Rob Adelman wrote in message ...
dan lavry wrote:

Dan, thanks for all of the information, though I have to admit most of
it is over my head. But somehow I am guessing that the above section is
a pretty critical piece of the puzzle.

-Rob

I am sorry. I have a massage and felt compelled to explain it with
some solid arguments. That I just got too technical. Let me rewind and
start again, though I may take some liberties to be less accurate and
more intuitive.

I someone told you that in order to draw a straight line, all you need
to know are 2 point, you will believe it. It talks to your common
sense. If some one tried to tell you that you need more points, you
will probably dismiss them.

Lets try drawing a circle through 3 points. Are 3 points enough to
draw that circle line? Slightly less intuitive is it not? But
manageable.

If I tell you that I have a curve that can go up or down or sideways
in a totally unpredictable way, you will realize that the more points
and the closer they are, the better the representation. So some folks
are saying: audio is like that complex curve, so give me more points –
increase the sample rate.

It may not be easy to grasp, but while audio is very complex, there
are some restrictions there, and it is not true that "anything goes".
The fact that we are dealing with some limited bandwidth (frequency
range) will, for example restrict that curved line representing the
sound from moving too fast (think of putting a restriction on the
maximum allowed slop). This is true for any wave, video, audio,
medical, instrumentation… The lower the bandwidth, the lower the
slope. I am not being completely accurate with slope, but higher
frequencies move faster.

The point is that the restrictions define the signal well enough so
that you do not need too many points to draw that line. Too few points
will not cut it, but you get to a certain level that allows you to
draw the line correctly. Just like 2 point for a straight line.

Some folks are trying to sell you on doubling the points, when you do
not need to. They call it 192KHz sampling. The extra points (samples)
take space, require you to double the processing power of your
machine, and in fact lower the quality of the outcome.

The argument is based on "more is better" which is often true, but not
always. Those that study EE and math know that it is not. Those that
do not have the background are just as likely to buy the BS, as they
are to buy the truth.

There are a lot of forces out there, from huge semiconductor houses to
huge workstation makers and their whole support network that have been
promoting that crock. With so much combined clout, few want to stand
up to it. And of course, such a myth gets propagated to the sales guys
that mostly lack the know how, and latch into that "more is merrier"
wrong explanation. We have a whole industry going in the wrong
direction.

But there are a lot of things in audio that are in the "gray area".
The above is not. We are dealing with fundamentals. The only argument
I can not deal with is: but I like it. Or It sounds great to me. Fine
if it does, but my point is: You do not need to go and double the data
and also double the processing to get that thing you like, If you have
a certain characteristic (distortion) you like, I can make it for you
with a 96KHz AD or lower.

I too want to improve quality, and there are things to do. But going
above 96KHz is screwing things up. Math engineering and science is on
my side of the argument. History will prove it, and hopefully very
soon. Meanwhile I am sorry to see folks pay good money to be taken to
a ride in the wrong direction.

Of course, those that got influenced to belive they are getting better
sound, are in a bind. It takes a "hack of a man" or a woman to go back
on it, to admitt you were wrong. Certainly such is the case in this
industry. And it is always "acceptble" to just say "but I like it". In
audio, you call it "an artistic decision" and no one will argue, well
almost no one...

I do not want to make anyone uncomfortable. I just want that 192 and
to have 96 accepted as morer than enough. I hoe it does and soon.
While observing some recording and mastering guys "go with the flow"
of faster is better, I am very pleased to see some top notch ears
that figured it out ""by ear". That is encouraging.

I hope this is clear and direct enough.


Dan Lavry

"dan lavry" wrote in message
om...

I do not want to make anyone uncomfortable. I just want that 192 and
to have 96 accepted as morer than enough. I hoe it does and soon.
While observing some recording and mastering guys "go with the flow"
of faster is better, I am very pleased to see some top notch ears
that figured it out ""by ear". That is encouraging.

I hope this is clear and direct enough.


Dan Lavry


Dan, you have explained your point about 192kHz problems very well!
It's sad to see some of these so-called 24-bit/192kHz converters which
really only have dynamics of about 100dB with bad distortion etc., marketed
with a 24/192 tag.
It appeals to some people because it's new, even though I must admit that I
myself didn't know about the 192kHz problems until you presented some
serious information.

Even though my head is still trying to figure out the various differences
between the variations of single- and multibit, shaping and non-shaping
converters,and their unique flaws/merits, your point is well made.

In article writes:

If I tell you that I have a curve that can go up or down or sideways
in a totally unpredictable way, you will realize that the more points
and the closer they are, the better the representation. So some folks
are saying: audio is like that complex curve, so give me more points –
increase the sample rate.

It may not be easy to grasp, but while audio is very complex, there
are some restrictions there, and it is not true that "anything goes".
The fact that we are dealing with some limited bandwidth (frequency
range) will, for example restrict that curved line representing the
sound from moving too fast (think of putting a restriction on the
maximum allowed slop).

The way I like to explain this is to imagine that you're driving a
race car on a waveform-shaped track. If that track is a perfect sine
wave, you'll have to go at a certain speed in order to complete a
cycle in a given time. If you now put some more bends in the track,
you'll be going a greater distance from end to end. If you want
to match your time for the simple curved track, you'll have to drive
faster. Put more kinks in it and you'll have to drive still faster.
When the track gets sufficiently contorted, ignoring things like
centrifical force and coefficient of friction that cause you to slow
down for turns, eventually the path will be long enough so that you
simply can't get up enough speed to get to the end of the track within
the proscribed time.

If you can somehow increase your maximum speed (like by putting a
bigger engine in the car) you can then again meet your mark. The speed
of the car represents the sample rate, the turns in the track
represent the frequency because they increase the distance traveled in
a fixed amount of time.



--
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
and reach me he double-m-eleven-double-zero at yahoo

some additional comments:

I did some measurements using an analog burst 10khz signal who is
complete not in sync with the sample clock.

