KVR Audio

Architeuthis · Post by **Architeuthis** » Mon Jun 29, 2015 7:01 am

Ok good.

Now we have arrived at the real question:

Is there a synthesis technique that naturally limits the rate of change of the waveform? A technique where you simply can't create waveforms that change faster than what the sampling rate allows with the controls given to the user and allows for a wide range of timbres. A set of instructions that creates a stream of samples from a simple algorithm, that is as low in aliasing as a standard sine wave.

aciddose · Post by **aciddose** » Mon Jun 29, 2015 7:06 am

No.

Closest you might get is something FFT-based.

Fact is, if it has harmonics those harmonics will alias. There is no way to auto-magically limit harmonic content so as to avoid ever generating aliases that could ever be described as "simple" or "a few instructions".

To avoid aliasing you must never sample a signal. You can't go from virtual/abstract (sine) to digital samples (sample of a sine) without the possibility of aliasing components of the signal you sample unless you apply a filter.

Remember that a theoretical "perfect" sine will also alias if sampled when its frequency is greater than or equal to nyquist.

Something like "if frequency >= nyquist, amplitude = zero" is a filter.

aciddose · Post by **aciddose** » Mon Jun 29, 2015 7:20 am

aciddose wrote: Remember that a theoretical "perfect" sine will also alias if sampled when its frequency is greater than or equal to nyquist.

In addition to this, consider the fact that even with this theoretical "perfect" sine below nyquist in frequency, it is only free of harmonics while its amplitude remains constant.

That means its amplitude never changes, ever... forever.

If you modulate the amplitude of the waveform, you introduce side-bands which are sum+difference of the content of the modulating waveform.

As an example if you modulate a 1000hz sine with a 50hz sine, you get side-bands at 950hz and 1050hz.

Frequency modulation is a bit more complicated. Okay, a lot more complicated. Nightmarishly so.

When we apply something like a Fourier transform we assume the signal is periodic and goes on infinitely in either direction. In the case of a sine, the only way to get a perfect spike in the transform is to have the frequency an exact integer fraction of the sample rate such that if the signal frame/window were "tiled" it would form a perfect continuous signal.

Try that with a rectangular window. When looking at periodic functions this is actually the best way to conduct your analysis as you don't get any windowing artifacts.

xoxos · Post by **xoxos** » Mon Jun 29, 2015 5:55 pm

your boy needs to quit asking questions on teh forum and configure some kinda development environment, so you can type in some numbers and observe the happenings.

*ahem*

digital sound synthesis? yall missed this one. fingersnap. possibly the handclap if you don't mind a bit of palm.

larf ask bedroom producers if they "do it all digitally".

arkmabat · Post by **arkmabat** » Mon Jun 29, 2015 6:25 pm

I was serious about the fish. It's organic and digital at once.

Architeuthis · Post by **Architeuthis** » Mon Jun 29, 2015 7:48 pm

Aciddose, I was never trying to discuss perfect anti-aliasing... maybe the title thread title is misleading. I'm only trying to discuss perceivable aliasing. Even good(?) anti-aliasing techniques simply limit aliasing rather than removing it completely.

I can only guess at what point aliasing is perceivable, be it via modulation of frequency or amplitude or whatever other modulation. I don't know who does what in terms of preventing aliasing in their VST products.

Alright, just... just... bare with me. Can you tell me if there's aliasing in a sound, can you take a look? I know the original intended waveform is not obvious and this is a poor scientific test in general due to lack of information, and my samplerate is 96000 which is maybe cheating, but check this out, just listen/see it for what it is.

http://www.elanhickler.com/_/elanhickle ... g_test.wav

Edit: So the point is that it fulfills the requirements of the first post. Not using an anti-aliasing technique (ok not really, I am intentionally limiting the rate of change of the waveform dependent on the frequency by adjusting the sound generating parameters), it's simple, and flexible enough to create a variety of timbres (not proven in this demo, just take my word for it for now).

Architeuthis · Post by **Architeuthis** » Mon Jun 29, 2015 8:24 pm

This goes back to the previous question:

Architeuthis wrote:Is there a synthesis technique that naturally limits the rate of change of the waveform?

To make it so you can't create waveforms that have foldover harmonics, that requires designing the synth patch so that any frequency you intend to play, you've set it up so that they don't create harmonics above a certain frequency. But unlike conventional sound generation techniques, you actually have a decent amount of reliability in removing harmonics. In essence, the anti-aliasing is intertwined with the sound you are trying to create. You have to make some sacrifices of course, you have to discover the sound you are trying to create, just like in FM, you can't know what you are going to get. You have to play around till you get something you like. Then adjust the sound to fit your intended frequency range.

What is the "perfect" digital sound synthesis technique?