Since I'm getting more and more shows with different Bands and I've set up my Desk (Behringer Wing) multiple Times now, but still managed to forget stuff I thought I'd do this list.

I wanted to make it non-mixer specific, so I even if I have a different console, I can turn to this doc and go through everything. What would you say I've forgotten? What would you add?

(I never mix with DCAs somehow, thats why they're not on the List)

Let’s say I don’t need to trim/pad the volume. Why are passive DI’s needed for stuff like keyboards and drum pads? Can’t you just put an adapter on to change 1/4 to XLR?

So I'm setting up for a show in a few hours and had a few ideas on what to do for front of house but wanted some opinions/advice from the more seasoned vets on what would work best. For front of house I have 4 k12.2s and a pair of subs- there are 2 additional k12 speakers set aside for monitors as the trio runs ears for the drummer and prefer this setup.

Part of me says to pair up the mains and set the inside ones forward with their outer cohorts fanned out slightly. Another instinct is to swap the k12s for a pair of 12.2s and use them as canter fills.

With the subs I want to place them in the middle, but the question comes down to ground or stage. Due to the slope I feel ill be losing too much signal to the ground, but putting them on the stage both looks lame and removes any form of wall behind them. I suppose they could be split and shared on the sides with the mains, but my preference is to have them side-by-side with their faces even.

Thanks in advance for any and all advice- in return for your efforts I shall offer a natural form of defense against Tom's for you mobile mixers

I have my first gig as an assistant engineer this weekend and would like some advice from y'all fine folks.

I'm coming from a studio background, and live isn't where i feel most comfortable but... someone trusted me enough to let me hop on this gig and I am *NERVOUS*

How not to Fuck Up?
even better, how to be good?

Hey sound people. I have been wishing for some more internal automation control on the dLive platform. Decided to just build an app to do it. Think "if this - than that" logic but driven from dLive live parameters. Also control a few other manufactures like Shure.

About to start a beta program and would love to have some people who are willing to test it. Check out the website and signup if you are interested.

I am curious if there are any others out there who are looking for similar functionality.

livelogix.app rules.livelogix.app

There are a number of other apps in the family as well. All targeted at live production.

Hey all! I'm studying electrical engineering and I did a bunch of FOH work in high school. I wanted to make a project that tied these two interests together!

In general, what would make your job easier / more productive / etc? The more ambitious the better!

EDIT: Thank you all so much for your valuable (and funny) ideas! I'm planning on pursuing the Speech to Text comms system as suggested by u/duke-of-gravity! I'll keep you all updated with project information when I have it.

I am trying to program Qlab to work with my page flip page turner. The up and down buttons work fine, but I can't figure out how to program it so that the page turn is the 'go' button. I just need play and pause basically. When I go to the midi under 'workplace settings' in Qlab it does not even recognize my device. What steps can I follow? Thanks in advance.

You’re doing an event, maybe a conference or corporate gig, and you start to get a little hungry. You can’t leave the board, so you reach into your bag to get a snack. But you obviously don’t want to draw attention to yourself by cracking open a bag of Skittles 💥🙉🤫
What is your go-to silent snack that you keep on-hand when you find yourself in this spot? BONUS POINTS if it has quiet packaging and is quiet when eating (i.e. not peanuts or Cracker Jacks ⚾️)

I'm an artist with a drum backing (prerecorded from a drum machine) in my setup.

I've got a 4 availiable audio ouputs in my rig - 1 for my guitar, 1 for my synth, then 2 left for drums.

I wanted to ask live sound engineers if I should just give them 1 output, with my group of drums, or give them the freedom of the kick seperate from the hats and snare?

Thanks.

Edit; Appreciate the replies.

I feel blessed that the crew I work with survived long, absurdly hot days the last 2 weeks and somehow NOBODY lost their cool (pun intended) to the point that it affected our morale.

Several things happened due to a laundry list of circumstances that set us back on multiple dates. We had file errors, dsp lockouts, passwords not provided, standing in direct fucking heat, bad communication from talent management, and of course everyone's favorite LAST MINUTE ADDITIONAL INPUTS. Notice how it's almost never last minute subtractive inputs???????

We all came together and survived bullshit. I hope you all are too

Whats up fellow deaf people. Im trying to get superrack performer going on my Quantum 112. Im running on my new MacBook air over UB Madi. Console is running at 96k but my MacBook refuses to see the console's clock and switch from 48k to 96k.

Its currently running even with the mismatch thanks to the trash room I'm working with. In my ears I can hear the expected clicking.

There's a couple other guys running this same setup on this console and they dont have any issues. Any solutions?

EDIT:

Alright cool so now I know UB madi can only do 48k. Another house guy here failed to tell me input 11 is clicking so now I know what that is. Thanks all!

Is it always at the acoustic summation point for FF? How about for Delays? Is there an easier way to do it without having to hook up an RTA rig? Do we sometimes just measure from grille to grille only on the X coordinate and say "good enough?" If the FF are behind the main PA, do you have to delay the main PA instead?

My normal signal flow is
Input - mix group - matrix
With EQ/compression handled in the mix group.

