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frequency response

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TOPIC 7 FREQUENCY RESPONSE AND FILTERING

Objectives

The definition of frequency response function, amplitude response and phase response

Bode plots

1

1.1

General

The oscilloscope enables us to observe the transient behaviour of a system's input and output

signals

In the frequency domain, circuits are characterised by their frequency response function

The laboratory instruments which measure frequency content of signals and systems are the

network analyser, the gain-phase test set and the spectrum analyzer

We begin with a definition of frequency response function and explore its use through examples

later

1.2

Definitions

Consider a general linear circuit having no internal independent sources, with a sinusoidal input (or

excitation) waveform x(t) and an output (or response) waveform y(t):

The input signal is represented by its phasor, having amplitude Xm and angle

The steady state sinusoidal response phasor will have amplitude Ym and angle +

The system function is defined by:

H ( ) =

Y Ym + Ym

=

=

X m

Xm

X

We have included the frequency as an argument for the system function because, in the case of

circuits containing inductors and/or capacitors, the ratio of the output phasor to the input phasor is

frequency dependent

It is common practice to write H(j) for H() because can arise only from the impedance of an

inductor (jL) or from the impedance of a capacitor 1/(jC), where it is always associated with j.

H() contains all the information we need to know about the circuit, provided we know its value for

each value of

1.3

The frequency-domain description of a signal consists of a phasor with amplitude and phase at a

particular frequency

We adopt the same idea for the system function H()

We define the amplitude response function:

A ( ) = H ( ) =

Ym

Xm

( ) = H ( )

Notice that since H() is a function of , both A and are functions of

We can now state that the amplitude of the circuit output signal is given by:

Ym = A ( ) X m

and its phase by

Y ( ) = ( ) +

Thus, we multiply the input signal amplitude by the system gain and add its phase angle to the

system phase shift to obtain the output amplitude and phase angle, respectively

1.4

We show a laboratory signal generator applying a cosine waveform with unit amplitude and general

frequency to our system under test

The AC steady-state response is measured at a frequency 1, then the input signal is changed to a

new frequency 2 and the response is measured again

By measuring the steady-state output signal, we can determine the value of the gain A() and the

phase shift () at each frequency

We can then plot the frequency response, that is, the gain and the phase versus frequency, as

follows:

Gain

Phase Shift

Laboratory instruments can automatically vary the frequency, determine the gain and phase shift at

each frequency; they are called gain and phase test sets or network analysers

1.5

Example

Suppose that we run a test by applying a sinusoidal source at the circuit input and adjusting it so

that its amplitude is one volt and its frequency is variable

We can use AC steady-state analysis to compute the response

The AC steady-state equivalent circuit is as follows:

Using the voltage divider rule, the response phasor is:

1

1

1

j C

Y=

10 =

10 =

tan 1 ( CR )

1

2

1

+

j

CR

1 + ( CR )

R+

j C

Since 100 is the input phasor X , we may write:

Y=

1

1

10 =

X

1 + j CR

1 + j CR

H ( ) =

Y

1

=

X 1 + j CR

Thus:

A( ) =

1

1+ (CR)

( ) = tan1 (CR)

The plot of the amplitude response function A() of the circuit is as follows:

The amplitude response of this circuit passes low frequencies and suppresses high frequencies

Higher-frequency components in any input signal will be attenuated much more than those at low

frequencies; a filter with such a response is called a lowpass filter

Furthermore, if we make the time constant RC larger, we reduce the high-frequency gain, and if we

make the time constant smaller, we increase the high-frequency gain

Thus a circuit of resistors and one or more reactive elements not only has a transient response but

also has a frequency response function which can effectively filter signals; we consider an example

2

2.1

Illustrative example

Suppose a sinusoidal signal is transmitted over a noisy channel that adds an interfering signal that

we wish to remove by a filter circuit

Lets assume the "noisy" signal is described by:

x ( t ) = cos(2t ) + 0.5cos(200t )

The term cos(2t ) is the desired signal and the term cos(200t ) represents the additive noise

A plot of one period of the signal with noise is shown:

:

If we could come up with a "filter" that would pass the sinusoid at the frequency of 2 rad/s and

block the one at 200 rad/s, we would have achieved our objective

We will assume that the noisy signal is available as a voltage source having the prescribed x(t) as its

waveform

Let us use the simple RC lowpass filter circuit for which we derived the amplitude response

function and plotted it:

4

A ( ) =

1 + ( CR )

The noise in our signal is a sinusoid with a frequency 100 times that of the signal

Consider making the RC time constant such that:

2 rad/s <<

1

<< 200 rad/s

RC

1

= 20 rad/s

RC

This is 10 higher than our signal frequency and 10 lower than the noise frequency

We can calculate the amplitude response of the circuit at the signal frequency and at the nose

frequency:

1

A ( ) =

1 + ( CR )

A ( 2 ) =

1

2

1+

20

A ( 200 ) =

1+

20

1

1

1+

10

1

200

1+

20

= 0.995

1

1 + (10 )

= 0.0995

We can use superposition to compute the circuit output signal resulting from our noisy signal as the

input signal:

where we have calculated the phase response at the two frequencies, but the phase is irrelevant in

this case

The signal component has been almost unchanged (amplitude reduced by about one-half of one

percent), but the noise waveform has been attenuated by a factor of 1/20

5

We have successfully used our frequency domain approach to come up with the circuit that did the

job we wanted it to do

The circuit we used is called a 1st order lowpass filter

The corresponding 1st order highpass filter shown:

The circuit is the same as the one we have worked with, except that the capacitor and resistor have

been interchanged

We can use the voltage divider rule to obtain the frequency response function and the amplitude and

phase responses:

H ( ) =

A ( ) =

Y

R

1

1

1

=

=

=

tan 1

1

CR

2

X R+ 1

1 j

1

1+

j C

CR

CR

1

1

1+

CR

1

( ) = tan 1

CR

We have:

A ( ) 0 = 0

A ( ) = 1

Hence, for this circuit, a high-frequency signal, considered to be the signal, is passed, and a lowfrequency signal, considered to be the noise, is blocked

