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Selective Filters

discriminates, according to some attribute of the objects

applied at its input, what passes through it.

• For example, an air filter allows air to pass through it but

prevents dust particles that are present in the air from

passing through.

• An oil filter performs a similar function, with the exception

that oil is the substance allowed to pass through the filter,

while particles of dirt are collected at the input to the filter

and prevented from passing through.

Linear Time-Invariant Systems as Frequency-

Selective Filters

• In photography, an ultraviolet filter is often used to prevent

ultraviolet light, which is present in sunlight and which is

not a part of visible light, from passing through and

affecting the chemicals on the film.

• A linear time-invariant system also performs a type of

discrimination or filtering among the various frequency

components at its input.

• The nature of this filtering action is determined by the

frequency response characteristics H(ω), which in turn

depends on the choice of the system parameters (e.g., the

coefficients (ak) and (bk) in the difference equation

characterization of the system).

Linear Time-Invariant Systems as Frequency-

Selective Filters

design frequency-selective filters that pass signals

with frequency components in some bands while

they attenuate signals containing frequency

components in other frequency bands.

Linear Time-Invariant Systems as Frequency-

Selective Filters

• In general, a linear time-invariant system modifies the input

signal spectrum X(ω) according to its frequency response

H(ω) to yield an output signal with spectrum 𝒀(𝝎) =

𝑯(𝝎)𝑿(𝝎).

• In a sense, H(ω) acts as a weighting function or a spectral

shaping function to the different frequency components in

the input signal.

• When viewed in this context, any linear time-invariant

system can be considered to be a frequency-shaping filter,

even though it may not necessarily completely block any or

all frequency components.

• Consequently, the terms“linear time-invariant

system”and“filter” are synonymous and are often used

interchangeably.

Linear Time-Invariant Systems as Frequency-

Selective Filters

• We use the term filter to describe a linear time -invariant

system used to perform spectral shaping or frequency-

selective filtering.

• Filtering is used in digital signal processing in a variety of

ways. For example,

-removal of undesirable noise from desired signals,

-spectral shaping such as equalization of communication

channels,

-signal detection in radar, sonar, and communications, and

-for performing spectral analysis of signals, and so on.

Ideal Filter Characteristics

• Filters are usually classified according to their

frequency-domain characteristics as

- low-pass

- high-pass

- band-pass

- band-stop or band-elimination filters

- all-pass

• The ideal magnitude response characteristics of these

types of filters are illustrated in Fig. 4.43 as shown,

these ideal filters have a constant-gain (usually taken as

unity-gain) pass-band characteristic and zero gain in

their stop-band.

Ideal Filter Characteristics

Ideal Filter Characteristics

Ideal Filter Characteristics

selective discrete-time filters.

Ideal Filter Characteristics

• Another characteristic of an ideal filter is a linear phase

response.

• To demonstrate this point, let us assume that a signal

sequence {x(n)} with frequency components confined to

the frequency range ω1 < ω < ω2 is passed through a filter

with frequency response

(4.5.1)

Ideal Filter Characteristics

The signal at the output of the filter has a spectrum

of the Fourier transform, we obtain the time-

domain output

Ideal Filter Characteristics

amplitude-scaled version of the input signal.

• A pure delay is usually tolerable and is not considered a

distortion of the signal. Neither is amplitude scaling.

• Therefore, ideal filters have a linear phase characteristic

within their pass-band, that is,

𝜃 𝜔 = −𝜔𝑛0 (4.5.4)

Ideal Filter Characteristics

• The derivative of the phase with respect to frequency has

the units of delay. Hence we can define the signal delay as a

function of frequency as

of the filter.

• We interpret τg(ω) as the time delay that a signal

component of frequency ω undergoes as it passes from the

input to the output of the system.

Ideal Filter Characteristics

• Note that when ϴ(ω) is linear as in (4.5.4),

τg(ω) = no = constant.

• In this case all frequency components of the input signal

undergo the same time delay.

