Discussion:
Re Re: Ambisonic Mic Comparison
(too old to reply)
Enda Bates
2017-06-24 17:07:15 UTC
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Hey Steve,

well actually I think our results match your impression to some extent. In our second listening test, the TetraMic was preferred to the Ambeo, particularly in terms of high frequency timbre. In terms of directional accuracy, our study did find the Ambeo to be slightly more accurate, but with the difference in capsule spacing that was expected.


Both the Ambeo and TetraMic were recorded with a MOTU 8m, with the stock Ambeo cables, and the PPAc cabling for the Tetra over a very short cable run, and yeah, the specific calibration was for sure used for the Tetra. In that scenario, the TetraMic recording was definitely noisier, purely due to the additional gain required. I've heard from quite a few people that given a high end preamp with sufficient clean gain, that's not so much of an issue.


Have you tried the latest version of the Ambeo A-to-B-format conversion plugin? The new filter sounds a lot better to my ears.

e


Steve
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Sampo Syreeni
2017-06-24 17:38:17 UTC
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Post by Enda Bates
In terms of directional accuracy, our study did find the Ambeo to be
slightly more accurate, but with the difference in capsule spacing
that was expected.
I glanced at the post, but don't seem to remember. Sorry about that. But
I just wanted to make sure: was the test double blind? You talk a *lot*
about what was to be expected, so that eliminating observer bias might
be doubly important.
Post by Enda Bates
Both the Ambeo and TetraMic were recorded with a MOTU 8m, with the
stock Ambeo cables, and the PPAc cabling for the Tetra over a very
short cable run, and yeah, the specific calibration was for sure used
for the Tetra.
In general, on a priori reasons, mics wouldn't be expected to be much
affected by cable considerations. Pretty much all professional mics of
today tend to be extremely low current devices, which means they aren't
too sensitive to resistance, and because of that, their proper cabling
is also rather thin, leading to low capacitance due to low effective
cross section between the cables, and low inductance because of close
lead spacing. Of course all that modulo shielding, but still.

Thus I think if the cabling needs to be mentioned, especially with high
end, broad diaphragm capsules, someone, somewhere did something Nasty.
It should be a given that with mics costing thousands of euros a piece
the cabling at least can be assumed to be beyond audible reproach.
Post by Enda Bates
In that scenario, the TetraMic recording was definitely noisier,
purely due to the additional gain required.
Is that because of smaller capsules, lower line levels, noise gain, or
what, you think?
Post by Enda Bates
I've heard from quite a few people that given a high end preamp with
sufficient clean gain, that's not so much of an issue.
How does that happen, precisely? I mean, analogue circuit wise speaking?
If the noise signal is there, no amount of even cleaner gain is going to
take it away. Input impedance issues might *generate* noise, true, but
then again what *is* the precise issue, here?

Inquiring minds want to know. :)
--
Sampo Syreeni, aka decoy - ***@iki.fi, http://decoy.iki.fi/front
+358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2
Steven Boardman
2017-06-24 17:59:39 UTC
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Hi Enda

Thanks for the update.

That all seems to make sense. I just wanted to confirm.

I haven’t been using the new Ambeo conversion plugin, so my 2 cents were based on the previous version. This may of skewed my perception! It’s good news anyway, as the top end needed eq’ing for my ears.
I have heard differences with different Tetramics, so there maybe discrepancies from one to the other. Especially when used out in the field, as mine sometimes suffers with humidity.
I mentioned the cable as there are quite a few different flavours, and I have noticed RF problems in some scenarios. It seems you have used the same as me, so it is comparable.
The Ambeo didn’t come with the extension cable originally, so that’s also why I asked. It doesn’t seem to suffer from RF so much anyway. It’s presumably lower noise because it is balanced at the mic.

I was pleasantly surprised at the accuracy of the H2n, and TBH for the price the sound isn’t to bad either! Shame you didn’t have any of the Brahma mics for test..

