Re: [time-nuts] Clocks for Audio gear

2012-05-13 Thread Sanjeev Gupta
On Fri, May 11, 2012 at 1:27 AM, Chris Albertson
albertson.ch...@gmail.comwrote:

 I've alway have thought that if nanosecond level jitter is bad then
 breathing while listening must be really bad.  If you inhale the path
 length from your ear to the speaker changes at the microsecond level.
  You'd think the resulting doppler shift would drive these audiophiles
 nuts.  All that pitch shifting.


+1


 So my new product for this is the head vice  It will have three
 wooden plates with lead screw clamps in a cast iron frame.   The iron
 frame is to be bolted to a wall of other massive object.   This will
 greatly reduce those annoying microsecond level audio path length
 variations that are caused by breathing, blood flow and eye blinks.
 For those sensitive to picosecond level jitter, the wood blocks can be
 removed and the steel clamp plate applied directly (wood after all is
 elastic and compressible)


!

+1.02104

Permission to resend to all my audiophile friends?

-- 
Sanjeev Gupta
+65 98551208 http://www.linkedin.com/in/ghane
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Re: [time-nuts] Clocks for Audio gear

2012-05-11 Thread MailLists
Ashihara's tests were with music/voice, taking into account 
psychoacoustics, for an average group of music savvy listeners, and even 
music professionals.
As uncorrelated jitter is practically raising the noise floor, most of 
it was masked by the signal, making it more difficult to detect. 
Benjamin and Gannon used sinusoidal jitter, which isn't appearing 
normally in signal chains (badly designed ones excepted).
In a real case, with higher probability (added) jitter would be 
correlated with the digital content transmitted over a path - S/PDIF, 
and AES/EBU are prone to jitter induced by the signal path 
characteristics, ISI - PSUs, and even external noise sources.

A more realistic simulation would take those into account.
OTOH there where tests on pure sine tones, with sine jitter, detectable 
by trained ears at even lower levels of jitter, which might indicate the 
lowest threshold of hearing, but using artificial conditions.

Who would listen to pure sine tones?

On 5/10/2012 8:25 PM, Heinzmann, Stefan (ALC NetworX GmbH) wrote:

Chris Albertson wrote:

If we are to believe the above paper,then those guys who claim to hear
pS jitter are wrong.


Note that the jitter spectrum matters for its audibility. Ashihara et.al. used 
random jitter, and it is not very suprising that the sensitivity for random 
jitter is lower than for jitter that has specially been shaped to improve 
detectability by human ears. Thus the results by Ashihara are credible, but 
they are not the lower limit on jitter audibility.

Benjamin and Gannon, the first reference in Ashihara's paper, come to lower 
figures for sinusoidal jitter with carefully selected frequencies relative to 
the main signal, which is also sinusoidal. Their results reach down to the 
single figure nanosecond range, and that can be regarded as the real limit of 
audibility.

Of course, that still leaves those who claim to hear jitter in the picoseconds 
range out in fairy-tale land. And jitter of just a few nanoseconds is still 
quite easy to achieve with crystal oscillators. No need for special and 
expensive parts, then. Normal developer diligence is enough.

Cheers
Stefan


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Re: [time-nuts] Clocks for Audio gear

2012-05-11 Thread Heinzmann, Stefan (ALC NetworX GmbH)
MailLists wrote:
 Ashihara's tests were with music/voice, taking into account
 psychoacoustics, for an average group of music savvy listeners, and
 even music professionals.
 As uncorrelated jitter is practically raising the noise floor, most of
 it was masked by the signal, making it more difficult to detect.
 Benjamin and Gannon used sinusoidal jitter, which isn't appearing
 normally in signal chains (badly designed ones excepted).
 In a real case, with higher probability (added) jitter would be
 correlated with the digital content transmitted over a path - S/PDIF,
 and AES/EBU are prone to jitter induced by the signal path
 characteristics, ISI - PSUs, and even external noise sources.
 A more realistic simulation would take those into account.
 OTOH there where tests on pure sine tones, with sine jitter, detectable
 by trained ears at even lower levels of jitter, which might indicate
 the lowest threshold of hearing, but using artificial conditions.
 Who would listen to pure sine tones?

Ashihara et.al. wanted to find out what level of jitter was likely to be 
audible under real-world conditions. Those conditions would likely include 
music as the main signal, and random jitter.

Benjamin/Gannon wanted to find out what levels of jitter could be detected if 
the conditions were as favorable as possible for detecting jitter. That is not 
the real-world situation, of course, but it can establish a baseline where you 
may legitimately say that if you stay below this line with jitter of whatever 
type, the effects are very unlikely to be audible. And, to add a comment 
towards Attila, one of the results by Benjamin/Gannon was that training matters 
a lot, and the best sensitivity was by trained listeners. Your comment is 
therefore warranted, but already accounted for.

Hence, even though their results appear to be very different, they are both 
valid, because it depends on the exact question asked. I would dare to say, 
that no matter how you set up your realistic simulation, the results are 
likely to be somewhere between the values by Benjamin/Gannon and by Ashihara 
et.al.

So, for the purpose of this group, I'd say the psychoacoustic stuff would lead 
too far, but it might be helpful to know at which jitter levels one can assume 
to be on the safe side in an audio system, regarding audibility of jitter 
effects. Judging from the mentioned studies, I concluded (for myself at least), 
that this boundary is somewhere in the single figure nanoseconds, until someone 
comes forth with hard evidence that it needs to be set lower.

