I’ve seen people complain about the length of the Shure push to talk button on the Axient handheld but what is this atrocity???

Hello World, this week I have calibrated a system for a small event. I'm familiar with Smaart and other calibrating tool. I have already seen this king of measurement but I doesn't know why it's happen : a small phase rotation at the crossover section.
I join some pictures of the mesure I made, the physical thing is in the red circle.

To resume : One RCF Art 915-A for the Top and TP218/1600 A (Thomann) for the Sub.
Crossover made with a DBX PA2 cut at 115 hz (LR 24)

The phase rotation appear and disappear with the mic position, but the majority of time the rotation is present as we can see on the third picture : 3 positions to made the average and the rotation is present.

I think it depend on the type of filter ( LR / But / 12,24 etc ) but it's change nothing to very little difference to the phase, and always on the cutoff frequency.

SO, here my question : how is this happening ?

I am a Junior in High school looking into going into live sound, I ideally want to go into concerts and live theater. I want to know what things I can do to accelerate my career

Context:
I fully run sound for my schools a cappella choir with 16 wireless handhelds on a Behringer x32 compact. They are all vocal mics including a bass pedal mic and a beatboxing mic. I am in charge of running the board, full gear setup including speakers and monitors.
I also do very basic sound for my church, all the setup is done for me so it’s mostly just muting and unmuting channels and occasionally slightly changing EQ, it is a podium, a handheld over the alter, 2 solo mics for the choir, 3 choir condensers, a guitar, and a mic for a violin
I am shadowing the lead engineer at a large theater near me at their upcoming production.
I am planning on going to college for audio engineering, my current top 2 are Belmont and MTSU, but let me know if you have any good colleges for live sound.

If you have any questions on anything please ask

I'm not exactly new to this business but I'm still unsure how to introduce myself to the artists or whether I should even introduce myself when working for a festival or a venue.

Sometimes it is really clear that they want to shake hands etc. and sometimes it's quite clear that it's better to be invisible, especially when doing FOH, but there are some situations where I can't read what the artist expects and sometimes this results in either me being too shy which makes the artists not feel welcome enough or me being too bold when offering a hand shake and then being met with a hesitation. Should I always offer a handshake and initiate the introduction and just think that whoever doesn't accept a handshake is a dick and simply not care or should I be more careful? I think that the handshake thing became much more of a problem after the pandemic.

So what do you think is the best approach for different genres, different positions and different level of the artists?

Our monitor X32 froze on us after a show, and when we powered it back on, the screen did not turn on, showing only dim lights. I know there is a post here stating it was the power supply unit; we have replaced the power supply and tried doing a force boot from a flash drive, and now we are lost regarding what is going on. Do you have any insight into what we did wrong or what our possible next steps could be?

I just got my x32 hooked back to my PC from 2 weeks away, and discovered that Microsoft has blocked the drivers for my console. The x-uf card drivers have been disabled by Microsoft. Both the USB and FireWire drivers. I know that I have an older system but very frustrating to suddenly not be able to use gear due to a windows update. Any one found a work around.

Hey guys, so I’m in a cover band and we’re probably gonna be playing small venues ~ 150 to 200 people. I’m currently running our backing tracks over usb through a linked stereo pair that I set up on the x32 and had a couple of questions.

Does it really matter? Is someone on the left side of the stage really gonna lose all that info from the right if I’m sending a true stereo source? Its not like I’m playing a 60 foot wide stage lol

Do I need to be concerned about sending playback tracks through my mono bus for smaller venues and getting an accurate mix no matter where somewhere one is in the room?

Do I need to pan that linked pair hard left and hard right, or keep them centered? From what I understand if it is a linked pair, the pan becomes a balance and to keep them centered. Is that true? 

What are most PA playback systems in venues? I’ve reading they are all mostly mono. In that case should I just hand FOH the left or right from my main and call it day or route thru my mono bus and hand them one xlr? 

Our stems seem to be true stereo files although some also look dual mono.

Any help is much appreciated and please tell me if I overthinking this lol.

It's all in the title. 🤦🏻‍♂️😩🙀

Our band has a Behringer X32 Unit that we use to mix our own ears for our shows. Everyone uses a phone or tablet and dials in their own in-ear mix. However, one of our band members recently got a brand new iPad Air, and the X32 mixing app doesn't work on it. He also tried the MX mix app, and that doesn't seem to work either. Does anyone know if there's another app available for newer Apple iOS that works with the X32? Or maybe we need to do a software or firmware update on the unit? It's about 4 years old.

