Hi,
A more efficient way to do this is by using a continuous time interpolator (ie.:
Lagrange 4th order interpolator). It gives you a polynomial function rather than
a sampled signal as any digital filter or regular interpolation turns out. So,
you can use this function to evaluate any arbitrary point in time, allowing
resampling to a any arbitrary sample rate.
Best,
- Felipe
Hi experts,
>
> I want to convert audio sampled at 192KHz to 44.1kHz. It
involves interpolation by 147 and decimation by 640. I'm designing filter
coefficients using fda tool, with FIR equiripple method. The parameters I gave
are
>
>fs - 192000*147
>fpass - 16000
>fstop - 22050
>
>Apass = 0.4
>Astop = -80dB
>
>But the fda tool gets hanged after some time. Is there any alternative to
generate coefficients from fda tool by changing parameters or design method?
>
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