William,

The ear is not perfectly linear.

I'm fully aware of this, and to me it's a completely different issue.

I sometimes play percussion in a local symphony, and that includes orchestra
bells. When playing two notes very loudly I can easily hear the IM products
that are generated inside my ears. How close musically the notes are to each
other determines which low frequency beat note I hear. I agree that some of
the beating is between ultrasonic components. But this does not mean an
audio system needs to capture ultrasonics! Moreover, this IM effect is not
audible out in the audience, and even I hear it only when playing very
loudly.

To take this to the logical conclusion, the "ultrasonics matter" camp would
have us capture everything up to 48 or 96 KHz or even higher, and they'd
have us buy loudspeakers that can reproduce that high, merely so we can play
it really loud while standing next to the speakers so our ears can generate
IM distortion.

No thanks.

--Ethan

ya mean read for yourself...??



"Arny Krueger" wrote in message
...
Willow wrote:
thanks for the reply,


Is there science out there that supports human hearing past 20K?
...Sounds
like you believe that to be the case?

You can make a good argument that, even though pure tones over 20KC
are not audible, that the added bandwidth makes transients more
accurate. That is, though you might not be able to hear a 21 KC
note, having the 21 KC components on a 7 KC note may make an audible
effect.

Change the waveform, then for sure you change the spectral content. Above
500-1000 KHz, the ear is very spectral-centric.

I have not seen a good well-conducted study show that this is the
case, but I haven't seen anything definitive saying that it's not
the case either.

The proof is in the filtering:

http://www.pcabx.com/technical/low_pass/index.htm

Yeah, i have hear that theory before. It is actually easy to test for
with a few recorded transients @ 48 and up-sample to 96. Then record
same transients at 96, Blind test those results....Anyone game?

Been there done that, except at a variety of sample rates:

http://www.pcabx.com/technical/sample_rates/index.htm

What Dan is pointing out is that the only thing the higher sample
rates buy
you is ultrasonic response, and sometimes they buy you ultrasonic
response only at the penalty of degraded linearity. So if the
ultrasonic stuff is NOT audible, then there is therefore no reason
to use the higher rates.

With ultrasonic response comes lots of ultrasonic noises. For example the
recording of a Bat I just downloaded lately had an interesting
agglomeration of noise above 30 KHz that didn't seem to vary with the
bat.

Actually there are more trade off than those mentioned. Doesn't Dan
build a box that sample up to 96 and why not 192 or 384 etc....?

All you need is Audition, it resamples up to 100 MHz. Seems like high
that
would be high enough... LynxTWO series audio interfaces among others,
perform well at up 192 KHz sampling.

I have not read the white paper yet, I would assume he answers these
q's there.

Listen for yourself!

ya mean read for yourself...??



"Arny Krueger" wrote in message
...
Willow wrote:
thanks for the reply,


Is there science out there that supports human hearing past 20K?
...Sounds
like you believe that to be the case?

You can make a good argument that, even though pure tones over 20KC
are not audible, that the added bandwidth makes transients more
accurate. That is, though you might not be able to hear a 21 KC
note, having the 21 KC components on a 7 KC note may make an audible
effect.

Change the waveform, then for sure you change the spectral content. Above
500-1000 KHz, the ear is very spectral-centric.

I have not seen a good well-conducted study show that this is the
case, but I haven't seen anything definitive saying that it's not
the case either.

The proof is in the filtering:

http://www.pcabx.com/technical/low_pass/index.htm

Yeah, i have hear that theory before. It is actually easy to test for
with a few recorded transients @ 48 and up-sample to 96. Then record
same transients at 96, Blind test those results....Anyone game?

Been there done that, except at a variety of sample rates:

http://www.pcabx.com/technical/sample_rates/index.htm

What Dan is pointing out is that the only thing the higher sample
rates buy
you is ultrasonic response, and sometimes they buy you ultrasonic
response only at the penalty of degraded linearity. So if the
ultrasonic stuff is NOT audible, then there is therefore no reason
to use the higher rates.

With ultrasonic response comes lots of ultrasonic noises. For example the
recording of a Bat I just downloaded lately had an interesting
agglomeration of noise above 30 KHz that didn't seem to vary with the
bat.

Actually there are more trade off than those mentioned. Doesn't Dan
build a box that sample up to 96 and why not 192 or 384 etc....?

All you need is Audition, it resamples up to 100 MHz. Seems like high
that
would be high enough... LynxTWO series audio interfaces among others,
perform well at up 192 KHz sampling.

I have not read the white paper yet, I would assume he answers these
q's there.

Listen for yourself!

Willow wrote:

ya mean read for yourself...??

No, I mean listen for yourself.

More specifically:

(1) Go to:

http://www.pcabx.com/technical/low_pass/index.htm

http://www.pcabx.com/technical/sample_rates/index.htm

(2) download files

(3) listen to them on your computer DAW using one or more of the supplied
double blind listening test comparators

Willow wrote:

ya mean read for yourself...??

No, I mean listen for yourself.

More specifically:

(1) Go to:

http://www.pcabx.com/technical/low_pass/index.htm

http://www.pcabx.com/technical/sample_rates/index.htm

(2) download files

(3) listen to them on your computer DAW using one or more of the supplied
double blind listening test comparators

"Arny Krueger" wrote in message news:jKKdnZGocuJbODndRVn-

I do not belive that people hear much above 20KHz. The theories about
alfa waves are taking less than 30KHz. I accepted 40KHz as good enough
to provide huge margins, so 88.2KHz is fast enough sampling.

But let us say that we hear or feel 70KHz (which I DO NOT BELIVE).
Than I would expect microphone makers to start making 70KHz bandwithe
mics. Same for speaker makers. Why is it they are pushing ONLY the AD
and DA to provide 96KH audio (192KHz sampling)?

Yes, one can make a measurnment mic to go 100KHz. A tiny device, so
small that you have a real bad signal to noise ratio. There is a
tradoff between speed and accuracy. But I do not see a mic maker or
speaker maker push for much above 20KHz. Until such time that you can
have a clean and complete audio chain, you can not even perform a
"normal listning tests.

That is unless one wants to argue that the 20KHz speaker emitts high
frequencies. I will not be surprised by such arguments. Not after
seing so many advocating 192KHz converters in a world of 20KHz mics
speakers and ears. Remember, it is the lowest bandwidth device that
dictates the overall outcome. The weakest link in the chain...

Sorry if I sound a bit stuffy. I have been talking against 192 on 4
sites.

