# Sampling Rate Conversion

Started by October 31, 2013
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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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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