I’ve been rethinking my habit of automatically applying high-pass filters to almost every track.
Lately, I’ve been experimenting with low shelves instead, reducing the low frequencies rather than cutting them out entirely.
My impression is that this can preserve more warmth, body, and a natural sense of depth, especially when working with acoustic instruments.
Of course, high-pass filters are great for removing unwanted rumble and cleaning up the low end.
But I’m curious about your approach:
When do you reach for a high-pass filter, and when do you prefer a low shelf?
Do you find that using fewer high-pass filters makes your mixes sound fuller and more organic, or does it just create unnecessary low-end buildup?

Are there any wives of audio engineers that have like hobbies or tips on how to handle your husband working out of home late at night at the music studio doing Audio engineering and recording? My husband works at one of the best studios in a large city and his clients are in the hip hop/rap industry but most sessions are late at night .. 7-3 am or 9-4 am or even an all nighter till 7 if it’s a super famous artist. Just looking for support or tips bc I have a really hard time sleeping when he is working at night and it’s very often.

I downloaded and tried out the Mercurial Tones plugin .. it seems pretty good, has some nice features. Before I went ahead and purchased I searched the web and forums to see if there's any reviews or background on the company itself and there is literally NOTHING, which makes me a bit nervous.

I don't want to buy something that's not going to be supported in a year or is vibe coded so wanted to see if anyone else is using the plugin or has experience with their stuff?

I'm getting into recording music, and I've realized that the monitoring signal I hear when tracking myself sounds very different than the same signal when played back.

For context, I'm using a Zoom L-20 digital mixer with Vic Firth SIH3 headphones on one of the auxilary outs.

For example, when I'm all plugged in, headphones on, in tracking mode, if I have a mic on my acoustic guitar, and I play and record it, I totally hear the signal of the microphone, and it sounds nice! Only problem is, when I play that take back through those same headphones, it sounds so much cleaner and better.

I thought it might be the fact that when I'm tracking, I'm not only hearing the mic signal go through the headphones, but also the vibrations of the guitar, but even when listening back to a take, and playing along identically with the microphone muted, the playback signal is still so much smoother and clearer.

Given this,

1. Is it realistic to want a monitoring signal similar to the playback signal? Or is this something that everyone deals with.

2. How does this affect learning microphone placements? What's the point in experimenting with mic placements while tracking if that's not even what the playback signal sounds like?

Yes, monitoring is off on my daw. I'm not getting double signals anywhere, as far as I'm aware. There is also little to no headphone bleed at all that could be going to the mic. These headphones grip tight.

If anyone can help, I would be so thankful. It would make everything a lot easier if what I heard in my headphones when tracking is what my playback signal sounds like.

Thanks!

I'm restoring a 2010 Mac Pro (5,1) as a studio machine on macOS Mojave, and my audio interface is a Mackie Onyx Blackbird. It worked on this machine before, but I wiped the drive and my copy of the driver installer is gone — and Mackie has pulled the downloads, so the official links are dead.

The specific file I need is the Onyx Blackbird Driver V4.1.0 + Control Panel V1.1.0 for Mac (.dmg). Note it's the Blackbird-specific package — the Onyx-i driver with the same version number doesn't install the Blackbird Control panel (already tried).

If anyone has this .dmg archived anywhere, a link would genuinely save this interface. Happy to mirror it afterward so the next person hunting for it finds it faster than I did.

In college, my first ever experience with audio engineering, I chose to record a jazz band live for my final. 10 piece drum mic set, every instrument mic'd, all in my college's recording studio. And when I finally ended the 3 hour recording session...

I'd forgotten phantom power on the drum overheads. Needless to say, my professor was not happy with me, lol.

Ever since, I have never forgotten to put phantom power on my condensers. Has anyone else made a mistake that they made sure never to repeat because of how bad it was?

I've been mixing and mastering my own music (mostly electronic/hyperpop type stuff) for a few months (or at least trying to), and when I have a mix/master I like, I put it in Audacity and use the Loudness Normalization effect to see/hear what my audio sounds like at -14 LUFS for Spotify and such.

Now I know I should use my ears, and I do, but when I compare my tracks to that of others, I notice that when I put in other tracks I like, downloaded either from Youtube or Bandcamp, I use the same loudness normalization effect on them so as to compare what my track sounds like after being normalized vs theirs.