The plot of the amplitude response is as for the lowpass filter but with the frequency scale inverted

2.2

Radio stations are limited by law to have a certain bandwidth

The frequency plot shows the permissible frequency limits for two hypothetical radio stations,

KOKA and KOLA

The continuous curve represents the amplitude spectrum of the very weak signal that is picked up

by the antenna of our radio receiver

The signal strength of the one we desire (say KOKA) is weaker than the other (KOLA)

If we do not select just one station and reject the other, we will hear a mixture of both stations in

our speaker

Our solution is to pass the composite signal through a bandpass filter

Ideally, the filter should have an amplitude response having the rectangular shape shown in the

figure

2.3

Consider the spectrum of the audio signal in a power amplifier in an audio sound system:

It is unfortunately true that an amplifier with a very high gain also picks up unwanted disturbances,

and one of the most common interfering signals comes from the AC power cables

This interference is in the form of a sinusoidal signal whose frequency is 50 Hz, or 100 rad/s,

which causes a hum in the speaker

Assuming that the audio signal components in a frequency range close to the interfering frequency

of 100 rad/s are not very important to the intelligibility of the waveform, we can pass the

composite waveform through a band-reject (or bandstop) filter having the frequency response

shown above

This eliminates only the interfering waveform and passes our desired audio signal relatively

unaffected, providing the "notch" is very narrow

Thus there are a number of standard filter frequency response types that can be applied in a host of

practical situations

They are called lowpass, highpass, bandpass and band-reject filters

Their ideal gain versus frequency templates are shown:

lowpass

highpass

bandpass

7

band-reject

Actual filters will not have these "brick wall" responses; that is, they will not change abruptly from

one value to another as the frequency changes

The first-order RC lowpass filter was far from its ideal template

With more circuit elements and more sophisticated design procedures, one can approximate the

ideal filter frequency response characteristic much more closely

There are catalogues containing tables of pre-designed filters based upon the standard types and

some standard types of approximating functions, such as Butterworth, Chebyshev and elliptic

In some applications, such as audio, the phase response of a filter has very little effect on perceived

sound so it is sufficient to consider only the amplitude response

In other applications, such as television, phase characteristics are important

In general, for distortion-less filtering of a signal, the filter gain should be constant and the phase

should be linear over the frequency range of the waveform

One standard filter type we have not mentioned is the all-pass filter

The gain is constant with frequency but the phase characteristic can be used to compensate for

nonlinear phase characteristic in another circuit

Amplitude

3

3.1

Phase

Definition

The simplest example of a bandpass filter is the LC tuned circuit

Ideal inductors and capacitors used in such a circuit should only store energy and not dissipate any

Practical inductors and capacitors do dissipate some energy

We start by defining a factor that measures the quality of an energy storage element

Consider the AC steady-state response of the two-terminal element shown:

Q = 2

w

= 2 P

energy dissipated per cycle

wD

In general, we will pick either the voltage or the current to have zero phase angle as a reference

For an ideal lossless inductor or capacitor, the energy dissipated per cycle is zero, implying that the

Q-factor is infinite

8

3.2

Q of a lossy inductor

Inductors are constructed in the form of a coil of wire having finite resistance

Thus, a practical model of an inductor consists of an ideal inductor in series with a small resistance

r s:

i( t ) = Im cos(t + )

The energy stored in an inductor is given by:

1

w ( t ) = Li 2 ( t )

2

In AC steady-state terms, the stored energy is:

1

w ( t ) = LIm2 cos 2 (t + )

2

The peak value of stored energy is:

1

w P = LIm2

2

The only element that dissipates (or absorbs) energy is the resistor which shares the same current as

the inductor

Energy is the integral of power; the energy dissipated over one full period is:

wD =

1

0 P (t)dt = T T 0 P (t)dt = T P (t)

T

2

2

wD = T rs i 2 ( t ) = T rs I m

cos2 ( t + ) = Trs I m

cos2 ( t + )

cos2 ( x ) =

1

1

1+ cos(2x )) =

(

2

2

Hence

1

w D = Trs Im2

2

Therefore, the Q of the lossy inductor is:

1 2

LIm

w

2L L

QL = 2 P = 2 2

=

=

1

wD

2

Tr

rs

s

Trs Im

2

9

Example 1

A lossy inductor has a series winding resistance of 10 and a nominal value of 10 mH

Find the quality factor at a frequency of 100 krad/s

We then have:

QL =

3.3

L 10 5 102

=

= 100

rs

10

Q of a Lossy Capacitor

A capacitor is constructed in the form of parallel metal plates (perhaps rolled up or folded in the

final construction phase) separated by some sort of dielectric

Thus the dielectrics finite resistance can be approximated by the equivalent circuit shown:

v ( t ) = Vm cos(t + )

The energy stored on the ideal capacitor as a function of time is:

1

1

w ( t ) = Cv 2 ( t ) = CVm2 cos2 (t + )

2

2

The peak energy stored is, therefore,

1

w P CVm2

2

The energy dissipated in one period in the resistor is:

1 2

TVm2

2

wD = T P(t) = T

Vm cos (t + ) =

rp

2rp

Hence the capacitor Q-factor is given by:

1

CVm2 2Cr

wP

p

QC = 2

= 2 2 2 =

= Crp

wD

T

TVm

2rp

Example 2

A lossy capacitor has a parallel dielectric resistance of 10 M and a nominal value of 10 nF

Find the quality factor at a frequency of 100 krad/s

QC = Crp = 10 5 10 7 108 = 10 4

10

3.4

We note that the inductor and capacitor Q-factors may be written in the following form:

QL =

L XL

=

rs

rs

QC = Crp =

rp

1 ( C )

rp

XC

We note also that the expressions are dimensionless ratios of impedances involving a wanted (X)

and an unwanted (r) quantity

Note also that the conditions that make the elements ideal, rs 0, rp , both make Q

These considerations allow us to predict Q expressions for other combinations

For example, inductor L with parallel resistance rp must have Q for rp ; so the Q

expression must be:

