--- license: mit tags: - audio - text-to-speech datasets: - ovieyra21/mabama-v8 - ovieyra21/mabama-v9 base_model: - microsoft/speecht5_tts pipeline_tag: text-to-speech metrics: - charcut_mt new_version: openai/whisper-large-v3-turbo library_name: espnet --- # SpeechT5 (TTS task) SpeechT5 model fine-tuned for speech synthesis (text-to-speech) on LibriTTS. This model was introduced in [SpeechT5: Unified-Modal Encoder-Decoder Pre-Training for Spoken Language Processing](https://arxiv.org/abs/2110.07205) by Junyi Ao, Rui Wang, Long Zhou, Chengyi Wang, Shuo Ren, Yu Wu, Shujie Liu, Tom Ko, Qing Li, Yu Zhang, Zhihua Wei, Yao Qian, Jinyu Li, Furu Wei. SpeechT5 was first released in [this repository](https://github.com/microsoft/SpeechT5/), [original weights](https://huggingface.co/mechanicalsea/speecht5-tts). The license used is [MIT](https://github.com/microsoft/SpeechT5/blob/main/LICENSE). ## Model Description Motivated by the success of T5 (Text-To-Text Transfer Transformer) in pre-trained natural language processing models, we propose a unified-modal SpeechT5 framework that explores the encoder-decoder pre-training for self-supervised speech/text representation learning. The SpeechT5 framework consists of a shared encoder-decoder network and six modal-specific (speech/text) pre/post-nets. After preprocessing the input speech/text through the pre-nets, the shared encoder-decoder network models the sequence-to-sequence transformation, and then the post-nets generate the output in the speech/text modality based on the output of the decoder. Leveraging large-scale unlabeled speech and text data, we pre-train SpeechT5 to learn a unified-modal representation, hoping to improve the modeling capability for both speech and text. To align the textual and speech information into this unified semantic space, we propose a cross-modal vector quantization approach that randomly mixes up speech/text states with latent units as the interface between encoder and decoder. Extensive evaluations show the superiority of the proposed SpeechT5 framework on a wide variety of spoken language processing tasks, including automatic speech recognition, speech synthesis, speech translation, voice conversion, speech enhancement, and speaker identification. - **Developed by:** Junyi Ao, Rui Wang, Long Zhou, Chengyi Wang, Shuo Ren, Yu Wu, Shujie Liu, Tom Ko, Qing Li, Yu Zhang, Zhihua Wei, Yao Qian, Jinyu Li, Furu Wei. - **Shared by [optional]:** [Matthijs Hollemans](https://huggingface.co/Matthijs) - **Model type:** text-to-speech - **Language(s) (NLP):** [More Information Needed] - **License:** [MIT](https://github.com/microsoft/SpeechT5/blob/main/LICENSE) - **Finetuned from model [optional]:** [More Information Needed] ## Model Sources [optional] - **Repository:** [https://github.com/microsoft/SpeechT5/] - **Paper:** [https://arxiv.org/pdf/2110.07205.pdf] - **Blog Post:** [https://huggingface.co/blog/speecht5] - **Demo:** [https://huggingface.co/spaces/Matthijs/speecht5-tts-demo] # Uses ## 🤗 Transformers Usage You can run SpeechT5 TTS locally with the 🤗 Transformers library. 1. First install the 🤗 [Transformers library](https://github.com/huggingface/transformers), sentencepiece, soundfile and datasets(optional): ``` pip install --upgrade pip pip install --upgrade transformers sentencepiece datasets[audio] ``` 2. Run inference via the `Text-to-Speech` (TTS) pipeline. You can access the SpeechT5 model via the TTS pipeline in just a few lines of code! ```python from transformers import pipeline from datasets import load_dataset import soundfile as sf synthesiser = pipeline("text-to-speech", "microsoft/speecht5_tts") embeddings_dataset = load_dataset("Matthijs/cmu-arctic-xvectors", split="validation") speaker_embedding = torch.tensor(embeddings_dataset[7306]["xvector"]).unsqueeze(0) # You can replace this embedding with your own as well. speech = synthesiser("Hello, my dog is cooler than you!", forward_params={"speaker_embeddings": speaker_embedding}) sf.write("speech.wav", speech["audio"], samplerate=speech["sampling_rate"]) ``` 3. Run inference via the Transformers modelling code - You can use the processor + generate code to convert text into a mono 16 kHz speech waveform for more fine-grained control. ```python from transformers import SpeechT5Processor, SpeechT5ForTextToSpeech, SpeechT5HifiGan from datasets import load_dataset import