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  #82   Report Post  
Arny Krueger
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Tommi" wrote in message

wrote in message
...
Justin Ulysses Morse wrote:

While it's true that the additional bits tack your extended
resolution onto "the bottom" of the dynamic range, it clearly
increases the resolution at all levels. You can have a -100dB
component to a -1dB signal, and you still want to hear it.


Is the ear even capable of hearing the -100 component against the
much louder -1? I thought masking pervented this.


Masking, it is frequency-dependent. However, this leads to thinking
about the fact that the human ear actually compresses dynamics at
higher sound pressures. My understanding is that we have roughly
80dB's worth of dynamic range at a time, which we then move according
to the sound pressure levels of the sound sources.


Probably less than 80 dB, more like 60 or 70 dB. The actual number depends
on the kinds of sounds you use to establish the 0 dB point. There's a big
difference in what happens if you set the 0 level with a sustained tone like
that from a piano, or a percussive tone, like that from a castanet or
triangle.

For example, if you'd be listening something at 110dB SPL for 5
minutes, after that you couldn't hear the same sound with 2dB SPL for
a while.


Yes, at sustained levels of 110 dB most people would experience quite a bit
of threshold shift.

It works the other way round too: If you're listening
something at 5dB spl for a while, and then suddenly the same sound
source produces a 120dB spl sound, your ear would compress it
lower(by stretching the eardrum, moving the hammer away from it etc)
in order to protect your hearing mechanism. This, however isn't true
with very short peaks because your protection mechanism takes some
time to wake up.


Agreed.


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

"Garthrr" wrote in message
...
Thanks Chris,
So is this point a matter of contention or is this agreed upon by all? If

it is
agreed upon then is the argument on the "24 bit sounds no better than 16

bit"
side that the effects of the ambiguity are inherently negligable or

perhaps
that interpolation or something else "repairs" the ambiguity adequately? I

hope
I have framed thew question well enough to be understood.

Garth~


24 bit sounds no better or may sound better simply because of the analog
front end of the converter and the quality of the converter itself in terms
of thermal noise. But beyond that, 24 bits is not greater resolution, and I
think that's what throws people off. It's greater bit depth, depth being
the representative word meaning it does down to further quiet (ideally -144
dBFS, but thermal noise plays a part along with the front end
implimentation).

Ostensibly this only means that RECORDINGS might sound better because
there's more available headroom above noise to fart around with, but great
quality digital recordings have been made on 16 bit machines, so it's not
inherent that 24 bits is better. It is inherent that one can be less
involved with watching levels on a session because now we can have peaks
that eek into -18 or so without worrying that one of the major hits is going
to freak into the clipping range. Otherwise, if you calibrate your system
where 0 dB VU = -20 dBFS or so then you still have the same headroom
regardless of bit depth.

The problem can lie in the fact that the noise floor is higher with 16 bit,
but it's more likely that almost any room (project studio concept) is going
to be far noisier than the theoretical lower limit of a 16 bit converter
anyway, so where's the great boost from 24 bit? Again, it comes down to
noise floor, and noise floor only. In the basic 16 bit world we just went
through one would be lucky to get a noise floor of -90 dBFS, but the
realworld range is more like -80 to -82 dBFS (cassette and vinyl were 10 to
25 dB noiser). In a 24 bit world, that figure drops maybe 20 dB or so, as I
often see about -104 dBFS for decent converters, and something like -110
dBFS on really good converters. Not even close to the theoretical lower end
that 24 bit converters have. But it is quieter, and this may make a
difference. It also may not. It is, by no means, the telling tale on
recording. It's simply a fact of the converters bit depth. Better ones are
quieter than cheaper ones. It's even possible that really good 16 bit
converters can sound better than really bad 24 bit ones.

However, DSD has been described as someone standing at a light switch,
turning it on and off. If they turn it on only once a second, then your
ability to resolve the room becomes haphazard has you try to scan the entire
contents of the room. The faster the room light is turned on and off, the
more perception there is that it's on and you have the greater resolving
power to see the contents. Before it was a chair here, a table there, etc.
Now as it goes faster and faster you see, not only the chair, but the
slipper protruding from underneath it that might trip you were you walking
around in the dark. As it goes even faster you see that there's little bits
of paper on the floor or dust or cat hair or cat hairball, etc.

I realize that it's not a technically correct description of what scanning
an input at 2.8 mHz is really like, but even in it's simple form it's easy
to see that, relative to the content, the faster the light goes off and on,
the more like being on it is. The more it seems like it's on, the more you
can see, or in the case of DSD, the more you can hear. Whether you NEED to
have all the content exposed is another question. If I don't intend to walk
across the room and kill the spider crawling up the wall, do I really need
the resolution to see it in the first place? Or do I just need to know the
layout of the room?

--


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.






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



  #84   Report Post  
Roger W. Norman
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Jay - atldigi" wrote in message
...
In article ,
(Garthrr) wrote:


Actually I was under the impresion that Jay was on the other side of the
fence--that he was saying that 24 bit was really no better than 16 bit
for any sort of real world audio.


Absolutely not. I know there's a lot of posting going on, and I've
written a lot in this thread, but I know I've stated several times that
the above is not what I'm saying. Instead of adding even more
confustion, please try to go back and read my posts again. Hopefully in
retrospect they'll make more sense.

