I've read where supposedly those who are accutely sensitive can hear
deficiencies in 44.1 /16-bit CD's. If so, how do those with "golden ears"
deal with eternally wallowing in inadequately reproduced sound?

"Doc" wrote in message
nk.net...
I've read where supposedly those who are accutely sensitive can hear
deficiencies in 44.1 /16-bit CD's. If so, how do those with "golden ears"
deal with eternally wallowing in inadequately reproduced sound?

I recently refurbished just about every component in my sound system and
have resumed my critical listening habits, so I have a few observations on
this:

A real eye-opening experience was a Crystal Clear Records direct-to-disc
recording that I have in my LP library. I have many Telarc CDs too. I also
have a Japanese recording on LP and CD and some curious differences were
observed.

The direct to disc was a performance of the Atlanta Brass Ensemble together
with an organ performance by Richard Morris at the Cathedral of Christ the
King, in Atlanta. After listening to a varied collection of CDs with digital
mastering, this recording stood out. Not only was it very quiet (having
bypassed the analog tape recording process), but it had a depth of
soundstage that most of the CDs lacked.

My other comparison was of the same drum performance, first appearing on LP
and later on a digitally remastered CD. The LP version had an immersive
3-dimensional space, giving a very palpable sense of 'being there'. The CD,
though lacking surface noise, didn't have quite the vastness and sense of
'space'.

My hunch? The sample rate was smearing subtle time arrival cues.

I think higher and deeper sample rates, such as 24/96, are the solution to
this difference.

Yes, I've heard the dither in very soft passages of classical recordings.
There are only 32,768 loudness levels with 16-bits. That increases to 16.7
million levels with 24-bit recording.


--
Best Regards,

Mark A. Weiss, P.E.
www.mwcomms.com
-

Mark & Mary Ann Weiss wrote:

My hunch? The sample rate was smearing subtle time arrival cues.

It is the bit depth that and sample rate together that
controls time resolution. Divide the sample period by the
number of discrete levels, (65536 for 16 bit) to get
approximately the time resolution. I think 16/44.1 at about
..35 nsec jitter is finer than the ear can detect.


I think higher and deeper sample rates, such as 24/96, are the solution to
this difference.

Yes, I've heard the dither in very soft passages of classical recordings.
There are only 32,768 loudness levels with 16-bits. That increases to 16.7
million levels with 24-bit recording.

And this translates purely to white noise, hiss. Stairstep
audibility is a myth. There was a pair of samples put up
here some time ago with music at 4 bit resolution compared
to full 16 bit resolution with the same amount of random
noise simply added and the sound is identical. If a
recording is anywhere near 0 dB, the -96 dB white noise due
to 16 bit samples (without dither) is absolutely inaudible.

There really isn't any valid argument for distribution and
consumer playback gear to be better than 16 bits. The wider
width is a headroom consideration for the recording and
mixing engineer only.

There is an argument that the reconstruction filters for
44.1 kHz introduces some small phase shift at the highest
frequencies that is audible but AFAIC that is very
equivocal. Most that claim to hear it are invested in being
known for having golden ears. Double blind testing has not
substantiated that audibility.


Bob
--

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

A. Einstein

On Sun, 30 Jan 2005 15:21:10 -0800, Bob Cain
wrote:

It is the bit depth that and sample rate together that
controls time resolution. Divide the sample period by the
number of discrete levels, (65536 for 16 bit) to get
approximately the time resolution. I think 16/44.1 at about
.35 nsec jitter is finer than the ear can detect.

This is very counter-intuitive. The extreme case of
infinite word length with no sample timing uncertainty
and no signal level modulation of sample point would
have, by implication, infinite bandwidth.

Wouldn't the reconstruction filtering process invalidate
this? Or is the answer over my head? Pretty likely.

Chris Hornbeck
"Don't be foolish, like the others." _Lola Montes_, 1955

Chris Hornbeck wrote:
On Sun, 30 Jan 2005 15:21:10 -0800, Bob Cain
wrote:


It is the bit depth that and sample rate together that
controls time resolution. Divide the sample period by the
number of discrete levels, (65536 for 16 bit) to get
approximately the time resolution. I think 16/44.1 at about
.35 nsec jitter is finer than the ear can detect.


