mirror of
https://github.com/SWivid/F5-TTS.git
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* group files in f5_tts directory * add setup.py * use global imports * simplify demo * add install directions for library mode * fix old huggingface_hub version constraint * move finetune to package * change imports to f5_tts.model * bump version * fix bad merge * Update inference-cli.py * fix HF space * reformat * fix utils.py vocab.txt import * fix format * adapt README for f5_tts package structure * simplify app.py * add gradio.Dockerfile and workflow * refactored for pyproject.toml * refactored for pyproject.toml * added in reference to packaged files * use fork for testing docker image * added in reference to packaged files * minor tweaks * fixed inference-cli.toml path * fixed inference-cli.toml path * fixed inference-cli.toml path * fixed inference-cli.toml path * refactor eval_infer_batch.py * fix typo * added eval_infer_batch to scripts --------- Co-authored-by: Roberts Slisans <rsxdalv@gmail.com> Co-authored-by: Adam Kessel <adam@rosi-kessel.org> Co-authored-by: Roberts Slisans <roberts.slisans@gmail.com>
306 lines
11 KiB
Markdown
306 lines
11 KiB
Markdown
# F5-TTS: A Fairytaler that Fakes Fluent and Faithful Speech with Flow Matching
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[](https://github.com/SWivid/F5-TTS)
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[](https://arxiv.org/abs/2410.06885)
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[](https://swivid.github.io/F5-TTS/)
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[](https://huggingface.co/spaces/mrfakename/E2-F5-TTS)
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[](https://modelscope.cn/studios/modelscope/E2-F5-TTS)
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[](https://x-lance.sjtu.edu.cn/)
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<img src="https://github.com/user-attachments/assets/12d7749c-071a-427c-81bf-b87b91def670" alt="Watermark" style="width: 40px; height: auto">
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**F5-TTS**: Diffusion Transformer with ConvNeXt V2, faster trained and inference.
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**E2 TTS**: Flat-UNet Transformer, closest reproduction from [paper](https://arxiv.org/abs/2406.18009).
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**Sway Sampling**: Inference-time flow step sampling strategy, greatly improves performance
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### Thanks to all the contributors !
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## Installation
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Clone the repository:
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```bash
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git clone https://github.com/SWivid/F5-TTS.git
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cd F5-TTS
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```
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Install torch with your CUDA version, e.g. :
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```bash
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pip install torch==2.3.0+cu118 --extra-index-url https://download.pytorch.org/whl/cu118
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pip install torchaudio==2.3.0+cu118 --extra-index-url https://download.pytorch.org/whl/cu118
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```
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Install other packages:
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```bash
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pip install -r requirements.txt
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```
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**[Optional]**: We provide [Dockerfile](https://github.com/SWivid/F5-TTS/blob/main/Dockerfile) and you can use the following command to build it.
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```bash
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docker build -t f5tts:v1 .
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```
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### Development
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When making a pull request, please use pre-commit to ensure code quality:
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```bash
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pip install pre-commit
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pre-commit install
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```
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This will run linters and formatters automatically before each commit.
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Manually run using:
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```bash
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pre-commit run --all-files
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```
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Note: Some model components have linting exceptions for E722 to accommodate tensor notation
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### As a pip package
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```bash
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pip install git+https://github.com/SWivid/F5-TTS.git
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```
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```python
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import gradio as gr
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from f5_tts.gradio_app import app
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with gr.Blocks() as main_app:
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gr.Markdown("# This is an example of using F5-TTS within a bigger Gradio app")
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# ... other Gradio components
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app.render()
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main_app.launch()
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```
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## Prepare Dataset
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Example data processing scripts for Emilia and Wenetspeech4TTS, and you may tailor your own one along with a Dataset class in `f5_tts/model/dataset.py`.
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```bash
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# switch to the main directory
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cd f5_tts
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# prepare custom dataset up to your need
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# download corresponding dataset first, and fill in the path in scripts
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# Prepare the Emilia dataset
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python scripts/prepare_emilia.py
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# Prepare the Wenetspeech4TTS dataset
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python scripts/prepare_wenetspeech4tts.py
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```
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## Training & Finetuning
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Once your datasets are prepared, you can start the training process.
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```bash
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# switch to the main directory
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cd f5_tts
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# setup accelerate config, e.g. use multi-gpu ddp, fp16
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# will be to: ~/.cache/huggingface/accelerate/default_config.yaml
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accelerate config
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accelerate launch train.py
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```
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An initial guidance on Finetuning [#57](https://github.com/SWivid/F5-TTS/discussions/57).
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Gradio UI finetuning with `f5_tts/finetune_gradio.py` see [#143](https://github.com/SWivid/F5-TTS/discussions/143).
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### Wandb Logging
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By default, the training script does NOT use logging (assuming you didn't manually log in using `wandb login`).
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To turn on wandb logging, you can either:
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1. Manually login with `wandb login`: Learn more [here](https://docs.wandb.ai/ref/cli/wandb-login)
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2. Automatically login programmatically by setting an environment variable: Get an API KEY at https://wandb.ai/site/ and set the environment variable as follows:
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On Mac & Linux:
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```
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export WANDB_API_KEY=<YOUR WANDB API KEY>
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```
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On Windows:
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```
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set WANDB_API_KEY=<YOUR WANDB API KEY>
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```
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Moreover, if you couldn't access Wandb and want to log metrics offline, you can the environment variable as follows:
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```
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export WANDB_MODE=offline
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```
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## Inference
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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`.
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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`.
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- To avoid possible inference failures, make sure you have seen through the following instructions.
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- A longer prompt audio allows shorter generated output. The part longer than 30s cannot be generated properly. Consider using a prompt audio <15s.
