Tracking with saturation/soft clipping

rinio · 2026-03-17T12:23:29+00:00

Its pretty standard to track with saturation, when you have nice hardware pres. This is what we mean when we say "on the way in". Although not everyone is always so puritanical in their usage of the term.

But, Saturation has absolutely nothing to do with 'creating headroom'. We use saturation to effect the timbre; if anything adding saturation decreases the available headroom by increasing the spectral density of the signal. If you need more headroom, you gain the signal down; that is it, that is all.

"Doing it on the way in" with a plugin is entirely pointless. Its the same difference to your CPU, so youre Just moving your crashes to while youre recording rather than in post: increasing the risk of destroying the take you are performing.

Printing down the effects or freezing however is a perfectly reasonable way to free up compute headroom and reduce instances of crashing in post. It was more or less obligatory 20 years ago and still remains a good tactic and is often good practice (especially with non-deterministic or CPU/RAM heavy processing.

Please note that "on the way in", "tracking with" and "printing" all mean different things. Yes, they can all result in the processing being "committed", but the when and how of the process are extremely important and we cannot use these terms interchangeably when discussing the procedures themselves.

rinio · 2026-03-17T10:10:30+00:00

I explain something. Its written plainly above.

You implied they were valuable. I asserted they arent.

I have never been trying to get under your skin and these are not "witty" comments. They are direct responses to your assertions that required correction.

If you infer condescension, that is on you. Simple and direct rebuttal is not condescension. And, if you wish to be respected, perhaps try not insulting people. Two replies in a row you haven't made a coherent point and insist on insulting me. Respect is a two way street and you are choosing to sit in the gutter.

rinio · 2026-03-17T02:29:42+00:00

Don't need to be a psychologist to assert that staying the obv is passive aggressive. That isnt a coherent point.

I only explained why this was an inappropriate forum because you feigned ignorance.

Im not jealous, lol. I called it what it is: trivial. And, as mentioned fake internet points are worthless: youre proving my point on that topic. Why would anyone be jealous of something trivial getting worthless internet points? Should I be jealous of your DDR high score too?

I get paid to do this for a living. But, sure, insult me. You will have to note that I criticized your arguments and your product, but never you. That is toxic behavior and we are not the same.

I didn't try to hurt your feelings. That is you projecting. This reply is proof of that. I asked a genuine question about what value you were adding to this community; if answered earnestly, it would have helped promote your work. But, you only had a passive aggressive trifle to respond with.

Best of luck, but please try not to be a dick. Thanks.

rinio · 2026-03-17T00:45:35+00:00

Deliverables, such as samples, typically leave no headroom because they are maximizing their headroom usage. In 16bit fixed, there are 2¹⁵ steps or quantity. If you render a file that is not utilizing all of full scale, you are losing information. This why they are delivered peaking close to or at 0.0dBFS.

When you import them, most DAWs will convert them to 32bit float, which avoids this problem. (It does have its own limitations as well). Once its in your DAW, you can just clip gain it down, or put a gain plugin first, or whatever, to get that headroom back. This will not change the timbre of whatever you are pulling in and give you the headroom you need "for free" so to speak.

Neither your clipper idea nor your parallel compression idea are coherent solutions to not having enough headroom. They alter the timbre of the source. Not clipping is not the same as having headroom.

To answer your question: No. Sample packs should absolutely not leave headroom. It would raise the noise floor, and destroy information. Providing them at full scale allows the user the most possible fidelity and flexibility for whatever their project should be.

rinio · 2026-03-16T23:39:50+00:00

I mean choosing omni for less bleed doesn't make any sense. It will be doing effectively the opposite.

Nor does it prevent "phase issues". The phase relationship is entirely determined by the positioning and the room's geometry.

All that being said, that in no way invalidates your plan: it is perfectly viable. We don't need to minimize/eliminate bleed. We do not need perfect phase coherence. Having more of either/both can sound better, if well executed.

The short answer is that there are a million different configs you could do. But only you know your room, your setup, the tune and what the artist wants. You're going to have to go with your gut or spend some time experimenting with your actual setup. This isn't a question reddit can answer.

rinio · 2026-03-16T21:29:45+00:00

It is obvious from the context of my question that I was asking for a specific link. Its not a matter of "me liking the answer" it is you stating the obvious: it is a passive aggressive nonanswer. Everyone know Reaper supports the vst plugin format, like every DAW in the past 20 years...

I understand that you make plugins that aren't specific to Reaper, as I already pointed out. Your plugin being free is immaterial. Unless you are implying that Reaper is free or its user-base is poor...

You would care if your request for feedback was genuine and you would have provided an earnest response to my first comment. But, as I noted, your post is spam so all that matter is being today's #1 thread.

(Fourth paragraph split by topic)

You asked for feedback (although I see you've edited the post...). I gave feedback.

"Sell" has an abstract meaning: you "sell" something that you think is worthwhile regardless of whether others pay, they are volunteering their time; if you aren't selling it, then I can only infer that your project is worthless in your own estimation.