This is very counter-intuitive. The extreme case of
infinite word length with no sample timing uncertainty
and no signal level modulation of sample point would
have, by implication, infinite bandwidth.

Infinite precision in time resolution is not at all the same
as infinite bandwidth. Even with infinite precision, the
Nyquist requirement of bandlimiting to half the sample rate
applies. Quantization jitter is just another form of noise.
I think it is more correlated with the signal than is
quantization noise but I'm not sure about that.


Wouldn't the reconstruction filtering process invalidate
this? Or is the answer over my head? Pretty likely.

Not sure why you would think that.


Bob
--

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

A. Einstein

On Sun, 30 Jan 2005 16:34:44 -0800, Bob Cain
wrote:

Infinite precision in time resolution is not at all the same
as infinite bandwidth. Even with infinite precision, the
Nyquist requirement of bandlimiting to half the sample rate
applies. Quantization jitter is just another form of noise.

I think of the A/D process in two steps, A, the bandwidth limiting
of anti-aliasing filters and the dynamic range limiting of
dither. And B, the potentially perfect conversion.

On the D/A end, a potentially perfect reconstruction of the
quantized samples yields a potentially perfectly exact duplicate
of the original bandwidth- and dynamic range- limited signal.

Or does it? There's one other mystery component.

I think it is more correlated with the signal than is
quantization noise but I'm not sure about that.

I'm thinking more in a perfect world scenario, where perfectly
dithered analog is sampled with no rounding errors.

But how does the reconstruction low-pass filter (pre- D/A)
effect, by its finite bandwidth, the effective precision
of the original A/D sampling? It seems to me that this
filter, although inaccessible and never discussed, has as
big an effect on reproduction as many more glamourous topics.

Wouldn't the reconstruction filtering process invalidate
this? Or is the answer over my head? Pretty likely.

Not sure why you would think that.

Sorry for being so flaky. Somebody in this very newsgroup once
said that if you could ask the right question, you wouldn't have
to ask. Lotsa good sense there.

Thanks, as always,

Chris Hornbeck
"Don't be foolish, like the others." _Lola Montes_, 1955

"Mark & Mary Ann Weiss" wrote in message
k.net

My other comparison was of the same drum performance, first appearing
on LP and later on a digitally remastered CD. The LP version had an
immersive 3-dimensional space, giving a very palpable sense of 'being
there'. The CD, though lacking surface noise, didn't have quite the
vastness and sense of 'space'.

One of the things that a recordist can know, but that very few hi fi
listeners if any can ever know, is exactly what the live performance that
was recorded sounds like.

It is very important to remember that high fidelity is about the
reproduction of music, not the production of music. You can imagine what the
live performance sounds like, but your reference sound is almost always a
presumption, not a fact. In contrast, the recordist was intimiately invovled
with the live performance, essentially by definition.


My hunch? The sample rate was smearing subtle time arrival cues.

It's a not uncommon old wive's tale that the sample rate limits the temporal
accuracy of recordings. This is a fallacy. The temporal accuracy of a
recording is proportional to the product of the sample rate and the number
of distinct levels that the data format can handle. This turns out to be an
immense number that represents time ambiguity that is almost completely
impossibly smaller than human perception.

I think higher and deeper sample rates, such as 24/96, are the
solution to this difference.

There's a simple fact - downsample *any* commercial recording at 24/96 down
to 16/44 and then upsample it back to 24/96 and compare it to the origional
in a bias-controlled listening test, and you find sonic equality.

Yes, I've heard the dither in very soft passages of classical
recordings. There are only 32,768 loudness levels with 16-bits. That
increases to 16.7 million levels with 24-bit recording.

Yet another not uncommon old wive's tale. All proper digital recordings are
dithered. Dither in essence eliminates the lower bound on distinguishable
loudness limits. Musical tones very much smaller the smallest step size can
be recorded and played back audibly and without distortion. The resolution
of both analog and digital systems is set by the same thing - the quotent of
the largest signal that can be recorded and the noise floor.