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- Uppercased letters will be uttered letter by letter, so use lowercased letters for normal words.
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- 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.
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### CLI Inference
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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_file` in `inference-cli.py`
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for change model use `--ckpt_file` to specify the model you want to load,
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for change vocab.txt use `--vocab_file` to provide your vocab.txt file.
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```bash
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# switch to the main directory
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cd f5_tts
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python inference-cli.py \
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--model "F5-TTS" \
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--ref_audio "tests/ref_audio/test_en_1_ref_short.wav" \
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--ref_text "Some call me nature, others call me mother nature." \
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--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."
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python inference-cli.py \
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--model "E2-TTS" \
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--ref_audio "tests/ref_audio/test_zh_1_ref_short.wav" \
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--ref_text "对,这就是我,万人敬仰的太乙真人。" \
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--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?"
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# Multi voice
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python inference-cli.py -c samples/story.toml
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```
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### Gradio App
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Currently supported features:
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- Chunk inference
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- Podcast Generation
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- Multiple Speech-Type Generation
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You can launch a Gradio app (web interface) to launch a GUI for inference (will load ckpt from Huggingface, you may also use 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`.
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```bash
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python f5_tts/gradio_app.py
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```
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You can specify the port/host:
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```bash
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python f5_tts/gradio_app.py --port 7860 --host 0.0.0.0
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```
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Or launch a share link:
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```bash
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python f5_tts/gradio_app.py --share
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```
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### Speech Editing
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To test speech editing capabilities, use the following command.
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```bash
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python f5_tts/speech_edit.py
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```
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## Evaluation
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### Prepare Test Datasets
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1. Seed-TTS test set: Download from [seed-tts-eval](https://github.com/BytedanceSpeech/seed-tts-eval).
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2. LibriSpeech test-clean: Download from [OpenSLR](http://www.openslr.org/12/).
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3. Unzip the downloaded datasets and place them in the data/ directory.
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4. Update the path for the test-clean data in `scripts/eval_infer_batch.py`
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5. Our filtered LibriSpeech-PC 4-10s subset is already under data/ in this repo
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### Batch Inference for Test Set
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To run batch inference for evaluations, execute the following commands:
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```bash
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# switch to the main directory
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cd f5_tts
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# batch inference for evaluations
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accelerate config # if not set before
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bash scripts/eval_infer_batch.sh
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```
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### Download Evaluation Model Checkpoints
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1. Chinese ASR Model: [Paraformer-zh](https://huggingface.co/funasr/paraformer-zh)
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2. English ASR Model: [Faster-Whisper](https://huggingface.co/Systran/faster-whisper-large-v3)
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3. WavLM Model: Download from [Google Drive](https://drive.google.com/file/d/1-aE1NfzpRCLxA4GUxX9ITI3F9LlbtEGP/view).
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### Objective Evaluation
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Install packages for evaluation:
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```bash
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pip install -r requirements_eval.txt
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```
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**Some Notes**
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For faster-whisper with CUDA 11:
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```bash
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pip install --force-reinstall ctranslate2==3.24.0
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```
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(Recommended) To avoid possible ASR failures, such as abnormal repetitions in output:
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```bash
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pip install faster-whisper==0.10.1
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```
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Update the path with your batch-inferenced results, and carry out WER / SIM evaluations:
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```bash
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# switch to the main directory
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cd f5_tts
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# Evaluation for Seed-TTS test set
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python scripts/eval_seedtts_testset.py
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# Evaluation for LibriSpeech-PC test-clean (cross-sentence)
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python scripts/eval_librispeech_test_clean.py
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```
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## Acknowledgements
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- [E2-TTS](https://arxiv.org/abs/2406.18009) brilliant work, simple and effective
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- [Emilia](https://arxiv.org/abs/2407.05361), [WenetSpeech4TTS](https://arxiv.org/abs/2406.05763) valuable datasets
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- [lucidrains](https://github.com/lucidrains) initial CFM structure with also [bfs18](https://github.com/bfs18) for discussion
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- [SD3](https://arxiv.org/abs/2403.03206) & [Hugging Face diffusers](https://github.com/huggingface/diffusers) DiT and MMDiT code structure
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- [torchdiffeq](https://github.com/rtqichen/torchdiffeq) as ODE solver, [Vocos](https://huggingface.co/charactr/vocos-mel-24khz) as vocoder
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- [FunASR](https://github.com/modelscope/FunASR), [faster-whisper](https://github.com/SYSTRAN/faster-whisper), [UniSpeech](https://github.com/microsoft/UniSpeech) for evaluation tools
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- [ctc-forced-aligner](https://github.com/MahmoudAshraf97/ctc-forced-aligner) for speech edit test
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- [mrfakename](https://x.com/realmrfakename) huggingface space demo ~
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- [f5-tts-mlx](https://github.com/lucasnewman/f5-tts-mlx/tree/main) Implementation with MLX framework by [Lucas Newman](https://github.com/lucasnewman)
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- [F5-TTS-ONNX](https://github.com/DakeQQ/F5-TTS-ONNX) ONNX Runtime version by [DakeQQ](https://github.com/DakeQQ)
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## Citation
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If our work and codebase is useful for you, please cite as:
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```
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@article{chen-etal-2024-f5tts,
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title={F5-TTS: A Fairytaler that Fakes Fluent and Faithful Speech with Flow Matching},
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author={Yushen Chen and Zhikang Niu and Ziyang Ma and Keqi Deng and Chunhui Wang and Jian Zhao and Kai Yu and Xie Chen},
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journal={arXiv preprint arXiv:2410.06885},
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year={2024},
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}
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```
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## License
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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.
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