Focus on 24/192 Misguided?.....

yes, I seem to concur that 24/192 makes no sense. I never was a believer of the hi-rez scam that pervades the industry.
There is also a nice article written by Dan Lavry on Lavry Engineering's website on why sampling upto a max of 24/96 makes sense & anything beyond that is bogus. Dan Lavry wrote this article back in 2004! This article is called "Sampling Theory" when you go to this link:
http://www.lavryengineering.com/index_html.html

I'm sorry Onhwy61 the referenced article is not nonsense. We are dealing with a pure digital signal until the output of the D/A. So, there are plenty of DSP techniques available to make this work without oversampling the heck out of the digital signal. We need to oversample just enough to ease the specifications of the analog reconstruction skirt so that it's not brickwall. That's where 96KHz sampling comes in.
I also bet that most people's systems (including yours & mine) do not have 96dB of dynamic range after all the sweat that we have put in to isolate & reduce noise.

If you really take the article seriously we should all be using 32kHz/12bit digital because the math works perfectly at that level too

no it does not. The Fs/2 freq would be 16KHz which would be less than 20KHz.
And, 12-b would be insufficient because one would add too much noise when going thru the mastering process & you'd effectively get 9-10 bits of music.

04-20-12: Onhwy61
And what's wrong with mathematically perfect response out to 16kHz? Most people, and certainly most middle age and older audiophiles' hearing doesn't go beyond 16kHz.

while it might be true that older ears do not have the 20-20K respone, the music is prepared for everyone. Like the article says there is a 100 yrs worth data that shows that 20-20K is the human hearing limit. So, when preparing digital music might as well keep the audio spectrum to its max limits. Younger people certainly can hear this range & so can many other older folks.

FM doesn't go beyond 15kHz and at its best it's pretty damn good sounding.

FM has (air) spectrum bandwidth limitations that force it to curtail bandwidth. If they could help it, they would have also transmitted in the 20-20K range. Air spectrum is very expensive so this compromise seems reasonable.

FM doesn't go beyond 15kHz and at its best it's pretty damn good sounding.

Like I wrote in my prev post & I'll write it again: if you start off w/ 12-b you'll end up with 9-10 bits after the mixing & mastering processes. If you start off w/ 16-b, you'll probably end up w/ 12-13 bits. The section "The dynamic range of 16 bits" explains quite well the DR of 16 bits & also how it might be possible to encode fainter signals using 16-b.
Since a lot of data already shows that sounds at absolute levels of +120dB, +130dB permanently damage ears, my understanding is that it might not be worth encoding sounds on a disk that cover the enitre 140dB dynamic range of human hearing. It appears that covering 120dB of dynamic range is sufficient. If one uses 12-b only & one attempts to encode very faint sounds my understanding is that 72dB could be a limiting factor trying to cover the entire 120 DR. 16-b & 96dB is adequate & the article shows a plot of a -105dB signal at 1KHz using clever dithering techniques.

Nyquist criteria applies only to continuous waves.

nonsense! The Nyquist criteria applies to any signal that needs to be quantized. The Nyquist criteria only gives the minimum requirement; it does not say that one is forced to have only 2 samples per highest frequency.

There is no filter in the world, digital or analog, that can do that (no matter how many poles) with even group delays (or linear phase if you prefer).

yeah, I know what you mean for analog filters & I agree w/ you in that respect but for digital FIR filters (linear phase) I'm not sure I totally agree with you. My understanding is that if you had a, say, 64-tap FIR you could have a very steep skirt digital filter that would have flat group delay & group delay distortion. I would have find some evidence of this before I contend this issue w/ you but for right now I'm skeptical that it cannot be done. I'll leave it that....

I see the case for upto 24/96 as it seems to alleviate most of the pressing issues such as noise creeping into the music signal during mixing/mastering, analog filters having too steep a skirt at 44.1KHz. I'm not sure that I buy the case for 24/192, etc.

If anyone is interested in looking at some signals look at the Powerpoint presentation at reference #17 in the article. SLides 20, 21, 24-28 show spectrum of instruments & spectra of music from commercial CDs. Look at the freq where the content dies off even for SACDs.

Short high frequency bursts like cymbals will suffer the most of distortion.