"Arny Krueger" wrote in message
...
"Mark & Mary Ann Weiss" wrote in message
k.net

My other comparison was of the same drum performance, first appearing
on LP and later on a digitally remastered CD. The LP version had an
immersive 3-dimensional space, giving a very palpable sense of 'being
there'. The CD, though lacking surface noise, didn't have quite the
vastness and sense of 'space'.

One of the things that a recordist can know, but that very few hi fi
listeners if any can ever know, is exactly what the live performance that
was recorded sounds like.

Even if you're at a live symphony concert, as with listening to speakers, is
it not true that what you hear depends on where you're sitting? That there's
not just one version of what it "really" sounds like live.

"Doc" wrote in message
nk.net
"Arny Krueger" wrote in message
...
"Mark & Mary Ann Weiss" wrote in message
k.net

My other comparison was of the same drum performance, first
appearing on LP and later on a digitally remastered CD. The LP
version had an immersive 3-dimensional space, giving a very
palpable sense of 'being there'. The CD, though lacking surface
noise, didn't have quite the vastness and sense of 'space'.

One of the things that a recordist can know, but that very few hi fi
listeners if any can ever know, is exactly what the live performance
that was recorded sounds like.

Even if you're at a live symphony concert, as with listening to
speakers, is it not true that what you hear depends on where you're
sitting?

Of course.

That there's not just one version of what it "really" sounds like live.

Also true.

Please be my guest as you punch even more holes in people's theories that
they know exactly what a recording should sound like, right down to the most
subtle details.

Mark & Mary Ann Weiss wrote:
Yes, I've heard the dither in very soft passages of classical recordings.
There are only 32,768 loudness levels with 16-bits.

65,536 loudness levels. A 16-bit signed integer ranges
from -32,768 to +32,767.

- Logan

"Doc" wrote in message
nk.net

I've read where supposedly those who are accutely sensitive can hear
deficiencies in 44.1 /16-bit CD's.

Interestingly enough, evidence that *anybody* has heard deficiencies due to
the 16/44 format as applied to normal music in bias-controlled tests is very
hard to find.

I've studied this problem on and off for over 20 years. The following was an
early attempt to understand this problem:

http://www.provide.net/~djcarlst/abx_digi.htm

"The Ampex 16 Bit Digital Delay Line vs. wire comparison was heard in a
recording studio control room on time aligned UREI 813 speakers with
McIntosh MC-2100 amplifiers. The audio source was a master 2-track 15 IPS
tape on a Scully 280. This master tape had been mixed from a 24-track master
tape on an Ampex MM-1000. The mixdown and playback was through an API
console. The listeners included professional recording engineers with years
of experience on major label projects, professional maintenance engineers,
and recording engineering students.. For those not familiar with studio
equipment, these are some of the most revered pieces of equipment of that
day. API consoles are still prized today for their high quality."

Then, there are the listening tests that you can do for yourself, using
audio files you can download from

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

"24 bit 96 KHz "reference" samples were made by using 2 B&K 4007 1/2"
condenser microphones powered by an Audio Technica phantom power unit,
preamplified using a Benchmark Media mic preamp, and recorded using a CardD
Deluxe in a 800 Mhz Pentium 3 computer located in another room. They were
closely miced on-axis in a fairly small dead space. Therefore the transients
are very well-defined and harmonic-rich, technically speaking. They also
have relatively low amounts of background noise (mostly acoustic). They may
sound quite "dry" to your ears."

"Each test file was prepared by downsampling the reference file to the
indicated sample rate, and then upsampled to the indicated sample rate of
either 16/44 or 24/96."

These files have been auditioned by virtually thousands of people, and still
nobody has reported signficiant deterioration of sound quality due to
downsampling to lower resolution formats, providing that lower resoltuion
format has higher resolution than 14/32. BTW, these are similar results as
those it is said to have been obtained by the BBC in the 1960's or 1970s.

If so, how do those with "golden
ears" deal with eternally wallowing in inadequately reproduced sound?

"Arny Krueger" wrote in message


"Doc" wrote in message
nk.net

If so, how do those with "golden
ears" deal with eternally wallowing in inadequately reproduced sound?

Pay them no mind if you've got more important things to worry about, and I
think we all do. They don't feel good unless they feel bad.

