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Author Topic: A question  (Read 12188 times)

kraster

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A question
« on: June 19, 2005, 02:09:45 AM »

I suspect this has probably been covered a million times before but I'm still confused. I recently read this following excerpt from an article from Apogee concerning recording ultrasonics:

"Why Record Ultrasonics?
As is widely recognized, most of us can ’t hear much above 18 kHz, but that does not mean that there isn’t anything up there that we need to record – and here's another reason for higher sampling rates. Plenty of acoustic instruments produce usable output up to around the 30 kHz mark – something that would be picked up in some form by a decent 30 in/s half-inch analog recording. A string section, for example, could well produce some significant ultrasonic energy.

Arguably, the ultrasonic content of all those instruments blends together to produce audible beat frequencies which contribute to the overall timbre of the sound. If you record your string section at a distance with a stereo pair, for example, all those interactions will have taken place in the air before your microphones ever capture the sound.You can record such a signal with 44.1 kHz sampling and never worry about losing anything –as long as your filters are of good quality and you have enough bits.

If, however, you recorded a string section with a couple of 48-track digital machines, mic on each instrument feeding its own track so that you can mix it all later, your close-mic technique does not pick up any interactions.The only time they can happen is when you mix – by which time the ultrasonic stuff has all been knocked off by your 48 kHz multitrack recorders, so that will never happen. It would thus seem that high sampling rates allow the flexibility of using different mic techniques with better results
."


Now I appreciate that most Mics won't be able to record 30khz but for arguments sake let's say we have some mics that do. Are the beat frequencies referred to in the article caused by non linearities such as the air and the ear and must we actually hear the original frequencies in order to hear the Beat frequencies?
In a sentence: Is there any truth in this article or is it more Voodoo?

Thanks,

Karl Odlum
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David Satz

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Re: A question
« Reply #1 on: June 19, 2005, 06:41:51 PM »

Karl, there are some possibly valid arguments in favor of wideband audio electronics and sampling rates higher than the bare minimum, but the statements that you've quoted here aren't among them.

Audible "beat" tones are a phenomenon that occurs in a listener's ears, not in the air of a hall. Thus these tones don't occur at all if either or both of the original "pure" frequencies lie beyond the listener's hearing range. A 50 kHz tone plus a 51 kHz tone, for example, even at very high sound pressure levels, won't produce audible 1 kHz difference energy unless there is non-linear distortion in the playback equipment.

--best regards
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kraster

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Re: A question
« Reply #2 on: June 20, 2005, 09:01:23 AM »

Thanks for that David. So the statement I quoted is a load of baloney? I always assumed in order to hear "beat" tones that one had to hear the original frequencies that cause the beating and the ear subsequently distorted them. Quotations like the one from apogee just serve to confuse matters.

Karl Odlum
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David Satz

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Re: A question
« Reply #3 on: June 20, 2005, 12:20:51 PM »

Karl, what I said was the simplified version. Air itself can be driven into non-linear behavior--at enormous sound pressure levels which can cause instant, traumatic physical injury or death. But musical performances, as heard by audiences at ordinary listening distances, probably never have anything above 20 kHz that gets within, say, 80 dB of such levels. Admittedly I may not know everything the kids are listening to these days ...

--best regards
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dcollins

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Re: A question
« Reply #4 on: June 20, 2005, 07:59:35 PM »

kraster wrote on Mon, 20 June 2005 06:01

Quotations like the one from apogee just serve to confuse matters.



Maybe someone from Apogee should come on here and explain what they are talking about, because I think Mr. Satz is 100% correct.

And it's not even controversial...

DC

kraster

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Re: A question
« Reply #5 on: June 21, 2005, 05:36:36 AM »

The fact that it states that it can improve your close Micing techniques is the biggest 'Red Herring' there as most Mics won't capture these ultrasonic frequencies and most speakers won't reproduce them. And even if they did we still wouldn't hear them. So the sample rate is a moot point.
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David Satz

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Re: A question
« Reply #6 on: June 21, 2005, 12:37:26 PM »

kraster, I'd nominate "arguably" as the biggest weasel word in the Apogee quote. Taken literally, it's a boast: "I can talk in a way that seems to make sense, as long as it never has to be tied to any actual reality." Then, unfortunately, they live up to their boast.

The thing is, an idea isn't necessarily wrong just because someone has tried to use a bogus argument in its favor. For example, you're right about the bandwidth limits of most microphones and speakers, but there are exceptions. And the way something is limited to a particular bandwidth can be more important than the bandwidth itself, as far as audible transparency is concerned.

There really are some other possibly valid arguments in favor of audio circuitry with wider (within reason) bandwidth than we can hear, or sampling rates higher (within reason) than 44.1 kHz. I won't go into them here, but they are for strictly practical reasons in particular situations--not because "wider bandwidth sounds better." The latter claim is widely believed by audiophiles, and it's the kind of statement which can't ever be disproved, so they go on believing it. But there hasn't been any proof of it in all these years, either, and one would think that it could rather easily be proved if it were true.