Best

Steve
Post by Enda Bates
Hey Steve,
well actually I think our results match your impression to some extent. In our second listening test, the TetraMic was preferred to the Ambeo, particularly in terms of high frequency timbre. In terms of directional accuracy, our study did find the Ambeo to be slightly more accurate, but with the difference in capsule spacing that was expected.
Both the Ambeo and TetraMic were recorded with a MOTU 8m, with the stock Ambeo cables, and the PPAc cabling for the Tetra over a very short cable run, and yeah, the specific calibration was for sure used for the Tetra. In that scenario, the TetraMic recording was definitely noisier, purely due to the additional gain required. I've heard from quite a few people that given a high end preamp with sufficient clean gain, that's not so much of an issue.
Have you tried the latest version of the Ambeo A-to-B-format conversion plugin? The new filter sounds a lot better to my ears.
e
Steve
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Gerard Lardner
2017-06-24 18:32:41 UTC
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Following on from Steve's comment, if it is of any interest to you Enda, I can offer an early Brahma (serial no 008), about 18 months old, and an old Oktava MK-012 D4 (maybe around 20 years old). I'm in Bray, so only about 11 miles from TCD.
I had some problems with the calibration files for the Brahma, but I had it recalibrated and now I am very satisfied with the result.
The Oktava was calibrated by Fons Adriaensen. It produces a very good sound on orchestral and large-scale choral recordings (the only ones I have used it in conjunction with conventional microphone techniques, where I could make a comparison), but the very large array size - 48-49 mm radius - means that the directionality is noticeably quite poor. The current Oktava MK-012 D4 mount is different, more stylish; but I have no idea if the array size is any smaller. 
Gerard

Sent from my Samsung Galaxy smartphone.
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Fons Adriaensen
2017-06-24 22:09:04 UTC
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... In that scenario, the TetraMic recording was definitely
noisier, purely due to the additional gain required.
That doesn't make much sense.

Noise level (relative to signal) shouldn't increase with gain.
If it does that means the amplifier has a problem.
A well designed mic preamp will in fact have better EIN at
high gain.
I've heard from quite a few people that given a high end
preamp with sufficient clean gain, that's not so much of
an issue.
The Tetramic capsules have an acoustic noise level of 19 dB(A),
and sensitivity is 7 mV/Pa. That means the electrical noise
level is -116 dBm(A). If the EIN of the preamp is say 6 dB
or more better then most of the noise comes from the mic.
So with an EIN of -122 dBm(A) you should be safe, and I
wouldn't call that 'high end'.

The specs for the Motu 8m don't even mention EIN (which
isn't a good sign). The 'dynamic range' figure of 112 dB
can mean all sorts of things and is pretty useless.

Ciao,
--
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)
Sampo Syreeni
2017-06-24 23:34:07 UTC
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Post by Fons Adriaensen
... In that scenario, the TetraMic recording was definitely
noisier, purely due to the additional gain required.
That doesn't make much sense.
I think as much as well.
Post by Fons Adriaensen
Noise level (relative to signal) shouldn't increase with gain.
At least when noise level is well defined...
Post by Fons Adriaensen
The Tetramic capsules have an acoustic noise level of 19 dB(A), and
sensitivity is 7 mV/Pa. [...]
Enda didn't quantify what s/he meant by "noise" too well. You talk about
A-weighted measurables, then, while Enda probably talked about program
level overall noise and distortion, or something like that.

Could it be that you're just talking about different perceptual
weightings? I mean, if we talk about noise, there we shouldn't ever go
with A-weighting, or even C-weighting, but the ITU 468 curve. The one
which peaks as fuck between 2-6kHz, and explains how things like Dolby B
and C sliding band companders work so well; the one which also fails to
explain the loudness of impulsive, nonstationary, nonlinear noise, yet.

I mean, the perceptual cognates of lower quality in this test appeared
to be in precisely that frequency range.
Post by Fons Adriaensen
That means the electrical noise level is -116 dBm(A). If the EIN of
the preamp is say 6 dB or more better then most of the noise comes
from the mic. So with an EIN of -122 dBm(A) you should be safe, and I
wouldn't call that 'high end'.
I'd call that just "sane engineering for a sane gain structure".
Post by Fons Adriaensen
The specs for the Motu 8m don't even mention EIN (which isn't a good
sign). The 'dynamic range' figure of 112 dB can mean all sorts of
things and is pretty useless.
Precisely.

