82 research outputs found

    CLiFF Notes: Research In Natural Language Processing at the University of Pennsylvania

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    CLIFF is the Computational Linguists\u27 Feedback Forum. We are a group of students and faculty who gather once a week to hear a presentation and discuss work currently in progress. The \u27feedback\u27 in the group\u27s name is important: we are interested in sharing ideas, in discussing ongoing research, and in bringing together work done by the students and faculty in Computer Science and other departments. However, there are only so many presentations which we can have in a year. We felt that it would be beneficial to have a report which would have, in one place, short descriptions of the work in Natural Language Processing at the University of Pennsylvania. This report then, is a collection of abstracts from both faculty and graduate students, in Computer Science, Psychology and Linguistics. We want to stress the close ties between these groups, as one of the things that we pride ourselves on here at Penn is the communication among different departments and the inter-departmental work. Rather than try to summarize the varied work currently underway at Penn, we suggest reading the abstracts to see how the students and faculty themselves describe their work. The report illustrates the diversity of interests among the researchers here, as well as explaining the areas of common interest. In addition, since it was our intent to put together a document that would be useful both inside and outside of the university, we hope that this report will explain to everyone some of what we are about

    Segmental Durations of Speech

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    This dissertation considers the segmental durations of speech from the viewpoint of speech technology, especially speech synthesis. The idea is that better models of segmental durations lead to higher naturalness and better intelligibility. These features are the key factors for better usability and generality of synthesized speech technology. Even though the studies are based on a Finnish corpus the approaches apply to all other languages as well. This is possibly due to the fact that most of the studies included in this dissertation are about universal effects taking place on utterance boundaries. Also the methods invented and used here are suitable for any other study of another language. This study is based on two corpora of news reading speech and sentences read aloud. The other corpus is read aloud by a 39-year-old male, whilst the other consists of several speakers in various situations. The use of two corpora is twofold: it involves a comparison of the corpora and a broader view on the matters of interest. The dissertation begins with an overview to the phonemes and the quantity system in the Finnish language. Especially, we are covering the intrinsic durations of phonemes and phoneme categories, as well as the difference of duration between short and long phonemes. The phoneme categories are presented to facilitate the problem of variability of speech segments. In this dissertation we cover the boundary-adjacent effects on segmental durations. In initial positions of utterances we find that there seems to be initial shortening in Finnish, but the result depends on the level of detail and on the individual phoneme. On the phoneme level we find that the shortening or lengthening only affects the very first ones at the beginning of an utterance. However, on average, the effect seems to shorten the whole first word on the word level. We establish the effect of final lengthening in Finnish. The effect in Finnish has been an open question for a long time, whilst Finnish has been the last missing piece for it to be a universal phenomenon. Final lengthening is studied from various angles and it is also shown that it is not a mere effect of prominence or an effect of speech corpus with high inter- and intra-speaker variation. The effect of final lengthening seems to extend from the final to the penultimate word. On a phoneme level it reaches a much wider area than the initial effect. We also present a normalization method suitable for corpus studies on segmental durations. The method uses an utterance-level normalization approach to capture the pattern of segmental durations within each utterance. This prevents the impact of various problematic variations within the corpora. The normalization is used in a study on final lengthening to show that the results on the effect are not caused by variation in the material. The dissertation shows an implementation and prowess of speech synthesis on a mobile platform. We find that the rule-based method of speech synthesis is a real-time software solution, but the signal generation process slows down the system beyond real time. Future aspects of speech synthesis on limited platforms are discussed. The dissertation considers ethical issues on the development of speech technology. The main focus is on the development of speech synthesis with high naturalness, but the problems and solutions are applicable to any other speech technology approaches.Siirretty Doriast

    Research in the Language, Information and Computation Laboratory of the University of Pennsylvania

