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  #121   Report Post  
Kurt Albershardt
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Roger W. Norman wrote:
"Kurt Albershardt" wrote in message
...

Also: due to the way computers store information, 18 bit PCM takes the
same amount of disk space as does 24 bit PCM (and often the same amount
as does 32 bit floating point PCM data.)




Well, my point wasn't that using 24 bit has anything wrong with it, nor,
with the cost of storage today, is there a problem with using 24 bit for any
project. In fact, the processing aspects of digital certainly require that
you use 24 bit, and save it as 32 bit floating point if you can. I was just
trying to make it clear that there are technical aspects of 24 bit
converters that wouldn't necessarily make them better than 16 bit in given
circumstances.



And I was merely pointing out one of the reasons we have so many 24-bit
converters on the market when their real performance is more like 19-22
bits.



  #122   Report Post  
Carey Carlan
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Justin Ulysses Morse wrote in
m:

Carey Carlan wrote:
Carey wrote:
Is your argument that this higher precision is inaudible?


and your answer is "Yes". That's all I needed to know.


What? Where do you get that? I've written a novel or two explaining
why it's theoretically audible, even though in practice it will be
covered up by noise most of the time. To say my answer is "yes" is to
miss the point which I have over-articulated here.


I didn't miss the point. I read your mini-novel. To summarize down to one
phrase, your thesis is that I have very little chance of hearing it. That
translates to 'no' in my book.
  #123   Report Post  
Justin Ulysses Morse
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Carey Carlan wrote:

I didn't miss the point. I read your mini-novel. To summarize down to one
phrase, your thesis is that I have very little chance of hearing it. That
translates to 'no' in my book.



Okay. Upon further consideration, "it's audible" isn't really what I
was getting at anyway. Not that I would know. My intuitive
understanding of the basic theory surpassed my listening experience.
Not that I know all that much about the theory either. Just the
basics.

ulysses
  #124   Report Post  
Jay - atldigi
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

In article , "Arny Krueger"
wrote:


makes pretty good use of it. However a converter with a relatively coarse
step size is still going to be noisier than one that has a smaller step
size.


This is the simplest way to state the advantage of greater bit depths -
less noisy. Dither is essential in a quantizer in audio and can be
thought of as levelling the playing field (to borrow another poster's
phrase) above the nosie floor of the system. However, to pick nits, it
is true that the error is not completely "cured" (again, to borrow a
poster's term). In practical terms, however, it's just turned into a
benign noise floor. The resolution of the audio in the available dynamic
range in terms of voltages being reproduced by the system is
continuously infinitely variable and not stepped with distortion and
aliasing.

So yes, the posters who have been asking the questions appear to
basically have a grasp on it now, though there are always little caveats
and distinctions. However, don't let the little technical nit picking
confuse the practical understanding you have gained. Digest the
conceptual big picture for a moment, and then you can suffer through the
distinction of why the extra steps and their gained precision only lower
the noise in a practical sense, even though the numbers themselves are
certainly more precise.

--
Jay Frigoletto
Mastersuite
Los Angeles
promastering.com
  #125   Report Post  
Jay - atldigi
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

In article , "Tommi"
wrote:

If it is so, then it..umm..is so. So this is what happens in the real
world, but in theory
24 bits represents the original signal more accurately than 16 bits.


"24 bits is more accurate" is actually true as a general statement, but
what that means specifically, and how that manifests itself in an audio
system has been the source of confusion. I hope that's the distinction
that has gotten accross in this discussion.

Thanks to everybody for participating.

--
Jay Frigoletto
Mastersuite
Los Angeles
promastering.com


  #126   Report Post  
EggHd
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Thanks to everybody for participating.

I didn't participate by posting but I enjoyed reading the threads on this
subject. It's good to keep learning.





---------------------------------------
"I know enough to know I don't know enough"
  #127   Report Post  
White Swan
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

I would also like to express my appreciation. I can't guarantee that I
understand everything that was posted here at this point, but I plan
on rereading it at my leisure.

My point is that hanging here is like getting a graduate level course
in engineering for free. I can't tell you how much I appreciate
everyone who takes the time to eloquently share their knowledge.
  #128   Report Post  
Garthrr
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

In article , Chris Hornbeck
writes:

The
process of quantizing has fundamental errors estimating the smallest
bit, errors which track the signal itself.


Is this in any way analogous to how a compressor with attack and release set
fast enough can track the waveform of a bass signal and cause audible
"distortion"?

Garth~


"I think the fact that music can come up a wire is a miracle."
Ed Cherney
  #129   Report Post  
Garthrr
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

In article , Jay -
atldigi writes:

"24 bits is more accurate" is actually true as a general statement, but
what that means specifically, and how that manifests itself in an audio
system has been the source of confusion. I hope that's the distinction
that has gotten accross in this discussion.


It has. Thank you guys for being patient and probably repeating yourselves ad
nauseum. Its sinking in.

Garth~


"I think the fact that music can come up a wire is a miracle."
Ed Cherney
  #130   Report Post  
Garthrr
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

In article , "Arny Krueger"
writes:

Dither clearly doesn't correct quantization error. Dither does change
quantization error into something that is sonically more benign.


Allow me to hazard a guess as to how dither works: First of all, do I
understand correctly that dither is used whether or not a file is being altered
in some way or not? In other words, even in a first generation digital
recording dither is already in use right? If the answer to that question is no
then I am really lost.

Anyway, if we have two adjacent samples of different voltages which are both
being rounded, does the dither sort of randomly insert other near voltages
around the two samples and in doing so smooth the differences between the two
samples' voltages? e.g: if we had one sample with a value of 1.736400 volts and
following sample with 1.736500 volts might the dither insert a voltage of
1.736450 (a voltage somewhere between the other two)? I realize those are not
16 bit numbers but I dont think it matters for this question.
Of course it has now occurred to me that I dont know where this dither signal
would be placed since there is no intermediate sample in which to put it.
Eeewww buoy!

