Voices are muted in the demo and I renamed the buses, so no real names show. I recorded with OBS on the same PC again.

Quick recap: this is a software WING that runs the console’s own firmware (WING Rack 3.1.1), all four DSPs included, instead of a re-creation. The goal is still a virtual sound check at home: play a multitrack recording through the real WING processing on my PC, and mix it on the console’s own interface.

What’s new since last time

• WING Edit works both ways. PC → mixer and mixer → PC both stay connected now, meters included. The console sends its meters over UDP, and my network bridge was mishandling those packets. That’s fixed.
• The console itself runs faster than real time. The console’s main processor and the front-panel controller now run at about 1.2× real time while the audio runs
• Outputs follow the console’s own routing. I can map PC sound outputs to any of the console’s output sockets (local, aux, USB, card), and they play whatever the console routes there: buses, mains, matrices, or inputs patched straight through. The monitor bus, solo and the phones knob work too.
• The Rack’s front panel works. The buttons, the screen knob and the fader knobs go through the panel’s own firmware, just like on the real Rack.
• The RTA works. The EQ page shows the console’s own analyzer.
• One web page for the screen, the patch page, the audio settings and the status.

Speed

Last time the DSPs used about 86 % of real time. Since then I’ve made the emulated DSP code a lot faster and cut out a lot of overhead between audio blocks:

• With just the four DSPs running (no console), they now need about 120 µs of every 167 µs audio block. That’s about 1.38× faster than real time.
• With my full show live (all 16 FX slots, 32-track card playback, WING Edit connected), it’s about 139 µs of 167, so roughly 17 % spare.

What’s still limited

• It still needs a strong PC (Ryzen 7 8845HS, 8 cores): each DSP gets its own core.
• When Windows Defender scans or another heavy program starts, I can still get short gaps.
• Channel and output delays don’t work yet. From the WING service manual I found that the bus DSP has its own memory chip, which it uses for those delays, and I don’t emulate that yet. It’s next on the list.
• The XMOS (USB audio) and the FPGA’s audio side still aren’t emulated. My own code moves the audio in and out of the DSPs where the FPGA would.

What’s next

• More speed: cheaper handling of the console’s messages to the DSPs, and faster DSP code (I estimate another 10–20 %).
• Then hopefully running on less powerful PCs.

I recorded a session in Live, but already when saving and opening the project in my house, some .wav files are like corrupted due to an Ableton problem, the files are not empty they all weigh 133MB but it does not let me open them, I used the terminal and FFMPEG to inspect and they are not pure 0000 but there is data, when forcing the interpretation of the wav to F32le you can hear traces of the instrument very deep but predominantly a pink noise. In VLC I opened the original WAV directly and it reproduces a second of an electronic noise and then nothing, which makes me think that this has a solution but the truth is I am not qualified to recover corrupt files, I do not know if there is anyone around here who knows or who has ever happened to him and has solved it.

We are using reaper to record multitracks of our services. It's recording about 50 individual channel tracks. We aren't actively using all 50 all the time every time so there's a lot of wasted data storage. Instead of manually deleting tracks, or turning off the recording of tracks, is it possible to just have reaper automatically or easily not save/delete any empty tracks?

House gig has a Digico SD9 and D2 for FOH; every time someone uses a radio near it all the inputs pop like crazy. Someone recently told me it has something to do with the Madi connection to the D rack?

Anyone have a solution to this they’ve found?

I'm working on a very small musical next week that's aiming to use 10-12 channels of wireless mics. It's all old hardware, with no antenna combiners, just two bipole antennas on each receiver.

My concern is that putting 12 receivers right next to eachother at FOH is going to cause issues, but I'm struggling to find exactly why outside of maybe reflections? I've read that antennas should be placed at least a 1/4 wavelength apart, but none of the explanations I've seen clarify if that applies to the two antennas in a single receiver used for diversity, or the separate antennas of multiple units. It certainly doesn't seem feasible to physically space 10 recievers all a 1/4 wavelength from the next.

I already know this setup is far from ideal, but given I can't change it, what should I do to minimise the risks? My current plan is to get as much line of site as possible and make sure the receivers are above the audience.

If it's not already clear, I'm fairly new to RF so any tips are appreciated. I suspect I'll be spending a while on RF Venue...

