No Need for a Lexicon? Evaluating the Value of the Pronunciation Lexica in End-to-End Models

For decades, context-dependent phonemes have been the dominant sub-word unit for conventional acoustic modeling systems. This status quo has begun to be challenged recently by end-to-end models which seek to combine acoustic, pronunciation, and language model components into a single neural network. Such systems, which typically predict graphemes or words, simplify the recognition process since they remove the need for a separate expert-curated pronunciation lexicon to map from phoneme-based units to words. However, there has been little previous work comparing phoneme-based versus grapheme-based sub-word units in the end-to-end modeling framework, to determine whether the gains from such approaches are primarily due to the new probabilistic model, or from the joint learning of the various components with grapheme-based units. In this work, we conduct detailed experiments which are aimed at quantifying the value of phoneme-based pronunciation lexica in the context of end-to-end models. We examine phoneme-based end-to-end models, which are contrasted against grapheme-based ones on a large vocabulary English Voice-search task, where we find that graphemes do indeed outperform phonemes. We also compare grapheme and phoneme-based approaches on a multi-dialect English task, which once again confirm the superiority of graphemes, greatly simplifying the system for recognizing multiple dialects

Statistical natural language generation for dialogue systems based on hierarchical models

Due to the increasing presence of natural-language interfaces in our life, natural language processing (NLP) is currently gaining more popularity every year. However, until recently, the main part of the research activity in this area was aimed to Natural Language Understanding (NLU), which is responsible for extracting meanings from natural language input. This is explained by a wider number of practical applications of NLU such as machine translation, etc., whereas Natural Language Generation is mainly used for providing output interfaces, which was considered more as a user interface problem rather than a functionality issue. Generally speaking, natural language generation (NLG) is the process of generating text from a semantic representation, which can be expressed in many different forms. The common application of NLG takes part in so called Spoken Dialogue System (SDS), where user interacts directly by voice with a computer- based system to receive information or perform a certain type of actions as, for example, buying a plane ticket or booking a table in a restaurant. Dialogue systems represent one of the most interesting applications within the field of speech technologies. Usually the NLG part in this kind of systems was provided by templates, only filling canned gaps with requested information. But nowadays, since SDS are increasing its complexity, more advanced and user-friendly interfaces should be provided, thereby creating a need for a more refined and adaptive approach. One of the solutions to be considered are the NLG models based on statistical frameworks, where the system’s response to user is generated in real-time, adjusting their response to the user performance, instead of just choosing a pertinent template. Due to the corpus-based approach, these systems are easy to adapt to the different tasks in a range of informational domain. The aim of this work is to present a statistical approach to the problem of utterance generation, which uses cooperation between two different language models (LM) in order to enhance the efficiency of NLG module. In the higher level, a class- based language model is used to build the syntactic structure of the sentence. Inthe second layer, a specific language model acts inside each class, dealing with the words. In the dialogue system described in this work, a user asks for an information regarding to a bus schedule, route schemes, fares and special information. Therefore in each dialogue the user has a specific dialogue goal, which needs to be met by the system. This could be used as one of the methods to measure the system performance, as well as the appropriate utterance generation and average dialogue length, which is important when speaking about an interactive information system. The work is organized as follows. In Section 2 the basic approaches to the NLG task are described, and their advantages and disadvantages are considered. Section 3 presents the objective of this work. In Section 4 the basic model and its novelty is explained. In Section 5 the details of the task features and the corpora employed are presented. Section 6 contains the experiments results and its explanation, as well as the evaluation of the obtained results. The Section 7 resumes the conclusions and the future investigation proposals

Compositional Morphology for Word Representations and Language Modelling

This paper presents a scalable method for integrating compositional morphological representations into a vector-based probabilistic language model. Our approach is evaluated in the context of log-bilinear language models, rendered suitably efficient for implementation inside a machine translation decoder by factoring the vocabulary. We perform both intrinsic and extrinsic evaluations, presenting results on a range of languages which demonstrate that our model learns morphological representations that both perform well on word similarity tasks and lead to substantial reductions in perplexity. When used for translation into morphologically rich languages with large vocabularies, our models obtain improvements of up to 1.2 BLEU points relative to a baseline system using back-off n-gram models.Comment: Proceedings of the 31st International Conference on Machine Learning (ICML

