Gain staging is a very hotly debated topic around the internet!
When getting started in music, gain staging meant something completely differently than it does now. In Ye Olden Days we were concerned with signal to noise ratio and tape hiss. We were always trying to print a signal that was super hot so that we would be well above the noise floor of the console and the tape machine’s hiss.
Later, when first switching to digital recording one of the pervasive theories was that you needed to have a hot signal because printing a quiet signal would produce a degradation in what was being recorded and converted into a digital audio file. Whether that is true or not really doesn’t matter these days because modern convertors are more than capable of accurately reproducing the source that is being recorded! You also don’t need to record a super hot signal, just a clean healthy signal that is above the noise floor! This brings us to our first tip.
- You don’t have to print hot. – What makes this easy in modern recording is that we are all recording at 24 Bit resolution and many are recording at 32 Bit floating point resolution. For the purpose of this article, let’s discuss 24 bit. Recording at 24 bit gives you plenty of headroom to get above that noise floor and still stay very far away from clipping your convertor. With each bit of resolution you have 6dB of headroom. Therefore, in a 24 bit recording we have 144dB of dynamic range to work with. So, you simply need to record a strong enough signal to get above the noise floor.
- Don’t Clip your individual Tracks or Master Buss – Maybe you’re working on mixing a song where you did recorded a really hot signal, and every 2 seconds you’re seeing that red clip light on your master fader. The most simple solution is to simply grab all of the faders and turn them down. 10dB will likely be enough and you can leave your master fader at 0 or unity and should not longer be clipping have extra headroom before the master fader (demonstrated in the video). You want this extra headroom so that you can do common mixing tasks like adding parallel busses for compression or adding effects sends etc. Every track that you have in your mix is adding some signal, and they are all being summed to 2 channels, the left and right output of you master fader. All that signal adds up, so make sure you’re not clipping and have plenty of head room before you start mixing.
- Start with Your Faders at Unity – Last but no means least, when you’re recording keep your faders ad 0 or unity. This is the default position in Pro Tools and many other DAW’s too. This is the optimum position for the faders, they have the most resolution right in the middle. You will notice that the there are numbered markings between the fader and the track meter. These numbers represent the amount of dB’s that the signal is being attenuated by when you move the fader. Notice how the distance Between -10 and +10 dB is nearly half the lenght of the fader, and the other half is goes from -10 all the way down to ∞. The adjustments you are making as you pull the fader get smaller and smaller. So, if you record a signal that is too hot and end up pulling the fader down to -30 to make up for that you will end up making it more difficult to make fine adjustments in the fader such as volume rides on a lead vocal. Give yourself the luxury of plenty of headroom and start by recording with those faders set at 0.
Do you still have questions about gain staging? Leave them in the comments below, I would love to hear from you and be able to help you make your best record yet.
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Here’s my most common challenge. I get a set of tracks and they sum to a clipped level on the master bus. I turn down all the faders and get headroom so that’s fine but I loose resolution on the fader movements. I would RATHER put a fader on the master bus and turn it down the- say – 8 db I need and keep my track faders closer to zero which makes adjustments easier. Any problem with this ? Does it change the resolution or the signal to noise ratio in for the worse ?
Gary does your DAW have a Trim control on each channel? I use the Trim control to set the gain leaving my faders at or near zero.
I am using pro tools. I”m a relative newbie at this so please advise.
Yes in the Mix window at the top of each channel there is a volume parameter for that track that when you click on it a fader will pop up letting you adjust the input volume. That is your Trim control. This is also used for your plugins as well so you don’t overdrive them with a hot signal. If you don’t see it at the top you may have to enable all fields in your Mix view.
Correction! You will only see that on virtual instrument tracks. Since you are more than likely going to be using plugins on each track you can use the output from your plugin as a trim control too. I’m new to Pro Tools too. The help file does discuss the input trim but I don’t know how to pull it up yet.
Okay after researching this you have to use any plugin or if you are like me a channel strip plugin. You can make a default setup for this or add one as needed. I don’t like clutter so I usually use one as needed. I haven’t tried the Avid channel strip yet but I will ( mainly because I have the Waves SSL channel strip). So far I have liked the Avid plugins that I have used. I mix a lot in Harrison Mixbus 32-C and it’s all there since it is a console emulation. Anyway I hope I was able to help.
Great video. Can you talk about the pros/cons of gain staging per track by using clip gain automation. Just boosting or attenuating by a few dB’s until each track is sitting around -18dB, this is just a round about number, but one I’ve seen thrown around on various social media sites.
Great Read and Video
Many forget that proper gain staging starts at the analog front end (and the same rules
apply to analog emu plugins – which are proper made. ie to have a
non-linear gain/signal to noise/distortion curve).
If you take care and do proper gain staging at your analog front end, the level into your DAW will take care of itself (before any processing that is). So many misunderstand this, and think they can record very hot into the DAW and solve it by clip gaining down aftewards. Then the damage to your signal may have already happened.
There’s reason most pro analog gear are calibrated to have their best Signal to Noise ratio (before adding to much distortion) at 0VU = +4dBu = 1.23V (all rms)….. Next in line, most often analog to digital converters are then again calibrated to have this 0VU to be set at -18dBFS.
