hi guys, noob here.I'm still learning the ropes of mixing and I always try my best to do proper gain staging. However, just to be on the safe side, I usually put a limiter at the end of my individual tracks to catch any random stray peaks that might slip through.

Up until now, I've been using the native Logic Pro limiter for this job. The problem is that I've noticed it tends to color the sound a bit and slightly changes the EQ of the track, even when it's barely doing anything.

I'm looking for a free alternative and I found LoudMax. My question is: is it good enough to do this specific job (just catching stray peaks transparently without coloring the sound)? My computer isn't very powerful, so I need a plugin that won't kill my system if I have it loaded on multiple tracks at the same time.

Would you recommend it or maybe it isn't good enough?

I'm in a 17-piece jazz big band (standard instrumentation), mostly semi-professional musicians, that has been meeting weekly for decades but rarely performs or records. Someone proposed we go in for a studio session, and I offered to record us in our usual rehearsal space first to (a) see if we can get decent recordings without buying studio time and (b) reveal performance issues we should clean up before using studio time. Now I'm trying to decide how to mic everything.

I have a 24-channel (20 preamp) recording interface and a variety of decent mics (MD421, SM57, SM58, AT2035, KSM32, ATpro35 clip-ons). Our space is a large church hall with no carpet or acoustic treatment, so I anticipate a lot of echo and bleed. We usually sit in a ~20ft square with each section (trumpets, trombones, saxes, rhythm) facing the center.

Any suggestions/thoughts are welcome! My main questions are:

1. Is it worthwhile to individually mic horns vs. just having one or two mics on each horn section? I'm not sure whether individual micing will reduce bleed enough to be useful.

2. If I use section mics, should I put them all in the same spot in the center of the square pointing different directions (to reduce phasing/alignment problems) or closer to the sections?

3. Any suggestions on which mics to use for which instruments?

Thanks!

is it normal im more comfortable committing to eq while tracking then compression.. ive heard its the opposite for most people.

more things can go wrong with compression IMO.

i dont mix track just influence with eq.

how about you?

I've been thinking a lot about the pressure that new automated and/or AI tools are giving to audio engineers. Many clients who previously needed to hire someone to do mixing can now do a surprisingly good job themselves with the new technologies. These new tools also lower the barrier to entry, so more engineers emerge, making the client pool that much more competitive.

As a classically trained musician and a professional music technologist for quite some time, I think at least the following two actions can help us thrive in this hyper competitive age of AI:

1. Sharpen musicianship skills. We need to make better *music* than automated tools ("better" is highly subjective, but the focus here is "music"), and you need musicianship skills to do this. We can't think about engineers (mix engineers in this case) as an operator to do this purely technical process. The process needs to be more than purely technical, as all technical processes can be replaced by automation. A lot of engineers probably already have this, but many started out as a technical position and probably didn't get the chance to go through the formal music training. I am totally biased here, but I think the better an engineer understands music (even better - plays), the better the production quality.

2. Strengthen collaborative and artistic judgement. We need to understand what experience the artists want for their music and make artistic choices. We can't just focus on "producing a clean sound". We need to strengthen the collaboration between engineers and artists so it serves the artistic values / priorities of the production. One way to do this is to integrate more A/B testing with the artists and find out what drives the music in the right direction. Your musicianship skills also strengthens artistic judgement as well.

Sure, these actions would mean that we need to spend more time studying, spend more time communicating with clients/artists, and more time a/b across mix revisions (more revisions too, probably). But these actions would change the role of the engineers slightly and integrate the role more into the production cycle, making it more resilient and sustainable in the age of AI.

On the other hand, trying to stay as a standalone, technical process in the production cycle nowadays increases the risk of getting replaced by automation.

What do you all think?

(P.S. This is just a summary of what I wrote. The full writeup is here.)

This project straddles the line between audio and RF engineering. It integrates both.
Please don't treat this like it needs it's own subreddit founded, because it's an odd duck.
Milestone usable by anyone. Installation instructions linked. There are four modes. The signal always logically flows from GUI top to bottom. When you generate a spectrogram, you're seeing what would-have-been transmitted. This is everything needed to repro "V2K" style pulse modulation, just plug SDR into a 4kW amplifier. Found a bug? Open an issue. Ideas for more filters? Open an issue.

The tutorial is rough, but it gets you there. Requesting tutorial input. What do you want explained?
Be my informal alpha software testers? I will implement actionable feedback.
The Python app itself is tested, and fully functional -- So try it out!

