[gameprogrammer] Re: mixing sounds

Thanks David, I was hoping you would respond to this one hehe

I'm actually using double's to work with the sound data before converting it
back to ints to go on disk.  So, could i just add all the voices together
then scan the samples, looking for the largest absolute value and if that's
greater than 1.0 divide all samples by that largest number to get back into
the range -1 to +1?

Hey btw do you know any sites that would help in learning how to code things
like chorus, reverb, echo and other effects?

Thanks for your help (:

On 8/4/08, David Olofson <david@xxxxxxxxxxx> wrote:
>
> On Monday 04 August 2008, Alan Wolfe wrote:
> > Hey,
> >
> > I was wondering does anyone know how to properly mix sound data?
> >
> > I'm working with some raw wav data and i had heard that just adding
> > sound wavs together mixes them, but what happens in the case of
> > overflow? (is that clipping?)
>
> That's the correct way of doing it, but indeed, bad things happen if
> the result overflows. However, it's actually much, much worse than
> clipping. It's wrapping! :-) (Unless you take special measures,
> you're just throwing high bits away, so there won't be any "proper"
> clipping.)
>
>
> > It seems like when i average the wav data that it sounds like it
> > mixes ok so not sure what is the best way :P
>
> Averaging is pretty close to what you do, from a practical POV.
> However, the "right" way of doing it is to decide on a 0 dB reference
> level somewhere below that of the output, so you have some headroom
> to avoid clipping.
>
> Theoretically, you can scale first (you'll probably need per-voice
> volume control anyway, so you can just scale the volume levels and
> avoid scaling every sample twice) and then mix. If you deal in
> floating point, you can actually do that - but with integers, you
> need to use fixed point (as in, reserving some fraction bits for
> correct rounding), and/or use langer integers, so you can do the
> scaling *after* the mixing.
>
> Either way, as long as you only need a few (say 4) simultaneous
> voices, you can get away by just scaling things so clipping cannot
> occur - but if you try that with 16+ voices, you'll end up with a
> very low output level.
>
> To get reasonably loud output (from a typical consumer sound system,
> at least) and a sensible dynamic range (for your average user), you
> need to do some dynamic processing, one way or another.
>
> You can find a very simple approach in Kobo Deluxe, where I use 32 bit
> mixing buffers (actually 8:24 fixed point, IIRC), and end the chain
> with a simple compressor/limiter, before converting to the output
> format.
>
> For better results, you should probably divide the frequency range
> into three or more bands, and process each band separately, as a
> single band compressor tends to generate that typical pumping
> compressor sound if you push it too hard. (That can be a useful
> effect to make explosions and stuff more "in your face", but
> normally, you want compression to sound as transparent as possible.)
>
>
> //David Olofson - Programmer, Composer, Open Source Advocate
>
> .-------  http://olofson.net - Games, SDL examples  -------.
> |        http://zeespace.net - 2.5D rendering engine       |
> |       http://audiality.org - Music/audio engine          |
> |     http://eel.olofson.net - Real time scripting         |
> '--  http://www.reologica.se - Rheology instrumentation  --'
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