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Gain-staging and plug-ins
Old 5th December 2010
  #1
Registered User
 
🎧 15 years
Gain-staging and plug-ins

There's a huge thread here about gain staging in the digital realm. In another thread some are lauding it as being among the most valuable tips they've ever come across on GS.

If I understand it correctly, on a channel basis one should regard the equivalent of analogue 0dbfs as -20db RMS in the digital realm.

I don't see how this can apply to purely ITB mixing.

Shouldn't plugins be hit at 0db peak? I've found almost all plugins to work fine, if not their best, that way.

What I do is make sure all signals (from recorded clips or midi instrument output) are close to 0db (peak), then simply use the faders to make the master hit close to 0db peak. I don't see any other meaningful way of doing it unless certain plugins benefit from lower levels. There will be differences in behaviour from say a compressor or guitar amp modeling unit, but that is almost always a question of internal tweaking (if necessary adjusting the input signal to the desirede range). As a mixing recipe, I just don't see the idea of maintaining -20db RMS - on the contrary why not rather watch the peaks and take full advantage of the available bit range?

Flame away.
Old 5th December 2010
  #2
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Mixocalypse's Avatar
 
🎧 15 years
Aside from hitting your converters to hard, in you daw, try putting a gain plugin on the first insert and lower the level until it's in that -15 -20dbfs range.

Then Mix for yourself and see if you hear a difference.
Old 5th December 2010 | Show parent
  #3
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🎧 15 years
Quote:
Originally Posted by Mixocalypse ➑️
Aside from hitting your converters to hard, in you daw, try putting a gain plugin on the first insert and lower the level until it's in that -15 -20dbfs range.

Then Mix for yourself and see if you hear a difference.
Hitting my converter too hard - how?

When I wrote:

Quote:
Originally Posted by Morten ➑️
simply use the faders to make the master hit close to 0db peak.
... I meant keeping the master right under 0db.
Old 5th December 2010
  #4
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Ethan Winer's Avatar
 
🎧 20 years
Quote:
Originally Posted by Morten ➑️
Shouldn't plugins be hit at 0db peak? I've found almost all plugins to work fine, if not their best, that way.
It depends on the plug-in and host program's bit depth. Most modern DAW programs use 32-bit floating point math for all operations. In that case the level passing through the plug-in is almost irrelevant. This is discussed in my AES Audio Myths video, in the section starting at 53:39, and specifically at 54:44. In that example I sent a normalized pop tune through an EQ plug-in after boosting the volume by 18 dB. So the volume was way over digital zero, yet the output of the EQ nulled completely with a parallel track that didn't have the volume boost. That part of the video also shows how you can do tests like this yourself.

--Ethan
Old 5th December 2010 | Show parent
  #5
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Mixocalypse's Avatar
 
🎧 15 years
Quote:
Originally Posted by Morten ➑️
Hitting my converter too hard - how?
on the way in.

I'm suggesting to try for yourself with material you already have recorded. By adding the gain plugin first and trimming the level. The comment about your converter... Coming itb Analog to digital Converters have an optimal range too. Clipping them or driving them to hard can have poor results. I noticed your post over on kvr too.

All I've suggested is to try for yourself and see how things turn out.
Old 5th December 2010 | Show parent
  #6
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Moatl's Avatar
 
🎧 10 years
Well,

first of all - depending on how the signal came into the box and you stay within the box it is not about hitting the converters too hard.

"...why not rather watch the peaks and take full advantage of the available bit range..."

Because as long as you stay in a reasonable range you won't have to worry about bit range anymore - at least when you track 24bit.


But what you really need to understand is that mixing with levels close to 0dbfs in the digital domain would be quite the same as mixing on analog console with keeping the levels close to something between +16dbu and +20dbu...

-> given the fact that many analog boards have a headroom of +15dbu max that'd mean you'd grill your board stike

That is also relevant for plugins in the digital domain - if you hit them too hard they sound like crap.

"0" on an analog mixing desk and "0" in your DAW is not the same!!!



second: Let's say you just have you have recorded some signals sitting in the channels of your DAW - pure - without any plugins and you recorded it pretty hot - let's say the channel peakmeter shows about -3dbfs at unity gain. And you don't bring the level down with a gainstaging plugin.

