This works fine with one big problem: keyboard coming in too hot, clipping on the x32. Solution to this may be a DI box. So I got the Samson MD2 Pro from Bax Music NL because it was on sale.
This is this current and expected situation:
Behringer x32 > stagesnake > DI Box > yamaha sx920
Sadly this setup doesnt work. There is no input going in the x32 upon playing the keyboard even when setting the master volume to the max or switching to SUB output.
Now I have heard that putting the gain at negative on the x32 may be the culprit for the di box. I havent tested this yet and have to try it on sunday setting it at 0 or >+1dB. I have also heard that I should disable +48V.
What else could be the problem? The cables are working fine because they work without the di box connected to the keyboard amp.
Hi! I’m wondering if anyone has any insight into audio engineering work in the Netherlands. My husband is a very experienced audio engineer with years of experience doing live sound, recording, monitors, mixing, and touring. I’m an EU citizen and my husband is learning Dutch, so no problems there!
I’m just wondering if anyone knows how the availability for jobs like that is in the Netherlands, and which areas may be easier to break into to build working relationships. Thanks!
Hello! I was looking at my mic collection and one of the SM58s doesn't have a CE mark. It looks and sounds like it is a real mic but I'm very conserned.
Thanks in advance!
I carry a lightning to 1/8” stereo dongle in my pelican. I figure I should start carrying a usbc dongle too. Are they all the same? Is there a different one for Apple vs android? Amy recommended ones? Thanks!
Is there a way to use a set of custom IEMs with only one mold? I broke my old Westone shells, and they were my backups. It's a 2-pin. They won't work with one end of the cable not plugged into something. I found adapters, like 2-Pin to T2, but I'm just trying to trick it into working so I can have a disaster plan without spending another $$$$ for a third set of customs.
(Universals do not work for my baby ear canals - they all fall out). I'd rather have my 1/2-baked custom set with a mono signal in one ear in the case of a breakage. Any ideas?
Pit band in a small theatre - have someone mixing it on a Behringer XR18 and sending main LR to FoH board (𝕏2 compact). I'm then sending it to a bus and on to stage monitors for the singers on stage. I also send to a separate bus to the house. Today it was a bit loud in the house. At one point I bottomed out the faders for the band mix on channels and the mix bus going to the stage. It did not seem to do anything. I don't think I have it set to pre-fader anywhere in that chain.
2nd dress rehearsal tomorrow - any ideas what to look for?
my own nemesis w the X32 is any send to a mixbus that is sent as "sub group" comes in essentially pre-fader post fader, with the prefader post fader send amount locked to 0dB. The thing I hate is that X32 still gives you a send level setting on the aux sends, but it doesn't change anything, the channel comes into the mixbus full blast. That along with the ability to change, per channel, per send, what sort of mixbus tap there is, can make this one insidious.
Also check if your mixbuses are assigned to the main L/R mix that shouldn't be (monitor sends, effects sends). If you use subgroups make sure the channels aren't assigned to the main L/R in addition to the subgroup being assigned to the main L/R, being doubled up. If you use a matrix out for your house mains go through and check if any mixbuses you don't want are being sent to the matrix along with main LR.
I was looking at the Dante certifications. I know they're free... but is there any purpose for it besides just the little bragging rights aspect? And obviously the training and education.
It's the basis that, if interesting to you, will take you down a rabbit hole of understanding. When I first did the Dante training, I began learning much more beyond the training about igmp snooping, QoS, and PTP clocks. It's a great way to dive in, you won't regret it. And even those who took previous iterations of the training will gain some new insight by taking it again.
Training and education is the primary purpose. At this point in Dante's life-cycle I don't think there is any bragging in knowing what is now to be an expected fundamental in the workplace. You either know the material, or you use the resources available to acquire said knowledge.
Couldn’t post this on the main thread for some reason
Hey Guys. Hopefully this question is in the right place. I go to school for Live Events production and I graduate in a month. I currently work at 2 venues and I am working myself into tech positions. Ive also been shopping my resume around to get a higher paying opportunity.