In article ,
"Arny Krueger" wrote:


These files have been auditioned by virtually thousands of people, and still
nobody has reported signficiant deterioration of sound quality due to
downsampling to lower resolution formats, providing that lower resoltuion
format has higher resolution than 14/32. BTW, these are similar results as
those it is said to have been obtained by the BBC in the 1960's or 1970s.

For what it's worth, I can clearly hear the difference between the
samples on your site of castanets low passed at 22kHz vs 18kHz. Using an
ABX utility, I can get above 99.5% confidence. This would indicate that
I could easily hear the difference between 14/44 and 14/32, given 32khz
sampling is equivalent to a low pass at 16kHz.

Monitoring chain nothing too special -- Motu 828mkII through HR624s in
an untreated room.

I'm also 19, though, so that may have something to do with the good HF
hearing.

-Todd

Todd Lipcon wrote:

I'm also 19, though, so that may have something to do with the good HF
hearing.

Yeah, like everything to do with it. :-)


Bob
--

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

A. Einstein

"Todd Lipcon" wrote in message


In article ,
"Arny Krueger" wrote:

These files have been auditioned by virtually thousands of people,
and still nobody has reported signficiant deterioration of sound
quality due to downsampling to lower resolution formats, providing
that lower resoltuion format has higher resolution than 14/32. BTW,
these are similar results as those it is said to have been obtained
by the BBC in the 1960's or 1970s.

For what it's worth, I can clearly hear the difference between the
samples on your site of castanets low passed at 22kHz vs 18kHz. Using
an ABX utility, I can get above 99.5% confidence. This would indicate
that I could easily hear the difference between 14/44 and 14/32,
given 32khz sampling is equivalent to a low pass at 16kHz.

If its so easy, why not give the other files a try?

Either you're hearing just the indicated difference, or your audio interface
and/or monitoring facility has problems with high frequency nonlinear
distortion.

Monitoring chain nothing too special -- Motu 828mkII through HR624s in
an untreated room.

Shouldn't be nonlinear enough to cause problems. However...

I'm also 19, though, so that may have something to do with the good HF
hearing.

That, and a relatively sheltered life could help quite a bit. However,
younger people have tried and failed for whatever reason.

In article ,
"Arny Krueger" wrote:

"Todd Lipcon" wrote in message



If its so easy, why not give the other files a try?

I haven't tried the other files with a proper ABX utility, since the one
I wrote for OSX doesn't support sample rates greater than 48k. At some
point when I have more free time I'll update it, but for now I can't do
proper ABX testing for the 96k files.

Also, by just listening to pure sine waves, I can tell that my hearing
drops off at just about 21khz, even when outputting with my 828 in 96khz
mode. This is through both my MDR7506s and my HR624s. Could be a
function of their lowpass filters, but more likely just the fact that
I'm not a bat.

I'm also 19, though, so that may have something to do with the good HF
hearing.

That, and a relatively sheltered life could help quite a bit. However,
younger people have tried and failed for whatever reason.

I'm probably fairly sheltered compared to lots of my peers. I don't go
clubbing or to rock concerts, and when I go to loud dance parties I try
to stay on the opposite side of the room from the PA and not be in that
room for more than 2 hours or so in a night. And of course I wear
hearing protection when operating a chop saw.

-Todd

"Todd Lipcon" wrote in message

In article ,
"Arny Krueger" wrote:

"Todd Lipcon" wrote in message



If its so easy, why not give the other files a try?

I haven't tried the other files with a proper ABX utility, since the
one I wrote for OSX doesn't support sample rates greater than 48k.

So, your ABX Comprator might be putting switching artifacts that you may be
unknowingly basing your judgements on. ABX Comparators are, as a rule not
artifact-free, but when they have audible artifacts, the artifacts are
themselves randomized.

At some point when I have more free time I'll update it, but for now I
can't do proper ABX testing for the 96k files.

Also, by just listening to pure sine waves, I can tell that my hearing
drops off at just about 21khz, even when outputting with my 828 in
96khz mode. This is through both my MDR7506s and my HR624s. Could be a
function of their lowpass filters, but more likely just the fact that
I'm not a bat.

