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JEPPIAAR ENGINEERING COLLEGE

DEPARTMENT OF C.S.E
QUESTION BANK
CS2403 DIGITAL SIGNAL PROCESSING

VII SEM/IV YEAR (2009-13 BATCH)


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.

UNIT II FREQUENCY TRANSFORMATIONS 9


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.

UNIT III IIR FILTER DESIGN 9


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.

UNIT IV FIR FILTER DESIGN 9


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.Find the Z transform of {1,0,2,0,3} May/ June 2007


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

2.Check whether the system y(n)=ex(n) is linear. May/ June 2007

The system is nonlinear.

3.What are the advantages of DSP? Nov/Dec 2009


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.

4.Define impulse signal. Nov/Dec 2009


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.

6. Find the period of x(n) = cos [8n/7 +2].


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

1.Find the convolution and correlation for x(n)={0,1,-2,3,-4} and


h(n)={0.5,1,2,1,0.5} April/May 2008
Ans. Refer page no 165 DSP by Nagoorkani

2. Determine the impulse response of the difference equation


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

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


1/(1-1/2Z-1)(1-1/4 Z-1)
Ans. Refer page no 461 DSP by Nagoorkani

4.check whether the following systems are linear time invarient.


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

5.Test the stability and causality of the following system


Nov/Dec2009

(i)y(n)=cosx(n)(8) )Refer page no 41 DSP by Nagoorkani


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

6. Find the one sided z-transform of discrete sequences generated by


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

1.Determine the DTFT of a sequence x(n) = an u(n). Nov/Dec 2006


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 )

2.What is FFT? Nov/Dec 2006


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

4. What are the advantages of FFT algorithm over direct computation


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.

5. State and prove Parsevals Theorem. Nov/Dec 2007


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

When y(n) = x(n), the above equation becomes


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

n=0 k=0

6. 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.

7. 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)]

8. Draw the basic butterfly diagram of radix -2 FFT. April/May 2008.

1 1
a A = a+ WNnk b
1
1
nk
WN
b B = a - WNnk b
-1

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

S.No DIT FFT Algorithm DIF FFT Algorithm


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:

By the definition of DFT;


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)

11. State Discrete Fourier Transform.


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

12. Distinguish between linear & circular convolution.

s.no Linear convolution circular convolution


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.

13. Define Zero padding? Why it is needed?


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.

14. State the shifting property of DFT.


Time shifting property states that
DFT {x(n-n0)} = X(K) e (j2n0k)/N

15. Why do we go for FFT?


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.

16. What do you mean by radix-2 FFT?


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.

17. Give any two application of 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.

19. What is Twiddle factor?


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

20. What is main advantage of FFT?


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 .

2. a) i) Derive the equation for Decimation in time algorithm for


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}.

4. a) i) From first principles obtain the signal flow graph for


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

Ans: i) Ref Pg.No 334-340, DSP by Salivahanan.


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

5. a) Two finite duration sequence are given by


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}.

UNIT III & IV


DIGITAL FILTERS(IIR & FIR ) DESIGN

PART-A

1. Obtain the block diagram representation of a FIR system?


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)
+ + + +

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


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.

4. Find the digital transfer function H(Z) by using impulse


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]

5. In the design of FIR digital filter, how is Kaiser Window


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.

7. What is the relationship between analog and digital frequency in


impulse invariant transformation? April/May 2008

Digital Frequency: = T
= analog frequency
T= Sampling interval

8. What is Prewarping? Why is it needed? Nov/Dec 2008


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.

9. What are the advantages of FIR filter?


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.

10.What is the necessary & sufficient condition of linear phase FIR


filter?
The condition for a linear phase filter is
1. = (N-1)/2
2. h(n) = h(N-1-n)

11.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.

12.Compare Rectangular & Hamming window.

S.No Rectangular 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
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

13.Compare Hamming window & Kaiser Window.


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 .

14.Compare FIR & IIR filter.

S.No FIR filter IIR filter


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

15.Define Frequency warping.


The non linear relation ship between analog and digital frequencies
introduced frequency distortion which is called as frequency
warping.

PART-B

1. a) Design a high pass filter hamming window by taking 9


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.

2. a) Design a digital Butterworth filter satisfying the constraints


0.707 | H()| 1.0 ; 0 /2
| H()| 0.2 ; 3/4 .
Nov/Dec 2006
Ans: Ref Pg.No 435-437, DSP by Salivahanan.

3. a) Design a digital Butterworth filter satisfying the constraints


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.

4. a) Describe the design of FIR filter using frequency sampling


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.

5. a)i) Design a digital BUTTERWORTH filter that satisfies the


following constraint using BILINEAR Transformation.
Assume T = 1 sec.
0.9 | H()| 1 ; 0 /2

| H()| 0.2 ; (3 /4)


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.

