蚂蚁 inclusionAI:GitHub 新仓库· inclusionAI·· 2025-09-29精选AI 评分57
inclusionAI发布MingTok-Audio:首个统一连续语音分词器
inclusionAI/MingTok-Audio
AI 导读
inclusionAI团队推出了MingTok-Audio,这是首个能有效融合语义与声学特征的统一连续语音分词器,适用于语音理解与生成任务。该模型基于纯因果Transformer架构,去除了卷积层以提升效率,并采用VAE进行连续特征建模以实现高质量音频重建。在语音重建性能上,其帧率为50,在SEED-ZH和SEED-EN测试集上的PESQ分别达到4.21和4.04,SIM为0.96,STOI为0.98,显著优于对比模型。在下游ASR任务中,其在多个方言数据集上取得了更低的错误率,例如在Hunan Minnan数据集上WER低至9.80%。
推荐理由
蚂蚁把语音 tokenizer 做到了 PESQ 4.2 的离谱分数,比第二名翻了快一倍,做语音理解和生成的团队值得拿这个当新 baseline 跑一下。
正文 · 原文
📝Technical Report 📖Project Page |🤗 Hugging Face| 🤖 ModelScope
Architecture
Key Features
- 🚀 First Unified Continuous Speech Tokenizer: the first continuous audio tokenizer to effectively integrate semantic and acoustic features, suitable for both understanding and generation tasks.
- 🎧 High-Quality Reconstruction: Achieve high-quality audio generation by modeling continuous features with a VAE, minimizing information loss and preserving intricate acoustic textures.
- 🌐 Convolution-Free Efficiency: Built on a pure causal transformer architecture, completely eliminating convolutional layers for superior efficiency and a simpler design.
Installation
pip install -r requirements.txt
Quick start
import torch
import torchaudio
from audio_tokenizer.modeling_audio_vae import AudioVAE
model = AudioVAE.from_pretrained('inclusionAI/MingTok-Audio')
model = model.cuda()
model.eval()
waveform, sr = torchaudio.load('data/1089-134686-0000.flac', backend='soundfile')
sample = {'waveform': waveform.cuda(), 'waveform_length': torch.tensor([waveform.size(-1)]).cuda()}
with torch.no_grad():
with torch.autocast(device_type='cuda', dtype=torch.bfloat16):
latent, frame_num = model.encode_latent(**sample)
output_waveform = model.decode(latent)
torchaudio.save('./1089-134686-0000_reconstruct.wav', output_waveform.cpu()[0], sample_rate=16000)
Performance
Speech reconstruction performance
| System | FrameRate | SEED-ZH | SEED-EN | ||||
|---|---|---|---|---|---|---|---|
| PESQ↑ | SIM↑ | STOI↑ | PESQ↑ | SIM↑ | STOI↑ | ||
| MiMo-Audio-Tokenizer | 25 | 2.71 | 0.89 | 0.93 | 2.43 | 0.85 | 0.92 |
| GLM4-Voice-Tokenizer | 12.5 | 1.06 | 0.33 | 0.61 | 1.05 | 0.12 | 0.60 |
| Baichuan-Audio-Tokenizer | 12.5 | 1.84 | 0.78 | 0.86 | 1.62 | 0.69 | 0.85 |
| XY-Tokenizer | 12.5 | 2.27 | 0.77 | 0.90 | 2.14 | 0.82 | 0.90 |
| Mimi | 75 | 2.05 | 0.73 | 0.89 | 2.01 | 0.77 | 0.89 |
| XCodec2.0 | 50 | 2.19 | 0.80 | 0.92 | 2.37 | 0.82 | 0.93 |
| BigCodec | 80 | 2.26 | 0.81 | 0.92 | 2.22 | 0.80 | 0.91 |
| MingTok-Audio(ours) | 50 | 4.21 | 0.96 | 0.98 | 4.04 | 0.96 | 0.98 |
The adaptation performance for downstream ASR tasks
| Datasets | Model | Performance | ||||||
|---|---|---|---|---|---|---|---|---|
| aishell2-ios | LS-clean | Hunan | Minnan | Guangyue | Chuanyu | Shanghai | ||
| Understanding ASR | Kimi-Audio | 2.56 | 1.28 | 31.93 | 80.28 | 41.49 | 6.69 | 60.64 |
| Qwen2.5 Omni | 2.75 | 1.80 | 29.31 | 53.43 | 10.39 | 7.61 | 32.05 | |
| Qwen2 Audio | 2.92 | 1.60 | 25.88 | 123.78 | 7.59 | 7.77 | 31.73 | |
| Ming-UniAudio-16B-A3B(ours) | 2.84 | 1.62 | 9.80 | 16.50 | 5.51 | 5.46 | 14.65 | |
The adaptation performance for downstream TTS tasks
| Datasets | Model | Performance | |||
|---|---|---|---|---|---|
| Seed-zh WER(%) | Seed-zh SIM | Seed-en WER(%) | Seed-en SIM | ||
| Generation | Seed-TTS | 1.12 | 0.80 | 2.25 | 0.76 |
| MiMo-Audio | 1.96 | - | 5.37 | - | |
| Qwen3-Omni-30B-A3B-Instruct | 1.07 | - | 1.39 | - | |
| Ming-Omni-Lite | 1.69 | 0.68 | 4.31 | 0.51 | |
| Ming-UniAudio-16B-A3B(ours) | 0.95 | 0.70 | 1.85 | 0.58 | |
Acknowledgements
- We borrowed a lot of code from X-Codec-2.0 for tokenizer training.
- We thank the OpenAI team for developing the Whisper model and making its weights publicly available.
License and Legal Disclaimer
This code repository is licensed under the MIT License, and the Legal Disclaimer is located in the LEGAL.md file under the project's root directory.
Citation
If you find our work helpful, feel free to give us a cite.
来源:蚂蚁 inclusionAI:GitHub 新仓库 · github.com