Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread DaveWr

Read my Dan Lavry post, that document gives the full proof !!

Nyquist is an absolute!


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Phil Leigh

bhaagensen;694440 Wrote: 
 Its a nice read and he's probably right in his conclusions - however one
 of his main poinsts, as I see it, can not be inferred (only) from what
 he's writing. In particular, the reconstruction of a discrete signal
 into a continuous one c.f. Nyquist. In fact to the contrary - the
 reconstruction part of the theorem directly requires an infinite sum.
 This is easy in mathematics, but how the engineers implement it is
 still something even folks like Monty does not explain in any detail -
 in fact, its often completely ignored... But why???
 
 
 Edit: So basically his whole sampling-rate argument relies on the *one*
 statement that the analog signal can be reconstructed losslessly.
 There is not a single word of argument in favor of this... its not
 enough to reference Nyquist, is it.

Yes it is enough in this case. Nyquist-Shannon Sampling Theorum came
long before the advent of digital audio. Somewhat like Einteins
theories of Relativity coming before space travel. For example,
Einstein predicted (amongs many things) that time would slow down in a
fast moving clock relative to a fixed observer. Very few people
believed this counter-intuitive assertion. Experiments in the 50's
later proved this assertion to be in fact true.

Sampling Theorum predicts perfect reconstruction provided that the
lowest frequency to be reconstructed is below the Nyquist limit. The
ONLY people who DON'T believe this to be true are:

1) some people selling outrageosuly priced audiophile snake oil
2) some people who write for hi-fi mags.
3) some people who have been listening to the nonsense spouted by 1 or
2 above.


The point is that increasing the sampling frequency does not and cannot
alter the sampling and reconstruction of frequencies below the Nyquist
limit. What it does do is to move the Nyquist limit. That's all it
does.

As others have mentioned, moving the limit upwards can be useful when
engineering to avoid side-effects from the necessary recovery filters
for a DAC... but this is only really relevant to analogue filters
(which are generally no longer used).

It is certainly NOT necessary to set a Nyquist limit 4x higher than the
extreme upper limit of human hearing!.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread darrenyeats

Thinking about the other end of the spectrum, this is one reason for the
sub versus no-sub debate. By nature, ears register higher frequency
distortion products more than the deeper fundamentals the sub is
producing. And keeping distortion low at low frequencies is very
difficult.
Darren

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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread bhaagensen

Its not that I'm questioning Nyquist - but its a *mathamatical* theory
involving the concept of infiniteness - in particular the
reconstruction theorem involves an infinite sum.

If you read the wikipedia page on this stuff, its noteted that for this
reason, any real-world implementation will only be able to approximate
the reconstruction part. Its clever and good stuff, but not implied by
the theorem itself.

What I've yet to come across, are good arguments comparing the
precision in approximation cf. any usefulness added information in
terms of higher sampling rates, could have?

(I read Dans paper long ago and AFAIR he doesnt dicsuss this much
either - but I'll have another look).

PS I've made this complaint on these forums eariler - no avail :)
PSS If you want to argue by scientific standards, this is a part not to
be left out.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread bhaagensen

Perhaps look at this this way. The circumference of a circle is pi*d -
easy. Very few people know how to compute this *exactly* on a computer
- not so easy :)

(However its easy to approximate within an error of...)


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread DaveWr

Irrelevant analogy, just because you use an irrational number in a
calculation, which is always approximated.  

On that basis sine waves, which most people believe in aren't accurate,
as they require an infinite series...


sin x = x - x3/3! + x5/5! - x7/7! +


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Re: [SlimDevices: Audiophiles] ZMIX Converter Test(60 generations of AD/DA bounce)

2012-03-07 Thread Mnyb

pski;694042 Wrote: 
 State the point. Point to evidence.

The audibility of ADC's and DAC's is highly exaggerated especially if
they are of half decent design .


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Mnyb

If it where not it would not work you would not get =20kHz upper limit
the fact that a DAC reproduce 20k from a 44.1 sampled signal proves
that it is actually working closely enough to Nyquist and likewise if
an ADC samples with 44.1k and still gets 20k it is working ? ( the
theoretical fs/2 would be 22.05kHz )

Most DAC's in test also produces textbook impulse response graphs and
have no more distortion and intermodulation etc close to 20kHz than
they have anywhere else , there are exceptions my crappy PC soundcard
or probably pulse audio re-sampling etc (ubuntu) produce clear
intermodulation products .

So when it's not working it's very evident it just fails and you have
imaging/aliasing all over the place , when I get home I could produce
links to interesting intermodulation test tones for DAC's these are
very close to fs so if your dac have problems close to fs you will hear
it :)

These tones are on 44.1 base and 48kHz base respectively

Also the crude visual examples in the article , where a handful of sinc
functions get you pretty close to real reconstruction , where in reality
practical gear have much higher precision .

The infinite is a red herring in this case , if it where huge problems
with practical implementations it would be very evident.

Practical gear is close enough
http://forums.slimdevices.com/showthread.php?p=694461#post694461


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread bhaagensen

The analogt may not be aces, but the point was that in regards to the
article it doesn't say much new that hasnt already been said a zillion
times. The final parts of the argument is missing though I'm sure the
engineers know. There must be some (scientific) epsilon-bound on how
little DACs miss the mark?

So there is still room for the audiophile crowd to go ballistic on
sampling rates - as I see it.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread bhaagensen

In conclusion i agree with you mnyb and others. Its just that wrt the
end result the rigidness of mathematical sampling theory does (alone)
not carry through all the way.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Mnyb

bhaagensen;694465 Wrote: 
 The analogt may not be aces, but the point was that in regards to the
 article it doesn't say much new that hasnt already been said a zillion
 times. The final parts of the argument is missing though I'm sure the
 engineers know. There must be some (scientific) epsilon-bound on how
 little DACs miss the mark?
 
 So there is still room for the audiophile crowd to go ballistic on
 sampling rates - as I see it.

Well as described oversampling is used both in recording and playback
.

So it's about the intermediate carrier to the listener .

In that case a modern DAC would not even theoretically have problems
playing back 44.1 it's doing at much higher rate anyway so it could be
made watertight that way .

And the reducing to 44.1 from something much higher is carried out in
software at the record company no messing around with analogue filters
in this case It is perfectly reconstructed sound 0-20khz in our audible
range .

So eventual non nyqvist ? behaviour is outside the bandwidth we care
about anyway, why cares if the DAC is not perfect at 100kHz it sure is
at 20kHz .

So the argument should be about the format of delivery to a consumer
for that format to carry all information that we can hear it's enough
if it is sampled to that limit. And as said it can be perfectly
reconstructed from such information, the weapon of choice is in many
cases oversampling .


