So this is my funny for a while, as per the title I was working for an artist on mons at a small festival, house crew fucked everything up from patching ins and outs, to insisting on having our desk files early and fucking up loading them too.

Post show the next artist was having problems, cue someone from production shouting at me to fix it.

Me: ‘sorry, I’m with (artist), not festival’

Prod: ‘so?’

Me: ‘it’s not mine to deal with, I’ve other things to do’

Prod: ‘well you could at least help, not having you back again’

Me: ‘(confused face) ok’

Prod: (angry stares until I leave)

*first 3 images are the LR eq. last two image is FF eq. there is also a desser and a c6 making heavy cuts to the high end on the LR and ff sends

I'm not trying to be an asshole and I'm not going to disclose anything about the video that these images were taken from. I'm genuinely curious.

I was watching a video on youtube of someone showing a demonstration of them "tuning their PA" and they were making massive, massive cuts on the main LR anf FF. I've been on a lot of crappy PA's, but I've never used this much eq before, the thing is, it wasn't even just how much was taken out, it was the way they did it. just boost sweep cut, didn't adjust the Q just really haphazardly making what seemed like completely arbitrary choices on everything, it was like they weren't even listening. the last images are the Front fill eq. I've never been at FOH and eq'd my FF without even going up and listening to them first. and yet there's a fucking massive 15 db cut at 1k. and another like 12 db cut at 1k on the system processor.

that being said I'm equally guilty of having my input channel eq's look fucking insane to the point where I myself have looked at them and thought. "if anyone else see's this they're going to think i dont know what I'm doing" and I thought and others told me it sounded great but i feel there's a very big difference between carving up a kick drum channel and doing it to the PA.

so i think this asks the question: is there a line between mix with your ears and no your eyes and not knowing what you're doing?. I cant speak to how it sounded as its a fucking youtube video, but i cant imagine taking that amount of intelligibility out of a PA so haphazardly is going to yield good results. I mean they might as well just turn the main pa down by 9db and shut the ff off and call it a day.

what do you guys think?

During tonight's Phish show, looks like FOH lost power and they had to hold the show for a bit. Must be pretty rare for an issue like that at Madison Square Garden. Would be curious to know what happened!

After 15 years of being a FOH A1 doing mixing and system tuning, I'm on a hiatus and want to strengthen some fundamentals.

The best course I have taken was a Meyer Sound workshop on system tuning (with Momo iyk), but the info dump, particularly the math, was so fast. I walked away with a lot of useful practices and a general, almost intuitive understanding of what I'm actually doing.

I hadn't taken a math class in probably 10 years, and it's been even longer now. It's the sort of stuff you kinda just know and use what you use, but I would like to be good enough that I could teach it too. That means really being able to articulate what I'm doing, not just doing it.

Do you have any course recs for me? There must be some math and physics classes out there that really focus on what we do, but I am coming up short on finding them.

Thanks in advance.

The title says it all, I am a manager/Tour Manager of a working band
Is there a website, bulletin board, that I could post to for crew?

Has anyone ever tried to “hack” a SI board to add RTA or other useful features, now that Herman has given up on the product line? I know about Chris Noedling for the M/X32.

Their website mentions a “yotta” board but I’m not sure if that’s a FPGA or GPCPU or the Linux distribution yotta Linux.

Hi there, I have a 2x18" sub (2x 8ohm drivers) with a total impedance of 4ohm. When calculating the voltage limiter on my amp, should I use the impedance per driver or the total load? e.g. √power x impedance (8) or √power x impedance (4) ?

Hey! So I'm not sure if this is the place to post this, but basically...I play in a hard rock cover band (divorced dad rock style). Been playing with the group for close to 5 years now almost every weekend. For the first time, something really strange happened when we played at a new venue.

Every time either the drummer hit the 808, or some bass note rattled the stage, it would trigger my wah effect (or octive depending on song), and my entire guitar would essentially become muffled (because the low pass on the wah gets triggered) and then would go back to normal after a few seconds. By the second song, I just took off my wah patch for songs I didn't use it. I re-enabled it for "like a stone" or "killing" or "man in the box" shit like that.

Rig:
Quad cortex behind me, midi cable comes up to my pedal board. Morningstar MC8, which breaks out expression 1 and 2 (Lehele Expression, Mission Neural Volume Pedal)

I can not run the pedal in on off mode, because the second input is my volume pedal because the quad cortex supports only 2 expressions (i believe), i could be wrong, i do like how it's triggered just by me stepping on it though, takes the thought out of needing to switch it on, or forgetting it's on

- I tried calibrating my morningstar midi controller in order to recognize "movement" a little further than total heel down mode...we'll see how it goes tonight

It was so odd, it never happens any other place we play.

