Anyone Using K-14 Metering?

[dcollins]

I just have a mark on the monitor attenuator. Sometimes I turn it up a couple clicks, other times down a couple. Mostly I leave it on the mark.

Mastering is not like film where there is a calibrated level, all you need to know is what 85 - 90 dB SPL sounds like in your room and do your "work" there.

If you listen too quietly you make it too bright/bassy, and if you listen too loud it will be dark/thin. How's that for a generlisation? But mostly true.

The K System is just to make folks cut at a lower level, isnt' it?

DC

[drakem]

Originally Posted by bob katz

Excuse my poor communication skills! :-). What I mean by not having to look at the meters is simply that if you apply enough monitor gain you will never record with an overload. And for 24 bit recording you just don't have to deal with the meters as long as, say, the maximum peak is somewhere above -10 dBFs, but it can even be lower than that and you will not perceive a signal to noise ratio degradation. So yes, you can mix "softer" (whatever that means) with the monitor gain at K-20 (0 dB on the monitor pot) and I'll bet you dollars to donuts that your max peak will still land at -10 dBFS or above. Go for the sound, trust your ears. Let us know how it works.

Thank You BOB and You all here for this great thread.
By "softer" I mean that we ( me and my wifey-assistant ) tend to listen and mix at quiet levels. I was reading a lot about SPL, db and so on and came to conclusion thats probably a personal problem of us.
We both are coming from classical music, both being professional orchester musicians.
I have the filling that what we hear (loud or soft) is what we remember from the days in orchestra. There is so much dynamic change, but you´ll notice that after a big FF ssimo there is ALLWAYS ( or almost, shame on Wagner) a quiet phrase, giving the listener time to breath. Thats brings us I guess to attenuate our monitoring level when mixing (specialy dance music ).

I have another question. Right now I read the book "Mixing with your mind" (great book by the way) by Mike Stavrou. In one chapter he explains his thoughts about mixing to digital.
He writes, one shoud aim for better peak/VU ratio (less peak more VU)than to analog, trying to mix close to digital 0VU. It looks like he thinks that digital audio tends to produce more distortion at lower VU than analog. (man, I hope I understood everything correct ).
Im not able to hear the difference between less or more digital 0, but I cant also tell the difference between 48kHz and 44kHz :(
Is it true? It makes all the things more confusing.
When we mix with K-14 (we love it ,our customers allmost allways...) we tend to stay between -10 -5db peak (but at monitor gain about 79db), than it sounds OK. ITs not close to digital 0 at all...

Thank you

Wojtek

[bob katz]

Originally Posted by drakem

Thank You BOB and You all here for this great thread.

SNIP

I have another question. Right now I read the book "Mixing with your mind" (great book by the way) by Mike Stavrou. In one chapter he explains his thoughts about mixing to digital.
He writes, one shoud aim for better peak/VU ratio (less peak more VU)than to analog, trying to mix close to digital 0VU. It looks like he thinks that digital audio tends to produce more distortion at lower VU than analog. (man, I hope I understood everything correct ).

Not true.

When we mix with K-14 (we love it ,our customers allmost allways...) we tend to stay between -10 -5db peak (but at monitor gain about 79db), than it sounds OK. ITs not close to digital 0 at all...

It's simply not a problem. If it sounds good, it is good. The vast majority of "good sounding" music recorded in the past century has about a 12-14 dB peak to average ratio. The more we realize that our recordings are going to be judged in the context of a very large group of music recorded between 1945 and 1999, whose average level is about a K-14, the more it makes sense to aim for that average. They're playing classic big band and jazz CDs in a local restaurant... They just "go together", no struggle at all. In another local restaurant they're having big trouble playing modern rock CDs together because of the smashing of the late 90's CDs versus even as "late" as 1990. And the horrid sound. So they went back to the radio.

BK

[PatrikT]

Bob,

Do you have any opinions on how well RME has implemented the K-system in their Digicheck meters?

Best Regards

[bob katz]

Originally Posted by PatrikT

Bob,

Do you have any opinions on how well RME has implemented the K-system in their Digicheck meters?

Best Regards

Perfectly! And you can alter it (customize it) if you really want. And one of the meters does a true peak oversampling/assessment of intersample peaks.

