Filter Design I

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Filter Design I
Sverre Holm
INF4480
Digital signalbehandling II
DEPARTMENT OF INFORMATICS
UNIVERSITY
OF OSLO
EKG signal i sty EKG signal i sty
Electrocardiogram (ECG
or EKG, abbreviated
f h G from the German
elektrokardiogramm)
m:\matlab\Aase- m:\matlab\Aase
EKG\Aase.m
Data fra Ottar Aase,
som den 8 oktober 1999 som den 8. oktober 1999
forsvarte sin avhandling
COMPUTER AIDED
DIAGNOSTICS OF
ACUTE CHEST PAIN ACUTE CHEST PAIN
Ullevl Universitets-
sykehus, Oslo
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Filterdesign Filterdesign
1. Spesifikasjon
Kj d l Kjenne anvendelsen
Kjenne designmetoder (hva som er mulig, FIR/IIR)
2. Approksimasjon pp j
Fokus her
3. Analyse
Filt l ifi t i f k d t Filtre er som regel spesifisert i frekvensdomenet
Ogs analysere i tid (fase, forsinkelse, ...)
4. Realiseringg
DSP, FPGA, PC: Matlab, C, Java ...
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Oppgave Oppgave
Designe FIR-filtre
Frekvenssampling (siste prioritet)
Vindusmetoden
Ekviripple Ekviripple
IIR-filtre
Transformerte standardfiltre: Butterworth, Tsjebysjeff,
elliptisk
Sammenligning
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Sources Sources Sources Sources
The slides about Digital Filter Specifications
h b d t d f lid b S Mit have been adapted from slides by S. Mitra,
2001
Butterworth Chebychev etc filters are based Butterworth, Chebychev, etc filters are based
on Wikipedia
Builds on Oppenheim & Schafer with Buck: Builds on Oppenheim & Schafer with Buck:
Discrete-Time Signal Processing, 1999.
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Ideal filters Ideal filters Ideal filters Ideal filters
Lavpass, hypass, bndpass, bndstopp
H
LP
(e
j
e
) H
HP
(e
j
e
)
1 1
t
e
0