Cheers
Stefan


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Re: [time-nuts] Clocks for Audio gear

2012-05-11 Thread shalimr9
The brain uses phase at low frequency and amplitude at higher frequencies to 
find the direction a signal is coming from. It works better for low frequencies 
than high when you have a steady tone, but high frequency positioning is better 
when the signal is pulsed.
It is almost impossible to locate the source of a continuous high pitch tone in 
a confined space because of standing waves.
My boat has a piezo buzzer to indicate low oil level. It was completely 
impossible for me to tell if it was located in the engine compartment or under 
the dash, or anywhere else, even though the cockpit is wide open. It turns out 
it was in the engine controller, to the right of the driving position. Someone 
had to tell me.

Didier KO4BB

 
Sent from my BlackBerry Wireless thingy while I do other things...

-Original Message-
From: Dan Kemppainen d...@irtelemetrics.com
Sender: time-nuts-boun...@febo.com
Date: Thu, 10 May 2012 16:16:44 
To: time-nuts@febo.com
Reply-To: Discussion of precise time and frequency measurement
time-nuts@febo.com
Subject: Re: [time-nuts] Clocks for Audio gear

Don't forget the human mind can compensate for a lot of things. Think 
of how we can triangulate a sound source in realtime even with the 
included echos in a small room. The only thing that I can think of 
that messes with that system is a single tone setting up standing 
waves. It's impressive if you think about it.

So, it's probably not much of a stretch to imagine the mind 
compensating for a little movement here and there (since we have 
controls and feedback to monitor that). It may just take a few 
thousand years for us to evolve to deal with distortion due to jitter 
in our digital recordings :)

All fun aside. This has been a worth while thread in my opinion. I'm 
learning more this week, than others watching this list!

On 5/10/2012 1:49 PM, time-nuts-requ...@febo.com wrote:
 I've alway have thought that if nanosecond level jitter is bad then
 breathing while listening must be really bad.  If you inhale the path
 length from your ear to the speaker changes at the microsecond level.
   You'd think the resulting doppler shift would drive these audiophiles
 nuts.  All that pitch shifting.

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Re: [time-nuts] Clocks for Audio gear

2012-05-11 Thread shalimr9
Breathing and heart beat are filtered out and corrected for by the brain,
otherwise we would have difficulties to hear a lot of things.

Interestingly, that works well in our natural environment, but not as well when 
you are somewhere else.

When free diving (when there is no noisy scuba gear and breathing), you can 
hear your own heartbeat and so can the fish, sometimes at significant distances 
as it propagates well under water.

Didier KO4BB

Sent from my BlackBerry Wireless thingy while I do other things...

-Original Message-
From: Attila Kinali att...@kinali.ch
Sender: time-nuts-boun...@febo.com
Date: Thu, 10 May 2012 23:25:50 
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] Clocks for Audio gear

On Thu, 10 May 2012 11:36:40 -0700
Hal Murray hmur...@megapathdsl.net wrote:

 Heartbeats may be more interesting than breathing.  Does anybody know of 
 spectrum domain data?  It should be possible to collect position info while 
 also monitoring heartbeat and chest diameter and then crunch some numbers do 
 see how much of the position correlates with heartbeat vs breathing and then 
 plot each part in the frequency domain.

Please do not forget that there is a quite sofisticated error correction
system attached to the hear, which we usually refere to as the brain.

Breathing and heart beat are filtered out and corrected for by the brain,
otherwise we would have difficulties to hear a lot of things.
For more infos, please have a look at perceptual psychology. A not too
bad introduction to that field is Sensation and Perception 
by E. B. Goldstein.

But please do not expect mathematical rigor in that field. It's still
a subfield of psychology.

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

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Re: [time-nuts] Clocks for Audio gear

2012-05-11 Thread Henk
Hi,

That jitter value, was that one period jitter? Or was it jitter over a
large number of periods, thus close by the carrier?

Henk

 MailLists wrote:
 Ashihara's tests were with music/voice, taking into account
 psychoacoustics, for an average group of music savvy listeners, and
 even music professionals.
 As uncorrelated jitter is practically raising the noise floor, most of
 it was masked by the signal, making it more difficult to detect.
 Benjamin and Gannon used sinusoidal jitter, which isn't appearing
 normally in signal chains (badly designed ones excepted).
 In a real case, with higher probability (added) jitter would be
 correlated with the digital content transmitted over a path - S/PDIF,
 and AES/EBU are prone to jitter induced by the signal path
 characteristics, ISI - PSUs, and even external noise sources.
 A more realistic simulation would take those into account.
 OTOH there where tests on pure sine tones, with sine jitter, detectable
 by trained ears at even lower levels of jitter, which might indicate
 the lowest threshold of hearing, but using artificial conditions.
 Who would listen to pure sine tones?

 Ashihara et.al. wanted to find out what level of jitter was likely to be
 audible under real-world conditions. Those conditions would likely include
 music as the main signal, and random jitter.

 Benjamin/Gannon wanted to find out what levels of jitter could be detected
 if the conditions were as favorable as possible for detecting jitter. That
 is not the real-world situation, of course, but it can establish a
 baseline where you may legitimately say that if you stay below this line
 with jitter of whatever type, the effects are very unlikely to be audible.
 And, to add a comment towards Attila, one of the results by
 Benjamin/Gannon was that training matters a lot, and the best sensitivity
 was by trained listeners. Your comment is therefore warranted, but already
 accounted for.

 Hence, even though their results appear to be very different, they are
 both valid, because it depends on the exact question asked. I would dare
 to say, that no matter how you set up your realistic simulation, the
 results are likely to be somewhere between the values by Benjamin/Gannon
 and by Ashihara et.al.

