A friend of mine signed me up for a catalog from the "Audio
Advisor".  He said I deserved this - I was afraid to ask what he
meant by that!  Spend a few minutes looking over this
site:  http://www.audioadvisor.com/  Be sure to check out their
"Power cords" at: http://home-audio.audioadvisor.com/search?w=Power+Cords

Burt, K6OQK

Burt I. Weiner Associates
Broadcast Technical Services
Glendale, California  U.S.A.
biwa@att.net
www.biwa.cc
K6OQK

I must get one of their line cords to see if it will improve my timing
system!!

You just have to laugh at this nonsense.

Rob K

-----Original Message-----
From: time-nuts-bounces@febo.com [mailto:time-nuts-bounces@febo.com] On
Behalf Of Burt I. Weiner
Sent: 07 May 2012 15:39
To: time-nuts@febo.com
Subject: [time-nuts] Oh dear

A friend of mine signed me up for a catalog from the "Audio Advisor".  He
said I deserved this - I was afraid to ask what he meant by that!  Spend a
few minutes looking over this
site:  http://www.audioadvisor.com/  Be sure to check out their "Power
cords" at: http://home-audio.audioadvisor.com/search?w=Power+Cords

Burt, K6OQK

Burt I. Weiner Associates
Broadcast Technical Services
Glendale, California  U.S.A.
biwa@att.net
www.biwa.cc
K6OQK

time-nuts mailing list -- time-nuts@febo.com To unsubscribe, go to
https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts
and follow the instructions there.

In message 226574.14407.qm@smtp104.sbc.mail.mud.yahoo.com, "Burt I. Weiner" w
rites:

I always wondered how the distortion could stop right at the power
outlet, but I see that somebody has cornered that market now.

Next step will be to try to sell them electricity produced on
turbogenerators aligned to the earths magnetic field in order
to deliver minimal low unharmonic distotion...

--
Poul-Henning Kamp      | UNIX since Zilog Zeus 3.20
phk@FreeBSD.ORG        | TCP/IP since RFC 956
FreeBSD committer      | BSD since 4.3-tahoe
Never attribute to malice what can adequately be explained by incompetence.

Nope.

Any Audiophool knows green electricity sounds much better, without any
artifacts produced by those annoying carbon atoms in coal, oil, or natural
gas, rattling around producing annoying distractions.

-John

================

On 5/7/12 7:39 AM, Burt I. Weiner wrote:

Well.. this is where folks on this list can do the world a service..

The whole thing about timing, stability, phase noise, Allan deviations,
etc. is complex, and it's tricky to come up with easy to understand,
short, descriptions of "why using a Rb for your CD player is BS".

We've all had to learn this stuff, and we do it in different ways, so
maybe the collective hive-mind is a good way to come up with decent
responses (after the initial wave of "can you believe it")

It's like explaining RF exposure limits.  There's a certain amount of
physics you have to know in order to understand how the limits work.

Most people do understand what's BS and what's not, once they understand
why.

-> the recent GPS filtering thing.. it took a YEAR for someone in the
PNT community to finally come up with a good, simple explanation of why
L^2 arguments were invalid.  And it comes down to the fact that GPS
isn't a communication link, so you can't use that conceptual model to
analyze it.  Once you get that, then people go "oh! That's why we can't
do that and have it still work"

And, on a more technically sophisticated level, there's lots of
engineers who are still wrapping their heads around the duality of time
domain (ADEV) and frequency domain (Phase noise) measurements, and when
you might use one or the other.  I've found a lot of good stuff on this
list for explaining it (and improving my own understanding.. nothing
like needing to explain it to someone else to test your own conceptual
understanding)

Interestingly, setting someone up with a counter, timelab, and a not so
hot function generator and letting them record and play for a couple
days (or over the weekend) is a great way. You see things like diurnal
variation, the HVAC cycling on and off, the sun shining through the window.

The spectrum analyzer does the phase noise thing fairly well (although
not for "close in"), and concepts like reciprocal mixing from a noisy LO
gunking up your narrow band signal are pretty obvious.

After that it's practical applications..

Just how bad can the noise be for a particular application?  Are you
interested in integrated jitter?

And for those who want a good debunking article to show to their
non-technical friends:

http://www.audioholics.com/education/cables/power-cables

On 5/7/2012 8:01 AM, Poul-Henning Kamp wrote:

In message 52252.12.6.201.2.1336403114.squirrel@popaccts.quikus.com, "J. Fors
ter" writes:

Unless, of course, you burn pure diamonds...