Thanks,


Well, you have to think in terms of the statement "real world audio", then
it's possible that 24 isn't better than 16 if real world audio recording is
accomplished in a room where quiet is represented by about -55 dB of room
noise, which would be a lot of live situations (or worse) and in a lot of
home/project studios where thousands haven't been spent in acoustic taming.
At that point, I believe there's a point where, on mic'd instruments,
masking starts taking place if one is playing that quietly, and it's
questionable whether having 24 bit converters actually does anything for
you. In your real world audio, Jay, as a mastering engineer, you obviously
should work at the highest resolution one can give you, hopefully allowing
enough real world audio headroom to do your job without having to step on
it. In 16 bit products it was normal to see -.1 dBFS levels with absolutely
no room for a mastering engineer to do anything but bring it down in order
to bring it back up. 24 bits doesn't even stop this stupid waste of
bandwidth, but at least it allows some level of having real headroom over
noise than 16 bits did. Does that matter on the above scenario when one is
working in a noisy room? Probably not. Does it matter if the 24 bit's
analog front end sucks and the 16 bit's analog front end doesn't? Probably.

I guess the real world problem is that one cannot define "real world audio"
that sticks for everyone. Personally I'd start at the bottom and say if the
room is live and going to be noisy with A/C turning on and off, people
shuffling their feet or programs or coughing, then 24 bit isn't a necessity
in that "real world audio" environment. Somewhat akin to the idea of having
EQ boosted in the HF range on a bass track where there's no content. Stupid
waste of space. Or dollars.

--


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.





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

In article , "Roger W. Norman"
wrote:

"real world audio", then it's possible that 24 isn't better than 16 if
real world audio recording is accomplished in a room where quiet is
represented by about -55 dB of room noise,


There certainly are a number of situations - probably a majority if
we're talking about most popular music styles, where more than 16 bits
most likely won't help. That's true enough. And also. there are no
situations where an ADC at 24 bits is strictly necessary since there
aren't any converters that can reproduce 144 dB dynamic range. 20 would
be a practical way to go, with the exceedingly rare converter perhaps
approaching the equivalent of 21 bits and change. But hey, with a 24 bit
chip you know that it can convey the best the converter electronics can
throw at it, and those are the chips available, so it makes sense that
they would be used by designers. And there are some recording situations
where you can use more than 16 bits, so I have no problem having it
available and using it since storage is cheap. That's quite different
from saying you NEED it always, or you need it for final delivery on the
new Metallica record. One needs to understand all the details of digital
audio to know when it helps, when it doesn't, and why. There's not a
"one size fits all" answer.

In your real world audio, Jay, as a mastering engineer, you
obviously should work at the highest resolution one can give you,


I can tell you that the minute you string a bunch of processors
together, especially if one has tubes, the noise floor in any mastering
room will likely be at least a little above 16 bits. however, there is
still useful and audible signal that can happen within the noise, so it
certainly pays to convert with more than 16 bits to be as certain as
possible you're getting everything you can. As far as processing goes,
you want as many bits as you can throw at it. That's different criteria.

In 16 bit products it was normal to see -.1 dBFS levels with
absolutely no room for a mastering engineer to do anything but bring it
down in order to bring it back up.


Yeah, that's a pain. Well, not really, but one more unneccesary step.

--
Jay Frigoletto
Mastersuite
Los Angeles
promastering.com


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


"Jay - atldigi" wrote in message
...

"increased resolution" on a full-scale signal is nothing more than the
added ability to resolve the quietest overtones, then he's right and is
actually in full agreement with Arny.


At least somebody understands me, but I thought I had already said this
somewhere in the thread. It's those quieter components that you are
getting from the extra bits. The louder components aren't represented
any better. In the end, it can be a more precise and better sounding
recording (provided the source is of a quality to benefit), but it's
because of the little things you can now record, not that the big ones
are better.



Still, isn't it so that the number of possibilities doubles each time we add
a new bit, thus a 24 bit converter has 16, 777, 217 values to choose from
when converting the voltage to numbers. This means, that the
most-significant-bit of a 24 bit converter has (16,777,217 / 2) 8, 388,608
values to choose from when it's giving a number to any signal in the region
of roughly -6dB to 0 dB Full Scale, right?

That leads to the following:
A 24 bit system has quantized a loud component somewhere between -6 to 0 dB
FS
a lot more accurately than a 16 bit system would with the same signal, since
it's rounded the original voltage more precisely.
Also, if you were recording the same source with a 24 bit system peaking
at -48dB FS _and_ a 16 bit system peaking at 0dB FS, after normalizing the
24bit file to 0 dB, you would essentially have two identical files,
_identically_ quantized, since the 24 bit system had used its 16
least-significant-bits.

So, louder components are also represented better in a 24 bit system.
Are THESE aforementioned things something we can ALL agree on?




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

Jay - atldigi wrote in
:

But that's exactly my point: only the -100 component is what you've
gained. The -1 component is not rendered any better than it was
before.


Jay, as I am a person steeped in computer bits but weak on audio theory, I
ask you to explain that statement.

A 24-bit signal offers +/- 8,388,608 possible volume levels.

A 16-bit signal supports only +/- 32,768 volume levels.

Only +/- 128 levels (48 dB) of the 24-bit signal are less than the lowest
bit of the 16-bit (providing lower threshold).

The other bits provide 256 more possible levels between each of the 16-
bits' levels.

Do you disagree with any of the above? It's just math.

Is your argument that this higher precision is inaudible?
  #88   Report Post  
Bob Cain
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain



Tommi wrote:


Still, isn't it so that the number of possibilities doubles each time we add
a new bit, thus a 24 bit converter has 16, 777, 217 values to choose from
when converting the voltage to numbers. This means, that the
most-significant-bit of a 24 bit converter has (16,777,217 / 2) 8, 388,608
values to choose from when it's giving a number to any signal in the region
of roughly -6dB to 0 dB Full Scale, right?


That's right.


