F5-TTS: Diffusion Transformer with ConvNeXt V2, faster trained and inference.
E2 TTS: Flat-UNet Transformer, closest reproduction from paper.
Sway Sampling: Inference-time flow step sampling strategy, greatly improves performance
Clone the repository:
git clone https://github.com/SWivid/F5-TTS.git
cd F5-TTS
Install torch with your CUDA version, e.g. :
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:
pip install -r requirements.txt
Example data processing scripts for Emilia and Wenetspeech4TTS, and you may tailor your own one along with a Dataset class in model/dataset.py
.
# 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
Once your datasets are prepared, you can start the training process.
# 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.
The pretrained model checkpoints can be reached at 🤗 Hugging Face and 🤖 Model Scope, 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.
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
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?"
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
.
python gradio_app.py
You can specify the port/host:
python gradio_app.py --port 7860 --host 0.0.0.0
Or launch a share link:
python gradio_app.py --share
To test speech editing capabilities, use the following command.
python speech_edit.py
- Seed-TTS test set: Download from seed-tts-eval.
- LibriSpeech test-clean: Download from OpenSLR.
- Unzip the downloaded datasets and place them in the data/ directory.
- Update the path for the test-clean data in
scripts/eval_infer_batch.py
- Our filtered LibriSpeech-PC 4-10s subset is already under data/ in this repo
To run batch inference for evaluations, execute the following commands:
# batch inference for evaluations
accelerate config # if not set before
bash scripts/eval_infer_batch.sh
- Chinese ASR Model: Paraformer-zh
- English ASR Model: Faster-Whisper
- WavLM Model: Download from Google Drive.
Install packages for evaluation:
pip install -r requirements_eval.txt
Some Notes
For faster-whisper with CUDA 11:
pip install --force-reinstall ctranslate2==3.24.0
(Recommended) To avoid possible ASR failures, such as abnormal repetitions in output:
pip install faster-whisper==0.10.1
Update the path with your batch-inferenced results, and carry out WER / SIM evaluations:
# 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
- E2-TTS brilliant work, simple and effective
- Emilia, WenetSpeech4TTS valuable datasets
- lucidrains initial CFM structure with also bfs18 for discussion
- SD3 & Hugging Face diffusers DiT and MMDiT code structure
- torchdiffeq as ODE solver, Vocos as vocoder
- mrfakename huggingface space demo ~
- FunASR, faster-whisper, UniSpeech for evaluation tools
- ctc-forced-aligner for speech edit test
@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},
}
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.