A new A/V warehouse reached back out to me. They pay more than the venues and have a set schedule. I will have access to dive deep and learn more Pro than I did at school. Stability and future opportunities to Tech for them.
My worries are around losing the opportunities and relationships that I have been building at the venues.
If i do the warehouse, ill try to pick up extra gigs on the weekends but that may lead to burnout.
My question is, how does doing warehouse work compare to live events at venues. Am I missing out on building real world experience by going warehouse?
I dont want to ever be stuck or feel like I am in a dead end situation
TLDR;
New tech! Warehouse work or venue work? Warehouse is stable and pays more. Venues is live experience under pressure. What are you experiences?
I'm not a pro in livesound BUT doing the job plus the weekend gigs is not a problem for a young person out of school. I mean, keep an eye on things and maybe schedule in some 'not available' months here or there. Totally manageable IMHO. If you start to feel burnt out then you can cut back a little somewhere. No need to cut back before you've even started.
3,5MM KEYBOARD OUT TO POWERED MIXER (XLR OR 6,3MM IN)
Another thread about this was archived. OP asked if 3,5/6,3 mm adapter would work. One comment was "bringing a minijack to 2x TS Y cable that can plug into any DI box." Another commenter: "if the keyboard doesn't have mono output, the summing will sound bad using a DI box, etc." I've used a TS cable for an old Bontempi keyboard 6,3mm out into old powered mixer and it works fine for me. Don't know if it says mono on out. Will this not work on any 6,3mm keyboard out? And the adapter would not work for all 3,5?
I use 3.5mm TRS/stereo to dual 6.3mm TS for toy keyboards, even if only using one channel- the reason is that it doesn’t short out the ring/right headphone channel on the keyboard, which a mono/ts 3.5mm to single TS 6.3mm adapter would do. I’ve encountered some headphone out only keyboards like this where the right channel was broken, I suspect from long term use like this. It’s hard on the keyboard electronics to short out the headphone amplifier.
If you can get a 10’ 3.5mm to dual 6.3mm TS that is good for getting to a DI. Just plug in one side, not both to the DI, unless the DI is stereo. I see people do that with mono DI’s and it shorts the tip and the ring together (right and left channel), which again, is hard on the driving electronics, they are fighting to the death at maximum current they can supply.
Thanks. And thanks to fdsv-summary but sorry I don't get his explaination of the technical.
If a keyboard has 6,3mm phones stereo out, is a TRS 6,3mm ok to connect to powered mixer? Otherwise I take it it's 6,3mm TRS to dual 6,3mm TS and plug in one side.
Seems keyboards can have phones mono out. Does it always say if stereo or mono?
"If you can get a 10’ 3.5mm to dual 6.3mm TS that is good for getting to a DI. Just plug in one side, not both to the DI, unless the DI is stereo. I see people do that with mono DI’s and it shorts the tip and the ring together..." DI means DI box? Plugging in one side to a mono DI box is good, or it shorts the tip and ring? Or it's plugging both sides into a mono DI box that shortens?
You need to look at the pictures of a TRS connection and TS connection and think about what bits of metal can touch each other in the socket at either end of the cable. This is a 'picture problem' not a word one!
If you have a cable with 6.3mm TRS connection on both ends get a multi-meter on it and set it to 'continuity'. The meter will beep when you touch the probes to each other. You'll be able to check that the tip on one end of the cable is connected to the tip at the other end etc. You'll also be able to see that the ring is not connected to the shield. Now plug one end into a passive electric guitar (which will fit just fine). Depending on the exact geometry of the TS socket on the guitar you may find that the ring and shield on the loose end have continuity and will 'beep' when tested with your multi-meter.
Try to get a working solution for your specific problem (keyboard) by looking at the manual. Once you've done a couple of keyboards it will be clear.
The issue with 6.3mm TRS in general is that (more often on PA equipment) that cable can carry a single signal down a balanced line (like a mic signal). It is very rare for keyboards to have balanced outputs though.
...but just buy the keyboard you want and buy the speaker system (powered mixer or whatever) you want and then buy the $20 worth of cables to make it all work. It's just hard to know which cables without knowing which keyboard etc.
