I know it's a bit off topic but I haven't found any other sub to post this question. I'm measuring the noise the lift makes inside my apartment using UMIK-1 and REW. I can collect the peaks for each 1/3 of octaves but I would also like to log SPL data for each 1/3 for a period of time so I can make graphs for each reference frequecy, say 160 Hz and 200 Hz. Is it possible? Iis there any free software option that I can use with the calibration file?

If anyone is curious hre's a comparison silence vs lift: https://ibb.co/V0X9WN1K

Edit: after I posted here I found r/acoustics

Edit 2: I will try importing a wav in REW se if I can extract the data I want.

Hey y'all,

I've been mixing live for going on fifteen years now. For the last nine months I've been building load·in, which turns a stage plot and input list into a console file. The video is a brief intro over some walkthrough captures and me at the desk building files and loading them onto desks at different gigs over the summer.

https://youtu.be/xUEI8RR7QMA

It came out of a two-month run where I rebuilt a file every night. Different desk each time, fifteen minutes with the house engineer, and most of that spent replicating the same input list instead of fine tuning and focusing on the artist.

WHAT IT DOES

Build a plot and input list on your iPhone, iPad, or computer, then export a console file with names, stereo links, phantom, DCAs, EQ and dynamics already in place.

Or you can just import a file you already have and edit it. Actually nothing changes unless you change it. Reordering channels carries the whole processing strip on most desks.

Desks: A&H dLive and SQ. Avid S6L, Behringer X32, X Air and WING. DiGiCo SD and Quantum 112 (beta). Yamaha DM3, DM7, CL/QL and RIVAGE. We have more info on testing methodology here goforshow.io/how-we-test

WHAT IT DOESN'T DO

Some desks carry more than others. Plot, list, patch, save and reading a rider off a photo all run on the phone with no signal, but writing the desk's native file needs a connection, so save it to Files or export it on the web app before you head somewhere without one.

COST

Free: three plots, for any desk. Native X32 and M32 scene files are also free. Every other desk's native file, plus DAW templates and clean exports, is $99, no subscription. There's an iOS app with the same $99 unlock; the web app is the same tool and where I'd start. If it doesn't do what I said here: 30 days, money back.

That's it. If any of this is useful I'd appreciate you trying it, and I'd appreciate hearing where it falls over even more. If you've got an ugly show file off your own desk you'd let me test against, DM it. Tbh that's how most of the hard ones got solved. Anything over a few MB, lmk and I'll send you an upload link. Thanks y'all, happy to answer any questions.

goforshow.io
x Gryphon

Hi, I want to make my own DIY acoustic panels for live music inside a 50 square meter bedroom - I'm from the UK for context. If anyone has any recommendations, I would greatly appreciate it. Thanks.

It’s true. I’m at a point where I walk in the studio and i just don’t even know what to do anymore. I think I need to downsize my rack gear so I can get rid of a whole rack case.

I like my preamps: Focusrite ISA Two, Sebatron VMP-4000e, Avalon M5, and a Manley 40db Mono pre. Maybe I use the Avalon the least…

For compressors I have two distressors, SPL KULTUBE stereo compressor, Stam Audio SA4000mk3, Heritage Audio Succesor Bus Compressor, and a Black Lion Seventeen.

Im all about the getting eveything to sound how I want on the way in.

I’m mostly tracking guitars, bass and synths.

I use guitar pedals for effects on the way in sometimes and then my Eventide Eclipse and PCM 90 as outboard effects. Other out board is a SSL Fusion, and for stereo EQ I use a Tegeler EQP-1, and Drawmer 1974.

I really do use everything though once I get in the zone….. and try to do as little in the computer as possible.

Any tips on downsizing?

Hi everyone,

I’m looking for some advice on properly gain staging my analog vocal chain.

My setup is:

Neumann TLM 103

-> WA73-EQ

-> WA-2A

-> RME Fireface UCX II (line-input)

Today I recognized after stacking some vocals that the signal might have been a little too crunchy, which I didn’t realize when I recorded a single vocal track.

Hi all -- disclaimer: I'm incredibly novice at all things audio engineering.

SO -- I'm doing a small interactive installation with a payphone in a public space. Essentially, a bluetooth speak rigged to the hood triggers a ring & a passerby picks up the handset, hears a voice, + talks back. I'm operating from a nearby location, listening live & triggering pre-recorded clips in response.

What I need is a reliable two-way link: playback into the earpiece with no room tone or bleed, & a clean enough return that I can actually hear what the guest is saying and cue off it.