That leads to the following:
A 24 bit system has quantized a loud component somewhere between -6 to 0 dB
FS
a lot more accurately than a 16 bit system would with the same signal, since
it's rounded the original voltage more precisely.
Also, if you were recording the same source with a 24 bit system peaking
at -48dB FS _and_ a 16 bit system peaking at 0dB FS, after normalizing the
24bit file to 0 dB, you would essentially have two identical files,
_identically_ quantized, since the 24 bit system had used its 16
least-significant-bits.


Right again.


So, louder components are also represented better in a 24 bit system.
Are THESE aforementioned things something we can ALL agree on?


Yes. What must be remembered, however, is how the
inaccuracy is perceived. Many think that the increased
resolution results in less perception of some kind of
stairstep effect. That is not the case. The preceived
situation with an N bit converter done properly and going
through the A/D and then the D/A process is _exactly_ the
same as an infinite resolution conversion at both stages
with a digital adder in between just adding in a noise
signal comprised of a random variable with values of 0 or
2^-N at each sample time. What is heard is additive noise
and only that iff the conversion is done without correlation
between the value of that bit and the value of the sample.
This is practically achievable.


Bob
--

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

A. Einstein
  #89   Report Post  
Kurt Albershardt
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Jay - atldigi wrote:

with a 24 bit
chip you know that it can convey the best the converter electronics can
throw at it, and those are the chips available, so it makes sense that
they would be used by designers. And there are some recording situations
where you can use more than 16 bits, so I have no problem having it
available and using it since storage is cheap.


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.)

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

In article , Carey Carlan
writes:

The other bits provide 256 more possible levels between each of the 16-
bits' levels.

Do you disagree with any of the above? It's just math.


This the question I keep trying to get an answer to but after trying a number
of times over several years I have not gotten one. Its like the question just
bounces right off.

Garth~


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


  #91   Report Post  
S O'Neill
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Right. There are more discrete 24-bit sample values available between
those provided by 16-bit depth, just as there are below the lowest
16-bit (non-zero) sample. Much of the discussion of sample size focuses
on absolute dynamic range and seems to ignore the additional resolution
between 16-bit values over the entire range. And the absolute dynamic
range is somewhat irrelevant because no analog circuitry exists with 144
dB of dynamic range. Though it does buy a LOT of headroom (or slop room).

Illustration:

Near zero, showing increased range:
16-bit: 00000000 00000001 (00000000) - the lowest 0's are all you get
24-bit: 00000000 00000000 (00000000-11111111) - you get the whole range

The same applies between any two arbitrary 16-bit sample values:
16-bit: 01010101 10101011 (00000000)
24-bit: 01010101 10101011 (00000000-11111111) - the actual value can be
defined 256 times more closely.


Carey Carlan wrote:

Jay - atldigi wrote in
:


But that's exactly my point: only the -100 component is what you've
gained. The -1 component is not rendered any better than it was
before.



Jay, as I am a person steeped in computer bits but weak on audio theory, I
ask you to explain that statement.

A 24-bit signal offers +/- 8,388,608 possible volume levels.

A 16-bit signal supports only +/- 32,768 volume levels.

Only +/- 128 levels (48 dB) of the 24-bit signal are less than the lowest
bit of the 16-bit (providing lower threshold).

The other bits provide 256 more possible levels between each of the 16-
bits' levels.

Do you disagree with any of the above? It's just math.

Is your argument that this higher precision is inaudible?



  #92   Report Post  
Justin Ulysses Morse
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

Carey Carlan wrote:

Jay - atldigi wrote in
:

But that's exactly my point: only the -100 component is what you've
gained. The -1 component is not rendered any better than it was
before.


Jay, as I am a person steeped in computer bits but weak on audio theory, I
ask you to explain that statement.

A 24-bit signal offers +/- 8,388,608 possible volume levels.

A 16-bit signal supports only +/- 32,768 volume levels.

Only +/- 128 levels (48 dB) of the 24-bit signal are less than the lowest
bit of the 16-bit (providing lower threshold).

The other bits provide 256 more possible levels between each of the 16-
bits' levels.

Do you disagree with any of the above? It's just math.

Is your argument that this higher precision is inaudible?



I thought this was the question we just got hammered out. I'm going to
use a "completely" unrelated analogy to illustrate a very similar
misunderstanding that should hopefully make this clear (but may just
muddle things up more). Bear with me.

----analogy----
When we're talking about SAMPLE RATE, people who don't entirely
understand digital or analog audio get the mistaken impression that
higher sample rates (more samples) will result in a better
representation of the audio due to its documentation of information
in-between samples. Those who understand how it works try to explain
that the additional information "in between the samples" consists
NOTHING other than higher-frequency information. If you were to
"subtract" the lower Fs audio from the higher Fs audio, all that would
remain would be frequencies above the Nyquist frequency of the lower
Fs. I know you understand this one.
----analogy----

Okay, so let's think about the bit depth again. Suppose you have a
loud-ass signal represented by 16 bits. If you move to 24 bits, you
are now able to "more accurately" represent that loud-ass signal
because you can represent an actual signal that falls in-between the
bits in the 16-bit system. But this additional information is NOTHING
more than low-level information. If you were to "subtract" the 16-bit
audio from the 24-bit audio, all that would remain would be a signal at
below -96dBFS.

Can you see how these are similar situations in different domains?

It's true that higher sample rates give a more accurate reproduction,
but that accuracy is nothing but higher-frequency information.

It's true that higher bit depth gives a more accurate reproduction, but
that accuracy is nothing but lower-level information.

If you want to understand this more intuitively, you should try it.
Start with a very clean 24-bit recording. make a copy of this data,
and truncate it to 16 bits. Then convert the 16-bit file back to
24-bits. You now have two 24-bit files that are identical in the top
16 bits but different in the bottom 24. One file has music down there,
the other has silence. Now create a difference file from these two
(invert one and then sum them). You now have nothing but the bottom 8
bits of the original 24-bit file. Listen to it. It probably won't
even sound like the music it came from because all the "loud" stuff is
gone. All that remains is the low-level information that was riding on
those taller waves.

