Do you struggle to get good mixes in the box?
There are many reasons why this could be, especially if you’re coming from the analog world (or have grown up learning from engineers who mixed on a console).
But there is one key issue when it comes to mixing (and summing) in the digital domain and it’s a simple one to fix, as I explain in today’s video.
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My name is Graham Cochrane and I'm the founder of The Recording Revolution, one of the world's most loved audio recording and mixing blogs with over 200,000 readers each month. 
Hey Graham,
The more I apply your “Technical tips” (The boring stuff), the greater improvement I see in my mixes. This week for me it was hearing a mix fall apart on a different system because I didn’t mix it in mono. All of these little things make a huge difference. It’s like EQ, Compression, mixing in mono, and gain staging are the cake, and everything else is just icing.
Thanks for another great tip!
Hey Graham – I just wanted to say how much I enjoy your videos, tips and tutorials – Really excellent stuff sir – keep up the great work. I’ve been an engineer for a while now and have worked on some cool records – but I have only just set up my own ‘mix room’ – so I’m a little late (at 46 years old) in joining the ‘mixing party’ ! Gain staging in the digital domain when tracking is something most producers are becoming wise to now in order to leave sufficient headroom for mixing ITB … I know for a fact that the last producers I have made records with as an engineer have been super vigilant at checking the levels of everything hitting the Pro Tools Inputs ! So it makes total sense to employ the same mind set when mixing ITB …. It has to be said that your point on using Clip Gain and / or a Trim Plug in to keep each channels levels conservative enough as to not exceed 0dbfs when summed ITB is such a great shout – as I have no idea about the ‘resolution’ deficiencies of running some channels with their faders set really low 😉 Anyway – Thank you for taking time to make such informative and helpful videos & tutorials !
Mike
If your recording your signal at around -18 to -12db there really should be no problems with headroom at all. I’m usually peaking around -10db when the mix is at its loudest on the mix buss. At the mastering stage my rms level is at around -18 to -14 which is nowhere near the levels mastered compared with most of today’s music. I’ve heard indie folk music that is crushed to death.
There is not a more important subject than this when working with digital audio.
Hi Graham
Don’t I have to set a new Thresholdlevel (Compressor…) if I just lower the levels of the files
Thanks
Couldn’t you bring down the levels of your groups instead? And do I see your using the Virtual console (VMR) on only your groups?
Probably using VCC, 2.0 which is embedded into VMR now.
Graham, another great tip,..thank you! When I’m mixing and I’m not sure if this will help or if this is what I should be doing, but it seems to help me a lot. I get a “good mix” watching my master fader meters all the time to make sure I’m not in the “danger zone” and I have headroom. I like to mix like you do at -15 to -18. If I’m near that, I just leave it alone and keep watching the master fader meters as I mix and add plugs, but If I don’t have enough head room, I will group all the tracks together and pull down the track fader and in-turn it will pull down all the track faders at once. I look at master bus when I’m doing this to gain back the headroom again and I turn up my speakers to compensate. Do you recommend grouping all the tracks and pulling down the faders this way? Seems to work for me, but I just want to make sure since you’re a pro!! Thanks again for all you do for us.
Hey John,
This can work, but remember Graham’s comment in the video about fader resolution. I personally like to have my fader’s close to zero so I have the maximum resolution. When I set up my static mix, I use gain plugins and the gain of my audio regions to ensure that I don’t need to move the faders any more than 5dB or so to get a good balance.
This is just a personal preference, but something to think about.
Alex
Alex, thank you. Yes I did hear that in the video. I will try trim plugs. Kind of strange though, because I never had any issues using the faders in groups before.
Graham, at what point on the fader do you start to lose resolution?
Hey, if you’ve never had any trouble with it, then don’t worry about it!
In Logic, my fader goes up to +6dB and when I pull it any lower than -15 to -20dB I start to lose enough resolution that it annoys me. Of course, every system is different though.
Awesome tips here! I’ve definitely been seeing a pretty quick improvement when I use these. Thanks so much for sharing this video!
