RRod
Headphoneus Supremus
- Joined
- Aug 25, 2014
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Fourier transform is how digital signal processing works:
http://en.wikipedia.org/wiki/Digital_signal_processing
Fourier analysis is at the heart of most analogue (waves you get from your mic or out of your headphones) and digital (0 and 1s) convertions.
Your mic records the signal in the time domain, then you take samples of it to convert it digitally, the sampling rate is chosen and limited byNyquist–Shannon sampling theorem(which is closely linked to fourier transforms)
"My point is we want a higher sampling rate therefore we need to increase the frequency range" <- this means we choose sampling rate which results in a higher frequency spectrum.
"NOT hearing extra sounds at 20+ kHz that has been suggested" <- clearly said what you have just said "And you can't hear those frequencies, so ask yourself how you are possibly sensing something like these "better" transients."
I suggest you read more on the subject. Nyquist-Shannon discusses the necessary sampling rate to fully capture an *already band-limited* signal. In a PCM recording chain, this bandlimiting is accomplished by an anti-aliasing filter at the front of the ADC. Once this band-limiting is accomplished, the signal can then be digitized at the required sampling rate. Note that so far no actual DSP has happened (the anti-aliasing filter is an analog device).
My point on audibility is that the ONLY differences between PCM formats are the bit depth and sampling rate. If we hold the bit-depth constant, as in 16/96 vs 16/48 PCM, then all perceivable differences are due to the difference in sampling rates, and thus to the effects of the additional frequencies of the higher rate.
Anyway, we're in full sound science mode, so I'll see you there if you want to continue.