A Bayesian Network View on Acoustic Model-Based Techniques for Robust Speech Recognition

This article provides a unifying Bayesian network view on various approaches for acoustic model adaptation, missing feature, and uncertainty decoding that are well-known in the literature of robust automatic speech recognition. The representatives of these classes can often be deduced from a Bayesian network that extends the conventional hidden Markov models used in speech recognition. These extensions, in turn, can in many cases be motivated from an underlying observation model that relates clean and distorted feature vectors. By converting the observation models into a Bayesian network representation, we formulate the corresponding compensation rules leading to a unified view on known derivations as well as to new formulations for certain approaches. The generic Bayesian perspective provided in this contribution thus highlights structural differences and similarities between the analyzed approaches

Modelo acústico de língua inglesa falada por portugueses

Trabalho de projecto de mestrado em Engenharia Informática, apresentado à Universidade de Lisboa, através da Faculdade de Ciências, 2007No contexto do reconhecimento robusto de fala baseado em modelos de Markov não observáveis (do inglês Hidden Markov Models - HMMs) este trabalho descreve algumas metodologias e experiências tendo em vista o reconhecimento de oradores estrangeiros. Quando falamos em Reconhecimento de Fala falamos obrigatoriamente em Modelos Acústicos também. Os modelos acústicos reflectem a maneira como pronunciamos/articulamos uma língua, modelando a sequência de sons emitidos aquando da fala. Essa modelação assenta em segmentos de fala mínimos, os fones, para os quais existe um conjunto de símbolos/alfabetos que representam a sua pronunciação. É no campo da fonética articulatória e acústica que se estuda a representação desses símbolos, sua articulação e pronunciação. Conseguimos descrever palavras analisando as unidades que as constituem, os fones. Um reconhecedor de fala interpreta o sinal de entrada, a fala, como uma sequência de símbolos codificados. Para isso, o sinal é fragmentado em observações de sensivelmente 10 milissegundos cada, reduzindo assim o factor de análise ao intervalo de tempo onde as características de um segmento de som não variam. Os modelos acústicos dão-nos uma noção sobre a probabilidade de uma determinada observação corresponder a uma determinada entidade. É, portanto, através de modelos sobre as entidades do vocabulário a reconhecer que é possível voltar a juntar esses fragmentos de som. Os modelos desenvolvidos neste trabalho são baseados em HMMs. Chamam-se assim por se fundamentarem nas cadeias de Markov (1856 - 1922): sequências de estados onde cada estado é condicionado pelo seu anterior. Localizando esta abordagem no nosso domínio, há que construir um conjunto de modelos - um para cada classe de sons a reconhecer - que serão treinados por dados de treino. Os dados são ficheiros áudio e respectivas transcrições (ao nível da palavra) de modo a que seja possível decompor essa transcrição em fones e alinhá-la a cada som do ficheiro áudio correspondente. Usando um modelo de estados, onde cada estado representa uma observação ou segmento de fala descrita, os dados vão-se reagrupando de maneira a criar modelos estatísticos, cada vez mais fidedignos, que consistam em representações das entidades da fala de uma determinada língua. O reconhecimento por parte de oradores estrangeiros com pronuncias diferentes da língua para qual o reconhecedor foi concebido, pode ser um grande problema para precisão de um reconhecedor. Esta variação pode ser ainda mais problemática que a variação dialectal de uma determinada língua, isto porque depende do conhecimento que cada orador têm relativamente à língua estrangeira. Usando para uma pequena quantidade áudio de oradores estrangeiros para o treino de novos modelos acústicos, foram efectuadas diversas experiências usando corpora de Portugueses a falar Inglês, de Português Europeu e de Inglês. Inicialmente foi explorado o comportamento, separadamente, dos modelos de Ingleses nativos e Portugueses nativos, quando testados com os corpora de teste (teste com nativos e teste com não nativos). De seguida foi treinado um outro modelo usando em simultâneo como corpus de treino, o áudio de Portugueses a falar Inglês e o de Ingleses nativos. Uma outra experiência levada a cabo teve em conta o uso de técnicas de adaptação, tal como a técnica MLLR, do inglês Maximum Likelihood