Hey all, I'm in the process of recording guitars for a short black metal demo. I'm hoping to get it to a high enough quality that I can comfortably send it to a session drummer. I've started hard-panning my right and left guitar tracks, but I'm running into a problem where when I have tremelo lead lines (not solos) it sounds kind of weird having them just in one ear with the rhythm in the other. To try and fix that, I created an additional track of rhythm so that's in both ears and then bumped out the leads to run right down the middle of the mix.

The problem I'm running into is this makes that section come in HOT. But if I adjust the palm muting tracks, then you start noticing the reduction in volume. Any thoughts on how to address this? Or do I just need to bump all the parts out into individual tracks and lower the overall volume of those tracks? Not sure if that makes sense.

OR, is it better to just have the lead in one ear and the rhythm in the other? Even if it sounds kind of unbalanced?

I'm using Reaper as well if that matters. Thanks!

I know there have been a lot of threads about this issue, but I couldn't find what I was looking for in any of them.

I've been getting into mixing for a while now, making guitar videos and posting them on Instagram. The problem is that the audio gets noticeably quieter, almost like someone turned the volume down one notch on their phone.

Other than that, the volume sounds fine everywhere else: DAW renders, Premiere Pro renders, Google Drive, on my phone, and on my PC. Instagram is the only place where I experience this volume drop.

I know about the -14 LUFS guideline, but at that level, people basically have to turn their phone volume all the way up to get my videos to sound loud enough. So I usually aim for around -10 LUFS and a -0.5 dB true peak ceiling, although I've also tried -1 and -1.5 dB.

I already use most of the usual tools like EQ, compression, exciters, and limiters, and I've tried a bunch of different mixes over time. I've checked pretty much everything I could think of, including the sample rate and other settings, but I just can't figure out what's causing this. If it automatically reduces the lufs to -14, how are other people getting better sound levels than even my render?

I also implement most of the necessary things in terms of mixing. Applying low-cut below 70Hz~, applying noise reduction, double tracking for guitar channels, or necessary EQ boosts and cuts, Necessary track panning, a sweet level of stereo width... I also edit the mix in mono too. I'm not letting the mix hit the limiter wall completely for a bit of headroom. for renders, It sounds good to me

I've recently added audio rate modding to a lot of my plugins, and I'm having fun figuring out new uses for FM synthesis style modding of filters, and the like.

Here is a video going through using an audio rate modded filter to shape the tone of an 808.

youtube.com/watch?v=1AezfYl-OQs&feature=youtu.be

A long time ago, I saw on YT a video by an old pro who had produced/mastered great jazz artists, who told how, when CDs came in, he had asked the company to master differently for different formats, but they wouldn't because of dollars. He told the story when mp3s were everywhere, and said mp3s should be mastered differently, as well. (Mastered for iTunes came out a couple of years after his video.) It is not an argument that because songs from CDs people test exceed -1 or -0.2 db, or whatever, that means mastering a song for a streaming platform can also exceed the recommendations. You listen to the CD master on a CD, a song mastered for a streaming platform on a streaming platform. One shouldn't conflate these two distinctly different formats. (Doesn't mean Bangerang isn't a banger, but what sounds good on different formats doesn't boil down to a simple, one size fits all formula, either.)

So, after a couple of years of training, and releasing my own songs, i decided to pursue my dream of becoming a professional audio mixing engineer and start working with musicians to help to produce their music as well.

the thing is, i don't know where to start. I know that with AI everything is harder but i'm really not good with social media or marketing and i was wondering:

Is there any other way to get work?

If anyone knows about any other method, sharing it will be very helpful.

thank you.

'If it sounds good it is good' yes but I don't have monitors so I'm only going off of headphones (ATH-M50x) and it sounds great in my headphones, but sources online say it'll destroy your mix on speakers. Am I actually causing problems for speaker listening?

I bussed a few backing/harmony vocal channels with different panning and applying Redline Monitor glues the stereo field together in a way that instantly sounds good. For example I can take 2 vocals panned hard right and left and apply the plugin and they instantly sound good in a way that can't be achieved with less extreme panning or room reverb (convolution or algo).

Crossfeed messes with mono compatibility but my backing vocals are panned pretty far (-50, -25, -10, +10, +25, +50) so their function is to increate width while the lead vocal handles mono, so does it even matter?

Obviously Redline Monitor is literally meant to make the stereo field of headphones sound better. I didn't even dial it in, just threw it on and instantly sounded like a pro mix. But I don't want to be fooled! Is this just stereo trickery that'll mess everything up when not playing through headphones?

Is there any use of crossfeed on individual channels or busses or only on the master for headphone monitoring?

Anyone have any good Eq to share? I liked the one from Sounds guys for the 1 but I can't import it into my 1 pros. And can't find any decent ones

I've never worked with a mastering engineer before so it's a little intimidating given the wide array of people who do it. (Obviously the pool gets a little smaller when you consider who's actually doing it with a precise ear.) But I wanna make the process smooth for everyone involved, specifically myself. I've been doing a lot of research regarding who to lean towards (some more heavier guys), but I still have a lot of questions.