--
Poul-Henning Kamp      | UNIX since Zilog Zeus 3.20
phk@FreeBSD.ORG        | TCP/IP since RFC 956
FreeBSD committer      | BSD since 4.3-tahoe
Never attribute to malice what can adequately be explained by incompetence.

In message 4FA7E639.9090604@earthlink.net, Jim Lux writes:

You seem to be working under the assumption that they care about
the measurable reality.

They do not.

This is about bling and about being better than the brother-in-law
at something, it has nothing to do with sound.

--
Poul-Henning Kamp      | UNIX since Zilog Zeus 3.20
phk@FreeBSD.ORG        | TCP/IP since RFC 956
FreeBSD committer      | BSD since 4.3-tahoe
Never attribute to malice what can adequately be explained by incompetence.

This is the most ridiculous discussion in the history of this group.  If
anyone could sympathize with the need for super-timing on audio it should be
those of you who think you need cesium clocks in your homes.

-RL

Robert Lutwak | SymmetricomR, Inc.
Chief Scientist

-----Original Message-----
From: time-nuts-bounces@febo.com [mailto:time-nuts-bounces@febo.com] On
Behalf Of Burt I. Weiner
Sent: Monday, May 07, 2012 10:39 AM
To: time-nuts@febo.com
Subject: [time-nuts] Oh dear

A friend of mine signed me up for a catalog from the "Audio
Advisor".  He said I deserved this - I was afraid to ask what he
meant by that!  Spend a few minutes looking over this
site:  http://www.audioadvisor.com/  Be sure to check out their
"Power cords" at: http://home-audio.audioadvisor.com/search?w=Power+Cords

Burt, K6OQK

Burt I. Weiner Associates
Broadcast Technical Services
Glendale, California  U.S.A.
biwa@att.net
www.biwa.cc
K6OQK

time-nuts mailing list -- time-nuts@febo.com
To unsubscribe, go to
https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts
and follow the instructions there.

A crummy crystal oscillator zero beated to WWV is good to 1 in 10E6, a Rb
disciplined to GPS maybe 1 in 10E11.

Do you seriously think you, or anybody, can hear a pitch difference of
0.001 Hz in the audio range?

A quartz crystal is plenty good for any audio application, IMO.

-John

=============

As an Audiophile and T-Nut I am often torn by what could affect sound quality, but I have realized that there are many things that affect sound that as engineers we have not learned to define. As a scientist I look forward to the day we can accurately rate how equipment will sound in mathematical terms. I fact in the next month I will be working on a project that covers both my Time Nut and Audio passions with one of the music industries leading engineers Gus Skinas who has lead the development of the SACD. Jitter is a concern in digital audio and we are going to use a Cesium standard and low phase noise clean-up oscillator during recording and playback to determine the degree that timing affects sound quality. I am still not ready to spent thousands on power cords.

Thomas Knox

I've been thinking I should be selling motor/generators to this
idiots.  Basically it is just an AC alternator that is belt driven by
an electric motor.  It produces very clean power as there is no
electrical connection to the grid.  The rubber belt is
non-conductive.  It is useless and not needed but at least uses "real
science" and works as advertised.

Not to try and one up you or anything but... I remember using 7th ed
on a Vax780.  It was a novelty we had no real use for it.  Must have
beed about the time timeframe as your Zilog machine.

Chris Albertson
Redondo Beach, California

On Mon, 07 May 2012 08:20:55 -0700
Dan Rae danrae@verizon.net wrote:

The thing is, that an Rb is good for one thing: Have a long term
stable and accurate frequency source that is better than 1 to some
billions for measurement or other stuff that take more than a few
hours or have to be repeated exactly in a couple of weeks.

