Francis Xavier Engineering College

EC1361 / Digital Signal Processing

UNIT 1
1.Define signal and system.
A signal is defined as a physical quantity which has
useful information and is dependent of one or more
independent variable(s) like time, speed, distance,
velocity, torque, pressure etc.
A system is a collection of components designed to
perform a specific task. g! amplifier, rectifier, filter
etc.

".#hat are the different forms of representation of signal$
A signal can be represented as
%equence form
&unctional form
'raphical form
(abular form
). Define Discrete time signal.
A discrete time signal * (n) is a function of an independent
variable that is an integer. A discrete time signal is not defined
at instant between two successive samples.
+. Define discrete time system.
A discrete or an algorithm that performs some prescribed
operation on a discrete time signal is called discrete time
system.
,. #hat are the elementary discrete time signals$
-nit sample sequence (unit impulse)
-nit step signal
-nit ramp signal
*ponential signal
.. %tate the classification of discrete time signals.
(he types of discrete time signals are
1. nergy and power signals
". /eriodic and Aperiodic signals
). %ymmetric(even) and Antisymmetric (odd) signals
0. Define energy and power signal.
1f is finite i.e, 233 infinity, then *(n) is called energy signal.
1f / is finite in the e*pression, the signal is
called a power signal.
4. Define periodic and aperiodic signal.
A signal * (n) is periodic in period 5, if * (n65) 7* (n) for all n.
1f a signal does not satisfy this equation, the signal is called
aperiodic signal.
8. Define symmetric and antisymmetric signal.
A real value signal * (n) is called symmetric (even) if * (9n) 7*
(n). :n the otherhand the signal is called antisymmetric (odd)
if * (9n) 7* (n).
12. %tate the classification of systems.
%tatic and dynamic system.
(ime invariant and time variant system.
;ausal and anticausal system.
<inear and 5on9linear system.
%table and -nstable system.
11.Define dynamic and static system.
A discrete time system is called static or memory less if its
output at any instant n depends almost on the input sample at
the same time but not on past and future samples of the input.
e.g. y(n) 7a * (n)
1n any other case the system is said to be dynamic and to have
memory.
e.g. (n) 7* (n)6) *(n91)
1".Define time variant and time invariant system.
A system is called time invariant if its output , input
characteristics dos not change with time.
e.g.y(n)7*(n)6*(n91)
A system is called time variant if its input, output
characteristics changes with time.
e.g.y(n)7*(9n).
1).Define linear and non9linear system.
<inear system is one which satisfies superposition principle.
%uperposition principle!
(he response of a system to a weighted sum of signals be
equal to the
corresponding weighted sum of responses of system to each of
individual input signal.
i.e., ( =a1*1(n)6a"*"(n)>7a1(=*1(n)>6a" (=*"(n)>
e.g.y(n)7n*(n)
A system which does not satisfy superposition principle is
known as non9linear
system.
e.g.(n)7*"(n)
1+.Define causal and anticausal system.
(he system is said to be causal if the output of the system at
any time ?n@ depends only on present and past inputs but does
not depend on the future inputs.
e.g.!9 y (n) 7* (n)9* (n91)
A system is said to be non9causal if a system does not satisfy
the above definition.
1,.Define stable and unable system.
A system is said to be stable if we get bounded output for
bounded input.
1.#hat are the steps involved in calculating convolution sum$
(he steps involved in calculating sum are
&olding
%hifting
Aultiplication
%ummation
