[SlimDevices: Audiophiles] 16bit versus 24bit versus dynamic range

[ralphpnj]

Disclaimer: I may be over simplifying things a bit for the sake of
clarity

Yes I know an odd title but here's the issue that I'm trying to
resolve:

An orchestra is recorded while playing a symphony and the dynamic range
of the performance is a maximum of 65 db above the noise floor - so if
the noise floor is at 30db then the maximum dynamic range required to
record the performance without distortion at the peak volume would be 95
db (30db + 65 db).

The maximum dynamic range of a 16 bit file is 96 db and that of 24 bit
is 144 db

So how does placing the above symphony recording into a 24 bit file
versus a 16 bit file increase the dynamic range of the recording?

Okay so I realize that putting the above symphony recording into a 24
bit file does not actually increase the inherent dynamic range of the
recording and this the point I'm trying to make. To put it another way,
I don't believe that any listening tests, either double blind or
sighted, are even need to determine if this is true since it is just a
purely mathematical function - increasing the bit depth of a digital
audio recording cannot and does not increase the dynamic range of the
music contained in the digital audio file. I like to think of the whole
thing using an envelope analogy - the recording of the performance would
be a letter on an 8 1/2" X 11" piece of paper with the 16 bit file being
a 9" X 12" envelope and the 24 bit file being an 11" X 14" envelope -
the letter nicely fits into both envelopes however the size of the
letter remains the same.

Am I missing something or is the whole 16 bit versus 24 bit for a fully
edited and mixed recording just another completely false marketing
claim?



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[TerryS]

In the example you picked, your point is correct. Since you stated that
the dynamic range of the original performance was 65 dB, then digitizing
it to more than 65 dB is a waste.
If a symphony was limited to 65 dB dynamic range, then 11 bits would be
enough.
But is 65 dB a realistic number for a symphony performance? Does that
allow for the very quietest moment (when no one is playing, speaking, or
coughing) to the very loudest drum crash or cannon (think 1812
Overture)?
The dynamic range of human hearing is often estimated at 140 dB
(threshold of hearing to threshold of pain). I don't know how much a
symphony performance could get to, but 65 dB seems pessimistic.

The other thing to keep in mind is that you would need to leave some
headroom at the high end unless you knew in advance what the highest
peak level of the performance was going to be. If you guess too low,
you will clip on the peak. If you guess too high, you will be 'throwing
away' that much dynamic range. So you might want 10 or 20 dB extra just
to be sure you stay out of clipping. Then your 65 dB symphony would
need to be recorded with 75 or 85 dB of dynamic range available.

Terry


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[cliveb]

Post by TerryS
The other thing to keep in mind is that you would need to leave some
headroom at the high end unless you knew in advance what the highest
peak level of the performance was going to be. If you guess too low,
you will clip on the peak. If you guess too high, you will be 'throwing
away' that much dynamic range. So you might want 10 or 20 dB extra just
to be sure you stay out of clipping. Then your 65 dB symphony would
need to be recorded with 75 or 85 dB of dynamic range available.

At the recording stage, yes. I don't think anyone would argue against
recording at 24 bits. Apart from the headroom issue, it also allows lots
of editing and DSP to be done without accumulating quantisation noise.
But once it's all done and dusted, there's no point packaging it for
distribution at anything more than 16 bit.



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[arnyk]

Post by cliveb
At the recording stage, yes. I don't think anyone would argue against
recording at 24 bits.

Think again. As I pointed out with simple math in a recent post, 24/96
adds about 200% wasted space to a 16/44 recording. If you are recording
stereo, this is manageable even negligible, but if you are multitracking
with a goodly number of channels, the file sizes can get to be
challenging to manage, and for what? Numbers for the sake of numbers?

Post by cliveb
Apart from the headroom issue, it also allows lots of editing and DSP to
be done without accumulating quantisation noise. But once it's all done
and dusted, there's no point packaging it for distribution at anything
more than 16 bit.

In the real world the actual performance probably has more like 65-70
dynamic range, so using 16 bits gives you 25-30 dB headroom, which any
recording engineer worth his salt can work with without problems.

Most DSP processing and editing uses 32 bits and up, so the noise it
adds to a 16 bit context is negligible.


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[pablolie]

the rational consensus i have read is that while a very few people have
hearing range that goes beyond 16/44, no human being can possibly
resolve beyond 20/44. 44 gives you the best possible range of human
hearing as established in innumerable tests, and provided to you by
Nyquist. 20bits covers the extreme best possibly case of dynamic hearing
capabilities.

there are some good tests online for those who want to be honest to
themselves. my hearing goes from 35-16.5k, and i can just hear a dynamic
range of <16, the 20 is wasted on me. it's the quiet passages that get
you with age. and I listen to the simple test (no wide staging no depth
stuff) with stellar headphones and a great headphone amplifier. and fail
when it comes to the theoretical 20/44 optimal human hearing limit.



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[cliveb]

Post by arnyk
Most DSP processing and editing uses 32 bits and up, so the noise it
adds to a 16 bit context is negligible.

Let's say you have a 16 bit recording and decide to perform some DSP on
it at 32 bit accuracy. So far so good. Now you need to store the
results, so you presumably dither it back down to 16 bits. Next day you
decide to do some more DSP, and the results are once more dithered back
down to 16 bits. Same again the next day, and the next. How many times
can you afford to do this before the multiple instances of dither noise
might become an issue?



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[arnyk]

Post by cliveb
Let's say you have a 16 bit recording and decide to perform some DSP on
it at 32 bit accuracy. So far so good. Now you need to store the
results, so you presumably dither it back down to 16 bits. Next day you
decide to do some more DSP, and the results are once more dithered back
down to 16 bits. Same again the next day, and the next. How many times
can you afford to do this before the multiple instances of dither noise
might become an issue?

In this case (the simple addition of spectrally similar uncorrelated
noise) noise accumulates at a rate of 3 dB times the log base 2 of the
number of passes. IOW 3 dB more noise after 2 passes, 3 dB more noise
on top of that for 4 total passes for 6 dB total, 3 dB more than that
for 8 total passes for 9 dB, etc. In a real world case where there is
already 20 dB or more headroom, it is not a big problem.


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[drmatt]

Post by arnyk
Incorrect. You might want to get up to speed with a thread (especially
one this short) before making mistakes like that. Please see post 8.

Sarcasm doesn't make you right.




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[arnyk]

Post by drmatt
Sarcasm doesn't make you right.

Trying to make a big show out of correcting someone doesn't buy you
much if they already admitted their mistake which of course I did some
hours ago. Do try to keep up!


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[docbob]

Post by arnyk
Trying to make a big show out of correcting someone doesn't buy you
much if they already admitted their mistake which of course I did some
hours ago. Do try to keep up!

Gracious acceptance of your own fallibility is not your strong suit on
any forum, is it?

I always respond with "oops, thanks for pointing that out", not a
childish attack rant with the admission embedded deep within. FYI, there
is no "oops" about my post 9, because you point out no errors, just
amusingly, flailingly shadow box.

But the OP wants to get back on topic, so I'm done with you.


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[pablolie]

this forum's ability to religiously and passionately recall each other's
audiophile and mensa license over a dispute about 4 bits of headroom
-that 99.9999% of the world's population doesn't give a hoot about-
never ceases to entertain me. :-D



...pablo
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Computer audio: workstation --USB->- audioengine D1 -> Grado
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[jfo]

Post by pablolie
this forum's ability to religiously and passionately recall each other's
audiophile and mensa license over a dispute about 4 bits of headroom
-that 99.9999% of the world's population doesn't give a hoot about-
never ceases to entertain me. :-D

Agreed. But it's not this forum, it's just 2 or 3 of the same old people
who like to engage in this mental masturbation.


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[arnyk]

Post by docbob
Gracious acceptance of your own fallibility is not your strong suit on
any forum, is it?

How would you know?

Post by docbob
I always respond with "oops, thanks for pointing that out", not a
childish attack rant with the admission embedded deep within. FYI, there
is no "oops" about my post 9, because you point out no errors, just
amusingly, flailingly shadow box.

Really? You've missed any number of opportunities to do so lately as you
have had to be corrected on many points.

Besides, the minor math error does not change the important point. If I
record a good sized multitracked musical work 16/44 I end up with a
large, awkward, but still somewhat manageable set of files running about
11 GB. Not bad to store as storage is cheap, but loading and saving
audio files this size is already time consuming. Of course most people
who advocate high sample rates have never actually had to struggle with
this sort of thing.

If I follow the well meaning but mistaken advice given by certain people
on this forum, and record at 24/192 I bloat the files for the same
musical work to about 60 GB, which is large enough that there is no
longer enough room for multiple generations of the files on any but the
largest storage units. Processing now takes six times longer, which can
run into large fractions of an hour, and yields no audible benefits. And
that is just for loading and saving files.


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[drmatt]

Post by arnyk
How would you know?

"A simple internet search will reveal...."




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[arnyk]

Post by drmatt
"A simple internet search will reveal...."

Prove it.


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[drmatt]

Post by arnyk
Prove it.

I think just flipping back through this thread is proof enough, let
alone searching the rest of the internet.

