I knew someone like Arny would jump in. Since I do this stuff for a living,
re-stating the obvious is torture.

"Gareth Hardy" wrote in message
...
Wrong. Since you admit you don't know what you are talking about, let me
tell you the basics.

Thanks, Arny. That's all I needed. The silence from buckaroo is deafening
(or is that a sampling rate problem).

I still don't really understand, but thanks for trying. My understanding
now
is that if decoded properly, the same WAV file (say a tuba) will sound the
same at 32KHz, 48Khz and 96KHz because having extra samples at low
frequency
will only smooth out to what the DAC would have done with fewer samples
anyway (I think).

"FDR" wrote in message

"Arny Krueger" wrote in message
...

Sasa, you've already shown what your attitude is with your claim that
www.pcabx.com is just a single page. It's a whole freakin' web site
that would educate you well, were you to actually lower yourself to
spend some time with it.

This says it all about that PAGE:

"This Page created 9/22/1999

This Page last updated 06/02/2003

(c) Copyright 1999, 2000, 2001 Arnold B. Krueger, All rights
reserved."

It's your PAGE and you even say it's a PAGE.

Yes, it's the site's home page and every page on the site is linked out of
it, directly or indirectly.

You've never heard of doing things this way before?

If you can't even remember what you wrote or what you have on the web
then what use are you?

I guess the idea that a web site has a home page and all other pages on the
site are linked out of it is way over your head.

But, for some odd reason, that's how most of the WWW world is made. Since
your powers of observation seem to be at a low ebb, you'll have to trust me
about this...

Arny,

I think these people are reading a lot of junk about perceptually based
music encoding and
crossing it into their conception of how PCM sampling works.

"Arny Krueger" wrote in message
...
"Jan Philips" wrote in message
news
On Thu, 26 Jun 2003 19:13:17 GMT, "FDR"
wrote:

Fourier transforms just transform from time domain to frequency
domain.

Yes, I realized that later, only the D/A is like a Fourier transform.

You can use Fourier to do processing, but for just digitizing an
analog signal a simple D/A is the way to go.

And a simple D/A doesn't use Fourier transform?

Nor does an A/D.

The FT would take the
data obtained form A/D and tell you which set of sine waves -
frequency and phase - (or any set of orthogonal functions) would
reproduce the original signal as closely as possible with the data you
have (subject to the Nyquist limit).

But you don't need sort of complexity that to just record, play back, and
transmit music.

From what I've been told, the D/A conversion interpolates a smooth
curve through the data points, not simply a stair-step.

The interpolation is generally done via a conceptually simple but
steep-sloped low-pass filter.

If it doesn't
use the FT then what does it do - a cubic interpolation or something
like that?

A conceptually-simple low pass filter suffices wonderfully. The low pass
filter gets a little complex when one tries to have good filtering at and
above Nyquist, but smooth response below. Nevertheless, its now usually
mostly done in the digital domain via oversampling, and has low analog
parts
count.

On Thu, 26 Jun 2003 16:38:50 -0400, "Arny Krueger"
wrote:

A conceptually-simple low pass filter suffices wonderfully.

Can you briefly explain what a "conceptually-simple low pass filter"
is and how it gets from digital to analog? I don't know much about
electronics but I do know that a low-pass filter lets low frequencies
through. But I don't see how that goes from digital to analog.

If the original is a sine wave (under the Nyquist limit), it is
digitally sampled, and a D/A conversion is done with the
conceptually-simple low pass filter, you get back a sine wave (not
some step approximation to a sine wave), right?

In mathematical information theory, info, as you call it, is that that
reduces uncertainty. Therefore, no information is lost
in digital sampling. You have the audiophiles erroneous belief that the
analog signal is somehow 'true' and 'complete' and the
digital signal is not. From the standpoint of mathematical information
theory, the digital signal is complete.

"FDR" wrote in message
.. .

"Lon Stowell" wrote in message
...
I do realize you are making this up as you go along.

