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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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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
Chapter 1
1. Introduction to Network Analysis & Synthesis
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
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.
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
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 non- linear 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
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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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.
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).
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
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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.
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.
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.
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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.
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.
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 ) .
say that network N1 is linear and N 2 is nonlinear as the principle of superposition holds for N1 but not
for N 2 .
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.
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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.
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.
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)
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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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
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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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
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.
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)
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.
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.
(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 …
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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.
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.
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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.
Exercise: check the stability of the system with characteristic equation S5 + 4s4+8s3 + 8s2+7s+4 = 0.
Exercise: Determine the range of K so that the system given by characteristic equation
S3+3Ks2+(K+2)s+4 = 0 is stable.
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Tutorial Exercises
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
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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)
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Chapter 3
−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
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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
∞ 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 ) .
∞
∫ 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.
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.
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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
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,
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
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.
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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.
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.
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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.
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
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
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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.
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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
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
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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.
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
K Kσ
Re = 2 > 0 . Therefore, F (s ) is p.r. If F (s ) is an impedance function, then the
s σ +ω
2
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
Z i ( s ) = Z in ( s + α ) (3.19)
where α , the dissipation constant, is real and positive.
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
b. F (s ) may have only simple poles on the jω -axis with real and positive residues.
Let us compare this new definition with the original one which requires the two conditions:
i. F (s ) is real when s is real.
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
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
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 .
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)
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
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
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.
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
The third condition can be checked by first finding the even part of F (s ) which is
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,
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
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 )] .
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
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 )] .
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
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
into a sum of simpler p.r. functions Z1 ( s ), Z 2 ( s),⋅ ⋅ ⋅, Z n ( s) and then to synthesize these individual
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
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
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.
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) .
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 )]
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
Fig.3.6
44
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CHAPTER 4
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)
Before we proceed to synthesis of L-C network, we will first examine some properties of L − C driving
point immittances.
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
(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.
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 .
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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.
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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 )
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.
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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 = ∞ .
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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.
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
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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. 4.11
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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 35 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.
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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.
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.
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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 )
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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)
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.
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.
remove Z (∞ ) as a resistor in the partial fraction expansion. In cases where the numerator is of lower
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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 (∞ ) .
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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.
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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.
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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.
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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 27 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.
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
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 .
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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
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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.
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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.
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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
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CHAPTER 5
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.
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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.
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.
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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).
V1 = Z11I1 + Z12 I 2
V2 = Z 21I1 + Z 22 I 2
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 .
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The equivalent circuit model of the above equations is shown in fig. 5.3.
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
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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,
∆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
Fig. 5.5
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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
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.
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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.
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
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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)
V1 V1
h11 = h12 =
I1 V2 = 0
V2 I1 = 0
I2 I2
h21 = h12 =
I1 V2 = 0
V2 I1 = 0
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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.
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
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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
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.
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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.
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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.
AD − BC =
(YA + Yc )(YB + Yc ) − (YAYB + YBYC + YAYC ) YC2
= =1
2 2
Y C YC
So that it is reciprocal.
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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.
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
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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,
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.
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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
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
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
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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
− 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
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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
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.
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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
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.
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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.
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
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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.
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.
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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
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
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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
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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)
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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
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
( )
Other transfer functions such as current ratios in terms of open circuit and short circuit parameters.
fig. 5.27
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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,
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
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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
fig. 5.30
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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.
Fig. 5.31
Fig. 5.32(a)
Fig. 5.33
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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
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
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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
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 = ∞.
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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 = ∞.
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.
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Exercise
1. Synthesize the transfer impedance given by the following equation:
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Chapter 6
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
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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 .
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
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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.
give the desired amplitude response. For example, if f (ω 2 ) = ω 2 n , then the amplitude function
can be written as
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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.
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.
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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.
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
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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 )( )
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0 1
1 ω
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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
ω ≤1
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
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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
(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
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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.
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
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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.
Finally, to obtain the demoralized Chebyshev poles, we simply multiply s ' k by cosh β κ ' that is,
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.
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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.
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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
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( )
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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.
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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
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
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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 )
Bo = 1
B1 = s + 1
⋅⋅⋅
Bn = (2n − 1) Bn−1 + s 2 Bn − 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
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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.
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
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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
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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
R-C network
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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).
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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.
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.
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Figure: smearing of pulses in systems with long rise and settling times
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.
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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.
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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.
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
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−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.
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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 :
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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: {∗ ∗ ∗}
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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
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
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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
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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
Where R0 is taken to be dimensionless here. Thus for a normalized resistor Rn the denormalized (actual)
resistance is R = R0 R n
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
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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:
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
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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
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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 .
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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
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
Ln ω 02 L n ∆ 1
Ln s n = s+ = Lb 1 s +
BW BWs C b1 s
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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.
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.
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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.
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
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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.
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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.
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.
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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:
1 +
fC
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
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AS&TU, EE Department Network Analysis & Synthesis – EEng 3209
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:
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
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Then the final circuit along with its frequency response is given below as:
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
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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.
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.
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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.
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
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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.
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.
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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.
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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.
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:
f
AF
Voltage − gain, ( Av ) =
Vout
= fC
Where:
Vin f
2
1 +
fC
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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:
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
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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.
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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
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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.
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.
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.
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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.
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-
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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.
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.
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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.
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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.
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− 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.
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.
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
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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
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
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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:
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Re sonant − Frequency
Q=
Bandwidth
Then for our simple example above the quality factor "Q" of the band pass filter is given as:
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.
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