@misc{BaumannGeiserMenzeletal., author = {Baumann, Timo and Geiser, Dorothee and Menzel, Wolfgang and Mohr, Mario and Neef, Svenja and Nykamp, S{\"o}ren and Rokita, Nils}, title = {Concurrent Sub-turn Interaction Specification and Dialogue Management with an Application to Interactive Storytelling}, series = {GSCL Workshop Gesprochene Sprache und Sprachverarbeitung (GSS), Darmstadt, Germany, 2013}, journal = {GSCL Workshop Gesprochene Sprache und Sprachverarbeitung (GSS), Darmstadt, Germany, 2013}, address = {Darmstadt, Germany}, abstract = {Conventional dialogue management centers around an interaction style that is best described as a ping-pong game, with full turns being the units at which speech is delivered (and expected) by the system, which greatly simplifies the interaction management, delivery and understanding components of the system. While the resulting mode of interaction works well for task-based systems, it is insufficient for more conversational interaction styles, where content is delivered and grounded in units finer than full turns (Poesio and Traum 1997) and where turns are delivered concurrently by both interlocutors and hence overlap more frequently. One domain with particularly frequent overlapping contributions is interactive storytelling: a storyteller that is responsive to listeners will integrate their feedback immediately while still speaking a current contribution, and listener's remarks or propositions regarding the story will typically be uttered immediately when the related content is delivered. We present our work on a dialogue manager that leverages recent advances in incremental speech delivery and reception (Baumann 2013) to provide for an interactive and concurrent storytelling experience. Our system uses an interaction graph, which uses a word-by-word granularity and allows to specify for individual stretches of speech where (and with what content) users may interrupt/comment, how this is interpreted, and how it is integrated into the storytelling process, thus providing for in-utterance alternatives to be spoken even without audibly interrupting the system's ongoing utterance. The system's interaction graph is specified in an XML language and may be hand-crafted by a story designer but can also be automatically generated. In our current system, speech recognition results are interpreted only when the user utterance is finished; however, we plan to integrate incremental speech recognition and understanding capabilities, and to immediately react to the start of user contributions.}, language = {en} } @inproceedings{SchenkeBaumann, author = {Schenke, Diana Marie and Baumann, Timo}, title = {Controlled Diversity: Length-optimized Natural Language Generation}, series = {Proceedings of the ISCA/ITG Workshop on Diversity in Large Speech and Language Models, February 20, 2025 in Berlin, Germany}, booktitle = {Proceedings of the ISCA/ITG Workshop on Diversity in Large Speech and Language Models, February 20, 2025 in Berlin, Germany}, editor = {M{\"o}ller, Sebastian and Knoeferle, Pia and Schulte, Britta and Feldhus, Nils}, doi = {10.48550/arXiv.2502.19347}, url = {http://nbn-resolving.de/urn:nbn:de:bvb:898-opus4-80534}, pages = {8}, abstract = {LLMs are not generally able to adjust the length of their outputs based on strict length requirements, a capability that would improve their usefulness in applications that require adherence to diverse user and system requirements. We present an approach to train LLMs to acquire this capability by augmenting existing data and applying existing fine-tuning techniques, which we compare based on the trained models' adherence to the length requirement and overall response quality relative to the baseline model. Our results demonstrate that these techniques can be successfully applied to train LLMs to adhere to length requirements, with the trained models generating texts which better align to the length requirements. Our results indicate that our method may change the response quality when using training data that was not generated by the baseline model. This allows simultaneous alignment to another training objective in certain scenarios, but is undesirable otherwise. Training on a dataset containing the model's own responses eliminates this issue.