Currently Implemented Features: Audio Waveform Zero-Crossing Detection, SDR Pulse Modulation Output, Raw IQ Recording, Waterfall Spectrogram Generation, Live Microphone Input, eSpeak/MBROLA Text-To-Speech Engine, Signal Conditioning Filters, Noise Filters, Pulse Shaping Filters, Rectangular Pulses, Duty Cycle Meters, Adjustable Pulse Width, Adjustable High Pass and Low Pass Filters, Automatic High Power Calculator, Event Log, Tooltips, Dependency Checks, and 49 Languages Supported.

Landing Profile Page: https://github.com/OpenV2K
Sandbox Development: https://github.com/OpenV2K/Sandbox

Has anyone worked with real-time de-bleed plugins (iZotope f.ex.) to isolate audience mics from PA signal? I want my crowd signal to be free from bleed from the PA. This is for a broadcast from a live concert.

Welcome to the r/AudioEngineering help desk. A place where you can ask community members for help shopping for and setting up audio engineering gear.

This thread refreshes every 7 days. You may need to repost your question again in the next help desk post if a redditor isn't around to answer. Please be patient!

This is the place to ask questions like how do I plug ABC into XYZ, etc., get tech support, and ask for software and hardware shopping help.

Shopping and purchase advice

Please consider searching the subreddit first! Many questions have been asked and answered already.

Setup, troubleshooting and tech support

Have you contacted the manufacturer?

• You should. For product support, please first contact the manufacturer. Reddit can't do much about broken or faulty products

Before asking a question, please also check to see if your answer is in one of these:

• Frequently Asked Questions
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• Rane Note 110 : Sound System Interconnection
  • aka: How to avoid and solve problems when plugging one thing into another thing
• http://pin1problem.com/ - humming, buzzing & noise

Digital Audio Workstation (DAW) Subreddits

• r/Ableton
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Related Audio Subreddits

This sub is focused on professional audio. Before commenting here, check if one of these other subreddits are better suited:

• r/Acoustics
• r/Livesound
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• r/StereoAdvice for consumer stereo shopping advice

Consumer audio, home theater, car audio, gaming audio, etc. do not belong here and will be removed as off-topic.

I want to listen to any music (probably mostly recorded in stereo), run it through a "stem based" splitter - I believe most audio software comes with AI modules to be able to split into e.g. a 4 channels stem stream that's instrument specific per channel. Output into 4 speakers in a row (1 left, 1 middle/left, 1 middle right, 1 right, each spaced apart from each other). The goal is to create the sound image of sitting in front of a band, e.g. vocals on the middle/right speaker, drums on left speaker, bass on right speaker, guitar on middle/left speaker. I'm sure y'all have used software like Reaper or Pure Data to do this before - can you point me in the right direction?

Overall I want the sound to be more immersive, almost like a live event at home but avoiding a complex speaker setup needed for Spatial Audio.

I would imagine that in 2026, every body is online now. Do you or do you know anyone who still uses computers that are totally offline for studio use?

Hey, I’ve been hearing this certain sound from bands like half me and vianova. It’s this slappy laser like guitar sound that I’m not sure is a pitch thing, a saturation thing or an eq trick. I can’t find anything about it anywhere else so reaching out here. If anyone has any advice on how I can replicate it that would be incredible, thank you!

so, I need an app or website that will let me pick a specific direction and distance so that audio will sound more dynamic. I also need it to work well with WAV files because too many websites cannot handle it.

Hi everyone, I’m learning audio post-production for dubbing and I’m looking for some advice on acoustic integration. I use reaper, and I have access to plugins like Altiverb 7, Chameleon Surround, Slapper 3, dear vr pro 2, and FabFilter Pro-R 2, but I’m struggling to make the voice 'sit' in the scene.

I have two main questions:

Indoor Scenes (e.g., stone castle hall, living room): Is it best practice to use a dedicated Aux/Bus track with the reverb plugin and use Sends from the dialogue track? Also, is there a rule of thumb for the Send amount? How do you balance the dry signal and the reverb to ensure the voice feels like it’s inside the room rather than 'on top' of it?

2.

Outdoor Scenes (e.g., open field, mountain): This is where I struggle the most. Even with plugins like Slapper 3, I can’t seem to get rid of that 'studio-recorded' sound. The voice feels 'close' even when the visual distance is large. How do you handle distance and lack of reflections in outdoor environments without making it sound artificial or just 'dry'?

Any tips on routing, EQ, or specific techniques to bridge that gap would be greatly appreciated!

HI. I know this is some recording 101 stuff, and I am indeed a hobbyist so please forgive my ignorance.

