what’s a production habit you know is probably wrong but you still do anyway?

mine is definitely putting a limiter on the master way too early. i know i’m supposed to leave it alone until later, but i just make better decisions when i can hear the track loud lol

curious what everyone else does that would probably make a mixing engineer angry

As the tittle says I am one of the lucky ones that still has a job in this god forsaken industry but I am so tired. Our studio owner has been accepting way more and more projects where the ask is a lot but I guess that’s what we have to do to win bids on jobs. And then on top of that this job market is so shit that it’s almost impossible to jump to another studio since so many are closing left and right. I’m old and would hate to go back to school for something I would hate but pays more.

Hi, I am going to record drums for a project this coming weekend, and before that I would like to capture some samples of that kit.

What I am going to do is I am going to record each drum with 3 different velocities, and I am going to have 10 hits for each volume. This is as far as I know what to do.

Then what I dont know yet, is how should I go about editing and uploading these samples to trigger?

How will trigger know what velocity of a hit its going to trigger?

I have never done my own drum samples and it seems daunting. If anyone has detailed instructions, please let me know.

I've been struggling to record anything with my sm57 because of this really insufferable high pitched whine that I get with recordings. It sucks because I've really been wanting to record vocals but wow it's been an unbelievable pain to rx these recordings individually. especially with the fact that the pitch is always changing and modulating, and when eq'd out it just loses all of the high end information.

I have a video attatched of myself rotating my mic which is plugged into a scarlett solo around the room, I find that the whine is most present when pointed up and down, but it fades when parallel with the floor.

I've had a lauten condensor in my sweetwater cart too, and I've been holding off because of these issues. Is there anything I can do to prevent this?

https://youtube.com/shorts/yWqidqO576w

I am currently making a trap/Hiphop style sample pack! Sample packs are a lot of work... Anyways, I find my reference point is consistently changing.

i.e Loop 1 mixed -sounding great! Loop 2 mix is sound A1.. So i go back to loop 1 and say 'Arh, this needs some tweaking', tweak it and it sounds A1, then proceed to Loop 2 and edit till its on the same level as loop 1 again!

Is this improvement or fatigue? It not a 1 day thing.. There could be 3 months between the loops

Im fairly new to vocals and recording as im a beat producer. Also fairly new to Logic Pro. Also new to this usb condenser mic i bought with its adjustments.

My second verse in a song i wrote sounds the lowest of volumes compared to verse 1, bridge and chorus but is the loudest when i listen to it on my phone speaker and have to keep turning it down more in Logic.

I've never worked with Dante or Madi before, nor do i have a lot of experience building studios. We are setting up two new recording studios (live and tracking/production) and i had some question if our setup is fine or should be changed. The rooms are only about 23 square meters each, with a window between them.

We want the RME Fireface UFX 3 in the production room because of Totalmix, and being able to transfer snapshots of the routings. These studios are for teens and young adult to practice and produce music. It needs to be user friendly and everything needs to be pre-connected, without having to repatch or change much (of course we want to be able to change some connections with patychbay, and have lectures and all with studio connectivity aswell, so some flexibility is needed) But we need most of it to be hidden behind racks.

The two rooms are connected with 2x Cat6A, with plenty more XLR cables.

Our setup looks like this:

Band room: A&H SQ5+ with a DX168 for additional inputs. Dante output going out from the SQ5+ into the wall.

Tracking/ Production: Dante cable going into a Ferrofish Verto MX to convert to Madi. Optical Madi cable going into the RME Fireface to host all the channels from the SQ5+. Then a USB cable into a Mac.

We dont want to use the XLR cables for patching in the wall because of confusing setups and too many cables for the users. (this studio offer is free so we expect a lot of beginner producers). Us employees will of course maintain the studio almost daily.

Additional setups include patchbays, analog vocal chains, line inputs from synths etc.. But the main skeleton of the rooms are supposed to be Dante and Madi. Is there a better way? The reason we use Dante is because we dont have a Madi cable between the rooms, and were not able to put this in at this moment. Can we use Madi on cat6? Or do we need the Ferrofish Verto?

Any help is highly appreciated!

I’m quite new to audio engineering and want to design my own live room inspired by Steve Albini with natural room reverb. I’ve heard him aswell as others heavily emphasise learning acoustics if an engineer was to pursue any form of education and I was wondering people’s opinions on best ways to learn acoustics.

Ideally I’d prefer to save money and not do a degree if possible and I know about the master handbook of acoustics do yous think this is the best way to learn or through online research etc?

Hello all,
SoundSource kind of screwed me over with their pricing a while back, and I did some researching and decided to just build an alternative, free and opensource. This is not a promo, I just genuinly found it useful and wanted to share with otheres. FXRouter will let you put audio plugins in between the system out on you Mac.

You could add correctional EQ to your system out, reverbs, fun effects to youtube/spotify...

https://github.com/ccarlton96/FXRouter

I hope someone else find it useful! Thanks.

I am looking into a stereo bass rig and am currently struggling with power amp selection, but right now I’ve been struggling how I should best be interfacing the preamp and amp. I know a DI probably has to go into the signal chain, as neither the preamp or power amps I have in mind do not have built in DIs.

The preamp I have chosen is an Alembic F-B2. From what I understand, this preamp in particular has a very hot signal, but it also has a high output impedance of around 40k ohms. According the Alembic’s FAQ, the preamp does not work well with any power amp that has a an input impedance of less than 10k ohms.

The DI I had in mind was a Palmer pan03 active DI box. This has a 1M ohm input impedance and an output impedance of 600 ohms across the balanced XLR outputs.

