I want to limit the audio of my social media posts (youtube, instagram).
As these posts are just sonic sketches, I would like to keep things simple. No mixing, no multiband compression, etc. Just a limiter.
What would be the ideal LUFS and peak settings? From Youlean site I read that, for Youtube, integrated LUFS should be -14 and max true peak -1 dB. Is this correct? And what would be Instagram standards? I couldn’t find them on the net.
(Any additional advice will be welcomed, I’m far from being an expert on this subject.)

channel strips, been thinking of adding one to my process…
anybody use them, swear by them? hate them? do they make an actual significant difference in your mixing or even composition?

I’m starting to use channel strips. I have a few from brainworx - Console N, Console SSL 4000 G and Focusrite SC that I have gradually picked up over time when they’ve been on sale.

I don’t have a background in using these kind of mixing consoles in real life, so I have to say that the skewmorphic interface is a bit of a pain to start with and still probably means I don’t use some features. The Focusrite is a bit easier to navigate than the N or G in that regard.

Part of my goal in picking these up was to get a bit more into doing a mix like one would on an actual console - as in, instead of reaching for lots of different plugins, just getting to really know the “essentials” of compression / gating / eq and gain staging in general, with the same interface on each track.

I have a default template for Ableton which currently has Focusrite SC loaded on a bunch of audio tracks as a starting point.

I think I’m still far from my goal of understanding the essentials, but I am finding that just having a channel strip loaded by default on a bunch of tracks does lead me in the direction of tweaking the gain and compression settings to get things sounding more full and present (as opposed to just pushing the volume fader on the track) and then doing some basic carving of the sonic space to get different parts sitting in their own spot in the mix. In that regard, having an integrated channel strip plug in is kind of doing the job of helping me become more familiar with what to reach for to get a certain sound.

Then if I don’t quite find what I’m looking for, I might add a more characterful plugin on that channel.

TBH there’s probably a lot more I could do with just a channel strip in terms of tweaking the settings that are there before reaching for something else, but I’m not trying to be too much of a purist (as if that makes much sense in the world of digital emulations anyway).

These particular plugins do take up a bunch of screen real estate. I think my workflow might be a bit better with a second monitor but I have a tiny studio space these days, which is another reason to doing everything in the box, at least as far as mixing goes.

I’m sure you could achieve similar things just by assembling other stock or standalone plugins that you might already have and setting them up in a templated way.

I’d love to hear how other people are using channel strips - I’m sure there’s lots more that I can learn.

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I wouldn’t invest too much money into channel strips. I find that the best thing I did was instead to buy a good Vumeter VST (there isn’t one in Ableton, the Klanghelm one is perfect and cheap) and save an effect chain inspired by channel strips using stock effects: Utility, Vumeter, Saturator, HPF & LPF, EQ8, Compressor, Expander/gate.

Easier on the CPU, wallet and brain

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My default Live channel has FirComp 2 into Crave EQ 2 preloaded, as well as whatever gain matching plugin I prefer then (don’t ask why, but I tend to alternate GainMatch, TBProAudio AB_LM and Volume Buddy). Saturation (when needed) depends so much on source I handle it on a track to track basis.
TrackComp 2 > Crave EQ 2 > TrackGate is also an amazing combo.
Channel strips can be absolutely awesome (first one I use was SSL Duende’s a while ago), but I’ve yet to find one that works all the time.

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I would agree with the concept of having a good (and unchanging) bus comp and limiter for the master bus. Keeping these two tools the same, and really getting to know them deeply is probably important to getting consistent mix output over time, regardless of skill level. It probably doesn’t matter which as long as you like them on 80% of your material.

However, I have recently reversed my position on ‘mixing into’ them. I used to mix into both, but I am finding more and more that my mix decisions are blunted by them, and sometimes a mix move gives a ‘nonlinear’ or unexpected result if I leave these in - particularly when it comes to panning, levels, and automation.

So now, I keep them in the template with my starting-point settings, but keep them off - maybe switching them on now and again to see how things are stacking up, and then turning them off again.

One other thing that hasn’t been mentioned much - I’ve also gotten into a better habit of setting track gain consistently using a VU / Peak meter, stopping when each track hits -6dBFS or 0VU (calibrated do -18dBFS). This is because many analogue-emulating VSTs are calibrated to behave a certain way at certain input levels. The -18 figure is not magic, it’s just what I’ve landed on, and it also makes gain staging a very simple (even meditative?) part of my mix setup, and then I’m using the faders as normal for setting balance relationships - and those fader positions reflect something meaningful about the mix as a whole.

Definitely a conversation in progress though…

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:+1: Keeping the track faders in DAW as a means of quick level balancing, and away from any automation or gain staging has also helped me greatly (all fades/automation then done with a simple gain plugin, I use Ableton Utility).

