I don't even know what more I can do. I read books, I practiced every day for hours, I learned plugins, but I can't reach a good sound. I posted my sound thinking it was a really good mix and master, but the feedback was "I like it, but it would be better with some more work." And yeah, I know it's true. I know my skills are limited, but I don't know how to improve anymore.

Another book or video just tells me to do this and that, and when I do that, it sounds awful. My version was better. More than that, I don't want to just copy what people say in tutorials. I really want to learn.

I listen to music that I like, and I really want to get that kind of sound, but sometimes it feels so far away.

I have my headphones, my interface, and yeah, it's a home studio, but I've seen people making music in a home studio with a really good sound, to the point where you can't even tell it was made at home unless you know where it was made.

I was thinking about paying an expert for feedback on my music production, but I'm kind of afraid of paying someone who doesn't understand the genre, the sound I like, etc., and then, when they finish the work, it sounds really bad. Because I've been to a studio before, and they gave me a really bad mix and master.

So I want to ask you guys, what can I do to improve? How can I figure out what I'm doing wrong and where the problem is in my mixes, yk?

And I'm not just talking about EQ, space, compression, or anything like that. I'm talking about effects, too. For example, you can listen to The Weeknd's music, and he has a really good sound, man. Or you can think of another artist, not just The Weeknd. Because you can have the most expensive microphone, everything you want, but if you mix badly, your sound is still going to be shit, even with the most expensive equipment.

So please, can someone tell me what I have to do? Do I need a mentor? If so, who should I look for, and how can I find someone who matches my style? If I have to watch a tutorial, which one?

I've read every post and subreddit talking about "the tutorial," and it doesn't work for me. But I have to keep trying again. I'm into it. I just want to understand what I'm doing wrong and how to get better.

I’m in a bit of a predicament here. I’m just a teenager learning audio engineering out of my room but I’m broke af. In all my recordings I notice lots of low end reverb and on the high end what I think is vibrations coming from the guitars on my wall?

Is there any treatments I can do to remove the reverb out of my room? Preferably cheap?

Hey everyone,
I’ve been meaning to post this update for a while, and today we hit another milestone that made me realize I really needed to share it.
Two pieces of legacy professional audio hardware are now working in SelahOS:
Digidesign/Avid MBox 2 ✅
PreSonus FireStudio Project ✅
The MBox 2 breakthrough happened first. After a lot of work investigating the original driver behavior and figuring out what the hardware needed, we were able to get the interface functioning under SelahOS.
That alone was exciting.
But recently I started working on a PreSonus FireStudio Project, which presents a very different challenge because we’re now dealing with a FireWire-era professional audio interface rather than the MBox 2’s USB architecture.
Today we took that beyond simply getting the FireStudio recognized.
I tested the physical inputs — and they’re working inside Ardour. 🎛️🔥
Actual audio from the FireStudio is making it into the DAW.
That was a big moment for me.
I’ve been building SelahOS around the idea that a creator-focused operating system shouldn’t just run music software. It should also help creators continue using the hardware they’ve already invested in — especially perfectly usable professional equipment that has gradually lost support as operating systems, drivers and computer architectures changed.
This work is becoming part of what we’re developing around SelahBridge.
The long-term goal isn’t:
“Here’s a weird workaround that gets one old interface running.”
It’s closer to:
Can we build a compatibility framework that makes supporting and preserving professional creator hardware repeatable?
We’re still early, and I’m being careful not to claim that every function of either device has been exhaustively validated yet.
For the FireStudio Project, the next round of testing includes things like:
multiple simultaneous inputs, playback, input/output together, sample rates, buffer settings/latency, longer recording sessions, reboot persistence, and general stability.
But getting from device detected → functional interface → live physical input into Ardour is a pretty meaningful step.
Claude Code has been assisting me throughout a lot of this engineering/debugging work, and it has been an interesting combination of studying the hardware and old driver behavior, testing ideas, breaking things, fixing them, and slowly turning those discoveries into something we can integrate more cleanly into SelahOS.
This is also making me think much more seriously about hardware preservation as part of SelahOS’s mission.
There are probably thousands of studios, musicians and producers with older interfaces and controllers sitting in closets — not because the hardware stopped sounding good, but because the software ecosystem moved on.
I don’t think “the manufacturer stopped updating the driver” necessarily needs to mean “the hardware is now garbage.”
That’s a problem I’d like SelahOS to help solve.
I’ll post more technical details and testing results as we validate everything.
And if anyone here has older professional audio hardware that has become difficult or impossible to use on modern systems, I’d genuinely love to know what it is.
That could help us figure out what hardware deserves attention next.
SelahOS — Pause. Reflect. Create.

