Not sure the sampling theorem is to blame, just not enough transition band for a practical filter to prevent aliasing (ADC) or images (DAC).
The 44.1 sampling frequency is based on a mis-application of the Nyquist theorem; yes, it is certainly not enough.
Well, if storage space and network bandwidth was no issue, I would take the 24 (as I do - 24 bit FLAC on home server, mp3 copies for mobile/portable use).
Ah, yes. More than happy to give JJ that privilege!
First point is true, I do not understand the second. Are you still talking about increased bit depth or sampling rate? Higher sampling rates often increase the noise (wider bandwidth, more settling artifacts). And what do you define as ideal and practical SNR? Aside: For high-resolution converters, SNR is often set by the analog noise floor rather than quantization noise. Similarly I am not aware of any converter that actually approaches the ideal SFDR in practice; IME distortion and other noise sources generally put the noise floor within 10 - 20 dB'ish of the SNR. That said, I have not looked closely at many audio DAC (chip) specs.
Hahahahaha! I'm sure you would
Man, I'm getting beat up tonight!
Tim,
Fine, lets complicate things. Mathematically higher bit depth is better because you have less quantization error. Increasing sampling makes the difference between ideal and practical SNR smaller. Mo' is betta. Period.
16bit advocates do not dispute this. What they do say is that anything more than 16bit depth and 44.1k samples is beyond the practical limit because anything else would be inaudible at least for most. Let's take a step back here and ask, is it really? Bit depth allows for more quantization levels between the highest input amplitude (Xmax) and the lowest (Xmin). Xmax in a 24bit device need not be the same as maximum corresponding voltage output of a 16bit device that being levels that would either hurt you or blow up your gear whichever comes first. Most likely max voltage out is the same for both limiting the dB but answering my own question I asked before, apparently amplitude q levels (steps) can be different. It's not as sexy as saying with 24bit you get higher dynamic range (which you can't use anyway) but it does say you get a better approximation of the continuous analog signal discreet in amplitude and time. Call it higher relative resolution or fidelity to the source, in this instance it is the same thing. So forget the extremes in amplitude for now and look at the quantization figures with 16 vs 24 bit in an ADC or DAC with a max of 2v.
Getting more to the point, why would anybody who can get a 24bit recording want to get a down converted version which will have either higher harmonic distortion or higher noise depending on how the down conversion was done because you have now created quantization error? I will guarantee that the answer will be based on practicality rather than what is really better/higher fidelity/higher performance EMPIRICALLY.
Can you please explain how the Nyquist theorem was mis-applied?
In practice, infinite sequences, perfect sampling, and perfect interpolation are all replaced by approximations, deviating from the ideal mathematical reconstruction.
Thread just seemed too much fun to resist though
It is the mark of a good engineer to understand when such things are negligible and when they are not. The Sony and Phillips cartel successfully pushed the early digital model on the claim that the errors were negligible. Obviously they were not; the digital analog debate is now about 35 years old...
Sure. It is based on samples of analog/infinite resolution, (16 bits would be an example of a limitation).
This quote is taken from the Wiki page, linked below:
http://en.wikipedia.org/wiki/Nyquist–Shannon_sampling_theorem
I've included the wiki page on this, as I have run into a lot of pushback over the years from people not completely familiar with the theorem. If one is to say "the difference between the ideal and the digital samples are so slight as to be negligible", what will occur is a debate that goes on for decades. In their recent book "Control Design And Simulation", Jack Golten and Andy Verwer discuss this phenomena in chapter two, with regard to applying mathematical models to the real world: "...mathematical models invariably involve simplification. Assumptions concerning operation are made, small effects are neglected and idealized relationships are assumed."
It is the mark of a good engineer to understand when such things are negligible and when they are not. The Sony and Phillips cartel successfully pushed the early digital model on the claim that the errors were negligible. Obviously they were not; the digital analog debate is now about 35 years old...
