Typically, the sound card does the A/D conversion at a fixed sample rate (say
48kHz), and then the audio is re-sampled to the desired sample rate either by
the sound card or by the OS. At least that is my understanding. So in effect,
the anti-aliasing filters are digital.
BTW, with modern oversampling ADCs, the anti-aliasing filters need not be very
steep. The audio is sampled at a rate considerably higher than the eventual
sample rate, so your anti-aliasing filters only have to cut-off frequencies
higher than the oversampled Nyquist rate (100's of kHz often), not the "actual"
Nyquist rate. Again, digital filters are used.
--
Jon Harris
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"Shytot" <Shytot@yme.com> wrote in message
news:orwOe.4893$iM2.495840@news.xtra.co.nz...
> How does a sound card set its anti-aliasing filters? After all, you can
> program a sound card to read at say 44.1kHz or 22,050Hz or half of that
> again so how do the ani-aliasing filters change? Switched cap filters are
> sampled filters so they would not be good and digital filters are no good
> either as we need analogue filters jus before sampling.
>
> Shytot
>
>
Reply by Shytot●August 23, 20052005-08-23
How does a sound card set its anti-aliasing filters? After all, you can
program a sound card to read at say 44.1kHz or 22,050Hz or half of that
again so how do the ani-aliasing filters change? Switched cap filters are
sampled filters so they would not be good and digital filters are no good
either as we need analogue filters jus before sampling.
Shytot