Shawn Riley wrote:
> I have 3 ideas, but I'm not sure if they'd even work, let alone how to code them if
> they did...
> 1- Try simulating the behaviour of ideal capacitors/resistors/op-amps/etc. Maybe a
> highpass filter could be "built" out of computer code & put just before the
>resampling
> function. (Or is *that* what the "convolution approach" is? Oops.)
This may result in phase shift in lowest frequencies, and, as most
CD recordings have a very limited dynamics range (they are continuously
at the max amplitude), the signal may easily run out of the
range of 16 bit short ints, resulting in audible distortion.
> 2- Resample the input file to something ridiculously low like 40Hz, & subtract this
> from the original WAV. (Although that would probably require the above method
>anyway.)
> 3- Remove the lowest frequency band(s) from the FFT data
Is the FFT size large enough for this (eg. a 1024 sample FFT has a
resolution
of about 40 Hz, and is not suitable for implementing a 5 Hz highpass
filter) ?
Generally, I do not think this highpass filter would not be
really useful, so it may be better to include it as an experimental
option.
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