ec

ec
t
t
e
0

ec

ec
t
H
BP
(e
j
e
)
H
BS
(e
j
e
)
1
1
1
6
t t
e

ec1

ec1

ec2

ec2 t t
e

ec1

ec1

ec2

ec2
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Prototype low-pass filter Prototype low pass filter
All filter design methods are specificed for
l l low-pass only
It can be transformed into a high-pass filter
O i b l d i i i h h Or it can be placed in series with others to
form band-pass and band-stop filters
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Digital Filter Specifications Digital Filter Specifications Digital Filter Specifications Digital Filter Specifications
As the impulse response corresponding to
h f th id l filt i l d f each of these ideal filters is noncausal and of
infinite length, these filters are not realizable
In practice the magnitude response In practice, the magnitude response
specifications of a digital filter in the passband
and in the stopband are given with some
acceptable tolerances
In addition, a transition band is specified
between the passband and stopband between the passband and stopband
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The magnitude response |G(e
je
)| of a digital
l filt b ifi d lowpass filter may be specified as:
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Passband: 0 e e
p
We require that |G(e
je
)| 1 with an error o
p
, i.e.,
p p
j
p
e G e e o o
e
s + s s , 1 ) ( 1
Stopband: e
s
e t
W i th t |G(
je
)| 0 ith o i
p p p
, ) (
We require that |G(e
je
)| 0 with an error o
s
, i.e.,
t e e o
e
s s s
j
e G , ) ( t e e o s s s
s s
e G , ) (
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e
p
- passband edge frequency
e
s
- stopband edge frequency
o
p
- peak ripple value in the passband
p
o
s
- peak ripple value in the stopband
Properties:
G(e
je
) is a periodic function of e
|G(e
je
)| of a real-coefficient digital filter is an even
function of e
Consequence: Filter specifications are given
only for 0 |e| t
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Specifications are often given in terms of loss
f ti function:
G(e)=-20 log
10
|G(e
je)
| in dB
Peak passband ripple
o
p
= -20log
10
(1-o
p
) dB o
p
20log
10
(1 o
p
) dB
Minimum stopband attenuation Minimum stopband attenuation
o
p
= -20log
10
(o
s
) dB
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Magnitude specifications may alternately be
i i li d f i di t d b l given in a normalized form as indicated below
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Normalized frequencies Normalized frequencies
Real values
Real frequencies: f
p
, f
s
, f
sample
Angular frequencies: e=2tf
Normalized values Normalized values
Angular frequencies: e=0...2t where f
sample
2t
Normalized frequencies f=0 ... 2 where 2 is the sampling
frequency (0...1 is the useful range): MATLAB filter
design
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FIR and IIR Digital Filter FIR and IIR Digital Filter FIR and IIR Digital Filter FIR and IIR Digital Filter
Difference equation
Transfer function Transfer function
General: IIR - Infinite Impulse Response
FIR Fi it I l R FIR - Finite Impulse Response
N=0, no feedback, always stable
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Pol- og nullpunktsplassering Pol og nullpunktsplassering
Poler innenfor enhetssirkel Stabil og kausal
Poler og nullpunkter i kompleks konjugerte par Reell Poler og nullpunkter i kompleks konjugerte par Reell
impulsrespons
Alle nullpunkter finnes speilet om enhetssirkelen Liner
fase.
Viktig! Liner fase og reelle koeffisienter Nullpunkter
finnes i grupper av fire.
Alle nullpunkter er speilbildet av en pol Allpass system
Alle nullpunkter innenfor enhetssirkel Minimum fase
system og inversfilter eksisterer
Alle nullpunkter utenfor enhetssirkel Kausal maksimum
fase
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Standard Analog Filter functions (magnitude)
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Standard Analog Filter Functions Standard Analog Filter Functions
