Vinciarelli A., Esposito A., Andre’ E., Bonin F., Chetouani M., Cohn F. J., Cristani M., Fuhrmann F., Gilmartin E., Hammal Z., Heylen D., Kaiser R., Koutsombogera M., Potamianos A., Renals S., Riccardi G., Salah A. G. Cognitive Computation, pp. 1-17, April 2015, 2015. (Abstract | Links | BibTeX | Tags: Conversational and Interactive Systems , Machine Learning, Speech Processing) Han S., Dinarelli M., Raymond C., Lefevre F., Lehnen P., De Mori R., Moschitti A., Ney H. and Riccardi G. Comparing Stochastic Approaches to Spoken Language Understanding in Multiple Languages (Article) IEEE Trans. on Audio, Speech and Language Processing, vol. 19, no. 6, pp. 1569-1583, 2011, 2014. (Abstract | Links | BibTeX | Tags: Signal Annotation and Interpretation, Speech Processing) Griol D., Callejas Z., Lopez-Cozar R. and Riccardi G. A Domain-Independent Statistical Methodology for Dialog Management in Spoken Dialog Systems (Article) Computer Speech and Language, to be published in 2014, 2014. (Abstract | Links | BibTeX | Tags: Conversational and Interactive Systems , Speech Processing) Dinarelli M., Moschitti A. and Riccardi G. Discriminative Reranking for Spoken Language Understanding (Article) IEEE Trans. on Audio, Speech and Language Processing, vol. 20, no. 2, pp. 526-539, 2012, 2012. (Abstract | Links | BibTeX | Tags: Signal Annotation and Interpretation, Speech Processing) De Mori R., Bechet F., Hakkani-Tur D., McTear M., Riccardi G. and Tur G. Spoken Language Understanding (Article) IEEE Signal Processing Magazine vol. 25, pp.50-58 ,2008, 2008. (BibTeX | Tags: Machine Learning, Natural Language Processing, Speech Processing) Hakkani-Tur D., Riccardi G. and Tur G. An Active Approach to spoken Language Processing (Article) ACM Transactions on Speech and Language Processing, Vol. 3, No. 3, pp 1-31, 2006, 2006. (Abstract | Links | BibTeX | Tags: Machine Learning, Natural Language Processing, Speech Processing) Gupta N., Tur G., Hakkani-Tur D., Bangalore S., Riccardi G. and Rahim M. The AT&T Spoken Language Understanding System (Article) IEEE Trans. on Audio, Speech and Language Processing, volume 14, Issue 1, pp. 213-22, 2006, 2006. (Abstract | Links | BibTeX | Tags: Machine Learning, Natural Language Processing, Speech Processing) Hakkani-Tur D., Bechet F., Riccardi G. and Tur G. Beyond ASR 1-Best: Using Word Confusion Network (Article) Computer Speech and Language, volume 20, Issue 4, pp. 495-514, 2006, 2006. (Abstract | Links | BibTeX | Tags: Language Modeling, Speech Processing) Riccardi G. and Baggia P. Spoken Dialog Systems: From Theory to Technology (Article) Edizione della Normale di Pisa, 2006, 2006. (BibTeX | Tags: Conversational and Interactive Systems , Machine Learning, Speech Processing) Potamianos A., Narayanan S. and Riccardi G. Adaptive Categorical Understanding for Spoken Dialogue Systems' (Article) Potamianos A., Narayanan S and Riccardi, G., 2005. (Abstract | Links | BibTeX | Tags: Conversational and Interactive Systems , Machine Learning, Speech Processing) Gorin A., Abella A., Alonso T., Riccardi G. and Wright J. Automated Natural Spoken Dialog (Article) IEEE Computer, vol. 35, n.4, pp. 51-56, April, 2002 (invited paper), 2002. (Abstract | Links | BibTeX | Tags: Conversational and Interactive Systems , Speech Processing) Rahim M., Riccardi G., Saul L., Wright J., Buntschuh B. and Gorin A. L. Robust Numeric Recognition in Spoken Language Dialogue (Article) Speech Communication, 34, pp. 195-212, 2001, 2001. (Abstract | Links | BibTeX | Tags: Machine Learning, Signal Annotation and Interpretation, Speech Processing) Rose R. C., Yao H., Riccardi G. and Wright J. H. Speech Communication, 34, pp. 321-331, 2001, 2001. (Abstract | Links | BibTeX | Tags: Machine Learning, Signal Annotation and Interpretation, Speech Processing) Riccardi G. and Gorin A. L. Spoken language adaptation over time and state in a natural spoken dialog system (Article) IEEE Trans. on Speech and Audio, vol. 8, pp. 3-10, 2000, 2000. (Abstract | Links | BibTeX | Tags: Conversational and Interactive Systems , Language Modeling, Speech Processing) Arai K., Wright J. H., Riccardi G. and Gorin A. L. Grammar fragment acquisition using syntactic and semantic clustering (Article) Speech Communication, vol. 27, no. 1, Jan. 1999, 1999. (Abstract | Links | BibTeX | Tags: Conversational and Interactive Systems , Language Modeling, Speech Processing) Gorin A. L., Riccardi G. and Wright J. H. How may I help you? (Article) Speech Communication, vol. 23, Oct. 1997, pp. 113-127., 1997. (Abstract | Links | BibTeX | Tags: Conversational and Interactive Systems , Speech Processing) Riccardi G., Pieraccini R. and Bocchieri E. Stochastic automata for language modeling (Article) Computer Speech and Language, vol. 10(4), 1996, pp. 265-293, 1996. (Abstract | Links | BibTeX | Tags: Conversational and Interactive Systems , Language Modeling, Speech Processing) Bocchieri E., Levin E., Pieraccini R. and Riccardi G. Understanding spontaneous speech (Article) J. of the Italian Assoc. of Artificial Intelligence, Sept. 1995, 1995. (BibTeX | Tags: Machine Learning, Signal Annotation and Interpretation, Speech Processing) Mumolo E., Rebelli A. and Riccardi G. Improved multipulse algorithm for speech coding by means of adaptive Boltzmann annealing (Article) European Transactions on Telecommunications, vol. 5, no. 6, Nov. 1994, 1994. (BibTeX | Tags: Speech Processing) Mian G. A. and Riccardi G. A localization property of line spectrum pairs (Article) IEEE Trans. on Speech and Audio Proc., vol. 2, no. 4, pp. 536-539, Oct. 1994, 1994. (Links | BibTeX | Tags: Speech Processing) Fratti M., Mian G. A. and Riccardi G. An approach to parameter reoptimization in multipulse based coders (Article) IEEE Trans. Speech & Audio Proc., vol. 1, no. 4, pp. 463-465, Oct. 1993, 1993. (Links | BibTeX | Tags: Speech Processing)2015
title = {Open Challenges in Modelling, Analysis and Synthesis of Human Behaviour in Human–Human and Human–Machine Interactions},
author = {Vinciarelli A., Esposito A., Andre’ E., Bonin F., Chetouani M., Cohn F. J., Cristani M., Fuhrmann F., Gilmartin E., Hammal Z., Heylen D., Kaiser R., Koutsombogera M., Potamianos A., Renals S., Riccardi G., Salah A. G.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2015/11/CogniComp15-ChallengesHHHM-Review.pdf},
year = {2015},
date = {2015-04-01},
journal = {Cognitive Computation, pp. 1-17, April 2015},
abstract = {Modelling, analysis and synthesis of behaviour are the subject of major efforts in computing science,
especially when it comes to technologies that make sense of human–human and human–machine interactions. This article outlines some of the most important issues that still need to be addressed to ensure substantial progress in the field, namely (1) development and adoption of virtuous data collection and sharing practices, (2) shift in the focus of interest from individuals to dyads and groups, (3) endowment of artificial agents with internal representations of users and context, (4) modelling of cognitive and semantic processes underlying social behaviour and (5) identification of application domains and strategies for moving from laboratory to the real-world products.},
keywords = {Conversational and Interactive Systems , Machine Learning, Speech Processing}
}
especially when it comes to technologies that make sense of human–human and human–machine interactions. This article outlines some of the most important issues that still need to be addressed to ensure substantial progress in the field, namely (1) development and adoption of virtuous data collection and sharing practices, (2) shift in the focus of interest from individuals to dyads and groups, (3) endowment of artificial agents with internal representations of users and context, (4) modelling of cognitive and semantic processes underlying social behaviour and (5) identification of application domains and strategies for moving from laboratory to the real-world products.2014
title = {Comparing Stochastic Approaches to Spoken Language Understanding in Multiple Languages},
author = {Han S., Dinarelli M., Raymond C., Lefevre F., Lehnen P., De Mori R., Moschitti A., Ney H. and Riccardi G.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2014/11/IEEETSLP10-MultSLU.pdf},
year = {2014},
date = {2014-01-01},
journal = {IEEE Trans. on Audio, Speech and Language Processing, vol. 19, no. 6, pp. 1569-1583, 2011},
abstract = {One of the first steps in building a spoken language understanding (SLU) module for dialogue systems is the extraction of flat concepts out of a given word sequence, usually provided by an automatic speech recognition (ASR) system. In this paper, six different modeling approaches are investigated to tackle the task of concept tagging. These methods include classical, well-known generative and discriminative methods like Finite State Transducers (FSTs), Statistical Machine Translation (SMT), Maximum Entropy Markov Models (MEMMs), or Support Vector Machines (SVMs) as well as techniques recently applied to natural language processing such as Conditional Random Fields (CRFs) or Dynamic Bayesian Networks (DBNs). Following a detailed description of the models, experimental