QL =

rp

XL

rp

Capacitor C in series with resistance rs must have Q for rs 0; so the Q expression must be:

QC =

3.5

XC 1 C

1

=

=

rs

rs

Crs

Let's compute the admittance of the series sub-circuit:

Y ( j ) =

1

r j L

= s

=

rs + j L rs2 + ( L )2

1 j

L

rs

L2

rs 1 +

rs

1 1 jQL

rs 1 + QL2

Y ( j ) =

1 1 jQL

1

1

1

1

1

1

=

j

=

+

=

+

rs QL2

QL rs QL2 rs j L r QL2 rs j L

QL2 rs

rs s

11

where

rp' = QL2 rs

L'= L

Note that this equivalent circuit depends on frequency because QL is a function of frequency

The series and parallel circuits can be equivalent only at one frequency

3.6

The equivalence that we have just derived is strictly valid only at a single frequency because rp'

depends on QL which depends on frequency

If we are only interested in small percentage frequency changes relative to the centre frequency, we

can assume that QL is a constant whose value is that assumed at the centre frequency

This is called the narrow-band approximation

Example 3

Find the parallel equivalent circuit for the lossy inductor of Example 1 with rs = 10 , L = 10 mH at

100 krad/s.

Then find the percentage error in rp varies over a frequency range of 99 krad/s to 101 krad/s

Solution

We have already computed the inductor Q to be 100 at 100 krad/s

Assuming that QL >> 1, we have the general expression for rp':

rp'

= QL2 rs

2

L 2

L )

(

= rs =

rs

rs

rp'

(L)

=

rs

10 5 102 )

(

=

10

= 10 5 = 100 k

At = 99 krad/s, we have:

rp' =

(L)

rs

99 10 3 102

= 98.01 k

10

2

10110 5 102

L )

(

'

rp =

=

rs

10

= 102.0 k

Thus, we see that a variation of 1 % in the frequency results in only a 2% variation in the

resulting parallel resistance

Hence, the use of a constant rp' of 100 k over this frequency range might be acceptable

12

4

4.1

.

We assume that the driving source is a sinusoidal current source

We wish to determine the AC steady-state response for the voltage v(t).as a function of frequency

The phasor form of the circuit is:

The frequency appears in the element impedances, so we do not have to make a special note of its

value on the phasor circuit diagram

The impedance of the two-terminal sub-circuit is:

1

L

j C

Z ( j ) =

= j

= jX ( )

2

1

1

LC

j L +

j C

j L

The reactance X() (the imaginary impedance without the j multiplier) can be plotted versus :

0 =

1

LC

This phenomenon is called resonance and the frequency 0 is called the resonant frequency

4.2

Now let's consider the practical situation where the inductor and the capacitor both have finite Q

13

If we perform the series-to-parallel transformation on the lossy inductor, we can combine the two

resistors into one

We then obtain the following equivalent sub-circuit:

We note that the parallel resistor is a composite of the loss resistance rp of the capacitor, the

(transformed) parallel equivalent resistance rp of the inductor, and any source resistance that might

be present (Norton equivalent for the driving source)

The narrow band approximation is based on the assumption that the resistance R is a constant over

the frequency range of interest

Our first objective will be to find the Q of the sub-circuit at the resonant frequency 0 = 1

LC

We apply a test current source to our sub-circuit and adjust it to be a sinusoid at frequency 0

At the resonant frequency 0 = 1 LC we know that the part of the sub-circuit consisting of the

capacitor and the inductor presents an infinite impedance, Z'(j0) =

(this part of the circuit is the same as a lossless tuned circuit we analysed earlier)

The impedance at the two terminals of the subcircuit is

Z ( j 0 ) = R

Thus, the phasor terminal voltage: is:

V = RI = RIm0

In other words:

Vm = RIm

We know that the energy stored on the capacitor as a function of time is:

14

wC ( t ) =

1 2

1

Cv ( t ) = CVm2 cos2 ( 0t )

2

2

To find the energy stored by the inductor, we first determlne the inductor current in phasor form:

IL =

V

V 0

V

= m

= m 90

j 0 L 0 L90

0L

iL ( t ) =

Vm

V

cos 0 t 90 = m sin( 0 t )

0L

0L

2

2

1 2

1 Vm

1 Vm

wL ( t ) = LiL ( t ) = L

sin ( 0t ) =

sin 2 ( 0t )

2

2

2 0 L

2 L

We can substitute 0 = 1

LC to show that:

1

w L ( t ) = CVm2 sin 2 ( 0 t )

2

wC ( t ) =

1

CVm2 cos2 ( 0t )

2

This shows that the peak energy stored in the inductor is the same as the peak energy stored in the

capacitor

It also shows that the sum of the inductor and capacitor energies is constant

This means that when the capacitor is storing its maximum energy, the inductor is storing no

energy, and vice versa

The energy is being swapped back and forth between the capacitor and the inductor, and none is

coming from the source

Because the impedance Z'(j0) = the current into the parallel LC combination is zero when =

Hence, at = the source current of our sub-circuit only feeds the resistor

The energy absorbed by the sub-circuit in one period is the same as the energy absorbed by the

resistor in that period:

wD =

1 Vm2

T0

2 R

15

The Q of the sub-circuit at = 0, which we will call Q0, is:

1

CVm2

wP

2

Q0 = 2

= 2

= 0 RC

wD

1 Vm2

T0

2 R

Using LC = 1/02, we can write an alternative expression for Q0:

1

R

Q0 = 0 R

=

2 L

L 0

0

Thus it is shown that the Q-factor of the tuned-circuit is the same as the Q-factor of the inductor or

the Q-factor of the capacitor where the same resistor R is used in each case

4.3

Now let's return to our parallel tuned-circuit and compute the impedance at its terminals as a

function of the general frequency :

We redraw the circuit in the phasor domain:

Z ( j ) =

1

1

1

=

=

2

1

1

1

1

0

+ jC +

+ j 0C 0

+

C

j

R

jL R

j R

0

R

R

=

0

0

1+ j 0 RC 1+ jQ0

0

0

The expression in the denominator brackets is called the fractional frequency deviation