Butterworth filter
no gain ripple in pass band and stop band slow cutoff no gain ripple in pass band and stop band, slow cutoff
Chebyshev filter (Type I)
no gain ripple in stop band, moderate cutoff
Chebyshev filter (Type II) y ( yp )
no gain ripple in pass band, moderate cutoff
Elliptic filter
gain ripple in pass and stop band, fast cutoff
Bessel filter
no group delay ripple, no gain ripple in both bands, slow gain cutoff
Linkwitz-Riley filter
Used for crossover filters for loudspeakers
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Laplace vs z-transform Laplace vs z transform
Laplace, s=o+jO:
Z-transform, z=e
je
:
Avbildning mellom s- og z-plan
ved biliner transform
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Filter Slopes Filter Slopes
Decade: 10 x frequency, e.g. 100 Hz 1 kHz
i d d is one decade
Octave: 2 x frequency (octave=8 for the white
piano keys) e g 100 Hz > 200 Hz piano keys), e.g. 100 Hz -> 200 Hz
1. order filter: rolls off at 6 dB per octave (20 dB per
decade)
2. order filter: the response decreases at 12 dB per
octave (-40 dB per decade)
3. order at 18 dB, and so on.
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Group Delay Group Delay
The group delay is the derivative of the phase
ith t t l f with respect to angular frequency
It is a measure of the distortion in the signal
introduced by phase differences for different introduced by phase differences for different
frequencies.
t
g
= -d[arg(H(e
je
))] / de
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Butterworth Filter Butterworth Filter
Maximally flat (has no ripples) in the passband, and
rolls off towards zero in the stopband rolls off towards zero in the stopband.
When viewed on a logarithmic plot, the response
slopes off linearly towards negative infinity.
Butterworth filters have a monotonically changing
magnitude function with .
First described by British engineer Stephen First described by British engineer Stephen
Butterworth in "On the Theory of Filter Amplifiers",
Wireless Engineer, vol. 7, 1930, pp. 536-541.
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3 Order Butterworth with e =1 3. Order Butterworth with e
p
1
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Plot of the gain of Butterworth low-pass filters of orders 1 through 5.
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Note that the slope is 20n dB/decade where n is the filter order.
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Analog Butterworth Analog Butterworth
3. order passive low pass
filter (Cauer topology). ( p gy)
2. order active filter
(Sallen-Key topology)
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Butterworth Butterworth
Frequency response:
Transfer function
Butterworth polynominals (n even, n odd)
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Butterworth Butterworth
n Factors of Polynomial B
n
(s)
1 (s + 1)
2 s
2
+ 1 4142s + 1 2 s + 1.4142s + 1
3 (s + 1)(s
2
+ s + 1)
4 (s
2
+ 0.7654s + 1)(s
2
+ 1.8478s + 1)
5 (s + 1)(s
2
+ 0.6180s + 1)(s
2
+ 1.6180s + 1) ( )( )( )
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Butterworth Filter Butterworth Filter
Nth order filter: all derivatives of the gain up
t d i l di th 2N 1th d i ti to and including the 2N-1th derivative are
zero at e=0, resulting in "maximal flatness".
In decibels the high frequency roll off is 20n In decibels, the high-frequency roll-off is 20n
dB/decade, or 6n dB/octave (not only
Butterworth all filters)
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Chebyshev filter Chebyshev filter
Norsk: Tsjebysjeff
Steeper roll-off and more passband ripple (type I) or
stopband ripple (type II) than Butterworth filters.
Minimize the error between the idealized filter
characteristic and the actual, but with ripples in the
passband.
Named after Pafnuty Chebyshev (1821-1894) Named after Pafnuty Chebyshev (1821 1894)
, because they are

