QwenLM/Qwen3-ASR

GitHub: QwenLM/Qwen3-ASR

Qwen3-ASR 是阿里通义千问团队开源的多语言语音识别模型系列,支持 52 种语言和方言的 ASR、语种识别及时间戳预测。

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# Qwen3-ASR

  🤗 Hugging Face   |   🤖 ModelScope   |   📑 Blog   |   📑 Paper  
🖥️ Hugging Face Demo   |    🖥️ ModelScope Demo   |   💬 WeChat (微信)   |   🫨 Discord   |   📑 API

## News ## Contents - [Overview](#overview) - [Introduction](#introduction) - [Model Architecture](#model-architecture) - [Released Models Description and Download](#released-models-description-and-download) - [Quickstart](#quickstart) - [Environment Setup](#environment-setup) - [Python Package Usage](#python-package-usage) - [Quick Inference](#quick-inference) - [vLLM Backend](#vllm-backend) - [Streaming Inference](#streaming-inference) - [ForcedAligner Usage](#forcedaligner-usage) - [DashScope API Usage](#dashscope-api-usage) - [Launch Local Web UI Demo](#launch-local-web-ui-demo) - [Gradio Demo](#gradio-demo) - [Streaming Demo](#streaming-demo) - [Deployment with vLLM](#deployment-with-vllm) - [Fine Tuning](#fine-tuning) - [Docker](#docker) - [Evaluation](#evaluation) - [Citation](#citation) ## Overview ### Introduction

### Model Architecture

### Released Models Description and Download Below is an introduction and download information for the Qwen3-ASR models. Please select and download the model that fits your needs. | Model | Supported Languages | Supported Dialects | Inference Mode | Audio Types | |---|---|---|---|---| | Qwen3-ASR-1.7B & Qwen3-ASR-0.6B | Chinese (zh), English (en), Cantonese (yue), Arabic (ar), German (de), French (fr), Spanish (es), Portuguese (pt), Indonesian (id), Italian (it), Korean (ko), Russian (ru), Thai (th), Vietnamese (vi), Japanese (ja), Turkish (tr), Hindi (hi), Malay (ms), Dutch (nl), Swedish (sv), Danish (da), Finnish (fi), Polish (pl), Czech (cs), Filipino (fil), Persian (fa), Greek (el), Hungarian (hu), Macedonian (mk), Romanian (ro) | Anhui, Dongbei, Fujian, Gansu, Guizhou, Hebei, Henan, Hubei, Hunan, Jiangxi, Ningxia, Shandong, Shaanxi, Shanxi, Sichuan, Tianjin, Yunnan, Zhejiang, Cantonese (Hong Kong accent), Cantonese (Guangdong accent), Wu language, Minnan language. | Offline / Streaming | Speech, Singing Voice, Songs with BGM | | Qwen3-ForcedAligner-0.6B | Chinese, English, Cantonese, French, German, Italian, Japanese, Korean, Portuguese, Russian, Spanish | -- | NAR | Speech | During model loading in the `qwen-asr` package or vLLM, model weights will be downloaded automatically based on the model name. However, if your runtime environment does not allow downloading weights during execution, you can use the following commands to manually download the model weights to a local directory: # Download through ModelScope (recommended for users in Mainland China) pip install -U modelscope modelscope download --model Qwen/Qwen3-ASR-1.7B --local_dir ./Qwen3-ASR-1.7B modelscope download --model Qwen/Qwen3-ASR-0.6B --local_dir ./Qwen3-ASR-0.6B modelscope download --model Qwen/Qwen3-ForcedAligner-0.6B --local_dir ./Qwen3-ForcedAligner-0.6B # Download through Hugging Face pip install -U "huggingface_hub[cli]" huggingface-cli download Qwen/Qwen3-ASR-1.7B --local-dir ./Qwen3-ASR-1.7B huggingface-cli download Qwen/Qwen3-ASR-0.6B --local-dir ./Qwen3-ASR-0.6B huggingface-cli download Qwen/Qwen3-ForcedAligner-0.6B --local-dir ./Qwen3-ForcedAligner-0.6B ## Quickstart ### Environment Setup The easiest way to use Qwen3-ASR is to install the `qwen-asr` Python package from PyPI. This will pull in the required runtime dependencies and allow you to load any released Qwen3-ASR model. If you’d like to simplify environment setup further, you can also use our official [Docker image](#docker). The `qwen-asr` package provides two backends: the transformers backend and the vLLM backend. For usage instructions for different backends, please refer to [Python Package Usage](#python-package-usage). We recommend using a **fresh, isolated environment** to avoid dependency conflicts with existing packages. You can create a clean Python 3.12 environment like this: conda create -n qwen3-asr python=3.12 -y conda activate qwen3-asr pip install -U qwen-asr pip install -U qwen-asr[vllm] If you want to develop or modify the code locally, install from source in editable mode: git clone https://github.com/QwenLM/Qwen3-ASR.git cd Qwen3-ASR pip install -e . # support vLLM backend # pip install -e ".[vllm]" Additionally, we recommend using FlashAttention 2 to reduce GPU memory usage and accelerate inference speed, especially for long inputs and large batch sizes. pip install -U flash-attn --no-build-isolation If your machine has less than 96GB of RAM and lots of CPU cores, run: MAX_JOBS=4 pip install -U flash-attn --no-build-isolation Also, you should have hardware that is compatible with FlashAttention 2. Read more about it in the official documentation of the [FlashAttention repository](https://github.com/Dao-AILab/flash-attention). FlashAttention 2 can only be used when a model is loaded in `torch.float16` or `torch.bfloat16`. ### Python Package Usage #### Quick Inference The `qwen-asr` package provides two backends: **transformers backend** and **vLLM backend**. You can pass audio inputs as a local path, a URL, base64 data, or a `(np.ndarray, sr)` tuple, and run batch inference. To quickly try Qwen3-ASR, you can use `Qwen3ASRModel.from_pretrained(...)` for the transformers backend with the following code: import torch from qwen_asr import Qwen3ASRModel model = Qwen3ASRModel.from_pretrained( "Qwen/Qwen3-ASR-1.7B", dtype=torch.bfloat16, device_map="cuda:0", # attn_implementation="flash_attention_2", max_inference_batch_size=32, # Batch size limit for inference. -1 means unlimited. Smaller values can help avoid OOM. max_new_tokens=256, # Maximum number of tokens to generate. Set a larger value for long audio input. ) results = model.transcribe( audio="https://qianwen-res.oss-cn-beijing.aliyuncs.com/Qwen3-ASR-Repo/asr_en.wav", language=None, # set "English" to force the language ) print(results[0].language) print(results[0].text) If you want to return timestamps, pass `forced_aligner` and its init kwargs. Here is an example of batch inference with timestamps output: import torch from qwen_asr import Qwen3ASRModel model = Qwen3ASRModel.from_pretrained( "Qwen/Qwen3-ASR-1.7B", dtype=torch.bfloat16, device_map="cuda:0", # attn_implementation="flash_attention_2", max_inference_batch_size=32, # Batch size limit for inference. -1 means unlimited. Smaller values can help avoid OOM. max_new_tokens=256, # Maximum number of tokens to generate. Set a larger value for long audio