10. Define causal <(1 system.
(he <(1 system is said to be causal if
h(n)72 for n32
14. Define stable <(1 system.
(he <(1 system is said to be stable if its impulse response is
absolutely
summable. 1.e.
18.#hat are the properties of convolution sum
(he properties of convolution sum are
;ommutative property.
Associative law.
Distributive law.
"2.%tate associative law
(he associative law can be e*pressed as
=*(n)Bh1(n)>Bh"(n)7*(n)=h1(n)Bh"(n)>
#here *(n)9input h1(n)9impulse response.
"1.%tate commutative law
(he commutative law can be e*pressed as
*(n)Bh(n)7h(n)B*(n)
"". %tate distributive law
(he distributive law can be e*pressed as
*(n)B=h1(n)6h"(n)>7*(n)Bh1(n)6*(n)Bh"(n)
").#hat is meant by sampling$
%ampling implements the representation of a continuous9time
signal by a discrete9time signal, which enables the processing
of signal by digital computers.
"+.%tate sampling theorem.
An ideal continuous9to9discrete9time (;CD) converter using
periodic sampling is shown in &ig.. 1t can be
mathematicallydescribed as
*=n> 7 *c(n() 9 3 n 3
where *c(t) is the input continuous time signal and *=n> is the
output discrete time signal. ( is the sampling period.
fs 7 1
( is the sampling frequency in samplesCsec. s 7 "
( is the sampling frequency in radiansCsec.
C/D
D
c
(t) *=n>7* (n() c
(
&igure ! An ideal ;CD converter.
-51( 11
",.Define E9transform
E9 transform can be defined as
"..Define Fegion of convergence
(he region of convergence (F:;) of D(E) the set of all values of
E for which D(E) attain final value.
"0.%tate properties of F:;.
(he F:; does not contain any poles and is in the form of circle.
#hen *(n) is of finite duration then F:; is entire E9plane
e*cept E72 or E7 infinity
1f D(E) is causal,then F:; includes E7infinity.
1f D(E) is anticasual,then F:; includes E72.
"4.state the properties of E9transform.
i) <inearity!9
ii)(ime shifting
iii)%caling in E9domain
iv)(ime reversal
v)Differtiation in E domain
vi);onvolution of two sequences
vii);orrelation
"8.Define circular convolution.
<et *1(n) and *"(n) are finite duration sequences both of
length 5 with D&(s
D1(G) and D"(k)
1f D)(k)7D1(k)D"(k) then the sequence *)(n) can be obtained
by circular
convolution defined as
)2. Differentiate <15AF H ;1F;-<AF convolution.
<15AF convolution ;1F;-<AF convolution
1. (he sequence need not
be of equal length.
". (he sequences need
not be periodic.
). <ength of the output
sequence will be 5)7
5165"91.
1. Ioth the sequence
should be of equal
length.
". Atleast one sequence
should be periodic.
). <ength of the output
sequence will be
5175"75), where 51
is length of *1(n) and
5" is the length of
*"(n) and 5) is the
length of *)(n).
)1.Jow to obtain the output sequence of linear convolution
through circular convolution$
;onsider two finite duration sequences *(n) and h(n) of
duration < samples and A samples. (he linear convolution of
these two sequences produces an output sequence of duration
<6A91 samples, whereas, the circular convolution of *(n) and
h(n) give 5 samples where 57ma*(<,A).1n order to obtain the
number of samples in circular convolution equal to <6A91,
both *(n) and h(n) must be appended with appropriate number
of Kero valued samples. 1n other words by increasing the
length of the sequences * (n) and h(n) to <6A91 points and
then circularly convolving the resulting sequences we obtain
the same result as that of linear convolution.
)".#hat is Kero padding$ #hat are its uses$
<et the sequence * (n) has a length <. 1f we want to find the 59
point D&((5L<) of the sequence *(n), we have to add (59<)
Keros to the sequence *(n). (his is known as Kero padding.
(he uses of Kero padding are
