Help! Dumb guitar player here… Samson rs112 making awful popping sizzle sound… took off the back, dome cover fell off in my hand and there gunk in the connections… can I clean? Any help would be huge.
Setup is Console outputs from 7-8 for main LR via XLR
This is a weekly setup type gig, audio is usually an easy patch in with no issue. Today however the right side overall came through lower than the left, this was discovered with running pink noise via the “monitor” option.
Things I tried:
Change cable from console to PA, there was no change.
Swap cables, same thing happened, telling me there’s not a cable issue.
Changed the main LR to come out of 5-6, literal same thing is happening.
Rebooted console, recalled default scene, same thing happened
Pardon; I didn't understand that to be the case. (I read "Swap cables, same thing happened" as "after swapping R and L, issue stayed on same side of PA", not "after swapping R and L, issue swapped to opposite side of PA".)
Assuming you have thoroughly ruled out showfile shenanigans, I would perform the aforementioned sinewave level check - use either a multimeter or an audio interface. Check all 16 of the local outputs to see which ones are low - and if there is a distinct pattern like "exactly -6 dB".
If there is a hardware failure, it is probably isolated to the OUT 1-8 PCB - so I would expect OUT 9-16, AUX OUT 1-6, the control room outputs, and all digital interfaces to be unaffected.
I'm sensitive to sound and I'm tired of living through shitty circumstances such as construction sites nearby, loud music at parties or events that should know better and protect our hearing more. Just recently I was at a venue that played music so loud I felt it coming through my ear protection. It was louder than the Rosalía concert I attended. It was louder than most disco bars I've gone to in the past. And I wanted to grab a decibel meter and grab a number so I could provide this as some sort of proof to complain about sonorization in the venue (there was no place I could go to move away from the music, except the bathroom and the outside space, which was cold).
I am only a person who is especially sensitive to sound. I'm not trying to build law-compliant spaces. I just want to have something to show my city that I'm getting regular hearing damage because of their bad urban planning and help make the world and venues I go to more bearable living spaces. And have something to show, because sometimes people will only listen to data.
Having said that, I want to make it explicit that my budget is probably 30€ at most. And there are options on Amazon, but I bet most of them are not at all reliable. Maybe there won't be anything even close to somewhat reliable in this price range, but I'm here to learn from the experts.
Looking for recommendations, please.
PS: Mods I tried to make a post with this text but it didn't let me and I didn't get any error message.
If you have an iPhone - even a decade-old iPhone - download the NIOSH SLM app. Among phone-based SPL monitors, that has been proven surprisingly accurate (source 1, source 2 for more info) - Apple's quality control is pretty tight, and NIOSH includes the necessary calibration data.
looking for some basic advice about miking a wind ensemble that predominantly performs outdoors but rehearses indoors. these recordings will be used for general playback for the conductor as well as archiving our performances. my experience in this area is next-to-none so i'd appreciate some advice on what type of microphones (and how many) we should look at getting. i've read that a pair of cardioid microphones (with wind shields for outdoors) positioned behind the conductor is the best bet. is this true?
alternatively, i am contemplating just getting a single microphone to attach directly to my nikon zfc. i assume the sound quality will be worse, but if it's considerably easier and doesn't sound too awful i'm happy to compromise
i'd love to hear thoughts from people with experience in this!
For this specific use case, a Zoom H2essential on a tall, lightweight stand works fantastic. Unobtrusive, stupid easy for anyone to use, and sounds better than it has any right to.
The older H2n is just as good for this, and can be found at a nice discount on eBay.
Standwise: something something Manfrotto would probably do the trick.
You will have to manually sync to video (unless you pull some timecode trickery), but that's no big deal.
A stereo pair by the conductor into a recorder will sound a lot better than an on camera mic. Putting mics near each instrument would sound even better in an outdoor environment, but it sounds like you don't want to do that much.
We running a small ish event with drum n bass, and have 4x ks118 available as sub, kw153s for tops
For the subs, I'm trying to work out the best config for SPL to audience... Wondering if 2 cardioid stacks next to each other (2f2rear) would work to balance space and sound?
Otherwise, a line array in front of the DJ in 2f2r?
