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Discussion in 'High-end Audio Forum' started by khaos974, May 12, 2011.
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  1. negura
    In terms of subjective preferences, there are a lot of good reviews for all the TotalDACs on Audiostream. The tube output one had a glowing review too. That's fine and it got some of my interest, but from experience I know to take reviews with a grain of salt.
    Following Vincent's reply I understand if I were to summarize:
    - the maximum amplitude is limited to 14 bit and there is a relation to 0.01% distortion as a limiting factor
    - ENOB not clear/unconfirmed at the moment
    - I don't dispute the measurements, but it's still unclear to me how the above allows for such a high DR

    It seems quite clear also from this article the relationship between ENOB and DR:
    ENOB = (dynamic range – 1.76)/6.02 
    As it stands either the DAC has just over 26 bits (ENOB) or the DR should be much lower.
    Very interestingly someone pointed me to this link: http://www.totaldac.com/boitier_stereo.htm

    Even for what was the entry level the exciting DR figures come in again: 134dBFs which should correlate 22 ENOB. If we compare this to MSBs Analog DAC's 133dB, the Yggdrasil's 122dB (measured by atomicbob) or the Theta V A's 110dB, the TotalDAC A1 at a fraction of the cost to the MSB could look like a winner in these areas.
    Now as it happens I have actually heard the TotalDAC A1, at home and at length, and my comments are posted elsewhere. Not only that it didn't come anywhere within sight of the performance of the MSB Analog or Yggdrasil, but 3 of us independently on various rigs compared it with the Theta V A. The TotalDAC A1 came awfully short in resolution, DR, clarity or precision to the very old Theta V A. The Theta V A has a DR of around 108-110dB, which gives 17-18 effective bits. Did I think the A1 sounded nice - yes. It was organic, engaging, had good tonality etc. But at the same time it could not perform or resolve even remotely well compared to the Theta. The final proof, if any was needed, was that the TotalDAC A1 owner has purchased a Theta V A.
    I am not saying the TotalDAC D1 and monos will not be much better. I am sure they are. But in this only example of hearing a TotalDAC, the DR spec, which is (about) the only performance related spec published, unfortunately did not translate to anything remotely comparable when hearing the DAC.
    I look forward hearing the TotalDAC D1 (hopefully before the EOY) to know first hand if it can actually match the MSB Analog's resolution, clarity, background blackness, and go beyond. That would be impressive and I would be glad there is more worthy competition out there, from much closer to me.
    dan.gheorghe, Beolab and mori39 like this.
  2. romaz
    These are the challenges of lab measurements, aren't they?  They can provide some objective quantification of what we are hearing or what we should hear but they really can't tell us if something sounds good.  My comments on resolution were only in response to your query and it was a valid question.  In the end, with respect to MSB vs TotalDac or other fine DACs at this level, it will amount to splitting hairs rather than major deltas.  I have always valued your comments and I look forward to hearing your take on the TotalDac D1 once you hear it.
    In an interview on Stereo and Mono, here is what Vincent had to say on this matter:
    “The truth is in listening, not in the theory or standard measurements. So the listening test decide what is good, even if it is unexpected or unbelievable. I like working like an instrument maker, not only as an electronic engineer. An "engineer only" work leads to very good measurements but most time a not so good sound.”
    I think this says it all.
  3. preproman

