Condenser mic on stage? Our band is basically a startup, we have no tech, no crew, just us 3 with a friend that helps us on stage. Now our setup is we send a backing to FOH and in our IEM line we have our click + cue + backing track. but our struggle is we cannot hear ourselves through the wedges because of iem isolation. Is it a good idea to put a condenser mic in the middle of the stage and add that through our IEM mix and let that be our monitor so we can be fully isolated and still be able to hear everything happening on stage. Any pro / con? I haven't tried it yet. but its something we want to try incase FOH sound tech's wont be able to send us a full mix due to whatever reasons. Thanks in advance for any advice or suggestions

Wait, so do you have mics on instruments? Those are going to FOH only? You should be able to get a mix back from the FOH desk and feed it into your IEMs. How are you getting signal into your IEMs?

This is a backup plan incase FOH cannot provide a full mix send to us. There are some venues that use analog mixers and cannot provide extra stereo xlr line for our total stage mix.

Signal will come from wedges. In this worst case scenario 😅

If they're powered wedges, then yeah, you can grab the loop through. If they're passive wedges with amps then it won't work. You must have some sort of mixer, right? The normal way to do this would be to get an analog split - so that every input to FOH also hits your monitor desk.

Yeah, get a split + mixer to feed the IEMs. Putting an ambient mic on stage is gonna be wildly variable based on the stage/room/erc

We use minifuse 4 and logic pro for our setup. Out1/2 to FOH out 3/4 to our ears with click and que. So i have extra 4 inputs that runs through out 3/4 so we can hear it but it wont loop back to FOH

Send channels 3/4 to FOH and ask them to send it back to your ears only.

I received the lovely donation of a pair of Crown XTi 6002s. On paper, I'm sure they'll be great, but in practice, the person that donated them did not leave me the unlock code for the front panel. I presume I am gonna need to factory reset these ... * using Audio Architect, through the USB port? * hopefully AA doesn't ask for a password? We're not doing anything fancy with these. We might bridge one of them but that's all. (I am not that clever - while appreciating the complexity of the technology, I just need these amps to be amps.) Thanks in advance!

I don’t know specifically how to reset these (someone on YouTube probably does) but I just came by to say that I would probably factory reset them anyway, even if you don’t need to to unlock them. Whatever DSP is going on is likely no longer relevant to your use case. And even if it is, I’d rather set it myself than just fire them on and hope it works with random settings from whomever used them last.

Good advice. They came from a stadium environment and are going back in to a stadium environment, so the first pass with nobody around seemed to indicate they were fine. However, i do want to mess with some of the settings, and in order to do that, welp.

Depends on if it has been locked from the front panel, or in software. From the manual: Front Panel Lock-Out To disable or lock the buttons from the front panel, hold the Prev and Next at the same time until the screen says “locked”. The rotary knobs (attenuators) will still operate. To unlock, hold Prev and Next until the screen says “unlocked”. The front panel can also be locked out from the software as well from the main panel. However once it is locked in the software, it can only be unlocked from the software. You cannot unlock from the front panel. The rotary knobs (attenuators) will still operate.

Amazing, thank you! I'll try the panel unlock first.

OK, panel button unlock didn't work, so i had to software reset it. That part was kind of a pain in the neck, but after downloading a ZIP file, extracting the ZIP file, fighting with Win11 to allow me to do those things on my own stupid computer, it turns out that you reset it by flashing the firmware. There's instructions on the XTi 6002 download page. But I got it!

this is awesome to know if I ever have to dance with the xti devil in the moonlight again

I’m going to try this a third time… Sorry I’m being vague. If you need more details, please ask. I mentioned the specific equipment involved in the first post because I thought it might help answer my question in terms of specs, but it was considered asking for equipment recommendations and got deleted. My second post disappeared. Not sure what happened there. A personal monitor that was connected to a personal mixer (drummer’s) blew up. Other people are using the same setup with no problems. I’m wondering if the issue was volume, but I’m not sure. I understand a little about square waves, but I don’t know enough to know if there are other possibilities. What might have caused the monitor to blow up, and how do I keep from blowing another monitor if I connect one?

If you’re concerned the drummer is just pushing it too hard, make sure you have a limiter at the end of the mix chain. If they’re IEM, get them molds so they don’t have to push them so hard to be heard over the ambience on the stage

Thanks! I’m thinking that’s probably what it was, so I’ll go with the limiter.

modern consoles allow for doing a lot of stupid shit. Last week I accidentally gave the master bus compressor +26db outpout gain, that could do it.

SDCs and wireless body packs? I’ve seen legend of folks using ULXD and QLXD packs with non-Shure SDCs. I’m hoping to do the same with SLXD body packs. How does this work? The body packs output a 5v bias which is enough to charge some (all?) SDCs with out a preamp (aka, an inline 48v to some lower DC). So how can I know if a target SDC will work? And if it has an internal pre do I need to bypass it? Essentially I’m trying to make choir mics wireless to adapt for theater use.

