# Real-Time Voice Cloning v2

### What is this?
It is an improved version of [Real-Time-Voice-Cloning](https://github.com/CorentinJ/Real-Time-Voice-Cloning).

## Installation
1. Install [ffmpeg](https://ffmpeg.org/download.html#get-packages). This is necessary for reading audio files.

2. Create a new conda environment with 
```
conda create -n rtvc python=3.7.13
```
3. Install [PyTorch](https://download.pytorch.org/whl/torch_stable.html).  Pick the proposed CUDA version if you have a GPU, otherwise pick CPU.
My torch version: `torch=1.9.1+cu111`
`torchvision=0.10.1+cu111`

4. Install the remaining requirements with 
```
pip install -r requirements.txt
```

5. Install spaCy model en_core_web_sm by 
`python -m spacy download en_core_web_sm`


## Training

### Encoder 

**Download dataset：** 

1. [LibriSpeech](https://www.openslr.org/12): train-other-500 for training, dev-other for validation
(extract as <datasets_root>/LibriSpeech/<dataset_name>)

2. [VoxCeleb1](https://mm.kaist.ac.kr/datasets/voxceleb/): Dev A - D for training, Test for validation as well as the metadata file `vox1_meta.csv` (extract as <datasets_root>/VoxCeleb1/ and <datasets_root>/VoxCeleb1/vox1_meta.csv)

3. [VoxCeleb2](https://mm.kaist.ac.kr/datasets/voxceleb/): Dev A - H for training, Test for validation
(extract as <datasets_root>/VoxCeleb2/)

**Encoder preprocessing：** 
```
python encoder_preprocess.py <datasets_root>
```

**Encoder training：** 

it is recommended to start visdom server for monitor training with
```
visdom
```
then start training with
```
python encoder_train.py <model_id> <datasets_root>/SV2TTS/encoder
```
### Synthesizer

**Download dataset：** 
1. [LibriSpeech](https://www.openslr.org/12): train-clean-100 and train-clean-360 for training, dev-clean for validation (extract as <datasets_root>/LibriSpeech/<dataset_name>)
2. [LibriSpeech alignments](https://drive.google.com/file/d/1WYfgr31T-PPwMcxuAq09XZfHQO5Mw8fE/view?usp=sharing): merge the directory structure with the LibriSpeech datasets you have downloaded (do not take the alignments from the datasets you haven't downloaded else the scripts will think you have them)
3. [VCTK](https://datashare.ed.ac.uk/handle/10283/3443): used for training and validation

**Synthesizer preprocessing:** 
```
python synthesizer_preprocess_audio.py <datasets_root>
python synthesizer_preprocess_embeds.py <datasets_root>/SV2TTS/synthesizer
```

**Synthesizer training:** 
```
python synthesizer_train.py <model_id> <datasets_root>/SV2TTS/synthesizer --use_tb
```
if you want to monitor the training progress, run
```
tensorboard --logdir log/synthesizer --host localhost --port 8088
```
### Vocoder

**Download dataset：** 

The same as synthesizer. You can skip this if you already download synthesizer training dataset.

**Vocoder preprocessing:** 
```
python vocoder_preprocess.py <datasets_root>
```

**Vocoder training:** 
```
python vocoder_train.py <model_id> <datasets_root> --use_tb
```
if you want to monitor the training progress, run
```
tensorboard --logdir log/vocoder --host localhost --port 8080
```
**Note:**

Training breakpoints are saved periodically, so you can run the training command and resume training when the breakpoint exists.

## Inference 

**Terminal:** 
```
python demo_cli.py
```
First input the number of audios, then input the audio file paths, then input the text message. The attention alignments and mel spectrogram are stored in syn_results/. The generated audio is stored in out_audios/.
## Dimension reduction visualization
**Download dataset：** 

[LibriSpeech](https://www.openslr.org/12): test-other
(extract as <datasets_root>/LibriSpeech/<dataset_name>)

**Preprocessing:** 
```
python encoder_test_preprocess.py <datasets_root>
```

**Visualization**
```
python encoder_test_visualization.py <model_id> <datasets_root>
```
The results are saved in dim_reduction_results/.

## Pretrained models
You can download the pretrained model from [this](https://drive.google.com/drive/folders/1oi5the9QxNbpOol_I5Qpr42hvoFXdhEF) and extract as saved_models/default
## What's new
**2022.05.19：** We calculated GE2E loss in encoder with CUDA rather than originally-configured CPU. It speeds up the encoder training speed.<br>
**2022.07.15：** We added Loss animation plot for synthesizer and vocoder.<br>
**2022.07.19：** We added response time and Griffin-Lim vocoder results for demo_toolbox.<br>
**2022.07.29：** We added model validation for encoder, synthesizer and vocoder.<br>
**2022.08.02：** We added voxceleb train and dev data for encoder. We added noise reduce method for the output wav from vocoder.
[noisereduce reference](https://github.com/timsainb/noisereduce)<br>
**2022.08.06：** We split the long text into short sentences using spacy for input of synthesizer. Make sure to install spaCy model en_core_web_sm by 
`python -m spacy download en_core_web_sm`<br>
**2022.09.02：** We set prop_decrease=0.6 for male and 0.9 for female in noisereduce function.(输出滤波，男女声使用不同的滤波参数)<br>
**2022.09.26：** We added speed adjustment(声音变速) for output audios using praat, install parselmouth using pip: `pip install praat-parselmouth`<br>
**2022.10.10：** We added voice filter functioning(声音美颜) for input audios, the weight ratio of the input audio embed and the standard audio embed is 7: 3. <br>
**2022.10.25：** We set small values(<0.06) to zeros in embed.(对嵌入向量较小值置零)<br>
**2022.10.26：** The split frequency for input audio is 170Hz. The split frequency for output noise reduce is 165Hz.<br>
**2022.12.01：** merge the single sentences to input.<br>
**2022.12.31：** added speaker embeddings dimension reduction visualzation results.<br>
**2023.01.01：** did more text preprocessing and text cleaning for TTS text input.<br>
**2023.02.27：** preprocessed ascii chars and abbreviations.<br>