Second Order Filter Functions

The standard form of the transfer function for second order filters is

Vout ( s ) n2 s 2 + n1 s + n0
T ( s) = =
Vin ( s ) ω 
s 2 +  o  s + ωo 2
Q

Because the denominator is a second-degree polynomial, all such filters have

two poles. Since the poles are the zeros of the denominator polynomial, we can
find them by applying the quadratic formula. The location, and nature, of the
poles depends only on the parameters ω o and Q :

If Q < 0.5 , then the two poles of T ( s ) are given by

2
ω  ωo 
p1, 2 = − o ±  −ω o
2

2Q  2Q 

Note that when Q < 0.5 that the radicand is positive so that the radical is real.
Also note that the radical is smaller than ω o 2 Q so that both poles lie on the
negative real axis of the complex s-plane.

If Q = 0.5 , then the two poles of T ( s ) are coincident and are given by

ωo
p1,2 = −
2Q
If Q > 0.5 , then the radicand above is negative so that the radical is pure
imaginary. Thus the two poles of T ( s ) are given by

ω  2  ω 2 
p1, 2 = − o ± ( −1)  ωo −  o  
2Q  2Q  


or

2
ω  ω 
p1, 2 = − o ± j ω o 2 − o 
2Q  2Q 

or

2
ω  1 
p1,2 = − o ± jω o 1 −  
2Q  2Q 

For Q > 0.5 , therefore, the poles form a complex conjugate pair and lie in the left
half of the complex s-plane.
Notice that as Q →∞ , both poles approach the jω axis. (Poles on the jω axis
correspond to constant amplitude oscillations of the kind we will encounter when
we consider oscillators.)

Although the denominator determines the poles of the filter transfer function, the
type of filter is determined by the numerator polynomial. We consider several
cases.

1. Low Pass Filter: n1 = n2 = 0

Vout ( s ) n0
T ( s) = =
Vin ( s ) ω 
s 2 +  o  s +ωo 2
 Q

If we let s = jω to obtain T ( jω ) we see that for frequencies ω << ω o that

n0
T ( jω ) →
ω <<ω o
, a constant
ωo 2

For frequencies ω >> ω o ,

n0
T ( jω ) →
ω >>ω o ω →∞
 →0
−ω 2

Thus, the transfer function with n1 = n2 = 0 is a low pass filter with a break
frequency of ω o . Because of the ω 2 factor in the denominator, the gain in dB,
20 log10 T ( jω ) , falls off at an asymptotic rate of 40 dB/decade at low
frequencies.
Any circuit that exhibits a transfer function of the form

Vout ( s ) n0
T ( s) = =
Vin ( s ) ω 
s 2 +  o  s +ωo 2
 Q

can act as a low pass filter. The equal component Sallen-Key circuit

has the transfer function

 RF 
1
1+ 
Vout ( s ) ( RC ) 
2
R 
T ( s) = =
Vin ( s ) 1  RF  1
s2 + 2− s +
( RC )
2
RC  R 

Thus, this Sallen-Key circuit acts as a low pass filter with

1
ωo =
RC

1 R
=2− F
Q R

1  RF 
no = 1 + 
( RC )
2
 R 
One of the advantages of an equal component Sallen-Key circuit for realizing a
second order low pass filter is that ω o and Q can be adjusted independently by
adjusting C and RF , respectively. We’ll appreciate this advantage more, later.

2. High Pass Filter: n0 = n1 = 0

Vout ( s ) n2 s 2
T ( s) = =
Vin ( s ) ω 
s 2 +  o  s +ωo 2
 Q

If we let s = jω to obtain T ( jω ) we see that for frequencies ω >> ω o that

n2 ( − ω 2 )
T ( jω ) → ω >>ω o
= n2
−ω 2 , a constant

For frequencies, ω << ω o

n2 ( − ω 2 )
T ( jω ) → ω <<ω o ω →0
 →0
ωo 2

Thus, the transfer function with n0 = n1 = 0 is a high pass filter with a break
frequency of ω o . Because of the ω 2 factor in the numerator, the gain in dB,
20 log 10 T ( jω ) , increases at an asymptotic rate of 40 dB/decade at frequencies
below ω o .

