How can WAV files cause any clipping red in DAW meters if floating point can represent much higher values than 24 bit?


Let's say we get a wav file in the daw, that is clipping so much, most values are around 65,000 (assume this is the highest in 24 bit depth). In daw however, to reach 0 dbFS, the bit values need to go way way higher than that, because floating-point systems can reach up to much higher values to reach 0 dbFS ( (1.11111111111111111111111)2x2^127 ). Given these numbers, if they are correct, I would expect us to be unable to even hit any clipping using ANY wav file in the world. However that is not the case. Why does it show any red in the meters at all, when WAV files to me seem incapable of causing any clipping mathmetically ?

Does the DAW proportionally increase wave's bit values to its own capability ?

Ozum Safa

Posted 2020-06-28T22:20:42.887

Reputation: 101

2This all sounds a bit confused. no decent DAW will clip internally. For your theoretical clipped track, it must have been recorded clipped. The whole idea of floating point inside the DAW is to make the theoretical headroom almost infinite. Red-lining it means that something else will give [i.e. your o/p stage], but the DAW won't clip itself. – Tetsujin – 2020-06-29T10:31:41.873



As Tetsujin states in their comment, any decently developed/implemented DAW will use floating-point arithmetics to increase their internal calculations headroom.

This does not mean, in any case, that you can arbitrarily increase gains (even if they are digital) and do any kind of processing without introducing distortion or artefacts.

There must always be a reference as to what is 0 dBFS and the conversion from fixed-point to floating-point numbers is not trivial (have a look at the internal representation of those two - most systems use the IEEE floating-point, a simple explanation can be found here).

To provide an insight as to how many different ways exist to convert fixed-to-floating points, you could consider using the floating-points to add levels in between the fixed point levels, below the fixed point levels linearly or logarithmically spaced, above the fixed point levels or below the fixed point levels! And all those are mere suggestions of what one could do. I am not sure exactly how the conversion is performed but you should keep in mind that there must be a way to represent the 0 dBFS point whether you are using fixed- or floating-point arithmetics and if a WAV file is clipped it should be clipped in the DAW too since the value of the clipped samples should be represented as 0 dBFS in both fixed- and floating-point numbers.


Posted 2020-06-28T22:20:42.887

Reputation: 209


Building on Tetsujin and ZaellixA responses:

If your wav isn't itself clipped, its encoding (if samples are floating points or are integer numbers, it's bit depth...) may be missing or wrong in the file, leading to a wrong interpretation of the values of the samples by your DAW, and thus exceeding the scale.

You can try to load your file using different encodings in Audacity (it's free), via the menu File->Import->Raw Data... If you succeed, now you just export your brand new recovered file (with another name, for precaution...)

Tiago Brizolara

Posted 2020-06-28T22:20:42.887

Reputation: 11