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    This report takes its name from the Computational Linguistics Feedback Forum (CLiFF), an informal discussion group for students and faculty. However the scope of the research covered in this report is broader than the title might suggest; this is the yearly report of the LINC Lab, the Language, Information and Computation Laboratory of the University of Pennsylvania. It may at first be hard to see the threads that bind together the work presented here, work by faculty, graduate students and postdocs in the Computer Science and Linguistics Departments, and the Institute for Research in Cognitive Science. It includes prototypical Natural Language fields such as: Combinatorial Categorial Grammars, Tree Adjoining Grammars, syntactic parsing and the syntax-semantics interface; but it extends to statistical methods, plan inference, instruction understanding, intonation, causal reasoning, free word order languages, geometric reasoning, medical informatics, connectionism, and language acquisition. Naturally, this introduction cannot spell out all the connections between these abstracts; we invite you to explore them on your own. In fact, with this issue it’s easier than ever to do so: this document is accessible on the “information superhighway”. Just call up http://www.cis.upenn.edu/~cliff-group/94/cliffnotes.html In addition, you can find many of the papers referenced in the CLiFF Notes on the net. Most can be obtained by following links from the authors’ abstracts in the web version of this report. The abstracts describe the researchers’ many areas of investigation, explain their shared concerns, and present some interesting work in Cognitive Science. We hope its new online format makes the CLiFF Notes a more useful and interesting guide to Computational Linguistics activity at Penn

    Statistical morphological disambiguation with application to disambiguation of pronunciations in Turkish /

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    The statistical morphological disambiguation of agglutinative languages suffers from data sparseness. In this study, we introduce the notion of distinguishing tag sets (DTS) to overcome the problem. The morphological analyses of words are modeled with DTS and the root major part-of-speech tags. The disambiguator based on the introduced representations performs the statistical morphological disambiguation of Turkish with a recall of as high as 95.69 percent. In text-to-speech systems and in developing transcriptions for acoustic speech data, the problem occurs in disambiguating the pronunciation of a token in context, so that the correct pronunciation can be produced or the transcription uses the correct set of phonemes. We apply the morphological disambiguator to this problem of pronunciation disambiguation and achieve 99.54 percent recall with 97.95 percent precision. Most text-to-speech systems perform phrase level accentuation based on content word/function word distinction. This approach seems easy and adequate for some right headed languages such as English but is not suitable for languages such as Turkish. We then use a a heuristic approach to mark up the phrase boundaries based on dependency parsing on a basis of phrase level accentuation for Turkish TTS synthesizers

    Perception and Acquisition of Natural Authentic English Speech for Chinese Learners Using DIT\u27s Speech Technologies

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    Given that Chinese language learners are greatly influenced by their mother-tongue, which is a tone language rather than an intonation language, learning and coping with authentic English speech seems more difficult than for learners of other languages. The focus of the current research is, on the basis of analysis of the nature of spoken English and spoken Chinese, to help Chinese learners derive benefit from ICT technologies developed by the Technological University Dublin (DIT). The thesis concentrates on investigating the application of speech technologies in bridging the gap between students’ internalised, idealised formulations and natural, authentic English speech. Part of the testing carried out by the present author demonstrates the acceptability of a slow-down algorithm in facilitating Chinese learners of English in re-producing formulaic language. This algorithm is useful because it can slow down audio files to any desired speed between 100% and 40% without distortion, so as to allow language learners to pay attention to the real, rapid flow of ‘messy’ speech and follow the intonation patterns contained in them. The rationale for and the application of natural, dialogic native-to-native English speech to language learning is also explored. The Chinese language learners involved in this study are exposed to authentic, native speech patterns by providing them access to real, informal dialogue in various contexts. In the course of this analysis, the influence of speed of delivery and pitch range on the categorisation of formulaic language is also investigated. The study investigates the potential of the speech tools available to the present author as an effective EFL learning facility, especially for speakers of tone languages, and their role in helping language learners achieve confluent interaction in an English L1 environment