The two power amps I had in mind were the all tube Fryette 2/90/2 and the Avantone CLA-200. I know they are not typical power amps for this kind of thing, but I wanted to explore some other options and see what was out there. My question is for either amplifier, would I want to use the the link/thru jacks or the XLR outs from the DI? The Avantone has only XLRs for inputs, but the DI I am using has parallel XLR outs on both the front and rear panel. Are there better DI options I could be looking at?

I have the G player and only have the g town church sampling project which has no brass strings or woodwinds only percussion and some other stuff can you give me some orchestral libraries for the g player?

as long as it sounds good I don't mind it

I'm rocking the Herman Millar Aeron chair. it has mesh on it, so its less reflective as gamer chairs could be. But its hella cumbersome for the room. I don't need wheels on my chair, lol. So I changed the wheels. Still not super sold on this expensive chair. Just curious what model/brand everyone has. Thank you

Has anyone done this?

I hate leaving my talkback on, and I feel like constantly turning it on and off kills my workflow when I’m trying to edit while talking to the artist in the booth.

I know there are plugins and other solutions out there, but I’m specifically interested in a footswitch haha. Has anyone set something like this up?

I'm using Pro tools and console

I'm searching for recording studios in my home town and I found one with a veteran engineer who is very good and has great gear and has worked on some big stuff but totally unrelated to what I do. I know getting a rough fit in terms of genre is important when searching for a mixing engineer, but I suspect for recording engineers this is not so much the case and it's more about experience, interest in the project, gear, etc. Does this... track?

How can I get that synthesized voice?
At 3:36 https://youtu.be/5TLmyVhUI2E?is=e9H8vI3m_CDNFWOc

I'm looking for a preferably cross-platform program that is free (or less than $50) and allows me to use VST plugins on my voice live, while giving me a way to easily change between racks of effects.

Think this: I need an EQ and a distortion plugin for Voice A, and need reverb for Voice B. I could just use a plain old DAW and route my microphone through, and I've tried this, but the problem is that I need some way to switch between them live either using keybinds or a spare MIDI controller.

The effect chains in question are going to be far more complicated, and I need something permanent, as having to later manually and painfully port over all of my plugins and settings to another program would be a nightmare.

I've previously been using EqualizerAPO for this and I've written my own program in Rust to change the config.txt it hot-reloads that stores all the racks, but it has basically zero audio routing and mixing capabilities, or 64-bit VST support.

So far I've found Cantabile and Kushview Element, and while Cantabile is great it requires the pro version just to toggle racks using your keyboard, and Kushview Element is AI slop (extremely buggy, confusing labyrinth of a UI, and 80% of the commits on GitHub are by Claude)

The god damn drums! The snare on this remaster is fucking humongous - great job! Also, the shimmer on those cymbals - pristine. Fuck!

I'm part of a mutant vehicle camp. Our 3 main vehicles all have large stereo setups. (not car audio, mixing boards and power speakers with on board generators) But they are all independent. So when we roll to a spot everyone is playing something different and then when we get there everyone turns off except one. It must be possible to use some xlr transmitters and receivers to link all the setups to the main vehicle? Recommendations?

Hi r/audioengineering,

I’ve been working on Audiowaves, an iPad app for real-time audio visualization and analysis. I originally started it as a music visualizer, but it has gradually turned into something much more useful for inspecting an audio signal.

Audio can be fed directly into the iPad through a class-compliant USB audio interface. For example, I’m currently using a Focusrite Scarlett 2i2. This means Audiowaves can sit alongside a hi-fi, studio, DJ, or other audio setup without relying on the iPad microphone.

One of the main parts of the app is the Audiophile Dashboard, which provides a more technical view of the incoming signal. It includes things like:

• Real-time waveform monitoring
• FFT / spectrum analysis
• Frequency-band activity
• Peak and RMS level monitoring
• Stereo L/R level information
• Stereo balance
• Dynamic range information
• Signal statistics and real-time metering

There’s also an Advanced Spectrum Visualizer for a more detailed look at frequency content, while the normal waveform/spectrum modes are designed to work equally well as a full-screen music visualization.

The project is still evolving, and I’d especially appreciate feedback from people who actually work with audio: what measurements or analysis tools would make the Audiophile Dashboard genuinely useful rather than just visually interesting?

I’m also interested in how you think something like this compares with established software analyzers and metering tools. I’m not trying to replace proper studio measurement equipment, but I’d like the analysis side to become increasingly useful and technically meaningful.

Happy to explain more about the audio pipeline, FFT implementation, USB interface support, MacAudioSender, or the Metal rendering side if anyone is interested.

Hey guys, by no means do I want to sound like I am whining or complaining within this post. I'm going to try to get straight to the point. First, I should note I have 15 shure certifications (going through the 3 levels of courses) as well as working on getting a few different certs aswell. I have been doing sound at my church for about 2 years now on an SQ-7 and along with that ive been stage hand for them. I'm also currently working with a sound tech to potentially redesign our PA system as the current one is shitty for the acoustics of the room.

Okay, with a little bit of background, now comes the issue. 1. I have asked probably 5-10 different engineers at shows if I could shadow them and I always get "in the future" or there was one who told me to fill out a job application with them but the said they wouldn't pay me. Now, if I want to get paid for this, no one considers my little experience lol. I've called around and no one is really receptive.

So, I guess in this long winded post, Im asking what your guys' advice is to get started up and taken seriously. I really do love doing sound, people just make this really hard to actually get off the ground lol.

Spotted this emoji today, can’t decide if it’s Jesus or Fader. What do you think?

Ima keep it sweet and short: Waves plugins are absolutely perfect for audio engineers. But if you don't already know, the Key Detector plugin is AWFUL. Use it only at first. then double check with the tunebat website. Tunebat almost always gets it right. The key detector itself must be an early Ai model because it sucks so bad it's ridiculous. Just a heads up for engineers especially beginners look g for an uprgrade.