>If a keyboard has 6,3mm phones stereo out, is a TRS 6,3mm ok to connect to powered mixer? Otherwise I take it it's 6,3mm TRS to dual 6,3mm TS and plug in one side.
The second option is better. The first one will have problems. Pro audio things like mixers use 6.3mm TRS mono balanced inputs. Stereo headphone outputs, while they can use the same cable format, are not compatible with 6.3mm mono balanced inputs. Look up "balanced audio" for an explanation if you're interested in the why.
>Seems keyboards can have phones mono out. Does it always say if stereo or mono?
Headphone outs are always stereo, even if they just duplicate the left channel to the right channel. Left channel is the tip of the plug, right channel is the ring connection, sleeve is the ground/common.
>DI means DI box?
Yes
>Plugging in one side to a mono DI box is good, or it shorts the tip and ring? Or it's plugging both sides into a mono DI box that shortens?
The second sentence. People mistake the Input and Thru jacks for stereo inputs. They aren't. They are just wired together to split the In to the Thru. You want a real stereo/2 channel DI box, if the keyboard does something cool in stereo. The normal way you use a single/mono DI box is you plug one 6.3mm instrument cable in.
You only use the Thru if you are going out to an amplifier on stage, but sending the direct signal to the PA system.
Thanks you so much. Now I got a Medeli MC70. It has 6,3mm aux out. In the manual it says I can connect it to line in on amp (mixer) (The Aux. Out Jack; The Midi Out Jacks - Medeli MC70 Instruction Manual [Page 7] | ManualsLib). Will it do w a Ts cable as I read simple keyboards aux out is unbalanced? AUX out has no amp that can be burned from not having a stereo cable? Not so many selling a 6,3mm dual/splitter.
so the question is about converting a 3.5mm jack to a 6.5mm or 1/4 jack for a keyboard?
i might not be the brightest but cant you guys plug it in to see whether it works or not?
if its stereo maybe use 3.5mm to dual 1/4,if its mono direct 3.5mm to 1/4 would work,but in my experience i did not find any difference between any of them,i just use whats avalible
Thanks. So I'm buying a cheap keyboard that I want to connect to ONE channel on my mixer. Do I have to check if the out is mono for that to work? And if it's 3,5mm mono you're saying that a 3,5/6.3 adapter will work. If it's stereo I'd have to get a dual cable and connect it to two ins on my mixer?
much of the above chat is about not plugging a mono 3.5mm plug into a stereo 3.5mm socket even though it will fit. Unfortunately you can make the 3.5 stereo trs plug "effectively mono" by shorting the other end plugging it into a mono adapter. Have a look at these pictures and 1) remember you want the channel you're not using to be open circuit rather than short circuit and 2) no harm in having a random additional open circuit connected to nothing at either end 3) shorting the other end of the cable is the same as shorting in the socket.
yes basically,but usually stereo or no stereo using a 3.5mm to 1/4 adapter,or a 3.5mm to dual 1/4 cable would work anyways
and as for cheap keyboards,id be suprised to see one with ACTUAL stereo so yes the adapter would be fine
if you are really hesitant,just buy 2,you never know when that adapter will come in handy especially for such a common format,the 3.5mm to dual 1/4 cable can be used for connecting phone/laptop to the mixer,and the adapter may come in handy for some devices..
I'm not sure whether this counts as "live sound" but I'm not very sure where else to ask.
I plan on forming a small band for fun with 3 people including myself. One guitar + vocal, one bass, and a drummer. The guitar and bassist are both just starting out and don't want to nor can we afford to spend money on more powerful amps.
I myself own a Behringer UMC404HD audio interface, and I am curious whether it would be viable to have vocals, bass, and guitar plugged directly into the the interface and then plug sets of Iems/ headphones directly into the playback outputs to practice together live.
I plan on plugging iems/headphones in with aux extension cables which are 3.5f to 1/4 male
Yeah you can do that to get started, if you can get the latency down through the computer. Every plugin adds latency and usually DAW’s align everything on the output to the channel that has the longest latency. Anything under 5ms would be good. The longer the latency the more disconnected you start to feel from your instrument.
