Leveraging Sequence-to-Sequence Speech Synthesis for Enhancing Acoustic-to-Word Speech Recognition

M. Mimura, Sei Ueno, H. Inaguma, S. Sakai, Tatsuya Kawahara
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引用次数: 41

Abstract

Encoder-decoder models for acoustic-to-word (A2W) automatic speech recognition (ASR) are attractive for their simplicity of architecture and run-time latency while achieving state-of-the-art performances. However, word-based models commonly suffer from the-of-vocabulary (OOV) word problem. They also cannot leverage text data to improve their language modeling capability. Recently, sequence-to-sequence neural speech synthesis models trainable from corpora have been developed and shown to achieve naturalness com- parable to recorded human speech. In this paper, we explore how we can leverage the current speech synthesis technology to tailor the ASR system for a target domain by preparing only a relevant text corpus. From a set of target domain texts, we generate speech features using a sequence-to-sequence speech synthesizer. These artificial speech features together with real speech features from conventional speech corpora are used to train an attention-based A2W model. Experimental results show that the proposed approach improves the word accuracy significantly compared to the baseline trained only with the real speech, although synthetic part of the training data comes only from a single female speaker voice.
利用序列到序列的语音合成增强声到词的语音识别
用于声到字(A2W)自动语音识别(ASR)的编码器-解码器模型因其架构的简单性和运行时延迟而具有吸引力,同时实现了最先进的性能。然而,基于单词的模型通常会遇到词汇表问题(OOV)。他们也不能利用文本数据来提高他们的语言建模能力。近年来,可从语料库中训练的序列到序列神经语音合成模型已经被开发出来,并被证明可以达到与人类语音记录相似的自然程度。在本文中,我们探讨了如何利用当前的语音合成技术,通过准备相关的文本语料库来为目标领域量身定制ASR系统。从一组目标域文本中,我们使用序列到序列语音合成器生成语音特征。这些人工语音特征与来自传统语音语料库的真实语音特征一起用于训练基于注意力的A2W模型。实验结果表明,尽管训练数据的合成部分仅来自单个女性说话人的声音,但与仅使用真实语音训练的基线相比,该方法显著提高了单词准确率。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
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