Traditionally, in speech recognition, the hidden Markov model state emission probability distributions are usually associated to continuous random variables, by using Gaussian mixtures. Thus, complex multimodal inter-feature dependencies are not accurately modeled by Gaussian models, since they are unimodal distributions and mixtures of Gaussians are needed in these complex cases, but this is done in a loose and inefficient way. Graphical models provide a precise and simple mechanism to model the dependencies among two or more variables. This paper proposes the use of discrete random variables as observations and graphical models to extract the internal dependence structure in the feature vectors. Therefore, speech features are quantized to a small number of levels, in order to obtain a tractable model. These quantized speech features provide a mechanism to increase the robustness against noise uncertainty. In addition, discrete random variables allow the learning of joint statistics of the observation densities. A method to estimate a graphical model with a constrained number of dependencies is shown in this paper, being a special kind of Bayesian network. Experimental results show that by using this modeling, better performance can be obtained compared to standard baseline systems.

In this paper, an unsupervised data-driven robust speech recognition approach is proposed based on a joint feature vector normalization and acoustic model adaptation. Feature vector normalization reduces the acoustic mismatch between training and testing conditions by mapping the feature vectors towards the training space. Model adaptation modifies the parameters of the acoustic models to match the test space. However, since neither is optimal, both approaches use an intermediate space between training and testing spaces to map either the feature vectors or acoustic models. The joint optimization of both approaches provides a common intermediate space with a better match between normalized feature vectors and adapted acoustic models. In this paper, feature vector normalization is based on a minimum mean square error (MMSE) criterion. A class dependent multi-environment model linear normalization (CD-MEMLIN) based on two classes (silence/speech) with a cross probability model (CD-MEMLIN-CPM) is used. CD-MEMLIN-CPM assumes that each class of clean and noisy spaces can be modeled with a Gaussian mixture model (GMM), training a linear transformation for each pair of Gaussians in an unsupervised data-driven training process. This feature vector normalization maps the recognition space feature vector to a normalized space. The acoustic model adaptation maps the training space to the normalized space by defining a set of linear transformations over an expanded HMM-state space, compensating for those degradations that the feature vector normalization is not able to model, like rotations. Experiments have been carried out with the Spanish SpeechDat Car database and Aurora 2 databases using both the standard Mel-frequency cepstral coefficient (MFCC) and advanced ETSI front-ends. Consistent improvements were reached for both corpora and front-ends. Using the standard MFCC front-end, a 92.08% average improvement on WER for Spanish SpeechDat Car and a 69.75% average improvement for clean condition evaluation of Aurora 2 was obtained, improving those results reached with ETSI advanced front-end (83.28% and 67.41%, respectively). Using the ETSI advanced front-end with the proposed solution, a 75.47% average improvement was obtained for the clean condition evaluation of Aurora 2 database.

In this paper we present two new methods for speech enhancement based on the previously publised fine pitch model (FPM) for voiced speech. The first method (FPM-NE) uses the FPM to produce a non-stationary noise estimate that can be used in any standard speech enhancement system. In this method, the FPM is used indirectly to perform speech enhancement. The second method we describe (FPM-SE) uses the FPM directly to perform speech enhancement. We present a study of the behavior of the two models on the standard Aurora 2 task, and demonstrate improvements of over 45% average word error rate reduction over the multi-style baseline.

We propose on-line unsupervised compensation technique for robust speech recognition that combines standard and throat microphone feature vectors. The solution, called Multi-Environment Model-based LInear Normalization with Throat microphone information, MEMLINT, is an extension of MEM-LIN formulation. Hence, standard microphone noisy space and throat microphone space are modelled as GMMs and a set of linear transformations are learnt from data associated to each pair of Gaussians (one for each GMM) using training stereo data. On the other hand, to compensate some kinds of degra-dation which are not considered in MEMLINT, we propose to use jointly an on-line unsupervised acoustic model adaptation method based on rotation transformations over an expanded HMM-state space (augMented stAte space acousTic dEcoder, MATE). Some experiments with an own recorded database were carried out, showing that the proposed approach significantly outperforms the single microphone approach. Index Terms: Throat microphone, robust speech recognition, feature vector normalization.

