Faders at Zero vs. Mixing With Faders Explained | Input Gain Settings for Different Mix Techniques

There are two major approaches to mixing live sound: one is mixing by setting all channel input gains to unity level and adjusting levels with faders, the other is setting faders at zero and adjusting levels with input gains. In the video I mix drums with both approaches, showing you the workflow and discussing pros and cons of each technique.
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[@G_handle]

Great video, Thank You! (Fair warning, this video triggered a stream of consciousness that resulted in this mess of a comment. Hopefully someone will benefit from me wasting my morning free time!) I’ve been mixing over 30 years now, and across a broad swath of the world that is audio. What you do in one scenario, you may do differently in another. While I tend to like Absolutes and Constants, I have yet to find any when it comes to essentially Gain Staging. IMHO, that’s because I’m always trying to achieve multiple goals, and which one takes priority usually determines which Technique or Approach I take. This video seems geared towards Live Sound using a Digital Mixer. In Live, my Master Fader and Main Outs are going to a PA System. My first concern is Feedback, then Gain-before-feedback. Then, how much is happening in this show, and how much can I sub-down and automate. So I’m working backwards from Not-Failing at my master, having the whole show on a few Subs/Groups & VCAs, and only then, worry about my individual channels. At my channels, which of your two approaches is dependent upon the board I’m using. Digital boards process After A/D conversion, and the quality of the DSP processing doesn’t really change based upon the input levels you hit it with, primarily because you can add or attenuate digital gain in multiple places with no loss. It’s just math. So there, the input level is mainly about hitting the converters properly. Healthy signals in, process at will, and sometimes the last processor has an output level, that gets sent to the fader. In the Studio, say Mixing, and say on Analog, everything is reversed. First, there’s no Time Pressure (yeah right) and if you screw up, nobody will ever know. Just fix it. So there I start with each individual Track. (Let’s assume the Production and Editing were well done) Every instrument and musician, vocal performance, microphone choice, recording technique, was chosen for a reason and I try to understand each one, help them achieve their goal, but also remove anything there that conflicts with goal of the overall project. So if those tracks are coming from a 2-inch tape machine, or a DAW timeline through D/A Converters (note the opposite of what we had before), then they hit input section of my Analog Console. Here, unlike the DSP, the board definitely does change its sound and performance relative to how much signal, voltage you send through its circuits. One thing that I don’t think was mentioned in this video, which is remarkable because it can now almost go unmentioned, is Noise and Noise-Floor. Seems like for a new generation of engineers, the noise floor is so low, it’s almost irrelevant. Well one of the main goals of Gain Staging used to be maximizing your signal to noise ratio. You tracked as hot as the Tape could handle, and even pushed the peaks into saturation (not primarily because it sounded good, that’s another reason altogether), then you hit the boards channels as hot as they could handle (again not For color), mixed, then output as hot as possible to the 2-Track. All to stay as far away from the noise floor as possible, at every “Stage” of the process, so that the 48-track Mix that you sent the Mastering engineer didn’t sound like left and right channels of a band playing on a beach with waves crashing and wind howling. The purpose of Compressors, or dynamic range compression, was to squeeze down the peaks, therefore allowing you to raise the overall signal higher above the noise floor than before. (I won’t talk about broadcast, but yeah most of the tools that we now use as creative instruments, were utilitarian broadcast devices first. I think Live world gave us Graphic EQs though.) Now the fun part. All those Tracks are piped into the Channels on your Analog Console. In addition to the two utilitarian techniques rightly discussed in this video, forget about the faders, pull them all down to infinity and use your Solo bus. Now you can think of each Channel-Strip as an extension of the instrument playing through it. You know that if you set the input too low, your in the noise floor of that channel’s circuitry. You know that if you set the input too high, you’ll be clipping the electronics, red lights will be flashing, all hell breaks loose. You should know, especially if you RTFM, that the console has a Nominal Level (which isn’t exactly the same as Unity), that the manufacturer has set as the intended range where you’re well above the noise floor, and have plenty of headroom before any distortion. (There are specs for this stuff) The Nominal level, is the usually Safest Cleanest signal you’re gonna get through the electronics, if used as intended. But who’s intentions really matter? The equipment manufacturers, or the Artists and Musicians passing through that equipment? The Producer or the Label that may be paying you? Or You, the Mix Engineer, entrusted by all these people, and with your hands on the controls? Bruce Swedien, one of the greatest audio engineers of all time (Who just passed away a few months ago. RIP Bruce! Thanks for everything you gave us.), once told me that... Bill Putnam, an even greater audio engineer and inventor of the half of all our tools not invented by Rupert Neve, and also Bruce’s mentor, told him that... “You don’t listen To the equipment, you listen Though the equipment!” That sage wisdom brings us full circle: With your ears locked in on Solo isolating just that One instrument trying to make its way off or that tape, or out of that converter, and through that channel to your ears, the Input knob is the pressure valve in your hand that determines how much PSI to let into your circuit. As the Signal’s electrons interact with the Console’s electronics, the Performance on the other side is telling you it’s intentions, and without any processing engaged, the channel itself starts responding. First the signal emerges from the noise darkness, then into the nominal light in all its glory, but then if you keep going, the signal pressure will start to press against the boundaries of the circuit, and the channel will start to complain a little. At some point, the channel will go from simply passing the signal along like a straight piece of wire, to adding its own contribution to the Signal that makes it out of the other side. In fact, that’s what Distortion means. Objectively, the circuit is ‘distorting’ the signal passing through it. A sine wave running through it into an oscilloscope will look distorted at some level of input gain. Every analog circuit will distort at some point. So electronic engineering 101 is design shit that doesn’t distort, let’s it perform its function, without mangling the original signals in ways not controlled by the Audio engineer. Audio engineering 101 is follow the nominal levels set out by the electronic engineers, stay above the noise, stay below the distortion, missionary sex. Audio engineering 2.0 (catch that) is to disregard the warning labels, and see what’s on the other side of the yellow tape. Audio engineering isn’t just Objective, it’s also Subjective. You’re making aesthetic decisions over not just what “should” something sound like, to an oscilloscope, but what does it sound like to your own ears, and ultimately to an audience. Now, “just because you can, doesn’t mean you should!” Things may sound amazing in solo, but totally not work ‘in the mix’. (I have a lot more to say, but I’m out of time, so to be continued.....possibly....)