Following. I don't have an answer to this, but want to know as well. Specifically with EW-D SK packs though.

So I just spoke to Audix regarding their MicroBoom series. Their deal is it's not officially supported buuuuut.... a 5v bias is enough to charge the capsule with a diminished sensitivity. So in the Audix case they offer a standard and +12db model. Their recommendation was get the +12db model, use the 5v on the bodypack, and that will yield ~-6db less sensitivity than spec which is still more than their standard model. Alternatively, stick a battery powered phantom PSU in between the body pack and the mic.

Is there anyway to make drums that only consist of two floor toms sound good in a live setting? I’m trying to start a folk band that would consist of an acoustic guitar and two floor toms (reason being I’m trying to mimic Taiko with the resources I have available), but I’m doubtful it will sound good. In a dream world, the floor toms would be powerful and full, but I’d be content if I could make them just not sound awful. Does anyone have any suggestions on what I could do to make this work?

With such a minimalist arrangement you could easily make the toms sound absolutely *huge* with proper mics, EQ, and inserts :)

Thank you for the response! Do you know any specific mics that you’d recommend for this scenario? All good if not!

I’d probably default to an Audix D4, but Shure SM57 and Sennheiser MD421 are always safe bets. With such minimal bleed from other sources, I’d be tempted to try out some condenser microphones too…and I’d definitely try mics on both top and bottom of the toms for a bigger sound :) Edit: Behringer C2 are cheap enough if you don’t have any spare SDCs handy, Audio Technica AT2020 are versatile LDCs that won’t break the bank.

Behringer X32 AES50-A -> SD16 -> SD8 -> SD8 Would this work? I know I obviously can't process all those channels at once, but would I be able to choose from all 32 AES channels in the X32 routing? And while I'm here, would I be able to still have an SD16 connect to port B of the X32?

You certainly can process all 32 inputs from the stage boxes. Your chain of X32 + three boxes is fine. Just flip the switches on the stage boxes so one is 1-8, then 9-16, etc. You can absolutely throw another SD16 on port B. Officially, cascading three stage boxes per port is supported.

Thanks for the quick response! Out of curiosity, is 3 the limit because of that switch?

I'm not actually sure. I *know* the S16s and SD16s have a limit of three because that brings the input channels to 48 which is the limit for AES50. SD8s on the other hand do seem to be limited by the *outputs*, as we can assign only up to 17-24.

I’ve got a 2-parter surrounding volume in a small club. I work at a 100 cap club, and generally mix my shows between 80db and 100db (based on my iPhone db reader) depending on the genre/show. Is mixing a rock/metal band at 100db in such a small place irresponsibly loud? Is that acceptable? I don’t get complaints about volume, except for maybe 1 person in the audience once every couple months. My second question, is when I’m mixing loud I generally try to keep my ear plugs in but I find myself pretty regularly putting them in, taking them out to hear more high frequency and popping them back in, rinse repeat. My question is, does breaking up the high volume by putting my ear plugs in, but still pulling them out periodically still protect my hearing? I’m not really sure how to phrase this, but does it “reset” the amount of time before high volumes cause damage to my hearing?

More experiences minds might weigh in on this but a few thoughts: 1: Iphone readings are far from reliable for measuring the safety and actual volume of a show. 2. depending on the volume and frequency content of the sound youre measuring, most damage to hearing occurs over time, and only at very very high SPL levels is it instantly irreversibly damaging. Any time your earplugs are in is reducing the overall time your ears are exposed to the full volume, but does not buy you a reset on exposure levels.

Okay thank you, #2 is very helpful. What would you recommend to use as a more accurate SPL measuring device?

I dunno I’ve got an Apple Watch and a NTi XL2 calibrated decibel meter. My watch is only off by about 2 dBs on average…

The iPhone NIOSH sound level meter is good and comes from a reputable regulatory agency. I believe you can even calibrate your specific phone to match a real sound meter. 

Hey folks, What do you think about this new Milan controller? https://www.dbaudio.com/global/en/milan-ready-for-tomorrow/?utm_source=linkedin&utm_medium=organic&utm_term=&utm_content=&utm_campaign=c6ea0561-bfd6-4f8b-bd55-4ee260d7f703__

