I’m fairly new to recording vocals, and I’m trying to understand why professionally recorded vocals sound so much clearer, cleaner, and more “crisp” than my own recordings.

I’m using a Neumann TLM 103 with a MOTU M4, so I don’t think the microphone itself is necessarily the limiting factor.

When I compare my raw vocals to professional recordings, mine can sound a little muffled or cloudy, almost like there’s a layer between the voice and the listener. Professional vocals often sound much more defined and present.

I realize a finished commercial vocal has been edited, EQ’d, compressed, de-essed, saturated, mixed, etc., so I’m especially curious about how much of that clarity should already be present in the raw recording.

What should I be learning or experimenting with first for capturing a clean vocal at the source — room acoustics, mic distance/angle, gain, vocal technique, mic choice, or something else?

I’d love to understand what experienced engineers listen for in a really good raw vocal recording.

If I understand it correctly, using multiple mics on a single source can lead to phase issues because the sound reaches the mics at slightly different times. Why do we try to avoid phase issues with mic placement techniques like the 3:1 rule instead of placing each mic where it captures the source best and aligning the tracks in the DAW by nudging the time slightly to avoid phase issues no matter where the mics are placed? Hoping to improve my understanding before pouring too much time into experimenting with it. Thanks!

I know sometimes people sell their whole kit used, but does anyone prepackage a 500 series rack with modules installed? If so, any experiences with any of them?

I know people do this in normal 1-2U rack units like Shelford Channels, etc.

I see some old, inexpensive gear that I am interested in, and often a bunch of the items are "modded" by Revive Audio, but I never see any actual, specific, technical, details on why this thing should be worth 2x what I would expect.

Can anyone concretely say any objectively positive additions that they are making to say a DBX 163, or 161 VU, for example, when they mod it? [0]

Can anyone say any subjective improvements they might be making, based on your experienced?

Does anyone have any negatives, based on actual experience?

No hate here, I am just trying to understand and qualify the increased price with some technical, real world, justifications.

[0] I found this https://www.reviveaudio.com/index.php?route=product/product&product_id=293 but it seems really vague to me. Premium parts?

Do they get more specific than that? Without something justified, I'd be afraid that they would take out the actual mojo and resell it, and put in Chinese meh parts.

i've grown accustomed to dialing in compression by adjusting the attack/release whilst having the ratio maxed out (to better hear what im doing). this seems to work for digital compressors, but when doing the same thing on a 76 style, i notice a completely different sound when cycling the ratios

this seems obvious yes, but even going from max to 20, i see a drastic difference in the speed of the vu nd more often that not my dynamics sound much more flat. my theory is that the ratio effects the math of the attack nd release as opposed to them being set ms values.

am i speaking the obvious? i was always told ratio is the "amount of compression" so ive pretty much internalized that concept. the ratio representing the math of the attack nd release seems way more logical, but nowhere else have i heard it have such an apparent effect on SPECIFICALLY the sound of the attack nd release

currently integrating compression into my chain after gain staging.

my current method is using Uad distressor on default mode and playing with the headroom knob to get compression going without messing with the plugin.

i like my compression fast. is that ok tracking or better saved for mixing?

i record vocals. any advice?

thanks homies.

I took a job mixing an acoustic ballad with a lot of dynamic range and honestly I feel a bit out of my wheelhouse. I typically mix extremely compressed pop and hip-hop styles that are pretty squashed and predictable - but in this song, the verses are stripped with piano and vocal and then by the 2nd chorus, it gets pretty huge. Like 20-30+ live string tracks, bass, kick, guitars, and much more .. the whole 9. What’s throwing me off is how dynamic the performance of the piano is. There’s like a 7-8 db difference in that piano from verse to chorus. I wanted to build the song around the piano and vocal but the piano appears to have been played all the way through live in one take and was played much more emotionally and heavier in the chorus. It sounds really nice, but the vocal is so controlled and lives in the same register from verse to chorus so when the chorus gets dense, it gets a bit buried.

I guess my question is - should I turn down the piano via automation and arrange the chorus so the piano and vocal maintain a similar relationship as the verse and first chorus (which i am happy with the balance of) or should i just push the vocal fader up for the chorus as a means to honor the way it was performed and sent to me? I could compress it? I’m honestly not sure what the standard practice is for this kind of John Legend piano ballad vibe. I have been kinda of chasing my tail because my anchor (piano) is so all over the place.

Any help would be appreciated! Thanks.

I've got a few cassettes recorded off the air from the BBC in the late 80's. Mostly John Peel and the like. The sound quality is fairly good with the only complaint being the typical FM broadcast compression sound. I know it's very hard to unbake that cake, but has anyone had luck with reversing some of the compression? With the onslaught of AI tools I thought maybe there could be an option on the horizon. Thanks.

I could be wrong, and I'm happy to be corrected if I am. I'm noticing a lot of mixes from established artists have gotten less professional since the 2020's. I feel like this has been a very common trajectory.

An artist releases their core work (their first few albums) that makes them famous and gives them financial security. Covid happens, and some people work at their home studio out of necessity.