Funnily enough, I'm about to buy meself four speakers right about now.
For the first time. So that I could finally, eventually, at least do
some pantophonics for myself before I *die*. Compose for at least a
simple setup of four identical floor standing speakers, and whatnot; for
the very minimum of proper spatial reproductive rigs.

It's then amazingly difficult to get a rig amenable to the job. At my
rather low price point, it's almost impossible to get any numbers on how
your tentative loudspeakers behave. Pretty much no speaker manufacturer
wants to publish even such basic measures as impedance curves at
contact, driver thermal constants/dynamic compression time constants,
polar response plots, waterfall plots, crossover frequencies, phase
plots, and the thing.

Undoubtedly it's more complicated on the speaker side. But the
difficulty is manifest on the mic side as well: all of the measurables
dual to those of a speaker can in fact sometimes affect the performance
of a mic, and then even there they don't just tell you outright what
those measurables *are*. Then because of the unknowns, you might well
end up paying several thousands of euros extra, for nothing at all. Even
at the high end, which people here talk about... :/
--
Sampo Syreeni, aka decoy - ***@iki.fi, http://decoy.iki.fi/front
+358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2
Fons Adriaensen
2017-06-25 11:46:12 UTC
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Post by Sampo Syreeni
Could it be that you're just talking about different perceptual
weightings? I mean, if we talk about noise, there we shouldn't ever
go with A-weighting, or even C-weighting, but the ITU 468 curve.
Electrical noise from active mics and mic preamps is normally white
and Gaussian, except at low frequencies where you will typically
find some 1/f noise. If that is not the case there is something
seriously wrong (but see remark about A/B conversion below).

If the noise is white then it doesn't matter much which weighting
filter is used, as long as it is specified.

Compared to a flat 20 kHz bandwidth, the A-filter will typically
show around 2 dB less, and ITU-468 will give around 11 dB more.
Most manufacturers provide A-weighted measurements, these are
more or less the de-facto standard. Very few will give ITU-468
figures.
Post by Sampo Syreeni
The one which peaks as fuck between 2-6kHz, and explains how things like
Dolby B and C sliding band companders work so well; the one which
also fails to explain the loudness of impulsive, nonstationary,
nonlinear noise, yet.
Not sure what you mean by the latter. The ITU-468 method is quite
sensitive to impulsive noise - it was designed to be. That is mainly
not the result of the filter but of the very peculiar pseudo-peak
detector specified by the standard.

In the case the issue is complicated by the EQ which is part of
the A/B conversion. The W signal normally requires some boost in
the high frequency range, how much depends on capsule directivity
and the array radius.

My original post was triggered by one of the various "this can
perhaps be explained by" remarks in the web article - none of
them make much sense IMHO.

Another thing which triggered my scepticism neurons is this
'timbre' evaluation of the various mics. Small differences of
the 'dull vs bright' and 'thin vs full' kind can usually be
corrected by some gentle EQ, so I really doubt if any of
this is relevant in practice.

More after I've read the AES papers.

Ciao,
--
FA

A world of exhaustive, reliable metadata would be an utopia.
It's also a pipe-dream, founded on self-delusion, nerd hubris
and hysterically inflated market opportunities. (Cory Doctorow)
Sampo Syreeni
2017-06-26 05:31:23 UTC
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Post by Fons Adriaensen
Post by Sampo Syreeni
Could it be that you're just talking about different perceptual
weightings? I mean, if we talk about noise, there we shouldn't ever
go with A-weighting, or even C-weighting, but the ITU 468 curve.
Electrical noise from active mics and mic preamps is normally white
and Gaussian, except at low frequencies where you will typically find
some 1/f noise. If that is not the case there is something seriously
wrong (but see remark about A/B conversion below).
Correct, but then soundfield type mics are compound beasts with things
like differencing and capsule matching going on as well. Those can lead
to issues foreign to the natural noise characteristic of any monolithic
mic, like a highpass component on the noise floor of the first stage
amplifier, and (guessing on theoretical grounds only since I'm no
practitioner like you) perhaps most crucially high frequency ripple in
the noise floor (too), because of capsule spacing and phasing
imperfections.

