In the worst case is a DA conversion at the incoming rate followed by an AD
at the computer rate.
In the best case the resampling is done in the digital domain, though I
don't know the mathematics involved when the clocks are only slightly
different...
Two clocks very rarely are in synch (a DAT output might be 48009
samples/sec while the board input might be 47995); resampling even if the
rates *should* be the same allows to avoid synchronizing and locking
(slaving) the input clock to the incoming clock.
The best way to verify if a board resamples or not its digital input is to
computer generate a .wav file with a signal at the Nyquist rate and full
scale amplitude. It is very simple to generate: a monophonic signal should
be -32768,+32767,-32768,+32767 and so on. A stereo should be
-32768,-32768,+32767,+32767,-32768,-32768,+32767,+32767...
Try also with amplitude reduced to 1/2.
Then you need a digital output board to send the SPDIF signal to the
board/device to be tested. If the signal you acquire is digitally resampled
you'll see immediately with no doubts......
Alternatively you could burn an audio CD and do the test with the digital
output of your CD player. Of course never send the sound to your
HiFi... I never tried the CD solution, though it should work.
Better explanations/hints are welcome,
Gianni
At 01.23 10/08/2002 +0000, you wrote:
>Hi,
>
> Is there a good writeup somewhere that describes what is meant
>when a device performes "needless resampling"? I've heard this said
>of the soundblaster and edirol products (and others) but must admit I
>don't understand it.
>
>Thanks,
>John
>
>
>
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--------------------------------------------------------------
Gianni Pavan
Email
Centro Interdisciplinare di Bioacustica e Ricerche Ambientali
Universita' degli Studi di Pavia
Via Taramelli 24, 27100 PAVIA, ITALIA
Tel/Fax +39-0382-525234 Laboratori
Tel/Fax +39-0382-526208 Segreteria
Web http://www.unipv.it/cibra
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