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High Rollers
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Posts posted by Filburt

  1. So...what is MQA then? The nearly-contentless descriptions in the links make it sound like a method of dynamically switching bit depth and sample rate based on analyzing the parametric requirments of the input, utilizing a psychoacoustic model. In which case, it's doing a less sophisticated form of what so-called "lossy" codecs do. I don't really care to pay AES $20 to read the white paper, though, in order to find out what they're actually up to.

  2. With modern off-the-shelf parts it really isn't as challenging to make something that sounds pretty good, than was the case several years ago, and it can be done for cheaper and in a smaller factor than was practicable back then. Just following the reference design is enough in many cases.

  3. Probably because they can report on paper that they have an analog volume control.  It's still a selling point unless you have a fair legacy in the digital realm -- Meridian, Wadia, et al.


    Ah, OK. So here's the worst part...


    Since the Sabre32 runs at 32 bit precision internally, it doesn't have to truncate in order to attenuate even a 24 bit input. I don't recall if they use floating or fixed-point math for this operation but even at fixed point one should be able to get at least around 50dB before truncation. Also note, everything will be attenuated, including the noise floor of the recording. In practice, you'll be limited by the analog noise floor of your system long before you're limited by internal precision on the DAC. Thanks to the exceptional SINAD of the ES9012/18, you could probably get at least 10-20dB of attenuation without significant loss in dynamic performance at its output (so, then, it's up to the rest of the system to keep up). On the other hand, with an analog volume control, additive noise will go up as you attenuate in addition to other potential maladies (e.g. crosstalk, distortion, and so forth).

  4. I'm not familiar with the LCD-X. I like the LCD-2 with GS-X. Not sure how similar it is to the X, though.


    Also...why did Oppo use an analog volume control on the HA-1? There is no "re-digitizing" involved using the internal ("digital") volume control on the Sabre32; it's just a multiply on the incoming data. This is more or less the sort of operation you're doing with an analog control too (well, ideally), except in practice you'll pick up some additional noise and distortion (e.g. phase and harmonic). I also don't recall pots typically having great crosstalk specs, but maybe that's changed since I last looked.

  5. I have a smaller model from Zojirushi that I received as a gift recently. So far, my bread has come out really nicely. I'd love to try sourdough, but I don't know much about how to make that one happen successfully at the moment. On another note, sourdough pancakes sounds amazing.


    Steve - very interesting info re: starters! I'd not heard that before.

  6. I'm still not particularly clear on what is supposed to be the advantage to transmitting DSD ("1 bit" PCM) over 24 bit LPCM in these cases. Why not just use the foobar plugin (or one of the other media players that plays this format)?

  7. So is there no noticeable difference to be heard with such cables, as I've read some arguing profusely that they do effect sound and also read some ridiculing the mere suggestion as here?


    Either way, if asked by cable gifter how good cable is after using it, I'm going to have to say its amazing so as not to hurt their feelings. Maybe I will hear a improvement anyway - will find out shortly :unsure:  


    The plan with your friend sounds reasonable. With regard to the more technical question, I guess it's a matter of whether you want the short(er) or the long answer. The short(er) answer is that I would estimate that the use of 'audiophile' cables that substantially depart from the design and construction of ordinary commercial-grade designs would generally correlate with reduced (rather than improved) performance if any change in system dynamics were observed at all. However, equipment destined for working primarily within audio-band frequencies, at least those with well-designed power supplies, should generally be able to handle less than ideal conditions with regard to the design of your power interconnect. So, you may be able to use the device pictured without degrading the performance of your equipment to a degree that would be noticed in ordinary use cases. As for the longer answer, I'm not sure I feel like writing it. However, probably the most germane aspect of that to note would be that the "weakest link" aphorism is apt to mislead intuition and mischaracterize the dynamics of systems, so I don't recommend it when thinking about how to improve the performance of one's audio equipment.

    • Like 1

    quote_icon.png Originally Posted by barrows viewpost-right.png
    I can understand if you are no longer monitoring this thread, but if you are I have a few tech questions on the new QB-9:

    1. I understand that you are running the ESS chip synchronously, that is with master clock and bit clock synchronous, disabling any intervention from the onboard ASRC and DPLL, right?

    2. I also understand that you are running your own OSF (on FPGA), and I would guess that this is the same set of filters as in the previous QB-9, outputting 705.6 kHz and 768 kHz to the ESS chip, with your minimum phase/slow roll off "listen" option, right?

    3. Given if my understanding above is correct, I suspect that you are using fixed oscillators at 45.184 and 49.152 as master to the FPGA and ESS chip, right?

    4. And, that the OSF in the ESS chip is off, eliminating the onboard OSF, right?

    Thanks again for sharing so much information here, I liked the previous QB-9, and I bet the new version sounds fantastic!

    Hello Barrows,

    Yes, yes, yes, yes, and I think so. YMMV. It's always best to listen for yourself especially with recording with which you are extremely familiar.



    Am I reading it wrong, is #1 referring to something independent of the ASRC that they licensed? I'm kind of a dumbass when it comes to this stuff tbh.



    The Sabre/Sabre32 has on-board asynchronous sample rate conversion as part of a coordinated system that also includes S/PDIF demultiplexing, rate estimation, and reclocking. I presume he's referring to that.

  9. I'm not entirely clear on what he means when talking about the relative amount of information encoded by DSD; the reference to Shannon (and Weaver) is somewhat opaque. Maybe he means entropy? If so, then feedback has to be factored in for DSD, since it is DPCM rather than PCM. DSD can also be entropy coded (DST) to save space. However, in the case of SACD, I imagine this is only used for dual-program discs (i.e. 2/5.1) since, otherwise, it doesn't matter much one way or another to the manufacturer (there is only one media specification, AFAIK).