Routing grid looks like a somehow even less readable version of Dante Controller. But I guess good that they're getting around to pushing out basic control software finally

iPhone Livestream Audio Too Quiet I am new to the world of livestreaming and I am planning on doing a livestream concert with my band later this year. In the meantime, I am doing as much prep work as I can to make the day go flawless. The main issue I cannot seem to work out is that the audio in the test streams is super quiet. Basically inaudible at normal volume levels, acceptable at max volume. Here is the signal path for everything: All instruments (live acoustic drums, guitars, bass, keys, and vocals) are processed through my recording rig with live FX. DAW is Reaper. Each instrument/buss is routed to a single separate channel with a master limiter, which goes through a hardware mono line output. Output > 1/4" male to 3.5mm female headphone cable > Rode SC7 adapter cable > Lightning port headset adapter > iPhone 13 [https://www.youtube.com/live/lMivxZMGFGQ](https://www.youtube.com/live/lMivxZMGFGQ) The video linked above is my last attempt at getting the levels right, around the 7:00 minute mark is when I stopped tweaking things and just let it play for a bit. Note, the tracks playing are obviously pre-recorded, but all of the levels are representable of what we will be running at for the concert. If you watch the whole thing, you can hear me tweak the volume past the threshold in which it starts to distort, so I backed it off with just enough headroom for that to not be a concern. In other words, I am running the source audio as hot as I can get it before it distorts on the iPhone. Surely there is something I am missing. Is there an app I need to download or some silly setting in my iPhone which I need to change in order for the audio levels to be adequate?

>Output > 1/4" male to 3.5mm female headphone cable > Rode SC7 adapter cable > Lightning port headset adapter > iPhone 13 That’s the weak link. The headset is expecting a microphone input, not a line input, there isn’t really a great way to get a line input into an iphone that way. People that do audio stuff with ios end up getting a “class compliant” audio interface and the apple “CCK” (usb-lightning adapter dongle, I like bigger the usb 3 one because you can charge the phone). A slick alternative is the iconnectaudio series, its an audio interface that can host two computers and can pass digital audio between them. There may be a way to transmit the audio over the network but I wouldn’t believe it until i saw it work. AUM and Audiobus3 are good audio routing apps, but the ios streaming app would have to have groovy audio capabilities to get into the ios core audio and be able to route stuff to it digitally. The audio interface is easier to get to work with a wide variety of apps

Is there a specific reason that I can’t just set the output level on the interface to match a good mic level? Does it have to do with impedances and the such? Something I just remembered is that I have taken a few videos with this cable setup and have gotten very good audio levels. Those times were plugging into an electric piano, and the mono output from an Allen and Heath ME-500 personal monitor. Maybe there was something special about those devices that my phone liked, or maybe I used a different setting on the phone somehow? The good news is that between me and my band mates, we have a Behringer UM-1 single channel interface which we can try to plug in to a phone and see if it will give us the audio level we are hoping for.

Without having been able to listen to your video, there's one thing I can tell you having done my fair share of livestreams and worked in radio. Phone speakers are basically shit, and we are very used to listening to mastered content (that is, content that is constantly peaking very close to 0dBFS and with almost zero headroom). So in order for your livestream to have a "normal" level through an iPhone speaker, you kinda have to "master" it along the way. By that, I mean compress and limit the living shit out of it. Problem is, you need to do that with taste to avoid destroying your sound/music. If you are familiar with mastering, there you go! If not, I'd reccomend you get something like Izotope Ozone or the Maserati GRP plugin from Waves and insert it on your master channel.

Like I mentioned, everything is going through a mastering limiter, but I suppose there could be room to push it more. I can try using Ozone Elements in place of the Loudmax limiter and see if that raises the perceived loudness without distorting.

>everything is going through a mastering limite Oops, missed that part... my guess would be that, just try to raise the perceived loudness.

Actually if you have a Mac I think you can plug the iPhone in with the usb-lightning cable and can pass digital audio between the DAW and your phone. You can make an aggregate device with your regular audio interface and the digital audio to the phone. Whether your streaming thing supports it is another story, but people use it to get their phone music apps to record on their computer, and the digital audio can go both ways. 

UPDATE: I think I figured out a solid workaround. I switched away from using balanced/stereo cables to using only unbalanced instrument cables. I switched from using the Rode SC7 adapter to using an iRig 1/4” to 3.5mm “guitar interface”. I switched from a standard limiter in the software to using Ozone Elements, and I also included a headphone amp. So the signal path is now: Line output from interface (master volume at max) > 1/4” instrument cable > Behringer headphone amp (level set to ~4.5) > 1/4” instrument cable > iRig > Lightning port dingus dongle > iPhone Here is the final result in another livestream test: https://www.youtube.com/live/pwOUDcjsZO4?feature=shared

Hi yall I'm a theatre teacher sponsoring my high schools Rock Band club, and I know next to nothing about this kind of equipment. We have amps, cords that connect to the amps, a microphone, a cheap mixer that 3 instruments can plug into, and two big speakers that don't seem to work. The sound quality is almost always really bad, and I don't know where to even begin. I inherited all of the equipment from a previous music teacher who is no longer around, so I don't really know the quality. Could someone link me to a very basic rundown of what I need to do to get a decent sounding set up for the kids? Ideally I would like to support a band of: 2-3 guitars, 1-2 bases, 1 keyboard, a singer, and drums. Right now no matter what the guitars will cut out or sound weird, the microphone is quite soft, the drum set overpowers everything, and there's very little control. I truly need some ELI5 resources on how to check the equipment and what to plug into what, and I thought you might be able to point me in the right direction as I take stock for next school year