Hi there,

With my band we are running backing tracks from Logic into our XR18 (for IEM) and sending a summed channel to one of the AUX outputs to FOH.

We frequently have requests from the FOH to give separated channels instead (so, one for Backing Guitars, one for Backing vocals, one for Synths).
For our other channels (live guitars, bass, vocals, etc) we have a splitter and send the clean signal to FOH.

Now we are evaluating how to do this properly to be able to send the individual channels for FOH as well and one of our options is the X32 rack, but I am not sure this would be ideal.
Our plan is to send the tracks to the X32 and use the Outs (not the Auxs) to FOH with each track. From what I understand, this would be a pre-gain signal (for the tracks comming from Logic at least).
Would this be a viable solution?

I saw a lot about the PlayAUDIO, but this would add one more layer of possible issues/noise to the system (plus 1U and of course the money). So if we can avoid this setup and use only the X32 it would be great :)

Hey, so, I've been recording almost all of my shows from gigs that I get to use a console to multitrack with- ie, if it's an M32, I'm taking a full multitrack recording of the show.

Past those multitracks, when I'm mixing down, what should my aims be? Like, how can I mix them down so the artist can send to a mastering engineer if they like it enough? I ask because I've never really been a recording engineer, just a live guy.

What are the general things people look for in a mixed down session?

Not even this one game in particular, but just generally... what's the thought process there? Is it because the acoustics within the stadium already add an effect of its own and they're using the same mix for live performance and stream? Even just a simple pinch of verb would sound so much better imo. Surely MSG can afford the routing and engineering capabilities to make two mixes.

i work in a setting with many different groups using the same mics, does something that adapts to source content exist out there? Hoping not to have to manage frequencies and thresholds for this

Hey doods!

Quick background...

I've been a drummer most of my life. I started mixing sound for local bands roughly 20 years ago. I have no desire to be a touring engineer, but I'm always looking to improve my skills as a local guy.

That said...

As a drummer, it's pretty easy for me to fix drum tones. These days, probably due to the age demographic of my clients, I don't have to deal with crappy drum tones. Drummers my age have enough exp3rienc3 and budget to bring good tone to the gig. I wish I could say the same for guitar players. Despite almost unlimited budgets and decades of 3xperience, sometimes the tone is - shall we say, "less than optimal." Bad bass guitar tone is a little easier to fix.

Where to start...

I always start at the front end. I try to repair tone at the source. I will tell guitar players or bass players what is bad about their tone. Sometimes I can even show them in the RTA. One bass player was sending signal that had nothing over 400Hz. I was able to show him this in the RTA and we fixed it together.

I know the the guitar is a midrange instrument. But not every guitar player seems to know this. The Loudness Curve becomes my worst enemy. The way they strum the guitar seems to elude them as they bang on the strings as hard as they can. We've all been there.

So my questions...

1. Is it our job to correct bad tone at the back end (at the console)?
2. If so, is there an algorithm you guys follow to do so?

Thanx!

D

UPDATE: Clearly, this community is split in half between "never screw with their tone" vs. "always help the musician any way you can." It's funny how sanctimonious some of you are. For that crowd, take it down a notch. For the rest of you, I appreciate your input.

Just ran sound at a small one-day festival of mostly punk and metal bands. Used the DI outputs of the bassist’s amps.

Most of the time this was OK but there were a couple of cases where the bass signal lacked all high mids and top end and this prevented it from sitting well in the mix and properly supporting overdriven guitar.

What is your go-to strategy for reliable bass tone in live settings if you only have one bass channel? Do you just make your bass players run through your personal DI box and call it a day?

If you use an XR18 do you do anything besides apply EQ and compression?

How do you make a snare "snap" in a live mix? Meaning a bright transient attack at the start of each snare hit that really cuts through the mix.

I figured an actual sound mixing discussion could be interesting.

Is it all done with EQ mostly, or compression, some fancy plugin, hardware insert, mic choice/placement, top and/or bottom mics, etc?? How do you do it and what flavor of each of these things do you like best?