The new model reduces the impact of local spectral and temporal variability by estimating a finite set of spectral and temporal warping factors which are applied to speech at the frame level. Optimum warping factors are obtained while decoding in a locally constrained search. The model involves augmenting the states of a standard hidden Markov model (HMM), providing an additional degree of freedom. It is argued in this paper that this represents an efficient and effective method for compensating local variability in speech which may have potential application to a broader array of speech transformations. The technique is presented in the context of existing methods for frequency warping-based speaker normalization for ASR. The new model is evaluated in clean and noisy task domains using subsets of the Aurora 2, the Spanish Speech-Dat-Car, and the TIDIGITS corpora. In addition, some experiments are performed on a Spanish language corpus collected from a population of speakers with a range of speech disorders. It has been found that, under clean or not severely degraded conditions, the new model provides improvements over the standard HMM baseline. It is argued that the framework of local warping is an effective general approach to providing more flexible models of speaker variability.

An on-line unsupervised hybrid compensation technique is proposed to reduce the mismatch between training and testing conditions. It combines multi-environment model based linear normalization with cross-probability model based on GMMs (MEMLIN CPM) with a novel acoustic model adaptation method based on rotation transformations. Hence, a set of rotation transformations is estimated with clean and MEMLIN CPM-normalized training data by linear regression in an unsupervised process. Thus, in testing, each MEMLIN CPM normalized frame is decoded using a modified Viterbi algorithm and expanded acoustic models, which are obtained from the reference ones and the set of rotation transformations. To test the proposed solution, some experiments with Spanish SpeechDat Car database were carried out. MEMLIN CPM over standard ETSI front-end parameters reaches 83.89% of average improvement in WER, while the introduced hybrid solution goes up to 92.07%. Also, the proposed hybrid technique was tested with Aurora 2 database, obtaining an average improvement of 68.88% with clean training.

This paper describes the performance of the combination of Multi-Environment Model-based LInear Normalization, MEMLIN, which provides an estimation of the uncorrupted feature vector, with Maximum Likelihood Linear Regression, MLLR, for the collected database under the auspices of the IST-EU STREP project HIWIRE. In this work the results for the nonnative adaptation task (NNA) are presented. The HIWIRE project database consist on command and control aeronautics application utterances pronounced by non-native speakers which are digitally corrupted with airplane cockpit noise. Thus, three noise conditions are defined: low, medium and high noise. In the proposed system, each MEMLIN-normalized feature vector is decoded using the MLLR-adapted acoustic models. The experiments show that an important improvement is reached combining MEMLIN and MLLR methods for all kinds of non-native speakers and noise conditions. Index Terms: Hiwire project, robust speech recognition, nonnative adaptation task.

In this paper, a set of feature vector normalization methods based on the minimum mean square error (MMSE) criterion and stereo data is presented. They include multi-environment model-based linear normalization (MEMLIN), polynomial MEMLIN (P-MEMLIN), multi-environment model-based histogram normalization (MEMHIN), and phoneme-dependent MEMLIN (PD-MEMLIN). Those methods model clean and noisy feature vector spaces using Gaussian mixture models (GMMs). The objective of the methods is to learn a transformation between clean and noisy feature vectors associated with each pair of clean and noisy model Gaussians. The direct approach to learn the transformation is by using stereo data; that is, noisy feature vectors and the corresponding clean feature vectors. In this paper, however, a nonstereo data based training procedure, is presented. The transformations can be modeled just like a bias vector (MEMLIN), or by using a first-order polynomial (P-MEMLIN) or a nonlinear function based on histogram equalization (MEMHIN). Further improvements are obtained by using phoneme-dependent bias vector transformation (PD-MEMLIN). In PD-MEMLIN, the clean and noisy feature vector spaces are split into several phonemes, and each of them is modeled as a GMM. Those methods achieve significant word error rate improvements over others that are based on similar targets. The experimental results using the SpeechDat Car database show an average improvement in word error rate greater than 68% in all cases compared to the baseline when using the original clean acoustic models, and up to 83% when training acoustic models on the new normalized feature space

This paper presents a two-channel speech reinforcement system for cars able to improve the communication between the front and the rear passengers. One of the problems of this kind of systems is that they must operate in closed-loop, as acoustic feedback paths appear due to the short distance between loudspeakers and microphones. This feedback paths can make the system become unstable and acoustic echo control is needed in order to ensure stability. The system must perform two plant identifications for each channel. One of them is an open-loop identification and the other one is closed-loop. We propose here the use of echo suppression filters specially designed for closed-loop subsystems along with echo suppression filters for open-loop subsystems based on the optimal filtering theory. Results about the performance of the proposed system are provided

