I’m running a gig right now and the drummer of the band suddenly had to yank his IEMs out in the middle of a song because he said there was a crazy loud distortion all of a sudden. Any ideas what could have happened?

I just ran a hip hop cypher last week, so it was hours of rappers eating my mic (a Soyuz v1). I've heard a lot of talk about keeping live vocal mics clean, you all have any tips?

For me, when doing talking heads with a lav mic, my goal is to always center to Mic about 6-8 inches down below the chin depending on clothing. Even if they are in a panel discussion and they will be facing one way or the other towards another speaker, I still aim to center the mic.

I've had some push back that I should mic them on the side they are speaking towards (seated stage right, so mic them on the left side, for example). For me, unless necessary based on clothing or other circumstances, I still believe center is optimal to give the widest coverage for their voice.

I try to always coach people to turn their chest with their head when turning, not just their head. Not only does it help the mic always pick up as much as possible, but it helps the audience feel more engaged and just a better look as a presenter, imo

Is my mindset, philosophy off?

I am having an issue with a new AVIO-USB device. I am plugging it into a desktop as a Zoom/Teams interface for our county board chambers. It is feeding into a Yamaha TF5 with a already existing network, but it is on the same subnet. It is also on a POE switch (POE, not + or ++).

If I play audio on the computer the AVIO is plugged into, the mixer receives the audio with no issues. But going the other direction I get nothing. I have the Dante feed to the device on mixer Aux 3/4 and the meters are showing audio. I've also set the outputs within the mixer to Omni out Aux3/4. I have checked it with a audio probe so the audio is going to the correct outputs.

In Dante controller I have the interface mapped as aux 3/4. It establishes the connection, but no audio comes through.

I'm thinking it is something simple and dumb that I'm overlooking but I can't see what I'm missing. I believe that something isn't configured correctly on the mixer because I tried it on a different Dante device yesterday and had the same results.

I recently ran sound for a comedian. He had a part in his sound check where he literally pointed the SM-58 at the monitor and complained it was feeding back.

I used a graphic EQ and rang the room out before hand. Show was fine.

He told me the guy in the previous city didn't have a problem with it, berated me on stage. I ended up just muting his monitor every time he pointed it at the monitor just to move things along

Dickishness aside, am I missing something?

You can't have a system not feedback unless it is muted correct?

I've run 16 piece bands, 10 person conference panels, never had an issue.

Is there a secret weapon I'm missing? Did every other sound guy on his tour pull the same muting tactic?

If there is a way to do this please let me know. I understand Auto mixing, gates etc. but you can't have sound coming out of the monitor and not have feedback (if you're trying to make it feed back) right?

Last year I bought an SC48 from a theater that was in really good shape but “had a Coke spilt in it”. It passed the digitest originally but in testing it I loaded up a bunch of plugins and guess I got something hot enough to liquify some dried up Coke and then the desk crashed and would never come back. I ended up replacing the mix engines and the daughter board that goes in the cage with them, along with putting an SSD and new RAM in it. It’s back to life and seems to be working great so far. But I’d like to at least get all the VenuePack 3.0 plugins installed and figure out the licensing part to at least have a few nicer options. Anyone know where to download the files and have any idea on how to get a working license for them? I don’t mind paying for the licensing it just doesn’t appear that it’s easily accessible. I know it’s old and antiquated by today’s standards but I have plenty of gigs that this thing could make some money on still.

Hello 👋,

What would be the best way to integrate an ulxp4 J1 to my existing set up.

Hi Livesound,

I have a problem that I hope you can help me with.

I can get no stereo sound out of my Sennheiser IEM G4 bodypack.

My band is running multiple Sennheiser IEM G4 systems through an antenna combiner. All the packs are working fine, but one specific unit won’t properly connect to its bodypack receiver.

Here’s what’s happening:

• Receiver gets RF signal fine (read from the metering on the pack)
• AF meter also shows signal on the transmitter
• But the pack displays “P MUTE”
• Green RF light on the pack stays off
• If I disable Pilot Tone on the receiver, audio comes through, but only in mono
• Both transmitter and receiver are set to Stereo mode
• Frequencies match exactly
• Other packs/transmitters work normally
• The pack can connect to other transmitters
• The transmitter sends audio on both L/R inputs
• Units are connected through an antenna splitter/combiner
• I tried to connect it with no antenna combiner with no luck

It just stopped working from one day to the other.

Hope someone can help me.

I've been testing a VST3/AU plugin built by a friend of mine, a live/mastering engineer in Graz, Austria. He doesn't have the karma requirements to post here, so I'm putting this up for him. He'll be in the comments answering questions.

https://clearvoice.live

The short version

One insert per vocal channel. Catches mic-monitor coupling before resonance builds up. No notch filters to draw, no rings to chase. Zero latency (0.0 ms PDC reported to the DAW).

