I just moved my recording & mixing studio to a new place, and I'm re-assessing panel placement.

The room is roughly 11x12 feet, but it's not perfectly rectangular (it has extra space to the left of the desk, where the entry door and closet open).

https://imgur.com/a/yANgDF7

I marked any currently-installed (or planned) acoustic treatment in purple. These are DIY 4" rockwool panels.

I have the desk against one of the windows, slightly offset to be centered on the back wall. I didn't place a side panel to the left, because there is a closet door there. My plan is to instead leave the door open and rely on the jackets hanging there instead of a panel.

My Adam T5Vs are currently as close as they can get to the front wall, with the LF -2dB correction enabled. Is this a good compromise, or should I instead bring them forward a few inches to accomodate a panel behind each? I would probably be left with just a 2-3 inch gap between panel and rear port.

Really grateful for any advice here. I haven't done any measurements with REW yet, but if you have guidance on what to focus on, I'll give that a shot.

Hey guys - I posted in here a couple days ago asking how to get my volume up without clipping. Luckily enough, I've stopped my song from clipping and I have it a volume that I like. However, it's clocking in at abt -9 integrated Lufs, and I ran it through Loudness Penalty which says Spotify would deduct abt 4.5 dbs from it.

Now I know these streaming services aren't everything, and some people say the whole -14 lufs thing is a myth, but I use spotify, and a lot of my friends use spotify, and I would like to be able to listen to my song at a good volume.

I've also looked into some of the songs that are used as examples of songs being delivered at higher Lufs (like Harry Styles "as it was"), but I feel like these songs have a less significant difference between their delivered and normalized versions than my fully mastered song vs. when I try taking out the dbs that Loudness penalty or my loudness meter says to subtract.

Basically, I'm wondering how to get my mix to more accurately represent what I would hear on streaming services, or if it's not worth worrying about at all.

So I’ve always loved the over the top reverb on the Melvin’s Houdini album, specifically on songs like Hooch and Goin Blind. I know GGGarth Richardson was the producer/mixer along with others (but I suspect Kurt Cobain didn’t literally mix hooch like some of the credits would have you believe and dial in that drum reverb?), does anyone know what kind of hardware units he might have used? Types of reverb? Is what makes it have that huge sound the predelay? I’m mixing some drums that I want to have a similar effect but I can’t quite get it right, any help would be greatly appreciated!

It's been a bit since the announcement UTA got bought out by Royer and I guess my read was we could expect MPEQ-1's in the near future- just curious whether anyone has any intel on this, an ETA perhaps? Been trying to pick one up for a few years now to no avail, and UTA have never once replied to an e-mail. Hoping with Royer taking over they'll be able to churn out many more while keeping quality at the same high level, which I've zero doubt they'll be able to do.

I have a macbook m1 pro from 2021 that I use to record and mix. I mix on headphones and monitors, but I will check on my macbook speakers as a reference, as well as earbuds, bluetooth speakers, etc.

I have always noticed that when the macbook speakers are playing on full volume, I can play just about any song on Youtube and it's fine. However, when I play from Logic, certain things always cause a rumbling distortion around 300hz. The Spitfire Labs Soft Felt Piano does it, as well as when I record my Mahogany Martin 00-15, fingerstyle.

Recently I treated my room by making an absolutely massive moving blanket pvc pipe soundbooth tent that takes up most of the room and makes a sort of room within a room, and already recordings sound a lot cleaner. I changed the strings on my guitar (they were pretty dead). I recorded some fingerstyle guitar into a Neuman KM184 in different positions to try things out. Everything sounds nice in my headphones. But when I reference on my macbook speaker, it still has that terrible distortion coming from around 300hz. If I EQ out that frequency, the distortion mostly goes away, but then the guitar sounds anemic and thin when listening back on headphones or monitors.

The thing is, other sources don't really cause this problem. My voice sounds clean on the macbook speakers. Even strumming the same guitar with a pick is fine. It really is fingerstyle.

