Classic Virtual Analog Phase Shifters

To create a *virtual analog* phaser, following closely the design
of typical analog phasers, we must translate each first-order allpass
to the digital domain. Working with the transfer function, we must
map from
plane to the
plane. There are several ways to
accomplish this goal [365]. However, in this case,
an excellent choice is the *bilinear transform* (see §7.3.2),
defined by

where is chosen to map one particular frequency to exactly where it belongs. In this case, can be chosen for each section to map the

Thus, given a particular desired break-frequency , we can set

Recall from Eq.
(8.19) that the transfer function of the
first-order *analog* allpass filter is given by

where is the break frequency. Applying the general bilinear transformation Eq. (8.20) gives

where we have denoted the pole of the digital allpass by

Figure 8.25 shows the digital phaser response curves corresponding to the analog response curves in Fig.8.24. While the break frequencies are preserved by construction, the notches have moved slightly, although this is not visible from the plots. An overlay of the total phase of the analog and digital allpass chains is shown in Fig.8.26. We see that the phase responses of the analog and digital allpass chains diverge visibly only above 9 kHz. The analog phase response approaches zero in the limit as , while the digital phase response reaches zero at half the sampling rate, kHz in this case. This is a good example of when the bilinear transform performs very well.

In general, the bilinear transform works well to digitize feedforward
analog structures in which the high-frequency warping is acceptable.
When frequency warping is excessive, it can be alleviated by the use
of *oversampling*; for example, the slight visible deviation in
Fig.8.26 below 10 kHz can be largely eliminated by increasing
the sampling rate by 15% or so. See the case of digitizing the Moog
VCF for an example in which the presence of feedback in the analog
circuit leads to a delay-free loop in the digitized system
[481,479].

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Center for Computer Research in Music and Acoustics (CCRMA), Stanford University