 So, for the purpose of this group, I'd say the psychoacoustic stuff would
 lead too far, but it might be helpful to know at which jitter levels one
 can assume to be on the safe side in an audio system, regarding audibility
 of jitter effects. Judging from the mentioned studies, I concluded (for
 myself at least), that this boundary is somewhere in the single figure
 nanoseconds, until someone comes forth with hard evidence that it needs to
 be set lower.

 Cheers
 Stefan


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Re: [time-nuts] Clocks for Audio gear

2012-05-11 Thread Jim Lux

On 5/11/12 12:48 AM, MailLists wrote:


Who would listen to pure sine tones?



As a youth, I listened to WWV, which is a pure sine tone, in between the 
ticks.  Drove my parents batty.




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Re: [time-nuts] Clocks for Audio gear

2012-05-11 Thread Stanley


- Original Message - 
From: Jim Lux jim...@earthlink.net

To: time-nuts@febo.com
Sent: Friday, May 11, 2012 8:10 AM
Subject: Re: [time-nuts] Clocks for Audio gear



On 5/11/12 12:48 AM, MailLists wrote:


Who would listen to pure sine tones?



As a youth, I listened to WWV, which is a pure sine tone, in between the 
ticks.  Drove my parents batty.




me too, I think replaying this in my head is a better way to measure time 
than the 1 thousand, 2 ... trick, having low music skills would wonder what 
the limit of human time keeping is ? Director , Drummer ... how would we 
collect data to produce a Allan variance graph ? Does a timing savant exist 
?


http://discovermagazine.com/2002/feb/featsavant

Stanley 



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Re: [time-nuts] Clocks for Audio gear

2012-05-11 Thread DaveH
Two things

1) - people in anechoic chambers will really notice the sound of their
heartbeats as well as the s-sh-s-sh sound of the blood flowing
through their heads. 

http://en.wikipedia.org/wiki/Anechoic_chamber

They are designed to have complete sound absorption and are dead quiet.
Very spooky to be in -- used to live in Boston and Harvard University had
one that I was able to visit for an hour.

2) - an interesting experiment in brain filtering is to stand near a
broadband noise source (fan or air conditioner or radio with someone
talking) and talk with someone. Have a recorder going and record your
conversation. You can understand the other person perfectly while face to
face but listening to the recorded conversation, it will be hard to hear
them over all that noise.

Dave

 -Original Message-
 From: time-nuts-boun...@febo.com 
 [mailto:time-nuts-boun...@febo.com] On Behalf Of shali...@gmail.com
 Sent: Friday, May 11, 2012 04:27
 To: Time-Nuts
 Subject: Re: [time-nuts] Clocks for Audio gear
 
 Breathing and heart beat are filtered out and corrected for 
 by the brain,
 otherwise we would have difficulties to hear a lot of things.
 
 Interestingly, that works well in our natural environment, 
 but not as well when you are somewhere else.
 
 When free diving (when there is no noisy scuba gear and 
 breathing), you can hear your own heartbeat and so can the 
 fish, sometimes at significant distances as it propagates 
 well under water.
 
 Didier KO4BB
 
 Sent from my BlackBerry Wireless thingy while I do other things...
 
 -Original Message-
 From: Attila Kinali att...@kinali.ch
 Sender: time-nuts-boun...@febo.com
 Date: Thu, 10 May 2012 23:25:50 
 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] Clocks for Audio gear
 
 On Thu, 10 May 2012 11:36:40 -0700
 Hal Murray hmur...@megapathdsl.net wrote:
 
  Heartbeats may be more interesting than breathing.  Does 
 anybody know of 
  spectrum domain data?  It should be possible to collect 
 position info while 
  also monitoring heartbeat and chest diameter and then 
 crunch some numbers do 
  see how much of the position correlates with heartbeat vs 
 breathing and then 
  plot each part in the frequency domain.
 
 Please do not forget that there is a quite sofisticated error 
 correction
 system attached to the hear, which we usually refere to as 
 the brain.
 
 Breathing and heart beat are filtered out and corrected for 
 by the brain,
 otherwise we would have difficulties to hear a lot of things.
 For more infos, please have a look at perceptual psychology. A not too
 bad introduction to that field is Sensation and Perception 
 by E. B. Goldstein.
 
 But please do not expect mathematical rigor in that field. It's still
 a subfield of psychology.
 
   Attila Kinali
 -- 
 Why does it take years to find the answers to
 the questions one should have asked long ago?
 
 ___
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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Attila Kinali
On Wed, 09 May 2012 14:25:34 -0700
Hal Murray hmur...@megapathdsl.net wrote:

 Back to technical stuff...
 
 As a practical matter, is clock jitter or phase noise from a typical low cost 
 crystal and decent board layout a significant problem in audio gear?  How 
 hard is it to measure?

Depends on how accurate it should be ;-)

But for audio purposes, the crystal itself has low enough noise. The
Problem is the oscillator circuit and the power supply. That's where
most designs mess up.

 
 Video geeks solved their clock distribution problems by using frame buffers.  
 Is there a similar trick for audio?  Is there a need for it?

Video is a lot less sensitive to jitter. An A/V desync starts to be
noticable from 10ms upwards for most people. Trained people notice
it even below 5ms. I guess that 1ms is beyond what anyone can notice.
But imagine you've a 1ms gap in your audio... People will scream at
the poor quality.

 Part of the problem is that they are doing magic down conversion in the ADC.  
 (I can't think of the term.)

Direct downconversion using sampling :-)

  Suppose you have a 100 MHz signal with a 1 MHz 
 bandwidth.  You don't have to sample at 200 MHz.  You can sample at 2 MHz and 
 your signal will alias down.  It's turning what is normally a bug into a 
 feature.  

