FunASR
modelscope · modelscope/FunASR
FunASR is an open-source speech recognition toolkit providing training, inference, and various audio processing pipelines, with OpenAI-compatible/MCP serving capabilities.
Open-source speech recognition toolkit for training, inference, streaming ASR, VAD, punctuation, speaker diarization pipelines, and OpenAI-compatible/MCP serving.
Preview
What you can do with it
- Transcribe spoken audio from meetings or voice notes into text for an AI agent to analyze.
- Build an agent that can process and respond to voice commands using real-time ASR.
- Develop an audio-event detection system to categorize sounds or speech patterns.
Safety check
No risky patterns found in the files we scanned.
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From the README
Industrial speech recognition toolkit for offline, streaming, and edge deployment. ASR · VAD · punctuation · speaker pipelines · emotion and audio-event models · OpenAI-compatible serving
Quick Start · Model selection · Models · Deployment matrix · Deployment hub · Docs · Benchmark · Contribute
Quick Start
Native Transformers
For Fun-ASR-Nano transcription with the Hugging Face API, start with the Transformers 5.17.0 CPU quickstart. No FunASR toolkit or remote Python code is needed.
Space · Notebook · Python / batch examples
FunASR toolkit and pipelines
No local setup? Open the Colab quickstart to transcribe a public sample or upload your own audio in a browser.
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# CPU-only installs can use the default PyPI wheels.
pip install torch torchaudio
pip install funasr
For GPU quickstarts, install the PyTorch and torchaudio wheels that match your NVIDIA driver from pytorch.org before installing FunASR. After installation, confirm the GPU is visible:
python - <<'PY'
import torch
print(torch.cuda.is_available())
PY
Only use device="cuda" when this prints True; otherwise use device="cpu"
or reinstall PyTorch with the correct CUDA wheel.
FunASR toolkit GPU example: Fun-ASR-Nano (Chinese, English, Japanese, and Chinese dialect groups and regional accents; the separate native Transformers CPU path is linked above):
from funasr import AutoModel
model = AutoModel(model="FunAudioLLM/Fun-ASR-Nano-2512", device="cuda")
result = model.generate(input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav")
print(result[0]["text"])
For the separate 31-language checkpoint, use Fun-ASR-MLT-Nano-2512. Language coverage is checkpoint-specific, so Nano and MLT-Nano should be treated as distinct model choices.
For a CPU-first example with five-language ASR plus emotion and audio-event tags, use SenseVoiceSmall. The pipeline below combines it with FSMN-VAD and CAM++ for speaker-aware VAD segments; these are not native speaker outputs of the SenseVoiceSmall checkpoint. See the SenseVoice paper, Hugging Face checkpoint, and GGUF edge checkpoint.
from funasr import AutoModel
from funasr.utils.postprocess_utils import rich_transcription_postprocess
model = AutoModel(model="iic/SenseVoiceSmall", vad_model="fsmn-vad", spk_model="cam++", device="cpu")
result = model.generate(
input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav",
batch_size_s=300,
)
# The AutoModel pipeline returns VAD segments with speaker ids and timestamps:
for seg in result[0]["sentence_info"]:
print(f"[{seg['start']/1000:.1f}s] Speaker {seg['spk']}: {rich_transcription_postprocess(seg['sentence'])}")
This prints each returned segment's start time in seconds, anonymous speaker index, and text with SenseVoice tags removed. Text and segment boundaries depend on the audio and checkpoint; no fixed transcript is asserted here.
CAM++ extracts spk_embedding vectors. AutoModel clusters those embeddings
and assigns speaker indices to VAD segments. Indices are local to a recording,
not known-person identities. See the SDK contract for
the component and result boundaries. Change to device="cuda" only after
verifying a compatible GPU environment as described above.
Scale & deploy the flagship
At scale, accelerate Fun-ASR-Nano with vLLM (batch processing):
from funasr.auto.auto_model_vllm import AutoModelVLLM
model = AutoModelVLLM(model="FunAudioLLM/Fun-ASR-Nano-2512", tensor_parallel_size=1)
results = model.generate(["audio1.wav", "audio2.wav"], language="auto")
Deploy as API server: Local SenseVoice CPU recipe · Nano GPU serving and pinned vLLM setup
Use with AI agents: MCP Server for Claude/Cursor · OpenAI API for LangChain/Dify/AutoGen
Use with voice agents: OpenClaw realtime plugin for self-hosted Talk and Voice Call transcription · Hugging Face speech-to-speech with SenseVoice (merged source checkout; not in v1.0.0)
Why FunASR?
FunASR is a toolkit: choose the task, checkpoint, and runtime separately. Support in one model or adapter does not imply support in every serving backend.
| Task | Checkpoint or pipeline | Runtime entrypoint | Important limitation |
|---|---|---|---|
| File transcription with emotion/event tags | SenseVoiceSmall | Python AutoModel, CPU or GPU | Five-language checkpoint; tags do not identify speakers. |
| LLM-based file transcription | Fun-ASR-Nano | AutoModel; split-engine AutoModelVLLM for the documented GPU path | Base Nano covers zh/en/ja and Chinese dialects/accents; timestamp support depends on checkpoint and path. |
| Broader multilingual transcription | Fun-ASR-MLT-Nano | Python AutoModel | Separate 31-language checkpoint; do not transfer its coverage to base Nano. |
| Chunked live transcription | Paraformer-zh-streaming | Streaming SDK or runtime WebSocket service | Use the streaming checkpoint and per-session cach |