FILTER DESIGN GUIDE

Frequency Devices, Inc.

August 2003
Source: http://frequencydevices.com/guide/fullguide.html, Oct. 14, 2004
Edited by William Rose, 2011

DIGITIZING SIGNALS AND ALIASING

Analog to Digital Conversion (A/D)

Most physical (real world) signals are analog. Operating on these signals efficiently often requires the
filtering, sampling and digitizing of the analog data using A/D converters. The converted digital data may
then be manipulated mathematically. Many data-acquisition systems must also construct a representation
of the original signal from the digital data stream.

Unfortunately, sampling often sacrifices accuracy for the sake of convenience. The digital version of a
signal may not resemble the original in some important respects. A graphic example is the movie scene
that apparently shows wagon wheels or helicopter blades turning backwards. This erroneous image,
known as an "alias", occurs because a "motion picture" camera actually samples continuous action into a
series of stills, and the frame rate (commonly 24 or 30 frames per second) is not fast enough or is nearly
an exact multiple of the object's rotation speed.

According to Nyquist's Theorem, accurately representing an analog signal with samples requires that the
original signal's highest frequency component be less than the Nyquist frequency, which is at least half the
sampling frequency. To correct the image in the movie example, the frame rate would have to exceed
twice the rotation speed of the wheel (or its spokes) or of the helicopter blades. No practical data-
acquisition system can sample fast enough to catch all of a real signal's components. Frequencies above
Nyquist appear as false low-frequency aliases. As an example, Figure 1 shows the result of sampling a
140 Hz signal at 100 Hz.

Figure 1

The process seems to indicate that the original signal was a 40 Hz sine wave, the difference between the
actual input wave and sampling frequencies.
Aliasing is a fundamental mathematical result of the sampling process. It occurs independent of any
physical sampling-system capabilities. Downstream processing cannot reverse its effect. Only filtering out
the alias high frequency components before sampling begins can prevent it.

When a signal undergoes A/D conversion, the amplitude of any frequency component above the Nyquist
frequency should be no higher than the converter's least significant bit (LSB). Some sources insist on
reducing the amplitude to below half of the LSB. For any full-scale undesirable signal component, then,
attenuation should be by at least a factor of 2n, where "n" is the number of bits in the A/D. For half of the
LSB, attenuation would be by a factor of 2n+1 . A 12-bit A/D, then, demands attenuation by a factor of at
least 4096 or 8192. To convert these attenuation requirements to decibels, we note that attenuation of
amplitude by a factor of 2 is equivalent to attenuation by 6 dB (since 20log10(2) = 6.02), and attenuation
by a factor of 2n = n* 6dB. Thus a 12 bit A to D should attenuate signals with frequencies above the
Nyquist frequency by 72 dB or 78 dB.

In practice, noise-signal amplitudes rarely match the amplitudes of signal components of interest, so this
attenuation calculation represents worst case.

IDEAL FILTER SHAPES (THEORETICAL)

Every electronic design project produces signals that require filtering, processing, or amplification, from
simple gain to the most complex DSP. The following presentation attempts to "de-mystify" some of these
signal-processing requirements. The concepts of ideal filters, commonly used filter transfer function
characteristics and implementation techniques will assist the reader in determining their electronic filter
and signal conditioning needs.

Real-world signals contain both wanted and unwanted information. Therefore, some kind of filtering
technique must separate the two before processing and analysis can begin.

An ideal filter transmits frequencies in its pass-band, unattenuated and without phase shift, while not
allowing any signal components in the stop-band to get through. All filters offer a pass-band, a stop-band
and a cutoff frequency or corner frequency (fc) that defines the frequency boundary between the pass-band
and the stop-band.

Figure 2 shows the four basic filter types: low-pass, high-pass, band-pass and band-reject (notch) filters.
The differences among these filter types depend on the relationship between pass- and stop-bands.
 Figure 2

Low-pass filters are by far the most common filter type, earning wide popularity in removing alias signals
and for other aspects of data acquisition and signal conversion. For a low-pass filter, the pass-band
extends from DC (0 Hz) to fc and the stop-band lies above fc .

In a high-pass filter, the pass-band lies above fc , while the stop-band resides below that point.

Combining high-pass and low-pass technologies permits constructing band-pass and band-reject filters.
Band-pass filters transmit only those signal components within a band around a center frequency fo .An
ideal band-pass filter would feature brick-wall transitions at fL and fH , rejecting all signal frequencies
outside that range. Band-pass filter applications include situations that require extracting a specific tone,
such as a test tone, from adjacent tones or broadband noise.

