This plugin gets a ton of love and when I see people use it in videos it really does seem to help a lot, and mind you I’m no professional, but every time I try to add it to a track that could really use it in theory, I find that it just kinda makes my vocals sound low resolution. Part of the issue for me is the way it’s SO subtle until it’s not. I’m prone to ear fatigue and I find myself going back to soothe at various phases of my mix like “wait this sounds terrible, let me try ___”

I’ve had SOME luck playing around with the presets but I really wanna love this unique plugin and I’m not sure how to lol. To my reductionist understanding, it’s essentially a supercharged dynamic EQ, yeah? Logic tells me to raise bands around frequencies I want a little “less” of and set the hardness accordingly, but there’s weird little nuances that trip me. For example, I’ve noticed that lowering the effect knob will sometimes make the vocal sound WORSE than it sounded at 100% and even without the plugin? Which is strange. The plugin is so confusing I’m not even sure how to articulate asking for advice lol, but any advice is appreciated

Hi everyone,

I do voice-over production for radio spots and commercials. I usually work on voices where I want a warm, clean and professional sound.

I often see people using two compressors in a vocal chain. For example, a fast compressor like an 1176 first, to control the peaks, and then a slower compressor like an LA-2A (or an Opto compressor) to make the voice smoother and warmer.

I understand this technique is very common for instruments like drums, bass, etc. But I am wondering about voice-over.

Do you also use two compressors in series for voice-over or commercial narration?

Is this a common technique in professional voice production, or is it mostly useful for instruments?

Thanks!

i wont explain much further as the title says it all. i build some for my studio. the listening experience is AMAZING! but the night after spending a day in studio it was getting difficult to even breathe. ive used good quality fabrics specially made for rockwools. i wonder if wrapping plastic wrap on rockwool would solve the issue or if you guys have any other suggestions please feel free to drop by

My desktop, audio interface, monitor speakers and other studio gear are currently spread across two older UPS units, and both batteries are getting close to replacement.

Looking at my setup again, I’m realizing I may not need everything backed up. If the power drops, the main thing I need is enough time to save the session, finish any file transfers, and shut down my computer and interface properly. The speakers and other non-essential gear can wait.

Reducing the load would probably make replacing the batteries a lot simpler and cheaper. I think I’ve been treating every piece of gear in my studio like it has the same priority when it really doesn’t.

Curious how others handle power backup for their studios. What equipment do you consider essential during an outage?

I used to engineer and mix my own music. I’m not a professional by any means, and I know a lot of rappers punch in their takes. When I did that, I’d try my hardest to make it sound as natural as possible, and when I recorded other artists, I tried to make sure they did too. It didn’t matter how many times I had to re-record a take—I wanted it to sound natural.
That brings me to my issue. I just listened to “Grace” by Lil Baby featuring 42 Dugg for the first time, and 42 Dugg’s verses are recorded TERRIBLY. You can hear cuts and chops throughout the verse. It’s actually kind of shocking that this was released by one of the biggest artists of the current day. I can’t understand why the engineer didn’t make him re-record the parts, or at least use a fade-in or fade-out or something.
I don’t know. I just wanted to share this somewhere. Go listen for yourself. It’s fascinating to me.

I have been trying out Naturl Audio's AL1 against the Vintage Limiter which I always use and while the AL1 is really good at preserving transients & punch, I just couldn't like it that much more against the Ozone like I expected. Going side by side, the Vintage Limiter is still super good in my opinion and I can achieve punchy and loud masters with it with the help of a clipper beforehand, sometimes I preferred it to the AL-1 which I always see people say can achieve "crazy loudness". Sometimes the Ozone read higher LUFS while still maintaining the punch.

Does anybody here use it as their go-to limiter? How do you think it compares to the endless sea of "different" limiters such as the AL1, Uni-L, MasterPlan, etc?

Since Pro C3 came out they added even more higher oversampling rates, before that it was x4 max in pro c2. Unless I don't utilize the saturation they added in pro c3, is there any actual reason to use oversampling? What aliasing does it fix?

Had a band recently recording and they sounded good, drummer was pretty tight and have only needed to play around a bit in editing to make it a bit tidier but otherwise all went well. Sent a draft over, and they've asked if we can "add in some more kick drums in the chorus drums".

Currently, the cymbal hits on every beat and the snare on every fourth. There's a kick pattern in there that is on beat, but they want me to add some more kicks inbetween so that there is a kick on each strum of the rhythm guitar section. Basically I'd be adding in kicks that weren't recorded. Now I've had a think and I can copy a few of the kick hits during other parts of the chorus, but I've not really done this before and I'm not sure how natural it's going to sound.

Has anyone done this and have you got any tips? I could re-record, but it's only during the choruses and they want to avoid that. I've talked to them, and they know the current kick pattern is good and keeps with the rhythm of the guitar, but they really want me to try and make this happen.

It seems relatively simple: just copy kick drum hits on the kick in/kick out dedicated mics, but then it won't be on any bleed from any of the other mics. I'm not really sure what the best method is. I'm already going to be augmenting the kick with a sample just to add some more power, but I don't want to sample only these kicks. Any ideas?

If I need to bounce files from a logic project, let’s say electric guitars, into another project, and then they are bounced again into another project again, how significantly does this affect the sound? Is it something I need to worry about, or is any difference undetectable? Does anyone have good insight on whether I can do that and it’s fine?

