[jcelerier]

> unless your code consists entirely of back-to-back operations with denormals and of nothing else.

Ending with the data being entirely in the denormal range is a common occurrence in some audio algorithms (and in there, intel CPUs dominate by such a large margin it's not even funny) ; if that happens at the beginning of your signal processing pipeline you're in for a rough time

[adrian_b]

I agree that this happens, but solving such cases with FTZ is a lazy solution, which is guaranteed to give bad results, due to the loss of precision.

Even when 32-bit floating point is used, for a greater dynamic range and for a 24-bit precision, instead of using 16-bit fixed-point numbers, proper DSP algorithm implementations still need to use some of the techniques that are necessary with fixed-point number algorithms, i.e. suitable scale factors must be inserted in various places.

A correct implementation must avoid almost all underflows and overflows, by appropriate scalings.

[jcelerier]

the problem is (speaking from the end user side), you can't guarantee that every plug-in you are going to use is going to be coded properly - and you don't want that 2007 plug-in whose author has been dead for a decade but is super important for your sound to bring your whole performance down when it gets silence-ish input