Mr. Lavry's 192kHz claims?

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On Sat, 06 Nov 2004 13:23:06 -0800, Bob Cain
<arcane@arcanemethods.com> wrote:


>When you consider the horrible things even a good
>loudspeaker (or a room) does to a signal it defies
>imagination that all these incredibly marginal effects could
>be of any real consequence. It's about marketing and gear
>churning as Dan implies if not directly states.


Bob,

While I understand your position, some of us actually enjoy seeking
out "marginal improvements" in audio quality. You might even call it a
passion.

JL
 
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John La Grou wrote:

> On Sat, 06 Nov 2004 13:23:06 -0800, Bob Cain
> <arcane@arcanemethods.com> wrote:
>
>
>
>>When you consider the horrible things even a good
>>loudspeaker (or a room) does to a signal it defies
>>imagination that all these incredibly marginal effects could
>>be of any real consequence. It's about marketing and gear
>>churning as Dan implies if not directly states.
>
>
>
> Bob,
>
> While I understand your position, some of us actually enjoy seeking
> out "marginal improvements" in audio quality. You might even call it a
> passion.
>
> JL

But only if they can be discriminated and I fail to see how
some of the extreme subtlety that is argued about could
possibly make it through the relatively large linear and
non-linear distortions imposed by speakers and rooms and
still be audible as improvements.

The effects are swamped by the variance just in the unit to
unit tolerances of speakers. They remain remarkably crude
elements of the system compared to the other components.
It's the old weakest link thing.

I well understand the passion for improvement, but with an
engineering background involving systems error analysis I am
extremely dubious that the marginal differences, and _so_
many are claimed in the audio world, are anywhere near as
signifigant as many who have a vested interest want to believe.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein
 
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"John La Grou" <jl@jps.net> wrote in message
news:lqjqo0h321r9aei8ll135qvf7mn1jfqk89@4ax.com...
> While I understand your position, some of us actually enjoy seeking
> out "marginal improvements" in audio quality. You might even call it a
> passion.

Don't get me wrong - it's a passion for me too, as a designer. And
obviously we've gotten to the point where we are now because of a succession
of people driven to say "sure, what we've got now is good enough, but I can
make something even better." After all, people claimed that Edison
cylinders were lifelike and realistic; thank heavens we didn't stop there.

But from the perspective of a musician or producer trying to make a good
record, I wonder whether we engineers have lost touch with where the most
urgent needs are, and I wonder whether our customers are well served by the
particular improvements we've chosen to focus on. If the goal is to make it
easier to get compelling recordings of great music in a time- and
cost-efficient way, how can we best support that goal?
 
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John La Grou wrote:
> On Sat, 06 Nov 2004 19:03:14 GMT, in rec.audio.pro you wrote:
>
>
>>Were they double-blind? What did the "subjective" listeners know going
>>into the tests?
>
>
>
> In essence:
>
> http://www.mil-media.com/docs/articles/design.shtml
>
> http://www.mil-media.com/docs/articles/preamps.shtml

I think these two pointers answered my question. I have to say that I
found them completely dissatisfying, even if they were convincing that
you strive for excellence in your company's pursuits.

>>Ah-ha! Then use the 192 KHz part, but do not run it at 192 KHz.
>
>
>
> We'll give users the choice of all available sample rates, and perhaps
> document our personal preferences in the manual. Even if raw 192kHz
> was generally inferior WRT subjective accuracy, it might offer a
> signature that some producers find useful to achieve a certain color.

I think that stating up front in your documentation to customers and
potential customers all the things you have said and disclaimed in this
thread would be of great value.

If you have a reputable position in the industry, people will look to
you for guidance. So this carries a great responsibility.

> I suspect that any deficiencies of 192kHz probably have more to do
> with DACs than ADCs. but coaxing the slicon designers to join these
> discussions is difficult. It's a small niche of designers who value
> their job security..

These are the salaried ones that must answer to the corporate powers
that be. Having spent many years in R&D myself in a large corporate
environment, I know well what is at stake.

They may be reading these discussions even, but they will say nothing.
 

