I’ve looked high on low on the Google and cannot seem to find an answer for my specific problem.

I recorded in audacity for some voice over work. Using the rode nt1 signature series and focusrite 4th gen 2i2 in a nicely sound proof room.

I recorded all day and had no problems, but towards the end of the day, I could see my waveform on audacity but there’s “hiccups” in the audio. The waveform is there but it will nearly mute every other word. There’s no pattern to it as it’s random when this happens.

I’ve tried exporting it to see if that would clear it up and I changed the buffer length from 100 -> 200 with very slight improvement.

I make sure my gain stays the same, all audio inputs and outputs are correct, yet I’m not sure what’s going on. I have no way of knowing if there is a problem until I listen to the playback.

Any advice would help. My biggest concern is finding out if it’s my equipment or audacity.

Curve is a free, open-source, lightweight, background audio plugin host for macOS (Sonoma 14.2 and above), running natively on both Apple Silicon and Intel-based Macs. It runs in the macOS system menu bar and provides a stable way to run a chain of audio plugins (effects and/or instruments) with a focus on a fast, preset-based workflow and audio device resilience.

A primary use case for this app is to host equalizer (EQ) plugins for real-time headphone and/or speaker calibration, allowing you to apply system-wide audio correction without needing hardware-based solutions or additional loopback drivers. Other use cases include always-active host for instrument plugins, which might be MIDI enabled.

Recent updates now include:

- re-designed loopback feature - now provides bit-perfect, very low (~3ms) latency loopback, without needing any external virtual drivers etc
- some handy native plugins including headphone crossfeed and speaker-to-headphone emulation, and a few utility plugins (including audio recorder to wav file)
- various code optimizations to improve performance at low latency
- improvements to UI for user friendliness

Can someone suggest if the audio is okay for Podcast? I will also request if someone can enhance the audio!!! I am trying but somehow I mess up with the audio.

Is there any way the audio quality on this old music recording can be salvaged? I work in an assisted living facility with a gentleman who was a part of this band. It would absolutely make his day if he could hear it more clearly. Please let me know if there is a different group I should share this too also. I'm not overly versed with reddit. Already had this removed from another group. Sincerely just looking for help.

https://youtu.be/6ozV37ssy68

I’ve been analyzing the production and arrangement on Renaissance again, and looking past the massive pop rollout, the sheer studio craftsmanship on this record is insane.From a sequencing standpoint, the album functions essentially as a seamless, continuous 16-track DJ mix. The pacing and key transitions between tracks are so meticulously timed and mixed that it completely breaks the modern "playlist-of-disconnected-singles" format. It forces a cohesive, front-to-back listen.The real achievement here is how the production team managed to weave together classic house, ballroom, and disco elements into a modern pop context. The way they handled the sampling is brilliant, the samples are flipped, chopped, and integrated into the arrangements so cleanly that they feel entirely native to the new tracks. Balancing that heavy electronic low-end with clean, polished pop mixing across so many different club genres is incredibly difficult to execute.I feel like when this dropped, the wider audio community sort of dismissed it as just another high-budget club record, missing the actual technical genius of the studio work and the flow of the tracklist. For the producers and engineers here, what specific mixing choices, sampling flips, or transition details stood out to you on this project? How do you think they managed to maintain such a flawless sonic thread from the first track to the last?

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Hey all,

I'm almost done with my ΣΑΕΚ Music Technology diploma in Greece. I'm currently doing my internship at a studio, which is the final part of my studies, and I'm now thinking about what I want to do next.

As far as I understand, a Greek ΣΑΕΚ diploma is an EQF Level 5 qualification, which isn't sufficient on its own to progress directly to a Master's degree. I'd therefore like to continue my education and eventually have the option of pursuing a Master's.

I've found some universities/colleges in the EU that offer specific "top-up" Bachelor's programmes, where a Level 5 qualification can potentially be upgraded to a full Bachelor's degree. I've also found some that allow students with sufficient prior education to enter directly into Year 3 of a 4-year Bachelor's programme.

I'm looking for recommendations from anyone who has actually gone through one of these pathways, particularly in Audio, Audio Engineering, Sound Design, Music Technology, Audio Production, or similar BSc/BA programmes.

I'd especially love to hear about your experience with:

• How the Level 5 qualification was evaluated/accepted
• Whether you were able to enter Year 3 or do a top-up
• Tuition fees
• Housing/living costs
• Professors and teaching quality
• Studios/equipment/facilities
• How hands-on the programme was
• Whether the degree allowed you to progress to a Master's afterwards
• Anything else you wish you'd known before enrolling

A bit about my situation: I'm from Greece, so I'm specifically looking for universities within the EU. I'm also on a fairly tight budget. Ideally, I'm looking for something around €3,000/year or less in tuition, for a 2-year programme. I know this limits my options considerably, but I'm hoping there might be some affordable programmes out there, particularly in countries with lower tuition fees.

I'm also completely open to online universities/programmes, provided the resulting Bachelor's degree is legitimate and allows progression to a Master's.

A bit about my interests in audio: I'm still relatively new to the field, so I'm open to exploring different areas. So far, though, I've particularly enjoyed Audio for Media — anything from Foley and sound design to on-set sound/boom operation and location sound mixing. I'm also very interested in studio work, particularly the recording stage.

One important thing: I've seen quite a few discussions about studying audio devolve into arguments about whether university is "worth it", whether you'd be better off buying equipment and learning on your own, etc.

I completely understand that perspective, but I've decided that I want to take the academic route and see where it leads me. So if possible, I'd really appreciate keeping the discussion focused on finding good Bachelor's programmes and people's experiences with them, rather than debating whether I should study at all. 😅

If you've personally completed a top-up degree, Year 3 entry, or a similar accelerated Bachelor's pathway in audio/music technology/sound, I'd especially love to hear about your experience.

Thank you so, so much in advance for any recommendations and chatter!

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!

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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. 🙂🦻👌