The reason I brought up the ITU curve is that simply by reading from the
response curve, such anomalies would be belittled by A-weighting with
its rapid HF rolloff, while being accentuated by the brutal peak of the
ITU curve, smack in the center of the speech intelligibility band and
reaching up from it to the 6kHz range where matching issues first start
to show. At least with lower end soundfield kinda designs, with smaller
diaphragms and more, detailed, physical geometry between the capsules,
instead of the more symmetrical, large diaphragm design of the likes of
the original SoundFields.

(I'm grasping a bit here, but wasn't it so that the MkIV and MkV only
started to exhibit mismatch above 10kHz, where the ITU curve already
rapidly rolls off?)

Plus of course the ITU curve was designed with a noise measurement goal
in mind, unlike the A-weighting and C-weighting ones (their only
difference being in the reference SPL, both being linear weightings
derived from equal loudness contours of the Fletcher-Munson and later
Robinson-Dadson kind). I believe it really is Better wrt to what Enda
started with, on top of this thread.

But of course its derivation or at least its most common intended
application also leaves a lot to be desired. ITU-R BS.468-4 as it stands
doesn't really tell much about what is being measured or how, except
that it's noise at electrical signal levels, and that we measure it
using a particular analogue-implementable circuit consisting of a
passive filter plus a quasi-peak detector. I.e. 1) the filter definition
itself is a remnant of analogue era gone by, and perhaps most saliently,
2a) the reasoning behind the precise processing engendered by the filter
and 2b) its proper ambit of application are *thoroughly* obscured. Most
importantly we don't know whether the standard was meant to quantify
noise in an idle channel, by itself, or noise relative to a utility
signal present at the same time -- a huge difference even when you apply
the idea to analogue noise reduction, as the

it's been mostly
applied to environmental noise pollution, where the noise is the utility
signal under study, but also to the quantification of background noise
in the presence of a utility signal, such as in noise reduction. Those
two scenarios aren't interchangeable by a long shot, even if the curve
itself seems to perform rather better than its primitive elders in both
its roles; the extra nonlinear processing applied to the signal in the
base standard,
Post by Fons Adriaensen
If the noise is white then it doesn't matter much which weighting
filter is used, as long as it is specified.
Doesn't it though, when you want to translate a purely objective
measurement into perceptual noisiness? I thought that was the very
essence of why all of the weighting curves were conceived in general, in
the first place, and the ITU one in particular. (Notwithstanding the
rest of the processing which goes along with the curve.) I mean,
essentially any weighting curve is a frequency-wise decision of what
matters and what does not; where the curve peaks, you'll have the
frequencies which most contribute to the aggregate noise figure, and
where you have the most attenuation, you're essentially downgrading the
importance of noise over that band.

Given my argument from compoundness of spatial mic designs above, I'd
argue a weighting which peaks around the frequencies which lead to
soundfield typical aberrations is more sensitive to the perceptual
shortcomings of this class of mics.

Granted, I can't really argue that such a weighting would be better
except by half-coincidence: while the ITU-468 curve really was designed
for the measurement of confounding noise in the presence (implicitly?)
of a utility signal, instead of audibility of signals as such like the
A, C and whatnot weightings, and so is probably better in the role we're
currently discussing in any case...on the other hand the perfect
weighting and whatever extra nonlinear machinery we might deem necessary
in order to appraise directional mic accuracy probably *would* differ
broadly from everything we have in our measurement toolkit right now.
So, my invoking the ITU curve is just a half measure...but I think a
relevant half-measure still.
Post by Fons Adriaensen
Compared to a flat 20 kHz bandwidth, the A-filter will typically show
around 2 dB less, and ITU-468 will give around 11 dB more.
I'll take the mean as given. What I'd however be more interested in is
the relative variance, and in particular how it behaves over weighting
and the binary condition of a mic being of soundfield type versus being
a high grade (thus flat as can be, on-axis as far as the mic has one)
monophonic measurement mic of any kind. And of course in the end it'd be
nice to see such figures being correlated with rigorous perceptual work,
MUSHRA's and all.

