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Understanding Sample Rate


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16 minutes ago, beerandmusic said:

 

what say you?

I doubt you can hang with the mansrs or jabbrs of the site....but please try to share anything besides a mock or a troll statement.

 

What I said of course.

 

You asked a question. I answered 100% factually. That is not mocking nor  trolling.

 

I do not know the totality of mansr's or jabbr's knowledge.

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1 minute ago, Blackmorec said:

It suddenly dawned on me why sampling at double the highest frequency is sufficient. May seem obvious to many but initially it wasn't to me.....All sound waves are sinusoidal. All you need is the amplitude and the number of oscillations per second (frequency) to perfectly describe the signal...whether you're recording (A to D) or replaying D to A.

 

I learned everything about ADCs and DACs from signal processing, and there you need at least 5 data points to describe a peak, a lot more would be better. But if the peaks are always sinusoidal i.e have the same, defined shape, all you need to know to describe it perfectly are amplitude and frequency.

 

Given the above, there's no reason why higher sampling rates should better describe a particular sound wave within the defined spectrum.  Ha! Got there in the end.

 

So, why is it the that higher sampling rates do sound better? A few months ago I had a demo of some YG Carmel IIs driven by AVM electronics.  I was listening to standard CDs and wasn't happy with the sound, which lacked air and acoustic resonance, replaced instead by the typical digital sting in the tail.  I complained to the dealer, who pushed a button on the amp, after which all was sweetness, light and all the air and acoustic resonance you could want, with no trace of digititis. Clearly it was unresolved components of the music that had caused the problem. As soon as they were properly resolved, away went the problem.  

So what had the dealer done? Switched to the amp's upsampling mode.

Got nothing especially  to do with sound. It's about 'information'.

 

Nyquist/Shannon  works on bus  timetables.

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24 minutes ago, beerandmusic said:

 

another white paper on nyquist and the theorem was for telegraph systems and should not be taken at face value...as it is for bandlimited signals.

 

The more i read, the more i believe that i would want to sample more than just twice the highest frequency.....unless you just want "good enough".

 

http://www.analog.com/media/en/training-seminars/tutorials/MT-002.pdf

THIS IS THE RELEVANT PART. Note carefully  what it says.

 

"Simply stated, the Nyquist criterion requires that the sampling frequency be at least twice the highest frequency contained in the signal, or information about the signal will be lost"

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18 minutes ago, beerandmusic said:

 

i am sorry it bothers you that i understand things you do not.

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

if you don't get it, i am sorry.

 

I thoroughly enjoy nazi flying saucers, alien flying saucers, ley  lines, the 'power' of stone circles, AG used for building the pyramids, witchcraft,  dirac communication, talking snakes, ancient airfields in Peru, the still used  Viking one in the UK,  why everyone is now being abused rather than stalked,  the Bermuda triangle, Merlin the Wizard, etc,  etc.

But  I don't understand any of them.

 

Please can you explain them? (Any ten will be sufficient.)

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14 minutes ago, sandyk said:

 

 It isn't.  24/48 would be better though, but even then ....

I would have chosen 24/48 had I been in charge  but apparently there were size limitations, both physical  (maybe vibrations?) and 'digital'  on the disc at the time.

 

We (IBM) had a very 'floppy' disc with a vaguely 'half cone' bump on the  stationary base over which it flew to make the also flying heads (heads  never touch)  closer to the surface but we never got it to work properly, The disc  always 'scrunched up' after a while.

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7 minutes ago, sandyk said:

 

The majority of those giving you a hard time are correct in most respects, and mean well, but they don't have a mortgage on the truth. There are still many things that we are currently learning , especially in the area of very clean power supplies etc.

I note also a backdown in some areas such as depth and height of stereo images from (in some cases) impossible, to accidental and non predictable accuracy .

Things may get even more interesting if Mani is able to convince Mansr  that he isn't imagining what he reports .

I meant well, but I eventually got irritated because he wouldn't even  listen to anyone who disagreed with his beliefs.

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6 minutes ago, sandyk said:

 

 Yes, I am aware of that being the preferred option , but back then, even 16 bit was hard to achieve, hence the 14 bit players available at the time including Nakamichi etc. ( I had a Nakamichi player at a later time)j

It was much  more fun when things wren't so 'advanced'.

