Richard at BitPerfect wrote a rather long piece on his blog and FB page called “On DSD vs. PCM…Again”, You can read it by clicking here. The catalyst for his 8000-word post was a couple of posts that I wrote about the DSD 64 vs. 96 kHz/24-bit PCM debate. You might recall that a reader over at Computer Audiophile ripped my entire post and posted it over at Chris Connaker’s site. Over the next few days, the CA community contributed 530 (it’s probably more now…I’ve dared to venture back there) comments trying to refute my contention that 96 kHz 24-bit PCM audio produces more fidelity than DSD 64.
He called me out in his article…challenging my knowledge of the “facts” and referring to my stance as “almost a point of theology”. I read his piece a few times and much of it just didn’t pass the smell test. In the first few paragraphs, he acknowledges, “technically, what he says is correct.” And then he goes on in paragraph after paragraph explaining the technical reasons why DSD tops 96 kHz/24-bit PCM.
So I sent the link along to the smartest guy I know in this regard and asked him whether my hackles were justifiably raised. Enter my good friend John Siau, a fellow PCM advocate, analog and digital expert, and principal of Benchmark Digital. I think his expertise and thoughtful responses are worthy of posting…with his permission so I’m going to spend a few posts talking about Richard’s piece, my thoughts, and John’s comments.
BitPerfect: “Another ‘fact’ is, though, that much to Waldrep’s chagrin, there is a substantial body of opinion out there that would prefer to listen to DSD over 24/96. Why should this be, given that the above technical arguments (and others that you could also add into the mix with which I might also tend to agree) evidently set forth ‘the facts’? Yes, why indeed…and the answer is simple to state, but complex in scope. The main reason is that the pro-PCM arguments conveniently ignore the most critical aspect that differentiates the sound quality, which is the business of getting the audio signal into the PCM format in the first place. Let’s take a look at that.”
John Siau This anecdotal evidence is meaningless. Very few people (if any) have ever had the opportunity to compare identical recordings delivered on PCM and DSD. There are many wonderful sounding DSD recordings that are being produced by highly skilled recording engineers who are putting a high emphasis on quality. The wonderful recordings that they are producing are a reflection of the care and skill that they put into the recording and production process. The quality is not due to the DSD delivery format. Based upon the mathematics, a 96/24 PCM dub of the DSD should be indistinguishable from the DSD original. This comparison would yield meaningful data. Anecdotal accounts of DSD sounding better than PCM are meaningless unless the same exact recording is available in both formats.
I’m not sure I understand why Richard uses quotes around the word “facts”…everything I stated in my post is true. I have no problem acknowledging that certain forces within the high-end audiophile community have succeeded in hyping DSD to the point where many make the claim that DSD is “warmer” or “more like analog”. They are certainly entitled to their opinion…just as one of the most prominent DSD supporters claimed on a panel at the Newport Show that with DSD “their is sort of an ease, a naturalness, roundness that I associate more with analog than I do with digital.” His opinions can’t be supported with facts.
I wrote a post about a German research project that compared DSD vs. PCM. The team recorded selections of music with exactly the same signal path and at the same level. You can read the post by clicking here. Here’s what they stated:
“The results showed that hardly any of the subjects could make a reproducible distinction between the two encoding systems. Hence it may be concluded that no significant differences are audible.”
But still audiophiles, writers, engineers, producers, and editors continue to insist that DSD rules.
BitPerfect: “If we are to encode an audio signal in PCM format, the most obvious way to approach the problem is using a sample-and hold circuit. This circuit looks at the incoming waveform, grabs hold of it at one specific instant, and ‘holds’ that value for the remainder of the sampling period. By ‘holding’ the signal, what we are doing is zeroing in on the value that we actually want to measure long enough to actually measure it.”
John Siau This entire discussion is irrelevant. It is a giant detour. Why take a tour of all of the A/D techniques that are rarely used for audio?
No one is using “sample and hold” methods any more! Anyone who has watched Monty’s video over at the xiph.com site knows, the values output by a current high-end ADC aren’t “held” between samples. There are no stair steps in analog to digital conversion.
Let’s stop here for today. And thanks to John for taking the time to read this article and response when Richard veers from the “facts”.