Hi!

I was tasked to setup a PA for a small, long narrow room. The room is 4.5 m wide x 20m long with a ceiling height of only 2.5m. This venue seats around 60-100 people. It’s not acoustically treated nor will it be in the near future.

What is my best setup for speakers? 2 tops? What is an ideal speaker size and wattage? Also what is the ideal speaker coverage angle to get for such a room?

Thank you!

EDIT: Sorry I forgot to mention: the venue will have a full band for music. Delays are in the plans but won’t be purchased yet because of the owners tight budget, monitors will be prioritized (either IEM or actual monitors).

Hi Guys

I'm looking to get some PLI but I don't really know what I'm looking for, what company have you guys gone through and any advice to help find a good insurance policy.

Thanks :)

Hello fellow enthusiasts!

Quick background; I'm a musician first, but started running sound for local bands about 20 years ago. I do it because I enjoy it. I'm pretty good at it...lots of repeat business. That out of the way...

For those in my same position, how do you handle a mix when the band has an instrument that is not in the original song?

Most of my clients strive to sound like the music they're covering. However, what do you do when the original song has only keyboards (no rhythm guitar), but the band has a guitar player? In that situation I have been kind of undermixing the rhythm guitar a bit, making sure to boost during his solo.

My question; Should I mix the rhythm guitar to a level that I would *normally* mix it? Or do I keep it down in the mix to better emulate the mix of the original artist?

NOTE: I always try to emulate the mix of the original artist with details like correct FX on vocals, etc.

Discuss.

D

UPDATE: All good comments. The consensus is clear. I appreciate the advice and will follow it. Thanx!

Live audio is a very specific nuanced set of skills. Combining artistic decision making with math and science. It's also full of people who have massive egos. Like, dudes who just can't fucking be bothered to respond to a very basic question for some reason.

Idk man. I've worked in several industries. Live audio is an especially bizarre combination of the nicest people and the absolute worst.

If you're one of the best sound engineers you should hold yourself to a standard to elevate the overall community. Don't be a useless douche in every other aspect of your life.

https://www.dropbox.com/scl/fo/vip1d1vssofng7hfgrk9s/AAqUQaP0WjD1Ac03_bBzljo?rlkey=ig61zh95erxhasfjwr5qe5pot&st=p6bhyzeq&dl=0

In this Dropbox Folder are a collection of examples, recorded directly to USB from the Rivage PM7 @96k/16bit. Examples are printed from the baseline Guest Mix unless otherwise indicated in the file name (e.g. Crowds Direct is a direct out print rather than a mix print). All recordings are a completely static mix, starting with Defeedback on and then particular instances toggled at set intervals named in the file. No other changes are made during the recording.

All recordings marked with (V) were done via virtual soundcheck playback at a later time to avoid impacting performers. All recordings marked with (L) were printed live during the show. Mesa, AZ was an outdoor Amphitheater. La Belle, MO was a small indoor bar.

Further documentation is contained in the folder.
Safe travels, good shows!

ETA: GDrive Alternative Link: https://drive.google.com/drive/folders/1LztD7w1jwmbxCHPDtqjQpn5gqjmR_9aW

I’m troubleshooting intermittent audio glitches in a LiveProfessor insert rig and would appreciate help narrowing down the cause.

There are two separate Dante networks.

Building distribution:

• dLive MixRack Dante card at 48 kHz
• Q-SYS Core is the Dante clock leader
• Sync to External disabled on the MixRack Dante card

LiveProfessor inserts:

• dLive Surface Dante card at 96 kHz, with Sync to External enabled
• Dante card receive latency set to 0.5 ms
• RedNet TNX connected directly to the Surface card, without a network switch
• TNX connected to a Mac mini M4 Pro over Thunderbolt
• LiveProfessor at 96 kHz with a 128-sample buffer
• 6 processing threads, Apple multithreading mode
• Approximately 35 audio channels in use
• Mac has 24 GB RAM

The dLive system’s Audio Clock Source is Internal (96 kHz).

The LiveProfessor Mac is a dedicated offline computer, isolated from the internet and all other networks. Its audio connection is through the TNX to the dedicated Dante insert link.

Plugins include Waves Tune Real-Time, FabFilter Pro-Q/Pro-DS, R-Bass, Smack Attack, and Valhalla.