BR
Dan Lavry
Lavry Engineering

"Arny Krueger" wrote in message news:jKKdnZGocuJbODndRVn-

I do not belive that people hear much above 20KHz. The theories about
alfa waves are taking less than 30KHz. I accepted 40KHz as good enough
to provide huge margins, so 88.2KHz is fast enough sampling.

But let us say that we hear or feel 70KHz (which I DO NOT BELIVE).
Than I would expect microphone makers to start making 70KHz bandwithe
mics. Same for speaker makers. Why is it they are pushing ONLY the AD
and DA to provide 96KH audio (192KHz sampling)?

Yes, one can make a measurnment mic to go 100KHz. A tiny device, so
small that you have a real bad signal to noise ratio. There is a
tradoff between speed and accuracy. But I do not see a mic maker or
speaker maker push for much above 20KHz. Until such time that you can
have a clean and complete audio chain, you can not even perform a
"normal listning tests.

That is unless one wants to argue that the 20KHz speaker emitts high
frequencies. I will not be surprised by such arguments. Not after
seing so many advocating 192KHz converters in a world of 20KHz mics
speakers and ears. Remember, it is the lowest bandwidth device that
dictates the overall outcome. The weakest link in the chain...

Sorry if I sound a bit stuffy. I have been talking against 192 on 4
sites.

BR
Dan Lavry
Lavry Engineering

dan lavry wrote:

But let us say that we hear or feel 70KHz (which I DO NOT BELIVE).
Than I would expect microphone makers to start making 70KHz bandwith
mics. Same for speaker makers. Why is it they are pushing ONLY the AD
and DA to provide 96KH audio (192KHz sampling)?

My speculation always has been they chose this one because this this is the
problem they can solve for negligable cost.

, one can make a measurnment mic to go 100KHz. A tiny device, so
small that you have a real bad signal to noise ratio. There is a
tradoff between speed and accuracy. But I do not see a mic maker or
speaker maker push for much above 20KHz. Until such time that you can
have a clean and complete audio chain, you can not even perform a
"normal listning tests.

There are still studio monitors in wide use that start rolling off at 16
KHz, and mics that start rolling off as low as 10 KHz.

How many people think their new 24/192 converters improve the sound of their
SM58s?

That is unless one wants to argue that the 20KHz speaker emitts high
frequencies. I will not be surprised by such arguments. Not after
seing so many advocating 192KHz converters in a world of 20KHz mics
speakers and ears. Remember, it is the lowest bandwidth device that
dictates the overall outcome. The weakest link in the chain...

....determines the strength of the whole chain.

Sorry if I sound a bit stuffy. I have been talking against 192 on 4
sites.

I've been speaking out against 96 since no later than Y2K. Check the
creation date on http://www.pcabx.com/technical/low_pass/index.htm .

dan lavry wrote:

But let us say that we hear or feel 70KHz (which I DO NOT BELIVE).
Than I would expect microphone makers to start making 70KHz bandwith
mics. Same for speaker makers. Why is it they are pushing ONLY the AD
and DA to provide 96KH audio (192KHz sampling)?

My speculation always has been they chose this one because this this is the
problem they can solve for negligable cost.

, one can make a measurnment mic to go 100KHz. A tiny device, so
small that you have a real bad signal to noise ratio. There is a
tradoff between speed and accuracy. But I do not see a mic maker or
speaker maker push for much above 20KHz. Until such time that you can
have a clean and complete audio chain, you can not even perform a
"normal listning tests.

There are still studio monitors in wide use that start rolling off at 16
KHz, and mics that start rolling off as low as 10 KHz.

How many people think their new 24/192 converters improve the sound of their
SM58s?

That is unless one wants to argue that the 20KHz speaker emitts high
frequencies. I will not be surprised by such arguments. Not after
seing so many advocating 192KHz converters in a world of 20KHz mics
speakers and ears. Remember, it is the lowest bandwidth device that
dictates the overall outcome. The weakest link in the chain...

....determines the strength of the whole chain.

Sorry if I sound a bit stuffy. I have been talking against 192 on 4
sites.

I've been speaking out against 96 since no later than Y2K. Check the
creation date on http://www.pcabx.com/technical/low_pass/index.htm .

"Ethan Winer" ethanw at ethanwiner dot com wrote in message ...
William,

The ear is not perfectly linear.

I'm fully aware of this, and to me it's a completely different issue.

I sometimes play percussion in a local symphony, and that includes orchestra
bells. When playing two notes very loudly I can easily hear the IM products
that are generated inside my ears. How close musically the notes are to each
other determines which low frequency beat note I hear. I agree that some of
the beating is between ultrasonic components. But this does not mean an
audio system needs to capture ultrasonics! Moreover, this IM effect is not
audible out in the audience, and even I hear it only when playing very
loudly.

To take this to the logical conclusion, the "ultrasonics matter" camp would
have us capture everything up to 48 or 96 KHz or even higher, and they'd
have us buy loudspeakers that can reproduce that high, merely so we can play
it really loud while standing next to the speakers so our ears can generate
IM distortion.

No thanks.

--Ethan

Hello kind folks.

I've been reading this thread for awhile, and this last message
(amongst others) has gotten me to the point of saying, "I'm despondent
as hell, and i can't take it anymore".


Let me start by saying that i'm not really speaking to any technical
issues involving the "96K vs. 192" question. Thanks to Dan Lavry, et.
al., I am well convinced that 96k would be the way to go for now.

What amazes me is how some folk here from the *scientific* camp seem
to be so out of date,as concerning the readilly available info and
research on sonics, audiology, the ear/brain system, etc. One person's
limited experience, related to others with further limited
understanding of the underlying phenomenea involved, do not make a
'scientific' contribution to the issue. To wit; "I can easily hear the
IM products generated in my ears". I must inform you here that the IM
phenomena spoken of emanated from the immediate accoustical space
surrounding the bells, long before any of that reached your ears. I am
in full agreement with you that it is unlikely that the audience in
question actually heard the resultant tone, as such. But neither does
that say that the composite tonal picture would or would not be
discernably altered to suchsaid audience. In various realms, both
Maurice Ravel and Brian Wilson understood these things very well.