input. forced_aligner="Qwen/Qwen3-ForcedAligner-0.6B", forced_aligner_kwargs=dict( dtype=torch.bfloat16, device_map="cuda:0", # attn_implementation="flash_attention_2", ), ) results = model.transcribe( audio=[ "https://qianwen-res.oss-cn-beijing.aliyuncs.com/Qwen3-ASR-Repo/asr_zh.wav", "https://qianwen-res.oss-cn-beijing.aliyuncs.com/Qwen3-ASR-Repo/asr_en.wav", ], language=["Chinese", "English"], # can also be set to None for automatic language detection return_time_stamps=True, ) for r in results: print(r.language, r.text, r.time_stamps[0]) For more detailed usage examples, please refer to the [example code](https://github.com/QwenLM/Qwen3-ASR/blob/main/examples/example_qwen3_asr_transformers.py) for the transformers backend. #### vLLM Backend If you want the fastest inference speed with Qwen3-ASR, we strongly recommend using the vLLM backend by initializing the model with `Qwen3ASRModel.LLM(...)`. Example code is provided below. Note that you must install it via `pip install -U qwen-asr[vllm]`. If you want the model to output timestamps, it’s best to install FlashAttention via `pip install -U flash-attn --no-build-isolation` to speed up inference for the forced aligner model. Remember to wrap your code under `if __name__ == '__main__':` to avoid the `spawn` error described in [vLLM Troubleshooting](https://docs.vllm.ai/en/latest/usage/troubleshooting/#python-multiprocessing). import torch from qwen_asr import Qwen3ASRModel if __name__ == '__main__': model = Qwen3ASRModel.LLM( model="Qwen/Qwen3-ASR-1.7B", gpu_memory_utilization=0.7, max_inference_batch_size=128, # Batch size limit for inference. -1 means unlimited. Smaller values can help avoid OOM. max_new_tokens=4096, # Maximum number of tokens to generate. Set a larger value for long audio input. forced_aligner="Qwen/Qwen3-ForcedAligner-0.6B", forced_aligner_kwargs=dict( dtype=torch.bfloat16, device_map="cuda:0", # attn_implementation="flash_attention_2", ), ) results = model.transcribe( audio=[ "https://qianwen-res.oss-cn-beijing.aliyuncs.com/Qwen3-ASR-Repo/asr_zh.wav", "https://qianwen-res.oss-cn-beijing.aliyuncs.com/Qwen3-ASR-Repo/asr_en.wav", ], language=["Chinese", "English"], # can also be set to None for automatic language detection return_time_stamps=True, ) for r in results: print(r.language, r.text, r.time_stamps[0]) For more detailed usage examples, please refer to the [example code](https://github.com/QwenLM/Qwen3-ASR/blob/main/examples/example_qwen3_asr_vllm.py) for the vLLM backend. In addition, you can start a vLLM server via the `qwen-asr-serve` command, which is a wrapper around `vllm serve`. You can pass any arguments supported by `vllm serve`, for example: qwen-asr-serve Qwen/Qwen3-ASR-1.7B --gpu-memory-utilization 0.8 --host 0.0.0.0 --port 8000 And send requests to the server via: import requests url = "http://localhost:8000/v1/chat/completions" headers = {"Content-Type": "application/json"} data = { "messages": [ { "role": "user", "content": [ { "type": "audio_url", "audio_url": { "url": "https://qianwen-res.oss-cn-beijing.aliyuncs.com/Qwen3-ASR-Repo/asr_en.wav" }, } ], } ] } response = requests.post(url, headers=headers, json=data, timeout=300) response.raise_for_status() content = response.json()['choices'][0]['message']['content'] print(content) # parse ASR output if you want from qwen_asr import parse_asr_output