So far so good - but you will want to add some EQ because you find it a little dull and you feel you need to push the upper frequencies a little to make it sitting in the mix a little bit better -> result:
the dullness is gone but the meter is in the red right away -> so what'd you do?
Of course you pull the channel fader down a little to make it stay outa the red, right?

Then you add the next plug-in in the chain and so on...

BUT - what you probably didn't think of, is that in the standard-setting (i.e. in Logic) the fader is at the the end of the chain and the peakmeter again is behind the fader.

Meaning: despite your peakmeter is in the green - your EQ is definitely hitting the red constantly -> the signal sounds harsh
You get the point?

And this effect gets carried through (or even multiplied) your whole plugin-chain that comes behind the EQ!

So the best is to set the metering to "pre-fader metering".

In this case you see the levels before they hit the fader and whenever you add a plugin and the signal runs too hot after adjusting it - you can change it IN THIS PLUGIN right away.

I mean you don't have to go down to a rms of -18dbfs but it still is a good orientation. Just try it and you will see how your mixes will become way more transparent and better sounding!

good luck
Old 5th December 2010 | Show parent
  #7
Registered User
 
🎧 15 years
Quote:
Originally Posted by Ethan Winer ➑️
It depends on the plug-in and host program's bit depth. Most modern DAW programs use 32-bit floating point math for all operations. In that case the level passing through the plug-in is almost irrelevant. This is discussed in my AES Audio Myths video, in the section starting at 53:39, and specifically at 54:44. In that example I sent a normalized pop tune through an EQ plug-in after boosting the volume by 18 dB. So the volume was way over digital zero, yet the output of the EQ nulled completely with a parallel track that didn't have the volume boost. That part of the video also shows how you can do tests like this yourself.

--Ethan
Ethan, thanks. Yet how come people on here insist that gain staging is a big issue in digital? I can see how it of course affects the simutaneous use of any outboard processing, but aside from that? Keeping everything ITB (with things handled by 32-bit floating processing) all we have to watch is the peak level on the master.
Old 5th December 2010 | Show parent
  #8
Registered User
 
🎧 15 years
Its about metering RMS vs. Peak

The only thing I would point out in Ethan's post is that we're talking about RMS values versus PEAK. Any signal can peak high in digital without a problem, agreed. However, some analog modeling plug-ins react to RMS levels differently and should be presented with real world levels in order to react in a real world way.

The key is having a meter that shows you BOTH RMS and PEAK so you can see what's actually going on.

For example:

Distorted guitars have very little difference between their peak and RMS (average) levels since they are already clipped and distorted by the amp. Cranking their recording levels up to a very high RMS level (say -5dBFS) is not necessary and can be a problem when you either use a modeling plug-in or go outside the box to a console/summer or outboard comp/eq. That RMS level is TOO loud and can cause gain staging issues.

A snare drum has a GREAT difference between its peak and RMS levels. It would be just fine and dandy to have it peaking very close to full scale (0dBFS) as its RMS value will be much lower, perhaps below -20dBFS. The snare is mostly transient as compared to the distorted guitar which has very little transient components.

So, on a PEAK meter, a snare would be MUCH louder "looking" than a distorted guitar for the same apparent perceived volume if both faders are set the same in the mix. We hear RMS levels but our meters in DAWS only read peaks!
Bottom line:

• If you are mixing totally in the box with all pure digital plugz on a 32-bit floating point system, do whatever the hell you want with levels. Ethan's right, it shouldn't really matter

• If you are using any sort of modeling plugz that have any level dependent saturation/compression/analogness and/or you are using some sort of hybrid mixing system with a summer/console and outboard gear, be conservative with RMS levels, keeping them between -20 and -10 dBFS thereabouts. It will make the mixing task a bit easier to manage gain staging.

And also to clarify: 0dBFS should NEVER = 0 VU or +4dBu! 0 dBFS is max level, clipping, bad news etc.... This is like +24dBu analog/electrical clipping in a console.... MAX.

-20dBFS should = +4dBu electrical or 0 VU. That is a reference operating level where most RMS signals should reside. Peaks will always be above that.


-ashley
Old 5th December 2010 | Show parent
  #9
Registered User
 
🎧 10 years
I'm running Cubase. If I remember correctly from the manual, that 32 bit float has to come down once you mixdown to 2-track.