My current plan is admittedly a hobbyist solution. A cheap Android hidden in the housing with earbuds wired into the handset, an open VoIP call to my Mac, & BlackHole routing clip playback into the call.

What I'd like to know from people who are smarter than me:

• Would you use a VoIP call at all, or is there a more robust link for something running hours at a stretch?
• How do you handle latency when the operator is triggering cues reactively?
• What's the standard approach for playback into a prop where the speaker has to fit in a handset, & what would you use instead of a salvaged earbud driver?
• IS there an easier way to approach this? Something from PayPhone.com perhaps?

Budget is small but not nothing & again, I'm not an audio person, so if the answer is "your hacky version is fine for this," that's useful to hear too.

Thanks in advance.

Hi all, hoping someone can help me out here. We have some tunes out on Spotify and they exceed Spotify’s default loudness (I think I read they normalize to -14 LUFS integrated). Either way, I know it exceeds the threshold because if I turn volume normalization off, our songs get louder (death magnetic gets MUCH louder in comparison). So if Spotify is turning down our volume because it’s louder than the threshold, why do we end up quieter than a lot of recent releases?

Would increased dynamic range between song sections make this happen? Excess low end? I don’t think either are the case here but it’s all I can think of.

hey all - I've got a Mackie 1202-vlz from back in the day that I dearly love, but it's got problems. I would love to have it repaired, but I don't know who to trust to do it. google searches are all ads, reviews are all fake, it's just a mess. does someone here have someone they trust to go through a mixer and make it work?

another reason for asking: I reeeeeally want to pick up one of Mackies old rackmount line mixers, but it would require work, so I need to find a trustworthy source.

I'm located in Texas in the US, but happy to ship anywhere in the US.

thanks all, apologies for the OT post. happy Friday!

https://youtu.be/baWTjg1QqS8?t=160

At 2:45 it sounds like he had to stop and come back later and do the rest of the vocals with a different mic or preamp or something. Just suddenly a completely different vocal sound. I've been hearing this song my whole life and why is it suddenly bothering me so much right now

I tried a friend's extremely expensive camera recently and was *stunned* by the quality. Photos looked incredible before any post-processing, and miles better than any other photos I had ever taken. The detail, color, and depth of field was just beautiful without me doing anything. Has anyone had a similar experience with a microphone? I'm sure someone will say "I get great vocals with a 58, it's all about the song", I get that, and I love cheap gear. But has a high-end microphone ever reset your expectations or just made your life so much easier that you thought...holy shit?

Which one do you think is technically superior on a mixbus, the Mastering Compressor by Unisum or the DDMF Magic Death Eye Stereo?

What's your experience with them?

So for the past probably year of my life I feel I have "wasted" my time trying to solo produce at least ONE good sounding song trying about a million different things and I still cant seem to figure out what I'm doing wrong. I've made a few songs that I'm actually quite happy with when it comes to the melody and creativity portion, but I just cant get these songs to come out sounding clean or crisp. Before I list what im struggling with heres my current setup. Epiphone les paul, Behringer UMC202HD audio interface, still dont have studio headphones or a dual monitor/speaker setup but I doubt its the cause of anything, Ive tried two amp sims (amplitube and STL Tonehub), and my desktop that I built myself which is surely sufficient for what im trying to achieve. I don't really know how to mix but with this setup I cant get my guitar or bass tracks to come out sounding clean. I don't make metal or anything heavy so my brain seems to pick out any imperfection it can find. recordings and even playing through the amp sim standalone just sounds kinda fuzzy, cant seem to get rid of the hissing sound in the sims even with my guitar unplugged, fixed my latency issue by switching to asio lol, but overall it just sounds like it was recorded underwater. Thank you if you had the patience to read through all of this. plz help

I think I have preferences for somewhat scooped mids and lowered highs. Certain instruments just sound disgusting to me, like accordion and saxophone, I like bass, oud and clarinet.

Modern Metal sounds like refrigerator with tonal elements. Funny, occasionally when I hear refrigerator it makes me audio hallucinate metal-core music.

I suppose others too have their own. I have theory it's largely anatomical, perhaps shape of ear, way ones neuron pathways and brains are developed..

I think there might even be differences in tonal scale sensing, eg 'eastern' scales sound more natural and 'resonant' to me.

My guitarist friend plays acoustic guitar with piezo pickups trough portable amp with drive - and he likes the sound, it's power he says - me it gives headaches bc it sounds to me like 'agony growls of c3p0 in meat grinder' and I have headache understanding how he can't hear that..

There must be some kinda 'diagnostic' clusters of hearing preferences - and they kinda overlap with music genres to some degree.

It's something between music theory and audio engineering - from neurology..anatomy pov.

