Building Automatic Speech Recognition (ASR) systems for code-switched speech has recently gained renewed attention due to the widespread use of speech technologies in multilingual communities worldwide. End-to-end ASR systems are a natural modeling choice due to their ease of use and superior performance in monolingual settings. However, it is well known that end-to-end systems require large amounts of labeled speech. In this work, we investigate improving code-switched ASR in low resource settings via data augmentation using code-switched text-to-speech (TTS) synthesis. We propose two targeted techniques to effectively leverage TTS speech samples: 1) Mixup, an existing technique to create new training samples via linear interpolation of existing samples, applied to TTS and real speech samples, and 2) a new loss function, used in conjunction with TTS samples, to encourage code-switched predictions. We report significant improvements in ASR performance achieving absolute word error rate (WER) reductions of up to 5%, and measurable improvement in code switching using our proposed techniques on a Hindi-English code-switched ASR task.
翻译:由于全球多语种社区广泛使用语音技术,最近人们重新关注了用于代码转换语音的自动语音识别系统(ASR)的建设。端到端 ASR系统是一种自然的建模选择,因为其使用方便,而且单语环境中的性能优异。然而,众所周知,端到端系统需要大量的标签式语音。在这项工作中,我们调查通过使用代码转换文本到语音合成(TTS)的数据增强,在低资源环境中改进代码转换的ASR系统。我们提出了两种有针对性的技术,以有效地利用 TTS 语音样本:1) Mixup,这是一种现有的技术,通过对现有样本进行线性内插,适用于 TTS 和真实语音样本,来创建新的培训样本。2)与 TTS 样本一起使用的新的损失功能,以鼓励代码转换预测。我们报告说,ASR的性能有了显著改善,实现了绝对字差率的降低5%,并利用我们提议的印地英语代码转换的ASR任务技术对代码转换作了可衡量的改进。