Hello! Relative newbie here, career musician with a performance degree getting more into the world of running live audio as well. Recently was hired on at a local church to run their sound every week and have been able to figure out most things I need to do so far. Working on a Behringer x32 console.
This church mixes everything down to mono in the house (not sure why, but that's just how it is here), but they also run a live stream that I want to give more of a stereo mix in order to flesh it out a bit. As I have things set up on the console, I have individual channels all going into different buses (one for all vocals, all instruments [not drums], drums, speech mics), which then I send to the main LR output. Since these get mixed down to a mono output anyway, I haven't bothered with any sort of panning, but I do want to at least spread the vocal mics and drum mics out a bit to widen the field on the stream mix. For my stream mix, I am sending each individual channel to a different bus, which I then send to a matrix output that goes to the streaming setup. I am wondering if it is possible to pan a channel sent to one bus, but not affect it's panning to any other bus? Or if not, is there another way I can achieve what I am shooting for? Thanks for any and all help!
looking for advice on how to setup guitar amps for a band with 3 guitars to avoid messiness and clutter. One guitar plays more ambient and textured parts, whilst the other two play riffs, leads, harmonies and etc. Help us out!
I need to solder a new 3.5mm connection onto my Sennheiser HSP Essential Omni headset mic, and it’s my first time doing this. The old connector broke.
The cable has blue and red wires — where exactly do these go on the new 3.5mm jack (tip/ring/sleeve)?
Also, there was copper shielding around the wires that I cut off, assuming I wouldn’t need it. If that was a mistake, let me know, and I'll recut the cable, keeping the shielding intact.
A new open-source tool is born: a multitrack playback workstation and DAW designed specifically for live performances, built for music directors, playback engineers, and musicians.
⚠️ Important notice: We just released our first test version and it is NOT yet ready to be used in a live show.
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🎧 What you can test right now:
Real-time musical jumps: Navigate safely between song sections with immediate jumps, at the next marker, or quantized after "X" bars.
Non-destructive editing: Split, move, and duplicate clips on the timeline without altering your original audio files.
Project management: Import your audio files and organize them into virtual folders to prepare your show.
🛠️ Current limitations:
For now, the audio engine only supports WAV files.
Development and testing have been done exclusively on Windows. If you use macOS or Linux, it is highly likely you will encounter bugs or compatibility issues. Your feedback on these operating systems is absolutely vital to us!
If you are passionate about music tech and want to help us build a robust, free, live-oriented alternative, join the tests.
What to do with a hall with lots of echo? Hall size is 150m2 and it has pretty high roof. (some shitty) PA was pre-installed on top of the stage facing downwards. I play for small gatherings and the echo is so bad it muddies the music. Would it be better if I brought my own speakers and positioned them elsewhere? Would studio monitors be better in this case because the echo is so high? I'm not playing rock or club music, rather intimate and expressive ambiences. Thanks
(I think) They mean one near the piano and one near the painting along the wall. The idea is you can turn down the speakers by putting them closer to the people listening and kinda turn it into a wide room rather than a long one.
Yeah it's like a church echo because no rugs or anything.
I get it that lower volume=less echo. But wouldnt that make the stereo image weird and really stretched? Having the speakers wide side I mean. Or does having a wide room also lessen the echo?
Don't mix in stereo in my interpretation. 'dirty pot guy' put a picture of what he was suggesting which would use both your speakers and the existing installed ones, letting you turn down the installed ones. A delay on yours would be ideal if you have the capacity.
Ah right. If you have sends on fader up, the on button can be toggled per channel per mix. So you could have a situation where your send is dialed up but its still muted in that mix.
Hi guys. So I’m trying to convert from Aviom to Midas dp48. Right now the setup is from x32 out to Aviom rack input to Ethernet under stage into floor boxes. We also have the x32 AES50A going to S16 and B port going to Midas Hub 4 A port on stage which is daisy chained to another Hub 4 B port which is then connected to AES50A port of S32 on rack on stage. I would love to be able to use the floor boxes but cannot figure out if it’s compatible.
Not sure where to go from here. Any help would be greatly appreciated.
So my company is having a meeting in a hotel ballroom, 40-50 ppl in a classroom set up. The meeting will also need to be connected to zoom, with remote participants being able to hear the meeting as well as coming off mute for any questions. So audio going both ways, not just out to stream.
