Robert Greene wrote:
> there are VERY serious problems of other kinds with using it at the kinds of 
>distances 
>
> (fractions of a meter less than 1/2 , much less often enough) where proximity
> effect becomes really major.

That is indeed true, except perhaps for the label of "Very".

I first noticed this in making measurements of soundfield microphones, not in 
analysis.  My measurement of the 'W' response showed proximity effect.  I 
believe that there are two reasons for this, both having to do with 
non-coincidence.  We model the outputs of the soundfield microphone array as 
'W', the sum of all the capsules (in the typical case) and X, Y, and Z, the 
differences between pairs of capsules.  These correspond to a coincident set of 
a monopole and three orthogonal dipoles at the center of the array.  We know 
that that model is not exactly accurate, especially in the high-frequency 
case.  
At frequencies where the distance across the array is a significant fraction of 
a wavelength there is a significant phase difference between the various 
capsule 
signals with the result that the frequency responses of the array change, and 
that they also are direction dependent.  One way of looking at that result is 
that the free-field and diffuse-field frequency responses begin to differ from 
each other at high frequencies.  This really happens fairly abruptly, with the 
diffuse-field omni response rolling off quite rapidly above 10 kHz for 
soundfield arrays with the typical 1.47 cm radius.

This is not a good thing.  What makes it tolerable is that the diffuse-field 
response of even a small (1/2") omni starts to roll off at just a few kHz.  By 
that measure a soundfield microphone is quite a good !

Likewise, we wish that the behavior of a soundfield microphone array were ideal 
at low frequencies, but it's not.  depending on the direction of arrival some 
of 
the capsules will be nearer to the source than others, so the level and the 
boost due to proximity effect is greater for the near capsules than for the far 
capsules.  I believe that this effect is more significant for 'W' than for the 
dipole outputs, basically because we expect the dipole outputs to have 
proximity 
effect but not the monopole output.

We should keep in mind that the near-field behavior of even conventional 
monopole (pressure) and dipole (figure eight) microphones is not ideal.  As I 
mentioned above, the DF response of omni microphones rolls off quite early.  
The 
DF response of figure-eight microphones tends to be a bit better, although 
still 
not ideal.  Also, the nearfield properties of conventional figure eight 
microphones isn't ideal.

If one were to need a soundfield microphone with ideal directional properties, 
it seems as though the only option is something like the Microflown.  But that 
has it's own set of problems.


----- Original Message ----
From: Robert Greene <gre...@math.ucla.edu>
To: Surround Sound discussion group <sursound@music.vt.edu>
Sent: Sat, July 23, 2011 5:45:48 PM
Subject: Re: [Sursound] the recent 2-channel 3D sound formats and their 
viability for actual 360 degree sound


I feel a little diffident in commenting on this in the presence of so many 
experts on the Soundfield mike in theory as well as in practice,
but unless I am misunderstanding how it works, there are VERY serious problems 
of other kinds with using it at the kinds of distances (fractions of a meter 
less than 1/2 , much less often enough) where proximity
effect becomes really major.

Namely, as I understand it, the way the B format signals are built is 
predicated 
upon the distances among the four capsules being quite small
compared to the distance of the source, for the following reason:
Compensation is needed for the fact that the capsules are on the faces of a 
tetrahedron, not coincident and all at the center. This compensation
is based on the fact that at reasonable distances to the source, the 
differences 
of the distances to the mikes is obtained by orthogonal projection on the axis 
of arrival of the sound(to a very good apporximation).

To make sense of this jargon, suppose a source is on the line that is equistant 
from three of the capsules.  Then its distance to those three
will always be the same, and if the source is reasonably far away the distance 
to the fourth capsule will be a constnat. This comes from the Pythagorean 
theorem limit case in effect: at large distances , the
difference between A to S and B to S is equal to the length of the projection 
of 
the line from A to B onto the line from A to S (or B to S these being parallel 
in the limit case).

If one does NOT have such large distance to the source, the variation of 
distances to the capsules will be extreme and also complicated.
Just think of how the distances to the four face centers of the tetrahedron 
will 
vary in odd ways when the source is close by!

So it seems to me(and I am prepared to be all wrong!) that
the Soundfield mike could not be expected to work at all well
except when the source is quite far away--a matter of meters, not
inches.  At close distances, there will be wild phase differentials among the 
four mike capsule outputs of a kind that depends on the distance
of the source from the center of the mike--something which the mike
does not "know" so that it cannot be compensated for.

Am I all wet here?