The magnitude and phase of Z(j) are:

Z ( j ) =

0

Z ( j ) = tan 1 Q0

1 + Q02

16

The response is plotted for two different values of Q0

The larger the value of Q0, the more selective is the tuned circuit

If we were tuning in a radio station, we would need a large Q0 if there was another station very

close in frequency

4.4

Bandwidth

The passband may be defined formally as the range of frequencies for which the normalized gain is

greater than 1/2

There are two frequencies, one above 0 and the other below 0, at which the response drops to this

value

We call these the upper cut-off frequency U and the lower cut-off frequency L, respectively

B = U L

Z ( j )

R

1 + Q02

Q02

0

2

0

=1

or

17

Q0 0 = 1

0

Q0 U 0 = +1

0 U

Q0 L 0 = 1

0 L

x

1 1

=

x Q0

or

x2

1

x 1 = 0

Q0

The solution is

1 2

1

x=

+1

2Q0

2Q0

1 2

1

U = 0

+

+ 1

2Q0

2Q0

1 2

1

L = 0

+

+ 1

2Q0

2Q0

B = U L =

0

Q0

and we have:

Q0 =

0

0

=

B U L

Notice that L and U are not arithmetically symmetric relative to 0 (this would imply

L + U = 2 0 )

It may be shown that they are geometrically symmetric, ie LU = 0

How might we measure 0 and Q0 in the laboratory?

Looking back at our resonance curve, we see that we can search for the peak frequency, which is 0

Next, we find the upper cut-off frequency by noting the frequency at which the response has

dropped to 1/2 times the peak value (1/2 = 0.707) and identifying it as U

18

Finally, we find L in a similar manner and subtract them to find the bandwidth

Then we can compute Q0

4.5

Phase Shift

Although we have considered the magnitude or gain characteristic, the phase shift is important for

some applications, including TV and digital transmission

It is of interest to see how the phase shift of the parallel tuned circuit varies with frequency

We now plot () as given by the equation we derived for two values of Q0:

0

Z ( j ) = tan 1 Q0

Observe that the phase shift is zero at resonance and approaches /2 rad (90o) for frequencies

below and above the resonant frequency

We can define resonance as the frequency at which the tuned-circuit impedance is purely resistive

(() = 0)

A look at the equation for () shows immediately that this frequency is 0

5

5.1

Its behaviour is very much like that of the parallel tuned circuit, except that all the properties of the

latter which we considered on an impedance basis hold for this circuit on an admittance basis

The admittance of the series tuned circuit is:

Y ( j ) =

1

1

1

j

L

=

=

=

j

= jB( )

Z ( j ) jL + 1

L ( j ) 2 + 02

02 2

jC

where

0 =

and B() is the susceptance

19

1

LC

Notice that the susceptance approaches infinity at = 0; this means that it is equivalent to a short

circuit at 0

The admittance of the series tuned circuit behaves precisely like the impedance of the parallel tuned

circuit

5.2

Y ( j ) = jC +

1

rP

Z ( j ) =

1

1

rP

rP

r jrPQC

=

=

=

= P

Y ( j ) 1 + j C 1 + j CrP 1 + jQC

1 + QC2

rp

Z ( j )

rP jrPQC

QC2

rP

QC2

rP

r

1

= P2 +

j CrP QC j C

with

rS =

rP

QC2

C'= C

Since QC depends on frequency, this equivalence is strictly valid only at the frequency for which QC

is determined

20

5.3

Look, once again, at the series tuned circuit, assuming that both the inductor and the capacitor are

lossy:

The series resistance R includes any driving source resistance (considered as a Thevenin equivalent)

along with the inductor series loss resistance and the transformed capacitor parallel loss resistance

Let's apply a sinusoidal test voltage source to our sub-circuit and test the Q of the entire sub-circuit

at the frequency 0

So Z(j0) = R

Thus, the current is in phase with the source voltage; furthermore, the voltage drop across the

inductor/capacitor series combination is zero because its series impedance is zero

Hence, we can immediately compute

1 Vm2

L 2

w

L

Q0 = 2 P = 2 2 2R = 0

wD

R

1 Vm

T0

2 R

where T0 is the period of a sinusoid at the resonant frequency 0

By making use of the equivalence:

0 =

1

LC

Q0 =

0 1

1

=

2

R C 0 0CR

As a check, we can see from the circuit that the infinite Q condition is R = 0

It is easy to show that the peak energies stored by the inductor and the capacitor are equal to one

another - and therefore to the peak energy stored by the sub-circuit itself.

5.4

21

Z ( j ) = R + jL +

L j

1

= R1+ 0

+ 0

jC

R 0 j

1

R

Y ( j ) =

0

1+ jQ0

0

If we compare this with the Z(j) for the parallel tuned circuit developed in the preceding section,

we will see that they are identical in form

The magnitude and phase are:

Y ( j ) =

1

R

0

1 + Q02

0

Y ( j ) = tan 1 Q0

The admittance of the series tuned-circuit has the same form as the impedance of the parallel tunedcircuit

5.5

A two-terminal circuit is said to be resonant at any nonzero frequency for which the AC steadystate impedance is purely real

This does not always occur where an inductor and a capacitor produce equal and opposite

reactances, as shown in the next example

Example 6

Find the resonant frequency of the following circuit:

22

Solution

The AC steady-state impedance is:

1

R (1 j CR )

R

j C

Z ( j ) = j L +

= j L +

= j L +

2

1

1 + j CR

1 + ( CR )

R+

j C

R

R

1 + ( CR )

CR 2

+ j L

2

1 + ( CR )

CR 2

L

1

L

02 =

1

LC CR 2

1+ ( 0CR) =

2

0 =

1

L

1

LC

CR 2

If R , we see that the resonant frequency approaches that of an ideal series tuned circuit

