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  1. Old Comment
    abraxalito's Avatar

    The real story about closed vs open loop class D amps

    Perhaps you are assuming that because Frank shares something here that I find the same as he does? No, I've yet to experience the 'degradation' of a resistive divider and I use them to set the right levels between my DAC and amp when I'm too lazy to wind a new trafo with the right turns ratio.

    No, I don't so much mean the decays of bass (organ) notes rather the whole LF 'bloom' or ambience which sets apart holographic soundstaging from the merely decent.

    As for how low, no there's no number as I've yet to decide a measurement. I'm sure its measurable though, just requires some ingenuity. Playing back a single tone - no, that wouldn't be a worthwhile exercise as its so unlike music. The test stimulus really does need to be as music-like as possible, if you've not come across such a stimulus I suggest you search on 'MTPR'.
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    Posted Today at 01:00 PM by abraxalito abraxalito is online now
  2. Old Comment
    miklos's Avatar

    The real story about closed vs open loop class D amps

    Maybe you two have way to sensitive ears (and I thought I have to) and very high demand on your electronics.
    It is much easier for me, once i have a pair of relative good speakers ,the sound quality is depending on the recording and a media carrying it. For example I don't understand how a simple, or a set of resistive dividers can degrade the sound so much at this would bother me. Not sure of the LF ambiance either, you mean like a bass pipe decaying in a church? The "dynamic noise (noise modulation) as low as possible" How low, is there a value, a number here? How would one measure if a DAC can retrieve the low level information, or not. Is playing back a -70dBFS tone and analyzing it would be good enough, or would one need a digital storage scope to see the decreased S/N on a dynamic signal? It's a surprise to me that the Led Zeppelin I would be any good for testing audio, other than maybe amplifier power output, no acoustic instruments in it, other than the drums.
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    Posted Today at 11:03 AM by miklos miklos is offline
  3. Old Comment
    abraxalito's Avatar

    The real story about closed vs open loop class D amps

    Thanks for that Frank, looks like there's more to the story than indicated in the initial paper so another update is in order. Before I write that though I'll need to study more in-depth the different OPS modulation styles as these seem to impact the PSRR....
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    Posted Today at 05:09 AM by abraxalito abraxalito is online now
  4. Old Comment
    fas42's Avatar

    The real story about closed vs open loop class D amps

    Richard, that paper you mention in your update has a powerpoint style summary of the paper here, http://wenku.baidu.com/view/cf49390b763231126edb118f, 2/3 of the way down. Might be enough to give you the gist of it ...
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    Posted Today at 03:23 AM by fas42 fas42 is offline
  5. Old Comment
    fas42's Avatar

    The real story about closed vs open loop class D amps

    Okay, that LF ambience I would term general ambience cues, not especially in the deep bass range - and, yes, it depends on the DAC working properly, much digital playback fails because that low level information is not retrieved well enough for the ears to decode what it's meant to represent, which, as you say, is the acoustic signature of the recording space.

    That same low level information is critical in unscrambling very 'messy' or 'difficult' pop recordings - the original Led Zeppelin I mastering has this component very strongly in its sound makeup, and typically fails to impress on playback because sounds are reproduced that don't make sense - the verdict is that it's a "bad recording" ...
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    Posted Yesterday at 05:54 AM by fas42 fas42 is offline
  6. Old Comment
    abraxalito's Avatar

    The real story about closed vs open loop class D amps

    I'm fairly cool with 16bits too, but customers do demand more on occasion (for whatever reason) so I'd like to have a solution for this.

    LF noise is a fairly new discovery for me. LF ambience is on many recordings, its what gives a venue its acoustic signature. On the TDA1387 without phenomenal amounts of decoupling the low level, low freq ambience information gets masked (presumably by noise modulation). Dan (Max Headroom) alerted me to the importance of low levels of LF noise - the difference is perceived as an enveloping LF 'bloom' when its low enough.
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    Posted Yesterday at 04:49 AM by abraxalito abraxalito is online now
  7. Old Comment
    fas42's Avatar