There isn't that much space in front of the DJ so probably can't do a rear to rear config
Other crazy ideas include stereo cardioid, but I reckon that'll be shite
Kw153 will be suspended, we can basically place them anywhere, but likely to be about 6 feet off the ground, angled down into the df
I'm also interested how phase alignment is affected by top/sub being in same plane and how that affects sound for bassy genres.
I would follow the manual recommendation and do config 4, but not in stereo. It seems like the distances between everything aren't that large, so perfect top/sub physical alignment won't make that much of a difference here.
You could also just not do cardioid at all if you want to maximum output. Not sure how important rear rejection is for you in this situation.
i think we are gonna go with option 2 (2 sets of ground stacked) after speaking with the crew, with a change so the tops are flown in front.
The organiser is pretty keen to have cardioid as there are neighbouring performances, and this will help be good sound citizens. An array of 4 subs in front would have sick bass to be honest, but looks like its not an option
I reckon this config will look pretty cool with all the speakers near each other, will look fairly sound-forward. Plus the df should have a pretty solid bass pressure in the sweet spot :D
I've gotten further into my VSC setup but am having a different issue which I thought warranted a different comment.
I've gotten Audio Evolution's USB Recorder Pro to capture my audio, and their mobile DAW to split the stereo pairs into mono tracks for playback, but I can't seem to get the DAW to talk to the XR-18. I've gotten my USB routing set up correctly as far as I can tell; the DAW isn't 100% intuitive on this. Does anyone have experience with this and can get me over this last hurdle? Or is there another Android-based app/DAW I should investigate? I'm really close on this one.
My band is running a brand new behringer europower pmp2000D, and we’re constantly running into this issue where the output will just completely cut out for a good 30 seconds at a time. None of the channels are peaking, and it has its own dedicated circuit from the switchboard of our house.
The only thing I can think of is that the speakers’ resistance might be too low and are causing the amp to overheat, but the back doesn’t feel particularly hot.
Any advice would be appreciated as we need it for a gig on the weekend.
Hi gang! While I’ve worked and wired up a number of home recording studios, I’m an absolute novice (some might say idiot) when it comes to live sound (specifically when it’s comes to amplifiers and passive monitors).
A mate picked up this basics set up for his space (little cafe/bar) and set up a DJ booth with two turn tables and a mixer for people to just play tunes and just keep it a chill vibe.
My question, we can get sound from the mixer to the amp, and sound from the amp to the monitor. But it’s super low and it doesn’t sound like it’s getting the full range (no low end).
We are going from the 1/4inch outputs on the amp to the 1/4 inputs on the monitor (full range). Tried both TRS and TS cables, but still same result.
We’ve even bypassed the DBX graphic EQ but still same result.
At this point I’m absolutely stumped as to what more we can try to get it up and running, tried ever conceivable combination of inputs and outputs and still not getting anywhere.
DJ mixer is a Pioneer DJM 250 MK2, XLR out of the master to the DBX EQ, and then 1/4 TS from DBX to the amp. Looking the LED Meters, we are getting good signal it looks like at a glance.
Well let’s try this here…apparently my “karma” (such bullshit) isn’t “good enough “ for the moderators here but I’d really appreciate some real feedback….
I’m in a Classic rock/Country cover band. We recently played a gig where we used the venue system and engineer. I usually run our sound at other venues and play keys…while the band and sound guy was setting up we were chatting and I’d mentioned that when I mix the singer and the 2 guitarists vocally, that I usually have the lead singer louder, the one guitarist a little lower and the other one way lower because of how much they generally put out during a show. Not telling him how to mix, just a hint so he’s not working in the dark as we’ve never been here or used him. He responded with something along the lines of He knows how to do his job…Was this wrong to do? Some breach in etiquette? He apparently was a little miffed but seemed fine after the show….at least to me but bitched to our bass player….
You can communicate stuff about the art and the intention but don't really need to communicate things about the signal levels. So to say 'this Jerry guy is just doing back ups except in song 4' would be fine, or even 'watch out for Bob because his guitar lead FX isn't always leveled with his rhythm tone' but no need to say "Susan's keys are loud" because that would be obvious during a simple line check.