I know something like that is a tall order, and asking for something
which *definitely* isn't available in the literature or the marketing
brochures right now. Also, I probably should have made my reasoning more
explicit to begin with, and noted that it might -- as usual -- go a bit
broadside with regard to the original discussion. So, my apologies, as
usual.

But it'd still be rather interesting to see what might come out of what
I mentioned above; I really don't think even the basic psychoacoustical
machinery of coincident mic design is too well developed as of date.
Perhaps that idea of relative variance over a four-field of discrete
choices, A-weighting/ITU-weighting, monophonic/periphonic, as correlated
to a MUSHRA score, might serve as a starting point for one more thesis
work? ;)
Post by Fons Adriaensen
Most manufacturers provide A-weighted measurements, these are more or
less the de-facto standard. Very few will give ITU-468 figures.
I'd argue, sadly so.
Post by Fons Adriaensen
Post by Sampo Syreeni
The one which peaks as fuck between 2-6kHz, and explains how things
like Dolby B and C sliding band companders work so well; the one
which also fails to explain the loudness of impulsive, nonstationary,
nonlinear noise, yet.
Not sure what you mean by the latter. The ITU-468 method is quite
sensitive to impulsive noise - it was designed to be.
Correct you are, once again. I often manage to to write precisely the
opposite of what I was thinking about. I really should take more care
with my output and flights of thought (fancy?) -- as you've in fact
pointed out a couple of times in the past already.

I also should point out that as far as implicit noise figures of a
coincident, spatial mic go, as here, I should have made it clearer that
I was referring only to the linear pre-equalization curve of the ITU
standard. Absent the further nonlinear detection machinery. As you say
below, that part of the standard is a mess in its own right.
Post by Fons Adriaensen
That is mainly not the result of the filter but of the very peculiar
pseudo-peak detector specified by the standard.
Yes. That step is highly suspect in the literature, in itself, wrt all
of the applications of the standard.

I think everybody will agree that something *like* it probably will need
to be in the measurement signal chain, if we want to account for the
particularities of impulsive noise. It's just that the ITU pseudo-peak
machinery is old as fuck, constrained by primitive analogue circuitry,
and quite possibly not too well thought out to begin with. Certainly
there's been as much critical commentary in the literature towards it as
there possibly could be, considering how peripheral to general audio
practice the field of noise metrology has over time proven to be.

Had I to guess, I'd also think we have a kind of anti-novelty bias at
work here: since few people know about the existence of the ITU
standard, as opposed to the more well known A-weighting and its ilk, we
tend to think the ITU thingy is somehow much newer. We think it it must
somehow represent a quantum leap over what came before, so that if it
doesn't, it must somehow be deficient.

In reality the ITU standard descends from 1970, which considering the
huge progress in audio theory and technology in recent decades seems
positively *ancient*. Sure, A-weighting was set in the thirties, so it's
much older, set against the human lifespan and the inevitable
generational shifts caused by it. But set against the background of
accelerating and at the same time *highly* uneven technological
progress...

In audio reproduction, the seventies were then *much* akin to the
thirties; especially on the practical side. Everybody already knew their
continuous time wave equations and what could be derived from them, all
of the analogue electronics equations were well know, and so on. So
compared to today, only one discrete revolution and one continuous
evolution really have made a difference: digital technology/the
explosion of computing power to back it up as the revolutionary kind,
and then materials plus modelling technology -- also speeded up by the
computer aided, digital side of things -- as the slow, difficult to
comprehend evolution. That which finally lets us have speakers, mics,
rooms, mixing consoles, whatever acoustical and electro-acoustical
implements, *finally* approximating the both the theoretical predictions
of pure acoustics, and the limits of psychoacoustical research, at the
same time.

So why would I rant as long as I have? Well, I think it's just idiotic
that we have to rely on even the ITU curve-or-standard when we talk
about noise measures and their perceptual import. Even the cutting edge
experimentalists and doers-shakers like you on the academic side and
Enda perhaps slightly on the more practical one, still rely on stuff
coming from the 30's and at most 70's. That's just insane, because
*especially* considering what we can now do with noise measurements,
after the empirical work which went into the development of first
analogue noise reduction plus especially after that into perceptual
noise shaping in A/D/A converters and lossy, digital, perceptual codecs,
we have an *abundance* of new, highly refined psychoacoustical theory at
our disposal. A veritable treasure trove of "stuff", applicable beyond
the wildest dreams of even the 80's engineer's imagination, to the very
problems we've always faced in trying to achieve verisimilitude of
reproduction.