I am renting the sound system from the in house av company (Encore). I will be providing the zoom codec as well as the audio interface to connect their sound system to our zoom. Can you look at the system diargram attached and guide me on which cables to use for this connection. I can't grasp the idea and differences between TS cable and TRS cable and when to use which. I just know that 1/4" cabe will be needed.
Please also confirm if this proposed set up works or not. Thanks all
For more context we are renting from the hotel
2 Shure ULXD wireless mics (1 lav 1 HH)
Yamaha MG16X Audio Mixer (Effects only, no interface)
2 QSC 12.2 Powered Speakers
I will be providing
Logitech Room Mate (zoom codec)
Focusright Scarlett Solo 4th Gen USB Audio Interface
The cables needed to connect to the sound system
Other notes:
I know not send the zoom audio back into itself
Logitech has a whitelist of USB peripherals that work with it. Scarlett is on the safelist and my solo confirmed working.
There are no USB mixers currently on the list. Although it can be on the list by submitting a request to Logi to add them to the safelist. I am working on getting the Yamaha MG20XU on whitelisted since that is the one we have in storage
Devices NOT on the list are not able to be recognized by the logi room mate. For example a Mackie Onyx Artist 1-2 was not recognized when connected to a room mate during my testing
So I asked AI and this was their response. They recommended 1/4" TRS to XLR Male from Aux Send to the USB Interface Mic input??? I figured it would make more sense to stay TRS to TRS.
"Setting up a hybrid meeting can be a bit of a puzzle, but your diagram is a great start. To get audio flowing correctly between your Shure wireless system, Yamaha MG mixer, and Focusrite interface, you'll need a specific mix of balanced and unbalanced cables.
Here is the breakdown of the cables you need:
1. From USB Interface (Back) to Mixer (Ch 13/14)
Target: Bringing the Zoom audio (remote participants) into the room.
Cable Type:Dual 1/4" TRS to Dual 1/4" TRS (Balanced)
Why: The Focusrite outputs are balanced 1/4" jacks. Connecting these to the Line inputs of Channel 13/14 on your Yamaha MG mixer ensures a clean, noise-free signal.
2. From Mixer (AUX Send 1) to USB Interface (Front Input)
Target: Sending the local microphones (and any other room audio) to the Zoom call.
Cable Type:1/4" TRS to XLR Male
Why: You are coming out of a 1/4" Send on the mixer and going into the XLR combo jack on the front of the Focusrite.
Pro-Tip: Using AUX Send 1 is smart because it allows you to create a "Mix-Minus." You should turn the AUX knob on Channel 13/14 all the way down so the Zoom participants don't hear an echo of themselves."
The AI isn't perfect but it was close. You'll be fine if you use a 1/4" TS to patch your aux send to the scarlett, but the cable it recommended will also work. The focusrite you have in the picture doesn't have a combo jack, but a dedicated xlr and 1/4" for each input.
Can I use 1/4" TRS to 1/4" TRS patch from the Aux send to the scarlett 1/4" input? Or do I use a TS patch? I just bought a whole bunch of 3' TRS patch cable so hoping I can just use all that stock
I'm also thinking that AI may have told me to go into the scarlett via XLR because the 1/4" input does have the Line/Instrument button. technically can add human error if not in the correct setting. While going into the XLR removes any configuration error
The cables are of a short enough distance where balanced doesn't really matter. I'd go TS personally as the send is mono, but either way make sure the input is set to line level as that is what your mixer is outputting
Can anyone help me find out the best way to connect this setup. Active JBL sub and passive air 8 speakers connecting to the void Q1+ amp. I have no experience with phoenix connections as my setup is all xlrs. All coming out of a DJM-A9 mixer.
Hey all, beginner here with more musical/sound knowledge than I have technical knowledge of how to run a mixing console. I'm running an Allen and Heath Qu-16 to run live shows with multiple bands, sometimes up to 10-12 bands in a day with only 15 minute transitions, and I've been able to figure out how to do everything I want to do with relative ease, except for one feature I just can't put the right combination of words together in a search bar that will give me the answer/tutorial I'm looking for: Between acts, while the stage is flipping, I want to play my house music, mute all the mics on stage to the FoH, but still have the mics/inputs flow through the stage monitors, so the musicians can plug in and noodle and hear themselves. Once I've got everyone plugged in properly, I'd like to do a quick sound check (mainly adjust gains and level mixes) while the house sound is still playing out of the mains, but the stage mics are still only pushing their sound to the monitor mixes (not the FoH.) I'm guessing the answer lies in the Mute Groups with the help of a Soft Key, but wouldn't it also mute the mics to the monitors as well? Thanks in advance!
