# Gain staging - my biggest sticking point



## ein fisch

i always mess my levels up so either stuff starts clipping or stuff is way too quiet during the mixing phase.. either way its very time consuming to spend 30min every second day just fixing volume levels..

so ima just go ahead and ask straightaway: do you guys have any workflow tips to share with me when it comes to gain staging? are you also making as big of a mess as i do?

cheers


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## Henu

Start your staging from the most loudest/ prominent instrument being at about -6dB or less and balance everything around it.


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## Morning Coffee

You could treat your DAW like an analogue mixing console in the sense that you could put a trim style plugin as your first plugin or even at the end of your plugin chain on each channel. This gives you an additional level of control as it allows you to increase or decrease the sound level before it even hits the faders in your DAW. 

Here is the same explanation


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## wst3

Gain Staging is not a simple matter, and while the rules from the dark ages (analog?) apply, but not completely. To complicate matters, not every DAW uses the same signal flow. So it gets tricky.

The goal, in either domain, is three-fold:

maximize signal to noise ratio
minimize clipping
provide adequate headroom from input to output
And like I said, it differs from DAW to DAW.

In the analog world one would take roughly the following steps:

trim the microphone input level so that it doesn't clip on the loudest peak, but only by a smidge. We do this because the first stage will provide much of the character, but also the lion's share of the noise.
Set the output of the channel - after all processing - to some predetermined level to maintain headroom in the summing bus.
For each processor, in-line or aux, do the same thing, except you'll be managing input level, not trim, for most plugins.
OK, that is grossly over-simplified, but it should get the idea across. Feel free to ask questions and I will feel free to write volumes<G>!


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## ZenFaced

Can someone clear this up for me? There are several well respected professional mixers who claim that when mixing ITB, unless you are trying to hit a sweet spot on certain analogue style plugins, gain staging is not necessary on each individual track and it's only the output buss you need to have the headroom on. So you as long as you have a limiter or trim on the output buss then it doesn't really matter how hot the individual tracks are because in the digital realm there really is no clipping until the digital signal is being converted back out to an analogue output signal.

Of course you would want gain staging on the way in (mic, preamp, etc) and get a nice recorded signal but once in the digital realm gain staging is not necessary except for the output bus. Is this correct?


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## JohnG

ZenFaced said:


> gain staging is not necessary on each individual track and it's only the output buss you need to have the headroom on. So you as long as you have a limiter or trim on the output buss then it doesn't really matter how hot the individual tracks are because in the digital realm there really is no clipping until the digital signal is being converted back out to an analogue output signal.



That ^^ is oversimplified and partly just plain wrong.

Follow @wst3 Bill Thompson's advice above and that way, whether you have all in the box or a mix of external / live signals and in the box you will have a good starting point.

The only thing I'd add is to set your monitoring level so it's always consistent, and something with which you're comfortable. To do that, get a VU meter (like from Radio Shack), decide on your overall sound monitoring level, and then set your amplifier so it's at that level using pink noise. 

I use 85 dB but some people use a different (much lower) level with great success.

That way, when you want something loud, you know it's "loud enough" because you're used to your monitoring level and can hear it.

Once you've set your overall monitoring level, you can use your ears, which is better than relying on meters.


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## Wally Garten

ZenFaced said:


> it doesn't really matter how hot the individual tracks are because in the digital realm there really is no clipping



I've heard the same advice -- that there's so much overhead in modern digital channels that you don't really experience clipping on individual tracks. I think technically this is true, and I've certainly let it go occasionally rather than trying to eliminate all peaking during recording and initial mixing -- usually on vocal tracks where there's so much variation in level that I know I'm going to end up needing to compress it a lot anyway. It won't kill you. _But_ usually I'd do what I John says and try to get the levels right, track by track, _because_....



