Understanding Sample Rate

[beerandmusic]

4 minutes ago, adamdea said:

Yes. That much became apparent over 10 pages ago. Having thought about it and already worked out the answer to your earlier question (why does OP bother asking a question if he doesn't care about the answer?) I answered in my own mind as "to get attention". So I think it's time to follow the logic of my own conclusion and engage the ignore button. 

 

You must have missed my post where i conceded the topic probably isn't even relevant to what my interest was because I learned that DSD doesn't use sampling rate related to nyquist....or that is what i believe i read....either way, i am not interested in said topic and only responding to those that insist continuing.

[marce]

Some excellent information, thanks to those who have perceived and provided the links and info.

 

[adamdea]

5 minutes ago, mansr said:

You can also ask the question, what is the minimum sample rate require to capture all the sound there is? The answer is roughly 500 kHz. Recordings done at 352.8 kHz are now readily available. If you examine one, you will see that they have precisely zero signal content above 100 kHz or so and very, very little above 50 kHz. This means that a sample rate of ~200 kHz is enough to capture sound in actual music, whether we are cats or humans.

True and I agree about the physics of sound and the spectrum of real musical sounds. 

Either way, the knowledge required comes from outside the sampling theorem itself. Of course for a whale or a creature on another planet the specs might be different.  

Similar points arise in relation to bit depth etc.

 

Ultimately you have to have a model of what matters. One of the problems of this hobby is that even those people who seem to take an orthodox approach to the sampling theorem like Rob Watts can take a pretty weird view of  what is audible -hence the 1 million tap filter and the claim somewhere to be able to hear things at -200dB . In fact that refusal to accept any limits to the spec drives a lot of arguments. I think the technical engineering side of it can become a bit of a red herring. You can show differences between the noise floor of dacs maybe 30dB below the 16 bit noise floor. You can show lowr phase noise in the clock. You can show objectively shinier paint on the router.

[marce]

3 hours ago, beerandmusic said:

 

nothing can change my belief because i am not willing to devote more time to it, and it is my honest belief that others have it wrong.

no one can control another's belief, let alone the person.

and attacks, mockery, ridicule doesn't change that.

 

If someone has the "brilliance" to be able to spoon feed me in laymans terms why i have difficulty accepting that 44.1K sampling rate is enough to capture infinite frequencies, infinite time slices, and infinite complex waveforms, that is audible to man, i would change my mind, but i highly doubt that is possible....i believe i am realistic.

 

PS i doubt you even understand the basic concepts either.

first page there was a link to a video...

[Spacehound]

17 minutes ago, mansr said:

If misunderstanding were an Olympic sport, you'd be winning the gold medal every time, summer and winter games.

Is there a gold  for  not getting ready on time?

[mansr]

2 minutes ago, Spacehound said:

Is there a gold  for  not getting ready on time?

They tried procrastination as an event once. The competitors never showed up.

[Spacehound]

11 minutes ago, adamdea said:

hTrue and I agree about the physics of sound and the spectrum of real musical sounds. 

Either way, the knowledge required comes from outside the sampling theorem itself. Of course for a whale or a creature on another planet the specs might be different.  

Similar points arise in relation to bit depth etc.

 

Ultimately you have to have a model of what matters. One of the problems of this hobby is that even those people who seem to take an orthodox approach to the sampling theorem like Rob Watts can take a pretty weird view of  what is audible -hence the 1 million tap filter and the claim somewhere to be able to hear things at -200dB . In fact that refusal to accept any limits to the spec drives a lot of arguments. I think the technical engineering side of it can become a bit of a red herring. You can show differences between the noise floor of dacs maybe 30dB below the 16 bit noise floor. You can show lowr phase noise in the clock. You can show objectively shinier paint on the router.

 

In audio 'spheres' there are more things that don't matter than do. 

[beerandmusic]

33 minutes ago, the_bat said:

I don't think this means what you think it means. 

