319
CHAPTER
19 Recursive Filters
Recursive filters are an efficient way of achieving a long impulse response, without having to
perform a long convolution. They execute very rapidly, but have less performance and flexibility
than other digital filters. Recursive filters are also called Infinite Impulse Response (IIR) filters,
since their impulse responses are composed of decaying exponentials. This distinguishes them
from digital filters carried out by convolution, called Finite Impulse Response (FIR) filters. This
chapter is an introduction to how recursive filters operate, and how simple members of the family
can be designed. Chapters 20, 26 and 33 present more sophisticated design methods.
The Recursive Method
To start the discussion of recursive filters, imagine that you need to extract
information from some signal, . Your need is so great that you hire an old x[ ]
mathematics professor to process the data for you. The professor's task is to
filter to produce , which hopefully contains the information you are x[ ] y[ ]
interested in. The professor begins his work of calculating each point in y[ ]
according to some algorithm that is locked tightly in his over-developed brain.
Part way through the task, a most unfortunate event occurs. The professor
begins to babble about analytic singularities and fractional transforms, and
other demons from a mathematician's nightmare. It is clear that the professor
has lost his mind. You watch with anxiety as the professor, and your algorithm,
are taken away by several men in white coats.
You frantically review the professor's notes to find the algorithm he was
using. You find that he had completed the calculation of points through y[0]
, and was about to start on point . As shown in Fig. 19-1, we will y[27] y[28]
let the variable, n, represent the point that is currently being calculated. This
means that is sample 28 in the output signal, is sample 27, y[n] y[n&1]
is sample 26, etc. Likewise, is point 28 in the input signal, y[n&2] x[n]
The Scientist and Engineer's Guide to Digital Signal Processing 320
y [n ] ' a0 x [n ] % a1 x [n &1] % a2 x [n &2] % a3 x [n &3] % þ
y [n ] ' a0 x [n ] % a1 x [n &1] % a2 x [n &2] % a3 x [n &3] % þ
% b1 y [n &1] % b2 y [n &2] % b3 y [n &3] % þ
EQUATION 19-1
The recursion equation. In this equation, is x[ ]
the input signal, is the output signal, and the y[ ]
a's and b's are coefficients.
is point 27, etc. To understand the algorithm being used, we ask x[n&1]
ourselves: "What information was available to the professor to calculate , y[n]
the sample currently being worked on?"
The most obvious source of information is the input signal, that is, the values:
. The professor could have been multiplying each point x[n], x[n&1], x[n&2],þ
in the input signal by a coefficient, and adding the products together:
You should recognize that this is nothing more than simple convolution, with
the coefficients: , forming the convolution kernel. If this was all the a0, a1, a2,þ
professor was doing, there wouldn't be much need for this story, or this chapter.
However, there is another source of information that the professor had access
to: the previously calculated values of the output signal, held in:
. Using this additional information, the algorithm y[n&1], y[n&2], y[n&3],þ
would be in the form:
In words, each point in the output signal is found by multiplying the values
from the input signal by the "a" coefficients, multiplying the previously
calculated values from the output signal by the "b" coefficients, and adding the
products together. Notice that there isn't a value for , because this b0
corresponds to the sample being calculated. Equation 19-1 is called the
recursion equation, and filters that use it are called recursive filters. The
"a" and "b" values that define the filter are called the recursion coefficients.
In actual practice, no more than about a dozen recursion coefficients can be
used or the filter becomes unstable (i.e., the output continually increases or
oscillates). Table 19-1 shows an example recursive filter program.
Recursive filters are useful because they bypass a longer convolution. For
instance, consider what happens when a delta function is passed through a
recursive filter. The output is the filter's impulse response, and will typically
be a sinusoidal oscillation that exponentially decays. Since this impulse
response in infinitely long, recursive filters are often called infinite impulse
response (IIR) filters. In effect, recursive filters convolve the input signal with
a very long filter kernel, although only a few coefficients are involved.