Apprentissage discriminant des modèles continus en traduction automatique

Over the past few years, neural network (NN) architectures have been successfully applied to many Natural Language Processing (NLP) applications, such as Automatic Speech Recognition (ASR) and Statistical Machine Translation (SMT).For the language modeling task, these models consider linguistic units (i.e words and phrases) through their projections into a continuous (multi-dimensional) space, and the estimated distribution is a function of these projections. Also qualified continuous-space models (CSMs), their peculiarity hence lies in this exploitation of a continuous representation that can be seen as an attempt to address the sparsity issue of the conventional discrete models. In the context of SMT, these echniques have been applied on neural network-based language models (NNLMs) included in SMT systems, and oncontinuous-space translation models (CSTMs). These models have led to significant and consistent gains in the SMT performance, but are also considered as very expensive in training and inference, especially for systems involving large vocabularies. To overcome this issue, Structured Output Layer (SOUL) and Noise Contrastive Estimation (NCE) have been proposed; the former modifies the standard structure on vocabulary words, while the latter approximates the maximum-likelihood estimation (MLE) by a sampling method. All these approaches share the same estimation criterion which is the MLE ; however using this procedure results in an inconsistency between theobjective function defined for parameter stimation and the way models are used in the SMT application. The work presented in this dissertation aims to design new performance-oriented and global training procedures for CSMs to overcome these issues. The main contributions lie in the investigation and evaluation of efficient training methods for (large-vocabulary) CSMs which aim~:(a) to reduce the total training cost, and (b) to improve the efficiency of these models when used within the SMT application. On the one hand, the training and inference cost can be reduced (using the SOUL structure or the NCE algorithm), or by reducing the number of iterations via a faster convergence. This thesis provides an empirical analysis of these solutions on different large-scale SMT tasks. On the other hand, we propose a discriminative training framework which optimizes the performance of the whole system containing the CSM as a component model. The experimental results show that this framework is efficient to both train and adapt CSM within SMT systems, opening promising research perspectives.Durant ces dernières années, les architectures de réseaux de neurones (RN) ont été appliquées avec succès à de nombreuses applications en Traitement Automatique de Langues (TAL), comme par exemple en Reconnaissance Automatique de la Parole (RAP) ainsi qu'en Traduction Automatique (TA).Pour la tâche de modélisation statique de la langue, ces modèles considèrent les unités linguistiques (c'est-à-dire des mots et des segments) à travers leurs projections dans un espace continu (multi-dimensionnel), et la distribution de probabilité à estimer est une fonction de ces projections.Ainsi connus sous le nom de "modèles continus" (MC), la particularité de ces derniers se trouve dans l'exploitation de la représentation continue qui peut être considérée comme une solution au problème de données creuses rencontré lors de l'utilisation des modèles discrets conventionnels.Dans le cadre de la TA, ces techniques ont été appliquées dans les modèles de langue neuronaux (MLN) utilisés dans les systèmes de TA, et dans les modèles continus de traduction (MCT).L'utilisation de ces modèles se sont traduit par d'importantes et significatives améliorations des performances des systèmes de TA. Ils sont néanmoins très coûteux lors des phrases d'apprentissage et d'inférence, notamment pour les systèmes ayant un grand vocabulaire.Afin de surmonter ce problème, l'architecture SOUL (pour "Structured Output Layer" en anglais) et l'algorithme NCE (pour "Noise Contrastive Estimation", ou l'estimation contrastive bruitée) ont été proposés: le premier modifie la structure standard de la couche de sortie, alors que le second cherche à approximer l'estimation du maximum de vraisemblance (MV) par une méthode d’échantillonnage.Toutes ces approches partagent le même critère d'estimation qui est la log-vraisemblance; pourtant son utilisation mène à une incohérence entre la fonction objectif définie pour l'estimation des modèles, et la manière dont ces modèles seront utilisés dans les systèmes de TA.Cette dissertation vise à concevoir de nouvelles procédures d'entraînement des MC, afin de surmonter ces problèmes.Les contributions principales se trouvent dans l'investigation et l'évaluation des méthodes d'entraînement efficaces pour MC qui visent à: (i) réduire le temps total de l'entraînement, et (ii) améliorer l'efficacité de ces modèles lors de leur utilisation dans les systèmes de TA.D'un côté, le coût d'entraînement et d'inférence peut être réduit (en utilisant l'architecture SOUL ou l'algorithme NCE), ou la convergence peut être accélérée.La dissertation présente une analyse empirique de ces approches pour des tâches de traduction automatique à grande échelle.D'un autre côté, nous proposons un cadre d'apprentissage discriminant qui optimise la performance du système entier ayant incorporé un modèle continu.Les résultats expérimentaux montrent que ce cadre d'entraînement est efficace pour l'apprentissage ainsi que pour l'adaptation des MC au sein des systèmes de TA, ce qui ouvre de nouvelles perspectives prometteuses

