u/rinio

If you want to de-ess on the way in, you buy a hardware desser. Simple as that. If you have a hardware multiband compress lying around, you can deploy it as a pseudo-deesser.

As you've intuited, mic selection and placement play a big role.

What you havent mentioned is EQ as well, but this is often a trade-off.

And performance plays a big role as well; no performer is going to never need deessing, but for most amateurs/hobbyists, this the root cause. Ofc you cant do anything about this unless you're also the vocalist.

A pop filter place little to no role in controlling siblance. We may as well consider it a complete non factor for this.

Seems solid. Some of the concerns I had with you previous post about the pre-built are addressed.

AIO is still an AIO. Up to you whether thats a problem for your audio workloads.

The nondescript "major brands" would be highly concerning to me. The "major brands" that would come to my mind for an high performance rig would be ~70% of your total price estimate for 64gb in 2026 retail prices. Do with that what you will.

Ad hominen and making things up about me is childish. Ive spoken only about the content you've written, not attacked you. The only one taking anything remotely personally is you in this thread.

I guarantee you many of those engineers are laughing at you behind your back. When you accompany it with a solo vocal, we know you are using it literally and not as a "shorthand".

Even used as a substitute for vocal *processing*, which is what I guess you mean by the "shorthand", any experienced engineer, like many others here, will tell you that it does not matter one iota outside of the full context. It can sound awful and harsh as a vocal, but great in the mix. Absent the rest of the product, it may as well be silence.

In my initial reply, I gave you actionable advice that you're choosing to ignore as "vague-ish" when it really isn't.

The "rant" in that reply is the context to explain where this nonsense comes from.

You are the one choosing the read it the way you are.

If you can assert that "[I] didn't have to leave a comment", I can assert that you shouldn't have made a post at all if you don't want to read the advice given. Or that you don't have to continue replying because "you're clearly bitter". But, I don't because its worse for everyone.

your whole "I'll hear your song on the radio" quip proves my point: this was never about who's [sic] better for me,

This was in response to an actual word-soup "rant", as you might call it where the only thing that was clear is that you didn't actually consider what was said. It is disrespectful on its own and contradicts the claim you made in it:

I'm immediately super grateful to anyone taking time off their day to listen

Its repeated twice that I did listen to you audio.

I dont see what would compel me to not make a quip in response to your disrespect. I guess you would also want to ban joking on this sub too?

Have you been on this or any adjacent sub for significant time? Or the internet?

Almost everyone on the internet *does* use it pretty literally. As are you by posting the solod track. It has never been a "shorthand" for anything: its origins are from self-important producers buying pre-made 2tracks and needing to justify an extra​ fee to the artist/label.

This only reinforces my point.

It would be equally useless to send the full song with the vocal on top because as I said I'm not happy with my processing and a raw vocal over a full production would just sound off and be harder to gauge in my opinion.

Youve taken this in a circle.

It wouldnt be useless, but it would break this subs rules.

Analyzing anything in solo is useless for everything to do with timbre.

You can insist that your opinion is right, I won't stop you. But you'll also notice that all the regulars on this sub, myself included, are telling you that this is a fine source and to consider it in context.

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But, since you know better than I, I'll just STFU, ignore the rest of the rambling in this reply and wait to hear your song on the radio... /s

So long as you think of it as "vocal mixing" you are truly and royally f***ed. The only, small, exception is for an a capella song/section.

Absolutely no one care about your vocal in solo and the sample provided is useless AF.

Forget everything about this and​ mix the song. With the vocal. Your source sounds usable or better. Any EQ moves need to be in service of the song and its areangement as a whole.

Given you're already 3 weeks behind and you've been delaying the single for nonsense reason, its probably time to delay the album and reconsider your core mixing philosophy.

I am not trying to be mean, but the other comments at time of writing are all blowing smoke up your ass and give you a very surface level take, when the root cause is abundantly clear. Universally, you'll get better results if you shift your focus to what actually matters rather than something that is entirely irrelevant.

Put simply, "vocal mixing" is bullshit from online content creators who want you to watch their "10 tips and tricks" video so they can make a buck on beginners not realizing that skipping the fundamentals will never be good for them. its snake oil and among the most harmful concepts propagated in the chronically online era.

