dBVU, dBFS, noise, and YOU! (correct mixer gain)

If people are into Wikipedia, there really is some nice information if you do a search for "decibel."

They offer many of the different dB references with explanations (dBm, dBu, dB(SPL), dbfs, and a lot more. I found it a nice refresher to what I had studied in the past.
 
mshilarious said:
If you can't, post your .mp3 & I'll do it for you :)
Audition has a pretty good FFT actually.

green = 0dBVU
yellow = +2dBVU
Purple = +4dBVU
red = +10dBVU

distortion.jpg


Is seems that, while noise is reduces.... THD is increased!

I think if I keep my board from getting much past +4dBVU, I'm OK.
And now it all holds true.
 

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Also keep in mind that some A/D convertors get really wonky when you get closer than -3 dBFS.

-18 dBFS is one defacto standard for average signal level; -12 dBFS is another.

We mostly agree that trying to pump an analog signal into an A/D convertor at 0 dBFS is probably a bad idea.
 
tarnationsauce2 said:
Is seems that, while noise is reduces.... THD is increased!

Maybe, maybe not. I'm assuming all the 1kHz peaks are normalized to 0dBFS. If so, you may just be seeing the reduction in noise, which was disguising the harmonic distortion.
 
tarnationsauce2 said:
Hmm, I see.

You can take a look at this 24bit wav file:

0, +2, +4. +7, +10 dBVU
2 seconds normalized tone, and then 2 seconds noise floor.

right-click, save-as 2.5mb
http://home.comcast.net/~tarnationsauce/db_test.wav

I get the same result. I think one thing you have discovered is that since your converter is 0dBFS = +16dBu, is that your gear is never going to be strained too much (it probably has another 6dB of headroom), and that distortion and noise shouldn't be a problem for you. And that is a good thing.
 
mshilarious said:
I get the same result. I think one thing you have discovered is that since your converter is 0dBFS = +16dBu, is that your gear is never going to be strained too much (it probably has another 6dB of headroom), and that distortion and noise shouldn't be a problem for you. And that is a good thing.
Cool, I guess my board has some decent headroom then? :D
 
Don't forget, you are testing using a 1khz test tone. 1k is very easy for a preamp to reproduce. Now what happens top your rpe's when you run a more broad range of sound through them? They may not hold up quite as well under heavier outputs. It is very standard and excepted for )db on an analog scale to be 1.23 to 1.24 volts of output at a 1 khz sine wave. It is also standard for your average level to hover around 0 in the analog domain. If you do this, you will have an average level in the digital domain of aroun -18 to -12 on most all equipment that is set up properly. If you follow this method you will be working within the primary operating range of all of your equipment. If you do not, than you will not be.
 
holy shit this thread is so full of mis-information that I can't even stop from laughing! Holy shit, i thought i'd seen it all. Holy shit! I cannot even help you my good folks, you've bought it hook, line and sinker! Later sucka's!
 
Cut and run eh? My guess is if you don't back up your claims, no one is going to bother listening to you, so why post? If you have seen some blatent mis-information, point it out, because you're probably right to some degree. Let's hear where you think the problems are.
 
jimmy2sticks said:
things seem to be heating up in this thread. i want to see RAK and sweetnubs go head-to-head.

oh that jimmy2sticks, always stirring up trouble.

I barely even posted in this thread myself so I'm not insulted personally, I just don't like someone coming in, saying everyone is totally wrong, and then vanishing with no explanation. But maybe sweetnubs posted that just to anger people for fun, who knows.

hey jimmy2sticks, I bet you'd like to see that :)
 
double post(I copied and pasted from my other post here, sorry for the duplicate)

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if you record with a 24 bit word, the noise floor is so low that setting levels that peak well below full scale is fine, still way above the noise floor.

Each bit you add to the word doubles the available values the word can represent, and therefore doubles the dynamic range (signal to noise ratio from full scale down to noise) that you can record.

A doubling of dynamic range equates to 6db. Therefore, each bit in the word contributes 6db of dynamic range. A 16 bit word therefore has a 96db signal to noise ratio, and 24 bit word can express 144db of signal.

In the real world, the audio electronics in the converter provide a higher noise floor than a 24 bit word can represnt, so a good 24 bit converter will give, lets say conservatively, 110db of signal to noise.

This means that if you record your audio with peaks no higher than -14db under Full Scale, you'll still be experiencing a recording with 96db of dynamic range, which is the best any 16 bit CD has every accomplished.


