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UnderTow
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Joined: 27/02/03
Posts: 317
Nichols article on 24 bit recording WRONG!
      #287670 - 26/04/06 05:53 PM
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


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PrinceXizor
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287674 - 26/04/06 06:25 PM
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

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My Home Studio Build Thread


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James Lehmann



Joined: 17/05/05
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287680 - 26/04/06 06:34 PM
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.


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The Producer



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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287684 - 26/04/06 06:47 PM
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.


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tex
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287740 - 26/04/06 09:12 PM
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)


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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287745 - 26/04/06 09:21 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


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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: tex]
      #287746 - 26/04/06 09:22 PM
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


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MadManDan



Joined: 13/09/04
Posts: 1863
Loc: Across the pond....New Yawk
Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287756 - 26/04/06 09:59 PM
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


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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: MadManDan]
      #287781 - 26/04/06 10:47 PM
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


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Mowens800



Joined: 16/06/05
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287783 - 26/04/06 10:48 PM
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


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tex
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Joined: 01/04/03
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287800 - 26/04/06 11:16 PM
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.


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gerard



Joined: 07/02/05
Posts: 2608
Loc: London, UK
Re: Nichols article on 24 bit recording WRONG! new [Re: Mowens800]
      #287802 - 26/04/06 11:18 PM


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)


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The Producer



Joined: 17/11/04
Posts: 97
Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287808 - 26/04/06 11:28 PM
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.


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UnderTow
member


Joined: 27/02/03
Posts: 317
Re: Nichols article on 24 bit recording WRONG! new [Re: The Producer]
      #287816 - 26/04/06 11:52 PM
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


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Guy Johnson



Joined: 02/05/03
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287826 - 27/04/06 12:13 AM
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

--------------------
Oh. It's Christmas.


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UnderTow
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Posts: 317
Re: Nichols article on 24 bit recording WRONG! new [Re: Guy Johnson]
      #287835 - 27/04/06 12:56 AM
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


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Nathan



Joined: 13/09/04
Posts: 1899
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287837 - 27/04/06 01:00 AM
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.


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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: Nathan]
      #287841 - 27/04/06 02:53 AM
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


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Anonymous
Unregistered




Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287844 - 27/04/06 03:27 AM
I'm gonna wait for Hugh to wake up so I can start reading this thread properly.

Night


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Michael Harrison
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Re: Nichols article on 24 bit recording WRONG! new [Re: ]
      #287850 - 27/04/06 05:30 AM
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


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gerard



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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287859 - 27/04/06 06:52 AM


i've got a headace already...

(still working on that percentage thingy)

har har


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Sheriton



Joined: 27/01/03
Posts: 1555
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287874 - 27/04/06 08:13 AM
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


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Nathan



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Re: Nichols article on 24 bit recording WRONG! new [Re: Sheriton]
      #287895 - 27/04/06 08:52 AM
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.


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Sam Inglis
SOS Features Editor


Joined: 15/12/00
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287897 - 27/04/06 08:53 AM
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?


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Lars Farm



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Posts: 66
Loc: Sundsvall, Sweden
Re: Nichols article on 24 bit recording WRONG! new [Re: Sheriton]
      #287909 - 27/04/06 09:04 AM
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


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Spandau-Staaken



Joined: 15/03/06
Posts: 647
Loc: N.E. U.K.
Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287911 - 27/04/06 09:05 AM
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...


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Steve Hill
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287919 - 27/04/06 09:12 AM
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.

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gerard



Joined: 07/02/05
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287922 - 27/04/06 09:13 AM


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...


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tomafd



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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287938 - 27/04/06 09:26 AM
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

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James PerrettModerator



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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287940 - 27/04/06 09:28 AM
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.

--------------------
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http://www.jrpmusic.net


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Nathan



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Re: Nichols article on 24 bit recording WRONG! new [Re: Lars Farm]
      #287954 - 27/04/06 09:34 AM
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.

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Hugh RobjohnsAdministrator
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287959 - 27/04/06 09:39 AM
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

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feline1
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287965 - 27/04/06 09:43 AM
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

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


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Steve Hill
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Re: Nichols article on 24 bit recording WRONG! new [Re: feline1]
      #287973 - 27/04/06 09:49 AM
And on a more serious topic by far, the technology has yet to be invented which would "finally make the banjo worth recording"

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Hugh RobjohnsAdministrator
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287984 - 27/04/06 10:05 AM
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

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steveman



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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #287998 - 27/04/06 10:21 AM
QED?


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cc.
getting into my stride


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Re: Nichols article on 24 bit recording WRONG! new [Re: Hugh Robjohns]
      #288011 - 27/04/06 10:40 AM
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).

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Midipicks - the all new MIDI Guitar forum...


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


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gerard



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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288016 - 27/04/06 10:45 AM


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

banjo? are you serious?


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Spandau-Staaken



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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288035 - 27/04/06 11:11 AM
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..



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seablade



Joined: 21/11/04
Posts: 4055
Re: Nichols article on 24 bit recording WRONG! new [Re: Lars Farm]
      #288048 - 27/04/06 11:27 AM
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;)


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Huge Longjohns
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288091 - 27/04/06 12:19 PM
So let me get this straight: if I record in 24 bit Hugh will come round and fit my carpets for me?

--------------------
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seablade



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Re: Nichols article on 24 bit recording WRONG! new [Re: Huge Longjohns]
      #288094 - 27/04/06 12:20 PM
Quote simonplent:

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


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Nathan



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Re: Nichols article on 24 bit recording WRONG! new [Re: Huge Longjohns]
      #288098 - 27/04/06 12:31 PM
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...


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Len
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Re: Nichols article on 24 bit recording WRONG! new [Re: Nathan]
      #288106 - 27/04/06 12:46 PM
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

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www.youtube.com/leonardngmusic


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Len
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288108 - 27/04/06 12:48 PM
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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Steve Hill
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Re: Nichols article on 24 bit recording WRONG! new [Re: Len]
      #288114 - 27/04/06 12:51 PM
Methinks the Grammy on the wall might be a killer argument...

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cc.
getting into my stride


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Re: Nichols article on 24 bit recording WRONG! new [Re: Len]
      #288115 - 27/04/06 12:53 PM
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.

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cc.
getting into my stride


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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288120 - 27/04/06 12:56 PM
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

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Shivanand
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Re: Nichols article on 24 bit recording WRONG! new [Re: Spandau-Staaken]
      #288143 - 27/04/06 01:35 PM
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.

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Lars Farm



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Re: Nichols article on 24 bit recording WRONG! new [Re: seablade]
      #288151 - 27/04/06 01:47 PM
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


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BigAl
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Re: Nichols article on 24 bit recording WRONG! new [Re: Len]
      #288183 - 27/04/06 02:35 PM
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!

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Doublehelix



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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288201 - 27/04/06 03:01 PM
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~


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BigAl
Just The Bass Player


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Re: Nichols article on 24 bit recording WRONG! new [Re: Doublehelix]
      #288205 - 27/04/06 03:09 PM
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.

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Mowens800



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Re: Nichols article on 24 bit recording WRONG! new [Re: Doublehelix]
      #288208 - 27/04/06 03:13 PM
Quote Doublehelix:


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.


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feline1
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288211 - 27/04/06 03:15 PM
I thought a "bitchin-izer" was what some of ran our posts through before submitting them to this forum

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desmond



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Re: Nichols article on 24 bit recording WRONG! new [Re: feline1]
      #288214 - 27/04/06 03:22 PM
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)


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feline1
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Re: Nichols article on 24 bit recording WRONG! new [Re: desmond]
      #288226 - 27/04/06 03:38 PM
Quote desmond:


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




Rosemary, the telephone operator?

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Re: Nichols article on 24 bit recording WRONG! new [Re: cc.]
      #288232 - 27/04/06 03:46 PM
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

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Richard Graham



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Re: Nichols article on 24 bit recording WRONG! new [Re: feline1]
      #288234 - 27/04/06 03:52 PM
Quote feline1:

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'.


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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: Sheriton]
      #288244 - 27/04/06 04:21 PM
Quote Sheriton:


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:


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:


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.


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seablade



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Re: Nichols article on 24 bit recording WRONG! new [Re: Lars Farm]
      #288247 - 27/04/06 04:25 PM
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:


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


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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: Lars Farm]
      #288251 - 27/04/06 04:33 PM
Quote Lars Farm:

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


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gerard



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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288267 - 27/04/06 05:06 PM


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

-=sigh=-


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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: Sam Inglis]
      #288271 - 27/04/06 05:16 PM
Quote Sam Inglis:


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


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Spyder
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288277 - 27/04/06 05:25 PM
"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.

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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288279 - 27/04/06 05:29 PM
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

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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: Hugh Robjohns]
      #288287 - 27/04/06 05:38 PM
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


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Hugh RobjohnsAdministrator
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288292 - 27/04/06 05:46 PM
Quote UnderTow:

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





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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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: Hugh Robjohns]
      #288309 - 27/04/06 06:18 PM
Quote Hugh Robjohns:


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.





Ok, we agree.

Quote:


Quote:

In reality we have the following:




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





Ok, we agree.

Quote:


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.





Ok, we agree.

Quote:


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).





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.

Quote:


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.





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.

Quote:


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.





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.

Quote:


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!





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.

Quote:


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.





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?

Quote:


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.





Good.

Quote:


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.





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.

Quote:


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








Lol! Typo.


UnderTow


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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: Hugh Robjohns]
      #288315 - 27/04/06 06:25 PM
Quote Hugh Robjohns:

Quote UnderTow:

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





I think you are confusing quantisation errors with bit errors.





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:


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





Agreed and stated by me in previous posts.

Quote:


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




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


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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: Spyder]
      #288317 - 27/04/06 06:29 PM
Quote Spyder:

"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.




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


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Re: Nichols article on 24 bit recording WRONG! new [Re: Lars Farm]
      #288318 - 27/04/06 06:32 PM
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




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


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Re: Nichols article on 24 bit recording WRONG! new [Re: Len]
      #288320 - 27/04/06 06:38 PM
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.

Len




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


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Crow



Joined: 20/04/06
Posts: 86
Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288340 - 27/04/06 07:11 PM
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:

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


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)


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abitfunkdub
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288352 - 27/04/06 07:41 PM
everybody step back !!!

the engineers are arguing !

this could turn nasty.



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


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Guy Johnson



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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288353 - 27/04/06 07:46 PM
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

--------------------
Oh. It's Christmas.


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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: Crow]
      #288355 - 27/04/06 07:49 PM
Quote Crow:



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






That was my reaction too ...

Quote:


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!





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


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Guy Johnson



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Re: Nichols article on 24 bit recording WRONG! new [Re: Guy Johnson]
      #288356 - 27/04/06 07:49 PM
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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Crow



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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288361 - 27/04/06 08:07 PM
Quote UnderTow:

Quote Crow:


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…




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.



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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Feefer
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288376 - 27/04/06 08:47 PM
Quote UnderTow:


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




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

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


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Lars Farm



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Re: Nichols article on 24 bit recording WRONG! new [Re: Crow]
      #288378 - 27/04/06 08:54 PM
Quote Crow:

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




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


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AndyJones
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Re: Nichols article on 24 bit recording WRONG! new [Re: Hugh Robjohns]
      #288380 - 27/04/06 09:03 PM
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...


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MadManDan



Joined: 13/09/04
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Re: Nichols article on 24 bit recording WRONG! new [Re: Feefer]
      #288383 - 27/04/06 09:09 PM
Quote Feefer:

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.


Well yeah, there's that, too.

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


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Crow



Joined: 20/04/06
Posts: 86
Re: Nichols article on 24 bit recording WRONG! new [Re: Lars Farm]
      #288391 - 27/04/06 09:18 PM
Quote Lars Farm:

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



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


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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: Feefer]
      #288395 - 27/04/06 09:24 PM
Quote Feefer:


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!





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!

Quote:


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.





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:

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.





Do you see now?

Quote:


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.





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.

Quote:


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....





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

Quote:


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





Lets stick to one article at a time.

Quote:


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





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

Quote:


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





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

UnderTow


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Studio Support Gnome
Not so Miserable Git


Joined: 22/07/03
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288418 - 27/04/06 09:57 PM
Quote UnderTow:

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





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

as others have already noted.

this

Quote:

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.






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

--------------------
Don't get the hump when i tell you it's going to be expensive, it's not my fault , you picked the site/building/room â


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Steve Hill
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288419 - 27/04/06 09:58 PM
Quote UnderTow:

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




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.

--------------------
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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: Studio Support Gnome]
      #288429 - 27/04/06 10:19 PM
Quote Max The Mac:

Quote UnderTow:

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





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





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:


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





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:


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





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

Quote:


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.





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.

Quote:


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





Not really but please do.

Quote:


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

Max




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


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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288430 - 27/04/06 10:21 PM

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

UnderTow


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MadManDan



Joined: 13/09/04
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288438 - 27/04/06 10:36 PM
Quote UnderTow:


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


AMEN!

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Crow



Joined: 20/04/06
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Re: Nichols article on 24 bit recording WRONG! new [Re: Crow]
      #288439 - 27/04/06 10:37 PM
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.


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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: Crow]
      #288443 - 27/04/06 10:53 PM
Quote Crow:


Anyone have an opinion on this?





Yes but I'll spare you all.

But I still do suggest people that are interested in the subject read the Dan Lavry article I reference in my first post.

UnderTow


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Studio Support Gnome
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288461 - 28/04/06 12:01 AM
Quote UnderTow:

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.






we'll start with an apology ( note the single p ) on my own behalf , as I must have missed those in amongst the rest of it... or at least forgotten them by the time i got to this end of the thread.

it does however rather seem that you failed to modify your later position in the light of your previously , now understood , incorrect positions.

in other words... it seemed very much like you kept banging on the same line despite having been shown and having admitted the errors..

in light of which, I really didn't much feel like expanding and wasting everybody's screen space with pointless effort.

