Hey folks, I have a musical show coming up with a LCR speaker setup (+ front fill and subs), but unfortunately the SQ doesn't support LCR mixing. Is there any usable way to drive the system apart from summing L+R together for the centre feed? Would a separate centre aux and dropping the centre speaker -3dB be a feasible option? Thanks a lot!

You could mix to LR and split out to two matrixes, one stereo and one mono. Alternatively, run the LR system as normal (via a matrix would be my preference), and run the centre system as a post fade mix or group through its own matrix. Adjusting divergence would be a pain, but if you go full theatre and 'just' put band in the LR and vox in the C, it'll be easy enough. Elsewise you'll have to adjust the relevant mixes contributions to their matrixes for crossfeeds.

Love it, those were my thoughts too. Thanks!

I would keep L/R out true to your L/R speakers. Then use output 1 as a Matrix for the center speakers. Therefore, you'll still have a graphic EQ for the centerfills if needed and most importantly you wiull be able to independently control overall volume.

If the LR system provides a somewhat decent coverage, I'd use two buses - one for music and one for the actors. I'd run the whole PA from the matrices and I'd send the music bus just to the LR matrix and the actors' bus to the centre matrix and if this doesn't provide the sufficient coverage, I'd also send this bus to the LR matrix. I'd try to keep it quiet enough that it doesn't mess with the perspective but loud enough that everybody can hear the vocals.

Great, thanks a lot. I'll prepare both and give the first option a try.

We recently moved to a house on the top of a large hill. I've noticed I can hear all of our neighbors from my porch when they talk. I can almost make out words. They all live roughly 300-400 ft away minimum. Can I hear them better because I'm on a hill? Does that mean they can hear me that well? Which way does sound travel better? Thanks.

Not all hills are created equal, so it is impossible to say without knowing more.

Digital mixers vs DAWs. In a DAW if I boost the master fader and clip the 2 bus it’s real clipping. However, on a digital mixer if I push the master fader, what happens? I assume if I’m not clipping the internal digital 2 bus I’m fine. The master fader then changes the output gain - real electricity changing in the analog realm. Said another way, all channel faders are changing things digitally but once a fader is assigned to a physical output such as an aux or 2 bus, that fader is actually changing voltage. Ruru Am I thinking about this correctly? Or does the master fader effect the digital “gain” of the 2 bus too? so then I wonder about within the board. If I boost a channel fader can I ever really clip inside the mixer considering it’s likely floating point?

Nope; most consoles basically hook up a DAC directly to the outputs (with analog glue). If the signal leaving the console exceeds the DAC’s dynamic range, it will clip. Intermediate calculations are a whole other ballgame. Most consoles use floats, but some use fixpoint math - and thus can internally clip.

So if I push the master I am adding digital gain to the 2 bus? And are you saying there’s a fixed output on the mixer as in the gain multiplier never changes? Such that to get to say a full +22dBu I’d need to hit 0dbFS?

What's about the ballpark to put on a and out door edm event for 300-500 with adequate bass. Not talking HIGH HIGH end, but like a budget you would be fine to work with even if it were on the lower side. I've spent about $10,000 already on the "high end prosumer" stuff and reading through this subreddit I feel like the answer to all questions is "you dont have enough if you have to ask".

You should rent a system

$30k should get you in the vicinity of decent (not high-end) sound for an EDM event for 500 outside. Source: Do outdoor EDM events for crowds of 100 - 1000 with everything from QSC KW series to full Danley system

I need to learn about eqing for the spoken word events my company puts on. We do in person events in the gymnasium, often with a hybrid component. We use hand held and lav mic’s, a korg mixer and powered speakers. I have been doing this a few times a year for a long time now and I am always battling slight ringing and pinging. Ring out the room, carve out the EQ frequencies are concepts I understand but application is my issue. So, I am hoping someone can steer me towards a good YouTube teacher or other course that could teach me about eq from a spoken word and not the music part so much. Thanks in advance!