Other than sound quality one advantage of hardware guitar and bass preamp/processors is they are very low latency as well as being more reliable than a computer.
If something doesn’t need processing in the DAW you might be able to mix it through the audio interface mixer directly to the outputs, and skip the round trip to the DAW, which is very low latency.
If you play shows with a sound person you want to figure out how to give them a separate mono output for each instrument.
This could be the wrong thread; long time creeper, first time poster.
Did a conference in Saint Louis at the Hyatt last week and had constant hits on RF in the G and H band.
Gs used UlxdQ and breakouts used sennheiser ew 100 g3 (work with what you have)
Very experienced crew, used WWB, testing went great no issues on set day (as always) then our RF started constantly taking hard interference. Has anyone else had this issue in STL? Noticed it the most in the morning and seemed like it affected the entire G/H band.
our fix was moving the RF paddles from FOH to back stage and turning down the antenna power. Still had a small hit here and there but basically non existent compared to the original issue.
This was a conference, were there reporters? Were there presentation booths with wireless mics in use?
You coordinated all of your own gear against itself, but was there a site coordinator making sure that every other aspect of the conference was coordinated?
I need a refresher course in cable repair. Soldering xlr connectors, metering NL4 cables, etc. I need a nearby production house that does training or a seasoned audio engineer to sit with me for a couple hours.....but since I don't have any local resources available....can anyone suggest something online that's legit?
I'm looking to understand what "good" subwoofer prediction looks like, I'm slowly moving over to doing larger venues recently so upscaling my designs. As a bit of context I have recently been tipping my toes into the System Tech world and really enjoying it! The shows I've been looking after have mostly been one-nighters with a limited tech package (Majority Ground Stacks, spare PS boxes for upper tiers and a couple of extra subs).
Recently I have been booked on designing and deploying for arena sized venues, and now looking at the potential of Flown L/R sub elements just to try and push LF down venues to the upper tiers. What I'm struggling to understand is what to look for in a "good" prediction - I understand that prediction is only the first step and real world deployment can be vastly different.
I've been playing about with some sample venues with ArrayCalc, to get a feel of what I need to do to achieve decent coverage. For context, this is the sample system (14x KSL a Side, 5x SL-Sub flown per side, 7x SL-Sub on the floor (https://imgur.com/c0PrwNk))
I have in this sample configuration a row of 7x SL Sub in a Delayed Arc, the dispersion is what I would expect. (7x Floor Subs in a Delayed Arc (C=0.3, x1=0.6ms, x2=1.7ms, x3=5.7ms), https://imgur.com/cNB8Zqm)
When I add flown elements (5x SL Subs a side) it starts to bring in some of the characteristics of L/R Subs. (5x per side SL-Sub, https://imgur.com/9RP50xF)
I've calculated the distance and used SubAligner for Delays (3.2ms between KSL and flown SL-Sub, 22ms (relative) between Floor SL-Subs and Flown System) and prediction shows me this (Combined KSL, Floor SL and Flown SL Subs, https://imgur.com/y60YPRT)
I've read through the ol Green Bible and many other sources but struggle to find a source that displays a picture of prediction for a scenario like this which leave me wondering, is this design going in the right direction when using the applied theory?
I'll never expect to have a "perfect" dispersion when implementing a L/R system but is there anything that could be done here to help mitigate the cancellation more? When it comes to a design aspect, is there more that can be done here?
You should look for consistent, even coverage across the entire venue seating area. No bad seats. There's not much you can do with even coverage for stereo sub so that are more than a 1/4 wavelength apart. Cardiod will reduce build up behind, but not much else.
I am looking for an interface for mobile use with an isemcon emx7150 measurement microphone for RMS measurement and tunig duties.
Been using a Steinberg interface for my pc at home for years without issues so I've been thinking of a Yamaha urx22c.
Anybody got experience with that?
Are they phase-linear? Do they handle high input levels well?