• In conclusion, ideal filters have a constant magnitude

characteristic and a linear phase characteristic within their

pass-band.

• In all cases, such filters are not physically realizable but

serve as a mathematical idealization of practical filters.

Ideal Filter Characteristics

• For example, the ideal low pass filter has an impulse

response

(4.5.6)

absolutely summable and therefore it is also unstable.

• Consequently, this ideal filter is physically unrealizable.

• Nevertheless, its frequency response characteristics can

be approximated very closely by practical, physically

realizable filters.

Design of simple digital filters

• We already know how the location of poles and zeros

affects the frequency response characteristics of the

system

• There is a graphical method which can be used for

computing the frequency response characteristics from the

pole-zero plot.

• This same approach can be used to design a number of

simple but important digital filters with desirable frequency

response characteristics.

Design of simple digital filters

• The basic principle underlying the pole-zero placement

method is to locate poles near points of the unit circle

corresponding to frequencies to be emphasized, and to

place zeros near the frequencies to be deemphasized.

Furthermore, the following constraints must be imposed:

1. All poles should be placed inside the unit circle in order for

the filter to be stable. However, zeros can be placed

anywhere in the z-plane.

2. All complex zeros and poles must occur in complex-

conjugate pairs in order for the filter coefficients to be real.

Design of simple digital filters

• for a given pole-zero pattern, the system function H(z)

can be expressed as

(4.5.7)

frequency response at some specified frequency . That is,

bo is selected such that

𝐻(𝜔0 ) = 1 (4.5.8)

Design of simple digital filters

• Usually, N is selected to equal or exceed M, so that the

filter has more nontrivial poles than zeros.

Lowpass, Highpass, and Bandpass Filters

• In the design of low pass digital filters, the poles should be

placed near the unit circle at points corresponding to low

frequencies (near ω = 0) and zeros should be placed near or

on the unit circle at points corresponding to high

frequencies (near ω = π).

• The opposite holds true for highpass filters.

• Figure 4.44 illustrates the pole-zero placement of three low

pass and three high pass filters.

Lowpass, Highpass, and Bandpass Filters

Figure 4.44

Lowpass, Highpass, and Bandpass Filters

• The magnitude and phase responses for the single-

pole filter with system function

• The gain G was selected as 1 − 𝑎, so that the filter

has unity gain at ω = 0.

• The gain of this filter at high frequencies is relatively

small.

Lowpass, Highpass, and Bandpass Filters

• The addition of a zero at z = -1 further attenuates

the response of the filter at high frequencies.

• This leads to a filter with a system function

illustrated in Fig. 4.45. In this case the magnitude of

H2(ω) goes to zero at ω = π.

Lowpass, Highpass, and Bandpass Filters

Lowpass, Highpass, and Bandpass Filters

• Similarly, we can obtain simple highpass filters by

reflecting (folding) the pole-zero locations of the

low pass filters about the imaginary axis in the z-

plane. Thus we obtain the system function

illustrated in Fig. 4.46 for a = 0.9.

Lowpass, Highpass, and Bandpass Filters

Lowpass, Highpass, and Bandpass Filters

Example 4.51

• A two-pole lowpass filter has the system function

frequency response H(ω) satisfies the conditions

H(0) = 1 and

Lowpass, Highpass, and Bandpass Filters

at 𝜔 = 𝜋/4

Lowpass, Highpass, and Bandpass Filters

• Hence

or equivalently

Consequently, the system function for the desired filter is

Lowpass, Highpass, and Bandpass Filters

Bandpass Filters

• The same principles can be applied for the design of

band pass filters.

• Basically, the band pass filter should contain one or

more pairs of complex–conjugate poles near the

unit circle, in the vicinity of the frequency band that

constitutes the passband of the filter.

• This is illustrated in the following example

Lowpass, Highpass, and Bandpass Filters

Example 4.5.2

• Design a two-pole bandpass filter that has the center of

its passband at ω = π/2, zero in its frequency response

characteristic at ω = 0 and ω = π, and its magnitude

response is 1Τ 2 at 𝜔 = 4𝜋Τ9 .

