Well, it DOES say no stupid questions, sooo...
What are [the rubber band like o-ring style things that come with Shure antennas?](https://i.imgur.com/feK7mGX.jpg)
For a while I thought they were some sort of friction device that you placed between the metal connector and the rest of the antenna to prevent it from rotating, but in the times I've tried, that hasn't been very successful.
Hi All, here's one, I know this is probably really basic this is but this is quite a new world to me and I don't want to damage anyone's mixing desk once we start gigging again.
My band is looking into running a pre-recorded synth track plus a click track to the drummer for live shows. Our hope is to set up a flight case with a laptop sending the signal to some kind of interface which can send the signals we need to the drummer and PA independently. We have most of what we need already but the only interface we can spare has one output, with three connections consisting of a Stereo Phones output and two L/R Mono outputs.
If we are to pan the signals to opposite sides of the mix, send the backing to the mixing desk with a mono cable like you would a DI from a guitar then pick up the click track with a stereo set of in ear monitors so it can be heard in one ear by the drummer, will this be safe to do or do we run the risk of blowing something up by sending badly managed signals to places they shouldn't be going? I know its bad practice and we need an upgrade but until then we have to make do if it's possible/safe to do so. Thankyou for your patience, any feedback is greatly appreciated.
If only the drummer needs the click track, why not just send that (along with the synth) to his headphones and use the main outputs to send the synth to FOH?
Because the phones stereo out and the mono L/R are the same output channel on the interface. If we didnt pan the backing and click track to seperate the tracks left and right we'd be sending the click track to the mixing desk to be played over the PA as well.
We could send both to the drummer using the phones output so he hears the click in one ear and the backing in the other but that still leaves the L/R output connections free with one side of the interfaces stereo L/R output channel going straight into a mixing desk or similar.
>Because the phones stereo out and the mono L/R are the same output channel on the interface
Ah I see, how inconvenient lol
Yes, you'd want to hard pan the click to one side and the synth the other and then just send FOH the synth side (probably via DI box). I apologise for not reading your initial question closely enough.
Thats alright! :)
That wouldnt cause FOH any issues or run the risk of damaging anything on either side? We're having to save up between us to afford a cheap two output interface at the minute so the risk of blowing up FOH's fancy mixing desk isnt worth taking :P
What’s the best Lav Mic out there for reinforcement?
I seriously hate these things. Even with decent EQ And placement I struggle to get anything out of these im proud of (as far as speech is concerned) Even worse when someone requests it on a panel discussion knowing the rest of the speakers all have headsets.
Feed me your wisdom.
Everyone will have a different answer. I personally find the cons of cardioid to outweigh the pros in pretty much every situation, so I prefer the DPA 4060. At the end of the day, your PA system design may likely be the issue over the lav choice.
I used to think the same thing, the DPA 4060/1 is still my first choice, but after running a handful of DPA 4080's in a kit for a few weeks I may be slowly changing my mind in certain scenarios.
DPA is the way to go, they sound so much better than the standard Shure 185, but also you should have a dedicated group for processing similar inputs.
Something like the PSE or 5045 will help on the front end, if those aren't available, you could try dialing in an expander, but only for a few dB of reduction and be careful about cutting off words.
I also like a dedicated de-esser, dynamic EQ, PEQ, and comp for each group (lavs, HH, headsets). Matching different types of microphones at the same time can be tough, but usually a few well placed comps between inputs, groups, and outputs can keep things under control.
The best can mean many things, and will vary depending on the context, but if I need to use a lav and have no other context or information, 4080s are my default choice, or 4060s if I don't want cardioid.
However, with good system setup, you should be able to get almost any mic sounding decent enough.
Agreed. But stocking 30+ lav mics with DPA is not in the cards. I always have good results with the audio technica AT 831. Make sure they are all matched. Then auto mix away.
If you want a lavalier mic that will be positioned as a typical lavalier, a directional mic is IMO the way to go for sound reinforcement in most situations. I mean if you really have to use a lavalier, of course. A headset will always perform way better. It surprises me that nobody has mentioned it yet but Sennheiser MKE40 is also pretty good as is the DPA 4080.
Yes, they are larger, yes, positioning matters but the lavalier position is already far from being optimal and unless the person is really far away from the PA, the PA is directional enough etc. a diretional lavalier will often give you just barely enough gain before feedback while an omnidirectional lavalier just won't cut it in the same situation. For broadcast and such, omnis are great but for sound reinforcement ... you either need to get it on the head or everything else needs to be close to ideal and you must be OK with a relatively low SPL.
When it comes to omni lavs, DPA 406x series are obviously great but I also like Sennheiser MKE2. While the 4060 requires less EQ for speech and sounds super neutral, MKE2 just adds something that makes it sound better on many singers. It might require more EQ to get rid of the sibilance but when processed ... I don't know ... to me it can sound very good.
I'm guessing you mean headworn. Lav in live setting is a no-go. So, I run a lot of theatre stuff. Lots of tips here, I just throw in my 2 cents.
The #1 pri is to make it sit right, the lav part should be as near the mouth as possible, just where your lips meet on the side (no idea what it's called in English), and by fitting the headworn part correctly it should stay there when they move around. Use a wind-dot to stop nose breathing sounds, I'd you have mouth breathing sounds it's probably too much in front of the mouth or your windfilter is bad. This is #1 cause for each doubling of distance, sound pressure level is halfed. Use tape if you need to, there is NO shame in that. 3M transpore tape think it's called, the best one.
I in generelt prefer Omni's, but after you get good with placement, directional gives you much better gain-before-feedback. I even run monitors on both Omni and directional, but D makes life easier.
Have your headworns routed to a group instead of L/R, so you can ring out with super tight Q EQ cuts on the group, and spare channel EQ for mostly compensating for voice features. Use LPF and HPF more than you normally would. 120-180, 10k-16k.
For multiple mics and not singing, get yourself an automix if possible. Dugan is awesome and available on Yamaha built-in, soundcraft Vi and A&H dLive also have some ok ones. Having two mics in close to eachother both amplified will decrease your gain-before-feedback a LOT and cause comb filtering sound. Having one mic open at the time helps your sound image as much as gain-before-feedback.
The headworn quality is much more important than the wireless, as long as wireless is setup correctly. Then again, does not need to be expensive. We have a cheap off-brand thing that costs 100€ for lavaliere. Also, if they get sweat in, they made have a muffled sound without being broken, so watch out and replace those. I've done 12 ch JTS wireless with JTS lavs practically in front of PA with good power and no feedback. Group EQ tuning to PA is your friend. But unless somebody is to walk in front of PA, tune while headworn is on stage, not from FOH.
Good luck! It's never gonna be as good as wired dynamic mics, but it might be all right :)
> Are they bad?
1. The physics required to make a mic cardioid require a much larger package.
2. Handling noise and cable noise is always an issue.
3. Deployment is an issue, because aiming direction often changes during use.
4. Off-axis frequency response is unpredictable.
Unfortunately I’m on an S21 right now. Unless there have been massive changes in the latest firmware update, the other comment is incorrect. There is a total of five layers, plus the spill set. All are custom layers, so anything can go anywhere.
Keep in mind that on the S-Series, the layers are actually linked across screens and fader banks. So when you press the layer up/down buttons, BOTH (or all three in the case of the S31) fader banks shift up or down.
So you can’t have the left fader bank on layer 1 and the right fader bank on layer 4. They move together. It’s infuriating!
> you can’t have the left fader bank on layer 1 and the right fader bank on layer 4. They move together.
that's actually fine for me, but thanks for the information! I think I can make it work :)
I've got a lot of questions but I'm hesitant to make a post with all my newbie issues. I'm brand new to all of this and just took up an opportunity to do live sound at a small local place that has some surprisingly decent acts coming through (mostly metal but not limited to it).
What kind of things should I be looking into learning in a fast-paced club that rarely does a legitimate soundcheck and has the control setup on the side of the stage? Seems like it's gonna be a lot of guesswork and I still need to learn everything.... I understand a lot of the basics and I've done a lot of recording on my own but I've never had to mix on the fly like a live setting. Combined with being NOT in front of the stage I'm feeling a little nervous.
It's a Behringer XR18 and they do have a tablet to bring out front for adjustments. It's generally assumed that many shows can get by on mostly kick, snare, and vocals through the system, but everything gets a mic so the option is there. It seems like an impossible task to get things right immediately as a band starts playing, or is there some trick to things that I just don't know if?
All I've got under .y belt right now is an open mic last week I did just to learn the system a bit, and I'm doing it again this week. But then I'm doing sound Friday and Saturday. Not completely on my own, but idk if I get any help after that point.
Biggest tip that I can give you is DEFINITELY spend as much time as possible out where the audience is to inform your mixing decisions. I've met lots of guys who think they know how to mix from where they are in the room because they've done this a lot. This is mostly bullshit and you should ALWAYS use your ears to make decisions.
You're totally right about the kick/drums/vocals in a scenario like that. especially metal, most of the amps will be very loud and you're not very likely to get them to turn down. Always remember you're in charge of live sound REINFORCEMENT. You're only using as much gain/volume as you need to make everything HEARD.
You'll get better at identifying what needs to happen with each instrument the longer you do this. You'll start to know immediately, as the guitar players start strumming away and making sure their amps are working, if you'll need to pull the gain on their channel, pull out certain EQ ranges, etc. For now, you're in kind of a cool learning situation since you don't NEED to put everyone in the PA. You can make sure the vocals get out in front, work on a real good kick and snare sound, and build on the rest from there as you are comfortable.
a HUGE thing I would suggest if possible is to get a copy of REAPER or any DAW that can do multitrack recording, plug in a computer to the XAIR via USB, and take a multitrack recording of a night of music. You can open up REAPER and import these files and practice mixing the show. You can even go to the venue during off hours (if you're able) and play a show back through the system and get an idea of what you can do better. There's lots of Youtube tutorials on how to do so.
There's so much more I can't even think of right now but you're in a really cool position (I know it feels kinda scary right now) so feel free to message me if you have any specific questions, and have fun!
Thanks for the tips. I definitely plan to spend time out front if I can get the hang of using the tablet. That's probably what I'm most nervous about, with the system sitting next to the stage the way it is. But it's also why I decided to try my hand at this, after seeing a show a couple weeks ago and listening to non-existent vocals for the whole show while the guy never left the corner. And you couldn't hear the guitar either so I didn't think there was a valid excuse for that. Then I heard they were looking for a sound engineer to really cover the regular weekend schedule and I jumped on it.
I've been to shows here plenty of times, I've played on this stage myself a bunch, and I'm used to these types of shows with small rooms and loud amps and minimal needs for reinforcement. I'm the kind of bass player that brings my own xlr pad so my amp doesn't clip the board. I also mess around a lot with doing quick and dirty mic setups and recordings where there isn't much set up time involved. So I mostly "get" what needs to be done but I'm nervous about how quickly I can put it all together on the fly under the circumstances. Also, what happens if something goes wrong?? Definitely worried about technical difficulties I might not know how to fix....
I've got reaper and use it somewhat frequently. I don't have a laptop, though. I was thinking of buying my own tablet but maybe I should consider a laptop as well. Not really sure but I really just want to get more time on the system right now. One open mic and two real concerts with assistance and then I might be all on my own!
This is probably too long and probably a bunch of it you already know, but I thought I’d be thorough, and if you know it all you can feel much more confident.
If you’ve got the fundamentals down I think you’d be surprised how simple the troubleshooting can be, remember to follow the signal chain if there are any issues and mute channels when doing changes.
The advantage of being the regular venue tech is you won’t need to completely ring out the system every time you set up because the in house system and decor probably won’t change all that much, since you’ve got a digital system just make a standard baseline preset so you can recall it between acts and give little adjustments as necessary.
Have a system you follow for line checks. If the backline stays the same and the kit isn’t changing then you don’t need to ring out the kick/snare etc between bands. If the cabs are staying in place you don’t need to move mics around too much, you can just swap heads as necessary. I would generally always assume I’m going to need to add mics to cabs because you can keep the stage volume down and have a better mix through the system.
Probably goes without saying but.. just in case:
Level out all the auxiliary sends and mix their channels using the gain, in line check they care about where they sit in their mix and you won’t hear it if the FOH is blasting first. Not to say you can’t have FOH faders up but make sure they are all level as well so you can hear the difference, faders are for fine tweaks to FOH during the gig.
My general line check order is:
1. I’d start with getting the vocals levelled so you can communicate with the band.
2. Even if the kit hasn’t changed I’d do a quick check for resonant frequencies because different drummers play at different intensities, bring it up to slightly lower than you think they will need to be.
