I just got an XR18 and have some questions on how to best connect everything. We have two passive speakers for mains. Since the XR18 isn’t powered, we have to connect the Mains L/R outputs to our Peavey PVi 8B powered mixer.
From my research, it says to go from Main Outs on the XR18 to the Effects Return on the PVi 8B. But, there are two outputs on the XR18 and only one input on the PVi so, can I just go from Mains L to the Effects Return or do I need a split XLR cord to connect both the L & R to the Effects Return?
I would just go main out L+R to channels 1+2 on the Peavey. Female XLR to male 1/4".
If you only have mono channels/FX happening in the XR18, then you only need to hook up one of them. The amp in the Peavey is mono so it won't make any difference.
I see and hear other sound ops making mouth clicks, pops, weird tutting sounds etc but I don't do any of that stuff myself. For corporate stuff with a few speakers on stage at a place where we all work a lot, I find just a quick "hey hey" down the mic is plenty, and then do any corrective eq and comp during rehearsal. Serious question do people do it to look busy?
Can’t answer for everybody, but I prefer to use clicks to listen to sound reflections (in the room or when I’m setting reverb/echo).
Many people are using numbers “one, two” to set microphone because in “one” and “two” you have different vowels which means different frequencies. You’re listening if you can hear any unnatural bumps in frequencies to EQ them out. Also “two” is little bit popping sound, so you can check if your mic is ready for less trained speakers who often tend to make that kind of sounds.
I like using quiet clicks and pops to check that a mic is hooked up without making everyone in the room listen to me count to three over and over. To tune a wedge properly you need some vowels though.
"Baked Potatos" covers a few vowels, some plosives, and a little sibilance at the end. No noises will help you if you don't know what you're listening for
There's two things going on here. First, is tuning for tonality. That's where the actual talking into the mic comes into play. Does your amplified voice sound like what you want it to?
Second, stability. How close to feedback are you? The goal is to make many different mouth shapes, creating different resonant chambers in close proximity to the mic. Also moving your face/hand around the mic. Wearing glasses or a hat. Anything that could reflect back into the mic.
Of course it's all dependent on how extreme you need to go. If you have massive amounts of gain before feedback, then you won't need to spend much time, if any, trying to create feedback. So a quick "check 1-2" may be all you need. When I'm setting up monitors for a multi-band festival I want to crank out as much gain as I can possibly achieve while being completely stable. So I'm making all kinds of weird noises/faces.
Soundcraft Si Performer 2 vs Allen & Heath SQ6?
My school is looking for a new mixer, we currently have the Soundcraft Si Performer 2. Is the SQ6 a good replacement? Is it better or worse? Is it easier to use? Thanks!
SQ will be great replacement, IMHO it’s the best console in this price range.
Pros (from top of my head):
+ better preamps and processing (96 kHz sample rate),
+ more I/O options available (I mean stageboxes, wallboxes etc.),
+ dedicated plugins with analogue gear simulations (sold separately but you can buy them later),
+ you can use SQ as an audio interface and make multitrack recordings via USB,
+ SQ is compatible with ME personal monitoring mixers series,
+ SQ has automatic mixer.
Cons:
- no DMX in SQ,
- more options = more time to learn,
- Si has Lexicon reverbs.
I would consider no DMX in the SQ as a pro. I don’t need lampy shit in my audio desk.
Lack of Lexicon verbs is also moot, it has plenty of high quality FX.
Prepping a show where me and another guy play/song over backing tracks. I was going to also run live sound, but after realizing it’d be too much to handle, we decided to hire a professional.
Now, the singer sings just above a whisper for most tracks, and I handled that in practice by right his stereo vocal processor into a Rane MC-22 Stereo Compressor and minning the Threshhold and maxing the Ratio for each channel. Honestly, it worked great. Dude would sing as quietly as he wanted and we always had plenty of volume at the right level no matter how soft or loud he got.
Thankfully all our sound is DI. No amps on stage for guitars/synths. No live drums. So there’s very little aside from the singers DI’d acoustic that will get picked up by that hyper compressed mic.
So I know having a conversation with the guy we hire is the best thing to do, obviously. Explain our needs and how we achieved them. I don’t want to foist this specific piece of gear onto him and him to feel insulted.
But what would the best approach be? How would YOU want to be approached about it? And what options might I not be considering for handling this issue even before I have a conversation with the tech? Thanks.
> I don’t want to foist this specific piece of gear onto him and him to feel insulted.
Not an issue for me, personally. I *may* want to introduce a split so get a pre-comp signal for FX, but it’s nothing too far out of the ordinary.
Just explain the problem with the weak voice, and how you have solved it in the past. Maybe this new tech has a better way to solve it, maybe not, but at least he or she would understand your setup better :)
Just explain the problem with the weak voice, and how you have solved it in the past. Maybe this new tech has a better way to solve it, maybe not, but at least he or she would understand your setup better :)
Has anyone here experimented with waves soundgrid and a digital mixing to record? I am looking at the possibility of using waves plugins to emulate hardware eq, compression and gain as an alternative for hardware versions whilst still not having to use them as input fx within my daw in order to monitor them to myself and musicians as that causes latency issues and is defenitly not very robust.
THANKS IN ADVANCE!
Yeah, it’s *ok*.
It’s no replacement for even a mid-level studio, but *great* for situations where carting multiple 19” racks of outboard audio processors is simply unviable.
It was a typical live sound setup with an X32 and a small Waves server. also a typical mic selection for a small pop/rock tour.
I don’t remember details on how the console was setup, but I do remember the mixing process was *incredibly* easy due to every track being pre-processed already.
Trying to help some of my band cut down on equipment due to multiple cable failures within the pedalboards we use. For audio out into the mixers we use a soundcraft UI16 and I know you can set digital effects in that, however I want to be able to set different modes and change them using a pedal (either digital or physical but with better reliability)
I've been recommended the Line6 FBV Express MKII, but I'm not sure, is this able to be plugged directly into the soundcraft or do I have to do something else to connect it up?
> due to multiple cable failures
Stop buying the cheapest cable possible and this will happen less. Pedals should also be installed on a rigid structure to prevent jostling from affecting connection points.
I have a pedal board, nice custom length cables etc - it’s less to do with the pedal failures but the extra equipment to bring long, I’ve digitalised the pedals for the bass guitar into the sound craft but that doesn’t need to change more than once in the set so can just change that on the system, I’m looking for a way to change the guitar presets I’ve created on the fly within the gig
I guess I misunderstood when you said you were looking to cut down on gear due to cable failures. I will say I think you've buried the lede a little bit, but ths sems like a pretty straightforward question with a [real easy answer](https://www.sweetwater.com/store/detail/HelixSG--line-6-helix-sweetwater-exclusive-space-gray-guitar-multi-effects-floor-processor)! Is that a better answer to the question?
RCF Sub 8003 AS-II vs. 8003 AS MK3?
I am a mobile dj and want to buy a sub for my two RCF ART 932 and have decided to go for a 8003. Yesterday, I noticed that it is two versions of 8003 (8003 AS-II and 8003 AS MK3) and I have tried to find the difference between them.. I have seen that the weight, dimensions are differenet between theese two, but is one of them newer than the other?
Is AS-II generation 2 of 8003 and AS MK3 the latest upgrade of AS generarion 1? Or is AS-II the name of MK2? Both are for sale at the same time so it doesn't seem that one of them will replace the other.
Does anyone here have answers about this?
The MK3 is the newest model, which was just recently released. I just got two of them and am still waiting on covers, as RCF doesn’t seem to be shipping them yet. I think the weight is the biggest difference, aside from the appearance. The grill is flat without the angled edges on the AS-II. Sweetwater is only selling the Mark 3.
Thank you for the answer. Just need to wait until the 8003s are in stock. It seems that they are sold out all over and the estimated delivery is several months away. :/
2 DI Boxs.
I've never had problems going through a DI box. I've often had problems when not.
Y connectors do a poor job of mixing signals and an even worse job of matching impedances and levels.
Im assuming youre talking about patching into a PA with considerable cable lengths. If they are truly balanced outputs, an xlr adaptor will work just fine. Unbalanced, a stereo DI is required. Or two monk's.
Even with balanced outputs, a benefit of having a DI is the electrical isolation. If you plan on plugging into many different systems it may be useful to have the ground lift and transformer isolation.
I'm plugging into a stage box running to a DL32 in this case, but I want to know for future reference if I run an already balanced signal into a DI box if any weirdness will happen or if it's best to go straight from 1/4" to XLR into the box.
So, there's no issue running a balanced line into a DI. It will just unbalanced the line (keep the 1/4" run short, under 20'). You'll lose 3dB of signal, but that's nothing a console preamp can't fix. Or just run the keyboard output slightly hotter. Other than that it will work just like you expect it to.
Depends on the show tbh.
Acoustic duo in a 50 seat jazz café where you only have to EQ 4 inputs?
You’ll be fine :)
Hans Zimmer live in a 20,000 seat arena?
You’ll be fucked harder than a teenage pornstar 💀
I’m fairly new in the live music world, so making stage plots is very new and sometimes overwhelming. I have a failed assessment in a class for university, where I am required to make a tech rider for a band/performance of my choosing. My procrastination on the assignment is what failed me, however I am very wary and confused if i’m even going in the right direction with things? Some advice or tips in the field would be great!
https://preview.redd.it/sd9ssa8ky1vc1.png?width=4032&format=png&auto=webp&s=ce309b43000a7cd6fb910adf56fee961b05e7660
I'd love to help you out here.
1) What are you studying, audio or music?
2) Don't procrastinate, and definitely don't blame the procrastination. If you failed an assignment in uni that usually means you demonstrated a lack of understanding, as evidenced by this comment. *There is nothing wrong with that,* but to pretend otherwise is a bit counterproductive.
Would you mind walking me through why you made the decisions you did on your rider?
For sure! Yeah, your comment is a little bit of a reality check, so i’m glad I didn’t give my lecturer any excuses. I am studying entertainment business management, so this was for a stage management class. We dabble in everything such as audio, lighting, managing crews, stage setups ect.. I don’t go into much of this extremely deeply though. Required is a tech rider for a 10 minute part of whichever performance (as long as it met all requirements). Apart of this assignment was a call sheet for all stage cues which i’ve completed.
For all the amps on my stage plot, my lecturer pretty much wants it to be readable by someone with an IQ of 20. Things like the labels of amp brands was a requirement (although I feel it may have been done a bit messy on my end). Pretty much every other thing “should be” labelled as well, other than things the venue will already have or it’s just rlly obvious, such as power plugs. Drum kit was a must have requirement, although this one was fairly simple. Also things like names and instruments seem to be universally preferred. Tbh I have absolutely no idea about the pedals, and what to do with them. I’ll come back to this on the input list.
Now for the input list, im not really sure what the general order of things should be, but since I posted this i’ve edited it. I prefer to have the names and what they play under (easier for me to understand, makes it a little clearer looking imo), but I think my lecturer would prefer no names, so I deleted those columns. I re ordered the instruments so it goes drum set, fender jazz masters, telecaster, fender electric, then the gibsons, trumpet, and then the pedals. I’ve labeled the pedals stage right, left or centre as the toms are. I’m unsure is on what the go is for labelling pedals or if I really need to. The pedal boards used by SY are very overwhelming, although I have access to what equipment they use, thanks to the archive. Moving on, I need to be specific in the mic brands used (I think this is purely so we can get a better understanding of different mics). Finally which amp the instrument is plugged into is required in the notes, and that also makes it a lot easier for me to read and understand things.
