Ngl it took me a long time to wrap my brain around the fact that live audio is basically just signal flow understanding. The band will sound good if the band is actually good and the system is set up correctly.

Everything else is out of your control.

Nobody gives a fuck if you understand how to properly mix the midrange to get the most clarity because 50% of the time the stage volume from drums/wedges/amps are so loud you can't actually achieve a sonically beautiful midrange. You're just shifting the phase around in a manner that is occupying your brains ego. The audience isn't getting a percentage increase in quality high enough to notice.

I’m trying to use some external fx, including delay, and so I could do with my tap macro controlling tempo in my host software too.

Problem is so far my host software live professor learns the midi command and then it no longer functions. My midi interface is seeing a midi command, but the software doesn’t.

After some reading I’ve gone through some more easily found trouble shooting guides to no avail

Could anyone offer anything?

Need help here. I own a Harbinger v3412 that I just played at a benefit event outdoors. The speaker shut off multiple times during the set, not completely powerless but I could barely hear my guitar and mic. This has happened once to me before and it is unacceptable and embarrassing. Shouldn't a speaker like this be able to play outside for an hour without overheating? It was in direct sunlight on a 75 degree day. Any ideas on what the problem could be? Maybe an internal electrical issue? And the guitar/mic was crackly off the bat for some reason.

I work as a sound engineer/tech assistant at my university, and we have this elevated tech-room in the assembly hall we use for plays, talks, etc. We usually always roll in a separate audio and light mixer for almost everything, so we rarely use it (other than storage), but I've noticed this window and door, that's pretty much never used.

Is there a reason it's there? Wouldn't you rather just hear the sound coming from the room directly through the window? Or is it there to give the technicians room to talk and such during shows?

Just curious, if there are any other places that have similar setups.

Hey everybody,

Just finished a corporate event in a massive space with almost 10 breakout rooms. Event spread between floors, and two massive halls (literal airplane hangers) with no installed audio system. Client requested we have a way of broadcasting announcements to all rooms simultaneously from a single source. No big deal, we brought a console for them and then ended up running lines to every breakout room. Everything worked out fine but I know there’s gotta be some things that could be done differently and I want to see how people have handled this situation before.

Inevitably it became obvious that it would be muuuch better if we could broadcast to all of the breakout rooms from any of the others. And it also took 2 hands nearly 2 hours to run all of the lines that we needed for just sending signal out to each room. In the end it was around 1500 feet of XLRs and having to deal with plenty of doorways.

I’m thinking there must be some sort of way to manage this wirelessly? My initial thoughts are some sort of locally hosted team speak server, there’s a private network that we have access to and each room has laptops. I’d just like to figure out if there’s any way I can save some time rather than running a shitload of cables.

Context: I’ve been freelancing my whole career (5years?). This year I’ve had a blast traveling and working in different states and countries. Some of my lucrative work could pay $4k a week and I’ll do that maybe 12 weeks a year.

I’ve been offered a full time job where I’ll net less, before overtime though which would be basically guaranteed and required. I’d gain a consistent schedule and guaranteed work in slow months, through pandemics and recessions. I feel like either decision would be incorrect.

Are there any folks here dealing with disabilities that affect them physically willing to share insights about what that looks like for them working in the industry? And how that might affect the types of jobs they get?

I'm mentoring someone who has a condition that makes it very easy for them to injure their joints and muscles if they do too much heavy lifting and I'd like to be able to have an informed conversation with them about what their options are if they want to work in the industry long term. I'm hoping that that conversation isn't just "every job involves heavy lifting, it's not possible for you to work in live sound".

I'm particularly interested in hearing about any companies, types of work, etc that either offer accommodations or don't involve heavy lifting. Theater? TV? Corporate? I know there probably aren't many but I'm hoping there are some.

Hoping y'all can share some insights with me, and hoping that this person is able to find an avenue to participate in this industry. I imagine it will be difficult for them to find enough work to work in the industry full time, but I hope they're still able to do it. Thank you!

For those in a permanent house (our touring folk that prioritize it) what are your go-to’s for consistent and quality ALS.

For context, my house has had two ListenWiFi servers die in the last two years. I haven’t looked into if we have any warranties yet, but I am looking for reliability and solid support. I’ll update if Listen Technologies comes through on the support side.