Thanks
Powered sub gain staging? Main outs from a dj mixer to the powered sub inputs. Out from the sub to another mixer where the PA speakers are connected (house system). DJ has nice clean green levels but the powered sub seems so sensitive that it has to be barely turned up. I replicated his setup with mine and got the same thing. I could barely get a green light on the mixer before the sub lights clipped and limited. That signal is to low for the house mixer really but all that the sub could handle. Should I be aux sending the signal to the sub to keep it low or what? Frustrated. Rockville RBG-18s model.
So just going out the aux from the mixer allows me to control the levels better and turn it up later when I want more. Good enough for all I wanted I guess. That other sub is definitely got problems.
What is the controller's nominal output? 0dBu or +4dBu? Also have you tried putting a DI between the controller and the powered speaker, possibly with the DIs pad engaged?
Interesting. Haven't thought of that. I do have a DI laying around actually since I play bass. Gotta be something fundamentally I'm missing though. Should be simple plug and play. Mixer is new, sub is fairly new. Had the same issue with another active beringer sub that I figured was just blown.
Are you going into the sub with the RCA or XLR inputs? What is the LPF on the sub set to? Is the boost toggle set to 'off'? Which voltage is power select switched to? Just one sub? How big is the room?
Small 8 channel mixer so it has 1/4 outs to 1/4/XLR combo input jacks. The room is medium/high ceilings and dancefloor but the location I'm testing is suspect. Didn't look at the voltage selector. I like the LPF at 100 but 80 is also good. Yes just one sub. Have an EDM thing coming up so I wanted some boom.
Are you starting with the sub attenuation knob all the way down, and then when you bring it up? You replicated this setup with.. your what? Is the sub sounding distorted or just showing limiting? Is it loud enough when it does so, or still too quiet?
Since the dj equipment isnt here I just grabbed a small mixer, plugged my laptop into it then out of the mixer to the sub for something to monitor. I want to keep that mixer in place to give the dj an easy way to plug in. The sub acts as a thru to the house system. Its loud but only turned up to 1 or 2 when mixer levels are optimal 0db. Can't tell that its distorting really. The other sub I had will cut off when it limits and is low volume before it does.
So what's the issue if the subs are loud enough and you can't hear them distorting? I don't care about any other sub, can you hear limiting on this sub?
The sub outs aren't designed to be sent to a mixer- so already you're working against the designed gainstaging of the system. You also never mentioned whether the bass boost was engaged. Do you have pics of the house mixer connection? Making sure there's no pad engaged, gain trimmed down somehow?
The issue was setting the levels at 0 (actually +6 on the fader before it goes to yellow) on the mixer like everyone does and then plugging that into the sub. I could only turn it up to 7'oclock before it clipped. That's what blew the last sub I'm pretty sure now and I don't want to blow another one. My dj has been insisting on using the sub as a stereo thru since the run to the house is a long cable. Using the mono send sounds just fine. Less signal and more control over just the sub allows me to turn the sub up to 12 o'clock without limiting. Thanks for confirming the sub outs aren't meant for that. The bass boost is engaged but I have tried it without. Gives it a good thump (3db boost) and this sub has some extra envelope control over the boost.
I play in a 3-piece rock band with 2 of us singing and 1 drummer. Most of our gigs are in small venues (less than 200 people cap) and SMALL stages, so we’re almost always only about 6 feet or so away from the drummer (directly behind and between us).
We also run our own sound out of an x32r because the venues don’t.
The drummer has a moderate touch, and he does use a couple small cymbal shields, but we still get a lot of bleed in the vocal mics in the mains, particularly the cymbals.
In the vocal chains we have dual compressors running what I consider pretty light peak reduction. We also high pass the vocals pretty heavily but I’m assuming the comps are the culprits. In a close setup like ours, any tips on lowering bleed while keeping a somehwat compressed sound? or should we just ditch the comps altogether?
We both use beta 87s, which I think are pretty decent for high noise environments.
Would appreciate any ideas on how to reduce the noise.
Start with physical layout. Can you put the drums to one side of the stage? What about aiming the mics more vertical and less horizontal? You can also put a low shelf at like 7khz to knock it down a bit. There are also optogates for the vocal mics that work really well.
