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SB3 volume control with 20-bit external DAC

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  • #16
    Originally posted by seanadams
    No - you are making this FAR more complicated than it is. Forget about decibels!

    All you do is set the jumpers so that when transporter is playing at full volume, you are at the maximum listening level you'd want to use. Just start at -30 and move them up towards 0, until you reach your max listening level. That is really all there is to it.

    If you are already using a preamp, or if your amplifier has gain controls, then you should use those controls and don't touch the jumpers.

    What you are trying to avoid is a situation where you are at your maximum listening level with the volume bar only half way up. This is what might happen if you connected transporter directly to an amp. That's when you need to change the jumpers.
    Thanks Sean,
    That's a simple way of setting and understanding!
    Bill

    Comment


    • #17
      Originally posted by seanadams
      This is completely, 100% wrong. It doesn't matter how many people say this, it is still wrong.
      Yes, you only maintain all the resolution of the original 16 bits by using all 24 bits - so truncating to 20 will loose you some information.
      Originally posted by seanadams
      You _ALWAYS_ lose SNR and dynamic range. The "you still have all the bits" concept is completely flawed.
      Well not completely flawed ;-) Any digital signal which doesn't swing the whole numerical range available to it, will loose some SNR (compared to the maximum available), at the DAC. However, this 'loss' of SNR may or may not be better than any SNR loss in a subsequent buffer and volume control circuit.

      Originally posted by seanadams
      Use the digital volume for day-to-day adjustment within your normal listening range, not to compensate for a badly matched system.
      Absolutely.

      Originally posted by seanadams
      The new 0 to 100 scale is, I believe, true decibels. (In the past when we used only 40 steps, it was a more complicated curve designed to stretch out the low end of the range). So -10 corresponds to 90, -20 to 80, and -30 to 70.
      Well it's your product (nice title BTW), but AFAIK each step corresponds to 0.5dB. So 90 is -5dB, 80 is -10dB and 70 is -15dB.

      The previous 0-40 range gave steps of 1.25dB each.
      Last edited by Patrick Dixon; 2006-12-26, 15:30.
      www.at-tunes.co.uk

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      • #18
        Originally posted by Patrick Dixon
        AFAIK each step corresponds to 0.5dB. So 90 is -5dB, 80 is -10dB and 70 is -15dB.
        You're correct, my bad.

        Comment


        • #19
          From another volume thread (2 months old)
          Originally posted by boybees
          Slim Server 6.5: If I have the volume control set at 80 instead of 100, does this take out information from the digital signal?
          Originally posted by Patrick Dixon
          No - it just gives you a slight reduction in the actual signal to noise ratio at the DAC.
          and now, in response to Sean's comments "This is completely, 100% wrong. It doesn't matter how many people say this, it is still wrong":
          Originally posted by Patrick Dixon
          Yes, you only maintain all the resolution of the original 16 bits by using all 24 bits - so truncating to 20 will loose you some information.
          I'm now more confused than I started.

          So Patrick is saying that reducing the digital volume will NOT take out information from the original signal, just slightly reduce the SNR.
          And Patrick is saying that even if you leave the volume at 100, you will lose original resolution of the 16 bits if you have a 20 bit DAC (as you are sending it a 24 bit signal).

          I apologise if I have miss interpreted everything, but it seems that Sean and Patrick are saying completely different things.

          The one bit that does seem clear, is that digitally reducing the volume, prior to converting to anologue, reduces the SNR.

          Comment


          • #20
            Originally posted by Triggaaar
            I'm now more confused than I started.
            Sorry about that.

            Originally posted by Triggaaar
            So Patrick is saying that reducing the digital volume will NOT take out information from the original signal, just slightly reduce the SNR.
            Yes - providing the input signal is 16 bit audio and you don't go lower than (IIRC) -35dB.
            Originally posted by Triggaaar
            And Patrick is saying that even if you leave the volume at 100, you will lose original resolution of the 16 bits if you have a 20 bit DAC (as you are sending it a 24 bit signal).
            No, I'm not saying that! At 100, the digits will be passed through untouched, it's only if you reduce volume you can loose resolution.