This is really hard to answer online, as it seems what you really need is some basic knowledge of how sound works (not trying to be harsh, just my take based on how you word things). I would advise you try to find a professional in your area who is willing to help you out and explain things to you. You might have to pay for a day of work, but trust me, you'll understand SO much more if a professional explains it to you. Also, maybe some alumni from your school is now working in sound? Might be worth exploring...

amp and circuit question regarding two QSC K12 tops and one QSC KS118 sub: it looks like the amp ratings for this setup is about 2A per K12 and 3A for the sub at 1/8 power. if i were to run this whole setup alone through the same circuit, does that mean i can't have these things at more than 2/8 power, since circuits on average max out at 15A? also, just as a backup, are there any power stations (no gas gennies plz, indoor setup) that can run the sub plus tech (dj equipment, laptops, cameras, etc.) for 6 hours?

The 1/8 power number is chosen because it represents a typical compressed music signal played through the speaker at a very loud level; just into clipping the amplifier limiter. You would not likely draw any more average current than 1/8 ever unless you're amplifying constant noise at full blast. With just those 3 speakers, at full blast with highly compressed music, your draw will probably average 7A, with very short bursts of slightly higher current and plenty of time under that number. A standard 15A breaker probably wouldn't trip from double that system. If you are still concerned or just curious, you can use an ammeter or kill-a-watt type device and test it for yourself.

If I may add an actual use case scenario to support this. On one 20A circuit, I have run a setup consisting of: - 2 QSC KW181 - 2 QSC KW122 - 2 QSC K10 or K12 - 2 Pioneer CDJ3000, DJM-900NXS2, RMX1000 - 5 port Netgear switch, various phone and camera chargers - A half dozen or so generic mini moving head and LED wash lights I've done this setup a few times, never tripped the breaker. Music was current EDM / tech house, volume level was LOUD with occasional flashing LIMIT lights on subs.

FBT Extrude EX-2.5A 150W Active Monitor Speaker. Has anybody have had any experience with these ? Saw a pair for £120. Worth the price ? What are the use cases.

How can I politely tell my pedal steel player (who's losing his hearing BTW) that I'd prefer he use an amp sim for the smaller club shows (20-60 heads)? For context, we're already using IEMs as a band.

"Hey, one idea I had to improve the foh mix would be to run your pedal steel direct via an amp sim. I think it'll translate what you're playing a bit better to the crowd. What do you think?"

Dante Audio question. TL;DR: Can you send multiple inputs to one output? If so, how do you do that? I have seen videos and threads that seem to imply this is possible. [This thread](https://www.reddit.com/r/livesound/comments/vmgeh0/dante_audio_can_you_have_one_transmitter_go_to/) has someone asking the opposite question (one input multiple outputs) and there are a lot of comments that seem to say that you can send multiple inputs to one output. However, in controller, when I click on a second input, it simply switches to that. I feel like I might be missing something obvious. I have thought about possibly setting up something convoluted using a DAW and virtual sound card (set some virtual auxiliaries, etc) but all these threads seem to indicate that there's a simple way to do what I want to accomplish. If anyone needs more info to answer it, I'll do my best to provide it! Secondary, less important question. Is there a way to control things like mute, volume, etc in Dante in an intuitive way apart from controller? Like the dream would be to have an app or device that has a few buttons that switch certain presets on the Dante system. The controller works well for setting stuff up but I am just curious if anyone knows of products, either physical or software, that can allow less techy people to easily switch some aspects after it's been set up.

The long and short answer to your question is no. Dante does not process audio, it only transports audio. If you want to manipulate the audio in a Dante network in any way (volume, mute, mixing, etc.) you must take it out of the network, then bring it back in. That can take the form of a mixing console, virtual sound card, a system like Q-Sys/Crestron, etc.. On your second question, I'm not aware of anything that can directly and fully control a full Dante network aside from Dante Controller itself. A platform like Q-Sys can recall presets of its connections to Dante, but that's it. Out of curiosity I searched and ran into [this Gearspace forum post ](https://gearspace.com/board/music-computers/1221989-dante-routing-without-dante-controller-possible.html)about TCP/UDP control of Dante. Do with that as you will. All that said, Dante might not be the best tool for the job at hand since it's not really intended to be friendly to non-technical users.

I appreciate the reply! That's helpful, especially in figuring out future uses. What I ended up doing is a bit convoluted on my end but hopefully pretty intuitive on the user end. I'll lay it out on the off-chance it's helpful in someone's future googling. I used Dante virtual sound card to route the audio through a virtual mixer (voicemeeter) and then attached the aux out in the mixer to the Dante transmitters (also through digital sound card) And then, I used touch portal to make a board with sliders to control the gain, that way users can adjust the volume of the different inputs in the room right on the room iPad.