Right. Also, sine-wave based listening tests don't address the *strongest*
reason that people can't hear the removal of high frequency information -
masking. Basically, what we hear is broken down into what are known as
critical bands - approximately 1/3 octave wide bands of frequencies
non-uniformly distributed across the audible range. The loudest tone in any
band tends to capture that band and obscure our perception of other tones in
that band. The frequency range of the highest band varies, but is something
like 13-18 KHz for most people. The strongest, lowest frequency in that band
tends to capture it and control what we hear in that band. So, if you are
listening to something that has very strong content at say 15 KHz, content
at 17 KHz tends to be irrelevant to your perceptions.


I'm also 19, though, so that may have something to do with the good
HF hearing.

That, and a relatively sheltered life could help quite a bit.
However, younger people have tried and failed for whatever reason.

I'm probably fairly sheltered compared to lots of my peers. I don't go
clubbing or to rock concerts, and when I go to loud dance parties I
try to stay on the opposite side of the room from the PA and not be
in that room for more than 2 hours or so in a night. And of course I
wear hearing protection when operating a chop saw.

I'm 58 and cultural norms for hearing protection have changed dramatically
over my lifetime. For example when I was in the Army, they didn't know how
to spell hearing protection, let alone provide it. There were some signs of
awareness up the chain of command, just near the end of my duty.

Todd Lipcon wrote:

I haven't tried the other files with a proper ABX utility, since the one
I wrote for OSX

Oops... is this available for others to use as well? If so, where?
Sources?

sincerely
Lars


--
lars farm // http://www.farm.se
lars is also a mail-account on the server farm.se
aim:

If so, how do those with "golden ears"
deal with eternally wallowing in inadequately reproduced sound? BRBR

Personally, I knelt on rice 1-2 hours a day from 1981 until Protools HD came
out.






please read as sarcasm
Kevin M. Kelly
"There needs to be a 12-step program for us gearheads"

deficences can be do to the actual mqastering involved, the quality of
the da and other such issues.
when cd first became commercial, many companies released their
catlogues using the LP master
with the RIAA curve intact and the players were outputting the audio
one sample out of time, L then R then L then R.

the quality of the playbck can be from inexpensive players. the issues
of reading and converting the digital to analogue can change the sound.

poor reading means error correction to replace misread data.
bad DA conversion and then the actual analogue out slurs the music.

and the inputs of the preamp and amp can be less then the audio being
sent to it.

the issue of 44.1 playback is an engineering concern.
just like those with the turntable.

buy cheap get less quality
the Linn turntable cost 10 grand, is belt driven and the tonearm is
manual
no auto set down and pick.
but it is a great sounding turntable

dale

"dale" wrote in message
ups.com...
deficences can be do to the actual mqastering involved, the quality of
the da and other such issues.
when cd first became commercial, many companies released their
catlogues using the LP master
with the RIAA curve intact and the players were outputting the audio
one sample out of time, L then R then L then R.

the quality of the playbck can be from inexpensive players. the issues
of reading and converting the digital to analogue can change the sound.

poor reading means error correction to replace misread data.
bad DA conversion and then the actual analogue out slurs the music.

and the inputs of the preamp and amp can be less then the audio being
sent to it.

the issue of 44.1 playback is an engineering concern.
just like those with the turntable.

Exactly. I think the real question for people comercial music is how audible
are the differences when played back over a typical consumer system, or over
the radio, or in a music video. Not to say that the pursuit of this level of
quality is frivolous, but it becomes less important when the target audience
that has the gear to appreciate it is a tiny fraction of your overall
audience.

Personally, I think that trying to engineer for both mono and stereo
playback is far more of a ball & chain than using 16bit @ 44.1KHz for
distribution.

Sean

"dale" wrote in message
ups.com...
deficences can be do to the actual mqastering involved, the quality of
the da and other such issues.
when cd first became commercial, many companies released their
catlogues using the LP master
with the RIAA curve intact

Mmm, no. They may have been using tapes that had been EQ'd for vinyl
mastering, but not RIAA. That was applied in the mastering process, not on
the tapes.

and the players were outputting the audio
one sample out of time, L then R then L then R.