    The real story about closed vs open loop class D amps

    Further on the DAC playing up when not fully conditioned, I'm listening to Perahia doing Mozart right now, at very low volume from switch on only a short time ago - the basic sound is very clean, good treble - but talk about noise modulation! On the right speaker a strong whistling or bzzz'ng, like someone blowing over a bottle, warbles all over the place depending upon the music; on the left, much better, like a white noise, gate switching on and off constantly - this channel has settled down much more already. The level of these artifacts modulates ferociously with the volume and content of the music - but, from a distance the music content is fine, comes through cleanly, has a distinct identity and presence separate from the these obvious distortion elements.
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    Posted Yesterday at 04:32 AM by fas42 fas42 is offline
    Updated Yesterday at 04:49 AM by fas42
  8. Old Comment
    fas42's Avatar

    The real story about closed vs open loop class D amps

    Okay, first an update on 'correct' sound - my take on that is that it's in place if I can put on a recording, any recording - and nothing about the sound bugs me, I can let go, and go with the sound emotionally - no artifact is irritating enough to distract me, making me aware that it's there - hence the "no bad recording" mantra.

    Personally, I've found that rough and ready pop and rock recordings are almost ideal for debugging a system - everything is in them which will stimulate discernable distortion artifacts, and the pattern of sounds repeats constantly through a song, making it easy to pick the 'problem' occurring. If a tough pop recording passes muster, then nearly every classical recording sails through with ease ...

    Volume controls are bad news, get rid of them! Use digital, or electronic chips, or similar - the impact of ordinary pot type mechanisms is far too audible.

    16 bits are fine, I'm happy with downsampling from hi-res if necessary, offline - I have not heard a single higher resolution recording that has made me say, "Gee, look at what I'm missing!!"

    I still don't get what you mean by LF noise - noise is low level distortion in my book, and I never separate it into high or low. That said, I'm happy with using regulation to getting LF ripple under control, at the moment.

    Any filtering I would do offline - prepare a doctored version of the track to play, to suit all the following equipment.
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    Posted Yesterday at 02:37 AM by fas42 fas42 is offline
  9. Old Comment
    abraxalito's Avatar

    The real story about closed vs open loop class D amps

    I've only been fiddling for maybe four years or so but I have ascertained a considerable number of ingredients in the recipe that works for me. I've not exhausted the details by any means but so far it has been

    a) Multibit DAC kitted out with enough caps on reference and supply to keep LF noise in check. One DAC per speaker driver.
    b) Passive I/V
    c) Steep filtering with passive filter into the band required by each driver of the speaker (2-way so far, 3-way still to try)
    d) As little active circuitry for voltage gain as humanly possible. AD815 with CCS loading and lots of caps on the rails comes closest so far to being a 'passive voltage amp'. With enough current from paralleled DACs I think perhaps this voltage amplification could be disposed of. This is an experiment to try.
    e) One amp per driver kitted out like DAC in a) with the addition of extreme HF noise decoupling from running in classB
    f) Transformers to couple between DACs and amps if they're not in the same box running from the same supply.

    Overall its a constant iteration of the basic principle of keeping the dynamic noise (noise modulation) as low as possible through attention to power supply related noise and IMD-generated noise. Frustration has been entirely optional

    Putting this another way I'm still working hard trying to break the original hypothesis that satisfying sound is characterized by having the lowest noise modulation. So far the hypothesis has stood up well in that changes to the system which have been intended to lower noise modulation have indeed resulted in improved listening satisfaction.

    A few things are on the 'still to be determined' list :

    i) the optimum place to put the volume control
    ii) How to incorporate more bits (hi-res recordings) than 16
    iii) Related to ii above - to what extent is the set-up limited by my choice of the TDA1387 as DAC?
    iv) Is there another solution to LF noise without requiring oodles of caps?
    v) Is there anything to be gained from digital filtering prior to the DACs?
    vi) Any advantage of applying the 'limit the bandwidth' principle to the LF cut off of the bass/mid driver?
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    Posted Yesterday at 01:43 AM by abraxalito abraxalito is online now
    Updated Yesterday at 02:10 AM by abraxalito
  10. Old Comment
    fas42's Avatar

    The real story about closed vs open loop class D amps

    Doing impressively well unconditionally is one of the hardest things to get happening, IME. I've felt extreme frustration over my years of fiddling, because I could only get the quality of sound I was after if enough factors were in place - and many of those required "warm up" times, hours often. This is what has made it so difficult to just give a recipe of what needs to be done, because there are so many variables, far too many. Hence, "conditional" is not a problem for me - it always has been conditional ...