Chapter 19- Recursive Filters 321
Sample number
0 10 20 30
-2
-1
0
1
2
a. The input signal, x[ ]
x[n-3]
x[n-2]
x[n]
x[n-1]
Sample number
0 10 20 30
-2
-1
0
1
2
b. The output signal, y[ ]
y[n-3]
y[n-2]
y[n]
y[n-1]
FIGURE 19-1
Recursive filter notation. The output sample being calculated, , is determined by the values from y[n]
the input signal, , as well as the previously calculated values in the output x[n], x[n&1], x[n&2], þ
signal, . These figures are shown for . y[n&1], y[n&2], y[n&3], þ n ' 28
Amplitude
Amplitude
100 'RECURSIVE FILTER
110 '
120 DIM X[499] 'holds the input signal
130 DIM Y[499] 'holds the filtered output signal
140 '
150 GOSUB XXXX 'mythical subroutine to calculate the recursion
160 ' 'coefficients: A0, A1, A2, B1, B2
170 '
180 GOSUB XXXX 'mythical subroutine to load X[ ] with the input data
190 '
200 FOR I% = 2 TO 499
210 Y[I%] = A0*X[I%] + A1*X[I%-1] + A2*X[I%-2] + B1*Y[I%-1] + B2*Y[I%-2]
220 NEXT I%
230 '
240 END
TABLE 19-1
The relationship between the recursion coefficients and the filter's response is
given by a mathematical technique called the z-transform, the topic of
Chapter 33. For example, the z-transform can be used for such tasks as:
converting between the recursion coefficients and the frequency response,
combining cascaded and parallel stages into a single filter, designing recursive
systems that mimic analog filters, etc. Unfortunately, the z-transform is very
mathematical, and more complicated than most DSP users are willing to deal
with. This is the realm of those that specialize in DSP.
There are three ways to find the recursion coefficients without having to
understand the z-transform. First, this chapter provides design equations for
several types of simple recursive filters. Second, Chapter 20 provides a
"cookbook" computer program for designing the more sophisticated Chebyshev
low-pass and high-pass filters. Third, Chapter 26 describes an iterative method
for designing recursive filters with an arbitrary frequency response.
The Scientist and Engineer's Guide to Digital Signal Processing 322
Sample number
0 10 20 30 40
-0.5
0.0
0.5
1.0
1.5
Sample number
0 10 20 30 40
-0.5
0.0
0.5
1.0
1.5
Time
0 10 20 30
-0.5
0.0
0.5
1.0
1.5
Time
0 10 20 30 40
-0.5
0.0
0.5
1.0
1.5
R
C
Digital Filter
Analog Filter
Recursive
Filter
a0 = 0.15
b1 = 0.85
FIGURE 19-2
Single pole low-pass filter. Digital recursive filters can mimic analog filters composed of resistors and
capacitors. As shown in this example, a single pole low-pass recursive filter smoothes the edge of a step input,
just as an electronic RC filter.
Amplitude
Amplitude Amplitude
Amplitude
Single Pole Recursive Filters
Figure 19-2 shows an example of what is called a single pole low-pass filter.
This recursive filter uses just two coefficients, and . For a0
' 0.15 b1
' 0.85
this example, the input signal is a step function. As you should expect for a
low-pass filter, the output is a smooth rise to the steady state level. This figure
also shows something that ties into your knowledge of electronics. This lowpass
recursive filter is completely analogous to an electronic low-pass filter
composed of a single resistor and capacitor.
The beauty of the recursive method is in its ability to create a wide variety of
responses by changing only a few parameters. For example, Fig. 19-3 shows
a filter with three coefficients: , and . As a0
' 0.93 a1
' &0.93 b1
' 0.86
shown by the similar step responses, this digital filter mimics an electronic RC
high-pass filter.