I have to agree that posting at 10 to 4 in the morning isn't always the best idea.... I've done it myself on a number of occasions...

it's rather a sweeping statement to say the entire article is 100% factually correct... but in relation to your original core complaints, yes it is.

The why has already been gone over... several times.. and you've already accepted it.

I should add that SOS readers, on the whole, have been rather spoilt ...... because the majority of "deep" technical explanation articles have been pulled out of the hands of one of the most easily understood writers in the field... Hugh...

Probably fair to say that his background as a trainer for Auntie has left him with a generally uncanny ability to put a point across with the minimum fuss and confusion on the vast majority of occasions.... most especially when he has time to think about how he wants to say it.... basically the BBC like people to come out of Wood Norton with some clue about what it is they're doing!

IMHO, if the average reader were to try getting the explanations contained any one of his more technical articles out of the scientific and engineering papers and manuals that originally contained the data , They'd be very tired, certainly confused and probably no wiser than when they started...


This is not to say that Hugh is ALWAYS right, as he himself is happy to point out.... but what he does say, he says very succinctly and intelligibly .



I cannot say that I find the same quality in the Writing of Roger Nichols, I want to be very clear on that... i am NOT his biggest fan..... in terms of his WRITING, but I cannot really fault his technical ability , experience, and general background knowledge.....


However, your initial post, and original "backing up" of your claims were factually and conceptually incorrect. others have stated so, and you've admitted to it piecemeal when shown in detail .... what's left is apparently largely a case of you fighting your corner because you feel put upon, not because you're actually right...

however, in response to your request I quote below your initial argument, and my rebuttals will be interspersed.


Quote:


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.


>>> Max .
Err Actually it (the sound quality issue ) is supported by all authorities on the subject, and is the reason for the multiple approaches to Dither by various parties, to get subjectively the best result... In context the statement is simple, and in my experience true... very very few people actually consciously consider the Delivery media during the recording process.. just look at how many people argue over Sample rate conversions from 24/96 to 16/44.1 , and how often the non integer relationship of the sample rate is accused of being the ONLY issue, utterly forgetting that there is also a bit depth reduction to be handled as well.
The Phrase "No-one" is often used as a generalisation in this kind of context, and his hardly an insult or defamation of anyone falling outside of the generalisation.

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.




>>>>>>> Max. First major technical issue., you've seemingly read his comment backwards... he never said what you implied. he never meant or implied it, and your comment is totally irrelevant in the context of the original article. it has no need to be a complete discourse on Digital Audio, it made a point within the article, and was all that was required to do so.

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.


>>>>>> Max. We've been here and done that, but this shows it in context. your statement here is utterly wrong.

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.


>>>>> Max...
Fact, he heard a difference ,
which in his experience, of recording audio, was significant... and his experience is not to be discounted or taken as being irrelevant... it is highly relevant.. it is only based on subjective experience and qualitative assessment that anyone can make a value judgement on anything... I submit that his experience is demonstrably long enough and at a high enough level for that value judgement to be given some credit.
a sort of "Expert Witness" in court kind of scenario... and i submit you were NOT there, and cannot make any kind of judgement based on that at all.
I also get the feeling that frankly, you don;t have a heap of experience with "antiquated" digital systems... or with recording "at the bleeding edge" with early Digital equipment... because those differences are very very real, as anyone who has had that experience will tell you...



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.


>>>>>> Max.
Been there, done that, again....



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!


>>>>>>> Max.
Okay this is where I feel the writing style to be the issue rather than the underlying concept or data.... I don't think Hugh would present that idea like this, and I sure as hell wouldn't... it's at this point I had to do a double take... for a while I had the feeling there'd been an editorial cock up with the diagram and text, until Hugh put my head back on the right way round...

Comparing them like this is I suppose, a valid way of presenting the idea... that there can be significant differences between 16 and 24 bit recordings of the same event... but it's definitely not the most concise or easily intelligible way... you CAN compare the two this way... you did it yourself just a moment beforehand... adding 8 zeroes remember? Dither is essentially a solution to a problem, not necessarily central to recording data at either resolution. But the fact that it ignores the existence of Dither as a solution does leave it somewhat short of an ideal description...


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.


>>>>>> left these two ,lumps together to answer in context.

Max.
Yes and err, no... in a very real, but somewhat general sense, ALL IC's are made that way... optical lithography, and ALL AD/DA systems have chips that are IC's at their heart........ AND no, he's not really talking about poxy PC or Mac sound cards, but real discrete Hardware, before the PC/Mac based DAW could so much as fart digital audio., in fact before the DAW we know it today existed .
The article was not meant as an educational discourse on cutting edge AD/DA , or PC hardware, but as a piece of useful background knowledge, that might one day save someone's bacon... . imagine if YOU had been given the job of re-archiving all this material.... would you have gone for the straight digital to digital transfer, that would seem so obvious a ploy , had you not known about some of the stuff contained in this article.... and others of it's type.... only those with experience with the systems of the day would really understand this without having to be told it... and there are increasingly few of them left and fewer still working almost at "tape op" level... archiving is not usually considered the preserve of the Grammy winning , working recording engineer.... and VERY few college courses teach anything about such "antiquated" hardware or it's operational idiosyncrasies.

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 totally 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 purpose.

UnderTow





The latter sections really need no comment... initial reactions to your post were about the manner of posting and the overtly aggressive "I'm oh so right and he's oh so wrong and should be hung" style...
all of which were pretty much to the point and fair comment... take it like a man and fess up to that.....

Sadly the ensuing posts did rather demonstrate the lack of sound argument


( a bad pun sorry, it is late, and I too am tired)

and your final comment was somewhat arrogant, or at least, one sided, since you pretty much asked everyone else to do exactly that.... to disprove your claims. virtually discounting their comment unless they did so
.

here endeth the lesson, that'll be £40 thanks.


best regards
Max


rearrange the following words.
Tea in cup a storm


--------------------
Don't get the hump when i tell you it's going to be expensive, it's not my fault , you picked the site/building/room â


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Guy Johnson



Joined: 02/05/03
Posts: 4487
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Re: Nichols article on 24 bit recording WRONG! new [Re: Studio Support Gnome]
      #288467 - 28/04/06 12:40 AM
Quote Max The Mac:



rearrange the following words.
Tea in cup a storm





24 bit molehills out of 16 bit mountains?


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UnderTow
member


Joined: 27/02/03
Posts: 317
Re: Nichols article on 24 bit recording WRONG! new [Re: Studio Support Gnome]
      #288475 - 28/04/06 01:48 AM
First up, I'll say that giving up was a good idea. It let me actually think about the content of my post and the tone I was using in that post.

I was in a very bad mood due to stuff that had nothing to do with anyone on this forum or anyone writing for SOS. I should never have let that spill into this forum. Or rather, I should never have posted anything in that state of mind. It clouded my judgement to say the very least!

I apologise to everyone here and Mr Nichols in particular.

Quote Max The Mac:


we'll start with an apology ( note the single p ) on my own behalf , as I must have missed those in amongst the rest of it... or at least forgotten them by the time i got to this end of the thread.





No need to apologise. I was talking out of my arse as well as being one.

Quote:


it does however rather seem that you failed to modify your later position in the light of your previously , now understood , incorrect positions.

in other words... it seemed very much like you kept banging on the same line despite having been shown and having admitted the errors..





Yes yes, *lowers head in shame*.

Quote:


I should add that SOS readers, on the whole, have been rather spoilt ...... because the majority of "deep" technical explanation articles have been pulled out of the hands of one of the most easily understood writers in the field... Hugh...

Probably fair to say that his background as a trainer for Auntie has left him with a generally uncanny ability to put a point across with the minimum fuss and confusion on the vast majority of occasions.... most especially when he has time to think about how he wants to say it.... basically the BBC like people to come out of Wood Norton with some clue about what it is they're doing!

IMHO, if the average reader were to try getting the explanations contained any one of his more technical articles out of the scientific and engineering papers and manuals that originally contained the data , They'd be very tired, certainly confused and probably no wiser than when they started...





Entirely agreed.

Quote:


However, your initial post, and original "backing up" of your claims were factually and conceptually incorrect. others have stated so, and you've admitted to it piecemeal when shown in detail .... what's left is apparently largely a case of you fighting your corner because you feel put upon, not because you're actually right...





Yes. Correct assessment.

Quote:


however, in response to your request I quote below your initial argument, and my rebuttals will be interspersed.





Thanks.

Quote:


>>> Max .
Err Actually it (the sound quality issue ) is supported by all authorities on the subject, and is the reason for the multiple approaches to Dither by various parties, to get subjectively the best result... In context the statement is simple, and in my experience true... very very few people actually consciously consider the Delivery media during the recording process.. just look at how many people argue over Sample rate conversions from 24/96 to 16/44.1 , and how often the non integer relationship of the sample rate is accused of being the ONLY issue, utterly forgetting that there is also a bit depth reduction to be handled as well.
The Phrase "No-one" is often used as a generalisation in this kind of context, and his hardly an insult or defamation of anyone falling outside of the generalisation.





Indeed. I shouldn't have taken personaly something that wasn't personal at all.


Quote:


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.

>>>>>>> Max. First major technical issue., you've seemingly read his comment backwards... he never said what you implied. he never meant or implied it, and your comment is totally irrelevant in the context of the original article. it has no need to be a complete discourse on Digital Audio, it made a point within the article, and was all that was required to do so.





Yup. In a sense I was supporting his statement. It isn't so much that I read his comment backwards, it is that I failed miserably to convey what I was trying to say: That the noise floor of 16 or 24 bit converters is not the only thing that contributes to the overall noise floor. This is certainly not a contradiction to what Mr Nichols says.

Quote:


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.


>>>>>> Max. We've been here and done that, but this shows it in context. your statement here is utterly wrong.





It is indeed. (I do know what I was trying to say but lets just forget it).

Quote:


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.


>>>>> Max...
Fact, he heard a difference ,
which in his experience, of recording audio, was significant... and his experience is not to be discounted or taken as being irrelevant... it is highly relevant.. it is only based on subjective experience and qualitative assessment that anyone can make a value judgement on anything... I submit that his experience is demonstrably long enough and at a high enough level for that value judgement to be given some credit.
a sort of "Expert Witness" in court kind of scenario... and i submit you were NOT there, and cannot make any kind of judgement based on that at all.
I also get the feeling that frankly, you don;t have a heap of experience with "antiquated" digital systems... or with recording "at the bleeding edge" with early Digital equipment... because those differences are very very real, as anyone who has had that experience will tell you...





You are entirely correct that I do not have that experience. I havn't been in the business long enough.

Quote:


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.


>>>>>> Max.
Been there, done that, again....





Agreed.

Quote:


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!


>>>>>>> Max.
Okay this is where I feel the writing style to be the issue rather than the underlying concept or data.... I don't think Hugh would present that idea like this, and I sure as hell wouldn't... it's at this point I had to do a double take... for a while I had the feeling there'd been an editorial cock up with the diagram and text, until Hugh put my head back on the right way round...

Comparing them like this is I suppose, a valid way of presenting the idea... that there can be significant differences between 16 and 24 bit recordings of the same event... but it's definitely not the most concise or easily intelligible way... you CAN compare the two this way... you did it yourself just a moment beforehand... adding 8 zeroes remember? Dither is essentially a solution to a problem, not necessarily central to recording data at either resolution. But the fact that it ignores the existence of Dither as a solution does leave it somewhat short of an ideal description...





Agreed. Just one point: Shouldn't an error percentage be calculated as a difference between the absolute value of the samples (lets say in 24 bits) and the difference the samples have to this value rather than just looking at the number of steps the samples are off and comparing these?

Quote:


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.

>>>>>> left these two ,lumps together to answer in context.

Max.
Yes and err, no... in a very real, but somewhat general sense, ALL IC's are made that way... optical lithography, and ALL AD/DA systems have chips that are IC's at their heart........ AND no, he's not really talking about poxy PC or Mac sound cards, but real discrete Hardware, before the PC/Mac based DAW could so much as fart digital audio., in fact before the DAW we know it today existed .





Ok, I did not find that Mr Nichols made it clear in his article that he was purely talking about these older converters when he started talking about 16 and 24 bit recording. At least not clear enough. Hence my questions.

Quote:


The article was not meant as an educational discourse on cutting edge AD/DA , or PC hardware, but as a piece of useful background knowledge, that might one day save someone's bacon... . imagine if YOU had been given the job of re-archiving all this material.... would you have gone for the straight digital to digital transfer, that would seem so obvious a ploy , had you not known about some of the stuff contained in this article.... and others of it's type.... only those with experience with the systems of the day would really understand this without having to be told it... and there are increasingly few of them left and fewer still working almost at "tape op" level... archiving is not usually considered the preserve of the Grammy winning , working recording engineer.... and VERY few college courses teach anything about such "antiquated" hardware or it's operational idiosyncrasies.





No, I would have done some research. Anyway, I am guessing that these machines don't interface with any modern connectors and protocols. I would assume anyone presented with the challenge would be obliged to do some research simply because of this.

Quote:


The latter sections really need no comment... initial reactions to your post were about the manner of posting and the overtly aggressive "I'm oh so right and he's oh so wrong and should be hung" style...
all of which were pretty much to the point and fair comment... take it like a man and fess up to that.....





Yes, I hope I have just done that.


Quote:



best regards
Max





And to you. Thanks for taking the time to respond.

Cup in a tea storm

UnderTow


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Anonymous
Unregistered




Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288482 - 28/04/06 05:03 AM
Quote UnderTow:

First up, I'll say that giving up was a good idea. It let me actually think about the content of my post and the tone I was using in that post.

I was in a very bad mood due to stuff that had nothing to do with anyone on this forum or anyone writing for SOS. I should never have let that spill into this forum. Or rather, I should never have posted anything in that state of mind. It clouded my judgement to say the very least!

I apologise to everyone here and Mr Nichols in particular.






That's ok just take it out on Hugh(that's what I do) and he'll give you a whooping,educate you and also make you feel good at the same time.
You think mag is called SOS for nothing!