Lowpass, Highpass, and Bandpass Filters

Solution

Clearly the filter must have poles at

𝑝1,2 = 𝑟𝑒 ±𝑗𝜋Τ2

and zeros at z = 1 and z = — 1.

Consequently, the system function is

Lowpass, Highpass, and Bandpass Filters

• The gain factor is determined by evaluating the

frequency response H(ω) of the filter at ω = π/2. Thus

we have

4π/9. Thus we have

Lowpass, Highpass, and Bandpass Filters

or equivalently,

Therefore, the system function for the desired filter is

k

Figure 4.47

Magnitude and phase

response of a simple

bandpass filter in

Example 4.5.2;

0.15[(1−𝑧 −2 )

𝐻(𝑧) =

(1+0.7𝑧 −2 )

Lowpass, Highpass, and Bandpass Filters

• It should be emphasized that the main purpose of the

foregoing methodology for designing simple digital

filters by pole-zero placement is to provide insight into

the effect that poles and zeros have on the frequency

response characteristic of systems.

• The methodology is not intended as a good method for

designing digital filters with well-specified passband

and stop band characteristics.

Lowpass, Highpass, and Bandpass Filters

A simple lowpass-to-highpass filter transformation

• Suppose that we have designed a prototype low pass filter

with impulse response hlp(n). By using the frequency

translation property of the Fourier transform, it is possible

to convert the prototype filter to either a bandpass or a

highpass filter.

• If hlp(n) denotes the impulse response of a low pass filter

with frequency response Hlp(ω), a highpass filter can be

obtained by translating HlP(ω) by π radians (i.e., replacing ω

by 𝜔 − 𝜋). Thus

Lowpass, Highpass, and Bandpass Filters

where HhP(ω) is the frequency response of the highpass

filter.

• Since a frequency translation of π radians is equivalent

to multiplication of the impulse response hlp(n) by 𝑒𝑗𝜋𝑛 ,

the impulse response of the highpass filter is,

(4.5.13)

• therefore , the impulse response of the highpass filter is

simply obtained from the impulse response of the low

pass filter by changing the signs of the odd-numbered

samples in hlp(n).

Lowpass, Highpass, and Bandpass Filters

• Conversely,

(4.5.14)

• If the low pass filter is described by the difference equation

(4.5.15)

its frequency response is

(4.5.16)

Lowpass, Highpass, and Bandpass Filters

• Now, if we replace ω by 𝜔 − 𝜋, in (4.5.16), then

(4.5.17)

Lowpass, Highpass, and Bandpass Filters

Example 4.5.3

Lowpass, Highpass, and Bandpass Filters

Solution

• The difference equation for the highpass filter, according to

(4.5.18), is

𝑦 𝑛 = −0.9𝑦 𝑛 − 1 + 0.1𝑥(𝑛)

Lowpass, Highpass, and Bandpass Filters

Digital Resonators

• A digital resonator is a special two-pole bandpass filter with

the pair of complex conjugate poles located near the unit

circle as shown in Fig. 4.48(a). The magnitude of the

frequency response of the filter is shown in Fig. 4.48(b).

• The name resonator refers to the fact that the filter has a

large magnitude response (i.e., it resonates) in the vicinity

of the pole location.

• The angular position of the pole determines the resonant

frequency of the filter.

• Digital resonators are useful in many applications, including

simple bandpass filtering and speech generation.

Lowpass, Highpass, and Bandpass Filters

Lowpass, Highpass, and Bandpass Filters

magnitude and phase response of a digital resonator with

(1) r =0.8 and (2) r = 0.95.

Lowpass, Highpass, and Bandpass Filters

• In the design of a digital resonator with a resonant peak at

or near ω = ωo , we select the complex-conjugate poles at

are many possible choices, two cases are of special interest.