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread chill

What a good article.  I loved his statement in the outro:
 The more that pseudoscience goes unchecked in the world at large, the
 harder it is for truth to overcome truthiness... 

Almost by way of an example (although he doesn't make the link
himself), in footnote 18 he includes a quote from Wired magazine:
 Some purists will tell you to skip FLACs altogether and just buy WAVs.
 [...] By buying WAVs, you can avoid the potential data loss incurred
 when the file is compressed into a FLAC. This data loss is rare, but it
 happens.

I wonder if Wired picked that up from TAS. :)


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread mlsstl

bhaagensen;694467 Wrote: 
 In conclusion i agree with you mnyb and others. Its just that wrt the
 end result the rigidness of mathematical sampling theory does (alone)
 not carry through all the way.

Human hearing does not carry through all the way either. We are also
not infinite. 

Audiophiles spend a lot of time rearranging deck hairs on the Titanic.
But it does make them feel better. ;-)


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread firedog

Unfortunately, the picture isn't as clear or as simple as he tries to
portray.

The Boston Audio DBT had flaws which are pointed out in many critiques
all over the net.

Among other things, he seems to thing the point of high res recordings
is so we can hear high frequenies (above 20k); that of course isn't the
point.

He talks about macro dynamics of music, but not microdynamics.

Finally, he may be correct. But it may also be irrelevant that he is.
Some have made the case that you can hear everything on a well produced
Redbook file that you can hear on a hi-res file, only some of the detail
is easier to hear on a high res file. This alone can account for the
subjective impression that the hi res sounds different or better.


In any case, even the authors of the DBT admit that there best sounding
files were those from SACD. This they attribute to better/different
mastering of the hi-res files, and not to any inherent superiority of
the format. That may be. But then that alone is a good reason to buy
them and listen to them.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Soulkeeper

firedog;694484 Wrote: 
 He talks about macro dynamics of music, but not microdynamics.

But what is microdynamics? Does it have a sensible definition, or is it
just one of those woo-words that can mean anything that the user wants
it to? (something like Spiritual Holistic Quantum Dynamics?)


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread mlsstl

firedog;694484 Wrote: 
 Unfortunately, the picture isn't as clear or as simple as he tries to
 portray.
 
 The Boston Audio DBT had flaws which are pointed out in many critiques
 all over the net
 
 Among other things, he seems to thing the point of high res recordings
 is so we can hear high frequenies (above 20k); that of course isn't the
 point.
 
 He talks about macro dynamics of music, but not microdynamics

The inability of humans to achieve perfection in blind testing seems to
be the touchstone used by those audiophiles to dismiss the results of
any test that challenges their beliefs. 

As many others have pointed out, conducting good blind tests is
difficult and even the best tests will never cover all theoretical
possibilities. That's true whether the subject is audio or something
else. 

OK, even if perfection in such tests is impossible, what does one do
with the accumulated evidence that says that many of the breathtaking
differences audiophiles hear under sighted conditions are either wildly
exaggerated or even imaginary? 

In the post quoted above, the case for supersonic frequency response is
damaged, so the issue is switched to microdynamics - another
audiophile term that has no specific meaning beyond the vague
impression assigned by each individual. 

Can one give an example of a sound that gets lost on a CD that would be
heard on a higher-rez recording? In addition to the Empire Brass (brass
is always a challenge to record well), I also listened to some Boston
Camerata last night, an early music choral group. I could clearly pick
out individual voices from the 30 or more singers, whether massed, lead
or background singers. What did the CD lose that a high-rez would have
revealed? Clearly and accurately presenting the voices of 30 people
singing together would seem a good test for the ability of a recording
to maintain clarity and not lose articulation. 

Yes, I've heard a number of high-rez recordings and they do tend to be
excellent. But I also have many CDs with superb audio quality. I rather
strongly suspect the quality of the high-rez recordings is far more due
to extra care given in the recording, mixing and mastering process than
any inherent technical advantage. These recordings are marketed to a
discriminating group of listeners and typically cost more, so they
better be giving them something for their money.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread bluegaspode

bhaagensen;694457 Wrote: 
 Its not that I'm questioning Nyquist - but its a *mathamatical* theory
 involving the concept of infiniteness - in particular the
 reconstruction theorem involves an infinite sum.

I also think this is the weakest spot in the argumentation of the
paper.
While in theory with Nyquist I can reconstruct the waveform with the
lower bound of samples (i.e. 2x frequency) perfectly, I need infinite
sums for it - so speaking in algorithms: I cannot compute it in
reasonable time.

So what is done in DACs is to approximate the original waveform with as
much sums as possible in a given timeframe.

Let's say for proper playback I can do 100calculations (take whatever
number you like) in 1ms.
Based on 100 calculations: can I get a better approximation of the
original waveform if I have 100 samples or 200 samples ?

Reading Wikipedia articles about Nyquist doesn't give me an answer for
that.
But that's the central question that needs to be answered, not if
Nyquist in theory can reproduce a waveform with a given number of
samples.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread DaveWr

bluegaspode;694492 Wrote: 
 
 
 So what is done in DACs is to approximate the original waveform with as
 much sums as possible in a given timeframe.
 
 

NO

The DAC generates a stepped ladder function, the reconstruction filter
turns this into the recovered waveform.

In the case of many modern, oversampling DACs, the reconstruction
filter is a simple 2nd order analogue filter.  No limited number of
calculations.  Yes the transform theory for the filter is a set of
infinite sums, but the filter is a filter.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread DaveWr

bluegaspode;694492 Wrote: 
 
 
 Let's say for proper playback I can do 100calculations (take whatever
 number you like) in 1ms.
 Based on 100 calculations: can I get a better approximation of the
 original waveform if I have 100 samples or 200 samples ?
 
 

NO

Assuming the 100 sampling frequency met Nyquist criteria and the ADC
didn't use old style brickwall anti-alias filters, and that the anti
alias filters were set to the same frequency, the post DAC signal would
be identical in both cases!  You could get Phil to check using his
difference test, there would be nothing.  

Above Nyquist sampling means all the data is available to recover the
total signal.  It doesn't need infinite processing power, it is an
analogue technique.

The modern approach using oversampling take the anti aliasing ADC
filters and the reconstruction DAC filters way out of audible effects
range.  
However there are still the matters of ADC and DAC linearity, are the
steps really equal;
what is the clock v data jitter presented to the DAC, that effectively
changes step sizes with time (note the higher the sample rate the
tighter the jitter spec needs to be);
what is the underlying noise level.  24 bit DACs don't usually achieve
much more than 21  to 22 bits of signal to noise performance.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread pippin

This is just wrong.
You don't need an infinite sum or anything, in theory you don't even
need a DAC, all you need is a low-pass filter. A DAC is nothing else
than a number of carefully tuned band-pass filters followed up by a
low-pass filter to clean things up.

