butterrworth digitaal filter usingg impulse innvariant transsformation.

10. Determine
D
thee system funnction H(z) of
o the chebyshevs low paass digital fiilter with thee specifications
=1dB ripp
ple in the passs band 0
=1dB ripple in the stopp band 0
U
Using
bilineaar transformaation (assum
me T=1sec)
11. Obtain
O
the dirrect form I, direct
d
form ii
i ,cascade, parallel
p
form
m realization for the systeem
y(n)= -0.1y(n
n-1)+0.2y(n-2)+3x(n)+3..6x(n-1)+0.66x(n-2)
A
Bilineaar Transform
mation to H(ss) =2/(S+2) (S+3) with T=0.1
T
sec.
12. Apply

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UNIT 4
FINITE IMPULSE RESPONSE DIGITAL FILTERS
PART A
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20

Compare FIR filters and FIR filters with regard to stability and complexity
List out the conditions for the FIR filter to be linear phase.
What is Gibbs phenomenon or Gibbs oscillation?
Write the equations for rectangular window and hamming window.
Write the equations for blackmanwindow.andhanning window.
Distinguish between FIR and IIR filters.
Compare the digital and analog filter.
What are the desirable properties of windowing technique?
Write the equation of Bartlett window.
Draw the Direct form I structure of the FIR filter.
Write the steps involved in FIR filter design.
Draw the direct form implementation of the FIR system having difference equation
y(n) = x(n) 2x(n-1) + 3x(n-2) 10x(n-6)
Obtain direct cascade realization of the system H(Z) = (1+5Z-1+6Z-2)(1+Z-1)
What are advantages and disadvantages of FIR filter?
What is the reason that FIR filter is always stable?
What is the necessary and sufficient condition for linear phase characteristic in FIR filter?
State the properties of FIR filter?
What are called symmetric and antisymmetric FIR filters?
State the condition for a digital filter to be causal and stable?
Write the procedure for designing FIR filter using windows.
21
Write the procedure for designing FIR filter using frequency sampling method.
PART - B

1. Design the first 15 coefficients of FIR filters of magnitude specification is


given below :

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

H(Z) = (1+5Z-1+6Z-2)(1+Z-1)
3. Design and obtain the coefficients of a 15 tap linear phase FIR low pass filter using Hamming
window to meet the given frequency response.

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4. Determine
D
thee coefficientts of a linearr phase FIR filter
f
of lenggth M = 15 which
w
has a symmetric
s
uunit

saample respon
nse and a freequency respponse that saatisfies the conditions

5. Design
D
an FIR
R filter usingg hanning window
w
with the followinng specification

6.

( Design a single
(i)
s
tier nootch filter too reject frequuencies in thee range 1 to 2 rad/sec ussing rectanguular
window with N =7 .
w
(iii) Compare Hamming window
w
and Kaiser
K
winddow.

7. Prove that an FIR filter has linear phaase it the uniit sample ressponse satisffies the conddition
h = h(N-1-n).also disccuss symmettric and anti symmetric cases
h(n)
c
of FIR
R filter when N is even.
8. (ii) Explain brriefly how thhe zeros in FIR filter is loocated.
( Using a rectangular
(ii)
r
w
window
techhnique, desiggn a low passs filter with pass band gaain of unity cut
off frequency
y of 1000Hz and workingg at a sampliing frequenccy of 5 kHz. The length of
o the impullse
reesponse shou
uld be 7.
9. D
Design an FIR
R low pass digital
d
filter using
u
the freequency sam
mpling methood for the folllowing
sppecificationss
Cut off frequency = 1500Hz
Sampling
g frequency = 15000Hz
Order of the
t filter N = 10
Filter Len
ngth requiredd L = N+1 = 11
10. Design
D
a FIR
R low pass fillter having thhe followingg specificatioons using Haanning winddow