- Can people actually hear the differences in sample rates above 48kHz?

On it's own, no. Nyquist-Shannon Sampling Theorem tells us that the highest frequency represented in the signal is half the sampling rate: 24kHz. The upper-limit of human hearing is right around 20kHz. 44,1kHz and 48kHz were deliberately chosen, partially for this reason.

It *can* provide downstream benefits if there will be further processing. Minimally, there is almost always an antialiasing filter at the DAC. In the context of distribution, this would include the distributor re-encoding. And, many listeners use some amount of device post-processing (Apple/Microsoft/Headphone brand EQ curves and such). All of these *can* perform better at higher rates by virtue of having less foldover distortion as one example

I will note that I am talking about a track that was produced at a higher rate from beginning to end. At every stage, information can only be lost, not gained. If a track was produced at 48kHz, for example, and upsampled to 96kHz at the end, all we have accomplished is doubling the file size. This applies to my entire reply.

- Can people hear the differences in sample rates above 48kHz in a final mix and master?

A layperson cannot.

An audiophile will claim they can. Regardless of whether that is truthful.

Trained listeners under ideal conditions can certainly hear the side-effects of , even if it's just the anti-aliasing filter. This is more apparent the more stages of processing were done at the higher rate. But they are NOT hearing the sample rate itself.

- Are there benefits to uploading the final masters at a sample rate higher than 48kHz?

Theoretically, if you produced the entire track at a higher rate, yes. As outlined above.

Practically and for your audience? I will refrain from be prescriptive, but I think it's pretty safe to say that the vast majority of listeners do not care for almost every artist.

What is important to you, as a producer, is not something I (or Reddit) can answer.

- In what situation would 96kHz be better than 88.2kHz?

This is identical to the question of 48kHz vs 44,1kHz.

48kHz was chosen for film applications because:

1. Its Nyquist frequency is above the upper limit of human hearing (24kHz > 20kHz)
2. It is easy to align with 24fps film. Each video frame is 2000 audio sample. (48000Hz / 24Hz = 2000samples)

44.1kHz was chosen for music (read: Compact Discs) because:

1. Its Nyquist frequency is above the upper limit of human hearing (22.05kHz > 20kHz)
2. It is a 'magic number' in computer science that is convenient for doing division in an era where floating-point operations (flops) were expensive.
   1. Nerd-shit you might want to skip:
      1. Floating point arithmetic and integer arithmetic are handles by different parts of a processor. Integer is obligatory, floating point is not. And, in the 90s, the cost of adding a floating point arithmetic unit to, for example, a CD player, was expensive.
      2. 44100 is the product of the square of the first four prime numbers (2x2x3x3x5x5x7x7 = 44100). Prime divisibility means a number can be divided by more numbers and still have the result be an integer, so we can 'cheat' not having an FP arithmetic unit better.

20+ years ago, sample rate conversion kinda' sucked, so we would choose 44.1kHz for musical content and 48kHz for audio-visual content. Today, conversion between the two is excellent so it doesn't matter very much so long as we don't convert back and forth many, many times.

The same applies to 96kHz vs 88.2kHz. You're either choosing the historical convention or not. But it makes very little difference nowadays.

Best practice is to choose what sample rate you will be using for your primary deliverable, *before* you start any work on the tune and use that from beginning to end. You can resample it to whatever you need for secondary deliverables. The notion is that we have the highest fidelity/accuracy to the original source if we resample as few times as possible. Again, in practice, sample rate conversion in 2026 is good enough as to make this a relative marginal concern to begin with.

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TLDR: I won't prescribe an answer for you. But there are theoretical benefits and practical drawbacks (file size/bandwidth). What balance is right for your work is up to you. But, hopefully, I've pointed you in the right direct if you want to learn about how and why one would make a choice in favor of higher rates so that you can make a competent choice for yourself.

Yeah, thats the one I meant.

I've seen tonnes of them around in IEM rigs for bands.

Art S8 comes the way you describe, but the ears are reversible. Its in/out on one face and outs on the other

It might be out of budget, but have you considered an AAoE Splitter/Solution.