The 32 bit float ensures no internal clipping. The "overs" or red clip indicators show the overs that will become distortion at 16 or 24 bit fixed (as in the exported wav file).

It's been a while since I read the manual. Someone correct anything I missed or got wrong.
Old 5th December 2010 | Show parent
  #10
Registered User
 
🎧 15 years
You are correct!

When the final output leaves the internal mix engine, its level must not exceed 0dBFS. You can simply use the master fader to ensure this happens. Also, pop on a dither plug (post-fader) set to 24 or 16 bits depending on your output format.

That's the only time in Cubendo where a level above 0dBFS will cause a problem. Anytime a signal must actually come out of a DA converter, it must not go above 0 or it will be clipped.
Old 5th December 2010 | Show parent
  #11
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Ethan Winer's Avatar
 
🎧 20 years
Quote:
Originally Posted by Morten ➑️
Yet how come people on here insist that gain staging is a big issue in digital?
How come people insist they saw big foot or were abducted by aliens? heh

Quote:
Keeping everything ITB (with things handled by 32-bit floating processing) all we have to watch is the peak level on the master.
Exactly.

--Ethan
Old 5th December 2010 | Show parent
  #12
Registered User
 
🎧 10 years
There have been a couple of outstanding tutorials here on GS in regards to gain staging itb. They were absolute gold in showing what the numbers mean as they go from analog to the daw.

The best stuff I learned was
1. keep the peaks under -12
2. low level sounds are perfectly fine around -24 or -18 when you are recording 24 bit
3. gain stage properly with your gear. this means pass an appropriate level off to the next device.

at the time I was mixing itb and didn't understand how fast my master bus was filling up. Now that I've moved to a hybrid setup, this info is just as important so I can pass good levels from my daw to my mixer.
Old 5th December 2010 | Show parent
  #13
Registered User
 
🎧 10 years
One thing is that most DAW's meters are (allegedly) not great and you can be having overs when your meter isn't telling you. I'm not the expert, I've just read the threads. Also, my mixes sound better to me with lower levels hitting everything.
Old 5th December 2010 | Show parent
  #14
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Mixocalypse's Avatar
 
🎧 15 years
Quote:
Originally Posted by quadrafunk ➑️
1. keep the peaks under -12
2. low level sounds are perfectly fine around -24 or -18 when you are recording 24 bit
3. gain stage properly with your gear. this means pass an appropriate level off to the next device.
I think this method also just makes thing easier during mix, giving you better/ more options while mixing. Lol for me anyway...
Old 6th December 2010 | Show parent
  #15
Registered User
 
🎧 10 years
I hear ya man. Some of the best advice I've read on GS.
Old 6th December 2010 | Show parent
  #16
Registered User
 
🎧 10 years
I usually build mixes around the kick peaking at about -18. I never hit it hard on the way in anyway, but this seemed to keep everything under control ITB and I would never have problems with peaking at the master. Logic is a little different though; I'm still getting comfortable with it.
Old 6th December 2010
  #17
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camus's Avatar
 
🎧 15 years
Quote:
Originally Posted by Morten ➑️
There's a huge thread here about gain staging in the digital realm. In another thread some are lauding it as being among the most valuable tips they've ever come across on GS.

If I understand it correctly, on a channel basis one should regard the equivalent of analogue 0dbfs as -20db RMS in the digital realm.

I don't see how this can apply to purely ITB mixing.

Shouldn't plugins be hit at 0db peak? I've found almost all plugins to work fine, if not their best, that way.

What I do is make sure all signals (from recorded clips or midi instrument output) are close to 0db (peak), then simply use the faders to make the master hit close to 0db peak. I don't see any other meaningful way of doing it unless certain plugins benefit from lower levels. There will be differences in behaviour from say a compressor or guitar amp modeling unit, but that is almost always a question of internal tweaking (if necessary adjusting the input signal to the desirede range). As a mixing recipe, I just don't see the idea of maintaining -20db RMS - on the contrary why not rather watch the peaks and take full advantage of the available bit range?

Flame away.
1. There is no such thing as analog 0 dBFS. dBFS is a measurement of decibel amplitude levels in digital systems.

2. Why on earth you would want every channel on your DAW to peak near 0 dBFS, only to have to pull all the faders down to ridiculous levels like -25 in order to keep your master buss from overloading, I have no idea.

