Commandment 9: you will have no peak in your recording above -10dbfs. You should in fact be AIMING considerably lower in order to ensure that the tracks never peak in single digits.
err, i use the Studio One included track utility called Mixtool to do Gain Staging. around -12 to -10 dB for each track. the more tracks the lower you got to go. -18 dB is a good starting point...
if i had lots and lots of tracks to gain-stage i would probably use the Hornet VU Meter to do that automatically.
Edited by moontan (02/08/1703:09 AM) Edit Reason: added stuff
Ok--so there's aspects, the calibration level....and the not increasing or decreasing with DSP....where am I losing you?
Do you understand that when I say calibrate for -18=0....that the adjustment you're making is the volume of your monitors' power amp? Which for powered monitors, is built in. You should NOT be able to play a CD through them without attenuation. After that adjustment, the rest is simply a window you're aiming for....which gets easier, because you're calibrated to the right level. So long as you want to play a full scale CD without any trim....you'll always be worried about "going over" because you're recording too hot.
How do you set the level for anything you record? Track number one. Whatever that is for you--how to you determine the level you record that? There's mic preamp gain knob, the first gain stage of the entire process....how do you determine where that knob gets set? How do you know you're at "the right level"....whatever that means to you right now?
I'd planned to do one on the importance of the NEW ability, only available mixing in software of what I've coined "zero sum gain staging". So, we'll talk about that below-- but it IS predicated on sort of gain staging 101 that can be found in discussions all of the net. the cliff notes of that are:
You track at the level your ADC's analog section is calibrated for....in the absence of that specific knowledge, you use -18dbfs=0vu. meaning what exactly? what reading on your master out and then your speaker controller, then where on the active monitors? What does this have to do with how loud you are listening to mixes?
I'm pretty sure Flatcat's post that inspired my novels on home recording was actually primarily about that need for studio calibration south of where it typically is.
The first thing I have to do in mixing any home studio's album, is nearly ALWAYS....trim it downtrim what down, where? to my studio's (loud relatively) standard of -14dbfs=0vu. More often than I'd like to admit that trim is -13db. -10db. Meaning someone is cutting at -1dbfs peak. which is often a solid +17db analog on their preamp.if they've normalized, how can you tell? This is significant because cheap preamps don't do +17db that well. Anyway--I don't want to rehash the CUTTING level....you can read about it 900 places on the web. I'm more interested in the next part:
Commandment: no DSP you use will increase the peak perceived gain of your recorded signal. An EQ can increase the perceived level, no? Fletcher Munson
Compressors can be used to raise the RMS level....reshape transients obviously....but, never raising the perceived peak. This is interesting to me, and has been ear opening because it was never available in the analog world I grew up in....boosting EQ made things louder....cutting made them quieter, with no output stage to compensate for that (other than the actual fader). But, in software, where all DSP happens via plug in....most of which (and ALL of which I rec using) have an output gain stage. Which means when you hit bypass it shouldn't get louder or quieter.
Now....obviously when you're raising the RMS level, if you bypass it at a CERTAIN point....it will get louder....because that's WHY you're using it....so, it's important to note that I mean during the loud parts, it shouldn't get louder.
Summing has a cumulative level effect....so, the more tracks, the more level gets added as you sum (mix) them....point being--that doesn't necessarily buy as much headroom as you think it mightwhat does "it" refer to?....but, the biggest thing here is in the evolution of "did I do harm or good"--being able to blindly flick the bypass switch and pick the "best" without it being just "the loudest" actually means that you do LESS DSP....because you realize when you've done well AND when you're just turning knobs hoping for something that isn't going to happen. I am the KIND of "here's some big compression and saturation and a rollercoaster EQ that makes this BOSS!!".....offset the gain and blindly pick "bypass" as better.
what you'll find is that plug in makers know this....and guys like Slate or Fab Filter will tend to make everything boost the level a little...just engaging it....it's just a touch louder. Even when they offer "auto offset" functions....remove that .5db or whatever manually and amazingly, they don't sound as "always better" as you thought they did. makes sense. louder sounds better in most tests
But, the power is there for you to make choices that were never possible on an SSL.....on a Neve....even on a 2480 or most hardware digital mixers. There are downsides to software mixers and recorders....this is a wholly unique upside. Don't let it go to wastemake what go to waste? the zero gain added plug ins?.