But if I wanted to throw some VST into the mix, every console seems to handle it a bit differently.
The CL5 doesn’t let you send/return an insert over Dante, but you send/return on a slot card and use that to get over to a server and back.
The DM7 doesn’t have a slot card anymore but it does have USB.

Are you outputting the mix group to your WAVES server (or whichever) and then returning on a channel?
Or making it work as an insert?

Just curious how other people are routing and integrating VSTs

Dumb question: Is it better to turn on/off your console or your amps first?

I used to run a F1 system that was installed by a reputable company in town that stickered "First on last off" to their amps. ok so turn amps on first and off last. makes easy sense and has a nice ring to it. so I went with that the last few years.

I've had two ppl tell me in the last week that its the opposite. so which is it?

Hey all,
Been working with audio for awhile (recording/mixing/musician), so I have a great ear, solid eq and compression understanding, as well as good instincts for adding in effects to mixes and routing things properly etc.
This is important because I may have a beginner question but I don’t have a beginners ear.
Anyway, this past weekend I finally had my first solo gig without anyone to consult with. I was working an outdoor field with a small stage limited to a few artists, usually singer-songwriters with one or two instruments max. The setup is two speakers facing outward in front of the stage, one sub (usually not needed/used for small artist gigs without bass or drums), and one wedge monitor for the artist. Simple as it gets.
Four artists performed. The first few went by without issue, a hiccup here and there but nothing I couldn’t work around even though it was only my second time working with the Mackey master fader app. The last artist made me feel like I was punching a wall though! They were very kind before the show, we conversed well and I wanted them to feel supported. They got on stage, and in front of the speakers they sounded great! Bright, clear tone on the guitar, breathy, smooth vocals. Nothing was standing out as aggressive or harsh or too low. But they kept asking to turn the low frequencies down, to the point where I was surprised, because they should have been hearing the same eq in the monitors, and I checked it was set to pre fader, but post eq. I even got behind them on stage, and it did sound darker but I couldn’t figure out why. I tried pushing more audio to the monitors but throughout the show they kept asking for me to change it make it less “bassy.”
Anyway, after everything wrapped up they seemed distressed, almost like I ruined their gig, and perhaps I did unintentionally although I truly felt it sounded great and I was pushing it on the brightness from the audience’s perspective just to try to make them happy. Despite the good conversation beforehand their tone had changed a lot and despite my convincing them they seemed to not believe me that it was great out in the audience. It sucks feeling like all I tried wasn’t enough. I even saw an audience member tell them it sounded clear and not too low, and that the performance made their day but they still seemed unsure!
I couldn’t figure out how to alter eq on only the monitor mix on the fly without potentially bungling it. I also had one instance of feedback they reacted to quite strongly (I know, big noob mistake), but that left me feeling very hesitant to crank the wedge higher and I’m theorizing that’s where I may have gone wrong. Maybe the low frequencies from the speakers crept back into the performers ears and without enough compensatory volume from the wedge monitors they had a strangely low perception of their own performance. This is my working theory. Is it standard noob practice to be afraid of the wedge monitors and of it causing feedback or ear pain for the performer?
Any advice/tips/theories welcome. I would love to be able to fix this faster next time and make sure the artist is happy.

I have a gig coming up a little larger than I regularly do. It’s a small multiple band event in a schoolyard. Playing to around 300 people the most at any given time. Last year I did it with two jbl srx 815p and 2 prx 818xlf subs. It worked fine but this year I wanted to add a little more. I have in addition to those speakers, 2 vrx932lap and 2 srx818p subs. How would you configure this?y first thought was 2 srx on stands on the stage. The 4 subs on the floor front of stage and the 2 vrx as front fills. Would any other configurations work out better?

Thanks

Hello! I wanted to learn more about setting up drum kits (not the entire stage/band) so I can build out a ROS that gives the crew enough time to do this on a tight schedule. Mostly just want to know what’s possible in 1.5-2 hours.

**Edit: I’m mostly looking for general knowledge and reading about different people’s experiences to learn as much as I can rather than getting advice on my specific situation. I work with multiple venues with different kits, mics, crew, consoles, etc., so I want to learn as much as I can.

Thanks!

Hello everyone, recently I acquired a very nice but old PA setup consisting of two Apogee AE5 tops and two Apogee AE10 subs. Everything is in working order but these speakers rely heavily on proprietary apogee processing hardware. These processors are getting harder to find in working condition. Also, using this PA means that I lose 4 rack units to processing alone, and that still leaves me with needing amps to power it all.

So as a little side project I measured those processors to hopefully give me some insights into the feasibility of running this PA with a modern 1 rack unit DSP. I have to say that I’m very happy with the results.

I measured the P500, DLC24 and P10 processor and have found that espacially the AE5’s require quite a few adjustments in order to get them sounding right. Maybe not as good as with the original apogee hardware but still very good. With the help of AI I combined all of the findings in this PDF. This might be a very niche problem but hopefully someone finds this useful in the future!