Linear Regression. Esta última permite a adaptação de uma determinada característica do orador, neste caso o sotaque estrangeiro, a um determinado modelo inicial. Com uma pequena quantidade de dados representando a característica que se quer modelar, esta técnica calcula um conjunto de transformações que serão aplicadas ao modelo que se quer adaptar. Foi também explorado o campo da modelação fonética onde estudou-se como é que o orador estrangeiro pronuncia a língua estrangeira, neste caso um Português a falar Inglês. Este estudo foi feito com a ajuda de um linguista, o qual definiu um conjunto de fones, resultado do mapeamento do inventário de fones do Inglês para o Português, que representam o Inglês falado por Portugueses de um determinado grupo de prestígio. Dada a grande variabilidade de pronúncias teve de se definir este grupo tendo em conta o nível de literacia dos oradores. Este estudo foi posteriormente usado na criação de um novo modelo treinado com os corpora de Portugueses a falar Inglês e de Portugueses nativos. Desta forma representamos um reconhecedor de Português nativo onde o reconhecimento de termos ingleses é possível. Tendo em conta a temática do reconhecimento de fala este projecto focou também a recolha de corpora para português europeu e a compilação de um léxico de Português europeu. Na área de aquisição de corpora o autor esteve envolvido na extracção e preparação dos dados de fala telefónica, para posterior treino de novos modelos acústicos de português europeu. Para compilação do léxico de português europeu usou-se um método incremental semi-automático. Este método consistiu em gerar automaticamente a pronunciação de grupos de 10 mil palavras, sendo cada grupo revisto e corrigido por um linguista. Cada grupo de palavras revistas era posteriormente usado para melhorar as regras de geração automática de pronunciações.The tremendous growth of technology has increased the need of integration of spoken language technologies into our daily applications, providing an easy and natural access to information. These applications are of different nature with different user’s interfaces. Besides voice enabled Internet portals or tourist information systems, automatic speech recognition systems can be used in home user’s experiences where TV and other appliances could be voice controlled, discarding keyboards or mouse interfaces, or in mobile phones and palm-sized computers for a hands-free and eyes-free manipulation. The development of these systems causes several known difficulties. One of them concerns the recognizer accuracy on dealing with non-native speakers with different phonetic pronunciations of a given language. The non-native accent can be more problematic than a dialect variation on the language. This mismatch depends on the individual speaking proficiency and speaker’s mother tongue. Consequently, when the speaker’s native language is not the same as the one that was used to train the recognizer, there is a considerable loss in recognition performance. In this thesis, we examine the problem of non-native speech in a speaker-independent and large-vocabulary recognizer in which a small amount of non-native data was used for training. Several experiments were performed using Hidden Markov models, trained with speech corpora containing European Portuguese native speakers, English native speakers and English spoken by European Portuguese native speakers. Initially it was explored the behaviour of an English native model and non-native English speakers’ model. Then using different corpus weights for the English native speakers and English spoken by Portuguese speakers it was trained a model as a pool of accents. Through adaptation techniques it was used the Maximum Likelihood Linear Regression method. It was also explored how European Portuguese speakers pronounce English language studying the correspondences between the phone sets of the foreign and target languages. The result was a new phone set, consequence of the mapping between the English and the Portuguese phone sets. Then a new model was trained with English Spoken by Portuguese speakers’ data and Portuguese native data. Concerning the speech recognition subject this work has other two purposes: collecting Portuguese corpora and supporting the compilation of a Portuguese lexicon, adopting some methods and algorithms to generate automatic phonetic pronunciations. The collected corpora was processed in order to train acoustic models to be used in the Exchange 2007 domain, namely in Outlook Voice Access