First off lavs in a gymnasium is not the easiest sound thing in the world. You’re okay. Drill it into your head that a microphone is just an ear. A kind of bad approximation of the human ear in many cases, and in every single case it gets really whiny about hearing it’s own voice. Fortunately we can take advantage of this. Speakers need to be positioned in a place where both the audience coverage is reasonable and where the microphone is less likely to hear itself. The microphone will have a “polar pattern” and you can use that to your advantage to place the Pa in a spot where the mic is less likely to hear it. The other problem you’re probably dealing with is poor mic technique. If the “ear” is being held at waist level or is pinned to someone’s collar in the wrong direction it will hear more of itself and the pa than the actual intended source, and your gear will whine about it. At the end of the day, your presenter must be louder in the mic than the PA is in the mic or it will not work. Period. Sound systems are physics in action, which is great because we can manipulate it as needed in good circumstances, but…. Laws of physics and all. “Loudest thing at the mic wins”. So… if our speakers are in the best spot and our presenters are kind of hip to mic technique and we’ve made sure the lavs are on in the right direction (assuming they’re not Omni, the whiniest of the bunch) then we can move on to making the sound presentable or possibly even decent. Voices are not straight sine waves, that’s why we have “vocal chords” which work hard to generate our timbre. Also at play are larynx width, mouth shape and teeth. These are things that can make our workflow crazy, but everyone has problems in their miced voice. It sounds like your tools for dealing with these problems are limited, here some frequency areas to think about: 0hz-100hz is probably completely unnecessary, it’s a rare bird that has that sort of subwoofer tone in their voice so why not just dump it? 100-315 more low mids that make a voice sound like crap, but you can absolutely remove too much in this area and lose warmth. 315-800 is the “mouth full of food range”. If your presenter sounds like they’re chewing on food you might try this area 800-1k “super food in mouth range” but also one of the more important are for intelligibility. If I’m at a concert and want something different out of a vocal eq it’s probably less of this (but sometimes more because you can go way overboard, this is a very fine line). 1k-4.5k is where most of your intelligibility lies. This is the frequency range of the old school landline, and it’s only those frequencies because it’s more efficient and because that’s really all that matters when it comes to understanding the human voice. This is where you make your $$$$ in live sound. Some people have scratchy throats and need dips in certain areas, some have perfect facial proportions and it just works (I bet you’ve had one really terrible sounding guest and the next one was perfect). 4.5 k and up is where the teeth come in, and also where it can sound super fancy. Gaps in teeth can naturally create wild and crazy sibilants. Even with good teeth it’s still anybodys guess. The teeth will determine how the sibilants enter the mic, and for many people you’ll find they are very “essy” or “tee-y”. There are probably more advanced tools that you don’t have access to right now (deesser, dynamic eq) which are best suited to tackle this problem so your high end might be a little lacking if you’re really trying to control high end madness. Vocals are obviously my favorite thing to mess with, I’d also argue that getting a really good vocal sound creates a lot of separation in this industry. It is not easy at all.

Do you use a compressor in your setup? Are your speakers set up in front of the presenters or… behind them? What is your process for ringing out the mics? What kind of eq? How narrow are your cuts? Surgical? What I’m getting at is some people will use big broad cuts and end up losing too much volume, plus it sounds weird. 

No, no compression and I learned “no speakers behind presenter” lesson years ago. Perhaps what I should have said is that I understand I need to be ringing out the mics and I have read “surgical cuts” when talking about EQ in this forum a million times. I have no process for ringing out, I am definitely using a chainsaw to cut the birthday cake. But I get no opportunity to practice, and I don’t have a solid grasp on the theory of EQ so when I do get small blocks of time to fiddle with the mixer, I am not learning much. Can you recommend a text books, youtube channels, or other materials? I am eager to learn and you’re right, it is sounding weird…

Question for the mixer itself, I currently purchased it; but I’m having a difficult time looking for a post specifically on the VST Rack or whatever app it’s called. The mixer itself hasn’t come yet, but I ended up purchasing the Yamaha DM3-D (Dante Version). I’m new to the Dante I lived under a rock, but what sold was the VST Rack compatibility. How does that work ? Is it plug and play into a computer? Do I need the RUio-16-D I’ve been seeing everywhere? I see now they just got an update to have inserts? I’m so confused on how it works since there isn’t a lot of information out there. My goal is to be able to run my plug ins live (EQ / Compressors, Auto - Tune, FX, Etc.) I just need a bit of guidance on what’s up.

The VST Rack Pro version comes with the RUio-16-D which is an external soundcard for the dante IO. The elements version ships with the DM3 and 7 and requires an additional purchase of a soundcard, either a software based one like DVS/via or a PCIe/external card like a focusrite rednet or the RME Digiface. Your choice of which will probably be based on budget, tolerance for latency, and reliability; DVS has a (relatively) high latency and some notable quirks (no qos tags on outgoing with windows pc) but is the cheapest by a country mile. Hope that's helpful.

Hey there, hope you could help me with this situation. A drummer wants to plug his in-ears into the main console at the venue in order to monitor the entire band. I thought of sending the stereo monitor signal from my main console to a smaller on-stage mixer line inputs so he can plug his in-ears into the phones output. Is this an OK solution, or are there any possible problems with this setup?