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Feefer
member


Joined: 10/04/03
Posts: 441
Loc: CA, USA
Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288483 - 28/04/06 05:40 AM
And due credit to Undertow for being big enough of a person to admit he was confused on this. The guy shows some hope, after all.

Wouldn't on-line forums be much more pleasant if more people had the courage to put their egos aside long enough to utter that little sentence: "I'm sorry. I was wrong"?

Chris

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


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Lars Farm



Joined: 11/11/04
Posts: 66
Loc: Sundsvall, Sweden
Re: Nichols article on 24 bit recording WRONG! new [Re: Crow]
      #288501 - 28/04/06 07:56 AM
Quote Crow:

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




It was in response to what you snipped from your own statement:-"My choice of sine waves in my example was unfortunate; replace that with a complex waveform and my point will be clearer.". Fourier shows that these are the same.


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Hugh RobjohnsAdministrator
SOS Technical Editor


Joined: 25/07/03
Posts: 22425
Loc: Worcestershire
Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288515 - 28/04/06 08:23 AM
Quote UnderTow:

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




But what is an 'amplitude resolution scale' ? This is a meaningless phrase to me (and I suspect everyone else here... ) I suspect this is the core of our communication problem

Quote:

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




Yep, with you so far...

Quote:

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.




Yes it does... but who or what is going to be flipping bits. This concerns data errors in recorded or transmitted files, and has nothing whatevere to do with tht intrinsic resolution of 16 or 24 bit data. I've commented on this area of confusion before...

Quote:

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.




To be fair, dithering (or not) doesn't really affect Roger Nichol's assertions or arguments.

Quote:

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




Arf yourself! There will always be quantisation errors, That is inherent when you quantise something because of the discrete nature of the quantisation levels. When the system is correctly dithered, those errors are distributed randomly, instead of being related to the signal amplitude, and hence manifest as noise instead of programme-related distortion.

I don't want to be rude, but there do appear to be several gaps and confusion in your understanding of the core concepts here. But ;et's plough on and try to resolve them...

Hugh

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


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Hugh RobjohnsAdministrator
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288525 - 28/04/06 08:30 AM
Quote UnderTow:

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.




I can only go by the evidence of what you have written. And from that there appears to be some misunderstandings and confusion. Maybe it is a language or use of language problem... but if we can find a consistent and common terminology perhaps we can all reach a common understanding too.

Quote:

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.




I can certainly agree with the first point. As for your mental gymnastics, I think it is further clouding and confusing the subject, not adding clarification.

Quote:

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.




In the nicest possible way, I'm yet to arrive at that conclusion

Hugh

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


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Hugh RobjohnsAdministrator
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Joined: 25/07/03
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288529 - 28/04/06 08:43 AM
Quote UnderTow:

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.




But we are yet to find a cohesive, well reasoned argument as to what is conceptually or technically wrong in Mr Nichol's arguments. I've not found anything and I have been looking hard...

I agree that the quoted numbers for step size differences in his text and diagrams seem a little abstract, and his percentage error calculation seems unfounded, but the principle of proportionate quantisation errors at LF and HF between 16 and 24 bit quantising seems to me to be reasonable.

Personally, Im not convinced that bass is conveyed significantly better than treble, subjectively, when using 24 bits. But I am convinced that everything generally sounds better... and obviously for the same core reasons. The resolution is considerably improved.

Quote:

I am saying that he should not write deep technical articles about digital audio.




Again, to be fair, it is not a deep technical article. He is describing a subjective effect he noticed, and his attempts to find a reason for that effect. As far as I can see, his technical arguments broadly hold water. I will happily agree that his explanations are less than perfectly clear, but as you have discovered yourself, UnderTow, it is actually very hard to get across complex technical issues in print.

hugh

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


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Hugh RobjohnsAdministrator
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Re: Nichols article on 24 bit recording WRONG! new [Re: Crow]
      #288545 - 28/04/06 09:08 AM
Quote Crow:

Can’t a low frequency signal be much more accurately represented than a high frequency one using the same sample rate?




No, it can't. But that's not the point of Nichol's argument.

What he is trying to argue is that since a low frequency waveform exhibits only relatively small changes in amplitude between adjacent samples, the aditional amplitude resolution af 24 bit quantising allows these small amplitude changes to be recorded with a smaller error (as a percentage of the amplitude change) than the equivalent HF signal.

Let me try to explain that more clearly...

For the sake of argument let's say that, when quantised to 16 bits, the first sample is rounded down to be quantised to some specific level and the next is rounded up to adjacent quantising level above it. (This rounding is, of course, the quantising error.)

However, if quantised to 24 bits, there will be 255 more quantising levels in between each and every 16 bit quantising level.

When quantised to 24 bits, then, the first sample might be coded to a quantising level which is now, say, 36 levels above the level corresponding to the 16 bit quantise level. There will still be a degree of rounding up or down involved, but clearly the amount of rounding error is now much smaller because the quantisation levels are much more closely spaced.

The second sample might be coded to a level which is 48 levels below that corresponding to the upper 16 bit quantising level.

So, if we measure the distance between the first and second samples described above, in terms of 24 bit quantising intervals, when quantised to 16 bits the distance is 256 steps -- because that is the spacing between the equivalenmt 16 bit quantisation levels.

While when quantised to 24 bits, the gap between these two samples is 172 (256-48-36).

So there is a relatively large percentage error here. The amplitude difference should be 172 'steps', but when quantised to 16 bits, it comes out at 256 steps.

256/172 suggests an error of 149%

In contrast, although the same kind of error mechanism exists for HF signals, there nature is such that adjacent sample are likely to be spaced over many more quantising levels in a 16 bit system.

Again, for the sake of argument, let's say that the first sample is rounded down to one quantising level, and the next sample is rounded up to a quantising level which is 35 16 bit levels higher. If we convert that amplitude difference to 24 bit quantising levels, remembering that there are 255 additional levels between each 16 bit quantising level, 35*255 = 8925 levels.

When quantised to 24 bit accuracy, let's say that the first sample ends up 36 levels higher than the 16 bit rounded value, and that the second sample is 48 levels below the 16 bit value -- in other words, exactly the same size of error as we had inthe LF example previously.

So the step difference between samples in the 24 bit system is 8925-36-48 = 8841 levels.

So the percentage error between 16 bit quantised values and 24 bit quantised values is 8925/8841 = 0.95%


That's it. That's the argument Nichols is making. 24 bit quantising typically results in much smaller percentage amplitude errors for LF signals, than HF signals.

From that premise, he argues that the increased resolution of 24 bit coding is more audible for LF sigals than HF signals, and that agrees with his subjective impressions.

Where is the problem? I'm intrigued to know

hugh

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


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gerard



Joined: 07/02/05
Posts: 2608
Loc: London, UK
Re: Nichols article on 24 bit recording WRONG! new [Re: Hugh Robjohns]
      #288548 - 28/04/06 09:16 AM
Quote Hugh Robjohns:

[Where is the problem? I'm intrigued to know

hugh





the banjo?

har har...


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BigAl
Just The Bass Player


Joined: 24/01/02
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Re: Nichols article on 24 bit recording WRONG! new [Re: Hugh Robjohns]
      #288556 - 28/04/06 09:27 AM
QUOTE:"UnderTow, it is actually very hard to get across complex technical issues in print."

I would agree Hugh.
I don't know if it really matters that much anyway to most on this forum. The complexities of digital audio are for the boffins who make the gear (convertors, recorders, software, etc..) and while it is a very interesting subject, this discussion proves that unless you know it in all its detail, I reckon we are better off using this wonderful gear which the boffins did invent, comment on the merits and shortfalls and continue to make music.
For my money, we see far too many posts about people judging the sound quality of various bits of gear by the technical spec (and sometimes brand name) - and not the actual sound itself.
The sound is always the most important thing.
Many people on this forum are fairly new to recording. They should be given good practical advice on getting the best from their gear rather than be bombarded with lots of technical mumbo-jumbo. This only adds to the confusion.

PS. If you put a bass guitar through a 16-bit guitar POD, in my opinion it actually gives you a nice tidy bass sound - from the persepctive of the bottom end.

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Hugh RobjohnsAdministrator
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288557 - 28/04/06 09:27 AM
Quote UnderTow:

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




He might be... but I still can't see the flaw in the argument. I can see the flaw in his numbers, but not the argument...

Quote:

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.




What he was trying to do was find an explanation to justify his subjective impressions: namely that the LF quality is greatly improved by 24 bit quantising (compared to 16 bit quantising) -- and far more so than the HF quality.

Now that might actually be a point worth debating in its own right. Do others here agree with that?

His (personal) theory for why there might be a greater improvement in LF resolution than HF resolution is an interesting one, and as far as I can see, is hard to argue against....

Quote:

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.




Er... it was (and is) perfectly compatible. What he was explaning was that the early A-D converters weren't very linear, and that each A-D was matched (by hand) to the respective D-A to try to minimise linearity errors between analogue in and analogue out. The A-D linearity errors are recorded on tape for all time in digital form, and that's why he was able to get better quality by transferring via the analogue domain and carefully optimising the D-A converter set up.

Which is all very interesting, but not actually relevant to his main issue which was about the subjective bass improvement he noticed when working with 24 bits.

Quote:

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




Seems clcear and accurate to me.
Quote:

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




... We all have our own opinions, and you are entitled to yours. But if you want to assert fault in his claims you'll have to do rather better than you have so far...

hugh

Edited to add: It has (understandably, I hope) taking me a while to wade through this huge thread and asnwer each relevant point. Consequently, this post was written long before I got to the end of the thread to realise that UnderTow has finally seen the light, thanks to Max's efforts.

So apologies for my continued battering long after the white flag had been raised! -- but I hope the further explanations and clrifications have helped others equally as confused about the claims and arguments.

hugh






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BigAl
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Re: Nichols article on 24 bit recording WRONG! new [Re: Hugh Robjohns]
      #288558 - 28/04/06 09:28 AM
"Where is the problem? I'm intrigued to know"

Ignorance?

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Sam Inglis
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Re: Nichols article on 24 bit recording WRONG! new [Re: BigAl]
      #288580 - 28/04/06 09:49 AM
Quote BigAl:

"Where is the problem? I'm intrigued to know"





Hugh and I had a long discussion before we put this article into print. I think both of us were quite surprised by the argument Roger Nichols makes about bass sounding better, but neither of us could fault his reasoning.

Thinking about it, though, there is one thing I'm not sure about. The argument makes sense in the context of a solo bass instrument with no high-frequency content, where the rate of change of the waveform is uniformly fairly slow. But if that is the explanation for bass sounding better at 24-bit, I can't see why the improvement would be noticeable in the context of a mix. After all, the waveform for a full mix typically displays a fairly rapid rate of change most of the time, meaning that the percentage error is low most of the time.


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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288583 - 28/04/06 10:04 AM
well I still think you should have gotten http://www.russandrews.com in to advise on the mouthwateringness of the resulting bass

Actually, when *ARE* you going to do an interview with Russ Andrews?

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BigAl
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Re: Nichols article on 24 bit recording WRONG! new [Re: Sam Inglis]
      #288589 - 28/04/06 10:08 AM
Sam,
It was Hugh who said that.

But on the subject - many of the arguments for using different mic's, preamps, bit-rates etc... only become apparent on solo testing and A/B tests etc...
In full mixes many of these differencies become less important, and there'e something called creativity which at times can upset the rule book.

PS. Saying the bass was better is subjective, regadless of the technical reasoning, even it was 'true' on paper (if you know what I mean).

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Re: Nichols article on 24 bit recording WRONG! new [Re: Hugh Robjohns]
      #288616 - 28/04/06 10:42 AM
Quote Hugh Robjohns:


He might be... but I still can't see the flaw in the argument. I can see the flaw in his numbers, but not the argument...





Well look at it this way: what he's saying is that a bass signal with the addition of some quantisation noise is going to sound worse than a treble signal with the same noise. That noise is pretty much like adding white noise to the signal. So would a bass signal with -96dB white noise added sound somehow much worse than a treble signal with the same noise added?

Well maybe there could be some psycoacoustic arguement about this, but then the same would hold true for analog equipment - the bass should sound even worse than the 16 bit digital signal because the noise floor is higher.

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Steve Hill
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288631 - 28/04/06 10:58 AM
I still want to hear evidence that the banjo was worth recording. People who make sweeping statements like that should put up or shut up!

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Crow



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Re: Nichols article on 24 bit recording WRONG! new [Re: Lars Farm]
      #288645 - 28/04/06 11:05 AM
Hugh, thanks very much for your detailed reply. I need to run my slide rule over the numbers and iron my brain before digesting it all; I’ll get back to you.


Quote Lars Farm:

Quote Crow:

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




It was in response to what you snipped from your own statement:-"My choice of sine waves in my example was unfortunate; replace that with a complex waveform and my point will be clearer.". Fourier shows that these are the same.



Sorry if I was a bit blunt with you, but I’m interested in the article being discussed here and this thread already has a very poor SNR, so I was hoping we could keep it on topic. In the context of the article being discussed here, I couldn’t see the relevance of your posts.


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BigAl
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Re: Nichols article on 24 bit recording WRONG! new [Re: Steve Hill]
      #288647 - 28/04/06 11:06 AM


"On top of old smokey...."

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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: Hugh Robjohns]
      #288688 - 28/04/06 12:17 PM
Quote Hugh Robjohns:


Where is the problem? I'm intrigued to know

hugh




Could it be that while the quantization error for each sample is larger for low frequency waves, the number of samples during the cycle of the wave is much bigger so the error averages out? So in practise, over the whole cycle of the wave, we have the same error percentage over the whole cycle?

There is a relationship between the number of samples during one cycle of a wave and the percentage of error due to quantization at each sample. Maybe someone that knows the exact maths can tell us what that relation is. Hugh?

Also, as the data goes through a reconstruction filter in the DAC, the graphs in the article seem to be a bit misleading. The angles (any angles) in the graph imply (infinitely) high frequency content which isn't realistic AFAIK. How does the quantization error in the waveform look at each point (including points between samples) after reconstruction?