For audio, you need a frequency source that is stable over a couple
of hours (probably a working day) and shows "low" jitter. Where as low
jitter is quite high in time-nuts terms and stable not stable at all.
A cycle-to-cylcle jitter of a couple of ns is not audioable at all,
but any Rb will have a much lower jitter. Or to have a different look at it,
you want to have very low phase noise, as this phase noise is mixed in
over the ADCs into your signal. But as we know, the phase noise of
an Rb is not defined by the Rb physics package, but by the OCXO they use.
(yes i know that the close in phase noise is defined by the reference
and not by the OCXO, but the "base level" is the OCXO, not the reference)

As for stability. You want the instruments to sound the same over an
recording. Ie the human ear has to preceive the recorded sound as the
same. The frequency resolution of the human ear is somewhere around 3Hz.
This makes for 150ppm (at 20kHz). Even a 32kHz tuning fork crystal
achieves an absolute accuracy that is better than this. Its stability is much
better than this....
Of course, you want to have enought headroom for other non ideal components.
So, lets say, go for a factor of 10, then we are at 15ppm. For absolute
accuracy, that's already a good XO. For stability, still most XO should
do that.

Or to say it differently: Using some good OCXO with low or very low
phase noise would be more than enough for even the most high end
audio equipment. You don't even have to discipline it, as a even
quite bad OCXO has variations much lower than 1ppm, which is definitly
not something anyone can hear.

IMHO getting a 20-50USD OCXO from ebay, some good, low noise power supply
(audio power supplies with low noise in the <40kHz region), some distribution
amplifier with low noise figure and you are set. All in all probably at
a cost of 200-300USD including rack mount. If you want to have "high fidelity"
you can use an GPSDO to get your OCXO within a couple mHz.

To summarize: Nobody here does want to insult anyone who does professional
audio recordings. But having the knowledge of what the stability and
accuracy numbers for an ordinary Rb mean, and being able to put that into
perspective with the not so good capabilties of the human sensory systems,
one wonders why people spend an awfull lot of money for something that has
no audiable effect over something a lot cheaper. Not to mention that other
things have a much higher impact on audio quality than the reference
oscillator: Like temperature and humidity during recording (do you control
them as well to the ppm level?), or the tuning of the instruments which
wanders quite a bit during use.

			Attila Kinali

--
Why does it take years to find the answers to
the questions one should have asked long ago?

Actually the numbers are quite real, play with the math, a small amount of jitter in a DAC (X) can
have a large difference (Y) when sampling a complex wave form especially
in the audiophile world where the sound of 24bit dac 16,777,216 discrete levels is clearly superior to older 16 bit dac 65,536 possible levels in 44.1 KHz to 192 KHz formats.

Thomas Knox

Actually in digital audio playback timing is just as important except that there is no was to remove jitter during poor recordings.
Thomas Knox

On Mon, 7 May 2012 10:02:25 -0600
Tom Knox actast@hotmail.com wrote:

Yes, i know that jitter is a pain when it comes to ADCs, but keep
in mind that your audio ADC does have a jitter of a couple
100ps itself. If it's a high end ADC that is. The standard ADCs are usually
in the ns range. For a normal 10MHz XO you measure the jitter in in the lower
10ps at most, a good one at lower than 1ps cycle-to-cycle. Of course, you have
to keep the clock signal clean of any disturbance that might add modulations
to it. But that's a matter of keeping the power supply clean and having the
signal shielded. It's not an inherent property of an Rb to have low jitter.
And as we all know from the recent  hype on the FE-5860As and the
following measurements, not all Rb's are low jitter.

		Attila Kinali

--
Why does it take years to find the answers to
the questions one should have asked long ago?

On Mon, 7 May 2012 18:19:19 +0200
Attila Kinali attila@kinali.ch wrote:

Addendum: Just to make sure it doesn't sound like i think that
engineering audio devices is easy. Dealing with audioable frequencies
is probably one of the most tedious tasks you can give to an electrical
engineer these days. You have to deal with very low frequencies and you
are not allowed to do averaging...

		Attila Kinali

--
Why does it take years to find the answers to
the questions one should have asked long ago?

It has nothing to do with engineering.

"Artists", and I use the word with a huge bag of salt, are often Prima
Donnas. They are under the illusion that their works are masterpieces,
because they sell millions of copies on iTunes or elsewhere, or theit
concerts are sold out in two minutes. So, naturally, every nuance of their
work needs THE most elaborate equipment to reproduce it in every
breathless detail.

So, to cater to the "talent", studios build bigger, more impressive,
facilities, to attract them.

It's almost entirely a marketing enterprise. The hucksters leading the
gullible at all levels.

I have a friend who is very into classical music. He spent tens of
thousands on a sound system. I then suggested he spend a few hundred and
go listen to the Boston Symphony live and in person. He was really bummed
out for months afterwards. Now he has taken up collecting records, yes
vinyl.

