Hey guys,
I'm just digging into the Sound.extract feature in FP10. I've been
using code found at http://www.bytearray.org/?p=329 to learn from, but
have a few questions. I'm able to create a waveform and draw to the
stage, but want to take it beyond that.
Basically the code I have is this:
var samples:ByteArray = new ByteArray();
var extract:Number = Math.floor
((_sound.length/1000)*44100);
_sound.extract(samples, extract);
var step:int = samples.length/4096;
do step-- while ( step % 4 );
samples.position = 0;
for (var c:int = 0; c < 4096; c++) {
left = samples.readFloat();
right = samples.readFloat();
samples.position = c*step;
}
My questions are:
1. I understand incrementing the byteArray read position by a step
amount. What I don't understand is why the step amount needs to be a
multiple of four? I get some very wacky results from my readFloat() if
I don't do this.
2. Anybody know why the number of samples we're taking is a power
of two? Is this important, maybe just for graphic performance?
3. Here's the big one - is it possible to isolate amplitude at
certain frequencies - to get something like you'd get from
computeSpectrum? I guess I'm overly confused by the data I'm getting
back from the byte array and how to dig deep with it. The way I imagine
this byteArray in my head is that each position in the byteArray, one by
one, would be a composite amplitude of all frequencies at a small point
in time. I think I must be imagining this data wrong - and you'll
probably cringe at my composite amplitude remark cause it'll probably
make no sense.
Thanks - and again, I really think I have some huge knowledge gap in how
sound data is used and read. Can anyone help?
ben
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