HMM Mixtures (HMM2) for Robust Speech Recognition
State-of-the-art automatic speech recognition (ASR) techniques are typically based on hidden Markov models (HMMs) for the modeling of temporal sequences of feature vectors extracted from the speech signal. At the level of each HMM state, Gaussian mixture models (GMMs) or artificial neural networks (ANNs) are commonly used in order to model the state emission probabilities. However, both GMMs and ANNs are rather rigid, as they are incapable of adapting to variations inherent in the speech signal, such as inter- and intra-speaker variations. Moreover, performance degradations of these systems are severe in the case of unmatched conditions such as in the presence of environmental noise. A lot of research effort is currently being devoted to overcoming these problems. The principal objective of this thesis is to explore new approaches towards a more robust and adaptive modeling of speech. In this context, different aspects of the modeling of speech data with HMMs and GMMs are investigated. Particular attention is given to the modeling of correlation. While correlation between different feature vectors (corresponding to temporal correlation) is typically modeled by the HMM, correlation between feature vector components (e.g., correlation in frequency) is modeled by the GMM part of the model. This thesis starts with the investigation of two potential ways to improve the modeling of correlation, consisting of (1) a shift of the modeling of temporal correlation towards GMMs, and (2) the modeling of correlation within each feature vector by a particular type of HMM. This leads to the development of a novel approach, referred to as ÒHMM2Ó, which is a major focus of this thesis. HMM2 is a particular mixture of hidden Markov models, where state emission probabilities of the temporal (primary) HMM are modeled through (secondary) state-dependent frequency-based HMMs. Low-dimensional GMMs are used for modeling the state emission probabilities of the secondary HMM states. Therefore, HMM2 can be seen as a generalization of conventional HMMs, which they include as a particular case. HMM2 may have several advantages as compared to standard systems. While the primary HMM performs time warping and time integration, the secondary HMM performs warping and integration along the frequency dimension of the speech signal. Frequency correlation is modeled through the secondary HMM topology. Due to the implicit, non-linear, state-dependent spectral warping performed by the secondary HMM, HMM2 may be viewed as a dynamic extension of the multi-band approach. Moreover, this frequency warping property may result in a better, more flexible modeling and parameter sharing. After an investigation of theoretical and practical aspects of HMM2, encouraging recognition results for the case of speech degraded by additive noise are given. Due to the spectral warping property of HMM2, this model is able to extract pertinent structural information of the speech signal, which is reflected in the trained model parameters. Consequently, such an HMM2 system can also be used to explicitly extract structures of a speech signal, which can then be converted into a new kind of ASR features, referred to as ÒHMM2 featuresÓ. In fact, frequency bands with similar characteristics are supposed to be emitted by the same secondary HMM state. The warping along the frequency dimension of speech thus results in an adaptable, data-driven frequency segmentation. In fact, as it can be assumed that different secondary HMM states model spectral regions characterized by high and low energies respectively, this segmentation may be related to formant structures. The application of HMM2 as a feature extractor is investigated, and it is shown that a system combining HMM2 features with conventional noise-robust features yields an improved speech recognition robustness. Moreover, a comparison of HMM2 features with formant tracks shows a comparable performance on a vowel classification task.
Published as: Weber, K. (2003). ``HMM Mixtures (HMM2) for Robust Speech Recognition''. Docteur ès Sciences thesis, Swiss Federal Institute of Technology Lausanne (EPFL), Lausanne, Switzerland.
Record created on 2006-03-10, modified on 2016-08-08