Qwen3-TTS 是 AI Skill Hub 本期精选AI工具之一。综合评分 8.0 分,整体质量较高。我们强烈推荐将其纳入你的 AI 工具库,帮助提升工作效率。
Qwen3-TTS 是一款基于 Python 开发的开源工具,专注于 ai、audio-generation、deep-learning 等核心功能。作为 GitHub 开源项目,它拥有活跃的社区支持和持续的版本迭代,代码完全透明可审计,支持本地部署以保护数据隐私。无论是个人使用还是集成到企业工作流,都能提供稳定可靠的解决方案。
Qwen3-TTS 是一款基于 Python 开发的开源工具,专注于 ai、audio-generation、deep-learning 等核心功能。作为 GitHub 开源项目,它拥有活跃的社区支持和持续的版本迭代,代码完全透明可审计,支持本地部署以保护数据隐私。无论是个人使用还是集成到企业工作流,都能提供稳定可靠的解决方案。
# 方式一:pip 安装(推荐)
pip install qwen3-tts-easyfinetuning
# 方式二:虚拟环境安装(推荐生产环境)
python -m venv .venv
source .venv/bin/activate # Windows: .venv\Scripts\activate
pip install qwen3-tts-easyfinetuning
# 方式三:从源码安装(获取最新功能)
git clone https://github.com/mozi1924/Qwen3-TTS-EasyFinetuning
cd Qwen3-TTS-EasyFinetuning
pip install -e .
# 验证安装
python -c "import qwen3_tts_easyfinetuning; print('安装成功')"
# 命令行使用
qwen3-tts-easyfinetuning --help
# 基本用法
qwen3-tts-easyfinetuning input_file -o output_file
# Python 代码中调用
import qwen3_tts_easyfinetuning
# 示例
result = qwen3_tts_easyfinetuning.process("input")
print(result)
# qwen3-tts-easyfinetuning 配置文件示例(config.yml) app: name: "qwen3-tts-easyfinetuning" debug: false log_level: "INFO" # 运行时指定配置文件 qwen3-tts-easyfinetuning --config config.yml # 或通过环境变量配置 export QWEN3_TTS_EASYFINETUNING_API_KEY="your-key" export QWEN3_TTS_EASYFINETUNING_OUTPUT_DIR="./output"
<p align="center"> <img src="https://img.shields.io/github/stars/mozi1924/Qwen3-TTS-EasyFinetuning?style=for-the-badge&color=ffd700" alt="GitHub Stars"> <img src="https://img.shields.io/github/license/mozi1924/Qwen3-TTS-EasyFinetuning?style=for-the-badge&color=blue" alt="License"> <img src="https://img.shields.io/badge/Python-3.11+-blue?style=for-the-badge&logo=python&logoColor=white" alt="Python 3.11+"> <img src="https://img.shields.io/badge/PyTorch-2.0+-ee4c2c?style=for-the-badge&logo=pytorch&logoColor=white" alt="PyTorch"> </p>
<p align="center"> <b>English</b> | <a href="./README_zh.md">简体中文</a> </p>
An easy-to-use workspace for fine-tuning the Qwen3-TTS model. This repository streamlines the entire process—from raw audio ingestion to creating high-stability, expressive custom voice models.
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Using Docker (Recommended) ```bash
docker compose up -d --build
**Using Python Virtual Environment**bash
python -m venv venv source venv/bin/activate pip install -r requirements.txt
pip install flash-attn==2.8.3 --no-build-isolation ```
For a comprehensive step-by-step guide with illustrations, please refer to my article:
👉 English Article | 中文文章
Information about the environment used for developing and testing this project: - OS: Ubuntu 24.04.4 LTS (Kernel 6.17.0-14-generic) - CPU: 2 x Intel(R) Xeon(R) Platinum 8259CL (KVM, 32 cores) - Memory: 32 GB - GPU: 2 x NVIDIA GeForce RTX 3080 (10 GB VRAM each) - Driver & CUDA: NVIDIA Driver 590.48.01 / CUDA 13.1 - Python: 3.11.14
To ensure stable training and avoid Out-of-Memory (OOM) errors, we recommend: - GPU: NVIDIA GPU with >= 16 GB VRAM (24 GB recommended for 1.7B model) - Memory: >= 32 GB RAM - Storage: SSD with at least 50 GB free space - OS: Linux (Ubuntu 20.04+ recommended) - Software: CUDA 12.4+ (v12.8+ recommended), Python 3.10+
⚠️ Special Note for Windows Users (GPU Training): Due to architectural limitations, do not use Rancher Desktop if you require GPU support on Windows, as it lacks native Nvidia GPU capabilities. Instead, choose one of the following: 1. Run on a native Linux GPU host (Highly Recommended: best performance and stability). 2. Use pure WSL2 (Ubuntu) (Recommended: install native Docker Engine or Python in WSL2 for seamless GPU access). 3. Use Docker Desktop (Supports GPU, but comes with significant performance overhead and is not guaranteed to be perfectly stable/usable on all Windows configurations).
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The src/cli.py serves as a unified entry point for all operations:
Step A: Prepare Data Place your raw .wav files in a directory (e.g., raw-dataset/my_speaker/).
python src/cli.py prepare --input_dir raw-dataset/my_speaker --speaker_name my_speaker
Step B: Start Training
python src/cli.py train --experiment_name exp1 --speaker_name my_speaker --epochs 3
For CustomVoice fine-tuning, generate the speaker embedding once after ASR and before training:
python src/cli.py embed --speaker_name my_speaker --init_model Qwen/Qwen3-TTS-12Hz-0.6B-CustomVoice
python src/cli.py train --experiment_name exp1 --speaker_name my_speaker --init_model Qwen/Qwen3-TTS-12Hz-0.6B-CustomVoice --epochs 3
Step C: Run Inference
python src/cli.py infer --checkpoint output/exp1/checkpoint-epoch-2 --speaker my_speaker --text "Hello world! This is my custom voice."
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高质量的语音合成和克隆工具
AI Skill Hub 为第三方内容聚合平台,本页面信息基于公开数据整理,不对工具功能和质量作任何法律背书。
建议在沙箱或测试环境中充分验证后,再部署至生产环境,并做好必要的安全评估。
✅ Apache 2.0 — 宽松开源协议,可商用,需保留版权声明和 NOTICE 文件,含专利授权条款。
经综合评估,Qwen3-TTS 在AI工具赛道中表现稳健,质量优秀。如果你已有明确的使用需求,可以直接上手体验;如果还在评估阶段,建议对比同类工具后再做决策。
| 原始名称 | Qwen3-TTS-EasyFinetuning |
| 原始描述 | 开源AI工具:Easy fine-tuning for Qwen3-TTS: Fast voice cloning and high-quality multilingual。⭐87 · Python |
| Topics | aiaudio-generationdeep-learningfine-tuning |
| GitHub | https://github.com/mozi1924/Qwen3-TTS-EasyFinetuning |
| License | Apache-2.0 |
| 语言 | Python |
收录时间:2026-05-27 · 更新时间:2026-05-27 · License:Apache-2.0 · AI Skill Hub 不对第三方内容的准确性作法律背书。