• One choice is to locate the zeros at the origin. The other

choice is to locate a zero at z = 1 and a zero at z = -1.

• This choice completely eliminates the response of the filter

at frequencies ω = 0 and ω = π , and it is useful in many

practical applications.

Lowpass, Highpass, and Bandpass Filters

• The system function of the digital resonator with zeros at

the origin is

(4.5.19)

(4.5.20)

gain bo so that |H(ωo)| = 1. From (4.5.19) we obtain

(4.5.21)

Lowpass, Highpass, and Bandpass Filters

and hence

expressed as

Lowpass, Highpass, and Bandpass Filters

• where U1(ω) and U2(ω) are the magnitudes of the vectors

from p1 and p2 to the point ω in the unit circle and Φ1(ω)

and Φ2(ω) are the corresponding angles of these two

vectors.

• The magnitudes U1(ω) and U2(ω) may be expressed as

ω = ωo. The product U1(ω)U2(ω) reaches a minimum value

at the frequency

Lowpass, Highpass, and Bandpass Filters

which defines precisely the resonant frequency of the filter.

• We observe that when r is very close to unity, ωr ≈ωo,

which is the angular position of the pole.

• We also observe that as r approaches unity, the resonance

peak becomes sharper because U1(ω) changes more rapidly

in relative size in the vicinity of ωo.

• A quantitative measure of the sharpness of the resonance

is provided by the 3-dB band width ∆ω of the filter.

Lowpass, Highpass, and Bandpass Filters

• For values of r close to unity

resonators with ωo = π/3 , r = 0.8 and ωo = π/3, r = 0.95.

• We note that the phase response undergoes its greatest

rate of change near the resonant frequency.

• If the zeros of the digital resonator are placed at z = 1 and z

= -1, the resonator has the system function

(4.5.27)

Lowpass, Highpass, and Bandpass Filters

magnitude and phase response of the resonator.

• For example, the magnitude response is

Lowpass, Highpass, and Bandpass Filters

frequency is altered from that given by the expression in

(4.5.25). The bandwidth of the filter is also altered .

Lowpass, Highpass, and Bandpass Filters

• Figure 4.49 illustrates the magnitude and phase

characteristics for ωo= π/3, r = 0.8 and ωo = π/3, r =

0.95.

• We observe that this filter has a slightly smaller

bandwidth than the resonator, which has zeros at

the origin.

• In addition, there appears to be a very small shift in

the resonant frequency due to the presence of the

zeros.

Lowpass, Highpass, and Bandpass Filters

resonator with zeros at ω = 0 and ω = π and (1) r = 0.8 and

(2) r = 0.95.

Lowpass, Highpass, and Bandpass Filters

Notch Filters

• A notch filter is a filter that contains one or more deep

notches or, ideally, perfect nulls in its frequency response

characteristic.

• Figure 4.50 illustrates the frequency response characteristic

of a notch filter with nulls at frequencies ωo and ω1.

• Notch filters are useful in many applications where specific

frequency components must be eliminated .

• For example, instrumentation and recording systems

require that the power-line frequency of 60Hz and its

harmonics be eliminated.

Lowpass, Highpass, and Bandpass Filters

notch filter.

Lowpass, Highpass, and Bandpass Filters

• To create a null in the frequency response of a filter at a

frequency ωo, we simply introduce a pair of complex-

conjugate zeros on the unit circle at an angle ωo. That is,

𝑧1,2 = 𝑒 ±𝑗𝜔0

• Thus the system function for an FIR notch filter is simply

(4.5.30)

Lowpass, Highpass, and Bandpass Filters

• As an illustration, Fig. 4.51 shows the magnitude response

for a notch filter having a null at ω = π/4.

Fig 4.51

Lowpass, Highpass, and Bandpass Filters

Lowpass, Highpass, and Bandpass Filters

All-Pass Filters

• An all-pass filter is defined as a system that has a constant

magnitude response for all frequencies, that is

system with system function

𝐻(𝑧) = 𝑧 −𝑘

• This system passes all signals without modification except

for a delay of k samples.

