Comment by cmer
14 years ago
There's a lot of scientific-sounded content in this, but unfortunately most of it couldn't be further from the truth. I'm an ex-audio engineer and studied digital and analog audio engineering; this has been debated to death over the last 15 years.
Digitally recording a triangle is the best example of why 48kHz is very limiting. The distinct sound of the triangle constitutes of a high fundamental frequency, ballpark 5kHz and of many very high-pitch harmonics. Most of these harmonics are above 20kHz. The harmonics are what makes it sound like a triangle, not the frequencies below 20kHz. This is why the triangle is one of the hardest instruments to digitally record. It always sounds like crap.
In theory, it's true that the human hear can't hear above ~18kHz, but it can hear the influence of the very high pitch harmonics on a lower frequency.
EDIT: here's more data backing what I said http://www.cco.caltech.edu/~boyk/spectra/spectra.htm
EDIT 2: typos, frequency mistake
> Digitally recording a triangle is the best example of why 48kHz is very limiting
The article's about distribution, not recording. I don't think anybody disputes the usefulness of higher sampling rates when recording.
> In theory, it's true that the human hear can't hear above ~18kHz, but it can hear the influence of the very high pitch harmonics on a lower frequency.
...and 48kHz audio contains those lower frequencies.
Stripping frequencies above 20kHz negates the effect on the lower frequencies since those lower frequencies are not "modified" by the higher ones. The human hear can actually hear the very high harmonics when they're combined with a lower fundamental frequency.
For example, the human hear will hear a 30kHz frequency if it's fundamental is 10kHz. If it's played at 44.1kHz, the 30kHz frequency is gone and all you'll hear is 10kHz, not a "different sounding" 10kHz.
For example, the human hear will hear a 30kHz frequency if it's fundamental is 10kHz
You are going to have to provide me with a citation to back that up because that goes against everything I've learned and experience in 17 years of working in acoustics.
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> For example, the human hear will hear a 30kHz frequency if it's fundamental is 10kHz.
No. It won't.
I think you're referring to this: http://en.wikipedia.org/wiki/Missing_fundamental
Which, since it's a psychoacoustic phenomenon, wouldn't hold true when the partials involved are above the audible spectrum.
I know where you're going with this I think, and I'm not disagreeing outright, but wouldn't this be captured during the high-bitrate (or good analog?) recording and mixing phase if the recording/mixing/mastering engineer were doing things right? At least, as well as possible?
> The article's about distribution, not recording. I don't think anybody disputes the usefulness of higher sampling rates when recording.
Didn't read the article, so commenting out of context, however it needs to be said that in sample-based music genres the distributed music gets used as if it were a recording. Maybe then it could be argued that higher sampling/bit rates should be available, if only for those who are sampling.
> In theory, it's true that the human hear can't hear above ~18kHz, but it can hear the influence of the very high pitch harmonics on a lower frequency.
That may well be true. But those mixed-down harmonics that are heard "live" would then be captured by the 16/44 (or whatever) sampling. IOW, the recording captures what you heard. Those upper harmonics have no emergent properties. Their effect is captured.
Bob
I'm no sound engineer, but as far as I can tell, the main point of that paper is that some instruments produce harmonics at frequencies greater than 20kHz, not that these frequencies matter to humans. However, section X references other papers that apparently make this claim.
Just because it is difficult to record a triangle does not necessarily mean it is impossible to accurately recreate the sound (to human ears) using 48kHz.
> I'm no sound engineer, but as far as I can tell, the main point of that paper is that some instruments produce harmonics at frequencies greater than 20kHz, not that these frequencies matter to humans. However, section X references other papers that apparently make this claim.
Yes, you're right.
In fact, some of the section X references don't even mention hearing, they talk about "alpha-EEG rhytms" (in this case "listeners explicitly denied that the reproduced sound was affected by the ultra-tweeter") and "bone-conducted ultrasonic hearing" trough the "saccule" ("organ that responds to acceleration and gravity and may be responsible for transduction of sound after destruction of the cochlea").
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In fact, most of the claims of the article are around the fact that there is energy over 20khz and how it can affect recording process.
This is a well known fact, and this is exactly why engineers filter out sub-sonic and super-sonic frequencies, especially today: stuff that you can't hear (or feel) will just suck your headroom and make you lose the loudness war.
