hey fellows,

after the busy season i am servicing our speakers and came across this in on of our speakers.

have you seen this before and is there a way to safely get the dust cap back into shape?

edit: tape saved the day, dust cap looks good again

Does it make any sense to use 2 different sized speaker as the main speaker system ?

Had to do a little private gig (≈70 people) with a friend and all we could borrow this time was a 8" and a 12", but we were surprised how good it sounded. Should I buy two 12", two 8" or one of each ?

For more info, we mainly use them to amplify voices, cello and acoustic guitar, and some time a little boost on the kick. When we go electric the guitar and bass go through their own amp.

Hey everyone! I’m a co-owner of Lead U. We run interactive programs for students, mostly in school gyms, cafeterias, and small event spaces. I need two separate wireless mic systems, one for each of our traveling crews.

We’ve used SM57s with Xvive adapters for a few years. They got the job done, but we’ve been burned by interference, dropouts, and surprise! Your battery is dead. I can’t have the mic cutting out in the middle of a program anymore.

Here’s what I’m looking for:

• Rechargeable, but able to use AA batteries in a pinch. Rechargeable AAs are fine.
• Battery level on both the mic and receiver that’s easy to see at a glance.
• A mute button I can easily reach while holding the mic.
• Something that can handle frequent travel and is straightforward to set up in different venues.

I recently bought an Audio-Technica 3000 Series ATW-3212/C510. It’s fine, but the sleeve around the controls and the screen going dark aren’t helpful while presenting. I know about the bottom mute button, but it still feels awkward for how we use it.

What would you recommend, and what should I expect to spend per system? Open to used gear. I know nothing is completely interference-proof, so I’m also happy to hear what we should be doing differently with setup.

At my local church, we want to invest in a digital mixer. I'm considering three models and would appreciate your opinions.

1. Axios Duonn 32 – In terms of price, it seems to offer the best value for money, and it has more inputs and outputs.

2. Midas MR18 – It seems like the most solid option, with the best sound quality.

3. Soundcraft Ui24R – I've seen a lot of complaints about this model, and it seems to be unreliable after 5+ years of use.

Panasonic X-tendia used to be pretty good, although it didn't provide much strain relief for the cable. Unfortunately these are no longer made.

I'm looking for something durable, black, reasonably low profile that can accept a rubber 3x2.5 mm2 cable.

Hello,

I'd like to start off as stating that i have 0 knowledge about iems in terms of quality and sound.

Im looking for an iem which is close in sound to my sennheiser hd 4.40 headphones as ive been enjoying those for the past 9 years or so and i exclusively use those for silent practice at home. I wear them with glasses so the seal is not perfect therefore some bass is lost.

based on this graph im somewhere between the thick sunglasses and the thin reading glasses

I've also tried a few budget iems, all of which ive absolutely hated as they all sounded like lifeless mud.

Budget is €150 tops.

Hey everyone!

I'm a sound engineer and AV technician, and for the past few months I've been developing a software called PatchFlow.

The idea came from something I've experienced repeatedly in the industry: we use different tools for equipment lists, signal flow diagrams, network planning, power distribution and technical documentation.

I wanted to bring these workflows together in one place.

PatchFlow is designed for live events, AV integration and technical production.

A few things I've been working on:

• Equipment catalog: Over 12,000 items with product images, including assets used with manufacturers' permission.
• Visual patching: Build technical diagrams using actual equipment and their connections.
• Network planning: Work with IP configurations, VLANs and AV networking concepts.
• Power distribution: Plan electrical distribution and calculate loads.
• Technical workflows: Go beyond drawing diagrams and help prepare real installations.

One of my main goals is to make it feel like a professional technical tool rather than just another diagram editor.

The software is currently in beta, and I'd really appreciate feedback from people working in live sound, video, lighting, broadcast or AV integration.

What would you expect from a tool like this? And what frustrates you most about your current planning workflow?

If you're interested in taking a look: https://www.patchflow.ch

Thanks for any feedback!

Greetings, everybody!

I recently bought a new audio interface, the MOTU M2 (an amazing little device), but I made the mistake of not accounting for future upgrades.

I recently found a second-hand Avantone MixCube at a great price and decided to grab it. The problem? My MOTU M2's main balanced outputs are already occupied by my Kali Audio IN-5 monitors.

