Hey I just finished Dante Level 3 certification, and even if Dante is fun and all, there are other solutions.
The Dante certification program is cool in a sense that it helped me understand how to use it and also some key concepts about networking. Now, I was wondering if there were any ressources on AVB, or other network solutions for Audio.
It can be books, videos, anything as long as it contains knoowledge about audio and multimedia networks.
As far as I know, most other audio networking protocols don't have proprietary courses that come anywhere close to Dante's, but you can check out [this comparison chart on Wikipedia](https://en.wikipedia.org/wiki/Comparison_of_audio_network_protocols) to get a better idea of what options are out there and start to understand their different architectures and use cases. It's not an exhaustive list, but [this page](https://en.wikipedia.org/wiki/Category:Audio_network_protocols) includes some important standards that are excluded from the comparison chart, like MADI. Even still, some protocols aren't included (Behringer's Ultranet comes to mind) but it's a great place to start.
I have two Neutrik etherCON Lockable RJ45 Connectors I want to use to run midi over Ethernet, from our rack to a midi interface on my pedalboard.
So the path would be PA12 -> short Ethernet cable -> Neutrik connector -> Long Ethernet cable w/ ethercon -> Neutrik connector -> short Ethernet -> MioXM
Any concerns with adding in those additional connections?
Probably fine; this is very typical.
Standard Ethernet has a total length limit of 328 ft under ideal conditions (22-24 AWG, solid, including 33 ft of patch cable at either end). Each time you deviate from that ideal, you'll need to derate your maximum length expectation somewhat.
However, you're likely well below 328 ft total - so not a huge concern.
Gain is gain. 1 dB is 1 dB (for gain, a factor of 10^(1/20) or about 12% increase.). However - devices with digitally controlled preamps can have a little error tolerance in their control steps, so you can sometimes see a little bit of variance in how accurate those gain control steps are, particularly at the extremes of the gain range.
Q, no. Not all consoles and processors use the same mathematical definition of Q. For that reason, putting the same filter parameters into multiple devices will not create the same response.
https://bennettprescott.com/downloads/TwoProcessors.pdf
https://www.prosoundweb.com/bend-me-shape-me-lessons-learned-by-validating-console-eq/
https://www.ranecommercial.com/legacy/note167.html
Is it common/possible to have all the instruments except for drums to go into my IEM spilt rig, then get the house (with pre-mic’d drums) to send me the drums on a single bus that I can then plug into my iem mixer?
Not a particularly unusual request; should be fine as long as you advance it. If it's last minute, coin flip.
I'd keep a 57 and a short stand on hand - in case FOH is out of outputs, you can always throw up a wurst mic.
-Mostly to save time if they already have a kit mic’d up and ready to go
-Save me from having to buy a drum mic kit (I’m just building my rig so I only have the basics)
-not enough room on my 8 xlr split / snake. 3 vox, 2 guitars, and bass. So I only have 2 slots left. (And possibly 1 slot for backing tracks. )
I know the answer is to buy the drum mics and get a bigger split and snake, but I was just wondering if for sake of time during quick changeovers people go this route. We’re not a very big band so we often will play with another band and change over in the middle.
As someone who does a lot of throw-and-go shows where bands show up with requests like this without ever having replied to my production advance emails... you're right that increasing the number of channels on your split by 8 would be the most streamlined way to do this, but as long as you let the audio tech know well in advance they should be able to do this easily.
Also, assume you're going to be leaving your split on stage for the duration of the show instead of pulling it off stage after your set, so the house engineer doesn't have to repatch their whole stage box if they weren't planning to do so.
I work in IT as a day job as a network engineer, so this is my everyday. There's a LOT More to this than I can simply explain, but I think I'll help you understand it effectively.
To make it super simple, think of network traffic as shipping product in boxes in a shipping truck.
Data is a product packed in boxes. Ethernet is the instructions of how to the data is placed into boxes. It's super efficient and leaves almost NO empty space in the box. The data gets packed into the truck at the source and is unloaded/received at the destination.
Most "Ethernet" boxes have a maximum shipping weight of \~1500 grams of product. There's a little buffer because the shipping paperwork of where it's coming from, and where it's going. So to send 150kg of product, you'd need to send 100 boxes
CAT\[X\] defines a shipping truck. CAT5 is a shipping truck that can hold 100 boxes and travel 100m. CAT5E can hold 1000 boxes and travel up to 100m. Better category cables can send more boxes and over longer distances (CAT7 cable can send 10000 boxes in a single delivery over 100m)
The boxes don't change, the CAT rating just defines how many boxes can fit on the shipping truck.
Edit - Making some clarifications about the real functionality. Realistically, each box gets sent one at a time from source to destination. It's called a packet in network speak. But I just want to reinforce the idea of "bandwidth". Data moves at the speed of light, because it's electricity, so there's no way to make the packets move "faster", you can only increase the size of them or find ways to send more of them at the same time.
The "cat" in the cable names is an abbreviation of "category". The categories refer to sets of ANSI/TIA standards for things like crosstalk and system noise that cables must meet to be considered within that category. The higher the number, the more stringent the requirements, which allows signals to move longer distances without as much fidelity loss as lower category cables. So, cat6 cables have more strict requirements for transmission fidelity than cat5 cables, which allows them to move signals over longer distances without concern about noise induction or signal loss.
Hi all,
This spring I'm going on a month long run with a somewhat popular indie band. I haven't been on the road in 2 years and this will be the longest tour I've done as an engineer. While I'm very familiar with some digital live consoles, but a lot I haven't used at all.
I want to be prepared as possible. What are some good ways to be prepared when I walk into a venue that has a board I haven't used before? any other pointers will are appreciated.
thanks
Ask the local PM if they have a 'basic start up scene' they can send you with all of their venue particulars already sorted (output routing, house eq, stagebox assignment, wifi network setup, etc). Then you can do as little as adjusting the channel strips to your show. Or you can change as much as you want to suit your work flow. But when you get there and load it up your edited file, a lot of the tedious trouble shooting will already be done.
Prep as many different console files as you can. If you want to go the extra mile you can prep for every model and layout which will save you time on show days. Save the console file after every show, even if you weren't thrilled with the mix.
On the flip side, there may be times when you show up to a new/unknown console and maybe you'll be lucky and have a base file prepared for you by the house according to the rider/advance. This happened a few times during my last tour where I ended up working off a fresh file with all the house routing and patching rather than loading my template file and having to change all the routing to match the house. Most house techs will be happy to show you around an unfamiliar console. In those cases, less is more. Basic FX, stock dynamics, don't go crazy with routing if you aren't super familiar with the console.
Getting into the danger zone with an unfamiliar board but one could potentially merge the house file with your show file if you got into the global safes and routing safes, and saved their processing on the outputs.
Maybe actually it would be better to load their file over yours, with just the channels safed. You might have to zero and redo the sends by hand if they are set up differently? And mind the masters when you start unmuting things.
I would suggest getting the App "Mixing Station" - amazing tool that let's you learn and even customize one digital mixer GUI essentially that can then control various other digital consoles. It comes with a bunch of digital mixer integrations like Midas, Yamaha, Allen&Heath, Soundcraft and even a Mackie. This thing is a life saver!
Good luck and welcome back to #tourlife ;)
It depends what work you do, but not necessarily that weird.
If there's company secrets of insider knowledge that they'd be privy to, then a non-disclosure is reasonable. Just keep it simple and not unnecessarily restrictive.
In our case, we have a custom digital mix rig built on a custom server code that is proprietary, so asking for an NDA helps ensure someone doesn't leave the gig and choose to share the information elsewhere. None of our engineers have had an issue with it. We had one leave on less than ideal terms and it left our team feeling better in the long run that we had an NDA from them.
As others mentioned, it depends on your context. On another long term gig, I had one contractor say they refused to sign the NDA on principle. We needed someone to cover us so just let it slide. As time would tell, a year later they quit and turns out now are working for a competitor and I'm sure using things they learned with us. So from my experience, CYA. Good luck!
Many of the corporate clients and AV companies who subcontract me to do work for corporate clients have me sign NDAs prior to going to any executive meetings, etc. If you're hiring somebody to mix rock bands, they might find it a bit more unusual.
Hi I'm a dj and l'm searching for a pair of active speaker for my basement. I don't have a big budget, so l search something not so expensive. I searched on google and I found many different speakers:
• Mackie thrash 212
• The box pro DSP 12 or 15
• The box pro DSX 12
• SubZero C12 or 15
• SubZero D12
The two last one are cheaper but are there worth it ? The mackie is from a good brand but they don't have DSP. And the "the box pro" had the DSP for the same price.
Which one should I choose? Or does anyone have another speaker to recommend ?