So let's at least draw from that history as best we can. First go with
the ITU weighting where we want to measure noise. But of course then
also acknowledge that it's a piece of *shit* compared to what we could
now do, and as such task a doctoral student or two to derive something
much better. ;)
Post by Fons Adriaensen
In the case the issue is complicated by the EQ which is part of the
A/B conversion. The W signal normally requires some boost in the high
frequency range, how much depends on capsule directivity and the array
radius.
It does, but as I said above, matching issues higher up the frequency
range could lead to differencing, could lead to a high pass noise
characteristic. Which could be picked up by our ears worse even in the W
channel than you might first think, and especially in the XYZ+ channels
higher up.

In particular because *nobody* has at least to my knowing applied
directional unmasking theory to soundfield type mics or signal chains.
They do that even now wrt Parametric Stereo, in MPEG perceptual audio
coding work, but none of that hard, psychoacoustical, measured theory
seems to really be translating back into the basic mic or other
physical-electrical work as of now. It's all computational.
Post by Fons Adriaensen
My original post was triggered by one of the various "this can perhaps
be explained by" remarks in the web article - none of them make much
sense IMHO.
Agreed. The objections pretty much sounded like undisciplined guessing
or grasping for straws to me as well. The test setup seemed a bit
unconventional as well, with speakers above the mid plane and whatnot.

However, Enda's work otherwise seems to me to be a bit more in the
classical vein of experiential research. Not what you'd could call
whackery or snakeoilmanship by any measure, but genuine striving for
better sound, via disciplined empiricism. Informed by modern acoustical
theory, of course, but maybe a bit less constrained by its central dogma
than is usual.

I like it. I also believe that sort of approach is necessary, as an
adjunct to the more theoretical minded research you and many others
on-list and off do. If not else, then even because of what we've been
talking about wrt the A/468-distinction above: we all know and agree
that there are tradeoffs here, and we'd all like to understand them
fully. We think we do, but then there are still surprises on the way;
like the way neither of us can say too much, with too much certainty,
about what the hell the quasi-peak machinery of 468 actually does or
why. About what or how we could or should put in its place.

In that regard the qualitative, experiential research of Enda's kind is
at least to me of the first importance. It serves as the first signal in
the human science which psychoacoustics is, which leads to closer
scrutiny, and eventually to a better physical-psychological
correspondence in measurement. Especially since Enda does *not* just
speculate willy-nilly, but quite evidently grounds his speculative
musings into proper acoustical theory; as such gives rise to testable
hypotheses, to be tried out by those of you in the hard, physical,
empirical, measurement business.
Post by Fons Adriaensen
Another thing which triggered my scepticism neurons is this 'timbre'
evaluation of the various mics. Small differences of the 'dull vs
bright' and 'thin vs full' kind can usually be corrected by some
gentle EQ, so I really doubt if any of this is relevant in practice.
Exactly so. If I remember correctly, attempts at quantifying what people
hear as different timbres, and the tests at quantifying the transparency
of various transmission channels, *always* after factor analysis/PCA
arrive at the same results: the principal component in the spectral
domain consists of a nigh-linear spectral tilt, after compensation for
some near-Weber-Fechner-law. Integrated over the whole of the human
frequency passband, sensitivity to such average tilt is just
ridiculously high, so that it for example tends to dominate loudspeaker
and headphone preference to something like 1.1-1.3 sigma level.

(Sorry once again, I never, ever remember where I got my info; I'm
clinically unable to remember any references, or faces, or numbers, or
sometimes even my own name. So, take it with a grain of salt; it
shouldn't be too difficult to find the relevant studies, given you
prolly have access to all of the best periodicals already.)