Deep Transfer Learning for Automatic Speech Recognition: Towards Better Generalization

Automatic speech recognition (ASR) has recently become an important challenge when using deep learning (DL). It requires large-scale training datasets and high computational and storage resources. Moreover, DL techniques and machine learning (ML) approaches in general, hypothesize that training and testing data come from the same domain, with the same input feature space and data distribution characteristics. This assumption, however, is not applicable in some real-world artificial intelligence (AI) applications. Moreover, there are situations where gathering real data is challenging, expensive, or rarely occurring, which can not meet the data requirements of DL models. deep transfer learning (DTL) has been introduced to overcome these issues, which helps develop high-performing models using real datasets that are small or slightly different but related to the training data. This paper presents a comprehensive survey of DTL-based ASR frameworks to shed light on the latest developments and helps academics and professionals understand current challenges. Specifically, after presenting the DTL background, a well-designed taxonomy is adopted to inform the state-of-the-art. A critical analysis is then conducted to identify the limitations and advantages of each framework. Moving on, a comparative study is introduced to highlight the current challenges before deriving opportunities for future research

Evaluation of innovative computer-assisted transcription and translation strategies for video lecture repositories

Nowadays, the technology enhanced learning area has experienced a strong growth with many new learning approaches like blended learning, flip teaching, massive open online courses, and open educational resources to complement face-to-face lectures. Specifically, video lectures are fast becoming an everyday educational resource in higher education for all of these new learning approaches, and they are being incorporated into existing university curricula around the world. Transcriptions and translations can improve the utility of these audiovisual assets, but rarely are present due to a lack of cost-effective solutions to do so. Lecture searchability, accessibility to people with impairments, translatability for foreign students, plagiarism detection, content recommendation, note-taking, and discovery of content-related videos are examples of advantages of the presence of transcriptions. For this reason, the aim of this thesis is to test in real-life case studies ways to obtain multilingual captions for video lectures in a cost-effective way by using state-of-the-art automatic speech recognition and machine translation techniques. Also, we explore interaction protocols to review these automatic transcriptions and translations, because unfortunately automatic subtitles are not error-free. In addition, we take a step further into multilingualism by extending our findings and evaluation to several languages. Finally, the outcomes of this thesis have been applied to thousands of video lectures in European universities and institutions.Hoy en día, el área del aprendizaje mejorado por la tecnología ha experimentado un fuerte crecimiento con muchos nuevos enfoques de aprendizaje como el aprendizaje combinado, la clase inversa, los cursos masivos abiertos en línea, y nuevos recursos educativos abiertos para complementar las clases presenciales. En concreto, los videos docentes se están convirtiendo rápidamente en un recurso educativo cotidiano en la educación superior para todos estos nuevos enfoques de aprendizaje, y se están incorporando a los planes de estudios universitarios existentes en todo el mundo. Las transcripciones y las traducciones pueden mejorar la utilidad de estos recursos audiovisuales, pero rara vez están presentes debido a la falta de soluciones rentables para hacerlo. La búsqueda de y en los videos, la accesibilidad a personas con impedimentos, la traducción para estudiantes extranjeros, la detección de plagios, la recomendación de contenido, la toma de notas y el descubrimiento de videos relacionados son ejemplos de las ventajas de la presencia de transcripciones. Por esta razón, el objetivo de esta tesis es probar en casos de estudio de la vida real las formas de obtener subtítulos multilingües para videos docentes de una manera rentable, mediante el uso de técnicas avanzadas de reconocimiento automático de voz y de traducción automática. Además, exploramos diferentes modelos de interacción para revisar estas transcripciones y traducciones automáticas, pues desafortunadamente los subtítulos automáticos no están libres de errores. Además, damos un paso más en el multilingüismo