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Garth D. Wiebe wrote:

> John La Grou wrote:
>> On Sat, 06 Nov 2004 19:03:14 GMT, in rec.audio.pro you wrote:
>>
>>>Were they double-blind? What did the "subjective" listeners know going
>>>into the tests?
>>
>> In essence:
>>
>> http://www.mil-media.com/docs/articles/design.shtml
>>
>> http://www.mil-media.com/docs/articles/preamps.shtml
>
> I think these two pointers answered my question. I have to say that I
> found them completely dissatisfying, even if they were convincing that
> you strive for excellence in your company's pursuits.

Strive??? I'd say he succeeds.... in spades. Just in case you weren't
aware, Mr. La Grou may very well make the most transparent microphone
preamplifier on the planet. What Dan Lavry is to A/D conversion, John La
Grou certainly is to preamplification.
 
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Bob Olhsson wrote:
> "Garth D. Wiebe" <gwiebe@audiorail.com> wrote in message
> news:418D1FF1.3050608@audiorail.com...
>
>>Ah-ha! Then use the 192 KHz part, but do not run it at 192 KHz. Do not
>>enable it at that speed.
>
>
> It's important to understand that Dan isn't arguing in favor of 96k chips.
> He's doing a lot more than just implimenting somebody else's parts so he
> sees this as a matter of putting a lot of resources into arguably inferior
> product performance.

In context, John La Grou argued that he was compelled to use the 192 KHz
parts because of other features that they had which the 96 KHz parts did
not. He did not say what specific features they were, but I used that
as the basis for my if-then clause.

I do agree with Dan's arguments that you refer to.
 
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agent86 wrote:
> Garth D. Wiebe wrote:
>
>
>>John La Grou wrote:
>>
>>>On Sat, 06 Nov 2004 19:03:14 GMT, in rec.audio.pro you wrote:
>>>
>>>
>>>>Were they double-blind? What did the "subjective" listeners know going
>>>>into the tests?
>>>
>>>In essence:
>>>
>>>http://www.mil-media.com/docs/articles/design.shtml
>>>
>>>http://www.mil-media.com/docs/articles/preamps.shtml
>>
>>I think these two pointers answered my question. I have to say that I
>>found them completely dissatisfying, even if they were convincing that
>>you strive for excellence in your company's pursuits.
>
>
> Strive??? I'd say he succeeds.... in spades. Just in case you weren't
> aware, Mr. La Grou may very well make the most transparent microphone
> preamplifier on the planet. What Dan Lavry is to A/D conversion, John La
> Grou certainly is to preamplification.

I am certainly not questioning the ultimate quality of his products and
work. Only whether a 192KHz converter was "better".
 
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agent86 wrote:

> Mr. La Grou may very well make the most transparent microphone
> preamplifier on the planet.

I have an HV-3D here, and I do enjoy it very much. But I take it from
your comment that you have not auditioned a Gordon Instruments preamp.
That preamp, for me, completely recalibrates "transparent". No, I cannot
afford one.

--
ha
 

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hank alrich wrote:

> agent86 wrote:
>
>> Mr. La Grou may very well make the most transparent microphone
>> preamplifier on the planet.
>
> I have an HV-3D here, and I do enjoy it very much. But I take it from
> your comment that you have not auditioned a Gordon Instruments preamp.
> That preamp, for me, completely recalibrates "transparent". No, I cannot
> afford one.

Nope, I haven't tried the Gordon. I'm usually careful to use phrases like
"may very well" for that very reason.

I can't even afford the Millenia. But it's one fine unit, no matter who it
belongs to.
 
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On Sat, 06 Nov 2004 18:50:50 -0800, Bob Cain
<arcane@arcanemethods.com> wrote:


>I well understand the passion for improvement, but with an
>engineering background involving systems error analysis I am
>extremely dubious that the marginal differences, and _so_
>many are claimed in the audio world, are anywhere near as
>signifigant as many who have a vested interest want to believe.


Bob,

This may sound callous, but I'm not concerned about what others think
about the marginal improvements I perceive. Nor am I concerned that
some may scoff at single-blind testing. It works for me, and I'll keep
striving for perceptible improvements in everything we do, regardless
of how insignificant others may judge those increments to be.

That said, I do understand and respect your position.

