Glen,
I've slept a few night since I read Bob Katz Master book, but I thought dithering was done to smooth jitter when converting from different sampling rates rather that sample sizes. i.e. Going from 96khz to 44.1khz. Am I mistaken? If so whats that called. And in the real world is "truncation noise" an issue?
Thanks
In my book, frankly, 98% of the time, none of what we're talking about is an issue, as 98% of the time it's all inaudible.
One has to take a lot of Bob Katz's stuff with a grain of salt. He's a very smart guy, but he tends to get purposely lost in the intricate detailed trees of the math and can't see the real woods of the situation. There's a whole lot of his stuff that while technically true, is largely irrelevant in the real world, IMHO.
As far as using dither to attack jitter from sample rate conversion, I just don't see how, myself. Maybe dither noise (for all dithering really is is is a form of noise) may help confuse the ears or mask other extreme low noise low imperfections, but it's not really designed to do anything other than to introduce noise at the last bit of a 16-bit word in order to average out just how inaccurate it winds up being.
Think of it like this. Take the value of pi, which we never can get quite exactly, because the decimal places just keep going. But let's say we "sample" it's value at 3.14159. But the spreadsheet cell we're sticking it in only has room for four decimal places, 3.1415. We might "dither" the last digit to say 3.1416 because it's actually closer to the "real" value for pi than 3.1415 actually is, even though the 5 is accurate for that decimal place.
Well, while not really exactly, that's kind of like what dithering is doing to the 16th bit of the binary sample value after chopping off the rest. But instead of rounding the value up or down specifically, dither is choosing a 1 or a 0 for that last bit based upon it's own pattern of selecting 1s and 0s that's not random, but is random in relation to the actual signal value. The argument is that this tends to "average out" the inaccuracy of that last bit in a way that is less audible or more pleasant to the human ear. The fact that this is happening at -96 dBFS on signals that rarely have any useful information at all below -75dbFS notwithstanding.
What that has to do with jitter, I don't know. Maybe it might help in smoothing out the last bit in the calculated values at the new sample rate. But I'd say that the best way to handle that is not to deal with sample rate conversion at all. Again Katz will probably say something different, and this is one of those never-ending forum arguments, but unless one is forced to convert from 48k audio for video, nobody here IMHO should ever be working in anything other than 44.1k.
Let the flames begin

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G.