[Pd] Complex audio signals

[Chuckk Hubbard]

On 6/21/06, Piotr Majdak <piotr at majdak.com> wrote:
> Chuckk Hubbard wrote:
> > On 6/20/06, Mathieu Bouchard <matju at artengine.ca> wrote:
> >
> >> On Mon, 19 Jun 2006, Chuckk Hubbard wrote:
>
> > Can it be heard?
>
> If you have any differences between the original and reconstructed
> signals, then they will be introduced by quantization (try a FFT with
> 8-bit fixed point DSP) or by overflow or by windowing effects - not by
> FT->IFT. This means: FT and IFT work as they are supposed to work -  all
> problems and differences in the perfect reconstruction of your signals
> are caused by inproper signal processing. And this means: if you have
> differences after FT->IFT then you will have differences after simple
> multiplications and/or additions too, because your system is not
> adequate to do this job.

For some reason I hadn't checked this before, and my system is indeed
adequate.  I wasn't sure if Pure Data's FFT has built-in windowing or
not, apparently it doesn't.

>
> > I'm specifically curious about seeing integration and convolution,
> > although I haven't found how to do that in Octave yet.
>
> If x is the sequence with your signal in MATLAB (Octave has the same
> syntax), then
>
> Integration is y=sum(x);
> Convolution is y=conv(x,f); where f is the sequence with the impulse
> response of the filter
> FT is X=fft(x);
> IFT is y=ifft(X);
>
> The syntax is quite easy - if you need some help about MATLAB, write me
> a personal mail - I'll do my best.

Cool, thanks, you might hear from me soon.  I wonder what's the
simplest way to do convolution natively in Pd?

-Chuckk

>
> br, Piotr
>


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