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# 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-orange.svg)](https://swivid.github.io/F5-TTS/) | |
[![hfspace](https://img.shields.io/badge/🤗-Space%20demo-yellow)](https://huggingface.co/spaces/mrfakename/E2-F5-TTS) | |
[![msspace](https://img.shields.io/badge/🤖-Space%20demo-blue)](https://modelscope.cn/studios/modelscope/E2-F5-TTS) | |
[![lab](https://img.shields.io/badge/X--LANCE-Lab-grey?labelColor=lightgrey)](https://x-lance.sjtu.edu.cn/) | |
<img src="https://github.com/user-attachments/assets/12d7749c-071a-427c-81bf-b87b91def670" alt="Watermark" style="width: 40px; height: auto"> | |
**F5-TTS**: Diffusion Transformer with ConvNeXt V2, faster trained and inference. | |
**E2 TTS**: Flat-UNet Transformer, closest reproduction from [paper](https://arxiv.org/abs/2406.18009). | |
**Sway Sampling**: Inference-time flow step sampling strategy, greatly improves performance | |
### Thanks to all the contributors ! | |
## 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 | |
``` | |
**[Optional]**: We provide [Dockerfile](https://github.com/SWivid/F5-TTS/blob/main/Dockerfile) and you can use the following command to build it. | |
```bash | |
docker build -t f5tts:v1 . | |
``` | |
## 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 & Finetuning | |
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). | |
Gradio UI finetuning with `finetune_gradio.py` see [#143](https://github.com/SWivid/F5-TTS/discussions/143). | |
## 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` | |
for change model use --ckpt_file to specify the model you want to load, | |
for change vocab.txt use --vocab_file to provide your vocab.txt file. | |
```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?" | |
# Multi voice | |
python inference-cli.py -c samples/story.toml | |
``` | |
### 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 | |
- [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 | |
- [mrfakename](https://x.com/realmrfakename) huggingface space demo ~ | |
- [f5-tts-mlx](https://github.com/lucasnewman/f5-tts-mlx/tree/main) Implementation of F5-TTS, with the MLX framework. | |
## Citation | |
If our work and codebase is useful for you, please cite as: | |
``` | |
@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. The pre-trained models are licensed under the CC-BY-NC license due to the training data Emilia, which is an in-the-wild dataset. Sorry for any inconvenience this may cause. | |