r/livesound

Hey all,

We're Sabrina and Martin from Sennheiser. We're running an anonymous survey for live sound and RF professionals (FOH, monitors, system/RF techs, touring and rental staff) and checked with the mods before posting.

The goal is to better understand real‑world wireless audio workflows in live environments, including:

• typical RF conditions and spectrum pressure
• how many channels people actually run
• how engineers cope when RF gets tight
• what works well with current systems and where limits appear

It is NOT a sales or lead‑gen survey.

• Anonymous
• ~10 minutes
• Aimed only at people working with wireless audio in live contexts
• Open for 2 weeks (until June 4th)

Survey link:
Real-world wireless audio workflows

Happy to hear feedback in the comments, and thanks for taking the time.

We’ll share high-level insights with the community once the results are evaluated.

Sabrina & Martin

Hi all,

I wanted to share y'all a project I started over a year ago. I already talked about it in the comment of a post here months ago and now that I have more convicing results with my latest prototypes, I wanted to show it in its own post.

Soooo : I am making and designing my own digital mixing console from the ground up. Why? you may ask, well because I can. The goal of this project is to make a complete system in a distributed manner, in the modular fashion that we could have seen with studer/soundcraft, or Avid for instance (or Innovason for the elderly). Every component of the console is an independant system, IO Boards, DSP, Control surface, they all live on their own rackable board and they all communicate via a switched Ethernet network only. In addition to that, the software I am building is designed with horizontal scaling in mind, so technically you would be only limited by your network capabilities (even though that wouldn't be completely true, as always, reality is more complex).

Of course, I am still pretty far from my goals. However, I have promising results already. Hardware wise, I built an IO Board prototype with programmable preamps and 24 bits ADC @ 96kHz sampling rate. Still a lot of audible noise (My noise floor is around -40 dBFS), but good enough for a prototype, I am actively working to push it back to -60 and below, "fortunately" it is mostly due to a bad hardware design decision...

The \"RackedIO\" IO Board prototype

SM58 Mini audio demo

On the software side, I built a software defined DSP with a plugin interface, a basic ugly GUI control software, and an Ethernet (L2) protocol for audio transport and control streams between the sub-components of the system. There are a lot of features missing, like routing protocol (coming soon 👀) or show management and not everything is perfectly working as intended. But again, for prototyping and PoC, that's way more that I could've ever imagined when I started the project.

My basic control software

Old demo of the software using STEMs as input

My current goal is the get at least 8 clean physical inputs and 2 physical outputs with <= 10ms latency in-to-out processing included before making the console more real-world proof or exploring more complex architectures.

The software is open-source, if anyone wants to see what its like. However I must warn you that this is certainly not production ready, easy to setup nor to compile (And at some places in the codebase, it is not the most beautiful code I have written in my life). This is a pretty complex environment for which I haven't yet written any proper documentation (I will, it is on my task list, but I haven't had the time to). With that said, if you feel like you want to help or contribute in any way, feel free to reach out to me 😄. I'd be glad to discuss about it.

https://github.com/OpenSoundStack/OpenAudioLiveSystem

Here the two things I am also considering :

- Making a series of videos to document and show how the project progresses

- If I see enough interest in that project, I'll probably open a discord server for it

It’s insane. Works great. Instantly bought it after demoing it lol. I have a Livebox MADI and I can easily throw a few instances on there. I didn’t realize till I used it, that the plugin itself has zero latency. Thats absolutely bonkers to me, I have no idea how they do it. So easy to use. Just download it and throw it in your VST3 folder and it loads into SuperRack Performer when you scan for plugins.

EDIT: So I measured it with SMAART for anyone curious about latency. At 0% and 50% it was literally 0.00 ms of latency but at 100% it reported 0.02ms of latency.

I started with a fresh Digico session on a SD10. I got a roundtrip latency of 1.88ms at 48K. Normally, once you edit the session, it changes to 1.90ms of latency. (Side-note if you didn’t know, the SD stuff has different latency depending on whether it’s an insert, output, group, or aux.) SuperRack Soundgrid had a roundtrip latency of 1.02ms. The MADI Livebox had a roundtrip latency of 2.27ms latency. The plugin itself is 0ms. So technically if you were to include the console and Livebox together, it would be 4.15ms of latency but that’s subject to change based on how you’re sending the channel whether via group or aux. This number would of course be lower at 96K.

