All this hype about Fs>48khz...

[Alécio Costa]

Hi... With all this hype about Fs>48khz and now every brand having its own hardware running under these magical superior frequencies, I ask myself why since day One of this revolution they did not go for a system running 44khz @ 48 bits, for example? I mean expanding wordlenght to a considerably higher number. Probably everything would have to be rewritten from ZERO.
Nika Aldrich comes to mind at this point to help us clarify this sibject.
Thanks

[wwittman]

It's not "hype" if it sounds better to you.


I won't record at less than 96k anymore unless I have no choice.



Now, "needing" lots of different mic pres for "colour".. THAT's hype!

[theblue1]

I don't see anything wrong with recording at higher resolutions if you've got the resources -- with the proviso that you don't end up having to put your (for instance) beautiful, silky 96 kHz signal through a nasty, gritty, inherently evil 'uneven' sample rate conversion down to 44.1 kHz.

[EDIT: Doh! Even back in '05, good SRC was just as accurate at 'uneven' rate conversion as from 'even multiples.' My bad. And if I say more stupid stuff in any posts below, ignore that too. ]

But, I agree -- and listening tests have strongly indicated -- that increasing sample rate does not have nearly the payoff in discernible, increased sound accuracy as increasing the word length of the samples themselves. (Apogee published a series of tests where listeners overwhelmingly preferred 48 kHz, 20 bit PCM audio to 96 khz, 16 bit. [96/16, of course, never caught on, but it was promoted for some time as the must-have format.] )

[dkatz42]

Originally Posted by Alécio Costa

One of this revolution they did not go for a system running 44khz @ 48 bits, for example? I mean expanding wordlenght to a considerably higher number.

48 bits can be useful in a mix engine (to avoid truncation errors during calculations) but as a format it's arguably useless, as it is beyond the precision of A/D converters (and the noise floor of the analog side.) This would be 288 dB of dynamic range (a ratio involving 28 zeros.) It's not clear that existing A/D converters give you 24 useful bits (144dB) as it is.

[jdunn]

Yes, I was going to post a similar answer. The analog side of the equation simply cannot make use of any more bits, at least not with current converter technology. It's not likely to change any time soon.

Interestingly, Sony DSD has gone the other direction..... 1 bit!

[dkatz42]

Originally Posted by jdunn

Interestingly, Sony DSD has gone the other direction..... 1 bit!

True, but at 2.8MHz. It works out to be about 4x the bit rate of 16/44 PCM, a reasonably good measure of information content.

[theblue1]

Originally Posted by jdunn

Yes, I was going to post a similar answer. The analog side of the equation simply cannot make use of any more bits, at least not with current converter technology. It's not likely to change any time soon.

Interestingly, Sony DSD has gone the other direction..... 1 bit!

Yes, but every time you process your dynamic values with multiplication or division (and that's, like, all the time) you end up with a value that can effectively double the resolution needed to store it. Multiple truncations during processing add alias error on top of alias error -- no matter how small those errors are, as they are 'compounded' in this fashion, there is an ever-greater risk that the errors will ultimately (perhaps after much processing) become audible (assuming one had a comparative baseline).

Anyhow, digital representation will always ultimately be compromised. True linearity is an unattainable grail. But the progress we've made so far has made digital audio a very practical medium for high (if not ultimate) quality reproduction.

It's all good but we should never turn down a few extra bits of resolution if someone's throwing them our way...

[dkatz42]

Originally Posted by theblue1

Yes, but every time you process your dynamic values with multiplication or division (and that's, like, all the time) you end up with a value that can effectively double the resolution needed to store it. Multiple truncations during processing add alias error on top of alias error -- no matter how small those errors are, as they are 'compounded' in this fashion, there is an ever-greater risk that the errors will ultimately (perhaps after much processing) become audible (assuming one had a comparative baseline).

I think we're in violent agreement here--there's no argument that having more bits to work with in the number crunching part of the process is useful. It's just that given the state of the art (and limitations of the analog world and our ears) there's not much to be gained with additional bits at the A/D/A and storage part of the system. Any additional bits will at best be picking up random thermal noise. Building a low-noise, highly linear analog circuit with even 144dB of dynamic range (24 bits) is extremely difficult, and the problems grow exponentially with the bit count.

I've heard it claimed that the current generation of 24 bit converters probably only deliver 20 bits accurately; this may have improved, but there's not much use in adding more bits if they can't be filled with useful data.

