24 Oct
2002
24 Oct
'02
11:31 a.m.
my feeling is
that oversampling alone will not do, although
you're probably right in that it will get rid of most of
the aliasing.
Its how most people tackle to problem. I think I can get the valve cheap
enough that it will be practical. i'm thinking about how to apply the technique
to produce
square clipping, or better a [0 .. 1] range of clipping,
but the integration approach is awkward.
I doubt that the electronics ever produces hard square clips. 
24 Oct
24 Oct
12:05 p.m.
New subject: [linux-audio-dev] [ann] unmatched - a LADSPA amp tone
nikodimka wrote:
You can calculate your tranformation for the input
signal S(x_i) once
And the same transformation for S(x_i)+1 again.
I mean two similar input signals differing in one bit of value.
So when you have two output saignals you can calculate
the error due to aliasing: abs( F(S(x_i)) - F(S(x_i)+1) )
(Or is it just a trivial thing and it doesnt work and
this is not what you discuss?)
it seems to me you're referring to what is commonly called
quantization error. quantization is rarely an issue with
32-bit float DSP, less so with doubles.
tim
12:25 p.m.
New subject: [linux-audio-dev] [ann] unmatched - a LADSPA amp tone
On Thu, Oct 24, 2002 at 06:27:59 -0700, nikodimka wrote:
You can calculate your tranformation for the input
signal S(x_i) once
And the same transformation for S(x_i)+1 again.
Won't that just give you the gradient at point x_i, ie. d/dt(S)?
We are talking about frequency domain aliasing here, which is when you
generate partials that would be above the nyquist frequencyi, so they get
reflected down into low frequencies. It is not directly related to the
differential of the signal, though a high differential is often indicative
of an aliasing problem.
Typically you prevent audio aliasing by generating the waveform in a way
so that it contains no partials above nyquist, or by generating it at a
sufficiently high sample rate that there are none, then decimating down.
- Steve
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Tim Goetze