This paper presents a novel acoustic modeling framework that naturally extends the Hidden Markov Model (HMM) approach. The novel models reduce the errors caused by speaker variability by means of a local spectral mismatch reduction. A more complex and flexible speech production scheme can be assumed, in which the local temporal and frequency elastic deformations of the speech are captured by the model. In the new framework the states of a standard HMM, which are usually associated with temporal transitions, are expanded so that a new degree of freedom for the model is provided and it is then possible to estimate an optimum frequency warping factor at the same time as the decoder finds the best state sequence. In the local spectral warping based models the states become time-frequency related states and the number of parameters of the model is comparable to the standard HMM since they share a certain amount of parameters as it will be shown. The novel models are evaluated in the noise-free TIDIGITS corpus, which includes connected digits uttered by male, female and children. It has been found that, under speaker group (age-gender) mismatch conditions, the local frequency warping reduced Word Error Rate (WER) in mean by a 70%, using the initial models. When matched speaker group conditions were tested the error was reduced in mean in a 9.7% after reestimating the models. Index Terms: speaker variability, local frequency warping.

In this work, we study the variations in the time and frequency domains inside a Spanish language corpus of speakers with nonpathological and pathological speech. We show how pathological speech has a greater variability in the duration of the words than non-pathological speech, while in the frequency domain we show that the vowels confusability increases by a 18%. The baseline experiments in Automatic Speech Recognition (ASR) with this corpus demonstrate that this variability causes a loss in the performance of ASR systems. To reduce the impact of time and frequency variability we use a recent Vocal Tract Length Normalization (VTLN) system: MATE (augMented stAte space acousTic modEl), as a way of improving the performance of ASR systems when dealing with speakers who suffer any kind of speech pathology. Experiments with MATE show a 17.04 % and 11.19 % WER reduction by using frequency and time MATE respectively. 1.

In this paper a Phoneme-Dependent Multi-Environment Models based LInear feature Normalization, PD-MEMLIN, is presented. The target of this algorithm is to learn the difference between clean and noisy feature vectors associated to a pair of gaussians of the same phoneme (one for a clean model, and the other one for a noisy model), for each basic defined environment. These differences are estimated in a previous training process with stereo data. In order to compensate some of the problems of the independence assumption of the feature vectors components and the mismatch error between perfect and proposed transformations, two approaches have been proposed too: a multi-environment rotation transformation algorithm, and the use of transformed space acoustic models. Some experiments with SpeechDat Car database were carried out in order to study the behavior of the proposed techniques in a real acoustic environment. The experimental results show an average improvement of more than 77% using PD-MEMLIN, and more than 85% using transformed space acoustic models and multienvironment rotation transformation, concerning the baseline.

This paper presents a decoding method for automatic speech recognition (ASR) that reduces the impact of local spectral and temporal variabilities on ASR performance. The procedure involves augmenting the standard Viterbi search for an optimum state sequence with a locally constrained search for optimum degrees of spectral warping or temporal warping applied to individual analysis frames. It is argued in the paper that this represents an efficient and effective method for compensating for local variability in speech which may have potential application to a broader array of speech transformations. The techniques are presented in the context of existing methods for frequency warping based speaker normalization and existing methods for computation of dynamic features for ASR. The modified decoding algorithms were evaluated in both clean and noisy task domains using

A multi-environment adaptation technique, based on minimum mean squared error estimation, is proposed. MEMLIN (multi-environment models based linear normalization) consists of a feature adaptation using stereo data and several basic defined environments. The target of this algorithm is to learn the difference between clean and noisy feature vectors associated to a pair of Gaussians (one for a clean model, and the other for a noisy model), for each basic environment. This knowledge, the associated Gaussians, the conditional probability between clean and noisy Gaussians, and the environment are the data used to compensate the mismatch between clean and noisy vectors. This algorithm obtains important improvements regarding other techniques that look for similar targets. The experimental results with the SpeechDat Car database shows an average improvement of more than 68%, concerning the baseline, over 7 different defined environments.