571 research outputs found

    Disambiguoiva morfologinen jäsennys probabilistisilla sekvenssimalleilla

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    A morphological tagger is a computer program that provides complete morphological descriptions of sentences. Morphological taggers find applications in many NLP fields. For example, they can be used as a pre-processing step for syntactic parsers, in information retrieval and machine translation. The task of morphological tagging is closely related to POS tagging but morphological taggers provide more fine-grained morphological information than POS taggers. Therefore, they are often applied to morphologically complex languages, which extensively utilize inflection, derivation and compounding for encoding structural and semantic information. This thesis presents work on data-driven morphological tagging for Finnish and other morphologically complex languages. There exists a very limited amount of previous work on data-driven morphological tagging for Finnish because of the lack of freely available manually prepared morphologically tagged corpora. The work presented in this thesis is made possible by the recently published Finnish dependency treebanks FinnTreeBank and Turku Dependency Treebank. Additionally, the Finnish open-source morphological analyzer OMorFi is extensively utilized in the experiments presented in the thesis. The thesis presents methods for improving tagging accuracy, estimation speed and tagging speed in presence of large structured morphological label sets that are typical for morphologically complex languages. More specifically, it presents a novel formulation of generative morphological taggers using weighted finite-state machines and applies finite-state taggers to context sensitive spelling correction of Finnish. The thesis also explores discriminative morphological tagging. It presents structured sub-label dependencies that can be used for improving tagging accuracy. Additionally, the thesis presents a cascaded variant of the averaged perceptron tagger. In presence of large label sets, a cascaded design results in substantial reduction of estimation speed compared to a standard perceptron tagger. Moreover, the thesis explores pruning strategies for perceptron taggers. Finally, the thesis presents the FinnPos toolkit for morphological tagging. FinnPos is an open-source state-of-the-art averaged perceptron tagger implemented by the author.Disambiguoiva morfologinen jäsennin on ohjelma, joka tuottaa yksikäsitteisiä morfologisia kuvauksia virkkeen sanoille. Tällaisia jäsentimiä voidaan hyödyntää monilla kielenkäsittelyn osa-alueilla, esimerkiksi syntaktisen jäsentimen tai konekäännösjärjestelmän esikäsittelyvaiheena. Kieliteknologisena tehtävänä disambiguoiva morfologinen jäsennys muistuttaa perinteistä sanaluokkajäsennystä, mutta se tuottaa hienojakoisempaa morfologista informaatiota kuin perinteinen sanaluokkajäsennin. Tämän takia disambiguoivia morfologisia jäsentimiä hyödynnetäänkin pääsääntöisesti morfologisesti monimutkaisten kielten, kuten suomen kielen, kieliteknologiassa. Tällaisissa kielissä käytetään paljon sananmuodostuskeinoja kuten taivutusta, johtamista ja yhdyssananmuodostusta. Väitöskirjan esittelemä tutkimus liittyy morfologisesti rikkaiden kielten disambiguoivaan morfologiseen jäsentämiseen koneoppimismenetelmin. Vaikka suomen disambiguoivaa morfologista jäsentämistä on tutkittu aiemmin (esim. Constraint Grammar -formalismin avulla), koneoppimismenetelmiä ei ole aiemmin juurikaan sovellettu. Tämä johtuu siitä että jäsentimen oppimiseen tarvittavia korkealuokkaisia morfologisesti annotoituja korpuksia ei ole ollut avoimesti saatavilla. Tässä väitöskirjassa esitelty tutkimus hyödyntää vastikään julkaistuja suomen kielen dependenssijäsennettyjä FinnTreeBank ja Turku Dependency Treebank korpuksia. Lisäksi tutkimus hyödyntää suomen kielen avointa morfologista OMorFi-jäsennintä. Väitöskirja esittelee menetelmiä jäsennystarkkuuden parantamiseen ja jäsentimen opetusnopeuden sekä jäsennysnopeuden kasvattamiseen. Väitöskirja esittää uuden tavan rakentaa generatiivisia jäsentimiä hyödyntäen painollisia äärellistilaisia koneita ja soveltaa tällaisia jäsentimiä suomen kielen kontekstisensitiiviseen oikeinkirjoituksentarkistukseen. Lisäksi väitöskirja käsittelee diskriminatiivisia jäsennysmalleja. Se esittelee tapoja hyödyntää morfologisten analyysien osia jäsennystarkkuuden parantamiseen. Lisäksi se esittää kaskadimallin, jonka avulla jäsentimen opetusaika lyhenee huomattavasi. Väitöskirja esittää myös tapoja jäsenninmallien pienentämiseen. Lopuksi esitellään FinnPos, joka on kirjoittaman toteuttama avoimen lähdekoodin työkalu disambiguoivien morfologisten jäsentimien opettamiseen

    Past, Present, and Future of Simultaneous Localization And Mapping: Towards the Robust-Perception Age