Any circuit that exhibits a transfer function of the form

Vout ( s ) n2 s 2
T ( s) = =
Vin ( s ) ω 
s 2 +  o  s +ωo 2
 Q

can act as a high pass filter. The equal component Sallen-Key circuit
has the transfer function

 RF  2
V ( s) 1+ s
T ( s ) = out =  R 
Vin ( s ) 1  RF  1
s2 + 2− s +
( RC )
2
RC  R 

Thus, this Sallen-Key circuit acts as a high pass filter with

1
ωo =
RC

1 R
=2− F
Q R

 R 
n2 = 1 + F 
 R 

One of the advantages of an equal component Sallen-Key circuit for realizing a

second order high pass filter is that ω o and Q can be adjusted independently by
adjusting C and RF , respectively. We’ll appreciate this advantage more, later.

3. Band Pass Filter: n0 = n2 = 0

Vout ( s ) n1s
T ( s) = =
Vin ( s ) ω 
s 2 +  o  s + ωo 2
 Q
It is straightforward to show that T ( jω ) exhibits a maximum at ω = ωo and, for
sufficiently large Q , the passband of the filter is quite narrow. An equal
component Sallen-Key band pass circuit does not work well in practice. Later,
we’ll consider a band pass filter circuit that contains more than one operational
amplifier, a requirement necessary to achieve narrow (say a few per cent of the
center frequency) passbands in practice.

4. Notch Filter: n1 = 0

Vout ( s ) s2 + ωn 2
T ( s) = = n2
Vin ( s ) ω 
s 2 +  o  s + ωo 2
 Q

where ω n is the notch frequency, a frequency corresponding to much reduced

output. Note that T ( s ) has zeros

z1, 2 = ± jω n

and T ( jω n ) = 0 so that the response, ideally, is zero at the notch frequency and
in a narrow band of frequencies about it. Far away from ω n , T ( jω ) is a
constant.

5. All Pass Filter:

ω 
s 2 −  o  s + ω o2
V ( s)  Q
T ( s ) = out = n2
Vin ( s ) ω 
s 2 +  o  s + ω o2
 Q

It is easy to see that T ( jω ) ≡1 . The all pass filter therefore affects only the
phase of the input, not its amplitude.

To view the poles and zeros of the various types of filters, as well as the
corresponding frequency responses, double click on the following:

Note especially the frequency response for a low pass filter. Regardless of your
choice of the coefficients, the frequency response, T ( jω ) differs considerably
from the ideal:
In the ideal low pass filter, note that the frequency response, T ( jω ) , is constant
up to the critical frequency, ω c , at which the response drops to zero. (The angle
of T ( jω ) , the phase shift, should be zero, ideally.) Regardless of how clever
your choice of ω o and Q , the ideal frequency response is impossible to realize in
practice. Indeed, the impulse response (the inverse Fourier transform of T ( jω ) )
that corresponds to the ideal frequency response is easily shown to be non-
causal. That is, the impulse response that corresponds to the ideal frequency
response for a low pass filter requires the filter to produce an output at times
before the impulse is applied.

To demonstrate this problem, we take a brief detour into Fourier transform

theory. Recall that a time function, x ( t ) , can be expressed in terms of its Fourier
transform, X ( ω ) , as a superposition of sinusoids (represented as e jωt ) as
follows:

1 +∞
x( t) = ∫ X ( ω ) e jωt dω
2π −∞

where X ( ω ) , typically a complex number, gives the amplitude and phase of the
component at each frequency, ω . Indeed, X ( ω ) is essentially the phasor
representation of the sinusoid with frequency, ω . The integral sums up all of the
frequency components. The constant 2π is merely cosmetic. What may seem
strange is that the integral sums over negative, as well as positive, frequencies,
ω . Recall, however, that if we write an ordinary sinusoid in terms of complex
exponential functions, that we write, from Euler’s identity,

1 jωt − jωt
cos ω t =
2
(e +e )
so that a sinusoid represented in terms of complex exponential functions requires
both positive and negative frequencies. The fact that X ( ω ) must exist for
negative as well as positive frequencies makes it slightly different from the usual
phasor representations of sinusoids. Nevertheless, the Fourier transform
representation and the phasor representation clearly are closely related.

