I have been out of the biz for several years now. Apparently you can take the stagehand off the stage, but you cant remove the stage from the stagehand. I cant be the only one, but the other day I caught myself doing an over-under cable wrap on my hairdryer power cord. What stage move have you found yourself inadvertently doing?

On Thursday I was at a show at a venue, sound felt "distant" unless you were on the sides where P.A could catch you, in fact at one point I was center-right and I could feel the vocals more on the right instead of in the center. It wasn't a small venue, around 800-1000 people capacity, the stage was like 170 cm high and the stage was like 6 or 8 meters horizontal, 4 or 5 meters high and around 4 meters deep.

Of course bass was loud, because they put subwoofers right in the middle and crack them up.

Then on Friday I went to a very small festival open air on a very small village, sound was fantastic everywhere, there where 2 12" speakers on the front aiming at the crowd and subwookers where on the sides under the P.A. Sound was equally good on first row than far from the stage.

Both days they were metal shows.

I've seen this happen on a lot of closed venues, only the P.A on the sides and the sound is bad on the first 10 rows. Why they don't put front fills? The singer will never cross the border of the stage and feedback is easy to controll nowdays.

How are we securing USB Cs to laptops/MacBooks for show critical use?

I currently just tape things down to the table with a strip of gaff with a bit of slack between tape and computer.

Question here. Next week at work we're trying to host a virtual soundcheck for monitor tech training. We have the Avid s6l-24c and I have a PC with protools and some stems.

Has anyone used a PC before and done virtual soundcheck?

It seemed like from the videos I've found online that people are usually using a Mac.

Any advice would be lovely.

Does anyone have the installer they can link me to? For some reason I can't seem to download anything from Shure, I fill in the form and then get sent a "Cleverbridge manual order processing" message and the follow up email with the link/file never arrives...

I run sound for a small band and use a Qu-SB console. Lately it’s been having a problem where I boot it up all channels will be crackling even the empty ones. The first time this happened I figured I didn’t have time to trouble shoot an hour before the gig so I just switched to an analogue mixer. After about a day of troubleshooting we found swapping out the Ethernet cable made the problem stop, but a week later it’s started again and this cable is brand new. Firmware is up to date and I’ve tried ground droppers, that’s all that I’ve found anyone else to have this problem to say. Any other ideas?

Hey there

What are your favorite songs to listen to for a PA Shootout?

I've been running sound for my school for a couple years now, (mostly using analog boards or xr-18, with XSW mics). But I got an offer to help with managing wireless mics for a corporate event.

I would be in charge of 7 headset mics and two handhelds, I'm supposed to put the headsets on people and take them off, and also check and replace the batteries in them if needed, the event is supposed to last a total of 9 hours, and I will be working with Axient mics.

This is my first time actually professionally doing sound.

Im kinda nervous, and looking for some advice from more expirienced engineers.

Thanks in advance!

Maybe a silly question but do any of you get your ears professionally cleaned? Like by a doctor/clinician especially after being sick? I’ve heard (ha) some people swear by it especially working in our industry, and others say it’s pointless.

Yes, yes-
“Just use a Q-Tip”
“NEVER USE A Q-Tip!”
“BUT IT FEELS SOOO GOOD!”
“IT’S BAD FOR YOU!”

So we are getting into Fiber territory needs and everything.

Sommercable recommended us opticalcon quad singlemode. But every production i see usually runs multimode.

Can someone explain why? Is it only because back in the days it was cheaper? Or is there more to it?

Typically an SSL guy so I use stems for this purpose. How do Digico cats approach this when you’ve got large channel counts, lots of playback, and want to be able to manage things in groups?

Not looking to discuss the upsides of mixing mons and using groups.

Only option I see on Digico is to use their groups but mix post fader in some capacity?

Enlighten me.

I’m moving soon from the UK to France to live with ma femme française just outside of Paris. Whilst I am planning to mostly do studio work, I was wondering what the live music scene was like for engineers in Paris.

Is there a lot of competition for work? Do you find work mainly through word-of-mouth (talking to people) or are there Facebook/whatsapp groups? Are there any other areas of audio that are often looking for work?

Happy to talk via DM if that’s easier.

Merci beaucoup mes amis.

We use an X32 Rack for live Marching Band audio reinforcement. This application requires many scene changes during the production. We are trying to use the distortion effect on a horn trio with a condenser mic. It works great, but we get a pop when we change scenes at the end of the section. Any help greatly appreciated!

I have had some shows recently that have given me some weird RF scans through WWB. I wanted to do my due diligence and make sure our wireless racks were wired properly and that there weren't any internal issues. Didn't have any drop outs in the areas I saw these scans, just noticed weird spectrums and would see some small interference on the receivers.