I know you're asking Jay and not me, but I'm confident we both
understand the issue and he just finished explaining that he doesn't
disagree with what you wrote above, and he hasn't said that the higher
precision is inaudible. What you need to realize about Jay's statement
(which was a reference to my explanation) is that the theoretical
-100dB and -1dB signals we're talking about are *simultaneous* and are
components of a single sound that can be considered separately for the
sake of analysis in attempting to understand this question of bit
depth, dynamic range, and resolution.

Does that make things any more clear?

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

In article , Justin Ulysses
Morse writes:

Okay, so let's think about the bit depth again. Suppose you have a
loud-ass signal represented by 16 bits. If you move to 24 bits, you
are now able to "more accurately" represent that loud-ass signal
because you can represent an actual signal that falls in-between the
bits in the 16-bit system. But this additional information is NOTHING
more than low-level information.


This is the first time I have heard this. Now this makes more sense to me. I
still dont understand why the info that falls between the 16 bits would
necessarily be low level information but I dont doubt that its true. Perhaps
someone could explain. Thanks Ulysses, thats a step forward in my
understanding.

Garth~


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

Garthrr wrote:

This is the first time I have heard this. Now this makes more sense to me.


I thought it was about the 4th or 5th time I said it in these threads
over the past 2 days, and I thought I was repeating myself. But I'm
glad to hear it's starting to gel.

I still dont understand why the info that falls between the 16 bits
would necessarily be low level information but I dont doubt that its
true.


Because the space between those bits is very tiny. Remember, we're not
talking about the whole 24-bit sample or the whole 16-bit sample.
We're talking about the DIFFERENCE between those two.

Think about a 16-bit sample as simply a 24-bit sample with 8 zeroes on
the end. So the difference between 24-bit audio and 24-bit audio
truncated to 16 bits is simply those last 8 bits dancing around. It
doesn't matter what the first 16 bits are doing because they're doing
it the same in both cases. We're only discussing what's in one sample
that's NOT in the other. Now, those last 8 bits can dance around as
rambunctiously as they like, but they'll never represent a DEVIATION
from the 16-bit sample of more than -96dBFS.

Suppose you have an input signal whose voltage at some arbitrary point
in time is 3.26534263219541623 volts. Now, off the top of my head I
estimate that the best approximation of this voltage you can represent
with 24 bits is maybe 3.2653426 volts. And 16 bits would round it off
to around 3.26534. So what's going on in the 24-bit audio that's
missing from the 16-bit audio? A signal in the neighborhood of 2.6
microvolts. Which is pretty dang low-level if you ask me. Even though
the signal we're listening to is up in the ballpark of full scale, the
"missing detail" we're talking about is down below -96dBFS.


Next question?

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

In article , Carey
Carlan wrote:

Jay - atldigi wrote in
:

But that's exactly my point: only the -100 component is what you've
gained. The -1 component is not rendered any better than it was
before.


First up, don't get hung up on this -1 and -100 example that was just in
reponse to another post where the example was given. The original
thought was useful enough conceptually, but shouldn't be thought of as a
real world practical example.


Jay, as I am a person steeped in computer bits but weak on audio theory,
I ask you to explain that statement.

A 24-bit signal offers +/- 8,388,608 possible volume levels.

A 16-bit signal supports only +/- 32,768 volume levels.

Only +/- 128 levels (48 dB) of the 24-bit signal are less than the lowest
bit of the 16-bit (providing lower threshold).

The other bits provide 256 more possible levels between each of the 16-
bits' levels.

Do you disagree with any of the above? It's just math.

Is your argument that this higher precision is inaudible?


That's the basic math for an ideal quantizer which is not what we use in
audio. Ideal (noiseless) quantizers are an excellent and necessary
learning aid, but aren't the ultimate implimentation we use in our audio
ADCs as they are non-linear and cause distortion (this non-linear
behavior being what you guys are giving examples of and worrying about).
When you look at a digital audio system as a whole, not just the
isolated parts (quantizer, sampler, filter etc.), it actually behaves a
little differently from how it's isolated parts first look on paper. We
deliberately use non-ideal quantizers to achieve linearity.

The key here, and some might not want to hear it, is dither. Dither is
not only used in the familiar requantizing. An example of requantizing
would be when you have a 24 bit file on your computer and "dither it"
down to 16 bits to go on a CD. In digital audio, all quantizers need to
be dithered. The quantizer in your ADC needs to be dithered to behave
properly.

Here's the important sentence taken right from Watkinson:

"The dither has resulted in a form of duty cycle modulation and the
resolution of the system has been extended indefinitely instead of being
limited by the size of the steps."

To explain all of this fully, the post would get very, very long. It may
be more effiecient to order a few AES papers (I could offer appropriate
references) and get Watkinson's and Pohlmann's books. However, I can
offer a few points that may be helpful.

The transfer function really does become linear with dither, at the
expense of a noise floor. Even in an ideal (noiseless) quantizer, loud,
complex signals' quantizing error will manifest as random noise. The
numerical values for the samples in such audio are so widely varying
that the quantizing errors will be independent or each other and will
distribute themselves in such a random pattern according to probability
that it will essentially behave as broadband noise (yet not of the same
gaussian distribution as thermal noise in an analog circuit).