Next is to learn have a VU meter actually works and respond. As an example, the VU meter responds to slow to really measure a kick or snare drum. When your kick hits the VU meter at -7 to -5VU its peak will end somewhere between -10 and -6 dBFS (about the same place that the max peak would be on a proper tape machine before been “shaved off” (aka tape compression) and started to go into severe distortion (aka tape distortion).
So remember that proper gain staging starts at the analog front end.
Check out hornets plugin vu mk2 meter plugin it stage gains for you!
I have a question about gain staging analog preamps.. I have a 1073 based preamp. it has the stepped controls. when doing vocals what is a good starting point for the input. I know how to set the output not to clip the AI. The problem I have is I’m getting so much different info. Some say run it hot 45 and over and turn the output down, other say for better vocals run the input gain down way lower and turn the output up. I know you get a bit more coloring if the input is hot but, is this really desired in most modern day vocal tracks ?
I keep experimenting with it but haven’t really come to my own conclusions. I thought maybe someone with more experience could give me some advice.
Digital gain staging… I use Digital Performer, which AFAIK has a standard 24-bit plug-in architecture, with 64-bit summing, so I Trim in 12db of headroom to prevent DSP clipping in-line. It’s a tricky subject, which is changing with newer DAW architectures… see this SOS article from a few years ago https://www.soundonsound.com/techniques/gain-staging-your-daw-software
The reason isn’t to give me extra summing headroom on the master bus (which has 64-bit headroom), but rather to ensure transient headroom within the channel’s DSP insert chain. Once you trim audio channels, any following aux channels (which may doing fx, parallel compression, distortion, etc.) tend to look after themselves, if you’re following a basic “unity” gain staging approach, it’s rare that I’ll trim an aux bus to stop its plug-ins overflowing.
– I never see a clip light (unless I’m cranking EQ all the way up to experiment on things) as this means I’ve lost data to overflow
– The weak link is your plug-in’s internal numeric headroom, not the DAW’s summing architecture
Next, theoretically this sort of trim workaround should not be necessary in the ideal world of Pro Tools 11, if you’re working with 64-bit AAX plug-ins. Older VST plug-ins formats are fixed by a “24-bit integer word-in, 24-bit integer word-out” protocol. But, some plug-ins can still have problems in newer architectures, e.g. http://avid.force.com/pkb/articles/en_US/troubleshooting/32-bit-reverb
So, IMHO, what will always be sensible / necessary (for any future DSP architecture) is to check gain staging by bypassing plug-ins to check audio is as loud in bypass as when enabled (or at least, whatever you intend). This is mainly to give better subjective listening, avoiding loudness effects, but also to prevent accidental clipping in the DSP chain when adding in processing stages later on, as well as avoiding headaches and re-work when re-ordering the chain.
@gambrosino:disqus … good question about signal to noise ratio… the raw system is silent enough for 20db of headroom not to be an issue, but when I’m adding headroom artificially using Trim, I do need to dial back the “analog” setting on modelling plug-ins that add noise to simulate real world circuits. Presumably this is because the plug-in is adding in noise 12db louder than designer anticipated – I also tend to use automation on master bus faders to tail out stems to keep intros / outros cleaner.
Perhaps a Q for @Warren on NOISE … the “analog” button… what does it do, e.g. on an SSL channel strip, other than add noise? The recent video on master bus suggested the API 2500 behaves quite differently on its analog setting – it’s v subtle 🙂 TBH, unless the “ANALOG” setting is adding someting sonically, most of the time I turn it off (eg. SSL channel strips and Neve VEQ / VComp).
Another well done video and very helpful info explained very concisely. I’m not sure which is cooler though…the video or the Ace Frehley guitar strap in the background 😉
Thanks for another interesting video Warren. I just joined PLAP and I am stoked. I am using RMS metering and try to average the recording level to -18 or so for an optimal level. When my fader is red while recording I adjust this to mix my headphone level for general monitoring. Am I OK?
Thank for being out there.
Thanks Warren! I’m still scratching the surface, especially with regard to this subject. I generally don’t gain stage unless I have a problem with a particular track, or tracks clipping, or near clipping. Then I’ll gain down to get the desired result, maybe only -5, -10 db. If it is a really hot track, I’ll gain down -20 to get it to work the way I want, not slamming the plugins, etc… I guess my question is, is that wrong? A friend of mine gains down all his tracks -18db out of the box, first thing. I haven’t found an easy way to do that in my DAW (still learning) and I’ve found that the stuff I recorded myself, I don’t have to do anything with as far as gain goes. It seems to be an issue with tracks I get to mix though, some are just hot. When they’re messing with the mix, fighting the plugins or me, I’ll gain it down then. I guess my approach has been, if it’s bugging me, fix it. I’m new, so it’s one of the first things I do starting a new mix after dragging in all the files, I’ll drag all the faders to off, play back and just look for the hot tracks and gain down all the tracks that peak, then I’ll gain down by how much they peaked, plus 5 more db to start and see who still clips. When the session plays through without any peaks, time to roll up the faders to see what’s up… Maybe I’m wrong about this, and I really need to take the time to learn the values of all the ways to look at volume, it”s a bit confusing to a noob 🙂
Hey Warren I totally appreciate the video’s you do, it’s an awesome thing that you’re doing by sharing all of your knowledge free of charge, I personally have learned a ton from your videos… thanks so much… Roger in Vermont…