Around -10 to -7 seems to work Ok for drums and perc usually on a VU.

heard it best on the second hit of the Hihat. I'll have to think on this - or rather try it out. Now the kind of music I work with has me shaving off (with a clipper) transients of snare and Kick anyway - and a Hihat is never loud enough that this stuff would matter for it. So In order to get my "heavy"-transient material in shape to achieve an overall loud mix in the end I am already making sacrifices. Setting the Soft-Clipper right is important to me though. I do not gainstage is what I said.. now I went into my last project and looked and I never come out of a track (and thus in of a bus) at more than -10 dB (I'm willing to bet that I did on other sessions). BUT I was shaving off about 3 dB off my Kick beforehand anyway... now mind you, that is a sample kick.. so it's already "loud". So let's be realistic: Getting a signal significantly louder (without increasing the peaks) in combination with the right compression and EQ will make that sound "come out at you" in a way that is far more relevant than whatever gainstaging mistake one could possibly make. Having said that: It will depend on the genre. If you're mixing live-Jazz sessions you probably won't find yourself clipping Kick and Snares as often. What I take away from this is that I will - in the future - leave even more headroom for mastering. I'm not gonna beat around the bush: I always peaked at ~-6 dB for my master - in my last project that meant turning up the output of the mixbus for 3 dB. I will not do that in the future. I'll hit the masterbus at -9/10 dB. So in my session that I just checked I was never over -15 Truepeak .. funnily enough I "made it" peak at -7,6 for the master - which I know now is probably a mistake. And I do not use VU meters.. I do not understand why I should ever use them ITB - but maybe I'm ignorant. PS: On a couple of quieter scources I had to use Input in order to hit the compressor hard enough and compensate for that change in volume after the compression which is suboptimal I guess. Note: I did crank the plugin-input.. I used a "clean" plugin solely for the purpose of boost.

Gain staging is kind of a 'how long is a piece of string' topic to me. Theres so many ways of doing it and there are vast opinions but I'd just say use your ears. I went back to the channel multitrack today and surprisingly enough when I gainstaged the snare to around -15vu I automatically heard top end crack come back out of nowhere. I didn't even know I was shaving it off as much as I was. So for me it worked but as you say, genre specific and there arent any hard rules but having a better understanding of how the transients are effected will help my ear more in the future 🤓

I didn't so much hear the difference as I felt it. There was more physical impact on the dry signal. That said, what I use in the mix varies. Maybe I'll saturate to remove some transient in a nice harmonic way but then compress with slow attack to bring some back.

I think that's what we can take away from it. It's about using our ears more and understanding how we want to affect our transients 🤓

Thanks Warren 🤓

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Thanks Bob.. The road continues to 10k 😜

Always a pleasure Andy 🤓

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It's quite simple. Just load it up and it'll default to VU @ -18dbfs. From there all you've got to do is loop the loudest part of your track, press the trigger tool when your loudest part hits and it'll bring the level down to -18dbfs at your loudest part. That's it really. If you want to gain stage to a different level all you have to do is change the reference level in the top left corner. That essentially changes 0VU to whatever dbfs value you want. If you say set -12bfs as your reference level (0vu) and you want to gain stage one track for more headroom then just type the dbfs you want in the trigger tool and that's you. Essentially its about deciding what you want 0vu to be, setting that as your reference level and then using the plugin as a VU meter

@@PaulThird Hmm Thx for the quick reply first off....But I think I was trying to understand when you mention setting the meter for "Peaking" and set to -6dbfs and then you mentioned trigger mode. So I guess thats why I'm confused...what is trigger mode and how do you use it and set it to -6 dbfs...I see where to set the peaking mode on the meter but that about it. Thx again for your assistance!

