5 Typical Mistakes Made By Behringer X32 Users - And Midas M32 - And How To Avoid Them X32 Tutorial
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- เผยแพร่เมื่อ 19 มิ.ย. 2024
- In-depth discussion on each of these points with visual examples. Live sound reinforcement how to video with tips and tricks. Behringer X32 and Midas M32 digital console tutorial. At the end, an example of a rather serious problem that arises and is often blamed on everything except operator error. A misunderstanding that is actually the root of the problem. That point illustrates and ties into all of the other steps.
X32 and M32 Tutorial
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Script files and other tutorials in PDF format as well as scene and library files are available for Patrons at Patreon Page:
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Affiliate Links-
Behringer X32 for sale at Sweetwater:
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Amazon Affiliate Links-
Behringer XR18 on Amazon:
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Midas MR18 on Amazon:
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Behringer X32 on Amazon:
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Shure SM58 Mic on Amazon:
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Shure Beta 58 on Amazon:
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Identifying common mistakes and the problems and symptoms they cause is what this video is for as well as to help eliminate these lurking issues. Of course to help explain why these might happen in the first place.
0:00 Start
1:01 Improper Gain Settings
3:36 Faders, Mutes, and Send usage at soundcheck
6:32 Not Using Channel High Pass Filters (AKA Low Cut)
9:21 Overuse of EQ/Improper Use of EQ
12:37 Mixing on the Gain Knobs/Constantly Tweaking Gain Knobs
13:54 Example of a Common But Very Problematic Issue That Occurs and Trips Up Many People
The information is also relevant to other consoles, including the very similar XR18 or Midas MR18, but the examples in the video are recorded using a Behringer X32 with FW4.04 installed (and this is functionally the same on the Midas M32). Also, the latest version of X-Edit software (X-Edit PC V4.2) can be seen in use in some places in the video. This is the same as the Midas M-Edit PC software. The Mac software is also very similar.
These general mixing topics are relevant to consoles from other manufacturers too, but obviously the surfaces and GUI will look different on those.
The information is mainly geared to live audio production but also has some recording fundamentals in it as well. It should be pertinent to techs, bands, schools, auditoriums, houses of worship, volunteers and those new to mixing, and anyone new to the X/M32 mixers, as well as those making a transition from analog to digital consoles. Perfect info for any venue with these consoles as part of their installed sound system.
The information about EQ is presented in a general sense. EQ is subjective, at least to a point, so I can only speak in general terms about EQ and EQ concepts. There is no perfect EQ setting, and the microphone, mic placement, instruments, backline (stage amps/drums) and musician/vocalist all play a factor in the final settings, as well as the operator's ears. Still, staying within the norms of these typical concepts presented here should help you build a more solid baseline to find and tweak settings.
Also, it should be noted, if you're coming at this from the POV of a studio engineer, the physics of a live performance are more limiting than what you might be used to in the studio. The concepts are the same, but typically, in a live situation, a more conservative approach will usually pay dividends. A couple of reasons here- One of which is that feedback isn't nearly the problem in the studio as it is on a live stage. Many venues don't have the best of acoustical environments. Right there with that- Volume, headroom, speaker dispersion, and the limits of sound system speaker abilities at show levels aren't really that much of a studio issue either. You can push some things in the studio without consequence that you can't get away with live. Especially, in smaller venues and with smaller sound systems.
Monitor Setup video for the Behringer X32 and Midas M32:
• Behringer X32 & Midas ...
X32/M32 Monitor Mixing/Setup playlist centered around this video:
• X32 / M32 Monitor Mixi...
Behringer X32 / Midas M32 Tutorials:
• Behringer X32 Tutorials
Suggested video:
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Five Tips For Better Live Vocal Mixes:
• 5 Tips For Better Live...
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Behringer Download Link for Firmware updates and PC (and Mac) X-Edit software:
www.behringer.com/downloads.html
Summary:
This is a Behringer X32 Tutorial / Live sound tutorial ( Midas M32 tutorial ) with mixing tips and tricks for bands, clubs, studios, church sound, and other venues and users including live sound reinforcement and recorded audio information (live, stage, or studio / home studio). Behringer X32 setup (and Midas MR32 Setup) information.
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As a Sweetwater Affiliate member I earn from qualifying purchases."
I just did a Monitor Setup video specifically for the Behringer X32 and Midas M32. If anyone is looking for console specific monitor setup info for the X32/M32 then check out this video:
th-cam.com/video/Vz9E6FaCatQ/w-d-xo.html
I also made an X32/M32 Monitor Mixing/Setup playlist centered around this new video:
th-cam.com/play/PLWtgwSNlxTjOcSjRK6m0JpvFQHABktu7O.html
Patreon Page:
www.patreon.com/AlanHamiltonAudio
@@greypoweroz Honestly, most of what you said is wrong. It's hard to parse the grains of truth from just misunderstood processes that you've been doing or advocating for. I can understand how it works for you, but it doesn't 'work' in the ways you think it's working. And it's not an optimal workflow either.
On a properly gain staged system, the faders will end up around unity, give or take anyway. But, when it's done by proper methods of following the input meters, you end up with something much closer to optimal gain into the console in the first place. Your way has no good way to tell if the rest of the system is anywhere close to optimally gained. If the goal was to just have the faders at unity, then why even have faders in the first place?
You said:
" keeps all the mic gains "lower" and reduces spill. It's a good way to SETUP a mix for live sound"
This is basically nonsense. You don't want to run your gains 'lower'. That should never be a goal. And "reduces spill"??? No... The mic's pickup pattern is the mic's pickup pattern. The mic will 'hear' what it hears. You're confusing yourself by using antiquated practices that you don't understand in the first place, and making 'correlation is causation' arguments when one thing is not the cause of the other.
Two identical consoles- One with the gain set by using the input meters to optimize the input level for maximum clean signal PLUS headroom. The other using your "set the faders at unity" and bring up the gains until the level I want is achieved". Everything else the same with a lone mic.
And make sure the systems have a 95dBa system level at FOH.
That mic will have IDENTICAL 'spill' (ambient sound pick up) for either setup.
Bring the gain up on either console by 3dB... yes, you'll hear ambient sounds picked up by the mic a little louder in the system (or headphones).
Turn down the fader 3dB on that same console that has the gain 3dB hotter. And NOW the ambient noise coming out of the system or headphones will be back to exactly the same between systems. The extra gain isn't what made the mic spill 'worse' (or heard better (if said more correctly))... the VOLUME did it. Just like feedback, the mic and system doesn't care where the volume comes from. It can come from the gain control or it can come from the fader. Or even the main fader. Just SOMEWHERE in the signal chain.
The gain is working in concert with other settings downstream. 3dB more gain on the input gain, 3dB less fader = same level in the house. The ability of the mic to pick up ambient sounds does not change. The amount you AMPLIFY those ambient sounds is what changes. But that's a function of volume. It could come from the gain control, the channel fader, the master fader, or even beyond the console and at some point in the signal chain after the mixer and before the raw speakers.
This is why I say that people that use your method don't actually understand why it works and why it doesn't. Nor why it's far from optimal. They're actually, usually, totally misunderstanding it. Thinking it's doing things it's not which they use to justify the method. And then want to die on that hill of misunderstanding.
I agree with the set it and forget it concept of channel gain. I especially love the fact that you emphasize the essential nature of having the channel gain properly set. Of course any adjustments in mid flow will change compression, gating, and monitor sends. I find that my vocalists will often sandbag me during setup, because they are not confident to sing out at full volume when they can't hide in the midst of all the vocalists. I find myself needing to adjust their gain stage, not to increase the gain, but to decrease it. This ensures that the signal hits the compressors at the right point, the gates at the right point, and prevents them from increasing their send to everyone's monitors because they are now singing at a higher level.
VERY comprehensive! Thanks so much for this video. My band Waking Tera is about to tackle IEM's with the X32 for the first time, and this def helps a TON! You sir are a gentleman and scholar.
There are a couple of monitor specific videos on the channel for the X32 so you might want to check out this video (and it links to another one at the end) if you haven't already seen it:
th-cam.com/video/Vz9E6FaCatQ/w-d-xo.html
Nice no nonsense video! I would add to #1 in that typically the offending player is playing louder because that's where he/she wants to be in the mix. Giving their instrument a slight bump in their monitor mix will often cause them to back off and put everything back into focus. (8/10 effective)
Good tip!
Among the several x32 tutorials I've watch on TH-cam, this is the the best and most useful. It helps that I've had good training in signal flow and doing mixes in a recoding studio setting. You are dead on on the subject of gain settings. Thanks for focusing on these common mistakes!
Thank you for watching and commenting!
Thanks for that! Not only helpful for these particular mixers but for mixing in a studio environment, generally speaking, as well.
Really interested in recording bands as well as live performances lately. This is gold!
Thanks for watching and commenting!
There IS a reason that professional FoH engineers use the concept of "set the faders near unity". Anyone that wants it set there to forget is doing it improperly, though. In fact, anyone that wants a channel set any point and leaves it there is probably not paying attention. A good engineer is always adjusting.
But to make that adjustment easier, you want your faders near unity. Just look at the fader dB labeling. The faders are logarithmic. That's why at the bottom of the faders the labeling is much closer than they are towards unity. As you mix closer to unity, you are much more precise in your adjustments, and that is actually how the console was designed to be used. There's a reason they start labeling the faders at that point.
While I can agree that pre-amping (not gaining, there is a difference) for -18 to -12 dBFS is a good target, you should ALSO be shooting to get your channels as close to unity as possible. In most cases, your venue probably has their house amps extremely hot, and you will think you need to bring the faders in the board down to compensate. It is NOT a sin to turn down the house amps. You should be shooting for an output of your board at -18 to -12dBFS as well, for proper gain structure.
Yes. And generally when the gains are adjusted properly, the faders will likely live near unity, give or take, for a natural sounding, modern mix. That is one area that will sort itself out over time as someone learns to use a consistent workflow and begins to understand how everything interacts.
Although, it can also be confusing to many volunteers and people starting out because they get mesmerized and think their faders HAVE to be at unity and make that their first priority. Then they don't develop a consistent work flow that maximizes an understanding of gain staging and signal flow that everything is building upon.
That all said, mixing in a venue where you have nearly complete control over the mix versus a room so small the stage sound can sometimes overwhelm the PA volume, means there are times that the signal might barely be in the PA at all. A fader doesn't have to live at unity or anywhere near it if the mix doesn't need it, or worse, is hurt by it. Putting it where it needs to be for the mix should be the priority.
In those cases, if fader resolution is important, that is where DCA/VCA assignments (or subgroups) can be a nice addition to the toolkit.
But that all also can be a bit overwhelming to add too much too fast, until the most basic concepts are understood.
I kind of look at it as fader resolution is more of a comfort factor thing. And proper gain structure, will automatically get you in that window mostly anyway.
Once someone understands the things that lead up to that, they can better grasp ways to get fader resolution into the picture if needed and get them in the window of unity as the mix calls for it. Things like where to be conservative on the pres, and where to be aggressive, using VCA/DCA's, being comfortable with something like the hat living at -10dB on the fader, etc...
But, for the mix, the fader ultimately needs to be where the fader needs to be. I don't want to teach people to be too freaked out if their fader isn't sitting at 0dB on the console, if the mix doesn't sound good there. I think because it's so easy to see the faders, that people who are inexperienced get too focused on that, forgetting or skipping over steps to accomplish that, and everything will happen with nothing learned about setting pre's, and signal flow in general.
And if the rig is too hot, turning down the amps, or just turning down the mains of the console to compensate is great advice. If everything coming into the console is gained properly, and none of the channel faders are able to get anywhere near unity for a proper mix/level, that's a pretty good sign the system is too hot. If the main needs to ride -10dB, or -20dB to compensate and allow the channel faders to get in their normal operating range, that's fine.
Of course if it's your own system, if you have to do that (and everything at the console level is operating normally in proper parameters) then it pretty much tells you your amps or processing downstream is too hot. If it's not your system, and not your place to be making system level adjustments, the main fader on the console is a perfectly acceptable compromise and quick fix.
EDIT: This conversation is relevant to this video that wasn't created at the time of the conversation. DCAs and Subgroups Explained:
th-cam.com/video/x6TKam2F50Q/w-d-xo.html
Sub groups.
If you're working with professionals on stage I think setting the gains (pre-amps) so faders are at unity is plausible...When mixing for all amateur musicians and vocalists you very rarely get their performance levels in rehearsal and setting gains using the unity approach doesn't leave enough room on the fader. As you stated they are logarithmic and there is not much adjustment left. I have been mixing live for 34 years and have never found any benefit to setting the gains using the unity method and the professional FOH engineers that I know discourage it. The other issues that I have run into with this is when you are sending direct outs for live-streaming you have to turn up your gain in your livestream rig so much that you introduce noise and hiss due to the level coming into the board being so low.
I definitely agree with changing amp settings to accommodate your mix being at the correct levels. It has been my experience that venues set all amps at unity and figure the guy mixing can adjust. That mindset can screw up all of your gain staging (pre-amping) at the board.
Thanks for this. I am new to the x32 compact (coming from a Mackie analog board) and its a little overwhelming but videos like this are super helpful and definitely appreciated. Thank you!
