Hi 1) From the docs:
http://help.adobe.com/en_US/AS3LCR/Flash_10.0/flash/media/Sound.html#extract() The audio data is always exposed as 44100 Hz Stereo. The sample type is a 32-bit floating-point value, which can be converted to a Number using ByteArray.readFloat(). So, each sample is 32 bits, that is 4 bytes. If you read less than four bytes, you're discarding part of the data; if you read more, you're mixing samples. This could also happen if your reads are not 4 bytes aligned. 2) Not sure if I misunderstood what you meant, but the number of samples extracted is calculated here: var extract:Number = Math.floor ((sound.length/1000)*44100); And is not a power of 2. It seems like the total samples, rounded down. (seconds of audio data * samples per second). 3) I'm by no means an audio programming expert, but I think that should be possible. Don't how performant it would be doing it in AS, in real time, though. Neither I know algorithms for it, but look up audio frequency analisys, you might find some pointers. By the way, I don't your imaging is wrong. Digital audio data is a finite/discrete representation of a sound wave. Cheers Juan Pablo Califano 2009/6/24 ben gomez farrell <b...@yellow5labs.com> > 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 > _______________________________________________ > Flashcoders mailing list > Flashcoders@chattyfig.figleaf.com > http://chattyfig.figleaf.com/mailman/listinfo/flashcoders > _______________________________________________ Flashcoders mailing list Flashcoders@chattyfig.figleaf.com http://chattyfig.figleaf.com/mailman/listinfo/flashcoders