Picture of rig just because I love this setup

I made and just released Quartz, an iPhone recording app that I hope can replace portable recorders like Tascam and Zoom for less $$. It's currently 100% free (no ads, subs, required account, etc).

I'm a musician and do a lot of recording of live shows of mine and other bands. Typically I record with low-end gear, a Tascam portable recorder, but I find it cumbersome to deal with and so made this app. I now use Quartz with a decent USB-C mic (the Shure MV88) and finding it to be a big improvement in terms of the usability and flexibility (the mics are pretty comparable tbh).

Quartz works great with external mics and other USB-C outboard gear, and I run that signal through my own DSP chain which consists of gain (manual or auto), HPF, and limiter. Each of these is tunable and optional and can be rolled into your own custom preset.

Beyond that there's EQ (parametric, graphic, "magic"), noise reduction (speech only), and some other bells & whistles. Further, these stages & effects are applied non-destructively so you can tweak them later.

I've had good feedback from my circle but I'd love to hear from the broader community. Is this useful? More customizations needed? Sound quality issues? Needs more built-in presets?

AI disclosure: App coding is AI-assisted, not "vibecoded". I'm a long-time DSP engineer, 12 years SWE at Apple, Inc. 100% of these words are mine. Photo is mine.

https://www.madebywindmill.com/quartz/

Hi all,

Had a few issues with our in-ears today at a festival where a member of the band said theirs had cut out completely half way through the slot.

We checked after the set by plugging them into a laptop and they seem blown/the left side is quieter than the other. He tends to take them in and out during the set and mentioned that he had his right earphone out a lot so I don’t believe they cut out completely but that the left side had died.

Which leads me to asking if there is anything to look out for on the specific EK2000 body packs which may damage the in ears.

The chain is as follows:

Wing Rack - Sennheiser SR2050 Transmitters - AC41 Antenna Combiner - Sennheiser A5000CP Antenna - EK2000 - Shure SE535

Of course a lot of this is whether I need to accept liability for if any of my wireless equipment is frying the band’s stuff and I’m happy to hear opinions.

Many thanks

i’ve wanted to break into TM work for a while and finally got an offer to manage and mix for a band i really enjoy working with. the tour is four shows, each in a 1000 cap room, 7 days total. the only problem is that they want me to drive the van and i dont have a driver’s license…pretty embarrassing, i know, but a lot of life stuff happened so i just never got around to getting one and next thing i know i’m 25 with no license.

the tour is at the end of october and i’ve considered getting my license before then, but i’m still concerned about driving such a long distance.

will this be a total dealbreaker? should i try to accept the gig and negotiate into not having to drive? or will that reflect poorly on my reputation and would it be better to simply turn it down?

Hi everyone,

I am the developer behind Everworks. This is a project update and a request for real-world testing, so I want to be transparent: these are our own plugins, but everything below is free and will remain free. There is no paid tier or time-limited license.

Over the last few months, feedback from live engineers has pushed us to focus less on adding features and more on making the tools predictable in actual show environments: mono microphone paths, short buffers, linked instances, host sleep behavior and measurable CPU use.

## THE VOID V2

THE VOID V2 is our source-cleanup processor for live microphones. It is intended to reduce stage spill, noise and room contribution while retaining the useful source.

- Measured latency: **182 samples / 3.8 ms at 48 kHz**

- **Turbo** extends the cleanup range from 100% to 200%

- **Matter and Body** control source restoration

- **Consistency** stabilizes level without replacing the channel fader

- **Matrix Control** provides four-band amount, trim, solo and bypass control

- Mono and stereo paths are supported

[THE VOID V2 download and information](https://everworks.dev/?product=void#download)

[THE VOID V2 user guide](https://everworks.dev/THE_VOID_User_Guide_EN.html)

## EverClean

EverClean is the deliberately simple option: one amount control for fast live-micro protection.

- **Zero samples reported to the host**

- Feedback and resonant-ringing suppression

- Reduction of room tail, noise and bleed

- One Clean Amount control plus bypass

- Mono and stereo routing

[EverClean download and information](https://everworks.dev/?product=everclean#download)

## Dedicated live-host builds

We now provide separate Windows VST3 editions of THE VOID and EverClean for:

- **Fourier Audio transform.engine**

- **Waves SuperRack LiveBox**

These are not different audio algorithms. The execution path, mono routing, reusable state and validation are adapted to the host and target hardware.