            Originally posted by Triggaaar
            The one bit that does seem clear, is that digitally reducing the volume, prior to converting to anologue, reduces the SNR.
            Almost everything you do to an analogue signal reduces its SNR too, so what I'm trying to say is that digital volume control vs analogue volume control is something of a trade-off.
            www.at-tunes.co.uk

            Comment


            • #21
              Originally posted by Patrick Dixon
              Sorry about that.

              Yes - providing the input signal is 16 bit audio and you don't go lower than (IIRC) -35dB.
              The bits don't have the same value that they had before you shifted them!! Attenuate by 6db and you're then using only half of the system's output range - you're closer to the noise floor and you have only half of the available "steps".

              I don't know what it means to "slightly reduce" the SNR. The reduction is exactly equal to the amount of attenuation.

              Comment


              • #22
                Thanks for your time trying to help me understand this. I am sorry if everyone else thinks this is clear, and I'm confusing the issue.

                Originally posted by Patrick Dixon
                No, I'm not saying that! At 100, the digits will be passed through untouched, it's only if you reduce volume you can loose resolution.
                Yes, I realise that at 100 volume, the squeezebox will pass the digits through without degrading the signal - I am trying to understand the following two points, which to me, seem contradictory (ignoring reduced SNR):
                "Yes, you only maintain all the resolution of the original 16 bits by using all 24 bits - so truncating to 20 will loose you some information."
                and
                reducing the digital volume will NOT take out information from the original signal, providing the input signal is 16 bit audio and you don't go lower than (IIRC) -35dB

                Thanks

                Comment


                • #23
                  Originally posted by Triggaaar
                  "Yes, you only maintain all the resolution of the original 16 bits by using all 24 bits - so truncating to 20 will loose you some information."
                  a 24-bit input value some of the lower attentuation values will not end up truncating any bits, in other words.

                  Thats part of why the input value is 24 bits even though the data is only 16 bits.

                  reducing the digital volume will NOT take out information from the original signal, providing the input signal is 16 bit audio and you don't go lower than (IIRC) -35dB
                  The same thing as above: the original value on a CD is only 16 bits.

                  So if you take a 16 bit value, and left-shift it (effectively) 8 times, rotating 0's into the low order bits, you have more wiggle room.

                  Or if binary isnt your cup of tea: think of it in decimal. You have a number, say, between 00 and 99, if you make it between 000 and 990, by multiplying by 10, you have a bit extra resolution when you do division without resorting to fractions. In this case, it actually gets you through the first N steps of volume attenuation without any truncation.

                  Comment


                  • #24
                    If you have ripped a file (say FLAC) from a CD, and then stream it from the PC to your squeezebox, does the squeezebox receive a 16 bit signal, and convert it, or does the slimserver stream a 24 bit signal?

                    Comment


                    • #25
                      There are two different issues here, and I think they might be causing some confusion. One is whether information is lost when you lower the volume; in other words, is it possible to reconstitute the original signal from the signal with lowered volume? The answer to this in the case of the SB (and probably TP) is that there is a certain range from 100 down for which this is possible, but that below some setting it isn't any longer (I think this is 35dB, so to lowest setting is 30 on the 100 point scale).
                      But of course this is not a question of much practical interest.

                      Another question is signal/noise ratio of the signal going to a DAC. In that case the issue above is probably tototally unimportant, and the only issue is how much of the dynamic range of the DAC is being used. Any digital signal with lower than max volume will suffer reduced S/N, and it shouldn't matter much, if at all, if it is slightly below 30 or slightly above 30.

                      Comment


                      • #26
                        Ignoring SNR, what is confusing me particularly, is how according to the statements I've quoted, reducing the volume to a certain point will not result in losing original information if using a 24 bit DAC, but it will if using an external 20 bit DAC.

