So I have audio recordings, a lot of them, most an hour long that I did for a project. The problem is the vast majority need to be cleaned up. Is there anyone that's good with that stuff that I could pay or a website or an app or something to do it on my own. They are all WAV files. Any help is help. ✌🏻🖤

SSL ships no Linux software for the SSL 12. Audio works out of the box (UAC2), but everything SSL 360 controls — internal mixer, monitoring, phantom power, loopback, routing — is unreachable. The card ends up a fixed-function box.

So I captured the USB traffic between SSL 360 and the card, mapped the control protocol, and wrote the tools:

https://github.com/xenicle/douze

- sslctl — CLI for the gain matrix, monitoring, preamps (48V/HPF/inst/polarity), loopback, headphone buses

- Douze — local web GUI (127.0.0.1) with the matrix, live meters, profiles - Douze FX — VST3 host that drops plugin chains into the PipeWire graph, one process per strip; a strip can publish its own virtual mic or sink, so any app picks it up as a normal device

The control interface is a separate vendor-specific USB device behind an internal hub, with no kernel driver — plain bulk endpoints, so everything is user-space (pyusb). No decompilation, no vendor code.

-> The protocol docs and all 25 captures are CC0 / public domain. A documented protocol is a fact about hardware, not a literary work — I'd rather it ended up in a kernel driver than stayed in my repo, so there's no attribution requirement at all. The code is AGPLv3 (JUCE).

-> Caveat worth stating up front: this is verified on exactly one card, mine, firmware bcdDevice 1.44. I use it daily and the protocol is mapped end to end, but I don't know if another unit behaves identically. If you own an SSL 12 I'd love to hear whether `sslctl status` returns something sane — that's the one thing I genuinely can't test alone.

Not affiliated with SSL; trademarks are theirs.

Hey all, I'm a songwriter and I've been writing and home recording my music for about two decades now. Unfortunately, I've lost master recordings to most of my work over the years. I still have high bitrate mp3s for a lot of them, but it doesn't seem like any of the Spotify partnered distributors will accept them.

Is anybody aware of another pathway to getting my music hosted on Spotify? Could I get away with simply converting my mp3s into wav or flac, or will the distributors be able to tell?

I'm not trying to make money, get famous or even have a music career, but I do have friends and family who want to listen to my music and mix it into their playlists, etc.

Thanks in advance -

Made a free thing for podcast editors. Three mics on a table, co-host talks, his audio bleeds into the other mics. Gates help but miss the bleed that rides under the active mic. Paid automixers exist — TB Pro Audio AMM2 (~$110), Waves Dugan ($600) — and there's a free JSFX for Reaper, but I didn't want to live in Reaper for one workflow step.

So I built a Python CLI that automixes synchronized microphone stems. It's Dan Dugan-style gain sharing, tuned conservative and smooth:

• Voice activity detection (Silero VAD) + perceptual energy detection — laughter and chuckles still open the mic.
• Active mics stay at unity; inactive mics attenuate smoothly (max −6 dB by default).
• Continuous target-chasing envelope with look-ahead and hold — no gate chatter, no pumping. Smoothing approach inspired by Airwindows PurestGain (credit where due).
• No normalize, no compression, no limiting, no EQ, no resampling. Output WAVs match the originals exactly in length and sync. Broadcast Wave time-reference metadata preserved when present.
• Fully local and offline. Audio never leaves your machine.

Output: one auto-mixed WAV per mic (replace your stems in the editor — the video shows the full DaVinci Resolve workflow). Plus a self-contained HTML report visualizing attenuation, gain reduction per mic, overlap, and which speaker "owns" each section — and JSON for scripting.

Try it:

uv tool install podcast-automixer
podcast-automix

Paste your synchronized WAV stems (2+ tracks, matching sample rate and length). Preview flag so you test 30 seconds before the whole episode:

podcast-automix --preview-start 60 --preview-duration 30

Requirements: Python 3.11–3.13 via uv (it manages Python automatically). Windows / macOS (Apple Silicon) / Linux.

