AET302

DIGITAL SIGNAL PROCESSING

S6 AEI
MODULE 3

BY,
ASWINI S H
ASSISTANT PROFESSOR
DEPARTMENT OF ECE, CET
 INTRODUCTION TO IIR FILTERS
➢ IIR filters are of recursive type, where the present output sample depends on the
present input, past input & past output samples.
➢ When compared to FIR filters, the IIR filter satisfies a given magnitude response
design objective with a lower order filter
➢ IIR filter does not exhibit linear phase or constant group delay behaviour
➢ If the principal objective of the digital filter design is to satisfy the specified
magnitude response alone, an IIR is usually the preferred choice
➢ Since the order of an IIR filter is significantly less than that of a FIR filter, IIR filter
would require fewer coefficients.

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 INTRODUCTION TO IIR FILTERS (Contd.)
Trade-offs:
When selecting between FIR and IIR filters:
➢ Linear Phase Requirement: If linear phase is critical (e.g., in applications like audio
processing), FIR filters might be preferred despite their higher computational
requirements.
➢ Transition Bandwidth and Efficiency: For applications where computational
efficiency is crucial or steeper transition bands are acceptable, IIR filters might be
chosen despite their nonlinear phase response.

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 INTRODUCTION TO IIR FILTERS (Contd.)
➢ IIR filters are described by the difference equation:

➢ On taking Z transform, we get:

Design of IIR filter for the given specifications is to find

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the filter coefficients aks and bks of the above equation
 DESIGN OF IIR FILTERS FROM ANALOG FILTERS
➢ The most common method of designing IIR digital filters is known as indirect
method, which involves first designing an analog prototype filter and then
transforming the prototype to a digital filter
➢ For the given specifications of a digital filter, the derivation of the digital filter
transfer function requires 3 steps:
1. Map the desired digital filter specifications into those for an equivalent
analog filter
2. Derive the transfer function for the analog prototype
3. Transform the transfer function of the analog prototype into an equivalent
digital filter transfer function
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DESIGN OF IIR FILTERS FROM ANALOG FILTERS (Contd.)
➢ 2 approaches can be followed:
(a) Approach 1:
➢ Apply frequency band transformation in s-domain to obtain other frequency
selective filters (high-pass, band-pass, and band-stop) H(s).
➢ Transform the analog filter H(s) to a digital filter H(z).

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 DESIGN OF IIR FILTERS FROM ANALOG FILTERS
(Contd.)
(b) Approach 2:
➢ Transform the analog LPF HLP(s) to a digital filter HLP(z).
➢ Apply frequency band transformation in z-domain to obtain other frequency
selective filters (high-pass, band-pass, and band-stop) H(z).

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DESIGN OF IIR FILTERS FROM ANALOG FILTERS
(Contd.)
➢ To convert an analog filter H(s) into a digital filter H(z), the transformation should
possess the following properties:
➢ The 𝐣𝜴 axis of the s-plane should map onto the unit circle in the z-plane.
➢ The left half of the s-plane maps to the inside of the unit circle. This implies
that the poles in the left half of the s-plane maps into the inside of the unit
circle. Hence a stable analog filter will be transformed to a stable digital filter.

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 DESIGN OF IIR FILTERS FROM ANALOG FILTERS
(Contd.)

Specifications for the magnitude response of a low pass filter (a) analog , (b) digital
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DESIGN OF IIR FILTERS FROM ANALOG FILTERS
(Contd.)
➢ The range of frequencies that are passed through the filter is called passband and those frequencies that are
blocked is called stopband.
➢ Transition band is specified between the passband and the stop band to permit the magnitude to drop off
smoothly.

𝜹𝒑 - passband error tolerance

𝜹𝒔 - maximum allowable
magnitude in the stopband

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DESIGN OF IIR FILTERS FROM ANALOG FILTERS (Contd.)

➢ Cutoff Frequency:
➢ The cutoff frequency is a critical parameter that defines the boundary between the passband
and the stopband in a digital filter's frequency response.
➢ For low-pass and high-pass filters, the cutoff frequency marks the point at which the filter's
response starts to transition from the passband to the stopband.
➢ For band-pass and band-stop filters, there are multiple cutoff frequencies defining the edges of
the passband and stopband.
➢ Transition Bandwidth:
➢ The transition bandwidth is the frequency range over which the filter transitions from the
passband to the stopband (or vice versa).
➢ It specifies the width of the transition region in the filter's frequency response and affects the
sharpness of the filter's cutoff.
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DESIGN OF IIR FILTERS FROM ANALOG FILTERS (Contd.)
➢ Passband Ripple:
➢ Passband ripple refers to the variation in magnitude response within the passband of the
filter.
➢ It quantifies the amount of ripple allowed in the passband and affects the filter's amplitude
response in the frequency domain.
➢ Stopband Ripple:
➢ Stopband ripple is the maximum variation in the magnitude of the filter's frequency response
within the stopband.
➢ It quantifies the deviation of the filter's response from the ideal stopband attenuation level.
➢ Passband Attenuation:
➢ Passband attenuation refers to the reduction in signal amplitude within the specified range of
frequencies that a filter is designed to allow or pass through without significant attenuation
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DESIGN OF IIR FILTERS FROM ANALOG FILTERS (Contd.)