The second radical represents the change in resonant frequency due to finite R

6

6.1

General

The tuned circuits which we have investigated have impedance and asmittance functions which

exhibit frequency selectivity

We now look into filters; a filter may be defined as a 2-port circuit whose frequency response

function H ( ) = Vo Vi exhibits frequency selectivity

We begin by considering some examples

6.2

Filtering Examples

Example 4

Find the voltage transfer function of the circuit shown and plot its gain and phase characteristics

23

Solution

Notice that there is a series tuned circuit between the voltage source (the filter input) and the output

terminal

Calling this impedance Z(j), we have:

H ( ) =

Vo

RL

=

Vs Z ( j ) + RL

Now we already know the functional form of Y(j) for the lossy series tuned circuit:

Y ( j ) =

Rs

0

1 + jQ0

Hence, we obtain

H ( ) =

RL

1

+ RL

Y ( j )

RL

0

RL + Rs + jRsQ0

RL

H0

RL + Rs

=

0

0

Rs

1+ j

Q0

1 + jQ0'

RL + Rs

0

0

where

H0 =

RL

Rs + RL

Q0' =

Rs

Q0

Rs + RL

Thus, our voltage transfer function has the same form as the admittance of the series tuned circuit or

the impedance of the parallel tuned circuit

There are only two differences: Qo has been decreased to Q0 by the additional resistance and Ho is

a voltage gain, not an admittance or impedance

The gain and phase variations with frequency are shown:

Note that they are identical in form with those for the parallel and series tuned circuits

The next example discusses a second-order bandstop filter

Example 5

Find the voltage gain transfer function of the circuit below and plot its gain and phase versus

24

Solution

We start by recognizing that a major component of this circuit is a parallel tuned circuit, whose

impedance we have already determined

Thus, we can write the voltage gain transfer function as:

0

RL 1 + jQ0

0

Vo

RL

RL

H ( ) =

=

=

=

RP

Vs Z ( j ) + RL

+ RL R + R 1 + jQ 0

P

L

0

0

0

1 + jQ0

0

0

0

1 + jQ0

1

+

jQ

0

0

RL

0

=

= H0

RP + RL

0

0

RL

1+ j

Q0

1 + jQ0'

RP + RL

0

0

where

H0 =

RL

RL + RP

Q0' =

RL

Q0

RL + RP

Let's investigate this function by converting it to Euler form by taking the magnitude and angle:

2

0

0

2

0

'2

1+ Q0

0

A( ) = H ( ) = H 0

1+ Q02

and

( ) = H ( ) = tan1Q0

0 tan1Q0'

0

0

0

25

6.3

For the above example, we can predict the type of response it has before we carry out detailed

analysis

Consider the behaviour of the inductor and capacitor at the extreme frequencies = 0 and

Z 0

Element Type

Impedance

Inductor

Z L = j L

ZL = 0

ZL =

(short-circuit) (open-circuit)

Capacitor

ZC =

1

j C

ZC =

ZC = 0

(open-circuit) (short-circuit)

Hence the inductor behaves like a short-circuit at zero frequency and an open-circuit as frequency

tends to infinity

The capacitor behaves like an open-circuit at zero frequency and a short-circuit as frequency tends

to infinity

Note that the equivalents for zero frequency are the same as the DC steady state equivalents we

used in transient analysis (setting frequency to zero implies that the analysis is a DC analysis)

Note further that at resonance, the parallel tuned-circuit is equivalent to an open-circuit and the

series tuned-circuit is equivalent to a short-circuit

We can now apply this to our band-stop circuit example:

At zero frequency the inductor is a short-circuit and at infinite frequency the capacitor is a shortcircuit; therefore vo = vs at these extreme frequencies; this implies that the amplitude response is

unity and the phase is zero

At resonance, the parallel tuned-circuit will become an open-circuit and the circuit simplifies to a

potential divider consisting of Rp and RL

The gain depends on the resistor values and is H0 = 0.1 in this case

The phase at resonance must be zero

Notice that a good notch filter (this one is not terribly good!) would have a large value of Rp relative

to the value of RL and a high Q0

Thus, the notch would be narrower and deeper

This by-inspection approach can be applied to most passive circuits in order to determine the type

of response (low-pass, high-pass, band-pass, band-stop, all-pass)

26

6.4

We now present a set of standard, or canonical, forms for the three most basic standard 2nd-order

filter types: lowpass, bandpass, highpass and bandstop:

H LP ( ) =

H BP ( ) =

H HP ( ) =

1

2

j

1 j

+ Q + 1

0

0 0

1 j

Q0 0

2

j

1 j

+ Q + 1

0

0 0

j

0

j

1 j

+ Q + 1

0

0 0

2

H BS ( ) =

j

+ 1

0

2

j

1 j

+ Q + 1

0

0 0

These functions all have the same denominator

If we wish, we can relate these expressions to those for tuned circuit impedances and/or admittances

by restructuring the denominator polynomial:

2

j

0

1 j

j 1

D ( ) =

+

+1=

1 + jQ0

Q0 0

0 Q0

0

0

The second factor is one that we easily recognize to be the denominator of our tuned circuit

admittance or impedance

The above forms clearly show the following limiting values:

0

HLP

jQ0

HBP

HHP

jQ0

HBS

If Q0 >> 1, then the lowpass and highpass filters have substantial gain at the frequency 0

However, for the lowpass filter, the actual maximum gain occurs slightly below 0 and is a little

higher than Q0; it may be shown that:

27

H LP ( ) max =

Q0

1

at

1

2Q02

p = 0 1

1

2Q02

We can plot amplitude and phase response for the lowpass filter for a range of Q values

It can be seen that for lower values of Q the frequency of peak gain is significantly less than 0

Notice also that the higher the value of Q, the steeper the phase curve at = 0

For the highpass filter, the maximum gain is the same but it occurs slightly above 0:

p =

1

2Q02

The plots of gain and phase are the same as for the lowpass filter but with the frequency scale

inverted about 0

7

7.1

General circuit architectures

Rather than looking at specific examples, in this section, we take a more general approach

Consider three general impedances connected in series:

We can regard this circuit as having been obtained by de-activating the sources in a number of

possible compete circuits; the short-circuit shows a possible positions for a voltage source

