[HN Gopher] Towards Nyquist Learners
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       Towards Nyquist Learners
        
       Author : sleepingreset
       Score  : 28 points
       Date   : 2024-11-17 08:15 UTC (1 days ago)
        
 (HTM) web link (gwern.net)
 (TXT) w3m dump (gwern.net)
        
       | gwbas1c wrote:
       | > In particular, this minimal frequency is twice the bandwitdh of
       | the function.
       | 
       | Careful, this is misleading.
       | 
       | If the peaks of the frequency align with your samples, you'll get
       | the full bandwidth.
       | 
       | If the 0-crossings align with your samples, you'll miss the
       | frequency.
       | 
       | These are why people swear by things like HD audio, SACD/DSD,
       | even though "you can't hear over 20khz"
        
         | 01HNNWZ0MV43FF wrote:
         | How bad is it around the frequencies I can hear as a
         | 30-something?
        
         | luma wrote:
         | You've misunderstood something about Nyquist. A sample rate of,
         | say, 44KHz, will capture ALL information below 22KHz and
         | recreate it perfectly.
         | 
         | There are of course implementation details to consider, for
         | example you probably want to have a steep filter so you don't
         | wind up with aliasing artifacts from content above 22KHz.
         | However it's important to understand: Nyquist isn't an
         | approximation. If your signal is below one half the sample
         | rate, it will be recreated with no signal lost.
        
           | StrangeDoctor wrote:
           | I was just about to post something saying similar. If I had
           | to guess,
           | 
           | >If the 0-crossings align with your samples, you'll miss the
           | frequency.
           | 
           | This is where the issue is. This isn't possible with more
           | than double the sampling rate.
        
             | kevin_thibedeau wrote:
             | It can only happen with a source exactly at N/2 and
             | correlated with your sampling clock. That doesn't happen in
             | the real world for audio.
        
               | marcosdumay wrote:
               | Yep, that's why people do things like 44kHz sampling
               | instead of 40kHz.
        
               | Sesse__ wrote:
               | No, 44 kHz is because you want to reconstruct the (20
               | kHz) bandlimited signal and it's (much) easier to realize
               | such a filter if you have a bit of a transition band.
        
               | mlyle wrote:
               | Anything _close_ to N /2 is going to have varying
               | magnitude that requires filtering and likely oversampling
               | to remove.
               | 
               | How close to the Nyquist bandwidth you can get depends
               | upon the quality of your filtering.
               | 
               | 44.1KHz is a reasonable compromise for a 20KHz passband.
               | 48KHz is arguably better now that bits are cheap-- get a
               | sliver more than 20KHz _and_ be less demanding on your
               | filter. Garbage has to be way up above 28KHz before it
               | starts to fold over into the audible region, too.
        
               | Sesse__ wrote:
               | > Garbage has to be way up above 28KHz before it starts
               | to fold over into the audible region, too.
               | 
               | You brick-wall everything at 20 kHz (with an analogue
               | filter) before you sample it; that's part of the CD
               | standard, and generally also what all other digital CD-
               | quality audio assumes. This ensures there simply is no 28
               | kHz garbage to fold. The stuff between 20 and 28 in your
               | reconstructed signal then is a known-silent guard band,
               | where your filter is free to do whatever it wants--which
               | in turn means that you can design it only for maximum
               | flatness (and ideally, zero phase) below 20 kHz and
               | maximum dampening above 28 kHz (where you will be seeing
               | the start of your signal's mirror image after digital-to-
               | audio conversion), not worrying about the 20-28 kHz
               | region.
        
               | mlyle wrote:
               | Your comment is mutually contradictory-- what is it, a
               | brick wall (impossible) analog filter or a more gentle
               | rolloff as things fold over?
               | 
               | What you really do, these days, is you sample at a higher
               | frequency; you can have an exceptionally gentle analog
               | filter (which will help you make it linear, too). E.g. if
               | you sample at 96KHz, you just need to roll to near zero
               | by 75KHz. Then you can digitally downsample/decimate to
               | 44.1KHz or 48KHz.
               | 
               | Also note for CD audio, it's more like 24KHz where you
               | get worried, not 28KHz.
        
               | Sesse__ wrote:
               | You're mixing up the two filters. The pre-sample filter
               | (before ADC) is defined to be a brickwall (of course
               | impossible in practice, so in reality, it will have to
               | start going off a bit before 20 kHz); the reconstruction
               | filter (after DAC) has a roll-off.
        
           | ImageXav wrote:
           | I think it is you who have misunderstood the Nyquist-Shannon
           | theorem. Aliasing and noise are real concerns. Tim Wescott
           | explains it very well [0] (Figures 3, 10 and 11). If your
           | signal is below one half the sample rate but the noise isn't,
           | you'll lose information about the signal. If your signal
           | phase is shifted wrt. the sampling, you'll lose information.
           | If your sampling period isn't representative, you'll lose
           | information. These are not implementation details.
           | 
           | [0]
           | https://www.wescottdesign.com/articles/Sampling/sampling.pdf
        
           | GlenTheMachine wrote:
           | Nyquist is a mathematical statement. As such, it has two
           | commonly overlooked requirements:
           | 
           | - the signal being sampled has to be stationary
           | 
           | - you have an infinite number of samples
           | 
           | In that case, a sampling frequency of 2N+epsilon will
           | perfectly reproduce the signal. Otherwise there can be
           | issues.
        
       | puttycat wrote:
       | Off topic, this thesis has one of the most concise and
       | straightforward acknowledgments section I saw.
        
       | ks2048 wrote:
       | Title has a misleading domain name (gwern.net). Link is to a PhD
       | thesis titled "Scaling Laws for Deep Learning" by Jonathan
       | Rosenfeld. Not sure why wasn't linked more directly,
       | 
       | https://arxiv.org/abs/2108.07686
       | 
       | https://arxiv.org/pdf/2108.07686#page=85
        
         | jll29 wrote:
         | Generally, if you google a person's name as follows:
         | "Jonathan S. Rosenfeld" +DBLP
         | 
         | you will get their computer science publication list.
         | 
         | From that, you can gather that the two main papers that form
         | the core of Rosenfeld's thesis are these:
         | 
         | https://openreview.net/pdf?id=ryenvpEKDr
         | 
         | https://proceedings.mlr.press/v139/rosenfeld21a.html
         | 
         | (if you prefer to read the gist in fewer pages.)
        
       | svantana wrote:
       | I think the basic premise of this paper is wrong. Very few
       | natural signals are bandlimited - if images were, they would be
       | no need to store in high resolution, you could just upsample.
       | Natural spectra tend to be pink (decaying ~3dB/octave), which can
       | be explained by the fractal nature of our world (zoom in on
       | details and you find more detail).
        
         | wbl wrote:
         | JPEG allocates very few bits to the higher frequency elements
         | of the blocks, especially in chroma. https://vicente-gonzalez-
         | ruiz.github.io/JPEG/#lossy-jpeg
        
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