Smaller or less complex signals, however, are more of a problem. This is
where the error becomes correlated to the signal and you get distortion.
What's worse, the distortion happens after the anti-aliasing filter, and
the harmonics of the distortion above Nyquist will alias causing
"birdies". Correct application of dither decorrelates the quantizing
error and linearizes the transfer function at the expense of a noise
floor. The resolution becomes, in effect, infinite. The limitation is
that the noise floor can be plainly audible in low resolution systems
and obscure low level detail in addition to being distracting. However,
within it's available dynamic range, the system is infinite. You can't
get better than that, even if you throw more bits at it. Taken as a
whole, the system works better than it would appear from the initial
learning aid of the ideal quantizer on paper. Add more bits, the noise
floor drops, and you can resolve smaller (quieter) details.

--
Jay Frigoletto
Mastersuite
Los Angeles
promastering.com


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

In article , "Tommi"
wrote:

"Jay - atldigi" wrote in message
...

"increased resolution" on a full-scale signal is nothing more than
the
added ability to resolve the quietest overtones, then he's right and
is
actually in full agreement with Arny.


At least somebody understands me, but I thought I had already said this
somewhere in the thread. It's those quieter components that you are
getting from the extra bits. The louder components aren't represented
any better. In the end, it can be a more precise and better sounding
recording (provided the source is of a quality to benefit), but it's
because of the little things you can now record, not that the big ones
are better.



Still, isn't it so that the number of possibilities doubles each time we
add a new bit, thus a 24 bit converter has 16, 777, 217 values to choose
from when converting the voltage to numbers. This means, that the
most-significant-bit of a 24 bit converter has (16,777,217 / 2)
8,388,608 values to choose from when it's giving a number to any signal
in the region of roughly -6dB to 0 dB Full Scale, right?

That leads to the following:
A 24 bit system has quantized a loud component somewhere between -6 to 0
dB FS a lot more accurately than a 16 bit system would with the same
signal, since it's rounded the original voltage more precisely.
Also, if you were recording the same source with a 24 bit system peaking
at -48dB FS _and_ a 16 bit system peaking at 0dB FS, after normalizing
the 24bit file to 0 dB, you would essentially have two identical files,
_identically_ quantized, since the 24 bit system had used its 16
least-significant-bits.

So, louder components are also represented better in a 24 bit system.
Are THESE aforementioned things something we can ALL agree on?


I'm going to repeat my last post in case you missed it or just glanced
over it because it's really the same question as Carey's. To preface:
what you see on paper with an ideal quantizer is not what you see in
actual practice with digital audio's intentionally non-ideal quantizers.
Also, we're ultimately dealing with voltages coming in and going out,
and the digits viewed at each isolated intermediate step don't tell the
whole story. Here goes:

--
That's the basic math for an ideal quantizer which is not what we use in
audio. Ideal (noiseless) quantizers are an excellent and necessary
learning aid, but aren't the ultimate implimentation we use in our audio
ADCs as they are non-linear and cause distortion (this non-linear
behavior being what you guys are giving examples of and worrying about).
When you look at a digital audio system as a whole, not just the
isolated parts (quantizer, sampler, filter etc.), it actually behaves a
little differently from how it's isolated parts first look on paper. We
deliberately use non-ideal quantizers to achieve linearity.

The key here, and some might not want to hear it, is dither. Dither is
not only used in the familiar requantizing. An example of requantizing
would be when you have a 24 bit file on your computer and "dither it"
down to 16 bits to go on a CD. In digital audio, all quantizers need to
be dithered. The quantizer in your ADC needs to be dithered to behave
properly.

Here's the important sentence taken right from Watkinson:

"The dither has resulted in a form of duty cycle modulation and the
resolution of the system has been extended indefinitely instead of being
limited by the size of the steps."

To explain all of this fully, the post would get very, very long. It may
be more effiecient to order a few AES papers (I could offer appropriate
references) and get Watkinson's and Pohlmann's books. However, I can
offer a few points that may be helpful.

The transfer function really does become linear with dither, at the
expense of a noise floor. Even in an ideal (noiseless) quantizer, loud,
complex signals' quantizing error will manifest as random noise. The
numerical values for the samples in such audio are so widely varying
that the quantizing errors will be independent or each other and will
distribute themselves in such a random pattern according to probability
that it will essentially behave as broadband noise (yet not of the same
gaussian distribution as thermal noise in an analog circuit).

Smaller or less complex signals, however, are more of a problem. This is
where the error becomes correlated to the signal and you get distortion.
What's worse, the distortion happens after the anti-aliasing filter, and
the harmonics of the distortion above Nyquist will alias causing
"birdies". Correct application of dither decorrelates the quantizing
error and linearizes the transfer function at the expense of a noise
floor. The resolution becomes, in effect, infinite. The limitation is
that the noise floor can be plainly audible in low resolution systems
and obscure low level detail in addition to being distracting. However,
within it's available dynamic range, the system is infinite. You can't
get better than that, even if you throw more bits at it. Taken as a
whole, the system works better than it would appear from the initial
learning aid of the ideal quantizer on paper. Add more bits, the noise
floor drops, and you can resolve smaller (quieter) details.

--
Jay Frigoletto
Mastersuite
Los Angeles
promastering.com
  #98   Report Post  
Arny Krueger
 
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"Garthrr" wrote in message


In article , Carey
Carlan writes:


The other bits provide 256 more possible levels between each of the
16- bits' levels.


Do you disagree with any of the above? It's just math.


This the question I keep trying to get an answer to but after trying
a number of times over several years I have not gotten one. Its like
the question just bounces right off.


It's doesn't bounce off everybody. I for one strongly affirm what Carey said
above. The relevant facts are very compelling to me. It's simply how things
work.


  #99   Report Post  
Roger W. Norman
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"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. 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).

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.

--


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.





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Arny Krueger
 
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Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

"Garthrr" wrote in message

In article , Justin
Ulysses Morse writes:

Okay, so let's think about the bit depth again. Suppose you have a
loud-ass signal represented by 16 bits. If you move to 24 bits, you
are now able to "more accurately" represent that loud-ass signal
because you can represent an actual signal that falls in-between the
bits in the 16-bit system. But this additional information is
NOTHING more than low-level information.


