diff --git a/README.en.md b/README.en.md new file mode 100644 index 0000000..4f29f49 --- /dev/null +++ b/README.en.md @@ -0,0 +1,317 @@ +([简体中文](./README_zh.md)|English|[日本語](./README_ja.md)|[한국어](./README_ko.md)) + +

+FunASR +

+ +

+ Industrial speech recognition. Up to 340x realtime, 26x faster than Whisper. 50+ languages.
+ Speaker diarization · Emotion detection · Streaming · One API call +

+ +

+ PyPI + Stars + Downloads + Docs +

+ +

+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](https://colab.research.google.com/assets/colab-badge.svg)](https://colab.research.google.com/github/modelscope/FunASR/blob/main/examples/colab/funasr_quickstart.ipynb) + +No local setup? Open the [Colab quickstart](./examples/colab/) to transcribe a public sample or upload your own audio in a browser. + +```bash +pip install torch torchaudio +pip install funasr +``` + +**Flagship model — Fun-ASR-Nano** (LLM-ASR, 31 languages; the default recommendation, needs a GPU): + +```python +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"]) +# 欢迎大家来体验达摩院推出的语音识别模型。 +``` + +On CPU (or for multilingual + emotion in one pass), use **SenseVoice** — which also returns speaker diarization and timestamps: + +```python +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") # use device="cpu" if you don't have a GPU +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, +) + +# 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. + +### Scale & deploy the flagship + +At scale, accelerate Fun-ASR-Nano with vLLM (batch processing): + +```python +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](examples/mcp_server/) for Claude/Cursor · [OpenAI API](examples/openai_api/) for LangChain/Dify/AutoGen + +### Why FunASR? + +Whisper is a single model; **FunASR is a toolkit** — you pick the right model per job: **Fun-ASR-Nano** (flagship LLM-ASR, GPU, 340x realtime with vLLM, 31 languages), **SenseVoice** (CPU-friendly, + emotion & audio events), **Paraformer** (low-latency streaming). The table shows what the toolkit delivers vs one Whisper model — each capability is labelled with the model that provides it: + +| | FunASR (toolkit) | Whisper | Cloud APIs | +|---|---|---|---| +| Top speed | **340x realtime** (Fun-ASR-Nano + vLLM) | 13x realtime | ~1x realtime | +| Speaker ID | ✅ Built-in | ❌ Needs pyannote | ✅ Extra cost | +| Emotion | ✅ via SenseVoice | ❌ | ❌ | +| Languages | 50+ (Qwen3-ASR 52, Nano 31) | 57 | Varies | +| Streaming | ✅ WebSocket (Paraformer) | ❌ | ✅ | +| CPU viable | ✅ 17x realtime (SenseVoice) | ❌ Too slow | N/A | +| Self-hosted | ✅ MIT license | ✅ MIT license | ❌ Cloud only | +| Cost | Free | Free | $0.006/min+ | + +Trying FunASR for the first time? Use the [Colab quickstart](./examples/colab/) before setting up a local environment. Choosing a first model? Start with the [model selection guide](./docs/model_selection.md). Planning a switch from Whisper or a cloud ASR provider? Use the [migration guide](./docs/migration_from_whisper.md) and [benchmark example](./examples/migration/) to test representative audio, map features, and roll out safely. + +--- + +## Installation + +```bash +pip install funasr +``` + +
From source / Requirements + +```bash +git clone https://github.com/modelscope/FunASR.git && cd FunASR +pip install -e ./ +``` +Requirements: Python ≥ 3.8. Install PyTorch + torchaudio first ([pytorch.org](https://pytorch.org/get-started/locally/)), then `pip install funasr`. + +