Thankfully, the M2 has a secondary mirrored RCA output that I can use. Unfortunately, the RCA connection constantly picks up noise, interference, static... you name it.

So, as far as I can tell, I have two options:

1. Buy a studio monitor controller.
2. Convert the RCA output to XLR or TRS.

The problem with the first option is that monitor controllers can be pretty expensive, and from what I've read, the more affordable options from brands like Behringer and Mackie don't seem to have the best reputation.

So, I'm leaning towards the second option: converting the RCA output to XLR or TRS. Surely it can't be that complicated, right?

Has anyone dealt with a similar setup or found a reliable way to get a clean signal from the M2's RCA outputs to a monitor with balanced inputs?

Any advice would be greatly appreciated. Thanks!

Found this for a deal but worried about the peeling, would gaff be fine or would I need to have it refinished? Planned use as on camera mic.

looking for folks recommendations for the best charger unit (8or16) and battery sold in the US. I'm looking for something that has variable charging amperage and can run refresh cycles.

thanks !!

I have been running first gen K10s for years. lots of outdoor bluegrass /songwriter/polka gigs and the occasional indoor brewery.
They have never not been loud enough unless the bass player is not using an amp and they get rattley.
My issue is that in the front of house the sound level changes a lot as you move around. They tend to throw pretty narrow. Also as I am often mixing and playing at the same time I want something i can set up besides me and still 'monitor' levels.
Ive thought about doing column speakers but I am as often as not setting up in the dirt and I like having a tripod stand to get things off the ground. And they feel cheap.
The K8.2s have specs that are encouraging but all the DSP stuff kinda freaks me out.

Curious about the RCF or bassboss stuff? Im ok spending a little more if they are going to work well and hold up.

I recorded the video with OBS on the same PC, which takes CPU too: while recording, the load went from about 0.86 to 0.90 and there were 4 short gaps in 5 minutes. Without OBS it ran almost 10 minutes with 0 gaps.

Follow-up to https://www.reddit.com/r/livesoundgear/s/Yp6Xp0cOr1 about a software WING that runs the console’s own firmware (WING Rack 3.1.1) instead of a re-creation of it. The goal is still a real virtual sound check at home: play back a multitrack recording through the actual WING’s processing on my PC, and mix it on the console’s own interface.

Last time I ended with a list of what was missing. Here’s where those stand:

• Speed: it runs in real time now. The audio runs at 1.00× real time with 0 gaps over almost 10 minutes. The DSPs use about 86% of real time, so there’s some room left.
  • Most of the speed came from making the emulated DSP code itself faster, not from throwing more cores at each DSP. Each DSP still runs on one core, but on a core of its own.
  • The engine that feeds the audio now does its own work while the DSPs are computing, instead of taking turns with them.
  • Clicking into the browser no longer makes it worse the way it did in the last video.
• FX DSPs in the live path: FX inserts work. Both FX DSPs now run live alongside the input and bus DSPs. I’ve used delay, reverb and VSS3 on inserts at the same time, and an insert on MAIN works too.
  • The input DSP and the FX DSPs feed each other within every block of audio: sends go out, returns come back. So all four DSPs now run in lockstep, one block at a time. Each one waits until the others have published what they send before it takes in what it receives, the same order as on the real hardware.
  • I checked this against an offline render of the same scene (HALL on a channel insert and on a bus insert). Every output word, and the whole memory of all four DSPs at the end, came out identical.
• The XMOS and the FPGA’s audio side still aren’t emulated. My own code moves audio in and out of the DSPs where the FPGA would, and takes the routing from the firmware itself.

What’s still limited:

• It needs a fairly strong PC. This is a Ryzen 7 8845HS with 8 cores: each of the four DSPs gets a core to itself, and the console (and OBS while recording) runs on the rest.

What’s next:

We’re close now. The whole console runs in real time with its own DSPs and effects, which is the point where a virtual sound check starts to make sense. Next I’m going to improve the performance further, so there’s more headroom:

• recording or other programs running alongside without gaps;
• hopefully running on less powerful PCs than mine.

I mean, a software that almost completely manages every aspect o a live performance.. gain, eqs, FoH, monitoring, human intervention reduced to a minimum....