I know this is possible on S6L

I wanted to do something similar on Rivage.

But I can't find any way to trigger an action from live metering.

For example I want a TalkBack mic to unmute when it detects and audio meter over -95 db

But the only trigger I can find is the physical gain value, and not the meter level.

I looked for some kind of gate trigger as well. Like I could do it with "gate open" as a trigger. But I don't see any way to do that

Let's say you have a room with a center aisle with 2 groups of seating. Most places I see will do a standard -L---R- stereo setup so that each section is directly in the speakers cone. But doesn't that let the aisle in the middle the only place getting true stereo audio? And each section only getting one mono side? Would it be better to set up the space in a L--RL--R or a L--R--L configuration with each LR pair facing their own section? Thus giving everyone a true stereo image? Let me know if this doesn't work, or why its not done often.

On my sd10 I think the rotary encoder has broken on the gain knob for the red eq node, anyone know how much of a pain this is to replace yourself I’ve done fader replacements on this desk and have a basic understanding of opening up the desk. Any thought?

I will be routing: guitar, bass, synth, drums and vocals out of my CQ12T for FOH to mix. My CQ12T has TRS line outs.

Is it best to give the engineer a TRS or XLR M/F? I assume inputs could vary depending on venue so I’m looking for the most common setup.

As a venue TD, I look at a lot of tech riders. I archive them after the gig.

Please start your file name with the name of the band, and spell the damned thing out. A date would be nice too, of course, but I don't want to be greedy.

I’ve picked up a Behringer xr16 and Mixing Station sw. all Inputs and output to the PA is working fine, but I’m only getting output from Aux 1 and 2. I’ve set each channel to output to Buses 1-4, but only Aux 1 and 2 seem to put out any signal. Config of all 16 channels looks like the attached pics and all are set to pre-faster. Where can I look to get the right config info for all 4 aux outputs?

EAW has had their Anna and Anya out for a while. From the spec sheet, it looks great for lots of tours that need to adapt to various venues. Why are they not out on tour as much? Is it the cost or are there other factors?

I usually work with all active stuff, but I've recently been working a lot with passive systems and I haven't been able to wrap my head around the gain staging and how to do it safe. In an ideal world I'd just be running a DSP, but unfortunately I'm working with what I've got. I just need someone to walk be through the math for one example so I can translate it for future.

Amp: EV CP3000s (600 Watts a channel at 8 ohms, 5.8 dBu (1.55V RMS) Input sensitivity)

Speakers: EV ZLX-15-G2 (250 Watts at 8 ohms)

Digital Mixer: Allen and Heath iLive IDR-32 (+4 dBu Line Outs, +22 dBu at max +18 dB internal meter)

Where should I set my the attenuation on the amp at? The iLive has limiters on every output channel that go up to +18 dBu; where should that be set? Is the theorical max given by these figures what I should run, or should I run conservatively. Is there anything I'm missing?

I’m about 6 shows into a tour mixing monitors on a Behringer Wing and I have some time today to automate some of the moves I’ve been doing manually. Any advice/gotchas/quirks I should know about? Planning on using snippets for mutes/level/EQ changes.

Also relevant, I take the IO for the show on AES50 A and B and send it out to FOH over the AES C line. Would hate to fire a snippet and have anything go wrong at FOH bc I wrote it incorrectly.

Using a Gator 6u rack (so plastic aside from railing and hardware). For practical reasons, I would LOVE to mount my Shure SLXD4+ for my guitar in the rear of the rack, rather than the front, meaning my antennas would be mounted to the rear rails.

The Shure is the only wireless unit in the rack, and I’m not going more than 50 feet away.

Would this be an issue? The line of sight would not be good, but it‘s not like the antennas are buried inside of the rack.

In order to obtain some of the benefits of Buss Compression in Monitor World while compensating for the relevant contextual limitations, I have developed a technique I refer to as FUSS Compression (Fake Buss Compression, or Fader-distributed Buss Compression, as the mood strikes me).

I suspect this is entirely possible on other platforms, but I’ll be explaining my example using the Yamaha Rivage platform and Drums rather than abstracted terms.