I'll try and help.
_________________________
kel
"I love what you guys are trying to do up there" ...from an audience member at one of my gigs. Gear: Fender Medium pick
Summing has a cumulative level effect....so, the more tracks, the more level gets added as you sum (mix) them....point being--that doesn't necessarily buy as much headroom as you think it mightwhat does "it" refer to?....
Lowering your peak recording levels to the calibration level.
Quote:
More often than I'd like to admit that trim is -13db. -10db. Meaning someone is cutting at -1dbfs peak. which is often a solid +17db analog on their preamp.if they've normalized, how can you tell?
Why would they do that? So, ok....true....they may be normalizing to seemingly random levels between 0 and -4dbfs. Or they could be trying to cut as hot as they can without going over. You can often zoom in an see square waves--but, by your logic, they COULD be exporting tracks with a lookahead limiter? This would all be a wholly different kind of bad gain staging.
Quote:
Commandment: no DSP you use will increase the peak perceived gain of your recorded signal. An EQ can increase the perceived level, no? Fletcher Munson
yes. And so you lower the actual volume. The whole thing is you should be able to hit bypass on the entire channel of plug ins and not hear a major shift in your basic level.
Quote:
this is a wholly unique upside. Don't let it go to wastemake what go to waste? the zero gain added plug ins?
yes, zero sum gain staging. The upside that is currently unique to software mixing.
the power is there for you to make choices that were never possible on an SSL.....on a Neve....even on a 2480 or most hardware digital mixers. There are downsides to software mixers and recorders....this is a wholly unique upside. Don't let it go to waste. Top Reply
My choice ::: since the beginning of recording time >> the volume/ attenuator knob ...it will never go wasted.
Edited by C Jo Go*Crystal Studios* (02/08/1705:20 AM)
How do you set the level for anything you record? Track number one. Whatever that is for you--how to you determine the level you record that? There's mic preamp gain knob, the first gain stage of the entire process....how do you determine where that knob gets set? How do you know you're at "the right level"....whatever that means to you right now?
Ms Manley :: You can run -20 or -18 if you like. -16 dbFS reading 0VU = +4 is more standard >> [ the right level comes built with gear -- color indicated even >
Edited by C Jo Go*Crystal Studios* (02/08/1705:59 AM)
Glad I'm not the only one. Pop generally has some good information, but he sure has a way of communicating it in a convoluted manner. Now, I'm more confused than I was before reading his thread....
And yet no one is answering the one question that you ALL have an answer to....because you functionally HAVE to....you do it every time you record something.
How do you determine the level of the very first track you record? You are looking at a meter of some sort? Listening for something?
Commandment 9: you will have no peak in your recording above -10dbfs. You should in fact be AIMING considerably lower in order to ensure that the tracks never peak in single digits.
this.
it is dirt easy.
in any modern DAW, you have meters, that you can set, to Peak AND/OR RMS....
at typical 44.1khz and 24 bit,
you could peak at -22 and still have plenty of signal to work with.
pros consider -12db peak about normal, pop suggests -10, i go as high as -8.
but RMS wise, i'm usually around -20 on my master output, with peaks as high as -8, so a fairly large crest factor.
And yet no one is answering the one question that you ALL have an answer to....because you functionally HAVE to....you do it every time you record something.
How do you determine the level of the very first track you record? You are looking at a meter of some sort? Listening for something?
Going thru this example might help put things in perspective.
My first track is going to be bass. I'll monitor thru my speakers. I pluck a string and the speakers are too loud. For now, I'll lower the speaker volume to where I want it.
Next, I'll look at the levels I'm hitting the DAW. My DAW track fader is set to 0 and my DAW master buss is set to 0. Cool. I play the song a bit and adjust my preamp or maybe the audio interface that has a built in preamp so that I'm peaking no more than say -10dBFS on my meters for the bass track. Even better, I might look to what my overall average signal is, and that's at around -18dBFS. If not, I make a little adjustment on the preamp. I record my bass part and after playing it all back and watching the meters, confirm I recorded the track at a good level. I might need to turn my speaker volume back up to what's comfortable, but I don't need to touch the faders in my DAW yet.