More of an observation, but it seems there’s no industry standard for ITB gain staging. I follow the advice of Graham and people like Fab Dupont that suggest recording and mixing individual tracks around -18 db but find it frustrating and wonder why default software instruments and channel strips in DAWS like Logic Pro X default to very high gain (e.g. -6 db+). I often find myself spending time restructuring DAW default instrument tracks to bring the gain down so they match my lower gain audio tracks, and so software instruments don’t slam plugins like Slate VMR that sit in the beginning of the track processing chain. But by the time I’ve restructured gain on default instrument tracks (e.g. attenuating software instrument volume, resetting track compressor threshold values, etc.) I often feel that some of the original track mojo is lost and in the tweaking process I’ve lost perspective on the larger mix. Most frustrating. Wondering if anyone else faces these issues with disparities in default instrument and plugin gain staging and how others deal with it?
I’ve noticed this in Logic’s instrument tracks too. I haven’t dealt with them much, but they’re definitely loud.
One way to deal with it could be just using volume faders or gain plugins. Like, if you’re instrument is feeding too loudly into VMR, but turning it down at the instrument level doesn’t sound good, maybe just slap a Gain plugin in the chain just before VMR and turn it down there? That way to retain the original sound of the instrument, but don’t feed it into VMR at too high a level.
Something to think about, anyway.
Alex
Thanks Graham!
Personally, I use “K-system metering” and mix at K-20. What this means for me is that I calibrate everything so that RMS -20dBfs hits my ears at 68dB of loudness. I usually mix the loudest part of the song to be in the red just a little bit on my K-system meter, meaning the RMS level is at about -16dBfs or so. Plenty of headroom, and my mixing volume in the room is consistent across mixes, which is nice.
A little hack that I use is once I get everything calibrated, I’ll play the loudest part of the song and see where the RMS value is on my K-system meter. Then I’ll highlight all my tracks and turn down all of their gain values at once until my K-system meter is reading around +2 to +4 RMS (which translates to -18 to -16 dBfs). That at least gives me a good start, and means I don’t have to mess around with faders quite so much.
I remember when I first started recording and mixing. The thing that I read *everywhere* online was to record things as loudly as possible without clipping. Of course, then I would have to yank all of my faders down because the master fader would be clipping all over the place. It was a real blessing to finally discover that this isn’t true in a digital environment, and recording and mixing at conservative levels is ok. It’s made a huge difference.
Thanks Graham! Great video.
Alex
Hi there Alex,
Are you talking about the metering system by Bob Katz?
Isn’t it for setting the appropriate monitoring levels for tracks with different compression rates?
I also use my own adapted Katz metering system and with many years of experience I too can blindly set my levels pretty acurately, without the visual aid of my meters (I always still check it, though).
But I don’t think that the mayority has the capacity that you have to succesfully adjust their gain levels this way, even with the use of the K-metering system; that takes experience.
Kind regards,
Hey Norman,
The K system is pretty in-depth, but I use it for calibrating my monitoring levels so that every time I mix a song, it’s at the same volume. I mix at K-20, which means that -20dBfs appears on my meter to be 0dB. The idea is that your mix’s RMS level should be around 0dB, pushing up to about +4dB in the louder parts.
To calibrate, I feed pink noise into my monitors, one at a time, and set the level (using a dB meter app on my iPhone) to 68dB. So again, for every song that I mix, as long as I set up my static mix so that my tracks are hitting around 0dB on my meter, they will be hitting my ears at around 68dB (actually slightly more since I calibrate each speaker at 68 individually). But the point is that it’s consistent, and I have plenty of headroom.
Hope that helps!
Hey Alex,
Thank you for the explanation; that’s sort of how I use the K-metering system too.
But you calibrate (pink noise) each time you mix?
I only did that once when I set up and marked my different listening levels.
And indeed it is very consistent.
Hey Norman,
Yeah, I calibrate each time I mix. It’s totally unnecessary. I should just put marks on my interface. But I just haven’t gotten around to it. Absolutely no need to do it every time, I’m just weird like that 🙂
Alex
The K-system sounds pretty interesting. Ill have to check that out. Also Fab Dupont will bring down his master fader to something like -6dbfs before he even starts mixing so he can keep his channel faders closer to zero. The mix buss itself has ridiculous headroom the problem is generally clipping your output converters not the mixbus itself. Check out this great SOS article: http://www.soundonsound.com/sos/jun10/articles/pt_0610.htm
good stuff graham
Very good advise once again, Graham!