JL
 
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"Bob Cain" <arcane@arcanemethods.com> wrote in message
news:cmjfbp012n5@enews3.newsguy.com...
>
>
> Walter Harley wrote:
>
>
>> Sheesh. I just read the many excellent discussions others have posted to
>> this thread, but I have to say I came away from AES overcome by dominant
>> thought: Does any music really *need* anything better than the fidelity
>> of, say, Kind Of Blue? Isn't 95% of this new gear just solutions to the
>> wrong problem?
>
> When you consider the horrible things even a good loudspeaker (or a room)
> does to a signal it defies imagination that all these incredibly marginal
> effects could be of any real consequence. It's about marketing and gear
> churning as Dan implies if not directly states.

Also comparisons of gear are unreliable with other as yet unaddressed
factors being variables such as in head position, auditory changes after
exposure to one set of music and a small period ofd time, etc.

What is more relevant (other than obvious real differences) is long term
imperssions of the qualities of a piece of gear, over a range of music and
time. But then it is hard to control the other necessary controls for
objective results.

geoff
 
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"Arny Krueger" <arnyk@hotpop.com> wrote in message
news:QoidnefLBLWNLBbcRVn-tw@comcast.com...
> "Bobby Owsinski" <polymedia@earthlink.net> wrote in message
> news:polymedia-1FFCCC.08334505112004@news1.west.earthlink.net
>
> > You know, I've followed Dan's claims and newsgroup threads and I must
> > admit that he presents a good case. But having done a fair amount of
> > 192k recording (as well as recording the same program and 44.1, 48, 96
> > and 192k), I can tell you that everyone involved in these recordings
> > are always very partial to the 192, especially after hearing the same
> > program at a lower rate.
>
> Tell you what, Bobby. Send me as much of as any high sample rate file(s)
as
> you think you need to make your point. My *real* email address is arnyk at
> comcast dot net .
>
> Comcast has a 10 meg final file size, or about 7.6 meg file size limit
for
> email attachments according to
> http://faq.comcast.net/faq/answer.jsp?name=17627&cat=Email&subcategory=1
If
> email won't handle the file size, I think I can provide you with some FTP
> upload space and a userid and password.
>
> I'll downsample your sample(s) down to various far lower sample rate and
> then upsample them back to whatever high sample rates they started out at.
> I'll then put up a web page at www.pcabx.com where people can download
them
> from, and listen for themselves.
>
>

Why the down/upsampling? Why not just post the samples?
I'm guessing editing...
 
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"Walter Harley" <walterh@cafewalterNOSPAM.com> wrote in
news:qLadndci0fGf9xDcRVn-uw@speakeasy.net:

<snip>

> I wonder whether our customers are well
> served by the particular improvements we've chosen to focus on. If
> the goal is to make it easier to get compelling recordings of great
> music in a time- and cost-efficient way, how can we best support that
> goal?

I think that we have reached a satisfying level in any single channel.

I still have not yet heard a believable full surround sound.

And I still believe that the next generation will combine sound with video
for portable entertainment.
 
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John La Grou <jl@jps.net> wrote in news:rj5ro0psfhcq8d2ub4dgi3i8n6jn9ssos1@
4ax.com:

> I'll keep
> striving for perceptible improvements in everything we do, regardless
> of how insignificant others may judge those increments to be.

Both are correct. Striving for perfection in each component aids us in
achieving perfection for the whole system. Let others work on the enormous
variation in reproduction formats.
 
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Thanks, Arny. Minimal difference, but the figures do speak to better 96 kHz
response, even though it's obviously insignificant in audability.

--


Roger W. Norman
SirMusic Studio

"Arny Krueger" <arnyk@hotpop.com> wrote in message
news:evSdnVQteqjOWBbcRVn-qQ@comcast.com...
> "Roger W. Norman" <rnorman@starpower.net> wrote in message
> news:g8OdnUGAXs8lJxbcRVn-3Q@rcn.net
> > Seems to me that regardless of the possible negatives, when used at a
> > lower bit rate than 192 kHz, these converters should perform close to
> > the stellar range. In other words, how much has anyone looked at 192
> > kHz converters running at 96/88.2?
>
> http://www.pcavtech.com/soundcards/LynxTWO/index.htm for one example.
>
>
 
