No! 20Hz----20kHz is NOT the range of human hearing!

[bp]

Heck, you can do it at home. Generate a random sequence of 25 kHz sine waves in your DAW and play them back (looping, so you don't know where the "play head" is at) through a very good speaker, earphones (or a cheap piezo will probably work).

Just don't try this with a 44.1 sample rate.

[Beck]

But I've been doing a lot of mastering lately, and one band sent in files some of which were recorded at 96kHz and some at 44.1kHz. I do think the 96kHz ones sounded better, but not because they reproduced higher frequencies - my assumption is that something about the 96kHz recording caused the audible frequencies to reproduce better. This wouldn't surprise me given my experience with amp sims, which sound night-and-day better when run at 96kHz than 44.1kHz. I'm not talking subtle improvement, but something even the tin-eared would perceive.

 

 

My own thoughts on this have always been that 2 sample points on the higher frequency waves are not really enough, and the more the merrier on mid frequencies as well. Since a guitar amp sim isn't going to be as concerned with anything beyond upper mids I can see how the higher sampling rate could make a difference that most people can detect.

 

I'm agnostic when it comes to the effect of infrasonic and ultrasonic frequencies, at least when it comes to music; although I know we can feel infrasonic, but not actually "hear" it with the ear. I'm especially aware of this when a certain neighbor drives by with his sub-woofers turned up to 11 and I feel it through my floor. I've thought about burying an axe in his windshield with a note that says, "Roll off the bass or else!" But I'd be out a good axe and have to by another one.

 

IMO, a digital recording will sound better as the highs are rolled off during recording to a certain point, depending on the sampling rate. This may seem counter intuitive, but I think the remaining highs actually sound better.

 

I'm always surprised at the number of people that don't even know where the "sparkle" and "air" live in the frequency spectrum. We don't even need 20 kHz. Even a recording with upper limits of 14 - 15 kHz will sound perfectly natural to most, and 16 kHz will be about as high as most upper harmonics produced by any instrument, and about as high as most adults can hear. I can hear above 17 kHz, but not quite 18 kHz these days. That's well above average for a person of my 40-something years. I know I'm not missing anything.

 

And oh yeah... to Ras, 20-to-20 is a general agreed upon average range that

[Rudolf von Hagenwil]

2) why do some people want so badly to believe there's more to hearing than the established known science? It's like the X-Files. --Ethan

that's a good question for scientists to address to Beethoven

-- Schubert

[Beck]

I dont know, but I know a girl that can sing "I Want To Kiss You All Over" by Exile in one long burp after a few beers.......

Does she have a sister?

[kwilson]

My own thoughts on this have always been that 2 sample points on the higher frequency waves are not really enough, and the more the merrier on mid frequencies as well.

 

 

I've always thought the same. At a sample rate of 44.1khz the highest note on a piano (with a frequency of 4186hz) is having its waveform represented by 10.5 samples. At a sample rate of 96khz that same waveform is represented by nearly 23 samples. I'll take the higher resolution for that reason alone.

[blue2blue]

that's a good question for scientists to address to Beethoven -- Schubert

I don't believe Schubert really said that.

 

 

 

Last Saturday I heard B's 9th in performance.

 

Not bad for a guy who couldn't hear anything below 20 kHz, either.

[MikeRivers]

 

My own thoughts on this have always been that 2 sample points on the higher frequency waves are not really enough

 

You think Mr. Nyquist was wrong? And Mr. Shannon?

 

It's true that some digital processes work better at high sample rates because there's less loss of ultrasonic information. De-clickers is one example. I'll trust Craig on the amplifier simulators, but I can see that they might be processing harmonics that we can't hear, and when they get through the mill, they become something that we can hear. In order to accurately simulate an amplifier, they need to accurately reproduce the losses as well as the distortions that the amplifier adds.

[rasputin1963]

 

And it doesn't just stop with frequency perception -- that uniqueness extends to every aspect of the human auditory system -- and it extends through time as well: what you hear today is not exactly what you will hear tomorrow, even if you could somehow be subjected to exactly the same sound phenomenon in exactly the same room with your head and body in exactly the same position. In practical reality, every minute turn of the head, change of position, as well as conditions within your body and particularly the interconnected system of the ears, nose, throat, oral cavities, change your perception in ways both small and large. Listen to a given sound, yawn or swallow, and listen again. Your raw perception of the sound will have changed. Our brains go a long way to sort of 'averaging out' those perceptions over time but, if you concentrate, you can to some degree 'override' that mental correction and, by carefully paying attention, get a practical demonstration of how tiny changes in both body, environment, and the body's position/orientation in that environment mean that, in essence, you never step in the 'same river' of auditory or other perceptual experience. Every moment is, in ways tiny and not, unique.