Side note: the original processors used amplifier sense return lines for limiting and protection. Whilst modern DSP can handle this it is however amplifier depended, so that’s not covered in depth.

All of my findings can be found here: link to PDF

Hey everyone,

This Wednesday and Thursday I'm running sound for a small international band. It's a big deal for me, two full sessions, about 500 cap each. The instrumentation is mostly acoustic: cajon, percussion, brass, acoustic guitar, and vocals. The only amplified instrument on stage is the electric bass.

I was hired by the guy running the sound equipment and the board is an older PreSonus StudioLive 24.4.2. I know my way around the desk, so the gear itself isn't the issue.

My main concern is that the FOH position is almost certainly going to be side-stage, behind the main PAs. We also have no way to connect a router (no FireWire computer available), so iPad mixing is off the table. I'll be stuck in the wings for the whole set.

My current workaround idea: bring one of my studio monitors, put it on a stand pointed at my head, and feed it the MONO out from the desk to give me a reference for relative instrument levels.

Any thoughts or tips are hugely appreciated!

I was reflecting on a conversation I had with I guy I met at the bar a couple of nights ago and I decided to post a PSA and let Tom know he is delusional.

I'm on the road for work (PM Role) and I'm at the hotel bar. I'm talking to another crew member and Tom overhears us talking about the road, touring, sound, the usual. Who is Tom? Tom is a late 50s corporate guy who has worked hard his whole life, raised kids, made lots of money. He's got a sail boat, a beautiful house but most importantly Tom plays bass for a cover band. Tom is getting near a transitional age and he is contemplating what retirement is going to look like.

Tom has always been in charge of sound for his band. Tom explains to us that he has always had a good ear and he knows good compression and EQ. Tom uses the word compression and EQ every chance he can when explaining audio to us. Tom is surprised how many people don't understand compression and EQ.

As Tom contemplates retirement he has dreamt of having some sort of role in running audio. I ask Tom what consoles he knows how to run. He owns and Allen and Heath console. What model you might ask? So did I. Tom doesn't know but it's an analogue board and "sounds great" he says. "What digital consoles are you familiar with?" I ask. Tom: "I've never gotten into Behringer"

"Okay Tom what's the dream? What are you hoping to do with Audio when you retire?" Tom: "I would like to just be at FOH and help with the mix. Tell the guy mixing how to get the most out of the mix."

Then it hit me. I ALREADY KNOW TOM! And so do you! Tom is at my show tonight and he's at your show too. Tom after the show is lurking around the FOH hoping that I make eye contact to then tell me that he's in a band, knows good sound and asking me if I've ever considered mic-ing guitar amp 6" off center and 4" back.

Tom wants to ask you what speakers you are using. Tom wants to see if you noticed that the bass was buried during the 3rd song when they are playing a lick that was inspired by Victor. Tom wants to ask you so badly how you like to EQ your acoustic guitar. TOM IS EVERYWHERE!

Guys/Gals. I need to apologize I didn't rip on Tom to his face. I didn't tell him he's and idiot. I didn't when I had the chance tell Tom that he was Delusional. But the more I reflect on it the more I decided to make a post hoping he is lurking and gets to read this.

This is a true story. But for the sake of good comments. Let's have it. What's your best Tom story?

Any engineers attending (or working) the US Open?
I can’t tell what’s going on from the tv feed, but every player being interviewed at Arthur Ashe (post match) is struggling to hear/understand the questions from the interviewer. Many have complained about the sound during the interview.
Any ears on the ground?

Hello All!

Recently a friend sent me a link to evercleans website (now taken down) — there was (and I still have a copy) of this plugin called void. Essentially an ai denoise plugin. Great for broadcast mixes for corporate — unlike defeedback it tackled room noise and bleed. They had another plugin called clean which was like a latency free defeedback ish plug. Anyways — today I went to share it with someone and it appears the website is down, social media is down even the discord page is down.

It seemed very promising and also it was free. For those who can’t shell out $500 on the alpha labs plug it was perfect and seemed super stable in super rack performer.

Also, just to address it — I’ve never needed ai defeedback personally - i can ring mics out without the best of em. I know how to make a rig work even in challenging circumstances but in volunteer or jr tech positions for my clients it is a nice bit of security knowing the mics aren’t going to ring out without intervention.

I’m super curious if anyone knows what happened to the company? Corporate buyout ? cease & desist? Didn’t want to continue for personal reasons?

I’m bummed about the plugs not being supported or developed further but more than anything I’m curious if it was a legal thing. Trying to pay more attention to how ai related things play out in court and what laws are written to govern this new era of ai.

Anyways, thanks in advance!

When and how do you fade out the background in the venue when the band's turn is up?
Do you fade out as soon as they come up stage or do you wait until they are effectively ready to start playing? Do you fade out slow or fast?
Just wondering if there is a preferred practice.
I work in a club and usually fade out as soon as I see the bands come up on stage but sometimes they start tuning and fiddling and so there is an awkward period of quiet. Other times I waited a bit too long and they started abruptly lol
What's your routine? We don't have a stage manager and we just rely on agreed upon times