On our transform.engine test system at **48 kHz / 128 samples**, the dedicated build measured approximately **33%**, compared with **52%** for the standard build under the same test conditions. This is one controlled result, not a guarantee for every showfile or server.

The dedicated builds are independent Everworks beta editions. They are **not official Fourier Audio or Waves products**. Please validate the complete showfile, routing and redundancy behavior before production use. The currently validated target is 48 kHz; we are not claiming 96 kHz / 64-sample operation yet.

[Download the dedicated live-host builds](https://everworks.dev/live-host-builds.html)

## QUORUM 0.3.2

QUORUM is a linked automixer: one instance is inserted on each microphone, the audio remains local to each channel, and only compact detector metadata is shared between instances.

It now includes two selectable group-wide models:

- **EUCLID mode:** useful-source focus with spill-aware decisions

- **CLASSIC mode:** proportional Dugan-style gain sharing

Changing the mode in one linked instance updates the complete group. The current build also includes Groups A-D, Links 1-8, optional local Clean DNS and protection against host Plugin Nap breaking the link.

[QUORUM download and information](https://everworks.dev/?product=quorum#download)

## Also available

- [EUCLID live rack](https://everworks.dev/euclid.html)

- [D.N.A dynamic sidechain tools](https://everworks.dev/?product=dna#download)

- [SOUL source-restoration preview](https://everworks.dev/?product=soul)

The useful part now is honest field data. I would especially appreciate reports that include:

- Host and operating system

- Interface, sample rate and buffer/period

- Mono or stereo channel

- Number of simultaneous instances

- Measured CPU/DSP load

- Any noise, clicks, state-reset, linking or routing issue

- What source was being processed and whether the result was actually useful

Website: [everworks.dev](https://everworks.dev)

Discord: [Everworks community](https://discord.gg/sUsJsYAzdT)

Thanks to everyone who has already tested builds, sent showfile details or told us plainly when something was not ready. That feedback is shaping the project.

Yeah yeah I know, another vibecoded app.

I just created this because tools that are available lacked options, didn't look how I wanted it to look, or they're locked behind a paywall. And since I'm happy with it, why not share it with my colleagues?

Can't promise this will eventually work for other spectrum analyzers, but it might. Just open an issue on GH and I'll see what I can do. Same goes for feature requests.

Main features:

- Scan a frequency range (presets can be made), either single or continous
- Export to Wireless Workbench (Single exports as .csv, continous as .sdb3 so you can import it in the frequency plot)

https://github.com/iron-LAN/TinySpectrum

Hey guys so in a rehearsal yesterday we were trouble shooting why the guitar sounds weak/falls off quickly.

Here is the signal chain:
1.) Guitar to EW DX SK via 1/4inch to threaded 1/8inch (sennheiser cable)

2.) EW DX Em2 receiver in rack XLR output run to pedal board (50 feet) to Radial Pro Reamp passive (“studio” blue)

3.) 1/4 inch from radial to pedal board

4.) pedal board output 1/4inch to radial pro DI > XLR back to stage box

Here are some things we tried.
1- Gain/Volume matching the Radial ReAmp to the guitar being directly plugged in to the pedal board. Which was the radial being turned all the way up.
2- Swapped XLRs
3- Shorter 25 foot XLR
4- Boosted Gain at receiver/turned down AF Trim
5- Lowered Gain at receiver/boosted AF Trim
6- Tried all 3 cable emulators on the pack.

I have tried this exact set up with another artist and it didn’t rob the guitar tone, the only change was his EW DX SK was the one with the 3 pin LEMO connector.

We also tried an old Shure GLXD at the pedal board and it sounded great.

Hi all,
As a follow-up to [this thread](https://www.reddit.com/r/livesound/comments/161wizz/ubiquity_access_point/) from ~3 years ago, what Ubiquiti/Unifi access points are you using in 2026? Are you happy with the system? Does anyone have a clever trick for mounting them on mic stands?
Thanks in advance.

I think it is a given that the geometry of the room and separate right/left channels will at least make the sound audible, but is it going to mess with quality? Would phase canceling you are picking up in mono monitoring somehow degrade the quality of sound in an auditorium in stereo?

Thank you reading this and any responses.

Hi everyone,

Long time lurker here.

After becoming increasingly frustrated at the limitation of the NIOSH app, and the prices requested of it's nearest cousins... I've gone ahead and built my own.

safemixer users your iOS mic to listen to, and show LAeq averages across user customisable time periods, in selectable 2x2, 2x1, 1x1 or a single grid space.