}, language = {en} } @unpublished{AmanpourBaumannHofmann, author = {Amanpour, Ali and Baumann, Timo and Hofmann, Ulrich G.}, title = {Optimizing Neural Data Analysis: Determining Minimum Recording Length for Unambigous Signal Processing}, doi = {10.35096/othr/pub-8060}, url = {http://nbn-resolving.de/urn:nbn:de:bvb:898-opus4-80606}, pages = {5}, abstract = {Advanced silicon electrode arrays facilitate the recording of thousands of neurons, generating extensive neural data that imposes a significant burden on researchers and processing algorithms. Thus, real-time analysis pipelines are gaining increasing value, while at the same time having to deal with non-stationary and noisy data. We intend to apply Machine Learning (ML) algorithms to a dense set of recordings from rat brains in order to prepare a functional atlas, correlating neuronal signals with anatomical position. While doing so, we needed to decide on a rational way which recording snippet length would best represent the original, longer source sequence and thus suffices to be further processed for anatomical correlation. We implemented an algorithm to evaluate the spectral information of systematically length varied records based on similarity to the original record. For our dataset a recording duration of 3 seconds satisfied moderate requirements across all channels, thus allowing us to reduce computational load for ongoing ML classification of microprobe sourced electrophysiologic signals.}, language = {en} } @incollection{MoralesBaumannPerezetal., author = {Morales, Juan A. and Baumann, Timo and P{\´e}rez, Jos{\´e} L. and Peinado, Antonio M. and G{\´o}mez, {\´A}ngel M.}, title = {Implementaci{\`o}n de un reconocedor distribuido de voz en tiempo real sobre IP}, series = {Actas de las IV Jornadas en Tecnolog{\´i}as del Habla}, booktitle = {Actas de las IV Jornadas en Tecnolog{\´i}as del Habla}, abstract = {En este trabajo se presenta una implementaci{\´o}n de unsistema de reconocimiento distribuido del habla en tiem-po real para su aplicaci{\´o}n en un entorno de Internet. Desa-rrollado como una aplicaci{\´o}n cliente-servidor, el clien-te hace uso del front-end est{\´a}ndar definido por la ETSI.Incluye un detector de voz para s{\´o}lo enviar informaci{\´o}ncuando el locutor habla y la transmisi{\´o}n de la informaci{\´o}nse hace de acuerdo al RFC 3557. El servidor realiza el re-conocimiento utilizando programaci{\´o}n din{\´a}mica, incluyea su vez un detector de voz y utiliza la t{\´e}cnica de mitiga-ci{\´o}n de p{\´e}rdida de paquetes propuesta en el est{\´a}ndar de laETSI. La modularidad del dise{\~n}o permite la utilizaci{\´o}n decualquier otro reconocedor sin que por ello se vean afec-tados los clientes. Las pruebas reales de funcionamientopara la aplicaci{\´o}n particular desarrollada han demostradouna alta fiabilidad en varias condiciones de transmisi{\´o}n.}, language = {es} } @article{NigelGWardVegaBaumann, author = {Nigel G. Ward, and Vega, Alejandro and Baumann, Timo}, title = {Prosodic and Temporal Features for Language Modeling for Dialog}, series = {Speech Communication}, volume = {54}, journal = {Speech Communication}, number = {2}, publisher = {ELSEVIER}, doi = {10.1016/j.specom.2011.07.009}, pages = {161 -- 174}, abstract = {If we can model the cognitive and communicative processes underlying speech, we should be able to better predict what a speaker will do. With this idea as inspiration, we examine a number of prosodic and timing features as potential sources of information on what words the speaker is likely to say next. In spontaneous dialog we find that word probabilities do vary with such features. Using perplexity as the metric, the most informative of these included recent speaking rate, volume, and pitch, and time until end of utterance. Using simple combinations of such features to augment trigram language models gave up to a 8.4\% perplexity benefit on the Switchboard corpus, and up to a 1.0\% relative reduction in word error rate (0.3\% absolute) on the Verbmobil II corpus.