These single pole recursive filters are definitely something you want to keep
in your DSP toolbox. You can use them to process digital signals just as
you would use RC networks to process analog electronic signals. This
includes everything you would expect: DC removal, high-frequency noise
suppression, wave shaping, smoothing, etc. They are easy to program, fast
Chapter 19- Recursive Filters 323
Sample number
0 10 20 30 40
-0.5
0.0
0.5
1.0
1.5
Sample number
0 10 20 30 40
-0.5
0.0
0.5
1.0
1.5
Time
0 10 20 30
-0.5
0.0
0.5
1.0
1.5
Time
0 10 20 30 40
-0.5
0.0
0.5
1.0
1.5
Recursive
Filter
a0 = 0.93
a1 = -0.93
R
C
Digital Filter
Analog Filter
b1 = 0.86
FIGURE 19-3
Single pole high-pass filter. Proper coefficient selection can also make the recursive filter mimic an electronic
RC high-pass filter. These single pole recursive filters can be used in DSP just as you would use RC circuits
in analog electronics.
Amplitude
Amplitude Amplitude
Amplitude
EQUATION 19-3
Single pole high-pass filter.
a0
' (1%x ) /2
a1
' &(1%x) /2
b1
' x
EQUATION 19-2
Single pole low-pass filter. The filter's
response is controlled by the parameter, x,
a value between zero and one.
a0
' 1&x
b1
' x
to execute, and produce few surprises. The coefficients are found from these
simple equations:
The characteristics of these filters are controlled by the parameter, x, a
value between zero and one. Physically, x is the amount of decay between
adjacent samples. For instance, x is 0.86 in Fig. 19-3, meaning that the
value of each sample in the output signal is 0.86 the value of the sample
before it. The higher the value of x, the slower the decay. Notice that the
The Scientist and Engineer's Guide to Digital Signal Processing 324
Sample number
0 100 200 300 400 500
-0.5
0.0
0.5
1.0
1.5
a. Original signal
Sample number
0 100 200 300 400 500
-0.5
0.0
0.5
1.0
1.5
b. Filtered signals
low-pass
high-pass
FIGURE 19-4
Example of single pole recursive filters. In (a), a high frequency burst rides on a slowly varying signal. In (b),
single pole low-pass and high-pass filters are used to separate the two components. The low-pass filter uses x
= 0.95, while the high-pass filter is for x = 0.86.
Amplitude
Amplitude
EQUATION 19-4
Time constant of single pole filters. This
equation relates the amount of decay
between samples, x, with the filter's time
constant, d, the number of samples for the
filter to decay to 36.8%.
x ' e &1/d
EQUATION 19-5
Cutoff frequency of single pole filters.
The amount of decay between samples, x,
is related to the cutoff frequency of the
filter, , a value between 0 and 0.5. fC
x ' e &2BfC
filter becomes unstable if x is made greater than one. That is, any nonzero
value on the input will make the output increase until an overflow occurs.
The value for x can be directly specified, or found from the desired time
constant of the filter. Just as R×C is the number of seconds it takes an RC
circuit to decay to 36.8% of its final value, d is the number of samples it takes
for a recursive filter to decay to this same level:
For instance, a sample-to-sample decay of corresponds to a time x ' 0.86
constant of samples (as shown in Fig 19-3). There is also a fixed d ' 6.63
relationship between x and the -3dB cutoff frequency, , of the digital filter: fC
This provides three ways to find the "a" and "b" coefficients, starting with the
time constant, the cutoff frequency, or just directly picking x.
Figure 19-4 shows an example of using single pole recursive filters. In (a), the
original signal is a smooth curve, except a burst of a high frequency sine wave.
Figure (b) shows the signal after passing through low-pass and high-pass
filters. The signals have been separated fairly well, but not perfectly, just as
if simple RC circuits were used on an analog signal.
Chapter 19- Recursive Filters 325
Frequency
0 0.1 0.2 0.3 0.4 0.5
0.0
0.5
1.0
1.5
fc = 0.25
0.05
0.01
c. Low-pass filter (4 stage)
Frequency
0 0.1 0.2 0.3 0.4 0.5
0.0
0.5
1.0
1.5
fc = 0.25
a. High-pass filter
0.05
0.01
Frequency
0 0.1 0.2 0.3 0.4 0.5
0.0
0.5
1.0
1.5
fc = 0.25
0.05
0.01
b. Low-pass filter
FIGURE 19-5
Single pole frequency responses. Figures (a)
and (b) show the frequency responses of highpass
and low-pass single pole recursive filters,
respectively. Figure (c) shows the frequency
response of a cascade of four low-pass filters.
