Syllable classification using static matrices and prosodic features

In this paper we explore the usefulness of prosodic features for syllable classification. In order to do this, we represent the syllable as a static analysis unit such that its acoustic-temporal dynamics could be merged into a set of features that the SVM classifier will consider as a whole. In the first part of our experiment we used MFCC as features for classification, obtaining a maximum accuracy of 86.66%. The second part of our study tests whether the prosodic information is complementary to the cepstral information for syllable classification. The results obtained show that combining the two types of information does improve the classification, but further analysis is necessary for a more successful combination of the two types of features

Phonetic Searching

An improved method and apparatus is disclosed which uses probabilistic techniques to map an input search string with a prestored audio file, and recognize certain portions of a search string phonetically. An improved interface is disclosed which permits users to input search strings, linguistics, phonetics, or a combination of both, and also allows logic functions to be specified by indicating how far separated specific phonemes are in time.Georgia Tech Research Corporatio

Discriminative and generative approaches for long- and short-term speaker characteristics modeling : application to speaker verification

The speaker verification problem can be stated as follows: given two speech recordings, determine whether or not they have been uttered by the same speaker. Most current speaker verification systems are based on Gaussian mixture models. This probabilistic representation allows to adequately model the complex distribution of the underlying speech feature parameters. It however represents an inadequate basis for discriminating between speakers, which is the key issue in the area of speaker verification. In the first part of this thesis, we attempt to overcome these difficulties by proposing to combine support vector machines, a well established discriminative modeling, with two generative approaches based on Gaussian mixture models. In the first generative approach, a target speaker is represented by a Gaussian mixture model corresponding to a Maximum A Posteriori adaptation of a large Gaussian mixture model, coined universal background model, to the target speaker data. The second generative approach is the Joint Factor Analysis that has become the state-of-the-art in the field of speaker verification during the last three years. The advantage of this technique is that it provides a framework of powerful tools for modeling the inter-speaker and channel variabilities. We propose and test several kernel functions that are integrated in the design of both previous combinations. The best results are obtained when the support vector machines are applied within a new space called the "total variability space", defined using the factor analysis. In this novel modeling approach, the channel effect is treated through a combination of linear discnminant analysis and kemel normalization based on the inverse of the within covariance matrix of the speaker. In the second part of this thesis, we present a new approach to modeling the speaker's longterm prosodic and spectral characteristics. This novel approach is based on continuous approximations of the prosodic and cepstral contours contained in a pseudo-syllabic segment of speech. Each of these contours is fitted to a Legendre polynomial, whose coefficients are modeled by a Gaussian mixture model. The joint factor analysis is used to treat the speaker and channel variabilities. Finally, we perform a scores fusion between systems based on long-term speaker characteristics with those described above that use short-term speaker features

Amélioration de la robustesse des systèmes de reconnaissance automatique du locuteur dans l'espace des i-vecteurs

Les systèmes nec plus ultra de reconnaissance du locuteur adoptent la représentation de la parole dans l’espace des i-vecteurs. Un i-vecteur n’est qu’un simple vecteur de faible dimension (typiquement dans les centaines) représentant une vaste gamme d’information véhiculée par le signal vocal. Bien que les performances de ces systèmes en matière des taux de reconnaissance aient atteint un niveau très avancé, une meilleure exploitation de ces systèmes dans les milieux réels de tous les jours nécessite encore plus d'efforts de la part des chercheurs en la matière. Dans le cadre de cette thèse, notre objectif principal est d'améliorer la robustesse des systèmes de reconnaissance du locuteur opérant dans l’espace des ivecteurs. Dans la première partie de ce travail, nous nous intéressons à la tâche de la vérification du locuteur. Nous nous focalisons plus particulièrement sur la conception d’un système de vérification à la fois indépendant du type du canal de transmission/enregistrement et du genre du locuteur. Dans le contexte des i-vecteurs, les classificateurs génératifs, tels que l’analyse discriminante linéaire probabiliste (PLDA), ont dominé le domaine de la reconnaissance du locuteur. Néanmoins, de simples classificateurs à base de la similarité angulaire du cosinus (SAC) restent concurrentiels. Ainsi, nous avons proposé dans cette partie deux solutions rendant respectivement les systèmes à base des deux classificateurs de l’état de l’art (le PLDA et la SAC) indépendants du type du canal et du genre du locuteur. En effet, nos systèmes conçus de la sorte sont considérés comme les deux premiers systèmes de vérification du locuteur atteignant les résultats de l’état de l’art (environ 2 % d’EER pour la parole téléphonique et 3 % pour la parole microphonique) sans pour autant profiter ni de l’information concernant le type du canal ni de celle concernant le genre du locuteur. Le regroupement en locuteurs est une autre tâche de la reconnaissance du locuteur qui représente notre centre d’intérêt dans la seconde partie de cette thèse. À nouveau, nos recherches seront menées uniquement dans le contexte de la représentation de la parole par des i-vecteurs. À vrai dire, il existe deux types d’applications à base du regroupement en locuteurs, soit, le regroupement en locuteurs des grands corpora des fichiers vocaux (speaker clustering) et la structuration en tours de parole d’un flux audio (speaker diarization). Une nouvelle version de l’algorithme non paramétrique de décalage de la moyenne (Mean Shift, MS) a été proposée afin de faire face au problème du regroupement en locuteurs. Nous avons démontré que les performances de notre nouvelle version de l’algorithme de MS à base de la distance angulaire du cosinus dépassent ceux de la version de base, une fois testés face à la tâche du regroupement en locuteurs. Le même algorithme nous a permis d’obtenir les résultats de l’état de l’art (DER égal à 12,4 %) de la structuration en tours de parole du corpus des données téléphoniques CallHome

