Hand Out

AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
Contents
Chapter 1 ..................................................................................................................................................... 3
1. Introduction to Network Analysis & Synthesis ................................................................................. 3
1.1. Lumped Circuit elements............................................................................................................ 4
Independent sources: ............................................................................................................................. 9
1.2. Classification of Networks ............................................................................................................... 10
1.2.1 Linear and Nonlinear Networks ................................................................................................. 10
1.2.2 Time-invariant and Time-variant Networks .............................................................................. 11
1.2.3 Passive and Active Networks..................................................................................................... 12
Chapter 2 ................................................................................................................................................... 14
Network Transform Representation and Analysis ...................................................................................... 14
2.1 Introduction ....................................................................................................................................... 14
2.1.1 Network Functions ..................................................................................................................... 14
2.2 Stability of a Network ....................................................................................................................... 18
Chapter 3 ................................................................................................................................................... 24
3 Elements of Realizability Theory ....................................................................................................... 24
3.1 Causality and Stability ...................................................................................................................... 24
3.2. Hurwitz Polynomials .................................................................................................................... 28
3.3. Positive Real Functions ................................................................................................................ 33
3.4. Elementary Synthesis procedures................................................................................................. 42
CHAPTER 4 .............................................................................................................................................. 45
Synthesis of One-Port Networks with Two Kinds Of Element .................................................................. 45
4.1. Introduction ...................................................................................................................................... 45
4.2. Properties and Synthesis of L-C Driving-point Immittances ........................................................... 45
4.2.1. Properties of L-C Immittance Functions................................................................................... 45
4.1.2. Synthesis of L-C Immittance Functions ................................................................................... 49
4.2. Properties and Synthesis of R-C Impedances and R-L Admittances ............................................... 55
4.2.1. Properties of R-C Impedances .................................................................................................. 56
4.2.2. Synthesis of R-C Impedances or R-L Admittances .................................................................. 59
4.3. Properties and Synthesis of R-L Impedances and R-C Admittances ............................................... 61
4.4. Synthesis of an R-L-C Functions ..................................................................................................... 64
CHAPTER 5 .............................................................................................................................................. 71
Two Port Networks ..................................................................................................................................... 71
5.1. Introduction .................................................................................................................................. 71
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5.2. Two-port Parameters .................................................................................................................... 72

5.2.1. Open circuit Impedance parameters (Z- Parameters) ........................................................... 73
5.2.2. Short circuit admittance parameters (Y- Parameters) ........................................................... 76
5.2.3. Hybrid Parameters ( h -parameters) .......................................................................................... 78
5.2.4. Transmission Parameters (ABCD Parameters) ........................................................................ 80
5.3. Relationships between two port Parameters................................................................................. 83
5.4. Transfer Functions Using Two-Port Parameters .......................................................................... 85
5.5. Interconnection of Two-Port Networks........................................................................................ 90
5.6. Elements of Transfer Functions Synthesis .................................................................................. 95
5.6.1. Properties of Transfer Immitance ......................................................................................... 96
5.6.2. Zeros of Transmission........................................................................................................... 98
5.6.3. Synthesis of Terminated Networks ....................................................................................... 99
Chapter 6 ............................................................................................................................................... 103
Topics in Filter Design.............................................................................................................................. 103
6.1. Introduction ................................................................................................................................ 103
6.1.1. The Filter Design Problem .................................................................................................. 103
6.1.2. The Approximation Problem in Network Theory ............................................................... 103
6.1.3. Time–domain approximation .............................................................................................. 104
6.1.4. Frequency-Domain Approximation .................................................................................... 105
6.3. Other Low-Pass Filter Approximations ......................................................................................... 109
6.3.1 The Chebyshev or Equal- Ripple Approximation ................................................................... 109
6.4. Monotonic filters with optimum cutoff.......................................................................................... 115
6.5. Linear phase filters ......................................................................................................................... 118
6.6. Transient Response of Low –Pass Filters ...................................................................................... 121
6.7 A Method of Reduce Overshoot in Filters ...................................................................................... 126
6.8 A Maximally Flat Delay and Controllable Magnitude Approximation .......................................... 128
6.9 Synthesis of Low –Pass Filters ....................................................................................................... 130
6.10 Magnitude and Frequency Normalization .................................................................................... 133
6.11 Frequency Transformations .......................................................................................................... 135
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Chapter 1
1. Introduction to Network Analysis & Synthesis
Network is a combination of electrical elements connected in any manner (conductively, inductively or

capacitively), whose impedance may be lumped or distributed or both. Further, it can be defined as a
collection of elements connected together to perform a certain task governed by a certain laws. Based on
the component from which network developed we have two classes: passive and active networks.
In generally accepted definitions of network analysis and synthesis, there are three key words: the
excitation, the network, the response as shown by fig. 1.1.
Excitation Response
Network
Fig.1.1. Key terms in network analysis & synthesis
A network in which the electrical properties are unaffected by interchanging input and output terminals is
called balanced network. In such networks, the elements are symmetrical with respect to ground
potential. To mention some examples of networks:
Filters − electrical networks which pass selective frequency signals.
Amplifiers/ Attenuators − electrical networks which are used to magnify or reduce signal power
level from input to output.
Equalizers − electrical networks used to counteract frequency or phase distortions.
Matching networks − electrical networks that match source and load impedances.
The study of network analysis and synthesis is sometimes referred to as network theory. Network theory
is based on fundamental electrical parameters. Network theory can be categorized into three classes
i. Mathematical modeling of electromagnetic phenomena
ii. Mathematical analysis of models of individual elements and systems
iii. Synthesis and design of systems & devices
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Modeling of a system should adequately describe the physical behavior of the system i.e. model is the
compromise between reality and theory (simplicity). Most electrical system analysis depends on two
fundamental theoretical concepts:
a. The lumped parameter circuit theory based on laws: KVL and KCL, ohms laws, etc…
b. Field theory based on Maxwell’s equations for networks composed of distributed
elements.
An electric network (amplifier, filter or equalizer circuits and the like) can be modeled using certain
lumped circuit elements. A basic problem faced by an engineer is to design a network model to satisfy
certain signal processing specifications and then fabricate physical components which approximate the
idealized elements. A prelude to design (and synthesis) is analysis, which is mainly an algebraic problem.
In this portion we discuss the characterization and modeling of lumped circuit elements. A network
consisting of lumped elements exhibits certain basic properties depending on the type of elements used
which themselves are classified according to their properties.
1.1. Lumped Circuit elements
A circuit element is said to be lumped if the instantaneous current entering one terminal is equal to the
instantaneous current leaving at the other terminals otherwise it is distributed elements. A television
antenna is not a lumped element as the current at the foot of the antenna is not the same as that at the tip
of the antenna. If the two were same there would be no radiation. The current through and the voltage
across a lumped element are well defined quantities and satisfy Kirchhoff’s laws.
If the physical dimension are small compared with the wave length of the highest signal frequency
applied to the network, then the element will be consider to be lumped. For example a frequency of
operation of 10 kHz (audio frequency) corresponds to a wavelength of 30km and any circuit built in a
laboratory has negligible physical dimension compared to this wavelength. If the frequency of operation
is 1GHz (microwave frequency) then the wave length is 0.3m and a circuits dimension becomes
comparable to the wavelength.
A large part of network theory deals with the study of lumped circuits. Lumped circuit is a result of
interconnection of lumped elements. Typical lumped elements are: resistors, capacitors, inductors, and
voltage and current sources.
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Resistors: An element which can be characterized by a curve in the V–I plane is called a resistor (V
represents voltage and I represent current). There are two types’ i.e. linear and non-linear resistors.
a. Linear resistors: I = GV or V=RI
I
I = GV or
slope=G=1/R
Fig.1.2. Linear resistor (a) characteristic curve (b) Circuit symbol
Another characterization of linear network is that the excitation and response of the network
are related by a linear differential equation.
b. Non linear resistors: i-v curve is not straight line; two nonlinear resistors are tunnel diode
and SCR (silicon controlled rectifier). The current through tunnel diode is single valued
function of the voltage, which makes it a voltage controlled resistor. This is depicted in fig.
1.3(a). In SCR, the voltage is single valued function of the current, which makes it act as the
current controlled resistor as shown in fig. 1.3(b). These nonlinear resistors V-I characteristic
have unique property of negative resistance in some ranges.
I
I=f(V)
Negative resistor
region
Fig.1.3. Non-linear characteristic curves
Evidently, nonlinear resistors cannot be characterized by a single valued resistance R, as in the case of
linear resistors. The resistance value of nonlinear resistor depends on the operating point and hence is
characterized by incremental resistance which is defined as the slope of the v-i characteristic at the
particular operating point. Nonlinear resistors find use in rectification, frequency multiplication, current
and voltage limiting, and many other electronic applications. Fig.1.4 shows the symbol and the v-i
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characteristic of a semiconductor diode (continuous curve). In analyzing a network with a nonlinear

element, such as a diode, one often resorts to piece-wise linearization. The piece-wise linear model of a
diode (broken line) is indicated in fig. 1.4.
Fig.1.4. Piece-wise linearization of characteristic curve
A distinct property of a linear resistor not usually possessed by a non-linear resistor is that the v-i
characteristic is symmetric with respect to the origin. An element exhibiting such a symmetric property is
called bilateral element. All linear circuits are bilateral but not all nonlinear circuits are bilateral.
Capacitors: An element which can be characterized by a curve in the v-q plane is called a capacitor (q
represents the charge). A capacitor is linear if its characteristic is a straight line passing through the origin
of the v-q plane.
Fig.1.5. Linear capacitor (a) characteristic curve (b) Circuit symbol

A linear capacitor is represented by the symbol shown above and is described analytically by q = CV
where the constant C is the slope of the straight line, and is called the capacitance. The unit of
capacitance is farad. (A farad is a very big unit for measuring capacitance; a more practical unit in a
network being µF -microfarad, or pF microfarads).
The current charge relation being i = dq dt , for a linear capacitor, we have
dv (t )
i=C
dt
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If a capacitor is characterized by a v-q curve other than a straight line through the origin, it is called a
nonlinear capacitor. Fig.1.6 shows the characteristic of a nonlinear capacitor (metal oxide semiconductor
capacitor). A nonlinear capacitor is characterized by q = f (v ) (for a voltage-controlled capacitor) and by
v = ψ (q ) (for a charge-controlled capacitor).
For a voltage controlled capacitor the current through the capacitor is

dq (t ) df (v ) dv (t )
i (t ) = = •
dt dv dt
dv (t )
Or i (t ) = C (v) where C (v ) = df (v ) / dv is called the incremental capacitance of the
dt
capacitor.
Fig.1.6. Non-linear capacitor characteristic curve
The instantaneous power entering an element is given by

p (t ) = v(t )i (t )
where v is the voltage in volts and i is the current in ampere. Hence, the energy in the element
from t 0 and t is
t
W (t 0 , t ) = ∫ v (τ )i (τ ) dτ
to
Let a capacitance be characterized by c = vˆ( q ) the
energy in the capacitor, from the above equations as
q (t )
t
W (t 0 , t ) = ∫ v(τ )i(τ )dτ = ∫ vˆ(q)dq
t0
q ( t0 )
If the initial charge on the capacitors, q(t 0 ), is zero, then the energy stored in the capacitor is
q (t )
W (t ) = ∫ vˆ(q)dq
0
For a linear capacitor (q = Cv ) it reduces to
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1 q 2 (t ) 1 2
W (t ) = = Cv (t )
2 C 2
Inductors: An element which can be characterized by a curve in the i − φ plane is called an inductor
(φ represents the flux). An inductor is called a linear inductor, if its characteristic is a straight line
passing through the origin of the i − φ plane as shown by fig.1.7.
Fig.1.7. Linear inductor (a) characteristic curve (b) Circuit symbol
A linear inductor is represented by the symbol shown in fig.1.7. (b) and described analytically by
φ = Li
where the constant L is the slope of the straight line and is called inductance. The unit of inductance is
Hennery. (Since inductance of one Henry is quite large, typical values of inductors are given mH- milli-
henrys.)
The voltage flux linkage relation being v = dφ / dt for a linear inductor, we have
di (t )
v=L
dt
If an inductor is characterized by a i − φ curve other than a straight line through the origin, then it is
called nonlinear inductor. A nonlinear inductor is characterized by
φ = f (i ) (For a current controlled inductor) and by i = ψ (φ ) (for a flux-controlled inductor) for a
current-controlled inductor the voltage across the inductor is
dφ (t ) df (i ) di (t )
v (t ) = = •
dt dt dt
di (t ) df (t )
Or v(t ) = L(i ) where L (i ) = is called the incremental inductance.
dt dt
Nonlinear inductors can be used in a frequency conversion, memory and storage.
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A special type of nonlinear inductor, such as a ferromagnetic-core inductor, has a characteristic that
exhibits the phenomenon of hysteresis.
Independent sources:
Fig.1.8 shows an arbitrary network N excited by a source of electric energy. If we change the network N,
then v and i in general also change. If the prescribed voltage is maintained then the source of energy is
called a voltage source. On the other hand if the prescribed current is maintained it is called current
source.
Fig.1.5. Independent source
A two terminal element is called an ideal independent voltage source, if it is capable of supplying any
current at the same prescribed voltage, i.e. the voltage across the source is independent of the current
drawn from the source as shown by fig.1.9. If the voltage of a voltage source is identically zero, the
voltage source is effectively a short circuit. The internal resistance of an ideal voltage source is
considered to be zero.
Fig.1.9. Voltage source:.symbol and the i-v characteristic curve
A two-terminal element is called an ideal independent current source, if it is capable of supplying a

prescribed current at any voltage, i.e. the current drawn from a current source is independent of the
voltage across the source. Fig.1.10. shows the symbol and the i-v characteristic of current source. If the
current drawn from a current source is identically zero, the current source is effectively an open circuit.
The internal shunt resistance of an ideal current source is considered to be infinite. In the physical world
there is no ideal independent current and voltage sources.
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Fig.1.10. Voltage source:.symbol and the i-v characteristic curve
A dependent voltage (or current) source is a source the voltage (or current) of which depends on another
voltage (or current). A dependent or controlled source is said to be voltage or current controlled if its
terminal behavior is controlled by another voltage or current. This leads to four different controlled
sources which are:
1. Voltage-controlled voltage source
2. Voltage-controlled current source
3. Current-controlled voltage source
4. Current-controlled current source
Some physical devices operate almost like ideal dependent sources. For example, an operational
amplifier is a voltage controlled voltage source, a field effect transistor a voltage controlled current
source.
1.2. Classification of Networks
The overall behavior of an electrical network can be predicated by the constituent element and their
interconnection. The behavior of the network, considered as a black box, leads to a number of
classifications like linear, nonlinear: time-invariant and time variant; passive, active.
1.2.1 Linear and Nonlinear Networks
In a linear network, the relationship between the voltage and current is described by a linear equation.
Consider two networks N1, and N2 as shown in fig.1.11(a) and (b) respectively. Network (a) is made up
of a linear resistor R, while (b) is made up of a semi-conductor diode and linear resistor R. let the cut-in
voltage of the diode be 0.6volt. In network (a) current I1 is given by V/R and exists for all values of V. In
network (b) if V is less than 0.6 volt the current I2 is zero and for voltages higher than 0.6volts I2 is given
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by (V-0.6)/(R+RF), where RF is the forward resistance of the diode. Obviously, in network (a) the current
response is linear in contrast to that in (b).
A system (network) is linear if (i) the principle of superposition and (ii) the principle of proportionality
hold. By the superposition principle, if, for a given network, [e1 (t ), r1 (t )] and [e2 (t ), r2 (t )] are excitation-
response pairs, then if the excitation were e(t ) = e1 (t ) + e2 (t ), the response would be r (t ) = r1 (t ) + r2 (t ) .
By the proportionality principle, if the excitation were C1e1 (t ), where C1 a constant is, then the response
would be C1 r1 (t ), i.e. the constant proportionality C1 is preserved by the linear network.
Let both the networks be excited by two serially connected voltage sources V1 and V2. Then it can be
seen that I 1 (V1 + V2 ) = I 1 (V1 ) + I 1 (V2 ) And I 2 (V1 + V2 ) ≠ I 2 (V1 ) + I 1 (V2 ) .
Fig.1.11. Linear and Non-linear networks

Where I i (V j ) is the current into the terminals of network N i when excited be a voltage source V j . We
say that network N1 is linear and N 2 is nonlinear as the principle of superposition holds for N1 but not
for N 2 .
Linearity of a network can be defined as follow: let a network be characterized by F ( xi ) = yi where xi is
the input and yi the output, and F (.) denotes some function. Then the network is linear if, and only if,
F (α 1 x1 + α 2 x 2 ) = α 1 F ( x1 ) + α 2 F ( x 2 ) = α 1 y1 + α 2 y 2
where α 1 and α 2 are arbitrary constants, and x1 and x 2 are any two allowable inputs. The principle
expressed by this equation is called the principle of superposition and homogeneity. Hence, we conclude
that a network is linear if it satisfies this principle; other-wise, it is nonlinear.
1.2.2 Time-invariant and Time-variant Networks
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Let a linear resistor be characterized by v(t ) = R (t )i (t ) where R (t ) is a prescribed time function. This can
be achieved, for example, by the sliding contact of potentiometer being moved back and forth by a motor.
Such a resistor is called a time-varying resistor. Similarly, it is possible to build time-varying capacitors
and inductors. The elements we considered previously were all time-invariant in that they were
characterized by parameters which were not dependent on time.
A time – invariant network is characterized by a constant coefficient equation whereas the time-variant
one by a time-variant coefficient equation. Mathematically, we can describe a time-invariant network by
F [x(t − t 0 )] = y(t − t 0 )
when the network is characterized by F [x(t )] = y (t ) i.e. the response (output) depends on the shape of the
excitation (input) but not on the time of application. A network composed of time-invariant elements is
necessarily time-invariant whereas network composed of time-variant elements may exhibit time-
invariant terminal behavior.
1.2.3 Passive and Active Networks
Consider a network made up of a single linear resistor. The energy supplied to (or dissipated by) the
resistor, from the previous relation is
t
W (t , t 0 ) = ∫ i 2 (t ) Rdt
to
If the resistor has to deliver power to the external world, R has to be negative. As long as R is positive the
resistor will consume power, and such a resistor is called a passive resistor.
Let v(t) and i(t) be the two voltage and current at the terminals of a network. Then the energy delivered to
the network is given by
t
W (t , t0 ) = ∫ v(τ )i(τ ) dτ
to
A network is said to be passive if, and only if, W (t , t 0 ) + E (t 0 ) ≥ 0 for all t and t 0 , and for all v(t )
and i (t ) , where E (t 0 ) is the energy in the network at t = t 0 . Otherwise, the network is said to be active.
In other words, if the energy delivered to the network is non-negative for all time and input, the network
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is said to be passive. The conditions for activity of an element can also be obtained by a study of its
characteristics. For example, we can state that a nonlinear resistor is passive if, and only if, its
characteristic, for all time, is in the first and third quadrants of v-i plane. Similarly, a capacitor (inductor)
is passive if, and only if, its characteristic is in the first and third quadrants of the v − q (φ − i ) plane.
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Chapter 2
Network Transform Representation and Analysis
2.1 Introduction
A concept of transform impedance and transform admittance is discussed in this chapter. Furthermore, a
function relating currents or voltages at a different parts of the network, called a transfer function, is
found to be mathematically similar to the transform impedance function. These functions are called
network functions.
2.1.1 Network Functions
In an electrical network, the word port has special meaning. It is a pair of terminals in which the current
into one terminal is equal to the current out of the other. The port at which the input/excitation is given is
called driving point or input port. The port at which output is taken is called output port. If the network
has only driving port, it is called one-port network and if the network has input and output ports, then it is
called two-port network. In fig.2.1. a general two-port network and a standard convention adopted in the
designation of voltages and currents are shown.
A one port network is completely specified when a voltage-current relationship at the terminals of the
port is given. The four variables (I1, I2, V1 and V2) of the two-port network, taken two at a time, results in
six set of equations that describe the two-port network. These equations are called network functions.
A network function is defined as the ratio of the zero – state response to the input, both the response and
the input expressed in Laplace domain, i.e.
(2.1)
where s the Laplace variable is the complex frequency variable .
Two sets of network functions can be defined: (1) Driving point functions (2) Transfer functions. These
are defined and illustrated by using the network shown in fig.2.1.
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Fig.2.1. Network used to define network functions

1. Driving Point Functions
If the excitation and the response are measured at the same set of terminals (terminal 1-1’, or 2-2’), the
network function is called driving point function. There are two driving point functions: impedance and
admittance.
(Driving point impedance)
(Driving point admittance) (2.2)
Because of the similarity of impedance and admittance, these two quantities are assigned one name
“Immitance” (a combination of impedance and admittance).
From the two relations, we can observe that the reciprocal of a driving point function is another driving
point function.
2. Transfer functions
If the excitation and response are measured at different sets of terminals, then the corresponding network
function is called a transfer function.
Transfer impedance
Transfer Admittance (2.3)
Transfer voltage ratio
Transfer current ratio
It can be observed from the definition of DP and transfer functions that for an impedance function,
excitation is a current source and response a voltage, and for an admittance function excitation a voltage
source and response a current. For simple networks DP function can be obtained by inspection. For
example, consider the ladder network shown in fig.2.2.
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Fig.2.2. Ladder network

(2.4)
i.e. the DP impedance of a ladder can be written in the form of a continued fraction expansion and such
an expansion is very useful in network analysis.
Exercise: Find the impedance and admittance for some simple networks containing,
a) Series b) Parallel
i. R – L i. R - L
ii. R – C ii. R - C
iii. L – C iii. L - C
iv. R – L – C iv. R – L - C
2.1. Poles and Zeros
Consider a linear time invariant network. Let x(t) be excitation and y(t) the response. Thus, in general, we
can express the relationship between x(t) and y(t) in the form of differential equation,
(2.5)
The coefficients in this differential equation depend on the elements and the topology of the network.
Laplace transforming and rearranging (with all initial conditions assumed to be zero) we have
, i.e., (2.6)
where p and q are polynomials in the complex frequency variable s. Thus we conclude that a network
function is a rational function of s with real coefficients. Alternatively, we can write equation (2.6) as
where (2.7)
Note: zi is called zero of H(s) and pi is called poles of H(s). The poles and zeros of a network need not be
distinct. The coefficients of the polynomials p(s) and q(s) being real any complex zeros and poles must
appear in conjugate pairs. From equation (2.7), it is clear that any network function is completely
specified by its poles and zeros and the scale factor k.
Exercise:
Consider the network shown in fig.2.3. Let x(t) be the input and y(t) (mesh current of the mesh II) be the
output. Find H(s).
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Fig.2.3.
The poles and zeros of a network function can be represented by a pole –zero plots. The pole – zero plot
of the network function is shown in fig.2.4.
Fig.2.4. pole-zero plot
If x(t) and y(t) are measured at the same set of terminals in equation (2.5), then H(s) would be a driving
point function. Without loss of generality, we assume that x(t) is current, and the response y(t) is voltage,.
Then
(2.8)
If we make the current x(t) go to zero, i.e., the network is being tested under open circuit conditions, then
the zero input voltage response is governed by the equation:
(2.9)
Hence the natural frequencies of the voltage response are determined by the roots of the equation,
(2.10)
Thus the roots of q(s) are the natural frequencies of the network under open circuit natural frequencies
(OCNF). On the other hand if we make the voltage y(t) go to zero, i.e., the network is being tested under
short circuit conditions, then the zero input current response is governed by the equation
(2.11)
And the natural frequencies of the current response are determined by the roots of the equation
(2.12)
Hence the roots of p(s) are called the short circuit natural frequencies (SCNF).
• Necessary conditions for driving point Immitance Functions (with common factors in N(s)
and D(s) cancelled):
1. The coefficients in the polynomials N(s) and D(s) must be real and positive.
2. Poles and zeros must be conjugate if imaginary or complex.
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3. The real part of all poles and zeros must be negative or zero, if the real part is zero, then that
pole or zero must be simple, i.e, all the roots of N(s) = 0 and D(s) = 0 lie on the left half of s-
plane and simple roots may lie on the imaginary or jw – axis.
4. The polynomials N(s) and D(s) may not have missing terms between those of highest and
lowest degrees, unless all even or all odd terms are missing.
5. The highest degree of N(s) and D(s) may differ by either zero or one only.
6. The lowest degree of N(s) and D(s) may differ by either zero or one only.
Exercise:
Check whether given functions are suitable in representing the driving point Immitance functions.
a)
b)
• Necessary conditions for transfer functions (with common factors in N(s) and D(s)
cancelled)
1. The coefficients in the polynomials N(s) and D(s) of T = N/D must be real and those for D(s)
must be positive.
2. Poles and zeros must be conjugate if complex.
3. The real part of poles must be negative or zero, if the real part is zero, then that pole must be
simple. This includes the origin.
4. The polynomial D(s) may not have any missing term between that of highest and lowest
degrees, unless all even or all odd terms are missing.
5. The polynomials N(s) may have terms missing, and some of the coefficients may be negative.
6. The degree of N(s) may be as small as zero independent of the degree of D(s).
7. (a) for G and α: the maximum degree of N(s) is equal to the degree of D(s).
(b) for Z and Y: the maximum degree of N(s) is equal to the degree of D(s) plus one.
Exercise
Check whether the given functions are suitable in representing the transfer functions.
a)
b)
2.2 Stability of a Network
Among the many forms of performance specifications used in design, the most important requirement is
that the system be stable. An unstable system is generally considered to be useless. For analysis and
design purposes, we can classify stability as absolute stability and relative stability. Absolute stability
refers to the condition of whether the system is stable or unstable; it is a yes or no answer. Once the
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system is found to be stable, it is of interest to determine how stable it is, and this degree of stability is a
measure of relative stability.
A system is said to be stable if its output (response) cannot be made to increase indefinitely by the
application of a bounded input excitation.
Relationship between impulse response and stability

The stability of a network function T(s) can be conveniently determined by considering its response to
an impulse function which is obtained by taking inverse laplace transform of the partial fraction
expansion of the function.
Stable system: A system is said to be stable if the impulse response approaches to zero for sufficiently
large time.
Ustable system: A system is said to be unstable if the impulse response grows without bound i.e,
approaches infinity for sufficiently large time.
Marginally stable system: A system is said to be marginally stable if the impulse response approaches a
constant non zero value or a constant amplitude oscillation for sufficiently large time.
Relationship between Poles positions and stability

The necessary and sufficient condition for the system to be stable is that all roots of characteristics
equation of the system lie in the negative half of the s–plane. In other words, we can say the poles of the
transfer function T(s) = N(s)/D(s) (i.e, roots of D(s) =0) lie in left half of the s–plane. On the other hand,
a system having any pole in the right half (or positive half) of the s–plane will be unstable.
Routh Hurwitz stability criterion

Routh Hurwitz stability criterion is method of determining number of roots of characteristic equation i.e.,
poles of transfer function with negative real parts, zero real parts and positive real parts and hence system
stability.
Consider that the characteristic equation of a system is of the form:
(2.13)
where all the coefficients ai are real. In order that equation (2.13) not have roots with positive real parts, it
is necessary that the following conditions hold:
1. All the coefficients of the equation have the same sign.
2. None of the coefficients vanishes, i.e, no missing term between that of highest and lowest
degrees.
Routh’s Tabulation
The first step in the Routh – Hurwitz criterion is to arrange the coefficients of the equation (2.13) into
two rows. The first row consists of the first, third, fifth…, coefficients and the second row consists of the
second, fourth, sixth,…, coefficients, all counting from the highest order term
an an-2 an-4 …
an-1 an-3 an-5 …
The next step is to form the array of numbers by the indicated operations
sn an an-2 an-4 …
19
sn-1 an-1 an-3 an-5 …

sn-2 bn bn-1 bn-2 …
sn-3 cn cn-1
. .
. .
. .
s1
s0
where
The array is called the Routh’s tabulation or Routh’s array. The last row of the Routh’s tabulation should
always be the s0 row.
The necessary and sufficient condition that all roots of equation (2.13) have negative real parts if all the
elements of the first column of the Routh’s tabulation are of the same sign. The number of the changes of
signs in the elements of the first column equals the number of roots with positive real parts.
Exercise:
1. Determine the stability of the systems whose characteristic equations are:
a) s4 +2s3+3s2+4s+5 = 0
b) s4+8s3+18s2+16s+5 = 0
2. Calculate the range of K for which the system given by characteristics equation S3+7s2+10s+10K
= 0 is stable.
Special cases when Routh’s tabulation terminates prematurely

Depending on the coefficients of the equation, sometimes the following difficulties may occur that
prevent Routh’s tabulation from completing properly:
1. The first element in any one row of Routh’s tabulation is zero, but the others are not.
2. All elements in one row of Routh’s tabulation are zero.
Case 1: If a zero appears in the first element of a row, the elements in the next row will all become
infinite, and Routh’s tabulation cannot be continued. To remedy this situation, we replace the zero
elements in the first column by an arbitrary small positive number ε, and then proceed with Routh’s
tabulation.
20
Exercise:
Check the stability of the system with characteristic equation 2s5+s4+6s3+3s2+s+1=0.
Case 2: If all the elements in one row of Routh’s tabulation are zero before the tabulation is properly
terminated, it indicates that one or more of the following conditions may exist:
1. The equation has at least one pair of real roots with equal magnitude but opposite signs.
2. The equation has one or more pairs of imaginary roots with equal magnitude opposite signs.
(e.g., s = ± j1, s= ±j2, ±j3).
3. The equation has pairs of complex conjugate roots forming symmetry about the origin of the
s–plane (e.g., s = -1 ± j1, s = 1 ± j1).
The situation with the entire row of zeros can be remedied by using the auxiliary equation A(s) = 0,
which is formed from the coefficients of the row just above the row of zeros in Routh’s tabulation. The
auxiliary equation is always even polynomials. The roots of the auxiliary equation also satisfy the
original equation. Thus, by solving the auxiliary equation, we also get some of the roots of the original
equation. To continue with Routh’s tabulation when a row of zeros appears, we conduct the following
steps:
1. Form the auxiliary equation A(s) = 0 by use of the coefficients from the row just preceding the
row of zeros.
2. Take the derivative of the auxiliary equation with respect to s; this gives = 0.
3. Replace the row of zeros with coefficients of = 0.