}, language = {en} } @incollection{LohmannEichhornBaumann, author = {Lohmann, Kris and Eichhorn, Ole and Baumann, Timo}, title = {Generating Situated Assisting Utterances to Facilitate Tactile-Map Understanding: A Prototype System}, series = {Proceedings of the Third Workshop on Speech and Language Processing for Assistive Technologies (SLPAT), Montreal, Canada, June 7-8, 2012}, booktitle = {Proceedings of the Third Workshop on Speech and Language Processing for Assistive Technologies (SLPAT), Montreal, Canada, June 7-8, 2012}, publisher = {Association for Computational Linguistics}, pages = {56 -- 65}, abstract = {Tactile maps are important substitutes for visual maps for blind and visually impaired people and the efficiency of tactile-map reading can largely be improved by giving assisting utterances that make use of spatial language. In this paper, we elaborate earlier ideas for a system that generates such utterances and present a prototype implementation based on a semantic conceptualization of the movements that the map user performs. A worked example shows the plausibility of the solution and the output that the prototype generates given input derived from experimental data.}, language = {en} } @inproceedings{Baumann, author = {Baumann, Timo}, title = {Feedback in Adaptive Interactive Storytelling}, series = {13th annual conference of the International Speech Communication Association 2012 (INTERSPEECH 2012): Portland, Oregon, USA, 9 - 13 September 2012, Interdisciplinary Workshop on Feedback Behaviors in Dialog}, booktitle = {13th annual conference of the International Speech Communication Association 2012 (INTERSPEECH 2012): Portland, Oregon, USA, 9 - 13 September 2012, Interdisciplinary Workshop on Feedback Behaviors in Dialog}, address = {Stevenson, USA}, isbn = {978-1-62276-759-5}, abstract = {Telling stories is different from reading out text: a speakerreponds to the listener's feedback and incorporates thisinto the ongoing talk. However, current computer sys-tems are unable to do this and instead just non-attentivelyread out a text, disregarding all feedback (or the absencethereof). I propose and discuss an idea for a small researchproject and a plan for how an attentive listening storytellercan be built.}, language = {en} } @inproceedings{Baumann, author = {Baumann, Timo}, title = {Integrating Prosodic Modelling with Incremental Speech Recognition}, series = {Proceedings of DiaHolmia, the 13th International Workshop on the Semantics and Pragmatics of Dialogue (SEMDIAL 2009), 24-26 JUNE, 2009, Sttockhom, Sweden}, booktitle = {Proceedings of DiaHolmia, the 13th International Workshop on the Semantics and Pragmatics of Dialogue (SEMDIAL 2009), 24-26 JUNE, 2009, Sttockhom, Sweden}, publisher = {Royal Institute of Technology}, pages = {143 -- 144}, abstract = {We describe ongoing and proposed work concerning incremental prosody extraction and classification for a spoken dialogue system. The system described will be tightly integrated with the SDS's speech recogntion which also works incrementally. The proposed architecture should allow for more control over the user interaction experience, for example allowing more precise and timely end-of-utterance vs. hesitation distinction, and auditive or visual back-channel generation.}, language = {en} } @masterthesis{Baumann, type = {Bachelor Thesis}, author = {Baumann, Timo}, title = {Automatische Erkennung von Akzentuierungen und Phrasierungen in Sprachsynthesekorpora}, address = {Hamburg}, language = {de} } @inproceedings{PenzkoferBaumann, author = {Penzkofer, Vinzent and Baumann, Timo}, title = {Evaluating and Fine-Tuning Retrieval-Augmented Language Models to Generate Text With Accurate Citations}, series = {Proceedings of the 20th Conference on Natural Language Processing (KONVENS 2024), September 10-13, 2024, Vienna, Austria}, booktitle = {Proceedings of the 20th Conference on Natural Language Processing (KONVENS 2024), September 10-13, 2024, Vienna, Austria}, publisher = {Association for Computational Linguistics}, organization = {Austrian Research Institute for Artificial Intelligence}, pages = {57 -- 64}, abstract = {Retrieval Augmented Generation (RAG) is becoming an essential tool for easily accessing large amounts of textual information. However, it is often challenging to determine whether the information in a given response originates from the retrieved context, the training, or is a result of hallucination. Our contribution in this area is twofold. Firstly, we demonstrate how existing datasets for information retrieval evaluation can be used to assess the ability of Large Language Models (LLMs) to correctly identify relevantsources. Our findings indicate that there are notable discrepancies in the performance of different current LLMs in this task. Secondly, we utilise the datasets and metrics for citation evaluation to enhance the citation quality of small open-weight LLMs through fine-tuning. We achieve significant performance gains in this task, matching the results of much larger models.