Band-reject (sometimes called band-stop or notch ) filters transmit all signals except those between fH
and fL . These filters can remove a specific tone - such as a 50 or 60 Hz line frequency pickup - from other
signals. Another common application is medical instrumentation, where high-impedance sensors pick up
line frequencies.

NON-IDEAL FILTERS (REAL WORLD)

Real-world signals contain both wanted and unwanted information. Therefore, some kind of filtering
technique must separate the two before processing and analysis can begin. Real filters are far from ideal.
They subject input signals to mathematical transfer functions with names like Butterworth, Bessel,
constant delay and elliptic that only approximate ideal behavior. Instead of the sharply defined transition
represented by ideal filters, real filters contain a transition region between the pass-band and the stop-band
as shown in Figure 3.

Figure 3

In addition, the pass-band is not flat like the ideal filter, may contain attenuation ripple, and the
attenuation in the stop-band may not be infinite. In order to simplify the analysis of various real world
filter types, filter response curves are normalized. When selecting a filter, this normalized data allows the
designer to compare the theoretical amplitude, phase and delay characteristics of each filter type.

Mathematics of filters

All the filters we will consider are designed to transform a sinusoidal input into a sinusoidal output with
the same frequency, but different magnitude and phase angle. The magnitude and phase effect of the filter
are different at different frequencies, of course. The filter’s frequency response, or gain, denoted H(ω) or
H(f), is the complex function of frequency that describes the magnitude and phase effect of the filter at
each frequency. (f = frequency in cycles/second; ω = frequency in radians/second = 2πf.) The frequency
response H(f) of an ideal filter would have magnitude=1 in the passband and magnitude=0 in the
stopband. The phase of an ideal H(f) would be 0 in the passband (i.e. no alteration in phase) and of no
importance in the stopband (since the magnitude would be 0 there). The frequency response, H(ω) or
H(f), of a real filter is described by the ratio of two polynomials:

H(ω) = N(ω) / D(ω) =

[1 + b1 (iω) +b2 (iω)2 +b3 (iω)3 + … + bM (iω)M] / [1 + a1 (iω) + a2 (iω)2 + a3 (iω)3 + … + aN (iω)N]

If the numerator polynomial is of degree M, the filter is said to have M zeroes; if the denominator
polynomial is of order N, the filter is said to have N poles. A high order filter, i.e. a filter with many poles
(and maybe zeroes) can have theoretically better (i.e. closer to ideal) performance, but it is more
complicated and expensive to implement, and it is more susceptible to degraded performance when any of
the polynomial coefficients are slightly off-spec. This means high order filters are finicky. Most common
“classical” filter types, including Butterworth, Bessel, Chebyshev, and elliptic, have no zeroes (the
numerator = 1). For them, the filter order equals the order of the denominator polynomial equals the
number of poles. The attenuation ratio is the reciprocal of the magnitude of H(ω):

A(ω) = 1 / | H(ω) | .

Example: Second order Butterworth low pass filter with cutoff frequency of 1 radian/sec.

The second order Butterworth low pass filter response is given by

H(ω) = 1 / [ 1 + sqrt(2) i ω – ω2 ] . (Take this on faith. We will not prove it here.)

Note that the denominator polynomial is of order 2, so the filter has 2 poles. The attenuation ratio is

A(ω) = 1 / | H(ω) | = | 1 + sqrt(2) i ω – ω2 |

= sqrt [ ( 1 + sqrt(2) i ω – ω2) ( 1 – sqrt(2) i ω – ω2) ] = sqrt [ 1 + ω4 ] .

(The last step in the above equations requires quite a bit of algebraic manipulation to prove, so don’t be
surprised if it is not obvious.) Therefore the attenuation ratio when ω<<1 is A(ω<<1) = sqrt (1) = 1 = 0
dB. The attenuation ratio when ω=1 is A(ω=1) = sqrt (2) = 3 dB. An attenuation of 3 dB is equal to a
gain of -3 dB, since attenuation and gain are reciprocals of one another. By convention, the frequency at
which a Butterworth or Bessel filter has a gain of -3 dB is called the filter cutoff frequency, because at that
frequency the amplitude is down by a factor of sqrt(2), and the power is down by a factor of 2.