I have the actual Randy Staub drum samples that were used during the tracking and recording of Metallica's Black Album, and I'm giving them away for free. DeathKick, MurderSNR, and 4 other one shot samples. If you want them, the link is in this video, where I explain how the revolutionary way they used them DURING TRACKING and mixing. They Kept These SECRET for 35 Years The link is in the description of the video. But if you don't want to watch the video that took me 3 months to make, you can just DM me and I'll send you the link. They're totally free, there's nothing to buy. I hope you guys enjoy it!

I have about 8-9 years of recording experience (mostly learning fro Youtube, here, and trial / error), primarily solo stuff. My usual workflow is drum machine > bass > rhythm guitar > lead guitar > sfx / perc / synth stuff. I have gotten pretty decent with recording techniques and mixing, and I am pretty happy with all that. My question is regarding taking my experience here to home recording a full band.

Ideally, I would like to play the songs live, with amps separated into either different bedrooms or decent sound proofing between amps. The room we will be recording in is a large, circular wooden room with some windows along one side and lots of scattered furniture. It's got great acoustics and doesnt have too many reflections (blankets & such on the windows).

Now, we have some elongated jam sections that have structure but feature improvisation on top of the structure. Originally I was thinking we would record drums first to a click, then layer like I always have, but I think I would like to try a live recording process for this album. Where can I go to learn more about recording like this? Ideally I would still be able to go back and overdub sections, which is why I am trying for isolation. Do you recomend isolating the amps into seperate rooms, or using smaller iso in one room? I have a headphone splitter so we would all be able to hear if it was in seperate rooms, but I've never recorded this way and dont know a lot about it.

I dont have super lofty 'I need this to sound polished' goals, in fact I am shooting for an early pavement / modest mouse sound (I know, super fkn original) ((we dont sound anything like these bands, I just mean from a purely sonic angle))

Any advice is appreciated. And before I get a 'just go to a studio' response, I have recorded in studios before but this project is really special to me and I want to have done it myself to some degree. I will send out for mastering, ofc.

Hi guys, noob here. Cowbell jokes aside, I'm a bitt puzzled when trying to figure out what's going on with the mixing and panning of Blue Oyster Cult's "Dont' fear the reaper". In the intro, it seems that there is only one guitar at the right side, and that (maybe?) just reverb from that guitar is heard on the left side. Or maybe it's two guitars and one of them only plays one note?

Same happens in the chorus: is there another guitar at the left or not? Are there two guitars panned or is it one on one side?

Also, percussion-wise, I seem to hear guiros and/or castanets? Or maybe it's a vibraslap?

So I've been making music as a hobby for about 5 years now. I was extremely happy with the RVox and CLA-2 Compressor combined with the SSL-Equalizer until Waves decided to pull out "Waves Central" and f*ck each and every Engineer with it. I'm in a basement so natuarally the Interent Connection isn't very good. So when I decide to record something, it's quite a pain to update each and every plugin, just for Waves to NOT update the existing plugins which already exist on a track. They just put you in Demo mode and want you to go over each and every Mixing Track you have, and painfully replace each and every plugin. Not only that, but I want a more ethical option than waves, because...well...I don't want to get into details, but just look where Waves is based out of and where they pay taxes. Bottom line is, if someone knows a better alternative to those 3 plugins I'd be extremely grateful, because yesterday they done me in so hard, I nearly lost my interest in Music as a whole.

​

Hey everyone!

My metalcore band is currently preparing to record an album, and I’m looking for some advice from people who have experience producing, mixing, or arranging modern metalcore.

We want to achieve a modern, polished, and massive sound, with big guitar layers, supporting synths, and an overall wall of sound, but without making everything sound overly processed, artificial, or unnecessarily complicated.