No sums involved. No infinity involved.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Wombat

firedog;694484 Wrote: 
 He talks about macro dynamics of music, but not microdynamics.
Can you explain what this term microdynamic correpondents with from a
technical standpoint of view?
I lately linked to that thread already if you talk about time
resolution:
http://www.hydrogenaudio.org/forums/index.php?showtopic=91126hl=
You can always ask J.J. Atkinson for some math that underlines your
statement. I am sure he will help you.


The issue playing back signals above 20kHz can damage playback in the
audible range is really interesting but not much sources i can find. 
The intermodulation of amps can be measured but even this is not easy
audible. The speakers themself do so much more harm it doesn´t really
matter.
What i recognize is that with some newer BW speakers like the CM8 i
see a hefty +10dB resonance on-axis at ~30kHz in frequency response
graphs i found on the net. So playing back any content at that
frequency may indeed trigger the tweeters break-up and distort sounds
below.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Mnyb

As promised link to aliasing tests.

http://www.audiocheck.net/audiotests_aliasing.php

My own ungodly soundblaster cheapest possible souncard aliase above
18kHz

And HF sweeps 22kHz down to 12kHz .

http://www.audiocheck.net/audiotests_frequencycheckhigh.php

These are 44.1 kHz files, so here you have it a 22k tone encoded by
44.1kHz sampling and it works

this nice discussion re dacs is besides the point .

The point is can a sample rate such as 44.1k or 48k encode the complete
wave form below 20kHz of-course .

Wanting higher rate is the same as wanting ultrasonics the quality
below 20kHz does not improve as we have already encoded all information
we need for that (nyqvist ) .
So what god comes with the ultrasonics ?


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Mnyb

Wombat;694512 Wrote: 
 Can you explain what this term microdynamic correpondents with from a
 technical standpoint of view?
 I lately linked to that thread already if you talk about time
 resolution:
 http://www.hydrogenaudio.org/forums/index.php?showtopic=91126hl=
 You can always ask J.J. Atkinson for some math that underlines your
 statement. I am sure he will help you.
 
 
 The issue playing back signals above 20kHz can damage playback in the
 audible range is really interesting but not much sources i can find. 
 The intermodulation of amps can be measured but even this is not easy
 audible. The speakers themself do so much more harm it doesn´t really
 matter.
 What i recognize is that with some newer BW speakers like the CM8 i
 see a hefty +10dB resonance on-axis at ~30kHz in frequency response
 graphs i found on the net. So playing back any content at that
 frequency may indeed trigger the tweeters break-up and distort sounds
 below.

Don't forget that by some perverse coincidence many that are very keen
on hirez also use tube amps wonder what happens around the ultrasonic
resonance in the output transformer ? or in general with such a design
that struggles a bit at normal hf ? especially with SACD a lot of air
and detail if you are unlucky , maybe ? but again this is stuff nobody
have tested either .

I have always wondered in general if not many exotic designs used by
many audiophiles have unforeseen technical limitations that no normal
engineer expects that can explain why some things are audible that
should not be ?


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread pippin

bhaagensen;694459 Wrote: 
 Perhaps look at this this way. The circumference of a circle is pi*d -
 easy. Very few people know how to compute this *exactly* on a computer
 - not so easy :)
 
 (However its easy to approximate within an error of...)

Again: you don't COMPUTE anything here. We are talking about analog
signals.

To stick with your example: even though you can't compute the exact
circumference of a circle to infinity, you can still draw one. And you
don't have to do any calculations to do so.

All you have to know is the radius, a single number, no additional
information required.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread ralphpnj

Mnyb;694320 Wrote: 
 So what god comes with the ultrasonics ?

I guess it would be this one?:


+---+
|Filename: futfic_5107.jpg  |
|Download: http://forums.slimdevices.com/attachment.php?attachmentid=13146|
+---+

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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread ralphpnj

pippin;694517 Wrote: 
 To stick with your example: even though you can't compute the exact
 circumference of a circle to infinity, you can still draw one. And you
 don't have to do any calculations to do so.
 
 All you have to know is the radius, a single number, no additional
 information required.

Plus this handy little tool:


+---+
|Filename: compass_drawing_circle.jpg   |
|Download: http://forums.slimdevices.com/attachment.php?attachmentid=13147|
+---+

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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread bluegaspode

DaveWr;694502 Wrote: 
 Above Nyquist sampling means all the data is available to recover the
 total signal.  It doesn't need infinite processing power, it is an
 analogue technique.

Well sorry to be playing devils advocate here.
But right now I read the argument as follows

a) there is a theorem which proves that under ideal (in the realworld
not achievable) conditions a sampling frequency of x will be good
enough
b) we use some totally different process to do the reconstruction but
still refer to the theorem and claim that it still applies.

For me this sounds like comparing Apples with Bananas.

Or to put it differently:

Mr. Dan Lavry begins his document with the statement:
 
 The great value offered by Nyquist's theorem is the realization that we
 have ALL the  information with 100% of the detail, and no distortions,
 without the burden of extra fast sampling.  
 
(sorry I didn't have time yet to read the paper to the end, will do it
later).

And quite opposite to this statement a quote from Wikipedia about
Nyquist:

 
 In practice, neither of the two statements of the sampling theorem
 described above can be completely satisfied, and neither can the
 reconstruction formula be precisely implemented. The reconstruction
 process that involves scaled and delayed sinc functions can be
 described as ideal. It cannot be realized in practice since it implies
 that each sample contributes to the reconstructed signal at almost all
 time points, requiring summing an infinite number of terms. Instead,
 some type of approximation of the sinc functions, finite in length, has
 to be used. The error that corresponds to the sinc-function
 approximation is referred to as interpolation error. 
 

So as the preconditions of the theorem cannot be met (according to
Wikipedia, sorry I don't have better source nor knowledge) I think it
is a valid question if we can overcome the (possible) deficiencies with
higher sample rates (or other means).

So for me this part of the argument has its flaws (while of course I'm
very happy to believe all the other proofs which include double blind
tests)


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Mnyb

ralphpnj;694519 Wrote: 
 I guess it would be this one?:

:P I corrected that now , ooh pulp sci fi mag..


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Soulkeeper

bluegaspode;694522 Wrote: 
 So as the preconditions of the theorem cannot be met (according to
 Wikipedia
Wikipedia is written by nerds. Therefore: As long as something that can
safely be ignored in practice, cannot be ignored in principle, Wikipedia
will not ignore it.

As an example, see the article on Evolution; it actually even mentions
Intelligent Design (in the last sentence, under Social and cultural
responses).