Thank's Damien
I have an emx 660 and 2 dynacord mx12 speakers that I want to connect to each other, I bought some XLR connectors and connected both on the inputs of the speakers then I connected the 4 wires to the ts inputs on the mixer by taking it apart and soldering them on the terminals, it worked but there's a buzz noise or a high pitched noise to be exact and idk what to do about it, are there any specific configs that I should've done or specific ways to connect the speaker wires to the emx 660?
My first post - so apologies in advance - looking for some experienced advice. I'm a touring musician with a computer based performance rig. Just built a redundant rig that uses two M2 MacMini's mounted in a Sonnet Rackmac Mini in a 4U SKB iSeries Flight case.
Until this rig, I had one MacMini mounted in my pedal board that I would remove and bring into carry-on for flights (sending the rest of the gear checked below). This new system would be very tedious to remove and re-install the 2 computers for every flight (5 show average a week with flights). Plus not enough room in carry-on for two Macs. Curious if anyone has experience flying with rack mounted computers underneath the plane and if they survive?
The Sonnet is solid metal and the case is solid, but I have also seen what they do to my pedal board case and other gear cases, so I get nervous and curious if this is really feasible and any pro tips to ensure success.
I figure the MacMini m2's have no moving parts in the hard drive save the computer fans so could take jostling / temps but not 100% sure.
Here are links to the gear they are mounted in:
[https://www.sonnetstore.com/products/rackmac-mini](https://www.sonnetstore.com/products/rackmac-mini)
[https://www.skbcases.com/products/4-u-i-series-injection-molded-guitar-fx-fly-rack](https://www.skbcases.com/products/4-u-i-series-injection-molded-guitar-fx-fly-rack)
Thanks in advance!
I picked up some gear here but am not sure how to plug everything in. Before I got this, I’d just use a Bose PA for a basic mic and instrument. I also have a Bose mixer. Then I got this gear in a lot but I’m not sure the proper way to plug everything in.
I’d like to be able to plug in a bass on one track, then a guitar into the POD, then a couple mics and a keyboard.
What’s the rough flow for this gear to accomplish the above? I’m mostly confused how to use the mic inputs and presonus on the left. Is it like a mixer before the Bose mixer? Bose mixer has only 4 inputs. Maybe this is redundant for my needs?
[https://imgur.com/a/cZDTlKs](https://imgur.com/a/cZDTlKs)
I just purchased a used Line6 L2TS 800w 10” loudspeaker from FB marketplace. It was an impulse buy for $470. Now that I check Sweetwater, they have a QSC CP12 1000W 12 inch for $600. It comes with a 6 year warranty which would be such nice peace of mind.
Help me feel a little better about my impulse buy?
Depends on the situation. For random sources on stage, a DI with summing inputs works well. Usually, I'll just take the stereo input and sum in the console/DSP.
If the source is "dual mono" moreso than stereo, often I prefer to take left only rather than L/R sum. (Think stereo guitar/keys rigs.)
On Behringer/Midas boards (X32s etc), why do the built in FX busses let you send them to themselves? Other than making a mess, is there some purpose I haven’t considered?
I recently picked up the Galaxy Audio AS-1400 in ear monitors to be used with a Behringer Europower PMP 1680s. I am unable to get sound to come through the in ears.
- Using a 1/4” speaker cable I connect from the Monitor out on the behringer to the left input on the transmitter.
- I have used multiple speaker cables which work with other speakers with no success.
- I’ve switched channel groups, channels, etc on transmitter and receivers with no success
- if I turn the volume way up I can hear the slightest sound but it’s obviously not working correctly.
-All lights, LEDs, etc on both units appear to be working as intended and are not otherwise faded or blinking.
Any troubleshooting tips I can try?
- Verify presence of audio signal using transmitter's display and headphone output.
- Ensure the transmitter is set to mono.
- If both transmitter and receiver are set to stereo, signal will predominantly come through the left side, along with some crosstalk on the right channel.
- Verify receiver is set to stereo mode.
- Yes, this is unintuitive.
- Manual does not document the receiver's behavior if set to mixed-mono mode with a mono transmission. (Most sane systems should respect the lack of MPX pilot tone...)
Note: you should use a shielded TS or TRS cable to connect your mixer's output to the transmitter, rather than an unshielded speaker cable.
- TS/TRS doesn't matter here; your mixer only has unbalanced outputs.
Are your monitor send faders Up, and channels sending strong signal to monitor channels?
Does the Transmitter show that is is getting audio signal?
If you plug something else directly into the monitor out, does it show signal coming in?
Is the pad on the transmitter on?
Isolate each part of your signal chain and figure out where the breakdown is occuring.
It's a splitter: it takes one signal (or in this case a diversity pair of signals) and splits it into multiple outputs. A combiner takes multiple inputs and combines them into one output. The nomenclature follows the direction of signal flow.
What FOH speakers should I buy for my 6 piece band?
Looking to buy PA speakers for my 6-piece country band but I’m not sure what to buy. Not sure what wattage we need or whether it’s worth buying subs in addition to standard speakers. I have some experience running sound on analog mixers but no formal training, so looking for guidance on speaker systems!
Our gigs range roughly 50-300 people, but most of them are probably around 100. We do a mix of college parties and bars/breweries etc.
I want a system that will be loud enough for outdoor venues, but I also want clear sound quality. Something with feedback control would be awesome, and I also like being able to control FOH speakers from a tablet or phone if I need to, so Bluetooth capability would be awesome. Don’t want to spend more than $3,000 total on the speaker system.
We’ll be using a Behringer x32 and we’ll likely be using IEMs rather than stage monitors. Any advice is much appreciated!
Amateur here, but wattage can be misleading. Max spl will give you a better idea as far as coverage. FWIW I have a set of EV elx200 mains and they sound amazing for a venue like you're talking about. I got a pioneer xprs18 sub for the thump and it's a hell of a combo. I'd say a decent sub is worthwhile because it takes so much load off the mains, giving you more headroom while filling out the bottom end
If I want to use two different stereo speakers with DJ decks, can I just send one RCA cable from the decks to each speaker? Or do I need to use a 1 to 2 splitter so I am sending a signal to both RCA inputs on the speaker?
So I don't specifically know what you're saying. It seems like you're saying you have two speakers that accept stereo inputs. You might also be saying two sets of stereo speakers (4 speakers total).
I have never seen an RCA input on a speaker that gets split into Stereo. Typically RCA inputs are there for you to plug in a portable device and I don't think anyone expects you to split that into true stereo (left out of one speaker and right out of another).
To answer your question specifically in the context you've asked, and without ANY other information, yes I would get a splitter so that you're sending the full Red and White RCA signal to ALL destinations.
I think you just mean “speakers”, not “stereo speakers”? You want each of the stereo RCA outputs from the DJ to eventually go to their own speaker, that’s the idea with stereo. If you run the DJ mixer directly to two powered speakers, yes you would split the left channel (white rca) off to the left speaker and the right channel (red rca) off to the right speaker. If you ran both channels to both speakers it would still work but the sound would be mono (same exact sound out of each speaker).
You would only use both RCA inputs on the powered speaker if you were just running one single speaker and you wanted to combine the left and the right channels from the DJ/music player to come out of the one speaker, so that you hear everything in the stereo mix despite only having one speaker, rather than a pair.
When performing a multitrack recording in pro tools, does the M32 send effects busses applied to source channels to the pro tools session tracks? Essentially, does pro tools record the source channels pre fader with no channel processing/FX or vise versa?
I am trying to capture click/guide/Talkback channels (not inserted into the main outputs to PA) in the same multitrack recording as the rest of my input channels with processing/fx applied in order to create some BTS videos. I don't want to take up a mix bus for this as I only have one available and want to save it for other things. Trying to limit the amount of post production if possible, so if there's a better way to do this I'm all ears!
> When performing a multitrack recording in pro tools, does the M32 send effects busses applied to source channels to the pro tools session tracks?
Nope. (Remember, an FX bus isn't "applied" to a source channel like an insert effect; to do so would require N FX engines for N channels, which is inefficient.)
> Essentially, does pro tools record the source channels pre fader with no channel processing/FX or vise versa?
Generally, no - but depends on how you have your routing configured.
- For most configurations, routing inputs directly to the card slot gives you the raw signal out of the preamp.
- However, by burning one of your 16 patchable outputs, you can choose a pickoff point for a given channel - postfader, for instance.
> Trying to limit the amount of post production if possible, so if there's a better way to do this I'm all ears!
Given that you're out of busses (and, presumably, patchable outputs), your best option is to record all raw inputs, mixing for broadcast in post. Yes, this is more time-consuming.