Normalization

See Figure 4 below for the theoretical performance characteristics and normalized response curve of an
8-pole, 6-zero constant delay filter. The frequency axis on the response plot is scaled so that the corner or
ripple frequency is always one Hertz instead of the actual intended corner or ripple frequency. This allows
one normalized curve to represent any filter that would have the same response shape. To convert a
normalized amplitude response curve to a curve representing a filter whose corner frequency is not at one
Hertz, multiplying any number on the frequency axis by the intended corner or ripple frequency scales the
frequency axis.
 Figure 4. Frequency Response

Amplitude Response

Amplitude Response is defined as the ratio of the output amplitude to the input amplitude versus
frequency and is usually plotted on a log/log scale as shown in Figure 5. Note how the steepness of the
transition band slope (roll-off) increases as the number of poles increase.

Figure 5. 2, 4, 6, and 8 Pole Butterworth Low Pass

Phase Response

All non-ideal filters introduce a time delay between the filter input and output terminals. This delay can be
represented as a phase shift if a sine wave is passed through the filter. The extent of phase shift depends
on the filter's transfer function. For most filter shapes, the amount of phase shift changes with the input
signal frequency. The normal way of representing this change in phase is through the concept of Group
Delay, or simply Delay: the time delay experienced by a sinusoidal wave of a particular frequency as it
passes through the filter.

Delay is also equal to the slope of the plot of phase versus frequency (on a plot with linear, not
logarithmic, axes). Figure 6 compares the group delay of some typical phase response curves

Figure 6: Delay for 8 Pole Low Pass Filters

Butterworth, Bessel, Constant Delay, Elliptic

Thus a point on a normalized group delay curve that has a group delay of one (1.0) would yield 1
millisecond Actual Delay for a filter with a 1KHz corner frequency.

Normalized Group Delay

Actual Delay =
Actual Corner Frequency (fc) in Hz
 1.0
Actual Delay = = 0.001 sec
1000 Hz

Analog Filter Specifications

Low Pass and High Pass

In order to define the limits of the filter pass-band in real circuits, most filter specifications define the
corner frequency (fc), as the frequency where attenuation reaches -3 dB or for elliptic filters, the ripple
frequency (fr), the point where the response curve last passes through the specified pass-band ripple.

Figure 7 is an elliptic filter with 0.05 dB passband ripple that attenuates to -80 dB at 1.56 fr . The
following table shows the theoretical filter attenuation versus frequency (expressed in terms of the ripple
frequency fr).

Filter Attenuation (Theoretical)

0.05 dB 1.00 fr
3.01 dB 1.05 fr
60.0 dB 1.45 fr
80.0 dB 1.56 fr

Also note that the elliptic transfer function attenuation is not monotonic in the stopband (i.e. not steadily
decreasing). Instead, the stopband attenuation has notches and humps. There is a stopband “floor” of
about -83 dB.

Figure 7
Band-Pass and Band Reject Filters

Specific items of interest for Band-Pass filters are the Center Frequency (geometric mean) fo and the
Filter Bandwidth.

Frequency fo represents the geometric mean of fH and fL. That is:

fo = (fH * fL ) 1/2

Bandwidth is defined as the difference between pass-band extremes:

Bandwidth = fH - fL

Figure 8 is a plot of a four pole-pair band-pass (i.e. 8 poles total) with a Butterworth transfer function.

Figure 8

FILTER SELECTION

Transfer functions can be classified into one of two basic categories, Amplitude filters and Phase filters.
Amplitude filters are designed for the best amplitude response for a given situation, for example zero
ripple in the amplitude response pass-band. Phase filters are designed for desired phase response, such as
linear phase with frequency throughout the filter amplitude pass-band.
Amplitude Filters
For many applications the design goal is to approximate ideal "brick wall" frequency response. Probably
the most common amplitude filter transfer function is the Butterworth. It yields the maximally flat
amplitude response in the pass-band (the first 2N - 1 derivatives of the frequency response are equal to
zero, where N is the filter degree, or number of poles). Therefore, amplitude response rolls-off
monotonically (uniform slope) as frequency increases in the stop-band.

The attenuation ratio, "A(ω)", of a Butterworth low-pass filter with a cutoff of 1 radian/sec is given by:

A(ω) = sqrt [ 1 + ω2N ] ,

where N = degree of the filter (number of poles).

Butterworth filters produce no pass-band ripple and provide theoretically infinite attenuation as frequency
increases when compared to fc . The primary limitation is that Butterworth filters produce slower roll-off
than some of the alternative transfer functions.