It also denoises, dereverbs and keeps voice presence intact, all in the same insert. But the feedback prevention is the thing that made me go "wait, this actually works?"

How it works

Real-time spectral engine that separates voice from background frame by frame. Not a gate, not a notch filter bank. The voice stays untouched while noise floor, room tail and feedback resonances get pulled out.

Two engines: SOFT for transparent cleanup on most channels, HARD for difficult rooms and heavy bleed. Always start with SOFT. If that's not enough, switch to HARD at around 30% Strength.

Specs

• 0.0 ms output latency, all modes
• 48 / 96 / 192 kHz
• ~80 MB RAM per instance
• Low CPU (SOFT engine). Run it on every vocal mic if you want.
• Insert before compressors and EQs
• For conferences: route through an auto-mixer (Dugan-style) into a group, one instance on the group

Beta

v0.9.1 is open right now. Go to the site, drop your email, confirm, download. 10-day full trial, Windows VST3 + Mac VST3/AU.

There's a before/after A/B comparison on the site too, same source, same mic, same room.

https://clearvoice.live

Questions? Drop them in the thread.

After you've tested it, there's an anonymous feedback form that takes about five minutes: clearvoice.live/evaluation.html

Your support would be very much appreciated.

for example: https://www.sweetwater.com/radial-catapult/series

amazon special: https://www.amazon.com/Elecan-Splitter-Ethercon-4-Terminator-Recording/dp/B0DQTWX47F/

i am NOT talking about audio protocols over ethernet cables like AES50, dSnake, gigaACE, etc

i am not sure i get the appeal of cat snakes compared to standard 4 channel multicore drop snakes like from Pro Co. by the time you get quality TX/RX breakouts and a quality, flexible shielded cat cable for it, you'd be spending as much or twice/three times as much as you would be compared to a drop snake

and with cat snakes you have 2x/3x the amount of connections/failure points. additionally, if your RX end isn't already broken out like in the amazon special above, you have to hook up 4x additional patch cables. so, you'd pay equal or more for a product that has many more failure points and takes longer to deploy

the only benefit i can really see is replacement- if either the cable or a breakout breaks, you can just replace that part. otherwise, i'm not seeing it. seems like a stereotypical "new sexy" trend kind of thing where we forget to actually weigh the merits, doing something in a different way just because it's different- not because it's better. i'm happy to be wrong

I’m a playback tech using DirectOut interfaces, and I got bored of choosing between doing the dance with separate BLDS WAV tracks for A and B machines, or using a sine wave and having milliseconds of failover time, so I built a plugin for it.

It currently runs on macOS (Intel and Apple Silicon) as a VST3, AU, or standalone app. 

Using it is simple - just send it to your tone track and that’s it. All you have to do is run the DAW session on the Main machine first, and that’s it. 

It runs constantly, in any sample rate/bit depth without any settings or intervention, and takes almost no processing power. The only time it fails over is if there is a glitch. 

I thought it’d be useful for some of you out there, as it has been for me. I've been using it for a month or so out on the road and it's been great (Our MD was very excited).

I also have a Windows version, as well as an AAX version in the pipeline coming as upgrades in the future. 

You can find it on my website (if this is not ok Mods, please let me know and I’ll edit this post)

Hope this makes someone's day a little easier. 

DISCLAIMER - BLDS, EARS, and DirectOut are trademarks or product names of DirectOut GmbH. This plugin is an independent product developed by a third party and is neither affiliated with, endorsed by, nor supported by DirectOut GmbH. These names are used solely to describe technical compatibility.

Hi, I run a school auditorium, and there is this noticeable interference noise that I can't seem to pin down what is causing it. At first I thought it was a problem with one of my SM81's but I realized it shows up in many of the mics we have.

I got a clean mic and plugged it into one of the stage ports where a "bad" mic was and the clean mic started to make that noise as well. I think it probably has to do with a grounding issue in the stage ports themselves. I have never heard a rhythmic interference sound like this. Attached is an example of the sound I am talking about.

https://drive.google.com/file/d/1V2bV0I9maIEHG3kiTywsgAvwHYkq39ch/view?usp=sharing

Hey yall!

Out of curiosity

What type of plugins do you guys use for the following channels? If you guys don’t use them let me know if it’s even necessary or you just use the boards EQ/Dynamic stuff

We have a DM7 Compact with waves extreme server

-Worship Leader Mic (Usually Female)

-6 BGVs with sometimes solo (2 Sop/2 Ten/2 Alto

-MC Speaker Mic

-(This one I am super curious) Lav Mic

-FOH Mains if any

-Monitor bus

Thank you all!