Garth~


"I think the fact that music can come up a wire is a miracle."
Ed Cherney


  #131   Report Post  
Jim Gilliland
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Garthrr wrote:
In article , "Arny Krueger"
writes:

Dither clearly doesn't correct quantization error. Dither does change
quantization error into something that is sonically more benign.


Allow me to hazard a guess as to how dither works.....


This topic has come up quite a few times here over the past few years -
a Google search will provide you with a wealth of material. For
example, there was a very informative thread on this topic that ran
through the first half of December two years ago.

Here's one article that I find to be very helpful in explaining exactly
how dither works:


On Dither
by Mithat Konar

Of the all the things that are commonly misunderstood about digital
audio, one of the most common is dither. Dither is the deliberate
introduction of a random signal (i.e., noise) prior to quantization in
an analog-to-digital converter or prior to reducing the wordlength
(e.g., 24 to 16 bits) in a DSP algorithm. Dither comes in various forms
and in varying levels levels of complexity, but all forms have in common
the introduction of a random signal into the signal chain.

It might seem like a stupid idea to add noise to a system, but when
dither is done correctly, it completely eliminates the low-level
distortion found in quantized systems while increasing the resolution of
the system to a level well below the noise floor. (In theory, the
resolution becomes infinite). In these respects, dither makes digital
systems behave exactly like analog ones.

The primary consequence of adding dither is a slight decrease in
signal-to-noise ratio -- typically a few decibels. However, advanced
schemes add the noise in such a way that its audibility is minimized.
The end result with 16-bit, 44.1 kHz systems is a digital signal with
all the benefits of dither (elimination low-level distortion and an
increase in resolution) but virtually no perceived increase in noise.

It is often stated that dither simply masks, not eliminates, the
low-level errors found in digital systems, that the errors are still
present, but they are simply buried in hiss. This is very much not the
case. The following thought experiments may help to put these issues
into intuitive perspective.

(1) Imagine you have a bipolar, DC coupled analog-to-digital converter
with a least significant bit (LSB) threshold of 1 volt. (Just go with it
-- this is a thought experiment!) Furthermore, assume the ADC stairstep
function is aligned such that it is symmetric about zero. In other
words, an ADC input over the 1-volt interval [-0.5, 0.5] produces an ADC
output of 0, an input over the range [0.5, 1.5] produces an output of 1,
and so on.

(2) Without applying any dither, apply a 0.25 volt signal to the input
of the ADC. The output of the ADC will be a string of zeros. In fact,
any signal between -0.5 and 0.5 volt will result in an ADC output of
zero. Any information below the LSB threshold is completely lost.

(3) Remove the 0.25 volt signal and apply dither to the input of the ADC
in the form of a completely random signal (i.e., noise) centered around
0 volts and large enough to just barely twiddle the LSBs of the ADC. The
output of the ADC will be a stream of very small random numbers.
However, the AVERAGE of all these values will be zero.

(4) Now let's apply our 0.25 volt signal again (with the dither on). The
output stream will again look like a stream of very small random
numbers, but guess what? The AVERAGE of all those numbers will now
be...you guessed it, 0.25. We have thus retained the information that
was previously lost (even though it's buried in "noise"). In other
words, our system's resolution has improved. The conversion is still
essentially random, but the presence of the 0.25 volt signal biases the
randomness. The mathematician would say that the characterization of the
system with dither is transformed from a completely deterministic one to
a statistical one.

(5) With the dither on, we can now change the input signal over a
continuous range and the average of the ADC output will track it
perfectly. An input signal of 0.373476 volts will have an average ADC
output of 0.373476. The same will hold true of inputs going over the +/-
0.5 volt LSB threshold: e.g., an input of 3.22278 will have an average
ADC output of 3.22278. So not only has the dither enhanced the
resolution of the system, but it has also eliminated quantization effects!

(6) The results in (4) and (5) will not happen by adding noise after the
A/D conversion. Go back to our first experiment and add the random
signal to the output of the ADC. As long as the input signal is between
-0.5 and 0.5 volt, the ADC's average output will still be zero. Between
0.5 and 1.5 volts the average will jump to 1. There is no resolution
enhancement and the quantization effects remain.

So, you can see that dither's resolution enhancement and error
elimination are truly physical/mathematical in nature and not a masking
trick. You should also keep in mind that human beings are able to hear
signals in the presence of noise of greater energy than the signal,
i.e., with negative signal-to-noise ratios. Therefore, even though a
given signal is below the noise-floor of the system, and therefore you
might think it irrelevant, it in fact may not be. Depending on what else
is going on around that signal, it can be audible at several decibels
below the noise-floor. Therefore, both benefits of dither -- eliminating
low-level quantization errors and increasing the system's resolution --
are truly beneficial, perceivable effects.

Source URL: http://www.birotechnology.com/articles/dither.html

  #134   Report Post  
Roger W. Norman
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Arny Krueger" wrote in message
...

I just didn't want any lurkers to think that only 24 bit converters
are a solution to any recording problems they might have. Again,
it's the use of the tools rather than the tools themselves.


I think that this is one of the most important messages that a group like
this has to present to newbies.




Maybe we should preface every post with it's not the fool's tool, it's the
tool's fool.
g

--


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




"Roger W. Norman" wrote in message

"Kurt Albershardt" wrote in message
...
Also: due to the way computers store information, 18 bit PCM takes
the same amount of disk space as does 24 bit PCM (and often the same
amount as does 32 bit floating point PCM data.)


