🚀 go-pugleaf

>As we all know, the classic technique of sample rate conversion would have us
>upsample to the lowest common multiple, low-pass filter, and downsample to the
>desired rate. The only problem is that the LCM of 44.1k and 48k is 21.168 Meg!

Why not, on the fly in a DSP chip for example, perform a first order hold
function (linear interpolation) and use the interpolated values to feed a
48K sampling rate output function. You could perform the first order hold
every 21.168M to proove your point, but all you need do is figure out the
interpolated value at a 48K rate. The DSP chip would still have to be pretty
fast since a TMS320C25 would be hard pressed to do this and perform it's
A/D and D/A interface functions ...

Sorry, no code, since that would be telling secrets :-) :-)

Ciao, Mark Salyzyn  mark@adec23.UUCP