Recall, also, the second part of the Fourier transform pair,

F { x ( t)} ≡ X ( ω ) = ∫
+∞
x ( t ) e− jω t dt
−∞

which permits the Fourier transform, X ( ω ) , to be calculated for a given time

function, x ( t ) . From this equation, we see that the complex conjugate of X ( ω ) ,
which we denote as X ( ω ) , is
*

(∫ )
*
=( X ( ω) ) = X* ( ω)
+∞ +∞
X ( −ω ) = ∫ x ( t ) e jω t dt = x ( t ) e− jω t dt
*

−∞ −∞

In specific cases in which X ( ω ) is real, this relation becomes

X ( −ω ) = X ( ω )

That is, if X ( ω ) is real, it is necessarily an even function of ω . Note, also,

specifically, that the Fourier transform of a unit impulse, δ ( t ) , applied at time
t = 0 is simply

F { δ ( t)} = ∫
+∞
δ ( t ) e− jω t dt =1
−∞

The fact that the Fourier transform of the impulse is a constant indicates that the
impulse contains equal components of all frequencies (with zero phase), from the
highest to the lowest. If an impulse were applied to the input of an ideal low pass
filter, therefore, the output would consist of constant components of frequencies
up to ω c and would have no components at higher frequencies. Because the
ideal low pass filter introduces no phase shift and the frequency components of
the impulse have zero phase angle, the spectrum, Vout ( ω ) , of the output voltage,
vout ( t ) , is real and hence must be an even function. Thus, Vout ( ω ) is a real
constant for frequencies between −ω c and ω c and zero outside this range. Thus,
the impulse response of the ideal low pass filter is:
 + ωc
1 +ωc K jω t K  e jωct − e − jωct  sin ω c t
vout ( t ) = ∫
jω t
Ke dω = e =  =
2π −ω c 2π jt −ω c
πt 2j  πt

Notice that the output of an ideal low pass filter in response to an impulse applied
to its input at time t = 0 begins (is non-zero) for negative times, before the
impulse is even applied. In effect, an ideal low pass filter therefore would need to
predict the future, ordinarily impossible, to respond as required.

The closest we can come to realizing the ideal low pass filter in practice is when
we do not require the output from the low pass filter to appear immediately, but
are willing to tolerate a long delay before the output is available. For example,
suppose we receive an analog data stream from a deep space probe and record
the entire stream before we begin the filtering operation. In such a case, we can
calculate the Fourier transform of the entire data stream and simply chop off all
frequencies greater than the critical frequency, ω c . In effect, the filter in this case
can know what happens in the future in that it can use later values of the signal in
calculating its output at a given time – the values at all times having been
recorded beforehand.

In the usual case in which we require immediate output from the low pass filter,
however, the ideal frequency response is simply impossible to realize. The ideal
frequency response for high pass and band pass filters is similarly impossible to
realize. The question then naturally arises as to what is the best approximation to
the ideal frequency responses that we can achieve in practice. The answer
varies depending on precisely what we mean by “best.” It turns out that the
response of all second order low pass filters at frequencies sufficiently below the
critical frequency, ω c , falls off at 40 dB per decade. Subject to that unavoidable
constraint, Butterworth filters give the flattest response at frequencies below ω c ,
Bessel filters exhibit the least overshoot in response to a step input, and
Chebyshev filters give the fastest fall-off for frequencies above ω c for a specified
“ripple” in the frequency response for frequencies in the pass band, that is, for
frequencies less than ω c .