Go figure,

-John

==================

[snip]

If you take into consideration that the best currently available DACs,
also true for analog circuits, have a dynamic range about 120-126dB, the
last 3-4 bits are quite irrelevant (random noise mostly)... a good 20bit
DAC already pushes the limits.
The marketingdroids swarming for the newest "32" bitters is even more
ludicrous.
On the other side, the dynamic range of the ear (if you care the least
for the future of your hearing), and of the quietest available listening
spaces, hardly gets to 100dB...

Of course, for the DSPs involved in the signal chain, 32bits integer
math might not be enough, due to rounding errors.

On 5/7/2012 7:02 PM, Tom Knox wrote:

Suppose you have a perfect, ideal clock that puts out 'convert' pulses at
an exact rate is used to strobe a high precision A/D.

Now suppose you add jitter to that perfect clock so that the rate stays
the same but time interval between successive pulses varies randomly
between P(1-x) and P(1+x).

How big would x have to be before anyone could detect any difference in
the sound?

I have my opinion, but what is yours and why?

-John

===============

On Mon, May 7, 2012 at 8:59 AM, Attila Kinali attila@kinali.ch wrote:

You are right about that.  But there are better clocks at 1/10th of the price.

Also in a recording studio, many times you need to phase lock to an
existing source and you do NOT want to be dead-on to some specific
frequency.  Jitter matters more then being frequency accurate.  And
end use would never hear a 1E-6 absolute frequency error, and I mean
"never".  But relative frequency errors and jitter is audible

As for audiophool's home playback systems there is no need at all for
an Rb clock.  They would do much better with an $50 OCXO in a box.

Chris Albertson
Redondo Beach, California

Hi

Be careful when you talk about jitter of any device, OCXO's included. There
is always an implied bandwidth in the conversion of phase noise to jitter.
If you extend the bandwidth down low enough (as in low audio) the jitter
goes up quite a bit. In the case of audio, jitter at low frequencies just
might be something to worry about.

Bob

-----Original Message-----
From: time-nuts-bounces@febo.com [mailto:time-nuts-bounces@febo.com] On
Behalf Of Attila Kinali
Sent: Monday, May 07, 2012 12:19 PM
To: Discussion of precise time and frequency measurement
Subject: Re: [time-nuts] Oh dear

On Mon, 7 May 2012 10:02:25 -0600
Tom Knox actast@hotmail.com wrote:

Yes, i know that jitter is a pain when it comes to ADCs, but keep
in mind that your audio ADC does have a jitter of a couple
100ps itself. If it's a high end ADC that is. The standard ADCs are usually
in the ns range. For a normal 10MHz XO you measure the jitter in in the
lower
10ps at most, a good one at lower than 1ps cycle-to-cycle. Of course, you
have
to keep the clock signal clean of any disturbance that might add modulations
to it. But that's a matter of keeping the power supply clean and having the
signal shielded. It's not an inherent property of an Rb to have low jitter.
And as we all know from the recent  hype on the FE-5860As and the
following measurements, not all Rb's are low jitter.

		Attila Kinali

--
Why does it take years to find the answers to
the questions one should have asked long ago?

time-nuts mailing list -- time-nuts@febo.com
To unsubscribe, go to
https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts
and follow the instructions there.

Nearly all modern recordings are "multiple mono". One microphone per instrument if not more. Multiple overdubs. If high ticket artists are collaborating, they may be recorded at different times. (Bruce Springsteen and Rosanne Cash duet for example.) They want a high bit depth so the final product doesn't have a high background noise.

The classic back of the envelope calculation regarding clock jitter is based on 44.1KHz sampling and a 20KHz sine wave. Take the maximum slew rate of the sine wave and the timing uncertainty (jitter), then compare to a LSB. It doesn't take much jitter even at 16 bits to be significant.

Modern ADCs are MASH. I don't know the analog to the argument for that technology.

