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On 2014-08-19 22:09, Alexandros Drymonitis wrote:
>
> Here's my perform routine: t_int *vanDerPol_perform(t_int *w) { //
> Copy the object pointer t_vanDerPol *x = (t_vanDerPol *) (w[1]);
>
> // Copy signal vector pointers t_float *frequency = (t_float *)
> (w[2]); t_float *factor = (t_float*) (w[3]); t_float *excitor_freq
> = (t_float*) (w[4]); t_float *excitor_amp = (t_float*) (w[5]);
> t_float *output = (t_float *) (w[6]);

the scalar type for signals is really *t_sample*.
you should NOT use *t_float* for these kind of things (though in
practice they two might always be the same).
if you are following a tutorial that suggests to use t_float, then
report a bug to the author of the tutorial.

>
> // Copy the signal vector size t_int n = w[7];
>
> // Dereference components from the object structure float twopi =
> x->x_twopi;

now that looks very much like a constant, which you don't need to
modify per-object...

> // excitor code si = *excitor_freq++ * si_factor; phase_local =
> phase / step; drive_sine = cos(phase_local * twopi) *
> *excitor_amp++;

i guess you are copying the CSound code here.
however,  i really think it would be much better to use an "external"
excitor, that is: instead of having an input for the frequency and
another input for the amplitude of the internal sine oscillator, you
might want to have a single input for an in-patch sine-oscillator.

so your patch would look like:
|
[osc~ 666]
|
[*~ 0.42]
|
[vanderpol~]
|

(it might make sense to make the excitor input the first one, so you
can use it

this has two huge advantages:
#1 Pd's oscillator is known to work; if it ever breaks, it will be
fixed immediately.
thus you don't need to re-code existing functionality, possibly
introducing bugs.

#2 you might want to use another exciter waveform, and get results
unheard of.
without any additional coding effort! for free!!

>
> // van der Pol code c = 2 - 2 * cos(*frequency++ * twopi / sr);

again, it might be better to use an external signal to drive this
oscillator.

> aa = (-c) * ax + *factor++ * (1 - ax * ax) * av; av += aa; ax = ax
> + av + drive_sine; *output++ = av;

hmm.
you are resetting "av" and "ax" to 0 for each DSP-block.
most likely this is not what you want, as the srtucture looks like a
feedback filter (IIR), so you need to store ax&av between DSP-blocks.

fgmsdar
IOhannes

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