and comparative results are presented on three corpora in different languages and with different complexity. The French MEDIA corpus has already been exploited during an evaluation campaign and so a direct comparison with existing benchmarks is possible. Recently collected Italian and Polish corpora are used to test the robustness and portability of the modeling approaches. For all tasks, manual transcriptions as well as ASR inputs are considered. Additionally to single systems, methods for system combination are investigated. The best performing model on all tasks is based on conditional random fields. On the MEDIA evaluation corpus, a concept error rate of 12.6% could be achieved. Here, additionally to attribute names, attribute values have been extracted using a combination of a rule-based and a statistical approach. Applying system combination using weighted ROVER with all six systems, the concept error rate (CER) drops to 12.0%.},
keywords = {Signal Annotation and Interpretation, Speech Processing}
}
title = {A Domain-Independent Statistical Methodology for Dialog Management in Spoken Dialog Systems},
author = {Griol D., Callejas Z., Lopez-Cozar R. and Riccardi G.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2014/11/CSL14-StatisticalDialogueManager1.pdf},
year = {2014},
date = {2014-01-01},
journal = {Computer Speech and Language, to be published in 2014},
abstract = {This paper proposes a domain-independent statistical methodology to develop dialog managers for spoken dialog systems. Our methodology employs a data-driven classification procedure to generate abstract representations of system turns taking into account the previous history of the dialog. A statistical framework is also introduced for the development and evaluation of dialog systems created using the methodology, which is based on a dialog simulation technique. The benefits and flexibility of the proposed methodology have been validated by developing statistical dialog managers for four spoken dialog systems of different complexity, designed for different languages (English, Italian, and Spanish) and application domains (from transactional to problem-solving tasks). The evaluation results show that the proposed methodology allows rapid development of new dialog managers as well as to explore new dialog strategies, which permit developing new enhanced versions of already existing systems. © 2013 Elsevier Ltd. All rights reserved.},
keywords = {Conversational and Interactive Systems , Speech Processing}
}
2012
title = {Discriminative Reranking for Spoken Language Understanding},
author = {Dinarelli M., Moschitti A. and Riccardi G.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2014/11/IEEETSLP11-DRMSLU1.pdf},
year = {2012},
date = {2012-01-01},
journal = {IEEE Trans. on Audio, Speech and Language Processing, vol. 20, no. 2, pp. 526-539, 2012},
abstract = {Spoken language understanding (SLU) is concerned with the extraction of meaning structures from spoken utterances. Recent computational approaches to SLU, e.g., conditional random fields (CRFs), optimize local models by encoding several features, mainly based on simple n-grams. In contrast, recent works have shown that the accuracy of CRF can be significantly improved by modeling long-distance dependency features. In this paper, we propose novel approaches to encode all possible dependencies between features and most importantly among parts of the meaning structure, e.g., concepts and their combination. We rerank hypotheses generated by local models, e.g., stochastic finite state transducers (SFSTs) or CRF, with a global model. The latter encodes a very large number of dependencies (in the form of trees or sequences) by applying kernel methods to the space of all meaning (sub) structures. We performed comparative experiments between SFST, CRF, support vector machines (SVMs), and our proposed discriminative reranking models (DRMs) on representative conversational speech corpora in three different languages: the ATIS (English), the MEDIA (French), and the LUNA (Italian) corpora. These corpora have been collected within three different domain applications of increasing complexity: informational, transactional, and problem-solving tasks, respectively. The results show that our DRMs consistently outperform the state-of-the-art models based on CRF.},
keywords = {Signal Annotation and Interpretation, Speech Processing}
}
2008
title = {Spoken Language Understanding},
author = {De Mori R., Bechet F., Hakkani-Tur D., McTear M., Riccardi G. and Tur G.},
year = {2008},
date = {2008-01-01},