The frequency response of recursive filters is
not always what you expect, especially if the
filter is pushed to extreme limits. For example,
the curve in (c) is quite useless. Many fC
' 0.25
factors are to blame, including: aliasing, roundoff
noise, and the nonlinear phase response.
Amplitude Amplitude
Amplitude
Figure 19-5 shows the frequency responses of various single pole recursive
filters. These curves are obtained by passing a delta function through the filter
to find the filter's impulse response. The FFT is then used to convert the
impulse response into the frequency response. In principle, the impulse
response is infinitely long; however, it decays below the single precision roundoff
noise after about 15 to 20 time constants. For example, when the time
constant of the filter is samples, the impulse response can be d ' 6.63
contained in about 128 samples.
The key feature in Fig. 19-5 is that single pole recursive filters have little
ability to separate one band of frequencies from another. In other words,
they perform well in the time domain, and poorly in the frequency domain.
The frequency response can be improved slightly by cascading several
stages. This can be accomplished in two ways. First, the signal can be
passed through the filter several times. Second, the z-transform can be used
to find the recursion coefficients that combine the cascade into a single
stage. Both ways work and are commonly used. Figure (c) shows the
frequency response of a cascade of four low-pass filters. Although the
stopband attenuation is significantly improved, the roll-off is still terrible.
If you need better performance in the frequency domain, look at the
Chebyshev filters of the next chapter.
The Scientist and Engineer's Guide to Digital Signal Processing 326
EQUATION 19-6
Four stage low-pass filter. These equations
provide the "a" and "b" coefficients for a
cascade of four single pole low-pass filters.
The relationship between x and the cutoff
frequency of this filter is given by Eq. 19-5,
with the 2B replaced by 14.445.
a0
' (1&x )4
b1
' 4x
b2
' &6x 2
b3
' 4x 3
b4
' &x 4
EQUATION 19-7
Band-pass filter. An example frequency
response is shown in Fig. 19-6a. To use
these equations, first select the center
frequency, f, and the bandwidth, BW. Both
of these are expressed as a fraction of the
sampling rate, and therefore in the range of
0 to 0.5. Next, calculate R, and then K, and
then the recursion coefficients.
a0
' 1&K
a1
' 2(K&R) cos( 2Bf )
a2
' R 2&K
b1
' 2R cos( 2Bf )
b2
' &R 2
EQUATION 19-8
Band-reject filter. This filter is commonly
called a notch filter. Example frequency
responses are shown in Fig. 19-6b.
a0
' K
a1
' &2K cos( 2Bf )
a2
' K
b1
' 2R cos(2Bf )
b2
' &R 2
K ' 1& 2R cos(2Bf )% R 2
2& 2cos(2Bf )
R ' 1 & 3BW
where:
The four stage low-pass filter is comparable to the Blackman and Gaussian
filters (relatives of the moving average, Chapter 15), but with a much faster
execution speed. The design equations for a four stage low-pass filter are:
Narrow-band Filters
A common need in electronics and DSP is to isolate a narrow band of
frequencies from a wider bandwidth signal. For example, you may want to
eliminate 60 hertz interference in an instrumentation system, or isolate the
signaling tones in a telephone network. Two types of frequency responses are
available: the band-pass and the band-reject (also called a notch filter).
Figure 19-6 shows the frequency response of these filters, with the recursion
coefficients provided by the following equations:
Chapter 19- Recursive Filters 327
Frequency
0 0.1 0.2 0.3 0.4 0.5
0.0
0.5
1.0
1.5
a. Band-pass frequency response
BW=0.0066
single stage
three stages
cascade of
Frequency
0 0.1 0.2 0.3 0.4 0.5
0.0
0.5
1.0
1.5
b. Band-reject frequency response
BW=0.0066
BW=0.033
FIGURE 19-6
Characteristics of narrow-band filters. Figure (a)
and (b) shows the frequency responses of
various band-pass and band-reject filters. The
step response of the band-reject filter is shown
in (c). The band-reject (notch) filter is useful
for removing 60 Hz and similar interference
from time domain encoded waveforms.