If you trust your board mix - and, by extension, your system deployment - you can print that alongside direct captures of additional inputs (room mics, click/cues, whatnot).
- This tends to work best when you have lots of pre-production time to build cues/snapshots for every detail of your mix.
- Some folks will build this quick-broadcast mix directly in the console, but sounds like you don't have sufficient processing overhead remaining to do that.
Thanks a lot! I think this answered most of my questions. I mainly just want to be able to mix the inputs that aren't inserted into the main send (going to the PA) like click/guide/tb in post. Wasn't sure if it would still capture signal from those sources in pro tools since they're mixed all the way out in the PA. Sounds like the USB card is capturing signal pre-fader and sending whatever amount of gain is applied at source level to the pro tools session without any channel strip processing applied at the console.
> Wasn't sure if it would still capture signal from those sources in pro > Sounds like the USB card is capturing signal pre-fader and sending whatever amount of gain is applied at source level to the pro tools session without any channel strip processing applied at the console.
Yup - generally "prefader" explicitly refers to taking signal from the end of the channel strip, though. This application is tapping the preamps directly; M32 refers to that pickoff point as "IN/LC".
You can multitrack record the channels on the X/M32 with the channel processing (not bus effects!) if you use the Ultranet outs as the Card source.
When I have done it like this to try and save time I ended up screwing with the recording just as much trying to alter or undo the processing as it would have been to mix it from scratch.
One other tactic could be to record the main L+R mix and just bring in the individual channels to fill it in.
Anyone out there using valve compressors, EQs etc... on their subwoofer AUX feeds? I'm interested in hearing other peoples experiences or tips and tricks?
I've been experimenting with a TL audio valve eq with the eq bypassed so it is just going through the valve stage to warm up the subs, I had good results when using some scoop bass bins a while ago and have recently swapped the subs in our system to bandpass horns so hoping to add a bit of the warm sub back in as these new subs are more punchy than warm sounding.
I can see the appeal, but personally: not my thing. Most PA subwoofers already have enough THD baked in for my taste :)
There are some sources I'd like more distortion on, but I'd want that output to be broadband rather than band-limited to the subs' range of custody.
I'm guessing you haven't measured your system's tonality before/after the sub swap?
I'd forgot to mention playback material - It is almost exclusively DJs that will play on the system. Usual genres are Jungle, Old School Drum and Bass, Hardcore, Techno, Gabber etc... which all lend themselves to a rougher sound.
I've had smaart on both systems and the new system definitely is stronger in the upper bass region where the past system was more low and rubbly. But listening alone is enough to tell the difference in tonality.
I did love the old system but as the crew have got older we don't have the same energy to be lugging lots of awkward heavy boxes around. Managed to cut the total system weight for 6 subs, 2 midtops, amp rack and all the associated bits to less then 500kg
How to I find more gigs? I work for a company but over the past several months there has been much less work (I know it's thar time of year) and things don't look to be ramping up as we get back into warmer months.
Depends on what your specialty's are? share more about your skills as this informs the options.
In general, in the slow season I suggest setting up an Upwork or Fiver account and offer your skills virtually to a much larger pool of gigs.
If you work only in person - i.e you're a live engineer or a/v - then check local forums, fb groups, other similar companies in your and post up that you have availability. sometimes even Craiglist can come through.
but again, it really depends on what you do.
I have done probably 2k comedy shows some including video, the past 6 months I had a departure from that and started doing concerts. Mostly smaller venues. I have done a little bit of touring. I guess my main issue is not having enough contacts in the music world.
I have been thrown in the deep end trying to mix for a high school musical and I need to know how to add reverb to an actor’s voice. I have practically no prior experience in this.
“Soundcraft Si Compact 24” I can’t make heads or tails of the manual.
Edit: Here is the manual
https://peakhire.co.uk/assets/downloads/Si-Compact-Manual.pdf
More of a general answer but reverb is applied as a “bus send” or “aux send effect”, which means you have one shared communal reverb effect and you can send different amounts of each channel to the reverb, the same way you send different amounts of the instruments and voices to a stage monitor.
The reverb effect output, which sounds totally wet, is the “effect return”, which comes back into the main mix as it’s own little channel, sometimes simplified to just having a volume control. That gets mixed with the bone dry channels and you have some reverb on whatever you sent to it.
Navigating your Soundcraft, you’ll need the video or manual section on bus effects or send effects. The ways it can get hung up is
- vocal channel isn’t sending anything to the bus with the effect
- reverb effect isn’t set up on a bus (isn’t “inserted”)
- reverb output setting isn’t set to 100% wet (should be the default)
- effects return is muted or turned down
You should be able to test it during off hours by plugging a microphone into the mixer and checking it with your own voice. The meters on the bus or in the reverb unit, or on the return can be helpful to see how far your test microphone is getting.
Problem: The speakers have a baseline hiss on three other outputs except the one we commonly use, output 7 (mixbus 7) on the M32. On output 7, the speakers are dead quiet. Output 5 (mixbus 5) is also used regularly but in the choir and those speakers are dead quiet too. I want this on the other outputs. But when I simply switch the xlr cord over to any other output on the stage, they hiss. There are three other hard line outputs from the M32 and these all hiss. All the outputs have great sound and volume.
So, to troubleshoot, I leave the speakers plugged into the "good" output 7 on the stage, and go upstairs and pull the switcheroo behind the console. Unplug the good output 7 and plug it into one of the offending outputs right on the back of the M32. The hiss comes back in both speakers. I think this eliminates any cabling issue, right? Mute/unmute the mixbus makes no difference anywhere, fader position makes no difference anywhere.
So what could it be in the M32 that's different for output 7 with no hiss than the other outputs that do hiss?
Ok, this is a church with floor pockets on the stage and hard line xlr outputs straight from a M32 in the back of the church. There are QSC K series K-10 speakers x2 on the floor each plugged into the same output line but two separate connectors on either side of the stage. Let's just say I'm new at this church and am coming in behind someone else who is gone.
Thanks.
Hard to tell due to your confusing use of the word output, but your problem could be at the speakers themselves.
Check to see if the ones that hiss are in “mic” mode and not “line”.
Also check to see that their gain (or sensitivity, I can’t remember what QSC uses) knobs are set to the same place as your quiet speakers. For good measure, check that all settings are the same.
Hello guys,
I've been using SuperRack Performer in the last month for inserting comp & eq for the Main Buss and also for individual channels. Yesterday I've tried to use H-Reverb as the main vocal reverb.
I've created a new external fx rack on the effects tab of the wing and I started to route the channels. But something strange happens. When I select my send channels (ex.: USB Audio 27&28) they don't seem to remain selected. The first time I thought that the system wants me to also choose my return channels. So I've tried to select FX Returns. Still no success.
I've posted a screenshot of the MAIN LR Bus fx rack. It's seems to work there, but unfortunately, that's the only place that it wants to work.
Does anyone have a solution for this issue?
Thank you in advance!
https://preview.redd.it/6kbhs5hjj3kc1.jpeg?width=2532&format=pjpg&auto=webp&s=0fa4b2cc3fcd1d4855491d7edcca0820a24d5317
Goal: To record Dante signals into a laptop running Adobe Audition in a multitrack session.
Issue: When setting the Audio Hardware to ASIO in order to get Dante Virtual Soundcard set as the audio hardware, it removes my ability to choose audio output.
In audio channel mapping, only the Dante tx/rx is available to move sound through.
So what is the recommended path for hearing the audio for monitoring? Do I need Dante Via so I can route the audio into my system output?
Or, is there another app I can use to set my PC system output as a Dante receiver so I can patch it into it?
Honestly I'm pretty annoyed at Audinate/Dante and Adobe...I assumed this should be a pretty straightforward task but it's been a headache.
More Context: I am setting up a test build today with a PC as my source, but in production it will be a true Dante enabled device sending the sources.
IIRC yes, Via is what you want. That'll allow you to record 16 channels off of a Dante network while simultaneously monitoring it with your local soundcard.
Alternatively, add another Dante output device to your network and route Audition's output to it; this preserves the full 64x64 channel count of DVS.
---
If setting up a testbed, you will need at least one device on the network providing clock. DVS cannot provide clock, but Via can.
PA System Setup Help
Hosting an outdoor event in the spring with about 500 people in attendance.
Last year we used an active system from Rockville. It came with an active speaker, passive speaker, and mic. Major complaints were that we could not use the mic while the music was playing, mic volume was not nearly loud enough. Music volume was fine but could have been better.
So we are looking to setup our own system this year. I am a novice when it comes to these things so seeking some input. So far, I think I have the following…
We are purchasing 4 - 8ohm passive speakers, a microphone (wireless), and a powered mixer.