[theblue1]

Well, I guess I'm focused not so much on the carrying capacity of the finished product format as the potential for long digital word lenths (deep bit depth) to minimize the problems introduced in level and EQ manipulation when long word lengths produced by multiplication (which is at the heart of much digital processing) are truncated to the internal processing resolution.

It's not so much whether or not a 24 bit or 32 bit format is capable of carrying a high resolution signal (24 bit, with roughly 17 million potential values per sample word, clearly has a darn lot of resolution) but whether a higher internal processing word length might minimize the accrued damage from processing and reprocessing audio files.

I think these issues certainly fit into the move to 'non-destructive' editing in DAW software. I'm sure there are plenty of other old-timers like myself who gritted our teeth every time we ran a process over one of our 16 bit clips back in the 90s. In fact, I used to often keep a 'virgin' track archived. Checking back against it could quickly reveal when destructive editing and processing had taken a toll. More than a few times, I went back to the archive track and started over, once I knew what I wanted from the track.

[dkatz42]

Originally Posted by theblue1

Well, I guess I'm focused not so much on the carrying capacity of the finished product format as the potential for long digital word lenths (deep bit depth) to minimize the problems introduced in level and EQ manipulation when long word lengths produced by multiplication (which is at the heart of much digital processing) are truncated to the internal processing resolution.

Yes, that's what I've been saying. Keep the intermediate results with extended resolution, then truncate at the end. 64 bit floating point hardware should bring us to nirvana (56 bits of resolution and 1500 dB of dynamic range.)

My point is that there's very little value for additional bits at the transition to/from the analog world. The samples coming in are padded with zeroes to extend them to the internal wordlength, and the samples going out are truncated. Wider ADCs would simply pad the samples with noise rather than zeroes.

[theblue1]

Got it and I'm in complete agreement.

Thank heaven, there are two people somewhere in the world that can agree about something.

[Johnny B]

I blame the Digital Math Scoundrels at the chip making companies for all the problems.

They should all be sued in a big Class Action Lawsuit.

[theblue1]

[George Necola]

here we go:

higher bitrates in conversion than 24 bits make no sense. The master of conversion, Dan Lavry (you know, the lavry blue stuff) had described this probleme here:
http://www.lavryengineering.com/docu...ing_Theory.pdf

and here (do you need 20bits):
http://www.lavryengineering.com/white_papers/dnf.pdf

what he wrote (copyright by dan lavry):
In my paper I mentioned 3 arguments why 192KHz is worse:
1. The file size increases thus the space requirement compared to say 96KHz is doubled, and data transfer is slower by 2.

2. The computational requirement gets to grow and often by more than a factor of 2. That is why people that bought into 192KHz often ended buying very expansive accelerator cards, and still came short.

3. That is the big one: There is a tradoff between speed and accuracy. Clearly, the accuracy of a 10Hz system is great, but it is too slow for audio. The accuracy of 1GHz is much poorer, and it is too fast for audio. The question is - what is the optimum rate?

It is not true that faster is better. It is not true that more is always better. A 6 foot person weighing 100lb is too thin, but the same person weighing 500lb is too heavy. There is such a thing as OPTIMAL RATE. In the case of audio, it is all about what people can hear. That is what dictates why most mic's and speakers are optimized to about 20-20KHz, not 20-96KHz. The same factors should apply to converters.

The speed accuracy tradoff is one of the general engineering concepts, and it manifests itself many ways. Most of them are practical, such as "you can charge the cap more accurately if you have more time", or "the amplifier will settle to a more accurate value if you give it more time". But with modern converters, mostly based on sigma delta, the tradoff starts on paper, before we get to "real world" circuits. The basic given set of design parameters for a sigma delta converter are 1. oversampling ratio 2. filter order 3. number of quantizer bits.
Say you have a given set of parameters.
You can design for the best 0-24KHz audio bandwidth
You can have less precision but more bandwidth 0-48KHz
You can have even less precision but more bandwidth 0-96KHz

This was regarding the paper design stage of sigma delta. Than you get into the real world circuitry and face the same tradoffs again...

There is no escape from speed vs accuracy tradoffs, sigma delta or not...

Regards
Dan Lavry



if you wanna follow the whole discussion:
http://recforums.prosoundweb.com/ind...7634b4445fac25