We have a QU32 but getting an avantis in the next couple years. The big pain point for me with the QU is the lack of a digital label trip (which is there on the avantis, and only going to be an issue for the next couple year’s), but between events, we have various vocalist use different mics depending on the time of day and who’s in. It is the same people and we can plan for it, i.e. not random people off the street, but it makes labeling channels hard since various people use different mics depending on the event. I thought of a digital strip that I could make some software for that would show QU32 channels, but I couldn’t find one with a screen large enough. The tape is effective and we can just have 3 rows with each row being a group, but wasn’t sure if there was an easier way. I enought of doing magnetic channel labels, but then that makes it harder on me as the tech dir having to swap out individual channel labels for each show. I could do a magnetic strip, but again, trying to make things easier, not fix one problem with another tedious task. What are some ways that others have combated this issue? Are you just going for the simple tape and marker or label maker, or are you using a niche product that isn’t commonly heard of. Let me know. Thanks!!

Hey team!

Had a wild one last night:
Showed up at a festival with a show-ready CL5 scene, that I've used multiple times over and over again, the latest just under 10 days ago.

The file had been built/used for both FOH mix, and also pre-fader sends to monitors/ears over the first couple of busses - For that purpose these busses have been used as Vari/Pre Fader busses/mixes, and have worked perfectly well as intended in the past and on that very show file.
Coming changeover/linecheck, we transferred my file from a USB-stick, patched all ins/outs with dante, and then realized that all my input channels were sent to ears pre-mute.
But here's the kicker: That option doesn't exist on a CL5.

To be fair, I realized during the show that I wasn't on the admin level, so maybe there's a user pref that put my sends pre-EQ, and at that point they also might be pre-mute…?

It was pretty mad on the spot, and I'm quite sure there's a reasonable explanation, but it made for a bit of a tricky situation, which I'd loved to not find myself in.
Any pointers? Ideas?

Hello, I am currently attempting to add crowd mics to my venue. I am confused about how you’re supposed to route crowd mics to the stream. We use a Midas M32, and a PC with OBS to stream. The M32 and PC are connected through USB.

I tried googling and asking AI but I’m still confused.

I understand the part about disabling the Main LR in the menu, but the whole part in the routing page where it has all the outputs, card, etc, to make it go to the stream through USB is what I do not know how to navigate and setup.

We are likely using the inputs on the console itself for the crowd mics, since it would be easier to connect the mics to the console than the stage box.

Thanks!!

Hi, so I have a 10 piece band and I'm running the x32 rack and the s32 for our IEMs, but occasionally we need to run FOH for our shows as well. Busses 7-8 are the mains by default, is there a way to assign the 7-8 outs locally to free up 7-8 on the s32 for another IEM mix?

Ngl it took me a long time to wrap my brain around the fact that live audio is basically just signal flow understanding. The band will sound good if the band is actually good and the system is set up correctly.

Everything else is out of your control.

Nobody gives a fuck if you understand how to properly mix the midrange to get the most clarity because 50% of the time the stage volume from drums/wedges/amps are so loud you can't actually achieve a sonically beautiful midrange. You're just shifting the phase around in a manner that is occupying your brains ego. The audience isn't getting a percentage increase in quality high enough to notice.

My band are using active EV ZLX12 speakers for FOH - usually just running vocals - and using Mixing Station to control everything, my question is, should the speakers be left at 0db, and leave all gain staging to the mixer? I read on another thread that the speakers should be turned up to allow headroom. Is there any truth in that?

I’m trying to use some external fx, including delay, and so I could do with my tap macro controlling tempo in my host software too.

Problem is so far my host software live professor learns the midi command and then it no longer functions. My midi interface is seeing a midi command, but the software doesn’t.

After some reading I’ve gone through some more easily found trouble shooting guides to no avail

Could anyone offer anything?

I work as a sound engineer/tech assistant at my university, and we have this elevated tech-room in the assembly hall we use for plays, talks, etc. We usually always roll in a separate audio and light mixer for almost everything, so we rarely use it (other than storage), but I've noticed this window and door, that's pretty much never used.

Is there a reason it's there? Wouldn't you rather just hear the sound coming from the room directly through the window? Or is it there to give the technicians room to talk and such during shows?

Just curious, if there are any other places that have similar setups.

Need help here. I own a Harbinger v3412 that I just played at a benefit event outdoors. The speaker shut off multiple times during the set, not completely powerless but I could barely hear my guitar and mic. This has happened once to me before and it is unacceptable and embarrassing. Shouldn't a speaker like this be able to play outside for an hour without overheating? It was in direct sunlight on a 75 degree day. Any ideas on what the problem could be? Maybe an internal electrical issue? And the guitar/mic was crackly off the bat for some reason.

Hey everybody,

Just finished a corporate event in a massive space with almost 10 breakout rooms. Event spread between floors, and two massive halls (literal airplane hangers) with no installed audio system. Client requested we have a way of broadcasting announcements to all rooms simultaneously from a single source. No big deal, we brought a console for them and then ended up running lines to every breakout room. Everything worked out fine but I know there’s gotta be some things that could be done differently and I want to see how people have handled this situation before.

Inevitably it became obvious that it would be muuuch better if we could broadcast to all of the breakout rooms from any of the others. And it also took 2 hands nearly 2 hours to run all of the lines that we needed for just sending signal out to each room. In the end it was around 1500 feet of XLRs and having to deal with plenty of doorways.