-----Original Message-----
From: MailLists lists@medesign.ro
Sender: time-nuts-bounces@febo.com
Date: Mon, 07 May 2012 19:31:10
To: Discussion of precise time and frequency measurementtime-nuts@febo.com
Reply-To: Discussion of precise time and frequency measurement
time-nuts@febo.com
Subject: Re: [time-nuts] Oh dear

If you take into consideration that the best currently available DACs,
also true for analog circuits, have a dynamic range about 120-126dB, the
last 3-4 bits are quite irrelevant (random noise mostly)... a good 20bit
DAC already pushes the limits.
The marketingdroids swarming for the newest "32" bitters is even more
ludicrous.
On the other side, the dynamic range of the ear (if you care the least
for the future of your hearing), and of the quietest available listening
spaces, hardly gets to 100dB...

Of course, for the DSPs involved in the signal chain, 32bits integer
math might not be enough, due to rounding errors.

On 5/7/2012 7:02 PM, Tom Knox wrote:

time-nuts mailing list -- time-nuts@febo.com
To unsubscribe, go to https://www.febo.com/cgi-bin/mailman/listinfo/time-nuts
and follow the instructions there.

On Mon, 07 May 2012 08:11:53 -0700
Jim Lux jimlux@earthlink.net wrote:

You mean the "Don't GPS your Rb" thread?

Uhm.. i don't understand at all. Could you give some pointers or explain
what L^2 and the rest is?

Thanks in advance

		Attila Kinali

--
Why does it take years to find the answers to
the questions one should have asked long ago?

On 05/07/2012 07:01 PM, Attila Kinali wrote:

Ligthsquared and their LTE system that threatend to noise out GPS from
US. Loads of messages on that on the list already.

Cheers,
Magnus

That was a big problem with the dynamic range of tape recorders, which
had to be solved with noise reduction circuits. Even good 16 bit ADCs
have a higher DR than the SNR of most instruments in quiet recording
studios. With the mixing of multiple dubs, the main problem is the
summed background noise, not that of the ADCs.
When doing the mix digitally, a DAW with higher bit depth is needed, to
conserve the DR: 16 tracks need another 4 bits. The downmix can then be
truncated to the final media bit depth (eventually with some dither
added, if not self-dithered due to noise).
The main problem with the old CD format wasn't actually the DR, the SR
was chosen too low.

One of the famous audiophile studios (Chesky Records) expressly avoids
overdubbing, and postprocessing, and puts accent on the microphone
placement. That's real art, unlike some "sound engineer" using heavy
processing, and turning up the compression control, for a "louder sound".

"Modern" AD/DA-Cs are mostly sigma-delta for technological, and cost
reasons. The better ones are also multi-bit...

On 5/7/2012 7:59 PM, lists@lazygranch.com wrote:

On Mon, 07 May 2012 19:07:12 +0200
Magnus Danielson magnus@rubidium.dyndns.org wrote:

Oh.. right... I completely forgot about that now that it's over.
(not to mention that it didn't interest me that much being on an
different continent and all)

		Attila Kinali

--
Why does it take years to find the answers to
the questions one should have asked long ago?

In message D2251F0F290D4B1AB54E1A4DBA34500E@vectron.com, "Bob Camp" writes:

Not with the kind of physical laws I live in.

At low audio frequencies, say 100 Hz, you have at least 441 samples
per period of audio, and the Y-difference from one sample to the
next is so small, that no amount of jitter will have sonic impact.

At a 20 kHz frequency however, you have sign reversal from sample
to sample and moving a sample in X has very high impact on the
energy of that and the surrounding samples.

This is exactly why we use oversampling in the first place:  You
get more gentle slopes from sample to sample which means that
the jitters effect is attenuated in the result.

The place where this audio-jitter-homoepathy comes from, is the
first generation of Philips CD players, CD-100 etc, which had
"jitter" come up from the poor mechanics, because there were
insufficient buffering before the de-interleaver.

--
Poul-Henning Kamp      | UNIX since Zilog Zeus 3.20
phk@FreeBSD.ORG        | TCP/IP since RFC 956
FreeBSD committer      | BSD since 4.3-tahoe
Never attribute to malice what can adequately be explained by incompetence.

In message 62172.12.6.201.2.1336409319.squirrel@popaccts.quikus.com, "J. Fors
ter" writes:

You have to tell us the sampling frequency before we can answer.

--
Poul-Henning Kamp      | UNIX since Zilog Zeus 3.20
phk@FreeBSD.ORG        | TCP/IP since RFC 956
FreeBSD committer      | BSD since 4.3-tahoe
Never attribute to malice what can adequately be explained by incompetence.

Analog Devices and Linear Technology have application notes on this
subject.  At least with sampling converters, jitter directly limits
dynamic range.