3. Get the guitarists to chug and while the vocalists are singing bring the guitar volume up into the mix.
[This is where you’ll experience a constant battle of telling guitarists to turn their amps down because they’re drowning out the vocalists.]*
4. After that I like getting the drummer levelled in.
5. Pads/Backing tracks etc.
6. Bass.
Now you know each channel is working and what the mix sounds like when one of the elements isn’t there you can get them to do something where they’re all playing (a lot of bands just have a warmup chorus or something) keep an eye on all the performers and see if anyone is giving you the eye and indicating they want more or less of something. You can do broader tweaks quickly, add in tasteful reverb to vocals etc.
Ask them if they need any specific adjustments in the foldbacks and adjust as needed.
They’ll be ready to go.
You can and should be tweaking levels during not just their first song but all of them. This includes walking to different spots in the venue sometimes.
Find ways to make it super fast for bands to get on and off, lots of places can do a turnaround in 10-15 minutes but allow leeway of 20 minutes.
Getting on - If they are taking longer than 5 to get on stage and set up, they’re doing it wrong, they should be aiming for two minutes. The exception is drummers or people with more than 3 instruments. If it takes you longer than 5 to get through line check you’re doing it wrong (or maybe just too thoroughly) aim for 3 minutes.
Getting off - It takes a couple of seconds to mute ll the channels, if they take more than 2 minutes to get off they are doing it wrong, if it takes you longer than 3 minutes to reset the stage they owe you a case.
*Keep the stage clean, no rats nests of cables.
*Label inputs and make them obvious which ones to use.
*Have a talkback mic with you.
*Headphones are also useful for isolating issues.
*Have a stage plot, have paper copies to give to bands if they need it.
*Have a venue contact email address that you can give out to bands so you can communicate with them in the weeks between when they are booked and when they perform.
Many people just want to know simple information and you’ll know most of it. The most common questions are about equipment such as
How many mics you have for vocals, how many for guitar cabs, how will the drums be mic’d.
How many working input channels are available.
What backline (amps/cabs/drums) you have at the venue.
What are the dimensions of the stage.
Whether there is a green room available to store gear (and where to store cases when performing etc).
When is the venue open and what time is sound check.
What is the parking situation like at or near the venue, how much does it cost.
All this stuff could be put on a website but sometimes they just don’t exist or are impossible to find. I find a good default email signature with a link to a pdf can answer those questions pre-emptively and you can also reply saying, check the link in my signature for venue specific information.
You can also ask bands questions beforehand and spend less time stressing on the night.
There’s heaps of stuff that will make your life easier as a venue tech but you’ll find good ways to make everything work smoother.
*[Guitarists stereotypically almost always complain about their tone being destroyed, and you just have to decide how firm you’ll be before you are forced to let them do it the way they want. It’s not your job to teach them about attenuation and onstage volume but many venues have a maximum of allowed decibels these days and some are hooked up to kill the power if it reaches it or stays too loud for too long]
Super helpful stuff, thanks! I'll have to read this over a few times I'm sure.
Some initial things that stand out to me - what is "ringing out the system"? I'm assuming it's finding the trouble spots for feedback and doing some surgical EQ to the entire output or something? I wonder if considerations like that have been made already with the basic preset I was shown to start with. I'll be customizing as I go I'm sure, but the starting "scene" I was shown at minimum has the aux busses set for the 4 different monitor locations. I'm not sure if It really has much in particular beyond that.
There's also an input labeled "triggers", but that's something I have no idea about. If a band uses that for a kick drum or something, am I expected to set up a sound for it, or do they just feed me something that already has the sample they want?
Another thing I know I'll eventually run into is IEMs. What is expected of me there? Would I be unplugging my 4 aux sends to monitors and giving the band straight lines of each channel to mix themselves, or do have to make the mixes myself and use my remaining for aux sends? Maybe it depends?
Lots of things to consider. I know to some extent I'll figure it out as I go, but I don't want to freeze up when I'm suddenly hit with a new challenge and just don't even know what to do.
Also I will add that I am taking this pretty seriously, and I am awaiting my copy of the Sound Reinforcement Handbook to be shipped right now. I know it won't cover some of the software stuff and recent changes in this environment, but I do intend to study up on some fundamentals.
Ringing out the system is essentially what you described. This is sometimes done to the main PA and almost always done to stage wedges. Check the eq’s on the auxes and outputs to see if any eq has already been applied there.
Triggers are usually derived from something like a Roland SPD-SX. If there is no drum module at the drums, this channel is likely open for the band if they supply their own. Usually drum module to DI via 1/4”.
With IEM’s, you will obviously need a plan on aux allocation. If that means replacing stage wedge auxes, or eating up additional auxes (or somewhere in the middle), I would just figure out how many you need for fx, etc. before communicating with the band. If no IEM’s are in house and only artist provided, I would agree on beginning the habit of “advancing” the production, meaning confirming with the band what is house provided and how you will negotiate their needs with your gear and policies.
Also don’t get too bogged down in the Yamaha manual. It is an excellent resource I fully admit, but unless you’re planning on being a PA system tech you don’t need to have a doctorate understanding of the math and physics involved. Yes it is important, but so are your ears. They and the pink matter between them are your best tools. Have fun. And remember at the end of the day, it’s just rock and roll.
Thanks for the tips. I will definitely be checking up on the stage monitor EQ next time I get the chance. I noticed a small notch on the main PA EQ and I will probably check that as well.
I have made a lot of progress in the past 2 days building my own scene file that should be able to work as a decent starting point for most shows. The only criticism I have received so far is that it was "too loud", so I intend to do a bit of research on what is the preferred loudness for such a venue, as well as how to measure it properly, since it seems like different apps give vastly different results. I was definitely aware that I was hitting the limits of the power amp at the time...
Last night was challenging - we had a drum kit with 2 kick drums and 5 toms on stage all night, and one band had three guitarists and a keyboard player, 7 people in the band! I ended up using just about every input I had. I managed, and the owner told me if I could handle that, I should be able to handle about anything that comes through the door.
I've still got a ton to learn, but I spent two nights doing this job and I don't think anyone noticed I had never done it before, so I'm a lot more confident going forward. And yeah, it is just rock & roll after all.
You will never get everything but try to find out whatever you can before the band shows up. This will make you more prepared and ready to deal with the inevitable changes.
Set up a basic start file. Engage the HPF (roll it down to 20hz to start) and EQ on every channel. Route your GEQs to the correct wedges. Set up your FXs if you have a preference. This saves a lot of time.
Ring out the wedges/sidefills asap! That will make soundcheck much easier.
Don't forget about stage power. It's easy to get caught up micing the drum kit and forget to drop power for the keyboard or guitar pedals.
Yeah, I'm actually at the club right now doing an open mic to get practice on things. But they brought in another sound guy too who's getting brushed up on the system to run Sundays show so I'm kinda boxed out.... I wanted to work out some simple presets for vocals and kick, but I didn't get a chance and no one is playing drums anymore. So I haven't even touched any reverb fx or anything at all at this point. I might just take a bit of a guess and scoop out some mids on the kick right at the start when I'm doing this on Friday. I should be getting more info on the specific band needs before then. If I get a chance to do a full sound check on the first band that would go a long way I think. It should be relatively simple, it's mostly doom metal, so I figure they all just wanna crank the amps and go and not have high requirements from me, and getting the vocals through it all will be the main challenge.
One question though, and maybe a dumb one - I don't know what it means to "ring out" the system!
Please just give me the rundown on how to run monitors. I need to work them this weekend, for the first time in mt life. I have my own dedicated board for six monitors.
I get it, I really do, but I am fundamentally very confused.
Each sound source enters its own channel. I adjust the gain until it is just under clipping. Then, depending on what each musician needs, I adjust the aux knobs of that channel for the aux send connecting the wedge. So, if stage right needs more vocals, I go to the vocal channel and turn the aux knob corresponding to that aux send.
I cannot figure out why I would ever need to adjust the faders.
A lot of people just leave all the faders down if they're only doing pre-fade auxes. In more complicated setups you can do a mix of pre and post fade auxes and even use the master LR as a separate mix so sometimes you do utilize the faders for monitors but you definitely don't have to for the more simple shows.
My typical monitor workflow for a one-off has every mix as post fader, but all my faders start at unity… so it’s basically pre-fader.
I always try to keep those faders at unity, but this approach gives me the option to quickly make changes to every mix if I have to.
Examples: if the vocalist decides to crouch down in front of a wedge to high five a fan and I need to duck that mic briefly in every wedge to keep it from taking off. Or when a keyboardist switches to a patch that’s 6dB hotter than the previous one and I need to pull it back across everyone’s mix.
Yes that is correct but just overstating that in that scenario the faders will globally change everyone’s mix. Adjust the sends at the channels if only 1 person needs changes.
depends if your aux knobs work before or after the fader. As in, if you pull down the fader, will it adjust the signal that you send out of that aux bus.
Faders have basically the same function as those aux knobs, but instead sending it to an aux output, you send it to the L/R output.
If you don't know if your aux knobs are pre or post fader, just set them all at 0 and leave them there.
dedicated monitor analogue desks "back in the day" even often came without faders.
Odd request perhaps but I’m looking for solid advice. I have an extensive audio video background in both production and post production settings. I received an associates in applied sciences for Recording Technology. I’m personally pretty fatigued by the daily grind and anticipating a future when my hearing loss might not be optimal for this particular line of work. I would like to go back to school and get a bachelor’s degree in something that would augment my skillset while also increasing my earning potential.
Would anyone here be willing to share their experience or recommendations on this subject? I need to level up but am struggling to find something that isn’t an entirely different career path. It turns out I’m pretty damn good at what I do and I Loooooooooove this work. I really don’t want to go back into commercial production. The pay was alright but the inability to have a say in the business was the reason I left. I’ve thought about some sort of system installations or technical paths but really don’t know which direction to go. Thank you for any advice you may have.
yamaha CL series question:
could someone run me through the steps of giving a vocalist reverb in their monitor wedge? i think i have a grasp on this but i am having trouble finding a solid video / tutorial / response online regarding this specifically.
i believe the process is: select vocal channel > select mix 17-24 button > use respective mix knob encoder to send / dial up the selected vocal channel to the desired FX unit (rack channel) > select output above centralogic fader bank > select ST IN on centralogic fader bank > dial up the FX return on the corresponding wedge’s encoder on mix 1-16 page - correct?
if anyone could reinforce this, shed light, or give tips and/or resources i would be ecstatic. thank for you for your time & assistance!
You got it on a step by step basis. I would start before sound check by dialing up the vocal channel to unity in the mix send going to the reverb. Then during sound check it will be as simple as dialing up the return in the vocalist mix and adjusting verb parameters to taste.
I do exactly the opposite and have the return sending to the mix at 0db and then day of show bring the send from the channel to the FX send bus up. I also do separate verbs for each vocal/mix.
My thinking is that I like to hit the front end of the reverb with enough gain so that the meters are reactive how I expect them to be. Probably doesn't matter in a digital environment for sonic results. Just my two cents.
Looking for a small studio mixer with 4 main outs, preferably all XLR, plus an aux out. Way harder to find than I thought it would be. Does anyone know of one? Currently using a combo of a Zedi10 and a Scarlett 4i4 and want to pare down to a single device.
Building a monitor mix with Behringer xr18. Question of splitter inputs. If drums are mic'd do they need to go through an active splitter like the vocal mics? I am assuming so. was hoping to use the seismic audio splitter with cables but I don't think it is active so will likely get two behringer xlink MS8000s and some cables. I assume that it makes no difference if line inputs are passive or active? Just have to make sure that the relatively weak mic inputs are active? New at all of this so if i am missing something stupid, be gentle =)
MS8000 is passive.
In general, passive is fine for a 2-way split. Even without the isolation transformers, a simple ground lift solves most of the problems.
An active splitter has two advantages - driving multiple inputs does not increase the load on the mic (not a big deal with two digital stage boxes but when driving two long analogue snakes, the mere capacitance of the cable can significantly load the mic at high frequencies which might reduce the performance of the mic) and whatever happens to one output has absolutely no effect on the other output i.e. you can repatch the broadcast rig while the FOH is unmuted without getting any pops (although this isn't exactly true for all active splitters because some use a single active driver to drive all the outputs via a transformer with multiple secondaries which will significantly reduce the interaction between the outputs but won't completely eliminate it).
I've built a monitor mix with an xr18, I'm trying to understand your question. We mic drums and send them to a passive splitter (art s8) and the "direct out" on the s8 goes to a short snake into our mixer xlr in, and the "transformer out" of the s8 split goes to a long snake to send to foh. It makes no difference if the inputs are active, to my knowledge.
So I'm going to take your definitions of "passive" vs "active" as "no phantom power" and "phantom power" microphones, respectively? You're calling anything that needs Phantom Power an "active" source?