Sample number
0 50 100 150 200
0.0
0.5
1.0
1.5
c. Band-reject step response
BW=0.0066
Amplitude Amplitude
Amplitude
Two parameters must be selected before using these equations: f, the center
frequency, and BW, the bandwidth (measured at an amplitude of 0.707). Both
of these are expressed as a fraction of the sampling frequency, and therefore
must be between 0 and 0.5. From these two specified values, calculate the
intermediate variables: R and K, and then the recursion coefficients.
As shown in (a), the band-pass filter has relatively large tails extending from
the main peak. This can be improved by cascading several stages. Since the
design equations are quite long, it is simpler to implement this cascade by
filtering the signal several times, rather than trying to find the coefficients
needed for a single filter.
Figure (b) shows examples of the band-reject filter. The narrowest bandwidth
that can be obtain with single precision is about 0.0003 of the sampling
frequency. When pushed beyond this limit, the attenuation of the notch will
degrade. Figure (c) shows the step response of the band-reject filter. There is
noticeable overshoot and ringing, but its amplitude is quite small. This allows
the filter to remove narrowband interference (60 Hz and the like) with only a
minor distortion to the time domain waveform.
The Scientist and Engineer's Guide to Digital Signal Processing 328
Phase Response
There are three types of phase response that a filter can have: zero phase,
linear phase, and nonlinear phase. An example of each of these is shown
in Figure 19-7. As shown in (a), the zero phase filter is characterized by an
impulse response that is symmetrical around sample zero. The actual shape
doesn't matter, only that the negative numbered samples are a mirror image of
the positive numbered samples. When the Fourier transform is taken of this
symmetrical waveform, the phase will be entirely zero, as shown in (b).
The disadvantage of the zero phase filter is that it requires the use of negative
indexes, which can be inconvenient to work with. The linear phase filter is a
way around this. The impulse response in (d) is identical to that shown in (a),
except it has been shifted to use only positive numbered samples. The impulse
response is still symmetrical between the left and right; however, the location
of symmetry has been shifted from zero. This shift results in the phase, (e),
being a straight line, accounting for the name: linear phase. The slope of this
straight line is directly proportional to the amount of the shift. Since the shift
in the impulse response does nothing but produce an identical shift in the output
signal, the linear phase filter is equivalent to the zero phase filter for most
purposes.
Figure (g) shows an impulse response that is not symmetrical between the left
and right. Correspondingly, the phase, (h), is not a straight line. In other
words, it has a nonlinear phase. Don't confuse the terms: nonlinear and
linear phase with the concept of system linearity discussed in Chapter 5.
Although both use the word linear, they are not related.
Why does anyone care if the phase is linear or not? Figures (c), (f), and (i)
show the answer. These are the pulse responses of each of the three filters.
The pulse response is nothing more than a positive going step response
followed by a negative going step response. The pulse response is used here
because it displays what happens to both the rising and falling edges in a
signal. Here is the important part: zero and linear phase filters have left and
right edges that look the same, while nonlinear phase filters have left and right
edges that look different. Many applications cannot tolerate the left and right
edges looking different. One example is the display of an oscilloscope, where
this difference could be misinterpreted as a feature of the signal being
measured. Another example is in video processing. Can you imagine turning
on your TV to find the left ear of your favorite actor looking different from his
right ear?
It is easy to make an FIR (finite impulse response) filter have a linear phase.
This is because the impulse response (filter kernel) is directly specified in the
design process. Making the filter kernel have left-right symmetry is all that is
required. This is not the case with IIR (recursive) filters, since the recursion
coefficients are what is specified, not the impulse response. The impulse
response of a recursive filter is not symmetrical between the left and right, and
therefore has a nonlinear phase.