My back of the envelope calculation comes up with about 25ps of RMS
jitter for an ideal 16 bit sampling converter at audio frequencies but
most delta-sigma converters should tolerate higher levels.  Analog
Devices says 100s of ps of clock jitter is acceptable for them.

How low can the dynamic range be before it becomes audible?

On Mon, 7 May 2012 09:48:39 -0700 (PDT), "J. Forster"
jfor@quikus.com wrote:

In message 4FA80913.7000703@medesign.ro, MailLists writes:

Not so fast there...

Yes, in theory your ADC could digitize a signal 14*6 = 84 dB below
reference level, but it would do so with 50% distortion, because
there would only be three distinct levels: {-1, 0, +1}

This is a much overlooked issue, in particular with classical music
where dynamics in the music can account for way more dB than people
realize.

We must start out by defining the acceptable level of total distortion,
if we choose 0.5% then we need 200 digital levels, roughly 8 of
your 16 bits for the signal.

That gives you a headroom of 7 bits (leaving one for the sign) and
that gives you 42 dB of S/N.

That isn't very much, headroom, 42dB, when the conductor waves the
entire philharmonic AND the full opera choir in, for for that wonderful
"Dies Ira" of Verdis.  Or Carmina Burana.  Or any of the many
other 'shock-effects' classical composers have enjoyed.

With digital, you get most distortion at weak signals, where your
ears are much better at detecting it, with vinyl you get more
distortion on strong signals, just like your ears, meaning the
level becomes unbearable sooner.

That is why, in plain and simple terms, classical struggles with
digital:  High distortion in weak passages.

It is also why the CD media has changed rythmic music, which
went from a love of distortion to a love of pure tones when
the CD media made it possible to play loud pure tones.

--
Poul-Henning Kamp      | UNIX since Zilog Zeus 3.20
phk@FreeBSD.ORG        | TCP/IP since RFC 956
FreeBSD committer      | BSD since 4.3-tahoe
Never attribute to malice what can adequately be explained by incompetence.

On Mon, 2012-05-07 at 18:15 +0000, Poul-Henning Kamp wrote:

Not true in a correctly dithered quantizer  (And they almost all are
these days)...

This is counter intuitive, but adding 1 LSB of uncorrelated noise having
the correct statistical properties (Triangular probability distribution)
has the effect of completely linearising the conversion process at the
cost of adding about 3dB of noise to the system.
With the noise added you can hear narrow tones well below the wideband
noise floor.

In a correctly dithered system the broadband noise floor is the only
thing determined by the word length, and narrow band signals can be
resolved to well below the noise floor.

Further, as the statistical properties of the noise are not all that
tightly coupled to its frequency domain properties, it is possible to
filter the noise to move most of the energy away from the regions where
the ear is most sensitive.

16 bits is actually fine as a distribution format, where is shows up as
a little short is as a capture format as at capture time you need
headroom to ensure nothing unexpected causes clipping, but once you are
done with the processing it is trivial to strip the headroom out and
dither down to 16 bits.

This discussion would be better over on the Pro audio list rather then
time nuts.

73, Dan.

In message 1336415866.16321.14.camel@laptop, Dan Mills writes:

Yes, I agree with that, and lets use that agreement to stop the
topic :-)

--
Poul-Henning Kamp      | UNIX since Zilog Zeus 3.20
phk@FreeBSD.ORG        | TCP/IP since RFC 956
FreeBSD committer      | BSD since 4.3-tahoe
Never attribute to malice what can adequately be explained by incompetence.

On Mon, May 7, 2012 at 11:15 AM, Poul-Henning Kamp phk@phk.freebsd.dk wrote:

You are mixing recording and distribution.  The 16-bit 44.1K "CD
Quality" is for distribution to consumers.  Few people record with
that format.  24-bits and 96K is a common recording format.  and then
later it is mastered to "fit" within the CD format.    And don't
forget that some tools the mastering engineer has are EQ, "dithering"
and frequency dependent compression.  It is VERY rare that a
performance would be linearly transliterated to the CD.  What you get
is something that was modified to sound good on consumer playback
equipment.  With "good" being the engineer's person opinion.

