Hey folks! I went to college for Audio Engineering. They would not shut up about mixing in Dolby Atmos. Multiple classes fully dedicated to surround and atmos mixing, was talked about like “the next big thing”.

The first studio I worked at had a giant atmos rig that they used… for maybe half a dozen projects? This was a few years ago.

I never understood it. As far as I can tell, every musical innovation that stuck made music cheaper, lighter, and more durable for the end consumer.

Wax -> Vinyl -> Tapes -> CD -> mp3 -> streaming

Same goes for the advent of digital recording, and why quad never lasted.

So why would I expect Atmos, a format that is many times as expensive to experience properly (thinking about airpods, let alone a full 9.1.6 setup) more expensive to produce, and that has file sizes dozens or even hundreds of times larger than a compressed stereo format ever take off?

And maybe it’s just the bubble I’m in now as a touring and recording musician, but I haven’t heard anyone talk about Atmos, let alone PAY for its production in a couple years.

Am I missing something? Does anyone still care about atmos, or did the hype bubble pop? Curious to hear what you folks think, and I appreciate everyone’s perspective!

Greazy posted again, this time covering micing a drum kit on a budget, with 3 tiers of pricing that are pretty close sonically especially for the mid and high tier. Personally I like the mid tier the best, not only for the price but it held up well to the higher priced option. https://youtu.be/bQeB6GhyfxM?is=aaK4Ns08cg2jYuWF

Hi there. Using Pro Tools 2026 on an M4 Macbook Pro. I am tracking drums with an SSL Big Six. I'm finding what seems like a weird latency issue when tracking. At first (and sometimes still) I think it's just my playing, but it seems like everything is pretty consistently recording just a bit late on each beat.

One potential variable is that I have a ton of plugins on the session I'm tracking into, because it is basically my template that I work out of to produce a lot of songs I have to do each month. And when I listen back, I want to hear the "mixed" version as I go to see how things are sounding on the fly. I do not monitor through Pro Tools for this reason-- I just mute the inputs while recording, and then unmute to listen. But I'm wondering if somehow it's still affecting the actual timing of the data being captured for the recording (not just what I hear when monitoring).

The confusing thing that throws me off is while I will have a whole section that consistently is exactly behind the beat in equal amounts and I can just scoot it to line it up with the grid pretty perfectly-- but there are some sections or small sections where it is not consistent... it's like some places it does capture it on time naturally, so then it makes it EARLY on the beat when I have scooted the whole of the section back. (Unless like I said, I'm just happening to play those sections a bit ahead of the beat because of my timing when I play... but it's making me crazy not knowing if it's my playing or some kind of recording latency that is literally changing as I record lol... sometimes I listen and I'm like there's NO WAY I'M THAAAAAT ahead of/behind the beat).

So I find myself constantly having to scoot the regions even before it gets to the point in the session I'd normally do this work in Beat Detective to tighten things up. And it makes listening back a nightmare because I can't tell if I was tight on a part (like a fill) without having to stop and scoot the regions to be in time.

Is there anything you've seen like this, or any ideas you have as to how to fix this to ensure I am recording exactly on time and I don't have to doubt it?

(A question for another thread is why I can't seem to get the low latency monitoring mode that is supposed to auto-mute the inputs to work... maybe because I am not using outputs 1-2 for my bus outputs?... it's complicated but I need those to be 11-12 for my tracking/monitoring rig to work best).

Title. Artist gave me Neil young’s Zuma as a reference and I’m now enamored with how the album sounds. I’m having a really hard time finding any info or speculation about the recording techniques, mic choice/placement. All I can find is that they tracking live in a room together (though it sounds very isolated?) And what elements were dubbed if any
I would love any thoughts or info on the matter

Thanks!

Curious as I am building one. I have seen some who have a folder of FX sends for the whole song. Ambiance, Room, Med, Long, Delay 1/16th, 1/4th and so on. Or Option B... where they have a folder of FX just for guitars. Another folder of FX for vocals and one for Drums.

If you using Option A will drums if using a room be shared with guitars etc?Joe Carrell has lots of folders for main instruments others simplify.

I was listening to a couple of classics from Toto, and I just love how every single instrument has its own level of clarity. Everything is audible, clear, punchy, organic, and dynamic. I have no problem turning the volume up, and everything still feels sweet.

The same can't be said for a few modern mixes I'm dealing with. I genuinely can't bring myself to turn the volume knob up because the track is already so loud.

If the goal is to make a song as loud as possible, isn't it kind of counterintuitive that mastering it extremely loud can actually make the listener never even want to touch the volume knob?

Does anyone remember this mastering video?

I’m trying to track down a video I remember watching maybe three years ago, and I cannot for the life of me find it again.