}, language = {en} } @inproceedings{BaumannEllerGagarina, author = {Baumann, Timo and Eller, Korbinian and Gagarina, Natalia}, title = {BERT-based Annotation of Oral Texts Elicited via Multilingual Assessment Instrument for Narratives}, series = {Proceedings of the the 6th Workshop on Narrative Understanding (WNU), November 12th to November 16th 2024, Miami, Florida, USA}, booktitle = {Proceedings of the the 6th Workshop on Narrative Understanding (WNU), November 12th to November 16th 2024, Miami, Florida, USA}, editor = {Lal, Yash Kumar and Clark, Elizabeth and Iyyer, Mohit and Chaturvedi, Snigdha and Brei, Anneliese and Brahman, Faeze and Chandu, Khyathi Raghavi}, publisher = {Association for Computational Linguistics}, address = {Stroudsburg, PA, USA}, doi = {10.18653/v1/2024.wnu-1.16}, pages = {99 -- 104}, abstract = {We investigate how NLP can help annotate the structure and complexity of oral narrative texts elicited via the Multilingual Assessment Instrument for Narratives (MAIN). MAIN is a theory-based tool designed to evaluate the narrative abilities of children who are learning one or more languages from birth or early in their development. It provides a standardized way to measure how well children can comprehend and produce stories across different languages and referential norms for children between 3 and 12 years old. MAIN has been adapted to over ninety languages and is used in over 65 countries. The MAIN analysis focuses on story structure and story complexity which are typically evaluated manually based on scoring sheets. We here investigate the automation of this process using BERT-based classification which already yields promising results.}, language = {en} } @inproceedings{MuehlhausenGomezLaueretal., author = {M{\"u}hlhausen, Sara and Gomez, Sarah and Lauer, Norina and Baumann, Timo}, title = {Cross lingual transfer learning does not improve aphasic speech recognition}, series = {Elektronische Sprachsignalverarbeitung 2025: Tagungsband der 36. Konferenz Halle/Saale, 05.-07. M{\"A}RZ 2025}, booktitle = {Elektronische Sprachsignalverarbeitung 2025: Tagungsband der 36. Konferenz Halle/Saale, 05.-07. M{\"A}RZ 2025}, editor = {Grawunder, Sven}, publisher = {TUDpress}, address = {Dresden}, isbn = {978-3-95908-803-9}, issn = {0940-6832}, doi = {10.35096/othr/pub-8051}, url = {http://nbn-resolving.de/urn:nbn:de:bvb:898-opus4-80518}, pages = {8}, abstract = {In addressing the particular linguistic challenges posed by patients suffering from aphasia, a language disorder, this paper proposes a fine-tuning approach to enhance the speech recognition capabilities of existing models. The available aphasic research data in German is highly limited. To address this constraint, we propose a cross-lingual transfer approach to utilize English data to improve performance in German. This advancement aims to support the development of a therapy platform tailored for patients with aphasia. For the base speech recognition model, we choose to use OpenAI's Whisper model, and for fine-tuning, we make use of TalkBank's AphasiaBank. The experimental findings demonstrate that the transcription of aphasic audio with Whisper is less successful than non-aphasic audio. However, fine-tuning the transcription in the respective language resulted in an enhancement of its quality. In contrast, fine-tuning the transcription in another language and expecting a transfer of the learned aphasic speech properties led to a deterioration in its quality.