This is the first time I have heard this. Now this makes more sense
to me. I still dont understand why the info that falls between the 16
bits would necessarily be low level information but I dont doubt that
its true. Perhaps someone could explain. Thanks Ulysses, thats a step
forward in my understanding.


The information that falls between the pairs of 16 bit values is obviously
very small, so I guess its fair to call it "low level information".

Imagine two 16 bit signals. One is a 1 KHz sine wave and one is a 10 KHz
sine wave. Both are integers that vary between 0 and 65535. Now, let's take
both sine waves and convert them to 24 bit numbers by adding trailing zeros.
They are now both numbers that vary between 0 and 16,777,215. Actually,
since the numbers were created by adding 8 zeroes to the end, no number goes
above 16,776,960. Both sets of numbers are a bit unusual, in that every
number ends with 8 zeroes.

Let's take the 10 KHz sine wave and divide it by 65536. It becomes very
low-level signal. It becomes a number that varies between 0 and 255.

Now, lets add the 10 KHz sine wave to the 1 KHz sine wave. Finally, we have
a set of numbers that don't all end in 8 zeroes!

The samples whose last 8 bits aren't zero represent the 10 KHz low level
signal, don't they? So the low level detail is, in a manner of speaking, in
the samples that are "in between".

If we want to separate the 1 KHz and 10 KHz signals, we have two possible
strategies:

(1) Do frequency domain filtering

(2) Take the top 16 bits of every sample which will give us back the
original 1 KHz sine wave, and take the bottom 8 bits every sample which will
give us back an 8 bit 10 KHz sine wave.




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

In article , Jay -
atldigi writes:

The transfer function really does become linear with dither, at the
expense of a noise floor.


Ok, there's another important piece of the puzzle (at least for me).
So if I understand correctly, its a situation where what would seem to be ideal
on paper is problematic as far as implementation because of the side effects of
distortion and aliasing. So dither cures these problems at the expense of the
noise floor by somehow either correcting the quantization error or rendering it
harmless. So what would be stairstep errors in voltage are smoothed over by
dither and thus the resolution of the system in the higher levels becomes, for
all practical purposes, perfect. Is that an essentially correct
oversimplification?

Garth~


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

In article , Justin Ulysses
Morse writes:

I thought it was about the 4th or 5th time I said it in these threads
over the past 2 days, and I thought I was repeating myself. But I'm
glad to hear it's starting to gel.


You may well have said it and I could have either missed the post or missed the
point. Either way it is starting to make sense to me even though its still a
little blurry.

Suppose you have an input signal whose voltage at some arbitrary point
in time is 3.26534263219541623 volts. Now, off the top of my head I
estimate that the best approximation of this voltage you can represent
with 24 bits is maybe 3.2653426 volts. And 16 bits would round it off
to around 3.26534. So what's going on in the 24-bit audio that's
missing from the 16-bit audio? A signal in the neighborhood of 2.6
microvolts. Which is pretty dang low-level if you ask me.

Here is a bblluurrrryy moment for me. Is it that there is a _signal_ which is
2.6 microvolts or... is it that there is an error of 2.6 microvolts in the
reproduction of a signal which is the above 3.26534263219541623 volts? To me
this seems like a qualitative difference (no matter how insignificant the
quantity in question may be).

I think its difficult for someone who knows a lot about a thing to explain it
to someone who knows nothing about it because the one who knows is apt to
assume certain understandings on the part of the one who doesnt.

Garth~


"I think the fact that music can come up a wire is a miracle."
Ed Cherney
  #103   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:


The information that falls between the pairs of 16 bit values is obviously
very small, so I guess its fair to call it "low level information".


Oh I never thought about it in that way. I took "low level" to mean low dB
level as in -100dB or something of that sort. I guess instead it was meant as
"insignificant".

Garth~


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

In article , Jay -
atldigi writes:

what you see on paper with an ideal quantizer is not what you see in
actual practice with digital audio's intentionally non-ideal quantizers.


I think this is really important for anyone who is struggling to get this stuff
as I am. Until this morning I was not aware of this although it doesnt surprise
me.

Garth~


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

In article , Jay -
atldigi writes:

There are probably a few other helpful things that
you may have missed or that didn't quite make sense the first time
through.


Good lord! You have a singular gift for understatement!

Garth~


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


  #106   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:

"Garthrr" wrote in message


In article , Carey
Carlan writes:


The other bits provide 256 more possible levels between each of the
16- bits' levels.


Do you disagree with any of the above? It's just math.


GW
This the question I keep trying to get an answer to but after trying
a number of times over several years I have not gotten one. Its like
the question just bounces right off.


AK
It's doesn't bounce off everybody. I for one strongly affirm what Carey said
above. The relevant facts are very compelling to me. It's simply how things
work.


I guess what I'm trying to find out is whether others do not agree with what
Carey said--whether this is in contention or not.
From something I just read I'm beginning to think that dither is somehow
responsible for negating the advantage of that extra resolution at least in
higher level signals.

Garth~



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

"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).


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.



  #108   Report Post  
Arny Krueger
 
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"Garthrr" wrote in message

In article , "Arny Krueger"
writes:

"Garthrr" wrote in message


In article , Carey
Carlan writes:


The other bits provide 256 more possible levels between each of the
16- bits' levels.


Do you disagree with any of the above? It's just math.


GW
This the question I keep trying to get an answer to but after trying
a number of times over several years I have not gotten one. Its like
the question just bounces right off.


AK
It's doesn't bounce off everybody. I for one strongly affirm what
Carey said above. The relevant facts are very compelling to me. It's
simply how things work.