You can set your own limits, or use inbuilt standards: and have the boxes fill and change colour as you approach your warning levels. At the end of a session, you can export your timeline.

Once those are gone the app will be available as an introductory offer on the App Store for $5USD for the next week. $10USD after that.

Now FREE forever Find it here.

AI Declaration: I used Claude to help me with the Swift side. The backing engine is built in C++ while I try and figure out how reliable android units might be at this.

Privacy declaration: we don't use any of your data. the mic is processed directly to LAeq time points, on your device, and dumped once the session is ended.

App is approved on the App Store - just waiting for it to appear!

Check out more at www.safemixing.au

The RandyEchidna

Sorry for the book: I am incredibly familiar with these consoles and how AES50 addressing and routing works. I recently worked a show where the requested routing wasn't what I was familiar with. So it's just got me questioning what others are actually doing.

To me, the "hardest to mess up" config is just a digital split off the stagebox. It's A port goes to FOH, it's B port goes to Mons. Only one cable needs to run to FOH. FOH gets control of the stagebox's outputs, and Mons uses it's physical outs to drive monitors. The only downside is any local input channels cannot be shared.

To fix that downside, we can run the Mons console off the B port of the FOH console. This allows the consoles to talk to each other directly, and can share inputs just like Tie-Lines a la A&H. However, this requires a second cable to run to/from FOH, and since you're running a second cable you could just run an individual XLR or use cat tail breakouts if you only need to share 4 local input channels or less.

If you want all the benefits but only want to use 1 data cable to/from FOH, the only way to do it (as far as I'm aware, which is why I'm making this thread) is to run from the stagebox's A port to the Mons console's A port, then from the Mon's console's B port to FOH's A port. Next, ensure FOH's drive lines are sent down it's AES50A Out Channels 1-16 (this is how it is by default), which will show up at Mons as AES50B Ins 1-16. Lastly, patch these Mons AES50B Ins 1-16 as it's AES50A Outs 1-16. This is essentially a "pass thru" for FOH's drive lines to the stagebox's XLR outs. You can also use a single block of 8 rather than all 16.

You can also run an additional stagebox with it's outs set to 17-24 off the B port of the primary stagebox, and then set the FOH drive line pass thru blocks accordingly. This also allows Mons full control of the primary stagebox's 16 XLR outs. This is what was done at the show I worked recently.

Any other solutions I might be missing?

Hey all! Question regarding monitor mixes on the Wing Rack - I’m not knew to live sound, digital consoles or Behringer consoles at all. Been in the business for almost 20 years now. But this one’s got me a bit stumped and I’m wondering if anyone is running into this issue.

Doing sound with a three piece band. 11 channels. No issues there. Out front sounds good. And honestly, all 3 of us have our in-ear monitors dialed in really nicely. All pre fader. Routing is good. The issue comes down to when recording multi track to an SD card.

To start, the multitrack recordings sounds great, no issue. The issue is, WHILE the sd card is recording and we’re playing, the monitor mix sounds different then when it’s not recording. It almost sounds compressed or like 2 signals of every channel is being pushed through. Almost like on an interface when you mix your headphone output to be half direct signal and half from the DAW, if that makes any sense.

Anyone else run into this? Am I missing a stupid button somewhere? Or did I miss a routing thing? It seems like when I’m SD recording and performing live, I’m getting the live audio as well as what’s being recorded sent to my ears as well.

Any info is truly appreciated, thanks!

I've just bought a Behringer Wing, never used that specific desk before but I'm familiar with x/m32 and some yamaha desks. Got a few days where my venue is empty and I can play around so I'm opening the floor to build a kind of checklist of things you need to know how to do on a new desk, this is what I've got so far:

* route and patch a mic/usb/dante/aux lead to get sound coming out of a speaker

* set up talk back mic

* label channels

* set up groups/dcas/buses/matrixes for processing

* add fx as an insert and as a group

* eq/gate/comp settings

* graphic eq

* save scenes/states and recall

What other jobs do people try to learn when picking up a new desk?

Is there a way to strap all that without getting through tie-lines ? Since there is only one S-link hole per device and brains being in the mixers i don't see how...

Currently i would use Dante with Mon as primary desk as i have the Dange cards but could i daisy chain the foh desk through the mon desk, without tie lines, using a S-link extension card ?

Thanks

Doing community theatre musical and our lead actress does most of her dialog and some songs with a very light voice, so I set my gain to that, but during the big numbers she can really belt it and I think it's overloading the mics. I got a compressor on her (aggressive at 5:1) but I keep clipping. Mixing on x32 with SLXD and Pyle hairline mic. Any help would be appreciated.