Chapter 19- Recursive Filters 329
Sample number
-25 0 25 50
-0.10
-0.05
0.00
0.05
0.10
0.15
0.20
0.25
a. Impulse response
Frequency
0 0.1 0.2 0.3 0.4 0.5
-12
-8
-4
0
4
8
12
b. Phase
Sample number
-25 0 25 50 75 100
-0.5
0.0
0.5
1.0
1.5
2.0
c. Pulse response
Sample number
-25 0 25 50
-0.10
-0.05
0.00
0.05
0.10
0.15
0.20
0.25
d. Impulse response
Frequency
0 0.1 0.2 0.3 0.4 0.5
-96
-64
-32
0
32
64
96
e. Phase
Sample number
-25 0 25 50 75 100
-0.5
0.0
0.5
1.0
1.5
2.0
f. Pulse response
Sample number
-25 0 25 50
-0.10
-0.05
0.00
0.05
0.10
0.15
0.20
0.25
g. Impulse response
Frequency
0 0.1 0.2 0.3 0.4 0.5
-12
-8
-4
0
4
8
12
h. Phase
Sample number
-25 0 25 50 75 100
-0.5
0.0
0.5
1.0
1.5
2.0
i. Pulse response
Zero Phase Filter
Linear Phase Filter
Nonlinear Phase Filter
FIGURE 19-7
Zero, linear, and nonlinear phase filters. A zero phase filter has an impulse response that has left-right symmetry
around sample number zero, as in (a). This results in a frequency response that has a phase composed entirely of
zeros, as in (b). Zero phase impulse responses are desirable because their step responses are symmetrical between
the top and bottom, making the left and right edges of pulses look the same, as is shown in (c). Linear phase filters
have left-right symmetry, but not around sample zero, as illustrated in (d). This results in a phase that is linear, that
is, a straight line, as shown in (e). The linear phase pulse response, shown in (f), has all the advantages of the zero
phase pulse response. In comparison, the impulse responses of nonlinear phase filters are not symmetrical between
the left and right, as in (g), and the phases are not a straight line, as in (h). The worst part is that the left and right
edges of the pulse response are not the same, as shown in (i).
Amplitude Amplitude
Phase (radians)
Amplitude
Phase (radians) Phase (radians)
Amplitude Amplitude Amplitude
The Scientist and Engineer's Guide to Digital Signal Processing 330
y [n ] ' a0 x [n ] % a1 x [n %1] % a2 x [n %2] % a3 x [n %3] % þ
% b1 y [n %1] % b2 y [n %2] % b3 y [n %3] % þ
EQUATION 19-9
The reverse recursion equation. This is the
same as Eq. 19-1, except the signal is filtered
from left-to-right, instead of right-to-left.
Analog electronic circuits have this same problem with the phase response.
Imagine a circuit composed of resistors and capacitors sitting on your desk. If
the input has always been zero, the output will also have always been zero.
When an impulse is applied to the input, the capacitors quickly charge to some
value and then begin to exponentially decay through the resistors. The impulse
response (i.e., the output signal) is a combination of these various decaying
exponentials. The impulse response cannot be symmetrical, because the output
was zero before the impulse, and the exponential decay never quite reaches a
value of zero again. Analog filter designers attack this problem with the
Bessel filter, presented in Chapter 3. The Bessel filter is designed to have as
linear phase as possible; however, it is far below the performance of digital
filters. The ability to provide an exact linear phase is a clear advantage of
digital filters.
Fortunately, there is a simple way to modify recursive filters to obtain a zero
phase. Figure 19-8 shows an example of how this works. The input signal to
be filtered is shown in (a). Figure (b) shows the signal after it has been
filtered by a single pole low-pass filter. Since this is a nonlinear phase filter,
the left and right edges do not look the same; they are inverted versions of each
other. As previously described, this recursive filter is implemented by starting
at sample 0 and working toward sample 150, calculating each sample along the
way.