Multi-level acoustic modeling for automatic speech recognition

Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2012.Cataloged from PDF version of thesis.Includes bibliographical references (p. 183-192).Context-dependent acoustic modeling is commonly used in large-vocabulary Automatic Speech Recognition (ASR) systems as a way to model coarticulatory variations that occur during speech production. Typically, the local phoneme context is used as a means to define context-dependent units. Because the number of possible context-dependent units can grow exponentially with the length of the contexts, many units will not have enough training examples to train a robust model, resulting in a data sparsity problem. For nearly two decades, this data sparsity problem has been dealt with by a clustering-based framework which systematically groups different context-dependent units into clusters such that each cluster can have enough data. Although dealing with the data sparsity issue, the clustering-based approach also makes all context-dependent units within a cluster have the same acoustic score, resulting in a quantization effect that can potentially limit the performance of the context-dependent model. In this work, a multi-level acoustic modeling framework is proposed to address both the data sparsity problem and the quantization effect. Under the multi-level framework, each context-dependent unit is associated with classifiers that target multiple levels of contextual resolution, and the outputs of the classifiers are linearly combined for scoring during recognition. By choosing the classifiers judiciously, both the data sparsity problem and the quantization effect can be dealt with. The proposed multi-level framework can also be integrated into existing large-vocabulary ASR systems, such as FST-based ASR systems, and is compatible with state-of-the-art error reduction techniques for ASR systems, such as discriminative training methods. Multiple sets of experiments have been conducted to compare the performance of the clustering-based acoustic model and the proposed multi-level model. In a phonetic recognition experiment on TIMIT, the multi-level model has about 8% relative improvement in terms of phone error rate, showing that the multi-level framework can help improve phonetic prediction accuracy. In a large-vocabulary transcription task, combining the proposed multi-level modeling framework with discriminative training can provide more than 20% relative improvement over a clustering baseline model in terms of Word Error Rate (WER), showing that the multi-level framework can be integrated into existing large-vocabulary decoding frameworks and that it combines well with discriminative training methods. In speaker adaptive transcription task, the multi-level model has about 14% relative WER improvement, showing that the proposed framework can adapt better to new speakers, and potentially to new environments than the conventional clustering-based approach.by Hung-An Chang.Ph.D

An artificial Intelligence Approach to improving Speech Recognition

Speech Recognition is a technology with promising applications. However, the performance of current speech recognizers greatly limit their widespread use. Approaches to reducing the word error rate have mainly been associated with statistical techniques. As a consequence, speech recognition results can still contain sentences that are nonsensical. The method proposed here, is to analize the output of any chosen speech recognition system, in order to determine whether a sentence contains syntactic or semantic errors. This is done via a software agent that uses the information from its knowledge base to attempt to correct the errors found. A system was implemented with a small vocabulary speaker-independent continuous speech recognition system, with limited sentence structures. The achieved increase in speech recognition accuracy, shows that there are bene ts in using this approach

Changing the way the world thinks about computer security.

Small changes in an established system can result in larger changes in the overall system (e.g. network effects, émergence, criticality, broken Windows theory). However, in an immature discipline, such as computer security, such changes can be difficult to envision and even more difficult to amplement, as the immature discipline is likely to lack the scientific framework that would allow for the introduction of even minute changes. (Cairns, P. and Thimbleby, H, 2003) describe three of the signs of an immature discipline as postulated by (Kuhn, 1970): a. squabbles over what are legitimate tools for research b. disagreement over which phenomenon are legitimate to study, and c. inability to scope the domain of study. The research presented in this document demonstrates how the computer security field, at the time this research began, was the embodiment of thèse characteristics. It presents a cohesive analysis of the intentional introduction of a séries of small changes chosen to aid in maturation of the discipline. Summarily, it builds upon existing theory, exploring the combined effect of coordinated and strategie changes in an immature system and establishing a scientific framework by which the impact of the changes can be quantified. By critically examining the nature of the computer security system overall, this work establishes the need for both increased scientific rigor, and a multidisciplinary approach to the global computer security problem. In order for these changes to take place, many common assumptions related to computer security had to be questioned. However, as the discipline was immature, and controlled by relatively few entities, questioning the status quo was not without difficulties. However, in order for the discipline to mature, more feedback into the overall computer security (and in particular, the computer malware/virus) system was needed, requiring a shift from a mostly closed system to one that was forced to undergo greater scrutiny from various other communities. The input from these communities resulted in long-term changes and increased maturation of the system. Figure 1 illustrates the specific areas in which the research presented herein addressed these needs, provides an overview of the research context, and outlines the specific impact of the research, specifically the development of new and significant scientific paradigms within the discipline