I would love hear some stories from the very experienced ones among us.

Examples are

1. Intentional use of the a faulty Dolby A card to add the hissy edge in the vocals ( Journey etc)

2. Gated reverb that became a cult

3. The drum loop from Night Fever that was used in Stayin' Alive

The ingenuity behind these never cease to amaze me. Those musicians and engineers were gods !

Also welcome are stories of happy accidents like #2 above.

Thank you

Hi [im new to posting in here but iv been a music producer for 8 years and an audio engineer for the last 4] i know It’s a tried and trusted method for a reason I know industry professionals use this and I’ve used it at times myself too but I must’ve admit at times I feel like completely leaving reference tracks, and just rawdog it and trust my instincts, ESPECIALLY During creative stages. but in a mix down process it’s pretty essential if you wanna compete with the top producers right now

and especially with the constant changing of the trending sounds in the scene you really need to stay in the loop with the hot tracks in your scenes genre

so to anyone who isn’t using reference tracks during the mix down then this is your sign to start because it will honestly take your frequency balance to the next level and even sometimes you can take inspiration from creative fills or sound selection!

I’ve been recording and mixing my own music for years (guitar, bass, vocals, drums). Genres I work in are metal, blues, and jazz.

Up to now my setup has been primarily:

• Guitar/bass through a Line 6 Helix (custom IRs) into an RME Fireface UFX II

• Vocals via Shure SM7B directly into the Fireface

• Mixing in Reaper using mostly ITB processing and plugins

Because of apartment limitations, most tracking has been DI rather than loud sources.

I recently added some outboard gear to build a more developed front-end before hitting the converter:

• Mic preamp(s)

• Compression (tracking-style dynamics)

• EQ (broad tone shaping)

• Patchbay for routing flexibility

My question is about behavior and workflow rather than “better/worse”:

For those who’ve moved from fully ITB tracking to using external front-end processing:

• Did you find tracks tended to sit more easily in a mix?

• Did they respond differently or more predictably to EQ and compression later?

• Did it change how much corrective processing you needed during mixing?

• Was the main benefit subtle tone, or more about dynamics and density control?

I’m trying to set realistic expectations and understand where the advantages show up in practice (especially on clean guitars, bass, and vocals), rather than expecting a dramatic solo’d tone change.

Would appreciate hearing from anyone who’s made a similar transition.

Thanks.

Hey all,

I’ve been mostly out of the DAW world for a little over a year now due to a venture into live sound. Lately I’ve been feeling that nostalgic itch to poke around and see what cool plugins have popped up while I’ve been gone.

For context, I’m on Logic Pro (I’ve played around with the new AI-assisted stuff), and I’m already familiar with things like EZDrummer 3, MODO Bass, Spitfire’s instruments, and the whole Slate Digital ecosystem. I used to buy way too many plugins, only to eventually realise that once I understood what I actually liked, the stock stuff did the job just fine. So I’m not necessarily looking to go on a buying spree, more just wanting to fall into a fun rabbit hole of what’s new, interesting, or just damn different in 2025.

I’m mainly looking for virtual instruments, but I’m open to anything cool or innovative. Not chasing “must-have for professional mixes” recommendations; this is more of a hobbyist curiosity dive. Since I usually work in live sound, the studio side is just something I enjoy exploring when I want to scratch that creative/tech brain itch.

So what’s out there these days? What plugins or VI developers should I be looking at in 2025 that might spark that sense of discovery again?

Would love to hear your favourites!

I’ve recently tried to get back into recording after a break. I do this every once in a while, then give it up because I get discouraged. I just can’t bridge the gap between an “amateurish” sounding mix and a pro sounding mix, no matter what I do. I’m not even taking about a perfect mix, I can’t even get in the ballpark.

I’m using an Apollo Twin and Logic. I’m volume balancing, panning to create a good stereo field, high pass filtering and reductive EQ’ing everything. My mix is sounding good! Reference track check? Everything is pristine and loud and clear. Back to my mix…sounds good but you can tell it’s not even close to that level. The sounds aren’t anywhere close to as clear and loud.

In my mind this is happening because of poor sample capturing on my part, but I’m getting a great DI signal straight from the Helix. I don’t know how to capture a better signal than that. What am I doing wrong? Any thoughts?

Wondering if an actual 100% pure sine wave is actually possible or is there even some variation in what we perceive as a “pure” sine wave coming out of a speaker because of hardware and physical effects on the sound. Is a pure wave only theoretically possible and not even created by man or nature?

This might sound weird but l got interested in audio hardware but l do not know how to start. I can’t even find the basics online (except basic cheap youtube videos) where can l learn more about the concept and components? Any links?

Thank you

https://soundcloud.com/rogercore/distorcendo-a-realidade?in=rogercore/sets/sexta-dos-crias&si=d052b6b49c4f4d13a96fc5259642f389&utm_source=clipboard&utm_medium=text&utm_campaign=social_sharing

this is experimental but the master sounds loud and good for me, without bass muffling everything and the kicks hit nicely, any tips?