BK

[Pete G]

Bob, et al

I feel like a broken record, but I just cant listen at 83, or even 77 - in fact my ratshack spl meter indicates I've taken 20dB off by the time I get to a level my ears can handle. My dilemma is that I feel I need to listen at this level for my health, but the discussion here would indicate my mixes/mastering will be less than optimum. What's the way round this?

cheers

Pete G

[Kiwiburger]

As long as you reference your mixes against commercial mixes that you respect, I don't see any problem. Most great mix engineers mix at low levels more often than not - I don't see any justification to mix at such high levels all the time - i'm looking after my ears too.

"Mixing with your mind" (great book by the way) by Mike Stavrou.

I love this book, and highly respect Mike Stavrou for his work at AIR studios on The Pretenders and a whole lot of other great stuff. I use some of the tracks he mixed as my reference audio files.

That book is different from most, and it's aimed at people who already have the basics of recording well sorted. It's more about taking it to the next level, and most of the stuff he is talking out is creative use of your mind using what you have, rather than focusing on gear specifics. Very refreshing. But occasionally I see people who don't like or understand this book - usually because the don't understand the basics to start with, and these new ideas can seem a bit whacky.

I think his comments on digital audio are probably more relevant to 16 bit audio - although he does recommend 24 bits, and the basic principles still apply. I haven't seen a better explaination of the perceptual differences between analog and digital anywhere else. Most discussions get into the maths and the theory, and missing the big picture. Mikes comparison of the two skyscrapers (to those who have the book) makes it very obvious what he is talking about. It's harder to describe in words, but i'll try - because I don't think he is wrong at all.

Analog tape isn't capable of perfect reproduction of sound (however much you may like the sound, it's not a perfect reproduction). Very quiet sounds get masked in noise. Very loud sounds get their transients squashed, so that leaves the middle range where reproduction is at it's best. This is illustrated by a picture of a skyscraper, that is very blurry at the bottom and at the top, but is very clear in the middle section.

Digital audio isn't capable of perfect reproduction of sound either. Very quiet sounds have to be represented with very few bits, hence the "distortion" of low resolution, grainy digital sound that is under-recorded. Noise is less of a problem, but just because the noise isn't there, doesn't mean that low level sounds are better than low level analog sounds - they are a lot worse, just with less noise. The louder the signal, the better the resolution - right up to the point where clipping takes place. So this is illustrated by a picture of a skyscraper that is very blurry at the bottom, and gets progressively better all the way up to the top, which is clearest.

I'm convinced he is 100% correct, and this is a valid way to compare analog and digital. I expect many people will mis-understand what he is saying - that would be semantic arguments in my view, and missing the point.

It seems to me that hybrid systems of analog and digital are actually having to deal with the worst of both mediums, sadly.

The skyscraper photo analogy actually works best with low quality consumer gear - probably slightly less of an issue with high end stuff. But since it affects consumer playback systems, I think it's highly relevant.

It also highlights to me the stupidity of forcing all your dynamic range into the top 6dB of a CD. Sure - as far as your digital "skyscraper" goes, that is where digital resolution is at it's best. But when you force the audio through the inevitable analog stage (especially cheap stuff), you are forcing it in the top "blurry" part of the analog "skyscraper".

Practically anything that convinces people to mix and master at lower levels, to avoid the "blurry" extremes is going to result in better audio.

FWIW.

[dcollins]

Originally Posted by Kiwiburger

Digital audio isn't capable of perfect reproduction of sound either. Very quiet sounds have to be represented with very few bits, hence the "distortion" of low resolution, grainy digital sound that is under-recorded. Noise is less of a problem, but just because the noise isn't there, doesn't mean that low level sounds are better than low level analog sounds - they are a lot worse, just with less noise. The louder the signal, the better the resolution - right up to the point where clipping takes place. So this is illustrated by a picture of a skyscraper that is very blurry at the bottom, and gets progressively better all the way up to the top, which is clearest.

Uh, no.

In digital systems with dither, there is NO difference between analog and digital audio at low or any levels below clipping.

If that's what he's saying, it's 100% wrong. Todays A/D converters have remarkably good low-level performance. Measured or audibly.

If by "blurry" you mean noisy, maybe, but if you are hearing grain at low-levels, you need to find out what is broken.

It also highlights to me the stupidity of forcing all your dynamic range into the top 6dB of a CD. Sure - as far as your digital "skyscraper" goes, that is where digital resolution is at it's best. But when you force the audio through the inevitable analog stage (especially cheap stuff), you are forcing it in the top "blurry" part of the analog "skyscraper".

No. The only difference is one of noise.

Your fades should go perfectly down to a benign noisefloor and that's all there is to it.