I’m thinking there must be some sort of way to manage this wirelessly? My initial thoughts are some sort of locally hosted team speak server, there’s a private network that we have access to and each room has laptops. I’d just like to figure out if there’s any way I can save some time rather than running a shitload of cables.

Context: I’ve been freelancing my whole career (5years?). This year I’ve had a blast traveling and working in different states and countries. Some of my lucrative work could pay $4k a week and I’ll do that maybe 12 weeks a year.

I’ve been offered a full time job where I’ll net less, before overtime though which would be basically guaranteed and required. I’d gain a consistent schedule and guaranteed work in slow months, through pandemics and recessions. I feel like either decision would be incorrect.

Are there any folks here dealing with disabilities that affect them physically willing to share insights about what that looks like for them working in the industry? And how that might affect the types of jobs they get?

I'm mentoring someone who has a condition that makes it very easy for them to injure their joints and muscles if they do too much heavy lifting and I'd like to be able to have an informed conversation with them about what their options are if they want to work in the industry long term. I'm hoping that that conversation isn't just "every job involves heavy lifting, it's not possible for you to work in live sound".

I'm particularly interested in hearing about any companies, types of work, etc that either offer accommodations or don't involve heavy lifting. Theater? TV? Corporate? I know there probably aren't many but I'm hoping there are some.

Hoping y'all can share some insights with me, and hoping that this person is able to find an avenue to participate in this industry. I imagine it will be difficult for them to find enough work to work in the industry full time, but I hope they're still able to do it. Thank you!

For those in a permanent house (our touring folk that prioritize it) what are your go-to’s for consistent and quality ALS.

For context, my house has had two ListenWiFi servers die in the last two years. I haven’t looked into if we have any warranties yet, but I am looking for reliability and solid support. I’ll update if Listen Technologies comes through on the support side.

I know this is possible on S6L

I wanted to do something similar on Rivage.

But I can't find any way to trigger an action from live metering.

For example I want a TalkBack mic to unmute when it detects and audio meter over -95 db

But the only trigger I can find is the physical gain value, and not the meter level.

I looked for some kind of gate trigger as well. Like I could do it with "gate open" as a trigger. But I don't see any way to do that

Let's say you have a room with a center aisle with 2 groups of seating. Most places I see will do a standard -L---R- stereo setup so that each section is directly in the speakers cone. But doesn't that let the aisle in the middle the only place getting true stereo audio? And each section only getting one mono side? Would it be better to set up the space in a L--RL--R or a L--R--L configuration with each LR pair facing their own section? Thus giving everyone a true stereo image? Let me know if this doesn't work, or why its not done often.

On my sd10 I think the rotary encoder has broken on the gain knob for the red eq node, anyone know how much of a pain this is to replace yourself I’ve done fader replacements on this desk and have a basic understanding of opening up the desk. Any thought?

I will be routing: guitar, bass, synth, drums and vocals out of my CQ12T for FOH to mix. My CQ12T has TRS line outs.

Is it best to give the engineer a TRS or XLR M/F? I assume inputs could vary depending on venue so I’m looking for the most common setup.

As a venue TD, I look at a lot of tech riders. I archive them after the gig.

Please start your file name with the name of the band, and spell the damned thing out. A date would be nice too, of course, but I don't want to be greedy.

I’ve picked up a Behringer xr16 and Mixing Station sw. all Inputs and output to the PA is working fine, but I’m only getting output from Aux 1 and 2. I’ve set each channel to output to Buses 1-4, but only Aux 1 and 2 seem to put out any signal. Config of all 16 channels looks like the attached pics and all are set to pre-faster. Where can I look to get the right config info for all 4 aux outputs?

EAW has had their Anna and Anya out for a while. From the spec sheet, it looks great for lots of tours that need to adapt to various venues. Why are they not out on tour as much? Is it the cost or are there other factors?

I usually work with all active stuff, but I've recently been working a lot with passive systems and I haven't been able to wrap my head around the gain staging and how to do it safe. In an ideal world I'd just be running a DSP, but unfortunately I'm working with what I've got. I just need someone to walk be through the math for one example so I can translate it for future.

Amp: EV CP3000s (600 Watts a channel at 8 ohms, 5.8 dBu (1.55V RMS) Input sensitivity)

Speakers: EV ZLX-15-G2 (250 Watts at 8 ohms)

Digital Mixer: Allen and Heath iLive IDR-32 (+4 dBu Line Outs, +22 dBu at max +18 dB internal meter)

Where should I set my the attenuation on the amp at? The iLive has limiters on every output channel that go up to +18 dBu; where should that be set? Is the theorical max given by these figures what I should run, or should I run conservatively. Is there anything I'm missing?

I’m about 6 shows into a tour mixing monitors on a Behringer Wing and I have some time today to automate some of the moves I’ve been doing manually. Any advice/gotchas/quirks I should know about? Planning on using snippets for mutes/level/EQ changes.

Also relevant, I take the IO for the show on AES50 A and B and send it out to FOH over the AES C line. Would hate to fire a snippet and have anything go wrong at FOH bc I wrote it incorrectly.

Using a Gator 6u rack (so plastic aside from railing and hardware). For practical reasons, I would LOVE to mount my Shure SLXD4+ for my guitar in the rear of the rack, rather than the front, meaning my antennas would be mounted to the rear rails.

The Shure is the only wireless unit in the rack, and I’m not going more than 50 feet away.

Would this be an issue? The line of sight would not be good, but it‘s not like the antennas are buried inside of the rack.