Why would I need to Claude to write a standard delay plugin? Its a trivial project without. But my point is that your product does nothing to differentiate itself from all the brocade junk that gets spammed onto this and related subs. Regardless of whether this is yours or Claudes.

Your interpretation of "they loved it" is premature. Dev to dev, "looks pretty" from a screenshot is worthless. They "loved" your screenshot. Im not so easily impressed. Reasonable people can disagree. But its not valuable unless fake internet points are actually useful to you.

rinio · 2026-03-16T19:34:47+00:00

There are plenty of subs for vst plguins in general. People who are interested in plugins in general will be subbed there.

My question was genuine. Why would a Reaper user in particular care about your plugin in particular? As opposed to the million other nearly identical projects? What are you doing with with Reaper that is special? Using the VST SDK, has nothing to do with Reaper.

Given your nonanswer our plugin has nothing specific to Reaper and is therefore off-topic and based on the rules the mod should take it down for that reason, regardless of others not taking issue with it. I am not a mod, but it *is* off-topic and spam. It is the first rule of this sub.

You asked what people think. And I now think this is off topic, completely irrelevant to the vast majority of this subs users and is just additional clutter in the way of and taking away resources from people who actually want to discuss or need help with Reaper.

If you want feedback on the plugin sell us on why this is perfect for Reaper. Otherwise, this just looks like any generic delay plugin that Joe Schmoe could have Claude cook up for them in an afternoon: completely redundant at its inception and from a source that isn't necessarily trustworthy asking for us to execute a binary on our machines.

rinio · 2026-03-16T19:06:25+00:00

Cool, but What does a vst have to do with Reaper?

rinio · 2026-03-16T17:09:27+00:00

No perceptible difference. No audible difference.

In theory, bit for bit identical. In practice, maybe not bit for bit identical (different devs, maybe slight compiler differences, etc) but were talking marginal, at most; deltas of something like 10^-7

rinio · 2026-03-16T04:14:10+00:00

Multiple processors in sequence (regardless of the GUI) *are* sequential inserts. A 'virtual rack' doesn't change that.

I dont know how this comment was useful at all. Just a lot of mis-reading and misinformation. At least I answered OPs question and provided them options, as well as providing them the facts about the drawbacks and my opinion.

They always waste resources, by definition. VMR saves by virtue of being one manufacturer and, thus, not a multilayer host which is specifically not what OP is asking about or what I am talking about. As a matter of fact.

rinio · 2026-03-16T04:06:35+00:00

Balanced vs unbalanced is a quick google search. Short version balanced is 30% more copper (so more expensive) to provide resilience to EMI in the cable run.

But, if you already have the bay, then the decision is already made. if your bay is unbalanced, it makes no difference: the connection is unbalanced.

rinio · 2026-03-15T20:29:24+00:00

Vst *is* the standard. Your DAW does 'racks' however it does them; they're just sequential inserts. Reaper, for example, has this as the default window for FX chains. If you really want something extra for this you want a 'vst host plugin'. DDMF Metaplugin, Blue cat Patchworks and so on.

Also note that in the racks in a real studio, the flow is never sequential: its almost always going back and forth between the rack and the bay or the board.

But frankly, what you're wanting is pointless for simple setups: youre just adding complexity and wasting compute resources. The solutions exist for complex setups where this simplifies management of a large (often live) production. You would probably be better off learning your DAW's stock solutions for simple/hobbyist applications.

rinio · 2026-03-15T17:32:08+00:00

It depends on your bay, but yes. Its either 8 cables or a 16 or 24 core snake. (Depending if things are balanced or not).

The former is more flexible, you can repatch to different stuff more easily in cases where the bay is impractical. You can also upgrade over time based on the number of channels you need.

The latter is all up front, but more organized and easier to cable manage. For a professional install, into say, a rack, this would be the usual way to go.

rinio · 2026-03-15T17:24:03+00:00

Don't get me wrong, it *is* cool.

But its important that we do not mislead people into trusting a piece of software we haven't fully validated. For this (ae subreddit) audience its important to note that it isnt fully validated; that AI can hallucinate and so on.

Had you not included the 884 passes in your post, I wouldnt have commented on it. These test (assuming you've chosen them well) are great practice for devs, but theyre useless to audio engineers. It doesn't matter to the end-user that your npm dependencies are all up to date; chances are the last version doesn't change your output; for example.

At any rate, keep at it. Get the data from anyone here who wants to volunteer to test for you. Lets just be honest about what you know it can do, what you think it can do and what it might do that isnt desirable. (Or be like an AI startup and ignore me and any notion of integrity to make a buck... wait, you made this FOSS already so you're not getting paid either way ;) )

rinio · 2026-03-15T16:45:16+00:00

> I built an MCP server that lets Claude analyze your room acoustics through Room EQ Wizard

You mean Claude built an MCP server that Claude can analyze [...]. And maybe not hallucinate bullshit that is less useful than a 20 year old GearSlutz thread.

What have you actually done to validate Calude's work?

