[sdiy] A new shade of pink (noise)
Richie Burnett
rburnett at richieburnett.co.uk
Tue Nov 24 21:14:46 CET 2020
Thanks for that link Eric. Very interesting, I always wondered how they
achieved 128 voice poly on the Streichfet.
The section towards the end about using BLEP techniques to emulate "variable
sample rate playback with Zero-Order-Hold" on a modern DSP with fixed sample
rate is very interesting. To me this is the secret to getting that gritty
PPG type sound when playing back wavetables at low pitches. I would call
the additional spectral content "Imaging" rather than "Aliasing" though,
because it is produced by repeated images of the spectrum due to the
sampling process that aren't being supressed properly during playback. Not
aliasing that I think of happened during recording of the wavetable. A lot
of people do seem to call the filter that follows a DAC an "Anti-aliasing
filter" though, when it is really an "anti-imaging filter" in my opinion, or
even "reconstruction filter" is a better description, I think.
-Richie,
-----Original Message-----
From: Eric Brombaugh
Sent: Tuesday, November 24, 2020 6:54 PM
To: synth-diy at synth-diy.org
Subject: Re: [sdiy] A new shade of pink (noise)
The shortcuts that Stenzel takes in the implementation bear a strong
resemblance to the band-limited interpolation oscillator bank approach
that he described in his ADC17 talk a few years ago.
https://www.youtube.com/watch?v=lpM4Tawq-XU&t=6s
I think I see a pattern in his thinking...
Eric
On 11/24/20 5:57 AM, Tom Wiltshire wrote:
> There’s an interesting paper by Stefan Stenzel on Github, describing a new
> digital pink noise generation algorithm:
>
>
> https://github.com/Stenzel/newshadeofpink/blob/master/newshadeofpink.pdf
>
>
> Has anyone seen this? I’ve read it, but there’s one thing I don’t
> understand. In the paper, he talks about taking multiple 1-bit noise
> sources (as you would in the Voss0-McCartney algorithm) but instead of
> using a “zero order hold” (e.g. “stretching” each sample) to decrease the
> sample rate, he uses linear interpolation.
> This is the bit I don’t get - how do you linearly interpolate a 1-bit
> signal? There’s nothing in between!
> He mentions at one point that the digital signal is to be interpreted
> as -1 or +1, which would mean that there is a 0 between the two values,
> but I still don’t understand how that makes sense when it’s a digital
> signal and not a bit of signal processing maths formula.
>
> Any clarifications appreciated. I’d like to understand this method better,
> but the paper is very brief, assumes quite a lot of background I don’t
> have, and doesn’t provide any worked examples for illustration.
>
> Many thanks,
> Tom
>
>
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