Please be aware that the article is not correct. Nichols does not understand digital audio and should not be writing technical articles for SOS.

If you want to learn about digital audio, I suggest you have a look a these articles instead:

http://www.lavryengineering.com/documents/Sampling_Theory.pdf

http://www.users.qwest.net/~volt42/cadenzarecording/DitherExplained.pdf

UnderTow

Hmmm...well, that was not very useful (other than the links). Do you have a specific quibble? I'm sure the technogeeks around here would appreciate your opinion.

P-X

--------------------
My Home Studio Build Thread

I've not read the article in question but I can spot faulty reasoning a mile off.

You have so far failed to provide:

1) Specific parts of the article with which you disagree; all you have said is "the article is not correct"
2) Evidence to support your assertion that "the article [or part of it] is not correct"
3) Evidence to support your assertion that "Nichols does not understand digital audio"
4) Reasons why Nichols "should not be writing technical articles for SOS"
5) Your real name and credentials

If you wish to start a sensible dialogue on this article (or anything for that matter) you first need to learn how to construct an argument - simply referencing other people's writing on the general topic of digital audio gives no credibility to your unsupported claims.

I agree with last comment.

UnderTow, you may be right but you should cite where 'Nichols' (what, not even worthy enough of full name?) is wrong to justify your argument.

You should really try to present a more balanced tone to gain support for your statement.

Regards to all,

The Producer.

Undertow knows hardly anything. He saw a book once and he heard his teacher say something about Nyquist theorem. Oh! He is only 12.

Edited by tex (26/04/06 09:13 PM)

I was shocked by the article and responded as such. Dinner got in the way of a proper explanation. Here it is:

Quote:

---

With most digital audio now being recorded at 24-bit, nobody thinks about the ultimate 16-bit destination and what that conversion does to the sound quality.

---

This comment is not supported by any data and is a bit of an insult to all the people that do think about it. He seems to be saying that he is the only person that has thought about this.

Quote:

---

The first impression is that you are just changing the noise floor from -144dB to -96dB, which isn't going to hurt the punchy high-level mix that you spent so much time on. But there is much more to this conversion than meets the ear.

---

This is incomplete to say the least. You could have a recording with a noise floor way above -96 dB FS. It won't suddenly go down to -144 dB FS just by converting to 24 bits.

Quote:

---

Just to refresh your memory, 24-bit audio gives you 256 times more resolution in the position of each sample on the waveform.

---

This isn't correct. It only gives more resolution to low-level signals. This can be easily demonstrated by converting a 16 bit file to a 24 bit file. The extra 8 bits are just filled with zeros. There is no requantization of the audio. A (loud) signal which lives in the upper bits of a 16 bit audio file does not gain any resolution by having extra zeros added in the extra bits.

Quote:

---

The first time I heard 24-bit, recording Bela Fleck and the Flecktones, I expected to hear more sheen and high-frequency clarity. I was incorrect. The most noticeable difference was in the low frequencies: the mouthwatering sound of the bass, the breathtaking realism of the kick drum, the clarity of vocals, and the low-end improvements that finally make the banjo worth recording. It took me a while to figure out why this was the case, but I finally got to the bottom of it.

---

This is anecdotal and isn't based on any facts. He seems to base his whole premise on "bad bass" in 16 bit audio on this recording.

Quote:

---

This 256 times higher resolution of a 24-bit sample is in effect everywhere on the waveform, from the lowest levels to the highest peaks. A sample point nearing 0dB full scale is 256 times more accurate than the same sample recorded at 16-bit.

---

This is incorrect. The resolution is only added at the lower end of the amplitude scale.

Quote:

---

To cut down on the confusion with bit sizes, let's use the size of the smallest bit in the 24-bit scale as a reference and call it a step. The difference between Sample A and Sample B in the 24-bit recording is 16 steps. The difference between the same samples in the 16-bit recording is 112 steps. That is 96 steps away from where it should have been — a 700 percent error in a low-frequency signal.

---

Not only can you not compare 24 and 16 bit signals like this (you need to to add dither during conversion to linearise the quantization steps) but the man can't even calculate percentages!

Quote:

---

The voltages generated in a converter are from small resistors that are trimmed by lasers during manufacturing. The small bits that are prone to error are different for every converter, even from the same manufacturer. This means that bit 23 on the converter on track 1 may be different than bit 23 on the converter on track 12.

---

Is he talking about segmented DACs? He does have this paragraph in his article:

Quote:

---

Current oversampling converters do not have the linearity and tracking problems that conventional converters have. Usually you are dealing with a 1-bit converter that samples at 256 times the sample rate. This type of converter just looks at the incoming signal and compares it to a reference voltage. The comparing circuit then decides whether the reference voltage should go up or down. This is done very fast many times between samples. Now the reference voltage that matched the incoming signal determines the resulting PCM value. Only one bit, and no linearity errors.

---

But nearly every soundcard you buy these days uses oversampling. I might be wrong but he seems to be looking at antiquated technology and basing a whole article on that. This just confuses things.

A few comments to the response:

Quote:

---

I've not read the article in question but I can spot faulty reasoning a mile off.

---

Not supporting my comment is not faulty reasoning. It's just lack of evidence.

Quote:

---

You have so far failed to provide:

1) Specific parts of the article with which you disagree; all you have said is "the article is not correct"
2) Evidence to support your assertion that "the article [or part of it] is not correct"
3) Evidence to support your assertion that "Nichols does not understand digital audio"
4) Reasons why Nichols "should not be writing technical articles for SOS"

---

See above.

Quote:

---

5) Your real name and credentials

---

This is totaly irrelevant. I prefer to let the facts talk rather than reputation. Actually I am against any form of reputation or lack thereof clouding the issue.

Quote:

---

If you wish to start a sensible dialogue on this article (or anything for that matter) you first need to learn how to construct an argument - simply referencing other people's writing on the general topic of digital audio gives no credibility to your unsupported claims.

---

Oh stop being so patronising. You have no idea if I can construct an argument or not as I didn't even attempt to argue! (Talking about faulty reasoning ...) Maybe you should have read the article before responding ...

Those articles are long and complex and it would be completely pointless and extremely time consuming to rewrite those articles in my own words just for this purpouse.

UnderTow

Quote tex:

---

Undertow knows hardly anything. He saw a book once and he heard his teacher say something about Nyquist theorem. Oh! He is only 12.

---

And another person jumping to conclusions without any facts ...

UnderTow

You made some decent technical points, but all we were saying was- maybe you should have prepared your backup arguments and made the post complete from the start rather than just starting a fire, fanning the flame a little, and walking away.

--------------------
Gear list: If you can't find it, grind it

Quote MadManDan:

---

You made some decent technical points, but all we were saying was- maybe you should have prepared your backup arguments and made the post complete from the start rather than just starting a fire, fanning the flame a little, and walking away.

---

Yes well, I was shocked. Anyway, the fire is still burning.

UnderTow

I'd like this thread brought to the attention of the article writer to clarrify the points. I'd hate to think the magazine was printing out inaccurate articles, it's also an interesting subject to be debated.

Carry on

Quote UnderTow:

---

Quote tex:

---

Undertow knows hardly anything. He saw a book once and he heard his teacher say something about Nyquist theorem. Oh! He is only 12.

---

And another person jumping to conclusions without any facts ...

UnderTow

---

Well I wasn't getting any facts from you at the time and I was interested to see the whole argument. Now I have thank you. I didn't say what 12.

--------------------
Success is round the corner. It's also round the bend.

wow...

interesting...

(i'm trying to figure out the correct percentage of that quoted figure)

(i'm a guitar player, it might take a while)

Dear UnderTow,

Your actual arguments are most valid. The article wavers between objective and subjective statements with plenty of inaccurate information. The editor must have had a couple of pages to fill to meet a deadline. Too many dependable staff writers away at shows. What?

Next time for brevity on your part simply pose your argument as a question in the first place i.e 'Nichols on 24 Bit,Wrong or Not?', point us towards the article and the sites for the real facts. Having thrown open the topic to investigation, argument, feedback & discussion here on what is a forum afterall we'd eventually come to our own decision which may concur with your's or not.

Regards to all (never intentionally patronising)
The Producer.

Quote The Producer:

---

Dear UnderTow,

Your actual arguments are most valid. The article wavers between objective and subjective statements with plenty of inaccurate information. The editor must have had a couple of pages to fill to meet a deadline. Too many dependable staff writers away at shows. What?

---

Apparantly Nichols has a monthly blurb in SOS since one or two issues. I found the previous one questionable at best. (Something about harddrives loosing their magnetization if you leave them on a shelf for three months...)

I remain with the notion that Nichols does not have the technical knowledge and understanding to write technical articles for SOS.

Quote:

---

Next time ...

---

Lets hope there isn't a next time ...

Quote:

---

for brevity on your part simply pose your argument as a question in the first place

---

I would if it was a question.

I know my approach was probably counter productive. On the other hand, it might convey my shock and dismay at reading such an article in a magazine I respect.

Quote:

---

Regards to all (never intentionally patronising)
The Producer.

---

Thanks for confirming my technical arguments.

An article about the merits of 24 bits vs 16 bits recording and not a single time is the word dither mentioned. I'm still amazed ...

UnderTow

What an odd rant. So Mr. Nichols (let's be polite, please) thinks long and hard, and comes up with a good, logically consistent theory about bass sounds with 24 bit. And writes an interesting and brain-engaging article.

It may not fit with some learned 'facts' as learnt by an anonymous ranting poster.

'nuff said

G

--------------------
PA stuff on FB

Quote Guy Johnson:

---

What an odd rant. So Mr. Nichols (let's be polite, please) thinks long and hard,

---

That is his job when writting such articles. Unfortunately he forgot one important aspect of his job: checking his facts.