The only real, attested to deviations from that idea/ideal of just the
spectral tilt governing all, are 1) speech formant like
characteristics,, i.e. waveguide-like resonances excited by
near-periodic waveforms with some nonlinearity so as to not *just*
"light up" the resonance using single harmonical series but having the
excitation be a bit more spread out as it is in human speach, 2) the
ridiculous sensitivity peak at 2-6kHz as attested to by the empirical
ITU BS.648 transfer function; believe it or not, even to date it pretty
much defies reduction to any basic psychoacoustical theory, and 3) the
unreasonable efficiency of the human hearing system to react to
wideband, binaural/dichotic onsets, and discern them beyond even high
static noise backgrounds.

If you doubt me here, just read through the perceptual audio coding
theory as a whole. All of the above has been explicitly taken advantage
of, there. Fully? I dunno. Probably the last, time-domain thingy is at
least a topic of contention. Especially since it has been cited as an
explanation for why wide bandwidths in digital audio of over 25kHz (cf.
ARA) could perhaps lead to better spatial resolution/spatiousness.

(BTW, Peter Craven seemed to provisionally buy into the argument, too.
As one of the Ambisonic masterminds. He once put out an AES paper about
the provisional benefits of minimum phase D/A reconstruction filters. I
don't really buy into that theory *per se*, but just as Craven, given
that we have extremely high sampling rates, arbitrary order digital
filters and reasonable lossless compression algorithms readily available
nowadays, I'd too advocate for wide bandwidths, slow rolloffs and
perhaps even for minimum phase reconstruction filters.

Because, what would you really lose? Nothing in time or frequency at
least, because of the *extreme* rates and filtering accuracies we
currently have. What might we gain? Well, unconditional freedom from
preringing. Which really *can*, at least in theory, be translated into
something nonlinearly hearable, even via your common speaker or
headphone. Thus, just to be sure...
Post by Fons Adriaensen
More after I've read the AES papers.
I'd really like to see your interpretation of them.

Not to mention, they really should go into the Motherlode. Somehow,
someone pirating them at their own peril, for communal benefit. I'm not
the one to say *you* should be the one to betray your licence with your
relevant publisher...except that I kind of am... ;)
--
Sampo Syreeni, aka decoy - ***@iki.fi, http://decoy.iki.fi/front
+358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2
David Pickett
2017-06-25 11:27:22 UTC
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Post by Sampo Syreeni
It's then amazingly difficult to get a rig amenable to the job. At
my rather low price point, it's almost impossible to get any numbers
on how your tentative loudspeakers behave. Pretty much no speaker
manufacturer wants to publish even such basic measures as impedance
curves at contact, driver thermal constants/dynamic compression time
constants, polar response plots, waterfall plots, crossover
frequencies, phase plots, and the thing.
In recent years, I have only seen such plots (and not all of them) in
reviews by Stereophile magazine, after seeing which, and listening to
in the shop, I bought four B&W DM603 S3 speakers for about $600 each
about 10 years ago. I am still very happy with them though, had I the
money and the space, I would be even happier wih their studio
monitors! Lower quality speakers that that I wouldnt expect to give
decent 4.0 results. But what price range are you looking at?
Post by Sampo Syreeni
Undoubtedly it's more complicated on the speaker side. But the
difficulty is manifest on the mic side as well: all of the
measurables dual to those of a speaker can in fact sometimes affect
the performance of a mic, and then even there they don't just tell
you outright what those measurables *are*.
No microphone known to me, at any price, has sufficiently defined or
credible specifications. You just have to use your ears, I am afraid!

David
Post by Sampo Syreeni
Then because of the unknowns, you might well end up paying several
thousands of euros extra, for nothing at all. Even at the high end,
which people here talk about... :/
John Leonard
2017-06-25 12:02:19 UTC
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Post by Fons Adriaensen
If it does that means the amplifier has a problem.
And that, in my experience anyway, is indeed often the problem. When I started using the Tetramic, it was with low-cost portable recorders and at the gain levels that I thought sensible for recording, the pre-amps were indeed adding noise: swapping to the Metric Halo ULN-8 made a big difference, as anyone who’s heard the recordings I’ve made with that combination will attest.

I’m not a MOTU user, so I can’t comment on how it behaves, but I’m very happy with the combination that I use.

Regards,

John


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