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Thoughts on a bunch of DACs (and why delta-sigma kinda sucks, just to get you to think about stuff)

Discussion in 'Dedicated Source Components' started by purrin, Dec 5, 2013.
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  1. Sonic Defender Contributor
    I'm curious, it seems in the current discussion one of my questions is creeping in again. What are the exact mechanisms by which an R2R DAC sound better than a D-S DAC? I have very little doubt that some of the better R2R designs we have been hearing about in this thread are actually more musical, better sounding DACs as compared with many of the garden variety D-S DACs, but it does seem reasonable that the real difference in the sound could be mostly attributed to the digital reconstruction filters and or the analogue stage of the DAC (as well as the digital input stage). So if this were at all plausible then couldn't a D-S DAC that was equally well designed in these key metrics sound equally as good?
    I guess a related question of mine would be is it the R2R design that allows the digital filters to work as implemented in say the Theta's and the Yggy? If I have misunderstood this discussion my apology.
  2. anetode
    Well there's your problem -- you're not using a vector display! Get outta here with that PCM/RASTER crap, come back when you can attach a polaroid of an oscilloscope.
    Wannabe audiophiles tryinna edit audio with their 4K "high-res" displays [​IMG] 
    SoupRKnowva and azteca x like this.
  3. PCM is stairsteps but this is not a darwback.  Imagine if you are taking elevator and sewgays every day, you will gradually become unhealthly and sad, you need stairs.  As Mitch Hedberg said, "crutches are strairs that help me walk."  So PCM is strairsteps but they help ear train, to here better and not be a trophy.  The R2R is not stiars, this is a porblem, it is like hndicapped ramp - this ramp is cause of handidcap! 
    Argo Duck likes this.
  4. KeithEmo
    The basic bottom line is that the purpose of a DAC is to convert a list of numbers into an analog voltage. Therefore, assuming that whatever method you use does that well enough, how it actually works is really not terribly important. The circuitry involved in a "basic R2R ladder DAC" is theoretically simpler than that of a Delta-Sigma DAC. In analog circuitry, there are certain drawbacks to making circuits more complicated - basically, since the analog signal picks up noise and distortion whenever it passes through a part, a more complex circuit has "more places for things to go wrong".  However, this really isn't the case with digital circuitry. (A modern computer has a lot more transistors in its CPU than a 25 year old one, and it runs faster, but it still delivers numbers that are "just as clean".)
    I've only ever heard two of what I would consider to be "credible technical arguments" about specific reasons why oversampling DACs in general would be "inferior" - and neither specifically applies to only Delta-Sigma DACs.
    First, ANY type of oversampling filter will introduce SOME ringing. It's a sort of side effect of how digital filters work, and is unavoidable with any current filter design - you can have super accurate frequency energy response or super accurate time response but not both (the ringing you see is there because some of the energy that belongs in the impulse has been "smeared" and appears at the wrong time; if you simply wipe out the smeared energy, then the total amount of energy is no longer perfectly correct). Now, pre-ringing and post-ringing sound different, at least according to some people, and it would be nice to minimize the amount of ringing to a bare minimum - or make sure none of it is in the audible frequency range - so this is one area where extra effort may indeed yield improvements (or, at least, differences). One current idea is that post-ringing is less audible than pre-ringing because it is masked better by the actual signal, so many current filters use some math tricks to "shift" the pre-ringing to post-ringing (you get less or no pre and more post). SOme people prefer the way these filters sound - other's don't.
    Second, jitter produces distortion - because, when there is jitter on the clock, then you have "correct samples at slightly incorrect times" - so, when you convert these samples into analog audio, you get errors. The amount of error you get for a given amount of jitter depends on how much jitter you have as a percentage of the clock period. So, if you have a file sampled at 44k, with 1 ps of jitter on each sample, and you upsample it to a 10x clock "locked" to the original clock, you may well end up with a data stream that is sampled at 440k, but still has 1 ps of jitter on each sample (assuming you use a simple clock, locked to the original, and no jitter reduction or filtering mechanism). Since that 1 ps of jitter is now a larger PERCENTAGE of your new higher sample clock, it may produce a larger amount of  distortion when your new signal is converted. This means that, in order to maintain an equivalent level of performance (in terms of jitter caused distortion), for any oversampling DAC, including a Delta-Sigma one, you must maintain a lower level of jitter. (In other words, all else being equal, a Delta-Sigma DAC may well be a lot more sensitive to jitter, and so you have to assure a lower level of jitter to avoid problems. (Or, to say it in reverse, if you have jitter that you can't eliminate, it will probably cause a Delta-Sigma DAC to distort more than it would a simple NOS DAC.... however, jitter isn't that difficult to reduce or eliminate, and oversampling offers more than enough benefits to offset this "limitation".)
    gikigill, Sko0byDoo and vhsownsbeta like this.
  5. blasjw

    FYI, scopes don't use CRTs for the most part nowadays either.  Predominantly LCD.
  6. anetode
    Na ah, not the r2r ones [​IMG]
  7. smitty1110
    But since of us miss the old CRTs. There's something special about them...