To get a fine none-overshouting (pre/post) sampled signal a high
sample rate is required to get ride of the ringing (about 10 times
related to burst sine freq.).

I read some time ago in a paper that 1/100% ringing is audible....

Cheers

Hp

Tommi wrote:

It's sad to see some of these so-called 24-bit/192kHz converters which
really only have dynamics of about 100dB with bad distortion etc., marketed
with a 24/192 tag.

Well, there are other converters that are spec'd at 24/192 that offer
120+ dB of dynamic range, very fine distortion specs and very low
noise. And they sound fine.

What surprises me most these days are the folks who complain about the
sound of high res digital audio but extol LPs (vinyl) with their 30 dB
max of separation and surface noise.


--
Len Moskowitz PDAudio, Binaural Mics, Cables, DPA, M-Audio
Core Sound http://www.stealthmicrophones.com
Teaneck, New Jersey USA http://www.core-sound.com
Tel: 201-801-0812, FAX: 201-801-0912

In article writes:

Dan, you have explained your point about 192kHz problems very well!
It's sad to see some of these so-called 24-bit/192kHz converters which
really only have dynamics of about 100dB with bad distortion etc., marketed
with a 24/192 tag.

They get away with it because people see "24/192" and don't see the
actual dynamic range specifications.

In the early days of 96 kHz products, a lot converters (in both
directions) that sounded pretty good at 48 kHz didn't sound quite as
good at 96 kHz. I think those problems have been fixed, but the
numbers for S/N ratio and dynamic range are limited (at least with
good quality chips) by analog circuitry, short term clock stability,
board layout (which affects all of the above) and physics.



--
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
and reach me he double-m-eleven-double-zero at yahoo

Hp Widmer wrote in message . ..
some additional comments:

I did some measurements using an analog burst 10khz signal who is
complete not in sync with the sample clock.

To get a fine none-overshouting (pre/post) sampled signal a high
sample rate is required to get ride of the ringing (about 10 times
related to burst sine freq.).

I read some time ago in a paper that 1/100% ringing is audible....

Cheers

Hp

A burst is not a band limited signal. In fact, making a sharp corner
takes infinite bandwidth. So if you take a zero signal and all of a
sudden you "shoot up" into the first cycle of the burst, and also the
ending of it- sudenly to zero, it takes huge bandwidth.

That is why we window FFT's, and why we also the best FIR filters are
done with the window method. Loosly speaking, windowing anounts to a
very gradual taper at the start and end of the wave, sort of like fade
in and fade out that mastering and music editors do.

I am not sugesting you need to window andything, but I sugest that you
need to make sure that the burst is FILTERED with proper anti alaising
filter. If you did not filter the high frequency content of the burst,
you will have alaising.

If you filter it properly and the rest is done correctly, your
conclusion will change. If you get ANY more detail when sampling
faster there are 2 possibilities:
1. Something is wrong with the test or the setup
2. Nyquist was wrong,. Shannon was wrong, math does not work and
science are wrong.

I will not get into much back and forth regarding issues that are as
solid as the law of garvity.

I do not know why after my long post, I have to deal with such a
response. You are saying "I tested it and sampling theorm is wrong". I
would be inclined to figure out what is wrong with the test or the
setup.

Dan Lavry

Arny Krueger wrote:

You said:

"For instance, attempting to design a 4th order Butterworth high
pass filter with a cutoff of 20Hz at a sample rate of 48kHz is
difficult in 32 bit floating point but it can be done in 64
bit double precision floating point."

So let's put your claim to the test. Cool Edit Pro has a variable frequency
high pass Butterworth filter of any reasonable order including 4. It can be
run at any sample rate up to 999,999 Hz including 48 Khz and 384 KHz. AFAIK
it is implemented in 32 bit floating point arithmetic.

Aaahhh. A discussion that's near and dear to my day's work.

You *did* misinterpret what Erik said. His point was that moving to
sample rates *complicates* matters considerably. In other words, it's
something that a DSP guy dreads.

I can assure you from extensive personal experience that if you design,
say, a highpass filter of fixed order and a cutoff frequency that's very
low relative to Fs, implementing a filter with the same cutoff frequency
(in absolute Hz, *not* in rad/sample) at 2Fs is much more difficult. 8Fs
is even worse. And maintaining the same *transition* band makes it even
uglier, as the required order grows with Fs.

What do I mean by difficult? Specifically, I'm talking about the
roundoff noise power and its attendant spectral distribution. For an IIR
filter, this metric gives an indication of the filter's output's
dependence on and vulnerability to re-circulated round-off noise errors.
And it also gives some indication of the relative amplitudes of the
filter's internal states. Here's a simple experiment you can do yourself
if you have a good analysis tool like Matlab. Design a HP filter with a
cutoff of 28 Hz and a transition band of 10 Hz and a sampling rate of
44.1 kHz. Now use those same specs and do it for 88.2 kHz. Now do it for
176.4 kHz. On each of those filters, use Matlab's qfilt objects to
generate finite-wordlength equivalents and set the quantization format
to {float,32,8}, architect the thing as a direct-form filter and run a
noise-load analysis on it. You'll see that direct-form is horrible at
44.1 kHz, hopeless at 88.2 kHz, and lethal at 176.4 kHz. We're talking
about so much noise that the filter won't work.

One's first thought is to re-implement the thing as cascaded
second-order sections. The results are stil bad, terrible, useless,
respectively. Doing it as a lattice/ladder only helps a little. In fact,
the only way to make it "work" is to resort to cascaded exotic
section-optimal minimum-roundoff structures, a la Roberts and Mullis.

The point here is not that it can't be done. But one must take heroic
measures to make it work properly at higher sample rates. (By
*properly*, I mean in the worst case, stable; in the best case, as good
as the "easy" filters). The theory is very, very solid on this. Have a
look at Dick Roberts and Cliff Mullis's book -- they give the most
elegant geometric explanation of the phenomenon to date. But try the
expermient yourself. It's maddening!