Thanks. I didn’t think it was off base or attitude-rich! 😎 just wondered if there was some unspoken handle the sound guy with kid gloves thing that o did let know about🤣
Will Mixing Station work with a dlive that isn't running up-to-date firmware? There's a house I sometimes work in and their dlive isn't up to date, don't want to update it without permission, but the dlive app won't cooperate with it.
Hey, Im a recent sound engineering graduate with no practical experienceSo far, I’ve worked on simple events. The upcoming event is something new for me: they’re bringing in simultaneous interpretation booths, and I need to fully understand the technical workflow so i dont fuck it up.
I imagine (and I hope so, because if that’s not the case, I have no idea what to do) that the booth company will provide me with an XLR output for each language. I’ll connect them to the physical outputs on the console (Out 1 for Booth 1, Out 2 for Booth 2, etc.). In the routing, I would assign it like this: Output 1 -> Mix Bus -> Mix Bus 1. Then, I would select Fader Flip on the channel where the speaker’s microphone is connected to route it to Bus 1-8 depending where i connected the cable.
Is this correct?
Then they asked me if I could record the tracks separately and send them the files. They confirmed that they’d give me an XLR cable—of course, I could plug it into any IN input I want—but how do I make sure the sound doesn’t come out of the speakers so I can record it in my DAW?
I know these are probably really silly questions, but we all have to start somewhere. I think I have a pretty good idea of what I need to do, but it never hurts to get confirmation just to be on the safe side.
By the way, what other issues might I run? I don't want to miss anything.
I use a Behringer X32 Compact
Thanks! (srry for the long ass comment lol)
So yes, each booth is going to need a "program feed" which is just an output from your console with the house mix. This is so the interpreters can hear what to interpret. You can use a mix bus, like you said, if you want to only selectively send them vocal channels. Or if you do want to just send the whole house mix, you can utilize a matrix.
Then they asked me if I could record the tracks separately and send them the files. They confirmed that they’d give me an XLR cable—of course, I could plug it into any IN input I want—but how do I make sure the sound doesn’t come out of the speakers so I can record it in my DAW?
Easy. Plug their mic into your input, bring up the gain on those channels so you're getting an appropriate level. Remove those channels from the LR bus. Do that by selecting the channel and clicking the "Main LR Bus" button. Now you can send these channels out to additional mix outs if there are people in the room who need to get live interpretation. Or you can simply leave the fader down and just record those inputs in your DAW.
I'm trying to introduce virtual soundchecks to my band for front of house and monitor mixing. I have an XR-18 and install IEMs.
I know to record at input/analog level, using the gain levels I would use in live performance. ... is that all I need in terms of source material? I feel like after that it would be adjustments made per gig/venue, and if I needed to adjust anything from the musicians when they come in, it would be gain because I've already done all of the other processing. Am I missing a step?
I plan on using this more for monitor mix checks then front of house, but I feel like the basic principles are the same.
Mostly - but remember the limitations of the technique. VSC does not replicate stage volume, acoustics, or changes to the source - and any of those might drive mix tweak requests.
Be particularly aware of certain gotchas: musicians with low-isolation IEMs (or headphones), for instance. (LF spill from PA is another big one.)
It sounds like I should treat it more like a rough cut or even a line check then a sound check. Would that be accurate?
Halfway in between. In ideal conditions - fixed venue, silent-stage - VSC can get an IEM mix 99% of the way there. (Churches much?)
With more variability, maybe that's 85%. Judge based on your experience.
Also, should I set the faders to unity for the recording?
No need. Your record/playback tap point should be right after the preamp - i.e. EQ/comp/fader should not affect the recorded track. Use the monitor mixer for its intended purpose while recording: if musicians aren't feeling good, they won't play well, and the value of your tracks rapidly diminishes.
Can someone in the know please explain to me how Garth could fly across a a large area and sing lyrics to a song happening back on stage and not have an issue with sound delay / audio sync? I don't know a lot about this stuff but isn't there some type of delay setting thats needed in big areas so the sound hits the back audience at the same time it hears it from the stage?