It’s a simple technique… but I wish I had learnt it sooner :smile:

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I agree. The quick workflow that seems to work best for my stuff that I’m not to precious about (i.e. it’s going on youtube or whatever) is:

  • fix issues and build arrangement. I’m usually working with 1-3 stereo stems from the synth or norns or whatever. Usually involves some combo of pro q-3 doing broad stroke stuff and tiny peak notching, plus either automating some of those bands or automating cutoff with auto-filter, as well as adding volume envelopes to the beginning and end. I’ll also mess with m/s stuff in this phase if I have to either with ableton’s utility or in pro-q 3 (usually the issue is too much bass in the side channel).
  • then I add the limiter, make sure monitors are set to my standard monitoring amount, and pull up the volume to be in the right ballpark. Usually some combo of how hard the limiter is working plus my ears lets me know if the glue comp stage is necessary, about how much i’m gonna need to push it (and if I’m going to want to do it parallel, and what the ballpark attack/release I should start at is.
  • then I add the glue comp, adjusting based on what I’m hearing.
  • then I usually listen another few times through, possibly on some different monitoring (headphones, mixcube) and make the little changes to make things be just right.
  • then I export and check lufs by dragging the file back in.

I think this order works for me because it progressively makes things sound better. I can understand mixing with the bus comp on, but I think I like doing it after because it makes me think “this was already sounding alright, now it sounds better!” hah

Depending on how many bands are in that first step, I can crank this out pretty quick now, and I’m usually happy enough with the results.

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I’m curious what the VU meters did for your process that the track meters lacked?

I’ve been using a large pair of real VUs for about a decade now. Takes a while to “learn” them, like any other meter, and is just another way of visualising level, but they do correspond pretty nicely to perceived level, and are great at showing the overall crest factor/PLR (or lack of) of the music. Of course they were designed over 80 years ago for low quality speech, and don’t take frequency response into account, but I wouldn’t want to be without mine now.

Here’s the original spec from 1940:

chinn_a-new-svi.pdf (aes.org)

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It’s mostly a “gain staging for noobs” habit. I read somewhere that despite being digital, most analog modelling plugins still have an “optimal” input level which is around -18 dbVu, so I just took the habit of making sure my tracks are at the correct level before entering an effect chain. It’s also a good way to make sure you have some headroom while still having latitude using the track faders for mixing.

I also like to use it to make sure my levels are consistant before and after a specific effect, it helps to hear the change in sound properly.

But the main answer is that the level displays in Ableton are small and difficult to read green bars, and that the levels measured by Vumeters are closer to “human hearing” than peak and RMS meters due to the time reponse which is a bit slower.

The way I use it the mostly is during mastering though, I just calibrate the meter to a desired level (-10 or -8 dBVU), put it after everything (glue comp with soft clipping and limiter) and make sure my track kisses 0 at the loudest part using the comp make up gain. That’s my “mastering for noobs” process :slight_smile:

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That might have been from Klanghelm themselves. I have their MJUC and SDRR and when I use them I always ensure I’ve used the trim pot for -18dbVu cause the manuals told me to :slight_smile:

I have to admit I don’t know the science to this - I believe you and @gregg - but anyway they certainly seem a lot more enjoyable to use than the channel meters.

To add to the topic at hand: I have on all my tracks a TDR Nova (clean EQ) → Soundtoys SieQ (character EQ) → TDR Koltelnikov all turned off by default. I’ll use almost always both EQs but I don’t love compression so try to avoid it (or super lightly). I’ll mix in mono first, do some adjustments for stereo and then turn a bunch of knobs in TDR Limiter 6 - while comparing them to reference tracks - to wrap things up. I’ll leave them for days and check again later.

I have no idea what I’m doing :slight_smile: So I am enjoying this thread.

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Very possible ! I have their mjuc compressor so i might have read it there.

Me neither, and i definitely don’t have the time, patience, ears or listening gear to dive deep in it, so I mostly try to only do things that I can hear sound good, definitely avoid doing things that have no effect i can hear or don’t sound good, and follow some meters I can trust. And mostly try to keep things simple, only use a few plugins and be systematic.

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All three (VU, RMS and peak) are not very close to perceived loudness, but perhaps VU is the closest. For that you want something with frequency weighting and time windows such LUFS (Momentary, Short Term, and Integrated, all useful in their own ways), which was created to be much closer to perceived loudness.

But there is something about the way a real VU reacts to audio that is very pleasing to the eye and easy to read, that I haven’t found in any digital meter (including emulations of VUs, and I beta test for Klanghelm etc.) It’s nice that a VU displays 23dB of headroom, not too much, not too little, and on my Crookwood VUs I have a series of 3dB attenuators so you easily switch the reference level to accommodate different levels (quiet single tracks, full mixes, mastered material, and heavily slammed music).