What are you even doing here anyway? Why do you even care what people on
the squeezebox forum think? Seems like a bit of a comedown to me given
your apparent reputation.




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[Julf]

Post by drmatt
I think just flipping back through this thread is proof enough, let
alone searching the rest of the internet.
What are you even doing here anyway? Why do you even care what people on
the squeezebox forum think? Seems like a bit of a comedown to me given
your view of your reputation.

Could we please keep the discussion factual instead of descending into
silly ad hominems?



"To try to judge the real from the false will always be hard. In this
fast-growing art of 'high fidelity' the quackery will bear a solid gilt
edge that will fool many people" - Paul W Klipsch, 1953
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[drmatt]

Post by Julf
Could we please keep the discussion factual instead of descending into
silly ad hominems?

If only I'd been the first, or worst, I'd agree with you on that.

But really, it's a media streamer from a company that makes mice and
keyboards. It's hardly an important part of the HiFi landscape. It's
rather like finding the queen delivering your milk.




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[arnyk]

Post by drmatt
If only I'd been the first, or worst, I'd agree with you on that.

You're not the first, but the other point seems to be one of your
personal goals.

But really, it's a media streamer from a company that makes mice and
keyboards. It's hardly an important part of the HiFi landscape. It's
rather like finding the queen delivering your milk.

This begs the question of why do you Dr Matt bother with this place, given
your apparent disrespect for their company, and their products.

In contrast to you, I admire Logitech, and am using several of their
products as I edit this post.


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[drmatt]

Post by arnyk
You're not the first, but the other point seems to be one of your
personal goals.
This begs the question of why do you Dr Matt bother with this place,
given your apparent disrespect for their company, and their products.
In contrast to you, I admire Logitech, and am using several of their
products as I edit this post.

You're not the first either, but you are very very good at it.

I love my squeezeboxes. All seven of them. It's a fantastic product that
was totally ahead of its time and totally misunderstood. But it was
never a big success, it was too expensive for most people and the rest
of the world just didn't care. Pragmatic is the word. Bravo Logitech for
picking it up but it didn't really work out.




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[arnyk]

Post by drmatt
I think just flipping back through this thread is proof enough, let
alone searching the rest of the internet.

What this thread shows is a lot of accurate relevant comments, and some
comments that are just paraphrases of common audiophile myths.

I know from decades of experience that those who need audiophile myths
to be comfortable are very dependent on these myths for reinforcement of
their emotions and atypical behaviors, and typically respond to
corrections of them with denial.

Post by drmatt
What are you even doing here anyway?

Discussing technical topics in audio, particularly as they relate to
digital audio hardware.

Post by drmatt
Why do you even care what people on the squeezebox forum think?

Many of the people on this forum feel about audiophile myths just like I
do, and we share information.

The Squeezebox is an effective, pragmatic piece of audio gear if a bit
legacy, and therefore runs afoul of any number of audiophile myths.

Post by drmatt
Seems like a bit of a comedown to me given your view of your reputation.

I have a reputation? ;-)

Do you think that my reputation should make me disinterested in what
well-informed audiophiles believe?


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[arnyk]

Post by ralphpnj
An orchestra is recorded while playing a symphony and the dynamic range
of the performance is a maximum of 65 db above the noise floor - so if
the noise floor is at 30db then the maximum dynamic range required to
record the performance without distortion at the peak volume would be 95
db (30db + 65 db).

IME close to being a real world example. Most adult audiences are
comfortable when the louder passages (but no necessarily the peaks) run
in the 90-95 dB range.

30 dB is a very quiet concert room, but put an audience into it, and
just their breathing and body noises will add another 10-25 dB. If the
room is empty, its the middle of the night in a quiet part of town, the
HVAC is turned off, and the streets are free of traffic and the air is
free of aid traffic, lower numbers such as 20 dB might be logged.

OTOH, your living room is probably running more like 40 dB at most
times.

Post by ralphpnj
The maximum dynamic range of a 16 bit file is 96 db and that of 24 bit
is 144 db.

With dither, maybe 1-3 poorer than that.

The recording gear adds appreciable noise of its own. The best recording
mics have equivalent noise of just a few dB, but a typical recording mic
will have equivalent noise on the order of 18 dB. If minimal micing is
used, there are only from 2 to 6 mics active, but that will raise the
noise level by from 3 to 5 dB. If multiple micing is used then the
background noise due to the mics increases by 3 dB times the logarithm
base 2 of the number of mics at best. IOW add 3 dB for the second mic, 6
dB if there are 4 mics, 9 dB if there are 8 mics, etc. For low
frequency noises that are in phase for all the mics, the backgound noise
adds linearly as the number of mics increase. IOW, YMMV.

Post by ralphpnj
So how does placing the above symphony recording into a 24 bit file
versus a 16 bit file increase the dynamic range of the recording?

If the recording engineer uses a dynamic range expander or rides the
gain, then a recording's dynamic range can exceed that of the live
performance.

Post by ralphpnj
Okay so I realize that putting the above symphony recording into a 24
bit file does not actually increase the inherent dynamic range of the
recording and this the point I'm trying to make. To put it another way,
I don't believe that any listening tests, either double blind or
sighted, are even need to determine if this is true since it is just a
purely mathematical function - increasing the bit depth of a digital
audio recording cannot and does not increase the dynamic range of the
music contained in the digital audio file. I like to think of the whole
thing using an envelope analogy - the recording of the performance would
be a letter on an 8 1/2" X 11" piece of paper with the 16 bit file being
a 9" X 12" envelope and the 24 bit file being an 11" X 14" envelope -
the letter nicely fits into both envelopes however the size of the
letter remains the same.
Am I missing something or is the whole 16 bit versus 24 bit for a fully
edited and mixed recording just another completely false marketing
claim?

It is very rare case where doing a good job of converting a 24 bit
recording to 16 bits makes an audible difference. This article
http://www.avsforum.com/forum/91-audio-theory-setup-chat/2370801-test-your-ability-hear-high-res-audio.html
shows tests you can do yourself to verify this:


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[Mike Sargent]

Post by arnyk
This article
http://www.avsforum.com/forum/91-audio-theory-setup-chat/2370801-test-your-ability-hear-high-res-audio.html

I had downloaded those files and discovered that it just doesn't matter.
I don't *ever* listen at the sorts of volume that would be necessary to
hear these differences. I'd be deaf in days. Right now I'm listening to
Joss Stone (sounds great) and my SPL meter says I'm only listening at
around 75 dB. So that 96 dB (or 110 or whatever) dynamic range of CD
(ripped to FLAC in this case) is unimportant because I'm never going to
listen to it loud enough that it would matter.

These days I only get high def audio if it's cheaper than the CD (yes,
that does happen), or it's multi-channel (which I can certainly hear and
love).

Mike


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[adamdea]

Just one comment on this- I'm not sure it's quite right to assume that
adding quantification noise at the same level as the noise floor of the
recording will have no effect. Surely it will double the noise floor.
Presumably the logical specification is that the added quantisation
noise should not perceptibly increase the noise floor of the recording.
I'm not sure how fare below the recording noise floor it would need to
be not to increase the total noise perceptibly any ideas 10dB?

Another point. It has always seemed to me that Bob Stuart, in his
well-known paper explaining why we might need hi rez audio, pretty much
proved why we didn't. He took the curve fro just perceptible sound at
various frequencies and then took the 16 bit noise floor assuming that
the stereo was amplified to play at 120dB peak. This shows that at that
level the quantisation noise is just perceptible in the most sensitive
zone.[but not if you noise shape- it is completely undetectable then.
And this of course ignores masking and the question of the level of your
amp's noise floor.] But it can quite easily be seen from the same graph
that if you are laying only a few dBs lower there is no problem. And I
know for a fact that I always listen at below that level - ergo........

So -"buy hi rez so that you don't run the risk of just hearing the
quantisation noise when playing a silent passage on a really good
recording on your hifi at full volume." I should have been in
advertising.


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[arnyk]

Post by adamdea
Just one comment on this- I'm not sure it's quite right to assume that
adding quantification noise at the same level as the noise floor of the
recording will have no effect. Surely it will double the noise floor.

I sense a misunderstanding of quantization (the correct term - not
quantification) noise. Quantization noise is not added in, rather it is
an inherent component of the digitized signal. Strictly speaking calling
it a noise is a questionable use of the word noise, as noise is usually
a random signal, while quantization noise is not random but rather 100%
periodic and predictable. Quantization error or quantization distortion
would be more accurate terms. At the most quantization error is pseudo
random since it is actually 100% periodic and predictable.

In the lexicon of electronics, it would probably be more logical to call
it a distortion. Calling it a noise is an artifact of the days when it
was not fully understood, and thought to be a noise. The convention
continues to be used. Everybody recognizes the word as an identifier
even though it is not a 100% accurate use of the terminology.

The peak value of quantization error is 1/2 of the LSB. Its average or
RMS values are proportionately less because RMS and average measurements
tend to average out the peaks.

If the signal is dithered, the total noise in the signal need not
increase, because the true function of dither is to overcome the
periodic nature of quantization error, and replace it with something
that is perceived by the human ear to be random.


