The Emperor's New Sampling Rate

Apr 1, 2008 12:00 PM, By Paul D. Lehrman



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But wait a minute — haven't we all heard the superiority of high-sample-rate audio? Leaving the tweak-heads aside, there are a huge number of people in this field for whom I have real respect — Moorer among them — who have experienced high-sample-rate audio to sound more “spacious” or “detailed” or “enveloping.” You might even be one of them.

As it happens, I'm not, which is not to say I think everyone else is full of beans; I've just never experienced it in an environment that I feel was controlled enough for me to be comfortable making that kind of judgment. It's not that I'm lazy: As Meyer and Moran realized, setting up a test that could really be considered objective is not trivial. Even if I were the sole subject of the test, I'd still want lots of time, multiple music sources, incontrovertibly great equipment, an excellent level-matching system and a very quiet (and consistent) room.

I have had one experience that came close to this, but the result was inconclusive. At the press roll-out a dozen years ago of DSD at Sony's studios in New York, a group of audio writers got a demonstration of how the new system compared with a 20-bit PCM digital stream, as well as with a direct analog feed from a live band in the studio. I could hear some differences. Yet how to describe them — or whether I would hear them again in another time and place — I couldn't tell you. I did, however, mention a preference at the session for the way instrument decays sounded in PCM, to which David Smith (R.I.P.) replied, “We've heard that from others. In fact, you'd be very flattered if you knew who else said that same thing.” What the significance of that was, I guess I'll never know, but it didn't seem to get in the way of DSD ending up with plenty of fans among the recording community.

But something is causing people to say they are hearing differences. If a double-blind test can't confirm those differences, then what's going on? For one possible reason, let's go back to Moorer's paper that I quoted earlier (called “New Audio Formats: A Time of Change and a Time of Opportunity,” which can be found on his Website, Later in the paper, Moorer noted that humans can distinguish time delays — when they involve the difference between their two ears — of 15 microseconds or less. Do the math, and you can see that while the sampling interval at 48 kHz is longer than 15 µs, the sampling interval at 96 kHz is shorter. Therefore, he says, we prefer higher sampling rates because “probably [my emphasis] some kind of time-domain resolution between the left- and right-ear signals is more accurately preserved at 96 kHz.” It's an interesting starting point for a discussion, but to my knowledge it's never gotten past that point — as a theory, it has never been expanded upon or tested. And judging from the results of Meyer and Moran's experiment, it doesn't seem to be a factor.

Some folks think it's all simply wishful thinking on everybody's part: The system costs more and has better specs; therefore, we make ourselves believe it sounds better. There's something to that reasoning. Humans are a notoriously imperfect lot and tend to see and hear what we want to hear. Another very plausible reason is something that the authors discovered in their research. Despite the fact that no one could hear the difference in playback systems, they reported that “virtually all of the SACD and DVD-A recordings sounded better than most CDs — sometimes much better.” As it wasn't the technology itself that was responsible for this, what was? The authors' conclusion is because they are simply engineered better. Because high-end recordings are a niche market, “Engineers and producers are being given the freedom to produce recordings that sound as good as they can make them, without having to compress or equalize the signal to suit lesser systems and casual listening conditions. These recordings seem to have been made with great care and manifest affection by engineers trying to please themselves and their peers.”


But there's one more reason worth examining, among whose proponents is Ethan Winer — a musician, engineer, studio owner, manufacturer and iconoclast who's been in the recording business for some 40 years — who is definitely of the “show-me” school of audio theory and is an outspoken critic of “subjectivism” — that school of thought that encourages people to discuss the performance of audio components and systems using vaguely definable and often irrelevant adjectives instead of hard data. Winer's company, RealTraps, manufactures modestly priced acoustic treatment products for studios, so it's not surprising that he contends that anomalies caused by the listening space and our place in it far outweigh any possible subtleties we might be picking up when we change sample rates.

In an article on his Website (, Winer points out that in a typical room, moving one's head or listening position as little as four inches can result in huge changes in the frequency-response curves one is hearing. What could be a 10dB dip in one spot at one frequency could be a 6dB boost a couple of inches away. These wide variations are caused primarily by comb-filtering effects from the speakers and from the various reflections bouncing around the room, which are present no matter how well the room is acoustically treated. Winer blames this phenomenon for most of the unquantifiable differences people report hearing when they are testing high-end gear.

He writes, “I am convinced that comb filtering is at the root of people reporting a change in the sound of cables and electronics, even when no significant change is likely. If someone listens to their system using one pair of cables, then gets up and switches cables and sits down again, the frequency response heard is sure to be very different because it's impossible to sit down again in exactly the same place. So the sound really did change, but probably not because the cables sound different!”

The test subjects in the Meyer/Moran experiment didn't get up and move around, and so the fact that they couldn't discern any differences in the two signal paths fits nicely into Winer's theory. In fact, his response when I sent him the article was, “Nothing in here surprises me.”

Am I sure that Winer is right? No, although I think he's onto something, the way I think Moorer's thoughts about microscopic phase differences may be important in some way we haven't yet figured out. But I am delighted to read Meyer and Moran's paper for two reasons: It confirms something I've long suspected and it throws down the gauntlet for further research to be done.

Paul D. Lehrman doesn't have much frequency response above 10 kHz, but considers himself more aware than ever.

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