I'm also of the opinion that the sampling frequency needs to be higher anyway so the brickwall filter can be avoided. IMO that is one of the more significant advances that has occurred on the record side, as you don't have such crazy effects from the filter.
Because 24/44.1 is practically non-existent, and the content above 20khz in 24/96, 24/192 has been demonstrated to cause IMD in the audible spectrum. Audio reproduction is all a series of compromises, and here's the one we face here: Which is more audible, more damaging? The harmonic distortion and noise created by down conversion, or the IMD genrated by electronics and transducers reproducing ultrasonic content?
I don't know the answer, but the proponents of hi-res don't seem to even want to ask the question, they want to assume that bigger numbers = better reproduction. And the odd thing is many of those proponents are the same folks who, when talking about analog, seem to think THD and noise are pretty harmless compared to IMD.
Tim
Yes- that's pretty funny. In the face of the facts I mean.In this video it is experimentally proven in real time that the Nyquist theorem is applied correctly in 16/44 digital:
http://www.youtube.com/watch?v=cIQ9IXSUzuM
Apparently, yes- something is wrong with this picture
I'm also of the opinion that the sampling frequency needs to be higher anyway so the brickwall filter can be avoided.
In a way this is sort of like proving the existence of Bigfoot...
What I have is my experience as an LP mastering engineer, so I am working with master tapes and files. But- if I go in that direction, the thread derails, this is *not* an analog/digital debate, is it??
Yes- that's pretty funny. In the face of the facts I mean.
I plea midnight posting Don. Yes I'm still talking about bit depth as substituting bitdepth n will show the change in SNR NOT substituting sampling rate.
Man, I'm getting beat up tonight!
Getting back to the spirit of my original post, it will take about 13 bucks for you to get an idea for yourself. Perhaps not what ultimately is better but at least what is better for you.
In other threads you will find that a few of us here (very few I think) believe depth has more importance than sampling rate given the same source. Perhaps not coincidentally we happen to be people who have recorded in 24 bit and have had to down convert. That is where I am coming from. These days you don't have to have been at the console or workstation. The 24bit/*kHz files are out there. For a little bit more you can get applications that will even allow you to choose your dither. Are the differences night and day? Honestly no. It's not like 16bit vs 4 bit but it is there. Twilight and night? The difference may or may not be great but they can be meaningful from an artistic point of view. In these debates somehow focus is on how loud things are. The differences I hear are in the low amplitude sound events. Brushes on the skin of a snare, decay trails, ambience. This is what I experience gets buried. Does it change the musical message? Ummm. No, I wouldn't say it does. Does it make the experience just that bit less enjoyable, many a time, yeah.
For some reason, I gravitate to 88 or 96. I prefer these to 176 and 192 as I find these generally lacking in energy. Don't ask me why, I don't know. At the same time what 24/44.1s I do have, I find only marginally better than 16/44.1. Continuous amplitude and time to discreet amplitude and discreet time back to continuous amplitude and continuous time. The 2 go hand in hand. I don't think one can think purely on digital to analog conversion without appreciating the analog to digital conversion too.
If maybe there is one assumption I am uncomfortable with its the one that says "audiophiles" are just being lemmings. Enthusiasm doesn't necessarily suggest the ol happy guy has been brainwashed by some evil guys orchestrating some conspiracy to separate them from their money. Like I said, there are many reasons the professional world records at those resolutions. That the products are decimated are for practical purposes. TV, Radio, mass distribution media are generally 16bit 44 or 48. Like Ulf (I hate you ) I convert to 320kbps for portable use, again practical considerations. I'll continue to do it too until a 2 TB ipod comes along. LOL.
Now I get the chance, albeit by proxy server , to get my hands on (hopefully) the original files or something close to it. It's an opportunity I'm not passing up on exploring. Same goes for DSD. In the end it's about the titles. I'll still get what I like in whatever format I can get it. Even if that means downloading some obscure one hit wonder from a UK MTV grab off of youtube.
Yeah, I'm a sick puppy.
BTW I'm still searching for Zerra One and Blue in Heaven. Maybe someone can help me!