However, in my experience as an analog OTB caveman, it is just engineers that have been trained on analog gear that do recognise and understand the importance of proper gainstructure.
Once they see the difference between the digital peak metering and the traditional VU RMS reading and what dBFS level corresponds with ‘0’ VU, they’re good to go.
More often I come across engineers that have no, or litle, OTB experience that are confused about the ‘headroom’ and gainstructure issue.
For those interested:
-18dBFS coresponds with 0VU on a (+4dBu) system that is rated for +22dBu max levels (22-4=18).
In such an environment, you thus have a headroom of 18dB.
For a system that can handle +28dBu maximum levels, ‘0’VU would equal -24dBFS (28-4=24), leaving a 24dB headroom.
This is the reason why some digital equipment advise -24dBFS, while others aim for -18dBFS (or -22dBFS).
Regards,
Hi Graham,
when I am in the middle of my mix, and need a track to be louder oder quieter I unsert a trim-plugin too, but I am getting used to insert as the last insert right before my fader in order to not change the gain going through my already inserted plugins.
Cheers!
Andreas
Hi Graham, Do you have (a video) of would you consider doing one about summing in the box?
Many thanks for all your your informative video’s. Db
I never watch my faders and I always end up wanting more volume for my recordings. As a result when I share my music with the world there is a lot of digital distortion and it sounds terrible. I’ve heard this advice before, but never utilize it. Time to start fresh. Can anyone here help me out with bringing up the volume during the mastering stage (step by step, plat by play) using a stock Limiter in Logic Pro-X? Was never too sure how to do that…
Hey Corey,
here s what I like to do:
1. route all of my tracks to a bus (I use bus 64 ALWAYS 🙂
2. insert the following chain of insert Plugins: Gain / Limiter / Gain
3. pull down the output of the Limiter to -10 and set the Gain of the Limiter to +10
4. Now you can drive the Limiter with the Gainplugin inserted before the Limiter, while it will clamp down every peak passing -10dbfs (almost)
5. set the Gainplugin inserted after the Limiter to +9 (leave 1 db of headroom for intersample peak spikes!!!) this will boost it to “Masteringlevel”.
6. you can easily switch between “Mixlevel” and “Masteringlevel” by enabling the final Gainplugin AND you can listen to your Mastered /Limited Mix at “Mixlevel” 🙂
7. all of that ONLY works if you do what Graham talked about in the Video.
🙂
Carl
That’s Great ! Thank so much Carl and Mike! I saved all that info and will try it out this weekend for sure!!! I am definitely guilty of never using a reference track, though the concept makes a lot of sense to me. I’m just lazy and afraid I’ll get confused or overwhelmed. #coward
Reference Track sir. Get a good reference track and bring it into your DAW. Then you bring the gain of the ref track down to match the volume of your mix. After that make any EQ changes to your mix to get it in the ball park (it will never match 100%). The balance, lows, highs and overall body should feel close. Compress it GENTLE to tame peaks or give it a little more oomph if necessary. Then set a limiter to just below 0db and start wrenching down. Don’t pulverize it, just bring it up to where it is knocking off a few dB’s of peaks.
There ya go – now everyone can master anything, right??? 🙂 jk
It takes patience and there is no real recipe. You have to listen and tweak just like mixing.
To make this stage a ton easier, why not pull in a reference track during mixing? It sounds like you don’t do this, but it will give you that ‘instant gut check’ we all need.
Good luck!
Helpful video Graham, thanks. I would like to ask you, what is the reason for having a “sub-mix”, a “mix-bus”, and a “Master” bus in your DAW? I use Sonar Platinum and do all my “pre-master” mixing on this one Master bus only, using its inserts. Would like to know your logic for having these three buses at the end of your channel chain. p.s. Blue Cat Audio has free and accurate gain plugins for those that don’t have them built in to their DAWs. (Sonar has them at the top of the Console strips). Appreciate your work and concern for us “in the box mixers”!
Would you know the name of the gain plugin from Blue Cat Audio? I use an older version of Sonar. Sonar 8.5 producer. The only gain I see is in the track view. No plugin. I’ve used that a few times, but find it much easier to group the faders to lower the master bus output if necessary.