Quote:

---

and comes up with a good, logically consistent theory about bass sounds with 24 bit.

---

He did nothing of the kind.

Quote:

---

And writes an interesting and brain-engaging article.

---

The part of my brain most engaged after reading the article was my amygdala.

Quote:

---

It may not fit with some learned 'facts' as learnt by an anonymous ranting poster.

---

Read those articles in the links I posted.

Quote:

---

'nuff said
G

---

That is the whole problem. Readers of SOS presumably trust SOS to bring them informative articles based on correct technical data. When they fail to do this for whatever reason, they fail as an informative source about Music Recording, and thus they fail at their primary purpouse.

You seem to be blindly believing the article simply because it is printed in SOS which prooves my point and confirms my reasons for concern.

Seriously, read the articles I linked in my first post. That should dispel any ideas that the article in question is technicaly correct.

UnderTow

i take issue with the belief that the extra resolution only applies to low-level signals, this is surely not correct. there are 256 times as many integers available between -FSD and +FSD, so the resolution is available to all signals.

it is of particular value to low-level signals, yes, but is available across the whole range. what an odd argument.

--------------------
planet nine
lincoln, uk.

Quote Nathan:

---

i take issue with the belief that the extra resolution only applies to low-level signals, this is surely not correct. there are 256 times as many integers available between -FSD and +FSD, so the resolution is available to all signals.

---

There are indeed 16777216 values in 24 bit instead of the 65536 values of 16 bit but the extra 16711680 values are all packed down at the bottom of the amplitude scale.

Check out this image of a sine wave: http://home.casema.nl/ajohnston/Bits.png

Look at the dB scale on the right of the image. Now look at the same wave heavily zoomed in verticaly: http://home.casema.nl/ajohnston/Bits-Zoomed.png

All those extra 16711680 values are packed arround the red center line between the green lines that represent -96 dB. They are not evenly distributed over the whole range.

If these extra values were distributed over the whole range, you would need to requantize when converting a 16 bit file to a 24 bit file when in reality the 8 extra bits added are just filled with zeros.

There is even a small section of the Nichols article that supports this:

Quote:

---

Each bit of a 24-bit sample has a different voltage value assigned to it. The voltage value of each bit is supposed to be exactly half the value of the bit above it.

---

So each bit you add to the bit depth of a digital word has an equivalent voltage value of half that of the previous least significant bit.

You can actually test this yourself: Take a 16 bit wave file. Any file will do. Convert it to 24 bits and then convert it back to 16 bits without doing any kind of processing (including any gain change or adding any dither!). The resulting file is 100% identical to the file you started with. Every sample has the exact same value. You can do this as many times as you want. It won't affect the signal in any way whatsoever.

Does this explain things or does it add more confusion?


UnderTow

I'm gonna wait for Hugh to wake up so I can start reading this thread properly.

Night

Quote Music Manic:

---

I'm gonna wait for Hugh to wake up so I can start reading this thread properly.

---

Hugh's gonna wake up to a headache.

Although, I do have to admit that upon reading the article I am inclined to share Under Tow's concerns about Mr Nicols reasoning. Subjects/principles such as dither, Nyquist, and reconstruction seem not to apply...

Mike

--------------------
www.ehsound.co.uk - Live Sound Hire & Services

i've got a headace already...

(still working on that percentage thingy)

har har

Quote UnderTow:

---

There are indeed 16777216 values in 24 bit instead of the 65536 values of 16 bit but the extra 16711680 values are all packed down at the bottom of the amplitude scale.

Check out this image of a sine wave: http://home.casema.nl/ajohnston/Bits.png

Look at the dB scale on the right of the image. Now look at the same wave heavily zoomed in verticaly: http://home.casema.nl/ajohnston/Bits-Zoomed.png

All those extra 16711680 values are packed arround the red center line between the green lines that represent -96 dB. They are not evenly distributed over the whole range.

If these extra values were distributed over the whole range, you would need to requantize when converting a 16 bit file to a 24 bit file when in reality the 8 extra bits added are just filled with zeros.

UnderTow

---

There's a fundemental misunderstanding of digital audio right here. Back to basics:

Lets start with a 2 bit signal (easier), assuming full scale is 1 volt (and ignoring signing for the moment):

00 - 0v
01 - 0.33v
10 - 0.67v
11 - 1v

Add another bit:

000 - 0v
001 - 0.14v
010 - 0.29v
011 - 0.43v
100 - 0.57v
101 - 0.71v
110 - 0.86v
111 - 1v

The extra bit doesn't just add extra resolution at the bottom end of the scale - it adds it throughout. Just because the extra zeros are added to the least significant end of the data doesn't mean that it just represents the quietest part of the converted signal.

I really wouldn't put much weight on what screen grabs of cool edit show you - a decent book (not internet site) on digital audio will explain all this far better than I can. Failing that, I'm sure Hugh will soon dispatch this when he returns.

Sheriton

--------------------
There's nothing we can't face... Except for bunnies

the bit about each less significant bit being half of the value of the one preceding it is just binary. in the same way that each number to the right is a tenth of the value of the preceding one in denary (base10).

you could think of 16-bit as being to the nearest integer, whereas 24-bit uses a few decimal places (this is an analogy, note). the extra resolution is available whatever the number, but yes, it would be more significant for lower numbers because they are smaller. when you convert from 24 to 16-bit and back again, it's like adding those decimal places (or working to them) and then lopping them off again -the number doesn't change if you didn't rescale (they're still "zeros"), the problems arise when you lop off (truncate) the "fractional" bit that is not zero. (end of analogy)

don't forget that those graphs show a dB logarithmic ruler on what i think is a linear scale, hence the dB numbers are going to cram in once you get nearer the zero line. the digital number representing a sample is linear, bottom to top, it represents the voltage not the dB value -just with more resolution (accuracy if you like) if it's a 24-bit sample.

--------------------
planet nine
lincoln, uk.

We've had a lot of comments about this article, and Roger is preparing a response or clarification for a future issue. I'll leave that to him, but a couple of points about UnderTow's claims:

Quote UnderTow:

---

Quote:

---

The first impression is that you are just changing the noise floor from -144dB to -96dB, which isn't going to hurt the punchy high-level mix that you spent so much time on. But there is much more to this conversion than meets the ear.

---

This is incomplete to say the least. You could have a recording with a noise floor way above -96 dB FS. It won't suddenly go down to -144 dB FS just by converting to 24 bits.

---

Roger Nichols is talking about converting from 24-bit to 16-bit. You seem to be talking about converting the other way.

Quote UnderTow:

---

Quote:

---

Just to refresh your memory, 24-bit audio gives you 256 times more resolution in the position of each sample on the waveform.

---

This isn't correct. It only gives more resolution to low-level signals. This can be easily demonstrated by converting a 16 bit file to a 24 bit file. The extra 8 bits are just filled with zeros. There is no requantization of the audio. A (loud) signal which lives in the upper bits of a 16 bit audio file does not gain any resolution by having extra zeros added in the extra bits.

---

I don't think he's claiming that converting a 16-bit recording to 24-bit will magically add resolution; he's simply stating that 24-bit recording is more accurate.

Quote UnderTow:

---

Quote:

---

The first time I heard 24-bit, recording Bela Fleck and the Flecktones, I expected to hear more sheen and high-frequency clarity. I was incorrect. The most noticeable difference was in the low frequencies: the mouthwatering sound of the bass, the breathtaking realism of the kick drum, the clarity of vocals, and the low-end improvements that finally make the banjo worth recording. It took me a while to figure out why this was the case, but I finally got to the bottom of it.

---

This is anecdotal and isn't based on any facts. He seems to base his whole premise on "bad bass" in 16 bit audio on this recording.

---

I think he is just using that record as an example of the kinds of difference he hears in 24-bit recording, not as the sole evidence for his theory. In any case, whether things sound better or worse is inevitably going to be subjective. What other 'fact' could there be about whether the bass sounded better to Roger Nichols on that particular recording?

Quote Sheriton:

---

Quote UnderTow:

---

the extra 16711680 values are all packed down at the bottom of the amplitude scale.

---

00 - 0v
01 - 0.33v
10 - 0.67v
11 - 1v

Add another bit:

000 - 0v
001 - 0.14v
010 - 0.29v
011 - 0.43v
100 - 0.57v
101 - 0.71v
110 - 0.86v
111 - 1v

The extra bit doesn't just add extra resolution at the bottom end of the scale - it adds it throughout. Just because the extra zeros are added to the least significant end of the data doesn't mean that it just represents the quietest part of the converted signal.

---

AFAIK… In the digital domain the interpretation of the bitpattern is:
- bit 1/MSB carries info up to 0dBFS
- bit 2 (that happens to be LSB in example 1) carries info about what happens up to -6dBFS
- bit 3 (that happens to be LSB in example 2) carries info about what happens up to -12dBFS

So, the added bit has added four new values at the bottom of the amplitude range. Logarithmic, not linear. Shouldn't it be so in the analouge domain too?

Lars

This is a clear cut case of having technical abilities, strengths and knowledge but absolutely no people, influencing or presentational skills.

Had the poster of this thread raised the subject in a adult, polite, fascinating way I would have been open to listen but instead I just picture some sad geek sitting with his mathematical formulas screaming "you're wrong you're wrong!".

There is absolutely no excuse for referring to a writer by their surname only - wherther you are a fan or a foe.


(Perhaps there are issues around the writer of this thread wanting to write for music magazines but being unsuccessful).

The key challenge around the accuracy of facts may have been correct but like any spoilt child that starts screaming, we're not so much interested in the noise coming out of their mouth.

So Mr Angry - why not shock the world and apologise for your brattish conduct, then perhaps we can start a sensible debate about 24bit to 16bit conversion ?

