Modularity and Neural Integration in Large-Vocabulary Continuous Speech Recognition

This Thesis tackles the problems of modularity in Large-Vocabulary Continuous Speech Recognition with use of Neural Network

Subspace and graph methods to leverage auxiliary data for limited target data multi-class classification, applied to speaker verification

Thesis (Ph. D.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2011.Cataloged from PDF version of thesis.Includes bibliographical references (p. 127-130).Multi-class classification can be adversely affected by the absence of sufficient target (in-class) instances for training. Such cases arise in face recognition, speaker verification, and document classification, among others. Auxiliary data-sets, which contain a diverse sampling of non-target instances, are leveraged in this thesis using subspace and graph methods to improve classification where target data is limited. The auxiliary data is used to define a compact representation that maps instances into a vector space where inner products quantify class similarity. Within this space, an estimate of the subspace that constitutes within-class variability (e.g. the recording channel in speaker verification or the illumination conditions in face recognition) can be obtained using class-labeled auxiliary data. This thesis proposes a way to incorporate this estimate into the SVM framework to perform nuisance compensation, thus improving classification performance. Another contribution is a framework that combines mapping and compensation into a single linear comparison, which motivates computationally inexpensive and accurate comparison functions. A key aspect of the work takes advantage of efficient pairwise comparisons between the training, test, and auxiliary instances to characterize their interaction within the vector space, and exploits it for improved classification in three ways. The first uses the local variability around the train and test instances to reduce false-alarms. The second assumes the instances lie on a low-dimensional manifold and uses the distances along the manifold. The third extracts relational features from a similarity graph where nodes correspond to the training, test and auxiliary instances. To quantify the merit of the proposed techniques, results of experiments in speaker verification are presented where only a single target recording is provided to train the classifier. Experiments are preformed on standard NIST corpora and methods are compared using standard evalutation metrics: detection error trade-off curves, minimum decision costs, and equal error rates.by Zahi Nadim Karam.Ph.D

Phonetic aware techniques for Speaker Verification

The goal of this thesis is to improve current state-of-the-art techniques in speaker verification (SV), typically based on âidentity-vectorsâ (i-vectors) and deep neural network (DNN), by exploiting diverse (phonetic) information extracted using various techniques such as automatic speech recognition (ASR). Different speakers span different subspaces within a universal acoustic space, usually modelled by âuniversal background modelâ. The speaker-specific subspace depends on the speakerâs voice characteristics, but also on the verbalised text of a speaker. In current state-of-the-art SV systems, i-vectors are extracted by applying a factor analysis technique to obtain low dimensional speaker-specific representation. Furthermore, DNN output is also employed in a conventional i-vector framework to model phonetic information embedded in the speech signal. This thesis proposes various techniques to exploit phonetic knowledge of speech to further enrich speaker characteristics. More specifically, the techniques proposed in this thesis are applied to various SV tasks, namely, text-independent and text-dependent SV. For text-independent SV task, several ASR systems are developed and applied to compute phonetic posterior probabilities, subsequently exploited to enhance the speaker-specific information included in i-vectors. These approaches are then extended for text-dependent SV task, exploiting temporal information in a principled way, i.e., by using dynamic time warping applied on speaker informative vectors. Finally, as opposed to train DNN with phonetic information, DNN is trained in an end-to-end fashion to directly discriminate speakers. The baseline end-to-end SV approach consists of mapping a variable length speech segment to a fixed dimensional speaker vector by estimating the mean of hidden representations in DNN structure. We improve upon this technique by computing a distance function between two utterances which takes into account common phonetic units. The whole network is optimized by employing a triplet-loss objective function. The proposed approaches are evaluated on commonly used datasets such as NIST SRE 2010 and RSR2015. Significant improvements are observed over the baseline systems on both the text-dependent and text-independent SV tasks by applying phonetic knowledge

Amélioration des systèmes de traduction par analyse linguistique et thématique (Application à la traduction depuis l'arabe)

