Integrated speech and morphological processing in a connectionist continuous speech understanding for Korean

A new tightly coupled speech and natural language integration model is presented for a TDNN-based continuous possibly large vocabulary speech recognition system for Korean. Unlike popular n-best techniques developed for integrating mainly HMM-based speech recognition and natural language processing in a {\em word level}, which is obviously inadequate for morphologically complex agglutinative languages, our model constructs a spoken language system based on a {\em morpheme-level} speech and language integration. With this integration scheme, the spoken Korean processing engine (SKOPE) is designed and implemented using a TDNN-based diphone recognition module integrated with a Viterbi-based lexical decoding and symbolic phonological/morphological co-analysis. Our experiment results show that the speaker-dependent continuous {\em eojeol} (Korean word) recognition and integrated morphological analysis can be achieved with over 80.6% success rate directly from speech inputs for the middle-level vocabularies.Comment: latex source with a4 style, 15 pages, to be published in computer processing of oriental language journa

The Speech-Language Interface in the Spoken Language Translator

The Spoken Language Translator is a prototype for practically useful systems capable of translating continuous spoken language within restricted domains. The prototype system translates air travel (ATIS) queries from spoken English to spoken Swedish and to French. It is constructed, with as few modifications as possible, from existing pieces of speech and language processing software. The speech recognizer and language understander are connected by a fairly conventional pipelined N-best interface. This paper focuses on the ways in which the language processor makes intelligent use of the sentence hypotheses delivered by the recognizer. These ways include (1) producing modified hypotheses to reflect the possible presence of repairs in the uttered word sequence; (2) fast parsing with a version of the grammar automatically specialized to the more frequent constructions in the training corpus; and (3) allowing syntactic and semantic factors to interact with acoustic ones in the choice of a meaning structure for translation, so that the acoustically preferred hypothesis is not always selected even if it is within linguistic coverage.Comment: 9 pages, LaTeX. Published: Proceedings of TWLT-8, December 199

Äärelliset tilamallit lukupuheen tunnistamisessa ja tarkastamisessa

An automatic speech recognition system has to combine acoustic and linguistic information. Therefore the search space spans multiple layers. Finite state models and weighted finite state transducers in particular can efficiently represent this search space by modeling each layer as a transducer and combining them using generic weighted finite state transducer algorithms. When recognising a text prompt being read aloud, the prompt gives a good estimate of what is going to be said. However human reading naturally produces some deviations from the text, called miscues. The purpose of this thesis is to create a system which accurately recognises recordings of reading. A miscue tolerant finite state language model is implemented and compared against two traditional approaches, an N-gram model and forced alignment. The recognition result will ultimately be used to validate the recording as fit for further automatic processing in a spoken foreign language exam, which Project DigiTala is designing for the Finnish matriculation examination. The computerization of the matriculation examination in Finland makes the use of such automatic tools possible. This thesis first introduces the context for the task of recognising and validating reading. Then it explores three methodologies needed to solve the task: automatic speech recognition, finite state models, and the modeling of reading. Next it recounts the implementation of the miscue tolerant finite state language models and the two baseline methods. After that it describes experiments which show that the miscue tolerant finite state language models solve the task of this thesis significantly better than the baseline methods. Finally the thesis concludes with a discussion of the results and future work.Automaattinen puheentunnistusjärjestelmä yhdistää akustista ja kielellistä tietoa, joten sen hakuavaruus on monitasoinen. Tämän hakuavaruuden voi esittää tehokkaasti äärellisillä tilamalleilla. Erityisesti painotetut äärelliset tilamuuttajat voivat esittää jokaista hakuavaruuden tasoa ja nämä muuttajat voidaan yhdistää yleisillä muuttaja-algoritmeilla. Kun tunnistetaan ääneen lukemista syötteestä, syöte rajaa hakuavaruutta hyvin. Ihmiset kuitenkin poikkeavat tekstistä hieman. Kutsun näitä lukupoikkeamiksi, koska ne ovat luonnollinen osa taitavaakin lukemista, eivätkä siis suoranaisesti lukuvirheitä. Tämän diplomityön tavoite on luoda järjestelmä, joka tunnistaa lukupuheäänitteitä tarkasti. Tätä varten toteutetaan lukupoikkeamia sietävä äärellisen tilan kielimalli, jota verrataan kahteen perinteiseen menetelmään, N-gram malleihin ja pakotettuun kohdistukseen. Lukupuheen tunnistustulosta käytetään, kun tarkastetaan, sopiiko äänite seuraaviin automaattisiin käsittelyvaiheisiin puhutussa vieraan kielen kokeessa. DigiTalaprojekti muotoilee puhuttua osiota vieraan kielen ylioppilaskokeisiin. Ylioppilaskokeiden sähköistäminen mahdollistaa tällaisten automaattisten menetelmien käytön. Kokeet sekä englanninkielisellä simuloidulla aineistolla että ruotsinkielisellä tosimaailman aineistolla osoittavat, että lukupoikkeamia sietävä äärellisen tilan kielimalli ratkaisee diplomityön ongelmanasettelun. Vaikealla tosimaailman aineistolla saadaan 3.77 ± 0.47 prosentuaalinen sanavirhemäärä

Spontal-N: A Corpus of Interactional Spoken Norwegian

Spontal-N is a corpus of spontaneous, interactional Norwegian. To our knowledge, it is the first corpus of Norwegian in which the majority of speakers have spent significant parts of their lives in Sweden, and in which the recorded speech displays varying degrees of interference from Swedish. The corpus consists of studio quality audio- and video-recordings of four 30-minute free conversations between acquaintances, and a manual orthographic transcription of the entire material. On basis of the orthographic transcriptions, we automatically annotated approximately 50 percent of the material on the phoneme level, by means of a forced alignment between the acoustic signal and pronunciations listed in a dictionary. Approximately seven percent of the automatic transcription was manually corrected. Taking the manual correction as a gold standard, we evaluated several sources of pronunciation variants for the automatic transcription. Spontal-N is intended as a general purpose speech resource that is also suitable for investigating phonetic detail

Automatic Speech Recognition for Low-resource Languages and Accents Using Multilingual and Crosslingual Information

This thesis explores methods to rapidly bootstrap automatic speech recognition systems for languages, which lack resources for speech and language processing. We focus on finding approaches which allow using data from multiple languages to improve the performance for those languages on different levels, such as feature extraction, acoustic modeling and language modeling. Under application aspects, this thesis also includes research work on non-native and Code-Switching speech

The listening talker: A review of human and algorithmic context-induced modifications of speech

International audienceSpeech output technology is finding widespread application, including in scenarios where intelligibility might be compromised - at least for some listeners - by adverse conditions. Unlike most current algorithms, talkers continually adapt their speech patterns as a response to the immediate context of spoken communication, where the type of interlocutor and the environment are the dominant situational factors influencing speech production. Observations of talker behaviour can motivate the design of more robust speech output algorithms. Starting with a listener-oriented categorisation of possible goals for speech modification, this review article summarises the extensive set of behavioural findings related to human speech modification, identifies which factors appear to be beneficial, and goes on to examine previous computational attempts to improve intelligibility in noise. The review concludes by tabulating 46 speech modifications, many of which have yet to be perceptually or algorithmically evaluated. Consequently, the review provides a roadmap for future work in improving the robustness of speech output