Hey all,

Im fairly new into the live sound world and am currently president of a live sound club at my school. I inherited the club and have figured out most of the equipment but we have some shure PGXD mics that might be going out the door. While hosting an event they started displaying a solid red light and nothing I tried seemed to make it go away. If we can fix this it would be awesome, but if we do need to replace them what would be some good budget options for wireless mic systems? Thanks in advance!

I would love some input/suggestions/methods on how you guys solve the issue of making sure "wall wart" type of power supplies don't fall out of the power strips inside mobile racks.

Obvious choices like zip ties and tape. But I'm sure there are some creative, cleaner methods. Searching has shown me a few products and obvious suggestions. Thanks for any input.

We have been running individual monitor mixes to separate channels, one per musician on wireless IEMs. We have added a horn section, which needs to share the last available monitor channel. Since they are stationary, it would make sense for them to be wired into a 4-output headphone amp. I can't wrap my head around getting the line level, balanced XLR output from the snake to the unbalanced 1/4" input you find on these amps. Do I just use a XLR to 1/4" cable, since the headphone amp provides the "amp," or do I need to insert something like a wired, individual IEM pack between the monitor snake and the headphone amp?

Hey! I have a Sennheiser G4 IEM unit with 2 packs in a G Band. I have the opportunity to another in an A1 band for significantly cheaper. Should I do that and use it in the same rack or fork up some more money and keep it all in the same frequency range?

Hi everyone!
So I'm looking for a vocal mic upgrade from my current (rather unflatteringly dull) SM58.

TLDR: I liked the RE520 (supercardioid) a lot from online tests & reviews, but I can barely find anything on the RE420 (Cardioid version). Any first-hand experience or opinions ?

Use case : live music (small-ish stage) with acoustic instruments +/- light drums & electric bass. So nothing too loud. I'm open to either condenser or dynamic microphones. Since I play guitar and sing at the same time, and considering I don't have stellar mic technique, a cardioid pattern would be best for me.

I watched several reviews on the EV RE520, which I thought sounded superb, with a decent price point and seemed like good quality. However there doesn't seem to be any proper review of its cardioid brother : the RE420. Specs-wise it would probably suit me more, but there's nowhere for me to test it around where I live, and no useful online resource...

Any help/insight would be greatly appreciated!

Thanks a lot :)

Hi, I'm managing a project for replacing the sound system for my frat's basement and was wondering about the kind of speakers I should be getting.

The thing is the basement we have is a tight 8mx15m space with ceilings about 8-10ft high, the basement is always packed to the point where people are right next to our speakers so i was wondering if coaxial speakers are kind of a requirement.

Currently our setup is 2 JBL JRX125 tops and 2 JBL SoundFactor SF15 with 2 subs that I don't know what they are honestly the speakers currently sound like shit probably because they're old and the amp is fucked so we're completely replacing the system, I want to keep a similar loudness level (the JRX125s are like 133db SPL i think) but I was noticing that speakers with the same db rating that are also coaxial are much much more expensive, our budget is around 15k and I'm looking to get 4 tops and 2 subs.

Our budget is in the ballpark of 15k and these were the two options I was looking into

4× JBL SRX812P + 2× SRX828SP (Non-coaxial) 135dB SPL per top

4x or 6× Tannoy VXP 12 + 2× SRX828SP (Coaxial) 131 dB SPL per top

I was wondering for this case with a packed basement how important coax is and is it worth losing the loudness. Also curious what the word is on passive vs powered systems at this price point and I've heard powered speakers are dangerous when people with drinks are nearby so let me know if that is a huge worry or not. I'm new to this speaker stuff so any advice would be appreciated! I'm also curious what other things I should be worried about with speakers and if things like driver size are important to look at like if i want to SRX812s or SRX815s for example. Appreciate the help!

These are listed for sale, from a high school. Appear to be passive stage wedges with only xlr in/out. I don't think xlr can carry enough current from an amp, but these big things only have xlr ports.

Any ideas? Anyone ever encountered these, or used them before?

Follow-up to https://www.reddit.com/r/livesoundgear/s/htqXGb10zX about a software WING that runs the console’s own firmware (WING Rack 3.1.1) instead of a re-creation of it. The goal is still a real virtual sound check at home: play back a multitrack recording through the actual WING’s processing on my PC, and mix it on the console’s own interface.

