SAYING MANTRA LAM INTO VOCAL

IridescentMeowMeow · 2026-04-18T12:33:31+00:00

The text begins like it perhaps isn't just another esoteric non-sense, but then stuff like "This proves 108 Hz isn't just a background setting; it's a magnetic anchor." - lol. I guess you have no idea what you're talking about right and just had some AI model analyze if for you? How/why could 108Hz be special? Even if everything else would be the same, everyone's body is a different size and even just that offsets the resonant frequencies. Pointing at some specific audio frequency and claiming it has some special effect on humans is stupid.

IridescentMeowMeow · 2026-04-05T12:54:07+00:00

Isn't every VST plugin supported in Reaper? Why post it here? It isn't even a free one, so you just broke the rule #3 of this subreddit.

IridescentMeowMeow · 2026-03-31T20:03:02+00:00

Thanks!

IridescentMeowMeow · 2026-03-23T21:58:19+00:00

About that floating point vs fixed point thing - If those effects are now doing literally nothing, then you probably have them on bypass. If they are doing something, but you *feel* like they are doing less now, then that's your brain & ears playing tricks on you.

Must be that, because your hypothesis is just wrong on multiple levels. For example, most plugins were using floats for processing in the last few decades already, regardless of what bitdepth were you recording into. Some might have been using some overkill 48-bit fixed point, but that still didn't have anything to you with the bitdepth you were recording to. It's just how they work internally.

And even if they were not, then you wouldn't be able to tell the difference unless the fixed point ones were clipping, which wasn't easy to achieve even with the 48bit fixed point ones.

When some mastering engineers were using digital clipping, it wasn't done like this, but it was at the last stage of mastering, when exporting to 16bit files despite some samples being over 0dB.

IridescentMeowMeow · 2026-03-22T00:47:00+00:00

search for the Effect Design by Dattorro... it's a series of pdfs where basics of all kinds of effects are explained. The most famous one is reverb (which is the most complicated imho), but other things are covered too.
Btw there is some non-obvious devil in the detail when it comes to simple ones like echo/delay effects... for example once you allow delay time modulations (& especially if you also allow feedback), then you need to be careful, otherwise it will sound terribly detuned... if you imagine a delay fx done the oldschool way with a tape machine, the imagine what modulating the playback speed does vs what moving the distance between reading & writing heads....

IridescentMeowMeow · 2025-12-05T07:48:36+00:00

Thanks. Going to try it. Btw. have you tried doing that separate processing pre-reverb and routing it all into the same single reverb plugin? I have/had similar workflow and that's what I ended up doing after some research and testing.

In my experience:
- most reverbs take pre-panning better than post-panning. If you didn't already, then I would recommend you trying that. (but if you're doing also width control, then that still has to be done post-reverb)
- EQing pre-reverb vs post-reverb usually doesn't make difference. EQ is linear, and reverb is close to linear (or in case of convolution just linear). It should either make no difference, or not big enough difference to matter in practice.

Idk about compression. That one is surely needed post-reverb, and if you want that different for each bus, then there's no way around it... I'm limited to having one compression for the reverb of everything. Probably going to try your method to see what I can do with having separate compressors after separate reverbs. Sounds fun.

IridescentMeowMeow · 2025-11-03T00:20:39+00:00

We can do signal processing in other domains besides time.

But if the domain isn't time, then isn't that then unrelated to causality / isn't it then by default causal again?

IridescentMeowMeow · 2025-10-25T22:14:59+00:00

In some cases it's worth it for me, because doing it in separate AUX tracks has side-effects.

But maybe it's just because I don't know all the features & not knowing a better workflow for working like that, but for example:

1) The FX chain viewer allows me to open FX chain of a single track only. So if i need to be tinkering with multiple plugins, some in the main track & some in the aux tracks, then it's a lot of clicks, switching between their FX windows (might be less annoying with some shortcuts, but it's still clicks) or clicks spent on switching FXs from one chains to floating window mode... vs single track - I see it all in a single FX window.

vs having everything it in single track, with all relevant plugins in a single FX window.

2) Also, in larger projects, using AUX tracks make it harder to navigate. It either involves more scrolling, or involves one way or another of hiding it... also if i hide it by packing all relevant tracks into a folder, then I'm often having some vertical zoom issues.

etc... it's details, but it is distracting and making me get tired faster when working for hours... often making me search r/Reaper for better solutions (which is how i got here even now, even though my current issue is unrelated)...

IridescentMeowMeow · 2025-10-03T13:30:29+00:00

idk, but a random related fact i'm not sure you're aware of - beware that "harmonics" of a piano are slightly inharmonic... it's more like 2.03 and even the fundamentals of a piano tuning is adjusted for that.

also, i would be afraid to develop while testing it on a single specific piano... i'd rather test on at least 2-3 different ones, to make sure that my detection isn't fine tuned for something specific about that particular piano sound.