The impedance looking into the short-circuit is:

Z = Z1 + Z 2 + Z 3

We have dropped the (j) or () because the use of upper case letters shows that we are working in

the frequency domain

We now form new 2-port circuits by introducing a source into the above circuit in such a way that

de-activation of the source gives the original circuit:

28

H ( j )

Circuit

Vo

Z3

=

Vi Z1 + Z 2 + Z 3

Vo

Z2 + Z 3

=

Vi Z1 + Z 2 + Z 3

We have used voltage division to determine the frequency response functions for these two circuits

Notice that in both cases the denominator of the expression is the same as the impedance of the deactivated circuit; therefore the denominator of the circuit depends on the de-activated circuit; it

governs the natural response of the circuit independently of sources

Consider now three general admittances connected in parallel:

We can regard this circuit as having been obtained by de-activating a source in a number of possible

compete circuits; the open-circuit shows a possible position for a current source

The admittance of the circuit looking into the open-circuit is:

Y = Y1 + Y2 + Y3

We now form new circuits by introducing sources into the above circuit in such a way that deactivation of the new circuits gives the original circuit:

H ( j )

Circuit

Vo

Y1

=

Vi Y1 + Y2 + Y3

Vo

Y1 + Y2

=

Vi Y1 + Y2 + Y3

We have used voltage division to determine the frequency response functions for these two circuits

29

Notice that in both cases the denominator of the expression is the same as the admittance of the deactivated circuit; again the denominator of the circuit depends on the de-activated circuit; it governs

the natural response of the circuit independently of sources

We are now ready to consider replacing general impedances and admittances by R, L and C

elements

7.2

The first of the four architectures, based on series impedances, can generate three useful circuits,

which we show together with their frequency response functions:

H(j)

Circuit

H ( j ) =

H ( j ) =

H ( j ) =

Type

( j )

LC + j CR + 1

( j )2 LC

( j )2 LC + j CR + 1

j CR

( j )2 LC + j CR + 1

Lowpass

Highpass

Bandpass

The frequency response functions can be confirmed by use of the voltage division rule

The second architecture generates only one useful circuit:

H(j)

Type

2

j ) LC + 1

(

H ( j ) =

( j )2 LC + j CR + 1

Bandstop

Circuit

Notice that all of these four circuits have the same denominator expression; this is expected because

they all reduce to the same form when the input voltage source is deactivated

The third architecture, based on parallel admittances, can generate three useful circuits:

H(j)

Circuit

H ( j ) =

H ( j ) =

30

Type

( j )2 LC + j L

R +1

( j )2 LC

( j )2 LC + j L

R +1

Lowpass

Highpass

H ( j ) =

j L R

( j )

LC + j L R + 1

Bandpass

H(j)

Circuit

H ( j ) =

Type

( j )2 LC + 1

( j )2 LC + j L

R +1

Bandstop

The four circuits based on parallel admittances have the same denominator expression as expected

The general 2nd order denominator expression is:

2

j

1 j

D ( ) =

+

+1

Q0 0

0

where 0 is the resonant frequency and Q0 is the Q-factor

Comparing this with the expressions in the above tables, the first design equation is:

LC =

02

The second design equation for the first set of circuits is:

CR =

R=

1

LC

=

0Q0

Q0

1

Q0

L

C

The second design equation for the first set of circuits is:

L

1

LC

=

=

R 0Q0

Q0

R = Q0

L

C

The 2nd order allpass response can not be produced by such simple circuits; we either need to use a

passive circuit with a lattice structure or we can realise it with an active circuit using an operational

amplifier or transistors

8

8.1

BODE PLOTS

Introduction

The Bode plot method is a way of rapidly sketching the gain and phase response of a circuit from its

transfer function

31

Let's start with an example:

Example 7

Find the transfer function H ( ) = Vo Vi for the circuit shown:

This circuit is a model for a voltage amplifier, such as the "preamp" in a stereo system

Note that the circuit includes a voltage-controlled voltage source

Solution

As we are interested in frequency response, we use the phasor equivalent circuit:

If use a unit of impedance of k. and a unit of current of mA, the unit of voltage will remain V

We can analyse the circuit in stages:

Vx

=

Vi

1

10 5

1 + 1 +

j

j

j2 + 10 5

1

j

2 j + 5 10 4

Let the voltage of the voltage source, 100Vx , be denoted Vy ; then we have:

Vy

Vx

= 100

and

Vo

Z2

1

1

1 10 7

=

=

=

=

1

2 j + 10 7

Vy Z1 + Z 2 Z1Y2 + 1

7

2 + j 10 + 1

2

H ( ) =

25 10 7 ( j )

Vo Vo Vy Vx 1 10 7

1

j

=

=

100

=

2 j + 5 10 4

Vi Vy Vx V1 2 j + 10 7

j + 5 10 4 j + 10 7

32

)(

H ( ) =

8.2

K ( j )

( j + p1 )( j + p2 )

By generalizing upon the preceding example, we see that the system function of any response

variable of a circuit constructed from our standard supply of elements, namely R, L, C, and

dependent sources, has the form

H ( ) =

Vo

( j + z1 )( j + z2 ) ... ( j + zm )

=K

Vi

( j + p1 )( j + p2 ) ... ( j + pn )

We call the factors (j + zl), (j + z2) ... (j + zm) the zero factors of H(w) and the factors (j + p1),

(j + p2) ... (j + n) the pole factors of H()

K is the scale factor

In the previous example, there is only one zero factor j with z1 = 0

There are two finite pole factors: p1 = 5 104 and p2 = 107

The scale factor is K = 25 107

These zero and pole factors are all simple, or of order one, ie none are repeated

If we plot the gain frequency response of a circuit such as the example circuit, we will obtain a

graph with the general shape shown:

The response is constant over a very wide frequency range; the value of gain over this frequency

range is called the midband gain Amb

8.3

Decibels

We investigate an experimental plot of A() taken in the lab and discover that Amb = 100