As an example, the Sound Tools Cat Rack. 1RU box, takes 12 XLR inputs. Rack it with your mixer. For each group of 4 ins, get a short tail and hard wire it to the mixer; now you patch those inputs via the Cat Rack always, rather than direct. On a gig where you need a split, you bring a Cat5/ethercon cable of whatever length you need (and feel like carrying) along with a tail/box for each group of 4 channels to split. You can packs the splits separately to keep em neat. On the gig just run as many 4 channel groups as you need.

Or on the end for the split you could have a second Cat rack to do 12 channels at a time and 3 Cat5/Ethercon cables (loomed if you want) to do the run for the split.

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Your plan seems reasonable as-is though, provided your existing splitter and are good.

At the same time, it really depends on how valuable your time is. In my area at least, a blank panel with connectors is pretty much the same cost as a standard 1RU 8 channel xlr split.

Is FabFilter noticeably better in terms of audio quality, or is the main difference the interface, workflow, and flexibility?

There is no "better or worse in terms of audio quality". Just different. But both are industry standard "quality" in every regard.

How has your experience been with Waves' Update Plan (WUP)? Has it actually caused you problems or additional expenses over the years?

I would never touch it with a ten foot pole. Even when I have owned active/valid licenses for their plugins, I would circumvent the DRM.

Would you personally choose seven FabFilter plugins for $365 over a larger Waves bundle for $299?

Over the past 20 years ive let all my Waves licenses lapse, and when changing computers never even bothered to try them again.

Fabfilter's stuff has been on every mix Ive done for the past 10 years. Often dozens or a hundred instances on each.

The math speaks for itself.

Are there any specific advantages to Waves that you feel FabFilter doesn't offer?

Fabfilter doesnt really do hardware emulation to my knowledge. But,

1. You dont actually need any of that. If you're mixes are great with emulation plugins, they'd also be great without. Crap with = crap without. And so on. It doesnt change you, your skill or your output.
2. Waves became popular by being some of the first to do good emulations ~20-25 years ago. They were the only game in town and the bundles were like $10k. Now there are tonnes of options and everything is like $20.

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So, buy a fabfilter bundle license and you'll have a solid set of base tools for your entire career.

Buy a Waves license and get a tonne of variety of crap you have to sift through every time you want to do anything so that they can Nickle and dime you in 5 years when your time runs out.

FF is the clear winner, imho.

Define "right" and "wrong".

If you tweak your wording from a value judgement (right/wrong) to conventional/unconventional your thought is then correct based on a historical context. For something like clipping, you could change your verbiage to nominal vs super (or sub) nominal. These become indisputable claims that, I think, are what you're getting at. Almost every example where we would say "right" or "wrong" in AE, we are just being obtuse and failing to communicate our meaning well.

So your friends are right, but I suspect the issue is that you aren't articulating what you actually mean when you say "right" and "wrong". Its ultimately just a semantics question.

I built an emulator so you can hear "real translation"... lol

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Pros, the target of this sub, aren't spending "all night" airdopping their mixes around to different devices. When we are, its just for a final sanity check, where we'll be rendering it anyways.

Beyond that, even as a beginner, you probably shouldnt be adopting a workflow like that. Quick intermittent "car tests" should be sufficient and doing more is probably a net negative. One learns translation by listening to the same music across different systems, not one's work in progress mix: thats the fastest way to never learn how good mixes translate.

We don't need more slop, even if Claude told you we do. This is pointless for the folk in this sub and a counterproductive grift for those in WeAreTheMusicMakers.

And, FFS, a SAAS model for trash like this from an unknown dev?

And do you actually think professionals (or even any sane person) is going to allow arbitrary binaries from an unknown dev from gumroad?

Are you inventing wav files...?

Slap the wavs and the text file in a zip. This is the same thing, and either way th text file isn't really helpful.

If you mean for the click, then an actual MIDI tempo map could help, in that we can set that in every DAW. But every DAW already exports/imports these. You aren't adding value.

For musicians, sure, i see what you're after. But getting musicians to adopt any app/tech is a chore and a half. Since band are likely already using some file sharing app, just dump these into folders and be done with it. Like, getting musicians to use a simple calendar tool for scheduling is already more complicated than it needs to be (and more impactful than this).