It's not a bug. It's a feature of the sampling process itself.
Think of sampling as multiplication of an comp of dirac pulses
with your input signal. Now remember that mixing is a multiplication
of two signalsAnd the idea doing that is actually quite old,
but it wasn't feasible with main stream components until a few years back.


Attila Kinali

-- 
The trouble with you, Shev, is you don't say anything until you've saved
up a whole truckload of damned heavy brick arguments and then you dump
them all out and never look at the bleeding body mangled beneath the heap
-- Tirin, The Dispossessed, U. Le Guin

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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread MailLists
Hearing tests showed the ability to discern jitter above a few hundred 
nanoseconds rms.

http://amorgignitamorem.nl/Audio/Jitter/Detection%20threshold%20for%20distortions%20due%20to%20jitter%20on%20digital%20audio%2026_50.pdf

Others claim the ability to detect jitter in the picoseconds range...

It would be a conservative assumption that jitter in the range of 
tens-hundreds of picoseconds will be practically not discernible.


Usually integrated oscillators are composed of the classical inverting 
gate oscillator, with external CQC, and selfbiasing R, which has 
practically no rejection (~6dB) of the power supply noise. As it's 
usually on the same die with noisy digital circuitry, the gate threshold 
will jump around, producing timing errors, also the slew-rate is quite 
low, which just worsens the situation.


As most digital circuitry is less affected by jitter, the best solution 
is to place a clean oscillator near the D/A conversion, where the most 
critical timing point is, and through buffers clock the rest of the 
digital circuits - eventually galvanic isolation might be implemented, 
to pollute less the analog part with digital noise.
To minimize jitter, digital clock inputs should be driven by fast 
slew-rate circuitry.



On 5/10/2012 12:25 AM, Hal Murray wrote:

was Subject: Re: [time-nuts] Faster than light of a different type
(Probably my fault.)

act...@hotmail.com said:

What I found funny was that the Audiophlie and light thread drew such
attacks when it hit home to me as exactly what the Time-Nuts mission is
about.  The Audio thread touched on some real world time and freq research
...


I too enjoyed the technical discussions.  Thanks for your contributions.

It's the audiophool bashing that people are complaining about.  Sure, it's
fun, but only at the right time and it gets old quickly.  The problem is that
with large groups, there are different opinions of when and how much is
appropriate.  The long tail on opinions of reasonable can annoy a lot of
people.

---

Back to technical stuff...

As a practical matter, is clock jitter or phase noise from a typical low cost
crystal and decent board layout a significant problem in audio gear?  How
hard is it to measure?

Is clock accuracy a practical problem?  How good are people with perfect
pitch?  It wouldn't surprise me if there are a few that are much much better
than others, but how good is that relative to 50 PPM which I can get in a low
cost crystal?

Video geeks solved their clock distribution problems by using frame buffers.
Is there a similar trick for audio?  Is there a need for it?



I know clocking is a serious problem in fancy DSP systems.  For example,
modern radar has gone digital.  In that context, clock jitter can be
important.  Standard procedure is don't run your clock through a FPGA because
it will add jitter.

Part of the problem is that they are doing magic down conversion in the ADC.
(I can't think of the term.)  Suppose you have a 100 MHz signal with a 1 MHz
bandwidth.  You don't have to sample at 200 MHz.  You can sample at 2 MHz and
your signal will alias down.  It's turning what is normally a bug into a
feature.  The catch is that the errors/noise due to clock jitter happens at
the high frequency, in this case multiplying the noise by 100.  (Your
sample/hold at the front end has to work at the high frequency and your
anti-aliasing filter gets more interesting.)

There has been an interesting change in the specs for ADCs and DACs over the
past 20(?) years.  They used to be specified using terms like DNL and INL and
No-missing-codes.  Modern high-speed ADCs are specified with terms like ENOB
and SFDR.  Data sheets often include several plots of a batch of samples run
through a DFT so you can see the noise floor and such.

Here is a reasonable glossary:
   http://pdfserv.maxim-ic.com/en/an/AN641.pdf

I don't remember comments/specs about clock jitter in the data sheets but I
haven't looked at one in a few years.  I'll have to keep an eye out the next
time I'm browsing.





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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Poul-Henning Kamp
In message 4fab74eb.1050...@medesign.ro, MailLists writes:

Others claim the ability to detect jitter in the picoseconds range...

It would be a conservative assumption that jitter in the range of 
tens-hundreds of picoseconds will be practically not discernible.

We're probably talking about one of those tripple-blind experiments ?

(http://www.cco.caltech.edu/~boyk/sdttest.htm)


-- 
Poul-Henning Kamp   | UNIX since Zilog Zeus 3.20
p...@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.

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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Heinzmann, Stefan (ALC NetworX GmbH)
Hal Murray wrote:

 As a practical matter, is clock jitter or phase noise from a typical
 low cost crystal and decent board layout a significant problem in audio
 gear?  How hard is it to measure?

The answer depends a lot on the circumstances (as usual).

If you refer to jitter effects on a conversion between analog and digital 
(either way), and you're interested in whether these can be audible, you will 
want to measure the effect on the audio signal, and not necessarily the jitter 
itself. For example, you might want to use a high frequency sine wave close to 
the upper bandwidth limit as the audio signal, and measure the jitter-related 
distortions using an FFT analyzer. It may be hard to distinguish jitter-related 
artifacts from other distortions, but as you are interested in the overall 
signal fidelity, that's probably what you want anyway. The good thing about 
that is that you need no extra gear that you don't already have as an audio 
developer. No expensive phase noise analyzer, for example.

If you refer to jitter effects on a digital transmission, you will be 
interested in what they do to the bit error probability, or whether you are 
still conformant to some standard that puts a limit to the allowable jitter. In 
that case you are more likely to find yourself looking at eye diagrams on an 
oscilloscope, or perhaps using a bit error analyzer.

