Recording too hot

[RAK]

I know I've said this in another post, but I'd like to say again here that dbVU is not a real measurment. The measurement that VU meters use is dBu or dBv which is a voltage measurement referenced to .775 volt (RMS)

There is also dBm which is a power measurement referenced to 1 mW.

db(SPL) - Sound Pressure Level - that is an acoustical measurement.

I apologize if I missed this in someone else's post. Bit depth does relate to headroom, 6 dB/bit (excluding the first bit) so 16 bit = 90 dB which would be your 0 dBFS. 24 bit = 138 dB

If your ultimate goal is for CD, then you're sample rate converting/dithering down to 44.1kHz/16 bit. That may be what those other threads meant by it all turning out the same in the end.

 

[rgraves]

Southside Glen to the rescue!
Thanks so much for the information...OK, so I went and did some experimenting and here's the deal:

First of all I didn't even realize it, but my cubase set up is already been set to "32 bit float" the whole time. So that's where everyone seems to recommend it right?

I am recording at 44.1 though and plan to keep it there since there's still some debate on the subject (unless someone is so passionate as to convince me otherwise.)

Let me explain my set up:

With the EMU 1820M audiodock I have my guitar going into the preamp, any direct in guitar also going into the preamp, and vocals go to the preamp.
The Keyboard, which is usually used in classical piano format, is sent to the "line in" of the EMU.

The preamp cannot be altered as far as "pro" or "consumer", but the "line in" can.
I've always had this set to "consumer". I went ahead and changed it to "pro +4" and the volume went way down on it. It is right at about -16db max, where as I used to be able to get up to -6db.

So here's my question(s): My keyboard being a Yamaha S90, doesn't seem to be something that would be considered "pro" even though it's a $2000 keyboard (or was anyways.) Is it right to record it with the "pro +4"?

The emulated output from my Marshall AVT150 doesn't fluxuate hardly at all from it's output, especially with high distortion, it is pretty much fixed at around -14db with the preamp all the way down (zero), there is no volume control for the emulated out, and the signal like I said only fluxuates maybe 2 or 3 db max. SOOO, with signals that are somewhat fixed should I record those hotter than -18dbfs? like at -4dbfs?

 

[The Flame]

Remember that is a more or less average level. Your big peaks could easily rise another 8-10dB above that.

What I sould have explained better was that 0dBVU come in at -18dBFS (give or take) if you have your input mixer levels on your soundcard and software set to untiy gain (0 gain).

The idea is you shouldn't have to "set" much of anything on your computer. If you have your levels set correctly on your gear going in, they should more or less automatically come into your computer (or digital recorder) at the right levels if you do not boost or cut your input levels in your software. Don't worry about the meter readings; if your gain staging is set properly onthe analog side, that's all you need to really worry about. This is the idea and the ideal.

G.

Hey man, im sorry but i dont understand that. I Shouldnt be having to constantly set levels? I am not sure what you mean by having the gain staging set properly on the analog side, as in using the volume level knobs on my firepod? Luckily the firepod has clip meters on the unit itself, which i watch. I never let it go into red. But i watch the levels on cubase more, because they seem accurate, when it goes into red, i hear clipping. This is something I always wondered about, and somethign that is constantly making my recordings so hard to work with. As i read your reply, i am getting it more and more. Are you saying to watch the hardware's "input"? I do that, but i usually let the level go to the top of green and sometimes the input from the start hits 0.0. is this ok?

 

[Farview]

Hey man, im sorry but i dont understand that. I Shouldnt be having to constantly set levels? I am not sure what you mean by having the gain staging set properly on the analog side, as in using the volume level knobs on my firepod? Luckily the firepod has clip meters on the unit itself, which i watch. I never let it go into red. But i watch the levels on cubase more, because they seem accurate, when it goes into red, i hear clipping. This is something I always wondered about, and somethign that is constantly making my recordings so hard to work with. As i read your reply, i am getting it more and more. Are you saying to watch the hardware's "input"? I do that, but i usually let the level go to the top of green and sometimes the input from the start hits 0.0. is this ok?