Seen some videos on Subwoofer shaping. have an event where two ks112's will be at sides of a 7foot DJ table with either a k8 or 10 above each. Any advantage to facing them any other direction than front? I'm at end of a long room (ratio 3:1 or so) with my back to window.

Hey all! Was hoping to get some help with an RF setup I'm running for my band. I think I'm in my head on this. I recently side-graded to Audio Technica ATW 3255's (5 channels) from the PSM900's. Main reason was wideband. I'm noticing the AT's are more prone to dropout where the PSM900's would hold connection. We're doing bar gigs and occasionally do some crowd interaction so we'll go out into the crowd. Definitely a challenging RF situation but I think its my setup not the ATs

Right now I'm running 4 channels (the 5th is spare we run on a whip if needed in a pinch) into an active phenyx pro combiner with gain compensation. a 25 foot amazon BNC cable to a Shure LPDA paddle antenna I typically run about 7 feet high pointing toward stage and crowd (so behind us stage left usually). The show last weekend we got a hard drop from our guitar player in crowd. This was a total blackout until he got back on stage. I was running the combiner at 0 compensation so I'm thinking the BNC cable quality and length might have been the culprit.

Just tested at home with the combiner gain set to +4 and it seems to get a noticeable more stable connection. Tested in basement and walk tested putting it out of sight and making have to punch through walls etc. (obviously a stress test as line of sight is always priority at shows).

The other members on stage btw had no dropouts the entire night.

Our PSM900's have been rock solid for the most part. Occasional drops here and there but we put them through some pretty interesting scenarios and they come through. I run those at 10mW on a PA411 combiner with the same paddle and cable.

Am I expecting too much from this wireless gear? The AT's have some great reviews, and again with clear line of sight they are REALLY great, but I'm thinking I can push them a bit more for our crowd interactions and just wasn't compensating for the cable length and quality.

Any help would be super appreciated!

I’ve been to many large-scale EDM shows in the last few years with L-Acoustics or D&B systems that are too top end-heavy.

Dance music doesn’t need to be as sizzly and bright especially when punters are in front of a PA for 6-8 hours. So many get fatigued and it’s unnecessary. Please help me make sense of this.

Hey everybody! I recently bought a used Behringer X32 from Facebook Marketplace and was unpleasantly surprised to see there aren't many videos showing how to fix these things when they break, so I recorded some tutorial videos as I was fixing this one.

The repairs are relatively easy to do, the only potentially challenging part is that some of them require soldering. If you have an X32 that has any issues, there are step-by-step instructions in the videos in this playlist. The repair tutorials there are for:

Fixing Slow or Dead X32 Faders
This is the most common one I see. This repair also requires soldering, unless you're lucky and it just needs a new rubber belt for the fader. This repair requires soldering.

Fixing a Broken X32 XLR Input Jack
Sometimes XLR cables get stuck and the input jack gets broken trying to get it out of there. This repair also requires soldering.

Fixing the Rotary Encoder Knobs Under the X32 Screen
On older X32's, these knobs get oil, dust, and gunk in the switches and the knobs don't respond very well when you try to use them. They're typically jumpy or they scroll back and forward when you're only scrolling one way. No soldering required for this repair.

Replacing an X32 Scribble Strip LCD
Sometimes the scribble strips go blank, have blurry text, or just die. This one is an easy replacement, no soldering required.

I hope these videos help some of you out there with X32's that need repairs. Don't listen to people who say they're trash or that they're not worth fixing. Anyone that tells you that is spending someone else's money on gear.

I am desperately trying to find the files for it for Linux, if anyone has them please please share. And to be more specific, I would rather use the native software than run the windows version through wine, which is what I saw someone say.

I managed to find the file

The only stupid questions are the ones left unasked.

Im a touring comedian who tries to always video my shows. I am trying to step up the audio on my videos by getting multiple sources like my camera mounted mic, recorder on the audience to pick up crowd noise, and I want to get the mic feed direct from the board.