language, text = parse_asr_output(content) print(language) print(text) #### Streaming Inference Qwen3-ASR fully supports streaming inference. Currently, streaming inference is only available with the vLLM backend. Note that streaming inference does not support batch inference or returning timestamps. Please refer to the [example code](https://github.com/QwenLM/Qwen3-ASR/blob/main/examples/example_qwen3_asr_vllm_streaming.py) for details. You can also launch a streaming web demo through the [guide](#streaming-demo) to experience Qwen3-ASR’s streaming transcription capabilities. #### ForcedAligner Usage `Qwen3-ForcedAligner-0.6B` can align text–speech pairs and return word or character level timestamps. Here is an example of using the forced aligner directly: import torch from qwen_asr import Qwen3ForcedAligner model = Qwen3ForcedAligner.from_pretrained( "Qwen/Qwen3-ForcedAligner-0.6B", dtype=torch.bfloat16, device_map="cuda:0", # attn_implementation="flash_attention_2", ) results = model.align( audio="https://qianwen-res.oss-cn-beijing.aliyuncs.com/Qwen3-ASR-Repo/asr_zh.wav", text="甚至出现交易几乎停滞的情况。", language="Chinese", ) print(results[0]) print(results[0][0].text, results[0][0].start_time, results[0][0].end_time) In addition, the forced aligner supports local paths / URLs / base64 data / `(np.ndarray, sr)` inputs and batch inference. Please refer to the [example code](https://github.com/QwenLM/Qwen3-ASR/blob/main/examples/example_qwen3_forced_aligner.py) for details. ### DashScope API Usage To further explore Qwen3-ASR, we encourage you to try our DashScope API for a faster and more efficient experience. For detailed API information and documentation, please refer to the following: | API Description | API Documentation (Mainland China) | API Documentation (International) | |------------------|-----------------------------------|------------------------------------| | Real-time API for Qwen3-ASR. | [https://help.aliyun.com/zh/model-studio/qwen-real-time-speech-recognition](https://help.aliyun.com/zh/model-studio/qwen-real-time-speech-recognition) | [https://www.alibabacloud.com/help/en/model-studio/qwen-real-time-speech-recognition](https://www.alibabacloud.com/help/en/model-studio/qwen-real-time-speech-recognition) | | FileTrans API for Qwen3-ASR. | [https://help.aliyun.com/zh/model-studio/qwen-speech-recognition](https://help.aliyun.com/zh/model-studio/qwen-speech-recognition) | [https://www.alibabacloud.com/help/en/model-studio/qwen-speech-recognition](https://www.alibabacloud.com/help/en/model-studio/qwen-speech-recognition) | ## Launch Local Web UI Demo ### Gradio Demo To launch the Qwen3-ASR web UI gradio demo, install the `qwen-asr` package and run `qwen-asr-demo`. Use the command below for help: qwen-asr-demo --help To launch the demo, you can use the following commands: # Transformers backend qwen-asr-demo \ --asr-checkpoint Qwen/Qwen3-ASR-1.7B \ --backend transformers \ --cuda-visible-devices 0 \ --ip 0.0.0.0 --port 8000 # Transformers backend + Forced Aligner (enable timestamps) qwen-asr-demo \ --asr-checkpoint Qwen/Qwen3-ASR-1.7B \ --aligner-checkpoint Qwen/Qwen3-ForcedAligner-0.6B \ --backend transformers \ --cuda-visible-devices 0 \ --backend-kwargs '{"device_map":"cuda:0","dtype":"bfloat16","max_inference_batch_size":8,"max_new_tokens":256}' \ --aligner-kwargs '{"device_map":"cuda:0","dtype":"bfloat16"}' \ --ip 0.0.0.0 --port 8000 # vLLM backend + Forced Aligner (enable timestamps) qwen-asr-demo \ --asr-checkpoint Qwen/Qwen3-ASR-1.7B \ --aligner-checkpoint Qwen/Qwen3-ForcedAligner-0.6B \ --backend vllm \ --cuda-visible-devices 0 \ --backend-kwargs '{"gpu_memory_utilization":0.7,"max_inference_batch_size":8,"max_new_tokens":2048}' \ --aligner-kwargs '{"device_map":"cuda:0","dtype":"bfloat16"}' \ --ip 0.0.0.0 --port 8000 Then open `http://:8000`, or access it via port forwarding in tools like VS Code. #### Backend Notes This demo supports two backends: transformers and vLLM. All backend-specific initialization parameters should be passed via `--backend-kwargs` as a JSON dict. If not provided, the demo will use sensible defaults. # Example: override transformers init args with flash attention --backend-kwargs '{"device_map":"cuda:0","dtype":"bfloat16","attn_implementation":"flash_attention_2"}' # Example: override vLLM init args with 65% GPU memory --backend-kwargs '{"gpu_memory_utilization":0.65}' #### CUDA Device Notes # Use GPU 0 --cuda-visible-devices 0 # Use GPU 1 --cuda-visible-devices 1 #### Timestamps Notes Timestamps are only available when `--aligner-checkpoint` is provided. If you launch the demo without a forced aligner, the timestamps UI will be hidden automatically. # No forced aligner qwen-asr-demo --asr-checkpoint Qwen/Qwen3-ASR-1.7B # With forced aligner qwen-asr-demo \ --asr-checkpoint Qwen/Qwen3-ASR-1.7B \ --aligner-checkpoint Qwen/Qwen3-ForcedAligner-0.6B #### HTTPS Notes To avoid browser microphone permission issues after deploying the server, it is recommended/required to run the gradio service over HTTPS (especially when accessed remotely or behind modern browsers/gateways). Use `--ssl-certfile` and `--ssl-keyfile` to enable HTTPS. First, generate a private key and a self-signed certificate (valid for 365 days): openssl req -x509 -newkey rsa:2048 \ -keyout key.pem -out cert.pem \ -days 365 -nodes \ -subj "/CN=localhost" Then run the demo with HTTPS: qwen-asr-demo \ --asr-checkpoint Qwen/Qwen3-ASR-1.7B \ --backend transformers \ --cuda-visible-devices 0 \ --ip 0.0.0.0 --port 8000 \ --ssl-certfile cert.pem \ --ssl-keyfile key.pem \ --no-ssl-verify Then open `https://:8000` to use it. If your browser shows a warning, that’s expected for self-signed certificates. For production, use a real certificate. ### Streaming Demo To experience Qwen3-ASR’s streaming transcription capability in a web UI, we provide a minimal Flask-based streaming demo. The demo captures microphone audio in the browser, resamples it to 16,000 Hz, and continuously pushes PCM chunks to the model. Run the demo with the following command: qwen-asr-demo-streaming \ --asr-model-path Qwen/Qwen3-ASR-1.7B \ --gpu-memory-utilization 0.9 \ --host 0.0.0.0 \ --port 8000 Then open `http://:8000`, or access it via port forwarding in tools like VS Code. ## Deployment with vLLM ### Installation You can run Qwen3-ASR with vLLM nightly wheel or docker image. To install the nightly version of vLLM, we recommend using `uv` as the environment manager uv venv source .venv/bin/activate uv pip install -U vllm --pre \ --extra-index-url https://wheels.vllm.ai/nightly/cu129 \ --extra-index-url https://download.pytorch.org/whl/cu129 \ --index-strategy unsafe-best-match uv pip install "vllm[audio]" # For additional audio dependencies ### Online Serving You can easily deploy Qwen3-ASR with vLLM by running the following command vllm serve Qwen/Qwen3-ASR-1.7B After the model server is successfully deployed, you can interact with it in multiple ways. #### Using OpenAI SDK import base64 import httpx from openai import OpenAI # Initialize client client = OpenAI( base_url="http://localhost:8000/v1", api_key="EMPTY" ) # Create multimodal chat completion request response = client.chat.completions.create( model="Qwen/Qwen3-ASR-1.7B", messages=[ { "role": "user", "content": [ { "type": "audio_url", "audio_url": { {"url": "https://qianwen-res.oss-cn-beijing.aliyuncs.com/Qwen3-ASR-Repo/asr_en.wav"} } } ] } ], ) print(response.choices[0].message.content) This model is also supported on vLLM with OpenAI transcription API. import httpx from openai import OpenAI # Initialize client client = OpenAI( base_url="http://localhost:8000/v1", api_key="EMPTY" ) audio_url = "https://qianwen-res.oss-cn-beijing.aliyuncs.com/Qwen3-ASR-Repo/asr_en.wav" audio_file = httpx.get(audio_url).content transcription = client.audio.transcriptions.create( model="Qwen/Qwen3-ASR-1.7B", file=audio_file, ) print(transcription.text) #### Using cURL curl http://localhost:8000/v1/chat/completions \ -H "Content-Type: application/json" \ -d '{ "messages": [ {"role": "user", "content": [ {"type": "audio_url", "audio_url": {"url": "https://qianwen-res.oss-cn-beijing.aliyuncs.com/Qwen3-ASR-Repo/asr_en.wav"}} ]} ] }' ### Offline Inference See the following example on using vLLM to run offline inference with Qwen3-ASR from vllm import LLM, SamplingParams from vllm.assets.audio import AudioAsset import base64 import requests # Initialize the LLM llm = LLM( model="Qwen/Qwen3-ASR-1.7B" ) # Load audio audio_asset = AudioAsset("winning_call") # Create conversation with audio content conversation = [ { "role": "user", "content": [ { "type": "audio_url", "audio_url": {"url": audio_asset.url} } ] } ] sampling_params = SamplingParams(temperature=0.01, max_tokens=256) # Run inference using .chat() outputs = llm.chat(conversation, sampling_params=sampling_params) print(outputs[0].outputs[0].text) ## Fine Tuning Please refer to [Qwen3-ASR-Finetuning](finetuning/) for detailed instructions on fine-tuning Qwen3-ASR. ## Docker First, pull the image and start a container: LOCAL_WORKDIR=/path/to/your/workspace HOST_PORT=8000 CONTAINER_PORT=80 docker run --gpus all --name qwen3-asr \ -v /var/run/docker.sock:/var/run/docker.sock -p $HOST_PORT:$CONTAINER_PORT \ --mount type=bind,source=$LOCAL_WORKDIR,target=/data/shared/Qwen3-ASR \ --shm-size=4gb \ -it qwenllm/qwen3-asr:latest After running the command, you will enter the container’s bash shell. Your local workspace (**replace** `/path/to/your/workspace` **with the actual path**) will be mounted inside the container at `/data/shared/Qwen3-ASR`. Port `8000` on the host is mapped to port `80` in the container, so you can access services running in the container via `http://:8000`. Note that services inside the container must bind to `0.0.0.0` (not `127.0.0.1`) for port forwarding to work. If you exit the container, you can start it again and re-enter it with: docker start qwen3-asr docker exec -it qwen3-asr bash To remove the container completely, run: docker rm -f qwen3-asr ## Evaluation During evaluation, we ran inference for all models with `dtype=torch.bfloat16` and set `max_new_tokens=1024` using vLLM. Greedy search was used for all decoding, and none of the tests specified a language parameter. The detailed evaluation results are shown below.

ASR Benchmarks on Public Datasets (WER ↓)
