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192 khz vs 48 khz poll


esldude

192 khz sampled digital audio will record and reproduce analog musical signals below 20 khz more acc  

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I think many, many audiophiles have this idea that with more dots (higher sample rate) you have a smoother and more accurate waveform.

 

Having read your back and forth with Mike Mcsweeney on another thread (Vinyl Conversion), I would not be surprised to find that the above statement does reflect a very common belief (misbelief). So if you are very simply asking whether you need more than 48,000 dots to accurately capture an audible sound, the fact that (a) so few people have responded and (b) the vote is nearly tied, suggests many of the Computer Audiophile readers are still confused on the learnings of Nyquist et al.

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I think the filtering makes a difference with hi-res above 96k. And as Jud often notes, many modern DACs upsample automatically as part of their processing, so if they are fed a high res source, then there is less processing (so also less noise/interference) going on within the DAC, and this could improve the sound of the output.

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the fact that it is debatable at all, and the fact that there is a near 50% split in opinion, would lean me to believe what i already do believe. That you will always increase the accuracy of the reproduction with the higher sampling rate, provided perfect circuitry to accomplish the sampling task does not cause distortion. I don't even believe it is possible to achieve perfect reproduction without an infinite sampling. If you have to use an algorithm to mathematically reproduce the signal from a "sampling", there will be inaccuracies in the reproduction. Unfortunately this is not a perfect world, and we don't have perfect circuitry.

 

What is the purpose of the debate? Are you of the belief that we cannot continue to improve the technology, from live music to reproduction. Do we stop technology from improving the great gift of hearing the music of the world?

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Yes, it is cynical. I think many, many audiophiles have this idea that with more dots (higher sample rate) you have a smoother and more accurate waveform. That isn't a benefit of higher sample rates. You get only more bandwidth. Within the stipulation I made of 20 khz and lower both at least theoretically are equally accurate below 20 khz. I think this idea hirez equals smoother or hirez equals more resolution is a problem. It is technically incorrect as a way to imagine what goes on and what it gets you. More bit depth actually gets more resolution.

 

As long as you are sure that everything of interest is occurring at less than half the sampling rate, then Nyquist says you can perfectly reconstruct it. Of course in the idealized world of Nyquist, perfect filters and infinite time in which to implement them are available. For information about the practicalities of implementing imperfect filters in limited time in the real world, I would look to someone with experience like Miska. But certainly in terms of whether one can exactly reconstruct a waveform as a matter of mathematics, once you have a sampling rate just more than double the highest frequency of interest, you're there.

 

As wgscott mentioned, and papers by Lavry indicate you could benefit perhaps with 96 khz.

 

Just to note a pet peeve, there are papers I have seen by Lavry that do not bother to provide factual support for some of the more significant and controversial statements they make, such as that 4x sample rates are actually inferior.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

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Strictly speaking, mathematically in terms of Nyquist/Shannon, no. In almost all practical instances you will ever encounter, the answer is however, a resounding yes. And that is driven by several things, including the mathematics of filtering, which inevitably alter either the time or frequency domain. (Usually both.)

 

And no, the same filters used with material at different sample rates will not produce identical results - not mathematically or in practice.

 

One cannot answer this poll accurately without choosing both yes and no, as both answers are correct. As you well know, or at least, you should. Very few DACs out there are non-oversampling, and there are good and sufficient reasons for that.

 

Can you give an example of a non oversampling DAC that reproduces 16/48 material as accurately as a DAc that oversamples to 192k or beyond? I would thing such an example would give you a more definitive answer, no?

 

Otherwise, in all practical terms, the way you write the poll question, I would answer the poll question with a YES.

Anyone who considers protocol unimportant has never dealt with a cat DAC.

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I voted FASLE

 

I probably polluted the results by voting, but had to, to see the results. My listening experience and a growing number of suggestions by others is causing me to believe 96khz is adequate for the vast majority of listeners. If there is an advantage in using a higher sample rate it may benefit a few that have very expensive components, probably not mine. So there is always merit in the discussion because exploration is what leads to breakthrough understandings. One area where my understanding is very weak is filtering; it's need and function in a DAC. Rather than ask for an explanation, can someone suggest an information source that would be comprehendable to a non-EE. Thanks, all!

That I ask questions? I am more concerned about being stupid than looking like I might be.

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As long as you are sure that everything of interest is occurring at less than half the sampling rate, then Nyquist

 

That's a big IF.

You will miss the beginning of new tone between samples....and that tone, could be a tone that has never been heard by mankind before.