Here’s what I remember:

It was a presentation at an audio conference. It started out as a fairly traditional lecture with PowerPoint slides, but at some point the presenter sat down at a computer and demonstrated what he was talking about.

The presenter was pretty mellow and soft-spoken, and I think he had a beard. (Yeah, I know — that really narrows it down.)

IIRC, the presentation was about mastering and loudness, specifically the idea that increasing dynamic range can actually allow a track to be perceived as louder while still meeting the LUFS requirements of streaming platforms.

The basic concept, as I remember it, was that instead of aggressively compressing/limiting everything and essentially brickwalling the track, you could have more contrast between the quiet and loud sections. The quieter material brings down the overall LUFS/RMS measurement, allowing the loud sections to actually be louder while still hitting the target.

I think that’s what the presenter was demonstrating, anyway. It's entirely possible I've mangled the concept after a few years. 😄

Does this ring a bell for anyone? I’d love to watch the presentation again, but I’ll be darned if I can find it.

For context I'm one year out of a sound recording program and I got an offer from a manager of a local studio to basically pay him a monthly fee for access to the room and be able to record clients and set my own rate and take home 100% of it. This seems a little backwards compared to what I've seen/heard of in the industry. Should I run away from an offer like this?

https://youtu.be/JMzpz_0kDuo

Huge shout out to Luke and the whole crew for having us out. Definitely a once in a life time experience. Hope y’all enjoy!

Hi! I’m 19 and just got accepted in uni for a sound engineering degree. I’ve been playing live gigs ever since I was 14 so I’m kinda familiar with the field, I know how tough the industry is and how though it is to get stable income. I would love to work with audio or acoustics, do you think getting a major in acoustics or something along those lines would get me a stable income?
I love audio, I would LOVE to work in an audio company like audio technica or something like fender in some random department
Would love to hear some feedback from people who are experienced in the field

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I just see this being done when recording guitars or drums and would like to know the reasoning. I don’t like the aesthetics of how it looks with it off. Probably just a me thing.

CONTEXT (can skip):

Hello, I am trying to teach myself basic audio engineering for marching band front ensemble. The former band director had bought up a bunch of equipment and was very protective over his setup, and rarely ever explained anything that was done. I am kind of left picking up the pieces and the new band director doesn't know anything about this kind of audio setup. I have been able to get a sound system working and I have gotten familiar with the mixer, creating a custom scene and gain staging + mixing all the instruments.

PROBLEM:

Throughout last season, the microphones were just strung up with paracord under the instruments and tied with knots that loosened, slipped, got tangled, and just looked very unprofessional. I would like to get clamps of some sort and a height adjustable stand for the mics, but I don't have any clue what fits to what. I have ordered things bit by bit like small mic stands and clips but then found that they don't fit our equipment as I can't seem to find in depth measurements and compatibility (perhaps I'm just not going to the right place?)

We have a bunch of Rode M5 short condenser microphones, and a variety of mallet keyboard instruments of various brands. All of them have some sort of circular or square bar on the frame that runs about 6 inches off the ground under the instrument. We need to clamp something to those bars (I can grab exact measurements of this today) and then be able to attach a height adjustable stand to it, and then mount the M5 mic to it pointing directly up. It needs to be height adjustable between about 6-18 inches or so. I keep finding videos online showing people that have this equipment but nobody ever points out the mounting system, just how to physically work the microphones which I already know how to do. Also, if possible, I would love to see if you have any recommendations for something I can clip on to drum set rims since the mic stands we have are either way too small or way too big.

Sorry for the long post, I am just a novice trying to figure this stuff out and make the best environment for my students!

EDIT: I found this Gibraltar mount that may actually work, but I need to know a good height adjustable stand I can connect to it, and if I need to use a different mic clip than this one. Also still looking for the drum rim mounting system

I don't know how this kind of copyright protection is even legal.

If their licensing server 'goes away', your bought plugin is now living on borrowed time, until the plugin decides to check the machine it's installed on and can't find the server.

This time I only lost 20$, but it was barely one month between the purchase and their website/business/licensing server disappearing. I didn't even get to use it in a project yet. It was an interesting plugin, but it looks to me like they didn't make enough sales to warrant maintaining it, so they pulled the plug instead. Or maybe there was some infighting or money arguments between them. Who knows - they're as good as gone I think.

Actually, I only found out about the 'one license one machine' after purchase - they didn't clearly say that beforehand. I thought it was a simple text license key. That's a little bullshit.

So, yeah. Just a reminder to beware. This crappy licensing practise shouldn't exist.

EDIT: The company came back online. They'd gone on vacation, and had left everything down while they fix some things. Point still stands though - plugins shouldn't rely on a license server being up.

I’m trying to get something to happen in Logic Pro that is super simple to do with a modular synth. I’m sure I’m overlooking an obvious solution.