To make the point even more graphically - this all assumed that the source signal has a dynamic range in excess of 96db too. I would bet you a beer that it isn't even close. There's no tube mic that operates that cleanly. Your studio room has noise higher than that. All your hardware compressors and EQs operate with a much higher noise floor.

If you were very careful, and ended up having a source with 70db of dynamic range (congratulations!) you could record it with peaks at -26dbFS (-26 under full scale) and still have preserved every ounce of dynamic range.

So its obvious that hitting full scale isn't necessary at all - why not preserve some headroom just in case? Let's say you do make it just under full scale. No harm in doing that if you don't go over, right?

Well, what do you do when you want to EQ something +2db? Where does that 2db go? Into clipping of course, unless you lower the input level of the plug in, which is going to lose any hypothetical S/N benefit you had preserved anyway.

Even more importantly, when you record this hot, I've got to ask - what did you do to your preamps, and analog chain to get this level? Most converters are set so that 0dbVU = -18dbFS.

That means that if you're getting -6 below full scale on your converter, that you're +12db over the 0VU point! Many analog electronics can crap out here, but almost all will sound different at least. Some times it may be "better" but usually, its a small accumulation of distortion that builds into a waxy fog that makes people blame "digital recording" for its pristine playback of their slightly distorted, but "pretty on the meter," tracks.

If you record with levels around your 0 point, some thing like -18dbFS or -14dbFS, depending on how your converter is calibrated, you'll have your analog electronics in their sweet spot, headroom for plug ins and summing, an appropriate analog friendly level if you use analog inserts later in the process, and on a modern 24bit converter with 110db S/N, the ability to faithfully record signals with a dynamic range of over 90db.

And by all means, 0dbVu is no glass ceiling like 0dbFS is. Keeping levels around 0dbVU doesn't mean that peaks won't exceed that by 6 (or more) db. If they do, your ability to record 96db of S/N (if you even have it in the source, and you don't), just like the best CD you ever heard will be preserved, if peaks don't exceed -12dbFS! More if they do.


) A 24 bit PCM word can express a theoretical limit of 144 db of S/N.

2) The analog electronics in the converter limit the performance to a functional 100 db of S/N. (slightly more in some cases, but I'll use a conservative figure and make the point even without those extra 6 db)

3) As long as the noise floor in any recording system is lower than the noise floor in the signal you're recording, you will record the full dynamic range perfectly.

4) No source you've ever recorded had a signal to noise ratio higher than 80 db, and most would be much much lower. Lynn suggests that he RARELY sees the source's noise floor lower than 70 db down, and even then, rarely. Assuming that his peaks are not at full scale, his typical source S/N ratio must be in the 50-60 db range?

This means that if you record your (best ever) 80db S/N source into a converter so that the highest peak just reaches -19 dbFS (below full scale) on the meter, that the noise floor in your signal will be louder than the noise floor in the converter. You needn't record it any hotter than that.

In the real world, you could get away with peaks around -28 dbFS, and be PERFECT. Any higher than that is totally unnecessary.

Conclusion: There is absolutely NO benefit to tracking hot.

But does it hurt to do it? Read on...

1) Your microphone preamp is set to perform best (gritty distorted choices aside) peaking around 0dbVU. This is where you'd have it set if you were recording to analog tape, hitting 0 on the VU meter. Plug that same source into most converters, and you get peaks around -20dbFS to -14dbFS, depending on how the converter is setup.

The scientists who developed this system understood the situation, even if the guys who wrote the digidesign manual don't! They EXPECT you to record with peaks around 0VU (-18dbFS on the digital scale). They KNOW about the signal to noise deal I explained earlier. That's why they chose to put the nominal level so "low" on the meter.

When you record hotter, with peaks at -6dbFS, lets say. You're driving your mic preamp 12 db hotter than you did yesterday in the analog world! That's going to add a subtle layer of distortion to your project. And they say analog sounds so much better than digital - maybe its because most people use their analog gear incorrectly when recording to digital. Maybe the "problem with Pro Tools summing" is really the effect of tracking too hot?

I've heard people say "My Neves can handle outputs +24db according to the spec, so what's the big deal?" My Neve 1073s are great sounding workhorses. They are rated for a LOT of gain. Still, they definitely sound very different even at +12. Very different. Maybe a good choice in some cases, but not the norm.

2) If you have a peak at -2dbFS, and you try to boost a mid range frequency +3db on an equalizer, you're going to clip.

