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Multi-Way Conventional loudspeakers with crossovers

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Old 18th May 2010, 02:04 AM   #211
dlr is offline dlr  United States
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Originally Posted by twest820 View Post
Yep. dlr's proposal, however, was to work around the phase limitations by creating a driver model. Which, from the standpoint of implementing a digital crossover, is essentially inputting the inverse of the parametric equalization one would like and then using UE as a parametric equalizer. I would certainly say the thread's starting traverse ground already covered, but I'm afraid you've lost me in asserting it's circular to express a preference for using an equalizer directly.
It's not a work around, it's addressing the issue head on. If you insist on only using direct measurements with "imprecise" minimum-phase (that's what that is, there is no measuring precise absolute acoustic centers), that is, having some amount of excess-phase remaining due to the limitation of sample point uncertainty, then you will have phase "warp" as you call it. If you want results as close to linear phase as possible, you have to create a model accurately. There is no substitute. You can certainly end up with "good enough", though I suppose. It doesn't matter whether you use SE or the Allocator, both require accurate driver data for optimal results.

It's not inputting the inverse of the equalization, it's inputting the direct minimum-phase response of both drivers with a precisely determined amount of relative acoustic offset, then calculating the inverse phase required to linearize the response. Any driver input such as one that is measured using only sample-point resolution as you propose will always have some amount of excess-phase remaining. Then, of course, the higher the sampling rate the better, but it's not required. Phase correction will be most accurate only from a true minimum-phase response of each driver regardless of the sample rate. Nothing more, nothing less. How one gets to that point is the issue.

Dave
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Old 18th May 2010, 02:26 AM   #212
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Originally Posted by twest820 View Post
Thanks, but (as I've mentioned earlier in this thread) my laptop runs everything else just fine and has a very standard OS. Most of the SoundEasy crashes I've experienced are just the usual native code AVs raised by apps with bad memory management; entirely consistent with the observable quality of the program. Mostly I'm surprised it works as well as it does for other people.


John, these responses are unjustified. Could we keep this civil, please?


Well that is the way you come off. Sorry if I offended you. The problems you seem to be having seem linked to your laptop install. Instead of being surprised about how well it works for others maybe you should consider that what it does for others is closer to the norm and your experience is the exception.

Anyway, back to the issue excess phase. You are worried about +/- a fraction of a sample and the effect on phase at 20 K. But consider the reality of linearizing phase. If 1 sample is 180 degrees at 20k, it's 18 degrees at 2k, 1.8 degrees at 200 Hz and .18 degrees at 20 Hz. If I have a speaker with a 30 Hz B2 low frequency cut off then at 20 Hz there is roughly 120 degrees of phase rotation. Thus, to linearize phase to 20 Hz, assuming no additional rotation at high frequency, at 20 K there must be a delay of 120/0.18 = 667 samples. So what is the big deal about one sample more or less at 20 K?

You still seem overly concerned with minimum phase. It simple isn't an issue. Consistency is. It is of academic interest and little more.
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