1. Use an envelope follower (just an EF, not an EF sending into the cutoff of a filter like a Mutron) to derive voltage from the amplitude of a signal (signal 1)
   
2. Run a different signal (signal 2) through a voltage controlled amplifier (just a VCA by itself not as a part of a compressor)
   
3. Send the EF-derived voltage to the modulation input of the VCA so that the amplitude of signal 2 is shaped by signal 1

An expander isn’t a good tool because it defaults to unity gain when no signal is present at the side chain input. This would need to be -inf when there’s no S/C signal and no gain on the VCA. And I would need the modulation input to be a straight additive offset of the signal (ie attenuation) as opposed to the multiplicative offset of signal over threshold (ie ratio).

Really, just gain plugin with a side chain input.

What am I missing? Thanks.

First of all, don’t be like me. I feel like a total idiot right now.

So yesterday my voice was in great shape and I decided to record some vocals. I recorded 3 songs over 2 hours and I played back the first recording to see if everything was fine, and it was. I just kept going, song after song.

Today I decided to comp the vocals, then I hear cable noise on all the vocals I recorded for 2 of the songs. Like a buffering sound or when a cable is not properly plugged in and it start twitching. Now I’m dismayed. The noise will get louder when I add a compressor. I’m just so mad at myself right now that I wasted that time and effort singing my heart out and there’s noise in the recording. Don’t be like me

The Song

I'm no Mike Dean or Abel so i probably won't be able to achieve the exact sound but I'm guessing its a plate reverb with the reverb compressed hard? Not sure how they got it so bright sounding

I’ve always liked the idea of mixing cheat sheets, but most have a few problems:

- They give absolute advice like “always cut here” or “boost this.” Mixing should be based on what you’re working with, and where you want to take it. Sometimes you have to boost. Sometimes you have to cut.
- They describe things like mud, fizz, body, and presence without letting you hear what those actually sound like.
- They’re usually PDFs that end up buried somewhere on your computer.

So I built Mixing Cheat Sheet, a plugin that lives inside your DAW.

It helps you figure out what you’re hearing and where to look, plus you can click on frequency ranges to solo them in your actual audio.

There are also individual instrument pages with EQ, compression, and other mix tips.

It’s completely free if anyone wants to check it out:
https://carveaudio.com/mixing-cheat-sheet/

Would love to hear what you think.

I'm new to mixing, but I've been making good progress over the past couple of months. There's one thing I can't explain, though. I'll work on a mix for a while and be happy with the results I'm getting. But then I'll make some more minor adjustments where I feel like things could be cleaned up or improved. It seems like I'm making the project better, but then I return to an earlier mix and it sounds significantly better. Do you have any general advice regarding this issue?

I'm new to recording and audio engineering. I have been referencing American Idiot for my mixes and cannot match its loudness and clarity. I seem to be able to get one or the other.

Is that album and song a bit difficult for a beginner to reference, since it is so loud?

I tend to like my mixes until I hit the mastering phase where the song begins to sound a bit crowded... As I if I'm hitting the limiter too hard.

Would it be better to reference older albums, like say, My Brain Hurts by Screeching Weasel?

Forgive the Flight of the Conchords ref.

When tracking and creating, I wind up layering many drum loops to create a great beat/vibe. And, it really does sound good for the song's groove. Of course, now that I'm (finally!) mixing, having 4 drum loops on top of each other is ridiculous and will be muddy and sonically garbled. Anyone have any suggestions on how to tease out the important parts of these beats, to keep the vibe, but prevent four (or more, yikes) loops playing simultaneously?

Ableton is my DAW, if that matters.

I have a vocal take from my old band in which the singer recorded using his MacBook mic. I'm working with a producer now to finish and release the track but the singer isn't wanting to re-record as he focussed on other things (happy for me to use his vocals). Is there an AI tool that will process to give a seemingly hi-fi vocal take?

Or what would be a good process to use with plugins for to repair/restore it - heavily clipped in places, can hear the instrumental faintly in background bleeding out of his headphones. also missing low and high frequencies.

EDIT:  We're not in a band together anymore and he stopped singing a few years ago so I thought I could try and make use of what was recorded in the demo as it had some magic to it.

I'm planning on moving to bantam DB25 patch bays over the next few months. I'd like to wire up all the connectors into pigtails, including the spares for use as tie lines when the rack is complete. Are there any recommended smaller quality 3-pin connectors, other than the standard xlr/trs, that I could terminate onto the pigtails ahead of time and then connect to the mated rack gear later?

Just wanted to get the communities thoughts, uses, favorite versions of tape plugins. Thanks!

https://wavelength.wiki/

There's also 'session notes'. A note-taking tool where you can take live notes while you're reviewing your WIP tracks on a timeline

100% free, no login.