3. How much you fudge around with levels AFTER they have been recorded doesn't matter as much, as most of the S/N issues with modern 24bit recording lie within the analog gain-staging portion of the production chain. Nudging everything close to 0 dBFS after they are recorded (which I presume is what you are doing) offers zero audible benefit and alot of major workflow headaches later. Especially if you need to spit stuff out to analog hardware anywhere in the process.
Old 6th December 2010
  #18
Lives for gear
 
🎧 15 years
Quote:
Originally Posted by Morten ➑️
Shouldn't plugins be hit at 0db peak? I've found almost all plugins to work fine, if not their best, that way.
apply a HP or LP filter after your signal and watch what happens on your meters
Old 6th December 2010 | Show parent
  #19
Registered User
 
🎧 15 years
Quote:
Originally Posted by camus ➑️
1. There is no such thing as analog 0 dBFS. dBFS is a measurement of decibel amplitude levels in digital systems.

2. Why on earth you would want every channel on your DAW to peak near 0 dBFS, only to have to pull all the faders down to ridiculous levels like -25 in order to keep your master buss from overloading, I have no idea.

3. How much you fudge around with levels AFTER they have been recorded doesn't matter as much, as most of the S/N issues with modern 24bit recording lie within the analog gain-staging portion of the production chain. Nudging everything close to 0 dBFS after they are recorded (which I presume is what you are doing) offers zero audible benefit and alot of major workflow headaches later. Especially if you need to spit stuff out to analog hardware anywhere in the process.
2. Why is -25db a ridiculous fader position? Though most of my main tracks will be at -6db or so.

3. What workflow headaches? Please remember I'm talking straight ITB mixing.
Old 6th December 2010 | Show parent
  #20
Registered User
 
🎧 10 years
Quote:
Originally Posted by Morten ➑️
2. Why is -25db a ridiculous fader position? Though most of my main tracks will be at -6db or so.
Because way down there, you lose a huge amount of fader resolution -- becomes much harder to make small, accurate adjustments.

Quote:
Originally Posted by Morten ➑️
3. What workflow headaches? Please remember I'm talking straight ITB mixing.
Because you're giving yourself now room to manoeuvre --- try putting a comp like the Waves V-comp on a track that's peaking at 10dbfs -- you'll pin the GR meter --- now I realise that what you would do would be to pull it's input gain all the way down, but that will make that comp sound much different compared to hitting it with a more reasonable level.


Don't you ever get any problems with noise either? Say you've got a project with say 70 tracks ---- are you telling me that after you've normalised them all to 0dbfs, on playback, you don't hear more noise than you'd like?

Why do you need everything to be up at 0dbfs anyway? Not everything in your mix needs to be that loud, so why go through an extra step before you start mixing, when you're gonna have to turn 3/4 of the elements you previously turned up, down again?! And as someone already pointed out --- do you crank distorted guitar up to 0dbfs? If so, I'd imagine you have quite a time hearing anything else......
Old 6th December 2010 | Show parent
  #21
Registered User
 
🎧 15 years
Quote:
Originally Posted by NeedsMoreFuzz ➑️
Because way down there, you lose a huge amount of fader resolution -- becomes much harder to make small, accurate adjustments.



Because you're giving yourself now room to manoeuvre --- try putting a comp like the Waves V-comp on a track that's peaking at 10dbfs -- you'll pin the GR meter --- now I realise that what you would do would be to pull it's input gain all the way down, but that will make that comp sound much different compared to hitting it with a more reasonable level.


Don't you ever get any problems with noise either? Say you've got a project with say 70 tracks ---- are you telling me that after you've normalised them all to 0dbfs, on playback, you don't hear more noise than you'd like?