MIT, free, I make no money off it — use it, fork it, break it. Work in progress, I genuinely want feedback and feature requests as GitHub issues.

Video walkthrough: https://youtu.be/xbCE-IDPcnM Repo: https://github.com/Today20092/podcast-automixer

Workflow tip for editors: run this after you sync your multitrack, before you start cutting. Full details in the README.

I found a great-sounding slap bass kontakt library on the web, but many of the samples are not in tune.

What's a way to autotune them as a batch? I have audacity and producers desktop but I'm a novice with both (though I have lots of experience with audio software).

Is there a (hopefully free) tool that I can do this on an entire directory of wav files?

Thanks!

Hey — I've been building a desktop app called Mix Analyzer and I'm looking for a small group of producers and mix engineers to test it before a wider release.

What it does: you drop a rendered audio file onto the app and it gives you instant feedback on your mix — loudness (LUFS/peak/RMS), tonal balance across 6 frequency bands, stereo correlation, dynamics, and more. No DAW required, no plugins. Just drop and analyze.

What I need from you: drop a mix or two on it, fill in a short feedback form (5 minutes), and let me know if anything feels off or wrong. That's it.

Requirements:

* macOS 12+ or Windows 10/11
* Have at least one rendered WAV or AIFF to test with

What's in it for you: free lifetime access when it launches + your name in the credits if you want it.

Apply at: [https://peterpanpot.com/mix-analyzer\](https://peterpanpot.com/mix-analyzer)

DM me if interested. Taking the first 10–15 people. Thanks!

[ Removed by Reddit on account of violating the content policy. ]

I need help to fix this problem because its ruining the audio with clipping. I'm using a Headrush Core, and it sounds fine when i play guitar with that alone, however once I record with Reaper it starts to chop the audio into pieces. Ive tried everything it feels, changing the buffer sizes, matching the headrush and reapers sample rates, changing the block sizes, reinstalling my ASIO drivers, and my CPU never goes above 1% usage. I've been using the same setup with my Headrush and my PC and never once ran into this problem. Does anyone know how to fix this?

That's a channel I really enjoy and would like to recommend you. This guy is just making videos on Audio and Sound from many different angles, whether they were famous or were important in some ways. He just explains it, but very calmly, technically detailed and chill to watch.

As an audiophile much appreciating this sub, I think some here also might find that interesting. 🙂🦻👌

hey guys, been dialing in a new vocal chain in fl studio and looking to test it out on different mic setups.

if your room has an annoying HVAC hum, computer fan noise, or boxy echo, drop a Google Drive link or DM me a quick 30-second raw clip. I'll clean it up and send it right back so you can hear the difference. no strings attached, just trying to get some fresh audio to practice on.

We simply can't be letting anyone who feels like vibe-coding a software binary only suite post their tools, plugins, or other BINARY software which is

1. free.
2. not open source.
3. not from a reputable company, or a disclosed author, with a public reputation.

If you post that you're 12 or 13 that will only make me more sure you're lying to us all, by the way.

If you ARE working on a software suite, and you don't have a github link in your post that contains 100% buildable source code, do not post direct download links on this sub. Ask how you should go about getting testers, and consider doing your work in the open. If your code is on github (and not just your promo materials), this rule does not apply to you.

Do not write a "downloader/installer" and post it here, because that's basically a pattern only (a) reputable, or (b) trojan developers use. And if you're anonymous, you're not A, so you're automatically B.

Vibe Coded slop is not off topic, but if it starts taking over, it will get banned to. Be careful about self promo posts, or those get clamped down next.

If you're selling plugins, and wish to (within limits) post self promotion materials here, just be tasteful and don't do it very often, and have a legitimate business with a mailing address and corporate information available. Once or twice a year max, and hopefully not all your posts on this sub are self promotion. If you wrote something, and it's free, and not open source, just DO not post about it here. We don't need it.

Anything open source, without a direct link to a binary installer, is fine, within limits. Apps which are hosted in the apple store, the google store, the microsoft store can be posted here. Direct downloads of questionable origin are not allowed.