We notice something strange about the plot in the low frequency range where A() has a value of

approximately 1

As we wish to investigate this effect more closely, we adjust the scale of our plot so that a gain of

one corresponds to a height of, say, 1 cm

If we wish, however, to closely investigate the shape of the plot in the midband range as well-where

A() is l00, we find that our plot must be at least 100 cm tall

This would be a very unwieldy plot to handle

For this reason, we perform a nonlinear transformation on our scale by setting:

AdB = 20log A( )

The logarithm is to the base 10, not to the base e

33

The Bel (after Alexander Graham Bell, inventor of the telephone) was first defined as the unit of the

logarithm of a power ratio:

P

Power ratio in Bels = log 2

P1

It was then discovered that for most applications this unit was too large; thus, the decibel became

the standard

This was defined by the equation

P

Power ratio in deciBels = 10log 2

P1

Assuming that powers are developed in resistors with sinusoidal voltage waveforms, we note that:

Pi =

1 Vm2

2 R

Thus, if both powers are measured relative to the same resistance value, then:

V 2

V

Power ratio in deciBels = 10log 22 = 20log 2

V1

V1

AdB = 20log A( )

There are a number of common values that occur frequently:

A()

AdB

0.001

60

0.01

40

0.1

20

1/2

1/2

-3

+3

+6

10

+20

100

+40

1000

+60

Notice that a gain of 0 dB does not mean that the response is zero-it means that it has exactly the

same magnitude as the input

Observe also the special values 2, 1/2 have dB values of 3, and 2, 1/2 have dB values of 6

A gain of 1/x has the same magnitude in dB as that of x but opposite sign

34

Multiplying or dividing gain by a factor of 10 corresponds to adding or subtracting 20 dB

Example 8

Find the values of gain corresponding to 26 dB, 100 dB, and 34 dB

Solution

We recall that log(xy) = log(x) + log(y); thus, adding dB values corresponds to multiplying the

actual gain values

26 dB = 20 dB + 6 dB

20 dB corresponds to a gain of 10 and 6 dB to a gain of 2

Thus, 26 dB corresponds to a gain of 20

100 dB = 201og(x)

x = 10100/20 = 105

34 dB = 40 dB 6 dB

Gain is 100 0.5 = 50.

Logarithms of a few basic positive integers are worth remembering:

log(2) = 0.301 log(3) = 0.477 log(5) = 0.699 log(7) = 0.845

Other values can be easily calculated:

log(4) = log(22) = 2log(2) = 0.602

log(9) = log(32) = 2 log(3) = 0.954

20log(30) = 20log(10 3) = 20log(10) + 20log(3) = 20 + 20 0.477 = 20 + 9.54 = 29.54 dB

8.4

Now look back at the typical amplifier gain curve and consider the frequencies U and L

Suppose we find in our lab experiment that U = 105 rad/s and L = 10 rad/s

Suppose we want to investigate both regions in the frequency plot carefully, so we make 10 rad/s

correspond to 1 cm of horizontal length on our plot

We find that in order to include U = 105 rad/s we must make the plot 104 = 10,000 cm long

For this reason, we perform a logarithmic transformation on the frequency axis:

'= log( )

Note that we do not have the factor of 20 present in this case as we do for the gain

Consider a normal and a log() scale superimposed:

35

We note that multiplying any given frequency value by 10 results in adding one unit of log

frequency; we call such a frequency change a decade

Dividing a given frequency by a factor of 10 results in subtracting one unit of log frequency

Similarly, multiplying or dividing by a factor of 2 results in adding or subtracting 0.3 unit of log

frequency

We call such a frequency change an octave

The name comes from music, where there are eight notes in a frequency increment of a factor of 2

Note that when using a logarithmic frequency scale, the scale may be labelled in or log() units

as shown above

Logarithmic frequency scales are also commonly used when the frequency is in Hz

8.5

A Bode plot is a plot of the gain A() in dB versus log() and of the phase () versus log()

We start from the general expression for the system function of a circuit:

H ( ) =

Vo

( j + z1 )( j + z2 ) ... ( j + zm )

=K

Vi

( j + p1 )( j + p2 ) ... ( j + pn )

We then take the absolute value and note that the absolute value of a product is the product of the

absolute values and that the absolute value of a ratio is the ratio of the absolute values:

A( ) = H ( ) = K

j + z1 j + z2 ... j + zm

j + p1 j + p2 ... j + pn

Finally, we take the base 10 logarithm and multiply by 20 to get the dB value:

AdB ( ) = 20log K + 20log j + z1 + 20log j + z2 + ...+ 20log j + zm

20log j + p1 20log j + p2 ... 20log j + pn

Our interest is to develop quick approximate methods for sketching the Bode plot

Let's investigate the different types of factor one at a time

We assume that all the zero and pole constants (zk and pk, respectively) are real

8.6

We have sketched this factor:

The solid line represents a scale factor with magnitude greater than unity (positive dB value) and

the dotted line one with a magnitude less than unity

36

The horizontal axis is labelled with values of , but distances are plotted in terms of log(). Thus,

the origin is at = 1 rad/s

8.7

Now suppose that q finite zeros or q finite pole constants are equal

Then we have zl = z2 = ... = zq = a or p1 = p2 = ... = pq = a in the general system function

In this case, there is a zero or a pole factor of order q

We will consider factors of the form.