The attenuation ratio of a Chebychev transfer function (Figure 6C) is given by:

A(ω) = sqrt [ 1 + ε2 CN2(ω) ] .

which generates a series of polynomials, where ε is pass-band ripple and CN(ω) represents the nth order
polynomial in the series. Table 1 shows the first five Chebychev polynomials.

Chebychev Polynomials CN(ω)

N CN(ω)
1 ω
2 2ω2 – 1
3 4ω3 – 3ω
4 8ω4 – 8ω2 + 1
5 16ω5 – 20ω3 + 5ω

Table 1

The Chebychev function provides faster roll-off in the transition band than a Butterworth filter would, but
at the expense of some variation in the pass-band called ripple. Ripple denotes that the amplitude in the
pass-band varies between 1 and (1 + ε 2), where ε is always less than 1. Like the Butterworth, Chebychev
stop-band roll-off is monotonic. Many designers avoid Chebychev filters in favor of Cauer elliptic (or
simply elliptic) filters, because elliptic filters provide faster roll-off in the transition-band.

The elliptic filter attenuation ratio is given by

A(ω) = sqrt [ 1 + ε2 ZN2(ω) ] .

where ZN is the nth order elliptic polynomial and ε determines pass-band ripple attenuation at the cutoff
frequency, ω = 1. Although an elliptic filter achieves faster roll-off than either Butterworth or Chebychev
varieties, it introduces ripple in both the pass- and stop-bands. Also, elliptic filter roll-off is not
monotonic, eventually reaching an attenuation limit, called the stop-band floor.
For elliptic filters, shape factor depends not on the -3 dB corner frequency (fc), but on ripple frequency
(fr), the highest pass-band frequency on a low-pass filter or the lowest pass-band frequency on a high-pass
filter where pass-band ripple occurs, as shown in Figure 11.

Figure 11

At the stop-band edge, a small frequency change produces a large change in attenuation. Another critical
element in the shape of an elliptic filter is frequency fs, which denotes the first frequency at which the
attenuation reaches the stop-band floor.

Figure 12 compares the amplitude response of eight-pole Butterworth, eight-pole 0.1 dB ripple
Chebychev, and eight-pole 0.1 dB ripple, -84 dB stop-band floor elliptic filters. The curves are normalized
to the -3 dB cutoff frequencies.
 Figure 12

Generally, filters that produce faster roll-off in the transition-band exhibit poorer phase response and
group delay characteristics (See Figure 6).

Phase Filters
For some filter applications it is desirable to preserve a transient waveform while removing higher
frequency noise components from the signal. If each of the frequency components of the input waveform
(from the Fourier series or the Fourier transform) is phase shifted an amount linearly proportional to
frequency, then they remain in the correct time relationship and sum together to create, at the output, the
original waveform that was present at the input of the filter, with the higher frequencies components
having been removed by the filter. When a filter has phase delay that varies linearly with frequency it is
called a Linear Phase filter. A linear phase filter has a constant group delay, at least through the pass-
band. Amplitude filters provide relatively constant group delay only from 0 Hz to about the mid pass-band
frequency range peak near fc.

As with amplitude filters, mathematicians have provided polynomial approximations of an ideal linear
phase transfer function. The most common linear phase filter is based on Bessel (sometimes called
Thompson) functions. Bessel filters provide very linear phase response and little delay distortion (constant
group delay) in the pass-band. They show no overshoot in response to step input and roll-off
monotonically in the stop-band. They also exhibit much slower attenuation in the transition-band than
amplitude filters. Figure 13 presents amplitude and delay response curves for an 8-pole Bessel. Other
types of phase filters include, constant-delay (a modified Bessel), equiripple phase, equiripple delay, and
Gaussian transfer functions. They either have more pass-band amplitude roll-off for only a small
improvement in phase linearity or only slightly less roll-off in the pass-band at the expense of degrading
the phase linearity.
 Figure 13

Compensated Filters
Some applications require filters offering the sharp roll-off characteristics of amplitude-type filters and the
linearity of phase-type transfer functions. Two techniques, amplitude equalization and delay equalization,
are available to achieve these ends. Both add complexity to filter design, and have theoretical and
practical limits.