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[adamdea]

Post by arnyk
I sense a misunderstanding of quantization (the correct term - not
quantification) noise. Quantization noise is not added in, rather it is
an inherent component of the digitized signal. Strictly speaking calling
it a noise is a questionable use of the word noise, as noise is usually
a random signal, while quantization noise is not random but rather 100%
periodic and predictable. Quantization error or quantization distortion
would be more accurate terms. At the most quantization error is pseudo
random since it is actually 100% periodic and predictable.
In the lexicon of electronics, it would probably be more logical to call
it a distortion. Calling it a noise is an artifact of the days when it
was not fully understood, and thought to be a noise. The convention
continues to be used. Everybody recognizes the word as an identifier
even though it is not a 100% accurate use of the terminology.
The peak value of quantization error is 1/2 of the LSB. Its average or
RMS values are proportionately less because RMS and average measurements
tend to average out the peaks.
If the signal is dithered, the total noise in the signal need not
increase, because the true function of dither is to overcome the
periodic nature of quantization error, and replace it with something
that is perceived by the human ear to be random.

My choice of words was a bit careless but the point remains. Whether
quantisation error is termed noise or distortion is neither here nor
there. The issue is what bit depth may capture -without degradation- a
recording with a given noise floor. Quantisation must add some
uncertainty. The better point which you might have made was that it may
not be necessary to dither given the analog noise which may make the
recording self dithering. Now whether that means that the total noise
power remains the same in the quanitsed and pre-quanatised signal
remains the question. I think it is increased and that the self
dithering merely makes the quantisation error noise rather than
distortion. S we add noise totalling -96dB (?) across the full spectrum
(slightyl better than if we had had to dither)

So the question remains- is that additional noise perceptible. I would
guess not, provided that it is 10dB or so below the existing noise
floor.


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[bakker_be]

Right now I'm listening to Joss Stone (sounds great) and my SPL meter
says I'm only listening at around 75 dB. So that 96 dB (or 110 or
whatever) dynamic range of CD (ripped to FLAC in this case) is
unimportant because I'm never going to listen to it loud enough that it
would matter.
Mike

I think this is the most important reply in this whole discussion: Who
among us, in a typical living room environment plays loud enough to have
the full dynamic range of said symphonic orchestra audible? In Belgium
music performances (live and recorded) are legally limited to
100dB(A)LAeq,60min, which means that even when the crowd is really
silent, on average only around 60dB remain for the music ...
QED: 16/44 is enough to faithfully reproduce a legal music performance
in Belgium :)



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[drmatt]

All fair points about capability of humans. But the capability of the
machines is much higher, so in my view why not master at 24 bit anyway?
The "massive" increase in data storage is utterly trivial actually and
if it helps one engineer avoid clipping or pull a badly levelled track
out of digital noise it's probably worth it.
Speaking as a photographer I love how my 8 bit per component jpgs look
on my screen, but I don't think twice about shooting in 14 bit per pixel
raw because the latitude you gain on post shooting exposure adjustment
is huge.




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[arnyk]

Post by drmatt
All fair points about capability of humans. But the capability of the
machines is much higher, so in my view why not master at 24 bit anyway?

To make the absurdity of this suggestion more clear, why not master at
48 bits? Unlike 24 bit mastering that triples the file size with no
audible benefit, this only doubles it.

Why not force people to drive 20 mph when freeways are capable of 60
mph? Why not pay $36 for $12 movie tickets?

Post by drmatt
The "massive" increase in data storage is utterly trivial actually

That's what I said, given that only a few tracks are being recorded.

Trouble is, a great deal of work is done with far more active tracks. In
professional recording and live sound, 24-32 channels is an average to
small console. That means that situations where 24-32 or more active
channels are in play. 32 tracks of 16 bits for recording a 1 hour
session is 81 gigabytes. Why not make it 3 times larger with no audible
benefit?

Post by drmatt
and if it helps one engineer avoid clipping or pull a badly levelled
track out of the noise it's probably worth it.

I suspect that some production houses with poorly trained and
inexperienced staff may go this route. Saying that you record everything
at 24/96 is actually a tacit admission of a lack of skill and/or
carelesness. As I pointed out in a recent post, there might be 24 dB of
overhead in a 16 bit recording. Not enough? Maybe a little more
attention to the job at hand might help.

Post by drmatt
Speaking as a photographer I love how my 8 bit per component jpgs look
on my screen, but I don't think twice about shooting in 14 bit per pixel
raw because the latitude you gain on post shooting exposure adjustment
is huge.

Equating photography with audio is well knokwn sign of not understanding
the vast differences between audio and video media and technology.

I mix, record, and edit both, and I know that the wise man wears
different hats when he switches between those horses.


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[docbob]

Post by arnyk
To make the absurdity of this suggestion more clear, why not master at
48 bits?

Without discussing audibility at all, one can say about 24/96: it is a
standard format, audio software deals with it, and as drmatt points out
humans can sometimes make mistakes. If your amp is exactly 10ft from
your speakers, do you cut the speaker wire to exactly 10ft? Or do you
add a little?

Post by arnyk
Unlike 24 bit mastering that triples the file size with no audible
benefit, this only doubles it.Some humans make mistakes. "24 bit triples it, but 48 bit only doubles

it" is some wacky math. I think you meant 48 only -further- doubles it.
That's okay though, we're all human, we all make mistakes.

Post by arnyk
In professional recording and live sound, 24-32 channels is an average
to small console. That means that situations where 24-32 or more active
channels are in play. 32 tracks of 16 bits for recording a 1 hour
session is 81 gigabytes.Uh, 44100Hz * 2 bytes * 32 channels * 3600 sec / 10^9 = 10.16 GB per

hour. You are off by a factor of 8. Did we forget the difference between
bits and bytes? That's okay though, we're all human, we all make
mistakes.
So 10ish GB will easily fit on a $10 16GB USB3 stick for easy transport
or take up about 50cents worth of an external 2TB HD. Cheaper for
internal. For studios with 24-32 channel consoles, are these prices
excessive?


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[docbob]

To refresh your memory...

Post by arnyk
Uh, 44100Hz * 2 bytes * 32 channels * 3600 sec / 10^9 = 10.16 GB per
hour. You are off by a factor of 8. Did we forget the difference between
bits and bytes? That's okay though, we're all human, we all make
mistakes.

Where's the error?


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[arnyk]

Post by docbob
Without discussing audibility at all, one can say about 24/96: it is a
standard format,

There are now so many so-called standard formats that being a standard
format conveys very little benefit.

However, 44/16 has been THE standard format for over 30 years and and
48/16 (used in video) can duke it out to determine which is the most
widely used. In that contest, 24 bit and high sample rate formats are
barley a blip.

Post by docbob
audio software deals with it, and as drmatt points out humans can
sometimes make mistakes.

Transmitting 70 to 85 percent useless data is hardly justified by the
few mistakes that it would address. Setting levels is only one of the
many possible and even common kinds of errors that are commonly made
during audio production.

Post by docbob
If your amp is exactly 10ft from your speakers, do you cut the speaker
wire to exactly 10ft? Or do you add a little?

Straw man, off topic, and ludicrous argument. Ignores the fact that 16
bit audio was chosen at a time when it was known that 13-14 bits would
suffice. 16 bits were chosen because they coincided with the prevailing
computer technology of the day, not because it was some kind of minimum
based on the requirements of audio recording or human hearing. Both the
sample rate and sample size of CD format audio are overkill. Your
speaker wire is already 15 eet long.

Post by docbob
Some humans make mistakes. "24 bit triples it, but 48 bit only doubles
it" is some wacky math.

It makes perfect sense if you realize that the word *it *refers to the
previous format.

Post by docbob
I think you meant 48 only -further- doubles it.

Glad you finally figured that out.

Post by docbob
That's okay though, we're all human, we all make mistakes.

After making a few mistakes of your own, you finally get to mine:

Uh, 44100Hz * 2 bytes * 32 channels * 3600 sec / 10^9 = 10.16 GB per
hour. You are off by a factor of 8. Did we forget the difference between
bits and bytes? That's okay though, we're all human, we all make
mistakes.


You seem to be outdistancing me quickly in terms of rapidly making many
mistakes.


So 10ish GB will easily fit on a $10 16GB USB3 stick for easy transport
or take up about 50cents worth of an external 2TB HD. Cheaper for
internal. For studios with 24-32 channel consoles, are these prices
excessive?


I guess you didn't understand the two times that I said that space was not
the issue, the problem was processing time. Just another mistake... You
are now even further ahead of me! ;-)

I guess making this mistake suggests that you've never ever tried to copy
like 10 GB of data between two hard drives or a hard drive and a flash
drive, or a SSD and a flash drive or hard drive.
\
I'm giving you the benefit of the doubt, and calling it an ignorant
mistake, nothing malevolent.


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[ralphpnj]

So with all this back and forth I see that my original "question" (in
quotes because I never actually phrased it as a question) of whether or
not increasing the bit depth from 16 bit to 24 bit for a recording with
only 85db of dynamic range (as in the difference between the quietest
and the loudest passages of the audio on the recording) increases the
dynamic range of the recording. I very firmly believe that increasing
the bit depth has absolutely no effect on the dynamic range of the
recording and I stated is just like putting a letter in a bigger
envelope, i.e. same size letter just a bigger envelope.

