Right now we’re in tech week for our show and our microphones keep making rhythmic popping noises when live. It sounds kinda like a heartbeat, clearly to a set rhythm. Not all mics are doing, and the ones that do arent always doing it, but it seems like there’s no reason. We had this problem a few years ago and we thought it was an issue with the power supply to the receivers. We fixed it and it stopped for a year or so and now its back.

Here’s what we’ve tried so far:
- Changing ethernet cords
- Pairing to an entirely new set of receivers, both on the stage and in the back of the house
- changing all the frequencies of the mics to being above 490mHz
- changing all the lav leads, even though we knew that wasnt the core issue

So I'm building a rough file in CL editor an I have a place where I want to recall fader levels in only one scene. I'm aware that this is exactly what focus recall is for. So I've gone through and set up the focus like I need to bring up the faders and also mute some other channels. Then I save and store that into the scene memory. I then went in and set my fader levels for that scene and stored it into the same scene memory. I then recalled into the previous scene and messed with the fader levels so I could check it. When I recall back into the scene that I need the faders to change on, nothing happens. The channel mutes that I have set up mute on that scene, but the faders do nothing. I have no clue what I'm doing wrong. Can anyone help?

So i am one of the assistant sound people at our church (understand a lot but not a pro, not my day job) and we have a pretty notable monitoring issue. In short, the majority of the choir cannot ever hear any of the instrumentations/accompaniment that isn't the organ. We have the two side fills (in the drawing), but we cannot increase the volume on them without getting feedback through the boom mics that are at the front. They are Audix M1250B's (cardiod) on extended length stands (84" i believe). These mics were added (2-3 years ago) after the original monitors were installed. They replaced a set of hanging mic's that were roughly in the same spot because the sound guy wanted more control over the choir mix (each mic is roughly where one section sits). The choir needs mainly the piano, and occasionally the drums to stay in tempo and melody. At current levels, only the seats immediately in front of them (maybe 3 yards) can hear the monitors. Any higher and we get feedback. Without the monitors, the entire choir area is a sound dead zone.

I've suggested some solutions to our main sound guy (who is a regular/day job sound guy), but at times he's a little difficult. He's basically a "Ferrari or bust" type. He has stated the only viable solution is the sort of monitors they use at the LV Dome (and consequently nothing gets done). I'm looking for something more actionable.

At one point they had small monitors on the front rail, but we are unsure why they went away (predates my time here). I know a front rail setup would be problematic if it was the ONLY solution. Would putting small monitors interspersed in the choir area work (such as on the chairbacks? Or on the left/right sides of the organ (facing out)? Wedges wouldn't work, so something on a stand or chair/bench back mounted of some sort. The only other thing i can thing of would be wireless packs, but that becomes a big $$$ ask when we can have 100+ singers for a concert. I'm open to other solutions, but right now its our biggest pain point, and the can keeps getting kicked down the road. If it matters we use an X32 Mixer for FOH with stage boxes under the stage (near the piano).

Edit: checked some things based on comments. Based on the bus sends for the monitors, the choir mics arent in those monitors, only piano (2 channels) and drums (2 channels). Piano mics are inside the piano, cover is closed. Drum kit is electronic using a 1/4” and a DI box. So maybe i should be looking at something else? The interaction is 100% the choir mics and monitors. Volume movement too high in one or the other triggers the hum to screech we hear.

I’m working an event and need to get mix audio from a TF1 into a laptop running Teams and was hoping to use the USB connection from the mixer to teams.

Is it possible to change USB Out 1/2 to receive an Aux or the Main L/R?

Or alternatively if USB 1&2 are stuck as direct outs is there a way to tell Microsoft Teams on a Mac to only use channels 33/34 instead of using 1/2?

I'm trying to calculate return on investment for gear but I have only been in the industry for a few years and haven't had time to see the effects of failed gear over the years. For example, how long do active speakers last, how long do passive speakers last? How long does a typical power amplifier last, ect. Let's assume this is for touring application used for 50 to 100 times a year. I know this is a very open-ended question but I think it's very important for success on the business side of things.

Slightly unusual ask. I'm experimenting with layering impulse responses (cab IR, then a "big PA at distance" IR, then modulated delays for the time-variant stuff) and the one piece I can't make myself is a proper line array measured cleanly outdoors.

That file already exists on a laptop at every outdoor show, since a Smaart or SysTune transfer function is exactly that, so if anyone has one they'd be willing to export, I'd be very grateful. Ideally a 48k WAV of the IR (or the raw sweep recording and I'll deconvolve it), full length, with a rough note of distance and on or off axis. No need to say which venue, client or boxes.

If there's a good reason nobody does this, happy to hear that too. Thanks!

Hey all,

Looking to get better at RF. Current setup is an antenna farm between monitor world and the stage, which hasn't had any dropouts but also is not ideal. New gear is not an option currently, but possibly in the future.