+ +--- + +## Model Zoo + +| Model | Task | Languages | Params | Links | +|-------|------|-----------|--------|-------| +| **Fun-ASR-Nano** | ASR + timestamps | 31 languages | 800M | [⭐](https://www.modelscope.cn/models/FunAudioLLM/Fun-ASR-Nano-2512) [🤗](https://huggingface.co/FunAudioLLM/Fun-ASR-Nano-2512) | +| **SenseVoiceSmall** | ASR + emotion + events | zh/en/ja/ko/yue | 234M | [⭐](https://www.modelscope.cn/models/iic/SenseVoiceSmall) [🤗](https://huggingface.co/FunAudioLLM/SenseVoiceSmall) | +| **Paraformer-zh** | ASR + timestamps | zh/en | 220M | [⭐](https://www.modelscope.cn/models/iic/speech_paraformer-large-vad-punc_asr_nat-zh-cn-16k-common-vocab8404-pytorch/summary) [🤗](https://huggingface.co/funasr/paraformer-zh) | +| Paraformer-zh-streaming | Streaming ASR | zh/en | 220M | [⭐](https://modelscope.cn/models/iic/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-online/summary) [🤗](https://huggingface.co/funasr/paraformer-zh-streaming) | +| Qwen3-ASR | ASR, 52 languages | multilingual | 1.7B | [usage](examples/industrial_data_pretraining/qwen3_asr) | +| GLM-ASR-Nano | ASR, 17 languages | multilingual | 1.5B | [usage](examples/industrial_data_pretraining/glm_asr) | +| Whisper-large-v3 | ASR + translation | multilingual | 1550M | [usage](examples/industrial_data_pretraining/whisper) | +| Whisper-large-v3-turbo | ASR + translation | multilingual | 809M | [usage](examples/industrial_data_pretraining/whisper) | +| ct-punc | Punctuation | zh/en | 290M | [⭐](https://modelscope.cn/models/iic/punc_ct-transformer_cn-en-common-vocab471067-large/summary) [🤗](https://huggingface.co/funasr/ct-punc) | +| fsmn-vad | VAD | zh/en | 0.4M | [⭐](https://modelscope.cn/models/iic/speech_fsmn_vad_zh-cn-16k-common-pytorch/summary) [🤗](https://huggingface.co/funasr/fsmn-vad) | +| cam++ | Speaker diarization | — | 7.2M | [⭐](https://modelscope.cn/models/iic/speech_campplus_sv_zh-cn_16k-common/summary) [🤗](https://huggingface.co/funasr/campplus) | +| emotion2vec+large | Emotion recognition | — | 300M | [⭐](https://modelscope.cn/models/iic/emotion2vec_plus_large/summary) [🤗](https://huggingface.co/emotion2vec/emotion2vec_plus_large) | + +--- + +## Usage + +> Full examples with parameter docs: [Tutorial →](https://modelscope.github.io/FunASR/tutorial.html) + +```python +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") + + +# Streaming real-time (feed audio chunk by chunk) +import soundfile as sf +model = AutoModel(model="paraformer-zh-streaming", device="cuda") +audio, sr = sf.read("speech.wav", dtype="float32") # 16 kHz mono +chunk_size = [0, 10, 5] # 600 ms chunks +chunk_stride = chunk_size[1] * 960 +cache = {} +n_chunks = (len(audio) - 1) // chunk_stride + 1 +for i in range(n_chunks): + chunk = audio[i * chunk_stride : (i + 1) * chunk_stride] + res = model.generate(input=chunk, cache=cache, is_final=(i == n_chunks - 1), + chunk_size=chunk_size, encoder_chunk_look_back=4, decoder_chunk_look_back=1) + if res[0]["text"]: + print(res[0]["text"], end="", flush=True) + +# Emotion recognition +model = AutoModel(model="emotion2vec_plus_large", device="cuda") +result = model.generate(input="audio.wav", granularity="utterance") +``` + + +### CLI (Agent-Friendly) + +```bash +# 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 + +```bash +# 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: + +```bash +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 +``` + +```bash +# Docker streaming service +docker pull registry.cn-hangzhou.aliyuncs.com/funasr_repo/funasr:funasr-runtime-sdk-online-cpu-0.1.12 +``` + +### CPU / Edge — llama.cpp / GGUF (no GPU, no Python) + +Run **SenseVoice / Paraformer / Fun-ASR-Nano** as a **single self-contained binary** on CPU and edge devices — this is to FunASR what [whisper.cpp](https://github.com/ggml-org/whisper.cpp) is to Whisper, but with **~3× lower CER than whisper.cpp on Chinese**. Built-in FSMN-VAD, no Python at runtime. + +```bash +# 1) Grab a prebuilt binary from Releases (Linux / macOS / Windows), then: +bash download-funasr-model.sh sensevoice ./gguf # or: paraformer | nano +llama-funasr-sensevoice -m ./gguf/SenseVoiceSmall-f16.gguf --vad ./gguf/fsmn-vad.gguf -a audio.wav +# → 欢迎大家来体验达摩院推出的语音识别模型 +``` + +**Prebuilt binaries:** [Releases](../../releases) · **Download & quickstart:** [funasr.com/llama-cpp](https://www.funasr.com/llama-cpp.html) · **GGUF models:** [Hugging Face](https://huggingface.co/FunAudioLLM) · **Docs & benchmarks:** [runtime/llama.cpp/](./runtime/llama.cpp/) + +[OpenAI API example →](./examples/openai_api/) · [Gradio demo →](./examples/openai_api/GRADIO.md) · [Client recipes →](./examples/openai_api/CLIENTS.md) · [JavaScript/TypeScript