[@lukeheald3562]

Thanks much for you’re intelligent and helpful commentary. As someone who has gone from the board, to the stage, and often times doing both during this transition from analog to digital, I appreciate comments like this. Critique given from the lens of experience, while acknowledging that which is subjective, is extremely helpful to those of us trying to figure it all out. Thanks again for your contribution and God bless!

[@uuuh9767]

I am humbled by your focus on the final outcome ..and getting paid by a smiling client! (the last part is ofcourse just me reading into things) instead of arguing theory on the internet (which too needs to be done to a point)

[@575motocross4]

Thanks for the good read!

[@brucelmnop2043]

Always run your console gain at optimum level coming in.. then all of your Comps,Gates, and Aux buses will run at optimum performance. If your faders end up to low, turn down the output bus feeding your PA .. Clean signal is best.

[@IanHarrisP]

Completely agreed.

[@paulmanoogian7646]

So to clarify, you suggest that mixing with the faders with all your gains at about -20bfs or 18dbfs RMS is better than running all your faders at zero and having different gain levels coming into he board.

[@HT-Events]

@@paulmanoogian7646 yes, but also keep in mind the purpose of the channel and the transients of the input. A kick and snare for example have more transients than a guitar, so the peaks of the kick can be hotter than the peaks of a guitar. Also you don’t have to have a hot input for a hihat as overheads also pick that up, so you would not end up having a hihat fader only half up and lose precision control with the fader, like explained in the video. Summup: with good gain, you have good S/N ratio and often good headroom. Also processing is most optimal with good gain input. Hope this helps.

[@paulmanoogian7646]

@@HT-Events Thanks! Thats really helpful. Thanks for replying!

[@garrickhanson]

You do realize that compressors, gates, and outputs to auxes all have thresholds and levels, don't you? I don't think anyone suggests setting the gain so far out of the norm that it doesn't fall within the operating parameters of your downstream processors, but the faders operate on a logarithmic scale, and your best ability to do any fine tuning comes when the fader is at zero.

[@crazygooner3398]

Thanks for sharing your knowledge as well as the PROs and CONs of each method!

[@Nontacticalboy]

Preamps/gains are like water pump, faders are like your tap in your house. Mixing with gains(preamps) with faders at unity is like opening your tap all the way, and everytime you want to wash your hands, you go to the water pump to adjust how much water you need.

[@ralphdeonarine9133]

I love this analogy and wholeheartedly agree. Also, wouldn't mixing with gains be a nightmare when setting gain before feedback and ring out monitors?