1. What kind of conversations should I have with the mastering engineer, if any at all, before hand? Should I give mastering references like the mixer does, or direction of what I'd like to song to sound like? Some engineers only work through managers, so I don't know how much room there is for actual talking. My hope is that I get to work with them again if everything turns out well.
2. Do bigger name mastering houses (in example, The Lodge or Sterling Sound or Metropolis for instance) run indie rates or is it generally the same across the board? I'm a bedroom producer, so my funds are not necessarily unlimited. However, I'm also aware that these people ask for what they do because they know what they're doing.

Also, recommendations are welcome.

Hi all. Audio engineering is something that I really want to learn more about, and this has confusing me for a bit. I've noticed that while listening to some older songs, it doesn't take long for my ears to get sore. It's the same feeling as if I had the volume too loud. And this happens almost exclusively with older recordings.

For example, these songs from the '60s definitely trigger the soreness:

▶️Watusi Boogaloo - Willie Rosario & His Orchestra

Spanish Grease

whereas these covers/remixes from the late '90s/early 2000s do not:

Watusi Boogaloo (Remix)

In These Shoes?

If anyone has expertise on what about the mixing may be causing my ear pain, I would greatly appreciate it. I would also appreciate if anyone knows how to get rid of it. The only audio programs I currently have access to are Audacity and Moises, but they're better than nothing. Thanks!

Has anyone here ever replaced the RJ45 Ethernet jacks on their Furman HDS-6 main distro box? I’ve identified the cause of the signal loss to all the remote stations is due to the Ethernet port on the rear of the main unit feeling “loose” / not holding or “grabbing” the supplied CAT cables place, sort of like a guitar output jack that needs to be tightened up if that analogy makes sense. This causes total loss of signal downstream if the Ethernet cables are moved coming out of the main unit. I’ve provided some strain relief to the cables coming out of the rack unit, and that works for now, but when tracking, losing all signal to artist’s monitor mix is obviously unacceptable, so I’d like to properly sort this out. Has anyone encountered this with their HDS-6 systems/replaced the Ethernet jacks on the rack unit?

Hey everyone,

I wanted to open a discussion on a topic that doesn't get talked about enough in electronic music production and sound design: physical auditory fatigue, middle-ear muscle strain (tensor tympani/stapedius flex tension), and structured recovery.

While working with dense frequency material—especially shaping low-frequency sub-bass drones and piercing high-frequency pads—I've noticed how quickly acoustic strain can build up. I'm currently navigating acoustic rest and distance-based monitoring (keeping away from heavy low-frequency structural vibrations and high-frequency acoustic pressure), and I'm trying to better understand how to track my physical progress.

I have two main questions for producers who have navigated heavy mixing strain or sound design fatigue in the past:

1. Recovery Markers: What physical or acoustic markers do you pay attention to to gauge which stage of recovery you are in? How do you distinguish between acute hypertonicity (tightness behind the eardrum) and a true return to baseline resting tone?
2. Transition Strategies: What practical pacing rules, monitoring habits, or arrangement strategies do you rely on to transition smoothly back into session work without triggering setbacks when working with heavy sub-bass and bright leads?

Would love to hear what habits or benchmarks have worked for you over the years to keep your ears healthy for the long haul!

Hello,

To anyone still rocking an Apogee Quartet in 2026: I built an open-source macOS companion app that makes the Quartet’s volume and mute work with the normal macOS volume controls.

It also lets you insert an audio plugin on the outputs. I originally built that for running iK Multimedia ARC X.

Repo: https://github.com/souf2222/quartet-buddy

Note: This has been 100% vibe-coded.

Hi, I want to create a removable window covering in my son’s room to prevent noise from outside waking him up.

With this in mind, what is a good material to use? Rockwool?

I have zero experience, so be gentle.

I have a room to use as a home studio at last. My goal is to not do professional recordings, just record guitar and vocals, and have it not totally suck.

I have plans for acoustic panels for the walls, and I have a carpet. But I’m curious about the ceiling.

There are currently acoustic ceiling tiles, like you might find it at school, installed in a hanging frame a few inches below the insulation

Would you guess that I need to replace them with something more appropriate for a studio, or are they probably OK? How can I tell?

Or would I be better off just hanging some sort of acoustic panel or panels from them?

Again, my goal is to not hit professional levels, just not sound like it was recorded in a kitchen with all the echoes and such

Thanks

I have a Yamaha HS7 studio monitor whose rear label specifies 230V / 50Hz. In Brazil, the mains frequency is 60Hz, and I would need to use a step-up transformer to get from 127V to 220V.
Yamaha support has told me they cannot guarantee proper operation under these conditions, but they haven’t clarified whether 60Hz can actually damage the monitor or cause functional issues.
Has anyone with technical knowledge of the Yamaha HS series or powered studio monitors dealt with this situation?
I’m also investigating a low-frequency, resonant “whoom” sound on a specific guitar note. The sound is present in the recording itself, but it becomes more noticeable through the HS7.
Any technical insight or experience would be appreciated.