➢ Stopband Attenuation:
➢ Stopband attenuation specifies the level of attenuation (or suppression) of frequencies in the
stopband of the filter.
➢ It defines how effectively the filter suppresses unwanted frequencies outside the passband.
➢ Filter Order:
➢ The filter order determines the complexity of the filter and the number of filter coefficients
required to achieve the desired frequency response.
➢ Higher-order filters typically offer sharper cutoffs and better stopband attenuation but may
require more computational resources.

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DESIGN OF IIR FILTERS FROM ANALOG FILTERS
(Contd.)
➢ There are 2 types of analog filter design
➢ They are:
➢ Butterworth filter
➢ Chebyshev filter

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 ANALOG LOW PASS BUTTERWORTH FILTER

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 ANALOG LOW PASS BUTTERWORTH FILTER (Contd.)
➢ Butterworth low pass filters are all pole filters with a flat passband and is
characterized by the magnitude frequency response

N = 1,2,3,…….

Where N is the order of the filter and Ω𝐶 is the cutoff frequency

➢ The function is monotonically decreasing where the maximum response is unity
at Ω=0
➢ Ideal response is shown by dash line
➢ Magnitude response approaches ideal low pass characteristics as the order N 𝜴𝒑
𝜴𝑪 =
𝜺𝟏/𝑵
increases
➢ For values Ω < Ω𝐶 ; 𝐻(𝑗Ω ≈ 1, for values Ω > Ω𝐶 ; 𝐻(𝑗Ω decreases rapidly
➢ At Ω =
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Ω𝐶 ; the curve passes through 0.707, which corresponds to -3dB point 16
ANALOG LOW PASS BUTTERWORTH FILTER (Contd.)

➢ The magnitude square response of a normalized Butterworth filter (Ω𝑐=1𝑟𝑎𝑑/sec) is:

N = 1,2,3,…….

➢ The above function has a total of 2N poles which lie on a unit circle - N poles on the
left half of the s plane as well as N poles on the right half of the s plane due to the
presence of H(s) & H(-s).
➢ Angular separation between the poles is given by:
3600
which is = 600 for N = 3
2𝑁

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ANALOG LOW PASS BUTTERWORTH FILTER (Contd.)
Q. Find the expression for order N of an analog low pass Butterworth filter .

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ANALOG LOW PASS BUTTERWORTH FILTER (Contd.)

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ANALOG LOW PASS BUTTERWORTH FILTER (Contd.)

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ANALOG LOW PASS BUTTERWORTH FILTER (Contd.)

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ANALOG LOW PASS BUTTERWORTH FILTER (Contd.)

N = 3.758

Rounding off to the next higher integer, we get N = 4

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ANALOG LOW PASS BUTTERWORTH FILTER (Contd.)

Q. Find the expression for cut-off frequency of an analog low pass Butterworth filter.

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ANALOG LOW PASS BUTTERWORTH FILTER (Contd.)

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STEPS TO DESIGN ANALOG BUTTERWORTH LOW PASS FILTER

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STEPS TO DESIGN ANALOG BUTTERWORTH LOW PASS FILTER
(Contd.)
Necessary Equations for Butterworth LPF Design:

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STEPS TO DESIGN ANALOG BUTTERWORTH LOW PASS FILTER
(Contd.)

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STEPS TO DESIGN ANALOG BUTTERWORTH LOW PASS FILTER
(Contd.)
Q. Design an analog Butterworth filter that has a -2dB passband attenuation at a
frequency of 20 rad/sec and at least -10 dB stopband attenuation at 30 rad/sec.

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STEPS TO DESIGN ANALOG BUTTERWORTH LOW PASS FILTER
(Contd.)

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STEPS TO DESIGN ANALOG BUTTERWORTH LOW PASS FILTER (Contd.)

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STEPS TO DESIGN ANALOG BUTTERWORTH LOW PASS FILTER
(Contd.)