    Brian and I are engaged.  I'm guessing I'm after you.   Thanks a lot for that information, it's going to be really interesting testing them out.
    The TotalDAC Server has the re clocker in it - correct? If so I see why adding the Regen would yield no improvements.  You put a re clocker on top of a re clocker - shouldn't expect to yield any positive results from that.  One the other hand it makes sense adding the Regen to the M1 / Aries and getting good results.
    30 seconds is a bit of a stretch. But I'm going to put it through all the paces..  If nothing else it looks like it's very user friendly, so that's plus right there.  If I could get grid of this big monitor for an iPad, then I'm all for that..
    Thanks again Roy for that tip..
  4. romaz
    Sure.  I'm literally not kidding about 30 seconds.  This is how long it took me.  The N10 didn't even have to warm up much but again your baseline system may be better then mine.  I'll be interested to hear your impression.  As for the iPad user interface, it is excellent.
  5. nepherte
    Just for clarification, what iPad interface are we talking about?
  6. romaz
    Aurender uses an iPad interface called Conductor to control it.  It is the only way to control it.  It is the equivalent of MPAD for the TotalDac but unlike MPAD, Conductor can also play Tidal.  Roon capability is also expected.
  7. castleofargh Contributor
     I imagine this should probably move to sound science, but until then, here I go.
    I effectively do not understand what you're talking about. sure I get the general concept, but I fail to imagine it in actual audible soundwaves. concept, that a percussion has a very fast rising time on impact, so to be reproduced correctly it needs fast sound reproduction and stuff like that.
    but in practice the sound is in the audible range and will last more than a period of an audible frequency, so we don't really need super small data points do we? for a cymbal, I would imagine the first part of the sound(before it start going up in frequencies) would last a few ms. that's several magnitude bigger than what you've been arguing about. and again if something of audible amplitude variation was to rise faster than some 20µs, then it would express ultrasonic content right?  I can understand the concept, and in very rare extreme test tracks passages, I even could hear something that could go with the idea. but that was with mp3 not 16/44 PCM. I have no concrete situation of CD music not sounding like highres(aside from remastering), so of course when I read about how we are able to identify stuff smaller than the sample size of a CD, I wonder what could those small stuff be?
    here is how I conceptualize super short time events that could possibly be recognized by ear:
    1/ we have an audible tone, we instantaneously stop it( would a headphone be able to stop fast enough?) for a few µs, and start it again with the same tone rising calmly as to become an audible frequency. in such a situation I imagine I could probably get to hear something out of a very small time delay. but what instrument does that, what transducer can record and replay it accurately?
    2/ left and right ear differences that serve to estimate the direction of a sound. those differences would be rather small when the source is rather close to be on the axis right in front of us. and I'm willing to believe we can identify those very small left right delays and use them to place the sound in space.  IMO what could be wrong would be in a worst case scenario, the first sample, and by as much as 1/44100s. the spacing between samples is much better than the sample resolution(super duper clocks) so the wave will still be created accurately over a few samples(as long as it's in the audible range), and phase difference will be properly estimated soon after the first few samples.
    even if at some point we were to mistake a sound at 0.3° in front of us for a sound IDK maybe at 1°(and I see no reason for that to happen on a CD), I'm not so sure we humans have that much accuracy when placing sounds in space. and I doubt the guy doing the mastering is setting the panning of tracks within 2degrees. most likely he's turning a knob until the sound is approximately where he wants it. there is no losing the real sound in this matter even if we were to be misguided by 16/44 and I'm not sure at all that we are.
    within the same kind of idea, if a phase error of 7µs is enough, then anything more than... let's go for 35° out of phase in the audible range, would potentially create such a delay. and our gears are usually minimum phase, not perfect phase. so this + that+ a great many headphones(mostly dynamic), and many of us end up with variations that are bigger than what could come because of 16/44. yet I don't see too many people crying about it and proving that they can distinguish the sound difference.
    3/ the starting moment of a sound. will the album be ruined if the guitar starts up to 22µs later than it did in the studio? who will ever know, let alone notice? did all recording engineers in the 70's align all their tracks withing 1µS margin? to me that one is about splitting air, not getting a better audio.
    4/ stuff like shockwaves from explosion(not my favorite music TBH ^_^)? but I know nothing about it except that it still needs to travel in the air, be recorded and replayed.
    I wont say I've ruled out everything, but all my other ideas right now involve ultrasonic sounds, so I just dismiss those as "not heard by me".
    conclusion, while I would never deny the potential for better accuracy in high res, I feel like I can deny the claim of 16/44 falling under what we can all distinguish because we can distinguish some 7µs stimulus. if only because of how few honest and controlled material we have showing that people can identify a 16/44 track(and not just the low pass filter of the DAC, or some massive IMD from loud ultrasonic content being badly handled by the sound system). I don't know under what circumstance a human can react to a 7µs stimulus, but I doubt it happens often in the waveform of audible music.