Thank you…
Seems I found it Blue Cat’s Gain Suite. Is that the correct plug?
Yes
In the top of the Pro Channel is a gain trim knob. You might need a newer version of Sonar to have it.
Yes, I need to upgrade to Sonar Platinum, but I need to upgrade my PC also. I’m still running XP with only 4 gigs of ram. It’s fine and I have no issues running 8.5, but to run Sonar Platinum, I need windows 7 or above and at least 8 gigs of ram. PC’s are pricey!! When I do the upgrade, I really want solid state drives. I heard they are incredible and very fast for audio.
Sorry John, but I meant the “gain trim knob” is on every track at the top of each channel in the console view (mixer view) of Sonar. The Pro Channel is within the channel strip. I think Sonar X1, X2, X3 is when the gain trim knob was introduced. I am very happy with the new Sonar Platinum. Much more reliable and a better sounding audio engine than ever before. I started on Sonar 8.
Another good reason to adjust clip gain pre-fader is the to control how much signal is going into your plugins. Plugins will react differently at different gain levels. Hit that sweetspot (like -18 to -14dB) and your plugins will thank you with sweet consistency 😉
Wisdom delivered on a silver platter !
Man, I wrestled with this concept for YEARS! ! !
It really took a while to sink in.
🙂
Hornet plugins? Yes!
Hey guys
Thanks for the helpful video Graham.
When recording DI guitars, the guitar’s signal is so strong (especially if you have high output pickups), that even with interface gain on 0 I get pretty hot tracks, not clipping of course, but still hot. I’m using Sonar and it has pre-fader gain knobs, but I don’t use them for this, because I use amp sims and reducing the gain before the plugin will change the sound. I always used faders, pulling them down from -3db to -10db. So my question is this. Is this range (-3db -10db) considered as close to zero, or should I export my tracks with amp sims on them and with reduced volume? Sorry if this doesn’t make sense.
Thanks.
The only level that matters in the level going in while you’re recording. The trim knob and fader in Sonar doesn’t affect the recorded level. If you’re peaking at -3 you might be clipping at some point too. You might want to invest in a passive pad which will attenuate the signal before it hits your interface. A conservative signal will make your sims sound better too.
Thanks for replying.
I didn’t mean peaks on -3db, I mean my fader on -3db. Also I can turn down the volume knob on the guitar itself, but it will affect the sound if the amp sim drastically.
Hi Adam,
It is especially your pre-fader knobs that should be used to set the proper gain (and monitoring the levels on PFL).
Your amp simuation should have a drive or input level knob to control how hot you want to drive the amp sim.
The channel faders then are used to balance the amp simulated guitar track with the rest of your instruments in the mix.
Maybe I didn’t understand your question correctly, but not all your channel faders need to be close to the ‘0’ marking. If your gain was set right (0VU/-18dBFS/-22dBFS/-24dBFS), all faders close to ‘0’ would mean that all your tracks/instruments are equally loud in the mix.
The range between the ‘0’ marking and -10 (more so between -5 and -10) give the best fader resolution; you can see that the distance between these points are bigger than the other positions on your faders.
I hope this helps.
Thanks Norman. So, if I get it right I’m not in a trouble, because my tracks aren’t even close to clipping and my faders don’t go below the -10 marking. I just wanted to know is it normal to have a fader on -10, or is it too low.
Thanks everyone.
Clear and concise. Clarity is wonderful thing.
Thanks Graham!
Since I starting using -18 dbFS as a general rule for RMS levels, I record with peaks never going over -12, and adjust soft synth volumes in the plugin’s so that levels are conservative. Makes for better clarity and room to play. It’s nice too that Sonar has a trim knob at the top of every track. You can always slap on a limiter at the very end of your chain to get that final boost in volume. Essential for anyone mixing in the box to know this.
-12db is a great number to strive for. There is no need to go any higher.
Quite frankly we need instruction on how to to get reasonably loud mixes when the audio signal is at -5db on the master-bus. Obviously at this level the volume is too low to be competitive, so how do we now get things loud without having to abuse a limiter? That’s what we need to know.
