CFWhitman

I was reading a little bit about this and realized that some of what I said in my last post is out of date. It reflected the state of affairs when CDs were introduced 30 years ago. However, current digitial recording equipment can record audio without ever passing it through an analog filter of any kind. Instead, the signal is digitized at a high sampling rate and passed through a digital filter before ever being saved at a more practical sampling rate. Apparently a lot of newer music is never even originally recorded at a higher rate than 48kHz (though 24 bit is of course used, and necessary, for the stages before mastering).

As to sample rates, recording has been done at high sample rates for a long time. My comments were only relevant to the playback. Once the recording has been digitized, you can safely apply a digital filter and master it at a much lower sample rate with no loss of relevant data.

As to interpolation, your comments are irrelevant (except that linear interpolation would never be used; that would be a guess, and a bad one). As long as you have enough sample points to accurately recreate a sound wave, you can accurately predict any sample point along the sound wave.

As I mentioned in a comment on the last article, It's similar in principle to predicting points along a straight line. Once you have the endpoints of a straight line, you can predict any point in between with relatively simple math. With sound waves, once you have enough samples to accurately describe the sound wave, you can predict any point in between two samples with perfect accuracy, though the math is not so simple.