DCO's, Anti-Aliasing, and Filters

[Martin Czech]

> ANY and ALL digital signals have aliasing...There is no way around it.
> This is because between the samples, or dots that represent the
> waveform, ANYTHING can happen, and we would never know about it from the
> digital recording. This is really where aliasing occurs. Its about the
> space between the dots.
> 

SNIPPED

Very good description.

I just want to add that some proposals to this and related threads
included sample rate conversion also, this is a related matter.

Eg.  the front end of a system runs with Fs 30 Hz, and the back end
with 50 Hz sample rate (low figures, so I can calculate it in my head)
or vice versa.  First you would go to an intermediate rate 300 Hz,
the smallest integer multiple of both. Ie. every 10th (30 Hz) or 6th
(50 Hz) sample has a real value, the others are zero. Then this would be
lowpass filtered so that the Nyquist constraint is kept for the wanted
output rate. This lowpass filtering will fill the zero samples with
interpolation information.  Then you would keep only  every 10th or 6th
sample and throw the others away. I think this is the only way for high
quality sample rate conversion, ie. using an intermediate rate that is
the smallest integer multiple of both rates.

What does this mean for the usuall 44100 to 48000 Hz conversion?
Intermediate rate of 7,056,000 Hz !  Now I know why the high quality
sample rate conversion algos need so long! There are faster algos that
need less computation time, but they sound not that clean.

Or am I completely wrong? Is there a clean method that avoids these
extreme high intermediate rates ?

m.c.