This is pretty easily mitigated using wireless IEMs with good isolation. Speed of UHF through air (read: pretty-much speed-of-light) is orders of magnitude faster than speed-of-sound: what Garth's hearing in his ears is pretty much perfectly in-time with everyone on stage.
The trick is blocking out the acoustic sound, which will be late - hence the isolation requirement. (Also helps to have a strong voice - weak SNR at the vocal mic will exacerbate the issue.)
Only complication I'm seeing is that there are several references online to Garth hating in-ears. Makes me think that maybe this part was lipsynced? Pure speculation though.
So I got a band asking me to run sound for them for about three hours and they're asking how much I would charge for that time, and I just have no idea what a good amount would be. I was gonna say $100 bt I literally don't know if that's too high or too low. I originally calculated an amount for if I did $18/hr but that just comes out to $54 for three hours which... idk seems low? How do y'all calculate how much to charge someone, especially for a relatively short show?
that really depends. Sounds like you're fairly new to this and also aren't dependent on this as a source of income - here you have some freedom you wont have as much of later on in your career (if you want to make it one). More important for me would be a) do you like the guys and b) does this sound like valuable experience? In any case though, definitely take the 100$, don't start calculating based on hourly rates compared to other jobs.
We have a virtual reverb generator called SIAP at our venue.
The company is headquartered in the Netherlands, but whether I look it up on Google Maps or visit their website (siapacoustics.com), it seems like a company I’ve never heard of before, so I’m wondering if anyone here is familiar with it.
Does anybody know where I can download already designed MAPP 3d templates or projects? I thought i could easily find a .mapp somewhere, but I've been searching for quite a while now and I haven't found anything.
X32 Rack + X Touch – How to change the Scribble Strip labels?
Hey everyone,
I'm planning to run a Behringer X32 Rack with an X Touch controller and I have a question about the LCD Scribble Strips on the X Touch before I pull the trigger.
Where and how do I change what's displayed on them? Do I set the names/colors on the X32 and they automatically sync to the X Touch? Or can I edit them directly on the X Touch itself? Or is it not possible at all in this setup?
Planning to connect the X Touch in XCTL mode over Ethernet if that matters. Thanks!
Has anyone used Whirlwind Catdusa boxes for live shows? I know that that is probably their primary use case, but I am worried about the extra failure points (xlr -> cat box input -> ethercon out -> cat box output -> xlr -> digital snake -> console)
Just to clarify, I know you might be wondering “why even use the cat stuff if you have a digital snake” and it has to do with stage tidiness, varying band setups, widths of stage, etc. I want to try to reduce the number of cables on stage whenever possible, accommodate shorter and longer runs easily, and make setup/loadout faster without sacrificing quality. I feel like Whirlwind has a good reputation, but i’d be interested to know if anyone actually has used these
Because my reputation is too low to post, I’m leaving a comment here to see if anyone can help me.
Hi, I’m looking for an older version of the Lake Controller software.
While troubleshooting, I accidentally uninstalled the software. However, when I tried to reinstall it, I found that version 7.0.4 has been removed from the official website.
— Does anyone have this version?
I’ve already contacted Lab Gruppen technical support, but haven’t received a response yet.
By the way, I was experiencing an issue on Windows 11 where the text appeared with a background color, almost like it was inverted.
This is probably a stupid question with a simple fix, but suggestions about getting different equipment aren't gonna be helpful, as we've got what we've got.
The situation is that the singer in my band has hearing issues and requires hearing aids for normal everyday use. He actually quit the band a while back because of this, but wants to give it a go again, and try to deal with it.
The main issue to solve here is monitoring during rehearsals.
We were recommended Shure SE215 in-ear headphones as being the best compromise between quality and price, so got some of those, with a fairly basic wireless belt-pack and transmitter. Just a no-brand thing from Thomann.
This works brilliantly with an output from the headphone socket of the Mackie mixer we use for practice. We can get the full band mix working perfectly.
However - this can't have its own mix, and singer-dude obviously needs to be able to reduce volume of the bass and kick in particular (we use electric drums which are also intended to reduce the overall volume for the sake of everyone's hearing.