Get all the levels where you want them to sound their best before mixing down, making sure your Master fader doesn’t exceed -4 to -6 dB. You want from -6 to -4 dB left as headroom for your final (pre-master) mixdown. The mastering engineer needs this headroom to do his/her magic and bring the levels up to “pro” commercial levels.
Ideally, this should be handled in mastering, not mixing. In mastering, you use a combination of EQ, Compression, and Limiting to increase the loudness of your track (I include EQ because a good mastering engineer can use it to increase perceived loudness without actually making the track much louder).
I find that rather than “abusing a limiter”, it works well to squash the dynamics a little bit in multiple stages. When I’m mastering I’ll usually have at least a multiband compressor, a regular compressor, and two different limiters, each taking off 1-2dB of gain reduction. To me, this sounds more transparent than working a single limiter really hard.
But it’s also worth noting that “competitive” volumes aren’t the real point of mastering, and I think it’s a shame that the world of digital music has put so much emphasis on loud masters. In the digital world, we have sooooo much dynamic range and headroom. It’s sad that the industry has forced engineers to squash the dynamics of music so much that it’s only using a tiny fraction of the dynamic range that we have available.
That said, when I’m mastering, I try to get it as loud as I can, while maintaining the dynamics as much as I can. Sure, my masters aren’t going to be as loud as a commercial master. But it’s sure going to sound a lot more dynamic.
Hope that helps!
Alex
Thanks for all the amazing videos Graham – great to have your pearls of wisdom 🙂
i too like Curtis turn my group buses down by -6db if all getting a bit too hot on the master….does this not work as well and if so why not? it’s a lot quicker that adding individual trim plugins to each track!
Hi Graham,
I have been watching your videos for quite some time and must thank you for all the great tips you have shared technical and musical but none more than the philosophy behind your approach. It has been inspirational and motivational. On the subject of this very video and learning to be conscious of overall levels thru the mix, I worked out a good, {slightly higher than comfortable final mix level} start position for my monitor volume. I’m constantly ending up in the final mix bus at a good level. Thanks
My mix bus levels are typically fairly low and one of my concerns is how much I’m forcing my mix bus processing to bring the level back up. Anyone have any comments or rules of thumb on how much make-up gain “work” a single stage can do? For example, is 10 db too much for a single fader or plug-in to make up on the mix bus? Do you use plug-ins in series with the fader to minimize how much gain each stage is making up? I’ve been trying to keep individual stage boosts no greater than +6db but I have no real reason for this. Thanks. I always appreciate these videos and the comments. Great community.
Hey Vadim,
Personally, I set the make up gain on all of my plugins so that the output is the same volume as the input. I don’t want any of my plugins to change the volume of my track, or my master fader.
The reason is so that I can A/B by bypassing the plugin. If it changes the volume, then I can’t make a fair comparison.
So personally, I try to get the output level right *before* I even start using any plugins. Then I make sure the plugins aren’t changing that level.
There are no rules, of course. If you want to use plugins to boost you overall gain, there’s no reason why you can’t. But I think that using make-up gain to ensure that plugins don’t change the overall gain is a great way to do it. Might be worth trying out sometime.
Hey Graham,
I am 64 and the doctors say my hearing is great. I have enjoyed your extremely clear and helpful videos on mixing. It has been just the right info at just the right time for me. I have other publications on mixing but your teaching style is excellent. Thank you kind sir for your interest in others of us who love music and composition.
Graham, at what point do you start losing resolution on the faders?
If I may; it’s not really resolution, but there’s more space for fader movement around the ‘0’ position.
The space between the ‘0’ and the ‘-10’ position is much bigger than between any of the 10dB intervals below the -10 position.
So, you’ll be able to set the fader more acurately above the -10 position.
Thank you for that. I’ve always track at -18,..then what I do is group all my faders together before I start to mix and bring them down to -15 so I have plenty of room on the faders to adjust them for a good mix. This seems to be working for me, but I’m not sure if this is the correct way of setting my DAW up before mixing or adding any plugs. Like I said, it seems to be working for me and my tracks sound good.
I will try the same thing, but I won’t go down past -10 if that’s what you’re saying not to do.
Oh no John,
You may go below the -10 position.
Though, if you indeed gainstage correctly (also when mixing, making sure that your PRE-FADER channel levels are at -18dBFS), you’ll find that the fader positions of your main instruments will automatically stay between -10 and 0, where you will also be able to make very subtle adjustments because of the room to manoeuvre.
