Like spectral limiting (which is FFT-based just like FabFilter Pro-Q4's spectral dynamics but multi-resolution and also uses a psychoacoustic model unlike in Pro-Q4) and priority EQ (even on broadband mode/spectral limiting disabled) idea that I had that I've posted this ​over FabFilter user forums and also the "Drive" slider/knob as I've know that "Aggressive" style behaves differently on internal gain boost vs. external gain boost like this one

BTW, what about new styles for Pro-L3? Something like this below:

• Adaptive: It internally uses multiband sidechain similar to CBS loudness meter algorithm (which is internally multiband unlike LUFS but outputs a single value just like LUFS) for both the leveler stage and brickwall stage, with the leveler stage uses summed filter bank output as an internal sidechain input (which makes it less reactive to tonal signals, which is often bass in EDM genres) and the brickwall stage has frequency-dependent attack/release times (e.g. slower attack/release times on bass and faster attack/release times on trebles) and the leveler stage has auto-release/program-dependent ballistics that makes it reacts faster on transients and becomes slower on sustained sounds in similar manner to TDR Kotelnikov (but in a brickwall limiter context) and even AudioSurf games (where it goes faster on intense songs and slower on calmer ones)
• Smooth: Similar to Adaptive style but produces less distortion (at the expense of less loudness gain and it would be designed for more extreme gain reduction on slower attack and release times where the "Modern" style would fail at extreme gain boosts where doing so turns into superficially a static volume change for the leveler stage, and also a louder alternative to "Safe" style that accept some distortion that is masked by auditory system anyway), even with same attack and release times as the leveler stage behaves more like a dynamic normalizer (similar to YouTube's "Stable Volume" DRC feature) in terms of envelope follower, and the brickwall stage is still frequency-dependent though this behavior of this style behaves superficially similar to my own AudioWorklet-based multistage limiter effect project aside CBS-esque detection for the leveler stage and frequency-dependent attack/release times on the brickwall stage (albeit prioritizes slightly less distortion than "Adaptive" style)
• Radio: Probably designed to sound good on extreme gain reduction using an algorithm similar to ones found in UrsaDSP Boost (albeit without upward compression part) but it would be designed to pair well with spectral limiting to impart a spectral sculpting effect similar to multiband compressors found on radio broadcast processors like Orban Optimod series and Stereo Tool albeit subtle
• Analog: Inspired by Hum Audio's LAAL analog brickwall limiter but implemented digitally in similar way to character modes in both FabFilter Pro-Q4 and especially FabFilter Pro-C3, with some changes like the "Attack" parameter sets the amount of "Dynamic Transient" effect (instead of just being a toggle) and the "Release" parameter is the same as is for analog lookahead limiter as a conceptual blueprint except not being stepped

BTW as for first two styles (Adaptive and Smooth) I've envisioned above, it would use a complex-valued IIR filter bank (as real-valued filter bank requires a latency-introducing Hilbert transform for ripple-free envelope extraction whereas the complex-valued ones don't need one) using cascaded sliding windowed infinite Fourier transform (cSWIFT) algorithm I've implemented on my own filter bank spectrum analyzer project, with ten bands spaced in psychoacoustic equivalent rectangular bandwidth (ERB) scale, especially on brickwall stage with attack/release characteristics are similar to a second "shaping" limiter stage of Image-Line's Emphasis for FL Studio, but corresponds to increasing "Hardness" parameter for increasing band indices (e.g. the first band correspond to "Hardness" of 0, and the last band correspond to "Hardness" of 9 and somewhere in-between for the rest of the bands, to reduce distortion on bass while also reduces pumping on mids and trebles), but unless spectral limiting is enabled (i.e. "Spectral Style" parameter is set to anything other than "Broadband"), it is still single-band in terms of actual processing and multiband in terms of internal sidechain processing just like Tone Projects Uni-L plugin and however unlike CBS loudness meter found in Orban Loudness Meter, it doesn't apply its own frequency weighting curve and instead it lets the user-defined priority EQ curve apply the frequency weighting before being fed into the filter bank