Kijanki, this one is for you: here is a wonderful thread showing frequency spectrum of various brand of cymbals: http://www.drummerworld.com/forums/showthread.php?t=66957

The top quarter of the thread shows some really very good spectra of various cymbals. You can see that by 40KHz the spectrum has died down to 30-40dB SPL. The major part of a cymbal crash freq content is in the 20-20K range & the content falling off rapidly thereafter. I agree there is content beyong 20KHz but atleast 30dB by the time you hit 30KHz.
So, one could make the case for a 96KHz sampling rate wherein all the freq content upto 48KHz would be included. This sounds reasonable. At 48KHz the analog filter spec becomes reasonable too. Looks like it's a win-win situation....

04-20-12: Kijanki
............
Second nonsense is that digital filter can suppress 96dB within 4kHz without any problem. There is no filter in the world, digital or analog, that can do that (no matter how many poles) with even group delays (or linear phase if you prefer).
.........

not true! Here is a link to paper written by Dan Lavry of Lavry Engineering who wrote this paper in 1997 that shows a 500-tap FIR filter that has a passband of 15KHz & a transition band of 1KHz & stopband starting at 16KHz. The attenution achieved in the 1KHz transition is a whopping 100dB!! See page 3 of 7:
http://www.lavryengineering.com/white_papers/fir.pdf
yeah, it came at a price: 88.2 million operations per second using a dedicated DSP. Very high # of MOPS but do-able.
If one opens the transition band to 4KHz like the paper referenced by the OP then I'm sure that the # of taps will come down.
The paper also goes not to say that the group delay of the FIR filter is flat all the way out to 15KHz.

here is another digital filter paper (from the AES) that shows brickwall digital FIR filters:
http://www.nanophon.com/audio/antialia.pdf.
it's possible to have these brickwall FIR filters with reasonable DSP capacity.

04-20-12: Kijanki
Al, On one hand he has whole chapter titled "192kHz considered harmful" describing harm that 192kHz can do to amplifiers and speakers to later say this about oversampling:......

Kijanki, all that the author is saying is that when sampling at higher freq like 96KHz or 192KHz, you get intermodulation products that fold down into the 20-20K audio band due to typical preamp, power amp bandwidth limitations of not being able to reproduce higher freq products distortion-free i.e. due to the non-linearities of the electronics. And, systems having smaller bandwidths have the situation worse in that the probability that they'll amplify the high freq signals is much higher. So, the point is that if you do not sample at 96K or 192K you won't have these higher freq intermod products, they won't fold down into 20-20K & your preamp/power amp will not amplify them due to its non-linearity.

it's clear to see that if a 96K or 192K sampled signal is downsampled to 48KHz then the anti-aliasing filter will cut off all these high freq intermod products. So, according to the author, since this signal is free of any ultrasonic content, it's safer to playback with the idea that distortion products due to ultrasonics are not being played back.

I do not think that it's unreasonable to say that ultrasonics created due to higher freq sampling can create in-band intermod products that can be amplified by the non-linearities of the playback electronics & that they are harmful to the playback listening pleasure.

I don't think that the author should have labeled the paragraph as "192KHz considered harmful". People like Kijanki have read this literally thinking that the very act of sampling at 192KHz is harmful. No, I don't think that the very act is harmful; it's those ultrasonics folded down & amplified that are harmful.....

Suppose we want to compare the fidelity of 48kHz sampling to a 192kHz source sample. A typical way is to downsample from 192kHz to 48kHz, upsample it back to 192kHz, and then compare it to the original 192kHz sample in an ABX test [21].

Al, Kijanki: I *think* that I might know what the author is intending to say here: To do an A/B comparison, the author would like to level the playing field. Thus, he does not want to use the original 192KHz signal as-is. What he wants to do is downsample on-the-fly the 192KHz signal to 48KHz & create signal A. Then, upsample this 48KHz signal on-the-fly back upto 192KHz & create signal B. Now, the playing field is level because the same machine downsampled & upsampled the signal & the same filters have screwed up the A & B signals. The signal X is the original 192KHz signal. If you were to use the original 192KHz which was created on some different machine against the 48KHz created on your CDP, you would have the effect of 2 different digital filters & you could not do a true A/B comparison. Does this make sense guys?

if you're not sold on hi-res.....fine. just don't tell me what i hear isn't for real and i'm "wrong".
Levy03

if you are sold on hi-rez, fine. I don't know what you are listening to - true hi-rez (which would mean the analog masters sampled at 96KHz or 192KHz & made available for purchase) OR bogus hi-rez (whcih would mean taking the 16-b CD data, resampling it at 96K or 192K & providing it for sale to the unsuspecting public).
There have been sooooooo many scams re. hi-rez (recently read something about HDTracks were the offenders. Here is the link:
http://www.computeraudiophile.com/content/Metallica-Black-Album-HDTracks-Download)
that it's really very difficult to tell what the manuf has provided for sale.
You are probably listening to some digital filter thinking it's hi-rez & you are in 7th Heaven.
Maybe we should let you be - ignorance would be a bliss for you.....