Of course some subtle tracks, like a shaker here, a triangle there, that are not supposed to be very loud in the mix may end up way below the -10 position.
Here is a proven strategy for setting up your mix:
1. Always start by first setting the PRE-FADER levels of all your channels to -18dBFS.
2. Put your master fader(s) on the 0 position (normally all the way up).
3. Start raising your channel faders (centre panning) and try to get a nice balance while watching your MASTER meter, aiming for a -18dBFS level on the master meters
You’ll soon discover that when your mix is hitting about -18dBFS, the channel faders for your driving instruments will also be around the 0 position.
4. Now you’re free to group, EQ, pan like you please.
5** Important is to maintain the gain structure (PFL) during the whole mixing process: after each EQ move, compression, plugin, etc., you check wether your PRE-FADER level is still at -18dBFS and adjust if necessary.
That’s why it’s called gainstaging; you maintain the proper gain at every stage of processing of the sound.
I hope that there may be something usefull for you above.
Regards,
Thank you Norman..! I thought I was doing something wrong. The way you explained it,.is the way I do it. Thank you again for your time.
So true Graham – Might even be good to mention that some mastering engineers will reject a mix that has digital clipping…
Great stuff as usual!
Hey Norman,
Would the same gain staging work for Software Instrument?
Thanks
Hi Adrian,
If you mean the mixing and processing of virtual/software instruments, the answer is; certainly.
Most waveforms, also software generated ones, will have transient peaks that are (up to and even over) 18dB above the main body (RMS value) of the waveform.
The headroom is to ensure that the whole waveform can pass through our system without being ‘shaved off’ and thus maintaining sonic integrity.
Of course, some sounds can withstand some lack of proper gainstructure/headroom better than others and the sonic deterioration can be very subtle and gradual.
But, just like the good stuff add up to a good end result, with inproper gaistruscture you most likely will notice your total mix sounding a little harsh, less open.
Gainstaging is like making sure your garage door is sufficiently open so you don’t scratch or dent your car trying to cram it into the garage.
Long answer to your question, but hopefully clear.
Regards,
Hey Norman,
Thanks for your answer to my question. You said in your comment to set pfl at -18 and shoot for the same at the master buss…..do u mean -18rms?
Thanks
Adrian
Hi Adrian,
Yes, -18dB RMS indeed.
Regards,
OMG!!! This is one of the best videos you’ve ever done!! Going into it, as the video was starting, I KNEW you were going to say something about gain staging… LOL… However what I didn’t anticipate is you saying you did the same thing that I’ve been doing for the past at least 10 years, and that’s just pulling the faders down to create headroom Instead of adjusting trim and/or clip gains on individual shells. I NOW see exactly why that’s so important because of this video… You changed my ITB life forever dude!!! Thank you! God bless!!
Glad to help!
Great Informative video that makes clear perfect sense, many thanks!!
I have a question for you Graham, about your sends/returns plug ins…
Your instruments and music sends (i.e. drums, bass, instr., vox, and FX) have the VMR, and of course the mix bus has both VMR and VTM. Have you found plugging in the VMR (saturation) on only the music sends as you have done to be the same as plugging in an instance of VMR & VTM on each and every individual track?
I plug in an instance of VMR, VTM on every channel, but it is taxing on the system before I even start to mix. I love the sound, but I hope to get some feedback on this by someone who may have already experimented with this config. Many Blessings!
One more question for all readers out there. Is it not a good practice to lower the Mix Bus fader in DAW?
yup, letting me know i’m on the rite ‘track’ once again! thanx to all the great tips on RecRev, we
know that we CAN get a great master on our own. now that’s not to say that i’m not considering
having our album polished up at SoundLab, or at least the singles, but if we can do a good mix and hand it off to a pro for 60$ a song,(sale price rite now) well, do the math-3-5 grand vs under 500$
YOU GUYS HAVE SAVED US ALOTTA $$$ AND HEADACHES!