There is little use for a proper phase noise analyzer in audio, and RD labs of 
most manufacturers I know don't have one.

 Is clock accuracy a practical problem?  How good are people with
 perfect pitch?  It wouldn't surprise me if there are a few that are
 much much better than others, but how good is that relative to 50 PPM
 which I can get in a low cost crystal?

Clock accuracy can be a problem, but not because of pitch perception. Crystal 
oscillators are easily accurate enough for human perception, even the crappy 
ones.

Clock accuracy does matter in system applications, when pieces of gear need to 
lock to each other, or to a house clock. Here, the clock accuracy needs to 
match the lock range of downstream PLLs, or else you can't lock reliably. The 
AES11 standard has rules for that.

 Video geeks solved their clock distribution problems by using frame
 buffers.
 Is there a similar trick for audio?  Is there a need for it?

A sample buffer of at least one sample is contained in pretty much every 
S/P-DIF or AES/EBU receiver chip, so the answer to your question would be yes. 
Storing a sample is cheap, however, compared to storing a video frame.

Cheers
Stefan


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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Chris Albertson
On Thu, May 10, 2012 at 12:57 AM, MailLists li...@medesign.ro wrote:
 Hearing tests showed the ability to discern jitter above a few hundred
 nanoseconds rms.
 http://amorgignitamorem.nl/Audio/Jitter/Detection%20threshold%20for%20distortions%20due%20to%20jitter%20on%20digital%20audio%2026_50.pdf

 Others claim the ability to detect jitter in the picoseconds range...

If we are to believe the above paper,then those guys who claim to hear
pS jitter are wrong.  Likely they can also here is a fuse is is place
in the holder backwards.

So by the above, no now can hear 250 nS of jitter.I really doubt
any decent system other then the most low-cost consumer level junk has
jitter at the 250 nS level.   Even a TTL can oscillator is better
than that.

A TTL can that is marked 4.096 MHz costs about $2 and will make a
square wave with a period of very close to 250 nS.   Then they divide
this down to the sample rate of 96KHz.   In order to see a 250 nS
jitter in the 96K signal the TTL can would have to skip a beat.
250 nS is is a huge error and you don't get there with digital noise

Chris Albertson
Redondo Beach, California

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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread J. Forster
In fact, I do believe the paper is a voice of rationality in an ocean oh
hype. Very expensive hype, promoted by shameless hucksters.

-John

===



 On Thu, May 10, 2012 at 12:57 AM, MailLists li...@medesign.ro wrote:
 Hearing tests showed the ability to discern jitter above a few hundred
 nanoseconds rms.
 http://amorgignitamorem.nl/Audio/Jitter/Detection%20threshold%20for%20distortions%20due%20to%20jitter%20on%20digital%20audio%2026_50.pdf

 Others claim the ability to detect jitter in the picoseconds range...

 If we are to believe the above paper,then those guys who claim to hear
 pS jitter are wrong.  Likely they can also here is a fuse is is place
 in the holder backwards.

 So by the above, no now can hear 250 nS of jitter.I really doubt
 any decent system other then the most low-cost consumer level junk has
 jitter at the 250 nS level.   Even a TTL can oscillator is better
 than that.

 A TTL can that is marked 4.096 MHz costs about $2 and will make a
 square wave with a period of very close to 250 nS.   Then they divide
 this down to the sample rate of 96KHz.   In order to see a 250 nS
 jitter in the 96K signal the TTL can would have to skip a beat.
 250 nS is is a huge error and you don't get there with digital noise

 Chris Albertson
 Redondo Beach, California

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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Heinzmann, Stefan (ALC NetworX GmbH)
Chris Albertson wrote:
 If we are to believe the above paper,then those guys who claim to hear
 pS jitter are wrong.

Note that the jitter spectrum matters for its audibility. Ashihara et.al. used 
random jitter, and it is not very suprising that the sensitivity for random 
jitter is lower than for jitter that has specially been shaped to improve 
detectability by human ears. Thus the results by Ashihara are credible, but 
they are not the lower limit on jitter audibility.

Benjamin and Gannon, the first reference in Ashihara's paper, come to lower 
figures for sinusoidal jitter with carefully selected frequencies relative to 
the main signal, which is also sinusoidal. Their results reach down to the 
single figure nanosecond range, and that can be regarded as the real limit of 
audibility.

Of course, that still leaves those who claim to hear jitter in the picoseconds 
range out in fairy-tale land. And jitter of just a few nanoseconds is still 
quite easy to achieve with crystal oscillators. No need for special and 
expensive parts, then. Normal developer diligence is enough.

Cheers
Stefan


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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Chris Albertson
I've alway have thought that if nanosecond level jitter is bad then
breathing while listening must be really bad.  If you inhale the path
length from your ear to the speaker changes at the microsecond level.
 You'd think the resulting doppler shift would drive these audiophiles
nuts.  All that pitch shifting.

So my new product for this is the head vice  It will have three
wooden plates with lead screw clamps in a cast iron frame.   The iron
frame is to be bolted to a wall of other massive object.   This will
greatly reduce those annoying microsecond level audio path length
variations that are caused by breathing, blood flow and eye blinks.
For those sensitive to picosecond level jitter, the wood blocks can be
removed and the steel clamp plate applied directly (wood after all is
elastic and compressible)



On Thu, May 10, 2012 at 10:03 AM, J. Forster j...@quikus.com wrote:
 In fact, I do believe the paper is a voice of rationality in an ocean oh
 hype. Very expensive hype, promoted by shameless hucksters.

