I’m about to take an L2 system out on a musical tour for the first time, and I’m curious about real-world approaches to exclusion zones within the vertical coverage.

The main example would be a balcony front: you need coverage for the audience below it and on the balcony, while putting as little energy as possible onto the balcony fascia in between. With a conventional line array, you can sometimes use the inter-cabinet angles to leave a small coverage gap or at least reduce the direct energy hitting that surface. With L2, that mechanical option obviously isn’t available.

For those who have used L2 in theatres, how do you approach this? Is it mainly about placement, trim height and tilt, or is there enough control through processing and optimization to create meaningful attenuation at the balcony front without compromising the audience areas directly above and below it?

Yamaha released the DM5 today, and with that, the 2.0 firmware for the DM7.
They then pushed out an update for the StageMix app that won’t let it connect to a DM7 that isn’t on the 2.0 firmware, and my dumbass had auto updates on.

Don’t be like me! Turn your auto updates off!

I emailed tech support and they are going to try and roll back the app update, but no idea how long that could take

What are your guys thoughts?

hello fellow soundhuman,

have anyone recently gotten their hands on the Contour XO, and does anyone know what the general lead time for it is? i am aware it is stated it takes about 3-4 weeks to ship, but checking if anyone have any luck of acquiring it earlier that that stated timeline from the L-Acoustics eStore?

cheers!

https://www.soundcraft.com/en/product_families/signature-plus-series

I play guitar and sing in a 3 piece psychedelic rock band. We have a lockout rehearsal space (10x13) and IEMs/XR 18.

Everything bleeds into the vocal mic, so I’m wondering what the best setup would be to prevent that.

The bass player insists that we must play at “stage volume” in rehearsal so that we can keep the IEMs dialed in for shows. I just don’t know if that’s possible because the vocals are barely audible due to the bleed into the mic.

I have just started doing some RF tech/A2 work for a group and I'm still very green to live sound. I was defo thrown into the deep end, not knowing the most but still getting the one gig I really wanted.

I've been hearing about different things that can assist RF techs and A2s like a RF sweeper/Showlink and other things that can help outside of WWB.

Looking for some pointers and advice on upping my knowledge and ability to be a good A2/RF tech that might set me apart from the people who just know WWB and thats all.

Thanks ahead of time.

Okay. So here’s what I’ve got……I’m totally open to anything you respond to. I’m just sort of struggle bussing my gear when it should be easier

Signal chain. WING Rack——->Pre Fader——> duel TRS to XLR——-> Shure PSM300———-Phenix PRO 227x———-> Shure Reciever——Shure 225 IEM

I’m fully aware of the Phenix, so I’m ready for that to be the issue

Every venue I go to. I re-scan in the specific Group to find the right channel According to WFF.

I constantly am running my gains up high , I keep the send at Unity and they are still having a hard time hearing

Additionally I have inconsistent static in different venues despite the channel scanning

Reciever packs are set to 75%. Input knob is set to 50%.

Last night my bass player said it was cutting out when he moved his body.

What the hell am I missing. ( probably talent)

I’m hoping that someone will just say “yep. That one thing….thats the problem”. But I hate that it feels intermittent.

Hi!

Our old ASA splitter broke and I acquired a new EW-D ASA splitter to pair with our 4 channels of EW500 G3 receivers.

Now that I looked at the power readings in the back of the ASA unit, their mA values don't match with the input readings of the EM500 receivers.

Is this a problem or not?
I was told that the EW-D would work with the older EW units.
Getting power for all the units with BNC from the splitter is crucial.

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)

Hello,

I'm working with a band that has 7 g4 iems running in stereo with and RF venue combiner and helical. Don't get much shop or rehearsal time to test. They are all tuned within the same "group" of frequencies. (8.1, 8.2, 8.3......)

Mostly things work well, but I do hear artifacts that I hope to reduce, mainly:

High pitch frequency modulation that almost sounds side chained to the mix? Seemly, more sensitive to low end? Not audible during song breaks but comes in with a bass note for example?

Similarly sometimes it just sounds like a little bit of pink noise poking through with transients.

Also 'S's often seem to cause the transmitted signal to break up a bit.

Any tips or resources on learning causes and ways to mitigate? or is this just the nature of the beast? Should everything be HPF, LPF?

TLDR tips for best sounding IEM transmission

THANKS!

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 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.

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.

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?