Cheers Acustica.. 😂 Suppose it must have a learning curve all of its own

Tbh there's a lot of stuff out there that makes me scratch my head. That's why I do so many tests cause people keep coming out with mental assumptions haha

still ragin hahaha

That's the problem. Because of digital headroom the plugin devs can't do much. Any user can slam the bajeezus out of something just because of digitals headroom BUT I agree there could be more work done in highlighting the gain structure of the Plugin when making Demos. If that was standard by every dev then at least buyers would know that there is a difference in how you go into them. There's just not enough discussion about it 🤔

@@PaulThird yep, making all hardware emulating plugins -18 and mastering/bus stuff 0 would solve a lot of issues. Proper input metering in every plug-in would also be nice too haha. I appreciate the video on this with audio examples. It helps tremendously to know what to listen for

agreed. I feel like there needs to be a serious discussion about wether Plugins should have an option (not default - option) to "auto-gainstage" (and adjust the output make-up-gainwise). It just seems pretty retarded to me to implement characteristics of the analogue world to the digital domain that most producers that come up in the digital-day-and-age can not possibly know anything about without getting proper education - and even then.. come on now.. let's not be that elitist cult that has weird obstacles in order to reach the top. Let's try to make everyones life easy - may the best music win.

@@akagerhard I remember when I first started to record digitally, the reality about gain stages in a DAW took me by surprise. If you pull up a software instrument like Kontakt and record a clip, by default it’s probably *way* over -18 dbfs coming out of the plug-in. And your channel fader is pre insert so it’s not going to help you when you add a plug-in behind it. You have to turn down clip gain or add a gain utility. Whatever you do, you’re going to need metering at every stage to know what’s going on until you get a feel for it. It’s why I wish every plug-in that’s gain sensitive had an input indicator for it at least. Needing to reference a separate VU meter in 2021 is a bit silly IMO

@@PASHKULI with all due respect, if you can’t hear the difference, particularly on the percussive elements there’s something wrong with your playback system, audio quality settings in KZfaq or you don’t know what to listen for. The main thing here is to listen for the transient, like the “pokey” part of the snare for example. It’s just plain and simple more detailed, more “pokey” in the properly gainstaged example. I wouldn’t even call it subtle on those two examples. The transient becomes progressively flattened as the input gain increases. In some cases, a person might like that more clipped sound better, but knowing it’s there is important. I’ve done so many of these blind tests, in some I *cant* tell the differences, but more often than not I can consistently identify things once I know what to listen for. Example: I have a Rupert Neve direct box I love. Took it to a studio with a very successful engineer, he couldn’t hear the difference between it and the Radial. Thought it was placebo. So he blind tested me. I got it right within seconds every time. He was shocked. Then I told him what I was hearing, listening for. After that he could pick it out in the same blind test every time. It wasn’t anything wrong with his ears, he’s worked on grammy nominated stuff his are better trained than mine, he just didn’t know what to listen for at first with something that was admittedly subtle. Would it make a difference to an average listener? Probably not. But he bought one of those DIs after that session ;)

I actually can't believe it comes on the day I say "yeah so I think that's me done with gain staging" 😂

I was like “whaaa? A new Acustica plug-in?” Turns out the basic version is free too. Love the concept. Hopefully it’s lightweight because I don’t use AA stuff while tracking (just mixing) due to latency. Either way it’s a sweet gift and idea

@@mrnelsonius5631 I'm curious too how it behaves and if i will give up my just introduced vu metering on every track when mixing.

I just went on Instagram there and.. Black.. Plugin for gain staging.. FFS! I dunno if it'll tie in well with what I've done but I'll check it out 🤓

@@PaulThird Sure, i'll definitely also try it - and the gain staging part seems to be at no cost.

