An on-line speaker adaptation method for HMM-based speech recognizers

In the past few years numerous techniques have been proposed to improve the efficiency of basic adaptation methods like MLLR and MAP. These adaptation methods have a common aim, which is to increase the likelihood of the phoneme models for a particular speaker. During their operation, these speaker adaptation methods need precise phonetic segmentation information of the actual utterance, but these data samples are often faulty. To improve the overall performance, only those frames from the spoken sentence which are well segmented should be retained, while the incorrectly segmented data should not be used during adaptation. Several heuristic algorithms have been proposed in the literature for the selection of the reliably segmented data blocks, and here we would like to suggest some new heuristics that discriminate between faulty and well-segmented data. The effect of these methods on the efficiency of speech recognition using speaker adaptation is examined, and conclusions for each will be drawn. Besided post-filtering the set of the segmented adaptation examples, another way of improving the efficiency of the adaptation method might be to create a more precise segmentation, which should then reduce the chance of faulty data samples being included. We suggest a method like this here as well which is based on a scoring procedure for the N-best lists, taking into account phoneme duration

Adaptation of voice sever to automotive environment

This project is embedded within an investigation Project named "Movilidad y Automoción para Redes de Transporte Avanzados" (MARTA).It has as a fundamental strategic goal to consolidate the scientifically and technological basis to 21th century mobility to allow Spanish ITS ("Intelligent Transport Systems") sector to answer the challenges of efficiency, sustainability, etc . which European society and especially Spanish society has to confront in the next years. In this project Telefónica I+D (TID) is in charge of the study, specification and implementation of speech technology in automotive environment considering vehicle usability conditions. The work of the student in this project is to adapt a voice server, that contains speech tools, to automotive environment. Add new libraries that annex new functions and extend and develop the communication with XML to use these new functions

Fully Automated Non-Native Speech Recognition Using Confusion-Based Acoustic Model Integration

This paper presents a fully automated approach for the recognition of non-native speech based on acoustic model modification. For a native language (L1) and a spoken language (L2), pronunciation variants of the phones of L2 are automatically extracted from an existing non-native database as a confusion matrix with sequences of phones of L1. This is done using L1's and L2's ASR systems. This confusion concept deals with the problem of non existence of match between some L2 and L1 phones. The confusion matrix is then used to modify the acoustic models (HMMs) of L2 phones by integrating corresponding L1 phone models as alternative HMM paths. In this way, no lexicon modification is carried. The modified ASR system achieved an improvement between 32% and 40% (relative, L1=French and L2=English) in WER on the French non-native database used for testing

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

PersoNER: Persian named-entity recognition

© 1963-2018 ACL. Named-Entity Recognition (NER) is still a challenging task for languages with low digital resources. The main difficulties arise from the scarcity of annotated corpora and the consequent problematic training of an effective NER pipeline. To abridge this gap, in this paper we target the Persian language that is spoken by a population of over a hundred million people world-wide. We first present and provide ArmanPerosNERCorpus, the first manually-annotated Persian NER corpus. Then, we introduce PersoNER, an NER pipeline for Persian that leverages a word embedding and a sequential max-margin classifier. The experimental results show that the proposed approach is capable of achieving interesting MUC7 and CoNNL scores while outperforming two alternatives based on a CRF and a recurrent neural network

Self-Supervised and Controlled Multi-Document Opinion Summarization

We address the problem of unsupervised abstractive summarization of collections of user generated reviews with self-supervision and control. We propose a self-supervised setup that considers an individual document as a target summary for a set of similar documents. This setting makes training simpler than previous approaches by relying only on standard log-likelihood loss. We address the problem of hallucinations through the use of control codes, to steer the generation towards more coherent and relevant summaries.Finally, we extend the Transformer architecture to allow for multiple reviews as input. Our benchmarks on two datasets against graph-based and recent neural abstractive unsupervised models show that our proposed method generates summaries with a superior quality and relevance.This is confirmed in our human evaluation which focuses explicitly on the faithfulness of generated summaries We also provide an ablation study, which shows the importance of the control setup in controlling hallucinations and achieve high sentiment and topic alignment of the summaries with the input reviews.Comment: 18 pages including 5 pages appendi

Fully Automated Non-Native Speech Recognition Using Confusion-Based Acoustic Model Integration And Graphemic Constraints

This paper presents a fully automated approach for the recognition of non-native speech based on acoustic model modification. For a native language (L1) and a spoken language (L2), pronunciation variants of the phones of L2 are automatically extracted from an existing non-native database as a confusion matrix with sequences of phones of L1. This is done using L1's and L2's ASR systems. This confusion concept deals with the problem of non existence of match between some L2 and L1 phones. The confusion matrix is then used to modify the acoustic models (HMMs) of L2 phones by integrating corresponding L1 phone models as alternative HMM paths. We introduce graphemic contraints in the confusion extraction process: the phonetic confusion is established for each couple of `L2-phone' and the grapheme(s) correspondig to that phone. We claim that prononciation errors may depend on the graphemes related to each phone. The modified ASR system achieved an improvement between 32% and 40% (relative, L1=French and L2=English) in WER on the French non-native database used for testing. The introduction of graphemic contraints in the phonetic confusion allowed further improvements

Low-Resource Unsupervised NMT:Diagnosing the Problem and Providing a Linguistically Motivated Solution

Unsupervised Machine Translation hasbeen advancing our ability to translatewithout parallel data, but state-of-the-artmethods assume an abundance of mono-lingual data. This paper investigates thescenario where monolingual data is lim-ited as well, finding that current unsuper-vised methods suffer in performance un-der this stricter setting. We find that theperformance loss originates from the poorquality of the pretrained monolingual em-beddings, and we propose using linguis-tic information in the embedding train-ing scheme. To support this, we look attwo linguistic features that may help im-prove alignment quality: dependency in-formation and sub-word information. Us-ing dependency-based embeddings resultsin a complementary word representationwhich offers a boost in performance ofaround 1.5 BLEU points compared to stan-dardWORD2VECwhen monolingual datais limited to 1 million sentences per lan-guage. We also find that the inclusion ofsub-word information is crucial to improv-ing the quality of the embedding