[@Nontacticalboy]

@@ralphdeonarine9133 every circuitry have noises correct? Set the preamp as hot before clipping to get the hottest signal in comparison to the circuitry noise (signal to noise ratio). Then open the master fader at 3/4 before unity for headroom, then you mix your band. Then you can mix your monitors with Aux and foh with faders. Stage monitors or IEM setup. This is gain structure, we are setting the levels as 'high fidelity' as we can. And also I don't refer to settings with 'db' as it's irrelevant in discussions, the input levels, preamp settings, consoles all varies, so mentioning setting this at this db is in reference to what?

[@phenylphenol]

Correct. Today, gain is only there to make everything your mixing console "hears" is exactly what it would be like if your ear was where the microphone is -- that is, the RMS is essentially equivalent, and it's not outside the bounds for things to distort or clip. Mixing using gain in a studio context means that the person either has no idea what they're doing, or is doing it deliberately as an artist, at least with modern boards equipped with faders. But there's more to the story than that. The reason lots of people started mixing using gain is because before the late 1970s, most live sound consoles easily available had no faders, and many mass market boards just had a single knob for "volume" or "gain." So some people now mix the same way, with the gain first -- and then use the faders for micromanagement on-the-fly. I see this all the time in live engineering. This essentially robs the musicians of their expressive intent, and I discourage it. IMO, the right way to do it with modern gear and mic setups is to use gain only to adjust each channel's RMS to be equivalent, and keep your signal safe from clipping as it enters your console. Then, set faders for the performer's preferred mix, and let them perform. If you're live, ideally just use VCAs if you have them available; this is also helpful for catastrophe management if the room wasn't EQed well or people start moving around mic positioning. If you're in studio, well, sub-group to your heart's content.

[@DodaGarcia]

Damn that's amazing, now it finally makes sense to me. Thank you!

[@trubador25]

Seriously one of the best videos I’ve watched for my studio education. Surely will be posting questions as I work with this on my studio. But you’ve already helped me create a better mix and a much better workflow.

[@MrJlau88]

Great video, now I know the difference between the two. I've been using gain trim for several years but now I'll test the faders at zero to get more precise understanding of the two. Thanks a lot.

[@trubador25]

Awesome video man. I’ve learned so much from watching this. Thank you so much!

[@66fitton]

Great explanation of the things to be aware of regardless of which method you choose. Subbed👍 How do you not have more subscribers!? Very well done videos!

[@scottsmith4145]

The benefit of being able to use presets on each channel for the eqs, compressors and gate thresholds etc is huge for me and why I always set all channel gain levels to the same level. There is always a little tweaking I have to do anyway,, but much less when Im already in the area. Whichever method you end up using you should stick with it so you become fluent with it so you can move quickly and achieve a good basic mix relatively fast.

[@singtall]

and let's not forget that mixing from the faders doesn't effect the aux volumes. if you are doing live sound and messing with the input gain of the channel it will change the monitors as well.

[@gainmedialab]

Absolutely, The input gain setting should be done during sound check and then adjusted only slightly if during the performance there is a significant change in the input signal.

[@Shawneverette]

Changing the input gain changes everything going through the channel. Compressors, Gates , effects, aux sends will all respond differently at different input levels. Not having proper input gain will starve the channel signal.

[@mitas3484]

Makes sense now why the order is Drums -> Guitar -> Bass -> Keys and VOX in the channel input order, with the loudest first, in the analog days. When you talk about the effect of the faders at 9:00, in analog, you want the cleanest possible signal passing though. Interesting!

[@onzkicg]

Hi! Have you tried using a bluetooth transmitter device plugged in at mixer console HP output, and simple using a good IEM earphones with bluetooth enabled? Thanks!

[@gainmedialab]

Hi. I stay clear of Bluetooth technology for professional work, as it introduces serious sonic degradation, instability and additional latency, which makes it unusable for demanding IEM applications. Haven't come across a BT system that would actually be useful. Have you tried any?