You do need to set levels everytime you record something different. That is what gain controls on the firepod are for. The reason you are having a hard time getting this is because you are using an all-in-one unit. If you had a separate mixer and interface, you would see the meter on the mixer and how that relates to the meter in cubase.

Until you get some time working with VU meters, it will be hard for you to visualize this concept. The easiest safe way to keep your levels good is to just never go above -6dbfs in cubase.

The whole gain staging and -18dbfs thing relates to rms or average level, not peak level. You are dealing with peak meters, thus the confusion.

 

[The Flame]

thanx alot man. Ya i actully went downstairs and tried recording the bass at a lower level than i usually do. i tried playign hard, then soft, slapped, popped, and just finger picked it. It didnt clip as it usually does with all those slaps.I had more control. I will def try this with the guitars. I read in the cubase manual that you have to try to get the most level without clipping, but i guess i was using too much level. I was trying to set the bass too hot from the beggining. I was recording about -6. But betwenn songs, like if one is heavier, i wouldnt change the levels for that song, or just get good levels generally for all the songs that are gonna be recorded one time? So thanks alot. That helped! Keep Rockin!

 

[Farview]

The advise on levels that you get in those manuals isn't very good. The 'get it as hot as you can without clipping' thing is a hold-over from the 16bit days, when converters were really only giving you 12 good bits. It's outdated.

 

[SouthSIDE Glen]

I know I've said this in another post, but I'd like to say again here that dbVU is not a real measurment. The measurement that VU meters use is dBu or dBv which is a voltage measurement referenced to .775 volt (RMS)

True that dBVU is actually based upon some other reference. Just what that reference is can change depending upon the gear (which is part of why this gets so confusing). For example, on some tape decks, the VU meter is calibrated so 0dBVU corresponds to a magnetic field (measured in nW) around the tape record (or playback) head equivalent to the rating of the calibration tape used for specing the deck. On other devices, 0dBVU may be calibrated to correspond to a voltage level at the device's input or output equivalent to the rated input out voltage for the device (e.g. +4dBu or -10dBV.) Still other devices like AM transmitters have 0VU calibrated to reflect 100% of the rated carrier modulation level.

But all that is for tweakheads who are ready for it only. For the rest who may just be wrapping their heads around analog gain staging - "gain staging" really just being the buzzword for "setting the right input and output levels all along your signal chain in order to hit the 'sweet spots' for where your analog devices are designed to operate most efficiently and deliver the [more or less] optimal signal - it's mostly important to remember that quality analog gear is usually designed so that the 0dBVU mark on it's metering indicates to the idealized "sweet spot" for that piece of gear. Note that I say "idealized." There are variances and caveats along the way (e.g. some analog gear sounds "k3wl" or simply has plenty of headroom when pushed to oversaturation a few +dB above 0dBVU.) But the general idea is that if you are keeping your gear riding around 0dBVU, you are doing OK.

Now, when I referred to the "automatic" aspect of levels on the PC or digital recorder, this just goes back to the quasi-standard for quality A/D converters that says that an input of 0dBVU on the analog side should output an average level of somewhere around -18dBFS on the digital dB scale on out the digital side. (Again, as Farview said and I aluded to in an earlier post, this is an average reading; one can easily get transient peaks that will momentarily read well above that.) So, if your converter is punching out the bits based upon that conversion formula, in theory if you have the input gain in your recording software set to zero so that it's not boosting or cutting the input level (what's called called "unity gain"), then you should "automatically" recording at the same level that your converter is pumping out.

Yes, you're right (as is Farview) in that that means setting the levels correctly on your converter box - which in your case is the firepod. Setting those knobs to read proper levels on your meters combined with unity gain settings on the software inputs should give you the recording levels you want.

G.

 

[BigRay]

"DONT RIDE THOSE LEVELS!

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.