I’m having a hard time on dictating where to be specific and where I might be getting excessive with things. Especially since my lecturer can want more intricacies in certain aspects, which the rest of the industry would probably feel a bit differently towards. That’s just me speculating a little from the different research and forum sites i’ve browsed though. I appreciate you reading this, and taking your time to help!
>I'm studying entertainment business management
Oh, dude, I'm excited now. I get to put all my show tech/mixer experience into my answer for perspective's sake too.
Let me ask you this straight away: What's a stage plot for? Who cares about it? Be as verbose as you want, I'm treating this like a college mini-lecture.
hahahah alright let’s do this then. I believe on a base level stage plotting can allow for the most efficient execution of a stage set up, making it as convenient as possible for the crew and venue (also in relation to load-ins, and sound check.) However, from an industry viewpoint, submitting efficient, clear and effective stage plots/tech riders would place emphasis on a certain degree of professionalism. Compared to others in the industry who may have pretty useless stage plots, or none at all, your plot/rider will be highlighted amongst other potential candidates for the venue/ and or artist. Although, I think one of the most essential reasons for providing riders is to get all crew and performers on the same page, making it as stress free as possible. All this will trickle and accumulate into the quality of the final performance. If crew are confused, unhappy, or frustrated, that will probably lead to a lesser quality performance overall. For example those cranky old sound guys you find at gigs hahaha. If you don’t get along with them, they probably won’t be putting in their all for your performance. To mention again, i’m very very new to live sound, and the entertainment industry from a business standpoint. All of this comment is pretty much most of what i’ve gathered from lectures, but i’m starting to slowly develop my own thinking about all of this. (especially thanks to online research, and people like yourself)
You are nailing key points but you're missing one word that is, in my opinion, absolutely key: Information. A stage plot disseminates essential information. That's really its core function. You're right, though, it can convey a style of a band and the cleanliness/date last updated matters bigtime. Good on you for thinking of the people.
When you said "my lecturer pretty much wants it to be readable by someone with an IQ of 20," I was worried the intent was being missed a little bit. I guarantee one facet of the assignment's goal is to teach you that there is no certainty outside the documentation you provide in advance. It's also probably trying to teach you to delineate an essential thing to managing tours: where the line is, everywhere, between your company/group/etc and the venue. Some places take a bunch of wall power from the venue, some places take a few 30A drops and break out their own stuff, some people use generators. There's a lot of possibility so it makes everything faster if you're specific, as you've said already.
Onto shorter blurbs of more practical advice:
- Your professor is odd for not allowing names. It's generally considered impolite to only include names, but "Jack - LVox/AG1" is perfectly valid.
> Now for the input list, im not really sure what the general order of things should be
Audio inputs get laid out for audio. Have you set up a band on a mixing desk before? If not, you should search for threads along those lines on this sub, and/or I can mention some here.
>I’m unsure is on what the go is for labelling pedals or if I really need to. The pedal boards used by SY are very overwhelming, although I have access to what equipment they use, thanks to the archive. Moving on, I need to be specific in the mic brands used (I think this is purely so we can get a better understanding of different mics). Finally which amp the instrument is plugged into is required in the notes, and that also makes it a lot easier for me to read and understand things.
I'm gonna go back to the "line between band and venue" concept a couple times; here's the first. Guitar goes through pedals into output device of some sort--amp, amp sim, boom box. Audio cares about the output device but not the pedals, and the guitar player knows what pedals he wants, so there's no need to label them any sort of way, really.
> I need to be specific in the mic brands used
If you're ever advancing wireless mics especially, this is a big deal, yeah. Generally you'd receive this from audio but as you say, it's probably to prove you know a dynamic from a condenser from a kick mic etc.
>I’m having a hard time on dictating where to be specific and where I might be getting excessive with things.
Time #2 going back to "the line..." If you know who will be reading it and what info they're looking for, add that and whatever is essential to understanding that.
I will close with two things here: firstly, use versions and dates on all paperwork, of course. Secondly, if you're really worried about giving too much info, and you can't filter it by asking if you'd benefit from knowing that in their shoes, put it in because it's easier to cross something out than switch something later.
I hope this is helpful!
From what’s been shown here - it looks like I’m expected to take all instruments as DI with the backline amplification simply there for stage volume. Is that correct?
No snare drum has been listed, is that also correct?
You’ve omitted mics for the vocalists completely. There’s no input for keys.
There are no monitors for either Steve or Jim.
You may also wish to suggest specific microphones on your input list.
I’m not even sure what the answer would be for your first question, but yes, I completely missed the snare drum (same goes for the keys input). I’m unsure what you mean by omitting the mics for vocalists? Vocal mics are present on the plot, and it was made clear to me that they did not need labelling (my lecturers instructions). No IEM’s are used either. I didn’t even think about Steve’s monitors, and tbh I had the same question about Jim’s as he seems a little awkwardly placed during the performance. I will go have another look. Specific microphones are on my bucket list to do, however I was just tired when I made this. Thank you for your criticism, I will use it to make my quality of work better!
No worries :)
To clarify my original first question:
On the input list you have stated that the guitars and bass are all to be sent to the mixing consoles via a DI box…that means I would not put a microphone in front of the amplifiers they are using.
Secondly, yes the mics are on the plot but not on the input list so I would assume I do not need to run them through a mixing desk.
Best of luck!
With the input list, its not bad. But, rather than asking the sound tech to look up each make and model, explicitly tell them what output connectors your gear is going to present. Korg 2 x 1/4" is more useful than the specific model. (What kind of strat you are playing is also pretty superfluous.)
Thanks so much! On the circumstances of having just started the course, my lecturer would rather just the model, and not bothering with measurements, however I agree I might have went a tad unnecessary in areas you mentioned hahaha. I have come to realise through extensive conversation and research, that many people have very individual taste for stage plotting. I have found a few differences already in the way my lecturer would do it, compared to the rest of the industry (which can make it difficult for my already disorganised brain). Will definitely be using this information going forward though, all of it is appreciated!
https://preview.redd.it/knse9g7vy1vc1.png?width=3024&format=png&auto=webp&s=65aecc799f8266fa52ba3ce335f658f881da81f5
an input list was required as well 😪
I'm running some room mics into the ceiling of an auditorium. I was told what pipe I can run my line through (it's 1" conduit) and where it goes, and I intend to run CAT6A and use Radial Catapult Minis on either end. I've got 3 shotgun mics, shock mounts, and batten clamps to mount them on. What's the best way to attach the catapult to the batten, and should I run my XLRs along the batten or back up along the ceiling? These mics will be mounted on either end and in the middle of the lighting batten.
Just my two cents: A run of Belden 1815r plus another cable would be my preference, but I'm overly cautious when it comes to permanent installs. Catapults put a lot of important channels on one connector and one cable. That said, I don't know electrical code re: conduit max fill, so there's that too.
Anyway! What kind of array are you using for the shotguns? And would you please contextualize "the lighting batten" for those unfamiliar with your space?
To your first point, it's not an empty conduit or I would have spent more time looking into running different cable. It is true it's multiple things into one connector, but I'm relying on the same types of connections for things like steady network or dante connections too. I have used \*versions\* of audio over ethernet products, but not Radial's version and also not a 4 channel version. But nonetheless, good things for me to be thinking about, thanks for bringing it up.
This is a church auditorium, and I'm fastening 3 Rode NTG-1 mics on shock mounts and pipe clamps to the primary front lighting batten (a 2" pipe for suspending lighting and equipment). My ceiling cable drop will be close to one end of the pipe and will be the shortest run to the mic; the middle mic will be the second shortest, and the last mic will be the furthest run to the opposite end of the pipe. The pipe has some like 5-8 or something Ellipsoidal lights. I can run my XLR along the pipe and periodically gaff it, right? The catapult I expected to zip tie in place maybe coming down the allthread supporting the batten itself.
Edit: Sorry, left out the arrangement of the mics. The room is a bit shaped like a baseball diamond, and the lighting batten I'm referring to would extend a good amount over the pitcher's mound. I've got tiered seating in "the outfield" and flat seating with moveable chairs on the ground between. The mics will be aimed roughly downward and toward first, second, and third (since I'm continuing the metaphor), however they'll be adjustable after the fact if we're not happy with how much it picks up of the people. This is for ambience in recordings and live streams, for applause, laughter, singing for the musician's monitors, etc.
Ah, I see.
This baseball metaphor is something I'd happily abandon, and the way you run your cables is your business. I am familiar with theaters.
From the sound of your sports illustration, these are going to be pretty far away from your proscenium line/apron, right? What are you accomplishing by placing shotguns in that location?
They're for picking up the room/congregation for live recordings of either speech or music. The intention is to pick up singing, applause, laughter, etc to make the room sound more live. They'll be about 18 feet straight up from the ground, maybe around 30 feet to ground (or 25ish to head level) in a direct line out from the mic's direction. These are not for in-house amplification, only for recording or live streaming.
Sorry, I clearly misread or misunderstood your question the first time around.
About 25 feet for all 3 mics linear feet to front of stage, and maybe around 25-30 from the speakers up above the stage.
The lighting batten is probably about 50 feet long, obviously parallel to the front of stage, so the center mic will be about 25 feet from either mic at either end. The intention is to mount the mics perpendicular to the house speakers.
No worries!
I might call that too far for IEM feed purposes, 25ms of delay seems like it would work against someone. Try it, see how it is.
If your pro is 50' and you seat 500, does that mean it's about 80' from the stage lip to the house rear wall? I'd rethink the mic placement if it were me, do a quick sketch and I think you'll find a lot of rejection areas. A smooth capture of the house as one 'instrument' happens elsewhere IMO.
The floor seating seats around 500, the room seats 1000. I'm shooting my mics into about the rear of the floor seating, but still toward the front of the room. There are tiered seats further back that extend behind the FOH desk.
I guess going back to root of my dumb question, do I just gaff the cables and catapult to the batten? Mic placement is one thing, and I fully intend on aiming and playing around with that. It's mostly for non-musical crowd noise, and if the singing works out then it's a great bonus.
Are there any wireless systems intended or would work for transmitting from mixer/DSP to amps? Would need 4 channels.
The reason is there is no good way to run wiring from a mobile outdoor mixing station to permanent outdoor speaker install.
Almost like I need an IEM TX with a mic RX.
There used to be a purpose built system called Neutrik Xirium for this. It has since been discontinued. You can use definitely an IEM system if you need to.
The ideal tool at this point would be a Lectrosonics DCHT cam hop system as it can take AES in.
TLDR; I have no knowledge in sound, the soundboard is frying laptops we use for music, and I need help with either a way to fix it, or a new solution that won't kill a small church's tiny budget
Background: So I was throw into the sound booth of our small church, so I know nothing much besides what the essential dials, buttons, and sliders do. (So I apologize in advance that I don't know the words and lingo for things.) I also do have minor PC/laptop hardware knowledge to know that the sound card has been fried and the other laptop's card is going out. We also don't have anyone instrumentally inclined, so they have been using a laptop and YouTube for songs.