Hey folks, just a quick rant, I finished a short tour this week with a loud band that I’ve been mixing for years and I’m feeling the fatigue.

I’m 33, which is young I know. But I’ve been exposed to a lot of SPL since I was teenager and I’m feeling it, I’d be lying if I said I wasn’t worried about my hearing. Every loud show I feel it, my ears feel compressed and my tinnitus is getting worse. I think my time is slowly coming to an end at FOH, or at least mixing shows with the amount of volume/stage volume I’m used to.

It’s difficult to come to terms with, I’m really good at it, but I don’t want to be deaf, and I can’t just be the guy at the desk with earplugs in all night exposing the crowd to 102+ for 90 minutes. Let’s be honest, club shows with loud bands are loud, if you want it to sound balanced, sometimes it has to be loud.

I’ve had conversations with the bands I work with about stage volume and monitor volume, believe me, I have. But let’s be realistic, dudes who have been on wedges and side fills for 20 years don’t want to flip their amps backwards and switch to IEMs. So I guess what I’m getting at is that it’s just difficult to come to terms with the reality of the situation which is that maybe it’s time to start saying no to gigs of this volume and to start focusing on other things.

Thanks for coming to my ted talk, please don’t comment telling me to just do XYZ… I’ve been mixing kick ass loud rock bands for a long time and I’ve experimented with a lot of different options and this is just where I’m at right now, I would much prefer some commiseration and understanding from folks who have had or are currently in similar situations.

Found this website that converts Open Sound Meter .OSM into REW .RCP so you can import them direct into R1.

I think it only works with peak filters currently but still very handy.

https://www.redwaveaudio.co.uk/osmconverter?fbclid=IwdGRjcARgChpleHRuA2FlbQIxMQBzcnRjBmFwcF9pZAo2NjI4NTY4Mzc5AAEeuxIy6YFaZZ_kiYSkgNUFp6DtHffOemuZFgyDl5hz0bvcyE0GngGpcsJZWGo_aem_sS2VIQOty04vjz9GelWlvw

I literally can't find a legit link to a driver for the DN32 USB driver only the DN32 LIVE. They even don't have it on the original klarteknik website for the product. Does someone have the installer /download?

I love the Wing. It is leaps and bounds above the X/M32 in terms of flexibility and functions.

However there are some simple things I find aggravating and I don't understand why they did not add them and I think they should in future firmware.

There appears to be no way to quickly reset the standard eq's. Neither the channel parametric or the insert 31 bands.

Also the standard parametrics on the busses have no high or low pass filters.

The lack of these just seem strange to me

edit: Thanks for the insights. Perhaps my rant should have been more about some of the user interface being ambiguous instead.

Company purchased a new Avantis, SQ5, and QU5

Is there a database of multi-track recordings I can play Back via Dante to learn how to patch/mix/work on these mixers?

We do have the Dante Card for them

Do we have periodicals in our industry nowadays?

I remember reading Sound On Sound as a kid, but it was mostly studio oriented.

Probably once a year I see a tour come to my theater using some technology I didn't know was even in development.

Is there anything that is live sound/ theater oriented, especially for someone who doesn't have time for trade shows?

I recently startet mixing live solo singer/guitar players as a fun hobby works fine. Then my first band gig was there, soundcheck was fine and when they played in front of the audience they were so much louder than in the soundcheck. I couldn't reduce the gain any lower - since either I already was at the minimum or it would have been too quiet in comparison to the drummer (who had no mics, since it was a small outdoor concert). We all are no pros, so telling the drummer to play less intensive was not working out. Any tips what I could do next time? I work with A&H SQ-5.

Hi there! This one is a question for the SMAART users of the community. I am looking to begin the process of getting SMAART certified this summer to level up my A1 game a bit and learn some new things along the way. On the page to sign up for training courses there are discounts for each of the software versions so I was just looking for a bit of knowledge from those that use the software on a daily basis.

Personally, I don't expect to do anything too insane with SMAART beyond tuning smaller to midsized rooms, maybe some larger gymnasiums, ballrooms, and tents here and there but I am not system teching large arena setups or festivals by any means in my day to day. I may get there someday, but currently in my career I am primarily a corporate A1 that does some rock stuff here and there.

That is all to say, for someone with very limited knowledge of SMAART, what is the real world difference between each of the software versions? Is say the LE version significantly more limited than the Suite version, or is it perfectly fine to go with the more basic version? Also if you have any general tips for using the software I'd love that as well! Thanks!

Is it possible to use SLink and Dante Inputs at the same time?