The main issue is that we’re going to record with a local engineer who will help us with the recording itself, but he won’t really be handling the kind of production and sound design we’re aiming for. So, a lot of that responsibility will probably fall on me.

I have some experience with music production, but producing an entire metalcore album at this level is a different story. I’d love to hear your experiences, tips, and things you wish you had known when you started producing this kind of music.

1. How do you find the sweet spot between modern and overproduced?

A lot of modern metalcore productions sound absolutely massive, but when you listen closely, there are often so many layers going on: multiple guitar tracks, synths doubling riffs, sub-bass layers, extra textures, impacts, transitions, and so on.

I love that sound, and it’s something we’d like to incorporate into our music. However, I don’t want to end up adding layers just for the sake of making everything sound bigger. I still want the guitars to feel like guitars, the riffs to have their own identity, and the arrangements to breathe.

How do you decide when a part actually needs another layer and when it’s better to leave it alone?

Are there any general rules you follow when arranging and producing modern metalcore? For example, do you tend to keep verses relatively simple and save the bigger layers for choruses and breakdowns, or do you build up the sound more consistently throughout the song?

I know this is highly dependent on the song, but I’d love to hear how you approach it.

1. Layering guitar riffs with distorted synths

I’ve watched a few production videos where producers take MIDI versions of their guitar riffs and use them to trigger heavily distorted synths. They sometimes layer these synths underneath the actual guitars, and in some cases, they also add an octave-down version of both the guitar riff and the synth to make the entire thing sound even heavier.

I really like the results, and I’m interested in experimenting with this technique.

However, I’m not entirely sure how to approach it properly.

- Do you use distorted synth layers throughout an entire song, or do you save them for specific sections, such as choruses, main riffs, and breakdowns?

- Do you usually have the synth follow every note of the guitar riff, or do you simplify the MIDI part to focus on the main rhythmic accents?

- How loud should these layers actually be in the mix? Should they be clearly audible as a separate synth, or should they mostly be felt rather than heard?

- Do you usually use the same synth sound throughout a song, or do you change the sound depending on the section?

- How do you prevent the low-end layers from muddying up the guitars and bass, especially when playing in lower tunings?

I’m particularly interested in making the guitars sound bigger and more aggressive without losing their definition or making the mix unnecessarily muddy.

1. Synth plugins and sound design

At the moment, I mainly use Vital. I think it’s an amazing synth, but one thing that frustrates me is the lack of presets specifically suited to the kind of sounds I’m looking for.

I’m not necessarily looking for cinematic orchestral sounds or huge EDM leads. I’m more interested in aggressive, distorted, gritty synths that can complement low-tuned guitars and modern metalcore riffs.

Are there any plugins you would recommend for this?

I’d especially appreciate recommendations for plugins with a good selection of usable presets, since I’m still learning sound design and don’t want to spend hours building every sound from scratch.

Alternatively, are there any good preset packs for Vital that would be useful for modern metalcore production?

I’m also open to recommendations for free plugins or more affordable options.

1. Double-tracking guitars: Should both guitarists play both sides?

This is something I’d really like to get some opinions on, because I want to make sure I understand the recording process correctly.

Our music is quite guitar-driven, and my second guitarist and I both have our strengths and weaknesses as players. We can play the same parts, but we’re not necessarily identical in terms of timing, picking consistency, and overall tightness.

During our first recording session, we both recorded rhythm guitar parts, with one guitarist playing one side and the other guitarist playing the opposite side.

The result wasn’t necessarily bad, but it didn’t sound as tight and cohesive as I had hoped.

I’ve been wondering whether it would be better to have just one guitarist record the same rhythm part twice, with one take panned hard left and the other hard right, instead of having each guitarist record one side.

My understanding is that double-tracking works best when the two performances are very consistent, and that differences in timing, attack, and note length can sometimes make the guitars sound less focused.

So, here are my questions:

- Would you generally recommend having the tighter guitarist record both rhythm takes, especially for the main riffs?

- Is it better to have each guitarist record both sides, even if one player is slightly less consistent?

- Could we use the tighter guitarist for the main rhythm layers and have the second guitarist record additional layers, harmonies, or different voicings?