To further elaborate, I feel that 24 bit is being used as a pure
marketing term and offers no real benefit to the end user aka consumer.



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[Apesbrain]

This thread has gotten a bit out there but I want to circle back to the
beginning because I'm a bit confused. Ralphpnj describes a live
orchestral concert recording scenario where he has 65dB of music on top
of 30dB of room noise and asks why anything more than 16-bits is needed
to render that 95dB performance. TerryS says if one has only 65dB of
music then all one needs is 11 bits. Arnyk seems to agree when he adds
that such a 16-bit recording would have 25-35dB of headroom. What is
the reality here?

Then there is the AES study to which TerryS linked which finds "that the
reproduction of music performances at natural levels requires the
ability to produce very loud sounds in the range of 120-129 dB." That
would lend credence to those who say 16 bits are not enough.


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[ralphpnj]

Post by Apesbrain
This thread has gotten a bit out there but I want to circle back to the
beginning because I'm a bit confused. Ralphpnj describes a live
orchestral concert recording scenario where he has 65dB of music on top
of 30dB of room noise and asks why any more than 16 bits are needed to
render that 95dB performance. TerryS says if one has only 65dB of music
then all one needs is 11 bits. Arnyk seems to agree when he adds that
such a 16-bit recording would have 25-35dB of headroom. What is the
reality here?
Then there is the AES study to which TerryS linked which finds "that the
reproduction of music performances at natural levels requires the
ability to produce very loud sounds in the range of 120-129 dB." That
would lend credence to those who say 16 bits are not enough.

I'm with you there - lots of confusion and lots of smoke and mirrors.
Somewhere lies the truth - something that is more often than not ignored
in high end audio.



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[TerryS]

It gets hard to separate facts from opinion. Some things are easy... If
you assume a certain dynamic range for the recording (like the 65 dB we
started the discussion with), then the resolution required is just
straight math and in this case 11 bits is sufficient. 20*log(2^11)= 66
dB.
And you are correct that encoding it with any more bits than that is a
waste. Your analogy of the letter in the envelope is spot on.

But things diverge when you start talking about opinions.
How much extra headroom should you allow for mistakes in setting record
levels?
What is really the dynamic range of the performance in the environment?

Do you want 'Just good enough', or some extra margin? How much margin?

I personally like the 24/96k 'format' just because I don't want the
transport mechanism to be what sets the performance limit. I agree that
it is overkill. But when has high end audio ever been about 'good
enough'.
Those that say the 16/44k is good enough are absolutely correct. And
their point that the extra bits are probably wasted is also correct.
Probably every recording I own could be adequately captured at 16/44k,
but I keep hoping that someday there will be one that has more dynamic
range than that. I'm an audiophile.

Terry


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[ralphpnj]

Post by TerryS
It gets hard to separate facts from opinion. Some things are easy... If
you assume a certain dynamic range for the recording (like the 65 dB we
started the discussion with), then the resolution required is just
straight math and in this case 11 bits is sufficient. 20*log(2^11)= 66
dB.
And you are correct that encoding it with any more bits than that is a
waste. Your analogy of the letter in the envelope is spot on.
But things diverge when you start talking about opinions.
How much extra headroom should you allow for mistakes in setting record
levels?
What is really the dynamic range of the performance in the environment?
Do you want 'Just good enough', or some extra margin? How much margin?
I personally like the 24/96k 'format' just because I don't want the
transport mechanism to be what sets the performance limit. I agree that
it is overkill. But when has high end audio ever been about 'good
enough'.
Those that say the 16/44k is good enough are absolutely correct. And
their point that the extra bits are probably wasted is also correct.
Probably every recording I own could be adequately captured at 16/44k,
but I keep hoping that someday there will be one that has more dynamic
range than that. I'm an audiophile.
Terry

Thanks for answering my question.

Now onto your points about opinions. All of these points have more to do
with the process used to initially record the music and are useful in
that context. What I'm trying to get are there any recordings where the
24 bit version being sold the the consumer has an actual dynamic range
that is GREATER than the 16 bit version.

Taking a step back into analog recordings, most of which were/are made
using many tracks but which are then edited and mastered down to two
channel stereo for sale to the consumer, I think that a good analogy,
although not an exact analogy, (yes, I do love analogies) with the
current high resolution craze would be for the multi-track recording
masters to be offered for sale to the consumer. In other words, there is
a big difference between the needs of a recording team (artist, engineer
and producer) and that of the end user. So while 24 bit may make sense
when used during the recording process, 24 bit makes little to no
difference to the end user of the finished (fully mixed and mastered)
recording. And to be totally blunt: I don't think that are any fully
mixed and mastered recordings where 16 bits does not provide the full
dynamic range of the musical event on the recording.



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[Julf]

Post by ralphpnj
Now onto your points about opinions. All of these points have more to do
with the process used to initially record the music and are useful in
that context. What I'm trying to get are there any recordings where the
24 bit version being sold the the consumer has an actual dynamic range
that is GREATER than the 16 bit version.

I have looked at quite a few, and so far I haven't come across one with
more than 16 bits of actual dynamic range. Anything beyond 12 bits or so
is very rare.



"To try to judge the real from the false will always be hard. In this
fast-growing art of 'high fidelity' the quackery will bear a solid gilt
edge that will fool many people" - Paul W Klipsch, 1953
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[arnyk]

Post by Julf
I have looked at quite a few, and so far I haven't come across one with
more than 16 bits of actual dynamic range. Anything beyond 12 bits or so
is very rare.

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[ralphpnj]

Post by Julf
I have looked at quite a few, and so far I haven't come across one with
more than 16 bits of actual dynamic range. Anything beyond 12 bits or so
is very rare.

Perhaps a better way to phrase this issue is:

Do the 16 bit versions of all the recordings currently being released in
a 24 bit versions have the -*actual*- dynamic range of the music
compressed to fit into the 16 bit container? And do the 24 bit versions
offer greater -*actual*- dynamic range?



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[arnyk]

Post by ralphpnj
Do the 16 bit versions of all the recordings currently being released in
a 24 bit versions have the -*actual*- dynamic range of the music
compressed to fit into the 16 bit container? And do the 24 bit versions
offer greater -*actual*- dynamic range?

I don't think that there is a living person who has gathered that kind
of evidence about *-all-*of the recordings that are being released in 24
bit versions.

Furthermore there is a possibility that the standard and high resolution
versions of any recording may have been mastered differently due to the
preferences of management. which means that the answer to the type of
question you are asking is about artistic choices, not the capabilities
of the respective technologies.


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[Julf]

Post by ralphpnj
Do the 16 bit versions of all the recordings currently being released in
a 24 bit versions have the -*actual*- dynamic range of the music
compressed to fit into the 16 bit container? And do the 24 bit versions
offer greater -*actual*- dynamic range?

There is not necessarily any compression going on. All I can tell is
that all the 24-bit recordings I have looked at would easily have fit
into 16 bits, and I think archimago came to similar conclusions.



"To try to judge the real from the false will always be hard. In this
fast-growing art of 'high fidelity' the quackery will bear a solid gilt
edge that will fool many people" - Paul W Klipsch, 1953
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[arnyk]

Post by Julf
There is not necessarily any compression going on. All I can tell is
that all the 24-bit recordings I have looked at would easily have fit
into 16 bits, and I think archimago came to similar conclusions.

FWIW, I have made similar investigations, and obtained similar results.


It has been repeated and explained on this and other threads in many
ways that finding a live performance whose dynamic range exceeds the
capabilities of the CD format is difficult.

So, there would rarely if ever be a need for any live performance
recorded in any format to have its dynamics compressed to be effectively
recorded on a CD. If it were compressed, it might be for artistic or
business reasons, not the limits of CD technology.


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[mlsstl]

Post by arnyk
It has been repeated and explained on this and other threads in many
ways that finding a live performance whose dynamic range exceeds the
capabilities of the CD format is difficult.

And, it wouldn't hurt to note (again) that many of the "we desperately
need high-res -- CD isn't good enough" advocates are simultaneously
quite enamored with the sound quality of LPs, which are lucky to have a
range of 60-ish dB. Of course that is typically accompanied by a magical
explanation as to why that limited range of a LP doesn't matter but the
50% greater range of a CD just doesn't cut it.


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[ralphpnj]

Post by arnyk
I don't think that there is a living person who has gathered that kind
of evidence about *-all-* of the recordings that are being released in
24 bit versions.
Furthermore there is a possibility that the standard and high resolution
versions of any recording may have been mastered differently due to the
preferences of management. which means that the answer to the type of
question you are asking is about artistic choices, not the capabilities
of the respective technologies.

Quite right which is why many people, like Archimago, suggest that for a
truer comparison between 16/44.1 and 24/88.2, 96 or 192 the high rez
version should be down sampled backdownto 16/44.1 to insure that the
same master is being used for comparison.

