Upsampling..........

Actually Josh, it's a very good question as there isn't complete agreement on the answere.

Josh, you should have posted your findings. There is still a lot of discussion about the subject. We have been talking about this a bit on the http://www.newaudiosociety.com site for a while. Let us know what truth you arrived at.

Ok. From what I have gathered I can surmise that the when the digitial signal is upsamapled to a higher frequency the converter can extract more data by interpolating ( using some sort of algorithm i assume ) which will give a smoother/fuller sound, closer to the original recording. But, what is the algorithm used?

It's been a while, but I think it goes like this ....

Upsampling does not increase the detail, nor does it increase the bandwidth of the signal, but it alters the sampling frequency, and so it pushes the aliased signals (see Nyquist's theory of sampled signal) to higher frequencies.

This allows the anti-aliasing low-pass filter to have a slower rolloff. It is easier to design a filter with linear phase and lower passband ripple if the rolloff is less steep.

So the whole point of upsampling was to simplify the anti-aliasing low pass filter, so that it causes less degradation to audible frequencies.

Performing any kind of interpolation would be akin to doing some of the filtering in the digital domain. I believe the interpolation bit may be the difference between oversampling (old technology) and upsampling, but I'm not exactly sure.

Funny you should ask about the algorithm. I was thinking about this last night (because I really have no life !)

A perfect CD player would have an anti-aliasing filter that was a perfect brick wall. Below 22kHz signals would be completely untouched, and above 22kHz would be infinite attenuation.

In the time domain this equates to a sin(x)/x impulse response.

So my guess is that a good place to start for the interpolation algorithm would be the sin(x)/x waveform.