𝝀
𝒍𝒐𝒈
𝑵≥ 𝜺
𝜴
𝒍𝒐𝒈 𝜴 𝑺
𝑷

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𝑵 ≥ 𝟑. 𝟑𝟒 ≈ 𝟒
STEPS TO DESIGN ANALOG BUTTERWORTH LOW PASS FILTER
(Contd.)

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 FREQUENCY TRANSFORMATIONS IN ANALOG DOMAIN

➢ So far we have discussed designing of a low pass filter for the given specifications
➢ Frequency transformations can be done so as to design low pass filters with
different passband frequencies, high pass filters, bandpass filters and band stop
filters from a normalised low pass analog filter (𝜴𝑪 = 𝟏 𝒓𝒂𝒅/𝐬𝐞𝐜)

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FREQUENCY TRANSFORMATIONS IN ANALOG DOMAIN

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FREQUENCY TRANSFORMATIONS IN ANALOG DOMAIN

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FREQUENCY TRANSFORMATIONS IN ANALOG DOMAIN

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FREQUENCY TRANSFORMATIONS IN ANALOG DOMAIN

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FREQUENCY TRANSFORMATIONS IN ANALOG DOMAIN

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FREQUENCY TRANSFORMATIONS IN ANALOG DOMAIN (Contd.)

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FREQUENCY TRANSFORMATIONS IN ANALOG DOMAIN (Contd.)

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FREQUENCY TRANSFORMATIONS IN ANALOG DOMAIN (Contd.)

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FREQUENCY TRANSFORMATIONS IN ANALOG DOMAIN (Contd.)

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OVERVIEW AND COMPARISON OF OTHER ANALOG FILTERS
ANALOG LOW PASS CHEBYSHEV FILTERS:
➢ There are two types of Chebyshev filters.
➢ Type I Chebyshev filters are that exhibit all-pole filters that exhibit equiripple behaviour in the
passband and a monotonic characteristics in the stopband.
➢ On the other hand, the family of type II Chebyshev filter contains both poles and zeros and exhibits
a monotonic behaviour in the passband and an equiripple behaviour in the stopband

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OVERVIEW AND COMPARISON OF OTHER ANALOG FILTERS (Contd.)
ANALOG LOW PASS CHEBYSHEV FILTERS (Contd.):
➢ Poles of the Chebyshev transfer function are located on an ellipse in the s-plane
➢ The equation of the ellipse is given by:

where a and b are minor and major axes of the ellipse respectively.

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OVERVIEW AND COMPARISON OF OTHER ANALOG FILTERS (Contd.)

COMPARISON BETWEEN CHEBYSHEV AND BUTTERWORTH FILTER:

➢ The magnitude response of Butterworth filter decreases monotonically as the frequency Ω
increases from 0 to ∞, whereas the magnitude response of the Chebyshev filter exhibits ripples in
the passband or stopband according to the type
➢ The transition band is more in Butterworth filter when compared to Chebyshev filter
➢ The poles of Butterworth filter lie on a circle, whereas the poles of the Chebyshev filter lie on an
ellipse
➢ For the same specifications, the number of poles in Butterworth are more when compared to the
Chebyshev filter i.e. the order of the Chebyshev filter is less than that of Butterworth. This is a
great advantage because less number of discrete components will be necessary to construct the
filter.

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OVERVIEW AND COMPARISON OF OTHER ANALOG FILTERS (Contd.)

COMPARISON BETWEEN CHEBYSHEV AND BUTTERWORTH FILTER (Contd.):

➢ Butterworth filters are easier to design compared to Chebyshev filters because they do not
require specifying ripple parameters. Chebyshev filters require specifying ripple parameters, which
can make their design more complex. However, they offer greater flexibility in controlling passband
and stopband characteristics.
➢ Butterworth filters are commonly used in applications where a flat passband response is desired,
such as audio equalization, data communication, and sensor signal conditioning. Chebyshev
filters are preferred in applications where sharp transitions between passband and stopband are
required, such as in radio frequency (RF) communication, instrumentation, and radar systems.