Wrong is wrong and all your hand waving and avoidance won't
change the fact that you simply do not have a clue what you
are talking about.

As for the ears hearing analog, you may want to check a good
physiology reference or articles in Scientific American as
to how the ear really perceives sound and how the signals
from the inner ear [past the bones and drum] are passed
to the brain. Not that it has one single solitary thing
to do with whether or not players have filters to "fill in
the missing data points", but you are wrong on the ears as
well.

I've done a report on the ear for a class referencing medical books that
my
wife has (she's a physician). I know how the ear works, so don't act
condescending to me. I also know that everyday sounds are analog in
nature.
Sound pressure gets transduced though a microphone to an analog voltage,
then gets converted to binary and loses some info because of sampling and
other errors, get's reproduced to analog with some interpolation to
recover
some lost info and amplified to a usable signal that can be put through a
speaker to recreate sound pressure.

"Arny Krueger" wrote in message
...
"FDR" wrote in message

"Arny Krueger" wrote in message
...

Sasa, you've already shown what your attitude is with your claim that
www.pcabx.com is just a single page. It's a whole freakin' web site
that would educate you well, were you to actually lower yourself to
spend some time with it.

This says it all about that PAGE:

"This Page created 9/22/1999

This Page last updated 06/02/2003

(c) Copyright 1999, 2000, 2001 Arnold B. Krueger, All rights
reserved."

It's your PAGE and you even say it's a PAGE.

Yes, it's the site's home page and every page on the site is linked out of
it, directly or indirectly.

You've never heard of doing things this way before?

You call it a page, then you get ****ed at those that call it a page and not
a website. If you thought it was a website, then why don't you say so
yourself????

"Arny Krueger" wrote in message
...

Nice, call your fellowgroupmembers crapheads. Agression comes from
fear. Yo do not have to be affraid, sometimes people makes mistakes,
even you...

Sasa, you were pretending to be an expert and continue to do so.

Not really. I just stand behind my opinion as you do (and that simple
opinion is that 96k is better than 44.1k). And by doing so, I do not deserve
to be called a craphead...

Listen 24/96 sound cards aren't all that expensive any more.

Good ones are always expensive. And just maybe I live in Croatia where 300$
sound card is stated as luxury in gonverment laws!

However, I
use Echoaudio mia 24/96 which is more than a good soundcard.

Agreed. The Mia is entirely suitable for your education in this matter.

Now hitch that Mia to the files and programs you can download from
www.pcabx.com and get started for some basic lessons in audio audibility.

I have my ears, and don't need programs to test them. And concerning
lessons, I had mine more than enough...

It is false that 96kHz is better sounding than 44.1kHz?!

Yes.

Man, this is going to my archive to laugh occasionally...

Come back when you have had your ears opened at www.pcabx.com .

Now I saw FDR's post that owner of the page is actually you. No wonder you
are byting everyone who says differently. People have different opinions
about certain subjects. And in audio, sometimes you can be right or wrong.
As you believe I'm wrong, I believe you are too. So what?

Sasa, you've already shown what your attitude is with your claim that
www.pcabx.com is just a single page. It's a whole freakin' web site that
would educate you well, were you to actually lower yourself to spend some
time with it.

Do not catch me by a single word. pcabx IS a single web page/site/whatever.
I am here to learn from people and share knowledge. Not to try to convince,
by all means, somebody in his possible denial of truth as you are firmly
doing last couple of posts.

But Sasa, you obviously know it all. Who am I to try to dispel your
erroneous thinking?

You can try to defend your statements. Not by, as you energically did,
denying others...

Anyhow, my appologies if U felt insulted in any way. It was not intended.
And next time, please, have more patience...

"FDR" wrote in message
.. .

"Buckaroo" wrote in message
...
In mathematical information theory, info, as you call it, is that that
reduces uncertainty. Therefore, no information is lost
in digital sampling. You have the audiophiles erroneous belief that the
analog signal is somehow 'true' and 'complete' and the
digital signal is not. From the standpoint of mathematical information
theory, the digital signal is complete.