DC

[eliottjames]

Bob, I respect your work, and maybe I'm just not getting it (I leave the volume set at a comfortable level 95% when I mix) but isn't this method going a long way to end up with the ratshack meter at about 76 dB? Why not just use the meter to determine the volume, set the monitor controller where it makes the meter read 76 dB, and be done with it?

I think the real point is to find a comfortable level and leave it there. Right?

[Kiwiburger]

DCollins - all I can do is shake my head and wonder if my english skills aren't good enough or something ...

I don't wish to argue - just curious as to how I can improve my communication skills, because the idea that Mike is expressing is so simple I fail to see how I can explain it any better.

Let's assume we are talking about a 16 bit CD. If you have very quiet audio information - say around 48dB, then that is only going to use around 8 bits of the total 16 bits. For the pedants - all 16 bits will be 'used', but only 8 of them would be significant. So the signal will have all the graininess of an 8 bit sample - while still in the relatively noiseless digital domain. Once it's converted, it's then buried in the analog noisefloor as well.

I don't think anyone would argue that 24 bit sound is better than 16 bit sound, and that 16 bit sound is better than 8 bit sound. It doesn't take much reasoning to realise that if you don't make full use of the available bit depth resolution, you are throwing away quality.

I think most people recording digitally do realise this, and tend to try to record too hot. That's when you run into the limitations of analog - the top 6dB range of many cheap consumer converters is very distorted - measurably and audibly.

He's saying the same things that are being said by other people, just different words.

He's not wrong.

[dcollins]

Originally Posted by Kiwiburger

Let's assume we are talking about a 16 bit CD. If you have very quiet audio information - say around 48dB, then that is only going to use around 8 bits of the total 16 bits. For the pedants - all 16 bits will be 'used', but only 8 of them would be significant. So the signal will have all the graininess of an 8 bit sample - while still in the relatively noiseless digital domain.

Get back to us when you include the magic linearizing powers of dither. The only difference is the noise-floor.

8 bits or 80.

Forget the grain.

DC

[djui5]

Bob...this is f'n genius.

Nice work slick

[PatrikT]

Originally Posted by bob katz

Perfectly! And you can alter it (customize it) if you really want. And one of the meters does a true peak oversampling/assessment of intersample peaks.

BK

Hooray! I've been glancing at those for some months now.

Anyways - Now I realize that the "stereo phase & level meter" and the "stereo level meter" are slightly different. The latter have those green dots dancing in the rms staples, which the former has not.

Are those dots the representation of the RMS according to your K-thing? If they are, I better understand how it works.

[Nordenstam]

Originally Posted by Kiwiburger

The louder the signal, the better the resolution - right up to the point where clipping takes place.

This is the 'picture way' of thinking digital. Say, a 600x600 pixel image, with a bright spot in the middle and a gradient towards black on all sides. The gradient will be smoother the more available colors there is. Having a higher bit depth does improve resolution within the image, as the range of available colors grows with bit depth.

Digital audio doesn't work like that. Changing audio bit depth is the same as adding pixels to the border of the image. Doubling the bit depth would be the same as making the picture 1200x1200 pixels, with the same old gradient still occupying 600x600 pixels in the middle. Making the surrounding area bigger will not not make the gradient itself smoother. It'll only enlarge the black area around it.

Sample rate sets the available frequencies, bit depth sets the noise floor. Resolution doesn't enter the equation(see 'q for Paul Frindle' thread).


Andreas

[bob katz]

Originally Posted by Pete G

Bob, et al

I feel like a broken record, but I just cant listen at 83, or even 77 - in fact my ratshack spl meter indicates I've taken 20dB off by the time I get to a level my ears can handle. My dilemma is that I feel I need to listen at this level for my health, but the discussion here would indicate my mixes/mastering will be less than optimum. What's the way round this?

cheers

Pete G

You mean your Ratshack meter tells you you are listening at 53 dB SPL? It must be broken. Human speech at 3 feet is about 74 dB SPL! I'm not saying you have to listen at 83..... it is the level approximately where the ear is most linear, and so your bass response will translate the best. But even as low as 6 dB lower and you will still be doing good work. My guess is your meter must be broken, you can't possibly be 20 dB down from this...

When you go into a THX movie theatre, and watch a normal (not effects) movie, does it sound too loud to you?

The Radio Shack meter's analog attenuator is notoriously inexact from range to range, but not by 20 dB, maybe as much as 3. See if you can borrow an accurate SPL meter from somone, set it to C weighted, slow, and I'll bet you're really working somewhere between 73 and 83 dBC.