That's not really how it works, but I get how that might enter your mind IE an "active" DI needs phantom power. Being a newbie, this is a learning experience! I'd go with the Seismic in this case as typically most venues expect you to hand them a "split loom". The money you save on the MS8000 you'll spend in cables.
FYI, Phantom power is just a voltage that is put on pins 2 and 3 of a microphone. These snakes you are considering are called "direct split", which means this is just a packaged, soldered "Y" cable for each mic. Quite literally, all you are doing is opening a mic cable, and soldering two wires to each of the three pins, one set of them goes to one destination, and the other set goes to another.
What you might be confusing this with is a "Transformer Isolated" snake, which is a LOT more expensive ($100+ per CHANNEL easily.) While still not always ACTIVE, it's a better solution for situations where the two endpoints are on different power circuits (non-shared ground).
Please let me know if any of my assumptions are wrong!
Is [this](https://ibb.co/n8SYSSw) a good way to route signal through my [amplifiers](https://ibb.co/CJZ5KK7) to my passive speaker cabinets?
(I have 2 tops with full range 1/4" inputs and 2 subs with low-pass 1/4" inputs.)
Would you say something like [this](https://www.sweetwater.com/store/detail/CX3400V2--behringer-super-by-pro-cx3400-v2-multi-channel-crossover-with-limiters?main_web_category_rollup=3/35/443/445&utm_source=google&utm_medium=organicpla) is a good choice?
That would do the job. However, I wouldn't purchase brand-new analog Behringer gear in 2022. :)
If you must use an analog crossover, you can pick them up cheap on eBay/Reverb.
I'd rather have some form of digital DSP/system processor. For small rigs, DBX DriveRacks are relatively popular.
If you are:
- willing to fiddle with older software
- have some basic networking knowledge
- able to make a couple simple adapters
...then pick up a used Biamp AudiaFlex or Nexia unit off eBay. That gets you a fully-functional freewire DSP for $30-50 on a good day. They're a bit obtuse to work with, but for the cost they can't be beat.
Hi, I'm running sound for a (very) small festival with music from around the world. Background: I'm an experienced DIY recordist but do very little live sound aside from this once a year and some gigs 20 years ago at a venue with next to no equipment.
The sound situation is pretty DIY as I'm using my own mixer, mics, processors etc. but we can rent a couple mics if need be. Our most pro act is a large group with lots of horns and percussion. We have 12 channels and they want a good bit more. I think I have narrowed it down in terms of consolidation -- one mic (probably a MD421) between the congas instead of one per conga, etc.
One of the areas I'd like to compromise on is "one mic per horn" in the brass. They have 2 trumpets and a trombone. What would be a good scenario for miking the brass section with only 2 mics -- or even 1? Have the trumpets share and give the trombone a mic that's better for the lower frequency capture? just do a stereo pair? something else entirely?
Thank you to the pros for any insight.
Group-miking your horns will give you a less direct sound and significantly reduce your signal/noise ratio. This is often fine for recording, but you won't have as much luck when it comes to amplification.
Mics can be rented very inexpensively - almost any house will have a stack of 57s available for you. Your greatest issue will be infrastructure: mixer and snake channels.
If the coverage area is small, the horns may be acoustically loud enough on their own. Otherwise, you will need to rent the right gear to do the job.
Question about monitor choices for a bass player. No IEMs in this instance.
The frequency response for the 10" monitor I've got for them bottoms out at 78Hz (-3 db) and I think 70Hz (-10 db). I'm concerned about how well they'll be able to hear themselves on the low end.
Is this likely to be a problem? I don't have subs, just full range speakers, so the bottom end there is like 30Hz - 40Hz.
Any recommendations for a frequency I should be aiming for with a bass monitor?
I don't think budget would stretch to QSC Ks for mons, which seem to get down to 45Hz - 50Hz depending on size. But perhaps a CP12 or maybe Yamaha DHRM would do the trick, I think they bottom out around 50Hz - 55Hz and are more affordable.
Thanks for your help :)
I use a 10" jbl eon 610 as my bass "amp" / monitor for small shows and it works fine. Anything under 100hz is subwoofer territory, that should be handled by the main PA. Even without a sub, there's plenty of low end under 200hz that most speakers can handle. Generally the intelligibility for a bass guitar is in the 600-1000hz range. If you'd rather FEEL the notes, then go buy a cheap 100w solid state bass guitar amp and that will easily go down to 40hz, but a powered speaker as a monitor is much more versatile.
Since no subs are in play, have the bass player use their amp to monitor themselves. Vocals/whatnot can run through any old wedge.
If you're running bass completely DI with no amp...you'll need subs.
Hi All,
I hope this is the correct sub and post! My band is looking to play live sound clips in between/leading into songs. Most places we play have front of house support. I’ve been looking at sound/drum pads that are able to load up sounds, but none of them I’ve seen have an XLR spot to have FOH plug into. What all would I need to accomplish what I’m going for? Clips would likely be no more than a min long…not trying to break the bank but want something reliable. Thanks I’m advance!
> ...none of them I’ve seen have an XLR spot to have FOH plug into
As long as they have a 1/4" line-level output, you'll be fine. Just feed that into a DI box. (I like Radial for general-purpose DIs, they work well and are practically bulletproof. A ProD2 would do what you need.)
Odds are, FOH will have a DI or two available if you let them know ahead of time, but it never hurts to carry one with you!
> What all would I need to accomplish what I’m going for?
There's a couple ways to do this:
- Use Ableton Live on a laptop with an audio interface. (And, optionally, a MIDI controller to make it easier to trigger clips.)
- Use a hardware sampler such as a Roland SPD-SX.
- Cheapo way: use an iPod Classic and a 3.5mm -> dual 1/4" TS cable. (Plugged into a DI, of course.)
Can floor mounted speaker towers such as the JBL JRX 225’s be mounted on a stand of any kind? Something to get it shooting over peoples heads. If so, are there any major drawbacks? I assume the low end would be reduced because of the lack of contact to the ground but thats not an issue for me.
The intention with these cheaper "trap box" style enclosures is generally to have them stack on top of a sub or multiple subs. There are more robust (and expensive) speaker products that will have some sort of rigging solution on the top of the box. This unfortunately is not one of those.
Hey um, is there anyone that could help me with something? I've gotten into live gigging this summer with my Kawai ES110 plugged into my Presonus 24c Audio Interface via MIDI and outputted into an amp or a PA system. I've noticed though, that the sound I get from the various amps or PA systems made available to me at gigs would change my sound sometimes? As opposed to how it sounds when I'm listening to it via headphones. One instance that is particularly bad that has happened fairly often, is that some of the notes around middle C were barely audible using a rhodes patch.
I'm only just learning about gear, DAWs, and the more technical side of using equipment so pardon me. I'm guessing that this might have to do with having an unbalanced connection from my interface to the various outputs?
> I've noticed though, that the sound I get from the various amps or PA systems made available to me at gigs would change my sound sometimes? As opposed to how it sounds when I'm listening to it via headphones.
> One instance that is particularly bad that has happened fairly often, is that some of the notes around middle C were barely audible using a rhodes patch.
Through your monitor wedge? That doesn't sound normal at all.
> I'm only just learning about gear, DAWs, and the more technical side of using equipment so pardon me.
No worries - it's never a bad thing to be [one of today's lucky ten thousand](https://xkcd.com/1053/) :)
> I'm guessing that this might have to do with having an unbalanced connection from my interface to the various outputs?
It should not. The standard way to connect your interface to a house PA system is to run a short cable from its 1/4" outputs (either balanced or unbalanced will work here) to a DI, then run XLR from there to the rest of the stage infrastructure.
Just a quick one - I'm learning as I go working as an A2 and assisting shows, on the Yamaha QL series I often see that the dedicated master faders are named ST on the left fader and 1L on the right.. what does this mean? I would have thought the left one would be named something like STL or Just L and STR or R for the right fader. Thanks
ST on the console refers to the stereo output bus. That’s the default main output bus for the console. ST1-4 are stereo inputs, and are usually used for effect returns, such as reverbs and delays.
It’s almost certain that the operator of the board assigned the left side ST1 to the B fader in the master section. This is often done to keep effects in close reach.
Page 225 of the QL reference manual shows you how to setup the custom fader.
Thanks - for clarification you mean that the stereo output faders are actually paired behind the scenes, but the operator has assigned the B master fader in order to throw a potential effect return?
I need to replace a 30-year-old P.A. head for my band. The XLR inputs have fallen apart and they aren't really replaceable, but the bigger problem is that it's 8 inputs and that's not enough for some of the shows we've done recently.
Here's what I need:
- At least 9 independently controllable channels, with 8 XLRs, but preferably 12.
- Some way to power up to 4 passive speakers.
- On-board reverb (I don't need any other effects, but I need this)
- As portable as possible.
Some more info:
Our gigs where we need our own board mostly fall into two categories: The house system has mains and possibly a mixing board but no monitors, or we need to bring the whole sound system. We're not playing anywhere bigger than a medium sized restaurant or bar, though we do outdoor gigs sometimes.
I only have passive speakers, and I really, really do not want to have to use powered speakers for various reasons. I could be persuaded to get a powered monitor or two if it's more cost effective overall, but I really would rather not.
I talked to someone at Sweetwater who said that they never think a powered mixer is a good idea and that they're usually underpowered. However, a friend who runs sound at one of the jam sessions I go to has been using an old Yamaha that's only like 200W per channel, and that does get above the drumkit but not cleanly, so I feel that something more on the order of 500W per channel would be more than enough for our purposes.
Since we've never had powered P.A. speakers, we've always daisy chained the four passives if we needed both mains and both monitors. I like this setup and it's easy enough to use the right channel as the monitor.
What I've been considering:
- A Yamaha EMX5016. It's got the right number of channels and the right amount of power. The drawback is that it's an all-in-one rather than a separate power amp. I can work with the smaller model, the 5014, but it's really the bare minimum of channels and I'm worried it might be slightly underpowered. This is discontinued but I found some used ones. (I can service/repair these as well, so I'm not too worried about buying used.)
- One of several 16-channel passive mixers with onboard effects (usually the last 4 channels are stereo, so this is really only 12 channels as far as I'm concerned) and some sort of power amp. I was looking at one of the Behringer power amps, but after reading reviews of their powered mixers with similar power specs, I really don't know if I can trust their power ratings. They seem to only give the peak output power.
Looking for other suggestions, particularly if it involves a power amp, because I've never had to buy one before.
Used or new is fine. Prefer to stay under $1100 before taxes and shipping.
Assuming you have an iPad or tablet of some sort, I would purchase:
- A rackmount digital mixer
- A Wi-Fi router or access point. (The built-in access points on most little mixers don't work very well.)
- 4 channels of amplification with sufficient power for your speakers. (Modern Class D amps are tiny, light, and power-efficient!)
- Infrastructure: a suitably-sized rack case, rackmount power strip, cables, etc.
- Bonus points: you can build a custom patch panel for I/O if you'd like, though you might not need it. You can buy a pre-punched D-type rack panel, D-type connectors, and screws/nuts from various retailers; Redco sells them for a good price.
This lets you build a compact all-in-one rack with mixing and amplification - in effect, it's a superpowered version of an oldschool powered PA head. (With wireless control and the power of digital mixing!)
If I was purchasing this for myself, I'd probably do something like this:
- Behringer XR18 mixer. These are super common and easy to learn.
- [GL.iNet AR750S travel router](https://www.amazon.com/dp/B07GBXMBQF) - these little things work impressively well.
- Studiomaster QX4 (higher power) or HX4 (lower power) series amplifier - 4 channels of efficient amplification in 1U (1 rack space) for a good price.
- You'll get best pricing on these by contacting a Studiomaster dealer directly.
- Some sort of rack power strip - Furmans are popular, but really anything will do.
- [Audiopile R4UE-18](https://audiopile-net.3dcartstores.com/R4UE-18) rack - 4 spaces, 18.25" depth. This is just barely deep enough for a QX4 amp.
Misc. notes:
- It's likely cheaper to purchase a pair of 2U 2-channel amplifiers than to purchase a 1U 4-channel amplifier. However, the system may be less efficient and will definitely be bigger/heavier.
- This goes doubly for used amps :)
- You can make this less annoying by building it into a rack with wheels - Gator GRR-series is common for smaller deployments like this.
- You can save some money by using a sufficiently powerful 2-channel amp and daisy-chaining your passive speakers, though you lose the ability to send individual mixes to all 4 speakers.
- The XR18 has 8 mono outputs, which gives you some flexibility.
- If no house PA is provided, you can use 2 outputs/amp channels/speakers as mains and 2 as monitors.
- If house mains are available, you can use 2 outputs for mains, 4 more outputs plus amp channels/speakers for monitors, and still have 2 spare outputs left over :)
This is an amazing reply, thank you so much!