4. Continue with Routh’s tabulation in the usual manner with the newly formed row of
coefficients replacing the row of zeros.
5. Interpret the change of signs, if any, of the coefficients in the first column of Routh’s
tabulation in the usual manner.
Exercise: check the stability of the system with characteristic equation S5 + 4s4+8s3 + 8s2+7s+4 = 0.
Deficiencies of Routh Hurwitz Criterion

1. It is valid only if the characteristic equation is algebraic with real coefficients. If any of the
coefficients is complex, or if the equation is not algebraic, such as containing exponential or
sinusoidal functions of s, the Routh Hurwitz criterion is simply cannot be applied.
2. It gives an information about roots of characteristic equation only with respect to the left half
or right half of the s plane i.e., it doesn’t give the information about the roots on the jw axis
(stability boundary).
3. It cannot be applied to discrete time systems.
Exercise: Determine the range of K so that the system given by characteristic equation
S3+3Ks2+(K+2)s+4 = 0 is stable.
21
Tutorial Exercises
1. Obtain the pole zero location for the function
2. Calculate the driving point impedances of the networks shown in the fig.2.5 and plot their pole
zero diagrams.
Fig 2.5
3. The pole zero plot of a voltage transfer function is shown in fig.2.6. Find the transfer function if
the gain is to be 10.
Fig.2.6
4. Fig.2.7shows an infinite resistive ladder. Find the input resistance of the ladder.
Fig.2.7
5. Determine the stability of the systems whose characteristic equations are given by
22
a)
b)
6. Using Routh’s Hurwitz criterion, determine the number of roots in the right half of s–plane for the
characteristic equations:
a)
b)
7. Determine the range of K for which the systems given by the following characteristic equations
are stable:
a)
b)
23
Chapter 3
3 Elements of Realizability Theory

3.1 Causality and Stability
In the preceding chapters we have been primarily concerned with the problem of determining the
response, given the excitation and he network; this problem lies in the domain of network analysis. The
starting point for any synthesis problem is the system function
R (s )
H (s ) = (3.1)
E ( s)
Our task is to synthesize a network from a given system function. The first step in a synthesis procedure
is to determine whether H (s ) can be realized as a physical passive network. There are two important
considerations: causality and stability. By causality we mean that the voltage cannot appear between any
pair of terminals in the network before a current is impressed, or vice versa. In other words, the impulse
response of the network must be zero for t < 0 , that is, h (t ) = 0 for t < 0 . As an example, the
−a t
impulse response h (t ) = e − at u (t ) is causal, where as h(t ) = e is not causal.
In certain cases, the impulse response could be made realizable (causal) by delaying it appropriately. For
example, the impulse response in fig. 3.1(a) is not realizable. If we delay the response by T seconds, we
find that the delayed response h (t − T ) as shown in fig. 3.1(b) is realizable.
(a) (b)
Fig.3.1: (a) Non-realizable impulse response. (b) Realizable impulse response
24
In the frequency domain, causality is implied when the paley-wiener criterion is satisfied for the
amplitude function H ( jω ) . The Paley-wiener criterion states that a necessary and sufficient condition
for an amplitude function H ( jω ) to be realizable (causal) is that
∞ log H ( jω )
∫ dω < ∞ (3.3)
−∞ 1+ ω2
The following conditions must be satisfied before the Paley-Wiener criterion is valid:
1. h(t ) must possess a Fourier transform H ( jω ) .
2. The square magnitude function H ( jω ) must be integral that is

2
∞
∫ H ( jω ) dω < ∞
2
−∞
(3.4)
The physical implication of the Paley-Wiener criterion is that the amplitude H ( jω ) of a realizable
network must be zero over a finite band of frequencies. Another way of looking at the Paley-wiener
criterion is that the amplitude function cannot fall off to zero faster than exponential order. For example,
the ideal low pass filter in fig. 3.2 is not realizable because beyond ωC the amplitude is drops to zero.
Fig. 3.2: Ideal filter characteristic
The Gaussian shaped curve H ( jω ) = e −ω shown by fig. 3.3 is not realizable because
2
∞ ω2
log H ( jω ) = ω and the integral
2
∫−∞ 1 + ω 2 dω is not finite.
25
Fig 3.3: Gaussian Filter charactersitc

On the other hand, the integral, the amplitude function
1
H ( jω ) = (3.5)
1+ ω2
does represent a realizable network.
In fact the voltage ratio transfer function of the R-C network in the figure below has an amplitude
characteristic given H ( jω ) by equation (3.5).
For the ideal filter, the inverse transforms h(t ) has the form
A0 sin ω C t
h(t ) = (3.6)
πt
sin x
where A0 is constant. From the curve, we can observe that h(t ) is nonzero for t less than zero. In
x
fact, in order to make h(t ) causal, it must be delayed by an infinite amount. In practice, however, if we
delay h(t ) by a large but finite amount of t d such that for t < 0 the magnitude of h(t − t d ) is less than a
very small quantity ε , that is h(t − t d ) < ε t < 0 .
We then can approximate h(t − t d ) by a causal response h1 (t ) which is zero for t < 0 . For a more detail
discussion of the Paley-Wiener criterion, then you have to refer to other materials focusing to this issue.
If a network is stable, then for a bounded excitation e(t ) the response r (t ) is also bounded. In other words,
26
if e(t ) < C1 for 0 ≤ t < ∞ then r (t ) < C 2 for 0 ≤ t < ∞ where C1 and C2 are real, positive, finite
quantities. If a linear system is stable, then from the convolution integral we obtain
∞
r (t ) < C1 ∫ h(τ ) dτ < C2 (3.7 )
0
Equation (3.7 ) requires that the impulse response be absolutely integrable, or

∞
∫ h(τ ) dτ < ∞ (3.8)
0
One important requirement for h(t ) to be absolutely integrable is that the impulse response approach
zero as t approaches infinity, that is,
lim h(t ) → 0
t →∞
(3.9)
Generally, it can be said that with the exception of isolated impulses, the impulse response must be
bounded for all t, i.e., h(t ) < C for all t where C is real, positive, finte number.
Observe that our defination of stability precludes such terms as sin ω 0 t from the impulse response
because sin ω 0 t is not absolutely integrable. These undammed sinusoidal terms are associated with
simple poles on the jω axis. Since pure L − C network have system functions with simple poles on the
jω - axis, and since we do not wish to all these networks unstable, we say that a system is marginally
stable if its impulse response is bounded according to (3.8) , but does not approach zero as t approaches
infinity.
In the frequency domain, the stability criterion requires that the system function possess poles in the left-
half plane or on the jω axis only. Moreover, the poles on the jω axis, if H (s ) is given as
a n s n + a n −1 s n −1 + a n − 2 s n − 2 + .... + a1 s + a 0
H ( s) = (3.10)
bm s m + bm −1 s m −1 + bm − 2 s m − 2 + .... + b1 s + b0
then the order of the numerator n cannot exceed the order of the denominator m by more than unity, that
is n − m ≤ 1 . If n exceeded m by more than unity, this would imply that s = jω = ∞ , and there would be a
multiple pole. To summarize, in order for a network to be stable, the following three conditions on its
system function H (s ) must be satisfied:
1. H (s ) cannot have poles in the right-half plane.
27
2. H (s ) cannot have multiple poles on the jω − axis.
3. The degree of numerator of H (s ) cannot exceed the degree of the denominator by more than
unity.
Finally, it should be pointed out that a rational function H (s ) with poles in the left half plane only has an
inverse transform h(t ) , which is zero for t < 0 . In this respect, stability implies causality. Since system
functions of passive linear networks with lumped elements are rational functions with poles in the left-
half plane or jω axis only, causality ceases to be a problem when we deal with the system functions of
this type. We are only concerned with the problem of causality when we have to design a filter for given
amplitude characteristic such as the ideal filter. We know we could never hope to realize exactly a filter
of this type because the impulse response would not be causal.
3.2. Hurwitz Polynomials

In section 3.1 we saw that in order for a system function be stable, its poles must be restricted to the left-
half plane or on the jω -axis. Moreover, the pole on the jω -axis must be simple. The denominator
polynomial of the system function belongs to a class of polynomials known as Hurwitz polynomials. A
polynomial P (s ) is said o be Hurwitz if the following conditions are satisfied:
a. P (s ) is real when s is real
b. The roots of P (s ) have real parts which are zero or negative.
As a result of this conditions, if P (s ) is a Hurwitz polynomial given by
P ( s ) = a n s n + an−1s n−1 + ⋅ ⋅ ⋅ + a1s + a0 (3.11)

then all the coefficients a i must be real and if s i = α i + jβ is the root of P (s ) , then α i must be negative.
The polynomial P ( s ) = ( s + 1)( s + 1 + j 2 )( s + 1 − j 2 ) is Hurwitz because all of its roots
have negative real parts. On the other hand, G ( s ) = ( s − 1)( s + 2)( s + 3) is not Hurwitz because of
the root s=1, which has positive real part.
Hurwitz polynomials have the following properties:
28
i. All the coefficients ai are non negative. This is readily seen by examining the three types of
roots that a Hurwitz polynomial might have. These are s = −γ i , s = ± jω i and Real,
s = −α i ± jβ i where γ i , α i & ωi real and positive.
The polynomial P (s ) which contains these roots can be written as

P ( s ) = ( s + γ i )( s 2 + ωi2 )[( s + α i ) 2 + β i2 ] ⋅ ⋅ ⋅ (3.12)
Since P (s ) is the product of terms with only positive coefficients, it follows that the
coefficients of P (s ) must be positive. A corollary is that between the highest order terms in
s and the lowest order term, none of the coefficients may be zero unless the polynomial is
even or odd. In other words, a n−1 , a n−2 ,⋅ ⋅ ⋅, a 2 , a1 must not be zero if the polynomial is
neither even nor odd. This is readily seen because the absence of term ai implies cancellation
brought about by a root s − γ i with a positive real part.

ii. Both the odd and even parts of a Hurwitz polynomial P (s ) have roots on the jω -axis only.
If we denote the odd part of P (s ) as n(s) and the even part as m(s ) , so that
P ( s ) = n ( s ) + m( s ) then m(s ) and n(s) both have roots on the jω -axis only. Refer
books for the proof.
iii. As a result of property (ii) above, if P (s ) is either even or odd, all its roots are on the jω -
axis.
The continued fraction expansion of the ratio of the odd to even parts or the even to odd parts of a
Hurwitz polynomial yields all positive quotient terms. Suppose we denote the ratios as
ψ ( s ) = n( s ) / m( s ) or ψ ( s ) = m( s ) / n( s ) , then the continued fraction expansion of ψ (s ) can be
written as
ψ (s ) = q1s +
1
1
q2 s +
1 (3.13)
q3 s +
1
q4 s +
.... 1
+
qn s 29
where the quotients q1, q2 , ..., qn must be positive if the polynomial P ( s ) = n( s ) + m( s ) is Hurwitz.
To obtain the continued fraction expansion, we must perform a series of long divisions. Suppose ψ (s ) is
ψ (s ) =
m( s )
where m (s ) is one higher degree than n(s ) . Then if we divide n(s ) into m (s ) , we obtain a
n( s )
single quotient and a remainder
R1 (s )
ψ ( s) = q1 s + (3.14)
n (s )
The degree of the term R1 (s ) is one lower than the degree of n(s ) . Therefore if we invert the remainder
term and divide, we have
n (s ) R (s )
= q2 s + 2 (3.15)
R1 (s ) R1 (s )
Inverting and dividing again, we obtain
R1 (s ) R (s )
= q3 s + 3 (3.16)
R2 (s ) R2 (s )
We see that the process of obtaining the continued fraction expansion of ψ (s ) simply involves division
and inversion. At each step we obtain quotient term q i s and a remainder term, Ri+1 ( s) / Ri ( s) . We then
invert the remainder term and divide Ri+1 ( s) into Ri (s) to obtain a new quotient. There is a theorem in the
theory of continued fractions which states that the continued fraction expansion of the even to odd or odd
to even parts of a polynomial must be finite in length. Another theorem states that, if the continued
fraction expansion of the odd to even or even to odd parts of polynomial yields positive quotient terms,
then the polynomial must be Hurwitz to within a multiplicative factor W (s ) . That is, if we write
F ( s) = W ( s) F1 ( s) (3.17 )
Then F (s ) is Hurwitz, if W (s ) and F1 ( s) are Hurwitz.
30
For example, let us test whether the polynomial F ( s) = s 4 + s 3 + 5s 2 + 3s + 4 is Hurwitz or not.
The even and the odd parts of F (s ) are m( s) = s 4 + 5s 2 + 4 and n( s) = s 3 + 3s .

We now perform a continued fraction expansion of ψ ( s ) = m( s ) / n( s ) by dividing n(s ) by m (s ) , and then
inverting and dividing again, as given by the operation
s 3 + 3s s 4 + 5 s 2 + 4 ( s
s 4 + 3s 2
2 s 2 + 4 s 3 + 3s ( s / 2
s 3 + 2s
s 2s 2 + 4 (2s
2s 2
4) s ( s / 4
s
So that the continued fraction expansion of ψ (s ) is

m( s ) 1
ψ ( s) = =s+
n( s ) s 1
+
2 1
2s +
s/4
Since all the quotient terms of the continued fraction expansion are positive, F (s ) is Hurwitz.
Example 1: Let us test whether the polynomial G ( s ) = s + 2 s + 3s + 6 is Hurwitz. The continued

3 2
fraction expansion of n ( s ) / m ( s ) is obtained from the division
2 s 2 + 6 s 3 + 3s (s / 2
s 3 + 3s
0
We see that the division has been terminated abruptly by a common factor s 3 + 3s . The polynomial can
then be written as
( )
 2
G ( s ) = s 3 + 3s  1 + 
 s
We know that the term 1+ 2 / s is Hurwitz. Since the multiplicative factor s 3 + 3s is also Hurwitz, then
G (s ) is Hurwitz. The term s 3 + 3s is the multiplicative factor W (s ) , which we referred to earlier.
Example 2: Next consider a case where W (s ) is non-Hurwitz
31
S 7 + 2 s 6 + 2 s 5 + s 4 + 4 s 3 + 8s 2 + 8 s + 4
The continued fraction expansion of F (s ) is now obtained.
n( s ) s 1
= +
m( s ) 2 4 1
s+
3 3 4
2
s s +4 ( )
s4 + 4 ( )
We thus see that W ( s ) = s + 4 , which can be factored into
4
( )(
W (s) = s 2 + 2s + 2 s 2 − 2s + 2 ).
It is clear that F (s ) is not Hurwitz since W(s) is not Hurwitz.
Example 3: Let us consider a more obvious non-Hurwitz polynomial
F ( s ) = s 4 + s 3 + 2 s 2 + 3s + 2
The continued fraction expansion is
s 3 + 3s s 4 + 2 s 2 + 2 ( s
s 4 + 3s 2
− s 2 + 2 )s 3 + 3s (− s
s 3 − 2s
5s ) − s 2 + 2(− s / 5
− s2
2 )5s
5
s
2
5s
We see that F (s ) is not Hurwitz because of the negative quotients.

Example 4: Consider the case where F (s ) is an odd or even function. It is impossible to perform a
continued fraction expansion on the function as it stands. However, we can test the ratio of F (s ) to its
derivative, F ' ( s ) . If the ratio F ( s ) / F ' ( s ) gives a continued fraction expansion with all positive
coefficients, then F (s ) is Hurwitz. For example, if F (s ) is given as
F ( s) = s 7 + 3s 5 + 2s 3 + s Then F ' ( s ) is F ' ( s ) = 7 s 6 + 15 s 4 + 6 s 2 + 1 without going into

the details, it can be shown that the continued fraction expansion of F ( s ) / F ' ( s ) does not yield all
positive quotients. Therefore F (s ) is not Hurwitz.
32
3.3. Positive Real Functions

In this section we will study the properties of a class of functions known as positive real functions. These
functions are important because they represent physically realizable passive driving-point Immitance. A
function F (s ) is positive real (p.r.) if the following conditions are satisfied:
i. F (s ) is real for real s ; that is, F (σ ) is real.
ii. The real part of F (s ) is greater than or equal to zero when the real part of s is greater than or
equal to zero, that is, Re[ F ( s )] ≥ 0 , for Re s ≥ 0 .
Let us consider a complex plane interpretation of a p.r. function. Consider the s -plane and F (s ) plane in
Fig. 3.4. If F (s ) is p.r., then a point σ 0 on the positive real axis of the s -plane would correspond to, or
map into, a point F (σ 0 ) which must be on the positive real axis of the F (s ) plane. In addition, a point
s i in the right half of the s plane would map onto a point F ( si ) in the right-half of the F (s ) plane. In
other words, for a positive real function, the right half of the s -plane maps onto the right half of the
F (s ) -plane. The real axis of the s plane maps onto the real axis of the F (s ) plane.
Fig. 3.4: Mapping of s -plane onto F (s ) -plane

A further restriction we will impose is that F (s ) be rational. Consider the following examples of p.r.
functions:
i. F ( s ) = Ls (where L is a real, positive number) is p.r. by definition. If F (s ) is an impedance
function, then L is an inductance.
ii. F ( s ) = R (where R is a real and positive) is p.r. by definition. If F (s ) is an impedance function,

then R is a resistance.
iii. F ( s ) = K / s (where K is a real and positive) is p.r. because, when s is real, F (s ) is real. In
addition, when the real part of s is greater than zero i.e., Re( s ) = σ > 0 , then
33
K Kσ
Re  = 2 > 0 . Therefore, F (s ) is p.r. If F (s ) is an impedance function, then the
 s  σ +ω
2
corresponding element is a capacitor of 1/K farads.
We thus see that the basic passive impedances are p.r. functions. Similarly, it is clear that the admittances
are positive real.
We now show that all driving point immittances of passive networks must be p.r. The proof depends
upon the following assertion: for a sinusoidal input, the average power dissipated by a passive network is
nonnegative. For the passive network in the fig. 3.5, the average power dissipated by the network is
1
Re[ Z in ( jω )] I ≥ 0 . We then conclude that, for any passive network
2
2
Re[Z in ( jω )] ≥ 0 (3.18)
We can now prove that for Re s = σ ≥ 0 Re Z in (σ + jω ) ≥ 0 . Consider the network in fig. 3.5, whose
driving point impedance is Z in (s ) . Let us load the network with incidental dissipation such that if the
driving-point impedance of the uniformly loaded network is Z i (s ) , then
Z i ( s ) = Z in ( s + α ) (3.19)
where α , the dissipation constant, is real and positive.
Fig. 3.5: Passive one-port network
Since Z i (s ) is the impedance of a passive network, Re Z i ( jω ) ≥ 0 So that: Re Z in (α + jω ) ≥ 0

Since α is an arbitrary real positive quantity, it can be taken to be σ . Thus the theorem is proved.
Next let us consider some useful properties of p.r. functions.
34
1
1. If F (s ) is p.r. then is also p.r. This property implies that if a driving point impedance is p.r.,
F ( s)
then its reciprocal, the driving point admittance is also p.r.
2. The sum of p.r. function is p.r. from an impedance standpoint; we see that if two impedances are
connected in series, the sum of the impedances is p.r. An analogous situation holds for two
admittances in parallel. Note that the difference of two p.r. functions is not necessarily p.r.; for
example, F ( s ) = s − 1 / s is not p.r.
3. The poles and zeros of a p.r. function cannot have positive real parts, i.e., they cannot be in the
right half of s plane.
4. Only simple poles with real positive residues can exist on the jω -axis.
5. The poles and zeros of a p.r. function are real or occur in conjugate pairs. We know that the poles
and zeros of a network function are functions of the elements in the network. Since the elements
themselves are real, there cannot be complex poles or zeros without conjugates because this
would imply imaginary elements.
6. The highest powers of the numerator and denominator polynomials may differ at most by unity.
This condition prohibits multiple poles and zeros at s = ∞ .
7. The lowest powers of the denominator and numerator polynomials may differ by at most by unity.
This condition prohibits the possibility of multiple poles or zeros at s = 0 .
8. The necessary and sufficient conditions for a rational function with real coefficients F (s ) to be
p.r. are
a. F (s ) must have no poles in the right-half plane.
b. F (s ) may have only simple poles on the jω -axis with real and positive residues.
c. Re[ F ( jω )] ≥ 0 for all ω .
Let us compare this new definition with the original one which requires the two conditions:
i. F (s ) is real when s is real.
35
ii. Re[ F ( s )] ≥ 0 when Re[ s ] ≥ 0
In order to test condition (ii ) of the original definition, we must test every single point in the right-half
plane. In the alternate definition, condition (c ) merely requires that we test the behavior of F (s) along the
jω axis. It is apparent that testing a function for the three conditions given by the alternate definition
represents a considerable saving of effort, except in simple cases as F ( s ) = 1 / s .
Let us examine the implications of each criterion of the second definition. Condition (a ) requires that we
test the denominator of F (s ) for roots in the right-half plane, i.e., we must determine whether the
denominator of F (s ) is Hurwitz. This is readily accomplished through a continued fraction expansion of
the odd to even or even to odd parts of the denominator. The second requirements- condition (b ) is tested
by making a partial fraction expansion of F (s ) and checking whether the residues of the poles on the
jω -axis are positive and real. Thus, if F (s ) has a pair of poles at s = ± jω , a partial fraction expansion
*
K1 K1
gives terms of the form + .
s − jω1 s + jω1
The residues of complex conjugate poles are themselves conjugates. If the residues are real-as they must
*
K1 K1 K s
be in order for F (s ) to be p.r. then K1 = K1 so that
* + = 2 2 2.
s − jω1 s + jω1 s + ω1
If K1 is found to be positive, then F (s ) satisfies the second of the three conditions.
In order to test for the third condition for positive realness, we must first find the real part of F ( jω ) from
the original function F (s ) . To do this, let us consider a function F (s ) given as a quotient of two
polynomials
F ( s) =
P(s)
(3.20)
Q( s )
We can separate the even parts from the odd parts of P(s) and Q (s ) so that F (s ) is
M 1 (s) + N1 ( s)
F (s) = (3.21)
M 2 (s) + N 2 ( s)
Where M 1 ( s ) is an even function and N1 ( s) is an odd function. F (s ) is now decomposed into its even
and odd parts by multiplying both P(s) and Q (s ) by M 2 − N 2 so that
36
M 1 + N1 M 1 − N1 M 1 M 2 − N1 N 2 M 2 N 2 − M 1 N 2
F (s) = = + (3.22)
M 2 + N2 M 2 − N2 M 2 − N2
2 2
M 2 − N2
2 2
We see that the products M 1 M 2 and N1 N 2 are even functions, while M 1 N 2 and M 2 N1 are odd
functions. Therefore, the even parts of F (s ) is
M 1 M 2 − N1 N 2
Ev[F (s)] = (3.23)
M 2 − N2
2 2
And the odd part of F (s ) is
M 2 N2 − M1N 2
Odd [F (s)] = (3.24)
M 2 − N2
2 2
If we let s = jω , we see that the even part of any polynomial is real, while the odd part of the polynomial
is imaginary, so that if F ( jω ) is written as F ( jω ) = Re[F ( jω )] + j Im[ F ( jω )] , it is clear that
Re[ F ( jω )] = Ev[ F ( s )] s = jω and j Im[ F ( jω )] = Odd [ F ( s )] s = jω .
Therefore, to test for the third condition for positive realness, we determine the real part of F ( jω ) by
finding the even part of F (s ) and then letting s = jω . We then check to see whether Re F ( jω ) ≥ 0 for
all ω .
The denominator of Re[ F ( jω )] is always a positive quantity because
M 2 ( jω ) − N 2 ( jω ) = M 2 (ω ) + N 2 (ω ) ≥ 0 .
2 2 2 2
That is, there is an extra j or imaginary term in N 2 ( jω ) , which, when squared, gives -1, so that the
denominator of Re F ( jω ) is the sum of two squared numbers and is always positive. Therefore, our task
resolves into the problem of determining whether A(ω ) = M 1 ( jω ) M 2 ( jω ) − N1 ( jω ) N 2 ( jω ) ≥ 0
Note that A(ω ) must not have positive real roots of the type shown in fig. 3.5(a); i.e., A(ω ) must never
have single real roots of ω . However, A(ω ) may have double roots as shown in fig. 3.5(b), because A(ω )
need not become negative in this case.
(a) b)
37
Fig. 3.5 A(ω ) can have a double root

As an example, consider the requirements for
s+a
F ( s) =
s + bs + c
2
To be p.r. first, we know that, in order for the poles and zeros to be in the left-half plane or on the jω -
axis, the coefficients a, b, c must be greater or equal to zero. Second, if b = 0 then F (s ) will possess poles
on the jω axis we can then write F (s ) as
s a
F (s) = + 2
s +c s +c
2
We will show later that the coefficient a must also be zeros when b = 0 . Let us proceed with the third
requirement, i.e., Re[ F ( jω )] ≥ 0 . From the equation M 1 ( jω ) M 2 ( jω ) − N1 ( jω ) N 2 ( jω ) ≥ 0
We have a(−ω 2 + c) + bω 2 ≥ 0 which simplifies to A( jω ) = (b − a)ω 2 + ac ≥ 0 .