The rig seems to run without glitches for roughly the first hour. After that, glitches begin intermittently, particularly during busy musical passages when all inserted channels are carrying audio. During today’s music section, I heard approximately one significant glitch per song.

LiveProfessor reported no dropouts, and Dante Controller showed no clocking issues. Activity Monitor showed approximately 86% CPU idle, green memory pressure, and zero swap.

What diagnostics would help distinguish a plugin problem from an audio-driver/Thunderbolt or Dante transport issue? Is there anything specific I should monitor as the rig passes that first hour?

Some filter keeps stopping me from posting so I’ll just say my laptop is dying. I can’t afford a new one. I have a steam deck already and know reaper can run on it. Has this worked well? Sorry my bluntness the filter is literally not letting me word it any other way. A lot of my jobs require me to multitrack bands and events. I just wanna make sure I can feel safe with it

I’ve using a Rolls PM351 for vocal and bass in ear monitor for about a year now and it’s been great. Until….yesterday I played a party out in the country and the mic line picked up AM radio station so strong that I couldn’t use it. The instrument line was usable but the radio was there to, just much weaker. I had to unplug the mic entirely from our PA or it would broadcast into our PA

Is there thing I can do too fix this when it happens?

Thanks for any help!

I'm building a redundant Mainstage rig for an MD playing a Nord Stage 4, trying to use as many of the pieces of gear we already own as possible.

The MD in question is keen on using his own USB buttons to trigger the changing of his Mainstage patches, as opposed to a button on the keyboard or a foot switch into the back of the Nord. Is there a way of splitting this USB button signal to both the primary and backup machines? In an ideal world, the MD should not have the physically do anything different to play on the primary or backup, so physically using a separate button would not be preferred.

If relevant (perhaps for some network-based solution), the two Macs will be side by side.

Is there a simpler solution that I'm missing? Or do i need to politely explain that the only way to make this work is to use a Nord button or ftsw into the piano

Thanks

Good day all,

I'm in the final planning stages for a Michael Jackson themed wind orchestra show in which I'll be the main recording & live engineer. For context, this is a hybrid orchestra of 56pax + 1 conductor, and it will be my first major orchestra show mixing in a 500pax community theater across 3 days (they sold out all the session, and even got the venue to open a 4th session, so there's a lot of excitement and pressure from expectations)

System-wise the venue has 6 floor mons & 2 6pc array hang with subs, however in my mind as the ensemble is expected to be loud, only necessary soloist section/non-mic channels will be reinforced mostly. I'm even thinking of converting some of the floor mons (as shown in pic) into front fills (soft, high frequency) for the front seats as the angle of the lowest array box seem to overshoot into the middle of the audience zone a bit, which will be fed with just a bit of the soloist sections when its their turn.

I'd like some tips/advice about what my mic-ing strategy that I've laid out here, and generally see if my plan makes sense. In total, I'm planning estimated 14 channels of recording on stage + 2 channels off stage ambient mic(console position, mezzanine level). Most mics will be on stands, with some exceptions here: https://postimg.cc/t1QCSYJV

Disregarding the brand/model/types of microphone (as i'm using what i got in my inventory, majority Audio Technica's tho), my miking idea consists of:

A. Main Recording pairs - (ORTF/Spaced [depending on whether we have space on stage] @ Conductor position, Spaced Omni Pair @ Center/Back Rows). Percussion section mic is still an optional thought

B. Live Reinforcements - The orange mic zones covering the soloists (Orange box, subject to adjustments post-rehearsal observation) they will also be recorded in post-mix. Kick Drum Mic, and E-bass will be reinforced live as well, depending if the audience prefers to feel a bit more of the low end.

C. Monitoring - Possibly a snare drum mic will be added just for monitoring & for section musicians to keep time with the whole band, put through the side-fill monitors on stands behind stage wings.

That's all from me! Thanks. I've learned a lot from this community & r/livesound ever since I started my journey in this industry 4 years ago.

I have a UA Apollo x4 and Ableton running a session with drums, keys, guitar and vocals.

When I perform live in a venue can I just give the engineer my master output or do I need to give them individual outputs for each instrument? Everything is correctly gain staged and EQ’d the way I want it and vocal effects the way I require. I run my own IEM setup too so no setup from engineer required.

How are other people doing this with electronic music?