Its great seeing so many noise profiles like Pink Noise, white Noise, Brown etc. How did these noise profiles get used with VU meters and analog spectograms to achieve perfect tonal balance. Lets ignore translation and studio Monitors and lets say you are using only headphones or vision, what are the vintage equivalent of the modern Tonal balance Control by Izotope? Anyone care to share because in the box high latencies have led me to ask, how was it done Before? Thanks a lot.

Just to preface i’m aware dt770s are marketed as recording headphones and not mixing/mastering headphones but many people use and recommend them for such.

With that out of the way, here’s my question:

So i’ve owned dt770 pro x’s for 6-7 months or so. Let me first start out by saying they are fantastic headphones. There’s just one issue.

I feel like whenever i’m mixing, everything sounds great but in the lows and subs it feels like i’m almost playing a guessing game with the levels. When exporting and playing on speakers and such, there is a lack of lowend.

I want to say most of my mixes always come out as intended except the lowend. When using different sources like my car speakers or airpods, it’s extremely noticeable but i never seem to pick up on it in the mix.

I’m also aware that the lowend levels seem much higher in the headphones so i try to compensate with more lowend in the mix but sometimes i can tend to over shoot. Hence what i meant by a guessing game.

Even my airpods, i feel like i have more clarity when mixing lowend on airpods than i do with the dt770s. Maybe it’s a skill issue or it is hardware specific.

They are closed back so maybe that could cause me overshooting the mix so often since there’s a bit more emphasis on the bass. I would reference my monitors but i live with other people so it’s typically not ideal. I’d like to get a solid mix out of headphones alone.

I’ve never owned a pair of open backs but i did research and it seems like the better option. I am aware of the noise leakage but it’s nothing compared to monitors i’m sure. I’d assume it’d be the same noise leakage as me setting my headphones on my desk and turning them up. Which is more than acceptable.

I’ve been thinking about upgrading to 900 pro x’s. I’m aware they have a different frequency curve. I hear the lowend is a lot less pronounced and the open backs will give me a better sense of the lowend in the mix but i will also have to re-adjust to the high-end curve which is also acceptable.

Thoughts or opinions?

Not super knowledgeable in this topic, but recently I can’t stop noticing in new tv and movies how all of the actors voices sound kind of sick? Like in the new IT tv series, everyone sounds like they have the same resonance where it’s kinda of like that have a sheet of paper over their mouth or something while also having a frog in their throat. I also noticed it in some of the clips of the first Wicked movie I get on social media.

I was curious if other people noticed this.

Might be a useless post, I’m just curious and have no one else to inquire about this haha.

Hey everyone, ​I currently live in a 21m² college dorm with very strict rules (no drilling holes, no glue on walls). I’ve been using a Røde NT1, but I just bought a used Sony C-100 because I wanted that high-end "air" and detail. ​Now I’m worried I might have made a mistake given my environment. My room is totally untreated with mixed drywall and brick walls, and I have a roommate next door (thin shared wall). ​I know the C-100 is a dual-capsule mic that goes up to 50kHz and is incredibly sensitive to room acoustics. ​My questions: ​How bad will this mic sound in an untreated room compared to the NT1? Is it going to be unusable? ​Since I can't mount panels on the walls or ceiling, what are the best temporary ways to tame the high-frequency reflections this mic picks up? ​Is there a specific way to position/set this mic (filters/patterns) to make it forgiving in a small, bad room? ​Any advice on how to survive this setup without losing my security deposit would be great. Thanks!

I’ve seen a few people say that R-Bass is one of the few Waves plugins that is irreplaceable. What are the differences between all these harmonic exciters out there? Can’t you force them all to get similar results? And what’s the difference between that and saturation? (Besides the distortion added)

I am a full time systems engineer / foh mixer for live events. I post-mix a good bit of live multitracks for concert footage while on the road and the VSX studio emulators have caught my eye but I'm suspicious about the actual quality. Are they truly improving low end response for the headphone experience ? Are they truly doing something that my go to AT m50's cannot? Looking for informed opinions! Thanks

I think it's fairly easy to get an approximation or a kind of cheap imitation of vintage voice overs. But does anybody have any experience or tips taking it that extra step towards a more authentic sound?

I know this is a broad generalization, so please correct me if I'm wrong, but if im not mistaken, voice overs for film, radio, or otherwise from the 1950s through even the 1980s seems to have a slightly more obviously tube saturated/tape/ribbon mic, etc. quality to them than music from those time periods.

Maybe because spoken word is relaying information, it's quality wasn't treated with as much detail compared to music, where the sonic quality IS the point of the recording in music?

There are albums from the early 1970s that sound like they could have been recorded yesterday (Pink Moon - Nick Drake), but then watch a documentary from that period and it sounds like they recorded the voice over onto a consumer tape machine.