Well, my point wasn't that using 24 bit has anything wrong with it,
nor, with the cost of storage today, is there a problem with using 24
bit for any project. In fact, the processing aspects of digital
certainly require that you use 24 bit, and save it as 32 bit floating
point if you can. I was just trying to make it clear that there are
technical aspects of 24 bit converters that wouldn't necessarily make
them better than 16 bit in given circumstances. Pretty much the
concept of choosing one's tools to fit the job.


Agreed. One thing to realize is that most if not all of the major chip
makers have unflinchingly released 24/96 and 24/192 converters with worse
measured dynamic range than some of their earlier 16/44 converters.

There's
a part of the market that is all about numbers.

When I go out and do
location recordings, I still use Tascam DA38s and I've not had one
client that wasn't happy with the recordings. They might not have
liked the performances, but that's a different story. My last
submission to A Fifth of RAP was recorded onto DA38, and it sounds
pretty good to me (obviously biased) with plenty of dynamics. The
room lacked something and 24 bits wouldn't necessarily have bought me
anything more. The same with Scott Dorsey's extremely dynamic
recording. And again, Tonebarge absolute knocks one out of the park
with his Mackie/Adat combination (although he uses different pres for
tracking).







  #136   Report Post  
Chris Hornbeck
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain


On Dither
by Mithat Konar


(6) The results in (4) and (5) will not happen by adding noise after the
A/D conversion. Go back to our first experiment and add the random
signal to the output of the ADC. As long as the input signal is between
-0.5 and 0.5 volt, the ADC's average output will still be zero. Between
0.5 and 1.5 volts the average will jump to 1. There is no resolution
enhancement and the quantization effects remain.


I'm probably just reading this wrong, but on it's face, this is
incorrect. Digital dither is used all the time.

Chris Hornbeck
"That is my Theory, and what it is too."
Anne Elk
  #137   Report Post  
dan lavry
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

(Garthrr) wrote in message
Dither clearly doesn't correct quantization error. Dither does change
quantization error into something that is sonically more benign.


Of course it has now occurred to me that I dont know where this dither signal
would be placed since there is no intermediate sample in which to put it.
Eeewww buoy!

Garth~


Lip****z and Vanderkooey from Toranto university have a lot of
articals on the dither subject. As academics they use a lot of math,
which may be difficult reading for some. I wrote an artical for folks
that do not want the math but wish to get some common sense
understanding. Included are plots, both time and frequncy domain. It
is under the suport section of
www.lavryengineering.com and the
artical name is "Do you need 20 Bits". This was written in 1997.
Dither was and still is importent for say 16 bits work.

Of course, if 24 bit format ever becomes a serious release standard,
dither becomes that "gold comb to a bold head"...

Dither is cool, but it is a bit of a tradeoff between dynamic range
and distortions. Noise shaped dither provides some controll over where
the noise gets added (obviously away from the hearing sensitive region
of say "mid KHZ"). This is is much more advanced. If you want to see
that, go to the same place (suport) and check Acostic Bit Correction.

Again, if 24 bits becomes a standard, it all becomes just history...

BR
Dan Lavry
  #138   Report Post  
Jim Gilliland
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Chris Hornbeck wrote:
On Dither
by Mithat Konar


(6) The results in (4) and (5) will not happen by adding noise after the
A/D conversion. Go back to our first experiment and add the random
signal to the output of the ADC. As long as the input signal is between
-0.5 and 0.5 volt, the ADC's average output will still be zero. Between
0.5 and 1.5 volts the average will jump to 1. There is no resolution
enhancement and the quantization effects remain.


I'm probably just reading this wrong, but on it's face, this is
incorrect. Digital dither is used all the time.


Dither is used when reducing the resolution of the signal. It is
effective only when it is applied before the resolution is reduced, not
after. His example uses an imaginary ADC as its basis, but it would be
equally valid if he were talking about reducing word length in the
digital domain.

So, yes, I think you _are_ reading this wrong. His only point is that
dither is NOT effective if it is applied after the resolution has been
reduced (ie, after truncation).

The process of digitizing an analog signal is a process in which
resolution is reduced. So dither is required there, just as it is when
reducing the bit depth of a digital signal. The basic principle is
exactly the same in either case.

  #139   Report Post  
Justin Ulysses Morse
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Mike Rivers wrote:

I haven't given this any thought, so I'll just throw out the question
to ponder. Is it possible that time resolution between components of a
complex wave could be better with a higher sampler rate? Could the
phase relationship between the fundamental and, say, third harmonic of
a distorted guitar, be more accurately preserved at a higher sample
rate?



If my understanding is correct, and it's based mostly on what I read
here from I think Arny (but it makes perfect sense to me), then the
answer to your question is that yes, a higher sample rate (or a higher
bit rate for that matter) would improve that time resolution; but that
the resolution is already WAY higher than necessary. People assume
that the time-domain accuracy is one sample period; but it's really one
sample period divided by the quantization range. So the time domain
accuracy for CD audio would be 1/(44100*2^16) or 346 picoseconds.
That's about a decimal place or two off of what I vaguely remember Arny
saying, so I've probably messed it up. But you get the idea? I
probably shouldn't be paraphrasing from memory, so go back and read
Arny's post from yesterday in one of these threads.


ulysses
  #140   Report Post  
Justin Ulysses Morse
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Chris Hornbeck wrote:

On Dither
by Mithat Konar


(6) The results in (4) and (5) will not happen by adding noise after the
A/D conversion. Go back to our first experiment and add the random
signal to the output of the ADC. As long as the input signal is between
-0.5 and 0.5 volt, the ADC's average output will still be zero. Between
0.5 and 1.5 volts the average will jump to 1. There is no resolution
enhancement and the quantization effects remain.