And thats the core of it. You're creating an app and a container that services the same purpose as a folder/directory, but with more friction. Even in the best cases, this isn't an idea that will gain significant traction.

​If a client handed this to you to start a project, would this be useful?

Not even remotely. Like, literally none of this is useful. Just send a two track, and we're good to go.

Consildated export is a triviality.

A wav click is pretty pointless.

The text file appears to all be information that is trivially obvious from a listen and effectively takes no time to annotate, however the prod/eng needs it.

It will be a small miracle if you actually find a way to make this cross-compatible with all the major DAWs and their idiosyncrasies. And, even if you succeed in that, it will ostensibly never fit into real user workflows.

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This really feels like an idea that an AI affirmed as a good idea when it doesn't materially help the artist, the producer or the engineer in any meaningful way. We don't need more slop and you probably need push back on the LLM that is telling you this is a good idea to spend your time on.

1. Mix well. Mastering doesnt do this.

2. You always have an infinite number of frequencies and any sound that is played back. Number of bands and the magnitude applied are both lower the better the mix is.

3. Whatever the tune requires. Maybe none. maybe a lot. depends on the vibe.

4. The better the mix, the less you need. The upper limit is when it sounds worse.

5. LUFSi is a distribution target, not a production one in music and you arent Spotify. Ignore it. Or measure it from reference masters.

6. True peaks is an estimate and also safe to ignore. Most converters have considerable enough headroom in 2026 so as to make TP kinda' irrelevant. If you never want to clip any converter ever made, you keep it below 0 (but you also will sacrifice a lot of perceived level and transient information).

7. To wherever it sounds best. If you cannot tell the difference, you may as well not do it at all. Same applies to notching something out or a hpf.

8. Whatever sounds best. No, its not exclusively for low end. De-essers are a subset of multiband compressors.

9. Record a great performance cleanly in a relatively dead space. Just do a good job of getting it clean at the source.

10. Whatever sounds best. It entirely depends on the source and desired outcome.

Bonus: Stock will do the job on a budget. But, given your questions, you should probably hire a mastering engineer/service or fork up for something like Ozone. It's pretty clear from your questions that you have a lot of learning to do before you'll be able to get passable/mediocre result.

TLDR: My main recommendation would be to get a book on audio engineering and to study a lot. Your questions demonstrate the you do not know the fundamentals of the tools or the process and are trying to skip to the end. Knowing the fundamentals will progress your work more quickly and help you to ask actually meaningful questions in the future. If AE were a game of paint by numbers, as your questions intimate, it would have been fully automated already.

Talk to your​ engineer. They should also provide the spec by default.

Almost always it is renders of each edited source. This is usually more granular than the stems (which are produced by the mix/mix eng), but not as granular as the full multitracks.

Some engineers will take the session, but its highly likely that there are compatibility issues and is less common. Its often the distinction between a longer term collaboration on a session vs a mix turnover.

And a good tip is to number your render files. They most likely get alohabetized at some point so this will keep all the drum sources together.

And the render should be consolidated. The start time of each source must be identical, even if that means adding a bunch of silence.

TLDR: Talk to your engineer.

The problem in answering this question is that "bus" is an incredibly broad term. It is any track-like object that can accept and sum multiple inputs.

For a quad tracked guitar, we might have 4 sources. We might pan them individually. Or we could assign each pair to a bus and pan that. Then each of those busses could feed an unplanned "rhythm guitars" bus.

Whether or not to pan a bus depends on what the bus is for.

1. Yes. For the special case where only one bus has panning that isnt fully out and/or centered. Otherwise its more complicated depending on your pan laws.

If anything it just sounds like a workflow preference if you really want to pan your buses. Thoughts?

In simple routing configs, yes. When cascaded and if/when stereo/nonlinear processing is involved, no.

1. Yes. Dual-mono is just the same signal to both channels. If you send to a mono bus, that bus (eventually) hits the master bus which is, usually stereo. Again leading to dual mono. (Dont forget that the master _is_ a bus). Ofc, your master (or whatever terminal bus) can itself be mono if you want it to be; some consoles are like this so you can have twice as many mono busses, if you want.

I gave you a respectful and realistic assessment of the situation you're in and a set of actionable remedies.

