First of all, my humble apologies to Marcuspocus. I completely misread your posts. I agree with everything you wrote. I inadvertently started a debate thinking that you were talking about "24 bit floats" and wasted bits on the ASIO "bus"...
A few links for anyone who's interested in the concept of "floating points":
http://www.wordiq.com/definition/Floating_point
http://research.microsoft.com/~hollasch ... float.html
Incidentally in searching for these definitions I came across a link to the Sun .au audio file format:
http://astronomy.swin.edu.au/~pbourke/dataformats/au/
Interesting that it provides 32 and 64 bit floating point representations. (According to the link on the same site, WAV files do not.)
Music Manic: Let me stress that, as far as my ears are concerned, there is no difference in audio quality switching from 24 bit integer to 32 bit float etc. For that matter my ears are not "tuned" enough to hear differences between 44.1 kHz sample rate and 96 kHz. Some people can apparently hear the difference in sampling rates, but not me.
However computer geeks have spent the last 40 years tweaking and improving math algorithms for computers. The only thing computers were ever meant to do, in the early days, was calculate solutions to math problems quickly. These days floating point math functions are
darned fast.
Audio is very much about math problems. Basic operations like adding, multiplying and subtracting are efficient and precise even using integers. Division is a bit more challenging.
Logarithms, sines and cosines, square roots, and so on, are all floating point operations. And they are all very much a part of audio processing. For example, if you want to calculate the dbFS amplitude ("volume") of an audio signal, the equation is:
Code: Select all
db = 20 log ( <signal level> / <full scale level> )
Therefore if, for argument's sake only, your signal amplitude is 1,000,000,000 and your full scale is 2,147,483,648, then the dbFS amplitude of your signal is -6.638597312 db.
Now of course I can't speak for CWA or sequencer software makers, who may use shortcuts to calculate the levels shown in peak meters. (*) But this calculation is certainly one way to do it; and it requires floating point math!
But again: nothing to do with "sound quality" exactly. It all has much more to do with processing efficiency. There is a performance hit every time you cast an integer to a floating point for math function calculations, and another one when you cast the floating point back to an integer. You lose precision, too, but that's not something I would worry too much about.
As far as more info about sound processing: Petal and BingoTheClowno recently brought up a couple of interesting audio processing "toolkits" in the
Announcements forum. Having only cracked the manuals (I haven't started using 'em yet, mind you) I'd recommend 'em to anyone who's interested in exploring the wonderful and not-quite-as-arcane-as-it-sounds

world of audio processing.
I suspect I've thoroughly bored everyone to tears by now. My apologies for being a floating point firebrand! I'll end with a couple of words (32 bit words, of course -- groan!) on VDAT:
1) I recently switched to 24 bit recordings and am now getting nasty jitter as long as my Nuendo window is open and not minimized. When I record to VDAT I don't get any jitter at all. I could buy a $500 USD word clock... Or I could use VDAT!
2) For those who are used to working in VST: don't you find working in SFP easier? I'm quite pleased with the single-window recording, myself. In particular the routing is infinitely more flexible than anything I've seen in the VST world. Thoughts?
3) Buy VDAT. It's good!
OK I'll shut up now,
Johann
(*) An old trick for certain types of math problems is to store a giant lookup table in memory, and retrieve math function values from memory. This works especially well for sine, cosine, etc. -- any function that repeats itself over the domain.
<font size=-1>[ This Message was edited by: blazesboylan on 2004-07-22 02:10 ]</font>