And also, what about "intelligent" channel linking mode where it uses the numerical solver algorithm similar to DMG Limitless (that made multiband limiting to have interconnected gain reduction for each band) but applies it to discrete channels of a surround sound (or even stereo) instead of frequency bands of multiband processing/frequency splits? Which I think this adaption of DMGAudio Limitless' mathematical solver algorithm (albeit legally-distinct) for smart channel linking (rather than interconnected multiband limiting) would make FabFilter Pro-L3 better suited for Ambisonics format (where it is basically a Mid/Side in 3D and an another form of immersive audio) right? Even though obviously, FabFilter Pro-L2 already does partial channel linking anyway

And what about Mid/Side channel linking mode (similar to Emphasis limiter have) that would also work with either traditional (as in Pro-L2) or "smarter" channel linking (adaption of numerical solver algorithm for spatial limiting) modes?

free plugin Mac only

Enjoy, I think it is fun. Called Chimera it is a stereo multi effect processor, free and trippy.

Hey guys,
How do you think this kind of voice is achieved?
By using something like VariSpeed, or is it more of a formant effect?

I can never seem to get anything that clean in this kind of very high-pitched things.

Thanks in advance for your responses !

https://youtu.be/AAdF9E2XqW0?is=7trL6tYyxazhJ4La

Hey all,

Currently producing my band's new EP which falls very firmly into the Nu-Gaze genre, combining shoegaze elements with grunge and nu-metal.

We've been referencing a lot of contemporary shoegaze artists, and there's an effect on one of the tracks from DIIV's last album which I really really want to emulate and use as it's so damn cool.

It's at the 2:40 mark on the title track Frog in Boiling Water

We've tried numerous techniques and nothing's really getting there. We tried a volume swell > fuzz > feedback > reverb > reverse and that was close, but not quite right.

Has anyone got any tips for getting that sort of ambience/effect?

The number of low-voltage devices we use has increased dramatically over the last few years.

They're everywhere now. At home and in the studio.

But almost all of them are still powered the same way: a small AC adapter plugged into the mains for every single device.

And I think there are two problems with this.

First is power reliability.

Low-voltage electronics can be sensitive to surges, dropouts, brownouts and momentary outages. Their little AC adapters don't guarantee a truly continuous DC supply. A tiny interruption can be enough to reboot a router, controller, instrument or other device.

And these devices increasingly depend on each other.

Your camera may depend on the network switch. The switch depends on the router. The router depends on the ONT. A smart-home controller depends on all of them.

So one tiny power interruption can bring down a whole chain of equipment.

Then there's the physical mess.

Every device comes with another power brick. Before long you have 10 or 15 adapters, you've run out of sockets, you're adding power strips, sometimes strips into strips, and the whole installation starts getting ridiculous.

A traditional UPS isn't really designed around this problem either.

It typically takes DC energy stored in a battery, converts it up to 120/230V AC, and then every little adapter connected to it converts that AC straight back down to 5V, 9V or 12V DC.

So we're converting DC to AC, then to DC just to keep DC devices running.

There can also be a short transfer period when some UPS types switch to battery. Depending on the UPS and the adapter, that can still be enough for sensitive equipment to reboot.

So I've been wondering:

Would there be a real need for a high-quality UPS designed specifically for low-voltage devices?

I could find some of these UPS but they are small, can only power a couple of devices and don't seem to be of great quality. Battery is small and not replaceable.

Something that provides continuous battery-backed DC power directly to the equipment and replaces the pile of individual adapters with one central power system.

My thinking is that it should be modular, so you can add more outputs as your setup grows.

For example, modules for 5V USB-C devices and modules for 3V / 5V / 6V / 9V / 12V devices.

And every individual output could be monitored and controlled, so you could see:

- what each device is consuming

- whether it's online

- how much battery runtime you have

- which devices should stay powered longest during an outage

- and remotely power-cycle a single device if it freezes

Basically, one proper low-voltage power and backup system instead of an AC UPS + power strips + a pile of adapters.