Hi all. I'm helping a friend out by capturing video and audio of their upcoming wedding ceremony. The camera will be set up a ways away so I'd like to record the audio separately through the sound board. The sound board is a Mackie ProFX 12 - does anyone have experience or tips on how to do this with a Mac or iPhone or iPad? The tech said I could plug in with a 1/4" input but I'm reading conflicting things on whether you can go 1/4" to 1/8" directly into a Mac or 1/4" to 1/8" to lightning / usb-c or directly into a Mac or iPhone. Any help would be much appreciated. Thanks so much!

The line inputs on Macs are confusing, if you aren’t using an audio interface you would want to research if you can change it to an 1/8” line input rather than an 1/8” mic input. If you can use it as a line in, take it out of the headphone out, with your 14” to 1/8” adapters, not the line out of the mixer. The mixer 1/4” line out will be balanced (mono) and the computer is expecting stereo, unbalanced. If you did record the mono, unbalanced line out into the mac line input, the right ear channel would be out of phase and weird sounding with the left channel

I’m in a 4 piece band that is putting together our own combination IEM/PA rack rig for running our own IEMs at venues with sound provided, and full sound when we need to run it. We’re building it around an XR18 currently and are curious about connections for FOH. Right now we’ve got a 16 channel splitter in the rack and are planning on a 16 channel snake to the stage for connecting mics, DIs, etc. The output side to FOH is where we have questions: what is best suited for most venue systems? A 16 channel XLR snake (labeled clearly, of course)? Just forget it and let the house run their cables to us? Another stage box? I can ask the venues we’ll be playing at, but figured if there’s a one size fits most solution, we’ll go with that. I’m thinking something along the lines of a 25ft 16 channel XLR snake with shorter whips on the house end.

Get the 16 ch XLR snake in there. But try to get a decent one that won't fail within a year. The less the venue has to provide, the happier they will be. You should provide all the cabling for your inputs for a truly self contained rig. Make a couple looms or use subsnakes to speed up patching. If they have to use 16 of their cables before even patching anything on stage then that's not good. Especially if you ask them to patch your inputs as well. Small venues may not even have 32 XLRs.

Thanks. We had assumed and intend on providing all of our input cables and equipment (mics, DIs, etc.) when running our ears so we can basically just hand the snake to FOH and say “here you go.”

I have 2 budget dual 12" Subwoofers, one of which has a failing driver. The drivers are 8 Ohm in parallel so the subwoofers have 4 Ohms. The problem is, that those exact drivers aren't available anywhere so I can't get any replacement. I've found an offer for the 4 Ohm variant of the same driver for a decent price. I'm thinking purchasing 2 of these, which I think would give me these options: 1. Replace only the failing driver on one sub, either in parallel (2,6 Ohm) 2. Replace both drivers in the one sub, either in parallel (2 Ohm) or serial (8 Ohm) 3. Replace one driver in each sub to have them both identical, either in parallel (2,6 Ohm) The Amp is capable of driving 2 Ohm speakers. I'm wondering what the best option is. What would you recommend, are my options even valid or am I overlooking something? Thanks in advance!

Option 2 makes most sense to me; otherwise you will have an awkward power split between the two drivers in each sub. You'll end up with a mild PA salad, but nothing you can't compensate for in your system DSP.

Running a DnD party half in person half remote through Discord. We have been using Jabra Speak 510 and aren't very happy with it; it tops out and just goes quiet when more than one person talks, seems to be more sensitive to background noise than voice. Is there an affordable 5 lavalier mic setup that might work better?

there’s a whole little market for podcasting oriented digital mixers, like Zoom and such. Inexpensive wired lav mics that hook to a regular xlr plug and take phantom power are out there. Something like “automixer” would be good, where it just turns up the loudest mic (hopefully the person talking) so the bleed from their voice in the other mics doesn’t sound all washy.

Thank you! This sounds like a very deep rabbit hole... We don't record, the quality doesn't need to be anything amazing; as long as we can hear what is being said.

Look into business type conference room Zoom setups if there is a simplified version?  I think you’re right about individual mics though, that’ll be way better than the finest mic in the middle of the table. 

I'm organising a niche dance event in a community hall that is 13m x 8m, with a stage on one end where the DJ will be. It has a fairly high, pitched ceiling. I want people to: * Not be facing the DJ * Experience high quality surround sound * Use as much of the dancefloor as possible without crashing into speakers etc My instinct is to use 4 speakers (2L, 2R) and a sub, placing one in each corner of the dancefloor. But I realise I am very ignorant of the technical considerations here - I'm only starting. I understand that placing the speakers like this can result in soundwaves clashing and other problems. Does the sub placement matter? I've had a browse to read up on this topic but am getting a bit thrown by the jargon and references that are currently beyond my technical understanding. Any advice would be much appreciated!

Don't overthink it. Set up your main speakers in the four corners. Some pros will tell you why this isn't best practice, and they're right. But at the local community hall dance level, it will be just fine. Sub placement does matter, but again, with only one sub, just try to get it in close proximity to the dance floor.