I'm probably just reading this wrong, but on it's face, this is
incorrect. Digital dither is used all the time.


I know Mithat, and he knows what he's talking about. If he didn't, he
wouldn't have written it. He's not saying that dither can't be applied
in the digital domain. He's saying it needs to be applied prior to the
quantization (or re-quantization) that's being dithered.

In an ADC, this could occur in the analog domain, or in the case of an
oversampling converter, it could be done digitally before the signal is
quantized to 16 or however many bits.

In the case of truncating 24 bit data into 16 bits, he's saying you
have to add the dither BEFORE you lop off those bottom 8 bits. The
dither and the LSBs have to be summed before truncation.

ulysses


  #141   Report Post  
Chris Hornbeck
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

On Fri, 21 Nov 2003 13:42:15 -0600, Justin Ulysses Morse
wrote:

I know Mithat, and he knows what he's talking about. If he didn't, he
wouldn't have written it. He's not saying that dither can't be applied
in the digital domain. He's saying it needs to be applied prior to the
quantization (or re-quantization) that's being dithered.

In an ADC, this could occur in the analog domain, or in the case of an
oversampling converter, it could be done digitally before the signal is
quantized to 16 or however many bits.

In the case of truncating 24 bit data into 16 bits, he's saying you
have to add the dither BEFORE you lop off those bottom 8 bits. The
dither and the LSBs have to be summed before truncation.


Hi Ulysses,
Thanks for your comments. I really don't know how to resolve this.
What you say makes perfect sense, but Watkinson says otherwise.
Maybe someone else could help out?.....


Chris Hornbeck
"That is my Theory, and what it is too."
Anne Elk
  #142   Report Post  
Jim Gilliland
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Chris Hornbeck wrote:
On Fri, 21 Nov 2003 13:42:15 -0600, Justin Ulysses Morse
wrote:


I know Mithat, and he knows what he's talking about. If he didn't, he
wouldn't have written it. He's not saying that dither can't be applied
in the digital domain. He's saying it needs to be applied prior to the
quantization (or re-quantization) that's being dithered.

In an ADC, this could occur in the analog domain, or in the case of an
oversampling converter, it could be done digitally before the signal is
quantized to 16 or however many bits.

In the case of truncating 24 bit data into 16 bits, he's saying you
have to add the dither BEFORE you lop off those bottom 8 bits. The
dither and the LSBs have to be summed before truncation.


Hi Ulysses,
Thanks for your comments. I really don't know how to resolve this.
What you say makes perfect sense, but Watkinson says otherwise.
Maybe someone else could help out?.....


OK, now _I'm_ the one who's confused. I thought that what Ulysses wrote
above was pretty well established and non-controversial, and furthermore
that you (Chris) were already in agreement with it (based upon your post
back to me just a few minutes earlier). So what exactly is
contradictory here? What does Watkinson say that seems to counter the
above?

  #143   Report Post  
Chris Hornbeck
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

On Fri, 21 Nov 2003 20:40:02 GMT, Jim Gilliland
wrote:

OK, now _I'm_ the one who's confused. I thought that what Ulysses wrote
above was pretty well established and non-controversial, and furthermore
that you (Chris) were already in agreement with it (based upon your post
back to me just a few minutes earlier). So what exactly is
contradictory here? What does Watkinson say that seems to counter the
above?


Watkinson's The Art of Digital Video, 2nd ed. pg. 143-144 (sorry,
don't have The Art of Digital Audio handy):
" the principles of analog and digital dither are identical; the
processes simply take place in different domains using two's
complement numbers which are rounded or voltages which are quantized
as appropriate. In fact quantization of an analog dithered
waveform is identical to the hypothetical case of rounding after
bipolar digital dither where the number of bits to be removed is
infinite, and remains identical for practical purposes when as few
as 8 bits are to be removed. Analog dither may actually be generated
from bipolar digital dither (which is no more than random numbers
with certain properties) using a DAC."

Hopefully, I'll be shown to have misunderstood the above, and can
go back to my comfortable but fuzzy understanding of things before
this blasted thread started. Ignorance was bliss, and I'm too old
to study.

Thanks!

Chris Hornbeck
"That is my Theory, and what it is too."
Anne Elk
  #144   Report Post  
Jay - atldigi
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

In article , Justin
Ulysses Morse wrote:

Mike Rivers wrote:

I haven't given this any thought, so I'll just throw out the question
to ponder. Is it possible that time resolution between components of a
complex wave could be better with a higher sampler rate? Could the
phase relationship between the fundamental and, say, third harmonic of
a distorted guitar, be more accurately preserved at a higher sample
rate?



If my understanding is correct, and it's based mostly on what I read
here from I think Arny (but it makes perfect sense to me), then the
answer to your question is that yes, a higher sample rate (or a higher
bit rate for that matter) would improve that time resolution; but that
the resolution is already WAY higher than necessary. People assume
that the time-domain accuracy is one sample period; but it's really one
sample period divided by the quantization range. So the time domain
accuracy for CD audio would be 1/(44100*2^16) or 346 picoseconds.
That's about a decimal place or two off of what I vaguely remember Arny
saying, so I've probably messed it up. But you get the idea? I
probably shouldn't be paraphrasing from memory, so go back and read
Arny's post from yesterday in one of these threads.


ulysses


And you wouldn't believe it, but dither actually affects the time domain
as well. No kidding. Little known, but true - it's similarly helpful in
the time domain as it is for amplitude. Most of the science geeks say
that time resolution also becomes essentially infinite with proper
dither. So Mike's concern is probably something that doesn't need to be
worried about as some say the limit doesn't exist in a well designed
system, and even the competing camp says it's so small as to not make a
difference. In this case it doesn't matter who's right since it just
plain doesn't seem to matter. There's a few stragglers left over that
say it's possible that imaging can be affected when you have
multichannel in play, but even the significance of this deserves some
skepticism as it is far from proven or agreed upon, and the "infinite
time resolution in a properly implimented system" guys actually seem to
have the science to back it up. My vote is with them.