--------------------
What it says on the tin...

Interesting fracas! And hopefully everyone can keep cool so we can have an informative discussion which will leave us all older and wiser!

I am not a scientific genius about digital recording; I started life with tape recorders and razor blades. My judgments are based on does it sound good, and does it work reliably time after time without a lot of farting about with the technology getting in the way of making music.

But it seems to me that some, if not indeed much, of Under Tow's argument is based on converting a 16 bit signal to 24 and, lo and behold! - it doesn't sound any better. Well of course it doesn't. You can't magically put back something that has already been taken away, never to return.

However, if you put a really great mic through a great pre-amp into a great AD converter, then what you record will sound a lot better at 24 bit than at 16 bit. And your noise floor is much better, and multi-tracking e.g. drums or even a whole band is easier because you are less concerned about the risk of clipping, because you can track with a lot more headroom.

If you stay in 24-bit land for as long as possilbe (mixing, adding DSP effects, reverbs EQ etc), that is all going to sound better than at 16 bits. If you save "downgrading" to 16-bit till the latest possible stage in the chain (burning to CD), you will have a better recording.

This is what my ears tell me.

--------------------
Dynamite with a laser beam...

ok, i'm american so maybe there is a language barrier here...

but what is the big deal about calling the author Bloggs as opposed to Joe Bloggs?

from my limited education, i thought it was common to refer to authors by surname only?

perhaps this is from a different sector?

although you can say Joe Bloggs at first mention and then after just say Bloggs?

anyway, what's the big deal?

its the internet, dude...

lighten up...

i think its time for a group hug...

Well I just use the toys to make a noise... and I'm slowly getting to grips with some of the maths that's actually behind what we do, so I've found this thread very interesting, if a little heated !

For what it's worth, I still record at 16 bit, simply because-

1. I'm essentially quite lazy, and take a while to get 'up to date' unless my clients are complaining (and they're not...)

2. Given that most people seem to be listening on a lossy format anyway(mp3) I'm more concerned with the 'meat' of the tune- melody, chords, feel, etc, not the 'sound', as long as it's adequate- (tho' I do understand that having a 'better' recording in the first place might improve the final compressed result).

3. To my ears, my 16 bit productions sound just as 'good', and very often (to bang my own drum for once) a helluva lot better (sound-wise) to a lot of modern productions which I can only assume were recorded at 24 bit...


With all due respect, folks... storm in a teacup ?


tomafd

--------------------
http://anotherfineday.bandcamp.com/ http://anotherfineday.co.uk http://apollomusic.co.uk

Plenty of successful recording engineers have blind spots when it comes to technical details. But equally it appears that Undertow has a few blind spots too. Undertow - I would suggest you read and digest a decent text book on digital representation of analogue signals before starting a thread like this. There are some very fundamental flaws in your reasoning - the major one being that the increased resolution affects the whole dynamic range - not just the quiet parts.

Cheers

James.

--------------------
JRP Music - Audio Mastering and Restoration.
http://www.jrpmusic.net

Lars, the maths gets a bit offset here because so few bits are being used, the more bits you add, the closer binary1000... gets to 0.5 - and don't forget that the sample has to represent positive and negative excursions, so here B100 would be the zero line (tho I believe that for 16 bit ADCs the twos-complement method is used, where the MSB represents the polarity of the sample).

it depends what you mean by "cariies info up to", but bit 1 represents about halfway up the scale, hence -6dBFS, bit 2 6dB less. remember that bit3 is toggling (and therefore representing numbers) all the way thru this table, it might represent -24dB changes, but thats from 0 to full scale.

--------------------
planet nine
lincoln, uk.

UnderTow Quote:

---

Just to refresh your memory, 24-bit audio gives you 256 times more resolution in the position of each sample on the waveform.

---

Quote UnderTow:

---

This isn't correct. It only gives more resolution to low-level signals. This can be easily demonstrated by converting a 16 bit file to a 24 bit file. The extra 8 bits are just filled with zeros. There is no requantization of the audio. A (loud) signal which lives in the upper bits of a 16 bit audio file does not gain any resolution by having extra zeros added in the extra bits.

---

I don't particularly want to defend the author of the original article, but your contradiction here isn't strictly correct either. Quantising to 24 bits theoretically provides 256 times more resolution than quantising to 16 bits because the amplitude differences between quantising levels is smaller. So the statement as quoted is correct.

However, I also take your point that simply placing a 16 bit file into a 24 bit system will not change the quantising resolution of the original file (unless you also perform some processing on that signal, in which case the result will be output with true 24 bit resolution).

UnderTow Quote:

---

This 256 times higher resolution of a 24-bit sample is in effect everywhere on the waveform, from the lowest levels to the highest peaks. A sample point nearing 0dB full scale is 256 times more accurate than the same sample recorded at 16-bit.

---

Quote UnderTow:

---

This is incorrect. The resolution is only added at the lower end of the amplitude scale.

---

Again, not strictly true. Quantising with more bits reduces the level of the noise floor and allows quieter signal to be heard more easily, but the resolution is actually improved across the entire dynamic range.

Think of it this way. You have an analogue signalto quantise that runs between 0mV at its quietest and 1000mV at its loudest. If you quantise with 16 bits, you effectively have to draw a grid over the audio waveform with 65536 equally spaced horizontal lines, each line representating the level of a valid quantising value.

If you quantise with 24 bits, that grid has to contain 16,777,216 lines -- in the same space. We don't only add the extra 16,711,680 lines underneath the bottom of the 16 bit grid! So yes, the resolution across the entire dynamic range is improved. We can define the position of a sample peak with considerably greater resolution regardless of its particular amplitude. That is why the quantising error is smaller, and that's why the noise floor is lower.

Quote UnderTow:

---

But nearly every soundcard you buy these days uses oversampling. I might be wrong but he seems to be looking at antiquated technology and basing a whole article on that. This just confuses things.

---

He seems to be talking about very simple delta-sigma converters, and while he right about the principle advantages, the practical application is a little different and there are also some inherent problems. There's not such thing as a free lunch...

It certainly was a, shall we say, interesting article...

hugh

--------------------
Technical Editor, Sound On Sound

yes but
"The most noticeable difference was in the low frequencies: the mouthwatering sound of the bass, the breathtaking realism of the kick drum, the clarity of vocals, and the low-end improvements that finally make the banjo worth recording."