The math might be right, but that doesn't mean using it in the real world
makes sense or doesn't transfer easily.

Mathematically, we know E=mc^2, but that doesn't mean that making a
nuclear
bomb ie easy or that a theoretical bomb actually works.

Maybe I'm getting off track here, but I know what my ears and my common
sense tells me. I know that statistically, taking more data points gives
a
higher accuracy result. I believe that data lost can never be retrieved.
What do you have against higher sampling rates?




Sigh...at once you invoke both empiricism and mysticism. Can't argue with
that (laugh).

"Jan Philips" wrote in message

On Thu, 26 Jun 2003 16:38:50 -0400, "Arny Krueger"
wrote:

A conceptually-simple low pass filter suffices wonderfully.

Can you briefly explain what a "conceptually-simple low pass filter"
is and how it gets from digital to analog? I don't know much about
electronics but I do know that a low-pass filter lets low frequencies
through. But I don't see how that goes from digital to analog.

Ever hear of google?

Like Buckaroo I get tired of doing people's homework for them.

If the original is a sine wave (under the Nyquist limit), it is
digitally sampled, and a D/A conversion is done with the
conceptually-simple low pass filter, you get back a sine wave (not
some step approximation to a sine wave), right?

right.

Well do everyone a favour then, and keep quiet when you notice someone is
wrong. It's selfish twice over to just state something is "tripe" and then
wait for someone else to fill in the gaps. Some of us use UseNet to try and
learn.

I knew someone like Arny would jump in. Since I do this stuff for a
living,
re-stating the obvious is torture.

Thanks, Arny. That's all I needed. The silence from buckaroo is
deafening
(or is that a sampling rate problem).

"Buckaroo" wrote in message
...

"FDR" wrote in message
.. .

"Buckaroo" wrote in message
...
In mathematical information theory, info, as you call it, is that that
reduces uncertainty. Therefore, no information is lost
in digital sampling. You have the audiophiles erroneous belief that
the
analog signal is somehow 'true' and 'complete' and the
digital signal is not. From the standpoint of mathematical
information
theory, the digital signal is complete.

The math might be right, but that doesn't mean using it in the real
world
makes sense or doesn't transfer easily.

Mathematically, we know E=mc^2, but that doesn't mean that making a
nuclear
bomb ie easy or that a theoretical bomb actually works.

Maybe I'm getting off track here, but I know what my ears and my common
sense tells me. I know that statistically, taking more data points
gives
a
higher accuracy result. I believe that data lost can never be
retrieved.
What do you have against higher sampling rates?




Sigh...at once you invoke both empiricism and mysticism. Can't argue with
that (laugh).

If common sense and life experience are mysticism, then I'm guilty. If my
engineering background is empiricism, then I'm also guilty.

Once again, what do you have against higher sampling rates?

Oh, and here's what Dallas Semiconductor has to say :

"Now moving into the mathematical realm, assume the wheel is a unit circle
with sine and cosine coordinates. If one samples at the positive and
negative peaks of the cosine values (which are 180 degrees out of phase),
then the Nyquist criteria is met and the original cosine values can be
reconstructed from the two sampled data points. Thus the Nyquist limit is
essential in reconstructing the original signal. As more and more points are
added, the ability to replicate the original signal improves."

http://www.dalsemi.com/appnotes.cfm/appnote_number/928

Wow, looky there, more points = improvement.

On Fri, 27 Jun 2003 06:03:04 -0400, "Arny Krueger"
wrote:

If the original is a sine wave (under the Nyquist limit), it is
digitally sampled, and a D/A conversion is done with the
conceptually-simple low pass filter, you get back a sine wave (not
some step approximation to a sine wave), right?

right.

And although the Nyquist theorem applies to Fourier transforms, and
you said that the Nyquist theorem is at work here, the Fourier
transform is _not_ involved?

On Thu, 26 Jun 2003 13:41:17 -0700, Buckaroo wrote:

I knew someone like Arny would jump in. Since I do this stuff for a
living, re-stating the obvious is torture.

