On Saturday, 9 April 2016 at 14:15:38 UTC, Nordlöw wrote:
Has anybody more than I thought about representing the sample
rate of a sampled signal collected from sources such as
microphones and digital radio receivers?
With it we could automatically relate DFT/FFT bins to real
frequencies and other cool stuff.
Maybe we could make it part of the standard solution for linear
algebra processing and units of measurement in D.
Destroy.
What problem would that solve?
In data formats like WAV the sample rate is stored inside.
In audio processors like plugins the sample rate is passed out of
band.
If you pass the sample rate along with the audio data, then
you'll have to support the hypothetical case where it would
change every buffer. Hence why out-of-band is often preferred.