It is evident that in order to prevent A(ω ) from having positive real roots of ω , b must be greater than or
equal to a , that is b ≥ a . As a result, if b = 0 , then a = 0 . To summarize, the conditions that must be
fulfilled in order for F (s ) to be real are
1. a , b, c ≥ 0
2. b > a
s+2 s +1 s+4
We see that F ( s ) = is p.r., while the functions F ( s ) = 2 & F ( s) = are not
s + 3s + 2
2
s +2 s + 2s + 1
2
p.r.
As a second example, let us determine the conditions for the biquadrate function
s 2 + a1 s + a 0
F (s) = 2
s + b1 s + b0
To be p.r. we will assume that the coefficients a1 , a0 , b1 , b0 are all real positive constants. Let us test
whether F (s ) is p.r. by testing each requirement of the second definition.
First, if the coefficients of the denominator b1 and b0 are positive, the denominator must be Hurwitz.
Second, if b1 is positive, we have no poles on the jω -axis. Therefore we can ignore the second condition.
38
The third condition can be checked by first finding the even part of F (s ) which is
( s 2 + a 0 )( s 2 + b0 ) − a1b1 s 2 s 4 + [(a 0 + b0 ) − a1b1 ]s 2 + a 0 b0

Ev[ F ( s )] = =
( s 2 + b0 ) 2 − b1 s 2
2
(s 2
+ b0 )
2
− b1 s 2
2
The real part of F ( jω ) is then
ω 4 − [(a 0 + b0 ) − a1b1 ]ω 2 + a 0 b0
Re[ F ( jω )] =
( −ω 2 + b0 ) 2 + b1 ω 2
2
We see that denominator of Re[ F ( jω )] is truly always positive so it remains for us to determine whether
the numerator of Re[ F ( jω )] ever goes negative. Factoring the numerator, we obtain
(a 0 + b0 ) − a1b1 1
ω2 1, 2 = ± [(a 0 + b0 ) − a1b1 ] 2 − 4a 0 b0
2 2
There are two situations in which Re[ F ( jω )] does not have a simple real root.
1. When the quantity under the radical sign of equation above is zero (double, real root) or negative
(complex roots). In other words,
[(a 0 + b0 ) − a1b1 ] 2 − 4a0 b0 ≤ 0 Or [(a 0 + b0 ) − a1b1 ] 2 ≤ 4a 0 b0
If (a0 + b0 ) − a1b1 ≥ 0 , then (a 0 + b0 ) − a1b1 ≤ 2 a 0 b0 or a1b1 ≥ ( a 0 − b0 ) 2
If (a0 + b0 ) − a1b1 < 0 , then a1b1 − (a0 + b0 ) ≤ 2 a 0 b0
but (a0 + b0 ) − a1b1 < 0 < a1b1 − (a0 + b0 )
So again a1b1 ≥ ( a 0 − b0 )
2
2. The second situation in which Re[ F ( jω )] does not have a simple real root is when ω 2 1, 2 in
equation above is negative so that the roots are imaginary. This situation occurs when
[(a 0 + b0 ) − a1b1 ]2 − 4a 0 b0 > 0 and (a0 + b0 ) − a1b1 < 0

From the above equation we have
a1b1 − (a 0 + b0 ) > 2 a 0 b0 > (a0 + b0 ) − a1b1 Thus a1b1 > ( a0 − b0 ) 2
We thus see some of the above equation is a necessary and sufficient condition for a biquadrate function
to be positive real. If we have a1b1 = ( a 0 − b0 ) 2 , then we will have double zeros for Re[F ( jω )] .
39
s 2 + 2s + 25
Consider the following example; F ( s) = we see that
s 2 + 5s + 16
a1b1 = 2 × 5 = 10 ≥ ( a 0 − b0 ) 2 = ( 25 − 16 ) 2 = 1
So that F (s ) is p.r. The examples just given are, of course, special cases. But they do illustrate the
procedure by which functions are tested for the p.r. property. Let us consider a number of other helpful
points by which a function might be tested quickly. First, if F (s ) has poles on the jω -axis, a partial
fraction expansion will show if the residues of these poles are positive and real.
3s + 5
For example, F ( s ) = has a pair of poles at s = ± j1 . The partial fraction expansion of
(
s s2 +1 )
− 2s 5
F (s ) , F ( s ) = +
s2 +1 s
shows that the residue of the poles at s = ± j1 is negative. Therefore F (s ) is not p.r.
Since impedances and admittances of passive time-invariant networks are p.r. functions, we can make use
of our knowledge of impedances connected in series or parallel in our testing for the p.r. property. For
example, if Z 1 ( s ) and Z 2 ( s) are passive impedances, then Z 1 connected in parallel with Z 2 gives an
overall impedance
Z1 (s )Z 2 (s)
Z (s) = (3.25)
Z 1 ( s) + Z 2 (s)
Since the connecting of the two impedances in parallel has not affected the passivity of the network, we
know that Z (s ) must also be p.r. We see that if F1 ( s) and F2 ( s) are p.r. functions, then
F1 ( s ) F2 ( s )
F (s) = must also be p.r. Consequently, the functions
F1 ( s ) + F2 ( s )
Ks K
F ( s) = and F ( s ) = where α and K are real and positive quantities, must be p.r. We then
s +α s +α
observe that functions of the type
s+β s β
F ( s) = = + α , β ≥ 0 must be p.r. also.
s +α s +α s +α
Ks
Finally, let us determine whether F ( s ) = α , K ≥ 0 is p.r.
s +α
2
If we write F (s ) as
40
1
F (s) =
s / K + α / Ks
we see that the terms s / K and α / Ks are p.r. Therefore, the sum of the two terms must be p.r. Since the
reciprocal of a p.r. function is also p.r. we conclude that F (s ) is p.r.
Tutorial Exercise
1. Test the following polynomials for the Hurwitz property.
a. s 3 + s 2 + 2s + 2 b. s4 + s2 + s +1
c. s7 + s5 + s3 + s d. s 3 + 4 s 2 + 5s + 2
e. s 5 + 2s 3 + s f. s + 2s + 2s + s + 4s + 8s + 8s + 4
7 6 5 4 3 2
2. Determine whether the following functions are p.r. For the following with the denominator
already factored, perform a partial fraction expansion first.
s2 +1
a. F ( s ) = 3 c. F ( s) =
(s + 2)(s + 4)
s + 4s ( s + 1)(s + 3)
2s 2 + 2s + 4 s2 + 4 5s 2 + s
b. F ( s ) = d. F ( s) = 3 e. F ( s ) = 2
(
(s + 1) s 2 + 2 ) s + 3s 2 + 3s + 1 s +1
s 2 + Xs
3. Given Z ( s) = ;
s 2 + 5s + 4
a. What are the restrictions on X for Z (s ) to be a p.r. function?
b. Find X for Re[Z ( jω )] to have a second-order zero at ω = 0 .
c. Choose a numerical value for X and synthesize Z (s ) .
2s 2 + s + 2
4. Z ( s) = 2 is p.r. determine min[Re Z ( jω )] and synthesize Z (s ) by first removing
s + s +1
min[Re Z ( jω )] .
5. Perform a continued fraction expansion on the ratio
41
s 3 + 2s 2 + 3s + 2
Z (s) =
s 3 + s 2 + 2s + 1
What does the continued fraction expansion if Y (s) is the driving-point admittance of a passive
network? Draw the network from the continued fraction.
6. The following functions are impedance functions. Synthesize the impedances by successive
removals of jω axis poles or by removing min[Re Z ( jω )] .
s 3 + 4s s +1
a. Z ( s) = b. Z ( s) =
s2 + 2 s ( s + 2)
2s + 4 s 2 + 3s + 1
b. . Z ( s ) = d. Z ( s) =
2s + 3 s2 +1
7. Check whether the following equations are p.r. functions or not.
s 2 + 10s + 4
a. F ( s) =
s+2
s 3 + 5s 2 + 9 s + 3
b. F ( s) =
s 3 + 4s 2 + 7 s + 9
3.4. Elementary Synthesis procedures

The basic philosophy behind the synthesis of driving point functions is to break up a p.r. function Z (s )
into a sum of simpler p.r. functions Z1 ( s ), Z 2 ( s),⋅ ⋅ ⋅, Z n ( s) and then to synthesize these individual
Z i (s ) as elements of the overall network whose driving-point impedance is Z (s ) .
Z ( s) = Z1 ( s) + Z 2 ( s) + ⋅ ⋅ ⋅ + Z n ( s)
First, consider the “breaking-up” process of the function Z (s ) into the sum of functions Z i (s ) . One
important restriction is that all Z i (s ) must be p.r. Certainly, if all Z i (s ) were given to us, we could
synthesized a network whose driving-point impedance is Z (s ) by simply connecting all the Z i (s ) in
series. How if we were to start with Z (s ) to give us the individual Z i (s ) ? Suppose Z (s ) is given in
general as
a n s n + a n−1 s n−1 + ⋅ ⋅ ⋅ + a1 s + a 0
Z (s) =
bm s m + bm−1 s m−1 + ⋅ ⋅ ⋅ + b1 s + b0
42
Consider the case where Z (s ) has a pole at s = 0 (that is, b0 = 0 ). Let us divide P (s ) and Q (s ) to give a
quotient D / s and a remainder R (s ) , which we can denote as Z 1 ( s ) and Z 2 ( s) .

D
Z ( s) = + R( s ) ≥0
s
= Z1 ( s) + Z 2 (s)
Are Z 1 and Z 2 p.r.? From previous discussions, we know that Z1 = D / s is p.r. Is Z 2 ( s) p.r. ? Consider
the p.r. criteria given previously.
1. Z 2 ( s) Must have no poles in the right half plane.
2. Poles of Z 2 ( s) on the imaginary axis must be simple, and their residues must be real and positive.
3. Re[ Z 2 ( jω )] ≥ 0 for all ω
Let us examine these cases one by one. Criterion 1 is satisfied because the poles of Z 2 ( s) are poles
of Z (s ) . Criterion 2 is satisfied by this same argument. A simple partial fraction expansion does not
affect the residues of the other poles. When s = jω , Re[Z ( jω ) = D / jω ] = 0 . Therefore, we have
Re Z 2 ( jω ) = Z ( jω ) ≥ 0
From the foregoing discussion, it is seen that if Z (s ) has a pole at s = 0 , a partial fraction expansion can
be made such that one of the terms is of the form K / s and the other terms combined still remain p.r. A
similar argument shows that if Z (s ) has a pole at s = ∞ (that is, n − m = 1) , we can divide the numerator
by the denominator to give a quotient Ls and a remainder term R (s ) , again denoted as Z1 ( s ) and Z 2 ( s) .
Then Z ( s) = Ls + R( s) = Z 1 ( s) + Z 2 ( s) . Here Z 2 ( s) is also p.r.
If Z (s ) has a pair of conjugate imaginary poles on the imaginary axis, for example, poles at s = ± jω1 ,
then Z (s ) can be expanded into partial fractions so that
2 Ks 2 Ks  j 2ωK 
Z ( s) = + Z 2 ( s ) Here Re( 2 ) = Re  = 0 So that Z 2 ( s) is p.r.
s + ω1
2 2
s + ω1 2
 − ω + ω1
2 2

Finally, if Re[Z ( jω )] is minimum at some point ω i , and Re[Z ( jωi )] = K i as shown in fig.3.6, we can
remove a constant K ≤ K i from Re[Z ( jω )] so that the remainder is still p.r. This is because Re[Z ( jω )]
will still be greater than or equal to zero for all values of ω .
43
Fig.3.6
44
CHAPTER 4
Synthesis of One-Port Networks with Two Kinds Of Element

4.1. Introduction
In this chapter we will study methods for synthesizing one-port network with two kinds of elements.
Since we have three elements to choose from, the networks to be synthesized is either
R − C , R − L or L − C network. We first discuss the properties of a particular type of one-port network
and then we will synthesize it.
A single-port network is a load on the source that excites it. As we have discussed already, the ratio of
source voltage to current drawn by the network is called driving-point impedance, and equation below
expresses how the driving-point impedance is obtained. The ratio of source current to source voltage is
called as the driving-point admittance. Both the driving-point impedance and the driving-point
admittance are the driving-point functions. The term, the driving-point immittance, is used to either the
driving-point impedance or the driving-point admittance.
V ( s) I (s)
Z ( s) = Y ( s) =
I ( s) V (s)
Fig. 4.1. A single port or two terminal network
4.2. Properties and Synthesis of L-C Driving-point Immittances
Before we proceed to synthesis of L-C network, we will first examine some properties of L − C driving
point immittances.
4.2.1. Properties of L-C Immittance Functions
Consider the impedance Z (s ) of a passive one-port network. Let us represent Z (s ) as
45
M 1 (s ) + N1 (s )
z (s ) = where M 1 , M 2 are even parts of the numerator and denominator and N 1 , N 2 are
M 2 (s ) + N 2 (s )
odd parts respectively. The average power dissipated by the one–port network is
Re[z ( jϖ )][I ]
1
Average power =
2
2
where I is the input current. For a pure reactive network, it is know that the power dissipated is zero. We
therefore conclude that the real part of Z(s) is zero. That is the same as
Re[Z ( jϖ )] = Ev[Z ( jϖ )] = 0
M 1 (s )M 2 (s ) − N 1 (s )N 2 (s )
where Ev[Z (s )] =
M 2 (s ) − N 2 (s )
2 2
In order for Ev[Z ( jω )] = 0 ⇒ M 1 ( jω )M 2 ( jω ) − N 2 ( jω )N 1 ( jω ) = 0 , either of the following case must hold

(a) M1 ( jω ) =0=N2 ( jω )
(b) M2 ( jω ) =0=N1 ( jω )
N1 (s)
In case (a), Z (s ) = and in case (b), Z (s ) =
M2
.
M 2 (s) N2
We see from this development the following two properties of L-C function
1. Z LC (s ) or YLC (s ) is the ratio of even to odd or odd to even polynomials
2. Since both M i (s ) and N i (s ) are Hurwitz, they have only imaginary roots with real parts equals to
zero and it follows that the poles and zeros of Z LC (s ) or YLC (s ) are on the jω -axis.
Consider the example of an L-C immittance function given by

a 4 s 4 + a 2 32 + a a
Z (s ) =
s2
b2 3 + b2 + b1
s s
Let us examine the constraints on the coefficients ai and bi. We know first of all that in order for the
impedance to be positive real, the coefficients must be real and positive. We also know that an impedance
function cannot have multiple poles or zeros on the jω -axis. Since ∞ is defined to be on the jω -axis
the highest powers of the numerator and the denominator polynomials can differ by at most unity. For
example, if the highest order of the numerator is 2n then the highest order of the denominator can either
be 2n-1 so that there is simple zero at s = ∞ or the order can be 2n + 1 so that there is a simple zero
at s = ∞ . Similarly, the lowest order of numerator and denominator can differ by at most unity, or else
there would be multiple poles or zeros of Z (s ) at s = ο .
46
Another property of the numerator and denominator polynomial is that if the highest power of the
polynomial is 2n for example the next highest order tem must be 2n − 2 and the succeeding power must
differ by two orders all the way through. There cannot be any missing terms, i.e. no two adjacent terms of
either polynomial may differ by more than two power. For Z (s ) given above, if b3 = 0 , then Z (s ) will
1/ 4
b 
have poles when b5 s 5 + b1 s = 0 so that the poles be at s = 0 and at sk =  1  e j ( 2 k −1)π / 4
 b5 
k = 0,1,2,3
It is clearly seen that none of the pole are even on the jω -axis thus violating one of the basic properties
of an immittance function. From the properties given in the above equation, we can write a general L-C
impedance or admittance as
Z ( s) =
( )( ) (
K s 2 + ω12 s 2 + ω32 ⋅ ⋅ ⋅ s 2 + ωi2 ⋅ ⋅ ⋅ )
( )( ) (
s s 2 + ω22 s 2 + ω42 ⋅ ⋅ ⋅ s 2 + ω 2j ⋅ ⋅ ⋅ )
Expanding into partial fraction, we obtain
K0 K s K s
Z (s) = + 2 2 2 + 2 4 2 + ⋅ ⋅ ⋅ + K∞s
s s + ω 2 s + ω4
where K i are the residues of the poles. Since poles are all on the jω -axis, the residues must be real and
positive in order for Z (s ) to be positive real. Letting s = jω , we see that Z ( jω ) has zero real part, and
can thus be written as a pure reactance jX (ω ) . Thus we have
 K K ω 
Z ( jω ) = j  − 0 + 2 2 2 + ⋅ ⋅ ⋅ + K ∞ ω  = jX (ω )
 ω ω2 + ω2 
Differentiating with respective to ω , we have
dX (ω ) K 0
= 2 + K∞ +
(
2 K 2 ω 2 + ω 22 )
+ ⋅⋅⋅
dω ω (
ω 22 − ω 2
2
)
Since all the residues K i are positive, it is seen that for an L-C function,
dX (ω )
dω
≥0
A similar development show that the derivative of Im[Y ( jω )] = B(ω ) is also positive. That is
47
dB (ω )
dω
≥0
Consider the following example Z(s) is given as
(
Ks s 2 +ω 3
2
)
)(s )
Z (s) =
(s 2
+ω 2
2 2
+ ω4
2
Letting s = jω we obtain X (ω )
Z ( jω ) = jX (ω ) = + j
(
Kω − ω 2 + ω 32 )
(
− ω 2 + ω 22 − ω 2 + ω 42 )( )
Let us draw a curve of X (ω ) versus ω beginning with the zero at ω = 0, and examine the sequence of
critical frequencies encountered as ω increase. Since the slope of the X (ω ) curve is always positive the
net critical frequency we encounter is when X (ω ) becomes infinitely large or the pole is at ω 2 . As we
pass ω 2 , X (ω ) change sign and goes from +ve to -ve. In general, whenever we pass though any critical
frequency, there is always a change of sign as seen from the way jX (ω ) is written in the last equation.
After we pass through ω 2 , with the slope of X (ω ) always positive it is easy to see that the next critical
frequency is the zero at ω3 . Thus, if impedance function is an L-C imittance, the poles and zeros of the
function must alternate. The particular X (ω ) under discussion takes the form shown in fig. 4.2. Since the
highest power of the numerator and the denominator always differ by unity and the lowest power also
differ by one we observe that at s = 0 and at s = ∞ there is always a critical frequency whether a zero or
a pole.
For the example just discussed there is a zero at s = 0 and a zero at s = ∞ the critical frequencies
at s = 0 and at s = ∞ are called external critical frequencies whereas the remaining unite critical
frequencies are referred to as internal thus in the previous example ω1 , ω 2 & ω 4 , and internal critical
frequencies.
Fig. 4.2. Poles and zeros of an L-C immittance function alternate
48
Finally let us summarize the properties of L-C impedance or admittance functions.

1. Z LC ( s ) or YLC ( s ) is the ratio of odd on even or even to odd polynomials.
2. The pole and zeros are simple and lie on the axis
3. The pole and zero interlace on the axis
4. The highest power of numerator and denominator differ by unity; the lowest power also differ by
unity
5. There must be either a zero or a pole at the origin and infinity.
Exercises: Check whether the following functions are L-C or not. If not give the reason.
1. z ( s ) =
(
Ks s 2 + 4 ) 2. Z ( s ) =
s 2 + 4 s 2 + 5s
(s 2 + 1)(s 2 + 3) 3s 2 + 6s 2
(
K s2 +1 s2 + 9
3. Z ( s ) = 2
)( ) 4. Z ( s ) =
( )(
2 s2 +1 s2 + 9 )
( )
s + 2 (s 2 + 10 ) (
s s2 + 4 )
4.1.2. Synthesis of L-C Immittance Functions
We saw in the above section that an L-C immittance is a positive real function with poles and zeros on
the axis only. The partial fraction expansion of an L-C function is expressed in general terms as
K0 Ks K s Ks
F ( s) = + 2 1 2 + 2 2 2 .⋅⋅⋅ + 2 i 2 + K∞s
s s + ω1 s + ω2 s + ωi
The synthesis is accomplished directly from the partial fraction expansion by associating the individual
terms in the expansion with network elements. If F(s) is an impedance Z(s), then the term K 0 / s
represents a capacitor of 1 / K 0 farads: the term K ∞ s is an inductor of K ∞ henrys, and the term
Ki s
is a parallel tank circuit that consists of a capacitor of 1 / K i farads in parallel with an inductor
s + ωi
2 2
of K i / ω i2 . Thus a partial fraction expansion of general L-C impedance would yield the network shown in
fig. 4.3. This method of synthesis that is based on partial fraction expansion is called Foster synthesis.
49
Fig. 4.3. Foster Synthesis of an L-C impedance function
For example, consider the following L-C function.

(
Z (s ) =
)
2 s 2 + 1 (s 2 + 9 )
( )
s s2 + 4
A partial fraction expansion of Z(s) gives
15
s
Z (s ) = 2 s +
9/2
+ 22
s s +4
We then obtain the synthesized network shown in fig. 4.4.
Fig. 4.4.
The partial fraction expansion method is based upon the elementary synthesis procedure of removing
poles on the jω axis. The advantage with L-C functions is that all the poles of the function lie on the jω -
axis so that we can remove all the poles simultaneously. Then the partial fraction expansion of Y(s)
which is given by the equation below gives us a circuit consisting of parallel branches as shown in fig.
4.5. This is Foster II synthesis method of an LC network.
K0 K s K s
Y (s) = + 2 2 2 + 2 4 2 + ⋅ ⋅ ⋅ + K∞ s
s s + ω2 s + ω4
50
Fig. 4.5. Foster II synthesis
For example consider the admittance function Y (s ) =

(
s s 2 + 2 s2 + 4 )(.
)
(s 2 + 1)(s 2 + 3)
The partial fraction expansion of Y(s) is
1 3
s s
Y ( s) = s + 2
+ 22
s + 3 s +1
2
from which we synthesize the network shown in fig. 4.6.
Fig. 4.6
The L.C networks synthesized by a partial fraction expansions are, as mentioned above, sometimes
called Faster–type networks the impedance from is sometimes called a foster series networks(Foster I)
and the admittance form is a foster parallel network (Foster II).
A useful property of L-C immittances is that the numerator and the denominator always differ in degree
by unity. Therefore, there is always a zero or pole at s = ∞ . Suppose we consider the case of an L-C
impedance Z(s) whose numerator is of degree 2n and denominator is of degree 2n-1, giving Z(s) a pole
at s = ∞ . We can remove this pole by removing impedance L1s so that the remainder function Z2(s) is still
L-C.
Z 2 ( s ) = Z ( s ) − L1 s
The degree of the denominator of Z 2 ( s) is 2n − 1 but the numerator is of degree 2n-2 because the
numerator and denominator must differ in degree by 1. Therefore we see that Z 2 ( s) has a zero at s = ∞ .
51
If we invert Z 2 ( s) to give Y2 ( s ) = 1 / Z 2 ( s ) , then Y2 ( s) will have a pole at s = ∞ which we can again

remove to give a capacitor C2s and a remainder Y3(s), which is
Y3 ( s) = Y2 ( s) − C 2 s.
We readily see that Y3 ( s) has a zero at s = ∞ , which we can invert and remove. This process continues
until the remainder is zero. Each time we remove a pole, we remove an inductor or a capacitor depending
upon whether the function is impedance or admittance. Note that the final structure of the network
synthesized is a ladder whose series arms are inductors and whose shunt arms are capacitors, as shown in
fig. 4.7.
Fig. 4.7 Cauer Synthesis of an L-C impedance function

Consider the following example,
2 s 5 + 12 s 3 + 16 s
Z ( s) =
s 4 + 4s 2 + 3
We see that Z (s ) has a pole at s = ∞ , which we can remove by first dividing the denominator into the
numerator to give a quotient 2s and a remainder Z 2 ( s ) , as shown in fig. 4.8.
Fig. 4.8
Then we have
4s 3 + 10s
Z 2 ( s) = Z ( s ) − 2 s = 4
s + 4s 2 + 3
Observe that Z 2 ( s ) has a zero at s = ∞ . Inverting Z 2 ( s ), we again remove the pole at infinity. Then we
1
realize a capacitor of farad and remainder Y3 ( s) , as may be seen in fig. 4.9.
4
52
3 2
s +3
1 2
Y3 ( s ) = Y2 ( s ) − s = 2
4 4 s + 10 s
8
Removing the pole at s = ∞ of Z 3 ( s ) = 1 / Y3 ( s ), gives a series inductor of h and
3
8 2s
Z 4 ( s) = Z 3 ( s) − s = as shown in fig. 4.10.
3 3 2
s +3
2
Fig. 4.9 Fig. 4.10

3
The admittance Y4 ( s ) = 1 / Z 4 ( s ) has a pole at s = ∞ , which we remove to give a capacitor of farad and
4
2
a remainder Y5 ( s) = 3 / 2s, which represents an inductor of h . Removing this inductor gives us zero
3
remainder. Our synthesis is therefore complete and the final network is shown in fig. 4.11.
Fig. 4.11
Since we always remove a pole at s = ∞ by inverting the remainder and dividing, we conclude that we
can synthesis an L-C ladder network by a continued fraction expansion. The quotients represent the poles
at s = ∞ , which we remove, and we invert the remainder successively until the remainder is zero. For the
previous example, the continued fraction expansion is
53
s 4 + 4 S 2 + 3 2s 5 + 12s 3 + 16s (2s ↔ Z

2 s 5 + 8s 3 + 6 s
1
4 s 3 + 10s) s 4 + 4 s 2 + 3 s ↔ Y
4
5
s4 + s2
2
3 2  8
s + 3 4 s 3 + 10 s s ↔ Z
2  3
4 s 3 + 8s
3
2 s ) s 2 + 3 s ↔ Y
3
2 4
3 2
s
2
2
3 )2 s s ↔ Z
3
2s
We see that the quotients of the continued fraction expansion give the elements of the ladder network.
Because the continued fraction expansion always invites each remainder and divide, the successive
quotients alternate between Z and Y and then again Z, as shown in the proceeding expansion. If the initial
function is impedance, the first quotient must necessarily be impedance. When the first function is an
admittance the first quotient is an admittance.
Since the lowest degrees of numerator and denominator of an L-C admittance must differ by unity, it
follows that there must be a zero or a pole at s = 0 . If we follow the same procedure we have just outlined
and remove successively poles at s = 0 , we will have an alternate realization in a ladder structure. To do
this by continued fraction, we arrange both numerator and denominator in ascending order and divide the
lowest power of the denominator into the lowest power of the numerator: then we invert the remainder
and divide again.
For example in the case of the impedance we have
Z (s ) =
(s 2
)( )
+ 1 s 2 + 3 3 + 4s 2 + s 4
=
(
s s2 + 2 ) 2s + s 3
The continued fraction expansion to give the alternate realization is
54
 3
2s + s 3 3 + 4s 2 + s 4  ← z
 2s
3
3 + s2
2
5 2   4
s + s 4 2s + s 3  ← Y
2   5s
4
2s + s 3
5
1 35 2  25
s  s + s4 ←Z
5 2  2s
5 2
s
2
s4 ) 15 s 3  1
  ← Y
  5s
1 3
s
5
The final synthesized network is shown in fig. 4.12. The ladder networks realized are called cauer ladder
network because W. Cauer discovered the continued fraction method for synthesis of a passive network.
Fig. 4.12
Note that for both the foster and the Cauer-form realizations the number of element is one greater than
the number of intern critical frequents which we defined previously as being all the poles and zeros of
the function excluding those at s = 0 and s = ∞ . Without going into the proof of the statement, it can be
said that both the foster and the Cauer forms give the minimum number of elements for a specified LC
driving .point function. These realizations are sometimes known as canonical forms.
4.2. Properties and Synthesis of R-C Impedances and R-L Admittances

As we have done for synthesis of impedance functions of L-C networks, we will first discuss properties
of R-C driving point impedance and then we proceed to its synthesis.
55
4.2.1. Properties of R-C Impedances