1) #e can get better display of the frequency spectrum.
")#ith Kero padding the D&( can be used in linear filtering.
)). Determine the circular convolution of the sequence *1(n)
7M1,",),1N and *"(n)7 M+,),",1N. (A- 5:O "220)
1 1 ) " + 1,
" 1 1 ) ) 7 1.
) " 1 1 " "1
1 ) " 1 1 14
)+.%tate the convolution property of E transform. (A- 5:O
"220)
-51( 111
1.Define D&( and 1D&( (or) #hat are the analysis and synthesis
equations of D&($
D&((Analysis quation)
591 nk
1D&((%ynthesis quation)
".%tate the properties of D&(.
1) /eriodicity
") <inearity and symmetry
)) Aultiplication of two D&(s
+) ;ircular convolution
,) (ime reversal
.) ;ircular time shift and frequency shift
0) ;omple* conPugate
4);ircular correlation
).#hat are the two methods used for the sectional
convolution$
(he two methods used for the sectional convolution are
1)the overlap9add method and ")overlap9save method.
+.#hat is overlap9add method$
1n this method the siKe of the input data block *i(n) is <. (o
each data block we append A91 Keros and perform 5 point
cicular convolution of *i(n) and h(n). %ince each data block is
terminated with A91 Keros the last A91 points from each
output block must be overlapped and added to first A91 points
of the succeeding blocks. (his method is called overlap9add
method.
,.#hat is overlap9save method$
1n this method the data sequence is divided into 5 point
sections *i(n).ach
section contains the last A91 data points of the previous
section followed by < new data points to form a data sequence
of length 57<6A91.1n circular convolution of *i(n) with h(n) the
first A91 points will not agree with the linear convolution of
*i(n) and h(n) because of aliasing, the remaining points will
agree with linear convolution. Jence we discard the first (A91)
points of filtered section *i(n) 5 h(n). (his process is repeated
for all sections and the filtered sections are abutted together.
..#hy &&( is needed$
(he direct evaluation D&( requires 5" comple* multiplications
and 5" Q5
comple* additions.(hus for large values of 5 direct evaluation
of the D&( is difficult.Iy using &&( algorithm the number of
comple* computations can be reduced. %o we use &&(.
0.#hat is &&($
(he &ast &ourier (ransform is an algorithm used to compute
the D&(. 1t makes use of the symmetry and periodicity
properties of twiddle factor to effectively reduce the D&(
computation time.1t is based on the fundamental principle of
decomposing the computation of D&( of a sequence of length
5 into successively smaller D&(s.
4.Jow many multiplications and additions are required to
compute 5 point D&( using
redi*9" &&($
(he number of multiplications and additions required to
compute 5 point D&( using radi*9" &&( are 5 log" 5 and 5C"
log" 5 respectively,.
8.#hat is meant by radi*9" &&($
(he &&( algorithm is most efficient in calculating 5 point D&(.
1f the number of output points 5 can be e*pressed as a power
of " that is 57"A, where A is an integer, then this algorithm is
known as radi*9" algorithm.
12.#hat is D1( algorithm$
Decimation91n9(ime algorithm is used to calculate the D&( of a
5 point
sequence. (he idea is to break the 5 point sequence into two
sequences, the D&(s of which can be combined to give the D&t
of the original 5 point sequence.(his algorithm is called D1(
because the sequence *(n) is often splitted into smaller sub-
sequences.
11.#hat D1& algorithm$
1t is a popular form of the &&( algorithm. 1n this the output
sequence D(k) is
divided into smaller and smaller sub9sequences , that is why
the name Decimation 1n &requency.
1".Draw the basic butterfly diagram of D1( algorithm.
(he basic butterfly diagram for D1( algorithm is