I repeat the same steps with adding an acoustic gtr track and an electric gtr track. Each individually are recorded in the range I was shooting for (average of -18dBFS).
First, I may notice that blend of the 3 instruments is not quite right. Maybe I pull down the fader a bit on the bass so it sits better.
Everyone that said they were confused is doing this already, right? Maybe what's confusing is the calibration part. The point is to have a target range for how loud you are recording your instruments. Generally speaking, it's recording around -18dBFS on average and/or no hotter than around -10dB on peaks. The speaker volume is independent of this. But, just a reminder than when you are tracking, and later mixing, your speaker volume will be at a higher setting than just normal listening to completed mixes.
Pretty basic stuff, I know. As an example for me, I know that when I'm tracking, my volume for my speakers is going to be around 12:00 and my volume for CD's, final mixes, etc. may be around 9:00. There's more sophisticated ways to calibrate all of this, but the general idea is to record your tracks within a safe zone and to compensate speaker volume with the speak volume knob. In a general sense, you are now "calibrated".
Per Gonzo's diagram, hitting the nominal value on a VU meter is hitting at -16dBFS for this particular AD converter. You still have to be aware of the analog rules of tracking levels as it's a big part of the AD. Lock that piece in and you won't ever have to worry about it again.
As far as internal DAW gain staging, you're generally better off to compensate for any gain added thru a plugin. You add it an EQ, boost some signals and now it sounds better. Is it better because it's louder or because you actually changed the tone as you had expected. If you boosted EQ then lower the output so the gain you added by boosting freqs is compensated for. You can tell by bypassing the plugin. Adjust the output volume of the EQ so that it's as close a match as you can get it in volume to raw track. One advantage is now you are comparing the non-EQ'd and EQ'd more fairly. The other is you've kept your gain for that track at unity. The volume of the track is still the same it was.
Now at some point, you've added enough tracks that your master buss is going into the Red. The individual tracks haven't gotten any louder but the sum of the tracks is leaving you with little overhead on your master buss.
There's different ways to approach this but what I normally do is select all the faders of the tracks I've recorded and pull them all down simultaneously so I'm not peaking over, say, -6dBFS on my master buss meters. There's things like fixed point and float that dictate what amount of overhead you have but let's just stick with this for now. As I add tracks, I usually adjust levels as I go by bringing down faders to get a general blend. So, if I pull them all down at the same time, I'm keeping my general balance intact.
No aha moments from what I've written, sorry to say. Just trying to help those that are having trouble getting on the same page as Pop.
Registered: 11/29/07
Posts: 7590
Loc: San Clemente, CA
Originally Posted By: tbrugh
...As far as internal DAW gain staging, you're generally better off to compensate for any gain added thru a plugin. You add it an EQ, boost some signals and now it sounds better. Is it better because it's louder or because you actually changed the tone as you had expected. If you boosted EQ then lower the output so the gain you added by boosting freqs is compensated for. You can tell by bypassing the plugin. Adjust the output volume of the EQ so that it's as close a match as you can get it in volume to raw track. One advantage is now you are comparing the non-EQ'd and EQ'd more fairly. The other is you've kept your gain for that track at unity. The volume of the track is still the same it was...
Thanks, tbrugh.
That is a good take-away from Pops OG post. I did understand that part, but the way you've worded it helps elucidate the point as well.
That is a simple concept, and *should* be a given, but I do think it is often overlooked, and I know that I've overlooked it myself at times, when in a hurry.
Registered: 11/29/07
Posts: 7590
Loc: San Clemente, CA
Originally Posted By: Popmann
And yet no one is answering the one question that you ALL have an answer to....because you functionally HAVE to....you do it every time you record something.
How do you determine the level of the very first track you record? You are looking at a meter of some sort? Listening for something?
I'm looking at meters more than listening, although if I heard some quality I did not like, that would prompt me quicker than what the meter says.