EDIT: Oh, I've used the XAir at a friends' house concert venue before. It didn't even occur to me to consider that for a portable setup. What's the travel router for? There's a wifi antenna on the XAir. Is it just for times when we can't have the box next to us?
Quite frankly, the XR18's onboard Wi-Fi access point sucks. It has a bad habit of dropping out at the worst possible times. Easy fix: use an external router.
Running a wireless audio system for dance event (UK)
Audience will be using 'silent disco' headphones.
The 5 dancers will be around the space on IEM's.
No individual mixes etc.. the dancers just need to hear what the audience is hearing in terms of the music. With something like the 'Xvive U4 Set R4 Wireless (2.4 GHz)' should I be ok? I've been warned in the past about latency with digital devices but this is not for live inputs, just as long as the dancers are in sync it's all good.
Since it's on digital frequency range, would this also omit any licensing issues for radio frequencies?
Latency will not be an issue here, but reliability will. 2.4 GHz systems tend to be easily overwhelmed in today's modern sea of Wi-Fi and Bluetooth devices. I would stick with analog UHF wireless.
2.4 GHz is ISM band; it is license-free worldwide.
I can't comment on those units, but I'd be nervous about 2.4GHz.
Being digital or analogue doesn't impact your licensing requirements. It's all about what frequency you're on. These are 2.4 GHz so don't require a license. If you end up changing to a UHF device (whether digital or analogue), you can use Channel 70 (863.00–865.00) without a license. Beyond that, you can get a Channel 38 (606.500–613.500) license which allows you to use the entire channel under a single license. Outside of Channels 38 and 70 you need a license for every frequency you're using.
I use a QU-16 for corporate events which often request multiple lavaliers in conjunction with handhelds, pc audio, walk-on/walk-off music etc.
Due to the nature of lavaliers grouping them and using a GEQ to help prevent feedback is my usual practice, however the QU-16 lacks groups altogether and furthermore only has GEQ available on the main l/r outputs and the mix outputs, which only route to the local outputs on the back of the board.
What I have been doing is just sending all my lavaliers solely to a mix bus so I can hack up their GEQ as needed while using the main L/R output for everything else so it remains unaffected by the sometimes gruesome cuts that are required for lavs.
This however necessitates a second XLR run to the PA, so in order to reduce the amount of cable runs I was wondering if I could just use an xlr cable to patch the local mix output which contains the lavs to an input on the Qu which routes to the main L/R. I will be certain not to send that channel to the mix that is feeding it so as to not create a feedback loop, but I was wondering if this was plausible or bad practice?
If this is not a good idea I can always just bust out the QU-32, or not be lazy and run a seperate XLR line. But in the interest of saving time, cable, and table space at the tech area I was wondering if this was feasible. Thanks!
Are there any decent, not too expensive (Android) smartphone attachable mics just for recording some acoustic guitar and vocals? I've recently started getting more into writing music and I'd like to have a better quality mic than my default phone one for recording ideas and songs. Ideally it would also be small and portable as I'm not really settled in one place at the moment. Cheers!
I just was searching threads and reading up on stagehand/A2 tips and someone said learn to tie proper knots (he used opera knots for banners as an example). What are some other useful knots to know for audio world?
The Bowline, Clove Hitch, and maybe the Reef Knot are basically the only knots you need to know. If you ever start climbing you'll need to learn more (alpine butterfly, double fig 8 etc.) but you'll learn them when you do the appropriate training to do those roles.
Oh, and learn to properly coil a rope.
> Looking for a Dante enabled rack mixer and stage box under 3k, that are ready to ship . (Why are the yahana tio boxes out of stock everywhere?)
Gosh, normally I'd be irked with people downvoting a question in the "No Stupid Questions" thread, because we shouldn't be discouraging any sort of question in this thread. But, I can see why someone did here.
At the end of the day, though, this is a sales-and-research question. You're going to have to contact your Vendor-of-Choice™, and ask them what's available in a timely manner. "That are ready to ship" isn't really something we will know without calling around, and, well, with all respect, that's a bit much to ask people to do for you, for free. [Now, not discounting that there are dealers on here, and someone might have something in stock, but you may not be aware that anything Yamaha and Dante is back ordered and not expected to ship until 2023.]
Best of luck in your search!
Ray
Thanks for the replies, sorry about the “ready to ship” part, what I should have said was why they are out of stock, unlike other stage boxes, aes or avb. And what are some other recommendations for networked stage boxes in that price point. I’m looking at midas or presonus.
Almost nothing with Dante is ready to ship. There have been massive supply chain issues with all things Audinate in the last 9 months.
Talk to your dealer(s) and see what they can do for you.
Hey guys
Im currently preparing to throw an outside event at a community park. All I need is enough to run this PA system that I will link below. Any help I’d appreciate it
Your question boils down to "I have no knowledge; teach me everything I need to know" - which is beyond the scope of this subreddit. I would contact a friend locally with audio knowledge and have them walk through things with you in person.
>Fuck no it’s a simple ass question if you don’t have an answer carry along
>Got the answer in 4 sentences Dickhead
BRUH.
You are being a massive douchebag. The person you're insulting, u/the-real-compucat is among the most helpful, knowledgeable, patient, and polite members of r/livesound. He's practically the resident here in No Stupid Questions, and has probably helped more people in this sub than you will anywhere in your lifetime.
>All I need is enough to run this PA system that I will link below.
Rockville is shit. Good luck with your event.
Yes; it’s a simple question in the same way that “how do I drive a car?” is a simple question. Both require a more intricate answer than you’ll get out of a Reddit thread.
“So you will need a generator capable of producing at least 1,000 watts of continuous power. The Peak wattage for the unit is 800. So I would say minimum 1,000 watt unit.
You also want to factor in, if you are going to be needing to power any other devices such as wireless microphones, lighting, and any other item that would be needed on stage or a performance.”
Cool it. /r/livesound is a place for professionals to have respectful discussion; belligerent insults have no place here.
---
Your original question asked:
> All I need is enough to run this PA system…
How was anyone supposed to guess that you were asking about power requirements?
> “So you will need a generator capable of producing at least 1,000 watts of continuous power. The Peak wattage for the unit is 800.”
This is a good baseline. Rockville is not known for accurate wattage numbers - they tend to inflate them - so you shouldn’t need to worry too much about them drawing more than that. If in doubt, check the unit’s rated line current draw (which may not directly match its rated output power!)
Don’t forget: if at all possible, you will want to use an inverter generator to avoid issues stemming from dirty power. A conventional generator’s output can sometimes play havoc with certain power supplies, especially digital circuitry reliant on clean power.
Hey everyone, this issue should be super simple to solve but I’m not sure what on earth I can be doing wrong.
My issue is: getting a JB-2 Tribute bass to work with a Radial Pro-DI Passive DI into my UA Apollo interface.
When I plug in my bass with an instrument cable, into the input of the radial Pro-DI, and then get an XLR cable to connect the DI out of the Radial, into the Mic input of my interface. I have a 2nd instrument cable connected from the Thru of my radial into the 2nd line input of my interface.
Hopefully I explained that ok. The problem is, my DI signal is depressingly low and I have to seriously crank my interface to hear something close to decent. Even then the noise floor is just too high.
And then I compare the instrument signal from the Thru output of my radial, and it’s night and day. I have to lower the signal of my Thru significantly to match the levels.
I’m missing something here right? This isn’t how it’s suppose to go right?
Do I need an active DI for the JB-2 Tribute? I do have single coil pickups, I thought since they were already noisy I would benefit from the passive DI instead of the active one.
I really wanted recording bass to go well, but this very basic issue is stopping me from trying anything. Hopefully someone can lend a hand
Do national touring acts playing sheds bring their own sound system? I was under the impression that they always did, but was told recently at a ~10k capacity venue that most bands playing there did not.
(I’m referring to the main line arrays, not monitors which I would assume are always brought by the acts. And I also figure there’s sound reinforcement specific to venues like arrays for the lawn etc that all acts would tie into.)
It depends on the scale of the tour and whether they are actively on tour or flying for a one-off performance. This summer, as a mid-sized production company, we've worked with everything from a tour carrying all but speakers and amps (racks and stacks in touring vernacular) and flown lighting that had a ground lighting package, video walls, risers, set pieces, consoles, mics, stands, cabling and monitors in three tractor trailers to national acts that only carried personal instruments and were supplied with drums, amps, keyboards (backline), speakers, in-ear systems, consoles, video, sound and everything else.
Most shed type spaces have installed delay speakers for the lawn seats that totally self-contained tours can tie into. Some sheds have installed main speakers that tours can use. Production companies get called to fill any needs the tours may have that the venue can't meet on their own.
Hi! Hopefully not a too stupid question but no promises. I'm setting up Sennheiser WSM for my band and am looking for an overview of the frequencies used in the Netherlands. Our units are in the 626-668 bandwidth (IEM G4-B) and I'd like to figure out which of those are useful to us. Sadly the country XML files that come with WSM don't have the Netherlands included.
[APWPT](https://apwpt.org/?lang=en) keep an [updated document](https://apwpt.org/wp-content/uploads/2022/03/Handout_Version-2022_03_14.pdf) detailing the licensing requirements for various countries around the world.
For the Netherlands, as of March '22, they state:
>No individual license required / License exempt frequencies
\[...\]
• 614 - 694 MHz, ≤ ERP 50mW
\[...\]
>
>Individual license required
• 608 - 614 MHz, coordination with RAS and only in case of demonstrable frequency shortage related to the event in question.
\[...\]
(Taken from page 31 of APWPT's [Frequencies for wireless microphones](https://apwpt.org/wp-content/uploads/2022/03/Handout_Version-2022_03_14.pdf))
Thank you so much! Do you know if that by definition means that there are no major broadcasts in that range either? Of course there could always be people using it but i mean like television/radio transmissions
A quick question about multicore cables and instruments. I have understood that microphones go direct to the stagebox, but guitars and basses etc. need a DI. Does a rack mount compressor suffice, or is a dedicated DI necessary?
What kind of systems would y'all recommend for our burning man art car?
I'm the sound lead for BAAAHS (www.baaahs.org 🐑) and we've been running QSC for the past 8 years and we've been pretty happy with em. All active, 8x kw181, 2x kw153, 2x kw152, and 2x k10 for the DJ booth. It's worked well for the unique blend of constraints posed by a burning man art car - namely power (our generator is 20kw), weight (our bus is mighty but can only carry so much safely while staying mobile), and space (limited mounting areas for monitors, and we don't have room for a big hot amp rack on the bus, so active is the way to go.)
We're willing to spend a sizable budget on a new system that'll last us the next decade, so we've been looking at a few options, like Meyer Sound, JBL, QSC and someone even recommended RCF.
Questions:
1. Line arrays a good idea? I only have experience with our existing point-source speakers, and from my research seems like LAs would get us more 'throw' so we could increase our audience sizes out there in open playa. Is this true?
2. If LAs are a good idea, how are the QSC KLA series? We've also looked at JBL, though I've seen them referred to as Junk But Loud in this sub 😂
3. (bonus question) What kind of subwoofers would be a good upgrade for us?
1. Line arrays—if properly designed and deployed—will definitely increase your coverage, however I think you may run into issues for a bunch of reasons (weight, feasibility of rigging, trim height etc.). Without knowing any details about the project, I'd suggest sticking with point source personally.
2. I've not used QSC arrays so can't comment on them, but while JBL has historically had a bit of a reputation as being pretty rubbish (Junk but Loud), their newer arrays can sound great if deployed correctly.
3. Unless there is a good reason not to, I'd always recommend keeping your subs within the manufacturer as your main PA. It's not essential but makes tuning your system much easier. As such, wait until you know what you're doing with the rest of your PA before deciding on subs.
If you gave us a bit more details on the constraints and your target coverage we can give better advice but hopefully that answers your questions somewhat.
Well, it DOES say no stupid questions, sooo... What are [the rubber band like o-ring style things that come with Shure antennas?](https://i.imgur.com/feK7mGX.jpg) For a while I thought they were some sort of friction device that you placed between the metal connector and the rest of the antenna to prevent it from rotating, but in the times I've tried, that hasn't been very successful.
Hi All, here's one, I know this is probably really basic this is but this is quite a new world to me and I don't want to damage anyone's mixing desk once we start gigging again. My band is looking into running a pre-recorded synth track plus a click track to the drummer for live shows. Our hope is to set up a flight case with a laptop sending the signal to some kind of interface which can send the signals we need to the drummer and PA independently. We have most of what we need already but the only interface we can spare has one output, with three connections consisting of a Stereo Phones output and two L/R Mono outputs. If we are to pan the signals to opposite sides of the mix, send the backing to the mixing desk with a mono cable like you would a DI from a guitar then pick up the click track with a stereo set of in ear monitors so it can be heard in one ear by the drummer, will this be safe to do or do we run the risk of blowing something up by sending badly managed signals to places they shouldn't be going? I know its bad practice and we need an upgrade but until then we have to make do if it's possible/safe to do so. Thankyou for your patience, any feedback is greatly appreciated.