[@AdamFoldes]

Easy lover… 😎

[@hooskerdu1]

I have a question for everyone.. I was taught to maximize gain for each channel by using PFL, then adjust the Faders to the desired mix. I don't even bring the Faders up until I have the gain structure set. I recently purchased an Allen & Heath Qu-SB digital rack board, and while I love the sound quality it produces, it also creates a fair amount of hiss when sound isn't coming through the speakers to cover it up. I just started mixing bands again, so I'm a little rusty. So my question is...doesn't maximizing gain reduce hiss to it's lowest possible level, since the signal doesn't require being amplified as much before going to the mains? Or to lower hiss do I need to be less aggressive with gain structure, and amplify the sound coming from the preamps more? I had always thought Allen & Health was the gold standard for small to medium sized rigs, but I had much less hiss with the Midas MR-18 that I used in a previous band. That thing was as quiet as a mouse. Any help you can provide would be appreciated. Thanks all..

[@IanHarrisP]

Hiss in a signal is due to the amount of electricity flowing through the channel. By maximizing the gains and electricity and then turn up faders you'll likely have more overall hiss. Using less gain, which is to say, less electricity you reduce the amount of hiss. Starting your mix with the faders at 0 will help to eliminate this as much as possible.

[@uuuh9767]

Let's take it to the next step: how can we regain speed of work without *apparently* common threshold levels? Firstly, I study the mixing technique described by Dave Rat in one of his videos. That Bus strategy can be translated to a per channel strategy. Secondly, I notice that we at least form signal groups of similar dynamic properties - these groups are based on crest factor and bandwidth. Thirdly, threshold levels can be a lot closer by using HP/BP in the SC of Gate & Comp, as well as using small ratios 1.2-1.5 (but lower thresholds) What is a universal start setting for a compressor on a board like x32 or sq6? What if it is: T: -34dBFS R: 1.5 A: 0,9ms R: 50ms Knee: Softest Detector: RMS (Peak for drums) HP: 200Hz (lower for BD and Bass) Should a gate cut more than 15dB?

[@gainmedialab]

There is food for thought there, but to me there are no universal settings. They depend on the music material and your sonic vision. Since compression can be used for dynamic control and/or sonic sculpting, it really depends on what it is you want to achieve. And yes, gates can cut waaaay more than 15dB. Again, it depends on how busy your mix is, what the musical content is like and what you want to achieve.

[@canastraroyal]

wha

[@trubador25]

I already have a question. So I have my console set to record into my daw and also process playback from the daw. When I mix the playback It seems like I have to start at the very beginning. Meeting that I have to go into the daw and go through each channel and set the channel levels as they’re playing so that their peaks are basically all within the same range as closest possible. Basically setting it so that each channel in the daw is peeking right around -6 dB or close to it. Once I get that done and then I go to the mixing Consol and set the input gains for each channel that the DAW’s playing back into and get those input gain set the same way so that they’re all peaking right around -6 dB or close to it. And then I use my faders to adjust the mix. From there I’m going into a hard disk recorder to record to two track and I have to do the same thing on that where I am setting the track input trims so that they’re peeking right around -6db or close to it while recording the console mix. So I start at the daw play back levels, then go through the Consol input gain levels, adjust the faders , set the hard disk recorder trim to record the mix. Am I doing this right? At first it seemed like I was working backwards. I wasn’t paying attention to the levels in the daw. I was just letting it play and adjusting the input levels on the con soul to try to get them balanced and then mixing with the feeders. But I was still getting all sorts of clipping on the main output meters and on some of the channel inputs and having to set the gains really low. But now that I’m starting my adjustments with in the DAW it seems like it’s all working out a lot better and tying together more. Does this sound correct to you? Or should I be setting all of the levels in the DAW to basically zero and adjusting the trim on the consul? Or basically setting all of the trim levels on the consul to zero and using just the feeders to control the mix? I feel like the way I first described is right but I’m just not sure. A little confused on the chain and how it’s tying together With the gain structure between all of the different components. And one last thing and I apologize for such a long question/comment. But when I solo one of the channels and I’m looking at the meters is it better to rely on the meters that are above each channel strip or is it better to rely on the bigger Single meter that comes out next to the main output meters when I hit the solo button? When somethings not solo they tend to just look at the meters above each individual channel and see how balanced they are. But then hitting the solo button obviously brings up the much larger meter to gauge the input signal. Is it better to use that solo stores to get things balance?

[@gainmedialab]

How are you connecting from the console into your DAW and back?

[@trubador25]

@@gainmedialab it’s connected via usb to my mac. Using logic for the most part. Sometimes Ableton

[@trubador25]

Also, thank you for the info explanation on trim and gain. Was a huge help. Like I was saying in my last reply my board is going into my mac through USB. So it’s definitely affected by both when recording. Need to learn more about understanding how processing works with it.

[@mgrgin]

Are you keeping everything in mono when you check for polarity?

