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Application of the Bilinear Transform
The impedance of a mass in the frequency domain is
In the

plane, we have
where the ``a'' subscript denotes ``analog''. For simplicity, let's choose
the free constant

in the
bilinear transform such that

rad/sec maps to one fourth
the
sampling rate,
i.e.,

maps to

which implies

. Then the
impedance relation maps across as
where the ``d'' subscript denotes ``digital.
Multiplying through by the denominator and applying the
shift theorem
for

transforms gives the corresponding
difference equation
This difference equation is diagrammed in Fig. 7.16.
We recognize this recursive digital filter as the direct form I
structure. The direct-form II structure is obtained by commuting the
feedforward and feedback portions and noting that the two delay
elements contain the same value and can therefore be shared [449].
The two other major
filter-section forms are obtained by transposing the two direct
forms by exchanging the input and output, and reversing all
arrows. (This is a special case of Mason's Gain Formula which works
for the single-input, single-output case.) When a filter structure is
transposed, its summers become branching nodes and vice versa.
Further discussion of the four basic filter section forms can be found
in [333].
Figure 7.16:
A direct-form-I digital filter
simulating a mass
created using the bilinear transform
.
![\includegraphics[width=4in]{eps/lmassFilterDF1}](http://www.dsprelated.com/josimages_new/pasp/img1687.png) |
Subsections
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About the Author: Julius Orion Smith III
Julius Smith's background is in electrical engineering (BS Rice 1975, PhD Stanford 1983). He is presently Professor of Music and Associate Professor (by courtesy) of Electrical Engineering at
Stanford's Center for Computer Research in Music and Acoustics (CCRMA), teaching courses and pursuing research related to signal processing applied to music and audio systems. See
http://ccrma.stanford.edu/~jos/ for details.