Back to recording.  It is common to have the master studio clock be
an OCXO.  This would drive the (96K) sample clock and it is the sample
clock that gets distributed inside the rack.  My cheep home system, I
think has a low cost XO inside and is pretty much jitter free.  Simple
XOs can be pretty good especially what you care more about "clean"
than accurate

Chris Albertson
Redondo Beach, California

On 05/07/2012 08:15 PM, Poul-Henning Kamp wrote:

The late Julian Dunn has covered this in AES papers and pre-prints.

It relates to side-band, modulation frequency and masking-effects. He
came up with a sinusoidal modulation mask.

Look up his work!

Cheers,
Magnus

In message CABbxVHuASDQ-mwug6fMwc4Ln-D3ZKhegvpVvbPCprWXeWGfX0A@mail.gmail.com
, Chris Albertson writes:

I'm old enough to have listend to comparisons when 16 bit 44.1KHz
was both recoding and distribution format :-)

As I said: one of the main drivers for oversampling is to relax
requirements for analog and clock precision.

Or in the case of an entire generation worth of european classical
recordings:  "good" being equal to "Karajan can hear it through
his increasingly severe deafness" :-)

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phk@FreeBSD.ORG        | TCP/IP since RFC 956
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Never attribute to malice what can adequately be explained by incompetence.

In message CABbxVHtad_Ewe_PTrMkifkRYWzjFHw3XmYpFeV6B5su3xw+8Pw@mail.gmail.com
, Chris Albertson writes:

Disadvantage: on playback you get both a sample and a standard deviation :-)

I don't think anybody uses random sampling unless they have to
(think "when can we actually see this star with this telescope" etc)

--
Poul-Henning Kamp      | UNIX since Zilog Zeus 3.20
phk@FreeBSD.ORG        | TCP/IP since RFC 956
FreeBSD committer      | BSD since 4.3-tahoe
Never attribute to malice what can adequately be explained by incompetence.

Wow! $1260 for a 4' power cord, but wait, there's more... It was named
'Power Cord of the Year'.

Mike

On 5/7/2012 9:39 AM, Burt I. Weiner wrote:

On Mon, May 07, 2012 at 08:28:41AM -0700, J. Forster wrote:

I completely agree, and far more significant than accuracy

is jitter (phase noise) in maybe the tenths of a Hz to thousands  of Hz
area.  This does modulate the sampled sound and perhaps is perceptable
at very low levels.

BUT Cesium, or Rb buys nothing in respect to phase noise

in those ranges... really good quality quartz oscillators have much better
close in phase noise than many Rb's or Cesiums...

What Cesium and Rb buy is good performance measured over much

larger taus... which cannot possibly have any impact on human hearing.

--
Dave Emery N1PRE/AE, die@dieconsulting.com  DIE Consulting, Weston, Mass 02493
"An empty zombie mind with a forlorn barely readable weatherbeaten
'For Rent' sign still vainly flapping outside on the weed encrusted pole - in
celebration of what could have been, but wasn't and is not to be now either."

Hadley
K7MLR

A fine is a tax for doing wrong.  A tax is a fine for doing well.

Peter Cooper, of Fermi Lab, says, "Every experimentalist knows
that the apparatus, or at least your understanding of it, is
always at fault until demonstrated otherwise." He also says,
"Nature is really unmoved by what I, or anyone else, believes."

David,

I haven't been following this thread so I suppose it has already been
answered, but how are you measuring "zero beat?"

Lee Mushel
----- Original Message -----
From: "David I. Emery" die@dieconsulting.com
To: jfor@quikus.com; "Discussion of precise time and frequency
measurement" time-nuts@febo.com
Sent: Monday, May 07, 2012 5:37 PM
Subject: Re: [time-nuts] Oh dear

On Mon, 07 May 2012 20:50:41 +0200
Magnus Danielson magnus@rubidium.dyndns.org wrote:

Could you tell a little bit more? Which Julian Dunn is it and
what does AES stand for? And do you have any links/papers at
hand that i could have a look at? :-)

		Attila Kinali

--
Why does it take years to find the answers to
the questions one should have asked long ago?

AES = Audio Engineering Society

Google "Julian Dunn" audio

-John

=================

On Wed, 9 May 2012 09:43:25 -0700 (PDT)
"J. Forster" jfor@quikus.com wrote:

Thanks a lot... now i have more to read for those rainy evening ;-)

		Attila Kinali

--
Why does it take years to find the answers to
the questions one should have asked long ago?