a A 7 a6b#
G

5


b I 7 a9b#
G

5

#
G
5

1). Draw the basic butterfly diagram of D1& algorithm.
(he basic butterfly diagram for D1& algorithm is
a A 7 (a6b)


b I 7 (a9b)#
G

5

#
G
5

1+.Jow efficient is the &&($
(he D&( takes 5R" operations for 5 points. %ince at
any stage the computation required to combine
smaller D&(s into larger D&(s is proportional to 5,
and there are log"(5) stages (for radi* "), the total
computation is proportional to 5 B log"(5).
(herefore, the ratio between a D&( computation and
an &&( computation for the same 5 is proportional to
5 C log"(n). 1n cases where 5 is small this ratio is not
very significant, but when 5 becomes large, this ratio
gets very large. (very time you double 5, the
numerator doubles, but the denominator only
increases by 1.)
1,. Are &&(Ss limited to siKes that are powers of "$
5o. (he most common and familiar &&(Ss are Tradi*
"T. Jowever, other radices are sometimes used,
which are usually small numbers less than 12. &or
e*ample, radi*9+ is especially attractive because the
Ttwiddle factorsT are all 1, 91, P, or 9P, which can be
applied without any multiplications at all.
Also, Tmi*ed radi*T &&(Ss also can be done on
TcompositeT siKes. 1n this case, you break a non9
prime siKe down into its prime factors, and do an &&(
whose stages use those factors. &or e*ample, an &&(
of siKe 1222 might be done in si* stages using
radices of " and ,, since 1222 7 " B " B " B , B , B ,. 1t
might also be done in three stages using radi* 12,
since 1222 7 12 B 12 B 12.
1.. ;an &&(Ss be done on prime siKes$
Ues, although these are less efficient than single9
radi* or mi*ed9radi* &&(Ss. 1t is almost always
possible to avoid using prime siKes.
10. #hat is an &&( Tradi*T$
(he Tradi*T is the siKe of an &&( decomposition. 1n
the e*ample above, the radi* was ". &or single9radi*
&&(Ss, the transform siKe must be a power of the
radi*. 1n the e*ample above, the siKe was )", which
is " to the ,th power.
14. #hat are Ttwiddle factorsT$
T(widdle factorsT are the coefficients used to
combine results from a previous stage to form inputs
to the ne*t stage.
18. #hat is an Tin placeT &&($
An Tin placeT &&( is simply an &&( that is calculated
entirely inside its original sample memory. 1n other
words, calculating an Tin placeT &&( does not require
additional buffer memory (as some &&(Ss do.)
"2. #hat is Tbit reversalT$
TIit reversalT is Pust what it sounds like! reversing
the bits in a binary word from left to write. (herefore
the A%ISs become <%ISs and the <%ISs become
A%ISs. Iut what does that have to do with &&(Ss$
#ell, the data ordering required by radi*9" &&(Ss
turns out to be in Tbit reversedT order, so bit9
reversed inde*es are used to combine &&( stages. 1t
is possible (but slow) to calculate these bit9reversed
indices in softwareV however, bit reversals are trivial
when implemented in hardware. (herefore, almost all
D%/ processors include a hardware bit9reversal
inde*ing capability (which is one of the things that
distinguishes them from other microprocessors.)
"1 #hat is Tdecimation in timeT versus Tdecimation in
frequencyT$
&&(Ss can be decomposed using D&(Ss of even and
odd points, which is called a Decimation91n9(ime
(D1() &&(, or they can be decomposed using a first9
halfCsecond9half approach, which is called a
TDecimation91n9&requencyT (D1&) &&(. 'enerally, the
user does not need to worry which type is being
used.
-51( 1O
1. What are the different types of filters based on impulse response
Based on impulse response the filters are of two types
1 II! filter
" FI! filter
The II! filters are of recursive type# where$y the present output sample
depends on the present input# past input samples and output samples
The FI! filters are of non recursive type# where$y the present output
sample depends on the present input sample and previous input samples
!. What are the different types of filters based on fre"uency response
Based on fre%uency response the filters can $e classified as
1 &owpass filter
" 'ighpass filter
( Bandpass filter
) Bandre*ect filter
3. What are the ad#antages and disad#antages of $%& filters
'd#antages(
1 FI! filters have e+act linear phase
" FI! filters are always sta$le
( FI! filters can $e reali,ed in $oth recursive and non recursive
structure
) Filters with any ar$itrary magnitude response can $e tac-led using
FI! se%uence
Disad#antages(
1 For the same filter specifications the order of FI! filter design can
$e as high as . to 1/ times that in an II! design
" &arge storage re%uirement is re%uirement
( 0owerful computational facilities re%uired for the implementation
). State the structure of %%& filter
II! filters are of recursive type where$y the present o1p sample depends on present
i1p# past i1p samples and o1p samples The design of II! filter is reali,a$le and sta$le