(In case anybody is wondering, I have to live with these rates because
of DSD. It wasn't my choice!)

What bad thing to look for when operating it? Bad frequency response?
Nonlinear distortion? Bad phase response?

Overwhelming noise and graininess on the output, clipping, limit cycles,
grunge, etc. Just make sure you force the transition band to be tight.
Or, if you like, try a peaking filter centered at 28 Hz with a Q of 10
or greater and a boost of 8 dB. And just because you don't hear any
"nasties" doesn't mean the filter is really doing what you asked of it!
I've seen many a designer cheat and implement a fatter filter in order
to avoid the problem.

Or, it may indeed be the case that your program *does* work properly, in
which case the designer applied some real TLC to the DSP programming.
But it's still waaaaaaay harder to make it work at higher Fs.

enjoy!

Glenn @ Z-Systems

(Mike Rivers) wrote in message news:znr1069858374k@trad...
In article writes:

Dan, you have explained your point about 192kHz problems very well!
It's sad to see some of these so-called 24-bit/192kHz converters which
really only have dynamics of about 100dB with bad distortion etc., marketed
with a 24/192 tag.

They get away with it because people see "24/192" and don't see the
actual dynamic range specifications.

You are correct. First, as I stated before, all the 192 gear and IC's
I saw, and I look all the time (!!!) is specfied with A weighting.
Most of the other gear 96KHz gear is is specified A weighted. Next,
many of the AD converter companies just copy the IC specifications as
if it "exsists in mid air". You look at the finished design, including
the front end circuitry, power supply, clocks... you will be surprised
at the discrapancy there.

I am not saying the "ordinary specs" are the only thing that matter.
There are a lot of things not on the spec sheet, and should be there,
that make it or break it.

Here is an example: I am often amazed by how folks insist on ,1dB
flatness response. Not that it is difficult to do. I glad to comply
with that. But did you look at a speaker resonse latley? It is up and
down by dB's. It is a mess. Somewhere smack in the middle, there is a
crossover network that breaks that tone made out of say 500Hz into
(Assume 1.2KHz cross over for the example):

A. 500Hz fundumental and 1KHz first harmonic go to to the large cone.
B. 1.5Khz, 2KHz,.... 3rd forth and so on harmonics to the smaller
speaker cone

And guess what, there is a huge phase shift in there around the cross
over range...

And guess what else, that single say 500Hz piano note is played so
that some of the sound comes from the lower cone and the rest from the
upper cone...
If you play 100Hz, now you have the fundumntal and 10 harmonics or so
come out of the bottom large speaker...

When folks demand flatness I give it, but I do think they are barking
up the wrong tree. When the talk 192 I know they are full of it. When
they like a well preserved vinal, I at least understand some of what
they like. .

But transfering vinal to a 192 cracks me up.

BR
Dan Lavry

In article writes:

You are correct. First, as I stated before, all the 192 gear and IC's
I saw, and I look all the time (!!!) is specfied with A weighting.

I've noticed that too. It's amazing how much power supply hum goes
away with that filter, and stray clock hash too. The first computer
audio interface that I reviewed, the Echo Layla, on quick measurement
had an outrageous amount of noise, though it didn't sound noisy. A
look at the output with a scope showed lots of stuff above 22 kHz.
Applying a 20 kHz filter ahead of the meter let me get closer to the
manufacturer's published noise spec.

many of the AD converter companies just copy the IC specifications as
if it "exsists in mid air".

I've noticed that, and also just using the theoretical number of
96 or 144 dB (depending on the word length) as well.

I am not saying the "ordinary specs" are the only thing that matter.
There are a lot of things not on the spec sheet, and should be there,
that make it or break it.

The sad thing is that a lot of people who buy this stuff have nothing
to go on but specifications. When you're dealing with people with so
little experience and knowledge that they think the only difference
between a mic and a line input is the size of the connector, it's easy
to absorb a lot of irreleveant information and then be confused when
trying to sort it out.

Here is an example: I am often amazed by how folks insist on ,1dB
flatness response. Not that it is difficult to do. I glad to comply
with that. But did you look at a speaker resonse latley?

There must be something to frequency response. I hear people talk all
the time about fixing a "problem" in mastering by boosting something a
couple of tenths of a dB. I can't say as I hear it myself, but then
I'm just average.



--
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
and reach me he double-m-eleven-double-zero at yahoo

Hp Widmer wrote in message . ..
I did some measurements using an analog burst 10khz signal who is
complete not in sync with the sample clock.

To get a fine none-overshouting (pre/post) sampled signal a high
sample rate is required to get ride of the ringing (about 10 times
related to burst sine freq.).

This is what bandlimiting does. That ringing is the result of the
bandlimiting, and the stuff you are seeing is all well above 20 KHz.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

In article znr1069943194k@trad, (Mike Rivers)
wrote:


There must be something to frequency response. I hear people talk all
the time about fixing a "problem" in mastering by boosting something a
couple of tenths of a dB. I can't say as I hear it myself, but then
I'm just average.

People say a lot of things. What the those engineers should realize, is
that a tenth dB "correction" is absoultely meaningless in the real
world! No speaker, ear, room, etc. is nearly that accurate...

It's just a desire on the part of the engineer to do _something_ rather
that cut it flat.

DC

"Len Moskowitz" wrote in message

Tommi wrote:

It's sad to see some of these so-called 24-bit/192kHz converters
which really only have dynamics of about 100dB with bad distortion
etc., marketed with a 24/192 tag.

Well, there are other converters that are spec'd at 24/192 that offer
120+ dB of dynamic range, very fine distortion specs and very low
noise. And they sound fine.

What surprises me most these days are the folks who complain about the
sound of high res digital audio but extol LPs (vinyl) with their 30 dB
max of separation and surface noise.