ZenFaced said:


> as long as you have a limiter or trim on the output buss



This is a double-edged sword. I put a limiter on the master on most tracks near the end of mixing, either to tame random little peaks, boost the soft bits, or both. BUT if you put the limiter on too early (i.e., before you've got the mix mostly right), relying on it to fix the levels on individual tracks, sometimes it really just hampers you. You start relying on the limiter to bring everything up or down, but you end up going too far and it all sounds squashed and weird. Or there ends up being one problem track that you can't hear in the mix no matter what you do, unless you dial back the limiter and let the master clip. I've had all kinds of freaky problems from relying too much on the limiter. (Just my experiences, of course, and I'm self-educated and still learning mixing.)


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## karelpsota

ein fisch said:


> do you guys have any workflow tips to share with me when it comes to gain staging?
> cheers



Me techniques would make engineering schools scream but that's how I get the best results.

Basically, I do the "*Skrillex* *way"*. Where the loudest elements (drums) hit 0dB. Then I mix everything around it.

Note that put a *hard-clipper on my master, so I can hear when stuff distorts*. That is particularly handy to make the bass + kick problem obvious.

*A master limiter will transparently absorb the problem*, *which is* *not good* *for mixing*. You want to anticipate every possible problem.
*Too much headroom will also not show you some obvious problems*. (like a sub hit that's 6 dB louder than what mastering will allow)
Once, my mix is done. I glue everything together with a multiband comp + limiter. If I want to preserve perceived transients, I'll replace the limiter with a clipper.

EDIT: This is how I mix/master anything that has drums.
Orchestral and ambient track are usually really quiet mixes... as you should! You can just bump that 10+ db into a limiter later.


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## Synetos

What do you set the Master Stereo output level to read on the DAW meter for dBFS? 

This has always confused me. If -18dBFS is equal to 0dBVU (analog), what I should my stereo bus be reading when I am setting my room monitor level to 85db with my SPL meter?


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## robgb

JohnG said:


> That ^^ is oversimplified and partly just plain wrong.



Kenny Gioia demonstrates in this video that the only thing you need to worry about is master buss distortion. The only exception to this is if you are using analog style plugins on a track that might react negatively to higher levels.


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## Divico

karelpsota said:


> Basically, I do the "*Skrillex* *way"*. Where the loudest elements (drums) hit 0dB.


Sry but this is too much. If your drums alone hit 0dB than drums+rest will be above.


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## ZenFaced

robgb said:


> Kenny Gioia demonstrates in this video that the only thing you need to worry about is master buss distortion. The only exception to this is if you are using analog style plugins on a track that might react negatively to higher levels.




That's exactly what I was getting at in my post above and what I thought was true. Thanks for clearing that up! BTW - Kenny Gioia knows a hell of a lot more than that Graham guy. It's like pro vs amatuer mixing.


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## GtrString

Well, its not that hard. I trim every track at the start of a session to -12dbfs, and usually have about -6dbfs on my master bus for a mix, before mastering.

I monitor everything closely (both mix and master) with the Levels plugin from Mastering The Mix. It's brilliant.


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## DS_Joost

Henu said:


> Start your staging from the most loudest/ prominent instrument being at about -6dB or less and balance everything around it.



This quite simple piece of advice is secretly really, really fantastic. -6db for your loudest drums or trailer hits and then work your way down from there. Listen to an orchestra. There is so much difference between loudness of instruments. You are not mixing pop, you are mixing the most dynamic of man's creations.

The -6db level should be used only and only for the hardest, loudest hits played at their loudest. I would suggest keeping the rest of the instruments between -12db or -18db, depending on preference in dynamics.

What's important about this ''rule'' (there are none but this is a very good line to draw so to speak, I find) is that you shouldn't just look at the volume of your instruments that are playing. You should consider the accumulated loudness on your master bus. That's where you want to have headroom when mixing, not on the individual tracks. If your summed signal already reaches -3db before even starting to mix... well good luck.

Try to stay conservative with gain. You can always add volume, but it's very hard to turn things down once you are used to them.

Plus it's silly fun throwing Waves L3 on the master bus and cranking that thing into oblivion! Such a fun plugin that one!


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## rhizomusicosmos

robgb said:


> Kenny Gioia demonstrates in this video that the only thing you need to worry about is master buss distortion. The only exception to this is if you are using analog style plugins on a track that might react negatively to higher levels.



