VibeVoice is an open-source family of frontier voice AI models from Microsoft. It covers both speech recognition and speech synthesis, with a focus on long-form audio and real-time streaming.
Note: The original VibeVoice-TTS training code was removed from the repository in September 2025 because of misuse inconsistent with its research purpose. The repository continues to host ASR and real-time TTS models, documentation, and inference examples.
A unified speech-to-text model that processes up to 60 minutes of audio in a single pass.
- Long-form recognition: 60-minute single-pass processing within a 64K token context.
- Rich structured output: Produces transcriptions with speaker labels (Who), timestamps (When), and content (What).
- Multilingual: Supports 50+ languages.
- Customized hotwords: Users can supply domain-specific names or terms to improve accuracy.
- Inference options: Transformers, vLLM, and an edge CPU engine (VibeVoice-ASR-BitNet) that runs without a GPU.
A long-form, multi-speaker text-to-speech model for conversational audio.
- 90-minute generation: Synthesizes long-form speech in a single pass.
- Multi-speaker: Supports up to 4 distinct speakers with natural turn-taking.
- Expressive: Captures conversational dynamics and emotional nuance.
- Multilingual: English, Chinese, and other languages.
The TTS training code is no longer in the official repository, but model weights remain available through community mirrors.
A lightweight real-time streaming TTS model designed for low-latency applications.
- Parameter size: 0.5B (deployment-friendly).
- Low latency: First audible speech in ~300 milliseconds.
- Streaming input: Accepts streaming text while an LLM is still generating.
- Long-form capable: Robust generation up to ~10 minutes.
Clone the repository and install the dependencies:
git clone https://github.com/microsoft/VibeVoice.git
cd VibeVoice
pip install -r requirements.txtFor real-time TTS inference with the 0.5B model:
# Follow the streaming model docs for the exact command
python -m vibevoice.realtime inference \
--model microsoft/VibeVoice-Realtime-0.5B \
--text "Hello, this is a real-time voice demo."For ASR on a long audio file:
python -m vibevoice.asr transcribe \
--model microsoft/VibeVoice-ASR \
--audio long_meeting.wavSee the repository's
docs/folder for the latest model-specific commands and notebooks.
- Continuous speech tokenizers: Acoustic and semantic tokenizers operate at a low 7.5 Hz frame rate, preserving audio fidelity while improving efficiency for long sequences.
- Next-token diffusion framework: An LLM handles textual context and dialogue flow, and a diffusion head generates high-fidelity acoustic details.
- Heterogeneous quantization: The ASR BitNet engine compresses the model from 4.62 GB to 1.58 GB while keeping real-time factor (RTF) below 1 on a 3+ thread CPU.
VibeVoice is intended for research and development. High-quality synthetic speech can be misused for impersonation or disinformation. Microsoft recommends disclosing AI-generated voice content and complying with applicable laws and regulations.
- GitHub Repository
- Project Page
- VibeVoice-ASR on Hugging Face
- VibeVoice-Realtime-0.5B on Hugging Face
VibeVoice is released under the MIT License by Microsoft.