--
Jay Frigoletto
Mastersuite
Los Angeles
promastering.com
  #145   Report Post  
Jay - atldigi
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

In article , Justin
Ulysses Morse wrote:

He's not saying that dither can't be applied
in the digital domain. He's saying it needs to be applied prior to the
quantization (or re-quantization) that's being dithered.

In an ADC, this could occur in the analog domain, or in the case of an
oversampling converter, it could be done digitally before the signal is
quantized to 16 or however many bits.

In the case of truncating 24 bit data into 16 bits, he's saying you
have to add the dither BEFORE you lop off those bottom 8 bits. The
dither and the LSBs have to be summed before truncation.



Yes. If you do it after, the distortion is there, and possibly
(probably) aliasing from it, and there's no way to get rid of it once
it's there. However the designer wants to do it for the aplication at
hand (digital or analog), it needs to be done before the quantization or
requantization.


In article , Chris Hornbeck
wrote:
Hi Ulysses,
Thanks for your comments. I really don't know how to resolve this.
What you say makes perfect sense, but Watkinson says otherwise.
Maybe someone else could help out?.....



I think this has already been reolveed in other posts in the thread, so
I'll just add a small Watkinson quote:

"The introduction of dither PRIOR to a conventional quantizer inevitibly
causes a slight reduction in the signal-to-noise ratio attainable, but
this reduction is a small price to pay for the elimination of
non-linearities."

So obviously Watkinson does say it must happen before quantization. Your
Watkinson quote was talking about something a little different.

--
Jay Frigoletto
Mastersuite
Los Angeles
promastering.com


  #146   Report Post  
Jim Gilliland
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Chris Hornbeck wrote:
On Fri, 21 Nov 2003 20:40:02 GMT, Jim Gilliland
wrote:

OK, now _I'm_ the one who's confused. I thought that what Ulysses wrote
above was pretty well established and non-controversial, and furthermore
that you (Chris) were already in agreement with it (based upon your post
back to me just a few minutes earlier). So what exactly is
contradictory here? What does Watkinson say that seems to counter the
above?


Watkinson's The Art of Digital Video, 2nd ed. pg. 143-144 (sorry,
don't have The Art of Digital Audio handy):
" the principles of analog and digital dither are identical; the
processes simply take place in different domains using two's
complement numbers which are rounded or voltages which are quantized
as appropriate. In fact quantization of an analog dithered
waveform is identical to the hypothetical case of rounding after
bipolar digital dither where the number of bits to be removed is
infinite, and remains identical for practical purposes when as few
as 8 bits are to be removed. Analog dither may actually be generated
from bipolar digital dither (which is no more than random numbers
with certain properties) using a DAC."

Hopefully, I'll be shown to have misunderstood the above, and can
go back to my comfortable but fuzzy understanding of things before
this blasted thread started. Ignorance was bliss, and I'm too old
to study.


OK, all of your quote makes sense to me. What I don't see, though, is
how anything in Ulysses's quote contradicts it. Here is what you quoted
from Ulysses earlier - can you help me see where the conflict is? I
apologize if I'm being dense or stupid here.

I know Mithat, and he knows what he's talking about. If he didn't, he
wouldn't have written it. He's not saying that dither can't be applied
in the digital domain. He's saying it needs to be applied prior to the
quantization (or re-quantization) that's being dithered.
In an ADC, this could occur in the analog domain, or in the case of an
oversampling converter, it could be done digitally before the signal is
quantized to 16 or however many bits.

In the case of truncating 24 bit data into 16 bits, he's saying you
have to add the dither BEFORE you lop off those bottom 8 bits. The
dither and the LSBs have to be summed before truncation.


  #147   Report Post  
S O'Neill
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

He's talking about the input (A/D), basically saying that dither must be
applied to the input of (ie, before) a conversion. You're thinking of
bit depth reduction or output (D/A) dithering, where the same is true so
it's done in the digital domain; before the conversion. Similarly,
adding analog noise after a D/A conversion would not have the desired
effect.


Chris Hornbeck wrote:
On Dither
by Mithat Konar



(6) The results in (4) and (5) will not happen by adding noise after the
A/D conversion. Go back to our first experiment and add the random
signal to the output of the ADC. As long as the input signal is between
-0.5 and 0.5 volt, the ADC's average output will still be zero. Between
0.5 and 1.5 volts the average will jump to 1. There is no resolution
enhancement and the quantization effects remain.



I'm probably just reading this wrong, but on it's face, this is
incorrect. Digital dither is used all the time.

Chris Hornbeck
"That is my Theory, and what it is too."
Anne Elk


  #148   Report Post  
Jim Gilliland
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Jay - atldigi wrote:

There's a few stragglers left over that
say it's possible that imaging can be affected when you have
multichannel in play....


I bet they read Stereophile. g

  #149   Report Post  
Chris Hornbeck
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

On Fri, 21 Nov 2003 22:26:03 GMT, Jim Gilliland
wrote:

OK, all of your quote makes sense to me. What I don't see, though, is
how anything in Ulysses's quote contradicts it. Here is what you quoted
from Ulysses earlier - can you help me see where the conflict is? I
apologize if I'm being dense or stupid here.


It's certainly not you being dense here. I suspect that I'm reading
the word "identical", twice, and drawing unwarranted conclusions
that it means interchangable.

Since everybody but me is marching out of step, my conclusion that
an ADC could be dithered digitally must be wrong. Thanks for your
help; also Ulysses, Jay Frigoletto and S O'Neill.

