Digitakt manual says these two things:
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A sample contains a 16 bit, 48 kHz, mono audio file. (p.23)
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D/A and A/D converters are 24-bit, 48 kHz (p.64)
It looks like audio is sampled at 24-bit and then has bit reduction applied before saving.
Huh?
If the audio is going to be turned into a 16-bit file, why are the A/D converters 24-bit?
Does this give some benefit?