Yes, this is the benefit of 32 or 64 bit floating point internal DSP resolution: thousands of decibels of dynamic range.

Some practical caveats from my perspective, though:

If you redline all your individual channels, then you are not using the channel meters in their optimal range. You lose a lot of the visual feedback that helps you balance instrument levels.
As soon as you drop in a plugin that is not using floating point DSP (rare these days, I know), it will probably clip.
When you mix ITB sound sources with live input processing or routing to external hardware you have to remember that ADCs and DACs are integer and won't have the same headroom.


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## MartinH.

When I load in a reference track that I bought as a FLAC file and the meter on that track shows it goes to +1 or maybe even +2 db, what conclusion am I to draw from that? I always thought I'm not supposed to have my exported final audio file _ever _go above 0db, is that wrong? I'm talking about a metal album, in case it matters, not music for a film or game that would be layered over other audio.


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## DS_Joost

MartinH. said:


> When I load in a reference track that I bought as a FLAC file and the meter on that track shows it goes to +1 or maybe even +2 db, what conclusion am I to draw from that? I always thought I'm not supposed to have my exported final audio file _ever _go above 0db, is that wrong? I'm talking about a metal album, in case it matters, not music for a film or game that would be layered over other audio.



It means that track is so compressed into oblivion that you should never use it as a reference track. And that you should get your money back... That track is distorting, it has to.

Either that or your meters are not set to dbfs. In which case it doesn't matter


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## MartinH.

DS_Joost said:


> It means that track is so compressed into oblivion that you should never use it as a reference track. And that you should get your money back... That track is distorting, it has to.
> 
> Either that or your meters are not set to dbfs. In which case it doesn't matter



Oblivion is fine, I want to make blackmetal after all ^^. I picked that album because I really liked the sound. I don't know what dbfs means or where I would set it up. Any advice? I googled reaper and dbfs, but didn't see any clear info. As far as I can tell it is dbfs already and there were different kinds of confusions among users about repears metering.

I'm going for a (brick)wall of sound aesthetic, no dynamics needed. I just don't know how I should tweak the master limiter to have my pink-noise-like brickwall at the "correct" height. I thought it's -0.1 db, but to a noob like me it's really confusing to buy a commercial track and have it register above that on the meter.


Here are the meter settings in reaper:












This is one of the songs from the reference album that I'm talking about:




And this is what the master meter looks like when I play it back in reaper:


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## rhizomusicosmos

Hi MartinH. The decibel scale is purely a relative one. Saying "0 dB" by itself only makes sense within a particular context, e.g. voltage levels, power levels, acoustic loudness levels, digital levels, etc. 

You will often see a suffix added to dB in order to tie "0 dB" to a particular reference level: dBV, dBv, dBu are tied to voltage references, dBm is tied to an electrical power level (1 milliwatt), dB SPL is tied to sound pressure level, and dBFS is tied to the "full scale" maximum level that can be represented by the digital data format (i.e. all "1"s). See this for an overview: http://www.jimprice.com/prosound/db.htm

The Master level meters you pasted from Reaper have two readings: 

The inner bars are peak meters that show dBFS. So 0 dB on these bars is the digital max and you're right, these should not go into the red.
The outer bars are RMS meters the behaviour of which and how they relate to 0 dBFS is set up in the Master VU settings under "RMS metering settings". 

The Display Offset of 14 dB means that 0 dB on these bars is -14 dBFS which practically means you have 14 dB of headroom/red above 0 dB before you hit digital clipping. If you set this to 12 dB you will reduce that headroom -- think of it as moving your 0 dB on the RMS meters closer to 0 dBFS and thus making it "louder".
The Display Gain compensates somewhat for crest factor. If you set it to 0, then there is no compensation and a sine wave (often used for setting levels between gear) will have a 3 dB level disparity on the Peak (inner) and RMS (outer) meters. Set it to 3 and they will match.