extendiendo nuestros hallazgos y evaluaciones a muchos idiomas. Por último, destacar que los resultados de esta tesis se han aplicado a miles de vídeos docentes en universidades e instituciones europeas.Hui en dia, l'àrea d'aprenentatge millorat per la tecnologia ha experimentat un fort creixement, amb molts nous enfocaments d'aprenentatge com l'aprenentatge combinat, la classe inversa, els cursos massius oberts en línia i nous recursos educatius oberts per tal de complementar les classes presencials. En concret, els vídeos docents s'estan convertint ràpidament en un recurs educatiu quotidià en l'educació superior per a tots aquests nous enfocaments d'aprenentatge i estan incorporant-se als plans d'estudi universitari existents arreu del món. Les transcripcions i les traduccions poden millorar la utilitat d'aquests recursos audiovisuals, però rara vegada estan presents a causa de la falta de solucions rendibles per fer-ho. La cerca de i als vídeos, l'accessibilitat a persones amb impediments, la traducció per estudiants estrangers, la detecció de plagi, la recomanació de contingut, la presa de notes i el descobriment de vídeos relacionats són un exemple dels avantatges de la presència de transcripcions. Per aquesta raó, l'objectiu d'aquesta tesi és provar en casos d'estudi de la vida real les formes d'obtenir subtítols multilingües per a vídeos docents d'una manera rendible, mitjançant l'ús de tècniques avançades de reconeixement automàtic de veu i de traducció automàtica. A més a més, s'exploren diferents models d'interacció per a revisar aquestes transcripcions i traduccions automàtiques, puix malauradament els subtítols automàtics no estan lliures d'errades. A més, es fa un pas més en el multilingüisme estenent els nostres descobriments i avaluacions a molts idiomes. Per últim, destacar que els resultats d'aquesta tesi s'han aplicat a milers de vídeos docents en universitats i institucions europees.Valor Miró, JD. (2017). Evaluation of innovative computer-assisted transcription and translation strategies for video lecture repositories [Tesis doctoral no publicada]. Universitat Politècnica de València. https://doi.org/10.4995/Thesis/10251/90496TESI

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

Integrating Language Identification to improve Multilingual Speech Recognition

The process of determining the language of a speech utterance is called Language Identification (LID). This task can be very challenging as it has to take into account various language-specific aspects, such as phonetic, phonotactic, vocabulary and grammar-related cues. In multilingual speech recognition we try to find the most likely word sequence that corresponds to an utterance where the language is not known a priori. This is a considerably harder task compared to monolingual speech recognition and it is common to use LID to estimate the current language. In this project we present two general approaches for LID and describe how to integrate them into multilingual speech recognizers. The first approach uses hierarchical multilayer perceptrons to estimate language posterior probabilities given the acoustics in combination with hidden Markov models. The second approach evaluates the output of a multilingual speech recognizer to determine the spoken language. The research is applied to the MediaParl speech corpus that was recorded at the Parliament of the canton of Valais, where people switch from Swiss French to Swiss German or vice versa. Our experiments show that, on that particular data set, LID can be used to significantly improve the performance of multilingual speech recognizers. We will also point out that ASR dependent LID approaches yield the best performance due to higher-level cues and that our systems perform much worse on non-native dat

Spoken dialogue systems: architectures and applications

171 p.Technology and technological devices have become habitual and omnipresent. Humans need to learn tocommunicate with all kind of devices. Until recently humans needed to learn how the devices expressthemselves to communicate with them. But in recent times the tendency has become to makecommunication with these devices in more intuitive ways. The ideal way to communicate with deviceswould be the natural way of communication between humans, the speech. Humans have long beeninvestigating and designing systems that use this type of communication, giving rise to the so-calledSpoken Dialogue Systems.In this context, the primary goal of the thesis is to show how these systems can be implemented.Additionally, the thesis serves as a review of the state-of-the-art regarding architectures and toolkits.Finally, the thesis is intended to serve future system developers as a guide for their construction. For that