It's got to be the sentimentality factor. It ain't the technical
performance, that's for sure!

In article writes:

There must be something to frequency response. I hear people talk all
the time about fixing a "problem" in mastering by boosting something a
couple of tenths of a dB.

It's just a desire on the part of the engineer to do _something_ rather
that cut it flat.

Wait a minute. Aren't you a mastering engineer? But then most of us
have "fixed" something by turning a control on something that we
didn't realize was bypassed.



--
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
and reach me he double-m-eleven-double-zero at yahoo

Mike Rivers wrote:

Wait a minute. Aren't you a mastering engineer? But then most of us
have "fixed" something by turning a control on something that we
didn't realize was bypassed.

Whew! I'm glad to hear you say that. I feared I was the
only one who had fallen into that trap. :-)


Bob
--

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

A. Einstein

In article znr1070019730k@trad, (Mike Rivers)
wrote:

Wait a minute. Aren't you a mastering engineer? But then most of us
have "fixed" something by turning a control on something that we
didn't realize was bypassed.

Yes, I am. One that is realistic enough to know that you can't "fix"
anything by boosting 0.1dB... I think it's more of a canine-equine show
for the client "Ooh, listen to the air when I add .2 @20k Q1" That's
bound to increase sales.

Back when everything sounded great, the eq's were in 2dB steps.....

DC

"Mike Rivers" wrote in message
news:znr1070019730k@trad...
Wait a minute. Aren't you a mastering engineer? But then most of us
have "fixed" something by turning a control on something that we
didn't realize was bypassed.

All too often.

--


Roger W. Norman
SirMusic Studio
Purchase your copy of the Fifth of RAP CD set at www.recaudiopro.net.
See how far $20 really goes.

Ah, but just the bigger engine (a step up in converter sampling) futzes with
the drag coefficient, weight and balance, and g force so that the bigger
engine doesn't turn in any better time. Like in drag racing, after 190 mph
it costs an addition 14 horse power to get even one mph more. But if you
don't spend the money to redesign the drive chain, etc., the extra horse
power won't get you going any faster.

--


Roger W. Norman
SirMusic Studio
Purchase your copy of the Fifth of RAP CD set at www.recaudiopro.net.
See how far $20 really goes.




"Mike Rivers" wrote in message
news:znr1069853646k@trad...

In article
writes:

If I tell you that I have a curve that can go up or down or sideways
in a totally unpredictable way, you will realize that the more points
and the closer they are, the better the representation. So some folks
are saying: audio is like that complex curve, so give me more points –
increase the sample rate.

It may not be easy to grasp, but while audio is very complex, there
are some restrictions there, and it is not true that "anything goes".
The fact that we are dealing with some limited bandwidth (frequency
range) will, for example restrict that curved line representing the
sound from moving too fast (think of putting a restriction on the
maximum allowed slop).

The way I like to explain this is to imagine that you're driving a
race car on a waveform-shaped track. If that track is a perfect sine
wave, you'll have to go at a certain speed in order to complete a
cycle in a given time. If you now put some more bends in the track,
you'll be going a greater distance from end to end. If you want
to match your time for the simple curved track, you'll have to drive
faster. Put more kinks in it and you'll have to drive still faster.
When the track gets sufficiently contorted, ignoring things like
centrifical force and coefficient of friction that cause you to slow
down for turns, eventually the path will be long enough so that you
simply can't get up enough speed to get to the end of the track within
the proscribed time.

If you can somehow increase your maximum speed (like by putting a
bigger engine in the car) you can then again meet your mark. The speed
of the car represents the sample rate, the turns in the track
represent the frequency because they increase the distance traveled in
a fixed amount of time.



--
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
and reach me he double-m-eleven-double-zero at yahoo

In article writes:

Yes, I am. [a mastering engineer] One that is realistic enough to know
that you can't "fix" anything by boosting 0.1dB...

Back when everything sounded great, the eq's were in 2dB steps.....

Back then, in mastering, what you "fixed" wasn't what's wrong that's
keeping the record from selling millions, you fixed what would keep
you from cutting it properly. When you got the pressings, you expected
them to be missing a little of what you heard in the recording studio.
You didn't expect them to sound a whole lot better.



--
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
and reach me he double-m-eleven-double-zero at yahoo

You're telling Dave?

--


Roger W. Norman
SirMusic Studio
Purchase your copy of the Fifth of RAP CD set at www.recaudiopro.net.
See how far $20 really goes.




"Mike Rivers" wrote in message
news:znr1070112071k@trad...

In article
writes:

Yes, I am. [a mastering engineer] One that is realistic enough to know
that you can't "fix" anything by boosting 0.1dB...

Back when everything sounded great, the eq's were in 2dB steps.....

Back then, in mastering, what you "fixed" wasn't what's wrong that's
keeping the record from selling millions, you fixed what would keep
you from cutting it properly. When you got the pressings, you expected
them to be missing a little of what you heard in the recording studio.
You didn't expect them to sound a whole lot better.



--
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
and reach me he double-m-eleven-double-zero at yahoo

(Scott Dorsey) wrote in message ...
Hp Widmer wrote in message . ..
I did some measurements using an analog burst 10khz signal who is
complete not in sync with the sample clock.

To get a fine none-overshouting (pre/post) sampled signal a high
sample rate is required to get ride of the ringing (about 10 times
related to burst sine freq.).

This is what bandlimiting does. That ringing is the result of the
bandlimiting, and the stuff you are seeing is all well above 20 KHz.
--scott

It is easier to relate to the ringing with low frequency squate waves.
That 10KHz burst is "mixing up" a lot of things.
Say you take a 1KHz square wave. Say we talk about 44.1KHz. Than there
are 11 harmonics (decaying in amplitude). For square wave only odds
exist (no even harmonics) and as you add them, you find the decaying
ringing, and yes it is at high frequency. If you go to 100Hz, you have
110 harmonics.
As scott pointed out, the added harmonics are never enough to express
a square wave. With limited bandwith, the missing high frequncy energy
shows up as deviation from perfect square, by means of high frequency
overshoot and ringing. the DIFFERENCE between the perfect and
bandlimited square wave IS HIGH FREQUENCY CONTENT.