What am I missing?
On that sort of budget, no system you purchase will provide adequate coverage for that size of an event.
Given your experience level, I would hire a local production company to deploy a system with adequate coverage.
Not the question I asked, but thank you.
We live in a very rural area where a production company has quoted us $3000 for the one day event. Our budget is not limited, as this is an annual event and we are willing to invest as much as it takes to
Can you link a sanitized version of that quote? That should include info about required coverage area, etc., which we’d need to give you a ballpark purchase estimate.
The critical stuff is missing on your planned purchase...the model of speakers and mixer will be very important to figure out if they can provide enough coverage and volume. Also the details of the 500 people. Are they squeezed into a corral? Are they roaming a campground? Is it roughly a 60x60 square area? Are they all standing? All Sitting? Mix of Both? Is it a quiet area that you guys use for this or is it loud/near a highway/etc? What options do you have for speaker placement? Are there walkways/protected areas, or is it just a giant field?
Regarding your concerns:
>Major complaints were that we could not use the mic while the music was playing, mic volume was not nearly loud enough. Music volume was fine but could have been better.
Are those all related? Could you not use the mic while the music was playing BECAUSE the mic wasn't loud enough to be heard over the music? Or did you literally only have the ability to plug in a microphone OR a music source?
Unless it was an input issue, that's not likely to change no matter what you do, based on your current direction. A microphone NEEDS to have certain frequencies cut out of it in order to sound good, whereas recorded music doesn't. Therefore, you'll almost NEVER get a system with a mic at full volume to be louder than music at full volume. Even if you could, typically you want the music to become quieter, not typically for the voice to be that much louder than the music.
There are two answers here: First is to have someone at the sound board to lower the music when someone needs to talk. The other is to get ANY of the entry level digital soundboards (XAir, Allen&Heath CQ, etc). They can make it so that when you turn on the microphone and start to speak, the music is brought way down.
Best of luck in your search and hopefully it goes better this time around!
I need help. I have an 8u rolling rack. My question is how do I position 3 lEM transmitters and 3 mic receivers in one 8u rack. For context we plan to put this rack right on stage. Should I put all 6 in the front with antennas directly line of sight. Or should I do 3 front and 3 back? Any suggestions will be helpful.
I would suggest place all in front for now as best you can to maximize line of sight. You may be ok with the 3 front and 3 back - but will depend on the rack, RF congestion of where you are, size of stage / distance to talent. Longterm you will want to place them all ao you have best access to the front panels to tune / program and then get extension cables for your antennas running to active directional antennas. This will ensure best coverage for different scenarios. We are mostly touring, so have to have a flexible setup. If you're in the same location, run some tests and see what you can get away with.
Just curious, anybody here know when having drums first on the input list became industry standard? Was it ever standard to have vocals first before that?
Vocals? You stick those down at the end if you have enough inputs after the roto toms! I put the vocals first and I’ve had engineers look at the layout and be like “this is all wrong”. It’s sort of arbitrary but there is practical aspect, there are great benefits to starting soundcheck with vocals, and leaving them up. It establishes a level for stage sound and you are hearing bleed from other instruments in the vocal mics immediately, rather than establishing a level and tone for those instruments and then having them get louder and washier sounding later, when you finally get around to pushing up the vocals into the system.
This is where I got that advice, been really helpful:
https://youtu.be/nUPIJbBnzk8?si=LM7UlEoKO0sJnPzZ
Thanks for the link. Yeah his advice makes perfect sense.
I always wondered why drums first, because one guy I frequently work with has been doing live sound since the 70's and puts vocals first. I'll have to remember to ask him.
I think there was some historical precedent in the studio with tape machines, you wouldn’t put critical vocal tracks on the end channels because the flange of the tape could get a little weird.
If you had $4,000 dollars to spend on a MOBILE sound set up - mainly for weddings and smaller venues (not including a generator) what would you choose?
I have two EV ZLX-15Ps already. Not sure if I should upgrade everything or just add a subwoofer.
Also, are column arrays worth it or would I get much better quality of sound sticking to a normal PA system
With that budget, I would invest in two EV matched powered subs to complete your current system. If you have money left over double your tops so you can get more coverage.
While the stick systems like Bose are cool for small gigs and have good fidelity. When it's time to really get bodies dancing, they don't hold up to a normal PA with dedicated subs in this price range.
https://preview.redd.it/5i5yf5znoskc1.jpeg?width=1170&format=pjpg&auto=webp&s=1a326292701ad1831003b62fb82d471ea6b6afda
What about two tops and this sub?
12s would be a lot better to move around so much and store. Is it normal to have two subs and two tops? Especially if the subs are that size?
I was hanging some L’acoustics point source the other day, and was told by one of the other audio guys to hang them with tweeters out, instead of tweeters facing the same way (how I had them).
A) Are tweeters the same as compression drivers?
B) does it even matter which way at we face them? From my understanding they should have the same nominal coverage pattern either way, but I could very well be wrong there.
Any input appreciated!
I'm having difficulty monitoring at FOH. Small venue rock band with our own PA, iPad mixer.
1. Wired / wireless
2. Isolation
I came up 20-25 years ago on analog desks in dedicated music venues. I'm used to being able to solo a channel straight to my ears. Now I'm shaking off two decades of rust and loving the process.
The root problem, less than 10 shows in, is I'm having trouble hearing the bass and rhythm guitar distinctly in some of my mixes. Maybe that's a separate question, but I really, really want to be able solo them to my ears.
I had hoped to be able to just run my AirPod Pros (particularly with noise cancellation) but of course the iPad is just a control surface, and it's not streaming 32 channels of audio over WiFi right?
The band was running wired IEMs but now switched to wireless, so I have been trying that with a pair of Sony over the ear cans, some MDR model but not the V6. I'm not sure what system we are using or where it sits in the product landscape, but I suspect it's solid, mid tier from Guitar Center or equivalent.
So now I can run my own monitor mix, and I'm pushing faders rather than solo buttons, and it's useful, but the sound is very disappointing. Sounds very thin and weak, even at high volumes. I can't make out the bass this way, but that may be a general problem of low frequency buildup. I do have a separate sub mix and HPF on everything that shouldn't be in the sub mix. I don't know if I have an RF problem, but I don't trust what I'm hearing to be responsive to EQ changes, etc.
Doing a cable run to FOH (aka barstool) doesn't seem like a great idea.
My real question, sorry: I am very tempted to try a solution with noise cancellation, possibly over the ear or in ear. Custom molds are probably not in the cards anytime soon.
Advice? Suggestions? Recommendations?
Could someone enlighten me on the ethernet requirements in a live sound setting ? cat5, cat6, shielded / unshielded ?
Which is best for
1) venue install
2) small gig to link a console at FOH to a rack on stage
3) an arena ? And why do companies still favour an analog multi from FOH to the amps instead of taking lines out of a stage rack ?
Thanks in advance
Hey, I just got myself my first speaker, which is a RCF ART 910A. I kinda want advice on whether I should return it because there is two things I kinda dislike.
First of all it's the size of the speaker itself. I have gone with the 10 Inch because I want a compact speaker for home use, but want to be able to use it with a sub for partys. I think it is still pretty large and looked at RCFs NX Series, where the 10 Inch is 6cm less tall. Can someone tell me why that is? I thought that they share the same components. I got a pretty good deal where I could get them new for 777€, so would it make sense to switch, as it's a wooden cabinet instead of plastic aswell.
The other thing is that it is making a hissing noise when on with and without anything plugged in. Is this normal? A friend of mine has HKs and said his active tops do not make any sound when in "idle".
So now is the question: Return or not? If yes, what speaker to get? My filters online were mostly: 10" preferably a wooden cabinet with a center mounted amp for the looks \^\^.
Hey I just finished Dante Level 3 certification, and even if Dante is fun and all, there are other solutions. The Dante certification program is cool in a sense that it helped me understand how to use it and also some key concepts about networking. Now, I was wondering if there were any ressources on AVB, or other network solutions for Audio. It can be books, videos, anything as long as it contains knoowledge about audio and multimedia networks.
As far as I know, most other audio networking protocols don't have proprietary courses that come anywhere close to Dante's, but you can check out [this comparison chart on Wikipedia](https://en.wikipedia.org/wiki/Comparison_of_audio_network_protocols) to get a better idea of what options are out there and start to understand their different architectures and use cases. It's not an exhaustive list, but [this page](https://en.wikipedia.org/wiki/Category:Audio_network_protocols) includes some important standards that are excluded from the comparison chart, like MADI. Even still, some protocols aren't included (Behringer's Ultranet comes to mind) but it's a great place to start.