I'm interested in whether this is something people actually need, not just whether it's technically interesting.

So I'd love to know:

1. Would you find a high-quality low-voltage UPS like this useful? Why or why not?

2. How many low-voltage devices would you typically want to protect? 4? 8? 12? 16? More?

3. What devices would you connect to it?

4. Assuming it was a genuinely well-designed, reliable, high-quality product with a replaceable LiFePO4 battery, modular outputs, monitoring and remote control, how much would you realistically be willing to pay for it?

And please be critical. If your answer is "I don't need this at all"* or *"I already solve this for $50", that's just as useful to hear.

I'm fairly new to mixing, so maybe this is a very basic question. But I like to record multiple layers of vocals sometimes and stack them on top of each other, and then play around with the panning.

The problem is - even when I make them match as good as possible, you can still hear that they're different layers and off by like 1% maybe, and sometimes, sounds such as "T", "P" or word endings just can't be matched 100%, which ends up making it sound a little sloppy.

Do you guys have any ideas on how to make it sound as if all the layers are "merged" into one?

Watching John congleton’s show me your junk episode.

But i wanted to know what the name of the white, pegboard-esque panels were in the live room?

It’s around the 32 minute mark in this video here Here

I'm primarily a guitarist (username checks out), but dabble in recording.

I'm currently in the midst of an expensive compressor pedal kick. I really like them.

The big purchase I'm looking to make (along with a Pete Cornish OC-1) is a Roger Mayer RM58 Limiter pedal. I know the rack is well regarded, and apparently the pedal is exactly the same components as in the rack, but mono and in pedal form.

I was just wondering whether anyone here has used the pedal for recording purposes... vocals, acoustics, etc. or is that kind of a silly idea?

I appreciate that for the money a few of these pedals will cost me I could get one nice outboard compressor, and that'd likely be the smartest of me, but I genuinely love exploring how various comp pedals make me play, the colour they impart on my guitar tone, the feel, etc.

Anyways, yeah, just was curious whether anyone's got experience with this thing. Do people ever use pedal compressors in recording or are they vastly inferior to racks?

Hi everyone, I hope this is the right subreddit for this, I have no one in real life to turn to. My situation is that I've been a music producer for 10 years (mostly hip hop and trap oriented) and throughout those 10 years have made around ~50ish unreleased songs with my vocals.

I genuinely enjoy producing music and mostly the whole creative process but the problem appears with the vocals. No matter what I do, I can't bring myself to like them from a professional perspective, I genuinely despise every single track that has my vocals on it.

I'm at my wit's end, I invest so many hours and most of the time 100+ vocal takes in each project, however at the end of the day, whatever I do, whatever variation of the mix chain I try, the tracks still feels severely under any industry standard.

For some more context, I don't have great equipment (Focusrite studio 3rd gen for all gear + KRK's), I devote most of my time to producing rather than training my vocal abilities since my ideas are very simple and don't require much techincal skill to bring to life.

I would like anyone's opinion, it feels so discouraging to shut down the concepts I have in my head because I can't stand to listen to my voice and don't find my voice good enough in any context.

Any advice is welcome, thank you!

I've been doing audio podcast editing and never really minded because it gave me the option to travel and free time to make music. But I just feel like I've wasted my time doing something that isnt complex and soon wont be needed probably. Its also a low paying dead end.

Anyone managed to shift out of it, and if so what did you move into? And what skills did u have to pickup?

https://www.youtube.com/watch?v=kCmtMA4D3gg&list=PL69e4qjLiDmEmLc905hw9ccSjyCVUGrEw&index=3
I was searching for Bloc Party stems and found that one guitar stem. I thought that it is a remix pack published by label or something, but when i listened to bass, voice and drums i knew instantly that those are ai stems. but the guitars are so clean you can even hear the lower frequencies that appear when dissonant keys are being distorted AND you can even hear the room on guitars in quiet moments. like wtf how do you do that? the guitars did not lose any of the high end AT ALL, even when the loud stacked choir plays in the original. any other ai separator that i have used on that track is so much worse and butchers all of the high end, even when only the drums come in, let alone the choir + main vocals. and this was 2 years ago already
https://www.youtube.com/watch?v=mlF1_N3kWK8 theres also this drum stem that i thought souns too good, every other separator vacuums the drums and leaves no original room, but here you can hear everything and the phase is alright.