The 'IM' question is one that i've seen only skimped over in this
whole discussion. This is somewhat surprising. As far as pitch
discernment, most of us over the age of 35 can't hear much above 17k,
true enough. It is also true that various harmonic content of musical
instruments have been measured up to 50k. It is also true that our
hearing, and our ear/brain systems respond to phenomenea outside of
the Fourier/sinusoidal paramaters. It is also true that our bodies and
our brains have been measurably responsive to various accoustical
stimuli that bypass the ear/brain system entirely. The Department of
Defense is well aware of this, as they have spent muchas dinero on
both audible and "audible bypass" weapons systems. I do digress,
sorry. Back to the IM question. Various and sundry partials (what
electronics folks refer to as 'harmonics') have a wonderful interplay
and dance with each other, in the real world. The subtraction part of
the 'sum and difference' thing comes into play, and the resultant
audibility, whether heard as individual 'tones', or more likely, as
having some overall timbral (tonal) effect, manefest themselves within
the pitch-discernable' region. For example, two partials of 22k and
34k combine IM-wise to give us a resultant of 12,000 cycles,
unarguably well within our range as to 'pitch discernment' issues, but
quite arguably, at an amplitude that could be 'measured' to be beyond
our capacity to hear it. But, and this is important,if you would
propose to be 'scientific' about the matter, you would do well to read
the most current scientific literature as to what is 'important' to
the ear/brain system. Sorry folks, but a few 'gotcha!', or snapping of
fingers in the face 'gimmie an answer! NOW!' tests won't cut it.

Piano tuners work by IM phenomenea all the time, by way of coincident
partials. It is beyond me to comprehend why a technique in use for
centuries seems to never be involved in the discussion of this topic.

Another aspect of this issue that i have not seen addressed at all
heretofore is the temporal discernment acuity of the human ear/brain
system. Well ok, some have, but only inadvertantly. For those of the
Fourier/sinusoidal camp, please allow me to point out that the human
ear has also been reliably measured as being capable of detecting time
differences as small as 5 microseconds. Inverting the math there, that
would translate to ~200kc. Were we to indulge in the "logic" (such as
it is) of the aforementioned F/s crowd there, we could be arguing that
we should be going to that standard accross the board. I'm fairly
certain in stating that 'we don't want to go there', at least not
anytime soon.

I should further state that those who would propose to artificially
align themseles to the *scientiific* side of things, whilst eschewing
any sense of the origins of real science, i.e. the long and extended
process of determining what the proper questions are to begin with,
would do well to read Bob Cain's et. al.'s indications and references
upon all this. Good gosh folks, the knowledge that there was more to
our ears than mere pitch discernment has been around for more than 25
years.

I regret to say that the primary basis of my knowledge was derived
~20-25 years ago. Allow me to relate an "old" story then. A group of
'senior citizens'
for whom it was determined that they could not 'hear' above 8'000
cycles, as determined by frequency tests, nevertheless could all
readilly determine when a 12k filter was invoked. Knowing what i know
now, i might ask as to the steepness of the filter, etc. But then
again, that shouldn't be a consideration amongst a good bit of the
crowd here. Who knows.

I am relatively new to computers, internet, and so forth. I don't wish
to weigh in much on the "96k vs. 192k" questions, or analog vs.
digital. And i ceartainly don't intend to 'slam', or 'flame' or 'bash'
or whatever the term, the honourable Ethan. Anyone who is involved in
non-reinforced accoustically rendered music is good in my book.

Oh well. I mistakenly thought that we had moved beyond the
'sinusoidal' or 'amplitude/frequency' thing 20 years ago.

Indeed, current technoledgy changes current perception.


JF

"Ethan Winer" ethanw at ethanwiner dot com wrote in message ...
William,

The ear is not perfectly linear.

I'm fully aware of this, and to me it's a completely different issue.

I sometimes play percussion in a local symphony, and that includes orchestra
bells. When playing two notes very loudly I can easily hear the IM products
that are generated inside my ears. How close musically the notes are to each
other determines which low frequency beat note I hear. I agree that some of
the beating is between ultrasonic components. But this does not mean an
audio system needs to capture ultrasonics! Moreover, this IM effect is not
audible out in the audience, and even I hear it only when playing very
loudly.

To take this to the logical conclusion, the "ultrasonics matter" camp would
have us capture everything up to 48 or 96 KHz or even higher, and they'd
have us buy loudspeakers that can reproduce that high, merely so we can play
it really loud while standing next to the speakers so our ears can generate
IM distortion.

No thanks.

--Ethan

Hello kind folks.

I've been reading this thread for awhile, and this last message
(amongst others) has gotten me to the point of saying, "I'm despondent
as hell, and i can't take it anymore".


Let me start by saying that i'm not really speaking to any technical
issues involving the "96K vs. 192" question. Thanks to Dan Lavry, et.
al., I am well convinced that 96k would be the way to go for now.

What amazes me is how some folk here from the *scientific* camp seem
to be so out of date,as concerning the readilly available info and
research on sonics, audiology, the ear/brain system, etc. One person's
limited experience, related to others with further limited
understanding of the underlying phenomenea involved, do not make a
'scientific' contribution to the issue. To wit; "I can easily hear the
IM products generated in my ears". I must inform you here that the IM
phenomena spoken of emanated from the immediate accoustical space
surrounding the bells, long before any of that reached your ears. I am
in full agreement with you that it is unlikely that the audience in
question actually heard the resultant tone, as such. But neither does
that say that the composite tonal picture would or would not be
discernably altered to suchsaid audience. In various realms, both
Maurice Ravel and Brian Wilson understood these things very well.

The 'IM' question is one that i've seen only skimped over in this
whole discussion. This is somewhat surprising. As far as pitch
discernment, most of us over the age of 35 can't hear much above 17k,
true enough. It is also true that various harmonic content of musical
instruments have been measured up to 50k. It is also true that our
hearing, and our ear/brain systems respond to phenomenea outside of
the Fourier/sinusoidal paramaters. It is also true that our bodies and
our brains have been measurably responsive to various accoustical
stimuli that bypass the ear/brain system entirely. The Department of
Defense is well aware of this, as they have spent muchas dinero on
both audible and "audible bypass" weapons systems. I do digress,
sorry. Back to the IM question. Various and sundry partials (what
electronics folks refer to as 'harmonics') have a wonderful interplay
and dance with each other, in the real world. The subtraction part of
the 'sum and difference' thing comes into play, and the resultant
audibility, whether heard as individual 'tones', or more likely, as
having some overall timbral (tonal) effect, manefest themselves within
the pitch-discernable' region. For example, two partials of 22k and
34k combine IM-wise to give us a resultant of 12,000 cycles,
unarguably well within our range as to 'pitch discernment' issues, but
quite arguably, at an amplitude that could be 'measured' to be beyond
our capacity to hear it. But, and this is important,if you would
propose to be 'scientific' about the matter, you would do well to read
the most current scientific literature as to what is 'important' to
the ear/brain system. Sorry folks, but a few 'gotcha!', or snapping of
fingers in the face 'gimmie an answer! NOW!' tests won't cut it.