[bob katz]

Originally Posted by PatrikT

Hooray! I've been glancing at those for some months now.

Anyways - Now I realize that the "stereo phase & level meter" and the "stereo level meter" are slightly different. The latter have those green dots dancing in the rms staples, which the former has not.

Are those dots the representation of the RMS according to your K-thing? If they are, I better understand how it works.

I don't have the meter in front of me, but (obviously :-) the meters with the K14, K12, and K20 presets are the ones. load the preset, and then compare it with a customized setting.

[bob katz]

Originally Posted by eliottjames

Bob, I respect your work, and maybe I'm just not getting it (I leave the volume set at a comfortable level 95% when I mix) but isn't this method going a long way to end up with the ratshack meter at about 76 dB? Why not just use the meter to determine the volume, set the monitor controller where it makes the meter read 76 dB, and be done with it?

I think the real point is to find a comfortable level and leave it there. Right?

Thanks. There are lot of people who don't get it. Let me try to explain why turning your monitor control down and up tells you a LOT OF INFORMATION, if you learn how to read it.

THE WATER FAUCET ANALOGY.

Your monitor control (volume control) is like a water facucet. When the incoming PRESSURE is high, you only have to open the faucet a little bit to get a high pressure out. If you put a mark on the water faucet, you can see how much rotation you had to give it to get a certain water pressure out. So the physical POSITION of the faucet combined with a knowledge of the pressure coming out will tell you how much pressure they must be delivering at the street.

Similarly, the amount of attenuation that you apply to your monitor control and the SPL you get out will tell you how much SOUND PRESSURE there is coming in.

Where the analogy breaks down is that in the audio system, there is a fixed peak pressure that's allowed, so some sort of governor (compressor or limiter) must be applied to keep that outgoing pressure from exceeding the "overload" point.

Inthe audio side, I'm talking about how you can assess to a great degree the amount of compression that you must apply to a recording by observing your monitor gain control. Assuming you are listening at a given SPL for the same musical passage, and that the musical passage is a busy "pop" or rock recording. The more acoustic the recording, the harder it is to assess, because of the acoustic advantage (no time to talk about that here).

You start with the 83 dB calibration point at -20 dBFS (one speaker). We call that POSITION of the monitor knob "0 dB". At that setting, you can run a very clean and open stereo mix or master without ever getting a medium overload. This position will produce an RMS level of -20 dBFS and an SPL (per speaker) of 83 dB, and a peak level of UP TO 20 dB above that without EVER overloading the digital system. Think about it, you will never have to look at the meter... your monitor control becomes your meter!

Now, turn up your RMS level, say, 6 dB hotter, to -14 dBFS. You are then forced to turn down your monitor gain by 6 dB to get the same SPL. But the peaks of your 20 dB-crest-factor recording will overload the digital system by 6 dB. So you have to get rid of 6 dB worth of peak level or you will overload. And you do that by compressing and/or limiting. And that's why the sound won't be the same, even if the SPL is the same.

This goes on and on.... The more you attenuate your monitor, if you work at the same SPL with the same music, the more you must compress! THUS, (ipso facto they say in Latin) the lower you turn your monitor gain control, the more compression you have to apply. The ability to MEASURE the amount of compression by the position of the control comes from experience, knowledge of the music, and of course, your ears, but this knowledge of how to use the position of the control as part of your toolset puts you ahead of the rest of the gang.

In the 20th century, we set our record (or mastering) levels, and we adjusted our monitor gain so it didn't hurt our ears. In the 21st century, I'm suggesting that engineers have a knowledge of where that monitor gain control is set, for many good reasons, including helping to standardize mastered levels, reduce the pressure for the loudness race, as a tool to help identify when we must be overcompressing, and many more good reasons.

The monitor gain in the film studio is known, fixed, calibrated. I went into a movie theatre the other day and watched a great independent film that had a loud, clean, exceptionally dynamic rock recording/soundtrack. I could tell immediately that they must have had to produce a special mix for this film. Because virtually no currently-released rock recording is this dynamic AND this loud, at this monitor gain. So, by knowing how loud the music is in combination with observing the monitor gain, you can tell a lot about the degree of compression. The snap of the snaredrum and the dynamics of the music gave it away to me. So, knowing your monitor level control's position becomes an integral part of your toolset. If they had instead, took Snoop Dogg, it would have sounded much too loud peaking to full scale. They would have had to attenuate its level by 12 to 14 dB and it would have sounded squashed, wimpy, and undynamic besides.

