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Motivation:
Analyse filters
Design low frequency filters without large capacitors
Design filters without inductors
Design electronically programmable filters
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Low Pass
High Pass
Band Pass
Band Reject
Frequency domain
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Time domain
E2.2 Analogue Electronics
Filter types
Low pass
High pass
Band pass
Band Reject
Observe that a real filter is not sharp, and its transmission is not constant!
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c1
APF
c2
Delay elements
APF
c3
Coefficients
c4
Output
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+
+
+
All pass filter based analogue filters are similar to the digital filters
encountered in Digital Signal Processing
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as
P (s)
H (s) =
=
Q (s)
b s
n
k =0
m
k =0
=
k
an ( s z1 )( s z2 )" ( s zn )
bm ( s p1 )( s p2 )" ( s pm )
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Autumn 2009
Families of filters
H (s) H * (s) =
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1 + 2 Pn2 ( s )
Pole locations determine filter response. The closer poles are to the
imaginary axis the steepest the transition from passband to stopband.
a: Butterworth: poles on a circle
b: Chebyshev: Poles on an ellipse (sharper)
c: Elliptic: Like Chebyshev, plus zeroes on the imaginary axis (sharpest)
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Autumn 2009
Rs
Z(s)
Vout
Vin
GL
v
1
1
H ( s ) = out =
Z (s) =
vin 1 + ( Z ( s ) + Rs ) GL
GL
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(1 + Rs GL )
H (s)
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Cauer forms
First Cauer form
L1
L2
L3
C1
Ln
C2
C1
C2
L2
L1
Cn
(a)
C3
Cn
Ln
(b)
Z in = sL1 +
1
sC1 +
1
1
sL2 +
"
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Autumn 2009
s3 + 2s
s
1
= s+ 2
= s+
2
1
s +1
s +1
s+
s
H ( s ) = H0
C ( s / 0 ) + 2 B s / 0 + A
2
( s / 0 ) + 2 s / 0 + 1
2
1
, Q=
2
Function
Low Pass
High Pass
Band Pass
Band Stop
All Pass
-1
2nd order filters are useful: we can always decompose higher order filters to a
cascade of 2nd order filters!
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1 d 2 2 d
C d2
2 d
+ 1 y ( t ) = H 0 2 2 + B
+ A x (t )
2 2+
n dt
n dt
n dt
n dt
Substitute:
x = X ( ) e jt =X ( s ) e st , y = Y ( ) e jt = Y ( s ) e st
To get:
Y (s)
C ( s / n ) + 2 B s / n + A
H (s) =
= H0
2
X (s)
( s / n ) + 2 s / n + 1
2
This is the transfer function of a 2nd order filter. It follows that the filter solves the ODE.
The impulse responses (IR) of lowpass, bandpass and highpass filters are related*:
The IR of the BP is proportional to the time derivative of the IR of the LP
The IR of the HP is proportional to the time derivative of the IR of the BP
It follows that a loop of 2 integrators can implement any 2nd order filter. Such a loop is
called a biquad.
* (remember that H(s) is the Laplace transform of the impulse response)
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H0
q 2 + 2 q + 1
H 0q2
= H HP ( q )
H LP (1/ q ) =
2
1 + 2 q + q
If the components of a filter are replaced so that any impedance
dependence on is replaced by a similar dependence on 1/ the filter
changes from low pass to high pass
In practice we replace C with L and L with C so that:
n C =
1
n L
The same transformation generates a low pass filter from a high pass filter.
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H0
a+q
H1 ( q + 1/ q ) =
H0
H q
= 2 0
= H 2 BP ( q )
a + ( q + 1/ q ) q + aq + 1
In practice we replace the low pass elements, following the following recipe:
all capacitors with parallel LC circuits, (open at resonance) and
all inductors with series LC circuits (short at resonance)
1
n C =
n L
n is the centre frequency of the filter. The BPF has the same BW as the LPF
f = 4n = 2n = f B , LPF
To get a band reject filter replace in the low pass prototype:
Cseries LC
L parallel LC
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C = C / Z 0 , L = Z 0 L
To scale a low pass so that its break frequency is the required f0 Hz:
C f 0 = C , Lf 0 = L
After these transformations we can use the transformations from low pass
to the required filter function as described before
Note:
it is unusual to treat signal sources as pure voltage or current sources in
professional engineering applications. (This would make circuits too noisy!)
In professional audio the standard impedance used is 600 Ohms.
In cable, video and television applications the standard is 75 ohms
In most other radio frequency applications the standard is 50 ohms.