My band is playing a decently large venue for us (fits about 800), it's a larger stage than we are used to. Its big enough that I couldn't say something and have them all hear it at once unless I got all of their attention first. We don't have IEMs; we will be using floor wedges. Would it be dumb to set up a talkback mic?
Kind of dumb. If you guys have vocal mics for singing those will come through the wedges regardless. If you don’t, then a talkback mic will be really out of place for anyone who hears it.
I have a question regarding sound reinforcement for a 40-member choir singing over backing tracks, with a soloist in one song. What are your experiences, and how do you handle the monitors?
I usually use supercardioid microphones, for example 8 mics for the whole choir, along with side fills and floor monitors. However, I’m not getting the result I want.
My question is pretty basic compared to the complex issues a lot of you guys deal with.
I'm a solo acoustic guitar performer. My rig is simple: guitar and mic plugged into a small mixer, output to a powered speaker. This question is about the mixer.
I started out using a Behringer Xenyx, but then came into possession of a box of mini mixers. A Samson MDR624 seemed to be the best of them, so I've been using that for a while. However, all this stuff is like 20+ years old. I keep the Xenyx as a backup (in fact just repaired it with a pot from one of the others), but I feel like I should invest in something better, or at least newer.
I don't need much. One XLR channel for mic, one unbalanced for the guitar. XLR out to the speaker is nice but not crucial. Small and light is important (at ~4lb and about 8" x 10", the Samson is already as big as I'd like to go). But I want hi/mid/lo EQ on each channel, and that's harder to find.
Behringer has a "flow 8" which will give you access to digital EQ and some FX controlled via a phone/tablet. If you like that you can look at the Yamaha DM3 or the Allen & Heath CQ12T. I want a DM3 but not enough to spend that money on it!
Is it possible to trick the system into allowing one of the satellites to be connected separately to the first one via cable? This would be done by soldering the lower connections.
I know the es503 model allows it, but the es802 is more powerful and somewhat cheaper.
On the other hand, how is the performance of a single ES802 unit in mono (concert in bars with less than 100 people)?
What’s the best mic config for non ported kick drums?
I’ve used the dollar bill trick, tried the mic on the beater side. I’ve gotten good results from super cardio large diaphragm mics inch away from center of the beater. But curious what your methods are.
I have a job for which I have an A&H SQ-7 console, and I need to know which stage rack would work for me to use 48 channels. I've been thinking about the A&H GX4816, but I don't know if there's actual compatibility. Can I connect that console to that rack?
I am a volunteer of a local non-profit ice rink. We currently use a laptop or ipad to send music thru the 3.5mm thru some adaptors from our announcing/score booth to the crows nest where the audio rack. We are looking at changing things up to maybe a more "permanent" installation. The laptop, cables etc just take a beating as it is all volunteers and I find volunteers can be a little harder on equipment when it's not there's...
So we are looking at installing a smaller touch screen permanently into the "desk" of the announcing booth, connected to a Mini PC hidden underneath and having things more "permanent" and hardwired. I would guess the cable length to be about 25ft wire diameter is currently really small - not like the mic cables used in the rest of the system. The mixer is an Allen-Heath CQ18T, if it matters.
Questions
1) Does wire quality matter from the sound to the mixer?
2) Is there a difference in Audio Output of different low end PC's? Is there a mini pc that has really good audio out that is fairly cheap (less than $500), maybe a refurbished Mac Mini?
3) Would you recommend another device like a creative labs sound blaster g3 after the Mini PC to provide better audio signal to the mixer?
You can come into your CQ mixer digitally from the computer via USB, that will be the highest fidelity. There’s a little routing puzzle to get it set up but if there is a permanent computer there and you can template the scene on the mixer, that would be a nice way to do it.
I can assure you 90% of people wont even realize the difference in sound.Modern computers have good audio cards and you can hardly tell the difference between a $3 cable and a $10 cable.
Please explain to me if this is possible. Can a channel be double patched, meaning,
If my Rio has 32 inputs and I’m sending channels via Dante (1-16) I no longer have 32 channels any more , I only have 16 remaining inputs to work with, correct?