    The ultimate goal is to have a complete system, which you can take anywhere, plug in anywhere, at any time - and within, say, 5 minutes it's running at near 100% of its potential. That's a pipe dream, I've never heard anything come anywhere near that - I can live with compromise!

    How the 5756 is implemented, in every area, will determine whether it's a winner or not. If it behaves in the fashion of sounding like absolute crap on turn-on, and takes 2 hours to come to life, when it becomes exceptional, I can live with that - the power used to to keep the chip trickling along when not listening to it is trivial, if talking about generating convincing sound.

    'Correct' sound will mean that the recording always sounds the same, the word "better" is the problem here, each person will have their interpretation of that. And, agreed, bass is probably the area where there may be the most "disagreement", people have different standards for such. I like intense, "tight" bass - no bass unless it's meant to be there; I've mentioned Boney M. a few times, the tracks as recorded have a bass quality which I find very satisfying, and systems nominally quite superior to what mine was at the time completely miss the boat in terms of reproducing that. Pete, the chap up the road who has a DEQX unit, with monster subwoofers, way beyond what I've ever had, has bass which sounds exactly right to my ears - it matches in tonal quality, in tracks that I'm very familiar with, in the the bass area with what I would expect.

    The running in, to my mind, is largely about capacitance behaviour stabilising - the storing and releasing of charge. The S-D is all about about switching on and off of a lone capacitor, the single bit of its design. That capacitor has to be as "perfect" in behaviour as possible, otherwise there will be some type of distortion - and since capacitance is ultimately a materials behaviour exercising it causes the characteristics to change. Also, how good are the switches, how perfect are they? Are waveforms of the transitions always identical, or do they settle down in behaviour over time?
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    Posted Yesterday at 01:10 AM by fas42 fas42 is offline
  11. Old Comment
    abraxalito's Avatar

    The real story about closed vs open loop class D amps

    Ah I'm not interested in 'doing impressively well' with such a qualification. If it does impressively well unconditionally then my ears'll prick up.

    One of the reasons I like the 5756 better than the TPA3116 is they don't seem to have included the on-chip regulator for the analog signal processing. To my ears that LDO reg craps out the bass performance, it must be too noisy at LF (no surprise really, its a CMOS/DMOS process). Seems not to be a reg on the 5756 as the power pin is brought out directly.

    Your 'correct' sound in the way you've talked about it seems to have only one dimension. My sound though has dimensions even once the speakers disappear. So I still hear that for example the classD amp's bass sounds better than the classB - does this mean both create 'correct' sound? Or is classAB bass 'incorrect' and just needs a few more farads of capacitance?

    Do you have any idea what changes on a digital chip as it runs in? Any hypothesis?
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    Posted 11th July 2014 at 11:57 PM by abraxalito abraxalito is online now
  12. Old Comment
    fas42's Avatar

    The real story about closed vs open loop class D amps

    Yes, in a closed loop, digital input unit how is the feedback actually done? Probably there is some explanation out there, but I'm being lazy at the moment ...

    Yes, the 5706 is mentioned several times on diyAudio, and generally given a thumbs up for doing impressively well, subjectively, for what it is.

    If the 5756 innards are really intelligently thought out then it may do remarkably well, there being no room for interference to creep in via sloppy, conventional implementation. Personally, as I've mentioned many times, the S-D DAC is not a problem, after switch-on conditioning always sorts that out, IME - if aiming for optimum sound in, say, 5 minutes from power up then just add extra, peripheral circuitry to keep the chip nicely simmering in the "off" times.

    Interestingly, in the my PC setup which uses a cheap Realtek onboard DAC, you can easily hear this process happening. It's configured now to run at maximum analogue volume always, the digital volume control is the gain setting. And from a cold start you can hear the DAC glitching and stumbling, the digital artifacts are obvious, as if it had only about effectively 6 bits working. Steadily, as the chip runs in, warms to its work, , these artifacts fade away and finally completely disappear, even with one's ear to the speaker.
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    Posted 11th July 2014 at 11:44 PM by fas42 fas42 is offline
  13. Old Comment
    abraxalito's Avatar

    The real story about closed vs open loop class D amps

    You are asking if the feedback for the 5706 is digital? Meaning there is no internal ADC ? The DS doesn't give enough detail to work out how they do it. Where have you learned about the SQ? On DIYA?