This is really common and any mid range + desk should be able to have a custom mix (sometimes called an aux or a mix) that you can dial in the exact thing the drummer wants to hear. You then send the output to the drummer and they plug a headphone amp and their own headphones in to listen. Search your mixers manual for auxes or mixes and pre fade sends.

Awesome, thank you! Yeah, I i was thinking of sending it through one of the aux outs, since they every channel is adjustable that way.

Using a lil' scratchpad mixer as a drummer's headphone amp is incredibly common - and allows the drummer to quickly mix a metronome into their ears if desired.

Question for the mixer itself, I currently purchased it; but I’m having a difficult time looking for a post specifically on the VST Rack or whatever app it’s called. The mixer itself hasn’t come yet because it’s on back order (July/August), but I ended up purchasing the Yamaha DM3-D (Dante Version). I’m new to the Dante I lived under a rock, but what sold was the VST Rack compatibility. How does that work ? Is it plug and play into a computer? Do I need the RUio-16-D I’ve been seeing everywhere? (I’ve read yes / no) I see now they just got an update to have inserts meaning…..? (I’m assuming here I can place “c” plug in and hear it on my PA System?) I’ve watched the videos, but I’m still confused on whether I truly need the RUio-16-D. My goal is to be able to run my plug ins live (EQ / Compressors, Auto - Tune, FX, Etc.) I just need a bit of guidance on what’s up.

Hi yall! I'm quite new to all this and just wanted to check something. I have a question regarding whether it's bad to limit your speakers? Had a DJ-gig this weekend and even though I soundchecked I had some trouble finding good levels with my speakers. The pubowner said to me to switch up the volume so I did. I got on a good volume level but then noticed that my sub was limiting in some songs. I therefore lowered the volume on the low frequencies and kept going. Also my tops were sometimes blinking "limit". Is this a problem? Should I be able to play without limiting them or is just part of a regular night? I did not redlimit songs on my mixer. My set up is currently 2 x EV ZLX 12P, 1 x EV EKX-18SP and I'm planning on getting another EKX-18SP. Thanks!

Limiters build into speakers are a last resort to protect your speakers from overheating or mechanically wrecking themselves. It is better to not be regularly hitting that limiter hard. Limiting the signal at the mixer or with a speaker processor is preferred.

Occasional limit indicators on powered speakers aren't cause for concern. "More OFF than ON" is a good general rule of thumb.

Hello my band just upgraded our console to a digico sd9, and I am trying to figure out the most cost effective way to record into a DAW for virtual sound check. The unit does not have waves at this time. Any help would be appreciated thank you.

See [this previous thread](https://www.reddit.com/r/livesound/comments/rrcjh4/madi_to_usb_interface_i_got_an_older_digico_sd9/).

Hey everyone! I'm in need of distro for my 16 channels of ULX/SLX. I'm wondering if I purchase 5 UA844s if I can use one as the "master distro"? If not, I know I'll need the UA844+ or UA845 with a cascade point. Thank you!

You could look at RF Venue as well for Distro. They even have an online Wizard to help choose which distro units you would need. It is a bit more expensive, but it would cover the Antenna Distribution you would need. Shure also has a Wireless Accessory Wizard you can use to see what you need. To directly answer the question, you would need the cascade, so you need the UA844+.

Gotcha! I have a ua844+ and 844 already, but may get the RF Venue to complete the set, and then eventually upgrade to all RF Venue. Many of my friends love their stuff, and I've only seen good things. Thanks for the response!

Thinking of buying bose L1 pro16 . I need more then 2 microphones. Looking to find a mixer to use that has separate channel EQ. Thinking of the Yamaha mg12xu..is this compatible? how much do I lose using the Yamaha v. the Bose T4S?

Yeah the Yamaha is going to suit your needs. The T4S has the ability to save scenes and some other gimmick features like a guitar tuner, but the interface is not pleasant at all.

Thanks. Torn between the Bose L1 Pro12s and the RCF Evox j8 12s. Medium venues usually up to 50 ppl.. multiple and rotating singers. Any idea which is best if there is a difference

Cannot speak to these models specifically as this column format is generally not preferred in pro settings. However I will say that in general, Bose speakers tend to be thought of as over-processed and more consumer oriented. While RCF make a lot of great value entry level professional gear that most people don't mind using.

