In order to generate 32,100 Hz instead of 12,000 Hz you also need to limit the bandwidth, therefore you need a filter. The only difference is that in that case you need a band-pass filter instead of low-pass one.
I won't argue that I interpreted it correctly, but I took that phrase from: JAES vol 52 no 3 / "Pulse-Code Modulation - An Overview" - Lipshitz, Vanderkooy
https://pdfcoffee.com/jaes-v52-3-all-pdf-pdf-free.html
Well, I should have put "not yet quantized" in quotes. It was a digital file, so of course it was quantized. But I used high enough bit-depth to pretend (or simulate) that it was not quantized.
Here's how the files can be generated (files in attachment). For our pretend-not-quantized signal...
You can rearrange that equation as: the quantized signal (yellow) is the sum of the original (green) and the error (red). So what the DAC outputs (yellow) is a mix of the original and the error.
If not "noise", then what do (or did) you think this "just modifying" should do to the signal?
If...
I also like the term "error". And then "distortion" if the error is correlated to the signal and "noise" if the error is not correlated to the signal.
Then you can say: "Quantization produces an error. This error could be a distortion. You can use dither to turn distortion into noise." and...
And how will you decide that the definite answer was found?
Consider two scenarios:
Scenario 1: the definite answer is "yes".
In that scenario you keep looking and eventually, after a year or 100 years or 1000 years, you find the proof that there's is an audible difference. Cool, you've found...
That's about the noise introduced by the quantizer in the DAC's modulator, the "Q" block in figure 1.2. It's not about the noise that's already present in the input file.
The signals generation and upsampling was done with SoX.
That was the illustration that when you do the quantization (to...
I believe he said more than that, both in the message you quoted and in earlier messages.
The next sentence after above quote says:
"Early CD players took advantage of this, and generally carried the arithmetic in the digital filter to more than N-bits".
So the increased SNR was used for more...
The majority of the article describes reasons for oversampling in ADC. Then, in this last paragraph, it says that the term "oversampling" is also used to describe a process in DA conversion. I don't see any basis to conclude that the reasons described earlier apply here too? It's rather clear...
Maybe I have the nomenclature wrong, but doesn't "shorter-length" mean fewer coefficients, which implies shallower and earlier roll-off?
Assuming that you actually can hear the difference, that very well may be, for you. However your preference can't really be a basis for claiming that this...