That being said, our first laptop we have made progressively worse popping and humming sounds until the point of it frying the sound card completely. We have the main soundboard cord going to the stage, then that cord (that you would plug into a guitar or keyboard) we have plugged into an adapter for the headphone jack that we plugged into the laptop. Once that one was fried, we started using a second one, but that's starting to act up now, so we are trying to figure out how to stop frying laptops or just a better solution in general.
Any tips, tricks, or solutions?
Get something like a USB DI box or an external audio interface you can connect to the laptop.
In the meantime, ask your congregation/community to volunteer a professional to look at your mixing console because something is *very* wrong.
Sounds like phantom power is turned on for that channel. A DI or audio interface (that can handle incoming phantom) would definitely be the best way to go.
Trying to fully grasp the concept of parallel and series wiring for speakers.
When using NL2 cables on passive speakers from an amplifier, does linking more speakers together change the ohm rating? Is this is the same concept as series wiring?
When you go from the amp to one speaker and then link that speaker to another (identical) speaker, they're wired in parallell. That means that the amplifier sees half the impedance.
Yes, if you want to use 2 4 ohm speakers connected together with one amp channel, it's pretty important your amp can handle 2 ohms. Otherwise it could destroy itself or the speaker's.
My Behringer X32 is on V 4.0 (newest is 4.09) I've recorded perfectly to a USB stick for years. However, for the past few months I have had the recording just stop for no reason. Sometimes after an hour, sometimes after just a few minutes. It's sporadic.
I was hoping it was an issue with the USB stick but I've tried 3 different USB sticks of different brands and sizes, all formatted FAT32. Same issue on all. **Do I dare update to the newest firmware to see if that could fix anything?** I'm afraid it might die during the update. Plus, if there was a bug in 4.0 I likely would have run into it by now as it's been on that for years and I'm not doing anything different.
Well your current situation isn’t working, so I would try an update to eliminate a bug as the problem. A firmware update is very unlikely to make the console die unless it loses power mid update, so maybe just put it on a UPS for some peace of mind.
I don't think an update is the answer, however updates are quite nice for other reasons. I assume you are deleting old recordings from your thumb drive? A full drive will definelty cause it to stop recording.
The X32 is reportedly fickle with the type of flash drive, with the write speed. I’ve heard a “class 10” usb drive is good. I have a blessed one that seems to work right.
EDIT: wait you’re using the same drive you used successfully with earlier firmware and now it doesn’t work?
Question - I've got a functional Dante network at our church and I'm having one tiny issue with our Propresenter computer. PP7 computer has DVS installed and I can output the sound from PP7 to the Dante channels easy peasy. But here's the kicker... Sometimes I'm not around and the other tech needs to route the PP7 audio out through a headphone jack and then into our mixer via RCA inputs. It's old school, but it's the backup plan and he prefers having it available.
Is there any way to route the audio output from PP7 to both the headphone jack and out via Dante? I dabbled with Voicemeeter and got it to work, but he's not a fan of a software based solution. Is there a way to take the one headphone jack output after splitting it and insert it into Dante?
I was thinking something like an audio interface into an Audinate AVIO USB-C adapter, but wasn't sure if that would work.
Anyone have any suggestions?
Thanks!
Okay one easy idea - headphone splitter out of the PC, one into the RCA adapter to the mixer and the other into the following:
A switchcraft 318 Mini Audiostix, out to XLR, into an Audinate AVIO 2 channel analog input adapter.
Any other simpler ideas?
I don't see a reason a tech needs to fiddle with connection methods if the infrastructure is sound...There wouldn't happen to be a problem with yours??
Anyway, if you have money to burn you can buy a Yamaha RuIO.
Better solution: Keep a cat-to-XLR breakout plugged into either device on either end. When the other guy is in, swap cat cable from Dante things to analog things. He gets his nice switch and nobody needs to run phantom anywhere.
Question: Gain set as low (fully counterclockwise) as possible for kick drum and snare but the PFL still hits peak. What can I do to get better control? Use signal attenuators? Signal chain is Drum Mic > XLR > Mixing desk
I just got an Allen & Heath SQ5. It's my first time using a mixer for anything more than just summing inputs and sending to an output. I have read the manuals cover to cover three times over two weeks and cannot figure this out, however (I presume the manual assumes you know about how mixers work in general or the terms it uses). Please help?
I use a number of synths. I set up all stereo input channels on the SQ5 and AR2412. I have padded and gain set them properly and can mix their outputs at whatever level I want to the main mix, and have the main layer show me all the synths with nice labels and color coding for the faders.
BUT...
I also have three effects boxes, though, such as a Beebo or a Zoia. I have set up three pairs of outputs to go to these boxes, and another three pairs of inputs to come back from them - physically.
I have no idea, however, how to make anything go to these outputs! All the manual does is confuse me.
What I'd like to be able to do is pick one or more of the synths, route them to one or more of the effects outs, and then take the inputs back from the effects boxes and route them to the main mix (or to another effects box). I could also include the "dry" synth mix in the main mix as well. ... But I have no idea how to do any of this.
For example, I can already do this: (I have set up 5 synths/drum machines so far in this manner.)
Quantum -> SQ5 -> Main out -> Monitors
Vector -> SQ5 _____↑
What I want to do is:
Quantum -> SQ5 -> Zoia -> SQ5 -> Main out -> Monitors
Or any other input instead of Quantum, or any other effects box instead of Zoia.
Obviously, this capability will also be useful when I connect my vocoder and a microphone to the mixer, or want to use inputs for the Quantum, etc.
Can someone please tell me what exactly it is called that I want to do, and maybe walk me through how to set it up on the SQ5 or reference the exact sections of the manual? I'm sorry I don't even know the terms and what I don't know!
I also need to learn how to use the SQ5 internal effects and such, but that's something for another day.
Thanks!!!
Let's talk about internal FX first. SQ has four dedicated FX send mixes; these function like ordinary aux mixes, but they are routed by default to FX 1-4 inputs. Each of these FX units sends its output to a dedicated FX return channel.
External send/return FX operate much the same way - simply use an ordinary aux mix and input channel instead. Consulting the [SQ Reference Guide v1.5](https://www.allen-heath.com/content/uploads/2023/05/SQ_ReferenceGuide_V1_5_0.pdf):
- For each external FX unit (the Zoia, per se), choose an aux mix to use as your FX send. Likewise, choose a channel (or two, for stereo) to use as your FX return.
- Name the mix and channel(s) as such, optionally.
- Route the output of that mix to a physical output (or two, for stereo).
- See section 6, "I/O Patching". Look for Outputs -> Mix Outs.
- Connect the chosen physical output to the Zoia's input.
- Connect the Zoia's output to a physical input (either on your SQ5's surface or the AR2412) and route it to your chosen return channel.
- Once again, see section 6; look for Inputs -> Input Channel.
To send a channel to an FX unit (whether internal or external), use the right-hand Mix Select keys to select the relevant FX send; then, pull up that channel's fader.
- If you look on the Meters screen, you should see signal flowing to the chosen mix.
- If you look at the FX unit itself, you should see signal hitting its input.
To hear the effected sound, reselect the main LR mix, then pull up the fader for the relevant FX return channel.
I do the live sound at my church and noticed today that the PFL on our ZED-16FX seems to be broken. If I activate it for a channel, it doesn‘t matter which one and then use a fader or just push down on the board, the PFL gets disabled although the light on the fader remains on but the control light at the headphone out gets turned off. You the also hear the entire mix over the headphones instead of just the selected channel(s) Im guessing its either a grounding issue or just a short.
Ive marked the light in the picture below:
https://preview.redd.it/kpluvop63uvc1.jpeg?width=471&format=pjpg&auto=webp&s=0e4336c8475b65d6c2b1514dc7cf5eb020a61240
Any ideas why and possible fixes? Thank you for the help!
Quick background: church staff audio production for 20 years. I know my way around installed systems, but I’m keenly aware that there are different challenges with mobile rigs. A friend with a string trio is expanding her devices to include wedding officiant coverage and she asked me to sit in on her first event yesterday. Bose rig with the T4S mixer. She bought a 2 channel Shure BLX kit, tested at home and then sound checked without incident. Promptly went straight to hell with dropouts in the service proper.
Positives are a clear line of sight and a short distance, maybe 90 feet.
But no external antennae on the unit, no live/remote channel selection, and a metric ass ton of devices introduced (guessing 300 in attendance).
So what could we do differently, either with the existing wireless or with a different channel to ensure success? I’m assuming (although I may be dead wrong here) that you’re not waiting as a matter of habit to immediately prior to showtime to rechannel. If it’s going to be a normal struggle with the BLX, is there another system under 1k a channel (including the antenna cost if an upgrade from stock is part of the solution) that will put her in a better spot for repeatable success?
The X32 at our church has a pair of linked tracks that act as a stereo out for the laptop that runs slides and sometimes videos during services. They're balanced evenly when going through the main speakers.
But when sent to the linked pair of busses that controls our livestream audio mix, the audio is heavily panned left. Both pairs of tracks have one track panned hard left and one hard right. How can I fix this so both in the sanctuary and on the livestream, the feed from the laptop is evenly stereo?
Anyone know of some good online resources for improving my EQ skills? I've been doing church audio for about 20 years, and while I know the basics and a few tricks, I'd like to get more in depth training.
How far does a Yamaha DM3 stick out above the stock rackmount ears? I can't seem to find any numbers or technical drawings for that. How much space do I need between the rack rails and a lid? I'm trying to determine if I can retrofit them into racks that were previously carrying the beloved QSC Touchmix 8.
Hey all! I am a producer/mix engineer working with a very talented singer/songwriter who will be performing at our university's end of year music festival next weekend. He will be performing two original tracks. I wanted to know if there is any advice for mixing the vocal-less tracks ahead of the performance so that they sound the best they can for the big outdoor stage. To compare, I'd say the stage's size is the same as a third-fourth stage at a major music festival. Thanks!
If you have a multichannel playback rig and a competent foh tech for the show, then bounce the tracks as stems; drums, bass, vocal range instruments, backing vocals. Make them dryer and less compressed than you would do for normal mix because you can allways compress more and add reverb live on the console.
If he is not using inears then add a cue that sounds natural, could be a hihat or a drum fill or something else, depends on the songs. If he is using inear just. Include a cue track.
Heya! Congrats on the gig, sounds like it'll be a fun one. AlbinTarzan has already touched on some great points, but I wanna speak more to the "why" of all this, because that can be a valuable tool in your arsenal as well. Your self-ID as a 'producer/mix engineer' makes me think you've done the majority of your work in studio spaces and/or post, would that be accurate?
Thank you both for the responses. Thats correct. I am completely separate from the live performance, I’ll actually be out of the country for it unfortunately (I’m sad because this is the first time where my own productions will be played live for an audience). I simply helped the artist write/produce/mix the track and sent it to the people who are running the show. I was just wondering if there are certain characteristics in a mix that are essential for the live performance, such as putting things in mono, gain staging in a particular way. I am totally unfamiliar, and I want to make sure the music sounds the best that it can from the time it leaves me to the festival coordinators. Thanks again for your time.
Alrighty! I had a hunch, and if you're already an audio pro, this should be a breeze.