}, subject = {Automatische Spracherkennung}, language = {en} } @incollection{BaumannHusseinMeyerSickendiek, author = {Baumann, Timo and Hussein, Hussein and Meyer-Sickendiek, Burkhard}, title = {Free Verse Prosodies: Identifying and Classifying Spoken Poetry Using Literary and Computational Perspectives (Rhythmicalizer)}, series = {Mixing Methods: Practical Insights from the Humanities in the Digital Age}, booktitle = {Mixing Methods: Practical Insights from the Humanities in the Digital Age}, editor = {Schneider, Birgit and L{\"o}ffler, Beate and Mager, Tino and Hein, Carola}, publisher = {Bielefeld University Press}, address = {Bielefeld}, doi = {10.1515/9783839469132-018}, pages = {167 -- 186}, abstract = {At least 80\% of modern and postmodern poems exhibit neither rhyme nor metrical schemes such as iamb or trochee. However, does this mean that they are free of any rhythmical features?TheUS American research onfree verse prosody claimsthe opposite: Modern poets like Whitman, the Imagists, the Beat poets and contemporary Slam poets have developed a postmetrical idea of prosody, using rhythmical features of everyday language, prose, and musical styles like Jazz or Hip Hop. It has spawned a large and complex variety intheir poetic prosodies which,however,appearto bemuchharderto quantify and regularize than traditional patterns. In our project, we examinethe largest portal for spoken poetry Lyrikline and analysed and classified such rhythmical patterns by using pattern recognition and classification techniques. We integrate a human-in-the-loop approach in which we interleave manual annotation with computational modelling and data-based analysis. Our results are integrated into the website of Lyrikline. Our follow-up project makes our research results available to a wider audience, in particular to high school-level teaching.}, language = {en} } @inproceedings{BaumannBussAttereretal., author = {Baumann, Timo and Buß, Okko and Atterer, Michaela and Schlangen, David}, title = {Evaluating the Potential Utility of ASR N-Best Lists for Incremental Spoken Dialogue Systems}, series = {Proceedings of Interspeech 2009 : speech and intelligence ; 6 - 10 September, 2009, Brighton, UK}, booktitle = {Proceedings of Interspeech 2009 : speech and intelligence ; 6 - 10 September, 2009, Brighton, UK}, publisher = {ISCA}, address = {Brighton, UK}, doi = {10.21437/Interspeech.2009-318}, pages = {1031 -- 1034}, abstract = {The potential of using ASR n-best lists for dialogue systems has often been recognised (if less often realised): it is often the case that even when the top-ranked hypothesis is erroneous, a bet- ter one can be found at a lower rank. In this paper, we describe metrics for evaluating whether the same potential carries over to incremental dialogue systems, where ASR output is consumed and reacted upon while speech is still ongoing. We show that even small N can provide an advantage for semantic process- ing, at a cost of a computational overhead.}, language = {en} } @inproceedings{BaumannAttererSchlangen, author = {Baumann, Timo and Atterer, Michaela and Schlangen, David}, title = {Assessing and Improving the Performance of Speech Recognition for Incremental Systems}, series = {NAACL '09: Proceedings of Human Language Technologies: The 2009 Annual Conference of the North American Chapter of the Association for Computational Linguistics, May 31 - June 5, 2009, Boulder, Colorado, USA}, booktitle = {NAACL '09: Proceedings of Human Language Technologies: The 2009 Annual Conference of the North American Chapter of the Association for Computational Linguistics, May 31 - June 5, 2009, Boulder, Colorado, USA}, publisher = {Association for Computational Linguistics}, isbn = {978-1-932432-41-1}, pages = {380 -- 388}, abstract = {In incremental spoken dialogue systems, par- tial hypotheses about what was said are re- quired even while the utterance is still ongo- ing. We define measures for evaluating the quality of incremental ASR components with respect to the relative correctness of the par- tial hypotheses compared to hypotheses that can optimize over the complete input, the tim- ingof hypothesisformationrelative to the por- tion ofthe inputthey areabout, andhypothesis stability, defined as the number of times they are revised. We show that simple incremen- tal post-processing can improve stability dra- matically, at the cost of timeliness (from 90\% of edits of hypotheses being spurious down to 10\% at a lag of 320ms). The measures are not independent,and we show how system de- signers can find a desired operating point for their ASR. To our knowledge, we are the first to suggest and examine a variety of measures for assessing incremental ASR and improve performance on this basis.