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"Natalie Drest" <mccoeyHAT@netspaceCOAT.net.au> wrote in message
news:cmkutq$113b$1@otis.netspace.net.au
> "Arny Krueger" <arnyk@hotpop.com> wrote in message
> news:QoidnefLBLWNLBbcRVn-tw@comcast.com...
>> "Bobby Owsinski" <polymedia@earthlink.net> wrote in message
>> news:polymedia-1FFCCC.08334505112004@news1.west.earthlink.net
>>
>>> You know, I've followed Dan's claims and newsgroup threads and I
>>> must admit that he presents a good case. But having done a fair
>>> amount of 192k recording (as well as recording the same program and
>>> 44.1, 48, 96 and 192k), I can tell you that everyone involved in
>>> these recordings are always very partial to the 192, especially
>>> after hearing the same program at a lower rate.
>>
>> Tell you what, Bobby. Send me as much of as any high sample rate
>> file(s) as you think you need to make your point. My *real* email
>> address is arnyk at comcast dot net .
>>
>> Comcast has a 10 meg final file size, or about 7.6 meg file size
>> limit for email attachments according to
>> http://faq.comcast.net/faq/answer.jsp?name=17627&cat=Email&subcategory=1
>> If
>> email won't handle the file size, I think I can provide you with
>> some FTP upload space and a userid and password.
>>
>> I'll downsample your sample(s) down to various far lower sample rate
>> and then upsample them back to whatever high sample rates they
>> started out at. I'll then put up a web page at www.pcabx.com where
>> people can download them from, and listen for themselves.

> Why the down/upsampling?

The purpose of the downsampling is to provide examples of what
low-sample-rate digital data formats do to high-sample-rate audio data.

The purpose of the subsequent upsampling is to provide samples that people
can compare using the same converters operating at the same sample rates.

>Why not just post the samples?

Because you can't isolate the sonic signatures of sample rates from the
sonic signature of hardware operating at different sample rates that way.

> I'm guessing editing...

No, its all about doing a comparison of just the sonic properties of digital
formats operating at different sample rates.

If you want to cut to the chase - I'll tell you what happens when you do
proper listening tests. You find out what has been shown many times - that
44/16 is actually sonic overkill. Audible artifacts of not enough data per
sample, and not enough samples per second sort of cut out when you go much
higher than about 14/38, presuming a good clean modern monitoring
environment. Ironically, substandard monitoring environments can be more
*sensitive* to high sample rate music, but that is due to artifacts that
they introduce due to their technical inadequacies.

The usual argument against tests with results like these, is that the
origional music was not pristene enough and/or that the monitoring
environment was not clean enough, or someones ears aren't good enough.
Therefore, it is helpful to get the person making the naive assertions to
provide the origional music for testing and perform the tests with their own
monitoring system, and of course use their own ears.

Here is an example of what happens when *name* people do their own tests
like these:

"George Massenburg" <gmlinc@ix.netcom.com> wrote in message
news:dc15750e.0301091707.5e40d7ce@posting.google.com

> Speaking of 'differences'. I hope that I live long enough to craft and
> demonstrate what a scientific listening/evaluation test is and what it
> isn't.
> What it isn't is what you might call the [golden-ear pantload name
> here] demonstration where this guy sits you down and plays you a
> couple of things (could be anything: the levels aren't calibrated and
> could be anywhere). [G.E.P.L.] proceeds to switch sounds for you
> saying, "O.K., listen to this. RIght, NOW listen to THIS!" (maybe he
> actually turns the monitor gain up) "Wow, that's great, huh?" And this
> other? HEY, you couldn't possibly like THAT, could you??? I mean,
> c'mon, you'd be an IDIOT not to hear the difference...
> Any test where you know which piece of gear you're listening to...any
> test that's not perfectly blindfolded and well-controlled cannot
> possibly be called scientific. As much as I don't like the downsides
> of the A-B-C-Hidden Reference it's a very useful discipline to reveal
> modest differences.
> The best listening tests demand that you objectify what you hear.
> An example of a useful, forthright listening test is the high-octave
> test suggested and implemented by Bob Katz, where he takes a 96/24
> file (presumably rich in >20kHz content), and filters it at 20kHz or
> so. Then he listens (through exactly the same hardware, and under
> exactly the same circumstances, removing conversion, to name one
> factor, as a possible variant) to see if he can tell the difference
> between the two (filtered and unfiltered) files. Can I be brave here
> and tell you the truth? Neither of us have had significant successes
> with differentiating between the samples. (Incidentally, this is a
> test that I proposed several years ago at the AES Technical Committee
> on Studio Production and Practices, and have finally implemented on
> the EdNet web site. Stay tuned.)
 
