Hy all!

I have got a theoretical question. Consider that I have sampled a function f(x) -> g(k) = f(k*dx) ; k=0,1,2,...,N ; dx = L/N
Than I fourier transform it, G = FFT(g). With the inverse DFT I can obtain the original series g(k). If I would like to use the IDFT for function interpollation(with some modification: k -> x*N/L )

N-1
---
\
f_int(x) = 1/N / G(n) exp( -i*2*pi*n*x/L) ,
---
n=0

than the values of f_int(x) only equal with f(x) at the original sample point, despite I 'am sampling f(x) = x^2 (for checking) with N=1024 and L = 1. Between the original sample points f_int(x) is badly oscillating and provides function results that are different from f(x) with sever ten percents. I have a feeling that there is something that I don't know.
I would like to interpolate a very badly oscillating function with nearly constant frequency, that is why I am trying to use some fourier method.

Can anybody help me?

Hey, Im facing the exact same problem. If you know the answer now, please do let me know. thanks.

...facing the exact same problem...

What problem? I can't understand what the Original Poster was trying to say in the first part of the post.

I wasn't hanging around this forum way back then (three and a half years ago), anyhow, and apparently no one else could help.

As for the last part:

I would like to interpolate a very badly oscillating function with nearly constant frequency,

That's called a bandpass filter.

I personally can't see trying to teach Digital Signal Processing fundamentals on a forum like this, which was created to help people with Numerical Recipes books and code, but...

If you have any specific questions maybe someone can help.

Regards,

Dave