recipes →](./examples/openai_api/JAVASCRIPT.md) · [Kubernetes template →](./examples/openai_api/kubernetes/) · [Workflow recipes →](./examples/openai_api/WORKFLOWS.md) · [Postman collection →](./examples/openai_api/POSTMAN.md) · [OpenAPI spec →](./examples/openai_api/OPENAPI.md) · [Security guide →](./examples/openai_api/SECURITY.md) · [Deployment matrix →](./docs/deployment_matrix.md) · [Deployment docs →](./runtime/readme.md) · [Agent integration →](https://modelscope.github.io/FunASR/agent.html) + +--- + +## Benchmark + +> 184 long-form audio files (192 min). [Full report →](https://modelscope.github.io/FunASR/benchmark.html) · [RTFx and reproducibility notes →](./docs/benchmark/rtf_reproducibility.md) + +| Model | Chinese CER ↓ | GPU Speed | CPU Speed | vs Whisper-large-v3 | +|-------|------|-----------|-----------|-------------------| +| **Fun-ASR-Nano** (vLLM) | **8.20%** | **340x** realtime | — | 🚀 **26x faster** | +| **SenseVoice-Small** | **7.81%** | **170x** realtime | **17x** realtime | 🚀 **13x faster** | +| **Paraformer-Large** | 10.18% | **120x** realtime | **15x** realtime | 🚀 **9x faster** | +| Whisper-large-v3-turbo | 21.71% | 46x realtime | ❌ | 3.4x faster | +| Whisper-large-v3 | 20.02% | 13x realtime | ❌ | baseline | + +> **Key takeaway:** FunASR models run on CPU faster than Whisper runs on GPU. + +--- + +## What's new + +- 2026/06/20: **llama.cpp / GGUF runtime** — run SenseVoice / Paraformer / Fun-ASR-Nano on CPU & edge as a single self-contained binary (a whisper.cpp-style alternative), built-in FSMN-VAD, no Python at runtime. Prebuilt binaries for Linux / macOS / Windows + **q8 quantized models (~half the size, same accuracy)**. [runtime/llama.cpp/](./runtime/llama.cpp/) · [Releases](../../releases) +- 2026/06/21: **v1.3.12** on PyPI — rolling fixes (qwen3-asr language codes, glm_asr, vLLM repetition_penalty). `pip install --upgrade funasr` +- 2026/05/24: **vLLM Inference Engine** — 2-3x faster LLM decoding for Fun-ASR-Nano. Streaming WebSocket service with VAD + Speaker Diarization. [Guide →](docs/vllm_guide.md) · [Realtime WS tuning →](docs/vllm_guide.md#67-production-concurrency-and-multi-process-deployment) · [API stability checklist →](docs/vllm_guide.md#production-api-stability-checklist) +- 2026/05/24: **Dynamic VAD** — adaptive silence threshold (default on). Short sentences stay intact, long segments get auto-split. [Details →](docs/vllm_guide.md#附录dynamicstreamingvad) +- 2026/05/24: **v1.3.3** — `funasr-server` CLI, OpenAI-compatible API, MCP Server for AI agents. `pip install --upgrade funasr` +- 2026/05/20: Added Qwen3-ASR (0.6B/1.7B) — 52 languages, auto detection. [usage](examples/industrial_data_pretraining/qwen3_asr) +- 2026/05/20: Added GLM-ASR-Nano (1.5B) — 17 languages, dialect support. [usage](examples/industrial_data_pretraining/glm_asr) +- 2026/05/19: Fun-ASR-Nano and SenseVoice now support speaker diarization. +- 2025/12/15: [Fun-ASR-Nano-2512](https://github.com/FunAudioLLM/Fun-ASR) — 31 languages, tens of millions of hours training. + +
Older + +- 2024/10/10: Whisper-large-v3-turbo support added. +- 2024/07/04: [SenseVoice](https://github.com/FunAudioLLM/SenseVoice) — ASR + emotion + audio events. +- 2024/01/30: FunASR 1.0 released. + +
+ +--- + +## Community + +| | | +|---|---| +| 📖 [Documentation](https://modelscope.github.io/FunASR/) | 🐛 [Issues](https://github.com/modelscope/FunASR/issues) | +| 💬 [Discussions](https://github.com/modelscope/FunASR/discussions) | 🤗 [HuggingFace](https://huggingface.co/funasr) | +| 🤝 [Contributing](./CONTRIBUTING.md) | 🌐 [funasr.com](https://www.funasr.com) | +| 🧩 [Community projects](./docs/community_projects.md) | 💡 [Use-case showcase](./docs/use_case_showcase.md) | + +## Star History + + + + + + Star History Chart + + + +## License + +[MIT License](./LICENSE) + +## Citations + +```bibtex +@inproceedings{gao2023funasr, + author={Zhifu Gao and others}, + title={FunASR: A Fundamental End-to-End Speech Recognition Toolkit}, + booktitle={INTERSPEECH}, + year={2023} +} +```