Truncating prior could get rid of some of the desired content spectrum. Truncating too late can cause some "imaging" of the audio spectrum, to exist in ultrasonic range. Ideal filter then keeps all the audible band, and none of the ultrasonic above 22.05 kHz. If this is not good enough, I have to dig deeper into signal processing. Do you want me to do that?
Ah I understand now, it's basically - how should I call it - "mapping" or "translating" the signal itself within the bandwidth established by the filter. So if it truncated to something ridiculous like 220kHz, that would mean basically everything in the audible range when we would listen to it, would be basically all the sub-bass and bass notes by translation? And all the details beyond 20Khz+ would be inaudible naturally due to headphones and such not reproducing such frequencies (and those that do by chance, do it in vein due to out natural hard-limit of hearing ideal being 20kHz in adolescence as a best case scenario). So we'd basically be only able to hear ~10% of our original audio if it was truncated at 220Khz for example while we played the file back?
I hope I understood that correctly?
Now that I have you, and if you don't mind. These digital filters, are they simply software or something hardware based? Like if a company had an API that can adjust firmware level operations of the DAC chip, would people themselves be able to "make their own" filter. Or is this some sort of silicon-level hardware processes that is set in stone, and making your own would be a question that only makes sense if you were making your own DAC chip itself (which for modern chips I presume would be impossible due to the manufacturing and financial limitations on working with Sigma-Delta chips of this caliber of performance?). What sort of process is it to create a good filter (like what's stopping companies from always hitting the ideal 22.05?) Like for noise and distortion we have a good idea of the factors that come into play that allow for their presence, so I was wondering what sort of thing would be a limitation of a "good" digital filter (I presume one is latency from what I imagine is a problem in the professional field of products).
Oh and just very quickly, is the creation of these filters equal amongst DAC chip varients? Like R2R and such. Is there such a thing as a "analogue" filter of sorts? (Apologies if this question doesn't even make sense to ask).
Thanks again bossman!