There is an audible difference that's a little harder to grasp/describe, but more noticeably, I'll see that my audio is kind of a flat square in a lot of parts, whereas there's, even if it looked AND SOUNDED similar to mine before both being normalized, will kind of have more of this fuzzier/hairier look to the waveform. And I typically will end up liking how the reference track sounds a LOT more than how my track, even considering my inevitable bias. With this information, I've concluded that my track's waveforms should probably *look* more like that in order to *sound* more like that. And duh yes I'm gonna use my ears, but I'd think myself a fool not to use my other senses as well.

How do I make it so my audio has more fuzz to it after being normalized to -14 LUFS, instead of being a flat square almost all throughout?

For further clarification if anyone's curious, I was comparing my track to Eyes Wide Closed by 2hollis.

I’ve been making music for about 5-6 years now, I’m 25 years old. I record, mix, and master all my own music from scratch and I feel like I’m decent at it. But I just wanted to see if anyone had any extra tips for creating this kind of music that I might find helpful. Thanks!

In a live take. I recorded a singer song writer today, and after trying many different mics and mic placements. I couldn't get his vocals louder than his guitar. More quiet singer, with a heavy rhythm hand. I tried a ksm32 on his vocal, and a sm57/lewitt240 on guitar. Guitar is too hot on vocal mic.

No success separating the guitar and vocals. Any tips and tricks?

I feel like I probably should have just coached him to play softer and sing louder.

Hi all,

I've created an inexpensive app for routing audio around macOS and, unlike similar apps, that can host VST3 and AU plugins. It can be downloaded (2 week trial) from www.segedunumaudio.co.uk

You can route any input to any output, choose whether to put a plugin in the middle and also use advanced mode, where you can add in a simple mixer (eg for live streaming). One of my consistent bug bears for online meetings is that I can't EQ/Compress my microphone in software very easily... So now I can easily mix feeds from my computer into calls/streams with my microphone and have it all run through an SSL Bus compressor 👌

Constructive feedback welcome.

Full disclosure: Agentic coding was used in addition to my own programming skills to create this.

Hi, Im 17 and have recorded a couple songs on my own and shared them with a few friends. Last night, one of those friends reached out and asked if i could help her band record. I have no experience recording actual amps or drums. I usually just use amp sims and a drum VST. I want to help them, and i’m more than sure i could do it to some extent, but i would like some tips on what i should look into and work on. I started cramming videos last night to open up my mind, but any help works!!!!!

Hello so I have an assignment with reaper and Youlean which is to have a -14 LUFS with a 0 true peak max but in Youlean I cannot get it to do both, it's either a -17 LUFS with 2 true peak or the contrary. I don't know what to do😥

Hi beautiful people!! I hope I'm not in the wrong sub, otherwise sorry, I will delete my post.

I would like to record some readings, some books in particular. I would love to create some audiobooks just for fun and for myself, so I bought a budget microphone on Amazon. The exact model is the TONOR Q9. It is a USB condenser microphone with a 16 mm diaphragm and a cardioid pattern. It comes with a complete kit including an extended boom arm, metal shock mount, desk clamp, pop filter, and a 1.8 m USB cable.

​The supported sampling rate is 44.1 kHz - 48 kHz, with a frequency range from 20 Hz to 20 kHz.

I would like to ask what software you suggest I can use on my laptop (not necessarily a free software), and some recording tips, if you have some.

Maybe if you have some guide I can use to record my voice, just because I don't know anything about settings etc.

Thank you in advance!!

Title pretty much says it all - Editing a skate video and I got a high res .wav straight from the producer. It sounds good on all of my devices, but when I play it on my friends old tv it sounds horrible. I would really like it to sound decent on all devices. Any tips? DM me if you want a link to the video in question!

hi, so I need advise on mixing. I'm 18, started producing pop at 11 and I can make pro productions and mix my vocals well but I just started learning mixing this year

when listening to my production it pumps on speakers but not on headphones, phone speakers or earphones

(I'm not even using a limiter or clipping past 0db but it's pumping and when kick and 808 bass stop, the volume of track significantly rises)

... at first I thought it was sharp transients, but even after clipping, it still pumped and I later realized the main issue in most of my mixes was heavy low end

I watched alot of tutorials, some saying to set kick to -12 db then layering everything around it (ofc in context of whole mix) but what i know is just setting the kick level doesn't magically fix the issue as you might end up having mids too low in comparison to low end thus the pumping on speakers would still happen