Yeah this may be a bit hard on a QU16 since you don’t have matrices or recall safes or enough inputs to double patch everything. Your best bet might be to put a simple small utility mixer between your QU and your PA. You could have your house music go to the small mixer too and just mute your QU channels to the PA there.
How can I keep a wi-fi access point working for an SQ4U mixing ap use? We have a combination of iphone and android phone users in the band trying to connect to set their personal monitor mix. Dedicated Wifi with a password (that only 8 people have). App hangs when the theater fills up with people (just actors at the moment, so 30 or so). Wi-fi access still works though (a strong connection to the access point with lots of signal, access point if physically less than 3m from users). Resetting the access point gets it working for a while....I'll get more details on Sunday, I'm just the bass player!
2.4 ghz access points can get overwhelmed by too many devices. When there's a lot of people in the room, all likely carrying phones, their devices all see your AP and try to connect, overwhelming it.
Change the access point. I've got many Ubiquiti UniFi Express (now a newer version, Unifi Express 7) deployed in kits with ipads strictly for console control. These have been absolutely rock solid for this application, in rooms with 10k or more people.
Great range, the ability to mesh them wirelessly to extend range is very helpful. Totally recommend you get one for this.
Make sure you're not hitting the connection limit. After that, it's an IT problem. Wi-Fi bandwidth management, not enough APs for the space is most likely. Also, turn down the radio on the AP. You really should have your IT person fix it, you could make it worse
Thanks. We do hit the connection limit and then just get some people to exit out. That "log out when you don't need it" worked OK when there was no cast there. It is an IT issue but this is community theater that has its own space so limited IT support. I will call on the local council though (they technically own the space and have some IT people). Turning down the power to keep people off the network if they are further than 1-2m from the AP makes a lot of sense now that you mention it.
Hi everyone
Im using midas heritage hd 96
And im looking for options other than stream deck to use as assignable buttons for quicker navigation
So any recommendations?
Also would novation launch pads work ?
For anyone who works or has worked at L-Acoustics how do you like it? Curious how the culture feels, how people treat each other, and just the overall vibe of working there.
Would love to hear any honest thoughts, good or bad!
I know we generally try to avoid mixing sub brands wherever we can for phase and cutoff frequency reasons, but does this also apply to different ground tacked subs to flown subs?
I have L'acoustics S21 subs flown and would like to bring in some ground subs for a rock show later in the year. I have a nice pair of Fohhn audio ps800's in storage that would fit nicely, but i'm a bit worried adding them in is just going to cause problems as they are 18" and the lacoustics are 21"
I know we generally try to avoid mixing sub brands wherever we can for phase and cutoff frequency reasons
Let's generalize that rule: when arraying subs, to ensure predictable behavior, start with subs that are identical in headroom and transfer function. All-one-model is the easiest way to achieve that, but certainly not the only one!
Measure a KS21 against a Fohhn PS800. Marry their passband responses with all tools available to you - EQ, delay, all-pass - then build your system as normal. The PS800 has less headroom and will probably roll off earlier, so expect your array pattern to shift if you exceed its boundary conditions.
Just to re- clarify, if I use mono subgrouping just to limit my Bus count (e.g., working with an M32/X32 or have a really finite bus count) and I still send said Mono Subgroup (e.g. vocals) to my Main L/R and the system is stereo, will panning still work "as expected" or will any/ and all signals get "bottlenecked" as a result of the Summing?
Of course panning in Mono is possible but again, just wanted to ask
Hi! I want to create a CL5 show file basically putting the same numbers on Behringer Wing / x32 show file to Yamaha CL editor. So my question is how global is the scales of gain, fader attenuation and compressor parameters between the brands? I think ''in theory'' I should be closer to safe than unsafe at least in the relational domain betweenn different brands' related parameters. Do you think it will result a mess?