The K-System was not necessary in the days of analog audio OR ANY medium with a standardized average record level. Our digital media do not have an average level standard, so my standard for calibrating a 1 dB/step monitor in combination with good ears/monitors and acoustics becomes our measurement standard for the 21st century. The meters also help :-)

I hope this helps, at least a little bit.

[Pete G]

Originally Posted by bob katz

You mean your Ratshack meter tells you you are listening at 53 dB SPL? It must be broken. Human speech at 3 feet is about 74 dB SPL! I'm not saying you have to listen at 83..... it is the level approximately where the ear is most linear, and so your bass response will translate the best. But even as low as 6 dB lower and you will still be doing good work. My guess is your meter must be broken, you can't possibly be 20 dB down from this...

When you go into a THX movie theatre, and watch a normal (not effects) movie, does it sound too loud to you?

The Radio Shack meter's analog attenuator is notoriously inexact from range to range, but not by 20 dB, maybe as much as 3. See if you can borrow an accurate SPL meter from somone, set it to C weighted, slow, and I'll bet you're really working somewhere between 73 and 83 dBC.

Thanks for taking the time Bob. Well, I got another SPL meter, and the first one was out a bit. Here's what I found though: put Donald Fagen on - RMS reads about 0 on K-20 (I'm using Inspector XL) but I want listen to him at a room reading of about 70dB or less. Having done my pink noise calibration. So I conclude that I would normally mix at between 60 and 68, with occasional excursions to 72/73 or so.

At the movies, the high points are always a bit loud!

I'll work at 70 for a while.....

Pete G

[PatrikT]

Originally Posted by bob katz

I don't have the meter in front of me, but (obviously :-) the meters with the K14, K12, and K20 presets are the ones. load the preset, and then compare it with a customized setting.

What I meant was that in one case the RMS staples follows the peak swing very closely (the first meter mentioned in my former post).

On the second meter there are additional dots with slower "reaction" in the RMS part. I guess those are the ones to glance at when aiming at a certain K-value.

[Nordenstam]

Have you compared the release and the hold settings?

The K presets in the RME software have a fixed bottom level at -60dB. Way too much for my use. -20dB gives a better range.

[bob katz]

Originally Posted by Lupo

Have you compared the release and the hold settings?

The K presets in the RME software have a fixed bottom level at -60dB. Way too much for my use. -20dB gives a better range.

You're right. I may change that "spec". But in the end, every K-Meter I've endorsed has had good looking ballistics. I don't use the RME meters day in and day out, I glance in the back of the room at a big analog Dorrough, so you know, I haven't really studied the ergonomics. The good news is you can take an RME meter and customize it!

[bob katz]

Originally Posted by Pete G

Thanks for taking the time Bob. Well, I got another SPL meter, and the first one was out a bit. Here's what I found though: put Donald Fagen on - RMS reads about 0 on K-20 (I'm using Inspector XL) but I want listen to him at a room reading of about 70dB or less. Having done my pink noise calibration. So I conclude that I would normally mix at between 60 and 68, with occasional excursions to 72/73 or so.

At the movies, the high points are always a bit loud!

I'll work at 70 for a while.....

Pete G

If it works for you, far be it from me to tell you how loudly you "must" listen. What speed is your meter, "slow", C weighted? If so, then you are an exceptional mixer. I think I like to hear 83-86 dB on the loudest peaks of very dynamic music with good transients.

[dcollins]

Originally Posted by bob katz

But in the end, every K-Meter I've endorsed has had good looking ballistics.

Why arent't the ballistics standardized? Do you get different readings from different versions of the meter?

DC

[bob katz]

Originally Posted by dcollins

Why arent't the ballistics standardized? Do you get different readings from different versions of the meter?

DC

The ballistics are standardized, 600 ms attack, and 2 second decay to 20 dB, but the problem is that the PERCEIVED ballistics change depending on the physical size of the meter, so it has to be tweaked. My current feeling is you can even do 60 dB of scale if you make the top 20 "prime". No surprise, VU meters look and feel different when they come in different physical sizes even when the ballistics for a given meter change are the same.

You do get the same numeric readings on all K-system meters, but the feeling of the "height of the bar" has to be tweaked so I've allowed a lot of leeway. In Spectrafoo's implementation, the meter is infinitely stretchable, and I find it seems faster if you make it larger (no surprise).