AND, YES, keep your headroom low then beef it up on the stereo bus, proving to be a wise approach, even tho i DO use make-up gain on weak tracks, MOST of the punch is on the out bus with our more recent stuff.
http://noisetrade.com/citizenstormwatch/revelation-station
check out these 5 jams that i recorded in 2012 with NO compression on my tracks, then just using Graham’s tips to remaster the stereo, it’s not stellar audio, but if you’ve been at this a while, you’ll hear the RecRev influence knowing that all the tricks were applied to my final L/R
thanx again, guys!
Hey guys,
Can anyone tell me why when using a VU meter to set pfl on tracks in Logic 8 at -18db rms it clip the channel (Virtual Instruments) in Logic before reaching -18db on the VU.
Thanks
Adrian
Adrian,
You probably need to (re-)calibrate the VU meter.
Hey Norman,
It’s calibrated to -18db rms when needle is at “0”.
Thanks
Adrian
Norman,
Is there a next number other than -18db that I can use to calibrate the VU meter?
Thanks
Adrian
Hey Adrian,
A VU meter is typically calibrated at ‘0’, so that 0VU would corespond with -18dBFS on a digital meter.
Now I don’t know if you have a real VU meter, VU peak or PPM, but here is the calibration procedure:
1. Play a 1kHz sinus test tone in your DAW/system.
2. measure the voltage at the main out of your soundcard/system with a multimeter.
3. Adjust your faders to get a reading of 1.23V if you have a +4dBu system or 0.775V for a 0dBm system.
4. Now you adjust the VU meter to show a ‘0’ readout. In case of PPM, the meter should now show -6dB.
Man, this stupid tablet is starting to work on my nerves; posted my reply in the middle of my explanation..
5. You’see that (with a test tone) -18dBFS is equal to ‘0’ on a VU and -6dB PPM, while with music you’rd goin to get -18dBFS = 0VU = 0dB PPM.
In case hooking up a multimeter, etc. is too technical for you, you can just do the calibration just trusting the digital meter of your DAW.
1. Play 1kHz test tone.
2. Adjust faders to get a -18dBFS reading on your DIGITAL METERS.
3. Now jou calibrate the VU to read ‘0’ or the VU peak/PPM to read -6dB.
4. With real music you now get -18dBFS = ‘0’VU = 0dB PPM.
I hope my explanation is a little clear.
Regards,
Just now I that I see that you asked another question before this one, I think that I did not really answer the correct question. Now I don’t even know what question I was answering, LOL.
The calibration process is correct, though!
You can indeed use -16dBFS = 0VU to -24dBFS = 0VU instead of -18dBFS.
And it is calibrated when -16/-18/-20/-24dBFS = 0VU.
But to be certain really, you should do the calibration process with the test tone and multmeter.
On using a DMM (digital multimeter): Note that most DMMs have a sample rate so slow that they are not accurate for RMS readings above a few hundred Hz. Yes, even Fluke meters costing a few hundred dollars are inaccurate for anything much over 300Hz (even if they have frequency counters much higher than that – this part is not related to Vrms). I have a high frequency bench meter that can measure Vrms up to 150KHz accurately, but few meters under $500 are good for even 1KHz.
If you have a fully analog meter (like an old Simpson), you might be able to get a good Vrms for 1KHz, though…
If you have a low-cost DMM, you might be better off using a test tone around 100Hz to 200Hz at the highest. Most audio interfaces should still be pretty accurate down there and an average DMM will be far more accurate down there.
Oscilloscopes can be good if you know what you’re doing with them and you verify it against a good DMM, but if you only have a low-cost DMM, keep the frequency low.
Hi Walt,
The DMM is just used to measure the voltage (+4dBu = 1.23V) at the output pins, not for actual frequency counting.
The 1kHz tone is just what manufacturers use to test/calibrate their equipment and rate certain aspects of their specifications.
In this case:
A 1kHz tone measuring 1.23Vac at the output, should mark ‘0’ on a VU meter, -6dB on a peak reading meter and -18dBFS (adjustable by choice -16 to -24) on a digital meter.
Multimeters can measure these voltages acurately and in my experience not even necessary to be a true RMS meter.
But if I’m missing something, please do share.
Regards,
Hi Norman,
Maybe I muddied up the subject by mentioning the freq counter.
I am referring to the Vrms (the “AC” setting on a DMM) measurement.
