r/audioengineering

I have recently received some multitracks to mix. Eight people (2 per mic, 2 per voice SATB) and one stereo pair of them all together (XY). Its supposed to be some church hymns accompanied by piano and organ. For some reason, hardware or software related (unknown) all of the vocal tracks have been recorded with 11kHz LPF. To make matters worse, from 10 recorded songs, 5 of them have been recorded with the LPF while the other 5 are "normal" and have the whole frequency range. I have tried mixing them both and the difference is quite audible.

I have tried saturation (both on inserts and parallel), various exciters, iZotope Spectral Repair with little to no success. The most success I had was Ozone 12 Exciter making it sound the most natural however when AB testing the mixes, there is still quite a difference.

Any advice is appreciated!

As many people have already suggested, you shouldn't mix your own songs. And I do understand the reason behind it.

Bias, attachment, and you tend to overthink everything. Which is why it's best to get a fresh set of ears to approach your song.

But how would you handle creative difference within the band?

We have three audio engineers, me included. And everyone just seems to have a different take on how we should mix our song. We decided to get a different person to eventually mix it, but I am afraid that our ideas would still clash.

If thats the case, is the engineer have the freedom to choose which direction to go? Or is that the producers job?

Granted that we started the band to actually have some fun with creating music, but I am afraid that having too many chefs in the kitchen would bite my ass and would just call it quits.

Hello everyone!

I recently saw a lot of producer use tape saturation on their drum bus (in dnb and house tracks) and i found it really adds warmth and glue. So i downloaded toTape 9 and chowtape (i think it is the name?) and tried to do the same but it dosnt sounds good at all!

I just wanted to know how you guys use that effect?

When i use saturn 2 for exemple i can keep it very subbtle and just stack 3 of them with very little drive and it works wonderfully (strangly more than one saturn with more drive), but with these plugins it sound like lacking of high end and too obvious.

I know they are highly praised plugins so i might not understand how to use them

Hey everyone,

quick context: I've been into metal for years (huge fan, been playing guitar for about 15 years), but I've never actually produced or mixed metal music before. My production background is 5 years of trap beats so I know my way around Logic Pro X and audio production in general, but I've never taken a proper course or lesson on mixing/mastering. In the trap world I never really had to dig into it myself. Now that I'm fully focused on metal guitar (recording covers, working on original riffs), I'm realizing I have zero real foundation in mixing, and I want to learn it properly instead of just guessing.

I want to start narrow: just guitars first, before even thinking about mixing a full track. My goal is to really understand tone shaping, gain staging, EQ carving, and how to make a DI'd metal guitar (I'm using Neural DSP Archetype: Abasi a lot for example) sit right in a mix, clarity, low-end control, that tight/aggressive modern metal sound.

I honestly don't know where to even start, so any direction helps. Specifically:

• What's the right starting point for someone with little to zero mixing training who wants to learn metal guitar mixing specifically and general mixing theory?
• Any structured resources/courses/YouTube channels that actually teach metal guitar mixing step by step?
• Which stock Logic Pro X plugins are genuinely good for shaping metal guitar tone (EQ, compression, saturation, multiband)? I keep seeing people recommend third-party plugins, but I'd like to know what's already possible with Logic's stock stuff.
• Any free plugins worth trying for metal guitar mixing? Doesn't have to be Logic-only.

Eventually I'd like to work up to mixing/mastering a full metal track (guitars, bass, drums, vocals), but I want to build a real foundation on guitars first instead of jumping around without knowing the basics.

Any advice, resources, or plugin suggestions are hugely appreciated!

I've been watching the World Cup and I'm curious about how they mic the pitch. I see the mics placed around the periphery, but I'm curious if anyone here has any insight on the details of how they capture the kicks and manage the crowd noise and what not.

OK,

So I have a bit to bring forward here, so sorry for the long post

#1 - I quit upgrading my iZotope everything bundle sometime after NI took over because of the stupidity that NI exercised

#2 - I'm concerned that in 5 years or so, Boris will retire, and we go thru stupid stuff again

#3 - I really only use Rx.

#4 - My cat can master better than Ozone10

#5 - I have other high quality tools that help me down the path

But when it was announced that Boris FX was buying iZotope, I reconsidered purchasing the update to my everything bundle. When I saw that it was $269 to upgrade, I pulled the trigger

My initial impressions

- Now $269 is not chump change, but I thought it a reasonable cost for what I was getting. The reasonable part is what has been missing form iZotope in the NI days. I'm just glad I could get the upgrade, and NOT have to use Native Access.

- Ozone 12 Ai actually does a decent job, and my cat is a bit weary.

- I just started doing audio books, and the little I did with Nectar is promising

- When I walked away from NI, I created mastering chains in Fabfilter. But it looks like my workflow will change again. I'm liking Ozone12 results

So, this is my $0.02

Hey everybody,

So I recently started working in a studio and sometimes see recording engineers and artists have the sm58 lips close while recording. I was wondering how the mouth/air sounds in some sessions destroy recordings and in others sound full and clean even before I dive in their mixes or clean the takes up?

My solo experimental experiences have been always messed up with plosives and harsh *Hs* all through.

Would love to hear your thoughts on this!

I’m curious to hear from people who have actually used both.

I had a pair of Sennheiser HD 560S with SoundID Reference including Virtual Monitoring. I knew SoundID already had room simulation, headphone calibration, and translation checks.

Still, I returned the HD 560S and bought Slate VSX because I assumed the dedicated hardware/software ecosystem would be a more trustworthy mixing solution than a general headphone profile + virtual room system.

