I would like to know if there's a standard way to obtain the FIR coefficients for a sampling frequency $F'$ when the coefficients are for a sampling frequency $F$.
I would like to implement the filter described in ITU R BS.1770-3 (the oversampling FIR filter, page 18). The filter coefficients are given for a sampling frequency of 48 kHz.
What would be the way to derive the coefficients for another sampling frequency?
Edit december 28 2012
I wonder if there's a meaning in my request. The described filter is used on a 48 kHz sampling frequency audio signal that has been transformed into a 192 kHz sampling frequency signal by stuffing 3 zeros between each original sample. It is the second step of an oversampling method. The signal $[x_0, x_1, ..., x_n]$ is first transformed into $[x_0, 0,0,0, x_1, 0,0,0, ...,x_n,0,0,0]$. The filter I have mentioned is applied on this zero-stuffed signal.
If I understand correctly (which might not be the case), the cut-off frequency of this low pass filter can be expressed relative to the sampling frequency. Given this context, and the desired result which is a sort of evaluation of the reconstruction of the audio signal into the analog domain, wouldn't this relative to sampling frequency cut-off frequency be appropriate for any sampling frequency ?
Let's say the cut-off frequency of the filter is $\alpha \times F_s$ ($\alpha$ should be around 0.125 for a 22 kHz analog upper bandwith ?), wouln't this $\alpha$ value be the good one for others sampling frequencies ? If I have an original signal at 8 kHz sampling frequency, a four time oversample would lead to a 32 kHz sampling frequency, and I can suppose that the reconstructed analog signal would have an upper bandwith limit at approximately 4 kHz ?
End of Edit december 28 2012