VibeVoice-ASR-HF Locally via LM Studio Quantized GGUF Step-by-Step

Deploying this model locally is quickest when done via a simple curl command.

Please follow the instructions listed below to get started.

The installer auto-downloads and deploys the entire model pack.

Without any user input, the software calibrates parameters for optimal hardware usage.

📊 File Hash: 3b8ca68dc09feabceeebacee773b36a6 — Last update: 2026-06-24



  • Processor: high single-core performance needed for token latency
  • RAM: fast 5600MHz+ required to avoid memory bottlenecks
  • Disk Space:70 GB free space for full FP16 weights storage
  • Graphics: TensorRT-LLM / vLLM inference engine compatible chip

The VibeVoice-ASR-HF leverages a transformer-based architecture optimized for low‑latency speech recognition in edge environments. It supports over 100 languages and dialects, delivering real-time transcription with an average word error rate below 5 %. The model achieves sub‑200 ms inference time on standard CPUs, making it suitable for live captioning and voice‑controlled applications. Integrated with popular frameworks through a lightweight API, developers can deploy the model without extensive hardware resources. A comparison of key metrics is provided below.

Parameter Value
Model size ≈ 150 M parameters
Supported languages 100+ languages & dialects
Average latency <200 ms on CPU
Word error rate <5 %
API compatibility REST & gRPC

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