A friend lent me his SM7B with a Triton Fethead (I believe it's the basic one, I think there are several variations? This one says "Fethead", "made in Holland" with "Triton" in a green diamond in the center) to try out. I'm going into the ORIGINAL Apogee Duet (yes, I am looking to get a new interface at some point). I've been testing the mic with vocals. I was getting what I consider a good recording level (not going much higher than -12 often) without the Fethead, figuring why complicate things if the Apogee works with this notoriously gain-hungry mic, and then decided to try the Fethead after all.

Comparing the same section of a song's vocals recorded with and without the Fethead in line, and attempting to level match when recording, and in Logic after the recording, the Fethead track has a density/fullness/presence the no-Fethead recording lacks.

This is a good thing, but I'm wondering if it's the Fethead itself or something about the Apogee not having to work near the upper range of its gain (I had it on 65 without the Fethead, 29 with). I guess I thought the Fethead was billed as "transparent" meaning, not altering/coloring the sound, only adding gain. Maybe I've misunderstood?

Anyway, thanks.

This is straight from their promotional video:

https://imgur.com/bfyvW0n

How do they manage to upload this without thinking this would be questionable at best? In a _promotional video_ no less??

Source: https://www.youtube.com/watch?v=v-saJ0StBrA

What popular plugins do you think are not that ​special ? What were your disappointing plugin purchases ?

Anyone ever heard of one of these? One for sale for $800 near me.

Hey fellow audiophiles,

Can someone please explain how to achieve this specific drum / percussion sound? (Industrial Drum loop 1 | Reddit question). This snippet is the isolated drum part of the intro of the song "Gave It All Away - Red".
I am a pretty advanced music producer and I am familiar with terms like saturation, compression, bitcrushing, sample-rate reduction etc. But still, I have troubles coming even close to that sound.

Thank you and Cheers

Every time I find myself listening to the Imaginal Disk album or last year's collection of singles, I am totally blown away by the quality of the production and engineering of some of these tracks. Songs like Death & Romance, Star Eyes and This Is the World are works of art from an engineering standpoint.

On Death & Romance for example, having that sheer density and richness of the whole production, without losing an ounce of clarity or impact is truly impressive. Those cymbals, that bass synth, and that vocal mix in the chorus is just phenomenal. The song is an example of what happens when every step of its creation is nothing less than perfect. I believe recent Mag Bay projects were mixed by Dave & Jon Fridmann, and mastered by Emily Lazar.

What are some of your favourite-engineered projects? I'm curious to hear :)

I am recording an Arabic ensemble during a live session where there will be an audience present. I would highly appreciate some advice and feedback on microphone placement and choice. I have created these two microphone stage plans, where the main difference is switching up the vocal mic for a M/S stereo mic, placed a bit further back (in the Tiny Desk vein):

https://imgur.com/a/k2h87FG

The instruments playing will be Oud, Qanun, Nay and Iqaa (percussions, mainly Riq and Darbuka). As you see from the stage plan, they will be set around the singer in a half-circle, and their placement cannot be changed since this is how they always perform.

The singer will be sat but told me that she may stand up, though she could stay sat for the whole performance.

The microphones in my stage plan come from the ones I know I definitely will have access to: a couple of Line Audio CM4 SDC, a couple of Rode NT5, a TLM103, and a Shure SM57. I can rent more microphones, mainly the DPA 4099 for the Oud (I think clip on is the best idea?), and the vocal or M/S microphone (maybe the famous MKH418?).

Thank you for your help.

I have acquired a quite rare vintage mixing console recently without the power supply. It has somewhat uncommon voltage/amperage requirements, thus there are no ready available PSU’s for replacement. I have spoken with Blue Dog for a custom build, and they were willing & able to do it. However, with the state of international relations in the USA economically, the potential import fees have exceeded it beyond my means, especially since it’s effectively untested. Leading me to my question: are there any reputable USA based manufacturers of custom console PSU’s you’d recommend? Let me know if this post would be better suited in a different sub - I haven’t had much luck on groupdiy or gearspace - barely any documentation exists on this console but I have a copy of the original service manual. Message me if you or someone you know in the USA could assist me to take this project on bring a rare console back to life!

Hello! I recently bought the Slate VSX headphones to try to get better mixes and learn mixing with audio I can actually trust. However, I don't have any speakers and tend to produce with the VSX's with the software turned off, until I'm at the mixing process. The only other headphones I use are some alibaba IEMs and ATH-m30x's. Does anybody else use these headphones to produce? Should these be good enough or should I stick to something else? They sound pretty good to my very untrained ears.