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DESIGN OF IIR FILTER FROM ANALOG FILTERS
IIR FILTER DESIGN BY IMPULSE INVARIANCE
➢ In impulse invariance method, the IIR filter is designed such that the unit impulse
response h(n) of digital filter is the sampled version of the impulse response of
analog filter
➢ The z-transform of an infinite impulse response is given by:

…(1)

➢ Consider the mapping of points from the s-plane to the z-plane implied by the
relation: …(2) 54
IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)
…(2)
➢ If we substitute 𝒔 = 𝝈 + 𝒋𝜴 and express the complex variable z in polar form as
𝒛 = 𝒓𝒆𝒋𝝎 we get:

…(3)

which gives:
…(4)

…(5)

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 IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)
…(3)

➢ The first term in the product in Eqn. (3), 𝑒 𝜎𝑇 , has a magnitude of 𝑒 𝜎𝑇 and an angle of
0 – a real number
➢ The second term 𝑒 𝑗Ω𝑇 , has unity magnitude and an angle of Ω𝑇
➢ The analog pole is mapped to a place in the z-plane of magnitude 𝒆𝝈𝑻 and angle 𝜴𝑻
➢ Real part of the analog pole determines the radius of the z-plane pole and the
imaginary part of the analog pole dictates the angle of the digital pole.

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 IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)
➢ Consider any pole on the 𝒋𝜴-axis, where 𝝈 = 𝟎
➢ These poles map to the z-plane at a radius 𝒓 = 𝒆𝟎.𝑻 = 𝟏
➢ Therefore, the impulse invariant mapping map poles from the s-plane’s 𝒋𝜴-axis to
the z-plane’s unit circle

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 IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)
➢ Consider the poles in the left half of s-plane where 𝝈 < 𝟎
➢ These poles map inside the unit circle, because 𝒓 = 𝒆𝝈𝑻 < 𝟏 for 𝝈 < 𝟎
➢ Therefore, all s-plane poles with negative real parts map to z-plane poles inside the
unit circle – stable analog poles are mapped to stable digital poles
➢ Impulse invariant mapping preserves the stability of the filter

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 IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)
➢ Consider the poles in the right half of s-plane where 𝝈 > 𝟎
➢ These poles map outside the unit circle, because 𝒓 = 𝒆𝝈𝑻 > 𝟏 for 𝝈 > 𝟎
➢ Therefore, all s-plane poles in the right half of s-plane map to digital poles outside
the unit circle

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 IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)
➢ Although the 𝒋𝜴-axis is mapped into the unit circle, it is not one-to-one mapping
rather it is many-to-one mapping, where many points in s-plane are mapped to a
single point in the z-plane
➢ Consider two poles in the s-plane with identical real parts, but with imaginary
𝟐𝝅
components differing by
𝑻

➢ Let the poles be:

…(6)

➢ These poles map to z-plane poles z1 and z2, via impulse invariant mapping
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 IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)
…(7)

…(8)
➢ From Eqn. (7) and (8), we find that these poles map to the same location in the z-plane

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 IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)
➢ There are an infinite number of s-plane poles that map to the same location in the
z-plane.
𝟐𝝅
➢ They must have the same real parts and that differ by some integer multiple of .
𝑻

➢ This is the main disadvantage of impulse invariant mapping.

𝝅 𝝅
➢ The s-plane poles having imaginary parts greater than or less than − cause aliasing,
𝑻 𝑻

when sampling analog signals

➢ The analog poles will not be aliased by the impulse invariant mapping if they are
𝝅
confined to the s-plane’s "Primary strip" (within of the real axis).
𝑻

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 IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)
➢ Let Ha(s) is the system function of an analog filter.
➢ This can be expressed in partial fraction form as:

…(9)

where 𝑝𝑘 are the poles of the analog filter & 𝑐𝑘 are the coefficients in the partial fraction
expansion.
➢ The inverse Laplace transform of Eq.(9) is:

…(10)
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 IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)
➢ If we sample ha(t) periodically at t = nT, we have:

…(11)

➢ We know:
…(12)

➢ Substituting Eqn. (11) in (12), we have:

…(13) 64
IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)

➢ For high sampling rates (for small T), the digital filter gain is high.
➢ Therefore, instead of Eq. (13), we can use

…(14)

➢ Due to the presence of aliasing, the impulse invariant method is appropriate

for the design of lowpass and bandpass filters only.
➢ The impulse invariance method is unsuccessful for implementing digital filters
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such as a high pass filter.
IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)

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 IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)
Q. For the analog transfer function, 𝑯(𝒔) = 𝟐/(𝒔𝟐 + 𝟑𝒔 + 𝟐), determine H(z) using
Impulse Invariance Method. Assume T=1 sec.

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IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)

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 IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)
Q) Design a third order Butterworth digital filter using impulse invariance technique.
Assume sampling period of T= 1 sec.

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IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)

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IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)

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IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)

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IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)

Some useful Laplace Transform formulas

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IIR FILTER DESIGN BY IMPULSE INVARIANCE (Contd.)

