The Spoken Language Translator (SLT) 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.

Most spoken language translation systems developed to date rely on a pipelined architecture, in which the main stages are speech recognition, linguistic analysis, transfer, generation and speech synthesis. When making projections of error rates for systems of this kind, it is natural to assume that the error rates for the individual components are independent, making the system accuracy the product of the component accuracies. The paper reports experiments carried out using the SRI-SICSTelia Research Spoken Language Translator and a 1000-utterance sample of unseen data. The results suggest that the naive performance model leads to serious overestimates of system error rates, since there are in fact strong dependencies between the components. Predicting the system error rate on the independence assumption by simple multiplication resulted in a 16% proportional overestimate for all utterances,

Spam deobfuscation is a processing to detect obfuscated words appeared in spam emails and to convert them back to the original words for correct recognition. Lexicon tree hidden Markov model (LT-HMM) was recently shown to be useful in spam deobfuscation. However, LT-HMM suffers from a huge number of states, which is not desirable for practical applications. In this paper we present a complexity-reduced HMM, referred to as dynamically weighted HMM (DW-HMM) where the states involving the same emission probability are grouped into super-states, while preserving state transition probabilities of the original HMM. DW-HMM dramatically reduces the number of states and its state transition probabilities are determined in the decoding phase. We illustrate how we convert a LT-HMM to its associated DW-HMM. We confirm the useful behavior of DW-HMM in the task of spam deobfuscation, showing that it significantly reduces the number of states while maintaining the high accuracy. 1

With maturing speech technology, spoken dialogue systems are increasingly moving from research prototypes to fielded systems. The fielded systems however generally employ much simpler linguistic and dialogue processing strategies than the research prototypes. We describe an implemented spoken-language dialogue system for a travel planning domain which supports a mixed initiative dialogue strategy. The system accesses a commercially available travel information web-server. The system architecture combines both shallow and deep linguistic processors, partly so that a robust if shallow analysis is always available to the dialogue manager, and partly so that we can begin to examine where significant gains can be made by employing more advanced linguistic processing. We present the results of a preliminary investigation using data from a Wizard of Oz experiment. The results lend limited support to our original hypothesis that deep linguistic processing will prove useful at points where the ...

Most spoken language translation systems developed to date rely on a pipelined architecture, in which the main stages are speech recognition, linguistic analysis, transfer, generation and speech synthesis. When making projections of error rates for systems of this kind, it is natural to assume that the error rates for the individual components are independent, making the system accuracy the product of the component accuracies. The paper reports experiments carried out using the SRI-SICS-Telia Research Spoken Language Translator and a 1000-utterance sample of unseen data. The results suggest that the naive performance model leads to serious overestimates of system error rates, since there are in fact strong dependencies between the components. Predicting the system error rate on the independence assumption by simple multiplication resulted in a 16 % proportional overestimate for all utterances, and a 19 % overestimate when only utterances of length 1-10 words were considered. 1 1