journal = {IEEE Signal Processing Magazine vol. 25, pp.50-58 ,2008},
keywords = {Machine Learning, Natural Language Processing, Speech Processing}
}
2006
title = {An Active Approach to spoken Language Processing},
author = {Hakkani-Tur D., Riccardi G. and Tur G.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2014/11/acm-tslp-06.pdf},
year = {2006},
date = {2006-01-01},
journal = {ACM Transactions on Speech and Language Processing, Vol. 3, No. 3, pp 1-31, 2006},
abstract = {State of the art data-driven speech and language processing systems require a large amount of human intervention ranging from data annotation to system prototyping. In the traditional supervised passive approach, the system is trained on a given number of annotated data samples and evaluated using a separate test set. Then more data is collected arbitrarily, annotated, and the whole cycle is repeated. In this article, we propose the active approach where the system itself selects its own training data, evaluates itself and re-trains when necessary. We first employ active learning which aims to automatically select the examples that are likely to be the most informative for a given task. We use active learning for both selecting the examples to label and the examples to re-label in order to correct labeling errors. Furthermore, the system automatically evaluates itself using active evaluation to keep track of the unexpected events and decides on-demand to label more examples. The active approach enables dynamic adaptation of spoken language processing systems to unseen or unexpected events for nonstationary input while reducing the manual annotation effort significantly. We have evaluated the active approach with the AT&T spoken dialog system used for customer care applications. In this article, we present our results for both automatic speech recognition and spoken language understanding. Categories and Subject Descriptors: I.2.7 [Artificial Intelligence]: Natural Language Processing—Speech recognition and synthesis; I.5.1 [Pattern Recognition]: Models—Statistical General Terms: Algorithms, Languages, Performance Additional Key Words and Phrases: Passive learning, active learning, adaptive learning, unsupervised learning, active evaluation, spoken language understanding, automatic speech recognition, spoken dialog systems, speech and language processing},
keywords = {Machine Learning, Natural Language Processing, Speech Processing}
}
title = {The AT&T Spoken Language Understanding System},
author = {Gupta N., Tur G., Hakkani-Tur D., Bangalore S., Riccardi G. and Rahim M.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2014/11/IEEE-SAP-2005-SLU.pdf},
year = {2006},
date = {2006-01-01},
journal = {IEEE Trans. on Audio, Speech and Language Processing, volume 14, Issue 1, pp. 213-22, 2006},
abstract = {Spoken language understanding (SLU) aims at extracting meaning from natural language speech. Over the past decade, a variety of practical goal-oriented spoken dialog systems have been built for limited domains. SLU in these systems ranges from understanding predetermined phrases through fixed grammars, extracting some predefined named entities, extracting users’ intents for call classification, to combinations of users’ intents and named entities. In this paper, we present the SLU system of VoiceTone ® (a service provided by AT&T where AT&T develops, deploys and hosts spoken dialog applications for enterprise customers). The SLU system includes extracting both intents and the named entities from the users’ utterances. For intent determination, we use statistical classifiers trained from labeled data, and for named entity extraction we use rule-based fixed grammars. The focus of our work is to exploit data and to use machine learning techniques to create scalable SLU systems which can be quickly deployed for new domains with minimal human intervention. These objectives are achieved by 1) using the predicate-argument representation of semantic content of an utterance; 2) extending statistical classifiers to seamlessly integrate hand crafted classification rules with the rules learned from data; and 3) developing an active learning framework to minimize the human labeling effort for quickly building the classifier models and adapting them to changes. We present an evaluation of this system using two deployed applications of VoiceTone},
keywords = {Machine Learning, Natural Language Processing, Speech Processing}
}
title = {Beyond ASR 1-Best: Using Word Confusion Network},
author = {Hakkani-Tur D., Bechet F., Riccardi G. and Tur G.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2014/11/CSL-pivot-slu.pdf},