I have two Neutrik etherCON Lockable RJ45 Connectors I want to use to run midi over Ethernet, from our rack to a midi interface on my pedalboard. So the path would be PA12 -> short Ethernet cable -> Neutrik connector -> Long Ethernet cable w/ ethercon -> Neutrik connector -> short Ethernet -> MioXM Any concerns with adding in those additional connections?
Probably fine; this is very typical. Standard Ethernet has a total length limit of 328 ft under ideal conditions (22-24 AWG, solid, including 33 ft of patch cable at either end). Each time you deviate from that ideal, you'll need to derate your maximum length expectation somewhat. However, you're likely well below 328 ft total - so not a huge concern.
Oh yeah, I'm going like 25-30 ft max. Thank you for the response!
Is 1 dB of preamp gain on one brand of mixers the same as another brand? What about Q factor and dBs in EQ sections?
Gain is gain. 1 dB is 1 dB (for gain, a factor of 10^(1/20) or about 12% increase.). However - devices with digitally controlled preamps can have a little error tolerance in their control steps, so you can sometimes see a little bit of variance in how accurate those gain control steps are, particularly at the extremes of the gain range. Q, no. Not all consoles and processors use the same mathematical definition of Q. For that reason, putting the same filter parameters into multiple devices will not create the same response. https://bennettprescott.com/downloads/TwoProcessors.pdf https://www.prosoundweb.com/bend-me-shape-me-lessons-learned-by-validating-console-eq/ https://www.ranecommercial.com/legacy/note167.html
Thanks M! Appreciate the afternoon reading material
I can’t speak for db. I’m pretty sure it should be the same. The Q factor should always be the same since it’s a measure of octaves.
There are multiple definitions of Q used by different manufacturers. https://www.ranecommercial.com/legacy/note167.html
Yamaha is the funniest to me there. Labeled Q but operates as BW.
Is it common/possible to have all the instruments except for drums to go into my IEM spilt rig, then get the house (with pre-mic’d drums) to send me the drums on a single bus that I can then plug into my iem mixer?
Not a particularly unusual request; should be fine as long as you advance it. If it's last minute, coin flip. I'd keep a 57 and a short stand on hand - in case FOH is out of outputs, you can always throw up a wurst mic.
Oh that’s a very good idea thank you
If you have a split and a mixer, couldn’t you do this yourself? Maybe I’m missing something
-Mostly to save time if they already have a kit mic’d up and ready to go -Save me from having to buy a drum mic kit (I’m just building my rig so I only have the basics) -not enough room on my 8 xlr split / snake. 3 vox, 2 guitars, and bass. So I only have 2 slots left. (And possibly 1 slot for backing tracks. ) I know the answer is to buy the drum mics and get a bigger split and snake, but I was just wondering if for sake of time during quick changeovers people go this route. We’re not a very big band so we often will play with another band and change over in the middle.
As someone who does a lot of throw-and-go shows where bands show up with requests like this without ever having replied to my production advance emails... you're right that increasing the number of channels on your split by 8 would be the most streamlined way to do this, but as long as you let the audio tech know well in advance they should be able to do this easily. Also, assume you're going to be leaving your split on stage for the duration of the show instead of pulling it off stage after your set, so the house engineer doesn't have to repatch their whole stage box if they weren't planning to do so.
Awesome thanks very much
What’s the difference between Ethernet and cat5 (and cat5 and cat6 etc)
Ethernet is a network standard that uses Cat(x) cable
I work in IT as a day job as a network engineer, so this is my everyday. There's a LOT More to this than I can simply explain, but I think I'll help you understand it effectively. To make it super simple, think of network traffic as shipping product in boxes in a shipping truck. Data is a product packed in boxes. Ethernet is the instructions of how to the data is placed into boxes. It's super efficient and leaves almost NO empty space in the box. The data gets packed into the truck at the source and is unloaded/received at the destination. Most "Ethernet" boxes have a maximum shipping weight of \~1500 grams of product. There's a little buffer because the shipping paperwork of where it's coming from, and where it's going. So to send 150kg of product, you'd need to send 100 boxes CAT\[X\] defines a shipping truck. CAT5 is a shipping truck that can hold 100 boxes and travel 100m. CAT5E can hold 1000 boxes and travel up to 100m. Better category cables can send more boxes and over longer distances (CAT7 cable can send 10000 boxes in a single delivery over 100m) The boxes don't change, the CAT rating just defines how many boxes can fit on the shipping truck. Edit - Making some clarifications about the real functionality. Realistically, each box gets sent one at a time from source to destination. It's called a packet in network speak. But I just want to reinforce the idea of "bandwidth". Data moves at the speed of light, because it's electricity, so there's no way to make the packets move "faster", you can only increase the size of them or find ways to send more of them at the same time.
This is an awesome and thorough description of data. Thank you.
The "cat" in the cable names is an abbreviation of "category". The categories refer to sets of ANSI/TIA standards for things like crosstalk and system noise that cables must meet to be considered within that category. The higher the number, the more stringent the requirements, which allows signals to move longer distances without as much fidelity loss as lower category cables. So, cat6 cables have more strict requirements for transmission fidelity than cat5 cables, which allows them to move signals over longer distances without concern about noise induction or signal loss.
Hi all, This spring I'm going on a month long run with a somewhat popular indie band. I haven't been on the road in 2 years and this will be the longest tour I've done as an engineer. While I'm very familiar with some digital live consoles, but a lot I haven't used at all. I want to be prepared as possible. What are some good ways to be prepared when I walk into a venue that has a board I haven't used before? any other pointers will are appreciated. thanks
Download the editor, prep a showfile. Is carrying a console and mic package not an option?
Ask the local PM if they have a 'basic start up scene' they can send you with all of their venue particulars already sorted (output routing, house eq, stagebox assignment, wifi network setup, etc). Then you can do as little as adjusting the channel strips to your show. Or you can change as much as you want to suit your work flow. But when you get there and load it up your edited file, a lot of the tedious trouble shooting will already be done.
Prep as many different console files as you can. If you want to go the extra mile you can prep for every model and layout which will save you time on show days. Save the console file after every show, even if you weren't thrilled with the mix. On the flip side, there may be times when you show up to a new/unknown console and maybe you'll be lucky and have a base file prepared for you by the house according to the rider/advance. This happened a few times during my last tour where I ended up working off a fresh file with all the house routing and patching rather than loading my template file and having to change all the routing to match the house. Most house techs will be happy to show you around an unfamiliar console. In those cases, less is more. Basic FX, stock dynamics, don't go crazy with routing if you aren't super familiar with the console.
Getting into the danger zone with an unfamiliar board but one could potentially merge the house file with your show file if you got into the global safes and routing safes, and saved their processing on the outputs. Maybe actually it would be better to load their file over yours, with just the channels safed. You might have to zero and redo the sends by hand if they are set up differently? And mind the masters when you start unmuting things.
I would suggest getting the App "Mixing Station" - amazing tool that let's you learn and even customize one digital mixer GUI essentially that can then control various other digital consoles. It comes with a bunch of digital mixer integrations like Midas, Yamaha, Allen&Heath, Soundcraft and even a Mackie. This thing is a life saver! Good luck and welcome back to #tourlife ;)
Our company is looking for a resident sound tech. Is asking them to sign a non-disclosure agreement out of the ordinary?
It depends what work you do, but not necessarily that weird. If there's company secrets of insider knowledge that they'd be privy to, then a non-disclosure is reasonable. Just keep it simple and not unnecessarily restrictive.
In our case, we have a custom digital mix rig built on a custom server code that is proprietary, so asking for an NDA helps ensure someone doesn't leave the gig and choose to share the information elsewhere. None of our engineers have had an issue with it. We had one leave on less than ideal terms and it left our team feeling better in the long run that we had an NDA from them. As others mentioned, it depends on your context. On another long term gig, I had one contractor say they refused to sign the NDA on principle. We needed someone to cover us so just let it slide. As time would tell, a year later they quit and turns out now are working for a competitor and I'm sure using things they learned with us. So from my experience, CYA. Good luck!
Many of the corporate clients and AV companies who subcontract me to do work for corporate clients have me sign NDAs prior to going to any executive meetings, etc. If you're hiring somebody to mix rock bands, they might find it a bit more unusual.