Piano tuners work by IM phenomenea all the time, by way of coincident
partials. It is beyond me to comprehend why a technique in use for
centuries seems to never be involved in the discussion of this topic.

Another aspect of this issue that i have not seen addressed at all
heretofore is the temporal discernment acuity of the human ear/brain
system. Well ok, some have, but only inadvertantly. For those of the
Fourier/sinusoidal camp, please allow me to point out that the human
ear has also been reliably measured as being capable of detecting time
differences as small as 5 microseconds. Inverting the math there, that
would translate to ~200kc. Were we to indulge in the "logic" (such as
it is) of the aforementioned F/s crowd there, we could be arguing that
we should be going to that standard accross the board. I'm fairly
certain in stating that 'we don't want to go there', at least not
anytime soon.

I should further state that those who would propose to artificially
align themseles to the *scientiific* side of things, whilst eschewing
any sense of the origins of real science, i.e. the long and extended
process of determining what the proper questions are to begin with,
would do well to read Bob Cain's et. al.'s indications and references
upon all this. Good gosh folks, the knowledge that there was more to
our ears than mere pitch discernment has been around for more than 25
years.

I regret to say that the primary basis of my knowledge was derived
~20-25 years ago. Allow me to relate an "old" story then. A group of
'senior citizens'
for whom it was determined that they could not 'hear' above 8'000
cycles, as determined by frequency tests, nevertheless could all
readilly determine when a 12k filter was invoked. Knowing what i know
now, i might ask as to the steepness of the filter, etc. But then
again, that shouldn't be a consideration amongst a good bit of the
crowd here. Who knows.

I am relatively new to computers, internet, and so forth. I don't wish
to weigh in much on the "96k vs. 192k" questions, or analog vs.
digital. And i ceartainly don't intend to 'slam', or 'flame' or 'bash'
or whatever the term, the honourable Ethan. Anyone who is involved in
non-reinforced accoustically rendered music is good in my book.

Oh well. I mistakenly thought that we had moved beyond the
'sinusoidal' or 'amplitude/frequency' thing 20 years ago.

Indeed, current technoledgy changes current perception.


JF

John Fowler wrote:

Indeed, current technology changes current perception.

Not at all. When it comes to just listening, technology can't change the
basic performance of the human animal. We can reliably perceive what we are
equipped to perceive, nothing more.

People can cite all the theoretical studies they want to, but if people
can't hear the effects, on broadband recordings of music, of an accurate
brick wall filter @ 16 KHz, then that is that.

Here; I've made it as simple as I can:
http://www.pcabx.com/technical/low_pass/index.htm . Download the files and
listen for yourself.

John Fowler wrote:

Indeed, current technology changes current perception.

Not at all. When it comes to just listening, technology can't change the
basic performance of the human animal. We can reliably perceive what we are
equipped to perceive, nothing more.

People can cite all the theoretical studies they want to, but if people
can't hear the effects, on broadband recordings of music, of an accurate
brick wall filter @ 16 KHz, then that is that.

Here; I've made it as simple as I can:
http://www.pcabx.com/technical/low_pass/index.htm . Download the files and
listen for yourself.

John,

I must inform you here that the IM phenomena spoken of emanated from the
immediate accoustical space surrounding the bells, long before any of that
reached your ears ... Piano tuners work by IM phenomenea all the time

No, the IM products are *not* generated in the acoustic space, and that
points up a fundamental flaw in this line of reasoning. There is a key
distinction between IM products and the beats a piano tuner hones in on: IM
products are the result of non-linearity. In this case the non-linearity is
within the ears. Acoustic space is very much linear, except in certain cases
like two surfaces that almost touch so a loud sound makes them touch and
they buzz or rattle. Or some other such mechanically contrived situation.
Otherwise, sum and difference frequencies are not generated in the air or in
any other linear medium. If they were, every time I play a double stop on my
cello I'd get additional tones that are grossly out of tune with the music.

In the large picture, even if it can be shown that in certain special
circumstances some people can hear a little past 20 KHz, so what? To me it
makes no sense to establish CD/DVD standards that require 2 to 3 times more
data storage and bandwidth just so once a week 5% of people can say "I think
I hear a tiny change." Even sillier is raising the bandwidth to way past
what's necessary, then applying *lossy compression* because the audio can no
longer fit in the allotted space.

Even more to the point, the problems real recording and mixing engineers
face when trying to make tracks sound pleasing has nothing at all to do with
extended bandwidth. A truly great sounding mix will sound great whether you
low-pass it at 15 KHz, 12 KHz, or even 8 KHz. That people will obsess over
the importance of supersonic content while ignoring completely the horribly
skewed low frequency response in their listening rooms simply astounds me.

And i ceartainly don't intend to 'slam', or 'flame' or 'bash' or whatever
the term, the honourable Ethan. Anyone who is involved in non-reinforced
accoustically rendered music is good in my book.

Thanks. I never take any of this stuff personally, and I'm always a little
surprised when others do. I have only one goal - as Joe Friday would say,
"Just the facts ma'am." If we can't discuss this stuff in a civil manner,
progress is impossible.

--Ethan

John,

I must inform you here that the IM phenomena spoken of emanated from the
immediate accoustical space surrounding the bells, long before any of that
reached your ears ... Piano tuners work by IM phenomenea all the time

No, the IM products are *not* generated in the acoustic space, and that
points up a fundamental flaw in this line of reasoning. There is a key
distinction between IM products and the beats a piano tuner hones in on: IM
products are the result of non-linearity. In this case the non-linearity is
within the ears. Acoustic space is very much linear, except in certain cases
like two surfaces that almost touch so a loud sound makes them touch and
they buzz or rattle. Or some other such mechanically contrived situation.
Otherwise, sum and difference frequencies are not generated in the air or in
any other linear medium. If they were, every time I play a double stop on my
cello I'd get additional tones that are grossly out of tune with the music.

In the large picture, even if it can be shown that in certain special
circumstances some people can hear a little past 20 KHz, so what? To me it
makes no sense to establish CD/DVD standards that require 2 to 3 times more
data storage and bandwidth just so once a week 5% of people can say "I think
I hear a tiny change." Even sillier is raising the bandwidth to way past
what's necessary, then applying *lossy compression* because the audio can no
longer fit in the allotted space.

Even more to the point, the problems real recording and mixing engineers
face when trying to make tracks sound pleasing has nothing at all to do with
extended bandwidth. A truly great sounding mix will sound great whether you
low-pass it at 15 KHz, 12 KHz, or even 8 KHz. That people will obsess over
the importance of supersonic content while ignoring completely the horribly
skewed low frequency response in their listening rooms simply astounds me.

