[wado1942]

You said: "The noise floor has little to do with usable dynamic range". Sorry, but that does not make sense. I think you are using the term "usable dynamic range" as a mixture of noise and distortion (you pointed out that an 8 bit did not sound good and it was not just the about dynamic range). Fell free to clarify, if you wish.

You're correct in your assumption. I think distortion and noise should be measured together, particularly since a lot of people don't know the difference, especialy where digital technology is involved. For instance, if I recorded a guitar to 12-bit digital, I'd have to keep the level pretty high in order to sound acceptible. That's due to the distortion being so high at lower levels. However, on 1/4" tape, I could keep it comparatively low and while there may be more noise, the overall sound is much less irritating. This means the usable dynamic range of tape is wider than 12-bit digital even though the 12-bit medium may have lower noise.

I assume you know that the lowest bandwidth device in the audio chain determines the bandwidth.

Of course, but there inlies a problem because the definitions are a bit blurry where this comes to play. For instance, I have some microphones that are +-3dB 10Hz to 25KHz or so. Now if you record those though my preamp with 50KHz +-1dB, that bandwidth is preserved. Comming back off of tape, the playback maybe -3dB at 20KHz but I can see on an FFT that there's valid information going clear up to the limits of my testing ability on 96KHz converters. What I'm saying is, even though our ears may be -45dB or something (just an uneducated guess) at 25KHz, we're still aware of and expecting information up there at least on a subconscious level. When you put a 50th order filter on the signal at 20KHz, that expectation is not met. As subtle as it may be, the natural world does not behave this way. So it's no surprise to me in that test I mentioned that even though the subjects did not HEAR the difference, they FELT it. Be aware, also, that your brain ramps up the gain and fills in blanks so to speak with what we percieve. Like with our eyes. The image is very blurry, distorted and missing all sorts of information. But we work in conjunction with memory, combining experiences from 10ms ago with 10 years ago, we can give a glance at something and see a sharp, detailed image. Case in point, people with cochlear implants who had hearing recently say that they can identify and follow songs they knew before they went deaf. However, a new song on the radio just sounds like noise to them. Granted, that's probably in part due to how music is made now, but all jokes aside, it's more evidence that no EE or MD or whoever else knows even a fraction of what there is to know about perception. BTW, did you know that you can actually hear sound directly with your brain?

Numbers have always been used to lie about audio gear. Dan refuses to play that game. I can't admire him enough for that.

No kidding. I have a live console that claims to be a +4 device but it hard clips at +8. I actually keep the average output around .38V because it starts to get a little fuzzy if I go too much above that. Then there's that little Behringer that I use for routing signals to my talkback monitors which claims to be 10-channel when in fact it's 4 because they're counting the tape & AUX returns, and the stereo channels are counted double.

Correct and nyquist frequency of 96kHz which is way beyond what earthly creatures can ever hear...

Well, humans anyway.

Why should 192 sample rate be avoided? is there a universal quantifiable disadvantage?

No, I said "at this stage" because there just aren't any 192KHz converters that outperform 96KHz converters. But that's part of the argument I've been making. Instead of upping the clock to get higher sample rates and continue decimating, we should just store the raw sampled data. Then we'd have all the sample rate we'd ever want. When advancements in filtering & such are made, it can be applied in software offline. If you ever wanted to use DSP on your recorded audio, you can just add in some empty less significant bits, maybe an extra 12 or so that can be dithered back to 4 or whatever and process to your heart's content without upsampling. We'd need to rewrite the books on how DSP is handled because we'd be dealing with Delta slopes instead of hard, set voltages but, that's why I say "this stage".
On playback, you'd only need analogue LPFs for anti imaging purposes.