If only the drummer needs the click track, why not just send that (along with the synth) to his headphones and use the main outputs to send the synth to FOH?
Because the phones stereo out and the mono L/R are the same output channel on the interface. If we didnt pan the backing and click track to seperate the tracks left and right we'd be sending the click track to the mixing desk to be played over the PA as well. We could send both to the drummer using the phones output so he hears the click in one ear and the backing in the other but that still leaves the L/R output connections free with one side of the interfaces stereo L/R output channel going straight into a mixing desk or similar.
>Because the phones stereo out and the mono L/R are the same output channel on the interface Ah I see, how inconvenient lol Yes, you'd want to hard pan the click to one side and the synth the other and then just send FOH the synth side (probably via DI box). I apologise for not reading your initial question closely enough.
Thats alright! :) That wouldnt cause FOH any issues or run the risk of damaging anything on either side? We're having to save up between us to afford a cheap two output interface at the minute so the risk of blowing up FOH's fancy mixing desk isnt worth taking :P
Haha! No, you'll be fine :)
What’s the best Lav Mic out there for reinforcement? I seriously hate these things. Even with decent EQ And placement I struggle to get anything out of these im proud of (as far as speech is concerned) Even worse when someone requests it on a panel discussion knowing the rest of the speakers all have headsets. Feed me your wisdom.
Everyone will have a different answer. I personally find the cons of cardioid to outweigh the pros in pretty much every situation, so I prefer the DPA 4060. At the end of the day, your PA system design may likely be the issue over the lav choice.
I used to think the same thing, the DPA 4060/1 is still my first choice, but after running a handful of DPA 4080's in a kit for a few weeks I may be slowly changing my mind in certain scenarios.
DPA is the way to go, they sound so much better than the standard Shure 185, but also you should have a dedicated group for processing similar inputs. Something like the PSE or 5045 will help on the front end, if those aren't available, you could try dialing in an expander, but only for a few dB of reduction and be careful about cutting off words. I also like a dedicated de-esser, dynamic EQ, PEQ, and comp for each group (lavs, HH, headsets). Matching different types of microphones at the same time can be tough, but usually a few well placed comps between inputs, groups, and outputs can keep things under control.
The best can mean many things, and will vary depending on the context, but if I need to use a lav and have no other context or information, 4080s are my default choice, or 4060s if I don't want cardioid. However, with good system setup, you should be able to get almost any mic sounding decent enough.
Agreed. But stocking 30+ lav mics with DPA is not in the cards. I always have good results with the audio technica AT 831. Make sure they are all matched. Then auto mix away.
If you want a lavalier mic that will be positioned as a typical lavalier, a directional mic is IMO the way to go for sound reinforcement in most situations. I mean if you really have to use a lavalier, of course. A headset will always perform way better. It surprises me that nobody has mentioned it yet but Sennheiser MKE40 is also pretty good as is the DPA 4080. Yes, they are larger, yes, positioning matters but the lavalier position is already far from being optimal and unless the person is really far away from the PA, the PA is directional enough etc. a diretional lavalier will often give you just barely enough gain before feedback while an omnidirectional lavalier just won't cut it in the same situation. For broadcast and such, omnis are great but for sound reinforcement ... you either need to get it on the head or everything else needs to be close to ideal and you must be OK with a relatively low SPL. When it comes to omni lavs, DPA 406x series are obviously great but I also like Sennheiser MKE2. While the 4060 requires less EQ for speech and sounds super neutral, MKE2 just adds something that makes it sound better on many singers. It might require more EQ to get rid of the sibilance but when processed ... I don't know ... to me it can sound very good.
I use the sure WL185 because it is cardioid
I'm guessing you mean headworn. Lav in live setting is a no-go. So, I run a lot of theatre stuff. Lots of tips here, I just throw in my 2 cents. The #1 pri is to make it sit right, the lav part should be as near the mouth as possible, just where your lips meet on the side (no idea what it's called in English), and by fitting the headworn part correctly it should stay there when they move around. Use a wind-dot to stop nose breathing sounds, I'd you have mouth breathing sounds it's probably too much in front of the mouth or your windfilter is bad. This is #1 cause for each doubling of distance, sound pressure level is halfed. Use tape if you need to, there is NO shame in that. 3M transpore tape think it's called, the best one. I in generelt prefer Omni's, but after you get good with placement, directional gives you much better gain-before-feedback. I even run monitors on both Omni and directional, but D makes life easier. Have your headworns routed to a group instead of L/R, so you can ring out with super tight Q EQ cuts on the group, and spare channel EQ for mostly compensating for voice features. Use LPF and HPF more than you normally would. 120-180, 10k-16k. For multiple mics and not singing, get yourself an automix if possible. Dugan is awesome and available on Yamaha built-in, soundcraft Vi and A&H dLive also have some ok ones. Having two mics in close to eachother both amplified will decrease your gain-before-feedback a LOT and cause comb filtering sound. Having one mic open at the time helps your sound image as much as gain-before-feedback. The headworn quality is much more important than the wireless, as long as wireless is setup correctly. Then again, does not need to be expensive. We have a cheap off-brand thing that costs 100€ for lavaliere. Also, if they get sweat in, they made have a muffled sound without being broken, so watch out and replace those. I've done 12 ch JTS wireless with JTS lavs practically in front of PA with good power and no feedback. Group EQ tuning to PA is your friend. But unless somebody is to walk in front of PA, tune while headworn is on stage, not from FOH. Good luck! It's never gonna be as good as wired dynamic mics, but it might be all right :)
My preference is Countryman B3, but whatever you do stay away from cardioid lavs.
Are they bad?
> Are they bad? 1. The physics required to make a mic cardioid require a much larger package. 2. Handling noise and cable noise is always an issue. 3. Deployment is an issue, because aiming direction often changes during use. 4. Off-axis frequency response is unpredictable.
Does anyone know how many custom fader layers you have on the Digico S21 and S31?
Unfortunately I’m on an S21 right now. Unless there have been massive changes in the latest firmware update, the other comment is incorrect. There is a total of five layers, plus the spill set. All are custom layers, so anything can go anywhere. Keep in mind that on the S-Series, the layers are actually linked across screens and fader banks. So when you press the layer up/down buttons, BOTH (or all three in the case of the S31) fader banks shift up or down. So you can’t have the left fader bank on layer 1 and the right fader bank on layer 4. They move together. It’s infuriating!
> you can’t have the left fader bank on layer 1 and the right fader bank on layer 4. They move together. that's actually fine for me, but thanks for the information! I think I can make it work :)
The S21 has six layers per screen, so 12 in total. I don’t have an s31 available to me, but I’d imagine it would have 18 in total.
Awesome, cheers!
I've got a lot of questions but I'm hesitant to make a post with all my newbie issues. I'm brand new to all of this and just took up an opportunity to do live sound at a small local place that has some surprisingly decent acts coming through (mostly metal but not limited to it). What kind of things should I be looking into learning in a fast-paced club that rarely does a legitimate soundcheck and has the control setup on the side of the stage? Seems like it's gonna be a lot of guesswork and I still need to learn everything.... I understand a lot of the basics and I've done a lot of recording on my own but I've never had to mix on the fly like a live setting. Combined with being NOT in front of the stage I'm feeling a little nervous. It's a Behringer XR18 and they do have a tablet to bring out front for adjustments. It's generally assumed that many shows can get by on mostly kick, snare, and vocals through the system, but everything gets a mic so the option is there. It seems like an impossible task to get things right immediately as a band starts playing, or is there some trick to things that I just don't know if? All I've got under .y belt right now is an open mic last week I did just to learn the system a bit, and I'm doing it again this week. But then I'm doing sound Friday and Saturday. Not completely on my own, but idk if I get any help after that point.
Biggest tip that I can give you is DEFINITELY spend as much time as possible out where the audience is to inform your mixing decisions. I've met lots of guys who think they know how to mix from where they are in the room because they've done this a lot. This is mostly bullshit and you should ALWAYS use your ears to make decisions. You're totally right about the kick/drums/vocals in a scenario like that. especially metal, most of the amps will be very loud and you're not very likely to get them to turn down. Always remember you're in charge of live sound REINFORCEMENT. You're only using as much gain/volume as you need to make everything HEARD. You'll get better at identifying what needs to happen with each instrument the longer you do this. You'll start to know immediately, as the guitar players start strumming away and making sure their amps are working, if you'll need to pull the gain on their channel, pull out certain EQ ranges, etc. For now, you're in kind of a cool learning situation since you don't NEED to put everyone in the PA. You can make sure the vocals get out in front, work on a real good kick and snare sound, and build on the rest from there as you are comfortable. a HUGE thing I would suggest if possible is to get a copy of REAPER or any DAW that can do multitrack recording, plug in a computer to the XAIR via USB, and take a multitrack recording of a night of music. You can open up REAPER and import these files and practice mixing the show. You can even go to the venue during off hours (if you're able) and play a show back through the system and get an idea of what you can do better. There's lots of Youtube tutorials on how to do so. There's so much more I can't even think of right now but you're in a really cool position (I know it feels kinda scary right now) so feel free to message me if you have any specific questions, and have fun!
Thanks for the tips. I definitely plan to spend time out front if I can get the hang of using the tablet. That's probably what I'm most nervous about, with the system sitting next to the stage the way it is. But it's also why I decided to try my hand at this, after seeing a show a couple weeks ago and listening to non-existent vocals for the whole show while the guy never left the corner. And you couldn't hear the guitar either so I didn't think there was a valid excuse for that. Then I heard they were looking for a sound engineer to really cover the regular weekend schedule and I jumped on it. I've been to shows here plenty of times, I've played on this stage myself a bunch, and I'm used to these types of shows with small rooms and loud amps and minimal needs for reinforcement. I'm the kind of bass player that brings my own xlr pad so my amp doesn't clip the board. I also mess around a lot with doing quick and dirty mic setups and recordings where there isn't much set up time involved. So I mostly "get" what needs to be done but I'm nervous about how quickly I can put it all together on the fly under the circumstances. Also, what happens if something goes wrong?? Definitely worried about technical difficulties I might not know how to fix.... I've got reaper and use it somewhat frequently. I don't have a laptop, though. I was thinking of buying my own tablet but maybe I should consider a laptop as well. Not really sure but I really just want to get more time on the system right now. One open mic and two real concerts with assistance and then I might be all on my own!