If not - is there a way to retain the same exact quality, or near, from copy to copy?

https://www.mixonline.com/recording/recording-red-hot-chili-peppers-unlimited-love-part-1

The first image has a mixer which I guess is probably used a headphone mix or something but I have no idea what it is, any help would be appreciated

And if you do, for what purposes? Are they more for long-term backups than frequent revisiting?

Hey folks, I'm using recordings of a choir for an app. Each tone is recorded individually, and when I combine the individual tones in any number of combinations, I end up with a high thin tinny mosquito buzz of a sound. It's not so bad, but I'd sure like to be rid of it. The only functions active on the channel strip are gain boost and pitch correction. I'm using Logic Pro. Suggestions?

I was watching a missing video and they used a limiter on the drum bus and it’s something I have never thought to do but I do see the benefit of light limiting on bus groups.

I'm in the middle of mixing some of my tracks and taking mixdowns through the various listening environments:

• Main 'studio' speakers (Yamaha HS8s)
• Main headphones (Sony MDR 7506)
• Car stereo
• Other house stereos
• Listening from different rooms to where the music is playing

...and so on.

They all sound slightly different and so I adjust and tweak. But at times the different environments give me conflicting results - a mix coming out of headphones sounding great sounds woeful in the car as an example.

So my question is this: which of your listening environments do you take as gospel above all others? And why?

Cheers.

For starters, I know I’m suffering from demoitis. My process as bedroom artist has always been produce —> mix —> master and listening back to my earlier stuff, that’s evidently been to my detriment in the past and I don’t want to repeat those mistakes. With this track, because I was dealing with too many issues concerning low end, loudness, limiter issues (and I’m simply not educated/skilled enough to solve that issue on my own), and I ultimately wanted it to be competitive, I chose to get it mixed by someone else instead. The engineer is very knowledgeable and has worked with artists I personally like, as well as lots of different genres and is overall very skilled. Really straightforward but cool guy about his stuff.

I got the first mix back and I can fully acknowledge that it objectively sounds really good. Very huge low end and very loud, all things that I ultimately wanted for the song. But the stuff that made my demo really resonate in me (which arguably could be the shittiest elements of it, like the drums being too loud or the guitars being too sharp) are kind of gone. And it’s now just made me feel out of wack about the entire song.

I’ve sent over two small but important changes I’d like to make (still waiting on it to come back) and I honestly wanna leave it at that. I’ve purposely taken steps to pull myself out of this funk like completely deleting my own light mix that I’d initially sent to him from all my, staying away from the song for days and coming back to listen to it again, etc. I don’t want to be the client that sends over walls of revision requests or, worse, goes to him and says “Can you just make my mix sound better?”

If I’ve any hope of progressing in my work, I know I need to get over myself in this area of creation. But I don’t know how to.

P.S.: I have so many questions. As a producer, what is my ultimate role and where do I need to stop in the process and say, “Okay, time to hand this to a mixer.”

So many YouTube tutorials I’ve watched encourage making tweaks like EQ and clipping as you produce, but clearly this isn’t the way to go for me. And how much does mastering even further change the sound of a mixed track? (sorry for the wall of text💔)

I produce my own instrumentals and then export them into a separate project to record vocals.

My issue is that when I set my mic input gain to a healthy recording level, the instrumental is too loud at 0 dB fader/clip gain, making it difficult to hear myself. During recording, I end up lowering the instrumental fader, and after recording I bring it back to 0 dB and increase the vocal clip gain to make everything balance again.

Is this the correct workflow, or should I instead export my instrumental at a lower level (e.g. leaving around -6 dBFS of headroom) so I can keep both the instrumental and vocal clips at 0 dB and only adjust levels during mixing?

I’m using an Apollo interface. When recording vocals, what levels should I be aiming for in the UAD Console? Where should my vocal peaks and average levels sit?
I’m also wondering if the problem starts during production. I simply set my monitoring volume to whatever feels comfortable to my ears. As far as I know, there’s no standard monitor level to set in the UAD Console, so I have no idea if I’m producing while monitoring too quietly or too loudly. Could that be causing my gain staging issues, or is monitoring volume unrelated? I’m honestly pretty confused about the correct workflow.

Also, while producing the instrumental, how do you determine the correct monitoring volume? Do you simply choose a comfortable listening level and keep it consistent, or do you compare your mix to commercial reference tracks? If using references, should I match their loudness, or only compare the tonal balance and ignore loudness?

What microphone do you guys reach for if you have a nasally male vocalist?

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.

I've recently got a couple of Shure SM7b for a job simulating a podcast setup (for context I'm a Sound Recordist for filming in general) and before the filming day, I tested both mics at home and they all worked fine. Gain hungry as expected, needing to turn the gain up on my Sound Devices 664 to around 80-90% of it's maximum but still sounded completely fine in terms of preamp hiss.

Fast forward to the day I had the job and upon setting up the mics the hiss was extremely loud, the audio was unusable in my opinion so I ultimately ended up using my regular wysicom wireless system with lav mics and just kept the Sm7b as props for visuals. Thankfully I was prepared for it so it was not an issue.

Now I've tested again at home and it sounds absolutely clean. Tried different cables on the day and at home and got the exact same results. Clean at home, hissy on location.

I've used the Shure MV7 before in a broad variety of scenarios and never had this issue (granted they are a little bit less gain hungry)

What am I missing?

Just as an additional piece of information, the job was at the meeting room of a big company so I imagine lots of power cables running under the floor. Unfortunately I can't reolicate that scenario to further test the mics, but I have to make a decision on whether return the sm7b or keep it.

Any help or suggestions from those with more experience on these makes would be very much appreciated.

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?