Then you'd be the perfect person to write up a web site on the technical
aspects of digital audio. Then whenever someone asks a question you can
point them to your web site. How does that sound?

--
Ian.

EOM

"Ian Hastie" wrote in message
news
On Thu, 26 Jun 2003 13:41:17 -0700, Buckaroo wrote:

I knew someone like Arny would jump in. Since I do this stuff for a
living, re-stating the obvious is torture.

Then you'd be the perfect person to write up a web site on the technical
aspects of digital audio. Then whenever someone asks a question you can
point them to your web site. How does that sound?

It would seem obvious, but this type of person doesn't seem interested in
teaching.


--
Ian.

EOM

On Fri, 27 Jun 2003 10:09:11 -0400, Jan Philips
wrote:

On Thu, 26 Jun 2003 14:00:22 -0700, "Buckaroo"
wrote:

In mathematical information theory, info, as you call it, is that that
reduces uncertainty. Therefore, no information is lost
in digital sampling.

With some conditions. If the signal can be broken down to pure sine
waves, none of them are above the Nyquist frequency, and the
measurements are absolutely precise.

Doesn't the Nyquist frequency also rely on using a sinc function for
reconstruction? It's been awhile since I've worked with this stuff
academically of professionally (hence keeping out of this thread except to
call someone on the still-false claim that DVD-Rs are cheaper or even as
cheap per unit storage than CD-Rs), but I seem to recall that perfect
reconstruction of a signal sampled at or above 1/2 its max frequency required
the use of a perfect sinc function, which is not so easy to replicate in real
life (and is usually approximated). IF my memory on this is correct, then it
seems likely that while information technically isn't lost in the ADC
process, it may be lost in the DAC process. Whether or not that loss (if
present) would be sufficient to be audible to an untrained (or even trained)
ear, I dunno.

Given that Arny and "Buckaroo" are so adamant about 44.1kHz sampling being
absolutely perfect (forgetting bit depth for the moment) for frequencies up
to about 20kHz, I'm definitely interested in their thoughts as to what I'm
recalling incorrectly.

--
Erik Harris
AIM: KngFuJoe http://www.eharrishome.com
Chinese-Indonesian MA Club http://www.eharrishome.com/cimac/

The above email address is obfuscated to try to prevent SPAM.
Replace each dollar sign with an "e" for the correct address.

Erik Harris writes:
On Fri, 27 Jun 2003 10:09:11 -0400, Jan Philips
wrote:

On Thu, 26 Jun 2003 14:00:22 -0700, "Buckaroo"
wrote:

In mathematical information theory, info, as you call it, is that that
reduces uncertainty. Therefore, no information is lost
in digital sampling.

With some conditions. If the signal can be broken down to pure sine
waves, none of them are above the Nyquist frequency, and the
measurements are absolutely precise.

Doesn't the Nyquist frequency also rely on using a sinc function for
reconstruction? It's been awhile since I've worked with this stuff
academically of professionally (hence keeping out of this thread

You and me both.

except to call someone on the still-false claim that DVD-Rs are
cheaper or even as cheap per unit storage than CD-Rs), but I seem to
recall that perfect reconstruction of a signal sampled at or above
1/2 its max frequency required the use of a perfect sinc function,
which is not so easy to replicate in real life (and is usually
approximated). IF my memory on this is correct, then it seems
likely that while information technically isn't lost in the ADC
process, it may be lost in the DAC process. Whether or not that
loss (if present) would be sufficient to be audible to an untrained
(or even trained) ear, I dunno.

Well, there's also that pesky matter of that band limiting filtering,
the required steepness of it to effectively eliminate aliasing
components, and the phase anomalies the filter presents in real life.

That (as I recall) is why practically oversampling is typically a Good
Thing--it relieved the burden of that perfect low pass filter, and
allowed it to be a) farther from the passband, b) less steep and c)
consequently able to introduce fewer phase anomalies than otherwise
possible.

