Hi Stuart

An elaboration on David's reply.

We want our audio BW to go to at least say 3.4 kHz . That is what fixed line telephones offer.

If we were to sample at 8kHz, our nyquist frequency is 4kHz of course.
Any spectral information above the nyquist rate will be aliased bay into the baseband, IE below the nyquist rate.

IF we assume we don't care about aliases between 3.4 and 4kHz, then this allows for signals up to 4.0 - 3.4 = 0.6,  and 0.6+Fyquist(4kHz) = 4.6kHz. (as a 4.5kHz signal will get aliased to 3.5 kHz.)

Assume we want our aliases 50dB down, so our aliasing filter must go from 0dB down at 3.4kHz to 50 dB down at 4.6kHz

Clearly alot of work in the analog department- probably needs an elliptic filter.
a transition bandwidth factor of (4.6/3.4) = 1.35

Now, if the sample rate was 16kHz, now, Fnyquist is 8 kHz. again, we dont care about aliases above 3.4kHz,
and Fnyquist, now 8kHz - 3.4kHz is 4.6kHz.

So, for freq up to Fnyquist + 4.6kHz - IE 12.6kHz  will not alias into problem areas.

so, now our analog filter spec if relaxed
0dB down at 3.4kHz, and 50dB down at 12.6kHz
a transition bandwidth factor of (12.6/3.4) = 3.7



QED




On 17/09/2015 7:53 PM, Stuart Longland wrote:
Hi all,

Just looking at the source code, it hit me.  When in analogue mode, we
run 16kHz sample rate, fair enough.

But in DV mode, we sample the ADC at 16kHz, downconvert to 8kHz, pass it
through the modem and codec, then upconvert back to 16kHz for the DAC.

We're dealing with voice frequencies, with SSB transmit bandwidths of
less than 3kHz.

Why not do the whole lot at 8kHz and save some CPU time?  Maybe some
rate-switching when in analogue or DV mode might be an option too, so we
run ADC/DAC at full-rate in analogue (for higher fidelity), then switch
to half-rate when we're doing DV.

Regards,


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