torch import soundfile as sf from datasets import load_dataset processor = SpeechT5Processor.from_pretrained("microsoft/speecht5_tts") model = SpeechT5ForTextToSpeech.from_pretrained("microsoft/speecht5_tts") vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan") inputs = processor(text="Hello, my dog is cute.", return_tensors="pt") # load xvector containing speaker's voice characteristics from a dataset embeddings_dataset = load_dataset("Matthijs/cmu-arctic-xvectors", split="validation") speaker_embeddings = torch.tensor(embeddings_dataset[7306]["xvector"]).unsqueeze(0) speech = model.generate_speech(inputs["input_ids"], speaker_embeddings, vocoder=vocoder) sf.write("speech.wav", speech.numpy(), samplerate=16000) ``` ### Fine-tuning the Model Refer to [this Colab notebook](https://colab.research.google.com/drive/1i7I5pzBcU3WDFarDnzweIj4-sVVoIUFJ) for an example of how to fine-tune SpeechT5 for TTS on a different dataset or a new language. ## Direct Use You can use this model for speech synthesis. See the [model hub](https://huggingface.co/models?search=speecht5) to look for fine-tuned versions on a task that interests you. ## Downstream Use [optional] [More Information Needed] ## Out-of-Scope Use [More Information Needed] # Bias, Risks, and Limitations [More Information Needed] ## Recommendations Users (both direct and downstream) should be made aware of the risks, biases and limitations of the model. More information needed for further recommendations. ## Training Data LibriTTS ## Training Procedure ### Preprocessing [optional] Leveraging large-scale unlabeled speech and text data, we pre-train SpeechT5 to learn a unified-modal representation, hoping to improve the modeling capability for both speech and text. ### Training hyperparameters - **Precision:** [More Information Needed] - **Regime:** [More Information Needed] ### Speeds, Sizes, Times [optional] [More Information Needed] # Evaluation ## Testing Data, Factors & Metrics ### Testing Data [More Information Needed] ### Factors [More Information Needed] ### Metrics [More Information Needed] ## Results [More Information Needed] ### Summary # Model Examination [optional] Extensive evaluations show the superiority of the proposed SpeechT5 framework on a wide variety of spoken language processing tasks, including automatic speech recognition, speech synthesis, speech translation, voice conversion, speech enhancement, and speaker identification. # Environmental Impact Carbon emissions can be estimated using the [Machine Learning Impact calculator](https://mlco2.github.io/impact#compute) presented in [Lacoste et al. (2019)](https://arxiv.org/abs/1910.09700). - **Hardware Type:** [More Information Needed] - **Hours used:** [More Information Needed] - **Cloud Provider:** [More Information Needed] - **Compute Region:** [More Information Needed] - **Carbon Emitted:** [More Information Needed] # Technical Specifications [optional] ## Model Architecture and Objective The SpeechT5 framework consists of a shared encoder-decoder network and six modal-specific (speech/text) pre/post-nets. After preprocessing the input speech/text through the pre-nets, the shared encoder-decoder network models the sequence-to-sequence transformation, and then the post-nets generate the output in the speech/text modality based on the output of the decoder. ## Compute Infrastructure [More Information Needed] ### Hardware [More Information Needed] ### Software [More Information Needed] # Citation [optional] **BibTeX:** ```bibtex @inproceedings{ao-etal-2022-speecht5, title = {{S}peech{T}5: Unified-Modal Encoder-Decoder Pre-Training for Spoken Language Processing}, author = {Ao, Junyi and Wang, Rui and Zhou, Long and Wang, Chengyi and Ren, Shuo and Wu, Yu and Liu, Shujie and Ko, Tom and Li, Qing and Zhang, Yu and Wei, Zhihua and Qian, Yao and Li, Jinyu and Wei, Furu}, booktitle = {Proceedings of the 60th Annual Meeting of the Association for Computational Linguistics (Volume 1: Long Papers)}, month = {May}, year = {2022}, pages={5723--5738}, } ``` # Glossary [optional] - **text-to-speech** to synthesize audio # More Information [optional] [More Information Needed] # Model Card Authors [optional] Disclaimer: The team releasing SpeechT5 did not write a model card for this model so this model card has been written by the Hugging Face team. # Model Card Contact [More Information Needed]