MR NICHOLS IS ACTUALLY RUSS ANDREWS, AND I CLAIM MY FIVE POUNDS

"mouthwatering bass" indeed

--------------------
~~~ A weasel hath not such a deal of spleen as you are tossed with! www.feline1.co.uk ~~~

And on a more serious topic by far, the technology has yet to be invented which would "finally make the banjo worth recording"

--------------------
Dynamite with a laser beam...

Quote UnderTow:

---

There are indeed 16777216 values in 24 bit instead of the 65536 values of 16 bit but the extra 16711680 values are all packed down at the bottom of the amplitude scale.

---

I haven't bothered to check out those sites, but on an amplitude scale, yes, it might appear that way. However, this is not strictly relevant to the quantising process.

What the quantiser is trying to do is define the amplitude of a sample. With a 16 bit quantiser the 65536 available scale points are spaced out equally across the full dynamic range. If you switch up to 24 bit quantising, you effectively add a further 256 new levels between each and every line onthe original 16 bit scale.

It's a bit like measuring the size of your living room for a new carpet. If you measure the room to the nearest metre and cut the carpet to that figure the measurement is pretty crude and you might end up with lots of carpet curling up over the skirting boards, or a big gap around the edge of the room! Measure the room to the nearest millimetre and cut the carpet accordingly, and the fit will be virtually perfect.

If you check out a tape measure, I don't think you'll find all the millimietre markings squeezed in under the first metre mark and none above it... They are evenely distributed along the entire length of the tape.

Quote UnderTow:

---

If these extra values were distributed over the whole range, you would need to requantize when converting a 16 bit file to a 24 bit file when in reality the 8 extra bits added are just filled with zeros.

---

Again, a misunderstanding on your part. Going back to my carpet analogy, I can quite happily work with figures in whole metres on a ruler marked in metres and millimetres without any trouble.... I don't have to do anything to the whole metres figure before I can use it... but it clearly doesn't make use of the precision available in a ruler with millimetre markings.

For example, we might have measured the carpet to the nearest metre as being 4 metres long. When measured to the nearest millimetre it might have been 4.236 metres.

The last number has the equivalent (for the sake of this argument) resolution as a 24 bit quantise, while the 4.000 figure is the equivalent of the 16 bit quantise. Notice that there are three zeros appended to metre-only figure when denoted with the same potential resolution as my millimetre scale.

Quote UnderTow:

---

There is even a small section of the Nichols article that supports this:

Quote:

---

Each bit of a 24-bit sample has a different voltage value assigned to it. The voltage value of each bit is supposed to be exactly half the value of the bit above it.

---

So each bit you add to the bit depth of a digital word has an equivalent voltage value of half that of the previous least significant bit.

---

Again, what Nichols says in this quote is technically correct. Your interpretation is wrong.

Each additional bit you add to the wordlength allows you to count twice as many quantising levels. Which means the space between each quatisation level is halved if you asdd an extra bit to the word length. Which means the accuracy of measuement has doubled, and the potential error is halved... which is why the noise floor (noise caused by random errors in the quantising measurement) halves (drops by 6dB) for each additional bit added to the wordlength.

hugh

--------------------
Technical Editor, Sound On Sound

QED?

Quote Hugh Robjohns:

---

It certainly was a, shall we say, interesting article...

hugh

---

I think misleading would be a better word. How you can discuss this subject without mentioning dither is beyong me. And surely the 'bass benefits more' idea is just plain wrong - the noise floor is reduced for all frequencies equally (until you get into shaping anyway).

--------------------
Midipicks - the all new MIDI Guitar forum...


Edited by cc. (27/04/06 10:40 AM)

yeah i think that other dude made the most valid point...

banjo? are you serious?

Quote:

---

ok, i'm american so maybe there is a language barrier here...

but what is the big deal about calling the author Bloggs as opposed to Joe Bloggs?

from my limited education, i thought it was common to refer to authors by surname only?

perhaps this is from a different sector?

although you can say Joe Bloggs at first mention and then after just say Bloggs?

anyway, what's the big deal?

its the internet, dude...

lighten up...

i think its time for a group hug...

---

Yo Gerard hey hows it hanging dude,

It's a question of common courtesy and hey here's always hoping that this will be one of the few forums that doesn't get completely Americanized (but that's a discussion for another day). Using just a surname could be done as an appropriate, good gestured reference but in this case it clearly wasn't, read the opening post again, he makes it personal even calling for the person to be sacked from their job!! - That's the point Dude / Homie.

I'm not defending the author of the SOS article or his views - I don't know him from Adam... It just pee's me off to see people who are just interested in starting a flaming session... all it takes is some manners and respect for people... And surely theres no distinction there between British and American communication.

Rant over..



--------------------
What it says on the tin...

Quote Lars Farm:

---

AFAIK… In the digital domain the interpretation of the bitpattern is:
- bit 1/MSB carries info up to 0dBFS
- bit 2 (that happens to be LSB in example 1) carries info about what happens up to -6dBFS
- bit 3 (that happens to be LSB in example 2) carries info about what happens up to -12dBFS

So, the added bit has added four new values at the bottom of the amplitude range. Logarithmic, not linear. Shouldn't it be so in the analouge domain too?

Lars

---

That would be a misunderstanding on how binary works. As most digital audio is stored in floating point values between 1 and -1 binary allows that number behind the decimal to be much more accurate, but in its most basic form binary describes any number in a set number of on/off signals.

Meaning in a 8 Bit/1 Byte number which can be up to 255, the first digit describes wether or not the number is at least 128, the second describes wether there is another 64 added onto that, the third describes wether there is another 32 added onto that etc. This allows numbers such as 129 to be possible by having a 1 in the first digit and last digit. So therefore it allows for more accurate depth description along the entire scale if you have a limited range, and larger number if dealing with integers(Well ever integer from 0 on up) with a non limited range.

So if most digital audio is dealing with floating point numbers between 1 and -1, increasing from 16 to 24 bits allows you to get the same dynamic range out of it overall, just much more accurate of one, which does allow for extended descriptions into the quieter range, but also extended descriptions between the limit of 65535 of a 16 bit number, and slightly quieter which in 16 bit would be 65534, but in 24 bit would be some number I cant think of right off hand that is 65534.999something. That can be a significant difference when dealing with a scale as large as the dB scale for our hearing as we hear a fairly large range of differences in sound.

But in other words, it doesnt just add the ability to hear quieter sounds, it allows for more accurate representation of what those quiet sounds actually are in terms of levels, allowing your noisefloor to shift from 0000000000000001 to 000000000000000000000001. This affects your entire scale if a signal is recorded in 24 bit allowing for much more 'depth' to your recordings. Still doesnt change the voltage value for the equivalent of every digit being 1 in a 16 bit or 24 bit recording, both of those will still be the exact same.

Hope that made sense at all.

Seablade

Not very good at explaining what goes on in his head;)

NOTE: Not all Digital audio deals in floating point numbers, it really depends on how a program was coded, but in general I find it better and it does seem to be a majority of software I deal with deals in it. It could easily be defined as a range between -32767 and +32767

NOTE 2: In many of my examples above to keep things simple I used unsigned bit numbers, however technically it owuld be a signed bit number as I noted, so one of those bits would be dedicated to identifying wether the signal was positive or negative. I cant remember right off hand which one, however it isnt exactly important unless you deal with the code on it;)

So let me get this straight: if I record in 24 bit Hugh will come round and fit my carpets for me?

--------------------
"Ignorance more frequently begets confidence than does knowledge" Charles Darwin.

Quote simonplent:

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So let me get this straight: if I record in 24 bit Hugh will come round and fit my carpets for me?

---

Thats pretty much the gist of it;)

Seablade

Quote:

---

So let me get this straight: if I record in 24 bit Hugh will come round and fit my carpets for me?

---

yes, but he'll probably put them on your wall -and then add some bass traps...


--------------------
planet nine
lincoln, uk.

Quote Nathan:

---

Quote:

---

So let me get this straight: if I record in 24 bit Hugh will come round and fit my carpets for me?

---

yes, but he'll probably put them on your wall -and then add some bass traps...

---

Don't forget - he won't be sticking the stuff on your wall -oh no - he wile be hanging them off a couple of hooks so that you don't end up with a sticky wall!

Len

--------------------
www.youtube.com/leonardngmusic

To get back on topic, am I going to trust the ears of a Grammy winner who has (IMHO) recorded some truly stunning work to a level I can only dream of, or some guy who comes along screaming that the math is questionable (and even then as Hugh has pointed out, not very accurately)?

Answers on a postcard, please.

Len

--------------------
www.youtube.com/leonardngmusic

Methinks the Grammy on the wall might be a killer argument...

--------------------
Dynamite with a laser beam...

Quote Len:

---

To get back on topic, am I going to trust the ears of a Grammy winner who has (IMHO) recorded some truly stunning work to a level I can only dream of, or some guy who comes along screaming that the math is questionable (and even then as Hugh has pointed out, not very accurately)?

Answers on a postcard, please.

---

Neither.

--------------------
Midipicks - the all new MIDI Guitar forum...

And I should point out that Hugh is completely wrong too. Carpets are discrete in nature (being composed of individual threads) and so should not be used in this discussion

--------------------
Midipicks - the all new MIDI Guitar forum...

Quote R2B: David Rogers:

---

There is absolutely no excuse for referring to a writer by their surname only - wherther you are a fan or a foe.

---

I don't think that the use of the surname only requires an excuse. It is common practice to refer to writers by their surname and does not imply any pro or anti viewpoint of their work.

--------------------
"Qui habet aures audiendi audiat"

Quote seablade:

---

That would be a misunderstanding on how binary works.

---

I was a bit hazy, but when we talk about 16-bit audio or 24 bit audio, we do talk about a model. An integer model where 16-bits can store 2^16-1 = 65535 discrete values. This model can be implemented by floating point, fixed point or integers as long as it behaves as the model, but the model is integers. So there are 16 and 24 bit resp.

I should have talked about precision and error instead. Would you agree that the error caused by the limited number of bits is in the least significant bit? Would you then agree that when you extend the representation with one bit the error due to limits in the representation is halved? The value you try to represent is the same, but the representation is done with greater precision. Wouldn't then errors from the last bit in a 16 bit representation be within the last 6dB of the 96 dB? If you add a bit to 17 bits, wouldn't the error then become smaller? 6dB smaller for every bit added? Isn't that what like the OP said?

Lars

There's is absolutely no substance in that statement.
Robbie Williams has one a few awards and I would trust quite a few pairs of ears before his, including my own.
This argument is more to do with the theory of it, not the sound.
People saying things sounds 'better' doesn't necessarily mean it IS technically better.
Remember that the human ear is a funny old thing and one man's Barbera Striesand is anoter man's Victoria Beckham.

Also, a decent producer who can record/produce good material and make it sound good doesn't have to know the mathematical theories of digital audio - not at all!

--------------------
Jack of all trades, master of some.

But... the *real* questions are:

"Which soundcard should I buy?"

"Which monitors for under $200?"

"Which is better for recording, a PC or a Mac?"


I do know that Mr. Nichols is well-respected in some circles, and not-so-well respected in others...just like the rest of us, eh? Go to some of the US forums like PSW or GearSlutz, and if you do a search for "Roger Nichols", you will find some very heated arguments over the credibility of Mr. Nichols.

Being in the position he is in, you would expect some scrutiny, but for some reason with Mr. Nichols, it is not a matter of "like vs. dis-like", but rather a matter of credibility. Most of what I have read has *nothing* to do with liking his style or his work, or whether folks agree with his methodology, etc., it seems to be centered on his credibility or lack therof.

Do a search for some other "big names" in the business, Terry Manning, Elloitt Schneiner, Chuck Ainley, Ross Hogarth, etc., and you will find differing opinions on the popularity of their styles, but you will not see many (if any?) credibility "issues"?

So now, I have to wonder: "Why Roger Nichols?" Seems a bit odd, eh?

Recently, he (or should I say his company, "Roger Nichols Digital") announced that they were releasing some Pro Tools plugins, and oh man... The flame wars began once again calling his qualifications into question!!! His company web page is HERE .

Look at the names of a couple of his plugins: "Bitchin-izer" and "Wendel-izer"... these names make me want to run out an buy them... NOT!!! What the hell is a Bitchin-izer anyway???

I ask again: "Why Roger Nichols?" He is obviously successful and has a decent track record with more Grammys than most of us have! So what is it with him? Maybe he tells it like he feels without treading down the politically correct lane? Maybe he speaks his mind and bucks the generally-accepted techniques???

Who knows??? I surely don't!

I only post this as a 3rd party observer, not to stir up the pot with some unsubstantiated rumors, but rather to point out that for some reason, he seems to be quite often associated with controversy. I urge everyone to form their own opinions, and a quick search on any of the above-mentioned forums will provide a lot of details.

Personally, I know nothing of the man other than to say that I like some of his work.

I must admit, I *was* rather "surprised" to see that he was now going to be publishing a regular SOS article considering his "controversial" status.

Actually, on second thought, maybe it is a good idea *because* of his controversial nature! Controversay sells copy!!!

Personal opinion: I found both of his SOS articles to be a bit "ho-hum" and a bit of a let-down.

--------------------
James
"Never interrupt your enemy when he is making a mistake" ~Napoleon Bonaparte~

QUOTE:"Personal opinion: I found both of his SOS articles to be a bit "ho-hum" and a bit of a let-down. "

Ho-humdy-ho...
Now that IS controversial.

--------------------
Jack of all trades, master of some.

Quote Doublehelix:

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Look at the names of a couple of his plugins: "Bitchin-izer"

What the hell is a Bitchin-izer anyway???

---

Clearly a pluging that makes your mixes 'Bitchin' regardless of your ability. Actually who needs ability, just use his auto mix first, then the bitchinator and your done.

I thought a "bitchin-izer" was what some of ran our posts through before submitting them to this forum

--------------------
~~~ A weasel hath not such a deal of spleen as you are tossed with! www.feline1.co.uk ~~~

Roger Nicholls is a weird one. Sometimes he comes across as a successful, talented and knowledgable guy, and other times he comes across as an opinionated amateur.

