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Re: From Wayne Parham - Re: Patentable claim

Posted by tomservo (M) on July 5, 2002 at 10:45:57

In Reply to: Re: From Wayne Parham - Re: Patentable claim posted by Wayne Parham on July 4, 2002 at 17:49:49:

> > Not having heard one, that would seem to be a remarkable call to make.

So, are you moving the discussion into the realm of the subjective, where we discuss what we think we hear
with our ears?
Or are we talking about the math involved to describe phase relationships and performance in the time domain.
Because the
former isn't something you could have used to impress the patent examiner.

No Wayne I was responding to "your" value judgment, passing judgment on a speaker and loudly proclaiming it on a
several public forums without ever even hearing it seemed like a bit much.

I see you have a forum for promoting your speakers that's nice, after looking through some of your posts I found another one which is puzzling
You assert that you are a little miffed at us that we (meaning me) would "copy" your approach and that the idea for
the Unity began as the result of a joke when you spent 20 years in the oven with your pi..
You must imagine the world revolves around you I guess and that no one has ever given any of this though before you.

There is a loudspeaker world outside of yours where even they contemplate these issues, do a search and see how
companies like Meyer, EAW, Reinkus Heinz, Nexo and others deal with this stuff.
Bottom line and publicly, I am saying there is no way anything about the Unity evolved with any awareness of your approach and many of my ideas, come in a flash followed by a tail of details much like the impulse response of a line source (followed by a great deal of work to make it work). BFD

Looking through your crossover paper I see you are in fact familiar with Spice etc.
You go through a long process showing all the circuit modeling but I did not see any indication that you actually
measured the acoustic results or are familiar with the time delays within the drivers.
Further, you go on to assert that people are "not sensitive to time" and so your arrangement of placing the hf AHEAD
in time is cool.

You also say

Crossover phase also represents a difference in time between the two diaphragms. Crossover phase shows when the signa
l is generated by each motor, and the graph shown describes all reactive effects – including speaker driver characteristics.

The graph plotted literally shows when each diaphragm makes a sound.

This is incorrect, one has to account for the intrinsic delay in each driver and other fixed delays only then do you see
the phase response of the driver.
I wonder in fact if you even have a way to measure time or real acoustic phase.

> > one can move the hf source even more foreword and find a spot
> > where the two outputs are temporarily in phase but the "Time" is
> > skewed and an impulse would still show the hf leading.

That's the key here. "Temporarily" That's the point of all this. Moving the offset forward or back, either one is
still a
compromise. Your defense of the time alignment technique you've used only makes my case even more clear.
There is no
way to truly time align two different point sources at every frequency and at different positions with techniques
as simple as
position offset or the use of single-node filters. You can make them aligned at one frequency, and at one position
and that is
all.

Again, it would seem that you do not have a good time measuring device at your disposal and given the choice of
making time and phase correct at crossover VS making the phase right but time off N 1/2 cycles (by moving the hf in
front of the lf) seems not like the right way to me but the wrong way.

Having the phase right does make for a good amplitude response but making the time wrong insures poor impulse response. Granted in your design you have no way (other than an electronic delay like a BSS366) to arrive at the correct "time", the Unity is simply a way to do it with out any extra electronics.

You could aproximate the correct situation on one of your speakers or for that matter anyone with a multiway horn system (assuming you have the hf driver able to move front to back) and all you need is an oscillator.
Set the oscillator at your crossover point, where the sound is "equal" between the upper/lower drivers, adjust the
sound level to comfortable and then reverse the phase of the hf (just temporarily). This should put a deep null "on
axis", this null is much narrower and easier to hear than the broad addition hump in the "right phase".

To compensate for the hf signal coming out of the crossover first and the LF driver having its own significant delay
and phase lag, move the hf unit to the rear of the lf driver by about 1/4 wavelength at crossover (at the half and half
subjective amplitude point the crossover phase shift is usually in the neighborhood of 90 degrees).
Move the driver back and fourth some to get the deepest notch on axis ( at the listening position).
If there is no notch, then reverse the phase again as this would mean that the previous forward offset was N
wavelengths instead of 1/2 wl's.
Once the distance physical distance is about 1/4 wl or a little more (depending on the LF driver's delay it may be
significantly more than 1/4 wl), then switch the hf driver to "the right phase" (additive) and both phase AND time
should be approximately correct.
Even without a TEF machine and proper crossover slopes, this can give decent results (but should be done outside for best) compared to out of time.
Since it would be so easy to try, I would suggest to anyone with multiple horn systems to try this.

I think Davies says it well, "It is one of the fundamental laws of linear systems that if their output depends only on
previous input - that is, if they cannot see into the future - then the phase response is completely determined by
the
amplitude response."

That being true, it is still a mystery why you think that spreading out a complex signal in time such as your hf forward
alignment does would be superior to an approach that doesn't.
By the way, your assertion above regarding minimum phase systems (where the phase response is completely determined by the amplitude response) is only true for a single, linear, point source driver operating well below breakup.
As soon as two or more sources are involved (in differnet locations or delays) or are non linear or have changing directivity the system is not minimum phase.
A speaker system like you describe would measure a large change in acoustic phase in the transition between the upper and
lower sections even though the amplitude could be perfectly flat.
Unfortunately at least some of the MLS measurement systems have a potential flaw in that they calculate acoustic phase based on amplitude using a hilbert transform (the assumption that you refer to, that the system is linear and minimum phase) .
While the average user never notices, that the acoustic phase measurement is often very different than one taken
where the phase is actually measured (as in a TDS system).
When dealing with acoustic phase (like where one has two sources adding together) only the real measurement of
acoustic phase is helpful.

On still a different post you go on to tell someone else that only the mid drivers are horn loaded, all I can say is your
a lot of magnitude but no vector on this (that is U.B Wrong)
The unity speaker has a compression driver at the apex (on all the products so far), the response of the horn/
compression driver is only changed by the presence of the small mid entry passages below about 1500 Hz which is in
the neighborhood of the crossover. Above that the power response changes little "with or without" the holes and so
for the compression driver raw, the sensitivity is about 116 dB 1w 1m at 2 KHz. As the horn is very much a constant
directivty device, its high end response looks just like the Plane Wave Tube response for the compression driver
(which reflects the drivers actual acoustic power).
As the power on all compression drivers rolls off way before the hf limit (a typical 1" rolls off at about 2-2.5 KHZ
and a tad 2001 at about 4 KHz), here the real maximum sensitivity is set by the desired HF cutoff.
In this case, the compression driver into a the 60X60 degree horn is at about 104-105 dB 1W 1M at 18 kHz so that
is neighborhood of the maximum sensitivity which can be realized with flat response and passive circuitry.
That sensitivity is set by the coverage angle of the horn and compression driver, like any other CD horn.

Tom

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