 

http://www.wescottdesign.com/articles/Sampling/sampling.pdf

 

How about this white paper, specifically this excerpt:

The difficulty with the Nyquist-Shannon sampling theorem is that it is based on the notion that the signal to be sampled must be perfectly band limited. This property of the theorem is unfortunate because no real world signal is truly and perfectly band limited. In fact, if a signal were to be perfectly band limited—if it were to have absolutely no energy outside of some finite frequency band—then it must extend infinitely in time. What this means is that no system that samples data from the real world can do so perfectly—unless you’re willing to wait an infinite amount of time for your results. If no system can sample data perfectly, however, why do we bother with sampled time systems? The answer, of course, is that while youcan never be perfect, with a bit of work you

can design sampled time systems that are good enough

 

 

[mansr]

9 minutes ago, adamdea said:

True and I agree about the physics of sound and the spectrum of real musical sounds. 

Either way, the knowledge required comes from outside the sampling theorem itself.

The difference is that this outside knowledge is easier to acquire and validate than that concerning our ability to perceive sounds.

[adamdea]

1 minute ago, Spacehound said:

 

In audio 'spheres' there are more things that don't matter than do. 

What saddens me is that all the noise about things that don't matter tends to drive out consideration of what does matter. The real science seems to be driven more by gaming than music. If ambisonics had caught on way back we could have a catalogue of proper surround recordings available to be played on whatever configuration you chose and basically future proof. Even if it had been at 13/32 it would be more use than any dsd stereo recording. 

[beerandmusic]

28 minutes ago, beerandmusic said:

 

http://www.wescottdesign.com/articles/Sampling/sampling.pdf

 

How about this white paper, specifically this excerpt:

The difficulty with the Nyquist-Shannon sampling theorem is that it is based on the notion that the signal to be sampled must be perfectly band limited. This property of the theorem is unfortunate because no real world signal is truly and perfectly band limited. In fact, if a signal were to be perfectly band limited—if it were to have absolutely no energy outside of some finite frequency band—then it must extend infinitely in time. What this means is that no system that samples data from the real world can do so perfectly—unless you’re willing to wait an infinite amount of time for your results. If no system can sample data perfectly, however, why do we bother with sampled time systems? The answer, of course, is that while youcan never be perfect, with a bit of work you

can design sampled time systems that are good enough

 

 

 

I have already conceded to "good enough" long ago....

 

there are criteria in the theorem that don't meet "real world" scenarios (e.g. only applicable to band limited signals)

 

most people agree that increasing the sample rate can improve SQ because of other reasons related to filters

 

 

[semente]

1 minute ago, beerandmusic said:

 

I have already conceded to "good enough"

 

Why? Just because some guy wrote something which you don't understand but think it somehow supports your beliefs?

[beerandmusic]

4 minutes ago, semente said:

 

Why? Just because some guy wrote something which you don't understand but think it somehow supports your beliefs?

 

My beliefs are based on the same reasons your beliefs are based on...

 

https://en.wikipedia.org/wiki/Belief

 

I never doubted the theorem, just it's application or comments many people make without knowing what they are saying.  Likewise, i knew it was not able to "perfectly capture all"....like some suggested.

[adamdea]

7 minutes ago, mansr said:

The difference is that this outside knowledge is easier to acquire and validate than that concerning our ability to perceive sounds.

Possibly as regards bandwidth- but just as easy to ignore if one takes the fluctuations of one's listening experience and one's personal interpretation of the meaning of those experiences as the only critical data, alongside a general faith in progress. At one point it looked as though DXD might become a de facto "this far and no further" standard. But forget it. Limits are offensive and if you play someone a dxd file and a 32/ 768 file pretty much the same group of people who can hear the difference between 16/44 and 24/96 will claim to hear the improvement in the 32/768. 

 

When first came to this hobby I assumed that the 16/44 spec must have been cobbled together as a result of the limits of technology in the early 80s and that it made sense that it must be easy to improve on it now. After all I remembered what computers were like in 1982 and assumed that analogies with video were closely applicable. It took me a long time to "unlearn" that.

 

It's useful to learn about the sampling theorem (it being beautifully elegant and worth knowing)  but I think that learning about perception provides what is in many cases the real answer to the eternal question- why do I hear a difference when....

 

And amongst the really clever technically minded people I have met on the internet forums are some who I feel just confuse the issue by finding brilliant and subtle technical explanations as to why there might be a peculiarity of the kit or set up leading to perceived result. (I don't mean you)

 

[jabbr]

9 hours ago, audiventory said:

Sometimes for improving of played back sound need to cut information. I mean ultrasound, that can cause audible distortions due intermodulations. The intermodulation products may be listened as noise. In this case cutting of ultrasound above 20 kHz can remove noise.

