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Old 7th April 2010, 07:12 PM   #11
qusp is offline qusp  Australia
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right, OK thanks, not sure if you saw the last analogy I added above and if it makes any sense to you. what do they say, a little information is a dangerous thing regardless the result is still my favored way of doing things and its a trade-off i'm willing to make given the circumstances. any other kind of pot will do the same thing but with different methods, generally with other complications not presented here; so i'm all good. wasnt so much thinking there was a free lunch involved, but more that I expected that since not all the bit depth of the chip was used for audio, that functions such as volume would be treated as 'system tasks' if that makes any sense. I do wonder why something like that isnt undertaken, given the bandwidth we have available in todays tech. anyway thanks again for clearing that up

Last edited by qusp; 7th April 2010 at 07:17 PM.
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Old 7th April 2010, 07:24 PM   #12
qusp is offline qusp  Australia
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LOL, I just realized the flaw in my thinking (or have I); in audio, even if a completely separate chip of equal resolution were to multitask this volume control (as a sort of 'overlay'), SNR would still be effected in the same way is that right?, being that SNR is just a yardstick for what is a finite resource measured over time. sorry, probably some conceptual wrestling I should do in the privacy of my own home ;D.

apologies for all the analogies, trying to put it into language I understand and concepts i'm more familiar with

Last edited by qusp; 7th April 2010 at 07:29 PM.
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Old 7th April 2010, 07:51 PM   #13
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Well the 'problem' in this scenario is that your trade off involves more added noise relative to the useful signal, so higher 'noise power' out of band because of the sigma delta working principle. In theory you could tweak the anti alias filtering on the analog side for lower cutoff / higher order to mitigate this.
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Old 7th April 2010, 09:05 PM   #14
markusA is offline markusA  Sweden
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I don't have the technical know how like many of you guys but this is how I understand it.
qusp is right that the 32bits used by the dac does give room for attenuation without loosning resolution/data, at least in the beginning. If used to attenuate a lot it will become more cumbersome.
So, designwise one should aim to match the amplification to your listening levels.
As long as the attenuation isn't big everything is fine.
SNR should be fine as well as long as the attenuation isn't too great.

Othoh, analog attenuation isn't perfect either...
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Old 8th April 2010, 12:40 AM   #15
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Originally Posted by markusG View Post
Othoh, analog attenuation isn't perfect either...
It's usually actually usually much less perfect.

Just measure the results and see. Any analog loss(attenuation) will bring with it added distortion and noise of one sort or another. You also have issues like channel matching and non-linearity to deal with in some cases.

If the DAC has sufficient dynamic range such as the ES9018 it will be difficult at best to match with an analog attenuator.
Less pulp more juice Twisted Pear Audio.
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Old 8th April 2010, 02:51 AM   #16
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Perhaps its time to tell you guys that I am using the Sabre "D" attenuation myself as well . Plan to use r2r but the project is slow.

As the Sabre is a class on its own ( hyperstream modulator - low , 4th order noise shaping) , D attenuation actually brings it closer to its sigma delta siblings , with their higher order noise shaping.
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