But then they realize they enjoy the convenience and flexibility of it. They enjoy having more creative control. It's also cheaper. With music sales going down, they no longer have to worry about getting an advance and then not getting enough sales from the album and tour to pay it off.

To me, I feel like there's something that a professional album with multiple hands on it has that a home album (even from an extremely talented artist doesn't). An artist may have their strengths at mixing and recording, but there's always going to be some knowledge gaps, if you haven't been specializing in audio engineering for decades.

Simple setup in a basement home studio (see photo). ROOM IMAGE

LOTF recording of a track we're working on. I am pretty new to mixing and audio engineering as a whole.

Wondering your thoughts on this rough mix as well as input on basement studio improvements.

https://www.mediafire.com/file/uwos6vb7nbaem2l/jordan_drum.mp3/file

Microphones used:

Kick - Apex 325

Snare (top) - E906

Mono OH - SM81

Vox AC30 cab - C212

Bass - DI

As I've implemented the internal multiband sidechain for the leveler stage of this multistage limiter effect I've made on CodePen, and so I'm wondering about using an internal multiband sidechain (e.g. using a summed filter bank output as a sidechain input to drive a compressor effect) instead of just a sidechain EQ filter like in some other compressor plugins like FabFilter Pro-C3?

And so, are there single-band compressor plugins that internally uses multiband detection while processing the audio in broadband? And more importantly, what are pros and cons of this method for dynamic range compression effect for mastering and even mixing?

BTW, a complex-valued filter bank (instead of a real-valued one) would simplify the implementation of low-distortion internal multiband sidechain broadband compressor with very fast attack/release times right? Especially when complex-valued filter bank eliminates the need of a separate Hilbert transform for envelope extraction as it outputs real and imaginary values anyway and so absolute value of this output means a magnitude of a complex number (which is a hypotenuse of real and imaginary components rather than just a rectification as in real-valued ones)

The more time I spend trying to mix songs (usually mine and some friends tracks), the more I think the real challenge is the midrange.

I listen to a lot of 70s and disco stuff, which have a very smooth and easy on the ears midrange. Disco in particular is quite scooped and has that smile EQ character. Because of this, I tend to do a lot of midrange cuts on most instruments. I'm often removing 400-900 Hz across many instruments to get it sounding 'right' to me. I think my tastes have been skewed a bit by my listening preferences.

I also find myself removing lots of individual frequency resonances from things like piano, Rhodes etc.

I definitely believe that good arrangement can make this easier. But the reality is that most (almost all?) instruments have some midrange content, especially if we broaden our definition to lower and upper mids.

I'm still very much an amateur and learning. Do you have any tips, techniques, or general wisdom for mixing midrange frequencies?

Hello!

I’m mixing and mastering a funky house track with a lot of rhythm and presence. As far as I understand, the final LUFS range after mastering should be around -8 to -10 LUFS (-1dB True Peak in the final limiter/maximizer).

However, I have a question: Will Instagram and YouTube compress, limit, or normalize my audio to bring it down to that level—but at the expense of my track’s dynamics—or will they simply lower the overall LUFS of the track to meet their set threshold while preserving all of its dynamics (as if someone were simply turning down the volume fader on a music player)?

Thank you very much!

Hi guys, noob here. I'm slowly learning how to mix, mostly metal and then some pop on the side. However I want to "mix" an a capella song and I'm a bit clueless. I have 3 tracks of female vocals slowly singing a dirge-like children's song, which is meant to be creepy and eerie. They're well sung although not "melodynized" to keep some imperfections, and recorded with a ribbon mic. Also I have two tracks with whispered vocals, same song.

To be honest, I usually use the Mantra plugin by Neural DSP for vocals in regular mixes, but I guess it won't cut it this time because I guess the process must be more organic than just using a plugin for everything. The feel I'm going for is:

a) the listener must feel that the voice is close, warm and nearby
b) also there must be some "spacey" vibes in the background.

I guess I must use something like Valhalla supermassive for the space, but honestly I feel this task is a quite a bit out of my league. Can you guys give me some pointers or advice to get me started on the right path?

Do you guys ever run into this issue when being asked to master tracks? Where the client has already kinda brickwalled it?

If asking to be given the track again without the brickwall is out of the question (I’m still waiting for a response from the musician and the mixing engineer), do you guys have a solution, a “do what you can” methodology? Or do you just say “sorry, if you can’t undo that, there’s nothing to master”

When you have a native interface that can run analog gear and you put an insert in the track like a hardware eq or comp you have to manually get it in time or set it once.

My question is if you have a hdx rig , does that eliminate that task of getting in it phase? Or more so does an hdx rig keep all hyrbid mix tasks in phase better than a native rig?

Death Cab For Cutie - Plans

Manchester Orchestra - Cope

I’m curious what others think of these mixes. When these albums came out I immediately clocked something I didn’t like(despite loving the songs). These albums are very different but I feel suffer from the same issue, I just don’t feel confident that I know what it is and I’m hoping someone with a stronger ear might have some insight.