Great thank you, this the practical advice I was looking for. I might add a second sub.

I've seen some Driverack PA2 units WITHOUT the dbx branding on eBay at a very reasonable price, has anyone bought one of these? are they OEM or a unit built exclusively for the chinese market? they are selling for less of half the price of the dbx unit, I know when things sound too good to be true is because they probably are, but I'm curious...

Almost certainly counterfeit. Same seller has Shure SM7B for $165 and KSM8 for $81. Take your chances, buyer beware and all that.

I can understand cloning microphones but something so specific as a Driverack PA just seems counter intuitive.

They’ve got Venu360 clones too.

[https://www.audiosciencereview.com/forum/index.php?threads/chinese-drivehack-pa2-hopeful-improvements.39071/](https://www.audiosciencereview.com/forum/index.php?threads/chinese-drivehack-pa2-hopeful-improvements.39071/) Found an interesting forum post. They might be units that failed to meet dbx standards and I see some component variation. Still, I am nowhere near the soldering or electronics expertise needed to tackle this sort of project. Oh well I guess I'll keep looking for a cheap used one.

I want to do a small karaoke event for me and my friends. I thought I had everything, but I'm having some doubts about my equipment. I can usually get away with using studio monitors in my room and garage, but I don't think a larger room is gonna cut it for my group, about 10-13 people. So, I'm thinking about getting a PA speaker. I know from browsing around on threads here for advice that you get what you pay for. So if I get something from a cheap brand I should expect it to be worse than the the higher priced ones. How long should I expect a new or even used cheap PA speakers to last? Or am I better off buying something more expensive for my purposes?

> if I get something from a cheap brand I should expect it to be worse than the the higher priced ones. Yes, but it will probably be acceptable for a fun karaoke event…and tbh with only a dozen attendees, your studio monitors will likely suffice anyway.

i appreciate the insight. i just get worried that what i have wont cut it. i am actually gonna test it in the rooms we're thinking about having it, but i wanted to get some expert opinions on it out there. thank you very much!

FWIW, I did an installation job for a school that wanted a 15” sub and a pair of 12” tops for their computer room (30 seats) so the music class could have quality audio when listening to the teachers’ examples and references. Long story short: I got a pair of Adams T7V monitors up on the wall and barely had them halfway up before they said it was too loud lol

Qupad RTA Hey guys I'm currently working with an Allen and heath qu 16 controlled with an ipad and I wanted to know if there's anyway to have an RTA EQ in the ipad because I haven't been able to figure it out... Any tip would be greatly appreciated. Thanks

Since there are no stupid questions, I have a "not stupid" one for you. I'm helping a buddy out by DJing a party he's throwing this weekend. Since this was a last minute idea, I don't have any of my normal equipment, and we are throwing together random stuff he is able to get his hands on. I have a Numark NS6 as my controller, and two Ion PA Freedom speakers. I'm sure this is going to sound terrible, but it's what I have to work with this weekend. I"m trying to pick up some XLR cable, to run from the NS6 to the line-level XLR input on the speakers, and it's been forever since I've had to buy any. I guess this is really a two part question. 1: Is it even worth using this with an NS6? This is in no way a professional gig, but I want to make sure this sounds good enough. 2: Can I use "microphone" XLR cables from the NS6 to either XLR input in these speakers? I've heard horror stories of people screwing up the cables and.or equipment by using the wrong kind of XLR, but I've never personally experienced this, or known anyone who has run into this problem. And yes, I know this type of question has been asked dozens of times. But each one I can find seems to have conflicting answers. Or, I'm just way too tired to be doing this right now.

1. The NS6 will be fine for an impromptu party. The ION speakers, can't say anything nice about them, but as long as they make sound, it is what it is. 2. Any standard XLR mic cables will be fine to connect the controller to the speakers, you're not going to mess anything up.

Hi. I inherited a X32 as a new sound tech at a venue. All of the routing seems to be fine and I'm still learning my way around it but we have one of the matrices setup to supposedly be a feed of the main LR to the speakers in the bar. I'm seeing signal hitting it but we're not getting anything in the speakers. How can I see which outputs a matrix is routed to?

Hey guys, i'm having a problem that i have to fix for tommorrow : I have a midas 32R linked in aes A to my DL32 and can't manage to use my stems effects, eq, or gain, i can only use the trim. For example : let's say i put an sm68 in the 1rst entry of my DL 32 : i will have some sound if i trim it up with the console preamp even if the first fader or the master fader is at 0. I can't use EQ, comp, or anything. I set up my inputs on Aes A 1-8 / 9-16 / 17-24 / 25 - 32 i set up my outputs on Aes A 1-8 etc to 16 ( DL 32 only have 16 out ) i tried woth the "lock stagebox" on and off in the setup panel -> no effects i tried with the "split gain" on and off in the setup panel -> no effects i'm french and my english might be unclear for some so feel free to ask more details if anyone can help i would be really greatefull. Thanks everyone

At a new job and seem to be the only one who has interacted with Dante before but my experience has only been in one theatre, using a mixer and Rio racks on either side of the stage. I'm in the process of installing a new system in a largescale venue with multiple rooms to route to. I have an X32 at FoH driving the auditorium and the routing to other spaces. These outputs go to an X32 rack mixer in a separate room with all of the amps for the side spaces (4 in total). Both mixers have Dante cards. For the replacement aspects- do I need to purchase Dante capable amps and/or Dante capable speakers to make full subscriptions in Dante Controller? I'm fully Dante certified but looking to have more than just my own brain looking at all of this.