    Speech-to-speech translation to support medical interviews

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    Projeto de mestrado em Engenharia Informática, apresentada à Universidade de Lisboa, através da Faculdade de Ciências, 2013Este relatório apresenta a criação de um sistema de tradução fala-para-fala. O sistema consiste na captação de voz na forma de sinal áudio que de seguida é interpretado, traduzido e sintetizado para voz. Tendo como entrada um enunciado numa linguagem de origem e como saída um enunciado numa linguagem destino. O sistema implementado tem como âmbito do seu funcionamento o domínio médico, tendo em vista apoiar o diálogo entre médico e utente em linguagens diferentes durante consultas médicas. No caso do presente trabalho, foram escolhidos o português e inglês, sendo possível a tradução fala-para-fala nos dois sentidos. A escolha destas duas línguas resulta sobretudo da disponibilidade de recursos para o desenvolvimento do sistema. Ao longo dos anos tem existido um esforço de pesquisa e desenvolvimento em tecnologia que permite quebrar as barreiras do multilinguismo. Uma dessas tecnologias, com resultados de qualidade crescentemente aceitável, são os sistemas de tradução fala-para-fala. Em geral, estes sistemas são compostos por três componentes: reconhecimento de fala, tradução automática e sintetização de voz. Neste projecto foram implementadas as três componentes. No entanto, uma vez que face às tecnologias disponíveis, a componente de tradução tem um maior impacto no desempenho final do sistema, a esta foi conferida uma maior atenção. Embora nós, como humanos, compreendamos facilmente a linguagem falada, isto é algo extremamente difícil e complexo de um ponto de vista computacional. O objectivo do reconhecimento de fala é abordar esta tarefa computacionalmente através da construção de sistemas que mapeiam um sinal acústico para uma sequência de caracteres. Os modelos actuais para reconhecimento de fala fazem uso de modelos estatísticos. Nestes, a fala é reconhecida através do uso de modelos de linguagem que possibilitam a estimativa das probabilidades para as palavras, independentemente do sinal de entrada, e de um modelo acústico onde as propriedades acústicas da fala estão contempladas. Os modelos actuais de tradução automática, assim como os de reconhecimento de fala, são na sua larga maioria estatísticos. Actualmente os modelos de tradução baseados em unidades frásicas de input são os que obtém os resultados com melhor qualidade. Esta abordagem consiste na tradução de pequenos segmentos de palavras, onde existe uma tradução lexical e um modelo de alinhamento. Os modelos estatísticos fazem uso de textos de duas línguas alinhados, tendo como princípio o facto de que através da frequência de cada segmento de palavras, em relação à outra linguagem, seja obtida uma distribuição probabilística. Deste modo torna-se possível calcular qual a palavra ou conjunto de palavras mais prováveis de ocorrer como tradução para determinado texto que se pretenda traduzir. A sintetização de voz consiste na geração de fala na forma de onda acústica tendo como ponto de partida uma palavra ou uma sequência de palavras. Envolve o processamento de linguagens naturais e processamento de sinal. O primeiro converte o texto numa representação fonética e o último converte essa representação em sinal acústico. Neste documento é apresentado o estado da arte das três áreas envolvidas. São também apresentados os sistemas de tradução fala-para-fala, fazendo ou não uso do domínio médico, e também os processos existentes para a avaliação de cada uma das componentes. Tendo em vista a implementação de um sistema com as diversas componentes, foi necessário efectuar um levantamento da tecnologia existente. O levantamento teve por objectivo a implementação de duas soluções aplicacionais. Uma aplicação disponível pela internet como página web e outra através de uma aplicação móvel, ambas permitindo o reconhecimento de fala, tradução automática e sintetização de voz em ambas as linguagens e direcções. Dois sistemas de reconhecimento de fala foram escolhidos, o Microsoft Speech Platform para a aplicação móvel e o reconhecimento de fala disponível pelo Google nos browsers Google Chrome. O primeiro a ser usado na aplicação móvel e o segundo na aplicação web. O sistema de tradução automática escolhido foi o Moses. Sendo um sistema de tradução estatístico que permite a criação de modelos de tradução diversos, como os modelos baseados em frase e os modelos baseados em fatores. O sistema de sintetização de voz escolhido foi o Microsoft Speech Platform. A aplicação móvel foi desenvolvida para a plataforma iOS da Apple tendo em vista o uso de um telemóvel iPhone. A integração dos componentes pelas diversas arquitecturas foi assegurada pela implementação de web services. O reconhecimento de fala na aplicação web foi desenvolvido recorrendo ao uso da W3C Speech Input API Specifications, onde a programação através de