GitHub - modelscope/FunASR: Industrial-grade speech recognition toolkit: 170x realtime, 50+ languages, speaker diarization, emotion detection, streaming, and OpenAI-compatible API. (original) (raw)

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FunASR

Industrial speech recognition. 170x faster than Whisper. 50+ languages.
Speaker diarization · Emotion detection · Streaming · One API call

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modelscope%2FFunASR | Trendshift

Quick Start · Colab · Benchmark · Model selection · Migration guide · Use cases · Deployment matrix · Models · Agent Integration · Docs · Contribute


Quick Start

Open In Colab

No local setup? Open the Colab quickstart to transcribe a public sample or upload your own audio in a browser.

pip install torch torchaudio pip install funasr

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="cuda") result = model.generate(input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav")

One call returns VAD segments with speaker id + timestamps — render them however you like:

for seg in result[0]["sentence_info"]: print(f"[{seg['start']/1000:.1f}s] Speaker {seg['spk']}: {rich_transcription_postprocess(seg['sentence'])}")

Output — structured text with speaker labels, timestamps, and punctuation:

[0.6s] Speaker 0: 欢迎大家来体验达摩院推出的语音识别模型

That's it. One model, one call — VAD segmentation, speech recognition, punctuation, speaker diarization all happen automatically.

LLM-powered ASR: Fun-ASR-Nano

For highest accuracy across 31 languages (including Chinese dialects), use Fun-ASR-Nano — an LLM-based ASR combining SenseVoice encoder with Qwen3-0.6B decoder:

from funasr import AutoModel

model = AutoModel(model="FunAudioLLM/Fun-ASR-Nano-2512", vad_model="fsmn-vad", device="cuda") result = model.generate(input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav")

With vLLM acceleration (16x faster, 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: funasr-server --device cuda → OpenAI-compatible endpoint at localhost:8000

Use with AI agents: MCP Server for Claude/Cursor · OpenAI API for LangChain/Dify/AutoGen

Why FunASR?

FunASR Whisper Cloud APIs
Speed 170x realtime 13x realtime ~1x realtime
Speaker ID ✅ Built-in ❌ Needs pyannote ✅ Extra cost
Emotion ✅ Happy/Sad/Angry
Languages 50+ 57 Varies
Streaming ✅ WebSocket
vLLM Acceleration ✅ 2-3x faster N/A
Self-hosted ✅ MIT license ✅ MIT license ❌ Cloud only
Cost Free Free $0.006/min+
CPU viable ✅ 17x realtime ❌ Too slow N/A

Trying FunASR for the first time? Use the Colab quickstart before setting up a local environment. Choosing a first model? Start with the model selection guide. Planning a switch from Whisper or a cloud ASR provider? Use the migration guide and benchmark example to test representative audio, map features, and roll out safely.


Benchmark

184 long-form audio files (192 min). Full report →

Model GPU Speed CPU Speed vs Whisper-large-v3
SenseVoice-Small 170x realtime 17x realtime 🚀 13x faster
Paraformer-Large 120x realtime 15x realtime 🚀 9x faster
Whisper-large-v3-turbo 46x realtime 3.4x faster
Fun-ASR-Nano 17x realtime 3.6x realtime 1.3x faster
Whisper-large-v3 13x realtime baseline

Key takeaway: FunASR models run on CPU faster than Whisper runs on GPU.


What's new


Installation

From source / Requirements

git clone https://github.com/modelscope/FunASR.git && cd FunASR pip install -e ./

Requirements: Python ≥ 3.8. Install PyTorch + torchaudio first (pytorch.org), then pip install funasr.


Model Zoo

Model Task Languages Params Links
Fun-ASR-Nano ASR + timestamps 31 languages 800M 🤗
SenseVoiceSmall ASR + emotion + events zh/en/ja/ko/yue 234M 🤗
Paraformer-zh ASR + timestamps zh/en 220M 🤗
Paraformer-zh-streaming Streaming ASR zh/en 220M 🤗
Qwen3-ASR ASR, 52 languages multilingual 1.7B usage
GLM-ASR-Nano ASR, 17 languages multilingual 1.5B usage
Whisper-large-v3 ASR + translation multilingual 1550M usage
Whisper-large-v3-turbo ASR + translation multilingual 809M usage
ct-punc Punctuation zh/en 290M 🤗
fsmn-vad VAD zh/en 0.4M 🤗
cam++ Speaker diarization 7.2M 🤗
emotion2vec+large Emotion recognition 300M 🤗

Usage

Full examples with parameter docs: Tutorial →

from funasr import AutoModel

Chinese production (VAD + ASR + punctuation + speaker)

model = AutoModel(model="paraformer-zh", vad_model="fsmn-vad", punc_model="ct-punc", spk_model="cam++", device="cuda") result = model.generate(input="https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/asr_example_zh.wav", hotword="关键词 20")

31 languages with timestamps

model = AutoModel(model="FunAudioLLM/Fun-ASR-Nano-2512", hub="hf", trust_remote_code=True, vad_model="fsmn-vad", vad_kwargs={"max_single_segment_time": 30000}, device="cuda") result = model.generate(input="audio.wav", batch_size=1)

Streaming real-time

model = AutoModel(model="paraformer-zh-streaming", device="cuda") result = model.generate(input="chunk.wav", cache={}, chunk_size=[0, 10, 5])

Emotion recognition

model = AutoModel(model="emotion2vec_plus_large", device="cuda") result = model.generate(input="audio.wav", granularity="utterance")

CLI (Agent-Friendly)

Transcribe audio (simplest)

funasr audio.wav

JSON output (for AI agents)

funasr audio.wav --output-format json

SRT subtitles

funasr audio.wav --output-format srt --output-dir ./subs

Speaker diarization + timestamps

funasr audio.wav --spk --timestamps -f json

Choose model and language

funasr audio.wav --model paraformer --language zh

Batch transcribe

funasr *.wav --output-format srt --output-dir ./output

Available models: sensevoice (default), paraformer, paraformer-en, fun-asr-nano


Deploy

OpenAI-compatible API (recommended)

pip install torch torchaudio pip install funasr vllm fastapi uvicorn python-multipart funasr-server --device cuda

→ POST /v1/audio/transcriptions at localhost:8000

Verify it with a public sample:

curl -L https://isv-data.oss-cn-hangzhou.aliyuncs.com/ics/MaaS/ASR/test_audio/BAC009S0764W0121.wav -o sample.wav curl http://localhost:8000/v1/audio/transcriptions
-F file=@sample.wav
-F model=sensevoice
-F response_format=verbose_json

Docker streaming service

docker pull registry.cn-hangzhou.aliyuncs.com/funasr_repo/funasr:funasr-runtime-sdk-online-cpu-0.1.12

OpenAI API example → · Gradio demo → · Client recipes → · JavaScript/TypeScript recipes → · Kubernetes template → · Workflow recipes → · Postman collection → · OpenAPI spec → · Security guide → · Deployment matrix → · Deployment docs → · Agent integration →


Community

📖 Documentation 🐛 Issues
💬 Discussions 🤗 HuggingFace
🤝 Contributing 📈 20k growth plan

Star History

Star History Chart

License

MIT License

Citations

@inproceedings{gao2023funasr, author={Zhifu Gao and others}, title={FunASR: A Fundamental End-to-End Speech Recognition Toolkit}, booktitle={INTERSPEECH}, year={2023} }