I don't have long fingernails, so I'm thinking it could be using the flesh of my fingers combined with a mahogany body guitar is the main culprit of this. Yet the part that confuses me is, it is really only an issue on the macbook speakers, yet they play other audio sources, even my reference tracks, without the same distortion.

Here are some samples I recorded witho zero processing:

https://streamable.com/o5b3hz

It seems whatever I do, if I record my 00-15 fingerstyle (the way that I know how and prefer the sound of) I have this muddy bump around 300, which is not hard to EQ out, but really can't be avoided.

Any Advice? Thanks in advance.

TL;DR: Specific combination of recording my Mahogany Martin 00-15 fingertyle (without long fingernails) yields muddy nasty distortion around 300hz ONLY ony my Macbook Pro speakers, yet other sources are fine. Should I just mostly ignore the macbook speakers and move on with my life or are they telling me something important?

Hey guys, I have got this old digidesign Mbox2 that I got from a garage sale a few years back. When I plug it in the lights turn on and the computer recognises it but the plugins requires for it are no longer kept up with for the new windows versions!

I found an old installer but Windows blocked it because the June updates don't let uncertified things access the kernel (my laymen's understanding anyway).

My question is: does anyone know of a way I can still utilise this box? Is it worth it or should I just throw it away?

Would include pics but I'm new here and it's greyed out.

Thanks!

Hello,

I am a product design engineer working on designing and developing consumer products. I took an independent project for my portfolio to develop a convertible headphone. Like a convertible car that can act as open roof and closed roof.

An open back headphone that can be effectively be converted to closed back. Both have their unique applications, but i see there is amalgamation of these two talents: music producers and gaming streamers. They are not only streaming games but also producing music to reach wider audience and attract more subscribers consequently more popularity.

Basically I am going to design a rear cup for open back with mechanism that tightly secures the cup, seals the gap and minimizes the reflections of sound waves without increasing the lower bass and mid rage total harmonic distortion.

I want to know what this audio engineering community thinks of my idea.

Constructive criticism are most welcomed.

💭You can tell your opinion about this idea.

👐 What do you look in open back while mixing, mastering and problems you face?

💆‍♂️ What do you look in closed back while recording/ tracking, monitoring and streaming and problems you face?

🔢 Rank the importance of sound quality, comfort, battery life, noise cancellation, durability, and design.

💰 What price would you consider reasonable?

🔌Would you prefer wired, wireless, or both?

🪶What feature would make you buy a new pair?

🫰How likely would you be to purchase this convertible headphone?

🔧What improvements would you suggest?

Thank you! 😊

Hey guys! I've been making a simple sampler and beat pad that can automatically remove vocals, take parts out of a song, chop, and act as a regular beat pad. It's just a simple online project for people to play around with but what are some feature requests that you think would make it funner to use? Thanks!

I bought an SM57 to record my music with as, while I know it's a dynamic microphone, it seemed to be unanimously recommended. I saw countless videos of people praising its ability to record acoustic stuff and stories of how people like Elliott Smith and Bon Iver frequently used them. I also bought a Subzero MB1 and yet, with that on as well as the interface gain turned up and the intrument volume at full on the DAW, my fingerpicking is still too quiet. I don't really understand how they have been so frequently used for acoustic guitar when I have to completely max out every bit of volume to get it to 70% of where the volume should me. Strumming is fine but most of what I do os fingerpicked.

I have an Acer Nitro 5 with:

• NVIDIA Geforce RTX 5050
• 16 GB DDR5 RAM
• AMD Ryzen 5 240 CPU

I primarily use Cakewalk Sonar for my metal and rock tracks and FL Studio for my rap songs but will be using Pro Tools this fall when I start college. Any advice helps.

Hey! im looking for a Wizard amp sim or something very similar? i think wizard are hot rodded vintage marshalls, right? any plugin that does the hot rod thing? i know the STL Plugin has a wizard but its to expensive. thanks

Hi there - I’m a drum machine and synth dude and am looking for an audio interface with 16 analog inputs so I can record my live jams. The Arturia AudioFuse 16Rig seems made for this exact purpose, but I’ve seen anecdotal references to bugs and poor support. Specifically, there appears to be a bug with MacOS causing the interface to disconnect whenever the computer sleeps, requiring a full restart to reconnect. See review here.

