Dear audiodsp group,
I have designed a function for audio resampling between arbitrary rates
using the common "windowed sinc" interpolation with FIR filter. The
function works well, but please could you answer a few questions about the
resampling quality for me?
* Do I understand correctly that the number of filter "taps" is equal to
the FIR window size in terms of input samples? i.e. increasing the FIR
window size increases the number of filter "taps" and the resampling
quality.
* I would like the resampling to be of a similar quality, regardless of
the input / output sampling rates which may vary over a wide range. To
achieve this, should I define the FIR window size in terms of seconds, so
that the number of filter "taps" is dependent on the input sampling
frequency (filter taps = window seconds * input frequency)? Or is there a
better way...
Thanking you in advance for your time,
dspdodge