Why do you need everything to be up at 0dbfs anyway? Not everything in your mix needs to be that loud, so why go through an extra step before you start mixing, when you're gonna have to turn 3/4 of the elements you previously turned up, down again?! And as someone already pointed out --- do you crank distorted guitar up to 0dbfs? If so, I'd imagine you have quite a time hearing anything else......
You don't seem to be getting what I'm saying at all. Once again, I use the faders to keep tracks where they need to be in the mix. Pre-fader I run close to 0db peak in the virtual signal chain, and adjust the input and output of plugins accordingly. I don't hit the master with 70 tracks of 0db peak. This is obvious from my posts, so why are you going there?
Old 6th December 2010 | Show parent
  #22
Registered User
 
🎧 10 years
Quote:
Originally Posted by Morten ➑️
You don't seem to be getting what I'm saying at all. Once again, I use the faders to keep tracks where they need to be in the mix. Pre-fader I run close to 0db peak in the virtual signal chain, and adjust the input and output of plugins accordingly. I don't hit the master with 70 tracks of 0db peak. This is obvious from my posts, so why are you going there?
As I said in my post above ---- you can pull the input gain of eg a V-comp down, but it will sound different to actually presenting the entire plugin with a lower signal level (closer to the range it's designed to work at)

I really don't understand the point of doing what you're doing, if you're having to juggle the input and output gains all the time? Why not just work with sensible levels?

In your opinion, is there a demonstrable sonic benefit, or a workflow one? IMHO, doing things the way you're doing them is only giving yourself more work. If you wanna make life pointlessly difficult for yourself, be my guest, but I am genuinely interested to know if you think there's some kind of sonic benefit to doing what you do. In my experience/opinion, I feel that by maintaining "proper" gain staging, and sensible levels throughtout, including through plugins (unless of course I'm going for an effect through a modelling plugin etc), gives me a better-sounding final product, but I'm interested to hear your reasons for thinking otherwise.


I can see your thinking behind "using the full bit range", but are you primarily recording at 24-bit, and mixing for cd? If so, you're going to have to knock things down to 16-bit at some point......
Old 6th December 2010 | Show parent
  #23
Registered User
 
🎧 10 years
one way to avoid all your faders down on individual tracks is to put a trim plugin on the master fader...
Old 6th December 2010 | Show parent
  #24
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🎧 10 years
Quote:
Originally Posted by r4uz ➑️
one way to avoid all your faders down on individual tracks is to put a trim plugin on the master fader...
Yes, but why give yourself the trouble of having to insert extra unnecessary plugins, and having to watch your levels all the time, when you could just record at a little bit of a lower level, so then you don't have to push faders/insert trim plugins everywhere to arrive at a decent level? Without getting into the whole discussion of whether it sounds better one way or another (that one will run til the end of time) --- you're creating unnecessary work for yourself. Sure, you may think "it's only inserting a plugin or two, it doesn't take any time" --- but why waste any time/effort at all having to do it when it's so easily avoidable?
Old 6th December 2010 | Show parent
  #25
Gear Maniac
 
Dirk Reinking's Avatar
 
🎧 10 years
to my ears it sounds better if the levels are only in the green !!
also busses summing up and overload the reverb bus..
or the master.. (as an example)
yes we can adjust the levels with the reverb input fader or
a gain plugin but why ?? for me its a BIG improvement to keep levels down.
let your ears decide ...
Old 6th December 2010 | Show parent
  #26
Registered User
 
🎧 10 years
"With every gain there is a loss grasshopper" (pun intended)

I find it ironic that in the age of DAWs that are almost perfect recording devices, with amazing S/N ratios, we have lost a simple concept that came from the common use of VU meters.

Back in "the day" when I was building audio systems, the average best case signal to noise of the line amps (Germanium RCA units) was 80 to 85 dB. Tape, running at 15 IPS, was 68dB, if you lined up the machine properly. So levels REALLY mattered. The noise was always ready to sneak into the sound and distortion was just around the corner.

So the concept I refer to is one called "reference level". Reference Level is not the peak, rather it is the level that the engineer has decided will be used to line up the equipment. It is for sine wave tones ONLY. A typical configuration in a recording studio was to use a level of +4dBm (or dBu if measured un-terminated) With this Reference Level on the VU meter, everyone understood that peaks could go much higher, as in the snare drum example or with a tambourine as another example.

How high do those peak go when the VU meter says zero and it's NOT a sine wave tone being measured. Well... it wasn't exact but the text books said "assume between 6 to 10 dB". However to avoid clipping with solid state line amps, people would design the amps to be capable of +24dBm. An extra 10dB of "headroom".

So there is the magic 20dB! Reference level is +4 (remember this is on the signal wires in the studio) and the clipping range of the amps is +24.

What are missing these days in the DAW is the "Reference Level". The recommendation in this thread is to establish it yourself and use -20dBfs.