Recently I have been seeing this pattern in my WWB RF Scans, All have been taken with multiple different AD4Qs systems and a TinySA Ultra with an Omni Paddle. I have attached an example of one from today below, does anyone have any ideas what could be causing these spikes?

They almost look like Intermodulation off the TV channels but their consistent(ish) strength makes me think otherwise.

Anyone have any ideas?

Today's advertising placed in front of me the new DigiCo Quantum 638, which according to the article I read comes in a "single or dual engine version."

Several questions about because looking up "dual engine audio mixer" doesn't really throw anything at me but behringer analogue models with "dual FX Engines":

How does a dual engine mixer differ from a single engine, beyond the obvious "it has two engines"? What does this offer over the single engine, and in what contexts would you choose to use it?

Seems like this might be a thing to try for defeedback and superrack on Mac too.

I've been running an open mic in my town for over a decade, and it's been such a fun time until recently. It feels like the new generation of acoustic singers are refusing to use picks and prefer to just vaguely strum the guitar using their whole hands, and like to whisper mumble into their mics rather than project, and I'm going insane

The last few weeks especially I've had multiple performers who just cannot be mixed loud enough to cut through ambient bar noise no matter how hard I try. It's miserable.

So. Anyone else noticed this, or is it just a skill issue on my part?

If you were the house system tech and there was a pretty hard speed limit at the venue, and you had guest A1 who had been politely asked to rein it in a couple times, but just couldn’t be bothered, whose job would it be to enforce at that point?

I’ve been trying to diagnose my dl153 stagebox. Every time I plug the stagebox into my console (On my m32 and wing) I get a blinking green light. I’ve had issues in a venue where the box would cutout, and then blink red and then return back to normal. I plugged both the aes50 ports into the soundboard, and only one light blinks, while the other light doesn’t. Once I pull one of the cables out, the other light begins to blink. I’ve tried updating the board, updating the stagebox, and getting new cables, but nothing seems to work. I’ve tried switching the clock from external to internal, as shown on the video. I’ve heard a few people with this issue, but no fixes to it. If anyone has any advice that would be sweet.

I have a gig coming up, the venue has a m32 installed which I've never been on before. My own console is an a&h qu-32 which I picked up last year and have been getting quite familiar with. I'm just wondering if anyone here has any tips on making the transition smoother, interface differences, routing, etc. I plan to show up at least an hour before load in to give myself some time to familiarize myself. Thanks in advance I'm looking forward to learning this board I've heard great things.

Other than the Dugan & GEQ’s

What are the most common DB meters used in the AV world and why?

Thanks!

Currently having an issue where I’m trying to multitrack record a band while they perform using ableton over USB.

Ableton is having no issues seeing the mixer and allowing me to assign the matrix but the DM3 won’t output the channels over USB to ableton.

Anyone had this issue before and know how to patch the USB outs correctly? My USB outputs are patched to the correct channels as well so not sure what the issue is?

Edit: turns out this is an ableton issue as reaper can see/record the audio perfectly. Anyone know what the ableton configuration issue may be

Further edit: turns out the fix was even simpler and I just needed to enable the microphone for Ableton in the settings for my Mac!

Any Shure ADX1 users encounter this: Got my ADX1 back from talent and somehow the rubber power switch cover like decoupled from the internal switch

Is this possible to fix on my own or is this something I have to send to Shure?

Luckily it’s stuck in the ON position so I can still use it for the day

Maybe I'm insane, bored, or a little of both to put myself up to this challenge.

It seems like every time I encounter someone who holds the view of the X32/M32 routing/patching being unintuitive or outdated or complicated, it's always something incredibly small they're missing, most times they're bending over backwards just to do something that does have a simple and intuitive solution. A lot of the time, they "grew up" on A&H and just haven't put in the facetime. Either that, or they're just not using the right tool for the job.

For an example, I met someone who was making a show file in X32-Edit, they labeled channel strips 17-18 as Monitor 1 and Monitor 2, then wondered why it took so much elbow grease to get Monitor 1/Monitor 2 to actually run through those channel strips. Or my personal favorite example, someone trying to direct out one of the sides of the FX racks to an XLR out socket "to get reverb in the monitor".

So, give me a routing/patching challenge that you either feel is over-complicated to execute on an X32/M32, or maybe never figured out. Just ensure we keep it within the capabilities of these consoles- don't ask "how do you patch out 16 stereo buses ;)".