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Arny Krueger wrote:

> "Natalie Drest" <mccoeyHAT@netspaceCOAT.net.au> wrote in message
>
>>Why the down/upsampling?
>
>
> The purpose of the downsampling is to provide examples of what
> low-sample-rate digital data formats do to high-sample-rate audio data.


Remember that Mr. Lavry maintained this was not a valid comparison
because the performance of the S/H was degraded at higher sample rates;
a convertor optimized for 192 could not perform as well as the same
convertor optimized for 96. That is, starting at 96 would necessarily
give better results than downsampling to 96.
 
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"John La Grou" <jl@jps.net> wrote in message
news:619qo092ncs07cpp4cmpc4lp1icphq7s4d@4ax.com
> On Sat, 06 Nov 2004 19:03:14 GMT, in rec.audio.pro you wrote:
>
>> Were they double-blind? What did the "subjective" listeners know
>> going into the tests?

> In essence:

> http://www.mil-media.com/docs/articles/design.shtml

> http://www.mil-media.com/docs/articles/preamps.shtml

I see nothing in these articles that should assure *anybody* that
time-synched, level-matched, bias-controlled listening tests are being used
in any way, size, shape or form.

What am I missing?

BTW, the *standard* document for judging the adequacy of a listening test
would be ITU recommendation BS-1116. More information about proper
listening tests can be found at www.pcabx.com .
 
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John La Grou wrote:

> Bob,
>
> This may sound callous, but I'm not concerned about what others think
> about the marginal improvements I perceive. Nor am I concerned that
> some may scoff at single-blind testing. It works for me, and I'll keep
> striving for perceptible improvements in everything we do, regardless
> of how insignificant others may judge those increments to be.

I want to make sure to emphasize again that for my part, all my comments
are for the purpose of zeroing in on the 192K sampling rate issue. That
is not a topic for subjective listening. That is science and
engineering. It is an area for theory and for regimented experimental
testing that must be double blind. It is something that should never
have gone to market without that R&D rigor.

Correct me if I'm mistaken, but I do not recall your website indicating
even "single blind" testing. Now, there is a time and a place for
subjective listening. I do a humble amount of studio recording and live
SR myself on the side, plenty enough to know that just listening is
important in many circumstances, and especially just listening with
clients to find out what each individual likes and wants. Non-blind
listening. I've developed a good reputation within my humble realm of
clientele, who depend on me to get "good sound" and "good recordings".
Choice of microphones and speakers, placement of microphones and
speakers, effects, the mix, and so on. You can't "double-blind" most of
that sort of thing.

You seem to have a good balanced view of things. But a reputable
company like yours could, even inadvertently, mislead people to the
wrong idea. One person in this thread utters the chant "When Yo Yo Ma
and John Williams ask for it, there must be something to it." Another
in this thread says "Just in case you weren't aware, Mr. La Grou may
very well make the most transparent microphone preamplifier on the
planet. What Dan Lavry is to A/D conversion, John La Grou certainly is
to preamplification."

So the question I am asking is, will we end up with people saying "John
La Grou prefers 192K, so there must be something to it."? That is the
question. You could in principle feed this, or you could disclaim it.
I would urge that if you feel compelled to implement 192K and make that
an option to your customers, then your marketing, press releases,
website, and owner's manual would make it clear that you are using 192K
converters not because you think 192K sampling rate is fundamentally the
way to go, but for the other reasons you stated, which have nothing
fundamentally to do with any desire to sample at 192K.
 
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In article <418F657B.8070300@audiorail.com> gwiebe@audiorail.com writes:

> So the question I am asking is, will we end up with people saying "John
> La Grou prefers 192K, so there must be something to it."? That is the
> question.

Substitute "may" for "must" and that's valid. John makes good
recordings, and if his recordings start sounding even better to him
(and presumably his customers) when he starts using 192 kHz
components, then so be it.