The number one mistake you mentioned here is one of the biggest problems I have with sound guys that don’t know how to use the board and I always get the same response “I didn’t touch your fader” I am so glad I found your video thank you
Those mistakes are general, certainly not specific for X32 users.
Gain setting at -18db is however specific to a digital console.
Yup and some digital consoles have some or all meters adjusted to have 0 around the -18 dB mark, but these Behringers do not. And clipping these sounds pretty bad, so don't let the signal go to 0 dB on these.
It's true I'm an old sound guy but I'm 100% a faders at unity guy. My main argument for it is around unity an inch of fader movement is about +/-10db change. If my fader is set to something like -20, an inch of travel represents about +/-40bd change. If I'm recording, I want to maximize my A/D for maximum dynamic range, thus I would set the gain your way. If I'm mixing for a live show, I care about what I hear more than what the meters look like (obviously not peaking though). So I set the faders to 0db and bring up the gain to I like the level I hear.
If gain structure is correct, then my way will still end up with the faders at or around unity. Plus, my way will instantly show you there's a gain structure problem somewhere in the signal path when the faders can't ride around unity for a proper mix.
Lastly, it makes all of the other settings consistent. If you consistently set your input gains to a known baseline, then everything else will operate at consistent settings or said better: You have a normal operating range and know what to expect when you see/use certain settings.
For example: Bus sends... Compressor and Gate thresholds... FX Returns...
And when any of these things don't end up in your normal operating range, then you know you have a problem somewhere in the signal chain, be it a bad cable or misadjusted system gain somewhere.
Setting the fader at unity and bringing the gain up until you like what you hear hides a lot of system problems that can cascade.
OTOH, if the system gain is correct from input to speakers, then it basically nets you the same thing anyway. But the fly in the oinment is when you work on multiple systems, it's hiding problems from you when you do find yourself on an improperly gained system.
I set the slider @ 0 and adjust the gain and once it’s near -18 db then I use the slider. Always sounds good 👌🏻
Good good! I have my assessment soon and this is good in-depth startup concepts
Looks like pre-gain level is most commented on)
I personally usually set my faders to like -6 dB or somewhere nearby, then set gain to hear the band as I want them to sound, loud and clear. This way I have at least 6 dB headroom to boost later at the show before reaching unity. And during the show I try to have my levels between -6 and +6 dB for the best resolution.
Sometimes I also pull master fader down 3 to 6 dB before check and add couple dbs immediately after the check to compensate for later gaining. But I usually do that at the venues I know well enough to guess the default needed gain.
Good to know others' workflow, this is how we educate. Thanks for the video!
Awesome Alan!! I'm a career bassist made Worship Leader at the church I play. Engineer is volunteer so I'm trying to unravel the mystery of the X32 (Digital) realm - I can't just look at the board and see what they've got going on. This gives me some great questions to ask and suggestions to make from the stage during rehearsals. Thanks for posting!
I hope it helps! Thanks for watching and commenting. LMK if any questions come up in the process.
Alan great video I am going to use this in my setup
I've been mixing for a while and I've always run up against all these when someone looks at my settings. One thing you should add is if you don't get that 18 dB gain you're going to be losing gain on busses. I usually mix corporate AV but occasionally do bands and I play in a band and I've followed these five guidance points pretty closely just intuitively. I can't believe how many people basically dropped their whole gain structure by EQing everything down. It's good to hear it some reinforcement to philosophies I've been using.
Glad to watch this and know I got the basics right at least 😊.
My only difference is on the gain go live where someone sends less signal during sound check, I set gain based on that, then they drive it on live sending more signal than the gain is set for. I then prefer to pull a little gain, it’s usually only a few db and it resets the volume on everything dependant on that channel, front of house, in ears and floor monitors, live stream etc. It rarely happens as I’m on stage with a tablet standing next to the artist and over time they know how to give me what I need.
Thank you for the great info.
Thanks!
Alan, what a fantastic video and so well thought out and put together. I’d love to see your take on X32 rack with p16’s. I have the rack and p16’s for a home rehearsal setup. I watched a lot of David mills stuff several years ago when I was using the X32 compact but for space I went to the rack. It’s funny how I’ve neglected/forgot a lot of that teaching since going to the rack so I’m having to revisit a lot since I’m not so happy with my p16 mix even after following a lot of basic p16 mixing videos but there are really important points here that apply to those initial things that need to be set that WILL affect monitor mixes. Thanks again for the time you put in here.
Thanks. I don't have a P-16 to do a video with. I've set them up for churches on installs and service calls, so I can probably answer specific questions, but don't actually have physical unit to demo and video.
Thanks for commenting! I did put a lot of time into the video. :)
@@AlanHamiltonAudio no problem. I always appreciate the time people put in. I’ve got my own channel as well so yes, lots of effort. Appreciate the info!
Nice explanation of gain structure. Too many people try and apply the "fat" studio sound to the live environment. "Turn everything in the signal path up to 11 and trim the masters!!!". My response is there are no transformers, valves or magnetic tape heads to nicely saturate on a sub $50k console.
Very true!
Got our Midas M32 Live 6 months ago, completely noob in digital boards and even mixing but don't have any of these mistakes. This is because I took David Wills M32 Training Tutorial on Udemy. When our board arrived I knew right away how to setup and start using the board and also basic sound mixing. No more hunting videos in TH-cam how to do this or do that.. ☺️
#1 is a one I usually agree with, except for when I tell the singer to sing something out loud, then they hold back, then get a lot Louder during the performance and it peaks.
The dreaded singer that sandbags you at soundcheck... Yes...
Also have to be careful of drummers that do that too.
Seriously the best X32 video I have ever seen.
Thanks!!!
Thanks for this very informative video. 👍🏻
09:28 something important about EQ ing and processing in general : before doing so (like an EQ boost), be aware of the tap points of your monitors !
If you boost some highs in your vocals, and these vocals are sent post EQ to your monitors, then you will be boosting your highs in your FOH speakers AND in your monitors, which might cause some feedback issues !
That is if you use your console to do FOH and monitors of course. One workaround that i have to try out would be to send vocals to a subgroup that goes through FOH, so that you can EQ this subgroup without affecting your monitors.
Once again... GREAT VIDEO! I especially love "DON'T MIX ON THE GAIN KNOBS!" What performer would ever want things going up and down in their mix?
I've been called in for service calls where I've literally had to referee some serious debates where the musicians have claimed their mons were constantly changing and the audio tech was claiming he/she never touched the monitors. So they each were blaming the other, or the equipment.
And you guessed it... After a bit of post mortem, it was revealed the tech was constantly messing with the gain knobs during the performance.
They were either trying to get the faders to live perfectly at zero or trying to get some 'perfect' input level on the meters. Or just literally, mixing on the gain knobs more than the faders. But they never realized or considered signal flow and actually were changing the mon mixes when they did it. It just never clicked for them, even though the musicians were telling them their monitors were constantly changing, that this one knob (gain) could actually impact other things (on other knobs) like their monitors.
@@AlanHamiltonAudio Aaaaaargh!
Very good tips Alan! Thank you!
Thanks for watching! :)
This was super helpful.
One big mistake that cost me : looking at the mute buttons on and fader positions at -inf and thinking you are safe and can play some sources without checking the tap points before !
I sent some white noise through a wedge with a fader really low just to test it and got blasted, because i didn't check before that the mixbus for this wedge was PREfader !
Now i will always check that what needs to be post fader post mute is that way before i send any sound sources thru the console !
Excellent information.. 100% Appreciated.. Thanks for sharing
Thanks!!!
Good point about using the Low Cut filter, this is also very helpful cleaning up IEM mixes. Likewise cutting using Hi Shelf EQ can be helpful removing content that is above an instrument’s natural range.
I am very cautious about adding positive shelving EQ on the high frequencies.
I’m primarily addressing Live sound, recording can be different.
Good point! Thanks for watching!
+1
Excellent video you explained things well
Thanks!
Alan I have watched every video you have on the xr. I heard you say not to trust the internal router. Unfortunately I had to learn the hard way not to trust its reliability. Could you please do a video on setting up an alternative router in detail. Thanks
Awesome video, thanks Alan.
Thanks! And thanks for watching! :)
Every time I get called to a church with an X32 I see those ridiculous EQ settings. HA! Great video.
Thanks!
I have folder with some photos of EQ settings I have ran across in service calls.
I usually see those ridiculous settings and the EQ is inactive. But they swear it fixed their issue
@@LucasWasson hahaha
Fantastic video !
Alan, why did you come for my head? I feel called out. 😂😂
Okay, fine, I need to stop mixing with the gain knobs.
But I need a strategy. We don't really have time for mixing everyone on time, so I've recorded a number of live services so I can mix myself in the room, but I know it's not good because I get a snapshot of the performance and not having people live.
Great video! My team and I were talking about this video and we came up with this question: when aiming to set the gain to -18 dBFS, why disable the Gate, Compressor, and EQ? Do those really impact what we are seeing on the input meter? Does the input meter only show the input at the gain setting, or does it show the level post gate/compressor/eq. Hopefully this question makes sense. My volunteers asked about this and I wasn't sure how to answer as I reasoned through it (it didn't help that we were meeting without the console right in front of us). Thanks in advance!
You can get away without disabling those things, but on any console it's good practice to disable them at soundcheck/line check on a channel. For one thing, and maybe the only thing really, on many consoles the clip light is typically looking at other parts of the circuit than just the input. But that said, unless you're starting with a baseline preset and you know what's in your preset, you could be setting up a channel with something dialed in that you don't want and might not notice. So it's good practice to make sure you're starting 'fresh'.
In my case I need a lowcut above 100 Hz, because the room has an overload in the low mids, what causes a honk feedback. The voices sounds thinny, but that's better than the unwanted feedbacks, I prefer a stable sound over a excellent sound, that does an excellent job in annoying the audience by feedbacks and also kill the tweeters.
About synthesizer, it depends to the quality. In some cases the sound is annoying and I use the equalizer to make it pleasing, in other cases I reduce about 2 kHz, that this instrument doesn't hide the vocals.
About guitar: Cheaper guitars has an midboost, that sounds cheap, so I turn down the middle frequency range, better guitars has a piezoelectric pickup's resonance about 4 kHz, I cut and boost above it.
Violin tends to sound like a cello, so I cut below 500 Hz and boost high frequencies.
I was expecting over or under compression as the biggest mistake. just came back from a gig tonight and was handed an ipad to control an AH Q16 first set had half the band faders maxed. between sets I told the band to lower their in ears because ill be raising the gain 6db on the vocals guitar and keys, since they all had apps to mix their own monitors it took only a few seconds to get clear vocals, helped that they started with 4 voice harmony intro. "there are stars in the northern sky" that let me do the usual 2.5/1 ratio with threshold set so the gain reduction is on average about 2 db. i never use it on kick rarely on snare and a bit on toms, gates are usually also unless the drummer sucks. The most compression will usually be keys followed by bass then acoustic guitars at maybe 4 db of gain reduction. the magic starts to happen when the comps are dancing along between 0 to -4 db more then that on vocals wil often cause the artists to ask for more vox in the monitors and that leads to feedback if over compressed. because im often in a small room it doesn't take much for the kick and floor tom skins to vibrate if over compressed. The second set got me shout out for good sound once the comps danced with the crowd.
I don't get the luxury of a sound check either because the venue is a busy restaurant. I got used to that working a venue that booked tours where the headliner gets to check with their tech while i basically patch and go.
Under compression is probably less of a problem than over-compression. Under compression could just be a place where the mix could be improved... But over compression can lead to all kinds of bad outcomes. From squashed/bad sound, compression in mons killing the vocalists voices fighting it, to feedback problems when an open mic rises above the threshold of compression and suddenly that gain reduction is gone. If the open mic is not at the GBF threshold... then no problem... but if it is.... Ouch!
Good comment and thoughts! Thanks!
very very smart explanation!
Thanks!
What's with these "your first mistake is buying a Behringer" comments? I've seen at least 5 or 6.... not original. I mean, why even watch this video and leave comments if you don't like them?
Their gear snobs and have normally used desks costing thousands more but people can't tell the difference. Really annoys them.
Actually, the X32 alignement is -19dbfs.
Check manual for specs : it says 0dbfs = 23dbu
Meaning that +4dbu = 0VU = 1.23Volt = -19dbfs
Totally awesome tutorial
Every young kid starting out should have to run an analog board first
Definitely would really help, if nothing else so they can wrap their heads around signal flow. It's so much more obvious, or at least it is easy to follow as someone points things out, when you have an XLR connector at the top of a channel strip, and bus assignments at the bottom of the strip.
Although, they probably won't ever know the 'fun' they missed in patching in a few dozen insert cables, carrying processing racks around, and troubleshooting cables that came unplugged in the rack in transport, broken connections, etc... ;)
I learned on an 80's soundcraft 32x8x8 when I was a younger teenager in the early 90s. That sea of knobs became intuitive and logical for me after a while. And if anything the x32 isn't actually that far off in workflow. At least not how I have mine set up.
Alan great video!!!!
Thanks!!🙂
Hi, can u give us tips for the range frequency for each instrument? Like for bass it s on a range 60-80 maybe, and each part of drum , and other instrument, thanks!!
