On Mon, 17 Feb 2003, Jaroslaw Sobierski wrote:
> > > b) sum overflow: we can lower volume of samples before sum; I think that
> > > hardware works in this way, too
> >
> > Here I don't understand you. Suppose we have 3 samples to mix:
> > a = 0x7500
> > b = 0x7400
> > c = 0x8300
> >
> > If you do a + b + c (in this order) you obtain:
> > d=0
> > d += a -> 7500
> > d += b -> 0xe900 -> 0x7fff
> > d += c -> 0x02ff
> >
> > while the correct result is 0x6c00. You see?
>
> AFAIK most hardware does not mix by reducing volume before the sum. On the
> contrary, it is usually summed "as is" to a wider register, and often even so
> used. For example, a sound card able to mix 16 chanels of 16 bits would have
> a 16+4 bits or 24 bit register were the channels are added and no saturation
> can occur. In good hardware this would not even be downscaled back to 16 bits,
> but a 24 bit D/A converter would be used instead. In older times (Gravis Ultra
> Sound and I think older SB AWE) this could easily be spotted by the difference
> in supported "hardware" channels and "software" channels. A card with a 32 bit
> sum register and 24 bit DA could support (as above) 16 hardware channels and
> for example 64 software channels (mixed together in quadrouplets to the 16 hw).
>
> In our case, such "solution" would have to affect the whole buffer, meaning
> we would need 3 (or better yet 4) bytes per sample, which would eventually get
> reduced back to 2 bytes on the way out to the sound card. This seems neither
> elegant nor memory efficient but would work, and also solves the "a)" problem
> because we don't need to saturate so an atomic add can be performed on each
> sample.
Yes, this solution is good. I've though about it, too. Unfortunately, it
adds additional transfers including saturation from the "sum" ring buffer
to the DMA buffer of hardware.
Jaroslav
-----
Jaroslav Kysela <[EMAIL PROTECTED]>
Linux Kernel Sound Maintainer
ALSA Project, SuSE Labs
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