[linux-audio-dev] mixing algorithms...

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Subject: [linux-audio-dev] mixing algorithms...
From: David E. Storey (dave_AT_tamos.net)
Date: Sun Sep 21 2003 - 22:10:46 EEST


Hey there...

I have an incredibly naive question... How does one mix n channels of
audio down to one channel? I've scoured the 'net as best I could and
haven't really found anything very authoritative. Suppose I'm dealing
with float point data between -1.0 and 1.0 and have n channels. I know
physically, it's all a summation of the individual waves, but strictly,
summation of multiple waves between -1.0 and 1.0 doesn't keep you
between -1.0 and 1.0. Looking through sox, they tend to multiply by some
gain value relative to the number of channels. (namely... an average) So
is averaging considered a valid professional audio method of digital
mixing? what if your source samples are all 16bit? Wouldn't you need to
dither in a case like that? Doesn't averaging also imply truncation of
your source signal? Am I missing something here? I've tried looking to
see what jack and ardour do, but was lefting wondering WHERE to look.

respectfully,
d!



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