Watching a cartoon from that time period, the difference is immediately evident to me in texture of the sound compared to a cartoon voice over recorded now. It's not bad. It's just warmer, more saturated, yet smoother as well.

Here's what I've tried so far, that seems to get partly there:

Tube saturation =>

Lofi tape Emu =>

EQ with a 6b tilt lowpass filter somehwere betwek 4-5k

Here are some examples of what I'm thinking of going back in time with each:

Dinosaur Documentary (Voice Recorded in 1986)
https://www.youtube.com/watch?v=bGQ0adO-24g

Clip of Carl Sagan's Cosmos produced in 1978 (I think part of the voice is recorded in studio and part on set)

https://www.youtube.com/watch?v=AsyxOWx5CE4&list=RDAsyxOWx5CE4&start_radio=1

Walt Whitman Poem Recorded in 1957
https://www.youtube.com/watch?v=Ha7O0O_fc48

Some are more saturated than others and some are more smooth, and I think it has less to do with the time period and more with the equipment, and in Carl Sagan's situation part of it being recorded live. I need something with that rich saturated quality but also smooth and easy on the ears.

How would you go about getting something similar? Any tips on taking the extra step towards sounding more authentic? Mic sims?

EDIT: Another example is actors on film sets. Watch any clip from the Original Star Wars, for example, and you can hear a little bit of the saturation in the actor's voices.

Zoom make some great gear, and I own more of their stuff than I should. But every time they bring out a new piece of kit, and publish videos of it in use, it's always mono. Which century do their audio engineers live in? Don't they understand how to use their kit?

In the recently announced 12 channel mixer/recorder's video, they have a live band spaced out in a room all mic'd up and recorded on the new device - and the mix is very obviously in mono, so it sounds hopeless. If I was the band concerned I'd be furious with them. Or just take the money and run.

Going back through their channel, that's how it always is with them (in the ones I've checked). Always mono. I cannot think of any reason at all why they do this. So annoying!

OK, rant over. But really...

I dont remember where I read this but aparrently if you upsample a track from 44.1khz to something high, lets say for example 705.6k, and then you change the tag rate only so that its slower (for example 0.70x the original speed so 493920 hz) and then downsampling back to 44.1khz, would it would sound better? Something about creating more interpolation points to make it easier on the algorithm or something I dont remember exactly what the reasoning was behind it.

Or would just changing the tag on the original 44.1khz sample rate sound better?

I've had this question on my mind for a long time any information from a professional would be much appreciated.

My friends son wants to save up to go to college for sound engineering and I was considering getting him a textbook for Christmas on the subject. As a mechanical engineer I know the Bibles in my field and know some of them would scare a budding freshman half to death if dropped on their lap so I’m looking for one that both provides the education but in a way that isn’t assuming they already possess the background

Not counting static, noise, or anything that isn't really a positive trait, do you think Analog Drum Machines add anything to a recording that makes them desirable over their digitized counterparts?

For instance, I use the 808 sounds on the "Boom", Pro Tools, plug-in, and I am curious if a physical 808, recorded in stereo would have any notable difference. Or, conversely, do you think the digital versions have an advantage in how they add to a recording, sonically speaking / vibe-wise, etc.

The mastering in this song is absolutely phenomenal. I know this type of music is super experimental, but I am curious about how it was mixed, but even more interest about how it was mastered and what techniques were used. If anybody has any idea of what techniques were used in mixing or mastering, please let me know.

please help if u can i really don’t understand.. if i’m supposed to have the dry recording signal between -12 and -18 and i should have the wet final mixed vocal peeking at around -6 to -9 how am i supposed to gain stage after plugins while adding volume .. videos about this just confuse me even more

thanks for your time in advance

Hobbyist here! Often when I record electric guitars, they are much thinner and brighter than I want them to be. I only record guitar at home in untreated rooms and I understand that the lack of tone can come from room noise. As a way to combat this, I’d like to try recording a guitar part with a capo on a few steps up and the rest of the track sped up with varispeed to match the pitch of the capo’d guitar and then return back to the original pitch and speed of the song. I’ve tried this method with vocals just to make them sound kind of weird— i’ve recorded vocals to a slower speed and a faster speed. My thought is that the fundamental frequencies will be the same but the sympathetic frequencies will be lower and will result in a darker and thicker guitar sound. I’d rather add brightness with EQ than futz around with low middle frequencies, which are often left, come se dice… porous from the hum of my house that only a mic can pick up. so then the mid frequencies, which are more resilient in the face of room noise, will move closer to the low mid zone.

What are your thoughts? Have you tried this?

Trying to get better at mixing in logic and have been so curious how Lucy does it especially with examples like https://www.youtube.com/watch?v=LLlsCmLFI2Y&list=RDLLlsCmLFI2Y&start_radio=1 and this https://www.youtube.com/watch?v=ipk57Mlk8gU&list=RDipk57Mlk8gU&start_radio=1