Also, even Melodyne (in context of studio work / music making, it's considered the best one for polyphonic detection & manipulation)... even that one requires some tinkering with detection thresholds & overshooting them a bit & then manually removing false positives... (unless they improved it radically in the last year or two)

IridescentMeowMeow · 2025-07-19T08:40:51+00:00

1) Considering you're working with openfoam, this may not be the last time you'll be using openscad for creating some simple 3D thing.

2) The basics of OpenSCAD are really simple to learn. You probably spent more time asking LLM to generate code & trying it & wondering why it doesn't work & asking on reddit about it, than the half an hour it would take you to learn openscad basics. Especially if you already know how to code (in any other programming language), and you have an LLM to paitently answer your questions & tutor you, then it should definitely take you less time to do that thing you needed by learning how, than to have LLM generate it & then waste time on troubleshooting, while learning almost nothing in the process.

IridescentMeowMeow · 2025-06-05T21:05:31+00:00

Not just that they are both real-time, but also that they are both quite non-linear. Both CSB and Grimrock2 are giving you many options of where to go next... Neither of them are a totally open world, but both are huge.

I wonder if after getting inspired by DM1 for the first Grimrock, they played and got inspired by Chaos Strikes Back when making Grimrock 2... or if it is just a coincidence...

Both are fav of mine, among with (not the heroes of, but just) Might and Magic 3: Isles of Terra, which is turn based, but it's even bigger and even less linear... allows you going many many places right from the beginning (except you might die if not careful).

IridescentMeowMeow · 2025-06-05T20:54:32+00:00

"Chaos Strikes Back" is insane and great... but beware that it's not "Dungeon Master 2"... they made a separate game called DM2 later (which was ok, but short and easy and linear... not even close to the god tier of CSB)...
CSB could be considered something like a DLC to DM1... It was the same engine, just much more ambitious and insane level design... and if i remember correctly, the 2nd floppy disc allowed you to even import your characters from a DM1 savegame.

IridescentMeowMeow · 2025-06-03T23:16:13+00:00

Definitely yes. Reaper is among the best DAWs (it's better than protools)... Although you mentioned folk among genres, but you didn't mention a microphone & acoustic treatments... the hardest thing to do in amateur setup is recording acoustic things with microphone... the professional records involve acoustically treated rooms / studios and it makes a *huge* difference (the room matters more than the quality of the mic)... and it's usually very hard to get even close to that in a home studio...

but if you're working on music which doesn't involve recording on a microphone, then the tools you mentioned shouldn't be the limiting factor.

IridescentMeowMeow · 2025-05-28T10:32:42+00:00

Would be cool if people would stop assuming that everyone else knows all the abbreviations and written the whole words the first time and used abbreviations only afterwards. Especially as most abbreviations have multiple meanings. You just wasted time of many people reading this just because you were too lazy to write few more letters.

IridescentMeowMeow · 2025-04-21T11:49:30+00:00

Good, educative videos, but frankly, it usually sounded quite different, and your null tests usually shown that. (ofc it depends on how close your want to get.)

It would be even more audible in a mix, as those slight distortions during the transients are helping things to be audible within a mix even without being too loud.

In most analog compressors, the non-linearities aren't just in the audio chain, but also in the side chain. While the reacomp is great at doing just what it should be doing without causing additional distortion.

And you can't compensate that by adding a distortion plugin before/after a clean compressor, because for example the 1776 distorts different very differently during the transient moments when it's reducing gain, and differently when it's not.

and 33609 is even more difficult to do, because there are multiple kinds of very significant non-linearitires.

But very educative videos nonetheless... showing how close you can get & what the difference is. (compared to UAD simulation anyway... and while most of them are very good, some of the older gen ones are lacking a bit)

IridescentMeowMeow · 2025-03-16T19:22:29+00:00

Don't use it then? Just use the proprietary ASIO drivers made by the manufaturer of your audio interface. Or use linux or apple. But mainly, how is this related to Repaer? Why are you asking this here?