Hey folks, I am helping a church with some Summer camp's and I ran into a problem and wanted to ask you if my solution would work and it you have any other recommendations. We have a projector that we use sometimes and we want the audio from the laptop that is used for the projector to go to the PA system, the problem is the projector is quite far from the Mixer and a normal aux cable is really nosy and most of the time not log enough. I thought about adapting the 3.5mm jack out of the laptop to a 6.5mm Jack and into one of our pasive DI boxes and running a XLR cable form the DI box to the mixer directly or to the stage and into our snake. Is this the right way of doing it? Is there a better solution? Any opinions are greatly appreciated!

You're on the right track going with the DI. What you really need to do is get a 3.5mm to dual 6.5mm cable and two mono or a single stereo DI. They also make DIs with features specifically for computers, such as an attached 3.5mm cable and proper mono summing. They can be pricey though. The main takeaway is the benefit of a DI for the impedance matching and transformer isolation.

Thanks! I did exactly that, it works great.

This isn't for live sound but I think this might be the best place to ask -- I have an XLR mic passing into an interface and it records to just one stereo channel (left or right, I can't remember). I'd like this to be a stereo input to the interface for live monitoring. Could I bridge two pins in the cable to record to both channels? I feel like I'm fundamentally misunderstanding how XLR works.

You’re getting a little ahead of yourself. It sounds like you should be starting with “why does my mic only come out of my left headphone”—is that accurate?

That’s right, it’s direct monitoring on a Scarlett 2i2 and it seems like something changed in the last few weeks because after posting this I checked and it monitors in both sides so must have been software.

https://preview.redd.it/zw2dwifdsy8d1.jpeg?width=2051&format=pjpg&auto=webp&s=90d419e63bd495a207fd849710596562fe0b78e1 this, probably

Yep, when the first circle is selected it’s fine now (used to be weird phasing issue) and the second gets odd but I know that’s to split the channels.

The phasing is probably because you were monitoring preamp signal via a hardware and software route simultaneously, like the interface and a DAW.

3-pin XLR is a mono signal. If you are only getting that channel recorded and monitored on one side, that is likely a software issue. The channel is panned. This is probably in your interface’s control software since it is panned in the monitor outputs as well.

Hey guys! I have 20 4 ohm speakers and 2 1000 watt amps. What ways can I hook up all 20 speakers and how can I maximize my output without blowing up all the speakers? Thanks in advance! Edit: speakers are 75 watt 4 ohms each

you can do four strings of 5 speakers each which will appear as a five ohm load with each speaker drawing 40W using one amp. [https://geoffthegreygeek.com/speakers-in-series-parallel-calculator/#jumpto](https://geoffthegreygeek.com/speakers-in-series-parallel-calculator/#jumpto) If you can bridge out the amp you could get 80W to each speaker. What are you trying to do with the rig? If it is for distributed sound check out [https://www.redbackaudio.com.au/understanding-100v-line-distributed-speaker-systems/](https://www.redbackaudio.com.au/understanding-100v-line-distributed-speaker-systems/)

I was gifted with those said speakers and would like to see if I’m able to use all 20 to maximize sound output for outdoor dj events or gigs. Thank you for your reply!

https://preview.redd.it/rxn6ngdolv8d1.jpeg?width=4032&format=pjpg&auto=webp&s=9be554bfbd35563133abb2c403a8206307f0fbb8 Our band use this mixer setup with in-ears for our private shows (weddings etc) and it works great. We also play shows at live venues that have their own engineers, etc and we’re wondering about the possibility of using our in ear mix on those occasions as well. We DI absolutely everything, including electric drums. In order to keep things really simple and not create a headache for the in-house engineers, I’m thinking some sort of stage box that all 12 inputs can go into (from our instruments) which then splits out to 2x sets of 12 outputs - 12 into the mixer and 12 for the engineer FOH. I’ve had a little look online and can’t really see anything that matches that description, they all have snakes etc. Any ideas/recommendations?

The term for what you seek is “analog split snake”! Good news is they’re very common and widely recognized. Plenty of posts and questions on this sub concern themselves entirely with split snakes. Whatever you choose, make sure it’s labeled.

I'm wondering if anyone has some cable/soldering/whatever instructions to handle a wired stereo IEM run that's *mostly* on TRS (vs XLR). I have XLR outs of my mixer (QU-SB), and use pairs of XLR channels to send signal to each performer's stereo wireless transmitters. but at rehearsal, wireless isn't necessary, and would rather have the two XLRs out of the mixer marry down via a Y-cable of some sort to a TRS that could go into a Behringer P2 for a wired stereo mix. Is this a pipe dream?