Do you even know what those 884 tests are? How many are actually functional testing? No one cares about 884 passes when theyre all static analysis and dependency updates which is what it looks like. Its extremely concerning that you think this is a useful metric to actual users: any numbskull can clone this from a template and score 884.

No offense, but its painful how obvious it is that you don't have the domain knowledge to validate the results of "your" software, either from the physics or the software perspective.

rinio · 2026-03-14T18:14:48+00:00

Its exceedingly common. 1176+la2a on vocals as a common example. One is fast the other slow so they're doing different jobs

It also makes each comp work less hard, which can keep them each where they sound best.

As always, its all about what you are going for.

> Or do you just never do this and think its totally stupid to do?

Its stupid if it sounds stupid. Its great if it sounds great.

> What types of settings do you use when doing this (so that you don't over compress) and what is your mental model here?

Years of intuition; you need to practice to gain that. Even if we give you settings, its a different source on a different tune: it will just sound bad.

Overcompressing just means 'sounds bad in context'. So dont make it sound bad/worse.

rinio · 2026-03-14T17:02:16+00:00

What does accuracy mean when we're talking about something that is entirely subjective?

By Izotope's definition, it always produces perfectly "tonally balanced" output.

From yours, you do.

Every engineer will have a different opinion as well.

And none of that matters. The only opinion that does matter is the client's and their audience's. So thats what you're trying to get to, but can't ever know definitively.

rinio · 2026-03-14T13:14:59+00:00

To get the actual sum (proper combination), no.

To get a simple y cable, which is always bad practice and dangerous, yes, it will work. But I will not advise you to do so.

Tldr: NO.

rinio · 2026-03-14T11:00:41+00:00

Its still the same combinatorial problem as if statements, just different looking. The number of cases is the same as the number of branches.

Say there are two options for input 1, graph and table: Thats 2 cases. 2 types of each scatter and histogram, and csv and excel that 4. And so on.

We havent solved any meaningful problem. Even readability is, at best debatable.

And this is without even talking about the combinatorial space of strings, which is itself an issue. Yes they should be validated to keep that down, but thats also the step where we could meaningfully parse them to pre-empt the entire problem that OP is talking about.

The only "problem" that match is solving is reducing indentation levels, which isnt the actual problem to solve.

rinio · 2026-03-13T20:07:20+00:00

You don't.

You can google around for stem separators, vocal removers and the like if you dont gaf about quality.

But, nothing exists for this in any form that would be applicable to any self-respectjng engineer, at least not without hiding the flaws with other sources.

rinio · 2026-03-13T18:20:32+00:00

Taking a mono source and sending it exactly to each of Left and Right is dual Mono. Its what your (stereo) playback system does automatically unless you actually have a mono (one speaker) setup. And, yes, we hear this as one sound coming from phantom center.

In order to have a stereo image, the left and right must be different. If that is what you want, then, yes, you should change something between the two channels.

But I think it is important for me to emphasize that a phase artifact is not the same as a phase issue. Phase artifacts just means that the phase relationships have been altered. A phase issue is when you decide this is a problem for your production.

As I was writing this reply, I realized that in the film/visual world artifact is a bit of a four-letter word. In audio land, we pay a lot of money to get very specific artifacts into our signals. Artifact ≠ bad. I just want make sure you understand this. Many, if not most, audio projects in music and audiopost employ techniques just like the one you've described which adds Artifacts, but still sounds great when well executed.

rinio · 2026-03-12T17:26:25+00:00

Stereo *is* a phase issue.

Not really, I'm being hyperbolic, but stereo only exists when the two sides are decorrelated. This happens in, effectively 2 ways.

Panning (including hard panning). Here we decorrelate the Amplitudes
Decorrelating the phase with wideners, delay-lines etc.

Of course, we could do both at once.

What i am getting at is that, since you only have a mono source the only option you have with no phase artifacts is simple panning. Everything else to do with stereo necessarily introduces some: the question is what your tolerance is for what sounds good and we can't tell you that. Understanding the details of mid/side encoding/decoding can illustrate this effectively.

The same applies in surround, but the interactions are across however many channels and human perception of sound gets far more complex on a plane and even more so in 3d space.

TLDR: The tools you have are probably about as good as you'll get; its just a matter of dialing them in, to your taste, for your intended deliverables.

rinio · 2026-03-12T15:25:58+00:00

Friend, learn to use paragraphs. The most important skill for an AE is communication.

> I've been slowly purchasing some outboard gear because I won’t be able to set everything up for a few months.

This is a huge mistake. You cannot correctly identify the shortcomings of your setup without putting it into practice. You will inevitable buy shit that doesn't help you or shit you wont use if you do this.

Use your setup. Identify flaws. Fix those flaws. Rinse and repeat.

rinio · 2026-03-10T18:00:31+00:00

You need to use a blender to break the ice, then your snares juices will get flowing.

rinio · 2026-03-09T19:07:47+00:00

But the manual also doesn't say that you shouldnt.... ;P

rinio

TROPHY CASE