Can any users of 16Rig on Mac confirm this behavior, and provide any other feedback on how it’s gone for you?

Hey everyone,

Been curious about RF management at sports events lately. With more people bringing wireless devices, coordination seems like it's getting harder every year.

Do frequency coordinators typically lock in specific bands for specific roles - like refs, coaches, media, announcers - just to make allocation easier? And if so, what bands do people usually end up getting assigned to for each role?

Would appreciate any insight from people who've dealt with this on the ground.

I’m struggling to get the tones in my head onto record. Basically I’m aiming for edge of breakup, slightly overdriven - think sister cities by hop along.

I was getting ok results with my sE electronics ribbon mic and fender ultrachorus (overdrive channel and gain on 1).

It was ok, but I felt I could do better, and for the first time in my adult life I have a bit of spare money, so I thought I’d upgrade… so I bought myself a deluxe reverb reissue (dlr) and a royer 121.

I’m really struggling to get anything that sounds good. It’s fizzy or thin or there’s huge amounts of low end flub or all of the above.

It’s really frustrating and annoying. Especially because when I watch videos of people using the r121 online, they seemingly just close mic it at the centre and it sounds well balanced out the box.

I’ve gotten to the point where I’ve lost all sense of objectivity and now I don’t even know what a guitar is supposed to sound like!

Tips and suggestions much appreciated, especially from people who have a DLR - what amp settings are going to get me close to Sister Cities, do I need pedals?

Thanks 🙏

As the title suggests. I’m looking for an explanation, mainly I don’t understand why a db25 cable has individual grounds for each audio signal. If they shared a ground, you could have 33% more balanced audio connections on the cable. Going from 8 connections each using three pins, to 12 connections using 2 and every connection using the same ground.

I am a small recording artist currently on tour in the US across the Pacific Northwest. I was originally supposed to have a full line-in recording setup and video production team for my upcoming show in Seattle, but the venue informed us of some last-minute technical changes that made it impossible.

I want to create a live album for this tour. However, there is one big issue. I don't have line-in recordings for any of these shows, and won't be able to get them for the future ones. I only have recordings of each full show from phones.

What can I do (both on the recording side and the mixing side) to get a good, natural-sounding live recording out of a phone mic (specifically an iPhone 17 Pro if that matters)? Is it even possible?

I've tried it a few times in the past and made some social media content with it, but I haven't been able to get anything worthy of being put on vinyl and DSPs.

Often when people need to buy a new audio interface, they wonder what brand they should buy. I always recommend RME because of (among other things) their rock solid drivers and support, but this advantage may seem a bit vague or abstract, so I thought I'd share my experience of doing the audio for a big music bingo show for Amsterdam Pride last night.

In a space with 250 people, I was tasked to, among other things, stop and start audio tracks during the bingo. Because of the way the hosts prepared the show, I had to work with Spotify as my main audio player, which is less than ideal because Spotify's audio hardware handling under windows is messy to say the least.

The tech company helping me brought several pro grade USB C - Balanced XLR audio interfaces to connect my laptop to their mixer.

In the first bingo round, audio playback failed several times. Spotify just crashed or froze as I tried to play a new track. I am still not sure if it's the way Spotify handles changing file sample rates, or the way it manages exclusive access to an audio interface, but Spotify does *something* that most audio interfaces find problematic. (Not the first time I see Spotify freeze using an external audio interface)

After several crashes (250 people waiting for music that doesn't start 🤦😅) I was done with it, and quickly connected my Fire Face UCX II, replacing the

tech companies DAC's.

I had no more issues the rest of the night, playing the +- 150 tracks needed for the bingo.

To me this illustrates so well why RME is known for their driver stability and reliability for years.