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Mnyb

Soulkeeper;694524 Wrote: 
 Wikipedia is written by nerds. Therefore: As long as something that can
 safely be ignored in practice, cannot be ignored in principle,
 Wikipedia will not ignore it.
 .

Exactly, now play that 44.1 encoded hf sweep 22  12 kHz does it work ?
the nyqvist freuency is 22.05 kHz and it begins with a 22k tone a pretty
close shave to me , we that are not 14 may take it for granted as we
will begin to hear the sweep at 17-13k , or just look at it in
audacity

Sine waves what does it really tell ? another long dead fella to the
rescue Fourier he discovered that a wave can be described as a sum of
sine functions , simplified.

Speaking of... for the complete math nerds of wikipedia
http://en.wikipedia.org/wiki/Fourier_series
look at the pictures don't read, headache inducing :-/

the inverse is known to us as these nice spectrum's we use to analyze
HD-tracks so called hirez files aka fft


-- 
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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread bluegaspode

You are all free to provide me with better resources than Wikipedia :)


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread darrenyeats

mlsstl;694489 Wrote: 
 I could clearly pick out individual voices from the 30 or more singers,
 whether massed, lead or background singers. What did the CD lose that a
 high-rez would have revealed? Clearly and accurately presenting the
 voices of 30 people singing together would seem a good test for the
 ability of a recording to maintain clarity and not lose articulation. 
 
It's hardly fair to ask people not to trust their ears and then provide
arguments like this. Shame on you (just kidding!)


Soulkeeper;694524 Wrote: 
 Wikipedia is written by nerds. Therefore: As long as something that can
 safely be ignored in practice, cannot be ignored in principle,
 Wikipedia will not ignore it.
 
 As an example, see the article on Evolution; it actually even mentions
 Intelligent Design (in the last sentence, under Social and cultural
 responses).
But would you say that about Wikipedia if Wikipedia had agreed with
your own views?

I am not against your conclusions but it's all getting a bit mob-like
around here!
Darren


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread darrenyeats

Mnyb;694516 Wrote: 
 I have always wondered in general if not many exotic designs used by
 many audiophiles have unforeseen technical limitations that no normal
 engineer expects that can explain why some things are audible that
 should not be ?
I find that a very interesting point.
Darren


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread mlsstl

darrenyeats;694534 Wrote: 
 It's hardly fair to ask people not to trust their ears and then provide
 arguments like this. Shame on you (just kidding!)

My apologies to the group. I am thoroughly chastened. Is purchasing a
album from HDTracks that I already have on CD sufficient penance?


-- 
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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Mnyb

mlsstl;694536 Wrote: 
 My apologies to the group. I am thoroughly chastened. Is purchasing a
 album from HDTracks that I already have on CD sufficient penance?

It must be a 40 year old rock classic ripped from SACD and get the 192k
version ;) bonus point if they used thier  cryogenic treated cables in
the process ( that was not a joke they do that, amazing ?)


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Phil Leigh

bluegaspode;694522 Wrote: 
 Well sorry to be playing devils advocate here.
 But right now I read the argument as follows
 
 a) there is a theorem which proves that under ideal (in the realworld
 not achievable) conditions a sampling frequency of x will be good
 enough
 b) we use some totally different process to do the reconstruction but
 still refer to the theorem and claim that it still applies.
 
 For me this sounds like comparing Apples with Bananas.
 
 Or to put it differently:
 
 Mr. Dan Lavry begins his document with the statement:
 
 (sorry I didn't have time yet to read the paper to the end, will do it
 later).
 
 And quite opposite to this statement a quote from Wikipedia about
 Nyquist:
 
 
 
 So as the preconditions of the theorem cannot be met (according to
 Wikipedia, sorry I don't have better source nor knowledge) I think it
 is a valid question if we can overcome the (possible) deficiencies with
 higher sample rates (or other means).
 
 So for me this part of the argument has its flaws (while of course I'm
 very happy to believe all the other proofs which include double blind
 tests)
It is trivial to show that the error inherent in the process is below
the range of detection.
A computer-generated sine wave  dataset (I.e. not recorded via an ADC)
can be passed through a DAC and null-compared. I've done this and the
result should be no surprise to anyone. 
PS don't try this with a nos DAC!

The human ear/brain also operates on a sampling basis (there's nothing
analogue about the ear by the way) and uses a reconstruction filter
just like a DAC to integrate the discrete samples into something we can
understand.

What some people are misunderstanding here is that it is the
reconstruction filter that recovers the analogue signal, not the DAC.
The DAC simply presents the filter with a set of voltages over time. It
is within the filter that the sinc function becomes manifest and this is
indeed an infinite series - the mathematical definition of a filter is a
continuous function over time. A filter is not a step function!

To be clear on this, what comes out of the filter IS a mathematically
perfect sine wave... All the way up to the Nyquist frequency. This is
both predicted by the theorem and demonstrable in practice. There is no
known way to differentiate between a 1khz sine wave sampled at 44.1 or
192. In every conceivable way of  measuring or analysing that sine
wave it they will be indistinguishable.

What is true for one sine wave is also of course true for any
combination of sine wAves (or as we usually refer to it... Music).

Or do we need to have a conversation about Fourier transforms and the
practical implications of mathematical infinite series as well?


Now there IS a way to break this model. Try a perfect square wave!.
This has to be done using a mathetmatically generated waveform/data set
because it is impossible to generate or record a perfect square wave
with infinite rise/fall times. The filter will introduce non-linear
ringing.

This is all predict by the theorem because an infinitely fast slope
requires an infinite number of samples... I'm sure you get the idea.

Bottom line is this; for real world sound distribution 44.1 is fine ...
Which is why there are many many fine sounding red book CD's... And why
there is no published evidence that stands scrutiny to support the idea
that anyone can tell the difference between the same master distributed
and played back at 192 or down sampled to 44.1.


-- 
Phil Leigh

You want to see the signal path BEFORE it gets onto a CD/vinyl...it
ain't what you'd call minimal...
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Re: [SlimDevices: Audiophiles] How low is low?

2012-03-07 Thread darrenyeats

Well, my sub journey continues.

I have now settled on placing my sub at the same distance as my main
speakers (centre of driver basis).

Because the sub gives better massage sub-sonic performance when it's
near to me, coffee table style, I've been trying to integrate it there.
However, there or anywhere in front of the plane of the speakers,
something sounds just wrong. I think it's the phase of the bass
wavefronts being mismatched at this frequency or that. Even if I play
with the phase control on the sub, this helps certain frequencies to be
in phase, but other bass frequencies are not. The crossover is set to
lowest frequency.