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]


http://www.analog.com/processors/processors/sharc/technicalLibrary/data_word_size.html


http://www.prorec.com/prorec/articl...ea68a9018c905afb8625675400514576?OpenDocument

http://www.prorec.com/prorec/articles.nsf/articles/3F50D03B51D16EF386256904007DFFC0





1) 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"

 

[jndietz]

So basically, if I set my mixer's pre-amp gain at -10dB, set the track volume on the mixer to 0dB, and the main out on the mixer to 0dB, it will come out at approximately -18dBFS in my DAW (Adobe Audition 2.0)?

Here is my mixer (Behringer UB802)

 

[Farview]

So basically, if I set my mixer's pre-amp gain at -10dB, set the track volume on the mixer to 0dB, and the main out on the mixer to 0dB, it will come out at approximately -18dBFS in my DAW (Adobe Audition 2.0)?

Here is my mixer (Behringer UB802)

The mixers preamp gain setting will depend on the level of the signal going into the preamp. If you have the track fader and the main out fader at 0db, you adjust the preamp gain so that your mixers meters read 0dbvu. The DAW should just take care of itself.

 

[jndietz]

The mixers preamp gain setting will depend on the level of the signal going into the preamp. If you have the track fader and the main out fader at 0db, you adjust the preamp gain so that your mixers meters read 0dbvu. The DAW should just take care of itself.

I was just thinking about this--so... lets say I have the track fader set to +5dBVU and the main out at +1dbVU, does that mean I should adjust the preamp gain so that it reads +6dbVU (yellow LED on my mixer)?

 

[Farview]

I was just thinking about this--so... lets say I have the track fader set to +5dBVU and the main out at +1dbVU, does that mean I should adjust the preamp gain so that it reads +6dbVU (yellow LED on my mixer)?

Assuming your mixer is calibrated correctly, yes.

The preamp gain sets the level of the signal coming into the board. In the case of a micrphone, it brings the mic up to line level. Everything coming into the board (out of the preamps) should be at line level.

The channel faders set the level of the channels signal relative to the other channels.

The subgroup faders set the level of groups of signals relative to each other.

The master fader sets the level of the mix on it's way out the board.

 

[jndietz]

Assuming your mixer is calibrated correctly, yes.

The preamp gain sets the level of the signal coming into the board. In the case of a micrphone, it brings the mic up to line level. Everything coming into the board (out of the preamps) should be at line level.

The channel faders set the level of the channels signal relative to the other channels.

The subgroup faders set the level of groups of signals relative to each other.

The master fader sets the level of the mix on it's way out the board.

And just to clarify, "line level" is -10dBVU?

A definition from Google said this:

This is the nominal or operating level of an audio system. This level normally corresponds with a '0 VU' meter reading. Standard Line levels are: +8 dBm (1.95 volts RMS), Broadcast +4 dBm (1.23 volts RMS), Pro Recording -10 dBv (310 millivolts RMS), Alternative Pro Recording

Going back to my previous post of the +5dBVU on the track and +1dBVU on the main out, that should theoretically come out as -12dbFS? What did you mean by "the DAW should just take care of itself"? Does that mean I shouldn't have to change anything in the DAW?

And finally my last questions before I head back into recording my guitars, vocals, and drums:

Should I be recording at line level and then mix accordingly, adjusting track volume as I need to, finally mastering the product into line level? Or should I record tracks as hot as possible without clipping, then mastering the product into line level?

 

[Farview]

And just to clarify, "line level" is -10dBVU?

read it again, it says -10dbv. Db by itself doesn't mean anything, you have to know which scale you are talking about.

Going back to my previous post of the +5dBVU on the track and +1dBVU on the main out, that should theoretically come out as -12dbFS?

yes, as long as everything is calibrated.

What did you mean by "the DAW should just take care of itself"? Does that mean I shouldn't have to change anything in the DAW?

I mean that you should have to worry (for most instruments) about the levels in the DAW if you are sending it a signal that is 0dbVU.

Should I be recording at line level and then mix accordingly, adjusting track volume as I need to,

yes

finally mastering the product into line level?

No. You want to mix the song out at line level. Mastering is where you get the volume way up to compete with commercial releases.

Or should I record tracks as hot as possible without clipping, then mastering the product into line level?

The entire rest if this thread is telling you not to "record tracks as hot as possible without clipping".