If you don't want answers to your questions or need people to blow smoke up your ass, perhaps you should "STFU" rather than asking.

One made by someone who knows what they are an expert on and can talk about it rather than ask reddit to be creative for them.

Depends on what you actually need.

The graphics card is pointless for audio.

The no name case fans are going to be loud AF. AIO have a less consistent (and less customizable) noise profile. This matters if you'll be recordin near it.

AIOs are also a PITA if you need to transport it.

8 performance and 16 efficiency cores makes that CPU pretty much unusable for a multichannel hybrid setup if that's your thing.

The LGA1851 socket for the CPU basically means that you will never be upgrading the CPU. When it isn't enough, your buying a new mobo and cpu.

The listing doesn't specify how the ram is configured, but for best performance, you always want exactly 2 stick. If it comes with a 1×32gb then you're golden for your upgrade. If its 2×16 you might end up throwing that ram in the bin to get a 2×32gb config. (Or god forbid they sell it with 4×8....)

They also dont specify what RAM you're getting, which is real sus given RAM prices nowadays. It leads me to believe its garbage in there. Good RAM is important if you care about latency.

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Is this a fine box for most producers who want to game? Yes.

Is this a machine I would in the same room Im going to track in? Hell no.

Is it a machine I would buy? No. Fuck efficiency cores, I want my hybrid studio.

Is it an exception production box? No. Its a gaming/office setup.

Frankly, I'd always just spec and build my own, but Im also a mega nerd.

Talk to the plant, the press operator and/or the cutting eng. Not reddit.

When reviewing test pressings, you are looking for systemic defects which are grounds for rejection. Since you say "all the test pressings" i am assuming your listening across multiple copies. If the pop is at the same place on all the tests, you have a systemic defect and should probably reject them. (Again, talk to the plant to coordinate the fix, if possible. They can attempt a repro).

"Fancy headphones" is meaningless.

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Fun anecdote:

I had some test pressings come back with an intermittent high pitched whine. I rejected the tests. The plant claimed that they couldn't repro the problem.

A week later they gave me call: "hey, sorry, everyone who works here is 60+ so we couldn't repro it. My grandson was in the plant on the weekend and he confirmed that this sounds like garbage. Sorry for the delay, we know what went wrong and we'll expedite the shipping on the next rounds of tests for free"

You're doing good by reviewing them carefully.

Listen to it and follow your ears.

You already screwed the pooch with having destroyed your own project files due to negligence.

Knowing you chains is irrelevant information.

If the information is there, its an EQ problem/solution.

If there's no information there, you're likely adding a post-mixbus layer and doing your best to polish the turd.

But, also, if your mix was "done" (and therefore client approved) you should be accepting that and moving on or consulting with the producer. If you are the client/prod, then you just need to acknowledge that you were never "done".

The best, time and budget permitting, is to recreate everything and make sure to not lose all your work. Back up on at least 3 drives in at least 2 physical locations or your work doesnt exist at all, or whatever the saying is nowadays.

It just isn't going to happen. Period.

You mention a lot about why Reaper is popular at the upper echelons without getting to the crux of why DAWs are difficult to displace at those levels. It doesn't matter that Reaper is easy to script, but that big production studio have already spent millions building and maintaining their pipelines and integrations.

The same applies to PT, even if its cumbersome or impossible for the average user. While film studios aren't interested in batch processing, they do need to distribute and collaborate amongst hundreds of users globally and across many spaces in audio post studios. The same applies for Reaper

So long as the top (in terms of spending) cannot be displaced, educational institutions won't focus on it. And it will never get the same kind of mainstream/industry traction.

Your arguments are focused on the bedroom level, who are also resigned to always be at the bedroom level. Frankly, Luna is expensive mediocrity that only gets decent if you buy in to an expensive closed garden of averageness.

If your main issue is that you get bored of audio engineering because you get bored of how your tools look, you probably aren't cut out to be an audio engineer. If you want to give it a go anyways, you've already mentioned your solution: Reaper. Just swap skins when you lose inspiration. Your projects stay perfectly intercompatible, plenty of us use Reaper outside of game audio and it'll do everything Ableton/Luna can (and usually more).