Digital Signal Processing
Spring 2024
1Zulaikha Kiran, 2024
Week 13
Material taken from:
Discrete time Signal Processing by Oppenheim and Schafer –3
rd
Edition
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Filter Design
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•The design of filters involves the following stages:
•the specification of the desired properties of the system
•Dependent on application
•the approximation of the specifications using a causal discrete timesystem
•the realization of the system.
•Dependent on available technologies
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Discrete Time IIR Filters from
Continuous Time Filters
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Why use continuous time filter as starting
point?
•The art of continuous-time filter design is highly advanced
•Many designs are simple
•Approximations that work for continuous time, do not directly work
for discrete time design, since frequency response is periodic
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Impulse Invariance
•Intheconceptofimpulseinvariance,adiscrete-timesystemis
definedbysamplingtheimpulseresponseofacontinuous-time
system
•providesadirectmeansofcomputingsamplesoftheoutputofabandlimited
continuous-timesystemforbandlimitedinputsignals
•iftheoverallobjectiveistosimulateacontinuous-timesystemina
discrete-timesetting,wemightdesignthediscrete-timesystemsuch
thatitsimpulseresponsecorrespondstosamplesofthecontinuous-
time
•ORitmightbedesirabletomaintain,inadiscretetimesetting,certain
time-domaincharacteristicsofwell-developedcontinuous-timefilters
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•in the context of filter design,
•impulseinvarianceisamethodforobtainingadiscrete-timesystemwhose
frequencyresponseisdeterminedbythefrequencyresponseofa
continuous-timesystem.
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Impulse Invariance
•the impulse response of the discrete-time filter is chosen
proportional to equally spaced samples of the impulse response of
the continuous-time filter
•where T
drepresents a sampling interval
•the parameter Td has no role whatsoever in the design process or the
resulting discrete-time filter
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•The frequency response of the discrete-time filter is related to the
frequency response of the continuous-time filter by
•If
•Then
•i.e., the discrete-time and continuous-time frequency responses are
related by a linear scaling of the frequency axis, namely, ω = ΩT
dfor
|ω| < π.
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Process
•forthedesignofadiscrete-timefilterwithgivenfrequencyresponse
specifications,thediscrete-timefilterspecificationsarefirst
transformedtocontinuous-timefilterspecificationsusing
•we obtain the specifications on Hc(jΩ) by applying the relation
•Afterobtainingacontinuous-timefilterthatmeetsthese
specifications,thecontinuoustimefilterwithsystemfunctionHc(s)is
transformedtothedesireddiscrete-timefilterwithsystemfunction
H(z)
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•in the transformation back to discrete-time frequency, H(e
jω
) will be
related to H
c(jΩ) through
•which again applies the transformation to the frequency axis.
•As a consequence, the “sampling” parameter T
dcannot be used to
control aliasing
•if Td is made smaller, then the cutoff frequency of the continuous-
time filter must increase in proportion
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•consider the system function of a causal continuous-time filter
expressed in terms of a partial fraction expansion
•Corresponding impulse response is
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•a pole at s = s
kin the s-plane transforms to a pole at z = e
SkTd
in the z-plane
and the coefficients in the partial fraction expansions of H
c(s) and H(z) are
equal, except for the scaling multiplier T
d
•If the continuous-time causal filter is stable (pole in the left half plane),
then the discrete-time causal filter is also stable (pole inside the unit circle)
•itisimportanttorecognizethattheimpulseinvariancedesignprocedure
doesnotcorrespondtoasimplemappingofthes-planetothez-plane
•thezerosinthediscrete-timesystemwillnotingeneralbemappedinthesameway
thepolesaremapped
•SeeExample2Chapter7Oppenheim
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•In the impulse invariance design procedure, the relationship between
continuous time and discrete-time frequency is linear; consequently,
except for aliasing, the shape of the frequency response is preserved.
•Theimpulseinvariancetechniqueisappropriateonlyforbandlimited
filters;highpassorbandstopcontinuous-timefilters,forexample,
wouldrequireadditionalbandlimitingtoavoidseverealiasing
distortionifimpulseinvariancedesignisused
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Bilinear Transformation
•Algebraic transformation b/w variables ‘s’and ‘z’
•One-one mapping between the s and z-planes
•Given a C.T. prototype filter H
c(s), the corresponding D.T. filter H(z) is
•TheparameterT
dhasnoinfluencewhatsoeverinthefilterdesign,itiscancelled
outaswestartwithdigitalspecsandreturntodigitalfilterwhilepassingby
analogfilter
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1
21
()
1
d
z
s
Tz
1
1
21
( ) ( ( ))
1
c
d
z
H z H
Tz
1 ( / 2)
1 ( / 2)
d
d
Ts
z
Ts
Substitute sj 1 2 2
1 2 2
dd
dd
T j T
z
T j T
//
//
(1) If <0, then |z|<1 for any
similarly, if >0, then |z|>1 for all
(2) If =0, then12
12
d
d
jT
z
jT
/
/ 1z 12
12
j d
d
jT
e
jT
/
/ 21
1
j
j
d
e
s
Te
/2
/2
2 sin / 222
tan / 2
2 cos / 2
j
j
dd
ej j
sj
T e T
2
tan( )
2
d
T
2arctan( / 2)
d
T
Since =0
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tan( )
2
d
T
2arctan( / 2)
d
T
•Whole of left-half s-plane mapped to inside the unit circle in z-plane
•The whole of imaginary axis on s-plane mapped to unit circle, no
aliasing problem
•0≤Ω≤∞maps to 0≤??????≤??????
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•The bilinear transformation
avoids the problem of
aliasing encountered with the
use of impulse invariance,
because it maps the entire
imaginary axis of the s-plane
onto the unit circle in the z-
plane.
•The price paid for this,
however, is the nonlinear
compression of the
frequency axis
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•the design of discrete-time
filters using the bilinear
transformation is useful only
when the nonlinear
compression of the frequency
axis can be tolerated or
compensated for
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Filter Design Comparisons
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Butterworth Filter
•Monotonic in both passbandand stopband.
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ChebyshevType I
•Equiripplein passband, Monotonic in stopband
•Where the Chebyshevpolynomial is given by
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1
|)(|
22
2
cN
c
V
jH
)coscos()(
1
xNxV
N
c
11
ChebyshevType II
•Monotonic in passband, Equiripplein stopband
•Where the Chebyshevpolynomial is given by
Zulaikha Kiran, 2024 35)coscos()(
1
xNxV
N
122
2
)]/([1
1
|)(|
cN
c
V
jH
1 c
Elliptic Filter
•Equiripplein both passbandand
stopband
•where U
N(Ω) is a Jacobianelliptic
function
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1
|)(|
22
2
N
c
U
jH
Frequency Transformations of LowpassIIR Filters to Other Filters
•Design a frequency-normalized prototype LPF
•Use algebraic transformation to derive the desired filter from prototype LPF
Given a lowpasssystem function
lp
HZ
Transform it to a new system function Hz
11
Z G z
11lp
Z G z
H z H Z
Constraints on the transformation:
1
Gz
must be a rational function of1
z
The inside of the unit circle of the Z-plane must map to the inside of the unit circle of the z-plane
The unit circle of the Z-plane must map onto the unit circle of the z-plane
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