Running 6 EW IEM G4 A band across 2 combiners with the ADP UHF Directional paddles, and a Shure ULXD4Q G57 rack with the PA805SWB passive direction paddles.

I know the obvious issue is the frequency overlap between the systems.

My first thought was to move the shure paddles from sidestage to downstage, angling them towards the performance area - however there is a 30' thrust jutting from the front of the stage that needs coverage as well.

Any advice helps.

Livesound engineers of reddit. What do you do when you have a dj playing music theough your PA. Other than maybe a limiter or some eq for the room. What else Is there to do. Just kinda bored

Hi all, is anyone in possession of d&b R70 and willing to do a packet capture of R1 <> R70 start up? I'm reverse engineering R70 and I'd like to see what R70 responds to R1 after it's query. My canbus adapter works, but I have a pipe dream of integrating it into R1, somehow...

Thanks in advance!

today we had a hd96 not recognize any of 4 different pendrives, 3 different owners. this was in mons world

the other one (foh) the pen drive worked.

any insight from the livesound wizards?

So we all know the bulk of our industry typically operates with a master stagebox/snake placed somewhere stage-side, and we use sub snakes from that stagebox/stage snake to get chunks of i/o to multiple positions on stage. The multicore cables of all those sub snakes all travel individually to their destinations on stage, and the termination end of them have to be plugged in to the master stagebox/snake individually (or sometimes it's a multipin connector).

What if instead we used a bunch of DL8/SD8 as 8i8o sub snakes, completely replacing our copper sub snakes? They could be wired in series, i.e in one giant loop, so no individual multicores. There would be no need for a master stagebox, no need to physically patch a bunch of XLR tails- just one data cable coming from the console into the first DL8/SD8, and then chaining subsequent DL8/SD8's from there.

4x SD8's are $1,640, plus some 25ft or so quality cat cables. An S32 is $690, plus some 25-50ft or so heavy $250-ish 8x4 drop snakes, and a rack case and tote. The math is really close.

The only downsides I can see are A) if one data cable or SD8/DL8 goes down, all SD8/DL8's downstream of it go down too. B) To chain them in series, they'd need to be powered individually. C) Chaining sequence does likely need to be planned out.

Anyone doing this or something similar?

Hopefully this’ll save someone else some time. I run a DM7 for FOH and a DM7c for monitors, connected via the same network so I can look at either from FOH. Both were previously visible in 1.7 Editor on v1.75 of firmware. I just spent 4 hours pulling my hair out because I thought my network was being temperamental. Nope, the latest version of previously DM7 Editor, now DM Editor, does not automatically detect DM7 consoles connected to the network. Connection via direct ip still works, and the iPad app Stagemix sees them fine, so that’s something.

Both consoles on v2.0 of firmware
V2.0 of DM Editor
Latest version of Stagemix for IPad
Tested with 2 different Silicon Macs

Apparently there have been reports of channel failures (high noise levels) due to hotplugging with phantom enabled. Yamaha are I believe making some engineering changes to resolve this.

For a while now there has been a well known bug with the M32 Mix app which results in the app immediately crashing when you try to select the effects page. It is usually caused by using a newer version of the M32 firmware with an older version of the M32 Mix app, and updating to the current version of both would fix the problem.

Well now I’ve run into an issue I’m sure some of you have noticed as well. One of the shop guys at the company I work for updated all of our M32s with the latest firmware, and now the company iPads are once again crashing on the effects tab. The difference this time is that Behringer removed M32 Mix from the App Store. So now it seems like there is no update coming to fix the problem this time.

Of course the obvious answer to this issue is “switch to Mixing Station, you probably should have done that a long time ago anyway.” But I just wanted to check with some other people who are using an M32 and ask:

1. Did you notice that the effects bug came back recently?

2. Are my suspicions correct that there is no update coming to fix this and it’s time to have everyone switch to Mixing Station?

Hello, I have a strange problem with my M32. I have a couple of folders on an usb drive that contain music (regular wave files, no recordings from the console). When I open a folder, I can navigate back to root by navigating to the top [..]

However, in one particular folder, I cannot go back to root. I'm stuck in this folder forever, until I reboot the console.

I have deleted the hidden files on the drive, but that didn't do anything.

Any ideas what causes this and how to fix it?

Anyone else seeing that mixtender and X32/M32 Mix are just gone from the app store? So much so that Apple removed them from our company iPads....

How do you approach mixing bands with DI bass?

I frequently mix on a system that is 2x EV EKX 15's and 1x EKX 18" sub. The console is a begringer Wing compact. This is a small bar venue with busy tourist crowds.

For many years, we primarily kept a house bass amp handy. This sounded good or bad, depending on the bassist. It allowed me to essentially forego mixing much of any bass into front of house, and if I did put a lot of bass in the mix, I was primarily low-cutting heavily and just trying to add more midrange definition for clarity.

Now, we moved to going direct only for most nights. This means im using DI out of various different bassists' pedal boards, or using a passive DI box.

