| > 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.)
| > 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
|
| I don't think that 3 would be good, since it would affect a rather
| large range of frequencies ( the lowest band is for 38 Hz, as someone
| recently mentioned, I think at least freq to 50 Hz would be affected )
Should you tell me WHY?
1. if you grab an audio CD, you dont need use HPF, they do not have DC
(only if mastering studio is fullfilled by idiots)
2. if you create your own music track, I am sure, that final process is DC
offset removal + fade-in/fade-out few milisec at begin and end of track...
3. if you use live encoding of audio from soundcard, you can use --highpass
0.015 --highpass-width 0 options for LAME (I use it)
4. I think that some kind of reason should be found forever
5. ...
6. ...
...
...
Best regards
Jaroslav Lukesh
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