Venues I work run the gamut from dive bar with a mixer board that I have to set up, to clubs and theaters with a sound person. I was figuring on the bars just plugging in one of my pocket audio recorders to an output. Curious if that’s the best way to go? Also for the clubs and theaters, would it be usually fine to just give them an sd card or thumb drive and request they record the audio? (And of course tipping them for the help).

Hey all,

So I’m currently running a musical at a theater I’ve done lots of shows with; here we have about 18 omni lavs involved. Mostly point source, which are the ones I’m most worried about. It feels like the frequency response is off on some of them, and that some of them are more prone to feedback than others. Is that common when they get handled too harshly? Or is it more common for them to just drop entirely without changing their response?

I would like to test their frequency response. I will probably just put a small speaker with white noise to them to see how different they are. What would be the best way to analyze the frequency spectrum?

Thanks everyone

How do yall handle stage lighting causing hum through the PA speakers? My band has been having a hard time with it recently and ive been trying to find a good way to keep our PA and lights isolated from each other on the circuit, but sometimes its not possible in smaller venues.

I'm covering for a long-term absence, so I've been thrown in, sink or swim.

I have a mic that gets signal on several scenes, but on one particular scene, I get no signal.

No signal light at all on channel 10. I've tried copying channel 9, which is a good mic signal, but still no luck.

On all other active scenes, channel 10 is a wireless mic. On this scene.... Channel 10 is not.

I have an instinct that this is a routing problem but I don't know how to approach it. I'm reading the manual now, but I'm in a near panic situation.

Wondering if I'm being an idiot...

I have a couple of Ethercon>XLR boxes that I connect with shielded CAT6E.

With traditional balanced XLR cables, everything works as expected.

I am trying to do something unconventional, and it may or may not be possible... Other posts indicate that analog audio should work with Ethercon, but that may be for balanced signals only.

For an IEM mix, I have two unbalanced AUX outputs from a mixer combined into a TRS (for stereo IEM). If I put this directly into my headphone amp, all is well, it sounds great.

But I take that TRS and connect it to the XLR of one of my Ethercon boxes (T=pos, R=neg, S=gnd). On my other Ethercon box, I take the XLR back to TRS and feed that to a headphone amp.

There is a ground issue. If I touch the metal case of the headphone amp there's a loud hum. If I touch the metal chassis of my mixer, the hum pretty much disappears. This makes me think the Ethercon setup is not properly passing ground from my mixer to my headphone amp.

Is what I am trying to do not possible, foolish, or dangerous (or all three?)

Is there a better approach to doing a wired IEM mix through this Ethercon snake setup?

(edit: I'm using CAT6E, not CAT5E)

Last weekend I worked in an event where the biggest challenge wasn’t sound quality, it was sound control. Venue had strict noise limits, multiple zones with different vibes and a crowd that didn’t want to feel restricted. Instead of pushing traditional PA setups to their limits, we tried something different: a wireless headphone party setup where everyone had headphones with multiple channels and each zone basically became its own mix without bleeding into the next.

What surprised me most wasn’t just the clean audio but how much easier it made coordination on the backend, no complaints from neighbors, no fighting with room acoustics and people could literally switch between DJs instantly. It felt like a weird mix of live sound and personal listening tech but it worked way better than expected.

Anyone here tried something like this for live events?

Looking for ideas, especially in a theatre context. New boss is trying to get us to adopt Streamdecks into our workflow. I'm not opposed, I just don't get it, and I'd like to find a way to be stoked about this instead of causing conflict. We already have and use Go-Boxes for triggering QLab playback, and I'm pretty happy with using User-Defined Keys for most console control functions. People seem pretty jazzed on these things, but all I've seen them used for so far is just... a more complicated way to do what I already do just fine with the aforementioned control solutions. If it matters, all house consoles are Yamaha but we frequently rent in other brands. Help me out, i know y'all probably have some solutions to problems I didn't even think about.

I updated my fader layout a month ago and have tried multiple things to get my fader layout changes on my mixer to carry over to my apps. I'm sure that the issue is on the mixer side because I tried a fresh mixing station install on a phone that had never had it installed, and the incorrect layout loaded. I'm thinking there's a setting or something I am missing on the mixer.

What I have tried:

Re-save mix file and reload

Reboot mixer

Delete cache/memory on both apps (multiple devices)

Remove and reinstall both apps completely (multiple devices)

What am I missing?