defined in terms of Chebyshev polynomials.
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Chebyshev type I and type II
The frequency response of a
fourth order type I Chebyshev fourth-order type I Chebyshev
low-pass filter with = 1
The frequency response of a
fifth-order type II Chebyshev
l filt ith 0 01 low-pass filter with = 0.01
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5 Order Chebyshev type I (=0 5) 5. Order Chebyshev type I ( 0.5)
There are
ripples in the pp
gain and the
group delay in
the passband
but not in the
stop band.
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5 Order Chebyshev type II (=0 1) 5. Order Chebyshev type II ( 0.1)
There are
ripples in the
gain in the stop
band but not in
th b d the pass band.
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Chebyshev Chebyshev
Frequency response (type I, type II)
is the ripple factor,
0
is the cutoff frequency
and T
n
() is a nth order Chebyshev polynomial.
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Elliptic Filters Elliptic Filters
An elliptic filter (also known as a Cauer filter)
h li d i l ( i i l ) b h i i has equalized ripple (equiripple) behavior in
both the passband and the stopband.
The amount of ripple in each band is The amount of ripple in each band is
independently adjustable.
No other filter of equal order can have a No other filter of equal order can have a
faster transition between the passband and
the stopband, for the given values of ripple
(whether the ripple is equalized or not) (whether the ripple is equalized or not).
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Cauer Cauer
Wilhelm Cauer (June 24, 1900 April 22, 1945) was a
German mathematician and scientist German mathematician and scientist.
He is most noted for his work on the analysis and
synthesis of electronic filters and his work marked the
b i i f th fi ld f t k th i beginning of the field of network synthesis.
Prior to his work, electronic filter design was an art,
requiring specialized knowledge and intuition. Cauer
placed the field on a firm mathematical footing,
providing a theoretical basis for the rational design of
electronic filters.
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4 Order Elliptic Filter 4. Order Elliptic Filter
The frequency
response of a fourth- response of a fourth
order elliptic low-
pass filter with =0.5
and =1.05.
Also shown are the
minimum gain in the
passband and the
maximum gain in the maximum gain in the
stopband, and the
transition region
between normalized
frequency 1 and
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4 Order Elliptic Filter 4. Order Elliptic Filter
A closeup of the
transition region transition region
of the previous
plot
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Elliptic filter Elliptic filter
Frequency response:
where R
n
() is the nth-order Jacobian elliptic
rational function and
0
is the cutoff
frequency, is the ripple factor, is the
selectivity factor selectivity factor
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Bessel Filter Bessel Filter
A Bessel filter is a filter with a maximally flat
group delay ( linear phase response) group delay ( linear phase response).
Analog Bessel filters are characterized by
almost constant group delay across the entire g p y
passband.
The filter that best preserves the wave shape
of filtered signals in the passband of filtered signals in the passband.
Named after Friedrich Bessel (17841846) as
the filter polynomial is expressed with Bessel
ffunctions
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4 Order Bessel Filter 4. Order Bessel Filter
A plot of the gain and
group delay for a fourth- group delay for a fourth
order low pass Bessel filter.
Note that the transition
from the pass band to the from the pass band to the
stop band is much slower
than for other filters, but the
group delay is practically
t t i th b d constant in the passband.
The Bessel filter maximizes
the flatness of the group
delay curve at zero delay curve at zero
frequency.
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Bessel filter Bessel filter
Transfer function
where
n
(s) is a reverse Bessel polynomial,
0
is the cut-off frequency
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Comparison Comparison
Butterworth: maximally flat amplitude response
Bessel: maximally flat group delay Bessel: maximally flat group delay
Compared with a Chebyshev Type I/Type II filter or an
elliptic filter, the Butterworth filter has a slower roll-off,
and thus will require a higher order to implement a
ti l t b d ifi ti particular stopband specification.
However, Butterworth filter will have a more linear
phase response in the passband than the Chebyshev
Type I/Type II and elliptic filters Type I/Type II and elliptic filters.
Chebyshev filters are sharper than the Butterworth
filter; they are not as sharp as the elliptic one, but they
show fewer ripples over the bandwidth.
Elliptic filters are sharper than all other filters, but they
show ripples on the whole bandwidth.
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2-way loudspeakers y
Goal: Control of spatial response at
crossover frequency:
1 G i i i 1. Gain sums to unity in crossover
region
2-element array:
How to ensure steering straight g g
ahead in crossover region?
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2-way loudspeaker
h t common phasecenter
Solutions with common phase center:
Coaxial
D'Appolito configuration: Tweeter positioned
between two low frequency drivers between two low frequency drivers.
Both are rather uncommon systems
48
Eltek AS, www.dynabel.no
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Avoiding beaming in loudspeakers
h h t t when phase centers are apart
Goal: absence of sudden
change in directivity with g y
frequency.
Crossover filter must:
1. Sum to unity gain
2 E h f th 2. Ensure same phase for the
two drivers in crossover
region
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Linkwitz-Riley Filter Linkwitz Riley Filter
A Linkwitz-Riley (L-R) filter is used in audio crossovers,
named after Siegfried Linkwitz and Russ Riley named after Siegfried Linkwitz and Russ Riley
S. Linkwitz, Activee Crossover Networks for Noncoincident
Drivers in Journ. Audio Eng. Soc., 24 (1), pp. 2-8; Jan 1976.
L R crossover = a low pass and a high pass L R filter L-R crossover = a low-pass and a high-pass L-R filter.
L-R = cascade of two Butterworth filters, each with 3
dB gain at the cut-off frequency (=Butterworth squared
filt ) filter).
The L-R filter has a 6 dB gain at the cutoff frequency.
The sum of the low-pass and high-pass outputs will be The sum of the low pass and high pass outputs will be
0 dB, so the crossover behaves like an all-pass filter:
flat amplitude response and a smoothly changing
phase response.
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phase response.
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Linkwitz-Riley filters orders 2 and 4 Linkwitz Riley filters, orders 2 and 4
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Butterworth cross-over Butterworth cross over
52
Dennis A. Bohn, Linkwitz-Riley Crossovers: A Primer
http://www.rane.com/note160.html
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Linkwitz-Riley crossover Linkwitz Riley crossover
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Linkwitz-Riley 2 order
Key: Parallel phase functions
for the two branches
Linkwitz Riley 2. order
Amplitude Phase Group delay
4th-order Linkwitz-Riley (LR-4) filters is the de facto
standard for professional audio active crossovers.
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Q value Q value
Quality factor higher Q for
sharper resonance
H
BP
(e
j
e
)
sharper resonance
Q = 2total energy stored /
energy lost in a single cycle t t
e
1
1
ec1
ec1
ec2
ec2
gy g y
A radio receiver with a high Q
would be more difficult to tune with
the necessary precision but would
H
LP
(e
j
e
)