BR

Dan Lavry

Hi Dan,

It is easier to relate to the ringing with low frequency squate waves.
That 10KHz burst is "mixing up" a lot of things.

The real question will be : Bandlimit vs. transients given from some
natural instruments and the required sample rate. In other words how
do we really hear or we sime mask above 1xkHz.

When I listen to my 192khz/24bit Stereo DVD-A Eagles (Hotel
California) or Neil Young (Harvest) there is so much more MUSIC coming
out...


Hp

"Hp Widmer" wrote in message
...
When I listen to my 192khz/24bit Stereo DVD-A Eagles (Hotel
California) or Neil Young (Harvest) there is so much more MUSIC coming
out...


Hp


Than what? How do you make a comparative when you give nothing by which the
comparison is made?

--


Roger W. Norman
SirMusic Studio
Purchase your copy of the Fifth of RAP CD set at www.recaudiopro.net.
See how far $20 really goes.

Hp Widmer wrote:
When I listen to my 192khz/24bit Stereo DVD-A Eagles (Hotel
California) or Neil Young (Harvest) there is so much more MUSIC coming
out...

How much of this is because the DVA-A is well done, and how much of it
is because the CD release was just butchered?

The CD release of Hotel California is just nasty-sounding. Harsh and
screechy, but with no real detail. It sounds nothing at all like the
original LP. That's no slam against the CD format, that is a slam against
the people who horribly bungled the CD release.

On the other hand, does it really matter? If the DVD-A sounds good, that
is a reason to go to DVD-A, whether it's a technical improvement or just
a social one. I mean, I'm the first one to acknowledge that there are serious
problems with LPs, but I'm not getting rid of my turntable because of the
number of recordings out there that sound better on LP because the CD release
was so poorly handled.
--scott
--
"C'est un Nagra. C'est suisse, et tres, tres precis."

Roger W. Norman wrote:

When I listen to my 192khz/24bit Stereo DVD-A Eagles (Hotel
California) or Neil Young (Harvest) there is so much more MUSIC coming
out...

Than what? How do you make a comparative when you give nothing by which the
comparison is made?

Then before he hits play?

Scott Dorsey wrote:


The CD release of Hotel California is just nasty-sounding. Harsh and
screechy, but with no real detail. It sounds nothing at all like the
original LP. That's no slam against the CD format, that is a slam against
the people who horribly bungled the CD release.

I have this LP. While the songs are classics, and some really good stuff
I was always disappointed with the overall sound. It seems a little 2d
to me. I never had the CD but I can imagine.

-Rob

Scott Dorsey wrote:

How much of this is because the DVA-A is well done, and how much of it
is because the CD release was just butchered?

The CD release of Hotel California is just nasty-sounding.


Yeah, but I'm pretty sure the Eagles played on the DVD-A release also.

ulysses

After you've worked with analog and digital formats for a while, you know
that 44.1k sampled reproduction does not retain the detail, air, and open
sounding quality of an analog source. This is not the case with 2Fs or 4Fs
sampling, at least not in practice. You get to recognize the closed in,
blurry quality of CDs in general vs the more natural sound of DVD-A or SACD.
Before someone rises up to smack me down with the sampling theorem, and how
1Fs is all we'll ever need because of the bandwidth the human auditory blah
blah blah, the fact is that in *practical implementation* the promise of the
sampling theorem is not fulfilled, even with the best available 1Fs dacs.
I've recently been doing some tests with 44.1k 16-bit recordings upsampled
to DSD with a new "trellis" algorithm from Philips. The playback of the
upsampled-to-DSD files through a DSD dac has better detail and transient
response than what can be heard from the original 44k file played through a
variety of the best 1Fs dacs in the industry. This may be due to the fact
that there is actually more information encoded in a good 44.1k recording (a
la the sampling theorem) than we can playback with real world
top-of-the-line PCM dacs, but more of that information does come through the
upsampled DSD playback. Not trying to prove anything, only reporting what I
can hear.


"Roger W. Norman" wrote in message
Than what? How do you make a comparative when you give nothing by which
the
comparison is made?

Remixer wrote:

Not trying to prove anything, only reporting what I
can hear.


But, but, but.....

Hp Widmer wrote in message . ..
Hi Dan,

It is easier to relate to the ringing with low frequency squate waves.
That 10KHz burst is "mixing up" a lot of things.

The real question will be : Bandlimit vs. transients given from some
natural instruments and the required sample rate. In other words how
do we really hear or we sime mask above 1xkHz.

When I listen to my 192khz/24bit Stereo DVD-A Eagles (Hotel
California) or Neil Young (Harvest) there is so much more MUSIC coming
out...


Hp

You seem to be insisting that in suport of 192KHz despite the fact
that it is against all aspects of physics and math. There is a
research artical done with the greatest care, that after searching
around for tons of cases, talks about some special case showing a
muted trumpet yielding -55dB ay around 45KHz and almost zero at 90KHz.
(96KHz AD will be enough for that).
The mics used are special measuring types. Not the kind folks use for
recording. "Ordinary" mics drop off way below the 120KHz test mic. The
speakers do also. The human ear does too.

So here is the old BS about the mysterious transients. Transients are
subject to bandwidth limitations. It may be true that a Fourier series
is based on periodic waves so it is not the way to approach
transients. So what? So you use different math. Nyquist was still
right. You bandlimit the signal to a frequency called Nyquist, and
sample over twice that rate, and you got ALL the information in the
data. Not just sine waves that last for ever. The transients you hear
are made out of energy in the same bandwidth called audio.