The impulse response h2n3 for a reali,a$le filter is
h2n34/ for n5 /
*. State the ad#antage of direct form +structure o#er direct form +structure.
In direct form 6structure# the num$er of memory locations re%uired is less than
that of direct form structure
6. ,o- one can design digital filters from analog filters
7 8ap the desired digital filter specifications into those for an e%uivalent analog
filter
7 9erive the analog transfer function for the analog prototype
7 Transform the transfer function of the analog prototype into an e%uivalent digital
filter transfer function
.. /ention the procedures for digiti0ing the transfer function of an analog filter.
The two important procedures for digiti,ing the transfer function of an analog
filter are
7 Impulse invariance method
7 Bilinear transformation method
1. What do you understand by bac2-ard difference
:ne of the simplest method for converting an analog filter into a digital filter is to
appro+imate the differential e%uation $y an e%uivalent difference e%uation
d1dt y2t34y2nT3;y2nT;T31T
The a$ove e%uation is called $ac-ward difference e%uation
3. What is the mapping procedure bet-een S4plane 5 64plane in the method of mapping
differentials What are its characteristics
The mapping procedure $etween <;plane = >;plane in the method of mapping of
differentials is given $y
'2>3 4'2<3?<421;>;131T
The a$ove mapping has the following characteristics
7 The left half of <;plane maps inside a circle of radius 1centered at >4 / in the >plane
7 The right half of <;plane maps into the region outside the circle of radius 1 in the
>;plane
7 The * @;a+is maps onto the perimeter of the circle of radius 1 in the >;plane
17. What is meant by impulse in#ariant method of designing %%& filter
In this method of digiti,ing an analog filter# the impulse response of resulting
digital filter is a sampled version of the impulse response of the analog filter
The transfer function of analog filter in partial fraction form#
11. 8i#e the bilinear transform e"uation bet-een S4plane 5 64plane.
<4"1T21;>;111A>;13
1!. What is bilinear transformation
The $ilinear transformation is a mapping that transforms the left half of <;plane
into the unit circle in the >;plane only once# thus avoiding aliasing of fre%uency
components
The mapping from the <;plane to the >;plane is in $ilinear transformation is
<4"1T21;>;111A>;13
13. What are the properties of bilinear transformation
7 The mapping for the $ilinear transformation is a one;to;one mapping that is for
every point ># there is e+actly one corresponding point <# and vice;versa
7 The * @;a+is maps on to the unit circle ?,?41#the left half of the s;plane maps to
the interior of the unit circle ?,?41 and the half of the s;plane maps on to the
e+terior of the unit circle ?,?41
1). Write a short note on pre4-arping.
The effect of the non;linear compression at high fre%uencies can $e compensated Bhen
the desired magnitude response is piece;wise constant over fre%uency# this compression
can $e compensated $y introducing a suita$le pre;scaling# or pre;warping the critical
fre%uencies $y using the formula
1*. What are the ad#antages 5 disad#antages of bilinear transformation
'd#antages(
7 The $ilinear transformation provides one;to;one mapping
7 <ta$le continuous systems can $e mapped into reali,a$le# sta$le digital systems
7 There is no aliasing
Disad#antage(
7 The mapping is highly non;linear producing fre%uency# compression at high
fre%uencies
7 Neither the impulse response nor the phase response of the analog filter is
preserved in a digital filter o$tained $y $ilinear transformation
16. What is the ad#antage of cascade reali0ation
Cuanti,ation errors can $e minimi,ed if we reali,e an &TI system in cascade form
1.. Define signal flo- graph.
D signal flow graph is a graphical representation of the relationships $etween the
varia$les of a set of linear difference e%uations
11. What is transposition theorem 5 transposed structure
The transpose of a structure is defined $y the following operations
7 !everse the directions of all $ranches in the signal flow graph
7 Interchange the input and outputs
7 !everse the roles of all nodes in the flow graph
7 <umming points $ecome $ranching points
7 Branching points $ecome summing points
Dccording to transposition theorem if we reverse the directions of all $ranch
transmittance and interchange the input and output in the flowgraph# the system function
remains unchanged
13.,o- -ill you con#ert a direct form % to a direct form %% structure
By reversing the output and input sections and cascading we get direct form II
!7. 9y impulse in#ariant method obtain the digital filter transfer function and differential
e"uation of the analog filter ,:s;< 1/=s>1?. :'@ ABC !77.;
Using impulse invariant transformation s;p
i
1;e
p
i
T
,
;1