 

 

 

Which rather implies to me the importance of an aspiring hitmaker to blanket his audience with playback in as many environments as possible..... So the listener's brain will hear your song under a wide variety of mental states and situations... and thus his brain can "average out" his listenings of your song. Only then can he construe your song as a "hit".

 

It sounds like I'm being glib or facile here, but I couldn't be more serious.

[Extreme Mixing]

I'll leave it to you guys to sort out the frequencies above 20K. I've stayed busy for my whole life trying to sort out the ones below.

 

Steve

[techristian]

 

Let me ask you something, Ethan: if you had your choice of two products that were impacted by frequency range -- perhaps a headphone amp -- and one had a filter with a sharp cut-off at 20Hz and 20kHz, while the other had a range of 10Hz-120kHz, and both were the same price, would you give any consideration to the one with the expanded frequency range? Based on your philosophy in the above statement, it would appear that it wouldn't make any difference.

 

 

I might be able to FEEL the 10hz, so I would rather spend my money on the system with the wider frequency range.

 

Dan

[UstadKhanAli]

I might be able to FEEL the 10hz, so I would rather spend my money on the system with the wider frequency range. Dan

And hopefully your TV, clock radio, home stereo, and car stereo can reproduce that!

[techristian]

And then there is the ROOM that I watch tv and listen to music in. It does have a rug, but also plaster walls. How will that room resonate with the extra frequency response? How does the extra frequency response effect the STEREO IMAGE? Phase? Digital "grittiness" ?

 

Dan

[Beck]

You think Mr. Nyquist was wrong? And Mr. Shannon?

 

 

Nope, I think many people that invoke Nyquist-Shannon in digital vs. digital or digital vs. analog discussions are wrong about what Nyquist means. So it

[Jimbroni]

I don't know about all this sixth sense utrasonic alien ear mumbo jumbo.

But I know I like 24 bit 48K worlds better than 16 bit 44.1.

Honestly, I think it has more to do with how the DAWs process summing and fx, than anything.

However, I do wonder how much creedance there is to the notion of say a 40khz signal affecting a 10khz signal. The reason I consider it, is that I know I can have bass line centered at 125hz, but when you boost the 500hz it sounds like more 125 without having to actually increase that energy. Think about it 40K is only the second harmonic of a 10K source. Now where it breaks down is that if your speaker, playback system, and ears can't create that upper harmonic, then it shouldn't have an impact on the 10K source. However, in the digital calculation realm, are these numbers jiving together in a way that maybe could benefit from higher rates? I dunno.
I only know what I like, and prefer 48K but I use 44.1K because thats where I end up anyway. And I'm not fond of SRC, I don't trust it. Down the road if given unlimited hd space, I'd probably bump it up to 88.2. As many have noted that converts nicely to 44.1, though its seems like a big space investment for me to diddle with now.

[Extreme Mixing]

If you only had the choice of listening to music from 10K up or 10k down, which do you think would sound more satisfying?

[Beck]

If you only had the choice of listening to music from 10K up or 10k down, which do you think would sound more satisfying?

 

 

 

10k down, without question.

 

Unless you gotta thing for upper harmonics and maybe hate fundamentals,

[DevilRaysFan]

If you only had the choice of listening to music from 10K up or 10k down, which do you think would sound more satisfying?

frequencies? 10K down

dollars? 10K up

[cyscrimum]

How great your info is!But Can you give some more sample questions and answers.I really want to get more info about this topicIt really useful for me. Thanks.

[Ethan Winer]

 

if you had your choice of two products that were impacted by frequency range -- perhaps a headphone amp -- and one had a filter with a sharp cut-off at 20Hz and 20kHz, while the other had a range of 10Hz-120kHz, and both were the same price, would you give any consideration to the one with the expanded frequency range?

 

 

Yes! But not for the reason you think. A well-designed amplifier should have no problem passing frequencies from far below to far beyond what we can hear. If a circuit doesn't pass a much wider range, that's a good indication it's not a good design. For example, if the designer had to roll off the high end just above 20 KHz, the circuit is probably unstable and the roll-off is a band aid fix to avoid oscillation.

 

But for a sound card or other digital recording device, I'd expect it to roll off sharply above 20 KHz. That's fine and normal.

 

--Ethan

[Ethan Winer]

 

Bob Katz tried to study this out a few years back. He created a set of brick wall filters to progressively chop off the top end that he was listening to in his mastering suite. He was quite surprised at how low he had to go before he considered the sound to be degraded, or even that he could hear a difference with and without the filter.

 

 

Yes, and that parallels a test I did back in the 1970s. A good friend of mine worked for Hewlett-Packard at the time, and had access to all sorts of cool test gear. One test we did was to jangle a set of keys in front of my Neumann U-47 microphone. We split the output of the console's mic preamp to a spectrum analyzer, and also patched in a precision sweepable low-pass filter. With the filter out of the circuit, the microphone output showed energy up to the spectrum analyzer's 50 KHz limit! Then we slowly reduced the filter's cut-off frequency listening carefully for the first sign of lost highs in the jangling keys.