year = {2006},
date = {2006-01-01},
journal = {Computer Speech and Language, volume 20, Issue 4, pp. 495-514, 2006},
abstract = {We are interested in the problem of robust understanding from noisy spontaneous speech input. With the advances in automated speech recognition (ASR), there has been increasing interest in spoken language understanding (SLU). A challenge in large vocabulary spoken language understanding is robustness to ASR errors. State of the art spoken language understanding relies on the best ASR hypotheses (ASR 1-best). In this paper, we propose methods for a tighter integration of ASR and SLU using word confusion networks (WCNs). WCNs obtained from ASR word graphs (lattices) provide a compact representation of multiple aligned ASR hypotheses along with word confidence scores, without compromising recognition accuracy. We present our work on exploiting WCNs instead of simply using ASR one-best hypotheses. In this work, we focus on the tasks of named entity detection and extraction and call classification in a spoken dialog system, although the idea is more general and applicable to other spoken language processing tasks. For named entity detection, we have improved the F-measure by using both word lattices and WCNs, 6–10% absolute. The processing of WCNs was 25 times faster than lattices, which is very important for real-life applications. For call classification, we have shown between 5% and 10% relative reduction in error rate using WCNs compared to ASR 1-best output. Ó 2005 Elsevier Ltd. All rights reserved.},
keywords = {Language Modeling, Speech Processing}
}
title = {Spoken Dialog Systems: From Theory to Technology},
author = {Riccardi G. and Baggia P.},
year = {2006},
date = {2006-01-01},
journal = {Edizione della Normale di Pisa, 2006},
keywords = {Conversational and Interactive Systems , Machine Learning, Speech Processing}
}
2005
title = {Adaptive Categorical Understanding for Spoken Dialogue Systems'},
author = {Potamianos A., Narayanan S. and Riccardi G.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2014/11/ieee_adapt-categ-05.pdf},
year = {2005},
date = {2005-01-01},
journal = {Potamianos A., Narayanan S and Riccardi, G.},
abstract = {IEEE Trans. on Speech and Audio, vol. 13, n.3 , pp. 321-329, 2005},
keywords = {Conversational and Interactive Systems , Machine Learning, Speech Processing}
}
2002
title = {Automated Natural Spoken Dialog},
author = {Gorin A., Abella A., Alonso T., Riccardi G. and Wright J.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2014/11/computer_magazine_2002.pdf},
year = {2002},
date = {2002-01-01},
journal = {IEEE Computer, vol. 35, n.4, pp. 51-56, April, 2002 (invited paper)},
abstract = {Engineers have long sought to design systems that understand and act upon spoken language. Extracting meaning from natural, unconstrained speech over the telephone is technically challenging, and quantifying semantic content is crucial for engineering and evaluating such systems.},
keywords = {Conversational and Interactive Systems , Speech Processing}
}
2001
title = {Robust Numeric Recognition in Spoken Language Dialogue},
author = {Rahim M., Riccardi G., Saul L., Wright J., Buntschuh B. and Gorin A. L.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2014/11/numericlang-speechcomm-2001.pdf},
year = {2001},
date = {2001-11-01},
journal = {Speech Communication, 34, pp. 195-212, 2001},
abstract = {This paper addresses the problem of automatic numeric recognition and understanding in spoken language dialogue. We show that accurate numeric understanding in ̄uent unconstrained speech demands maintaining robustness at several dierent levels of system design, including acoustic, language, understanding and dialogue. We describe a robust system for numeric recognition and present algorithms for feature extraction, acoustic and language modeling, discriminative training, utterance veri®cation and numeric understanding and validation. Experimental results from a ®eld-trial of a spoken dialogue system are presented that include customers\' responses to credit card and telephone number requests. Ó 2001 Elsevier Science B.V. All rights reserved.},
keywords = {Machine Learning, Signal Annotation and Interpretation, Speech Processing}
}
title = {Integration of Utterance Verification with Statistical Language Modeling and Spoken Language Understanding},
author = {Rose R. C., Yao H., Riccardi G. and Wright J. H.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2014/11/uv-speechcomm-2001.pdf},
year = {2001},
date = {2001-01-01},