Oohh ... a system that produces IM distortion with information only above 20 kHz is not so well designed. 

 

Typical DSD DACs might use 80-100 kHz as a corner for the filter.

[firedog]

1 hour ago, mansr said:

If misunderstanding were an Olympic sport, you'd be winning the gold medal every time, summer and winter games.

Hey, he even posted asking me why I responded to him when I “promised’ not to. Only two problems: I was responding to you, and I didn’t make any promises. I responded to another post saying in a previous post I wasn’t responding to him.  

So apparently he is in the running for a medal.

[beerandmusic]

nvm...not worth it.

[firedog]

3 minutes ago, beerandmusic said:

remember you said you won't respond to me...

 

I actually didn’t say that. Just another misunderstanding of yours. But I forgive the mistake, I realize now that your ability to read and understand something in context isn’t the best. 

[psjug]

44 minutes ago, beerandmusic said:

 

there are criteria in the theorem that don't meet "real world" scenarios (e.g. only applicable to band limited signals)

 

Sincere question:  do you get that anti-aliasing filters are applied to MAKE the signal band limited?  Would calling the filters band-limiters help?

[beerandmusic]

34 minutes ago, beerandmusic said:

 

My beliefs are based on the same reasons your beliefs are based on...

 

https://en.wikipedia.org/wiki/Belief

 

I never doubted the theorem, just it's application or comments many people make without knowing what they are saying.  Likewise, i knew it was not able to "perfectly capture all"....like some suggested.

 

But to expand on "why"....if you look at the examples and posts that i provided in this thread, you will see that the problem i had, had to do with "infinite number of time and frequencies".  And sure enough, the theorem had a criteria that did not allow this (e.g. the criteria of bandlimited).   Now that i know that criteria, I am comfortable in accepting based on the criteria of the theorem, especially since white papers suggest that because of this criteria, that it is NOT able to "perfectly" capture all.

 

So again, i concede the theorem is "good enough" ...as that meets my belief.

 

So to conclude:

I have already conceded to "good enough" long ago....

 

there are criteria in the theorem that don't meet "real world" scenarios (e.g. only applicable to band limited signals)

 

most people agree that increasing the sample rate can improve SQ because of other reasons related to filters

[beerandmusic]

7 minutes ago, psjug said:

Sincere question:  do you get that anti-aliasing filters are applied to MAKE the signal band limited?  Would calling the filters band-limiters help?

no i didnt know that...but isn't that one of the reasons to increase the sample rate to move this filtering outside of the audio band?

 

Or if you want to expand on why it makes sense to increase the sample rate related to filtering, some may be interested?

 

I personally have already been content some time ago.

[jabbr]

5 hours ago, beerandmusic said:

the theorem does not speak to infinite frequencies that really exist.

 

it has criteria of a "band limited signal", which i do not understand, and it is my current belief, that it is in that criteria that makes it not applicable.

 

You remain stuck on this idea of infinite frequencies. ??‍♂️ Do you accept that a physical velocity cannot exceed the speed of light? Why would you think a physical frequency could be infinite? Consider calculating the kinetic energy of a particle having such a frequency ...

 

In any case as @mansr noted above, when doing wideband recordings we don’t see frequencies above 100 kHz (at best) so in that case you should be quite happy with 192kHz sampling and call it a day. 

[beerandmusic]

1 minute ago, jabbr said:

 

You remain stuck on this idea of infinite frequencies. ??‍♂️ Do you accept that a physical velocity cannot exceed the speed of light? Why would you think a physical frequency could be infinite? Consider calculating the kinetic energy of a particle having such a frequency ...

 

In any case as @mansr noted above, when doing wideband recordings we don’t see frequencies above 100 kHz (at best) so in that case you should be quite happy with 192kHz sampling and call it a day. 

 

infinite number of frequencies and infinite number of time slices.

[jabbr]

2 minutes ago, beerandmusic said:

 

infinite number of frequencies and infinite number of time slices.

Nope

[beerandmusic]

1 minute ago, jabbr said:

Nope

nope what?