Here’s an exercise we performed in our electrical engineering labs in college that might help explain:
We would use a signal generator and create a sine wave at 50Hz – let’s say at 1.0V for an example. The signal would be viewed on an oscilloscope and also measured on a DMM. (The AC measurement on a DMM is Vrms.)
We would then raise the frequency to 1Khz and note that the amplitude had not changed on the oscilloscope, but now the DMM was reading a lower voltage – maybe 0.8V for example. Then we would move up to 10KHz and the DMM would read 0.6V, but the oscilloscope would show that the amplitude was still actually the same – thus the actual Vrms was still 1.0V, but the DMM would read 0.6V. Then we would measure it with a more serious bench meter (DMM designed for lab use) and it would measure 1.0V.
This may be due to the sample rate of the DMM or the general method used to measure it (some high frequency “true” rms meters use a significantly different measurement method than sampling with an ADC). I use bench meters that, in the example above, would still measure 1.0V at 150KHz, but the hand-held Fluke meters we use the most at work would probably only measure less than 0.9V at 1KHz.
Analog meters generally don’t have this problem – at least in the audio frequency spectrum, but DMMs, which are the vast majority of meters sold today, have this problem unless they were specifically designed to work with higher frequencies like some lab (bench) meters.
Does that explain it better – or did I muddy it up more with details? 🙂
Hey there Walt,
Your explanation was very clear.
I’m going to give using a lower frequency a try next time, or at least check the acuracy of my multimeters.
These are the little details that give depth to ones knowledge; thank you for that!
Regards,
Glad I could help.
I know it would be frustrating if you had trusted a meter that is reporting 1.23V when the signal might actually be higher – and then clipping when you end up running the mix hotter due to the low voltage reported by the DMM. Maybe an exaggeration, but it sure would be frustrating if that happened.
I wish all DMMs would at least be accurate on Vac through the audio spectrum, but I think the manufacturers assume you’ll only be measuring 60Hz.
Hey Graham,
After using the trim or clip gain to adjust the tracks level with the faders at unity, at what point do u start to use the track faders?
Adrian
I’m sure there are many acceptable answers here, but for what it’s worth, here’s what I do:
I usually start with he fader until I have the track volume roughly where I want it. Then I’ll apply a gain plugin to get my fader close to zero. For example, if I have to turn it down by 8dB, maybe I’ll use a gain plugin to turn it down by 10dB so I can turn the fader back up to 2dB. I’m not so concerned with leaving the faders at unity. Just keeping them close so I have decent resolution.
Hope that’s helpful!
Thanks Graham.
Haha. Not Graham, but hopefully still helpful!
Thanks Alex.
Jees, this is such an important video, thank you so much Graham.
I am shocked at a) how long its taken for me to figure this out, and b) what a difference it makes to the musicality of the mix. I seriously could not believe my ears between the difference in tonality and musicality when leaving average levels around -15 db compared to how I always tried to push the mix to sit between -6 and 0 db’s. Its almost night and day in terms of quality of sound. For the first time I am not upset at how my mixes sound flattened out, but am instead enjoying the multi-dimensionality of my mixes.
Only problem now is. I need to master my stuff. And I am afraid to do so cos I am worried to lose the 3-d musicality. Can anyone offer any suggestion about how to take a decent sounding mix and master whilst keeping the musicality (and please, I am not talking about the effect of compression, limiting etc, I am specifically referring to the negative effect of playing around with the audio in the -6 to 0db range). I have standard plugins by Waves etc. Thanks
Gain-staging is not only needed in the box you also need it OTB.
Even if you do proper gain staging. ITB still sounds not that three dimensional as mixing on the console. Some people just don’t want to hear it. Its a different sound. Are there great ITB mixes. Sure. Is it easy to do ITB.
No, its way more difficult ITB to mimic that 3D feeling of mixing on a console.
BTW hitting your Converter Outs at minus 6 dbfs could make some prosumer converters sound stressed. Watch out the specs of your manual. RME Fire Faces for example is Input/Output level for 0 dBFS @ +4 dBu: +13 dBu. That means if you set your zero VU at – 20 dbfs you just have 9 db headroom over this, Peaking around – 11 to 12 dbfs now is your new zero. Read the manual, read the specs of the converter factory. You don’t want to stress your ears with stressed output stages at you converter.