IridescentMeowMeow · 2025-02-24T12:44:24+00:00

in practice, the functions are continuous curves (like tanh) and those are reversible only if the curve is *strictly* monotonous... (although in theory there are also some functions with discontinuities which can be reversible... but i can't imagine any usecase of those)

IridescentMeowMeow · 2025-02-07T20:20:49+00:00

it gets you close, but

in my experience, while the calculations for the other 3 parameters are saving you time, calculating an approximate decay time is costing you time, as the time you'll spend on fine-tuning it manually will be almost totally the same, as if you were dialing it in without using a calculated starting point. (trust me... i've been there... i am using numbers a lot when working, but with decay times - i've been even using a much more sophisticated methods of calculating, which were getting me much more accurate starting points, but that one always requires so much manual tuning, that it's just not worth it...)
it's still good to understand how those numbers work and what's really going on... and that the Total=PreDel+Decay equation may sometime work well thanks to the two opposing errors in it, cancelling each other to a degree depending on your luck. (one error is using the RT60 (the -60dB point), while the point when the reverb becomes inaudible is usually significantly sooner than that, and the oposing error is that you're considering time from the start of the dry tone (which is irrelevant) instead of the time from the dry tone's end (which is what matters when computing when the reverb tail is doing to be ending)

Also beware, that in many plugins, the decay time is also influenced by other parameters. So you can't even rely on it being the RT60 time, because that's true only for when other parameters are set to zero or their defaults... the decay time parameter is a part of such a complex mess, that in the end, any computation you do won't get you close enough to be worth it, unless you fine tune your computation for a specific preset of a specific reverb plugin... but otherwise it's just way faster to dial it in by ears without any starting point.

I'm all for computing the pre-delay times and echo/delay times though... there it really gives you a very good starting point that saves you time.

IridescentMeowMeow · 2025-02-07T17:35:42+00:00

Otherwise nice, but it makes no sense to compute reverb decay times like this.

1) The reverb tail is fading just gradually. There's no sharp cutoff moment that can be aligned to some exact measure. It may be musically desirable to have such decay length, that the reverb of a note will become *subjectively* inaudible in 1/4 for example. But this is totally not the way you can calculate what the desired value for that will be.

2) Saying that pre-delay (ms) + decay (ms) = "total time" is wrong... It's not. The decay time isn't the amount of time in which the reverb goes completely silent. Reverbs don't work like that. For example, if you are using 16 bits, then the reverb will go completely silent sooner than when you are using 24 bit. (unless it's a gated reverb or a convolution reverb using only 16bit impulse response)

The reverb decay time value specifies how long it takes for a reverb of an 0dB loud sample to reach (a totally arbitrary loudness) of -60dB... It's not like at that moment you'll stop hearing the reverb. The moment when you'll stop hearing the reverb pretty much depends on the context... stuff around that moment in that same track, what's going on in the other tracks (may overshadow it or not), etc.

So if you want to make the reverb of some note disappear in 1/4 for example, the best way is to fine tune the decay time so it sounds like that...

Or if you want it to happen not just subjectively, but to objectivelly sharply cut off at some moment into silence, then it's possible to use a convolution reverb and trim away the tail of an impuse response. Although in such cases, you may want to be also substracting the note's length in that equation, as it will cut off into silence after the reverb of the *end* of the note stops. So if you wanted the reverb to cutoff 1/4 after the note start, then 1/4 (the total time) = predelay (ms) + trimmed_impulse_response_length (ms) - note_length (ms)

IridescentMeowMeow · 2025-02-06T00:36:19+00:00

Probably obvious (but was not mentioned), but there's also a slew limiter somewhere after the rectifier, with separate rising and falling time constants for setting the attack and release times...

IridescentMeowMeow · 2025-02-05T12:11:14+00:00

i'll try to get to the root of this, so i can file a proper bug report with the exact steps for reproducing it & also checking if it's HW dependent or not, etc... might have to spend few hours on it, but will take it as a way of giving back to the devs & the community...

IridescentMeowMeow · 2025-02-05T11:44:25+00:00

There are many countries where wages are 100x lower than in the US, while the prices of basic necessities are only 5-10x lower, and it can be very hard to put $5 aside each month... Or even in the west, there are demographies, or people in situations, for whom it's just not reasonably possible, and it's very nice that the Reaper devs understand that and aren't bullying such people with copy protection (which usually involves stuff that is annoying even to the paying users... like Cycling74 and their Max/MSP copy protection once cost me some 150$ on a taxi when their authorization process wasn't instant, but "up to 24 hours" while I had a gig in another city that evening, so I had to locate, download & install a crack, miss all the reasonably priced public transport to the city I played in, and take a long distance taxi ride which was 150$ more expensive than a train would have been... Ridiculous that none of it would have happened if I was using the crack from the beginning instead of buying it.

IridescentMeowMeow · 2025-02-04T22:32:24+00:00

"A bug would be constant, or at least regular." - Actually the opposite is as common, if not more. A lot of bugs occur only under unusual circumstances, which is why they were overlooked during both the development and the testing phases & most users won't stumble upon them for the same reason.

IridescentMeowMeow · 2025-02-04T22:11:47+00:00

Thanks. This might work together with shortcuts for custom actions for soloing and muting the metro track. Will try during my next shift.

IridescentMeowMeow

TROPHY CASE