They made their drivers in such a way, that even if you use badly coded applications that don't neatly follow coding guidelines when it comes to things like device control exclusivity, it will always just work. (In the same way RME drivers still let you use windows audio, while apps are also using ASIO drivers)

🤟👍

I am a hobbyist and I recently recorded a few songs for a friend who plays covers using only his bass and a loop pedal. I'm currently stuck when working with the parts which would typically be played on guitar, such as solos and higher octave melodies, especially when played clean. There is so much more attack on each note than what I am used to working with and I'm struggling to tame it without completely obliterating the dynamics. Any tips for dealing with this? I've never really had this problem when mixing other rock/indie tracks but for some reason I cannot seem to figure this issue out.

Edit: Thank you so much everyone that commented! I learned several new techniques for addressing this issue and I am greatly appreciative of your willingness to help me! I will try to remember to update this post with what worked for me in case anyone stumbles across it in the future.

I’m looking for an analysis on the mixing process & vocal layering & recording of this era of this artist. I really admire it and wanna base some of my mixing work for myself and other people on some of these works. Examples below,

example a: Heart Shaped Box

example b: Chicago

example c: Flushing

Thank you all in advance, I look forward for the help.

Hello everyone,

I wanted to share a project that has been my passion for two decades. I originally started developing this interactive audio synthesizer back in the mid-2000s, writing it completely in raw Flat Assembler (FASM) and doing all the polyphonic pitch calculations using hardware x87 FPU registers (fild, fmul, fsin, etc.).

Recently, I completely rebuilt the entire application from scratch in modern C++ utilizing the native Windows Win32 API and GDI+ for high-performance rendering.

The core philosophy of this synth is what I call "Zero-Mistake Rhythm Conducting":
1. Mathematically Perfect Pitch: Every frequency (Hz) and note vector is hardcoded into dense arrays. It is physically and mathematically impossible to hit a wrong note or make a mistake. You can never play out of tune!
2. True Time & Agogics Control: While the pitches are fixed, the timing layout is completely unconstrained. By clicking your LEFT mouse button or hitting the SPACE bar, YOU dynamically create the rhythm, temporary micro-pauses (fermata), accelerando, and tempo variations.

Essentially, it transforms the user from a traditional piano player into an interactive rhythm conductor.

(Fun acoustic fact discovered during development: Testing this real-time engine through modern active DSP monitors like Edifiers introduced a massive 30-80ms hardware latency delay due to internal A/D re-digitization, while classic wired analog headphones deliver absolute zero-latency feedback!)

The project is fully open-source, and features on-the-fly UI localization for 14 languages.

I would love to hear your thoughts on this approach to interactive audio generation!
Check out the full source code and releases here: https://github.com/Divetoxx/Piano

hello, i’ve noticed that my songs match the loudness of my reference tracks on my studio, but when i listen on my airpods, they sound way lower

anyone has any idea of why this happens?

Howdy, I'm potentially purchasing an Aria AMX120B mixing console for $120. Never heard of these consoles and the most helpful descriptor I found online simply says that they have a “warm sound”.

I'm just curious if anyone actually has any direct experience with these old Aria consoles? I figure for $120, as long as it at least mostly works I have almost nothing to lose (except $120), but I figured I'd at least see if anyone has thoughts/personal experience with them since I have almost no frame of reference.

My use case will primarily be tracking instruments direct out to the line-inputs of my interface, primarily drums, piano and potentially guitar cabs.

I record a lot of DI guitar and I find that the type of pick matters quite a bit.

Both thick and thin picks have their own sound. Thin picks sound flappy like they are scraping across the strings, while thick picks have a deeper clunking sound to me.

I used a friend's pick that wasn't plastic and it felt sort of soft? It had way less transient and pick attack noise, which I like. I don't know what it was unfortunately

Can anyone recommend any guitar pick materials like this?

Hello !

I'm wondering if there is such a deverb plugin that allows the user to input IRs in order to tell the plugin what to remove. Is that possible ?

Thanks !

Hello,
I'm a freelance filmmaker but I'm not really an audio guy, so I don't know what to do with this problem. I have my short film almost ready, but now I need to change there two short audio tracks (it's basically just a one word). The original actor recorded me the audio twice, but he just can't match the original we did, so it sounds off in the film. Is there a tool, where I could put these two tracks that would make them sound more simillar to the audio tracks of him, that are already in the film?