Who he *really* is is anybody's guess..

I guess his credibility issues stem from the fact he writes/reviews in many US mags, who are not as unbiased as some more reputable UK mages *ahem* and thus when he talks about gear (and to some extent, techniques) people are not quite sure whether he means what he says or he's doing the endorser/reviewer thing.

Edited by desmond (27/04/06 03:23 PM)

Quote desmond:

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Who he *really* is is anybody's guess..

---

Rosemary, the telephone operator?

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~~~ A weasel hath not such a deal of spleen as you are tossed with! www.feline1.co.uk ~~~

Quote cc.:

---

And surely the 'bass benefits more' idea is just plain wrong - the noise floor is reduced for all frequencies equally (until you get into shaping anyway).

---

I think psychoacoustics raise their head at this point. I think we would probably all agree that recording with 24 bit converters generally sounds a lot better than recording with 16 bit converters. There have been countless threads on this aspect before on the forum.

So if one person subjectively feels that the bass benefits in particular, who am I to argue. The fact remains that 24 bits does generally sound better than 16 bits....

The search for an explanation might be a bit wobbly... but there's nothing surprising in that. Very few people have a completely infallible understanding of this subject, myself included.

hugh

--------------------
Technical Editor, Sound On Sound

Quote feline1:

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I thought a "bitchin-izer" was what some of ran our posts through before submitting them to this forum

---

Personally I prefer 'Auto-Sneer' or 'Maxi-Sarc'.

Quote Sheriton:

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There's a fundemental misunderstanding of digital audio right here. Back to basics:

Lets start with a 2 bit signal (easier), assuming full scale is 1 volt (and ignoring signing for the moment):

00 - 0v
01 - 0.33v
10 - 0.67v
11 - 1v

Add another bit:

000 - 0v
001 - 0.14v
010 - 0.29v
011 - 0.43v
100 - 0.57v
101 - 0.71v
110 - 0.86v
111 - 1v

---

This is my understanding of things and if I am wrong please correct me:

You seem to be taking a big endian binary word and adding the Least Significant bit at the Most Significant end. You need to add that extra bit to the right, not to the left.

If what you say above is correct, when you convert a 16 bit file to a 24 bit file, the voltages of the equivalent sample points would change. We would have the following:


3 bits____________4 bits
000 - 0.00v_______0000 - 0v
001 - 0.14v_______0001 - 0.07v
010 - 0.29v_______0010 - 0.15v
011 - 0.43v_______0011 - 0.22v
100 - 0.57v_______0100 - 0.29v
101 - 0.71v_______0101 - 0.36v
110 - 0.86v_______0110 - 0.43v
111 - 1,00v_______0111 - 0.50v


(Sorry for the underscores. I couldn't get the formating right).

This would be a big problem. You wouldn't be able to convert a 16 bit file into a 24 bit file (without requantization) because on playback the volumes would have dropped by arround 48 dB.

In reality we have the following:


3 bits_____________4 bits
000 - 0.00v________0000 - 0.00v
--------____________0001 - 0.07v
001 - 0.14v________0010 - 0.14v
--------____________0011 - 0.21v
010 - 0.29v________0100 - 0.29v
--------____________0101 - 0.36v
011 - 0.43v________0110 - 0.43v
--------____________0111 - 0.50v
100 - 0.57v________1000 - 0.57v
--------____________1001 - 0.74v
101 - 0.71v________1010 - 0.71v
--------____________1011 - 0.78v
110 - 0.86v________1100 - 0.86v
--------____________1101 - 0.92v
111 - 1v___________1110 - 1.00v

Every word in the 3 bit word has an equivalent word with the exact same value in the 4 bit word. There are of course extra values in the 4 bit list.

So as I said, the extra bits are "only" giving extra amplitude resolution at the lower end of the amplitude resolution scale. Or put differently, the quantization errors are reduced in amplitude at the lowest resolution scale. The quantization errors are at a lower amplitude so the distortion produced by these quantization errors is lower.

I think the way I phrased my comments was what was confusing the issue. (I seem to have forgotten the word resolution at a few strategic locations. I guess that is what happens when posting at 10 to 4am after a 15 hour work day).

Quote:

---

The extra bit doesn't just add extra resolution at the bottom end of the scale - it adds it throughout.

---

Sorry if I wasn't clear. I am talking about the amplitude resolution scale. Not the full scale of the values themselves. In other words, I am mentaly plotting the resolution of a digital word against its bit size. I am jumping out of one way of thinking into another without making that completely clear.

Quote:

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Just because the extra zeros are added to the least significant end of the data doesn't mean that it just represents the quietest part of the converted signal.

---

No it indeed doesn't only represent the quietest part of the converted signal. It represents the lowest amplitude resolution scale of the sample values.

This does indeed affect the whole signal but, to my understanding, not in the way that Nichols[1] describes. Quantization errors at the lowest bits (at any bits actually) will be heard as distortion spread out over the whole bandwidth (and aliasing back into the audible band when bit reducing by truncation without application of dither as there are no anti-aliasing filters applied when bit reducing).

Quote:

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I really wouldn't put much weight on what screen grabs of cool edit show you

---

Actually it is very good compared to others. It shows the reconstructed waveform as it would look after going through the Digital to Analogue process unlike many other wave editors which just "join the dots". Also, because it is in a dB scale, it shows us things relative to how we hear them.

This is one fo the problems with Nichols'[1] article. He writes this:

Quote:

---

A sample point nearing 0dB full scale is 256 times more accurate than the same sample recorded at 16-bit.

---

Nichols[1] implies a logarithmic scale and then goes on to talk about a linear scale but this is incorrect relative to how humans hear things. We hear things on a logarithmic scale so the amplitude of errors and the increased resolution should also be calculated on a logarithimic scale. 24 bit audio is not 256 times more accurate to our ears.

Now, about my outburst. As I said before, I was shocked by reading the article. I was in a very stressfull session and escaped home to have dinner. Instead of happily relaxing while reading an SOS article, I encountered this article which is questionable to say the least.

UnderTow

[1] I'm treating Mr Roger Nichols' article as a technical article. In my book, in a technical discussion, refering to him as Mr Nichols would be a bigger insult than refering to him only by his sir name.

Quote Lars Farm:

---

This model can be implemented by floating point, fixed point or integers as long as it behaves as the model, but the model is integers. So there are 16 and 24 bit resp.

I should have talked about precision and error instead. Would you agree that the error caused by the limited number of bits is in the least significant bit?

---

To be perfectly honest I am not certain I understand the concept you are trying to explain to be able to answer, this shows more my experience of being largely self taught as a programmer and a realtive newcomer in the field of DSP programming;) Sorry. I probably have an opinion on it, I just dont understand the terminology used.

Quote:

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Would you then agree that when you extend the representation with one bit the error due to limits in the representation is halved?

---

I would agree with this. That is a funciton of binary and the fact it is a base 2 system, you get twice as precie with each added digit.

Quote:

---

The value you try to represent is the same, but the representation is done with greater precision. Wouldn't then errors from the last bit in a 16 bit representation be within the last 6dB of the 96 dB?

---

Now this is where I disagree, though it may come down to misunderstanding what you are trying to express.

If I am understanding you correctly, the shortcoming in this line of thought is the fact that it is not only that single digit that represents that range, but rather many digits(All of them for the largest section)

Quote:

---

If you add a bit to 17 bits, wouldn't the error then become smaller? 6dB smaller for every bit added? Isn't that what like the OP said?

Lars

---

I am not sure that I would quantify the error reduction in terms of dB in that regards. We already decided that it would be half the errors, so I suppose it would be correct, but mentioning it in terms of dB is in my opinion a bit misleading,, even if it is technically correct. Though in this instance I think it would be a difference of 3dB anyways.

The reason I wouldnt do this is I consider it a bit misleading in that it makes it seem that you are trying to say it merely adds that much onto it(Either making the signal louder, or capable of being quieter, the second is technically correct, but does not describe the entire process).

Of course as I said, it is possible we are trying to express the same opinion, and I am just misunderstanding you.

Seablade

Quote Lars Farm:

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Quote seablade:

---

That would be a misunderstanding on how binary works.

---

I should have talked about precision and error instead. Would you agree that the error caused by the limited number of bits is in the least significant bit? Would you then agree that when you extend the representation with one bit the error due to limits in the representation is halved? The value you try to represent is the same, but the representation is done with greater precision. Wouldn't then errors from the last bit in a 16 bit representation be within the last 6dB of the 96 dB? If you add a bit to 17 bits, wouldn't the error then become smaller? 6dB smaller for every bit added? Isn't that what like the OP said?

Lars

---

This is what I was driving at. We are talking about errors, precision and resolution. In these terms my comments were correct AFAIK.

Thanks for confirming that.

