Support
Looper's Delight
|
Support |
|
|
At 5:40 PM +0100 11/27/07, Rainer Thelonius Balthasar Straschill wrote: > >DSD is a delta-sigma (1bit) conversion technology which samples at 'round >2.88MHz. So how does this work, compared to typical PCM (e.g. CD, DVD) >audio? Now, I believe this type of one-bit sampling technology has actually been around since the late 80's. IIRC, there was some sampling technology touting one-bit conversion (Yamaha TX16W perhaps?) which had ties back to the original Sequential Circuits engineering team, which was originally sold to Yamaha then Korg when SCI went out of business in the mid-80's. This is one of those many occasions where I really miss Dr. Zvonar, dammit. :( >Is it worth four times the price? Well, if you believe the marketing. I think I've got a couple of evaluation points worth considering. That's not to say they aren't addressed with current DSD technology, but I feel they're certain worth having asked and answered. Okay, I am pulling up some crap from my severely mis-fired and perhaps slightly drug-addled memory cells of twenty years ago, so take the historical part with a grain of salt. The main issue levelled at one-bit recording back in the 80's , if I remember my history, was that it had substandard transient response. For instance, if you record in 16-bits, it's possible for one value to be 0000000000000000, while its neighboring value to spike immediately to 1111111111111111 -- no necessary interpolation between. If you take that same transient recorded in DSD (one-bit) and illustrate it using the same 16-bit terminology, this would translate to an initial value of 0000000000000000, followed by another of 0000000000000001, then 0000000000000010, all the way up until you finally reach 1111111111111111. There's the potential to observe softening of the transients, which could result in a digital compression that might be pleasing or irritating. Which one, I don't know. Regardless, it is another type of digital distortion. The supposed workaround for this is that, as you pointed out, the sampling rate is phenomenally high. These days, it's certainly much higher than was ever possible back when DSD was first put out. The way it's supposed to work is that the super-fast sampling rate compensates for having to approach these bit values "one step at a time". Does it work? Again, I'm not making a value judgment here. But try to approach your critical listening tests with that *potential* fault in mind. The next possible criticism is much more modern, and dovetails into an argument put forward by Dan Lavry, of Lavry Engineering in his white paper here: http://www.lavryengineering.com/documents/Sampling_Theory.pdf Now, of those of you that don't know, Lavry Engineering makes *extremely* high-end A/D/A converters. These are some of the converters favored by people who think even the top-line Apogee's are crap. So, if you're a mastering engineer and wanna go spend in the neighborhood of five figures for top-end converters, go Lavry. In other words, IMNSHO, the guy knows his sh**. In this paper, he's explaining (for one thing) why 192KHz sampling rates are nothing more than industry hype. In fact, Dan Lavry refuses to support 192K, and even argues that it is sub-standard to a good-quality converter operating at 88.2K or 96K. Why? Because, amongst many reasons, the tolerances of modern electronics (especially mass-produced designs) can't really keep up with 192K as a stable rate. The main operative point here is "stable". Micro-fluctuations induced by the components start to really come out when they are driven at such a high rate. I think it's roughly analogous to driving a 30 watt guitar amp full-on at 30 watts ("mine goes to 11", in other words). What happens is that you end up with distortion, where the guitar waveform becomes clipped off so that it's transformed into more of a square wave at the peaks. Now, of course, on a guitar amp this distortion is pleasing, and amps are actually designed to take advantage of this. On a digital converter, however, distortions produced by having to perform near the limits of component tolerance are not nearly so desirable. This can also be backed up by some of Bob Katz's jitter tests (I'm reading his excellent "Mastering Audio" book right now). Katz found that, in evaluating clock specs, it was not really so much the ability of the clock to generate a spot-on 44.1KHz (or whatever rate) clock. Rather, what mattered more to the converters and the overall quality of the sound was the *stability* of the clock source. In other words, it didn't matter quite so much that the clock generated at a rate of, say, 43.915KHz, as long as the clock was stable at that rate and as free of clock jitter as possible. This points out again that what matters a great deal is the stability and consistency of the sampling rate, not so much the speed of the rate itself. So, bringing it back to DSD.... Now you've got a converter that is performing at 2.88MHz. Is that going to be similarly susceptible to component tolerance factors? Again, I don't know for certain, but that's got to be a factor to consider when evaluating this technology. --m. -- _____ "take one step outside yourself. the whole path lasts no longer than one step..."