We present a new neural text to speech method that is able to transform text to speech in voices that are sampled in the wild. Unlike other text to speech systems, our solution is able to deal with unconstrained samples obtained from public speeches. The network architecture is simpler than those in the existing literature and is based on a novel shifting buffer working memory. The same buffer is used for estimating the attention, computing the output audio, and for updating the buffer itself. The input sentence is encoded using a context-free lookup table that contains one entry per character or phoneme. Lastly, the speakers are similarly represented by a short vector that can also be fitted to new speakers and variability in the generated speech is achieved by priming the buffer prior to generating the audio. Experimental results on two datasets demonstrate convincing multi-speaker and in-the-wild capabilities. In order to promote reproducibility, we release our source code and models: PyTorch code and sample audio files are available at ytaigman.github.io/loop.
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