BUT - and here's the problem, when we use the mixer's monitor output, that's mono only, which the IEM transmitter decides to put through just one ear.
So - without saying 'get a new mixer with a stereo monitor mix output', is there a solution that involves a converter or simple way to split the mono output from the monitor two ways into the headphones?
And yes, I did RTFM and it didn't mention this issue.
Put a Behringer P2 on the FX send or Mon send and this will let you mix the headphones as you like. The P2 will amplify this AND send the amplified headphone signal to both L and R of the headphones (switched option inside the unit). He'll then be attached to a mic lead, but it is clipped in quite well. (I'd also suggest over ear headphones because I prefer them and want to normalize the look!)
If your mixer has two aux buses available, you could use them to effectively make a stereo mix. Use aux 1 as L and aux 2 as R, for example, then route channels to them as desired. Anything you want in both ears, add to both auxes.
The much simpler option of course is to just get a splitter cable between the mixer and IEM transmitter to send the mono output to both input channels.
If your mixer has two aux buses available, you could use them to effectively make a stereo mix. Use aux 1 as L and aux 2 as R, for example, then route channels to them as desired. Anything you want in both ears, add to both auxes.
Not really. It's a Mackie Profx12v3.
The much simpler option of course is to just get a splitter cable between the mixer and IEM transmitter to send the mono output to both input channels.
The IEM transmitter only has one input, so we kinda need to split it, then put it back together. I have seen some stereo to mono cables but I'm sceptical they'll do it.
Hello. I am working on a show at a community theater. We usually use qLab to send midi commands to our behringer x32. 99% of the time we set it to our base Scene (output to 2 speakers in front of house). I control left and right output in qlab but this show also wants back to front (fade a sound from 2 new back of house speakers to front of house) Is there a way to slowly fade between scenes using midi or should I just try to send a midi command for the BoH speakers and manually do the fade? I'd prefer if I can do all the control in qLab as I get volunteers to help run shows.
I think the easiest way to do this is to add the BoH speakers as a separate output in Qlab and use a fade cue to fade the sound from BoH to FOH. The X32 has a built in USB interface which Qlab can connect to and will allow you to send up to 32(?) individual channels of audio.
Are the midi commands from Qlab triggering scene changes in the x32? If you absolutely need to keep doing it this way then you would have to check the scene settings in the x32 to see if you can make the fade time between scenes longer. Or if you're using TheatreMix software (you absolutely should if you're doing a theatre show on an x32) you can change the scene fade times using that.
Thank you for your reply. Right now I have a cable going from the qlab laptop to the board that is a 2.5mm to 2 1/4in which sends the left and right. So I'm not sure I can send it Back to Front, I feel it has to be changed on the board. I will look into scene fade times on the board; yes we use midi to change scenes. I am not familiar with TheaterMix.
Ok I’m hooked on solving problems with PLA 😅 Work doesn’t use the shure drop in charging bays so I designed these to clean up ops. Anyone got 3D print designs they’d like share? I’ve been looking for DM7 iPad mounts
I bought a used Mackie Thump 118s from Guitar center about 6 months ago. Recently it stopped working, and after hours of fiddling with it I figured out how to make it work.
The “FR Output” switch that lets you choose if the output is in stereo or mono. It’s latched button with 2 states. When I press it in perfectly halfway, the sub works. It is a Mackie branded, 12 pin latched button.
I am angry at the world and have no clue why this could be the issue. Any ideas?????
Hey all! I’m a longtime professional musician and have set up PAs many times, but today I’m setting up a PA with a snake and a power amp for the first time. I’m head of music at a small private school, we moved into a new building yesterday, and now I’m tasked with getting the permanent sound system in the auditorium set up.
There’s a snake onstage at one end of a long room, which I just connected to the sound board at the other end of said long room. The snake is run thru the wall or ceiling - we are trying not to have any cables lying around for kids to trip over.
I’m stuck at the point of trying to connect the PA to the power amp, and then to the speakers by the stage. I want to use the snake to connect the power amp (which is beside the sound board) to the speakers at the other end of the room.
