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SB3 volume control with 20-bit external DAC

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  • #31
    Originally posted by Phil Leigh
    ... all in all, it's probably best just to alter level in the analogue domain - at least we all get to sleep nights! ...
    You got that right Phil! I've been struggling over this issue for awhile now with the new system I've been putting together.

    I came to the same conclusion the other day and decided to replace my PS Audio A-100 amp with their integrated C-100. I was connecting the SB3 directly to the A-100 amp. I received the new unit today and, while I've only had the unit a few hours, the C-100 integrated seems to produce an improved sound over connecting the SB3 directly to the A-100 amp. Some of my recordings that sounded a bit thin before now sound more substantial.


    • #32
      Originally posted by USAudio
      the C-100 integrated seems to produce an improved sound over connecting the SB3 directly to the A-100 amp.
      Apples and Oranges IMHO. As we've learnt, relying on the digital volume controls is a handicap to the Sb3's SNR. A proper comparison might be the integrated amp or a seperate amp+pre versus fixed/stepped passive attenuators and limited use of the volume control.

      The digital volume control is great, and with the integrated you should not be afraid to use it for casual to semi serious listening. When you're all ears though, setting the Sb3 to max is the best idea, and if a comparison is to be made it should be at maximum digital volume for both setups.
      'The Buddha resides quite as comfortably in the circuits of a computer as he does at the top of a mountain or in the petals of a flower'.
      -Robert M. Pirsig, Zen and the Art of Motorcycle Maintenance


      • #33
        Originally posted by Patrick Dixon
        The normal approach with a 16 bit original represented as a 24 bit signal, would be to make the bottom 8 bits 0s - so using the volume control, some of those bits may be turned in 1s, but no 1s will ever be required lower than the 24th bit. If you then throw away the bottom 4 of those 24 bits, you may be throwing away some 1s, and therefore some information. If you use all of the 24 bits you aren't throwing away any information.
        This is very useful, thanks. I had wrongly assumed that as you reduced volume, you lost a bit of information at a time, and if you didn't go below the 16 bit original, which DAC you used would be irrelevant.

        If we had represented the original 16 bit signal as 20 bits, I assume we'd be able to slightly reduce the volume and pass to an external 20 bit DAC, without losing information (although reducing SNR - which I'm not saying is irrelevant, but in my case, I use a pre amp to set my maximum volume, and then use the sq box to adjust slightly for convenience)[/QUOTE]


        • #34
          Originally posted by seanadams
          I think the easiest way to understand this is to forget about numbers, decibels, and bits per sample for a minute, and just think about what's coming out of the DAC.

          To oversimply only slightly: there are two things always coming from the DAC. 1) signal and 2) noise.

          The level of the noise output stays the same no matter what signal level is being produced. That is really important to understand!

          When the DAC is making a loud signal, there is a lot of signal and a little noise. That's a high SNR, which is good.

          However, when the DAC is making a quiet signal, you have a little signal and a little noise. If we now consider the noise level in relation to the signal level, the noise is now louder. The noise level hasn't gone up in absolute terms (eg volts), but relative to the signal it has, so you now have a bad SNR.

          Now consider a simple resistor attenuator being fed by a loud (good SNR) signal from the DAC. When the voltage passes through the resistor divider, everything gets attenuated - the signal and noise together. You have the same* SNR coming out of the divider as you had going in, i.e., the DAC's optimal SNR is preserved.

          OK, now back to bits per sample. As you can see, the above effects really don't have much at all to do with bits per sample. We could send a million bits per sample, and it would still be the same. So why does bit depth matter? What is the significance of 16 vs 24 bit?

          What matters is that we send enough bits per sample that the DAC's full dynamic range is utilized. It is important to realize that the DAC's dynamic range is finite, and is less than its input word size - more like 20 bits, since it is limited by its output noise level.

          By "expanding" a 16 bit signal to 24 bit, all we are doing is saying "these 16 bits go in the most significant slots of the 24 bit word". We haven't improved the SNR of the signal, any more than you can "enhance" a digital photo the way they do on CSI.

          If we attenuate the 16 bit signal, yes, the zeroes and ones will migrate down into the least significant bits of the 24 bit word, and yes, if we still "have all the bits" we could then mathematically go in reverse and get back to the same data. But that is not what the DAC does with the signal! The bits represent a smaller signal now than they did before. We still have exactly the same decreasing SNR effect. Sending 24 bits into the DAC just means we aren't making it any worse than it already is. We haven't "bought more headroom"... it does NOT mean that those first 8 bits of attenuation are "free".

          To prove this, you could play a sine wave through the DAC and measure the SNR at each volume step. We would expect to see the SNR decrease as the volume is decreased. If there were anything special about the point where we start "losing bits", or if we were really getting "extra headroom", then the plot would decrease slowly (or not at all) until it reaches that point, and then there would be an inflection.

          However, that is not what you'll see. The SNR will simply decrease with the signal level, all the way down.

          I hope this helps... for extra credit maybe someone will try testing this?

          * Actally, there are a number of secondary effects which reduce the SNR by the time it gets through the amplifier, but these are vanishingly small in comparison.
          Hi Sean!
          I'm sorry for activating an old thread - but it features my question :-) Just a couple of weeks ago I got a Squeezebox Classic and I make use of the replay gain function. But even after adjusting the volume of my amplifier (which is connected via analog output of my SB) I think that the sound with RG enables is not that clear dynamic than with RG disabled. So my question is: does RG reduce the quality or is it just because it reduces the digital level so the SNR is lower with RG enabled (as the noise floor remains as it is...).

          Thanks for helping and understanding :-)