inclusionAI/MingTok-Audio
Python
Captured source
source ↗inclusionAI/MingTok-Audio
Language: Python
License: MIT
Stars: 88
Forks: 9
Open issues: 4
Created: 2025-09-29T03:19:13Z
Pushed: 2026-02-24T04:10:19Z
Default branch: main
Fork: no
Archived: no
README:
📝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
Speech reconstruction performance comparison on various audio benchmark datasets. The best results are in bold.
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
Understanding ASR performance comparison on various audio benchmark datasets. The best results are in bold.
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
Performance comparison on various audio benchmark datasets. The best results are in bold.
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
1. We borrowed a lot of code from X-Codec-2.0 for tokenizer training. 2. 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](./LICENSE), and the Legal Disclaimer is located in the [LEGAL.md file](./LEGAL.md) under the project's root directory.
Citation
If you find our work helpful, feel free to give us a cite.
Notability
notability 6.0/10New audio repo with moderate community interest