Acoustic modeling using transform-based phone-cluster adaptive training

Vimal Manohar, S. C. Bhargav, S. Umesh
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引用次数: 7

Abstract

In this paper, we propose a new acoustic modeling technique called the Phone-Cluster Adaptive Training. In this approach, the parameters of context-dependent states are obtained by the linear interpolation of several monophone cluster models, which are themselves obtained by adaptation using linear transformation of a canonical Gaussian Mixture Model (GMM). This approach is inspired from the Cluster Adaptive Training (CAT) for speaker adaptation and the Subspace Gaussian Mixture Model (SGMM). The parameters of the model are updated in an adaptive training framework. The interpolation vectors implicitly capture the phonetic context information. The proposed approach shows substantial improvement over the Continuous Density Hidden Markov Model (CDHMM) and a similar performance to that of the SGMM, while using significantly fewer parameters than both the CDHMM and the SGMM.
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基于变换的电话簇自适应训练声学建模
在本文中,我们提出了一种新的声学建模技术,称为电话簇自适应训练。该方法利用经典高斯混合模型(GMM)的线性变换自适应,对多个单声道聚类模型进行线性插值,得到与上下文相关的状态参数。该方法的灵感来自于用于说话人自适应的聚类自适应训练(CAT)和子空间高斯混合模型(SGMM)。在自适应训练框架中更新模型参数。插值向量隐式地捕获语音上下文信息。该方法比连续密度隐马尔可夫模型(CDHMM)有了很大的改进,性能与SGMM相似,同时使用的参数比CDHMM和SGMM都少得多。
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Learning filter banks within a deep neural network framework Efficient nearly error-less LVCSR decoding based on incremental forward and backward passes Porting concepts from DNNs back to GMMs Discriminative piecewise linear transformation based on deep learning for noise robust automatic speech recognition Acoustic modeling using transform-based phone-cluster adaptive training
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