[hpsdr] sampling

[Darrell Bellerive]

On April 21, 2007 11:25 pm, Glenn Thomas wrote:
> ***** High Performance Software Defined Radio Discussion List *****
> The only other explanation I can think of due to the fact that analog
> audio low pass filters have a significant phase shift, starting at
> about half the cutoff frequency and increasing to 90 degrees or so at
> cutoff. This phase shift is a form of distortion that, when it occurs
> between 10 KHz and 20 KHz, makes the "imaging" beloved by audiophiles
> difficult to achieve. The faster sampling rate allows the use of an
> anti-alias filter with a higher cutoff frequency, thus moving the
> frequencies with significant phase shift out of the audible range.
>
> Some years ago, the CD marketeers were pushing "oversampling" as the
> latest advance in HiFi sound. I went to the Hi-end HiFi show and
> asked most of them "what does over sampling buy me?" I collected no
> more than a large set of dumb looks. What might be going on (though I
> was unable to confirm it) is that the 44 KHz 16 bit samples were
> being interpolated to provide an apparent higher sample rate with
> increased resolution. Of course, the only information they had to
> work with is what is contained in 44KHz/16bit samples, so the output
> is theoretically the same no matter. However, providing interpolated
> samples at a faster rate would allow them to use an anti-aliasing
> filter with a higher cutoff frequency, thus avoiding the phase
> distortion present in a 20 KHz LPF. OTOH, I don't know if they
> actually did this or if the CD oversampling was just marketing bull.

Oversampling is just a low cost way of getting around the expense of 
designing and building high quality filters. The accounting department 
forces the engineering department to reduce production costs, and the 
marketing department heralds this new low cost technology as the next best 
thing to sliced bread.

-- 
Darrell Bellerive
Amateur Radio Stations VA7TO and VE7CLA
Grand Forks, British Columbia, Canada

 1177292140.0