UnderTow

you obviously haven't filtered that post through the bitch-inizer...

-=sigh=-

Quote Sam Inglis:

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Roger Nichols is talking about converting from 24-bit to 16-bit. You seem to be talking about converting the other way.

---

Not really. I am using my examples to show the bit relationship between 16 and 24 bit recording formats. In my example you will see that you can convert back and forth at will. (Note: This is only applicable in this example. Under normal conditions the signal needs to be dithered when bit reduced).

Quote:

---

What other 'fact' could there be about whether the bass sounded better to Roger Nichols on that particular recording?

---

For one he could have mentioned what he is comparing to. Judging by the rest of the article he seems to be comparing modern 24 bit converters to early 3M digital machines (with a combination of a 12-bit converter with an additional four bits of an 8-bit converter for gain ranging). I could be wrong but as he isn't clear, I can't be sure.

So it would appear that the conclusion of his whole article is: Modern 24 bit converters sound better than 24 year old converters with 12 bit resolution and 4 bit gain scaling.

No [ ****** ] sherlock!

UnderTow

"This is incomplete to say the least. You could have a recording with a noise floor way above -96 dB FS. It won't suddenly go down to -144 dB FS just by converting to 24 bits.
" - Undertow

Undertow, you seem to be fixated with converting a 16 bit file to 24 bit, as demonstrated by your 3 bit - 4bit example and the quote above.
What is it that you are arguing? That 24 bits isn't more accurate than 16? If you do the crude convertion that you propose, you don't gain any resolution, but it wasn't audio recorded at 16 bits, and you haven't started processing it yet.
There was also comments mixing up quantisation noise (roughly no. bits * 6 dB), with the studio signal chain noise floor. They are not the same thing.

--------------------
www.wildhope.com - pike in a lilypond

Quote UnderTow:

---

This is my understanding of things and if I am wrong please correct me:

---

Er... in the nicest possible way, myself and others have already attempted to correct you...

Quote:

---

If what you say above is correct, when you convert a 16 bit file to a 24 bit file, the voltages of the equivalent sample points would change.

---

Quite. That is an error on his part in providing those particualr sample value examples. Clearly, sample values don't change when importing a 16 bit file into a 24 bit environment.

Quote:

---

In reality we have the following:

---

Yes, that table looks right (as far as I can see with the tricky formatting...)

Quote:

---

Every word in the 3 bit word has an equivalent word with the exact same value in the 4 bit word. There are of course extra values in the 4 bit list.

---

Correct.

Quote:

---

So as I said, the extra bits are :

---

nly" giving extra amplitude resolution at the lower end of the amplitude resolution scale.

---

No, I still can't agree. The extra amplitude resolution applies at all signal amplitudes. The amplitude of a sample can be defined with greater resolution whatever it's amplitude if you have more quantising levels (more bits in the wordlength).

Quote:

---

Or put differently, the quantization errors are reduced in amplitude at the lowest resolution scale.

---

No again. The quantisation errors are reduced in amplitude full stop.

Quote:

---

The quantization errors are at a lower amplitude so the distortion produced by these quantization errors is lower.

---

If the converter is correctly dithered, there won't be any distortion, just random noise -- but yes, the quantisation errors are smaller.

Quote:

---

In other words, I am mentaly plotting the resolution of a digital word against its bit size. I am jumping out of one way of thinking into another without making that completely clear.

---

"Resolution of a digital word in relation to its bit size"... huh? You are still not making yourself clear to me!

Quote:

---

Quantization errors at the lowest bits (at any bits actually) will be heard as distortion spread out over the whole bandwidth

---

No, if properly dithered there will be no distortion, just noise.

Quote:

---

We hear things on a logarithmic scale so the amplitude of errors and the increased resolution should also be calculated on a logarithimic scale. 24 bit audio is not 256 times more accurate to our ears.

---

You are correct in that a 24 bit system isn't 256 times more accurate to our ears than 16 bit system, and yes, our sense of hearing is broadly logarithmic.

However, the quantisation scale is linear, and the size of quantisation errors is linearly related to the quantisation resolution. In that context, I think it is fair to say that the amplitude of a sample nearing 0dBFS can theoretically be determined with 256 times greater precision in a 24 bit system than is possible in a 16 bit system.

Quote:

---

In my book, in a technical discussion, refering to him as Mr Nichols would be a bigger insult than refering to him only by his sir name.

---

I don't think he has been knighted...

Hugh

--------------------
Technical Editor, Sound On Sound

Quote Hugh Robjohns:

---

Quote:

---

This 256 times higher resolution of a 24-bit sample is in effect everywhere on the waveform, from the lowest levels to the highest peaks. A sample point nearing 0dB full scale is 256 times more accurate than the same sample recorded at 16-bit.

---

Quote UnderTow:

---

This is incorrect. The resolution is only added at the lower end of the amplitude scale.

---

Again, not strictly true. Quantising with more bits reduces the level of the noise floor and allows quieter signal to be heard more easily, but the resolution is actually improved across the entire dynamic range.

---

Indeed. Sorry for confusing things. The terminology I used was inadequate to convey my thoughts. I hope I have clarified things in my later posts.

Quote:

---

Think of it this way. You have an analogue signalto quantise that runs between 0mV at its quietest and 1000mV at its loudest. If you quantise with 16 bits, you effectively have to draw a grid over the audio waveform with 65536 equally spaced horizontal lines, each line representating the level of a valid quantising value.

If you quantise with 24 bits, that grid has to contain 16,777,216 lines -- in the same space. We don't only add the extra 16,711,680 lines underneath the bottom of the 16 bit grid! So yes, the resolution across the entire dynamic range is improved. We can define the position of a sample peak with considerably greater resolution regardless of its particular amplitude. That is why the quantising error is smaller, and that's why the noise floor is lower.

---

Agreed. Again, my terminology was wanting. The errors at lower bits are much smaller than errors at higher bits.

Quote:

---

It certainly was a, shall we say, interesting article...

hugh

---

I think this sums it up.

UnderTow

Quote UnderTow:

---

The errors at lower bits are much smaller than errors at higher bits.

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I think you are confusing quantisation errors with bit errors.

A quantisation error has the same maximum amplitude regradless of the amplitude of the audio signal being quantised.

A bit error would produce an erroneous sample amplitude, in which an error of the LSB would produce a far smaller amplitude error than an error in the MSB. But this particular issue is not relevant in any way to the topic being discussed.

Hugh

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Technical Editor, Sound On Sound

Quote Hugh Robjohns:

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If what you say above is correct, when you convert a 16 bit file to a 24 bit file, the voltages of the equivalent sample points would change.

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Quite. That is an error on his part in providing those particualr sample value examples. Clearly, sample values don't change when importing a 16 bit file into a 24 bit environment.

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Ok, we agree.

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In reality we have the following:

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Yes, that table looks right (as far as I can see with the tricky formatting...)

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Ok, we agree.

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Every word in the 3 bit word has an equivalent word with the exact same value in the 4 bit word. There are of course extra values in the 4 bit list.

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Correct.

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Ok, we agree.

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So as I said, the extra bits are :

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nly" giving extra amplitude resolution at the lower end of the amplitude resolution scale.

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No, I still can't agree. The extra amplitude resolution applies at all signal amplitudes. The amplitude of a sample can be defined with greater resolution whatever it's amplitude if you have more quantising levels (more bits in the wordlength).

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Arf! I am not talking about the amplitude scale nor am I talking about the amplitude resolution! I am talking about an amplitude resolution scale

Imagine the following plot: Y-axis = amplitude resolution of LSB. i.e. the amplitude precision of the LSB value. X-axis = number of bits.

The plot line starts on the left with Y=-6dB FS and X=1 bit. The next dot on the line is Y=-12dB FS X=2 bit, then Y=-18 dB FS and X=3 bits. At the end we have Y=-144 dB FS and X=24 bits.

So the line decreases. The amplitude resolution of the LSB of each digital word decreases as the word length increases.

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Or put differently, the quantization errors are reduced in amplitude at the lowest resolution scale.

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No again. The quantisation errors are reduced in amplitude full stop.

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Arf! By "at the lowest resolution scale" I mean at the least significant bits. As you go through the bits in a digital word, starting at the MSB and ending at the LSB, the amplitude of the error caused by flipping a particular bit decreases.

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The quantization errors are at a lower amplitude so the distortion produced by these quantization errors is lower.

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If the converter is correctly dithered, there won't be any distortion, just random noise -- but yes, the quantisation errors are smaller.

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One of my points about the article is that the word dither is not mentioned once. But I agree with what you say. The dither linearises the quantae.

But as I am talking about quantization errors, you could have given me the benefit of the doubt and assumed I was talking about inproper or absent dither.

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In other words, I am mentaly plotting the resolution of a digital word against its bit size. I am jumping out of one way of thinking into another without making that completely clear.

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"Resolution of a digital word in relation to its bit size"... huh? You are still not making yourself clear to me!

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Oh come on, put a bit of effort into it.

As the word length increases, the resolution increases. I know that you know and understand that so you must be able to understand that sentence above.

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Quantization errors at the lowest bits (at any bits actually) will be heard as distortion spread out over the whole bandwidth

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No, if properly dithered there will be no distortion, just noise.

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Arf! If you properly dither, you won't have quantization errors! So if you have quantization errors, you havn't properly dithered!

You wouldn't be trying to be disingenuous would you?

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We hear things on a logarithmic scale so the amplitude of errors and the increased resolution should also be calculated on a logarithimic scale. 24 bit audio is not 256 times more accurate to our ears.

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You are correct in that a 24 bit system isn't 256 times more accurate to our ears than 16 bit system, and yes, our sense of hearing is broadly logarithmic.

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Good.

