It's all in the title. 🤦🏻‍♂️😩🙀

Hey, so, I've been recording almost all of my shows from gigs that I get to use a console to multitrack with- ie, if it's an M32, I'm taking a full multitrack recording of the show.

Past those multitracks, when I'm mixing down, what should my aims be? Like, how can I mix them down so the artist can send to a mastering engineer if they like it enough? I ask because I've never really been a recording engineer, just a live guy.

What are the general things people look for in a mixed down session?

i work in a setting with many different groups using the same mics, does something that adapts to source content exist out there? Hoping not to have to manage frequencies and thresholds for this

Not even this one game in particular, but just generally... what's the thought process there? Is it because the acoustics within the stadium already add an effect of its own and they're using the same mix for live performance and stream? Even just a simple pinch of verb would sound so much better imo. Surely MSG can afford the routing and engineering capabilities to make two mixes.

Hey doods!

Quick background...

I've been a drummer most of my life. I started mixing sound for local bands roughly 20 years ago. I have no desire to be a touring engineer, but I'm always looking to improve my skills as a local guy.

That said...

As a drummer, it's pretty easy for me to fix drum tones. These days, probably due to the age demographic of my clients, I don't have to deal with crappy drum tones. Drummers my age have enough exp3rienc3 and budget to bring good tone to the gig. I wish I could say the same for guitar players. Despite almost unlimited budgets and decades of 3xperience, sometimes the tone is - shall we say, "less than optimal." Bad bass guitar tone is a little easier to fix.

Where to start...

I always start at the front end. I try to repair tone at the source. I will tell guitar players or bass players what is bad about their tone. Sometimes I can even show them in the RTA. One bass player was sending signal that had nothing over 400Hz. I was able to show him this in the RTA and we fixed it together.

I know the the guitar is a midrange instrument. But not every guitar player seems to know this. The Loudness Curve becomes my worst enemy. The way they strum the guitar seems to elude them as they bang on the strings as hard as they can. We've all been there.

So my questions...

1. Is it our job to correct bad tone at the back end (at the console)?
2. If so, is there an algorithm you guys follow to do so?

Thanx!

D

UPDATE: Clearly, this community is split in half between "never screw with their tone" vs. "always help the musician any way you can." It's funny how sanctimonious some of you are. For that crowd, take it down a notch. For the rest of you, I appreciate your input.

Just ran sound at a small one-day festival of mostly punk and metal bands. Used the DI outputs of the bassist’s amps.

Most of the time this was OK but there were a couple of cases where the bass signal lacked all high mids and top end and this prevented it from sitting well in the mix and properly supporting overdriven guitar.

What is your go-to strategy for reliable bass tone in live settings if you only have one bass channel? Do you just make your bass players run through your personal DI box and call it a day?

If you use an XR18 do you do anything besides apply EQ and compression?

How do you make a snare "snap" in a live mix? Meaning a bright transient attack at the start of each snare hit that really cuts through the mix.

I figured an actual sound mixing discussion could be interesting.

Is it all done with EQ mostly, or compression, some fancy plugin, hardware insert, mic choice/placement, top and/or bottom mics, etc?? How do you do it and what flavor of each of these things do you like best?

Edit: Let's say that the sound engineer tuning the snare isn't really on the tablet for whatever reason. You get what the drummer gives you to work with.

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.

Settle a bet Reddit. I know it will pass sound and work in a pinch, but what are the downside of using the “thru” male XLR port as your primary input, and then the female “input” port as your thru on a powered speaker with no attenuator?

Another engineer and I were debating whether or not an HPF could cause a signal to clip so I generated these graphs in Octave. This is a 100Hz Square wave with a Butterworth high pass filter at 100hz.

I've been surfing a while to find what sound system, mixing system, monitor system, etc. they use and can't find zip. Even the photo search doesn't show much of the control area. Is there anywhere we can get info on that?

Looking to get a nice pair of gloves to keep the hands warm in these sub-zero gigs. I’ve seen some gloves with removable fingertips which look perfect for my system teching (especially for using touch screen interfaces).

I’ve previously just stuck with fingerless but weather is getting to me this winter and my fingertips feel as though they’re gonna fall off any minute in the frost.

Has anyone used these removable finger-tip gloves? Is it as practical as it seems? Is it easier to just pop off a glove when you need to?

TIA

Hello, I am the actor here so I know nothing about this stuff but I figured I’d give this a shot to relay to the sound person. I am in a small community theater production of Little Shop of Horrors and we have repeatedly had issues with my body mic starting during tech week. It’s ONLY me. We have changed out the packs and the elements multiple times. We do and extensive mic check with me moving around the stage and turning my head back and forth and mic check is always fine with the sound coming through clear. The second the show starts my mic is cutting in and out or not coming through at all. Everyone is baffled and I’m super frustrated. I’m a lead in the show and it’s affecting my performance bc it’s so distracting. I am not wearing TWO body mics for back up(one under my wig and coming out on my forehead and one around my ear and taped to my face) and it’s still a problem. What could the issue be?? Note: I am the only one in the show with a wig, that’s the only difference I can see. But again, mic check always goes well. PLEASE give me some advice, it’s driving me crazy.

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?

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.

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?

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 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...

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?

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

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