Assume N = 7
11. (ii) Realize thee following FIR system using minim
mum numberr of multiplieers
-1
-2
-3
-4
(ii)H(Z) = 1 + 2Z + 0.5Z - 0.5Z - 0.5Z
0
(4)
-1
-2
-3
-4
-5
-6
(iii) H(Z) = 1 + 2Z + 3Z + 4Z + 3Z
Z + 2Z + Z
(44)
(iiii) Design a digital FIR band pass fiilter with low
wer cut off frequency
fr
20000Hz and uppper cut off
frrequency 320
00 Hz usingg Hamming window
w
of leength N = 7. Sampling rate is 100000Hz.
(8))
12. (ii) Consider an
a FIR lattice filter with coefficientss k1 = 1/2 ;k2 = 1/3 ; k3 = 1/4. Determ
mine the FIR
R
fiilter coefficieents for the direct
d
form structure
s
(8))
(iii) Using a reectangular window
w
technnique, designn a low passs filter with pass
p band gaain of unity ccut
off frequency
y of 1000Hz and workingg at a sampliing frequenccy of 5 kHz. The length of
o the impullse
reesponse shou
uld be 7. (8)

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UNIT V

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FINITE WORD LENGTH EFFECTS


PART A
1. What is truncation?
2. What is product quantization error?
3. What is meant by fixed point arithmetic? Give example
4. Explain the meaning of limit cycle oscillator
5. What is overflow oscillations?
6. What are the advantages of floating point arithmetic?
7. Compare truncation with rounding errors.
8. What is dead band of a filter?
9. What do you understand by input quantization error?
10. State the methods used to prevent overflow?
11. Compare fixed point and floating point arithmetic?
12. What are the two types of quantization employed in a digital system?
13. What is rounding and what is the range of rounding?
14. What is quantization step size?
15. Define Noise transfer function?
16. What are limit cycles?
17. What is meant by block floating point representation? What are its advantages?
18. What are the three-quantization errors to finite word length registers in digital filters?
19. What is coefficient quantization error? What is its effect?
20. Why rounding is preferred to truncation in realizing digital filter?
21. State the need for scaling in filter implementation.
22. What is product round off noise?
PART B
1.

Discuss in detail the errors resulting from rounding and truncation?

2.

Explain the limit cycle oscillations due to product round off and overflow errors? (Nov2010)

3.
Explain the characteristics of limit cycle oscillations with respect to the system described by the
difference equation y(n)=0.95y(n-1)+x(n). x(n)=0; y(n-1) =13. Determine the dead band of the system

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4.
T output off A/D converter is applieed to digital filter with thhe system fuunction
The
Find thee output noisse power froom the digitaal filter whenn the input siignal is quanntized to havve 8 bits.
c
q
quantization
in FIR filterrs?
5.
(ii)Explain thee effects of co-efficient
(ii)Distiinguish betw
ween fixed pooint and floaating point arrithmetic
6.

W respect to finite worrd length efffects in digitaal filters, with examples discuss aboout
With

(i)

O
Over
flow lim
mit cycle osccillation

(ii)

Signal scaling
g

7.

C
Consider
a seecond order IIR
I filter witth

Find thee effect on quantization


q
on pole locaations of the given system
m function inn direct form
m and in casccade
form. Assume
A
b = 3 bits.
8.

W is called
What
d quantizatioon noise? Deerive the exppression for quantizationn noise poweer.

9.

H to preveent limit cyclle oscillationns? Explain.


How

10.

(ii) Compare the


t truncatioon and roundding errors ussing fixed pooint and floaating point reepresentationn.

(ii) Reppresent the fo


ollowing num
mbers in floaating point format
fo
with five
f bits for mantissa
m
andd three bits ffor
exponennt.
(a) 710
(b) 0.2510
(c) -7710
(d) -00.2510
11. (ii) Explain th
he characteriistics of lim
mit cycle osciillation withh respect to the
t system described
d
byy the
differennce equation : y(n) = 0.955 y(n-1) + x (n) ; x(n)= 0
andd y(n-1)= 13..
(ii) Expplain the effeects of coeffiicient quantization in FIR
R filters.
12.
E
Explain
the characteristtics of a lim
mit cycle osciillation withh respect to the
t system described
d
byy the
equationn

D
Determine
thee dead band of the filter x(n) = (3/4))(n).

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