I really do like the waveform preview in Audition/Cool Edit as it shows the wave after reconstruction. You can even see inter sample peaks going over 0 dB FS on some material if you zoom in enough but I disgress.


Hugh, you write the following:

Quote:


There will always be quantisation errors, That is inherent when you quantise something because of the discrete nature of the quantisation levels. When the system is correctly dithered, those errors are distributed randomly, instead of being related to the signal amplitude, and hence manifest as noise instead of programme-related distortion.





Assuming a theoretical ideal system with ideal TPDF dither, are we not effectively increasing the resolution of the system by linearising the quantization steps and trading that in for noise? In other words, arn't the quantae steps "removed" for all practical purpouses?

If I take a 16 bit recording, convert it to 24 bit, reduce the volume by 96 dB or more, convert back to 16 bit with TPDF dither applied and then normalise, I still hear the original recording below the dither noise.

At the point in this process after the volume reduction by 96 dB or more, the entire signal is encoded in the lower 8 bits of the 24 bit words (bits that don't exist in a 16 bit word). Yet, after bit reduction with dither, the original signal is still there. Doesn't this support my comment that the quantization distortion is removed (but traded in for noise)?

And if what I describe above is correct, isn't it so that discussing this topic without including dither is a bit pointless?

UnderTow


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Mowens800



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Re: Nichols article on 24 bit recording WRONG! new [Re: Hugh Robjohns]
      #288691 - 28/04/06 12:20 PM
Quote Hugh Robjohns:


However, if quantised to 24 bits, there will be 255 more quantising levels in between each and every 16 bit quantising level.

hugh




What happens when you then turn it back to 16bit to put on a CD. The level the sample is, cannot be represented in 16bit format? Is that what dither is about? Do the articles undertow mentioned hold the answers?


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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288697 - 28/04/06 12:39 PM
Quote UnderTow:


Could it be that while the quantization error for each sample is larger for low frequency waves, the number of samples during the cycle of the wave is much bigger so the error averages out? So in practise, over the whole cycle of the wave, we have the same error percentage over the whole cycle?





I don't think it's quite that, but nearly. What he's done by looking at a one sample interval is to compare the high frequence component of the quantization noise to the high frequency component of the signal (which is obviously lower in a bass than a treble sound).

You could make a opposite arguement: for treble signals sounding worse if you compare the low frequency component of the quantisation noise (which is just as large as the hf component) to the low frequency component of the signal. But you couldn't do that with a little picture that everybody could 'understand' (and so be mislead by).

In fact you could make a better case that way because treble signals in music are at a much lower level than bass signals (something else the article doesn't mention).


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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288746 - 28/04/06 02:01 PM
Quote UnderTow:

Could it be that while the quantization error for each sample is larger for low frequency waves, the number of samples during the cycle of the wave is much bigger so the error averages out? So in practise, over the whole cycle of the wave, we have the same error percentage over the whole cycle?




Hmmm... time averaged noise... now that is an interesting concept...

Quote:

Also, as the data goes through a reconstruction filter in the DAC, the graphs in the article seem to be a bit misleading. The angles (any angles) in the graph imply (infinitely) high frequency content which isn't realistic AFAIK.




er... it was a deliberately simplistic diagram intended to explain the difference in quantising errors in an A-D. Wave reconstruction in the D-A isn't relevant to that argument.

Quote:

How does the quantization error in the waveform look at each point (including points between samples) after reconstruction?




The quantisation error only exists at the sampling instants, by definition, and at those points the graph as shown is correct. What is technically incorrect is the straight line used to join the dots.... but I can't help thinking you are looking for flaws here when none really exist (or are relevant at any rate)

Quote:

Assuming a theoretical ideal system with ideal TPDF dither, are we not effectively increasing the resolution of the system by linearising the quantization steps and trading that in for noise? In other words, arn't the quantae steps "removed" for all practical purpouses?




Yes, you can certainly look at it that way. But this view point, and my explanation above are not at odds with each other, just alternative ways of looking at the same thing.

Quote:

If I take a 16 bit recording, convert it to 24 bit, reduce the volume by 96 dB or more, convert back to 16 bit with TPDF dither applied and then normalise, I still hear the original recording below the dither noise.




I won't ask why you'd want to, but yes, assuming your DAW has sufficient internal dynamic range, that would indeed be the case.

Quote:

At the point in this process after the volume reduction by 96 dB or more, the entire signal is encoded in the lower 8 bits of the 24 bit words (bits that don't exist in a 16 bit word).




Not strictly true. The full 16 bit original signal would be encoded using floating point maths to retain the maximum resolution. Only at the point of outputting the 24 bit file would it be truncated and dithered to sit in the bottom eight bits.

Quote:

Doesn't this support my comment that the quantization distortion is removed (but traded in for noise)?




Er... it's a very odd way of demonstrating the point, and I'm not sure what you're getting at anyway. Was there ever any doubt that dithering converts nasty quantisation errors into benign noise? I'm not sure what point you are trying to address here. Sorry.

Quote:

And if what I describe above is correct, isn't it so that discussing this topic without including dither is a bit pointless?




I think what you are trying to describe is correct... but no, dither still isn't relevant to the point Nichols was arguing.

He is arguing that an LF signal coded with 24 bits has less quantising error than when coded at 16 bits, but that the proportionate error for HF signals does not change improve anything like as much. That is true. Yes, if the system is properly dithered, those errors are essentially random noise not distortion artefacts, but the facts remain.

Hugh

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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288777 - 28/04/06 02:48 PM
There's a part I don't get:

Ok so we know that there is a difference in the amount of levels 24Bit give us over 16Bit and 24Bit gives us more to represent the sound,but isn't this just part of the A/D D/A conversion aren't there other factors which contribute to the sound that appears at the I/O?
What about the filters,clocks etc?

Would like to know more about how Lo and Hi frequencies are dealt with in analogue and digital domains and what problems they cause.

Thanks

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UnderTow
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Re: Nichols article on 24 bit recording WRONG! new [Re: Hugh Robjohns]
      #288790 - 28/04/06 03:15 PM
Quote Hugh Robjohns:

Quote UnderTow:

Could it be that while the quantization error for each sample is larger for low frequency waves, the number of samples during the cycle of the wave is much bigger so the error averages out? So in practise, over the whole cycle of the wave, we have the same error percentage over the whole cycle?




Hmmm... time averaged noise... now that is an interesting concept...






Well isn't that how we hear things? Isn't that why we talk about RMS values as they are closer representations of how we as humans hear things?

Quote:


Quote:

Also, as the data goes through a reconstruction filter in the DAC, the graphs in the article seem to be a bit misleading. The angles (any angles) in the graph imply (infinitely) high frequency content which isn't realistic AFAIK.




er... it was a deliberately simplistic diagram intended to explain the difference in quantising errors in an A-D. Wave reconstruction in the D-A isn't relevant to that argument.





How so? We can only hear stuff after it has gone through the DAC with the relevant reconstruction filters so I don't see how looking at a plot that doesn't represent the final waveform explains what we do or don't hear.

Maybe I am missunderstanding something here. Would you care to explain?

Quote:


Quote:

How does the quantization error in the waveform look at each point (including points between samples) after reconstruction?




The quantisation error only exists at the sampling instants, by definition, and at those points the graph as shown is correct.






Yes but how does it affect the wave after reconstruction? Surely this is relevant?


Quote:


What is technically incorrect is the straight line used to join the dots.... but I can't help thinking you are looking for flaws here when none really exist (or are relevant at any rate)





I am trying to understand things. At the moment I don't quite understand why the reconstruction filters are irrelevant.

Quote:


Quote:

Assuming a theoretical ideal system with ideal TPDF dither, are we not effectively increasing the resolution of the system by linearising the quantization steps and trading that in for noise? In other words, arn't the quantae steps "removed" for all practical purpouses?




Yes, you can certainly look at it that way. But this view point, and my explanation above are not at odds with each other, just alternative ways of looking at the same thing.





Well if the dithering process linearises the quantization steps, I don't see how the quantization error (when properly dithering) affects the low frequency waves in the way that Mr Nichols seems to be describing. Basicly, I don't understand his explanation and I still doubt it is correct. That might be due to my lack of understanding but that is why I continue this discussion.

Quote:


Quote:

If I take a 16 bit recording, convert it to 24 bit, reduce the volume by 96 dB or more, convert back to 16 bit with TPDF dither applied and then normalise, I still hear the original recording below the dither noise.




I won't ask why you'd want to, but yes, assuming your DAW has sufficient internal dynamic range, that would indeed be the case.





Obviously to try and understand things. Not something I would do in every day recording.

Quote:


Quote:

At the point in this process after the volume reduction by 96 dB or more, the entire signal is encoded in the lower 8 bits of the 24 bit words (bits that don't exist in a 16 bit word).




Not strictly true. The full 16 bit original signal would be encoded using floating point maths to retain the maximum resolution. Only at the point of outputting the 24 bit file would it be truncated and dithered to sit in the bottom eight bits.





Indeed and that is why you have to save the intermediary file as a 24 bit file to make sure that SF is not just using a 32 bit float intermediary file. (Sorry for not claryfing that in my original description).

Anyway, even with saving the file as 24 bit file, the signal is still there after reopening the file, bit reducing to 16 bit and normalizing.

After conversion to 16 bit, the entire signal is encoded in the lowest 2 bits of the 16 bit format. (2 bits because TPDF has a 2 bit amplitude peak to peak). In other words, the quantization steps have been linearised and are not actual steps any more (and we trade this in for noise).

Do you see what I am driving at? If I am completely wrong, please explain me where my misunderstanding lies.

Quote:


Quote:

Doesn't this support my comment that the quantization distortion is removed (but traded in for noise)?




Er... it's a very odd way of demonstrating the point, and I'm not sure what you're getting at anyway. Was there ever any doubt that dithering converts nasty quantisation errors into benign noise? I'm not sure what point you are trying to address here. Sorry.





I am trying to demonstrate the linearization of the quantae steps by dithering. And if I am correct in my understanding, this contradicts Mr Nichols' explanation AFAICS.

Quote:


Quote:

And if what I describe above is correct, isn't it so that discussing this topic without including dither is a bit pointless?




I think what you are trying to describe is correct... but no, dither still isn't relevant to the point Nichols was arguing.





I don't get this. If the steps are linearized, then his explanation is incorrect. Again, I might be misunderstanding things but I need convincing of that.

Quote:


Quote:


He is arguing that an LF signal coded with 24 bits has less quantising error than when coded at 16 bits, but that the proportionate error for HF signals does not change improve anything like as much. That is true. Yes, if the system is properly dithered, those errors are essentially random noise not distortion artefacts, but the facts remain.

Hugh







Hmmm ... I'm still not convinced. I think his argument is only valid in an un-dithered system. But AFAIK, dither is an essential part of digital audio. If we leave it out, we are not talking about practical implementations of sampling theory. (Idem for reconstruction filters in the DACs).

UnderTow


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Richard Graham



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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288806 - 28/04/06 03:35 PM
I've just read this entire thread, and it's taken me all day, but I don't understand it AT ALL. I thought digital audio was perfect quality, like CDs? It would help if everyone stopped going on about 'dithering', 'converters', 'bits', 'resolution' and all the rest of it. I mean wtf?? Doesn't anyone around here know ANYTHING about audio?


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Crow



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Re: Nichols article on 24 bit recording WRONG! new [Re: Richard Graham]
      #288818 - 28/04/06 03:50 PM
Quote Richard Graham:

I've just read this entire thread, it's taken me all day, and I don't understand it AT ALL. I thought digital audio was perfect quality, like CDs?



Listening to 'Dueling Banjos' on my portable CD player whilst buggering a goat after drinking a 6 pack; THAT'S PERFECT.


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Re: Nichols article on 24 bit recording WRONG! new [Re: ]
      #288850 - 28/04/06 04:46 PM
Quote Music Manic:

There's a part I don't get:




Oh no...!

Quote:

aren't there other factors which contribute to the sound that appears at the I/O? What about the filters,clocks etc?




Yes, all these things can make a difference. But Nichols was seeking to find a way of explaning a specific phenomena he felt (subjectively) was a benefit to low frequencies when recording with 24 bits (as opposed to 16)

Quote:

Would like to know more about how Lo and Hi frequencies are dealt with in analogue and digital domains and what problems they cause.




Wouldn't we all!

hugh

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Crow



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Re: Nichols article on 24 bit recording WRONG! new [Re: Crow]
      #288864 - 28/04/06 05:09 PM
Quote Crow:

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 I’m not sure that it’s at all a relevant comparison.




I just sat down with a notepad and pencil and worked out some examples that have satisfied my curiosity. Looking at it purely mathematically it does stand out and the numbers seem to shout at me, ‘This is significant’. But, a part of me says, is it really significant? Now that I’m happy with the maths, I’m still wondering whether the comparison has any bearing on how low and high frequencies actually sound at differing bit depths. I don’t have access to equipment to test it out subjectively myself, but I’m now wondering if it’s a common perception that bass frequencies sound ‘better’ when sampled at higher bit depths. And if so, is this improvement more noticeable than it is for higher frequencies?
I guess it comes down just as much to psychoacoustics in the end and I don’t mean by that the aural study of Jason Voorhees’s chainsaw in action.


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Steve Hill
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Re: Nichols article on 24 bit recording WRONG! new [Re: Crow]
      #288872 - 28/04/06 05:20 PM
Quote Crow:

Quote Richard Graham:

I've just read this entire thread, it's taken me all day, and I don't understand it AT ALL. I thought digital audio was perfect quality, like CDs?



Listening to 'Dueling Banjos' on my portable CD player whilst buggering a goat after drinking a 6 pack; THAT'S PERFECT.




Thank you for bringing us back to the essential point of this thread: WHAT ABOUT THAT BANJO?

We have a right to know!

We demand to know!!

We shall not rest until we hear The Banjo Worth Recording!!!

