Transcript: Smaart Impulse & Phase Measurement

Going up, the phase is also zero at the R min point in the impedance, this is where the series L in the VC is equal but opposite the mass reactance (C) and these two terms cancel out leaving the Rdc in series with the acoustic load and losses (a small R). So you see, you all ready have a “thing” which has a different delay for each frequency, a woofer and most speakers. At low frequencies, unwrapping the acoustic phase back to nominally zero degrees can be done without dsp and when done makes a wonderful sounding subwoofer. Unlike conventional woofers, the zero phase and flat response yields a system which CAN reproduce a complex waveshape.

The normal, non zero acoustic phase is the main thing which has stopped many attempts at active sound cancellation in its simplest form. I have spent a great deal of time working on speakers which had as little acoustic phase change over the widest frequency range possible as well and would also say that makes a significant audible difference.

On the other hand, I do everything with drivers, crossovers and horns and physical placement, partly because I want to actually attack the real problem but also because I am not to hip actually working with dsp. I know it is possible to correct all the phase stuff this way too and there is at least one hifi dsp correction product which claims to do this, at least at the microphone location.This is one area where an efficient horn can have an edge, to the extent they are dominated by the acoustic load, a resistance, there acoustic phase is resistive about zero degrees (output pressure and input voltage coincide over a wide range of frequencies).

Reply posted by Chip on September 19, 2001
Tom, Two questions:
1) would you consider preparing a “idiots guide to LF phase”? I’m very interested in the phase artifacts caused by different types of boxes and venting / porting methods. I never realy considered this as such a huge factor in the differing performance of different types of boxes.
2) How would you best describe an all-pass filter, and it’s effects?
David, et all, please respond as well. This type of shared information is why we are here.

Reply posted by Nathan Butler on September 18, 2001
Whenever I think of this, I like to reference some simple mathematics. Bear with me… A pure tone can be described by:
1) cos(wt) where w = frequency, t = time

With a phase shift (1) becomes:
2) cos(wt + p) where p = phase in degrees

With a time delay, (1) becomes:
3) cos(w(t + td)) where td = time delay

As an example, let’s say w = 100, p = 90, and td = 0.9
(2) becomes cos(100t + 90)
(3) also becomes cos(100t + 90)

Now let’s make the frequency, w = 200
(2) becomes cos(200t + 90)
(3) becomes cos(200t + 180)

Essentially, a time delay yields similar results to a phase shift, except that a time delay increases the phase shift with frequency. Hope this helps.

Reply posted by Patrik Arnekvist on September 18, 2001
I gotta go and try if changing phase angle on my omnidrive to, say 90 deg changes the delay, I guess it will. but how does one come to the conclusion that here I need to shift the phase angle 90 deg? I can’t really inderstand why one would need to do that. But changing the delay must be different, cause the time of 90 deg at 5kHz is quite different than 90deg at 100 Hz..right? I’m getting a headache here, excuse me for thinking out loud.