    Actually, that might have been the xrays messing with my brain XD
  8. jcx
    one thing that may be going on with the communication problem is the ambiguous use of the word "DAC"
    it can be a the generic name for the whole box from Benchmark, Schiit... with USB or whatever digital inputs and RCA or XLR analog Voltage, with smooth continuous time audio signal on the output in competent implementations
    it can refer to the function block, today mostly monolithic integrated circuit chip from the likes of TI, AKM, ESS
    it can be a conceptual function label
    the box has continuous, smooth analog output on the RCA because it has analog low pass filters between the DAC chip/module's analog output and the outside world - these are known as reconstruction filters or anti-image/image reject filters
    the internal DAC chips/modules may have current or voltage outputs, may be Zero Order Hold - keeps the analog output quantity constant between update intervals - at whatever internal processing rate used - here you may see "stair steps" - varying amounts of analog circuitry is needed after different DAC chips depending on it and your analog output quality requirements
    NOS, ZOH DACs need very steep analog filters, in recording you need filters with very similar performance , called "anti-alias filters" - these have to be very high order for Redbook - I believe up to 18th order analog filters were used on both ends of the process giving huge phase distortion, lots of post ringing in pure analog filter implementations
    for an idea of the complexity of filtering this steeply in analog - a 18th order filter would likely be made with 9 op amps, 36 precision resistors and capacitors, any pretense to accuracy could require much better than 1% tolerance parts - and its very hard ( where hard==expensive ) to find capacitors sold in even 2% tolerances
    even high oversampling systems need low pass analog filters on their outputs - but they can be much higher frequency and lower order (simpler, cheaper)
  9. goobicii
    I never said its drawback,what you mean R2R is like ramp,tell me more about it ,thats interesting
  10. Sapientiam
    I'd ask your question the other way around as it seems to me, both subjectively and objectively that multibit DACs are adding in less of their own characteristics to the music than D-S DACs.
    So 'Why do D-S DACs sound worse?' Without considering the digital filters there are at least a couple of technical reasons, related to the back-end of the DAC - the modulator and low-bit multibit DAC itself.
    The first reason is that the quantizer can't be correctly dithered because its in a feedback loop, hence the optimum level of dither can't be established. Non-optimal dither levels result in noise modulation - signal correlated shifts in the noise floor. I suspect this is an issue that ESS worked hard on in their 'hyperstream' DACs - reducing noise modulation in the modulator - at least its hinted at in Martin Mallison's RMAF presentation.
    The second issue I don't believe Mallinson talked about at all - that's the fact that the low-bit DAC used isn't a very good one, in terms of the element matching. The apparent ability to use a not-so-good DAC is the whole point of designing D-S converters. Its this that makes them cheap to produce - the old multibit DACs needed resistor laser trimming which takes time with very expensive hardware hence translates to considerably higher prices. In order to get around the limitations of using a DAC with poorer than 10bit precision a lot of signal processing 'tricks' have to be used otherwise the measured THD would look very bad. The tricks used reduce to something quite simple - conversion of harmonic distortion into noise. I take it its assumed in doing this that 'harmonic distortion' = bad and 'noise' = benign but this looks to me to be a questionable assumption for audio. So long as the noise remains totally constant with signal level its reasonable, but that's the rub - 'linearizing' a poor DAC by turning its distortion into noise ISTM generates non-constant noise levels because its distortion isn't constant with signal level. Its this effect I believe which is responsible for the 'bump' in the THD+N vs signal level graph, around -35dB seen on some plots from ES9018 devices.
  11. evillamer
    Not sure if you guys will agree on this?
    A perfect Digital to Analogue Converter is supposed to convert the original recorded digital audio file into analogue wave form without adding or subtracting anything else with perfect timing and precision. e..g Garbage in, Garbage Out in computing terms or DAC Linearity in audio engineering terms.
    (However no such perfect electrical dac will ever exist as we all working with limits of Johnson Noise and other forms of EMF distortion and Jitter etc.)
    If you wanted a DAC is as close to the described perfect dac as possible given the same filter design costing/expertise.  Which will offer a closer result? R2R or Sigma Delta?
  12. jcx
    yes Head-Fi has a problem recognizing intellectual/engineering "critical analysis" - fanboys don't understand anything beyond you may have said something could be/have been different so you must be dissing their guru/objects of adoration
    n3rdling, snip3r77 and altrunox like this.
  13. jimmypowder

    May the Yggdrasil envelope you in madness!

    All hail Schiit .
  14. wahsmoh
    Zach, still waiting for your Theta Gen V vs Progeny A comparison. I want to know if I am hearing atleast 98% of the real deal and I might just wait til I have a real job til I buy a Yggy cause the Progeny is sex to your ears.
  15. zach915m

    Hah! I'm fascinated as well. Tonight I plan to get a little session in - I'll get some thoughts down on here later tonight or in the am.
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