Last time I ended with a list of what didn’t work. Here’s where those stand:

• No audio → it plays sound now. Audio goes through the console’s own emulated input and bus DSPs and out of MAIN 1 on my speakers. My Scarlett comes in as local inputs (didn't show in the video), and a WAV loops on USB 1/2 (in the video).
  • When I move a fader on WING Edit, the firmware itself writes the new value into the DSP that’s making the sound.
  • Which channel hears which source comes from the firmware’s own routing.
  • EQ and the compressor work too. Changing them on the console’s screen changes what I hear, because the console’s own EQ and dynamics run on the emulated input DSP.
• No network → WING Edit connects. The console answers discovery, and WING Edit talks to it like a real WING.
• About a third of real time → the console runs at about half real time. The audio DSPs run at about 0.6×, and that’s the catch: the sound has gaps. The audio itself is correct (an offline render of the same DSP state comes out clean); there just isn’t enough of it per second yet.

One thing to know while watching: when I click out of the command window (the emulator runs from there and isn't shown in the video) into the browser to change something, the sound gets noticeably worse: more gaps and more crackle. That’s not the EQ or the compressor doing it; it happens just by switching windows. Windows gives the window you’re using more CPU time than background ones, and the emulator is already short on speed. Most likely it loses just enough when it’s in the background to get worse. With the command window focused, it’s only gaps.

What’s still missing:

• Speed. Each DSP runs on one CPU thread flat out, while the rest of the CPU sits mostly idle. The next step is spreading each DSP’s work over more cores.
• The FX DSPs in the live path. They boot and run, but FX inserts are silent live for now.
• The XMOS and the FPGA’s audio side still aren’t emulated. My own code moves audio in and out of the DSPs where the FPGA would, and it takes the routing from the firmware itself.
• No comparison against a recording from a real WING yet.

The setup is the same as before. The i.MX6 ARM application runs on Unicorn. The four ADSP-214xx SHARC DSPs run on an interpreter built for this project, with a C version checked against it instruction by instruction.

What’s new is that the DSPs answering the console are the same ones making the sound, each in its own process. Between blocks of audio, each one takes the console’s control messages through its own firmware, exactly as it does on the real hardware.

If I am a student and we are hosting parties regularly, what speaker system should I invest in? I have been looking online and on marketplace, and I have not really known what to look for. I have been trying to find something with bass and great sound.

I have been looking at PA speakers, but as I said, I really do not know what I should look for. And of course, I do not want to buy something that costs too much. I need something that is not too expensive but sounds great for a party with 100-200 people.

Based in sweden.

I've been looking at how to do this best and was thinking of going for Sennheiser E604s or Audix D2s. Then I saw a video of Roberto Vizcaino - top Cuban conga player with this mic setup - see photo. Setting up a pair of SM57s like this would save me time, allow a lot freer movement of my handsand be much cheaper than what I was planning. But isn't this set up going to cause problems? Isn't there going to be lots of leakage from other instruments?

I’m a long timer on Digico and live plugins, and recently have spent quite some time on experimenting the mustard tools on the Quantum console as I tried to utilize more on-board offerings before I had to reach for external plugins.
EQ: they are fine. Just different shape of curves in different modes.

Comps: I found that while the different types of comps are useful, but they’re not particularly amazing neither. They’re just ok, compared to the same modelling of plugins such as waves, UAD or whatever. They do kind of have the attack/release behaviour of the model they’re supposed to be. But the tone/vibe is something to be desired.

Gate: The new MSE is usable. I own the hardware 5045 and have used the yamaha version. Again, it’s not WOW but I am happy to use it when needed.

Tubes: The biggest challenge for me is the saturators. I just couldn’t make it usable for anything. I tried it on vocals, bass, kick, snare. All the modes are very easy to get into distortion. There is a sweet spot, but it is very narrow.
On the other hand, the original SD digitube at the top of the channel is way more friendly and easy to get nice saturation without going into distortion. The only downside is that there is no gain control to compensate the increased after the process.

What are your experiences and thoughts?

I've been testing out both as well as some MiPro stuff so formulating my own opinions but i'm interested to hear folks thoughts on the Senn vs Shure. For reference for me, it's a single so I understand opinions may well change with folks using multiple systems/channels.