}, language = {en} } @inproceedings{BaumannBussSchlangen, author = {Baumann, Timo and Buß, Okko and Schlangen, David}, title = {InproTK in Action: Open-Source Software for Building German-Speaking Incremental Spoken Dialogue Systems}, series = {Electronic speech signal processing 2010 : proceedings of the 21st conference, Berlin, 8 - 10 September 2010}, booktitle = {Electronic speech signal processing 2010 : proceedings of the 21st conference, Berlin, 8 - 10 September 2010}, publisher = {TUDpress}, address = {Berlin, Germany}, pages = {204 -- 211}, abstract = {We present INPROTK, a toolkit for building incremental spoken dia-logue systems. Incremental spoken dialogue systems (systems that may react whilethe user's utterance is ongoing) are a fairly recent research topic and allow for ex-citing new features. Even though toolkits exist that help in building conventionaldialogue systems, INPROTK offers both a tested architecture for building incre-mental SDSs as well as many of the building blocks necessary when building suchsystems. With INPROTK a researcher can avoid many of the technical difficulties,which hopefully further fosters research in this area.}, language = {en} } @inproceedings{AttererBaumannSchlangen, author = {Atterer, Michaela and Baumann, Timo and Schlangen, David}, title = {No Sooner Said Than Done: Testing the Incrementality of Semantic Interpretations of Spontaneous Speech}, series = {Proceedings of Interspeech 2009 : 6 - 10 September 2009, Brighton, U.K.}, booktitle = {Proceedings of Interspeech 2009 : 6 - 10 September 2009, Brighton, U.K.}, publisher = {International Speech Communication Association}, address = {Brighton, UK}, doi = {10.21437/Interspeech.2009-539}, pages = {1855 -- 1858}, abstract = {Ideally, a spoken dialogue system should react without much delay to a user's utterance. Such a system would already select an object, for instance, before the user has finished her utterance about moving this particular object to a particular place. A prerequisite for such a prompt reaction is that semantic representations are built up on the fly and passed on to other modules. Few approaches to incremental semantics construction exist, and, to our knowledge, none of those has been systematically tested on a spontaneous speech corpus. In this paper, we develop measures to test empirically on transcribed spontaneous speech to what extent we can create semantic interpretation on the fly with an incremental semantic chunker that builds a frame semantics.}, language = {en} } @inproceedings{BaumannSchlangen, author = {Baumann, Timo and Schlangen, David}, title = {Predicting the Micro-Timing of User Input for an Incremental Spoken Dialogue System that Completes a User's Ongoing Turn}, series = {SIGDIAL 2011 Conference, 12th annual meeting of the Special Interest Group on Discourse and Dialogue, Co-located with ACL HLT 2011, Portland, Oregon, 17 - 18 June 2011}, booktitle = {SIGDIAL 2011 Conference, 12th annual meeting of the Special Interest Group on Discourse and Dialogue, Co-located with ACL HLT 2011, Portland, Oregon, 17 - 18 June 2011}, publisher = {Curran}, address = {Red Hook, NY}, isbn = {978-1-61839-242-8}, pages = {120 -- 129}, abstract = {We present the novel task of predicting tem-poral features of continuations of user input,while that input is still ongoing. We show that the remaining duration of an ongoing word, aswell as the duration of the next can be predicted reasonably well, and we put this information touse in a system that synchronously completesa user's speech. While we focus on collaborative completions, the techniques presented here may also be useful for the alignment of back-channels and immediate turn-taking in anincremental SDS, or to synchronously monitorthe user's speech fluency for other reasons.