journal = {Speech Communication, 34, pp. 321-331, 2001},
abstract = {Methods for utterance veri®cation (UV) and their integration into statistical language modeling and understanding formalisms for a large vocabulary spoken understanding system are presented. The paper consists of three parts. First, a set of acoustic likelihood ratio (LR) based UV techniques are described and applied to the problem of rejecting portions of a hypothesized word string that may have been incorrectly decoded by a large vocabulary continuous speech recognizer. Second, a procedure for integrating the acoustic level con®dence measures with the statistical language model is described. Finally, the eect of integrating acoustic level con®dence into the spoken language understanding unit (SLU) in a call-type classi®cation task is discussed. These techniques were evaluated on utterances collected from a highly unconstrained call routing task performed over the telephone network. They have been evaluated in terms of their ability to classify utterances into a set of 15 call-types that are accepted by the application. Ó 2001 Elsevier Science B.V. All rights reserved.},
keywords = {Machine Learning, Signal Annotation and Interpretation, Speech Processing}
}
2000
title = {Spoken language adaptation over time and state in a natural spoken dialog system},
author = {Riccardi G. and Gorin A. L.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2014/11/IEEETSLP00-LMAdapt.pdf},
year = {2000},
date = {2000-01-01},
journal = {IEEE Trans. on Speech and Audio, vol. 8, pp. 3-10, 2000},
abstract = {We are interested in adaptive spoken dialog systems for automated services. Peoples’ spoken language usage varies over time for a given task, and furthermore varies depending on the state of the dialog. Thus, it is crucial to adapt automatic speech recognition (ASR) language models to these varying conditions. We characterize and quantify these variations based on a database of 30K user-transactions with AT&T’s experimental How May I Help You? spoken dialog system. We describe a novel adaptation algorithm for language models with time and dialog-state varying parameters. Our language adaptation framework allows for recognizing and understanding unconstrained speech at each stage of the dialog, enabling context-switching and error recovery. These models have been used to train state-dependent ASR language models. We have evaluated their performance with respect to word accuracy and perplexity over time and dialog states. We have achieved a reduction of 40% in perplexity and of 8.4% in word error rate over the baseline system, averaged across all dialog states.},
keywords = {Conversational and Interactive Systems , Language Modeling, Speech Processing}
}
1999
title = {Grammar fragment acquisition using syntactic and semantic clustering},
author = {Arai K., Wright J. H., Riccardi G. and Gorin A. L.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2014/11/fragclustering-speechcomm-19981.pdf},
year = {1999},
date = {1999-01-01},
journal = {Speech Communication, vol. 27, no. 1, Jan. 1999},
abstract = {A new method for automatically acquiring Fragments for understanding ̄uent speech is proposed. The goal of this method is to generate a collection of Fragments, each representing a set of syntactically and semantically similar phrases. First, phrases observed frequently in the training set are selected as candidates. Each candidate phrase has three associated probability distributions: of following contexts, of preceding contexts, and of associated semantic actions. The similarity between candidate phrases is measured by applying the Kullback±Leibler distance to these three probability distributions. Candidate phrases that are close in all three distances are clustered into a Fragment. Salient sequences of these Fragments are then automatically acquired, and exploited by a spoken language understanding module to classify calls in AT&T\'s ``How may I help you?\'\' task. These Fragments allow us to generalize unobserved phrases. For instance, they detected 246 phrases in the test-set that were not present in the training-set. This result shows that unseen phrases can be automatically discovered by our new method. Experimental results show that 2.8% of the improvement in call-type classi®catio},
keywords = {Conversational and Interactive Systems , Language Modeling, Speech Processing}
}
1997
title = {How may I help you?},
author = {Gorin A. L., Riccardi G. and Wright J. H.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2014/11/specom97.pdf},
year = {1997},