And i ceartainly don't intend to 'slam', or 'flame' or 'bash' or whatever
the term, the honourable Ethan. Anyone who is involved in non-reinforced
accoustically rendered music is good in my book.

Thanks. I never take any of this stuff personally, and I'm always a little
surprised when others do. I have only one goal - as Joe Friday would say,
"Just the facts ma'am." If we can't discuss this stuff in a civil manner,
progress is impossible.

--Ethan

Ethan Winer wrote:

...snips..

No, the IM products are *not* generated in the acoustic space, and that
points up a fundamental flaw in this line of reasoning. There is a key
distinction between IM products and the beats a piano tuner hones in on: IM
products are the result of non-linearity. In this case the non-linearity is
within the ears. Acoustic space is very much linear, except in certain cases
like two surfaces that almost touch so a loud sound makes them touch and
they buzz or rattle. Or some other such mechanically contrived situation.
Otherwise, sum and difference frequencies are not generated in the air or in
any other linear medium. If they were, every time I play a double stop on my
cello I'd get additional tones that are grossly out of tune with the music.


Ah yes, but air is not a linear medium! True, for most small amplitude
perturbations
the nonlinear effects are negligible but none the less still exist. As the SPL
is increased
the nonlinear effects become noticeable.

It's been years since I did any nonlinear acoustics work but here's a somewhat
simplified
version of some of the associated physics:
In a sound wave the density of the gas changes rapidly. Compressing a gas
results in
heating, and the velocity of sound is temperature dependent. The compressed part
of the
wave moves a little faster than the the rarefied portion resulting in wave front
distortion.
The higher the SPL the greater the distortion.

....that's enough for now, gotta set up for tonight's show.

Later...

Ron Capik
--

Ethan Winer wrote:

...snips..

No, the IM products are *not* generated in the acoustic space, and that
points up a fundamental flaw in this line of reasoning. There is a key
distinction between IM products and the beats a piano tuner hones in on: IM
products are the result of non-linearity. In this case the non-linearity is
within the ears. Acoustic space is very much linear, except in certain cases
like two surfaces that almost touch so a loud sound makes them touch and
they buzz or rattle. Or some other such mechanically contrived situation.
Otherwise, sum and difference frequencies are not generated in the air or in
any other linear medium. If they were, every time I play a double stop on my
cello I'd get additional tones that are grossly out of tune with the music.


Ah yes, but air is not a linear medium! True, for most small amplitude
perturbations
the nonlinear effects are negligible but none the less still exist. As the SPL
is increased
the nonlinear effects become noticeable.

It's been years since I did any nonlinear acoustics work but here's a somewhat
simplified
version of some of the associated physics:
In a sound wave the density of the gas changes rapidly. Compressing a gas
results in
heating, and the velocity of sound is temperature dependent. The compressed part
of the
wave moves a little faster than the the rarefied portion resulting in wave front
distortion.
The higher the SPL the greater the distortion.

....that's enough for now, gotta set up for tonight's show.

Later...

Ron Capik
--

Ron,

Ah yes, but air is not a linear medium!

Any nonlinearities of air are many orders of magnitude below the stuff I've
been addressing. In the overall scheme of things, air is linear enough that
cleanly reproduced music still sounds clean. The gross distortion you can
hear when playing orchestra bells loudly is not caused by the nonlinearity
of air. That high level of distortion happens entirely in the ears.

--Ethan

Ron,

Ah yes, but air is not a linear medium!

Any nonlinearities of air are many orders of magnitude below the stuff I've
been addressing. In the overall scheme of things, air is linear enough that
cleanly reproduced music still sounds clean. The gross distortion you can
hear when playing orchestra bells loudly is not caused by the nonlinearity
of air. That high level of distortion happens entirely in the ears.

--Ethan

dan lavry wrote:

I do not belive that people hear much above 20KHz. The theories about
alfa waves are taking less than 30KHz. I accepted 40KHz as good enough
to provide huge margins, so 88.2KHz is fast enough sampling.

But let us say that we hear or feel 70KHz (which I DO NOT BELIVE).
Than I would expect microphone makers to start making 70KHz bandwithe
mics. Same for speaker makers. Why is it they are pushing ONLY the AD
and DA to provide 96KH audio (192KHz sampling)?

I don't understand the marketing.

Hell, my ancient ATR-100 has a -3 dB point around 35 KHz, and I regularly
record with B&K 4133 microphones that are pretty clean up to around 40 KHz.
They sound good although I doubt the extended response has anything to do
with why they sound good.

Yes, one can make a measurnment mic to go 100KHz. A tiny device, so
small that you have a real bad signal to noise ratio. There is a
tradoff between speed and accuracy. But I do not see a mic maker or
speaker maker push for much above 20KHz. Until such time that you can
have a clean and complete audio chain, you can not even perform a
"normal listning tests.

It's not about whether something is audible, it's about whether it can
sell. I'm not sure how the marketing guys have convinced a large portion
of the population that high sampling rates are so important, but it's
worked. But since they don't understand what high sampling rates actually
do, I don't think they'll be in any way be motivated to buy other extended
bandwidth products without a totally different marketing campaign.

That is unless one wants to argue that the 20KHz speaker emitts high
frequencies. I will not be surprised by such arguments. Not after
seing so many advocating 192KHz converters in a world of 20KHz mics
speakers and ears. Remember, it is the lowest bandwidth device that
dictates the overall outcome. The weakest link in the chain...

Not mine. My monitors are down 3 dB at 16 KHz in my room, and probably fall
like a rock above 20 KHz.

But, the Tannoy guys have been remarkably good recently about selling
the Ellipse, with a supertweeter to reproduce ultrasonics. I can't tell
any difference with the supertweeter disabled or enabled (well, maybe it
sounds a little better with it disabled), but it's selling.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."

dan lavry wrote:

I do not belive that people hear much above 20KHz. The theories about
alfa waves are taking less than 30KHz. I accepted 40KHz as good enough
to provide huge margins, so 88.2KHz is fast enough sampling.

But let us say that we hear or feel 70KHz (which I DO NOT BELIVE).
Than I would expect microphone makers to start making 70KHz bandwithe
mics. Same for speaker makers. Why is it they are pushing ONLY the AD
and DA to provide 96KH audio (192KHz sampling)?

I don't understand the marketing.

Hell, my ancient ATR-100 has a -3 dB point around 35 KHz, and I regularly
record with B&K 4133 microphones that are pretty clean up to around 40 KHz.
They sound good although I doubt the extended response has anything to do
with why they sound good.