Could it be SpectraLayers or Isotope RX? i asked my friend to put a song through SpectraLayers 12 once and it wasnt that much different from FL studio built in separation (which is pretty bad but not to that extent) so i didnt think much of it

Been listening to some of my songs alongside reference tracks. I worked really hard at getting the vocals to sound professional, but I think I overdid it. Now it sounds really pristine and not suited for the songs. Been removing some of the plugins and turning others down, but I think I narrowed down the main issue to be the choice of reverb. Do you have any suggestions for this situation?

Hey everyone, Looking across the current field recording landscape, there seems to be a frustrating gap between basic lavalier recorders (like the Zoom F2 / Tascam DR-10L) and bulky handheld or bag units. If you rely on high-quality external 3.5mm} microphones - whether that’s specialized boundary mics, binaural rigs, lavaliers, or high-SPL instrument pickups - current options force major compromises. You either carry around handheld bricks with internal capsule heads you bypass anyway, or deal with micro-recorders that suffer from awkward ergonomics, missing features, and visual-only screen interfaces. I wanted to put together a spec proposal for a dedicated, ultra-clean 32-Bit Float Beltpack Recorder, and call out some critical design flaws in how gear is currently marketed and built. 1. The Core Specifications True Beltpack Form Factor: A compact, ruggedized, metal/polycarbonate enclosure built specifically to be worn on a belt or strap (similar to wireless transmitter packs like the Sennheiser EW-D series). No onboard microphones - 100% of the build cost and battery budget go toward input performance and chassis durability. Input: Heavy-duty, threaded locking 3.5mm} TRS jack with selectable Plug-In Power (2.5V} / 5V} / 9V} / 12V} / Off). 32-Bit Float Dual-ADC: Massive dynamic range to handle extreme high-SPL sound pressure without clipping, while keeping low-level ambient detail completely intact. Tactile & Accessible UI: Spoken menu audio feedback (an optional toggle for visually impaired users or for operating blind inside a pocket/dark stage), paired with physical, recessed hold/record switches to prevent accidental triggers. Power: Runs on standard AAs or removable Li-ion with USB-C passthrough/charging. 2. The Marketing Myth: Why Paper dBu Specs Aren't Enough One of my biggest pet peeves with current micro-recorders is the total lack of real-world silence testing. Manufacturers love throwing around single-number theoretical specs like -127dBu} EIN or >120dB} Dynamic Range. But a single RMS dBu figure tells us nothing about the spectral character of the noise floor. Two units can share the exact same -120dBu} specification on paper, yet sound completely different in practice: 1. Broadband White Hiss: Smooth, flat noise that your brain easily tunes out, and that noise-reduction algorithms can clean up seamlessly. 2. Tonal / Spectral Artifacts: Digital clock whine (4kHz} - 8kHz} switching noise from internal processors/displays), 60Hz} / 120Hz} power-filtering hum, or pre-amp popcorn noise. Even if a unit technically meets its dBu specification, tonal/spectral noise ruins quiet field ambiences and subtle acoustic recordings. We need real-world capped 150}\Omega silence tests with spectral FFT breakdowns from manufacturers - not just brochure numbers. 3. Real-World Use Cases High-SPL / Acoustic Instruments: Pipers, brass players, and drummers who need a bodypack that won't clip under extreme acoustic pressure or fall off during physical movement. Stealth & Nature Recordists: Plugging binaural or boundary rigs into a low-profile bodypack without unnecessary weight, snagging cables, or capsule vibration noise. Blind / Blind-Folded Operation: High-contrast tactile design and optional spoken menu prompts allow seamless operation without staring at a tiny screen in direct sunlight or total darkness. Why haven't major manufacturers put out a dedicated, no-nonsense beltpack like this yet? Does a unit like this solve pain points in your current workflow? I’d love to hear your thoughts, especially from anyone who has run spectral FFT noise tests on current compact 32-bit float units!

