[music-dsp] FIR filter question
padawan12 at obiwannabe.co.uk
Mon Dec 6 11:28:12 EST 2010
Without being quite sure how you intend to
use the filter I don't know if this advice is useful;
But for wavetable synthesis you don't really need
a filter, as such.
For preparation of tables it is usual to do it
by construction, to build the table additively
from a sum of partials. That way your period
perfectly fits the table, by definition.
For inbetweening tables the method is often nothing
more complicated than taking an average.
If you then choose your table mixtures carefully
as a function of note, highest table harmonic
and sampling rate, you can be sure never to get
any foldover in the first place.
On Mon, 6 Dec 2010 16:46:49 +0100 (CET)
"cparodi.ugemi at libero.it" <cparodi.ugemi at libero.it> wrote:
> Dear Members,
> I'm trying to figure out how can I use the FIR filter in the Freeverb3 open
> source library as an anti-aliasing filter in a wavetable synth.
> More in detail: I'd like to use it in the initialization phase to pre-
> calculate a lookup table of filtered waveforms starting from single-cycle
> waveforms. I would then interpolate-read from this table during playback using
> the most appropriate index (based on closest pitch of the note being played).
> Is this a completely crazy idea or something feasible? Did any of you ever
> used those FIR filters?
> Thanks in advance for any help on this.
> dupswapdrop -- the music-dsp mailing list and website:
> subscription info, FAQ, source code archive, list archive, book reviews, dsp links
Andy Farnell <padawan12 at obiwannabe.co.uk>
More information about the music-dsp