The only "good sounding" triangles you'll hear are those buried in a mix. Alone, it always sounds weird and "muted".
EDIT: Listen to the triangle at the beginning of Rush's YYZ. It's an old recording, but it sounds significantly worse than the analog version. It's been digitally mastered some time ago so if it was mastered today, it would probably sound better, but still not great. I heard a rumor that Rush is remastering all their albums "for iTunes" at the moment, so hopefully we'll be able to compare soon!
Not a very good example, because that's a Crotale (A Flat, ~4" cymbal, basically), not a triangle.
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Yes, but our ears only hear 20Hz-20KHz. So, according to Nyquist theory, you can recreate the entire signal that the human ear hears by recording those sonic artifacts that result from interference between supersonic harmonics.
So while it's true that the human ear can't hear well above ~18KHz, and the interference between high order harmonics are audible, it's also true that a properly recorded signal, sampled at 44.1KHz, oversampled, and filtered, can reproduce the exact signal the human ear is capable of hearing. At least according to theory.
The human ear is capable of detecting sound pressure as well as sound intensity, and while playback of the interference between harmonics can be reproduced faithfully in the sound intensity realm, the sound pressure levels will differ, and it is theorized that people may be able to tell the difference between the two. However, as far as I am aware, nobody has been able to demonstrate this reliably in practice.
What about sound outside 20-20k that affects us via mechanisms other than being directly sensed in the air by our ears? For instance, consider frequencies below 20 Hz that we can feel with our feet as vibrations in the floor, instead of hear with our ears? Or what about the possibility of sound above 20k causing a vibration in something other than our ears, which could have a subharmonic in 20-20k that gets conducted to our ears via bone?
I'd prefer recording technology to err on the side of capturing what we need to reproduce all of that, even if we aren't sure that we need it.
>I'd prefer recording technology to err on the side of capturing what we need to reproduce all of that, even if we aren't sure that we need it.
Again, this article is about distribution, not recording.
Nyquist is true for static signals. Music is not static. Brick wall at 20 kHz and get audible phasing artifacts! (Even if your filter is phase linear)
I'm an audio engineer, too, and I agree that this has been debated to death. And I agree that frequencies above the threshold of hearing are more important than standard dogma (based on Nyquist theory combined with Pure tone audiometry) allows. It helps explain how audio gear with a 100kHz bandwidth sounds clearer than gear with a 20kHz bandwidth even when they measure the same in the audible band.
Have you read the Audio Technology magazine interview with Rupert Neve?
Greg Simmons: Geoff Emerick, the famous British Producer ?
Rupert Neve: Yes, he started me off on this trail. A 48 input console had been delivered to George Martin's Air Studios, and Geoff Emerick was very unhappy about it. It was a new console, made not long after I had sold the Neve company in 1977. George Martin called me and said, "please come and make Geoff happy, while he's unhappy we can't do any work".
They'd had engineers from the company there, and so on. The danger is that if you are not sensitive to people like Geoff Emerick, and you don't respect them for what they have done, then you are not going to listen to them. Unfortunately, there was a breed of young engineers in the company ( I hasten to say this was after I sold it !) who couldn't understand what he was bitching about. So they went back to the company and just made a report saying the customer was mad and there wasn't really a problem. Leave it alone, forget it, the problem will go away. They were acting like used car salesmen. I was very angry with it. So I went and spent time there, at George Martin's request, and Geoff finally managed to show me what it was that he could hear, and then I began to hear it, too.
Now Geoff was The Golden Ears - and he still is - and he was perceiving something that I wasn't looking for. And it wasn't until I had spent some time with him, as it were, being lead by him through the sounds, that I began to pick up what he was listening to. And once I'd heard it, oh yes, then I knew what he was talking about. We measured it and found that in three out of the full 48 channels, the output transformers had not been correctly terminated and were producing a 3dB rise at 54kHz. And so people said, "oh no, he can't possible hear that". But when we corrected that problem, and it was only one capacitor that had to be added to each of those three channels, I mean, Geoff's face just lit up ! Here you have the happiness/ unhappiness mood thing the Japanese were talking about.
copy here: http://poonshead.com/Reading/Articles.aspx
The article doesn't suggest only using 48kHz for recording and mixing. I don't think the author would disagree that recording triangles is difficult. He would argue that once you've decided what final audible frequencies you want to present to the listener, the best way to distribute them is at 16-bit 44.1/48kHz. It's a compelling case.