[@4le5ko]

Matej Grginic No, but in this video you are listening to Monitor Cue Bus, and soloing a channel will make it mono.

[@jimpemberton]

I use a both/and approach. I try to get the faders close to unity, but there are times when the physical amount of electrons being drawn through a microphone matters to the behavior of the microphone or the gain can mitigate a noisy signal and you need to focus your gain on that and adjust the levels in the processing and/or the faders. There are times when the gain doesn't matter so much and if you can just keep it from clipping, you're golden. The key is to understand what you are doing with your gain and your faders regardless of where you are on the spectrum between the two approaches.

[@fernandovenegas5374]

I was mainly confusing on the fader positioning while adjusting th input gain.. so basically Just keep the faders all the way down while adjusting the input gain to around -10 to -20 decibels, and then bring up faders to nominal position 0db? Thanks!

[@gainmedialab]

The faders won’t necessarily go to nominal level, because you have to adjust them to create a balanced mix, when you match the input gain levels.

[@jthunderbass1]

What happens to floor monitors when you change input gain on channels mid show? How does the artist react when you change input gain midshow?

[@gainmedialab]

The techniques in the video explain the initial approach of console setup. I would not recommend drastic gain changes mid show. However, if you see a drastic change in your signal input between the sound check and the show, then adjusting gain can bring the mix back in check. Make sure those changes are as minimal as possible.

[@shangomensy7403]

this is explained, if you have set the gain of the Guitar to -18dB and during the show he sends you a signal with 6dB more for example by improving it on his guitar and this generates at the same time the signal entering into dynamic effects, you probably need to reduce it to keep an optimal level. @@gainmedialab

[@shankarjogdand6239]

Sir teach gain structure on unlock mixer❤❤❤

[@mickymalibu]

I had noticed when I set the input gain on each channel to optimal setting that some instruments are difficult to control and seem louder, more overpowering than others. I assume this is to do with frequencies. I will try faders at zero next as it seems to offer more control over levels once set up.

[@Gauseltown]

Question: What is the difference between the two methods? With the faders down, you pull your gain up to a certain level and with the faders up at 0dB you do the same. And in the first method you pull the faders also to 0dB. What are the advantages?

[@gainmedialab]

With getting optimal input gain, your faders don't line up at 0. With Faders at zero, you don't get optimal input gain on all channels, but get a good visual staring point for your mix. It is a technique left over from the days of analogue, where faders would introduce sonic degradation when not at 0 position (still used in studios with analogue consoles).

[@Gauseltown]

@@gainmedialab Newbie here...sorry. What does it mean that when I have my faders at 0dB I won't get optimal input gain? Do the faders change the gain of the channel? I thought gain and faders would be separate from each other. Thank you.

[@gainmedialab]

@@Gauseltown They are separate. But if you first put the faders at 0 and then adjust gain, then the input gain might not necessarily be at optimal level. Check out the different values for hihat in the video, where I talk about this.

[@jtrjammer]

@@gainmedialab I believe also that the fader throw is not linear, but exponential. So when set at 0dB or Unity Gain the fader is at the point which the least amount of movement has the greatest amount of change.

[@GuyLaurentSuizo]

the drum track sounds like Easy Lover

[@4le5ko]

Good ear!

[@MUTEant44]

I wonder why, while people complain that their faders land to low after setting gain, they always seem to forget about TRIM!. We never had trims in analog desks, but neither we had A/Ds. I tend to set my gains so the signal hits Preamp and A/D at proper level then if I need I can adjust with trim which is also before all processing so you can mix both ways - have similar levels @Preamp and A/D AND faders @zero if you wish. Since digital consoles I have seen literally ZERO! guest engineers ever touching trim pots.

[@victorginervalverde6550]

really interested in your point.... you mean i can set gain to unity, and adjust the trim to place the fader at zero.... but.... how much level is going to receive the compresor?, just unity from gain, or trimmed one (lower)?

[@trubador25]

This is probably a dumb question, but I thought the input gain setting was the trim. What is the difference between setting the input gain and the trim? And where is it generally on the console? I basically just see the preamp gain adjustment for the channel and then the channel faders. What do you mean by trim? Sorry to keep editing. On my board there’s something called HA again split that plots trim and gain. Normally adjusting the trim adjusted the preamp. But with the split the the channel gain control adjusts the trim and the preamp gain has to be changed form the preamp tab on the set up menu. And it says A/D gain. What is the difference between these? What is A/D gain? Thank you for reading this and answering, if you’re able to.