This is an all ATC set up and it sounds like the crossover is designed
to integrate well...assuming the drivers are roughly in the same plane.
Certainly I've heard the exact same equipment positioned similarly in a
bigger room and I got the rightness, bass reinforcement and
massage factors at the same time. In my smaller room I get only the
first two. This may be physicscould be factors of construction or
being physically nearer the mains skewing my hearing/feeling balance
(think about headphones playing loud...a high sonic SPL doesn't equate
to body shaking amounts of air movement because the drivers are nearer,
much nearer in this example, to your ears).

Not everything is better in the current position because a big room
mode lives in the crossover region. However, I have Helmholtz
Resonators managing that. Overall, it is better. Maybe I will play with
the gain a bit.

Any advice or suggestions?
Darren

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Re: [SlimDevices: Audiophiles] ZMIX Converter Test(60 generations of AD/DA bounce)

2012-03-07 Thread TheOctavist

you got it mynb


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread maggior

Phil Leigh;694539 Wrote: 
 Now there IS a way to break this model. Try a perfect square wave!. This
 has to be done using a mathetmatically generated waveform/data set
 because it is impossible to generate or record a perfect square wave
 with infinite rise/fall times in the analogue domain. The filter will
 introduce non-linear ringing.
 
 This is all predict by the theorem because an infinitely fast slope
 requires an infinite number of samples... I'm sure you get the idea.
 

Just trying to educate myself here...

Does this mean that there could be audible distortion introduced due to
ringing in a recording that has clipped samples?  Taken to an extreme,
clipping could start to approximate a square wave.

BTW, I'm finding this discussion to be very interesting...I'm learning
a lot.


-- 
maggior

Rich
-
Setup: 2 SB3s, 4 Booms, 1 Duet, 1 Receiver, 1 Touch, iPeng on iPod
Touch, SqueezeCommander on Xoom.  SuSE 11.0 Server running
SqueezeBoxServer 7.5.5 and SqueezeSlave.  
Current library stats: 37,509 songs, 2,934 albums, 515 artists.
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Re: [SlimDevices: Audiophiles] ZMIX Converter Test(60 generations of AD/DA bounce)

2012-03-07 Thread Mnyb

This could be an interesting DAC test, wonder how NOS DAC's and DAC's
with unorthodox filters would fare ?


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread darrenyeats

mlsstl;694536 Wrote: 
 My apologies to the group. I am thoroughly chastened. Is purchasing a
 album from HDTracks that I already have on CD sufficient penance?

That more than covers it!

Sent from my HTC Sensation Z710e using Tapatalk


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Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0

2012-03-07 Thread TheOctavist

they dont make any difference, so no. :-)


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Mnyb

Hmm thinking about it there can not really be any invalid combinations
of samples ?

Whatever gets coded inside the 16/44.1 code space or whatever must be
recoverable you might not want to listen to it and the square wave is
approximate as you can't have infinite slope and it do ring .
But afaik the ringing are above 20kHz .


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Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0

2012-03-07 Thread cfraser

eduardoo;694433 Wrote: 
 
 Since then, I have been diligently reading through this thread (now up
 to page 50 or so), and I see that they were then mentioning finetuning
 of priorities.  Given that I just downloaded my TT3.0 a few days ago
 and I would probably need quite some time to go through the hundred
 pages of threads, can someone tell me if those priority tweaks are
 still required/applicable? If so, can you please point me to the right
 resource for instructions (I'm a little confused by the numerous
 replies)? Also, I see that there is a Dynabot mod out there that people
 have applied in tandem with TT2.0.  Should that be applied to 3.0 as
 well?

I don't think we want to go down that dung-strewn path again, so by all
means read it all and try what you want and form your own opinion.

My favorite and most-used setup is all the TT3.0 defaults EXCEPT with a
5k buffer and Logitech priorities. My second favorite, and what I
believe to be the most accurate setup of those presented here (as best
as I can test), is the TT3.0 default and everything Klaus, except
with a 4k buffer for here. [There has never been a decent explanation
here as to why the ALSA buffer size can make such a diff.]

I listen to internet radio quite a lot, and so far I have *never* heard
the Logitech priorities screw up the audio, whereas I have a few times
with stock TT3.0 priorities...so that's another advantage of them for
me, besides that I like the sound better (less thin). I am always using
an external DAC, and I am astonished what it does to IR via the SBT
compared to the IR I get directly via my Denon AVR (this sounds truly
dreadful, all other radio incl. AM/FM/Sirius does too).


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Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0

2012-03-07 Thread guidof

eduardoo;694433 Wrote: 
 
 Since then, I have been diligently reading through this thread (now up
 to page 50 or so), and I see that they were then mentioning finetuning
 of priorities.  Given that I just downloaded my TT3.0 a few days ago
 and I would probably need quite some time to go through the hundred
 pages of threads, can someone tell me if those priority tweaks are
 still required/applicable? If so, can you please point me to the right
 resource for instructions (I'm a little confused by the numerous
 replies)? Also, I see that there is a Dynabot mod out there that people
 have applied in tandem with TT2.0.  Should that be applied to 3.0 as
 well?
 
 Sorry for being a little too anxious to get the answer, but your help
 is appreciated.
 
 Thanks.

To the extent that any priority settings provide any benefits, these
are probably dependent on your system, your expectations re: sound
quality, and the kind of music you listen to. So, there really is no
short cut. Try a few, and stick with one you like best.

Happy listening!

Guido F.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread DaveWr

We do have to distinguish properly between CD at 44.1k sampling with
16bits  and 44.1k - 24 bits.  The 24 bits do have an effect.  They
allow a significantly greater dynamic range, assuming the mastering
engineer does't get into loudness wars compression.

It may be for some people that they compare 96/24 with CD and it is the
24bit resolution not the sampling frequency differences that are being
heard.

Also if its old music, my late 60s and 70s youth, any new releases will
be from analogue tape remastered.  This remastering may (and I know
several cases of did) produce entirely different mix from early CD
versions.  IMHO not always better.

Phil Leigh would have more input on these recording and mastering
issues.

Today with the freedom to use anti-aliasing and reconstruction filters
that are low order and relatively benign, due to very high frequency
oversampling techniques, I believe we have an almost untainted record /
replay chain.

Dave


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Soulkeeper

The article linked in the OP covers bit depth too. 

 The incorrect '96dB' figure ignores the spectral power density of a
 signal. 16 bit audio can go considerably deeper than 96dB, and deeper
 yet with proper use of dither. Handled correctly, the dynamic range of
 16 bit audio reaches 120dB in practice [10], more than twenty times
 deeper than the 96dB claim.
 
 That's greater than the difference between a mosquito somewhere in the
 same room and a jackhammer a foot away or the difference between a
 deserted 'soundproof' room and a sound loud enough to cause hearing
 damage in seconds.
 