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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288876 - 28/04/06 05:26 PM
Quote UnderTow:

Well isn't that how we hear things? Isn't that why we talk about RMS values as they are closer representations of how we as humans hear things?




RMS measurements don't have an explicit time constant involved and tend to be closer to a measurement of the energy contained in a signal rather than peak amplitude... But yes, RMS levels do generally approximate better to perceived loudness in some reagrds than peak levels. Not sure howe RMS measurements can be related specifically to the concept you were suggesting earlier though.

Quote:

How so? We can only hear stuff after it has gone through the DAC with the relevant reconstruction filters so I don't see how looking at a plot that doesn't represent the final waveform explains what we do or don't hear.




He was trying to explain what is, fundamentally, a very simple point about the relative amplitude of quantisation errors during A-D conversion between 16 bit quantising and 24 bit quantising. That's what the graph is intended to show. It's not the clearest graph I grant you, but I think it more or less does what it was intended to do. The subsequent D-A conversion does nothing to change the principle that he is talking about, and would only cloud the explanation further...

Quote:

Maybe I am missunderstanding something here. Would you care to explain?




I have done, and I'm worn out!

Quote:

Yes but how does it affect the wave after reconstruction? Surely this is relevant?




Not to his argument, no. The relative amplitude of error is casued by the A-D process -- the quantisation accuracy. The D-A process plays no part in that whatsoever.

Quote:

I am trying to understand things. At the moment I don't quite understand why the reconstruction filters are irrelevant.




Because they don't have any effect on the quantisation levels. The filters simply remove the sampling artefacts. They play no role whatsoever in quantisation.

Quote:

Well if the dithering process linearises the quantization steps, I don't see how the quantization error (when properly dithering) affects the low frequency waves in the way that Mr Nichols seems to be describing.




I think it fair to say that his argument has overlooked the effect of dither to linearise the quantisation process. Putting his argument another way, perhaps he is saying that the signal-noise ratio improvement when stepping up from 16 bit quantisation to 24 bit quantisation, is more apparent for low frequencies than high frequencies... and when you look at it that way, it does seem harder to justify....

Quote:

Do you see what I am driving at? If I am completely wrong, please explain me where my misunderstanding lies.




Yes, I do see what you are driving at... I quite understand the action of dither. The pertinent question is the what is the relationship between the dither noise and the signal frequency...

Quote:

I don't get this. If the steps are linearized, then his explanation is incorrect. Again, I might be misunderstanding things but I need convincing of that.




I think I am seeing the light. I have been trying to explain Nichols argument as presented... you are trying to relate it to a real-world converter that is properly dithered. We have been talking cross-purposes. I think I can now see where you are coming from.

The issue, perhaps, is one of how the signal frequency and dither signal interact, because the randomised error signal (dither noise) associated with each sample would be proportionately smaller for LF signals than HF signals in 24 bit quantising when compared to 16 bit quantising. My head is starting to hurt... I need to go away and think more about this...

Quote:

Hmmm ... I'm still not convinced. I think his argument is only valid in an un-dithered system.




Certainly his explanation has assumed that case, and yes, that is not a valid way to conceive a well engineered digital audio system...

But I'm not yet convinced that the intrinsic quantising error won't still have an effect along the lines of his proposal...

Hugh

--------------------
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UnderTow
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Joined: 27/02/03
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Re: Nichols article on 24 bit recording WRONG! new [Re: Hugh Robjohns]
      #288923 - 28/04/06 06:37 PM
Quote Hugh Robjohns:


I think I am seeing the light. I have been trying to explain Nichols argument as presented... you are trying to relate it to a real-world converter that is properly dithered. We have been talking cross-purposes. I think I can now see where you are coming from.





Hurray hurray!

Quote:


The issue, perhaps, is one of how the signal frequency and dither signal interact, because the randomised error signal (dither noise) associated with each sample would be proportionately smaller for LF signals than HF signals in 24 bit quantising when compared to 16 bit quantising. My head is starting to hurt... I need to go away and think more about this...





Ok, I have to think about this too. Please do share your findings when you have had some time to think about this.

Quote:

Hmmm ... I'm still not convinced. I think his argument is only valid in an un-dithered system.




Certainly his explanation has assumed that case, and yes, that is not a valid way to conceive a well engineered digital audio system...





Ok we agree on this.

Quote:


But I'm not yet convinced that the intrinsic quantising error won't still have an effect along the lines of his proposal...

Hugh




Fair enough. I am not convinced either way but I am tending away from Mr Nichols proposition.

UnderTow


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cc.
getting into my stride


Joined: 11/03/03
Posts: 945
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #288936 - 28/04/06 06:57 PM
I give up.

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


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TImellis



Joined: 06/09/04
Posts: 354
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #289007 - 28/04/06 09:17 PM
My "2p worth", I have been warming myself at this gentle blaze for a few days now, but the weather's getting much warmer and I could do without this extra heat.
Any sound in the lower frequencies (other than a sine wave) surely is going to contribute more upper partials that fall within the limits of our hearing range than a higher frequency sound. The more of these that can be portrayed by a higher bit rate, obviously the more accurate (fuller) the sound will be. Surely, don't the bass sounds benefit more from this accuracy simply because they have the advantage (even acoustically) of having more upper partials within the range to capture in the first place?

--------------------
Don't ya jus luvvit?


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TImellis



Joined: 06/09/04
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Re: Nichols article on 24 bit recording WRONG! new [Re: TImellis]
      #289011 - 28/04/06 09:24 PM
. . . you must surely all know the joke about the BANJO player, booked for a New Year's Eve gig.
When it was over, the hotel manager (who liked what he'd heard) gave him an advance booking for the next New Year.
"That's fantastic" said the banjo player "can I leave my gear here?"

--------------------
Don't ya jus luvvit?


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Michael Harrison
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Re: Nichols article on 24 bit recording WRONG! new [Re: Richard Graham]
      #289068 - 28/04/06 11:31 PM
Quote Richard Graham:

I thought digital audio was perfect quality, like CDs? It would help if everyone stopped going on about 'dithering', 'converters', 'bits', 'resolution' and all the rest of it. I mean wtf?? Doesn't anyone around here know ANYTHING about audio?




This has to be a joke, right?

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


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Steve Hill
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Re: Nichols article on 24 bit recording WRONG! new [Re: Michael Harrison]
      #289115 - 29/04/06 06:09 AM
Quote Michael Harrison:

Quote Richard Graham:

I thought digital audio was perfect quality, like CDs? It would help if everyone stopped going on about 'dithering', 'converters', 'bits', 'resolution' and all the rest of it. I mean wtf?? Doesn't anyone around here know ANYTHING about audio?




This has to be a joke, right?




What? The whole thread? I thought a couple of bits made sense.

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


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Richard Graham



Joined: 10/04/06
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Re: Nichols article on 24 bit recording WRONG! new [Re: Michael Harrison]
      #289145 - 29/04/06 09:18 AM
Quote Michael Harrison:

Quote Richard Graham:

I thought digital audio was perfect quality, like CDs? It would help if everyone stopped going on about 'dithering', 'converters', 'bits', 'resolution' and all the rest of it. I mean wtf?? Doesn't anyone around here know ANYTHING about audio?




This has to be a joke, right?




See my other thread, about next month's SOS article:

http://www.soundonsound.com/forum/showflat.php?Cat=&Number=288006&page=0&v iew=collapsed&sb=5&o=&fpart=1#288006

--------------------
"if you don't have much soul left and you know it, you still got soul" - Bukowski


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Wonks
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Re: Nichols article on 24 bit recording WRONG! new [Re: Steve Hill]
      #289169 - 29/04/06 10:36 AM
Quote Steve Hill:

.. I thought a couple of bits made sense.



So Steve, you advocate a 2-bit recording system over a 16- or 24- bit one. Interesting!


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Crow



Joined: 20/04/06
Posts: 86
Re: Nichols article on 24 bit recording WRONG! new [Re: Richard Graham]
      #289173 - 29/04/06 10:42 AM
I have my own methods for spicing up old digital multi-track tapes for CD release. Step back Mr Nichols, please….

Stick your old 3M digital tape reels in the washing machine with a big dollop of Aurial Diabolical. There is of course a lot of debate as to which conditioner to use; I prefer to use Bit-Shitter, as it really gets rid of those NASTY stains.
When it comes to line drying the tapes, it’s been scientologically proven that an East wind will add 3 dB of OOMPH to the underlying BAD ASSNESS of the recordings. A West wind will add 10 chart placings to any recording without a banjo. Recordings with a banjo will of course immediately be returned to the Ku Klux Klan and automatically released on the next volume of Now That’s What I Call Incest Vol X.
A Northern wind will subtly mix in a choir of strangled pygmies at the choruses. The pygmy chorus will be phase-shifted and flanged as per the EEC guidelines on these matters.
A Southern wind will virally mutate your recordings and produce a line full of extended mix tapes, at least one of which will induce epilepsy in guinea pigs.
A full moon is best avoided as it maliciously overdubs a bagpipes solo at the bridge.

Edited by Crow (29/04/06 10:55 AM)


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Studio Support Gnome
Not so Miserable Git


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Re: Nichols article on 24 bit recording WRONG! new [Re: Steve Hill]
      #289192 - 29/04/06 11:52 AM
Quote Steve Hill:

We shall not rest until we hear The Banjo Worth Recording!!!




remind me to drop it round..... I have several recordings to prove that it's possible.... some of them are even musical....

Max

--------------------
Don't get the hump when i tell you it's going to be expensive, it's not my fault , you picked the site/building/room â


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Studio Support Gnome
Not so Miserable Git


Joined: 22/07/03
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Re: Nichols article on 24 bit recording WRONG! new [Re: Studio Support Gnome]
      #289195 - 29/04/06 12:00 PM
some of Stockhausen's followers have written a collaborative piece called.


"Sonata for car crusher and Gross (144) Banjos"



--------------------
Don't get the hump when i tell you it's going to be expensive, it's not my fault , you picked the site/building/room â


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Crow



Joined: 20/04/06
Posts: 86
Re: Nichols article on 24 bit recording WRONG! new [Re: Studio Support Gnome]
      #289196 - 29/04/06 12:09 PM
Quote Max The Mac:

Quote Steve Hill:

We shall not rest until we hear The Banjo Worth Recording!!!




remind me to drop it round..... I have several recordings to prove that it's possible.... some of them are even musical....
Max




I once heard John Mclaughlin playing a banjo in his inimitable machine gun stylee. This wasn’t busking in Alabama or Finsbury Park, but at a gig with Jack Bruce, Bilham Cobly and Stu Goldberg. Whether that was worth recording is down to your personal peccadilloes.


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Studio Support Gnome
Not so Miserable Git


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Re: Nichols article on 24 bit recording WRONG! new [Re: Crow]
      #289214 - 29/04/06 01:04 PM
in actuality, i have recorded and mixed several albums of "traditional" (mostly Irish) music, complete with Banjos, and while I find the Banjo as humorous as the next guy, i'd have to admit that these albums were worth recording.... including the Banjo parts....

make of that what you will.

--------------------
Don't get the hump when i tell you it's going to be expensive, it's not my fault , you picked the site/building/room â


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The Producer



Joined: 17/11/04
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #289383 - 29/04/06 10:20 PM
'Steely Dan Play Acoustic Live on Banjo' engineered by the very capable producer, yet somewhat (dither)ing writer, Roger Nichols.

I'd pay good money for that.

What a wonderful world it would be.


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Steve Hill
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #289764 - 30/04/06 11:03 PM
Caught a bit of Eagles in concert, 1973-vintage, on BBC4 on Friday night. Jeez, that boy could play a banjo...

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


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Hugh RobjohnsAdministrator
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #294491 - 10/05/06 11:38 AM
Apologies for the time it has taken me to get back to this thread. Illness in the family...

Anyway, have given this much more thought now, and talked to a few boffins and eggheads about it. The general concensus is -- as Under Tow was suggesting -- that Roger Nichols' arguments don't apply to a correctly dithered system. The quantisation errors he is comparing between 16 and 24 bit systems simply don't exist when the system is dithered correctly.

A correctly dithered system is inherently linear and quantisation errors translate into a constant noise floor -- which is lower or smaller for 24 bit systems than 16 bit systems.

Therefore 24 bits does sound better than 16 bits because the noise floor is lower. Some may perceive the improvement to be greater at low frequencies than high frequencies.... but Nichols' quantisation arguments don't provide the explanation I'm afraid.

Thanks to UnderTow for his patience and perserverence in arguing his point.... after a confused and frustrating start I think we got there in the end

hugh


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PrinceXizor
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Re: Nichols article on 24 bit recording WRONG! new [Re: Hugh Robjohns]
      #294674 - 10/05/06 04:24 PM
I'll just add as a post-mortem (as an engineer and problem-solver)...this whole thread illustrates the keen importance of:

1. Clearly defining your assumptions and givens.

2. NOT applying conclusions made under one set of assumptions to another set of assumptions.

In this case, a conclusion based upon an undithered system has been applied to a dithered system.

You'd be awfully surprised how many "problems" are solved by forcing someone to go back and state all of their assumptions and givens to their argument.

Anway, I'll get off of my teacher/soap-box now.

P-X

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


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James Lehmann



Joined: 17/05/05
Posts: 2011
Loc: Europe
Re: Nichols article on 24 bit recording WRONG! new [Re: PrinceXizor]
      #294731 - 10/05/06 05:10 PM
Quote PrinceXizor:

...this whole thread illustrates the keen importance of:

1. Clearly defining your assumptions and givens.

2. NOT applying conclusions made under one set of assumptions to another set of assumptions.



Agreed 100%! Erroneous logic like you describe is applied with alarming abandon to discussions everywhere on the Internet - at the least sowing mass confusion, and at the worst causing serious misinformation overload with potentially dire consequences for someone down the line.