Amplitude compensation modifies the amplitude response of phase filters to produce a filter that is
sometimes called a constant delay filter. This technique can achieve a factor-of-two improvement in
Bessel roll-off to a -80 dB floor, comparable to Butterworth-filter performance. For comparison, Figure
15 shows the amplitude response of an 8-pole Bessel, an 8-pole, 6-zero constant delay, and a 8-pole
Butterworth response.
 Figure 15

OUTPUT SIGNAL ERRORS

Besides inaccuracies of theoretical approximation, the most significant side effects of signal filtering are
the following:

Settling time is not strictly an output signal error because it is mathematically related to the filter transfer
function, but is usually deemed to be an undesirable filter side effect. All filters serve to delay the input
signal by a certain minimum amount as well as increasing rise and fall time of any fast changing input
signal. A general rule for settling time is that the more the filter approaches a "brick-wall" approximation,
the longer it will take to settle. Therefore, an eight-pole filter will take longer to settle than a four-pole
filter.

Step Response for amplitude type filters may exhibit substantial overshoot (ringing) when presented with
a sudden change in voltage amplitude at the filter input. See Figure 17 for typical 8 pole transfer function
step response curves.
 Figure 17

SELECTING THE RIGHT ANALOG FILTER

Choosing the correct filter shape for a particular application requires defining properties of the incoming
signal that the filter must remove, as well as the properties that it must retain. In most situations, there is
some overlap between these two areas, demanding a degree of compromise. Digital filters (i.e. filters
implemented in software, which act on the digitized waveform) offer several advantages over analog
filters (to be dicussed later). However, ant-aliasing filtering (lowpass filtering to remove frequencies
above the Nyquist frequency) can only be done by an analog filter. Therefore, the most common
application for analog filtering is anti-aliasing.

Time Domain Waveform Preservation

Filters for such applications feature linear phase response in the pass-band, and must not introduce ringing
or overshoot. Phase-derived filters, such as Bessel or constant-delay (equiripple-phase) and their
amplitude-compensated derivatives, work best in these cases.
High Selectivity in the Frequency Domain
Situations where removal of undesired components is the overriding concern and some distortion in the
time domain of the signal's shape is of less importance generally require sharper roll-off filters with
Butterworth or elliptic transfer functions. Spectrum analysis, for example, involves only the amplitude of
each frequency component of the input signal. Most voice and data transmission also requires integrity
only of amplitudes, as do many forms of modal analysis, which determines resonant frequencies of
structures and objects.

Compromise Filters
Although linear-phase filters preserve critical information, many applications also require rapid transition-
band roll-off. Anti-aliasing filters fit into this category. A balance between these mutually exclusive
requirements can often be achieved by phase-derived types and amplitude-compensated versions of phase
filters.

Further Comments on Anti-aliasing Analog Filters (by WCR)

If the Nyquist frequency is not that much higher than the highest frequency of interest in the signal, then it
will be hard to find an anti-aliasing filter that will do the job. If a filter is found, it will probably be
expensive. However, it may be easier and cheaper to simply sample at a higher rate, which raises the
Nyquist frequency, and reduces the demands on the filter. For example, if the frequencies of interest
extend to 300 Hz, and the sampling rate is 1 kHz, then the Nyquist frequency is 500 Hz. This means the
antialiasing filter’s passband should extend up to 300 Hz and the stopband should start at 500 Hz, a factor
of 1.67 difference in frequency. If the signal is digitized with 12 bit resolution, then the stopband
attenuation should be >= 72 dB (see above). It would take a very high order filter to meet these
requirements. But if the sampling rate were increased to 2 kHz (so Nyquist frequency = 1 kHz), then the
passband and stopband frequencies for the filter would differ by a factor of 3.33, which reduces the
demands on the filter performance.
If a high performance anti-aliasing filter is needed, consider the 8 pole, 6 zero filters available from
Frequency Devices Inc. They are available with -80 dB stopband floor (perfect for 12 bit A to D
conversion). They have excellent phase linearity in the passband (which means good time domain
performance, very little overshoot or ringing), and rapid rolloff above the cutoff frequency. They are
available as fixed frequency units (the cutoff is set at the factory set and is not adjustable) and with
programmable cutoff frequency (adjustable according to user needs). The following figures show
frequency and time domain behavior of the 8 pole Bessel filter and the 8 pole, 6 zero filter.
8-pole 6-zero constant delay low pass (-80 dB) Low pass 8 pole Bessel (D828L8L)
(D828L8D80)
(Source: www.frequencydevices.com,10/18/2004.)