 

The 'top' Philips non-portable cassette players had 'dynamic  noise reduction'. It was much  better than  Dolby when well-adjusted and didn't need  a special recording 'curve'.

 

In production Philips never adjusted  it at all. I made one, exactly to Philips circuit,  slowly  turned the  'preset type' pot, and you could gradually hear the noise vanish and come back again if you turned it too far.

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26 minutes ago, adamdea said:

Whilst I mainly agree with you, I would say that there is far more established knowledge in audio than most people seem to think or draw on (see for example the dreadful farrago of a discussion of auditory localisation on another thread); it's not quite so mathematical as the sampling theorem (being amongst other things experimentally derived, and in more developing fields)  but sill it's solid scientific understanding, fairly easily accessible  and frankly no one seems to give a damn. 

Most people insist on starting from what they think they know and only cherry-picking technical stuff so far as it seems to support that. I find it extraordinary that we still have discussions about the sampling theorem; I would have assumed that everyone involved would either have understood it by now, given up hope or decided that they didn't care. I would also have hoped that people would stop posturing and pretending either to understand it or to be interested in trying to understand it. Why bother?

Our intuitive understanding of every process involved (maths, engineering, psychoacoustics no doubt more else) is muddled and about as helpful as the intuition that thunder must be caused by the gods moving chairs around. . As long as that intuitive understanding is valued above any actual knowledge we get the wrong answer. Where much of it is debatable that may not make much difference. Where it isn't debatable is just leads to a very long very boring river of sludge, enlivened by the occasional interesting anecdote, charming generous and warm-hearted character or flash of wit.

'Digital', audio or not,  is fully understood  and has been for a very long time. We made it  from scratch to work as we intended it to, we didn't 'discover' it. At first it was entirely 'mechanical' but the principles have not  changed at all.

 

The 'small area' where it changes from digital to analog (if reguired) is only fully understood in its digital part.

 

The only 'digital' advance needed (or possible) are in computing speed, which is not important in audio as it already vastly faster than audio will ever need. .   

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49 minutes ago, sandyk said:

 

He doesn't need to. He only needs to be convinced that Mani can hear them. I wouldn't sell Mani's hearing or equipment short either.

Despite what many appear to believe, there can be a big gulf between the SQ of typical gear and more well designed equipment where fewer compromises have been made, especially in the areas of power supply isolation between Digital and Analogue areas. The ready availability of extremely low noise voltage regulators such as the LT3045 etc. will result in further audible improvements as their use becomes more common place.

He's already pre-judged mansr before he's  even got started on anything.

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59 minutes ago, pkane2001 said:

every ....OP made up his mind and doesn’t want anyone to keep telling him he’s wrong. His approach to knowledge allows him to bypass logic and reason and to stick to an unprovable belief based on an incorrect intuition. What else is there to discuss?

 

Nothing.

 

After the first  few posts there never was. Because he's  not only ignorant of science he's ACTIVELY against it despite it works for him 24/7.

 

So whyTF did he ever ask the question?

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2 minutes ago, adamdea said:

Yes but you still can't get answers out without (consciously or unconsciously) applying a psychacoustic model. If we were cats, we might need a higher sampling rate than 44.1 kHz.

That is not because of any limitation in the maths, it's just that our specification of what we  need from the system comes from something outside. 

I have been dying to get my hands on someone who really understands current thinking on psychacoustics because the information about time resolution of the perceptual system is a bit difficult to get hold of. I have got as far as understanding that there seems to be some current thinking about the system integrating over a 6ms window. I have a hunch that this might be important to certain current controversies.

Cats and psychoacoustics have zero to do with computing or the  sampling theorem.

As I previously said it even works fine on bus timetables.  And as mansr said it will work  100% on  things not yet discovered.

 

The rest is 'vague (not your fault). It's like  'logic machines' such as computers. Be very  exact in  what you wish for :) 

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1 minute ago, adamdea said:

Of course not, but they do have a lot to do with the question what is the minimum sample rate required to capture all the information you can hear. You need to understand the sampling theorem and know something about human hearing to answer that question. Like durrr

He appears to point blank refuse to understand anything he doesn't understand already. Unless perhaps  our sample rate is too low.

 

The "something  about" is the maximum frequency we can hear.  All the 'time' and 'impulse' stuff follows from that and most people's 'understanding' of it is nonsense because they don't understand that fundamental point.

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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. 

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