}, language = {en} } @inproceedings{BuschmeierBaumannDorschetal., author = {Buschmeier, Hendrik and Baumann, Timo and Dorsch, Benjamin and Kopp, Stefan and Schlangen, David}, title = {Combining Incremental Language Generation and Incremental Speech Synthesis for Adaptive Information Presentation}, series = {SIGDIAL '12: Proceedings of the 13th Annual Meeting of the Special Interest Group on Discourse and Dialogue, Seoul, South Korea, July 5 - 6, 2012}, booktitle = {SIGDIAL '12: Proceedings of the 13th Annual Meeting of the Special Interest Group on Discourse and Dialogue, Seoul, South Korea, July 5 - 6, 2012}, publisher = {Association for Computational Linguistics}, doi = {10.5555/2392800.2392852}, pages = {295 -- 303}, abstract = {Participants in a conversation are normally receptive to their surroundings and their interlocutors, even while they are speaking and can, if necessary, adapt their ongoing utterance. Typical dialogue systems are not receptive and cannot adapt while uttering. We present combin-able components for incremental natural lan-guage generation and incremental speech syn-thesis and demonstrate the flexibility they can achieve with an example system that adapts to a listener's acoustic understanding problems by pausing, repeating and possibly rephrasing problematic parts of an utterance. In an evaluation, this system was rated as significantly more natural than two systems representing the current state of the art that either ignore the interrupting event or just pause; it also has a lower response time.}, language = {en} } @inproceedings{HeintzeBaumannSchlangen, author = {Heintze, Silvan and Baumann, Timo and Schlangen, David}, title = {Comparing Local and Sequential Models for Statistical Incremental Natural Language Understanding}, series = {SIGDIAL '10: Proceedings of the 11th Annual Meeting of the Special Interest Group on Discourse and Dialogue, Tokyo, Japan,September 24 - 25, 2010}, booktitle = {SIGDIAL '10: Proceedings of the 11th Annual Meeting of the Special Interest Group on Discourse and Dialogue, Tokyo, Japan,September 24 - 25, 2010}, publisher = {Association for Computational Linguistics}, address = {Tokyo, Japan}, isbn = {978-1-932432-85-5}, doi = {10.5555/1944506.1944508}, abstract = {Incremental natural language understanding is the task of assigning semantic representations to successively larger prefixes of utterances. We compare two types of statistical models for this task: a) local models, which predict a single class for an input; and b), sequential models, which align a sequence of classes to a sequence of input tokens. We show that, with some modifications, the first type of model can be improved and made to approximate the output of the second, even though the latter is more informative. We show on two different data sets that both types of model achieve comparable performance (significantly better than a baseline), with the first type requiring simpler training data. Results for the first type of model have been reported in the literature; we show that for our kind of data our more sophisticated variant of the model performs better.}, language = {en} } @article{BaumannBussSchlangen, author = {Baumann, Timo and Buß, Okko and Schlangen, David}, title = {Evaluation and Optimisation of Incremental Processors}, series = {Dialogue \& Discourse}, volume = {2}, journal = {Dialogue \& Discourse}, number = {1}, publisher = {OJS}, doi = {10.5087/dad.2011.106}, pages = {113 -- 141}, abstract = {Incremental spoken dialogue systems, which process user input as it unfolds, pose additionalengineering challenges compared to more standard non-incremental systems: Their processingcomponents must be able to accept partial, and possibly subsequently revised input, and mustproduce output that is at the same time as accurate as possible and delivered with as little delay aspossible. In this article, we define metrics that measure how well a given processor meets thesechallenges, and we identify types of gold standards for evaluation. We exemplify these metrics inthe evaluation of several incremental processors that we have developed. We also present genericmeans to optimise some of the measures, if certain trade-offs are accepted. We believe that thiswork will help enable principled comparison of components for incremental dialogue systems andportability of results.}, language = {en} }