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24/96 digitalization - can it be audible, blind test results
Axon
post Jul 13 2011, 01:57
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QUOTE (DonP @ Jun 28 2011, 17:44) *
QUOTE (Axon @ Jun 28 2011, 15:33) *
:F

If it's easier with speakers, then I'm inclined to at least suspect that the difference might be distortion-induced.


That's the second post you've made in this thread suggesting that a small difference in volume (the other time, .05 dB) was detectable due to distortion.

By what mechanism would distortion differ significantly over a .05dB power change? Or even 1 dB?

I can certainly accept that speakers can have audible distortion. The question is how much the distortion can change, beyond just scaling with the main signal, over a very small range of level.

I haven't forgotten about this, and I think you may be right. I can't really back up my original claim, on continued reflection.

I was sorta thinking that, in the context of harmonics increasing faster than increases in the magnitude of the fundamental, there ought to be some sort of continuum between "static nonlinear distortion" and "clipping". Ie, a tiny change in volume could cause a huge relative change in harmonic distortion at the onset of clipping, but if you define the distortion statically, ie y=x+0.001x^2, there's no relative change whatsoever -- the second harmonic is always some fixed level below the first harmonic. And my hypothesis only works if the relative change is so high as to cause some harmonic to no longer get masked.

But for that to be the case, I think the system would have to be well on its way to being limited/clipped to begin with. If a 0.05db volume difference causes say a 1db level change for some particular harmonic, making it audible, then if the system has another 10db of clean headroom, we ought to expect that this harmonic ought to b e able to increase by another 200db....
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2Bdecided
post Jul 13 2011, 12:58
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If one of the speaker cones has a hard physical restriction that is reached when the volume is increased a little, then that would be a good example of what you describe.

Trivial to set up with test tones and a little speaker modification wink.gif

Hard to imagine how this could happen with real music and a system with demonstrably useful amounts of headroom.

Though I'm often amazed how much hidden distortion there is in some systems, which is only revealed by narrow/low bandwidth sources.

Cheers,
David.
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Axon
post Jul 13 2011, 16:37
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The best idea I could come up with is if the distortion was somehow "multiplexed" into several harmonics, but irregularly. That is, for a given volume increase, harmonic A would bump up by a significant amount, but the other harmonics would stay the same. Then for some further volume increase, harmonic B would increase, but A and all the others would stay the same.

This would give the observed behavior I would need for my hypothesis. But it's ridiculously thin. Also, it's ridiculous.
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Arnold B. Kruege...
post Jul 13 2011, 20:04
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QUOTE (RonaldDumsfeld @ Jul 1 2011, 18:29) *
Is there a possibility that some convertor chips are designed from scratch to convert at multiples of 44.1 but have the ability to resample to multiples of 48 and vice versa? Are these things supposed to be equally good at conversion (as far as the human ear is capable) at all sample rates?


As a rule converter chips rely on an external clock to control the speed at which they work. IOW, they just follow the clock and work the same no matter what it is.
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knutinh
post Jul 13 2011, 21:48
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So what is the conclusion on the topic at hand?

-k
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AndyH-ha
post Jul 14 2011, 06:59
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My apologies for not having the energy to read enough of yet another 24 bit argument, and maybe getting very off topic here, but while converter chips may function to the clock given them at the moment, the results are not always the same. Maybe mine is a special case, somehow defective: I have a Echo Mia that does rather poorly at 22050Hz, in spite of seeming to be very good at higher sample rates.

I've recorded many hundreds of hours of spoken audio that ended up at 22050Hz final product. It finally dawned on me that recording there to begin with would be more efficient. I normally use the Mia computer for those recordings.

Results were not so satisfactory, although not really unusable. There were definitely different than recording at 44.1kHz and downsampling. I verified that the Audiophile 2496 containing computer has no problem at 22050 but I did not find it convenient to switch for general use, so I just continue to record at 44.1 and downsample.
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