Transfer tasks in text-to-speech (TTS) synthesis -where one or more aspects of the speech of one set of speakers is transferred to another set of speakers that do not feature these aspects originally -remains a challenging task.One of the challenges is that models that have high-quality transfer capabilities can have issues in stability, making them impractical for user-facing critical tasks.This paper demonstrates that transfer can be obtained by training a robust TTS system on data generated by a less robust TTS system designed for a high-quality transfer task; in particular, a CHiVE-BERT monolingual TTS system is trained on the output of a Tacotron model designed for accent transfer.While some quality loss is inevitable with this approach, experimental results show that the models trained on synthetic data this way can produce high quality audio displaying accent transfer, while preserving speaker characteristics such as speaking style.

Sequential models achieve state-of-the-art results in audio, visual and textual domains with respect to both estimating the data distribution and generating desired samples. Efficient sampling for this class of models at the cost of little to no loss in quality has however remained an elusive problem. With a focus on text-to-speech synthesis, we describe a set of general techniques for reducing sampling time while maintaining high output quality. We first describe a single-layer recurrent neural network, the WaveRNN, with a dual softmax layer that matches the quality of the state-of-the-art WaveNet model. The compact form of the network makes it possible to generate 24 kHz 16-bit audio 4 times faster than real time on a GPU. Secondly, we apply a weight pruning technique to reduce the number of weights in the WaveRNN. We find that, for a constant number of parameters, large sparse networks perform better than small dense networks and this relationship holds past sparsity levels of more than 96%. The small number of weights in a Sparse WaveRNN makes it possible to sample high-fidelity audio on a mobile phone CPU in real time. Finally, we describe a new dependency scheme for sampling that lets us trade a constant number of non-local, distant dependencies for the ability to generate samples in batches. The Batch WaveRNN produces 8 samples per step without loss of quality and offers orthogonal ways of further increasing sampling efficiency.

The recently-developed WaveNet architecture is the current state of the art in realistic speech synthesis, consistently rated as more natural sounding for many different languages than any previous system. However, because WaveNet relies on sequential generation of one audio sample at a time, it is poorly suited to today’s massively parallel computers, and therefore hard to deploy in a real-time production setting. This paper introduces Probability Density Distillation, a new method for training a parallel feed-forward network from a trained WaveNet with no significant difference in quality. The resulting system is capable of generating high-fidelity speech samples at more than 20 times faster than real-time, a 1000x speed up relative to the original WaveNet, and capable of serving multiple English and Japanese voices in a production setting.

This is the workshop report for WOMRAD 2010, the Workshop on Music Recommendation and Discovery held on 26th September 2010 in conjunction with ACM RecSys in Barcelona Spain.