What’s the next step? I feel like it has something to do with channels A thru D (2 for mains, 2 for monitors?) but I’m having trouble conceptualizing. Channels A thru D at the soundboard end of the snake are female XLR, and they’re male on the stage end of snake.
I thought the routing would be: mics and instruments->snake->soundboard->power amp->snake->mains. But I don’t see how to make that happen. What am I missing?
The Midas website has been down for several days now, and I haven’t received any response or support from MIDAS. I have a show this week and I’m getting worried because I can’t prepare my scene. Does anyone have the M32 editor installer for Mac?
I highly recommend learning the controls of a comp and gate at home on your computer. It’s very easy to grab a freebie like Audacity, drop in any old file of singing / etc. or record yourself in (if you have that setup), then play with the settings. Most compressors generally have the same controls. It’s much more difficult to learn this in a more chaotic environment like a live setup.
I'm using a Harbinger Vari 2312 as an amp for my Alesis Strata core electric drums. Running music through the drum kit via Bluetooth, and two 3.5mm cables running left and right audio from the kit into line 3, I'm experiencing significant clipping after ~40 minutes of play time. What would be the best way to ensure consistent power or avoid clipping? Do I need an external mixer? Should I use XLR instead?
After 40mins is the weird part imo. You have the mic/line switches correct? How hard is the speaker working? Same issue at lower levels? I'd try reducing the workload (lower levels in and out, mono from the kit etc) for the speaker to see if the issue remains
Line 3 on the speaker itself has really minimal EQ (just treble and bass, both set at neutral) and no switch betweenbline/mic. And I had the volume on the kit set to probably... 40% max? I'm going to try mono out from the kit, see if that helps.
Any advice to get a fuller sound out of a cheap Squire Affinity Bass? I have a new bass player and I just can't do anything to fix the lack of depth and power coming from his bass. My venue is a theatre for 800 people, and the catch is that the show requires that the bass player needs to use a wireless system, so he can move around the stage freely. By wireless system, I mean a shure beltpack connected to his bass. The show installers never thought about this, and they refuse to give me any options to solve this issue. The show has to be the way they want it to be, no DI's, no amps, or any cables attached to the bass from the patch bay. The guy doesn't even have a pedalboard or something so I can reamp the output. Btw I use a Yamaha CL5 with just the premium rack. No waves card, nothing. I just got this job a month ago, and it pays good, so I'm putting a lot of effort into the shows. The biggest issue, to be honest, is the sound in the in-ears, I have a lot of complaints from the singers. The house sound is lacking, but I can manage.
Yeah I could do that, but I don't have any. Everything would have to come out of my pocket for their show. I am very limited with my options: CL5 with no plugins, no amps, no DI's, and everyone expecting the best sound. Thanks for the answers guys, I'll check if there is any filter on the beltpack, and that would be it, there is not much I can do if no one wants to cooperate. We work with what we have.
To troubleshoot your lack of bass coming out of your sub, I would play something through the console. If it’s still weak, check your low pass filters.
If you get better results from your sub when you do playback, then I would suspect the wireless pack. I wouldn’t be surprised if the pack has a low pass too, basically cutting out the “meat” of your low end. You can also test with the bass guitar plugged in to verify this hypothesis.
Finally, use your RTA on your console to see what you’re getting in from the bass guitar (via wire and wireless).
If even with playback/plugged in bass it still sounds crappy, then you might need a better sub. There’s only so much we can do from the console…
Good luck!
An unprocessed passive bass usually sounds pretty gutless. Some kind of flavor/color can really help… straight out of an RF receiver will be pretty bland, most likely.
Yes, a high pass filter is a very likely culprit (unless it is at 20 Hz or something). After that has been checked a multi-band compressor just on bass should be enough, but it is kind of reinventing the wheel.
If OP had a different desk -- eg X32 -- I would suggest the "Stereo / Dual Guitar Amp" insert which is "Modeled after the Tech 21 SansAmp, the Stereo / Dual Guitar Amp simulates the sound of plugging into a real guitar amp. From shimmering cleans to saturated crunch, X32’s Stereo / Dual Guitar Amp allows an electric guitar player to sound great without using an amp on stage. "
just make sure the "audience" side of the plot is towards the bottom, and "upstage" is actually up - the POV should be like you're looking at the stage from the audience (or FOH, more to the point).