Great thread, ladies and germs,

Chris Hornbeck
"That is my Theory, and what it is too."
Anne Elk
  #150   Report Post  
S O'Neill
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Chris Hornbeck wrote:

Watkinson's The Art of Digital Video, 2nd ed. pg. 143-144 (sorry,
don't have The Art of Digital Audio handy): " the principles of
analog and digital dither are identical; the processes simply take
place in different domains using two's complement numbers which are
rounded or voltages which are quantized as appropriate. In fact
quantization of an analog dithered waveform is identical to the




hypothetical


Here's the problem. Assuming an infinite number of bits to be removed,
you're right; dither is necessary to truncate to N bits, like 24 or 16.
But they're saying that Analog is equivalent to infinite bit depth,
which is clearly impossible; an A/D convertor gets you some finite
number of bits. After it's done that, dither below its resolution buys
nothing. It just stays below that resolution, and rather than adding
information, it only adds noise.




case of rounding after bipolar digital dither where the number of
bits to be removed is infinite, and remains identical for practical
purposes when as few as 8 bits are to be removed. Analog dither may
actually be generated from bipolar digital dither (which is no more
than random numbers with certain properties) using a DAC."

Hopefully, I'll be shown to have misunderstood the above, and can go
back to my comfortable but fuzzy understanding of things before this
blasted thread started. Ignorance was bliss, and I'm too old to
study.

Thanks!

Chris Hornbeck "That is my Theory, and what it is too." Anne Elk




  #151   Report Post  
Garthrr
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Jim Gilliland writes:

Here's one article that I find to be very helpful in explaining exactly
how dither works:


Thanks Jim!
Garth~


"I think the fact that music can come up a wire is a miracle."
Ed Cherney
  #153   Report Post  
dan lavry
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

In general, with audio it is better to sample at high depth and resolution
and then downgrade afterwards than it is to stay true to the final output.

Jon


I followed all the postings, and there are a lot of fine comments out
there. I guess I'll add my 2 bits:

First, going back to some of the earlier dither comments: The concept
of digital dither was covered pretty well - word length reduction. I
would probably magnify the fact that reduction with no dither causes 2
bad things. The first is harmonic distortions, the second is noise
modulation. Think for a second of a tiny sine wave, less than 1
quantization peak to peak, residing between 2 quantization levels. It
yields nothing. That is and incorect result. Now shift it so it
"seats" on a quantization level, going over and under and over and
under... by only one bit. That gives you a 1 bit amolitude square
wave... This is not a good signal either... This is just a starting
point to think about. Again you can see pictures on
www.lavryengineering.com (under suport, "Do you need 20 Bits" artical.

But I was more interested in analog dither. Years ago, I was
experimenting with in. Some of it was around Nyquist frequencies, some
was "subtuctive dither" (remove much in the digital domain what you
injected in the analog domain). Why analog dither? Well, nothing
happens if you feed a signal into a circuit and it does not cvause
some digital transitions... Say your circuit generates a different
code every 1 millivolt, but the signal is small and is always between
5.1 and 5.7mV... No action. So you need to "force it" to go over and
under some threshold (transition) levels. So we add noise. It does get
pretty complicated in a hurry. First knee jerk reaction is to add all
the noise in high frequncey. It is not that simple, because after
truncation, a lot of the high frequncey reapears in the audio range.

But you "have to do somthing"- add some random noise, thus trade off
some noise (dynamic range). For word reduction it is done to fix the
distortions and noise modulation problems. On the analog side, it was
about "getting anything to move" (and more than that). So the question
beacme, was is the best tradeoff? Least energy added for fixing the
problem? That was when triangle dither (2 least significant bits peak
to peak) solution came about.

But than there was a period where analog dither died (at least for
me). The quantization levels got so close to each other, that the
circuit noise was way bigger than I wished for. Take say a 10V range
and try for 20 bits - 10uV steps
(10 microvolts). Getting the signal in there with such low noie (10 uV
peak to peak) is not a walk in the park... You "call" the noise "too
much dither" and move farward...

But than analog dither came back - BIG TIMES! It started with 1 bit
(like DSD) and is now an everyday thing wth the newer multibit ADC's.
Those have just very few bits, thus the quantization levels are very
far apart. No longer in the microvolts, but in the volts (or near it).
This are all noise shaping converters. You feed back a filtered
quantized output and subtuct from the input.. This will take too long
to explain, but such a raw structure will have all sorts of problems
thus it requires dither, and often very large amplitude. The dither
randomness tend to "break up" what would otherwise be consistant
repetative patterns (called limit cycles), not exactly what you want
in the audio...

Regarding the comments on dynamic range and bits. Yes it was well
said, and about 6dB per bit is right. So 24 bits is about 144dB, and
we can all realize it is also a bit of crock. My AD122 MKII has the
largest dynamic range (127dB unweighted) so it is 21+ bits. My first
generation AD122 had 122dB (20 bits) and I called it a 20 bit
converter. Than all those 100-110dB devices apeared on the market with
"24 bits" on the panel and the sales guys insisted that I call mine a
24 bits. The bits are there, but the last few just bounce around
mindlesly - no realtion to the audio.

I have a cheap circuit for andom number generation. If I add 100 bits
of noie at the thend (least significant bits) will it be a 124 bits
machine? Lets go for it

I could probably design a 24 bit AD, and it will be an unbelivable
energy hog. Remember that we are on a log scale. The differance
between 20 and 24 is not "just 4 more". It is a factor of 16 more.
Sort of like a 6 earthquake vs an 8 earthquake. Big differance.