I guess what I'm trying to find out is whether others do not agree
with what Carey said--whether this is in contention or not.


On Usenet, *anything* can be in contention! ;-)

From something I just read I'm beginning to think that dither is
somehow responsible for negating the advantage of that extra
resolution at least in higher level signals.


Not if dither is used correctly, and these days it often is used correctly.

Usually, dither is sized to match the actual size of the smallest
quantization step. This number has to be chosen carefully because the
smallest step being quantized by just about all 24-bit converters isn't the
theoretical 1 sixteen-millionth of full scale. It's something vastly bigger,
and it varies quit a bit from converter to converter. But the education
process has worked and most chip designers know about this.

Dither makes the size of the quantization steps much more sonically
palatable, no matter how big or small they are.

From the standpoint of technical accuracy, dither doesn't negate any
advantages, and it doesn't level any playing fields.

These days, *everybody* uses dither in their quantizers, and IME *everybody*
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.



  #109   Report Post  
Arny Krueger
 
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"Garthrr" wrote in message

In article , Justin
Ulysses Morse writes:

I thought it was about the 4th or 5th time I said it in these threads
over the past 2 days, and I thought I was repeating myself. But I'm
glad to hear it's starting to gel.


You may well have said it and I could have either missed the post or
missed the point. Either way it is starting to make sense to me even
though its still a little blurry.


Suppose you have an input signal whose voltage at some arbitrary
point in time is 3.26534263219541623 volts. Now, off the top of my
head I estimate that the best approximation of this voltage you can
represent with 24 bits is maybe 3.2653426 volts. And 16 bits would
round it off to around 3.26534. So what's going on in the 24-bit
audio that's missing from the 16-bit audio? A signal in the
neighborhood of 2.6 microvolts. Which is pretty dang low-level if
you ask me.


Here is a bblluurrrryy moment for me. Is it that there is a _signal_
which is 2.6 microvolts or... is it that there is an error of 2.6

microvolts
in the reproduction of a signal which is the above
3.26534263219541623 volts?


Both. You can think of 3.26534 volts as 3.26534263219541623 volts with an
approximately 2.6 microvolt error, or you can think of .26534 volts as a
3.26534263219541623 volt signal with an approximately 2.6 microvolt error
voltage added.

To me this seems like a qualitative
difference (no matter how insignificant the quantity in question may
be).


It is a small qualitative difference. It's an error that is about 120 dB
down in the presence of a signal which is close to full scale (0 dB), which
means don't worry about it. If the size of a significant signal was -100 dB,
then it would be worth worrying about.




  #110   Report Post  
Arny Krueger
 
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"Garthrr" wrote in message

In article ,
Jay - atldigi writes:

The transfer function really does become linear with dither, at the
expense of a noise floor.


Ok, there's another important piece of the puzzle (at least for me).
So if I understand correctly, its a situation where what would seem
to be ideal on paper is problematic as far as implementation because
of the side effects of distortion and aliasing. So dither cures these
problems at the expense of the noise floor by somehow either
correcting the quantization error or rendering it harmless.


Dither clearly doesn't correct quantization error. Dither does change
quantization error into something that is sonically more benign. What dither
does is sort of like turning sewage into beer. The sewage hasn't been
exactly been turned into chemically pure water. But, the sewage has been
turned into beer, which is something that is more palatable to most of us
than sewage.

So what
would be stairstep errors in voltage are smoothed over by dither and
thus the resolution of the system in the higher levels becomes, for
all practical purposes, perfect. Is that an essentially correct
oversimplification?


The stairstep errors don't get smoothed, they become randomized. Think of
quantization errors as being splats of paint on the wall. Without dither the
splats of paint would tend to be grouped, so they would look like bigger
messier splats.

Dither makes the splats more uniformly distributed so they appear to be
smaller and actually more like some nicely shaded, more neutral color.

BTW, dithering pretty much works this way when it's applied to computer
graphics.





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Justin Ulysses Morse
 
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Garthrr wrote:


Here is a bblluurrrryy moment for me. Is it that there is a _signal_ which is
2.6 microvolts or... is it that there is an error of 2.6 microvolts in the
reproduction of a signal which is the above 3.26534263219541623 volts? To me
this seems like a qualitative difference (no matter how insignificant the
quantity in question may be).


It's exactly the same thing either way you think about it. The 16-bit
signal is the what you'd end up with if you started with the 24-bit
signal and added (or subtracted) the error signal. Since we're
"rounding" (kind of), sometimes the 16-bit value will be a little
bigger, some times it'll be a little smaller than the 24-bit value. So
when you look at the error "signal" over the course of time, it's
constantly bouncing up and down, not quite randomly but sort of
arbitrarily because the music doesn't care where those 16-bit
quanitizations are.

Of course I'm ignoring the topic of dither for the sake of theory, but
as Jay points out Dither changes everything. Dither is another "error"
signal added to the 24-bit signal just before you do the "rounding."
The dither signal, which averages just under -96dB, plus those 8 lost
bits which average under -96dB, add up to a "music plus noise" signal
which averages just OVER -96dB, so it is able to show up in the 16-bit
signal.


ulysses
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Justin Ulysses Morse
 
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Arny Krueger wrote:

Now, lets add the 10 KHz sine wave to the 1 KHz sine wave. Finally, we have
a set of numbers that don't all end in 8 zeroes!


I like this example because it equates to something musically
meaningful. That 10kHz sine wave is a high-order even harmonic of the
1kHz sine wave. Though our example created it synthetically, it could
just as easily have been a distortion component in an amplifier, or an
overtone in an acoustic instrument. It contributes to the tonal
character of the louder 1kHz tone. Removing this harmonic, either by
filtering out the high frequencies or by swamping it with noise or
quantization error, will change the musical character of the music just
like adding a harmonic that wasn't originally there would.


