Newsgroups: rec.audio.high-end From: prls!max@uwm.UUCP (Max Hauser) Subject: MASH, Bitstream, oversampling, etc. Date: 17 May 91 13:53:09 GMT In article <12047@uwm.edu>, smithjh@math.orst.edu (Jeremy Smith) inquires: (1) what is the MASH 1-bit D/A conversion? (2) Is this better than oversampling? Intelligent questions deserve an intelligent answer. In fact these and related questions arise regularly, so I will try a more general response. I last did so on the "MASH" topic in (I think) summer 1989, on the group rec.audio. I will limit myself to the definitions and the technical context and thus leave the field open to others to comment intelligently on personal experience and impressions of specific products containing these techniques. "MASH" is a commercial acronym to describe a class of oversampling data converters (either A-to-D or D-to-A) introduced and popularized by T. Hayashi of Nippon Telephone and Telegraph in February 1986 at an integrated-circuits conference [Note 1]. Thus "MASH" denotes a subset of "oversampling" rather than a competing technique. The acronym derives from and means "MultistAge noise SHaping." While the technique was popularized by Hayashi, he did not use the actual acronym "MASH," which came somewhat later, from K. Uchimura et al. [Note 2]. Also, the underlying idea is much older still [Note 3]. The technique now dubbed "MASH" is a series of small one-bit oversampling data converters each of which does a partial, and noise-shaped, conversion and then passes a residue (conversion error) to the next stage. The individual outputs of all of the stages are then combined (this is nontrivial) to form a composite output. That output is passed to a digital filter (for an A-to-D function) or an analog filter (for a D-to-A function). (Yes, that's right -- I've proofread it.) This brutally short explanation does not do the subject justice and may even leave unfamiliar readers with the mistaken impression that MASH resembles successive-approximation or pipelined data conversion (well-defined in the art), which it does not, differing deeply in philosophy. However it's an accurate nutshell description in that it will endure deeper scrutiny. For further details you may wish to look in one of my survey papers, such as the AES Journal January/February 1991. The technique dubbed "MASH" by NTT is also called, less commercially, "feedforward" or (less precisely) "cascaded" higher-order noise-shaping [Note 3]. It is one of several different topologies that can implement the "noise-shaping modulator" portion of an oversampling data converter. Other such topologies include multibit and one-bit pure-feedback modulators. The latter (one-bit) class are called delta-sigma in the research literature, and "Bitstream" by NV Philips in digital-audio products (these techniques also have many applications outside of audio). "MASH" data converters are themselves not truly "one-bit" data converters in a meaningful sense, as I have explained in more depth in print, although they are made up of one-bit subsections and this often causes confusion. Each of these competing modulator topologies has technical strengths and weaknesses that are very involved and do not lend themselves to reductive explanation. The signal fidelity in the "MASH" technique can be very good but depends on a different set of circuit elements than in the one-bit schemes. It is all a matter of "second-order" electrical effects; if the components are all perfect (as they invariably are assumed to be, in popular "explanations" of this subject matter), then all the techniques work well. When Hayashi introduced the "MASH" technique to the international community in 1986 (I was present), it was suggested from the floor that this method, although a viable alternative to one-bit oversampling, did introduce certain additional vulnerabilities to circuit imperfections, and moreover that the real justification for the method was not clear (Hayashi's actual motivating statement at the time was dubious). It has since been suggested, perhaps insightfully, that the sheer novelty of the "MASH" method might afford a commercial, rather than a performance, advantage since the other basic oversampling methods occupy expired patents and are therefore in the public domain. (My use of passive verbs in this paragraph is not accidental.) But I think it even more important, in the context of this newsgroup and the usual questions, that none of these considerations need correlate at all with audible differences among finished audio products that employ these data converters! Offhand I would suggest that audible quality could easily be affected far more by such prosaic outboard factors as the choice of output buffer amplifiers, and the degree of analog-digital isolation in the conversion-reconstruction circuitry, than by whether the internal modulator is realized via "MASH," or "Bitstream," or some other topology. This has not, of course, prevented legions of advertising copywriters and