Therefore#
'2s34 1

s;2;13
'2,34 1
1;e
2;13T
,
;1

&et T41 sec
The differential e%uation is o$tained $y ta-ing inverse > transform
h2n34 e
2;13n
u2n3 y2n34 e
2;13n
u2n3+2n3
"1.#hat are the advantages and disadvantages of &1F filters$
Advantages!
1. FIR fters have exact near phase.
2. FIR fters are aways stabe.
3. FIR fters can be reazed n both recursve and non recursve
structure.
4. Fters wth any arbtrary magntude response can be tacked usng
FIR sequence.
Disadvantages!
1. For the same fter specfcatons the order of FIR fter desgn can
be as hgh as 5 to 10 tmes that n an IIR desgn.
2. Large storage requrement s requrement
3. Powerfu computatona factes requred for the mpementaton.
"". #hat are the design techniques of designing &1F filters$
There are three we known methods for desgnng FIR fters wth near
phase .They are (1.)Wndow method (2.)Frequency sampng method
(3.)Optma
or mnmax desgn.
"). #hat is 'ibb@s phenomenon$ (A- 5ov "220)
One possbe way of fndng an FIR fter that approxmates H(e|w) woud be
to
truncate the nfnte Fourer seres at n=(N-1/2).Drect truncaton of the
seres
w ead to fxed percentage overshoots and undershoots before and after an
approxmated dscontnuty n the frequency response.
"+. <ist the steps involved in the design of &1F filters using
windows.
1.For the desred frequency response Hd(w), fnd the mpuse response
hd(n) usng Equaton
2.Mutpy the nfnte mpuse response wth a chosen wndow sequence
w(n) of ength N to obtan fter coeffcents h(n),.e.,
3.Fnd the transfer functon of the reazabe fter
",. #hat are the desirable characteristics of the window function$
The desrabe characterstcs of the wndow are
1.The centra obe of the frequency response of the wndow shoud
contan most of the energy and shoud be narrow.
2.The hghest sde obe eve of the frequency response shoud be sma.
3.The sde obes of the frequency response shoud decrease n energy
rapdy as m tends to n.
".. 'ive the equations specifying the following windows.
a. Rectanguar wndow
b. Hammng wndow
c. Hannng wndow
d. Bartett wndow
e. Kaser wndow
a. Rectanguar wndow:
The equaton for Rectanguar wndow s gven by

b. Hammng wndow:
The equaton for Hammng wndow s gven by
c. Hannng wndow:
The equaton for Hannng wndow s gven by
d. Bartett wndow:
The equaton for Bartett wndow s gven by
e. Kaser wndow:
The equaton for Kaser wndow s gven by
"0. #hat is the necessary and sufficient condition for linear phase
characteristic in &1F
filter$
The necessary and suffcent condton for near phase characterstc n
FIR fter s, the mpuse response h(n) of the system shoud have the
symmetry
property .e.,
where N s the duraton of the sequence.
"4. #hat are the advantages of Gaiser window$
o It provdes fexbty for the desgner to seect the sde obe eve
and N
o It has the attractve property that the sde obe eve can be vared
contnuousy from the ow vaue n the Backman wndow to the
hgh vaue n the rectanguar wndow
"8. #hat is the principle of designing &1F filter using frequency
sampling method$
In frequency sampng method the desred magntude response s
samped and a near phase response s specfed .The sampes of desred
frequency response are dentfed as DFT coeffcents. The fter coeffcents
are
then determned as the IDFT of ths set of sampes.
)2. &or what type of filters frequency sampling method is suitable$
Frequency sampng method s attractve for narrow band frequency
seectve fters where ony a few of the sampes of the frequency response
are
non zero.
)1. Draw the direct form realiKation of &1F system.
X(Z)
Y(Z)
)". Draw the direct form realiKation of a linear /hase &1F system for
5 even.
X(Z)
Y(Z)
)). Draw the direct form realiKation of a linear /hase &1F system for
5 odd
X(Z)
Y(Z)
)+. #hen are the cascade form realiKation is preferred in &1F filters$
The cascade form reazaton s preferred when compex zeros wth absoute
magntude s ess than one.
),. Distinguish between &1F filters and 11F filters. (A- 5:O "220)