--best regards
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danlavry

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Re: A question
« Reply #7 on: June 21, 2005, 06:04:40 PM »

“Thanks for that David. So the statement I quoted is a load of baloney? I always assumed in order to hear "beat" tones that one had to hear the original frequencies that cause the beating and the ear subsequently distorted them. Quotations like the one from apogee just serve to confuse matters.

Karl Odlum”


Hi Karl,

Yes it is a bunch of baloney.

1. The "beat tones” do NOT occur when adding musical material with LINEAR summation. A proper addition IS A LINEAR processes, be it circuit or software.

2. To have “beat tones” one must have a NON LINEAR processing, and the outcome is very non musical. Say for simplicity sake you have some instrument A playing 1KHz and its harmonics (2,3,4,5….30KHz), and instrument B playing 1.3KHz (nearly a third chord) with harmonics (2.6,  3.9…. 28.6…29.9KHZ). Do you want to have the difference of say 28.6 and 30KHz (it is 1.4KHz) in the audio? And at the same time also have 29.9-30KHz (which is 100Hz)?... In fact by the time you have the various sums and differences the beats are all over the place, and your best cure is LINEARITY thus no beats.

3. If you have beats, you have non linearity. Is the non linearity restricted to high frequency extension? If not, then the beats will occur with low frequencies and real mics and ears, and that is bad news, counter to transparency.

4. Say the non linearity is restricted only to signals above 20KHz (magic, is it not), then the Apogee argument advocates taking high frequency harmonics that we do not normally hear in live performance, and throwing combinations of sums and differences of those ultrasonic back into the audio band we do hear…

When I see such “educational” material, I wonder. I wonder about the caliber of the “educators”…I wonder about the motivation to writing such stuff...

Regards
Dan Lavry
www.lavryengineering.com

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kraster

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Re: A question
« Reply #8 on: June 22, 2005, 07:06:35 AM »

Thanks for your reply, Dan. I had already assumed what you said in your post was the case until I read the Apogee article I quoted which left me confused. I then assumed what I had learnt before was wrong because Apogee know what they're talking about and make some decent converters.

This is pretty unfair on the average Joe (myself included). I and many others make purchasing decisions based on this kind of info. Their (apogee's) argument at first glance seems plausible. But they're twisting the facts to suit their own ends.

Karl
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Max

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Re: A question
« Reply #9 on: June 22, 2005, 01:09:26 PM »

kraster wrote on Sun, 19 June 2005 07:09

I suspect this has probably been covered a million times before but I'm still confused. I recently read this following excerpt from an article from Apogee concerning recording ultrasonics:

"Why Record Ultrasonics?
As is widely recognized, most of us can ?t hear much above 18 kHz, but that does not mean that there isn?t anything up there that we need to record ? and here's another reason for higher sampling rates. Plenty of acoustic instruments produce usable output up to around the 30 kHz mark ? something that would be picked up in some form by a decent 30 in/s half-inch analog recording. A string section, for example, could well produce some significant ultrasonic energy.

Arguably, the ultrasonic content of all those instruments blends together to produce audible beat frequencies which contribute to the overall timbre of the sound. If you record your string section at a distance with a stereo pair, for example, all those interactions will have taken place in the air before your microphones ever capture the sound.You can record such a signal with 44.1 kHz sampling and never worry about losing anything ?as long as your filters are of good quality and you have enough bits.

If, however, you recorded a string section with a couple of 48-track digital machines, mic on each instrument feeding its own track so that you can mix it all later, your close-mic technique does not pick up any interactions.The only time they can happen is when you mix ? by which time the ultrasonic stuff has all been knocked off by your 48 kHz multitrack recorders, so that will never happen. It would thus seem that high sampling rates allow the flexibility of using different mic techniques with better results
."


Now I appreciate that most Mics won't be able to record 30khz but for arguments sake let's say we have some mics that do. Are the beat frequencies referred to in the article caused by non linearities such as the air and the ear and must we actually hear the original frequencies in order to hear the Beat frequencies?
In a sentence: Is there any truth in this article or is it more Voodoo?

Thanks,

Karl Odlum


Hi Karl,

Thank you for bringing this to our attention. The Purple Pages and the Apogee Guide to Digital Audio were written many years ago by folks that no longer work for Apogee. While much of the material is useful for beginners, there are some inaccuracies that the current Apogee team found unacceptable and the decision was made to pull these documents. Unfortunately, they were still on the website as of this morning, an oversight on my part. They have since been removed.

For the record, we agree that this is just non-sense. In re-reading this together, Lucas and I are not even sure what point the writer was trying to make here. Dan, I think this was written around when you still worked at Apogee, perhaps you can shed some light on this? (just kidding Wink ).