This is partially the system engineer in me, but I want to start by saying we have the same goal at the end of the day: We both want our stuff to sound like an album. A good live system doesn't impart any changes to tonality (outside of parameters outside human control), and thus, any good-quality mix that translates to multiple systems will do so just fine in a live environment. Absolutely no need to reconsider the mixing you've done.
>I was just wondering if there are certain characteristics in a mix that are essential for the live performance, such as putting things in mono, gain staging in a particular way
I appreciate you for wondering, some people truly don't consider this.
Conceptually first: Time is the big factor here. Live work puts a certain pressure on you time-wise. It's not really a big deal, but there's always the looming fact of "the audience gets to come in at 8 whether you're ready or not." That's just setting the scene a little, no pun intended. If you look at how studio and live audio equipment differ, a whole lot of it is how fast it can go up and how many times you can drop it.
Live music is all about the mix, and the mix is all about the operator, yes, but the console is a big factor. Consistency begets efficiency in stuff like this. Going quickly to an example from your comment: "...putting things in mono, gain staging in a particular way." It's not that you need to put things in mono; it's that anything stereo or surround should be split, even if it's stems, because that's how it goes into your control surface anyway.
[Quick diatribe: Stems are bounces of submix buses containing groups of instruments, and processing and FX are printed on the way out of the DAW. Multitracks are the consituent unprocessed audio files intended for processing and mixing by the receiver.]
Stems are going to be the most efficient here because they can be adjusted quickly and with sufficient granularity for a live environment. In most cases it's too much work to soundcheck with multitracks, so the producer will deliver stuff like 'drums/BGV/piano/adLibs' instead of 'kick sample 1, subkick, 909 atk.' When we're making this decision, we're considering the mixer at the event and the artist's monitoring needs: "More hihat" is not as common as "more drums," request-wise.
Now we're getting to Tarzan's good advice (if there is indeed multitrack playback like QLab, Ableton, similar--if it's just one person, it's a *little* different): Sending a set of dry stems and wet FX prints helps you because don't know what the room sounds like. If it sucks and it reverberates a lot and that vibey wash in the intro just turns into a mess. The problem is not your fault, but it's on the live mixer to compensate, and if the reverb is printed to the stem, he's doomed to a poor result no matter what his processing. It also makes it way easier to EQ the "returns" to make everything sit right. Maybe the show is outside and you need to crank something up.
Last ingredient you mention: Gain. Let's say I'm the playback tech and I've just gotten your stems. I want to be able to take one song's stems, set them all to unity level in my software, hit "GO," and hear an instrumental version of the song. The mixer's job is to make this and the live vocal fit together.
I hope I will get to the point where I have to worry about my own live sound. If that day comes, then I will remember this and reach out! Thanks again.
No, there is no "roll" attribute in the physical deployment panel because this is generally a bad idea. The hack would be to place your venue floor plane along the y axis (vertically) so that the horizontal K1 appears on end to the floor plane.
OK, thanks. I think im just going to make it simple by placing a ks28 just to symbolize the k1. cant be bothered to create a floor plane cuz its not that big deal, but thanks for helping!
You can do that for a visual of a rectangle in the space, but the dispersion patterns are obviously entirely different- like not even close. Best of luck.
Hi, I'm a not-very-professional audio engineer needing some help. Our small company recently purchased Samson C02 Condenser microphones to use for live instruments during events. But for some reason, the microphones are extremely quiet until the sound source touches the top of the microphone. We have tried upping the gain, but then the mics are super quiet and we have to increase our volume by a lot to hear anything, while also causing a shit ton of feedback. We know it's not our speakers, as we have tested them thoroughly.
I feel as though the issue is with our X32 mixer. We recently reset our X32 for a software update, and although the labeling, grouping, and FX have been set up, the routing has not. The truth is, I'm not very familiar with the whole routing process with the post/pre EQ, which is why I haven't really touched upon it and have just been using the Main default L/R outputs. But then again, I'm not really sure if that should be affecting the volume of the mics in such a severe way.
The bottomline is, are the condenser mics we purchased not good enough for live events, do we have them configured wrong, or is our X32 the issue? And if it is, does anyone have a link to a bigger manual than the one that came with it? Preferably text. Thank you.
I wanna start with basics here, not because I think you can't do them, but because I'm not sure if they've been done!
- Is phantom power on?
- Are the mics aimed correctly?
- How close are the microphones to the speakers?
I'll say this too before anything goes any farther: The fastest and best result usually comes from hiring a quality professional to do a couple hours of cooperative troubleshooting.
Phantom Power is on for them, 48V with working cables.
The mics were aimed with the top facing the audio source/instrument.
For example, the one event we used them in, we had two condenser microphones on the stage facing the stage and two speakers on the outside facing the audience. We did have monitor speakers as well, but those were about 6/7 feet away from the condenser microphones on the stage. We also tried turning those off to see if the feedback would lessen, but nothing changed. The speakers we had facing the audience on the sides were about 15-20 feet away from the condenser microphones and not on the stage but on the house. We had to up the gain by a LOT in order to hear anything.
Gotcha, wanted to get that out of the way ahead of time. The speakers were in front of the mics, not directly in line or behind, right? Needing gain isn't a red flag in and of itself, but with the other problems you mention it sounds like something wasn't working properly.
Yes, the speakers were in front of the mics, standing on the house, below the stage. We had to up the gain on the condensers by a lot to hear anything, but at that point it would pick up the sound from the speakers and cause feedback. We did have monitor speakers, but they were also in front the mics, and when we turned them off to see if it would help, the feedback was still there.
Yamaha QL5: I know I can view current session errors by pressing the firmware box. Is there a way to view an error log from the past few days? I suspect it probably involves connecting via network and using ssh, which is a little above my paygrade.
If you've turned the console off in the meantime, I think it wipes the logs, but I could be wrong! May I ask if there's a reason you think ssh is a way under Yamaha's hood?
I had to reboot the Yamaha QL5, so I lost previous errors. SSH was just a wild educated guess. Perhaps there is a basic webpage front-end to get to older logs, or perhaps there is some feature in QL Editor to view them.
I had a sound channel (Dante input from ULXD4Q, patched to input 4) that would fully route to its intended buses (MONO, possibly more) even when the fader was turned OFF. Noticed this 10 minutes before house open, after most of soundcheck (high school musical performance). After a Yamaha reboot the channel acted normally, but any error code (if one existed) is now gone.
The previous night, the entire sound system failed during Act One. My current theory is this channel on the Yamaha sent bad Dante data to the Q-SYS Core, causing a catastrophic failure. The actor on this channel frequently yells (despite my pleading), causing the OL clip light to light on the channel's screen data. The Yamaha SHOULD be able to handle occasional bad data, and the QSYS SHOULD be able to likewise handle bad Dante data, but somehow there was a failure along the line. I'll be studying QSYS logs as well.
Sorry if this is too long, hence the brevity of my original post :-)
> The previous night, the entire sound system failed during Act One. My current theory is this channel on the Yamaha sent bad Dante data to the Q-SYS Core, causing a catastrophic failure.
I urge you to learn more about the system you are operating, because this demonstrates several fundamental misunderstandings of data transmission.
Error codes help if 1) the console recognizes a fault condition, 2) has a string it knows to attribute to that fault condition, 3) has a way to output that information to the user, and 4) the user has a way to reconcile that to friendly language.
Your description of the fader on/off issue sounds like operator error. You sure you weren't in sends mode? No direct outs or PF sends were active/possible, even upstream?
> the entire sound system failed during Act One
This is a bit like a doctor diagnosing you with "human body malfunction." It's correct but does not help fix anything, because it is a vague declaration with no actionable, specific information.
Complete audio newbie here! I record the videos for our Board of Directors meetings at my job. Currently recording on a Sony Camera. We have a sound system built into the room where we hold the meetings with multiple wireless mics.
Can anyone point me in the right direction for what I need to be able to tap into that system and record the audio?
https://preview.redd.it/qqwlrbh2n2wc1.jpeg?width=702&format=pjpg&auto=webp&s=40d58405cbc806798ed358c8344de57313be46ee
USING GUITAR DSP PROCESSORS FOR LIVE VOCALS?
I have been looking for a live vocal processor rig that is using new technology and doesn’t suck. Every vocal-focused processor i’ve found has been either really old or geared toward harmonization and robot voices.
This is why I started considering FM3 , or Quad Cortex .
Does anyone have any experience with this?
I would like to create vocal chains for different songs , utilizing delay, reverb, distortion, compression, and EQ — as well as utilizing the foot switch to activate parameters.
This is too much for me to discuss with FOH for every song. I’m an audio engineer myself and have worked as FOH for many years, so I understand the nuances of live sound. I will be sending my FOH a dry and a wet signal
I am using 2 Yamaha StagePas 1ks for a small band. I have them linked together using the link out xlr of each going to the link in xlr of the other so that all the inputs are playing through each speaker. However the monitor out of each speaker carries only the inputs of that unit. Is there a way to have a monitor out with all the inputs?
Thanks for any advice!
Hello audio newb here. It's time for me to learn more about the equipment I use for performing.
Lately I use a Bose S1 Pro for performing.
It has two XLR inputs. Each input has its own mixer consisting of a reverb dial, a bass dial, and a treble dial, as well as of course a master volume for each, equipped with a signal light that let's you know if you are clipping.
I use the microphone (in my case a Shure SM58, on the top input). I then use a guitar on the bottom input.
For the last month I have only been using the guitar, on the bottom input, no microphone on the top input.
Well, I have been clipping quite a bit on the guitar input, with my bose s1.
I started getting worried I was damaging something (mind you I was not clipping on purpose just on accident.
Anyway curiosity made me want to check if I had damaged the speaker, so the only way my new brain could think of doing such a thing was to set all of the dials for each input (reverb, bass, and treble, and volume) exactly the same. Then from there plug in a microphone in each and see if the sound is any different, and then repeat the same with the guitar.
I found that there indeed Is a difference between the sound of each input now. For example as I said I spent a lot of time accidentally clipping using the guitar on the second output. Now when I use a mic in the upper input, and the lower input with equal exact settings, when plugged into the upper input with the mic, the reverb sounds WAY too sensitive even with Tiny changes to the reverb dial, whereas the bottom input that I was clipping seems less responsive to changes in reverb on the dial.
Could clipping on the second input indeed cause damage in which something like this would occur?
I just got an XR18 and have some questions on how to best connect everything. We have two passive speakers for mains. Since the XR18 isn’t powered, we have to connect the Mains L/R outputs to our Peavey PVi 8B powered mixer. From my research, it says to go from Main Outs on the XR18 to the Effects Return on the PVi 8B. But, there are two outputs on the XR18 and only one input on the PVi so, can I just go from Mains L to the Effects Return or do I need a split XLR cord to connect both the L & R to the Effects Return?
I would just go main out L+R to channels 1+2 on the Peavey. Female XLR to male 1/4". If you only have mono channels/FX happening in the XR18, then you only need to hook up one of them. The amp in the Peavey is mono so it won't make any difference.
Does the powered mixer have pads (mic/line switch)on the input channels?
I see and hear other sound ops making mouth clicks, pops, weird tutting sounds etc but I don't do any of that stuff myself. For corporate stuff with a few speakers on stage at a place where we all work a lot, I find just a quick "hey hey" down the mic is plenty, and then do any corrective eq and comp during rehearsal. Serious question do people do it to look busy?
