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Darren New <dne### [at] san rrcom> wrote:
> Remember when "don't bank on it" actually meant something? :-)
>
LOL
Stephen
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Warp wrote:
> Sorry for quoting the entire long post, but I find it rather amusing
> that this amount of text doesn't really tell exactly *what* you are doing.
I want to split a sound signal into several frequency bands.
Actually, technically I *have* split a sound signal into several
frequency bands... but not in a very useful way. In particular, the
lowest frequency band still contains large amounts of high frequencies.
It's really frustrating to have worked this long on something and not
got very far. I should probably go read a book or something...
--
http://blog.orphi.me.uk/
http://www.zazzle.com/MathematicalOrchid*
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Stephen wrote:
> Why are we here?
We're bored?
> What is the meaning of 42?
Would it be wrong to reply "4 * 10^2 + 2 * 10^2"? ;-)
--
http://blog.orphi.me.uk/
http://www.zazzle.com/MathematicalOrchid*
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>> What is the meaning of 42?
>
> Would it be wrong to reply "4 * 10^2 + 2 * 10^2"? ;-)
Yes... yes it would... PWN3D! >_<
--
http://blog.orphi.me.uk/
http://www.zazzle.com/MathematicalOrchid*
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Orchid XP v8 <voi### [at] devnull> wrote:
> Warp wrote:
> > Sorry for quoting the entire long post, but I find it rather amusing
> > that this amount of text doesn't really tell exactly *what* you are doing.
> I want to split a sound signal into several frequency bands.
Why?
(Btw, did you know that even if a sound signal has only one single
frequency, a discrete fourier transform is usually completely unable
to find that single frequency, and instead will find a large (potentially
infinite) amount of frequencies around that real frequency?)
--
- Warp
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>> I want to split a sound signal into several frequency bands.
>
> Why?
>
> (Btw, did you know that even if a sound signal has only one single
> frequency, a discrete fourier transform is usually completely unable
> to find that single frequency, and instead will find a large (potentially
> infinite) amount of frequencies around that real frequency?)
I think you just answered the question, right there. :-)
No, in seriousness... I'm playing around with designing an audio codec.
Most of the ones I've seen are based on chopping the signal into bits,
taking some kind of Fourier-related transform, selectively removing some
of the data, and then trying to hide the discontinuities between
segments. My plan is to seperate a signal into several continuous
signals, and process those.
--
http://blog.orphi.me.uk/
http://www.zazzle.com/MathematicalOrchid*
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On 04-Nov-08 20:14, Orchid XP v8 wrote:
> I should probably go read a book or something...
Why?
Remember Westheimer's Discovery: A couple of months in the laboratory
can frequently save a couple of hours in the library
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Would help if I actually wrote something before posting... ;-)
> I want to split a sound signal
Don't worry about the phase of the signals then, your ears don't care. In
hardware filters the phase is rarely preserved, and actually often varies as
a function of frequency. Just make sure the phase shift is the same for the
Left and Right channels, otherwise you might mess up some stereo effect.
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> (Btw, did you know that even if a sound signal has only one single
> frequency, a discrete fourier transform is usually completely unable
> to find that single frequency, and instead will find a large (potentially
> infinite) amount of frequencies around that real frequency?)
It just depends how many samples you use for your fourier transform, the
more samples you take, the more accurately you can divide up the frequency
spectrum. If you have a couple of seconds of normal audio recording (so
like 100k samples), you should be able to find a single frequency pretty
accurately, certainly to within 1 Hz.
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scott wrote:
> Would help if I actually wrote something before posting... ;-)
>
>> I want to split a sound signal
>
> Don't worry about the phase of the signals then, your ears don't care.
> In hardware filters the phase is rarely preserved, and actually often
> varies as a function of frequency. Just make sure the phase shift is
> the same for the Left and Right channels, otherwise you might mess up
> some stereo effect.
More significantly, if the same frequency is present in more than one
channel, phase will matter. (I still need to add the channels back
together again and end up with the same signal as before, which won't
happen if phase differences cause spurious boosting/cutting.)
Besides, in a digital FIR filter, it's trivial to make the filter
zero-phase.
--
http://blog.orphi.me.uk/
http://www.zazzle.com/MathematicalOrchid*
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