lora-svc

lora-svc

开源AI歌声转换系统,结合Whisper和BigVGAN的先进技术

lora-svc是一个开源的歌声转换系统,集成了OpenAI的Whisper、Nvidia的BigVGAN和Microsoft的Adapter技术。该项目利用多语言语音识别、反混叠语音生成和高效微调等技术,实现高质量的声音转换。lora-svc提供完整的数据处理、模型训练和推理流程,支持自定义训练和灵活推理,适合研究声音转换技术的开发者和研究人员使用。

Singing Voice ConversionWhisperBigVGANLoRA人工智能Github开源项目
<div align="center"> <h1> Singing Voice Conversion based on Whisper & neural source-filter BigVGAN </h1> <img alt="GitHub Repo stars" src="https://img.shields.io/github/stars/PlayVoice/lora-svc"> <img alt="GitHub forks" src="https://img.shields.io/github/forks/PlayVoice/lora-svc"> <img alt="GitHub issues" src="https://img.shields.io/github/issues/PlayVoice/lora-svc"> <img alt="GitHub" src="https://img.shields.io/github/license/PlayVoice/lora-svc"> </div>
Black technology based on the three giants of artificial intelligence:

OpenAI's whisper, 680,000 hours in multiple languages

Nvidia's bigvgan, anti-aliasing for speech generation

Microsoft's adapter, high-efficiency for fine-tuning

LoRA is not fully implemented in this project, but it can be found here: LoRA TTS & paper

use pretrain model to fine tune

https://user-images.githubusercontent.com/16432329/231021007-6e34cbb4-e256-491d-8ab6-5ce4e822da21.mp4

Dataset preparation

Necessary pre-processing:

  • 1 accompaniment separation, UVR
  • 2 cut audio, less than 30 seconds for whisper, slicer

then put the dataset into the data_raw directory according to the following file structure

data_raw ├───speaker0 │ ├───000001.wav │ ├───... │ └───000xxx.wav └───speaker1 ├───000001.wav ├───... └───000xxx.wav

Install dependencies

  • 1 software dependency

    pip install -r requirements.txt

  • 2 download the Timbre Encoder: Speaker-Encoder by @mueller91, put best_model.pth.tar into speaker_pretrain/

  • 3 download whisper model multiple language medium model, Make sure to download medium.pt,put it into whisper_pretrain/

    Tip: whisper is built-in, do not install it additionally, it will conflict and report an error

  • 4 download pretrain model maxgan_pretrain_32K.pth, and do test

    python svc_inference.py --config configs/maxgan.yaml --model maxgan_pretrain_32K.pth --spk ./configs/singers/singer0001.npy --wave test.wav

Data preprocessing

use this command if you want to automate this:

python3 prepare/easyprocess.py

or step by step, as follows:

  • 1, re-sampling

    generate audio with a sampling rate of 16000Hz

    python prepare/preprocess_a.py -w ./data_raw -o ./data_svc/waves-16k -s 16000

    generate audio with a sampling rate of 32000Hz

    python prepare/preprocess_a.py -w ./data_raw -o ./data_svc/waves-32k -s 32000

  • 2, use 16K audio to extract pitch

    python prepare/preprocess_f0.py -w data_svc/waves-16k/ -p data_svc/pitch

  • 3, use 16K audio to extract ppg

    python prepare/preprocess_ppg.py -w data_svc/waves-16k/ -p data_svc/whisper

  • 4, use 16k audio to extract timbre code

    python prepare/preprocess_speaker.py data_svc/waves-16k/ data_svc/speaker

  • 5, extract the singer code for inference

    python prepare/preprocess_speaker_ave.py data_svc/speaker/ data_svc/singer

  • 6, use 32k audio to generate training index

    python prepare/preprocess_train.py

  • 7, training file debugging

    python prepare/preprocess_zzz.py -c configs/maxgan.yaml

data_svc/ └── waves-16k │ └── speaker0 │ │ ├── 000001.wav │ │ └── 000xxx.wav │ └── speaker1 │ ├── 000001.wav │ └── 000xxx.wav └── waves-32k │ └── speaker0 │ │ ├── 000001.wav │ │ └── 000xxx.wav │ └── speaker1 │ ├── 000001.wav │ └── 000xxx.wav └── pitch │ └── speaker0 │ │ ├── 000001.pit.npy │ │ └── 000xxx.pit.npy │ └── speaker1 │ ├── 000001.pit.npy │ └── 000xxx.pit.npy └── whisper │ └── speaker0 │ │ ├── 000001.ppg.npy │ │ └── 000xxx.ppg.npy │ └── speaker1 │ ├── 000001.ppg.npy │ └── 000xxx.ppg.npy └── speaker │ └── speaker0 │ │ ├── 000001.spk.npy │ │ └── 000xxx.spk.npy │ └── speaker1 │ ├── 000001.spk.npy │ └── 000xxx.spk.npy └── singer ├── speaker0.spk.npy └── speaker1.spk.npy

Train

  • 0, if fine-tuning based on the pre-trained model, you need to download the pre-trained model: maxgan_pretrain_32K.pth

    set pretrain: "./maxgan_pretrain_32K.pth" in configs/maxgan.yaml,and adjust the learning rate appropriately, eg 1e-5

  • 1, start training

    python svc_trainer.py -c configs/maxgan.yaml -n svc

  • 2, resume training

    python svc_trainer.py -c configs/maxgan.yaml -n svc -p chkpt/svc/***.pth

  • 3, view log

    tensorboard --logdir logs/

final_model_loss

Inference

use this command if you want a GUI that does all the commands below:

python3 svc_gui.py

or step by step, as follows:

  • 1, export inference model

    python svc_export.py --config configs/maxgan.yaml --checkpoint_path chkpt/svc/***.pt

  • 2, use whisper to extract content encoding, without using one-click reasoning, in order to reduce GPU memory usage

    python whisper/inference.py -w test.wav -p test.ppg.npy

  • 3, extract the F0 parameter to the csv text format

    python pitch/inference.py -w test.wav -p test.csv

  • 4, specify parameters and infer

    python svc_inference.py --config configs/maxgan.yaml --model maxgan_g.pth --spk ./data_svc/singers/your_singer.npy --wave test.wav --ppg test.ppg.npy --pit test.csv

    when --ppg is specified, when the same audio is reasoned multiple times, it can avoid repeated extraction of audio content codes; if it is not specified, it will be automatically extracted;

    when --pit is specified, the manually tuned F0 parameter can be loaded; if not specified, it will be automatically extracted;

    generate files in the current directory:svc_out.wav

    args--config--model--spk--wave--ppg--pit--shift
    nameconfig pathmodel pathspeakerwave inputwave ppgwave pitchpitch shift
  • 5, post by vad

    python svc_inference_post.py --ref test.wav --svc svc_out.wav --out svc_post.wav

Source of code and References

Adapter-Based Extension of Multi-Speaker Text-to-Speech Model for New Speakers

AdaSpeech: Adaptive Text to Speech for Custom Voice

https://github.com/nii-yamagishilab/project-NN-Pytorch-scripts/tree/master/project/01-nsf

https://github.com/mindslab-ai/univnet [paper]

https://github.com/openai/whisper/ [paper]

https://github.com/NVIDIA/BigVGAN [paper]

编辑推荐精选

问小白

问小白

全能AI智能助手,随时解答生活与工作的多样问题

问小白,由元石科技研发的AI智能助手,快速准确地解答各种生活和工作问题,包括但不限于搜索、规划和社交互动,帮助用户在日常生活中提高效率,轻松管理个人事务。

热门AI助手AI对话AI工具聊天机器人
Transly

Transly

实时语音翻译/同声传译工具

Transly是一个多场景的AI大语言模型驱动的同声传译、专业翻译助手,它拥有超精准的音频识别翻译能力,几乎零延迟的使用体验和支持多国语言可以让你带它走遍全球,无论你是留学生、商务人士、韩剧美剧爱好者,还是出国游玩、多国会议、跨国追星等等,都可以满足你所有需要同传的场景需求,线上线下通用,扫除语言障碍,让全世界的语言交流不再有国界。

