[sdiy] Digital accumulator VCO core?

cheater cheater cheater00social at gmail.com
Tue Feb 16 02:09:19 CET 2021


I'd much rather prefer some randomness, so that the transitions don't
all sound the same.

On Mon, Feb 15, 2021 at 11:35 PM Tom Wiltshire <tom at electricdruid.net> wrote:
>
> The point as not to remove the aliases by *filtering* but by *cancellation*.
>
> Imagine you could create a waveform that consisted of only “anti-aliases” that would eliminate the aliased harmonics when applied to the original signal. That’s what we’re doing.
>
> Yes, the usual scenario is that the sample rate is fixed and therefore the band-limited data only needs to be derived once. You can do all the hard sums offline, generate one massive look-up table of the band-limited edge correction, and then you just paste the right bits into the right places and all the aliasing disappears. That’s why I described it as “magical”!
>
> Tom
>
>
> > On 15 Feb 2021, at 22:02, Brian Willoughby <brianw at audiobanshee.com> wrote:
> >
> > I don't agree with this 5-step simplification. Once you create a naive raw impulse, the data already includes an infinite series of harmonics that are aliased and cannot be removed by filtering. Even two passes of a filter cannot remove the aliases that fall in between the in-band harmonics.
> >
> > In contrast, you can synthesize a band-limited impulse easily using the sin(x)/x formula. There's no need to low-pass filter a synthesized band-limited impulse. The windowed version just leaves out values of sin(x)/x for large values of abs(x)
> >
> > What I don't know about is the math to convert a standard band-limited impulse into a minimum-phase band-limited impulse. Anyone?
> >
> > Either way, a synthesized band-limited impulse can be converted to a band-limited step via integration, and this can be done in advance. I'm assuming that the sample rate of the converter is fixed, and thus the band-limited data need only be synthesized once.
> >
> > Brian Willoughby
> >
> >
> > On Feb 15, 2021, at 04:06, Richie Burnett <rburnett at richieburnett.co.uk> wrote:
> >>> If you do a polyblep you correct the sample just before and just after the transition to "blend" the fractional-sample-position step into whole samples, but I don't get what the curve is supposed to be or how you design it.
> >>
> >> Ok, here goes...  (See attached JPG image.)
> >>
> >> 1. Start with a raw impulse (A).
> >> 2. Pass this through a boxcar FIR filter to get a band-limited impulse (B).
> >> 3. Pass this through the same boxcar filter again to get an even more band-limited impulse (C).
> >> 4. Now integrate this band-limited impulse to get a band-limited step (D).
> >> 5. Finally subtract a naive step from the band-limited step to get (E).
> >
> >
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