Now, suppose that instead of moving from sample 0 toward sample 150, we
start at sample 150 and move toward sample 0. In other words, each sample
in the output signal is calculated from input and output samples to the right of
the sample being worked on. This means that the recursion equation, Eq. 19-1,
is changed to:
Figure (c) shows the result of this reverse filtering. This is analogous to
passing an analog signal through an electronic RC circuit while running time
backwards. !esrevinu eht pu-wercs nac lasrever emit -noituaC
Filtering in the reverse direction does not produce any benefit in itself; the
filtered signal still has left and right edges that do not look alike. The
magic happens when forward and reverse filtering are combined. Figure (d)
results from filtering the signal in the forward direction and then filtering again
in the reverse direction. Voila! This produces a zero phase recursive filter.
In fact, any recursive filter can be converted to zero phase with this
bidirectional filtering technique. The only penalty for this improved
performance is a factor of two in execution time and program complexity.
Chapter 19- Recursive Filters 331
Sample number
0 25 50 75 100 125 150
-0.5
0.0
0.5
1.0
1.5
a. Original signal
Sample number
0 25 50 75 100 125 150
-0.5
0.0
0.5
1.0
1.5
b. Filtered
FIGURE 19-8
Bidirectional recursive filtering. A rectangular
pulse input signal is shown in (a). Figure (b)
shows the signal after being filtered with a
single pole recursive low-pass filter, passing
from left-to-right. In (c), the signal has been
processed in the same manner, except with the
filter moving right-to-left. Figure (d) shows the
signal after being filtered both left-to-right and
then right-to-left. Any recursive filter can be
made zero phase by using this technique.
Sample number
0 25 50 75 100 125 150
-0.5
0.0
0.5
1.0
1.5
c. Filtered
Sample number
0 25 50 75 100 125 150
-0.5
0.0
0.5
1.0
1.5
d. Filtered
Amplitude Amplitude Amplitude
Amplitude
How do you find the impulse and frequency responses of the overall filter? The
magnitude of the frequency response is the same for each direction, while the
phases are opposite in sign. When the two directions are combined, the
magnitude becomes squared, while the phase cancels to zero. In the time
domain, this corresponds to convolving the original impulse response with a
left-for-right flipped version of itself. For instance, the impulse response of a
The Scientist and Engineer's Guide to Digital Signal Processing 332
single pole low-pass filter is a one-sided exponential. The impulse response of
the corresponding bidirectional filter is a one-sided exponential that decays to
the right, convolved with a one-sided exponential that decays to the left. Going
through the mathematics, this turns out to be a double-sided exponential that
decays both to the left and right, with the same decay constant as the original
filter.
Some applications only have a portion of the signal in the computer at a
particular time, such as systems that alternately input and output data on a
continuing basis. Bidirectional filtering can be used in these cases by
combining it with the overlap-add method described in the last chapter. When
you come to the question of how long the impulse response is, don't say
"infinite." If you do, you will need to pad each signal segment with an infinite
number of zeros. Remember, the impulse response can be truncated when it
has decayed below the round-off noise level, i.e., about 15 to 20 time constants.
Each segment will need to be padded with zeros on both the left and right to
allow for the expansion during the bidirectional filtering.
Using Integers
Single precision floating point is ideal to implement these simple recursive
filters. The use of integers is possible, but it is much more difficult. There are
two main problems. First, the round-off error from the limited number of bits
can degrade the response of the filter, or even make it unstable. Second, the
fractional values of the recursion coefficients must be handled with integer
math. One way to attack this problem is to express each coefficient as a
fraction. For example, 0.15 becomes 19/128. Instead of multiplying by 0.15,
you first multiply by 19 and then divide by 128. Another way is to replace the
multiplications with look-up tables. For example, a 12 bit ADC produces
samples with a value between 0 and 4095. Instead of multiplying each sample
by 0.15, you pass the samples through a look-up table that is 4096 entries long.
The value obtained from the look-up table is equal to 0.15 times the value
entering the look-up table. This method is very fast, but it does require extra
memory; a separate look-up table is needed for each coefficient. Before you
try either of these integer methods, make sure the recursive algorithm for the
moving average filter will not suit your needs. It loves integers.