I know I did not say it that way earlier. I though it would be enough
to say that if the mics don't pick it, the speakers do not sound it...
it places the same limitation on the bandwidth of all music - which
includes transients.
Anything outside constant preiodic waves that started at the beginng
of time and will last for ever, includes some transient energy! I
guess you are trying to talk about "fast transients". There are huge
misconceptions around that. For example, many folks thing that a
bandlimited (say to 22KHz) 1KHz square wave rises faster (more slew
rate - volts per second) than a sine wave at 20KHz. I can go on but
will not.

I have been designing for a long time and know about transient
problems such as capacitor dialectric absorption in sample hold. That
is not a 192KHz related. It is at the oposite side - low frequencies
way within the ausio band. If it fixing it took changing from 96 to
192K sampling, you would not hear it! That is the whole point. 192KHz
will not fix a thing for audio.

Folks keep throwing misleading garbage based on lack of understanding,
and I can not answer it for ever. I need to work for a living, but I
am right, so the billion dollar conglamerates will chock me and the
truth but constatly throwing garbage at such volume that it will be
unanswered.

And whan it all gets too much for them to handel, there will always be
that last resort. That last argument that no one can stand to: It
sounded better. This argument will yield best results if you get some
real big names. Guess what- I have some real big names agreeing with
me that 96KHz is better than 192. I did not ask permision to say who,
but I put my inegrity on the line.

In either case, I came accross a big name that refused to audition 96
on the grounds that 192 contains "something" that staed with the music
at 44.1 (after decimation. Now, I will not ever calim that I know 1%
about recording when compared to that gentelman. I just wish that I
was recipicated with the same respect regarding Math and Engineering.

I would like to belive that there is a differance between the
stereophile and the pro community. Not all stereophile stuff is bad,
but much of it is very ridiculess. You do not hear a differance
between a yellow and orange cable. Arrowa on speakers is insanly
stupid. The argument is always about "It sounds better to me"...
Followed by someone getting ripped off.

Folks this is about getting our industry back on track. Double the
data. double the processing and LOWERING THE QUALITY OF CONVERSION is
a high price to pay. Science, and egineering and math on one hand, led
by greats such as Shanon and Nyquist, and the billion dollars
industial companies using every trick in the book (including avoiding
the physics and engineering books) to sell thiere BS. I belive it is
about marketing and money.

I realize that if one fell for it, it is difficult to admit. But
192KHz for audio is a crock! The only argumet left is "but I like it".
No one can stand to that one. If I tell you you added 10% distortions,
and you say "I like it", you win. So lets compromise: I say 192KHz s a
crock. You agree it is but you get to like it.

Sorry for my tone. I put out some energy to do good. I lost a big sale
because I refuse to play that 192KHz game. I try to educte folks about
it. First it was "more dots are better". Than it was "better
antialiasing". Than "transients". All very wrong and based on far from
sufficient understanding of the basics. What's next?

I am still here, but getting tired of it.

Dan Lavry

In article ,
(dan lavry) wrote:

Hp Widmer wrote in message
. ..
Hi Dan,

It is easier to relate to the ringing with low frequency squate waves.
That 10KHz burst is "mixing up" a lot of things.

The real question will be : Bandlimit vs. transients given from some
natural instruments and the required sample rate. In other words how
do we really hear or we sime mask above 1xkHz.

When I listen to my 192khz/24bit Stereo DVD-A Eagles (Hotel
California) or Neil Young (Harvest) there is so much more MUSIC coming
out...


Hp

You seem to be insisting that in suport of 192KHz despite the fact
that it is against all aspects of physics and math. There is a
research artical done with the greatest care, that after searching
around for tons of cases, talks about some special case showing a
muted trumpet yielding -55dB ay around 45KHz and almost zero at 90KHz.
(96KHz AD will be enough for that).
The mics used are special measuring types. Not the kind folks use for
recording. "Ordinary" mics drop off way below the 120KHz test mic. The
speakers do also. The human ear does too.

So here is the old BS about the mysterious transients. Transients are
subject to bandwidth limitations. It may be true that a Fourier series
is based on periodic waves so it is not the way to approach
transients. So what? So you use different math. Nyquist was still
right. You bandlimit the signal to a frequency called Nyquist, and
sample over twice that rate, and you got ALL the information in the
data. Not just sine waves that last for ever. The transients you hear
are made out of energy in the same bandwidth called audio.

I know I did not say it that way earlier. I though it would be enough
to say that if the mics don't pick it, the speakers do not sound it...
it places the same limitation on the bandwidth of all music - which
includes transients.
Anything outside constant preiodic waves that started at the beginng
of time and will last for ever, includes some transient energy! I
guess you are trying to talk about "fast transients". There are huge
misconceptions around that. For example, many folks thing that a
bandlimited (say to 22KHz) 1KHz square wave rises faster (more slew
rate - volts per second) than a sine wave at 20KHz. I can go on but
will not.

I have been designing for a long time and know about transient
problems such as capacitor dialectric absorption in sample hold. That
is not a 192KHz related. It is at the oposite side - low frequencies
way within the ausio band. If it fixing it took changing from 96 to
192K sampling, you would not hear it! That is the whole point. 192KHz
will not fix a thing for audio.

Folks keep throwing misleading garbage based on lack of understanding,
and I can not answer it for ever. I need to work for a living, but I
am right, so the billion dollar conglamerates will chock me and the
truth but constatly throwing garbage at such volume that it will be
unanswered.

And whan it all gets too much for them to handel, there will always be
that last resort. That last argument that no one can stand to: It
sounded better. This argument will yield best results if you get some
real big names. Guess what- I have some real big names agreeing with
me that 96KHz is better than 192. I did not ask permision to say who,
but I put my inegrity on the line.

