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No idea of the technicals, other than sampling at double the audible frequency does- to this layman - seem logical it would give you a wave more diverged from r

No idea of the technicals, other than sampling at double the audible frequency does- to this layman - seem logical it would give you a wave more diverged from reality as frequency increases - and at the absurd extreme a 22khz (inaudible to most) square wave if sampling at 44, 4 points to represent 11k etc. I don't see how it can be otherwise, just as interpolating and anti aliasing is often right but cannot always correctly insert missing pixels - see the attempts to use AI to make hires versions of Mario and other 8 bit assets, some work, many don't. It was only an idle conjecture after the main point, which was meant to be about mastering, so I'll expand that a little. One of the places I worked at in the 90s was connected with a well known recording studio. We got to blind demo from master tapes how their latest attempts at going digital were going compared to analogue a few times. At least a couple with whatever graphs they'd measured. 25-30 years later, I don't have much recollection of the how or what. Aurally, it wasn't something subtle to listen out carefully for, flaws were plain, therefore memorable. Surprising in the era of digital and CD growth and their supposed "perfect" sound forever, and DDD being marketed as always preferable to ADD or AAD. Not so surprising when you realise it was the era of all ADCs and DACs at any cost being shit. By the time I left, analogue mastering was still preferred by the engineers and still the studio's first choice for albums they recorded. Can't explain that technically either, though they clearly thought, and could demonstrate analogue was still doing a measurably better job. Were they misunderstanding? shrug.

jancsika 8y ago
> No idea of the technicals, other than sampling at double the audible frequency does- to this layman - seem logical it would give you a wave more diverged from reality as frequency increases - and at the absurd extreme a 22khz (inaudible to most) square wave if sampling at 44, 4 points to represent 11k etc. I gave a link that clearly explains and tests the technical aspects using analog equipment. It's an extremely well-thought out and well-researched video. Did you watch it? I'm not saying you didn't hear something. I'm just saying the phenomenon you described does not accurately characterize the current state of digital audio, even when using commodity hardware.
rcxdude 8y ago
> No idea of the technicals, other than sampling at double the audible frequency does- to this layman - seem logical it would give you a wave more diverged from reality as frequency increases - and at the absurd extreme a 22khz (inaudible to most) square wave if sampling at 44, 4 points to represent 11k etc. I don't see how it can be otherwise, just as interpolating and anti aliasing is often right but cannot always correctly insert missing pixels - see the attempts to use AI to make hires versions of Mario and other 8 bit assets, some work, many don't. It was only an idle conjecture after the main point, which was meant to be about mastering, so I'll expand that a little. This is not correct. If the input signal is entirely below nyquist, the sampled values (ignoring quantisation of the levels, which is a different thing) exactly capture the signal, should you appropriately reconstruct it. Now, a perfect reconstruction (or antialias) filter does not exist, which is one of the reasons why we use 44kHz or 48kHz instead of the 'ideal' 40kHz, but this recreation is extremely accurate, to the point that modern digital approaches will outright outperform analogue techniques. The same is true for images, incidentally, but most media does not exceed the limits of human vision yet (nor do most display mediums make for very good reconstruction filters). I highly recommend watching the video, it demonstrates this very well.

jancsika 8y ago

rcxdude 8y ago

> No idea of the technicals, other than sampling at double the audible frequency does- to this layman - seem logical it would give you a wave more diverged from reality as frequency increases - and at the absurd extreme a 22khz (inaudible to most) square wave if sampling at 44, 4 points to represent 11k etc. I don't see how it can be otherwise, just as interpolating and anti aliasing is often right but cannot always correctly insert missing pixels - see the attempts to use AI to make hires versions of Mario and other 8 bit assets, some work, many don't. It was only an idle conjecture after the main point, which was meant to be about mastering, so I'll expand that a little. This is not correct. If the input signal is entirely below nyquist, the sampled values (ignoring quantisation of the levels, which is a different thing) exactly capture the signal, should you appropriately reconstruct it. Now, a perfect reconstruction (or antialias) filter does not exist, which is one of the reasons why we use 44kHz or 48kHz instead of the 'ideal' 40kHz, but this recreation is extremely accurate, to the point that modern digital approaches will outright outperform analogue techniques. The same is true for images, incidentally, but most media does not exceed the limits of human vision yet (nor do most display mediums make for very good reconstruction filters). I highly recommend watching the video, it demonstrates this very well.