r/livesound

Hello, I am currently attempting to add crowd mics to my venue. I am confused about how you’re supposed to route crowd mics to the stream. We use a Midas M32, and a PC with OBS to stream. The M32 and PC are connected through USB.

I tried googling and asking AI but I’m still confused.

I understand the part about disabling the Main LR in the menu, but the whole part in the routing page where it has all the outputs, card, etc, to make it go to the stream through USB is what I do not know how to navigate and setup.

We are likely using the inputs on the console itself for the crowd mics, since it would be easier to connect the mics to the console than the stage box.

Thanks!!

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.

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.

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.

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.

Edit: completely forgot to add my country, UK Based

Thanks :)

One of our new "techs" had his first monitor shift and this person loves to dig into settings that he doesn't know about.

Basically now whenever a scene is recalled all of the gains are set to +50 and some of them are -6

Does anyone who knows more about Yamaha than me tell me what this might be?

I have my own "preset" setting in the settings that I recall with my own user settings so I do not think it's any setting within there.

I don't share any gains with foh so it's not that

Thank you

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!

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?

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?

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!

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.

The only stupid questions are the ones left unasked.

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

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

Yes it's back! Please keep all show and tell type posts in these weekly threads. Unless you have a specific question about your setup, keep those types of pics here. Bonus points if you include a list of equipment with your picture.

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

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!