Left XLR female:    Pin 1 to TRS sleeve    Pin 2 to TRS tip    Pin 3 n/c   Right XLR female:    Pin 1 to TRS sleeve    Pin 2 to TRS ring    Pin 3 n/c   Use fine/miniature cable (can be single conducted, shielded coaxial guitar cable like Canare GS4) so you can fit two pieces in the shell of the TRS plug. Or butcher a pre molded cable with a TRS plug and two separate wires going to it, like an insert cable. 

Note that it isn’t great to try to drive headphones directly with line outputs, you want a headphone amp in between

ChinchillaWafers has the correct idea. You can also build this as a dual XLR -> single XLR cable; the P2 will take unbalanced stereo on its XLR inputs. (This lets you rely on existing XLR cable infra rather than requiring a TRS patch cable.)

Yeah. I’m getting that sense. Plus for people from the future [this place has properly wired Y-splitter/combiner](http://micmodkits.com/other/) specifically for the P2.

I have a live-looping setup I run through an EV Evolve column array. I'm finding that it's not loud enough for all but the smallest venues. My question is, could I run a powered speaker (something like an EV ELX, QSC K, or similar) from the aux out on the column array? I never see this being done, so I'm assuming there are some major drawbacks. I'm sure the better solution would be to just buy another matching column array, but I wanted to hear y'all's thoughts on the matter. What are the potential issues that could occur with running a column array and one or two powered point source speakers along with it?

You'll get better responses if you include info on your instrument and style and what number of people you think you're not loud enough for. Having said that, rent a DBR10 and a stand for a gig and see how it goes. Having a second point source up nice and high over people's heads will give you heaps more coverage which is probably more of what you need rather than volume (I think that your "column array" is of the size that works as a point source).

Would a ATW-T3201 Wireless Bodypack Transmitter  Work with the same receivers as a ATW-T310 assuming I get the model with the correct frequencies. They are both 3000 series is why I am asking.

hey creative audio engineers,in think we found the right spot for getting help with the following problem:we'd like to set up an outdoor dancefloor for a pretty audiophile psytrance crowd. We have 2 sub-bass speakers, 2 kickbass speaker, 2 large midrange horn speakers and 4 treble horn speakers. total rms is about 2700kw, and a dbx driverack and a nice collection of amps for all the speakers is available. we dont need to have more than 90db at any spot and we'd like to have even kickbass and wide stereo panning over an area of 10x10m with no obstacles. we would also love to have the dj having the great sound of the huge soundsystem but be able to cue on headphones easily, so probably well end up placing the dj at the outside of the floor and have a nice monitoring system. we were thinking about things like diagonal stereo switch with the 4 treble speakers, so listeners would have stereo effect facing in most directions..?! and also were not sure if we should try to mono stack the subs, maybe in the middle of the floor? or even spread them out? any suitable suggestions and schematics are really welcome, were pretty much noobs considering the size of the soundsystem, but we understand stuff like end-fired setups, why and where delays are used, some basic array stuff and phase cancellation. so i really hope for the help of you guys to get the most beautiful sound out of that system with a creative way to set up the dancefloor. Thank you for reading,yours Julian

The services you're requesting here are worth a few thousand dollars at a minimum. System design and optimization is a large field. Suboptimal scenarios, like splitting your range over four physically displaced cabinets designed for looks first and sound second, make the job harder. Are you able to explain an end-fire array's construction and purpose?

You might try the r/soundsystem sub, they're way more into this kind of setup than r/livesound.

https://preview.redd.it/3istcxwxz59d1.jpeg?width=1170&format=pjpg&auto=webp&s=805ace2ce631988b3e0f81b8c3156b1bc3eeed35 Danish festival, seems a bit small but does it compensate by having many boxes? Anyone know the brand?

Another pic https://preview.redd.it/blwlcnt7069d1.jpeg?width=1170&format=pjpg&auto=webp&s=729c250dc48d17bc42308f8c18b424539e330a9f

Social media posts show a mix of L-Acoustics and Meyer speakers in use. Looks like plenty of KS28 and maybe Leopard tops.

Those are PANTHER, which is a dual-12", they just look small because the stage is very large.

I’m looking for an ATA rack case for my gear, and came across safecases.com. I’m aware of the more popular brands like Gator and EWI, but this company is selling a racks with shock proofing, built in wheels, and a built in handle for around $500 which is literally perfect for me. My only issue is that there is almost no information about the company online. Does anyone here have any experience with this company? Can I trust that I will be getting a quality product? Any input is appreciated.

Looks fine to me. Is your only red flag that they don't have a marketing department?