The properties of R-C driving point impedances can be derived from known properties of L-C functions
by a process of mapping the jω -axis on to the − σ -axis. Here we will assume that all driving point
function that can be realized with two kinds of elements can be realized in a foster form. Based upon this
assumption we can derive all the pertinent properties or R-C or R-L driving point functions.
Let us consider first the properties or R-C driving point impedance functions.
Referring to the series foster form for an L-C impedance given in the fig. 4.3, we can obtain a foster
realization of an R-C impedance by simply replacing all the inductances by resistance so that a general R-
C impedance could be represented as shown in Fig. 4.13.
Fig. 4.13 Foster realization of an R-C Impedance function
Z (s ) =
K0 K1 K2 Ki
+ K∞ + + + ... +
s s +σ1 s +σ 2 s +σi
Ki
where C 0 = 1 , R∞ = K ∞ , C i = 1 , Ri =
K0 Ki σ i.
From this development two major properties of R-C impedance are obtained and are listed in the
following
1. The poles of R-C driving point impedance are on the negative real (− σ ) -axis it can be shown
from a parallel foster form that the poles of an R-C admittance function are also on the axis we
can thus conclude that the zeros of an R-C impedances are also on the (− σ ) -axis.
2. The residues of the poles K i are real and positive. We shall see later that this property does not
apply to R-C admittances.
56
Since the poles and zero of R-C impedances are on the (− σ ) -axis, let us examine the slope of Z (σ )
dZ (σ )
along the - σ -axis. To find the slope, , we first let s = σ in Z (s ) and then we take the derivative of
dσ
Z ( σ ) w. r. t σ . Thus we have,
k0 K2 K2
Z (σ ) = + K∞ + +
σ σ + σ1 σ + σ 2
dZ (σ ) −K0 − K1 − K2
and = + + + ...
dσ σ2
(σ + σ 1 ) 2
(σ + σ 2 ) 2
dZ (σ )
It is clear that <0
dσ
Let us now look at the behavior of Z(s) at the two points where the real axis and the imaginary axis
interest, namely at σ = ω = 0 and at σ = ω = ∞ . This is readily done by examining the general R-C
network in fig. 4.13 at these two frequencies. At σ = 0 (dc), if the capacitor Co is in the circuit, it is an
open circuit and there is a pole of Z(s) at σ = 0 . If Co is not in the circuit, then Z (0) is simply the sum of
all the resistances in the circuit because all of the capacitors are open circuits at σ = 0 .
. Z (0) = R1 + R2 + ... + R∞
At σ = ∞ all the capacitors are short circuits. Thus if R∞ is in the circuit, Z (∞ ) = R∞ . If R∞ is missing
then Z (∞ ) =0. To summarize these last two statements we have
∞ C0 Pr esent

1. Z (0) =  ∞
∑ Ri C0 mis sin g
 i =1
0 R∞ mis sin g
2. Z (∞) = 
 R∞ R∞ Pr esent
If we examine the two cases for Z (0) and Z (∞ ) , we see that Z (0) ≥ Z (∞ ) .
Next let us see whether the pole and zeros of an R-C impedance function alternative. We have already
established that the critical frequency nearest the origin must be a pole and the critical frequency nearest
σ = ∞ must be a zero. Therefore if Z(s) is given as
Z ( s) =
(s +σ 2 )(s + σ 4 )
(s + σ 1 )(s + σ 3 )
57
Then, if Z (s ) is R-C, the singularity nearest the origin must be a pole which we will assume to be
at s = −σ 1 ; the singularity furthest from the origin must be a zero, which we will take to be s = −σ 4 . Let
us plot Z (σ ) =
(σ +σ 2 )(σ + σ 4 ) versus − σ beginning at σ = 0 and extending to σ = −∞ . At
(σ + σ 1 )(σ + σ 3 )
σ = 0, Z (0) is equal to a positive constant.
σ 2σ 4
Z ( 0) =
σ 1σ 3
Since the slope of Z (σ ) is always positive as − σ increase, Z (σ ) must increase until the poles s = −σ 1 is
reached. At σ = −σ 1 , Z (σ ) change sign and is negative until the next critical frequency is reached. We
see that this next critical frequency must be the zero, s = −σ 2 . Since Z (σ ) increase for increasing − σ ,
the third critical frequency must be the pole, s = −σ 3 . Because Z (σ ) change sign at − σ 3 , the final
critical frequency must be the zero, s = −σ 4 . Beyond σ = −σ 4 , the curve becomes asymptotic
to Z (∞ ) = 1 . From this analysis we see that the poles and zeros of R-C impedance must alternate. For the
σ 2σ 4
case being considered ∞ > σ 4 > σ 3 > σ 2 > σ 1 ≥ 0 . In addition we see that > 1 which shows that
σ 1σ 3
z (0) > z (∞ )
Fig. 4.14. Poles and zeros alternate on − σ -axis for an R-C impedance
To summarize the three properties we need to recognize an R-C impedance are

1. Pole and zeros lie on the negative real axis and they alternate.
58
2. The singularity nearest to (or at) the origin must be a pole whereas the singularity nearest to (or
at) σ = −∞ must be zero.
3. The residues of the poles must be real and positive
Example: Check if the following impedance functions are R-C impedance.
1. Z ( s ) =
(s + 1)(s + 4)(s + 8) 3. Z (s ) =
(s + 2)(s + 4)
3(s + 2)(s + 6) (s + 1)
2. Z (s ) =
(s + 1)(s + 8) 4. Z (s ) =
(s + 1)(s + 2)
(s + 2)(s + 4) s(s + 3)
Let us re-examine the partial fraction expansion of a general R-C impedance
F (s ) =
K0 K1 K2 Ki
+ K∞ + + + ⋅⋅⋅ +
s s + σ1 s + σ 2 s +σi
Instead of letting F (s ) represent an impedance consider the case where F (s ) is an admittance Y (s ) . If we
associate the individual terms in the expansion to network elements, we then obtain the net work shown
in Fig. 4.15.
Fig. 4.15. R-C impedance function can be realized as R-L admittance
We see that an R-C impedance ZRC(s) can also be realized as R-L admittance YRL(s). All the properties of
R-C impedances it is therefore important to specify where a function is to be realized as an R-C
impedance or an R-L admittance.
4.2.2. Synthesis of R-C Impedances or R-L Admittances

We postulated in section 4.3.1 that the foster form realization exists for an R-C impedance or an R-L
admittance. Since foster network are synthesized by partial fraction expansions the synthesis is
accomplice with ease. An important point to remember is that we must remove the minimum real part of
z ( jω ) in the partial fraction expansion . It can be shown that min [Re Z ( jω ) = Z (∞ )] so that we have to
remove Z (∞ ) as a resistor in the partial fraction expansion. In cases where the numerator is of lower
59
degree than the denominator, Z (∞ ) = 0 . When the numerator and the denominator are of the same
degree, then Z (∞ ) can be obtained by dividing the denominator to numerator. The quotient is then Z (∞ ) .
Consider the following example.

3(s + 2 )(s + 4 )
F (s ) =
3(s + 3)
The partial fraction expansion of the remainder function is obtained as
8 1
F ( s) = + +3
s s+3
Here F (∞ ) = 3 . If F (s ) is an impedance Z (s ) , it must be an R-C impedance and it is realized in the
series foster form as shown in fig. 4.16.
Fig. 4.16.
On the other hand if F (s ) represents an admittance, we realize Y (s) as an R-L network in the parallel
foster form as in fig. 4.17.
Fig. 4.17.
An alternate method of synthesis is based on the following fact. If we remove min Re[Z ( jω )] = Z (∞ )
from Z (s ) , we create a zero at s = ∞ for the remainder Z 2 ( s) . If we invert Z 2 ( s) , we then have a pole at
s = ∞ which we can remove to give Z 3 (s) . Since min Re[Y3 ( jω )] = Y3 (∞ ) , if we remove Y3 (∞ ) we

would have a zero at s = ∞ again which we again invert and remove. The process of extracting Z (∞ ) or
Y (∞ ) and the removal of a pole of the reciprocal of the reminder involve dividing the numerator by the
60
denominator. Consequently, we see that the whole synthesis process can be resolved by a continued
fraction expansion. The quotients represent the elements of a ladder network.
3(s + 2 )(s + 4 )
For example the continued fraction expansion of F (s ) =
3(s + 3)
)
s 2 +3s 3s 2 + 18s + 24(3 ← Z / Y
3s 2 + 9 s
s
9 s + 24 )s 2 + 3s ← Y / Z
9
8
s2 + s
3
1 
s 9s + 24(27 ← Z / Y
3 
9s
 1
24 ) 13 s s ← Y / Z
 72
1
3s
If F(s) is impedance Z(s) the resulting network is shown in Fig. 4.18. If F(s) is admittance Y(s), we have
the R-L network of fig. 4.19.
Fig. 4.18. Fig. 4.19.
4.3. Properties and Synthesis of R-L Impedances and R-C Admittances
The immittance that represents series foster R-L impedance or a parallel foster R-C admittance is given
as.
F (s ) = K ∞ s + K 0 +
K 1s K 2s Kis
+ + ... +
s +σ1 s +σ 2 s +σi
61
The significant difference between an R-C impedance and R-L impedance is that the partial fraction
expansion term for the R-C tank circuit is K i / (s + σ i ) ; whereas for the R-L impedance the corresponding
term must be multiplied by an s in order to give an R-L tank circuit consisting of a resistor in parallel
with an inductor.
The properties of R-L impedance or R-C admittance function can be derived in much the same manner as
the properties of R-C impedance functions. Without going into the derivation of the properties, the more
significant ones are given in the following
1. Poles and zeros of an R-L impedance or R-C admittance are located on the negative real axis and
they alternate.
2. The singularity nearest to or at the origin is a zero .The singularity nearest to or at s = ∞ must be a
pole.
3. The residues of the pole must be real and negative.
Because of the third property a partial fraction expansion of an R-L impedance function would yield
Ki
terms as −
s +σi
This does not present any trouble, as the term above does not represent an R-L impedance at all. To
obtain the foster from of an R-L impedance we expand Z ( s ) / s into partial fraction. If Z(s) is and R-L
impedance, we will state without proof that the partial fraction expansion of Z ( s ) / s yields positive
residues. Thus we have.
Z (s) K K1 K2 Ki
= K∞ + 0 + + + ... +
s s s +σ1 s +σ 2 s +σi
where K 0, K 1 , ... K ∞ >0 . If we multiply both sides by s we obtain Z (s ) in the desired form for
synthesis.
2( s + 1)(s + 3)
Consider the following function F ( s ) = .
( s + 2)(8 + 6)
F(s) represents an R-L impedance or an R-C admittance because it satisfies the fires two criteria cited.
The partial fraction Expansion of F (s ) is
1 15
F (s) = 2 − 2 − 2
(s + 2) (8 + 6)
So we see that the residues are negative.
62
The partial fraction expansion of F ( s ) / s on the other hand is
1 1 5
F (s)
= 2+ 4 = 4
s s s+2 s+6
If we multiply both sides by s we obtain
1 5
s s
1 4 4
F (s) = + +
2 s+2 s+6
If F (s ) represents an impedance Z (s ) it is synthesized in series foster form giving the R-L networks in
fig. 4.20. If F (s ) is an admittance Y(s), then it is synthesized in parallel foster form resulting in the R-C
networks shown in fig. 4.21.
Fig. 4.20 Fig. 4.20
To synthesize an R-L impedance in ladder form, we make use of the fact that min Re[Z ( jω )] = Z (0) . If
we remove Z(0) from Z(s), the remainder function Z1(s) will have a zero at s = 0 . After inverting Z1(s),
we can then remove the pole at s = 0 . Since the value Z (0) is obtained by dividing the lowest power
term of the denominator into the lowest power term of the numerator the synthesis could be carried out
by a continued fraction expansion by arranging the numerator and denominator polynomials in ascending
order and then dividing.
For example the following function is either an R-L impedance or an R-C admittance.
2( s + 1)(s + 3) 6 + 8s + 2s 2
F ( s) = =
( s + 2)(s + 6) 12 + 8s + s 2
The continued fraction expansion of F (s ) is
63
1
12 + 8s + s 2 6 + 8s + 2 s 2 (
2
1 2
6 + 4s + s
2
3 2 3
4s + s 12 + 8s + s 2 
2  s
9
12 + s
2
9  3 8
s + s 2 4 s + s 2 
2  2 7
8
4s + s 2
7
5 27  49
s  s + s2 
14  2  5s
7
s
2
s2 )5 2 5
s 
14  14
5 2
s
14
0
If F (s ) is an impedance function, the resulting network is shown in fig. 4.22. If, on the other hand, F (s )
is an R-C admittance Y (s ) , then the network is synthesized as fig. 4.23.
Fig. 4.22. Fig. 4.23.
4.4. Synthesis of an R-L-C Functions

Under certain conditions R-L-C driving point function may be synthesized with the use of either partial
fractions (Foster Form) or continued fractions (Cauer Form). For example the function,
64
s 2 + 2s + 2
Z (s) = 2
s + s +1
Z (s ) is neither L-C, R-C nor R-L. Nevertheless the function can be synthesized by continued fraction
expansion as shown below.
s 2 + s + 1 s 2 + 2 s + 2 (1 ← Z
s2 + s +1
s + 1)s 2 + s + 1(s ← Y
s2 + s
1)s + 1(s + 1 ← Z
s +1
0
The network derived from this expansion is given in fig. 4.24.
Fig. 4.24
Consider the following admittance function. The poles and zeros of the admittance function are all on the
negative real axis but they do not alternate.
s + 2)(s + 3)
Y ( s) =
( s + 1)(s + 4
The partial fraction expansion for Y(s) is,
2
−3
Y ( s) = 1 + s
+
s +1 s+4
Since one of residues is negative we cannot use this expansion for synthesis. An alternate method would
be to expand Y ( s ) / s and then multiply the whole expansion by s .
65
3 2 1
Y ( s) 2
= − 3 + 6
s s s +1 s + 4
Multiply by s we obtain,
2 1
s s
3 3 6
Y ( s) = − +
2 s +1 s + 4
Note that Y(s) also has a negative term. If we divide the numerator of this negative term to the
denominator we can rid ourselves of any terms with negative signs.
 2  1 2 1
3  2  s s
5
Y ( s) = − − 3 + 6 = + 3 + 6
2  3 s +1 s + 4 6 s +1 s + 4
 
 
The network that is realized from the expanded function is given in Fig. 4.25 below.
Fig. 4.25
Considering the same example. To synthesis in cauer form if we try to expand Y (s ) by continued
fractions, we see that negative quotients result. However we can expand Z ( s ) = 1 / Y ( s ) by continued
fraction. Although the expansion is not as simple or straight forward as in the case of an R-C functions,
because we sometimes have to reverse the order of division to make the quotients all positive. The
continued fraction expansion of Z (s ) is
66
6 + 5s + s 2 4 + 5s + s 2 (2 / 3 ← Z
10s 2 2
4+ + s
3 3
5 1   18
s + s 2 6 + 5 s + s 2  ← Y
3 3   5s
6
6+ s
5
19 1 5 1
s + s 2  s 2 + s ← Z
5 3 3 3
1 2 19
s + s
3 15
6  19  19
s s + s2 ← Y
15  5 2
19
s
5
s2 )6  6
s
15  15s
←Z
6
s
15
As we see the division process giving the quotient of 1 / 3 involves a reversal of the order of the
polynomials involved. The resulting ladder network is given on fig. 4.26.
Fig. 2.26
At the beginning of this section it was stated that only under special conditions can an R-L-C driving
point function be synthesized with the use of a ladder form or the foster forms. These conditions are not
given here because they are rather involved instead. When a positive real function is given and it is
found that the function is not synthesizable by using two kinds of element only, it is suggested that a
continued fraction expansion or a partial fraction expansion be tried first.
67
Tutorial Problems
1. a. Which of the following functions are L-C driving point impedances? Why?
s ( s 2 + 4)( s 2 + 16)
i. Z1 ( s ) =
( s 2 + 9)( s 2 + 25
( s 2 + 1)( s 2 + 8
ii. Z 2 ( s ) =
s ( s 2 + 4)
b. Synthesize the realizable impedances in a foster and Cauer forms.
2. Indicate the general form of the two foster and the two Cauer networks that could be used to
synthesize the following L-C impedance.
(
( s 2 + 1)( s 2 + 9) s 2 + 25
Z (s) =
)
s ( s 2 + 4)( s 2 + 16)
There is no need to calculate the element values of the network.
6s 4 + 42s 2 + 48
3. Synthesize the L-C driving point impedance Z ( s ) = in the form shown the
s 5 + 18s 3 + 48s
figure below i.e., determine the element values of the network
4. There exists an L-C network with the same driving-point impedance as the network shown in the
figure. This alternate network should contain only two elements. Find this network.
5. The input impedance for the network shown is

2s 2 + 2
Z in = 2
s + 2s 2 + 2 s + 2
68
If Z 0 is an L-C network,
(a) Find the expression for Z 0 .
(b) Synthesize Z 0 in a foster series form.
6. Indicate which of the following function is either R-C, R-L of L-C impedance functions
s 3 + 2s s 2 + 6s + 8
a) Z ( s ) = 4 b) Z ( s ) = 2
s + 4s 2 + 3 s + 4s + 3
s + 4s + 3
2
s 2 + 2s + 6
c) Z ( s ) = 2 d) Z ( s ) =
s + 6s + 8 s2 + s
s 4 + 5s 2 + 6
e) Z ( s ) =
s3 + s
7. An impedance function has the pole zero pattern shown in the figure below. If Z ( −2) = 3
synthesize the impedance in a foster form and a Cauer form.
8. From the following functions pick out the ones which are R-C admittances and synthesize in one
foster and one Cauer form.
2( s + 1)( s + 3) 4( s + 1)(s + 3)
a) Y ( s ) = c) Y ( s ) =
( s + 2)(s + 4) s ( s + 2)
s ( s + 4)(s + 8) ( s + 1)(s + 4)
b) Y ( s ) = d) Y ( s ) =
( s + 1)( s + 6) s ( s + 2)
9. Find the networks for the following function. Both foster and ladder forms are required.
( s + 1)(s + 4) 3( s + 1)(s + 4)
a) Z ( s ) = b) Y ( s ) =
s ( s + 2) ( s + 3)
10. For the network shown, find Y when
V 1 s ( s 2 + 3)
G(s) = 2 = = 2
V0 2 + Y 2 s + s 2 + 6 s + 1
Synthesize Y as L-C admittance
69
11. Synthesize by continued fractions the function

s 3 + 2 s 2 + 3s + 1)
Y ( s) = 3
s + s 2 + 2 s + 1)
12. synthesis the following functions in Cauer form.
s 3 + 2s 2 + s + 1
a) Z ( s ) = ,
s3 + s2 + 2
s 3 + s 2 + 2s + 1
b) Z ( s ) =
s 4 + s 3 + 3s 2 + s + 1
4s 3 + 3s 2 + 4s + 2
c) Z ( s ) =
2s 2 + s
13. Of the three pole-zero diagrams shown below, pick the diagram that represents an R-L impedance
function and synthesize in series Foster form.
70
CHAPTER 5
Two Port Networks

5.1. Introduction
In most cases we encountered that networks with terminals are connected in pairs to other networks. A
two-port network (a kind of four-terminal network or quadripole) is an electrical circuit or device
with two pairs of terminals connected together internally by an electrical network. Two terminals
constitute a port if they satisfy the essential requirement known as the port condition: the same current
must enter and leave a port. Examples include small-signal models for transistors (such as the hybrid-
model), filters and matching networks.
A two-port network makes possible the isolation of either a complete circuit or part of it and replacing it
by its characteristic parameters. Once this is done, the isolated part of the circuit becomes a "black box"
with a set of distinctive properties, enabling us to abstract away its specific physical buildup, thus
simplifying analysis. Any linear circuit with four terminals can be transformed into a two-port network
provided that it does not contain an independent source and satisfies the port conditions.
There are a number of alternative sets of parameters that can be used to describe a linear two-port
network, the usual sets are called z, y, h, g, and ABCD parameters, each described individually below.
These are all limited to linear networks since an underlying assumption of their derivation is that any
given circuit condition is a linear superposition of various short-circuit and open circuit conditions. They
are usually expressed in matrix notation, and they establish relations between the variables which are
shown in fig. 5.1.
Fig. 5.1 Two port network

V1 = Input voltage, V2 = Output voltage, I1 = Input current, I 2 = Output current
These current and voltage variables are most useful at low-to-moderate frequencies. At high frequencies
(e.g., microwave frequencies), the use of power and energy variables is more appropriate, and the two-
port current–voltage approach is replaced by an approach based upon scattering parameters.
71
A pair of terminals can be called a port only if the current entering one terminal is equal to the current
leaving the other; this definition is called the port condition. A four-terminal network can only be
properly called a two-port when the terminals are connected to the external circuitry in two pairs both
meeting the port condition.
By analogy with transmission network, one of the ports is called the input port, while the other termed as
output port. There are certain properties of two-ports that frequently occur in practical networks and can
be used to greatly simplify the analysis. These include:
Reciprocal networks: A network is said to be reciprocal if the voltage appearing at port 2 due to a
current applied at port 1 is the same as the voltage appearing at port 1 when the same current is applied to
port 2. In general, a network will be reciprocal if it consists entirely of linear passive components (that is,
resistors, capacitors and inductors).
Symmetrical networks: A network is symmetrical if its input impedance is equal to its output
impedance. Most often, but not necessarily, symmetrical networks are also physically symmetrical.
Sometimes also antisymmetrical networks are of interest. These are networks where the input and output
impedances are the duals of each other.
Lossless network: A lossless network is one which contains no resistors or other dissipative elements.
5.2. Two-port Parameters

A two port network is a special case of multiport network. Each port consists of two terminals one for
entering the current and the other for leaving. In order to describe the relationship between the port
voltage and port current of a linear two port network, two linear equations are required among the four
variables.
A general two-port network, shown in fig. 5.1, has two pairs of voltage current relationships. The
variables are V1 , V2 , I 1 , I 2 . Two of these are dependent variables; the other two are independent variables.
The number of possible combinations generated by four variables taken two at a time is six. Thus there
are six possible sets of equations describing a two-port network. We will discuss the four most useful
descriptions here.
A one-port network is completely characterized by its driving-point impedance or admittance function. In

contrast a two-port network requires in general, the specification of four parameters. In the realization of
72
filter networks an often specified quantity is the open circuit voltage ratio T ( s ) = V2 / V1 , (V2 is the output
or the load end voltage and V1 the input or the source end voltage).
Figure → 2' . equivalent circuit of a two port network in terms of Z-parameters.
5.2.1. Open circuit Impedance parameters (Z- Parameters)

Expressing two-port voltages in terms of two-port currents, i.e., (V1 , V2 ) = f ( I 1 , I 2 )
V1 = Z11I1 + Z12 I 2
V2 = Z 21I1 + Z 22 I 2
V1   Z11 Z12   I1 

V  = Z   or [V ] = [Z ][I ]
 2   21 Z 22   I 2 
Where [Z ] is the open circuit impedance matrix of the two port network and impedances Z ij are the
open circuit impedance parameters and the equations of the two port networks given above are called the
z-parameter equations. In these equations the variables V1 and V2 are dependent, and I1 , I 2 are
independent.
The individual z parameters are defined by
V1 V1 V2 V2
z11 = , z12 = , z21 = , z22 =
I1 I 2 =0
I2 I1 =0
I1 I 2 =0
I2 I1 =0
All these parameters are measured under open circuit conditions as shown in the fig.5.2. Hence these
circuit parameters are called open circuit impedance parameters. z11 relates the current and voltage in the
1-1’ port only; whereas z22 gives the current-voltage relationship for the 2-2’ port. Such parameters are
called open-circuit driving-point impedances. On the other hand, the parameters z12 and z 21 relate the
voltage in one port to the current in the other. These are known as (open-circuit) transfer impedances. If
the two port network is reciprocal, then z12 = z 21 .
Fig. 5.2 Determination of the Z-parameters
73
The equivalent circuit model of the above equations is shown in fig. 5.3.
Fig. 5.3 equivalent circuit model of Z-parameter
It is observed that all the -parameters have the dimensions of impedance. Moreover, the individual
parameters are specified only when the current in one of the ports is zero. This corresponds to one of
ports being open circuited, from which the z parameters also derive the name open-circuit parameters.
As an example, let us find the open-circuit parameters for the T circuit in fig. 5.4. We can obtain the
z − parameters by inspection.
Fig. 5.4
V1 V2 V1 V1
z11 = = Z a + Zb , z 22 = = Z b + Z c , z12 = = Z b , z12 = = Zb
I1 I 2 =0
I2 I 2 =0
I2 I1 =0
I2 I1 =0
Observe that z12 = z21 and hence the network is reciprocal. In general, most passive time-invariant
networks are reciprocal.
Most two-port networks, whether passive or active, can be characterized by a set of open-circuited
parameters. Usually, the network is sufficiently complicated so that we cannot obtain the z-parameters by
inspection, as we did for the T circuit in fig. 5.4. The question now, ‘how do we obtain the z-parameters
for any circuit in general? The procedure is as follows. We write a set of node equations with the voltages
74
at the ports V1 and V2 , and other node voltages within the two-port V3,V4 , ,⋅ ⋅ ⋅,Vk as the dependent
variables. The independent variables are the currents I 1 and I 2 , which we will take to be current sources.
We then proceed to write a set of node equations.
I1 = n11V1 + n12V2 + n13V3 + ⋅ ⋅ ⋅ + n1kVk
I 2 = n21V1 + n22V2 + n23V3 + ⋅ ⋅ ⋅ + n2 kVk
0 = n31V1 + n32V2 + n33V3 + ⋅ ⋅ ⋅ + n3kVk
M M M
0 = nk1V1 + nk 2V2 + nk 3V3 + ⋅ ⋅ ⋅ + nkkVk
1
where nij represents the admittance between the i th and j th nodes, that is, nij = Gij + sCij +
sLij
If the circuit is made up of R-L-C elements only, then it is clear that nij = n ji . As a result, the ji th
cofactor of the determinant of the node equations ∆ ij , must be equal to the ji th cofactor ∆ ji , that is,
∆ ij = ∆ ji . This result leads directly to the reciprocity condition z 21 = z12 .
∆11 ∆ ∆12 ∆
V1 = I 1 + 21 I 2 , V2 = I 1 + 22 I 2
∆ ∆ ∆ ∆
In relating this last set of equations to the defining equations for the z parameters, it is clear that
∆ 11 ∆ 21 ∆ 12 ∆
z11 = z12 = z 21 = and z 22 = 22
∆ ∆ ∆ ∆
Since for a passive network ∆ 12 = ∆ 21 , it follows that z12 = z 21 , the network is then reciprocal.
As an example, let us find the z parameters of the circuit in fig. 5.5. First, the node equations are
I1 = (YA + YC )V1 − YCV2
I 2 = (YB + YC )V2 − YCV1
Fig. 5.5
75
The determinant for this set of equations is ∆ = YAYB + YAYC + YBYC in terms of ∆ , the open-circuit
parameters for the circuit are
YB + YC YC YC YA + YC
z11 = z 21 = z12 = z 22 =
∆ ∆ ∆ ∆
5.2.2. Short circuit admittance parameters (Y- Parameters)
If we choose V1 and V2 as independent variables, we can characterize the network by
 I1   y11 y12  V1 

I  = y y 22  V2 
 2  21
Where y11 , y12 , y 21 and y 22 are called y-parameters or short circuit admittance parameters. The
y − parameters are expressed explicitly as
I1 I1 I2 I1
y11 = , y12 = , y 21 = and y 22 =
V1 V2 =0
V2 V1 =0
V1 V2 =0
V1 V =0
1
The reason that the y − parameters are also called short-circuit admittance parameters is now apparent. In
obtaining y11 and y 21 , the 2 − 2' port must be short circuited, and when we find y 22 and y12 , the 1− 1' port
must be short circuited, as shown in fig. 5.6. When y11 = y22 or z11 = z 22 , the network is symmetrical.
y11 and y 22 are short circuit driving point admittances at the two port networks and y 21 & y12 are short
circuit transfer admittances. In particular y 21 is the forward transfer admittance i.e. the ratio of current
response in port-2 to voltage response in port 1 and y12 is reverse transfer admittance.
Fig. 5.6 Determination of y-parameters
Suppose we were to write a set of mesh equations for the general two-port network. Then the voltages V1
and V2 would become independent sources, and the currents I1 and I 2 would be just two of the
dependent mesh currents. The general set of mesh equations are
76
V1 = m11 I1 + m12 I 2 + m13 I 3 + ⋅ ⋅ ⋅ + m1k I k