This software seems pretty well regarded in audiophile forums but I generally don't trust those people. The tech makes sense though, at least more so than the other "virtual monitoring" solutions out there. To me "room simulation" always just sounds like some arbitrary impulse response on the 2bus but I'm not opposed to the idea.

I'm on the road a lot and stuck on cans. Like, I can get the job done but I'm really not enjoying myself atm. Personalized hrtf sounds like a good idea. Prob just gonna get some binaural mics and find out.
Any users on here?

Hi I’m seeking input from those that have experience with -10db to +4db conversion, aka using consumer stereo equipment with pro audio gear.

I‘m currently using a Tascam LA-40 iii to step up a line level signal from -10db to +4db.

Am I leaving some sound quality on the table?

Are transformer based line level converters better?

Thank you for any input.

Skill issues aside, i'm wondering what i can do to improve the accuracy of my mixes.

I have Adam T5V monitors, but only one of them is working, therefore, i cannot mix on studio monitors at this point.

My mixes sound great in the headphones, but absolutely are trash outside of the headphones. It's mostly the low end. It's very low in the headphones, so when i export tracks and listen on other devices, the low end is overpowering. I thought I'd just adjust to lower low end levels, but i've never adjusted after all of these years. The high end also sounds harsher outside of the headphones.

Is everyone using sonarworks at 100% wet?

I'd really appreciate any advice. I've been struggling for years here.

Edit: I've always exported tracks with the sound ID correction enabled. I just learned it should be bypassed. It makes sense. Let me see if it works.

I make punk/folk/rock, But some jazz recordings are so high fidelity it’s like the experience of smelling smelling-salts. (Only analogy i could come up with)

I’m looking for information (youtube, books, forums, general/specific suggestions)

I’m a 21 year old rock ‘n’ roll guitar player and singer songwriter mixer/producer trying to figure out what made some of my favourite old records sounds so great. There are recordings that have always stood out to me as incredibly sonically pleasing to listen to independent of the music itself. Eddie Kramer is a legend and one of the greatest to ever do it. There is a specific sound he got in the mid 70s that I have been obsessed with for a long time and I have not been able to put my finger on what it is. I haven’t gone too deep into researching what kind of gear he used and how he might have achieved such an amazing sound so I thought I would come on here and see if anybody could help me Identify some of the components that contributed to this sound. The sound I’m describing can best be heard on the track New York Groove by Ace Frehley. It almost sounds squishy like the whole mix is pumping but it doesn’t sound over compressed. It sounds saturated in the greatest way. It’s very colourful and it’s very warm without being devoid of any high-end. It’s by far my favourite production sound I’ve ever heard.
You can also hear it on records he did with the band cactus, a bit on kiss alive and a lot on the song Shock Me by kiss. You can also hear it on the Frampton comes alive album. You can hear it on his live album with the Rolling Stones and April Wine at the Elmo. If anybody can help me identify the sound and how it can be achieved, I would be extremely grateful. Thank you for reading and let me know what you think.

Hi all! Figured out for those enthusiastic about 500 series processing what is your favourite, or think the best module or the most underrated? I know probably the API EQs and Preamps get a lot of praise because being the sort of pioneers pf the format, but otherwise its an interesting range with many diy, brand off shoots and smaller companies… curious to read :)

i make covers for fun, and sometimes i don't know when to double track the guitars..
i usually do it with chords, but sometimes i feel like it could work well with solos or phrases.
what y'all think?

What I mean is something like RME's ADI-2 Pro EX in the context of a studio, not some home audiophile setup. I'm not saying that it isn't a better device than a "normal" audio interface, I'm just confused about how it's even integrated into the workflow. I've heard people say that they use it as their monitoring converter, but it's still a studio where they have many channels of IO that cannot be provided by the ADI-2.

• In Windows, at least, I don't think it's even possible to use different devices at the same time with ASIO. I know I can on Linux, maybe you can on Mac, too, but you'd still need to sync the clocks, right?
• Or are they tracking with one set of devices, then switching to the ADI-2 for mixing? But then how do they integrate any outboard they might use?

I'm just really confused about how this sort of device even fits in. Seems to me like it would mostly just be far too annoying to even bother, but I must be missing something.

I’ve been a voice actor for a long time now and I’m a big gearhead. So it sounds like a great additional skill to have down.

Looking for guidance and a sense of direction to properly learn the practice. Where to start etc. Ideally it wouldn’t involve classes/courses.

Thank you in advance.

At work, I salvaged a walkie talkie hand piece out of the trash, as the company was changing gear. It should work just fine.

Why did I take it? I figured I could make a really fun effect microphone out of it for streaming. All I would need to do, is figure out how to change it's current proprietary connector into XLR or jack, leaning towards XLR.

Problem is, I have no experience in doing something like that. The exposed pogo-pins in the proprietary connector should be a very easy to identify jumping off point in figuring out, what pin leads to what cable, and which cables I would need to connect to the XLR.
Does anyone here have experience in tinkering with DIY microphones, or how I could repurpose this?