Edit: Let's say that the sound engineer tuning the snare isn't really on the tablet for whatever reason. You get what the drummer gives you to work with.

Voices are muted in the demo and I renamed the buses, so no real names show. I recorded with OBS on the same PC again.

Quick recap: this is a software WING that runs the console’s own firmware (WING Rack 3.1.1), all four DSPs included, instead of a re-creation. The goal is still a virtual sound check at home: play a multitrack recording through the real WING processing on my PC, and mix it on the console’s own interface.

What’s new since last time

• WING Edit works both ways. PC → mixer and mixer → PC both stay connected now, meters included. The console sends its meters over UDP, and my network bridge was mishandling those packets. That’s fixed.
• The console itself runs faster than real time. The console’s main processor and the front-panel controller now run at about 1.2× real time while the audio runs
• Outputs follow the console’s own routing. I can map PC sound outputs to any of the console’s output sockets (local, aux, USB, card), and they play whatever the console routes there: buses, mains, matrices, or inputs patched straight through. The monitor bus, solo and the phones knob work too.
• The Rack’s front panel works. The buttons, the screen knob and the fader knobs go through the panel’s own firmware, just like on the real Rack.
• The RTA works. The EQ page shows the console’s own analyzer.
• One web page for the screen, the patch page, the audio settings and the status.

Speed

Last time the DSPs used about 86 % of real time. Since then I’ve made the emulated DSP code a lot faster and cut out a lot of overhead between audio blocks:

• With just the four DSPs running (no console), they now need about 120 µs of every 167 µs audio block. That’s about 1.38× faster than real time.
• With my full show live (all 16 FX slots, 32-track card playback, WING Edit connected), it’s about 139 µs of 167, so roughly 17 % spare.

What’s still limited

• It still needs a strong PC (Ryzen 7 8845HS, 8 cores): each DSP gets its own core.
• When Windows Defender scans or another heavy program starts, I can still get short gaps.
• Channel and output delays don’t work yet. From the WING service manual I found that the bus DSP has its own memory chip, which it uses for those delays, and I don’t emulate that yet. It’s next on the list.
• The XMOS (USB audio) and the FPGA’s audio side still aren’t emulated. My own code moves the audio in and out of the DSPs where the FPGA would.

What’s next

• More speed: cheaper handling of the console’s messages to the DSPs, and faster DSP code (I estimate another 10–20 %).
• Then hopefully running on less powerful PCs.

Settle a bet Reddit. I know it will pass sound and work in a pinch, but what are the downside of using the “thru” male XLR port as your primary input, and then the female “input” port as your thru on a powered speaker with no attenuator?

Another engineer and I were debating whether or not an HPF could cause a signal to clip so I generated these graphs in Octave. This is a 100Hz Square wave with a Butterworth high pass filter at 100hz.

I've been surfing a while to find what sound system, mixing system, monitor system, etc. they use and can't find zip. Even the photo search doesn't show much of the control area. Is there anywhere we can get info on that?

Looking to get a nice pair of gloves to keep the hands warm in these sub-zero gigs. I’ve seen some gloves with removable fingertips which look perfect for my system teching (especially for using touch screen interfaces).

I’ve previously just stuck with fingerless but weather is getting to me this winter and my fingertips feel as though they’re gonna fall off any minute in the frost.

Has anyone used these removable finger-tip gloves? Is it as practical as it seems? Is it easier to just pop off a glove when you need to?

TIA

Hello, I am the actor here so I know nothing about this stuff but I figured I’d give this a shot to relay to the sound person. I am in a small community theater production of Little Shop of Horrors and we have repeatedly had issues with my body mic starting during tech week. It’s ONLY me. We have changed out the packs and the elements multiple times. We do and extensive mic check with me moving around the stage and turning my head back and forth and mic check is always fine with the sound coming through clear. The second the show starts my mic is cutting in and out or not coming through at all. Everyone is baffled and I’m super frustrated. I’m a lead in the show and it’s affecting my performance bc it’s so distracting. I am not wearing TWO body mics for back up(one under my wig and coming out on my forehead and one around my ear and taped to my face) and it’s still a problem. What could the issue be?? Note: I am the only one in the show with a wig, that’s the only difference I can see. But again, mic check always goes well. PLEASE give me some advice, it’s driving me crazy.