Singing Voice Conversion based on Whisper & neural source-filter BigVGAN
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
Dataset preparation
Necessary pre-processing:
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
intospeaker_pretrain/
-
3 download whisper model multiple language medium model, Make sure to download
medium.pt
,put it intowhisper_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/
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 name config path model path speaker wave input wave ppg wave pitch pitch 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]