    Yes I found the 5756 and mentioned it on the TPA3116 thread. No-one else has commented on it so far but I agree it could be great (even ignoring the DAC). Yes its different from the 5706 as there's an analog classD amp, not a digital-to-PWM converter. This tells me that most likely even they don't like what they've done in the 'direct-to-digital' chippery and are reverting to good old analog classD.....
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    Posted 11th July 2014 at 01:39 PM by abraxalito abraxalito is online now
  14. Old Comment
    fas42's Avatar

    The real story about closed vs open loop class D amps

    That feedback is digital, is it not? That's a whole different ball game, how does the chip do its tricks internally is the question - I note generally positive comments about SQ, in spite of very uninspiring distortion figures ...

    And, I also just discovered the TAS5756, which I note you have also come across. This might do quite spectacular things if handled well - is this done in a different way from the TAS5706? Plenty of scope here for fine detail tweaking - everything done in one tiny box, erh, chip might just work nicely if fed with very clean power and signals.
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    Posted 11th July 2014 at 01:07 PM by fas42 fas42 is offline
  15. Old Comment

    D/A

    I like very much the cuted Aya pcb ... and the ??? on the top of the caps .
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    Posted 9th July 2014 at 02:17 AM by Eldam Eldam is offline
  16. Old Comment
    wintermute's Avatar

    Amplifier-centric two-way XO (and system) design

    Thanks, that confirms what I was thinking. I shall scale and do some sims with a more friendly load

    Tony.
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    Posted 7th July 2014 at 08:11 AM by wintermute wintermute is offline
  17. Old Comment
    abraxalito's Avatar

    Tweeter DAC

    Accidental, thanks for the heads up! I will go investigate now.... Found an unticked box, its ticked now. Let me know if you get problems.
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    Posted 5th July 2014 at 02:53 PM by abraxalito abraxalito is online now
    Updated 5th July 2014 at 03:00 PM by abraxalito
  18. Old Comment
    fas42's Avatar

    Tweeter DAC

    Richard, you appear to have disabled comments for your latest blog, the XO-centric setup - accidental or intentional ... ?
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    Posted 5th July 2014 at 03:59 AM by fas42 fas42 is offline
  19. Old Comment
    abraxalito's Avatar

    Amplifier-centric two-way XO (and system) design

    When planet10 and I got into a conversation about PLLXOs some time ago he wasn't hungry enough (it seemed to me) to be prepared to make them work into a low enough impedance. I got the distinct impression he wanted a PLLXO that worked into 100k load. In which case, sure he'd have desperate problems with getting high enough Q. If your load is just 1ohm, then 0.1R is about 1dB droop already. I have a filter on my blog which has a 0.93ohm load and doesn't suffer droop but its made from custom ferrite bead inductors. As a general rule of thumb for not-too-demanding filters the total DCR of all the series inductors shouldn't exceed 15% of the load impedance. My AIF with 4 inductors around 1.6R each falls within this guideline for example with a 50R load. It does still exhibit about 1dB of droop but seeing as the zero-order hold of the DAC causes droop beyond this, there's already a network in place to correct for it.
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    Posted 4th July 2014 at 11:20 AM by abraxalito abraxalito is online now
  20. Old Comment
    wintermute's Avatar

    Amplifier-centric two-way XO (and system) design

    Hi Richard, I was only simming passive to get a baseline for my active implementation of the passive filter. I wound my own air cored inductors for my passive crossovers but they are speaker level not line level.

    I know planet10 has commented a number of times about pllxo's getting "droopy" when cascading more than one order, and my sims certainly confirmed that.

    I wasn't using smd so was putting in 6nH for esl for caps (which is the spec of the rifa MKP1387's I'm using. for coils even 0.1 ohms is enough to cause some serious deviation from ideal, I was quite surprised.... In my case though I was simming with 1 ohm load since that is the "normalised" filter for doing the active calculations... perhaps I should have used a more realistic load and then scaled the component values appropriately....

    Tony.
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    Posted 4th July 2014 at 08:26 AM by wintermute wintermute is offline
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