The VS2480 (Yes...I'm one of the holdouts...) has a setting where you can set the point at which the "over" LED's light up. I have mine set at the most conservative level, so that IF they light up, I KNOW I'm over - way over.
I'll take a pass at the track with the "peak level indicator" engaged. I aim to keep the highest level while capturing right around "OdB" on the meter. I know an occasional transient will go above that, but there is so much "padding" in place that it shouldn't cause a problem.
On my analog equipment, I'm old school, and going for "as close to 0dB as I can get," to get the benefits of tube saturation. Though, again, if that didn't sound "good," I'd back it off, and fault the meters.
Registered: 11/29/07
Posts: 7590
Loc: San Clemente, CA
As for monitoring levels, I generally monitor when mixing at "conversation" levels. I forget the exact db, but I got it from Owlsinkskis (Sp?) "Mixing Engineers Handbook."
I actually have a stand-alone dB meter, and have measured my monitor system at the sweet spot, and marked the "monitor" knob on my VS2480 as to where that is.
I have the JBL LSR4328P monitors that self-calibrate, and are connected through it's own network, so all volumes are equal. They are on stands, and have been carefully placed, with measuring distances from the walls/corners, to listening position, and appropriate attention paid to acoustic treatment.
Registered: 11/29/07
Posts: 7590
Loc: San Clemente, CA
Originally Posted By: Bat
...pros consider -12db peak about normal, pop suggests -10, i go as high as -8...
I'm in that neighborhood. But, Math was never my strong suite, and each device seems to meter differently? So, I'm really not sure where I'm hitting on what scale...
Better to use the meters on the input channels....not the little clip indicator light. Certainly they have a peak hold, right? So you belt your loudest line, look at the meter and it tells you what level that peaked at? Adjust the preamp up or down until that peak is -14dbfs. In practice, you'll end up going over that....thus why a lot of people use -18, which is also fine.
Registered: 11/29/07
Posts: 7590
Loc: San Clemente, CA
Originally Posted By: Popmann
...Certainly they have a peak hold, right? So you belt your loudest line, look at the meter and it tells you what level that peaked at? Adjust the preamp up or down until that peak is -14dbfs. In practice, you'll end up going over that....thus why a lot of people use -18, which is also fine.
Yes. "Peak Hold." That is what I meant. I just brain-farted on what it was called. AND, I am looking at the actual channel meters.
I will start setting my levels more conservatively though. Thanks!
Commandment 9: you will have no peak in your recording above -10dbfs. You should in fact be AIMING considerably lower in order to ensure that the tracks never peak in single digits.
maybe we should add that after gain staging all our tracks: >The volume of the Master bus should not go higher than -6 dB or -5 dB, from the volume of all the tracks combined.< if it does, re-adjust your gain staging.
that 5-6 dB headroom on the Master is to leave some leeway for the mastering process.
So, your contention is that if my mix peaks at -2dbfs I've left mastering "less headroom" (an analog concept FWIW) than -6? Do explain how you figure. What defines "headroom" in a mastering studio?
So, your contention is that if my mix peaks at -2dbfs I've left mastering "less headroom" (an analog concept FWIW) than -6? Do explain how you figure.
from what i've gathered, no more than -6 or -5 dB on the master bus seems to be the general consensus (i know our own Falcon Eddy here would confirm this). again, that's from the volume of all the tracks combined after gain-staging them. (you might have to re-adjust a few times during mixing, especially the closer you get to the final mix)
i would agree. EQuing, stereo widening and other processing at the mastering stage will change the volume, usually boosting it. the processing i do at mastering will always eat up 2 or 3 db, sometimes more. it's been my experience anyway.
myself, i never go higher than -5 dB on the master bus to leave some leeway for the mastering. i'm happy if it's anywhere between -6 and -5 dB. it's a good and safe ballpark figure.
a new rule: if you use Peak and/or RMS values to do gain staging, stop right now. use the new LUFS standards, especially the Momentary (last 400 ms) to do proper gain staging. you still need to use Peak/True Peak to make sure you're not clipping though.
the old RMS standard can be thrown in the garbage. it was not invented to measure audio, but wattage.
Da Bears 