HTML permite a captação de áudio no Google Chrome. Para a implementação do sistema tradução fala-para-fala foi necessário a obtenção de corpora paralelos de forma a se poder treinar os modelos estatísticos, sendo este um dos factores cruciais para o bom desempenho dos componentes. Uma vez que o sistema tem como domínio de aplicação o diálogo médico, corpora neste domínio seria o mais vantajoso. No entanto, a inexistência de tais corpora para o par Inglês-Português levou à aquisição de corpora alternativos. Através de uma experiência exploratória foi abordado o tipo de implementação mais adequado da componente de reconhecimento de fala, tendo como foco o modelo de linguagem. Três experiências foram então conduzidas de forma a decidir entre a aplicação de um modelo de linguagem baseado em regras ou um modelo estatístico. Para implementar um modelo de linguagem baseado em regras foi necessário a criação de um corpus médico que reflectisse um diálogo entre médico e paciente. Para tal, com a ajuda de um médico, criei um diálogo de um caso hipotético de lesão num braço devido a um acidente de carro. Este diálogo teve como base para a sua estruturação a aplicação do processo de anamnesis. A anamnesis consiste numa metodologia médica que através de um conjunto de perguntas chave permite adquirir a informação necessária para a formulação de um diagnóstico médico e decisão sobre o tratamento necessário. O corpus médico foi também transformado num corpus de fala de forma a este ser avaliado ao longo das experiências. Numa primeira experiência foi criada uma gramática básica cuja implementação foi obtida recorrendo à Speech Recognition Grammar Specification de forma a ser usada como modelo de linguagem pela componente de reconhecimento de fala. A segunda experiência tinha como objectivo a criação de uma gramática mais complexa que a primeira. Para tal foi criada uma gramática livre de contexto. Após a criação da gramática livre de contexto esta foi convertida manualmente para uma gramática SRGS. Na terceira experiência foram criados dois modelo de linguagem estatísticos, o primeiro fazendo uso do mesmo corpus que o usado nas experiências anteriores e o segundo composto por 30.000 frases independentes. Obteve-se melhores resultados com o modelo de linguagem estatístico e este ficou como a escolha para a implementação do componente de reconhecimento de fala. No treino da componente de tradução automática foram usados dois modelos estatísticos, baseados em frases e em factores. Pretendeu-se comparar os resultados entre os dois modelos de forma a escolher o modelo mais vantajoso. Para fazer uso do modelo baseado em factores foi necessária a preparação de corpora. Com os corpora já adquiridos foi concretizada a sua anotação para ambas as linguagens. Recorrendo ao LX-Suite e ao CoreNLP, foram criados corpora anotados com lemmas e informação morfossintáctica, com a primeira ferramenta para o português e a última para o inglês. Uma vez que a componente de sintetização de voz permitia uma implementação célere, esta foi implementada recorrendo aos modelos já existentes para ambas as linguagens e disponibilizados pela ferramenta. Por fim, são apresentados os resultados obtidos e a sua avaliação. Tanto a avaliação do sistema de reconhecimento de fala como o de tradução automática demonstraram um desempenho muito competitivo, do nível do estado da arte. A componente de reconhecimento de fala, assim como a componente de tradução automática, obtiveram melhores resultados fazendo-se uso de modelos de linguagem estatístico.This report presents the development of a speech-to-speech translation system. The system consists in the capture of voice as an audio signal that is then interpreted, translated and synthesized to voice for a target language. The three main components of the system, speech recognition, machine translation and speech synthesis, make use of statistical models, such as hidden Markov models. Given the technology available, the machine translation component has a greater impact on the performance of the system, a greater attention has thus been given to it. The system assumes the support to medical interviews between doctor and patient in different languages as its applicational domain. Two application solutions were developed: an online service on a website and a mobile application. This report begins by presenting the general concepts of the relevant areas involved. It proceeds with an overview of the state of the art relating to each area as well as to the methods used for the evaluation of the different components. It provides also an overview of existing technology and the criteria for choosing the tools to be used in the development of the system. It explains the acquisition and creation of the corpora used, and the process of development and integration of the components: speech recognition, machine translation and text-to-speech. Finally, the evaluation results are presented, as well as the final conclusions