 16 bits is enough to store all we can hear, and will always be enough.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread bluegaspode

Phil Leigh;694539 Wrote: 
 I
 What some people are misunderstanding here is that it is the
 reconstruction filter that recovers the analogue signal, not the DAC.
 The DAC simply presents the filter with a set of voltages over time. It
 is within the filter that the sinc function becomes manifest and this is
 indeed an infinite series - the mathematical definition of a filter is a
 continuous function over time. A filter is not a step function!
 To be clear on this, what comes out of the filter IS a mathematically
 perfect sine wave... 

I read through the paper posted before in the meantime
(http://www.lavryengineering.com/documents/Sampling_Theory.pdf )

Is there any more (hopefully easy to understand) information about how
these reconstruction filters work in practice? The paper comes close
around page 18, where it is shown how at least the right half of a sinc
function can be produced by some circuit.
I'm missing the left part of the sinc function because based on the
explanation it is need as well to recreate the original wave-form.

I obviously never cared about how DA converters work, based on the
paper I now think of thousands of  of sinc-producing circuits which all
add up to the final waveform. 
Probably this is not how it works in practice but this a missing link
for me now to agree that Nyquist theorem and good circuits is all that
we need.


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Re: [SlimDevices: Audiophiles] How low is low?

2012-03-07 Thread guidof

darrenyeats;694542 Wrote: 
 Well, my sub journey continues.
 
 I have now settled on placing my sub at the same distance as my main
 speakers (centre of driver 
 Not everything is better in the current position because a big room
 mode lives in the crossover region. However, I have Helmholtz
 Resonators managing that. Overall, it is better. Maybe I will play with
 the gain a bit.
 
 Any advice or suggestions? Even random ramblings like mine are
 welcome.
 Darren
 
 Sent from my HTC Sensation Z710e using Tapatalk

Personally, I found integrating a sub with main speakers is a rather
frustrating journey.

In the end, and in my room/system, the best result has come from NOT
trying for massage. Instead, I systematically tried a variety of gain
settings at the minimum crossover (30Hz), then again at a slightly
higher crossover. I focused not so much on listening for bass, but
rather for correct timbres and credible soundstage. I found the
DSpeaker (see sig) processor to be helpful in correcting for room
modes. When I finally got piano music to sound close to that coming
from a real piano, I stopped.

In my room, the best sub location is on the centerline, about two feet
farther away than the main speakers.

Keep trying! Patience has it's rewards!

Guido F.


-- 
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Marantz TT 15S1 Turntable, Virtuoso Wood Cartridge-Conrad Johnson
Motif preamp
Oppo BDP-83 Universal Player-Cambridge Azur 840C DAC
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Re: [SlimDevices: Audiophiles] 192 downsampling when decoding at server side

2012-03-07 Thread JohnSwenson

I'm going to stick my neck out here and say that I am one of those
people that can tell the difference between flac decode on server or
Touch. I have tested it with DBT and it is there. Its not huge, many
other things make a bigger difference. I have not been able to tell any
difference between the source file on the server being flac or wav. I
did do a trial with streaming flac with different compression levels,
that one was pretty inconclusive. There was maybe a hint that the
higher compression levels are slightly worse, but I wouldn't stake
anything on it. 

As to the exact mechanism, I don't know, I have a guess, but no easy
way to test it. The standard explanation that its processor load I'm
pretty sure is false, my guess is that its differences in processor
cache hit rates. The lower the cache hit rate the more the main memory
has to be accessed. Every memory access puts a large electrical load on
the system which shows up as noise on the PS and groundplane. I have
done some tests with a simple ground plane analyzer and DO see
differences in noise level and spectrum with different things going on
in the Touch. I'm in the process of building a much more sensitive
device and will be sure to do some tests with streaming type once I get
it up and running. 

I run my system wired to the Touch with everything sent PCM, I have not
had any downside to streaming PCM, and it does have a slight improvement
in sound for me. If doing so caused any issues (not playing things,
dropouts, cliks, pops, etc) I would have no qualms about streaming
flac. 

John S.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread DaveWr

maggior;694553 Wrote: 
 Just trying to educate myself here...
 
 Does this mean that there could be audible distortion introduced due to
 ringing in a recording that has clipped waveforms?  Taken to an extreme,
 clipping could start to approximate a square wave.
 
 BTW, I'm finding this discussion to be very interesting...I'm learning
 a lot.

Clipping itself is an extreme distortion, at medium power levels due
the high frequency content of these waveforms, this can easily destroy
loudspeaker tweeters.

The clipping waveform will usually have some of this high frequency
removed by the anti-alias filter that is the first part of the ADC
(analogue to digital convertor).  Although it will be now more benign,
it is still a distortion.

The ringing issue from square waves is no longer any issue, due to the
way ADCs are designed.  they don't have a very sharp (often 7th order)
'brickwall' filter at 20khz anymore.  This was always the source of
some low level errors, also repeated in the playback DAC chain.  10
years ago it was trendy to have multiple filter choices in your DAC
chain of your CD player, and they did sound slightly different.

As with all things digital speeds have gone up, and the digital guys
made it easier to get more linear ADC / DAC systems by using only a few
bits but at very high oversampled rates to achieve better results to
standard multibit DACs.

Dave


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Soulkeeper

bluegaspode;694580 Wrote: 
 Is there any more (hopefully easy to understand) information about how
 these reconstruction filters work in practice?

I found 'this'
(http://skywired.net/blog/2011/05/introducing-the-delta-sigma-modulator/),
which looks promising. I'll start reading it myself, now.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread DaveWr

Soulkeeper;694585 Wrote: 
 I found 'this'
 (http://skywired.net/blog/2011/05/introducing-the-delta-sigma-modulator/),
 which looks promising. I'll start reading it myself, now.

Those systems are 1 bit A/D and D/A systems.  This is the technology
used by Sony in their DSD techniques as used in SACDs.

Dave


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Soulkeeper

AFAIU, delta-sigma DACs are used for PCM decoding. It's the most
widespread type of audio DAC, at least according to some of what I've
read.

Unfortunately the article I linked to went into more detail about ADC
than DAC.


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread DaveWr

Soulkeeper;694588 Wrote: 
 AFAIU, delta-sigma DACs are used for PCM decoding. It's the most
 widespread type of audio DAC, at least according to some of what I've
 read.
 
 Unfortunately the article I linked to went into more detail about ADC
 than DAC.
 
 After a quick scan of the DSD article on Wikipedia, I get the idea that
 DSD stores audio in a delta-sigma modulated format, while delta-sigma
 DACs convert a PCM signal to a delta-sigma modulated signal as part of
 the DA conversion.