So why is your Metal track showing as peaking above 0 dBFS (+1 dB on the inner meters)? This is because the meters are warning you of probable inter-sample overs. This is good and means the Peak meters in Reaper are giving some indication of True Peaks (https://www.masteringthemix.com/blogs/learn/inter-sample-and-true-peak-metering). These overs can be produced in some DACs when they reconstruct the analogue waveform for output and can sound as clipping distortion on peaks. How bad this sounds is entirely dependent on the DAC and some deal with it quite gracefully and some are unforgiving, so YMMV.

I suspect the track you are using as a reference is rather aggressively brickwall limited at 0 dBFS. My advice would be to re-normalise the track in Audacity or similar program to allow some margin -- maybe to -1.0 dBFS or even -2.0 dBFS -- and then use it as a reference in Reaper. Similarly try to keep your own tracks from going into the red on the Peak meters. You can then still have the brickwall sound without the uncontrollable nastiness of DAC-related inter-sample peaks.


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## oks2024

I just saw this video today, there is a lot of information about loudness, db, RMS, LUFS, and mastering for different platforms:


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## MartinH.

rhizomusicosmos said:


> Hi MartinH. The decibel scale is purely a relative one. Saying "0 dB" by itself only makes sense within a particular context, e.g. voltage levels, power levels, acoustic loudness levels, digital levels, etc.
> 
> You will often see a suffix added to dB in order to tie "0 dB" to a particular reference level: dBV, dBv, dBu are tied to voltage references, dBm is tied to an electrical power level (1 milliwatt), dB SPL is tied to sound pressure level, and dBFS is tied to the "full scale" maximum level that can be represented by the digital data format (i.e. all "1"s). See this for an overview: http://www.jimprice.com/prosound/db.htm
> 
> The Master level meters you pasted from Reaper have two readings:
> 
> The inner bars are peak meters that show dBFS. So 0 dB on these bars is the digital max and you're right, these should not go into the red.
> The outer bars are RMS meters the behaviour of which and how they relate to 0 dBFS is set up in the Master VU settings under "RMS metering settings".
> The Display Offset of 14 dB means that 0 dB on these bars is -14 dBFS which practically means you have 14 dB of headroom/red above 0 dB before you hit digital clipping. If you set this to 12 dB you will reduce that headroom -- think of it as moving your 0 dB on the RMS meters closer to 0 dBFS and thus making it "louder".
> The Display Gain compensates somewhat for crest factor. If you set it to 0, then there is no compensation and a sine wave (often used for setting levels between gear) will have a 3 dB level disparity on the Peak (inner) and RMS (outer) meters. Set it to 3 and they will match.
> 
> So why is your Metal track showing as peaking above 0 dBFS (+1 dB on the inner meters)? This is because the meters are warning you of probable inter-sample overs. This is good and means the Peak meters in Reaper are giving some indication of True Peaks (https://www.masteringthemix.com/blogs/learn/inter-sample-and-true-peak-metering). These overs can be produced in some DACs when they reconstruct the analogue waveform for output and can sound as clipping distortion on peaks. How bad this sounds is entirely dependent on the DAC and some deal with it quite gracefully and some are unforgiving, so YMMV.
> 
> I suspect the track you are using as a reference is rather aggressively brickwall limited at 0 dBFS. My advice would be to re-normalise the track in Audacity or similar program to allow some margin -- maybe to -1.0 dBFS or even -2.0 dBFS -- and then use it as a reference in Reaper. Similarly try to keep your own tracks from going into the red on the Peak meters. You can then still have the brickwall sound without the uncontrollable nastiness of DAC-related inter-sample peaks.





oks2024 said:


> I just saw this video today, there is a lot of information about loudness, db, RMS, LUFS, and mastering for different platforms:





Thank you so much for the detailed description and the video links! I think I get it now and I finally understand what that "true peak" setting in Ozone Elements means. I tended to activate it because it says "prevents clipping in the analog domain", which sounded good to me.
One of the videos also said that for combatants in the loudness wars it wasn't too uncommon to clip +1 db there, so that might explain the levels I'm getting from the reference track. I think I'll just turn down that reference track a little and set my own limiter to target -0.1 dbfs true peak, since I have no need to "compete on loudness". Thanks a lot for your help! This made a couple of things click for me.


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