                        Comment


                        • #27
                          Originally posted by Triggaaar
                          Ignoring SNR, what is confusing me particularly, is how according to the statements I've quoted, reducing the volume to a certain point will not result in losing original information if using a 24 bit DAC, but it will if using an external 20 bit DAC.
                          The volume control settings are carefully chosen so that with a 16 bit original no bits lower than the 24th one will be changed. 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.

                          It may be helpful to think of SNR as an 'analogue' thing and information as a 'digital' thing. The digital signal in this case is a representation of an analogue signal, and how you represent it digitally has an implication on the maximum SNR attainable in the analogue domain.

                          In the digital domain you can represent and manipulate the signal in all kinds of different ways, but so long as you don't discard any bits, you retain all the information and you can still get back to the original digital signal.
                          www.at-tunes.co.uk

                          Comment


                          • #28
                            So Patrick - to (try and) cut a long story short...

                            In the SB implementation, if you take a 16-bit file, the SB internally upsamples to and outputs at 24 bits...if those 24 bits are sent to a 20-bit DAC do you lose any information (at full volume).

                            My guess is NO since the original 16 bits are preserved (within the 20). This is surely the case otherwise the SB would only be "bit perfect" with 24-bit DACs which is not the case AFAIK.

                            I'm happy with the idea that any lowering of the digital volume potentially loses information, regardless of bit-depth
                            You want to see the signal path BEFORE it gets onto a CD/vinyl...it ain't what you'd call minimal...
                            Touch(wired/W7)+Teddy Pardo PSU - Audiolense 3.3/2.0+INGUZ DRC - MF M1 DAC - Linn 5103 - full Aktiv 5.1 system (6x LK140's, ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Pekin Tuner, Townsend Supertweeters,VdH Toslink,Kimber 8TC Speaker & Chord Signature Plus Interconnect cables
                            Stax4070+SRM7/II phones
                            Kitchen Boom, Outdoors: SB Radio, Harmony One remote for everything.

                            Comment


                            • #29
                              Originally posted by Phil Leigh
                              So Patrick - to (try and) cut a long story short...
                              ... if only.

                              Originally posted by Phil Leigh
                              In the SB implementation, if you take a 16-bit file, the SB internally upsamples to and outputs at 24 bits
                              Err, that's not upsampling - upsampling is something completely different!

                              Originally posted by Phil Leigh
                              ...if those 24 bits are sent to a 20-bit DAC do you lose any information (at full volume).

                              My guess is NO since the original 16 bits are preserved (within the 20). This is surely the case otherwise the SB would only be "bit perfect" with 24-bit DACs which is not the case AFAIK.
                              Please see my previous post - at full volume the bits are passed through unchanged - therefore there can be no loss of information.

                              Originally posted by Phil Leigh
                              I'm happy with the idea that any lowering of the digital volume potentially loses information, regardless of bit-depth
                              It doesn't! (It has implications on the SNR when the signal is converted back to analogue though.)
                              www.at-tunes.co.uk

                              Comment


                              • #30
                                Yes sorry I didn't (obviously!) mean upsample...what is the correct term for altering the bit-depth of a sample? "re-fathoming"?

                                a-ha!...

                                I get it (at last) you can alter the extra bits as much as you like until you happen to cause an effect on one of the original 16...and once you do that you lose information? - that all makes perfect sense.

                                all in all, it's probably best just to alter level in the analogue domain - at least we all get to sleep nights!
                                Cheers
                                Phil
                                You want to see the signal path BEFORE it gets onto a CD/vinyl...it ain't what you'd call minimal...
                                Touch(wired/W7)+Teddy Pardo PSU - Audiolense 3.3/2.0+INGUZ DRC - MF M1 DAC - Linn 5103 - full Aktiv 5.1 system (6x LK140's, ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Pekin Tuner, Townsend Supertweeters,VdH Toslink,Kimber 8TC Speaker & Chord Signature Plus Interconnect cables
                                Stax4070+SRM7/II phones
                                Kitchen Boom, Outdoors: SB Radio, Harmony One remote for everything.

                                Comment

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