Paul McGowan writes:
We all like the idea of getting something without having to pay a penalty: that free sample at the market, a kind gesture, a door being opened when your arms are full. Closer to home, more digital audio information than we started with.
When we upsample a 44.1kHz 16-bit file to a higher rate and depth, like 96kHz 24 bits, we typically get better sound quality. And since the magic of upsampling just sort of works at the touch of a button, we seem to be getting more for nothing. After all, the file size is considerably bigger. There must be more there. Right?
So, how does that work? How can a program know what went missing from the original recording so it can add it back in?
There are actually two things going on. The first, and least important, is interpolation. Interpolation is a mathematical process that adds more data points through intelligent guesswork and statistical analysis. Simply put, if our steps are moving in a predictable pattern: 1, 3, 5, 7 then it’s likely we can add the missing steps: 2, 4, and 6, as additional data points so we wind up with 1,2,3,4,5,6,7.
Perhaps more important is the choice of filters. With standard CD rates of 44.1kHz we need to have a fairly steep filter so we don’t run into trouble above 22kHz. By increasing to 96kHz we can apply a much gentler and better sounding filter to our digital data and this, in my experience, is responsible for the majority of what we might consider better sound from upsampling.
I put together a video on the subject you can watch here.