- How much does the choice of guitarist actually matter compared to editing, re-amping, amp tones, and the overall arrangement?

- At what point does editing become too much, and how do you retain some natural feel without sacrificing tightness?

I understand that having two different performances can add width and character, so I’m not assuming that one guitarist recording both sides is automatically the better solution. I’m just trying to understand what tends to work best in modern, tightly produced metalcore.

1. Any other production tips for a guitar-driven metalcore album?

If you have experience producing modern metalcore, I’d love to hear any additional advice you might have.

For example:

- Common mistakes to avoid when recording rhythm guitars.

- How to make guitars sound huge without simply adding more tracks.

- How to balance guitars, bass, synths, and programmed drums.

- How to make breakdowns feel heavier without relying entirely on sub-bass drops and sound effects.

- Any useful references, tutorials, plugins, or production breakdowns worth checking out.

I’m not trying to copy a particular band or achieve a completely artificial, hyper-edited sound. I just want our album to sound competitive with modern metalcore releases while retaining our own identity.

Thanks in advance to anyone willing to share their experience! I’d really appreciate any practical advice, especially from people who have actually recorded and produced this kind of music.

To cut to the chase, I work at a commercial recording studio and more recently have gotten more work doing voice overs. Particularly ones with outside producers or team members for audio books/video game stuff.

I’m just curious if anyone else who frequents this space more often has any thoughts/preferences on how they route these sorts of sessions? Mainly thinking about easier ways of getting video call talk back into the clients ears, and getting DAW audio (I use pro tools) to the remote guests. I’ve tried a few different options and it’s not rocket science or anything. It just seems to me there might be a super efficient way to get it done that I may not be doing.

I mostly specialize in full band tracking and song writing spaces, as does the studio owner. Since I am the one working the projects I am limited to what software the studio has.

Thanks for any insight!

Im obsessed with this EP especially with its vocals sounding like no other artist other than Imogene heap and wanted to get a similar vocal chain in logic

I recently got a Zoom H5 Handy Studio field recorder as a present along with a 3.5 mm male connector to dule XLR aux cable. In my free time, I will often hang out with friends on a conferencing client called Team Talk, the only such client that will transmit high fidelity stereo without an extra price tag. Just because I could, I decided to hook my Zoom to an old Echo Dot, create an extra instance of Team Talk, plug the Zoom into my USB hub where my audio equipment lives, and set the input of the second instance of Team Talk to my Zoom in Audio Interface mode. In this way, my friend group could torture my Echo device and we could hear it perfectly through the setup.

Yes, I know this was unnecessary and simple in it's way. However, it was a fun signal flow experiment, and I have always loved playing with new toys in interesting ways. My fellow engineers, what's your "I definitely over-engineered this because I could" story? I know I am not the only nerd up in here.

• I have an untreated 8ftx10ft bedroom with a 2ftx5ft closet.
• I'm running a Rode PodMIc and an Audio Technica AT-2035 through an Audient EVO 8.
• I like to record YouTube covers for fun of J-pop (Kenshi Yonezu, Natori, Imase), vocaloid, and artists like Radiohead, Sufjan Stevens, Keshi, and Local Natives.
• I would also like to possibly do voice overs for fun
• I notice my voice sounding super sibilant with both mics.
• The PodMic is angled 45° from my mouth a fist away, at mouth level while the AT-2035 is a shaka brah fist away at mouth level (I stand to use it) and pointed at my mouth. I'm not sure if it's the mics, mic placement, my voice, my technique, improper gainstaging, or my room not being treated
• I was thinking of making my closet into a blanket fort (ignore that I wrote "curtain"), but don't know if this would help get me better recordings of if it's too boxy?
• I was also thinking of getting a SM58 to combat sibilance
• I also listened to the recordings from this person and thought the blanket over head, fabric cube, and fabric cube with blanket hanging behind sounded good. However, I hear that the box will end up sounding boxy, and isolation shields don't do anything/much?
• Would something like this alone help out or would I also need the treat the top and/or cover all of the sides?
• Any cheaper alternatives are welcomed. Again, this is just for fun. I'm new to singing and don't expect to blow up and voice acting would be cool, but I just want to make my recordings easier for my mixer

Curious how people handle this:

You start with one EQ, then another for a resonance, then a third for air… and suddenly you’ve got 3–4 instances and no clear picture of the combined response.