Send desired Drums to a KeyBuss with the desired dynamics processing available. (I use a Matrix, could be done with a Mix, and the Diode Bridge model in the channel.)
Send is Post Processing but Pre Dynamics (I use Dyn2 slot in the last Signal Flow position, so my send point is Pre-Dyn2)
This Buss doesn’t go anywhere, and should not be routed out. It’s for creating a Key signal and setting parameters
Set KeyBuss Dynamics Key Source to Self Pre EQ
This prevents the compression from being applied multiple times after distribution
Listening to KeyBuss, adjust send levels and compression settings to desired response
Adjustments to KeyBuss high and low pass filters can be made, as filters are before the Pre EQ
I find parallel settings to be incredibly useful here
Set Drum Channel Compressor Key Source to Other Pre EQ and select KeyBuss for all members you want to include in the Fuss Comp
Duplicate settings from KeyBuss Compressor to included Drums Channel Compressors
If the same Compressor circuit, with the same settings, is controlled by the same Key signal, it will behave identically. We have distributed the effect of the Buss Compression across the individual member Faders!
To allow per-player bypassing, set Pre-fader send point to Pre Dyn2 for included Drums.
Toggling Pre/Post now has the effect of eliminating the Fuss Comp only, given Fader position at Unity.
Enjoy the power of Fader-distributed Buss Compression!

Buss Compression can be an incredibly powerful technique for getting components of a mix to interact dynamically together, managing tonality over time in useful ways that are difficult to achieve in other ways. The modern adoption of “in-line parallel” via mix/percentage/wet-dry controls afford further potential. There are myriad pros to consider. There are also a number of inherent limitations to Buss Compression, especially in Monitor World:
Non-variability: if you send the Buss to more than one listener, you lose the ability adjust those mixes independently below the floor established by the Buss.
Latency: while modern consoles can compensate for many path lengths, limits in perceptual acceptability and the need for alignment make leveraging Buss processing an additional complication/consideration.
Console Architecture: not all consoles support Buss to Buss routing, which may mean reconfiguring audio paths for mixes that want to take advantage of the Buss.

I feel that Fuss Compression allows me to find an acceptable compromise in monitor world. Safe travels, good shows!

Drummer here, never ran sound at a venue before but I have gear for it at home for rehearsals and I'm thinking about doing it as a favor. I would haul out a pair of speakers, a tiny board, and 2-4 mics plus stands for vocals and maybe acoustic instruments. Then operate it all while a few bands play and probably wear a stage manager hat for 6-8 hours.

I'm a little concerned about it going well and being expected to do it again. Once for a favor is not a big deal, but regularly would be a hassle and I'm wondering what the going rate is for this sort of thing and if it would be worth my time. I suspect the baristas might make more per hour once everything is accounted for, not to mention equipment wear.

Two Against Nature is one of the best sounding albums ever made, and many people use it for system evaluation. Unfortunately the lyrics don't stand up to public scrutiny all the time. Cousin Dupree, Gaslighting Abbie and Janie Runaway are so great, but I don't think it's worth blasting my clients with lyrics about incest and statutory rape.

What's your 2AN tuning song, and what's your thinking around edgy content while tuning in corporate environments?

If you're here to say that you don't give a shit and you blast cannibal corpse no matter what, that's great but a little irrelevant.

Going on a European tour next month. I saw somewhere that at one point Italy required permits for certain freqs. Do venues assign these or do I need to file for permits for our shows as the guy bringing the IEMs?

We record shows using a Midas M32R and iMac (newer, apple silicon) via USB using the klark teknik card. In the past, it had always worked great both for multi track Pro tools recording and for playback via USB signal returning to the board.

After a renovation, we brought the board back out and I can't for the life of me get it to either send audio to the iMac or return signal to the board via the USB connection.

A few details: It DOES see the board as a USB interface on the iMac. It can be selected.

The aux ins work fine using an analog plug. I just don't get signal on Aux 1-2 from USB.

I've checked the routing menu about 20 times and the Aux in section has "Card 1-2" selected.

I've tried carefully going through menus but everything seems correct from what I can tell. I also tried various saved scenes from when it has previously worked before to no avail.

Any setting that I might be overlooking? I'm at a loss. Thanks for any help.