date = {1997-01-01},
journal = {Speech Communication, vol. 23, Oct. 1997, pp. 113-127.},
abstract = {We are interested in providing automated services via natural spoken dialog systems. By natural, we mean that the machine understands and acts upon what people actually say, in contrast to what one would like them to say. There are many issues that arise when such systems are targeted for large populations of non-expert users. In this paper, we focus on the task of automatically routing telephone calls based on a user’s fluently spoken response to the open-ended prompt of ‘‘How may I help you?’’. We first describe a database generated from 10,000 spoken transactions between customers and human agents. We then describe methods for automatically acquiring language models for both recognition and understanding from such data. Experimental results evaluating call-classification from speech are reported for that database. These methods have been embedded within a spoken dialog system, with subsequent processing for information retrieval and formfilling. q 1997 Elsevier Science B.V.},
keywords = {Conversational and Interactive Systems , Speech Processing}
}
1996
title = {Stochastic automata for language modeling},
author = {Riccardi G., Pieraccini R. and Bocchieri E.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2014/11/csl96.pdf},
year = {1996},
date = {1996-01-01},
journal = { Computer Speech and Language, vol. 10(4), 1996, pp. 265-293},
abstract = {Stochastic language models are widely used in spoken language understanding to recognize and interpret the speech signal: the speech samples are decoded into word transcriptions by means of acoustic and syntactic models and then interpreted according to a semantic model. Both for speech recognition and understanding, search algorithms use stochastic models to extract the most likely uttered sentence and its correspondent interpretation. The design of the language models has to be effective in order to mostly constrain the search algorithms and has to be efficient to comply with the storage space limits. In this work we present the Variable N-gram Stochastic Automaton (VNSA) language model that provides a unified formalism for building a wide class of language models. First, this approach allows for the use of accurate language models for large vocabulary speech recognition by using the standard search algorithm in the one-pass Viterbi decoder. Second, the unified formalism is an effective approach to incorporate different sources of information for computing the probability of word sequences. Third, the VNSAs are well suited for those applications where speech and language decoding cascades are implemented through weighted rational transductions. The VNSAs have been compared to standard bigram and trigram language models and their reduced set of parameters does not affect by any means the performances in terms of perplexity. The design of a stochastic language model through the VNSA is described and applied to word and phrase class-based language models. The effectiveness of VNSAs has been tested within the Air Travel Information System (ATIS) task to build the language model for th},
keywords = {Conversational and Interactive Systems , Language Modeling, Speech Processing}
}
1995
title = {Understanding spontaneous speech},
author = {Bocchieri E., Levin E., Pieraccini R. and Riccardi G.},
year = {1995},
date = {1995-01-01},
journal = {J. of the Italian Assoc. of Artificial Intelligence, Sept. 1995},
keywords = {Machine Learning, Signal Annotation and Interpretation, Speech Processing}
}
1994
title = {Improved multipulse algorithm for speech coding by means of adaptive Boltzmann annealing},
author = {Mumolo E., Rebelli A. and Riccardi G.},
year = {1994},
date = {1994-01-01},
journal = {European Transactions on Telecommunications, vol. 5, no. 6, Nov. 1994},
keywords = {Speech Processing}
}
title = {A localization property of line spectrum pairs},
author = {Mian G. A. and Riccardi G.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2014/11/IEEE_LSP.pdf},
year = {1994},
date = {1994-01-01},
journal = {IEEE Trans. on Speech and Audio Proc., vol. 2, no. 4, pp. 536-539, Oct. 1994},
keywords = {Speech Processing}
}
1993
title = {An approach to parameter reoptimization in multipulse based coders},
author = {Fratti M., Mian G. A. and Riccardi G.},
url = {https://sisl.disi.unitn.it/wp-content/uploads/2014/11/IEEE_Multipulse.pdf},
year = {1993},
date = {1993-01-01},
journal = {IEEE Trans. Speech & Audio Proc., vol. 1, no. 4, pp. 463-465, Oct. 1993},
keywords = {Speech Processing}
}