Post by arnyk
There is not necessarily any compression going on. All I can tell is
that all the 24-bit recordings I have looked at would easily have fit
into 16 bits, and I think archimago came to similar conclusions.
FWIW, I have made similar investigations, and obtained similar results.
It has been repeated and explained on this and other threads in many
ways that finding a live performance whose dynamic range exceeds the
capabilities of the CD format is difficult.
So, there would rarely if ever be a need for any live performance
recorded in any format to have its dynamics compressed to be effectively
recorded on a CD. If it were compressed, it might be for artistic or
business reasons, not the limits of CD technology.

Just as I thought.

Post by arnyk
And, it wouldn't hurt to note (again) that many of the "we desperately
need high-res -- CD isn't good enough" advocates are simultaneously
quite enamored with the sound quality of LPs, which are lucky to have a
range of 60-ish dB. Of course that is typically accompanied by a magical
explanation as to why that limited range of a LP doesn't matter but the
50% greater range of a CD just doesn't cut it.

I do love the way the high end audio gurus either use
numbers/measurements as a weapon to make their point or just ignore
numbers/measurements when they disprove their point. It's like having
your cake and eating it too!



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[TerryS]

Post by ralphpnj
Thanks for answering my question.
Now onto your points about opinions. All of these points have more to do
with the process used to initially record the music and are useful in
that context. What I'm trying to get are there any recordings where the
24 bit version being sold the the consumer has an actual dynamic range
that is GREATER than the 16 bit version.
Taking a step back into analog recordings, most of which were/are made
using many tracks but which are then edited and mastered down to two
channel stereo for sale to the consumer, I think that a good analogy,
although not an exact analogy, (yes, I do love analogies) with the
current high resolution craze would be for the multi-track recording
masters to be offered for sale to the consumer. In other words, there is
a big difference between the needs of a recording team (artist, engineer
and producer) and that of the end user. So while 24 bit may make sense
when used during the recording process, 24 bit makes little to no
difference to the end user of the finished (fully mixed and mastered)
recording. And to be totally blunt: I don't think that are any fully
mixed and mastered recordings where 16 bits does not provide the full
dynamic range of the musical event on the recording.

I have no idea if there are actually any existing recordings that would
benefit from the extra bits. It would require extraordinary care in
setting up the recording environment. It would have to have a very low
noise floor (should be easily doable in a studio, but what about a
concert venue?). It would also have extremely large peak levels (might
happen in a concert venue, but seems unlikely in a studio). So the
combination of low noise floor and large peaks seems difficult to
achieve. These are just my uneducated opinions. I know next to nothing
about recording.

The other issue is the mics and other equipment used in the recording
chain. The link I referenced in my first response talks about this in
some detail.

It may be like the 'chicken and egg' thing. The recording engineers may
be slow to try to achieve extraordinary dynamic range in the recordings
until more people have shown the desire to purchase the higher
resolution products. Again, a guess on my part.

In my opinion, Hi-Fidelity is the pursuit of the highest attainable
performance, at least to the limit of human hearing. If I 'settle' for
16/44k, then at what point in my pursuit of perfection does my source
material become the limiting factor in my reproduction chain. The link
I referenced makes a pretty good case that the current state of the art
is not yet sufficient to reproduce the full dynamic range the human ear
is capable of hearing. Still work to be done, and 16 bits falls short.
But I fully understand the argument that 16 bits is good enough given
the limitations many others have cited (environmental noise, limits to
the maximum allowable volume, equipment limitations, and even personal
preferences). I don't necessarily agree. These are not hard limits,
just current state of the art. They can be pushed further.

Terry


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[Julf]

Post by TerryS
It may be like the 'chicken and egg' thing. The recording engineers may
be slow to try to achieve extraordinary dynamic range in the recordings
until more people have shown the desire to purchase the higher
resolution products. Again, a guess on my part.

The irony is that recording engineers have been pretty constantly
*reducing* dynamic range of their recordings, as most listeners prefer a
compressed dynamic range.

Post by TerryS
If I 'settle' for 16/44k, then at what point in my pursuit of perfection
does my source material become the limiting factor in my reproduction
chain.

The source material (rather than the number of bits) is already the
limiting factor.

Post by TerryS
The link I referenced makes a pretty good case that the current state of
the art is not yet sufficient to reproduce the full dynamic range the
human ear is capable of hearing. Still work to be done, and 16 bits
falls short.

Not sure I agree, see arnyk's comments about the effects of dithering.

Post by TerryS
But I fully understand the argument that 16 bits is good enough given
the limitations many others have cited (environmental noise, limits to
the maximum allowable volume, equipment limitations, and even personal
preferences). I don't necessarily agree. These are not hard limits,
just current state of the art. They can be pushed further.

But a think to remember is that if I have access to the original
recording, I can tell you, unequivocally, if the source material "fits"
in a given number of bits without compression or limiting, or not. The
recording engineer can measure the noise floor, and determine what the
highest peak is in the material, and thus knows how many bits are
required. No guesswork needed. As I wrote, so far I haven't come across
source material that pushes the 16-bit limit once the gain has been
normalized.



"To try to judge the real from the false will always be hard. In this
fast-growing art of 'high fidelity' the quackery will bear a solid gilt
edge that will fool many people" - Paul W Klipsch, 1953
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[adamdea]

Post by TerryS
It gets hard to separate facts from opinion. Some things are easy... If
you assume a certain dynamic range for the recording (like the 65 dB we
started the discussion with), then the resolution required is just
straight math and in this case 11 bits is sufficient. 20*log(2^11)= 66
dB.
Terry

WE still keep ploughing with this (slight) fallacy. It is not enough to
say that the level of quantisation noise (shut up Arnie) is equal to
that of the recording in order to capture it. It surely has to be
necessary that the additional noise from quantisation will not
perceptibly increase the total noise. Adding the same power of noise
again =3dB increase which I think should be noticeable
http://www.ncbi.nlm.nih.gov/pmc/articles/PMC4335553

I would have thought you'd want an extra bit to be on the safe side.


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[TerryS]

Post by adamdea
WE still keep ploughing with this (slight) fallacy. It is not enough to
say that the level of quantisation noise (shut up Arnie) is equal to
that of the recording in order to capture it. It surely has to be
necessary that the additional noise from quantisation will not
perceptibly increase the total noise. Adding the same power of noise
again =3dB increase which I think should be noticeable
http://www.ncbi.nlm.nih.gov/pmc/articles/PMC4335553
I would have thought you'd want an extra bit to be on the safe side.

Agreed. I was trying to keep the discussion as simple as possible. An
extra bit would add 6 dB to the dynamic range. It would seem foolish
(to me) to push it to the last dB.

Terry


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[arnyk]

Post by adamdea
WE still keep ploughing with this (slight) fallacy. It is not enough to
say that the level of quantisation noise (shut up Arnie) is equal to
that of the recording in order to capture it.

Attack me as a person as you will. It still won't make the myths you
seem to believe in, and your utter misunderstanding of the topic at
hand, relevant or true.

I never said: "That the level of quantisation noise is equal to that of
the recording in order to capture it." Given the insulting tone of your
post it is hard to say if you are intentionally publishing untruths, or
simply innocently unaware of how digital audio technology works. In any
case your post and the relevant facts about digital audio have nothing
to do with each other.

A key sentence: "Difficulty hearing speech in background noise—a poor
SNR—often leads to dissatisfaction with hearing-assistance devices." In
fact speech and music in modern recordings that meet the usual
professional standards is at a far higher level than that of any noise
in the recording as compared to that related to the use of hearing
assistance devices, which may higher than that of any speech or music.

Post by adamdea
It surely has to be necessary that the additional noise from
quantisation will not perceptibly increase the total noise. Adding the
same power of noise again =3dB increase which I think should be
noticeable
http://www.ncbi.nlm.nih.gov/pmc/articles/PMC4335553

I suggest that one and all read this article and see that in fact it has
nothing at to do with quantization noise. It is about the effects of
natural ambient noise such as crowd noise, on people's ability to hear
natural sounds such as speech and music in situations where the actual
SNR is very poor. In the situations addressed by the cited paper, the
voice and music may even be at a lower level than the ambient noise.

Post by adamdea
I would have thought you'd want an extra bit to be on the safe side.

That's a very hand wavy claim, and one that seems to attempt to
supercede science with speculation.


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[adamdea]

Post by arnyk
Attack me as a person as you will. It still won't make the myths you
seem to believe in, and your utter misunderstanding of the topic at
hand, relevant or true.
I never said: "That the level of quantisation noise is equal to that of
the recording in order to capture it." Given the insulting tone of your
post it is hard to say if you are intentionally publishing untruths, or
simply innocently unaware of how digital audio technology works. In any
case your post and the relevant facts about digital audio have nothing
to do with each other.
A key sentence: "Difficulty hearing speech in background noise—a poor
SNR—often leads to dissatisfaction with hearing-assistance devices." In
fact speech and music in modern recordings that meet the usual
professional standards is at a far higher level than that of any noise
in the recording as compared to that related to the use of hearing
assistance devices, which may higher than that of any speech or music.
I suggest that one and all read this article and see that in fact it has
nothing at to do with quantization noise. It is about the effects of
natural ambient noise such as crowd noise, on people's ability to hear
natural sounds such as speech and music in situations where the actual
SNR is very poor. In the situations addressed by the cited paper, the
voice and music may even be at a lower level than the ambient noise.
That's a very hand wavy claim, and one that seems to attempt to
supercede science with speculation.