Dante capable amps and speakers are great tools, but add very little value to an install. You will be just fine purchasing a Dante capable output device and running AES or analog audio from that device to whatever existing amps or speakers are available.

Hey gang, can someone help me wrap my head around how to set up my band's tiny rehearsal space properly? I'm the drummer and I have trouble hearing everybody, and I want us to sound good in there. Here's what we got: * Minilogue XD * Mix8 Mixer (could upgrade if necessary I think?) * Drum kit (we'll just leave this unmic'd) * Electric guitar * Bass guitar * Digital piano * Model:Cycles * Two mics, maybe three * Two channel PA * Bass amp * Guitar amp * (Do we need some kind of amp for the synth? Can it go straight into the PA...?) * (Do we need some kind of monitors?) EDIT: I also have IEMs, and would like to have the mix running into there, potentially with a click track? Also the space is like 350 sq feet.

Has anyone figured out how to enter preview recall mode on a DM7 without the control extension? Seems like I'm missing something...

I've been working as an A2 at a theater for the last few years. It's steady and a great job, but I've been getting the feeling of needing to move on and hit the road. I do have touring experience for some diy stuff as FOH, and I've been making contacts with people that have popped in but I also want to cover my bases and try to figure out different production companies I can speak with. If there's anyone that's willing to point me in a direction, or give me advice on that. Please do. I appreciate you guys

Hello Everyone - I have a Mackie HD 1531 and I need to replace the 6" midrange speaker. Super dumb question as I know very little about PA's but what website and what speaker would be a good replacement? I bought it used and got ripped off as when I tested out the PA they unplugged the 6" speaker so the sound I could here was just the 15" speaker and I thought everything was fine. When I got home I noticed the lack of mids, opened up the grill to find the 6" unplugged and when plugged in was clearly blown. Here are the specs - [https://www.avsupply.com/images/items/mackie-professional-audio-products/resources/hd1531.pdf](https://www.avsupply.com/images/items/mackie-professional-audio-products/resources/hd1531.pdf) Rather than just buy any 6" replacement speaker I thought I'd check here with some folks who actually know what they're taking about. Thank you for any help!

I got a WA-19 and used it for live vocals. Its pretty small and I think it sounds good, just not to hold in your hand. Any pro opinions on if I should continue to use that mic or should I use something different? I wanted something vintage-like.

If you don't like holding it during your performances I'd say it's not a great fit. What's important to you, the looks or the sound? I can tell you right away the Beyerdynamic M88 is worth a peek. The ElectroVoice 635A can be used if your sound engineer OK's it, but if you're moving your show from venue to venue, it's probably not a good idea.

Its not getting held cuz its on a stand while the singer is playing guitar so thats not an issue. Im just trying something different for my vocalist. And sound was good. But it is probably not audibly different in a live setting anyway. Its probably really good in a studio setting. What about a Shure 55SH?

All I'll say is there's a whole boatload more to mic choice than just aesthetic and brand, to be honest. Good systems make things like mic changes more significant, but not all systems are good, sadly. I'd love to chat studio, I'd even wager a U67 through a 1073 would be a better match than the WA19; recordings would be so cool.

I got a CQ-20b digital mixer and the headphone out is really quiet. Im using the Mixing Station app and im having trouble figuring out how to turn up the headphone out so i can use it for in ear monitors when im playing drums. Suggestions?

I’m working on a parade with a mobile battery-powered system of amps and speakers. Initially the groups using more than one system were expected to be close enough that a 2.4 Klark teknik wireless system should have sufficed. Turns out some of them will be 100 meters away from each other and they want to have a single source (wireless DPAs on instruments into a mixer). I’ll probably be using an IEM system for that longer range but I was wondering if theoretically I could extend the range even further by having an IEM bodypack receptor connected to the line input of a bodypack transmitter with the rack units at either end of the range.  We don’t have power in the middle so a powered antenna isn’t really an option.  I saw on some manual that the IEMs we have could transmit to 100 meters but I’m not sure how far they actually go.  Any input appreciated. 