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    Simultaneous Localization and Mapping (SLAM)consists in the concurrent construction of a model of the environment (the map), and the estimation of the state of the robot moving within it. The SLAM community has made astonishing progress over the last 30 years, enabling large-scale real-world applications, and witnessing a steady transition of this technology to industry. We survey the current state of SLAM. We start by presenting what is now the de-facto standard formulation for SLAM. We then review related work, covering a broad set of topics including robustness and scalability in long-term mapping, metric and semantic representations for mapping, theoretical performance guarantees, active SLAM and exploration, and other new frontiers. This paper simultaneously serves as a position paper and tutorial to those who are users of SLAM. By looking at the published research with a critical eye, we delineate open challenges and new research issues, that still deserve careful scientific investigation. The paper also contains the authors' take on two questions that often animate discussions during robotics conferences: Do robots need SLAM? and Is SLAM solved

    Spoken content retrieval: A survey of techniques and technologies

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    Speech media, that is, digital audio and video containing spoken content, has blossomed in recent years. Large collections are accruing on the Internet as well as in private and enterprise settings. This growth has motivated extensive research on techniques and technologies that facilitate reliable indexing and retrieval. Spoken content retrieval (SCR) requires the combination of audio and speech processing technologies with methods from information retrieval (IR). SCR research initially investigated planned speech structured in document-like units, but has subsequently shifted focus to more informal spoken content produced spontaneously, outside of the studio and in conversational settings. This survey provides an overview of the field of SCR encompassing component technologies, the relationship of SCR to text IR and automatic speech recognition and user interaction issues. It is aimed at researchers with backgrounds in speech technology or IR who are seeking deeper insight on how these fields are integrated to support research and development, thus addressing the core challenges of SCR

    Analysis and modeling of non-native speech for automatic speech recognition

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    Thesis (S.M.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1999.Includes bibliographical references (p. 75-77).by Karen Livescu.S.M

    Segmentation, Diarization and Speech Transcription: Surprise Data Unraveled

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    In this thesis, research on large vocabulary continuous speech recognition for unknown audio conditions is presented. For automatic speech recognition systems based on statistical methods, it is important that the conditions of the audio used for training the statistical models match the conditions of the audio to be processed. Any mismatch will decrease the accuracy of the recognition. If it is unpredictable what kind of data can be expected, or in other words if the conditions of the audio to be processed are unknown, it is impossible to tune the models. If the material consists of `surprise data' the output of the system is likely to be poor. In this thesis methods are presented for which no external training data is required for training models. These novel methods have been implemented in a large vocabulary continuous speech recognition system called SHoUT. This system consists of three subsystems: speech/non-speech classification, speaker diarization and automatic speech recognition. The speech/non-speech classification subsystem separates speech from silence and unknown audible non-speech events. The type of non-speech present in audio recordings can vary from paper shuffling in recordings of meetings to sound effects in television shows. Because it is unknown what type of non-speech needs to be detected, it is not possible to train high quality statistical models for each type of non-speech sound. The speech/non-speech classification subsystem, also called the speech activity detection subsystem, does not attempt to classify all audible non-speech in a single run. Instead, first a bootstrap speech/silence classification is obtained using a standard speech activity component. Next, the models for speech, silence and audible non-speech are trained on the target audio using the bootstrap classification. This approach makes it possible to classify speech and non-speech with high accuracy, without the need to know what kinds of sound are present in the audio recording. Once all non-speech is filtered out of the audio, it is the task of the speaker diarization subsystem to determine how many speakers occur in the recording and exactly when they are speaking. The speaker diarization subsystem applies agglomerative clustering to create clusters of speech fragments for each speaker in the recording. First, statistical speaker models are created on random chunks of the recording and by iteratively realigning the data, retraining the models and merging models that represent the same speaker, accurate speaker models are obtained for speaker clustering. This method does not require any statistical models developed on a training set, which makes the diarization subsystem insensitive for variation in audio conditions. Unfortunately, because the algorithm is of complexity O(n3)O(n^3), this clustering method is slow for long recordings. Two variations of the subsystem are presented that reduce the needed computational effort, so that the subsystem is applicable for long audio recordings as well. The automatic speech recognition subsystem developed for this research, is based on Viterbi decoding on a fixed pronunciation prefix tree. Using the fixed tree, a flexible modular decoder could be developed, but it was not straightforward to apply full language model look-ahead efficiently. In this thesis a novel method is discussed that makes it possible to apply language model look-ahead effectively on the fixed tree. Also, to obtain higher speech recognition accuracy on audio with unknown acoustical conditions, a selection from the numerous known methods that exist for robust automatic speech recognition is applied and evaluated in this thesis. The three individual subsystems as well as the entire system have been successfully evaluated on three international benchmarks. The diarization subsystem has been evaluated at the NIST RT06s benchmark and the speech activity detection subsystem has been tested at RT07s. The entire system was evaluated at N-Best, the first automatic speech recognition benchmark for Dutch
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