Text-to-Speech (TTS) has recently seen great progress in synthesizing high-quality speech owing to the rapid development of parallel TTS systems, but producing speech with naturalistic prosodic variations, speaking styles and emotional tones remains challenging. Moreover, since duration and speech are generated separately, parallel TTS models still have problems finding the best monotonic alignments that are crucial for naturalistic speech synthesis. Here, we propose StyleTTS, a style-based generative model for parallel TTS that can synthesize diverse speech with natural prosody from a reference speech utterance. With novel Transferable Monotonic Aligner (TMA) and duration-invariant data augmentation schemes, our method significantly outperforms state-of-the-art models on both single and multi-speaker datasets in subjective tests of speech naturalness and speaker similarity. Through self-supervised learning of the speaking styles, our model can synthesize speech with the same prosodic and emotional tone as any given reference speech without the need for explicitly labeling these categories.

Paper: https://arxiv.org/abs/2107.10394

Audio samples: https://styletts.github.io/

1. Python >= 3.7
2. Clone this repository:

git clone https://github.com/yl4579/StyleTTS.git
cd StyleTTS

3. Install python requirements:

pip install SoundFile torchaudio munch torch pydub pyyaml librosa git+https://github.com/resemble-ai/monotonic_align.git

4. Download and extract the LJSpeech dataset, unzip to the data folder and upsample the data to 24 kHz. The vocoder, text aligner and pitch extractor are pre-trained on 24 kHz data, but you can easily change the preprocessing and re-train them using your own preprocessing. I will provide more receipes and pre-trained models later if I have time. If you are willing to help, feel free to work on other preprocessing methods. For LibriTTS, you will need to combine train-clean-360 with train-clean-100 and rename the folder train-clean-460 (see val_list_libritts.txt as an example).

First stage training:

python train_first.py --config_path ./Configs/config.yml

Second stage training:

python train_second.py --config_path ./Configs/config.yml

You can run both consecutively and it will train both the first and second stage. The model will be saved in the format "epoch_1st_%05d.pth" and "epoch_2nd_%05d.pth". Checkpoints and Tensorboard logs will be saved at log_dir.

The data list format needs to be filename.wav|transcription, see val_list_libritts.txt as an example.

Please refer to inference.ipynb for details.

The pretrained StyleTTS and Hifi-GAN on LJSpeech corpus in 24 kHz can be downloaded at StyleTTS Link and Hifi-GAN Link.

The pretrained StyleTTS and Hifi-GAN on LibriTTS corpus can be downloaded at StyleTTS Link and Hifi-GAN Link. You also need to download test-clean from LibriTTS if you want to run the zero-shot demo.

Please unzip to Models and Vocoder respectivey and run each cell in the notebook. You will also need to install phonemizer to run this inference demo.

The pretrained text aligner and pitch extractor models are provided under the Utils folder. Both the text aligner and pitch extractor models are trained with melspectrograms preprocessed using meldataset.py.

You can edit the meldataset.py with your own melspectrogram preprocessing, but the provided pretrained models will no longer work. You will need to train your own text aligner and pitch extractor with the new preprocessing.

The code for training new text aligner model is available here and that for training new pitch extractor models is available here.

I will provide more recepies with existing preprocessing like those in official HifiGAN and ESPNet in the future if I have extra time. If you are willing to help, feel free to make receipes with ESPNet.