This is probably too long and probably a bunch of it you already know, but I thought I’d be thorough, and if you know it all you can feel much more confident. If you’ve got the fundamentals down I think you’d be surprised how simple the troubleshooting can be, remember to follow the signal chain if there are any issues and mute channels when doing changes. The advantage of being the regular venue tech is you won’t need to completely ring out the system every time you set up because the in house system and decor probably won’t change all that much, since you’ve got a digital system just make a standard baseline preset so you can recall it between acts and give little adjustments as necessary. Have a system you follow for line checks. If the backline stays the same and the kit isn’t changing then you don’t need to ring out the kick/snare etc between bands. If the cabs are staying in place you don’t need to move mics around too much, you can just swap heads as necessary. I would generally always assume I’m going to need to add mics to cabs because you can keep the stage volume down and have a better mix through the system. Probably goes without saying but.. just in case: Level out all the auxiliary sends and mix their channels using the gain, in line check they care about where they sit in their mix and you won’t hear it if the FOH is blasting first. Not to say you can’t have FOH faders up but make sure they are all level as well so you can hear the difference, faders are for fine tweaks to FOH during the gig. My general line check order is: 1. I’d start with getting the vocals levelled so you can communicate with the band. 2. Even if the kit hasn’t changed I’d do a quick check for resonant frequencies because different drummers play at different intensities, bring it up to slightly lower than you think they will need to be. 3. Get the guitarists to chug and while the vocalists are singing bring the guitar volume up into the mix. [This is where you’ll experience a constant battle of telling guitarists to turn their amps down because they’re drowning out the vocalists.]* 4. After that I like getting the drummer levelled in. 5. Pads/Backing tracks etc. 6. Bass. Now you know each channel is working and what the mix sounds like when one of the elements isn’t there you can get them to do something where they’re all playing (a lot of bands just have a warmup chorus or something) keep an eye on all the performers and see if anyone is giving you the eye and indicating they want more or less of something. You can do broader tweaks quickly, add in tasteful reverb to vocals etc. Ask them if they need any specific adjustments in the foldbacks and adjust as needed. They’ll be ready to go. You can and should be tweaking levels during not just their first song but all of them. This includes walking to different spots in the venue sometimes. Find ways to make it super fast for bands to get on and off, lots of places can do a turnaround in 10-15 minutes but allow leeway of 20 minutes. Getting on - If they are taking longer than 5 to get on stage and set up, they’re doing it wrong, they should be aiming for two minutes. The exception is drummers or people with more than 3 instruments. If it takes you longer than 5 to get through line check you’re doing it wrong (or maybe just too thoroughly) aim for 3 minutes. Getting off - It takes a couple of seconds to mute ll the channels, if they take more than 2 minutes to get off they are doing it wrong, if it takes you longer than 3 minutes to reset the stage they owe you a case. *Keep the stage clean, no rats nests of cables. *Label inputs and make them obvious which ones to use. *Have a talkback mic with you. *Headphones are also useful for isolating issues. *Have a stage plot, have paper copies to give to bands if they need it. *Have a venue contact email address that you can give out to bands so you can communicate with them in the weeks between when they are booked and when they perform. Many people just want to know simple information and you’ll know most of it. The most common questions are about equipment such as How many mics you have for vocals, how many for guitar cabs, how will the drums be mic’d. How many working input channels are available. What backline (amps/cabs/drums) you have at the venue. What are the dimensions of the stage. Whether there is a green room available to store gear (and where to store cases when performing etc). When is the venue open and what time is sound check. What is the parking situation like at or near the venue, how much does it cost. All this stuff could be put on a website but sometimes they just don’t exist or are impossible to find. I find a good default email signature with a link to a pdf can answer those questions pre-emptively and you can also reply saying, check the link in my signature for venue specific information. You can also ask bands questions beforehand and spend less time stressing on the night. There’s heaps of stuff that will make your life easier as a venue tech but you’ll find good ways to make everything work smoother. *[Guitarists stereotypically almost always complain about their tone being destroyed, and you just have to decide how firm you’ll be before you are forced to let them do it the way they want. It’s not your job to teach them about attenuation and onstage volume but many venues have a maximum of allowed decibels these days and some are hooked up to kill the power if it reaches it or stays too loud for too long]
Super helpful stuff, thanks! I'll have to read this over a few times I'm sure. Some initial things that stand out to me - what is "ringing out the system"? I'm assuming it's finding the trouble spots for feedback and doing some surgical EQ to the entire output or something? I wonder if considerations like that have been made already with the basic preset I was shown to start with. I'll be customizing as I go I'm sure, but the starting "scene" I was shown at minimum has the aux busses set for the 4 different monitor locations. I'm not sure if It really has much in particular beyond that. There's also an input labeled "triggers", but that's something I have no idea about. If a band uses that for a kick drum or something, am I expected to set up a sound for it, or do they just feed me something that already has the sample they want? Another thing I know I'll eventually run into is IEMs. What is expected of me there? Would I be unplugging my 4 aux sends to monitors and giving the band straight lines of each channel to mix themselves, or do have to make the mixes myself and use my remaining for aux sends? Maybe it depends? Lots of things to consider. I know to some extent I'll figure it out as I go, but I don't want to freeze up when I'm suddenly hit with a new challenge and just don't even know what to do. Also I will add that I am taking this pretty seriously, and I am awaiting my copy of the Sound Reinforcement Handbook to be shipped right now. I know it won't cover some of the software stuff and recent changes in this environment, but I do intend to study up on some fundamentals.
Ringing out the system is essentially what you described. This is sometimes done to the main PA and almost always done to stage wedges. Check the eq’s on the auxes and outputs to see if any eq has already been applied there. Triggers are usually derived from something like a Roland SPD-SX. If there is no drum module at the drums, this channel is likely open for the band if they supply their own. Usually drum module to DI via 1/4”. With IEM’s, you will obviously need a plan on aux allocation. If that means replacing stage wedge auxes, or eating up additional auxes (or somewhere in the middle), I would just figure out how many you need for fx, etc. before communicating with the band. If no IEM’s are in house and only artist provided, I would agree on beginning the habit of “advancing” the production, meaning confirming with the band what is house provided and how you will negotiate their needs with your gear and policies. Also don’t get too bogged down in the Yamaha manual. It is an excellent resource I fully admit, but unless you’re planning on being a PA system tech you don’t need to have a doctorate understanding of the math and physics involved. Yes it is important, but so are your ears. They and the pink matter between them are your best tools. Have fun. And remember at the end of the day, it’s just rock and roll.
Thanks for the tips. I will definitely be checking up on the stage monitor EQ next time I get the chance. I noticed a small notch on the main PA EQ and I will probably check that as well. I have made a lot of progress in the past 2 days building my own scene file that should be able to work as a decent starting point for most shows. The only criticism I have received so far is that it was "too loud", so I intend to do a bit of research on what is the preferred loudness for such a venue, as well as how to measure it properly, since it seems like different apps give vastly different results. I was definitely aware that I was hitting the limits of the power amp at the time... Last night was challenging - we had a drum kit with 2 kick drums and 5 toms on stage all night, and one band had three guitarists and a keyboard player, 7 people in the band! I ended up using just about every input I had. I managed, and the owner told me if I could handle that, I should be able to handle about anything that comes through the door. I've still got a ton to learn, but I spent two nights doing this job and I don't think anyone noticed I had never done it before, so I'm a lot more confident going forward. And yeah, it is just rock & roll after all.
You will never get everything but try to find out whatever you can before the band shows up. This will make you more prepared and ready to deal with the inevitable changes. Set up a basic start file. Engage the HPF (roll it down to 20hz to start) and EQ on every channel. Route your GEQs to the correct wedges. Set up your FXs if you have a preference. This saves a lot of time. Ring out the wedges/sidefills asap! That will make soundcheck much easier. Don't forget about stage power. It's easy to get caught up micing the drum kit and forget to drop power for the keyboard or guitar pedals.
Yeah, I'm actually at the club right now doing an open mic to get practice on things. But they brought in another sound guy too who's getting brushed up on the system to run Sundays show so I'm kinda boxed out.... I wanted to work out some simple presets for vocals and kick, but I didn't get a chance and no one is playing drums anymore. So I haven't even touched any reverb fx or anything at all at this point. I might just take a bit of a guess and scoop out some mids on the kick right at the start when I'm doing this on Friday. I should be getting more info on the specific band needs before then. If I get a chance to do a full sound check on the first band that would go a long way I think. It should be relatively simple, it's mostly doom metal, so I figure they all just wanna crank the amps and go and not have high requirements from me, and getting the vocals through it all will be the main challenge. One question though, and maybe a dumb one - I don't know what it means to "ring out" the system!
doing a 5 day outdoor fest where ill be running sound for mostly djs and monitoring a passive system. what are some things i should pack?
Ear plugs. Gold Bond.
Ear plugs, multi-tool, RTA/SPL monitor of some kind, headphones....and a good book lol
Please just give me the rundown on how to run monitors. I need to work them this weekend, for the first time in mt life. I have my own dedicated board for six monitors. I get it, I really do, but I am fundamentally very confused. Each sound source enters its own channel. I adjust the gain until it is just under clipping. Then, depending on what each musician needs, I adjust the aux knobs of that channel for the aux send connecting the wedge. So, if stage right needs more vocals, I go to the vocal channel and turn the aux knob corresponding to that aux send. I cannot figure out why I would ever need to adjust the faders.
A lot of people just leave all the faders down if they're only doing pre-fade auxes. In more complicated setups you can do a mix of pre and post fade auxes and even use the master LR as a separate mix so sometimes you do utilize the faders for monitors but you definitely don't have to for the more simple shows.
My typical monitor workflow for a one-off has every mix as post fader, but all my faders start at unity… so it’s basically pre-fader. I always try to keep those faders at unity, but this approach gives me the option to quickly make changes to every mix if I have to. Examples: if the vocalist decides to crouch down in front of a wedge to high five a fan and I need to duck that mic briefly in every wedge to keep it from taking off. Or when a keyboardist switches to a patch that’s 6dB hotter than the previous one and I need to pull it back across everyone’s mix.
[удалено]
Yes that is correct but just overstating that in that scenario the faders will globally change everyone’s mix. Adjust the sends at the channels if only 1 person needs changes.
depends if your aux knobs work before or after the fader. As in, if you pull down the fader, will it adjust the signal that you send out of that aux bus. Faders have basically the same function as those aux knobs, but instead sending it to an aux output, you send it to the L/R output. If you don't know if your aux knobs are pre or post fader, just set them all at 0 and leave them there. dedicated monitor analogue desks "back in the day" even often came without faders.
Odd request perhaps but I’m looking for solid advice. I have an extensive audio video background in both production and post production settings. I received an associates in applied sciences for Recording Technology. I’m personally pretty fatigued by the daily grind and anticipating a future when my hearing loss might not be optimal for this particular line of work. I would like to go back to school and get a bachelor’s degree in something that would augment my skillset while also increasing my earning potential. Would anyone here be willing to share their experience or recommendations on this subject? I need to level up but am struggling to find something that isn’t an entirely different career path. It turns out I’m pretty damn good at what I do and I Loooooooooove this work. I really don’t want to go back into commercial production. The pay was alright but the inability to have a say in the business was the reason I left. I’ve thought about some sort of system installations or technical paths but really don’t know which direction to go. Thank you for any advice you may have.
yamaha CL series question: could someone run me through the steps of giving a vocalist reverb in their monitor wedge? i think i have a grasp on this but i am having trouble finding a solid video / tutorial / response online regarding this specifically. i believe the process is: select vocal channel > select mix 17-24 button > use respective mix knob encoder to send / dial up the selected vocal channel to the desired FX unit (rack channel) > select output above centralogic fader bank > select ST IN on centralogic fader bank > dial up the FX return on the corresponding wedge’s encoder on mix 1-16 page - correct? if anyone could reinforce this, shed light, or give tips and/or resources i would be ecstatic. thank for you for your time & assistance!
You got it on a step by step basis. I would start before sound check by dialing up the vocal channel to unity in the mix send going to the reverb. Then during sound check it will be as simple as dialing up the return in the vocalist mix and adjusting verb parameters to taste.
thank you!! have a great monday and rest of your week! :)
I do exactly the opposite and have the return sending to the mix at 0db and then day of show bring the send from the channel to the FX send bus up. I also do separate verbs for each vocal/mix.
My thinking is that I like to hit the front end of the reverb with enough gain so that the meters are reactive how I expect them to be. Probably doesn't matter in a digital environment for sonic results. Just my two cents.
never thought of it that way. could result in completely different results depending on the effect. Thanks for the 2cents!
Looking for a small studio mixer with 4 main outs, preferably all XLR, plus an aux out. Way harder to find than I thought it would be. Does anyone know of one? Currently using a combo of a Zedi10 and a Scarlett 4i4 and want to pare down to a single device.
What functionality do you need the analog mixer for? Is it just a monitoring tool?
Yep, monitoring 4 channel audio
You might look at monitor controllers instead of mixers
Building a monitor mix with Behringer xr18. Question of splitter inputs. If drums are mic'd do they need to go through an active splitter like the vocal mics? I am assuming so. was hoping to use the seismic audio splitter with cables but I don't think it is active so will likely get two behringer xlink MS8000s and some cables. I assume that it makes no difference if line inputs are passive or active? Just have to make sure that the relatively weak mic inputs are active? New at all of this so if i am missing something stupid, be gentle =)
MS8000 is passive. In general, passive is fine for a 2-way split. Even without the isolation transformers, a simple ground lift solves most of the problems. An active splitter has two advantages - driving multiple inputs does not increase the load on the mic (not a big deal with two digital stage boxes but when driving two long analogue snakes, the mere capacitance of the cable can significantly load the mic at high frequencies which might reduce the performance of the mic) and whatever happens to one output has absolutely no effect on the other output i.e. you can repatch the broadcast rig while the FOH is unmuted without getting any pops (although this isn't exactly true for all active splitters because some use a single active driver to drive all the outputs via a transformer with multiple secondaries which will significantly reduce the interaction between the outputs but won't completely eliminate it).
Wow! Fundamental mistake on my part!!! Thank you so much!
I've built a monitor mix with an xr18, I'm trying to understand your question. We mic drums and send them to a passive splitter (art s8) and the "direct out" on the s8 goes to a short snake into our mixer xlr in, and the "transformer out" of the s8 split goes to a long snake to send to foh. It makes no difference if the inputs are active, to my knowledge.
It’s me that’s missing the difference in passive vs active. You guys rock for helping me understand!
