On 8/10/15 10:02 AM, Peter S wrote:
On 10/08/2015, robert bristow-johnson<r...@audioimagination.com> wrote:
the thing that i *think* Peter is missing in this is the same as some of
the early manufacturers when they truncated the 30-bit words (or
whatever they had in the decimation filters) to 18 "meaningful" bits.
that simply adds one more source of quantization noise. and, especially
when our DSPs were 24 bits or more, we could make use of those bits.
i'd much rather have a 24-bit word, even if the dynamic range divided by
6.02 is only 18 or 20.
Are you *sure* truncating adds extra quantization noise?
---->[sigma-delta modulator]---->[decimation filter]---->[quantizer]--->
adds extra quantization noise to:
---->[sigma-delta modulator]---->[decimation filter]---->
When the
lowest bits are already noise,
they're not completely noise. they are signal buried in noise.
try this: take a flute solo or some other monophonic lick and fade it
down by a rate of, i dunno, 3 dB per second. eventually it will hit
your quantization floor. do that and tell us you can't hear a
difference. there will be a point where the quantized output will be
buzzing because your flute solo has become a square wave and then
silence, but in the unquantized version, you can still hear it (but it's
buried in noise of some spectral shape).
a way to *really* hear it is to compute the net error signal (output
minus input) and listen to that. and boost the error signal by 40 dB
just so that you can scrutinize the difference. i have some old .wav
files and MATLAB code for you (and a powerpoint from 2008 that spells it
out).
there is no reason to bother zeroing those bottom 4 or 6 or 8 bits (if
your normal word width is 24 bits). give the DSP whatever word width is
natural for the DSP. those bouncing bottom bits are actually useful for
most audio DSP processing. sometimes we have to *add* dither to the
input so that non-linear breakpoints get to be a little "rounded". i
would much rather have the *less* noisy input that also has the least
significant bits bouncing around keeping them discontinuities (like
internal clipping or quantization) in my algorithm in check.
another common experiment: reduce your amplitude by a factor of
32767/32768, quantize to the word size, then bump it back up with the
reciprocal factor. and subtract the original and listen to the error
signal boosted by 40 dB.
I highly doubt that truncation would
add harmonic distortion or limit cycle oscillations. I tested the
difference, and if the noise floor is high enough, in blind ABX tests,
I personally hear no difference between the non-truncated and the
truncated signal (whatever bit depth).
Have you measured the difference? Do you actually hear the difference?
Can your ear distinguish between the two in blind listening tests?
Mine doesn't, and my analysis shows no difference in harmonic
distortion, so I believe that those lowest, discarded bits of noise
practically add nothing to the signal, and they can be discarded
without any loss of quality or any introduced artifacts.
That's my belief, and I base this on analysis and blind ABX listening
tests between the two.
i have a better idea (it's also what we should force the Monster Cable
folks to do, or maybe the DSD folks):
AB testing. the subject is asked to determine if A and B are the same
or different. have as many trials where A and B are identical as trials
where they are different. remove any bias from the subjects by
subtracting the number of false positives from the number of times the
subject correctly identifies a positive. same for false negatives.
then see if the subjects do better than random guessing.
there is *still* a part of your signal in those bottom 6 bits. why
ditch them?
Because adding further bits of noise add no perceptional difference.
you didn't say anything about the audio level. if you claim this for
*any* level of audio, including audio that is going down into the noise
floor, you're decidedly mistaken.
--
r b-j r...@audioimagination.com
"Imagination is more important than knowledge."
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