V2 = m21 I1 + m22 I 2 + m23 I 3 + ⋅ ⋅ ⋅ + m2k I k
0 = m31I1 + m32 I1 + m33 I 3 + ⋅ ⋅ ⋅ + m3k I k
M M M
0 = mk1 I1 + mk 2 I 2 + mk 3 I 3 + ⋅ ⋅ ⋅ + mkk I k
Where mii represents the sum of the impedances in the i th mesh and mij is the common impedance
between mesh i and mesh j , we note here again that for an R-L-C network, mij = m ji for all i and j .
Thus reciprocity holds. Solving the set of meshes equations for I 1 and I 2 using Cramer’s rule, we obtain
the following equations.
∆ 11 ∆ ∆ ∆
I1 = V1 + 21 V2 and I 2 = 12 V1 + 22 V2
∆ ∆ ∆ ∆
This equation define the short-circuit admittance parameters as
I 1 = y11V1 + y12V2
I 2 = y 21V1 + y 212V2
∆ ij
where yij = for all i,& j .
∆
Fig. 5.7 shows an equivalent circuit model of y-parameter equations.
Fig. 5.7Equivalent circuit model for y-parameter equation
As an example, let us obtain the y − parameters of the circuit depicted in fig. 5.5. To obtain y11 and y 21 ,
we short circuit terminals 2 − 2' . We then have
y11 = YA + YC , y 21 = −YC
We next short-circuit terminals 1− 1' to obtain y 22 = YB + YC , y12 = −YC
Let us find the y parameters for the bridged-T circuit given in fig.5.8. The mesh equations for the circuit
are
77
1  1
V1 =  + 1 I1 + I 2 − I 3
s  s
1  1
V2 = I1 +  + 1 I 2 + I 3
s  s
1 1 1 
0 = − I1 + I 2 + 2 + 1 I 3
s s s 
In straightforward fashion we obtain
2(2 s + 1) 2s 2 + 4s + 1 2s 2 + 2s + 1
∆= , ∆11 = ∆ 22 = , ∆12 = ∆ 21 = −
s2 s2 s2
Fig. 5.8
The short-circuit parameters
2s 2 + 4s + 1 2s 2 + 2s + 1
y11 = y 22 = y 21 = y12 = −
s (2s + 1) s (2 s + 1)
5.2.3. Hybrid Parameters ( h -parameters)

A set of parameters that is extremely useful in describing transistor circuits are the h parameters given
by the equations
V1 = h11 I 1 + h12V2
I 2 = h21 I 1 + h22V2
The individual parameters are defined by the relationships
V1 V1
h11 = h12 =
I1 V2 = 0
V2 I1 = 0
I2 I2
h21 = h12 =
I1 V2 = 0
V2 I1 = 0
78
It can be seen that h- parameters are interpreted under mixed set of terminal conditions, some of them
under open circuit and some under short circuit conditions. h11 and h21 are short-circuit type parameters,
and h12 and h22 are open-circuit type parameters. The parameter h11 can be interpreted as the input
impedance at port 1 with port 2 short circuited. It is easily seen that h11 is merely the reciprocal of y11 i.e.,
1 1
h11 = . The parameter h22 is an open-circuit admittance parameter and is related to z 22 by h22 = .
y11 z 22
Both the remaining h -parameters are transfer functions; h21 is a short-circuit current ratio, and h12 is an
open-circuit voltage ratio. Their relationships to z and y-parameters is discussed later in this chapter.
Fig. 5.7 shows an equivalent circuit model of y-parameter equations.
Fig. 5.9 Equivalent circuit of h-parameter equation

For the circuit in fig. 5.5, the h -parameters are
1 YC YC YAYC
h11 = , h12 = , h21 = − , h22 = YB +
YA + YC YA + YC YA + YC YA + YC
Observe that for the circuit, h21 = −h12 . This is the reciprocity condition for the h -parameters and can
be derived from their relationships to either the z or y parameters.
Next let us consider the h -parameters of an ideal device called the negative impedance converter (NIC),
which converts positive load impedance into a negative impedance at its input port. Consider the NIC
with load impedance Z L shown in fig, 5.10. Its input impedance is
V V
Z in = −Z L , which can be rewritten as 1 = 2
I1 I 2
The following voltage-current relationships hold for the NIC.
V1 = kV2 , I1 = kI 2 .
If we interpret these equations using h parameters, we arrive at the following conditions.
1
h11 = h22 = 0 , h12 = =k
h21
79
Fig. 5.10 Negative impedance converter with load impedance
We see that since h12 ≠ − h21 , the NIC is nonreciprocal. In matrix notation, the h matrix of the NIC is
 h11 h12   0 k 
h  = 1 
 21 h22   k 0 
The NIC is a convenient device in the modeling of active circuits. It is not, however, a device that exists
only in the imagination. Practical realizations of NIC’s have been achieved using transistors.
I 1 = g11V1 + g12 I 2 
 g − parameters
V2 = g 21V1 + g 22 I 2 
5.2.4. Transmission Parameters (ABCD Parameters)

Let us take as the dependent variables the voltage and current at the port 1, and define the following
equation.
V1   A B   V2 
 I  = C C  − I 
 1   2 
This matrix equation defines the A, B, C, D parameters, whose matrix is known as the transmission
matrix because it relates the voltage and current at the input port to their corresponding quantities at the
output. The reason the current I 2 carries a negative sign is that most transmission engineers like to regard
their output current as coming out of the output port instead of going into the port, as per standard usage.
In explicit form, the ABCD parameters can be expressed as
V1 V1 I1 I1
A= , B=− , C= , D=−
V2 I 2 =0
I2 V2 = 0
V2 I 2 =0
I2 V2 =0
From these relations we see that A represents an open-circuit voltage transfer function; B is a short-
circuit transfer impedance; C is an open-circuit transfer admittance; and D is a short-circuit current ratio.
Note that all four parameters are transfer functions so that the term transmission matrix is a very
appropriate one.
80
Let us describe the short-circuit transfer functions B and D in terms of y parameters, and the open-circuit
transfer functions A and C in terms of z parameters. Using straight forward algebraic operations, we
obtain
Z11 1 1 Y11
A= , B=− , C= , D=−
Z 21 Y21 Z 21 Y21
For the ABCD parameters, the reciprocity condition is expressed by the equation
A B
det   = AD − BC = 1
C D 
Let us find, as an example, the ABCD parameter for the ideal transformer in fig. 5.11, whose defining
equations are
V1 = nV2 I1 =
1
(− I 2 )
n
Fig. 5.11
If we express the above equation in matrix form, we have
V1  n 0  V 
V  = 0 1  2 
−I
 2   n   2 
So that the transmission matrix of the ideal transformer is
 A B  n 0
C D  = 0 1
   n 
Note, incidentally, that the ideal transformer does not possess an impedance or admittance matrix because
the self- and mutual inductances are infinite.
For the ideal transformer terminated in load impedance shown in fig. 5.12, the following set of equations
apply.
81
V2
V1 = nV2 I1 =
nZ L
Taking the ratio of V1 to I 1 , we find the input impedance at port 1 to be
V1
Z1 = = n2ZL
I1
Thus we see that an ideal transformer is an impedance transformer. If the load element were an inductor
L (fig. 5.12 (b)), at port 1 we would see an equivalent inductor of value n 2 L. Similarly, a capacitor C at
the load (fig. 5.12 (c)) would appear as a capacitor of value C / n 2 at port 1.
Fig. 5.12
As a second example indicating the use of the transmission matrix in network analysis, consider the
ABCD parameter of the circuit in fig. 5.5.
YB + YC 1 YAYB + YBYC + YAYC Y +Y

A= , B= , C= , D= A C
YC YC YC YC
If we check for reciprocity of the above equations, we see that
AD − BC =
(YA + Yc )(YB + Yc ) − (YAYB + YBYC + YAYC ) YC2
= =1
2 2
Y C YC
So that it is reciprocal.
82
To conclude this section, the possible combinations generated by the four variables, taken two at a time is
six. We have discussed four of them in the previous section. The remaining two are the inverse h-
parameters or g-parameters and the inverse ABCD-parameters or A' B' C ' D' -parameters. The six two-
port parameters are summarized in table 5.1 below.
Table 5.1: Summary of two-port parameters

Name Function
Matrix Equation
Express In terms of
Open circuit impedance [Z ] V1 ,V2 I1 , I 2 V1   Z11 Z12   I1 

V  = Z  
 2   21 Z 22   I 2 
Short-Circuit admittance [Y ] I1 , I 2 V1 ,V2  I1  Y11 Y12  V1 
 I  = Y  V 
 2   21 Y22   2
Transmission or Chain [T ] V1 , I1 V2 ,− I 2 V1   A B   V2 
 I  = C D − I 
 1   2 
Inverse Transmission [T ] V2 , I 2 V1 ,− I1 V2   A' B'   V1 
 I  = C ' D' − I 
 2   1 
Hybrid [h] V1 , I 2 I1 , V2 V1   h11 h12   I1 
 I  = h h22  V 
 2   21  2
Inverse Hybrid [g ] I1 , V2 V1 , I 2  I 1   g11 g12  V1 
V  =  g g 22   I 2 
 2   21
5.3. Relationships between two port Parameters
The relationships between two-port parameters are quite easily obtained because of the simple algebraic
1 1
nature of the two-port equations. For example, we have seen that h11 = and h22 = . To drive
y11 z 22
h22 in terms of open circuit parameters, consider the z parameter equations when port 1 is open
V1 z12
circuited I1 = 0 : V1 = z12 I 2 , V2 = z 22 I 2 Therefore, we have h12 = I1 = 0 =
V2 z 22
83
Similarly, since h21 is defined as short-circuit type parameter, we drive h21 in terms of y parameters as
y21
h21 = We can express all the h parameters as functions of the z parameters or y parameters alone.
y11
An easy way to accomplish this task is by finding out what the relationships are between z and
y parameters themselves. Certainly, by their very nature, the z and y parameters are not simply
reciprocals of each other (as the novice might guess), since one set of parameters is defined for open-
circuit conditions and the other for short circuit. The z and y relationships can be obtained very easily by
using matrix notation. If we define the z matrix as
[Z ] = 
z11 z12   y11 y12 
 And the y matrix as [Y ] = 
 z 21 z 22   y 21 y 22 
In simplified notation we can write the two sets of equations as [V ] = [Z ][I ] and [I ] = [Y ][V ]
Replacing [I ] by [Y ][V ] from equation above, we obtain [V ] = [Z ][Y ][V ] so that the product [Z ][Y ] must
yield the unit matrix [U ] . The matrices [Y ] and [Z ] must therefore be inverses of each other, that is,
[Z ]−1 = [Y ] and [Y ]−1 = [Z ]

From the relationship, we can find the relations between the individual z and y parameters.
z 22 z11 z12 z 21
y11 = y 22 = y12 = − y 21 = −
∆z ∆z ∆z ∆z
where ∆ z = z11 z 22 − z12 z 21 ; and

y 22 y11 y12 y 21
z11 = z 22 = z12 = − z 21 = −
∆y ∆y ∆y ∆y
where ∆ y = y11 y 22 − y12 y 21 .
Using these identifies, we can drive the h or ABCD parameters in terms of either the z or y parameters.
Table 5.2 provides a conversion table to facilitate the process. Note that in the table ∆ T = AD − BC.
84
Table 5.2: Parameter Conversion Matrix
5.4. Transfer Functions Using Two-Port Parameters

In this section we will examine how to determine driving point and transfer functions of two-port by use
of two-port parameters. These functions fall into two broad categories. The first applies to two-ports
without load and source impedances. These transfer functions can be described by means of z or
y parameters alone. For example, let us derive the expressions for the open-circuit voltage ratio V2 V1 by
using z -parameters first and y parameters next. Consider the z parameter equations for two-port
network. When port 2 is open circuited,
V2 = z 21 I 1 V1 = z11 I1
If we take the ratio of V2 to V1 , we obtain
V2 z 21
=
V1 z11
85
By letting I 2 of the second y parameter equation go to zero, we derive the open-circuit voltage ratio as
V2 y
= − 21
V1 y 22
In similar manner we can derive the short-circuit current ratio of two-port as

I 2 y 21 I2 z
= and = − 21
I 1 y11 I1 z 22
The open and short-circuit transfer functions are not those we usually deal with in practice, since there
are frequently source and load impedances to account for. The second category of tow-port transfer
functions are those including source or load impedances. These transfer functions are functions of the
two-port parameters z , h, or y and the source and /or load impedance.
I2
For example, let us derive the transfer admittance of two-port network that is terminated in
V1
resistor of R ohms, as given in fig. 5.13. For this two port network, the following equations apply.
I 2 = y 21V1 + y 22V2 V2 = − I 2 R
Fig 5.13 Two-port network terminated in load resistance R
By eliminating the variable V2 , we obtain the transfer admittance as
I2 y 21 R
Y21 = =
V1 y 22 + 1 / R
Note that Y21 and y 21 are not the same. Y21 is the transfer admittance of the two-port network terminated
in a resistor R, and y 21 is the transfer admittance when port 2 is short circuited . We must be careful to
make this distinction in other cases of a similar nature.
In order to solve for transfer functions of two-ports terminated at either port by an impedance Z L , it is
convenient to use the equivalent circuit of the two port network given in terms of its z parameter. The
equivalent voltage sources z12 I 2 and z 21 I 1 are called controlled voltage sources because they depend
86
upon a current or voltage somewhere in the network. Similarly the current sources y12V2 and y 21V1 are
controlled sources. For the circuit in fig. 5.24, let us find the transfer impedance Z 21 = V2 / I 1 , with port 2
terminated in load impedance, Z L .
Fig 5.14 Two-port equivalent circuit terminated in Z L
If we write the mesh equation for the I 2 mesh we have
− z 21 I 1= ( z 22 + Z L ) I 2 but V2 = − I 2 Z L
V2 z 21 Z L
Thus, Z 21= I = z + Z
1 22 L
It also is clear that the current- ratio transfer function for the terminated two-port network is
I2 − z 21
= . In similar fashion, we obtain the voltage-ratio transfer function for the circuit represented
I 1 z 22 + Z L
in the fig. 5.15 as:
V2 y 21
=−
V1 Y 2+ y 22
Fig. 5.15 Two port equivalent
87
V2
Next, suppose we are required to find the transfer function for the two –port network terminated at
Vg
both ends, as shown in fig. 5.16. We first write the two mesh equations
Vg = ( R 1 + z 11 ) I 1+ z 12 I 2
0 = z 21 I 1+ ( z 22 + R 2 ) I 2
Next, we solve for I2 to give
V2 z 21
I 2=
( R1 + z11 )( R2 + z 22 ) − z12 z 21
From the equation V2 = R2 I 2 , we may now arrive at the following solution.

V2 RI z 21 R2
=− 2 2 =
Vg Vg ( R1 + z 11 )( R 2 + z 22 ) − z 21 z12
Fig. 5.16 Two-port network terminated at both ports
Note that the equivalent circuits of the two-port networks are not unique. Two other examples are given
in fig. 5.17 and fig.1.18. Observe that the controlled sources are nonzero in these equivalent circuits only
if the circuit is nonreciprocal.
Fig. 5.17 Two-Port equivalent circuit with one controlled-voltage source
88
Fig. 5.18 Two port equivalent circuit with one controlled-current source
Finally, let us consider the hybrid equivalent circuit shown in fig. 5.19. Observe the voltage–controlled
source h21V2 at input port and the current controlled source h11 I1 at the output port. Let us find the input
impedance Z in . The pertinent equations are
V 1=h11 I 1+ h12V2
V 2= − Z L I 2= −(h 21 I 1+ h 22V2 ) Z I )
Solving this equation for V2 , we find
h Z I
V 2= − 21 L 1
1 + h 22 Z L
By substitution, we have
 h h Z 
V 1=  h11 − 12 21 L  I 1
 1 + h 22 Z L 
So that
V h h Z
Z in = 1 = h11 − 12 21 L
I1 1 + h22 Z L
Note that h11 has the dimensions of impedance h22 is an admittance, and h12 , h21 are dimensionless since
they represent voltage and current ratios, respectively.
Fig. 5.19 Hybrid Equivalent Circuit
89
5.5. Interconnection of Two-Port Networks

In this section we will consider various interconnections of two–port networks. We will see that when a
pair of two-ports is cascaded, the overall transmission matrix is equal to the product of the individual
transmission matrices of the two-port networks. When two-port networks are connected in series their z-
matrices add; when they are connected in parallel, their y matrices add.
First let us consider the case in which we connect a pair of two ports N a and N b in cascade or in tandem,
as shown in fig. 5.20.
Fig. 5.20 Cascade connection of two port networks

V 2 a  V1b 
We see that I  = I 
 2 a   1b 
The transmission matrix equation for Na is
V 1   Aa Ba  V 2 a 
 I  = C D a  − I 2 a 
 1  a
Correspondingly for Nb we have
V 2 a  V1b   Ab Bb  V 2 
− I  =  I  = C Db  − I 2 
 2 a   1b   b
Substituting the second matrix into the first, we obtain
V 1   Aa Ba   A b Bb  V 2 
I  =  ⋅ .
D b  − I 2 
 1  C a Da  C b
We see that the transmission matrix of the overall two two-port networks connected in cascade is simply
the product of the transmission matrices of the individual two ports.
As an example, let us calculate the overall transmission matrix of a gyrator in tandem with a T network
shown in fig. 5.21. The ideal gyrator is an impedance inversion device whose input impedance Z in is
related to its load impedance Z L by
a2
Z in = a 2YL =
ZL
90
The constant a in above is defined as the gyration resistance. If we regard the gyrator as a two-port
network, its defining equations are
V2
V1 = a ( − I 2 ) , I 1 =
a
0 a
So the transmission matrix of the gyrator is  1 
 a 0

We see that for the gyrator AD − BC = −1. Therefore, the gyrator is a nonreciprocal device, although it is
passive.
Fig. 5.21 Gyrator in tandem with T circuit

 z a + zb z a z b + zb z c + z a z c 
 z zb 
For the T-network the transmission matrix is  b  . Therefore, the overall
 1 zb + zc 
 zb zb 
transmission matrix of the configuration in fig. 5.21 is obtained by the product of the individual
transmission matrices
 za + zb z a zb + zb zc + z a zc   a a( z b + z c ) 
A B 0 a  z zb   z zb 
C D  =  1  = 
b b
   a 0  1 zb + zc +
  a zb
z z a zb + zb zc + z a zc 

 z b zb   az b az b 
If we check the configuration in fig. 5.21 for reciprocity, we see that for its transmission matrix
AD − BC = −1 i.e., AD − BC ≠ 1 and hence it is not reciprocal. However, the T-network is reciprocal.
We thus see that any reciprocal network connected in tandem with a gyrator yields a configuration that is
nonreciprocal.
Next consider the situation in which a pair of two-port networks N a and N b are connected in parallel, as
shown in fig. 5.22.
91
Fig. 5.22 Cascade connection of two port networks
Let us find the y parameters for the overall two-port network. The y-matrix equations for the individual
two-ports are
 I1a   y11a y12 a  V1a  I  y y12b  V1b 
I  =  y    and  1b  =  11b  
 2 a   21a y 22 a  V2 a   I 2b   y 21b y 22b  V2b 
From fig. 5.22 see that the following equations must hold.
V1 = V1a = V1b , I1 = I 1a + I1b
V2 = V2 a = V2b , I 2 = I 2 a + I 2b
In matrix notation, the sum of the individual [I i ] matrices of two-port networks in parallel must equal the
[I ] matrix of the overall two-port network. Thus we have

 I1   I 1a   I 1b   y11a y12 a   y11b y12b  V1 
 I  =  I  +  I  =  y +
y 22 a   y 21b

y 22b  V2 
 2   2 a   2 b   21a
So that the y parameters of the overall two-port network can be expressed in terms of the y parameters of
the individual two-port as
 y11 y12   y11a y12 a   y11b y12b   y11a + y11b y12 a + y12 a 
= + =
y
 21 y 22   y 21a y 22 a   y 21b y 22b   y 21a + y 21b y 22 a + y 22b 
If we connect two-port networks in series as in fig. 5.23, in similar way as derived above, we can express
the z -parameters of the overall two-port network in terms of z -parameters of individual two-port
networks as
 z11 z12   z11a + z11b z12 a + z12 a 

=
z
 21 z 22   z 21a + z 21b z 22 a + z 22b 
92
Fig. 5.23 Series connection of two-port networks
Note that in connecting two-port networks in series or in parallel, we must be careful that the individual
character of a two-port network is not altered when connected in series or parallel with another two-port
network. For example, when we connect the two-port networks in parallel as shown in fig. 2.24, the
impedances Z 7 and Z 8 will be short circuited. Therefore, to ensure that a two port network does not
interfere with the internal affairs of the other, ideal transformers are used to provide the necessary
isolation.
Fig. 2.24
We may summarize the interconnection of two port networks by the following three points:
1. When two-ports are connected in parallel, find the y parameters first, and, from y parameters,
derive the other two-port parameters.
2. When two-ports are connected in series, it is usually easiest to find the z parameters.
3. When two-ports are connected in tandem, the transmission matrix is generally easier to obtain.
As a final example, let us find the y parameters of the bridge-T circuit in fig. We see that the bridge-T-
circuit would be decomposed into a parallel connection of two-port networks as shown in 5.25. Our task
is to first find the y-parameters of the two-port networks N a and N b . Since both N a and N b are
symmetrical networks, y11 = y 22 and y12 = y 21 .
93
Fig. 5.25
The y parameters of N b are obtained by inspection and are
1
y12 b = y 21b = −
2
1
y11b = y 22b =
2
N a is a T-circuit so that the z-parameters can be obtained by inspection.
s +1
z11a = z 22 a =
s
z12a = z 21a =1
Using the conversion matrix, the y-parameters can be found as
z 22a s (s + 1)
y11a = y 22a = =
∆z a 2s + 1
z12 a s2
y12 a = y 21a = =−
∆z a 2s + 1
Since both N a and N b are symmetrical networks, the overall network is also symmetrical and its y-
parameters y11 = y 22 and y12 = y 21 . We know that for parallel connection two two-port networks, the y-
parameters of the overall network is equal to the sum of the respective y-parameters of constituent two-
port networks, i.e.,
 y11 y12   y11a + y11b y12 a + y12 a 
y =
 21 y 22   y 21a + y 21b y 22 a + y 22b 
s (s + 1) 1 2s 2 + 4s + 1
y11 = y11a + y11b = + = = y 22
2s + 1 2 2(2s + 1)
s2 1 2s 2 + 2s + 1
y12 = y12a + y12b = − − =− = y 21
2s + 1 2 2(2s + 1)
94
5.6. Elements of Transfer Functions Synthesis
As we have defined in chapter 2 a transfer function is a function which relates the current or voltage at
one port to the current or voltage at another port. So far in this chapter we have discussed various
descriptions of two port networks in terms open circuit parameters Zij and short circuit parameters Yij and
other two port parameters. As derived in this section 5.4 transfer functions can be expressed in terms of
theses parameters.
Examples
1) in terms of the open circuit impedance parameters, voltage ratio transfer function is given by
2) In terms of the short-circuit parameters, the voltage ratio is shown to be
When the network is terminated at port two by a resistor R as shown in figure 5.26, the transfer
impedance of the overall network is
And the overall transfer admittance is
( )
Fig. 5.26 a Fig. 5.26 b

When both ports are terminated as figure 5.27, the voltage ratio transfer function is
Other transfer functions such as current ratios in terms of open circuit and short circuit parameters.
fig. 5.27
95
5.6.1. Properties of Transfer Immitance
In this chapter we synthesize transfer immittances Z21 and Y21 for passive reciprocal networks. Therefore,
knowing properties of such network functions is essential. Let say the transfer function is denoted by
H(s). Without going to proof , we present properties of H(s) as follow:
1. H(s) is real for real s. This property is satisfied when H(s) is a rational function with real
coefficients.
2. T(s) has no poles in the right-half plane and no multiple poles on the jw axis. If T(s) is given as
T(s) = P(s)/ Q(s), the degree of P(s) cannot exceed the degree of Q(s) by more than unity. In
addition, Q(s) must be a Hurwitz polynomial.
3. Suppose P(s) and Q(s) are given in terms of even and odd parts, that is,
Mi(s) even and Ni(s) is odd. Then H(jw) is
the amplitude response of H(jw) is
And is an even function of w.

The phase response is
Now let us discuss some specific properties of open circuit and short circuit parameters
a. The poles of Z21( S ) are also the poles of Z11 ( S ) and Z22( S ) . However, not all the poles of Z11(S)
and Z22( S ) are the poles of Z21(S) .
If there is no cancellation between each numerator and denominator of Z11, Z22 & Z21, then the poles are
the roots of the determinant , and all three functions have the same poles. Consider the two port network
described by the black box in figure 5.28 (a). Let Z11’, Z22’ and Z21’ be the Z – parameters of the network.
If we attach the impedances Z1 and Z2 to ports one and two as shown in figure 5.28 (b), then the Z –
parameters for the two port network are
96
Fig. 5.28
It is clear that the poles of Z11 include the poles of Z1; the poles of Z22 include the poles of Z2, However,
the poles of Z12 include neither the poles of Z1 nor Z2 Consequently, we see that all the poles of Z21 are
also poles of Zll and Z22 . The reverse is not necessarily true.
b. Similarly, the poles of Y21(s) are also the poles Y11(s) and Y22(s). However, not all of the poles of
Y11(s) and Y22(S) are the poles of Y21(s). This can be examining the two port network in figure 5.29
fig. 5.29
The Y parameters are
As an example consider the network in figure 5.30
fig. 5.30
97
It is clear that from the above equations that Y11(s) and Y22(s) have poles at s = 0 and s =∞, whereas
Y21(s) only has a pole at s =∞.
c. Suppose Yll(S) , Y22(s), and Y21(S) all have poles at S = S1. Let us denote by k 11 the residue of the pole at
Sl of the function Y11(S), The residue of the pole S = S1 of Y22(s) will be denoted as k22 and the residue of
the same pole of Y12(S) will be denoted as k12• Without going into the proof, a general property of L-C,
R-C, or R-L two-port networks is that
This equation is known as the residue condition. For example, for the L-C network in Fig.5.30, the
residue condition applied to the pole at s = ∞ gives 3 x 3 – 32 = 0; whereas for the pole at s = 0, we have
2 x 4 – 02 = 8 > O. Thus we see that the residue condition is fulfilled for both poles.
5.6.2. Zeros of Transmission

A zero of transmission is a zero of a transfer function. At a zero of transmission, there is zero output for
an input of the same frequency. For the network in figure 5.31, the capacitor is an open circuit at s = 0, so
there is a zero of transmission at s = 0. For the networks in figure 5.32 (a) and (b), the zero of
transmission occurs at s = ± .
Fig. 5.31
Fig. 5.32(a)
Fig. 5.32 (b)

For the network in Fig. 5.33, the zero of transmission occurs at s = -1/RC.
Fig. 5.33
98
In general, all the transfer functions of a given network have the same zeros of transmission, except in
certain special cases. For example if Z12(S) has a zero of transmission at S = S1 than Y12(S), V2(s)/ V1(s),
etc., will also have a zero at S = S1, This fact is clearly seen when we examine the relationships between
the transfer functions. For example, we have
5.6.3. Synthesis of Terminated Networks
In this section we will consider synthesis of two port networks terminated with 1Ω resistor and constant
resistor R at one port (normally port two).
5.6.4. Synthesis of Y21 & Z21 with a 1 Ω Termination
Here we consider the synthesis of an LC ladder network with 1Ω resistor termination to meet a specified
transfer impedance Z21 or a transfer admittance Y21.
Fig. 5. 34
In terms of the open circuit and short circuit parameters of LC network, the transfer functions Z21(s) and
Y21(s) are given by
We know that the ratio of even to odd or odd to even parts of a Hurwitz polynomial yields all positive
quotients on continued fraction expansion. This quotients in turn can be associated with reactances.
Therefore it is clear that the ratio of even to odd or odd to even parts of a Hurwitz polynomial is an LC
driving point function.
Another point to be taken in mind is the fact that the open circuit transfer impedance Z21 or the short
circuit transfer admittance Y21 of an LC is an odd function.
Now suppose that the transfer admittance Y21(s) is given as quotient of two polynomials
99
Where P(s) is either even or odd. Now, how to determine the short circuit parameters Y21 and Y22 from
above equation to get it into the form
Which is that of a two port network terminated by 1Ω resistor. The answer is quite simple. We divide
both the numerator P(s) and the denominator Q(s) by M(s) or N(s) , the even or the odd part of Q(s).
Since Y21 must be odd, if P(s) is even, we divide by N(s) so that
From this expression we obtain,
On the other hand, if P(s) is odd, we divide by M(s) so that
We assume that P9s), M(s) and N(s) do not possess common roots. For our purposes, we will consider
only synthesis of Y21 and Z21 with zeros of transmission either at s = 0 or s = ∞. In ladder network, a zero
of transmission at s = 0 corresponds to a single capacitor in a series branch or a single inductor in a
parallel branch.
In terms of transfer impedance
the presence of n zeros of Z21(S) at s = 0 implies that the coefficients an-1>an-2, ••• , a1> ao are all zero.
The number of zeros of Z21(S) at s = ∞ is given by the difference between the highest powers of the
denominator and the numerator, m - n. We know that n can exceed m by at most unity, while m can be
greater than n by more than one. For example, if m - n = 2, and n = 3 with an-1>…> a1> ao = 0, we know
that the transfer function has three zeros of transmission at s = 0 and two zeros of transmission at s = ∞.
100
We can now proceed with the matter of synthesis. Consider the following example
We see that all three zeros of transmission are at s = ∞. Since the numerator P(s) is a constant, it must be
even, so we divide by the odd part of the denominator by s3 + 4s. We obtain
We see that both Z21 and Z22 have the same poles. Our task is thus simplified to the point where we must
synthesize Z22 so that the resulting network has the transmission zeros of Z21. This requires that we first
examine the possible structures of the networks which have the required zeros of transmission and see if
we can synthesize Z22 one of those forms. For the example that we are considering, a network which gives
us three zeros of transmission at s = ∞.
We can synthesize Z22 in this structure by following continued fraction expansion of
Z22 is synthesized from 1Ω termination toward the input end, the final network takes the form shown
below
Fig. 5.35
Examining the synthesized network more closely, we see it takes the form of low pass filter. Thus the
specification of all zeros at s = ∞ is equivalent to the specification of a low pass filter.
101
Exercise
1. Synthesize the transfer impedance given by the following equation:
2. Synthesize the transfer admittance function
102
Chapter 6
Topics in Filter Design

6.1. Introduction
Filters are network which passes signals with a specific frequency range and rejects or attenuates the
signals whose frequencies are outside of the desired frequency range. Filter may be categorized as low
pass, high pas, band pass and band rejection (elimination) filters depending on the frequency range they
pass. They are also classified as passive and active filters depending on constituent circuit components.
6.1.1. The Filter Design Problem

In the preceding chapter we examined different methods of synthesizing a driving point or transfer
function H(s). Most problems have as their initial specification amplitude or phase characteristic, or an
impulse response characteristic instead of the system function H(s). The problem is to obtain a realizable
system function from the given amplitude or phase characteristic. For example, a typical design problem
might be to synthesize a network to meet a given low pass filter characteristic. The specifications might
consist of the cutoff frequency ω C , the maximum allowed deviation from a prescribed amplitude within
the pass band, and the rate of fall off in the stop band. We must then construct the system function from
the amplitude specification. We will consider selected topics in approximation theory and then present
examples of filter design where both the approximation and the synthesis problems must be solved.
6.1.2. The Approximation Problem in Network Theory

The essence of the problem is the approximation of a given function f(x) by another function
f a ( x; α1 ,....,α n ) in an interval x1 ≤ x ≤ x 2 the parameter α1 ,....,α n in the approximating function are fixed
by the particular error criterion chosen. When we let ∈= f (x) − f a (α , α 1 ,....,α n ), the following error
criteria most common:
x2 2
1. Least Squares. The value of I (α1 ,....,α n ) is minimized where I (α1 ,....,α n ) = ∫x ∈ ω ( x)dx
1
And ω (x ) is a weighting function which stresses the error in certain subintervals.