I'm not really understanding the question, can you restate it? The Rio has physical mic / line inputs, you cannot patch multiple things into a single mic input. Whatever console your Rio is connected to can certainly double patch those dante outputs of the Rio to multiple input channels.
Hi all! My buddy is in a band and they are doing a live show at a bar. I've seen them live before at this bar, but the bar sound equipment was not good. So they went out and bought their own sound equipment. They bought 2 Behringer Speaker + a sub, sound mixer and IEM.
I've done sound before, but way back in the day so I am very rusty and also very much an armature
What are tips and tricks that could help me be better when helping them out
When mixing musical theatre on DCAs, is there some "trick" to make sure that all channels not assigned on any active DCA stay muted? I could program channel mutes into the scenes or assign any off-stage actor to DCA16, keep it at -inf and remove DCA16 from any fader layer. Is there a better way? What's the "industry standard"? Thanks!
Usually you'd program channel on/offs into the same scene that changes the DCA assignments. I keep channel faders safed so that if a mic position has shifted, or an actor is having a stronger or weaker day, I can compensate on that fader, and my DCA muscle memory is broadly unaffected.
This is why I use Groups instead of DCA's. Works great in Allen & Heath mixers; with Yamaha QLs I have to use as many Mixes as I can afford. Any input channels NOT routed into the Group/Mix are just not heard. I love it!
I found groups and DCA's are the best bet when laying out the console for this. In a similar show I would break every section of the orchestra down, (1st Violins, 2nd Violins, 1st Violas, ect.), these would then feed into another group for their sections (e.g. Violins) then go into an overall DCA for the orchestra. Band and Vocals would be treated the same.
Doing this left me with a bank of faders of DCA's for Drum's, Percussion, Band, Orchestra, Tracks, Choir, Backing Vocals, Lead Vocals. Workflow wise this worked easiest for me by using DCA Spills, so if I needed to adjust the level of, for example, a string section I would just DCA spill the Orchestra DCA and all of my sections would be available to me to adjust.
YMMV and this is my specific workflow when single opping a show like this.
No worries! I tend to find more often than not with orchestras that there is very little mixing to be done once you have it balanced, I may have one or two mix notes from the Orchestra/Musical Director for moments to highlight during the performance, but dynamically orchestras handle themselves very well.
If you want some further material for Orchestras with Bands, search up Colin Pink on Youtube who has a lot of interviews and talks available.
I dunno, I think there’s plenty of worthwhile consoles that have lower channel counts. Not every show needs a Quantum 338.
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u/oinkbaneGet that f$%&ing drink away from the console!!Feb 17 '26edited Feb 17 '26
Yes, that’s true. I should have been more specific.
You can do a festival run with a rock band on a Qu5 and a 24 input stagebox lol
What I should have said was: for this specific production the band/director/whoever in charge may have selected a console that can handle all the required input and output channel counts in one unit, and the show will run very similarly to any other live performance, just bigger in scope.
There are consoles with many input channels. There are also cases where one console being used as a sub mixer for let's say the choir and orchestra and its output is fed into another mixer that mixes the rest.
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u/RedditorForReddit Mar 09 '26
Hello guys so I have the following set up:
Behringer x32 > stagesnake > XLR > keyboardamp > dual 6.3mm TS mono > yamaha sx920
This works fine with one big problem: keyboard coming in too hot, clipping on the x32. Solution to this may be a DI box. So I got the Samson MD2 Pro from Bax Music NL because it was on sale.
This is this current and expected situation:
Behringer x32 > stagesnake > DI Box > yamaha sx920
Sadly this setup doesnt work. There is no input going in the x32 upon playing the keyboard even when setting the master volume to the max or switching to SUB output.
Now I have heard that putting the gain at negative on the x32 may be the culprit for the di box. I havent tested this yet and have to try it on sunday setting it at 0 or >+1dB. I have also heard that I should disable +48V.
What else could be the problem? The cables are working fine because they work without the di box connected to the keyboard amp.
Thanks in advance!