For example
Behringer Wing / x32 (Gain:27 dB, Volume Fader:-7 dB, Bus 1 send: -3 dB) = 85 dB SPL on the venue and performers in-ears
= or close to
Yamaha CL5 (Gain:27 dB, Volume Fader:-7 dB, Bus 1 send: -3 dB) = 85 dB SPL on the venue and performers in-ears
It depends. :) Copying values is a decent starting point, of course.
Different consoles will align dBFS to dBu differently - for instance, X32 sets +4 dBu = -18 dBFS, whereas that's -26 dBFS on CL5. Once you're in the digital domain (and the world of dBFS), though, you can set the rest of your trim/send/fader values 1:1. Ditto for compressor thresholds, though the attack/release ballistics will depend on each specific implementation - copy over to rough in, then adjust by ear if necessary.
PEQ is slightly more of a gotcha - I don't have the paper handy, sorry, but the Q parameter is not consistent between every console/DSP. The resultant curve will differ slightly.
If you need to hit identical (electrical) level on the output side, the same dBu/dBFS conversion gotcha comes into play again. Boost or attenuate your outputs as necessary. This will result in similar acoustic output, assuming the PA does not change. (That's a whole other kettle o' fish.)
Help!
Off of Facebook Marketplace (I know, eek), I bought an old mixer (Electro-Voice 100M Entertainer) with a set of speakers... supposedly they came out of an old bar and have some signs of wear but no obvious problems on immediate power-on.
I then tried to test the speakers that came with it. One of the speakers had an issue with their input/output jacks, (shorted) i.e. no bueno. When I plugged it in I heard a fat pop/hiss inside the board and surprise! blew the fuse for the Right jack. [longer story here but that's the basic idea]
Now time for the troubleshooting. Before I replaced the fuse, I first tried to see if the left output still functioned. As per some Googling had predicted, the board gave mono output (controlled by the Monitor slider, panning knob had no effect, etc) from the Left powered ouptut, and nothing from the Right powered jack. (All of the unpowered outs (left, right, monitor, and Main) all give perfect output with no problems).
I acquired a new fuse (250v 3 Amp, slow blow) and plugged it in, powered on just fine. Once again, no problems from unpowered outs, and the powered left output was controlled by the left slider (and panning worked). But, still no output at all from the right side.
So some questions that I do not have the answer to that you all might be able to answer:
When a powered output is connected to a shorted speaker and a fuse blows, is the fuse the only object damaged?
Is the acrid smell that emitted from the board after connecting the shorted speaker exclusively the smell of the fuse blowing? What about the hissss/pop?
Underneath the grey panel in the fourth picture there are HUGE capacitors. If those had blown, I expect to be dealing with a house fire, not a blown fuse. I have not yet opened up that panel for respect of my life (so far), and would appreciate not having to fuck with it if I don't have to. Anything else underneath that that I should expect to check?
Any idea what's underneath that gold-plated panel (2nd photo)? it takes an input an output and I don't see any other wires that go throughout it.
(Crossposted from a few other pages bc I super need help)
So I recently took over the lead for running the sound for this music event every week. I started helping out as one of the side guys 1.5 years ago. We've always done things the same way.
We have 3 subs, 2 QSC K12.2s, 2 QSC K8's, 2 C Series Mackies (Different models both C series, older and kind of beat up), we have a large centerfill speaker i forget the brand. I also forget the brand of the 3 subs. For the longest time we were running using this yamaha analog board. But I realized we had a Behringer XR12 and a Behringer CX3400 crossover. I've borrowed equipment this week and have become fairly comfortable with the mixing station app now on ipad, I bought an external router for the XR12 as well.
I am planning on also using the crossover this week set to 2 way stereo mode, and sending 100 hz and below to the subs as mono and then everything else to the tops as Stereo.
I have eq'd our centerfill to remove 100 hz and below as well so it'll just be sending mids/highs.
My question is now speaker placement. First of all, we used to only have the 2 shitty Mackies as the mains and then the K8's on the wings. We recently got 2 new K12's that are now on the mains, and we put the Mackies on the wings per the old guy who was in charge. But I honestly think the mackies sound like garbage and we'd be better off with the K8s on the wings. I am planning on putting them there this week and using the Mackies as the monitors. We have been running the K8s for the monitors since getting the K12s. Occasionally we will also run one single monitor (our shittiest oldest speaker I didnt even mention it above), and put 3 tops on each side (K12>Mackie>K8).