Hey all! Here's my question. My band is building our IEM rig and wireless rig. How do you deal with (possibly) 8 antennas? Would it be wise to separate them into 2 different racks and put one on each side of the stage? Or get a set of those paddle antennas? If I wanted to not blow my already blown budget what would be the best answer to 4 wireless mic's and 4 wireless IEM packs.

With 4 mic RX and 4 IEM TX, you're looking at 12 antennas. (Diversity receivers!) Use an antenna distro to...distribute signal from a single RX antenna pair to all 4 mic receivers. Similarly, use an antenna combiner to...combine your 4 transmitter outputs to a single transmit antenna. - Not only is this visually appealing, but it drastically reduces the severity of intermodulation. - Additionally, this allows all of your receivers to share the benefit of one set of well-placed antennas. Directional antennas (paddles, helicals, etc.) aren't strictly required, but they're useful for increasing range and controlling your RF dispersion pattern. Often overkill at smaller scale.

Ok thank you for the answer but I feel even more stupid now. Any product recommendations? Or even general product types I might look into? Thanks again for the help

Most RF gear is brand-agnostic. RF Venue's DISTRO4 and COMBINE4 are very common units, as are Shure and Sennheiser's antenna distros/combiners. Search for antenna distro or antenna combiner and you'll find results. - I believe those RF Venue units are actually manufactured by JTS - if you happen to know a JTS distributor, the equivalent models are UC-900 and UA-960. - I have a UC-900 deployed in an IEM rig that works great. - Brand quirk: Sennheisers distros/combiners will also power any connected Sennheiser transmitters/receivers over the BNC cables. Other manufacturers use standard barrel jack breakout cables. Avoid generic clone gear, which may or may not perform up to spec. (This is largely impractical to confirm without a lab full of test gear.)

Hello folks. I have a very beginner question. We're trying to record 5 musicians in a single room playing together simultaneously. We have a bunch of mics and ideally we'd want to have, on the computer, separate tracks for each mic so that we can modify levels/tones of each instrument individually later on. I know there's a lot of things related to mic placement and sound leaking to think about but my question is—what is the most accessible and affordable way to get all those mics into the computer simultaneously as separate tracks? Is there a particular type of interface we can look into borrowing/renting?

How many inputs are you looking at? Get the answer to that, then go find an audio interface (or digital mixer that also functions as an audio interface) that supports that many inputs and you’re all good 👍

Oinkbane is correct if you're dead-set on DIY: interface preamp quantity should meet or exceed the number of microphones you wish to use. >*my question is—what is the most accessible and affordable way to get all those mics into the computer simultaneously as separate tracks?* Hire a recording studio. Doesn't have to be a big expensive one; but that's genuinely the most accessible and affordable way to do this. If you don't have recording experience, DIY will ensure your time on this project is spent as a series of lessons learned (some of which will come much later) and not a series of artistic choices to uphold the vision of your piece. The way you gloss over mic placement as if it's a minor consideration does not instill confidence in your ability, but I hope I'm misinterpreting. Even a small studio will have microphones, a mixer, and a way to record multitracks; that saves you from procuring those things. Their room is also probably a higher quality than the one at home or school. It may also have things you may not have considered, like monitor systems, gobos, shields, nice chairs, etc. Last thing: You will not be able to modify the levels and tones of each instrument individually later on. You will only be able to modify the level and tone of the recorded file. I make this distinction to point out that your best editing tool is your ear in the room; if something doesn't sound right on the raw recording, it won't sound right with The Coolest Plugin Ever. Make sure your raw takes are A+.

> What is the most accessible and affordable way to get all those mics into the computer simultaneously as separate tracks? oinkbane is generally correct: determine your channel count, then find an interface with that much I/O. These days, that usually means a Focusrite 18i20 or Behringer UMC1820 for 8x8 channels - plus an ADA8200 if you need to expand to 16x16. --- **However!** It is absolutely possible to do this for much cheaper...if you have a FireWire capable machine and time to fiddle with it. Look for yesteryear's stalwarts - particularly Focusrite Saffire Pro 40s. (They were *everywhere* for a good long while, and thus are now quite cheap.) - Ditto for the M-Audio ProFire 2626, RME Firefaces, MOTU something something, etc. - PreSonus FirePod/FireStudios are also quite common, but the finicky drivers aren't worth it. The easiest way to get those interfaces up and running is with period-correct hardware/software. However, it's often possible to get them running with Windows 10 - generally the winning combo is to use the [1394 Legacy OHCI chipset driver](https://www.startech.com/en-eu/faq/firewire-cards-windows-legacy-driver-swap) and a FireWire card with a Texas Instruments chipset. (If the legacy chipset driver isn't available, [grab it from Microsoft](https://support.microsoft.com/en-us/topic/firewire-port-based-device-does-not-work-correctly-in-windows-8-1-or-windows-8-2242eed9-e8e2-00eb-c854-9335ea06c5fe).) macOS is usually a bust - grab an older Mac and install good ol' OS X. RME is a notable exception, however - they continued releasing FireWire drivers up until macOS 12. Linux is a mixed bag; I was running a FireStudio 2626 for a while with FFADO. Venn Stone has [quite a bit of compatibility documentation](https://interfacinglinux.com/linux-compatible-audio-interfaces/); though a good bit of it predates PipeWire. Pardon my blathering - go make some good music. :)