Do you need 24 bits AD? Probably not, short of some of the headroom
comments. What is the best Mic preamp out there? Say -130dBu? How much
gain is it set for? Say I use 30dB gain, than the noise floor is at
-100dBu and a peak to peak siganl out of the preampp is 24Bu driving
an AD. So we have 100+24=124dB
dynamic range. My AD122 MKII gives you 3dB margin. But say you need
40dB mic pre gain. Now you can use the 114dB dynamic range device... I
am not a recording engineer, but I think that 127dB is already only
for close mics that put out serious signal followed by a great mic
pre...

While I rather have folks specify dynamic range (not bits), the 24 bit
number is pretty good in the sense that it is a multiple of 8 bits
(thus 3 bytes). It is a good number for computers and hardware
(multiple of 8). I just wish 24 bits did not get used by industry
salesman as measure of quality.

Last comment for now: Whatever I said regarding 24 bits and what is
needed is ONLY for AD's and DA's. Let me call them CONVERSION BITS.
There are other type of bits. Let me call them PROCESSING BITS. If 24
conversion bits may be an overkill, 24 processing bits is way too
little. Folks need to realize it. A guy comes with some digital EQ and
tells you that it has 56 bits, or 32 floating point DSP, and he does
have a point to make. Just do not later go and look for a 56 bits
AD... Different issue.

I think it was already explained, but here we go: Say I want to
avarage 50 cent (money) and 51 cent. It is 50.5 cents. I can only deal
with cents so we either call it 50 or 51. We have 1/2 cent inacuracy.
In a bianry wold, I lost a bit of acuracy. So lets agree to have a new
coin - 1/2 cent. Now we avarage 50, 50 and 51. It yields 50.3333333...
Well ae can call it 50 or 50.5.... The general statment is that when
you add more and more computations, you reduce more and more of the
accuracy. So yes, we typicaly need a lot of computational bits.

To all, pardon me if soem of the above is a repeat of what you said, I
tried to fill some gaps, and probably covered things that needed no
help. I am new to audio NG.

BR
Dan Lavry
  #155   Report Post  
Mike Rivers
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain


In article writes:

My first
generation AD122 had 122dB (20 bits) and I called it a 20 bit
converter. Than all those 100-110dB devices apeared on the market with
"24 bits" on the panel and the sales guys insisted that I call mine a
24 bits. The bits are there, but the last few just bounce around
mindlesly - no realtion to the audio.


You're doing better than most, if not all. However, the term that we
throw around with reckless abandon is "24-bit" and not "24 bits of
resolution." Nobody said that those lowest order bits had to actually
carry information, they just have to be there so that a 24-bit
receiver will recognize the format.

I suspect that this may have been started by the A/D converter chip
manufacturers who wanted to produce chips with a 24-bit data word.
What goes in isn't their problem, it's the problem of the person
designing the application around the chip.

Do you need 24 bits AD? Probably not, short of some of the headroom
comments. What is the best Mic preamp out there? Say -130dBu? How much
gain is it set for? Say I use 30dB gain, than the noise floor is at
-100dBu and a peak to peak siganl out of the preampp is 24Bu driving
an AD. So we have 100+24=124dB
dynamic range. My AD122 MKII gives you 3dB margin. But say you need
40dB mic pre gain. Now you can use the 114dB dynamic range device...


I've just stirred up a discussion over on the Pro-Audio mailing list
about a related subject. How does someone who thinks that the
difference between a line level and mic level input is the kind of
connector used compare a the gain and noise performance of a preamp
which has only a digital output (integrated A/D converter with no
user-adjustable calibration) with a straight analog preamp and an A/D
converter of unknown input sensitivity for full scale (needless to say
and also unknown noise performance)? You can compare volts out to
volts in and get gain, or volts out for no volts in and get dB of
quiescent noise. But how do you relate volts in to dBFS on your DAW's
meter or headroom indicator? It's a different ball game, but trying ot
explain that you have to think differently about these things requires
more learning than some people (who buy by looking at spec sheets)
want to bother with.



--
I'm really Mike Rivers - )
However, until the spam goes away or Hell freezes over,
lots of IP addresses are blocked from this system. If
you e-mail me and it bounces, use your secret decoder ring
and reach me he double-m-eleven-double-zero at yahoo


  #156   Report Post  
Arny Krueger
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Chris Hornbeck" wrote in message

On Thu, 20 Nov 2003 16:04:43 GMT, Carey Carlan
wrote:

Dithering smooths out the differences
between the 65,535 steps, making them as smooth as the 16 million
steps.


Dither is just noise, but noise has a special property in this case.
Although it can't smooth out differences, it can remove errors.


Dither doesn't remove the errors, it just makes them more palatable to the
ear.

Undithered quantization error is among the nastier, more unpredictable forms
of distortion around.

From http://www.pcabx.com/technical/bits44/index.htm (which has freely
downloadable .wav files that are practical examples of how dither changes
things)

Artifacts of improperly dithered quantization:

(1) Background noise level, from dither or quantization noise. Is it steady
or does it follow the intensity or tonality of the test tone?

(2) Intermittent spurious little whistles or "birdies" due to lack of dither
and the steady frequency change in the tone.

(3) Loss of low level signal due to lack of dither.

(4) Raspy distorted sound due to lack of dither.

OTOH, properly dithered quantization error sounds like background noise. We
even have some control over what the background noise sounds like. As the
signal goes down in amplitude, dither keeps it audible and recognizable even
when it is smaller than one quantization step.




  #157   Report Post  
Arny Krueger
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Justin Ulysses Morse" wrote in message
m

Mike Rivers wrote:


I haven't given this any thought, so I'll just throw out the question
to ponder. Is it possible that time resolution between components of
a complex wave could be better with a higher sampler rate? Could the
phase relationship between the fundamental and, say, third harmonic
of a distorted guitar, be more accurately preserved at a higher
sample rate?