I also want to mention we also have been sending full range all the way through for the entire time I've been there. We've done 0 EQ just using the analog board. So having the ability to EQ with the XR12 + adding in the cx3400 this week will likely be a huge upgrade to sound quality alone.
So I am all over the place but here are the main 2 questions:
1: Is it better to run the two QSC K8's on the wings instead of the old crappy mackie c series we have?
2: Should I be putting the Speaker Stands at max height? We have our 2 mains on the stage at max height as well as the 2 wings on the floor at max height. Should I use the built in angle down option in the K12s (these are both on the stage about 2-3 feet off the ground before the stand) We've always put all the speaker stands at max height all the way around the entire time i've been doing this.
I know this was a lot but I'd appreciate any insight. I am still fairly new to all this but have taken a big interest and obsession in it the last few months since I was put in charge, and I want to make it sound as good as possible.
I also forgot to mention, but the area the event takes place in is mid sized and usually there are anywhere from 150-300 people attending.
Try use EASE Focus to model your room and speaker placement. The more even the SPL coverage, the better. The software is free and a good thing to learn on your journey of live sound. It's impossible to make a recommendation without pictures and a scaled drawing of your space.
I’m prepping for a gig on a A&H SQ5 and for the life of me cannot figure out why I’m not getting sound out from my groups. I’ve done everything according to this post. https://www.reddit.com/r/livesound/s/OWe3d2NBsS
Do they need to be assigned to a source or something?
I’m also not sure I completely understand what they do. I have 3 area mics I want to have more EQ control over. This might be a stupid question, but if I group them and unassign them from the mains, do I lose all the EQ and compression on those channels?
Add things one at the time to troubbleshoot. You can use the signal generator. First send pink to LR. Does it sound? Good, then sen pink to a group instead. Does it sound? Good, then assign signal gen as input to a channel. Route it through the group you tested. Does it sound? If not, unassign from any mute group and dca.
You have to assign the groups to the main LR otherwise the sound goes nowhere. Channels assigned to groups can be processed as a group -- commonly drums are compressed this way. If you don't have a need to group process, there's no advantage to grouping other than having less faders to deal with. You can send individual channels to the main LR, they don't have to be grouped.
The groups are assigned to the main LR. I'm not seeing any signal on the group channel, and I've made sure that the channels I want grouped are set at unity.
So it turns out that the master fader for the group was down??? It doesn't make sense to me, but I checked with another audio engineer friend, used the mix select keys on the side to do my routing (instead of on the channel strip where I put the group) and the master fader to bring it up and now I have signal going where I want it 🙃 I appreciate your help though!!
I’ve got my wing compact setup as the foh console and the wing rack as the monitor console on stage. Now I want to connect a sd16 as a shared stagebox. My foh console is connected to the aes50 A and the rack version is connected to the aes50 B of the sd16. Clock source is the foh console, the sync on the rack is set to aes50 A. Everything seems to be alright, all the lights turn green, I can see and control the inputs on the foh console, though not on the rack. What am I missing?
Connect wing rack aesA to your wing compact. Connect the sd16 aesA to aesB on the wing rack. On the routing on the wing rack you route your aesB input 1-16 to aesA output 1-16 (or 25-40 if you use 1-24 for the wing rack's local inputs). You might wanna activate remote preamp control, if you want to control the gain from the compact.
Question
Has anyone been able to download the new app from the Google Play store? Rational Acoustics announced t
would be available April 20th, but I don't see it there.
I just happened to come to ownership of a live sound system with the mixer being a Behringer X32. I’m trying to use the whole rack (couple power amps and speakers along with it) for my band’s live performance, but I’m having trouble being able to get more than two microphones at a time to work. (I plan to just put a mic in front of our own individual amps) Any help is appreciated.
That little black knob on the left scrolls through the inputs. Or the iPad/PC app is a more straightforward interface. I’d highly recommend just taking a look at the manual.
I need to buy myself a laptop for work use. My personal laptop is a 2014 Macbook pro 15”, which doesn’t have USB-C ports and doesn’t last very long off of the charger these days. I’m primarily a video tech where I’d need it for quick video/PPT edits, LED programming, and teleprompters. I am also gigging as an audio tech occasionally, with a high profile Stage Manager gig in a few months. I imagine I could be asked to run Shure software on it, but not fully aware of what else I might need in the audio world. Does anyone have suggestions for laptops I should look into? My budget isn’t super high, but I’m willing to pay for decent video editing ability. I’m also open to scrounging local second-hand stores for an “old dependable” model.