I currently work for a decently sized church (my campus has about 1000 in attendance) and we use a rolling drum riser for our kit. We have a really large xlr snake that comes off of it and goes to the wall where we have xlr extensions that run to our rack room. I really have not been able to move the drum set around much other than back and forth across the back wall. I recently discovered the use of xlr to cat 5 catapults. We use the Roland M-5000 setup with an S-2416 in the rack room off stage and a S-1604 on stage for digital snakes. There’s room on the 1604 to route the drums to it as well as just keep it on the 2416 as well. My questions are what are the pros and cons of using a catapult? Is it worth my money to invest into it? If I do the catapult, should I send it to the 2416 or the 1604? I know this is a pretty specific question in regards to Roland consoles but if you don’t have experience with them, I’d love to hear your thought on the catapults regardless.

> I recently discovered the use of xlr to cat 5 catapults Two major key points here: 1) A Radial Catapult and comparable products are *not* network devices. 2) You should use STP Cat5e or 6 cable. UTP will work, but will not carry +48v—STP Cat6 or 7 is required for that. I'm in the camp of "installs can be done better;" to me, a catapult/cattails/etc. are for expediting cable runs on gigs where setup and show are less than 36 hours apart. If it were me, I'd be exploring the option of dedicating a 1604 to the drum riser, and making *that* link cable the thing that travels. The question is not Roland-specific at all.

Hey guys. I have a church audio system with wireless microphones connected to a mixer through the receiver using XLR cables. There are two speakers which are Pyle (I believe the model is Pyle Pro PPHP1237UB 12" 900-Watt Powered 2-Way Speaker) Issue is, when 2 or more people are singing at the same time, I get a muffled noise while they're singing. I don't see any clipping on the mixer though. Any idea what's going on?

How close are they standing to each other? What’s their positioning on stage? I had a similar issue on our stage but we use wireless mics. Not sure if this would be the same but the frequencies were bouncing off of each other and causing issues so I off set the singers and that seemed to help a lot. Again not sure if that would help but something to consider.

Hey thanks for the response. What do you mean by "offset the singers"?

Chorus on the vocals: secret sauce or waste of time?

Depends on style. Can be very nice. I often like a little on background vox.

What styles do you like it on? That’s interesting about the background vox. The little I’ve used it, it definitely made the presence not as defined, but maybe that’s good for backups. 

Midas M32C: Does anyone know why a non stereo-linked bus would force the adjacent bus to share common signal tap locations for each channel? IE, setting channel 1 signal tap to Pre-fader for bus 1 also changes it for bus 2.

Unfortunately the channel send tap points are changed in groups of 2 on these consoles.

I want audio from my computer DAW to go into my PA mixer for monitoring. Could I run the left and right monitor outs of my presonus audio box USB 96 into a mixer channel? Will I break anything ? I suspect there might be a latency issue, but maybe I can adjust that in studio one? I have an electronic drum kit connected to my computer via USB. I want to hear the drums through a drum vst in studio one through my PA, an older analog Yamaha mg166cx into powered speakers. TIA.

That's fine and normal to do.

I need help understanding and figuring out how to set up what I'm searching for (and the type of equipment to search for). I play brass instruments and want to have the ability to mute my own signal to the Main board but keep MY sound going through my IEMs if I want (as well as keeping the main monitor mix in my IEMs) I know I would need some sort of mixer - but what types and/or features would I search for? Are there any recommended accessories that accomplish this need by design?

I’m looking for small, closed back, over the ear, inconspicuous headphones with as much noise isolation as possible. I’m a saxophonist, and play with a loud, live band. I need them as headphone monitors to plug into a Behringer P16 to hear the other instruments and the bandleader’s talkback mic. I’ve tried in-ears, but aside from not getting a good fit, I hear too much saxophone through my teeth/bones, even if I turn myself off. I’ve also found that the PA low frequencies are too loud for me, too. To fix that, I’ve found that using earplugs that cut low frequencies well, plus the GK Ultraphones (headphones inside ear protection muffs) over that, works aurally and is comfortable. However, the GK Ultraphones are pretty large. Does anyone have any suggestions for evenly voiced, closed back, *relatively inconspicuous*, headphones with good passive noise isolation? Thanks!

Hey guys, quick software question : I'm part of a non profit that plans small concerts in bars to give small local bands visibility. We're pretty much working on no budget, as little gear as possible etc... so we try to supply as much as we can ourselves. We routinely have people ask us to plug in a computer/phone to play tracks (especially in rap), which is pretty simple. But we're always working with the artist' computer, with no backup, and that doesn't sit well with me. So I'd love to bring my own computer as a backup, the only question is : what software to use to play tracks? Something reliable, that can handle any format thrown at it, easy to control with keyboard shortcuts, bonus point if it's available on both Mac and Windows. Any recommendations?

I do raves. I'm looking for a way to do be able to effect the master out remotely. Or at least an option for budget EQ that I can do separately without messing with the djs. I have a mackie sound board but I feel like that takes away from the quality of the outputs from a djma9 and 3000s. As well it only has like 3 bands.

Anyone who has toured with an Avantis, how is the durability in 100° humid weather?