|
![](/i/fill.gif) |
> (You might think, for example, that you could just snip your 5-minute song
> into, say, 50 ms chunks and take the Fourier transform of each chunk.
> Alas, snipping it up introduces phantom frequencies that aren't really
> there.)
Use something other than a rectangular window function then, that helps with
the phantom frequencies.
> I've spent a significant amount of time trying to come up with some
> mathematics for analysing sound the way that the human auditry system
> does... So far, nothing works.
If you don't need it in realtime then just run a band-pass filter over the
whole song at varying frequencies, you can then read off the amplitude at
any frequency at any time during the song.
Post a reply to this message
|
![](/i/fill.gif) |