You're just being silly, go back and think about it again. Quantisation
will (unless undithered) always add noise. The issue is simply how much
additional noise is perceptible. No hand waving here. Adding -96dB of
noise to -90dB will raise the noise floor to -91. Is that a change? Yes,
is it perceptible? Who knows. Actually someone probably does. Not you.

My original post invited thought and information on the subject not
knee- jerk ranting. I was not suggesting I had produced a definitive
answer, merely establishing the incontrovertible point-that it is clear
that only once one has established the just-noticeable increase in
volume level can one be clear about the bit depth required to produce
transparent quantisation of a given signal.

That I'm not even arguing with any proposition in particular and you're
just arguing out of habit, like a drunken bore at a dinner party.




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[Julf]

merely establishing the incontrovertible point-that it is clear that
only once one has established the just-noticeable increase in volume
level can one be clear about the bit depth required to produce
transparent quantisation of a given signal.

Not quite sure what you are saying here. Are you talking about a
just-noticeable increase in noise level, or a just-noticeable change in
signal amplitude? The former might relate to the needed number of bits,
the latter less so.



"To try to judge the real from the false will always be hard. In this
fast-growing art of 'high fidelity' the quackery will bear a solid gilt
edge that will fool many people" - Paul W Klipsch, 1953
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[adamdea]

Post by Julf
Not quite sure what you are saying here. Are you talking about a
just-noticeable increase in noise level, or a just-noticeable change in
signal amplitude? The former might relate to the needed number of bits,
the latter less so.

Noise level, Julf. Provided that the half lsb is spectrally flat by
dithering of otherwise the only effect of quantisation is to add an
error equivalent to spectrally-flat noise. This is quantisation 101.
So the question is whether this noise will when added to the noise in
the signal pre-quantisation, produce a noticeable increase.
Incidentally if anyone thinks I'm going off piste here I can quote from
a basic text
http://www.dspguide.com/ch3/1.htm
"First we will look at the effects of quantization. Any one sample in
the digitized signal can have a maximum error of ±? LSB (Least
Significant Bit, jargon for the distance between adjacent quantization
levels). Figure (d) shows the quantization error for this particular
example, found by subtracting (b) from (c), with the appropriate
conversions. In other words, the digital output (c), is equivalent to
the continuous input (b), plus a quantization error (d). An important
feature of this analysis is that the quantization error appears very
much like random noise.

This sets the stage for an important model of quantization error. *In
most cases, quantization results in nothing more than the addition of a
specific amount of random noise to the signal. *The additive noise is
uniformly distributed between ±? LSB, has a mean of zero, and a standard
deviation of 1/&#8730;12 LSB (~0.29 LSB). For example, passing an analog
signal through an 8 bit digitizer adds an rms noise of: 0.29/256, or
about 1/900 of the full scale value. A 12 bit conversion adds a noise
of: 0.29/4096 &#8776; 1/14,000, while a 16 bit conversion adds:
0.29/65536 &#8776; 1/227,000. Since quantization error is a random
noise, the number of bits determines the precision of the data. For
example, you might make the statement: "We increased the precision of
the measurement from 8 to 12 bits."

This model is extremely powerful, because *the random noise generated by
quantization will simply add to whatever noise is already present in the
analog signal. *For example, imagine an analog signal with a maximum
amplitude of 1.0 volts, and a random noise of 1.0 millivolts rms.
Digitizing this signal to 8 bits results in 1.0 volts becoming digital
number 255, and 1.0 millivolts becoming 0.255 LSB. As discussed in the
last chapter, random noise signals are combined by adding their
variances. That is, the signals are added in quadrature: &#8730;(A2 +
B2) = C. The total noise on the digitized signal is therefore given by:
&#8730;(0.2552 + 0.292) = 0.386 LSB. This is an increase of about 50%
over the noise already in the analog signal. Digitizing this same signal
to 12 bits would produce virtually no increase in the noise, and nothing
would be lost due to quantization. When faced with the decision of how
many bits are needed in a system, ask two questions: (1) How much noise
is already present in the analog signal? (2) H*ow much noise can be
tolerated in the digital signa*l?"*

You will thereofre note that the point I posed in my first post remains,
unmolested by the imaginary correction by Arnyk, namely that one cannot
say that a channel has been perfectly captured just because its inherent
noise level is the same as the noise which will be added by dithering.
The only way of knowing that is by knowing the just noticeable level of
increase *of noise* in that channel. This will undoubtedly vary with the
level of the inherent noise and may be affected by whether the noise is
shaped or not, but let's assume it's spectrally flat.

I have had to go round the houses with this point but it's really not
that complicated. Now where is the psycho-acoustic evidence of the just
noticeable level of noise increase. I'm hoping that may trusty
introduction to the psychology of hearing might have it in. In the
absence of precisely determined levels I reckon that keeping the
increase in noise to 1db seems sensible, which implies 16 bit being good
to capture capture a self-dithering channel with a noise level of --90dB
(or -87dB if we feel we have to add tpdf dither). Whether we have the
evidence or not let me make this clear there is no privileged position
in adding quantisation noise at a level equal to the inherent noise in
the channel- this might be either above or below audibility givne that
it doubles the noise level. Imagine a channel with noise at -18db: I'm
pretty sure that quantising at 3 bits WILL audibly increase the noise in
the channel. At 16 bit level the quantisation noise is demonstrably
audible in really extreme cases (if playing at 120dB peak level- this
can be shown using fletcher munson and is what sparked the interest in
noise shaped dither) so reallly really strictly doubling it is probably
an audible difference, albeit probably not at ordinary listening
levels.

* NB the non-noise model of quantisation error is irrelevant in this
context because the issue is capturing an audio channel and this entails
that the signal is either self-dithering or dithered. There is no point
discussing incompetent digitisation.


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[arnyk]

Post by adamdea
Noise level, Julf. Provided that the half lsb is spectrally flat by
dithering of otherwise the only effect of quantisation is to add an
error equivalent to spectrally-flat noise. This is quantisation 101.

Modern audio ADCs (of sigma-delta design) generally noise shape the
quantization error, in effect turning it into dither without adding any
additional noise:

http://www.ti.com/lit/an/slyt423/slyt423.pdf page 16:

"Multi-order modulators shape the quantization noise to even higher
frequencies than do the lower-order modulators."

Post by adamdea
So the question is whether this noise will when added to the noise in
the signal pre-quantisation, produce a noticeable increase.

There is no such question that is relevant to the vast majority of audio
DACs because they use Sigma-Delta technology to turn the quantization
error signal into dither by shaping it in the time domain.

Post by adamdea
Incidentally if anyone thinks I'm going off piste here I can quote from
a basic text
http://www.dspguide.com/ch3/1.htm
"First we will look at the effects of quantization. Any one sample in
the digitized signal can have a maximum error of ±? LSB (Least
Significant Bit, jargon for the distance between adjacent quantization
levels). Figure (d) shows the quantization error for this particular
example, found by subtracting (b) from (c), with the appropriate
conversions. In other words, the digital output (c), is equivalent to
the continuous input (b), plus a quantization error (d). An important
feature of this analysis is that the quantization error appears very
much like random noise.

The cited text does not seem to be updated to describe the vast majority
of ADCs that are used in audio which are Sigma-Delta designs. This adds
more complexity, and if included right up front might make an
introductory text in DSP technology overly complex for newbies.

If one measures real world gear on a test bench, this is what one
typically finds.


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[adamdea]

Post by arnyk
Modern audio ADCs (of sigma-delta design) generally noise shape the
quantization error, in effect turning it into dither without adding any
"Multi-order modulators shape the quantization noise to even higher
frequencies than do the lower-order modulators."
There is no such question that is relevant to the vast majority of audio
ADCs because they use Sigma-Delta technology to turn the quantization
error signal into dither by shaping it in the time domain.
The cited text does not seem to be updated to describe the vast majority
of ADCs that are used in audio which are Sigma-Delta designs. This adds
more complexity, and if included right up front might make an
introductory text in DSP technology overly complex for newbies.
If one measures real world gear on a test bench, this is what one
typically finds.

Arny I donlt think the delta sigma thing is relevant if we are
discussing 16 bit quantisation. The adcs will produce a single or
multibit stream at several megaherz. Getting it to pcm is a different
story. Leaving aside gettign it to 24 bits of pcm, once it is in 24 bit
form (at which it will probably be mastered) it WILL be downsampled in
software to 16 bits. Delta sigma does not come into it. The downsampling
may or may not be done using noise shaping but is quantisation in the
sense described in the text. The point remains that there is nothing
particularly special about adding quantisation noise of the same power
as the existing noise in the channel.


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[arnyk]

Arny I don't think the delta sigma thing is relevant if we are
discussing 16 bit quantisation.

Please explain why I should believe this, as opposed to standard auio
engineering texts, formal classes and lab tests.

The ADCs will produce a single or multibit stream at several megaherz.

That is true for the internal operation of many Sigma Delta ADCs.
However, up until now you've been talking about older designs. Which do
you think you are talking about now, and what's our clue that you
changed topics?