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lots of great ideas here, it’s interesting to see people different perspective regarding these types of things, I think I’m going to spring for the new Focusrite CS. I like the idea of it behaving like an actual console, where if you have one on each channel they “speak” to each other and real console things like crosstalk begin to happen, hopefully helping to open up things. also includes compressor on each channel, also something I never use(maybe I should, lol) I suppose this bridges into the project template idea, which is something I never do as well, so we’ll see how this changes anything. vu metering is something new to me and seem with exploring as well, though I feel the metering in reaper is much better than Ableton

thanks all for chipping in

seems to be much more difficult to find info on mixing which just isn’t about rock, pop or hip hop :wink:

I have to admit I don’t know the science to this

As far as I know the -18 dbVu thing is related to analogue gear standard. After that point, circuit and components inside the gear would start to get warm and/or being overwhelmed by the electrical current, producing saturation, clipping and other kinds of distortion.

Thing is a lot of engineers started to perceive the distorted sound sometimes was a better pick for the piece, and then that is where the mojo and legend of analogue consoles started. That is why plugins that properly emulate analogue gear in general have a point where clean operation starts to distort (around -18dbVU, although the plugin’s manufacturer can change this), hopefully in pleasing ways.

That is also why a lot of oldschool engineers are so tight about gain staging (besides the sound to noise ratio): it is different to have all your channel faders up high producing distortion and the master channel fader down, than having your channel faders down and the master fader up. If you screwed the gain, you would have to manually adjust all the faders and probably lose the mix balance.

On the other hand it is the same thing in Ableton, for example, to have all your channels down by 6db and the master up by 6db. Digital gain does not provide any distortion, unless of course you hit the digital clipping point at 0db.

Engineers would also figure that specific channels on the console distorted differently (that is where a technology like Brainworx TMT finds its purpose).


also…sorry for the long long post!

tldr: working on music that is not yours is a great way to learn compression.


About compression, I must say I feel you! In the beginning I couldn’t compress anything “”“properly”“” in my music.

Thing about compression is that for me I could hear it better only on music from other people!

Maybe that is the case for @Gregg too, I don’t know, but please share! To me it was only after I’ve done some mastering for colleagues here in the Brazilian scene, first just for fun and then as someone who could actually improve the overall sound and balance, that I could start to hear and use compression properly in my own stuff.

In my own sound, I could never make my mind about it, and that was because it could never be better: only different. The composition mind set was never normative. If you’re discovering and searching for new sounds all the time, it becomes very hard to judge. It is all you.

When you receive a new piece of music from other people, it is way easier to judge! You bring in your experiences to contrast with the composer experiences. That gives you reference, makes you feel fulfilled when sound matches expectations, and make you feel uneasy where it doesn’t.

But now you can’t change almost any composition choices, and you start to value (a lot) EQ and compression.

You bring the snare fundamental down with EQ, because it was mudding the mix, clashing with the synth bass line. But now the power of the snare vanished. You bring a compressor and make so the it will avoid the kick and bass, but act on the snare. Tweak the attack and release, and now the compressor ducks everything by 2 or 3db after the snare hits. Now the bass and snare live without mud, but the pump after the snare will make it feel powerful again. Since you only ever brought things down, you now have a clearer sound and more headroom to make it louder. Win!

It is easier to use compression with instruments, because in the traditional instrument form, choice and function are more clearly connected, in general. By that I mean, you would almost never want a super resonant kick, for example, you would have gone for a tom, or maybe a tabla, or maybe a surdo. Texture, intensity, time and pitch are already determined by the composer/performer and by the instrument choice. So the engineer picks a alt rock, a punk or a jazz track and is more secure in making decisions.

Things become less normative on “avant garde” music where there is no defined “instrument” sounds. Then compression really must serve, even more, a creative outlet of the engineer. It is also the pursue of a feeling already in the music, as before, but the idea becomes harder to translate and accomplish. For a man that is a genius in that field, please check Rashad Becker.

After you develop your “mixing/mastering/engineer” persona with music from other people, you learn to switch with less problems from composer mode to engineer mode, and that is when you come back to your music and cringe hahaha! But cringe is good, cringe is great, because now you have found contrast and can start to bring in the right tools for the job :+1:

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has anyone experimented with analog summing emulation for digital mixes?
primarily curious about the UAD LUNA Neve system… i’ve also seen the WAVES NLS plugin, but i don’t get how that can truly model summing since it is a per-channel effect and is inherently relying on the DAW’s summing.

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Not exactly what you’re talking about but I’m a big fan of using the Plugin Alliance Brainworx SSL plugins on every single track and mix bus including the master. Hitting the randomize all button emulates an entire SSL board’s subtle differences per channel. Doing this really gave my mixes a more glued and homogeneous sound. I still use other plugins along with this but I try to get away with using these channel strips for most of the compression, filtering, EQ and saturation on each track and mix bus. It really simplified the mixing process for me too.

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i wasn’t aware of that company/plugin. thanks! the randomization of tolerances is very clever.

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I activate HEAT on Pro Tools all the time, it adds warmth to the mix.