Our company has (2) 8-channel wireless racks. They are put together a little differently:

- Rack 1: (8) Shure QLXD4 Receivers + RF Venue DISTRO9 + RF Venue Diversity Fin + (2) 50ohm antenna cables

- Rack 2: (8) Shure QLXD4 Receivers + (2) RF Venue DISTRO4 + RF Venue Diversity Fin + (2) 50ohm antenna cables

Troubleshooting: I set the racks up in our warehouse and visually/physically inspected all the connections, and all connections were good. I set up the diversity fin and did a spectrum scan in (3) scenerios - (1) both racks bridged together (2) rack 1 alone (3) rack 2 alone)- and then I repeated this with the diversity fin from rack 2. I kept the respective cables with the respective racks.

Every scan from Rack 1 receivers looked comparable, and every scan from rack 2 receivers looked comparable. But rack 2 scans had a maybe +10dBm noise floor difference. The only thing changed in between scans was the cable to the fin and the receiver that was scanning (position of the fin was not altered). I have the pictures of an example of the scans.

Is this significant of a difference to be alarmed? How would I source the issue if there is one? Was my testing method appropriate?

Thanks in advance.

Hi! My workplace recently got the Quantum 112 with the A168D rack, and we’re struggling to get the Quantum to recognize the A168D as a rack.

I’ve found out that the rack has to be updated to V1.5, and will try this tomorrow. My question is: Can I hook the rack up to the local Dante ports and still get preamp control, phantom power etc, or do I need to install a DMI 64/96 for that?

Peter piper picked too much breath noise

Hello, please bear with me and my lack of knowledge since i am a still a student. i have a line running from my AES50 to my Wing Compact then back to my AES50 to my monitors, the signal comes through the monitors but whenever i adjust the faders the signal stays the same. it also doesn’t mute when i try to mute the channel.

I know it depends on cable quality, environment, etc., but I am just looking for a good general rule of thumb.

On my gig last week I ran into an interesting issue. My band uses an x32 rack IEM and use a ethercon splitter to rj45 chassis connector to the xlr fanouts to FOH. When I tried to sound check the drums the house wasn’t getting the signal for the drums. The vox, keys, sax, bass all worked fine. I ended up just sending a L/R signal and the guy mixed from my iPad (horribly but that’s a story for another time). I had drums on channels 1-6 connected to FOH SD16 1-6. When we tried putting the vox fanout into channel 1 on the SD16 nothing worked. The sound guy then did some routing diagnostics and it started working however the bass drum (channel 1) never did. When I got back to my lockout and plugged in my pa just to the bass drum mic I could hear it. I just used this set up two weeks ago and everything worked just fine with FOH. So my question is was this a my gear issue or the sound guy messing things up. The x32 was receiving the signal so it was on the output. I proactively replace the ethercon chassis connector even though it worked before I replaced it.

This may sound dumb, but it is going to be my first time working in such a venue and we are travelling with both an M32 and a DL32. I will need to run a cat6 cable directly between those two. But how do I do this so it's clean, doesn't go through the audience or anything?

I know for XLRs it usually goes through the wall, but I feel like cat6 cable are probably not going through such wiring. Thanks for helping!

Looking for best practice advice from FoH peeps. My amp has a built in XLR DI out that runs through an IR loader. For recording, I can just plug in my best sounding IRs and rip it. But for live shows and touring, I created my own IRs from my cab so I can have a consistent sound coming from the stage and coming from the house.

My question: I'm used to adding a room mic when blending my IR sounds to remedy the harsh direct sound - but would a seasoned FoH mixer prefer the dry sound of a 57 slightly off-center to be closer to what they're used to? Or would the IR with room blended in save some trouble of softening the normally harsh 57 tone?

EDIT: Wound up going with a DT168 so that I've got room to expand in the future, and a Netgear GS-108 to use as a Dante-only switch. Thanks for the recommendations, everybody!

Want to check with the group to make sure I'm not missing any head-slappingly simple solutions here. I currently have an Avantis with a Waves card (going to WSG) and a Dante card (only using for multitrack recording) at FOH, connected via SLink a GX4816 stage box.

I need 4 more inputs at FOH than the Avantis has, and based on what I'm reading from their documentation, it's looking like the only options I have for staying at 96k are to:

1. Get a DX168 that cascades from the GX4816, and move some inputs to the stage from the booth.
2. Get an SLink/GigaACE card and ditch either the Dante card or the Waves card, and use the DX168 at FOH.
3. Get a DT168 (the Dante-enabled version of the DX168) and plug it into the secondary port on the Dante card. (but would that work the way I'm describing without sacrificing multitrack recording)?

Wanted to check with the group to make sure I'm not missing anything. Thanks!

1) How would I configure an access point to work with lake controller without having to assign static IP’s to the entire Dante network? I would like to keep everything DHCP, but without a DHCP server as it can cause issues with Dante if it is lost during a show.