GPT-4o
-Transcribe
Gemini-2.5
-Pro
Doubao-ASR Whisper
-large-v3
Fun-ASR
-MLT-Nano
Qwen3-ASR
-0.6B
Qwen3-ASR
-1.7B
English (en)
Librispeech
clean | other
1.39 | 3.75 2.89 | 3.56 2.78 | 5.70 1.51 | 3.97 1.68 | 4.03 2.11 | 4.55 1.63 | 3.38
GigaSpeech 25.50 9.37 9.55 9.76 - 8.88 8.45
CV-en 9.08 14.49 13.78 9.90 9.90 9.92 7.39
Fleurs-en 2.40 2.94 6.31 4.08 5.49 4.39 3.35
MLS-en 5.12 3.68 7.09 4.87 - 6.00 4.58
Tedlium 7.69 6.15 4.91 6.84 - 3.85 4.50
VoxPopuli 10.29 11.36 12.12 12.05 - 9.96 9.15
Chinese (zh)
WenetSpeech
net | meeting
15.30 | 32.27 14.43 | 13.47 N/A 9.86 | 19.11 6.35 | - 5.97 | 6.88 4.97 | 5.88
AISHELL-2-test 4.24 11.62 2.85 5.06 - 3.15 2.71
SpeechIO 12.86 5.30 2.93 7.56 - 3.44 2.88
Fleurs-zh 2.44 2.71 2.69 4.09 3.51 2.88 2.41
CV-zh 6.32 7.70 5.95 12.91 6.20 6.89 5.35
Chinese Dialect
KeSpeech 26.87 24.71 5.27 28.79 - 7.08 5.10
Fleurs-yue 4.98 9.43 4.98 9.18 - 5.79 3.98
CV-yue 11.36 18.76 13.20 16.23 - 9.50 7.57
CV-zh-tw 6.32 7.31 4.06 7.84 - 5.59 3.77
WenetSpeech-Yue
short | long
15.62 | 25.29 25.19 | 11.23 9.74 | 11.40 32.26 | 46.64 - | - 7.54 | 9.92 5.82 | 8.85
WenetSpeech-Chuan
easy | hard
34.81 | 53.98 43.79 | 67.30 11.40 | 20.20 14.35 | 26.80 - | - 13.92 | 24.45 11.99 | 21.63
ASR Benchmarks on Internal Datasets (WER ↓)
GPT-4o
-Transcribe
Gemini-2.5
-Pro
Doubao-ASR Whisper
-large-v3
Fun-ASR
-MLT-Nano
Qwen3-ASR
-0.6B
Qwen3-ASR
-1.7B
Accented English
Dialog-Accented English 28.56 23.85 20.41 21.30 19.96 16.62 16.07
Chinese Mandarin
Elders&Kids 14.27 36.93 4.17 10.61 4.54 4.48 3.81
ExtremeNoise 36.11 29.06 17.04 63.17 36.55 17.88 16.17
TongueTwister 20.87 4.97 3.47 16.63 9.02 4.06 2.44
Dialog-Mandarin 20.73 12.50 6.61 14.01 7.32 7.06 6.54
Chinese Dialect
Dialog-Cantonese 16.05 14.98 7.56 31.04 5.85 4.80 4.12
Dialog-Chinese Dialects 45.37 47.70 19.85 44.55 19.41 18.24 15.94

Dialect coverage: Results for Dialog-Accented English are averaged over 16 accents, and results for Dialog-Chinese Dialects are averaged over 22 Chinese dialects.

Multilingual ASR Benchmarks (WER ↓)
GLM-ASR
-Nano-2512
Whisper
-large-v3
Fun-ASR
-MLT-Nano
Qwen3-ASR
-0.6B
Qwen3-ASR
-1.7B
Open-sourced Benchmarks
MLS 13.32 8.62 28.70 13.19 8.55
CommonVoice 19.40 10.77 17.25 12.75 9.18
MLC-SLM 34.93 15.68 29.94 15.84 12.74
Fleurs 16.08 5.27 10.03 7.57 4.90
Fleurs 20.05 6.85 31.89 10.37 6.62
Fleurs†† 24.83 8.16 47.84 21.80 12.60
Qwen-ASR Internal Benchmarks
News-Multilingual 49.40 14.80 65.07 17.39 12.80

Language coverage: MLS includes 8 languages: {da, de, en, es, fr, it, pl, pt}.
CommonVoice includes 13 languages: {en, zh, yue, zh_TW, ar, de, es, fr, it, ja, ko, pt, ru}.
MLC-SLM includes 11 languages: {en, fr, de, it, pt, es, ja, ko, ru, th, vi}.
Fleurs includes 12 languages: {en, zh, yue, ar, de, es, fr, it, ja, ko, pt, ru }.
Fleurs includes 8 additional languages beyond Fleurs: {hi, id, ms, nl, pl, th, tr, vi}.
Fleurs†† includes 10 additional languages beyond Fleurs: {cs, da, el, fa, fi, fil, hu, mk, ro, sv}.
