Show HN: Moonshine Open-Weights STT models – higher accuracy than WhisperLargev3 (github.com)

316 points by petewarden ↗ HN
I wanted to share our new speech to text model, and the library to use them effectively. We're a small startup (six people, sub-$100k monthly GPU budget) so I'm proud of the work the team has done to create streaming STT models with lower word-error rates than OpenAI's largest Whisper model. Admittedly Large v3 is a couple of years old, but we're near the top the HF OpenASR leaderboard, even up against Nvidia's Parakeet family. Anyway, I'd love to get feedback on the models and software, and hear about what people might build with it.

42 comments

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This is awesome, well done guys, I’m gonna try it as my ASR component on the local voice assistant I’ve been building https://github.com/acatovic/ova. The tiny streaming latencies you show look insane
No idea why 'sudo pip install --break-system-packages moonshine-voice' is the recommended way to install on raspi?

The authors do acknowledge this though and give a slightly too complex way to do this with uv in an example project (FYI, you dont need to source anything if you use uv run)

How does this compare to Parakeet, which runs wonderfully on CPU?
haven't tested yet but I'm wondering how it will behave when talking about many IT jargon and tech acronyms. For those reason I had to mostly run LLM after STT but that was slowing done parakeet inference. Otherwise had problems to detect properly sometimes when talking about e.g. about CoreML, int8, fp16, half float, ARKit, AVFoundation, ONNX etc.
onnx models for browser possible?
For those wondering about the language support, currently English, Arabic, Japanese, Korean, Mandarin, Spanish, Ukrainian, Vietnamese are available (most in Base size = 58M params)
According to the OpenASR Leaderboard [1], looks like Parakeet V2/V3 and Canary-Qwen (a Qwen finetune) handily beat Moonshine. All 3 models are open, but Parakeet is the smallest of the 3. I use Parakeet V3 with Handy and it works great locally for me.

[1]: https://huggingface.co/spaces/hf-audio/open_asr_leaderboard

To this comment and all the other comments talking about handy below this comment. I tried handy right now and it's super amazing. I'm speaking this from Handy. This is so cool, man.

And handy even takes care of all the punctuation, which is really nice.

Thanks a lot for suggesting it to me. I actually wanted something like this, and I was using something like Google Docs, and it required me to use Chrome to get the speech to text version, and I actually ended up using Orion for that because Orion can actually work as a Chrome for some reason while still having both Firefox and Chrome extension support. So and I had it installed, but yeah.

This is really amazing and actually a sort of lifesaver actually, so thanks a lot, man.

Now I can actually just speak and this can convert this to text without having to go through any non-local model or Google Docs or whatever anything else.

Why is this so good man? It's so good

man, I actually now am thinking that I had like fully maxed out my typing speed to like hundred-120. But like this can actually write it faster. you know it's pretty amazing actually.

Have a nice day, or as I abbreviate it, HAND, smiley face. :D

Any plans regarding JavaScript support in the browser?

There was an issue with a demo but it's missing now. I can't recall for sure but I think I got it working locally myself too but then found it broke unexpectedly and I didn't manage to find out why.

Very cool. Anyway to run this in Web assembly, I have a project in mind
Accuracy is often presumed to be english, which is fine, but it's a vague thing to say "higher" because does it mean higher in English only? Higher in some subset of languages? Which ones?

The minimum useful data for this stuff is a small table of language | WER for dataset

Streaming transcription is crazy fast on an M1. Would be great to use this as a local option versus Wispr Flow.
I've helped many Twitch streamers set up https://github.com/royshil/obs-localvocal to plug transcription & translation into their streams, mainly for German audio to English subtitles.

I'd love a faster and more accurate option than Whisper, but streamers need something off-the-shelf they can install in their pipeline, like an OBS plugin which can just grab the audio from their OBS audio sources.

I see a couple obvious problems: this doesn't seem to support translation which is unfortunate, that's pretty key for this usecase. Also it only supports one language at a time, which is problematic with how streamers will frequently code-switch while talking to their chat in different languages or on Discord with their gameplay partners. Maybe such a plugin would be able to detect which language is spoken and route to one or the other model as needed?

I released a OBS plugin (and optional RTMP relay) that does exactly this. It can do real time translated captions and voice cloning/dubbing. The plugin lets you choose an audio source, then creates each language's captions and dub as new Sources. Use them however you'd like! check it out! https://streamfluent.ai
Implemented this to transcribe voice chat in a project and the streaming accuracy in English on this was unusable, even with the medium streaming model.
Claiming higher accuracy than Whisper Large v3 is a bold opening move. Does your evaluation account for Whisper's notorious hallucination loops during silences (the classic 'Thank you for watching!'), or is this purely based on WER on clean datasets? Also, what's the VRAM footprint for edge deployments? If it fits on a standard 8GB Mac without quantization tricks, this is huge.
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Tangentially, have you got any idea what the equivalent "partial tokens revised" rate for humans is? I know I've consciously experienced backtracking and re-interpreting words before, and presumably it happens subconsciously all the time. But that means there's a bound on how low it's reasonable to expect that rate to be, and I don't have an intuition for what it is.
Do you also support timestamps the detected word or even down to characters?
fyi the typepad link in your bio is broken
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> Models for other languages are released under the Moonshine Community License, which is a non-commercial license.

Weird to only release English as open weights.

Very exciting stuff!

    hear about what people might build with it
My startup is making software for firefighters to use during missions on tablets, excited to see (when I get the time) if we can use this as a keyboard alternative on the device. It's a use case where avoiding "clunky" is important and a perfect usecase for speech-to-text.

Due to the sector being increasingly worried about "hybrid threats" we try to rely on the cloud as little as possible and run things either on device or with the possibility of being self-hosted/on-premise. I really like the direction your company is going in in this respect.

We'd probably need custom training -- we need Norwegian, and there's some lingo, e.g., "bravo one two" should become "B-1.2". While that can perhaps also be done with simple post-processing rules, we would also probably want such examples in training for improved recognition? Have no VC funding, but looking forward to getting some income so that we can send some of it in your direction :)

The streaming architecture looks really promising for edge deployments. One thing I'm curious about: how does the caching mechanism handle multiple concurrent audio streams? For example, in a meeting transcription scenario with 4-5 speakers, would each stream maintain its own cache, or is there shared state that could create bottlenecks?