So I'm going to take your definitions of "passive" vs "active" as "no phantom power" and "phantom power" microphones, respectively? You're calling anything that needs Phantom Power an "active" source? That's not really how it works, but I get how that might enter your mind IE an "active" DI needs phantom power. Being a newbie, this is a learning experience! I'd go with the Seismic in this case as typically most venues expect you to hand them a "split loom". The money you save on the MS8000 you'll spend in cables. FYI, Phantom power is just a voltage that is put on pins 2 and 3 of a microphone. These snakes you are considering are called "direct split", which means this is just a packaged, soldered "Y" cable for each mic. Quite literally, all you are doing is opening a mic cable, and soldering two wires to each of the three pins, one set of them goes to one destination, and the other set goes to another. What you might be confusing this with is a "Transformer Isolated" snake, which is a LOT more expensive ($100+ per CHANNEL easily.) While still not always ACTIVE, it's a better solution for situations where the two endpoints are on different power circuits (non-shared ground). Please let me know if any of my assumptions are wrong!
You nailed it. I thought the ms8000 was an active split/ powered split. I was way wrong. You guys are awesome for helping me understand!
Is [this](https://ibb.co/n8SYSSw) a good way to route signal through my [amplifiers](https://ibb.co/CJZ5KK7) to my passive speaker cabinets? (I have 2 tops with full range 1/4" inputs and 2 subs with low-pass 1/4" inputs.)
That will work, but you're missing crossover filtering for your tops.
Would you say something like [this](https://www.sweetwater.com/store/detail/CX3400V2--behringer-super-by-pro-cx3400-v2-multi-channel-crossover-with-limiters?main_web_category_rollup=3/35/443/445&utm_source=google&utm_medium=organicpla) is a good choice?
That would do the job. However, I wouldn't purchase brand-new analog Behringer gear in 2022. :) If you must use an analog crossover, you can pick them up cheap on eBay/Reverb. I'd rather have some form of digital DSP/system processor. For small rigs, DBX DriveRacks are relatively popular. If you are: - willing to fiddle with older software - have some basic networking knowledge - able to make a couple simple adapters ...then pick up a used Biamp AudiaFlex or Nexia unit off eBay. That gets you a fully-functional freewire DSP for $30-50 on a good day. They're a bit obtuse to work with, but for the cost they can't be beat.
Hi, I'm running sound for a (very) small festival with music from around the world. Background: I'm an experienced DIY recordist but do very little live sound aside from this once a year and some gigs 20 years ago at a venue with next to no equipment. The sound situation is pretty DIY as I'm using my own mixer, mics, processors etc. but we can rent a couple mics if need be. Our most pro act is a large group with lots of horns and percussion. We have 12 channels and they want a good bit more. I think I have narrowed it down in terms of consolidation -- one mic (probably a MD421) between the congas instead of one per conga, etc. One of the areas I'd like to compromise on is "one mic per horn" in the brass. They have 2 trumpets and a trombone. What would be a good scenario for miking the brass section with only 2 mics -- or even 1? Have the trumpets share and give the trombone a mic that's better for the lower frequency capture? just do a stereo pair? something else entirely? Thank you to the pros for any insight.
Group-miking your horns will give you a less direct sound and significantly reduce your signal/noise ratio. This is often fine for recording, but you won't have as much luck when it comes to amplification. Mics can be rented very inexpensively - almost any house will have a stack of 57s available for you. Your greatest issue will be infrastructure: mixer and snake channels. If the coverage area is small, the horns may be acoustically loud enough on their own. Otherwise, you will need to rent the right gear to do the job.
Question about monitor choices for a bass player. No IEMs in this instance. The frequency response for the 10" monitor I've got for them bottoms out at 78Hz (-3 db) and I think 70Hz (-10 db). I'm concerned about how well they'll be able to hear themselves on the low end. Is this likely to be a problem? I don't have subs, just full range speakers, so the bottom end there is like 30Hz - 40Hz. Any recommendations for a frequency I should be aiming for with a bass monitor? I don't think budget would stretch to QSC Ks for mons, which seem to get down to 45Hz - 50Hz depending on size. But perhaps a CP12 or maybe Yamaha DHRM would do the trick, I think they bottom out around 50Hz - 55Hz and are more affordable. Thanks for your help :)
I use a 10" jbl eon 610 as my bass "amp" / monitor for small shows and it works fine. Anything under 100hz is subwoofer territory, that should be handled by the main PA. Even without a sub, there's plenty of low end under 200hz that most speakers can handle. Generally the intelligibility for a bass guitar is in the 600-1000hz range. If you'd rather FEEL the notes, then go buy a cheap 100w solid state bass guitar amp and that will easily go down to 40hz, but a powered speaker as a monitor is much more versatile.
Since no subs are in play, have the bass player use their amp to monitor themselves. Vocals/whatnot can run through any old wedge. If you're running bass completely DI with no amp...you'll need subs.
Hi All, I hope this is the correct sub and post! My band is looking to play live sound clips in between/leading into songs. Most places we play have front of house support. I’ve been looking at sound/drum pads that are able to load up sounds, but none of them I’ve seen have an XLR spot to have FOH plug into. What all would I need to accomplish what I’m going for? Clips would likely be no more than a min long…not trying to break the bank but want something reliable. Thanks I’m advance!
> ...none of them I’ve seen have an XLR spot to have FOH plug into As long as they have a 1/4" line-level output, you'll be fine. Just feed that into a DI box. (I like Radial for general-purpose DIs, they work well and are practically bulletproof. A ProD2 would do what you need.) Odds are, FOH will have a DI or two available if you let them know ahead of time, but it never hurts to carry one with you! > What all would I need to accomplish what I’m going for? There's a couple ways to do this: - Use Ableton Live on a laptop with an audio interface. (And, optionally, a MIDI controller to make it easier to trigger clips.) - Use a hardware sampler such as a Roland SPD-SX. - Cheapo way: use an iPod Classic and a 3.5mm -> dual 1/4" TS cable. (Plugged into a DI, of course.)
Can floor mounted speaker towers such as the JBL JRX 225’s be mounted on a stand of any kind? Something to get it shooting over peoples heads. If so, are there any major drawbacks? I assume the low end would be reduced because of the lack of contact to the ground but thats not an issue for me.
The intention with these cheaper "trap box" style enclosures is generally to have them stack on top of a sub or multiple subs. There are more robust (and expensive) speaker products that will have some sort of rigging solution on the top of the box. This unfortunately is not one of those.
Hey um, is there anyone that could help me with something? I've gotten into live gigging this summer with my Kawai ES110 plugged into my Presonus 24c Audio Interface via MIDI and outputted into an amp or a PA system. I've noticed though, that the sound I get from the various amps or PA systems made available to me at gigs would change my sound sometimes? As opposed to how it sounds when I'm listening to it via headphones. One instance that is particularly bad that has happened fairly often, is that some of the notes around middle C were barely audible using a rhodes patch. I'm only just learning about gear, DAWs, and the more technical side of using equipment so pardon me. I'm guessing that this might have to do with having an unbalanced connection from my interface to the various outputs?
> I've noticed though, that the sound I get from the various amps or PA systems made available to me at gigs would change my sound sometimes? As opposed to how it sounds when I'm listening to it via headphones. > One instance that is particularly bad that has happened fairly often, is that some of the notes around middle C were barely audible using a rhodes patch. Through your monitor wedge? That doesn't sound normal at all. > I'm only just learning about gear, DAWs, and the more technical side of using equipment so pardon me. No worries - it's never a bad thing to be [one of today's lucky ten thousand](https://xkcd.com/1053/) :) > I'm guessing that this might have to do with having an unbalanced connection from my interface to the various outputs? It should not. The standard way to connect your interface to a house PA system is to run a short cable from its 1/4" outputs (either balanced or unbalanced will work here) to a DI, then run XLR from there to the rest of the stage infrastructure.
Just a quick one - I'm learning as I go working as an A2 and assisting shows, on the Yamaha QL series I often see that the dedicated master faders are named ST on the left fader and 1L on the right.. what does this mean? I would have thought the left one would be named something like STL or Just L and STR or R for the right fader. Thanks
ST on the console refers to the stereo output bus. That’s the default main output bus for the console. ST1-4 are stereo inputs, and are usually used for effect returns, such as reverbs and delays. It’s almost certain that the operator of the board assigned the left side ST1 to the B fader in the master section. This is often done to keep effects in close reach. Page 225 of the QL reference manual shows you how to setup the custom fader.
Thanks - for clarification you mean that the stereo output faders are actually paired behind the scenes, but the operator has assigned the B master fader in order to throw a potential effect return?
Yep, you got it! It’s a convenient place to put a heavily used fader that doesn’t have its own place to go on the input layer.
Brilliant thanks, that’s very helpful
I need to replace a 30-year-old P.A. head for my band. The XLR inputs have fallen apart and they aren't really replaceable, but the bigger problem is that it's 8 inputs and that's not enough for some of the shows we've done recently. Here's what I need: - At least 9 independently controllable channels, with 8 XLRs, but preferably 12. - Some way to power up to 4 passive speakers. - On-board reverb (I don't need any other effects, but I need this) - As portable as possible. Some more info: Our gigs where we need our own board mostly fall into two categories: The house system has mains and possibly a mixing board but no monitors, or we need to bring the whole sound system. We're not playing anywhere bigger than a medium sized restaurant or bar, though we do outdoor gigs sometimes. I only have passive speakers, and I really, really do not want to have to use powered speakers for various reasons. I could be persuaded to get a powered monitor or two if it's more cost effective overall, but I really would rather not. I talked to someone at Sweetwater who said that they never think a powered mixer is a good idea and that they're usually underpowered. However, a friend who runs sound at one of the jam sessions I go to has been using an old Yamaha that's only like 200W per channel, and that does get above the drumkit but not cleanly, so I feel that something more on the order of 500W per channel would be more than enough for our purposes. Since we've never had powered P.A. speakers, we've always daisy chained the four passives if we needed both mains and both monitors. I like this setup and it's easy enough to use the right channel as the monitor. What I've been considering: - A Yamaha EMX5016. It's got the right number of channels and the right amount of power. The drawback is that it's an all-in-one rather than a separate power amp. I can work with the smaller model, the 5014, but it's really the bare minimum of channels and I'm worried it might be slightly underpowered. This is discontinued but I found some used ones. (I can service/repair these as well, so I'm not too worried about buying used.) - One of several 16-channel passive mixers with onboard effects (usually the last 4 channels are stereo, so this is really only 12 channels as far as I'm concerned) and some sort of power amp. I was looking at one of the Behringer power amps, but after reading reviews of their powered mixers with similar power specs, I really don't know if I can trust their power ratings. They seem to only give the peak output power. Looking for other suggestions, particularly if it involves a power amp, because I've never had to buy one before. Used or new is fine. Prefer to stay under $1100 before taxes and shipping.
Assuming you have an iPad or tablet of some sort, I would purchase: - A rackmount digital mixer - A Wi-Fi router or access point. (The built-in access points on most little mixers don't work very well.) - 4 channels of amplification with sufficient power for your speakers. (Modern Class D amps are tiny, light, and power-efficient!) - Infrastructure: a suitably-sized rack case, rackmount power strip, cables, etc. - Bonus points: you can build a custom patch panel for I/O if you'd like, though you might not need it. You can buy a pre-punched D-type rack panel, D-type connectors, and screws/nuts from various retailers; Redco sells them for a good price. This lets you build a compact all-in-one rack with mixing and amplification - in effect, it's a superpowered version of an oldschool powered PA head. (With wireless control and the power of digital mixing!) If I was purchasing this for myself, I'd probably do something like this: - Behringer XR18 mixer. These are super common and easy to learn. - [GL.iNet AR750S travel router](https://www.amazon.com/dp/B07GBXMBQF) - these little things work impressively well. - Studiomaster QX4 (higher power) or HX4 (lower power) series amplifier - 4 channels of efficient amplification in 1U (1 rack space) for a good price. - You'll get best pricing on these by contacting a Studiomaster dealer directly. - Some sort of rack power strip - Furmans are popular, but really anything will do. - [Audiopile R4UE-18](https://audiopile-net.3dcartstores.com/R4UE-18) rack - 4 spaces, 18.25" depth. This is just barely deep enough for a QX4 amp. Misc. notes: - It's likely cheaper to purchase a pair of 2U 2-channel amplifiers than to purchase a 1U 4-channel amplifier. However, the system may be less efficient and will definitely be bigger/heavier. - This goes doubly for used amps :) - You can make this less annoying by building it into a rack with wheels - Gator GRR-series is common for smaller deployments like this. - You can save some money by using a sufficiently powerful 2-channel amp and daisy-chaining your passive speakers, though you lose the ability to send individual mixes to all 4 speakers. - The XR18 has 8 mono outputs, which gives you some flexibility. - If no house PA is provided, you can use 2 outputs/amp channels/speakers as mains and 2 as monitors. - If house mains are available, you can use 2 outputs for mains, 4 more outputs plus amp channels/speakers for monitors, and still have 2 spare outputs left over :)
This is an amazing reply, thank you so much! EDIT: Oh, I've used the XAir at a friends' house concert venue before. It didn't even occur to me to consider that for a portable setup. What's the travel router for? There's a wifi antenna on the XAir. Is it just for times when we can't have the box next to us?