If my understanding is correct, and it's based mostly on what I read
here from I think Arny (but it makes perfect sense to me), then the
answer to your question is that yes, a higher sample rate (or a higher
bit rate for that matter) would improve that time resolution; but that
the resolution is already WAY higher than necessary. People assume
that the time-domain accuracy is one sample period; but it's really
one sample period divided by the quantization range. So the time
domain accuracy for CD audio would be 1/(44100*2^16) or 346
picoseconds. That's about a decimal place or two off of what I
vaguely remember Arny saying, so I've probably messed it up. But you
get the idea? I probably shouldn't be paraphrasing from memory, so
go back and read Arny's post from yesterday in one of these threads.


Bascially, you've the concept right.

Ditto what Jay said about dither, too. It inifinitizes things in both the
amplitude and time domain.


  #158   Report Post  
dan lavry
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

(Mike Rivers) wrote in message

You're doing better than most, if not all. However, the term that we
throw around with reckless abandon is "24-bit" and not "24 bits of
resolution." Nobody said that those lowest order bits had to actually
carry information, they just have to be there so that a 24-bit
receiver will recognize the format.


Well, perhapse you are correct for most cases. I will try and explain
my angle. I am not a recording or mastering engineer. I am an
equipment designer and manufacturer, so it does make a differance to
me when someone says: You sell 24 bits, your competition does too.
Both are 24 bits, and you cost more. It is the same thing with that
192Khz crock. I actualy did lose a big sale because I did not want to
join the "king has no cloth" parade and do 192KHz.

I do not know the precentage of folks that would be influenced by such
nonsense. But there are some out there that would buy a 192KHz 24Bits
machine that has over 1 ns jitter, 103dB dynamic range, bad
distortion, 10 msec recovary from clipping (overdrive)... and not take
the time to look at a 96KHz, 20 bits 110dB range low distortions, 20ps
jitter, 2usec recovary... for about the same price. After all, one
machine does 192/24. The other is only 96/20

As far as I know, digital recivers can handel various data length up
to 24 bits just fine. They assume "a bunch of trailing zeros" so a 16
bits with 8 trailing zeros look like a 24 bit. Yes, there is the
coding in the information side (declaring sample rate, bit,
emphasis...) and I do belive it defaults to 24 bits. It is not wise to
assume it. Most companies do not pay too much attention to the
information bits, certainly not on the DA side.

I've just stirred up a discussion over on the Pro-Audio mailing list
about a related subject. How does someone who thinks that the
difference between a line level and mic level input is the kind of
connector used compare a the gain and noise performance of a preamp
which has only a digital output (integrated A/D converter with no
user-adjustable calibration) with a straight analog preamp and an A/D
converter of unknown input sensitivity for full scale (needless to say
and also unknown noise performance)? You can compare volts out to
volts in and get gain, or volts out for no volts in and get dB of
quiescent noise. But how do you relate volts in to dBFS on your DAW's
meter or headroom indicator? It's a different ball game, but trying ot
explain that you have to think differently about these things requires
more learning than some people (who buy by looking at spec sheets)
want to bother with.


Good luck. I understand what you are saying. When you make mic pre,
you must specify analog in and analog out. You make an AD, you must
specify AD... and so on. For an oveall system one needs to incluse the
mic as well. Going for a high end system, one can disable the mic
pickup and still account for the mic noise by replacing the mic with a
noise equivalent physical resistor. One can measure the whole system,
or take the various parts and figure it out. But yes, once passed the
AD, you have to look at the digital outcome. I use FFT's and digital
distortion meters...

I tend to be blessed with customers that know what they are doing,
both in terms of gear and the musical ear and artistic tast. So I
should not complain too much. But often, the big time customers can
plain a simple do what it takes by using the best gear money can buy.
That is one way to go, if you have deep pockets, like the movie
industry. But there are a lot of folks that can not go up against the
stops on everything. At that point, it is wise to look for the
performance bottlenack and improve things one at a time... That is
when that knowladge can come handy...

BR

Dan L
  #159   Report Post  
Jay - atldigi
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

In article , "Arny Krueger"
wrote:

"Chris Hornbeck" wrote: steps.

Dither is just noise, but noise has a special property in this case.
Although it can't smooth out differences, it can remove errors.


Dither doesn't remove the errors, it just makes them more palatable to
the ear.


It can remove distortion but the error still exists as broadband noise.
I think Chris probably knew this but chose the wrong word.

--
Jay Frigoletto
Mastersuite
Los Angeles
promastering.com
  #160   Report Post  
Arny Krueger
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Jay - atldigi" wrote in message


In article , "Arny Krueger"
wrote:


"Chris Hornbeck" wrote: steps.


Dither is just noise, but noise has a special property in this case.
Although it can't smooth out differences, it can remove errors.


Dither doesn't remove the errors, it just makes them more palatable
to the ear.


It can remove distortion but the error still exists as broadband
noise.


Well, we call it noise but in fact it's 100% deterministic given that we
created the randomizing signal so we should know what it is.

I think Chris probably knew this but chose the wrong word.


Could be. But the point needs to be clearly made.

IMO, there's a lot of etymological weirdness in this area. Quantization
error is often called quantization noise. Spectral shaping of quantization
error is commonly called "noise shaping". Quantization error is noisy, but
it's noisy in the sense that loud neighbors are *noisy*. It's not noise in
the sense of random noise, because quantization error is 100% predictable.

I became impressed with how deterministic quantization error can be when I
started looking at measurements of a series of samples of the same sound
cards and saw how consistent they are. IME pure analog gear is not this
consistent. The consistency comes from the fact that the quantizers in the
converters were the major source of noise, and the quantizers were all
digital and therefore extremely similar.



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