讯飞智文

讯飞智文

一键生成PPT和Word,让学习生活更轻松

讯飞智文是一个利用 AI 技术的项目,能够帮助用户生成 PPT 以及各类文档。无论是商业领域的市场分析报告、年度目标制定,还是学生群体的职业生涯规划、实习避坑指南,亦或是活动策划、旅游攻略等内容,它都能提供支持,帮助用户精准表达,轻松呈现各种信息。

AI办公办公工具AI工具讯飞智文AI在线生成PPTAI撰写助手多语种文档生成AI自动配图热门
讯飞星火

讯飞星火

深度推理能力全新升级,全面对标OpenAI o1

科大讯飞的星火大模型,支持语言理解、知识问答和文本创作等多功能,适用于多种文件和业务场景,提升办公和日常生活的效率。讯飞星火是一个提供丰富智能服务的平台,涵盖科技资讯、图像创作、写作辅助、编程解答、科研文献解读等功能,能为不同需求的用户提供便捷高效的帮助,助力用户轻松获取信息、解决问题,满足多样化使用场景。

热门AI开发模型训练AI工具讯飞星火大模型智能问答内容创作多语种支持智慧生活
Spark-TTS

Spark-TTS

一种基于大语言模型的高效单流解耦语音令牌文本到语音合成模型

Spark-TTS 是一个基于 PyTorch 的开源文本到语音合成项目,由多个知名机构联合参与。该项目提供了高效的 LLM(大语言模型)驱动的语音合成方案,支持语音克隆和语音创建功能,可通过命令行界面(CLI)和 Web UI 两种方式使用。用户可以根据需求调整语音的性别、音高、速度等参数,生成高质量的语音。该项目适用于多种场景,如有声读物制作、智能语音助手开发等。

Trae

Trae

字节跳动发布的AI编程神器IDE

Trae是一种自适应的集成开发环境(IDE),通过自动化和多元协作改变开发流程。利用Trae,团队能够更快速、精确地编写和部署代码,从而提高编程效率和项目交付速度。Trae具备上下文感知和代码自动完成功能,是提升开发效率的理想工具。

AI工具TraeAI IDE协作生产力转型热门
咔片PPT

咔片PPT

AI助力,做PPT更简单!

咔片是一款轻量化在线演示设计工具,借助 AI 技术,实现从内容生成到智能设计的一站式 PPT 制作服务。支持多种文档格式导入生成 PPT,提供海量模板、智能美化、素材替换等功能,适用于销售、教师、学生等各类人群,能高效制作出高品质 PPT,满足不同场景演示需求。

讯飞绘文

讯飞绘文

选题、配图、成文,一站式创作,让内容运营更高效

讯飞绘文,一个AI集成平台,支持写作、选题、配图、排版和发布。高效生成适用于各类媒体的定制内容,加速品牌传播,提升内容营销效果。

热门AI辅助写作AI工具讯飞绘文内容运营AI创作个性化文章多平台分发AI助手
材料星

材料星

专业的AI公文写作平台,公文写作神器

AI 材料星,专业的 AI 公文写作辅助平台,为体制内工作人员提供高效的公文写作解决方案。拥有海量公文文库、9 大核心 AI 功能,支持 30 + 文稿类型生成,助力快速完成领导讲话、工作总结、述职报告等材料,提升办公效率,是体制打工人的得力写作神器。

openai-agents-python

openai-agents-python

OpenAI Agents SDK,助力开发者便捷使用 OpenAI 相关功能。

openai-agents-python 是 OpenAI 推出的一款强大 Python SDK,它为开发者提供了与 OpenAI 模型交互的高效工具,支持工具调用、结果处理、追踪等功能,涵盖多种应用场景,如研究助手、财务研究等,能显著提升开发效率,让开发者更轻松地利用 OpenAI 的技术优势。

下拉加载更多