Getting it to pcm is a different story. Leaving aside gettign it to 24
bits of pcm, once it is in 24 bit form (at which it will probably be
mastered) it WILL be downsampled in software to 16 bits.

Tht is true only if the distribution medium handles 24 bits. For
example, that excludes DSD but you don't seem to be able to qualify what
you say for the limited set circumstances that it applies to.

Delta sigma does not come into it.

I thought you were talking about ADCs, not mastering software. When did
you change that? If you are talking about ADCs then how Delta Sigma
works is very relevant because its how quantization error and dither are
handled.

The downsampling may or may not be done using noise shaping

That may be true of some mastering software, but most mastering software
emphasizes their options for noise shaping. Noise shaping during
mastering provides a means to produce the subjective equivalent of up to
120 dB dynamic range with 16 bit recordings. Most well informed persons
who are engaged in mastering are interested in advantages of that
nature.

If you critize 16 bit media while intentionally excluding well-known
effective technololgy such as noise shaping then you have seem to have
the makings of a straw man arugment.

but is quantisation in the sense described in the text.

The text is incomplete and does not mentioin more recent but generally
accepted technology. I hope that its authors are doing that to mislead
people, but rather are doing it to keep the story as simple as possible
for newbies.

The point remains that there is nothing particularly special about
adding quantisation noise of the same power as the existing noise in the
channel.

The above still ignores the fact that quantization noise is not
intentially added, but comes into the picture as an inherent
characteristic of digitization.

Using your own reference: http://www.dspguide.com/ch3/1.htm

"When isn't this model of quantization valid? Only when the quantization
error cannot be treated as random."

I.ve shown that most modern quality ADCs (e.g. Sigma Delta) for audio
do indeed treat quantization error as random.

It appears that you are cherry picking information from your reference
to bolster some erroneous thinking, and apparently ignoring parts of
your own reference that don't agree with your erroneous ideas about
quantization error.


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[Julf]

Post by adamdea
Noise level, Julf.

Thanks for the clarification. I am familiar with the theory, but wasn't
sure about what your "just-noticeable increase in volume level" was
referring to.



"To try to judge the real from the false will always be hard. In this
fast-growing art of 'high fidelity' the quackery will bear a solid gilt
edge that will fool many people" - Paul W Klipsch, 1953
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[arnyk]

Post by adamdea
You're just being silly, go back and think about it again. Quantisation
will (unless undithered) always add noise.

I'm happy to see that you are finally seeing the light, and finally
admit that quantization noise/distortion is not added separately but is
inherent in digitization. Of course claiming I said otherwise suggests
that someone doesn't understand the point I was trying to get across.

Post by adamdea
The issue is simply how much additional noise is perceptible

The audibility of a signal is highly dependent on the context (whatever
other signals are there at the same time, its amplitude, its spectral
content and whether or not it is random (IOW a true noise) or periodic
(IOW a true distortion) and thus deterministic.

The following paragraph seems to consider only amplitude, and therefore
it is in error.

Post by adamdea
No hand waving here. Adding -96dB of noise to -90dB will raise the noise
floor to -91. Is that a change? Yes, is it perceptible? Who knows.
Actually someone probably does. Not you.

Given the incomplete data given, it is impossible to estimate
audibility. Therefore I am proud to admit that I don't know whether it
is audible because I'm being insulted and hectored with incomplete
data. Anybody who really understands quantization noise, dither, and
audibility will do the same. The problem is the over-simplistically
stated question, not to mention the use of incorrect terminology.

Post by adamdea
My original post invited thought and information on the subject not
knee- jerk ranting.

Your original post and its sequels had just about everything wrong, as I
showed. They invited correction, which they obtained. These objections
suggest that correct information was not the goal.

So much for me trying to help people who can't be helped because they
know little, but seem to think they know it all. Many of them hide
behind pseudonyms like dr-this and dr-that which protects their
ignorance. They thus avoid accountability, and can tell what ever lies
and make whatever insults they wish. They can be as gratuitously
insulting as seems to be their preference, without any accountability.


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[Julf]

Post by Apesbrain
Then there is the AES study to which TerryS linked which finds "that the
reproduction of music performances at natural levels requires the
ability to produce very loud sounds in the range of 120-129 dB." That
would lend credence to those who say 16 bits are not enough.

If you assume the totally over-the-top 129 dB peak level, 16 bits still
gives you a range of 33-129 dB. Your listening room probably has a
background noise level above 33 dB, so 16 bits is still more than
enough.



"To try to judge the real from the false will always be hard. In this
fast-growing art of 'high fidelity' the quackery will bear a solid gilt
edge that will fool many people" - Paul W Klipsch, 1953
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[arnyk]

Post by Julf
If you assume the totally over-the-top 129 dB peak level, 16 bits still
gives you a range of 33-129 dB. Your listening room probably has a
background noise level above 33 dB, so 16 bits is still more than
enough.

When we discuss subjective dynamic range, we need to remember that audio
data well below the LSB is audible on properly dithered digital system.
Characterizing 16 bits as having 96 dB dynamic range does not include
this fact.

If you include this effect, then the subjective dynamic range of 16 bits
is well in excess of 96 dB.

In addition, dither shaped to exploit the characteristics of human
hearing can provide up to 120 dB effective subjective dynamic range.


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[Julf]

Post by arnyk
When we discuss subjective dynamic range, we need to remember that audio
data well below the LSB is audible on properly dithered digital system.
Characterizing 16 bits as having 96 dB dynamic range does not include
this fact.
If you include this effect, then the subjective dynamic range of 16 bits
is well in excess of 96 dB.
In addition, dither shaped to exploit the characteristics of human
hearing can provide up to 120 dB effective subjective dynamic range.

Agreed. My point is that even the undithered, simplistic 96 dB is more
than enough.



"To try to judge the real from the false will always be hard. In this
fast-growing art of 'high fidelity' the quackery will bear a solid gilt
edge that will fool many people" - Paul W Klipsch, 1953
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[arnyk]

And to reproduce that mythical 129 dB peak level mentioned with my 86
dB/1W sensitivity speakers, I would need 20 kW out of my amps...

The 129 dB peak number that I am familiar with comes from a number of
related papers by Fielder and Cohen such as :
http://www.zainea.com/Dynamic%20range.htm

This number was apparently observed at musical performances that
included "Electronic Augmentation" 20596.

The fallacy of these observations is that as many of us who have been
subjected to loud concerts with "Electronic Augmentation" IOW electric
guitars and PA systems for boosting the vocals, the electronics used in
concerts often has fairly acute and audible noise problems of its own.
If I had a nickel for every concert with "Electronic Augmentation" where
hum and noise from guitar amplifiers and hastily erected portable PA
systems are clearly audible throughout the concert...

So the fallacy of musical performances with high peak levels created
electronically but with no reported background noise is a centerpiece of
Fielder's landmark ca. 1980s and 1990s work.

Its instructive to see where this fallacy led the industry. The first
industry response to these alleged new dynamic range requirements was
known as HDCD, which was a clever scheme that encoded gain riding
controls into what is usually the bits that reproduce the noise floor.
We now know that while there was some industry and public acceptance of
HDCD, and despite its relatively simple implementation and potential to
become a mainstream augmentation to the audio CD, it never gained
critical mass and it is now largely consigned to long lost history.
FAIL! It was replaced in the marketplace by more elaborate schemes
such as DVD-A and SACD but again they failed to gain mainstream
acceptance and are now at best niche products. FAIL again!

It was apparent to many who did their homework and conducted DBTs and
found that so-called high resolution audio was not effective at
providing improved sound quality. These findings were first made at the
beginning of the new millennium and have never been overcome by tests
with comparable scientific rigor. Some are finding that any new music
format that hangs its hat on so-called high resolution audio is built on
anti-scientific wish fulfillment and will ultimately fail in the
marketplace.


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[mlsstl]

Post by arnyk
This number was apparently observed at musical performances that
included "Electronic Augmentation"...
The fallacy of these observations is that as many of us who have been
subjected to loud concerts with "Electronic Augmentation" IOW electric
guitars and PA systems for boosting the vocals, the electronics used in
concerts often has fairly acute and audible noise problems of its own.
If I had a nickel for every concert with "Electronic Augmentation" where
hum and noise from guitar amplifiers and hastily erected portable PA
systems are clearly audible throughout the concert...

Anyone who thinks a loud rock concert has a lot of "dynamic range" has
been inhaling way too much smoke at the event. It is simply loud. There
is nothing at the other end.


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[arnyk]

Post by mlsstl
Anyone who thinks a loud rock concert has a lot of "dynamic range" has
been inhaling way too much smoke at the event. It is simply loud. There
is nothing at the other end.

It turns out that the test data about peak SPLs from Fielder's 1985
paper http://www.aes.org/e-lib/browse.cfm?elib=3829 was about half taken
at concerts where Electronic Amplification was used and the loudest
instrument being played was a drum kit. IOW, a rock concert.

Here's the data from that paper:

20600

About 19 of the approximately 42 data samples were taken at performances
that relied on Electronic Augmentation. Two of those 19 Electronically
Augmented concerts were jazz while the remainder were rock or
miscellaneous. In short the paper's authors apparently wanted the
readers to consider rock concerts as some kind of high standard for
dynamic range.