Another unintended detriment to tracking hot is that you no longer have any headroom in your plug ins! It is true that in Pro Tools, you can recover lost headroom in the mix bus by lowering the master fader. This isn't true in an analog console, where the distortion has happened in a summing amp "upstream" on the master fader. In that case, the master fader only lowers the volume of the distorted signal, which remains distorted.

In Pro Tools, the master fader is actually a co-efficient with each individual fader before summing. This means that if you're clipping the mix bus, you can pull the master fader down, and fix it. Great. But what about the plug ins across each fader? They aren't affected by the master fader (thank god, or your compression levels would change etc) but neither are they protcted by the master fader. If you're clipping the mix bus, and have your master fader at values lower than unity, then odds are that you're clipping some plug ins too.

3) Most analog gear doesn't like inputs that are 12db and more over 0, even if the spec says they can take it. If you track hot, you're causing a nightmare for analog gear that you may choose to insert during the mix. Keep your levels around 0dbVU, and you can leave the digital domain freely without adding more sonic grunge.

Conclusion: Tracking hotter than 0dbVU can easily cause distortion in any number of places in the chain.

So, to reiterate:

1) There is absolutely NO benefit to tracking hot.

2) Tracking hotter than 0dbVU can easily cause distortion in any number of places in the chain.

If you want to hear the result of tracking too hot, and what it does to Pro Tools, listen to any Lenny Kravitz record. believe me, he uses all the best vintage gear, with gobs of headroom etc. There is no shortage of Neve, Helios, Fairchild, Neumann, Telefunken or whatever on his sessions. The sound of those records is entirely due to the tracking and mixing levels.

"But how do I get my product hot?"

There is a point to having a final mix that peaks at -0.1dbFS. if you are going to have a 16 bit version, if you want to be commercially competitive, if you like to see all the lights light up - sure, I do it every time. The point is i bump it up LAST in plug ins across the master fader. That way, the mix is all properly gain staged, with lots of headroom right up until the last thing juncture. Then if I raise the result to just below clipping after having the benefit of proper levels all the way through, everything is beautiful.

If you are a non believer, try it. The amount of air, detail and image is astonishing. In fact, eventually you may find that Pro Tools is actually TOO CLEAN and transparent! Then you'll start introducing purposeful distortion in your mix - distortion that YOU control at the mix is a very different animal than the unwitting accumulation of crud that comes from tracking too hot all along.

So all this means is that the noise is SO low in a good modern 24 bit converter that you can keep gobs of headroom for proper interfacing with analog gear, and still get the full 96+db of dynamic range, just 12 to 18 db lower on the meter. Your analog gear will thank you too.

So all of this results in a pristine, beatiful, airy, detailed 24 bit mix with peaks around -12dbFS? Cool! .[/i]
 
BigRay said:
A doubling of dynamic range equates to 6db. Therefore, each bit in the word contributes 6db of dynamic range. A 16 bit word therefore has a 96db signal to noise ratio, and 24 bit word can express 144db of signal.

I wasn't sure what you meant by a doubling of dynamic range equates to 6 dB, but my understanding is that dynamic range of a digital system was calculated as 6 dB/bit not counting the least significat bit (or something like that). So that a 16 bit word actually has 90 (15x6) dB of dynamic range and a 24 bit word has 138 (23x6) dB of dynamic range.

Doubling the distance from a source equates to a 6 dB decrease (generally).
Doubling the power is 3 dB increase (generally)
Doubling the volume is a 10 dB increase (generally)
But these are acoustic properties, not tied to bit depth.
 
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BigRay said:
But what about the plug ins across each fader? They aren't affected by the master fader (thank god, or your compression levels would change etc) but neither are they protcted by the master fader. If you're clipping the mix bus, and have your master fader at values lower than unity, then odds are that you're clipping some plug ins too.

This is incorrect.
The plugins on the master fader in Pro Tools are post fader. This is different than the plugins on audio/aux tracks which are pre fader. So any volume changes on the master fader affects levels going to the plugins (yes, including compression levels).
 
bennychico11 said:
This is incorrect.
The plugins on the master fader in Pro Tools are post fader.
Nuendo/Cubase have inserts pre and post fader on the master. I'm not usre about samplitude and other DAWs.
 
Farview said:
Nuendo/Cubase have inserts pre and post fader on the master. I'm not usre about samplitude and other DAWs.
yeah, you are right.

my PT homework isnt up to date(I dont use the POS, I use Sequoia and Samp) :p

but MOST others are as farview described.
 
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