A speech interface for air traffic control terminals

Several issues concerning the current use of speech interfaces are discussed and the design and development of a speech interface that enables air traffic controllers to command and control their terminals by voice is presented. A special emphasis is made in the comparison between laboratory experiments and field experiments in which a set of ergonomics-related effects are detected that cannot be observed in the controlled laboratory experiments. The paper presents both objective and subjective performance obtained in field evaluation of the system with student controllers at an air traffic control (ATC) training facility. The system exhibits high word recognition test rates (0.4% error in Spanish and 1.5% in English) and low command error (6% error in Spanish and 10.6% error in English in the field tests). Subjective impression has also been positive, encouraging future development and integration phases in the Spanish ATC terminals designed by Aeropuertos Españoles y Navegación Aérea (AENA)

Design and Evaluation of a Presentation Maestro: Controlling Electronic Presentations Through Gesture

Gesture-based interaction has long been seen as a natural means of input for electronic presentation systems; however, gesture-based presentation systems have not been evaluated in real-world contexts, and the implications of this interaction modality are not known. This thesis describes the design and evaluation of Maestro, a gesture-based presentation system which was developed to explore these issues. This work is presented in two parts. The first part describes Maestro's design, which was informed by a small observational study of people giving talks; and Maestro's evaluation, which involved a two week field study where Maestro was used for lecturing to a class of approximately 100 students. The observational study revealed that presenters regularly gesture towards the content of their slides. As such, Maestro supports several gestures which operate directly on slide content (e.g., pointing to a bullet causes it to be highlighted). The field study confirmed that audience members value these content-centric gestures. Conversely, the use of gestures for navigating slides is perceived to be less efficient than the use of a remote. Additionally, gestural input was found to result in a number of unexpected side effects which may hamper the presenter's ability to fully engage the audience. The second part of the thesis presents a gesture recognizer based on discrete hidden Markov models (DHMMs). Here, the contributions lie in presenting a feature set and a factorization of the standard DHMM observation distribution, which allows modeling of a wide range of gestures (e.g., both one-handed and bimanual gestures), but which uses few modeling parameters. To establish the overall robustness and accuracy of the recognition system, five new users and one expert were asked to perform ten instances of each gesture. The system accurately recognized 85% of gestures for new users, increasing to 96% for the expert user. In both cases, false positives accounted for fewer than 4% of all detections. These error rates compare favourably to those of similar systems

Towards robust real-world historical handwriting recognition

In this thesis, we make a bridge from the past to the future by using artificial-intelligence methods for text recognition in a historical Dutch collection of the Natuurkundige Commissie that explored Indonesia (1820-1850). In spite of the successes of systems like 'ChatGPT', reading historical handwriting is still quite challenging for AI. Whereas GPT-like methods work on digital texts, historical manuscripts are only available as an extremely diverse collections of (pixel) images. Despite the great results, current DL methods are very data greedy, time consuming, heavily dependent on the human expert from the humanities for labeling and require machine-learning experts for designing the models. Ideally, the use of deep learning methods should require minimal human effort, have an algorithm observe the evolution of the training process, and avoid inefficient use of the already sparse amount of labeled data. We present several approaches towards dealing with these problems, aiming to improve the robustness of current methods and to improve the autonomy in training. We applied our novel word and line text recognition approaches on nine data sets differing in time period, language, and difficulty: three locally collected historical Latin-based data sets from Naturalis, Leiden; four public Latin-based benchmark data sets for comparability with other approaches; and two Arabic data sets. Using ensemble voting of just five neural networks, a level of accuracy was achieved which required hundreds of neural networks in earlier studies. Moreover, we increased the speed of evaluation of each training epoch without the need of labeled data

Deep Learning for Distant Speech Recognition

Deep learning is an emerging technology that is considered one of the most promising directions for reaching higher levels of artificial intelligence. Among the other achievements, building computers that understand speech represents a crucial leap towards intelligent machines. Despite the great efforts of the past decades, however, a natural and robust human-machine speech interaction still appears to be out of reach, especially when users interact with a distant microphone in noisy and reverberant environments. The latter disturbances severely hamper the intelligibility of a speech signal, making Distant Speech Recognition (DSR) one of the major open challenges in the field. This thesis addresses the latter scenario and proposes some novel techniques, architectures, and algorithms to improve the robustness of distant-talking acoustic models. We first elaborate on methodologies for realistic data contamination, with a particular emphasis on DNN training with simulated data. We then investigate on approaches for better exploiting speech contexts, proposing some original methodologies for both feed-forward and recurrent neural networks. Lastly, inspired by the idea that cooperation across different DNNs could be the key for counteracting the harmful effects of noise and reverberation, we propose a novel deep learning paradigm called network of deep neural networks. The analysis of the original concepts were based on extensive experimental validations conducted on both real and simulated data, considering different corpora, microphone configurations, environments, noisy conditions, and ASR tasks.Comment: PhD Thesis Unitn, 201

Multi-touch interaction for interface prototyping

Tese de mestrado integrado. Engenharia Informática e Computação. Faculdade de Engenharia. Universidade do Porto. 201