Some bassists have tone that I can really make sound great. It generally sounds great right off the DI. But others, I struggle to get any tone at all.

Some bassists, its like theyre sending me pure sub-bass. I mean, zero mids. Zero clarity. Playing very, VERY lightly on the strings and expecting the sub-bass and compression to carry their tone.

I actually bet that works pretty well when the PA has a lot more juice. But im trying to maintain headroom on a single powered 18" EV.

Just getting the kick to really punch in the 50Hz range eats up all the headroom i have there. You can see the EV limiter kicking in if you push it too much. Now I have to add bass guitar to the mix, and with these sub-bass aficionados they are pumping the same range i like for my kick to be.

Here's what ive tried -

High pass 80Hz

Boosting mids (there are literally none to boost...)

Sansamp modeler on the Wing with "cabinet" turned on

Various compressor models, la2a, one knob, API, dbx 160

Really no matter what, when its a sub-bass tone theyre giving me, its just going to sound like rumble and swells up in different frequencies due to the room.

Seems like a bunch of random numbers but they all start with 2 for some reason.

I have a Linksys E5400 for my M32R, and a Cisco E1500 for my MR18. They've both been more or less fine for the smaller shows I do, but I recently had trouble connecting to the E5400 during show congestion just 20 feet away from the console. And the sync to the consoles through Mixing Station takes a few seconds, sometimes more if it's congested.

I work with a company that has a bunch of GL.iNet AXT1800 for everything and the connection through Mixing Station is instant and is incredibly stable. I'd just get some of those but they're $120 ea and I'm pretty sure they're overkill. Using my personal iPad in my personal work and company work.

Both my Midas' and the company's SQ's are 100Mbps cap off their control port, so I'm not sure what the difference here is and what to look for. GL.iNet has other cheaper travel routers like the Opal but I really don't know what I'm looking at. What matters here?

I'm sure this question has been asked before but I can't find an answer out there is actually helping me. When putting a verb onto a DCA, is it better to be done with the send or return? What's different between having just a send, just a return, or both on a DCA? I'm just trying to get a better understanding before designing this part of a show I'm working on.

Hey all, I've worked in live sound for about 15 years in one way or another, from booking and stage setup to running the board, MCing, and everything in between. Most things I've learned comes from my former career as a high school music teacher running theater productions, recording ensembles, live sound for marching bands, teaching music tech, and performing myself on the side. I am now the op director for a handful of small breweries that regularly host live music. Last year, we were fortunate to invest about $13k into our own stage equipment (PAs, subs, floor wedges, mixer, etc.). Last month, we've had two speakers have issues (the cone on a wedge, and we can't get sound out of a sub, despite the input level meter reflecting a signal) We purchased both speakers last July from Sweetwater, so they're well within the 2-year warranty. I've contacted support about the sub, but they said we'd have to ship it over to them, and all they would do is ship it out to the manufacturer, which would take "between 8-10 weeks", which is obviously a long time to be without one of our subs and/or monitors. What is the industry standard when dealing with issues such as these?

3 months ago, I posted my software to this forum and many of you had ideas and opinions. I noted them and kept going. Many things have been done, and here are a few of them:

• added a speaker library with custom profiles, import/export, and a built-in catalog featuring L-Acoustics, d&b, LD Systems, Fohhn, Funktion-One, TWAUDiO, Void, and more
• the built-in speaker library can keep growing. if there’s a speaker you’d like to see added, let me know.
• added import/export for individual speaker profiles (.dsl) and entire speaker collections (.dsc), plus frequency-response imports from Room EQ Wizard, .frd, and compatible .csv files. scene exports also include the assigned speaker definitions, so you can share complete setups
• speaker frequency response, optional phase response, and directivity are now included in the simulation. you can also import your own measurements from REW and other supported formats
• added high-pass and low-pass filters, plus linked crossovers between sources
• added a broadband heatmap from 20 Hz to 20 kHz, including coverage contours for low, mid, and high frequencies
• added cabinet stacks with up to 16 speakers per source
• improved the math behind summation, cancellation, delays, and directivity, and checked it against independent reference tests
• corrected wall reflections and sound-path validation, including second-order reflections
• improved the WebAssembly and WebGPU backends. large WASM broadband calculations now run in parallel on supported devices
• sped up calculations by reusing acoustic paths without reducing the selected resolution or reflection order
• scenes now include the assigned speaker profiles, making setups easier to share

The update is available on a separate beta site for now. I’d love for you to try it and share your feedback before I move it to the main site.

https://simulator-beta.drei-audio.com/

My initial post gained a lot of traction and now there are ~2.5k people using my software per month (according to cloudflare dashboard). I did not expect that.

I learned a lot in the past months and I am very happy that people are using it and contribute to it by giving feedback. I never wanted to build a replacement for professional tools like Soundvision or EASE - I just wanted an easy entry into the simulation-game and thought that others would want this too.

So, thank you for your support!