This video gives some EQ tips:
th-cam.com/video/tEX2Prwfh-g/w-d-xo.html
If you go to the channel homepage www.youtube.com/@AlanHamiltonAudio you can find videos specific to EQing/mixing bass guitar, toms, kick, snare, etc...
Well done for pointing out the importance of the low cut! 😁👍
If I am really under the cosh at a festival (and I mean a two minute line check is all I have time for), I just set the gains and the high-pass filters. Nine times out of ten, when the band starts playing it will sound OK and then you can start on any EQ or dynamics.
(and YES. High pass on Bass drum and bass guitar!)
Thanks! Yes, HPF and gain on a throw and go festival stage gets you a headstart on that first song soundcheck! :)
I'm also always trying to sneak in some quick line checks as soon as I hear a musician start noodling. That "is my amp working" power chord and pedal check lead for a guitarist, and that bang bang bang, slide the snare an inch to the right drummer check, are time savers in and of themselves. If they're going to make noise anyway, might as well not waste it... ;)
So many people do not understand hi pass .. lo cut
Good video, especially for beginners. Gain staging *is* important, however it's not neccessary to set every channel to -18dBFS when you're working fully digital. As long as your signal is not clipping and you're far enough away from the noise floor, it doesn't matter. Turn the gain up and just leave enough headroom so your signal doesn't clip when very high transients come across. The whole -18dBFS thing is pretty much just an arbitrary set number that makes little sense when working digital only. If you're using hybrid systems with analog outboard gear, gain staging is a whole other story but one of the main reasons ppl buy these consoles is to have no outboard gear I guess.
Yes, but for the inexperienced, they need to have some frame of reference beyond "0" that they are accustomed to from the analog world and VU meters...
And some understanding about the level of the signal they are sending out of the console to their DSP, DAW, amps, stream mix, etc...
And there's something to be said for consistency. If you use the same gain reference typically, then you'll quickly recognize when you have a problem bringing a send up when you know it should be screaming loud, and it's not. Especially someone who doesn't have the experience to recognize the differences they should expect if they intentionally don't use a repeatable gain strategy.
Also, in the Behringer, the vintage plugins have 0dBVU metering, and reactions, so they are expecting the operator to be operating at -18dBFS range. Which is another area that when you understand that, you can address it, but if you're just randomly going for 0 on the meters because that''s how that person always set their Mackie VLZ16.4, then it could lead to issues.
And there's a fade resolution discussion in this topic too.
Mostly, it's a case of - you need to know the rules to break the rules.
Thanks for watching and commenting!
@@AlanHamiltonAudio It's definetly good to have a "reference point" when starting out. Agreed. It's a minor detail, but what bugs me a little is that you present -18 as the end all be all number that you have to set your gain to. Often times beginners think way too much about rules someone gave them on the internet instead of trying things out themselves. But yeah, really just a minor detail.
I didn't know the part about the vintage plugins, as I barely use X32/M32 consoles. Thanks for sharing that info!
Midas Pro pre-amps do even sound better when they are driven "hot". Too bad the X32 and M32 do not have a separate trim knob to compensate like on the Midas Pro desks, so you won't overload the gate/comp and eq. Good video, especialy about gain staging and about NOT mixing with gain, but with the faders. I see too many vids on YT from church soundguys telling to set faders to unity and then adding gain to get the right level. Just don't😅
@@HT-Events I was just reading where someone was recommending exactly that! LOL.
About the last mistake, sometimes the gain needs some adjustments, artists can say "that's really the maximum level" and surprise me. I'm aware, that it affects the monitor, so I adjust monitor levels as well.
Yes. The main point is not to never adjust it, post show start, but to do everything you can to avoid that situation. Nor to think you have to chase a few dB here or there if you didn't get it perfect. That's also why I said, if it's clipping, or threatening to, then you need to address that. But to know (like you point out) that changing the gain now will have other consequences to prepare for (like changing the mons).
In some ways, even that might not be too big of an issue, because if a guitarist turns WAY up, and you have to turn down the gain, it's likely everyone will have wanted him turned down in their mix anyway. Of course, many times the person being louder than at soundcheck is perfectly happy that they also got louder in their own mon mix, so will instantly hear that reduction when the gain is reduced. And probably not like it... So a bit of both psychological as well as mix issues to sort thru. They were only louder in their mons because they got louder in the first place. And reducing the gain only reduced their monitor level to what it was at soundcheck (assuming you follow your metering rules and restore things to meter as it soundcheck)... but they just heard their monitor go down.
I see so many people that want to continually monkey with the gain for no reason. Looking for a 'perfect' setting or whatever. Or who got sandbagged at soundcheck, but not enough to really threaten the channel might clip. But yet they still have to change the gain. And then they end up with complaints that the monitors are changing.
If the gain ended up a little hot, or a little cool, it's not the end of the world if it's not threatening to clip and you have plenty of fader to make your mix adjustments. Just get it closer/better next time.
Pretty much your point- Adjust when you absolutely have to and understand how it impacts what follows so you can reset things accordingly to keep things in balance and seamless as possible.
Hey Alan, congratulations on the job! Is it possible to turn on subtitles for Portuguese (BR)? Thanks for the content, hugs from Brazil!
Just made an attempt. Does that work?
Hi Alan,
I sometimes see the clippings on the FX or the Main out. However, I can’t find which channel or frequency is clipped.
Would you please tell me how could I find which channel and or the frequency, or the FX that’s clipping, so I can low it down to remove the clippings.
Thanks in advance!
Solo the feeds and identify which signal sounds the loudest. If it sounds equal across all the possible inputs, simply pull down the 'master input' until the clipping goes away.
EXAMPLE:
You're getting clipping on the input to the vocal reverb. Solo the reverb send bus and listen to which vocal sounds the loudest, then drop the channel send to that bus. If they all sound balanced, then lower the gain on the vocal reverb input a bit and see if that takes care of it.
If not, rinse and repeat..
@@KeyboardsJR
Thanks so much for your excellent information!
Great video, thank you. We are running a two Midas mixers with the head amp split which turns the gain control to a digital trim. How do the preamps affect this? Does the trim affect the sends in the same manner? I assume it does. Looking for information on how to deal with the actual preamp gain or how to set them up in the first place. Thanks again!
Yes, one of the consoles should have the preamp gain and be the 'main' console. Typically this would be the monitor console. Its synchronization should be set to internal (and then the other should be set to AES50)... and you must have that setup correctly or your setup wouldn't be working as it is. ...And then the HA Gain Split should be enabled.
Then, at soundcheck, you tab over to the PREAMP tab in the window of the monitor world X32/M32 and set the gains from the window and the encoders under the window. To get there, press the SETUP button, then tab over to PREAMPS. That's also where the HA Gain Split check box is. Then, once here, set your gains here in this window for that -18dBFS window as explained in the video. Same rules apply.
(Note: That's where they were before V4.0 FW... I'm working without the console in front of me but I don't remember them moving away from this location).
Then, as you noted, the console surface just has digital trim. Ideally, you'd use this the same way as the video describes for gain, since it's still about keeping your gain structure consistent. But it's only going to have an impact on things on that particular console. Nothing done on digital trim of the FOH console impacts MON world, and nothing done on the digital trim of MON world console impacts FOH. But, anything done on the actual gain page of the console handling gains will affect both consoles. So it's still important to do that page first in soundcheck.
OTOH, if someone did get sandbagged at soundcheck, or someone's stage fright kicked in and they aren't singing as loud as at soundcheck, FOH can adjust that digital trim without impacting the mons at all (but would impact anything on FOH like FX sends, comp thresholds, FOH levels, etc.). On the mon console, adjusting digital trim would still impact the mons though, but during soundcheck, the digital trim would still be a way for mon world to make some additional tweaks to channel gain structure without going back to the PREAMP tab and making changes there that would impact FOH.
Usually, FOH and MON engineers work as a team at soundcheck when gains are shared like this. So there still needs to be a level of "set and forget" because otherwise it would be constant 'push pull' and the two engineers fighting each other. Actual gain changes made at the mon world console on the preamp tab do impact FOH. Digital trim changes, OTOH, only impact that particular console.... they aren't really affecting preamp gain so FOH won't notice a MON console digital trim change.
...So, with the mon console setup as being the console where the actual gains are controlled, and FOH starting with the digital trims all at unity (in a perfect world both consoles should start that way with digital trims at unity), then if the FOH engineer notices a channel being really hot (or just the opposite) he can ask the mon engineer to tweak that preamp gain before mon levels and mixing really begins. Ideally, the mon engineer will catch it him or herself, but sometimes they'll be focused on another channel, and you (FOH) notice something when the musicians are noodling around (and annoying everyone while they do it! ;) ). This should be before mon levels are set, or at least be prepared to compensate on mon levels if a change like this is required (maybe a channel gain setting was missed or the engineer grabbed the wrong encoder... or something like that to allow a channel to be missed and it's too far out of the window to be dealt with normally).
But essentially, the digital trims allow both engineers to mix the show at the gain levels they prefer (which in turns allows send levels and such (gate... comp thresholds...etc) to be in easily recognizable 'normal' levels for their preferred mixing styles/workflow. Some people might prefer to allow the peaks to hit above -18dBFS and some might prefer to average -18dBFS. While the actual gain can be set one way at the preamps, according to the A1 engineer's preferences usually, the digital trims allow either engineer to have the normal gain staging preferences. That's one reason I said there's a margin of error with the gains and that -18dBFS. Some people might prefer to be conservative, others might prefer to be a bit more aggressive with the peaks. Neither way is wrong as long as they are consistent, and it's still based on that -18dBFS window.
@@AlanHamiltonAudio Awesome detailed explanation. We are fortunate to have two of these consuls, but, we've never had to run them in tandem like this until recently. The Second mixer is for live streaming only. The FOH handles all of the monitor needs, and since we started with this gain split, things have been mostly ok, but not always. There is surprisingly little information about this out there, you have helped solve the last of our problems. For now at least, thanks again!
@@mhgreen I didn't think about one of the consoles being a broadcast/streaming console, but everything still applies the same whether it's a mon console or recording console that is connected.
Glad I could help. LMK if any other questions come up!
Awesome thanks 😊 Keep it up
Thanks!
Great tips. God bless you
Thank you!! :)
03:01 using solo to use the big meter for gain staging
HA - I feel like such a pro - I don't make any of these mistakes! Excellent tips, and a very well-made video. Keep up the great work.
Thanks for the comment and thanks for watching! I'll keep 'em coming as long as people keep watching.
Yes, I agree with Matt. So many tutorials I've seen tell you what to do or what not to do... but by not properly explaining WHY, it doesn't always compute the way it should. Thank-you.
@@kevnic777 Thanks!!!
Brilliant. Absolute gold. Still clarifying and learning things after years of doing this. The times I’ve come back to the console to find that whoever else I’m working with has taken all the high passes off! Now I know I’ve been doing it right!
I gained a lot by monitoring this video.
Is there a way to listen to monitor mix through the phones?? Is it possible to solo the monitor mix sends and listen through the headphones connected to the board??
If someone hasn't changed the default settings for the headphone/solo section of the console, all you have to do is press the solo button on the main monitor buss output that you want to hear and it'll rout it to the headphones, while not affecting the actual monitor mix going to the stage.
So yes it is easily possible to solo the monitor mixes. If that is not working, then someone has changed the default settings for the headphone/solo section in the console.
Thanks a ton brother ❤️🔥 happy mixing!!
6:30 The lowest tone on a 4 strings bass guitar is above 40 Hz, so you could cut below.
Like he says in the video, the filter doesn't suddenly cut off everything below the corner frequency, it's a gentle roll off. Plus, you actually don't need the lowest frequencies in order to recognize pitch, so it actually isn't a problem to filter out the lowest frequencies an instrument can produce. I regularly filter guitars at well over 100 Hz even though they can go down to 80 Hz and it isn't a problem, or I sometimes even filter bass or kick drum over 50 Hz. Especially if you're mixing live with no subs, you need to get the best bang for your buck from your speakers, and filtering out everything below like 100 to 150 Hz will help your speakers prioritize things.
Great video, thanks for taking the time and sharing, but your video title is wrong ...these are general techniques that apply to any mixers...then why mention those models at all ?
Thanks! I do mention in the video text description these concepts apply to other mixers, but the examples shown are the Behringer X platform GUI and surface. I know there are a lot of these mixers in use, whether installed in clubs, churches, or traveling with bands. A lot of people coming to them from analog. A lot of church volunteers using. Different experience levels.
Many times, these are people's first foray into digital consoles. Heck, at this point, they are many people's first foray into consoles in general...
Beyond that, had I not niched down, I'm afraid a lot of users wouldn't see the videos. On one hand, I could've opened it up to many other consoles, but on the other it would've made it hard for YT to promote the video to any particular audience. Someone searching for X32 tutorials might not ever find a wide-ranging version of the video. And at that, it might've gotten confusing having the examples on different consoles instead of focusing on one demographic and the platform(s) (GUI/surface) they are familiar with.
I even did another version of this video for the XR18/X-air series for that same reason. Even though the information in the X32 easily transfers to the XR, an XR user wouldn't be likely to find the video in the first place. I did make the X version more in depth so they aren't the same video with different visuals.