You are exactly right - virtually all modern DACs are multibit
delta-sigma DACs.  These bring another filter that does affect sound
quality - the interpolation filter.  This is used to manufacture
samples that don't exist in the original sampling. This is usually
where designers claim all their specialities.  For example, Linn in
their DS products don't use the DAC interpolation, but their own design
of interpolation and noise shaping digital filters.  Whether this is
different / better is probably a mute point.  I think it is very much a
low level effect.


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Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0

2012-03-07 Thread sckramer

cfraser;694561 Wrote: 
 I don't think we want to go down that dung-strewn path again, so by all
 means read it all and try what you want and form your own opinion.
 
 My favorite and most-used setup is all the TT3.0 defaults EXCEPT with a
 5k buffer and Logitech priorities. My second favorite, and what I
 believe to be the most accurate setup of those presented here (as best
 as I can test), is the TT3.0 default and everything Klaus, except
 with a 4k buffer for here. [There has never been a decent explanation
 here as to why the ALSA buffer size can make such a diff.]
 
 I listen to internet radio quite a lot, and so far I have *never* heard
 the Logitech priorities screw up the audio, whereas I have a few times
 with stock TT3.0 priorities...so that's another advantage of them for
 me, besides that I like the sound better (less thin). I am always using
 an external DAC, and I am astonished what it does to IR via the SBT
 compared to the IR I get directly via my Denon AVR (this sounds truly
 dreadful, all other radio from it incl. AM/FM/Sirius does too).

I recommend using 2.0 or wait for Klaus to release 4.0 --  also leave
the audio buffer mod disabled (with 2.0)


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread pippin

bluegaspode;694580 Wrote: 
 
 I obviously never cared about how DA converters work, based on the
 paper I now think of thousands of  of sinc-producing circuits which all
 add up to the final waveform. 
 Probably this is not how it works in practice but this a missing link
 for me now to agree that Nyquist theorem and good circuits is all that
 we need.


We have to do away with a huge misunderstanding here, I believe.
This article is NOT about how a DAC should work, which technology is
best and what kind of limitations practical recording and playing
equipment may have.

All of this has nothing, really NOTHING to do with the topic at hand.
It just doesn't matter how good or bad the DAC or the ADC or the
speaker or the microphone is. All of this has absolutely NOTHING to do
with the storage format for the music.


All that Nyquist/Shannon says is: if you have a frequency X, which is
the maximum frequency you are interested in (here: the highest
frequency you could probably ever hear), then if you use a sampling
frequency of 2*X to store your sampled data, then ALL information
contained in the signals below X will be in included in the information
you store. There is NO additional information you get by using a higher
sampling frequency. Nothing. All you get is additional information
about frequencies ABOVE X but not below, the information on the
frequencies below X is already there and it's complete.


What we have to understand here, is that this has nothing to do with
any imperfections in the recording or playback process, these will of
course be there, but if your recording is crap, i won't get better just
because you STORE more of it and at a higher sample rate. And if your
DAC is distorting, then it will not get any better just because you
throw higher frequencies at it - to the contrary, the article implies
it's getting actually WORSE because of effects letting distortions from
the higher frequencies leak into the lower frequency spectrum.
To be clear: this is NOT missing information that was just not
recorded due to the low sample rate, it's DISTORTED information due to
the bad reproduction process in the DAC.


What the article does NOT say is that it doesn't make sense to use
different sample rates or sample sizes for processing. It can make
sense to use something different while processing your data, for
example because of limitations of the technology you use. A good
example is the 24 bit sample size used in the Squeezebox internally.
This makes perfect sense because what the Squeezebox does is it does
digital processing to change the volume. If you stick to 16 bit data,
you would get rounding errors and information losses due to this
processing that you can avoid if you go to 24 bit in processing.
But it does NOT mean, that anything gets better if the data you throw
at it is already 24 bit.

To use a somewhat different analogy: When a bank calculates interest,
it will use 4 additional digits behind the cent (so 1$ is 1.00  for
them). Why? Because if you get, for example, 1% of interest for your
dollar per year and that interest is paid monthly, the monthly interest
you get would amount to 0.00 0833 ct. If you round that, you just get 0
so you would never get any interest which would be plain wrong because
to calculate things right, they will have to pay you 1 ct per year.
HOWEVER, they will never actually PAY you 0.0833 ct because there is no
such thing. and your Dollar doesn't get any different just because you
write it as 1.00  $, it's still exactly the same thing as 1$ and
actually everything behind the last cent digit has no meaning at all
(or it would already be a rounding error).


Likewise, nobody says that there will be no way to invent some fancy
technology that does a more accurate recording of the analog audio
signal at 2 MHz sample rate and this can be superior to a 16 bit 44.1
kHz microphone.
HOWEVER: If you then take the digital output of that hypothetical
processor and down-convert it to 44.1 kHz samplerate audio, then for
all frequencies below 22.05 kHz, there will be NO, not even the
slightest loss of information.
So it doesn't make sense to STORE and TRANSMIT the data at higher
frequencies.

There are a few good arguments for 48 kHz, most notably that since it's
common to use 96kHz or 192 kHz equipment in processing (remember: it can
STILL make a lot of sense to do PROCESSING at higher frequencies,
especially in the digital domain) you get pretty much simplified
up/downsampling logic.
I can't argue about 24 vs. 16 bit, that does indeed depend on the
actual dynamic range you can record and reproduce and I don't know
where technology is here, purely from an information  theory
standpoint, 24 bit word size DOES contain more information than 16 bit
word size, that's different from the sample rate thing.


Now there is a third thing, and that's the trust your ears thing.
1. Yes, you should, because in the end it's all that matters
2. Normal people do that but as of my 

Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0

2012-03-07 Thread lake_eleven

sckramer;694591 Wrote: 
 .also --unsolder the toslink!-- this made a very big difference
 relative to everything in this scope of things

can you elaborate how you did this. May be a pic


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Re: [SlimDevices: Audiophiles] 192 downsampling when decoding at server side

2012-03-07 Thread pippin

Cache miss rates would be the same or lower with flac. In the end you
are processing the same amount of data and it's a lot and it's size is
determined by the DECODED file size, the ENCODED file size is lower
with flac and you actually have to process all of it, even if wav is
just a raw pcm stream.

So if your rationale holds, wav must be worse than flac.


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Re: [SlimDevices: Audiophiles] soundcheck's Touch Toolbox 3.0

2012-03-07 Thread eduardoo

cfraser;694561 Wrote: 
 I don't think we want to go down that dung-strewn path again, so by all
 means read it all and try what you want and form your own opinion.
 
 My favorite and most-used setup is all the TT3.0 defaults EXCEPT with a
 5k buffer and Logitech priorities. My second favorite, and what I
 believe to be the most accurate setup of those presented here (as best
 as I can test), is the TT3.0 default and everything Klaus, except
 with a 4k buffer for here. [There has never been a decent explanation
 here as to why the ALSA buffer size can make such a diff.]
 