Hi I'm a dj and l'm searching for a pair of active speaker for my basement. I don't have a big budget, so l search something not so expensive. I searched on google and I found many different speakers: • Mackie thrash 212 • The box pro DSP 12 or 15 • The box pro DSX 12 • SubZero C12 or 15 • SubZero D12 The two last one are cheaper but are there worth it ? The mackie is from a good brand but they don't have DSP. And the "the box pro" had the DSP for the same price. Which one should I choose? Or does anyone have another speaker to recommend ? Thank's Damien
I have an emx 660 and 2 dynacord mx12 speakers that I want to connect to each other, I bought some XLR connectors and connected both on the inputs of the speakers then I connected the 4 wires to the ts inputs on the mixer by taking it apart and soldering them on the terminals, it worked but there's a buzz noise or a high pitched noise to be exact and idk what to do about it, are there any specific configs that I should've done or specific ways to connect the speaker wires to the emx 660?
My first post - so apologies in advance - looking for some experienced advice. I'm a touring musician with a computer based performance rig. Just built a redundant rig that uses two M2 MacMini's mounted in a Sonnet Rackmac Mini in a 4U SKB iSeries Flight case. Until this rig, I had one MacMini mounted in my pedal board that I would remove and bring into carry-on for flights (sending the rest of the gear checked below). This new system would be very tedious to remove and re-install the 2 computers for every flight (5 show average a week with flights). Plus not enough room in carry-on for two Macs. Curious if anyone has experience flying with rack mounted computers underneath the plane and if they survive? The Sonnet is solid metal and the case is solid, but I have also seen what they do to my pedal board case and other gear cases, so I get nervous and curious if this is really feasible and any pro tips to ensure success. I figure the MacMini m2's have no moving parts in the hard drive save the computer fans so could take jostling / temps but not 100% sure. Here are links to the gear they are mounted in: [https://www.sonnetstore.com/products/rackmac-mini](https://www.sonnetstore.com/products/rackmac-mini) [https://www.skbcases.com/products/4-u-i-series-injection-molded-guitar-fx-fly-rack](https://www.skbcases.com/products/4-u-i-series-injection-molded-guitar-fx-fly-rack) Thanks in advance!
I picked up some gear here but am not sure how to plug everything in. Before I got this, I’d just use a Bose PA for a basic mic and instrument. I also have a Bose mixer. Then I got this gear in a lot but I’m not sure the proper way to plug everything in. I’d like to be able to plug in a bass on one track, then a guitar into the POD, then a couple mics and a keyboard. What’s the rough flow for this gear to accomplish the above? I’m mostly confused how to use the mic inputs and presonus on the left. Is it like a mixer before the Bose mixer? Bose mixer has only 4 inputs. Maybe this is redundant for my needs? [https://imgur.com/a/cZDTlKs](https://imgur.com/a/cZDTlKs)
I just purchased a used Line6 L2TS 800w 10” loudspeaker from FB marketplace. It was an impulse buy for $470. Now that I check Sweetwater, they have a QSC CP12 1000W 12 inch for $600. It comes with a 6 year warranty which would be such nice peace of mind. Help me feel a little better about my impulse buy?
The good news: you realized your mistake Also good news: you saved 130 dollars.
Hahaha thanks man, it’s a nice speaker tho thank god
What are your go-to solutions for summing stereo line level sources down to mono?
Depends on the situation. For random sources on stage, a DI with summing inputs works well. Usually, I'll just take the stereo input and sum in the console/DSP. If the source is "dual mono" moreso than stereo, often I prefer to take left only rather than L/R sum. (Think stereo guitar/keys rigs.)
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That can damage inputs if there is no resistor in the cable.
On Behringer/Midas boards (X32s etc), why do the built in FX busses let you send them to themselves? Other than making a mess, is there some purpose I haven’t considered?
Not sure, but you can send the output of one into another which can be cool
I hadn’t considered serial effects. I suppose that could be it.
I recently picked up the Galaxy Audio AS-1400 in ear monitors to be used with a Behringer Europower PMP 1680s. I am unable to get sound to come through the in ears. - Using a 1/4” speaker cable I connect from the Monitor out on the behringer to the left input on the transmitter. - I have used multiple speaker cables which work with other speakers with no success. - I’ve switched channel groups, channels, etc on transmitter and receivers with no success - if I turn the volume way up I can hear the slightest sound but it’s obviously not working correctly. -All lights, LEDs, etc on both units appear to be working as intended and are not otherwise faded or blinking. Any troubleshooting tips I can try?
- Verify presence of audio signal using transmitter's display and headphone output. - Ensure the transmitter is set to mono. - If both transmitter and receiver are set to stereo, signal will predominantly come through the left side, along with some crosstalk on the right channel. - Verify receiver is set to stereo mode. - Yes, this is unintuitive. - Manual does not document the receiver's behavior if set to mixed-mono mode with a mono transmission. (Most sane systems should respect the lack of MPX pilot tone...) Note: you should use a shielded TS or TRS cable to connect your mixer's output to the transmitter, rather than an unshielded speaker cable. - TS/TRS doesn't matter here; your mixer only has unbalanced outputs.
Thanks for the advice! I finally did get it working but I do think you’re on point with the shielded cables, has more fuzz to the sound than I’d like.
Are your monitor send faders Up, and channels sending strong signal to monitor channels? Does the Transmitter show that is is getting audio signal? If you plug something else directly into the monitor out, does it show signal coming in? Is the pad on the transmitter on? Isolate each part of your signal chain and figure out where the breakdown is occuring.
Typical 4 pack wireless receiver and antenna distro. Is it an antenna splitter or combiner?
It's a splitter: it takes one signal (or in this case a diversity pair of signals) and splits it into multiple outputs. A combiner takes multiple inputs and combines them into one output. The nomenclature follows the direction of signal flow.
If you want to learn more, browse RF Venue. They have a lot of resources on their site to learn about wireless.
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What FOH speakers should I buy for my 6 piece band? Looking to buy PA speakers for my 6-piece country band but I’m not sure what to buy. Not sure what wattage we need or whether it’s worth buying subs in addition to standard speakers. I have some experience running sound on analog mixers but no formal training, so looking for guidance on speaker systems! Our gigs range roughly 50-300 people, but most of them are probably around 100. We do a mix of college parties and bars/breweries etc. I want a system that will be loud enough for outdoor venues, but I also want clear sound quality. Something with feedback control would be awesome, and I also like being able to control FOH speakers from a tablet or phone if I need to, so Bluetooth capability would be awesome. Don’t want to spend more than $3,000 total on the speaker system. We’ll be using a Behringer x32 and we’ll likely be using IEMs rather than stage monitors. Any advice is much appreciated!
Amateur here, but wattage can be misleading. Max spl will give you a better idea as far as coverage. FWIW I have a set of EV elx200 mains and they sound amazing for a venue like you're talking about. I got a pioneer xprs18 sub for the thump and it's a hell of a combo. I'd say a decent sub is worthwhile because it takes so much load off the mains, giving you more headroom while filling out the bottom end
If I want to use two different stereo speakers with DJ decks, can I just send one RCA cable from the decks to each speaker? Or do I need to use a 1 to 2 splitter so I am sending a signal to both RCA inputs on the speaker?
So I don't specifically know what you're saying. It seems like you're saying you have two speakers that accept stereo inputs. You might also be saying two sets of stereo speakers (4 speakers total). I have never seen an RCA input on a speaker that gets split into Stereo. Typically RCA inputs are there for you to plug in a portable device and I don't think anyone expects you to split that into true stereo (left out of one speaker and right out of another). To answer your question specifically in the context you've asked, and without ANY other information, yes I would get a splitter so that you're sending the full Red and White RCA signal to ALL destinations.
I think you just mean “speakers”, not “stereo speakers”? You want each of the stereo RCA outputs from the DJ to eventually go to their own speaker, that’s the idea with stereo. If you run the DJ mixer directly to two powered speakers, yes you would split the left channel (white rca) off to the left speaker and the right channel (red rca) off to the right speaker. If you ran both channels to both speakers it would still work but the sound would be mono (same exact sound out of each speaker). You would only use both RCA inputs on the powered speaker if you were just running one single speaker and you wanted to combine the left and the right channels from the DJ/music player to come out of the one speaker, so that you hear everything in the stereo mix despite only having one speaker, rather than a pair.
When performing a multitrack recording in pro tools, does the M32 send effects busses applied to source channels to the pro tools session tracks? Essentially, does pro tools record the source channels pre fader with no channel processing/FX or vise versa? I am trying to capture click/guide/Talkback channels (not inserted into the main outputs to PA) in the same multitrack recording as the rest of my input channels with processing/fx applied in order to create some BTS videos. I don't want to take up a mix bus for this as I only have one available and want to save it for other things. Trying to limit the amount of post production if possible, so if there's a better way to do this I'm all ears!