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

Carey Carlan writes:

The other bits provide 256 more possible levels between each of the
16- bits' levels.



Garthrr wrote:

I guess what I'm trying to find out is whether others do not agree with what
Carey said--whether this is in contention or not.


I haven't heard anybody disagree. I don't think anybody will.

From something I just read I'm beginning to think that dither is somehow
responsible for negating the advantage of that extra resolution at least in
higher level signals.


It uses the statistical average of the dither noise to bring the "extra
resolution" up into the range of the smaller sample size. This is
really a whole new can of worms though. It complicates the "bits"
question, which really does make sense even before you start talking
about dither.



ulysses
  #114   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:

Think about a 16-bit sample as simply a 24-bit sample with 8 zeroes on
the end. So the difference between 24-bit audio and 24-bit audio
truncated to 16 bits is simply those last 8 bits dancing around. It
doesn't matter what the first 16 bits are doing because they're doing
it the same in both cases. We're only discussing what's in one sample
that's NOT in the other. Now, those last 8 bits can dance around as
rambunctiously as they like, but they'll never represent a DEVIATION
from the 16-bit sample of more than -96dBFS.


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


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

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


"Jay - atldigi" wrote in message
...
Correct application of dither decorrelates the quantizing
error and linearizes the transfer function at the expense of a noise
floor. The resolution becomes, in effect, infinite. The limitation is
that the noise floor can be plainly audible in low resolution systems
and obscure low level detail in addition to being distracting. However,
within it's available dynamic range, the system is infinite. You can't
get better than that, even if you throw more bits at it. Taken as a
whole, the system works better than it would appear from the initial
learning aid of the ideal quantizer on paper. Add more bits, the noise
floor drops, and you can resolve smaller (quieter) details.


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.
Thank you Jay for your information!







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

Jay - atldigi wrote in
:

In article , Carey
Carlan wrote:

Is your argument that this higher precision is inaudible?


excerpts of Jay's reply :

The transfer function really does become linear with dither.
The limitation is that the noise floor can be plainly
audible in low resolution systems.
However, within it's available dynamic range, the system is infinite.
You can't get better than that, even if you throw more bits at it.


Therefore, your answer is "No". Dithering smooths out the differences
between the 65,535 steps, making them as smooth as the 16 million steps.

Add more bits, the noise floor drops, and you can resolve smaller
(quieter) details.


So, while the bits between the higher samples aren't necessary because
dithering smooths out the differences, the bottom 256 values between 16-bit
1 and 16-bit 0 allow you to use finer, quieter dithering that drops the
noise floor by 48 dB (ideally).

Thank you! I think I understand now.
  #118   Report Post  
2mb
 
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but I'll bet you could sell 8-bit pop music CDs today and nobody would
complain about the sound quality.

Given the quality of the mp3's I hear in my thieving friend's cars and
computers... quality doesn't seem to matter to a good percentage of
listeners. They sound decidedly Ensoniq Samplerish (circa 1988, 8bit/22k?) A
lot are tuned for 96kbps download speeds, not to be confused with 96khz,
128kbps is approximately like a cd which has some data compression problems
(depending on the mp3 encoder and settings). A lot of casual listeners don't
know (and probably don't care) what they are missing.

Sad that they are stealing, tragic that they don't care what they are
missing in terms of quality. It is a testament to their engineering that
these recordings are even 1/2 listenable after such abuse. Maybe RIAA will
help fix this: )

These people wouldn't even notice the sound quality if they were to (ahem)
buy such an 8 bit cd. Their ears are already accustomed to it. It would
sound like the rest of the stuff they listen to.


"Mike Rivers" wrote in message
news:znr1069010813k@trad...

In article

writes:

So what has happened? Yes, we have increased out dynamic range by 6 dB
between the loudest and softest signals the system can represent. But
we have increased the resolution throughout the system: from a 6dB
increment to a 4 dB increment.

Continuing to a 3 bit system: dynamic range is 18 dB. Values are 0dB,
2.25 dB, 4.5 dB, etc. to 18 dB. Wth each additional bit the dynamic
range is increasing, but ALSO the resolution is increasing everywhere
in the system.


So your next question should be: "What do I HEAR that's different?"

A good way to answer that question is to listen to some very low bit
rate recordings. I know this sounds like blasphemy, but once you get
up to about 8 bits, you don't get the sense that you're increasing
resolution, you get the sense that you're reducing the background
noise level into which the signal disappears. So yes, your ears are
able to RESOLVE a lower level signal in the presence of noise because
the resolution down there is better.

However, on the practical side, since most of the music we listen to
today has a dynamic range of less than 10 dB and is played back well
above the system and ambient noise floor, you don't get much of a
chance to take advantage of the added resolution. Of course it doesn't
hurt to have it there (for the occasions where you actually can use
it) but I'll bet you could sell 8-bit pop music CDs today and nobody
would complain about the sound quality.




--
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



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

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. The
process of quantizing has fundamental errors estimating the smallest
bit, errors which track the signal itself. Dither randomizes this
error completely, leaving only noise.

A 24 bit conversion of the signal with the dither level of a 16 bit
conversion would have no more (or less) information than a 16 bit
conversion.

So, while the bits between the higher samples aren't necessary because
dithering smooths out the differences, the bottom 256 values between 16-bit
1 and 16-bit 0 allow you to use finer, quieter dithering that drops the
noise floor by 48 dB (ideally).


Perzactly!

Chris Hornbeck

"That is my theory, and what it is too."
Anne Elk
  #120   Report Post  
Justin Ulysses Morse
 
Posts: n/a
Default 16 bit vs 24 bit, 44.1khz vs 48 khz <-- please explain

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.

ulysses
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