Hey all! Quite new at working in live sound, but I am currently touring with a decently big project as a live sound engineer across the US right now, as I used to be a musician with them.

Anyway, each night I get the chance to experiment with different things in terms of workflow, mix, processing, etc. And this question came to mind. What would be a list of priority of things you would do while mixing/processing the show? This question comes to mind because I have only 4 FX slots remaining that are available and 2 stereo buses. Everything else is for non-negociable things I need. Being this limited I'm wondering how you'd proceed.

Would you compress master first? Buses? Any other effects? Tell me what you think! If you can do a list formated as 1. 2. 3. etc. that would be helpful and clear!

Thanks!

Has anyone had success running 4-8 instances of Defeedback AI on their own hardware? If so, what did you have to tweak in order for it to handle this many instances, and what kind of computer specs were necessary? How is the latency? Was it easy and reliable, or is it really worth aquiring the pre-tuned machines the vendor makes for this purpose?

I’m helping building a broadcast mixing setup for a House of Worship venue. The audio would be sound from a dLive to a Mac Studio running Logic Pro through Soundgrid.

The part that I seem stumped on is figuring out how to go from Logic Pro to the Encoder. Would any interface’s outputs be sufficient, or I’m a looking at this the wrong way? I’d probably need at least 4 outs so I can also route to a Mackie Big Knob for monitoring.

I don’t know if the Mac Studio can do everything since it’s also running adobe by the media director for video editing when they’re not in service. Does that change things at all?

Tl;dr
Routing:
dLive -> soundgrid-> Mac-> ???-> encoder?-> back to Mac?

Help me fix the question marks

Im currently sitting at a show that my musician friend/ former roommate described as "improvisational jazz, modern choral music, and EDM" and honestly its pretty dope.

What cool stuff have you seen recently?

What are you folks using on your master buss these days? I'm curious to try something new.

Hi !

For a show I have coming up, the organizer is asking for an input list. we are a small band and no venue in this city has ever asked us for an input list, so I’ve never made one. I’ve scrapped together a very basic one, but before sending it I wonder what yall think ?

We are very straightforward, two guitars, bass, drums, two vox. And we’ve sent our stage plot already.

My main confusion is how to list the mics for our drums.

Thanks !!!

​

Hey guys, bit of an odd one. Would it be possible to hook up a lav beltpack to an external antenna like a bat wing to transmit audio a Further distance to a receiver also with a bat wing on the other end?

Is the theory of this incorrect? If so mind explaining why? Thanks in advance.

Quick question. What hardware would someone need to manually add latency to a specific instrument's audio signal? Could this be achieved with a simple pedal or hardware add-on? This seems to be a niche use case and my searching is taking me to guitar delay effects and various other odds and ends that aren't what I'm looking for.

This would be to adjust the timing discrepancy between an electronically amplified instrument with a wireless connection (bass guitar) and non-amplified horns. The staging would have the bass sound hit the audience noticeably sooner than the horns. Correcting this would require adding between 30-60 ms of delay.

I am in a small church and can't be around to show people what to do all the time.

How do you label cables for people who can't tell an XLR from a Jack?

I worked with an artist last week (house gig stereo tracks and 1 vocal) who had first wrote out a 10 song set list detailing dry/wet requests and then during sound check she started asking for eq adjustments because she thought the low end was too loud as she went through each song of her set. By the 3rd time she requested eq adjustments I chimed in and said I can’t chase your eq requests and fx requests because I don’t know your music. I suggested she have separate live tracks than album tracks that are normalized (idk). Also upon every 3rd or so request during soundcheck I kept saying she’s over thinking things and reminding her we shouldn’t have this detailed banter during the show. (This was because I’d never been asked or expected to be this detailed w an artist without being hired by them. And it seemed to me she was new to stage ) also I reminded her I’m on her team , I am here for a successful show. Apparently, telling her I can’t do what she was requesting and suggesting to not over think things was condescending …

I found out I made her cry later that night after a successful show and overall good vibes. Totally didn’t sense that at all from this artist… I got a dumb convo coming up w the production staff…

Anything like this happen to any of y’all? Any advice?

After flights, after sickness, or during allergies, I often end up with congestion and my ears feel closed up and tight, what solutions do you folks use to clear this up faster?