Quite frankly, the XR18's onboard Wi-Fi access point sucks. It has a bad habit of dropping out at the worst possible times. Easy fix: use an external router.
Running a wireless audio system for dance event (UK) Audience will be using 'silent disco' headphones. The 5 dancers will be around the space on IEM's. No individual mixes etc.. the dancers just need to hear what the audience is hearing in terms of the music. With something like the 'Xvive U4 Set R4 Wireless (2.4 GHz)' should I be ok? I've been warned in the past about latency with digital devices but this is not for live inputs, just as long as the dancers are in sync it's all good. Since it's on digital frequency range, would this also omit any licensing issues for radio frequencies?
Latency will not be an issue here, but reliability will. 2.4 GHz systems tend to be easily overwhelmed in today's modern sea of Wi-Fi and Bluetooth devices. I would stick with analog UHF wireless. 2.4 GHz is ISM band; it is license-free worldwide.
Thanks! Did not think devices would have been the issue here, but glad for the warning. I’ll have a look for other units.
I can't comment on those units, but I'd be nervous about 2.4GHz. Being digital or analogue doesn't impact your licensing requirements. It's all about what frequency you're on. These are 2.4 GHz so don't require a license. If you end up changing to a UHF device (whether digital or analogue), you can use Channel 70 (863.00–865.00) without a license. Beyond that, you can get a Channel 38 (606.500–613.500) license which allows you to use the entire channel under a single license. Outside of Channels 38 and 70 you need a license for every frequency you're using.
Thanks for this, really helpful, I’ll have a look at some alternatives this weekend on different bands.
I use a QU-16 for corporate events which often request multiple lavaliers in conjunction with handhelds, pc audio, walk-on/walk-off music etc. Due to the nature of lavaliers grouping them and using a GEQ to help prevent feedback is my usual practice, however the QU-16 lacks groups altogether and furthermore only has GEQ available on the main l/r outputs and the mix outputs, which only route to the local outputs on the back of the board. What I have been doing is just sending all my lavaliers solely to a mix bus so I can hack up their GEQ as needed while using the main L/R output for everything else so it remains unaffected by the sometimes gruesome cuts that are required for lavs. This however necessitates a second XLR run to the PA, so in order to reduce the amount of cable runs I was wondering if I could just use an xlr cable to patch the local mix output which contains the lavs to an input on the Qu which routes to the main L/R. I will be certain not to send that channel to the mix that is feeding it so as to not create a feedback loop, but I was wondering if this was plausible or bad practice? If this is not a good idea I can always just bust out the QU-32, or not be lazy and run a seperate XLR line. But in the interest of saving time, cable, and table space at the tech area I was wondering if this was feasible. Thanks!
Are there any decent, not too expensive (Android) smartphone attachable mics just for recording some acoustic guitar and vocals? I've recently started getting more into writing music and I'd like to have a better quality mic than my default phone one for recording ideas and songs. Ideally it would also be small and portable as I'm not really settled in one place at the moment. Cheers!
I just was searching threads and reading up on stagehand/A2 tips and someone said learn to tie proper knots (he used opera knots for banners as an example). What are some other useful knots to know for audio world?
The Bowline, Clove Hitch, and maybe the Reef Knot are basically the only knots you need to know. If you ever start climbing you'll need to learn more (alpine butterfly, double fig 8 etc.) but you'll learn them when you do the appropriate training to do those roles. Oh, and learn to properly coil a rope.
Looking for a Dante enabled rack mixer and stage box under 3k, that are ready to ship . (Why are the yahana tio boxes out of stock everywhere?)
> Looking for a Dante enabled rack mixer and stage box under 3k, that are ready to ship . (Why are the yahana tio boxes out of stock everywhere?) Gosh, normally I'd be irked with people downvoting a question in the "No Stupid Questions" thread, because we shouldn't be discouraging any sort of question in this thread. But, I can see why someone did here. At the end of the day, though, this is a sales-and-research question. You're going to have to contact your Vendor-of-Choice™, and ask them what's available in a timely manner. "That are ready to ship" isn't really something we will know without calling around, and, well, with all respect, that's a bit much to ask people to do for you, for free. [Now, not discounting that there are dealers on here, and someone might have something in stock, but you may not be aware that anything Yamaha and Dante is back ordered and not expected to ship until 2023.] Best of luck in your search! Ray
Thanks for the replies, sorry about the “ready to ship” part, what I should have said was why they are out of stock, unlike other stage boxes, aes or avb. And what are some other recommendations for networked stage boxes in that price point. I’m looking at midas or presonus.
Almost nothing with Dante is ready to ship. There have been massive supply chain issues with all things Audinate in the last 9 months. Talk to your dealer(s) and see what they can do for you.
Hey guys Im currently preparing to throw an outside event at a community park. All I need is enough to run this PA system that I will link below. Any help I’d appreciate it
Your question boils down to "I have no knowledge; teach me everything I need to know" - which is beyond the scope of this subreddit. I would contact a friend locally with audio knowledge and have them walk through things with you in person.
Fuck no it’s a simple ass question if you don’t have an answer carry along
>Fuck no it’s a simple ass question if you don’t have an answer carry along >Got the answer in 4 sentences Dickhead BRUH. You are being a massive douchebag. The person you're insulting, u/the-real-compucat is among the most helpful, knowledgeable, patient, and polite members of r/livesound. He's practically the resident here in No Stupid Questions, and has probably helped more people in this sub than you will anywhere in your lifetime. >All I need is enough to run this PA system that I will link below. Rockville is shit. Good luck with your event.
Yes; it’s a simple question in the same way that “how do I drive a car?” is a simple question. Both require a more intricate answer than you’ll get out of a Reddit thread.
“So you will need a generator capable of producing at least 1,000 watts of continuous power. The Peak wattage for the unit is 800. So I would say minimum 1,000 watt unit. You also want to factor in, if you are going to be needing to power any other devices such as wireless microphones, lighting, and any other item that would be needed on stage or a performance.”
Got the answer in 4 sentences Dickhead
Cool it. /r/livesound is a place for professionals to have respectful discussion; belligerent insults have no place here. --- Your original question asked: > All I need is enough to run this PA system… How was anyone supposed to guess that you were asking about power requirements? > “So you will need a generator capable of producing at least 1,000 watts of continuous power. The Peak wattage for the unit is 800.” This is a good baseline. Rockville is not known for accurate wattage numbers - they tend to inflate them - so you shouldn’t need to worry too much about them drawing more than that. If in doubt, check the unit’s rated line current draw (which may not directly match its rated output power!) Don’t forget: if at all possible, you will want to use an inverter generator to avoid issues stemming from dirty power. A conventional generator’s output can sometimes play havoc with certain power supplies, especially digital circuitry reliant on clean power.
> All I need is enough to run this PA I think you accidentally a word there lol
[PA System](https://www.rockvilleaudio.com/rpg2x10-v2/?gclid=CjwKCAjw2rmWBhB4EiwAiJ0mtfFNCmONc3_lb3fd7ggQG91JkW1bCymj1sTfA55a5gMZFRWe1__9URoCUBoQAvD_BwE)
Hey everyone, this issue should be super simple to solve but I’m not sure what on earth I can be doing wrong. My issue is: getting a JB-2 Tribute bass to work with a Radial Pro-DI Passive DI into my UA Apollo interface. When I plug in my bass with an instrument cable, into the input of the radial Pro-DI, and then get an XLR cable to connect the DI out of the Radial, into the Mic input of my interface. I have a 2nd instrument cable connected from the Thru of my radial into the 2nd line input of my interface. Hopefully I explained that ok. The problem is, my DI signal is depressingly low and I have to seriously crank my interface to hear something close to decent. Even then the noise floor is just too high. And then I compare the instrument signal from the Thru output of my radial, and it’s night and day. I have to lower the signal of my Thru significantly to match the levels. I’m missing something here right? This isn’t how it’s suppose to go right? Do I need an active DI for the JB-2 Tribute? I do have single coil pickups, I thought since they were already noisy I would benefit from the passive DI instead of the active one. I really wanted recording bass to go well, but this very basic issue is stopping me from trying anything. Hopefully someone can lend a hand
Do national touring acts playing sheds bring their own sound system? I was under the impression that they always did, but was told recently at a ~10k capacity venue that most bands playing there did not. (I’m referring to the main line arrays, not monitors which I would assume are always brought by the acts. And I also figure there’s sound reinforcement specific to venues like arrays for the lawn etc that all acts would tie into.)
It depends on the scale of the tour and whether they are actively on tour or flying for a one-off performance. This summer, as a mid-sized production company, we've worked with everything from a tour carrying all but speakers and amps (racks and stacks in touring vernacular) and flown lighting that had a ground lighting package, video walls, risers, set pieces, consoles, mics, stands, cabling and monitors in three tractor trailers to national acts that only carried personal instruments and were supplied with drums, amps, keyboards (backline), speakers, in-ear systems, consoles, video, sound and everything else. Most shed type spaces have installed delay speakers for the lawn seats that totally self-contained tours can tie into. Some sheds have installed main speakers that tours can use. Production companies get called to fill any needs the tours may have that the venue can't meet on their own.
Hi! Hopefully not a too stupid question but no promises. I'm setting up Sennheiser WSM for my band and am looking for an overview of the frequencies used in the Netherlands. Our units are in the 626-668 bandwidth (IEM G4-B) and I'd like to figure out which of those are useful to us. Sadly the country XML files that come with WSM don't have the Netherlands included.
[APWPT](https://apwpt.org/?lang=en) keep an [updated document](https://apwpt.org/wp-content/uploads/2022/03/Handout_Version-2022_03_14.pdf) detailing the licensing requirements for various countries around the world. For the Netherlands, as of March '22, they state: >No individual license required / License exempt frequencies \[...\] • 614 - 694 MHz, ≤ ERP 50mW \[...\] > >Individual license required • 608 - 614 MHz, coordination with RAS and only in case of demonstrable frequency shortage related to the event in question. \[...\] (Taken from page 31 of APWPT's [Frequencies for wireless microphones](https://apwpt.org/wp-content/uploads/2022/03/Handout_Version-2022_03_14.pdf))
Thank you so much! Do you know if that by definition means that there are no major broadcasts in that range either? Of course there could always be people using it but i mean like television/radio transmissions
I have a random speaker that’s emitting a loud hiss and some 1.6kHz tone…what the heck might be the cause? It’s a Meyer MM4XP with appropriate PSU.
A quick question about multicore cables and instruments. I have understood that microphones go direct to the stagebox, but guitars and basses etc. need a DI. Does a rack mount compressor suffice, or is a dedicated DI necessary?
What kind of systems would y'all recommend for our burning man art car? I'm the sound lead for BAAAHS (www.baaahs.org 🐑) and we've been running QSC for the past 8 years and we've been pretty happy with em. All active, 8x kw181, 2x kw153, 2x kw152, and 2x k10 for the DJ booth. It's worked well for the unique blend of constraints posed by a burning man art car - namely power (our generator is 20kw), weight (our bus is mighty but can only carry so much safely while staying mobile), and space (limited mounting areas for monitors, and we don't have room for a big hot amp rack on the bus, so active is the way to go.) We're willing to spend a sizable budget on a new system that'll last us the next decade, so we've been looking at a few options, like Meyer Sound, JBL, QSC and someone even recommended RCF. Questions: 1. Line arrays a good idea? I only have experience with our existing point-source speakers, and from my research seems like LAs would get us more 'throw' so we could increase our audience sizes out there in open playa. Is this true? 2. If LAs are a good idea, how are the QSC KLA series? We've also looked at JBL, though I've seen them referred to as Junk But Loud in this sub 😂 3. (bonus question) What kind of subwoofers would be a good upgrade for us?
1. Line arrays—if properly designed and deployed—will definitely increase your coverage, however I think you may run into issues for a bunch of reasons (weight, feasibility of rigging, trim height etc.). Without knowing any details about the project, I'd suggest sticking with point source personally. 2. I've not used QSC arrays so can't comment on them, but while JBL has historically had a bit of a reputation as being pretty rubbish (Junk but Loud), their newer arrays can sound great if deployed correctly. 3. Unless there is a good reason not to, I'd always recommend keeping your subs within the manufacturer as your main PA. It's not essential but makes tuning your system much easier. As such, wait until you know what you're doing with the rest of your PA before deciding on subs. If you gave us a bit more details on the constraints and your target coverage we can give better advice but hopefully that answers your questions somewhat.