I've started using metric AB and voxengo Span plus a reference track of my track's genre and vibe which helps with low end balance... But I atleast need to know how y'all get a production that's balanced and no heavy pump even before mixing (that is having kick and 808 bass not too loud than mids)

cause it's my main problem now

at first i thought his speakers where just old but when i went in the car my song literally sounded like it came through an old pay phone. i went back upstairs to figure out what’s wrong and it turns out the mic was distorted and crackling because the wires weren’t plugged in right. so i can’t even take my raw vocals and bring them to a different engineer for mixing, the song is just cooked. btw out of my 3 hours he mixed for 1 hour and 40 minutes of it. this is a new place that opened up and my first time going. i never had this problem at any other studio before. i never left a session completely unsatisfied by the mix
before. i even asked him to show me raw vocals from a previous client and they were clear as day compared to mine…. am i right to be refunded for this session? i literally got nothing out of it.

This might be slightly off topic so I apologize but to me its relevant.

When I am working on recording to my computer (Logic), I often times have to jump over to web browser to watch a tutorial or listen to a reference track. These sources can be way louder than the track I am working in, and I end up getting blasted. These days I am mostly working on a couple scratch tracks or loops while my kids sleep, so my overall system output volume is much louder compared to when I am simply watching or streaming online.

I understand gain structure and the recording levels on the way in and the output master track seem healthy - its just not nearly as loud as the mastered audio from Spotify web or Youtube video I open.

I was hoping for a *simple slider universal audio output extension for chrome, but all I can find are bloated 'boost' extensions that are more for cranking up youtube videos on quiet speakers.

Perhaps this is more suited for some sort of virtual mixer in my computer... but my dead simple solution was a simple slider extension in chrome that allows you to adjust/limit the overal output from the chrome app.

any ideas?

Title. I’ve had roles focused exclusively on vocals for several years. Worked w people that some would consider the authorities on that kind of thing. The box really does earn its job. If you track through it on the right settings, it does 95% of your work. Not saying you don’t have to perfect, but just like a channel strip does most of your heavy lifting, voxbox does the same w vox, and does such a great job. Recorded some incredible vocals tn w just one additional plugin after voxbox. The results just rock. No paid anything, I just had a great session and that plugin did most of the work, so giving it a shoutout. Really love that unit.

Hi guys, I’m new here and I wanted to ask what you guys think is the hardest thing to learn in audio engineering and why?

I spent three years building my studio and I've been running it for a year now. The biggest thing I've changed my mind about is the business model.

Going in, I thought the goal was a full calendar: lots of different people through the door, record them, send them off, next booking. Catch and release.

What's actually worked is the opposite. Focus on one or two acts at a time, make full albums with them, and stay with them through the release. Instead of only charging for the hours, I take a cut of the record and put real effort into pushing it with them.

It's slower and riskier per project, but the work is better because I'm invested past the final bounce. The artists also get someone who cares whether the record actually lands, not just whether the session ran on time.

Curious how others running rooms think about this. Do you do any back-end or points deals, or development-style relationships with a few acts? Or is hourly/day-rate still the only thing that keeps the lights on for you? What's gone wrong when you've tried it?

I run an archival channel where I post full sets of mostly hardcore/metal/punk bands. I get the multitracks and mix them back home. Cymbal bleed can be a pretty huge problem and I usually manually cut them out of vocal tracks but it's way too time consuming. Is there a good gate/denoiser type plugin that I can rely on for this? Friends recommend me Waves Clarity VX. Haven't tried but I'm looking for more suggestions

I'm a writer and the main character of something I'm working on right now is a sound engineer. She's a freelancer working in the mid-90s for small to mid-level bands––mostly in dive bars––and I'd appreciate help with what exactly her equipment & setup would look like.

First, when a sound engineer is coming in as NOT part of the venue's staff, where in the venue do they set up? The same private booth/area used by the venue's people? If it's a shitty dive bar and an act the venue doesn't really care about (like in this story), would they ever just be thrown off to a random table in the bar area?

Second, what equipment would they bring in that would be time period & low-budget appropriate? From the research I've done, I was thinking a soundboard/mixer, preamp (Adcom? do they need just one or multiple?), cable tester, XLR cables, and various smaller stuff like electrical tape, cable ties, adapters, etc. I've found so many options that I'm not sure what the base necessities are.

Sorry in advance if these are dumb questions and thanks!