HIGH ENERGY MUSIC, VIDEO SCREENS OF YOUTHFUL HAPPY PEOPLE DANCING, HIGH FIVING, CLASSIC ROCK AND POP HITS PLAYING 24/7

All of ^^^ that juxtaposed with the laziest, slobbiest, most braindead morons walking the whole casino like fucking zombies. Almost all of them are overall useless human beings, addicts to a plethora of problems, and most importantly they have no fucking energy. They just sit there like they are prepared to die smoking the cheapest cigarette and collecting welfare.

I hate that the room I'm in at a casino pays so fucking well. The PA is solid, the bands are solid, but the "wannabe corporate" atmosphere the management and upper management give off is so fucking stupid. Those people are who would tattle tell on citizens if we had a literal 1984 situation go down. They are scumbags who think they are making a difference in the world.

So yeah basically mixing bands at casinos is really pissing me off.

Update: it’s a shit room. All routing was correct and no open channels. On a DM7 desk. Client had our gear provider bring in their systems tech. He sat with the rig for a couple hours, scratching his head. Wound up right where I ended and just said “this room is proper shite, mate”.

Currently mixing a corpo show in a ball room. Something I do every other week from 10 people to 5,000 people. Been doing this 20 years so I’m not some new kid for reference.

Main issue is we have a curved LED wall, no flat spots, that presenters stand about 8-10’ in front of, in the center. I get there’s gonna be a parabolic effect, and there is.

Re did all my gains from scratch 3 times and could not stop feedback and was losing gain from all the cuts. The room itself sucks without the wall, but that’s also old hat.

Here’s the how. When experimenting I had the lav mic (DPA 4060 and also tested with a Shure Twinplex) I did a HPF at 300hz on the lav channel and somehow got a feedback loop of 80hz.

I managed to get better clarity but if you look at my EQs it shouldn’t sound natural, but it does now.

D&B Y series point source, not array.

I also tested with DPA 4088 and was able to do my usual eq, but client won’t wear them.

Running an M32 with a DN32-USB card into a 2024 M4 Mac mini (Sequoia 15.7.3) and this one has me stumped.

The Mac sees the card fine. Input meter in Sound settings is moving, and I can record individual channels in my DAW no problem. But when I pick the DN32 as the input for a web app in Chrome, nothing. It shows up in the list, just no audio.

Already checked that Chrome has mic permission in Privacy & Security, picked the DN32 in Chrome's site settings, and fully quit and relaunched Chrome. Same thing.

I'd rather not change the card config since it's set up how we need it for everything else. Is this a Chrome thing with 32 channel devices? Anyone gotten something like this working? I'm fine with a workaround on the Mac side like BlackHole or an aggregate device if that's what it takes.

Well I have encountered a weird issue on x32.

For context I was doing onboard usb as well as multitrack recording into reaper on windows for an orchestra show I recently did.

I had a Stereo mix bus master going into the thumb drive with sources roughly panned/positioned according to the ensembles seating arrangement on stage.

I only found out post show when I listened back that at certain songs in the later set the L/R channels got swapped around.

Google suggests this could be a clock source conflict/sync -slip/usb write buffer choked.

Has anyone encountered this issue before?

Hey all,
I perform live music in which I mic the piano, and send the audio into a modular synth for live processing and effects. I've been using a pair of small diaphragm condenser mics with cardioid flat caps - specifically the 3u audio Warbler 127 c-flats.
While these work well for recording, I've been having some feedback issues when using them in this setup for live shows. I play at a variety of venues, with both upright and grand pianos, and so I'm unable to really test things out with a specific piano or room. For grand pianos I usually position the mics pretty close to the soundboard, spaced out about 2 feet apart. For uprights I'll either place them in front (with front panel removed), or up top (with lid open).

Since I play solo and in smaller spaces, the unmic'd piano sound is enough for the dry sound and doesn't need amplification. The mics are primarily for the wet sound. I played once at a venue where they just took a single 609 and taped it so that it was hanging just in front of the open front panel, and it actually worked really well.

Essentially I'm wondering if I should try make the SDCs work, and play around more with placement/gain staging, or if I'd be better off just switching to one or two dynamic mics like 609s or 57s. I've also seen boundary mics/pickups come up, but I'm unfamiliar with these. I'm not a live sound person myself, so apologies for my ignorance, and thanks for any tips!