• This is a trivial all-pass system that has a linear phase

response characteristic.

Lowpass, Highpass, and Bandpass Filters

• A more interesting all-pass filter is described by the system

function

• If we define the polynomial A(z) as

Lowpass, Highpass, and Bandpass Filters

• then (4.5.43) can be expressed as

(4.5.44)

Since

FIR FILTER DESIGN

• The frequency response of an Nth-order causal FIR filter is

h(n) that result in a frequency response that satisfies a given set

of filter specifications.

FIR filters have two important advantages over IIR filters.

• First, they are guaranteed to be stable, even after the filter

coefficients have been quantized.

• Second, they may be easily constrained to have (generalized)

linear phase.

Because FIR filters are generally designed to have linear phase,

in the following we consider the design of linear phase FIR

filters.

Linear Phase FIR Design Using Windows

• Let ℎ𝑑 (𝑛) be the unit sample response of an ideal

frequency selective filter with linear phase,

𝐻𝑑 𝑒 𝑗𝜔 = 𝐴 𝑒 𝑗𝜔 𝑒 −𝑗(𝛼𝜔−𝛽)

• Because ℎ𝑑 (𝑛) will generally be infinite in length, it is

necessary to find an FIR approximation to 𝐻𝑑 (𝑒 𝑗𝜔 )

• With the window design method, the filter is designed

by windowing the unit sample response,

ℎ 𝑛 = ℎ𝑑 𝑛 𝜔(𝑛)

where ω(n) is a finite-length window that is equal to zero

outside the interval 0 ≤ n ≤ N and is symmetric about its

midpoint:

𝜔(𝑛) = 𝜔(𝑁 − 𝑛)

Linear Phase FIR Design Using Windows

• The effect of the window on the frequency

response may be seen from the complex

convolution theorem,

the discrete-time Fourier transform of the window,

W(𝑒 𝑗𝜔 )

• There are many different types of windows that

may be used in the window design method, a few

of which are listed in Table 9-1 .

Linear Phase FIR Design Using Windows

Linear Phase FIR Design Using Windows

• How well the frequency response of a filter designed with

the window design method approximates a desired

response, 𝐻𝑑 𝑒 𝑗𝜔 is determined by two factors (see Fig. 9-

2):

1. The width of the main lobe of W(𝑒 𝑗𝜔 )

2. The peak side-lobe amplitude of W(𝑒 𝑗𝜔 )

• Ideally, the main-lobe width should be narrow, and the

side-lobe amplitude should be small.

• However, for a fixed-length window, these cannot be

minimized independently.

Fig. 9-2. The DTFT of a typical window, which is characterized by the

width of its main lobe ,∆, and the peak amplitude of its side lobes, A,

relative to the amplitude of W(𝑒 𝑗𝜔 ) at ω = 0.

Linear Phase FIR Design Using Windows

Some general properties of windows are as follows:

1. As the length N of the window increases, the width of the

main lobe decreases, which results in a decrease in the

transition width between passbands and stopbands.

This relationship is given approximately by

𝑁∆𝑓 = 𝑐 9.1

where ∆f is the transition width, and c is a parameter that

depends on the window.

2. The peak side-lobe amplitude of the window is determined

by the shape of the window, and it is essentially independent

of the window length.

3. If the window shape is changed to decrease the side-lobe

amplitude, the width of the main lobe will generally increase.

Linear Phase FIR Design Using Windows

• Listed in Table 9.2 are the side-lobe amplitudes of several

windows along with the approximate transition width and

stopband attenuation that results when the given window

is used to design an Nth-order low-pass filter.