Given that Arny and "Buckaroo" are so adamant about 44.1kHz sampling
being absolutely perfect (forgetting bit depth for the moment) for
frequencies up to about 20kHz, I'm definitely interested in their
thoughts as to what I'm recalling incorrectly.

Myself as well. There's a big difference bewteen the academics of the
Nyquist rate in discrete mathemtics theory, and the realities of the
componentry and actual filters one can manufacture and wrap their ears
around.


Best Regards,
--
Todd H.
http://www.toddh.net/

You need to learn about frames of reference. The quote is correct....but
you cannot hear the improvement, which is only an
improvement in the physical frame of reference. You can find
double blind tests that prove this conclusively if you will Goggle for them.

Engineers are the worst offenders and failing to understand the slightest
thing about psychoacoustics.

Quite true. The reason that the sample rate was set at 44.1 Ks/sec was to
allow for
filter roll off at the Nyquist at the upper limit of hearing, back when the
designs called
for analog output filters after the sample and hold. With improved digital
filtering,
oversampling, and other algorithms, it is a moot point now.

As an aside:

Most individuals in industrialized nations have severe attenuation of
hearing acuity under laboratory conditions above
ca. 16, 000 Hz. Another point to consider.

I only teach post docs.

"Buckaroo" wrote in message
...
You need to learn about frames of reference. The quote is correct....but
you cannot hear the improvement, which is only an
improvement in the physical frame of reference. You can find
double blind tests that prove this conclusively if you will Goggle for
them.

Wow, it took me a whole 5 seconds to find a website that says 96kHz is
better than 44.1 khz. Audibly better. And it's done by a mastering
engineer no-less.

http://www.audiomedia.com/archive/fe...teningtest.htm

Here is another article that sheds some light on the limits of human hearing
and how higher sampling rates can result in better sound:

http://www.ocf.berkeley.edu/~ashon/a...ltrasonics.htm


Engineers are the worst offenders and failing to understand the slightest
thing about psychoacoustics.

I feel so hurt.

"Buckaroo" wrote in message
...

Most individuals in industrialized nations have severe attenuation of
hearing acuity under laboratory conditions above
ca. 16, 000 Hz. Another point to consider.

Thats not the point of the discussion right now.

The point is that this topic became a flame, as it went from audio to totaly
mathematical/phisycal discussion. I pressume that nobody actually plugged
his/hers SACD player and heard for themselves. Or maybe recorded some stuff
and mixed in 44.1k and 96k, because of that everything of this was invented.
For the end, I can assure you, that my 96k mixes sound way much better than
pittiful 44.1k. If we already have a 96k, why the hell not use it, for
mother's sake!!!

End

Wow. Hopeless.

The first article is voodoo. Laughable .. it would be rejected at any
meeting of scientists as opinion.

The second? Sorry, but those data say 'Harmonics visible' .. Who cares?
Percussion harmonics go to 100 kHz +

What do you not understand about science, perception, and how they connect
and relate to engineering/acoustics?

The second article is a missive from an undergraduate student in a lower
division psychology class....give me a break.

"Buckaroo" wrote in message
...
Wow. Hopeless.

The first article is voodoo. Laughable .. it would be rejected at any
meeting of scientists as opinion.

The second? Sorry, but those data say 'Harmonics visible' .. Who cares?
Percussion harmonics go to 100 kHz +

What do you not understand about science, perception, and how they connect
and relate to engineering/acoustics?

"Sasa [Sason] Miocic" wrote in message
...

"Buckaroo" wrote in message
...

Most individuals in industrialized nations have severe attenuation of
hearing acuity under laboratory conditions above
ca. 16, 000 Hz. Another point to consider.

Thats not the point of the discussion right now.

The point is that this topic became a flame, as it went from audio to
totaly
mathematical/phisycal discussion. I pressume that nobody actually plugged
his/hers SACD player and heard for themselves. Or maybe recorded some
stuff
and mixed in 44.1k and 96k, because of that everything of this was
invented.
For the end, I can assure you, that my 96k mixes sound way much better
than
pittiful 44.1k. If we already have a 96k, why the hell not use it, for
mother's sake!!!

