hi all,
thank you for the highlight on the subject. subsidiary question you know
your hardware and your algorithm do you use any high precision libraries
in an realtime environment ?
best,
j
On 12/04/2023 05:25, brianw wrote:
On Apr 11, 2023, at 3:27 AM, STEFFAN DIEDRICHSEN wrote:
- Stefan Stenzel proposed to gain up the input by some hundred dBs and gain
down the output accordingly to push out the likelihood of an underflow, which
leads to ahen interesting compander scheme.
I'm philosophising here, but I don't think it's accurate to call this option a
"compander," because there's no compression of expansion.
The fact that floating point signal samples are normalized to the range of -1.0
to +1.0 is basically arbitrary. It seems like one of those decisions made for
human convenience, and not mathematical optimization.
Even with this standard, there's a conversion required before the DAC to
convert floating point to fixed point (I'm going to ignore floating point DAC
chips). In other words, there's already a gain in and out for the
ADC-to-sample-to-DAC flow, so changing the specifics is nothing like companding.
Given the natural range of IEEE floats, it might make sense to dispense with
the normal -1.0 to +1.0 assumption and use something larger. One might merely
need to consider the maximum number of channels that need to be mixed together,
and not bother with any more headroom than would be needed.
Of course, all the software would have to be rewritten if the -1.0 to +1.0
standard were abandoned. However, that's not too far fetched because we already
have DSP systems that are entirely proprietary at the sample level (the digital
audio I/O, ADC and DAC are operating with 24-bit fixed point, and thus remain
compatible no matter what the internal floating point normalization is used).
Thanks to Stefan Stenzel for bringing something to my attention that I hadn't
stopped to consider: The natural range of floating point, and how digital audio
signals might best fit within that range.
Brian Willoughby