To realize a transfer function,

Vout ( s ) n0
T ( s) = =
Vin ( s ) ω 
s 2 +  o  s +ωo 2
 Q

for any one of the low pass filters with critical frequency, ω c , choose ω o and Q
according to the following table:
 Filter Type ωo 1
ωc Q
Butterworth 1.000 2
Chebyshev (1 dB ripple) 0.863 1.045
Chebyshev (2 dB ripple) 0.852 0.895
Chebyshev (3 dB ripple) 0.841 0.767
Bessel 1.274 1.732

The following Excel spreadsheet gives comparative plots of these second order
low pass filters:

L P F IL T E R . X L S

Here is one of the graphs from the spreadsheet:

Notice that the Chebyshev filter with 3 dB ripple in the pass band offers the
largest attenuation above ω c , as well as the largest hump in response near ω c .
The Bessel filter achieves the least attenuation above ω c . Notice that the
response of all of these filters falls off at the same rate – 40 dB per decade – at
frequencies higher that a few ω c . This slope is determined, of course, by the
degree of the denominator – 20 dB for each power of s in the denominator.
A happy note is that the “best” choices for ω o and Q turn out to be the identical
for both high pass and low pass filters.

A steeper fall off outside the pass band can be achieved by increasing the
degree of the denominator in the transfer function, T ( s ) . A simple way of
effectively increasing the degree of the denominator is to cascade two second
order filters. Cascading two filters effectively multiplies the transfer functions,
thus multiplying their denominators and thereby increasing the order. Although it
may not be obvious, cascading two second order Butterworth filters does not give
a fourth order Butterworth filter (flattest passband). To achieve a fourth order
Butterworth (or other) filter by cascading two second order filters, ω o and Q for
each filter should be chosen differently, according to the following table:

First Section Second Section

Type of Filter ω 01 1 ω 02 1
ωc Q1 ωc Q2
Butterworth 1.000 1.848 1.000 0.765
Chebyshev (1 dB) 0.502 1.275 0.943 0.281
Chebyshev (2 dB) 0.466 1.088 0.946 0.224
Chebyshev (3 dB) 0.443 0.929 0.950 0.179
Bessel 1.436 1.916 1.610 1.241

State Variable Second Order Filter System

Consider the following circuit (originally used for solving second order differential
equations during the heyday of analog computers) which turns out to be a
surprisingly versatile filter circuit:
For simplicity, we consider the analysis of this circuit in sections. First consider
the summing operational amplifier, the one on the left:

First, we write node equations at the two inputs of the operational amplifier and
write the relationship between the amplifier output and its inputs:

v− ( t ) − vLP ( t ) v − ( t ) − vin ( t ) v − ( t ) − v HP ( t )
( 1) + + =0
R R R
v+ ( t ) − vBP ( t ) v + ( t )
( 2) + =0
RF R
( 3) vout ( t ) = A  v+ ( t ) − v− ( t ) 

From equations (1) and (2), we find

( 1) ' 3v− ( t ) −  vLP ( t ) + vin ( t ) + v HP ( t )  = 0

 R 
( 2) ' v+ ( t )  1 + F  − vBP ( t ) = 0
 R 

 R 
If we multiply equation (1)' by 1 + F  and equation (2)' by 3, we find:
 R

 R   RF 
( 1) " 3 1 + F  v− ( t ) − 1 +  v LP ( t ) + v in ( t ) + v HP (t )  = 0
 R   R 
 R 
( 2) " 3v+ ( t ) 1 + F  − 3v BP ( t ) = 0
 R 

If we subtract equation (1)" from equation (2)", we find:

 R   R 
3 1 + F   v+ ( t ) − v− ( t )  + 3v BP ( t ) −  1+ F   vLP ( t )+ vin ( t )+ vHP ( t ) = 0
 R  R
The combination of equation (3) with this result gives:

 R  v ( t)  R 
3  1 + F  HP + 3vBP ( t ) − 1 + F  v LP ( t ) +v in( t) +v HP( t)  =0
 R  A  R 

