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Dusty Chalk

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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.


I want to print this out and hang it on my wall.

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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).

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"19) We have recently introduced a two channel A/D converter that will output both PCM and DSD. But this converter has a few tricks up its sleeves. Specifically we have taken a page from the "What is so great about DSD" manual and applied it to PCM. And it turns out that with quad rate PCM you can get ALL OF THE SONIC ADVANTAGES OF DSD WITH NONE OF THE PRACTICAL DISADVANTAGES."


What does he mean by applying what is great about DSD to PCM? By his previous points it doesn't sound like it has any advantages over high res PCM.


Also is quad rate PCM = 24/176.4 or some other novel thing?

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