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.
Quote from: [JAZ] on 18 October, 2011, 05:09:35 PMAnd 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.
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.
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.
This still leaves you with having to split up the 44.1 thousand samples amongst several instruments in a full band.
How is it possible to know absolutely nothing...and still have enough self-confidence...(to invent your own "facts".)
I wonder why those topics are such often repeated, even if I in the past also contributed to their length.
48 kHz is the optimal sample rate for human audio
theoretically better source for subsequent conversions
The highest frequency at 44.1 kHz 16 bit rate is 22.05 kHz according to Nyquist. Right?.........To wrap it all up, a higher sample rate would catch even the smallest differences so that what overlaps at 44.1 kHz doesn't overlap at 192 kHz. Then you don't need some complex work around to try and figure out where the bass notes are.
Quote from: jumpingjackflash5 on 20 November, 2016, 12:53:33 PMtechnical/futureproof reasons.Quote from: jumpingjackflash5 on 20 November, 2016, 12:53:33 PMtheoretically better source for subsequent conversionsThe needle on my speculative//hand-waving bullshit meter just moved. How fun!
This, however, does not claim that using 16 bit source for that purpose would be practically (audibly) different.
Just got done recording a bat conversation @ 576 KHz. The problem is I couldn't hear any of it.
What sort of bandwidth does a bat chirp cover? As in, what are the typical lowest and highest frequencies present in a chirp?