> > Hello
> >
> > I would vote for a default 14Hz high pass filter, removable with the -k
> > option.
> >
> > Why an high pass filter:
> > *theorical minimum audible freq for humans is 20Hz (also very discutable)
> > *most soundcards are unable to reproduce less than 20Hz frequencies
> > *most speakers are unable to reproduce less than 19Hz freq
> > *most (if not all) headphones are unable to reproduce such low freq (even if
> > their specs tell they are)
> > *ISO doc recommends an high pass filter
>
> At 5 Hz, IIRC.
>
> > *even if it's a small part of the spectrum, it's still a few bits usable for
> > other more critical parts.
>
> _very_ few. Every mdct sample is about 38 Hz wide, so you wouldn't even cut
> away the first (or zeroeth, whatever) sample.
>
> The reason ISO recommends highpass filtering is to cut away the DC portions,
> which mp3 can't directly represents (it is a combination of the 0'th and
> 575'th sample. The 575'th normally gets set to zero, especially if you're
> lowpass filtering).
>
> You get DC in a long block, if the sound contains some components < 19 Hz.
>
> DC leads to high-frequency echoes. (Some kind of phantom image of the true
> sound at lower freqs).
>
>
> Dag dag,
>
> Segher
>
Does anyone know how a high pass filter is usually implemented? The
convolution approach (like the low pass filter) seems like it would be
expensive and require a lot of extra internal buffering: a 10Hz
signal takes 4410 samples to represent one period. To get good
frequency resolution (so you can tell the difference between the 10Hz
signal and a 20Hz signal, for example) I would guess your window size
would have to be at least twice that, or 8820 samples?
Mark
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