Official model for
Spark-TTS: An Efficient LLM-Based Text-to-Speech Model with Single-Stream Decoupled Speech Tokens
Spark-TTS is an advanced text-to-speech system that uses the power of large language models (LLM) for highly accurate and natural-sounding voice synthesis. It is designed to be efficient, flexible, and powerful for both research and production use.
Inference Overview of Voice Cloning![]() |
Inference Overview of Controlled Generation![]() |
Clone and Install
git clone https://github.com/SparkAudio/Spark-TTS.git
cd Spark-TTS
conda create -n sparktts -y python=3.12
conda activate sparktts
pip install -r requirements.txt
# If you are in mainland China, you can set the mirror as follows:
pip install -r requirements.txt -i https://mirrors.aliyun.com/pypi/simple/ --trusted-host=mirrors.aliyun.com
Model Download
Download via python:
from huggingface_hub import snapshot_download
snapshot_download("SparkAudio/Spark-TTS-0.5B", local_dir="pretrained_models/Spark-TTS-0.5B")
Download via git clone:
mkdir -p pretrained_models
# Make sure you have git-lfs installed (https://git-lfs.com)
git lfs install
git clone https://huggingface.co/SparkAudio/Spark-TTS-0.5B pretrained_models/Spark-TTS-0.5B
Basic Usage
You can simply run the demo with the following commands:
cd example
bash infer.sh
Alternatively, you can directly execute the following command in the command line to perform inference:
python -m cli.inference \
--text "text to synthesis." \
--device 0 \
--save_dir "path/to/save/audio" \
--model_dir pretrained_models/Spark-TTS-0.5B \
--prompt_text "transcript of the prompt audio" \
--prompt_speech_path "path/to/prompt_audio"
UI Usage
You can start the UI interface by running python webui.py, which allows you to perform Voice Cloning and Voice Creation. Voice Cloning supports uploading reference audio or directly recording the audio.
| Voice Cloning | Voice Creation |
|---|---|
![]() | ![]() |
@misc{wang2025sparktts, title={Spark-TTS: An Efficient LLM-Based Text-to-Speech Model with Single-Stream Decoupled Speech Tokens}, author={Xinsheng Wang and Mingqi Jiang and Ziyang Ma and Ziyu Zhang and Songxiang Liu and Linqin Li and Zheng Liang and Qixi Zheng and Rui Wang and Xiaoqin Feng and Weizhen Bian and Zhen Ye and Sitong Cheng and Ruibin Yuan and Zhixian Zhao and Xinfa Zhu and Jiahao Pan and Liumeng Xue and Pengcheng Zhu and Yunlin Chen and Zhifei Li and Xie Chen and Lei Xie and Yike Guo and Wei Xue}, year={2025}, eprint={2503.01710}, archivePrefix={arXiv}, primaryClass={cs.SD}, url={https://arxiv.org/abs/2503.01710}, }
The model's license has been updated from Apache 2.0 to CC BY-NC-SA due to the licensing terms of some training data.
Key Changes:
The model can only be used for non-commercial purposes.
Any modifications or derivatives must also be released under CC BY-NC-SA 4.0.
Proper attribution is required when using or modifying the model.
Please ensure compliance with the new license terms.
This project provides a zero-shot voice cloning TTS model intended for academic research, educational purposes, and legitimate applications, such as personalized speech synthesis, assistive technologies, and linguistic research.
Please note:
Do not use this model for unauthorized voice cloning, impersonation, fraud, scams, deepfakes, or any illegal activities.
Ensure compliance with local laws and regulations when using this model and uphold ethical standards.
The developers assume no liability for any misuse of this model.
We advocate for the responsible development and use of AI and encourage the community to uphold safety and ethical principles in AI research and applications. If you have any concerns regarding ethics or misuse, please contact us.