--- In Simmons_Drums@yahoogroups.com, Patrice Jacquot
<Jacquot.Patrice@...> wrote:
>
> Hi Roland ,
> any sampler , even best ones like Fairlights need to get a very good
> signal in !
Uhm, the Fairlight was a *terrible* sampler, long surpassed by the
sampling capabilities of musical greetings cards and Tickle Me Elmo.
> "Garbage in / garbage out" !
This is extremely true.
>
> I highly recommend to use a good preamp to do so ...
> a Universal Audio Solo 610 (valve version) , or even solo 110
> (transistor version) for example ...
> I just tryed it ! you won't recognize the sounds . it gives them
> more air & a so nice low end.
> It should be another quality level in the SDX ;- )
Aye right. And don't forget to use solid gold phono plugs, and
unidirectional grain-orientated oxygen-free copper interconnects.
> If you haven't access to that sort of preamp it's really worthwhile
> renting one for 1 or 2 days ...
Make sure you get pictures of yourself with it in the studio. It'll
lend a lot of geek cred to your website pics. Won't help the sound
any, but it will look cool...
More seriously, the important bit is filtering off high frequencies
that cannot be captured by the sampler and which will cause aliasing.
I'm sure we're all familiar with Nyquist's Theorem, which says that
you need to sample at twice the maximum signal frequency or more. In
practice, the highest frequencies present in the sampled signal should
be somewhat lower than half the sample rate.
For a good example, let's look at the venerable Ensoniq Mirage. Used
normally, it passed the sample input through one of its 24dB/octave
filters. What does that mean exactly? Well, the frequency response
tails off as we go beyond the cutoff frequency, so that by the time
we've gone to twice the frequency the output is 24dB quieter - about
1/250th. For a sample rate of 32kHz we'd have a Nyquist frequency of
16kHz. If we set the cutoff frequency to 8kHz then by the time we
reach 16kHz then a full-scale signal would be reduced to 1/250, which
for an 8-bit sampler like the Mirage puts it down into the noise. In
practice, it's only the upper harmonics of a signal that are likely to
be a problem, and since these are generally well down on the
fundamental we could increase the cutoff to around 12kHz before
noticing much aliasing.
Now, the Mirage had another trick up its sleeve in the form of the
ISF-1 Input Sampling Filter, which included a 40dB/octave lowpass
filter. Now, this slopes off at a much steeper rate, so that by the
time we've gone up an octave the filtered signal is 1/10000 - one
ten-thousandth - of the input! This can be tuned far closer to the
Nyquist frequency without letting appreciable amounts of harmonics go
"over the edge".
In short - if you want the best out of an elderly sampler, you're
better to sample from the output of a PC, and apply brickwall passband
filtering to the signal in some sort of audio editing app to remove
anything that's likely to alias. If you don't fancy doing that, get a
filter with a very sharp cutoff, and possibly a parametric EQ to
sweeten up the input signal.
The preamps and converters in the SDX aren't particularly linear or
"good". Get the cleanest signal you can, filter carefully, and watch
those levels.
Or, of course, leave the input filter wide open and let it alias
horribly. It gives a really cool dirty effect. Very industrial - you
should try it at least once, even if you don't end up liking it.
HTH
Gordon