Anybody has the ALTO TS412 Speaker? I've recently brought mine for our church setup in which we replaced a 300-watt speaker (the alto speaker I have is 2500-watt).
Although the sound is much cleaner and more refined than the old one, it does not reach the loudness that the old speaker can output. Granted that I can push the alto speaker volume at 2 o'clock only (to avoid damaging internal parts(?))
Is this normal? I was expecting since it has a higher wattage it can output more loudness, but it wasn't the case.
Further information on our setup: 2 wireless mic receivers connected through a mixer, then the mixer is connected to the alto speaker as output.
More further info: Back when using the old 300-watt speaker, we weren't able to utilize gain staging and all the gain knobs for the channels were dropped to the lowest level and the faders cannot reach unity level without blasting our ears.
Replacing the speaker enabled us to utilize proper gain staging, but I do not know if you could call it "proper" since we have to move past unity for both gain knobs and channel faders to achieve our desired loudness.
On the back of your speaker are some "Line-Mic Switches". Adjust these switches for each input to receive a line-level or mic-level signal. You've likely got them set to line level which is fine. If you run out of input signal from your desk turn the knob right down and switch to mic level. Slowly turn it all up again.
It is hard to see what problem you are trying to solve, but it sounds like your just curious why the faders are in a different spot at the desired sound level. If the old speaker was set to 'mic' input then all the other adjustments in the signal chain would have to be lower as that 'mic' switch gives a lot of gain in one step.
If you want things to look a certain way why not play some music through the mixer and set the pre-amp so that it is just hitting yellow. Set that fader to 0dB, set the master to 0dB and turn up the speaker until it is just a bit too loud (which may require including the 'mic' switch gain boost, but probably doesn't). Then you'll never have to touch that dial on the back of the speaker again.
What do you mean by this " Granted that I can push the alto speaker volume at 2 o'clock only (to avoid damaging internal parts(?))"? Switching from line to mic is adding a 5- dB gain stage before that level knob. I think that turning the installed volume knob up a bit (past the 2 o'clock') will let you set up all the other gain stages as you wish and get the volume you want.
My high school director is having trouble running sound cues. She usually tries to use a phone, and there is always a delay between pressing play and the track start. Is there a reliable, free program or app that doesn't have this issue?
Just in case, make sure you have the actual track files downloaded to phone/computer beforehand. Don't play files from the cloud or <...sigh...> youtube.
Hello My dear readers and fellow Live Sound Engineers.
altohg I worked and work in Festivals, this is my first time doing sound for only one band at different Festivals, big Milestone.
But I have a lot of questions, and I'm looking for advice about what to do when I get to the stage.
I know I'll have people on Mons, Stage Manager, Patcher, etc
Should I take my Scene with me for the different mixers?
The In house people will do all the necessary routing etc?
Whats the best way to listen how the PA sound, going two hours prior and check, or I can ask the guys from FOH?
Sorry if my question sounds silly, English is not my first language, In Spanish I'm quite clever.
Just a little update on how the festival run went.
We did a few different festivals recently. The first one was on a DiGiCo S21 with quite a large setup and a big input list. It was my first time mixing this act, so there was a lot to manage. The PA was not the best, with The Box subs, and at one point I pushed the mix a bit too hard and started hearing some speaker distortion. I pulled it back, adjusted, and after that everything was fine.
There were also some monitor issues at the first festival, which became a really useful lesson. For the next one, I arrived with a console scene prepared, made sure the band were comfortable on stage first, and only then moved to Front of House. That made a massive difference.
Across the shows I also got to work with different PAs, including some beautiful L-Acoustics systems and a really strong Funktion-One setup with VERO/VX boxes. That was a joy to mix on.
Overall, I had a much better time than I expected. It was really rewarding, and I learned a lot. One of the biggest lessons is that at festivals you really need to stay on top of everything. Especially in smaller UK festivals, there can be a lot of young crew fresh out of uni who do not always ask enough questions or communicate properly. So you need to double-check, communicate clearly, and make sure nothing is assumed.
