(This repository is in the experimental stage. The content may change without notice.)

Other languages

• 日本語

The structure of the decoder is designed with reference to FastSVC, StreamVC, Hifi-GAN, etc. Low latency is achieved by using a "causal" convolution layer that does not refer to future information.

• Realtime conversion
• Low latency (approximately 0.2 seconds, subject to change based on the environment and optimizations)
• Stable phase and pitch (based on the source-filter model)
• Speaker style conversion using k-nearest neighbors (kNN) method

• Python 3.10 or later
• PyTorch 2.0 or later with GPU environment
• When training from scratch, prepare a large amount of human speech data (e.g., LJ Speech, JVS Corpus)

1. clone this repository.

git clone https://github.com/uthree/fastersvc.git

2. install requirements

pip3 install -r requirements.txt

The model pretrained with the JVS corpus is published here.

Train a model for basic voice conversion. At this stage, the model is not specialized for a specific speaker, but having a model that can perform basic voice synthesis allows for easy adaptation to a specific speaker with minimal adjustments.

Here are the steps:

1. Preprocess.

python3 preprocess.py <Dataset directory>

2. Train pitch estimator. Distill pitch estimation using a fast and parallelizable 1D CNN with the harvest algorithm from WORLD.

python3 train_pe.py

3. Train content encoder Distill HuBERT-base.

python3 train_ce.py

4. Train decoder The goal of the decoder is to reconstruct the original waveform from pitch and content.

sh

python3 train_dec.py

By adjusting the pre-trained model to a model specialized for conversion to a specific speaker, it is possible to create a more accurate model. This process takes much less time than pre-learning.

1. Combine only the audio files of a specific speaker into one folder, and preprocess.

python3 preorpcess.py <Dataset directory>

2. Fine tune the decoder.

python3 train_dec.py

3. Create a dictionary for vector search. This eliminates the need to encode audio files each time.

python3 extract_index.py -o <Dictionary output destination (optional)>

4. When inferring, you can load arbitrary dictionary data by adding the -idx <dictionary file> option.

• add -fp16 True to accelerate training with float16 if you have RTX series GPU.
• add -b <number> to set batch size. default is 16.
• add -e <number> to set epoch. default is 60.
• add -d <device name> to set training device, default is cuda.

1. Create an directory inputs
2. Put audio files in inputs
3. Run inference script

python3 infer.py -t <target audio file>

• You can set the transparency of the original audio information with -a <number from 0.0 to 1.0>.
• You can normalize the volume with --normalize True.
• You can change the calculation device with =d <device name>. Although it may not make much sense since it is originally high speed.
• Pitch shift can be performed with -p <scale>. Useful for voice conversion between men and women.

1. Confirm the ID of the audio device

python3 audio_device_list.py

2. Run inference

python3 infer_streaming.py -i <input device id> -o <output device id> -l <loopback device id> -t <target audio file>

(The loopback option is optional.)

• FastSVC
• kNN-VC
• WavLM (Fig. 2)
• StreamVC
• Hifi-GAN

This document is translated from Japanese using ChatGPT.