[PD] Combat aliasing!

[Derek Holzer]

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 <derek at umatic.nl <mailto:derek at umatic.nl>>
>
>     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"