&1F filters 11F filters
Can be easy desgned to
have a perfect near phase
1.These fters do not have
near phase.
Can be reazed recursvey
and non recursvey.
2.Can be reazed
recursvey ony.
Great fexbty to contro
the shape of ther
magntude response.
3.Less fexbe, usuay
mted to some specfc
fters.
Errors due to round off
nose s ess severe, many
because feedback s not
used.
4.Round off nose s severe.
-51( O
1.#hat are the different types of arithmetic in digital systems.$
There are three types of arthmetc used n dgta systems. They are fxed
pont
arthmetc, foatng pont ,bock foatng pont arthmetc.
".#hat is meant by fi*ed point number$.
In fxed pont number the poston of a bnary pont s fxed. The bt to the
rght
represent the fractona part and those to the eft s nteger part.
).#hat are the different types of fi*ed point arithmetic$
Dependng on the negatve numbers are represented there are three forms
of fxed pont
arthmetc. They are sgn magntude,1s compement,2s compement
+. #hat is meant by sign magnitude representation$
For sgn magntude representaton the eadng bnary dgt s used to
represent the sgn.
If t s equa to 1 the number s negatve, otherwse t s postve.
,. #hat is meant by 1@s complement form$
In 1,s compement form the postve number s represented as n the sgn
magntude
form. To obtan the negatve of the postve number ,compement a the bts
of the
postve number.
.. #hat is meant by "@s complement form$
In 2s compement form the postve number s represented as n the sgn
magntude
form. To obtan the negatve of the postve number ,compement a the bts
of the
postve number and add 1 to the LSB.
0. #hat is meant by floating pint representation$
In foatng pont form the postve number s represented as F =2CM,where s
mantssa,
s a fracton such that1/2<M<1and C the exponent can be ether postve or
negatve.
4. #hat are the advantages of floating pint representation$
1.Large dynamc range 2.overfow s unkey.
8.#hat are the quantiKation errors due to finite word length
registers in digital filters$
1.Input quantzaton errors2.Coeffcent quantzaton errors3.Product
quantzaton errors
12.#hat is input quantiKation error$.
The fter coeffcents are computed to nfnte precson n theory. But n
dgta computaton the
fter coeffcents are represented n bnary and are stored n regsters. If a b
bt regster s used the fter coeffcents must be rounded or truncated to b
bts ,whch produces an error.
11 .#hat is product quantiKation error$.
The product quantzaton errors arse at the out put of the mutper.
Mutpcaton of a b bt
data wth a b bt coeffcent resuts a product havng 2b bts. Snce a b bt
regster s used the
mutper output w be rounded or truncated to b bts whch produces the
error.
1". #hat is input quantiKation error$.
The nput quantzaton errors arse due to A/D converson.
1).#hat are the different quantiKation methods$
Truncaton and Roundng
1+.#hat is truncation$
Truncaton s a process of dscardng a bts ess sgnfcant than LSB that s
retaned
1,. #hat is Founding$
Roundng a number to b bts s accompshed by choosng a rounded resut
as the b bt number
cosest number beng unrounded.
1.. #hat are the two types of limit cycle behavior of D%/$.
1.Zero mt cyce behavor 2.Over fow mt cyce behavor
10.#hat are the methods to prevent overflow$
1. Saturaton arthmetc and2.Scang
14.%tate some applications of D%/$
Speech processng, Image processng, Radar sgna processng.
16 /ar2s
@A%D %
1 What are the different types of signals 8i#e eEample.
!. What are the different types of systems 8i#e eEample.
3. EEplain in details the different types of 'nalog to digital Con#erter
). State and pro#e sampling theorem.
*. What is Fuanti0ation error and deri#e an eEpression for the same
6. Determine -hether the gi#en systems are lineer or not.
:i;
dt
t dy 3 2
> *y :t; >! < E:t;
:ii; *
dt
t dy 3 2
>y :t; < *E :t;
.. Determine the energy of the follo-ing se"uence
E=n? <
n

