File size: 7,885 Bytes
626f70a ad63082 626f70a ad63082 626f70a |
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 |
# F5-TTS: A Fairytaler that Fakes Fluent and Faithful Speech with Flow Matching
[![python](https://img.shields.io/badge/Python-3.10-brightgreen)](https://github.com/SWivid/F5-TTS)
[![arXiv](https://img.shields.io/badge/arXiv-2410.06885-b31b1b.svg?logo=arXiv)](https://arxiv.org/abs/2410.06885)
[![demo](https://img.shields.io/badge/GitHub-Demo%20page-blue.svg)](https://swivid.github.io/F5-TTS/)
[![space](https://img.shields.io/badge/🤗-Space%20demo-yellow)](https://huggingface.co/spaces/mrfakename/E2-F5-TTS)
**F5-TTS**: Diffusion Transformer with ConvNeXt V2, faster trained and inference.
**E2 TTS**: Flat-UNet Transformer, closest reproduction.
**Sway Sampling**: Inference-time flow step sampling strategy, greatly improves performance
## Installation
Clone the repository:
```bash
git clone https://github.com/SWivid/F5-TTS.git
cd F5-TTS
```
Install torch with your CUDA version, e.g. :
```bash
pip install torch==2.3.0+cu118 --extra-index-url https://download.pytorch.org/whl/cu118
pip install torchaudio==2.3.0+cu118 --extra-index-url https://download.pytorch.org/whl/cu118
```
Install other packages:
```bash
pip install -r requirements.txt
```
## Prepare Dataset
Example data processing scripts for Emilia and Wenetspeech4TTS, and you may tailor your own one along with a Dataset class in `model/dataset.py`.
```bash
# prepare custom dataset up to your need
# download corresponding dataset first, and fill in the path in scripts
# Prepare the Emilia dataset
python scripts/prepare_emilia.py
# Prepare the Wenetspeech4TTS dataset
python scripts/prepare_wenetspeech4tts.py
```
## Training
Once your datasets are prepared, you can start the training process.
```bash
# setup accelerate config, e.g. use multi-gpu ddp, fp16
# will be to: ~/.cache/huggingface/accelerate/default_config.yaml
accelerate config
accelerate launch train.py
```
An initial guidance on Finetuning [#57](https://github.com/SWivid/F5-TTS/discussions/57).
## Inference
The pretrained model checkpoints can be reached at [🤗 Hugging Face](https://huggingface.co/SWivid/F5-TTS) and [⭐ Model Scope](https://www.modelscope.cn/models/SWivid/F5-TTS_Emilia-ZH-EN), or automatically downloaded with `inference-cli` and `gradio_app`.
Currently support 30s for a single generation, which is the **TOTAL** length of prompt audio and the generated. Batch inference with chunks is supported by `inference-cli` and `gradio_app`.
- To avoid possible inference failures, make sure you have seen through the following instructions.
- A longer prompt audio allows shorter generated output. The part longer than 30s cannot be generated properly. Consider using a prompt audio <15s.
- Uppercased letters will be uttered letter by letter, so use lowercased letters for normal words.
- Add some spaces (blank: " ") or punctuations (e.g. "," ".") to explicitly introduce some pauses. If first few words skipped in code-switched generation (cuz different speed with different languages), this might help.
### CLI Inference
Either you can specify everything in `inference-cli.toml` or override with flags. Leave `--ref_text ""` will have ASR model transcribe the reference audio automatically (use extra GPU memory). If encounter network error, consider use local ckpt, just set `ckpt_path` in `inference-cli.py`
```bash
python inference-cli.py \
--model "F5-TTS" \
--ref_audio "tests/ref_audio/test_en_1_ref_short.wav" \
--ref_text "Some call me nature, others call me mother nature." \
--gen_text "I don't really care what you call me. I've been a silent spectator, watching species evolve, empires rise and fall. But always remember, I am mighty and enduring. Respect me and I'll nurture you; ignore me and you shall face the consequences."
python inference-cli.py \
--model "E2-TTS" \
--ref_audio "tests/ref_audio/test_zh_1_ref_short.wav" \
--ref_text "对,这就是我,万人敬仰的太乙真人。" \
--gen_text "突然,身边一阵笑声。我看着他们,意气风发地挺直了胸膛,甩了甩那稍显肉感的双臂,轻笑道,我身上的肉,是为了掩饰我爆棚的魅力,否则,岂不吓坏了你们呢?"