XR18 Version- th-cam.com/video/EilVDp39A9g/w-d-xo.html
When the channel is more established, I might try and conceptualize a version that would cover multiple platforms... or I might just do another version for a different brand of console.... or consoles.
It just seems like YT has an easier time promoting videos that the algorithm understands exactly who the audience is.
In fact, I think, you should mainly mix on the Gain Knobs. Mixing is mainly adjusting for Changes in the Signal. Say a guitarist is changing from finger picking to hitting entire Chords. At that Point, you want to take down the entire signal path. Including Compressors, effects and Monitors.
If you are running some Cues like a Solo part, where the Soloist is supposed to be louder, that's where you should change the Fader. Faders are for adjusting signals relative to each other, Gain is for compensating Signal changes
I'm going to have to disagree except in outlier situations. First, the gain isn't there to mix with. That is the job of the faders. The gain is there to optimize the input level and to create a consistent baseline input level from channel to channel. You cannot continually chase gain all night long. So typically, you have to find an average, a happy medium, and roll with it. Something that gives you enough headroom for the entirety of the performance, while maximizing the input level to be as strong as it can be, while still leaving acceptable headroom.
If monitors are from FOH, then it's even more important to not monkey with the gains during the performance in order to try and find some continuous 'perfect' gain level. As an engineer, you have to allow the music to breathe and that means dynamics are a part of all of this. You don't want to fight the natural dynamics of the band too much. And you have compression as a tool to limit that dynamic range and/or knock back the peaks when it is a part of the toolkit you want/need to use.
The band's stage levels shouldn't vary so much that their stage mixes are getting thrown off when they play a different song. Once again, there should be a happy medium that can be found. The guitarist's lead patch shouldn't be so loud that he is happy and everyone else in the band, including the FOH mixer, are being overwhelmed. That is a problem that should be fixed at the source: Either the guitarist needs to adjust his patch so the lead volume doesn't jump THAT much, or the guitarist needs to be changed out for more of a team player. Same would be true of a keyboard player and key patches.
The natural dynamics of the band should fit inside the norms of the genre... and the size of the room/stage. And the channel gains at the console should be set to allow for those normal dynamics.
While it's true if a guitarist turns up, then you've lost some headroom in your input gain setting. But whether you need to readjust is a question of whether you were so aggressive in the first place that you have no margin of error and are now threatening to clip, and whether that change has negatively impacted monitor mixes. For one thing, if the musician is a team player, then that turn up might've just been something that all wanted/needed anyway. So it might not be NEGATIVELY impacting everyone's monitor mixes. Nor, does it mean they turned up so much they are now clipping or threatening to clip the mixer's input. So really, at that point, you probably should roll with it. OTOH, if they overdid it, experience should be your ultimate guide here. Before you have that experience, you can learn from the band's cues: Does the singer(s) suddenly want more vocals after that guitar or other instrument turned up? Are bandmembers asking for less guitar? Either scenario is telling you in all likelihood the guitarist turning up has wrecked the monitor mixes and bringing down the gain to you original levels is a potential best answer at that point. If you DO address the gain, does the guitarist turn up MORE again (if so, then that's a clue he/she needed more monitor all along so even if you do turn down channel gain, you should compensate with bringing their monitor level UP)? This last issue could be one that might just be one that needs to happen a bit anyway... if the guitarist is turning up then they probably DID need more monitor, or like it, since that is the effective change they created by turning up.
Of course just because THEY wanted more guitar doesn't mean anyone else in the band wanted it. Nor FOH wanted it. But, then again, it's possible guitar was weak for all, so you have to tread carefully when making pre-emptive changes.
In the case of the guitarist going from finger picking to full chords you mentioned... That COULD be an outlier situation where your best bet is to address the gain. It COULD be a situation where it's simply the natural dynamics of the material and you SHOULDN'T be getting that much in the way of things and you simply need a happy medium for an average gain setting at soundcheck and roll with it.
It could be a place where compression is all that is really needed, especially for the house and monitors are fine as-is. It could be a place where two channels are needed... one for the fingerpicking songs, and one for the full chords. Odds are if the two things are happening within one song, then it IS the natural dynamics of the song and you don't want to be chasing those dynamics and countering them with the gain control.
It's definitely not a scenario where there's a clear cut reason to be monkeying with the gain in all cases. And especially if the changes aren't wanted, nor appreciated in the monitor mixes... because any gain changes made on the console are going to be making monitor changes (and stream and recording changes) and whatever else is fed from that console. So first and foremost, the tech NEEDS to be aware that IS happening when they touch the gain control.
But beyond that, the potential outlier situation of an acoustic guitar being fingerpicked versus full chords in some cases, doesn't really change the overall advice that it is bad practice to be monkeying on the gains versus setting them properly in the first place at sound check and leaving them alone and mixing on the faders. Mixing on the faders doesn't create downstream issues like changing monitor mixes, dynamics thresholds, stream mixes, etc...
A lot of people would like to have the validation that constantly twiddling on the gains during the performance, like they like to do for usually poor reasoning and for some (needless) 'perfect' meter reading at all times is perfectly fine. I'm not going to give them that validation because more times than not, monkeying with the gains during the show is a very bad idea unless you know EXACTLY why you are doing it and EXACTLY what it is going to change if you do touch it. That includes understanding music has dynamics so something will fall and rise over the course of a song, let alone over the course of the night. And if you don't have a good reason to change the gain (and perfect meter readings, 'feedback', etc... are NOT good reasons to monkey with the gains).
Which is not to say "never" change the gain after soundcheck. For one thing, if you got sandbagged, or someone turns up so much they are now clipping the input or threatening to, that IS a validate reason to adjust the gain. And maybe the band does something that is so eclectic, so off the beaten path, that maybe the gain needs changed for that reason (like massive instrument swaps or something)....But the default position, and goal, needs to be to set the gain and forget it. Especially when monitors are from that console. Finding a happy medium and average level that will work. Being aggressive or conservative as you learn what suits you or the band.
There could be times when you break that rule, but to break that rule you have to fully understand the rule and fully grasp what that change means, why you're doing it, and the full consequences of it.
Totally wrong
@@AlanHamiltonAudio YEP, YEP, YEP!!
If the 'outlier' happens (guitar goes from one style to another or one instrument to another), almost ALL the digital desks out there are setup up with an input matrix, so assign 2 channels from the same input source (you can also reproduce this by using an XLR Y cable for analog). Finger picking vs. full chords is going to be a completely different set of dynamics, EQ, and possibly even routing, so use what ya brung.
Yes, allways if i can i am mixing on gain because - i want to see guy who have gain setup in all channels included - overhead mic to -18db and who try mixing it on faders. If i good remember around -30/40db on faders it is a few mm, but around 0db to +/-10db we have something around 20/30mm fader space.
@@Rolnick666 You can certainly set the gain more conservatively on a channel for fader resolution reasons. Especially if you do it consistently and not go overboard. That is not mixing on the gain knobs... Unless you set the faders all to be at exactly unity and continually tweak the gains during the show rather than moving on to mixing the show on the faders. Otherwise, why even have faders?
It's 2023... mixing on the gain knobs, ignoring input gain meters, random gain settings... are all just wrong. It's a bad habit/practice on a modern sound system. It's a bad practice from the 70's that just needs to go away and creates further problems, especially for people doing mons from FOH but other issues as well. And anyone learning now has no reason to learn bad techniques that are holdovers from 40-50 years ago when metering was limited and proper gain structure were so much harder to achieve.
No law requires a fader to be at exactly 0dB. And the faders are there to create a mix... otherwise why do they put faders on the console?
General question. When working in smaller venues/clubs I've constantly met players/singers that want ALL of the monitor. Meaning if I ever gain at -18 or -12 I will really struggle to send enough signal to a pair of JBLs without having to send egregious amounts. Tips for better monitoring with excessively low gain?
-18dBFS.... or -12dBFS is not really low. It's essentially 0dBVu in comparison to an analog console meter.
What is the monitor output meter reading on the mixer?
Is there a compressor engaged on the vocals, or on the monitor bus itself, that is into high levels of gain reduction? That could be sucking your level out of the mons before it leaves the mixer.
That would be my first thought... Something in the signal chain in the console choking back the what is actually leaving the console. A compression is always a suspect.
If you have GEQ's inserted in the FX Rack, those do have gain adjustments on them, so it's possible the gain has been pulled back there.
Or... lots of EQ cuts reducing gain by reducing so many frequency areas. That might not even show up on the meters because if the highs and mids are badly cut, the monitors might sound like a blanket is over them... yet getting a lot of signal, but muddy signal. So it won't 'cut' thru... and won't 'sound' loud, especially once the band is playing, yet the meters could 'look' loud.
Those are all assuming the problem is at the mixer....
Are the JBLs set for line input? Assuming they are powered...
What is their signal light/clip light doing?
So recently some changes have been made to our board that seem to have really messed with our drums. We have 3 of the drum channels that clip even when the gain is set to -18. We ordered a new S16 thinking maybe there might be something wrong with it but after watching this video, I'm back to thinking its single path. Would compression on the drums bus affect the gain levels?
@@shelbyhanneman I've not used the S16 so I might be speculating down the wrong path... But on your X32, on the screen, does it call the gain control "gain" or "trim"?
If you look on the screen and it calls it "trim" that that means you need to go into the SETUP (button) and tab over to PREAMPS and set the gain there.
When the console is in gain sharing mode, TRIM is not really 'gain', per se'... It's not setting the preamp, though it is giving you some ability to use your normal gain structure in the console. BUT... being that it's digital trim, it's not really manipulating the preamps. So IF your actual gain is too hot, and you clip the pre, the console will show you that clip... even if your console trim is nowhere near reaching 0dBFS itself. The 'console' is not... but the preamp IS...
Unless the console can't control the S16 gain and it 'has' to be set on the S16.
I'm sorry I don't have more info about the S16... but I've not used one to know how it works compared to interconnecting to a DL32 or connecting 2 X32's for example.
So... I'm kind of making an educated guess based on how things work when you use a second X32 via AES50 as an external pre.
Long story short... I'm betting the actual gain levels at the S16 are hot, and the surface control on your X32 is in TRIM mode, not GAIN mode, because that is the mode needed for the external pres of the S16.
I'm trying to figure out how to connect this to fm transmitter
I never mix on the gain knobs on my own mixer. but I had to add a mic once to a mixer that had a show going on at the same time, and the guy who mixed it before did a terrible job with the settings.
the only way to get my mic in there in a decent level without messing with the settings in-use was to... mix on the gain knob...
Yeah, sometimes it's "Any port in a storm". In 'combat conditions' sometimes you just have to do what you have to do when you're handed a situation where there's no time to re-do things an set it up properly.
Great point.
When people do things wrong it tends to snowball. Maybe it works for a while, but then the complexity and the mistakes catch up and burn them. Or worse, put someone else behind the 8 ball when it's their turn to take the reins. Especially like on a throw and go multi-band show.
@@AlanHamiltonAudio it was a food truck festival. I had a small booth in the back and suddenly 4 women came to me "hi yeah, we the dj is busy now on the main stage, and we have a concert afterwards but they don't have any microphones. The organisers said we had to ask if you had some spare ones laying around...". Which I had, luckily. So I went there, good! They had an X32 (same as I had) with no one behind it, single mic connected to input 8, which was re-routed to channel 1 and dj connected to aux in 5-6 re-routed to channel 3-4 and I was like "WHY😱😱" Gains were fucked up completely. I had to connect the mics whilst the DJ was busy and pre-mix them using my headphones. Luckily with only 3 mics it's not that difficult to make it sound ok enough but man... For me as an amateur having to fix their "professional" job?! It's just sad...
Even worse, they didn't use the main mix, they used 2 mix-busses so they could "turn one slider down when someone with the mic walked in front of that speaker". But this results in having to create the same mix twice! WTF just use the matrix bus for that!!😱😱
@@NiekNooijens Wow... Some things you see just make your head spin!
Odly I learned to mix with monitors post fader. So mixing a bit on the gains was not a big deal.
I have a questions. Do why there is +18db on scale when its dbfs?
On pod Yamaha 01v scale is on dbfs
I don't see +18dB on the meters. I only see meters that top out at 0.
Analog mixers set their range like that. Consider +18dbvu to be 0dbfs and set your gains between -6 to 0 and try not to go past +10. Digital is less forgiving and sounds terrible when clipped compared to an analog console that distorts slightly before clipping in a less aggressive manner. Some older analog boards will show red while still having a few extra db of headroom.
Mixing Station re-gain feature for the win!
Agreed!
The German translation has some mistakes.
I understand, the musician is requested in soundcheck to play loud as possible and in the performance it's possible much more, so the 18 dB headroom is good.
I usually go over YT's auto-generated English and correct any words/phrases it got wrong, and then let it use that to create the other translations. But this video is so old that I don't recall if I even went over the auto-generated transcript in the first place. If not, that might be where YT is messing up the translation and starting from some mistakes in the first place. Either way though, it's YT doing the translations. And I'm sure YT's German ability is better than mine ;)
.
07:00 engaging the low cut at 20hz by default
Great video!
Thanks for watching and commenting!
As a newbie trying to learn as much as possible about how to get consistently great mixes, this video is simple, straightforward and a wonderful tool. I will watch this one over again a few times and take notes.
