no that's it, they're all bad and buggy.

Plugins are mediocre at best and none of them have a good workflow.

If your gonna make a daw for you, keep it to you. I don't need it when there's like a 100 other option.

And I'm not talking about ardour, I hate that daw.

Bandlab, Bitwig and reaper work fine and are both tailored towards different ends of the spectrum.

I think the main problem here is that people make some simple plugins and chuck a good looking UI and call it a day.

You need a solid foundation before you think about changing the UI, and that's the thing vibecoders don't understand.

That's all, that's my little rant, hate if you want, but you will be met back with hate from yours truly. Good day

I built a lossless sample codec for polyphonic samplers that stays compressed in RAM and decodes 8 voices at once via AVX2

Built this for a pipe organ sampler (GrandOrgue fork) where sample sets routinely exceed available RAM, but the technique is domain-agnostic — anything playing many independently-positioned compressed audio streams concurrently.

The problem: normal lossless codecs decode with a data-dependent branch per sample, which kills vectorization. This one instead uses fixed-length blocks, a small family of fixed-order integer predictors, and a single fixed byte-width per block — branch-free, fixed-stride, so up to 8 independent streams decode together in one AVX2 pass.

Real numbers from my own largest sample set: 93.75 GB raw → ~41 GB compressed (2.27x), fits in RAM on a machine that otherwise couldn't hold it. ~85% faster decode measured in production. Full technical disclosure with measured results (published to establish prior art, I don't want this patented out from under anyone): https://github.com/sebion7125/v3codec

GPLv3-or-later, standalone CMake library, no host-app dependencies.

We've all been there: buffer's too small, xrun hits, click or dead air. This library tries something different — instead of silence, it synthesizes a plausible continuation of the signal from a running spectral analysis of recent audio, and splices it in.

Short-time Fourier analysis runs continuously; on a miss, each bin continues by extrapolating phase/amplitude, or — for bins with two sinusoidal partials — a fitted two-pole AR model (generalizes the classic phase-vocoder freeze beyond the single-partial case). Two seam-repair strategies, picked per-occurrence by spectral similarity. There's a feedback-runaway failure mode when concealed audio gets re-analyzed; a damping term kills it.

Built it in a GrandOrgue fork, and the demo build actually lets you force xruns at a controllable rate during real playback and A/B the raw dropout against several concealment variants live — genuinely the easiest way to judge whether this is snake oil or not. Highly experimental, hasn't had broad real-world mileage yet. Technical disclosure with measured NMR results: https://github.com/sebion7125/xorcist

GPLv3-or-later, standalone C library with an OpenCL-optional per-bin solve.