 

Sampling is to sample the current audio at specified times. Call them t1 through t100

if you are recording live "music" of the jungle, and at T50.5 a new animal begins his "music", you will lose his "music" between T50.5 and T51, and maybe his "music" is at a different pitch at the beginning. No algorithm is going to reproduce what was lost.

 

Likewise, you will always lose what "music" is lost between sampling that which is not "predictable". That is why they call them "predictable algorithms".

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I voted FASLE

 

I probably polluted the results by voting, but had to, to see the results. My listening experience and a growing number of suggestions by others is causing me to believe 96khz is adequate for the vast majority of listeners.

 

The poll wasn't about "would you be satisifed with a lower sampling", the question was of accuracy.

Your false answer was an accurate response.

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That's a big IF.

You will miss the beginning of new tone between samples....and that tone, could be a tone that has never been heard by mankind before.

 

Sampling is to sample the current audio at specified times. Call them t1 through t100

if you are recording live "music" of the jungle, and at T50.5 a new animal begins his "music", you will lose his "music" between T50.5 and T51, and maybe his "music" is at a different pitch at the beginning. No algorithm is going to reproduce what was lost.

 

F nyquist

 

Actually doesn't work that way, which is Dennis's point. I know it's unintuitive, but it is mathematically provable. I wish I could draw and make this clearer, but I unfortunately don't have an artistic bone in my body. So here's my attempt to do it in words:

 

Think about what music, or really any harmonic tone, including animal calls or songs, looks like on an oscilloscope. For instance, here's a trace of a tone from a marimba:

 

130-1300-overlay.jpg

 

What Nyquist says is that once you know where three points on that wave are located, and the amplitude of the wave doesn't vary too fast, there's only one wave that'll fit those three points. (That's why the requirement to have a sample rate greater than 2x the highest frequency of interest.) So you can trace *all* points on the wave continuously from beginning to end, and thus even if the marimba was first struck between sample points, it doesn't matter.

 

What *wouldn't* be reconstructed is a sudden transient that began *and* ended in between two consecutive samples. But that is equivalent to the "frequency of interest" being half the sample rate or higher - in other words, if you have an event that occurs within 1/23,000th of a second and your sample rate is 44.1kHz, that event won't be captured for the same reason a 44.1kHz sample rate is inadequate to reconstruct a 23kHz waveform.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical Ethernet to Fitlet3 -> Fibbr Alpha Optical USB -> iFi NEO iDSD DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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What Nyquist says is that once you know where three points on that wave are located,

 

What *wouldn't* be reconstructed is a sudden transient that began *and* ended in between two consecutive samples.

 

In my example, you will never know the 3 points on the wave, because the pitch is unique just as the very beginning of the wave.....and what you speak of that wouldn't be reconstructed is what i am referring to.

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In my example, you will never know the 3 points on the wave, because the pitch is unique just as the very beginning of the wave.....and what you speak of that wouldn't be reconstructed is what i am referring to.

 

All nyquist says it what logic states. As long as you have enough sampling data, you can predict the resultant. There is nothing mysterious about that. He even states himself that his theorem doesn't hold true in certain instances.

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Don't forget that perfect reproduction - even mathematical - via Shannon/Nyquist requires an infinite number of samples. It's why we never quite make perfect reproduction in the real world, though it comes close. Also, the sampling rate must be >2B where B is the highest frequency of interest. It breaks down at exactly two times. :)

 

 

Actually doesn't work that way, which is Dennis's point. I know it's unintuitive, but it is mathematically provable. I wish I could draw and make this clearer, but I unfortunately don't have an artistic bone in my body. So here's my attempt to do it in words:

 

Think about what music, or really any harmonic tone, including animal calls or songs, looks like on an oscilloscope. For instance, here's a trace of a tone from a marimba:

 

[ATTACH=CONFIG]10956[/ATTACH]

 

What Nyquist says is that once you know where three points on that wave are located, and the amplitude of the wave doesn't vary too fast, there's only one wave that'll fit those three points. (That's why the requirement to have a sample rate greater than 2x the highest frequency of interest.) So you can trace *all* points on the wave continuously from beginning to end, and thus even if the marimba was first struck between sample points, it doesn't matter.

 

What *wouldn't* be reconstructed is a sudden transient that began *and* ended in between two consecutive samples. But that is equivalent to the "frequency of interest" being half the sample rate or higher - in other words, if you have an event that occurs within 1/23,000th of a second and your sample rate is 44.1kHz, that event won't be captured for the same reason a 44.1kHz sample rate is inadequate to reconstruct a 23kHz waveform.

Anyone who considers protocol unimportant has never dealt with a cat DAC.

Robert A. Heinlein

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