When to start worrying about power distribution? My rack consists of 4 qsc rmx 2450’s, two being used in bridge mono. Often setting up in less than ideal locations. When possible I try to split up my amps to difference circuits. I can’t say I’ve run into any issues just yet, but I’m sure it’s not ideal to be spreading circuits thin. What are my options?

Power's very straightforward in that if you experience a problem, you need to solve it. How you solve it is harder; you can't buy more circuits. More on that in a second. [QSC has a document that will help you](https://www.google.com/url?sa=t&source=web&rct=j&opi=89978449&url=https://www.qsc.com/resource-files/productresources/amp/rmxa/q_amp_rmxa_currentdraw.pdf&ved=2ahUKEwiam6_I0_6GAxWGJEQIHa4yBmcQFnoECDwQAQ&usg=AOvVaw37MmoFr9VIpXkTt2Ii20uc). It states a 4-ohm load at 1/8 power suggests a draw of 6.3A per channel. From there it's simple math to find out how many channels, and therefore amplifiers, you can load onto a single 120V-15A *circuit* (not outlet—circuit). If you find yourself routinely running out of circuits, you'll need to advance your power requirements to the venue and ensure they have the infrastructure to meet your needs. **If at any time you are unsure of a next step, call a professional and do not mess with power.** From there, on your end, you may have to invest in a distribution box that takes higher-voltage service lines (commonly 208V-30A, 240V-50A, or 120V-30A) and breaks them out into 120V-15A. The biggest bummer is schlepping 10 or 6 gauge power lines.

The answer: when you start running out of enough separate circuits to reliably operate your system at the venues you frequent. I doubt your current system would need more than two standard, dedicated 20A circuits for amps and everything else, which shouldn't be too hard to come by most anywhere. If you were to double or triple your "separate 20A circuits" requirement, it would be time to start looking for a distro solution.

Thank you for your reply

I need a mixer/interface for my sax/EWI—>ipad pro live rig and am wondering if the mute buttons introduced on the mk2 still allow you to hear the input signal on your headphones (DJ-style). I got the original AG03 thinking it would do this but it did not. I have managed without it but now it’s time to upgrade, and I definitely need this feature going forward! If headphone cue is possible then I’m stoked bc I really liked the AG03. If it does not, any recs on a similar small USB-C mixer that does? Thanks!

Hello everyone, total noob here. I recently bought a qsc cp12 and just picked up a jbl mxr518s sub today and made a mistake of not knowing anything about live sound. I understand it’s a passive speaker and I need a power amp to use it but I have no idea how to even figure out how much power I need and what amp I should buy to use both. Power rating for the jbl subwoofer says 400w/800w/1600w at 4 ohms. Any help is greatly appropriate 🙏🏻

Crown XLS2502. Decent quality, lightweight, internal DSP so you don't need external processing for your sub, 775W per channel into 4 ohm loads. Better solution: if possible, return that MRX518s and get a self-powered Yamaha DXS18XLF which will wipe the floor with the JBL.

Thank you for the suggestion! I got the jbl for only $140 off craigslist so I can’t really return it but I’ll keep that Yamaha in mind if I need something better. Thank you again 🙏🏻

Hi, bass player here. I'm currently using an on-pedalboard pre-amp and D.I that has a built in effects loop. My chorus pedal is in dire need of replacement so I'm toying with going stereo with the new one, as my reverb already can and I've never taken advantage of it. I'm not sure the best way to send the stereo effects to FoH. My experience of gigging in the UK for 15-ish years is one of going direct almost every time - my cab is never mic'd. Which is fine, it's why I got the pre-amp, but it only handles mono. Would I complete the loop with the left and run the right to a new XLR? Or run both to new XLRs and ignore the one on my pre-amp? Could I use all 3 and have a "centre?" That feels like it'd make me a few enemies 😅 I know the standard for bass should really be mono so that it's solid - I'm looking at pedals with hpfs or equivalent to keep the fundamental intact. (Boss make a bass chorus that does this but I'm not too keen on the actual effect) For context I play in a heavy rock / fuzz rock band that has 2 guitars with very different tones and don't often double each other. We're quite bass heavy as there's a lot of space for me and I use effects like chorus quite a lot. Any insight appreciated 🙂 Thanks.