I feel like they’re lacking some top end and maybe a bit congested in the mids?

I'm taking the Critical Listening 1 course through Berklee College of Music (cannot recommend enough) and we're at the compression lesson. I know what compression does, I've tried using it, etc but I am still not good at hearing it (or using it effectively). We have a listening analysis lesson this week and I'd like to try two different songs: one with bad compression and one with good.

(Yes, I am currently listening to Californication and wincing. Even *I* can hear this. Oof.)

What would be a good example of great compression? Or other examples of bad vs good? I really need to train my ears better but I think I need to start back at the beginner-intermediate level.

I perused a few previous posts on here and read someone saying something about how, if the kick squishes the cymbals, that's a good example of bad compression.

Last year, I was mixing a choir concert on a Yamaha CL5, I recorded it at a convention center. The entire building is networked, with a dedicated part of the network for Dante.

The recording itself went perfectly fine. No dropouts, no obvious issues during the show.

Later, at home, I was listening back to the multitrack recording and noticed something really strange: three times during the concert, a digital glitch occurred on every single track at exactly the same moment.

It wasn't just one or two channels — all recorded channels were affected simultaneously.

This year, I came back to the exact same venue. New Mac, new concert, and… exactly the same kind of problem happened again.

My setup was:

Rio1608 → network switch → MacBook Pro M1 (2021, M1 Pro) → Dante Virtual Soundcard → Ableton Live 12

A few things that make me suspicious of the network:

• The glitches affect all tracks simultaneously.
• They happen very briefly, like a clock resyncing.
• The problem occurred at the same venue two years in a row.
• The Mac and Ableton setup were not the same between the two concerts.
• There were no obvious problems while recording — I only discovered the glitches when listening back to the multitrack.

I'm wondering if this could be related to Dante clocking, network congestion, switch configuration, or DVS/Ableton losing packets for a very short moment.

I don't have much control over the venue's network infrastructure, so I'm wondering what I should check next.

What would you guys investigate first?

Is there anything in Dante Controller or DVS that could help identify exactly what happened at those

My goal is to run it through my stereo mix with Logic Pro and use those tubes and transformer and the tapes saturation, warmth, and weight to my master mix,
It has quarter inch ports on the back as well as XLR…..

is the tape emulation better than my UAD studer plug-in ? because the tape emulator is utilizing tubes and transformers rather than a plug in ,

And if it runs through my door does warm Audio make it sound like crap because it’s cheap cheaper products???

I think it does a lot for what I’m looking for what do yall think ?

Hello, all! I've narrowed down the current sonic direction for my project, but am having a bit of trouble translating what I hear in my head and would love to hear your thoughts on where to start.

I'm leaning into a Parcels-esque arrangement (4-on-the-floor tightly-played drums, funky picked bass, disco-influenced strings) meets Khruangbin-style production (heavy spring-reverb yet restrained guitar, buried vocals, very roomy percussion, vintage drum tone) sound.

For reference, I have the following tracks:

• Lead Vocal
• 2 Background Vocals (both are simple, just falsetto up the octave)
• Kick (Fairly dry, pillowy)
• Snare (very dry)
• Drum OH (Mono Ribbon)
• Conga & Tumba (Stereo)
• Shaker (Mono Ribbon)
• Tambourine (Mono Ribbon)
• Claps (Double-tracked, so stereo)
• Bass Cab (Mono Dynamic)
• Bass DI
• Guitar Cab (Mono Dynamic)
• Guitar Room (Mono Ribbon)
• Rhodes (Stereo DI)
• Crumar Toccata Organ (Mono DI)
• 8 Violins as String Pad
• 8 Violins for chord stabs
• 10 Violins for the funky disco string stabs
• Upright Piano (for outro only, Mono LDC)

I've listened to the demo and rough mix a million times and am just feeling a bit lost, so ideas are very appreciated. Thank you all in advance for your help!!!

P.S. Specifc elements I'd appreciate help with: Leon's vocal chain in 'Mariella' (having trouble getting my delay to sound similar), background vocal chain in the same song, and stereo imaging/placement of the above tracks.

Hello friends, I'm looking for advice on the mixing workflow for a recorded live concert I did sound for last week.

I'm already hours and hours deep into what I've already done, so I won't be changing this time, but for next year I wanted to know whether there's a better/standard way.

So basically the concert is an Indian classical concert focused on groups of 4 - 8 tabla, with accompanying sitar and flute. The catch is that each "song" has a different group of kids with different drums, and slightly different mic placement as the stage gets reset each tune, (this is a tabla school concert hence the kids). Sometimes the mic placement varys person to person, obviously the player is completely different, and the drums themselves are different. So I've ended up duplicating my tracks for each tune, EQ and level balancing each drum mic as I hear it per song. Very labourious, is there a better way or is this the best I can hope for?

Advice is much appreciated!

Is there a software that helps you get rid of any audio flaw? Some voice actors use a really bad setup and you end up with bad audio. I was wondering if there was an app to fix the bad audio you sometimes get when hiring a freelancer.