微软开源语音转文本模型VibeVoice简介与实测
microsoft/VibeVoice
微软于2026年1月发布开源语音转文本模型VibeVoice,采用MIT许可证并内置说话人日志功能。社区提供的4位量化MLX版本约5.71GB。实测在128GB内存的M5 Max MacBook Pro上,使用mlx-audio工具处理一段60分钟音频耗时约8分45秒,峰值内存占用达30.44GB。模型默认支持最长25分钟音频,通过调整参数可处理至多1小时音频,输出为带时间戳和说话人ID的JSON格式,便于用Datasette Lite浏览分析。
Simon Willison’s Weblog
27th April 2026 - Link Blog
microsoft/VibeVoice. VibeVoice is Microsoft's Whisper-style audio model for speech-to-text, MIT licensed and with speaker diarization built into the model.
Microsoft released it on January 21st, 2026 but I hadn't tried it until today. Here's a one-liner to run it on a Mac with uv, mlx-audio (by Prince Canuma) and the 5.71GB mlx-community/VibeVoice-ASR-4bit MLX conversion of the 17.3GB VibeVoice-ASR model, in this case against a downloaded copy of my recent podcast appearance with Lenny Rachitsky:
uv run --with mlx-audio mlx_audio.stt.generate \
--model mlx-community/VibeVoice-ASR-4bit \
--audio lenny.mp3 --output-path lenny \
--format json --verbose --max-tokens 32768
The tool reported back:
Processing time: 524.79 seconds
Prompt: 26615 tokens, 50.718 tokens-per-sec
Generation: 20248 tokens, 38.585 tokens-per-sec
Peak memory: 30.44 GB
So that's 8 minutes 45 seconds for an hour of audio (running on a 128GB M5 Max MacBook Pro).
I've tested it against .wav and .mp3 files and they both worked fine.
If you omit --max-tokens it defaults to 8192, which is enough for about 25 minutes of audio. I discovered that through trial-and-error and quadrupled it to guarantee I'd get the full hour.
That command reported using 30.44GB of RAM at peak, but in Activity Monitor I observed 61.5GB of usage during the prefill stage and around 18GB during the generating phase.
Here's the resulting JSON. The key structure looks like this:
{
"text": "And an open question for me is how many other knowledge work fields are actually prone to these agent loops?",
"start": 13.85,
"end": 19.5,
"duration": 5.65,
"speaker_id": 0
},
{
"text": "Now that we have this power, people almost underestimate what they can do with it.",
"start": 19.5,
"end": 22.78,
"duration": 3.280000000000001,
"speaker_id": 1
},
{
"text": "Today, probably 95% of the code that I produce, I didn't type it myself. I write so much of my code on my phone. It's wild.",
"start": 22.78,
"end": 30.0,
"duration": 7.219999999999999,
"speaker_id": 0
}
Since that's an array of objects we can open it in Datasette Lite, making it easier to browse.
Amusingly that Datasette Lite view shows three speakers - it identified Lenny and me for the conversation, and then a separate Lenny for the voice he used for the additional intro and the sponsor reads!
VibeVoice can only handle up to an hour of audio, so running the above command transcribed just the first hour of the podcast. To transcribe more than that you'd need to split the audio, ideally with a minute or so of overlap so you can avoid errors from partially transcribed words at the split point. You'd also need to then line up the identified speaker IDs across the multiple segments.
27th April 2026
来源:Simon Willison 博客 · simonwillison.net