cult audio pundits from pontificating about the "revolutionary" aspects of this or that topology and deploying buzzwords like "noise-shaping" and "one-bit" with impressive vacuity. They apparently prefer this to focusing on the sonically and technically more vital, but less glamorous, issues. There is my terse explanation of the relationship between "MASH," "one-bit," "Bitstream," "delta-sigma," and "oversampling" data converters. Be skeptical of anyone who asserts glibly that one of these techniques is "better" than the others. You might be able to hear differences among CD players using the competing schemes, but almost certainly not for the reasons everybody talks about. Such differences could easily be due to any of the numerous important factors other than the particular choice of data-converter topology. Notes from text: Note 1: Hayashi et al., ISSCC 1986. Printed version in that year's ISSCC digest, pages 182-183. Note 2: Uchimura et al., ICASSP 1986. Printed version in 1986 ICASSP Proceedings pp. 1545-1548. Research papers quite regularly misattribute the modern origin of these circuits to this ICASSP paper rather than to the earlier ISSCC paper [Note 1] by co-workers. Note 3: The technique has existed in various forms, including a small paper in "Electronics Letters" in 1969. Candy and Temes have told me that they would cite this in their forthcoming broad research overview of oversampling. I find it still more intriguing that the technique is also a data-conversion implementation of Black's multistage linear amplifier (US patent 1,686,792, issued 1928), a classic invention actually predating the invention and patent of negative feedback, also by Black. Max Hauser {mips,philabs,pyramid}!prls!max prls!max@mips.com Copyright (c) 1991 by Max W. Hauser. All rights reserved. "A lot of things can be solved by the use of jargon -- for example, the effort of thinking, or the danger of saying something that someone else may not like. You don't have to be clever, and you're always on the side of whoever has the money or power ..." -- Stanislav Andreski, author of "Social Sciences as Sorcery" Newsgroups: rec.audio From: exspes@gdr.bath.ac.uk (P E Smee) Subject: Re: Oversampling for kids Date: 6 Jun 91 14:06:37 GMT In article <1991Jun4.174051.594@cs.sfu.ca> rosen@cs.sfu.ca (Wilf Rosen) writes: > >Could somebody please explain what oversampling really means with regard >to CD players. The only information I have been getting is from friends >who claim to know, or from (ulp) salesmen. No explanation I have seen yet >is really believable. (The most common explanation is that e.g. 4x oversampling >means the disk gets read 4 times. Somehow this seems silly). Layman's explanation follows. I guarantee it to be sufficiently accurate for your expressed needs, but to keep it simple I'm going to skip some of the technical complexities at a detail level where they shouldn't matter. First, the problem with non-oversampling players: The mechanics of the sampling process mean that you also get a spurious signal at half the sampling frequency (22.05 kHz for CD 44.1 kHz sampling); and that you also get a sort of 'mirror image' of the real audio signal, reflected around this 1/2-sampling freq. E.g. a 20KhZ note will also create an artifact at 25 kHz. These artifacts ARE beyond the range of human hearing, but they can cause follow-on side effects which impinge into the audible range, so they need to be filtered out. However, it is also very difficult (and or expensive) to build a sharp-cutoff filter which will not do nasty things to the bit of the signal which is passed through. So, how do you fix this? You increase the sampling rate. However, there are only 44.1 khZ worth of samples on the disk. What an oversampling player does is to take the samples it reads (reading the disk only once) and by use of various 'curve-fitting' types of algorithms, works out intermediate values. Like extrapolation, only fancier. A 2x oversampler works out one intermediate value between each pair of read values. A 4x works out 3 intermediates. The effect is to increase the apparent sampling rate, at which (or reflected around which) the artifacts I first mentioned occur. Thus, they can be filtered out by a much less aggressive filter, which will not mess up the desired part of the signal. (Some very high x oversampling players can actually dispense with this filter entirely, on the grounds that the artifacts have been moved to such a high frequency that they won't interfere with the final result even if they are left in.) If you're going to try to find references, a likely keyword would be 'digital filtering'. -- Paul Smee, Computing Service, University of Bristol, Bristol BS8 1UD, UK P.Smee@bristol.ac.uk - ..!uunet!ukc!bsmail!p.smee - Tel +44 272 303132 Newsgroups: rec.audio From: jpb@calmasd.Prime.COM (Jan Bielawski) Subject: Re: Oversampling for kids Date: 7 Jun 91 19:45:48 GMT In article <1991Jun4.174051.594@cs.sfu.ca> rosen@cs.sfu.ca (Wilf Rosen) writes: <