Seriously, I apologize for the oversight and thanks again for pointing it out.
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Max Gutnik
Apogee Electronics

David Satz

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Re: A question
« Reply #10 on: June 22, 2005, 02:31:39 PM »

Now, that was a classy response; no cheesy denials, no bogus counterattacks. I say, hats off to Apogee for this.
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danlavry

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Re: A question
« Reply #11 on: June 22, 2005, 07:06:12 PM »

Max said:

“Hi Karl,

For the record, we agree that this is just non-sense. In re-reading this together, Lucas and I are not even sure what point the writer was trying to make here. Dan, I think this was written around when you still worked at Apogee, perhaps you can shed some light on this? (just kidding ).

Seriously, I apologize for the oversight and thanks again for pointing it out”



Max

It is refreshing to see such immediate acknowledgment that their was badly flawed material on your website. Strange that others found it. It did do some damage, steering some people towards 192KHz, which for a while was highly promoted by many companies including the one you are salesman for.I was under the impression it was put there as one of the attempts to sell 192KHz, which is relatively a recent development.  No feather in your cap.

Regarding the “mysterious unknown writer”, are you suggesting that your company stood behind the statements about digital audio, not knowing who the writer is? Also you, not I, should be in a position to know the date of the publication. Such knowladge even with an error of 5 years, would set me apart from that “educational material” by many years. My relationship with Apogee ended in 1990 even though my electronic designs continued to be used (some still are).

While you are making changes to your website you may want to correct another mistake. You say Jerry Goodwin designed what you call UV22. I put Nyquist band dither in the first A/D for Dorian Recordings in about 1988. Vince did the digital part. I built the unit before partnering with Apogee and before I even met Jerry. Jerry and I improved the statistical properties of the signal and Apogee called it UV22. The concept while old is being marked as something which it is not. The concept of Nyquist band dither and the statistical improvements came before “noise shaping” a new and very powerful concept.

Noise shaping is the foundation to modern conversion, and is also used by modern dither algorithms, providing a psychoacoustic advantage (shifting the error signal from audible to less audible hearing range). The “HR” in the new UV22HR is misleading because HR is commonly used to indicate high resolution. In fact, the latest improvements were done to fix a computability problem between UV22 and data compressed signals, not to provide high resolution.

Dan Lavry
www.lavryengineering.com
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kraster

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Re: A question
« Reply #12 on: June 23, 2005, 12:04:35 AM »

David Satz wrote on Wed, 22 June 2005 19:31

Now, that was a classy response; no cheesy denials, no bogus counterattacks. I say, hats off to Apogee for this.



Yes indeed. You can't say fairer than that. I guess that clears up the confusion!

Karl
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Greg Reierson

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Re: A question
« Reply #13 on: June 23, 2005, 10:26:30 AM »

Max wrote on Wed, 22 June 2005 12:09



Thank you for bringing this to our attention. The Purple Pages and the Apogee Guide to Digital Audio were written many years ago by folks that no longer work for Apogee. While much of the material is useful for beginners, there are some inaccuracies that the current Apogee team found unacceptable and the decision was made to pull these documents. Unfortunately, they were still on the website as of this morning, an oversight on my part. They have since been removed.

For the record, we agree that this is just non-sense. In re-reading this together, Lucas and I are not even sure what point the writer was trying to make here. Dan, I think this was written around when you still worked at Apogee, perhaps you can shed some light on this? (just kidding Wink ).

Seriously, I apologize for the oversight and thanks again for pointing it out.


Any chance you might publish that in Mix, EQ, etc. so we don't have to relive this discussion over and over and over.......


GR

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Terry Demol

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Re: A question
« Reply #14 on: June 23, 2005, 09:13:31 PM »

David Satz wrote on Sun, 19 June 2005 23:41

Karl, there are some possibly valid arguments in favor of wideband audio electronics and sampling rates higher than the bare minimum, but the statements that you've quoted here aren't among them.

Audible "beat" tones are a phenomenon that occurs in a listener's ears, not in the air of a hall. Thus these tones don't occur at all if either or both of the original "pure" frequencies lie beyond the listener's hearing range. A 50 kHz tone plus a 51 kHz tone, for example, even at very high sound pressure levels, won't produce audible 1 kHz difference energy unless there is non-linear distortion in the playback equipment.

--best regards


I've been thinking about this over the last few days and
maybe there is something we haven't considered here.

We know for a fact that air itself manifests 2nd harmonic
distortion on any sound wave travelling through it due to
the density difference between the high and low pressure
parts of a wave (compression and rarefaction).

We also know that any medium that imposes 2nd harmonic
distortion on a wave will also impose intermodulation
distortion.

So it appears to me that there will be some intermodulation
occuring by the air "carrier" itself before the sound reaches
our ears.

Does this make sense?

Cheers,

Terry
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