Which gives the effect of moving one of your speakers approximately 7.8mm
closer to you than the other.

There were problems with early digital recording, plenty of them, and they
haven't been entirely solved yet. But these two items were not among them.

Peace,
Paul

Paul Stamler wrote:

"dale" wrote

and the players were outputting the audio
one sample out of time, L then R then L then R.

Which gives the effect of moving one of your speakers approximately 7.8mm
closer to you than the other.

There were problems with early digital recording, plenty of them, and they
haven't been entirely solved yet. But these two items were not among them.

That last one actually bit me in the butt. When we bought a second
player, a Philips changer, I bought it home and hooked it up to a pair
of Paradigm booksheklf speakers (Titans, I think - among those Dorsey
hates) driven by a Tandberg integrated unit. Something bothered me for
about a week. I could swear stuff was moving, horn attacks shifted as
they happend, etc. Eventually I made a call to tech info and was told
that player output the left and then the right sample. I could not
believe I heard that, but I did. I returned the Philips and bought a
Denon that doesn't do that. In the case of that particular Philips, it
was a problem. I found that ironic given Philip's role in the creation
of the format.

--
ha

hank alrich wrote:

That last one actually bit me in the butt. When we bought a second
player, a Philips changer, I bought it home and hooked it up to a pair
of Paradigm booksheklf speakers (Titans, I think - among those Dorsey
hates) driven by a Tandberg integrated unit. Something bothered me for
about a week. I could swear stuff was moving, horn attacks shifted as
they happend, etc. Eventually I made a call to tech info and was told
that player output the left and then the right sample. I could not
believe I heard that, but I did. I returned the Philips and bought a
Denon that doesn't do that. In the case of that particular Philips, it
was a problem. I found that ironic given Philip's role in the creation
of the format.

I think by the time the changers came out, Philips had stopped doing that.
The original 14-bit chipset did it, but I think the second-generation set
that came out around 1985 didn't.

More likely it was something else crappy about the converters, like uneven
response between the two channels, etc. Most converters on consumer gear
even today have enough different things wrong with them it's not worth
worrying which thing wrong is causing a given issue.
--scott

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

On 30 Jan 2005 23:47:35 -0500, (Scott Dorsey) wrote:

Most converters on consumer gear
even today have enough different things wrong with them it's not worth
worrying which thing wrong is causing a given issue.

But, but, but... Isn't digital *perfect* ?

I've been mislead. I thought it would all be easy, and free.
Like everything else in life...

Chris Hornbeck
"Don't be foolish, like the others..." _Lola Montes_, 1955

Scott Dorsey wrote:

I think by the time the changers came out, Philips had stopped doing that.
The original 14-bit chipset did it, but I think the second-generation set
that came out around 1985 didn't.

I got hold of a guy in a tech shop that was factory warranty for the
Philips line and he toldhme that the unit I had purchased did that:
first left channel and then right. And that's just how it sounded. I was
flabbergasted.

--
ha

"dale" wrote in message
ups.com
deficences can be do to the actual mqastering involved, the quality of
the da and other such issues.
when cd first became commercial, many companies released their
catlogues using the LP master
with the RIAA curve intact

Cloose conceptually, but no, not that.

There were several layers of mastering during the production of LPs,
necessitated by the fact that the dies used to press LPs had relatively
short lives and were relatively difficult to produce. A mixdown tape might
go through some pre-mastering to correct overall sonic balance - both
spectral and loudness changes might need to be made. Then the results of pre
mastering, would be further remastered to be a good fit with the various
kinds of dynamic range limitations. The results of this would be called a
cutting master. Cutting masters were then cut into lacquer, and the RIAA
pre-emphasis would be applied at this time. The lacquer was futher cloned at
the pressing plant, as a large number of die sets would be required, and
only a limited number of die sets could be made from each lacquer.

and the players were outputting the audio
one sample out of time, L then R then L then R.

This is a disturbing-sounding process, but it has surprisingly subtle
effects. Time-shared DACs have shown up in various audio products from time
to time, including the original CDP 101 CD player, some early ADAT products
(I'm told), and some computer audio interfaces.

the quality of the playbck can be from inexpensive players. the
issues of reading and converting the digital to analogue can change
the sound.