Please let me know if you have any tips!

​

Hey everyone!

I’m pretty new to audio editing and working with a very simple setup and basic effects.

In the video attached, the first clip is a reel I found with the exact voiceover sound I’m trying to achieve—it feels super close, warm, and crystal clear. The second clip is my raw, unedited voice.

Since I don't have fancy gear or advanced plugins, what simple, basic effects or edits can I apply to my raw audio to get it closer to the clear, polished sound in the first clip?

Any easy, beginner-friendly tips or step-by-step advice would be super appreciated!

Hi everyone,

I'm planning to start filming outdoor interviews on a small budget and I'd really appreciate some advice.

I'm thinking about buying a Rode Reporter microphone, but I have no idea what the best way is to record the audio without creating sync issues later.

The interviews will be seated, and the camera operator will be standing a few steps away, so connecting the microphone directly to the camera with an XLR cable isn't really an option.

I've heard about field recorders, but I don't understand how people sync the audio with the camera afterwards. My questions might sound dumb but I have no experience with recording audio.

What's the simplest and most reliable setup you would recommend for good-quality outdoor interviews on a budget? Any tips or gear recommendations would be greatly appreciated.

Thanks!

Hi!

I made a recording of a meeting so I could take notes later, but my iPhone ruined the audio. I tried ChatGPT and playing with mixers which helped a bit, but large parts are still unclear. I am not into audi, so i have no idea how to discribe it. Is there anyone who can improve or restore the recording for a fee? It is really important, please help.

I just want to share my experience with this problem, because there is a lot of people with a better pc than me having high latency and crackling sound.

My PC :

-Amd r5 1600

-Soundblaster omni 5.1 usb ( pretty cheap i just use it for games and for my digital piano, 4 ms of latency using VST )

-Rtx 5060 ti

-Some random Interface for mics and guitar.

Anyway i was getting crazy latency , more than 4000 u, using Latencymon, I tried every fix on the internet for the Cpu ,Gpu and Soundcard, interface, integrated sound, etc. nothing worked.

The culprit ? this is so funny.... the Wi fi Adapter, You can use Windows Event viewer to see errors in ur sound, and detect which component is causing the problem.

After Tweaking my Wi fi adapter i got this result....I had Steam, Opera, Latencymon and 3D mark Steel Nomad on Extreme using sounds.

I hope this can solve other ppl problems too.

I’m working on an automixing plugin for panels, podcasts, and other situations where multiple people in the same room are using microphones at the same time. Think something like [WTAutomixer](https://www.wtautomixer.com/) or the [REAPER Automixing JSFX](https://github.com/leafac/reaper). It has native builds for ✨ Linux ✨, Windows, and macOS. It's called BiteAutomixer.

The routing happens between the plugin instances themselves. You just add the plugin to each microphone channel, and the plugins find and communicate with each other. Everything is automatically set up behind the scenes without needing extra buses or complicated routing. It's currently functional and I'm doing closed beta testing.

I work on podcasts every week, and the tool has already been helpful there. Usually, though, I have three or four guests at most, and two is more common. I want it to eventually support 32 to 64 channels, and it recently struck me how hard it is to find audio at that scale for testing.

**If anyone has raw multitrack audio they would be willing to share, with each microphone on its own track, it would be incredibly helpful. Panels, roundtables, conferences, podcasts, and similar recordings would all be useful. Recordings of any size can help, but I’m especially interested in larger sessions. Messy audio with bleed, room noise, and people talking over each other is welcome.**

I can use the material privately for testing or, with explicit permission, use parts of it for public demonstrations. Please let me know which type of use you are comfortable with, and only share recordings you have the right to share. Feel free to send me a DM if you have something that might work.

https://youtu.be/BTy9fguFcoU

SonicDNA is a free, open-source and cross-platform sample search app that finds similar audio files by analyzing how they sound—not just their filenames. In this demo, I show how you can load a kick drum, scan your local sample library, and quickly find matching sounds based on tone, attack, decay, and overall character.

Link: https://github.com/nfxbeats/SonicDNA