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However, the quantisation scale is linear, and the size of quantisation errors is linearly related to the quantisation resolution. In that context, I think it is fair to say that the amplitude of a sample nearing 0dBFS can theoretically be determined with 256 times greater precision in a 24 bit system than is possible in a 16 bit system.

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Only if we qualify that statement be precising that we are talking about the linear scale as you did. I have no objection to what you are stating.

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In my book, in a technical discussion, refering to him as Mr Nichols would be a bigger insult than refering to him only by his sir name.

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I don't think he has been knighted...

Hugh

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Lol! Typo.


UnderTow

Quote Hugh Robjohns:

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Quote UnderTow:

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The errors at lower bits are much smaller than errors at higher bits.

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I think you are confusing quantisation errors with bit errors.

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Arf! Not at all. I am doing some mental acrobatics to explain a point!

I think you are working under the assumption that I don't understand what I am talking about instead of working under the assumption that I do and trying to follow my logic.

Of course, the first does require less effort on your part.

Quote:

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A quantisation error has the same maximum amplitude regradless of the amplitude of the audio signal being quantised.

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Agreed and stated by me in previous posts.

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A bit error would produce an erroneous sample amplitude, in which an error of the LSB would produce a far smaller amplitude error than an error in the MSB. But this particular issue is not relevant in any way to the topic being discussed.

Hugh

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Not directly but it is as a bit of mental exercise to understand the subject. You know as well as me that digital audio is not intuitive at first glance.

As I have explained in earlier posts, I was describing my mental processes to clarify that my terminology was wrong but not my thought processes.

Again, try and follow my logic instead of assuming that I don't understand. Then! you can say I am wrong.

UnderTow

Quote Spyder:

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"This is incomplete to say the least. You could have a recording with a noise floor way above -96 dB FS. It won't suddenly go down to -144 dB FS just by converting to 24 bits.
" - Undertow

Undertow, you seem to be fixated with converting a 16 bit file to 24 bit, as demonstrated by your 3 bit - 4bit example and the quote above.

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Arf! No! *sigh* These are just examples to clarify a point. Unfortunately, people are not trying to following the logic and are assuming I am wrong from the start.

UnderTow

Quote Lars Farm:

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AFAIK… In the digital domain the interpretation of the bitpattern is:
- bit 1/MSB carries info up to 0dBFS
- bit 2 (that happens to be LSB in example 1) carries info about what happens up to -6dBFS
- bit 3 (that happens to be LSB in example 2) carries info about what happens up to -12dBFS

So, the added bit has added four new values at the bottom of the amplitude range. Logarithmic, not linear. Shouldn't it be so in the analouge domain too?

Lars

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If expressed in dBV or dbm, yes. Of course if expressed in absolute voltages, then no.

You seem to be the only one that understands what I am driving at ...

UnderTow

Quote Len:

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To get back on topic, am I going to trust the ears of a Grammy winner who has (IMHO) recorded some truly stunning work to a level I can only dream of, or some guy who comes along screaming that the math is questionable (and even then as Hugh has pointed out, not very accurately)?

Answers on a postcard, please.

Len

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And this is the crux of the problem. He has a Grammy (or more) and people believe him blindly. That is why he should either, check his facts or have his articles checked by somone at SOS that is more knowledgable about digital audio.

I am not saying that Mr Nichols should be fired. I am saying that he should not write deep technical articles about digital audio. I feel it would best serve the subscribers of SOS if he shared his knowledge about recording and mixing. That knowledge that has, amongst other things, earned him a Grammy award and respect in (part of) the recording industry.

UnderTow

I stumbled upon this thread before reading the article and it made me want to read it to find out what all the ‘fuss’ is about.
To quote from the article, ‘As a percentage of the difference between adjacent samples, the quantisation error is much greater for a low-frequency signal than a high-frequency one’.

After reading this I had to double check the cover of the magazine to make sure that it wasn’t the April issue.
But seriously, is this comparison actually meaningful? Can’t a low frequency signal be much more accurately represented than a high frequency one using the same sample rate?

e.g. a 48 Hz sine wave sampled at 48 kHz contains 1,000 samples per sine wave.
Whereas, a 4.8 kHz sine wave sampled at 48 kHz contains 10 samples per sine wave.
I’m not saying that this means that lower frequency sounds will ‘sound better’, I’m just trying to make another comparison, to juxtaposition it against Roger’s.
Is it anymore meaningful? My mind says yes, but I really don't know!

I found Roger’s style of writing difficult to follow; it reminded me that I use certain words in conversation that I’d find impossible to define accurately, but when I look them up in the dictionary I usually see that I’ve been using them in the correct context. There’s a big difference between being able to do something and to be able to talk about what you are doing in a clear and concise way. I’m looking forward to his next column; although I'm not sure that my motives are pure.

Quote Steve Hill:

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Methinks the Grammy on the wall might be a killer argument...

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That’s assuming you can still see it after Hugh’s popped round and carpeted your lounge wall from floor to ceiling

Edited by Crow (27/04/06 07:20 PM)

everybody step back !!!

the engineers are arguing !

this could turn nasty.



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http://profile.myspace.com/abitfunkdub/

I'll repeat part of my old post:

" [Mr Nichols] thinks long and hard, and comes up with a good, logically consistent theory about bass sounds with 24 bit. And writes an interesting and brain-engaging article."

So? What's wrong?

His argument is logically self-consistent.

SOS is not a peer-reviewed science magazine.

If someone stimulates some thinking, that's a good thing. Theories will always provoke criticism - it's the way science and knowledge progress. The article is obviously personal effort at understanding a phenomenon.

G

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PA stuff on FB

Quote Crow:

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After reading this I had to double check the cover of the magazine to make sure that it wasn’t the April issue.

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That was my reaction too ...

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But seriously, is this comparison actually meaningful? Can’t a low frequency signal be much more accurately represented than a high frequency one using the same sample rate?

e.g. a 48 Hz sine wave sampled at 48 kHz contains 1,000 samples per sine wave.
Whereas, a 4.8 kHz sine wave sampled at 48 kHz contains 10 samples per sine wave.
I’m not saying that this means that lower frequency sounds will ‘sound better’, I’m just trying to make another comparison, to juxtaposition it against Roger’s.
Is it anymore meaningful? My mind says yes, but I really don't know!

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Actually no. You only need two sample points to accurately represent a sine wave. Any sine wave up to half the sampling rate (the so called nyquist frequency) can be accurately reproduced. For a full explanation of this, please read this article:

http://www.lavryengineering.com/documents/Sampling_Theory.pdf

UnderTow

I'll repeat part of my old post:

" [Mr Nichols] thinks long and hard, and comes up with a good, logically consistent theory about bass sounds with 24 bit. And writes an interesting and brain-engaging article."

So? What's wrong?

His argument is logically self-consistent.

SOS is not a peer-reviewed science magazine.

If someone stimulates some thinking, that's a good thing. Theories will always provoke criticism - it's the way science and knowledge progress. The article is obviously personal effort at understanding a phenomenon.

G

BTW: Nice comment about the banjo . . . though I like them, myself, and Bela Fleck is of course, a god.

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PA stuff on FB

Quote UnderTow:

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Quote Crow:

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But seriously, is this comparison actually meaningful? Can’t a low frequency signal be much more accurately represented than a high frequency one using the same sample rate?

e.g. a 48 Hz sine wave sampled at 48 kHz contains 1,000 samples per sine wave.
Whereas, a 4.8 kHz sine wave sampled at 48 kHz contains 10 samples per sine wave…

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Actually no. You only need two sample points to accurately represent a sine wave. Any sine wave up to half the sampling rate (the so called nyquist frequency) can be accurately reproduced.

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My choice of sine waves in my example was unfortunate; replace that with a complex waveform and my point will be clearer.

Quote UnderTow:

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For one he could have mentioned what he is comparing to. Judging by the rest of the article he seems to be comparing modern 24 bit converters to early 3M digital machines (with a combination of a 12-bit converter with an additional four bits of an 8-bit converter for gain ranging). I could be wrong but as he isn't clear, I can't be sure.

So it would appear that the conclusion of his whole article is: Modern 24 bit converters sound better than 24 year old converters with 12 bit resolution and 4 bit gain scaling.

No [ ****** ] sherlock!

UnderTow

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It's been a few weeks since I read that article, but WOW: that statement alone makes it clear you missed the forest for the trees, and got so hung up on tech details that you entirely missed his point for writing the article!

The SUBJECT of his article is to point out issues encountered when transferring between digital formats. OF COURSE HE'S TALKING ABOUT ANTIQUATED GEAR, since that's the same gear that was used to make the original recordings he's archiving!

In his case, he's archiving digital recordings made in the early 1980's on 3M multitrack recorder, and he's simply saying that understanding the workings of the gear that the original music was recorded on will only aid in ending up with a more accurate rendition of the original material in current digital format. Yes, he recorded on that gear back when it WAS state-of-the-art, and knows it's inner workings quite intimately.

Now if he was unaware of problems with the earlier gear, or was a young whipper-snapper engineer with no first-hand experience of the 3M machine, he might have been tempted to use a digital interface board to do the transfers digitally, thinking a digital transfer would be preferable. Bit no, he decided NOT to do that, since it would've induced non-linear errors that the 3M's original D-A converters compensate for.

So in his words, "analogue transfers it was to be". He recorded to 24-bit from the analog outs.... Why? To make a more accurate archive by recording in 24-bit with modern gear....


You also referred to his last article, where he mentioned hard drives. You completely missed the point there, too: he was providing anecdotal experience that leaving data on hard drives in NOT the most reliable archiving method. This is hardly just his theory: computer scientists will tell you hard drives are subject to data corruption from stray cosmic radiation, and the damage is cumulative. Have an error induced in the wrong sector, and the entire drive may be left unuseable. I think he mentioned the concern that some of the moving parts inside of hard drives tend to seize/bind if the drives are allowed to sit without use for long periods (decades), which is entirely true due to bearing grease and other issues.