 -John

 ===



 On Thu, May 10, 2012 at 12:57 AM, MailLists li...@medesign.ro wrote:
 Hearing tests showed the ability to discern jitter above a few hundred
 nanoseconds rms.
 http://amorgignitamorem.nl/Audio/Jitter/Detection%20threshold%20for%20distortions%20due%20to%20jitter%20on%20digital%20audio%2026_50.pdf

 Others claim the ability to detect jitter in the picoseconds range...

 If we are to believe the above paper,then those guys who claim to hear
 pS jitter are wrong.  Likely they can also here is a fuse is is place
 in the holder backwards.

 So by the above, no now can hear 250 nS of jitter.    I really doubt
 any decent system other then the most low-cost consumer level junk has
 jitter at the 250 nS level.   Even a TTL can oscillator is better
 than that.

 A TTL can that is marked 4.096 MHz costs about $2 and will make a
 square wave with a period of very close to 250 nS.   Then they divide
 this down to the sample rate of 96KHz.   In order to see a 250 nS
 jitter in the 96K signal the TTL can would have to skip a beat.
 250 nS is is a huge error and you don't get there with digital noise

 Chris Albertson
 Redondo Beach, California

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-- 

Chris Albertson
Redondo Beach, California

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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Tom Knox

Great dialog. The Time Nuts form can be very humbling and often has me 
questioning my own knowledge base. This thread is no exception. My take on the 
effect of jitter is when an d toA converter is reproducing a pure sine wave 
even a single point of slight jitter will show up in the FFT as false analog 
content. In a complex wave it can easily change the relationship between a 
fundamental and harmonics. This is a major problem in speaker crossovers design 
where drivers with different masses can only be aligned at one freq. It has 
been my experience this relationship between fundmentals and harmonics is 
noticeably audible at levels that are difficult to measure. To me, it is this 
difficulty to mathematically define certain aspects of audible distortion that 
makes audio design fascinating. (It also makes some voodoo science and snake 
oil products difficult to disprove, especially to those lacking an electronics 
or physics background)  Will timing improvements affect sound quality, because 
of the complexity of human hearing it seems to me still worthy of 
investigation. Time will tell.
Best Wishes;
Thomas Knox



 From: albertson.ch...@gmail.com
 Date: Thu, 10 May 2012 09:48:21 -0700
 To: time-nuts@febo.com
 Subject: Re: [time-nuts] Clocks for Audio gear
 
 On Thu, May 10, 2012 at 12:57 AM, MailLists li...@medesign.ro wrote:
  Hearing tests showed the ability to discern jitter above a few hundred
  nanoseconds rms.
  http://amorgignitamorem.nl/Audio/Jitter/Detection%20threshold%20for%20distortions%20due%20to%20jitter%20on%20digital%20audio%2026_50.pdf
 
  Others claim the ability to detect jitter in the picoseconds range...
 
 If we are to believe the above paper,then those guys who claim to hear
 pS jitter are wrong.  Likely they can also here is a fuse is is place
 in the holder backwards.
 
 So by the above, no now can hear 250 nS of jitter.I really doubt
 any decent system other then the most low-cost consumer level junk has
 jitter at the 250 nS level.   Even a TTL can oscillator is better
 than that.
 
 A TTL can that is marked 4.096 MHz costs about $2 and will make a
 square wave with a period of very close to 250 nS.   Then they divide
 this down to the sample rate of 96KHz.   In order to see a 250 nS
 jitter in the 96K signal the TTL can would have to skip a beat.
 250 nS is is a huge error and you don't get there with digital noise
 
 Chris Albertson
 Redondo Beach, California
 
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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Hal Murray

albertson.ch...@gmail.com said:
 I've alway have thought that if nanosecond level jitter is bad then
 breathing while listening must be really bad.  If you inhale the path length
 from your ear to the speaker changes at the microsecond level.
  You'd think the resulting doppler shift would drive these audiophiles nuts.
  All that pitch shifting.

Perhaps the spectrum of the jitter matters.  If the frequency is low 
enough, I call it wander rather than jitter.  Audio doesn't need DC or low 
frequencies so wander is easy to filter out with a simple high-pass filter.

Heartbeats may be more interesting than breathing.  Does anybody know of 
spectrum domain data?  It should be possible to collect position info while 
also monitoring heartbeat and chest diameter and then crunch some numbers do 
see how much of the position correlates with heartbeat vs breathing and then 
plot each part in the frequency domain.


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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Dan Kemppainen
One interesting note however. Years ago we had a standard old 4040 
ripple counter in our shop that displayed a low occurrence of jitter 
of several times it's input frequency period at it's lowest frequency 
output (Sort of what you are describing below). I wish I had the 
numbers handy, but the output would be good for most of the time, then 
every once in a while it would jump to a longer delay. It was hard to 
catch with a scope, but when we measured every single pulse width it 
showed up fairly well. The high speed clock (A TTL OSC in a can) never 
skipped, as far as we know.
We never did figure that one out. From what I remember we switched IC 
manufacturers, and the problem when away.



Dan


On 5/10/2012 1:49 PM, time-nuts-requ...@febo.com wrote:

A TTL can that is marked 4.096 MHz costs about $2 and will make a
square wave with a period of very close to 250 nS.   Then they divide
this down to the sample rate of 96KHz.   In order to see a 250 nS
jitter in the 96K signal the TTL can would have to skip a beat.
250 nS is is a huge error and you don't get there with digital noise

Chris Albertson
Redondo Beach, California


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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Jim Lux

On 5/10/12 11:36 AM, Hal Murray wrote:


albertson.ch...@gmail.com said:

I've alway have thought that if nanosecond level jitter is bad then
breathing while listening must be really bad.  If you inhale the path length
from your ear to the speaker changes at the microsecond level.
  You'd think the resulting doppler shift would drive these audiophiles nuts.
  All that pitch shifting.