As for filters - look at typical response of 2and 8 pole 20kHz Bessel filter in dB:

2pole 8pole
20kHz -3 -3
22kHz -3.63 -3.67
40kHz -9.82 -13.68
80kHz -20.32 -51.81

Kijanki, you provided us w/ the freq resp of ANALOG Bessel filters. I agree with you & I did write this in my prev post - analog filters cannot creat a sharp cutoff like what the author has shown - large attenuation between 20K-24K.
But, how about digital FIR filters? Can they create such a sharp roll-off?
Yeah, sure they can! Did you bother to read any of the links I referenced in my post? The paper from Dan Lavry shows 1 example & then there is that AES paper by Julian Dunn that also shows 4 filters that have 100dB atten & only a modest # of taps. All FIRs have flat group delay in-band.

I found this nice article called "A Beginner’s Guide to High Resolution Downloads of Music". here is the link:
http://audaud.com/2012/03/a-beginners-guide-to-high-resolution-downloads-of-music/

In this article is a para called "How hi-res should you go?" towards the bottom (scroll almost to the end). That para cut & paste:
'Unless you have extremely youthful hearing ability plus the highest-end speakers and audio gear, many of us feel that the improvement of 192K over 96K is inaudible. The word length expansion from 16 bits to 24 bits makes a much greater enhancement in the sound. 24/96 or 24/88.2 is fine for nearly everything. Also, remember that 192K and 176.4K files take up much more memory on hard drives, for little audible improvement.'

Looks like many people think alike: there's a case for 88.2K or 96K sampling but not beyond......

put it into a device that could only add noise and jitter and eventually will fail (drive, optics), ....

hey Audiofreak32, don't talk about hardware failures! With your being sold on computer playback you don't have a leg to stand on when it comes to hardware failures! How often does computer hardware fail compared to CD drive & its optics? Even the cheapo $40 DVD players from Walmart outlast almost all HDDs & other computer hardware.....

Yeah, the convenience of HDD playback is immense, I have to agree.

If you have not heard a properly setup DAC with hi-res files, you owe it to yourself to do so...

I have - dedicated computer for music playback, going into a dCS upsampler, going into a dCS DAC. Both dCS upsampler & DAC were clocked by a dCS Verona Master CLock. All interconnects were some very expensive WireWorld stuff. The total $ outlay on this whole setup made my knees weak - I could never afford anything like this for a long time! The sonics were easily beaten by my 1-box CDP.....There was no body or soul to playback MUSIC but the SOUND was stellar.

Kijanki thanx for that info on power amp freq bandwidth. I suppose that many more amps are higher freq bandwidth than I imagined but I don't know how many of these have the same number for power bandwidth? Any idea?

04-23-12: Audiofreak32
To summarize....

I guess we have concluded just the opposite - that 24/192 is a VERY GOOD idea actually,....

I don't know how you concluded this? I concluded that there is no need for 24/192 - 24/96 suffices as it solves the issue of steep skirt analog filter for 16/44.1 & that once this issue is resolved & we have more DR w/ 24 bits there is no real need for 24/192.

Onhwy61, yes most of the electronics today is still well below 100KHz bandwidth. Hi-end does not mean hi bandwidth; it means better sonics in the 20-20K band. It might be easier for preemie to extend to 10s of KHz above 20K but for power amps to have a power bandwidth of 100KHz will cost you very close to a 2nd mortgage. Don't believe me, do some research yourself & find out just how many power amps have a power bandwidth that even touches 50KHz. Find out what your gears' bandwidths are. Almost all audio gear was never meant to amplify ultrasonics. The Pro studio gear might be a different ball of wax. Thanx.

Kijanki, u could have answered your own question about ultrasonics from 192K sampling now that you have answered the question of power bandwidths of power amps. Now you can see why a power amp would be nonlinear in the ultrasonic range & why those ultrasonic intermod distortion products folded down to the 20-20K. Now , the amplitude of these ultrasonics might not be large enough but better not to have them at all.....