Having played around with flite a decade ago, and at that time feeling that it was already then nowhere close to the fidelity of other speech synthesis examples. I find it surprising that there still isn't anything better than festival/flite? It sounded then like a clunky robot, and still does today. Surely some of the many research projects have released their work as open source? Work like, say https://arxiv.org/abs/1806.04558 [paper] https://github.com/CorentinJ/Real-Time-Voice-Cloning [repo]
Transfer Learning from Speaker Verification to Multispeaker Text-To-Speech Synthesis
Ye Jia, Yu Zhang, Ron J. Weiss
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Quan Wang, Jonathan Shen, Fei Ren, Zhifeng Chen, Patrick Nguyen, Ruoming Pang, Ignacio Lopez Moreno, Yonghui Wu
2018 · arXiv
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From the abstract
We describe a neural network-based system for text-to-speech (TTS) synthesis that is able to generate speech audio in the voice of many different speakers, including those unseen during training. Our system consists of three independently trained components: (1) a speaker encoder network, trained on a speaker verification task using an independent dataset of noisy speech from thousands of speakers without transcripts, to generate a fixed-dimensional embedding vector from seconds of reference speech from a target speaker; (2) a sequence-to-sequence synthesis network based on Tacotron 2, which generates a mel spectrogram from text, conditioned on the speaker embedding; (3) an auto-regressive WaveNet-based vocoder that converts the mel spectrogram into a sequence of time domain waveform samples. We demonstrate that the proposed model is able to transfer the knowledge of speaker variability learned by the discriminatively-trained speaker encoder to the new task, and is able to synthesize natural speech from speakers that were not seen during training. We quantify the importance of training the speaker encoder on a large and diverse speaker set in order to obtain the best generalization performance. Finally, we show that randomly sampled speaker embeddings can be used to synthesize speech in the voice of novel speakers dissimilar from those used in training, indicating that the model has learned a high quality speaker representation.
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This is not correct. The security page only shows that they use Google for Speech to Text. Instead, Text to Speech is done using technology developed by Lyrebird.ai, which Descript has bought over. Descript rebranded it as Overdub. Note that style transfer learning of voices is a hard problem and Overdub seems to have nailed it perfectly. I speculate that the underlying technology of Overdub is based on sv2tts (https://arxiv.org/abs/1806.04558). The closest comparision to Overdub would be https://www.resemble.ai/.
This can be trained using only 5 Seconds of reference audio: https://google.github.io/tacotron/publications/speaker_adapt... https://arxiv.org/pdf/1806.04558.pdf It's been mentioned a bit already, but thought it was worth calling out. This may be one of the lowest-overhead ways to start experimenting, at least in terms of data collection.
Hey, speech ML researcher here. Make sure you have different recordings of different contexts. fifteen.ai's best TTS voices use ~90 min of utterances, some separated by emotion. If you're having her read a text, make sure it's engaging--we do a lot of unconscious voicing when reading aloud. Tbh, if she has a non-Anglophone accent, you're going to need more because the training data is biased towards UK/US speakers. If you want to read up on the basics, check out the SV2TTS paper: https://arxiv.org/pdf/1806.04558.pdf Basically you use a speaker encoding to condition the TTS output. This paper/idea is used all over, even for speech-to-speech translation, with small…
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changes. There's a few open-source version implementations but mostly outdated--the better ones are either private for business or privacy reasons. There's a lot of work on non-parallel transfer learning (aka subjects are saying different things) so TTS has progressed rapidly and most public implementations lag a bit behind the research. If you're willing to grok speech processing, I'd start with NeMo for overall simplicity--don't get distracted by Kaldi. Edit: Important note! Utterances are usually clipped of silence before/after so take that into account when analyzing corpus lengths. The quality of each utterance is much much more important than the length--fifteen.ai's TTS is so good primarily because they got fans of each character to collect the data.
The paper "Generalization Of Audio Deepfake Detection" gives an overview. The paper https://arxiv.org/abs/1904.05441 has a list of spoofing methods. Here's one method as paper https://arxiv.org/pdf/1806.04558.pdf And here on GitHub https://github.com/CorentinJ/Real-Time-Voice-Cloning
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