[ErroneousBosh]

Denormals in audio code are kind of the "perfect storm", because they take ages to deal with - you're suddenly back into softfloat land - and because you have to deal with many thousands of them in a few hundred microseconds.

We take how fast hardware floating point is for granted. I suspect it would be interesting to compare something compiled with softfloat with a normal benchmark and see just how bad it is.

It's a great reason to do your DSP code in fixed-point, which is just integer with a couple of steps you have to write down on paper to keep straight until you get to the end. Or, I do, because I suck at arithmetic. Just do it all in machine-length signed ints, and forget all the mystical world of tiny tiny floating point values ;-)

[lomase]

Fixed floating point has been a mistery to me, and to be fair floting point is too. I know digital synths like Virus or Waldorf all used 24 bit fixed point math DSP.

I remember this dps site with lost of c and delphi code, there is where I found what denormals are.

Nowdays I dont see DPS code dealing with denormals. Maybe the CPU does not have to do it in software anymore? I don't really know.

[ErroneousBosh]

> I remember this dps site with lost of c and delphi code, there is where I found what denormals are.

musicdsp.org?

> Fixed floating point has been a mistery to me, and to be fair floting point is too. I know digital synths like Virus or Waldorf all used 24 bit fixed point math DSP.

If you imagine scaling a 16-bit value for like a volume control from 0 to 1, then you'd have maybe 32767 for maximum positive, and -32768 for maximum negative. You could convert those to floats, multiply, and convert back to a 16-bit integer.

But you don't want to use floats, you want to keep it all integer. So you make the volume range be from 0 to 255, and multiply your 16-bit value by that. Now you've got a 24-bit value, with a "binary point" between bits 7 and 8. Now the output is way off scale for the 16-bit DAC but if we chop off the fractional part by just shifting the result of the multiply left 8 bits, you've now got your volume control.

Some DSPs will actually do a 16 bit by 16 bit multiply which just discards the lower 16 bits of the result, with the assumption being that both 16-bit values mean "-1 to 1".