April /May 2008

Ans: a) i) Ref Pg.No 437-439, DSP by Salivahanan.


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.

FINITE WORD LENGTH EFFECTS

PART-A

1. Express the fraction 7/8 and 7/8 in sign magnitude, 2s


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

3. Identify the various factors which degrade the performance of


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

4. What is meant by limit cycle oscillation in digital filter?


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.

5. Express the fraction (-7/32) in signed magnitude and 2s


complement notations using 6 bits. Nov/Dec 2007 &Nov/Dec 2008
In Signed Magnitude: 1.001110
In 2s complement: 1.110010

6. Compare fixed & floating point number representation.

S.no Fixed point number Floating point number


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

7. What are the two types of quantization employed in digital


system?
1. Rounding
2. Truncation

8. Define Rounding & 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.

9. What is dead band?


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.For the given transfer function H(Z) = H1(Z) .H2(Z) ,where


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)

1. a)i) Explain the characteristics of a limit cycle oscillation w.r.t the


system described by the difference equation
y(n) = 0.95y(n-1)+x(n).Determine the dead band of the filter.

ii) Draw the product quantisation noise model of second order


IIR filter.
Nov/Dec 2006 & Nov/Dec 2008
Ans: a) i) Dead band = [-10,10]
ii) Ref Pg.No 513-514, DSP by Salivahanan.

2. a)i) Consider the truncation of negative fraction number


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

ii) The coefficients of a system defined by


1
H(Z) =
(1-0.4Z-1)(1-0.55Z-1)
are represented in anumber with a sign bit and 3 data bits.

4. Determine the new pole location for 1) Direct realization and 2)


Cascade realization of first order systems.Compare the
movements of the new pole away from the original ones in
both the cases.

iii) Consider the (b+1) bit bipolar A/D converter.Obtain an


expression for signal to quantization noise ratio .
May /June 2007& Nov/Dec 2007&April/May2008 & Nov/Dec 2008

Ans: a) i) Ref Pg.No 496-499, DSP by Salivahanan.


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

3.Give the applications of multirate digital signal processing.


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

IT 1252-DIGITAL SIGNAL PROCESING


(REGULATION 2004)

TIME:3 HOURS
MAX:100 MARKS

Answer all questions

PART A-(10*2=20 MARKS)

1.Find the Z transform of {1,0,2,0,3}


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

PART B_(5*16=80 marks)

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

1. What are the advantages of DSP?


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

11.a) Test the stability and causality of the following system


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

OR

b) Find the one sided z-transform of discrete sequences generated


by mathematically sampling of the following continuous time
function
(i)x(t)=sin wt (8)
(ii)x(t)=cos wt (8)

12.a)(i) calculate the percentage of savings in calculation in


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

13.a) Apply the bilinear transformation for the following:


(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

b)(i) Compare the impulse invariant and bilinear transformation.


(6)
(ii) explain the design procedure for lowpass digital butterworth
IIR filter(10)

14.a) Design a lowpass filter using rectangular window by taking 9


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

OR

b) Design a linear phase lowpass FIR filter with cutoff frequency of


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

15.a)(i) Compare the fixed point and floating point arithmetic,


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)
www.2freshesworld.com
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.

www.2freshesworld.com
PART B (5 16 = 80 Marks)

www.2freshesworld.com

11. (a) (i) Find the convolution x(n) * h(n) , where


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.
www.2freshesworld.com
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 .
www.2freshesworld.com
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

Answer ALL Questions


PART A (10 X 2 = 20 marks)

1. State Sampling Theorem?


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

1. 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 .
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

1. State and prove Parsevals Theorem. Nov/Dec 2007


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

When y(n) = x(n), the above equation becomes


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)]

4.Draw the basic butterfly diagram of radix -2 FFT. April/May 2008.

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

S.No DIT FFT Algorithm DIF FFT Algorithm


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.

6.Compare DFT AND FFT based on number of computations.

7.What is twiddle factors

7 compare radix 2 and radix 4 algorithms. .


8. Butterfly structure for radix 4 algorithms.

PART B

1. Derive the equation for Decimation in time algorithm for FFT.


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

1. What are the advantages and disadvantages of FIR filters?


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.

2..Distinguish between FIR filters and IIR filters.


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.

3. What are the design techniques of designing FIR filters?


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.

4.What is Gibbs phenomenon?


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.

5. What is the necessary and sufficient condition for linear phase


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.

3. What is Prewarping? Why is it needed? Nov/Dec 2008


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.

4.What are the advantages of FIR filter?


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.

5.What is the necessary & sufficient condition of linear phase FIR


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.