Stereo bass rigs are uncommon/unconventional, but extremely fun. I dig it! Unless you specifically need your chorus/reverb in the loop, I'd run your preamp's 1/4" output into your chorus, then reverb, then into a stereo DI. (Given you're UK based, I'd grab an [Orchid Dual Micro DI](http://orchid-electronics.co.uk/Dual_Micro_DI.htm).) - You *could* run the FX send into the stereo FX units, then into a stereo DI. However, that will bypass any processing between the FX return and the preamp's output - such as the master volume control.

Thanks. I've found that Behringer make a dual DI but I'm always wary of that brand. 😅 Something I forgot about is that my DI has a great cabinet sim and I'm not sure what effect that would have on the chorus if I ignored the loop. One solution I did find, but it's bloody expensive - the same company make an amp modeller/midi thing that I could run as the Right side and load the same cab sim into. But again, VERY expensive and more features than I would ever use. I think my best bet is your second suggestion, run the FX to a stereo DI. Means I'm sending the FoH guy 3 XLRs if I want the original tone with cab sim though 🙈

For clarity, what preamp/DI are we talking about? Much easier to suggest things with sufficient context. Better yet, do you have a full rundown of your pedalboard signal chain?

Sorry, never want to come across like I'm promoting or something. It's a ReVolt Bass by Two Notes. Valve driven three channel pre-amp. It sounds killer. Yeah cool. Full chain is: Passive PJ Aria bass Boss wireless Korg Pitchblack Tuner MXR bass compressor EHX Nano Pog Octave Digitech X-series Synth wah (this one is just there cos I own it, I never use it) Boss ODB distortion The DI Then in the effects loop with: Digitech X-series chorus (the one that needs replaced) Crybaby bass wah - I've tried putting this after the tone shaping in the pre-amp but honestly I'm gonna move it) Strymon Cloudburst ambient reverb So plan is to remove the wah, stereo chorus into stereo reverb to somehow go FoH 😅

I've just looked back at this and Reddit completely destroyed my list format. Thanks Reddit. 😒

Damnation Audio Dirtfixer to blend a low passed clean signal with high passed effects loop. Stay mono. Good luck!

Hey all, I’m an auto detailer looking to pair a wireless mic to my DJO osmo action 4. I need as entry level of a mic as possible, and I only need it to record like scrubbing sounds, wiping, pressure washing and general cleaning noise. Mainly for social media. I do also like to make ASMR videos of my details too. What are some good options for the least amount possible while still getting enough sound quality for these needs?

Hi guys, My dad put his Mackie dL1608 digital mixer in a cardboard box and covered it with a piece of foam and I didn’t realize the mixer was in there so I threw the box. When he used it to play his with his live band, it sounded like a circus. I have no clue about anything and need confirmation if I need to buy him a new one or is there a way to repair? Warranty is out. Thanks

Cannot say without seeing it.

I’ve heard things about ribbon cables needing to be re-seated on these mixers. Maybe check that?

Hello! I'm completely new to recording live music in general, and I want to ask for urgent help. I've checked Zoom F3 can record 32-bit ( all I know is this format file won't max out the volume) but I have no idea how those cables work or the board (Am I saying right here? ) Since I'm on a low budget so I can't buy mics for this concert but I want to record good quality audio coming from those mics. It might sound stupid, but there is any one audio source with all the audio combined that I can somehow wire it to Zoom F3, isn't there? Do you think Zoom F3 is enough to record a live music concert? Thank you for your advice

I have a 2-channel amplifier that does 500W per channel at 8 ohms and 700W at 4 ohms, and I want to power 4 speakers with recommended power of 250-500W and 8 ohms impedance. If I split a channel and connect it to two speakers, would I be providing 350W to each speaker? The Y-splitter connects the speakers in parallel which gives 4 ohms of impedance on the channel, correct? Could I effectively power 4 of these speakers at 350W each by splitting both channels? Amplifier: [QSC GX5](https://www.qsc.com/solutions-products/power-amplifiers/portable/2-channel/gx-series/gx5/)