It is in the sense that you are manipulating the signal in the first place at a level that is well above the resolution of human hearing. The mistake is always the same: too many assumptions on what a human can resolve. You just cannot apply known theorems to decide this. It doesn't work in non-invariant conditions. Again...!
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The sampling rate doesn't need any citation. You should demonstrate that is not relevant for the human perception. Especially when the signal is music and not just simple test signals!
As demonstrated in that experiment it isn't just a simple number but something more complex related to temporal structure as well. You don't have any control on this in the digital domain as time is just a parameter, not a variable!
Even worse, they're dangerous around machinery!
Shhhhhhhhhhhhhhh.
Fourier's transform demonstrates you can make any music by combinations of simple test signals.
Key to me for producing the highest quality sound is to lose that attitude - one works with the belief that "there is no such thing as a bad recording" and that then becomes a very powerful tool for tracking down system problems.
Why this is so is because for a "poor" recording every exaggeration or highlighting of recording problems is extremely apparent - if the playback system distorts then it distorts even further the recording abberations, making the latter quite intolerable at times. Using a very clean, 'pure' recording is one of the hardest ways of assessing where a system is misbehaving, because the mind finds it very easy to reconstruct how the sound should be - you don't test drive a new car for handling behaviour on the smoothest stretch of highway in your area.
Personally, I find that reducing the playback misbehaviour to the lowest levels means that I run out of recordings to label as "bad" - the problems they have become subjectively irrelevant, and they can be enjoyed for the musical message they were meant to convey ...
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Oh but there is! I know because I've made many of them myself.
BTW, I'm interested to know what measure of time resolution gives an answer of picoseconds for a CD player. If you put in a delta function, you could maybe look at the corresponding sinc function and interpolate the exact time of the event to that kind of accuracy.
But it certainly can't resolve two separate events unless they are spaced more than one sample apart.
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Shannon-Nyquist. The only thing that changes relative timing (or phase, if you prefer) between different frequency components below 1/2 the sampling frequency is jitter.
I use decent headphones (Audeze LCD-2s) to do my final judgement. Trust me, there are plenty of crappy recordings out there.
I am not sure at all! I think we are saying different things. In digital domain you have no control of time (as variable in the signal which carries information for sure when talking about music). What I mean is what you find here at the beginning, for example:
5.10 Jitter | 5) Sampling issues | Audio Quality in Networked Systems | Self Training | Training & Support | Yamaha
There are a lot of assumptions, imperfections and hypothesis in the digital process, as in any other field, including the practical things related to production of equipment and choices made by people in the music recording who are not necessarily experts of digital electronics! It is anything but perfect as theory and specs would imply and we have not considered yet that frequency and time are not interchangeable when it comes to humans and music.
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Quite the opposite, you have exact control of time to the picosecond level. If you're now claiming that Shannon-Nyquist is incorrect, you'll need to do some rigorous analysis to show why generations of engineers are wrong, despite successfully using this very basic piece of mathematics to design CAT scans, MRI, radio telescopes, fighter jets...
Likewise, you seem to be denying the Fourier theorem. You'll need even more rigorous analysis and evidence to convince anyone of that, given a couple of centuries of its successful application.
Interestingly, the page you linked to states all of this as well.
Hello all,
What a discussion here, if I might add I almost gave up on vinyl as having any superiority in the low level details up until I designed and built a decent performing mechanical linear tonearm "see Diy Linear Arm" thread. I'd say the front end and phono stage are some of the big limitations. The phono stage end in my case needs a large slew rate since its passive/active, and the first stage needs a large gain for the passive 2122hz filter following, with the riaa pre emphasis cut to vinyl this requires the high slew to be able to handle this without overload before the next active stage with gain.
Enough rambling, between all of this and a very affordable at120e dynamic range is almost identical on a well cut lp and its cd, beig actually better on the vinyl in many cases vs the dynamically limited but loud cd. Low level detail can be just as good, if not often audibly better than th cd, but in some cases in the hf you can hear the roll off placed to protect the cutterhead. Ultimately I've learned no one side of the debate is right, there are crappy recordings in both formats, but some are more limited by their gear than they like to think at the low level signal end to make this a fair comparison.
Colin
What a discussion here, if I might add I almost gave up on vinyl as having any superiority in the low level details up until I designed and built a decent performing mechanical linear tonearm "see Diy Linear Arm" thread. I'd say the front end and phono stage are some of the big limitations. The phono stage end in my case needs a large slew rate since its passive/active, and the first stage needs a large gain for the passive 2122hz filter following, with the riaa pre emphasis cut to vinyl this requires the high slew to be able to handle this without overload before the next active stage with gain.
Enough rambling, between all of this and a very affordable at120e dynamic range is almost identical on a well cut lp and its cd, beig actually better on the vinyl in many cases vs the dynamically limited but loud cd. Low level detail can be just as good, if not often audibly better than th cd, but in some cases in the hf you can hear the roll off placed to protect the cutterhead. Ultimately I've learned no one side of the debate is right, there are crappy recordings in both formats, but some are more limited by their gear than they like to think at the low level signal end to make this a fair comparison.
Colin
Ok. But you need to have corrections to limit jitter which has also different forms. As you have no control on time how can you be sure that you are only removing "the bad and not the good" when it comes to musical signal? I have never found anything satisfactory about this and usually the reasoning stops a simple level relying on the usual hypothesis.
Depends what criteria you use for 'crappy' ... in one sense the worst of the worst are the latest pop recordings: massive compression and studio manipulation, these are truly dynamite to get to present a reasonable face to the world. However, if one uses every little trick in the book one can think of, these can be brought back from the brink - whether it's worth doing so, from the point of view of the 'value' of the musical content, is another matter ...
You have control on the sampling rate that is made by clock! I am not claiming the Shannon-Nyquist is wrong don't confuse things. Rather, all those things you citing are not living objects!
In the same way I am not denying the Fourier theorem. It is not valid once you put the listener in the chain. It is valid until the signal leaves the speakers if you have a listener or even after if you have a microphone instead of a human being. It is written in the theorem why.
The audio "science" is so successful that you cannot tell how one thing will sound and you have to listen to it to decide if it is good or not. No such thing as CAT scans, MRI, radio telescopes, fighter jets...I am afraid.
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