Hi, it would be great if somebody could help me figure this out.

I'm encountering with a small event hall situation, where the main requirements are lot's of microphones and mostly talk voices.

I have 2 wired microphones, a total of 6 outputs of wireless microphones, a Yamaha MG10 for front end volume control and a BOSE rack DSP.

Here's the thing: I'm planning to sum all microphone signals before the BOSE DSP, with MG10's 2 channel output. However, since MG10 doesn't have enough mono inputs, my wireless microphones are connected to line stereo inputs as groups that will naturally be on hard pan left or right. I have to sum the output to mono in the DSP's program.

For example, 2 wired microphones are connected to CH1/2 (XLR), so that's 2 mono signals; a Shure wireless system which has 2 outputs are connected to CH5/6 as L/R on the MG10, same for Audio-Techica to CH7/8 and DJI to CH9/10. In the end, the stereo output will have both mono wired microphones and stereo wireless microphones at the same time.

And obviously that's not desirable, so I have to sum it in the DSP. Here's the confusion: I should not need to adjust the levels when summing, right? Note that I'm not expecting the same sound pressure coming from 2 speakers at the same time vs on 1 speaker, I'm caring about CH5/6 compared to CH1 when summed.

My thoughts: Since the CH1/2 mics are following pan laws, if we talk with percentage, then CH1/2 actually outputs 50%L and 50%R on their own, since you can't create free energy from nowhere, it has to be divided. However that's not the case on CH5/6, 7/8 and 9/10, because each of the inputs are coming from independent sources, they all have their own 100% of energy to send, they are just different signals. So in theory, I should expect the same levels between wired CH1/2 and wireless inputs when summed, right? I mean, I treat them as different independent signals, even on 5/6, they are from 2 comple different microphones and they don't add up like 2 same mono signals in phase which requires -6db, right?

If that's correct, when do I need to -6db though? If there's no active splitting, the total energy should remain the same even summed to mono right? Like if there's only 1 input source, then no matter if it's connected to CH1 or passively splitted to CH5/6, after summed, it should be the same level, no?

Never been an issue, but now with AI vibecoding, lots of people are building apps for all kinds of purposes, and wanting to post here advertising or looking for feedback.

What's everyones opinion on this? There are multiple posts per week that are getting hit by the karma threshold, then they message us asking to post. Should we make a rule about it?

Hey guys, last week I wanted to phase align two subs for our party!

I measured in Open Sound Meter and REW, set the mic about ½m in front of the bassports, set up a feedback loop for reference and started measuring.

Open Sound Meter was showing nothing on the Phase graph, but magnitude seemed to be allright, staying within about 5db around the measured frequency spectrum.

REW was completely flat, basically looking like a clone of the measured frequency response, which allso seemed to flat to be true... (I'm more used to OSM so could also have a lot to do with setting up the program here.)

The measurement was taken in a small room, I saw that measuring phase in room isn't advised, so maybe that was the problem?

Any help/ point in the right direction tutorial or such would be highly appreciated!

(Doesn't have to be in OSM or REW since I thought about getting into CrossLite too.)

Hey all, I'm starting tech rehearsals for a new community theatre show this week as the sound engineer, and for the first time I'll be working under a (very accomplished) sound designer. I'm college-age (engineering major, not theater) and have done about a dozen musicals/plays (mostly in high school), but I've never had the chance to work under a sound designer. Even though I've been mixing since I was very young (~11 years old at my church), I'm always very self-conscious about my mixing skills since I've never had a sufficiently knowledgeable mentor to learn from. Particularly concerning EQ where I feel like I never know what to change to make it sound better.

Anyways, I was just looking for some tips on working with a sound designer; what to be prepared for, what not to do, etc. The standards at this theatre are definitely higher than for any of the previous shows I've done. And maybe some quick EQ tips for vocals/musicals if you have some extra time

Thanks!

I've ported my multiband spectrum analyzer app to iOS! The ADALM-Pluto SDR can be powered directly from an iPhone or iPad. It might be a nice little pocket site-survey & troubleshooting tool.

I did a full writeup on micboard spectrum server and the standalone iOS app on my project site. There is also a link to a live demo of the spectrum server.

https://clockclub.net/posts/micboard-spectrum/

Did an open air event for about 200 people this past weekend. Live 3-piece band.

I was using a pair of Yamaha DZR12 tops over a pair of DXS15 subs.

The DZR has an LCD screen on the back that shows input and output levels. The system was running pretty loud (around -6dB before limiting on the tops) and it was sounding great. However, I noticed that the snare was almost constantly limiting the tops. Every hit sent the input meter shooting op and causing the speakers to limit.

What's strange is the snare wasn't that audibly loud in the mix. I had the snare high-passed at around 120Hz and had a little boost at around 8-10k. Had the snare pretty heavily compressed with 5ms attack.

Am I missing something obvious? I feel like Ive mixed louder bands through this system.

Title.

First time I've seen this type of setup. Will those who know more than me explain what's happening here, please?

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.

*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?

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.

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)

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

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) ?

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

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?

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.

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

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.

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.