 

I think we noticed a change around 18 or 19 KHz. However - and this is very important! - the JBL pro-grade monitors I had at the time rolled off around that same frequency. I point out the speaker roll-off because most speakers, including those sold today, do not put out much past 20 KHz. It kills me when people claim to hear, or be affected by, ultrasonic frequencies, for example:

 

 

There was Rupert Neve's classic demonstration that it was possible for him (and others, presumably) to hear the difference between an 8 kHz sine wave and 8 kHz square wave, but apparently that was ultimately attributed to something oscillating

 

 

Exactly. As I recall Rupert claimed that he and/or Geoff Emerick heard a circuit oscillating at 50 KHz. But what speakers can output 50 KHz? None! At least none available at that time. So it's clear what was heard was more likely IM distortion that aliased down into the audible range.

 

As for high frequency square waves versus sine waves, that's another case where people doing the tests don't understand basic electronics. If you switch a typical function generator from a sine wave to a square wave, the peak voltage stays the same but the RMS level at the fundamental frequency changes. So of course the sound will change, though it's due only to the level change and not the presence or absence of ultrasonic content.

 

--Ethan

[Ethan Winer]

 

My only remark is that "established known science" obviously doesn't know everything or no progress would ever be possible.

 

 

Of course. But as skeptics would say, just because we don't know everything doesn't mean we don't know anything!

 

 

Surely there's more to hearing than the current state of science understands?

 

 

I can tell you one thing that "science" knows for a fact, but that most people miss completely: Hearing is very fragile and short-term. Every time we hear something it sounds different, even when nothing changes. At my recent AES "audio myths" workshop, James Johnston of DTS (lossy encoding experts) spoke for 30 minutes about how unreliable hearing is and why ears are totally unsuitable as a "measuring" device.

 

--Ethan

[blue2blue]

 

Which rather implies to me the importance of an aspiring hitmaker to blanket his audience with playback in as many environments as possible..... So the listener's brain will hear your song under a wide variety of mental states and situations... and thus his brain can "average out" his listenings of your song. Only then can he construe your song as a "hit". It sounds like I'm being glib or facile here, but I couldn't be more serious.

No... it actually did make sense to me...

 

Certainly, whatever the rationale behind it, that's certainly the practice of the big media machine. Get that song everywhere.

[Beck]

I can tell you one thing that "science" knows for a fact, but that most people miss completely: Hearing is very fragile and short-term. Every time we hear something it sounds different, even when nothing changes. At my recent AES "audio myths" workshop, James Johnston of DTS (lossy encoding experts) spoke for 30 minutes about how unreliable hearing is and why ears are totally unsuitable as a "measuring" device.

This is basically what I was trying to say for like 30 hours when we were discussing the methods used by Meyer and Moran in their Audibility of a CD-Standard A/DA/A Loop... study. :poke:

[blue2blue]

 

And then there is the ROOM that I watch tv and listen to music in. It does have a rug, but also plaster walls. How will that room resonate with the extra frequency response? How does the extra frequency response effect the STEREO IMAGE? Phase? Digital "grittiness" ? Dan

For 10 Hz, it's all about size and shape (pallel-ness of walls/ceiling/floor). Those big old 10 Hz waves don't really 'care' if the walls are glass-shiny or wrinkle-coat, or, I suspect, carpeted. I would not make a 56x56 ft room with parallel walls if I was expecting to be pumping a lot of 10 Hz. I doubt you'd 'hear' it as sound but I'm pretty sure that at some point you would feel it.

 

But, hey, this stuff is Ethan's bailiwick.

[Beck]

I don't know about all this sixth sense utrasonic alien ear mumbo jumbo. But I know I like 24 bit 48K worlds better than 16 bit 44.1. Honestly, I think it has more to do with how the DAWs process summing and fx, than anything.

 

 

Even 20/48 can do wonders for me, but I don't do ITB or use plug-ins. If the interface has 8 analog outputs, I use 8 tracks, each going into a channel of the analog console. There the tracks are processed with the analog EQ on the console and sent to outboard effects through the effect sends just like my analog tape tracks. This is the secret (one of the secrets) to making digital sound less digital.

 

When using digital I've always believed in minimal meddling within the box. IMO, people can be too plug-in and process happy, which degrades a digital sample even more. So depending on how one works with the raw sample (recording) you can end up with something pretty good, or barely passable. Of course analog tape is my point of reference when judging sound quality, or maybe better expressed as the sound I'm looking for.

 

I still like my 10-year-old Echo Layla20. The ubiquitous Crystal CS5335 A/D and CS4327 D/A converters do a great job