@@PaulThird I saw it in the comments here and thought it was a funny coincidence. Paul posts a video on gainstaging, then Acustica releases a plug-in about it. The Universe is playing games! 😂

I wouldn't personally no. If its completely clean then you shouldn't really have to worry. Most mastering limiters are as clean as a whistle. You can smash waves L1 at 24db GR and really only all your getting is low level noise. No aliasing as its set up not to generate harmonics. Plus your limiter needs a good level going into it so it's not working as hard. I normally bounce my mixes at around -6dbfs before going into any mastering. In terms of mixbus eq's, compressors etc etc those analog emulations will normally recieve 0vu going in or maybe more headroom as I want to maintain cleanliness on my mixbus. These are all things I've picked up for myself quite recent being honest the last plugin output of the chain will be the one feeding the desired level to the limiter or dictating the final master level. In mixing engineers handbook they advise to leave tons of headroom on mixbus processing. Im learning more and more to use my ear for mixbus gain staging but I'll hardly ever go in the red. I go in way cleaner on mixbus using analog emulations. In terms of clippers your shaving the transients off anyway so gain staging would be pointless to me UNLESS it saturates heavily and it has no oversampling options. May be a way to possibly reduce aliasing but when standard clip is like 20 quid and it has control of saturation and crazy oversampling options I wouldn't worry too much about clippers haha

hello, the Classic Clipper is my clipper of choice ..now the nature of a clipper is, that you always shave off the transients.. that means if you are coming too quiet, you'll have to internally turn it up in order to "clip" it anyway. The magic happens with the "slope". I always take a couple of minutes just playing with the slope and look how much I can shave off without distorting or ruining the sound. The same is certainly true with limiters - but limiters are clean by design, while clippers aren't. Clipping will saturate in ways limiting doesn't. I don't know why, probably gonna need Paul to do a video on in-depth comparison of limiting and clipping in order to find that out - and he'll probably get his book out once more. But this is what it is. I use clippers on single heavy-transient scources like kicks and snares mostly. Sometimes I use it even in mastering-stage on the whole mix.. but you better watch out, because that will ruin your sound before you know it - it certainly changes it. And that's why I like to get my transients in shape before mastering. I strongly recommend you try it out yourself. You'll be surprised how much cleaner you can hit your limiter then. If you clip it in mastering-stage you'll add a lot of saturation while doing so - which CAN be a good thing, but in most cases I don't advise it - but that's just me.

@@PaulThird I never knew that limiters are that clean. Don’t they cause farty distortion if you push them way too hard? I know the Ozone Maximizer, for example, has oversampling when TP mode is enabled (I hate TP limiting so I never do) - why would they bother with this if limiters don’t alias?

I stopped compressing individual kick and snare about a month ago as melda gave me the key to the Palace. I use their drum leveller which means that the kick and snare hits are all at the same level but they haven't been compressed so the transients are completely in tact. I then use a drum bus compressor doing about a db to glue everything together and round off the pokeyness of the snare whilst also adding a little more of a natural sound to the kit overall. When it comes to transients, the better the timing the less the compressor needs to work so everything sounds more even so the snare gets rounded off and the rooms and overheads get brought up a little as well so everything sounds a little more like its being played in the same room

I don't compress kick or snare hard bring honest. 1-2db max. I want a snap out the snare so I normally do a really fast attack and release. Is there any reason why you compress snares so hard? I genuinely stopped over compressing my snares, kick, OH etc years ago after watching a CLA video with Steven Slate where he advised how he doesn't heavily compress the direct mics, if he does its in parallel or rooms or whatever. He maintains the dynamics of his direct mics and his drums sound HUGE!

@@PaulThird thats good advice! i should do the same now. usually i just want to bring up more body of my snare so thats why im compressing it hard.

@@Hirokee_ just out of curiosity - are you working with live-drums or are you working with samples? Live-drums need a lot more work than samples do.

@@akagerhard samples! Unfortunately i use a drum library and i dont know how to play drums.