    Speech Recognition

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    Chapters in the first part of the book cover all the essential speech processing techniques for building robust, automatic speech recognition systems: the representation for speech signals and the methods for speech-features extraction, acoustic and language modeling, efficient algorithms for searching the hypothesis space, and multimodal approaches to speech recognition. The last part of the book is devoted to other speech processing applications that can use the information from automatic speech recognition for speaker identification and tracking, for prosody modeling in emotion-detection systems and in other speech processing applications that are able to operate in real-world environments, like mobile communication services and smart homes

    Effects of errorless learning on the acquisition of velopharyngeal movement control

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    Session 1pSC - Speech Communication: Cross-Linguistic Studies of Speech Sound Learning of the Languages of Hong Kong (Poster Session)The implicit motor learning literature suggests a benefit for learning if errors are minimized during practice. This study investigated whether the same principle holds for learning velopharyngeal movement control. Normal speaking participants learned to produce hypernasal speech in either an errorless learning condition (in which the possibility for errors was limited) or an errorful learning condition (in which the possibility for errors was not limited). Nasality level of the participants’ speech was measured by nasometer and reflected by nasalance scores (in %). Errorless learners practiced producing hypernasal speech with a threshold nasalance score of 10% at the beginning, which gradually increased to a threshold of 50% at the end. The same set of threshold targets were presented to errorful learners but in a reversed order. Errors were defined by the proportion of speech with a nasalance score below the threshold. The results showed that, relative to errorful learners, errorless learners displayed fewer errors (50.7% vs. 17.7%) and a higher mean nasalance score (31.3% vs. 46.7%) during the acquisition phase. Furthermore, errorless learners outperformed errorful learners in both retention and novel transfer tests. Acknowledgment: Supported by The University of Hong Kong Strategic Research Theme for Sciences of Learning © 2012 Acoustical Society of Americapublished_or_final_versio

    Proceedings of the 60th Annual Meeting of the Association for Computational Linguistics: Student Research Workshop

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    Indonesian and Malay are underrepresented in the development of natural language processing (NLP) technologies and available resources are difficult to find. A clear picture of existing work can invigorate and inform how researchers conceptualise worthwhile projects. Using an education sector project to motivate the study, we conducted a wide-ranging overview of Indonesian and Malay human language technologies and corpus work. We charted 657 included studies according to Hirschberg and Manning's 2015 description of NLP, concluding that the field was dominated by exploratory corpus work, machine reading of text gathered from the Internet, and sentiment analysis. In this paper, we identify most published authors and research hubs, and make a number of recommendations to encourage future collaboration and efficiency within NLP in Indonesian and Malay

    Phraseology in Corpus-based transaltion studies : stylistic study of two contempoarary Chinese translation of Cervantes's Don Quijote

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    The present work sets out to investigate the stylistic profiles of two modern Chinese versions of Cervantes???s Don Quijote (I): by Yang Jiang (1978), the first direct translation from Castilian to Chinese, and by Liu Jingsheng (1995), which is one of the most commercially successful versions of the Castilian literary classic. This thesis focuses on a detailed linguistic analysis carried out with the help of the latest textual analytical tools, natural language processing applications and statistical packages. The type of linguistic phenomenon singled out for study is four-character expressions (FCEXs), which are a very typical category of Chinese phraseology. The work opens with the creation of a descriptive framework for the annotation of linguistic data extracted from the parallel corpus of Don Quijote. Subsequently, the classified and extracted data are put through several statistical tests. The results of these tests prove to be very revealing regarding the different use of FCEXs in the two Chinese translations. The computational modelling of the linguistic data would seem to indicate that among other findings, while Liu???s use of archaic idioms has followed the general patterns of the original and also of Yang???s work in the first half of Don Quijote I, noticeable variations begin to emerge in the second half of Liu???s more recent version. Such an idiosyncratic use of archaisms by Liu, which may be defined as style shifting or style variation, is then analyzed in quantitative terms through the application of the proposed context-motivated theory (CMT). The results of applying the CMT-derived statistical models show that the detected stylistic variation may well point to the internal consistency of the translator in rendering the second half of Part I of the novel, which reflects his freer, more creative and experimental style of translation. Through the introduction and testing of quantitative research methods adapted from corpus linguistics and textual statistics, this thesis has made a major contribution to methodological innovation in the study of style within the context of corpus-based translation studies.Imperial Users onl
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