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Vin
Vout
Vin
(a)
ideal integrator
Vout
Lossy integrator
(b)
1
Av =
RC
R1
1
Av =
R 1 + j R1C
Note: The ideal integrator is unstable at DC, and can only be used inside a
feedback loop
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R1
Vout
Vin
Vout
Vin
(a)
Ideal differentiator
(b)
Lossy differentiator
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C1
Vin
Vout
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R2
Vout
Vin
C1
C2
1
1
H (s) = 2
= 2
s R1C1 R2C2 + s ( R1C1 + R2C2 + R1C2 ) + 1 s 1 2 + s ( 1 + 2 + 12 ) + 1
0 = 1/ 1 2
1
2
12
1
1
, 2 = =
+
+
>2Q<
Q 2
2
1
21
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Note: Although we show these filters made with op-amps, they can be
made with ANY amplifying device, e.g. with bipolar transistors or
FETs.
The actual device we use will have input and output impedance which
we need to account for in the filter element value calculation.
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R2
Vout
K
Vin
C1
C2
K
H
A=
1
s 2 R1C1 R2C2 + s ( R1C1 + R1C2 + R2C2 ) + 1
sR1C1
B= 2
s R1C1 R2C2 + s ( R1C1 + R1C2 + R2C2 ) + 1
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K
H
AK
H=
1 BK
A and B are both rational functions, with the same denominator:
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A=
sR C
1
,B= 1 1
Q (s)
Q (s)
H=
K
K
= 2
Q KR1C1 s R1C1 R2C2 + s ( (1 K ) R1C1 + R1C2 + R2C2 ) + 1
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R2
B
Vout
K
Vin
C1
C2
K
H
H=
1
n2
H0
K
=
s 2 / n2 + 2 s / n + 1 s 2 R1C1 R2C2 + s ( (1 K ) R1C1 + R1C2 + R2C2 ) + 1
= R1C1 R2C2 n =
1
R1C1 R2C2
H0 = K
2
R1C1
R2C2
R1C2
1
1
= (1 K ) R1C1 + R1C2 + R2C2 2 = = (1 K )
+
+
Qn
Q
R2C2
R1C1
R2C1
For large enough K the circuit will have Q<0 and will become dynamically
unstable, i.e. it will become an oscillator
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C2
Vout
R1
Vin
K
R2
K
H
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C2
Vout
K
Vin
C1
R2
This has identical in form passive band pass filters in the forward
and feedback paths, shown on the middle. The block diagram in the
right is the same form as the other SK filters. If R1=R3 then the two
filters are identical and A=B . The transfer function of each path filter is:
A= B=
s 2
, 1 = R1C1 , 2 = R2C2 , 12 = R1C2
2
s 1 2 + ( 2 2 + 1 + 12 ) s + 2
H=
Ks 2 / 2
AH
= 2
1 AH s 1 2 / 2 + ( ( 2 K ) 2 + 12 + 1 ) s / 2 + 1
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2C
R
Vout
Vin
R/2
B
A
K
H
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Band Pass
All Pass
Op-amp is ideal
Inverting input is virtual GND, V=0, i=0
Nodal analysis usually simple
Tee-Pi transforms may simplify algebra
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state variable filters treat both the signal and its derivatives as variables
A low pass filter performs time integration on signal waveforms
A high pass filter performs time differentiation on signal waveforms
Recall that filters are analogue computers which solve ODEs
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C
z
= RC , K1 = K 2 =
2 R1 / / R2
, K 3 = 1
R2 + R1 / / R2
x = K1vi + K 2 y K 3 z , x = y = 2
z
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a x
P (s)
H (s) =
=
Q (s)
b x
n
i =0
m
i =0
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s z0 )( s z1 )" ( s zn )
(
=
( s p0 )( s p1 )" ( s pn )
P(s) and Q(s) have real coefficients. To make a higher order filter:
factor Q(s) into quadratic and linear factors
Implement factors as biquads
Cascade biquad sections to obtain the original transfer function
Note that P and Q have real coefficients, so that their roots are either
real or come in conjugate pairs.
The centre frequencies and damping factors of the sections required to
implement standard forms (Butterworth, Chebyshev, Elliptic etc) are
tabulated in reference books.
Tables are also included in CAD software for automated filter synthesis
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Alternate symbol
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Passive Gyrators
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A grounded component
between two gyrators appears
gyrated and in series
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K = 1/ j
which
Transforms R C
Transforms L R
C FDNR
FDNR is a fictitious circuit element with:
Y = D2
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Note that we can scale the filter coefficients by any factor of our
choice, including j. All we need is that the voltage divider works as
intended at all frequencies!
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S1
S2
C
V1
R
V2
V2
V1
(a)
(b)
(a) And (b) circuits are equivalent as long as signal frequency is much
smaller than switching frequency
The SC equivalent resistance is proportional to frequency
S
S1
S
Vout=-V
C
S1
Vout=2V
(a)
S1
S1
C
(b)
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Summing junction
Inverting
Lossy integrator
Inverter
Inverting
Ideal integrator
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Summary
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