News-Multilingual includes 15 languages: {ar, de, es, fr, hi, id, it, ja, ko, nl, pl, pt, ru, th, vi}.

Language Identification Accuracy (%) ↑
Whisper-large-v3 Qwen3-ASR-0.6B Qwen3-ASR-1.7B
MLS 99.9 99.3 99.9
CommonVoice 92.7 98.2 98.7
MLC-SLM 89.2 92.7 94.1
Fleurs 94.6 97.1 98.7
Avg. 94.1 96.8 97.9

Language coverage: The language sets follow Multilingual ASR Benchmarks. Here, Fleurs corresponds to Fleurs†† in Multilingual ASR Benchmarks and covers 30 languages.

Singing Voice & Song Transcription (WER ↓)
GPT-4o
-Transcribe
Gemini-2.5
-Pro
Doubao-ASR
-1.0
Whisper
-large-v3
Fun-ASR-MLT
-Nano
Qwen3-ASR
-1.7B
Singing
M4Singer 16.77 20.88 7.88 13.58 7.29 5.98
MIR-1k-vocal 11.87 9.85 6.56 11.71 8.17 6.25
Opencpop 7.93 6.49 3.80 9.52 2.98 3.08
Popcs 32.84 15.13 8.97 13.77 9.42 8.52
Songs with BGM
EntireSongs-en 30.71 12.18 33.51 N/A N/A 14.60
EntireSongs-zh 34.86 18.68 23.99 N/A N/A 13.91
ASR Inference Mode Performance (WER ↓)
Model Infer. Mode Librispeech Fleurs-en Fleurs-zh Avg.
Qwen3-ASR-1.7B Offline 1.63 | 3.38 3.35 2.41 2.69
Streaming 1.95 | 4.51 4.02 2.84 3.33
Qwen3-ASR-0.6B Offline 2.11 | 4.55 4.39 2.88 3.48
Streaming 2.54 | 6.27 5.38 3.40 4.40
Forced Alignment Benchmarks (AAS ms ↓)
Monotonic-Aligner NFA WhisperX Qwen3-ForcedAligner-0.6B
MFA-Labeled Raw
Chinese 161.1 109.8 - 33.1
English - 107.5 92.1 37.5
French - 100.7 145.3 41.7
German - 122.7 165.1 46.5
Italian - 142.7 155.5 75.5
Japanese - - - 42.2
Korean - - - 37.2
Portuguese - - - 38.4
Russian - 200.7 - 40.2
Spanish - 124.7 108.0 36.8
Avg. 161.1 129.8 133.2 42.9
MFA-Labeled Concat-300s
Chinese 1742.4 235.0 - 36.5
English - 226.7 227.2 58.6
French - 230.6 2052.2 53.4
German - 220.3 993.4 62.4
Italian - 290.5 5719.4 81.6
Japanese - - - 81.3
Korean - - - 42.2
Portuguese - - - 50.0
Russian - 283.3 - 43.0
Spanish - 240.2 4549.9 39.6
Cross-lingual - - - 34.2
Avg. 1742.4 246.7 2708.4 52.9
Human-Labeled
Raw 49.9 88.6 - 27.8
Raw-Noisy 53.3 89.5 - 41.8
Concat-60s 51.1 86.7 - 25.3
Concat-300s 410.8 140.0 - 24.8
Concat-Cross-lingual - - - 42.5
Avg. 141.3 101.2 - 32.4
## Citation @article{Qwen3-ASR, title={Qwen3-ASR Technical Report}, author={Xian Shi, Xiong Wang, Zhifang Guo, Yongqi Wang, Pei Zhang, Xinyu Zhang, Zishan Guo, Hongkun Hao, Yu Xi, Baosong Yang, Jin Xu, Jingren Zhou, Junyang Lin}, journal={arXiv preprint arXiv:2601.21337}, year={2026} }
标签:AI模型, ASR, 多语言识别, 时间戳预测, 深度学习, 语音识别