La traduction automatique des documents est considérée comme l une des tâches les plus difficiles en traitement automatique des langues et de la parole. Les particularités linguistiques de certaines langues, comme la langue arabe, rendent la tâche de traduction automatique plus difficile. Notre objectif dans cette thèse est d'améliorer les systèmes de traduction de l'arabe vers le français et vers l'anglais. Nous proposons donc une étude détaillée sur ces systèmes. Les principales recherches portent à la fois sur la construction de corpus parallèles, le prétraitement de l'arabe et sur l'adaptation des modèles de traduction et de langue.Tout d'abord, un corpus comparable journalistique a été exploré pour en extraire automatiquement un corpus parallèle. Ensuite, différentes approches d adaptation du modèle de traduction sont exploitées, soit en utilisant le corpus parallèle extrait automatiquement soit en utilisant un corpus parallèle construit automatiquement.Nous démontrons que l'adaptation des données du système de traduction permet d'améliorer la traduction. Un texte en arabe doit être prétraité avant de le traduire et ceci à cause du caractère agglutinatif de la langue arabe. Nous présentons notre outil de segmentation de l'arabe, SAPA (Segmentor and Part-of-speech tagger for Arabic), indépendant de toute ressource externe et permettant de réduire les temps de calcul. Cet outil permet de prédire simultanément l étiquette morpho-syntaxique ainsi que les proclitiques (conjonctions, prépositions, etc.) pour chaque mot, ensuite de séparer les proclitiques du lemme (ou mot de base). Nous décrivons également dans cette thèse notre outil de détection des entités nommées, NERAr (Named Entity Recognition for Arabic), et nous examions l'impact de l'intégration de la détection des entités nommées dans la tâche de prétraitement et la pré-traduction de ces entités nommées en utilisant des dictionnaires bilingues. Nous présentons par la suite plusieurs méthodes pour l'adaptation thématique des modèles de traduction et de langue expérimentées sur une application réelle contenant un corpus constitué d un ensemble de phrases multicatégoriques.Ces expériences ouvrent des perspectives importantes de recherche comme par exemple la combinaison de plusieurs systèmes lors de la traduction pour l'adaptation thématique. Il serait également intéressant d'effectuer une adaptation temporelle des modèles de traduction et de langue. Finalement, les systèmes de traduction améliorés arabe-français et arabe-anglais sont intégrés dans une plateforme d'analyse multimédia et montrent une amélioration des performances par rapport aux systèmes de traduction de base.Machine Translation is one of the most difficult tasks in natural language and speech processing. The linguistic peculiarities of some languages makes the machine translation task more difficult. In this thesis, we present a detailed study of machine translation systems from arabic to french and to english.Our principle researches carry on building parallel corpora, arabic preprocessing and adapting translation and language models. We propose a method for automatic extraction of parallel news corpora from a comparable corpora. Two approaches for translation model adaptation are explored using whether parallel corpora extracted automatically or parallel corpora constructed automatically. We demonstrate that adapting data used to build machine translation system improves translation.Arabic texts have to be preprocessed before machine translation and this because of the agglutinative character of arabic language. A prepocessing tool for arabic, SAPA (Segmentor and Part-of-speech tagger for Arabic), much faster than the state of the art tools and totally independant of any other external resource was developed. This tool predicts simultaneously morphosyntactic tags and proclitics (conjunctions, prepositions, etc.) for every word, then splits off words into lemma and proclitics.We describe also in this thesis, our named entity recognition tool for arabic, NERAr, and we focus on the impact of integrating named entity recognition in the preprocessing task. We used bilingual dictionaries to propose translations of the detected named entities. We present then many approaches to adapt thematically translation and language models using a corpora consists of a set of multicategoric sentences.These experiments open important research perspectives such as combining many systems when translating. It would be interesting also to focus on a temporal adaptation of translation and language models.Finally, improved machine translation systems from arabic to french and english are integrated in a multimedia platform analysis and shows improvements compared to basic machine translation systems.PARIS11-SCD-Bib. électronique (914719901) / SudocSudocFranceF

Constrained MLLR for speaker recognition

International audienceabstrac