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OmarHash



Joined: 12/02/06
Posts: 158
Loc: West Vancouver, British Columb...
Oh my head hurts..... new [Re: UnderTow]
      #302465 - 24/05/06 10:03 PM
I spent the last 10 hours reading this entire thread .... with 15-minute breaks after every 3 minutes of reading. It wasn't the maths and scientific lingo ...... honestly ....... but rather it was that banjo sound invading my head!

Tylenol-3 isn't helping much. Does anyone know where I can get morphine from the black-market? Disassociation of mind and body seems to be the only solution.




OH MY GOD ...... thoughts of banjo mariachi bandits invading my head again!
HELP!!!!!

--------------------
Adopt kids, not styles!!!!!


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Crow



Joined: 20/04/06
Posts: 86
Re: Oh my head hurts..... new [Re: OmarHash]
      #302641 - 25/05/06 10:01 AM
Quote OmarHash:

OH MY GOD ...... thoughts of banjo mariachi bandits invading my head again!HELP!!!!!



If you plan on imbibing ayahuasca any time soon, please let me know so that I can arrange for a phalanx of banjo strumming skateboarding dwarves to enliven your experience

P.S.
No offence intended to skateboarders or banjo players…. or people of restricted growth.


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__
Who's never been here


Joined: 28/11/02
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Re: Oh my head hurts..... new [Re: OmarHash]
      #302647 - 25/05/06 10:08 AM
Quote OmarHash:

I spent the last 10 hours reading this entire thread ....




Sit down and fekin learn, the rest of us have to! Have some Peyote, I find it helps keep me awake.

The Banjo is all part of learning... They pump it in here... subliminal like, they are engineers, they have techniques.


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Andi



Joined: 02/09/04
Posts: 1207
Loc: Berkshire, UK
Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #302686 - 25/05/06 10:53 AM
Shame really that Undertow has alredy issued a rather splendid apology for his obviously correct argument and has therefore morally abdicated the right to bask in the glow of having been correct all along.

Incidentally, I was just going to support you there mate

The only thing I don't get is................oh stuff it, I'm going on holiday tomorrow so I don't care.



A.

--------------------
Andi, www.thedustbowlaudio.com Mixing, Mastering, Audio Editing (and articles) at The Dustbowl Audio


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blueintheface



Joined: 24/11/04
Posts: 645
Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #308709 - 07/06/06 10:01 PM
Wow. I've just read this thread. Phew!

I read the the RN article a month or so ago and its 'logic' has been rattling round my head.

I subsequently started two threads questioning RN's assertions - one on this board - and got a reply from Hugh Robjohns saying 'Let's not start the fight all over again' - and eventually got pointed to this thread. Well, I didn't know this topic had closure - I haven't read anything in SOS (magazine) about the accuracy (or otherwise) of that article.

My observations: most people believed what they read in SOS - and then defended it like it was their religion. And like a religion, they took it at its word, rather than questioning the accuracy of the science behind it.

UnderTow stuck to his guns - and there was a lot of resistance - but he was right, and that seems to be generally accepted now, and maybe even understood. Has anyone acknowledged that? (I've just seen that Hugh did - kudos)

And where has RN been during all of this? Mocking up plug-in UI's in MS Paint?

The best that could come of this is that people allow their accepted beliefs to be challenged - and defend their positions with logic rather than the prejudice they've doemonstrated here.

For some, SOS magazine is a reference, and as such there's a greater responsiblility to the facts and readership at large, rather than to the staff covering each other's asses.

After all, it's not New Labour . . .


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gerard



Joined: 07/02/05
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #308789 - 08/06/06 06:49 AM


can someone please condense all the good bits for me to read...

in particular banjo jokes and jokes about funny plugin names...

thanks dudes...


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__
Who's never been here


Joined: 28/11/02
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Re: Nichols article on 24 bit recording WRONG! new [Re: gerard]
      #308801 - 08/06/06 08:03 AM
Quote gerard:



can someone please condense all the good bits for me to read...

in particular banjo jokes and jokes about funny plugin names...

thanks dudes...




It's a load of old bollocks isnt it? What we have here is a bloke, an American bloke no less. He made a shitload of money designing nuclear bombs [theyre bombs when something goes wrong]. He then made a load more out of working with some of best artists in the world. In fact he made the benchmark digital recording. The one people probably a/b with more than any other... Possibly still unsurpassed...

Here are the credits Fagin's 'The Nightfly'...

Recorded and mixed entirely on 3M digital 32 track and 4 track machines at Soundworks Digital Audio/Video Recording Studios, N.Y., Village Recorders, L.A., and Automated Sound, N.Y.

Chief Engineer: Roger Nichols

There are few people on this forum who wouldnt piss-spunk if he asked them to make his tea for an album. The bloke virtually invented digital recording!

He has now upset a few people by procuring a plug-in company, putting groovy facias on the apps and quadrupaling the price. Something that we will all do given a resurgance in the third party recording studio market, and the cash to buy a cockey facility somewhere.

Will these accusors be better recordists because they have an oak door on their enormous and infinitely well equipped studios? no they won't. But they will quadrupal their prices given the market to support it!

So what we have is a load of jealousy, hypocricy, back-biting, politics and big knobishness. It stinks!!!

Personally, as a long time reader of SoS, i'm extremely pleased that someone of this calibre is writing a column. I think it's a real coup. I like his style, its friendly, a little brash perhaps, but down to earth enough for me to understand.

"Nichols article on 24 bit recording WRONG!" Don't make me laugh... how many global number one benchmark albums have you engineered?

Edited by Hugh Robjohns (08/06/06 11:14 AM)


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blueintheface



Joined: 24/11/04
Posts: 645
Re: Nichols article on 24 bit recording WRONG! new [Re: __]
      #308997 - 08/06/06 01:43 PM
Quote ow:


There are few people on this forum who wouldnt piss-spunk if he asked them to make his tea for an album.




You do realize you can only speak for yourself?

Quote ow:



"Nichols article on 24 bit recording WRONG!" Don't make me laugh... how many global number one benchmark albums have you engineered?





He is right because of a having had a 'global number one', rather than the accuracy of what he says?

Enjoy your church, brother.


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Doublehelix



Joined: 04/12/02
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #309023 - 08/06/06 02:38 PM
Let's look at those list of credits on that album again:


Quote:


Chief Engineer: Roger Nichols
Overdub engineer: Daniel Lazerus
Tracking and mixdown: Elliot Scheiner

Sequencing, percussion and special effecs: Roger Nichols and WENDEL II
Digital maintenance: Wayne Yurgelun, Mike Morongell (Soundworks); Bill Roach and Jiri Donovsky (3M)
Assistant engineers: Wayne Yurgelun, Mike Morongell; also Cheryl Smith and Robin Lane
Original Mastering: Bob Ludwig at Masterdisk, N.Y.C.





So what does a chief engineer do when Elliot Scheiner is the tracking and mixdown engineer, and Daniel Lazerus is the Overdub engineer? There are also 4 assitant engineers listed! Wow!

It is pretty hard to go wrong when you have these 2 guys in your corner, and then mastered by Bob Ludwig to boot! There is no way this album could have sounded bad with that technical crew and then with the musicianship that Donald assembled as well. Once again... "WOW!"

This is not meant to be a condemnation, but a question. What does a chief engineer do in a situation like this??? Obviously something pretty special, otherwise Donald would not have listed him as such.

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


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UnderTow
member


Joined: 27/02/03
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Re: Nichols article on 24 bit recording WRONG! new [Re: Doublehelix]
      #309083 - 08/06/06 04:47 PM
Quote Doublehelix:



So what does a chief engineer do when Elliot Scheiner is the tracking and mixdown engineer, and Daniel Lazerus is the Overdub engineer? There are also 4 assitant engineers listed! Wow!





I can think of two things off the top of my head: Having the money to buy the studio in the first place and/or being good friends with the band and thus arranging to have the credits. This is speculation of course.

Anyway, the guy is still wrong in his article.

UnderTow


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tipex
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #309126 - 08/06/06 06:17 PM
"I can think of two things off the top of my head: Having the money to buy the studio in the first place and/or being good friends with the band and thus arranging to have the credits. "

now that's what I call fighting talk! You've gone from pointing out a technical error (possibly) to inferring that the guy has no skills at all! I don't think so. We simply must have donald or walter in to settle this


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UnderTow
member


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Re: Nichols article on 24 bit recording WRONG! new [Re: tipex]
      #309209 - 08/06/06 10:35 PM
Quote tipex:



now that's what I call fighting talk! You've gone from pointing out a technical error (possibly)





Make that certainly.

Quote:


to inferring that the guy has no skills at all!




Not at all. It is just idle speculation. There are many other possible reasons for him to have those credits. I just gave two possibilities. Make of that what ever you whish.

I was not in the sessions nor do I know anyone that was so I have absolutely no idea what Mr Nichols did or didn't do. Still, if someone else did the actual recording AND the mix it does make me wonder, as others, what the title of chief engineer means.

UnderTow


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GJordan



Joined: 09/06/06
Posts: 2
Re: Nichols article on 24 bit recording WRONG! new [Re: seablade]
      #309284 - 09/06/06 06:00 AM
I also got pointed to this forum and thread as the 'answer' to the question of this article having 'technical inaccuracies'.

I've read the article, and this thread, and other forums/comments on this. (I'm glad Undertow kind of got things communicated eventually, but his terminology is far from standard, and kudos to Hugh for his efforts). I'm referring specifically to the section titled 'The Bits'.

I don't really see a good explaination. I also don't see any comments in the June SOS (US/International version at least). All I see is effectively "what the article says is irrelevent with proper dither", but I also say that what the article says is on shaky ground from the start and right through.

I don't see any responses to some of the biggest technical problems I see in it, although one or two people hinted at them, wondering, but no responses.

My first comment, he is doing error as a percentage of the 'step size', since when has this been a valid measure of error for audibility? Error signals/noise are normally in absolute terms or relative to the signal level, not the instantaneous rate of change of 'voltage' level.
This may give you bigger numbers in some places with a high frequency signal, but remember the peaks and troughs of the waves have very small differences between samples. The errors he talks about, are basically at the same signal level irrespective of frequency.

Also, his whole premise that the noise level is 'less', and hence less audible, for higher frequencies is, er, inaccurate.

Let's ignore dithering for now, and just focus on the quantization noise aspect, as the article does (rightly or wrongly).

Standard signal analysis methods look at 3 main signals:
1. the error(noise)-free perfect desired signal (analog input signal for this situation)
2. the quantized signal (A/D output - that inherently defines signal between samples)
3. the error (noise) signal.
Then:
A/D output = perfect signal + error signal
i.e. A/D output is the same as adding (mixing) a noise signal to your 'prisine' signal.
This error signal is effectively a real signal that you can hear. If you regard 24-bit as perfect (or effectively perfect), you can hear this error signal for 16-bit quantisation by simply having a full 24-bit signal, truncating it to 16-bit then taking the difference between this and the original.
Math does the same.
You can then analyse the error signal.
To be least offensive on the ear, this noise signal should be truly random, with no tones, or other strange noises. This is why you use dither, to make this error signal random noise (flat frequency content).
With straight truncation the error is not audibly nice, and has artifacts that are dependent on the original signal.

Anyway, getting back how this relates to what the article says, for 16-bit quantization compared to 24-bit, the error signal, which is the same as adding a real/desired signal to the original, itself has a max peak-to-peak amplitude of 256 24-bit LSBs (i.e. 256 x the least significant bit voltage value for 24-bit).
The point is that this signal amplitude can be the same across all frequencies.
What the article is suggesting is that when you add this noise to a high frequency signal you hear it less than with a low frequency signal. If you have a regualr quiet mid-range sound that you mix with just a louder low frequency signal, then mix it just with a louder high frequency signal
does that mid-range sound sound quieter with the high freq added, than with the low freq added? From the article's logic it would.

Now back to dither a little bit, and A/D at 16-bit (i.e. old stuff), if your input signal has a better noise floor than the 16-bit converter, then without 'analog dithering', the A/D is going to be dominated by equantization noise, as the accuracy of the input signal is effectively being truncated (just like truncating 24-bit). If you don't purposefully add noise to the incoming signal to a simple, 16-bit A/D, you'll get the signal dependent quantization noise (i.e. bad noise). Plus older A/D had low end problems anyway. But I'm starting to gert of my area of knowledge now - I'm not a device historian, I can't say for sure what old 16-bit A/D did, but it's far better today.

Which kind of ends up with the 'answer' given before: what the article says on this is irrelevent in any system that has dither implemented correctly. But I'll add, please don't apply the reasoning to anything else, as it is fundamentally flawed anyway.

After starting with, what I consider to be a rather shaky premise and dubious error calculation method, the article does make several numerical (can only have multiples of 256 'steps' between 16-bit values), arithmetic (those number do not make 700%) and diagramatic errors (straight lines between samples, the 'low frequency' diagram zooms in to a higher frequency component).

I have a lot of respect for fighter pilots, but I'm not going to give much weight to their explanations of the nuts and bolts of aerodynamics. Roger Nichols is an ace fighter pilot.

Bring on the flames, abuse, or even plain ostracism. I'm unlikely to mind. Although intelligent criticism would be nicer.

G


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gerard



Joined: 07/02/05
Posts: 2608
Loc: London, UK
Re: Nichols article on 24 bit recording WRONG! new [Re: GJordan]
      #309285 - 09/06/06 06:18 AM


...anyone got anymore banjo jokes?


by the way, nicely worded post GJordan... are you new to the forum? don't worry we will soon have you talk'n trash and crap like the rest of us...

har har...


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__
Who's never been here


Joined: 28/11/02
Posts: 6263
Re: Nichols article on 24 bit recording WRONG! new [Re: gerard]
      #309292 - 09/06/06 07:21 AM
Quote gerard:

...don't worry we will soon have you talk'n trash and crap like the rest of us...




Yes, I'm running overnight tutorials. This course is designed to be available when 'you' need it. Here at the ow college of trash & crap we structure our courses around you!. We make it easy to study! Click here for more information, course availability and rates. Book Now! places are limited! Success guaranteed!


