Skip to content

[ICLR 2025] SOTA discrete acoustic codec models with 40/75 tokens per second for audio language modeling

License

Notifications You must be signed in to change notification settings

jishengpeng/WavTokenizer

Repository files navigation

WavTokenizer

SOTA Discrete Codec Models With Forty Tokens Per Second for Audio Language Modeling

arXiv demo model

πŸŽ‰πŸŽ‰ with WavTokenizer, you can represent speech, music, and audio with only 40 tokens per second!

πŸŽ‰πŸŽ‰ with WavTokenizer, You can get strong reconstruction results.

πŸŽ‰πŸŽ‰ WavTokenizer owns rich semantic information and is build for audio language models such as GPT-4o.

πŸ”₯ News

  • 2025.02.25: We update WavTokenizer camera ready version for ICLR 2025 and update WavTokenizer-large-v2 checkpoint on huggingface.
  • 2024.10.22: We update WavTokenizer on arxiv and release WavTokenizer-Large checkpoint.
  • 2024.09.09: We release WavTokenizer-medium checkpoint on huggingface.
  • 2024.08.31: We release WavTokenizer on arxiv.

result

Installation

To use WavTokenizer, install it using:

conda create -n wavtokenizer python=3.9
conda activate wavtokenizer
pip install -r requirements.txt

Infer

Part1: Reconstruct audio from raw wav

from encoder.utils import convert_audio
import torchaudio
import torch
from decoder.pretrained import WavTokenizer


device=torch.device('cpu')

config_path = "./configs/xxx.yaml"
model_path = "./xxx.ckpt"
audio_outpath = "xxx"

wavtokenizer = WavTokenizer.from_pretrained0802(config_path, model_path)
wavtokenizer = wavtokenizer.to(device)


wav, sr = torchaudio.load(audio_path)
wav = convert_audio(wav, sr, 24000, 1) 
bandwidth_id = torch.tensor([0])
wav=wav.to(device)
features,discrete_code= wavtokenizer.encode_infer(wav, bandwidth_id=bandwidth_id)
audio_out = wavtokenizer.decode(features, bandwidth_id=bandwidth_id) 
torchaudio.save(audio_outpath, audio_out, sample_rate=24000, encoding='PCM_S', bits_per_sample=16)

Part2: Generating discrete codecs

from encoder.utils import convert_audio
import torchaudio
import torch
from decoder.pretrained import WavTokenizer

device=torch.device('cpu')

config_path = "./configs/xxx.yaml"
model_path = "./xxx.ckpt"

wavtokenizer = WavTokenizer.from_pretrained0802(config_path, model_path)
wavtokenizer = wavtokenizer.to(device)

wav, sr = torchaudio.load(audio_path)
wav = convert_audio(wav, sr, 24000, 1) 
bandwidth_id = torch.tensor([0])
wav=wav.to(device)
_,discrete_code= wavtokenizer.encode_infer(wav, bandwidth_id=bandwidth_id)
print(discrete_code)

Part3: Audio reconstruction through codecs

# audio_tokens [n_q,1,t]/[n_q,t]
features = wavtokenizer.codes_to_features(audio_tokens)
bandwidth_id = torch.tensor([0])  
audio_out = wavtokenizer.decode(features, bandwidth_id=bandwidth_id)

Available models

πŸ€— links to the Huggingface model hub.

Model name HuggingFace Corpus Token/s Domain Open-Source
WavTokenizer-small-600-24k-4096 πŸ€— LibriTTS 40 Speech √
WavTokenizer-small-320-24k-4096 πŸ€— LibriTTS 75 Speech √
WavTokenizer-medium-320-24k-4096 πŸ€— 10000 Hours 75 Speech, Audio, Music √
WavTokenizer-large-600-24k-4096 πŸ€— 80000 Hours 40 Speech, Audio, Music √
WavTokenizer-large-320-24k-4096 πŸ€— 80000 Hours 75 Speech, Audio, Music √

Training

Step1: Prepare train dataset

# Process the data into a form similar to ./data/demo.txt

Step2: Modifying configuration files

# ./configs/xxx.yaml
# Modify the values of parameters such as batch_size, filelist_path, save_dir, device

Step3: Start training process

Refer to Pytorch Lightning documentation for details about customizing the training pipeline.

cd ./WavTokenizer
python train.py fit --config ./configs/xxx.yaml

Citation

If this code contributes to your research, please cite our work, Language-Codec and WavTokenizer:

@article{ji2024wavtokenizer,
  title={Wavtokenizer: an efficient acoustic discrete codec tokenizer for audio language modeling},
  author={Ji, Shengpeng and Jiang, Ziyue and Wang, Wen and Chen, Yifu and Fang, Minghui and Zuo, Jialong and Yang, Qian and Cheng, Xize and Wang, Zehan and Li, Ruiqi and others},
  journal={arXiv preprint arXiv:2408.16532},
  year={2024}
}

@article{ji2024language,
  title={Language-codec: Reducing the gaps between discrete codec representation and speech language models},
  author={Ji, Shengpeng and Fang, Minghui and Jiang, Ziyue and Huang, Rongjie and Zuo, Jialung and Wang, Shulei and Zhao, Zhou},
  journal={arXiv preprint arXiv:2402.12208},
  year={2024}
}

Releases

No releases published

Packages

No packages published

Languages