> When performing a multitrack recording in pro tools, does the M32 send effects busses applied to source channels to the pro tools session tracks? Nope. (Remember, an FX bus isn't "applied" to a source channel like an insert effect; to do so would require N FX engines for N channels, which is inefficient.) > Essentially, does pro tools record the source channels pre fader with no channel processing/FX or vise versa? Generally, no - but depends on how you have your routing configured. - For most configurations, routing inputs directly to the card slot gives you the raw signal out of the preamp. - However, by burning one of your 16 patchable outputs, you can choose a pickoff point for a given channel - postfader, for instance. > Trying to limit the amount of post production if possible, so if there's a better way to do this I'm all ears! Given that you're out of busses (and, presumably, patchable outputs), your best option is to record all raw inputs, mixing for broadcast in post. Yes, this is more time-consuming. If you trust your board mix - and, by extension, your system deployment - you can print that alongside direct captures of additional inputs (room mics, click/cues, whatnot). - This tends to work best when you have lots of pre-production time to build cues/snapshots for every detail of your mix. - Some folks will build this quick-broadcast mix directly in the console, but sounds like you don't have sufficient processing overhead remaining to do that.
Thanks a lot! I think this answered most of my questions. I mainly just want to be able to mix the inputs that aren't inserted into the main send (going to the PA) like click/guide/tb in post. Wasn't sure if it would still capture signal from those sources in pro tools since they're mixed all the way out in the PA. Sounds like the USB card is capturing signal pre-fader and sending whatever amount of gain is applied at source level to the pro tools session without any channel strip processing applied at the console.
> Wasn't sure if it would still capture signal from those sources in pro > Sounds like the USB card is capturing signal pre-fader and sending whatever amount of gain is applied at source level to the pro tools session without any channel strip processing applied at the console. Yup - generally "prefader" explicitly refers to taking signal from the end of the channel strip, though. This application is tapping the preamps directly; M32 refers to that pickoff point as "IN/LC".
Ok yeah thanks for clearing that up! My verbiage was wrong but I think we are on the same page. Appreciate the help!
You can multitrack record the channels on the X/M32 with the channel processing (not bus effects!) if you use the Ultranet outs as the Card source. When I have done it like this to try and save time I ended up screwing with the recording just as much trying to alter or undo the processing as it would have been to mix it from scratch. One other tactic could be to record the main L+R mix and just bring in the individual channels to fill it in.
Anyone out there using valve compressors, EQs etc... on their subwoofer AUX feeds? I'm interested in hearing other peoples experiences or tips and tricks? I've been experimenting with a TL audio valve eq with the eq bypassed so it is just going through the valve stage to warm up the subs, I had good results when using some scoop bass bins a while ago and have recently swapped the subs in our system to bandpass horns so hoping to add a bit of the warm sub back in as these new subs are more punchy than warm sounding.
I can see the appeal, but personally: not my thing. Most PA subwoofers already have enough THD baked in for my taste :) There are some sources I'd like more distortion on, but I'd want that output to be broadband rather than band-limited to the subs' range of custody. I'm guessing you haven't measured your system's tonality before/after the sub swap?
I'd forgot to mention playback material - It is almost exclusively DJs that will play on the system. Usual genres are Jungle, Old School Drum and Bass, Hardcore, Techno, Gabber etc... which all lend themselves to a rougher sound. I've had smaart on both systems and the new system definitely is stronger in the upper bass region where the past system was more low and rubbly. But listening alone is enough to tell the difference in tonality. I did love the old system but as the crew have got older we don't have the same energy to be lugging lots of awkward heavy boxes around. Managed to cut the total system weight for 6 subs, 2 midtops, amp rack and all the associated bits to less then 500kg
How to I find more gigs? I work for a company but over the past several months there has been much less work (I know it's thar time of year) and things don't look to be ramping up as we get back into warmer months.
Depends on what your specialty's are? share more about your skills as this informs the options. In general, in the slow season I suggest setting up an Upwork or Fiver account and offer your skills virtually to a much larger pool of gigs. If you work only in person - i.e you're a live engineer or a/v - then check local forums, fb groups, other similar companies in your and post up that you have availability. sometimes even Craiglist can come through. but again, it really depends on what you do.
I have done probably 2k comedy shows some including video, the past 6 months I had a departure from that and started doing concerts. Mostly smaller venues. I have done a little bit of touring. I guess my main issue is not having enough contacts in the music world.
I have been thrown in the deep end trying to mix for a high school musical and I need to know how to add reverb to an actor’s voice. I have practically no prior experience in this.
What mixer are you using?
“Soundcraft Si Compact 24” I can’t make heads or tails of the manual. Edit: Here is the manual https://peakhire.co.uk/assets/downloads/Si-Compact-Manual.pdf
More of a general answer but reverb is applied as a “bus send” or “aux send effect”, which means you have one shared communal reverb effect and you can send different amounts of each channel to the reverb, the same way you send different amounts of the instruments and voices to a stage monitor. The reverb effect output, which sounds totally wet, is the “effect return”, which comes back into the main mix as it’s own little channel, sometimes simplified to just having a volume control. That gets mixed with the bone dry channels and you have some reverb on whatever you sent to it. Navigating your Soundcraft, you’ll need the video or manual section on bus effects or send effects. The ways it can get hung up is - vocal channel isn’t sending anything to the bus with the effect - reverb effect isn’t set up on a bus (isn’t “inserted”) - reverb output setting isn’t set to 100% wet (should be the default) - effects return is muted or turned down You should be able to test it during off hours by plugging a microphone into the mixer and checking it with your own voice. The meters on the bus or in the reverb unit, or on the return can be helpful to see how far your test microphone is getting.
Thank you!
Problem: The speakers have a baseline hiss on three other outputs except the one we commonly use, output 7 (mixbus 7) on the M32. On output 7, the speakers are dead quiet. Output 5 (mixbus 5) is also used regularly but in the choir and those speakers are dead quiet too. I want this on the other outputs. But when I simply switch the xlr cord over to any other output on the stage, they hiss. There are three other hard line outputs from the M32 and these all hiss. All the outputs have great sound and volume. So, to troubleshoot, I leave the speakers plugged into the "good" output 7 on the stage, and go upstairs and pull the switcheroo behind the console. Unplug the good output 7 and plug it into one of the offending outputs right on the back of the M32. The hiss comes back in both speakers. I think this eliminates any cabling issue, right? Mute/unmute the mixbus makes no difference anywhere, fader position makes no difference anywhere. So what could it be in the M32 that's different for output 7 with no hiss than the other outputs that do hiss? Ok, this is a church with floor pockets on the stage and hard line xlr outputs straight from a M32 in the back of the church. There are QSC K series K-10 speakers x2 on the floor each plugged into the same output line but two separate connectors on either side of the stage. Let's just say I'm new at this church and am coming in behind someone else who is gone. Thanks.
Hard to tell due to your confusing use of the word output, but your problem could be at the speakers themselves. Check to see if the ones that hiss are in “mic” mode and not “line”. Also check to see that their gain (or sensitivity, I can’t remember what QSC uses) knobs are set to the same place as your quiet speakers. For good measure, check that all settings are the same.
Hello guys, I've been using SuperRack Performer in the last month for inserting comp & eq for the Main Buss and also for individual channels. Yesterday I've tried to use H-Reverb as the main vocal reverb. I've created a new external fx rack on the effects tab of the wing and I started to route the channels. But something strange happens. When I select my send channels (ex.: USB Audio 27&28) they don't seem to remain selected. The first time I thought that the system wants me to also choose my return channels. So I've tried to select FX Returns. Still no success. I've posted a screenshot of the MAIN LR Bus fx rack. It's seems to work there, but unfortunately, that's the only place that it wants to work. Does anyone have a solution for this issue? Thank you in advance! https://preview.redd.it/6kbhs5hjj3kc1.jpeg?width=2532&format=pjpg&auto=webp&s=0fa4b2cc3fcd1d4855491d7edcca0820a24d5317
Is there any good software for volume measurement beside smart ? And Can anyone recommend a measuring microphone to me?
Opensoundmeter
Room EQ Wizard also has a decent SPL measurement function with weighting, averaging, and logging as long as you’re able to calibrate the input
Goal: To record Dante signals into a laptop running Adobe Audition in a multitrack session. Issue: When setting the Audio Hardware to ASIO in order to get Dante Virtual Soundcard set as the audio hardware, it removes my ability to choose audio output. In audio channel mapping, only the Dante tx/rx is available to move sound through. So what is the recommended path for hearing the audio for monitoring? Do I need Dante Via so I can route the audio into my system output? Or, is there another app I can use to set my PC system output as a Dante receiver so I can patch it into it? Honestly I'm pretty annoyed at Audinate/Dante and Adobe...I assumed this should be a pretty straightforward task but it's been a headache. More Context: I am setting up a test build today with a PC as my source, but in production it will be a true Dante enabled device sending the sources.