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Steve Hill
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Joined: 07/01/03
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Re: Nichols article on 24 bit recording WRONG! new [Re: gerard]
      #309296 - 09/06/06 07:38 AM
Quote gerard:

...anyone got anymore banjo jokes?




Q. What's the difference between a 4-string and a 5-string banjo?

A. A 5-string burns for longer

Q. What's the definition of an optimist?

A. A banjo player with a pager

Q. What's the definition of perfect pitch?

A. Throwing a banjo into a skip/dumpster without touching the sides.

(The old ones are the best)

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


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Hugh RobjohnsAdministrator
SOS Technical Editor


Joined: 25/07/03
Posts: 22425
Loc: Worcestershire
Re: Nichols article on 24 bit recording WRONG! new [Re: GJordan]
      #309325 - 09/06/06 09:16 AM
Quote GJordan:

My first comment, he is doing error as a percentage of the 'step size', since when has this been a valid measure of error for audibility?




I think Roger was simply using his (bizarrely calculated) 'percentages' as a way of highlighting the relative sizes of the proportional errors.

Quote:

Error signals/noise are normally in absolute terms or relative to the signal level, not the instantaneous rate of change of 'voltage' level.




The voltage increment denoted by the least significant bit in a digital quantiser is, by definition, an absolute, fixed amount, with a clearly defined relationship to the peak signal level. Roger wasn't talking about instantaneous changes as such, just the possible level differences between adjacent samples.

As I undertsand it, essentially he was comparing the difference in error signal amplitudes between 16 and 24 bit quantisation... although as we know, in a properly dithered system there is no error signal, only random noise...

Quote:

but remember the peaks and troughs of the waves have very small differences between samples. The errors he talks about, are basically at the same signal level irrespective of frequency.




Sorry, don't follow the first sentence at all. If you compare the area where an analogue signal passes from the negative half cycle to the positive half cycle (the zero-crossing region) then, when sampled, adjacent samples of a low frequency signal in this are are likely to have relatively similar quantisation values -- becaue the rate of change of the signal is reltively low. Adjacent samples of an HF signal are likely to have significantly different quantisation values because the rate of change of the signal is relatively high.

As I see it, this was the fundamental assertion on which the rest of Roger's thoughts were based... and this bit, at least, is correct.

Quote:

Also, his whole premise that the noise level is 'less', and hence less audible, for higher frequencies is, er, inaccurate.




Quite

Quote:

Let's ignore dithering for now, and just focus on the quantization noise aspect, as the article does (rightly or wrongly).




This is treading the same dodgy ground as Mr Nichols... and strictly, we should be talking about quantising error rather than noise, to avoid further confusion. By definition, noise is random whereas quantisation error is statistically related to the input signal. A correctly dithered system exhibits a noise floor. An incorrectly (or absent) dithered system exhibits no noise floor at all, but signal errors (distortions) which are related to the source sound.

Quote:

A/D output is the same as adding (mixing) a noise signal to your 'prisine' signal.




This is getting messy. The output of an A-D is a sequence of binary values. However, in a properly dithered system, the signal encoded in those binary values is, essentially, the source signal plus a very small amount of random noise.

In an undithered system, the encoded signal is essentially the source plus a raft of source-related distortions. The amplitude of those distortions remains fixed (and related to the size of the LSB) -- and so take on a greater significance and audibility as the source signal decreases in amplitude.

Quote:

If you regard 24-bit as perfect (or effectively perfect), you can hear this error signal for 16-bit quantisation by simply having a full 24-bit signal, truncating it to 16-bit then taking the difference between this and the original.




First, in a perfect world, an undithered 24 bit system exhibits the same problems as an undithered 16 bit system...low level signals will still produce quntisation errors and the same distortions. The only difference is the absolute amplitude of those distortions.

However, it's not a perfect world and the inherent gaussian noise generated by the interface electronics will automatically act to dither a 24 bit system into near perfection anyway... making the comparison rather difficult.

Of course, truncating a 24 bit signal to 16 bits strips off any dithering at the 24 bit level, leaving you with an undithered (and hence distorted) 16 bit signal. Interesting to play with, perhaps, but not really relevant to the
practical difference between recording with a properly dithered 24 bit A-D versus a properly dithered 16 bit A-D -- which was the premise of Roger's article.

Quote:

With straight truncation the error is not audibly nice, and has artifacts that are dependent on the original signal.




Agreed...

Quote:

the error signal itself has a max peak-to-peak amplitude of 256 24-bit LSBs (i.e. 256 x the least significant bit voltage value for 24-bit). The point is that this signal amplitude can be the same across all frequencies.




You're saying the maximum error is potentially one LSB, which is correct, and if comparing a 16 bit system with a 24 bit one, the maximum 16 bit error amplitude equates to 256 LSBs in a 24 bit system... yes, with you so far...

Quote:

What the article is suggesting is that when you add this noise to a high frequency signal you hear it less than with a low frequency signal.





ROger was suggesting that if you count up the number of quantising levels crossed (in terms of 24 bit LSBs), the relative proportion of maximum potential error compared to the overall difference in quantising level for two adjacent samples is far greater for LF signals than for HF signals. The pure logic of what he said seems right to me, but the concept itself is deeply flawed and not applicable to practice.

Quote:

If you have a regular quiet mid-range sound that you mix with just a louder low frequency signal, then mix it just with a louder high frequency signal
does that mid-range sound sound quieter with the high freq added, than with the low freq added? From the article's logic it would.




Well, we might have to take noise masking into account here... but really, you are talking about mixing dissimilar, unrelated sounds. He was talking about distortion products generated by and related to the specific source... but we are getting bogged down debating irrelevant points.

Quote:

I have a lot of respect for fighter pilots, but I'm not going to give much weight to their explanations of the nuts and bolts of aerodynamics. Roger Nichols is an ace fighter pilot.




Amusing analogy! I like it...

hugh

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


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Jumpeyspyder



Joined: 20/01/06
Posts: 1251
Loc: Yorkshire
Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #309348 - 09/06/06 09:46 AM
Hi Guys
I've been reading this thread and Rogers article and trying to work out how 24 bit may be superior to 16 bit for bass sounds.

I know reletivly little about digital conversion but I have a theory that may suggest something to support Rogers article.
I don't want to spread missinformation so please correct me if I'm wrong !!!

My thoughts are (assuming no dither);-

What happens to a signal that is between two quantise levels, at the peak of its waveform for a long period (most likely with a slow bass note)

If you look at my

example
(the pink area is a low bit 'recording' and the green area is a higher bit recording)

At the top of the waveform the 'low bit recording' never reaches the next highest quantise level. My understanding is that the top of the recorded waveform will be flat this would introduce a mild distortion which would muddy the sound slightly. The same would happen to higher frequencies but the distortion would be at a higher frequency possibly above human hearing or just making the sound slightly harsher.

Would these distortions (if they exist )be noticeable after dithering??

What do you think ???


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Henry-S
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Joined: 11/07/04
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Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #309389 - 09/06/06 10:55 AM
I hate to make fun of this thread, because it is infact pretty amazing to see the "battle of minds" if you will.

I would say however, we all phrase things differently, we all have our "opinions". But nobody has mentioned

The Flux Capacitor





--------------------
There is nothing Grim about this Reaper
We Fell From The Sky


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UnderTow
member


Joined: 27/02/03
Posts: 317
Re: Nichols article on 24 bit recording WRONG! new [Re: Jumpeyspyder]
      #309427 - 09/06/06 11:47 AM
Quote Jumpeyspyder:



If you look at my

example
(the pink area is a low bit 'recording' and the green area is a higher bit recording)

At the top of the waveform the 'low bit recording' never reaches the next highest quantise level. My understanding is that the top of the recorded waveform will be flat this would introduce a mild distortion which would muddy the sound slightly.





There would be a red block at the top that goes beyond the blue curve up to the next horizontal red line. However, in reality, besides the effect of dithering to linearise the quantae steps, you have a reconstruction filter on the output of the DAC so what comes out of your converters is not a stepped waveform like in your graph but rather a smoothed curve that would look like the blue line.

Quote:


The same would happen to higher frequencies but the distortion would be at a higher frequency




I'm not exactly sure what frequency the distorton would be at but yes, in an undithered system, the quantization distortion is correlated to the input signal.

Quote:


possibly above human hearing or just making the sound slightly harsher.





Assuming that the converters are limited to audio bandwidth, in a proper system, anything above the bandwidth (above human hearing) has to be filtered out by anti-aliasing filters or it will fold back down into the audible bandwidth.

Quote:


Would these distortions (if they exist )be noticeable after dithering??

What do you think ???




Simply put, the idea of dither is to remove the quantization distortion trading it in for broadband noise. Dithering the signal after any quantization distortion is introduced has no effect on that distortion. It is too late.

Dither, anti-alias filters, reconstruction filters etc are integral part of sampling theory. If they are not part of the system, the system is broken.

PS: Check out this article by Nika Aldrich about dither: http://www.users.qwest.net/%7Evolt42/cadenzarecording/DitherExplained.pdf< /a>

UnderTow

Edited by UnderTow (09/06/06 11:52 AM)


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GJordan



Joined: 09/06/06
Posts: 2
Re: Nichols article on 24 bit recording WRONG! new [Re: Hugh Robjohns]
      #309585 - 09/06/06 05:57 PM
Thanks for the comments Hugh. Very civil. See my comments...

Quote Hugh Robjohns:

Quote GJordan:

My first comment, he is doing error as a percentage of the 'step size', since when has this been a valid measure of error for audibility?




I think Roger was simply using his (bizarrely calculated) 'percentages' as a way of highlighting the relative sizes of the proportional errors.



Yes, that's what he's doing, but he's totally misrepresenting the proportional size of the errors, as he's calculating them as a proportion of the change in sample value, whereas he should be showing them as a proportion of the actual sample value. But even more correctly, he should be (as you mention later)looking at the rnage of error values and relating this to signal level. If he did that, you would see no dependency to the slope or frequency of the signal in the way he describes.

Quote:

Quote:

Error signals/noise are normally in absolute terms or relative to the signal level, not the instantaneous rate of change of 'voltage' level.




The voltage increment denoted by the least significant bit in a digital quantiser is, by definition, an absolute, fixed amount, with a clearly defined relationship to the peak signal level. Roger wasn't talking about instantaneous changes as such, just the possible level differences between adjacent samples.

As I undertsand it, essentially he was comparing the difference in error signal amplitudes between 16 and 24 bit quantisation... although as we know, in a properly dithered system there is no error signal, only random noise...



I wasn't really clear here, but I was meaning instantantaneous change = change from one sample to the next. And this is what he's using as his base for calculating the error size. And this 'instantaneous change' is not necessarily a single 24-bit LSB, but many.

Also, he isn't comparing the error signal amplitudes, he's actually comparing the change in error from one sample to the next. Calculating the error signal amplitude doesn't involve any differencing between sample values; the error is the difference between the 'right' sample value and the actual value, for a single sample. For a single sample this is the instantaeous amplitude. If you look at these over many samples you can see how the signal behaves and calculate other useful measures (RMS level, peak-peak max, frequency content, etc.)

Quote:

Quote:

but remember the peaks and troughs of the waves have very small differences between samples. The errors he talks about, are basically at the same signal level irrespective of frequency.




Sorry, don't follow the first sentence at all. If you compare the area where an analogue signal passes from the negative half cycle to the positive half cycle (the zero-crossing region) then, when sampled, adjacent samples of a low frequency signal in this are are likely to have relatively similar quantisation values -- becaue the rate of change of the signal is reltively low. Adjacent samples of an HF signal are likely to have significantly different quantisation values because the rate of change of the signal is relatively high.

As I see it, this was the fundamental assertion on which the rest of Roger's thoughts were based... and this bit, at least, is correct.



Yes, if you compare samples at the zero-crossing region (for a single sinewave), but not if you compare samples at the top of the sinewave, where the voltage is changing from increasing to decreasing (and vice-versa at the botttom). I didn't see anything in the article alluding to having to measure at zero-crossings. But again, my point is that this difference is irrelevent to the level of quantization error signal. This difference does not define the level of the main signal.

Quote:

Quote:

Also, his whole premise that the noise level is 'less', and hence less audible, for higher frequencies is, er, inaccurate.




Quite



Phew, I'm glad we agree on that. I'm just trying to point out why his explaination that it is, is wrong.

Quote:

Quote:

Let's ignore dithering for now, and just focus on the quantization noise aspect, as the article does (rightly or wrongly).




This is treading the same dodgy ground as Mr Nichols... and strictly, we should be talking about quantising error rather than noise, to avoid further confusion. By definition, noise is random whereas quantisation error is statistically related to the input signal. A correctly dithered system exhibits a noise floor. An incorrectly (or absent) dithered system exhibits no noise floor at all, but signal errors (distortions) which are related to the source sound.



OK, a little bit of terminology mismatch here. I'd have to check the technical texts, but I thought the technical definition of noise in a signal, is anything that is not part of the actual signal you are trying to capture/measure. And this is not necessarily random. But, no big deal. I'll be sure to refer to quantization 'noise' as quantization error, and error signal.

Quote:

Quote:

A/D output is the same as adding (mixing) a noise signal to your 'prisine' signal.




This is getting messy. The output of an A-D is a sequence of binary values. However, in a properly dithered system, the signal encoded in those binary values is, essentially, the source signal plus a very small amount of random noise.

In an undithered system, the encoded signal is essentially the source plus a raft of source-related distortions. The amplitude of those distortions remains fixed (and related to the size of the LSB) -- and so take on a greater significance and audibility as the source signal decreases in amplitude.



Let me restate it better, the signal represented by the A/D output (i.e. the perfecty reconstructed signal) is the same as your 'pristine' analog signal (approriately bandwidth limited of course) mixed with the error signal (i.e. the reconstructed version of the error signal). Remember I am not incorporating dither here. I'm just rying to illustrate that the effect of quantising (without dither) is just the same as, in a 'perfect' analog domain, mixing another undesired real signal to your desired signal. Hence, the error signal can be, and should be, analysed as any other signal (which the acticle does not do).