If A→∞ (ideal operational amplifier), we see that

3
vLP ( t ) + vin ( t ) + vHP ( t ) = v (t )
RF BP
1+
R

or

3
vHP ( t ) = v ( t ) − vLP ( t ) − vin ( t )
RF BP
1+
R

In terms of Laplace transforms, we have, neglecting initial conditions,

3
( A) VHP ( s ) = V ( s ) − VLP ( s ) − Vin ( s )
R BP
1+ F
R

The remaining two operational amplifiers form two integrators of the type we
considered earlier:

Recall our earlier result for this circuit:

1 t
vout ( t ) = vout ( 0 ) − ∫ v ( τ ) dτ
( R + R1 ) C 0 in
In the present case, each of the integrators is driven by the output of an
operational amplifier, for which the Thevenin output impedance is very low. Thus,
R + R1 becomes, in the present case, simply R . If we neglect initial conditions, as
well, then we have

1 t
vout ( t ) = − vin ( τ ) d τ
R C ∫0

In the Laplace transform domain,

1
Vout ( s ) = − Vin ( s )
R Cs

For our particular circuit, therefore, we can write:

1
( B) VLP ( s ) = − V BP ( s )
R Cs

1
(C ) VBP ( s ) = − V HP ( s )
R Cs

or

( B) ' VBP ( s) = − R CsVLP ( s)

(C ) ' VHP ( s ) = − R CsVBP ( s)

From equations (B)' and (C)', we obtain:

( D) VHP ( s) = R2 C2 s 2 VLP ( s)

Thus, equation (A) becomes

3
R 2 C 2 s 2 VLP ( s ) = − RCsVLP ( s )  − VLP ( s ) − Vin ( s )
RF 
1+
R

 
 2 2 2 3RCs 
VLP ( s )  R C s + + 1  = −V in( s )
R
 1+ F 
 R 
 1
VLP ( s ) = − V in ( s )
 
 3 1 
R2 C 2 s 2 + s + 
 RF  ( RC )
2
 
 1+  RC 
 R 

1
( RC )
2

VLP ( s ) = − V in( s)
3 1
s +
2
s +
 RF  ( RC )
2

1 +  RC
 R 

The LP output therefore clearly corresponds to a low pass filter with

1
ωo =
RC

1 R 
Q =  1+ F 
3 R

Note that at low frequencies, the voltage gain (the modulus of the transfer
function) is unity.

Next, we calculate the output VBP ( s ) . From equation (B)', note that

1
( RC )
2

VBP ( s ) = − R CsV LP ( s ) = − ( − R Cs) V in( s)

3 1
s2+ s +
 RF  ( RC )
2

1 +  RC
 R 

1
s
VBP ( s ) = RC Vin ( s )
3 1
s2+ s +
 RF  ( RC )
2

1 +  RC
 R 

The BP output therefore clearly corresponds to a band pass filter with

1
ωo =
RC
 1 R 
Q =  1+ F 
3 R 

Next, we calculate the output VHP ( s ) . From equation (D), note that

1
( RC )
2

VHP ( s ) = R C s V LP ( s ) = − ( R C s
2 2 2 2 2 2
) V in( s)
3 1
s2+ s +
 RF  ( RC )
2

1 +  RC
 R 

s2
VHP ( s ) = − Vin ( s )
3 1
s2+ s +
 RF  ( RC )
2

1 +  RC
 R 

1
ωo =
RC

1 R 
Q =  1+ F 
3 R

Note that at high frequencies, the voltage gain (the modulus of the transfer
function) is unity.

We see, therefore, that this one configuration gives high pass, low pass
and band pass filter outputs, simultaneously. Note, also, that by taking
appropriate weighted sums of these outputs, we can synthesize any
second order filter, including notch and all pass filters, as well. This
configuration is available in an IC that uses switched capacitor
technology to make the extremely small capacitors possible on ICs
behave like much larger capacitors.