3 Feb
2004
3 Feb
'04
7:02 a.m.
Greetings
I wondered if it is possible to add resonance to my filters in wcnt. They
only use an array
and calculate the mean average of the contents according to cut off freq.
At the moment
I've removed my p**s poor efforts at creating resonance and added a feedback
loop instead.
Is it possible to code resonant linear filters which use an array as I do,
or will I have to go on a higher mathamatics course and get my head round
fancy bi-carbonated equations?
Oh yeh, wcnt is open source... and uses a text file interface only and is
not-real-time
toodle do
James ~(sirromseventyfive)~
What's wrong with *ahem*RTsynth/didgeridoo*ahem* - I've not got the
processing power to try all this stuff out.
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3 Feb
3 Feb
8:31 a.m.
James W. Morris wrote:
Is it possible to code resonant linear filters which
use an array as I do,
or will I have to go on a higher mathamatics course and get my head round
fancy bi-carbonated equations?
resonant filters use recursion; the algorithm sometimes is coded as
'take this input sample and give me an output sample' for readability
but is easily adapted to arrays. the maths behind them are intricate
(to me at least), but you don't need too firm an understanding of the
theory to actually use them. you'll find code examples for the
algorithm and the computation of the recursion coefficients in almost
every major audio synthesis/processing package or at musicdsp.org .
steve's ladspa set has a very nice state-variable filter (svf) which
lends itself well for resonance and reasonably quick parameter
changes. you may want to try it first before resorting to coding your
own.
tim
ps: the svf sounds better to me with the 'very slight waveshape for
extra stability' removed.
8:56 a.m.
On Tue, Feb 03, 2004 at 02:21:53 +0100, Tim Goetze wrote:
ps: the svf sounds better to me with the 'very
slight waveshape for
extra stability' removed.
Yeah, but without it the filter tends to explode with highish cutoff
values (above fs/4 IIRC). I should probably move it to doubles and remove
the waveshape.
- Steve
9:18 a.m.
On Tue, Feb 03, 2004 at 01:52:13PM +0000, Steve Harris wrote:
On Tue, Feb 03, 2004 at 02:21:53 +0100, Tim Goetze
wrote:
Moving it to doubles will not help. The problem is that at high F, the
simple mapping from the continuous time domain to the sampled domain breaks
down. You need to modify the params as non-linear functions of F and Q
in order to correct for this.
(had the same problem with the M**G VCF plugins)
--
Fons
ps: the svf sounds better to me with the
'very slight waveshape for
extra stability' removed.
Yeah, but without it the filter tends to explode with highish cutoff
values (above fs/4 IIRC). I should probably move it to doubles and remove
the waveshape.
9:38 a.m.
On Tue, Feb 03, 2004 at 03:13:54 +0100, Alfons Adriaensen wrote:
On Tue, Feb 03, 2004 at 01:52:13PM +0000, Steve Harris
wrote:
OK, thanks. Yes, that rings a bell now - I think I made it a bit better by
oversampling the input (2 or 4x). I'l have a look at your plugin to see
what you've done.
- Steve
On Tue, Feb 03, 2004 at 02:21:53 +0100, Tim
Goetze wrote:
Moving it to doubles will not help. The problem is that at high F, the
simple mapping from the continuous time domain to the sampled domain breaks
down. You need to modify the params as non-linear functions of F and Q
in order to correct for this. ps: the svf sounds better to me with the
'very slight waveshape for
extra stability' removed.
Yeah, but without it the filter tends to explode with highish cutoff
values (above fs/4 IIRC). I should probably move it to doubles and remove
the waveshape.
7393
days inactive
7393
days old
linux-audio-dev@lists.linuxaudio.org
4 comments
4 participants
participants (4)
-
Alfons Adriaensen -
James W. Morris -
Steve Harris -
Tim Goetze