What if you want to sample the song later?
That's one thing I find concerning with the move to digital. With analog media, you can go back, re-record and get an improved result (provided the source is good) but District 9 (which was shot on Red One) will never have improved quality other than resampling because the source is set to a particular digital format with associated data quality.
There seems to be some strange idea that analogue means 'infinite detail'. In this particular case, there's no significant difference between being limited by the original digital recording resolution and the grain size of a film recording.
"[...] provided the source is good" is begging the question; it's no different from saying "District 9 could be better if they hadn't recorded in 4k (or whatever the Red One was using) and downsampled it for my DVD" The nature of the source is irrelevant, barring the fact that film might provide a higher resolution, if film scanning technology increases, and you can afford to both capture on film, process and store your film properly (archiving film is rather difficult, I believe), and get the best quality digitisation possible.
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I completely disagree with the article having heard the difference many times myself. You can't record at 192kHz and hope to keep the same quality by distributing the final mix in 44.1kHz. It just doesn't work that way.
Well there is also the aliasing in that resampling. Recording at 192 for shipping at 48 makes more sense than shipping at 44.1 surely? Some audio seems to do 88.1 but rarely 176.2.
Would you like to post double-blind test results?
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Hey cmer, thanks for posting
I don't think I understand quite what you're saying and wondered if you could explain more. You and the article both say that humans can't hear above about 20kHz. If there are higher frequencies that create a harmonic at a lower frequency (e.g. a 33kHz harmonic that produces a sound at 16.5kHz) then surely that lower harmonic (16.5kHz in this case) will be recorded by the original recording equipment assuming it is recording at a frequency at least twice that of the highest audible frequency (let's say that this would be 48kHz, although there might be other DAC-related reasons to go higher).
I'm possibly being very daft here!
Let's make things super simple. Let's say you record 4 sine waves at a 192kHz sampling rate: 15kHz, 30kHz, 45kHz and 60kHz. All 4 frequencies will be captured and the 15kHz frequency will sound different to your hear because its harmonics.
If you take this recording and master it for a CD (44.1kHz), you'll effectively get up to ~20kHz (since they're a low pass filter starting at around 16-18kHz). This means that only our first frequency will be captured: 15kHz. It will be exactly the same as if you only recorded 15kHz alone. The harmonics don't modify the fundamental frequency, they just trick the human hear. But when they're gone, they have no effect whatsoever.
Hope this helps!
EDIT: the frequency numbers I used are actually somewhat of a bad example. Harmonics are never exactly double, triple the fundamental. Those would be mostly inaudible. But you get the idea.
I don't think I understand how it could sound different to my ear. My understanding is that my ear doesn't have the sensory equipment to detect signals above ~20kHz - this is what I was told at university, and a decent trawl of the web suggests this is still true. If there is any sound that is in the range 20Hz-20kHz then why doesn't the microphone pick it up?
Or am I wrong, and the ear is able to detect frequencies above 20kHz?
> Harmonics are never exactly double, triple the fundamental. Those would be mostly inaudible. But you get the idea.
Actually, they are: https://en.wikipedia.org/wiki/Harmonics
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> The harmonics don't modify the fundamental frequency, they just trick the human hear. But when they're gone, they have no effect whatsoever.
This is the part I really do not understand... either my ear CAN pick up those frequencies, maybe the harmonics are "tickling" the little hairs inside my cochlea and ultimately the frequencies I can actually hear were altered in my perception that way - or I can not hear or sense the harmonics and they physically alter the "original" wave that I end up actually hearing.
Either way, pretty much the exact same thing should happen in a studio microphone. Those all do have frequency limitations and AKG, Royer, Rode, Shure, Sennheiser, Audio Tech, what-have-you pretty much all go up to 15kHz or 20kHz according to specs, if I understand them correctly, but not further than that. If it isn't even recorded, those frequencies I also cannot hear can NOT alter my perception so they HAVE to somehow change the frequencies I can hear and are being recorded... on top of that you are making "room" for frequencies up to, say, 60kHz but I very strongly doubt your mics can go even remotely that high.
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?
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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.
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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.
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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.
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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.
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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.
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That data doesn't back your point at all. That data concerns what frequencies are present, not what frequencies can be heard.