Correct, nothing played back at original sampling rate will alias. If
you speed that sample up, then some of the recorded harmonics will go
over the Nyquist number and alias.
Please read the page I sent you on aliasing:
http://en.flossmanuals.net/PureData/Antialiasing
and also
http://en.flossmanuals.net/PureData/WhatIsDigitalAudio
Aliasing happens when you try to synthesize or play back a frequency
higher than 1/2 the sampling rate (this is called the Nyquist number).
In non-sine wave oscillators, it often comes from the highest harmonics.
In samples, it comes from playing them back faster than the original
sampling rate. It happens at the moment those frequencies are
synthesized, and cannot be removed later. Thus the oversampling approach
documented in the FLOSS Manual (and taken directly from Miller's Pd
manual patches).
A Karplus-Strong resonator is a delay line and as I understand it, so
long as no pitch shifting is going on then it can't alias. You would not
be able to create a Karplus-Strong resonator at 30KHz unless you have a
sampling rate of 60KHz, because the shortest delay time you can get is
still one sample (1/44100 of one second at normal sampling rate). Again,
math gurus are welcome to correct my calculations.
D.
On 3/31/10 6:39 PM, Pierre Massat wrote:
I m not sure i understand aliasing well... So anything that's sampled
and played back without altering the pitch would not suffer from
aliasing? When exactly does aliasing occur? during the DAC conversion,
or before that? Let's say i set a karplus-strong resonator to a
frequency of 30 KHz (assuming i'm a dog and i can hear a pitch that
high), at a 44.1 KHz sampling rate, than what happens? No aliasing at all?
2010/3/31 Derek Holzer <[email protected] <mailto:[email protected]>>
I was thinking about this the other day.... is it possible to have
aliasing with Karplus-Strong? Because it's a delay line, nothing is
being played back at any higher rate than it was sampled at, so no
aliasing should be possible. Right? Math-gurus correct me if I'm wrong.
Otherwise, any signal generator needs to be bandlimited or oversampled:
http://en.flossmanuals.net/PureData/Antialiasing
http://en.flossmanuals.net/PureData/GeneratingWaveforms
Frank Barknecht has some spliced-transition trick he uses as well,
I'm sure it will come up in a reply or two on this thread as well...
D.
On 3/31/10 6:27 PM, Pierre Massat wrote:
Hi!
I ve been reading the on-going debate about interpolation for a few
days, and it just occured to me that i don't how go about avoiding
aliasing more generally than with band-limited wavetables. If i
wanted
to play a sample at a pitch higher than the original, or if i
wanted to
use a karplus-strong resonator to generate notes, what would be the
proper way of ensuring that no aliasing occurs? Do people
generally use
low-pass filters with a cut-off somewhere below the Nyquist
frequency?
Or is there a trick that one can use earlier on in the signal
path of a
patch?
--
::: derek holzer ::: http://macumbista.net :::
---Oblique Strategy # 139:
"Revaluation (a warm feeling)"
--
::: derek holzer ::: http://macumbista.net :::
---Oblique Strategy # 151:
"Take away the important parts"
_______________________________________________
[email protected] mailing list
UNSUBSCRIBE and account-management ->
http://lists.puredata.info/listinfo/pd-list
