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Linux and music? - Page 2

post #11 of 14
Could I possibly get this explained a bit more? What is the optimal setting under alsamixer? By 0 db you mean 100 on the master volume? What about other volume settings? Also how does this change (if at all) using optical out or head phones as brought up?

Thanks
post #12 of 14
Quote:
Originally Posted by evermooingcow View Post

Could I possibly get this explained a bit more? What is the optimal setting under alsamixer? By 0 db you mean 100 on the master volume? What about other volume settings? Also how does this change (if at all) using optical out or head phones as brought up?

Thanks

Optical/spdif doesn't exactly send a signal the same way. Clipping happens when your audio out signal gets too high for the device pushing it, this causes lost signal (clipping). I believe spdif follows optical in the sense that the signal doesn't increase in strength as apposed to analog, so you can't really clip. At least, that's my understanding of it as I haven't really dissected audio signals that well.

As far as alsamixer, if you hear clipping you've gone up too far. I normally keep channels set to 90 or something. It actually has a red indicator which would give you an idea if you are pushing the signal too high. I just set everything but master to around 90 and then use master through the DE (kmix ect...) and go with that. You will know if it clips, it's that "distorted" sound you get.
Edited by mushroomboy - 1/24/13 at 2:46pm
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post #13 of 14
I see. I'm currently using optical out and I can confirm that the volume settings in alsamixer do not affect the actual output from the speakers. Thanks for the info.
post #14 of 14
100% is fine. The issue is with people amping past 100% (which is what values greater than 0db actually mean). And the clipping there happens in software so the interface you output from becomes irrelevant.

It's worth noting that clipping can only happen with digital signals. The issue is where the wave form is amped (which makes the wave form bigger) past the headroom available and thus the excess gets forcefully removed. In software terms, it's like trying to store a string of 500 characters in a variable / database which is only assigned to hold 400 characters. You'd get the 1st 400 characters stored fine, but the overflow of the last 100 chars will be lost. And thus the peaks of the waveform on digital signals are lost when the signal clips. (interestingly, some producers deliberately clip samples to create a raw, punchy sound - but those are individual samples clipped in a controlled way).

Analogue will obviously also distort if pushed past it's headroom, but analogue will degrade in a more progressive way rather than a simple slicing of the peaks of the waveform.

As with all music theory though, the best advice is to listen to your ears. While 100% / 0db is the best setting to output at in terms of signal to noise ratios, your external amps, speaker cables, or even the speakers themselves, may not like signals coming in that hot so they might add their own distortions. This is compounded by the fact that each passthrough your audio goes through can eat at the headroom of your signals.
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