Thank you for sharing this with us. You have earned a new subscriber. Cheers 🍻
@@stephentyler4352 Thanks for watching and I appreciate the comments and sub! :)
ANALOG OR DIGITAL DESK? ONLY ONE DESK OR MONITOR DESK TOO? I'M AGREE WITH YOU THESE ARE FOR ME THE BASIC AND MOST IMPORTANT TOOLS YOU CAN ADJUST( AFTER THE SOUND COMING FROM THE STAGE !) I LIKE TO MIX WITH A GOOD GAIN STRUCTURE AND CLEANING THE SOUND A MAXIMUM WITH THE LPF AND THE "Q" OF THE EQ OF CHANNELS, WE HAVE A SMOOTHER SOUND AND LESS GENERAL PHASE CURIOSITIES, HAVE COOL NEXT SOUNDS TO MIX,FRIENDLY FROM SWISS MOUNTAINS,
Changing the monitor levels is a BIG one. Another big mistake often made is using compression wrong and affecting the monitors.
Very true!
+1 same goes with EQ if your channel send to monitors is post EQ as well...
14:18 don't touch the gain during the set because it affects FOH, but also FX sends, and monitors ! (and multitrack recording, separate livestream mixes, etc)
of course you touch the gain during show, for example if the guitarist pulls himself up. then you HAVE to adjust. and you have to realize the effects of adjusting on monitors/fx/etc... that's your job as a soundguy.
@@fabianschafer5947 Well you do if only you have no choice, like now it's clipping like crazy, etc, or if changing gain has not a PITA effect elsewhere (multitrack recording, etc).
- Maybe he pulled himself up but not to a point where it's clipping or anything, and if you just wanna leave the gain untouched (because it affects a multitrackor recording or something else), so he's just louder, and you might just wanna compensate for the extra loudness later in the chain in the bus send levels of his channel and/or compression/gate thresholds
- In this case you can use the "regain" feature on Mixing Station so that any gain adjustment is automatically compensated after (sends, gate and comp thresholds, etc)
"if the guitarist pulls himself up"
This is also your job to let him know during souncheck to not pull himself up later on :) If he wants to hear himself more, he can just let you know later on and you just adjust his monitor mix.
Use Mixing Station’s re-gain feature!
@@airlinepilotonewheelguy2800 If you multitrack record, MS Regain doesn't take care of that though :)
Love that feature btw.
Oh, also, GOOD consoles need to have post fade dynamics. For conferences with a lot of speakers, compression AFTER the fader is helpful so you don't have to constantly play with the gain control to match the threshold. The best consoles allow post fade compression. Too many don't because everyone thinks that all audio gear is used for pop musicians!
I don't disagree with that, but that is where subgroups can come into play for corporate work if the console won't do it directly. And Automixing, like Dugan, fits the bill too. The X32's auto mixing isn't exactly Dugan, but close enough. Plus it's limited to 8 channels of automixing while the smaller XR18 has 16 channels of automixing capable channels.
I'm a noob to the x32.. so last night was our first attempt at using our new x32.. and there was no audio coming from the main L&R.. the mix busses had signal to the in ear monitors. The card out was sending signal to reaper. The master L&R fader had signal but there was no sound coming out the speakers. Any ideas?
There are so many variables it's hard to say. Did you build the scene offline and then load it into the console? Did you start from a clean slate (initialized console) or one that someone else had used, or that you'd been testing with?
I'm wondering if before you connected everything, did you try just connecting one mic and making sure that one mic worked in the console and out to the speakers? That SHOULD work in a fresh, out of the box console... and is pretty easy to troubleshoot. But once routing changes get made, DCA's and/or subgroups assigned, channel assignments changed, mutes, etc... it gets pretty hard to diagnose without just seeing everything.
One mistake that I see people making is some people connect the L-R speakers/amps to the L-R control room outputs on the back of the X32. There are 16 XLR outputs all together PLUS these extra 2 XLR outputs that are in a different area on the back of the console.
You want to connect the PA to a pair of the 16 XLR outs. I think the default is 15 - Left and 16 - Right (although that can be changed). You don't connect the PA to the control room L-R outputs. There is a volume control on the surface of the mixer for the control room outs, so if you connected there, and didn't bring that level control up, nothing would've happened out of those outputs. But had you connected there and turned that control up and gotten sound, you would've gotten control room style audio... For example, when you press SOLO that signal ONLY would've went to your PA.
So make sure you're connecting your PA to a pair of the 16 XLR outputs and not the control room outputs. That's just a guess amongst several possible things the issue could be.
@@AlanHamiltonAudio mine is a 32 rack so the control room are 1/4. So I know that couldn't be it. It's a brand new console but the only routing changes I made were the card routing for recording purposes.. hmm .ok.. I guess I'll have to back up everything and also save them in snippets.. initialize and load each piece one by one until I find where it goes wrong.. thanks Alan.. I did appreciate the feedback.. this x32 is way more complex than my XR18 . I thought since they were both behringer and social mixers that everything would be just as easy.. I'm learning that is clearly not the case.. routing is way easier on the 18.. but thank you. I'll keep at it
@@christopherheadcase6886 I was hoping it was the cue outs thing...
If I was you, I think I'd save your scene to USB, then reinitialize the console and connect one mic to channel 1, and connect an amp/speaker to your main out(s), and make 100% sure signal from channel 1, to the L-R out was actually working in the first place. At least that way, you can rule out a console issue (before you do a bunch of troubleshooting piece by piece) plus, have a really easy signal path to troubleshoot so that you can spot any "aha!" things along the way.
Damned good video. Sound advice (pun intended)
Hello, I never post on tech video like this but this time I will. I have an issue with my gain.
The gain at my church is set really since I join the service. For the mics I know for sure that if I increase them above +15 it will go into the speakers and make a big larsen.
I see on the compressor that a another gain could be set.
My question is, if I turn down the gain on the compressor section can I increase the gain from the gain/preamp section?
Thanks for the vid btw
No. Just think of it as 'loudness'. If you take 8dB away from one point, and add it back at some other point, then the loudness out of the speakers is exactly the same as it was before.
What I would look at if I was you, is your compression threshold and compression ratio, and how much gain reduction is taking place on the meters. One common problem that can cause feedback is high levels of compression taking a signal below the feedback threshold, but when the signal in the mic drops (singer stops or sings quieter), then the compression is lessened. So, for example, if there was 12db of gain reduction happening, when the singer stops singing, or sings quieter, some or all of that gain reduction goes away on the compression. If that gain reduction from the comp was the only thing keeping the mic from feeding back, then now it's going to feedback.
And if that compression gain reduction was causing the operator to push the fader higher because it was pulling the signal too far back in the mix, then that's exactly a perfect storm for feedback to occur.
If this is an issue, try disengaging the compressor and resetting gains and levels and see how everything behaves. Once it's stable, you can re-engage the compression, but do so with lighter settings- lower the ratio, raise the threshold... Watch you gain reduction meter on the compressor and don't let it be so active with so much gain reduction.
I'm not sure if this applies to your issue or not, but I thought it was worth mentioning.
@@AlanHamiltonAudio Thanks a lot for the answer. I try this tomorrow :)
@@theoaglaganian1448 Good luck! Let me know how it turns out.
@Alan Hamilton @Alan Hamilton Hello. I think it turns out "well". The first thing I did was placing the mic few feet away than normal. I think this help me to go higher in gain (the mic is the one we use during the sermon).
I don't really know if I hit the -18 DBfs because I was alone, but tomorrow I asked to start the service early so we can adjust the gain properly.
Aside of this mic we also use some HF headset. I didn't have much time so I'll check that tomorrow. When I hit -18DBfs we can ear every noise on 20 meters. I didn't touch the gain on the headset yet but I will.
I wanted to say thanks you because this video really motivated me to do this gain adjustment. I already know that it will improve the sound for our Zoom. So huge thanks again 😁
@@theoaglaganian1448 Thanks! Glad to help!
Totally agree! 👏👏👏
Thanks!!
The mixing console has more than 16 Bit, so there is no need to set gain very high.
Hi Alan, how to setup for live streaming on the x32.
I haven't done one specific to the X32. I probably should do one. It's essentially the same routine as the XR18, which I have done-
th-cam.com/video/aNlBOOhWbZw/w-d-xo.html
...But the X32 has more options since it has matrixes for one thing. ..And of course different PC drivers are needed.
@@AlanHamiltonAudio
Ok, thank you
thanks
Hi. Do you know if I can use the Midas DL32 stagebox with the Behringer X32 Rack? Are they totally compatible?
Definitely should be compatible.
@@AlanHamiltonAudio "should be" can lead to worry :D
@@Jinaci1732 No reason for the X32 rack to be any different than any of the other units on the platform. As long as your AES50 ports are working good... and you use Cat5E STP cable with their recommended Ethercon shells with tested continuity from shell to shell for your shields, it will be good.
th-cam.com/video/UjJjR2J_aSg/w-d-xo.html
@@AlanHamiltonAudio
My worry was more about the Behringer and Midas compatibility, but also about that. Thanks! Very helpful!
@@Jinaci1732 No problem! Good luck!
Good stuff here.
Thanks!!
There's something that confuses me about the Behringer x32. I would usually use the mix buses for different monitors. Why does the mixer use some of these for effects? Let's say I have 5 musicians playing and 3 singers, and I want them all to have stereo IEMs. There are 16 outputs in the 32-channel stagebox. I'll need to use all 16 outputs. How do I do this if mix buses 13-16 are being used for effects? Can someone please explain?
Good question. I think part of the problem is the way they are just there tempting you... so close, yet so far. 16 buses, 16 outs. Meanwhile, on the XR18, you get 10 buses, but 6 outs, plus 2 stereo. And those 4 FX sends are called FX sends (not just buses) and they cannot be assigned anywhere else. So it doesn't 'seem' to taunt you in the same way.
But really, what the X32 did differently, is they didn't dedicate, or simply label, any outs for L-R, or L-R-S-F (or L-C-R, sub). Nor did they dedicate any 4 buses as 100% FX sends. If they would've dedicated 12 outs as auxes, and 4 outs as mains... yet those 4 main outs had the ability to be assigned anywhere (not have to be dedicated as main outs), and any bus could be an FX send, or not, then it would've seemed to be a bonus. More of a "And if I want to throw two of my FX sends away, I can have 14 outs for mons, and still have 2 outs left for left-right mains... or throw them all 4 away, reassign the main outs to mons, and I can do 8 stereo mixes)".
But by having those 16 XLR outs in the rear, and 16 buses on the console, it makes it seem like there should be a way to use them all as mon mixes, yet still have everything else. But it's really more a flexibility thing. They had to draw a line somewhere on the number of XLR outs, and the number of buses.
And if they would've just dedicated 4 buses as FX sends with no ability to use them as mon sends (like the XR18 for example), and if they would've numbered the outs 1-12, and L-R-spare-spare then it wouldn't seem like you're having something taken away, when really, they're just giving you maximum flexibility to work with the available buses and outputs.
Keep in mind, there are also matrixes to be considered that need outs when they are used.
And there are the aux outs on the board too.
If you're normally using left/right/fill/sub, or left/right/matrix 1/matrix2, then all of a sudden it doesn't seem like those 4 FX buses should be available to you for additional mon sends.
I think they probably always factored in their users would at least be using 2 outputs for L-R...
Anyway, I kind of look at it more as a illusion that there should or could be 16 outs and 16 mon mixes (or 8 stereo), AND 4 FX buses... it just seems that way because they left maximum flexibility in assigning buses and outputs. Make those last 4 buses dedicated FX buses with no assignment flexibility, dedicate 4 outputs as house outputs, and all of a sudden it seems perfectly normal the way it is.
But they just went ahead and gave users maximum flexibility, to even strike FX sends and turn them into mon sends if a user wants. And configure main out, matrix outs, etc, wherever the user wants.
@@AlanHamiltonAudio thanks for the super detailed response
Nice!
Thanks!🙂
Where do you get the concept of -18dB for gain setting on the pre-amp? I heard no reasoning, I'm not saying it's wrong, but the way it was presented, seemed arbitrary at best. Can you give me a reason/s for why THAT specific number?
The quick answer is that 0dBVu essentially is -18dBFS if you were balancing levels.
People are coming to the digital platform and assuming the meters they were accustomed to seeing on analog consoles are exactly the same. But it's a different scale. Which brings us back to -18dBFS (the scale of the Behringer X series platform and most digital mixers) that is essentially the same as 0dBVU that they'd see on their analog console. If you run the channel gains up to 0dBFS on the X32 then you'd be more equal to +18dbVU on your analog console.
That is some screaming hot inputs, and if you mix at that level throughout the mixer, you're going to be wondering why you're hitting your IEM transmitters so hard, your amps, DSP, etc... And if you're recording then you're going to be hitting your DAW really hard too.
If you connect that mixer to a properly gained sound system, the problems will be quickly apparent. I'm sure some users have compensated downstream of the mixer and didn't even realize they were compensating.
Also, the vintage plugins inside the X platform ARE scaled to 0dbVU on their meters. So, if you're gained to -18dBFS on your channel, sending that to a subgroup with a Leisure comp inserted on the bus, then it's seeing what it expects, a signal normally in the 0dBVU range. But if you gained your channels to 0dBFS, then you're hitting that comp at +18dBVU. You then get whatever they programmed the emulation to do when it sees signals that hot... and that's likely whatever the real hardware did with signal that hot. Definitely, you've thrown your headroom away.