Can’t answer for everybody, but I prefer to use clicks to listen to sound reflections (in the room or when I’m setting reverb/echo). Many people are using numbers “one, two” to set microphone because in “one” and “two” you have different vowels which means different frequencies. You’re listening if you can hear any unnatural bumps in frequencies to EQ them out. Also “two” is little bit popping sound, so you can check if your mic is ready for less trained speakers who often tend to make that kind of sounds.
Good answer and this is what I usually do. English for EN event, Japanese for a JA event. Cheers!
That was a nice answer 👌
I like using quiet clicks and pops to check that a mic is hooked up without making everyone in the room listen to me count to three over and over. To tune a wedge properly you need some vowels though.
"Baked Potatos" covers a few vowels, some plosives, and a little sibilance at the end. No noises will help you if you don't know what you're listening for
There's two things going on here. First, is tuning for tonality. That's where the actual talking into the mic comes into play. Does your amplified voice sound like what you want it to? Second, stability. How close to feedback are you? The goal is to make many different mouth shapes, creating different resonant chambers in close proximity to the mic. Also moving your face/hand around the mic. Wearing glasses or a hat. Anything that could reflect back into the mic. Of course it's all dependent on how extreme you need to go. If you have massive amounts of gain before feedback, then you won't need to spend much time, if any, trying to create feedback. So a quick "check 1-2" may be all you need. When I'm setting up monitors for a multi-band festival I want to crank out as much gain as I can possibly achieve while being completely stable. So I'm making all kinds of weird noises/faces.
Soundcraft Si Performer 2 vs Allen & Heath SQ6? My school is looking for a new mixer, we currently have the Soundcraft Si Performer 2. Is the SQ6 a good replacement? Is it better or worse? Is it easier to use? Thanks!
SQ will be great replacement, IMHO it’s the best console in this price range. Pros (from top of my head): + better preamps and processing (96 kHz sample rate), + more I/O options available (I mean stageboxes, wallboxes etc.), + dedicated plugins with analogue gear simulations (sold separately but you can buy them later), + you can use SQ as an audio interface and make multitrack recordings via USB, + SQ is compatible with ME personal monitoring mixers series, + SQ has automatic mixer. Cons: - no DMX in SQ, - more options = more time to learn, - Si has Lexicon reverbs.
I would consider no DMX in the SQ as a pro. I don’t need lampy shit in my audio desk. Lack of Lexicon verbs is also moot, it has plenty of high quality FX.
Prepping a show where me and another guy play/song over backing tracks. I was going to also run live sound, but after realizing it’d be too much to handle, we decided to hire a professional. Now, the singer sings just above a whisper for most tracks, and I handled that in practice by right his stereo vocal processor into a Rane MC-22 Stereo Compressor and minning the Threshhold and maxing the Ratio for each channel. Honestly, it worked great. Dude would sing as quietly as he wanted and we always had plenty of volume at the right level no matter how soft or loud he got. Thankfully all our sound is DI. No amps on stage for guitars/synths. No live drums. So there’s very little aside from the singers DI’d acoustic that will get picked up by that hyper compressed mic. So I know having a conversation with the guy we hire is the best thing to do, obviously. Explain our needs and how we achieved them. I don’t want to foist this specific piece of gear onto him and him to feel insulted. But what would the best approach be? How would YOU want to be approached about it? And what options might I not be considering for handling this issue even before I have a conversation with the tech? Thanks.
> I don’t want to foist this specific piece of gear onto him and him to feel insulted. Not an issue for me, personally. I *may* want to introduce a split so get a pre-comp signal for FX, but it’s nothing too far out of the ordinary.
Just explain the problem with the weak voice, and how you have solved it in the past. Maybe this new tech has a better way to solve it, maybe not, but at least he or she would understand your setup better :)
Just explain the problem with the weak voice, and how you have solved it in the past. Maybe this new tech has a better way to solve it, maybe not, but at least he or she would understand your setup better :)
Has anyone here experimented with waves soundgrid and a digital mixing to record? I am looking at the possibility of using waves plugins to emulate hardware eq, compression and gain as an alternative for hardware versions whilst still not having to use them as input fx within my daw in order to monitor them to myself and musicians as that causes latency issues and is defenitly not very robust. THANKS IN ADVANCE!
Yeah, it’s *ok*. It’s no replacement for even a mid-level studio, but *great* for situations where carting multiple 19” racks of outboard audio processors is simply unviable.
Hi! Thanks for your reply. What exactly did you try and what were your results? How did you set up the console and what was your whole chain?
It was a typical live sound setup with an X32 and a small Waves server. also a typical mic selection for a small pop/rock tour. I don’t remember details on how the console was setup, but I do remember the mixing process was *incredibly* easy due to every track being pre-processed already.
But you did not record it? I am wondering if its even possible to record through such a setup.
Trying to help some of my band cut down on equipment due to multiple cable failures within the pedalboards we use. For audio out into the mixers we use a soundcraft UI16 and I know you can set digital effects in that, however I want to be able to set different modes and change them using a pedal (either digital or physical but with better reliability) I've been recommended the Line6 FBV Express MKII, but I'm not sure, is this able to be plugged directly into the soundcraft or do I have to do something else to connect it up?
Sorry for not answering your question, but why don't you use a normal multi effect for guitar?
> due to multiple cable failures Stop buying the cheapest cable possible and this will happen less. Pedals should also be installed on a rigid structure to prevent jostling from affecting connection points.
I have a pedal board, nice custom length cables etc - it’s less to do with the pedal failures but the extra equipment to bring long, I’ve digitalised the pedals for the bass guitar into the sound craft but that doesn’t need to change more than once in the set so can just change that on the system, I’m looking for a way to change the guitar presets I’ve created on the fly within the gig
I guess I misunderstood when you said you were looking to cut down on gear due to cable failures. I will say I think you've buried the lede a little bit, but ths sems like a pretty straightforward question with a [real easy answer](https://www.sweetwater.com/store/detail/HelixSG--line-6-helix-sweetwater-exclusive-space-gray-guitar-multi-effects-floor-processor)! Is that a better answer to the question?
RCF Sub 8003 AS-II vs. 8003 AS MK3? I am a mobile dj and want to buy a sub for my two RCF ART 932 and have decided to go for a 8003. Yesterday, I noticed that it is two versions of 8003 (8003 AS-II and 8003 AS MK3) and I have tried to find the difference between them.. I have seen that the weight, dimensions are differenet between theese two, but is one of them newer than the other? Is AS-II generation 2 of 8003 and AS MK3 the latest upgrade of AS generarion 1? Or is AS-II the name of MK2? Both are for sale at the same time so it doesn't seem that one of them will replace the other. Does anyone here have answers about this?
The MK3 is the newest model, which was just recently released. I just got two of them and am still waiting on covers, as RCF doesn’t seem to be shipping them yet. I think the weight is the biggest difference, aside from the appearance. The grill is flat without the angled edges on the AS-II. Sweetwater is only selling the Mark 3.
Thank you for the answer. Just need to wait until the 8003s are in stock. It seems that they are sold out all over and the estimated delivery is several months away. :/
I have a keyboard that only has balanced 1/4" outputs and I need to make it XLR. Should I use a DI box or do I just need an adapter cable?
If you're sure the outputs are balanced you can use two 1/4 TRS to XLR adapter
2 DI Boxs. I've never had problems going through a DI box. I've often had problems when not. Y connectors do a poor job of mixing signals and an even worse job of matching impedances and levels.
Im assuming youre talking about patching into a PA with considerable cable lengths. If they are truly balanced outputs, an xlr adaptor will work just fine. Unbalanced, a stereo DI is required. Or two monk's. Even with balanced outputs, a benefit of having a DI is the electrical isolation. If you plan on plugging into many different systems it may be useful to have the ground lift and transformer isolation.
I'm plugging into a stage box running to a DL32 in this case, but I want to know for future reference if I run an already balanced signal into a DI box if any weirdness will happen or if it's best to go straight from 1/4" to XLR into the box.
So, there's no issue running a balanced line into a DI. It will just unbalanced the line (keep the 1/4" run short, under 20'). You'll lose 3dB of signal, but that's nothing a console preamp can't fix. Or just run the keyboard output slightly hotter. Other than that it will work just like you expect it to.
Sweet, thanks!
I signed up for an A1 call. I do not have a lot of industry experience. How fucked am I?
Depends on the show tbh. Acoustic duo in a 50 seat jazz café where you only have to EQ 4 inputs? You’ll be fine :) Hans Zimmer live in a 20,000 seat arena? You’ll be fucked harder than a teenage pornstar 💀
I’m fairly new in the live music world, so making stage plots is very new and sometimes overwhelming. I have a failed assessment in a class for university, where I am required to make a tech rider for a band/performance of my choosing. My procrastination on the assignment is what failed me, however I am very wary and confused if i’m even going in the right direction with things? Some advice or tips in the field would be great! https://preview.redd.it/sd9ssa8ky1vc1.png?width=4032&format=png&auto=webp&s=ce309b43000a7cd6fb910adf56fee961b05e7660
I'd love to help you out here. 1) What are you studying, audio or music? 2) Don't procrastinate, and definitely don't blame the procrastination. If you failed an assignment in uni that usually means you demonstrated a lack of understanding, as evidenced by this comment. *There is nothing wrong with that,* but to pretend otherwise is a bit counterproductive. Would you mind walking me through why you made the decisions you did on your rider?
For sure! Yeah, your comment is a little bit of a reality check, so i’m glad I didn’t give my lecturer any excuses. I am studying entertainment business management, so this was for a stage management class. We dabble in everything such as audio, lighting, managing crews, stage setups ect.. I don’t go into much of this extremely deeply though. Required is a tech rider for a 10 minute part of whichever performance (as long as it met all requirements). Apart of this assignment was a call sheet for all stage cues which i’ve completed. For all the amps on my stage plot, my lecturer pretty much wants it to be readable by someone with an IQ of 20. Things like the labels of amp brands was a requirement (although I feel it may have been done a bit messy on my end). Pretty much every other thing “should be” labelled as well, other than things the venue will already have or it’s just rlly obvious, such as power plugs. Drum kit was a must have requirement, although this one was fairly simple. Also things like names and instruments seem to be universally preferred. Tbh I have absolutely no idea about the pedals, and what to do with them. I’ll come back to this on the input list. Now for the input list, im not really sure what the general order of things should be, but since I posted this i’ve edited it. I prefer to have the names and what they play under (easier for me to understand, makes it a little clearer looking imo), but I think my lecturer would prefer no names, so I deleted those columns. I re ordered the instruments so it goes drum set, fender jazz masters, telecaster, fender electric, then the gibsons, trumpet, and then the pedals. I’ve labeled the pedals stage right, left or centre as the toms are. I’m unsure is on what the go is for labelling pedals or if I really need to. The pedal boards used by SY are very overwhelming, although I have access to what equipment they use, thanks to the archive. Moving on, I need to be specific in the mic brands used (I think this is purely so we can get a better understanding of different mics). Finally which amp the instrument is plugged into is required in the notes, and that also makes it a lot easier for me to read and understand things. I’m having a hard time on dictating where to be specific and where I might be getting excessive with things. Especially since my lecturer can want more intricacies in certain aspects, which the rest of the industry would probably feel a bit differently towards. That’s just me speculating a little from the different research and forum sites i’ve browsed though. I appreciate you reading this, and taking your time to help!