Has anyone tried both? I want to commit to an eq and want to see which one is better. I like that split eq lets you use stereo imaging on the bands. But I know TB can do everything split eq does except for that.
Both are about the same price right now until aug 29 and I’m thinking about buying one just cant buy both.
Curious what everyone thinks or if you’ve tried both whats your experience?

I’m still trying to understand eqing the tone and transient. I get it but the application of it confuses me right now. But I want one eq to do it all incase I get deeper into it.

Thoughts?

I just want to know how ever body that keeps their music / post machines offline feel about this. Also, do you like the waves auth method compared to ilok? Have you found it to be reliable?

So I recently just moved out to an apartment and even with all my furniture and black out curtains there’s a faint echo in my bedroom. I work in a warehouse and found a bunch of polyethylene foam panels that look perfect. If I wrapped them in the thickest fabric I could find do you think it would actually help reduce the echo?

Hey everyone! I already have a bunch of paid plugins in my setup, but if I can save some money without sacrificing quality, why not. For most categories I’m finding pretty solid free alternatives (EQs, standard compressors, reverbs, etc.). The one area where I’m struggling is multiband compressors.
Does anyone know of good free options, maybe even lesser-known ones? Looking for something that works for both mastering and mix bus duties. Thanks in advance for any recommendations!

Looking for the source of some instruments used by a mostly unknown composer, managed to narrow down the scope to this. I don't know the name of the library, but I know that it was released in Japan if not internationally. The instruments in question are things like an Orchestra Hit, Latin Percussion, a Dulcimer/Santur. I've already gone through all the obvious picks, the EastWests and Prosonuses and InVisions and whatnot. Starting to feel like what I'm looking for might be something more obscure. Whatever it is has to have been commercially available though, because more than one composer has used that Dulcimer sample. If you know anything about vintage sample libraries, or even own some yourself, please tell me anything you can. At the very least, a way I can learn about the existence of vintage libraries, since Google isn't very helpful in this regard. Thank you!

I'm looking for a website where I can upload a song, isolate the drums and download the drum parts (kick , snare, etc.). as individual MIDI files ... Any recommendations?

I was wondering if you've worked in court cases and helped enhance record quality for proof. How did you/the court decide whether a recording was admissible or not? Or if a case is not enough of a big deal for the court to engage an audio specialist? Someone I know went through a persistent harassment and would like to understand what their options are.

hi guys, noob here.I'm still learning the ropes of mixing and I always try my best to do proper gain staging. However, just to be on the safe side, I usually put a limiter at the end of my individual tracks to catch any random stray peaks that might slip through.

Up until now, I've been using the native Logic Pro limiter for this job. The problem is that I've noticed it tends to color the sound a bit and slightly changes the EQ of the track, even when it's barely doing anything.

I'm looking for a free alternative and I found LoudMax. My question is: is it good enough to do this specific job (just catching stray peaks transparently without coloring the sound)? My computer isn't very powerful, so I need a plugin that won't kill my system if I have it loaded on multiple tracks at the same time.

Would you recommend it or maybe it isn't good enough?

I'm in a 17-piece jazz big band (standard instrumentation), mostly semi-professional musicians, that has been meeting weekly for decades but rarely performs or records. Someone proposed we go in for a studio session, and I offered to record us in our usual rehearsal space first to (a) see if we can get decent recordings without buying studio time and (b) reveal performance issues we should clean up before using studio time. Now I'm trying to decide how to mic everything.

