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Re: Two Questions on Re(or Up)sampling

Posted by Jim F. (A) on December 28, 2006 at 09:09:00

In Reply to: Two Questions on Re(or Up)sampling posted by Xenithon on December 28, 2006 at 07:21:14:

> #1. if the original files are 44.1kHz, and the hardware
> you use (DAC, USB interface etc.) only has 48kHz or 96kHz
> upsampling, is it better to upsample to one of those or
> keep the original 44.1kHz.

There's no sense in going from 44.1 to 48 just for sound
(unless, of course, **you** think it makes an improvement).

44.1 to 96 is considered (by those who "believe" in upsampling
at all) to be a "legitimate" path.

> The reason I ask is that I read a number of times that it's
> best to upsample in whole multiples, e.g. 88.2kHz etc.

There's a theoretical basis for this, but whether it translates
to a difference in sound is up to you. I don't take it very
seriously. Back in '99, when upsampling became a "hot"
issue thanks to dCS (using their 974 digital-to-digital converter),
and Bascom King and Jonathan Scull (in _Stereophile_) and
others raved about it, the upsampling conversions initially
reported on were 44.1 -> 96 or 44.1 -> 192 (non-integral
multiples).

> (still output 24-bit, so changing bit-depth but not sample-rate)

? 44.1 -> 88.2 is certainly a change in sample rate.

> if keeping 44.1kHz, in a program such as foobar, would it
> be better to leave out any sampling plugins in the DSP loop

Yes.

> or is there still an advantage to using something like PPHS
> to resample 44.1kHz ultra-linear mode; or the secret-rabbit
> resampler at 44.1kHz etc.

I can't imagine why there would be an advantage, again unless
it happens to sound better to you.

If you're a DAC manufacturer (like, say, Bel Canto with their
DAC 1 and DAC 2, or PS Audio with their Link III) there's
a possible reason to use an ASRC (asynchronous sample-rate
converter) chip (a Crystal CS8420 in the case of the old
Bel Cantos; a TI/Burr-Brown SRC4192 in the case of the
new Link III) -- namely, to reclock the data going into
the DAC chip and eliminate jitter. In those cases, the
upsampling (to 96 in the case of the DAC 1; to 192 in the
case of the DAC 2 or Link III) isn't defeatable, so
if you happen to be using a 96KHz source (in the case
of the DAC 1) or a 192 kHz source (in the case of the
Link III) you'll be going through the ASRC with a 1:1 clock
ratio. But this is for jitter rejection, which you won't
get by using an upsampling DSP in Foobar.

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