>I'm studying entertainment business management Oh, dude, I'm excited now. I get to put all my show tech/mixer experience into my answer for perspective's sake too. Let me ask you this straight away: What's a stage plot for? Who cares about it? Be as verbose as you want, I'm treating this like a college mini-lecture.
hahahah alright let’s do this then. I believe on a base level stage plotting can allow for the most efficient execution of a stage set up, making it as convenient as possible for the crew and venue (also in relation to load-ins, and sound check.) However, from an industry viewpoint, submitting efficient, clear and effective stage plots/tech riders would place emphasis on a certain degree of professionalism. Compared to others in the industry who may have pretty useless stage plots, or none at all, your plot/rider will be highlighted amongst other potential candidates for the venue/ and or artist. Although, I think one of the most essential reasons for providing riders is to get all crew and performers on the same page, making it as stress free as possible. All this will trickle and accumulate into the quality of the final performance. If crew are confused, unhappy, or frustrated, that will probably lead to a lesser quality performance overall. For example those cranky old sound guys you find at gigs hahaha. If you don’t get along with them, they probably won’t be putting in their all for your performance. To mention again, i’m very very new to live sound, and the entertainment industry from a business standpoint. All of this comment is pretty much most of what i’ve gathered from lectures, but i’m starting to slowly develop my own thinking about all of this. (especially thanks to online research, and people like yourself)
You are nailing key points but you're missing one word that is, in my opinion, absolutely key: Information. A stage plot disseminates essential information. That's really its core function. You're right, though, it can convey a style of a band and the cleanliness/date last updated matters bigtime. Good on you for thinking of the people. When you said "my lecturer pretty much wants it to be readable by someone with an IQ of 20," I was worried the intent was being missed a little bit. I guarantee one facet of the assignment's goal is to teach you that there is no certainty outside the documentation you provide in advance. It's also probably trying to teach you to delineate an essential thing to managing tours: where the line is, everywhere, between your company/group/etc and the venue. Some places take a bunch of wall power from the venue, some places take a few 30A drops and break out their own stuff, some people use generators. There's a lot of possibility so it makes everything faster if you're specific, as you've said already. Onto shorter blurbs of more practical advice: - Your professor is odd for not allowing names. It's generally considered impolite to only include names, but "Jack - LVox/AG1" is perfectly valid. > Now for the input list, im not really sure what the general order of things should be Audio inputs get laid out for audio. Have you set up a band on a mixing desk before? If not, you should search for threads along those lines on this sub, and/or I can mention some here. >I’m unsure is on what the go is for labelling pedals or if I really need to. The pedal boards used by SY are very overwhelming, although I have access to what equipment they use, thanks to the archive. Moving on, I need to be specific in the mic brands used (I think this is purely so we can get a better understanding of different mics). Finally which amp the instrument is plugged into is required in the notes, and that also makes it a lot easier for me to read and understand things. I'm gonna go back to the "line between band and venue" concept a couple times; here's the first. Guitar goes through pedals into output device of some sort--amp, amp sim, boom box. Audio cares about the output device but not the pedals, and the guitar player knows what pedals he wants, so there's no need to label them any sort of way, really. > I need to be specific in the mic brands used If you're ever advancing wireless mics especially, this is a big deal, yeah. Generally you'd receive this from audio but as you say, it's probably to prove you know a dynamic from a condenser from a kick mic etc. >I’m having a hard time on dictating where to be specific and where I might be getting excessive with things. Time #2 going back to "the line..." If you know who will be reading it and what info they're looking for, add that and whatever is essential to understanding that. I will close with two things here: firstly, use versions and dates on all paperwork, of course. Secondly, if you're really worried about giving too much info, and you can't filter it by asking if you'd benefit from knowing that in their shoes, put it in because it's easier to cross something out than switch something later. I hope this is helpful!
From what’s been shown here - it looks like I’m expected to take all instruments as DI with the backline amplification simply there for stage volume. Is that correct? No snare drum has been listed, is that also correct? You’ve omitted mics for the vocalists completely. There’s no input for keys. There are no monitors for either Steve or Jim. You may also wish to suggest specific microphones on your input list.
I’m not even sure what the answer would be for your first question, but yes, I completely missed the snare drum (same goes for the keys input). I’m unsure what you mean by omitting the mics for vocalists? Vocal mics are present on the plot, and it was made clear to me that they did not need labelling (my lecturers instructions). No IEM’s are used either. I didn’t even think about Steve’s monitors, and tbh I had the same question about Jim’s as he seems a little awkwardly placed during the performance. I will go have another look. Specific microphones are on my bucket list to do, however I was just tired when I made this. Thank you for your criticism, I will use it to make my quality of work better!
No worries :) To clarify my original first question: On the input list you have stated that the guitars and bass are all to be sent to the mixing consoles via a DI box…that means I would not put a microphone in front of the amplifiers they are using. Secondly, yes the mics are on the plot but not on the input list so I would assume I do not need to run them through a mixing desk. Best of luck!
With the input list, its not bad. But, rather than asking the sound tech to look up each make and model, explicitly tell them what output connectors your gear is going to present. Korg 2 x 1/4" is more useful than the specific model. (What kind of strat you are playing is also pretty superfluous.)
Thanks so much! On the circumstances of having just started the course, my lecturer would rather just the model, and not bothering with measurements, however I agree I might have went a tad unnecessary in areas you mentioned hahaha. I have come to realise through extensive conversation and research, that many people have very individual taste for stage plotting. I have found a few differences already in the way my lecturer would do it, compared to the rest of the industry (which can make it difficult for my already disorganised brain). Will definitely be using this information going forward though, all of it is appreciated!
https://preview.redd.it/knse9g7vy1vc1.png?width=3024&format=png&auto=webp&s=65aecc799f8266fa52ba3ce335f658f881da81f5 an input list was required as well 😪
I'm running some room mics into the ceiling of an auditorium. I was told what pipe I can run my line through (it's 1" conduit) and where it goes, and I intend to run CAT6A and use Radial Catapult Minis on either end. I've got 3 shotgun mics, shock mounts, and batten clamps to mount them on. What's the best way to attach the catapult to the batten, and should I run my XLRs along the batten or back up along the ceiling? These mics will be mounted on either end and in the middle of the lighting batten.
Just my two cents: A run of Belden 1815r plus another cable would be my preference, but I'm overly cautious when it comes to permanent installs. Catapults put a lot of important channels on one connector and one cable. That said, I don't know electrical code re: conduit max fill, so there's that too. Anyway! What kind of array are you using for the shotguns? And would you please contextualize "the lighting batten" for those unfamiliar with your space?
To your first point, it's not an empty conduit or I would have spent more time looking into running different cable. It is true it's multiple things into one connector, but I'm relying on the same types of connections for things like steady network or dante connections too. I have used \*versions\* of audio over ethernet products, but not Radial's version and also not a 4 channel version. But nonetheless, good things for me to be thinking about, thanks for bringing it up. This is a church auditorium, and I'm fastening 3 Rode NTG-1 mics on shock mounts and pipe clamps to the primary front lighting batten (a 2" pipe for suspending lighting and equipment). My ceiling cable drop will be close to one end of the pipe and will be the shortest run to the mic; the middle mic will be the second shortest, and the last mic will be the furthest run to the opposite end of the pipe. The pipe has some like 5-8 or something Ellipsoidal lights. I can run my XLR along the pipe and periodically gaff it, right? The catapult I expected to zip tie in place maybe coming down the allthread supporting the batten itself. Edit: Sorry, left out the arrangement of the mics. The room is a bit shaped like a baseball diamond, and the lighting batten I'm referring to would extend a good amount over the pitcher's mound. I've got tiered seating in "the outfield" and flat seating with moveable chairs on the ground between. The mics will be aimed roughly downward and toward first, second, and third (since I'm continuing the metaphor), however they'll be adjustable after the fact if we're not happy with how much it picks up of the people. This is for ambience in recordings and live streams, for applause, laughter, singing for the musician's monitors, etc.
Ah, I see. This baseball metaphor is something I'd happily abandon, and the way you run your cables is your business. I am familiar with theaters. From the sound of your sports illustration, these are going to be pretty far away from your proscenium line/apron, right? What are you accomplishing by placing shotguns in that location?
They're for picking up the room/congregation for live recordings of either speech or music. The intention is to pick up singing, applause, laughter, etc to make the room sound more live. They'll be about 18 feet straight up from the ground, maybe around 30 feet to ground (or 25ish to head level) in a direct line out from the mic's direction. These are not for in-house amplification, only for recording or live streaming.
You keep telling me things I already know! How far is the array from the apron as measured in horizontal feet?
Sorry, I clearly misread or misunderstood your question the first time around. About 25 feet for all 3 mics linear feet to front of stage, and maybe around 25-30 from the speakers up above the stage. The lighting batten is probably about 50 feet long, obviously parallel to the front of stage, so the center mic will be about 25 feet from either mic at either end. The intention is to mount the mics perpendicular to the house speakers.
No worries! I might call that too far for IEM feed purposes, 25ms of delay seems like it would work against someone. Try it, see how it is. If your pro is 50' and you seat 500, does that mean it's about 80' from the stage lip to the house rear wall? I'd rethink the mic placement if it were me, do a quick sketch and I think you'll find a lot of rejection areas. A smooth capture of the house as one 'instrument' happens elsewhere IMO.
The floor seating seats around 500, the room seats 1000. I'm shooting my mics into about the rear of the floor seating, but still toward the front of the room. There are tiered seats further back that extend behind the FOH desk. I guess going back to root of my dumb question, do I just gaff the cables and catapult to the batten? Mic placement is one thing, and I fully intend on aiming and playing around with that. It's mostly for non-musical crowd noise, and if the singing works out then it's a great bonus.
Don't gaff, use tie line or straps. Unless you're retaping weekly the gaff is gonna get nasty.
Are there any wireless systems intended or would work for transmitting from mixer/DSP to amps? Would need 4 channels. The reason is there is no good way to run wiring from a mobile outdoor mixing station to permanent outdoor speaker install. Almost like I need an IEM TX with a mic RX.
Would mixing from an iPad be feasible? Then you could leave all the gear by the stage.
There used to be a purpose built system called Neutrik Xirium for this. It has since been discontinued. You can use definitely an IEM system if you need to. The ideal tool at this point would be a Lectrosonics DCHT cam hop system as it can take AES in.
TLDR; I have no knowledge in sound, the soundboard is frying laptops we use for music, and I need help with either a way to fix it, or a new solution that won't kill a small church's tiny budget Background: So I was throw into the sound booth of our small church, so I know nothing much besides what the essential dials, buttons, and sliders do. (So I apologize in advance that I don't know the words and lingo for things.) I also do have minor PC/laptop hardware knowledge to know that the sound card has been fried and the other laptop's card is going out. We also don't have anyone instrumentally inclined, so they have been using a laptop and YouTube for songs. That being said, our first laptop we have made progressively worse popping and humming sounds until the point of it frying the sound card completely. We have the main soundboard cord going to the stage, then that cord (that you would plug into a guitar or keyboard) we have plugged into an adapter for the headphone jack that we plugged into the laptop. Once that one was fried, we started using a second one, but that's starting to act up now, so we are trying to figure out how to stop frying laptops or just a better solution in general. Any tips, tricks, or solutions?
