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What's the point of higher sampling rates in audio?, Such as 96 and 192kHz.
Roseval
post Nov 15 2011, 22:49
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Having digitized some vinyl I know for sure I won’t go through that again.
A very time consuming affair.
If I had to do it over again, I would go for as much overkill I can afford.
Better save then sorry



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2Bdecided
post Nov 16 2011, 13:38
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QUOTE (pappaapa @ Nov 9 2011, 00:10) *
But still, it leaves out all the sounds that hasn't been recorded through a mic. Synths, or a lined bass.
And more important (i'm guessing) it leaves out all the sounds together, in a mix.
It's a good job none of this really matters, because the measurable distortion in speakers is several orders of magnitude larger than the measurable distortion in air. If this mattered, these recordings with high levels of ultrasonics would be unlistenably bad.

Luckily, they're mostly OK. And any distortion in air, 10s of dBs lower than the distortion in the speakers, is irrelevant.

Unless, I suppose, you're running a PA rig at home - but then it's the distortion (and damage) in your ears that should concern you.

Cheers,
David.
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HTS
post Mar 20 2012, 03:35
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QUOTE (Arnold B. Krueger @ Nov 1 2011, 15:44) *
QUOTE ([JAZ] @ Oct 18 2011, 17:09) *

And are you 100% sure that the plugin is samplerate aware???

I've been part of this past year fixing most of the plugins of my program* so that they don't perform differently at different sample rates.


If nonlinear processing is done in the digital domain, certain high harmonics can reflect back down into the audible range. Any signal that is generated in the digital domain above the Nyquist frequency aliases around the Nyquist frequency. For example, an 8 KHz tone in a 44.1 KHz sampled sytem is distorted by a fourth order digitally-implemented nonlinearity and would be expected to produce a fourth harmonic at 32 KHz. Since 32 KHz is 10 Khz higher than the 44.1 KHz Nyquist frequency of 22 KHz, the 4th harmonic is aliased down to 12 KHz where it is far more audible than it would be in a digital system with a much higher sample rate or an analog system. In a 96 KHz system, the same nonlinearity would produce a foruth harmonic at 32 KHz where it would not be audible.

Aliasing can also impact certain dynamic range compression or expansion algorithms that are based on nonlinarities implemented in the digital domain.

Most kinds of processing that are used in music synthesis and mixing are linear, so aliasing is not a common problem.


Aren't algorithmic reverbs based on non-linear transformations? Especially these days they offer many types of adjustments to the parameters for effects.
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Woodinville
post Apr 7 2012, 05:41
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QUOTE (HTS @ Mar 19 2012, 19:35) *
Aren't algorithmic reverbs based on non-linear transformations? Especially these days they offer many types of adjustments to the parameters for effects.


How so? huge IIR filters are not nonlinear.


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knutinh
post Apr 9 2012, 12:20
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QUOTE (Woodinville @ Apr 7 2012, 06:41) *
How so? huge IIR filters are not nonlinear.

I was under the impression that standard reverb algorithms introduced time-variant "stuff" in the filter topology to reduce audible artifacts, making them non-LTI?

Nevertheless, I assume that the kind of nonlinearities "warned" about by Arnold in terms of aliasing was of a very different kind: Signal-dependent gain with non-smooth envelope.

-k

This post has been edited by knutinh: Apr 9 2012, 12:22
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