Any headline with a question...
Of course it depends on what you want to do and what you are used to.
Technically, regardless of the rest of the product design there are high quality potentiometers available both as linear faders or rotary knobs. I guess dirt is more likely to get into a linear fader and make it scratchy - especially in a club environment.
I think what's going on with consonance is that there is an implied lower pitch that all the partials are harmonics of - that also works when there are just sine waves.
The Sony PCM-F1 was one of these - and was the first time I got confused between "is this live or is this the playback?" because until then it had been obvious.
or if the lock is on some smaller part you might be able to still render the audio but e.g. not update the settings that changed yet, and try again next time.
Phase itself looks random, but what makes the sound blurry is that the phase doesn't line up like it should across frequencies at transients. Maybe something the model could grab hold of better is phase discontinuity (deviation from the expected phase based on the previous slices) or relative phase between peaks, encoded as colour?
But the same thing could be done as a post-processing step, finding points where the spectrum is changing fast and resetting the phases to make a sharper transient.
Yes it's a good decision, and 'the standard' for audio in the same way as 32-bit ARGB makes everything easier for graphics.
Of course there are some things that are better done with ints or maybe doubles, like keeping track of the time position, but for the sine wave generator it doesn't need to know how long it's been running in total, just the position in the current cycle, so floats are fine for that.
Each string has it's fundamental frequency and harmonics. Some of them coincide or are very close in frequency, but some are unique to one string so can be picked out and their frequency measured with a FFT of sufficient resolution and used to indicate the tuning of the string it must be a harmonic of.
(Polytune is more robust than just doing this and does work really well, but if the guitar is really out of tune the method falls apart and you have to use its one-string-at-a-time mode)
Diminishing returns - the polynomial is already good relative to the unpredictability of the waveform. Adding degrees of freedom to the analyzed/predicted waveform shape at some point needs as many bits to store as a larger residual signal would have.
> One way to get around this would be to decrease bucket size by increasing the FFT size.
A better way is parabolic interpolation, which is in the source code but not mentioned in the article - and that works for finding the fractional position of peaks in the FFT or in the autocorrelation.
An even better way is by comparing the phase of the peak in two successive FFTs: If the signal phase has changed by X degrees after T seconds, what's the nearest frequency to the bin centre that can be true for? (this is the main thing a "phase vocoder" does)
I guess there are enough in existence for those that want them (and not a digital recreation). An electro-mechanical remake would maybe not be any cheaper than a restored original, and might lack (imaginary) mojo.
There's a lot less going on inside a Rhodes piano.
If we sensed sound like we sense colour we'd only hear the amount of bass/mid/treble not the individual pitches.
But if there's such a thing as a non-linear medium for light, then transmitting two wavelengths through it should do interesting things depending on the ratio of the wavelengths?