Do you:

• leave the stack and move on
• bounce/print and re-EQ
• manually rebuild into one instance by ear
• something else?

I got tired of guessing at the sum, so I made a small free browser thing that pastes TDR Nova Copy States, shows the summed curve, and lets you match it back into fewer bands.

Happy to share the link if that’s allowed / useful. Mostly interested in how others solve this workflow, and if it is a real problem for anyone else.

Hello everyone, what's the most difficult concept to grasp in audio to you?

I've been making records with artists for over 10 years (I also have a background in physics). There are some concepts that seem basic to me but are difficult to understand for artists AND some concepts that seem convoluted to me but are obvious to artist.

I started posting on socials this summer, I talk about my vision on making records, my tastes, some music but I also try to demystify some concepts. I'm really inspired by some of the posts in this sub. It's giving me a fresh perspective about what needs clarification and also interesting points of view.

For example, I think one of the most misunderstood topic in audio engineering for music is: LUFS.
A few years ago I was producing an EP for a band. I delivered the master but they insisted for me to print a version at -14LUFS. I explained to them in length that it was not what they wanted. But it's hard to argue against "that's what Spotify says!". After them insisting I delivered them one version at -14LUFS and my proper version of the mastering.
The morning of their single's released they were all texting me about their song being way quieter than everything else on the platform...

That was years ago but I keep seeing so many misconception about LUFS online. I understand this topic is very confusing. So without going in detail about measurement (not everyone has worked as metrologist) I tried to demystify that topic for artists and made short video about -14 LUFS in my socials.

What are other topics or concepts you think are not well understood?

Might be a silly question but I’ve been struggling for so long to get a good acoustic guitar mix like this one.
song

Also, how do you guys think this guitar is mixed?

How do you commit to processing on your track? A distressor for example.

I’ve always changed the output of the track with the distressor insert on it to “create a new track”, and then I just arm that track for recording and hit record. Done.

Someone told me I should be routing it to a bus and then create a new track and make the input from the bus..

Why the extra steps? My way seems simpler.

original post and video: https://www.reddit.com/r/livesoundgear/comments/1x3i7wk/update_2_my_virtual_wing_running_the_real/

Follow-up to https://www.reddit.com/r/livesoundgear/s/htqXGb10zX about a software WING that runs the console’s own firmware (WING Rack 3.1.1) instead of a re-creation of it. The goal is still a real virtual sound check at home: play back a multitrack recording through the actual WING’s processing on my PC, and mix it on the console’s own interface.

Last time I ended with a list of what didn’t work. Here’s where those stand:

• No audio → it plays sound now. Audio goes through the console’s own emulated input and bus DSPs and out of MAIN 1 on my speakers. My Scarlett comes in as local inputs (didn't show in the video), and a WAV loops on USB 1/2 (in the video).
  • When I move a fader on WING Edit, the firmware itself writes the new value into the DSP that’s making the sound.
  • Which channel hears which source comes from the firmware’s own routing.
  • EQ and the compressor work too. Changing them on the console’s screen changes what I hear, because the console’s own EQ and dynamics run on the emulated input DSP.
• No network → WING Edit connects. The console answers discovery, and WING Edit talks to it like a real WING.
• About a third of real time → the console runs at about half real time. The audio DSPs run at about 0.6×, and that’s the catch: the sound has gaps. The audio itself is correct (an offline render of the same DSP state comes out clean); there just isn’t enough of it per second yet.

One thing to know while watching: when I click out of the command window (the emulator runs from there and isn't shown in the video) into the browser to change something, the sound gets noticeably worse: more gaps and more crackle. That’s not the EQ or the compressor doing it; it happens just by switching windows. Windows gives the window you’re using more CPU time than background ones, and the emulator is already short on speed. Most likely it loses just enough when it’s in the background to get worse. With the command window focused, it’s only gaps.