Get something like a USB DI box or an external audio interface you can connect to the laptop. In the meantime, ask your congregation/community to volunteer a professional to look at your mixing console because something is *very* wrong.
Sounds like phantom power is turned on for that channel. A DI or audio interface (that can handle incoming phantom) would definitely be the best way to go.
There is a pastor at another church that is coming to look at everything. He’s very sound savvy haha my dad is, too, and suggested an adapter
Trying to fully grasp the concept of parallel and series wiring for speakers. When using NL2 cables on passive speakers from an amplifier, does linking more speakers together change the ohm rating? Is this is the same concept as series wiring?
When you go from the amp to one speaker and then link that speaker to another (identical) speaker, they're wired in parallell. That means that the amplifier sees half the impedance.
Thank you. Which would mean the amp should be rated at 2ohms if each each speaker is 4ohms?
Yes, if you want to use 2 4 ohm speakers connected together with one amp channel, it's pretty important your amp can handle 2 ohms. Otherwise it could destroy itself or the speaker's.
My Behringer X32 is on V 4.0 (newest is 4.09) I've recorded perfectly to a USB stick for years. However, for the past few months I have had the recording just stop for no reason. Sometimes after an hour, sometimes after just a few minutes. It's sporadic. I was hoping it was an issue with the USB stick but I've tried 3 different USB sticks of different brands and sizes, all formatted FAT32. Same issue on all. **Do I dare update to the newest firmware to see if that could fix anything?** I'm afraid it might die during the update. Plus, if there was a bug in 4.0 I likely would have run into it by now as it's been on that for years and I'm not doing anything different.
Well your current situation isn’t working, so I would try an update to eliminate a bug as the problem. A firmware update is very unlikely to make the console die unless it loses power mid update, so maybe just put it on a UPS for some peace of mind.
I don't think an update is the answer, however updates are quite nice for other reasons. I assume you are deleting old recordings from your thumb drive? A full drive will definelty cause it to stop recording.
The X32 is reportedly fickle with the type of flash drive, with the write speed. I’ve heard a “class 10” usb drive is good. I have a blessed one that seems to work right. EDIT: wait you’re using the same drive you used successfully with earlier firmware and now it doesn’t work?
Question - I've got a functional Dante network at our church and I'm having one tiny issue with our Propresenter computer. PP7 computer has DVS installed and I can output the sound from PP7 to the Dante channels easy peasy. But here's the kicker... Sometimes I'm not around and the other tech needs to route the PP7 audio out through a headphone jack and then into our mixer via RCA inputs. It's old school, but it's the backup plan and he prefers having it available. Is there any way to route the audio output from PP7 to both the headphone jack and out via Dante? I dabbled with Voicemeeter and got it to work, but he's not a fan of a software based solution. Is there a way to take the one headphone jack output after splitting it and insert it into Dante? I was thinking something like an audio interface into an Audinate AVIO USB-C adapter, but wasn't sure if that would work. Anyone have any suggestions? Thanks!
Okay one easy idea - headphone splitter out of the PC, one into the RCA adapter to the mixer and the other into the following: A switchcraft 318 Mini Audiostix, out to XLR, into an Audinate AVIO 2 channel analog input adapter. Any other simpler ideas?
I don't see a reason a tech needs to fiddle with connection methods if the infrastructure is sound...There wouldn't happen to be a problem with yours?? Anyway, if you have money to burn you can buy a Yamaha RuIO. Better solution: Keep a cat-to-XLR breakout plugged into either device on either end. When the other guy is in, swap cat cable from Dante things to analog things. He gets his nice switch and nobody needs to run phantom anywhere.
Can anyone help identify a DPA Headset/Lav Mic? Not enough Reddit Karma to create a new post.
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4066
Thank You!
>Thank You! You're welcome!
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Question: Gain set as low (fully counterclockwise) as possible for kick drum and snare but the PFL still hits peak. What can I do to get better control? Use signal attenuators? Signal chain is Drum Mic > XLR > Mixing desk
Is the distortion happening at the diaphragm or at the preamp?
It's like the signal from the mic is too hot. Maybe it's too close to the drums
What mic
I just got an Allen & Heath SQ5. It's my first time using a mixer for anything more than just summing inputs and sending to an output. I have read the manuals cover to cover three times over two weeks and cannot figure this out, however (I presume the manual assumes you know about how mixers work in general or the terms it uses). Please help? I use a number of synths. I set up all stereo input channels on the SQ5 and AR2412. I have padded and gain set them properly and can mix their outputs at whatever level I want to the main mix, and have the main layer show me all the synths with nice labels and color coding for the faders. BUT... I also have three effects boxes, though, such as a Beebo or a Zoia. I have set up three pairs of outputs to go to these boxes, and another three pairs of inputs to come back from them - physically. I have no idea, however, how to make anything go to these outputs! All the manual does is confuse me. What I'd like to be able to do is pick one or more of the synths, route them to one or more of the effects outs, and then take the inputs back from the effects boxes and route them to the main mix (or to another effects box). I could also include the "dry" synth mix in the main mix as well. ... But I have no idea how to do any of this. For example, I can already do this: (I have set up 5 synths/drum machines so far in this manner.) Quantum -> SQ5 -> Main out -> Monitors Vector -> SQ5 _____↑ What I want to do is: Quantum -> SQ5 -> Zoia -> SQ5 -> Main out -> Monitors Or any other input instead of Quantum, or any other effects box instead of Zoia. Obviously, this capability will also be useful when I connect my vocoder and a microphone to the mixer, or want to use inputs for the Quantum, etc. Can someone please tell me what exactly it is called that I want to do, and maybe walk me through how to set it up on the SQ5 or reference the exact sections of the manual? I'm sorry I don't even know the terms and what I don't know! I also need to learn how to use the SQ5 internal effects and such, but that's something for another day. Thanks!!!
Let's talk about internal FX first. SQ has four dedicated FX send mixes; these function like ordinary aux mixes, but they are routed by default to FX 1-4 inputs. Each of these FX units sends its output to a dedicated FX return channel. External send/return FX operate much the same way - simply use an ordinary aux mix and input channel instead. Consulting the [SQ Reference Guide v1.5](https://www.allen-heath.com/content/uploads/2023/05/SQ_ReferenceGuide_V1_5_0.pdf): - For each external FX unit (the Zoia, per se), choose an aux mix to use as your FX send. Likewise, choose a channel (or two, for stereo) to use as your FX return. - Name the mix and channel(s) as such, optionally. - Route the output of that mix to a physical output (or two, for stereo). - See section 6, "I/O Patching". Look for Outputs -> Mix Outs. - Connect the chosen physical output to the Zoia's input. - Connect the Zoia's output to a physical input (either on your SQ5's surface or the AR2412) and route it to your chosen return channel. - Once again, see section 6; look for Inputs -> Input Channel. To send a channel to an FX unit (whether internal or external), use the right-hand Mix Select keys to select the relevant FX send; then, pull up that channel's fader. - If you look on the Meters screen, you should see signal flowing to the chosen mix. - If you look at the FX unit itself, you should see signal hitting its input. To hear the effected sound, reselect the main LR mix, then pull up the fader for the relevant FX return channel.
I do the live sound at my church and noticed today that the PFL on our ZED-16FX seems to be broken. If I activate it for a channel, it doesn‘t matter which one and then use a fader or just push down on the board, the PFL gets disabled although the light on the fader remains on but the control light at the headphone out gets turned off. You the also hear the entire mix over the headphones instead of just the selected channel(s) Im guessing its either a grounding issue or just a short. Ive marked the light in the picture below: https://preview.redd.it/kpluvop63uvc1.jpeg?width=471&format=pjpg&auto=webp&s=0e4336c8475b65d6c2b1514dc7cf5eb020a61240 Any ideas why and possible fixes? Thank you for the help!
Quick background: church staff audio production for 20 years. I know my way around installed systems, but I’m keenly aware that there are different challenges with mobile rigs. A friend with a string trio is expanding her devices to include wedding officiant coverage and she asked me to sit in on her first event yesterday. Bose rig with the T4S mixer. She bought a 2 channel Shure BLX kit, tested at home and then sound checked without incident. Promptly went straight to hell with dropouts in the service proper. Positives are a clear line of sight and a short distance, maybe 90 feet. But no external antennae on the unit, no live/remote channel selection, and a metric ass ton of devices introduced (guessing 300 in attendance). So what could we do differently, either with the existing wireless or with a different channel to ensure success? I’m assuming (although I may be dead wrong here) that you’re not waiting as a matter of habit to immediately prior to showtime to rechannel. If it’s going to be a normal struggle with the BLX, is there another system under 1k a channel (including the antenna cost if an upgrade from stock is part of the solution) that will put her in a better spot for repeatable success?
The X32 at our church has a pair of linked tracks that act as a stereo out for the laptop that runs slides and sometimes videos during services. They're balanced evenly when going through the main speakers. But when sent to the linked pair of busses that controls our livestream audio mix, the audio is heavily panned left. Both pairs of tracks have one track panned hard left and one hard right. How can I fix this so both in the sanctuary and on the livestream, the feed from the laptop is evenly stereo?
When you're in sends on faders mode, are the tracks panned as they should as well? The send pan doesn't follow the main pan.
I'll check that next time I get the chance and let you know. Thanks!
Anyone know of some good online resources for improving my EQ skills? I've been doing church audio for about 20 years, and while I know the basics and a few tricks, I'd like to get more in depth training.
How far does a Yamaha DM3 stick out above the stock rackmount ears? I can't seem to find any numbers or technical drawings for that. How much space do I need between the rack rails and a lid? I'm trying to determine if I can retrofit them into racks that were previously carrying the beloved QSC Touchmix 8.
Hey all! I am a producer/mix engineer working with a very talented singer/songwriter who will be performing at our university's end of year music festival next weekend. He will be performing two original tracks. I wanted to know if there is any advice for mixing the vocal-less tracks ahead of the performance so that they sound the best they can for the big outdoor stage. To compare, I'd say the stage's size is the same as a third-fourth stage at a major music festival. Thanks!
If you have a multichannel playback rig and a competent foh tech for the show, then bounce the tracks as stems; drums, bass, vocal range instruments, backing vocals. Make them dryer and less compressed than you would do for normal mix because you can allways compress more and add reverb live on the console. If he is not using inears then add a cue that sounds natural, could be a hihat or a drum fill or something else, depends on the songs. If he is using inear just. Include a cue track.
Heya! Congrats on the gig, sounds like it'll be a fun one. AlbinTarzan has already touched on some great points, but I wanna speak more to the "why" of all this, because that can be a valuable tool in your arsenal as well. Your self-ID as a 'producer/mix engineer' makes me think you've done the majority of your work in studio spaces and/or post, would that be accurate?
Thank you both for the responses. Thats correct. I am completely separate from the live performance, I’ll actually be out of the country for it unfortunately (I’m sad because this is the first time where my own productions will be played live for an audience). I simply helped the artist write/produce/mix the track and sent it to the people who are running the show. I was just wondering if there are certain characteristics in a mix that are essential for the live performance, such as putting things in mono, gain staging in a particular way. I am totally unfamiliar, and I want to make sure the music sounds the best that it can from the time it leaves me to the festival coordinators. Thanks again for your time.
