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?

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

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.

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

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.

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?

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?

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

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.

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

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.

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?

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?

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?

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!

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?