[VALL-E](https://arxiv.org/abs/2301.02111) describes how treating text-to-speech synthesis as a language problem can easily be solved with a language model. The original paper utilizes a basic transformer as the underlying architecture to perform zero-shot text-to-speech synthesis using a short audio prompt as reference.
# Why VALL-E?
At the time, state-of-the-art neural-based TTS solutions were sparing. TorToiSe had a similar approach to treating TTS as a language problem, but required a ton of additional cruft on top of its ensemble. Thus, when VALL-E's paper released, it was simple yet effective with it requiring, at the time, just an AR and a NAR model, and leaving EnCodec to handle the rest (feature extraction, encoding audio, decoding audio). Vocos then improves upon EnCodec's decoding to produce better quality audio.
However, at this point and time, the implementation is rather divorced from VALL-E and its derivating papers, but the core principle is still followed.
* [ ] extend to addditional tasks ([SpeechX](https://arxiv.org/abs/2308.06873)).
-`stt` (Speech-to-Text) seems to be working fine for the most part.
- other tasks seem to require a ton of VRAM......
* [ ] extend using [VALL-E 2](https://arxiv.org/pdf/2406.05370)'s features (grouped code modeling + repetition aware sampling)
- desu these don't seem to be worthwhile improvements, as inferencing is already rather fast, and RAS is just a fancy sampler.
* [ ] audio streaming
- this *technically* can work without any additional architecture changes, just clever tricks with sampling-then-decoding-to-audio.
- something similar to HiFiGAN (or the one for TorToiSe) trained on the last hidden states of the AR *might* also enable an alternate way for streaming.
* [ ] speed up inferencing
- KV caching both yields broken output and quadratically slow output, unless I'm doing something grossly wrong.
- A pure HF model is the only way to fix this, but converting the model to one is a bit of a chore.
* [ ] replace the phonemizer with something that doesn't depend on espeak
* [ ] train the model to handle text => phoneme (without a hit to the rest of the model)
* [ ] ...and phonemes => text
* [ ] allow raw text as input instead
- espeak is nice, but I can only really put my whole trust with phonemizing English.
- a small model trained to handle converting text to phonemes might work, but has it's own problems (another model to carry around, as accurate as the dataset it was trained against, requires training for each language... etc).
* [ ] smarter/clever inferencing, such as:
* [ ] "rolling" context, where the last generated sentence is the prefix for the next sentence.
* [ ] explore exotic features like:
* using a pure text vocab rather than IPA phonemes (as a transformer should be "smart" enough to map text tokens)
* interleaving by using summed embedding tokens:
* for example, `<RVQ 0-7><RVQ 0>` => `<RVQ 0-7><RVQ 0-1>` => `<RVQ 0-7><RVQ 0-2>` (etc.)
* however, I imagine the sequences to train for this are *too* exotic.
* mixing multiple speakers through summing input prompt embeddings
* I do not expect this to work, but you never know...
## Notices and Citations
Unless otherwise credited/noted in this repo or within the designated Python file, this repository is [licensed](LICENSE) under AGPLv3.
- [EnCodec](https://github.com/facebookresearch/encodec) is licensed under CC-BY-NC 4.0. If you use the code to generate audio quantization or perform decoding, it is important to adhere to the terms of their license.
- This implementation was originally based on [enhuiz/vall-e](https://github.com/enhuiz/vall-e), but has been heavily, heavily modified over time. Without it I would not have had a good basis to muck around and learn.
```bibtex
@article{wang2023neural,
title={Neural Codec Language Models are Zero-Shot Text to Speech Synthesizers},
author={Wang, Chengyi and Chen, Sanyuan and Wu, Yu and Zhang, Ziqiang and Zhou, Long and Liu, Shujie and Chen, Zhuo and Liu, Yanqing and Wang, Huaming and Li, Jinyu and others},
journal={arXiv preprint arXiv:2301.02111},
year={2023}
}
```
```bibtex
@article{defossez2022highfi,
title={High Fidelity Neural Audio Compression},
author={Défossez, Alexandre and Copet, Jade and Synnaeve, Gabriel and Adi, Yossi},
