Comment by leephillips
14 years ago
The linked article was accurate. You are confused.
"I'm an ex-audio engineer"
Hard to believe.
"The distinct sound of the triangle constitutes of a high fundamental frequency, ballpark 10kHz"
That's a pretty high note - higher than the top key on the piano. But an "audio engineer" would know that.
"many very high-pitch harmonics"
Since the next harmonic after the fundamental would be at 20khz, which only young people can hear, and none of the others are audible to any human, I don't understand what you are talking about.
"Most of these harmonics are 20kHz."
OK, you don't either.
"it can hear the influence of the very high pitch harmonics on a lower frequency."
Sure....
You clearly have little to no musical background, and think that your basic math skills are a substitute. The overtones present in a cymbal or triangle are not straight multiples of the fundamental, they are chaotic, and are very important in determining the timbre. Anyone (and I mean that) can easily tell the difference between a cymbal with and without a low-pass filter with the threshold around 22kHz, because these "inaudible" frequencies are lost.
If anyone can hear it, then surely it must have been verified through a double-blind test. Can you provide a citation?
I don't know of any to point you to. They probably exist, but I haven't read them. Let me know if you stir some up.
This is a much more polite response than what I had in mind. Better this way I guess :)
The undertones created by the high overtones are realized in the anti-aliasing filter during recording. That's not actually the reason 44 kHz sampling isn't enough.
1: He said "harmonics", not overtones. 2: You can not hear inaudible frequencies. Because they are inaudible.
1: You're still wrong. One person's typo is not a slight against physics. 2: That's what "sarcastic quotes" are for.
Playing Devil's Advocate...
The statement that frequencies above 20kHz don't matter rests upon the assumption that the ear is linear. If the ear is not linear (I don't know whether it is not not) then frequencies above 20kHz will matter, as the ear will be able to mix higher frequencies down to less than 20kHz. For example, if we have frequencies of 56kHz and 59kHz, the ear MIGHT be able to discern a difference frequency of 3kHz. No doubt this effect could be reproduced by signal with a sampling rate of 44.1KHz, but only if the analogue systems, before the sampling stage, reproduce any non-linearity in the human ear.
Incidentally, you can get speakers that create a localised beam of sound, that the person sitting next to you cannot hear. They work by transmitting frequencies above the audible range. These high frequencies can be beamformed by a relaitively small speaker array, so the sound is localised. They then rely on the non-linearity of the ear (or maybe the air around the ear?) to mix the ultrasonic frequencies down to audible frequencies. I guess there must be non-linearity in the human auditory system!
On the subject of 24-bits my understanding is that 16-bits is adequate, provided the levels (scaling) are set correctly in the recording. What 24-bits delivers is the ability to do a crappy job of the mixing, and still end up with the full dynamic range of the human ear. 24-bits is probably a temporary solution though, as manufacturers will engage in the usual Loudness War [1], and push the signal to the top of the dynamic range. Before long 24-bit audio will be equivalent to 16-bits (since the 8 least significant bits will be unused) and the next big thing will be 32-bit audio.
Having said all that, I'd guess that the speakers will be the limiting factor in most sound systems, not the recording format.
[1] http://en.wikipedia.org/wiki/Loudness_war
Nonlinearity of the ear is thought to be the explanation for sum and difference tones, which most certainly exist: https://en.wikipedia.org/wiki/Combination_tone.
> Having said all that, I'd guess that the speakers will be the limiting factor in most sound systems, not the recording format.
Yes. And DACs, which normally have filters too.
Yes, though I tend to think of the reconstruction filters as being part of the recording format.
Here's an interesting article:
In 1975, the Canadian Broadcasting Corporation was using a head shaped microphone, which was presumably an attempt to reproduce the non-linearity of the ear. It would be interesting to do such experiments with digital sampling.
Thinking about it, if every person has a different non-linear response, in theory the only way to reproduce sound beyond a certain threshold of fidelity would be to reproduce the ultrasonic components, so each person would hear their own non-linearity. (That would be beyond what I can hear or care about, but it would be fun to play with. Beyond a certain level we also get to the point where we need to ask what it means to hear a sound.)
I disagree -- in most sound systems, the room is generally the most limiting factor.
Pardon the reductio ad absurdum, but would you prefer to listen to $1,000 speakers in a dry, padded listening room, or to $100,000 speakers in a tile bathroom? Obviously the room matters; I think most people underestimate by how much.
Probably. I should have left it at "it's not the recording format" and not nominated a limiting factor.
I'd take the bathroom, given that my singing voice sounds less worse there! :-)
Another "ex audio engineer" here, you can believe or not at your leisure. Many hours spent in high-end recording and mastering environments.
I'm not sure what your background in audio is, but everything he says is correct. High end frequencies well past 15k and up (22.1k actually) are widely acknowledged to influence the lower frequencies and play a huge role in the perception of the quality of a recording. This is an old debate with pros and cons on both sides, but in general you'll find the "Golden Ears" mastering engineers (Stephen Marcussen, Bob Ludwig, etc.) come down on the side of higher sampling rates.
Now, if your original recording was mastered to 16/44.1, then a transfer by way of 24/192 will probably actually hurt the recording. But if you're mastering from an original analog or high-quality digital, in my experience there's no question, higher sampling rates deliver better experiences.
I have also spent many hours spent in high-end recording and mastering environments, and it's my observation that most engineers suffer from confirmation bias just like everyone else on the planet.
I've caught engineers using L1-Ultramaximizer (or similar) to bounce a recording down to 16-bit/44.1khz as part of the mastering process, and they're always surprised when they're completely unable to hear the difference even in the most simple cover-the-screen-and-toggle-bypass test.
Audio, perhaps like the wine industry, is a vast bastion of confirmation bias and subjectivity, no argument there.
But I know what my ears hear, and IMO there is absolutely a vast different between 44.1 and 192. I'm not sure how you can even question it. Someone else on the thread was saying it's impossible to hear the difference between 16bit and 24bit. I don't even know what to say to that. It's like telling me the glass of Gallo "Table Red" you're drinking is as good as my '75 Lafite. All I can say is "cheers" and just enjoy.
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As much respect as I have for Bob Ludwig's hard won mastering skills, he also strongly believes in $n,000/foot speaker cable, which is what he has installed at Gateway. So by all means give him well deserved props, but don't assume he's an expert on all aspects of audio theory or practice.
I've always thought the most expensive speaker cable sounds a lot better... to the wallet of the salesperson.