@@Hirokee_ that's just fine, most hits nowadays work with samples. Samples usually require less compression. To be honest I usually don't compress them - saturation/clipping, distortion, limiting, "transient-shapers", eq - yes, regular compression? Not really. But that's just me

Remember to check out my autism channel if you want to learn more about my life 🤓🤓

For kick I gain stage to 0vu (-18dbfs)

@@PaulThird thanks for the quick reply... Much appreciated!

maybe next week :)

Yup. I went back to the session im working on for the channel and went in at -15vu for the snare in AA brown and the difference was extremely audible.. To me anyway. Felt I had a bit more clarity in the snare, bit more clean punch. I had it going in at -18dbfs originally from MvMETER2

@@PaulThird Ok fine, i will try that with some percussion elements. When measuring with mvMeter the signal is +/- 3dB louder with -15 VU, if i see this correctly.

The solution using the filtering can lead to eq cramping. Like what you see in the noiseAsh need. The phase invert thing.. 🤷‍♂️ I'd probably say if that could be done and it worked they'd do it.. I'd imagine so anyway but might not be easy as it sounds

@@PaulThird Not sure the filtering will work at all without oversampling, as the damage is already done in the clipping stage. If we take a 10k sine as input and distort that, we would get harmonics at 20k 30k etc. At 48khz samplerate that 30k overtone will alias down to 18khz (as I understand it), so a steep filter at 20k will not help anything. With oversampling the 30k overtone would not alias down and can be filtered away before downsampling.

I agree. Oversampling is the most natural way of reducing aliasing but it obviously has the most cpu consumption. Probably the reason why many devs try to use oversampling combined with anti aliasing filters. Shifting niquist higher therotically makes that 'harmonic brick wall' further away so you can make more higher order harmonics without hitting nuquist and coming back as aliasing. To me it's not about removing it all together, it's more about reducing it to a level that's more inaudible. Aliasing is something that is managed more than it is eradicated I've noticed 🤓

If you like the end result then just keep doing what your doing 🤓

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I'm not saying gain staging is an issue with anything thats not a fast transient. So for everything else a vu meter will read correctly. The reason you use a vu is so you can stay in tact with the working of the analog emulation which works in correlation with the hardware it's emulating. It means the metering is correlating to the gain staging behaviour of the hardware. That's why many analog emulations have vu meters as we are dealing with analog headroom, not digital headroom. In a clean digital plugin you could do what you like.. As long as your not clipping in the daw but analog emulations are modelled to act like hardware does with analog headroom This is lifted directly from the mixing engineers handbook when talking about "rules" of gain staging "leave lots of headroom. You can raise the level later" The level is something that can be easily brought up at the end using the last plugin output. Or you can stick a gain plugin or MvMETER 2 and raise the output level to the desired level at the end of the chain. This was all in a discussion with Nathaniel Kunkel who is as skilled as any. This is a quote from him in the book; "I also use tons of headroom. When was the last time you took a console that clipped at +25 and ran it at +24? You just don't do that in analog, so why would you do that in digital? I use at least 10 or 15db headroom in my buss. If I'm going to print a loud version, I'll take it out to tc electronics m6000 or something that does a really outstanding job of handling overlevels, and then bring it back to pro tools and not change it." My video is all about analog emulations and its my belief that you treat the plugin as if you were using the gear to get the best results. Again no hard rules but that's a bit of extra info I could give you 🤓

This is confusing to me. I'm reading the excerpt you pulled from the manual. He is talking about a console that clips at +25. That's WAY off the VU scale. Are you sure he is not referring to dBU? On this console he is leaving 10 to 15 dB of headroom on his buss. I interpret this to mean +10 to +15 dBu which is -10 dBfs to -5 dBfs. That seems to make sense to me. I personally like to keep my levels around -12 dBfs which gives me a similar amount of headroom with peaks considered (ok maybe a bit less but close). I've never mixed on a pro console but I know that signal to noise ratio was a concern so you needed to record hotter. I can't imagine the noise you would have with a level at -15VU...

I read elsewhere that a good level to mix at for digital or analog is -18dBfs. This will provide all the headroom needed. Goes along with what the manual said (10-15dB of headroom).

I'm 32 so was in my early 20s lol

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