Yes, one can make a measurnment mic to go 100KHz. A tiny device, so
small that you have a real bad signal to noise ratio. There is a
tradoff between speed and accuracy. But I do not see a mic maker or
speaker maker push for much above 20KHz. Until such time that you can
have a clean and complete audio chain, you can not even perform a
"normal listning tests.

It's not about whether something is audible, it's about whether it can
sell. I'm not sure how the marketing guys have convinced a large portion
of the population that high sampling rates are so important, but it's
worked. But since they don't understand what high sampling rates actually
do, I don't think they'll be in any way be motivated to buy other extended
bandwidth products without a totally different marketing campaign.

That is unless one wants to argue that the 20KHz speaker emitts high
frequencies. I will not be surprised by such arguments. Not after
seing so many advocating 192KHz converters in a world of 20KHz mics
speakers and ears. Remember, it is the lowest bandwidth device that
dictates the overall outcome. The weakest link in the chain...

Not mine. My monitors are down 3 dB at 16 KHz in my room, and probably fall
like a rock above 20 KHz.

But, the Tannoy guys have been remarkably good recently about selling
the Ellipse, with a supertweeter to reproduce ultrasonics. I can't tell
any difference with the supertweeter disabled or enabled (well, maybe it
sounds a little better with it disabled), but it's selling.
--scott

--
"C'est un Nagra. C'est suisse, et tres, tres precis."

Ron Capik wrote:

Ah yes, but air is not a linear medium! True, for most small amplitude
perturbations the nonlinear effects are negligible but none the less
still exist.

We know that the ear is incredibly insensitive to nonlinear distortion,
compared to test equipment.

As the SPL is increased
the nonlinear effects become noticeable.

I see no evidence to support this exceptional claim.

It's been years since I did any nonlinear acoustics work but here's a
somewhat simplified version of some of the associated physics:

In a sound wave the density of the gas changes rapidly. Compressing a
gas results in
heating, and the velocity of sound is temperature dependent. The
compressed part of the
wave moves a little faster than the the rarefied portion resulting in
wave front distortion.

Seems like an overly elaborate explanation. In chemistry there is this
equation PV = NRT which suggests that at constant temperature, compression
of an ideal gas is linear. This would be called isothermal compression. To
explain the nonlinearity of air, you don't have to invoke any nonlinear
properties of air. All you have to do is observe that when compressed by
sound, the compression is not isothermal. When compressed acoustically, the
air heats up due to the compression, and it doesn't have the time or a ready
heat sink that would required to maintain itself at a constant temperature.

Here's another example of an explanation of this effect:

http://lungster.com/l/speakers/BassListArchive.html

This article makes the point that compared to many entities involved in the
construction of loudspeakers, the air is relatively linear.

The higher the SPL the greater the distortion.

Right, but without any quantification of the effect, this is just an
elaborately-described old wive's story. If you drink to much water it will
kill you. It's all about quantification.

Ron Capik wrote:

Ah yes, but air is not a linear medium! True, for most small amplitude
perturbations the nonlinear effects are negligible but none the less
still exist.

We know that the ear is incredibly insensitive to nonlinear distortion,
compared to test equipment.

As the SPL is increased
the nonlinear effects become noticeable.

I see no evidence to support this exceptional claim.

It's been years since I did any nonlinear acoustics work but here's a
somewhat simplified version of some of the associated physics:

In a sound wave the density of the gas changes rapidly. Compressing a
gas results in
heating, and the velocity of sound is temperature dependent. The
compressed part of the
wave moves a little faster than the the rarefied portion resulting in
wave front distortion.

Seems like an overly elaborate explanation. In chemistry there is this
equation PV = NRT which suggests that at constant temperature, compression
of an ideal gas is linear. This would be called isothermal compression. To
explain the nonlinearity of air, you don't have to invoke any nonlinear
properties of air. All you have to do is observe that when compressed by
sound, the compression is not isothermal. When compressed acoustically, the
air heats up due to the compression, and it doesn't have the time or a ready
heat sink that would required to maintain itself at a constant temperature.

Here's another example of an explanation of this effect:

http://lungster.com/l/speakers/BassListArchive.html

This article makes the point that compared to many entities involved in the
construction of loudspeakers, the air is relatively linear.

The higher the SPL the greater the distortion.

Right, but without any quantification of the effect, this is just an
elaborately-described old wive's story. If you drink to much water it will
kill you. It's all about quantification.

Arny Krueger wrote:

Ron Capik wrote:

Ah yes, but air is not a linear medium! True, for most small amplitude
perturbations the nonlinear effects are negligible but none the less
still exist.

We know that the ear is incredibly insensitive to nonlinear distortion,
compared to test equipment.

As the SPL is increased
the nonlinear effects become noticeable.

I see no evidence to support this exceptional claim.
...snips..

For the most part I was responding to Ethan's rather broad,
unqualified, unquanitfied statement:

"Acoustic space is very much linear, except in
certain cases like two surfaces that almost touch
so a loud sound makes them touch and they buzz
or rattle. Or some other such mechanically
contrived situation."

The thread at that point was addressing inter-modulation.
Helmlholtz observed inter-modulation (sum/difference frequencies)
resulting from interaction of pipe organ notes. The effect is strongly
dependent on interaction volume as well as frequency and sound
pressure level.

Anomalous harmonic content due to nonlinear interactions may well
exist in the sound field of an orchestra bell, or not. I don't know,
I haven't done the experiment. In ear distortion (saturation effects, etc.)
from being near said very loud bell likely overwhelm any anomalous
content due to nonlinear effects. The complexity of the bell tone in all
probability likewise overwhelms notice of any nonlinear contributions.

For a more extensive discussion of nonlinear acoustics see:
http://www.atcsd.com/pdf/HSSWHTPAPERRevE.pdf

I don't know that any of this has any bearing on the sampling rate discussion.
For me it's just a sidebar. My equipment (read: ears) isn't good enough to
join the experimental fray... :-(
Why 192kHz? ...because we can. ;-}


Later...

Ron Capik
--
[ I now return you to the battle in progress. ]

Arny Krueger wrote:

Ron Capik wrote:

Ah yes, but air is not a linear medium! True, for most small amplitude
perturbations the nonlinear effects are negligible but none the less
still exist.

We know that the ear is incredibly insensitive to nonlinear distortion,
compared to test equipment.

As the SPL is increased
the nonlinear effects become noticeable.

I see no evidence to support this exceptional claim.
...snips..