In either case, I came accross a big name that refused to audition 96
on the grounds that 192 contains "something" that staed with the music
at 44.1 (after decimation. Now, I will not ever calim that I know 1%
about recording when compared to that gentelman. I just wish that I
was recipicated with the same respect regarding Math and Engineering.

I would like to belive that there is a differance between the
stereophile and the pro community. Not all stereophile stuff is bad,
but much of it is very ridiculess. You do not hear a differance
between a yellow and orange cable. Arrowa on speakers is insanly
stupid. The argument is always about "It sounds better to me"...
Followed by someone getting ripped off.

Folks this is about getting our industry back on track. Double the
data. double the processing and LOWERING THE QUALITY OF CONVERSION is
a high price to pay. Science, and egineering and math on one hand, led
by greats such as Shanon and Nyquist, and the billion dollars
industial companies using every trick in the book (including avoiding
the physics and engineering books) to sell thiere BS. I belive it is
about marketing and money.

I realize that if one fell for it, it is difficult to admit. But
192KHz for audio is a crock! The only argumet left is "but I like it".
No one can stand to that one. If I tell you you added 10% distortions,
and you say "I like it", you win. So lets compromise: I say 192KHz s a
crock. You agree it is but you get to like it.

Sorry for my tone. I put out some energy to do good. I lost a big sale
because I refuse to play that 192KHz game. I try to educte folks about
it. First it was "more dots are better". Than it was "better
antialiasing". Than "transients". All very wrong and based on far from
sufficient understanding of the basics. What's next?

I am still here, but getting tired of it.

Dan Lavry


While I don't even dream to have even a portion of the technical
knowledge and experience that you have, Dan, I have done emperical tests
of the same material at 48, 96 and 192k.

We used acoustic sources; piano, light percussion (bells, shakers,
etc.), acoustic guitar, cymbals, room tone and a ping-pong match. Every
person involved (5 of us) agreed that there was no comparison; all 192k
material from every source sounded far better than the others.

After listening to the 192k material, we were able to pick out
undesirable characteristics at the lower sampling rates that were not
that evident before. 96k had a slight hardness in the upper mids and
48k was really closed, small and hard (not scientific descriptions, I
admit) with even more of the upper mid edgyness. 192k had none of the
hardness that the lower sample rates exhibited. After listening to the
192k stuff, 48k was practically unlistenable and 96k only slightly
moreso. In blind tests, the musicians especially picked the 192k
material every time.

So why is this? I always thought it was because of the higher sampling
rate but if you can provide an alternative explanation I'm certainly
open to it.

Until then, I've got to say that 192k sounds A LOT better to me than
anything else.

--
Bobby Owsinski
Surround Associates
http://www.surroundassociates.com

"Remixer" wrote in message ...

Before someone rises up to smack me down with the sampling theorem, and how
1Fs is all we'll ever need because of the bandwidth the human auditory blah
blah blah...

And the laws of garvity? and basic math? and the whole of science just
because you hear something good enough to have every thing wrong. This
is pretty disrespectfull to engineering and math.

Why do not figure what is it about your setup blha blah blah blah...

the fact is that in *practical implementation* the promise of the
sampling theorem is not fulfilled, even with the best available 1Fs dacs.

We are talking 192KHz fs, not a 1fs DA. It has been years simce 1fs
DA. Stay on the subject. You are mixing it up blah blah blah blah

I've recently been doing some tests with 44.1k 16-bit recordings upsampled
to DSD with a new "trellis" algorithm from Philips. The playback of the
upsampled-to-DSD files through a DSD dac has better detail and transient
response than what can be heard from the original 44k file played through a
variety of the best 1Fs dacs in the industry.

The best 1fs Dac in the in the industry is long gone. Wake up to
upsampling. The problem with 1 bit DA was the need for exreamly high
order anti imaging filter. So we went to upsdampling. Also, mixing 16
bits into the argument is out of place. Clearly 20 bits is better than
16. We are talking sample rates.

This may be due to the fact
that there is actually more information encoded in a good 44.1k recording (a
la the sampling theorem) than we can playback with real world
top-of-the-line PCM dacs, but more of that information does come through the
upsampled DSD playback. Not trying to prove anything, only reporting what I
can hear.

I really do not think you know how significent the sampling theorm is.
It is not about a couple of jerks drinking beer a spouting garbage. It
is about the fundumentals of modern signal theory, by great minds. Not
much of the electronics around you would work if the therory was
wrong. Which camp do I respect? Reality science math and engineering.
And a few top notch audio engineers that know their stuff.

Dan Lavry



"Roger W. Norman" wrote in message
Than what? How do you make a comparative when you give nothing by which
the
comparison is made?

dan lavry wrote:

"Remixer" wrote in message ...

Before someone rises up to smack me down with the sampling theorem, and how
1Fs is all we'll ever need because of the bandwidth the human auditory blah
blah blah...

And the laws of garvity?

Hey, dood, I dropped a feather and a BB that weighed the
same from my window yesterday and they definitely did _not_
fall at the same rate. Those Norton and Bernstein guys got
something wrong.

Seriously, it is a pleasure having someone with your
recognized authority debunking all the nonsense that plugs
the gullet here. I wish you luck in convincing people that
they should be looking for the factors that are really
causing the perceived differences.


Bob
--

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

A. Einstein

"dan lavry" wrote in message
om...
And the laws of garvity? and basic math? and the whole of science just
because you hear something good enough to have every thing wrong. This
is pretty disrespectfull to engineering and math.

Why do not figure what is it about your setup blha blah blah blah...


I realize my use of "blah blah blah" was sufficiently ambiguous to offend
just about anybody, but it was not intended for Mr. Lavry and was not meant
to disparage any of the well-reasoned arguments he has put forth in this
thread.


We are talking 192KHz fs, not a 1fs DA. It has been years simce 1fs
DA. Stay on the subject. You are mixing it up blah blah blah blah


This thread has already split off into tangents. My post was in response to
Mr. Norman's questioning of HP's ability to recognize the perceived
superiority of a DVD-A without the benefit of a direct comparison under
scientifically controlled circumstances. (That's why I back quoted Mr.
Norman's post.) Sorry if it was taken any other way.



