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High Rollers
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Everything posted by Filburt

  1. Ah, OK. I definitely wouldn't describe that as 'plenty of room'. The floor is already at -100 by 30K. If using a digital filter, it's enough room that you can keep 20K out of the transition band and still have ample stopband attenuation by 26K. However, if you're using an analog lowpass, you won't get comparable attenuation without using a high-order filter that would assuredly breach the "purity" ethic. In any case, I wouldn't put much stock (essentially none) in the claim that DSD can be converted with just a lowpass filter. If you physically attach an SACD to an analog lowpass filter, you aren't getting music on the other end. You still have to (1) obtain a code-equivalent voltage output (since the physical disc inconveniently fails to supply it outright), and (2) send that output to a lowpass filter. All digital audio, once you have the code-equivalent voltage output, can be "converted with just a lowpass filter" (hence the term "reconstruction filter"). Obtaining said output is precisely what one uses a DAC to do. As far as I am aware, real-world converters (multibit delta-sigma) run DSD through their modulator loop and DSP just like anything else, in order to gain anything like 'high resolution' performance, so Sony's block diagram is inapplicable even when one ignores the realities about how DSD is handled on the production side. I guess you could try some alternatives, although it would be difficult to avoid something functionally very similar, if not analogous, while still getting worthwhile performance. I certainly wouldn't describe a modern class D amplifier as architecturally minimalist, or analogous to what I think people interpret Sony's marketing claim to entail. Once factoring in the practical reality of taking in data and processing it, such an amplifier would include most (if not all) of the functional architecture of the aforementioned converters. The closest thing I can think of to Sony's idea would be a switch driving a filter, but this would essentially be the back-end of an uncontrolled switching amplifier; the performance is likely to be rather abysmal (and I'm not sure, in any relevant sense, 'pure').
  2. Uh...you don't get "purity" by using a mechanically simpler conversion process in digital audio. You will probably get various types of errors (noise, distortion, flatness, linearity), which are not generally considered to contribute to "purity" of the reconstructed output. I'm not sure what is meant by converting "without silicon" in this case.
  3. [Note: In the interest of brevity and helpfulness, I sacrificed precision and detail. Also I just got bored trying to correct anything, and I noticed something nearly as soon as I posted it. So...whatever at this point; probably close enough] The Scarlet Book standard (SACD) effectively forces 'better' behavior on the part of production engineers, while building in more margin for error. Anecdotally, I've heard that the peak meter on a lot of PCM recording and processing tools simply reports the instantaneous amplitude of incoming samples. Unfortunately, samples rarely fall along the maxima or minima of the incoming signal, which means analog peak is often in excess of the recorded peak value. Note that this does not mean that analog peak is unrecoverable; the familiar strictures of sampling apply. What it does mean is that the downstream playback device needs a certain amount of headroom to avoid clipping, and/or some means of rescaling the input. Historically, most consumer digital playback devices (e.g. CD players, 'DACs,' and so forth) made no such accommodations; meaning, you'll get clipping. One notable exception is that the PMD100 does rescale on the order of about 1dB, so devices using this filter do potentially have some headroom.* The development of the SACD standard took a more systems-oriented approach and ultimately imposes a measure of conservatism in attempting to maximize peak and average amplitude. It restricts what can pass as "legal" data for the purposes of producing an SACD; that is, it imposes a concrete penalty (the SACD won't go to press) if one passes illegal data. Chiefly, it restricts peak modulation depth, as measured by an unweighted moving average of uh...I think it's 28 samples (I don't have the specification in front of me so I'll have to go by memory). A sort of industry 'rule of thumb' for DS modulators is that they are adequately linear up to about a 50% modulation depth, and 50% is what was ultimately chosen for 0dBFS. In practice, acceptable linearity tends to persist somewhere in excess of 50% MD for most modulators in commercial devices, while the SB restriction works out to (IIRC) something like 71% or about +3dBFS. So, some amount of headroom above 0dBFS can be built-in on the playback side if following the specification, though note that this comes at some cost to maximizing e.g. SNR specification at 0dBFS, so I suppose it isn't guaranteed that a manufacturer will opt to do so.** Secondly, a simple unweighted average in this case is going to integrate out of band noise and as a result will overestimate analog peak, with this effect tending to increase with the use of dynamic compression. Third, 0dBFS is still the canonical target peak, on the presumption that downstream headroom will not be built-in. If you combine these factors, what you get is some reduction in the likelihood of clipping on the playback side (though the degree to which this is the case will vary between devices), due to (1) the potential of violating the restrictions when following popular conventions in PCM recording, and (2) likelihood of having some measure of headroom built into the playback system. So, given these considerations, why did I express the prior opinion in the other post? Because sensible restrictions can be built into a PCM recording system (and safeguards in the playback system), too, and result in superior performance. *Another example would be a 'NOS' R-2R DAC in cases where the reconstruction filter has adequate headroom. I don't recommend this solution, though - such DACs do not adequately filter aliases, which in turn intermodulate with the passband to produce in-band distortion and noise. **Nor are manufacturers necessarily forthcoming about whether they opted to do so or not. Note that the ESS Sabre's modulator is linear beyond 71% MD and, as I recall, this particular converter does build in headroom although, so far as I understand, it does so for both DSD and PCM data.
  4. Heh...I was typing up a post on this topic, but it looks like that is largely superfluous now. I also didn't want to really belabor the point - as Dusty noted, these discussions are generally not geared towards putting others' feet to the fire and aiming towards consensus. In any case, Siau covers a lot of the issues with DSD (though, not all of the issues...) and I agree with him regarding PCM being a more favorable archival format.
  5. Hmm...why ES9016 instead of 9012/9018? Product differentiation (from higher-end)?
  6. OK. Looks like an LME part (maybe 47920 or 49990) on I/V, NP0 caps, SMD (thin film? I hope at least that...) resistors. Not optimal, but better than what I've seen in some other cases. Great form factor, though.
  7. What is the output stage design on the DAC 2?
  8. Maybe he needs a $1000 USB cable
  9. Is anyone going to be around on Sunday?
  10. Filburt


    Happy Birthday, Al!
  11. I've generally found that I can taste differences between vodkas, though I don't know much about describing those differences. I've tried maybe 15 or so. OTOH I think Luksusowa was my favorite, though it's been a while since I've had vodka.
  12. Happy birthday! Hope it was a good one (sorry about lateness >.<)
  13. What the review lacked in detail it made up for in vagueness.
  14. The impression compares substantially to a notable quantity of reviews I've seen on many sites.
  15. Looks like it needs to be resoldered. If you are experienced at soldering can do it quickly, I'd reflow w/ a bit of new solder then wick it out, and then resolder. However, you can probably just reflow w/ a bit of solder and go from there.
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