What’s still missing:

• Speed. Each DSP runs on one CPU thread flat out, while the rest of the CPU sits mostly idle. The next step is spreading each DSP’s work over more cores.
• The FX DSPs in the live path. They boot and run, but FX inserts are silent live for now.
• The XMOS and the FPGA’s audio side still aren’t emulated. My own code moves audio in and out of the DSPs where the FPGA would, and it takes the routing from the firmware itself.
• No comparison against a recording from a real WING yet.

The setup is the same as before. The i.MX6 ARM application runs on Unicorn. The four ADSP-214xx SHARC DSPs run on an interpreter built for this project, with a C version checked against it instruction by instruction.

What’s new is that the DSPs answering the console are the same ones making the sound, each in its own process. Between blocks of audio, each one takes the console’s control messages through its own firmware, exactly as it does on the real hardware.

Hi, fellow subredditors!

I'm in need of some moral guidance.

We all know the plugin market is full of grifters, sponsored YouTubers who seem to love absolutely everything, questionable marketing tactics, and increasingly, AI slop.

And I really don't want to become part of that.

That said, through the power of AI, spousal neglect, and relentlessly pestering my developer friends, I've built a few plugins of my own.

I mainly started because I was dissatisfied with the workflow of established tools, particularly when songwriting. I got tired of disappearing down an engineering rabbit hole every time I was just looking for a sound. I wanted tools that let me experiment creatively without constantly interrupting the process.

Some of these plugins are getting close to being commercially viable, and I'd like to release them at affordable prices.

But affordable pricing also means I need to reach quite a few people to make this sustainable. And that's where my dilemma begins.

So I'd love to hear your thoughts:

• Which YouTubers or reviewers do you actually trust for plugin recommendations? Who do you actively avoid?
• Do you consider sponsorships, affiliate links, or free review licenses inherently fishy? Where do you draw the line?
• What's the last plugin you bought, and how did you discover it?
• What makes you immediately distrust a plugin developer or brand?
• What could a small, unknown developer do to earn your trust?
• Do you ever click on plugin ads, or is that just money down the drain?

Bonus question: What would you like plugin developers to stop doing altogether?

I'm not here to promote anything, and I'm deliberately leaving out the brand and product names. I'm genuinely interested in how people here feel about marketing in our industry.

Since I already have a decent day-job, I'd much rather build a small side-business people respect than become another company shouting "GAME-CHANGING ANALOG WARMTH" at everyone.

Cheers!

Given all the recent developments with AI reverse engineering of software, do you think it’d be possible to convert PT sessions to Cubase, Logic or DAW Project?

I extracted the vocals using ai and i am trying hard to figure out how they achieved this, i hear some flanger, huge reverb but still the vocals remain upfront and the there are doubles ads and even low octave underneath the main …can someone help ?

https://youtu.be/FJ63j_90BfY?si=Z2RTanMJ9KK5S-nJ

Here's what I wanted :

1. Sit at a coffee shop, and plug in my VSX into the Macbook headphone out
   
2. Have all system audio + DAW going through VSX
   
3. Be able to sample off YouTube pre-VSX processing

This has proven to be the messiest possible thing, for what seems like such a simple request. Having to make aggregate devices, routing in Audio Hijack, setting System Audio + DAW Audio, every single time I wanted to simply sample the system audio.

This little utility fully solves this, with a single click. Audio is duplicated pre-VSX, and routed to the device input as a microphone/loopback. This allows running your entire system through VSX, and in your DAW the ability to record dry system audio without any external tools.

If anyone is interested in testing this, please send me a message. I'm currently sitting in my daily coffee shop, recording YouTube directly into my DAW, hearing everything through VSX but capturing dry!

I was testing some clippers and i made that discovery, sorry for the sound it might not be your cup of tea, but i found that very interesting :

https://streamable.com/6c5ebm

I’m considering a new kick drum/bass mic, and the sE electronics V-Kick is on the list. I know that microphone EQ switches that cut low frequencies can increase headroom, but I’m wondering if EQ switches for changing tonal characteristics generally, like on the V-Kick, impart any distinct tonal difference on the captured sound compared to EQing it afterward in a DAW. Is it entirely a matter of convenience dialing in a sound, or is having the EQ done by circuitry within the mic itself, closer to the raw capturing of the sound, somehow advantageous to doing the exact same EQ adjustments during post-processing?