At this time even $40 optical disc players can have amazingly good technical
performance - essentially fully exploiting the capabilties of the CD audio
format. 24/192 stereo DAC chips with 90 dB dynamic range can run under $1
each in production quantities.

poor reading means error correction to replace misread data.
bad DA conversion and then the actual analogue out slurs the music.

The current standard for CD playback is really very good. Any reasonably
competent player is unlikely to be a sonic stumbling block.

Doc wrote:
I've read where supposedly those who are accutely sensitive can hear
deficiencies in 44.1 /16-bit CD's. If so, how do those with "golden ears"
deal with eternally wallowing in inadequately reproduced sound?

I have spent my life being frustrated with the fact that recordings don't
sound very much like the live event. It's getting better and better every
day, and if anything the frustration is reduced.

For the most part, though, I think we are at the point where the actual
storage medium is the least of the problems. There are huge deficiencies
in 44.1/16 bit CDs but most of them have nothing at all to do with the
recording format.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

In article et writes:

I've read where supposedly those who are accutely sensitive can hear
deficiencies in 44.1 /16-bit CD's. If so, how do those with "golden ears"
deal with eternally wallowing in inadequately reproduced sound?

They go to concerts in good halls where there's no sound reinforcement.

--
I'm really Mike Rivers )
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
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In article et
writes:

I've read where supposedly those who are accutely sensitive can hear
deficiencies in 44.1 /16-bit CD's. If so, how do those with "golden ears"
deal with eternally wallowing in inadequately reproduced sound?

On 1/30/05 11:27 AM, in article znr1107089811k@trad, "Mike Rivers"
wrote:
They go to concerts in good halls where there's no sound reinforcement.


Gosh... Do you really suppose there IS such a place Toto?



--
JV
--

On Sun, 30 Jan 2005 05:44:45 GMT, Doc wrote:
I've read where supposedly those who are accutely sensitive can hear
deficiencies in 44.1 /16-bit CD's. If so, how do those with "golden ears"
deal with eternally wallowing in inadequately reproduced sound?


Shannon's sampling theory hasn't changed since it was derrived during
WWII.

When I was working directly with telecoms and military types, the "rule
of thumb" we used was 2.5x the intended bandwidth, but that was generous
for unknown signals. The telecoms sampled at 8kHz and had been doing so
basically forever, at least in computer terms, but they'd been doing it
more or less forever and know a lot about reconstruction intelligible
voice.

By that criteria, 44.1 is a bit low for a 20kHz bandwidth, 96kHz a bit
generous. OTOH, a lot of beam forming types were delighted when,
because of the application to audio, inexpensive 100kHz multichannel ADC
and DAC chips became available.

The next time a relative gets an Ultrasound done, much of the resolution
you get nowadays is indirectly related to the pro audio market.

the think i hate most is having to scale down to 16 bits from 24.
kills me. i'd be happy at 48khz/24bit, but that's not a consumer
format.

In article . com,
wrote:
the think i hate most is having to scale down to 16 bits from 24.
kills me. i'd be happy at 48khz/24bit, but that's not a consumer
format.


I dunno, 96 dB of dynamic range sure sounds like a lot to me. If only
people would actually use it....
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

Scott Dorsey wrote:

I dunno, 96 dB of dynamic range sure sounds like a lot to me. If only
people would actually use it....

Yes, it is interesting that as the dynamic range of the equipment has
been increasing, the dynamic range being utilized has been decreasing.

"Scott Dorsey" wrote in message

In article . com,
wrote:
the think i hate most is having to scale down to 16 bits from 24.
kills me. i'd be happy at 48khz/24bit, but that's not a consumer
format.


I dunno, 96 dB of dynamic range sure sounds like a lot to me. If only
people would actually use it....

Name a room with that kind of dynamic range for acoustical instruments or
unassisted vocalists under regular performance conditions.

wrote:
the think i hate most is having to scale down to 16 bits from 24.
kills me. i'd be happy at 48khz/24bit, but that's not a consumer
format.