Do a Google on "hard drive bearing grease".


@@@@

Dude (and you come off as a Dude, BTW), I'd think long and hard before calling Roger Nichols to the carpet on either his knowledge or experience: he's worked for the major labels for at least 4 decade, and has FORGOTTEN more about digital audio and recording engineering than some wanna-be anorak freshman straight out of of basic audio theory classes (earning a "C", no doubt) could hope to know. The "Immortal" Roger Nichols has been working in the studios as an engineer even before many of us were waddling about in diapers, and the major audio manufacturers were banging on his door for his technical assistance to help them usher in digital audio gear in the early 1980s!

If nothing else, he has the respect of Donald Fagan/Walter Becker, and if he's good enough to be the engineer for Steely Dan records, then that's good enough for me.

BTW, SOS, Kudos on adding Roger Nichols to the roster: a definite feather in the cap.

Chris

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1.5GHz Al 17" Powerbook G4 (2.0GB RAM, Hitachi 60GB 7,200 rpm drive), running Logic Pro 7 under OSX 10.4.5

Quote Crow:

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My choice of sine waves in my example was unfortunate; replace that with a complex waveform and my point will be clearer.

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Any complex waveform can be decomposed into a sum of sine-waves of different frequencies. Your complex waveform can thus simply be completely described by simple sine waves of varying frequencies. Simple sine waves and only simple sine waves.

The limitation here is mostly how small (in time) irregularities you can represent. You can't describe a sine-wave with less than two samples per cycle. So, the sample frequency sets a limit on the complexity of the waveform by limiting the highest frequency that can be represented. Bit depth sets another limit. Quantization error/noise depends on bit depth. This affects how soft sounds you can represent.

Lars

Phew! After reading this thread I thought I was going nuts. Hugh's comments on this subject have put my mind at rest again, thankfully...

Quote Feefer:

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Dude (and you come off as a Dude, BTW), I'd think long and hard before calling Roger Nichols to the carpet on either his knowledge or experience: he's worked for the major labels for decades, and has FORGOTTEN more about digital audio and recording engineering than some anorak freshman straight out of of basic audio theory classes (earning a "C", no doubt) could hope to know. The "Immortal" Roger Nichols been working in the studios as an engineer before most of us were waddling about in diapers, and the major audio manufacturers were banging on his door for his technical assistance to help them usher in digital audio gear!If nothing else, he has the respect of Donald Fagan/Walter Becker, and if he's good enough to be the engineer for Steely Dan records, then that's good enough for me.

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Well yeah, there's that, too.

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Gear list: If you can't find it, grind it

Quote Lars Farm:

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Any complex waveform can be decomposed into a sum of sine-waves of different frequencies. Your complex waveform can thus simply be completely described by simple sine waves of varying frequencies. Simple sine waves and only simple sine waves. Lars

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Yeah, hats off to Monsieur Fourier, but this is completely off topic!

Quote Feefer:

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The SUBJECT of his article is to point out issues encountered when transferring between digital formats. OF COURSE HE'S TALKING ABOUT ANTIQUATED GEAR, since that's the same gear that was used to make the original recordings he's archiving!

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But he is not! He is comparing 16 bit to 24 bit. He is not comparing "a unique combination of a 12-bit converter with an additional four bits of an 8-bit converter for gain ranging."

Get your blinders off and accept that Nichols might be wrong!

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In his case, he's archiving digital recordings made in the early 1980's on 3M multitrack recorder, and he's simply saying that understanding the workings of the gear that the original music was recorded on will only aid in ending up with a more accurate rendition of the original material in current digital format. Yes, he recorded on that gear back when it WAS state-of-the-art, and knows it's inner workings quite intimately.

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I agree with that but it is just an introductory paragraph. In the rest of his article he is comparing straight 16 bit with straight 24 bit. He is not comparing 24 bit to "16 bit" implementation of the 3M digital machines!

This should have been a hint to you if you were not so starstruck:

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With most digital audio now being recorded at 24-bit, nobody thinks about the ultimate 16-bit destination and what that conversion does to the sound quality.

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Do you see now?

Quote:

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Now if he was unaware of problems with the earlier gear, or was a young whipper-snapper engineer with no first-hand experience of the 3M machine, he might have been tempted to use a digital interface board to do the transfers digitally, thinking a digital transfer would be preferable. Bit no, he decided NOT to do that, since it would've induced non-linear errors that the 3M's original D-A converters compensate for.

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I am not disagreeing with his choice but he could have summed it up by saying that the 3M implementation is not compatible with modern implementations of digital audio.

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So in his words, "analogue transfers it was to be". He recorded to 24-bit from the analog outs.... Why? To make a more accurate archive by recording in 24-bit with modern gear....

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Again, I do not disagree with his choice. I do disagree with his explanation.

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You also referred to his last article, where he mentioned hard drives.

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Lets stick to one article at a time.

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Dude (and you come off as a Dude, BTW), I'd think long and hard before calling Roger Nichols to the carpet on either his knowledge or experience: he's worked for the major labels for decades, and has FORGOTTEN more about digital audio and recording engineering

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Bla bla bla. It doesn't change the fact that his explanations are not correct.

Quote:

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than some anorak freshman straight out of of basic audio theory classes (earning a "C", no doubt) could hope to know.

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Don't shoot the messenger. Anyway, do not assume anything about me just because you can't follow my arguments.

UnderTow

Quote UnderTow:

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Get your blinders off and accept that Nichols might be wrong!

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how about considering that YOU might be wrong... (and Are)

as others have already noted.

this

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Quote:


Just to refresh your memory, 24-bit audio gives you 256 times more resolution in the position of each sample on the waveform.




This isn't correct. It only gives more resolution to low-level signals. This can be easily demonstrated by converting a 16 bit file to a 24 bit file. The extra 8 bits are just filled with zeros. There is no requantization of the audio. A (loud) signal which lives in the upper bits of a 16 bit audio file does not gain any resolution by having extra zeros added in the extra bits.

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clearly shows that you lack a full understanding of numeric data systems. or perhaps appropriate terminology.

to put it bluntly. and in Base 10 to be utterly crystal clear...

if in addition to expressing integer values, you are capable of expressing partial values, like 0.00001 to 0.99999 then you are capable of expressing 15.00001 to 15.99999 as well.

this by definition is finer resolution than only being capable of expressing integers

and it's effectively this that the extra bits allow.

In addition, the fact that you seem to fail to appreciate WHY your example of simply converting a 16 bit file to a 24 bit file, (and back) in the suggested manner is SO not relevant, shows a decided logical blind spot.

Since this is a core tenet of your entire argument... I won't bother pulling the rest apart.,..

I would suggest an apology might be in order....

Max

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if you don't know who i am, i aint gonna tell you.

Quote UnderTow:

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Get your blinders off and accept that Nichols might be wrong!

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Why does this matter so much? Why are you banging this drum so hard? Is it personal or what?

We all know what sounds good, or not. Maybe 0.00001% of the population give a flying toss about the underlying science as to WHY is sounds the way it does. A fractionally higher percentage of the population of sound engineers might give a toss, but I assure you it is very marginal.

Yes, he might be wrong, or at least guilty of some infelicitous language. So what? I'm not going to frame his article and build a candle-lit shrine round it. It's tomorrow's fish and chips wrapper.

--------------------
Dynamite with a laser beam...

Quote Max The Mac:

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Quote UnderTow:

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Get your blinders off and accept that Nichols might be wrong!

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how about considering that YOU might be wrong... (and Are)
as others have already noted.

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I posted some wrong stuff and I think I have cleared that up. And I did say sorry for the confusion. So I did appologise for that. If that wasn't clear, I appologise again.

Quote:

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clearly shows that you lack a full understanding of numeric data systems. or perhaps appropriate terminology.

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Yes that was wrong and I shouldn't have posted at 10 to 4 in the morning. But I did explain later on what I meant. Basicly, in my tired state, I was thinking of error magnitudes but describing it as just amplitude. I should have said errors in the amplitutde or something like that.

Quote:

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and it's effectively this that the extra bits allow.

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Yes agreed. My point was that the error gets smaller.

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In addition, the fact that you seem to fail to appreciate WHY your example of simply converting a 16 bit file to a 24 bit file, (and back) in the suggested manner is SO not relevant, shows a decided logical blind spot.

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It was relevant to demonstrate one particular point in response to one remark. My bad terminology confused the whole issue. That doesn't mean I have a logical blind spot. It means I shouldn't post at 10 to 4 in the morning.

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Since this is a core tenet of your entire argument... I won't bother pulling the rest apart.,..

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Not really but please do.

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I would suggest an apology might be in order....

Max

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Are you saying that the article is factualy correct? If so, and you can explain why, I will appologise. Maybe I do misunderstand that article completely but so far this hasn't been demonstrated AFAICS.

UnderTow

Ok I give up. And I appologise to this forum for ever bringing this up.

UnderTow

Quote UnderTow:

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Ok I give up. And I appologise to this forum for ever bringing this up.
UnderTow

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AMEN!

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Gear list: If you can't find it, grind it

To quote from the article, ‘As a percentage of the difference between adjacent samples, the quantisation error is much greater for a low-frequency signal than a high-frequency one’.

I’m really curious about this postulation, but unfortunately possibly the only person that is! I can see where he’s coming from with this, but I’m not sure that it’s at all a relevant comparison. Anyone have an opinion on this?

I found it an interesting article, although a bit ‘strange’ in some way I can’t quite describe. I’ve never heard of the guy and I don’t really care how many awards he has, I’m just trying to understand this particular postulation. I’m intrigued and curious to know what others make of this idea of his.