the necessary precision, but would
do a better job of filtering out
signals from other stations that lay
nearby on the spectrum
1
e
nearby on the spectrum. t
e
0

ec
ec t
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Q value for filters Q value for filters
A graph of a filter's gain
magnitude illustrating the magnitude, illustrating the
concept of -3 dB at a gain of
0.707 or half-power
bandwidth (Af) bandwidth (Af).
Q = f
0
/Af
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2 Order High Pass Filter 2. Order High Pass Filter
f
0
=100 Hz
Critically damped y p
Q=1/2, -6 dB at f
0
Linkwitz-Riley (Butterworth
2
)
Best pulse response
Q=1/3
1/2
0.577
B l filt Bessel filter
Flattest passband frequency
response,
Q=1/2
1/2
0.707, -6 dB at f
0
Butterworth filter Butterworth filter
Closed box loudspeaker
Tuning of box volume vs. element
stiffness (surround, spider) and
mass
Q 1 2 more apparent bass Q 1.2 more apparent bass
Q 1: warm sound, saleable
Q 0.8 more detailed,
somewhat shallow
Q 0.5 overdamped
V Dickason: Loudspeaker Cookbok
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V. Dickason: Loudspeaker Cookbok
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2 Order High Pass Filter: phase 2. Order High Pass Filter: phase
Critically damped
Q 1/2 6 dB t f Q=1/2, -6 dB at f
0
Best pulse response
Q=1/3
1/2
0.577
Bessel filter
Phase is closest to a linear
function in passband
Flattest passband
frequency response,
Q=1/2
1/2
0.707, -6 dB at f
0
Butterworth filter
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IIR filters IIR filters
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Transform analog prototype to
di it l d i digital domain
s-plane to z-domain: s=o+jO z=e
je
Frequency axis: s=jO maps to |z|=1
Stability, causality maintained
O=0 e=0 always: LP LP
60
http://en.wikibooks.org/wiki/Digital_Signal_Processing/Bilinear_Transform
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Transform from s-plane to z-plane Transform from s plane to z plane
1 Impulse invariance 1. Impulse invariance
Impulse response is a sampled version of the analog one
Aliasing as O=O
s
e=t
2. Bilinear transform
Let O=be mapped to e=t
N li t f t f H( ) t H( ) Nonlinear transform to go from H(s) to H(z)
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Bilinear transform Bilinear transform
No aliasing, maintains stability and causality,
but warps frequency axis but warps frequency axis
Best for lowpass, as passband is little
distorted
For other filters, start with analog LP =>
digital LP => digital HP etc
http://wwwborescom/courses/intro/iir/5 warphtm
62
http://www.bores.com/courses/intro/iir/5_warp.htm
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Matlab IIR Digital Filter Design Matlab IIR Digital Filter Design Matlab IIR Digital Filter Design Matlab IIR Digital Filter Design
Order Estimation
For IIR filter design using bilinear transformation,
MATLAB statements to determine the order and
bandedge are:
[N, Wn] = buttord(Wp, Ws, Rp, Rs);
[N, Wn] = cheb1ord(Wp, Ws, Rp, Rs);
[N W ] h b2 d(W W R R ) [N, Wn] = cheb2ord(Wp, Ws, Rp, Rs);
[N, Wn] = ellipord(Wp, Ws, Rp, Rs);
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Filter Design
For IIR filter design using bilinear transformation,
MATLAB statements to use are:
[b, a] = butter(N, Wn) , ,
[b, a] = cheby1(N, Rp, Wn)
[b, a] = cheby2(N, Rs, Wn)
[b, a] = ellip(N, Rp, Rs, Wn)
No need to think about bilinear transform
Transfer function can be computed using freqz(b a Transfer function can be computed using freqz(b, a,
w) where w is a set of angular frequencies
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Design an elliptic IIR lowpass filter with
F 0 8 kH F 1 kH F 4 kH 0 5 F
p
=0.8 kHz, F
s
=1 kHz, F
sample
=4 kHz, o
p
= 0.5
dB, o
s
=40 dB
Code fragments used are: Code fragments used are:
[N,Wn] = ellipord(0.8/(4/2), 1/(4/2), 0.5, 40);
Result: N=5 order Result: N=5. order
[b, a] = ellip(N, 0.5, 40, Wn);
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0
Elliptic IIR Lowpass Filter
-01
0
Passband Details
-20
G
a
i
n
,

d
B
-0.3
-0.2
-0.1
G
a
i
n
,

d
B
0 0.2 0.4 0.6 0.8 1
-60
-40
G
0 0.1 0.2 0.3 0.4
-0.5
-0.4
G
0 0.2 0.4 0.6 0.8 1
e/t
0 0.1 0.2 0.3 0.4
e/t
66

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