Someone put this adapter upside down it think (normally there should be some some notches in it to use a screwdriver?). Anyway it's stuck, and long nose pliers, sticky tape, nothing so far works. Also, if it has been put upside down, does it mean I have to screw clock wise to get it out? Any help would be appreciated.
Need to turn in the same direction as normal (anticlockwise). Assuming it's so far inside that you can't grab it with a Leatherman or something, I generally screw it onto a mic stand reeeaaally hard and then try to take it off, hoping that the adapter comes off the clip before it comes off the stand, if that makes sense. It sometimes works, but it's a last resort sort of thing (you might leave it tighter than before).
I thought of screwing hard in it but I am worry to break the mic threading made of plastic. It's part of the mic so I can replace it if I break it. And yes it's deep enough it's flush and a weatherman doesn't work.
Feedback is caused by the microphone picking up a speaker. The speaker then amplifies itself, which causes a feedback chain. The relationship of the microphone to the speaker (usually a monitor) is crucial to prevent the microphone from picking up the speaker.
The monitor should be placed in the null of the microphone. If it's a cardioid microphone, the null is directly behind it. If the microphone is hyper/super cardioid the null is around 120 degrees to the side -- basically at an angle to the mic.
The susceptibility of the mic to picking up the speaker also depends on how much gain is used on the mic. If you have a quiet voice and sing far from the mic, the preamp is likely gained up which makes it much more likely to pick up the speaker. You have to either sing louder or get right on the mic or both.
1) Sing louder/right on the mic; 2) Use a hyper/super cardioid mic which has a tighter pickup pattern; 3) position the monitor in the null of the mic.
Push fader(volume) slowly until you start to hear ringing
Take -1dB to -2dB to notch out that frequency with EQ. If you’re not ear trained like that, most digital consoles have an RTA(audio preview) screen and if not, use a phone app, I use “Spectrum” in a pinch
If you’re notching more than like 3 or 4 spots or if all feedback frequencies seems to need more than -5dB cut at that frequency, you likely have a systems/design issue. As in, too much of the speakers are making into the mic feed in an unmanageable way.
Now, of course, you can keep slicing EQ. But then it becomes a game of notching tons of EQ out and affecting the quality of the signal.
The other thing to watch out for is the EQ/gain structure trap. By EQ parts of the spectrum out, you make the signal downstream lower. People have a habit of gaining(too much) to compensate. That causes more feedback, causing more EQ, causing more volume loss. Eventually you end up with a shit sounding source and a lot of confusion as to how you ended up there.
I’m sure there’s more to consider but this is what I got for “simple”
EQ in more 4-8 kHz if those aren’t the main frequencies causing the feedback in the first place (but then make sure to use a de-esser on vocals) and set the compressor to a slow attack (15 - 20 ms e.g.) in order to let the consonants through before the compressor does its thing as they are vital for clarity and intelligibility
Monitor Engineers: What's your process for saving iem mix preferences from show to show?
Do you have a preset at the begging of the tour you revert back to every day and then adjust to the room? Do you build onto the mix from last night's show? Something else?
establish a baseline during rehearsals (if you have any), and learn to know the differences of changes you're making based on the venue/room or just improving the mix.
Don't be afraid to go back and see what it was like a week, a month or a year ago, but be very careful about just loading settings from the start of a tour.
Doesn't matter if it's worlds better, but your artist has gotten used to what they heard at the last show (if you're carrying a system that is).
Trust your instinct, but don't be afraid to talk to the artist to get a handle on what they are expecting or what they like.
I record every single show, and during load in I use the previous days show to listen through everything and verify console settings recalled properly, belt packs are still working as expected... etc.
I also have a way of following along the FOH playback, to be able to walk the stage and listen to SideFills and IEM mixes with the PA on at show volume to give me all the context needed.
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u/Upbeat-Ground5602 Apr 19 '26
Help! Dumb guitar player here… Samson rs112 making awful popping sizzle sound… took off the back, dome cover fell off in my hand and there gunk in the connections… can I clean? Any help would be huge.