On my Sonar it is something like 80 dB below signal, so that is fine.Max M. wrote: ↑Sat Mar 16, 2019 1:23 pmlfm
And this is what we see in Sonic Analyzer printing all these multiple bands showing other frequency components than original 1 KHz.
In properly smoothed 4s fade-out of 1kHz sine, the distortion is hundreds dB below the usual noise level. So there's something wrong with your measurement method (or with that Sonic Analyzer).
But StudioOne is just 50 dB below doing the same thing.
And as I recall as bad for Logic as well.
So think the tool Sonic Visualiser is correct somehow. It has really nice sense of each cursor position you can see all the way how these extra harmonics occur and what levels and freqency and all.
I spent almost 6 hours doing the tests on various daws and various settings I thought would interfer. Difference is massive.
No change in automation all is fine, doing the ramp - all goes wild.
Even recording one track to another - moving a fader manually doing that - same thing.
Got it thanks.---
For the rest, your main mistake is that you assume that a single sine-shaped period (half or full cycle) is itself a "sinewave" having only a particular frequency (w/o any other bands).
It is not. The cycle becomes a sinewave w/o any extra-band harmonics only when repeated infinitely (ideally infinitely, in reality it does not have to be infinite but just long enough for the parasite harmonics to go below SNR).
Thus your theory is wrong right at its first step by expecting that you either don't have any distortion by chaining two sine-shaped cycles of different levels. Or that this instant X distortion is somehow lower than "N times X/N" distortion of smoothing, just because you preserve the shape of a single wave cycle.
(+ a few more mistakes at next steps).
So assumption that instead of destroying frequency content of each period, and half period for that matter - doing it once a half period would be less distortion is wrong?
Thinking is that 48 samples all differ from what a proper sine should be - would be worse - than differing once every 24 samples?
As far as I can tell - HF content is blurred more if moving each samples value - since intersamples between fundamentals is the HF content, the harmonics on that or fundamentals of other higher pitched instruments - if we are talking full music content and not just one sine.
So not getting the reasoning that you would move that into LF and below?
It got to be some technique you know of that daw makers don't.
Or is it costing to much cpu to do it right, so they cheat?
Thanks for input anyway, I'm learning every post....