Perhaps the spectrum of the jitter matters.  If the frequency is low
enough, I call it wander rather than jitter.  Audio doesn't need DC or low
frequencies so wander is easy to filter out with a simple high-pass filter.

Heartbeats may be more interesting than breathing.  Does anybody know of
spectrum domain data?  It should be possible to collect position info while
also monitoring heartbeat and chest diameter and then crunch some numbers do
see how much of the position correlates with heartbeat vs breathing and then
plot each part in the frequency domain.





oddly, I happen to have just that data to hand, having been looking at 
ballistocardiography.  If you put someone in a bed, suspended by 4 wires 
(one at each corner), your heart beat results in about 1mm displacement 
(head to foot).  1 degree phase shift at 1 kHz, or thereabouts.



in terms of displacement in general, breathing is on the order of 1cm 
(at 0.1 Hz) and heartbeat is on the order of 0.1-1mm, depending on where 
you look.  (look up microwave cardiography for instance)


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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Jim Lux

On 5/10/12 12:44 PM, Chris Albertson wrote:

On Thu, May 10, 2012 at 10:25 AM, Heinzmann, Stefan  (ALC NetworX
GmbH)stefan.heinzm...@alcnetworx.de  wrote:


Benjamin and Gannon, the first reference in Ashihara's paper, come to lower 
figures for sinusoidal jitter with carefully selected frequencies relative to 
the main signal, which is also sinusoidal. Their results reach down to the 
single figure nanosecond range, and that can be regarded as the real limit of 
audibility.



Are there any real audio systems with sinusoidal jitter.


powerline ripple on a signal going into a threshold detector that drive 
the sample clock would be a nice way to generate sinusoidal jitter.


I've got a nice example where 66MHz processor clock modulates a 49MHz 
sample clock (well, it's not perfectly sinusoidal, but if you digitize a 
clean sine wave, you get nice aliases of the modulation frequency).





  I'd goes

that it would all be random.  I can see where I could build a
system with that defect if I wanted to but are there any systems on
the market like this?


Chris Albertson
Redondo Beach, California

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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Dan Kemppainen
Don't forget the human mind can compensate for a lot of things. Think 
of how we can triangulate a sound source in realtime even with the 
included echos in a small room. The only thing that I can think of 
that messes with that system is a single tone setting up standing 
waves. It's impressive if you think about it.


So, it's probably not much of a stretch to imagine the mind 
compensating for a little movement here and there (since we have 
controls and feedback to monitor that). It may just take a few 
thousand years for us to evolve to deal with distortion due to jitter 
in our digital recordings :)


All fun aside. This has been a worth while thread in my opinion. I'm 
learning more this week, than others watching this list!


On 5/10/2012 1:49 PM, time-nuts-requ...@febo.com wrote:

I've alway have thought that if nanosecond level jitter is bad then
breathing while listening must be really bad.  If you inhale the path
length from your ear to the speaker changes at the microsecond level.
  You'd think the resulting doppler shift would drive these audiophiles
nuts.  All that pitch shifting.


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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Chris Albertson
On Thu, May 10, 2012 at 1:09 PM, Jim Lux jim...@earthlink.net wrote:
 On 5/10/12 12:44 PM, Chris Albertson wrote:


 powerline ripple on a signal going into a threshold detector that drive the
 sample clock would be a nice way to generate sinusoidal jitter.

I can think of other ways to design a defective system.  But what I
was wondering if there are in fact any audio systems on the market
like this.

The real reason that a recording studio will use a master clock is not
because you can hear nS level jitter.  It's that they like to use a
sample accurate editor and they want to be able to over dub tracks
recorded at different times, maybe months apart

Chris Albertson
Redondo Beach, California

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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Hal Murray

 Are there any real audio systems with sinusoidal jitter.   I'd goes that
 it would all be random.  I can see where I could build a system with
 that defect if I wanted to but are there any systems on the market like
 this? 

I could easily imagine jitter with a significant sinusoidal component.  
Consider noise from the power supply leaking through.  If you think of it as 
phase noise rather than jitter it would show up as a spike and harmonics.


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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Hal Murray

d...@irtelemetrics.com said:
 One interesting note however. Years ago we had a standard old 4040  ripple
 counter in our shop that displayed a low occurrence of jitter  of several
 times it's input frequency period at it's lowest frequency  output (Sort of
 what you are describing below). I wish I had the  numbers handy, but the
 output would be good for most of the time, then  every once in a while it
 would jump to a longer delay. It was hard to  catch with a scope, but when
 we measured every single pulse width it  showed up fairly well. The high
 speed clock (A TTL OSC in a can) never  skipped, as far as we know. We never
 did figure that one out. From what I remember we switched IC  manufacturers,
 and the problem when away. 

I expect ever old timer who did serious glitch chasing 25-30 years ago has 
similar stories.

Mine involved a dual port RAM, 16x4.  I think it was a 74S189.  It was made 
by many companies.  Chips from one vendor just occasionally screwed up.  We 
were using them in FIFOs on ethernet boards.  (They were probably used other 
other places too.  I was worked on networking.)

Back then, most large electronics companies had incoming inspection and 
qualification groups.  I remember stories of somebody contacting that group 
for help and getting a response of roughly That's why we didn't qualify 
them.  (We were a research group.  We purchased small batches of chips 
wherever we could get them without going through the official qualified list.)