For the most part I was responding to Ethan's rather broad,
unqualified, unquanitfied statement:

"Acoustic space is very much linear, except in
certain cases like two surfaces that almost touch
so a loud sound makes them touch and they buzz
or rattle. Or some other such mechanically
contrived situation."

The thread at that point was addressing inter-modulation.
Helmlholtz observed inter-modulation (sum/difference frequencies)
resulting from interaction of pipe organ notes. The effect is strongly
dependent on interaction volume as well as frequency and sound
pressure level.

Anomalous harmonic content due to nonlinear interactions may well
exist in the sound field of an orchestra bell, or not. I don't know,
I haven't done the experiment. In ear distortion (saturation effects, etc.)
from being near said very loud bell likely overwhelm any anomalous
content due to nonlinear effects. The complexity of the bell tone in all
probability likewise overwhelms notice of any nonlinear contributions.

For a more extensive discussion of nonlinear acoustics see:
http://www.atcsd.com/pdf/HSSWHTPAPERRevE.pdf

I don't know that any of this has any bearing on the sampling rate discussion.
For me it's just a sidebar. My equipment (read: ears) isn't good enough to
join the experimental fray... :-(
Why 192kHz? ...because we can. ;-}


Later...

Ron Capik
--
[ I now return you to the battle in progress. ]

Ron,

For the most part I was responding to Ethan's rather broad,
unqualified, unquanitfied statement:

Okay, so what's the worst case IM distortion - caused only by the
nonlinearity of air - that you'll likely measure at "normal" volume levels?
The gross distortion I hear when playing bells very loudly is probably
around 20-50 percent. That is, the sum and difference tones seem about as
loud as the primary tones.

--Ethan

Ron,

For the most part I was responding to Ethan's rather broad,
unqualified, unquanitfied statement:

Okay, so what's the worst case IM distortion - caused only by the
nonlinearity of air - that you'll likely measure at "normal" volume levels?
The gross distortion I hear when playing bells very loudly is probably
around 20-50 percent. That is, the sum and difference tones seem about as
loud as the primary tones.

--Ethan

Ron Capik wrote:


For a more extensive discussion of nonlinear acoustics see:
http://www.atcsd.com/pdf/HSSWHTPAPERRevE.pdf

On page 4 we find the necessary quantification in the form of a chart
showing the nonlinearity in the compression of air.

The chart runs from 0 to 2 bars (a measure of pressure).

Then we recall from readhing micrphone spec sheets that 94 dB-SPL = 10
microbars.

1 bar is 100,000 times 10 microbars. IOW 1 bar is 100 dB above 10
microbars.

IOW, a sound whose amplitude is 1 bar is equivalent to a SPL of 194 dB.

'nuff said, eh?

Maybe not.

Moving right along to http://www.atcsd.com/pdf/HSSWHTPAPERRevE.pdf page 5
we see another practical demonstration of the nonlinear distoriton being
discussed. The stated SPL is 140 dB.

On page 10 we see mention of SPLs in the range of 130 to 140 dB.

On page 11 we see mention of 134.5 dB SPL.

On page 22 we see mention of 120 and 200 dB SPL.

On page 24 we see mention of 155 dB SPL.

On page 27 we see mention of 135 dB SPL.

Bottom line, it's highly unlikely that the nonlinearity of air is relevant
at so-called normal listening levels in the studio, control room, or home.

I'm glossing over a lot of interesting technology in-between. That
technology seems to relate to the problems of obtaining a satisfactory audio
signal by means of nonlinear distortion, and effciently obtaining
ultrasonic signals with sufficient acoustical amplitude to cause the
required nonlinear distoriton.

ATC's technology seems to be useful, but it doesn't justify the use of
higher sampling rates than 44.1 KHz for normal listening in the studio,
control room, or home. It involves SPLs that are something like 2 or more
orders of magnitude higher. We have to remember that while natural musical
acoustical signals have components above 20 KHz, they are only a fraction of
the total energy involved, and for a variety of physical reasons, they are
in a region where the power levels are naturally rolling off.

Ron Capik wrote:


For a more extensive discussion of nonlinear acoustics see:
http://www.atcsd.com/pdf/HSSWHTPAPERRevE.pdf

On page 4 we find the necessary quantification in the form of a chart
showing the nonlinearity in the compression of air.

The chart runs from 0 to 2 bars (a measure of pressure).

Then we recall from readhing micrphone spec sheets that 94 dB-SPL = 10
microbars.

1 bar is 100,000 times 10 microbars. IOW 1 bar is 100 dB above 10
microbars.

IOW, a sound whose amplitude is 1 bar is equivalent to a SPL of 194 dB.

'nuff said, eh?

Maybe not.

Moving right along to http://www.atcsd.com/pdf/HSSWHTPAPERRevE.pdf page 5
we see another practical demonstration of the nonlinear distoriton being
discussed. The stated SPL is 140 dB.

On page 10 we see mention of SPLs in the range of 130 to 140 dB.

On page 11 we see mention of 134.5 dB SPL.

On page 22 we see mention of 120 and 200 dB SPL.

On page 24 we see mention of 155 dB SPL.

On page 27 we see mention of 135 dB SPL.

Bottom line, it's highly unlikely that the nonlinearity of air is relevant
at so-called normal listening levels in the studio, control room, or home.

I'm glossing over a lot of interesting technology in-between. That
technology seems to relate to the problems of obtaining a satisfactory audio
signal by means of nonlinear distortion, and effciently obtaining
ultrasonic signals with sufficient acoustical amplitude to cause the
required nonlinear distoriton.

ATC's technology seems to be useful, but it doesn't justify the use of
higher sampling rates than 44.1 KHz for normal listening in the studio,
control room, or home. It involves SPLs that are something like 2 or more
orders of magnitude higher. We have to remember that while natural musical
acoustical signals have components above 20 KHz, they are only a fraction of
the total energy involved, and for a variety of physical reasons, they are
in a region where the power levels are naturally rolling off.

"Arny Krueger" wrote in message ...
John Fowler wrote:

Indeed, current technology changes current perception.

Contemporary perception would have been a better choice of words.

Not at all. When it comes to just listening, technology can't change the
basic performance of the human animal. We can reliably perceive what we are
equipped to perceive, nothing more.

So then, you equate 'perception' with 'performance'? Interesting.
Human performance has not changed to any significant degree for
thousands of years. Are you trying to say that our 'perception' has
"kept pace", as it were?
A big part of the reason that humans are what they are is that, unlike
other species, the evolution of our perception has far outpaced the
evolution of our physical performance.

Any type of ear trainig that i am aware of has historically been for
the express purpose of increasing conscious perceptual abilities, or
aural acuity, to be more precise.

