miniDSP kits, our answers to your technical questions

[OllBoll]

Originally Posted by IMSTOOPID

the analog volume control smd resistor ladder will maintain a 112db s/n ratio from 0v to 2.5v input.. with max out 2.8/5.6v bal output.
and yes when you are mixing balanced and unbalanced gear you need at least -60 db of gain in some cases.. regardless of gain staging.. my system routinely runs at -20 db for normal listening because of the mix of 4 different amps all with different gains 23/26/29/32 and some are balanced input some are unbalanced..
each was chosen for a specific job for a specific driver and it';s power out..(and cost)

oh yea and 4 way active with speakrs ranging form 85-93 db sensitivity.. makes the whole gain thing a total cluster .lol

I would like to run it lower but I run out of gain cut on the dcx right now..
a 4 way active system gets LOUD fast with log control pots..
this is one place I prefer linear pots/.

Then why not just fix the gain structure by individual L-pads for each channel?

[IMSTOOPID]

UGH!
how about a properly designed digital/analog output in the the first place?
no need for L pads if the system si done right and what now i got to gain aroun 8 lpads??
Id be better of designing and building my own 8 channel preamp around the cs3318
which I don't have time for..
guess i'll stick with cascaded behringers for now.

[OllBoll]

Originally Posted by IMSTOOPID

UGH!
how about a properly designed digital/analog output in the the first place?
no need for L pads if the system si done right and what now i got to gain aroun 8 lpads??
Id be better of designing and building my own 8 channel preamp around the cs3318
which I don't have time for..
guess i'll stick with cascaded behringers for now.

It would be enough to make one -23 db L-pad for each channel, the 9 db difference to the most attenuated channel is so small so it wouldn't impact SQ.

[curryman]

Originally Posted by IMSTOOPID

UGH!
how about a properly designed digital/analog output in the the first place?
no need for L pads if the system si done right and what now i got to gain aroun 8 lpads??
Id be better of designing and building my own 8 channel preamp around the cs3318
which I don't have time for..
guess i'll stick with cascaded behringers for now.

mhh, the analog outputs of the Behringer DCX is +22dBu FS = 9.75Vrms which is more compatible to pro gear than hifi equipment. Thus with respect to propper gain structure with hifi amps the miniDSPs are more compatible

How does a properly designed digital/analog output look like, having in mind that you have to interface with a wide variety of systems (amps with different gains and particularly different speakers ranging from <80dB to >110dB) and not to forget the really affordable price? It's always easy to ask for a product that does fit ones own individual needs and doesn't cost too much

[IMSTOOPID]

mine are 5 v modded unit..
and the stk behringer has -15 db of cut

a proper digital volume is done as said before in the cpu post processing. just like the jriver or foobar media players do..
analog is easy after you get the signal form the dac.. making 8 channel track close enough is tough..

[OllBoll]

I don't see how it would matter where the calculations are done, the only thing that matters is that it is done and with what precision the calculations are done in. It makes no difference if the attenuation is done in the media player, or in the OS volume controller, or in the MiniDSP or in the built in digital volume controller in the DAC itself.

[IMSTOOPID]

REREAD THE ESS SABER DOCS..

WHERE the volume contorl is done in the digital domain makes a HUGE difference..
you wouldn't want the dac making the volume changes to a 16 bit signal post processing .. you could end up with 8 bit s or less of signal resolution.

my point is with 32/64 bit FP available there is no reason not to do it correctly.

[OllBoll]

Originally Posted by IMSTOOPID

REREAD THE ESS SABER DOCS..

WHERE the volume contorl is done in the digital domain makes a HUGE difference..
you wouldn't want the dac making the volume changes to a 16 bit signal post processing .. you could end up with 8 bit s or less of signal resolution.

my point is with 32/64 bit FP available there is no reason not to do it correctly.

I don't think you understand how digital attenuation actually works : /

You do digital attenuation like this: You have a full signal in 24 bit words. You want to attenuate the signal by ~ 48 db, so you divide the 24 bit words by 256 since 2^8 = 256. You might also want to add dither before but lets ignore that for now. The important thing is that the output of the attenuation is the original signal / X, where X is how much you want to attenuate. And now for the even more important fact: You could do this calculation with 24 bits of information in a 99999 bit word length and the output would be exactly the same except for lots of unused zeroes. Sure if your DAC can input 32/64 bit floating point input words this would be another story but I have yet to see one.

And if you have to do volume changes to a 16 bit signal which would be a 24 bit signal that already has ~ 48 db of attenuation then your gain structure is messed up. If your gain structure is not messed up that means full signal volume in to the DAC means max volume in your actual system. And that means that cutting bits from this full signal equals attenuation.

If your system has such a gain structure and a 24 bit DAC then it doesn't matter where you do the attenuation because any noise that is introduced due to the resolution loss cannot be heard by humans. You don't need 32 bit DACs, or floatnig point DACs because nice and normal 24 bit DACs are good enough.

[curryman]

OllBoll, I generally agree but want to add that it can make a difference if you do attenuate (significantly) before the DSP doing a lot of math for filter and EQ calculations vs. after the DSP. Thus from the technical point of view (not marketing or subjectivism ) it should be done after the DSP, equal in the DSP itself or in the DAC.

The ESS Paper seems to be quite old since they are talking about 16Bit DACs vs their 32Bit implementation. Or maybe they just want to emphasize how good they are Do the same math with a 24Bit system which is actually standard and you will not see significant influence. Finally as long as one really believes digital volume control is evil, he will not be satisfied with such a system

[OllBoll]

Originally Posted by curryman

OllBoll, I generally agree but want to add that it can make a difference if you do attenuate (significantly) before the DSP doing a lot of math for filter and EQ calculations vs. after the DSP. Thus from the technical point of view (not marketing or subjectivism ) it should be done after the DSP, equal in the DSP itself or in the DAC.

Of course, but in practice with this level of attenuation the signal level will be so low so you shouln't hear it as long as said before the gain structure is ok. A prime example of this is that most of all music today is stored in 16 bit format since you don't need more and thus you'd have to attenuate more than 48 db to even start to reduce SQ.