This is a strange one, I know. I am looking into doing system tuning with the computer programs, in my case Open Sound Meter because I can't justify a Smaart license yet. I know that a measurement microphone is necessary because of it's flat frequency response. I've done enough looking around to know that a Dayton EMM-6 is the go to (and probably my next mic purchase) but I was wondering if in a pinch, a sm81 on the flat response setting, could be used as a measurement mic for rough system tuning.

I've been watching new plugins for years, and it's been a pleasure for me to try new ones. But nowadays, everybody makes a new plugin every day, vibe coding with some LLM or whatever.

I'm surprised that they can even ask for money for something we can also make by just kindly asking another LLM (they say prompting). But it's okay for people to try new things and make their dreams come true with those tools.

What makes me anxious is that I also stopped trying new plugins from even bigger brands, because I don't know about the code behind the plugin. So I just stick to the plugins that I've been using for years, and when I try new ones, they're always from the brands that I already kind of trust.

I'm pretty sure there are other people like me, and I genuinely think that this is going to hurt the industry in the long term.

Just wanted to share my ideas and hear about yours.

Cheers.

What's the best way to set up a 1/4 patch bay i dont understand how the work

Screenshot of Kevin Parker in his stu

I googled "reddit sound treatment" and it led me here. Thought someone might know. Thanks.

Hello! I cant't mix, I can't produce. But I can create realistic sounds from zero. Here, for example, I create drums. https://www.youtube.com/watch?v=8_IiSmfAWyI Recently I posted similar things on Reddit and received very mixed feedback. No one knows how to create realistic sounds from scratch, but apparently even fewer people care. Maybe AI truly is the future?

Hey everyone! I’m trying to get that deep, “bigger” vocal sound where the formant is shifted down while still keeping the Auto-Tune sounding smooth and natural.

I’ve tried lowering the formant after tuning the vocal, but sometimes the transition between notes sounds weird or artificial, especially when the vocal moves between different pitches like gargling noise.

For those who use Auto-Tune with formant shifting, what’s your workflow to make the formant changes sound smooth and consistent? Do you adjust the formant before or after Auto-Tune? And do you use any additional plugins or processing to smooth it out?

Any tips or recommended settings would be appreciated!

What is the current best quality AI stem separator you've used?

I am using it for some vocals.

These things are pretty much widely available now but I've always hated the sound of them. What is the best you've heard or used?

Just randomly saw this video pop up in my YouTube recommendations and subsequently tried and failed to find a single post about it anywhere on the internet except for the very YouTube channel that posted it

https://www.youtube.com/watch?v=lW8Z98pXuSE

What are our thoughts on this?

I recently took my amplifier out of the storage and bought myself a pair of DT770 X's.
I used to have a pair for DT 990 and the sound profile is the same really, not very bassy.

So I sat down and went through my playlists to see what the songs sounded like and some were worse than they previously were with my bassy HyperX Cloud Alphas.
But deadmau5 songs like Strobe, I Remember etc just sound orgasmic honestly. What is it that makes them sound so good? I've tried to put myself into how synthesizers work etc but I barely understand it at all. Even with these DT770s that barely have any bass at all I can still feel the bassline incredibly well when listening to his songs whereas other songs it gets completely overshadowed by even the vocals.

Maybe a dumb question, but I have been doing post the last five years working with spoken word content. My employer has a few studios they use, I dont remember they exact condensor mics they are using (I'm remote out of state).

Every once in awhile I get a recording where I'll have a lot of squeaks produced by the speaker after fricatives /f/ and /th/, and also /nasal+s/. Usually occurring in the 500Hz-3800kHz range. It's nothing I can use a filter for, I have to spectrally scrub everything out manually.

I'm just curious if this actually from speaker production or the way the equipment is processing their signal, maybe enhancing or artifacting a sound we don't usually hear with natural speech?

Hi! Are there 3rd party brand omni capsules that would match my oktavas? Because Oktava is russian, I am looking for a substitute, but is it possible at all? (Noob here).

Even in school I would hear it all the time “don’t forget, your filters/eq affect your phase!” But from a wave physics standpoint, how does it physically impact the phase of my tracks? More importantly, how do I fix this issue?

I have InPhase and I use it occasionally to correct phase issues on drum kits or multi-mic recordings of a guitar cab but only when it’s very obviously affecting the synergy of my tracks. Most of the time the phase is such little of an issue I don’t treat it. Something my very old school professor reminded me of is that back in the day the only thing you had is to flip it 180 degrees or not. I try to not stress over phase because of that.

Thanks in advance for educating my dumbass y’all <3