Because Nyquist says so!!!!!!!!!!!!!


End

"Jan Philips" wrote in message

On Fri, 27 Jun 2003 06:03:04 -0400, "Arny Krueger"
wrote:

If the original is a sine wave (under the Nyquist limit), it is
digitally sampled, and a D/A conversion is done with the
conceptually-simple low pass filter, you get back a sine wave (not
some step approximation to a sine wave), right?

right.

And although the Nyquist theorem applies to Fourier transforms, and
you said that the Nyquist theorem is at work here, the Fourier
transform is _not_ involved?

right.

"FDR" wrote in message

"Buckaroo" wrote in message
...
You need to learn about frames of reference. The quote is
correct....but you cannot hear the improvement, which is only an
improvement in the physical frame of reference. You can find
double blind tests that prove this conclusively if you will Goggle
for
them.

Wow, it took me a whole 5 seconds to find a website that says 96kHz is
better than 44.1 khz. Audibly better. And it's done by a mastering
engineer no-less.


http://www.audiomedia.com/archive/fe...teningtest/uk-
0400-listeningtest.htm


This page provides the following conclusions:

"This led to the following (preliminary) conclusions:
"1. A properly-designed 20kHz digital filter can be sonically invisible in a
96kHz sampled environment.

"2. Experience and this experiment suggests that 44.1kHz sampling digital
systems can sound much better simply by use of better digital filters. This
includes all the filters in compact disc players, A/Ds, etc. The effects of
cumulative filters must also be considered -
a situation similar to the familiar effects of group delay in successive
bandpass limited analogue circuits.

"3. 96kHz sampling systems do not sound better because of increased
bandwidth. The ear does not use information above 20kHz to evaluate sound.

IOW the site you cited says that 96 KHz is *NOT* audibly better than 44.1
KHz.


Here is another article that sheds some light on the limits of human
hearing and how higher sampling rates can result in better sound:

http://www.ocf.berkeley.edu/~ashon/a...ltrasonics.htm

It's just an article written by an undergraduate student who was taking a
psychology course.

Engineers are the worst offenders and failing to understand the
slightest thing about psychoacoustics.

I feel so hurt.

Listen for yourself at www.pcabx.com.

"Buckaroo" wrote in message
...
Wow. Hopeless.

The first article is voodoo. Laughable .. it would be rejected at any
meeting of scientists as opinion.

Here's the opininon of another:

"In the Handbook for Sound Engineers Steve Dove says anti-aliasing filters
"....exhibit serious frequency dependent delay and convoluted
frequency/phase characteristics... leaving mangled audio in their wake". He
also advocates sampling around 100 kHz, and says the result is a more open
and spacious sound. "

http://www.silcom.com/~aludwig/EARS.htm

But then again, what would a sound engineer know?

As for finding all these articles supporting your claim on Google, I didn't
find any.

"Arny Krueger" wrote in message
...


IOW the site you cited says that 96 KHz is *NOT* audibly better than 44.1
KHz.

"My experiences with 96kHz/24-bit sampled recording have been exceptional.
The sound is more open, transparent, and dynamic than previous 44.1kHz
recordings, with a purer mid range and more apparent depth and space closer
to the analogue source."




Here is another article that sheds some light on the limits of human
hearing and how higher sampling rates can result in better sound:

http://www.ocf.berkeley.edu/~ashon/a...ltrasonics.htm

It's just an article written by an undergraduate student who was taking a
psychology course.

Do you dispute this in his report:

"Additional work supporting James Boyk's findings was done by John Atkinson,
editor of Stereophile magazine, the preeminent audiophile journal. In his
October 2000 editorial he describes spectral analyses of audio recordings,
all of which demonstrate more or less activity above 20 kHz. An interesting
finding he reports is that it is not just acoustic instruments that exhibit
ultrasonic activity - the electric guitar in bluegrass music, where
intentional feedback produces rampant clipping and the characteristic
electric guitar sound, also results in spectral content extending above 20
kHz. Furthermore, Atkinson noticed that even old analog recordings from the
'60s and earlier have captured this ultrasonic content."