 I listen to internet radio quite a lot, and so far I have *never* heard
 the Logitech priorities screw up the audio, whereas I have a few times
 with stock TT3.0 priorities...so that's another advantage of them for
 me, besides that I like the sound better (less thin). I am always using
 an external DAC, and I am astonished what it does to IR via the SBT
 compared to the IR I get directly via my Denon AVR (this sounds truly
 dreadful, all other radio from it incl. AM/FM/Sirius does too).

Thanks, guys... and please allow me a very dumb question.

Can you give me some brief instructions or point me to the right thread
as to how this priority tweak is done?  I was lost amidst the hundreds
of threads about it (mostly about tweaks of the values) and can't
locate where the instructions began (i.e. commands or modifying the
files)

Thanks.


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Re: [SlimDevices: Audiophiles] 192 downsampling when decoding at server side

2012-03-07 Thread Mnyb

...and connected to an outboard DAC of good quality, not much left of
any plausible mechanism for this to happen all other mechanism not
related to anything electrical or audio are still overwhelmingly more
plausible.

Leaves electrical noise , not even jitter ( as almost everything any of
us own have lower jitter levels than is commonly agreed to be humanly
audible by a factor 100 or more ) tiny differences in jitter in a
product that does not have fantastic but ok values science fiction if
anyone detects those ? And noise levels why would they differ
significantly ?


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread Mnyb

Thanks pippin .

It is not about specific techniques let's assume that it is sota the
best 192k stuff is used during mastering and playback 

I to so tries to make this point if we just leave bithdept depth aside
for a while .

Ponder for a moment that no analog signal recorded had any content
above  24 kHz

Then a 48k signal and a 192k signal would contain exactly the same
signal it would reconstruct to exactly the same wave form you would not
even forensically be able to tell them apart .

What is said is that a digital signal has the necessary data to
completely describe any signal at fs/2.

this includes the time domain there is a fallacy to believe that the
content between samples are lost that there is an inaccuracy of +_ 1/fs
this is false.

So arguments for higher fs necessary implies that it should be any
merit of playing back ultrasonics .


There may be 1000's of reason for a studio or DAW software to operate
at any level .


As pippin and many others say to get that high quality 44.1 or 48
signal any kind of higher fs recording and processing can be involved
but that's not really the point , on the contrary employing such
methods will probably yield a perfect such signal for all intents and
purposes and actually prove the point that fs at 44.1 encodes all info
in the frequency domain we can hear.
But all beside the point


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread bluegaspode

pippin;694598 Wrote: 
 
 All of this has nothing, really NOTHING to do with the topic at hand.
 It just doesn't matter how good or bad the DAC or the ADC or the
 speaker or the microphone is. All of this has absolutely NOTHING to do
 with the storage format for the music.
 

You are missing my point.
It's fundamental that I am asking about how DACs work, because
otherwise the Nyquist theorem is of no value in our practical world.

To make my argument clear lets go back to circles: I think there is
some theory that claims that all data you need to draw a perfect circle
is to have it's center point and radius (that would be 2 samples).
Without asking how the circle will be drawn later, it is WRONG to just
conclude that it doesn't help to record more samples (like 100 points
on the perimeter of the circle).

Let's say the hardware that is used to draw the circle, is only able to
work with a rough estimation of PI (2 instead of 3.14...). The circle
that this device would be drawing would be a very bad approximation of
the original circle and every graphophile lover of circles would
complain. Now in such a scenario you will draw BETTER circles with 100
samples of points on the perimeter and even better circles with 1000
samples of points on the perimeter.

So as long as a DAC (or the reproduction filter) isn't working with
good enough sinc waveforms the Nyquist theory remains just a nice
theory.
Maybe you all know so much more about DACs than me, that you are not
questioning the sinc functions in your DAC anymore.

But I am :) ... and maybe also many audiophiles that read that article,
so don't stop at claiming 'Nyquist is enough, there is no more to talk
about'.
If you want to prove that in nowadays world higher samplerates don't
help you cannot stop at Nyquist but you need to go all the way to the
loudspeaker.

 
 You are exactly right - virtually all modern DACs are multibit
 delta-sigma DACs. These bring another filter that does affect sound
 quality - the interpolation filter. This is used to manufacture samples
 that don't exist in the original sampling.

  Uhh ohhh. And why is this better than just using on of the original
  samples if I had the double sample rate? Or are we in DAC snake oil
  territory already when it comes to creating interpolation filters that
  invent some new samples?


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Re: [SlimDevices: Audiophiles] 192kHz considered harmful

2012-03-07 Thread pippin

bluegaspode;694626 Wrote: 
 
 Let's say the hardware that is used to draw the circle, is only able to
 work with a rough estimation of PI (2 instead of 3.14...). The circle
 that this device would be drawing would be a very bad approximation of
 the original circle and every graphophile lover of circles would
 complain. Now in such a scenario you will draw BETTER circles with 100
 samples of points on the perimeter and even better circles with 1000
 samples of points on the perimeter.
 
Sorry, but that's nonsense.

We are talking about digital information processing here.
If your hardware that does the drawing of the circle only can use rough
approximations, you need to convert your perfect circle (center plus
radius) to whatever strange approximation your machine needs to draw a
good circle.
All you need for that is a computer that understands pi and how to do a
perfect circle from the center point and the radius (and of course your
machine's limitation).

The information (center point and radius) is still complete. There is
nothing else you will ever need to describe it.

Even worse. What you are postulating assumes that a WORSE
representation of the circle could actually lead to better results
through a process that involves that the CREATOR of the limited
approximation actually has a better understanding of the limitations of
your reproduction machine than that machine itself (because it's
creating a worse approximation of the circle to assist the latter's
reproduction process).

Actually a pretty accurate description of a lot of stuff being done in
audio but still by no means sensible.

 
 So as long as a DAC (or the reproduction filter) isn't working with
 good enough sinc waveforms the Nyquist theory remains just a nice
 theory.
 
No, that, too, is wrong. The theory is correct. Just because your
filter is bad doesn't mean that any different data will serve it
better. Worse. Now you need to do an end-to-end optimization just to
avoid using perfect information. Doesn't make sense and will certainly
not work.

 
 Maybe you all know so much more about DACs than me, that you are not
 questioning the sinc functions in your DAC anymore.
 
No, we have understood Shannon and so we know it's two completely
separate problems.

Again (that's what I wrote above): if your DAC needs something else
than 44.1/16, it can CREATE IT FROM THAT. There is no issue with
upsampling 44.1/16 to 12.3GHz@486 bits if that's what your DAC needs.
There will not be any information loss. But you gain exactly nothing
from transmitting and storing data in that format.


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