This guy has some charts, but there's a lot of information about this on the web. The thing is the scale of the meters and to realize you're not looking at the same scale as you would be on an analog console.
www.tedlandstudio.com/gain-staging-the-x32
@@AlanHamiltonAudio Thanks, so here's a question, do the pres in the X/M32's happen before or after the A/D conversion? And either was, is the level meter POST conversion?
@@gotscottgreen Are you wondering about whether the signal can be clipped if it hits zero on the input?
-behringerwiki.musictribe.com/index.php?title=Clipping_-_What_happens_when_the_signal_goes_into_clipping%3F
@@AlanHamiltonAudio Not at all, I am curious as if the preamp is analog and then it hits the A/D converter, and THEN is monitored, or where the A/D conversion happens in the chain. Is it pre or post preamp
@@gotscottgreen I'm under the impression the metering is in the digital domain. Analog pre... ...A/D... metering.
I've got some bookmarks, but I'm not sure anything actually clears that up beyond telling you enough to make an assumption. I'll see if the bookmarks still work and will post them.
Watching guys n Girls today try and pre-empt a show on the Digital Console, cracks me up.. no end ..
Here's 24yrs to date, half in the old school Analogue world and Half in the new digital work on Live sound touring internationally.
While Live sound is my Passion, and my dirty Mistress and my absolute Drug, I can't help but feel this way....As I'm sure many who are as seasoned as me , fell the same way !
This is Still my process on any console as I lost count of how many times Id have to set up an XL4 , or an Pro9 etc or even MANY an x32 in a shitty club with Analogue Outboard dynamics, FX, blind as a starting point etc, especially at festivals to get it right in the first 10 bars of the band's first song using only Gain to start with VERY QUICKLY so ALL instruments including the voice could actually be heard, within reason, ignoring the audible mess that appeared onstage no matter what setting I had in an Analogue or Digital console as a very basic starting point.
Now time to organize the mess!
By song 2, I had Vocals and Drums dialed in where they kinda need to be as a starting point, Cymbals, Snare n Kick first priority after lead Vocal - keep in mind, all instruments after that point of the show starting just to be heard in a ROUGH mix, right ???
By song 3 at the lastest point, I had, hope to have the Bass Gat dialed in using quickly formed Dynamics, more Dynamics, and more, Serial, Parallel, upsidedown dynamic Processing, whats ever needed inserted to get that person's Bass just heard after removing the audible shit of Bass and Natural Room Acoustics coupled with that person's setup, out front as a starting point!!
+
In the Same song TRY n get the Guitar lead break patch changes under control Beyond the speed typically needed to Swipe Right on Tinda if you saw someone *clearly* outside your typical reach but they said:"I just want anyone to f$%K me" ! , so the Guitar man can do his *Widdly Widdly, Wow wow* , " *Im the lead Guitarist* " Moments without cutting off the heads of the first 7 rows!
By song 2ish to 4 id have hoped I got the BV's and keys happily where I wanted them as a starting point if not too distracted with Vocalist and Guitarist ego's and a (mature) Manager in my Ear telling how big the band is " *Absolutely Going to be* " and If I ... ".... *play ya cards right* ...... " yada yadda yadda yadda ,, blah blah blah .. , and some!
Song 5,, as a Goal, to ultimately ignore *all of the above* for, and just fuckn try in smash it and aim for to now start *POLISHING A TURD* using EQ, creative Dynamic Processing, with expensive outboard Dynamic processors on the Inserts etc ETC ETC ...
By the last Chorus on the last song on their 2nd Encore, I typically had it perfect enough for me to want to listen to it at home...
.....*except* , thanks to being overexposed to MANY MANY bands I'd have to TRY n make sound good, the number of bands playing live, Id heard over 25+ years, telling me that they think they are the next biggest new thing, n..
"...did you want a Job" ?? , Because their friends and Mom n Dad etc out front said it sounded great .. ,
I already despised that unknown band, musically,....by the time they got to the 9th bar of their first song!
I Hate my Job, actually, I fuckn Despise it because it,
Yet, It would so cool if I didn't have to deal with the Artists and the trailing Entourage....
OK, so now I understand why the old practice of starting with the fader at Unity is old news (as am I!). 🙂
But...even when I set my fader at Unity, I still use the meters to set gain and go for the top of green with only an occasional yellow, which it turns out is the -18 you say to shoot for. Then I go back and tweak the fader. I suppose starting with the fader at the bottom isnt much different so I'll start doing that.
Here's my question...My world is H.S. theater and most channels are mics. On singers. I don't have a prayer of getting a signal that's even remotely consistent enough to set it this way, especially considering they're supposed to be singing at many different levels. I've had to resort to tell them to "sing your loudest part" and set the gain based on that. Otherwise, I end up with lots of red during the show. What's a better way to get the right gain level that I can set-and-forget?
Nothing really wrong with having them sing their loudest part for you and basing the gain off of that. After all, the idea is to optimize the gain and keep it out of the red and away from clip. Of course, making sure the kids understand and use proper mic technique and stay on mic is also part of the journey. But with kids that can be like herding cats sometimes too.
If you know the kids are mostly going to be singing quiet, you can certainly gain things a little more aggressively... BUT only to the point that you also know their loudest part won't be clipping.
Also, keep in mind the goal is to AVERAGE -18dBFS. You can and should PEAK higher. An extremely rare and random -3dBFS being the absolute peak can still be OK as long as that is really the absolute peak and it stays out of clip. As long as the 'normal' peaks are a little more in line with common mixing practices, and your regular signal can average 'around' -18dBFS. And a -6dBFS absolute peak for a shout/scream/loudest part of an extremely dynamic vocal wouldn't be unheard of.
But your first goal is to keep the signal out of clip AND maintain headroom. Whatever that takes.
Compression is great for keeping weak singer louder, while clamping down on their more 'confident' parts where they suddenly get louder. BUT remember, the built in compression is AFTER the preamp, so compression won't help you keep from clipping the input. But as long as your input stays clean, compression CAN help to raise the lower parts of the vocal... if needed... and can help keep louder parts tamed... if needed too...
It never hurts to mention some psychology here either... Especially when working with inexperienced performers and talent. Or talking heads. Be careful giving them monitor levels that they don't ask for. It can make them sing or talk quieter which in turn creates a vicious cycle of the sound person turning them up (in the house) and that just adds to what they are already hearing from the monitors, so they speak or sing even quieter.
Of course for singers you don't want to take monitors away from them... they need what they need... But they might not need as much as they think if they don't test the monitors with them singing at show levels. A whispered "1 2 3... more monitor please" might lead to them getting way more monitor than they expect when they start belting. And that can lead to quieter singing, singer further away from the mic, or both.
Back in the 80's I trained under John McBride. His reasoning for unity on the fader was that the fader should never be able to be pushed into clipping. I have always set it up that way. The faders may move up and down but won't clip no matter what.
@@greypoweroz "Also .... Feedback on dynamic mics is related to the "gain on the mic". The more gain you have, the more "live" the mic is. Typically in live sound you want the LEAST gain you can get away with at the input stage, in my humble opinion....."
This couldn't be any more wrong. Gain should be optimized and optimal gain is not running it as low as possible. Input gain, by itself, doesn't ever cause feedback. Set it to whatever you want... it'll never feedback until you actually pot up the channel or monitor send.
There is no "perfect" gain setting... just an acceptable gain setting. A 'window of acceptability'. A gain setting that gives the console as much gain as possible to work with, while still allowing for headroom and never clipping should always be the goal. Since modern consoles have input and output metering, it's simple to do this with repeatable and consistent methods these days. We don't have to rely on guess work and workaround from the 60's and 70's in the 21st century. And we can have the entire system properly gain staged. And work with different amps/speakers, even different musicians, each night and still have consistent and predictable baseline settings on the console. At least as far as they relate to gain settings and optimizing gain.
Old myths die hard. Especially ones that are rooted in some semi-truths and misunderstandings.
Correlation is not causation: I'm sure plenty of people have been mixing a show, and for whatever reason, reached for the gain knob. And turned it up. And got feedback. And immediately in their mind a lightbulb went off: "The gain control causes feedback!!"
Well, volume causes feedback... and turning the gain up certainly brought the volume up on that channel. And to every send on that channel. So whatever points were close to feedback, would now feedback. But the would've had the same feedback if you never touched the gain and just turned that channel up at wherever it was feeding back. Be it monitor send or the house feed.
I'm also sure someone has had feedback problems, maybe in mons from FOH, and cluelessly grabbed the gain control, turned it down, and it stopped the feedback. But again, correlation is not causation. Turning the gain down turned EVERYTHING in that channel strip down. Of course the feedback stopped. But it wasn't any 'special' gain setting that did it. It was using a hammer on a problem with you needed a scalpel.
The gain control needs to be set where the input meter says it needs to be set. Whatever consistent window of acceptability works for the operator to maintain enough workable signal into the channel (as much as possible) while leaving plenty of headroom to not clip or threaten to clip. And modern metering gives us a general baseline for that. So we don't have to do it blindly.
Like I said, the gain control, by itself, does not cause feedback. It's not on an island. If it did, it would cause feedback when you turn it up, before you pot up the channel. It doesn't. It's the interaction between the gain and the channel and channel sends (and main and mon outs)...
BUT... that interaction HAS a FIXED variable: The input gain itself has a meter to show us optimum levels. It's the one setting we can actually KNOW where to set it within some reasonable boundaries. We have a meter to guide us. It's the other points AFTER the gain where mix subjectivity (and physical placement issues) creep into the equation... where good and bad choices can actually cause feedback.
All of this is why setting the faders to unity and then bringing the gains up is a poor way to mix. It was fine 50-60 years ago... when we needed workarounds and mixing hacks to try and find some workable way to get SOME kind of baselines as a starting point. But that is because metering was weak back in the day. Especially accurate and scaling peak metering.
When you worked with ONE PA, and especially one band, it would become repeatable, once you finally tweaked everything as best you could in the entire system for gain staging.
But change one variable, and you're back to square one. Bigger amps, more efficient or less efficient speakers, a new singer, new mics, the drummer tuning his drums different... a new guitar amp... an extra member...
All of that is easily accountable for, via meters.... Your entire mix as far as gains and basic levels can be within repeatable norms. If you connect the mixer to a different rig, you still set your mixer just like always, following your input meters and you KNOW roughly where your comp thresholds and gate thresholds should be... your FX sends... your monitor sends if from FOH...
IF you then bring a fader up and realize it's way too loud too quick... then you KNOW the system itself (AFTER the console) is gained too hot. And you then have the option of choking things back on your master (the bandaid fix) or finding where in the sound system you are getting too much gain. Then address it. That way, your standard mix baselines on the console remain consistent and repeatable. Within easily tweaked windows.
Bringing up faders to unity, and then bringing up gains until loud enough means you have absolutely no consistency from rig to rig/mix to mix/band to band. Nor are you EVER optimizing your gain structure, unless you just luck into it. It's a terrible way to learn to mix in the 21st century. In fact, we've had the tools to easily avoid this archaic approach even on club systems since the late 70's and early 80's.
I know old myths and habits die hard, but the truth is, sometimes that's just what needs to happen.
i read it all.. thank you for the ideas.. @@AlanHamiltonAudio
people saying the first mistake was buying a Behringer: we heard you the first 400 times and it was never funny.
It wasn't meant to be funny. Just stating facts.
@@image66media what facts?
@@image66media That used to be true until the day Midas was acquired and the X32 was released. Since then they've become the most popular option around, at least in my area. Everywhere that has an installed PA uses something from the X/M32 ecosystem.
I’m trying to figure out how to send only a band of frequencies to my subs from one channel. (Electronic drum kit, I only need bass level things to go to the subs at the club owners insistence) and I can’t figure the process on this board if it’s even possible! Argh 🤦♀️
It's possible. I do it a lot (although when I do it, I'm still sending full range signal from kick, toms, bass, etc. to the mid/hi's, and then using a sep feed for the subs... The separate sub on an aux control lets me keep vocals, cymbals, etc. OUT of the subs). But the process is the same, except if you're just in a situation where you don't need full range reinforcement on your instruments, but DO need/want reinforcement of the low end (to give things a little oomph at low volume levels) then you simply unassign those channels from the mix (or use subgroups and balance there if you want a little bit beyond stage volume for the rest of the signal) while sending things hotter to the subs.
This video is for the XR18 but the gist of the info is the same to at least get you thinking how the concept works:
th-cam.com/video/ZBXaR2W2E1I/w-d-xo.html
The X32 DOES have crossover filters which means you can do this without outboard gear (like is mentioned for the XR18... and the X32 also has that CENTER control that can be your sub feed versus using an aux (although the concept is the same and you can do it either way on the X32).
To do this for one channel is kind of using a sledge hammer to pound in a carpet tack, but I do this ALL the time with my systems.
Each channel can be set to route the output to the 'MONO/CENTER' output (go to MTX/MAIN layer in the BUS section of the desk - bottom button) right next to the L/R Main fader.
In the SETUP area there is a setting for "L/C/R" or "L/R/M"...set it for L/R/M.