```
### Gradio App
Currently supported features:
- Chunk inference
- Podcast Generation
- Multiple Speech-Type Generation
You can launch a Gradio app (web interface) to launch a GUI for inference (will load ckpt from Huggingface, you may set `ckpt_path` to local file in `gradio_app.py`). Currently load ASR model, F5-TTS and E2 TTS all in once, thus use more GPU memory than `inference-cli`.
```bash
python gradio_app.py
```
You can specify the port/host:
```bash
python gradio_app.py --port 7860 --host 0.0.0.0
```
Or launch a share link:
```bash
python gradio_app.py --share
```
### Speech Editing
To test speech editing capabilities, use the following command.
```bash
python speech_edit.py
```
## Evaluation
### Prepare Test Datasets
1. Seed-TTS test set: Download from [seed-tts-eval](https://github.com/BytedanceSpeech/seed-tts-eval).
2. LibriSpeech test-clean: Download from [OpenSLR](http://www.openslr.org/12/).
3. Unzip the downloaded datasets and place them in the data/ directory.
4. Update the path for the test-clean data in `scripts/eval_infer_batch.py`
5. Our filtered LibriSpeech-PC 4-10s subset is already under data/ in this repo
### Batch Inference for Test Set
To run batch inference for evaluations, execute the following commands:
```bash
# batch inference for evaluations
accelerate config # if not set before
bash scripts/eval_infer_batch.sh
```
### Download Evaluation Model Checkpoints
1. Chinese ASR Model: [Paraformer-zh](https://huggingface.co/funasr/paraformer-zh)
2. English ASR Model: [Faster-Whisper](https://huggingface.co/Systran/faster-whisper-large-v3)
3. WavLM Model: Download from [Google Drive](https://drive.google.com/file/d/1-aE1NfzpRCLxA4GUxX9ITI3F9LlbtEGP/view).
### Objective Evaluation
Install packages for evaluation:
```bash
pip install -r requirements_eval.txt
```
**Some Notes**
For faster-whisper with CUDA 11:
```bash
pip install --force-reinstall ctranslate2==3.24.0
```
(Recommended) To avoid possible ASR failures, such as abnormal repetitions in output:
```bash
pip install faster-whisper==0.10.1
```
Update the path with your batch-inferenced results, and carry out WER / SIM evaluations:
```bash
# Evaluation for Seed-TTS test set
python scripts/eval_seedtts_testset.py
# Evaluation for LibriSpeech-PC test-clean (cross-sentence)
python scripts/eval_librispeech_test_clean.py
```
## Acknowledgements
- [E2-TTS](https://arxiv.org/abs/2406.18009) brilliant work, simple and effective
- [Emilia](https://arxiv.org/abs/2407.05361), [WenetSpeech4TTS](https://arxiv.org/abs/2406.05763) valuable datasets
- [lucidrains](https://github.com/lucidrains) initial CFM structure with also [bfs18](https://github.com/bfs18) for discussion
- [SD3](https://arxiv.org/abs/2403.03206) & [Hugging Face diffusers](https://github.com/huggingface/diffusers) DiT and MMDiT code structure
- [torchdiffeq](https://github.com/rtqichen/torchdiffeq) as ODE solver, [Vocos](https://huggingface.co/charactr/vocos-mel-24khz) as vocoder
- [mrfakename](https://x.com/realmrfakename) huggingface space demo ~
- [FunASR](https://github.com/modelscope/FunASR), [faster-whisper](https://github.com/SYSTRAN/faster-whisper), [UniSpeech](https://github.com/microsoft/UniSpeech) for evaluation tools
- [ctc-forced-aligner](https://github.com/MahmoudAshraf97/ctc-forced-aligner) for speech edit test
## Citation
```
@article{chen-etal-2024-f5tts,
title={F5-TTS: A Fairytaler that Fakes Fluent and Faithful Speech with Flow Matching},
author={Yushen Chen and Zhikang Niu and Ziyang Ma and Keqi Deng and Chunhui Wang and Jian Zhao and Kai Yu and Xie Chen},
journal={arXiv preprint arXiv:2410.06885},
year={2024},
}
```
## License
Our code is released under MIT License. |