基于宏观时间演化系统的适应方法与间接/直接适应方法的统一解释

Shinji Watanabe, Atsushi Nakamura
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引用次数: 4

摘要

语音识别的增量自适应技术旨在快速、稳定地调整声学模型以适应由于说话人、说话方式、噪声源等时间变化而产生的时变声学特征。提出了一种基于宏观时间演化系统的增量自适应框架,通过连续更新声学模型参数的后验分布来模拟时变特征。在本文中,我们对该提议和两种主要的传统方法进行了统一的解释,即通过转换参数间接自适应(例如最大似然线性回归(MLLR))和直接自适应分类器参数(例如最大后验(MAP))。分析和实验结果表明,增量自适应既包括常规自适应方法,也包括组合自适应方法,并同时具有快速稳定的自适应特点。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
A unified interpretation of adaptation approaches based on a macroscopic time evolution system and indirect/direct adaptation approaches
Incremental adaptation techniques for speech recognition are aimed at adjusting acoustic models quickly and stably to time-variant acoustic characteristics due to temporal changes of speaker, speaking style, noise source, etc. We proposed a novel incremental adaptation framework based on a macroscopic time evolution system, which models the time-variant characteristics by successively updating posterior distributions of acoustic model parameters. In this paper, we provide a unified interpretation of the proposal and the two major conventional approaches of indirect adaptation via transformation parameters (e.g. maximum likelihood linear regression (MLLR)) and direct adaptation of classifier parameters (e.g. maximum a posteriori (MAP)). We reveal analytically and experimentally that the proposed incremental adaptation involves both the conventional and their combinatorial approaches, and simultaneously possesses their quick and stable adaptation characteristics.
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