There's also a setting right beneath for "M/C follows L/R"...set this to ON.
Whatever channel your lick drum is coming in on, select the channel and in the MAINS section (next to the screen on the left) push the button that says MONO and it will light. Turn the encoder right above the button and that will affect the level of the M/C output. You will need to set up a separate XLR feed to amp channel(s) to power up the subs (or go direct to the subs if they're self-powered.
And Bob's your uncle!
Your can select filters and EQ on the M/C output channel to suit your purposes just the way you do on each channel.
I just route a separate output for my subs through the mono center out. Because it has its own send knob next to your pan nob. Being able to control subs separately is really important.
My old Yamaha board, my first digital board, took me a bit to learn as I was used to analog. But the manual, which was quite extensive compared to a lit of manuals to newer consoles these days, it was stated to get metering close to "0". Why would anyone change it if that's kinda how it's been for decades? Even with the difference between digital and analog, the algorythem is written by man and you would think could be designed any way one would like. So wouldn't it make more sense to just keep it simple? Thats like having different size lug nuts on the same wheel of your car. Kinda stupid.
In Behringer's case they have some vintage plugins that would be working at their clip point if you pushed the inputs to their max. They, for whatever reason, emulated the analog metering, and scale for those plugins.
Of course when you think about it, on an analog console where there's +16dBu, +18dBu, +24dBu above zero where it clips, then it kind of drops back to an apples to apples comparison, as long as you know what the meter is showing you.
On the M7CL they're basically saying push the input and create your own margin of error and headoom because they don't want you hitting that "Overload". Their meters are dBFS too.
But, they apparently don't have any plugins that would freak out if you pushed close to 0dBFS when that means the plugin is seeing +18dB above it's nominal/zero point).
If you gain stage an X32 to -18dBFS and allow some 3 to 6dB peaks, you could drop that console into the same rig as an analog console that was normally running 0dBu on its meters... And basically be an invisible change to your gain staging and the rest of the rig.
I did have some of links talking about all of this.
Here are a couple of them:
behringerwiki.musictribe.com/index.php?title=Clipping_-_What_happens_when_the_signal_goes_into_clipping%3F
support.biamp.com/General/Audio/Gain_structure%3A_input_and_output_levels
Five examples of common mistakes I've seen people making when mixing. Especially on digital consoles like the Behringer X32 and XR18 (and Midas versions)... And at the end, an example to tie it all together and show one common problem I've been called in to diagnose at various service calls... with people thinking they had a system problem, and it was operator error and a misunderstanding about how things all interacted in the console.
I did a similar video for the XR18, but this one is more in depth, longer, with more background info in the sections and specifically targeted to the X32/M32 and filmed using an X32 and the X32-Edit software.
Let me know if there are any questions or further topics like this you'd like to see explored.
Patreon Page:
www.patreon.com/AlanHamiltonAudio
Amazon Affiliate Links-
Behringer X32 on Amazon:
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Midas M32 on Amazon:
amzn.to/3ovhi8n
Behringer XR18 on Amazon:
amzn.to/2LfTpmO
Midas MR18 on Amazon:
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Suggested Videos:
How to EQ a Kick Drum:
th-cam.com/video/CQg1_-ZJ4MU/w-d-xo.html
How to EQ a Snare Drum:
th-cam.com/video/gITnB0hQ9as/w-d-xo.html
Behringer Vintage Reverb Tutorial:
th-cam.com/video/6r5k1P_ZGsQ/w-d-xo.html
X32/M32 Gain Staging and Signal Flow:
th-cam.com/video/7MJ3rfj7HNc/w-d-xo.html
X32 Tech Tips Playlist:
th-cam.com/play/PLWtgwSNlxTjPJCmx6FF1igBWd30PxLbCG.html
Tech Videos Playlist:
th-cam.com/video/5gzsEErKdb8/w-d-xo.html
Nice video. But where do you get that accent?
Living too close to Kentucky! ;)
It's also why I don't listen to older videos, or any of my videos much once they get uploaded and tested... ;)
Although I have changed the monitoring setup and script process over time to be a little more mindful of it and stay better focused on the narration and not too lax with it.
I don't even have an X32 but I learnt alot! Thanks
Thanks for watching! Glad it was helpful! :)
👍
I would disagree saying that unity gain is a bad method, I'd say its a very good method, it creates a very balanced mix, very quickly. As engineers we shouldnt rely on meters, we should use our ears, plus unity gain works on any console, analogue or digital. It also means it puts the fader with the most incremental control over level on that channel. Engineers have been using this method for a very long time, and it hasnt failed me yet.
You'll get that anyway if your system gain is correct everywhere. And if it's not correct, it'll never be correct if the meters aren't used to get it correct. Meanwhile, setting faders at unity and dialing on the gain knobs with little concern for the meters creates inconsistencies from show to show if the rig is changing (such as provided racks and stacks at each venue) and your gains aren't consistent. Everything (sends) after the gain, except the channel fader, will not be in the same basic range if the gain isn't the same for the same source.
And your frame of reference will be constantly changing. You can't just look at a send, such an FX send, or a mon send if you're doing mons from FOH, and see that it's in the ballpark of correct (or see a problem quickly developing) if you're gains aren't consistent.
It's also a terrible method to try and teach someone about gain structure, because there really is no structure. It's also almost ignoring the point of the faders too... Your method is a system that works best when gain structure is already set everywhere correctly. And the system and/or gain structure then remains the same.
I could do your method on my system, and it would roughly end up about the same... because the system gain structure is already correct. But I could take my console and connect to a different system, with questionable gain structure on that provided system, and your method would have no consistency and now everything in the console would be off. It would still work, but I couldn't really rough anything in, let alone recall a baseline mix or preset and already be close. I'd effectively be starting from scratch. I'd have to make a mental note that all my gains are off by whatever dB +/- and then start applying that math to my sends... and still needing to listen and adjust as that would happen anyway. And my gains being off by XdB +/- wouldn't truly be consistent either, because those would still be relying on my source to be the same level... which may or may not be true. So that math would be on a channel by channel basis. So it's back to scratch for every send and resetting them.
Meanwhile, I could've started with proper gain structure using the meters, that I set using the same method every time, that will get me consistent results, even if the source changes level, and give me consistent and recallable send levels.... just by using my input meters.
Worst case scenario, I have to adjust my main output level (or mon/aux main outs) to counter the gain structure issues downstream of the console. And this is something that is teachable and recreate-able versus a spaghetti on the wall approach that will never be the same when you start changing downstream variables. Monitor sends... FX sends... stream feeds... fills... etc....
I'm not saying a person can't learn to deal with those changing variables and compensate... But I'm saying "Why should they?" when there's a perfectly fine input meter(s) to use that allows for 100% consistency and simplifies this? And guarantees, no matter what else is happening, you at least can know your gain structure in the console is 100% correct and set exactly the same, each and every gig.
This is especially important for those that have to mix mons from FOH.
And it actually is much easier to teach someone what the meters should be... and that if their meter is correct and they can't get the channel fader to reach unity and not be screaming loud, feeding back uncontrollably, or not nearly loud enough, then they know they have a gain structure issue down stream of the console. Which they can solve with the main outs of the console in quick combat conditions... or by sorting out the equip downstream of the console if they have the time. Your method will never teach them that.
Your method also gives people this mistaken impression faders need to ride 100% at unity (even if that isn't what you mean), and if they don't, they need to tweak and/or mix on the gain knobs. Which, leads to FX going up and down, and when mons are from FOH, inconsistent/changing monitor levels, and inconsistent stream and broadcast mixes when it's all done from the same console. Which is more and more prevalent these days.
We have meters, we have faders, we have ears. I don't know why people wouldn't want to use all of these and bring order and consistency into the equation and then mix from that foundation. And it sure makes it easier to teach proper gain structure methods, and how to recognize gain structure issues throughout the entire system when you start with something consistent.
I tend to find myself using a mixture of both. The thing is, different instruments/inputs often have a different perceived loudness, regardless of the peak level on the meter. Typically I set the gain based on getting a healthy meter, and typically the faders will give me a good visual reference of where I have something set. HOWEVER, if an input is hyper compressed, the peak will have been brought down. Meaning if I set the gain to the same metering point, it will be much louder than other, less compressed (or should I say saturated) inputs. This would be very bad in the scenario I'm setting a monitor mix, since I now don't have a frame of reference for how loud it is based on where my slider is.
Wow, Alan schooled you lol.
@@AlanHamiltonAudio Ive read what you said and although I agree with bits, others I dont. Mixing with unity gain ive found to be very consistant, because once the input gain has been set correctly for the source, the mix balances itself very quickly, and then you only need to adjust the faders, and you get the most amount of adjustment at unity.
Meters are a tool, not absolute, I rely on my ears to mix, not the meters. Especially when the atmosphere of the venue changes as the evening goes on, it gets hotter sound changes in the room, If I relied on keeping it at -18, then I couldnt adjust for the room temperature.
The way and many other engineers mix is keeping audio between -18 and -12, however some venues have to be louder to overpower the noise floor of a natural reverb in the room, so then the source is louder than the room.
@@StraightNoChaser86 Seems like you have nothing contructive to say, no worthy input into the convosation. Just ignorance.
Regain feature in mixing station app fix the last problem
Yes, that's a great feature and something that isn't really possible with analog consoles... outside of math and memory and working as fast as you can to do the resets ;)
Be aware of gain changes if you multitrack the show (pre processing) though :)
But why do these topics not apply to any other desk ?
They do, and it's mentioned in the text description of the video:
"The information is also relevant to other consoles, including the very similar XR18 or Midas MR18, but the examples in the video are filmed using a Behringer X32 with FW4.04 installed (and this is functionally the same on the Midas M32). Also, the latest version of X-Edit software (X-Edit PC V4.2) can be seen in use in some places in the video. This is the same as the Midas M-Edit PC software. The Mac software is also very similar.
That said, these general mixing topics are relevant to consoles from other manufacturers too, but obviously the surfaces and GUI will look different on those."
But that said, the target audience are Behringer/Midas users, many coming to digital for the first time. Besides the fact they are purchased by many bands, they are also installed in many venues (clubs, churches, etc.) with users that might not be as familiar with these consoles, or digital consoles. And for many, they will be searching specifically for the Behringer (or Midas) X32/M32 topics and would likely overlook a video focusing this info on another desk.
Applies to all consoles. Number one is getting your gain right. It doesn't matter if its analog, or the most expensive digital, get the gain right. Not only is it input level, it is also channel tone.
I believe the advice on the unity gain setting is misguided. It is a standardized technique that nearly all mix engineers embrace--especially for live sound. I should be able to set down at any preconfigured sound system and expect the standardized fader position to be at zero. This takes the guesswork and other surprises out of a mix. This has nothing to do with 50 year old technology that dictated this rule-of-thumb, it's just a best-practice thing. It makes no sense to have to have a microphone fader set to -8dB forever just because somebody is holding to the -18 dBFS meter reading on the gain. While I understand why you made this suggestion, and in some circumstances it is good advice, it is also misleading. However, where you are correct is when we are using compressors on the inputs because we can "trim" for unity gain using the output gain knob of the compressor. Regardless, I do teach everyone to strive for unity gain setting on the mixer so there is no confusion or mistakes made during a show.
The problem is, you won't be sitting down at preconfigured sound systems that are always correctly setup as far as gain staging. If they are then either method will roughly equal the same basic result as the faders will end up in the neighborhood of unity anyway. But if the system is not properly configured, then your gains will either be way too cool or way too hot trying to compensate for poor downstream gain structure practices.
And no consistency on the channel strip from gig to gig since the system is always changing and you can't be certain proper gain staging has been followed always. An experienced engineer can probably recognize it quickly if his or her gains are way off the mark... but then, that doesn't necessarily hold true of they aren't on a console they are very familiar with. And it still won't yield consistency.
When you simply follow the meters and optimize your inputs, then that is always going to be the same from into the console to the output busses. If the channel fader can't reach unity, then that tells you there are gain issues after the console... but you can address those at the system level, or at the master fader (and master bus level faders).
And if you're on a console with matrixes, then that is truly the place to address the master out since even your master fader, and any inserted compression or limiting would still remain at normal operating levels from gig to gig.
So we're really talking about two means to an end... but only one method has repeatable consistency and optimized gain structure from the input of the console to the output bus. The other relies on all of the variables after the console to be correct each and every time. Even when the system is changing. And 'correct' is a window. Just like with your input meters and the operator being a little more or less aggressive on certain inputs, systems can be setup more or less aggressive too. So the only method that has any guaranteed consistency for the mix engineer is if they learn to use their input meters to set gain and properly and consistently deploy them. Then use the master faders, or matrixes, to account for gain differences between a properly gained console and a questionably or poorly gained sound system.
And, these days, with consoles also doing more secondary mixes like stream feeds, rough mix feeds, feeds to video, fills, and USB feeds to a DAW for tracking... it becomes even more important to have a consistent and repeatable form of setting gain and optimizing gain structure.
For those that have to mix mons from FOH, this also comes into play because you need that consistency there as well to keep your channel mon sends in normal operating ranges.
So, while the bring the fader to zero method and start turning up the gain knob method 'can' work, it relies on the downstream pieces to be consistent for that method to be consistent. And unless the console is connected to the same system every gig, that's not ever going to be the case.
