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Re: From Wayne Parham - Re: Patentable claim

You said

But I maintain the position that the Unity horn is not acting as an acoustic transformer for each of its drivers.
I use similarly-shaped, large format conical horns as wideband devices too. But I realize that the horn is only loading
the driver over a portion of its range, and certainly not loading the entire audio bandwidth.

If not acting as an acoustic transformer, what then explains the impedance curves which clearly show "radiation
resistance" and how then do the mid drivers reach an electroacoustic efficiency greater than the compression driver?
You are correct when you say a single driver will not load the horn over a wide frequency range and I have explained
several times about the horns low cutoff being a high pass filter who's frequency is a function of position in the horn.

I am not familiar with your stuff enough to know what horn you use or how big it is, but if you have one to spare, try
adding mid drivers forward of the compression driver (a place to start is where the horn area is about 10 sq. ins).

The point is that if one had a very large horn, that the lf cutoff is still dictated here by the expansion rate and that if
one wanted to use the horn at a lower frequency, one would have to place the drivers where the expansion rate was
appropriate for the frequency in question.
To date I have not made a horn which loads "the entire audio bandwidth" ( the 10 octave span from
20 Hz to 20 kHz ), the largest span is from about 200 Hz to 20 kHz but I am casually working on one which I hope
will cover 100 Hz to 20 kHz. Since the math says it should work, I don't see why it wouldn't but the issue is my
efforts are governed by what makes the most logical product.


you said

I also maintain the position that you cannot make this configuration time-linear or "unity summed." I am sure that you
have put a lot of work into time-aligning your drivers, but I insist that the time alignment method you chose is an
approximation, just like everyone else who uses this sort of technique.
Said another way, being an approximation, the device is still not acting as a point source.


You insist it won't work yet Energy vs time measurements show it does, you insist it can't be a point source yet polar
measurements show it is.
You insist it is an "approximation" while it was designed, built and fine tuned entirely based on actual measurements of
time and phase of the parts used.

You assert that by using even a fastwoofer like the 2226 which only has an internal delay equal to .64 feet (I just measured one)
and by using a crossover where signals above Xo come out significantly AHEAD in time of those coming out of the low pass, that
by moving the hf source still further ahead in time, that "this" fixes the time problem. Maybe fixes isn't what you mean since you do say the ear is not sensitive to time.
You say that by measuring this time discrepancy and accounting for it until the measurements show ONE time, that this is wrong somehow.

Picture the Indy 500, your suggesting that by the woofer being a full lap behind, that it is in the same phase and time
as the lead car at the finish. The woofer may be "phase" with the lead car but that is not the same as time, the woofer is still behind 1 lap.
A second issue is what does that misalignment sound like, it may be favorable, for example there is a Pro Sound
subwoofer called a bassmaxx that has the higher frequency direct radiation arriving ahead of the horn radiation by
about 1/2 wl at mid band (corresponding to an HF lead distance of about 10 feet). Some people really like what that does to the sound of music, it adds an "attack" not present when the sound is reproduced strictly according to the input signal.
Lets not confuse that with accuracy however.


I am not sure if you really don't understand or are posturing in defense of "your approach" that you refer to but if you really don't get it, explain at what point the concept is inconsistent with the physics and I will try to explain it further.

Also in another post you are vigorously slamming people about the notch you saw at 4 KHz on Nicks web site.
Each new design takes some time to evolve, at best I can only develop about 3 or 4 new speaker products a year
and even then I have been known to go back and change something.
For example our td-1, one of the products that
has been for sale the longest is having a new crossover shortly.

When the new horn shape that Nick used was made, it also took a bit to get the kinks figured out.
Where the mid holes are, there is a small impedance discontinuity (due to the change in cross section), making the
holes as small as possible makes this as small as possible.
At about 4 KHz, that location in the horn is governing the directivity in the 4 KHz area and that causes a reflection.
This reflection causes the dispersion pattern to widen temporarily at 4 kHz which because the acoustic power spread
over a wider angle produces a "hole" (the notch you saw) exactly on axis.
This notch is not present off axis fwiw.
The reflection can be eliminated (or at least its effects once the sound exits the mouth) by placing 4 small foam
absorbers at the right location on the mouth and then the dispersion and amplitude response at 4 kHz are "fixed"
without other impact.
Even when not "fixed" the effect is minor and a narrower slice in frequency than one can discern.

Again I would offer to send you some measurements if it really is an issue of understanding and I would urge you to
go to the speaker fest and hear them and think about it before you say much more publicly .

While I do have some computer programs which are not publicly available, there is at least one program capable of
this kind of design. It is called AKABAK and can model just about anything one can think of.
You might want to look into it .


Tom



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