2. Maximally flat. The first n-1 derivatives of E are made to vanish at x=xo.
3. Chebyshev .the value of µ is minimized in the interval x1 ≤ x ≤ x 2 where µ ε max
103
4. Interpolation. The value of ε is made to vanish at a set of points in the interval x1 ≤ x ≤ x 2 .
After an error criterion is chosen, we must determine the particular form of the approximating function.
This depends up on whether we choose to approximate in the time or frequency domain suppose f(x)
represents a magnitude function in the frequency domain and the approximating function is to be rational
in ω 2 ; then
α1 + α 3 x + α 5 x 2 + ⋅ ⋅ ⋅
f a (α , α 1 ,...., α n )= Where x = ω 2 in addition, the values of α k must be
α 2 + α 4 x + α 6 x + ....
2
restricted to insure that f a (α ;α 1 ,....,α n ) ≥ 0 in the f(x) might represent an impulse response of a system
to be synthesized. In the case of an R-C transfer function, we have
f a (α ;α 1 ,....,α n ) ≥ 0 = α1 ∈α x +α 3 ∈α x +...
2 4
Where x=t. since an R-C transfer function must have its poles on the negative real axis , the values of
α k , k even, are restricted to negative real numbers. The keystone of any approximation problem lies in
the choice of a suitable error criterion subject to reliability restrictions. The problem can be simplified
when some of the α ' s are assigned before applying the error criteria. All the error criteria cited, except
the Chebyshev, can then be reduced to a set of linear algebraic equations for the unknowns α1 ,..., α n .
6.1.3. Time–domain approximation

The principal problem of time-domain approximation consists of approximating an impulse response
h(t ) by an approximating function h * (t ) such that the squared error ε = ∫ [h(t ) − h * (t )] dt is a

∞ 2
0
minimum. A generally effective procedure in time –domain approximation utilizes orthonormal

n
function φk (t ) . The approximating function h*(t) takes the form h * (t ) = ∑ α k φ k (t )
k =1
2
∞  n

So that the error ε =∫
Ο 

h (t ) − ∑
k =1
α k φ k (t ) dt

is minimized when
∞
α k = ∫ h(t )φk (t )dt k = 1,2,..., n
0
sk t
If the orthonormal set is made up of a sum of exponentials e , then the approximate impulse response.
n n
αk
h * (t ) = ∑ α k e sk t
Has a transform H * (s) = ∑
k =1 k =1 s − sk
104
Reliability is insured if in the orthonormal set {α k }

e sk t , Re sk ≤ 0 ; k = 1,2,⋅ ⋅ ⋅, n . Synthesis then proceeds
from system function H*(s).
6.1.4. Frequency-Domain Approximation

In frequency-domain approximation the principal problem is to find a rational function H(s) whose
magnitude H ( jω ) approximates the ideal low–pass characteristic in figure below according to a
predetermined error criterion. In the next few sections we examine several different ways to approximate
the ideal low-pass: the maximally flat or Butterworth approximation, the equal –ripple or Chebyshev
approximation, and the optimal or Legendre approximation, elliptic or Bessel.
Figure: ideal low-pass filter characteristic.

Another major problem is that of obtaining transfer function H(s), whose phase is approximately linear or
whose delay is approximately flat over a given range of frequencies. Here again there are two different
methods: the maximally flat or the equal –ripple methods. Our discussion will center on the maximally
flat method/the joint problem of approximating both magnitude and phase over a given frequency range
is possible.
6.2. The Maximally Flat Low-Pass Filters Approximation.

In previous chapter we saw that the ideal low–pass filter in figure above is not realizable because its
associated impulse response is zero for t < 0 however, if we use a rational function approximation to this
low-pass filter characteristic, the Paley- wiener criterion will be automatically satisfied we will therefore
restrict ourselves to rational function approximations. In low pass filter design, if we assume that all the
zeros of the system function are at infinity, the magnitude function takes the general form.
K0
M (ω ) = Where K 0 is the d-c gain constant and f (ω 2 ) is the polynomial to be selected to
[1 + f (ω )]
2 1/ 2
give the desired amplitude response. For example, if f (ω 2 ) = ω 2 n , then the amplitude function
can be written as
105
K0
M (ω ) =
( 1 + ω 2n ) 1
2
We see that M (0) = K 0 , and that M (ω ) is monotonically decreasing with ω . In addition, the 0.707 or 3-
Ko
decible point is at ω = 1 for all n, that is, M (1) = for all n
2
The cut off frequency is thus seen to be ω = 1 the parameter n controls the closeness of approximation in
both the pass band and the stop band. Curves of M (ω ) for different n are shown in figure below. Observe
that the higher n , is the better the approximation.
Figure: Amplitude response of Butterworth low-pass filters

The amplitude approximation of the above equation is called Butterworth or maximally flat response.
The reason for the term ‘maximally flat” is that when we expand M (ω ) in a power series about ω = 0 we
have
35 8 n
M (ω ) = K 0 (1 − 12 ω 2 n + 83 ω 4 n − 165 ω 6 n + ω + ⋅ ⋅ ⋅)
128
We see that the first 2n-1 derivatives of M (ω ) are equal to zero at ω = 0 for ω >> 1, the amplitude
response of a Butterworth function can be written as (with K 0 normalized to be unity)

1
M (ω ) ≈ ω >> 1
ωn
We observe that asymptotically, M (ω ) falls off as ω − n for a Butterworth response in terms of decibels,
the asymptotic slope is obtained as.
106
20 log M (ω ) = −20 n log ω

Consequently, the amplitude response falls asymptotically at a rate 6ndb / octave or 20n db / decade .
One question remains. How do we obtain a transfer function H (s ) from only the amplitude
characteristics M (ω ) ? The procedure is as follow. We first note that the amplitude response M (ω ) and
the complete system function H ( jω ) are related by M 2 (ω ) = H ( jω ) H (− jω )
If we define a new function h( s 2 ) such that h( s 2 ) = H ( s ) H ( − s )
We see that M 2 (ω ) = h(−ω 2 )
From h(−ω 2 ) all we need to do is to substitute s 2 = −ω 2 to give h( s 2 ) . Then we factor h( s 2 ) in to the

product H ( s ) H (− s ) . Since the poles and zeros of H (s ) are the mirror images of the poles and zeros
of H ( − s ) , we simply choose the Hurwitz factors of h( s 2 ) as H (s ) . An example will serve to clarify this
discussion. Consider the third order (n=3) Butterworth response given by
1 1 1
M 2 (ω ) = = We see that h( s 2 ) is h( s 2 ) =
1+ω 6
1 − ( −ω 2 ) 3 1 − (s 2 ) 3
1 1
Factoring h( s 2 ) , we obtain h( s 2 ) = = H ( s) H (− s)
1 + 2s + 2s + s 1 − 2s + 2s 2 − s 3
2 3
1 1
We then have H ( s ) = =
s 3 + 2s 2 + 2s + 1 ( s + 1)(s + 12 + j 3
2 )(s + 12 − j 3 / 2 )
The poles of H(s) and H (-s) are shown in figure below. Observe that the poles of H (-s) are mirror
images of the poles of H(s), as given by the theorem on Hurwitz polynomials.
Figure: Poles of H ( s ) H ( − s ) for an n=3 are Butterworth filter
For a Butterworth response, the poles of H ( s ) H ( − s ) are the roots of (−1) n s 2 n = −1 = e j ( 2 k −1)π
s k = e j [( 2 k −1)π ] n even
k = 0,1,2,⋅ ⋅ ⋅, 2n − 1 . The poles sk are then given by
= e j [( k / n )π ] n odd
107
Or simply by sk = e j [(2 k + n−1)/ 2 n ]π k = 0,1, , 2,⋅ ⋅ ⋅,2 n Expressing s k as

2k + n − 1  2k − 1  π
sk = σ k + jωk , the real and imaginary parts are given by ωk = sin π = cos 2
2n  n 
2k + n − 1  2k − 1  π
σ k = cos π = sin  2
2n  n 
It is seen from the above equations that all the poles of H ( s ) H ( − s ) are located on the unit circle in the s
plane, and are symmetrical about both the σ and jω axes to satisfy reliability conditions, we associate
the poles in the right-half plane with H ( − s ) , and the poles in the left–half plane with H (s ) .
As an example, consider the construction of H (s ) that gives an n=4 Butterworth responses from the
above equations it is seen that the poles are given by sk = e j[( 2 k +3) / 8]π H (s ) is then given as
1
H ( s) =
(s + e j ( )π
5
8
)(s + e ( ) )(s + e ( ) )(s + e
j 7
8
π j 9
8
π j ( 11
8
)π
)
1
If we express sk in complex form and expand, we obtain H ( s ) =
( )
s + 0.76536 s + 1 ( s 2 + 1.84776 s + 1)
2
To simplify the use of Butterworth function, H(s) is given in tables below for n=1 to n=8 in factored form
as in equation above or multiplied out as
1
H (s) = n −1
a n s + a n −1 s
n
+ ... + a1 s + 1
Table 1: Butterwort Polynomials (Factored Form)
1. s + 1
2. s 2 + 2s + 1
3. (s 2
)
+ s + 1 (s + 1)
4. (s 2
)(
+ 0.76536s + 1 s 2 + 1.84776s + 1 )
5. (s + 1)(s 2 + 0.6180s + 1)(s 2 + 16180s + 1)
6. (s 2
)(
+ 0.5176 s + 1 s 2 + 2 s + 1) s 2 + 19318s + 1 ( ))
7. (s + 1)(s 2 + 0.4450s + 1)(s 2 + 1.2456s + 1)(s 2 + 1.8022s + 1)
8. (s 2
)( )(
+ 0.3986s + 1 s 2 + 1.1110s + 1 s 2 + 1.6630s + 1 s 2 + 1.9622s + 1 )( )
108
Table 2: Butterworth polynomials

n a1 a2 a3 a4 a5 a6 a7 a8
1. 1
2. 2 1
3. 2 2 1
4. 2.613 3.414 2.613 1
5. 3.236 5.236 5.236 3.236 1
6. 3.864 7.464 9.141 7.464 3.864 1
7. 4.494 10.103 14.606 14.606 10.103 4.494 1
8. 5.126 13.138 21.848 25.691 21.848 13.138 5.126 1
6.3. Other Low-Pass Filter Approximations

In section 6.2 we examined the maximally flat approximation to a low–pass filter characteristic we will
consider other low pass filter approximants in this section.
6.3.1 The Chebyshev or Equal- Ripple Approximation

We have seen that the maximally flat approximation to the ideal low pass filter is best at ω = 0 whereas
as we approach the cutoff frequency ω = 1 , the approximation which “ripples” about unity in the pass
band and falls off rapidly beyond the cutoff ω = 1 the approximation is equally good at ω = 0 and
ω = 1 and, as a result is called an equal-ripple approximation. The equal–ripple property is brought about
by the use of Chebyshev cosine polynomials defined as
C n (ω ) = cos(n cos −1 ω ) ω ≤1
= cosh(n cosh −1 ω ) ω >1
For n=0 we see that C0 (ω ) = 1 and for n=1, we have C1 (ω ) = ω

Higher order Chebyshev polynomials are obtained through the recursive formula
C n (ω ) = 2ωC n −1 (ω ) − C n −2 (ω )
Thus for n=2 we obtain C 2 (ω ) as C 2 (ω ) = 2ω (ω ) − 1 = 2ω 2 − 1

In the table below Chebyshev polynomials of orders up to n= 10 are given.
Table 3:
n Chebyshev polynomials C n (ω ) = cos( n cos −1 ω )
0 1
1 ω
109
2 2ω 2 − 1
3 4ω 3 − 3ω
4 8ω 4 − 8ω 2 + 1
5 16ω 5 − 20ω 3 + 5ω
6 32ω 6 − 48ω 4 + 18ω 2 − 1
7 64ω 7 − 112ω 5 + 56ω 3 − 7ω
8 128ω 8 − 256ω 6 + 160ω 4 − 32ω 2 + 1
9 256ω 9 − 576ω 7 + 432ω 5 − 120ω 3 + 9ω
10 512ω 10 − 1280ω 8 + 1120ω 6 − 400ω + 50ω 2 − 1
Figure: C3 (ω ) and C 4 (ω ) Chebyshev polynomials

The pertinent properties of Chebyshev polynomials used in the low –pass filter approximation are:
1. The zeros of the polynomials are located in the interval ω ≤ 1 as seen by the plots of C3( ω ) and
C4( ω )in figure above.

2. within the interval ω ≤ 1 , the absolute value of C n (ω ) never exceeds unity, that is Cn (ω ) ≤ 1 for
ω ≤1
3. beyond the interval ω ≤ 1, Cn (ω ) increases rapidly for increasing values of ω
Now, how do we apply the Chebyshev polynomials to the low-pass filter approximation? Consider the
function ∈2 C n (ω ), where ∈ real and small is compared to 1. It is clear that ∈2 C n (ω ), will vary
2 2
between 0 and ∈2 in the interval ω ≤ 1 . Now we add 1 to this function making it 1+ ∈2 C n (ω ), . This new
2
function varies between 1 and 1+ ∈2 , , a quantity slightly greater than unity, for ω ≤ 1 . Inverting this
function, we obtain the function which we will associate with H ( j ω ) 2 ; thus
110
H ( jω ) =
2 1
1+ ∈ C n (ω )
2 2
Within the interval ω ≤ 1 , H ( jω ) oscillates about unity such that the maximum value is 1 and the
2
( )
minimum is 1 / 1+ ∈2 . Outside this interval C n2 (ω ) becomes very large so that as ω increase, a point will
be reached where ∈2 C n2 (ω ) >> 1 and H ( jω ) approaches zero very rapidly with further increase in ω .
2
Figure below shows a Chebyshev approximation to ideal low-pass filter. We see that within the pass band
and 0 ≤ ω ≤ 1, H ( jω ) ripples between the value 1 and (1+ ∈2 ) 2 . The ripple height or distance between
−1
Figure: Chebyshev approximation to low-pass filter

Maximum and minimum in the pass band is given as
1 1
Ripple = 1 − At ω = 1, H ( jω ) is H ( j1) = Because C n (1) = 1
2
(1+ ∈2 ) 12 (1+ ∈ ) 2
2 1
In the stop band that is for ω > 1, asω increase, we reach a point ωκ , where C n (ω ) >> 1 so that
2
1
H ( jω ) ≅ ω > ωκ
∈ C n (ω )
The loss in decibels is given as Loss = −20 log10 H ( jω ) ≅ 20 log ∈ +20 log C n (ω )
But for large ω, Cn (ω ) can be approximated by its leading term 2 n−1 ω n , so that
Loss = 20 log ∈ +20 log 2 n −1 ω n = 20 log ε + 6( n − 1) + 20 n log ω

We see that the Chebyshev response also falls off at the rate of 20ndb/decade after an initial drop of
20 log ∈ +6( n − 1) decibels. However in most applications, ∈ is a very small number so that the 20 log ∈
111
term is actually negative. It is necessary, therefore, to compensate for this decrease in loss in the stop
band by choosing a sufficiently large n.
From the preceding discussion, we see that a Chebyshev approximation depends up on two variables, ∈
and n which can be determined from the specifications directly the maximum permissible ripple puts a
bound on ∈ . Once ∈ is determined any desired value of attenuation in the stop band fixes n .
The derivation of the system function H (s ) from a Chebyshev amplitude approximation H ( jω ) is
somewhat involved and will not be given here instead, we will simply give the results of such a
derivation. First we introduce a design parameter.
Figure: Locus of Poles of Chebyshev filters
βκ = 1n sinh −1 ∈1
Where n is the degree of the Chebyshev polynomial and ∈ is the factor controlling ripple width. The
poles, sk = σ k + jωκ , of the equal- ripple approximant H (s ) are located on an ellipse in the s plane,
given by.
σ k2 ωk2
+ =1
sinh 2 β κ cosh 2 βκ
The major semi axis of the ellipse is on the jω -axis and has a value ω = ± cosh β k , the minor semi axis
has a value σ = ± sinh β k , and the foci are at ω = ±1 . The half power point of the equal ripple amplitude
response occurs at the point where the ellipse intersects the jω axis i.e. at ω = cosh β k . Recall that for
the Butterworth response, the half-power point occurs at ω = 1 let us normalize the Chebyshev poles
s k such that the half-power point also
112
falls at ω = 1 instead of at ω = cosh β k i.e. let us choose a normalizing factor cosh β k Such that the
sk σk jω ∆
s'k = = + = σ ' k + jω ' k
cosh β κ cosh β κ cosh β κ
normalized pole locations s' k are given by
 2k − 1  π
The normalized pole locations can be derived as σ 'κ = tanh β κ sin  2
 n 
 2k − 1  π
ω 'κ = cos 2
 n 
Comparing the normalized Chebyshev pole locations with the Butterworth pole locations, we see that the
imaginary parts are the same, while the real part σ ' k of the Chebyshev pole location is equal to the real
part of the Butterworth poles times the facto tanh β κ . For example, with n=3 and tanh β κ = 0.444, the
butterwort poles are.
s1 = −1 + j 0 s 2,3 = −0.5 ± j 0.866

So that the normalized Chebyshev poles are given by
s1 = −1(0.444) + j 0 = −0.444 + j 0 s2,3 = −0.5(0.444) ± j 0.866 = −0.222 ± j 0.866
Finally, to obtain the demoralized Chebyshev poles, we simply multiply s ' k by cosh β κ ' that is,
s k = (σ ' k + jω ' k )cosh β κ

There is an easier geometrical method to obtain the Chebyshev poles, given only the semi axis
information and the degree n . First we draw two circles, the smaller of radius sinh β κ and the larger of
radius cosh β κ as shown in figure below. Next we draw radial lines according to the angles of
Butterworth poles. Finally we draw vertical dashed lines from the intersection of the smaller circle and
the radial lines and horizontal dashed lines from the intersection of the large circle and the radial lines.
The Chebyshev poles are located at the intersection of the vertical and horizontal dashed lines as shown
in the next figure.
113
Figure: n=3 Chebyshev filter poles

Consider the following example we would like to obtain a system function H (s ) that exhibits a
Chebyshev characteristic with not more than 1- decibel ripple in the pass band and is down at least 20
decibels at ω = 2 . When we design for 1- decibel ripple, we know that at ω = 1 , H ( j1) is down 1 decibel
so that 20 log H ( j1) = 20 log

1 1
= −1 We than obtain = 0.891, and ∈ = 0.509
(1+ ∈2 ) 2
1 1
(1+ ∈2 ) 2
Our next task is to find n from the 20 decibels at ω = 2 specification. From the previous relation the loss
can be given as approximately. 20 ≅ 20 log 0.509 + 6( n − 1) + 20 n log 2
Solving for n, =2.65 since n must be an integer, we let n=3 with the specification of n and ∈ , the pole
locations are completely specified. Our next task is to determine these pole locations. First we must find
1 1
β κ β κ = 1n sinh
−1
= sinh −1 1.965 = 0.476
∈ 3
In order to find the normalized Chebyshev poles from the Butterworth poles, we must first determine tan
β κ Here we have tanh β k = tanh 0.476 = 0.443
Fro table 1, the n=3 Butterworth poles are s1 = −1.0, s 2 , 3 = −0.5 ± j 0.866
Multiplying the real parts of these poles by 0.443, we obtain the normalized Chebyshev poles.
s1 = −0443, s 2,3 = −0222 ± j 0.866
Finally, the denormalized Chebyshev poles are obtained by multiplying the normalized ones by
cosh β κ = 1.1155 so that the denormalized poles are s1 = −0494 and s 2,3 = −0249 ± j 0.972
0.502
H(s) is then H ( s) =
( s + 0.494)( s + 0.249 − j 0.972)( s + 0.249 + j 097)
0.502
=
s + 0992 s + 1.255s + 0.502
3 2
In figure below, the amplitude responses of the Chebyshev and n =3 Butterworth filter are shown.
114
6.4. Monotonic filters with optimum cutoff

In comparing Butterworth filters with Chebyshev filters, the following can be said. The Butterworth
response is a maximally flat, monotonic response, whereas the Chebyshev response is equal ripple in the
pass band. In the stop band, the Chebyshev response falls off more rapidly than the Butterworth (except
when e is very, very small). In this respect, the Chebyshev filter is a better filter than the Butterworth.
However, as we shall see in the next section, the transient response often Chebyshev filter is very poor. If
we require sharp cutoff characteristics for a given degree n, however, the Butterworth filter is quite
unsatisfactory.
Figure: Amplitude response of n=3 Chebyshev filter with 1.0-decibel ripple in pass band and Butterworth
response (n=3)
A class of filters called optimum or ‘L’ filters, which have the following properties:
1. The amplitude response is monotonic
2. The fall –off rate at ω cutoffs the greatest possible, if monotonic is assumed
3. The zeros of the system function of the L filter are all at infinity.
Recall that the magnitude response of a low –pass filter with all zeros at infinity can be express as
k0
M (ω ) = Let us denote the polynomial generating the L filter by
[1 + f (ω )]
1
2 2
( )
f (ω 2 ) = Ln ω 2
115
( )
The polynomial Ln ω 2 has the following properties.
a. Ln (0 ) = 0 b. Ln (1) = 1 c.
( )
dLn ω 2
≤0 d.
( )
dLn ω 2
= M (max imum)
dω dω ω =1
Properties a, b and c are the same as for the Butterworth generating polynomial f (ω 2 ) = ω 2 n . Property c
insures that the response M (ω ) is monotonic and property‘d’ requires that the slope of Ln ω 2 ( ) at
ω = 1 be the steepest to insure sharpest cutoff. Papoulis originally derived the generating equation for the
polynomials Ln ( for n odd ) to be
2
2ω 2 −1 k 
Ln (ω ) = ∫ ∑ ai Pi ( x) dx
2
−1
 i =0 
Where n = 2k + 1 and the Pi (x) are the Legender polynomials of the first kind
P0 ( x ) = 1 P1 ( x ) = x, P2 ( x ) =
1
2
( ) 1
3x 2 − 1 , P3 ( x ) = (5 x 3 − 3x ) ⋅ ⋅ ⋅
2
a1 a 2 ak 1
And the constants at are given by ao = = = ⋅⋅⋅ = =
3 5 2k + 1 2 (k + 1)
Later Papoulis and, independently, showed that the even order Ln polynomials can be given by.
2
2ω 2 −1 k 
L 2 k + 2 (ω ) = ∫
2
( x + 1) ∑ a i Pi ( x) dx , n = 2k + 2 Where the constants a1 are given by:
−1
 i =o 
a2 a 1
Case 1(k even): ao = = ... k = a1 = a3 = ... = a k −1 = 0
5 2k + 1 (k + 1)(k + 2)
a1 a3 a 1
Case 2(k odd): = = ... k = ao = a 2 = ... = a 2 k = 0
3 7 2k + 1 (k + 1)(k + 2)
( )
Fukada tabulated the Ln ω 2 polynomials up to n=7 together with
dLn ω 2 ( )
evaluated at ω = 1 to give an
dω
indication of the steepness of the cut off. This is shown in table 4. To obtain the system function H (s ) for
( )
the " L" filter, we must factor the equation for h s 2 and choose the Hurwitz factors as H (s ) .
1
h( s 2 ) = H ( s ) H ( − s ) =
1 + Ln ( − s 2 )
For example for n = 3 the magnitude response squared is
1 1
M 2 (ω ) = =
1 + L3 (ω ) 1 + ω − 3ω 4 + 3ω 6
2 2
116
1
Substituting − ω 2 = s 2 , we obtain h( s 2 ) = H ( s ) H (− s ) =
1 − s 2 − 3s 4 − 3s 6
Table 4 Ln (ω 2 ) polynomials
dLn (1)
N ( )
Ln ω 2
dω
2 ω4 4
3 3ω 6 − 3ω 4 + ω 2 8
4 6ω 8 − 6ω 6 + 3ω 4 12
5 20ω 10 − 40ω 8 + 28ω 6 − 8ω 4 + ω 2 18
6 50ω 12 − 120ω 10 + 105ω 8 − 40ω 6 + 6ω 4 24
7 175ω 14 − 525ω 12 + 615ω 10 − 355ω 8 + 105ω 6 − 15ω 4 + ω 2 32
0.577
After we factor h( s 2 ) , we obtain H (s) =
s + 1.31s + 1.359 s + 0.577
3 2
Where the numerator factor, 0.577, is chosen too let the d-c gain be unity. The poles of
H (s) are s1 = −0.62; s 2,3 = −0.345 ± j 0.901 . The amplitude response of third –order optimum
(L) and Butterworth filters are compared in figure below. Not that the amplitude response of the optimum
filter is not maximally flat, although still monotonic. However, the cut off characteristic of the optimum
filter is shaper than the cutoff of the Butterworth filter.
117
Figure: Amplitude response of optimum Versus Butterworth filters
6.5. Linear phase filters
Suppose a system function is given by H ( s ) = Ke −sT ∗∗