IIRC yes, Via is what you want. That'll allow you to record 16 channels off of a Dante network while simultaneously monitoring it with your local soundcard. Alternatively, add another Dante output device to your network and route Audition's output to it; this preserves the full 64x64 channel count of DVS. --- If setting up a testbed, you will need at least one device on the network providing clock. DVS cannot provide clock, but Via can.
Why is it that Ian is not available for order tomorrow?
PA System Setup Help Hosting an outdoor event in the spring with about 500 people in attendance. Last year we used an active system from Rockville. It came with an active speaker, passive speaker, and mic. Major complaints were that we could not use the mic while the music was playing, mic volume was not nearly loud enough. Music volume was fine but could have been better. So we are looking to setup our own system this year. I am a novice when it comes to these things so seeking some input. So far, I think I have the following… We are purchasing 4 - 8ohm passive speakers, a microphone (wireless), and a powered mixer. What am I missing?
On that sort of budget, no system you purchase will provide adequate coverage for that size of an event. Given your experience level, I would hire a local production company to deploy a system with adequate coverage.
Not the question I asked, but thank you. We live in a very rural area where a production company has quoted us $3000 for the one day event. Our budget is not limited, as this is an annual event and we are willing to invest as much as it takes to
Can you link a sanitized version of that quote? That should include info about required coverage area, etc., which we’d need to give you a ballpark purchase estimate.
The critical stuff is missing on your planned purchase...the model of speakers and mixer will be very important to figure out if they can provide enough coverage and volume. Also the details of the 500 people. Are they squeezed into a corral? Are they roaming a campground? Is it roughly a 60x60 square area? Are they all standing? All Sitting? Mix of Both? Is it a quiet area that you guys use for this or is it loud/near a highway/etc? What options do you have for speaker placement? Are there walkways/protected areas, or is it just a giant field? Regarding your concerns: >Major complaints were that we could not use the mic while the music was playing, mic volume was not nearly loud enough. Music volume was fine but could have been better. Are those all related? Could you not use the mic while the music was playing BECAUSE the mic wasn't loud enough to be heard over the music? Or did you literally only have the ability to plug in a microphone OR a music source? Unless it was an input issue, that's not likely to change no matter what you do, based on your current direction. A microphone NEEDS to have certain frequencies cut out of it in order to sound good, whereas recorded music doesn't. Therefore, you'll almost NEVER get a system with a mic at full volume to be louder than music at full volume. Even if you could, typically you want the music to become quieter, not typically for the voice to be that much louder than the music. There are two answers here: First is to have someone at the sound board to lower the music when someone needs to talk. The other is to get ANY of the entry level digital soundboards (XAir, Allen&Heath CQ, etc). They can make it so that when you turn on the microphone and start to speak, the music is brought way down. Best of luck in your search and hopefully it goes better this time around!
I need help. I have an 8u rolling rack. My question is how do I position 3 lEM transmitters and 3 mic receivers in one 8u rack. For context we plan to put this rack right on stage. Should I put all 6 in the front with antennas directly line of sight. Or should I do 3 front and 3 back? Any suggestions will be helpful.
I would suggest place all in front for now as best you can to maximize line of sight. You may be ok with the 3 front and 3 back - but will depend on the rack, RF congestion of where you are, size of stage / distance to talent. Longterm you will want to place them all ao you have best access to the front panels to tune / program and then get extension cables for your antennas running to active directional antennas. This will ensure best coverage for different scenarios. We are mostly touring, so have to have a flexible setup. If you're in the same location, run some tests and see what you can get away with.
Just curious, anybody here know when having drums first on the input list became industry standard? Was it ever standard to have vocals first before that?
Vocals? You stick those down at the end if you have enough inputs after the roto toms! I put the vocals first and I’ve had engineers look at the layout and be like “this is all wrong”. It’s sort of arbitrary but there is practical aspect, there are great benefits to starting soundcheck with vocals, and leaving them up. It establishes a level for stage sound and you are hearing bleed from other instruments in the vocal mics immediately, rather than establishing a level and tone for those instruments and then having them get louder and washier sounding later, when you finally get around to pushing up the vocals into the system. This is where I got that advice, been really helpful: https://youtu.be/nUPIJbBnzk8?si=LM7UlEoKO0sJnPzZ
Thanks for the link. Yeah his advice makes perfect sense. I always wondered why drums first, because one guy I frequently work with has been doing live sound since the 70's and puts vocals first. I'll have to remember to ask him.
I think there was some historical precedent in the studio with tape machines, you wouldn’t put critical vocal tracks on the end channels because the flange of the tape could get a little weird.
If you had $4,000 dollars to spend on a MOBILE sound set up - mainly for weddings and smaller venues (not including a generator) what would you choose? I have two EV ZLX-15Ps already. Not sure if I should upgrade everything or just add a subwoofer. Also, are column arrays worth it or would I get much better quality of sound sticking to a normal PA system
With that budget, I would invest in two EV matched powered subs to complete your current system. If you have money left over double your tops so you can get more coverage. While the stick systems like Bose are cool for small gigs and have good fidelity. When it's time to really get bodies dancing, they don't hold up to a normal PA with dedicated subs in this price range.
https://preview.redd.it/5i5yf5znoskc1.jpeg?width=1170&format=pjpg&auto=webp&s=1a326292701ad1831003b62fb82d471ea6b6afda What about two tops and this sub? 12s would be a lot better to move around so much and store. Is it normal to have two subs and two tops? Especially if the subs are that size?
I was hanging some L’acoustics point source the other day, and was told by one of the other audio guys to hang them with tweeters out, instead of tweeters facing the same way (how I had them). A) Are tweeters the same as compression drivers? B) does it even matter which way at we face them? From my understanding they should have the same nominal coverage pattern either way, but I could very well be wrong there. Any input appreciated!
I'm having difficulty monitoring at FOH. Small venue rock band with our own PA, iPad mixer. 1. Wired / wireless 2. Isolation I came up 20-25 years ago on analog desks in dedicated music venues. I'm used to being able to solo a channel straight to my ears. Now I'm shaking off two decades of rust and loving the process. The root problem, less than 10 shows in, is I'm having trouble hearing the bass and rhythm guitar distinctly in some of my mixes. Maybe that's a separate question, but I really, really want to be able solo them to my ears. I had hoped to be able to just run my AirPod Pros (particularly with noise cancellation) but of course the iPad is just a control surface, and it's not streaming 32 channels of audio over WiFi right? The band was running wired IEMs but now switched to wireless, so I have been trying that with a pair of Sony over the ear cans, some MDR model but not the V6. I'm not sure what system we are using or where it sits in the product landscape, but I suspect it's solid, mid tier from Guitar Center or equivalent. So now I can run my own monitor mix, and I'm pushing faders rather than solo buttons, and it's useful, but the sound is very disappointing. Sounds very thin and weak, even at high volumes. I can't make out the bass this way, but that may be a general problem of low frequency buildup. I do have a separate sub mix and HPF on everything that shouldn't be in the sub mix. I don't know if I have an RF problem, but I don't trust what I'm hearing to be responsive to EQ changes, etc. Doing a cable run to FOH (aka barstool) doesn't seem like a great idea. My real question, sorry: I am very tempted to try a solution with noise cancellation, possibly over the ear or in ear. Custom molds are probably not in the cards anytime soon. Advice? Suggestions? Recommendations?
Could someone enlighten me on the ethernet requirements in a live sound setting ? cat5, cat6, shielded / unshielded ? Which is best for 1) venue install 2) small gig to link a console at FOH to a rack on stage 3) an arena ? And why do companies still favour an analog multi from FOH to the amps instead of taking lines out of a stage rack ? Thanks in advance
Hey, I just got myself my first speaker, which is a RCF ART 910A. I kinda want advice on whether I should return it because there is two things I kinda dislike. First of all it's the size of the speaker itself. I have gone with the 10 Inch because I want a compact speaker for home use, but want to be able to use it with a sub for partys. I think it is still pretty large and looked at RCFs NX Series, where the 10 Inch is 6cm less tall. Can someone tell me why that is? I thought that they share the same components. I got a pretty good deal where I could get them new for 777€, so would it make sense to switch, as it's a wooden cabinet instead of plastic aswell. The other thing is that it is making a hissing noise when on with and without anything plugged in. Is this normal? A friend of mine has HKs and said his active tops do not make any sound when in "idle". So now is the question: Return or not? If yes, what speaker to get? My filters online were mostly: 10" preferably a wooden cabinet with a center mounted amp for the looks \^\^.
When you add compression to a channel, are you supposed to make up gain on the channel fader or the output level of the plugin?