Pat Farrell;182627 Wrote: 
> Muggy wrote:
> > The upsampling involves simply adding zeros after each data point to
> > pad the data out - 146 zeros after every data point.  This leaves
> the
> > frequency content unmolested over the bandwidth of the original
> > signal.
> 
> I don't think this is correct.
> I'm not even sure where you would "add zeros" to anything.
> 
> There are two approaches to upsampling. I'll take an easy example.
> Assume you have a 44.1khz signal and want to to be 88.2kHz. Also assume
> 
> that the PCM data looks like
> 0, 2, 4, 6, 7 for the first samples.
> To upsample it, you first double the number of data points.
> 0, 0, 2, 2, 4, 4, 6, 6, 7, 7
> Then you can decide if you want to leave the data 'as is' with 
> duplication, or if you want to do some interpolation.
> You could, for example, do simple linear interpolation, giving
> 0, 1, 2, 3,  4, 5, 6, 7, 7, 7,
> Or you could do a higher order quadratic or other functional
> smoothing.
> 
> Then to reduce the sample rate, you just throw away data. So to go from
> 
> this 88.2 to 44.1, you just pick every other sample.
> this would give you, tada, 0, 2, 4, 6, 7
> 
> Now, if you want to widen the data, say from 16 bits to 24, the crude 
> approach is to just add 8 bits of zeros to each value. But adding zeros
> 
> doesn't address the sample rate.

I think that both those method would change the frequency content of
the signal.  By adding zeros between samples, the frequency content is
mirrored/repeated at higher frequencies and this can then be removed by
filtering leaving an upsampled signal with the same frequency content as
the original.

Taking your example:-
0, 2, 4, 6, 7 @44.1kHz
becomes
0, 0, 2, 0, 4, 0, 6, 0, 7, 0 @88.2kHz
this is then passed through a low-pass digital filter with a cut-off
frequency set to the bandwidth of the original signal.  The result will
be an upsampled signal with the same spectral content as the original.


-- 
Muggy
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