And remember Reference Level is measured with Sine waves only.

Hope that helps a little.

BF
Old 6th December 2010 | Show parent
  #27
Registered User
 
🎧 15 years
Quote:
Originally Posted by NeedsMoreFuzz ➑️
As I said in my post above ---- you can pull the input gain of eg a V-comp down, but it will sound different to actually presenting the entire plugin with a lower signal level (closer to the range it's designed to work at)

I really don't understand the point of doing what you're doing, if you're having to juggle the input and output gains all the time? Why not just work with sensible levels?

In your opinion, is there a demonstrable sonic benefit, or a workflow one? IMHO, doing things the way you're doing them is only giving yourself more work. If you wanna make life pointlessly difficult for yourself, be my guest, but I am genuinely interested to know if you think there's some kind of sonic benefit to doing what you do. In my experience/opinion, I feel that by maintaining "proper" gain staging, and sensible levels throughtout, including through plugins (unless of course I'm going for an effect through a modelling plugin etc), gives me a better-sounding final product, but I'm interested to hear your reasons for thinking otherwise.


I can see your thinking behind "using the full bit range", but are you primarily recording at 24-bit, and mixing for cd? If so, you're going to have to knock things down to 16-bit at some point......
My point being: these are sensible levels.
Old 6th December 2010 | Show parent
  #28
Registered User
 
🎧 10 years
Quote:
Originally Posted by Morten ➑️
My point being: these are sensible levels.
lol, ok man, well you've obviously made up your mind, so I guess there's not much more to talk about. In your opinion, you're working with sensible levels -- that's cool, if you're happy doing that, go for it. I completely disagree with you, but hey, we all have different approaches

If you don't mind, I'll quote myself here --- I'm genuinely interested to hear your response to this:

Quote:
Originally Posted by NeedsMoreFuzz ➑️
In your opinion, is there a demonstrable sonic benefit, or a workflow one? IMHO, doing things the way you're doing them is only giving yourself more work. If you wanna make life pointlessly difficult for yourself, be my guest, but I am genuinely interested to know if you think there's some kind of sonic benefit to doing what you do. In my experience/opinion, I feel that by maintaining "proper" gain staging, and sensible levels throughtout, including through plugins (unless of course I'm going for an effect through a modelling plugin etc), gives me a better-sounding final product, but I'm interested to hear your reasons for thinking otherwise.
Old 6th December 2010 | Show parent
  #29
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Mixocalypse's Avatar
 
🎧 15 years
My question is, did you try to work this way yet? proper gain staging.

And why do you insist on working the other way... What's your benefit ?
Old 6th December 2010
  #30
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hobson's Avatar
 
🎧 15 years
Gain-staging and plug-ins

Quote:
Originally Posted by theBF
I find it ironic that in the age of DAWs that are almost perfect recording devices, with amazing S/N ratios, we have lost a simple concept that came from the common use of VU meters.

Back in "the day" when I was building audio systems, the average best case signal to noise of the line amps (Germanium RCA units) was 80 to 85 dB. Tape, running at 15 IPS, was 68dB, if you lined up the machine properly. So levels REALLY mattered. The noise was always ready to sneak into the sound and distortion was just around the corner.

So the concept I refer to is one called "reference level". Reference Level is not the peak, rather it is the level that the engineer has decided will be used to line up the equipment. It is for sine wave tones ONLY. A typical configuration in a recording studio was to use a level of +4dBm (or dBu if measured un-terminated) With this Reference Level on the VU meter, everyone understood that peaks could go much higher, as in the snare drum example or with a tambourine as another example.

How high do those peak go when the VU meter says zero and it's NOT a sine wave tone being measured. Well... it wasn't exact but the text books said "assume between 6 to 10 dB". However to avoid clipping with solid state line amps, people would design the amps to be capable of +24dBm. An extra 10dB of "headroom".

So there is the magic 20dB! Reference level is +4 (remember this is on the signal wires in the studio) and the clipping range of the amps is +24.

What are missing these days in the DAW is the "Reference Level". The recommendation in this thread is to establish it yourself and use -20dBfs.

And remember Reference Level is measured with Sine waves only.

Hope that helps a little.

BF
Often pointed out & discussed. Rarely understood (IMHO).

Hobson

Sent from my iPhone using Gearslutz
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