I've found each manufacturer's implementations have to be tweaked unless he did exactly the same size steps (physical size) on the screen.

On a similar subject, has anyone found the SIFAM brand VU meters (used in some Neve products) seem slower than the standard, though they claim to meet the standard? I have an NAB test disc with a VU meter test signal but I currently do not have any SIFAMs to test. Wasn't the VU meter first determined empirically (what feels right) and then standardized by measuring the prototype?

[bob katz]

I wrote...

Originally Posted by bob katz

If it works for you, far be it from me to tell you how loudly you "must" listen. What speed is your meter, "slow", C weighted? If so, then you are an exceptional mixer. I think I like to hear 83-86 dB on the loudest peaks of very dynamic music with good transients.

This is a good discussion to have! I'll have to put on that Donald Fagen (Nightfly?) and see where I prefer to hear it, what sounds good to me. Because it is so acoustic, people tend to want to listen to this recording at a lower average SPL than you would a rock recording with the same K-Level. It's worth discussing. So your preferences for a lower SPL on the Donald Fagen may well be what most people want. Steely Dan, though, (when I'm in the mood) I'm fairly certain I tend to listen to fortes somewhere around 83-85 dB.

[Paul Frindle]

Originally Posted by Kiwiburger

Digital audio isn't capable of perfect reproduction of sound either. Very quiet sounds have to be represented with very few bits, hence the "distortion" of low resolution, grainy digital sound that is under-recorded. Noise is less of a problem, but just because the noise isn't there, doesn't mean that low level sounds are better than low level analog sounds - they are a lot worse, just with less noise. The louder the signal, the better the resolution - right up to the point where clipping takes place. So this is illustrated by a picture of a skyscraper that is very blurry at the bottom, and gets progressively better all the way up to the top, which is clearest.
FWIW.

This is completely wrong. If this is the impression someone has given in a well read book - I am not surprised people are confused. Whoever wrote this simply did not understand :-(

[Kiwiburger]

Thanks Paul - i've been googling some of your stuff and it's changing my view of things ...

[Paul Frindle]

Originally Posted by Kiwiburger

Thanks Paul - i've been googling some of your stuff and it's changing my view of things ...

I'm glad I can help in any way :-)

This basic misconception goes round and round, I have been fighting it for at least a decade, but it keeps resurfacing. The problem is that however wrong this misconception actually is, it's apparently so compelling and seemingly 'obvious' that it immediately poisons people's understanding of how digital audio works fundamentally. This causes terrible damage as it dictates the way that people actually work - and then (like catch 22) it provides it's own self justification by actually producing the very same bad results that convince people that digital audio is 'flawed' and suffers from limited 'resolution'.

It really is like some malevolent virus, in that it propagates with amazing ease (because it seems so obvious and people accept it so readily) despite being completely wrong.

A few years ago (in one of my more depressed moments) the notion that people could be 'poisoned' on mass in such a fashion caused me some panic. I mean luckily this 'resolution virus' only affects digital audio. But what if someone came up with a similarly compelling idea that was similarly damaging, that was accepted immediately by everyone on first exposure to it (because it was so apparently obvious an indisputable) and yet it was destroying the whole of our society? But nothing and no one could oust the idea - whatever they said or did and however much evidence there was to show it was wrong and damaging - because applying the idea itself actually caused the symptoms and produced the very damage that convinced people the idea was valid in the first place!!!!

Now this really is a plot for a Stephen King horror novel? Or is it happening already (that's another subject)?

[mars]

Originally Posted by Paul Frindle

A few years ago (in one of my more depressed moments) the notion that people could be 'poisoned' on mass in such a fashion caused me some panic. I mean luckily this 'resolution virus' only affects digital audio. But what if someone came up with a similarly compelling idea that was similarly damaging, that was accepted immediately by everyone on first exposure to it (because it was so apparently obvious an indisputable) and yet it was destroying the whole of our society? But nothing and no one could oust the idea - whatever they said or did and however much evidence there was to show it was wrong and damaging - because applying the idea itself actually caused the symptoms and produced the very damage that convinced people the idea was valid in the first place!!!!

Um... happens all the time, Paul. Google "Eugenics" or "Bell Curve", not to mention "Global Warming" or "Overpopulation".

[Brian Foraker]

BK and all,

Anything to consider differently about using the K System for mastering only?

I'm using the Penguin meters mainly.


Thank you!

brian f