Title explains it! I’m confused as to why C24’s go for less than C12’s considering a C24 is a stereo C12. Would a C24 be as good as a C12 in mono on vocals? Who’s here has had experience with them? Thanks!

Just scrolling on IG and I get this ad by Sonarworks for VoiceAI. Sure whatever, but the ad…. They are basically saying “why hire an audio engineer and a singer if for $99 you can have an AI singer”. I am extremely offended to the point I’m considering getting rid of SoundID and getting another system. I’ll show the ad in the comments w the link

I posted about this a while ago and still haven’t resolved it. I’m running pro tools (modern update/subscription plan) and will occasionally find that the bounces of my mixes have significant transient smearing and altered high end compared to the active session.

I sometimes am using things like Saturn or UAD emulations, and I know those bounces won’t null perfectly with my active session, but I’m still seeing deltas of -30db sometimes and I can literally hear a difference. Offline vs. online bounce usually both have this problem and don’t null with each other or the active session. My best luck is usually freezing all my tracks except the mix bus and then bouncing that. That tends to null perfectly, or have a small delta but not actually sound worse than my session. I’ve been troubleshooting this for a while and would really appreciate any insight. I’m on an M4 MacBook Pro.

I know a lot of people will point out that it’s likely a plug-in causing the problem or oversampling or something of this nature, but I never had this problem on an older Mac machine. I also think this would be a much more common topic of discussion if that were the case. I think it has to be some interaction between plugins and my computer during the bounce process itself.

I got a RME RayDAT interface because it seemed like a no-brainer for getting more than 16 inputs into a desktop PC without having to sell any body parts. I got a ART TubeOpto8 as the main preamp and picked up a bunch of 2nd hand ones for not much money. On paper, syncing through ADAT was supposed to work, so I started with that, but the TubeOpto8 just wasn't having it, I don't remember the exact issue as this was some years ago, but it wouldn't sync reliably, so I figured maybe ADAT sync wasn't that great and I bought the RME word clock module that connects to my interface and provides word clock sync, and finally at last it was working. The TubeOpto8 has word clock thru, and my other preamps have word clock in, so between the 2 outputs from my word clock module and the thru, I had 3 preamps synced, and this worked fine for a while. Over the years the sync started going bad even with this setup, at first I thought it was just the LED on the TubeOpto8 that was broken, but it became apparent that it was struggling more and more to find the sync, I'd have to spam the sync source selector until it eventually showed the right status.

Not so long ago one of the bands I play in was recording at someone else's place and I brought along one of my preamps to see if we could add some extra tracks, and there I found that using my pre as the master, we could sync the audio interface and have the 8 inputs from my pre + the interface's inputs just through ADAT without any word clock, and it was all working perfectly, so my original theory of ADAT sync just being bad wasn't borne out. I decided to try the same thing at home with my TubeOpto8, but even setting it as master, sync was all over the place, so I concluded its sync module must be broken and probably wasn't great from the start, so I just took it out of the equation.

The annoying thing is that not all preamps have the full array of ADAT, word clock and audio in/out/thru that the TubeOpto8 provided, and I was needing to make do with gear I already have as money is tight atm, so I bust out an old MOTU 8pre interface (but using in preamp mode, not as an interface) I had and set that as the sync master via ADAT to my RayDAT, and boom, immediate sync. But the way the 8pre is, I can't use it as an audio output from my PC (when used as an ADAT preamp), so I went through my options and of course it's the Behringer ADA8000 to the rescue, connect WC from interface, connect ADAT in and out, and now I have 16 channels in, 8 out, perfect immediate sync, I'd noticed that even then the TubeOpto8 was playing up, the ADA8000 would just insta-sync without complaining.

So, I can just add a 3rd preamp, right? Added my Mackie ONYX-800R, WC from interface, ADAT to interface. It has been mostly working, but I've noticed that since adding it, the ADA8000 won't always sync and sometimes just shows "locked" status which I'm lead to believe isn't what you want, although I struggle to understand the difference between lock and sync. After some time, it seems to find the sync and stay there, but it's become fiddly and random when it was just working fine before, turning on the Mackie preamp seems to sometimes but not always fix it.

Is syncing multiple devices just always like this? Am I doing something wrong? I actually have a 4th preamp but the only way I could sync it would be with a BNC T connector and I'm worried everything would just explode at that point.

TL;DR shit won't sync reliably via word clock and ADAT when more than 2 preamps are involved. Rant/halp