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[cliveb]

Post by arnyk
If I had a nickel for every concert with "Electronic Augmentation" where
hum and noise from guitar amplifiers and hastily erected portable PA
systems are clearly audible throughout the concert....

That menacing hum and hiss before the gig starts is an integral part of
any self-respecting rock concert - it adds to the anticipation.
A guitar amp that doesn't hum and hiss should only be used for jazz.



Transporter -> ATC SCM100A
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[Wirrunna]

And to reproduce that mythical 129 dB peak level mentioned with my 86
dB/1W sensitivity speakers, I would need 20 kW out of my amps...

Here you go - http://sound.westhost.com/dynamic-range.htm



A camel is a racehorse designed by a committee.
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[docbob]

Post by Julf
Could we please keep the discussion factual instead of descending into
silly ad hominems?

First, ad hominems -can be- factual.
But I get your point that you want the thread to focus on the facts of
the topic, not fact or fiction about members. Your request would be so
much more powerful if you applied it evenly to both those you agree with
and those you disagree with. I believe keeping those -with whom you
agree- on the high road is more important, but that’s just me. And how

Post by Julf
Prove it.

I debated whether to post or PM drmatt or PM arnyk or just sit on the
“proof”. I don’t have links ready to go, but the right keywords will
lead anyone to 4 or 5 “biggies”. Rather than belabor this further, I’ll
let Arny decide how I should respond. Does he really want to relive this
"proof"?

Julf, I think it would be great if all members focussed on the
substance, the topics, the facts about audio and not the people, but
many members here attack audiophiles, placebophiles, cool-aid drinkers
and believers (in the wrong beliefs). If you would call on all members
uniformly to attack myths, wrong beliefs or fallacies, rather than the
holders thereof, this would be a better place.


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[drmatt]

Yup




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[docbob]

If you would call on all members uniformly to attack myths, wrong
beliefs or fallacies, rather than the holders thereof, this would be a
better place.
Point taken.

:-) I just realized that I offered to give my path (high road vs. low

Rather than belabor this further, I’ll let *someone else* decide how I
should respond. *Do we* really want to relive this "proof"?

Offer withdrawn.

I'm glad drmatt deleted his response to this. Let's all stick to the
topic.


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[arnyk]

I’ll let Arny decide how I should respond.

If you have something that is correct and relevant to say, bring it on.


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[Julf]

Post by Wirrunna
Here you go - http://sound.westhost.com/dynamic-range.htm

I love (and sorely miss) good old Wireless World!



"To try to judge the real from the false will always be hard. In this
fast-growing art of 'high fidelity' the quackery will bear a solid gilt
edge that will fool many people" - Paul W Klipsch, 1953
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[arnyk]

Post by Julf
I love (and sorely miss) good old Wireless World!

Unfortunately the project amp is said by the author to be capable of 500
watts into a 20 ohm load which falls well short of the 20 KW
requirement.

Many modern high powered amps are designed to drive very low impedance
loads and avoid the stated problems with high voltage signals by not
requiring them. Others are switchmode amps, which is a very different
technology.


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[arnyk]

Point taken.

Some people here confuse criticisms of incorrect thinking and beliefs
with personal attacks. The most erroneous of them here lately seem to
affect aliases that start out "Dr".


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[Julf]

Post by arnyk
Some people here confuse criticisms of incorrect thinking and beliefs
with personal attacks. The most erroneous of them here lately seem to
affect aliases that start out "Dr".

The first sentence is probably a statement of fact. The second sentence
is a personal attack / slur. Can we try to avoid the latter kind?



"To try to judge the real from the false will always be hard. In this
fast-growing art of 'high fidelity' the quackery will bear a solid gilt
edge that will fool many people" - Paul W Klipsch, 1953
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[arnyk]

Post by Julf
The first sentence is probably a statement of fact. The second sentence
is a personal attack / slur. Can we try to avoid the latter kind?

Saying that certain posters make copious errors of fact when that is
what they do is also a statement of fact.

Both of them also seem to entertain themselves by personally attacking
me. Then they try to engage well-meaning people as allies by complaining
that they are being personally attacked when all that is happening is
that the more egregious of their errors are being corrected. It is a
strategy that works.

I'm avoiding making true personal attacks the best I can, but they seem
to be successfully gaming some people.


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[drmatt]

I didn't say 24/96 just 24 for the extra latitude. 81GB for a master
file from a recording studio doesn't really sound very scary to me. I
have more capacity than that in my pocket.

Are we defining a defacto standard here or discussing possibilities?

I can't deny that talented folks don't need any help to get things
sounding good, but the talented folks who actually provide the music I
listen to have thus far provided a mixed bag of quality and poor 16/44
recordings. Perhaps some of them could have been better with a higher
res digital master?

Do please educate us poor souls on the vast difference between exposure
mapping from 14 bit raw Bayer array data to display for photography and
down mixing from 24 bit master data to 16 bit in an audio device, we're
all ears.




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[arnyk]

Post by drmatt
I didn't say 24/96 just 24 for the extra latitude. 81GB for a master
file from a recording studio doesn't really sound very scary to me. I
have more capacity than that in my pocket.

81 GB is the size of the file if it were 16/44. If it was 24/96 it would
be 3 times larger, and if it was 24/192 it would be 6 times larger. You
correctly observe that storage is very cheap these days. The problem
with large files is the time it takes to process them. For example, have
you ever loaded an 81 GB file onto a flash drive? How long does it
actually take?

Post by drmatt
Are we defining a defacto standard here or discussing possibilities?

The de facto standard is 16/44.

Post by drmatt
I can't deny that talented folks don't need any help to get things
sounding good, but the talented folks who actually provide the music I
listen to have thus far provided a mixed bag of quality and poor 16/44
recordings. Perhaps some of them could have been better with a higher
res digital master?

Thanks for admitting that you are just speculating, and don't really
know. People who go hands on with this stuff such as myself need not
speculate because we have the real world experiences. Of course, why
think that people who actually work with this stuff know anything about
how it works? ;-)

Post by drmatt
Do please educate us poor souls on the vast difference between exposure
mapping from 14 bit raw Bayer array data to display for photography and
down mixing from 24 bit master data to 16 bit in an audio device, we're
all ears. (Apart from the obvious completely different maths.)

This is an audio forum, not a video forum. If you want to go that far
off topic, please do so on your own time.


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[docbob]

81 GB is the size of the file if it were 16/44.81 GB is the size of what file? 32 channels for 8 hours? Is that

realistic?


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[drmatt]

81 GB is the size of what file? 32 channels for 8 hours? Is that
realistic?

Yes this is correct. Uncompressed.




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[docbob]

Post by drmatt
Yes this is correct. Uncompressed.

I meant: is 8 hours a realistic recording time to worry about file
manipulations (copying, etc)? I don't file size as a compelling reason
to avoid hi-res.


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[arnyk]

81 GB is the size of what file? 32 channels for 8 hours? Is that
realistic?

Incorrect. You might want to get up to speed with a thread (especially
one this short) before making mistakes like that. Please see post 8.


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[docbob]

Post by arnyk
Incorrect. You might want to get up to speed with a thread (especially
one this short) before making mistakes like that. Please see post 8.

I saw post 8 and answered in post 9 with my calculation for 1 hour
(=10.16 GB). Where's the error?


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[mlsstl]

Post by drmatt
I can't deny that talented folks don't need any help to get things
sounding good, but the talented folks who actually provide the music I
listen to have thus far provided a mixed bag of quality and poor 16/44
recordings. Perhaps some of them could have been better with a higher
res digital master?

I seriously doubt they would have been better with a high-res master. If
one CD quality file can sound good, they all can, and I have hundreds of
excellent recordings and many more that sound pretty good. When they
don't sound that way, the source of the poor quality is from the hands
of the engineers, producers and musicians.

Of course, often the sound is bad because that is exactly the way they
want it to sound. The "loudness wars" are completely due to intentional
choices made during the recording, mixing and mastering process. You
also have some people on the production end that just aren't as talented
as others -- no different from any other industry. A certain percentage
of the medical doctors out there graduated at the bottom of their class
yet still have a license.

I don't mind people working toward technical improvements; that's always
welcome. However, high-res is never, ever going to fix the quality
problems of recorded music because those deficiencies track back to
conscious or unconscious choices made by those involved in the recording
and production process.

That is one of the things I find so annoying about audiophiles. They
seem to think that everything they hear has a technical fix with a new
product or storage format -- often a subjectively magical one -- when in
reality most music sounds exactly the way the musicians, recording
engineers and producers want it to.

That's my 2 cents.


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[drmatt]

They'd definitely tell me it was off topic in a /video/ forum...

Anyway, yes I know how long it takes to copy 81 or 162 GB files around,
it's pretty tedious. Still a hell of a lot quicker than duplication of
master tapes I would imagine and getting quicker year on year.

You have real world experience. Good. Good for you. My real world
experience is that the tech is ready, the file size is not an issue,
/some/ people would benefit, and frankly if you want to stick with 16/44
don't you just sound like the "vinyl is good enough" brigade did twenty
years ago?




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