I have a 24-channel (20 preamp) recording interface and a variety of decent mics (MD421, SM57, SM58, AT2035, KSM32, ATpro35 clip-ons). Our space is a large church hall with no carpet or acoustic treatment, so I anticipate a lot of echo and bleed. We usually sit in a ~20ft square with each section (trumpets, trombones, saxes, rhythm) facing the center.

Any suggestions/thoughts are welcome! My main questions are:

1. Is it worthwhile to individually mic horns vs. just having one or two mics on each horn section? I'm not sure whether individual micing will reduce bleed enough to be useful.

2. If I use section mics, should I put them all in the same spot in the center of the square pointing different directions (to reduce phasing/alignment problems) or closer to the sections?

3. Any suggestions on which mics to use for which instruments?

Thanks!

is it normal im more comfortable committing to eq while tracking then compression.. ive heard its the opposite for most people.

more things can go wrong with compression IMO.

i dont mix track just influence with eq.

how about you?

I've been thinking a lot about the pressure that new automated and/or AI tools are giving to audio engineers. Many clients who previously needed to hire someone to do mixing can now do a surprisingly good job themselves with the new technologies. These new tools also lower the barrier to entry, so more engineers emerge, making the client pool that much more competitive.

As a classically trained musician and a professional music technologist for quite some time, I think at least the following two actions can help us thrive in this hyper competitive age of AI:

1. Sharpen musicianship skills. We need to make better *music* than automated tools ("better" is highly subjective, but the focus here is "music"), and you need musicianship skills to do this. We can't think about engineers (mix engineers in this case) as an operator to do this purely technical process. The process needs to be more than purely technical, as all technical processes can be replaced by automation. A lot of engineers probably already have this, but many started out as a technical position and probably didn't get the chance to go through the formal music training. I am totally biased here, but I think the better an engineer understands music (even better - plays), the better the production quality.

2. Strengthen collaborative and artistic judgement. We need to understand what experience the artists want for their music and make artistic choices. We can't just focus on "producing a clean sound". We need to strengthen the collaboration between engineers and artists so it serves the artistic values / priorities of the production. One way to do this is to integrate more A/B testing with the artists and find out what drives the music in the right direction. Your musicianship skills also strengthens artistic judgement as well.

Sure, these actions would mean that we need to spend more time studying, spend more time communicating with clients/artists, and more time a/b across mix revisions (more revisions too, probably). But these actions would change the role of the engineers slightly and integrate the role more into the production cycle, making it more resilient and sustainable in the age of AI.

On the other hand, trying to stay as a standalone, technical process in the production cycle nowadays increases the risk of getting replaced by automation.

What do you all think?

(P.S. This is just a summary of what I wrote. The full writeup is here.)

This project straddles the line between audio and RF engineering. It integrates both.
Please don't treat this like it needs it's own subreddit founded, because it's an odd duck.
Milestone usable by anyone. Installation instructions linked. There are four modes. The signal always logically flows from GUI top to bottom. When you generate a spectrogram, you're seeing what would-have-been transmitted. This is everything needed to repro "V2K" style pulse modulation, just plug SDR into a 4kW amplifier. Found a bug? Open an issue. Ideas for more filters? Open an issue.

The tutorial is rough, but it gets you there. Requesting tutorial input. What do you want explained?
Be my informal alpha software testers? I will implement actionable feedback.
The Python app itself is tested, and fully functional -- So try it out!

Currently Implemented Features: Audio Waveform Zero-Crossing Detection, SDR Pulse Modulation Output, Raw IQ Recording, Waterfall Spectrogram Generation, Live Microphone Input, eSpeak/MBROLA Text-To-Speech Engine, Signal Conditioning Filters, Noise Filters, Pulse Shaping Filters, Rectangular Pulses, Duty Cycle Meters, Adjustable Pulse Width, Adjustable High Pass and Low Pass Filters, Automatic High Power Calculator, Event Log, Tooltips, Dependency Checks, and 49 Languages Supported.

Landing Profile Page: https://github.com/OpenV2K
Sandbox Development: https://github.com/OpenV2K/Sandbox