And that's why it's a poor method to teach, learn, or rely upon.
Why deal with inconsistency when you can use the meters and consistently maintain optimized gain structure throughout the console? It's the one piece of the system where you can have absolute consistency. And that you will be standing or sitting behind each show. Like I said, the faders are still going to roughly end up at unity anyway, but the difference is, the only change to get them there will be the master fader(s) to account for the system gain structure of a provided racks and stacks rig. Not every channel send and insert. That all can remain consistent to your baselines and visual cues remain the same too.
It's also not something that nearly all mix engineers embrace. Maybe in 1970... and even 1980... But as meter bridges and proper channel metering came onto the scene, it's faded away and is now 'a technique' but no longer 'the technique'. It's actually pretty rare. More complex systems and technology have brought better tools to the job. It's pretty rare to see someone mixing by bringing the fader up and THEN setting the gain, ignoring or almost ignoring the input meter.
What has been embraced is technology and taking advantage of all of the tools at our disposal these days to give us solid baselines and platforms to work from.
And it's much easier to teach and explain a method that relies on consistency and meters at the very first stage of the console. And it allows even inexperienced operators to see there's a downstream problem easily enough... and know exactly what can be done about it.
Setting proper gain structure throughout the system is paramount, especially input channel gain. Part of getting that correct is getting the source correct to begin with.
Unity gain is all about SNR. With proper gain structure, unity will give you the best signal with the least amount of noise. It is one of the least understood things about a desk.
While digital is quieter than analog by a long shot, electronic components still have their fair share of noise. It simply stands to reason that we want the best sound possible. If not, don't worry about gain structure.
Far too many that don't understand unity gain think we're saying park the fader at unity and never move it. As well, they have the misconception that we also believe just turn the gain up to whatever you like.
This couldn't get farther from the truth and for those few that actually teach that, they don't have a clue.
Whether you put your fader at unity before you set the gain or after makes zero difference. The fact is that if you actually get your gain set properly, your fader is going to land around unity and if it doesn't, with very few exceptions, your gain isn't set properly somewhere in the system.
Here's another big one. The fader had to stay exactly at unity. NOT AT ALL! If things are set properly unity is a "home position" if you will. A place the faders should be close to but doesn't have to be exact. With very few exceptions, if you're wondering further than 3-6dB from unity, things are probably not set properly. Is it your fault? Could be but it may not be at all. Talent is notorious (especially unseasoned or just the idiot ones) at turning things up after soundcheck.
An exception to this could be overheads on drums, especially if the talent is using ears. The fact may just have to ride lower for FOH.
We're also accused of mixing with our eyes. Wrong as well. I run my fader up, set my gain hardly ever looking at the meter (which I don't suggest unless you're very seasoned) but anytime I've been questioned on it but an A2 or whoever, check it and it's darn near always where it should be.
Having said that, I run things hotter than most because just like analog, most digital desks (not all) sound better if you drive three preamp a bit. Obviously you don't want to clip digital like we used to drive analog well above 0dBVU.
If I'm training someone, I always teach them -18dBFS as the equivalent to 0dBVU which should be their average.
Many people also don't seem to grasp what happens at a bus send or output when you have 30 or 40 channels stacked up. We can easily overload the bus, subgroup, output, etc quickly.
Back to proper game staging through the system, brings up one more thing that so many people felt a realize...
The knobs on the front of a power amp or not an output volume control, but an input gain attenuator. Big difference!
Depending on the voltage, the mixer is putting out, you can drive an amp to a full output power even though the input knob might be only set to halfway or anywhere under full-on. This is an extremely important part of getting the entire system from front to back set up properly.
They don't understand that EQ is nothing more than a volume controller particular frequency which can drive the gain up on a channel if they're using the improper method of boosting EQ instead of cutting it, nearly 99% of the time.
I see so many people hear a sound that's muffled so they just naturally grab the high frequencies and turn them up. It's like putting a Band-Aid on the thing and all we're doing is introducing more noise, instead of going and finding what's muffled, and cutting those frequencies for the clarity we're looking for.
I can go on and on. I know I'm getting a little sidetracked here, but I'd like to mention one other thing real quick just since I'm commenting...
On your video you talked about using a HPF on every channel and I couldn't agree more. The one thing I believe that needs clarified is when you mentioned that unless you have a bunch of subs with a bunch of power, you need to do this.
A HPF has nothing to do with how many subs or how much power you have but it has to do with the roll off of the box. Whether you have a single 500w sub that rolls off at 40Hz or 100,000 Watts of subs that roll off at 40Hz, they are identical when it comes to the HPF. The only difference is with all that power you may not notice it stealing your low end strength, but certainly you'll probably notice the low end mess , but the principle is the same. Wherever your box rolls off is where your HPF needs to be at the very least. Obviously on most channels it's going to be considerably higher.
Thanks for making good videos because there's so many people out there giving bad advice, which does nothing more than race people with bad habits. Eventually we'll have a bunch of people saying I've been doing this for 5 decades, the only problem is they've been doing it wrong for 5 decades.
@@sisterpete Thanks for the backup.
I don't think I said not to use a HPF if you have large subs and power though. At least that wasn't what I meant. It was more along the lines of unless you had large subs and power your system likely doesn't reach down all that low in the first place so a conservative HPF is what you likely need. IOW, no use setting a HPF setting of 30Hz on a rig without subs that doesn't go below 50Hz-60Hz in the first place. And not only no use, but also how you get into over-excursion on the cones.
So the point was to still basically always to use the HPF, but without subs the system cutoff wasn't likely to reach all that far down in the first so would need a higher cutoff than a system with subs. But not that a system with subs didn't need the HPF.
@@sisterpete, I find that properly setting levels "old school method" and today's "gain staging method" are actually about the same. Clipping is clipping no matter which method you choose. Unfortunately, the modern-day proponents of "gain staging" are failing to understand that we're saying the same thing, but just talking past each other. It's easy to denigrate the old way because we failed to properly learn the old way!
Digital mixers and audio processing that uses 32-bit floating point do not add noise to the signal in any meaningful manner. However, it is true that the processing itself, and the algorithms used WILL add noise just because that's literally what they do. A compressor with significant make-up gain will add noise by amplifying the noise already present in the signal. Part of this is the nature of the channel pre-amps as well as the AD converters.
A proper old-school method for setting levels is based on the principles of not only keeping SNR to a minimum, but also minimizing the amount of signal modification. A signal passed through near "unity" with little levels adjustment will retain phase accuracy. Clipping distortion is always an issue (unless you are using tape saturation) on the one side, and SNR is an issue on the other side. The ONLY main difference between an analog signal processing chain and a digital processing chain is that clipping distortion is far less forgiving in digital and digital is usually more forgiving in SNR. But in the end, it's almost always best to adjust your input gain in the neighborhood of 0dB on an old-school VU meter or -12 to -18dB dBFS. The two are about equivalent. (not exact, but for purposes of this discussion, they are).
There are two reasons why I almost always tweak the gain as to land my faders at 0dB. My faders at -dB allow for more granularity in level adjustments. A half-inch physical adjustment in the fader position has a much different amount of adjustment than it does if the fader position is at -20dB. Also, there is the "at a glance" identification of something being amiss. When I soundcheck and zero out my faders, if during the show I'm having to jam the fader up against the top stop, I can tell without even PFLing if the 9V battery in the guitar is dying and we can fix it between sets. When the faders are all over the place, it's hard to identify when things change.
The proponents of the modern day gain staging method create a lot of work for themselves and almost always have a less consistent mix throughout the show. Unfortunately, they are caught up in this modern religion of rejecting the "old ways" because they happen to be the "old ways" for when we all used analog gear.
Proper gain staging and level adjustment is definitely more important with digital mixers where we have gates and compressors on EVERY channel. But this leads to another whole discussion as to the purpose and intent of the dynamics processing. Far too many of us are using it as a crutch instead of an effect. I do both, of course, and the input level into the processor is dependent upon the purpose of the processor. The output level from the processor is then adjusted to provide a "normalized" level. Rarely should a vocal mike have any makeup gain. Use it more as a limiter instead of a compressor. A bass guitar needs a compressor with makeup gain. Proper gain staging is dependent upon this as we want to use the processor properly for the intended purpose.
Yes and no. There’s more to it than the video covers. I’d rather give my hihat mic less gain and have more fader resolution at unity. Same with bottom snare. But most other sources I’ll gain up to -18 just as the video describes. If the faders are way down then there’s way less control.
Your mistake is to use -18 as a gain point in FS for peak content. The -18 to -21 dBFS is for matching 0 VU dB REFERENCE. Most analog gear has at least 20 to 24 dBm of headroom, that's why 0VU reference tones are -18 to -21 dBFS, depending upon manufacturer. Some, like Sony, recommended -20, Tascam used to recommend -18 dBFS. Utlimately is depends upon your musical content. If you have wide dynamic range, like classical music, set the 0 reference to -20 or 21. For more pop and limited dynamics, use -18 and you will maintain a tighter dynamic range in your mix.
You can take advantage of the headroom and set gain to above -18. Usually -6dBFS gives plenty of headroom with most sound sources. Not sure why you espouse -18 for gain levels. That's totally bogus. Otherwise, you stuff is spot on. The -18 dBFS is for matching 0 dB VU reference tones!
This advice comes with more than 40 years of pro audio experience in recording, video, FOH, MONS and radio production on almost every type of gear imaginable. Look closer into the purpose of -18 dBFS for reference tone. Peaks can certainly go much higher with no problem.
First off, it's Behringer's reference. Secondly, it's not the peak but the average. Third, the operator can be more aggressive, or less aggressive, to a point. Experience plays a factor here.
PEAKING to -6dBFS is pretty aggressive but should be OK since it's still allowing 6dB of headroom. That ASSUMES, that -6dBFS peak reading really IS the peak. Then it should be OK, though the margin of error is only 6dB so the operator better have the experience him or herself to know that's really the peak. But, I'd never recommend someone set the console for a -6dBFS AVERAGE. -18dBFS is a much safer and conservative average. -15dBFS could work as a more aggressive average... even -12dBFS... But those ever more aggressive averages leaves less and less margin of error for peaks.
Also, for someone's first foray into mixing, it's much smarter to be conservative rather than aggressive because they don't have the experience to realize soundcheck and performance levels can be different. Especially when the performers aren't that experienced themselves. Let alone the ones that play games and soundcheck quieter on purpose so they can be louder for the show.
But I digress...
Yes, -18dBFS aligns, essentially, with 0dBVu. Which is the point... Most gear roughly has +16dB to 24dB above 0dBVU for headroom before clipping. Downstream gear will see that -18dBFS on the output meter as basically 0dBVu. So, building the inputs to average -18dBFS doesn't start you in a hole that will have too much gain/output in the output section of the console. And leaves you without headroom on either end. Meanwhile, for years, 0dBVu was the rough goal on an analog console (still is), so -18dBFS is essentially just that on a digital console/meter.
The whole point is to have headroom above the PEAKS, and nobody said to set the inputs to PEAK at -18dBFS. BUT, certainly, setting them to average -18dBFS is a good, safe gain structure practice that should see that nothing is PEAKING too close to the "bucket full" number of 0dBFS.
I push digital desks harder than most, but for most people that are used to analog gear, the first thing they should do is be sure to understand that -18dBFS is the new 0dBVU. Get your input gain stages set correctly with that as your average and life will be happy.
While there's not one "official" standard of measurement for this, and certainly, even though behringer suggests -18dBFS, so does nearly every other manufacturer of digital desks, so it's not "totally bogus" by any stretch of the imagination. Heck, I recall back in the late 80's, using my first digital recorder made by Hybrid Arts. Even then they suggested -15dBFS be referenced as 0dBVU.
The reason for this is fairly simple. Back in the day, hitting an analog preamp hard was not only ok, it was very desirable in many cases. If you've been in the game for over 4 decades, as have I, you'll certainly recall we use to saturate the heck out of tape also. But with digital, there is no headroom above 0dBFS, it does not exist, in other words, you've hit the digital ceiling. it's a clipped signal and that's not pretty in the digital domain.
If you recall, in the analog world, we'd generate a sine wave at 1.228 volts and calibrate our meters to read 0dBVU at that level. dBVU wasn't as matter of fact as dBFS, which is absolute because it's all math, but it meant something if the system was calibrated at the 1.228 volts, which was an excepted standard back in the day, just as -18dBFS is an accepted standard in today's world. Accepted, not official.
Also for what it's worth, converters will vary some. So depending on your converters specifications, with very few exceptions, at the same voltage, 1.228, 0dBVU will fall somewhere between -18 and -22bBFS
Many digital consoles sound better when driven harder, I do it all the time. But most certainly I wouldn't even remotely make a video telling unseasoned mix engineers to do so. I would do exactly what he did here and give them the proper guidelines.
There's FAR too many people that know little to nothing about sound offering advice these days. Blind leading the blind. He does'nt need to add to those numbers.
I agree. -18 came out of the "DAT days", so to speak. It was a safe reference level for tone. This was for tapes going to duplication or mastering and they would could set their level.
Easy method for obtaining auto tune using x32?
Not that I've tried.
Why?
We do this with ableton through dante but only for our live feed.
That's on the singer. Or buy a unit.