The small group of people that gathered outside a theatre in the
1890's to hear the first rendition of a performance through passive
elecrtroacoustical means, said "it sounds just like the real thing!"
With AM radio, 5k cps bandwidth was deemed as plenty. We are stuck
with that one forever.

People can cite all the theoretical studies they want to,

Ok, that's a new one on me, "theoretical studies"? Most studies test
one theory to be proven or not. If the various tests prove or disprove
that theory, is it still a "theoretical" study? The tests themselves
are real, as far as i know.

but if people can't hear the effects, on broadband recordings of music, of an accurate
brick wall filter @ 16 KHz, then that is that.

That sentiment was prevalent when they set up the AM band limits, way
back when, and was actually "proven", by the criteria of that day.

Here; I've made it as simple as I can:
http://www.pcabx.com/technical/low_pass/index.htm . Download the files and
listen for yourself.

Well, my stereo is in storage for awhile yet,and these tiny computer
speakers would not be up to the task, i'm afraid. I'm sure that i will
find the tests useful, eventually. But, the little that i did
download, out of curiosity, made me wonder, "how scienticic is the
setting of parameters in the first place?"

To enumerate but a few considerations; what microphones are to be used
for the task at hand? Which mic technique vs. another? Who sets the
standard there? Ditto for the mic pre and all the rest of the
recording chain. Ditto again for the playback chain.

You have, indeed, made it as simple as you can, that's what good tests
for many things do, abitrarily define the variables so as to reduce or
eliminate a great number of other considerations, so as to be able to
isolate one thing we wish to focus on.
Not sure though, as regards sound recording,that what we might hear or
not on two tracks would measure up the same way accross 48 tracks,
plus mix down, plus mastering, plus printing, but, a useful test
nonetheless. Lets not mention 'plugins', even.

A car that reached 40 mph was considered to be nearly a permanent
limit at the time. The head of the US patent office seriously
considered shutting it down in the 1890's because, in his estimation
(*perception*?), there was nothing of significance more to be
invented.

However imperfectly worded, i stand by my earlier statement.

jf

"Arny Krueger" wrote in message ...
John Fowler wrote:

Indeed, current technology changes current perception.

Contemporary perception would have been a better choice of words.

Not at all. When it comes to just listening, technology can't change the
basic performance of the human animal. We can reliably perceive what we are
equipped to perceive, nothing more.

So then, you equate 'perception' with 'performance'? Interesting.
Human performance has not changed to any significant degree for
thousands of years. Are you trying to say that our 'perception' has
"kept pace", as it were?
A big part of the reason that humans are what they are is that, unlike
other species, the evolution of our perception has far outpaced the
evolution of our physical performance.

Any type of ear trainig that i am aware of has historically been for
the express purpose of increasing conscious perceptual abilities, or
aural acuity, to be more precise.

The small group of people that gathered outside a theatre in the
1890's to hear the first rendition of a performance through passive
elecrtroacoustical means, said "it sounds just like the real thing!"
With AM radio, 5k cps bandwidth was deemed as plenty. We are stuck
with that one forever.

People can cite all the theoretical studies they want to,

Ok, that's a new one on me, "theoretical studies"? Most studies test
one theory to be proven or not. If the various tests prove or disprove
that theory, is it still a "theoretical" study? The tests themselves
are real, as far as i know.

but if people can't hear the effects, on broadband recordings of music, of an accurate
brick wall filter @ 16 KHz, then that is that.

That sentiment was prevalent when they set up the AM band limits, way
back when, and was actually "proven", by the criteria of that day.

Here; I've made it as simple as I can:
http://www.pcabx.com/technical/low_pass/index.htm . Download the files and
listen for yourself.

Well, my stereo is in storage for awhile yet,and these tiny computer
speakers would not be up to the task, i'm afraid. I'm sure that i will
find the tests useful, eventually. But, the little that i did
download, out of curiosity, made me wonder, "how scienticic is the
setting of parameters in the first place?"

To enumerate but a few considerations; what microphones are to be used
for the task at hand? Which mic technique vs. another? Who sets the
standard there? Ditto for the mic pre and all the rest of the
recording chain. Ditto again for the playback chain.

You have, indeed, made it as simple as you can, that's what good tests
for many things do, abitrarily define the variables so as to reduce or
eliminate a great number of other considerations, so as to be able to
isolate one thing we wish to focus on.
Not sure though, as regards sound recording,that what we might hear or
not on two tracks would measure up the same way accross 48 tracks,
plus mix down, plus mastering, plus printing, but, a useful test
nonetheless. Lets not mention 'plugins', even.

A car that reached 40 mph was considered to be nearly a permanent
limit at the time. The head of the US patent office seriously
considered shutting it down in the 1890's because, in his estimation
(*perception*?), there was nothing of significance more to be
invented.

However imperfectly worded, i stand by my earlier statement.

jf

John Fowler wrote:
"Arny Krueger" wrote in message
...
John Fowler wrote:

Indeed, current technology changes current perception.

Contemporary perception would have been a better choice of words.

Not at all. When it comes to just listening, technology can't change
the basic performance of the human animal. We can reliably perceive
what we are equipped to perceive, nothing more.

So then, you equate 'perception' with 'performance'?

Not at all.

John Fowler wrote:
"Arny Krueger" wrote in message
...
John Fowler wrote:

Indeed, current technology changes current perception.

Contemporary perception would have been a better choice of words.

Not at all. When it comes to just listening, technology can't change
the basic performance of the human animal. We can reliably perceive
what we are equipped to perceive, nothing more.

So then, you equate 'perception' with 'performance'?

Not at all.

"Arny Krueger" wrote in message ...
John Fowler wrote:
"Arny Krueger" wrote in message
...
John Fowler wrote:

Indeed, current technology changes current perception.

Contemporary perception would have been a better choice of words.

Not at all. When it comes to just listening, technology can't change
the basic performance of the human animal. We can reliably perceive
what we are equipped to perceive, nothing more.

So then, you equate 'perception' with 'performance'?

Not at all.

If you can get enough people to agree with you that you did not
actually say what you actually said, then you need to run for office.

"Arny Krueger" wrote in message ...
John Fowler wrote:
"Arny Krueger" wrote in message
...
John Fowler wrote:

Indeed, current technology changes current perception.

Contemporary perception would have been a better choice of words.

Not at all. When it comes to just listening, technology can't change
the basic performance of the human animal. We can reliably perceive
what we are equipped to perceive, nothing more.

So then, you equate 'perception' with 'performance'?

Not at all.

If you can get enough people to agree with you that you did not
actually say what you actually said, then you need to run for office.