Were K is a positive real constant. Then the frequency response of the system can be expressed as
H ( jω ) = Ke − jωT
So that the amplitude response M (ω ) is a constant K , and the phase response is linear in ω .
φ (ω ) = −ωT
− sT
The response of such a system to an excitation denoted by R( s ) = KE ( s )e
So that the inverse transform r (t ) can be written as r (t ) = ξ

−1
[R(s )] = Ke(t − T )u(t − T )
We see that the response r (t ) is simply the excitation delayed by a time T, and multiplied by a constant.
Thus no signal distortion results from transmission through a system described by H(s) in equation (∗∗) .
We note further that the delay T can be obtained by differentiating the phase response φ (ω ) by ω ; that is,
dφ (ω )
Dealy = − =T
dω
Consequently, in a system with linear phase, the delay of the system is obtained by differentiating the
phase response φ (ω ) . A system with linear phase and constant amplitude is obviously desirable from a
pulse transmission viewpoint. However, the system function H(s) in Eq (∗∗) is only realizable in terms of
a lossless transmission line called a delay line. If we require that the transmission network be made up of
lumped elements, then we must approximate H ( s) = Ke − st by a rational function in s . The approximation
method we shall describe here is due to Thomson. We can write H (s ) as
K0 K0
H (s) = =
e sT
sinh sT + cosh sT
Where K 0 is chosen such that H (0) = 1 . Let the delay T be normalized to unity and let us divide both
numerator and denominator of H (s ) by sinh s to obtain.
K 0 / sinh s
H (s) = If sinh s and cosh s are expanded in power series, we have
coth s + 1
s2 s4 s6 s3 s5 s7
cosh s = 1 + + + + ... sinh s = s + + + + ...
2! 4! 6! 3! 5! 7!
From these series expansions, we then obtain a continued fraction expansion of coth s as
118
1 1
coth s = +
s 3 1
+
s 5 1
+
s 7
+ ...
s
Ko
If the continued fraction is terminated in n terms, then H (s ) can be written as H ( s ) =
Bn ( s )
Where Bn (s) is Bessel polynomials defined by the formulas
Bo = 1
B1 = s + 1
⋅⋅⋅
Bn = (2n − 1) Bn−1 + s 2 Bn − 2
From these formulas, we obtain B 2 = s + 3s + 3 , B3 = s 3 + 6 s 2 + 15 s + 15

2
Higher order Bessel polynomials are given in table 5 and the roots of Bessel polynomials are given in
table 6. Note that the roots are all in the left-half plan. The amplitude and phase response of a system
function employing normalized third-order Bessel polynomial are given by the solid lines in figure
below.
15
H (s) =
s + 6 s + 15s + 15
3 2
Table 5coeffcients of Bessel polynomials

N b0 b1 b2 b3 b4 b5 b6 b7
0 1
1 1 1
2 3 3 1
3 15 15 6 1
4 105 105 45 10 1
5 945 945 420 105 15 1
6 10,395 10,395 4,725 1,260 210 21 1
7 135,135 135,135 62.370 17,325 3,150 378 28 1
119
These are compared with the amplitude and phase of an un-normalized third-order Butterworth function
given by the dotted lines. Note that the phase response of the constant-delay function is more liner than
the phase of the Butterworth function. Also, the amplitude cutoff of the constant-delay curve is more
gradual than that of the Butterworth.
Table 6 Roots of Bessel Polynomials

n Roots of Bessel Polynomials
1 -1.0+j0
2 − 1.5 ± j 0.866025
− 2.32219 + j 0
3 
− 1.83891 ± j1.75438
− 2.89621 ± j 086723
4 
− 2.10379 ± j 265742
− 3.64674 + j 0
5 
− 3.35196 ± j1.74266
− 4.24836 ± j 0.86751

6 − 3.73571 ± j 2.62627
− 2.51593 ± j 4.4967

− 4.97179 + j0
− 4.75829 ±
 ji.73929
7 
− 4.07014 ± j 3.51717
− 2.68568 ± j 5.42069
120
Figure: Amplitude response of n=3 Bessel and Butterworth filters.
Figure: Phase responses of low-pass filters
6.6. Transient Response of Low –Pass Filters

In this section we will compare the transient response of the filters discussed in section above in
particular, we will compare the step response of the filters according to the following figures of merit:
1. Rise time t R : The rise time of the step response is defined here as the time required for the step
response to rise from 10% to 90% of its final value as depicted below.
121
2. Ringing: Ringing is an oscillatory transient occurring in the response of a filter as result of a sudden
change in input (such as a step) .A quantitative measure of the ringing in a step response is given by its
settling time.
3. Settling time: The settling time is that time t s beyond which the step response does not differ from the
final value by more than 2% as depicted in fig. below.
4. Delay time, t D : Delay time is the time which the step response requires to reach 50% of its final value
as shown in fig. below.
5. Overshoot. The overshoot of the step response is defined as the difference between the peak value and
the final value of the step response expressed as a percentage of the final value.
Most of the foregoing figures of merit are related to frequency response particularly bandwidth and phase
linearity. Some of the quantities such as rise time and delay time are intimately related to each other but
have rather tenuous ties with overshoot, Let us examine qualitatively the relationship between the
transient response criteria just cited and frequency response.
Figure: figure of merit for step response. t R = Rise time; t s = setting time; t D = delay time
Rise time and bandwidth have an inverse relationship in a filter. The wider the bandwidth, the smaller the
rise time; the narrower the band width, the longer the rise time .physically, the inverse relationship could
be explained by noting that the limited performance of the filter at high frequencies slows down the
abrupt rise in voltage of the step and prolongs the rise time. Thus we have.
TR × BW = cons tan t
122
Rise time is a particularly important criterion in pulse transmission. In an article on data transmission it
was shown that in transmitting a pulse of width T1 though a system with adjustable bandwidths, the
following results wee obtained:
Bandwidth ( f ° = 1 / T 1) Rise time (milliseconds)
fC 0.5
2 fC 0.25
3 fC 0.16
4 fC 0.12
5 fC 0.10
The table shows a definite inverse relationship between rise time and bandwidth. A definition of time
delay is given by Elmore as the first moment or centroid of the impulse response
∞
TD = ∫ t × h(t )dt
°
Provided the step response has little or no overshoot .Elmore’s definition or rise time if given as the
second moment.
∞ 1/ 2
TR = 2π ∫ (t − TD ) 2 h(t )dt 
 ° 
These definitions are useful because we can obtain rise time and delay time directly from the coefficients
of the system function H (s ) without going into the roof, which is in Elemore and sands, if H (s ) is given
1 + a1 s + a2 s 2 + ...an s n
as H ( s ) =
1 + b1s + b2 s 2 + ..... + bm s m
{ [
The time delay TD is TD = b1 − a1 . And the rise time is TR = (2π b1 − a1 + 2(a 2 − b2 )
2 2
]}
1/ 2
For the R-C network in Figure below, H (s ) is

v( s ) R
H (s) = = , TD = RC , TR = 2π RC
I ( s) 1 + sRC
R-C network
123
Is should be emphasized that Elmore’s definitions are restricted to step response without overshoot
because to the moment definition. The more general definition of rise time is the 10-90% on cited earlier,
which has no formal mathematical definition. Overshoot is generally caused by “excess” gain at high
frequencies .By excess gain we normally mean a magnitude characteristic with a peak such as the shunt
peaked response shown by the dashed curve in fig below. A magnitude characteristic with no overshoot
is the magnitude characteristic of an R-C interstage shown by the solid curve
.
Figure: Comparison of shunt-peaked and simple R-C magnitudes
The step response of the n=3, n=7, and n=10 Butterworth filters are shown in fig. below. Note that as n
increases, the overshoot increase, this is because the higher order Butterworth filters have flatter
magnitude characteristics (i.e. there is more gain at frequencies just below the cutoff).
Figure: Step response of normalized Butterworth low-pass filters

Ringing is due to sharp cutoff in the filter magnitude response, and is accentuated by a rising gain
characteristic preceding the discontinuity. The step response of an n=3 Bessel (liner phase) filter is
124
compared to the response of an n=3 Chebyshev filter with 1- decibel ripple in Fig below. We cannot
compare their rise times since the bandwidths of the two filters have not been adjusted to be equal.
However, we can compare their ringing and settling time’s .The Chebyshev filter has a sharper cutoff,
and therefore has more ringing and longer settling time than the Bessel filter. Note also the negligible
overshoot of the Bessel filter that is characteristic of the entire class of Bessel filters.
Figure: Comparison of filter transient responses.

The decision as to which filter is best depends upon the particular situation. In certain applications, such
as for transmission of music, phase is not important. In these cases, the sharpness of cutoff may be the
dominate factor so that the Chebyshev or the optimum filter is better than the other s, suppose we were
dealing with a pulse transmission system with the requirements that the output sequence have
approximately the same shape as the input sequence, except for a time delay of T = T2 − T1 , as shown in
fig “a” below. It is clear that a filter with a long rise time is not suitable, because the pulse would: smear”
over each other.
As seen in fig “b” below, the same can be said for long settling times. Since a pulse transmission system
must have linear phase to insure undistorted harmonic reconstruction at the receiver, the best filter for the
system is a linear phase filter with small rise and settling times.
125
Figure: smearing of pulses in systems with long rise and settling times
6.7 A Method of Reduce Overshoot in Filters
We present here a method to reduce the overshoot and ringing of a filter step response. The step response
of a tenth – order Butterworth filter is shown in fig below. It is seen that the overshoot is about 18% we
note that after the first peak, the ringing of the step response has an approximate sinusoidal wave shape.
Let us now consider the second derivative of the step response shown by the dashed curve in below.
Beyond the first peak of the step response, the second derivative is also (approximately)sinusoidal, and is
negative when the step response is less than unity if we add the second derivative to the step response,
we reduce the over shoot and ringing.
126
Suppose α (t ) is the step response and H (s ) is the system function of the filter. The corrected step
d 2α (t )
response can be written as. α 1 (t ) = α (t ) + K Where K is a real, positive constant
dt 2
Taking the Laplace transform, we have
 H ( s )  = ( Ks 2 + 1) H ( s)
Γ[α 1 (t )] =
H (s)
+ Ks 2 
s  s  s
We see that by adding a pair of zeros on the jω -axis at s = ± j / k , the overshoot and ringing are
reduced. For low –pass filters, the factor 1 / k , must in general be greater than the bandwidth of the
system. For normalized Butterworth filters the bandwidth is ω = 1 so that k < 1 . The factor k also controls
the amount of overshoot reduction. If K is too small, adding the zeros on the jω axis will have
negligible effect.
127
Therefore the zeros should be added some where near the band edge. Figure above shows the effects of
adding zeros at ω = ±1 (i.e. right at the band edge) and at ω =1.5 we see that the further away the zeros
are placed from the band edge the less effect they will have. The addition of the zeroes will decrease the
3-decible band width of the filter, however as seen in figure above. Therefore a compromises must be
reached between reduction band width and reduction overshoot.
6.8 A Maximally Flat Delay and Controllable Magnitude Approximation.

In this section we will examine an interesting result, which is due to Budak. The result deals with linear
phase approximation with controllable magnitude. In previous section, we discussed approximation of a
flat delay using Bessel polynomials. The resulting rational approximant was an all- pole function (all
transmission zeros at s = ∞ ) whose denominator was a Bessel polynomial. There was no control of the
magnitude using the all –pole approximant. In Budak’s method the magnitude is controllable, while the
phase is as linear as the standard Bessel approximation.
Budak’s approximation is obtained by introducing the parameter k to split e − s in to two parts such that.
−s e − ks
e = − ( k −1) s
0 < k ≤1
e
And then approximate independently e − ks , and e − ( k −1) s with all–pole Bessel polynomial
approximations. Thus the resulting approximation for e − s we have Bessel polynomials for both
numerator and denominator. The poles of the e − ( k −1) s approximant will be the zeros in the final
approximate, while the poles of the e − ks approximate remain as poles in the final approximate, For
reliability, the degree of the e − ( k −1) s approximant should be less than the degree of the
e − ks approximant.
As an example, consider the approximation with three zeros and four poles
105
e − ks ≅
( ks ) + 10 ( ks ) + 45 ( ks ) 2 + 105 ( ks ) + 105
4 3
15
e − ( k −1 ) s ≅
[( k − 1) s ] + 6 [( k − 1s ]2 + 15 [( k − 1) s ] + + 15
3
We then perform the operation as indicated above to obtain
128
−s
e ≅
{ 3 2
}
7 [(k − 1) s ] + 6[(k − 1) s ] + 15[(k − 1) s ] + 15
(ks) 4 + 10(ks) 3 + 45(ks) 2 + 105(ks) + 105
In Fig “a” below the magnitude characteristic of the above equation is plotted with k as a parameter. The
phase characteristic is given in terms of deviation of phase from linearity ∆φ = ω − φ (ω ) and is shown in
figure “b”. The improvement in phase linearity over the all- pole Bessel approximation k=1 is shown by
these curves. Note as the bandwidth is increased (k decreasing) phase linearity is improved.
Figure below shows the step response of the same equation also with k as a parameter. Since the effect of
decreasing k increases band width, the corresponding effect in the time domain is to decrease rise time.
Budak also observes that as k decrease from unity, the poles and zeros migrate to keep the phase liner.
The zeros move in ward from infinity along radial lines, while the poles move outward along radial lines.
129
6.9 Synthesis of Low –Pass Filters

Given the system function of the low–pass filter as derived by the methods described in previous section,
we can proceed with the synthesis of the filter network. If we consider the class of filters terminated in a
1 − Ω load, and if we let the system function be transfer impedance,
z 21 y 21
Z 21 ( s ) = Or a transfer admittance Y21 ( s ) =
1 + z 22 1 + y 22
We can synthesize the low–pass filter, consider then n=3 Optimum (L) filter function.
Given as a transfer impedance
0.577
Z 21 ( s ) =
s + 1.31s + 1.359 s + 0.577
3 2
We see that the zeros of transmission are all at infinity. Since the numerator of Z21 is even, we divide
both numerator and denominator by the odd part of the denominator s2+1.359s . Thus we have
0.577 1.31s 2 + 0.577
z 21 = , z =
s 3 + 1.359s s 3 + 1.359s
22
The structure of the low –pass filter with three zeros of transmission at infinity is given in chapter 12 we
must synthesize z22 to give the π reactance structure. This we accomplish through the following
continued fraction expansion 1 / z 22 :
130
1.31s 2 + 0.577 s 3 + 1.359s (0.763s

s 3 + 0.440s
0.919s )1.3s 2 + 0.577(1.45s
1.31s 2
0.577)0.919s (1.593s
0.919s
The optimum filter is shown in figure below for the n= 3 Butterworth filter given by the transfer
impedance.
1
z 21 ( s ) =
1
Z 21 ( s ) =
1
We have z 22 ( s ) =
s + 2s
3
s 3 + 2s 2 + 2s + 1 s 3 + 2s
we then synthesize z22(s) by a continued fraction expansion to give the filter shown in figure below
Figure: {∗ ∗ ∗}
Figure: Canonical forms for filters described in tables 7, 8, and 9 below.

In table 7, 8 and 9 are listed element values (up to n=7) for single-terminated Butterworth Chebyshev (1-
decible ripple) and Bessel filters, respectively. These apply to the canonical realization for transfer
impedance Z 21 ( s) shown in the immediate above figure. If Y21 (s) realization is desired, we simply
replace all shunt capacitors by series inductors and vice versa. Some common interstage structures are
shown in figure below. In this figure a structure known as the shunt peaked network is shown. The
transfer impedance of the shunt-peaked network is
1 s + R/L
Z 21 ( s ) =
C s + sR / L + (1 / LC )
2
Table 7 Normalized Element Values for a single Terminated Butterworth filter

N C1 L2 C3 L4 C5 C6 C7
131
1 1.000
2 0.707 1.414
3 0.500 1,333 1.500
4 0.383 1.082 1.577 1.553
5 0.309 0.894 1.382 1.694 1.531
6 0.259 0.758 1.202 1.553 1.759 1.553
7 0222 0.656 1.055 1.397 1.659 1.799 1.558
Table 8 Normalized Element values for single Terminated Chebyshev filter with 1 db Ripple
N C1 L2 C3 L4 C5 C6 C7
1 0.509
2 0.911 0.996
3 1.012 1.333 1.509
4 1.050 1.413 1.909 1.282
5 1.067 1.444 1.994 1.591 1.665
6 1.077 1.460 2.027 1.651 2.049 1.346
7 1.083 1.496 2.044 1.674 2.119 1.649 1.712
Table 9 Normalized Element values for a single Terminated Bessel Filter
N C1 L2 C3 L4 C5 C6 C7
1 1.000
2 0.333 1.000
3 0.167 0.480 0.833
4 0.100 0.290 0.463 0.710
5 0.067 0.195 0.310 0.422 0.623
6 0.048 0.140 0.225 0.301 0.382 0.560
7 0.036 0.106 0.170 0.229 0.283 0.349 0.511
We see that Z 21 ( s) has a real zero and a pair of poles which may be complex depending upon the values
of R, L, & C. In figure “b” below a simple R-C interstage is shown whose transfer impedance is
1 1
Z 21 ( s ) =
C S + 1 / RC
132
Observe that all the filter transfer function considered up to this point is made up of pairs of conjugate
poles and simple poles on the jω axis. It is clear that if we cascade shunt-peaked stages and R-C stages,
we can adjust the R, L, and C elements to give the desired response characteristic. The only problem is to
cancel the finite zero of the shunt- peaked stage. for example , if we whish to design an amplifier with an
n =3 low –pass Butterworth characteristic , we first break up the system function into complex pole pairs
and real pole terms ,a s given by
1 s +1 1 1
Z 21 ( s ) = = 2
( s + s + 1( s + 1) s + s + 1 s + 1 s + 1
2
(a) Shunt-peaked interstage (b) R-C interstage

We then associate the individual factors with shunt-peaked or simple R-C stages and solve for the
element values. The n=3 Butterworth amplifier is given in figure below.
Figure: Butterworth Amplifier
6.10 Magnitude and Frequency Normalization

Filters designed with these restrictions are considered to be normalized in both cutoff frequency and
impedance level. we will now discuss methods whereby the normalized filters can be converted in to
filters which meet arbitrary cutoff frequency and impedance level specification. Let us denote by a
subscript n the normalized frequency variable s n and the normalized element values Ln , Rn , and Cn , The
normalized frequency variables s n is related to the actual frequency s by the relation
133
s
sn = Where ω0 the normalizing constant, is dimensionless and is often taken to be the actual cut off
ω0
frequency. Since the impedance of an element remains invariant under frequency normalization, we
obtain the actual element values from the normalized values by setting the impedances in the two cases
equal to each other. For example, for an inductor, we have; sn Ln = sL = ω0 sn L from this equation
Ln
we then obtain the denormalized value of inductance as L=
ω0
1
Similarly, from the impedance of a frequency normalized capacitor Cn we obtain the denormalized
snCn
1 1
value of capacitance thorough the equation. =
s n C n sC
C
So that actual value of the capacitance is C = n
ω 0
Since resistances, ideally, are independent of frequency, they are unaffected by frequency normalization.
Consider, next, impedance denormalization. Suppose the actual impedance level should be R0 ohms
instead of 1Ω then denormalized impedance Z is related to normalized impedance Z n by Z = R0 Z n
Where R0 is taken to be dimensionless here. Thus for a normalized resistor Rn the denormalized (actual)
resistance is R = R0 R n
For an inductance, the corresponding relationship is sL = R0 (sLn )
So that the actual inductance value is L = R0 Ln

1 R
Similarly, for a capacitor we have = 0
sC sCn
Cn
So that the actual capacitances is C =
R0
For combined frequency and magnitude denormalization, we simply combine the two sets of equations to
R = R0 Rn C=
Cn
L=
R0 Ln
give
R0ω0 ω0
134
Let us consider an actual example in design. In section above, we synthesized transfer impedance Z 21
with an n = 3 Butterworth amplitude characteristics with cutoff frequency of 1 rad/sec and a load
impedance of 1Ω . Let us redesign this filter for a cutoff frequency of 10 4 rad/sec to work in to a load of
500 Ω . From the original network in Fig {∗ ∗ ∗} above, we take the element values and denormalize
with the normalizing factors, ω0 = 104 and R0 = 500
Then the denormalized element values are:
R = 500 RL = 500
1 4 3
(500)
C1 = 2
= 0.1µf , L= 3
= 0.0067h , C2 = 2
= 0.3µf
500(10 4 ) 10,000 500(10 4 )
The final design is shown in the next figure:
Figure: Denormalized Low-pass Filter
6.11 Frequency Transformations
Up to this point, we have discussed only the design of low –pass filters, while neglecting the equally
important designs of high-pass; band-pass-and band–elimination filters. We will remedy this situation
here, not by introducing new design procedures but through a technique known as a frequency
transformation, whereby, beginning from a normalized low-pass filter, we can generate any other form of
filter. Using frequency transformation, the elements of the normalized low –pass filter are changed in to
elements of a high-pass, band –pass or band –elimination filter.
Analytically, a frequency transformation simply changes on L-C driving–point function in to another L-C
function themselves. Also, since we proceed from normalized low-pass filter, the transformation equation
include built- in frequency denormalization factors so that the resulting networks need only be scaled for
135
impedance level. Consider the simplest transformation equation that of low–pass to high pass, which is
ω0
s=
sn
Where s n represents the normalized low-pass frequency variable, s is the regular frequency variable,
and ωo is the cutoff frequency of the high-pass filter. In terms of real and imaginary parts, we have
ω0 ω 0 (σ n − jω n )
σ + jω = =
σ n + jω n σ n2 + ω n2
Since we are interested principally in how the jω n axis maps into the jω axis we let σ n = 0 so that
ω0
ω=
ωn
Which is the equation that transforms normalized low-pass filters to denormalized high-pass filters. From
immediate above equation, we see that the point ω = ±1 corresponds to the point ω = ±ω o it is also clear
that the transformation maps the segment ω n ≤ 1 on to the segments defined by ω 0 ≤ ω ≤ ∞, as

shown in figure below.
Now let us see how the frequency transformations change the network elements. For convenience, let us
denote the normalized low–pass network element with a subscript n, the high–pass element with a
subscript h, the band-pass elements with a subscript b, and the band-elimination elements with a
subscript e. For the low –pass to high-pass case; let us first consider the changes for the capacitor Cn
Figure: Low-pass to high-pass Transformation

The transformation is given by the equation.
136
1 s
Ln , we have ω0 1
= ∆ Lh s For the inductor Ln sn = Ln ∆
C n sn ω oC n s Ch s
We have observed that a capacitor changes in to an inductor and an inductor changes in to a capacitor in a
low-pass to high-pass transformation (figure below). The element values of the high-pass filter are given
in terms of the normalized low –pass filter elements as
1 1
Lh = And Ch =
ω0Cn ω 0 Ln
Consider the following example. From the normalized third –order Butterworth filter given in Fig
{∗ ∗ ∗}, let us design a corresponding high pass filter with its cutoff frequency ω o = 10 6 rad / sec and
the impedance level of 500 Ω .
Figure: Element changes resulting from frequency transformations
Figure: transformation of low-pass filter in figure {∗ ∗ ∗}into high –pass filter.

From the low-pass filter, we can draw by inspection the high –pass –filter circuit shown in figure above.
Its element values are:
137
500 1 500
R L = 500Ω , L1h = 6 1
= 10 −3 h , Cn = 6 4
= 1.5 × 10 −9 f , L2 h = 6 2
= 0.333 × 10 −3 h
10 ( 2 ) (500)10 ( 3 ) 10 ( 3 )
Next, let us examine the low –pass to band –pass transformation (also an L-C function):
ω0  s ω0 
sn =  + 
BW  ω0 s 
Where, if ωC 2 and ωC1 denote the upper and lower cutoff frequencies of the band –pass filter, BW is the
bandwidth BW = ω C 2 − ω C1
And ω 0 is the geometric mean of ωC 2 and ωC1 ω 0 = ω C 2 ω C1
The low-pass to band –pass transformation map the segment ω n ≤ 1 to the segments
ω C 2 ≥ ω ≥ ω C1 shown in figure below. The normalized low-pass elements are then modified
according to the following equations.
Ln ω 02 L n ∆ 1
Ln s n = s+ = Lb 1 s +
BW BWs C b1 s
Figure: Low-pass to band-pass transformation

We not the inductor Ln is transformed in to a series-tuned tank, shown in fig. above, whose elements are
given as
Ln Cn
Lb1 = , Cb1 =
BW ω02 Ln
138
The capacitor Cn is transformed into a parallel –tuned tank (figure above) whose elements are
BW Cn
Lb 2 = , Cb 2 =
ωo 2Cn BW
Let us transform the third-order Butterworth low-pass filter in figure {∗ ∗ ∗}in to a band –pass filter
with a 1Ω impedance level, whose bandwidth is , BW = 6 x104rad / sec and its band –pass is “centered”
at ωo =4x104rad/sec . We draw the band-pass filter shown in figure below by the rule given above.
Figure: Band-pass filter transformed from low-pass filter in figure {∗ ∗ ∗}

The element values of the band-pass filter are given in the following equation.
6 x10 4 4
1
1 4
L1 = = o.75 × 10 − h , C 1= 2
= × 10 −4 f , L2 = 3
= 92 × 10 −4 h
(4 × 10 ) (
4 2 1
2 ) 6 × 10 4
12 6 × 10 4
Fi
6 × 10 4 9 6 × 10 4 3
C 2= = × 10 − 4 f , L3 = = 0.25 × 10 −4 h , C 3= 2
= 0.25 × 10 −4
f
(4 × 10 ) ( 2 ) 32
4 2 3
( 4 × 10 4 ) 2( 2 )
3
6 × 10 4
BW
sn =
nally, the band –elimination filter is obtained through the transformation  s ω0 
ω 0  + 
ω0 s 
Where BW and ω0 are defined in a manner similar to that for the band –pass filter .The transformation
maps the segment of the jω n − axis in figure “a” below onto the segments shown on the jω − axis in
figure “b” below.
139
Figure: Low-pass to band-elimination transformation