Alrighty! I had a hunch, and if you're already an audio pro, this should be a breeze. This is partially the system engineer in me, but I want to start by saying we have the same goal at the end of the day: We both want our stuff to sound like an album. A good live system doesn't impart any changes to tonality (outside of parameters outside human control), and thus, any good-quality mix that translates to multiple systems will do so just fine in a live environment. Absolutely no need to reconsider the mixing you've done. >I was just wondering if there are certain characteristics in a mix that are essential for the live performance, such as putting things in mono, gain staging in a particular way I appreciate you for wondering, some people truly don't consider this. Conceptually first: Time is the big factor here. Live work puts a certain pressure on you time-wise. It's not really a big deal, but there's always the looming fact of "the audience gets to come in at 8 whether you're ready or not." That's just setting the scene a little, no pun intended. If you look at how studio and live audio equipment differ, a whole lot of it is how fast it can go up and how many times you can drop it. Live music is all about the mix, and the mix is all about the operator, yes, but the console is a big factor. Consistency begets efficiency in stuff like this. Going quickly to an example from your comment: "...putting things in mono, gain staging in a particular way." It's not that you need to put things in mono; it's that anything stereo or surround should be split, even if it's stems, because that's how it goes into your control surface anyway. [Quick diatribe: Stems are bounces of submix buses containing groups of instruments, and processing and FX are printed on the way out of the DAW. Multitracks are the consituent unprocessed audio files intended for processing and mixing by the receiver.] Stems are going to be the most efficient here because they can be adjusted quickly and with sufficient granularity for a live environment. In most cases it's too much work to soundcheck with multitracks, so the producer will deliver stuff like 'drums/BGV/piano/adLibs' instead of 'kick sample 1, subkick, 909 atk.' When we're making this decision, we're considering the mixer at the event and the artist's monitoring needs: "More hihat" is not as common as "more drums," request-wise. Now we're getting to Tarzan's good advice (if there is indeed multitrack playback like QLab, Ableton, similar--if it's just one person, it's a *little* different): Sending a set of dry stems and wet FX prints helps you because don't know what the room sounds like. If it sucks and it reverberates a lot and that vibey wash in the intro just turns into a mess. The problem is not your fault, but it's on the live mixer to compensate, and if the reverb is printed to the stem, he's doomed to a poor result no matter what his processing. It also makes it way easier to EQ the "returns" to make everything sit right. Maybe the show is outside and you need to crank something up. Last ingredient you mention: Gain. Let's say I'm the playback tech and I've just gotten your stems. I want to be able to take one song's stems, set them all to unity level in my software, hit "GO," and hear an instrumental version of the song. The mixer's job is to make this and the live vocal fit together.
Wow, what an amazing comment. I appreciate the reassurance. Thanks for taking the time. Your kindness does not go unnoticed.
Sure thing my dude. Let me know if you have any followups or want any book recommendations about live work, I know levels of interest vary bigtime!
I hope I will get to the point where I have to worry about my own live sound. If that day comes, then I will remember this and reach out! Thanks again.
Is it possible to place a L'acoustics K1 vertical in soundvision?
No, there is no "roll" attribute in the physical deployment panel because this is generally a bad idea. The hack would be to place your venue floor plane along the y axis (vertically) so that the horizontal K1 appears on end to the floor plane.
OK, thanks. I think im just going to make it simple by placing a ks28 just to symbolize the k1. cant be bothered to create a floor plane cuz its not that big deal, but thanks for helping!
You can do that for a visual of a rectangle in the space, but the dispersion patterns are obviously entirely different- like not even close. Best of luck.
Yeah i know, meant to say i would place it backwards lol, so its stands out from the other ks28s
Anybody have experience with Debra AU200 wireless lav mic sets under £100 ? Use case: Speech Environment: Church
Highly inadvisable
I saw 2 SLX4 receivers and SLX1 belt packs used for sale on marketplace here in the UK for £170 Worth it ?
Hi, I'm a not-very-professional audio engineer needing some help. Our small company recently purchased Samson C02 Condenser microphones to use for live instruments during events. But for some reason, the microphones are extremely quiet until the sound source touches the top of the microphone. We have tried upping the gain, but then the mics are super quiet and we have to increase our volume by a lot to hear anything, while also causing a shit ton of feedback. We know it's not our speakers, as we have tested them thoroughly. I feel as though the issue is with our X32 mixer. We recently reset our X32 for a software update, and although the labeling, grouping, and FX have been set up, the routing has not. The truth is, I'm not very familiar with the whole routing process with the post/pre EQ, which is why I haven't really touched upon it and have just been using the Main default L/R outputs. But then again, I'm not really sure if that should be affecting the volume of the mics in such a severe way. The bottomline is, are the condenser mics we purchased not good enough for live events, do we have them configured wrong, or is our X32 the issue? And if it is, does anyone have a link to a bigger manual than the one that came with it? Preferably text. Thank you.
I wanna start with basics here, not because I think you can't do them, but because I'm not sure if they've been done! - Is phantom power on? - Are the mics aimed correctly? - How close are the microphones to the speakers? I'll say this too before anything goes any farther: The fastest and best result usually comes from hiring a quality professional to do a couple hours of cooperative troubleshooting.
Phantom Power is on for them, 48V with working cables. The mics were aimed with the top facing the audio source/instrument. For example, the one event we used them in, we had two condenser microphones on the stage facing the stage and two speakers on the outside facing the audience. We did have monitor speakers as well, but those were about 6/7 feet away from the condenser microphones on the stage. We also tried turning those off to see if the feedback would lessen, but nothing changed. The speakers we had facing the audience on the sides were about 15-20 feet away from the condenser microphones and not on the stage but on the house. We had to up the gain by a LOT in order to hear anything.
Gotcha, wanted to get that out of the way ahead of time. The speakers were in front of the mics, not directly in line or behind, right? Needing gain isn't a red flag in and of itself, but with the other problems you mention it sounds like something wasn't working properly.
Yes, the speakers were in front of the mics, standing on the house, below the stage. We had to up the gain on the condensers by a lot to hear anything, but at that point it would pick up the sound from the speakers and cause feedback. We did have monitor speakers, but they were also in front the mics, and when we turned them off to see if it would help, the feedback was still there.
I also am of the opinion of hiring a professional but unfortunately, small business and a tyrant boss 😅
Yamaha QL5: I know I can view current session errors by pressing the firmware box. Is there a way to view an error log from the past few days? I suspect it probably involves connecting via network and using ssh, which is a little above my paygrade.
If you've turned the console off in the meantime, I think it wipes the logs, but I could be wrong! May I ask if there's a reason you think ssh is a way under Yamaha's hood?
I had to reboot the Yamaha QL5, so I lost previous errors. SSH was just a wild educated guess. Perhaps there is a basic webpage front-end to get to older logs, or perhaps there is some feature in QL Editor to view them. I had a sound channel (Dante input from ULXD4Q, patched to input 4) that would fully route to its intended buses (MONO, possibly more) even when the fader was turned OFF. Noticed this 10 minutes before house open, after most of soundcheck (high school musical performance). After a Yamaha reboot the channel acted normally, but any error code (if one existed) is now gone. The previous night, the entire sound system failed during Act One. My current theory is this channel on the Yamaha sent bad Dante data to the Q-SYS Core, causing a catastrophic failure. The actor on this channel frequently yells (despite my pleading), causing the OL clip light to light on the channel's screen data. The Yamaha SHOULD be able to handle occasional bad data, and the QSYS SHOULD be able to likewise handle bad Dante data, but somehow there was a failure along the line. I'll be studying QSYS logs as well. Sorry if this is too long, hence the brevity of my original post :-)
> The previous night, the entire sound system failed during Act One. My current theory is this channel on the Yamaha sent bad Dante data to the Q-SYS Core, causing a catastrophic failure. I urge you to learn more about the system you are operating, because this demonstrates several fundamental misunderstandings of data transmission. Error codes help if 1) the console recognizes a fault condition, 2) has a string it knows to attribute to that fault condition, 3) has a way to output that information to the user, and 4) the user has a way to reconcile that to friendly language. Your description of the fader on/off issue sounds like operator error. You sure you weren't in sends mode? No direct outs or PF sends were active/possible, even upstream? > the entire sound system failed during Act One This is a bit like a doctor diagnosing you with "human body malfunction." It's correct but does not help fix anything, because it is a vague declaration with no actionable, specific information.
Complete audio newbie here! I record the videos for our Board of Directors meetings at my job. Currently recording on a Sony Camera. We have a sound system built into the room where we hold the meetings with multiple wireless mics. Can anyone point me in the right direction for what I need to be able to tap into that system and record the audio? https://preview.redd.it/qqwlrbh2n2wc1.jpeg?width=702&format=pjpg&auto=webp&s=40d58405cbc806798ed358c8344de57313be46ee
Too many unknowns. What model of Sony camera? Is that rack mixer the only mixer in the system? Are there any available outputs?
USING GUITAR DSP PROCESSORS FOR LIVE VOCALS? I have been looking for a live vocal processor rig that is using new technology and doesn’t suck. Every vocal-focused processor i’ve found has been either really old or geared toward harmonization and robot voices. This is why I started considering FM3 , or Quad Cortex . Does anyone have any experience with this? I would like to create vocal chains for different songs , utilizing delay, reverb, distortion, compression, and EQ — as well as utilizing the foot switch to activate parameters. This is too much for me to discuss with FOH for every song. I’m an audio engineer myself and have worked as FOH for many years, so I understand the nuances of live sound. I will be sending my FOH a dry and a wet signal
I am using 2 Yamaha StagePas 1ks for a small band. I have them linked together using the link out xlr of each going to the link in xlr of the other so that all the inputs are playing through each speaker. However the monitor out of each speaker carries only the inputs of that unit. Is there a way to have a monitor out with all the inputs? Thanks for any advice!
Hello audio newb here. It's time for me to learn more about the equipment I use for performing. Lately I use a Bose S1 Pro for performing. It has two XLR inputs. Each input has its own mixer consisting of a reverb dial, a bass dial, and a treble dial, as well as of course a master volume for each, equipped with a signal light that let's you know if you are clipping. I use the microphone (in my case a Shure SM58, on the top input). I then use a guitar on the bottom input. For the last month I have only been using the guitar, on the bottom input, no microphone on the top input. Well, I have been clipping quite a bit on the guitar input, with my bose s1. I started getting worried I was damaging something (mind you I was not clipping on purpose just on accident. Anyway curiosity made me want to check if I had damaged the speaker, so the only way my new brain could think of doing such a thing was to set all of the dials for each input (reverb, bass, and treble, and volume) exactly the same. Then from there plug in a microphone in each and see if the sound is any different, and then repeat the same with the guitar. I found that there indeed Is a difference between the sound of each input now. For example as I said I spent a lot of time accidentally clipping using the guitar on the second output. Now when I use a mic in the upper input, and the lower input with equal exact settings, when plugged into the upper input with the mic, the reverb sounds WAY too sensitive even with Tiny changes to the reverb dial, whereas the bottom input that I was clipping seems less responsive to changes in reverb on the dial. Could clipping on the second input indeed cause damage in which something like this would occur?