  10. Based on the datasheets it looks like, at least for the PCM1792, AD1955, and ES9018, DSD is run through both the DSP and modulator. That would make sense, considering the output stage is designed for a multibit input. They also use DEM, which can be used as an additional form of noise shaping. Interestingly, only ESS indicates a filter resembling the analog filter specification from Sony.


    On the subject of complexity, I'd like to revisit the 'lowpass filter' claim, having now introduced the notion of audio as a system. The recording format in idealized terms is a means of feedforward control of the playback device by the recording device. However, this form of control requires that the playback device is adequately modeled. Sony's marketing takes advantage of the perception that lowpass filters are typically more homogenous in behavior across particular cases, and closer to ideal, as compared to DACs. In other words, they are claiming that DSD will, in practice, be a superior feedforward system compared to PCM. Sony's description, however, is incomplete. The full description is that, given an ideal voltage source, all you need is a lowpass. This claim, however, is true of any digital audio system; PCM, DSD, or otherwise. Likewise, the truncated claim is false - SACDs are not voltage sources, nor is a format in and of itself. In the case of DSD, the modeled device is an ideal switch; the noise shaping in DSD is modeled for that ideal. What is the closest thing we have, in practice? The same thing as in the case of PCM - a DAC :)

  11. The point regarding filters isn't that you can hear ultrasonic noise; it's about avoiding audible intermodulation of noise with your passband. Ideally, the noise is uncorrelated but in practice cyclical errors are likely to show up in a 7th order 1-bit modulator as you can't employ dither. One would have to find out experimentally if this turned out to be audible. Digital filtering is already used in DSD playback; you can see that in datasheets for DACs that support DSD playback (and probably in the case of 'digital amplifiers' that support DSD playback, if they specify such things in their datasheets). However, the design of the filter may vary, particularly depending on how DSD is handled by the specific DAC. Aside from that issue, I disagree that DSD is relevantly 'simpler' except when working with very loose, largely undescriptive metaphors for the functional architecture of an audio system. That is, if one thinks that DSD permits a 'dacless' playback architecture, one has made too much of the notion of a DAC. DSD is not more of an 'analog' format than PCM, nor does it in some intrinsic sense somehow get you closer to a "pure" representation of 'the signal.' I somewhat obliquely suggested this earlier, but audio systems are probably best modeled as control systems. The 'signal' is a metaphor for an idealized reconstruction of the input; the sense in which the signal is in the format is relational to the system. Under Sony's functional diagram, all that is happening is the functional blocking is done slightly differently than CD Audio, such that the D/A conversion block is divided differently, where the front-end and back-end of a delta-sigma DAC is physically split. Sony's functional diagram isn't representative of real-world implementations of DSD playback, in any case, but this is essentially what that diagram represents.  Certain functional blocks of the total system are distributed in time and place, but they are all relatable in terms of a total system, and whether the reconstructed output of the playback device is 'pure' is only really intelligible within the confines of thinking about the system.  It makes little sense to describe a playback device, that is lacking ordinary features of the control system, as providing a more pure output simply on the basis of lacking such features. It may be a mechanically simpler device, but omitting certain processing (such as filtering, noise shaping/dither, and so forth) may effectively represent a failure in the control system and a less accurate or 'pure' reconstruction of the input. Therefore, you cannot tell that a DAC design contains superfluous parts, or is likely to adulterate 'the signal,' simply on the basis of its complexity and the extent of processing involved. You can only tell when taking into account the total system. People generally get around this caveat by inserting a tacit ceteris paribus clause - that is, all other things being equal; but that's the problem, they aren't. Design requirements are domain-specific, and simplicitly is not a particularly good universal indicium of fidelity in audio design. That doesn't mean that all complex designs are good (or that complexity itself is good); it means evaluation is domain-specific. In the specific case of DACs, simplicity often leads to a lower-fidelity reconstruction of the original recording (subject to production variables like post-processing). To the extent that DSD allows for 'simplicity,' it does in the sense that you could (in theory) use it as a control signal for what is essentially an open-loop switching amplifier, which is architecturally "simpler" in its most basic sense than an R-2R latter. However, as I noted before such a device would have abysmal performance that is not in any relevant sense 'pure' or architecturally equivalent to high-performance switching amplifiers designed for audio reproduction.

    • Like 1
  12. If you find (most likely, experimentally) that having little (if any) attenuation of noise doesn't have excessively deliterious effects, I guess you could use something like a second-order filter simply to avoid passing noise at f > 100KHz. If you wanted attenuation by the time the noise floor starts to rise (~26KHz in that graph), or even by the end of that graph, you'd need a very high-order filter. A digital filter would make this operation trivial.


    Second, my earlier point was that DSD isn't any more 'analog' than PCM, and it doesn't bring one substantively closer to 'DACless' in any practical sense, outside of marginal cases (that I'd still be tempted to call d/a conversion) that no one interested in audio fidelity would use. Also, if the 'digital amplifier' (i.e. class D) uses a multi-level modulator, it might run it through the modulator loop anyway as well (in addition to whatever else would be done in DSP). In Sony's idealized case (which is rarely the actual case) one at best is able to omit the modulator from the playback device, which systemically just means you've functionally divided the d/a process between devices. Calling this process 'DACless' is roughly as credible as my describing running decoded PCM (relative to the device in question) into a non-segmented ladder DAC as a 'DACless' process.

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