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION

➢ The bilinear transformation is a conformal mapping that transforms the 𝒋𝜴 axis into
the unit circle in the z - plane only once, thus avoiding aliasing of frequency
components
➢ All points in the LHP of ‘s’ are mapped inside the unit circle in the z – plane and all
points in the RHP of ‘s’ are mapped into corresponding points outside the unit
circle in the z – plane
➢ Consider an analog linear filter with system function: …(1)

which can be written as:

…(2) 75
IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)
➢ This can be characterized by the differential equation:

…(3)

➢ y(t) can be approximated by the trapezoidal formula.

➢ Thus,

…(4)

where y'(t) denotes the derivative of y(t).

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)
➢ The approximation of the integral in Eqn. (4) by the trapezoidal formula at t = nT and
t0 = nT – T yields:

…(5)

➢ From the differential Eq.(3) we obtain:

…(6)

➢ Substituting Eq.(6) in Eq.(5) we get:

…(7)
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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)

…(7)

➢ which implies:

…(8)

➢ With y(n) = y(nT) and x(n) = x(nT) we obtain the result:

➢ The Z - transform of this difference equation is:

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)

➢ The system function of the digital filter is:

𝑇
➢ Dividing numerator and denominator by (1 + 𝑧 −1 ) we get:
2

…(9)

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)
➢ Comparing Eqn. (1) and Eq.(9), the mapping from s-plane to the z-plane can be
obtained as:
…(10)

➢ This relationship between s and z is known as bilinear transformation

➢ Let 𝒛 = 𝒓𝒆𝒋𝝎 and
➢ 𝒔 = 𝝈 + 𝒋𝜴 …(11)

➢ Then Eqn. (10) can be expressed as:

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)
➢ Separating imaginary and real parts, we have:

…(12)

➢ Comparing Eq. (11) and Eq. (12), we have:

…(13)

➢ From Eqn.(13), we find that if r ≤ 1, then 𝜎 < 0 and if r > 1, then 𝜎 > 0
➢ Consequently, the LHP in 's' maps into the inside of the unit circle in the z – plane and the RHP in the
‘s’ maps into the outside of the unit circle
➢ When r = 1, 𝜎 = 0 and

…(15)

…(14) 81
IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)
THE WARPING EFFECT:
➢ Let Ω and ω represent the frequency variables of the analog filter and the derived
digital filter respectively
➢ From Eqn.(14), we have:
➢ For small value of 𝜔:

…(16)

➢ For low frequencies, the relationship between Ω and ω are linear, as a result the
digital filter have the same amplitude response as the analog filter.

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)
THE WARPING EFFECT (Contd.):
➢ For high frequencies, however, the relationship between Ω and ω becomes non-
linear and distortion is introduced in the frequency scale of the digital filter to that
of the analog filter
➢ This is also known as warping effect

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)
THE WARPING EFFECT (Contd.):
➢ The influence of the warping effect on the amplitude response is shown in the
below figure by considering an analog filter with a number of passbands centered at
regular intervals:
➢ The derived digital filter will have same
number of passbands.
➢ But the center frequencies and bandwidth
of higher frequency passband will tend to
reduce disproportionately

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)
THE WARPING EFFECT (Contd.):
➢ The influence of the warping effect on the phase response is shown in the below
figure:

➢ Considering an analog filter with linear

phase response, the phase response of the
derived digital filter will be non-linear.

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)
PREWARPING:
➢ The warping effect can be eliminated by prewarping the analog filter.
➢ This can be done by finding prewarping analog frequencies using the formula:

𝟐 𝝎
𝛀 = 𝒕𝒂𝒏 …(17)
𝑻 𝟐

➢ Therefore, we have:

…(18)

…(19)

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)
STEPS TO DESIGN DIGITAL FILTER USING BILINEAR TRANSFORM TECHNIQUE:

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 IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)

𝟐
Q. Apply bilinear transformation to 𝑯(𝒔) = with T=1 sec and find H(z)
(𝒔+𝟏)(𝒔+𝟐)

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)

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IIR FILTER DESIGN BY BILINEAR TRANSFORMATION (Contd.)

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FREQUENCY TRANSFORMATION IN DIGITAL DOMAIN

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FREQUENCY TRANSFORMATION IN DIGITAL DOMAIN (Contd.)

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FREQUENCY TRANSFORMATION IN DIGITAL DOMAIN (Contd.)

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FREQUENCY TRANSFORMATION IN DIGITAL DOMAIN (Contd.)

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FREQUENCY TRANSFORMATION IN DIGITAL DOMAIN (Contd.)

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