Now I’m wondering if I overestimated that difference.

The HD 560S sounded very good to me as open-back headphones, while VSX is a more expensive closed-back system. Since SoundID already offers headphone-specific calibration, Virtual Monitoring, and more translation checks, I’m trying to understand what the real-world advantage of VSX is today.

For people who have used both:
- Is VSX still meaningfully better for mix translation?
- Is the room/imaging realism noticeably ahead of SoundID Virtual Monitoring?
- Would you choose VSX over HD 560S + SoundID VM if you already had that setup?
- Or is the difference mostly workflow/preference rather than a clear upgrade?

Not trying to start a brand war. I’m genuinely looking for practical experiences from people who have mixed on both.

Recently I recorded a track through my cheap USB mixer's interface. I didn't hear the clipping on my output but very quickly learned that I had clipped the interface into oblivion.. But I loved it. The extra layer of noise washed over the whole mix really took it to the places I wanted it to go. But obviously the digital clipping isn't the most tasteful and I'd like to replace it with something put in my FX loop that has a little nicer of a crunch to it, but still won't colour the mix too aggressively. What options do I have here? I know I could use a tape distortion plugin but I need to run it in a no laptop live rig. I'm considering trying the tape delay preamp in the Zoom multistomp. Any other recommendations

Or your top three things? I'm brand new to the field, and I'm sure a lot of heads here have forgotten more than I'll ever know, but I'd like to know what general direction to go because it's a vast subject.

Was there ever a conclusion to which song won? What’s the loudest song you’ve ever heard?

Any help would be really appreciated:
The film was shot in the basement bar of a post-war NYC building. It's narrow and long. In this scene, two people are talking, with background bar noise and music coming from the speakers. How do I get the music to sound like it's being played in this kind of space in ADOBE?

I managed to get the effect in DaVinci but cannot replicate it in Adobe to save my life. Please help.

THIS is what the bar looks like.

I saw lots of engineers saying it's useless since the quantization errors will be so low due to the bit depth dynamic range of 24, yet if you do dither, you will fix the quantization errors but get very low hiss. I saw both in practice, what's your take on this?

So use UVR5 for ripping audio files apart into Vocal and Imstrumental using Demucs model.
And i noticed that one of the old GPUs that i own is crazy fast comparing to any other GPU i tryed. GTX 1080 can rip audio file that is 2h long in 15-20min when GPUs like GTX1660 RTX3070 RTX5070ti do it in about 2.5-3h or mostly fail and give errors about not enough system memory.

I have no idea why GTX1080 is SO GOOD. Maybe because of 2 decoder chips but 5070ti also has 2 decoder chips onboard,

Maybe someone could explain?

I've been using it for a while but it has been slow for me so what model do you recommend using for fast conversion im really busy

sorry I meant vocals i was not awake

> Hey everyone,
> I’ve been playing drums for over 40 years, played in countless bands, and recorded in dozens of studios. Never in my life had my timing or setup been an issue. Until a year ago, when I joined my current project.
> For the past 12 months, all we talk about is "latency" – and it is **exclusively** about my drums.
> Here is the setup: The bassist plays through a highly complex, digital multi-effects unit. The keyboarder runs 5 different synths (Nord Stages, Roland, Moog, Vintage) with various internal latencies and external hardware delays straight into the board. I am running my electronic kit/setup through a Zoom L-12, taking a completely analog cable signal out of the headphone jack into the main mixer.
> Yes, my pedal has a digital latency of maybe 2 ms. Yes, acoustic drums (when we micro-manage individual stems in solo mode) have natural air-travel latency to the overheads. But when you listen to the full mix, the tracks sound completely tight and round.
> Yet, the bassist (who spent 18 years as a part-time sound engineer) and the keyboarder constantly dissect my individual tracks under a microscope in "solo mode," complaining about milliseconds. Now, we are choosing a external mixing engineer, and before he even heard a single proper take, he is already talking about "potential issues with the drums." It feels like a herd mentality at this point.
> Honestly, I am exhausted. It feels like I’m under constant surveillance, and it completely killed my joy of playing. When I called them out on it, the bassist sent me a wall of text lecturing me about transients and delay compensation, basically saying "I know your setup better than you do."
> Am I losing my mind here? Is it normal for modern DAWs and digital-synth-heavy bands to put the entire blame on the drummer just because a drum hit has a sharp transient that is easy to point at? How do you deal with band members who act like absolute studio-scientists instead of just playing rock 'n' roll?
> Appreciate any insights, because right now, I'm ready to walk away and take a long break.

Thanks.

Hi all, Beside my head I have my mic processing rack with a bunch of preamps, voice strips, EQs, etc. I'm in a very quiet room, and the transformer buzz is audible, which is undesirable for my kind of work. I've been wondering if anyone ever tried converting their analog racks to DC input, and how that went. All the racks I'm talking about are internally running at +15,0,-15V rails, so I could just have a single linear (or even high quality SMPS) power supply, distribute it via XT or XLR style connectors, and have simple filter boards inside each unit that filter the incoming DC, with switch to disconnect the mains PSU and connect the DC board. I was wondering if this is a good idea in general and in particular how this relates to issues like ground loops and EMI pickup. My run from the DC power supply to the rack would be about 2-3 meters long, but it would be running parallel to some digital cabling. I would also be interested in suggestions on connectors to use, especially if I want to make daisy chain cables that go from device to device.

I would appreciate suggestions / experiences on the topic.

Thanks!