Robert

On Sat, 23 Jul 2011, dave.mal...@york.ac.uk wrote:

> Hi Folks,
>    I have an interesting question (well, I think it's interesting). The 
>Soundfield microphone, like any directional microphone, has a boosted bass 
>response to close sounds. When listening to this through a speaker rig, we 
>hear 
>this boost and tend to interpret it as meaning the sound is close especially 
>in 
>a dry acoustic with a Greene-Lee head brace etc., etc.,. However, surely 
>(unless 
>I am being more dense than usual tonight) this is a learnt response based on 
>the 
>behaviour we have heard from directional mics? After all, taken individually, 
>at 
>those sort of frequencies our ears are essentially omnidirectional and not 
>subject to bass boost (to anything like the same degree).
> 
> Any thoughts, anyone?
> 
>  Dave
> On Jul 23 2011, Dave Hunt wrote:
> 
>> Hi again,
>> 
>>> Date: Thu, 21 Jul 2011 21:01:41 +0300 (EEST)
>>> From: Sampo Syreeni <de...@iki.fi>
>>> 
>>> On 2011-07-21, Dave Hunt wrote:
>>> 
>>>> There is certainly no consideration of values outside the unit  sphere.
>>>> [...]
>>> 
>>> Correct, and we've been here before.
>> 
>> We certainly have.
>> 
>>> As BLaH points out, even the first
>>> order decoder handles distance as well as it possibly can. So does the
>>> SoundField mic on the encoding side.
>> 
>> The encoding and decoding are well matched. In some ways hardly surprising.
>> 
>>> But the classical synthetic
>>> encoding equation is for infinitely far away sources only, that is,
>>> plane waves. Running the result through a proper, BLaH compliant  decoder
>>> then reconstructs a simulacrum of such a plane wave, with first order
>>> directional blurring, spatial aliasing caused by the discrete rig, and
>>> the purposely imposed psychoacoustic optimizations overlaid on top of
>>> the original, extended soundfield. So in fact it's wrong to say  that the
>>> source is produced at the distance of the rig: instead it's produced
>>> infinitely far away, modulo the above three complications. (That is
>>> bound to be one part of why even synthetically panned sources localise
>>> so nicely even when listening from outside the rig.)
>> 
>> I have already admitted the error of my original statement. You're  right 
>> that 
>>POA assumes plane waves. The encoded signals are  reproduced at the distance 
>>of 
>>the loudspeakers. The shelf filters in  a BLaH compliant decoder are (as I 
>>understand it) an attempt to  compensate for the speakers finite distance, 
>>and 
>>that they don't  produce plane waves at the listener. This is often referred 
>>to 
>>as  'distance compensation'.
>> 
>>> If you want to synthetically encode a near-field source so to speak  "by
>>> the book", you'll have to lift the source term from Daniel, Nicol and
>>> Moreau's NFC work. I seem to remember it amounts to a first order  filter
>>> on the first order part of the source signal in the continuous domain,
>>> which you'll then have to discretize. (But don't take my word for it,
>>> it's been a while since I went through DN&R.)
>> 
>> Me too, but as I remember it tries to build the 'distance  compensation' 
>> into 
>>the encoding, and thus is dependent on the  distance of the loudspeakers. 
>>Thus 
>>the encoding is only suitable for  an identical or similar rig, and is not 
>>transferable to other rigs.  Amplitude/delay based systems such as WFS, Delta 
>>stereophony and  TiMax have similar problems. The encoding has to be matched 
>>to 
>>the  speaker rig.
>> 
>>>  Simply
>>> manipulating the relative amplitude or even the spectral contour  doesn't
>>> in theory cut it, though it's a cheap way to get some of the
>>> psychoacoustic effects of a nearby source.
>> 
>> Agreed that it is far from perfect, but this is obviously not a  trivial 
>>problem. What I'm suggesting is a fudge, though it can  produce simulations 
>>of 
>>sources both inside and outside the  loudspeaker radius which can be 
>>psychoacoustically effective, and are  transferable to different rigs.
>> 
>> We're still left with the "40 foot high geese" problem.
>> 
>>>  The only minor nit is that synthetic
>>> panning needs a bit more refinement for near sources that wasn't being
>>> handled by the older literature.
>> 
>> The "(potentially nasty) bass boost" you refer to is obviously a  problem. 
>> You 
>>could limit it from going extremely large at very small  distances, and 
>>ensure 
>>that  the output only went to 0dBFS maximum,  but this would require a huge 
>>dynamic range throughout the whole  system: large bit depth, good DACs, very 
>>quiet amplifiers etc..
>> 
>> If you could do the encoding assuming a given speaker distance, then modify 
>> the 
>>decoding for a different distance it might help, though  I've no idea how to 
>>do 
>>this.
>> 
>> Ciao,
>> 
>> Dave Hunt
>> 
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