[@gainmedialab]

It truly is a good workflow, but not all consoles have that ability. Plus, the trim still affects your processing levels for EQ and dynamics. Great for running preamps hotter on some high end stage boxes, but not all users are fans.

[@gainmedialab]

@@trubador25 Hey. The Gain is the control for the analogue stage of the preamplifier. It controls the level of analogue signal amplification before it hits the converter from analogue to digital (A/D gain). Trim is the control for the digital signal level AFTER the converter has already done its job, so in the digital domain. For example, if you have 2 digital consoles sharing the same stage rack with the preamps, one of the consoles would control the level of the preamp (the analogue gain). That is why it would be called the master console. The second console would only have control of the digital signal after the converter (using the trim, with no access to the preamp level) - that one would be called a slave console.

[@MUTEant44]

@@trubador25 What's usually called gain or your A/D gain or HA is a gain of an analogue stage preamp. After that signal hits D/A converter and from now becomes digital. That's where your trim sit. It allows to adjust signal level without any sound effects. While usually setting some analogue gain levels affect the sound and hitting an A/D at certain level is often also important trim itself has no effect on sound. It allows you to set how much signal goes to further processing as your dynamics/EQ require certain levels to work best as well.

[@wackerburg]

I see another variant of "Faders at zero" with some electronic music artists. Like Paul Kalkbrenner [ kzbin.info/www/bejne/nnyzf3l4rceLgKc ] or Octave One [ kzbin.info/www/bejne/m2XZp4SDrceMpKM ]. Here we have "Faders at +10", which makes the live performance a tad easier, because you can basically shove the channels in and out dynamically while not having to put too much attention to "hitting the perfect mixing level again". So I guess it's more an ergonomical decision here. They both do use Midas Venice consoles, the first series - made in Germany. Which sound nice when being driven, but you're driving the channel and not the preamp. I think this way of "all faders up"-workflow clearly derives from the world of DJ mixers, where the upmost fader position resembles "0dB". I use the same console and yes, especially with the hi-hats, after setting the preamp gain slightly north of 0dB using PFL, my channel fader ends up pretty low. So when performing on the console - and THIS is probably the huge difference in usage - the "travel-distance" becomes quite short and it's difficult to hit this fader position correctly again after like fading it out for some bars. My guess is that "Faders at +10dB" will even increase the problems of the "Faders at zero"-method. In regards of "driving the board", which of course can be highly desireably with some consoles... we are talking about two very different ways of "drive", because we hit different amplifying sections of the mixing console. My guess is that the preamp section is usually of higher quality compared to the amp used in the fader section.

[@KurtCollier]

she's an easy lover (dun da dun) she'll get a hold on you belive it! (Dun da dun)

[@phenylphenol]

Mixing with faders at zero seems kinda crazy to me -- you're pushing aesthetic things way too far back in the signal chain. For this heavily multi-track engineering, use gain only to match the trim of the inputs so you have as clean and strong a signal as possible (without clipping). You should really never mix using gain, since that was never the idea, but a lot of people do that based on how the tech developed over time. Set the console's gain when faders are all at zero, obviously, but all you're doing is amplifying or attenuating so you have clean, "equivalent" streams. The goal there is to overcome differences is microphones, cables, or direct boxes / inputs. Make everything in the cans for each channel sound like what you think it would sound like if your ear was where the mic physically is. That's what gain is for. After that, you can start mixing using faders -- that's what they're there for.

[@cdubb9781]

is there an analog board that exists that you can do processing straight across it? spoiler alert, it doesn't. Even if it did, why would you want that? this is a terrible video for live mixing

[@gainmedialab]

Hey, thanks for your comment. Could you elaborate on what you mean by “processing straight across it”? Not sure what you mean by that.

[@cdubb9781]

@@gainmedialab just using your words at about the 14:00 minute mark

[@cdubb9781]

not to mention, what if you have monitors? hence, live sound.

[@gainmedialab]

Aah, I see what you mean. If you have all channels coming in with the same input gain setting, regardless of the fader settings, then all auxiliary sends for monitors can be set much more easily, because you can reference levels between them for all channels (which works well on analog consoles and even better on digital consoles with sends on faders). When using the Faders at zero approach, the input levels will vary, so you cannot reference them for a quick setting.

[@cdubb9781]

@@gainmedialab if your post fade. which no one does if doing monitors from FOH