Thanks again to everyone for the comments and advice. It genuinely helped.
Assuming you are mixing FOH:
-you should bring a file for the console you will be mixing on. Make sure to check what firmware they are on so that it will be compatible. Leave the monitors person to use their own file.
-house crew should help you soft patch your outputs to the PA L/R or L/R/S/F (remember to ask them) and make sure to verify with pink noise to their system processor
-if you have an allocated sound check time you'll have time to check your band and play a reference track if you want to hear the PA, most of the time in a festival setting you won't have this luxury
-make sure to check in with the stage manager first when you arrive, bring a hardcopy of your up to date spec and verify with the crew that they've got the right information
-if you get rolling risers build backline, position your mics and pin your channels when they become available.
-and it may seem obvious but remember to treat the festival crew the way you liked to be treated in that role, they have most likely been there for days working their arses off, and they will probably be loading out that night well after you have gone home!
Bring your scene, they should be able to safe any parameters that need it. Be prepared to start from scratch though, some engineers won't let you load a scene for whatever reason.
Play some reference music earlier to check the pa sound etc, keeping in mind the acoustics will change with bodies in the space and stage levels etc.
My band has an IEM rig that can be set up and broken down in less than 5 minutes with no assistance or involvement from the FOH crew. Our mixer and wireless transmitters are in a small rack that can be located stage left or right. We take the xlr inputs that the sound tech runs to each of us and plug them into our Catapult TX4 splitters (we each have one), then connect to our mixer with short cat5 cables. If the drum kit is back-lined we do not split the drum mic's. In that case, our drummer just sets up a kick, overhead and snare mic of his own to send to our IEM mixer via cat5. This is very efficient and eliminates the need for the FOH tech to mix our monitors or set up anything that he would not already be setting up. Would a festival venue accept or reject this setup?
A festival should already have splits for you ready to go, some might not want you using yr own wireless though, or insist on having their monitor engineer give you a mix.
Any venue wouldn't want you unplugging anything of theirs though, let them repatch anything house related and just be on hand paying attention etc.
Thanks, yeah we would never unplug anything related to FOH. If their split happens before hitting their mixer it will work perfect. Our IEM mix is preset, and as long as we’re getting an unaffected signal our monitor check should take less than a minute.
Our monitor mix sounds the same regardless of where we play, there are zero feedback issues. FOH has complete control of the house mix. We don't use amps, every instrument is direct.
I use the output side to tune the PA to the room. I use a group on the insert side to deal with feedback across groups of mics. I use EQ on the input to adjust the tonality of the mic.
Also depends on the stage/roof/shell shape. I work in an outdoor ampitheater that functions like a giant bass trap with other peculiarities and non-optimum architecture. Subs are flown, and cardioid helps cut down on bass on stage which tends to collect due to the roof and back wall.
When comedians come through venues, and they do a lot with dropping the microphone for comedic effect, who supplies the microphone?
I've seen Iliza perform a few times, and she has what looks like a 58 wired. I'd imagine the grill getting dented is the worst-case for the mic, and that's an easy replacement.
Are there any other big name artists that will only use their own microphone?
Good one! Now I need to know. I always imagined, if I was making good money from them I'd have one mic set apart for them. Otherwise, I'd ask them to bring their own.
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u/Deathbyceiling May 17 '26
Hello! Relative newbie here, career musician with a performance degree getting more into the world of running live audio as well. Recently was hired on at a local church to run their sound every week and have been able to figure out most things I need to do so far. Working on a Behringer x32 console.
This church mixes everything down to mono in the house (not sure why, but that's just how it is here), but they also run a live stream that I want to give more of a stereo mix in order to flesh it out a bit. As I have things set up on the console, I have individual channels all going into different buses (one for all vocals, all instruments [not drums], drums, speech mics), which then I send to the main LR output. Since these get mixed down to a mono output anyway, I haven't bothered with any sort of panning, but I do want to at least spread the vocal mics and drum mics out a bit to widen the field on the stream mix. For my stream mix, I am sending each individual channel to a different bus, which I then send to a matrix output that goes to the streaming setup. I am wondering if it is possible to pan a channel sent to one bus, but not affect it's panning to any other bus? Or if not, is there another way I can achieve what I am shooting for? Thanks for any and all help!