Quote:

Quote:

If you regard 24-bit as perfect (or effectively perfect), you can hear this error signal for 16-bit quantisation by simply having a full 24-bit signal, truncating it to 16-bit then taking the difference between this and the original.




First, in a perfect world, an undithered 24 bit system exhibits the same problems as an undithered 16 bit system...low level signals will still produce quntisation errors and the same distortions. The only difference is the absolute amplitude of those distortions.



Absolutely, my point is that you can tell someting about the nature of the 16-bit quantization error signal by using 24-bit, as the quantisation error signal level of pure 24-bit is 48dB less than the signal you're looking at, the 16-bit quantization error signal. This is a simple 'mathematical' exercise to investigate the characteristics of the 16-bit quantization error signal that can be done in a 24-bit audio editor.

Quote:

Quote:

Interesting to play with, perhaps, but not really relevant to the
practical difference between recording with a properly dithered 24 bit A-D versus a properly dithered 16 bit A-D -- which was the premise of Roger's article.






Er, not quite, Roger's article completely ignores properly dithered signals. I thought this was already understood. Otherwise, he's just looking at random noise levels, not quantization errors.

Quote:

Quote:

What the article is suggesting is that when you add this noise to a high frequency signal you hear it less than with a low frequency signal.




ROger was suggesting that if you count up the number of quantising levels crossed (in terms of 24 bit LSBs), the relative proportion of maximum potential error compared to the overall difference in quantising level for two adjacent samples is far greater for LF signals than for HF signals. The pure logic of what he said seems right to me, but the concept itself is deeply flawed and not applicable to practice.



Jackpot! This should be requoted in it's own entry. Very well put - although I might have said "and not applicable to anything in audio (theory or practice, dither or no dither)." Hopefully I've covered some of the reasons why.

Quote:

Quote:

If you have a regular quiet mid-range sound that you mix with just a louder low frequency signal, then mix it just with a louder high frequency signal
does that mid-range sound sound quieter with the high freq added, than with the low freq added? From the article's logic it would.




Well, we might have to take noise masking into account here... but really, you are talking about mixing dissimilar, unrelated sounds. He was talking about distortion products generated by and related to the specific source... but we are getting bogged down debating irrelevant points.



Right, signal masking does start to come in at this point, but that would be getting too far off into deep technicalities. Ones in which I'm less well versed (that's the domaian of the audio compression research gurus and others).
My main point was that the quantization error signal is a 'real' signal, and since is the prodcut of a non-linear process produces frequency components that are not present in the original signal. But the article is implying that these components are heard less when present with a high frequency signal than with a low one. Of course further investigation would be, what are the frequency components that are produced for different input signals? That is a very complex question. I don't know as much as I'd like about the answer to that question. Be we all know that they don't sound good.

Quote:

Quote:

I have a lot of respect for fighter pilots, but I'm not going to give much weight to their explanations of the nuts and bolts of aerodynamics. Roger Nichols is an ace fighter pilot.




Amusing analogy! I like it...

hugh




Thanks.


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gerard



Joined: 07/02/05
Posts: 2608
Loc: London, UK
Re: Nichols article on 24 bit recording WRONG! new [Re: Steve Hill]
      #309833 - 11/06/06 12:57 AM
Quote Steve Hill:

Quote gerard:

...anyone got anymore banjo jokes?




Q. What's the difference between a 4-string and a 5-string banjo?

A. A 5-string burns for longer

Q. What's the definition of an optimist?

A. A banjo player with a pager

Q. What's the definition of perfect pitch?

A. Throwing a banjo into a skip/dumpster without touching the sides.

(The old ones are the best)





love it!

thanks dude!


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Hugh RobjohnsAdministrator
SOS Technical Editor


Joined: 25/07/03
Posts: 22425
Loc: Worcestershire
Re: Nichols article on 24 bit recording WRONG! new [Re: GJordan]
      #310592 - 13/06/06 08:40 AM
Quote GJordan:

Yes, that's what he's doing, but he's totally misrepresenting the proportional size of the errors, as he's calculating them as a proportion of the change in sample value, whereas he should be showing them as a proportion of the actual sample value.




I take your points, but can't help thinking this is all a bit pointless. The technical content of the article was flawed in so many ways that picking at one small bit seems churlish...

Quote:

I thought the technical definition of noise in a signal, is anything that is not part of the actual signal you are trying to capture/measure.




Hmmm.. That sounds like a mathematician's definition to me. In the engineering world, there has always been a difference between truly random errors -- NOISE -- and programme-related errors -- DISTORTION. In some cases noise can be benign (as in dither noise) and sometimes rather annoying and intrusive. Likewise, disortion is acceptable (or even desirable) in some situations and highly irritating in others. It all depends on the context and the human subjectivity to it. This is audio we are talking about, after all...

Quote:

I'm just rying to illustrate that the effect of quantising (without dither) is just the same as, in a 'perfect' analog domain, mixing another undesired real signal to your desired signal.




But it's not. An undithered digital system is inherently non-linear, whereas an analogue system is (generally) linear. You really can't compare the two in the way you are trying to.

Quote:

Er, not quite, Roger's article completely ignores properly dithered signals. I thought this was already understood.




Yes, agreed, ... but he was treating his analysis as if the systems were working properly when his explanation was based on undithered quantisation errors. Basically what we have here is a large bucket of smelly stuff and we have spent five pages poking at it with a stick to see what animal it has come from! But it really doesn't matter -- it's still a big bucket of smelly stuff, and the best thing we can all do is step away...

Quote:

My main point was that the quantization error signal is a 'real' signal, and since is the prodcut of a non-linear process produces frequency components that are not present in the original signal.




Again, this is all flight-of-fancy stuff. Non-dithered digital systems aren't relevant to the practical use of modern digital systems. Other than when cocking things up when changing word lengths, no one is going to come across undithered digital systems. So the artefacts generated by undithered systems, while 'real' enough in broken sysrtems, don't exist in properly working ones, and so aren't relevant in the practical world either.

Quote:

But the article is implying that these components are heard less when present with a high frequency signal than with a low one.




The article was simply Roger's attempt to justify his entirely subjective impression that bass sounded better when recorded with 24 bit systems than with 16 bit systems.

First, I'm not sure we have any agreement that this is an impression shared with anyone else. "4 bit systems certainly sound better than 16 bit ones, but whether the bass is markedly better than the treble I'm not so sure.

Secondly, as we have agreed, his attempts to find a technical justification have been shown to be deeply flawed, both in fundamental concept and its argument. It's probably best to just draw a line there and move on...

Quote:

Of course further investigation would be, what are the frequency components that are produced for different input signals? That is a very complex question.




But utterly irrelevant since who would want to record audio with a broken, non-dithered A-D converter?

Hugh

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


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Feefer
member


Joined: 10/04/03
Posts: 441
Loc: CA, USA
Re: Nichols article on 24 bit recording WRONG! new [Re: Hugh Robjohns]
      #310856 - 13/06/06 08:51 PM
Quote Hugh Robjohns:

But utterly irrelevant since who would want to record audio with a broken, non-dithered A-D converter?




Exactly. Unless that's all we had in the early 1980's.

Are you saying older 16-bit systems as used in early 80's digital recorders were always properly dithered, completely linear properly-designed systems?

It's been a long time since I read his article (and I'm not digging it up again, Thank You), but I took the two concepts discussed in his article as being related: both pertained to a design problem that plagued early digital recorders, and problems still relevant to anyone in 2006 who deals with digital recordings made on equipment used in the early days of digital audio, or who is familiar with the typical "sound" of these systems.

1) The first issue pertained to problems with archiving, since trying to do a straight digital transfer from 16-bit to 24-bit (as a young engineer might be tempted to do) would be inherently flawed: such data transfer would bypass the D/A converters which were chosen by the designers in an attempt to compensate for the non-linearity of inherently flawed A/D converters. That was his point as to why he did the archival transfers via audio to a 24-bit recorder.

2) I read the second topic (his commenting on the perception that bass sounds better in 24-bit vs 16-bit systems) not as an explanation of the effect those (8) additional bits would make using a "properly-dithered and linear" modern 24-bit vs. properly-dithered 16-bit recorder, but as a statement that the inherent non-linearity of the D/A converters used in older 16-bit machines was not as mitigated as the designers might've hoped, with this as evidence that the 'matched' converter approach apparently had limitations.

In other words, apparently enough error is introduced in the flawed A/D converter that is not overcome by passing thru the "compensating" D/A converter to still allow the types of effects that result in subjectively poorer detail in the low end. The non-linear errors on the front end were still present, regardless of compensation on the A/D end.

Hence the point that people may think they're comparing the quality of 16-bit to 24-bit recorders in a fair manner, but may be making a mistake thinking both machines are properly dithered, as if the ONLY variable they're changing is bit depth. It's obviously not so simple, as they're actually comparing old improperly designed 16-bit systems to modern 24-bit machines.


I don't know if this was his intent, but that's how I read it. Admittedly, if it was his point, he did a poor job of explaining it, as this required alot of reading between the lines (harder to understand than a Steely Dan lyric, I'd say).

Likewise, I don't know if this explanation is even technically valid. I do think some of the technical boffins here got lost in the details, and lost the forest for the trees rather than going with the gestalt of what he was saying. We'll let you guys tear it apart, and see if the concept survives scrutiny.

As another disclaimer, I also don't know if his words were published exactly as submitted, or if someone made attempts to "clarify" his article: it's possible the editor didn't really get his point in the first place, and just made matters worse by trying to 'fix' it! Maybe a statement on this would be helpful.

But anyway you cut it, it's a mess.

Perhaps the better question is why SOS didn't work out these concerns amongst it's editorial staff BEFORE running with an article that some must've felt was full of holes and hot air? Is this not the job of the editors? Why didn't they contact RN BEFORE to clarify what he meant? I know everyone wants to give RN a black eye over this, but it's just as embarrassing to SOS, IMO.

On second thought, maybe we should just move on, as it's really quibbling at this point.

Chris

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


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gerard



Joined: 07/02/05
Posts: 2608
Loc: London, UK
Re: Nichols article on 24 bit recording WRONG! new [Re: Feefer]
      #310874 - 13/06/06 09:55 PM
Quote Feefer:

On second thought, maybe we should just move on, as it's really quibbling at this point.

Chris





unless anybody has got any more banjo jokes they want to share?


anybody?


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Doublehelix



Joined: 04/12/02
Posts: 4162
Loc: USA
Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #311139 - 14/06/06 01:03 PM
Quote:

I know everyone wants to give RN a black eye over this, but it's just as embarrassing to SOS, IMO.





I happen to agree here, and also wondered why it was not scrutinized a bit better before going to print.

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


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UnderTow
member


Joined: 27/02/03
Posts: 317
Re: Nichols article on 24 bit recording WRONG! new [Re: Feefer]
      #311171 - 14/06/06 02:09 PM
Quote Feefer:


Perhaps the better question is why SOS didn't work out these concerns amongst it's editorial staff BEFORE running with an article that some must've felt was full of holes and hot air? Is this not the job of the editors? Why didn't they contact RN BEFORE to clarify what he meant? I know everyone wants to give RN a black eye over this, but it's just as embarrassing to SOS, IMO.





Agreed. I'm sure alot of people that have read this article and don't visit the forum or this thread are confused and/or misinformed.

If SOS have any respect for their readers and correctness of information, a full unconditional retraction in print is in order. Anything less would be very disapointing and an insult to the readership.

UnderTow


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gerard



Joined: 07/02/05
Posts: 2608
Loc: London, UK
Re: Nichols article on 24 bit recording WRONG! new [Re: UnderTow]
      #311320 - 14/06/06 07:39 PM

perhaps we can pick the best banjo jokes for the printed correction?

shall we have a vote on the top 5 banjo jokes?


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Hugh RobjohnsAdministrator
SOS Technical Editor


Joined: 25/07/03
Posts: 22425
Loc: Worcestershire
Re: Nichols article on 24 bit recording WRONG! new [Re: Feefer]
      #311443 - 15/06/06 12:15 AM
Quote Feefer:

Are you saying older 16-bit systems as used in early 80's digital recorders were always properly dithered, completely linear properly-designed systems?




No, definitely not. Many of those early systems were pretty ropey and did the 'digital cause' a lot of damage which we are still fighting today...

Quote:

1) The first issue pertained to problems with archiving, since trying to do a straight digital transfer from 16-bit to 24-bit (as a young engineer might be tempted to do) would be inherently flawed: such data transfer would bypass the D/A converters which were chosen by the designers in an attempt to compensate for the non-linearity of inherently flawed A/D converters. That was his point as to why he did the archival transfers via audio to a 24-bit recorder.




Yes. It was an interesting observation, but an entirely correct one. With that particular recorder's design, the digital data on tape was (slightly) non-linear because of the inherent A-D inaccuracies.

However, I'm less convinced of the validity of finding a replay machine 20 years later in which the D-A converters would compensate perfectly for an unknown set of A-D converters.... It's one thing to optimise each channel's A-D and D-A on a single machine, quite another to match every machine everywhere...

Quote:

2) In other words, apparently enough error is introduced in the flawed A/D converter that is not overcome by passing thru the "compensating" D/A converter to still allow the types of effects that result in subjectively poorer detail in the low end. The non-linear errors on the front end were still present, regardless of compensation on the A/D end.




Hmmm... see my comment above!

Quote:

Hence the point that people may think they're comparing the quality of 16-bit to 24-bit recorders in a fair manner, but may be making a mistake thinking both machines are properly dithered, as if the ONLY variable they're changing is bit depth. It's obviously not so simple, as they're actually comparing old improperly designed 16-bit systems to modern 24-bit machines.




Indeed so. Lots of things have improved in 20 years. Dither is one, but so is the anti-alias and reconstruction filtering, clocking accuracy and stability, and all manner of converter issues such as droop and step non-linearities.

I won't comment on the last question....

hugh

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


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