Both DVD-Video and DVD-Audio support 48khz/24bit:

http://www.disctronics.co.uk/technol...daud_audio.htm

--
http://www.mat.uc.pt/~rps/

..pt is Portugal| `Whom the gods love die young'-Menander (342-292 BC)
Europe | Villeneuve 50-82, Toivonen 56-86, Senna 60-94

Doc wrote:
I've read where supposedly those who are accutely sensitive can hear
deficiencies in 44.1 /16-bit CD's. If so, how do those with "golden
ears"
deal with eternally wallowing in inadequately reproduced sound?

I believe that Ruper NEve's comment on Fletcher's site adn Dr Oohashi's
research which people have refered to hear are the tip of the iceberg
in proving that digital audio is responsible for the dismal record
sales that we've seen lately.

People listen to music for a reason, and digital audio doesn't deliver
as completely as analog audio.

(And yes, I use ProTools [through and analog board] becuase I have no
choice these days)

"Mike Caffrey" wrote in message
oups.com
Doc wrote:
I've read where supposedly those who are accutely sensitive can hear
deficiencies in 44.1 /16-bit CD's. If so, how do those with "golden
ears" deal with eternally wallowing in inadequately reproduced sound?

I believe that Ruper NEve's comment on Fletcher's site adn Dr
Oohashi's research which people have refered to hear are the tip of
the iceberg in proving that digital audio is responsible for the
dismal record sales that we've seen lately.

In fact Oohashi's research is irrelevant to any general presumed failings of
digital audio because the whole context of the paper is digital audio.
Basically, the paper is about comparing between one flavor of digital to
another.

As far as Rupert Neve's comments on the Mercenary site goes, how about a
URL? I just spent 10 minutes fruitlessly going through it.

People listen to music for a reason, and digital audio doesn't deliver
as completely as analog audio.

Mike, there you are provably wrong. There never was an analog format with
the bandpass and dynamic range that we can easily obtain digitally.

Apparently you don't know that analog tape has its own brickwall filter due
to the width of the head gap. Good high speed analog tape has a brick wall
in the 22-28 KHz range, always did, still does. SACD and DVD-A
transcriptions of the best analog tapes give a clear picture of this
limitation.

(And yes, I use ProTools [through and analog board] becuase I have no
choice these days)

But you do have the choice to record at 192/24 which gives about 4 times the
bandpass of the best commercial analog tape, not to mention about 20 dB or
more dynamic range. I'm not saying you should do this as a rule, but
perhaps you should stop claiming that digital audio has limitations that it
clearly doesn't have.

I'm not saying that analog tape and digital audio sound the same, but the
reasons are due to things analog tape adds, not things that digital
necessarily takes way.

Arny Krueger wrote:

Good high speed analog tape has a brick wall
in the 22-28 KHz range, always did, still does.

Studer B67, 15 ips, -3 dB @ 30 KHz for the first ten years of its life.

This is in no way intended to argue with the rest of your treatise. But
you do keep giving analog tape machine bandwidth slightly short shrift.
g

Now, the TEAC 3340 I was given a while back...

--
ha

"hank alrich" wrote in message

Arny Krueger wrote:

Good high speed analog tape has a brick wall
in the 22-28 KHz range, always did, still does.

Studer B67, 15 ips, -3 dB @ 30 KHz for the first ten years of its
life.

OK, which is pretty close to 28 KHz, right? ;-)

And, there was the same-old, some-old gap-related brick wall just above
that, right?

I have to admit that I'm kinda overcome by all the complaining about digital
system brick wall filters, when the analog tape that some others seem to
want to deify had a pretty healthy built-in brick wall of its own. As I
recall, the group delay near the null due to gap length was fairly strong,
as well.

I am also comparing those kind of numbers to digital formats like 192/24
with ca. 93 KHz bandpass or 96/24 with ca. 45 KHz bandpass...

This is in no way intended to argue with the rest of your treatise.

OK. ;-)

But you do keep giving analog tape machine bandwidth slightly short
shrift. g

OK, I was thinking of typical numbers - those Studers were from near the top
of a relatively new pile, right?

I imagine that the last round of Otaris were pretty extended, as well.

Now, the TEAC 3340 I was given a while back...

Oh, oh! ;-)