For the low –pass to band-elimination transformation we, therefore, have the following element changes.
1 ∆ 1 1 ω0  s ω0  ∆ 1
Ln s n = = =  +  = Le 2 s +
( s / Ln BW ) + (ω 0 / L n BWs ) C e1 s + (1 / Le1 s )
2
C n s n C n BW  ω0 s  Ce2 s
Observe that the normalized low-pass inductor goes in to a parallel tuned circuit and the capacitor Cn
goes into a series –tuned circuit.
Table 6.10: Table of various Frequency Transformations s
Transformation Low –pass to Equation
ωo
High –pass sn =
s
ωo  s ωo 
Band-pass sn =  + 
BW ω
 o s 
BW
Band –elimination sn =
 s ωo 
ω o  + 
 ωo s 
6.12 Active Low Pass Filter
Limitations of passive filters:
140
1. At low frequency (approximately 100Hz to few hundred kHz) increases the size and weight of
inductors for better device value, which deteriorated sharpness at cut off frequency.
2. Impossibility of integrating inductors with practical value of inductances in the micro-
miniaturization circuit element.
3. Necessity of buffer (isolation) amplifiers to prevent loading while cascading sections of filters.
4. Need external amplifier to adjust the required gain.
Merits are: They are stable, non frequency limited and operate under unlimited power
Active filters are networks which use active device together with RC elements. In the Passive Filter
discussion, we saw how a basic first-order filter circuits, such as the low pass and the high pass filters can
be made using just a single resistor in series with a non-polarized capacitor connected across a sinusoidal
input signal. We also noticed that the main disadvantage of passive filters is that the amplitude of the
output signal is less than that of the input signal, i.e., the gain is never greater than 1 and that the load
impedance affects the filters characteristics. With passive filter circuits containing multiple stages, this
loss in signal amplitude called "Attenuation" can become quiet severe. One way of restoring or
controlling this loss of signal is by using amplification through the use of Active Filters.
As their name implies, Active Filters contain active components such as operational amplifiers,
transistors or FET's within their circuit design. They draw their power from an external power source and
use it to boost or amplify the output signal. Filter amplification can also be used to either shape or alter
the frequency response of the filter circuit by producing a more selective output response, making the
output bandwidth of the filter narrower or even wider.
An active filter generally uses an operational amplifier (op-amp) within its design and in the Operational
Amplifier we can realize that as an Op-amp has high input impedance, low output impedance and a
voltage gain determined by the resistor network within its feedback loop. Unlike a passive high pass filter
which has in theory an infinite high frequency response, the maximum frequency response of an active
filter is limited to the Gain/Bandwidth product (or open loop gain) of the operational amplifier being
used. Still, active filters are generally easier to design than passive filters; they produce good
performance characteristics, very good accuracy with a steep roll-off and low noise when used with a
good circuit design.
141
6.12.1 Active Low Pass Filter
The most common and easily understood active filter is the active Low Pass Filter. Its principle of
operation and frequency response is exactly the same as those for the previously seen passive filter; the
only difference this time is that it uses an op-amp for amplification and gain control. The simplest form
of a low pass active filter is to connect an inverting or non-inverting amplifier, similar to the basic RC
low pass filter circuit as shown.
First Order Active Low Pass Filter
This first-order low pass active filter consists simply of a passive RC filter stage providing a low
frequency path to the input of a non-inverting operational amplifier. The amplifier is configured as a
voltage-follower(Buffer) giving it a DC gain of one, Av = +1 or unity gain as opposed to the previous
passive RC filter which has a DC gain of less than unity. The advantage of this configuration is that the
op-amps high input impedance prevents excessive loading on the filters output while its low output
impedance prevents the filters cut-off frequency point from being affected by changes in the impedance
of the load. While this configuration provides good stability to the filter, its main disadvantage is that it
has no voltage gain above one. However, although the voltage gain is unity the power gain is very high as
its output impedance is much lower than its input impedance. If a voltage gain greater than one is
required we can use the following filter circuit.
142
Active Low Pass Filter with Amplification
The frequency response of the circuit will be the same as that for the passive RC filter, except that the
amplitude of the output is increased by the pass band gain, AF of the amplifier. For a non-inverting
amplifier circuit, the magnitude of the voltage gain for the filter is given as a function of the feedback
resistor (R2) divided by its corresponding input resistor (R1) value and is given as:
R1
DC − gain =
R1 + R2
Therefore, the gain of an active low pass filter as a function of frequency will be:
Gain of a first-order low pass filter

Vout AF
Voltage − gain, ( Av ) = = Where:
Vin  f 
2
1 +  
 fC 
• AF = the pass band gain of the filter, (1 + R2 / R1 )

• f = the frequency of the input signal in Hertz, (Hz)
• f C = the cut-off frequency in Hertz, (Hz)
Thus, the operation of a low pass active filter can be verified from the frequency gain equation above as:
Vout
1. At very low frequencies, f < fC , ≅ AF 2. At the cut-off frequency, f = fC ,
Vin
Vout AF
= = 0.707 AF
Vin 2
143
Vout
3. At very high frequencies, f > f C , < AF
Vin
Thus, the Active Low Pass Filter has a constant gain AF from 0Hz to the high frequency cut-off point,
f C . At f C the gain is 0.707 AF , and after f C it decreases at a constant rate as the frequency increases.
That is, when the frequency is increased tenfold (one decade), the voltage gain is divided by 10. In other
words, the gain decreases 20dB (= 20log 10) each time the frequency is increased by 10. When dealing
with filter circuits the magnitude of the pass band gain of the circuit is generally expressed in decibels or
dB as a function of the voltage gain and this is defined as:
Magnitude of Voltage Gain in (dB)
V   2Vout 
Av (dB ) = 20 log10  out  ∴ −3dB = 20 log10 
 V


 Vin   in 
Example1: Design a non-inverting active low pass filter circuit that has a gain of ten at low
frequencies, a high frequency cut-off or corner frequency of 159Hz and an input impedance of 10KΩ.
R2
The voltage gain of a non-inverting operational amplifier is given as: AF = 1 + = 10
R1
Assume a value for resistor R1 of 1kΩ rearranging the formula above gives a value for R2 of
R2 = (10 − 1) × R1 = 9 × 1kΩ = 9kΩ Then, for a voltage gain of 10, R1 = 1kΩ and R2 = 9kΩ. However, a
9kΩ resistor does not exist so the next preferred value of 9k1Ω is used instead.
Converting this voltage gain to a decibel dB value gives: Gain in dB 20 log A = 20 log 10 = 20 dB
The cut-off or corner frequency ( f C ) is given as being 159Hz with an input impedance of 10kΩ. This
cut-off frequency can be found by using the formula:
1
fC = Hz Where f C = 159Hz and R = 10kΩ. Then, by rearranging the above formula we can find
2πRC
1 1
the value for capacitor C as: C = = = 100nF
2πRf C 2π × 159 × 10kΩ
144
Then the final circuit along with its frequency response is given below as:
Low Pass Filter Circuit
Frequency Response Curve
If the external impedance connected to the input of the circuit changes, this change will also affect the
corner frequency of the filter (components connected in series or parallel). One way of avoiding this is to
place the capacitor in parallel with the feedback resistor R2. The value of the capacitor will change
slightly from being 100nF to 110nF to take account of the 9k1Ω resistor and the formula used to calculate
the cut-off corner frequency is the same as that used for the RC passive low pass filter.
1
fC = Hz An example of the new Active Low Pass Filter circuit is given as.
2πR 2 C
145
Simplified non-inverting amplifier filter circuit Equivalent inverting amplifier filter circuit
Applications of Active Low Pass Filters are in audio amplifiers, equalizers or speaker systems to direct
the lower frequency bass signals to the larger bass speakers or to reduce any high frequency noise or
"hiss" type distortion. When used like this in audio applications the active low pass filter is sometimes
called a "Bass Boost" filter.
Second-order Low Pass Active Filter
As with the passive filter, a first-order low pass active filter can be converted into a second-order low
pass filter simply by using an additional RC network in the input path. The frequency response of the
second-order low pass filter is identical to that of the first-order type except that the stop band roll-off
will be twice the first-order filters at 40dB/decade (12dB/octave). Therefore, the design steps required of
the second-order active low pass filter are the same.
146
Second-order Active Low Pass Filter Circuit
When cascading together filter circuits to form higher-order filters, the overall gain of the filter is equal to
the product of each stage. For example, the gain of one stage may be 10 and the gain of the second stage
may be 32 and the gain of a third stage may be 100. Then the overall gain will be 32,000, (10 x 32 x 100)
as shown below.
Cascading Voltage Gain
AV = AV1 × AV2 × AV3 ⇒ AV = 10 × 32 × 100 = 32,000 → AV (dB ) = 20 log10 (32,000)
AV (dB ) = 90 dB Or 90dB = 20dB + 30dB + 40dB

Second-order (two-pole) active filters are important because higher-order filters can be designed using
them. By cascading together first and second-order filters, filters with an order value, either odd or even
up to any value can be constructed. In the next tutorial about filters, we will see that Active High Pass
Filters, can be constructed by reversing the positions of the resistor and capacitor in the circuit.
6.12.2 Active High Pass Filter
The basic operation of an Active High Pass Filter (HPF) is exactly the same as that for its equivalent
RC passive filter circuit, except that this type of circuit has an operational amplifier or op-amp included
within its design for amplification and gain control. Like the previous active low pass filter circuit, the
147
simplest form of an active high pass filter is to connect a standard inverting or non-inverting operational
amplifier to the basic RC high pass passive filter circuit as shown.
First Order Active High Pass Filter
Technically, there is no such thing as an active high pass filter. Unlike Passive High Pass Filters which
have an "infinite" frequency response, the maximum pass band frequency response of an active high pass
filter is limited by the characteristics or bandwidth of the operational amplifier being used, making them
appear as if they are band pass filters with a high frequency cut-off determined by the selection of op-
amp and gain.
In the Operational Amplifier we saw that the maximum frequency response of an op-amp is limited to the
Gain/Bandwidth product or open loop voltage gain ( AV ) of the operational amplifier being used giving
it a bandwidth limitation, where the closed loop response of the op amp intersects the open loop response.
A commonly available operational amplifier such as the uA741 has a typical "open-loop" (without any
feedback) DC voltage gain of about 100dB maximum reducing at a roll off rate of -20dB/Decade (-
6db/Octave) as the input frequency increases. The gain of the uA741 reduces until it reaches unity gain,
(0dB) or its "transition frequency" ( Ft ) which is about 1MHz. This causes the op-amp to have a
frequency response curve very similar to that of a first-order low pass filter and this is shown below.
148
Frequency response curve of a typical Operational Amplifier
Then the filters performance at high frequencies is limited by this unity gain crossover frequency which
determines the overall bandwidth of the open-loop amplifier. The gain-bandwidth product of the op-amp
starts from around 100 kHz for small signal amplifiers up to about 1GHz for high-speed digital video
amplifiers and op-amp based active filters can achieve very good accuracy and performance provided that
low tolerance resistors and capacitors are used. Under normal circumstances the maximum pass band
required for a closed loop active high pass or band pass filter is well below that of the maximum open-
loop transition frequency. However, when designing active filter circuits it is important to choose the
correct op-amp for the circuit as the loss of high frequency signals may result in signal distortion.
149
Active High Pass Filter
A first-order (single-pole) Active High Pass Filter as its name implies, attenuates low frequencies
and passes high frequency signals. It consists simply of a passive filter section followed by a non-
inverting operational amplifier. The frequency response of the circuit is the same as that of the
passive filter, except that the amplitude of the signal is increased by the gain of the amplifier and
for a non-inverting amplifier the value of the pass band voltage gain is given as 1 + R2/R1, the same
as for the low pass filter circuit.
Active High Pass Filter with Amplification
This first-Order high pass filter, consists simply of a passive filter followed by a non-inverting amplifier.
The frequency response of the circuit is the same as that of the passive filter, except that the amplitude of
the signal is increased by the gain of the amplifier. For a non-inverting amplifier circuit, the magnitude of
the voltage gain for the filter is given as a function of the feedback resistor (R2) divided by its
corresponding input resistor (R1) value and is given as:
Gain for an Active High Pass Filter
 f 
AF  
Voltage − gain, ( Av ) =
Vout
=  fC 
Where:
Vin  f 
2
1 +  
 fC 
• AF = the pass band gain of the filter, (1 + R2 / R1 )

• f = the frequency of the input signal in Hertz, (Hz)
150
• f C = the cut-off frequency in Hertz, (Hz)
Just like the low pass filter, the operation of a high pass active filter can be verified from the frequency
gain equation above as:
Vout
1. At very low frequencies, f < f C , < AF
Vin
Vout AF
2. At the cut-off frequency, f = f C , = = 0.707 AF
Vin 2
Vout
3. At very high frequencies, f > f C , ≅ AF
Vin
Then, the Active High Pass Filter has a gain AF that increases from 0Hz to the low frequency cut-off
point, f C at 20dB/decade as the frequency increases. At f C the gain is 0.707 AF , and after f C all
frequencies are pass band frequencies so the filter has a constant gain AF with the highest frequency
being determined by the closed loop bandwidth of the op-amp. When dealing with filter circuits the
magnitude of the pass band gain of the circuit is generally expressed in decibels or dB as a function of the
voltage gain and this is defined as:
Magnitude of Voltage Gain in (dB)
V   2Vout 
Av (dB ) = 20 log10  out  ∴ −3dB = 20 log10 
 V


 Vin   in 
For a first-order filter the frequency response curve of the filter increases by 20dB/decade or 6dB/octave
up to the determined cut-off frequency point which is always at -3dB below the maximum gain value. As
with the previous filter circuits, the lower cut-off or corner frequency ( f C ) can be found by using the
1
same formula: f C = Hz
2πRC
151
The corresponding phase angle or phase shift of the output signal is the same as that given for the passive
RC filter and leads that of the input signal. It is equal to +45o at the cut-off frequency f C value and is
given as:
 1 
Phase Shift φ = tan −1  
 2πfRC 
A simple first-order active high pass filter can also be made using an inverting operational amplifier
configuration as well, and an example of this circuit design is given along with its corresponding
frequency response curve. A gain of 40dB has been assumed for the circuit.
Inverting Operational Amplifier Circuit
Frequency Response Curve
152
Example: A first order active high pass filter has a pass band gain of two and a cut-off corner frequency
of 1 kHz. If the input capacitor has a value of 10nF, calculate the value of the cut-off frequency
determining resistor and the gain resistors in the feedback network. Also, plot the expected frequency
response of the filter.
With a cut-off corner frequency given as 1 kHz and a capacitor of 10nF, the value of R will therefore be:
1 1
R= = = 15.92kΩ
2πf C C 2π × 1000 × 10 × 10 −9
R2 R R
The pass band gain of the filter, AF is given as being, 2. AF = 1 + , ∴ 2 = 1 + 2 And 2 = 1
R1 R1 R1
As the value of resistor, R2 divided by resistor, R1 gives a value of one. Then, resistor R1 must be equal to
resistor R2, since the pass band gain, AF = 2 . We can therefore select a suitable value for the two
resistors of say, 10kΩ's each for both feedback resistors. So for a high pass filter with a cut-off corner
frequency of 1 kHz, the values of R and C will be, 10kΩ's and 10nF respectively. The values of the two
feedback resistors to produce a pass band gain of two are given as: R1 = R2 = 10kΩ's
The data for the frequency response bode plot can be obtained by substituting the values obtained above
over a frequency range from 100Hz to 100 kHz into the equation for voltage gain:
 f 
AF  
Voltage − gain, ( Av ) =
Vout
=  fC 
This then will give us the following table of
Vin  f 
2
1 +  
 fC 
data.
Frequency, ƒ Voltage Gain Gain, (dB) Frequency, ƒ Voltage Gain Gain, (dB)
( Hz ) ( Vo / Vin ) 20log( Vo / Vin ) ( Hz ) ( Vo / Vin ) 20log( Vo / Vin )
100 0.20 -14.02 3,000 1.90 5.56
200 0.39 -8.13 5,000 1.96 5.85
500 0.89 -0.97 10,000 1.99 5.98
153
800 1.25 1.93 50,000 2.00 6.02

1,000 1.41 3.01 100,000 2.00 6.02
The frequency response data from the table above can now be plotted as shown below. In the stop band
(from 100Hz to 1 kHz), the gain increases at a rate of 20dB/decade. However, in the pass band after the
cut-off frequency, f C = 1 kHz, the gain remains constant at 6.02dB. The upper-frequency limit of the
pass band is determined by the open loop bandwidth of the operational amplifier used as we discussed
earlier. Then the bode plot of the filter circuit will look like this.
The Frequency Response Bode-plot for our example
Applications of Active High Pass Filters are in audio amplifiers, equalizers or speaker systems to direct
the high frequency signals to the smaller tweeter speakers or to reduce any low frequency noise or
"rumble" type distortion. When used like this in audio applications the active high pass filter is
sometimes called a "Treble Boost" filter.
Second-order High Pass Active Filter
As with the passive filter, a first-order high pass active filter can be converted into a second-order high
pass filter simply by using an additional RC network in the input path. The frequency response of the
second-order high pass filter is identical to that of the first-order type except that the stop band roll-off
will be twice the first-order filters at 40dB/decade (12dB/octave). Therefore, the design steps required of
the second-order active high pass filter are the same.
Second-order Active High Pass Filter Circuit
154
Higher-order high pass filters, such as third, fourth, fifth, etc are formed simply by cascading together
first and second-order filters. For example, a third order high pass filter is formed by cascading in series
first and second order filters, a fourth-order high pass filter by cascading two second-order filters together
and so on. Then an Active High Pass Filter with an even order number will consist of only second-order
filters, while an odd order number will start with a first-order filter at the beginning as shown.
Cascading Active High Pass Filters
Although there is no limit to the order of a filter that can be formed, as the order of the filter increases so
to does its size. Also, its accuracy declines, that is the difference between the actual stop band response
and the theoretical stop band response also increases. If the frequency determining resistors are all equal,
R1 = R2 = R3 etc, and the frequency determining capacitors are all equal, C1 = C2 = C3 etc, then the cut-
155
off frequency for any order of filter will be exactly the same. However, the overall gain of the higher-
order filter is fixed because all the frequency determining components are equal. In the next tutorial about
filters, we will see that Active Band Pass Filters, can be constructed by cascading together a high pass
and a low pass filter.
6.12.3 Active Band Pass Filter
As we saw previously in the Passive Band Pass Filter tutorial, the principal characteristic of a Band Pass
Filter or any filter for that matter is its ability to pass frequencies relatively unattenuated over a specified
band or spread of frequencies called the "Pass Band". For a low pass filter this pass band starts from 0Hz
or DC and continues up to the specified cut-off frequency point at -3dB down from the maximum pass
band gain. Equally, for a high pass filter the pass band starts from this -3dB cut-off frequency and
continues up to infinity or the maximum open loop gain for an active filter.
However, the Active Band Pass Filter is slightly different in that it is a frequency selective filter circuit
used in electronic systems to separate a signal at one particular frequency, or a range of signals that lie
within a certain "band" of frequencies from signals at all other frequencies. This band or range of
frequencies is set between two cut-off or corner frequency points labeled the "lower frequency" (ƒL) and
the "higher frequency" (ƒH) while attenuating any signals outside of these two points. Simple Active
Band Pass Filter can be easily made by cascading together a single Low Pass Filter with a single High
Pass Filter as shown.
The cut-off or corner frequency of the low pass filter (LPF) is higher than the cut-off frequency of the
high pass filter (HPF) and the difference between the frequencies at the -3dB point will determine the
"bandwidth" of the band pass filter while attenuating any signals outside of these points. One way of
making a very simple Active Band Pass Filter is to connect the basic passive high and low pass filters
we look at previously to an amplifying op-amp circuit as shown.
156
Active Band Pass Filter
This cascading together of the individual low and high pass passive filters produces a low "Q-factor" type
filter circuit which has a wide pass band. The first stage of the filter will be the high pass stage that uses
the capacitor to block any DC biasing from the source. This design has the advantage of producing a
relatively flat asymmetrical pass band frequency response with one half representing the low pass
response and the other half representing high pass response as shown.
The higher corner point (ƒH) as well as the lower corner frequency cut-off point (ƒL) are calculated the
same as before in the standard first-order low and high pass filter circuits. Obviously, a reasonable
separation is required between the two cut-off points to prevent any interaction between the low pass and
high pass stages. The amplifier provides isolation between the two stages and defines the overall voltage
gain of the circuit. The bandwidth of the filter is therefore the difference between these upper and lower -
3dB points. For example, if the -3dB cut-off points are at 200Hz and 600Hz then the bandwidth of the
filter would be given as: Bandwidth (BW) = 600 - 200 = 400Hz. The normalized frequency response and
phase shift for an active band pass filter will be as follows.
157
Active Band Pass Frequency Response
While the above passive tuned filter circuit will work as a band pass filter, the pass band (bandwidth) can
be quite wide and this may be a problem if we want to isolate a small band of frequencies. Active band
pass filter can also be made using inverting operational amplifiers, and by rearranging the positions of the
resistors and capacitors within the circuit we can produce a much better filter circuit as shown below. The
lower cut-off -3dB point is given by ƒC2 while the upper cut-off -3dB point is given by ƒC1.
Inverting Band Pass Filter Circuit
158
− R2 1 1
Voltage − gain = , f C1 = , f C2 =
R1 2πR1C1 2πR2 C 2
This type of band pass filter is designed to have a much narrower pass band. The centre frequency and
bandwidth of the filter is related to the values of R1 , R2 , C1 and C 2 . The output of the filter is again taken
from the output of the op-amp.
Multiple Feedback Band Pass Active Filter
We can improve the band pass response of the above circuit by rearranging the components again to
produce an infinite-gain multiple-feedback (IGMF) band pass filter. This type of active band pass design
produces a "tuned" circuit based around a negative feedback active filter giving it a high "Q-factor" (up to
25) amplitude response and steep roll-off on either side of its centre frequency. Because the frequency
response of the circuit is similar to a resonance circuit, this centre frequency is referred to as the resonant
frequency, (ƒr). Consider the circuit below.
Infinite Gain Multiple Feedback Active Filter
This band pass filter circuit uses the full gain of the operational amplifier, with multiple negative
feedbacks applied via resistor, R2 and capacitor C2. Then we can define the characteristics of the IGMF
filter as follows:
fr 1 R2
fr =
1 QBP = = Maximum Gain, AV = −
R2
= −2Q 2
2π R1 R 2 C1C 2 BW(3dB ) 2 R1 2 R1
We can see then that the relationship between resistors, R1 and R2 determines the band pass "Q-factor"
and the frequency at which the maximum amplitude occurs, the gain of the circuit will be equal to − 2Q 2 .
Then as the gain increases so to does the selectivity. In other words, high-gain and high-selectivity
159
Example: An active band pass filter that has a gain Av of one and a resonant frequency, f r of 1 kHz is
constructed using an infinite gain multiple feedback filter circuit. Calculate the values of the components
required to implement the circuit.
Firstly, we can determine the values of the two resistors, R1 and R2 required for the filter using the gain
of the circuit to find Q as follows.
2
 
1 1 R2 R2  0.7071 
AV = 1 = −2Q 2 ∴ Q BP = = 0.7071 Q = 0.7071 = ∴ =  =2
2 2 R1 R1  1 
 
 2 
Then we can see that a value of Q = 0.7071 gives a relationship of resistor, R2 being twice the value of
resistor R1. Then we can choose any suitable value of resistances to give the required ratio of two. Then
resistor R1 = 10kΩ and R2 = 20kΩ. The centre or resonant frequency is given as 1 kHz. Using the new
resistor values obtained, we can determine the value of the capacitors required assuming that C = C1 =
C2 .
1 1 1
f r = 1000Hz = ∴C = = = 11.2nF
2πC R1 R2 2πf r R1 R2 2π 1000 10,000 × 20,000
The closest standard value is 10nF.
Resonant Frequency
The actual shape of the frequency response curve for any passive or active band pass filter will depend
upon the characteristics of the filter circuit with the curve above being defined as an "ideal" band pass
response. An active band pass filter is a 2nd Order type filter because it has "two" reactive components
(two capacitors) within its circuit design and will have a peak response or Resonant Frequency ( f r ) at
its "centre frequency", f C . The centre frequency is generally calculated as being the geometric mean of
the two -3dB frequencies between the upper and the lower cut-off points with the resonant frequency
(point of oscillation) being given as: fr = fL × fH Where:
• f r is the resonant or Centre Frequency
160
• f L is the lower -3dB cut-off frequency point

• f H is the upper -3db cut-off frequency point and in our simple example above the resonant centre
frequency is given as: f r = 200 × 600 = 120,000 = 346 HZ
The "Q" or Quality Factor
In a Band Pass Filter circuit, the overall width of the actual pass band between the upper and lower -3dB
corner points of the filter determines the Quality Factor or Q-point of the circuit. This Q Factor is a
measure of how "Selective" or "Un-selective" the band pass filter is towards a given spread of
frequencies. The lower the value of the Q factor the wider is the bandwidth of the filter and consequently
the higher the Q factors the narrower and more "selective" is the filter. The Quality Factor, Q of the filter
is sometimes given the Greek symbol of Alpha, (α) and is known as the alpha-peak frequency where:
1
α=
Q
As the quality factor of a band pass filter (Second-order System) relates to the "sharpness" of the filters
response around its centre resonant frequency ( f r ) it can also be thought of as the Damping Factor or
Damping Coefficient because the more damping the filter has the flatter is its response and likewise, the
less damping the filter has the sharper is its response. The damping ratio is given the Greek symbol of
Xi, (ξ ) where: ξ = 2α
The "Q" of a band pass filter is the ratio of the Resonant Frequency, ( f r ) to the Bandwidth, (BW)
between the upper and lower -3dB frequencies and is given as:
161
Re sonant − Frequency
Q=
Bandwidth
Then for our simple example above the quality factor "Q" of the band pass filter is given as:
346Hz / 400Hz = 0.865. Note that Q is a ratio and has no units.
When analysing active filters, generally a normalized circuit is considered which produces an "ideal"
frequency response having a rectangular shape, and a transition between the pass band and the stop band
that has an abrupt or very steep roll-off slope. However, these ideal responses are not possible in the real
world so we use approximations to give us the best frequency response possible for the type of filter we
are trying to design. Probably the best known filter approximation for doing this is the Butterworth or
maximally-flat response. In the next tutorial we will look at higher order filters and use Butterworth
approximations to produce filters that have a frequency response which is as flat as mathematically
possible in the pass band and a smooth transition or roll-off rate.
162

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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209

5.2. Two-port Parameters .................................................................................................................... 72

Network is a combination of electrical elements connected in any manner (conductively, inductively or

Fig.1.1. Key terms in network analysis & synthesis

1.1. Lumped Circuit elements

Fig.1.2. Linear resistor (a) characteristic curve (b) Circuit symbol

Fig.1.3. Non-linear characteristic curves

characteristic of a semiconductor diode (continuous curve). In analyzing a network with a nonlinear

Fig.1.4. Piece-wise linearization of characteristic curve

Fig.1.5. Linear capacitor (a) characteristic curve (b) Circuit symbol

The current charge relation being i = dq dt , for a linear capacitor, we have

For a voltage controlled capacitor the current through the capacitor is

Fig.1.6. Non-linear capacitor characteristic curve

The instantaneous power entering an element is given by

For a linear capacitor (q = Cv ) it reduces to

Fig.1.7. Linear inductor (a) characteristic curve (b) Circuit symbol

Fig.1.5. Independent source

Fig.1.9. Voltage source:.symbol and the i-v characteristic curve

A two-terminal element is called an ideal independent current source, if it is capable of supplying a

Fig.1.10. Voltage source:.symbol and the i-v characteristic curve

1.2. Classification of Networks

1.2.1 Linear and Nonlinear Networks

Fig.1.11. Linear and Non-linear networks

Linearity of a network can be defined as follow: let a network be characterized by F ( xi ) = yi where xi is

1.2.2 Time-invariant and Time-variant Networks

1.2.3 Passive and Active Networks

2.1.1 Network Functions

where s the Laplace variable is the complex frequency variable .

Fig.2.1. Network used to define network functions

(Driving point impedance)

(Driving point admittance) (2.2)

Transfer Admittance (2.3)

Transfer voltage ratio

Transfer current ratio

Fig.2.2. Ladder network

2.1. Poles and Zeros

Fig.2.4. pole-zero plot

2.2 Stability of a Network

Relationship between impulse response and stability

Relationship between Poles positions and stability

Routh Hurwitz stability criterion

sn-1 an-1 an-3 an-5 …

Special cases when Routh’s tabulation terminates prematurely

3. Replace the row of zeros with coefficients of = 0.

Deficiencies of Routh Hurwitz Criterion

1. Obtain the pole zero location for the function

3 Elements of Realizability Theory

for an amplitude function H ( jω ) to be realizable (causal) is that

2. The square magnitude function H ( jω ) must be integral that is

Fig. 3.2: Ideal filter characteristic

Fig 3.3: Gaussian Filter charactersitc

very small quantity ε , that is h(t − t d ) < ε t < 0 .

Equation (3.7 ) requires that the impulse response be absolutely integrable, or

2. H (s ) cannot have multiple poles on the jω − axis.

3.2. Hurwitz Polynomials

b. The roots of P (s ) have real parts which are zero or negative.

As a result of this conditions, if P (s ) is a Hurwitz polynomial given by

P ( s ) = a n s n + an−1s n−1 + ⋅ ⋅ ⋅ + a1s + a0 (3.11)

The polynomial P ( s ) = ( s + 1)( s + 1 + j 2 )( s + 1 − j 2 ) is Hurwitz because all of its roots

Hurwitz polynomials have the following properties:

The polynomial P (s ) which contains these roots can be written as

brought about by a root s − γ i with a positive real part.

For example, let us test whether the polynomial F ( s) = s 4 + s 3 + 5s 2 + 3s + 4 is Hurwitz or not.