F ( ) = ( j + a)

If the F() term is a qth-order zero factor, it will appear in the numerator

If it is a pole factor, it will appear in the denominator

Consider the first case for which a = 0

Let's consider:

F ( ) = ( j )

F ( ) = ( j ) = j = q

q

We have used the fact that the absolute value of a product is the product of the absolute values

Finally, we take the logarithm and multiply by 20:

( )

F ( ) dB = 20log q = 20qlog( )

As one must subtract the factor if it represents a pole, we have merely changed the sign in order to

obtain the plot for a pole factor:

A 10-fold increase or decrease in frequency results in a 20q dB increase or decrease in the factor

37

It is also 6q dB/octave

Suppose, now, that a 0:

We now have:

F ( ) = ( j + a)

F ( ) = ( j + a) = j + a

q

F ( ) dB = 20log ( j + a) = 20qlog j + a = 20qlog 2 + a 2

q

For this reason, we resort to an approximation, noting that:

20qlog( ); >> a

F ( ) dB

20qlog a; << a

Note that we have used magnitude signs around the parameter a for it could possibly be negative

8.8

We now show the approximation involved in constructing a linearised Bode gain plot where we

assume that the factor is equal to 20qlog() for |a| and to 20q log|a| 1 for |a|

The resulting plots for a zero factor and for a pole factor are as follows:

The frequency = |a| where the staight lines intersect is called the break frequency

Note that the extrapolation of the sloping lines always pass through the origin at (1 rad/sec, 0 dB)

8.9

It can be sown that the maximum error between the straight line approximation and the actual curve

occurs at the break frequency w = |a| and is given by:

Error( = a ) = 3q dB

The plus sign will hold for a zero factor and the negative sign for a pole factor

The true plot is above the approximate one by 3q dB at a zero break frequency and is below it by 3q

dB at a pole break frequency

For a single zero or single pole for which q = 1, the maximum error is 3 dB

An octave above and below the break frequency, we have:

38

1

Error( = 2 a ) = Error = a = 0.97q q dB

1

Error( = 10 a ) = Error =

10

a = 0.04q 0 dB

We now give an example showing how to sketch a Bode gain plot.

Example 9

Suppose that a circuit or system has the system function

H ( ) =

Vo

( j )

= 10 5

Vi

( j + 10)( j + 1000) 2

Solution

The scale factor is K = 105, or 100 dB

It is plotted as follows:

The corresponding factor is plotted as follows:

For the simple pole factor (j + 10) the break frequency is 10 rad/s; the plot is as follows:

Finally, we plot the Bode factor corresponding to the pole factor (j + 1000)2 of order 2:

In this last plot, we have not preserved the same scale as for the others because this would lead to

problems in presentation

Now we simply add all four plots order to obtain the overall Bode gain plot:

39

Notice that we have sketched in the curve, which falls below the linearised approximation

Because the break frequencies are widely separated, the error at each is almost that computed for

each in isolation, i.e., 3 dB at 10 rad/s and 6 dB at 1000 rad/s

The gain for the flat part of the curve may be fixed by adding the individual figures at any

convenient frequency, eg at 10 rad/sec

8.10 The Quick Method

There is a much faster way of doing this type of plot

Looking at the Bode plot produced in the above example, we could have started at very low values

of frequency with a line of positive slope dictated by the order of the zero factor at = 0

Then we could make the slope break upward by an amount 20q dB/dec at each zero break

frequency or downward by an amount 20q dB/dec at each pole break frequency, where q is the

order of each zero or pole

When any frequency being considered is below a given break frequency, we use the constant value;

when it is above that break frequency, we use 20qlog()

Example 10

Use the quick method to sketch the Bode gain plot for the following transfer function:

H ( ) = 4 10 5

( j ) 2 ( j + 1000)( j + 10 4 )

Solution

The first step is to mark the zeros and poles on the log() axis; we use for o for zeros and for

poles:

The bulk of the solution consists merely of steps in sketching the graph, so we merely sketch the

final shape, starting from the low end of the frequency scale:

We can evaluate the approximate magnitude at = 1 rad/s to fix the vertical axis scale:

H (1) 4 10 5

(10)(100)(500) = 2 10 4 86

(1) 2 (1000)(10 4 )

40

dB

The exact plot is 3 dB above our linearized one at the corner frequency at 10 rad/s and 3 dB below

at 104 rad/s

The errors at the other break frequencies cannot be determined so easily because they are not

widely separated from their nearest neighbours

The quick method procedure for sketching Bode gain plots can be summarised as follows:

1. Draw a frequency axis.

2. Evaluate the approximate magnitude at one single value of frequency to fix the vertical scale

(typically at a low value of frequency, perhaps = 1 rad/s) and draw the vertical axis. Note that

"approximate" means to use the linear approximation for each of the factors.

3. Label all the finite break frequencies with an o for a zero and an x for a pole. If the associated

zero or pole is of order higher than one, label its order in parentheses.

4. Start the Bode plot at low values of frequency with a slope of 20q dB/dec., where q is the order

of the zero or pole at = 0, if any. If there are none, q = 0 and the initial slope is zero.

5. Imagine the frequency to increase slowly. Continue drawing the plot, causing the slope to break

upward by +20q dB/dec. at each break frequency associated with a zero of order q and downward

by -20q dB/dec. at each break frequency associated with a pole of order q. Continue this until the

last break frequency has been included.

Example 12

Find the transfer function and sketch the linearised Bode gain and phase plots for the circuit shown:

Solution

We let the common resistor value be R and the capacitor value be C

The phasor domain equivalent circuit is as follows:

H ( ) =

R

1+ jCR j + 100

=

=

1

2

+

j

CR

j + 200

R

jC

+R

1

R+

jC

41

There is one finite zero factor (j + 100) and one finite pole factor (j + 200)

This gives the following gain plot:

Notice that a slope of 20 dB/decade is the same as a slope of 6 dB/octave; so the high-frequency

assymptotic gain value is 0 dB

This checks with the circuit, because the capacitor is a short circuit for high frequencies and an

open circuit for low frequencies

The Bode plot method is not applicable to complex poles and zeros:

Eg 2nd order lowpass filter based on LC tuned-circuit:

H LP ( ) =

1

2

j

1 j

+

+1

0 Q0 0

Denominator can only be factorised into two real factors D( ) = ( j + )( j + ) only if Qo < 1/2

In the case of complex poles and zeros, a computer or calculator is needed to plot A() and ()

9

CONCLUSIONS

In this topic, we have extended the concept of phasor by allowing the frequency to be a variable

rather than a constant

We defined frequency response function, amplitude response and phase response

We looked at series and parallel tuned circuits and defined Q-factor and bandwidth

This led to a discussion of filtering and we considered 2nd-order passive filters

Finally we looked at Bode plots for rapid approximate sketching of circuit amplitude responses

42