I think the issue is that we (as an industry of practicioners rather
than pure scientists) tend to talk in shortspeak. "192 kHz" doesn't
mean simply "generating 192,000 samples for each one second of audio"
but rather means "building new gear based on components capable of
generating . . ." As John said, his 192 kHz results may not be
isolated to just the higher sample rate, but a combination of that and
better design of the parts that he buys off the shelf as well as
better surrounding designs based on what he's learned and how what he
designs affects what he hears.

As Dan Lavry suggests, simply changing sample rate and leaving
everything else the same doesn't make for a good experiment.


--
I'm really Mike Rivers (mrivers@d-and-d.com)
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me here: double-m-eleven-double-zero at yahoo
 
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"S O'Neill" <nopsam@nospam.net> wrote in message
news:OJ6dnf-ItsrDChLcRVn-ow@omsoft.com
> Arny Krueger wrote:
>
>> "Natalie Drest" <mccoeyHAT@netspaceCOAT.net.au> wrote in message
>>
>>> Why the down/upsampling?
>>
>>
>> The purpose of the downsampling is to provide examples of what
>> low-sample-rate digital data formats do to high-sample-rate audio
>> data.
>
>
> Remember that Mr. Lavry maintained this was not a valid comparison
> because the performance of the S/H was degraded at higher sample
> rates; a convertor optimized for 192 could not perform as well as the
> same convertor optimized for 96.

I think you've got me confused with someone who has a controversy with Mr.
Lavry. If you go back and look at the post I was responding to, it was by
Bobby Oswinsky, not Dan Levry.

> That is, starting at 96 would necessarily give better results than
> downsampling to 96.

Agreed, but that was not the issue that I was addressing. I was addressing
what appeared to be a claim that recording at 192 has audible benefits.
 
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"Mike Rivers" <mrivers@d-and-d.com> wrote in message
news:znr1099919189k@trad

> As Dan Lavry suggests, simply changing sample rate and leaving
> everything else the same doesn't make for a good experiment.

I'm not sure that's what Dan was trying to suggest, but my omniscience
module is not performing as desired lately.

Simply changing the sample rate and leaving everything else the same does
make for a good experiment, depending on the question you are trying to
answer. There seem to be a lot of different quesitons that various people
have in mind.

One question that many might find itneresting might be: Does changing the
sample rate and leaving everything else pretty much the same make a
difference? Looking at extant controversies, even narrower questions such
as: "Does increasing the sample rate above 44.1 KHz and leaving everything
else pretty much the same make a difference?" seem to be interesting to some
people.
 
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Arny Krueger wrote:


> If you want to cut to the chase - I'll tell you what happens when you do
> proper listening tests. You find out what has been shown many times - that
> 44/16 is actually sonic overkill. Audible artifacts of not enough data per
> sample, and not enough samples per second sort of cut out when you go much
> higher than about 14/38, presuming a good clean modern monitoring
> environment. Ironically, substandard monitoring environments can be more
> *sensitive* to high sample rate music, but that is due to artifacts that
> they introduce due to their technical inadequacies.

Careful, Arny. People here are likely to shoot the
messenger. :)

For all the argument about faith vs fact as guiding
principles you'd think all golden ears were neocons.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein
 
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On Mon, 08 Nov 2004 14:25:05 +0100, Roger W. Norman wrote:
> Thanks, Arny. Minimal difference, but the figures do speak to better 96
> kHz response, even though it's obviously insignificant in audability.

There are also some theoretical benefits for signal processing (FFT) at a
higher sampling rate. Here too the question is if they are audible.
IMHO there are no real grounds to spend time and money on improvements if
you have a good performing 96/24 or even a 44.1/24 set.
Better spend your time and money on acoustics, microphones, microphone
position etc.

For a consumer playback environment I think 44.1/16 allmost never is the
limitting factor. Speakers, speaker placement and room acoustics normally
are by far the most limitting factors.

--
Chel van Gennip
Bezoek Serg van Gennip's site http://www.serg.vangennip.com
 
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Chel van Gennip wrote:


> For a consumer playback environment I think 44.1/16 allmost never is the
> limitting factor. Speakers, speaker placement and room acoustics normally
> are by far the most limitting factors.

By at least an order of magnitude WRT all relevant
paramaters of accuracy.


Bob
--

"Things should be described as simply as possible, but no
simpler."

A. Einstein
 
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