    well if errors in time huge enough to actually change the pitch of sounds doesn't impact time resolution, I wonder what does ^_^.
    to me oversampling is mainly a mean to use more convenient low pass filters, not a way to fill the gaps(that said I'm 100% pro OS). if you question the number of samples and the need to reduce the gaps, I guess you question either Nyquist and pretty much all that is digital sound to this day, or our inability to hear ultrasounds. else 16/44 seems just fine for most uses.
    I would attribute your testimony of feeling more dynamic to 2 ideas:
    1/ the coloration and added distortions/noise/crosstalk end up really sounding different and perhaps at times, more engaging?
    2/ non linearity. the needle going far too high on some trebles and giving an output several db higher than the signal on the groove intended. but as this would happen on louder parts with the most magnitude, we can end up with a wider range of amplitudes. quiet part quieter and loudest part louder.
  8. romaz
    Sorry, I wish I could speak French because clearly you have good intelligence on the science of sound and I would love to engage you in meaningful conversation but I am simply struggling to understand you.  I still believe you are understanding time resolution differently than I am.  Perhaps another analogy I can use is soundstage, both width and depth.  Have you ever tried listening to a large and complex orchestral performance recorded in 96k mp3?  Quite likely, you will find it to sound very 2 dimensional.  You might recognize the presence of woodwind instruments but you probably will struggle to know where on the stage the woodwind section is relative to the string section.  To take it further, within the string section, even though they travel at different frequencies, you will probably struggle to differentiate the differences between an upper standup bass versus the lower registers of a cello.  Then play that same recording via a properly mastered SACD which is 64x the resolution of a CD and all of a sudden, you know exactly where everything is on the stage.  This 3D quality has everything to do with time resolution and what oversampling hopes to achieve.
  9. castleofargh Contributor

    oh don't worry, it's also horrible to read my posts in french. ^_^
    how about you explain to me, or point me toward those 7µs tests? if I can get my head around an actual situation where such time difference can be noticed, maybe I'll get it. my only personal experience of minute changes being audible is when changing phase only in one channel. but it's out of context here.
    your example of a 96kbps mp3 has IMO too many other components to be a valid example. just removing so much content in the trebles as 96kbps does, will do what you talk about. we're bound to lose some sense of precision because trebles are easier to pinpoint in space than low frequencies. so by losing some audible trebles we do lose some of that pinpoint accuracy.  also the artifacts become clearly audible and must be shouting to our brain "something isn't right!!!!". 
    but at least I understand how missing some audible frequencies can do what you explain. but that doesn't apply to 16/44(at least not to my ears^_^).
  10. a1uc
    Does anyone have any suggestions ?  I would like to use the D1 Dual with headphones and to drive my mono blocks .  I dont want to keep changing XLR  Cables since there is only 1 pair of XLR outputs . In order to take full advantage of the D1 I need to use the XLR outputs  
    My Mono blocks are single ended so I use the XLR to RCA  Transformers from Totaldac 
  11. Yoga
    What's the general opinion of the d1-integrated? Which is the d1-single and server/clock in a single box if I'm not mistaken.

    Quite a lot cheaper than the mono or d1-dual + server/clock. I wonder what degree of gains you get for 1/3 or 1/2 the price.
  12. bmichels
    The d-1 intégral do NOT have a reclocker. This is also why It is cheaper than a D1 single + a D1 server ( which includes a reclocker)
    Yoga likes this.
  13. romaz

    I wish I could speak French half as good as you can speak English.  There are actually numerous studies and this threshold has been reported to be anywhere from 5-10µs.  Here is a link to one of those studies and it references others:
    Here is also a video by Hans Beekhuyzen that you might find helpful:
  14. cat6man
    and congratulations.
    i'm currently optimizing my d1-dual/bhse/009 setup and am very interested in hearing of your results.
  15. romaz
    Yes, this is somewhat of a dilemma, and you might want to base your decision on which system you listen to the most.  With the 6.35mm (1/4") jack in the back, I found the SQ to be quite good, just under powered for certain headphones.  If you intend to drive an LCD-X which has an impedance of 20 ohms, the power output from this jack will probably be adequate given its high sensitivity although you can A/B and decide for yourself.  For the HD800, that headphone port will drive it just fine.  For something like the HE-1000, I found that port to not be powerful enough.  Even though the output from that jack is unbalanced and only takes advantage of half of the ladder, I found it to be excellent.
    The alternative would be to drive your monoblocs single ended from the RCA outputs and keep your headphones connected via XLR.  Personally, this is probably what I would do.
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