Wow, old analog beats new 44.1 khz digital.

The report cites references that possibly higher frequencies are important
to perception. So if that's true, limiting sampling rates to 44.1 may be
discarding important perceptual information.

But hey, 44.1 is all we need right? Maybe all we need is to druve Hyundais
and eat soy food too.


Engineers are the worst offenders and failing to understand the
slightest thing about psychoacoustics.

I feel so hurt.

Listen for yourself at www.pcabx.com.

Is that a page or a website?

Here's anpther non-scientific, but interesting article on 96 kHz including
some ABX tests:

"I know... I know... I can hear many of you saying there is absolutely NO
need for recording with a 96kHZ Sample Rate. Two weeks ago, I would have
agreed with you! I emphasize *would have* agreed with you! Let me state this
very clearly... YOU CAN INDEED HEAR THE DIFFERENCE when recording with a
96kHz Sample Rate!

I wouldn't have believed it myself if I hadn't heard the results. Bottom
line is that the highs sound more open and detailed. By the way... two other
folks here in my studio could pick the 96kHz track EVERY time in a blind
listening test (when compared with a 44.1kHz version). To hell with theory,
my EARS tell me there is a difference.

Want a real dose of Blasphemy? I compared recording at 96kHz and Sample Rate
converting down to 44.1, to simply recording at 44.1kHz. I couldn't believe
my ears! The track originally recorded at 96kHz and Sample Rate converted
down to 44.1kHz had much better sounding highs, maintaining much of the
character from recording at 96kHz.

This goes against everything that I have learned over the years... and goes
against accepted practice. So I don't make this statement lightly! You CAN
hear a difference... anyone who tells you otherwise hasn't tried recording
at 96kHz! Period. "

http://www.prorec.com/prorec/article...256688000FBE08

Did you click on it? Did you listen? Audiophiles are even more mislead
than engineers.

THAT is YOUR problem...the fact that you do not see the utter stupidity of
this as you read it.

"...as it went from audio to totaly
mathematical/phisycal discussion.


Absolutely NOT. I complain that you and others have not read anything about
the
results of psychoacoustical experimentation on hearing that links physics to
perception.


Hence your mysticism in what you believe you hear.

"Buckaroo" wrote in message
...
THAT is YOUR problem...the fact that you do not see the utter stupidity of
this as you read it.

No, that is the classic repsonse of condescension and feigned superiority.
You add little else to this discussion. than to tell people they are wrong
and that you have secret knowledge that can't be revealed.

Your smug attitude will never change readers minds.

"Buckaroo" wrote in message
...
Did you click on it? Did you listen? Audiophiles are even more mislead
than engineers.

Arny says I have to have aq $5000 audio card so I won't bother.

"Buckaroo" wrote

RE the DVD issue. It is often perceptually based coding....compressed based
on psychoacoustic models and thus
not at all isomorphic with the original digital PCM signal (which is
supposed to be included in stereo and possibly dolby surround,
but may not be).

Oh, OK, I got 'cha... NOT! What might this mean in practical terms? I
guess you're saying that this might somewhat change the sound? I have
some things that have a fair amount of hiss in the original recording,
could you say if what you're talking about would have an impact on
recordings in which hiss is present? Thanks.

"FDR" wrote in message


"Buckaroo" wrote in message
...

Did you click on it? Did you listen? Audiophiles are even more
mislead than engineers.

Arny says I have to have aq $5000 audio card so I won't bother.

More lies.

"FDR" wrote in message

"Buckaroo" wrote in message
...
THAT is YOUR problem...the fact that you do not see the utter
stupidity of this as you read it.

No, that is the classic repsonse of condescension and feigned
superiority. You add little else to this discussion. than to tell
people they are wrong and that you have secret knowledge that can't
be revealed.

Your smug attitude will never change readers minds.

Highly ironic.