The best 1fs Dac in the in the industry is long gone. Wake up to
upsampling. The problem with 1 bit DA was the need for exreamly high
order anti imaging filter.

The pcm dacs in my test were all oversampling dacs. I guess I should have
been more explicit in comparing the DSD upsampling of a 44.1k source against
the same source played through an upsampling PCM dac. Yet another tangent
and another target for a flame war.


I really do not think you know how significent the sampling theorm is.
It is not about a couple of jerks drinking beer a spouting garbage. It
is about the fundumentals of modern signal theory, by great minds. Not
much of the electronics around you would work if the therory was
wrong. Which camp do I respect? Reality science math and engineering.
And a few top notch audio engineers that know their stuff.


Again, no attempt to question the validity of a proven mathematical theorem,
but it is all too often used as a blunt instrument to bludgeon those who
find fault with 1Fs recording and reproduction, upsampled or not.


Mr Lavry, please don't get caught by the remarkable power of newsgroups to
get one's dander up. With so many cross-currents going on at once, and
without the benefit of nuances, it is easy to take offense when none was
intended, or even directed.. Your recent posts have already caused me to
rethink a lot of what I have taken for granted about 4Fs. Take it all in
stride and keep contributing. Every little bit of good information helps
somebody somewhere in spite of the punting, trolling, and sniping that does
go on.

dan lavry wrote:

And the laws of garvity? and basic math? and the whole of science just
because you hear something good enough to have every thing wrong. This
is pretty disrespectfull to engineering and math.

The problem is that on paper, you could have 2 sets of criteria, one
seeming to be far more significant than the other. But in reality, the
lesser one could have far more significance in the way the brain
interprets it for reasons we just don't understand.

-Rob

"Remixer" wrote in message
...

I've recently been doing some tests with 44.1k 16-bit recordings upsampled
to DSD with a new "trellis" algorithm from Philips. The playback of the
upsampled-to-DSD files through a DSD dac has better detail and transient
response than what can be heard from the original 44k file played through a
variety of the best 1Fs dacs in the industry.

First of all, you don't have a 1fs dac. 64fs is entry level, yours is
probaby more. Second, that "better detail and transient response" could
just be distortion that you like better. Nothng wrong with that. A
little 3rd harmonic added sounds like detail.... How much new music
would you say the upsampling creates?

I guess the question is: If you run it through the trellis 20 times,
does it just get better and better?

DC

First of all, you don't have a 1fs dac. 64fs is entry level, yours is
probaby more. Second, that "better detail and transient response" could
just be distortion that you like better. Nothng wrong with that. A
little 3rd harmonic added sounds like detail.... How much new music
would you say the upsampling creates?

I guess the question is: If you run it through the trellis 20 times,
does it just get better and better?

DC

I don't think it's 3rd harmonic distortion, it doesn't really sound like
that well known effect, (I've been doing this since before the Aphex Aural
Exciter and HEDD) and doubtful that Philips would let enough distortion
creep into their rather pricey trellis algorithm to make such an audible
difference.

"Bobby Owsinski" wrote in message
news
While I don't even dream to have even a portion of the technical
knowledge and experience that you have, Dan, I have done emperical tests
of the same material at 48, 96 and 192k.

We used acoustic sources; piano, light percussion (bells, shakers,
etc.), acoustic guitar, cymbals, room tone and a ping-pong match. Every
person involved (5 of us) agreed that there was no comparison; all 192k
material from every source sounded far better than the others.

After listening to the 192k material, we were able to pick out
undesirable characteristics at the lower sampling rates that were not
that evident before. 96k had a slight hardness in the upper mids and
48k was really closed, small and hard (not scientific descriptions, I
admit) with even more of the upper mid edgyness. 192k had none of the
hardness that the lower sample rates exhibited. After listening to the
192k stuff, 48k was practically unlistenable and 96k only slightly
moreso. In blind tests, the musicians especially picked the 192k
material every time.

So why is this? I always thought it was because of the higher sampling
rate but if you can provide an alternative explanation I'm certainly
open to it.

Until then, I've got to say that 192k sounds A LOT better to me than
anything else.

Where do I begin?
What converters did you use? What speakers? Mics? Preamps? How can you say
anything about the characteristics of different sample rates without any
valid data?
It's like saying your kitchen television has a better picture than the
monitors at Lucasfilm, no one can argue about your opinion, but there is no
science backing it up.

Presumably they used the same converters at each sample rate, eliminating
the converter variable, and compared with the feed from the studio,
eliminating the mic and pre-amp variables. BTW, when you go to the
supermarket and ask for change for your twenty do you demand proof of the
validity of integer arithmetic? At least some of the people who post here
are professional listeners who don't need to prove the earth is a planet to
take a walk. Dismissing the reasoned observations of working engineers
because they're not EEs under a double-blind test is such an old saw and not
particularly productive.




"Tommi" wrote in message
news

Where do I begin?
What converters did you use? What speakers? Mics? Preamps? How can you say
anything about the characteristics of different sample rates without any
valid data?
It's like saying your kitchen television has a better picture than the
monitors at Lucasfilm, no one can argue about your opinion, but there is
no
science backing it up.

Bobby Owsinski wrote in message .. .

In blind tests, the musicians especially picked the 192k
material every time.

So you are saying you were able to pick apart the 48/96/192 KHz
material, under blind conditions, with no problems at all, being the
differences evident, aren't you? This is an extraordinary claim, that
I'd like to know more about. Were the test samples properly level
matched ( 0.1 dB difference) and time aligned? Was the test
double-blind? How many trials and correct identifications did you get?
Could you give us more information about the equipment used at the
test? Could you provide us with same of the samples used at the test?