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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Attila Kinali
On Thu, 10 May 2012 19:25:33 +0200
Heinzmann, Stefan  (ALC NetworX GmbH) stefan.heinzm...@alcnetworx.de wrote:

 Benjamin and Gannon, the first reference in Ashihara's paper, come to lower 
 figures for sinusoidal jitter with carefully selected frequencies relative 
 to the main signal, which is also sinusoidal. Their results reach down to 
 the single figure nanosecond range, and that can be regarded as the real 
 limit of audibility.
 
 Of course, that still leaves those who claim to hear jitter in the 
 picoseconds range out in fairy-tale land.

I'm not so sure there. Having been active in the video coding scene
for quite a while i know that a trained ear/eye can be easily a factor
10 better than the average. Eg, after a couple of months of hunting for
bugs in the A/V sync code, i got sensitive to A/V desync down to 3-5ms.
Yes, i know this is below the frame rate. But i cannot otherwise explain
it as something started to feel odd, when the desync got over that limit.
From somewhere between 5 and 10ms i could usually tell in which direction
the desync was. Yes.. still below the frame rate.

And i know that there are eyes and ears out there that are much better
trained than mine ever could be. So i wouldnt imediatly dismiss it, if
someone would tell me he could detect a A/V desync of 1ms (given proper
frame rate).

So, if someone proves that 1ns jitter is audiable for an average person,
i would definitly not decline the possibility that a trained ear can hear
100ps jitter.

But that is still a jitter level you can get with an crystall quite easily.
You need to take care of a few things to not introduce any jitter from
power supply or bad shielding, but nothing too difficult.

Attila Kinali

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

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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Attila Kinali
On Thu, 10 May 2012 11:36:40 -0700
Hal Murray hmur...@megapathdsl.net wrote:

 Heartbeats may be more interesting than breathing.  Does anybody know of 
 spectrum domain data?  It should be possible to collect position info while 
 also monitoring heartbeat and chest diameter and then crunch some numbers do 
 see how much of the position correlates with heartbeat vs breathing and then 
 plot each part in the frequency domain.

Please do not forget that there is a quite sofisticated error correction
system attached to the hear, which we usually refere to as the brain.

Breathing and heart beat are filtered out and corrected for by the brain,
otherwise we would have difficulties to hear a lot of things.
For more infos, please have a look at perceptual psychology. A not too
bad introduction to that field is Sensation and Perception 
by E. B. Goldstein.

But please do not expect mathematical rigor in that field. It's still
a subfield of psychology.

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

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Re: [time-nuts] Clocks for Audio gear

2012-05-10 Thread Tom Van Baak

This audio thread had some interesting information; thank you.

Now I welcome you to get back to the focus of the group; please.

Thanks,
/tvb

p.s. If we need to start another mailing list that includes audio let me know; 
contact me off-line.



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Re: [time-nuts] Clocks for Audio gear

2012-05-09 Thread Azelio Boriani
You can refer to this for a relation between SFDR and INL:

http://www-inst.eecs.berkeley.edu/~ee247/fa07/files07/lectures/L14_f07.pdf


On Wed, May 9, 2012 at 11:25 PM, Hal Murray hmur...@megapathdsl.net wrote:

 was Subject: Re: [time-nuts] Faster than light of a different type
 (Probably my fault.)

 act...@hotmail.com said:
  What I found funny was that the Audiophlie and light thread drew such
  attacks when it hit home to me as exactly what the Time-Nuts mission is
  about.  The Audio thread touched on some real world time and freq
 research
  ...

 I too enjoyed the technical discussions.  Thanks for your contributions.

 It's the audiophool bashing that people are complaining about.  Sure, it's
 fun, but only at the right time and it gets old quickly.  The problem is
 that
 with large groups, there are different opinions of when and how much is
 appropriate.  The long tail on opinions of reasonable can annoy a lot of
 people.

 ---

 Back to technical stuff...

 As a practical matter, is clock jitter or phase noise from a typical low
 cost
 crystal and decent board layout a significant problem in audio gear?  How
 hard is it to measure?

 Is clock accuracy a practical problem?  How good are people with perfect
 pitch?  It wouldn't surprise me if there are a few that are much much
 better
 than others, but how good is that relative to 50 PPM which I can get in a
 low
 cost crystal?

 Video geeks solved their clock distribution problems by using frame
 buffers.
 Is there a similar trick for audio?  Is there a need for it?

 

 I know clocking is a serious problem in fancy DSP systems.  For example,
 modern radar has gone digital.  In that context, clock jitter can be
 important.  Standard procedure is don't run your clock through a FPGA
 because
 it will add jitter.

 Part of the problem is that they are doing magic down conversion in the
 ADC.
 (I can't think of the term.)  Suppose you have a 100 MHz signal with a 1
 MHz
 bandwidth.  You don't have to sample at 200 MHz.  You can sample at 2 MHz
 and
 your signal will alias down.  It's turning what is normally a bug into a
 feature.  The catch is that the errors/noise due to clock jitter happens at
 the high frequency, in this case multiplying the noise by 100.  (Your
 sample/hold at the front end has to work at the high frequency and your
 anti-aliasing filter gets more interesting.)

 There has been an interesting change in the specs for ADCs and DACs over
 the
 past 20(?) years.  They used to be specified using terms like DNL and INL
 and
 No-missing-codes.  Modern high-speed ADCs are specified with terms like
 ENOB
 and SFDR.  Data sheets often include several plots of a batch of samples
 run
 through a DFT so you can see the noise floor and such.

 Here is a reasonable glossary:
  http://pdfserv.maxim-ic.com/en/an/AN641.pdf

 I don't remember comments/specs about clock jitter in the data sheets but I
 haven't looked at one in a few years.  I'll have to keep an eye out the
 next
 time I'm browsing.



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