Linear Phase FIR Design Using Windows

EXAMPLE 9.3.1

Suppose that we would like to design an FIR linear phase

low-pass filter according to the following specifications:

we may use a Hanning window. Although we could also

use a Hamming or a Blackman window, these windows

would overdesign the filter and produce a larger

stopband attenuation at the expense of an increase in

the transition width. Because the specification calls for

a transition width of ∆𝜔 = 𝜔𝑠 − 𝜔𝑝 = 0.02𝜋, or ∆f =

0.01, with

𝑁∆𝑓 = 3.1

Linear Phase FIR Design Using Windows

• for a Hanning window (see Table 9.2), an estimate

of the required filter order is

the ideal low-pass filter that is to be windowed.

𝜔𝑠 +𝜔𝑝

With a cut-off frequency of 𝜔𝑐 = ൗ2 = 0.2𝜋,

and a delay of α = N/2 = 155, the unit sample

response is

Design of IIR Filters from Analog Filters

The design of a digital filter from an analog prototype

requires that we transform ℎ𝑎 (𝑡) to h(n) or 𝐻𝑎 (𝑠) to

H(z).

• A mapping from the s-plane to the z-plane may be

written as

this transformation to produce an acceptable digital

filter, the mapping m(z) should have the following

properties:

Design of IIR Filters from Analog Filters

1. The mapping from the jΩ-axis to the unit circle, 𝑧 = 1,

should be one to one and onto the unit circle in order to

preserve the frequency response characteristics of the

analog filter.

2. Points in the left-half s-plane should map to points inside

the unit circle to preserve the stability of the analog filter.

3. The mapping m(z) should be a rational function of z so

that a rational 𝐻𝑎 (𝑠) is mapped to a rational H(z).

Design of IIR Filters from Analog Filters

Two approaches that are commonly used to map analog

filters into digital filters are impulse invariance method and

the Bilinear Transformation.

• The bilinear transformation is a mapping from the s-plane

to the z-plane defined by

2 1−𝑧 −1

𝑠=

𝑇 1+𝑧 −1

• Given an analog filter with a system function 𝐻𝑎 (𝑠) the

digital filter is designed as follows:

Design of IIR Filters from Analog Filters

2 1 − 𝑧 −1

𝐻 𝑧 = 𝐻𝑎

𝑇 1 + 𝑧 −1

• The bilinear transformation is a rational function that maps

the left-half s-plane inside the unit circle and maps the jΩ-axis

in a one-to-one manner onto the unit circle.

• However, the relationship between the jΩ-axis and the unit

circle is highly nonlinear and is given by the frequency

warping function

Ω𝑇𝑠

𝜔 = 𝑡𝑎𝑛 (9.12)

2

Design of IIR Filters from Analog Filters

• As a result of this warping, the bilinear transformation will

only preserve the magnitude response of analog filters that

have an ideal response that is piecewise constant.

• Therefore, the bilinear transformation is generally only

used in the design of frequency selective filters.

• The parameter 𝑇𝑠 , in the bilinear transformation is normally

included for historical reasons. However, it does not enter

into the design process, because it only scales the jΩ-axis in

the frequency warping function, and this scaling may be

done in the specification of the analog filter.

• Therefore, 𝑇𝑠 , may be set to any value to simplify the

design procedure.

Design of IIR Filters from Analog Filters

• The steps involved in the design of a digital low-pass filter

with a passband cutoff frequency 𝜔𝑝 , stopband cutoff

frequency 𝜔𝑠 , passband ripple 𝛿𝑝 and stopband ripple 𝛿𝑠 ,

are as follows:

1. Prewarp the passband and stopband cutoff frequencies of

the digital filter, 𝜔𝑝 and 𝜔𝑠 using the inverse of Eq. (9.12)

to determine the passband and cutoff frequencies of the

analog low-pass filter.

With 𝑇𝑠 = 2, the prewarping function is

𝜔

Ω = 𝑡𝑎𝑛

2

Design of IIR Filters from Analog Filters

2. Design an analog low-pass filter with the cutoff

frequencies found in step 1 and a passband and

stopband ripple 𝛿𝑝 and 𝛿𝑠 respectively.

3. Apply the bilinear transformation to the filter

designed in step 2.

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