"
1
for n G7
7 for nH7
1. Determine -hich of the se"uence is periodic or aperiodic
:i; E
1
=n? <sin

E
F n
:ii; E
!
=n?< sin

G
n
3. Chec2 -hether the gi#en system is time #ariance or time in#ariance
:i; y=n? < )n E=n?
:ii; y=n? < n =E =n??
!
17. $ind -hether the gi#en signal is periodic or a periodic E:t; < e
I-t>
J/)
@A%D %%
1. $ind the output y=n? of a casual discrete K time LD% system -hich is characteri0ed
by the difference e"uation
M=n?43/) y=n41? >
G
1
y=n4!? <!E=n?
!. $ind the DD$D of the follo-ing se"uence
N=n? < sin

"
n
u=n?
3. EEplain the properties of DD$D -ith proof.
). EEplain the properties of 04transform -ith proof.
*. $ind the fre"uency response ,:e
I-
; and impulse response h:n; of a casual discrete
time LD% system -hich is characteri0ed by the gi#en difference e"uation.
M=n?43/)y=n41?>
G
1
y=n4!? < !E=n?
6. Determine the 04transform of the se"uence
N=n?<cos-
o
n u=n?
.. Determine E=n? for the gi#en se"uence h=n? <O1P!P3Q
y=n?<O1P1P!P41P3Q using long di#ision method.
1. Determine the in#erse 04transform of the follo-ing E:0; by the partial fraction
eEpansion method.
N:0; <
" E
"
"
+
+
z z
z
3. @se &esidue method to find the in#erse 04transform E:n; for E:0;<
3 " 32 1 2 z z
z
17. @sing the &esidue method find the in#erse 04transform of
E:0;<
3 . / 32 ". / 2
1
z z
&BC( 1!1G7.*
11. $ind the transfer function and %mpulse response of a discrete time LD% system
described by linear constant4coefficient difference e"uation gi#en as under
y=n?<
"
1
y=n41?>E=n?
(
1
E=n41?
1!. Sol#e the difference e"uation by using 04transform method
E:n>!;>3E:n>1;>!E:n;<7
@A%D %%%
1. Bbtain D$D of the follo-ing se"uence
E =n? < = R P R P RP RP 7P 7P 7P 7?
using decimation in fre"uency $$D algorithms.
!. Dra- the flo- diagram of D%D $$D for A < 16
3. Let E :n; be the finite duration se"uence of length 1 such that
E :n; < O41P 7P !P 7P 4)P 7P !P 7 Q $ind E :i
e
; using D%D $$D flo- graph.
). @sing radiE 4! D%D $$D find E :t; from the follo-ing se"uence
E =n? < O R P RP RP RP 7P 7P 7P 7QQ
*. Deri#e the &adiE 4! D%D 'lgorithm.
6. Deri#e the &adiE 4! D%$ 'lgorithm.
.. EEplain the different properties of D$D -ith proof.
1. Determine the DD$ of E=n? is
E =n? < 1 for ! n 6
7 for n < 7P 1P .P 1P 3
3. Compute the %n#ase D$D of N :2; < O1P !P 3P )Q
17. $ind the %D$D of N :2; < O3P :!>I;P :!4I; Q
11. Determine the Con#olution of t-o finite duration se"uence gi#en belo-.
E =n? < 1 for 41 n > 1
7 other-ise
1!. Determine the circular Con#olution of the follo-ing
E =n? < O 1P 7.*P 1P 7.*P 1P 7.*P 1P 7.* Q
h =n? < O 7P 1P !P 3 Q
@A%D %C
1. Design a linear phase $%& band pass filter to pass the fre"uencies in the range 1 to !
rad/sec using ,anning -indo- -ith A<*
!. Describe the impulse %n#ariance and 9ilinear transformation methods used for
designing %%& filters.
3. EEplain in detail about %%& filter design by the bilinear transformation.
). Design a filter -ith
,
d
:e
I-
; < e
4I3-
4S/) T U T S/)
7 S/) H VUVT S
using ,anning -indo- -ith A<..
*.:i; Pro#e that an $%& filter has linear phase if the unit sample response satisfies the
condition
h:n;< Wh:A414n;P n<7P1P!PXXXXPA41. 'lso discuss symmetric and
antisymmetric cases of $%& filter.
:17;
:ii; EEplain the need for the use of -indo- se"uences in the design of $%& filter.
Describe the -indo- se"uences generally used and compare their properties.
:6;
6.EEplain in detail about the steps in#ol#ed in the design of 9utter-orth 5
Chebysche# filters.
.. :i;Deri#e bilinear transformation for an analog filter -ith system function ,:s;<
b/:s>a; :1;
:ii; Design a single pole lo- pass digital filter -ith 43d9 band -idth of 7.!S by use
of bilinear transformation.
:1;
1. :i;Bbtain the direct form %P direct form %%P cascade and parallel reali0ations for
the follo-ing system
M:n;<47.1y:n41;>7.!y:n4!;>3E:n;>3.6E:n41;>7.6E:n4!;
:11;
:ii;Discuss the limitation of designing an %%& filter using impulse in#ariant
method.
@A%D C
1. Describe in detail the architectural aspects of D/S 3!7C*) digital signal processor
using an illustrati#e bloc2 diagram.
!. ' second order %%& filter is described by its transfer function
1
,:6; < 444444444444444444444444444
:147.* 0
41
;:147.)*0
41
;
Determine the effect of coefficient "uanti0ation due to truncation -ith 3 bit -ord
length registers in :i; direct form and :ii; cascade form.
3. EEplain in detail about Fuanti0ation in floating point reali0ation of %%& digital
filters.
). Deri#e the Steady state input 5 output noise po-er.
*. Write short notes on
:i; Fuanti0ation by truncation :6;
:ii; Fuanti0ation by rounding :6;
:iii; Sample 5 ,old operation :);